From ed70f3a264e9f746eaf17c96ccc4c9b7eda742dc Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 4 Jun 2014 10:11:06 +0100 Subject: ASoC: arizona: Implement TDM support for Arizona devices Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 78 ++++++++++++++++++++++++++++++++++++++++++---- 1 file changed, 72 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 29e198f57d4c..e77f61c387f7 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1185,7 +1185,10 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, int base = dai->driver->base; const int *rates; int i, ret, val; + int channels = params_channels(params); int chan_limit = arizona->pdata.max_channels_clocked[dai->id - 1]; + int tdm_width = arizona->tdm_width[dai->id - 1]; + int tdm_slots = arizona->tdm_slots[dai->id - 1]; int bclk, lrclk, wl, frame, bclk_target; if (params_rate(params) % 8000) @@ -1193,18 +1196,27 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, else rates = &arizona_48k_bclk_rates[0]; - bclk_target = snd_soc_params_to_bclk(params); - if (chan_limit && chan_limit < params_channels(params)) { + if (tdm_slots) { + arizona_aif_dbg(dai, "Configuring for %d %d bit TDM slots\n", + tdm_slots, tdm_width); + bclk_target = tdm_slots * tdm_width * params_rate(params); + channels = tdm_slots; + } else { + bclk_target = snd_soc_params_to_bclk(params); + } + + if (chan_limit && chan_limit < channels) { arizona_aif_dbg(dai, "Limiting to %d channels\n", chan_limit); - bclk_target /= params_channels(params); + bclk_target /= channels; bclk_target *= chan_limit; } - /* Force stereo for I2S mode */ + /* Force multiple of 2 channels for I2S mode */ val = snd_soc_read(codec, base + ARIZONA_AIF_FORMAT); - if (params_channels(params) == 1 && (val & ARIZONA_AIF1_FMT_MASK)) { + if ((channels & 1) && (val & ARIZONA_AIF1_FMT_MASK)) { arizona_aif_dbg(dai, "Forcing stereo mode\n"); - bclk_target *= 2; + bclk_target /= channels; + bclk_target *= channels + 1; } for (i = 0; i < ARRAY_SIZE(arizona_44k1_bclk_rates); i++) { @@ -1324,9 +1336,63 @@ static int arizona_set_tristate(struct snd_soc_dai *dai, int tristate) ARIZONA_AIF1_TRI, reg); } +static void arizona_set_channels_to_mask(struct snd_soc_dai *dai, + unsigned int base, + int channels, unsigned int mask) +{ + struct snd_soc_codec *codec = dai->codec; + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + int slot, i; + + for (i = 0; i < channels; ++i) { + slot = ffs(mask) - 1; + if (slot < 0) + return; + + regmap_write(arizona->regmap, base + i, slot); + + mask &= ~(1 << slot); + } + + if (mask) + arizona_aif_warn(dai, "Too many channels in TDM mask\n"); +} + +static int arizona_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + int base = dai->driver->base; + int rx_max_chan = dai->driver->playback.channels_max; + int tx_max_chan = dai->driver->capture.channels_max; + + /* Only support TDM for the physical AIFs */ + if (dai->id > ARIZONA_MAX_AIF) + return -ENOTSUPP; + + if (slots == 0) { + tx_mask = (1 << tx_max_chan) - 1; + rx_mask = (1 << rx_max_chan) - 1; + } + + arizona_set_channels_to_mask(dai, base + ARIZONA_AIF_FRAME_CTRL_3, + tx_max_chan, tx_mask); + arizona_set_channels_to_mask(dai, base + ARIZONA_AIF_FRAME_CTRL_11, + rx_max_chan, rx_mask); + + arizona->tdm_width[dai->id - 1] = slot_width; + arizona->tdm_slots[dai->id - 1] = slots; + + return 0; +} + const struct snd_soc_dai_ops arizona_dai_ops = { .startup = arizona_startup, .set_fmt = arizona_set_fmt, + .set_tdm_slot = arizona_set_tdm_slot, .hw_params = arizona_hw_params, .set_sysclk = arizona_dai_set_sysclk, .set_tristate = arizona_set_tristate, -- cgit v1.2.3 From 4091d3425a2e4947a2d353c33356b05418240ace Mon Sep 17 00:00:00 2001 From: Robert Jarzmik Date: Mon, 9 Jun 2014 21:59:12 +0200 Subject: ASoC: pxa2xx-ac97: prepare and unprepare the clocks Add the clock prepare and unprepare call to the driver initialization phase. This will remove a warning once the PXA architecture is migrated to the clock infrastructure. Signed-off-by: Robert Jarzmik Signed-off-by: Mark Brown --- sound/arm/pxa2xx-ac97-lib.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 66de90ed30ca..39c3969ac1c7 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -152,9 +152,9 @@ static inline void pxa_ac97_cold_pxa27x(void) gsr_bits = 0; /* PXA27x Developers Manual section 13.5.2.2.1 */ - clk_enable(ac97conf_clk); + clk_prepare_enable(ac97conf_clk); udelay(5); - clk_disable(ac97conf_clk); + clk_disable_unprepare(ac97conf_clk); GCR = GCR_COLD_RST | GCR_WARM_RST; } #endif @@ -299,14 +299,14 @@ static irqreturn_t pxa2xx_ac97_irq(int irq, void *dev_id) int pxa2xx_ac97_hw_suspend(void) { GCR |= GCR_ACLINK_OFF; - clk_disable(ac97_clk); + clk_disable_unprepare(ac97_clk); return 0; } EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_suspend); int pxa2xx_ac97_hw_resume(void) { - clk_enable(ac97_clk); + clk_prepare_enable(ac97_clk); return 0; } EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_resume); @@ -368,7 +368,7 @@ int pxa2xx_ac97_hw_probe(struct platform_device *dev) goto err_clk; } - ret = clk_enable(ac97_clk); + ret = clk_prepare_enable(ac97_clk); if (ret) goto err_clk2; @@ -403,7 +403,7 @@ void pxa2xx_ac97_hw_remove(struct platform_device *dev) clk_put(ac97conf_clk); ac97conf_clk = NULL; } - clk_disable(ac97_clk); + clk_disable_unprepare(ac97_clk); clk_put(ac97_clk); ac97_clk = NULL; } -- cgit v1.2.3 From 8b9920e3f4019313803419923b8c5c486fa82d84 Mon Sep 17 00:00:00 2001 From: Bo Shen Date: Mon, 9 Jun 2014 11:31:44 +0800 Subject: ASoC: wm8904: switch to CCF Enable WM8904 to support common clock framework. Signed-off-by: Bo Shen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8904.c | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index f7c549949c54..da46c2ad0566 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -11,6 +11,7 @@ * published by the Free Software Foundation. */ +#include #include #include #include @@ -49,6 +50,7 @@ static const char *wm8904_supply_names[WM8904_NUM_SUPPLIES] = { /* codec private data */ struct wm8904_priv { struct regmap *regmap; + struct clk *mclk; enum wm8904_type devtype; @@ -1828,6 +1830,7 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: + clk_prepare_enable(wm8904->mclk); break; case SND_SOC_BIAS_PREPARE: @@ -1894,6 +1897,7 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec, regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); + clk_disable_unprepare(wm8904->mclk); break; } codec->dapm.bias_level = level; @@ -2110,6 +2114,13 @@ static int wm8904_i2c_probe(struct i2c_client *i2c, if (wm8904 == NULL) return -ENOMEM; + wm8904->mclk = devm_clk_get(&i2c->dev, "mclk"); + if (IS_ERR(wm8904->mclk)) { + ret = PTR_ERR(wm8904->mclk); + dev_err(&i2c->dev, "Failed to get MCLK\n"); + return ret; + } + wm8904->regmap = devm_regmap_init_i2c(i2c, &wm8904_regmap); if (IS_ERR(wm8904->regmap)) { ret = PTR_ERR(wm8904->regmap); -- cgit v1.2.3 From 53e3030b4ba10ef50bbae2c7bd344fcb10539299 Mon Sep 17 00:00:00 2001 From: Bo Shen Date: Mon, 9 Jun 2014 11:31:43 +0800 Subject: ASoC: atmel_wm8904: switch to CCF Signed-off-by: Bo Shen Signed-off-by: Mark Brown --- sound/soc/atmel/atmel_wm8904.c | 50 ------------------------------------------ 1 file changed, 50 deletions(-) (limited to 'sound') diff --git a/sound/soc/atmel/atmel_wm8904.c b/sound/soc/atmel/atmel_wm8904.c index b4e36901a40b..4052268ce462 100644 --- a/sound/soc/atmel/atmel_wm8904.c +++ b/sound/soc/atmel/atmel_wm8904.c @@ -18,10 +18,6 @@ #include "../codecs/wm8904.h" #include "atmel_ssc_dai.h" -#define MCLK_RATE 32768 - -static struct clk *mclk; - static const struct snd_soc_dapm_widget atmel_asoc_wm8904_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Mic", NULL), @@ -61,26 +57,6 @@ static struct snd_soc_ops atmel_asoc_wm8904_ops = { .hw_params = atmel_asoc_wm8904_hw_params, }; -static int atmel_set_bias_level(struct snd_soc_card *card, - struct snd_soc_dapm_context *dapm, - enum snd_soc_bias_level level) -{ - if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { - switch (level) { - case SND_SOC_BIAS_PREPARE: - clk_prepare_enable(mclk); - break; - case SND_SOC_BIAS_OFF: - clk_disable_unprepare(mclk); - break; - default: - break; - } - } - - return 0; -}; - static struct snd_soc_dai_link atmel_asoc_wm8904_dailink = { .name = "WM8904", .stream_name = "WM8904 PCM", @@ -94,7 +70,6 @@ static struct snd_soc_dai_link atmel_asoc_wm8904_dailink = { static struct snd_soc_card atmel_asoc_wm8904_card = { .name = "atmel_asoc_wm8904", .owner = THIS_MODULE, - .set_bias_level = atmel_set_bias_level, .dai_link = &atmel_asoc_wm8904_dailink, .num_links = 1, .dapm_widgets = atmel_asoc_wm8904_dapm_widgets, @@ -153,7 +128,6 @@ static int atmel_asoc_wm8904_probe(struct platform_device *pdev) { struct snd_soc_card *card = &atmel_asoc_wm8904_card; struct snd_soc_dai_link *dailink = &atmel_asoc_wm8904_dailink; - struct clk *clk_src; int id, ret; card->dev = &pdev->dev; @@ -170,30 +144,6 @@ static int atmel_asoc_wm8904_probe(struct platform_device *pdev) return ret; } - mclk = clk_get(NULL, "pck0"); - if (IS_ERR(mclk)) { - dev_err(&pdev->dev, "failed to get pck0\n"); - ret = PTR_ERR(mclk); - goto err_set_audio; - } - - clk_src = clk_get(NULL, "clk32k"); - if (IS_ERR(clk_src)) { - dev_err(&pdev->dev, "failed to get clk32k\n"); - ret = PTR_ERR(clk_src); - goto err_set_audio; - } - - ret = clk_set_parent(mclk, clk_src); - clk_put(clk_src); - if (ret != 0) { - dev_err(&pdev->dev, "failed to set MCLK parent\n"); - goto err_set_audio; - } - - dev_info(&pdev->dev, "setting pck0 to %dHz\n", MCLK_RATE); - clk_set_rate(mclk, MCLK_RATE); - ret = snd_soc_register_card(card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed\n"); -- cgit v1.2.3 From b5d4f4a53f275afa1d066ace6a9fa6995337ead9 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 3 Jun 2014 14:56:35 +0800 Subject: ASoC: rl6231: Remove unneeded inclusion of header files Signed-off-by: Axel Lin Tested-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rl6231.c | 19 ------------------- 1 file changed, 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rl6231.c b/sound/soc/codecs/rl6231.c index 7b82fbe0d14c..56650d6c2f53 100644 --- a/sound/soc/codecs/rl6231.c +++ b/sound/soc/codecs/rl6231.c @@ -11,25 +11,6 @@ */ #include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include #include "rl6231.h" -- cgit v1.2.3 From cc9e92431ee9c7fe974266e0e6533a1a68e45539 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 6 Jun 2014 14:14:05 +0100 Subject: ASoC: wm5102: Add controls to allow shaping of ultrasonic response Add controls to allow custom shaping of the ultrasonic response. This custom shaping can be turned on/off at runtime, although, it should be noted that settings will not affect a currently open audio stream, they will be applied when the next audio stream is started. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- include/linux/mfd/arizona/core.h | 3 ++ sound/soc/codecs/arizona.c | 34 ++++++++++++++++++++++ sound/soc/codecs/wm5102.c | 62 ++++++++++++++++++++++++++++++++++++++++ 3 files changed, 99 insertions(+) (limited to 'sound') diff --git a/include/linux/mfd/arizona/core.h b/include/linux/mfd/arizona/core.h index 11783b511b9a..55926517d50b 100644 --- a/include/linux/mfd/arizona/core.h +++ b/include/linux/mfd/arizona/core.h @@ -113,6 +113,9 @@ struct arizona { int tdm_width[ARIZONA_MAX_AIF]; int tdm_slots[ARIZONA_MAX_AIF]; + + uint16_t dac_comp_coeff; + uint8_t dac_comp_enabled; }; int arizona_clk32k_enable(struct arizona *arizona); diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index e77f61c387f7..41b56ee6ff51 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1127,6 +1127,31 @@ static int arizona_startup(struct snd_pcm_substream *substream, constraint); } +static void arizona_wm5102_set_dac_comp(struct snd_soc_codec *codec, + unsigned int rate) +{ + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + struct reg_default dac_comp[] = { + { 0x80, 0x3 }, + { ARIZONA_DAC_COMP_1, 0 }, + { ARIZONA_DAC_COMP_2, 0 }, + { 0x80, 0x0 }, + }; + + mutex_lock(&codec->mutex); + + dac_comp[1].def = arizona->dac_comp_coeff; + if (rate >= 176400) + dac_comp[2].def = arizona->dac_comp_enabled; + + mutex_unlock(&codec->mutex); + + regmap_multi_reg_write(arizona->regmap, + dac_comp, + ARRAY_SIZE(dac_comp)); +} + static int arizona_hw_params_rate(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -1153,6 +1178,15 @@ static int arizona_hw_params_rate(struct snd_pcm_substream *substream, switch (dai_priv->clk) { case ARIZONA_CLK_SYSCLK: + switch (priv->arizona->type) { + case WM5102: + arizona_wm5102_set_dac_comp(codec, + params_rate(params)); + break; + default: + break; + } + snd_soc_update_bits(codec, ARIZONA_SAMPLE_RATE_1, ARIZONA_SAMPLE_RATE_1_MASK, sr_val); if (base) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index dcf1d12cfef8..7bf2397fc25a 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -612,6 +612,62 @@ static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w, return 0; } +static int wm5102_out_comp_coeff_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); + uint16_t data; + + mutex_lock(&codec->mutex); + data = cpu_to_be16(arizona->dac_comp_coeff); + memcpy(ucontrol->value.bytes.data, &data, sizeof(data)); + mutex_unlock(&codec->mutex); + + return 0; +} + +static int wm5102_out_comp_coeff_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); + + mutex_lock(&codec->mutex); + memcpy(&arizona->dac_comp_coeff, ucontrol->value.bytes.data, + sizeof(arizona->dac_comp_coeff)); + arizona->dac_comp_coeff = be16_to_cpu(arizona->dac_comp_coeff); + mutex_unlock(&codec->mutex); + + return 0; +} + +static int wm5102_out_comp_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); + + mutex_lock(&codec->mutex); + ucontrol->value.integer.value[0] = arizona->dac_comp_enabled; + mutex_unlock(&codec->mutex); + + return 0; +} + +static int wm5102_out_comp_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); + + mutex_lock(&codec->mutex); + arizona->dac_comp_enabled = ucontrol->value.integer.value[0]; + mutex_unlock(&codec->mutex); + + return 0; +} + static const char *wm5102_osr_text[] = { "Low power", "Normal", "High performance", }; @@ -843,6 +899,12 @@ SOC_SINGLE_TLV("Noise Gate Threshold Volume", ARIZONA_NOISE_GATE_CONTROL, ARIZONA_NGATE_THR_SHIFT, 7, 1, ng_tlv), SOC_ENUM("Noise Gate Hold", arizona_ng_hold), +SND_SOC_BYTES_EXT("Output Compensation Coefficient", 2, + wm5102_out_comp_coeff_get, wm5102_out_comp_coeff_put), + +SOC_SINGLE_EXT("Output Compensation Switch", 0, 0, 1, 0, + wm5102_out_comp_switch_get, wm5102_out_comp_switch_put), + WM5102_NG_SRC("HPOUT1L", ARIZONA_NOISE_GATE_SELECT_1L), WM5102_NG_SRC("HPOUT1R", ARIZONA_NOISE_GATE_SELECT_1R), WM5102_NG_SRC("HPOUT2L", ARIZONA_NOISE_GATE_SELECT_2L), -- cgit v1.2.3 From 2488708f5c13e7bcb884dbb31b767e314ea80016 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 6 Jun 2014 14:09:17 +0200 Subject: ASoC: sigmadsp: Split regmap and I2C support into separate modules When the SigmaDSP module is built-in, but the I2C core is build as a module we'll get a undefined reference: sound/built-in.o: In function `sigma_action_write_i2c': :(.text+0x5d8d4): undefined reference to `i2c_master_send' This can happen if a audio driver that is using the regmap SigmaDSP interface is built into the kernel, but core I2C support is build as a module. To fix this split the SigmaDSP module into three modules, one module providing the core infrastructure and two small modules implementing the regmap and I2C interfaces. This allows e.g. the core infrastructure and regmap support to be built into the kernel while I2C support can still be build as a module. Fixes: dab464b60 ("ASoC: Add ADAU1361/ADAU1761 audio CODEC support") Reported-by: Arnd Bergmann Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 12 +++++-- sound/soc/codecs/Makefile | 4 +++ sound/soc/codecs/sigmadsp-i2c.c | 35 ++++++++++++++++++++ sound/soc/codecs/sigmadsp-regmap.c | 36 +++++++++++++++++++++ sound/soc/codecs/sigmadsp.c | 65 ++------------------------------------ sound/soc/codecs/sigmadsp.h | 20 ++++++++++++ 6 files changed, 107 insertions(+), 65 deletions(-) create mode 100644 sound/soc/codecs/sigmadsp-i2c.c create mode 100644 sound/soc/codecs/sigmadsp-regmap.c (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index cbfa1e18f651..0b9571c858f8 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -225,11 +225,11 @@ config SND_SOC_ADAU1373 config SND_SOC_ADAU1701 tristate "Analog Devices ADAU1701 CODEC" depends on I2C - select SND_SOC_SIGMADSP + select SND_SOC_SIGMADSP_I2C config SND_SOC_ADAU17X1 tristate - select SND_SOC_SIGMADSP + select SND_SOC_SIGMADSP_REGMAP config SND_SOC_ADAU1761 tristate @@ -476,6 +476,14 @@ config SND_SOC_SIGMADSP tristate select CRC32 +config SND_SOC_SIGMADSP_I2C + tristate + select SND_SOC_SIGMADSP + +config SND_SOC_SIGMADSP_REGMAP + tristate + select SND_SOC_SIGMADSP + config SND_SOC_SIRF_AUDIO_CODEC tristate "SiRF SoC internal audio codec" select REGMAP_MMIO diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index be3377b8d73f..1bd6e1cf6f82 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -77,6 +77,8 @@ snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o snd-soc-alc5632-objs := alc5632.o snd-soc-sigmadsp-objs := sigmadsp.o +snd-soc-sigmadsp-i2c-objs := sigmadsp-i2c.o +snd-soc-sigmadsp-regmap-objs := sigmadsp-regmap.o snd-soc-si476x-objs := si476x.o snd-soc-sirf-audio-codec-objs := sirf-audio-codec.o snd-soc-sn95031-objs := sn95031.o @@ -240,6 +242,8 @@ obj-$(CONFIG_SND_SOC_RT5651) += snd-soc-rt5651.o obj-$(CONFIG_SND_SOC_RT5677) += snd-soc-rt5677.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o +obj-$(CONFIG_SND_SOC_SIGMADSP_I2C) += snd-soc-sigmadsp-i2c.o +obj-$(CONFIG_SND_SOC_SIGMADSP_REGMAP) += snd-soc-sigmadsp-regmap.o obj-$(CONFIG_SND_SOC_SI476X) += snd-soc-si476x.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif-rx.o snd-soc-spdif-tx.o diff --git a/sound/soc/codecs/sigmadsp-i2c.c b/sound/soc/codecs/sigmadsp-i2c.c new file mode 100644 index 000000000000..246081aae8ca --- /dev/null +++ b/sound/soc/codecs/sigmadsp-i2c.c @@ -0,0 +1,35 @@ +/* + * Load Analog Devices SigmaStudio firmware files + * + * Copyright 2009-2011 Analog Devices Inc. + * + * Licensed under the GPL-2 or later. + */ + +#include +#include +#include + +#include "sigmadsp.h" + +static int sigma_action_write_i2c(void *control_data, + const struct sigma_action *sa, size_t len) +{ + return i2c_master_send(control_data, (const unsigned char *)&sa->addr, + len); +} + +int process_sigma_firmware(struct i2c_client *client, const char *name) +{ + struct sigma_firmware ssfw; + + ssfw.control_data = client; + ssfw.write = sigma_action_write_i2c; + + return _process_sigma_firmware(&client->dev, &ssfw, name); +} +EXPORT_SYMBOL(process_sigma_firmware); + +MODULE_AUTHOR("Lars-Peter Clausen "); +MODULE_DESCRIPTION("SigmaDSP I2C firmware loader"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/sigmadsp-regmap.c b/sound/soc/codecs/sigmadsp-regmap.c new file mode 100644 index 000000000000..f78ed8d2cfb2 --- /dev/null +++ b/sound/soc/codecs/sigmadsp-regmap.c @@ -0,0 +1,36 @@ +/* + * Load Analog Devices SigmaStudio firmware files + * + * Copyright 2009-2011 Analog Devices Inc. + * + * Licensed under the GPL-2 or later. + */ + +#include +#include +#include + +#include "sigmadsp.h" + +static int sigma_action_write_regmap(void *control_data, + const struct sigma_action *sa, size_t len) +{ + return regmap_raw_write(control_data, be16_to_cpu(sa->addr), + sa->payload, len - 2); +} + +int process_sigma_firmware_regmap(struct device *dev, struct regmap *regmap, + const char *name) +{ + struct sigma_firmware ssfw; + + ssfw.control_data = regmap; + ssfw.write = sigma_action_write_regmap; + + return _process_sigma_firmware(dev, &ssfw, name); +} +EXPORT_SYMBOL(process_sigma_firmware_regmap); + +MODULE_AUTHOR("Lars-Peter Clausen "); +MODULE_DESCRIPTION("SigmaDSP regmap firmware loader"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c index 4068f2491232..f2de7e049bc6 100644 --- a/sound/soc/codecs/sigmadsp.c +++ b/sound/soc/codecs/sigmadsp.c @@ -34,23 +34,6 @@ enum { SIGMA_ACTION_END, }; -struct sigma_action { - u8 instr; - u8 len_hi; - __le16 len; - __be16 addr; - unsigned char payload[]; -} __packed; - -struct sigma_firmware { - const struct firmware *fw; - size_t pos; - - void *control_data; - int (*write)(void *control_data, const struct sigma_action *sa, - size_t len); -}; - static inline u32 sigma_action_len(struct sigma_action *sa) { return (sa->len_hi << 16) | le16_to_cpu(sa->len); @@ -138,7 +121,7 @@ process_sigma_actions(struct sigma_firmware *ssfw) return 0; } -static int _process_sigma_firmware(struct device *dev, +int _process_sigma_firmware(struct device *dev, struct sigma_firmware *ssfw, const char *name) { int ret; @@ -197,50 +180,6 @@ static int _process_sigma_firmware(struct device *dev, return ret; } - -#if IS_ENABLED(CONFIG_I2C) - -static int sigma_action_write_i2c(void *control_data, - const struct sigma_action *sa, size_t len) -{ - return i2c_master_send(control_data, (const unsigned char *)&sa->addr, - len); -} - -int process_sigma_firmware(struct i2c_client *client, const char *name) -{ - struct sigma_firmware ssfw; - - ssfw.control_data = client; - ssfw.write = sigma_action_write_i2c; - - return _process_sigma_firmware(&client->dev, &ssfw, name); -} -EXPORT_SYMBOL(process_sigma_firmware); - -#endif - -#if IS_ENABLED(CONFIG_REGMAP) - -static int sigma_action_write_regmap(void *control_data, - const struct sigma_action *sa, size_t len) -{ - return regmap_raw_write(control_data, be16_to_cpu(sa->addr), - sa->payload, len - 2); -} - -int process_sigma_firmware_regmap(struct device *dev, struct regmap *regmap, - const char *name) -{ - struct sigma_firmware ssfw; - - ssfw.control_data = regmap; - ssfw.write = sigma_action_write_regmap; - - return _process_sigma_firmware(dev, &ssfw, name); -} -EXPORT_SYMBOL(process_sigma_firmware_regmap); - -#endif +EXPORT_SYMBOL_GPL(_process_sigma_firmware); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/sigmadsp.h b/sound/soc/codecs/sigmadsp.h index e439cbd7af7d..c47cd23e9827 100644 --- a/sound/soc/codecs/sigmadsp.h +++ b/sound/soc/codecs/sigmadsp.h @@ -12,6 +12,26 @@ #include #include +struct sigma_action { + u8 instr; + u8 len_hi; + __le16 len; + __be16 addr; + unsigned char payload[]; +} __packed; + +struct sigma_firmware { + const struct firmware *fw; + size_t pos; + + void *control_data; + int (*write)(void *control_data, const struct sigma_action *sa, + size_t len); +}; + +int _process_sigma_firmware(struct device *dev, + struct sigma_firmware *ssfw, const char *name); + struct i2c_client; extern int process_sigma_firmware(struct i2c_client *client, const char *name); -- cgit v1.2.3 From 5c3fc7a79aae474670096d73da21aec9f0750abb Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 11 Jun 2014 12:50:58 +0100 Subject: ASoC: wm5102: Convert snd_kcontrol_chip to snd_soc_kcontrol_codec Controls for shaping the ultrasonic frequency response were introduced in this commit: commit 720630c002ffc7b0fa2ed5b3f4bfb36fd8f87ca6 ASoC: wm5102: Add controls to allow shaping of ultrasonic response However, they mistakenly used snd_kcontrol_chip instead of snd_soc_kcontrol_codec, which has replaced it now the framework is moving to componentisation. This patch fixes this. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm5102.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 7bf2397fc25a..58913580cc5a 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -615,7 +615,7 @@ static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w, static int wm5102_out_comp_coeff_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct arizona *arizona = dev_get_drvdata(codec->dev->parent); uint16_t data; @@ -630,7 +630,7 @@ static int wm5102_out_comp_coeff_get(struct snd_kcontrol *kcontrol, static int wm5102_out_comp_coeff_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct arizona *arizona = dev_get_drvdata(codec->dev->parent); mutex_lock(&codec->mutex); @@ -645,7 +645,7 @@ static int wm5102_out_comp_coeff_put(struct snd_kcontrol *kcontrol, static int wm5102_out_comp_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct arizona *arizona = dev_get_drvdata(codec->dev->parent); mutex_lock(&codec->mutex); @@ -658,7 +658,7 @@ static int wm5102_out_comp_switch_get(struct snd_kcontrol *kcontrol, static int wm5102_out_comp_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct arizona *arizona = dev_get_drvdata(codec->dev->parent); mutex_lock(&codec->mutex); -- cgit v1.2.3 From c8cfbec882b408cd933d24882302cacd39985e18 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 3 Jun 2014 10:56:41 +0800 Subject: ASoC: rt5677: Convert to use module_i2c_driver Signed-off-by: Axel Lin Tested-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 13 +------------ 1 file changed, 1 insertion(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 833231e27340..60fff7fdb85e 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -3480,18 +3480,7 @@ static struct i2c_driver rt5677_i2c_driver = { .remove = rt5677_i2c_remove, .id_table = rt5677_i2c_id, }; - -static int __init rt5677_modinit(void) -{ - return i2c_add_driver(&rt5677_i2c_driver); -} -module_init(rt5677_modinit); - -static void __exit rt5677_modexit(void) -{ - i2c_del_driver(&rt5677_i2c_driver); -} -module_exit(rt5677_modexit); +module_i2c_driver(rt5677_i2c_driver); MODULE_DESCRIPTION("ASoC RT5677 driver"); MODULE_AUTHOR("Oder Chiou "); -- cgit v1.2.3 From 30f14b439f0cc0c7776722c0b1c2836769fe0ce7 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 10 Jun 2014 08:57:36 +0800 Subject: ASoC: rt5677: Convert to use rl6231_get_clk_info Signed-off-by: Axel Lin Tested-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 2 ++ sound/soc/codecs/rt5677.c | 18 ++---------------- 2 files changed, 4 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index cbfa1e18f651..9d88845dc7c4 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -445,9 +445,11 @@ config SND_SOC_RL6231 default y if SND_SOC_RT5640=y default y if SND_SOC_RT5645=y default y if SND_SOC_RT5651=y + default y if SND_SOC_RT5677=y default m if SND_SOC_RT5640=m default m if SND_SOC_RT5645=m default m if SND_SOC_RT5651=m + default m if SND_SOC_RT5677=m config SND_SOC_RT5631 tristate diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 60fff7fdb85e..1ffc6891fcbc 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -27,6 +27,7 @@ #include #include +#include "rl6231.h" #include "rt5677.h" #define RT5677_DEVICE_ID 0x6327 @@ -2798,21 +2799,6 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "PDM2R", NULL, "PDM2 R Mux" }, }; -static int get_clk_info(int sclk, int rate) -{ - int i, pd[] = {1, 2, 3, 4, 6, 8, 12, 16}; - - if (sclk <= 0 || rate <= 0) - return -EINVAL; - - rate = rate << 8; - for (i = 0; i < ARRAY_SIZE(pd); i++) - if (sclk == rate * pd[i]) - return i; - - return -EINVAL; -} - static int rt5677_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { @@ -2822,7 +2808,7 @@ static int rt5677_hw_params(struct snd_pcm_substream *substream, int pre_div, bclk_ms, frame_size; rt5677->lrck[dai->id] = params_rate(params); - pre_div = get_clk_info(rt5677->sysclk, rt5677->lrck[dai->id]); + pre_div = rl6231_get_clk_info(rt5677->sysclk, rt5677->lrck[dai->id]); if (pre_div < 0) { dev_err(codec->dev, "Unsupported clock setting\n"); return -EINVAL; -- cgit v1.2.3 From 9a53581efabec8d9c69933c360e2ab036a1da9bc Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 3 Jun 2014 10:58:58 +0800 Subject: ASoC: rt5677: Convert to use rl6231_calc_dmic_clk Signed-off-by: Axel Lin Tested-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 16 +--------------- 1 file changed, 1 insertion(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 1ffc6891fcbc..45f99a824c15 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -637,21 +637,7 @@ static int set_dmic_clk(struct snd_soc_dapm_widget *w, { struct snd_soc_codec *codec = w->codec; struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); - int div[] = {2, 3, 4, 6, 8, 12}, idx = -EINVAL, i; - int rate, red, bound, temp; - - rate = rt5677->sysclk; - red = 3000000 * 12; - for (i = 0; i < ARRAY_SIZE(div); i++) { - bound = div[i] * 3000000; - if (rate > bound) - continue; - temp = bound - rate; - if (temp < red) { - red = temp; - idx = i; - } - } + int idx = rl6231_calc_dmic_clk(rt5677->sysclk); if (idx < 0) dev_err(codec->dev, "Failed to set DMIC clock\n"); -- cgit v1.2.3 From 294e8a75a19623c3d8a3b7f692d3b869932c7f8f Mon Sep 17 00:00:00 2001 From: Anssi Hannula Date: Mon, 16 Jun 2014 02:56:41 +0300 Subject: ASoC: spdif_transmitter: Allow 192kHz sample rate Transmitters and receivers may support a 192kHz sample rate. Tested with a Cubox-i imx6 system and an Onkyo TX-SR607 receiver. Signed-off-by: Anssi Hannula Signed-off-by: Mark Brown --- sound/soc/codecs/spdif_transmitter.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/spdif_transmitter.c b/sound/soc/codecs/spdif_transmitter.c index a078aa31052a..e0df537dd4b7 100644 --- a/sound/soc/codecs/spdif_transmitter.c +++ b/sound/soc/codecs/spdif_transmitter.c @@ -24,7 +24,7 @@ #define DRV_NAME "spdif-dit" -#define STUB_RATES SNDRV_PCM_RATE_8000_96000 +#define STUB_RATES SNDRV_PCM_RATE_8000_192000 #define STUB_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE) -- cgit v1.2.3 From f3a30baa2876f910cbfb15b9a7442bacd8437e9d Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Tue, 6 May 2014 16:42:25 +0800 Subject: ASoC: fsl_spdif: Improve coding style 1) Apply better indentations 2) Drop braces for single statement. 3) Use simpler ternary to reduce code. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_spdif.c | 72 +++++++++++++++++++++++------------------------ sound/soc/fsl/fsl_spdif.h | 2 ++ 2 files changed, 37 insertions(+), 37 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index b912d45a2a4c..fe4d9e3da793 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -32,10 +32,13 @@ #define FSL_SPDIF_TXFIFO_WML 0x8 #define FSL_SPDIF_RXFIFO_WML 0x8 -#define INTR_FOR_PLAYBACK (INT_TXFIFO_RESYNC) -#define INTR_FOR_CAPTURE (INT_SYM_ERR | INT_BIT_ERR | INT_URX_FUL | INT_URX_OV|\ - INT_QRX_FUL | INT_QRX_OV | INT_UQ_SYNC | INT_UQ_ERR |\ - INT_RXFIFO_RESYNC | INT_LOSS_LOCK | INT_DPLL_LOCKED) +#define INTR_FOR_PLAYBACK (INT_TXFIFO_RESYNC) +#define INTR_FOR_CAPTURE (INT_SYM_ERR | INT_BIT_ERR | INT_URX_FUL |\ + INT_URX_OV | INT_QRX_FUL | INT_QRX_OV |\ + INT_UQ_SYNC | INT_UQ_ERR | INT_RXFIFO_RESYNC |\ + INT_LOSS_LOCK | INT_DPLL_LOCKED) + +#define SIE_INTR_FOR(tx) (tx ? INTR_FOR_PLAYBACK : INTR_FOR_CAPTURE) /* Index list for the values that has if (DPLL Locked) condition */ static u8 srpc_dpll_locked[] = { 0x0, 0x1, 0x2, 0x3, 0x4, 0xa, 0xb }; @@ -137,10 +140,9 @@ static void spdif_irq_sym_error(struct fsl_spdif_priv *spdif_priv) dev_dbg(&pdev->dev, "isr: receiver found illegal symbol\n"); - if (!spdif_priv->dpll_locked) { - /* DPLL unlocked seems no audio stream */ + /* Clear illegal symbol if DPLL unlocked since no audio stream */ + if (!spdif_priv->dpll_locked) regmap_update_bits(regmap, REG_SPDIF_SIE, INT_SYM_ERR, 0); - } } /* U/Q Channel receive register full */ @@ -335,8 +337,8 @@ static void spdif_write_channel_status(struct fsl_spdif_priv *spdif_priv) u32 ch_status; ch_status = (bitrev8(ctrl->ch_status[0]) << 16) | - (bitrev8(ctrl->ch_status[1]) << 8) | - bitrev8(ctrl->ch_status[2]); + (bitrev8(ctrl->ch_status[1]) << 8) | + bitrev8(ctrl->ch_status[2]); regmap_write(regmap, REG_SPDIF_STCSCH, ch_status); dev_dbg(&pdev->dev, "STCSCH: 0x%06x\n", ch_status); @@ -433,13 +435,12 @@ clk_set_bypass: spdif_set_cstatus(ctrl, IEC958_AES3_CON_FS, csfs); /* select clock source and divisor */ - stc = STC_TXCLK_ALL_EN | STC_TXCLK_SRC_SET(clk) | STC_TXCLK_DF(txclk_df); - mask = STC_TXCLK_ALL_EN_MASK | STC_TXCLK_SRC_MASK | STC_TXCLK_DF_MASK; + stc = STC_TXCLK_ALL_EN | STC_TXCLK_SRC_SET(clk) | + STC_TXCLK_DF(txclk_df) | STC_SYSCLK_DF(sysclk_df); + mask = STC_TXCLK_ALL_EN_MASK | STC_TXCLK_SRC_MASK | + STC_TXCLK_DF_MASK | STC_SYSCLK_DF_MASK; regmap_update_bits(regmap, REG_SPDIF_STC, mask, stc); - regmap_update_bits(regmap, REG_SPDIF_STC, - STC_SYSCLK_DF_MASK, STC_SYSCLK_DF(sysclk_df)); - dev_dbg(&pdev->dev, "set sample rate to %dHz for %dHz playback\n", spdif_priv->txrate[rate], sample_rate); @@ -553,7 +554,7 @@ static int fsl_spdif_hw_params(struct snd_pcm_substream *substream, return ret; } spdif_set_cstatus(ctrl, IEC958_AES3_CON_CLOCK, - IEC958_AES3_CON_CLOCK_1000PPM); + IEC958_AES3_CON_CLOCK_1000PPM); spdif_write_channel_status(spdif_priv); } else { /* Setup rx clock source */ @@ -569,9 +570,9 @@ static int fsl_spdif_trigger(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai); struct regmap *regmap = spdif_priv->regmap; - int is_playack = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); - u32 intr = is_playack ? INTR_FOR_PLAYBACK : INTR_FOR_CAPTURE; - u32 dmaen = is_playack ? SCR_DMA_TX_EN : SCR_DMA_RX_EN;; + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + u32 intr = SIE_INTR_FOR(tx); + u32 dmaen = SCR_DMA_xX_EN(tx); switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -662,9 +663,8 @@ static int fsl_spdif_capture_get(struct snd_kcontrol *kcontrol, u32 cstatus, val; regmap_read(regmap, REG_SPDIF_SIS, &val); - if (!(val & INT_CNEW)) { + if (!(val & INT_CNEW)) return -EAGAIN; - } regmap_read(regmap, REG_SPDIF_SRCSH, &cstatus); ucontrol->value.iec958.status[0] = (cstatus >> 16) & 0xFF; @@ -693,15 +693,14 @@ static int fsl_spdif_subcode_get(struct snd_kcontrol *kcontrol, struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai); struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control; unsigned long flags; - int ret = 0; + int ret = -EAGAIN; spin_lock_irqsave(&ctrl->ctl_lock, flags); if (ctrl->ready_buf) { int idx = (ctrl->ready_buf - 1) * SPDIF_UBITS_SIZE; memcpy(&ucontrol->value.iec958.subcode[0], &ctrl->subcode[idx], SPDIF_UBITS_SIZE); - } else { - ret = -EAGAIN; + ret = 0; } spin_unlock_irqrestore(&ctrl->ctl_lock, flags); @@ -726,15 +725,14 @@ static int fsl_spdif_qget(struct snd_kcontrol *kcontrol, struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai); struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control; unsigned long flags; - int ret = 0; + int ret = -EAGAIN; spin_lock_irqsave(&ctrl->ctl_lock, flags); if (ctrl->ready_buf) { int idx = (ctrl->ready_buf - 1) * SPDIF_QSUB_SIZE; memcpy(&ucontrol->value.bytes.data[0], &ctrl->qsub[idx], SPDIF_QSUB_SIZE); - } else { - ret = -EAGAIN; + ret = 0; } spin_unlock_irqrestore(&ctrl->ctl_lock, flags); @@ -799,10 +797,10 @@ static int spdif_get_rxclk_rate(struct fsl_spdif_priv *spdif_priv, regmap_read(regmap, REG_SPDIF_SRPC, &phaseconf); clksrc = (phaseconf >> SRPC_CLKSRC_SEL_OFFSET) & 0xf; - if (srpc_dpll_locked[clksrc] && (phaseconf & SRPC_DPLL_LOCKED)) { - /* Get bus clock from system */ + + /* Get bus clock from system */ + if (srpc_dpll_locked[clksrc] && (phaseconf & SRPC_DPLL_LOCKED)) busclk_freq = clk_get_rate(spdif_priv->sysclk); - } /* FreqMeas_CLK = (BUS_CLK * FreqMeas) / 2 ^ 10 / GAINSEL / 128 */ tmpval64 = (u64) busclk_freq * freqmeas; @@ -826,12 +824,12 @@ static int fsl_spdif_rxrate_get(struct snd_kcontrol *kcontrol, { struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai); - int rate = spdif_get_rxclk_rate(spdif_priv, SPDIF_DEFAULT_GAINSEL); + int rate = 0; if (spdif_priv->dpll_locked) - ucontrol->value.integer.value[0] = rate; - else - ucontrol->value.integer.value[0] = 0; + rate = spdif_get_rxclk_rate(spdif_priv, SPDIF_DEFAULT_GAINSEL); + + ucontrol->value.integer.value[0] = rate; return 0; } @@ -1238,12 +1236,12 @@ static int fsl_spdif_probe(struct platform_device *pdev) spin_lock_init(&ctrl->ctl_lock); /* Init tx channel status default value */ - ctrl->ch_status[0] = - IEC958_AES0_CON_NOT_COPYRIGHT | IEC958_AES0_CON_EMPHASIS_5015; + ctrl->ch_status[0] = IEC958_AES0_CON_NOT_COPYRIGHT | + IEC958_AES0_CON_EMPHASIS_5015; ctrl->ch_status[1] = IEC958_AES1_CON_DIGDIGCONV_ID; ctrl->ch_status[2] = 0x00; - ctrl->ch_status[3] = - IEC958_AES3_CON_FS_44100 | IEC958_AES3_CON_CLOCK_1000PPM; + ctrl->ch_status[3] = IEC958_AES3_CON_FS_44100 | + IEC958_AES3_CON_CLOCK_1000PPM; spdif_priv->dpll_locked = false; diff --git a/sound/soc/fsl/fsl_spdif.h b/sound/soc/fsl/fsl_spdif.h index 16fde4b927d3..b13e30ea1c9e 100644 --- a/sound/soc/fsl/fsl_spdif.h +++ b/sound/soc/fsl/fsl_spdif.h @@ -93,6 +93,8 @@ #define SCR_USRC_SEL_RECV (0x1 << SCR_USRC_SEL_OFFSET) #define SCR_USRC_SEL_CHIP (0x3 << SCR_USRC_SEL_OFFSET) +#define SCR_DMA_xX_EN(tx) (tx ? SCR_DMA_TX_EN : SCR_DMA_RX_EN) + /* SPDIF CDText control */ #define SRCD_CD_USER_OFFSET 1 #define SRCD_CD_USER (1 << SRCD_CD_USER_OFFSET) -- cgit v1.2.3 From c7dfeed10928f00f8c99ad3b58a4f4c5cfd7c581 Mon Sep 17 00:00:00 2001 From: Anssi Hannula Date: Mon, 16 Jun 2014 02:56:42 +0300 Subject: ASoC: fsl_spdif: Add support for output sample rates 96kHz and 192kHz Add support for the output sample rates 96kHz and 192kHz. Tested with a Cubox-i imx6 system and an Onkyo TX-SR607 receiver. Signed-off-by: Anssi Hannula Acked-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_spdif.c | 14 +++++++++++--- sound/soc/fsl/fsl_spdif.h | 8 ++++++-- 2 files changed, 17 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index fe4d9e3da793..6e5e369d9693 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -99,7 +99,7 @@ struct fsl_spdif_priv { struct platform_device *pdev; struct regmap *regmap; bool dpll_locked; - u16 txrate[SPDIF_TXRATE_MAX]; + u32 txrate[SPDIF_TXRATE_MAX]; u8 txclk_df[SPDIF_TXRATE_MAX]; u8 sysclk_df[SPDIF_TXRATE_MAX]; u8 txclk_src[SPDIF_TXRATE_MAX]; @@ -392,6 +392,14 @@ static int spdif_set_sample_rate(struct snd_pcm_substream *substream, rate = SPDIF_TXRATE_48000; csfs = IEC958_AES3_CON_FS_48000; break; + case 96000: + rate = SPDIF_TXRATE_96000; + csfs = IEC958_AES3_CON_FS_96000; + break; + case 192000: + rate = SPDIF_TXRATE_192000; + csfs = IEC958_AES3_CON_FS_192000; + break; default: dev_err(&pdev->dev, "unsupported sample rate %d\n", sample_rate); return -EINVAL; @@ -1044,7 +1052,7 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv, struct clk *clk, u64 savesub, enum spdif_txrate index, bool round) { - const u32 rate[] = { 32000, 44100, 48000 }; + const u32 rate[] = { 32000, 44100, 48000, 96000, 192000 }; bool is_sysclk = clk == spdif_priv->sysclk; u64 rate_ideal, rate_actual, sub; u32 sysclk_dfmin, sysclk_dfmax; @@ -1103,7 +1111,7 @@ out: static int fsl_spdif_probe_txclk(struct fsl_spdif_priv *spdif_priv, enum spdif_txrate index) { - const u32 rate[] = { 32000, 44100, 48000 }; + const u32 rate[] = { 32000, 44100, 48000, 96000, 192000 }; struct platform_device *pdev = spdif_priv->pdev; struct device *dev = &pdev->dev; u64 savesub = 100000, ret; diff --git a/sound/soc/fsl/fsl_spdif.h b/sound/soc/fsl/fsl_spdif.h index b13e30ea1c9e..00bd3514c610 100644 --- a/sound/soc/fsl/fsl_spdif.h +++ b/sound/soc/fsl/fsl_spdif.h @@ -166,8 +166,10 @@ enum spdif_txrate { SPDIF_TXRATE_32000 = 0, SPDIF_TXRATE_44100, SPDIF_TXRATE_48000, + SPDIF_TXRATE_96000, + SPDIF_TXRATE_192000, }; -#define SPDIF_TXRATE_MAX (SPDIF_TXRATE_48000 + 1) +#define SPDIF_TXRATE_MAX (SPDIF_TXRATE_192000 + 1) #define SPDIF_CSTATUS_BYTE 6 @@ -177,7 +179,9 @@ enum spdif_txrate { #define FSL_SPDIF_RATES_PLAYBACK (SNDRV_PCM_RATE_32000 | \ SNDRV_PCM_RATE_44100 | \ - SNDRV_PCM_RATE_48000) + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_192000) #define FSL_SPDIF_RATES_CAPTURE (SNDRV_PCM_RATE_16000 | \ SNDRV_PCM_RATE_32000 | \ -- cgit v1.2.3 From 7eced3ec080d7868715f8af4628c36726ef913e7 Mon Sep 17 00:00:00 2001 From: Christian Engelmayer Date: Fri, 13 Jun 2014 19:19:28 +0200 Subject: ASoC: wm8985: Remove unused pointer in wm8985_remove() Commit a0b148b4 (ASoC: wm8985: Use devm_regulator_bulk_get()) removed the last user of pointer wm8985 to struct wm8985_priv. Thus remove it. Detected by Coverity CID 1222150. Signed-off-by: Christian Engelmayer Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8985.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c index 0f5780c09f3a..cbbdfbbd0a9f 100644 --- a/sound/soc/codecs/wm8985.c +++ b/sound/soc/codecs/wm8985.c @@ -980,9 +980,6 @@ static int wm8985_resume(struct snd_soc_codec *codec) static int wm8985_remove(struct snd_soc_codec *codec) { - struct wm8985_priv *wm8985; - - wm8985 = snd_soc_codec_get_drvdata(codec); wm8985_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } -- cgit v1.2.3 From acf2c60a60b3d6d7080854b9483f37d99ded9b23 Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Fri, 13 Jun 2014 07:42:40 -0500 Subject: ASoC: fsl-ssi: fix do_div build warning in fsl_ssi_set_bclk() do_div() requires that the first parameter is a 64-bit integer, which but clkrate was defined as an unsigned long. This caused the following warnings: CC sound/soc/fsl/fsl_ssi.o sound/soc/fsl/fsl_ssi.c: In function 'fsl_ssi_set_bclk': sound/soc/fsl/fsl_ssi.c:593:3: warning: comparison of distinct pointer types lacks a cast sound/soc/fsl/fsl_ssi.c:593:3: warning: right shift count >= width of type sound/soc/fsl/fsl_ssi.c:593:3: warning: passing argument 1 of '__div64_32' from incompatible pointer type include/asm-generic/div64.h:35:17: note: expected 'uint64_t *' but argument is of type 'long unsigned int *' Signed-off-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 9bfef55d77d1..3043d576856b 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -590,8 +590,8 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, else clkrate = clk_round_rate(ssi_private->baudclk, tmprate); - do_div(clkrate, factor); - afreq = (u32)clkrate / (i + 1); + clkrate /= factor; + afreq = clkrate / (i + 1); if (freq == afreq) sub = 0; -- cgit v1.2.3 From b38314179c9ccb789e6fe967cff171fa817e8978 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Mon, 16 Jun 2014 21:24:03 +0100 Subject: ASoC: wm8994: Prevent double lock of accdet_lock mutex on wm1811 wm1811_micd_stop takes the accdet_lock mutex, and is called from two places, one of which is already holding the accdet_lock. This obviously causes a lock up. This patch fixes this issue by removing the lock from wm1811_micd_stop and ensuring that it is always locked externally. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm8994.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 247b39013fba..9719d3ca8e47 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3505,6 +3505,7 @@ static irqreturn_t wm8994_mic_irq(int irq, void *data) return IRQ_HANDLED; } +/* Should be called with accdet_lock held */ static void wm1811_micd_stop(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); @@ -3512,14 +3513,10 @@ static void wm1811_micd_stop(struct snd_soc_codec *codec) if (!wm8994->jackdet) return; - mutex_lock(&wm8994->accdet_lock); - snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, WM8958_MICD_ENA, 0); wm1811_jackdet_set_mode(codec, WM1811_JACKDET_MODE_JACK); - mutex_unlock(&wm8994->accdet_lock); - if (wm8994->wm8994->pdata.jd_ext_cap) snd_soc_dapm_disable_pin(&codec->dapm, "MICBIAS2"); @@ -3560,10 +3557,10 @@ static void wm8958_open_circuit_work(struct work_struct *work) open_circuit_work.work); struct device *dev = wm8994->wm8994->dev; - wm1811_micd_stop(wm8994->hubs.codec); - mutex_lock(&wm8994->accdet_lock); + wm1811_micd_stop(wm8994->hubs.codec); + dev_dbg(dev, "Reporting open circuit\n"); wm8994->jack_mic = false; -- cgit v1.2.3 From 5ba4059c385544001031ecff1335e10b3365112c Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Thu, 5 Jun 2014 12:40:04 +0200 Subject: ASoC: MMP audio needs sram support Building the pxa/mmp audio driver without support for the mmp sram driver enabled results in this link error: sound/built-in.o: In function `mmp_pcm_free_dma_buffers': :(.text+0x3e734): undefined reference to `sram_get_gpool' sound/built-in.o: In function `mmp_pcm_new': :(.text+0x3e7c0): undefined reference to `sram_get_gpool' The sram driver is cannot be manually enabled and needs to be turned on by selecting MMP_SRAM from each module that needs it, which is what this patch does. Ideally, MMP should move over to the generic SRAM support, but for the moment, we can avoid the build error. Signed-off-by: Arnd Bergmann Signed-off-by: Mark Brown --- sound/soc/pxa/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index ea6372b15dd5..a3a7385c969d 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -11,6 +11,7 @@ config SND_PXA2XX_SOC config SND_MMP_SOC bool "Soc Audio for Marvell MMP chips" depends on ARCH_MMP + select MMP_SRAM select SND_SOC_GENERIC_DMAENGINE_PCM select SND_ARM help -- cgit v1.2.3 From 64eae986fc1e3a281b00f04b7c9c00b145ec8a57 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 18 Jun 2014 17:54:43 +0900 Subject: ASoC: rsnd: fixup rsnd_gen_dma_addr() for Gen1 ad32d0c7b0e993433df152ae747652647eb65a27 (ASoC: rsnd: add rsnd_gen_dma_addr() for DMAC addr) added rsnd_gen_dma_addr() to calculate DMA addr, but, it is necessary only for Gen2. This patch ignores Gen1 case. Kernel will be panic without this patch. Special thanks to Simon Reported-by: Simon Horman Signed-off-by: Kuninori Morimoto Tested-by: Simon Horman Signed-off-by: Mark Brown --- sound/soc/sh/rcar/gen.c | 33 +++++++++++++++++++++------------ 1 file changed, 21 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 1dd2b7d38c2c..0280a11c0899 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -184,7 +184,7 @@ static int rsnd_gen_regmap_init(struct rsnd_priv *priv, #define RDMA_CMD_O_N(addr, i) (addr ##_reg - 0x004f8000 + (0x400 * i)) #define RDMA_CMD_O_P(addr, i) (addr ##_reg - 0x001f8000 + (0x400 * i)) -void rsnd_gen_dma_addr(struct rsnd_priv *priv, +static void rsnd_gen2_dma_addr(struct rsnd_priv *priv, struct rsnd_dma *dma, struct dma_slave_config *cfg, int is_play, int slave_id) @@ -226,17 +226,6 @@ void rsnd_gen_dma_addr(struct rsnd_priv *priv, } }; - cfg->slave_id = slave_id; - cfg->src_addr = 0; - cfg->dst_addr = 0; - cfg->direction = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; - - /* - * gen1 uses default DMA addr - */ - if (rsnd_is_gen1(priv)) - return; - /* it shouldn't happen */ if (use_dvc & !use_src) { dev_err(dev, "DVC is selected without SRC\n"); @@ -250,6 +239,26 @@ void rsnd_gen_dma_addr(struct rsnd_priv *priv, id, cfg->src_addr, cfg->dst_addr); } +void rsnd_gen_dma_addr(struct rsnd_priv *priv, + struct rsnd_dma *dma, + struct dma_slave_config *cfg, + int is_play, int slave_id) +{ + cfg->slave_id = slave_id; + cfg->src_addr = 0; + cfg->dst_addr = 0; + cfg->direction = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; + + /* + * gen1 uses default DMA addr + */ + if (rsnd_is_gen1(priv)) + return; + + rsnd_gen2_dma_addr(priv, dma, cfg, is_play, slave_id); +} + + /* * Gen2 */ -- cgit v1.2.3 From c08c3b088053cec1465051258844e7934d3e3e37 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 18 Jun 2014 17:55:09 +0900 Subject: ASoC: rsnd: fixup loop exit timing of dma name search Current dma name search loop didn't care about SSI index This patch fixes it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 4e86265f625c..ed76901f8202 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -297,7 +297,6 @@ static void rsnd_dma_of_name(struct rsnd_dma *dma, for (i = 1; i < MOD_MAX; i++) { if (!src) { mod[i] = ssi; - break; } else if (!dvc) { mod[i] = src; src = NULL; @@ -308,6 +307,9 @@ static void rsnd_dma_of_name(struct rsnd_dma *dma, if (mod[i] == this) index = i; + + if (mod[i] == ssi) + break; } if (is_play) { -- cgit v1.2.3 From fb668e735bef560291429d0694a5020695062cbb Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 27 Mar 2014 21:42:14 +0100 Subject: ASoC: ak5386: add regulator consumer support The chip has two power supplies, VA and VDD. Enable them both as long as the codec is in use. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/codecs/ak5386.c | 50 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 50 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/ak5386.c b/sound/soc/codecs/ak5386.c index 72e953b2cb41..a30be5ce3fa4 100644 --- a/sound/soc/codecs/ak5386.c +++ b/sound/soc/codecs/ak5386.c @@ -14,12 +14,18 @@ #include #include #include +#include #include #include #include +static const char const *supply_names[] = { + "va", "vd" +}; + struct ak5386_priv { int reset_gpio; + struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)]; }; static const struct snd_soc_dapm_widget ak5386_dapm_widgets[] = { @@ -32,7 +38,42 @@ static const struct snd_soc_dapm_route ak5386_dapm_routes[] = { { "Capture", NULL, "AINR" }, }; +static int ak5386_soc_probe(struct snd_soc_codec *codec) +{ + struct ak5386_priv *priv = snd_soc_codec_get_drvdata(codec); + return regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies); +} + +static int ak5386_soc_remove(struct snd_soc_codec *codec) +{ + struct ak5386_priv *priv = snd_soc_codec_get_drvdata(codec); + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); + return 0; +} + +#ifdef CONFIG_PM +static int ak5386_soc_suspend(struct snd_soc_codec *codec) +{ + struct ak5386_priv *priv = snd_soc_codec_get_drvdata(codec); + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); + return 0; +} + +static int ak5386_soc_resume(struct snd_soc_codec *codec) +{ + struct ak5386_priv *priv = snd_soc_codec_get_drvdata(codec); + return regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies); +} +#else +#define ak5386_soc_suspend NULL +#define ak5386_soc_resume NULL +#endif /* CONFIG_PM */ + static struct snd_soc_codec_driver soc_codec_ak5386 = { + .probe = ak5386_soc_probe, + .remove = ak5386_soc_remove, + .suspend = ak5386_soc_suspend, + .resume = ak5386_soc_resume, .dapm_widgets = ak5386_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(ak5386_dapm_widgets), .dapm_routes = ak5386_dapm_routes, @@ -122,6 +163,7 @@ static int ak5386_probe(struct platform_device *pdev) { struct device *dev = &pdev->dev; struct ak5386_priv *priv; + int ret, i; priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); if (!priv) @@ -130,6 +172,14 @@ static int ak5386_probe(struct platform_device *pdev) priv->reset_gpio = -EINVAL; dev_set_drvdata(dev, priv); + for (i = 0; i < ARRAY_SIZE(supply_names); i++) + priv->supplies[i].supply = supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(priv->supplies), + priv->supplies); + if (ret < 0) + return ret; + if (of_match_device(of_match_ptr(ak5386_dt_ids), dev)) priv->reset_gpio = of_get_named_gpio(dev->of_node, "reset-gpio", 0); -- cgit v1.2.3 From 2ad76541026ba5c6c93b5a5a6f1418aa89fa5f34 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Fri, 28 Mar 2014 19:05:04 +0100 Subject: ASoC: ak5386: Add regulators to documentation and fix sparse warning Document the newly added regulators to the DT binding document. Also, "static const char const *x" is not identical to "static const char * const x", which sparse now complains about. Fix it. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/ak5386.txt | 4 ++++ sound/soc/codecs/ak5386.c | 2 +- 2 files changed, 5 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/ak5386.txt b/Documentation/devicetree/bindings/sound/ak5386.txt index dc3914fe6ce8..ec3df3abba0c 100644 --- a/Documentation/devicetree/bindings/sound/ak5386.txt +++ b/Documentation/devicetree/bindings/sound/ak5386.txt @@ -10,10 +10,14 @@ Optional properties: - reset-gpio : a GPIO spec for the reset/power down pin. If specified, it will be deasserted at probe time. + - va-supply : a regulator spec, providing 5.0V + - vd-supply : a regulator spec, providing 3.3V Example: spdif: ak5386@0 { compatible = "asahi-kasei,ak5386"; reset-gpio = <&gpio0 23>; + va-supply = <&vdd_5v0_reg>; + vd-supply = <&vdd_3v3_reg>; }; diff --git a/sound/soc/codecs/ak5386.c b/sound/soc/codecs/ak5386.c index a30be5ce3fa4..8107a1cac876 100644 --- a/sound/soc/codecs/ak5386.c +++ b/sound/soc/codecs/ak5386.c @@ -19,7 +19,7 @@ #include #include -static const char const *supply_names[] = { +static const char * const supply_names[] = { "va", "vd" }; -- cgit v1.2.3 From 6479285d8aa1cbf22d21706370e812e7af51241c Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 27 Mar 2014 11:27:40 +0100 Subject: ASoC: davinci-mcasp: set up channel status bits for S/PDIF mode In DIT (S/PDIF) mode, program the transmitted user bits to reflect the configured sample rate, along with some other details. Signed-off-by: Daniel Mack Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 49 +++++++++++++++++++++++++++++++++++++-- 1 file changed, 47 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 9afb14629a17..121971e1371e 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -27,6 +27,7 @@ #include #include +#include #include #include #include @@ -637,8 +638,12 @@ static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream) } /* S/PDIF */ -static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp) +static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp, + unsigned int rate) { + u32 cs_value = 0; + u8 *cs_bytes = (u8*) &cs_value; + /* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0 and LSB first */ mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXROT(6) | TXSSZ(15)); @@ -660,6 +665,46 @@ static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp) /* Enable the DIT */ mcasp_set_bits(mcasp, DAVINCI_MCASP_TXDITCTL_REG, DITEN); + /* Set S/PDIF channel status bits */ + cs_bytes[0] = IEC958_AES0_CON_NOT_COPYRIGHT; + cs_bytes[1] = IEC958_AES1_CON_PCM_CODER; + + switch (rate) { + case 22050: + cs_bytes[3] |= IEC958_AES3_CON_FS_22050; + break; + case 24000: + cs_bytes[3] |= IEC958_AES3_CON_FS_24000; + break; + case 32000: + cs_bytes[3] |= IEC958_AES3_CON_FS_32000; + break; + case 44100: + cs_bytes[3] |= IEC958_AES3_CON_FS_44100; + break; + case 48000: + cs_bytes[3] |= IEC958_AES3_CON_FS_48000; + break; + case 88200: + cs_bytes[3] |= IEC958_AES3_CON_FS_88200; + break; + case 96000: + cs_bytes[3] |= IEC958_AES3_CON_FS_96000; + break; + case 176400: + cs_bytes[3] |= IEC958_AES3_CON_FS_176400; + break; + case 192000: + cs_bytes[3] |= IEC958_AES3_CON_FS_192000; + break; + default: + printk(KERN_WARNING "unsupported sampling rate: %d\n", rate); + return -EINVAL; + } + + mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRA_REG, cs_value); + mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRB_REG, cs_value); + return 0; } @@ -692,7 +737,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, return ret; if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) - ret = mcasp_dit_hw_param(mcasp); + ret = mcasp_dit_hw_param(mcasp, params_rate(params)); else ret = mcasp_i2s_hw_param(mcasp, substream->stream); -- cgit v1.2.3 From e73a2571988870372f7f2b6072a98f49e414be66 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 19 Jun 2014 07:50:01 +0200 Subject: ASoC: wm5100/wm8903/wm8996: Replace open-coded snd_soc_dapm_to_codec() We now have a generic helper function to cast from a DAPM context to a CODEC. Make use of it in the places which previously open-coded it. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm5100.c | 3 +-- sound/soc/codecs/wm8903.c | 3 +-- sound/soc/codecs/wm8996.c | 3 +-- 3 files changed, 3 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index eca983fad891..42a3ff3a1b68 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -735,8 +735,7 @@ WM5100_MIXER_CONTROLS("LHPF4", WM5100_HPLP4MIX_INPUT_1_SOURCE), static void wm5100_seq_notifier(struct snd_soc_dapm_context *dapm, enum snd_soc_dapm_type event, int subseq) { - struct snd_soc_codec *codec = container_of(dapm, - struct snd_soc_codec, dapm); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); u16 val, expect, i; diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index b0084a127d18..2116e79085ec 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -281,8 +281,7 @@ static int wm8903_dcs_event(struct snd_soc_dapm_widget *w, static void wm8903_seq_notifier(struct snd_soc_dapm_context *dapm, enum snd_soc_dapm_type event, int subseq) { - struct snd_soc_codec *codec = container_of(dapm, - struct snd_soc_codec, dapm); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); int dcs_mode = WM8903_DCS_MODE_WRITE_STOP; int i, val; diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index c6cbb3b8ace9..8f4d2cddb962 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -690,8 +690,7 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, u16 mask) static void wm8996_seq_notifier(struct snd_soc_dapm_context *dapm, enum snd_soc_dapm_type event, int subseq) { - struct snd_soc_codec *codec = container_of(dapm, - struct snd_soc_codec, dapm); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); u16 val, mask; -- cgit v1.2.3 From 4adeb0ccf86a5af1825bbfe290dee9e60a5ab870 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Thu, 19 Jun 2014 09:32:05 +0300 Subject: ASoC: max98090: Fix missing free_irq max98090.c doesn't free the threaded interrupt it requests. This causes an oops when doing "cat /proc/interrupts" after snd-soc-max98090.ko is unloaded. Fix this by requesting the interrupt by using devm_request_threaded_irq(). Signed-off-by: Jarkko Nikula Cc: Stable # 3.10+ Signed-off-by: Mark Brown --- sound/soc/codecs/max98090.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index f5fccc7a8e89..d97f1ce7ff7d 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2284,7 +2284,7 @@ static int max98090_probe(struct snd_soc_codec *codec) /* Register for interrupts */ dev_dbg(codec->dev, "irq = %d\n", max98090->irq); - ret = request_threaded_irq(max98090->irq, NULL, + ret = devm_request_threaded_irq(codec->dev, max98090->irq, NULL, max98090_interrupt, IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "max98090_interrupt", codec); if (ret < 0) { -- cgit v1.2.3 From 6c49a986957bc5fe39b29166cb7fad573dc242ba Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Thu, 19 Jun 2014 09:44:26 +0300 Subject: ASoC: max98090: Remove needless defines and line feeds Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/codecs/max98090.c | 13 ------------- 1 file changed, 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index f5fccc7a8e89..3aec3ae78fe0 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -26,10 +26,6 @@ #include #include "max98090.h" -#define DEBUG -#define EXTMIC_METHOD -#define EXTMIC_METHOD_TEST - /* Allows for sparsely populated register maps */ static struct reg_default max98090_reg[] = { { 0x00, 0x00 }, /* 00 Software Reset */ @@ -820,7 +816,6 @@ static int max98090_micinput_event(struct snd_soc_dapm_widget *w, else val = (val & M98090_MIC_PA2EN_MASK) >> M98090_MIC_PA2EN_SHIFT; - if (val >= 1) { if (w->reg == M98090_REG_MIC1_INPUT_LEVEL) { max98090->pa1en = val - 1; /* Update for volatile */ @@ -1140,7 +1135,6 @@ static const struct snd_kcontrol_new max98090_mixhprsel_mux = SOC_DAPM_ENUM("MIXHPRSEL Mux", mixhprsel_mux_enum); static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { - SND_SOC_DAPM_INPUT("MIC1"), SND_SOC_DAPM_INPUT("MIC2"), SND_SOC_DAPM_INPUT("DMICL"), @@ -1304,7 +1298,6 @@ static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { }; static const struct snd_soc_dapm_widget max98091_dapm_widgets[] = { - SND_SOC_DAPM_INPUT("DMIC3"), SND_SOC_DAPM_INPUT("DMIC4"), @@ -1315,7 +1308,6 @@ static const struct snd_soc_dapm_widget max98091_dapm_widgets[] = { }; static const struct snd_soc_dapm_route max98090_dapm_routes[] = { - {"MIC1 Input", NULL, "MIC1"}, {"MIC2 Input", NULL, "MIC2"}, @@ -1493,17 +1485,14 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = { {"SPKR", NULL, "SPK Right Out"}, {"RCVL", NULL, "RCV Left Out"}, {"RCVR", NULL, "RCV Right Out"}, - }; static const struct snd_soc_dapm_route max98091_dapm_routes[] = { - /* DMIC inputs */ {"DMIC3", NULL, "DMIC3_ENA"}, {"DMIC4", NULL, "DMIC4_ENA"}, {"DMIC3", NULL, "AHPF"}, {"DMIC4", NULL, "AHPF"}, - }; static int max98090_add_widgets(struct snd_soc_codec *codec) @@ -1531,7 +1520,6 @@ static int max98090_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(dapm, max98091_dapm_routes, ARRAY_SIZE(max98091_dapm_routes)); - } return 0; @@ -2221,7 +2209,6 @@ static void max98090_handle_pdata(struct snd_soc_codec *codec) dev_err(codec->dev, "No platform data\n"); return; } - } static int max98090_probe(struct snd_soc_codec *codec) -- cgit v1.2.3 From a28d167fbbef1f31d79ad3ad65a59ea6fa4d1b1f Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 13 Jun 2014 09:57:45 +0530 Subject: ASoC: mc13783: Add missing of_node_put of_get_child_by_name expects of_node_put be called when done. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/mc13783.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 9965277b595a..388f90a597fa 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -766,11 +766,11 @@ static int __init mc13783_codec_probe(struct platform_device *pdev) ret = of_property_read_u32(np, "adc-port", &priv->adc_ssi_port); if (ret) - return ret; + goto out; ret = of_property_read_u32(np, "dac-port", &priv->dac_ssi_port); if (ret) - return ret; + goto out; } dev_set_drvdata(&pdev->dev, priv); @@ -783,6 +783,8 @@ static int __init mc13783_codec_probe(struct platform_device *pdev) ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_mc13783, mc13783_dai_async, ARRAY_SIZE(mc13783_dai_async)); +out: + of_node_put(np); return ret; } -- cgit v1.2.3 From 7f28f357840fc857348b7ed42d7ee697cd221a59 Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Fri, 13 Jun 2014 12:49:59 +0300 Subject: ASoC: davinci-mcasp: Add dependecy to SND_DAVINCI_SOC or SND_OMAP_SOC Fixes build with SND_DAVINCI_SOC or SND_OMAP_SOC alone and adds build dependecy to SND_DAVINCI_SOC or SND_OMAP_SOC. Signed-off-by: Jyri Sarha Signed-off-by: Mark Brown --- sound/soc/davinci/Kconfig | 1 + sound/soc/davinci/davinci-mcasp.c | 8 ++++++++ 2 files changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 50a098749b9e..fdbb16fffd30 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -6,6 +6,7 @@ config SND_DAVINCI_SOC_I2S tristate config SND_DAVINCI_SOC_MCASP + depends on SND_DAVINCI_SOC || SND_OMAP_SOC tristate config SND_DAVINCI_SOC_VCIF diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 9afb14629a17..0ee4986038cc 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1223,14 +1223,22 @@ static int davinci_mcasp_probe(struct platform_device *pdev) goto err; switch (mcasp->version) { +#if IS_BUILTIN(CONFIG_SND_DAVINCI_SOC) || \ + (IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \ + IS_MODULE(CONFIG_SND_DAVINCI_SOC)) case MCASP_VERSION_1: case MCASP_VERSION_2: case MCASP_VERSION_3: ret = davinci_soc_platform_register(&pdev->dev); break; +#endif +#if IS_BUILTIN(CONFIG_SND_OMAP_SOC) || \ + (IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \ + IS_MODULE(CONFIG_SND_OMAP_SOC)) case MCASP_VERSION_4: ret = omap_pcm_platform_register(&pdev->dev); break; +#endif default: dev_err(&pdev->dev, "Invalid McASP version: %d\n", mcasp->version); -- cgit v1.2.3 From 0929878f93be5534974e058bc1e4b3abb36478b5 Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Fri, 13 Jun 2014 12:50:00 +0300 Subject: ASoC: davinci-mcasp: Allow best effort in selecting BCLK divider Do not fail if the exact BLCK rate can not be produced, just print a warning. Check that sysclk frequency is set before implicitly setting the BCLK divider. Signed-off-by: Jyri Sarha Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 121971e1371e..5b81adb3c93e 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -721,14 +721,18 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, int ret; /* If mcasp is BCLK master we need to set BCLK divider */ - if (mcasp->bclk_master) { + if (mcasp->bclk_master && mcasp->sysclk_freq) { unsigned int bclk_freq = snd_soc_params_to_bclk(params); + unsigned int div = mcasp->sysclk_freq / bclk_freq; if (mcasp->sysclk_freq % bclk_freq != 0) { - dev_err(mcasp->dev, "Can't produce required BCLK\n"); - return -EINVAL; + if (((mcasp->sysclk_freq / div) - bclk_freq) > + (bclk_freq - (mcasp->sysclk_freq / (div+1)))) + div++; + dev_warn(mcasp->dev, + "Inaccurate BCLK: %u Hz / %u != %u Hz\n", + mcasp->sysclk_freq, div, bclk_freq); } - davinci_mcasp_set_clkdiv( - cpu_dai, 1, mcasp->sysclk_freq / bclk_freq); + davinci_mcasp_set_clkdiv(cpu_dai, 1, div); } ret = mcasp_common_hw_param(mcasp, substream->stream, -- cgit v1.2.3 From c7099eb1c19e60251e6725d6302354dfabae5303 Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Fri, 13 Jun 2014 13:04:36 +0300 Subject: ASoC: simple-card: Make u32 DT parameter handling 64-bit proof Passing unsigned int pointers as u32 ponters may be dangerous on 64-bit system. Signed-off-by: Jyri Sarha Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 03a7fdcdf114..159e517fa09a 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -116,6 +116,7 @@ asoc_simple_card_sub_parse_of(struct device_node *np, { struct device_node *node; struct clk *clk; + u32 val; int ret; /* @@ -151,10 +152,8 @@ asoc_simple_card_sub_parse_of(struct device_node *np, } dai->sysclk = clk_get_rate(clk); - } else if (of_property_read_bool(np, "system-clock-frequency")) { - of_property_read_u32(np, - "system-clock-frequency", - &dai->sysclk); + } else if (!of_property_read_u32(np, "system-clock-frequency", &val)) { + dai->sysclk = val; } else { clk = of_clk_get(node, 0); if (!IS_ERR(clk)) @@ -303,6 +302,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, { struct snd_soc_dai_link *dai_link = priv->snd_card.dai_link; struct simple_dai_props *dai_props = priv->dai_props; + u32 val; int ret; /* parsing the card name from DT */ @@ -325,8 +325,9 @@ static int asoc_simple_card_parse_of(struct device_node *node, } /* Factor to mclk, used in hw_params() */ - of_property_read_u32(node, "simple-audio-card,mclk-fs", - &priv->mclk_fs); + ret = of_property_read_u32(node, "simple-audio-card,mclk-fs", &val); + if (ret == 0) + priv->mclk_fs = val; dev_dbg(dev, "New simple-card: %s\n", priv->snd_card.name ? priv->snd_card.name : ""); -- cgit v1.2.3 From 7ed9de76ff342cbd717a9cf897044b99272cb8f8 Mon Sep 17 00:00:00 2001 From: Qiao Zhou Date: Wed, 4 Jun 2014 19:42:06 +0800 Subject: ASoC: pcm: fix dpcm_path_put in dpcm runtime update we need to release dapm widget list after dpcm_path_get in soc_dpcm_runtime_update. otherwise, there will be potential memory leak. add dpcm_path_put to fix it. Signed-off-by: Qiao Zhou Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/soc-pcm.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 54d18f22a33e..4ea656770d65 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2069,6 +2069,7 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card) dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK); } + dpcm_path_put(&list); capture: /* skip if FE doesn't have capture capability */ if (!fe->cpu_dai->driver->capture.channels_min) -- cgit v1.2.3 From 1657caf5d8c5f96528bb6b041028174ee2681681 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 10 Jun 2014 11:35:56 +0800 Subject: ASoC: rt5640: Remove unneeded goto in rt5640_i2c_probe Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/rt5640.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index de80e89b5fd8..6bc6efdec550 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2215,14 +2215,8 @@ static int rt5640_i2c_probe(struct i2c_client *i2c, rt5640->hp_mute = 1; - ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640, - rt5640_dai, ARRAY_SIZE(rt5640_dai)); - if (ret < 0) - goto err; - - return 0; -err: - return ret; + return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640, + rt5640_dai, ARRAY_SIZE(rt5640_dai)); } static int rt5640_i2c_remove(struct i2c_client *i2c) -- cgit v1.2.3 From dd56ebadf4bfccb47a3b82d0b01a99c22967abc8 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 10 Jun 2014 11:36:41 +0800 Subject: ASoC: rt5645: Remove unneeded goto in rt5645_i2c_probe Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 02147be2b302..a7762d0a623e 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2345,14 +2345,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, } - ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5645, - rt5645_dai, ARRAY_SIZE(rt5645_dai)); - if (ret < 0) - goto err; - - return 0; -err: - return ret; + return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5645, + rt5645_dai, ARRAY_SIZE(rt5645_dai)); } static int rt5645_i2c_remove(struct i2c_client *i2c) -- cgit v1.2.3 From d0bdcb9181873bc085ca0b7fabefb92eb2d9e708 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 10 Jun 2014 11:37:24 +0800 Subject: ASoC: rt5677: Remove unneeded goto in rt5677_i2c_probe Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 45f99a824c15..a18740c31b82 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -3426,14 +3426,8 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5677->regmap, RT5677_IN1, RT5677_IN_DF2, RT5677_IN_DF2); - ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5677, - rt5677_dai, ARRAY_SIZE(rt5677_dai)); - if (ret < 0) - goto err; - - return 0; -err: - return ret; + return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5677, + rt5677_dai, ARRAY_SIZE(rt5677_dai)); } static int rt5677_i2c_remove(struct i2c_client *i2c) -- cgit v1.2.3 From 3d0c03d9c6a0cfd7e3a12c29ca5eefed02da16ff Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Tue, 10 Jun 2014 14:35:23 +0800 Subject: ASoC: rt5677: Replace the string "micbias1" to "MICBIAS1" The patch replaces the string "micbias1" to "MICBIAS1". Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index a18740c31b82..6153e35ed428 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -1672,7 +1672,7 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { /* Input Side */ /* micbias */ - SND_SOC_DAPM_SUPPLY("micbias1", RT5677_PWR_ANLG2, RT5677_PWR_MB1_BIT, + SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5677_PWR_ANLG2, RT5677_PWR_MB1_BIT, 0, rt5677_set_micbias1_event, SND_SOC_DAPM_POST_PMU), /* Input Lines */ -- cgit v1.2.3 From 1b7fd76ad93e7f121ae76d9b3843c873bff50587 Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Tue, 10 Jun 2014 14:35:24 +0800 Subject: ASoC: rt5677: Replace the string "source" to "Source" The patch replaces the string "source" to "Source". Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 122 +++++++++++++++++++++++----------------------- 1 file changed, 61 insertions(+), 61 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 6153e35ed428..2919f4417b3e 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -938,7 +938,7 @@ static const struct snd_kcontrol_new rt5677_ob_7_mix[] = { /* Mux */ -/* DAC1 L/R source */ /* MX-29 [10:8] */ +/* DAC1 L/R Source */ /* MX-29 [10:8] */ static const char * const rt5677_dac1_src[] = { "IF1 DAC 01", "IF2 DAC 01", "IF3 DAC LR", "IF4 DAC LR", "SLB DAC 01", "OB 01" @@ -949,9 +949,9 @@ static SOC_ENUM_SINGLE_DECL( RT5677_DAC1_L_SEL_SFT, rt5677_dac1_src); static const struct snd_kcontrol_new rt5677_dac1_mux = - SOC_DAPM_ENUM("DAC1 source", rt5677_dac1_enum); + SOC_DAPM_ENUM("DAC1 Source", rt5677_dac1_enum); -/* ADDA1 L/R source */ /* MX-29 [1:0] */ +/* ADDA1 L/R Source */ /* MX-29 [1:0] */ static const char * const rt5677_adda1_src[] = { "STO1 ADC MIX", "STO2 ADC MIX", "OB 67", }; @@ -961,10 +961,10 @@ static SOC_ENUM_SINGLE_DECL( RT5677_ADDA1_SEL_SFT, rt5677_adda1_src); static const struct snd_kcontrol_new rt5677_adda1_mux = - SOC_DAPM_ENUM("ADDA1 source", rt5677_adda1_enum); + SOC_DAPM_ENUM("ADDA1 Source", rt5677_adda1_enum); -/*DAC2 L/R source*/ /* MX-1B [6:4] [2:0] */ +/*DAC2 L/R Source*/ /* MX-1B [6:4] [2:0] */ static const char * const rt5677_dac2l_src[] = { "IF1 DAC 2", "IF2 DAC 2", "IF3 DAC L", "IF4 DAC L", "SLB DAC 2", "OB 2", @@ -975,7 +975,7 @@ static SOC_ENUM_SINGLE_DECL( RT5677_SEL_DAC2_L_SRC_SFT, rt5677_dac2l_src); static const struct snd_kcontrol_new rt5677_dac2_l_mux = - SOC_DAPM_ENUM("DAC2 L source", rt5677_dac2l_enum); + SOC_DAPM_ENUM("DAC2 L Source", rt5677_dac2l_enum); static const char * const rt5677_dac2r_src[] = { "IF1 DAC 3", "IF2 DAC 3", "IF3 DAC R", "IF4 DAC R", "SLB DAC 3", @@ -987,9 +987,9 @@ static SOC_ENUM_SINGLE_DECL( RT5677_SEL_DAC2_R_SRC_SFT, rt5677_dac2r_src); static const struct snd_kcontrol_new rt5677_dac2_r_mux = - SOC_DAPM_ENUM("DAC2 R source", rt5677_dac2r_enum); + SOC_DAPM_ENUM("DAC2 R Source", rt5677_dac2r_enum); -/*DAC3 L/R source*/ /* MX-16 [6:4] [2:0] */ +/*DAC3 L/R Source*/ /* MX-16 [6:4] [2:0] */ static const char * const rt5677_dac3l_src[] = { "IF1 DAC 4", "IF2 DAC 4", "IF3 DAC L", "IF4 DAC L", "SLB DAC 4", "OB 4" @@ -1000,7 +1000,7 @@ static SOC_ENUM_SINGLE_DECL( RT5677_SEL_DAC3_L_SRC_SFT, rt5677_dac3l_src); static const struct snd_kcontrol_new rt5677_dac3_l_mux = - SOC_DAPM_ENUM("DAC3 L source", rt5677_dac3l_enum); + SOC_DAPM_ENUM("DAC3 L Source", rt5677_dac3l_enum); static const char * const rt5677_dac3r_src[] = { "IF1 DAC 5", "IF2 DAC 5", "IF3 DAC R", "IF4 DAC R", @@ -1012,9 +1012,9 @@ static SOC_ENUM_SINGLE_DECL( RT5677_SEL_DAC3_R_SRC_SFT, rt5677_dac3r_src); static const struct snd_kcontrol_new rt5677_dac3_r_mux = - SOC_DAPM_ENUM("DAC3 R source", rt5677_dac3r_enum); + SOC_DAPM_ENUM("DAC3 R Source", rt5677_dac3r_enum); -/*DAC4 L/R source*/ /* MX-16 [14:12] [10:8] */ +/*DAC4 L/R Source*/ /* MX-16 [14:12] [10:8] */ static const char * const rt5677_dac4l_src[] = { "IF1 DAC 6", "IF2 DAC 6", "IF3 DAC L", "IF4 DAC L", "SLB DAC 6", "OB 6" @@ -1025,7 +1025,7 @@ static SOC_ENUM_SINGLE_DECL( RT5677_SEL_DAC4_L_SRC_SFT, rt5677_dac4l_src); static const struct snd_kcontrol_new rt5677_dac4_l_mux = - SOC_DAPM_ENUM("DAC4 L source", rt5677_dac4l_enum); + SOC_DAPM_ENUM("DAC4 L Source", rt5677_dac4l_enum); static const char * const rt5677_dac4r_src[] = { "IF1 DAC 7", "IF2 DAC 7", "IF3 DAC R", "IF4 DAC R", @@ -1037,7 +1037,7 @@ static SOC_ENUM_SINGLE_DECL( RT5677_SEL_DAC4_R_SRC_SFT, rt5677_dac4r_src); static const struct snd_kcontrol_new rt5677_dac4_r_mux = - SOC_DAPM_ENUM("DAC4 R source", rt5677_dac4r_enum); + SOC_DAPM_ENUM("DAC4 R Source", rt5677_dac4r_enum); /* In/OutBound Source Pass SRC */ /* MX-A5 [3] [4] [0] [1] [2] */ static const char * const rt5677_iob_bypass_src[] = { @@ -1049,35 +1049,35 @@ static SOC_ENUM_SINGLE_DECL( RT5677_SEL_SRC_OB01_SFT, rt5677_iob_bypass_src); static const struct snd_kcontrol_new rt5677_ob01_bypass_src_mux = - SOC_DAPM_ENUM("OB01 Bypass source", rt5677_ob01_bypass_src_enum); + SOC_DAPM_ENUM("OB01 Bypass Source", rt5677_ob01_bypass_src_enum); static SOC_ENUM_SINGLE_DECL( rt5677_ob23_bypass_src_enum, RT5677_DSP_IN_OUTB_CTRL, RT5677_SEL_SRC_OB23_SFT, rt5677_iob_bypass_src); static const struct snd_kcontrol_new rt5677_ob23_bypass_src_mux = - SOC_DAPM_ENUM("OB23 Bypass source", rt5677_ob23_bypass_src_enum); + SOC_DAPM_ENUM("OB23 Bypass Source", rt5677_ob23_bypass_src_enum); static SOC_ENUM_SINGLE_DECL( rt5677_ib01_bypass_src_enum, RT5677_DSP_IN_OUTB_CTRL, RT5677_SEL_SRC_IB01_SFT, rt5677_iob_bypass_src); static const struct snd_kcontrol_new rt5677_ib01_bypass_src_mux = - SOC_DAPM_ENUM("IB01 Bypass source", rt5677_ib01_bypass_src_enum); + SOC_DAPM_ENUM("IB01 Bypass Source", rt5677_ib01_bypass_src_enum); static SOC_ENUM_SINGLE_DECL( rt5677_ib23_bypass_src_enum, RT5677_DSP_IN_OUTB_CTRL, RT5677_SEL_SRC_IB23_SFT, rt5677_iob_bypass_src); static const struct snd_kcontrol_new rt5677_ib23_bypass_src_mux = - SOC_DAPM_ENUM("IB23 Bypass source", rt5677_ib23_bypass_src_enum); + SOC_DAPM_ENUM("IB23 Bypass Source", rt5677_ib23_bypass_src_enum); static SOC_ENUM_SINGLE_DECL( rt5677_ib45_bypass_src_enum, RT5677_DSP_IN_OUTB_CTRL, RT5677_SEL_SRC_IB45_SFT, rt5677_iob_bypass_src); static const struct snd_kcontrol_new rt5677_ib45_bypass_src_mux = - SOC_DAPM_ENUM("IB45 Bypass source", rt5677_ib45_bypass_src_enum); + SOC_DAPM_ENUM("IB45 Bypass Source", rt5677_ib45_bypass_src_enum); /* Stereo ADC Source 2 */ /* MX-27 MX26 MX25 [11:10] */ static const char * const rt5677_stereo_adc2_src[] = { @@ -1089,21 +1089,21 @@ static SOC_ENUM_SINGLE_DECL( RT5677_SEL_STO1_ADC2_SFT, rt5677_stereo_adc2_src); static const struct snd_kcontrol_new rt5677_sto1_adc2_mux = - SOC_DAPM_ENUM("Stereo1 ADC2 source", rt5677_stereo1_adc2_enum); + SOC_DAPM_ENUM("Stereo1 ADC2 Source", rt5677_stereo1_adc2_enum); static SOC_ENUM_SINGLE_DECL( rt5677_stereo2_adc2_enum, RT5677_STO2_ADC_MIXER, RT5677_SEL_STO2_ADC2_SFT, rt5677_stereo_adc2_src); static const struct snd_kcontrol_new rt5677_sto2_adc2_mux = - SOC_DAPM_ENUM("Stereo2 ADC2 source", rt5677_stereo2_adc2_enum); + SOC_DAPM_ENUM("Stereo2 ADC2 Source", rt5677_stereo2_adc2_enum); static SOC_ENUM_SINGLE_DECL( rt5677_stereo3_adc2_enum, RT5677_STO3_ADC_MIXER, RT5677_SEL_STO3_ADC2_SFT, rt5677_stereo_adc2_src); static const struct snd_kcontrol_new rt5677_sto3_adc2_mux = - SOC_DAPM_ENUM("Stereo3 ADC2 source", rt5677_stereo3_adc2_enum); + SOC_DAPM_ENUM("Stereo3 ADC2 Source", rt5677_stereo3_adc2_enum); /* DMIC Source */ /* MX-28 [9:8][1:0] MX-27 MX-26 MX-25 MX-24 [9:8] */ static const char * const rt5677_dmic_src[] = { @@ -1115,44 +1115,44 @@ static SOC_ENUM_SINGLE_DECL( RT5677_SEL_MONO_DMIC_L_SFT, rt5677_dmic_src); static const struct snd_kcontrol_new rt5677_mono_dmic_l_mux = - SOC_DAPM_ENUM("Mono DMIC L source", rt5677_mono_dmic_l_enum); + SOC_DAPM_ENUM("Mono DMIC L Source", rt5677_mono_dmic_l_enum); static SOC_ENUM_SINGLE_DECL( rt5677_mono_dmic_r_enum, RT5677_MONO_ADC_MIXER, RT5677_SEL_MONO_DMIC_R_SFT, rt5677_dmic_src); static const struct snd_kcontrol_new rt5677_mono_dmic_r_mux = - SOC_DAPM_ENUM("Mono DMIC R source", rt5677_mono_dmic_r_enum); + SOC_DAPM_ENUM("Mono DMIC R Source", rt5677_mono_dmic_r_enum); static SOC_ENUM_SINGLE_DECL( rt5677_stereo1_dmic_enum, RT5677_STO1_ADC_MIXER, RT5677_SEL_STO1_DMIC_SFT, rt5677_dmic_src); static const struct snd_kcontrol_new rt5677_sto1_dmic_mux = - SOC_DAPM_ENUM("Stereo1 DMIC source", rt5677_stereo1_dmic_enum); + SOC_DAPM_ENUM("Stereo1 DMIC Source", rt5677_stereo1_dmic_enum); static SOC_ENUM_SINGLE_DECL( rt5677_stereo2_dmic_enum, RT5677_STO2_ADC_MIXER, RT5677_SEL_STO2_DMIC_SFT, rt5677_dmic_src); static const struct snd_kcontrol_new rt5677_sto2_dmic_mux = - SOC_DAPM_ENUM("Stereo2 DMIC source", rt5677_stereo2_dmic_enum); + SOC_DAPM_ENUM("Stereo2 DMIC Source", rt5677_stereo2_dmic_enum); static SOC_ENUM_SINGLE_DECL( rt5677_stereo3_dmic_enum, RT5677_STO3_ADC_MIXER, RT5677_SEL_STO3_DMIC_SFT, rt5677_dmic_src); static const struct snd_kcontrol_new rt5677_sto3_dmic_mux = - SOC_DAPM_ENUM("Stereo3 DMIC source", rt5677_stereo3_dmic_enum); + SOC_DAPM_ENUM("Stereo3 DMIC Source", rt5677_stereo3_dmic_enum); static SOC_ENUM_SINGLE_DECL( rt5677_stereo4_dmic_enum, RT5677_STO4_ADC_MIXER, RT5677_SEL_STO4_DMIC_SFT, rt5677_dmic_src); static const struct snd_kcontrol_new rt5677_sto4_dmic_mux = - SOC_DAPM_ENUM("Stereo4 DMIC source", rt5677_stereo4_dmic_enum); + SOC_DAPM_ENUM("Stereo4 DMIC Source", rt5677_stereo4_dmic_enum); -/* Stereo2 ADC source */ /* MX-26 [0] */ +/* Stereo2 ADC Source */ /* MX-26 [0] */ static const char * const rt5677_stereo2_adc_lr_src[] = { "L", "LR" }; @@ -1162,7 +1162,7 @@ static SOC_ENUM_SINGLE_DECL( RT5677_SEL_STO2_LR_MIX_SFT, rt5677_stereo2_adc_lr_src); static const struct snd_kcontrol_new rt5677_sto2_adc_lr_mux = - SOC_DAPM_ENUM("Stereo2 ADC LR source", rt5677_stereo2_adc_lr_enum); + SOC_DAPM_ENUM("Stereo2 ADC LR Source", rt5677_stereo2_adc_lr_enum); /* Stereo1 ADC Source 1 */ /* MX-27 MX26 MX25 [13:12] */ static const char * const rt5677_stereo_adc1_src[] = { @@ -1174,23 +1174,23 @@ static SOC_ENUM_SINGLE_DECL( RT5677_SEL_STO1_ADC1_SFT, rt5677_stereo_adc1_src); static const struct snd_kcontrol_new rt5677_sto1_adc1_mux = - SOC_DAPM_ENUM("Stereo1 ADC1 source", rt5677_stereo1_adc1_enum); + SOC_DAPM_ENUM("Stereo1 ADC1 Source", rt5677_stereo1_adc1_enum); static SOC_ENUM_SINGLE_DECL( rt5677_stereo2_adc1_enum, RT5677_STO2_ADC_MIXER, RT5677_SEL_STO2_ADC1_SFT, rt5677_stereo_adc1_src); static const struct snd_kcontrol_new rt5677_sto2_adc1_mux = - SOC_DAPM_ENUM("Stereo2 ADC1 source", rt5677_stereo2_adc1_enum); + SOC_DAPM_ENUM("Stereo2 ADC1 Source", rt5677_stereo2_adc1_enum); static SOC_ENUM_SINGLE_DECL( rt5677_stereo3_adc1_enum, RT5677_STO3_ADC_MIXER, RT5677_SEL_STO3_ADC1_SFT, rt5677_stereo_adc1_src); static const struct snd_kcontrol_new rt5677_sto3_adc1_mux = - SOC_DAPM_ENUM("Stereo3 ADC1 source", rt5677_stereo3_adc1_enum); + SOC_DAPM_ENUM("Stereo3 ADC1 Source", rt5677_stereo3_adc1_enum); -/* Mono ADC Left source 2 */ /* MX-28 [11:10] */ +/* Mono ADC Left Source 2 */ /* MX-28 [11:10] */ static const char * const rt5677_mono_adc2_l_src[] = { "DD MIX1L", "DMIC", "MONO DAC MIXL" }; @@ -1200,9 +1200,9 @@ static SOC_ENUM_SINGLE_DECL( RT5677_SEL_MONO_ADC_L2_SFT, rt5677_mono_adc2_l_src); static const struct snd_kcontrol_new rt5677_mono_adc2_l_mux = - SOC_DAPM_ENUM("Mono ADC2 L source", rt5677_mono_adc2_l_enum); + SOC_DAPM_ENUM("Mono ADC2 L Source", rt5677_mono_adc2_l_enum); -/* Mono ADC Left source 1 */ /* MX-28 [13:12] */ +/* Mono ADC Left Source 1 */ /* MX-28 [13:12] */ static const char * const rt5677_mono_adc1_l_src[] = { "DD MIX1L", "ADC1", "MONO DAC MIXL" }; @@ -1212,9 +1212,9 @@ static SOC_ENUM_SINGLE_DECL( RT5677_SEL_MONO_ADC_L1_SFT, rt5677_mono_adc1_l_src); static const struct snd_kcontrol_new rt5677_mono_adc1_l_mux = - SOC_DAPM_ENUM("Mono ADC1 L source", rt5677_mono_adc1_l_enum); + SOC_DAPM_ENUM("Mono ADC1 L Source", rt5677_mono_adc1_l_enum); -/* Mono ADC Right source 2 */ /* MX-28 [3:2] */ +/* Mono ADC Right Source 2 */ /* MX-28 [3:2] */ static const char * const rt5677_mono_adc2_r_src[] = { "DD MIX1R", "DMIC", "MONO DAC MIXR" }; @@ -1224,9 +1224,9 @@ static SOC_ENUM_SINGLE_DECL( RT5677_SEL_MONO_ADC_R2_SFT, rt5677_mono_adc2_r_src); static const struct snd_kcontrol_new rt5677_mono_adc2_r_mux = - SOC_DAPM_ENUM("Mono ADC2 R source", rt5677_mono_adc2_r_enum); + SOC_DAPM_ENUM("Mono ADC2 R Source", rt5677_mono_adc2_r_enum); -/* Mono ADC Right source 1 */ /* MX-28 [5:4] */ +/* Mono ADC Right Source 1 */ /* MX-28 [5:4] */ static const char * const rt5677_mono_adc1_r_src[] = { "DD MIX1R", "ADC2", "MONO DAC MIXR" }; @@ -1236,7 +1236,7 @@ static SOC_ENUM_SINGLE_DECL( RT5677_SEL_MONO_ADC_R1_SFT, rt5677_mono_adc1_r_src); static const struct snd_kcontrol_new rt5677_mono_adc1_r_mux = - SOC_DAPM_ENUM("Mono ADC1 R source", rt5677_mono_adc1_r_enum); + SOC_DAPM_ENUM("Mono ADC1 R Source", rt5677_mono_adc1_r_enum); /* Stereo4 ADC Source 2 */ /* MX-24 [11:10] */ static const char * const rt5677_stereo4_adc2_src[] = { @@ -1248,7 +1248,7 @@ static SOC_ENUM_SINGLE_DECL( RT5677_SEL_STO4_ADC2_SFT, rt5677_stereo4_adc2_src); static const struct snd_kcontrol_new rt5677_sto4_adc2_mux = - SOC_DAPM_ENUM("Stereo4 ADC2 source", rt5677_stereo4_adc2_enum); + SOC_DAPM_ENUM("Stereo4 ADC2 Source", rt5677_stereo4_adc2_enum); /* Stereo4 ADC Source 1 */ /* MX-24 [13:12] */ @@ -1261,7 +1261,7 @@ static SOC_ENUM_SINGLE_DECL( RT5677_SEL_STO4_ADC1_SFT, rt5677_stereo4_adc1_src); static const struct snd_kcontrol_new rt5677_sto4_adc1_mux = - SOC_DAPM_ENUM("Stereo4 ADC1 source", rt5677_stereo4_adc1_enum); + SOC_DAPM_ENUM("Stereo4 ADC1 Source", rt5677_stereo4_adc1_enum); /* InBound0/1 Source */ /* MX-A3 [14:12] */ static const char * const rt5677_inbound01_src[] = { @@ -1403,7 +1403,7 @@ static SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5677_dac3_mux = SOC_DAPM_ENUM("Analog DAC3 Source", rt5677_dac3_enum); -/* PDM channel source */ /* MX-31 [13:12][9:8][5:4][1:0] */ +/* PDM channel Source */ /* MX-31 [13:12][9:8][5:4][1:0] */ static const char * const rt5677_pdm_src[] = { "STO1 DAC MIX", "MONO DAC MIX", "DD MIX1", "DD MIX2" }; @@ -1413,28 +1413,28 @@ static SOC_ENUM_SINGLE_DECL( RT5677_SEL_PDM1_L_SFT, rt5677_pdm_src); static const struct snd_kcontrol_new rt5677_pdm1_l_mux = - SOC_DAPM_ENUM("PDM1 source", rt5677_pdm1_l_enum); + SOC_DAPM_ENUM("PDM1 Source", rt5677_pdm1_l_enum); static SOC_ENUM_SINGLE_DECL( rt5677_pdm2_l_enum, RT5677_PDM_OUT_CTRL, RT5677_SEL_PDM2_L_SFT, rt5677_pdm_src); static const struct snd_kcontrol_new rt5677_pdm2_l_mux = - SOC_DAPM_ENUM("PDM2 source", rt5677_pdm2_l_enum); + SOC_DAPM_ENUM("PDM2 Source", rt5677_pdm2_l_enum); static SOC_ENUM_SINGLE_DECL( rt5677_pdm1_r_enum, RT5677_PDM_OUT_CTRL, RT5677_SEL_PDM1_R_SFT, rt5677_pdm_src); static const struct snd_kcontrol_new rt5677_pdm1_r_mux = - SOC_DAPM_ENUM("PDM1 source", rt5677_pdm1_r_enum); + SOC_DAPM_ENUM("PDM1 Source", rt5677_pdm1_r_enum); static SOC_ENUM_SINGLE_DECL( rt5677_pdm2_r_enum, RT5677_PDM_OUT_CTRL, RT5677_SEL_PDM2_R_SFT, rt5677_pdm_src); static const struct snd_kcontrol_new rt5677_pdm2_r_mux = - SOC_DAPM_ENUM("PDM2 source", rt5677_pdm2_r_enum); + SOC_DAPM_ENUM("PDM2 Source", rt5677_pdm2_r_enum); /* TDM IF1/2 SLB ADC1 Data Selection */ /* MX-3C MX-41 [5:4] MX-08 [1:0]*/ static const char * const rt5677_if12_adc1_src[] = { @@ -1446,21 +1446,21 @@ static SOC_ENUM_SINGLE_DECL( RT5677_IF1_ADC1_SFT, rt5677_if12_adc1_src); static const struct snd_kcontrol_new rt5677_if1_adc1_mux = - SOC_DAPM_ENUM("IF1 ADC1 source", rt5677_if1_adc1_enum); + SOC_DAPM_ENUM("IF1 ADC1 Source", rt5677_if1_adc1_enum); static SOC_ENUM_SINGLE_DECL( rt5677_if2_adc1_enum, RT5677_TDM2_CTRL2, RT5677_IF2_ADC1_SFT, rt5677_if12_adc1_src); static const struct snd_kcontrol_new rt5677_if2_adc1_mux = - SOC_DAPM_ENUM("IF2 ADC1 source", rt5677_if2_adc1_enum); + SOC_DAPM_ENUM("IF2 ADC1 Source", rt5677_if2_adc1_enum); static SOC_ENUM_SINGLE_DECL( rt5677_slb_adc1_enum, RT5677_SLIMBUS_RX, RT5677_SLB_ADC1_SFT, rt5677_if12_adc1_src); static const struct snd_kcontrol_new rt5677_slb_adc1_mux = - SOC_DAPM_ENUM("SLB ADC1 source", rt5677_slb_adc1_enum); + SOC_DAPM_ENUM("SLB ADC1 Source", rt5677_slb_adc1_enum); /* TDM IF1/2 SLB ADC2 Data Selection */ /* MX-3C MX-41 [7:6] MX-08 [3:2] */ static const char * const rt5677_if12_adc2_src[] = { @@ -1472,21 +1472,21 @@ static SOC_ENUM_SINGLE_DECL( RT5677_IF1_ADC2_SFT, rt5677_if12_adc2_src); static const struct snd_kcontrol_new rt5677_if1_adc2_mux = - SOC_DAPM_ENUM("IF1 ADC2 source", rt5677_if1_adc2_enum); + SOC_DAPM_ENUM("IF1 ADC2 Source", rt5677_if1_adc2_enum); static SOC_ENUM_SINGLE_DECL( rt5677_if2_adc2_enum, RT5677_TDM2_CTRL2, RT5677_IF2_ADC2_SFT, rt5677_if12_adc2_src); static const struct snd_kcontrol_new rt5677_if2_adc2_mux = - SOC_DAPM_ENUM("IF2 ADC2 source", rt5677_if2_adc2_enum); + SOC_DAPM_ENUM("IF2 ADC2 Source", rt5677_if2_adc2_enum); static SOC_ENUM_SINGLE_DECL( rt5677_slb_adc2_enum, RT5677_SLIMBUS_RX, RT5677_SLB_ADC2_SFT, rt5677_if12_adc2_src); static const struct snd_kcontrol_new rt5677_slb_adc2_mux = - SOC_DAPM_ENUM("SLB ADC2 source", rt5677_slb_adc2_enum); + SOC_DAPM_ENUM("SLB ADC2 Source", rt5677_slb_adc2_enum); /* TDM IF1/2 SLB ADC3 Data Selection */ /* MX-3C MX-41 [9:8] MX-08 [5:4] */ static const char * const rt5677_if12_adc3_src[] = { @@ -1498,21 +1498,21 @@ static SOC_ENUM_SINGLE_DECL( RT5677_IF1_ADC3_SFT, rt5677_if12_adc3_src); static const struct snd_kcontrol_new rt5677_if1_adc3_mux = - SOC_DAPM_ENUM("IF1 ADC3 source", rt5677_if1_adc3_enum); + SOC_DAPM_ENUM("IF1 ADC3 Source", rt5677_if1_adc3_enum); static SOC_ENUM_SINGLE_DECL( rt5677_if2_adc3_enum, RT5677_TDM2_CTRL2, RT5677_IF2_ADC3_SFT, rt5677_if12_adc3_src); static const struct snd_kcontrol_new rt5677_if2_adc3_mux = - SOC_DAPM_ENUM("IF2 ADC3 source", rt5677_if2_adc3_enum); + SOC_DAPM_ENUM("IF2 ADC3 Source", rt5677_if2_adc3_enum); static SOC_ENUM_SINGLE_DECL( rt5677_slb_adc3_enum, RT5677_SLIMBUS_RX, RT5677_SLB_ADC3_SFT, rt5677_if12_adc3_src); static const struct snd_kcontrol_new rt5677_slb_adc3_mux = - SOC_DAPM_ENUM("SLB ADC3 source", rt5677_slb_adc3_enum); + SOC_DAPM_ENUM("SLB ADC3 Source", rt5677_slb_adc3_enum); /* TDM IF1/2 SLB ADC4 Data Selection */ /* MX-3C MX-41 [11:10] MX-08 [7:6] */ static const char * const rt5677_if12_adc4_src[] = { @@ -1524,21 +1524,21 @@ static SOC_ENUM_SINGLE_DECL( RT5677_IF1_ADC4_SFT, rt5677_if12_adc4_src); static const struct snd_kcontrol_new rt5677_if1_adc4_mux = - SOC_DAPM_ENUM("IF1 ADC4 source", rt5677_if1_adc4_enum); + SOC_DAPM_ENUM("IF1 ADC4 Source", rt5677_if1_adc4_enum); static SOC_ENUM_SINGLE_DECL( rt5677_if2_adc4_enum, RT5677_TDM2_CTRL2, RT5677_IF2_ADC4_SFT, rt5677_if12_adc4_src); static const struct snd_kcontrol_new rt5677_if2_adc4_mux = - SOC_DAPM_ENUM("IF2 ADC4 source", rt5677_if2_adc4_enum); + SOC_DAPM_ENUM("IF2 ADC4 Source", rt5677_if2_adc4_enum); static SOC_ENUM_SINGLE_DECL( rt5677_slb_adc4_enum, RT5677_SLIMBUS_RX, RT5677_SLB_ADC4_SFT, rt5677_if12_adc4_src); static const struct snd_kcontrol_new rt5677_slb_adc4_mux = - SOC_DAPM_ENUM("SLB ADC4 source", rt5677_slb_adc4_enum); + SOC_DAPM_ENUM("SLB ADC4 Source", rt5677_slb_adc4_enum); /* Interface3/4 ADC Data Input */ /* MX-2F [3:0] MX-30 [7:4]*/ static const char * const rt5677_if34_adc_src[] = { @@ -1551,14 +1551,14 @@ static SOC_ENUM_SINGLE_DECL( RT5677_IF3_ADC_IN_SFT, rt5677_if34_adc_src); static const struct snd_kcontrol_new rt5677_if3_adc_mux = - SOC_DAPM_ENUM("IF3 ADC source", rt5677_if3_adc_enum); + SOC_DAPM_ENUM("IF3 ADC Source", rt5677_if3_adc_enum); static SOC_ENUM_SINGLE_DECL( rt5677_if4_adc_enum, RT5677_IF4_DATA, RT5677_IF4_ADC_IN_SFT, rt5677_if34_adc_src); static const struct snd_kcontrol_new rt5677_if4_adc_mux = - SOC_DAPM_ENUM("IF4 ADC source", rt5677_if4_adc_enum); + SOC_DAPM_ENUM("IF4 ADC Source", rt5677_if4_adc_enum); static int rt5677_bst1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) -- cgit v1.2.3 From 80220f29d6b7f4c30aaccbe81058bdc4a83a727a Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Tue, 10 Jun 2014 14:35:25 +0800 Subject: ASoC: rt5677: Replace the string "Gain" to "Volume" The patch replaces the string "Gain" to "Volume". Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 2919f4417b3e..f81909004fdc 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -605,19 +605,19 @@ static const struct snd_kcontrol_new rt5677_snd_controls[] = { adc_vol_tlv), /* ADC Boost Volume Control */ - SOC_DOUBLE_TLV("STO1 ADC Boost Gain", RT5677_STO1_2_ADC_BST, + SOC_DOUBLE_TLV("STO1 ADC Boost Volume", RT5677_STO1_2_ADC_BST, RT5677_STO1_ADC_L_BST_SFT, RT5677_STO1_ADC_R_BST_SFT, 3, 0, adc_bst_tlv), - SOC_DOUBLE_TLV("STO2 ADC Boost Gain", RT5677_STO1_2_ADC_BST, + SOC_DOUBLE_TLV("STO2 ADC Boost Volume", RT5677_STO1_2_ADC_BST, RT5677_STO2_ADC_L_BST_SFT, RT5677_STO2_ADC_R_BST_SFT, 3, 0, adc_bst_tlv), - SOC_DOUBLE_TLV("STO3 ADC Boost Gain", RT5677_STO3_4_ADC_BST, + SOC_DOUBLE_TLV("STO3 ADC Boost Volume", RT5677_STO3_4_ADC_BST, RT5677_STO3_ADC_L_BST_SFT, RT5677_STO3_ADC_R_BST_SFT, 3, 0, adc_bst_tlv), - SOC_DOUBLE_TLV("STO4 ADC Boost Gain", RT5677_STO3_4_ADC_BST, + SOC_DOUBLE_TLV("STO4 ADC Boost Volume", RT5677_STO3_4_ADC_BST, RT5677_STO4_ADC_L_BST_SFT, RT5677_STO4_ADC_R_BST_SFT, 3, 0, adc_bst_tlv), - SOC_DOUBLE_TLV("Mono ADC Boost Gain", RT5677_ADC_BST_CTRL2, + SOC_DOUBLE_TLV("Mono ADC Boost Volume", RT5677_ADC_BST_CTRL2, RT5677_MONO_ADC_L_BST_SFT, RT5677_MONO_ADC_R_BST_SFT, 3, 0, adc_bst_tlv), }; -- cgit v1.2.3 From f58c3b915287af02d192cc8fb64ae8689107ec5f Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Tue, 10 Jun 2014 14:35:26 +0800 Subject: ASoC: rt5677: Add a PMD case to MICBIAS1 event The patch adds a PMD case to MICBIAS1 event. Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index f81909004fdc..8ecfb323b1b2 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -1657,6 +1657,13 @@ static int rt5677_set_micbias1_event(struct snd_soc_dapm_widget *w, RT5677_PWR_CLK_MB, RT5677_PWR_CLK_MB1 | RT5677_PWR_PP_MB1 | RT5677_PWR_CLK_MB); break; + + case SND_SOC_DAPM_PRE_PMD: + regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG2, + RT5677_PWR_CLK_MB1 | RT5677_PWR_PP_MB1 | + RT5677_PWR_CLK_MB, 0); + break; + default: return 0; } @@ -1673,7 +1680,8 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { /* Input Side */ /* micbias */ SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5677_PWR_ANLG2, RT5677_PWR_MB1_BIT, - 0, rt5677_set_micbias1_event, SND_SOC_DAPM_POST_PMU), + 0, rt5677_set_micbias1_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), /* Input Lines */ SND_SOC_DAPM_INPUT("DMIC L1"), -- cgit v1.2.3 From b60f363b7f226daf40025ab13972dc82e6780be3 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 10 Jun 2014 18:41:02 +0100 Subject: ASoC: wm5110: Power both channels for differential mono output On the wm5110 CODEC both the left and right channel must be powered when an output is being used as a mono output, although no audio is routed to the right output channel. This patch adds additional DAPM routes to link the right channel to the left in the case where an output is marked as mono. Audio must always be brought in on the left channel for mono operation. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 25 +++++++++++++++++++++++++ sound/soc/codecs/arizona.h | 1 + sound/soc/codecs/wm5110.c | 1 + 3 files changed, 27 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 29e198f57d4c..747c71e59c04 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -243,6 +243,31 @@ int arizona_init_spk(struct snd_soc_codec *codec) } EXPORT_SYMBOL_GPL(arizona_init_spk); +static const struct snd_soc_dapm_route arizona_mono_routes[] = { + { "OUT1R", NULL, "OUT1L" }, + { "OUT2R", NULL, "OUT2L" }, + { "OUT3R", NULL, "OUT3L" }, + { "OUT4R", NULL, "OUT4L" }, + { "OUT5R", NULL, "OUT5L" }, + { "OUT6R", NULL, "OUT6L" }, +}; + +int arizona_init_mono(struct snd_soc_codec *codec) +{ + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + int i; + + for (i = 0; i < ARIZONA_MAX_OUTPUT; ++i) { + if (arizona->pdata.out_mono[i]) + snd_soc_dapm_add_routes(&codec->dapm, + &arizona_mono_routes[i], 1); + } + + return 0; +} +EXPORT_SYMBOL_GPL(arizona_init_mono); + int arizona_init_gpio(struct snd_soc_codec *codec) { struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 05ae17f5bca3..942cfb197b6d 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -249,6 +249,7 @@ extern int arizona_set_fll(struct arizona_fll *fll, int source, extern int arizona_init_spk(struct snd_soc_codec *codec); extern int arizona_init_gpio(struct snd_soc_codec *codec); +extern int arizona_init_mono(struct snd_soc_codec *codec); extern int arizona_init_dai(struct arizona_priv *priv, int dai); diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 2e5fcb559e90..62ef54456499 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -1596,6 +1596,7 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) arizona_init_spk(codec); arizona_init_gpio(codec); + arizona_init_mono(codec); ret = snd_soc_add_codec_controls(codec, wm_adsp2_fw_controls, 8); if (ret != 0) -- cgit v1.2.3 From dfaf535665faa4b5aba4b59633f6b724a467c96e Mon Sep 17 00:00:00 2001 From: Bo Shen Date: Wed, 11 Jun 2014 18:14:40 +0800 Subject: ASoC: atmel_ssc_dai: enable fslen extension feature When SSC work as master, it will generate the frame sync signal. On old SoCs, it only supports frame sync length less or equal to 16bits, on newer SoCs, it supports frame sync length extension, which can support frame size larger than 16 bits. So, add this to make it supports playback 24/32 bits audio clips. Signed-off-by: Bo Shen Acked-by: Nicolas Ferre Signed-off-by: Mark Brown --- include/linux/atmel-ssc.h | 12 ++++++++++++ sound/soc/atmel/atmel_ssc_dai.c | 34 ++++++++++++++++++---------------- 2 files changed, 30 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/include/linux/atmel-ssc.h b/include/linux/atmel-ssc.h index e8dd40873d55..7c0f6549898b 100644 --- a/include/linux/atmel-ssc.h +++ b/include/linux/atmel-ssc.h @@ -72,6 +72,12 @@ void ssc_free(struct ssc_device *ssc); #define SSC_RFMR_DATNB_OFFSET 8 #define SSC_RFMR_FSEDGE_SIZE 1 #define SSC_RFMR_FSEDGE_OFFSET 24 +/* + * The FSLEN_EXT exist on at91sam9rl, at91sam9g10, + * at91sam9g20, and at91sam9g45 and newer SoCs + */ +#define SSC_RFMR_FSLEN_EXT_SIZE 4 +#define SSC_RFMR_FSLEN_EXT_OFFSET 28 #define SSC_RFMR_FSLEN_SIZE 4 #define SSC_RFMR_FSLEN_OFFSET 16 #define SSC_RFMR_FSOS_SIZE 4 @@ -110,6 +116,12 @@ void ssc_free(struct ssc_device *ssc); #define SSC_TFMR_FSDEN_OFFSET 23 #define SSC_TFMR_FSEDGE_SIZE 1 #define SSC_TFMR_FSEDGE_OFFSET 24 +/* + * The FSLEN_EXT exist on at91sam9rl, at91sam9g10, + * at91sam9g20, and at91sam9g45 and newer SoCs + */ +#define SSC_TFMR_FSLEN_EXT_SIZE 4 +#define SSC_TFMR_FSLEN_EXT_OFFSET 28 #define SSC_TFMR_FSLEN_SIZE 4 #define SSC_TFMR_FSLEN_OFFSET 16 #define SSC_TFMR_FSOS_SIZE 3 diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index de433cfd044c..f403f399808a 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -347,6 +347,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, u32 tfmr, rfmr, tcmr, rcmr; int start_event; int ret; + int fslen, fslen_ext; /* * Currently, there is only one set of dma params for @@ -387,18 +388,6 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - /* - * The SSC only supports up to 16-bit samples in I2S format, due - * to the size of the Frame Mode Register FSLEN field. - */ - if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S - && bits > 16) { - printk(KERN_WARNING - "atmel_ssc_dai: sample size %d " - "is too large for I2S\n", bits); - return -EINVAL; - } - /* * Compute SSC register settings. */ @@ -413,6 +402,17 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, * from the MCK divider, and the BCLK signal * is output on the SSC TK line. */ + + if (bits > 16 && !ssc->pdata->has_fslen_ext) { + dev_err(dai->dev, + "sample size %d is too large for SSC device\n", + bits); + return -EINVAL; + } + + fslen_ext = (bits - 1) / 16; + fslen = (bits - 1) % 16; + rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) | SSC_BF(RCMR_STTDLY, START_DELAY) | SSC_BF(RCMR_START, SSC_START_FALLING_RF) @@ -420,9 +420,10 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, | SSC_BF(RCMR_CKO, SSC_CKO_NONE) | SSC_BF(RCMR_CKS, SSC_CKS_DIV); - rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + rfmr = SSC_BF(RFMR_FSLEN_EXT, fslen_ext) + | SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | SSC_BF(RFMR_FSOS, SSC_FSOS_NEGATIVE) - | SSC_BF(RFMR_FSLEN, (bits - 1)) + | SSC_BF(RFMR_FSLEN, fslen) | SSC_BF(RFMR_DATNB, (channels - 1)) | SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_LOOP, 0) @@ -435,10 +436,11 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, | SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) | SSC_BF(TCMR_CKS, SSC_CKS_DIV); - tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + tfmr = SSC_BF(TFMR_FSLEN_EXT, fslen_ext) + | SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | SSC_BF(TFMR_FSDEN, 0) | SSC_BF(TFMR_FSOS, SSC_FSOS_NEGATIVE) - | SSC_BF(TFMR_FSLEN, (bits - 1)) + | SSC_BF(TFMR_FSLEN, fslen) | SSC_BF(TFMR_DATNB, (channels - 1)) | SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATDEF, 0) -- cgit v1.2.3 From 4131eceb4a5e471f5a866ee10e680f0081376e3b Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 6 Jun 2014 11:10:11 +0300 Subject: ASoC: Intel: Show Baytrail SST DSP firmware details during init DSP initialization complete message IPC_IA_FW_INIT_CMPLT is a large message carrying firmware details in mailbox. Read and show those details during init in order to be able to get that information to QA reports. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/intel/sst-baytrail-ipc.c | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/sst-baytrail-ipc.c b/sound/soc/intel/sst-baytrail-ipc.c index d207b22ea330..67673a2c0f41 100644 --- a/sound/soc/intel/sst-baytrail-ipc.c +++ b/sound/soc/intel/sst-baytrail-ipc.c @@ -122,6 +122,26 @@ struct sst_byt_tstamp { u32 channel_peak[8]; } __packed; +struct sst_byt_fw_version { + u8 build; + u8 minor; + u8 major; + u8 type; +} __packed; + +struct sst_byt_fw_build_info { + u8 date[16]; + u8 time[16]; +} __packed; + +struct sst_byt_fw_init { + struct sst_byt_fw_version fw_version; + struct sst_byt_fw_build_info build_info; + u16 result; + u8 module_id; + u8 debug_info; +} __packed; + /* driver internal IPC message structure */ struct ipc_message { struct list_head list; @@ -868,6 +888,7 @@ int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata) { struct sst_byt *byt; struct sst_fw *byt_sst_fw; + struct sst_byt_fw_init init; int err; dev_dbg(dev, "initialising Byt DSP IPC\n"); @@ -929,6 +950,15 @@ int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata) goto boot_err; } + /* show firmware information */ + sst_dsp_inbox_read(byt->dsp, &init, sizeof(init)); + dev_info(byt->dev, "FW version: %02x.%02x.%02x.%02x\n", + init.fw_version.major, init.fw_version.minor, + init.fw_version.build, init.fw_version.type); + dev_info(byt->dev, "Build type: %x\n", init.fw_version.type); + dev_info(byt->dev, "Build date: %s %s\n", + init.build_info.date, init.build_info.time); + pdata->dsp = byt; byt->fw = byt_sst_fw; -- cgit v1.2.3 From c9a8e3bd3df0e25d4ac9f6be1ba294004bb0bc9a Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 9 Jun 2014 14:39:06 +0300 Subject: ASoC: Intel: byt-rt5640: Enable headset mic bias voltage Connect "Headset Mic" to "MICBIAS1" supply widget of RT5640 in order to enable bias voltage for headset microphones. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/intel/byt-rt5640.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/intel/byt-rt5640.c b/sound/soc/intel/byt-rt5640.c index 53d160d39972..234a58de3c53 100644 --- a/sound/soc/intel/byt-rt5640.c +++ b/sound/soc/intel/byt-rt5640.c @@ -34,6 +34,7 @@ static const struct snd_soc_dapm_widget byt_rt5640_widgets[] = { }; static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = { + {"Headset Mic", NULL, "MICBIAS1"}, {"IN2P", NULL, "Headset Mic"}, {"IN2N", NULL, "Headset Mic"}, {"DMIC1", NULL, "Internal Mic"}, -- cgit v1.2.3 From 6cc0f4e63994a2b77fb6cd7c3bc1e25b7bdb9881 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Fri, 13 Jun 2014 18:03:51 +0530 Subject: ASoC: Intel: mfld_pcm: move stream handling to dai_ops This helps us to handle pcm and compress ops seperately and per dai Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-platform-pcm.c | 112 ++++++++++++++++++-------------- 1 file changed, 63 insertions(+), 49 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 7c790f51d259..0d46005752bc 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -230,19 +230,12 @@ static int sst_platform_init_stream(struct snd_pcm_substream *substream) } /* end -- helper functions */ -static int sst_platform_open(struct snd_pcm_substream *substream) +static int sst_media_open(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { + int ret_val = 0; struct snd_pcm_runtime *runtime = substream->runtime; struct sst_runtime_stream *stream; - int ret_val; - - pr_debug("sst_platform_open called\n"); - - snd_soc_set_runtime_hwparams(substream, &sst_platform_pcm_hw); - ret_val = snd_pcm_hw_constraint_integer(runtime, - SNDRV_PCM_HW_PARAM_PERIODS); - if (ret_val < 0) - return ret_val; stream = kzalloc(sizeof(*stream), GFP_KERNEL); if (!stream) @@ -251,50 +244,54 @@ static int sst_platform_open(struct snd_pcm_substream *substream) /* get the sst ops */ mutex_lock(&sst_lock); - if (!sst) { + if (!sst || + !try_module_get(sst->dev->driver->owner)) { pr_err("no device available to run\n"); - mutex_unlock(&sst_lock); - kfree(stream); - return -ENODEV; - } - if (!try_module_get(sst->dev->driver->owner)) { - mutex_unlock(&sst_lock); - kfree(stream); - return -ENODEV; + ret_val = -ENODEV; + goto out_ops; } stream->ops = sst->ops; mutex_unlock(&sst_lock); stream->stream_info.str_id = 0; - sst_set_stream_status(stream, SST_PLATFORM_INIT); + stream->stream_info.mad_substream = substream; /* allocate memory for SST API set */ runtime->private_data = stream; - return 0; + /* Make sure, that the period size is always even */ + snd_pcm_hw_constraint_step(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_PERIODS, 2); + + return snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); +out_ops: + kfree(stream); + mutex_unlock(&sst_lock); + return ret_val; } -static int sst_platform_close(struct snd_pcm_substream *substream) +static void sst_media_close(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct sst_runtime_stream *stream; int ret_val = 0, str_id; - pr_debug("sst_platform_close called\n"); stream = substream->runtime->private_data; str_id = stream->stream_info.str_id; if (str_id) ret_val = stream->ops->close(str_id); module_put(sst->dev->driver->owner); kfree(stream); - return ret_val; + return; } -static int sst_platform_pcm_prepare(struct snd_pcm_substream *substream) +static int sst_media_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct sst_runtime_stream *stream; int ret_val = 0, str_id; - pr_debug("sst_platform_pcm_prepare called\n"); stream = substream->runtime->private_data; str_id = stream->stream_info.str_id; if (stream->stream_info.str_id) { @@ -316,6 +313,41 @@ static int sst_platform_pcm_prepare(struct snd_pcm_substream *substream) return ret_val; } +static int sst_media_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + memset(substream->runtime->dma_area, 0, params_buffer_bytes(params)); + return 0; +} + +static int sst_media_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + return snd_pcm_lib_free_pages(substream); +} + +static struct snd_soc_dai_ops sst_media_dai_ops = { + .startup = sst_media_open, + .shutdown = sst_media_close, + .prepare = sst_media_prepare, + .hw_params = sst_media_hw_params, + .hw_free = sst_media_hw_free, +}; + +static int sst_platform_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime; + + if (substream->pcm->internal) + return 0; + + runtime = substream->runtime; + runtime->hw = sst_platform_pcm_hw; + return 0; +} + static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { @@ -377,32 +409,14 @@ static snd_pcm_uframes_t sst_platform_pcm_pointer pr_err("sst: error code = %d\n", ret_val); return ret_val; } - return stream->stream_info.buffer_ptr; -} - -static int sst_platform_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); - memset(substream->runtime->dma_area, 0, params_buffer_bytes(params)); - - return 0; -} - -static int sst_platform_pcm_hw_free(struct snd_pcm_substream *substream) -{ - return snd_pcm_lib_free_pages(substream); + return str_info->buffer_ptr; } static struct snd_pcm_ops sst_platform_ops = { .open = sst_platform_open, - .close = sst_platform_close, .ioctl = snd_pcm_lib_ioctl, - .prepare = sst_platform_pcm_prepare, .trigger = sst_platform_pcm_trigger, .pointer = sst_platform_pcm_pointer, - .hw_params = sst_platform_pcm_hw_params, - .hw_free = sst_platform_pcm_hw_free, }; static void sst_pcm_free(struct snd_pcm *pcm) @@ -413,15 +427,15 @@ static void sst_pcm_free(struct snd_pcm *pcm) static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; int retval = 0; - pr_debug("sst_pcm_new called\n"); - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream || - pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { + if (dai->driver->playback.channels_min || + dai->driver->capture.channels_min) { retval = snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), + snd_dma_continuous_data(GFP_DMA), SST_MIN_BUFFER, SST_MAX_BUFFER); if (retval) { pr_err("dma buffer allocationf fail\n"); -- cgit v1.2.3 From 9daa5bd34f84e43f23ce996d43da5f39348ae8fd Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Fri, 13 Jun 2014 18:03:52 +0530 Subject: ASoC: Intel: mfld-pcm rename period callback arg The argument was called mad_substream which is no longer apt as older driver is not used anymore so rename as arg Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-platform-pcm.c | 10 +++++----- sound/soc/intel/sst-mfld-platform.h | 4 ++-- 2 files changed, 7 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 0d46005752bc..4528946f5e9e 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -192,9 +192,9 @@ static int sst_platform_alloc_stream(struct snd_pcm_substream *substream) return ret_val; } -static void sst_period_elapsed(void *mad_substream) +static void sst_period_elapsed(void *arg) { - struct snd_pcm_substream *substream = mad_substream; + struct snd_pcm_substream *substream = arg; struct sst_runtime_stream *stream; int status; @@ -218,7 +218,7 @@ static int sst_platform_init_stream(struct snd_pcm_substream *substream) pr_debug("setting buffer ptr param\n"); sst_set_stream_status(stream, SST_PLATFORM_INIT); stream->stream_info.period_elapsed = sst_period_elapsed; - stream->stream_info.mad_substream = substream; + stream->stream_info.arg = substream; stream->stream_info.buffer_ptr = 0; stream->stream_info.sfreq = substream->runtime->rate; ret_val = stream->ops->device_control( @@ -255,7 +255,7 @@ static int sst_media_open(struct snd_pcm_substream *substream, stream->stream_info.str_id = 0; - stream->stream_info.mad_substream = substream; + stream->stream_info.arg = substream; /* allocate memory for SST API set */ runtime->private_data = stream; @@ -363,7 +363,7 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, pr_debug("sst: Trigger Start\n"); str_cmd = SST_SND_START; status = SST_PLATFORM_RUNNING; - stream->stream_info.mad_substream = substream; + stream->stream_info.arg = substream; break; case SNDRV_PCM_TRIGGER_STOP: pr_debug("sst: in stop\n"); diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index 6c5e7dc49e3c..6d929c7d7bbb 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -39,8 +39,8 @@ extern struct sst_device *sst; struct pcm_stream_info { int str_id; - void *mad_substream; - void (*period_elapsed) (void *mad_substream); + void *arg; + void (*period_elapsed) (void *arg); unsigned long long buffer_ptr; int sfreq; }; -- cgit v1.2.3 From 2a6358250081c24cc1e564cb82ecbfd19d8c7238 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Fri, 13 Jun 2014 18:03:53 +0530 Subject: ASoc: Intel: mfld-pcm: report pcm delay Now the DSP is capable of reporting the delay, report it to upper layers Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-platform-pcm.c | 1 + sound/soc/intel/sst-mfld-platform.h | 1 + 2 files changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 4528946f5e9e..80879e5fcb49 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -409,6 +409,7 @@ static snd_pcm_uframes_t sst_platform_pcm_pointer pr_err("sst: error code = %d\n", ret_val); return ret_val; } + substream->runtime->delay = str_info->pcm_delay; return str_info->buffer_ptr; } diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index 6d929c7d7bbb..33a0a2776238 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -42,6 +42,7 @@ struct pcm_stream_info { void *arg; void (*period_elapsed) (void *arg); unsigned long long buffer_ptr; + unsigned long long pcm_delay; int sfreq; }; -- cgit v1.2.3 From 914bc160efda323347a8f6a6e2da4c0b35aeba7b Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 19 Jun 2014 09:40:27 +0200 Subject: ASoC: tlv320aic31xx: Remove duplicate const SOC_ENUM_SINGLE_DECL() already adds the const qualifier, so there is no need to manually specify it. Fixes the following warnings from sparse: sound/soc/codecs/tlv320aic31xx.c:253:1: warning: duplicate const sound/soc/codecs/tlv320aic31xx.c:255:1: warning: duplicate const sound/soc/codecs/tlv320aic31xx.c:257:1: warning: duplicate const sound/soc/codecs/tlv320aic31xx.c:260:1: warning: duplicate const sound/soc/codecs/tlv320aic31xx.c:262:1: warning: duplicate const Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic31xx.c | 21 ++++++++++----------- 1 file changed, 10 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 23419109ecac..5128e6685762 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -249,17 +249,16 @@ static const char * const mic_select_text[] = { "Off", "FFR 10 Ohm", "FFR 20 Ohm", "FFR 40 Ohm" }; -static const -SOC_ENUM_SINGLE_DECL(mic1lp_p_enum, AIC31XX_MICPGAPI, 6, mic_select_text); -static const -SOC_ENUM_SINGLE_DECL(mic1rp_p_enum, AIC31XX_MICPGAPI, 4, mic_select_text); -static const -SOC_ENUM_SINGLE_DECL(mic1lm_p_enum, AIC31XX_MICPGAPI, 2, mic_select_text); - -static const -SOC_ENUM_SINGLE_DECL(cm_m_enum, AIC31XX_MICPGAMI, 6, mic_select_text); -static const -SOC_ENUM_SINGLE_DECL(mic1lm_m_enum, AIC31XX_MICPGAMI, 4, mic_select_text); +static SOC_ENUM_SINGLE_DECL(mic1lp_p_enum, AIC31XX_MICPGAPI, 6, + mic_select_text); +static SOC_ENUM_SINGLE_DECL(mic1rp_p_enum, AIC31XX_MICPGAPI, 4, + mic_select_text); +static SOC_ENUM_SINGLE_DECL(mic1lm_p_enum, AIC31XX_MICPGAPI, 2, + mic_select_text); + +static SOC_ENUM_SINGLE_DECL(cm_m_enum, AIC31XX_MICPGAMI, 6, mic_select_text); +static SOC_ENUM_SINGLE_DECL(mic1lm_m_enum, AIC31XX_MICPGAMI, 4, + mic_select_text); static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -6350, 50, 0); static const DECLARE_TLV_DB_SCALE(adc_fgain_tlv, 0, 10, 0); -- cgit v1.2.3 From afb7bb45bb904da3704aad47adc4615a81f515c5 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 19 Jun 2014 09:40:28 +0200 Subject: ASoC: cs42xx8: Make of match table static The cs42xx8_of_match table is not used outside of the driver, hence it can and should be made static. Fixes the following warning from sparse: sound/soc/codecs/cs42xx8.c:425:27: warning: symbol 'cs42xx8_of_match' was not declared. Should it be static? Signed-off-by: Lars-Peter Clausen Acked-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42xx8.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c index a25bc6061a30..ec53ffc4d8ce 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -422,7 +422,7 @@ const struct cs42xx8_driver_data cs42888_data = { }; EXPORT_SYMBOL_GPL(cs42888_data); -const struct of_device_id cs42xx8_of_match[] = { +static const struct of_device_id cs42xx8_of_match[] = { { .compatible = "cirrus,cs42448", .data = &cs42448_data, }, { .compatible = "cirrus,cs42888", .data = &cs42888_data, }, { /* sentinel */ } -- cgit v1.2.3 From 76a77f47123945f6e5a4dd167e54768012ef9669 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 19 Jun 2014 09:40:29 +0200 Subject: ASoC: omap-pcm: Include omap-pcm.h omap_pcm_platform_register() is declared in omap-pcm.h and defined in omap-pcm.c. To make sure that the function signature matches for both omap-pcm.c should include omap-pcm.h Fixes the following warning from sparse: sound/soc/omap/omap-pcm.c:235:5: warning: symbol 'omap_pcm_platform_register' was not declared. Should it be static? Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/omap/omap-pcm.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 8d809f8509c8..f4b05bc23e4b 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -31,6 +31,7 @@ #include #include #include +#include #ifdef CONFIG_ARCH_OMAP1 #define pcm_omap1510() cpu_is_omap1510() -- cgit v1.2.3 From cd7bcc6000165f6215d15e2e32b58a646e5de5ec Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 19 Jun 2014 09:40:30 +0200 Subject: ASoC: rcar: Fix dma direction type dmaengine_prep_slave_single() expects a enum dma_transfer_direction and not a enum dma_data_direction. Since the integer representations of both DMA_TO_DEVICE and DMA_MEM_TO_DEV aswell as DMA_FROM_DEVICE and DMA_DEV_TO_MEM have the same value the code worked fine even though it was using the wrong type. Fixes the following warning from sparse: sound/soc/sh/rcar/core.c:227:49: warning: mixing different enum types sound/soc/sh/rcar/core.c:227:49: int enum dma_data_direction versus sound/soc/sh/rcar/core.c:227:49: int enum dma_transfer_direction Signed-off-by: Lars-Peter Clausen Acked-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 2 +- sound/soc/sh/rcar/rsnd.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 91880156e1ae..7bdda8fbde87 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -363,7 +363,7 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, if (ret < 0) goto rsnd_dma_init_err; - dma->dir = is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE; + dma->dir = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; INIT_WORK(&dma->work, rsnd_dma_do_work); return 0; diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 39d98af5ee05..067a89e9f25c 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -158,7 +158,7 @@ struct rsnd_dma { struct sh_dmae_slave slave; struct work_struct work; struct dma_chan *chan; - enum dma_data_direction dir; + enum dma_transfer_direction dir; int submit_loop; int offset; /* it cares A/B plane */ -- cgit v1.2.3 From 9f98cd69c1517e05aa00ab843f054457cc3488c1 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 19 Jun 2014 09:40:31 +0200 Subject: ASoC: sh/fsi: Make one-bit bitfields unsigned One-bit signed bitfields have two possible values: 0 and -1. This sometimes leads to unexpected results (e.g. foo.bar = 1; foo.bar == 1 => false) which is why it is recommended to make one-bit bitfields unsigned. This fixes the following sparse warnings: sound/soc/sh/fsi.c:267:25: error: dubious one-bit signed bitfield sound/soc/sh/fsi.c:268:22: error: dubious one-bit signed bitfield sound/soc/sh/fsi.c:269:20: error: dubious one-bit signed bitfield sound/soc/sh/fsi.c:270:28: error: dubious one-bit signed bitfield sound/soc/sh/fsi.c:271:26: error: dubious one-bit signed bitfield sound/soc/sh/fsi.c:272:25: error: dubious one-bit signed bitfield Signed-off-by: Lars-Peter Clausen Acked-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 710a079a7377..2c95d85c69cb 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -264,12 +264,12 @@ struct fsi_priv { u32 fmt; int chan_num:16; - int clk_master:1; - int clk_cpg:1; - int spdif:1; - int enable_stream:1; - int bit_clk_inv:1; - int lr_clk_inv:1; + unsigned int clk_master:1; + unsigned int clk_cpg:1; + unsigned int spdif:1; + unsigned int enable_stream:1; + unsigned int bit_clk_inv:1; + unsigned int lr_clk_inv:1; }; struct fsi_stream_handler { -- cgit v1.2.3 From 94f99c875c109e51decf0d8c25ec2c946db20c56 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 16 Jun 2014 18:13:01 +0200 Subject: ASoC: Move name_prefix from CODEC to component Move the name_prefix from the CODEC struct to the component struct. This will eventually allow to specify prefixes for all types of components. It is also necessary to make the DAPM code component type independent (i.e. a DAPM context does not need to know whether it belongs to a CODEC or a platform or something else). Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 2 +- sound/soc/soc-core.c | 12 ++++++------ sound/soc/soc-dapm.c | 36 ++++++++++++++++++++++-------------- 3 files changed, 29 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index ed9e2d7e5fdc..e1cce0042f8a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -682,6 +682,7 @@ struct snd_soc_component_driver { struct snd_soc_component { const char *name; int id; + const char *name_prefix; struct device *dev; unsigned int active; @@ -710,7 +711,6 @@ struct snd_soc_component { /* SoC Audio Codec device */ struct snd_soc_codec { const char *name; - const char *name_prefix; int id; struct device *dev; const struct snd_soc_codec_driver *driver; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b87d7d882e6d..ba822e927187 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1108,7 +1108,7 @@ static void soc_remove_dai_links(struct snd_soc_card *card) } static void soc_set_name_prefix(struct snd_soc_card *card, - struct snd_soc_codec *codec) + struct snd_soc_component *component) { int i; @@ -1117,11 +1117,11 @@ static void soc_set_name_prefix(struct snd_soc_card *card, for (i = 0; i < card->num_configs; i++) { struct snd_soc_codec_conf *map = &card->codec_conf[i]; - if (map->of_node && codec->dev->of_node != map->of_node) + if (map->of_node && component->dev->of_node != map->of_node) continue; - if (map->dev_name && strcmp(codec->name, map->dev_name)) + if (map->dev_name && strcmp(component->name, map->dev_name)) continue; - codec->name_prefix = map->name_prefix; + component->name_prefix = map->name_prefix; break; } } @@ -1135,7 +1135,7 @@ static int soc_probe_codec(struct snd_soc_card *card, codec->card = card; codec->dapm.card = card; - soc_set_name_prefix(card, codec); + soc_set_name_prefix(card, &codec->component); if (!try_module_get(codec->dev->driver->owner)) return -ENODEV; @@ -2403,7 +2403,7 @@ int snd_soc_add_codec_controls(struct snd_soc_codec *codec, struct snd_card *card = codec->card->snd_card; return snd_soc_add_controls(card, codec->dev, controls, num_controls, - codec->name_prefix, &codec->component); + codec->component.name_prefix, &codec->component); } EXPORT_SYMBOL_GPL(snd_soc_add_codec_controls); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a74b9bf23d9f..2f29b289a333 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -375,6 +375,13 @@ static void dapm_reset(struct snd_soc_card *card) } } +static const char *soc_dapm_prefix(struct snd_soc_dapm_context *dapm) +{ + if (!dapm->component) + return NULL; + return dapm->component->name_prefix; +} + static int soc_widget_read(struct snd_soc_dapm_widget *w, int reg, unsigned int *value) { @@ -570,11 +577,7 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w, const char *name; int ret; - if (dapm->codec) - prefix = dapm->codec->name_prefix; - else - prefix = NULL; - + prefix = soc_dapm_prefix(dapm); if (prefix) prefix_len = strlen(prefix) + 1; else @@ -2371,14 +2374,16 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, const char *source; char prefixed_sink[80]; char prefixed_source[80]; + const char *prefix; int ret; - if (dapm->codec && dapm->codec->name_prefix) { + prefix = soc_dapm_prefix(dapm); + if (prefix) { snprintf(prefixed_sink, sizeof(prefixed_sink), "%s %s", - dapm->codec->name_prefix, route->sink); + prefix, route->sink); sink = prefixed_sink; snprintf(prefixed_source, sizeof(prefixed_source), "%s %s", - dapm->codec->name_prefix, route->source); + prefix, route->source); source = prefixed_source; } else { sink = route->sink; @@ -2439,6 +2444,7 @@ static int snd_soc_dapm_del_route(struct snd_soc_dapm_context *dapm, const char *source; char prefixed_sink[80]; char prefixed_source[80]; + const char *prefix; if (route->control) { dev_err(dapm->dev, @@ -2446,12 +2452,13 @@ static int snd_soc_dapm_del_route(struct snd_soc_dapm_context *dapm, return -EINVAL; } - if (dapm->codec && dapm->codec->name_prefix) { + prefix = soc_dapm_prefix(dapm); + if (prefix) { snprintf(prefixed_sink, sizeof(prefixed_sink), "%s %s", - dapm->codec->name_prefix, route->sink); + prefix, route->sink); sink = prefixed_sink; snprintf(prefixed_source, sizeof(prefixed_source), "%s %s", - dapm->codec->name_prefix, route->source); + prefix, route->source); source = prefixed_source; } else { sink = route->sink; @@ -2968,6 +2975,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_widget *widget) { struct snd_soc_dapm_widget *w; + const char *prefix; int ret; if ((w = dapm_cnew_widget(widget)) == NULL) @@ -3008,9 +3016,9 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, break; } - if (dapm->codec && dapm->codec->name_prefix) - w->name = kasprintf(GFP_KERNEL, "%s %s", - dapm->codec->name_prefix, widget->name); + prefix = soc_dapm_prefix(dapm); + if (prefix) + w->name = kasprintf(GFP_KERNEL, "%s %s", prefix, widget->name); else w->name = kasprintf(GFP_KERNEL, "%s", widget->name); -- cgit v1.2.3 From f4333203ec933f9272c90c7add01774ec2cf94d3 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 16 Jun 2014 18:13:02 +0200 Subject: ASoC: Move name and id from CODEC/platform to component The component struct already has a name and id field which are initialized to the same values as the same fields in the CODEC and platform structs. So remove them from the CODEC and platform structs and used the ones from the component struct instead. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 4 ---- include/trace/events/asoc.h | 6 ++--- sound/soc/codecs/tlv320dac33.c | 2 +- sound/soc/soc-cache.c | 7 +++--- sound/soc/soc-compress.c | 8 ++++--- sound/soc/soc-core.c | 53 ++++++++++++++++-------------------------- sound/soc/soc-dapm.c | 8 +++---- sound/soc/soc-pcm.c | 4 ++-- 8 files changed, 38 insertions(+), 54 deletions(-) (limited to 'sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index e1cce0042f8a..f64bf9452466 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -710,8 +710,6 @@ struct snd_soc_component { /* SoC Audio Codec device */ struct snd_soc_codec { - const char *name; - int id; struct device *dev; const struct snd_soc_codec_driver *driver; @@ -848,8 +846,6 @@ struct snd_soc_platform_driver { }; struct snd_soc_platform { - const char *name; - int id; struct device *dev; const struct snd_soc_platform_driver *driver; diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h index c75c795a377b..0194a641e4e2 100644 --- a/include/trace/events/asoc.h +++ b/include/trace/events/asoc.h @@ -296,17 +296,17 @@ TRACE_EVENT(snd_soc_cache_sync, TP_ARGS(codec, type, status), TP_STRUCT__entry( - __string( name, codec->name ) + __string( name, codec->component.name) __string( status, status ) __string( type, type ) __field( int, id ) ), TP_fast_assign( - __assign_str(name, codec->name); + __assign_str(name, codec->component.name); __assign_str(status, status); __assign_str(type, type); - __entry->id = codec->id; + __entry->id = codec->component.id; ), TP_printk("codec=%s.%d type=%s status=%s", __get_str(name), diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index df3a7506c023..ff006cc95520 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1404,7 +1404,7 @@ static int dac33_soc_probe(struct snd_soc_codec *codec) if (dac33->irq >= 0) { ret = request_irq(dac33->irq, dac33_interrupt_handler, IRQF_TRIGGER_RISING, - codec->name, codec); + codec->component.name, codec); if (ret < 0) { dev_err(codec->dev, "Could not request IRQ%d (%d)\n", dac33->irq, ret); diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 00e70b6c7da2..a9f82b5aba9d 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -78,7 +78,7 @@ int snd_soc_cache_init(struct snd_soc_codec *codec) mutex_init(&codec->cache_rw_mutex); dev_dbg(codec->dev, "ASoC: Initializing cache for %s codec\n", - codec->name); + codec->component.name); if (codec_drv->reg_cache_default) codec->reg_cache = kmemdup(codec_drv->reg_cache_default, @@ -98,8 +98,7 @@ int snd_soc_cache_init(struct snd_soc_codec *codec) int snd_soc_cache_exit(struct snd_soc_codec *codec) { dev_dbg(codec->dev, "ASoC: Destroying cache for %s codec\n", - codec->name); - + codec->component.name); kfree(codec->reg_cache); codec->reg_cache = NULL; return 0; @@ -192,7 +191,7 @@ int snd_soc_cache_sync(struct snd_soc_codec *codec) return 0; dev_dbg(codec->dev, "ASoC: Syncing cache for %s codec\n", - codec->name); + codec->component.name); trace_snd_soc_cache_sync(codec, name, "start"); ret = snd_soc_flat_cache_sync(codec); if (!ret) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 10f7f1da2aca..f96fb96b2678 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -37,7 +37,8 @@ static int soc_compr_open(struct snd_compr_stream *cstream) if (platform->driver->compr_ops && platform->driver->compr_ops->open) { ret = platform->driver->compr_ops->open(cstream); if (ret < 0) { - pr_err("compress asoc: can't open platform %s\n", platform->name); + pr_err("compress asoc: can't open platform %s\n", + platform->component.name); goto out; } } @@ -84,7 +85,8 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) if (platform->driver->compr_ops && platform->driver->compr_ops->open) { ret = platform->driver->compr_ops->open(cstream); if (ret < 0) { - pr_err("compress asoc: can't open platform %s\n", platform->name); + pr_err("compress asoc: can't open platform %s\n", + platform->component.name); goto out; } } @@ -680,7 +682,7 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) ret = snd_compress_new(rtd->card->snd_card, num, direction, compr); if (ret < 0) { pr_err("compress asoc: can't create compress for codec %s\n", - codec->name); + codec->component.name); goto compr_err; } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ba822e927187..b8cc88a9f947 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -274,7 +274,7 @@ static void soc_init_codec_debugfs(struct snd_soc_codec *codec) { struct dentry *debugfs_card_root = codec->card->debugfs_card_root; - codec->debugfs_codec_root = debugfs_create_dir(codec->name, + codec->debugfs_codec_root = debugfs_create_dir(codec->component.name, debugfs_card_root); if (!codec->debugfs_codec_root) { dev_warn(codec->dev, @@ -306,8 +306,8 @@ static void soc_init_platform_debugfs(struct snd_soc_platform *platform) { struct dentry *debugfs_card_root = platform->card->debugfs_card_root; - platform->debugfs_platform_root = debugfs_create_dir(platform->name, - debugfs_card_root); + platform->debugfs_platform_root = debugfs_create_dir( + platform->component.name, debugfs_card_root); if (!platform->debugfs_platform_root) { dev_warn(platform->dev, "ASoC: Failed to create platform debugfs directory\n"); @@ -335,7 +335,7 @@ static ssize_t codec_list_read_file(struct file *file, char __user *user_buf, list_for_each_entry(codec, &codec_list, list) { len = snprintf(buf + ret, PAGE_SIZE - ret, "%s\n", - codec->name); + codec->component.name); if (len >= 0) ret += len; if (ret > PAGE_SIZE) { @@ -406,7 +406,7 @@ static ssize_t platform_list_read_file(struct file *file, list_for_each_entry(platform, &platform_list, list) { len = snprintf(buf + ret, PAGE_SIZE - ret, "%s\n", - platform->name); + platform->component.name); if (len >= 0) ret += len; if (ret > PAGE_SIZE) { @@ -528,7 +528,7 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec) codec->ac97->dev.release = soc_ac97_device_release; dev_set_name(&codec->ac97->dev, "%d-%d:%s", - codec->card->snd_card->number, 0, codec->name); + codec->card->snd_card->number, 0, codec->component.name); err = device_register(&codec->ac97->dev); if (err < 0) { dev_err(codec->dev, "ASoC: Can't register ac97 bus\n"); @@ -857,7 +857,7 @@ static struct snd_soc_codec *soc_find_codec(const struct device_node *codec_of_n if (codec->dev->of_node != codec_of_node) continue; } else { - if (strcmp(codec->name, codec_name)) + if (strcmp(codec->component.name, codec_name)) continue; } @@ -945,7 +945,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) dai_link->platform_of_node) continue; } else { - if (strcmp(platform->name, platform_name)) + if (strcmp(platform->component.name, platform_name)) continue; } @@ -1177,7 +1177,7 @@ static int soc_probe_codec(struct snd_soc_card *card, WARN(codec->dapm.idle_bias_off && codec->dapm.bias_level != SND_SOC_BIAS_OFF, "codec %s can not start from non-off bias with idle_bias_off==1\n", - codec->name); + codec->component.name); } if (driver->controls) @@ -1647,7 +1647,8 @@ static struct snd_soc_codec *soc_find_matching_codec(struct snd_soc_card *card, if (aux_dev->codec_of_node && (codec->dev->of_node != aux_dev->codec_of_node)) continue; - if (aux_dev->codec_name && strcmp(codec->name, aux_dev->codec_name)) + if (aux_dev->codec_name && + strcmp(codec->component.name, aux_dev->codec_name)) continue; return codec; } @@ -4131,11 +4132,6 @@ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, { int ret; - /* create platform component name */ - platform->name = fmt_single_name(dev, &platform->id); - if (platform->name == NULL) - return -ENOMEM; - platform->dev = dev; platform->driver = platform_drv; platform->dapm.dev = dev; @@ -4161,7 +4157,8 @@ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, list_add(&platform->list, &platform_list); mutex_unlock(&client_mutex); - dev_dbg(dev, "ASoC: Registered platform '%s'\n", platform->name); + dev_dbg(dev, "ASoC: Registered platform '%s'\n", + platform->component.name); return 0; } @@ -4205,8 +4202,7 @@ void snd_soc_remove_platform(struct snd_soc_platform *platform) mutex_unlock(&client_mutex); dev_dbg(platform->dev, "ASoC: Unregistered platform '%s'\n", - platform->name); - kfree(platform->name); + platform->component.name); } EXPORT_SYMBOL_GPL(snd_soc_remove_platform); @@ -4312,13 +4308,6 @@ int snd_soc_register_codec(struct device *dev, if (codec == NULL) return -ENOMEM; - /* create CODEC component name */ - codec->name = fmt_single_name(dev, &codec->id); - if (codec->name == NULL) { - ret = -ENOMEM; - goto fail_codec; - } - if (codec_drv->write) codec->component.write = snd_soc_codec_drv_write; if (codec_drv->read) @@ -4368,19 +4357,17 @@ int snd_soc_register_codec(struct device *dev, codec, dai_drv, num_dai, false); if (ret < 0) { dev_err(codec->dev, "ASoC: Failed to regster component: %d\n", ret); - goto fail_codec_name; + goto fail_codec; } - dev_dbg(codec->dev, "ASoC: Registered codec '%s'\n", codec->name); + dev_dbg(codec->dev, "ASoC: Registered codec '%s'\n", + codec->component.name); return 0; -fail_codec_name: +fail_codec: mutex_lock(&client_mutex); list_del(&codec->list); mutex_unlock(&client_mutex); - - kfree(codec->name); -fail_codec: kfree(codec); return ret; } @@ -4408,10 +4395,10 @@ found: list_del(&codec->list); mutex_unlock(&client_mutex); - dev_dbg(codec->dev, "ASoC: Unregistered codec '%s'\n", codec->name); + dev_dbg(codec->dev, "ASoC: Unregistered codec '%s'\n", + codec->component.name); snd_soc_cache_exit(codec); - kfree(codec->name); kfree(codec); } EXPORT_SYMBOL_GPL(snd_soc_unregister_codec); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 2f29b289a333..074e9ce136f8 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3400,8 +3400,8 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card) source = cpu_dai->playback_widget; sink = codec_dai->playback_widget; dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n", - cpu_dai->codec->name, source->name, - codec_dai->platform->name, sink->name); + cpu_dai->component->name, source->name, + codec_dai->component->name, sink->name); snd_soc_dapm_add_path(&card->dapm, source, sink, NULL, NULL); @@ -3412,8 +3412,8 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card) source = codec_dai->capture_widget; sink = cpu_dai->capture_widget; dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n", - codec_dai->codec->name, source->name, - cpu_dai->platform->name, sink->name); + codec_dai->component->name, source->name, + cpu_dai->component->name, sink->name); snd_soc_dapm_add_path(&card->dapm, source, sink, NULL, NULL); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 54d18f22a33e..9b78bb619bec 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -376,7 +376,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) ret = platform->driver->ops->open(substream); if (ret < 0) { dev_err(platform->dev, "ASoC: can't open platform" - " %s: %d\n", platform->name, ret); + " %s: %d\n", platform->component.name, ret); goto platform_err; } } @@ -707,7 +707,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, ret = platform->driver->ops->hw_params(substream, params); if (ret < 0) { dev_err(platform->dev, "ASoC: %s hw params failed: %d\n", - platform->name, ret); + platform->component.name, ret); goto platform_err; } } -- cgit v1.2.3 From bb13109d85ba5f8009f1a26d840e70198537a4e3 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 16 Jun 2014 18:13:03 +0200 Subject: ASoC: Split component registration into two steps Split snd_soc_component_register() into snd_soc_component_initialize() and snd_soc_component_add(). Using a 2-stage registration approach has the advantage that it is possible to modify the component after it has been initialized, but before it is made visible to the system. This e.g. allows CODECs or platforms to overwrite some of the default settings made in snd_soc_component_initialize(). Similar snd_soc_component_unregister() is split into two steps as well, snd_soc_component_delete(), which removes the component from the system, and snd_soc_component_cleanup(), which frees all the resources allocated by the component. Furthermore this patch makes sure that if a component is visible on two list (e.g. the component list and the CODEC list) it is added or removed to both lists atomically. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 177 ++++++++++++++++++++++++++------------------------- 1 file changed, 91 insertions(+), 86 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b8cc88a9f947..0944da66538d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3922,16 +3922,14 @@ static void snd_soc_unregister_dais(struct snd_soc_component *component) * snd_soc_register_dais - Register a DAI with the ASoC core * * @component: The component the DAIs are registered for - * @codec: The CODEC that the DAIs are registered for, NULL if the component is - * not a CODEC. * @dai_drv: DAI driver to use for the DAIs * @count: Number of DAIs * @legacy_dai_naming: Use the legacy naming scheme and let the DAI inherit the * parent's name. */ static int snd_soc_register_dais(struct snd_soc_component *component, - struct snd_soc_codec *codec, struct snd_soc_dai_driver *dai_drv, - size_t count, bool legacy_dai_naming) + struct snd_soc_dai_driver *dai_drv, size_t count, + bool legacy_dai_naming) { struct device *dev = component->dev; struct snd_soc_dai *dai; @@ -3940,6 +3938,9 @@ static int snd_soc_register_dais(struct snd_soc_component *component, dev_dbg(dev, "ASoC: dai register %s #%Zu\n", dev_name(dev), count); + component->dai_drv = dai_drv; + component->num_dai = count; + for (i = 0; i < count; i++) { dai = kzalloc(sizeof(struct snd_soc_dai), GFP_KERNEL); @@ -3972,7 +3973,6 @@ static int snd_soc_register_dais(struct snd_soc_component *component, } dai->component = component; - dai->codec = codec; dai->dev = dev; dai->driver = &dai_drv[i]; dai->dapm.dev = dev; @@ -3995,60 +3995,52 @@ err: return ret; } -/** - * snd_soc_register_component - Register a component with the ASoC core - * - */ -static int -__snd_soc_register_component(struct device *dev, - struct snd_soc_component *cmpnt, - const struct snd_soc_component_driver *cmpnt_drv, - struct snd_soc_codec *codec, - struct snd_soc_dai_driver *dai_drv, - int num_dai, bool allow_single_dai) +static int snd_soc_component_initialize(struct snd_soc_component *component, + const struct snd_soc_component_driver *driver, struct device *dev) { - int ret; - - dev_dbg(dev, "component register %s\n", dev_name(dev)); - - if (!cmpnt) { - dev_err(dev, "ASoC: Failed to connecting component\n"); + component->name = fmt_single_name(dev, &component->id); + if (!component->name) { + dev_err(dev, "ASoC: Failed to allocate name\n"); return -ENOMEM; } - mutex_init(&cmpnt->io_mutex); + component->dev = dev; + component->driver = driver; - cmpnt->name = fmt_single_name(dev, &cmpnt->id); - if (!cmpnt->name) { - dev_err(dev, "ASoC: Failed to simplifying name\n"); - return -ENOMEM; - } + INIT_LIST_HEAD(&component->dai_list); + mutex_init(&component->io_mutex); - cmpnt->dev = dev; - cmpnt->driver = cmpnt_drv; - cmpnt->dai_drv = dai_drv; - cmpnt->num_dai = num_dai; - INIT_LIST_HEAD(&cmpnt->dai_list); + return 0; +} - ret = snd_soc_register_dais(cmpnt, codec, dai_drv, num_dai, - allow_single_dai); - if (ret < 0) { - dev_err(dev, "ASoC: Failed to regster DAIs: %d\n", ret); - goto error_component_name; - } +static void snd_soc_component_add_unlocked(struct snd_soc_component *component) +{ + list_add(&component->list, &component_list); +} +static void snd_soc_component_add(struct snd_soc_component *component) +{ mutex_lock(&client_mutex); - list_add(&cmpnt->list, &component_list); + snd_soc_component_add_unlocked(component); mutex_unlock(&client_mutex); +} - dev_dbg(cmpnt->dev, "ASoC: Registered component '%s'\n", cmpnt->name); - - return ret; +static void snd_soc_component_cleanup(struct snd_soc_component *component) +{ + snd_soc_unregister_dais(component); + kfree(component->name); +} -error_component_name: - kfree(cmpnt->name); +static void snd_soc_component_del_unlocked(struct snd_soc_component *component) +{ + list_del(&component->list); +} - return ret; +static void snd_soc_component_del(struct snd_soc_component *component) +{ + mutex_lock(&client_mutex); + snd_soc_component_del_unlocked(component); + mutex_unlock(&client_mutex); } int snd_soc_register_component(struct device *dev, @@ -4057,32 +4049,38 @@ int snd_soc_register_component(struct device *dev, int num_dai) { struct snd_soc_component *cmpnt; + int ret; - cmpnt = devm_kzalloc(dev, sizeof(*cmpnt), GFP_KERNEL); + cmpnt = kzalloc(sizeof(*cmpnt), GFP_KERNEL); if (!cmpnt) { dev_err(dev, "ASoC: Failed to allocate memory\n"); return -ENOMEM; } + ret = snd_soc_component_initialize(cmpnt, cmpnt_drv, dev); + if (ret) + goto err_free; + cmpnt->ignore_pmdown_time = true; cmpnt->registered_as_component = true; - return __snd_soc_register_component(dev, cmpnt, cmpnt_drv, NULL, - dai_drv, num_dai, true); -} -EXPORT_SYMBOL_GPL(snd_soc_register_component); + ret = snd_soc_register_dais(cmpnt, dai_drv, num_dai, true); + if (ret < 0) { + dev_err(dev, "ASoC: Failed to regster DAIs: %d\n", ret); + goto err_cleanup; + } -static void __snd_soc_unregister_component(struct snd_soc_component *cmpnt) -{ - snd_soc_unregister_dais(cmpnt); + snd_soc_component_add(cmpnt); - mutex_lock(&client_mutex); - list_del(&cmpnt->list); - mutex_unlock(&client_mutex); + return 0; - dev_dbg(cmpnt->dev, "ASoC: Unregistered component '%s'\n", cmpnt->name); - kfree(cmpnt->name); +err_cleanup: + snd_soc_component_cleanup(cmpnt); +err_free: + kfree(cmpnt); + return ret; } +EXPORT_SYMBOL_GPL(snd_soc_register_component); /** * snd_soc_unregister_component - Unregister a component from the ASoC core @@ -4099,7 +4097,9 @@ void snd_soc_unregister_component(struct device *dev) return; found: - __snd_soc_unregister_component(cmpnt); + snd_soc_component_del(cmpnt); + snd_soc_component_cleanup(cmpnt); + kfree(cmpnt); } EXPORT_SYMBOL_GPL(snd_soc_unregister_component); @@ -4132,6 +4132,11 @@ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, { int ret; + ret = snd_soc_component_initialize(&platform->component, + &platform_drv->component_driver, dev); + if (ret) + return ret; + platform->dev = dev; platform->driver = platform_drv; platform->dapm.dev = dev; @@ -4143,17 +4148,8 @@ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, if (platform_drv->read) platform->component.read = snd_soc_platform_drv_read; - /* register component */ - ret = __snd_soc_register_component(dev, &platform->component, - &platform_drv->component_driver, - NULL, NULL, 0, false); - if (ret < 0) { - dev_err(platform->component.dev, - "ASoC: Failed to register component: %d\n", ret); - return ret; - } - mutex_lock(&client_mutex); + snd_soc_component_add_unlocked(&platform->component); list_add(&platform->list, &platform_list); mutex_unlock(&client_mutex); @@ -4195,12 +4191,14 @@ EXPORT_SYMBOL_GPL(snd_soc_register_platform); */ void snd_soc_remove_platform(struct snd_soc_platform *platform) { - __snd_soc_unregister_component(&platform->component); mutex_lock(&client_mutex); list_del(&platform->list); + snd_soc_component_del_unlocked(&platform->component); mutex_unlock(&client_mutex); + snd_soc_component_cleanup(&platform->component); + dev_dbg(platform->dev, "ASoC: Unregistered platform '%s'\n", platform->component.name); } @@ -4299,6 +4297,7 @@ int snd_soc_register_codec(struct device *dev, int num_dai) { struct snd_soc_codec *codec; + struct snd_soc_dai *dai; struct regmap *regmap; int ret, i; @@ -4308,6 +4307,11 @@ int snd_soc_register_codec(struct device *dev, if (codec == NULL) return -ENOMEM; + ret = snd_soc_component_initialize(&codec->component, + &codec_drv->component_driver, dev); + if (ret) + goto err_free; + if (codec_drv->write) codec->component.write = snd_soc_codec_drv_write; if (codec_drv->read) @@ -4337,7 +4341,7 @@ int snd_soc_register_codec(struct device *dev, dev_err(codec->dev, "Failed to set cache I/O:%d\n", ret); - return ret; + goto err_cleanup; } } } @@ -4347,27 +4351,27 @@ int snd_soc_register_codec(struct device *dev, fixup_codec_formats(&dai_drv[i].capture); } + ret = snd_soc_register_dais(&codec->component, dai_drv, num_dai, false); + if (ret < 0) { + dev_err(dev, "ASoC: Failed to regster DAIs: %d\n", ret); + goto err_cleanup; + } + + list_for_each_entry(dai, &codec->component.dai_list, list) + dai->codec = codec; + mutex_lock(&client_mutex); + snd_soc_component_add_unlocked(&codec->component); list_add(&codec->list, &codec_list); mutex_unlock(&client_mutex); - /* register component */ - ret = __snd_soc_register_component(dev, &codec->component, - &codec_drv->component_driver, - codec, dai_drv, num_dai, false); - if (ret < 0) { - dev_err(codec->dev, "ASoC: Failed to regster component: %d\n", ret); - goto fail_codec; - } - dev_dbg(codec->dev, "ASoC: Registered codec '%s'\n", codec->component.name); return 0; -fail_codec: - mutex_lock(&client_mutex); - list_del(&codec->list); - mutex_unlock(&client_mutex); +err_cleanup: + snd_soc_component_cleanup(&codec->component); +err_free: kfree(codec); return ret; } @@ -4389,15 +4393,16 @@ void snd_soc_unregister_codec(struct device *dev) return; found: - __snd_soc_unregister_component(&codec->component); mutex_lock(&client_mutex); list_del(&codec->list); + snd_soc_component_del_unlocked(&codec->component); mutex_unlock(&client_mutex); dev_dbg(codec->dev, "ASoC: Unregistered codec '%s'\n", codec->component.name); + snd_soc_component_cleanup(&codec->component); snd_soc_cache_exit(codec); kfree(codec); } -- cgit v1.2.3 From 7df3788410e674423375b88e2d95c46e4015f5f5 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 16 Jun 2014 18:13:04 +0200 Subject: ASoC: Auto disconnect pins from all DAPM contexts Currently only pins in CODEC DAPM contexts are automatically marked as non-connected if the card has the fully_routed flag set. This makes sense since widgets which qualify for auto-disconnection are only found in CODEC DAPM contexts. But with componentisation this is going to change, so consider all widgets for auto-disconnection. Also it is probably faster to walk the widgets list only once rather than once for each CODEC. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 2 +- sound/soc/soc-core.c | 3 +-- sound/soc/soc-dapm.c | 27 +++++++++++---------------- 3 files changed, 13 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 6b59471cdf44..8db627cc2fbe 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -431,7 +431,7 @@ int snd_soc_dapm_force_enable_pin_unlocked(struct snd_soc_dapm_context *dapm, const char *pin); int snd_soc_dapm_ignore_suspend(struct snd_soc_dapm_context *dapm, const char *pin); -void snd_soc_dapm_auto_nc_codec_pins(struct snd_soc_codec *codec); +void snd_soc_dapm_auto_nc_pins(struct snd_soc_card *card); /* Mostly internal - should not normally be used */ void dapm_mark_io_dirty(struct snd_soc_dapm_context *dapm); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0944da66538d..bca8a7150a4a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1928,8 +1928,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) } if (card->fully_routed) - list_for_each_entry(codec, &card->codec_dev_list, card_list) - snd_soc_dapm_auto_nc_codec_pins(codec); + snd_soc_dapm_auto_nc_pins(card); snd_soc_dapm_new_widgets(card); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 074e9ce136f8..3ccbf9ba04b2 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3763,36 +3763,31 @@ static bool snd_soc_dapm_widget_in_card_paths(struct snd_soc_card *card, } /** - * snd_soc_dapm_auto_nc_codec_pins - call snd_soc_dapm_nc_pin for unused pins - * @codec: The codec whose pins should be processed + * snd_soc_dapm_auto_nc_pins - call snd_soc_dapm_nc_pin for unused pins + * @card: The card whose pins should be processed * - * Automatically call snd_soc_dapm_nc_pin() for any external pins in the codec - * which are unused. Pins are used if they are connected externally to the - * codec, whether that be to some other device, or a loop-back connection to - * the codec itself. + * Automatically call snd_soc_dapm_nc_pin() for any external pins in the card + * which are unused. Pins are used if they are connected externally to a + * component, whether that be to some other device, or a loop-back connection to + * the component itself. */ -void snd_soc_dapm_auto_nc_codec_pins(struct snd_soc_codec *codec) +void snd_soc_dapm_auto_nc_pins(struct snd_soc_card *card) { - struct snd_soc_card *card = codec->card; - struct snd_soc_dapm_context *dapm = &codec->dapm; struct snd_soc_dapm_widget *w; - dev_dbg(codec->dev, "ASoC: Auto NC: DAPMs: card:%p codec:%p\n", - &card->dapm, &codec->dapm); + dev_dbg(card->dev, "ASoC: Auto NC: DAPMs: card:%p\n", &card->dapm); list_for_each_entry(w, &card->widgets, list) { - if (w->dapm != dapm) - continue; switch (w->id) { case snd_soc_dapm_input: case snd_soc_dapm_output: case snd_soc_dapm_micbias: - dev_dbg(codec->dev, "ASoC: Auto NC: Checking widget %s\n", + dev_dbg(card->dev, "ASoC: Auto NC: Checking widget %s\n", w->name); if (!snd_soc_dapm_widget_in_card_paths(card, w)) { - dev_dbg(codec->dev, + dev_dbg(card->dev, "... Not in map; disabling\n"); - snd_soc_dapm_nc_pin(dapm, w->name); + snd_soc_dapm_nc_pin(w->dapm, w->name); } break; default: -- cgit v1.2.3 From 68f831c2724ab72c0088471b2ed1dc99e81948ef Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 16 Jun 2014 18:13:05 +0200 Subject: ASoC: Add a set_bias_level() callback to the DAPM context struct Currently the DAPM code directly looks at the CODEC driver struct to get a handle to the set_bias_level() callback. This patch adds a new set_bias_level() callback to the DAPM context struct. The DAPM code will use this new callback instead of the CODEC callback. For CODECs the new callback is set up to call the CODEC specific set_bias_level callback(). Not looking directly at the CODEC driver struct will allow non CODEC DAPM contexts to implement a set_bias_level() callback. This is also similar to how the seq_notifier() and stream_event() callbacks are currently handled. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 2 ++ sound/soc/soc-core.c | 10 ++++++++++ sound/soc/soc-dapm.c | 11 +++-------- 3 files changed, 15 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 8db627cc2fbe..3a5c4f969c04 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -601,6 +601,8 @@ struct snd_soc_dapm_context { struct list_head list; int (*stream_event)(struct snd_soc_dapm_context *dapm, int event); + int (*set_bias_level)(struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level); #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_dapm; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index bca8a7150a4a..10e13c43bc54 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4285,6 +4285,14 @@ static int snd_soc_codec_drv_read(struct snd_soc_component *component, return 0; } +static int snd_soc_codec_set_bias_level(struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); + + return codec->driver->set_bias_level(codec, level); +} + /** * snd_soc_register_codec - Register a codec with the ASoC core * @@ -4322,6 +4330,8 @@ int snd_soc_register_codec(struct device *dev, codec->dapm.component = &codec->component; codec->dapm.seq_notifier = codec_drv->seq_notifier; codec->dapm.stream_event = codec_drv->stream_event; + if (codec_drv->set_bias_level) + codec->dapm.set_bias_level = snd_soc_codec_set_bias_level; codec->dev = dev; codec->driver = codec_drv; codec->component.val_bytes = codec_drv->reg_word_size; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index fab1a8813abf..6c94a6b3fce7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -427,15 +427,10 @@ static int snd_soc_dapm_set_bias_level(struct snd_soc_dapm_context *dapm, if (ret != 0) goto out; - if (dapm->codec) { - if (dapm->codec->driver->set_bias_level) - ret = dapm->codec->driver->set_bias_level(dapm->codec, - level); - else - dapm->bias_level = level; - } else if (!card || dapm != &card->dapm) { + if (dapm->set_bias_level) + ret = dapm->set_bias_level(dapm, level); + else if (!card || dapm != &card->dapm) dapm->bias_level = level; - } if (ret != 0) goto out; -- cgit v1.2.3 From ce0fc93ae56e2ba50ff8c220d69e4e860e889320 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 16 Jun 2014 18:13:06 +0200 Subject: ASoC: Add DAPM support at the component level This patch adds full DAPM support at the component level. Previously there was only full DAPM support for CODECs and partial DAPM support (e.g. no Mixers nor MUXs) for platforms. Having DAPM support at the component level will allow all types of components to use DAPM and also help in consolidating the DAPM support between CODECs and platforms. Since the DAPM context is directly embedded into the snd_soc_codec and snd_soc_platform struct and the 'dapm' field is directly referenced in a lot of drivers moving the field just right now is not possible without causing code churn. The approach this patch takes is to add two new fields to the component struct. One field which is the pointer to the actual DAPM context used by the component and one DAPM context that will be used as the default if no other context was specified. For CODECs and platforms the pointer is initialized to point to the CODEC or platform DAPM context. All generic code when referencing a component's DAPM struct will go via the pointer. This will make it possible to eventually seamlessly move the DAPM context from snd_soc_codec and snd_soc_platform struct over once all direct references have been eliminated. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 2 + include/sound/soc.h | 30 ++++++++++++++ sound/soc/soc-core.c | 19 ++++++--- sound/soc/soc-dapm.c | 100 ++++++++++++++++++++++++++++++----------------- 4 files changed, 110 insertions(+), 41 deletions(-) (limited to 'sound') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 3a5c4f969c04..e292683ee694 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -441,6 +441,8 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, struct snd_soc_dapm_widget_list **list); struct snd_soc_codec *snd_soc_dapm_kcontrol_codec(struct snd_kcontrol *kcontrol); +struct snd_soc_dapm_context *snd_soc_dapm_kcontrol_dapm( + struct snd_kcontrol *kcontrol); /* dapm widget types */ enum snd_soc_dapm_type { diff --git a/include/sound/soc.h b/include/sound/soc.h index f64bf9452466..a21dfecba56b 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -706,6 +706,10 @@ struct snd_soc_component { int val_bytes; struct mutex io_mutex; + + /* Don't use these, use snd_soc_component_get_dapm() */ + struct snd_soc_dapm_context dapm; + struct snd_soc_dapm_context *dapm_ptr; }; /* SoC Audio Codec device */ @@ -1160,6 +1164,21 @@ static inline struct snd_soc_platform *snd_soc_component_to_platform( return container_of(component, struct snd_soc_platform, component); } +/** + * snd_soc_dapm_to_component() - Casts a DAPM context to the component it is + * embedded in + * @dapm: The DAPM context to cast to the component + * + * This function must only be used on DAPM contexts that are known to be part of + * a component (e.g. in a component driver). Otherwise the behavior is + * undefined. + */ +static inline struct snd_soc_component *snd_soc_dapm_to_component( + struct snd_soc_dapm_context *dapm) +{ + return container_of(dapm, struct snd_soc_component, dapm); +} + /** * snd_soc_dapm_to_codec() - Casts a DAPM context to the CODEC it is embedded in * @dapm: The DAPM context to cast to the CODEC @@ -1187,6 +1206,17 @@ static inline struct snd_soc_platform *snd_soc_dapm_to_platform( return container_of(dapm, struct snd_soc_platform, dapm); } +/** + * snd_soc_component_get_dapm() - Returns the DAPM context associated with a + * component + * @component: The component for which to get the DAPM context + */ +static inline struct snd_soc_dapm_context *snd_soc_component_get_dapm( + struct snd_soc_component *component) +{ + return component->dapm_ptr; +} + /* codec IO */ unsigned int snd_soc_read(struct snd_soc_codec *codec, unsigned int reg); int snd_soc_write(struct snd_soc_codec *codec, unsigned int reg, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 10e13c43bc54..f519a9f7571c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3997,6 +3997,8 @@ err: static int snd_soc_component_initialize(struct snd_soc_component *component, const struct snd_soc_component_driver *driver, struct device *dev) { + struct snd_soc_dapm_context *dapm; + component->name = fmt_single_name(dev, &component->id); if (!component->name) { dev_err(dev, "ASoC: Failed to allocate name\n"); @@ -4006,6 +4008,14 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, component->dev = dev; component->driver = driver; + if (!component->dapm_ptr) + component->dapm_ptr = &component->dapm; + + dapm = component->dapm_ptr; + dapm->dev = dev; + dapm->component = component; + dapm->bias_level = SND_SOC_BIAS_OFF; + INIT_LIST_HEAD(&component->dai_list); mutex_init(&component->io_mutex); @@ -4131,6 +4141,8 @@ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, { int ret; + platform->component.dapm_ptr = &platform->dapm; + ret = snd_soc_component_initialize(&platform->component, &platform_drv->component_driver, dev); if (ret) @@ -4138,9 +4150,7 @@ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, platform->dev = dev; platform->driver = platform_drv; - platform->dapm.dev = dev; platform->dapm.platform = platform; - platform->dapm.component = &platform->component; platform->dapm.stream_event = platform_drv->stream_event; if (platform_drv->write) platform->component.write = snd_soc_platform_drv_write; @@ -4314,6 +4324,8 @@ int snd_soc_register_codec(struct device *dev, if (codec == NULL) return -ENOMEM; + codec->component.dapm_ptr = &codec->dapm; + ret = snd_soc_component_initialize(&codec->component, &codec_drv->component_driver, dev); if (ret) @@ -4324,10 +4336,7 @@ int snd_soc_register_codec(struct device *dev, if (codec_drv->read) codec->component.read = snd_soc_codec_drv_read; codec->component.ignore_pmdown_time = codec_drv->ignore_pmdown_time; - codec->dapm.bias_level = SND_SOC_BIAS_OFF; - codec->dapm.dev = dev; codec->dapm.codec = codec; - codec->dapm.component = &codec->component; codec->dapm.seq_notifier = codec_drv->seq_notifier; codec->dapm.stream_event = codec_drv->stream_event; if (codec_drv->set_bias_level) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 6c94a6b3fce7..4702b926a6a0 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -349,13 +349,28 @@ static bool dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol, return true; } +/** + * snd_soc_dapm_kcontrol_dapm() - Returns the dapm context associated to a + * kcontrol + * @kcontrol: The kcontrol + * + * Note: This function must only be used on kcontrols that are known to have + * been registered for a CODEC. Otherwise the behaviour is undefined. + */ +struct snd_soc_dapm_context *snd_soc_dapm_kcontrol_dapm( + struct snd_kcontrol *kcontrol) +{ + return dapm_kcontrol_get_wlist(kcontrol)->widgets[0]->dapm; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_kcontrol_dapm); + /** * snd_soc_dapm_kcontrol_codec() - Returns the codec associated to a kcontrol * @kcontrol: The kcontrol */ struct snd_soc_codec *snd_soc_dapm_kcontrol_codec(struct snd_kcontrol *kcontrol) { - return dapm_kcontrol_get_wlist(kcontrol)->widgets[0]->codec; + return snd_soc_dapm_to_codec(snd_soc_dapm_kcontrol_dapm(kcontrol)); } EXPORT_SYMBOL_GPL(snd_soc_dapm_kcontrol_codec); @@ -382,23 +397,31 @@ static const char *soc_dapm_prefix(struct snd_soc_dapm_context *dapm) return dapm->component->name_prefix; } -static int soc_widget_read(struct snd_soc_dapm_widget *w, int reg, +static int soc_dapm_read(struct snd_soc_dapm_context *dapm, int reg, unsigned int *value) { - if (!w->dapm->component) + if (!dapm->component) return -EIO; - return snd_soc_component_read(w->dapm->component, reg, value); + return snd_soc_component_read(dapm->component, reg, value); } -static int soc_widget_update_bits(struct snd_soc_dapm_widget *w, +static int soc_dapm_update_bits(struct snd_soc_dapm_context *dapm, int reg, unsigned int mask, unsigned int value) { - if (!w->dapm->component) + if (!dapm->component) return -EIO; - return snd_soc_component_update_bits_async(w->dapm->component, reg, + return snd_soc_component_update_bits_async(dapm->component, reg, mask, value); } +static int soc_dapm_test_bits(struct snd_soc_dapm_context *dapm, + int reg, unsigned int mask, unsigned int value) +{ + if (!dapm->component) + return -EIO; + return snd_soc_component_test_bits(dapm->component, reg, mask, value); +} + static void soc_dapm_async_complete(struct snd_soc_dapm_context *dapm) { if (dapm->component) @@ -454,7 +477,7 @@ static int dapm_connect_mux(struct snd_soc_dapm_context *dapm, int i; if (e->reg != SND_SOC_NOPM) { - soc_widget_read(dest, e->reg, &val); + soc_dapm_read(dapm, e->reg, &val); val = (val >> e->shift_l) & e->mask; item = snd_soc_enum_val_to_item(e, val); } else { @@ -498,7 +521,7 @@ static void dapm_set_mixer_path_status(struct snd_soc_dapm_widget *w, unsigned int val; if (reg != SND_SOC_NOPM) { - soc_widget_read(w, reg, &val); + soc_dapm_read(w->dapm, reg, &val); val = (val >> shift) & mask; if (invert) val = max - val; @@ -1306,16 +1329,18 @@ static void dapm_seq_check_event(struct snd_soc_card *card, static void dapm_seq_run_coalesced(struct snd_soc_card *card, struct list_head *pending) { + struct snd_soc_dapm_context *dapm; struct snd_soc_dapm_widget *w; int reg; unsigned int value = 0; unsigned int mask = 0; - reg = list_first_entry(pending, struct snd_soc_dapm_widget, - power_list)->reg; + w = list_first_entry(pending, struct snd_soc_dapm_widget, power_list); + reg = w->reg; + dapm = w->dapm; list_for_each_entry(w, pending, power_list) { - WARN_ON(reg != w->reg); + WARN_ON(reg != w->reg || dapm != w->dapm); w->power = w->new_power; mask |= w->mask << w->shift; @@ -1324,7 +1349,7 @@ static void dapm_seq_run_coalesced(struct snd_soc_card *card, else value |= w->off_val << w->shift; - pop_dbg(w->dapm->dev, card->pop_time, + pop_dbg(dapm->dev, card->pop_time, "pop test : Queue %s: reg=0x%x, 0x%x/0x%x\n", w->name, reg, value, mask); @@ -1337,14 +1362,12 @@ static void dapm_seq_run_coalesced(struct snd_soc_card *card, /* Any widget will do, they should all be updating the * same register. */ - w = list_first_entry(pending, struct snd_soc_dapm_widget, - power_list); - pop_dbg(w->dapm->dev, card->pop_time, + pop_dbg(dapm->dev, card->pop_time, "pop test : Applying 0x%x/0x%x to %x in %dms\n", value, mask, reg, card->pop_time); pop_wait(card->pop_time); - soc_widget_update_bits(w, reg, mask, value); + soc_dapm_update_bits(dapm, reg, mask, value); } list_for_each_entry(w, pending, power_list) { @@ -1490,7 +1513,8 @@ static void dapm_widget_update(struct snd_soc_card *card) if (!w) return; - ret = soc_widget_update_bits(w, update->reg, update->mask, update->val); + ret = soc_dapm_update_bits(w->dapm, update->reg, update->mask, + update->val); if (ret < 0) dev_err(w->dapm->dev, "ASoC: %s DAPM update failed: %d\n", w->name, ret); @@ -2672,7 +2696,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) /* Read the initial power state from the device */ if (w->reg >= 0) { - soc_widget_read(w, w->reg, &val); + soc_dapm_read(w->dapm, w->reg, &val); val = val >> w->shift; val &= w->mask; if (val == w->on_val) @@ -2703,8 +2727,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_new_widgets); int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); - struct snd_soc_card *card = codec->card; + struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol); + struct snd_soc_card *card = dapm->card; struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; int reg = mc->reg; @@ -2713,17 +2737,20 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; unsigned int val; + int ret = 0; if (snd_soc_volsw_is_stereo(mc)) - dev_warn(codec->dapm.dev, + dev_warn(dapm->dev, "ASoC: Control '%s' is stereo, which is not supported\n", kcontrol->id.name); mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - if (dapm_kcontrol_is_powered(kcontrol) && reg != SND_SOC_NOPM) - val = (snd_soc_read(codec, reg) >> shift) & mask; - else + if (dapm_kcontrol_is_powered(kcontrol) && reg != SND_SOC_NOPM) { + ret = soc_dapm_read(dapm, reg, &val); + val = (val >> shift) & mask; + } else { val = dapm_kcontrol_get_value(kcontrol); + } mutex_unlock(&card->dapm_mutex); if (invert) @@ -2731,7 +2758,7 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, else ucontrol->value.integer.value[0] = val; - return 0; + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_get_volsw); @@ -2747,8 +2774,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_get_volsw); int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); - struct snd_soc_card *card = codec->card; + struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol); + struct snd_soc_card *card = dapm->card; struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; int reg = mc->reg; @@ -2762,7 +2789,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, int ret = 0; if (snd_soc_volsw_is_stereo(mc)) - dev_warn(codec->dapm.dev, + dev_warn(dapm->dev, "ASoC: Control '%s' is stereo, which is not supported\n", kcontrol->id.name); @@ -2780,7 +2807,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, mask = mask << shift; val = val << shift; - reg_change = snd_soc_test_bits(codec, reg, mask, val); + reg_change = soc_dapm_test_bits(dapm, reg, mask, val); } if (change || reg_change) { @@ -2819,12 +2846,13 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_put_volsw); int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); + struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int reg_val, val; + int ret = 0; if (e->reg != SND_SOC_NOPM) - reg_val = snd_soc_read(codec, e->reg); + ret = soc_dapm_read(dapm, e->reg, ®_val); else reg_val = dapm_kcontrol_get_value(kcontrol); @@ -2836,7 +2864,7 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, ucontrol->value.enumerated.item[1] = val; } - return 0; + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_double); @@ -2852,8 +2880,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_double); int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); - struct snd_soc_card *card = codec->card; + struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol); + struct snd_soc_card *card = dapm->card; struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int *item = ucontrol->value.enumerated.item; unsigned int val, change; @@ -2876,7 +2904,7 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); if (e->reg != SND_SOC_NOPM) - change = snd_soc_test_bits(codec, e->reg, mask, val); + change = soc_dapm_test_bits(dapm, e->reg, mask, val); else change = dapm_kcontrol_set_value(kcontrol, val); -- cgit v1.2.3 From bc9af9fa9b89cb74eed020066d882abf238fad69 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 16 Jun 2014 18:13:07 +0200 Subject: ASoC: Use component DAPM context for platforms The snd_soc_platform dapm field is not accessed outside of the ASoC core. Switch it over to using the snd_soc_component DAPM context. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 4 +--- sound/soc/soc-core.c | 25 ++++++++++++------------- 2 files changed, 13 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index a21dfecba56b..84ebc079c92f 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -862,8 +862,6 @@ struct snd_soc_platform { struct snd_soc_component component; - struct snd_soc_dapm_context dapm; - #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_platform_root; #endif @@ -1203,7 +1201,7 @@ static inline struct snd_soc_codec *snd_soc_dapm_to_codec( static inline struct snd_soc_platform *snd_soc_dapm_to_platform( struct snd_soc_dapm_context *dapm) { - return container_of(dapm, struct snd_soc_platform, dapm); + return snd_soc_component_to_platform(snd_soc_dapm_to_component(dapm)); } /** diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f519a9f7571c..711e99c43be3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -314,7 +314,7 @@ static void soc_init_platform_debugfs(struct snd_soc_platform *platform) return; } - snd_soc_dapm_debugfs_init(&platform->dapm, + snd_soc_dapm_debugfs_init(&platform->component.dapm, platform->debugfs_platform_root); } @@ -974,7 +974,7 @@ static int soc_remove_platform(struct snd_soc_platform *platform) } /* Make sure all DAPM widgets are freed */ - snd_soc_dapm_free(&platform->dapm); + snd_soc_dapm_free(&platform->component.dapm); soc_cleanup_platform_debugfs(platform); platform->probed = 0; @@ -1210,7 +1210,7 @@ static int soc_probe_platform(struct snd_soc_card *card, struct snd_soc_dai *dai; platform->card = card; - platform->dapm.card = card; + platform->component.dapm.card = card; if (!try_module_get(platform->dev->driver->owner)) return -ENODEV; @@ -1218,7 +1218,7 @@ static int soc_probe_platform(struct snd_soc_card *card, soc_init_platform_debugfs(platform); if (driver->dapm_widgets) - snd_soc_dapm_new_controls(&platform->dapm, + snd_soc_dapm_new_controls(&platform->component.dapm, driver->dapm_widgets, driver->num_dapm_widgets); /* Create DAPM widgets for each DAI stream */ @@ -1226,10 +1226,11 @@ static int soc_probe_platform(struct snd_soc_card *card, if (component->dev != platform->dev) continue; list_for_each_entry(dai, &component->dai_list, list) - snd_soc_dapm_new_dai_widgets(&platform->dapm, dai); + snd_soc_dapm_new_dai_widgets(&platform->component.dapm, + dai); } - platform->dapm.idle_bias_off = 1; + platform->component.dapm.idle_bias_off = 1; if (driver->probe) { ret = driver->probe(platform); @@ -1244,13 +1245,13 @@ static int soc_probe_platform(struct snd_soc_card *card, snd_soc_add_platform_controls(platform, driver->controls, driver->num_controls); if (driver->dapm_routes) - snd_soc_dapm_add_routes(&platform->dapm, driver->dapm_routes, - driver->num_dapm_routes); + snd_soc_dapm_add_routes(&platform->component.dapm, + driver->dapm_routes, driver->num_dapm_routes); /* mark platform as probed and add to card platform list */ platform->probed = 1; list_add(&platform->card_list, &card->platform_dev_list); - list_add(&platform->dapm.list, &card->dapm_list); + list_add(&platform->component.dapm.list, &card->dapm_list); return 0; @@ -4141,8 +4142,6 @@ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, { int ret; - platform->component.dapm_ptr = &platform->dapm; - ret = snd_soc_component_initialize(&platform->component, &platform_drv->component_driver, dev); if (ret) @@ -4150,8 +4149,8 @@ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, platform->dev = dev; platform->driver = platform_drv; - platform->dapm.platform = platform; - platform->dapm.stream_event = platform_drv->stream_event; + platform->component.dapm.platform = platform; + platform->component.dapm.stream_event = platform_drv->stream_event; if (platform_drv->write) platform->component.write = snd_soc_platform_drv_write; if (platform_drv->read) -- cgit v1.2.3 From 14e8bdebfbc1d5c8804b3520233b2d4e516056bc Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 16 Jun 2014 18:13:08 +0200 Subject: ASoC: Add component level stream_event() and seq_notifier() support This patch adds stream_event() and seq_notifier() callbacks similar to those found in the snd_soc_codec_driver and snd_soc_platform driver struct to the snd_soc_component_driver struct. This is meant to unify the handling of these callbacks across different types of components and will eventually allow their removal from the CODEC and platfrom driver structs. The new callbacks are slightly different from the old ones in that they take a snd_soc_component as a parameter rather than a snd_soc_dapm_context. This was done since otherwise casting from the DAPM context to the component would typically be the first thing to do in the callback. And the interface becomes slightly cleaner by passing a snd_soc_component to all callbacks in the snd_soc_component_driver struct. The patch also already removes the stream_event() callback from the snd_soc_codec_driver and snd_soc_platform_driver structs as it is currently unused. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 9 +++------ sound/soc/soc-core.c | 25 ++++++++++++++++++++++--- 2 files changed, 25 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index 84ebc079c92f..9a5b4f6fe847 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -677,6 +677,9 @@ struct snd_soc_component_driver { int (*of_xlate_dai_name)(struct snd_soc_component *component, struct of_phandle_args *args, const char **dai_name); + void (*seq_notifier)(struct snd_soc_component *, enum snd_soc_dapm_type, + int subseq); + int (*stream_event)(struct snd_soc_component *, int event); }; struct snd_soc_component { @@ -792,9 +795,6 @@ struct snd_soc_codec_driver { void (*seq_notifier)(struct snd_soc_dapm_context *, enum snd_soc_dapm_type, int); - /* codec stream completion event */ - int (*stream_event)(struct snd_soc_dapm_context *dapm, int event); - bool ignore_pmdown_time; /* Doesn't benefit from pmdown delay */ /* probe ordering - for components with runtime dependencies */ @@ -836,9 +836,6 @@ struct snd_soc_platform_driver { /* platform stream compress ops */ const struct snd_compr_ops *compr_ops; - /* platform stream completion event */ - int (*stream_event)(struct snd_soc_dapm_context *dapm, int event); - /* probe ordering - for components with runtime dependencies */ int probe_order; int remove_order; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 711e99c43be3..5fe732fdfd59 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3995,6 +3995,22 @@ err: return ret; } +static void snd_soc_component_seq_notifier(struct snd_soc_dapm_context *dapm, + enum snd_soc_dapm_type type, int subseq) +{ + struct snd_soc_component *component = dapm->component; + + component->driver->seq_notifier(component, type, subseq); +} + +static int snd_soc_component_stream_event(struct snd_soc_dapm_context *dapm, + int event) +{ + struct snd_soc_component *component = dapm->component; + + return component->driver->stream_event(component, event); +} + static int snd_soc_component_initialize(struct snd_soc_component *component, const struct snd_soc_component_driver *driver, struct device *dev) { @@ -4016,6 +4032,10 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, dapm->dev = dev; dapm->component = component; dapm->bias_level = SND_SOC_BIAS_OFF; + if (driver->seq_notifier) + dapm->seq_notifier = snd_soc_component_seq_notifier; + if (driver->stream_event) + dapm->stream_event = snd_soc_component_stream_event; INIT_LIST_HEAD(&component->dai_list); mutex_init(&component->io_mutex); @@ -4150,7 +4170,6 @@ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, platform->dev = dev; platform->driver = platform_drv; platform->component.dapm.platform = platform; - platform->component.dapm.stream_event = platform_drv->stream_event; if (platform_drv->write) platform->component.write = snd_soc_platform_drv_write; if (platform_drv->read) @@ -4336,8 +4355,8 @@ int snd_soc_register_codec(struct device *dev, codec->component.read = snd_soc_codec_drv_read; codec->component.ignore_pmdown_time = codec_drv->ignore_pmdown_time; codec->dapm.codec = codec; - codec->dapm.seq_notifier = codec_drv->seq_notifier; - codec->dapm.stream_event = codec_drv->stream_event; + if (codec_drv->seq_notifier) + codec->dapm.seq_notifier = codec_drv->seq_notifier; if (codec_drv->set_bias_level) codec->dapm.set_bias_level = snd_soc_codec_set_bias_level; codec->dev = dev; -- cgit v1.2.3 From 9420d97b3f8672478696ae9c3e33051243e1f4a0 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 16 Jun 2014 18:13:10 +0200 Subject: ASoC: dapm: Remove DAI DAPM context The DAI DAPM context was added in commit be09ad90 ("ASoC: core: Add platform DAI widget mapping") and the only user was removed again in commit ae10e7e8f ("ASoC: core: Only add platform DAI widgets once."). Now that we have a per component DAPM context it is unlikely that we'll need the DAI DAPM context again. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 1 - sound/soc/soc-core.c | 12 +----------- 2 files changed, 1 insertion(+), 12 deletions(-) (limited to 'sound') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 688f2ba8009f..031be2ab75d0 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -257,7 +257,6 @@ struct snd_soc_dai { struct snd_soc_dapm_widget *playback_widget; struct snd_soc_dapm_widget *capture_widget; - struct snd_soc_dapm_context dapm; /* DAI DMA data */ void *playback_dma_data; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 5fe732fdfd59..995a4b7f87eb 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1048,11 +1048,8 @@ static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order) cpu_dai->name, err); } cpu_dai->probed = 0; - - if (!cpu_dai->codec) { - snd_soc_dapm_free(&cpu_dai->dapm); + if (!cpu_dai->codec) module_put(cpu_dai->dev->driver->owner); - } } } @@ -1510,11 +1507,8 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) if (!cpu_dai->probed && cpu_dai->driver->probe_order == order) { if (!cpu_dai->codec) { - cpu_dai->dapm.card = card; if (!try_module_get(cpu_dai->dev->driver->owner)) return -ENODEV; - - list_add(&cpu_dai->dapm.list, &card->dapm_list); } if (cpu_dai->driver->probe) { @@ -3975,13 +3969,9 @@ static int snd_soc_register_dais(struct snd_soc_component *component, dai->component = component; dai->dev = dev; dai->driver = &dai_drv[i]; - dai->dapm.dev = dev; if (!dai->driver->ops) dai->driver->ops = &null_dai_ops; - if (!dai->codec) - dai->dapm.idle_bias_off = 1; - list_add(&dai->list, &component->dai_list); dev_dbg(dev, "ASoC: Registered DAI '%s'\n", dai->name); -- cgit v1.2.3 From 88a8fe3df65fa0229b04f1c03411062230091cdd Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 16 Jun 2014 18:13:11 +0200 Subject: ASoC: dapm: Remove platform field from widget and dapm context struct The platform field in the snd_soc_dapm_widget and snd_soc_dapm_context structs is now unused can be removed. New code that wants to get the platform for a widget or dapm context should use snd_soc_dapm_to_platform(w->dapm) or snd_soc_dapm_to_platform(dapm). Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 2 -- sound/soc/soc-core.c | 1 - sound/soc/soc-dapm.c | 1 - 3 files changed, 4 deletions(-) (limited to 'sound') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index e292683ee694..aac04ff84eea 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -526,7 +526,6 @@ struct snd_soc_dapm_widget { const char *name; /* widget name */ const char *sname; /* stream name */ struct snd_soc_codec *codec; - struct snd_soc_platform *platform; struct list_head list; struct snd_soc_dapm_context *dapm; @@ -595,7 +594,6 @@ struct snd_soc_dapm_context { struct device *dev; /* from parent - for debug */ struct snd_soc_component *component; /* parent component */ struct snd_soc_codec *codec; /* parent codec */ - struct snd_soc_platform *platform; /* parent platform */ struct snd_soc_card *card; /* parent card */ /* used during DAPM updates */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 995a4b7f87eb..19c1e9588e05 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4159,7 +4159,6 @@ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, platform->dev = dev; platform->driver = platform_drv; - platform->component.dapm.platform = platform; if (platform_drv->write) platform->component.write = snd_soc_platform_drv_write; if (platform_drv->read) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 4702b926a6a0..4bf08cffd1f1 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3097,7 +3097,6 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, w->dapm = dapm; w->codec = dapm->codec; - w->platform = dapm->platform; INIT_LIST_HEAD(&w->sources); INIT_LIST_HEAD(&w->sinks); INIT_LIST_HEAD(&w->list); -- cgit v1.2.3 From 5264d0e6ef0a926eaf11313715c15de737b2f0b3 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Wed, 4 Jun 2014 11:40:40 +0200 Subject: ASoC: samsung: Add I2C dependency for snow Both codecs used by snow, max98090 and max98095 require the use of I2C, so we can only select this driver if I2C is there, otherwise we get a build error like: codecs/max98090.c:2494:1: warning: data definition has no type or storage class [enabled by default] module_i2c_driver(max98090_i2c_driver); ^ codecs/max98095.c:2443:1: warning: data definition has no type or storage class [enabled by default] module_i2c_driver(max98095_i2c_driver); ^ This adds one more I2C dependency to the hundreds we already have. Signed-off-by: Arnd Bergmann Acked-by: Tushar Behera Signed-off-by: Mark Brown --- sound/soc/samsung/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 753b8c93ab51..7745629e2c88 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -236,7 +236,7 @@ config SND_SOC_LITTLEMILL config SND_SOC_SNOW tristate "Audio support for Google Snow boards" - depends on SND_SOC_SAMSUNG + depends on SND_SOC_SAMSUNG && I2C select SND_SOC_MAX98090 select SND_SOC_MAX98095 select SND_SAMSUNG_I2S -- cgit v1.2.3 From ff40260f79dc0436604452bccd449bffd25ebafb Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Tue, 3 Jun 2014 14:11:56 +0200 Subject: ASoC: fsl: refine DMA/FIQ dependencies Commit 31ee2bfd724ab ("ASoC: fsl: select SND_SOC_IMX_PCM_DMA where needed") started selecting SND_SOC_IMX_PCM_DMA and SND_SOC_IMX_PCM_FIQ for two drivers when building for i.MX. This has turned out too aggressive, as FIQ is only available for i.mx2 through i.mx5, but not i.mx6 or vybrid. Further, two more drivers have become user-selectable in the meantime, and they both depend on DMA for the imx platform as well. This changes the selection of FIQ to depend on the TZIC or AVIC interrupt controllers that actually export the imx specific FIQ interfaces, and adds the missing select statements for SAI and ESAI. Signed-off-by: Arnd Bergmann Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 37933629cbed..52bbe9f80111 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -5,6 +5,7 @@ comment "Common SoC Audio options for Freescale CPUs:" config SND_SOC_FSL_SAI tristate "Synchronous Audio Interface (SAI) module support" select REGMAP_MMIO + select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y if you want to add Synchronous Audio Interface (SAI) @@ -15,7 +16,7 @@ config SND_SOC_FSL_SAI config SND_SOC_FSL_SSI tristate "Synchronous Serial Interface module support" select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n - select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && ARCH_MXC + select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && (MXC_TZIC || MXC_AVIC) select REGMAP_MMIO help Say Y if you want to add Synchronous Serial Interface (SSI) @@ -27,7 +28,7 @@ config SND_SOC_FSL_SPDIF tristate "Sony/Philips Digital Interface module support" select REGMAP_MMIO select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n - select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && ARCH_MXC + select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && (MXC_TZIC || MXC_AVIC) help Say Y if you want to add Sony/Philips Digital Interface (SPDIF) support for the Freescale CPUs. @@ -37,6 +38,7 @@ config SND_SOC_FSL_SPDIF config SND_SOC_FSL_ESAI tristate "Enhanced Serial Audio Interface (ESAI) module support" select REGMAP_MMIO + select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n select SND_SOC_FSL_UTILS help Say Y if you want to add Enhanced Synchronous Audio Interface -- cgit v1.2.3 From aa9b045f70160c664291d5482270baf2ed89cc1b Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Fri, 13 Jun 2014 18:03:54 +0530 Subject: ASoC: Intel: add the mrfld fw IPC definations This will be used to update current driver as well as in support for the mrfld patches Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-dsp.h | 414 +++++++++++++++++++++++++++++++++++++++-- 1 file changed, 403 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-mfld-dsp.h b/sound/soc/intel/sst-mfld-dsp.h index 8d482d76475a..2c887855e7d8 100644 --- a/sound/soc/intel/sst-mfld-dsp.h +++ b/sound/soc/intel/sst-mfld-dsp.h @@ -3,7 +3,7 @@ /* * sst_mfld_dsp.h - Intel SST Driver for audio engine * - * Copyright (C) 2008-12 Intel Corporation + * Copyright (C) 2008-14 Intel Corporation * Authors: Vinod Koul * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ * @@ -19,6 +19,142 @@ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ */ +#define SST_MAX_BIN_BYTES 1024 + +#define MAX_DBG_RW_BYTES 80 +#define MAX_NUM_SCATTER_BUFFERS 8 +#define MAX_LOOP_BACK_DWORDS 8 +/* IPC base address and mailbox, timestamp offsets */ +#define SST_MAILBOX_SIZE 0x0400 +#define SST_MAILBOX_SEND 0x0000 +#define SST_TIME_STAMP 0x1800 +#define SST_TIME_STAMP_MRFLD 0x800 +#define SST_RESERVED_OFFSET 0x1A00 +#define SST_SCU_LPE_MAILBOX 0x1000 +#define SST_LPE_SCU_MAILBOX 0x1400 +#define SST_SCU_LPE_LOG_BUF (SST_SCU_LPE_MAILBOX+16) +#define PROCESS_MSG 0x80 + +/* Message ID's for IPC messages */ +/* Bits B7: SST or IA/SC ; B6-B4: Msg Category; B3-B0: Msg Type */ + +/* I2L Firmware/Codec Download msgs */ +#define IPC_IA_PREP_LIB_DNLD 0x01 +#define IPC_IA_LIB_DNLD_CMPLT 0x02 +#define IPC_IA_GET_FW_VERSION 0x04 +#define IPC_IA_GET_FW_BUILD_INF 0x05 +#define IPC_IA_GET_FW_INFO 0x06 +#define IPC_IA_GET_FW_CTXT 0x07 +#define IPC_IA_SET_FW_CTXT 0x08 +#define IPC_IA_PREPARE_SHUTDOWN 0x31 +/* I2L Codec Config/control msgs */ +#define IPC_PREP_D3 0x10 +#define IPC_IA_SET_CODEC_PARAMS 0x10 +#define IPC_IA_GET_CODEC_PARAMS 0x11 +#define IPC_IA_SET_PPP_PARAMS 0x12 +#define IPC_IA_GET_PPP_PARAMS 0x13 +#define IPC_SST_PERIOD_ELAPSED_MRFLD 0xA +#define IPC_IA_ALG_PARAMS 0x1A +#define IPC_IA_TUNING_PARAMS 0x1B +#define IPC_IA_SET_RUNTIME_PARAMS 0x1C +#define IPC_IA_SET_PARAMS 0x1 +#define IPC_IA_GET_PARAMS 0x2 + +#define IPC_EFFECTS_CREATE 0xE +#define IPC_EFFECTS_DESTROY 0xF + +/* I2L Stream config/control msgs */ +#define IPC_IA_ALLOC_STREAM_MRFLD 0x2 +#define IPC_IA_ALLOC_STREAM 0x20 /* Allocate a stream ID */ +#define IPC_IA_FREE_STREAM_MRFLD 0x03 +#define IPC_IA_FREE_STREAM 0x21 /* Free the stream ID */ +#define IPC_IA_SET_STREAM_PARAMS 0x22 +#define IPC_IA_SET_STREAM_PARAMS_MRFLD 0x12 +#define IPC_IA_GET_STREAM_PARAMS 0x23 +#define IPC_IA_PAUSE_STREAM 0x24 +#define IPC_IA_PAUSE_STREAM_MRFLD 0x4 +#define IPC_IA_RESUME_STREAM 0x25 +#define IPC_IA_RESUME_STREAM_MRFLD 0x5 +#define IPC_IA_DROP_STREAM 0x26 +#define IPC_IA_DROP_STREAM_MRFLD 0x07 +#define IPC_IA_DRAIN_STREAM 0x27 /* Short msg with str_id */ +#define IPC_IA_DRAIN_STREAM_MRFLD 0x8 +#define IPC_IA_CONTROL_ROUTING 0x29 +#define IPC_IA_VTSV_UPDATE_MODULES 0x20 +#define IPC_IA_VTSV_DETECTED 0x21 + +#define IPC_IA_START_STREAM_MRFLD 0X06 +#define IPC_IA_START_STREAM 0x30 /* Short msg with str_id */ + +#define IPC_IA_SET_GAIN_MRFLD 0x21 +/* Debug msgs */ +#define IPC_IA_DBG_MEM_READ 0x40 +#define IPC_IA_DBG_MEM_WRITE 0x41 +#define IPC_IA_DBG_LOOP_BACK 0x42 +#define IPC_IA_DBG_LOG_ENABLE 0x45 +#define IPC_IA_DBG_SET_PROBE_PARAMS 0x47 + +/* L2I Firmware/Codec Download msgs */ +#define IPC_IA_FW_INIT_CMPLT 0x81 +#define IPC_IA_FW_INIT_CMPLT_MRFLD 0x01 +#define IPC_IA_FW_ASYNC_ERR_MRFLD 0x11 + +/* L2I Codec Config/control msgs */ +#define IPC_SST_FRAGMENT_ELPASED 0x90 /* Request IA more data */ + +#define IPC_SST_BUF_UNDER_RUN 0x92 /* PB Under run and stopped */ +#define IPC_SST_BUF_OVER_RUN 0x93 /* CAP Under run and stopped */ +#define IPC_SST_DRAIN_END 0x94 /* PB Drain complete and stopped */ +#define IPC_SST_CHNGE_SSP_PARAMS 0x95 /* PB SSP parameters changed */ +#define IPC_SST_STREAM_PROCESS_FATAL_ERR 0x96/* error in processing a stream */ +#define IPC_SST_PERIOD_ELAPSED 0x97 /* period elapsed */ + +#define IPC_SST_ERROR_EVENT 0x99 /* Buffer over run occurred */ +/* L2S messages */ +#define IPC_SC_DDR_LINK_UP 0xC0 +#define IPC_SC_DDR_LINK_DOWN 0xC1 +#define IPC_SC_SET_LPECLK_REQ 0xC2 +#define IPC_SC_SSP_BIT_BANG 0xC3 + +/* L2I Error reporting msgs */ +#define IPC_IA_MEM_ALLOC_FAIL 0xE0 +#define IPC_IA_PROC_ERR 0xE1 /* error in processing a + stream can be used by playback and + capture modules */ + +/* L2I Debug msgs */ +#define IPC_IA_PRINT_STRING 0xF0 + +/* Buffer under-run */ +#define IPC_IA_BUF_UNDER_RUN_MRFLD 0x0B + +/* Mrfld specific defines: + * For asynchronous messages(INIT_CMPLT, PERIOD_ELAPSED, ASYNC_ERROR) + * received from FW, the format is: + * - IPC High: pvt_id is set to zero. Always short message. + * - msg_id is in lower 16-bits of IPC low payload. + * - pipe_id is in higher 16-bits of IPC low payload for period_elapsed. + * - error id is in higher 16-bits of IPC low payload for async errors. + */ +#define SST_ASYNC_DRV_ID 0 + +/* Command Response or Acknowledge message to any IPC message will have + * same message ID and stream ID information which is sent. + * There is no specific Ack message ID. The data field is used as response + * meaning. + */ +enum ackData { + IPC_ACK_SUCCESS = 0, + IPC_ACK_FAILURE, +}; + +enum ipc_ia_msg_id { + IPC_CMD = 1, /*!< Task Control message ID */ + IPC_SET_PARAMS = 2,/*!< Task Set param message ID */ + IPC_GET_PARAMS = 3, /*!< Task Get param message ID */ + IPC_INVALID = 0xFF, /*! Date: Fri, 13 Jun 2014 18:03:55 +0530 Subject: ASoC: Intel: mfld-pcm: modularize stream allocation code Tis will be used to add table based support for pcm front ends in subsequent patches Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-platform-pcm.c | 102 ++++++++++++++++++++++---------- sound/soc/intel/sst-mfld-platform.h | 2 +- 2 files changed, 71 insertions(+), 33 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 80879e5fcb49..6e7bfb1bc4aa 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -143,52 +143,90 @@ static inline int sst_get_stream_status(struct sst_runtime_stream *stream) return state; } +static void sst_fill_alloc_params(struct snd_pcm_substream *substream, + struct snd_sst_alloc_params_ext *alloc_param) +{ + unsigned int channels; + snd_pcm_uframes_t period_size; + ssize_t periodbytes; + ssize_t buffer_bytes = snd_pcm_lib_buffer_bytes(substream); + u32 buffer_addr = virt_to_phys(substream->dma_buffer.area); + + channels = substream->runtime->channels; + period_size = substream->runtime->period_size; + periodbytes = samples_to_bytes(substream->runtime, period_size); + alloc_param->ring_buf_info[0].addr = buffer_addr; + alloc_param->ring_buf_info[0].size = buffer_bytes; + alloc_param->sg_count = 1; + alloc_param->reserved = 0; + alloc_param->frag_size = periodbytes * channels; + +} static void sst_fill_pcm_params(struct snd_pcm_substream *substream, - struct sst_pcm_params *param) + struct snd_sst_stream_params *param) { + param->uc.pcm_params.num_chan = (u8) substream->runtime->channels; + param->uc.pcm_params.pcm_wd_sz = substream->runtime->sample_bits; + param->uc.pcm_params.sfreq = substream->runtime->rate; - param->num_chan = (u8) substream->runtime->channels; - param->pcm_wd_sz = substream->runtime->sample_bits; - param->reserved = 0; - param->sfreq = substream->runtime->rate; - param->ring_buffer_size = snd_pcm_lib_buffer_bytes(substream); - param->period_count = substream->runtime->period_size; - param->ring_buffer_addr = virt_to_phys(substream->dma_buffer.area); - pr_debug("period_cnt = %d\n", param->period_count); - pr_debug("sfreq= %d, wd_sz = %d\n", param->sfreq, param->pcm_wd_sz); + /* PCM stream via ALSA interface */ + param->uc.pcm_params.use_offload_path = 0; + param->uc.pcm_params.reserved2 = 0; + memset(param->uc.pcm_params.channel_map, 0, sizeof(u8)); + +} +int sst_fill_stream_params(void *substream, + struct snd_sst_params *str_params, bool is_compress) +{ + struct snd_pcm_substream *pstream = NULL; + struct snd_compr_stream *cstream = NULL; + + if (is_compress == true) + cstream = (struct snd_compr_stream *)substream; + else + pstream = (struct snd_pcm_substream *)substream; + + str_params->stream_type = SST_STREAM_TYPE_MUSIC; + + /* For pcm streams */ + if (pstream) + str_params->ops = (u8)pstream->stream; + if (cstream) + str_params->ops = (u8)cstream->direction; + + return 0; } -static int sst_platform_alloc_stream(struct snd_pcm_substream *substream) +static int sst_platform_alloc_stream(struct snd_pcm_substream *substream, + struct snd_soc_platform *platform) { struct sst_runtime_stream *stream = substream->runtime->private_data; - struct sst_pcm_params param = {0}; - struct sst_stream_params str_params = {0}; - int ret_val; + struct snd_sst_stream_params param = {{{0,},},}; + struct snd_sst_params str_params = {0}; + struct snd_sst_alloc_params_ext alloc_params = {0}; + int ret_val = 0; /* set codec params and inform SST driver the same */ sst_fill_pcm_params(substream, ¶m); + sst_fill_alloc_params(substream, &alloc_params); substream->runtime->dma_area = substream->dma_buffer.area; str_params.sparams = param; - str_params.codec = param.codec; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - str_params.ops = STREAM_OPS_PLAYBACK; - str_params.device_type = substream->pcm->device + 1; - pr_debug("Playbck stream,Device %d\n", - substream->pcm->device); - } else { - str_params.ops = STREAM_OPS_CAPTURE; - str_params.device_type = SND_SST_DEVICE_CAPTURE; - pr_debug("Capture stream,Device %d\n", - substream->pcm->device); - } - ret_val = stream->ops->open(&str_params); - pr_debug("SST_SND_PLAY/CAPTURE ret_val = %x\n", ret_val); + str_params.aparams = alloc_params; + str_params.codec = SST_CODEC_TYPE_PCM; + + /* fill the device type and stream id to pass to SST driver */ + ret_val = sst_fill_stream_params(substream, &str_params, false); if (ret_val < 0) return ret_val; - stream->stream_info.str_id = ret_val; - pr_debug("str id : %d\n", stream->stream_info.str_id); + stream->stream_info.str_id = str_params.stream_id; + + ret_val = stream->ops->open(&str_params); + if (ret_val <= 0) + return ret_val; + + return ret_val; } @@ -300,8 +338,8 @@ static int sst_media_prepare(struct snd_pcm_substream *substream, return ret_val; } - ret_val = sst_platform_alloc_stream(substream); - if (ret_val < 0) + ret_val = sst_platform_alloc_stream(substream, dai->platform); + if (ret_val <= 0) return ret_val; snprintf(substream->pcm->id, sizeof(substream->pcm->id), "%d", stream->stream_info.str_id); diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index 33a0a2776238..aa5ddbb26d93 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -125,7 +125,7 @@ struct compress_sst_ops { }; struct sst_ops { - int (*open) (struct sst_stream_params *str_param); + int (*open) (struct snd_sst_params *str_param); int (*device_control) (int cmd, void *arg); int (*close) (unsigned int str_id); }; -- cgit v1.2.3 From eb72cbdf51d0b12aa2a8169110950d163b5d693b Mon Sep 17 00:00:00 2001 From: Shahina Shaik Date: Fri, 13 Jun 2014 11:18:47 +0530 Subject: ASoC: tlv320aic32x4: Fixed Coding style Issues Fixed a brace coding style issue in the tlv320aic32x4.c Signed-off-by: Shahina Shaik Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 1d9b117345a3..1e50f15022bc 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -626,21 +626,18 @@ static int aic32x4_probe(struct snd_soc_codec *codec) snd_soc_write(codec, AIC32X4_MICBIAS, AIC32X4_MICBIAS_LDOIN | AIC32X4_MICBIAS_2075V); } - if (aic32x4->power_cfg & AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE) { + if (aic32x4->power_cfg & AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE) snd_soc_write(codec, AIC32X4_PWRCFG, AIC32X4_AVDDWEAKDISABLE); - } tmp_reg = (aic32x4->power_cfg & AIC32X4_PWR_AIC32X4_LDO_ENABLE) ? AIC32X4_LDOCTLEN : 0; snd_soc_write(codec, AIC32X4_LDOCTL, tmp_reg); tmp_reg = snd_soc_read(codec, AIC32X4_CMMODE); - if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_LDOIN_RANGE_18_36) { + if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_LDOIN_RANGE_18_36) tmp_reg |= AIC32X4_LDOIN_18_36; - } - if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_HP_LDOIN_POWERED) { + if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_HP_LDOIN_POWERED) tmp_reg |= AIC32X4_LDOIN2HP; - } snd_soc_write(codec, AIC32X4_CMMODE, tmp_reg); /* Mic PGA routing */ -- cgit v1.2.3 From 43bf38ba566831867836569b3958064cd5c37eac Mon Sep 17 00:00:00 2001 From: Shahina Shaik Date: Fri, 13 Jun 2014 11:18:48 +0530 Subject: ASoC: tlv320aic32x4: Fixed Coding Style Issues Fixed Coding style issues of lines over 80 characters. Signed-off-by: Shahina Shaik Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 1e50f15022bc..89e41d2f7586 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -642,13 +642,17 @@ static int aic32x4_probe(struct snd_soc_codec *codec) /* Mic PGA routing */ if (aic32x4->micpga_routing & AIC32X4_MICPGA_ROUTE_LMIC_IN2R_10K) - snd_soc_write(codec, AIC32X4_LMICPGANIN, AIC32X4_LMICPGANIN_IN2R_10K); + snd_soc_write(codec, AIC32X4_LMICPGANIN, + AIC32X4_LMICPGANIN_IN2R_10K); else - snd_soc_write(codec, AIC32X4_LMICPGANIN, AIC32X4_LMICPGANIN_CM1L_10K); + snd_soc_write(codec, AIC32X4_LMICPGANIN, + AIC32X4_LMICPGANIN_CM1L_10K); if (aic32x4->micpga_routing & AIC32X4_MICPGA_ROUTE_RMIC_IN1L_10K) - snd_soc_write(codec, AIC32X4_RMICPGANIN, AIC32X4_RMICPGANIN_IN1L_10K); + snd_soc_write(codec, AIC32X4_RMICPGANIN, + AIC32X4_RMICPGANIN_IN1L_10K); else - snd_soc_write(codec, AIC32X4_RMICPGANIN, AIC32X4_RMICPGANIN_CM1R_10K); + snd_soc_write(codec, AIC32X4_RMICPGANIN, + AIC32X4_RMICPGANIN_CM1R_10K); aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY); -- cgit v1.2.3 From 423ca88eb5288d3bf381bd5b4601a496ab1ae463 Mon Sep 17 00:00:00 2001 From: Shahina Shaik Date: Fri, 13 Jun 2014 11:56:54 +0530 Subject: ASoC: tlv320aic31xx: Fixed Coding Style Issues Fixed coding style issues of "Missing Blank line after declaration" Signed-off-by: Shahina Shaik Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic31xx.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 23419109ecac..a9c4a8baba88 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -329,6 +329,7 @@ static int aic31xx_wait_bits(struct aic31xx_priv *aic31xx, unsigned int reg, unsigned int bits; int counter = count; int ret = regmap_read(aic31xx->regmap, reg, &bits); + while ((bits & mask) != wbits && counter && !ret) { usleep_range(sleep, sleep * 2); ret = regmap_read(aic31xx->regmap, reg, &bits); @@ -435,6 +436,7 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w, { struct snd_soc_codec *codec = w->codec; struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + switch (event) { case SND_SOC_DAPM_POST_PMU: /* change mic bias voltage to user defined */ -- cgit v1.2.3 From 61b165caa686b8334379293d0e241f740fac195a Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Fri, 13 Jun 2014 18:03:56 +0530 Subject: ASoC: Intel: add mrfld pipelines Merrifield DSP used various pipelines to identify the streams and processing modules. Add these defination in the pcm driver and also add a table for device entries to firmware pipeline id conversion Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- arch/x86/include/asm/platform_sst_audio.h | 78 +++++++++++++++++++++++ sound/soc/intel/sst-atom-controls.h | 30 +++++++++ sound/soc/intel/sst-mfld-platform-pcm.c | 100 +++++++++++++++++++++++++++--- sound/soc/intel/sst-mfld-platform.h | 18 ++++++ 4 files changed, 217 insertions(+), 9 deletions(-) create mode 100644 arch/x86/include/asm/platform_sst_audio.h create mode 100644 sound/soc/intel/sst-atom-controls.h (limited to 'sound') diff --git a/arch/x86/include/asm/platform_sst_audio.h b/arch/x86/include/asm/platform_sst_audio.h new file mode 100644 index 000000000000..0a4e140315b6 --- /dev/null +++ b/arch/x86/include/asm/platform_sst_audio.h @@ -0,0 +1,78 @@ +/* + * platform_sst_audio.h: sst audio platform data header file + * + * Copyright (C) 2012-14 Intel Corporation + * Author: Jeeja KP + * Omair Mohammed Abdullah + * Vinod Koul ,vinod.koul@intel.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; version 2 + * of the License. + */ +#ifndef _PLATFORM_SST_AUDIO_H_ +#define _PLATFORM_SST_AUDIO_H_ + +#include + +enum sst_audio_task_id_mrfld { + SST_TASK_ID_NONE = 0, + SST_TASK_ID_SBA = 1, + SST_TASK_ID_MEDIA = 3, + SST_TASK_ID_MAX = SST_TASK_ID_MEDIA, +}; + +/* Device IDs for Merrifield are Pipe IDs, + * ref: DSP spec v0.75 */ +enum sst_audio_device_id_mrfld { + /* Output pipeline IDs */ + PIPE_ID_OUT_START = 0x0, + PIPE_CODEC_OUT0 = 0x2, + PIPE_CODEC_OUT1 = 0x3, + PIPE_SPROT_LOOP_OUT = 0x4, + PIPE_MEDIA_LOOP1_OUT = 0x5, + PIPE_MEDIA_LOOP2_OUT = 0x6, + PIPE_VOIP_OUT = 0xC, + PIPE_PCM0_OUT = 0xD, + PIPE_PCM1_OUT = 0xE, + PIPE_PCM2_OUT = 0xF, + PIPE_MEDIA0_OUT = 0x12, + PIPE_MEDIA1_OUT = 0x13, +/* Input Pipeline IDs */ + PIPE_ID_IN_START = 0x80, + PIPE_CODEC_IN0 = 0x82, + PIPE_CODEC_IN1 = 0x83, + PIPE_SPROT_LOOP_IN = 0x84, + PIPE_MEDIA_LOOP1_IN = 0x85, + PIPE_MEDIA_LOOP2_IN = 0x86, + PIPE_VOIP_IN = 0x8C, + PIPE_PCM0_IN = 0x8D, + PIPE_PCM1_IN = 0x8E, + PIPE_MEDIA0_IN = 0x8F, + PIPE_MEDIA1_IN = 0x90, + PIPE_MEDIA2_IN = 0x91, + PIPE_RSVD = 0xFF, +}; + +/* The stream map for each platform consists of an array of the below + * stream map structure. + */ +struct sst_dev_stream_map { + u8 dev_num; /* device id */ + u8 subdev_num; /* substream */ + u8 direction; + u8 device_id; /* fw id */ + u8 task_id; /* fw task */ + u8 status; +}; + +struct sst_platform_data { + /* Intel software platform id*/ + struct sst_dev_stream_map *pdev_strm_map; + unsigned int strm_map_size; +}; + +int add_sst_platform_device(void); +#endif + diff --git a/sound/soc/intel/sst-atom-controls.h b/sound/soc/intel/sst-atom-controls.h new file mode 100644 index 000000000000..14063ab8c7c5 --- /dev/null +++ b/sound/soc/intel/sst-atom-controls.h @@ -0,0 +1,30 @@ +/* + * Copyright (C) 2013-14 Intel Corp + * Author: Ramesh Babu + * Omair M Abdullah + * Samreen Nilofer + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + */ + +#ifndef __SST_CONTROLS_V2_H__ +#define __SST_CONTROLS_V2_H__ + +enum { + MERR_DPCM_AUDIO = 0, + MERR_DPCM_COMPR, +}; + + +#endif diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 6e7bfb1bc4aa..7de87887d9f8 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -1,7 +1,7 @@ /* * sst_mfld_platform.c - Intel MID Platform driver * - * Copyright (C) 2010-2013 Intel Corp + * Copyright (C) 2010-2014 Intel Corp * Author: Vinod Koul * Author: Harsha Priya * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ @@ -27,7 +27,9 @@ #include #include #include +#include #include "sst-mfld-platform.h" +#include "sst-atom-controls.h" struct sst_device *sst; static DEFINE_MUTEX(sst_lock); @@ -92,6 +94,13 @@ static struct snd_pcm_hardware sst_platform_pcm_hw = { .fifo_size = SST_FIFO_SIZE, }; +static struct sst_dev_stream_map dpcm_strm_map[] = { + {0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF}, /* Reserved, not in use */ + {MERR_DPCM_AUDIO, 0, SNDRV_PCM_STREAM_PLAYBACK, PIPE_MEDIA1_IN, SST_TASK_ID_MEDIA, 0}, + {MERR_DPCM_COMPR, 0, SNDRV_PCM_STREAM_PLAYBACK, PIPE_MEDIA0_IN, SST_TASK_ID_MEDIA, 0}, + {MERR_DPCM_AUDIO, 0, SNDRV_PCM_STREAM_CAPTURE, PIPE_PCM1_OUT, SST_TASK_ID_MEDIA, 0}, +}; + /* MFLD - MSIC */ static struct snd_soc_dai_driver sst_platform_dai[] = { { @@ -175,12 +184,36 @@ static void sst_fill_pcm_params(struct snd_pcm_substream *substream, memset(param->uc.pcm_params.channel_map, 0, sizeof(u8)); } + +static int sst_get_stream_mapping(int dev, int sdev, int dir, + struct sst_dev_stream_map *map, int size) +{ + int i; + + if (map == NULL) + return -EINVAL; + + + /* index 0 is not used in stream map */ + for (i = 1; i < size; i++) { + if ((map[i].dev_num == dev) && (map[i].direction == dir)) + return i; + } + return 0; +} + int sst_fill_stream_params(void *substream, - struct snd_sst_params *str_params, bool is_compress) + const struct sst_data *ctx, struct snd_sst_params *str_params, bool is_compress) { + int map_size; + int index; + struct sst_dev_stream_map *map; struct snd_pcm_substream *pstream = NULL; struct snd_compr_stream *cstream = NULL; + map = ctx->pdata->pdev_strm_map; + map_size = ctx->pdata->strm_map_size; + if (is_compress == true) cstream = (struct snd_compr_stream *)substream; else @@ -189,11 +222,32 @@ int sst_fill_stream_params(void *substream, str_params->stream_type = SST_STREAM_TYPE_MUSIC; /* For pcm streams */ - if (pstream) + if (pstream) { + index = sst_get_stream_mapping(pstream->pcm->device, + pstream->number, pstream->stream, + map, map_size); + if (index <= 0) + return -EINVAL; + + str_params->stream_id = index; + str_params->device_type = map[index].device_id; + str_params->task = map[index].task_id; + str_params->ops = (u8)pstream->stream; - if (cstream) - str_params->ops = (u8)cstream->direction; + } + + if (cstream) { + index = sst_get_stream_mapping(cstream->device->device, + 0, cstream->direction, + map, map_size); + if (index <= 0) + return -EINVAL; + str_params->stream_id = index; + str_params->device_type = map[index].device_id; + str_params->task = map[index].task_id; + str_params->ops = (u8)cstream->direction; + } return 0; } @@ -206,6 +260,7 @@ static int sst_platform_alloc_stream(struct snd_pcm_substream *substream, struct snd_sst_params str_params = {0}; struct snd_sst_alloc_params_ext alloc_params = {0}; int ret_val = 0; + struct sst_data *ctx = snd_soc_platform_get_drvdata(platform); /* set codec params and inform SST driver the same */ sst_fill_pcm_params(substream, ¶m); @@ -216,7 +271,7 @@ static int sst_platform_alloc_stream(struct snd_pcm_substream *substream, str_params.codec = SST_CODEC_TYPE_PCM; /* fill the device type and stream id to pass to SST driver */ - ret_val = sst_fill_stream_params(substream, &str_params, false); + ret_val = sst_fill_stream_params(substream, ctx, &str_params, false); if (ret_val < 0) return ret_val; @@ -321,7 +376,22 @@ static void sst_media_close(struct snd_pcm_substream *substream, ret_val = stream->ops->close(str_id); module_put(sst->dev->driver->owner); kfree(stream); - return; +} + +static inline unsigned int get_current_pipe_id(struct snd_soc_platform *platform, + struct snd_pcm_substream *substream) +{ + struct sst_data *sst = snd_soc_platform_get_drvdata(platform); + struct sst_dev_stream_map *map = sst->pdata->pdev_strm_map; + struct sst_runtime_stream *stream = + substream->runtime->private_data; + u32 str_id = stream->stream_info.str_id; + unsigned int pipe_id; + pipe_id = map[str_id].device_id; + + pr_debug("%s: got pipe_id = %#x for str_id = %d\n", + __func__, pipe_id, str_id); + return pipe_id; } static int sst_media_prepare(struct snd_pcm_substream *substream, @@ -498,10 +568,22 @@ static const struct snd_soc_component_driver sst_component = { static int sst_platform_probe(struct platform_device *pdev) { + struct sst_data *drv; int ret; + struct sst_platform_data *pdata = pdev->dev.platform_data; + + drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL); + if (sst == NULL) { + pr_err("kzalloc failed\n"); + return -ENOMEM; + } + + pdata->pdev_strm_map = dpcm_strm_map; + pdata->strm_map_size = ARRAY_SIZE(dpcm_strm_map); + drv->pdata = pdata; + mutex_init(&drv->lock); + dev_set_drvdata(&pdev->dev, drv); - pr_debug("sst_platform_probe called\n"); - sst = NULL; ret = snd_soc_register_platform(&pdev->dev, &sst_soc_platform_drv); if (ret) { pr_err("registering soc platform failed\n"); diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index aa5ddbb26d93..33891a86b3e7 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -144,10 +144,28 @@ struct sst_device { char *name; struct device *dev; struct sst_ops *ops; + struct platform_device *pdev; struct compress_sst_ops *compr_ops; }; +struct sst_data; + void sst_set_stream_status(struct sst_runtime_stream *stream, int state); +struct sst_algo_int_control_v2 { + struct soc_mixer_control mc; + u16 module_id; /* module identifieer */ + u16 pipe_id; /* location info: pipe_id + instance_id */ + u16 instance_id; + unsigned int value; /* Value received is stored here */ +}; + +struct sst_data { + struct platform_device *pdev; + struct sst_platform_data *pdata; + struct mutex lock; +}; + int sst_register_dsp(struct sst_device *sst); int sst_unregister_dsp(struct sst_device *sst); + #endif -- cgit v1.2.3 From 0ec66fed40e31e74a762dd7166a9bf62ebbae5da Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Fri, 13 Jun 2014 18:03:57 +0530 Subject: ASoC: Intel: use common stream allocation method for compressed stream As added in previosu patch along with stream to piep conversion si required for compressed audio too Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-platform-compress.c | 11 ++++++++--- sound/soc/intel/sst-mfld-platform.h | 7 +++---- 2 files changed, 11 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-mfld-platform-compress.c b/sound/soc/intel/sst-mfld-platform-compress.c index 02abd19fce1d..29c059ca19e8 100644 --- a/sound/soc/intel/sst-mfld-platform-compress.c +++ b/sound/soc/intel/sst-mfld-platform-compress.c @@ -100,14 +100,19 @@ static int sst_platform_compr_set_params(struct snd_compr_stream *cstream, int retval; struct snd_sst_params str_params; struct sst_compress_cb cb; + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct sst_data *ctx = snd_soc_platform_get_drvdata(platform); stream = cstream->runtime->private_data; /* construct fw structure for this*/ memset(&str_params, 0, sizeof(str_params)); - str_params.ops = STREAM_OPS_PLAYBACK; - str_params.stream_type = SST_STREAM_TYPE_MUSIC; - str_params.device_type = SND_SST_DEVICE_COMPRESS; + /* fill the device type and stream id to pass to SST driver */ + retval = sst_fill_stream_params(cstream, ctx, &str_params, true); + pr_debug("compr_set_params: fill stream params ret_val = 0x%x\n", retval); + if (retval < 0) + return retval; switch (params->codec.id) { case SND_AUDIOCODEC_MP3: { diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index 33891a86b3e7..9dc962ff1e1d 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -149,8 +149,10 @@ struct sst_device { }; struct sst_data; - void sst_set_stream_status(struct sst_runtime_stream *stream, int state); +int sst_fill_stream_params(void *substream, const struct sst_data *ctx, + struct snd_sst_params *str_params, bool is_compress); + struct sst_algo_int_control_v2 { struct soc_mixer_control mc; u16 module_id; /* module identifieer */ @@ -158,14 +160,11 @@ struct sst_algo_int_control_v2 { u16 instance_id; unsigned int value; /* Value received is stored here */ }; - struct sst_data { struct platform_device *pdev; struct sst_platform_data *pdata; struct mutex lock; }; - int sst_register_dsp(struct sst_device *sst); int sst_unregister_dsp(struct sst_device *sst); - #endif -- cgit v1.2.3 From 87b132bc0315fdfe7677449da1fb1ce12c5dda35 Mon Sep 17 00:00:00 2001 From: Vasily Khoruzhick Date: Mon, 23 Jun 2014 23:24:04 +0300 Subject: ASoC: samsung: s3c24{xx,12}-i2s: port to use generic dmaengine API Use dmaengine instead of legacy s3c24xx DMA API for s3c24xx and s3c2412 Signed-off-by: Vasily Khoruzhick Signed-off-by: Mark Brown --- sound/soc/samsung/Kconfig | 9 +++---- sound/soc/samsung/dmaengine.c | 3 +++ sound/soc/samsung/s3c-i2s-v2.c | 17 +------------ sound/soc/samsung/s3c2412-i2s.c | 47 +++++++++++++++++----------------- sound/soc/samsung/s3c24xx-i2s.c | 56 +++++++++++++++++++---------------------- 5 files changed, 56 insertions(+), 76 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 7745629e2c88..e88e598fbe0c 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -1,11 +1,10 @@ config SND_SOC_SAMSUNG tristate "ASoC support for Samsung" depends on PLAT_SAMSUNG - select S3C2410_DMA if ARCH_S3C24XX + select S3C24XX_DMAC if ARCH_S3C24XX select S3C64XX_PL080 if ARCH_S3C64XX - select SND_S3C_DMA if !ARCH_S3C24XX - select SND_S3C_DMA_LEGACY if ARCH_S3C24XX - select SND_SOC_GENERIC_DMAENGINE_PCM if !ARCH_S3C24XX + select SND_S3C_DMA + select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for codecs attached to the Samsung SoCs' Audio interfaces. You will also need to @@ -19,7 +18,6 @@ config SND_S3C_DMA_LEGACY config SND_S3C24XX_I2S tristate - select S3C24XX_DMA config SND_S3C_I2SV2_SOC tristate @@ -27,7 +25,6 @@ config SND_S3C_I2SV2_SOC config SND_S3C2412_SOC_I2S tristate select SND_S3C_I2SV2_SOC - select S3C2410_DMA config SND_SAMSUNG_PCM tristate diff --git a/sound/soc/samsung/dmaengine.c b/sound/soc/samsung/dmaengine.c index a0e4e7948909..506f5bf6d082 100644 --- a/sound/soc/samsung/dmaengine.c +++ b/sound/soc/samsung/dmaengine.c @@ -17,6 +17,7 @@ #include #include +#include #include #include @@ -29,6 +30,8 @@ #ifdef CONFIG_ARCH_S3C64XX #define filter_fn pl08x_filter_id +#elif defined(CONFIG_ARCH_S3C24XX) +#define filter_fn s3c24xx_dma_filter #else #define filter_fn NULL #endif diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index 0ff4bbe23af3..de6c321b8b68 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -392,8 +392,6 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); unsigned long irqs; int ret = 0; - struct s3c_dma_params *dma_data = - snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); pr_debug("Entered %s\n", __func__); @@ -424,13 +422,6 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, local_irq_restore(irqs); - /* - * Load the next buffer to DMA to meet the reqirement - * of the auto reload mechanism of S3C24XX. - * This call won't bother S3C64XX. - */ - s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); - break; case SNDRV_PCM_TRIGGER_STOP: @@ -644,12 +635,6 @@ int s3c_i2sv2_probe(struct snd_soc_dai *dai, /* record our i2s structure for later use in the callbacks */ snd_soc_dai_set_drvdata(dai, i2s); - i2s->regs = ioremap(base, 0x100); - if (i2s->regs == NULL) { - dev_err(dev, "cannot ioremap registers\n"); - return -ENXIO; - } - i2s->iis_pclk = clk_get(dev, "iis"); if (IS_ERR(i2s->iis_pclk)) { dev_err(dev, "failed to get iis_clock\n"); @@ -729,7 +714,7 @@ int s3c_i2sv2_register_component(struct device *dev, int id, struct snd_soc_component_driver *cmp_drv, struct snd_soc_dai_driver *dai_drv) { - struct snd_soc_dai_ops *ops = dai_drv->ops; + struct snd_soc_dai_ops *ops = (struct snd_soc_dai_ops *)dai_drv->ops; ops->trigger = s3c2412_i2s_trigger; if (!ops->hw_params) diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 08c059be9104..d9d27cc0657c 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -33,25 +33,19 @@ #include "regs-i2s-v2.h" #include "s3c2412-i2s.h" -static struct s3c_dma_client s3c2412_dma_client_out = { - .name = "I2S PCM Stereo out" -}; - -static struct s3c_dma_client s3c2412_dma_client_in = { - .name = "I2S PCM Stereo in" -}; - static struct s3c_dma_params s3c2412_i2s_pcm_stereo_out = { - .client = &s3c2412_dma_client_out, + .client = + (struct s3c_dma_client *)&s3c2412_i2s_pcm_stereo_out, .channel = DMACH_I2S_OUT, - .dma_addr = S3C2410_PA_IIS + S3C2412_IISTXD, + .ch_name = "tx", .dma_size = 4, }; static struct s3c_dma_params s3c2412_i2s_pcm_stereo_in = { - .client = &s3c2412_dma_client_in, + .client = + (struct s3c_dma_client *)&s3c2412_i2s_pcm_stereo_in, .channel = DMACH_I2S_IN, - .dma_addr = S3C2410_PA_IIS + S3C2412_IISRXD, + .ch_name = "rx", .dma_size = 4, }; @@ -63,6 +57,9 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) pr_debug("Entered %s\n", __func__); + samsung_asoc_init_dma_data(dai, &s3c2412_i2s_pcm_stereo_out, + &s3c2412_i2s_pcm_stereo_in); + ret = s3c_i2sv2_probe(dai, &s3c2412_i2s, S3C2410_PA_IIS); if (ret) return ret; @@ -70,10 +67,9 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) s3c2412_i2s.dma_capture = &s3c2412_i2s_pcm_stereo_in; s3c2412_i2s.dma_playback = &s3c2412_i2s_pcm_stereo_out; - s3c2412_i2s.iis_cclk = clk_get(dai->dev, "i2sclk"); + s3c2412_i2s.iis_cclk = devm_clk_get(dai->dev, "i2sclk"); if (IS_ERR(s3c2412_i2s.iis_cclk)) { pr_err("failed to get i2sclk clock\n"); - iounmap(s3c2412_i2s.regs); return PTR_ERR(s3c2412_i2s.iis_cclk); } @@ -94,8 +90,6 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) static int s3c2412_i2s_remove(struct snd_soc_dai *dai) { clk_disable(s3c2412_i2s.iis_cclk); - clk_put(s3c2412_i2s.iis_cclk); - iounmap(s3c2412_i2s.regs); return 0; } @@ -105,18 +99,10 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct s3c_i2sv2_info *i2s = snd_soc_dai_get_drvdata(cpu_dai); - struct s3c_dma_params *dma_data; u32 iismod; pr_debug("Entered %s\n", __func__); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = i2s->dma_playback; - else - dma_data = i2s->dma_capture; - - snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); - iismod = readl(i2s->regs + S3C2412_IISMOD); pr_debug("%s: r: IISMOD: %x\n", __func__, iismod); @@ -169,6 +155,19 @@ static const struct snd_soc_component_driver s3c2412_i2s_component = { static int s3c2412_iis_dev_probe(struct platform_device *pdev) { int ret = 0; + struct resource *res; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(&pdev->dev, "Can't get IO resource.\n"); + return -ENOENT; + } + s3c2412_i2s.regs = devm_ioremap_resource(&pdev->dev, res); + if (s3c2412_i2s.regs == NULL) + return -ENXIO; + + s3c2412_i2s_pcm_stereo_out.dma_addr = res->start + S3C2412_IISTXD; + s3c2412_i2s_pcm_stereo_in.dma_addr = res->start + S3C2412_IISRXD; ret = s3c_i2sv2_register_component(&pdev->dev, -1, &s3c2412_i2s_component, diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index 9aba9fb7df0e..6f3ee87da722 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -31,25 +31,19 @@ #include "dma.h" #include "s3c24xx-i2s.h" -static struct s3c_dma_client s3c24xx_dma_client_out = { - .name = "I2S PCM Stereo out" -}; - -static struct s3c_dma_client s3c24xx_dma_client_in = { - .name = "I2S PCM Stereo in" -}; - static struct s3c_dma_params s3c24xx_i2s_pcm_stereo_out = { - .client = &s3c24xx_dma_client_out, + .client = + (struct s3c_dma_client *)&s3c24xx_i2s_pcm_stereo_out, .channel = DMACH_I2S_OUT, - .dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO, + .ch_name = "tx", .dma_size = 2, }; static struct s3c_dma_params s3c24xx_i2s_pcm_stereo_in = { - .client = &s3c24xx_dma_client_in, + .client = + (struct s3c_dma_client *)&s3c24xx_i2s_pcm_stereo_in, .channel = DMACH_I2S_IN, - .dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO, + .ch_name = "rx", .dma_size = 2, }; @@ -231,18 +225,12 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c_dma_params *dma_data; + struct snd_dmaengine_dai_dma_data *dma_data; u32 iismod; pr_debug("Entered %s\n", __func__); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = &s3c24xx_i2s_pcm_stereo_out; - else - dma_data = &s3c24xx_i2s_pcm_stereo_in; - - snd_soc_dai_set_dma_data(rtd->cpu_dai, substream, dma_data); + dma_data = snd_soc_dai_get_dma_data(dai, substream); /* Working copies of register */ iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); @@ -251,11 +239,11 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, switch (params_width(params)) { case 8: iismod &= ~S3C2410_IISMOD_16BIT; - dma_data->dma_size = 1; + dma_data->addr_width = 1; break; case 16: iismod |= S3C2410_IISMOD_16BIT; - dma_data->dma_size = 2; + dma_data->addr_width = 2; break; default: return -EINVAL; @@ -270,8 +258,6 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { int ret = 0; - struct s3c_dma_params *dma_data = - snd_soc_dai_get_dma_data(dai, substream); pr_debug("Entered %s\n", __func__); @@ -290,7 +276,6 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, else s3c24xx_snd_txctrl(1); - s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: @@ -380,14 +365,12 @@ static int s3c24xx_i2s_probe(struct snd_soc_dai *dai) { pr_debug("Entered %s\n", __func__); - s3c24xx_i2s.regs = ioremap(S3C2410_PA_IIS, 0x100); - if (s3c24xx_i2s.regs == NULL) - return -ENXIO; + samsung_asoc_init_dma_data(dai, &s3c24xx_i2s_pcm_stereo_out, + &s3c24xx_i2s_pcm_stereo_in); - s3c24xx_i2s.iis_clk = clk_get(dai->dev, "iis"); + s3c24xx_i2s.iis_clk = devm_clk_get(dai->dev, "iis"); if (IS_ERR(s3c24xx_i2s.iis_clk)) { pr_err("failed to get iis_clock\n"); - iounmap(s3c24xx_i2s.regs); return PTR_ERR(s3c24xx_i2s.iis_clk); } clk_enable(s3c24xx_i2s.iis_clk); @@ -474,6 +457,19 @@ static const struct snd_soc_component_driver s3c24xx_i2s_component = { static int s3c24xx_iis_dev_probe(struct platform_device *pdev) { int ret = 0; + struct resource *res; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(&pdev->dev, "Can't get IO resource.\n"); + return -ENOENT; + } + s3c24xx_i2s.regs = devm_ioremap_resource(&pdev->dev, res); + if (s3c24xx_i2s.regs == NULL) + return -ENXIO; + + s3c24xx_i2s_pcm_stereo_out.dma_addr = res->start + S3C2410_IISFIFO; + s3c24xx_i2s_pcm_stereo_in.dma_addr = res->start + S3C2410_IISFIFO; ret = devm_snd_soc_register_component(&pdev->dev, &s3c24xx_i2s_component, &s3c24xx_i2s_dai, 1); -- cgit v1.2.3 From ae602456e83c9242f752f8176ebe658e37d4f90d Mon Sep 17 00:00:00 2001 From: Vasily Khoruzhick Date: Mon, 23 Jun 2014 23:24:05 +0300 Subject: ASoC: samsung: drop support for legacy S3C24XX DMA API Signed-off-by: Vasily Khoruzhick Signed-off-by: Mark Brown --- sound/soc/samsung/Kconfig | 3 - sound/soc/samsung/Makefile | 2 - sound/soc/samsung/dma.c | 454 --------------------------------------------- 3 files changed, 459 deletions(-) delete mode 100644 sound/soc/samsung/dma.c (limited to 'sound') diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index e88e598fbe0c..3be49cd2e984 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -13,9 +13,6 @@ config SND_SOC_SAMSUNG config SND_S3C_DMA tristate -config SND_S3C_DMA_LEGACY - tristate - config SND_S3C24XX_I2S tristate diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 6d0212ba571c..6469199acea5 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -1,6 +1,5 @@ # S3c24XX Platform Support snd-soc-s3c-dma-objs := dmaengine.o -snd-soc-s3c-dma-legacy-objs := dma.o snd-soc-idma-objs := idma.o snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o @@ -11,7 +10,6 @@ snd-soc-pcm-objs := pcm.o snd-soc-i2s-objs := i2s.o obj-$(CONFIG_SND_S3C_DMA) += snd-soc-s3c-dma.o -obj-$(CONFIG_SND_S3C_DMA_LEGACY) += snd-soc-s3c-dma-legacy.o obj-$(CONFIG_SND_S3C24XX_I2S) += snd-soc-s3c24xx-i2s.o obj-$(CONFIG_SND_SAMSUNG_AC97) += snd-soc-ac97.o obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c deleted file mode 100644 index d9dc7bcc0336..000000000000 --- a/sound/soc/samsung/dma.c +++ /dev/null @@ -1,454 +0,0 @@ -/* - * dma.c -- ALSA Soc Audio Layer - * - * (c) 2006 Wolfson Microelectronics PLC. - * Graeme Gregory graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com - * - * Copyright 2004-2005 Simtec Electronics - * http://armlinux.simtec.co.uk/ - * Ben Dooks - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ - -#include -#include -#include - -#include -#include - -#include -#include -#include - -#include "dma.h" - -#define ST_RUNNING (1<<0) -#define ST_OPENED (1<<1) - -static const struct snd_pcm_hardware dma_hardware = { - .info = SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID, - .buffer_bytes_max = 128*1024, - .period_bytes_min = PAGE_SIZE, - .period_bytes_max = PAGE_SIZE*2, - .periods_min = 2, - .periods_max = 128, - .fifo_size = 32, -}; - -struct runtime_data { - spinlock_t lock; - int state; - unsigned int dma_loaded; - unsigned int dma_period; - dma_addr_t dma_start; - dma_addr_t dma_pos; - dma_addr_t dma_end; - struct s3c_dma_params *params; -}; - -static void audio_buffdone(void *data); - -/* dma_enqueue - * - * place a dma buffer onto the queue for the dma system - * to handle. - */ -static void dma_enqueue(struct snd_pcm_substream *substream) -{ - struct runtime_data *prtd = substream->runtime->private_data; - dma_addr_t pos = prtd->dma_pos; - unsigned int limit; - struct samsung_dma_prep dma_info; - - pr_debug("Entered %s\n", __func__); - - limit = (prtd->dma_end - prtd->dma_start) / prtd->dma_period; - - pr_debug("%s: loaded %d, limit %d\n", - __func__, prtd->dma_loaded, limit); - - dma_info.cap = (samsung_dma_has_circular() ? DMA_CYCLIC : DMA_SLAVE); - dma_info.direction = - (substream->stream == SNDRV_PCM_STREAM_PLAYBACK - ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM); - dma_info.fp = audio_buffdone; - dma_info.fp_param = substream; - dma_info.period = prtd->dma_period; - dma_info.len = prtd->dma_period*limit; - - if (dma_info.cap == DMA_CYCLIC) { - dma_info.buf = pos; - prtd->params->ops->prepare(prtd->params->ch, &dma_info); - prtd->dma_loaded += limit; - return; - } - - while (prtd->dma_loaded < limit) { - pr_debug("dma_loaded: %d\n", prtd->dma_loaded); - - if ((pos + dma_info.period) > prtd->dma_end) { - dma_info.period = prtd->dma_end - pos; - pr_debug("%s: corrected dma len %ld\n", - __func__, dma_info.period); - } - - dma_info.buf = pos; - prtd->params->ops->prepare(prtd->params->ch, &dma_info); - - prtd->dma_loaded++; - pos += prtd->dma_period; - if (pos >= prtd->dma_end) - pos = prtd->dma_start; - } - - prtd->dma_pos = pos; -} - -static void audio_buffdone(void *data) -{ - struct snd_pcm_substream *substream = data; - struct runtime_data *prtd = substream->runtime->private_data; - - pr_debug("Entered %s\n", __func__); - - if (prtd->state & ST_RUNNING) { - prtd->dma_pos += prtd->dma_period; - if (prtd->dma_pos >= prtd->dma_end) - prtd->dma_pos = prtd->dma_start; - - if (substream) - snd_pcm_period_elapsed(substream); - - spin_lock(&prtd->lock); - if (!samsung_dma_has_circular()) { - prtd->dma_loaded--; - dma_enqueue(substream); - } - spin_unlock(&prtd->lock); - } -} - -static int dma_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct runtime_data *prtd = runtime->private_data; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - unsigned long totbytes = params_buffer_bytes(params); - struct s3c_dma_params *dma = - snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - struct samsung_dma_req req; - struct samsung_dma_config config; - - pr_debug("Entered %s\n", __func__); - - /* return if this is a bufferless transfer e.g. - * codec <--> BT codec or GSM modem -- lg FIXME */ - if (!dma) - return 0; - - /* this may get called several times by oss emulation - * with different params -HW */ - if (prtd->params == NULL) { - /* prepare DMA */ - prtd->params = dma; - - pr_debug("params %p, client %p, channel %d\n", prtd->params, - prtd->params->client, prtd->params->channel); - - prtd->params->ops = samsung_dma_get_ops(); - - req.cap = (samsung_dma_has_circular() ? - DMA_CYCLIC : DMA_SLAVE); - req.client = prtd->params->client; - config.direction = - (substream->stream == SNDRV_PCM_STREAM_PLAYBACK - ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM); - config.width = prtd->params->dma_size; - config.fifo = prtd->params->dma_addr; - prtd->params->ch = prtd->params->ops->request( - prtd->params->channel, &req, rtd->cpu_dai->dev, - prtd->params->ch_name); - if (!prtd->params->ch) { - pr_err("Failed to allocate DMA channel\n"); - return -ENXIO; - } - prtd->params->ops->config(prtd->params->ch, &config); - } - - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - - runtime->dma_bytes = totbytes; - - spin_lock_irq(&prtd->lock); - prtd->dma_loaded = 0; - prtd->dma_period = params_period_bytes(params); - prtd->dma_start = runtime->dma_addr; - prtd->dma_pos = prtd->dma_start; - prtd->dma_end = prtd->dma_start + totbytes; - spin_unlock_irq(&prtd->lock); - - return 0; -} - -static int dma_hw_free(struct snd_pcm_substream *substream) -{ - struct runtime_data *prtd = substream->runtime->private_data; - - pr_debug("Entered %s\n", __func__); - - snd_pcm_set_runtime_buffer(substream, NULL); - - if (prtd->params) { - prtd->params->ops->flush(prtd->params->ch); - prtd->params->ops->release(prtd->params->ch, - prtd->params->client); - prtd->params = NULL; - } - - return 0; -} - -static int dma_prepare(struct snd_pcm_substream *substream) -{ - struct runtime_data *prtd = substream->runtime->private_data; - int ret = 0; - - pr_debug("Entered %s\n", __func__); - - /* return if this is a bufferless transfer e.g. - * codec <--> BT codec or GSM modem -- lg FIXME */ - if (!prtd->params) - return 0; - - /* flush the DMA channel */ - prtd->params->ops->flush(prtd->params->ch); - - prtd->dma_loaded = 0; - prtd->dma_pos = prtd->dma_start; - - /* enqueue dma buffers */ - dma_enqueue(substream); - - return ret; -} - -static int dma_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct runtime_data *prtd = substream->runtime->private_data; - int ret = 0; - - pr_debug("Entered %s\n", __func__); - - spin_lock(&prtd->lock); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - prtd->state |= ST_RUNNING; - prtd->params->ops->trigger(prtd->params->ch); - break; - - case SNDRV_PCM_TRIGGER_STOP: - prtd->state &= ~ST_RUNNING; - prtd->params->ops->stop(prtd->params->ch); - break; - - default: - ret = -EINVAL; - break; - } - - spin_unlock(&prtd->lock); - - return ret; -} - -static snd_pcm_uframes_t -dma_pointer(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct runtime_data *prtd = runtime->private_data; - unsigned long res; - - pr_debug("Entered %s\n", __func__); - - res = prtd->dma_pos - prtd->dma_start; - - pr_debug("Pointer offset: %lu\n", res); - - /* we seem to be getting the odd error from the pcm library due - * to out-of-bounds pointers. this is maybe due to the dma engine - * not having loaded the new values for the channel before being - * called... (todo - fix ) - */ - - if (res >= snd_pcm_lib_buffer_bytes(substream)) { - if (res == snd_pcm_lib_buffer_bytes(substream)) - res = 0; - } - - return bytes_to_frames(substream->runtime, res); -} - -static int dma_open(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct runtime_data *prtd; - - pr_debug("Entered %s\n", __func__); - - snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); - snd_soc_set_runtime_hwparams(substream, &dma_hardware); - - prtd = kzalloc(sizeof(struct runtime_data), GFP_KERNEL); - if (prtd == NULL) - return -ENOMEM; - - spin_lock_init(&prtd->lock); - - runtime->private_data = prtd; - return 0; -} - -static int dma_close(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct runtime_data *prtd = runtime->private_data; - - pr_debug("Entered %s\n", __func__); - - if (!prtd) - pr_debug("dma_close called with prtd == NULL\n"); - - kfree(prtd); - - return 0; -} - -static int dma_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - pr_debug("Entered %s\n", __func__); - - return dma_mmap_writecombine(substream->pcm->card->dev, vma, - runtime->dma_area, - runtime->dma_addr, - runtime->dma_bytes); -} - -static struct snd_pcm_ops dma_ops = { - .open = dma_open, - .close = dma_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = dma_hw_params, - .hw_free = dma_hw_free, - .prepare = dma_prepare, - .trigger = dma_trigger, - .pointer = dma_pointer, - .mmap = dma_mmap, -}; - -static int preallocate_dma_buffer(struct snd_pcm *pcm, int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = dma_hardware.buffer_bytes_max; - - pr_debug("Entered %s\n", __func__); - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - buf->area = dma_alloc_writecombine(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - if (!buf->area) - return -ENOMEM; - buf->bytes = size; - return 0; -} - -static void dma_free_dma_buffers(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - pr_debug("Entered %s\n", __func__); - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - - dma_free_writecombine(pcm->card->dev, buf->bytes, - buf->area, buf->addr); - buf->area = NULL; - } -} - -static int dma_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_card *card = rtd->card->snd_card; - struct snd_pcm *pcm = rtd->pcm; - int ret; - - pr_debug("Entered %s\n", __func__); - - ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); - if (ret) - return ret; - - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { - ret = preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } - - if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - ret = preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } -out: - return ret; -} - -static struct snd_soc_platform_driver samsung_asoc_platform = { - .ops = &dma_ops, - .pcm_new = dma_new, - .pcm_free = dma_free_dma_buffers, -}; - -void samsung_asoc_init_dma_data(struct snd_soc_dai *dai, - struct s3c_dma_params *playback, - struct s3c_dma_params *capture) -{ - snd_soc_dai_init_dma_data(dai, playback, capture); -} -EXPORT_SYMBOL_GPL(samsung_asoc_init_dma_data); - -int samsung_asoc_dma_platform_register(struct device *dev) -{ - return devm_snd_soc_register_platform(dev, &samsung_asoc_platform); -} -EXPORT_SYMBOL_GPL(samsung_asoc_dma_platform_register); - -MODULE_AUTHOR("Ben Dooks, "); -MODULE_DESCRIPTION("Samsung ASoC DMA Driver"); -MODULE_LICENSE("GPL"); -- cgit v1.2.3 From 77ea6bf777b08ef1c5a44b6c1548cff449a96de7 Mon Sep 17 00:00:00 2001 From: Vasily Khoruzhick Date: Mon, 23 Jun 2014 23:24:06 +0300 Subject: ASoC: samsung: s3c2412-i2s: Move to clk_prepare_enable/clk_disable_unprepare Use clk_prepare_enable/clk_disable_unprepare to make the driver work properly with common clock framework. Signed-off-by: Vasily Khoruzhick Signed-off-by: Mark Brown --- sound/soc/samsung/s3c2412-i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index d9d27cc0657c..d6bc6dc0aafb 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -76,7 +76,7 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) /* Set MPLL as the source for IIS CLK */ clk_set_parent(s3c2412_i2s.iis_cclk, clk_get(NULL, "mpll")); - clk_enable(s3c2412_i2s.iis_cclk); + clk_prepare_enable(s3c2412_i2s.iis_cclk); s3c2412_i2s.iis_cclk = s3c2412_i2s.iis_pclk; @@ -89,7 +89,7 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) static int s3c2412_i2s_remove(struct snd_soc_dai *dai) { - clk_disable(s3c2412_i2s.iis_cclk); + clk_disable_unprepare(s3c2412_i2s.iis_cclk); return 0; } -- cgit v1.2.3 From c1ae59c7bd75e5f861d913a238f659b38ff4a07b Mon Sep 17 00:00:00 2001 From: Vasily Khoruzhick Date: Mon, 23 Jun 2014 23:24:07 +0300 Subject: ASoC: samsung: s3c24xx-i2s: Move to clk_prepare_enable/clk_disable_unprepare Use clk_prepare_enable/clk_disable_unprepare to make the driver work properly with common clock framework. Signed-off-by: Vasily Khoruzhick Signed-off-by: Mark Brown --- sound/soc/samsung/s3c24xx-i2s.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index 6f3ee87da722..e8b98528e356 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -373,7 +373,7 @@ static int s3c24xx_i2s_probe(struct snd_soc_dai *dai) pr_err("failed to get iis_clock\n"); return PTR_ERR(s3c24xx_i2s.iis_clk); } - clk_enable(s3c24xx_i2s.iis_clk); + clk_prepare_enable(s3c24xx_i2s.iis_clk); /* Configure the I2S pins (GPE0...GPE4) in correct mode */ s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), @@ -397,7 +397,7 @@ static int s3c24xx_i2s_suspend(struct snd_soc_dai *cpu_dai) s3c24xx_i2s.iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON); s3c24xx_i2s.iispsr = readl(s3c24xx_i2s.regs + S3C2410_IISPSR); - clk_disable(s3c24xx_i2s.iis_clk); + clk_disable_unprepare(s3c24xx_i2s.iis_clk); return 0; } @@ -405,7 +405,7 @@ static int s3c24xx_i2s_suspend(struct snd_soc_dai *cpu_dai) static int s3c24xx_i2s_resume(struct snd_soc_dai *cpu_dai) { pr_debug("Entered %s\n", __func__); - clk_enable(s3c24xx_i2s.iis_clk); + clk_prepare_enable(s3c24xx_i2s.iis_clk); writel(s3c24xx_i2s.iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); writel(s3c24xx_i2s.iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); -- cgit v1.2.3 From fb6f806967f6fe36fa40334e5551a5892d48f36f Mon Sep 17 00:00:00 2001 From: Paul Handrigan Date: Mon, 23 Jun 2014 17:29:53 -0500 Subject: ASoC: Add support for the CS4265 CODEC This patch adds support for the Cirrus Logic CS4265 Stereo I2C CODEC. Signed-off-by: Paul Handrigan Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs4265.c | 682 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs4265.h | 64 +++++ 4 files changed, 754 insertions(+) create mode 100644 sound/soc/codecs/cs4265.c create mode 100644 sound/soc/codecs/cs4265.h (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index cbfa1e18f651..3960a57124fa 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -47,6 +47,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS42L52 if I2C && INPUT select SND_SOC_CS42L56 if I2C && INPUT select SND_SOC_CS42L73 if I2C + select SND_SOC_CS4265 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI select SND_SOC_CS42XX8_I2C if I2C @@ -338,6 +339,11 @@ config SND_SOC_CS42L73 tristate "Cirrus Logic CS42L73 CODEC" depends on I2C +config SND_SOC_CS4265 + tristate "Cirrus Logic CS4265 CODEC" + depends on I2C + select REGMAP_I2C + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate "Cirrus Logic CS4270 CODEC" diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index be3377b8d73f..2e62bad23ddc 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -37,6 +37,7 @@ snd-soc-cs42l51-i2c-objs := cs42l51-i2c.o snd-soc-cs42l52-objs := cs42l52.o snd-soc-cs42l56-objs := cs42l56.o snd-soc-cs42l73-objs := cs42l73.o +snd-soc-cs4265-objs := cs4265.o snd-soc-cs4270-objs := cs4270.o snd-soc-cs4271-objs := cs4271.o snd-soc-cs42xx8-objs := cs42xx8.o @@ -202,6 +203,7 @@ obj-$(CONFIG_SND_SOC_CS42L51_I2C) += snd-soc-cs42l51-i2c.o obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o obj-$(CONFIG_SND_SOC_CS42L56) += snd-soc-cs42l56.o obj-$(CONFIG_SND_SOC_CS42L73) += snd-soc-cs42l73.o +obj-$(CONFIG_SND_SOC_CS4265) += snd-soc-cs4265.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o obj-$(CONFIG_SND_SOC_CS42XX8) += snd-soc-cs42xx8.o diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c new file mode 100644 index 000000000000..c9c04d2987c2 --- /dev/null +++ b/sound/soc/codecs/cs4265.c @@ -0,0 +1,682 @@ +/* + * cs4265.c -- CS4265 ALSA SoC audio driver + * + * Copyright 2014 Cirrus Logic, Inc. + * + * Author: Paul Handrigan + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "cs4265.h" + +struct cs4265_private { + struct device *dev; + struct regmap *regmap; + struct gpio_desc *reset_gpio; + u8 format; + u32 sysclk; +}; + +static const struct reg_default cs4265_reg_defaults[] = { + { CS4265_PWRCTL, 0x0F }, + { CS4265_DAC_CTL, 0x08 }, + { CS4265_ADC_CTL, 0x00 }, + { CS4265_MCLK_FREQ, 0x00 }, + { CS4265_SIG_SEL, 0x40 }, + { CS4265_CHB_PGA_CTL, 0x00 }, + { CS4265_CHA_PGA_CTL, 0x00 }, + { CS4265_ADC_CTL2, 0x19 }, + { CS4265_DAC_CHA_VOL, 0x00 }, + { CS4265_DAC_CHB_VOL, 0x00 }, + { CS4265_DAC_CTL2, 0xC0 }, + { CS4265_SPDIF_CTL1, 0x00 }, + { CS4265_SPDIF_CTL2, 0x00 }, + { CS4265_INT_MASK, 0x00 }, + { CS4265_STATUS_MODE_MSB, 0x00 }, + { CS4265_STATUS_MODE_LSB, 0x00 }, +}; + +static bool cs4265_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS4265_PWRCTL: + case CS4265_DAC_CTL: + case CS4265_ADC_CTL: + case CS4265_MCLK_FREQ: + case CS4265_SIG_SEL: + case CS4265_CHB_PGA_CTL: + case CS4265_CHA_PGA_CTL: + case CS4265_ADC_CTL2: + case CS4265_DAC_CHA_VOL: + case CS4265_DAC_CHB_VOL: + case CS4265_DAC_CTL2: + case CS4265_SPDIF_CTL1: + case CS4265_SPDIF_CTL2: + case CS4265_INT_MASK: + case CS4265_STATUS_MODE_MSB: + case CS4265_STATUS_MODE_LSB: + return true; + default: + return false; + } +} + +static bool cs4265_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS4265_INT_STATUS: + return 1; + default: + return 0; + } +} + +static DECLARE_TLV_DB_SCALE(pga_tlv, -1200, 50, 0); + +static DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 0); + +static const char * const digital_input_mux_text[] = { + "SDIN1", "SDIN2" +}; + +static SOC_ENUM_SINGLE_DECL(digital_input_mux_enum, CS4265_SIG_SEL, 7, + digital_input_mux_text); + +static const struct snd_kcontrol_new digital_input_mux = + SOC_DAPM_ENUM("Digital Input Mux", digital_input_mux_enum); + +static const char * const mic_linein_text[] = { + "MIC", "LINEIN" +}; + +static SOC_ENUM_SINGLE_DECL(mic_linein_enum, CS4265_ADC_CTL2, 0, + mic_linein_text); + +static const char * const cam_mode_text[] = { + "One Byte", "Two Byte" +}; + +static SOC_ENUM_SINGLE_DECL(cam_mode_enum, CS4265_SPDIF_CTL1, 5, + cam_mode_text); + +static const char * const cam_mono_stereo_text[] = { + "Stereo", "Mono" +}; + +static SOC_ENUM_SINGLE_DECL(spdif_mono_stereo_enum, CS4265_SPDIF_CTL2, 2, + cam_mono_stereo_text); + +static const char * const mono_select_text[] = { + "Channel A", "Channel B" +}; + +static SOC_ENUM_SINGLE_DECL(spdif_mono_select_enum, CS4265_SPDIF_CTL2, 0, + mono_select_text); + +static const struct snd_kcontrol_new mic_linein_mux = + SOC_DAPM_ENUM("ADC Input Capture Mux", mic_linein_enum); + +static const struct snd_kcontrol_new loopback_ctl = + SOC_DAPM_SINGLE("Switch", CS4265_SIG_SEL, 1, 1, 0); + +static const struct snd_kcontrol_new spdif_switch = + SOC_DAPM_SINGLE("Switch", SND_SOC_NOPM, 0, 0, 0); + +static const struct snd_kcontrol_new dac_switch = + SOC_DAPM_SINGLE("Switch", CS4265_PWRCTL, 1, 1, 0); + +static const struct snd_kcontrol_new cs4265_snd_controls[] = { + + SOC_DOUBLE_R_SX_TLV("PGA Volume", CS4265_CHA_PGA_CTL, + CS4265_CHB_PGA_CTL, 0, 0x28, 0x30, pga_tlv), + SOC_DOUBLE_R_TLV("DAC Volume", CS4265_DAC_CHA_VOL, + CS4265_DAC_CHB_VOL, 0, 0xFF, 1, dac_tlv), + SOC_SINGLE("De-emp 44.1kHz Switch", CS4265_DAC_CTL, 1, + 1, 0), + SOC_SINGLE("DAC INV Switch", CS4265_DAC_CTL2, 5, + 1, 0), + SOC_SINGLE("DAC Zero Cross Switch", CS4265_DAC_CTL2, 6, + 1, 0), + SOC_SINGLE("DAC Soft Ramp Switch", CS4265_DAC_CTL2, 7, + 1, 0), + SOC_SINGLE("ADC HPF Switch", CS4265_ADC_CTL, 1, + 1, 0), + SOC_SINGLE("ADC Zero Cross Switch", CS4265_ADC_CTL2, 3, + 1, 1), + SOC_SINGLE("ADC Soft Ramp Switch", CS4265_ADC_CTL2, 7, + 1, 0), + SOC_SINGLE("E to F Buffer Disable Switch", CS4265_SPDIF_CTL1, + 6, 1, 0), + SOC_ENUM("C Data Access", cam_mode_enum), + SOC_SINGLE("Validity Bit Control Switch", CS4265_SPDIF_CTL2, + 3, 1, 0), + SOC_ENUM("SPDIF Mono/Stereo", spdif_mono_stereo_enum), + SOC_SINGLE("MMTLR Data Switch", 0, + 1, 1, 0), + SOC_ENUM("Mono Channel Select", spdif_mono_select_enum), + SND_SOC_BYTES("C Data Buffer", CS4265_C_DATA_BUFF, 24), +}; + +static const struct snd_soc_dapm_widget cs4265_dapm_widgets[] = { + + SND_SOC_DAPM_INPUT("LINEINL"), + SND_SOC_DAPM_INPUT("LINEINR"), + SND_SOC_DAPM_INPUT("MICL"), + SND_SOC_DAPM_INPUT("MICR"), + + SND_SOC_DAPM_AIF_OUT("DOUT", NULL, 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("SPDIFOUT", NULL, 0, + SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_MUX("ADC Mux", SND_SOC_NOPM, 0, 0, &mic_linein_mux), + + SND_SOC_DAPM_ADC("ADC", NULL, CS4265_PWRCTL, 2, 1), + SND_SOC_DAPM_PGA("Pre-amp MIC", CS4265_PWRCTL, 3, + 1, NULL, 0), + + SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, + 0, 0, &digital_input_mux), + + SND_SOC_DAPM_MIXER("SDIN1 Input Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("SDIN2 Input Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("SPDIF Transmitter", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SWITCH("Loopback", SND_SOC_NOPM, 0, 0, + &loopback_ctl), + SND_SOC_DAPM_SWITCH("SPDIF", SND_SOC_NOPM, 0, 0, + &spdif_switch), + SND_SOC_DAPM_SWITCH("DAC", CS4265_PWRCTL, 1, 1, + &dac_switch), + + SND_SOC_DAPM_AIF_IN("DIN1", NULL, 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("DIN2", NULL, 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("TXIN", NULL, 0, + CS4265_SPDIF_CTL2, 5, 1), + + SND_SOC_DAPM_OUTPUT("LINEOUTL"), + SND_SOC_DAPM_OUTPUT("LINEOUTR"), + +}; + +static const struct snd_soc_dapm_route cs4265_audio_map[] = { + + {"DIN1", NULL, "DAI1 Playback"}, + {"DIN2", NULL, "DAI2 Playback"}, + {"SDIN1 Input Mixer", NULL, "DIN1"}, + {"SDIN2 Input Mixer", NULL, "DIN2"}, + {"Input Mux", "SDIN1", "SDIN1 Input Mixer"}, + {"Input Mux", "SDIN2", "SDIN2 Input Mixer"}, + {"DAC", "Switch", "Input Mux"}, + {"SPDIF", "Switch", "Input Mux"}, + {"LINEOUTL", NULL, "DAC"}, + {"LINEOUTR", NULL, "DAC"}, + {"SPDIFOUT", NULL, "SPDIF"}, + + {"ADC Mux", "LINEIN", "LINEINL"}, + {"ADC Mux", "LINEIN", "LINEINR"}, + {"ADC Mux", "MIC", "MICL"}, + {"ADC Mux", "MIC", "MICR"}, + {"ADC", NULL, "ADC Mux"}, + {"DOUT", NULL, "ADC"}, + {"DAI1 Capture", NULL, "DOUT"}, + {"DAI2 Capture", NULL, "DOUT"}, + + /* Loopback */ + {"Loopback", "Switch", "ADC"}, + {"DAC", NULL, "Loopback"}, +}; + +struct cs4265_clk_para { + u32 mclk; + u32 rate; + u8 fm_mode; /* values 1, 2, or 4 */ + u8 mclkdiv; +}; + +static const struct cs4265_clk_para clk_map_table[] = { + /*32k*/ + {8192000, 32000, 0, 0}, + {12288000, 32000, 0, 1}, + {16384000, 32000, 0, 2}, + {24576000, 32000, 0, 3}, + {32768000, 32000, 0, 4}, + + /*44.1k*/ + {11289600, 44100, 0, 0}, + {16934400, 44100, 0, 1}, + {22579200, 44100, 0, 2}, + {33868000, 44100, 0, 3}, + {45158400, 44100, 0, 4}, + + /*48k*/ + {12288000, 48000, 0, 0}, + {18432000, 48000, 0, 1}, + {24576000, 48000, 0, 2}, + {36864000, 48000, 0, 3}, + {49152000, 48000, 0, 4}, + + /*64k*/ + {8192000, 64000, 1, 0}, + {1228800, 64000, 1, 1}, + {1693440, 64000, 1, 2}, + {2457600, 64000, 1, 3}, + {3276800, 64000, 1, 4}, + + /* 88.2k */ + {11289600, 88200, 1, 0}, + {16934400, 88200, 1, 1}, + {22579200, 88200, 1, 2}, + {33868000, 88200, 1, 3}, + {45158400, 88200, 1, 4}, + + /* 96k */ + {12288000, 96000, 1, 0}, + {18432000, 96000, 1, 1}, + {24576000, 96000, 1, 2}, + {36864000, 96000, 1, 3}, + {49152000, 96000, 1, 4}, + + /* 128k */ + {8192000, 128000, 2, 0}, + {12288000, 128000, 2, 1}, + {16934400, 128000, 2, 2}, + {24576000, 128000, 2, 3}, + {32768000, 128000, 2, 4}, + + /* 176.4k */ + {11289600, 176400, 2, 0}, + {16934400, 176400, 2, 1}, + {22579200, 176400, 2, 2}, + {33868000, 176400, 2, 3}, + {49152000, 176400, 2, 4}, + + /* 192k */ + {12288000, 192000, 2, 0}, + {18432000, 192000, 2, 1}, + {24576000, 192000, 2, 2}, + {36864000, 192000, 2, 3}, + {49152000, 192000, 2, 4}, +}; + +static int cs4265_get_clk_index(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(clk_map_table); i++) { + if (clk_map_table[i].rate == rate && + clk_map_table[i].mclk == mclk) + return i; + } + return -EINVAL; +} + +static int cs4265_set_sysclk(struct snd_soc_dai *codec_dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs4265_private *cs4265 = snd_soc_codec_get_drvdata(codec); + int i; + + if (clk_id != 0) { + dev_err(codec->dev, "Invalid clk_id %d\n", clk_id); + return -EINVAL; + } + for (i = 0; i < ARRAY_SIZE(clk_map_table); i++) { + if (clk_map_table[i].mclk == freq) { + cs4265->sysclk = freq; + return 0; + } + } + cs4265->sysclk = 0; + dev_err(codec->dev, "Invalid freq parameter %d\n", freq); + return -EINVAL; +} + +static int cs4265_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs4265_private *cs4265 = snd_soc_codec_get_drvdata(codec); + u8 iface = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + snd_soc_update_bits(codec, CS4265_ADC_CTL, + CS4265_ADC_MASTER, + CS4265_ADC_MASTER); + break; + case SND_SOC_DAIFMT_CBS_CFS: + snd_soc_update_bits(codec, CS4265_ADC_CTL, + CS4265_ADC_MASTER, + 0); + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= SND_SOC_DAIFMT_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + iface |= SND_SOC_DAIFMT_RIGHT_J; + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= SND_SOC_DAIFMT_LEFT_J; + break; + default: + return -EINVAL; + } + + cs4265->format = iface; + return 0; +} + +static int cs4265_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + + if (mute) { + snd_soc_update_bits(codec, CS4265_DAC_CTL, + CS4265_DAC_CTL_MUTE, + CS4265_DAC_CTL_MUTE); + snd_soc_update_bits(codec, CS4265_SPDIF_CTL2, + CS4265_SPDIF_CTL2_MUTE, + CS4265_SPDIF_CTL2_MUTE); + } else { + snd_soc_update_bits(codec, CS4265_DAC_CTL, + CS4265_DAC_CTL_MUTE, + 0); + snd_soc_update_bits(codec, CS4265_SPDIF_CTL2, + CS4265_SPDIF_CTL2_MUTE, + 0); + } + return 0; +} + +static int cs4265_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs4265_private *cs4265 = snd_soc_codec_get_drvdata(codec); + int index; + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE && + ((cs4265->format & SND_SOC_DAIFMT_FORMAT_MASK) + == SND_SOC_DAIFMT_RIGHT_J)) + return -EINVAL; + + index = cs4265_get_clk_index(cs4265->sysclk, params_rate(params)); + if (index >= 0) { + snd_soc_update_bits(codec, CS4265_ADC_CTL, + CS4265_ADC_FM, clk_map_table[index].fm_mode); + snd_soc_update_bits(codec, CS4265_MCLK_FREQ, + CS4265_MCLK_FREQ_MASK, + clk_map_table[index].mclkdiv); + + } else { + dev_err(codec->dev, "can't get correct mclk\n"); + return -EINVAL; + } + + switch (cs4265->format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + snd_soc_update_bits(codec, CS4265_DAC_CTL, + CS4265_DAC_CTL_DIF, (1 << 4)); + snd_soc_update_bits(codec, CS4265_ADC_CTL, + CS4265_ADC_DIF, (1 << 4)); + snd_soc_update_bits(codec, CS4265_SPDIF_CTL2, + CS4265_SPDIF_CTL2_DIF, (1 << 6)); + break; + case SND_SOC_DAIFMT_RIGHT_J: + if (params_format(params) & SNDRV_PCM_FORMAT_S16_LE) { + snd_soc_update_bits(codec, CS4265_DAC_CTL, + CS4265_DAC_CTL_DIF, (1 << 5)); + snd_soc_update_bits(codec, CS4265_ADC_CTL, + CS4265_SPDIF_CTL2_DIF, (1 << 7)); + } else { + snd_soc_update_bits(codec, CS4265_DAC_CTL, + CS4265_DAC_CTL_DIF, (3 << 5)); + snd_soc_update_bits(codec, CS4265_ADC_CTL, + CS4265_SPDIF_CTL2_DIF, (1 << 7)); + } + break; + case SND_SOC_DAIFMT_LEFT_J: + snd_soc_update_bits(codec, CS4265_DAC_CTL, + CS4265_DAC_CTL_DIF, 0); + snd_soc_update_bits(codec, CS4265_ADC_CTL, + CS4265_ADC_DIF, 0); + snd_soc_update_bits(codec, CS4265_ADC_CTL, + CS4265_SPDIF_CTL2_DIF, (1 << 6)); + + break; + default: + return -EINVAL; + } + return 0; +} + +static int cs4265_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + snd_soc_update_bits(codec, CS4265_PWRCTL, + CS4265_PWRCTL_PDN, 0); + break; + case SND_SOC_BIAS_STANDBY: + snd_soc_update_bits(codec, CS4265_PWRCTL, + CS4265_PWRCTL_PDN, + CS4265_PWRCTL_PDN); + break; + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, CS4265_PWRCTL, + CS4265_PWRCTL_PDN, + CS4265_PWRCTL_PDN); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +#define CS4265_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | \ + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000) + +#define CS4265_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_U24_LE) + +static const struct snd_soc_dai_ops cs4265_ops = { + .hw_params = cs4265_pcm_hw_params, + .digital_mute = cs4265_digital_mute, + .set_fmt = cs4265_set_fmt, + .set_sysclk = cs4265_set_sysclk, +}; + +static struct snd_soc_dai_driver cs4265_dai[] = { + { + .name = "cs4265-dai1", + .playback = { + .stream_name = "DAI1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CS4265_RATES, + .formats = CS4265_FORMATS, + }, + .capture = { + .stream_name = "DAI1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS4265_RATES, + .formats = CS4265_FORMATS, + }, + .ops = &cs4265_ops, + }, + { + .name = "cs4265-dai2", + .playback = { + .stream_name = "DAI2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CS4265_RATES, + .formats = CS4265_FORMATS, + }, + .capture = { + .stream_name = "DAI2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS4265_RATES, + .formats = CS4265_FORMATS, + }, + .ops = &cs4265_ops, + }, +}; + +static const struct snd_soc_codec_driver soc_codec_cs4265 = { + .set_bias_level = cs4265_set_bias_level, + + .dapm_widgets = cs4265_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs4265_dapm_widgets), + .dapm_routes = cs4265_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs4265_audio_map), + + .controls = cs4265_snd_controls, + .num_controls = ARRAY_SIZE(cs4265_snd_controls), +}; + +static const struct regmap_config cs4265_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS4265_MAX_REGISTER, + .reg_defaults = cs4265_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs4265_reg_defaults), + .readable_reg = cs4265_readable_register, + .volatile_reg = cs4265_volatile_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int cs4265_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs4265_private *cs4265; + int ret = 0; + unsigned int devid = 0; + unsigned int reg; + + cs4265 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs4265_private), + GFP_KERNEL); + if (cs4265 == NULL) + return -ENOMEM; + cs4265->dev = &i2c_client->dev; + + cs4265->regmap = devm_regmap_init_i2c(i2c_client, &cs4265_regmap); + if (IS_ERR(cs4265->regmap)) { + ret = PTR_ERR(cs4265->regmap); + dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + + cs4265->reset_gpio = devm_gpiod_get(&i2c_client->dev, + "reset-gpios"); + if (IS_ERR(cs4265->reset_gpio)) { + ret = PTR_ERR(cs4265->reset_gpio); + if (ret != -ENOENT && ret != -ENOSYS) + return ret; + + cs4265->reset_gpio = NULL; + } else { + ret = gpiod_direction_output(cs4265->reset_gpio, 0); + if (ret) + return ret; + mdelay(1); + gpiod_set_value_cansleep(cs4265->reset_gpio, 1); + + } + + i2c_set_clientdata(i2c_client, cs4265); + + ret = regmap_read(cs4265->regmap, CS4265_CHIP_ID, ®); + devid = reg & CS4265_CHIP_ID_MASK; + if (devid != CS4265_CHIP_ID_VAL) { + ret = -ENODEV; + dev_err(&i2c_client->dev, + "CS4265 Device ID (%X). Expected %X\n", + devid, CS4265_CHIP_ID); + return ret; + } + dev_info(&i2c_client->dev, + "CS4265 Version %x\n", + reg & CS4265_REV_ID_MASK); + + regmap_write(cs4265->regmap, CS4265_PWRCTL, 0x0F); + + ret = snd_soc_register_codec(&i2c_client->dev, + &soc_codec_cs4265, cs4265_dai, + ARRAY_SIZE(cs4265_dai)); + return ret; +} + +static int cs4265_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct of_device_id cs4265_of_match[] = { + { .compatible = "cirrus,cs4265", }, + { } +}; +MODULE_DEVICE_TABLE(of, cs4265_of_match); + +static const struct i2c_device_id cs4265_id[] = { + { "cs4265", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, cs4265_id); + +static struct i2c_driver cs4265_i2c_driver = { + .driver = { + .name = "cs4265", + .owner = THIS_MODULE, + .of_match_table = cs4265_of_match, + }, + .id_table = cs4265_id, + .probe = cs4265_i2c_probe, + .remove = cs4265_i2c_remove, +}; + +module_i2c_driver(cs4265_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS4265 driver"); +MODULE_AUTHOR("Paul Handrigan, Cirrus Logic Inc, "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs4265.h b/sound/soc/codecs/cs4265.h new file mode 100644 index 000000000000..0a80a8dcec67 --- /dev/null +++ b/sound/soc/codecs/cs4265.h @@ -0,0 +1,64 @@ +/* + * cs4265.h -- CS4265 ALSA SoC audio driver + * + * Copyright 2014 Cirrus Logic, Inc. + * + * Author: Paul Handrigan + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#ifndef __CS4265_H__ +#define __CS4265_H__ + +#define CS4265_CHIP_ID 0x1 +#define CS4265_CHIP_ID_VAL 0xD0 +#define CS4265_CHIP_ID_MASK 0xF0 +#define CS4265_REV_ID_MASK 0x0F + +#define CS4265_PWRCTL 0x02 +#define CS4265_PWRCTL_PDN 1 + +#define CS4265_DAC_CTL 0x3 +#define CS4265_DAC_CTL_MUTE (1 << 2) +#define CS4265_DAC_CTL_DIF (3 << 4) + +#define CS4265_ADC_CTL 0x4 +#define CS4265_ADC_MASTER 1 +#define CS4265_ADC_DIF (1 << 4) +#define CS4265_ADC_FM (3 << 6) + +#define CS4265_MCLK_FREQ 0x5 +#define CS4265_MCLK_FREQ_MASK (7 << 4) + +#define CS4265_SIG_SEL 0x6 +#define CS4265_SIG_SEL_LOOP (1 << 1) + +#define CS4265_CHB_PGA_CTL 0x7 +#define CS4265_CHA_PGA_CTL 0x8 + +#define CS4265_ADC_CTL2 0x9 + +#define CS4265_DAC_CHA_VOL 0xA +#define CS4265_DAC_CHB_VOL 0xB + +#define CS4265_DAC_CTL2 0xC + +#define CS4265_INT_STATUS 0xD +#define CS4265_INT_MASK 0xE +#define CS4265_STATUS_MODE_MSB 0xF +#define CS4265_STATUS_MODE_LSB 0x10 + +#define CS4265_SPDIF_CTL1 0x11 + +#define CS4265_SPDIF_CTL2 0x12 +#define CS4265_SPDIF_CTL2_MUTE (1 << 4) +#define CS4265_SPDIF_CTL2_DIF (3 << 6) + +#define CS4265_C_DATA_BUFF 0x13 +#define CS4265_MAX_REGISTER 0x2A + +#endif -- cgit v1.2.3 From a5b37bf36f541dd06d876f9a3743abc224a25fd3 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 23 Jun 2014 16:29:37 +0300 Subject: ASoC: Intel: byt-max98090: Move MICBIAS widget to supply of Headset Mic Move "MICBIAS" as a supply widget to "Headset Mic" instead of keeping it between input pin "IN34" and "Headset Mic". Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/intel/byt-max98090.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/byt-max98090.c b/sound/soc/intel/byt-max98090.c index 5fc98c64a3f4..a3c5f02a60fc 100644 --- a/sound/soc/intel/byt-max98090.c +++ b/sound/soc/intel/byt-max98090.c @@ -39,8 +39,7 @@ static const struct snd_soc_dapm_widget byt_max98090_widgets[] = { static const struct snd_soc_dapm_route byt_max98090_audio_map[] = { {"IN34", NULL, "Headset Mic"}, - {"IN34", NULL, "MICBIAS"}, - {"MICBIAS", NULL, "Headset Mic"}, + {"Headset Mic", NULL, "MICBIAS"}, {"DMICL", NULL, "Int Mic"}, {"Headphone", NULL, "HPL"}, {"Headphone", NULL, "HPR"}, -- cgit v1.2.3 From 725a6dfd014d02b4ad1d00df364a6d4a15ff2ebb Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 23 Jun 2014 16:29:38 +0300 Subject: ASoC: Intel: byt-max98090: Fix mic detect GPIO polarity Mic detect GPIO is active low when headset microphone is detected. Found both by debugging and checking the schematics. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/intel/byt-max98090.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/intel/byt-max98090.c b/sound/soc/intel/byt-max98090.c index a3c5f02a60fc..3d06440278ec 100644 --- a/sound/soc/intel/byt-max98090.c +++ b/sound/soc/intel/byt-max98090.c @@ -83,6 +83,7 @@ static struct snd_soc_jack_gpio hs_jack_gpios[] = { { .name = "mic-gpio", .idx = 1, + .invert = 1, .report = SND_JACK_MICROPHONE | SND_JACK_LINEIN, .debounce_time = 200, }, -- cgit v1.2.3 From 6a0cdccad87a7032440f204c04a599e4bcc8656b Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 23 Jun 2014 16:29:39 +0300 Subject: ASoC: Intel: byt-max98090: Do not report SND_JACK_LINEIN Headset jack has only mono microphone input. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/intel/byt-max98090.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/byt-max98090.c b/sound/soc/intel/byt-max98090.c index 3d06440278ec..165ab6d6c4a3 100644 --- a/sound/soc/intel/byt-max98090.c +++ b/sound/soc/intel/byt-max98090.c @@ -84,7 +84,7 @@ static struct snd_soc_jack_gpio hs_jack_gpios[] = { .name = "mic-gpio", .idx = 1, .invert = 1, - .report = SND_JACK_MICROPHONE | SND_JACK_LINEIN, + .report = SND_JACK_MICROPHONE, .debounce_time = 200, }, }; -- cgit v1.2.3 From 24988992932612ba1c069ad597125dbce3e19621 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 23 Jun 2014 16:29:40 +0300 Subject: ASoC: Intel: byt-max98090: Fix jack type in order to report correctly Pass actual jack type bitmask to snd_soc_jack_new() in order to report also microphone detections and not only headphone. While at it change also jack name and pass also SND_JACK_LINEOUT type. Reported-by: Jin Yao Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/intel/byt-max98090.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/byt-max98090.c b/sound/soc/intel/byt-max98090.c index 165ab6d6c4a3..3e2e7152ae66 100644 --- a/sound/soc/intel/byt-max98090.c +++ b/sound/soc/intel/byt-max98090.c @@ -108,7 +108,8 @@ static int byt_max98090_init(struct snd_soc_pcm_runtime *runtime) } /* Enable jack detection */ - ret = snd_soc_jack_new(codec, "Headphone", SND_JACK_HEADPHONE, jack); + ret = snd_soc_jack_new(codec, "Headset", + SND_JACK_LINEOUT | SND_JACK_HEADSET, jack); if (ret) return ret; -- cgit v1.2.3 From ab6f7d0d932313cedfb1696e0536bdbe661cd9c9 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 23 Jun 2014 16:29:42 +0300 Subject: ASoC: Intel: byt-max98090: Do not enable MAX98090 microphone detection It turned out there is no need to enable microphone detection in MAX98090 codec. Headset microphone is anyway detected by a GPIO signal from another chip and headset button presses cannot be detected either because a signal needed for it is not connected. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/intel/byt-max98090.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/byt-max98090.c b/sound/soc/intel/byt-max98090.c index 3e2e7152ae66..5cfb41ec3fab 100644 --- a/sound/soc/intel/byt-max98090.c +++ b/sound/soc/intel/byt-max98090.c @@ -118,13 +118,9 @@ static int byt_max98090_init(struct snd_soc_pcm_runtime *runtime) if (ret) return ret; - ret = snd_soc_jack_add_gpiods(card->dev->parent, jack, - ARRAY_SIZE(hs_jack_gpios), - hs_jack_gpios); - if (ret) - return ret; - - return max98090_mic_detect(codec, jack); + return snd_soc_jack_add_gpiods(card->dev->parent, jack, + ARRAY_SIZE(hs_jack_gpios), + hs_jack_gpios); } static struct snd_soc_dai_link byt_max98090_dais[] = { -- cgit v1.2.3 From b29d7c5f7126d3c0b9984fcfd74ea82ec4fb3510 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 23 Jun 2014 16:29:41 +0300 Subject: ASoC: Intel: byt-max98090: Do not change speaker and DMIC with jack state Kernel should not enable/disable speakers and digital microphone whenever jack is inserted/removed. This is more use-case than kernel specific decision. For instance one may want to play VoIP ring tones using both speakers and headphone but play music only from one of them. Because of above reason remove "Ext Spk" and "Int Mic" update when jack state is changed. Also this update was illogical anyway: "Ext Spk" was enabled when jack was inserted and disabled when jack was removed. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/intel/byt-max98090.c | 8 -------- 1 file changed, 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/byt-max98090.c b/sound/soc/intel/byt-max98090.c index 5fc98c64a3f4..da0b2ee28dff 100644 --- a/sound/soc/intel/byt-max98090.c +++ b/sound/soc/intel/byt-max98090.c @@ -64,14 +64,6 @@ static struct snd_soc_jack_pin hs_jack_pins[] = { .pin = "Headset Mic", .mask = SND_JACK_MICROPHONE, }, - { - .pin = "Ext Spk", - .mask = SND_JACK_LINEOUT, - }, - { - .pin = "Int Mic", - .mask = SND_JACK_LINEIN, - }, }; static struct snd_soc_jack_gpio hs_jack_gpios[] = { -- cgit v1.2.3 From 41adf9056a26cf2e03139feefccf1c1d236a1c63 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Thu, 26 Jun 2014 12:17:24 +0530 Subject: ASoC: samsung: Remove unused variable from idma.c MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit ‘iiscon’ is not used in the function. Remove it. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/samsung/idma.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c index 8cc5770abb39..db6cefa18017 100644 --- a/sound/soc/samsung/idma.c +++ b/sound/soc/samsung/idma.c @@ -261,10 +261,9 @@ static int idma_mmap(struct snd_pcm_substream *substream, static irqreturn_t iis_irq(int irqno, void *dev_id) { struct idma_ctrl *prtd = (struct idma_ctrl *)dev_id; - u32 iiscon, iisahb, val, addr; + u32 iisahb, val, addr; iisahb = readl(idma.regs + I2SAHB); - iiscon = readl(idma.regs + I2SCON); val = (iisahb & AHB_LVL0INT) ? AHB_CLRLVL0INT : 0; -- cgit v1.2.3 From 0e0327f2ab722e9e1e55c225cb86e1b02336ec96 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 20 Jun 2014 15:28:57 +0530 Subject: ASoC: cs4270: Remove redundant OOM message Let memory subsystem handle the error logging. Signed-off-by: Sachin Kamat Acked-by: Paul Handrigan Signed-off-by: Mark Brown --- sound/soc/codecs/cs4270.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 9947a9583679..e6d4ff9fd992 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -664,10 +664,8 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, cs4270 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs4270_private), GFP_KERNEL); - if (!cs4270) { - dev_err(&i2c_client->dev, "could not allocate codec\n"); + if (!cs4270) return -ENOMEM; - } /* get the power supply regulators */ for (i = 0; i < ARRAY_SIZE(supply_names); i++) -- cgit v1.2.3 From 10d95ad48b4bf49b9853466654654eec2d9df031 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 20 Jun 2014 15:28:58 +0530 Subject: ASoC: cs42l73: Remove redundant OOM message Let memory subsystem handle the error logging. Signed-off-by: Sachin Kamat Acked-by: Paul Handrigan Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l73.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index ae3717992d56..e3b601624794 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1408,10 +1408,8 @@ static int cs42l73_i2c_probe(struct i2c_client *i2c_client, cs42l73 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs42l73_private), GFP_KERNEL); - if (!cs42l73) { - dev_err(&i2c_client->dev, "could not allocate codec\n"); + if (!cs42l73) return -ENOMEM; - } cs42l73->regmap = devm_regmap_init_i2c(i2c_client, &cs42l73_regmap); if (IS_ERR(cs42l73->regmap)) { -- cgit v1.2.3 From be81333415fa06440a6fc788053db3622caab194 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 20 Jun 2014 15:28:59 +0530 Subject: ASoC: sgtl5000: Remove redundant OOM message Let memory subsystem handle the error logging. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 3d39f0b5b4a8..249fadbdb42e 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -843,10 +843,8 @@ static int ldo_regulator_register(struct snd_soc_codec *codec, ldo = kzalloc(sizeof(struct ldo_regulator), GFP_KERNEL); - if (!ldo) { - dev_err(codec->dev, "failed to allocate ldo_regulator\n"); + if (!ldo) return -ENOMEM; - } ldo->desc.name = kstrdup(dev_name(codec->dev), GFP_KERNEL); if (!ldo->desc.name) { -- cgit v1.2.3 From 5c1573a342ad6452f2d7af284612baae5d532a0a Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 20 Jun 2014 15:29:00 +0530 Subject: ASoC: sta529: Remove redundant OOM message Let memory subsystem handle the error logging. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/sta529.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c index a40c4b0196a3..fcdf11026aff 100644 --- a/sound/soc/codecs/sta529.c +++ b/sound/soc/codecs/sta529.c @@ -380,10 +380,8 @@ static int sta529_i2c_probe(struct i2c_client *i2c, return -EINVAL; sta529 = devm_kzalloc(&i2c->dev, sizeof(struct sta529), GFP_KERNEL); - if (sta529 == NULL) { - dev_err(&i2c->dev, "Can not allocate memory\n"); + if (!sta529) return -ENOMEM; - } sta529->regmap = devm_regmap_init_i2c(i2c, &sta529_regmap); if (IS_ERR(sta529->regmap)) { -- cgit v1.2.3 From b1117f5294ed845aefa499d67ca4faf2493bdf99 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 20 Jun 2014 15:29:01 +0530 Subject: ASoC: tlv320aic3x: Remove redundant OOM message Let memory subsystem handle the error logging. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index e12fafbb1e09..446303198ffd 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1477,10 +1477,8 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, u32 value; aic3x = devm_kzalloc(&i2c->dev, sizeof(struct aic3x_priv), GFP_KERNEL); - if (aic3x == NULL) { - dev_err(&i2c->dev, "failed to create private data\n"); + if (!aic3x) return -ENOMEM; - } aic3x->regmap = devm_regmap_init_i2c(i2c, &aic3x_regmap); if (IS_ERR(aic3x->regmap)) { @@ -1498,10 +1496,8 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, } else if (np) { ai3x_setup = devm_kzalloc(&i2c->dev, sizeof(*ai3x_setup), GFP_KERNEL); - if (ai3x_setup == NULL) { - dev_err(&i2c->dev, "failed to create private data\n"); + if (!ai3x_setup) return -ENOMEM; - } ret = of_get_named_gpio(np, "gpio-reset", 0); if (ret >= 0) -- cgit v1.2.3 From 656e3435752a66810ee0f8bcb8e012e9e6f26a42 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 20 Jun 2014 15:29:02 +0530 Subject: ASoC: tpa6130a2: Remove redundant OOM message Let memory subsystem handle the error logging. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/tpa6130a2.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 8fc5a647453b..6fac9e034c48 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -381,10 +381,8 @@ static int tpa6130a2_probe(struct i2c_client *client, dev = &client->dev; data = devm_kzalloc(&client->dev, sizeof(*data), GFP_KERNEL); - if (data == NULL) { - dev_err(dev, "Can not allocate memory\n"); + if (!data) return -ENOMEM; - } if (pdata) { data->power_gpio = pdata->power_gpio; -- cgit v1.2.3 From 04cc41a809fb13867073ff76acf23e7df5c5b23d Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 20 Jun 2014 15:29:03 +0530 Subject: ASoC: twl4030: Remove redundant OOM message Let memory subsystem handle the error logging. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 69e12a311ba2..955df35c5ab6 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -2162,10 +2162,8 @@ static int twl4030_soc_probe(struct snd_soc_codec *codec) twl4030 = devm_kzalloc(codec->dev, sizeof(struct twl4030_priv), GFP_KERNEL); - if (twl4030 == NULL) { - dev_err(codec->dev, "Can not allocate memory\n"); + if (!twl4030) return -ENOMEM; - } snd_soc_codec_set_drvdata(codec, twl4030); /* Set the defaults, and power up the codec */ twl4030->sysclk = twl4030_audio_get_mclk() / 1000; -- cgit v1.2.3 From ac872d3d72773d1ac8824fa6c04430efc72e2a43 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 20 Jun 2014 15:29:04 +0530 Subject: ASoC: wl1273: Remove redundant OOM message Let memory subsystem handle the error logging. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/wl1273.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index 4ead0dc02b87..5d8ba779085b 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -461,10 +461,8 @@ static int wl1273_probe(struct snd_soc_codec *codec) } wl1273 = kzalloc(sizeof(struct wl1273_priv), GFP_KERNEL); - if (wl1273 == NULL) { - dev_err(codec->dev, "Cannot allocate memory.\n"); + if (!wl1273) return -ENOMEM; - } wl1273->mode = WL1273_MODE_BT; wl1273->core = *core; -- cgit v1.2.3 From 84cbc75f9a72d646701ec4fb4accdfdc5f6f3ea2 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 20 Jun 2014 15:29:05 +0530 Subject: ASoC: wm0010: Remove redundant OOM message Let memory subsystem handle the error logging. Signed-off-by: Sachin Kamat Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm0010.c | 12 +----------- 1 file changed, 1 insertion(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 71ce3159a62e..982467c1d56a 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -413,7 +413,6 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec) xfer = kzalloc(sizeof(*xfer), GFP_KERNEL); if (!xfer) { - dev_err(codec->dev, "Failed to allocate xfer\n"); ret = -ENOMEM; goto abort; } @@ -423,8 +422,6 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec) out = kzalloc(len, GFP_KERNEL | GFP_DMA); if (!out) { - dev_err(codec->dev, - "Failed to allocate RX buffer\n"); ret = -ENOMEM; goto abort1; } @@ -432,8 +429,6 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec) img = kzalloc(len, GFP_KERNEL | GFP_DMA); if (!img) { - dev_err(codec->dev, - "Failed to allocate image buffer\n"); ret = -ENOMEM; goto abort1; } @@ -526,14 +521,12 @@ static int wm0010_stage2_load(struct snd_soc_codec *codec) /* Copy to local buffer first as vmalloc causes problems for dma */ img = kzalloc(fw->size, GFP_KERNEL | GFP_DMA); if (!img) { - dev_err(codec->dev, "Failed to allocate image buffer\n"); ret = -ENOMEM; goto abort2; } out = kzalloc(fw->size, GFP_KERNEL | GFP_DMA); if (!out) { - dev_err(codec->dev, "Failed to allocate output buffer\n"); ret = -ENOMEM; goto abort1; } @@ -679,11 +672,8 @@ static int wm0010_boot(struct snd_soc_codec *codec) } img_swap = kzalloc(len, GFP_KERNEL | GFP_DMA); - if (!img_swap) { - dev_err(codec->dev, - "Failed to allocate image buffer\n"); + if (!img_swap) goto abort; - } /* We need to re-order for 0010 */ byte_swap_64((u64 *)&pll_rec, img_swap, len); -- cgit v1.2.3 From 611d7a7ba8678dfc2f597e8be6a1a6e4f2bdb8a4 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 20 Jun 2014 15:29:06 +0530 Subject: ASoC: wm1250-ev1: Remove redundant OOM message Let memory subsystem handle the error logging. Signed-off-by: Sachin Kamat Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm1250-ev1.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c index 6e6b93d4696e..8011f75fb6cb 100644 --- a/sound/soc/codecs/wm1250-ev1.c +++ b/sound/soc/codecs/wm1250-ev1.c @@ -164,7 +164,6 @@ static int wm1250_ev1_pdata(struct i2c_client *i2c) wm1250 = devm_kzalloc(&i2c->dev, sizeof(*wm1250), GFP_KERNEL); if (!wm1250) { - dev_err(&i2c->dev, "Unable to allocate private data\n"); ret = -ENOMEM; goto err; } -- cgit v1.2.3 From a0f62118b73483d0e3a980759b5df869e81d4477 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 20 Jun 2014 15:29:07 +0530 Subject: ASoC: wm2000: Remove redundant OOM message Let memory subsystem handle the error logging. Signed-off-by: Sachin Kamat Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm2000.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index a4c352cc3464..34ef65c52a7d 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -826,10 +826,8 @@ static int wm2000_i2c_probe(struct i2c_client *i2c, wm2000 = devm_kzalloc(&i2c->dev, sizeof(struct wm2000_priv), GFP_KERNEL); - if (wm2000 == NULL) { - dev_err(&i2c->dev, "Unable to allocate private data\n"); + if (!wm2000) return -ENOMEM; - } mutex_init(&wm2000->lock); -- cgit v1.2.3 From d931099bebdd057e77d47e5a990db6e769374bd5 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 20 Jun 2014 15:29:08 +0530 Subject: ASoC: wm8904: Remove redundant OOM message Let memory subsystem handle the error logging. Signed-off-by: Sachin Kamat Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8904.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index f7c549949c54..e2792980ecf2 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2013,12 +2013,8 @@ static void wm8904_handle_pdata(struct snd_soc_codec *codec) /* We need an array of texts for the enum API */ wm8904->drc_texts = kmalloc(sizeof(char *) * pdata->num_drc_cfgs, GFP_KERNEL); - if (!wm8904->drc_texts) { - dev_err(codec->dev, - "Failed to allocate %d DRC config texts\n", - pdata->num_drc_cfgs); + if (!wm8904->drc_texts) return; - } for (i = 0; i < pdata->num_drc_cfgs; i++) wm8904->drc_texts[i] = pdata->drc_cfgs[i].name; -- cgit v1.2.3 From 2cec4ff7f05e4f8721dc9c02d4ef4a8ad1ada7e8 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 20 Jun 2014 15:29:09 +0530 Subject: ASoC: wm8958: Remove redundant OOM message Let memory subsystem handle the error logging. Signed-off-by: Sachin Kamat Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8958-dsp2.c | 24 ++++-------------------- 1 file changed, 4 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index b2ebb104d879..0dada7f0105e 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -934,12 +934,8 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) /* We need an array of texts for the enum API */ wm8994->mbc_texts = kmalloc(sizeof(char *) * pdata->num_mbc_cfgs, GFP_KERNEL); - if (!wm8994->mbc_texts) { - dev_err(wm8994->hubs.codec->dev, - "Failed to allocate %d MBC config texts\n", - pdata->num_mbc_cfgs); + if (!wm8994->mbc_texts) return; - } for (i = 0; i < pdata->num_mbc_cfgs; i++) wm8994->mbc_texts[i] = pdata->mbc_cfgs[i].name; @@ -963,12 +959,8 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) /* We need an array of texts for the enum API */ wm8994->vss_texts = kmalloc(sizeof(char *) * pdata->num_vss_cfgs, GFP_KERNEL); - if (!wm8994->vss_texts) { - dev_err(wm8994->hubs.codec->dev, - "Failed to allocate %d VSS config texts\n", - pdata->num_vss_cfgs); + if (!wm8994->vss_texts) return; - } for (i = 0; i < pdata->num_vss_cfgs; i++) wm8994->vss_texts[i] = pdata->vss_cfgs[i].name; @@ -993,12 +985,8 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) /* We need an array of texts for the enum API */ wm8994->vss_hpf_texts = kmalloc(sizeof(char *) * pdata->num_vss_hpf_cfgs, GFP_KERNEL); - if (!wm8994->vss_hpf_texts) { - dev_err(wm8994->hubs.codec->dev, - "Failed to allocate %d VSS HPF config texts\n", - pdata->num_vss_hpf_cfgs); + if (!wm8994->vss_hpf_texts) return; - } for (i = 0; i < pdata->num_vss_hpf_cfgs; i++) wm8994->vss_hpf_texts[i] = pdata->vss_hpf_cfgs[i].name; @@ -1024,12 +1012,8 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) /* We need an array of texts for the enum API */ wm8994->enh_eq_texts = kmalloc(sizeof(char *) * pdata->num_enh_eq_cfgs, GFP_KERNEL); - if (!wm8994->enh_eq_texts) { - dev_err(wm8994->hubs.codec->dev, - "Failed to allocate %d enhanced EQ config texts\n", - pdata->num_enh_eq_cfgs); + if (!wm8994->enh_eq_texts) return; - } for (i = 0; i < pdata->num_enh_eq_cfgs; i++) wm8994->enh_eq_texts[i] = pdata->enh_eq_cfgs[i].name; -- cgit v1.2.3 From 549f66e0286bb800e024d80582f9c95c6f4f5947 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 20 Jun 2014 15:29:10 +0530 Subject: ASoC: wm8994: Remove redundant OOM message Let memory subsystem handle the error logging. Signed-off-by: Sachin Kamat Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 247b39013fba..621c442160e1 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3296,12 +3296,8 @@ static void wm8994_handle_pdata(struct wm8994_priv *wm8994) /* We need an array of texts for the enum API */ wm8994->drc_texts = devm_kzalloc(wm8994->hubs.codec->dev, sizeof(char *) * pdata->num_drc_cfgs, GFP_KERNEL); - if (!wm8994->drc_texts) { - dev_err(wm8994->hubs.codec->dev, - "Failed to allocate %d DRC config texts\n", - pdata->num_drc_cfgs); + if (!wm8994->drc_texts) return; - } for (i = 0; i < pdata->num_drc_cfgs; i++) wm8994->drc_texts[i] = pdata->drc_cfgs[i].name; -- cgit v1.2.3 From 0463585ce5186e617aa5a04fa9f3f27160810a21 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 20 Jun 2014 15:29:11 +0530 Subject: ASoC: wm9090: Remove redundant OOM message Let memory subsystem handle the error logging. Signed-off-by: Sachin Kamat Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm9090.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index 87934171f063..a13f0725611a 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -613,10 +613,8 @@ static int wm9090_i2c_probe(struct i2c_client *i2c, int ret; wm9090 = devm_kzalloc(&i2c->dev, sizeof(*wm9090), GFP_KERNEL); - if (wm9090 == NULL) { - dev_err(&i2c->dev, "Can not allocate memory\n"); + if (!wm9090) return -ENOMEM; - } wm9090->regmap = devm_regmap_init_i2c(i2c, &wm9090_regmap); if (IS_ERR(wm9090->regmap)) { -- cgit v1.2.3 From ba546687083b68f43c26fcce9771278faae2bcf5 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 20 Jun 2014 15:29:12 +0530 Subject: ASoC: wm_hubs: Remove redundant OOM message Let memory subsystem handle the error logging. Signed-off-by: Sachin Kamat Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_hubs.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 916817fe6632..374537d5e179 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -183,10 +183,8 @@ static void wm_hubs_dcs_cache_set(struct snd_soc_codec *codec, u16 dcs_cfg) return; cache = devm_kzalloc(codec->dev, sizeof(*cache), GFP_KERNEL); - if (!cache) { - dev_err(codec->dev, "Failed to allocate DCS cache entry\n"); + if (!cache) return; - } cache->left = snd_soc_read(codec, WM8993_LEFT_OUTPUT_VOLUME); cache->left &= WM8993_HPOUT1L_VOL_MASK; -- cgit v1.2.3 From 5bca396919aed5fbed61197019824bd3ff9f127d Mon Sep 17 00:00:00 2001 From: Fabian Frederick Date: Tue, 27 May 2014 08:57:55 +0200 Subject: ASoC: wm0010.c: add static to local variable Also add const to array text data bss dec hex filename 10946 2904 3528 17378 43e2 sound/soc/codecs/wm0010.o-before 10891 2840 3512 17243 435b sound/soc/codecs/wm0010.o-after Signed-off-by: Fabian Frederick Signed-off-by: Mark Brown --- sound/soc/codecs/wm0010.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 71ce3159a62e..65d7c79716a0 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -144,7 +144,7 @@ static const struct snd_soc_dapm_route wm0010_dapm_routes[] = { static const char *wm0010_state_to_str(enum wm0010_state state) { - const char *state_to_str[] = { + static const char * const state_to_str[] = { "Power off", "Out of reset", "Boot ROM", -- cgit v1.2.3 From 52b896cfef00289b5966b9b0e22b865511238216 Mon Sep 17 00:00:00 2001 From: Russell King Date: Thu, 26 Jun 2014 15:22:55 +0100 Subject: ASoC: kirkwood-i2s: provide helper KIRKWOOD_RECCTL_ENABLE_MASK definition Add a KIRKWOOD_RECCTL_ENABLE_MASK definition to complement the existing PLAYCTL definition, and make use of it where we wish to clear both enable bits. Signed-off-by: Russell King Tested-by: Andrew Lunn Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-i2s.c | 7 +++---- sound/soc/kirkwood/kirkwood.h | 3 +++ 2 files changed, 6 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 9f842222e798..55af6c8e4603 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -322,8 +322,7 @@ static int kirkwood_i2s_rec_trigger(struct snd_pcm_substream *substream, else ctl &= ~KIRKWOOD_RECCTL_I2S_EN; /* spdif */ - value = ctl & ~(KIRKWOOD_RECCTL_I2S_EN | - KIRKWOOD_RECCTL_SPDIF_EN); + value = ctl & ~KIRKWOOD_RECCTL_ENABLE_MASK; writel(value, priv->io + KIRKWOOD_RECCTL); /* enable interrupts */ @@ -347,7 +346,7 @@ static int kirkwood_i2s_rec_trigger(struct snd_pcm_substream *substream, /* disable all records */ value = readl(priv->io + KIRKWOOD_RECCTL); - value &= ~(KIRKWOOD_RECCTL_I2S_EN | KIRKWOOD_RECCTL_SPDIF_EN); + value &= ~KIRKWOOD_RECCTL_ENABLE_MASK; writel(value, priv->io + KIRKWOOD_RECCTL); break; @@ -411,7 +410,7 @@ static int kirkwood_i2s_init(struct kirkwood_dma_data *priv) writel(value, priv->io + KIRKWOOD_PLAYCTL); value = readl(priv->io + KIRKWOOD_RECCTL); - value &= ~(KIRKWOOD_RECCTL_I2S_EN | KIRKWOOD_RECCTL_SPDIF_EN); + value &= ~KIRKWOOD_RECCTL_ENABLE_MASK; writel(value, priv->io + KIRKWOOD_RECCTL); return 0; diff --git a/sound/soc/kirkwood/kirkwood.h b/sound/soc/kirkwood/kirkwood.h index bf23afbba1d7..ab21de090938 100644 --- a/sound/soc/kirkwood/kirkwood.h +++ b/sound/soc/kirkwood/kirkwood.h @@ -38,6 +38,9 @@ #define KIRKWOOD_RECCTL_SIZE_24 (1<<0) #define KIRKWOOD_RECCTL_SIZE_32 (0<<0) +#define KIRKWOOD_RECCTL_ENABLE_MASK (KIRKWOOD_RECCTL_SPDIF_EN | \ + KIRKWOOD_RECCTL_I2S_EN) + #define KIRKWOOD_REC_BUF_ADDR 0x1004 #define KIRKWOOD_REC_BUF_SIZE 0x1008 #define KIRKWOOD_REC_BYTE_COUNT 0x100C -- cgit v1.2.3 From 6772190632ebce6c5c6010d2bc77d5de866831b6 Mon Sep 17 00:00:00 2001 From: Russell King Date: Thu, 26 Jun 2014 15:23:00 +0100 Subject: ASoC: kirkwood-i2s: fix RECCTL masking Since we wish to disable capture inputs for some formats, we need to ensure that we clear the enable bits in our cached record control register. This seems to have been missed, resulting in the register only accumulating enable bits. Signed-off-by: Russell King Tested-by: Andrew Lunn Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-i2s.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 55af6c8e4603..ef1a164d8703 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -212,7 +212,8 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream, KIRKWOOD_PLAYCTL_SIZE_MASK); priv->ctl_play |= ctl_play; } else { - priv->ctl_rec &= ~KIRKWOOD_RECCTL_SIZE_MASK; + priv->ctl_rec &= ~(KIRKWOOD_RECCTL_ENABLE_MASK | + KIRKWOOD_RECCTL_SIZE_MASK); priv->ctl_rec |= ctl_rec; } -- cgit v1.2.3 From 2fbc38219c0af91afbeb3c9d97c62e1c7c74df61 Mon Sep 17 00:00:00 2001 From: Russell King Date: Thu, 26 Jun 2014 15:23:05 +0100 Subject: ASoC: kirkwood-i2s: fix mute handling The spec requires that the mute bits must be set while the channel is disabled. Ensure that this is the case by providing a helper which ensures that the appropriate mute bit is set while the enable bit is clear. Signed-off-by: Russell King Tested-by: Andrew Lunn Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-i2s.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index ef1a164d8703..b601ad680d7b 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -222,6 +222,15 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream, return 0; } +static unsigned kirkwood_i2s_play_mute(unsigned ctl) +{ + if (!(ctl & KIRKWOOD_PLAYCTL_I2S_EN)) + ctl |= KIRKWOOD_PLAYCTL_I2S_MUTE; + if (!(ctl & KIRKWOOD_PLAYCTL_SPDIF_EN)) + ctl |= KIRKWOOD_PLAYCTL_SPDIF_MUTE; + return ctl; +} + static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -257,7 +266,7 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, ctl &= ~KIRKWOOD_PLAYCTL_SPDIF_EN; /* i2s */ else ctl &= ~KIRKWOOD_PLAYCTL_I2S_EN; /* spdif */ - + ctl = kirkwood_i2s_play_mute(ctl); value = ctl & ~KIRKWOOD_PLAYCTL_ENABLE_MASK; writel(value, priv->io + KIRKWOOD_PLAYCTL); @@ -296,6 +305,7 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: ctl &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE | KIRKWOOD_PLAYCTL_SPDIF_MUTE); + ctl = kirkwood_i2s_play_mute(ctl); writel(ctl, priv->io + KIRKWOOD_PLAYCTL); break; -- cgit v1.2.3 From 4d2097e51795b760c392d3fbc6ca6b6f77c83419 Mon Sep 17 00:00:00 2001 From: Russell King Date: Thu, 26 Jun 2014 15:23:10 +0100 Subject: ASoC: kirkwood-i2s: fix pause handling some more We still see the occasional timeout waiting for busy to clear. As the spec is contradictory, and we know that the current implementation doesn't work, try an alternative interpretation from the spec. This one appears to work - I have yet to find any issue with it during my testing over several months. Signed-off-by: Russell King Tested-by: Andrew Lunn Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index b601ad680d7b..e98650c01eba 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -238,7 +238,7 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, uint32_t ctl, value; ctl = readl(priv->io + KIRKWOOD_PLAYCTL); - if (ctl & KIRKWOOD_PLAYCTL_PAUSE) { + if ((ctl & KIRKWOOD_PLAYCTL_ENABLE_MASK) == 0) { unsigned timeout = 5000; /* * The Armada510 spec says that if we enter pause mode, the -- cgit v1.2.3 From a622251c01b628cbbd1b02a877a6469303ec2b58 Mon Sep 17 00:00:00 2001 From: Russell King Date: Thu, 26 Jun 2014 15:23:15 +0100 Subject: ASoC: kirkwood: allow smaller audio periods and smaller number of periods There is no hardware restriction requiring a minimum of 8 periods, or a minimum of 2048 bytes in a period. Let's drop these values so that userspace has more flexibility in choosing these parameters. Signed-off-by: Russell King Tested-by: Andrew Lunn Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/kirkwood/kirkwood.h b/sound/soc/kirkwood/kirkwood.h index ab21de090938..90e32a781424 100644 --- a/sound/soc/kirkwood/kirkwood.h +++ b/sound/soc/kirkwood/kirkwood.h @@ -124,9 +124,9 @@ /* Theses values come from the marvell alsa driver */ /* need to find where they come from */ -#define KIRKWOOD_SND_MIN_PERIODS 8 +#define KIRKWOOD_SND_MIN_PERIODS 2 #define KIRKWOOD_SND_MAX_PERIODS 16 -#define KIRKWOOD_SND_MIN_PERIOD_BYTES 0x800 +#define KIRKWOOD_SND_MIN_PERIOD_BYTES 256 #define KIRKWOOD_SND_MAX_PERIOD_BYTES 0x8000 #define KIRKWOOD_SND_MAX_BUFFER_BYTES (KIRKWOOD_SND_MAX_PERIOD_BYTES \ * KIRKWOOD_SND_MAX_PERIODS) -- cgit v1.2.3 From 920ec4e595faf89f7db022a068a4729a4d2c48ae Mon Sep 17 00:00:00 2001 From: Russell King Date: Thu, 26 Jun 2014 15:23:20 +0100 Subject: ASoC: kirkwood: implement NO_PERIOD_WAKEUP support Permit ALSA to run without hardware interrupts from the audio interface. Instead, ALSA will use a kernel timer to decide when to check the buffer state, resulting in a lighter workload for the CPU. Signed-off-by: Russell King Tested-by: Andrew Lunn Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-dma.c | 11 ++++++----- sound/soc/kirkwood/kirkwood-i2s.c | 9 ++++++--- 2 files changed, 12 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index aac22fccdcdc..4cf2245950d7 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -28,11 +28,12 @@ static struct kirkwood_dma_data *kirkwood_priv(struct snd_pcm_substream *subs) } static struct snd_pcm_hardware kirkwood_dma_snd_hw = { - .info = (SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_PAUSE), + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, .buffer_bytes_max = KIRKWOOD_SND_MAX_BUFFER_BYTES, .period_bytes_min = KIRKWOOD_SND_MIN_PERIOD_BYTES, .period_bytes_max = KIRKWOOD_SND_MAX_PERIOD_BYTES, diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index e98650c01eba..0704cd6d2314 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -234,6 +234,7 @@ static unsigned kirkwood_i2s_play_mute(unsigned ctl) static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { + struct snd_pcm_runtime *runtime = substream->runtime; struct kirkwood_dma_data *priv = snd_soc_dai_get_drvdata(dai); uint32_t ctl, value; @@ -271,9 +272,11 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, writel(value, priv->io + KIRKWOOD_PLAYCTL); /* enable interrupts */ - value = readl(priv->io + KIRKWOOD_INT_MASK); - value |= KIRKWOOD_INT_CAUSE_PLAY_BYTES; - writel(value, priv->io + KIRKWOOD_INT_MASK); + if (!runtime->no_period_wakeup) { + value = readl(priv->io + KIRKWOOD_INT_MASK); + value |= KIRKWOOD_INT_CAUSE_PLAY_BYTES; + writel(value, priv->io + KIRKWOOD_INT_MASK); + } /* enable playback */ writel(ctl, priv->io + KIRKWOOD_PLAYCTL); -- cgit v1.2.3 From e73f3de5c5eb5285a1b1d8eed111eb229702b20f Mon Sep 17 00:00:00 2001 From: Russell King Date: Thu, 26 Jun 2014 15:22:50 +0100 Subject: ASoC: fix debugfs directory creation bug Avoid creating duplicate directories by prefixing codecs and platforms with their separate identifiers. This avoids snd-soc-dummy (which can appear both as a dummy platform and a dummy codec on the same card) from clashing. Signed-off-by: Russell King Tested-by: Andrew Lunn Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 28 ++++++++++++++++++++++++---- 1 file changed, 24 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b87d7d882e6d..91120b8e283e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -270,12 +270,32 @@ static const struct file_operations codec_reg_fops = { .llseek = default_llseek, }; +static struct dentry *soc_debugfs_create_dir(struct dentry *parent, + const char *fmt, ...) +{ + struct dentry *de; + va_list ap; + char *s; + + va_start(ap, fmt); + s = kvasprintf(GFP_KERNEL, fmt, ap); + va_end(ap); + + if (!s) + return NULL; + + de = debugfs_create_dir(s, parent); + kfree(s); + + return de; +} + static void soc_init_codec_debugfs(struct snd_soc_codec *codec) { struct dentry *debugfs_card_root = codec->card->debugfs_card_root; - codec->debugfs_codec_root = debugfs_create_dir(codec->name, - debugfs_card_root); + codec->debugfs_codec_root = soc_debugfs_create_dir(debugfs_card_root, + "codec:%s", codec->name); if (!codec->debugfs_codec_root) { dev_warn(codec->dev, "ASoC: Failed to create codec debugfs directory\n"); @@ -306,8 +326,8 @@ static void soc_init_platform_debugfs(struct snd_soc_platform *platform) { struct dentry *debugfs_card_root = platform->card->debugfs_card_root; - platform->debugfs_platform_root = debugfs_create_dir(platform->name, - debugfs_card_root); + platform->debugfs_platform_root = soc_debugfs_create_dir(debugfs_card_root, + "platform:%s", platform->name); if (!platform->debugfs_platform_root) { dev_warn(platform->dev, "ASoC: Failed to create platform debugfs directory\n"); -- cgit v1.2.3 From ffb83e8cb14cace1b08ceb56695b580c808d8a41 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Sun, 22 Jun 2014 17:55:18 -0700 Subject: ASoC: fsi: use SNDRV_DMA_TYPE_DEV for sound buffer Current fsi driver is using SNDRV_DMA_TYPE_CONTINUOUS for snd_pcm_lib_preallocate_pages_for_all(). But, it came from original dma-sh7760.c, and no longer needed. This patch exchange its parameter, and removed original dma mapping and un-needed dma_sync_single_xxx() from driver. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- arch/arm/mach-shmobile/board-armadillo800eva.c | 4 +++ arch/arm/mach-shmobile/board-kzm9g.c | 2 ++ arch/arm/mach-shmobile/board-mackerel.c | 4 +++ arch/sh/boards/mach-ecovec24/setup.c | 2 ++ sound/soc/sh/fsi.c | 42 +++----------------------- 5 files changed, 17 insertions(+), 37 deletions(-) (limited to 'sound') diff --git a/arch/arm/mach-shmobile/board-armadillo800eva.c b/arch/arm/mach-shmobile/board-armadillo800eva.c index 30fcac73a540..689c121157ec 100644 --- a/arch/arm/mach-shmobile/board-armadillo800eva.c +++ b/arch/arm/mach-shmobile/board-armadillo800eva.c @@ -998,6 +998,8 @@ static struct platform_device fsi_wm8978_device = { .id = 0, .dev = { .platform_data = &fsi_wm8978_info, + .coherent_dma_mask = DMA_BIT_MASK(32), + .dma_mask = &fsi_wm8978_device.dev.coherent_dma_mask, }, }; @@ -1021,6 +1023,8 @@ static struct platform_device fsi_hdmi_device = { .id = 1, .dev = { .platform_data = &fsi2_hdmi_info, + .coherent_dma_mask = DMA_BIT_MASK(32), + .dma_mask = &fsi_hdmi_device.dev.coherent_dma_mask, }, }; diff --git a/arch/arm/mach-shmobile/board-kzm9g.c b/arch/arm/mach-shmobile/board-kzm9g.c index f94ec8ca42c1..01e0d1386db7 100644 --- a/arch/arm/mach-shmobile/board-kzm9g.c +++ b/arch/arm/mach-shmobile/board-kzm9g.c @@ -603,6 +603,8 @@ static struct platform_device fsi_ak4648_device = { .name = "asoc-simple-card", .dev = { .platform_data = &fsi2_ak4648_info, + .coherent_dma_mask = DMA_BIT_MASK(32), + .dma_mask = &fsi_ak4648_device.dev.coherent_dma_mask, }, }; diff --git a/arch/arm/mach-shmobile/board-mackerel.c b/arch/arm/mach-shmobile/board-mackerel.c index 0ff4d8e45cf7..112553f0f9bf 100644 --- a/arch/arm/mach-shmobile/board-mackerel.c +++ b/arch/arm/mach-shmobile/board-mackerel.c @@ -523,6 +523,8 @@ static struct platform_device fsi_hdmi_device = { .id = 1, .dev = { .platform_data = &fsi2_hdmi_info, + .coherent_dma_mask = DMA_BIT_MASK(32), + .dma_mask = &fsi_hdmi_device.dev.coherent_dma_mask, }, }; @@ -919,6 +921,8 @@ static struct platform_device fsi_ak4643_device = { .name = "asoc-simple-card", .dev = { .platform_data = &fsi2_ak4643_info, + .coherent_dma_mask = DMA_BIT_MASK(32), + .dma_mask = &fsi_ak4643_device.dev.coherent_dma_mask, }, }; diff --git a/arch/sh/boards/mach-ecovec24/setup.c b/arch/sh/boards/mach-ecovec24/setup.c index 85d5255d259f..0d3049244cd3 100644 --- a/arch/sh/boards/mach-ecovec24/setup.c +++ b/arch/sh/boards/mach-ecovec24/setup.c @@ -874,6 +874,8 @@ static struct platform_device fsi_da7210_device = { .name = "asoc-simple-card", .dev = { .platform_data = &fsi_da7210_info, + .coherent_dma_mask = DMA_BIT_MASK(32), + .dma_mask = &fsi_da7210_device.dev.coherent_dma_mask, }, }; diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 710a079a7377..7a6b632c3db8 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -233,7 +233,6 @@ struct fsi_stream { */ struct dma_chan *chan; struct work_struct work; - dma_addr_t dma; int dma_id; int loop_cnt; int additional_pos; @@ -1279,11 +1278,6 @@ static irqreturn_t fsi_interrupt(int irq, void *data) */ static int fsi_dma_init(struct fsi_priv *fsi, struct fsi_stream *io) { - struct snd_pcm_runtime *runtime = io->substream->runtime; - struct snd_soc_dai *dai = fsi_get_dai(io->substream); - enum dma_data_direction dir = fsi_stream_is_play(fsi, io) ? - DMA_TO_DEVICE : DMA_FROM_DEVICE; - /* * 24bit data : 24bit bus / package in back * 16bit data : 16bit bus / stream mode @@ -1293,19 +1287,7 @@ static int fsi_dma_init(struct fsi_priv *fsi, struct fsi_stream *io) io->loop_cnt = 2; /* push 1st, 2nd period first, then 3rd, 4th... */ io->additional_pos = 0; - io->dma = dma_map_single(dai->dev, runtime->dma_area, - snd_pcm_lib_buffer_bytes(io->substream), dir); - return 0; -} - -static int fsi_dma_quit(struct fsi_priv *fsi, struct fsi_stream *io) -{ - struct snd_soc_dai *dai = fsi_get_dai(io->substream); - enum dma_data_direction dir = fsi_stream_is_play(fsi, io) ? - DMA_TO_DEVICE : DMA_FROM_DEVICE; - dma_unmap_single(dai->dev, io->dma, - snd_pcm_lib_buffer_bytes(io->substream), dir); return 0; } @@ -1317,7 +1299,8 @@ static dma_addr_t fsi_dma_get_area(struct fsi_stream *io, int additional) if (period >= runtime->periods) period = 0; - return io->dma + samples_to_bytes(runtime, period * io->period_samples); + return runtime->dma_addr + + samples_to_bytes(runtime, period * io->period_samples); } static void fsi_dma_complete(void *data) @@ -1325,12 +1308,6 @@ static void fsi_dma_complete(void *data) struct fsi_stream *io = (struct fsi_stream *)data; struct fsi_priv *fsi = fsi_stream_to_priv(io); struct snd_pcm_runtime *runtime = io->substream->runtime; - struct snd_soc_dai *dai = fsi_get_dai(io->substream); - enum dma_data_direction dir = fsi_stream_is_play(fsi, io) ? - DMA_TO_DEVICE : DMA_FROM_DEVICE; - - dma_sync_single_for_cpu(dai->dev, fsi_dma_get_area(io, 0), - samples_to_bytes(runtime, io->period_samples), dir); io->buff_sample_pos += io->period_samples; io->period_pos++; @@ -1369,8 +1346,6 @@ static void fsi_dma_do_work(struct work_struct *work) for (i = 0; i < io->loop_cnt; i++) { buf = fsi_dma_get_area(io, io->additional_pos); - dma_sync_single_for_device(dai->dev, buf, len, dir); - desc = dmaengine_prep_slave_single(io->chan, buf, len, dir, DMA_PREP_INTERRUPT | DMA_CTRL_ACK); if (!desc) { @@ -1495,7 +1470,6 @@ static int fsi_dma_remove(struct fsi_priv *fsi, struct fsi_stream *io) static struct fsi_stream_handler fsi_dma_push_handler = { .init = fsi_dma_init, - .quit = fsi_dma_quit, .probe = fsi_dma_probe, .transfer = fsi_dma_transfer, .remove = fsi_dma_remove, @@ -1850,16 +1824,10 @@ static void fsi_pcm_free(struct snd_pcm *pcm) static int fsi_pcm_new(struct snd_soc_pcm_runtime *rtd) { - struct snd_pcm *pcm = rtd->pcm; - - /* - * dont use SNDRV_DMA_TYPE_DEV, since it will oops the SH kernel - * in MMAP mode (i.e. aplay -M) - */ return snd_pcm_lib_preallocate_pages_for_all( - pcm, - SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), + rtd->pcm, + SNDRV_DMA_TYPE_DEV, + rtd->card->snd_card->dev, PREALLOC_BUFFER, PREALLOC_BUFFER_MAX); } -- cgit v1.2.3 From d403e24908836cf80531519530a9be84426e3a3f Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Sun, 22 Jun 2014 17:55:41 -0700 Subject: ASoC: fsi: add fsi_pointer_update() for common pointer method fsi PIO/DMA handler are using each own pointer update method, but these can be share. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 57 +++++++++++++++++++++--------------------------------- 1 file changed, 22 insertions(+), 35 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 7a6b632c3db8..820a40833c04 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1041,6 +1041,26 @@ static int fsi_clk_set_rate_cpg(struct device *dev, return ret; } +static void fsi_pointer_update(struct fsi_stream *io, int size) +{ + io->buff_sample_pos += size; + + if (io->buff_sample_pos >= + io->period_samples * (io->period_pos + 1)) { + struct snd_pcm_substream *substream = io->substream; + struct snd_pcm_runtime *runtime = substream->runtime; + + io->period_pos++; + + if (io->period_pos >= runtime->periods) { + io->buff_sample_pos = 0; + io->period_pos = 0; + } + + snd_pcm_period_elapsed(substream); + } +} + /* * pio data transfer handler */ @@ -1107,31 +1127,11 @@ static int fsi_pio_transfer(struct fsi_priv *fsi, struct fsi_stream *io, void (*run32)(struct fsi_priv *fsi, u8 *buf, int samples), int samples) { - struct snd_pcm_runtime *runtime; - struct snd_pcm_substream *substream; u8 *buf; - int over_period; if (!fsi_stream_is_working(fsi, io)) return -EINVAL; - over_period = 0; - substream = io->substream; - runtime = substream->runtime; - - /* FSI FIFO has limit. - * So, this driver can not send periods data at a time - */ - if (io->buff_sample_pos >= - io->period_samples * (io->period_pos + 1)) { - - over_period = 1; - io->period_pos = (io->period_pos + 1) % runtime->periods; - - if (0 == io->period_pos) - io->buff_sample_pos = 0; - } - buf = fsi_pio_get_area(fsi, io); switch (io->sample_width) { @@ -1145,11 +1145,7 @@ static int fsi_pio_transfer(struct fsi_priv *fsi, struct fsi_stream *io, return -EINVAL; } - /* update buff_sample_pos */ - io->buff_sample_pos += samples; - - if (over_period) - snd_pcm_period_elapsed(substream); + fsi_pointer_update(io, samples); return 0; } @@ -1307,20 +1303,11 @@ static void fsi_dma_complete(void *data) { struct fsi_stream *io = (struct fsi_stream *)data; struct fsi_priv *fsi = fsi_stream_to_priv(io); - struct snd_pcm_runtime *runtime = io->substream->runtime; - io->buff_sample_pos += io->period_samples; - io->period_pos++; - - if (io->period_pos >= runtime->periods) { - io->period_pos = 0; - io->buff_sample_pos = 0; - } + fsi_pointer_update(io, io->period_samples); fsi_count_fifo_err(fsi); fsi_stream_transfer(io); - - snd_pcm_period_elapsed(io->substream); } static void fsi_dma_do_work(struct work_struct *work) -- cgit v1.2.3 From 8457e0e9e274cae4898f84dd5aaeb5d2098126c8 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Sun, 22 Jun 2014 17:55:59 -0700 Subject: ASoC: fsi: use dmaengine_prep_dma_cyclic() for DMA transfer Current FSI driver is using DMAEngine directly, but, ASoC is requesting to use common DMA transfer method, like snd_dmaengine_pcm_trigger() or dmaengine_pcm_ops. It is difficult to switch at this point, since Renesas driver is also supporting PIO transfer. This patch uses dmaengine_prep_dma_cyclic() instead of dmaengine_prep_slave_single(). It is used in requested method, and is good first step to switch over. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 94 ++++++++++++++++-------------------------------------- 1 file changed, 27 insertions(+), 67 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 820a40833c04..a57eb96e57eb 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -232,10 +232,7 @@ struct fsi_stream { * these are for DMAEngine */ struct dma_chan *chan; - struct work_struct work; int dma_id; - int loop_cnt; - int additional_pos; }; struct fsi_clk { @@ -1281,24 +1278,9 @@ static int fsi_dma_init(struct fsi_priv *fsi, struct fsi_stream *io) io->bus_option = BUSOP_SET(24, PACKAGE_24BITBUS_BACK) | BUSOP_SET(16, PACKAGE_16BITBUS_STREAM); - io->loop_cnt = 2; /* push 1st, 2nd period first, then 3rd, 4th... */ - io->additional_pos = 0; - return 0; } -static dma_addr_t fsi_dma_get_area(struct fsi_stream *io, int additional) -{ - struct snd_pcm_runtime *runtime = io->substream->runtime; - int period = io->period_pos + additional; - - if (period >= runtime->periods) - period = 0; - - return runtime->dma_addr + - samples_to_bytes(runtime, period * io->period_samples); -} - static void fsi_dma_complete(void *data) { struct fsi_stream *io = (struct fsi_stream *)data; @@ -1307,53 +1289,37 @@ static void fsi_dma_complete(void *data) fsi_pointer_update(io, io->period_samples); fsi_count_fifo_err(fsi); - fsi_stream_transfer(io); } -static void fsi_dma_do_work(struct work_struct *work) +static int fsi_dma_transfer(struct fsi_priv *fsi, struct fsi_stream *io) { - struct fsi_stream *io = container_of(work, struct fsi_stream, work); - struct fsi_priv *fsi = fsi_stream_to_priv(io); - struct snd_soc_dai *dai; + struct snd_soc_dai *dai = fsi_get_dai(io->substream); + struct snd_pcm_substream *substream = io->substream; struct dma_async_tx_descriptor *desc; - struct snd_pcm_runtime *runtime; - enum dma_data_direction dir; int is_play = fsi_stream_is_play(fsi, io); - int len, i; - dma_addr_t buf; - - if (!fsi_stream_is_working(fsi, io)) - return; - - dai = fsi_get_dai(io->substream); - runtime = io->substream->runtime; - dir = is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE; - len = samples_to_bytes(runtime, io->period_samples); - - for (i = 0; i < io->loop_cnt; i++) { - buf = fsi_dma_get_area(io, io->additional_pos); - - desc = dmaengine_prep_slave_single(io->chan, buf, len, dir, - DMA_PREP_INTERRUPT | DMA_CTRL_ACK); - if (!desc) { - dev_err(dai->dev, "dmaengine_prep_slave_sg() fail\n"); - return; - } - - desc->callback = fsi_dma_complete; - desc->callback_param = io; - - if (dmaengine_submit(desc) < 0) { - dev_err(dai->dev, "tx_submit() fail\n"); - return; - } + enum dma_data_direction dir = is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE; + int ret = -EIO; + + desc = dmaengine_prep_dma_cyclic(io->chan, + substream->runtime->dma_addr, + snd_pcm_lib_buffer_bytes(substream), + snd_pcm_lib_period_bytes(substream), + dir, + DMA_PREP_INTERRUPT | DMA_CTRL_ACK); + if (!desc) { + dev_err(dai->dev, "dmaengine_prep_dma_cyclic() fail\n"); + goto fsi_dma_transfer_err; + } - dma_async_issue_pending(io->chan); + desc->callback = fsi_dma_complete; + desc->callback_param = io; - io->additional_pos = 1; + if (dmaengine_submit(desc) < 0) { + dev_err(dai->dev, "tx_submit() fail\n"); + goto fsi_dma_transfer_err; } - io->loop_cnt = 1; + dma_async_issue_pending(io->chan); /* * FIXME @@ -1370,13 +1336,11 @@ static void fsi_dma_do_work(struct work_struct *work) fsi_reg_write(fsi, DIFF_ST, 0); } } -} -static int fsi_dma_transfer(struct fsi_priv *fsi, struct fsi_stream *io) -{ - schedule_work(&io->work); + ret = 0; - return 0; +fsi_dma_transfer_err: + return ret; } static int fsi_dma_push_start_stop(struct fsi_priv *fsi, struct fsi_stream *io, @@ -1437,15 +1401,11 @@ static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io, struct dev return fsi_stream_probe(fsi, dev); } - INIT_WORK(&io->work, fsi_dma_do_work); - return 0; } static int fsi_dma_remove(struct fsi_priv *fsi, struct fsi_stream *io) { - cancel_work_sync(&io->work); - fsi_stream_stop(fsi, io); if (io->chan) @@ -1618,9 +1578,9 @@ static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, if (!ret) ret = fsi_hw_startup(fsi, io, dai->dev); if (!ret) - ret = fsi_stream_transfer(io); + ret = fsi_stream_start(fsi, io); if (!ret) - fsi_stream_start(fsi, io); + ret = fsi_stream_transfer(io); break; case SNDRV_PCM_TRIGGER_STOP: if (!ret) -- cgit v1.2.3 From d9288d0ba12de1b5efb830b9128e4cc6877318fc Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Sun, 22 Jun 2014 17:56:23 -0700 Subject: ASoC: rsnd: SSI + DMA can select BUSIF Sound data needs to be sent to R-Car sound SSI when playback. But, there are 2 interfaces for it. 1st is SSITDR/SSIRDR which are mapped on SSI. 2nd is SSIn_BUSIF which are mapped on SSIU. 2nd SSIn_BUSIF is used when DMA transfer, and it is always used if sound data came from via SRC. But, we can use it when SSI+DMA case too. (Current driver is assuming 1st SSITDR/SSIRDR for it) 2nd SSIn_BUSIF can be used as FIFO. This is very helpful/useful for SSI+DMA. But DMA address / DMA ID are not same between 1st/2nd cases. This patch care about these settings. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/renesas,rsnd.txt | 1 + include/sound/rcar_snd.h | 1 + sound/soc/sh/rcar/core.c | 22 +++++--- sound/soc/sh/rcar/gen.c | 64 ++++++++++++++-------- sound/soc/sh/rcar/rsnd.h | 10 +++- sound/soc/sh/rcar/src.c | 37 +++++++++---- sound/soc/sh/rcar/ssi.c | 33 ++++++++++- 7 files changed, 123 insertions(+), 45 deletions(-) (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 8346cab046cd..41a120c2389d 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -21,6 +21,7 @@ SSI subnode properties: - interrupts : Should contain SSI interrupt for PIO transfer - shared-pin : if shared clock pin - pio-transfer : use PIO transfer mode +- no-busif : BUSIF is not ussed when [mem -> SSI] via DMA case SRC subnode properties: no properties at this point diff --git a/include/sound/rcar_snd.h b/include/sound/rcar_snd.h index f4a706f82cb7..d76412b84b48 100644 --- a/include/sound/rcar_snd.h +++ b/include/sound/rcar_snd.h @@ -34,6 +34,7 @@ * B : SSI direction */ #define RSND_SSI_CLK_PIN_SHARE (1 << 31) +#define RSND_SSI_NO_BUSIF (1 << 30) /* SSI+DMA without BUSIF */ #define RSND_SSI(_dma_id, _pio_irq, _flags) \ { .dma_id = _dma_id, .pio_irq = _pio_irq, .flags = _flags } diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 7f68b33dcbbb..8c3707a68603 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -138,6 +138,17 @@ char *rsnd_mod_name(struct rsnd_mod *mod) return mod->ops->name; } +char *rsnd_mod_dma_name(struct rsnd_mod *mod) +{ + if (!mod || !mod->ops) + return "unknown"; + + if (!mod->ops->dma_name) + return mod->ops->name; + + return mod->ops->dma_name(mod); +} + void rsnd_mod_init(struct rsnd_priv *priv, struct rsnd_mod *mod, struct rsnd_mod_ops *ops, @@ -261,7 +272,7 @@ static int _rsnd_dma_of_name(char *dma_name, struct rsnd_mod *mod) { if (mod) return snprintf(dma_name, DMA_NAME_SIZE / 2, "%s%d", - rsnd_mod_name(mod), rsnd_mod_id(mod)); + rsnd_mod_dma_name(mod), rsnd_mod_id(mod)); else return snprintf(dma_name, DMA_NAME_SIZE / 2, "mem"); @@ -343,11 +354,8 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, dma_cap_zero(mask); dma_cap_set(DMA_SLAVE, mask); - if (dev->of_node) - rsnd_dma_of_name(dma, is_play, dma_name); - else - snprintf(dma_name, DMA_NAME_SIZE, - is_play ? "tx" : "rx"); + rsnd_dma_of_name(dma, is_play, dma_name); + rsnd_gen_dma_addr(priv, dma, &cfg, is_play, id); dev_dbg(dev, "dma name : %s\n", dma_name); @@ -359,8 +367,6 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, return -EIO; } - rsnd_gen_dma_addr(priv, dma, &cfg, is_play, id); - ret = dmaengine_slave_config(dma->chan, &cfg); if (ret < 0) goto rsnd_dma_init_err; diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 0280a11c0899..46677af6c748 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -165,15 +165,19 @@ static int rsnd_gen_regmap_init(struct rsnd_priv *priv, * * ex) R-Car H2 case * mod / DMAC in / DMAC out / DMAC PP in / DMAC pp out - * SSI : 0xec541000 / 0xec241008 / 0xec24100c / 0xec400000 / 0xec400000 + * SSI : 0xec541000 / 0xec241008 / 0xec24100c + * SSIU: 0xec541000 / 0xec100000 / 0xec100000 / 0xec400000 / 0xec400000 * SCU : 0xec500000 / 0xec000000 / 0xec004000 / 0xec300000 / 0xec304000 * CMD : 0xec500000 / 0xec008000 0xec308000 */ #define RDMA_SSI_I_N(addr, i) (addr ##_reg - 0x00300000 + (0x40 * i) + 0x8) #define RDMA_SSI_O_N(addr, i) (addr ##_reg - 0x00300000 + (0x40 * i) + 0xc) -#define RDMA_SSI_I_P(addr, i) (addr ##_reg - 0x00141000 + (0x1000 * i)) -#define RDMA_SSI_O_P(addr, i) (addr ##_reg - 0x00141000 + (0x1000 * i)) +#define RDMA_SSIU_I_N(addr, i) (addr ##_reg - 0x00441000 + (0x1000 * i)) +#define RDMA_SSIU_O_N(addr, i) (addr ##_reg - 0x00441000 + (0x1000 * i)) + +#define RDMA_SSIU_I_P(addr, i) (addr ##_reg - 0x00141000 + (0x1000 * i)) +#define RDMA_SSIU_O_P(addr, i) (addr ##_reg - 0x00141000 + (0x1000 * i)) #define RDMA_SRC_I_N(addr, i) (addr ##_reg - 0x00500000 + (0x400 * i)) #define RDMA_SRC_O_N(addr, i) (addr ##_reg - 0x004fc000 + (0x400 * i)) @@ -204,26 +208,36 @@ static void rsnd_gen2_dma_addr(struct rsnd_priv *priv, struct dma_addr { dma_addr_t src_addr; dma_addr_t dst_addr; - } dma_addrs[2][2][3] = { - { /* SRC */ - /* Capture */ - {{ 0, 0 }, - { RDMA_SRC_O_N(src, id), 0 }, - { RDMA_CMD_O_N(src, id), 0 }}, - /* Playback */ - {{ 0, 0, }, - { 0, RDMA_SRC_I_N(src, id) }, - { 0, RDMA_SRC_I_N(src, id) }} - }, { /* SSI */ - /* Capture */ - {{ RDMA_SSI_O_N(ssi, id), 0 }, - { RDMA_SSI_O_P(ssi, id), RDMA_SRC_I_P(src, id) }, - { RDMA_SSI_O_P(ssi, id), RDMA_SRC_I_P(src, id) }}, - /* Playback */ - {{ 0, RDMA_SSI_I_N(ssi, id) }, - { RDMA_SRC_O_P(src, id), RDMA_SSI_I_P(ssi, id) }, - { RDMA_CMD_O_P(src, id), RDMA_SSI_I_P(ssi, id) }} - } + } dma_addrs[3][2][3] = { + /* SRC */ + {{{ 0, 0 }, + /* Capture */ + { RDMA_SRC_O_N(src, id), 0 }, + { RDMA_CMD_O_N(src, id), 0 } }, + /* Playback */ + {{ 0, 0, }, + { 0, RDMA_SRC_I_N(src, id) }, + { 0, RDMA_SRC_I_N(src, id) } } + }, + /* SSI */ + /* Capture */ + {{{ RDMA_SSI_O_N(ssi, id), 0 }, + { 0, 0 }, + { 0, 0 } }, + /* Playback */ + {{ 0, RDMA_SSI_I_N(ssi, id) }, + { 0, 0 }, + { 0, 0 } } + }, + /* SSIU */ + /* Capture */ + {{{ RDMA_SSIU_O_N(ssi, id), 0 }, + { RDMA_SSIU_O_P(ssi, id), RDMA_SRC_I_P(src, id) }, + { RDMA_SSIU_O_P(ssi, id), RDMA_SRC_I_P(src, id) } }, + /* Playback */ + {{ 0, RDMA_SSIU_I_N(ssi, id) }, + { RDMA_SRC_O_P(src, id), RDMA_SSIU_I_P(ssi, id) }, + { RDMA_CMD_O_P(src, id), RDMA_SSIU_I_P(ssi, id) } } }, }; /* it shouldn't happen */ @@ -232,6 +246,10 @@ static void rsnd_gen2_dma_addr(struct rsnd_priv *priv, return; } + /* use SSIU or SSI ? */ + if (is_ssi && (0 == strcmp(rsnd_mod_dma_name(mod), "ssiu"))) + is_ssi++; + cfg->src_addr = dma_addrs[is_ssi][is_play][use_src + use_dvc].src_addr; cfg->dst_addr = dma_addrs[is_ssi][is_play][use_src + use_dvc].dst_addr; diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 067a89e9f25c..a1466c1570bc 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -185,6 +185,7 @@ enum rsnd_mod_type { struct rsnd_mod_ops { char *name; + char* (*dma_name)(struct rsnd_mod *mod); int (*probe)(struct rsnd_mod *mod, struct rsnd_dai *rdai); int (*remove)(struct rsnd_mod *mod, @@ -224,6 +225,7 @@ void rsnd_mod_init(struct rsnd_priv *priv, enum rsnd_mod_type type, int id); char *rsnd_mod_name(struct rsnd_mod *mod); +char *rsnd_mod_dma_name(struct rsnd_mod *mod); /* * R-Car sound DAI @@ -391,8 +393,12 @@ struct rsnd_mod *rsnd_src_mod_get(struct rsnd_priv *priv, int id); unsigned int rsnd_src_get_ssi_rate(struct rsnd_priv *priv, struct rsnd_dai_stream *io, struct snd_pcm_runtime *runtime); -int rsnd_src_ssi_mode_init(struct rsnd_mod *ssi_mod, - struct rsnd_dai *rdai); +int rsnd_src_ssiu_start(struct rsnd_mod *ssi_mod, + struct rsnd_dai *rdai, + int use_busif); +int rsnd_src_ssiu_stop(struct rsnd_mod *ssi_mod, + struct rsnd_dai *rdai, + int use_busif); int rsnd_src_enable_ssi_irq(struct rsnd_mod *ssi_mod, struct rsnd_dai *rdai); diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 200eda019bc7..4d39505c21cf 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -106,18 +106,17 @@ struct rsnd_src { /* * Gen1/Gen2 common functions */ -int rsnd_src_ssi_mode_init(struct rsnd_mod *ssi_mod, - struct rsnd_dai *rdai) +int rsnd_src_ssiu_start(struct rsnd_mod *ssi_mod, + struct rsnd_dai *rdai, + int use_busif) { - struct rsnd_dai_stream *io = rsnd_mod_to_io(ssi_mod); - struct rsnd_mod *src_mod = rsnd_io_to_mod_src(io); int ssi_id = rsnd_mod_id(ssi_mod); /* * SSI_MODE0 */ rsnd_mod_bset(ssi_mod, SSI_MODE0, (1 << ssi_id), - src_mod ? 0 : (1 << ssi_id)); + !use_busif << ssi_id); /* * SSI_MODE1 @@ -143,6 +142,29 @@ int rsnd_src_ssi_mode_init(struct rsnd_mod *ssi_mod, 0x2 << shift : 0x1 << shift); } + /* + * DMA settings for SSIU + */ + if (use_busif) { + rsnd_mod_write(ssi_mod, SSI_BUSIF_ADINR, + rsnd_get_adinr(ssi_mod)); + rsnd_mod_write(ssi_mod, SSI_BUSIF_MODE, 1); + rsnd_mod_write(ssi_mod, SSI_CTRL, 0x1); + } + + return 0; +} + +int rsnd_src_ssiu_stop(struct rsnd_mod *ssi_mod, + struct rsnd_dai *rdai, + int use_busif) +{ + /* + * DMA settings for SSIU + */ + if (use_busif) + rsnd_mod_write(ssi_mod, SSI_CTRL, 0); + return 0; } @@ -467,9 +489,6 @@ static int rsnd_src_set_convert_rate_gen2(struct rsnd_mod *mod, if (ret < 0) return ret; - rsnd_mod_write(mod, SSI_BUSIF_ADINR, rsnd_get_adinr(mod)); - rsnd_mod_write(mod, SSI_BUSIF_MODE, 1); - rsnd_mod_write(mod, SRC_SRCCR, 0x00011110); rsnd_mod_write(mod, SRC_BSDSR, 0x01800000); @@ -554,7 +573,6 @@ static int rsnd_src_start_gen2(struct rsnd_mod *mod, rsnd_dma_start(rsnd_mod_to_dma(&src->mod)); - rsnd_mod_write(mod, SSI_CTRL, 0x1); rsnd_mod_write(mod, SRC_CTRL, val); return rsnd_src_start(mod, rdai); @@ -565,7 +583,6 @@ static int rsnd_src_stop_gen2(struct rsnd_mod *mod, { struct rsnd_src *src = rsnd_mod_to_src(mod); - rsnd_mod_write(mod, SSI_CTRL, 0); rsnd_mod_write(mod, SRC_CTRL, 0); rsnd_dma_stop(rsnd_mod_to_dma(&src->mod)); diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 2df723df5d19..34e84009162b 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -90,6 +90,20 @@ struct rsnd_ssi { #define rsnd_ssi_mode_flags(p) ((p)->info->flags) #define rsnd_ssi_dai_id(ssi) ((ssi)->info->dai_id) +static int rsnd_ssi_use_busif(struct rsnd_mod *mod) +{ + struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); + struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); + int use_busif = 0; + + if (!(rsnd_ssi_mode_flags(ssi) & RSND_SSI_NO_BUSIF)) + use_busif = 1; + if (rsnd_io_to_mod_src(io)) + use_busif = 1; + + return use_busif; +} + static void rsnd_ssi_status_check(struct rsnd_mod *mod, u32 bit) { @@ -289,8 +303,6 @@ static int rsnd_ssi_init(struct rsnd_mod *mod, ssi->cr_own = cr; ssi->err = -1; /* ignore 1st error */ - rsnd_src_ssi_mode_init(mod, rdai); - return 0; } @@ -389,6 +401,8 @@ static int rsnd_ssi_pio_start(struct rsnd_mod *mod, /* enable PIO IRQ */ ssi->cr_etc = UIEN | OIEN | DIEN; + rsnd_src_ssiu_start(mod, rdai, 0); + rsnd_src_enable_ssi_irq(mod, rdai); rsnd_ssi_hw_start(ssi, rdai, io); @@ -405,6 +419,8 @@ static int rsnd_ssi_pio_stop(struct rsnd_mod *mod, rsnd_ssi_hw_stop(ssi, rdai); + rsnd_src_ssiu_stop(mod, rdai, 0); + return 0; } @@ -457,6 +473,8 @@ static int rsnd_ssi_dma_start(struct rsnd_mod *mod, /* enable DMA transfer */ ssi->cr_etc = DMEN; + rsnd_src_ssiu_start(mod, rdai, rsnd_ssi_use_busif(mod)); + rsnd_dma_start(dma); rsnd_ssi_hw_start(ssi, ssi->rdai, io); @@ -482,11 +500,19 @@ static int rsnd_ssi_dma_stop(struct rsnd_mod *mod, rsnd_dma_stop(dma); + rsnd_src_ssiu_stop(mod, rdai, 1); + return 0; } +static char *rsnd_ssi_dma_name(struct rsnd_mod *mod) +{ + return rsnd_ssi_use_busif(mod) ? "ssiu" : SSI_NAME; +} + static struct rsnd_mod_ops rsnd_ssi_dma_ops = { .name = SSI_NAME, + .dma_name = rsnd_ssi_dma_name, .probe = rsnd_ssi_dma_probe, .remove = rsnd_ssi_dma_remove, .init = rsnd_ssi_init, @@ -595,6 +621,9 @@ static void rsnd_of_parse_ssi(struct platform_device *pdev, */ ssi_info->dma_id = of_get_property(np, "pio-transfer", NULL) ? 0 : 1; + + if (of_get_property(np, "no-busif", NULL)) + ssi_info->flags |= RSND_SSI_NO_BUSIF; } rsnd_of_parse_ssi_end: -- cgit v1.2.3 From ccd01559ead29b59918458e9b412ff18b88fc6cf Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Sun, 22 Jun 2014 17:56:41 -0700 Subject: ASoC: rsnd: use dmaengine_prep_dma_cyclic() instead of original method Current R-Car sound driver is using DMAEngine directly, but, ASoC is requesting to use common DMA transfer method, like snd_dmaengine_pcm_trigger() or dmaengine_pcm_ops. It is difficult to switch at this point, since Renesas driver is also supporting PIO transfer. This patch uses dmaengine_prep_dma_cyclic() instead of dmaengine_prep_slave_single(). It is used in requested method, and is good first step to switch over. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 76 ++++++++++++------------------------------------ sound/soc/sh/rcar/rsnd.h | 4 --- 2 files changed, 18 insertions(+), 62 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 8c3707a68603..5149fe2dae9f 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -164,26 +164,8 @@ void rsnd_mod_init(struct rsnd_priv *priv, /* * rsnd_dma functions */ -static void __rsnd_dma_start(struct rsnd_dma *dma); -static void rsnd_dma_continue(struct rsnd_dma *dma) -{ - /* push next A or B plane */ - dma->submit_loop = 1; - schedule_work(&dma->work); -} - -void rsnd_dma_start(struct rsnd_dma *dma) -{ - /* push both A and B plane*/ - dma->offset = 0; - dma->submit_loop = 2; - __rsnd_dma_start(dma); -} - void rsnd_dma_stop(struct rsnd_dma *dma) { - dma->submit_loop = 0; - cancel_work_sync(&dma->work); dmaengine_terminate_all(dma->chan); } @@ -191,11 +173,7 @@ static void rsnd_dma_complete(void *data) { struct rsnd_dma *dma = (struct rsnd_dma *)data; struct rsnd_mod *mod = rsnd_dma_to_mod(dma); - struct rsnd_priv *priv = rsnd_mod_to_priv(rsnd_dma_to_mod(dma)); struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); - unsigned long flags; - - rsnd_lock(priv, flags); /* * Renesas sound Gen1 needs 1 DMAC, @@ -208,57 +186,40 @@ static void rsnd_dma_complete(void *data) * rsnd_dai_pointer_update() will be called twice, * ant it will breaks io->byte_pos */ - if (dma->submit_loop) - rsnd_dma_continue(dma); - - rsnd_unlock(priv, flags); rsnd_dai_pointer_update(io, io->byte_per_period); } -static void __rsnd_dma_start(struct rsnd_dma *dma) +void rsnd_dma_start(struct rsnd_dma *dma) { struct rsnd_mod *mod = rsnd_dma_to_mod(dma); struct rsnd_priv *priv = rsnd_mod_to_priv(mod); struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); - struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + struct snd_pcm_substream *substream = io->substream; struct device *dev = rsnd_priv_to_dev(priv); struct dma_async_tx_descriptor *desc; - dma_addr_t buf; - size_t len = io->byte_per_period; - int i; - - for (i = 0; i < dma->submit_loop; i++) { - buf = runtime->dma_addr + - rsnd_dai_pointer_offset(io, dma->offset + len); - dma->offset = len; + desc = dmaengine_prep_dma_cyclic(dma->chan, + substream->runtime->dma_addr, + snd_pcm_lib_buffer_bytes(substream), + snd_pcm_lib_period_bytes(substream), + dma->dir, + DMA_PREP_INTERRUPT | DMA_CTRL_ACK); - desc = dmaengine_prep_slave_single( - dma->chan, buf, len, dma->dir, - DMA_PREP_INTERRUPT | DMA_CTRL_ACK); - if (!desc) { - dev_err(dev, "dmaengine_prep_slave_sg() fail\n"); - return; - } - - desc->callback = rsnd_dma_complete; - desc->callback_param = dma; + if (!desc) { + dev_err(dev, "dmaengine_prep_slave_sg() fail\n"); + return; + } - if (dmaengine_submit(desc) < 0) { - dev_err(dev, "dmaengine_submit() fail\n"); - return; - } + desc->callback = rsnd_dma_complete; + desc->callback_param = dma; - dma_async_issue_pending(dma->chan); + if (dmaengine_submit(desc) < 0) { + dev_err(dev, "dmaengine_submit() fail\n"); + return; } -} - -static void rsnd_dma_do_work(struct work_struct *work) -{ - struct rsnd_dma *dma = container_of(work, struct rsnd_dma, work); - __rsnd_dma_start(dma); + dma_async_issue_pending(dma->chan); } int rsnd_dma_available(struct rsnd_dma *dma) @@ -372,7 +333,6 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, goto rsnd_dma_init_err; dma->dir = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; - INIT_WORK(&dma->work, rsnd_dma_do_work); return 0; diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index a1466c1570bc..60b5e9260600 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -156,12 +156,8 @@ u32 rsnd_get_adinr(struct rsnd_mod *mod); */ struct rsnd_dma { struct sh_dmae_slave slave; - struct work_struct work; struct dma_chan *chan; enum dma_transfer_direction dir; - - int submit_loop; - int offset; /* it cares A/B plane */ }; void rsnd_dma_start(struct rsnd_dma *dma); -- cgit v1.2.3 From 65f459923b7926f6a7f156970d83360bd80cb169 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Sun, 22 Jun 2014 17:57:04 -0700 Subject: ASoC: rsnd: enable DVC when capture Current DVC can be enabled only when playback, but, this came from misunderstanding. It is not correct. DVC <-> DMA relationship is... Playback: MEM -> DMAC -> SRC -> DVC -> DMACp -> SSI Capture: SSI -> DMACp -> SRC -> DVC -> DMAC -> MEM DVC can be used for both Playback/Capture Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/dvc.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index ed0007006899..deef310c75dc 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -191,24 +191,20 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, struct snd_soc_pcm_runtime *rtd) { struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); - struct rsnd_priv *priv = rsnd_mod_to_priv(mod); - struct device *dev = rsnd_priv_to_dev(priv); struct snd_card *card = rtd->card->snd_card; struct snd_kcontrol *kctrl; static struct snd_kcontrol_new knew = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Playback Volume", .info = rsnd_dvc_volume_info, .get = rsnd_dvc_volume_get, .put = rsnd_dvc_volume_put, }; int ret; - if (!rsnd_dai_is_play(rdai, io)) { - dev_err(dev, "DVC%d is connected to Capture DAI\n", - rsnd_mod_id(mod)); - return -EINVAL; - } + if (rsnd_dai_is_play(rdai, io)) + knew.name = "Playback Volume"; + else + knew.name = "Capture Volume"; kctrl = snd_ctl_new1(&knew, mod); if (!kctrl) -- cgit v1.2.3 From 59f5cbecf9531e56b1da16c9343349e3e3ea972b Mon Sep 17 00:00:00 2001 From: Paul Handrigan Date: Sat, 28 Jun 2014 11:34:25 -0500 Subject: ASoC: cs4265: Change return values to boolean. The cs4265_volatile_register reutrns a bool. The function now returns true or false vs 1 and 0. Signed-off-by: Paul Handrigan Signed-off-by: Mark Brown --- sound/soc/codecs/cs4265.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index c9c04d2987c2..4c4bf216d51a 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -87,9 +87,9 @@ static bool cs4265_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { case CS4265_INT_STATUS: - return 1; + return true; default: - return 0; + return false; } } -- cgit v1.2.3 From 25ccb22ed55cf4ed1b94e2627b80f9ef44637558 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 26 Jun 2014 08:06:55 +0300 Subject: ASoC: tlv320aic3x: Correct S24_3LE support Correct the hw_params callback to configure the codec correctly in case of S24_3LE format since in case of S24_3LE the codec has been configured to 16bit format mode. S24_LE is not defined as supported format for the codec. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index e12fafbb1e09..5360772bc1ad 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -879,7 +879,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S20_3LE: data |= (0x01 << 4); break; - case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S24_3LE: data |= (0x02 << 4); break; case SNDRV_PCM_FORMAT_S32_LE: -- cgit v1.2.3 From 2a11a10abe1b88b86e44480511262c72da97e55a Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 26 Jun 2014 08:06:56 +0300 Subject: ASoC: tlv320aic3x: Add support for S24_LE format The codec need to be configured to 24bit mode in case of S24_LE format. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 5360772bc1ad..d275890a6827 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -880,6 +880,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, data |= (0x01 << 4); break; case SNDRV_PCM_FORMAT_S24_3LE: + case SNDRV_PCM_FORMAT_S24_LE: data |= (0x02 << 4); break; case SNDRV_PCM_FORMAT_S32_LE: @@ -1194,7 +1195,8 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec, #define AIC3X_RATES SNDRV_PCM_RATE_8000_96000 #define AIC3X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ - SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) static const struct snd_soc_dai_ops aic3x_dai_ops = { .hw_params = aic3x_hw_params, -- cgit v1.2.3 From 182bef863cc37a9a387ae9bc0f1b05243234bd4a Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 26 Jun 2014 08:09:24 +0300 Subject: ASoC: davinci-mcasp: Fix S24_LE and U24_LE support In case of S24_LE/U24_LE modes we expect 24bits on the bus while the samples are stored and transferred in memory on 32bits (lower 3 bytes of the 4 bytes). Signed-off-by: Peter Ujfalusi Tested-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 0ee4986038cc..bfcc6c3dc2fd 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -720,6 +720,10 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_U24_LE: case SNDRV_PCM_FORMAT_S24_LE: + dma_params->data_type = 4; + word_length = 24; + break; + case SNDRV_PCM_FORMAT_U32_LE: case SNDRV_PCM_FORMAT_S32_LE: dma_params->data_type = 4; -- cgit v1.2.3 From 099d334e3d5c0b26480dffc44fe6272c90898237 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 17 Jun 2014 12:41:31 +0800 Subject: ASoC: rt5677: Convert to use rl6231_pll_calc The implementation of rt5677_pll_calc() has the same logic of rl6231_pll_calc(). The only difference is the lower boundary checking for freq_in. This patch calls rl6231_pll_calc() instead of open-coded. The k_bp of struct rt5677_pll_code is always false, thus also remove the code to check pll_code.k_bp. Signed-off-by: Axel Lin Tested-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 69 ++++++----------------------------------------- sound/soc/codecs/rt5677.h | 8 ------ 2 files changed, 8 insertions(+), 69 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 8ecfb323b1b2..98370ae3b38d 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -2996,62 +2996,12 @@ static int rt5677_set_dai_sysclk(struct snd_soc_dai *dai, * Returns 0 for success or negative error code. */ static int rt5677_pll_calc(const unsigned int freq_in, - const unsigned int freq_out, struct rt5677_pll_code *pll_code) + const unsigned int freq_out, struct rl6231_pll_code *pll_code) { - int max_n = RT5677_PLL_N_MAX, max_m = RT5677_PLL_M_MAX; - int k, red, n_t, pll_out, in_t; - int n = 0, m = 0, m_t = 0; - int out_t, red_t = abs(freq_out - freq_in); - bool m_bp = false, k_bp = false; - - if (RT5677_PLL_INP_MAX < freq_in || RT5677_PLL_INP_MIN > freq_in) + if (RT5677_PLL_INP_MIN > freq_in) return -EINVAL; - k = 100000000 / freq_out - 2; - if (k > RT5677_PLL_K_MAX) - k = RT5677_PLL_K_MAX; - for (n_t = 0; n_t <= max_n; n_t++) { - in_t = freq_in / (k + 2); - pll_out = freq_out / (n_t + 2); - if (in_t < 0) - continue; - if (in_t == pll_out) { - m_bp = true; - n = n_t; - goto code_find; - } - red = abs(in_t - pll_out); - if (red < red_t) { - m_bp = true; - n = n_t; - m = m_t; - if (red == 0) - goto code_find; - red_t = red; - } - for (m_t = 0; m_t <= max_m; m_t++) { - out_t = in_t / (m_t + 2); - red = abs(out_t - pll_out); - if (red < red_t) { - m_bp = false; - n = n_t; - m = m_t; - if (red == 0) - goto code_find; - red_t = red; - } - } - } - pr_debug("Only get approximation about PLL\n"); - -code_find: - - pll_code->m_bp = m_bp; - pll_code->k_bp = k_bp; - pll_code->m_code = m; - pll_code->n_code = n; - pll_code->k_code = k; - return 0; + return rl6231_pll_calc(freq_in, freq_out, pll_code); } static int rt5677_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, @@ -3059,7 +3009,7 @@ static int rt5677_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, { struct snd_soc_codec *codec = dai->codec; struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); - struct rt5677_pll_code pll_code; + struct rl6231_pll_code pll_code; int ret; if (source == rt5677->pll_src && freq_in == rt5677->pll_in && @@ -3117,15 +3067,12 @@ static int rt5677_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, return ret; } - dev_dbg(codec->dev, "m_bypass=%d k_bypass=%d m=%d n=%d k=%d\n", - pll_code.m_bp, pll_code.k_bp, - (pll_code.m_bp ? 0 : pll_code.m_code), pll_code.n_code, - (pll_code.k_bp ? 0 : pll_code.k_code)); + dev_dbg(codec->dev, "m_bypass=%d m=%d n=%d k=%d\n", + pll_code.m_bp, (pll_code.m_bp ? 0 : pll_code.m_code), + pll_code.n_code, pll_code.k_code); regmap_write(rt5677->regmap, RT5677_PLL1_CTRL1, - pll_code.n_code << RT5677_PLL_N_SFT | - pll_code.k_bp << RT5677_PLL_K_BP_SFT | - (pll_code.k_bp ? 0 : pll_code.k_code)); + pll_code.n_code << RT5677_PLL_N_SFT | pll_code.k_code); regmap_write(rt5677->regmap, RT5677_PLL1_CTRL2, (pll_code.m_bp ? 0 : pll_code.m_code) << RT5677_PLL_M_SFT | pll_code.m_bp << RT5677_PLL_M_BP_SFT); diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index af4e9c797408..08252e7c1ab4 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1425,14 +1425,6 @@ enum { RT5677_AIFS, }; -struct rt5677_pll_code { - bool m_bp; /* Indicates bypass m code or not. */ - bool k_bp; /* Indicates bypass k code or not. */ - int m_code; - int n_code; - int k_code; -}; - struct rt5677_priv { struct snd_soc_codec *codec; struct rt5677_platform_data pdata; -- cgit v1.2.3 From 053e69d57cc6253b19ea661f929c8c1b6a907bff Mon Sep 17 00:00:00 2001 From: Wonjoon Lee Date: Fri, 20 Jun 2014 13:33:15 +0530 Subject: ASoC: max98090: Add max98091 compatible string The MAX98091 CODEC is the same as MAX98090 CODEC, but with an extra microphone. Existing driver for MAX98090 CODEC already has support for MAX98091 CODEC. Adding proper compatible string so that MAX98091 CODEC can be specified from device tree. Signed-off-by: Wonjoon Lee Signed-off-by: Doug Anderson Signed-off-by: Tushar Behera Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/max98090.txt | 2 +- sound/soc/codecs/max98090.c | 2 ++ 2 files changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/max98090.txt b/Documentation/devicetree/bindings/sound/max98090.txt index a5e63fa47dc5..c454e67f54bb 100644 --- a/Documentation/devicetree/bindings/sound/max98090.txt +++ b/Documentation/devicetree/bindings/sound/max98090.txt @@ -4,7 +4,7 @@ This device supports I2C only. Required properties: -- compatible : "maxim,max98090". +- compatible : "maxim,max98090" or "maxim,max98091". - reg : The I2C address of the device. diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 3aec3ae78fe0..c00b36872dfe 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2447,12 +2447,14 @@ static const struct dev_pm_ops max98090_pm = { static const struct i2c_device_id max98090_i2c_id[] = { { "max98090", MAX98090 }, + { "max98091", MAX98091 }, { } }; MODULE_DEVICE_TABLE(i2c, max98090_i2c_id); static const struct of_device_id max98090_of_match[] = { { .compatible = "maxim,max98090", }, + { .compatible = "maxim,max98091", }, { } }; MODULE_DEVICE_TABLE(of, max98090_of_match); -- cgit v1.2.3 From 46aed597527384b30a6d49bff1806f6b1ed1fd77 Mon Sep 17 00:00:00 2001 From: Tushar Behera Date: Fri, 20 Jun 2014 13:33:16 +0530 Subject: ASoC: samsung: Extend snow driver to support MAX98091 Peach-pi board has MAX98091 CODEC. Extend snow machine driver to support this board. Signed-off-by: Tushar Behera Reviewed-by: Doug Anderson Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/snow.txt | 1 + sound/soc/samsung/snow.c | 1 + 2 files changed, 2 insertions(+) (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/snow.txt b/Documentation/devicetree/bindings/sound/snow.txt index 678b191c37b8..e0b7a8207f5f 100644 --- a/Documentation/devicetree/bindings/sound/snow.txt +++ b/Documentation/devicetree/bindings/sound/snow.txt @@ -3,6 +3,7 @@ Audio Binding for Snow boards Required properties: - compatible : Can be one of the following, "google,snow-audio-max98090" or + "google,snow-audio-max98091" or "google,snow-audio-max98095" - samsung,i2s-controller: The phandle of the Samsung I2S controller - samsung,audio-codec: The phandle of the audio codec diff --git a/sound/soc/samsung/snow.c b/sound/soc/samsung/snow.c index 014c177840ba..8bbd348358dd 100644 --- a/sound/soc/samsung/snow.c +++ b/sound/soc/samsung/snow.c @@ -103,6 +103,7 @@ static int snow_probe(struct platform_device *pdev) static const struct of_device_id snow_of_match[] = { { .compatible = "google,snow-audio-max98090", }, + { .compatible = "google,snow-audio-max98091", }, { .compatible = "google,snow-audio-max98095", }, {}, }; -- cgit v1.2.3 From d1a792f3b4072bfac4150bb62aa34917b77fdb6d Mon Sep 17 00:00:00 2001 From: Russell King - ARM Linux Date: Wed, 25 Jun 2014 13:00:33 +0100 Subject: Update imx-sdma cyclic handling to report residue I received a report this morning from one of the Novena developers that the behaviour of the iMX6 ASoC codec driver (using imx-pcm-dma.c) was sub-optimal under high system load. While there are issues relating to system load remaining, upon reviewing the ASoC imx-pcm-dma.c driver, it was noticed that it not using the residue support, because SDMA doesn't support it. This has the effect that SDMA has to make multiple calls into the ASoC and ALSA code, one for each period. Since ALSA's snd_pcm_elapsed() does not need to be called multiple times and it is entirely sufficient to call it once to update ALSA with the current buffer position via the pointer method, we can do better here. We can also avoid stopping the DMA entirely, just like real cyclic DMA implementations behave. While this means that we replay some old samples, this is a nicer behaviour than having audio stop and restart. The changes to achieve this are relatively minor - imx-sdma.c can track where the DMA is to the nearest descriptor boundary - it does this already when deciding how many callbacks to issue. In doing this, buf_tail always points at the descriptor which will complete next. The residue is defined by the bytes remaining to the end of the buffer, when the buffer is viewed as a single block of memory [start...end]. So, when we start out, there's a full buffer worth of residue, and this counts down as we approach the end of the buffer, eventually becoming zero at the end, before returning to the full buffer worth when we wrap back to the start. Moving the walking of the descriptors into the interrupt handler means that we can update the BD_DONE flag at interrupt time, thus avoiding a delayed tasklet stopping the cyclic DMA. This means that the residue can be calculated from (total descriptors - buf_tail) * descriptor size. This is what the change below does. We update imx-pcm-dma.c to remove the NO_RESIDUE flag since we now provide the residue. Signed-off-by: Russell King Tested-by: Shawn Guo Signed-off-by: Vinod Koul --- drivers/dma/imx-sdma.c | 22 ++++++++++++++++++---- sound/soc/fsl/imx-pcm-dma.c | 1 - 2 files changed, 18 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/drivers/dma/imx-sdma.c b/drivers/dma/imx-sdma.c index 128714622bf5..14867e3ac8ff 100644 --- a/drivers/dma/imx-sdma.c +++ b/drivers/dma/imx-sdma.c @@ -255,6 +255,7 @@ struct sdma_channel { enum dma_slave_buswidth word_size; unsigned int buf_tail; unsigned int num_bd; + unsigned int period_len; struct sdma_buffer_descriptor *bd; dma_addr_t bd_phys; unsigned int pc_from_device, pc_to_device; @@ -592,6 +593,12 @@ static void sdma_event_disable(struct sdma_channel *sdmac, unsigned int event) } static void sdma_handle_channel_loop(struct sdma_channel *sdmac) +{ + if (sdmac->desc.callback) + sdmac->desc.callback(sdmac->desc.callback_param); +} + +static void sdma_update_channel_loop(struct sdma_channel *sdmac) { struct sdma_buffer_descriptor *bd; @@ -611,9 +618,6 @@ static void sdma_handle_channel_loop(struct sdma_channel *sdmac) bd->mode.status |= BD_DONE; sdmac->buf_tail++; sdmac->buf_tail %= sdmac->num_bd; - - if (sdmac->desc.callback) - sdmac->desc.callback(sdmac->desc.callback_param); } } @@ -669,6 +673,9 @@ static irqreturn_t sdma_int_handler(int irq, void *dev_id) int channel = fls(stat) - 1; struct sdma_channel *sdmac = &sdma->channel[channel]; + if (sdmac->flags & IMX_DMA_SG_LOOP) + sdma_update_channel_loop(sdmac); + tasklet_schedule(&sdmac->tasklet); __clear_bit(channel, &stat); @@ -1129,6 +1136,7 @@ static struct dma_async_tx_descriptor *sdma_prep_dma_cyclic( sdmac->status = DMA_IN_PROGRESS; sdmac->buf_tail = 0; + sdmac->period_len = period_len; sdmac->flags |= IMX_DMA_SG_LOOP; sdmac->direction = direction; @@ -1225,9 +1233,15 @@ static enum dma_status sdma_tx_status(struct dma_chan *chan, struct dma_tx_state *txstate) { struct sdma_channel *sdmac = to_sdma_chan(chan); + u32 residue; + + if (sdmac->flags & IMX_DMA_SG_LOOP) + residue = (sdmac->num_bd - sdmac->buf_tail) * sdmac->period_len; + else + residue = sdmac->chn_count - sdmac->chn_real_count; dma_set_tx_state(txstate, chan->completed_cookie, chan->cookie, - sdmac->chn_count - sdmac->chn_real_count); + residue); return sdmac->status; } diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c index 0849b7b83f0a..0db94f492e97 100644 --- a/sound/soc/fsl/imx-pcm-dma.c +++ b/sound/soc/fsl/imx-pcm-dma.c @@ -59,7 +59,6 @@ int imx_pcm_dma_init(struct platform_device *pdev) { return devm_snd_dmaengine_pcm_register(&pdev->dev, &imx_dmaengine_pcm_config, - SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | SND_DMAENGINE_PCM_FLAG_COMPAT); } EXPORT_SYMBOL_GPL(imx_pcm_dma_init); -- cgit v1.2.3 From 3f901a028feb916d17d6c01b9010f3f56f675d26 Mon Sep 17 00:00:00 2001 From: Benoit Cousson Date: Tue, 1 Jul 2014 09:47:54 +0200 Subject: ASoC: core: Change soc_link_dai_widgets signature for multiple codecs Since multiple codecs DAI will be usable in the future, remove explicit unique codec_dai and cpu_dai parameters. Replace them with snd_soc_pcm_runtime pointer that will contain every instances. No functionale change. Signed-off-by: Benoit Cousson Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b87d7d882e6d..075e0ae5fade 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1450,9 +1450,10 @@ static int soc_probe_codec_dai(struct snd_soc_card *card, static int soc_link_dai_widgets(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link, - struct snd_soc_dai *cpu_dai, - struct snd_soc_dai *codec_dai) + struct snd_soc_pcm_runtime *rtd) { + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dapm_widget *play_w, *capture_w; int ret; @@ -1570,8 +1571,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) codec2codec_close_delayed_work); /* link the DAI widgets */ - ret = soc_link_dai_widgets(card, dai_link, - cpu_dai, codec_dai); + ret = soc_link_dai_widgets(card, dai_link, rtd); if (ret) return ret; } -- cgit v1.2.3 From c8dd1fec47d0b1875f292c40bed381b343e38b40 Mon Sep 17 00:00:00 2001 From: Benoit Cousson Date: Tue, 1 Jul 2014 09:47:55 +0200 Subject: ASoC: pcm: Refactor soc_pcm_apply_msb for multicodecs Refactor the function to facilitate the migration to multiple codecs. Fix a trailing space in the header as well. No functional change. Signed-off-by: Benoit Cousson Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 35 ++++++++++++++++++++++++----------- 1 file changed, 24 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 54d18f22a33e..320a904b6b69 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -7,7 +7,7 @@ * Copyright (C) 2010 Texas Instruments Inc. * * Authors: Liam Girdwood - * Mark Brown + * Mark Brown * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the @@ -284,15 +284,10 @@ static int sample_sizes[] = { 24, 32, }; -static void soc_pcm_apply_msb(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) +static void soc_pcm_set_msb(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai, int bits) { - int ret, i, bits; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - bits = dai->driver->playback.sig_bits; - else - bits = dai->driver->capture.sig_bits; + int ret, i; if (!bits) return; @@ -310,6 +305,25 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream, } } +static void soc_pcm_apply_msb(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + unsigned int bits = 0, cpu_bits; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + bits = codec_dai->driver->playback.sig_bits; + cpu_bits = cpu_dai->driver->playback.sig_bits; + } else { + bits = codec_dai->driver->capture.sig_bits; + cpu_bits = cpu_dai->driver->capture.sig_bits; + } + + soc_pcm_set_msb(substream, codec_dai, bits); + soc_pcm_set_msb(substream, cpu_dai, cpu_bits); +} + static void soc_pcm_init_runtime_hw(struct snd_pcm_runtime *runtime, struct snd_soc_pcm_stream *codec_stream, struct snd_soc_pcm_stream *cpu_stream) @@ -433,8 +447,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) goto config_err; } - soc_pcm_apply_msb(substream, codec_dai); - soc_pcm_apply_msb(substream, cpu_dai); + soc_pcm_apply_msb(substream); /* Symmetry only applies if we've already got an active stream. */ if (cpu_dai->active) { -- cgit v1.2.3 From 37523034851d6a4b1dc951e24bf2f11bc28fe58d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Sun, 22 Jun 2014 17:58:26 -0700 Subject: ASoC: rsnd: DMA cleanup for flexible SSI/SRC selection Current R-Car sound SSI/SRC/DVC selection has feature limit. (It is assuming that SSI/SRC are using same index number) So that enabling SSI/SRC flexible selection, this patch modifies DMA settings. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 45 ++++++++++++++++++++---------- sound/soc/sh/rcar/gen.c | 71 +++++++++++++++++++++--------------------------- sound/soc/sh/rcar/rsnd.h | 7 ++--- 3 files changed, 64 insertions(+), 59 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 5149fe2dae9f..4435a31ecdad 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -239,8 +239,21 @@ static int _rsnd_dma_of_name(char *dma_name, struct rsnd_mod *mod) } -static void rsnd_dma_of_name(struct rsnd_dma *dma, - int is_play, char *dma_name) +static void rsnd_dma_of_name(struct rsnd_mod *mod_from, + struct rsnd_mod *mod_to, + char *dma_name) +{ + int index = 0; + + index = _rsnd_dma_of_name(dma_name + index, mod_from); + *(dma_name + index++) = '_'; + index = _rsnd_dma_of_name(dma_name + index, mod_to); +} + +static void rsnd_dma_of_path(struct rsnd_dma *dma, + int is_play, + struct rsnd_mod **mod_from, + struct rsnd_mod **mod_to) { struct rsnd_mod *this = rsnd_dma_to_mod(dma); struct rsnd_dai_stream *io = rsnd_mod_to_io(this); @@ -248,7 +261,6 @@ static void rsnd_dma_of_name(struct rsnd_dma *dma, struct rsnd_mod *src = rsnd_io_to_mod_src(io); struct rsnd_mod *dvc = rsnd_io_to_mod_dvc(io); struct rsnd_mod *mod[MOD_MAX]; - struct rsnd_mod *src_mod, *dst_mod; int i, index; @@ -285,17 +297,12 @@ static void rsnd_dma_of_name(struct rsnd_dma *dma, } if (is_play) { - src_mod = mod[index - 1]; - dst_mod = mod[index]; + *mod_from = mod[index - 1]; + *mod_to = mod[index]; } else { - src_mod = mod[index]; - dst_mod = mod[index - 1]; + *mod_from = mod[index]; + *mod_to = mod[index - 1]; } - - index = 0; - index = _rsnd_dma_of_name(dma_name + index, src_mod); - *(dma_name + index++) = '_'; - index = _rsnd_dma_of_name(dma_name + index, dst_mod); } int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, @@ -303,6 +310,8 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, { struct device *dev = rsnd_priv_to_dev(priv); struct dma_slave_config cfg; + struct rsnd_mod *mod_from; + struct rsnd_mod *mod_to; char dma_name[DMA_NAME_SIZE]; dma_cap_mask_t mask; int ret; @@ -315,10 +324,16 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, dma_cap_zero(mask); dma_cap_set(DMA_SLAVE, mask); - rsnd_dma_of_name(dma, is_play, dma_name); - rsnd_gen_dma_addr(priv, dma, &cfg, is_play, id); + rsnd_dma_of_path(dma, is_play, &mod_from, &mod_to); + rsnd_dma_of_name(mod_from, mod_to, dma_name); + + cfg.slave_id = id; + cfg.direction = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; + cfg.src_addr = rsnd_gen_dma_addr(priv, mod_from, is_play, 1); + cfg.dst_addr = rsnd_gen_dma_addr(priv, mod_to, is_play, 0); - dev_dbg(dev, "dma name : %s\n", dma_name); + dev_dbg(dev, "dma : %s %pad -> %pad\n", + dma_name, &cfg.src_addr, &cfg.dst_addr); dma->chan = dma_request_slave_channel_compat(mask, shdma_chan_filter, (void *)id, dev, diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 46677af6c748..73ce4c90efda 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -168,7 +168,7 @@ static int rsnd_gen_regmap_init(struct rsnd_priv *priv, * SSI : 0xec541000 / 0xec241008 / 0xec24100c * SSIU: 0xec541000 / 0xec100000 / 0xec100000 / 0xec400000 / 0xec400000 * SCU : 0xec500000 / 0xec000000 / 0xec004000 / 0xec300000 / 0xec304000 - * CMD : 0xec500000 / 0xec008000 0xec308000 + * CMD : 0xec500000 / / 0xec008000 0xec308000 */ #define RDMA_SSI_I_N(addr, i) (addr ##_reg - 0x00300000 + (0x40 * i) + 0x8) #define RDMA_SSI_O_N(addr, i) (addr ##_reg - 0x00300000 + (0x40 * i) + 0xc) @@ -188,14 +188,13 @@ static int rsnd_gen_regmap_init(struct rsnd_priv *priv, #define RDMA_CMD_O_N(addr, i) (addr ##_reg - 0x004f8000 + (0x400 * i)) #define RDMA_CMD_O_P(addr, i) (addr ##_reg - 0x001f8000 + (0x400 * i)) -static void rsnd_gen2_dma_addr(struct rsnd_priv *priv, - struct rsnd_dma *dma, - struct dma_slave_config *cfg, - int is_play, int slave_id) +static dma_addr_t +rsnd_gen2_dma_addr(struct rsnd_priv *priv, + struct rsnd_mod *mod, + int is_play, int is_from) { struct platform_device *pdev = rsnd_priv_to_pdev(priv); struct device *dev = rsnd_priv_to_dev(priv); - struct rsnd_mod *mod = rsnd_dma_to_mod(dma); struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); dma_addr_t ssi_reg = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN2_SSI)->start; @@ -206,76 +205,68 @@ static void rsnd_gen2_dma_addr(struct rsnd_priv *priv, int use_dvc = !!rsnd_io_to_mod_dvc(io); int id = rsnd_mod_id(mod); struct dma_addr { - dma_addr_t src_addr; - dma_addr_t dst_addr; + dma_addr_t out_addr; + dma_addr_t in_addr; } dma_addrs[3][2][3] = { /* SRC */ {{{ 0, 0 }, /* Capture */ - { RDMA_SRC_O_N(src, id), 0 }, - { RDMA_CMD_O_N(src, id), 0 } }, + { RDMA_SRC_O_N(src, id), RDMA_SRC_I_P(src, id) }, + { RDMA_CMD_O_N(src, id), RDMA_SRC_I_P(src, id) } }, /* Playback */ {{ 0, 0, }, - { 0, RDMA_SRC_I_N(src, id) }, - { 0, RDMA_SRC_I_N(src, id) } } + { RDMA_SRC_O_P(src, id), RDMA_SRC_I_N(src, id) }, + { RDMA_CMD_O_P(src, id), RDMA_SRC_I_N(src, id) } } }, /* SSI */ /* Capture */ {{{ RDMA_SSI_O_N(ssi, id), 0 }, - { 0, 0 }, - { 0, 0 } }, + { RDMA_SSIU_O_P(ssi, id), 0 }, + { RDMA_SSIU_O_P(ssi, id), 0 } }, /* Playback */ {{ 0, RDMA_SSI_I_N(ssi, id) }, - { 0, 0 }, - { 0, 0 } } + { 0, RDMA_SSIU_I_P(ssi, id) }, + { 0, RDMA_SSIU_I_P(ssi, id) } } }, /* SSIU */ /* Capture */ {{{ RDMA_SSIU_O_N(ssi, id), 0 }, - { RDMA_SSIU_O_P(ssi, id), RDMA_SRC_I_P(src, id) }, - { RDMA_SSIU_O_P(ssi, id), RDMA_SRC_I_P(src, id) } }, + { RDMA_SSIU_O_P(ssi, id), 0 }, + { RDMA_SSIU_O_P(ssi, id), 0 } }, /* Playback */ {{ 0, RDMA_SSIU_I_N(ssi, id) }, - { RDMA_SRC_O_P(src, id), RDMA_SSIU_I_P(ssi, id) }, - { RDMA_CMD_O_P(src, id), RDMA_SSIU_I_P(ssi, id) } } }, + { 0, RDMA_SSIU_I_P(ssi, id) }, + { 0, RDMA_SSIU_I_P(ssi, id) } } }, }; /* it shouldn't happen */ - if (use_dvc & !use_src) { + if (use_dvc & !use_src) dev_err(dev, "DVC is selected without SRC\n"); - return; - } /* use SSIU or SSI ? */ if (is_ssi && (0 == strcmp(rsnd_mod_dma_name(mod), "ssiu"))) is_ssi++; - cfg->src_addr = dma_addrs[is_ssi][is_play][use_src + use_dvc].src_addr; - cfg->dst_addr = dma_addrs[is_ssi][is_play][use_src + use_dvc].dst_addr; - - dev_dbg(dev, "dma%d addr - src : %x / dst : %x\n", - id, cfg->src_addr, cfg->dst_addr); + return (is_from) ? + dma_addrs[is_ssi][is_play][use_src + use_dvc].out_addr : + dma_addrs[is_ssi][is_play][use_src + use_dvc].in_addr; } -void rsnd_gen_dma_addr(struct rsnd_priv *priv, - struct rsnd_dma *dma, - struct dma_slave_config *cfg, - int is_play, int slave_id) +dma_addr_t rsnd_gen_dma_addr(struct rsnd_priv *priv, + struct rsnd_mod *mod, + int is_play, int is_from) { - cfg->slave_id = slave_id; - cfg->src_addr = 0; - cfg->dst_addr = 0; - cfg->direction = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; - /* * gen1 uses default DMA addr */ if (rsnd_is_gen1(priv)) - return; + return 0; - rsnd_gen2_dma_addr(priv, dma, cfg, is_play, slave_id); -} + if (!mod) + return 0; + return rsnd_gen2_dma_addr(priv, mod, is_play, is_from); +} /* * Gen2 diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 60b5e9260600..425b22ee6fb9 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -279,10 +279,9 @@ int rsnd_gen_probe(struct platform_device *pdev, void __iomem *rsnd_gen_reg_get(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg); -void rsnd_gen_dma_addr(struct rsnd_priv *priv, - struct rsnd_dma *dma, - struct dma_slave_config *cfg, - int is_play, int slave_id); +dma_addr_t rsnd_gen_dma_addr(struct rsnd_priv *priv, + struct rsnd_mod *mod, + int is_play, int is_from); #define rsnd_is_gen1(s) (((s)->info->flags & RSND_GEN_MASK) == RSND_GEN1) #define rsnd_is_gen2(s) (((s)->info->flags & RSND_GEN_MASK) == RSND_GEN2) -- cgit v1.2.3 From 340371005d399b595f4044fc5e336f4a66ead101 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Sun, 22 Jun 2014 17:59:02 -0700 Subject: ASoC: rsnd: DMA start address is properly used for each DMAC R-Car sound uses Audio DMAC and Audio DMAC peri peri. Audio DMAC peri peri transfers data inside circuit. DMA transfer needs source / destination address, and destination address can be set via dmaengine_slave_config(). The source address can be set when starting DMAEngine. Because Audio DMAC peri peri always ignores its value, current driver always used same source address for Audio DMAC / Audio DMAC peri peri (Audio DMAC peri peri source / destination address is always fixed value) But, This is not good match for DT booting. This patch properly uses DMA start address for Audio DMAC / Audio DMAC peri peri. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 2 ++ sound/soc/sh/rcar/rsnd.h | 1 + 2 files changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 4435a31ecdad..49d9b31b185c 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -200,6 +200,7 @@ void rsnd_dma_start(struct rsnd_dma *dma) struct dma_async_tx_descriptor *desc; desc = dmaengine_prep_dma_cyclic(dma->chan, + (dma->addr) ? dma->addr : substream->runtime->dma_addr, snd_pcm_lib_buffer_bytes(substream), snd_pcm_lib_period_bytes(substream), @@ -347,6 +348,7 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, if (ret < 0) goto rsnd_dma_init_err; + dma->addr = is_play ? cfg.src_addr : cfg.dst_addr; dma->dir = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; return 0; diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 425b22ee6fb9..631b149df08f 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -158,6 +158,7 @@ struct rsnd_dma { struct sh_dmae_slave slave; struct dma_chan *chan; enum dma_transfer_direction dir; + dma_addr_t addr; }; void rsnd_dma_start(struct rsnd_dma *dma); -- cgit v1.2.3 From 34cb6123f90d264d63bdcd3ee0df0d2cb4b36aab Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Sun, 22 Jun 2014 17:59:28 -0700 Subject: ASoC: rsnd: add DT support to DVC Now, DVC can use DT Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/renesas,rsnd.txt | 8 +++++ sound/soc/sh/rcar/core.c | 13 ++++++-- sound/soc/sh/rcar/dvc.c | 38 ++++++++++++++++++++++ 3 files changed, 57 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 41a120c2389d..aa697abf337e 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -13,6 +13,9 @@ Required properties: - rcar_sound,src : Should contain SRC feature. The number of SRC subnode should be same as HW. see below for detail. +- rcar_sound,dvc : Should contain DVC feature. + The number of DVC subnode should be same as HW. + see below for detail. - rcar_sound,dai : DAI contents. The number of DAI subnode should be same as HW. see below for detail. @@ -40,6 +43,11 @@ rcar_sound: rcar_sound@0xffd90000 { <0 0xec540000 0 0x1000>, /* SSIU */ <0 0xec541000 0 0x1280>; /* SSI */ + rcar_sound,dvc { + dvc0: dvc@0 { }; + dvc1: dvc@1 { }; + }; + rcar_sound,src { src0: src@0 { }; src1: src@1 { }; diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 49d9b31b185c..907d4802fd5c 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -286,7 +286,13 @@ static void rsnd_dma_of_path(struct rsnd_dma *dma, mod[i] = src; src = NULL; } else { - mod[i] = dvc; + if ((!is_play) && (this == src)) + this = dvc; + + mod[i] = (is_play) ? src : dvc; + i++; + mod[i] = (is_play) ? dvc : src; + src = NULL; dvc = NULL; } @@ -719,12 +725,13 @@ static void rsnd_of_parse_dai(struct platform_device *pdev, struct device_node *dai_node, *dai_np; struct device_node *ssi_node, *ssi_np; struct device_node *src_node, *src_np; + struct device_node *dvc_node, *dvc_np; struct device_node *playback, *capture; struct rsnd_dai_platform_info *dai_info; struct rcar_snd_info *info = rsnd_priv_to_info(priv); struct device *dev = &pdev->dev; int nr, i; - int dai_i, ssi_i, src_i; + int dai_i, ssi_i, src_i, dvc_i; if (!of_data) return; @@ -750,6 +757,7 @@ static void rsnd_of_parse_dai(struct platform_device *pdev, ssi_node = of_get_child_by_name(dev->of_node, "rcar_sound,ssi"); src_node = of_get_child_by_name(dev->of_node, "rcar_sound,src"); + dvc_node = of_get_child_by_name(dev->of_node, "rcar_sound,dvc"); #define mod_parse(name) \ if (name##_node) { \ @@ -785,6 +793,7 @@ if (name##_node) { \ mod_parse(ssi); mod_parse(src); + mod_parse(dvc); if (playback) of_node_put(playback); diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index deef310c75dc..9096fb03d001 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -235,6 +235,42 @@ struct rsnd_mod *rsnd_dvc_mod_get(struct rsnd_priv *priv, int id) return &((struct rsnd_dvc *)(priv->dvc) + id)->mod; } +static void rsnd_of_parse_dvc(struct platform_device *pdev, + const struct rsnd_of_data *of_data, + struct rsnd_priv *priv) +{ + struct device_node *node; + struct rsnd_dvc_platform_info *dvc_info; + struct rcar_snd_info *info = rsnd_priv_to_info(priv); + struct device *dev = &pdev->dev; + int nr; + + if (!of_data) + return; + + node = of_get_child_by_name(dev->of_node, "rcar_sound,dvc"); + if (!node) + return; + + nr = of_get_child_count(node); + if (!nr) + goto rsnd_of_parse_dvc_end; + + dvc_info = devm_kzalloc(dev, + sizeof(struct rsnd_dvc_platform_info) * nr, + GFP_KERNEL); + if (!dvc_info) { + dev_err(dev, "dvc info allocation error\n"); + goto rsnd_of_parse_dvc_end; + } + + info->dvc_info = dvc_info; + info->dvc_info_nr = nr; + +rsnd_of_parse_dvc_end: + of_node_put(node); +} + int rsnd_dvc_probe(struct platform_device *pdev, const struct rsnd_of_data *of_data, struct rsnd_priv *priv) @@ -246,6 +282,8 @@ int rsnd_dvc_probe(struct platform_device *pdev, char name[RSND_DVC_NAME_SIZE]; int i, nr; + rsnd_of_parse_dvc(pdev, of_data, priv); + nr = info->dvc_info_nr; if (!nr) return 0; -- cgit v1.2.3 From 5e8351de740d9eff26cc146a6591a4e7517496b0 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Mon, 30 Jun 2014 20:31:13 +0800 Subject: ASoC: add RT5670 CODEC driver This patch adds a minimum support of Realtek ALC5670 codec. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- include/sound/rt5670.h | 27 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt5670-dsp.h | 54 + sound/soc/codecs/rt5670.c | 2692 +++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5670.h | 2000 ++++++++++++++++++++++++++++++ 6 files changed, 4781 insertions(+) create mode 100644 include/sound/rt5670.h create mode 100644 sound/soc/codecs/rt5670-dsp.h create mode 100644 sound/soc/codecs/rt5670.c create mode 100644 sound/soc/codecs/rt5670.h (limited to 'sound') diff --git a/include/sound/rt5670.h b/include/sound/rt5670.h new file mode 100644 index 000000000000..bd311197a3b5 --- /dev/null +++ b/include/sound/rt5670.h @@ -0,0 +1,27 @@ +/* + * linux/sound/rt5670.h -- Platform data for RT5670 + * + * Copyright 2014 Realtek Microelectronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __LINUX_SND_RT5670_H +#define __LINUX_SND_RT5670_H + +struct rt5670_platform_data { + int jd_mode; + bool in2_diff; + + bool dmic_en; + unsigned int dmic1_data_pin; + /* 0 = GPIO6; 1 = IN2P; 3 = GPIO7*/ + unsigned int dmic2_data_pin; + /* 0 = GPIO8; 1 = IN3N; */ + unsigned int dmic3_data_pin; + /* 0 = GPIO9; 1 = GPIO10; 2 = GPIO5*/ +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9d88845dc7c4..f31b132e75c5 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -78,6 +78,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_RT5640 if I2C select SND_SOC_RT5645 if I2C select SND_SOC_RT5651 if I2C + select SND_SOC_RT5670 if I2C select SND_SOC_RT5677 if I2C select SND_SOC_SGTL5000 if I2C select SND_SOC_SI476X if MFD_SI476X_CORE @@ -445,10 +446,12 @@ config SND_SOC_RL6231 default y if SND_SOC_RT5640=y default y if SND_SOC_RT5645=y default y if SND_SOC_RT5651=y + default y if SND_SOC_RT5670=y default y if SND_SOC_RT5677=y default m if SND_SOC_RT5640=m default m if SND_SOC_RT5645=m default m if SND_SOC_RT5651=m + default m if SND_SOC_RT5670=m default m if SND_SOC_RT5677=m config SND_SOC_RT5631 @@ -463,6 +466,9 @@ config SND_SOC_RT5645 config SND_SOC_RT5651 tristate +config SND_SOC_RT5670 + tristate + config SND_SOC_RT5677 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index be3377b8d73f..c6648789073b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -72,6 +72,7 @@ snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o snd-soc-rt5645-objs := rt5645.o snd-soc-rt5651-objs := rt5651.o +snd-soc-rt5670-objs := rt5670.o snd-soc-rt5677-objs := rt5677.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o @@ -237,6 +238,7 @@ obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o obj-$(CONFIG_SND_SOC_RT5645) += snd-soc-rt5645.o obj-$(CONFIG_SND_SOC_RT5651) += snd-soc-rt5651.o +obj-$(CONFIG_SND_SOC_RT5670) += snd-soc-rt5670.o obj-$(CONFIG_SND_SOC_RT5677) += snd-soc-rt5677.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o diff --git a/sound/soc/codecs/rt5670-dsp.h b/sound/soc/codecs/rt5670-dsp.h new file mode 100644 index 000000000000..a34d0cdb8198 --- /dev/null +++ b/sound/soc/codecs/rt5670-dsp.h @@ -0,0 +1,54 @@ +/* + * rt5670-dsp.h -- RT5670 ALSA SoC DSP driver + * + * Copyright 2014 Realtek Microelectronics + * Author: Bard Liao + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __RT5670_DSP_H__ +#define __RT5670_DSP_H__ + +#define RT5670_DSP_CTRL1 0xe0 +#define RT5670_DSP_CTRL2 0xe1 +#define RT5670_DSP_CTRL3 0xe2 +#define RT5670_DSP_CTRL4 0xe3 +#define RT5670_DSP_CTRL5 0xe4 + +/* DSP Control 1 (0xe0) */ +#define RT5670_DSP_CMD_MASK (0xff << 8) +#define RT5670_DSP_CMD_PE (0x0d << 8) /* Patch Entry */ +#define RT5670_DSP_CMD_MW (0x3b << 8) /* Memory Write */ +#define RT5670_DSP_CMD_MR (0x37 << 8) /* Memory Read */ +#define RT5670_DSP_CMD_RR (0x60 << 8) /* Register Read */ +#define RT5670_DSP_CMD_RW (0x68 << 8) /* Register Write */ +#define RT5670_DSP_REG_DATHI (0x26 << 8) /* High Data Addr */ +#define RT5670_DSP_REG_DATLO (0x25 << 8) /* Low Data Addr */ +#define RT5670_DSP_CLK_MASK (0x3 << 6) +#define RT5670_DSP_CLK_SFT 6 +#define RT5670_DSP_CLK_768K (0x0 << 6) +#define RT5670_DSP_CLK_384K (0x1 << 6) +#define RT5670_DSP_CLK_192K (0x2 << 6) +#define RT5670_DSP_CLK_96K (0x3 << 6) +#define RT5670_DSP_BUSY_MASK (0x1 << 5) +#define RT5670_DSP_RW_MASK (0x1 << 4) +#define RT5670_DSP_DL_MASK (0x3 << 2) +#define RT5670_DSP_DL_0 (0x0 << 2) +#define RT5670_DSP_DL_1 (0x1 << 2) +#define RT5670_DSP_DL_2 (0x2 << 2) +#define RT5670_DSP_DL_3 (0x3 << 2) +#define RT5670_DSP_I2C_AL_16 (0x1 << 1) +#define RT5670_DSP_CMD_EN (0x1) + +struct rt5670_dsp_param { + u16 cmd_fmt; + u16 addr; + u16 data; + u8 cmd; +}; + +#endif /* __RT5670_DSP_H__ */ + diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c new file mode 100644 index 000000000000..879d42e1a1bd --- /dev/null +++ b/sound/soc/codecs/rt5670.c @@ -0,0 +1,2692 @@ +/* + * rt5670.c -- RT5670 ALSA SoC audio codec driver + * + * Copyright 2014 Realtek Semiconductor Corp. + * Author: Bard Liao + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "rl6231.h" +#include "rt5670.h" +#include "rt5670-dsp.h" + +#define RT5670_DEVICE_ID 0x6271 + +#define RT5670_PR_RANGE_BASE (0xff + 1) +#define RT5670_PR_SPACING 0x100 + +#define RT5670_PR_BASE (RT5670_PR_RANGE_BASE + (0 * RT5670_PR_SPACING)) + +static const struct regmap_range_cfg rt5670_ranges[] = { + { .name = "PR", .range_min = RT5670_PR_BASE, + .range_max = RT5670_PR_BASE + 0xf8, + .selector_reg = RT5670_PRIV_INDEX, + .selector_mask = 0xff, + .selector_shift = 0x0, + .window_start = RT5670_PRIV_DATA, + .window_len = 0x1, }, +}; + +static struct reg_default init_list[] = { + { RT5670_PR_BASE + 0x14, 0x9a8a }, + { RT5670_PR_BASE + 0x38, 0x3ba1 }, + { RT5670_PR_BASE + 0x3d, 0x3640 }, +}; +#define RT5670_INIT_REG_LEN ARRAY_SIZE(init_list) + +static const struct reg_default rt5670_reg[] = { + { 0x00, 0x0000 }, + { 0x02, 0x8888 }, + { 0x03, 0x8888 }, + { 0x0a, 0x0001 }, + { 0x0b, 0x0827 }, + { 0x0c, 0x0000 }, + { 0x0d, 0x0008 }, + { 0x0e, 0x0000 }, + { 0x0f, 0x0808 }, + { 0x19, 0xafaf }, + { 0x1a, 0xafaf }, + { 0x1b, 0x0011 }, + { 0x1c, 0x2f2f }, + { 0x1d, 0x2f2f }, + { 0x1e, 0x0000 }, + { 0x1f, 0x2f2f }, + { 0x20, 0x0000 }, + { 0x26, 0x7860 }, + { 0x27, 0x7860 }, + { 0x28, 0x7871 }, + { 0x29, 0x8080 }, + { 0x2a, 0x5656 }, + { 0x2b, 0x5454 }, + { 0x2c, 0xaaa0 }, + { 0x2d, 0x0000 }, + { 0x2e, 0x2f2f }, + { 0x2f, 0x1002 }, + { 0x30, 0x0000 }, + { 0x31, 0x5f00 }, + { 0x32, 0x0000 }, + { 0x33, 0x0000 }, + { 0x34, 0x0000 }, + { 0x35, 0x0000 }, + { 0x36, 0x0000 }, + { 0x37, 0x0000 }, + { 0x38, 0x0000 }, + { 0x3b, 0x0000 }, + { 0x3c, 0x007f }, + { 0x3d, 0x0000 }, + { 0x3e, 0x007f }, + { 0x45, 0xe00f }, + { 0x4c, 0x5380 }, + { 0x4f, 0x0073 }, + { 0x52, 0x00d3 }, + { 0x53, 0xf0f0 }, + { 0x61, 0x0000 }, + { 0x62, 0x0001 }, + { 0x63, 0x00c3 }, + { 0x64, 0x0000 }, + { 0x65, 0x0000 }, + { 0x66, 0x0000 }, + { 0x6f, 0x8000 }, + { 0x70, 0x8000 }, + { 0x71, 0x8000 }, + { 0x72, 0x8000 }, + { 0x73, 0x1110 }, + { 0x74, 0x0e00 }, + { 0x75, 0x1505 }, + { 0x76, 0x0015 }, + { 0x77, 0x0c00 }, + { 0x78, 0x4000 }, + { 0x79, 0x0123 }, + { 0x7f, 0x1100 }, + { 0x80, 0x0000 }, + { 0x81, 0x0000 }, + { 0x82, 0x0000 }, + { 0x83, 0x0000 }, + { 0x84, 0x0000 }, + { 0x85, 0x0000 }, + { 0x86, 0x0008 }, + { 0x87, 0x0000 }, + { 0x88, 0x0000 }, + { 0x89, 0x0000 }, + { 0x8a, 0x0000 }, + { 0x8b, 0x0000 }, + { 0x8c, 0x0007 }, + { 0x8d, 0x0000 }, + { 0x8e, 0x0004 }, + { 0x8f, 0x1100 }, + { 0x90, 0x0646 }, + { 0x91, 0x0c06 }, + { 0x93, 0x0000 }, + { 0x94, 0x0000 }, + { 0x95, 0x0000 }, + { 0x97, 0x0000 }, + { 0x98, 0x0000 }, + { 0x99, 0x0000 }, + { 0x9a, 0x2184 }, + { 0x9b, 0x010a }, + { 0x9c, 0x0aea }, + { 0x9d, 0x000c }, + { 0x9e, 0x0400 }, + { 0xae, 0x7000 }, + { 0xaf, 0x0000 }, + { 0xb0, 0x6000 }, + { 0xb1, 0x0000 }, + { 0xb2, 0x0000 }, + { 0xb3, 0x001f }, + { 0xb4, 0x2206 }, + { 0xb5, 0x1f00 }, + { 0xb6, 0x0000 }, + { 0xb7, 0x0000 }, + { 0xbb, 0x0000 }, + { 0xbc, 0x0000 }, + { 0xbd, 0x0000 }, + { 0xbe, 0x0000 }, + { 0xbf, 0x0000 }, + { 0xc0, 0x0000 }, + { 0xc1, 0x0000 }, + { 0xc2, 0x0000 }, + { 0xcd, 0x0000 }, + { 0xce, 0x0000 }, + { 0xcf, 0x1813 }, + { 0xd0, 0x0690 }, + { 0xd1, 0x1c17 }, + { 0xd3, 0xb320 }, + { 0xd4, 0x0000 }, + { 0xd6, 0x0400 }, + { 0xd9, 0x0809 }, + { 0xda, 0x0000 }, + { 0xdb, 0x0001 }, + { 0xdc, 0x0049 }, + { 0xdd, 0x0009 }, + { 0xe6, 0x8000 }, + { 0xe7, 0x0000 }, + { 0xec, 0xb300 }, + { 0xed, 0x0000 }, + { 0xee, 0xb300 }, + { 0xef, 0x0000 }, + { 0xf8, 0x0000 }, + { 0xf9, 0x0000 }, + { 0xfa, 0x8010 }, + { 0xfb, 0x0033 }, + { 0xfc, 0x0080 }, +}; + +static bool rt5670_volatile_register(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rt5670_ranges); i++) { + if ((reg >= rt5670_ranges[i].window_start && + reg <= rt5670_ranges[i].window_start + + rt5670_ranges[i].window_len) || + (reg >= rt5670_ranges[i].range_min && + reg <= rt5670_ranges[i].range_max)) { + return true; + } + } + + switch (reg) { + case RT5670_RESET: + case RT5670_PDM_DATA_CTRL1: + case RT5670_PDM1_DATA_CTRL4: + case RT5670_PDM2_DATA_CTRL4: + case RT5670_PRIV_DATA: + case RT5670_ASRC_5: + case RT5670_CJ_CTRL1: + case RT5670_CJ_CTRL2: + case RT5670_CJ_CTRL3: + case RT5670_A_JD_CTRL1: + case RT5670_A_JD_CTRL2: + case RT5670_VAD_CTRL5: + case RT5670_ADC_EQ_CTRL1: + case RT5670_EQ_CTRL1: + case RT5670_ALC_CTRL_1: + case RT5670_IRQ_CTRL1: + case RT5670_IRQ_CTRL2: + case RT5670_INT_IRQ_ST: + case RT5670_IL_CMD: + case RT5670_DSP_CTRL1: + case RT5670_DSP_CTRL2: + case RT5670_DSP_CTRL3: + case RT5670_DSP_CTRL4: + case RT5670_DSP_CTRL5: + case RT5670_VENDOR_ID: + case RT5670_VENDOR_ID1: + case RT5670_VENDOR_ID2: + return true; + default: + return false; + } +} + +static bool rt5670_readable_register(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rt5670_ranges); i++) { + if ((reg >= rt5670_ranges[i].window_start && + reg <= rt5670_ranges[i].window_start + + rt5670_ranges[i].window_len) || + (reg >= rt5670_ranges[i].range_min && + reg <= rt5670_ranges[i].range_max)) { + return true; + } + } + + switch (reg) { + case RT5670_RESET: + case RT5670_HP_VOL: + case RT5670_LOUT1: + case RT5670_CJ_CTRL1: + case RT5670_CJ_CTRL2: + case RT5670_CJ_CTRL3: + case RT5670_IN2: + case RT5670_INL1_INR1_VOL: + case RT5670_DAC1_DIG_VOL: + case RT5670_DAC2_DIG_VOL: + case RT5670_DAC_CTRL: + case RT5670_STO1_ADC_DIG_VOL: + case RT5670_MONO_ADC_DIG_VOL: + case RT5670_STO2_ADC_DIG_VOL: + case RT5670_ADC_BST_VOL1: + case RT5670_ADC_BST_VOL2: + case RT5670_STO2_ADC_MIXER: + case RT5670_STO1_ADC_MIXER: + case RT5670_MONO_ADC_MIXER: + case RT5670_AD_DA_MIXER: + case RT5670_STO_DAC_MIXER: + case RT5670_DD_MIXER: + case RT5670_DIG_MIXER: + case RT5670_DSP_PATH1: + case RT5670_DSP_PATH2: + case RT5670_DIG_INF1_DATA: + case RT5670_DIG_INF2_DATA: + case RT5670_PDM_OUT_CTRL: + case RT5670_PDM_DATA_CTRL1: + case RT5670_PDM1_DATA_CTRL2: + case RT5670_PDM1_DATA_CTRL3: + case RT5670_PDM1_DATA_CTRL4: + case RT5670_PDM2_DATA_CTRL2: + case RT5670_PDM2_DATA_CTRL3: + case RT5670_PDM2_DATA_CTRL4: + case RT5670_REC_L1_MIXER: + case RT5670_REC_L2_MIXER: + case RT5670_REC_R1_MIXER: + case RT5670_REC_R2_MIXER: + case RT5670_HPO_MIXER: + case RT5670_MONO_MIXER: + case RT5670_OUT_L1_MIXER: + case RT5670_OUT_R1_MIXER: + case RT5670_LOUT_MIXER: + case RT5670_PWR_DIG1: + case RT5670_PWR_DIG2: + case RT5670_PWR_ANLG1: + case RT5670_PWR_ANLG2: + case RT5670_PWR_MIXER: + case RT5670_PWR_VOL: + case RT5670_PRIV_INDEX: + case RT5670_PRIV_DATA: + case RT5670_I2S4_SDP: + case RT5670_I2S1_SDP: + case RT5670_I2S2_SDP: + case RT5670_I2S3_SDP: + case RT5670_ADDA_CLK1: + case RT5670_ADDA_CLK2: + case RT5670_DMIC_CTRL1: + case RT5670_DMIC_CTRL2: + case RT5670_TDM_CTRL_1: + case RT5670_TDM_CTRL_2: + case RT5670_TDM_CTRL_3: + case RT5670_DSP_CLK: + case RT5670_GLB_CLK: + case RT5670_PLL_CTRL1: + case RT5670_PLL_CTRL2: + case RT5670_ASRC_1: + case RT5670_ASRC_2: + case RT5670_ASRC_3: + case RT5670_ASRC_4: + case RT5670_ASRC_5: + case RT5670_ASRC_7: + case RT5670_ASRC_8: + case RT5670_ASRC_9: + case RT5670_ASRC_10: + case RT5670_ASRC_11: + case RT5670_ASRC_12: + case RT5670_ASRC_13: + case RT5670_ASRC_14: + case RT5670_DEPOP_M1: + case RT5670_DEPOP_M2: + case RT5670_DEPOP_M3: + case RT5670_CHARGE_PUMP: + case RT5670_MICBIAS: + case RT5670_A_JD_CTRL1: + case RT5670_A_JD_CTRL2: + case RT5670_VAD_CTRL1: + case RT5670_VAD_CTRL2: + case RT5670_VAD_CTRL3: + case RT5670_VAD_CTRL4: + case RT5670_VAD_CTRL5: + case RT5670_ADC_EQ_CTRL1: + case RT5670_ADC_EQ_CTRL2: + case RT5670_EQ_CTRL1: + case RT5670_EQ_CTRL2: + case RT5670_ALC_DRC_CTRL1: + case RT5670_ALC_DRC_CTRL2: + case RT5670_ALC_CTRL_1: + case RT5670_ALC_CTRL_2: + case RT5670_ALC_CTRL_3: + case RT5670_JD_CTRL: + case RT5670_IRQ_CTRL1: + case RT5670_IRQ_CTRL2: + case RT5670_INT_IRQ_ST: + case RT5670_GPIO_CTRL1: + case RT5670_GPIO_CTRL2: + case RT5670_GPIO_CTRL3: + case RT5670_SCRABBLE_FUN: + case RT5670_SCRABBLE_CTRL: + case RT5670_BASE_BACK: + case RT5670_MP3_PLUS1: + case RT5670_MP3_PLUS2: + case RT5670_ADJ_HPF1: + case RT5670_ADJ_HPF2: + case RT5670_HP_CALIB_AMP_DET: + case RT5670_SV_ZCD1: + case RT5670_SV_ZCD2: + case RT5670_IL_CMD: + case RT5670_IL_CMD2: + case RT5670_IL_CMD3: + case RT5670_DRC_HL_CTRL1: + case RT5670_DRC_HL_CTRL2: + case RT5670_ADC_MONO_HP_CTRL1: + case RT5670_ADC_MONO_HP_CTRL2: + case RT5670_ADC_STO2_HP_CTRL1: + case RT5670_ADC_STO2_HP_CTRL2: + case RT5670_JD_CTRL3: + case RT5670_JD_CTRL4: + case RT5670_DIG_MISC: + case RT5670_DSP_CTRL1: + case RT5670_DSP_CTRL2: + case RT5670_DSP_CTRL3: + case RT5670_DSP_CTRL4: + case RT5670_DSP_CTRL5: + case RT5670_GEN_CTRL2: + case RT5670_GEN_CTRL3: + case RT5670_VENDOR_ID: + case RT5670_VENDOR_ID1: + case RT5670_VENDOR_ID2: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); +static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); + +/* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */ +static unsigned int bst_tlv[] = { + TLV_DB_RANGE_HEAD(7), + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0), + 3, 5, TLV_DB_SCALE_ITEM(3000, 500, 0), + 6, 6, TLV_DB_SCALE_ITEM(4400, 0, 0), + 7, 7, TLV_DB_SCALE_ITEM(5000, 0, 0), + 8, 8, TLV_DB_SCALE_ITEM(5200, 0, 0), +}; + +/* Interface data select */ +static const char * const rt5670_data_select[] = { + "Normal", "Swap", "left copy to right", "right copy to left" +}; + +static const SOC_ENUM_SINGLE_DECL(rt5670_if2_dac_enum, RT5670_DIG_INF1_DATA, + RT5670_IF2_DAC_SEL_SFT, rt5670_data_select); + +static const SOC_ENUM_SINGLE_DECL(rt5670_if2_adc_enum, RT5670_DIG_INF1_DATA, + RT5670_IF2_ADC_SEL_SFT, rt5670_data_select); + +static const struct snd_kcontrol_new rt5670_snd_controls[] = { + /* Headphone Output Volume */ + SOC_DOUBLE("HP Playback Switch", RT5670_HP_VOL, + RT5670_L_MUTE_SFT, RT5670_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_TLV("HP Playback Volume", RT5670_HP_VOL, + RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, + 39, 0, out_vol_tlv), + /* OUTPUT Control */ + SOC_DOUBLE("OUT Channel Switch", RT5670_LOUT1, + RT5670_VOL_L_SFT, RT5670_VOL_R_SFT, 1, 1), + SOC_DOUBLE_TLV("OUT Playback Volume", RT5670_LOUT1, + RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, 39, 1, out_vol_tlv), + /* DAC Digital Volume */ + SOC_DOUBLE("DAC2 Playback Switch", RT5670_DAC_CTRL, + RT5670_M_DAC_L2_VOL_SFT, RT5670_M_DAC_R2_VOL_SFT, 1, 1), + SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5670_DAC1_DIG_VOL, + RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, + 175, 0, dac_vol_tlv), + SOC_DOUBLE_TLV("Mono DAC Playback Volume", RT5670_DAC2_DIG_VOL, + RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, + 175, 0, dac_vol_tlv), + /* IN1/IN2 Control */ + SOC_SINGLE_TLV("IN1 Boost Volume", RT5670_CJ_CTRL1, + RT5670_BST_SFT1, 8, 0, bst_tlv), + SOC_SINGLE_TLV("IN2 Boost Volume", RT5670_IN2, + RT5670_BST_SFT1, 8, 0, bst_tlv), + /* INL/INR Volume Control */ + SOC_DOUBLE_TLV("IN Capture Volume", RT5670_INL1_INR1_VOL, + RT5670_INL_VOL_SFT, RT5670_INR_VOL_SFT, + 31, 1, in_vol_tlv), + /* ADC Digital Volume Control */ + SOC_DOUBLE("ADC Capture Switch", RT5670_STO1_ADC_DIG_VOL, + RT5670_L_MUTE_SFT, RT5670_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_TLV("ADC Capture Volume", RT5670_STO1_ADC_DIG_VOL, + RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, + 127, 0, adc_vol_tlv), + + SOC_DOUBLE_TLV("Mono ADC Capture Volume", RT5670_MONO_ADC_DIG_VOL, + RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, + 127, 0, adc_vol_tlv), + + /* ADC Boost Volume Control */ + SOC_DOUBLE_TLV("STO1 ADC Boost Gain Volume", RT5670_ADC_BST_VOL1, + RT5670_STO1_ADC_L_BST_SFT, RT5670_STO1_ADC_R_BST_SFT, + 3, 0, adc_bst_tlv), + + SOC_DOUBLE_TLV("STO2 ADC Boost Gain Volume", RT5670_ADC_BST_VOL1, + RT5670_STO2_ADC_L_BST_SFT, RT5670_STO2_ADC_R_BST_SFT, + 3, 0, adc_bst_tlv), + + SOC_ENUM("ADC IF2 Data Switch", rt5670_if2_adc_enum), + SOC_ENUM("DAC IF2 Data Switch", rt5670_if2_dac_enum), +}; + +/** + * set_dmic_clk - Set parameter of dmic. + * + * @w: DAPM widget. + * @kcontrol: The kcontrol of this widget. + * @event: Event id. + * + * Choose dmic clock between 1MHz and 3MHz. + * It is better for clock to approximate 3MHz. + */ +static int set_dmic_clk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + int idx = -EINVAL; + + idx = rl6231_calc_dmic_clk(rt5670->sysclk); + + if (idx < 0) + dev_err(codec->dev, "Failed to set DMIC clock\n"); + else + snd_soc_update_bits(codec, RT5670_DMIC_CTRL1, + RT5670_DMIC_CLK_MASK, idx << RT5670_DMIC_CLK_SFT); + return idx; +} + +static int is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + unsigned int val; + + val = snd_soc_read(source->codec, RT5670_GLB_CLK); + val &= RT5670_SCLK_SRC_MASK; + if (val == RT5670_SCLK_SRC_PLL1) + return 1; + else + return 0; +} + +static int is_using_asrc(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + unsigned int reg, shift, val; + + switch (source->shift) { + case 0: + reg = RT5670_ASRC_3; + shift = 0; + break; + case 1: + reg = RT5670_ASRC_3; + shift = 4; + break; + case 2: + reg = RT5670_ASRC_5; + shift = 12; + break; + case 3: + reg = RT5670_ASRC_2; + shift = 0; + break; + case 8: + reg = RT5670_ASRC_2; + shift = 4; + break; + case 9: + reg = RT5670_ASRC_2; + shift = 8; + break; + case 10: + reg = RT5670_ASRC_2; + shift = 12; + break; + default: + return 0; + } + + val = (snd_soc_read(source->codec, reg) >> shift) & 0xf; + switch (val) { + case 1: + case 2: + case 3: + case 4: + return 1; + default: + return 0; + } + +} + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt5670_sto1_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5670_STO1_ADC_MIXER, + RT5670_M_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5670_STO1_ADC_MIXER, + RT5670_M_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_sto1_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5670_STO1_ADC_MIXER, + RT5670_M_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5670_STO1_ADC_MIXER, + RT5670_M_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_sto2_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5670_STO2_ADC_MIXER, + RT5670_M_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5670_STO2_ADC_MIXER, + RT5670_M_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_sto2_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5670_STO2_ADC_MIXER, + RT5670_M_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5670_STO2_ADC_MIXER, + RT5670_M_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_mono_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5670_MONO_ADC_MIXER, + RT5670_M_MONO_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5670_MONO_ADC_MIXER, + RT5670_M_MONO_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_mono_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5670_MONO_ADC_MIXER, + RT5670_M_MONO_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5670_MONO_ADC_MIXER, + RT5670_M_MONO_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_dac_l_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5670_AD_DA_MIXER, + RT5670_M_ADCMIX_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5670_AD_DA_MIXER, + RT5670_M_DAC1_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_dac_r_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5670_AD_DA_MIXER, + RT5670_M_ADCMIX_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5670_AD_DA_MIXER, + RT5670_M_DAC1_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_sto_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5670_STO_DAC_MIXER, + RT5670_M_DAC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5670_STO_DAC_MIXER, + RT5670_M_DAC_L2_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5670_STO_DAC_MIXER, + RT5670_M_DAC_R1_STO_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_sto_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5670_STO_DAC_MIXER, + RT5670_M_DAC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5670_STO_DAC_MIXER, + RT5670_M_DAC_R2_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5670_STO_DAC_MIXER, + RT5670_M_DAC_L1_STO_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_mono_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5670_DD_MIXER, + RT5670_M_DAC_L1_MONO_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5670_DD_MIXER, + RT5670_M_DAC_L2_MONO_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5670_DD_MIXER, + RT5670_M_DAC_R2_MONO_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_mono_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5670_DD_MIXER, + RT5670_M_DAC_R1_MONO_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5670_DD_MIXER, + RT5670_M_DAC_R2_MONO_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5670_DD_MIXER, + RT5670_M_DAC_L2_MONO_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_dig_l_mix[] = { + SOC_DAPM_SINGLE("Sto DAC Mix L Switch", RT5670_DIG_MIXER, + RT5670_M_STO_L_DAC_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5670_DIG_MIXER, + RT5670_M_DAC_L2_DAC_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5670_DIG_MIXER, + RT5670_M_DAC_R2_DAC_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_dig_r_mix[] = { + SOC_DAPM_SINGLE("Sto DAC Mix R Switch", RT5670_DIG_MIXER, + RT5670_M_STO_R_DAC_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5670_DIG_MIXER, + RT5670_M_DAC_R2_DAC_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5670_DIG_MIXER, + RT5670_M_DAC_L2_DAC_R_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt5670_rec_l_mix[] = { + SOC_DAPM_SINGLE("INL Switch", RT5670_REC_L2_MIXER, + RT5670_M_IN_L_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5670_REC_L2_MIXER, + RT5670_M_BST2_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5670_REC_L2_MIXER, + RT5670_M_BST1_RM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_rec_r_mix[] = { + SOC_DAPM_SINGLE("INR Switch", RT5670_REC_R2_MIXER, + RT5670_M_IN_R_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5670_REC_R2_MIXER, + RT5670_M_BST2_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5670_REC_R2_MIXER, + RT5670_M_BST1_RM_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_out_l_mix[] = { + SOC_DAPM_SINGLE("BST1 Switch", RT5670_OUT_L1_MIXER, + RT5670_M_BST1_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("INL Switch", RT5670_OUT_L1_MIXER, + RT5670_M_IN_L_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5670_OUT_L1_MIXER, + RT5670_M_DAC_L2_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5670_OUT_L1_MIXER, + RT5670_M_DAC_L1_OM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_out_r_mix[] = { + SOC_DAPM_SINGLE("BST2 Switch", RT5670_OUT_R1_MIXER, + RT5670_M_BST2_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("INR Switch", RT5670_OUT_R1_MIXER, + RT5670_M_IN_R_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5670_OUT_R1_MIXER, + RT5670_M_DAC_R2_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5670_OUT_R1_MIXER, + RT5670_M_DAC_R1_OM_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_hpo_mix[] = { + SOC_DAPM_SINGLE("DAC1 Switch", RT5670_HPO_MIXER, + RT5670_M_DAC1_HM_SFT, 1, 1), + SOC_DAPM_SINGLE("HPVOL Switch", RT5670_HPO_MIXER, + RT5670_M_HPVOL_HM_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_hpvoll_mix[] = { + SOC_DAPM_SINGLE("DAC1 Switch", RT5670_HPO_MIXER, + RT5670_M_DACL1_HML_SFT, 1, 1), + SOC_DAPM_SINGLE("INL Switch", RT5670_HPO_MIXER, + RT5670_M_INL1_HML_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_hpvolr_mix[] = { + SOC_DAPM_SINGLE("DAC1 Switch", RT5670_HPO_MIXER, + RT5670_M_DACR1_HMR_SFT, 1, 1), + SOC_DAPM_SINGLE("INR Switch", RT5670_HPO_MIXER, + RT5670_M_INR1_HMR_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_lout_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5670_LOUT_MIXER, + RT5670_M_DAC_L1_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5670_LOUT_MIXER, + RT5670_M_DAC_R1_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("OUTMIX L Switch", RT5670_LOUT_MIXER, + RT5670_M_OV_L_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("OUTMIX R Switch", RT5670_LOUT_MIXER, + RT5670_M_OV_R_LM_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_hpl_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5670_HPO_MIXER, + RT5670_M_DACL1_HML_SFT, 1, 1), + SOC_DAPM_SINGLE("INL1 Switch", RT5670_HPO_MIXER, + RT5670_M_INL1_HML_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_hpr_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5670_HPO_MIXER, + RT5670_M_DACR1_HMR_SFT, 1, 1), + SOC_DAPM_SINGLE("INR1 Switch", RT5670_HPO_MIXER, + RT5670_M_INR1_HMR_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new lout_l_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5670_LOUT1, + RT5670_L_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new lout_r_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5670_LOUT1, + RT5670_R_MUTE_SFT, 1, 1); + +/* DAC1 L/R source */ /* MX-29 [9:8] [11:10] */ +static const char * const rt5670_dac1_src[] = { + "IF1 DAC", "IF2 DAC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_dac1l_enum, RT5670_AD_DA_MIXER, + RT5670_DAC1_L_SEL_SFT, rt5670_dac1_src); + +static const struct snd_kcontrol_new rt5670_dac1l_mux = + SOC_DAPM_ENUM("DAC1 L source", rt5670_dac1l_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_dac1r_enum, RT5670_AD_DA_MIXER, + RT5670_DAC1_R_SEL_SFT, rt5670_dac1_src); + +static const struct snd_kcontrol_new rt5670_dac1r_mux = + SOC_DAPM_ENUM("DAC1 R source", rt5670_dac1r_enum); + +/*DAC2 L/R source*/ /* MX-1B [6:4] [2:0] */ +/* TODO Use SOC_VALUE_ENUM_SINGLE_DECL */ +static const char * const rt5670_dac12_src[] = { + "IF1 DAC", "IF2 DAC", "IF3 DAC", "TxDC DAC", + "Bass", "VAD_ADC", "IF4 DAC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_dac2l_enum, RT5670_DAC_CTRL, + RT5670_DAC2_L_SEL_SFT, rt5670_dac12_src); + +static const struct snd_kcontrol_new rt5670_dac_l2_mux = + SOC_DAPM_ENUM("DAC2 L source", rt5670_dac2l_enum); + +static const char * const rt5670_dacr2_src[] = { + "IF1 DAC", "IF2 DAC", "IF3 DAC", "TxDC DAC", "TxDP ADC", "IF4 DAC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_dac2r_enum, RT5670_DAC_CTRL, + RT5670_DAC2_R_SEL_SFT, rt5670_dacr2_src); + +static const struct snd_kcontrol_new rt5670_dac_r2_mux = + SOC_DAPM_ENUM("DAC2 R source", rt5670_dac2r_enum); + +/*RxDP source*/ /* MX-2D [15:13] */ +static const char * const rt5670_rxdp_src[] = { + "IF2 DAC", "IF1 DAC", "STO1 ADC Mixer", "STO2 ADC Mixer", + "Mono ADC Mixer L", "Mono ADC Mixer R", "DAC1" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_rxdp_enum, RT5670_DSP_PATH1, + RT5670_RXDP_SEL_SFT, rt5670_rxdp_src); + +static const struct snd_kcontrol_new rt5670_rxdp_mux = + SOC_DAPM_ENUM("DAC2 L source", rt5670_rxdp_enum); + +/* MX-2D [1] [0] */ +static const char * const rt5670_dsp_bypass_src[] = { + "DSP", "Bypass" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_dsp_ul_enum, RT5670_DSP_PATH1, + RT5670_DSP_UL_SFT, rt5670_dsp_bypass_src); + +static const struct snd_kcontrol_new rt5670_dsp_ul_mux = + SOC_DAPM_ENUM("DSP UL source", rt5670_dsp_ul_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_dsp_dl_enum, RT5670_DSP_PATH1, + RT5670_DSP_DL_SFT, rt5670_dsp_bypass_src); + +static const struct snd_kcontrol_new rt5670_dsp_dl_mux = + SOC_DAPM_ENUM("DSP DL source", rt5670_dsp_dl_enum); + +/* Stereo2 ADC source */ +/* MX-26 [15] */ +static const char * const rt5670_stereo2_adc_lr_src[] = { + "L", "LR" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo2_adc_lr_enum, RT5670_STO2_ADC_MIXER, + RT5670_STO2_ADC_SRC_SFT, rt5670_stereo2_adc_lr_src); + +static const struct snd_kcontrol_new rt5670_sto2_adc_lr_mux = + SOC_DAPM_ENUM("Stereo2 ADC LR source", rt5670_stereo2_adc_lr_enum); + +/* Stereo1 ADC source */ +/* MX-27 MX-26 [12] */ +static const char * const rt5670_stereo_adc1_src[] = { + "DAC MIX", "ADC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo1_adc1_enum, RT5670_STO1_ADC_MIXER, + RT5670_ADC_1_SRC_SFT, rt5670_stereo_adc1_src); + +static const struct snd_kcontrol_new rt5670_sto_adc_l1_mux = + SOC_DAPM_ENUM("Stereo1 ADC L1 source", rt5670_stereo1_adc1_enum); + +static const struct snd_kcontrol_new rt5670_sto_adc_r1_mux = + SOC_DAPM_ENUM("Stereo1 ADC R1 source", rt5670_stereo1_adc1_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo2_adc1_enum, RT5670_STO2_ADC_MIXER, + RT5670_ADC_1_SRC_SFT, rt5670_stereo_adc1_src); + +static const struct snd_kcontrol_new rt5670_sto2_adc_l1_mux = + SOC_DAPM_ENUM("Stereo2 ADC L1 source", rt5670_stereo2_adc1_enum); + +static const struct snd_kcontrol_new rt5670_sto2_adc_r1_mux = + SOC_DAPM_ENUM("Stereo2 ADC R1 source", rt5670_stereo2_adc1_enum); + +/* MX-27 MX-26 [11] */ +static const char * const rt5670_stereo_adc2_src[] = { + "DAC MIX", "DMIC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo1_adc2_enum, RT5670_STO1_ADC_MIXER, + RT5670_ADC_2_SRC_SFT, rt5670_stereo_adc2_src); + +static const struct snd_kcontrol_new rt5670_sto_adc_l2_mux = + SOC_DAPM_ENUM("Stereo1 ADC L2 source", rt5670_stereo1_adc2_enum); + +static const struct snd_kcontrol_new rt5670_sto_adc_r2_mux = + SOC_DAPM_ENUM("Stereo1 ADC R2 source", rt5670_stereo1_adc2_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo2_adc2_enum, RT5670_STO2_ADC_MIXER, + RT5670_ADC_2_SRC_SFT, rt5670_stereo_adc2_src); + +static const struct snd_kcontrol_new rt5670_sto2_adc_l2_mux = + SOC_DAPM_ENUM("Stereo2 ADC L2 source", rt5670_stereo2_adc2_enum); + +static const struct snd_kcontrol_new rt5670_sto2_adc_r2_mux = + SOC_DAPM_ENUM("Stereo2 ADC R2 source", rt5670_stereo2_adc2_enum); + +/* MX-27 MX26 [10] */ +static const char * const rt5670_stereo_adc_src[] = { + "ADC1L ADC2R", "ADC3" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo1_adc_enum, RT5670_STO1_ADC_MIXER, + RT5670_ADC_SRC_SFT, rt5670_stereo_adc_src); + +static const struct snd_kcontrol_new rt5670_sto_adc_mux = + SOC_DAPM_ENUM("Stereo1 ADC source", rt5670_stereo1_adc_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo2_adc_enum, RT5670_STO2_ADC_MIXER, + RT5670_ADC_SRC_SFT, rt5670_stereo_adc_src); + +static const struct snd_kcontrol_new rt5670_sto2_adc_mux = + SOC_DAPM_ENUM("Stereo2 ADC source", rt5670_stereo2_adc_enum); + +/* MX-27 MX-26 [9:8] */ +static const char * const rt5670_stereo_dmic_src[] = { + "DMIC1", "DMIC2", "DMIC3" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo1_dmic_enum, RT5670_STO1_ADC_MIXER, + RT5670_DMIC_SRC_SFT, rt5670_stereo_dmic_src); + +static const struct snd_kcontrol_new rt5670_sto1_dmic_mux = + SOC_DAPM_ENUM("Stereo1 DMIC source", rt5670_stereo1_dmic_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo2_dmic_enum, RT5670_STO2_ADC_MIXER, + RT5670_DMIC_SRC_SFT, rt5670_stereo_dmic_src); + +static const struct snd_kcontrol_new rt5670_sto2_dmic_mux = + SOC_DAPM_ENUM("Stereo2 DMIC source", rt5670_stereo2_dmic_enum); + +/* MX-27 [0] */ +static const char * const rt5670_stereo_dmic3_src[] = { + "DMIC3", "PDM ADC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo_dmic3_enum, RT5670_STO1_ADC_MIXER, + RT5670_DMIC3_SRC_SFT, rt5670_stereo_dmic3_src); + +static const struct snd_kcontrol_new rt5670_sto_dmic3_mux = + SOC_DAPM_ENUM("Stereo DMIC3 source", rt5670_stereo_dmic3_enum); + +/* Mono ADC source */ +/* MX-28 [12] */ +static const char * const rt5670_mono_adc_l1_src[] = { + "Mono DAC MIXL", "ADC1" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_mono_adc_l1_enum, RT5670_MONO_ADC_MIXER, + RT5670_MONO_ADC_L1_SRC_SFT, rt5670_mono_adc_l1_src); + +static const struct snd_kcontrol_new rt5670_mono_adc_l1_mux = + SOC_DAPM_ENUM("Mono ADC1 left source", rt5670_mono_adc_l1_enum); +/* MX-28 [11] */ +static const char * const rt5670_mono_adc_l2_src[] = { + "Mono DAC MIXL", "DMIC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_mono_adc_l2_enum, RT5670_MONO_ADC_MIXER, + RT5670_MONO_ADC_L2_SRC_SFT, rt5670_mono_adc_l2_src); + +static const struct snd_kcontrol_new rt5670_mono_adc_l2_mux = + SOC_DAPM_ENUM("Mono ADC2 left source", rt5670_mono_adc_l2_enum); + +/* MX-28 [9:8] */ +static const char * const rt5670_mono_dmic_src[] = { + "DMIC1", "DMIC2", "DMIC3" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_mono_dmic_l_enum, RT5670_MONO_ADC_MIXER, + RT5670_MONO_DMIC_L_SRC_SFT, rt5670_mono_dmic_src); + +static const struct snd_kcontrol_new rt5670_mono_dmic_l_mux = + SOC_DAPM_ENUM("Mono DMIC left source", rt5670_mono_dmic_l_enum); +/* MX-28 [1:0] */ +static const SOC_ENUM_SINGLE_DECL( + rt5670_mono_dmic_r_enum, RT5670_MONO_ADC_MIXER, + RT5670_MONO_DMIC_R_SRC_SFT, rt5670_mono_dmic_src); + +static const struct snd_kcontrol_new rt5670_mono_dmic_r_mux = + SOC_DAPM_ENUM("Mono DMIC Right source", rt5670_mono_dmic_r_enum); +/* MX-28 [4] */ +static const char * const rt5670_mono_adc_r1_src[] = { + "Mono DAC MIXR", "ADC2" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_mono_adc_r1_enum, RT5670_MONO_ADC_MIXER, + RT5670_MONO_ADC_R1_SRC_SFT, rt5670_mono_adc_r1_src); + +static const struct snd_kcontrol_new rt5670_mono_adc_r1_mux = + SOC_DAPM_ENUM("Mono ADC1 right source", rt5670_mono_adc_r1_enum); +/* MX-28 [3] */ +static const char * const rt5670_mono_adc_r2_src[] = { + "Mono DAC MIXR", "DMIC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_mono_adc_r2_enum, RT5670_MONO_ADC_MIXER, + RT5670_MONO_ADC_R2_SRC_SFT, rt5670_mono_adc_r2_src); + +static const struct snd_kcontrol_new rt5670_mono_adc_r2_mux = + SOC_DAPM_ENUM("Mono ADC2 right source", rt5670_mono_adc_r2_enum); + +/* MX-2D [3:2] */ +static const char * const rt5670_txdp_slot_src[] = { + "Slot 0-1", "Slot 2-3", "Slot 4-5", "Slot 6-7" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_txdp_slot_enum, RT5670_DSP_PATH1, + RT5670_TXDP_SLOT_SEL_SFT, rt5670_txdp_slot_src); + +static const struct snd_kcontrol_new rt5670_txdp_slot_mux = + SOC_DAPM_ENUM("TxDP Slot source", rt5670_txdp_slot_enum); + +/* MX-2F [15] */ +static const char * const rt5670_if1_adc2_in_src[] = { + "IF_ADC2", "VAD_ADC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_if1_adc2_in_enum, RT5670_DIG_INF1_DATA, + RT5670_IF1_ADC2_IN_SFT, rt5670_if1_adc2_in_src); + +static const struct snd_kcontrol_new rt5670_if1_adc2_in_mux = + SOC_DAPM_ENUM("IF1 ADC2 IN source", rt5670_if1_adc2_in_enum); + +/* MX-2F [14:12] */ +static const char * const rt5670_if2_adc_in_src[] = { + "IF_ADC1", "IF_ADC2", "IF_ADC3", "TxDC_DAC", "TxDP_ADC", "VAD_ADC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_if2_adc_in_enum, RT5670_DIG_INF1_DATA, + RT5670_IF2_ADC_IN_SFT, rt5670_if2_adc_in_src); + +static const struct snd_kcontrol_new rt5670_if2_adc_in_mux = + SOC_DAPM_ENUM("IF2 ADC IN source", rt5670_if2_adc_in_enum); + +/* MX-30 [5:4] */ +static const char * const rt5670_if4_adc_in_src[] = { + "IF_ADC1", "IF_ADC2", "IF_ADC3" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_if4_adc_in_enum, RT5670_DIG_INF2_DATA, + RT5670_IF4_ADC_IN_SFT, rt5670_if4_adc_in_src); + +static const struct snd_kcontrol_new rt5670_if4_adc_in_mux = + SOC_DAPM_ENUM("IF4 ADC IN source", rt5670_if4_adc_in_enum); + +/* MX-31 [15] [13] [11] [9] */ +static const char * const rt5670_pdm_src[] = { + "Mono DAC", "Stereo DAC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_pdm1_l_enum, RT5670_PDM_OUT_CTRL, + RT5670_PDM1_L_SFT, rt5670_pdm_src); + +static const struct snd_kcontrol_new rt5670_pdm1_l_mux = + SOC_DAPM_ENUM("PDM1 L source", rt5670_pdm1_l_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_pdm1_r_enum, RT5670_PDM_OUT_CTRL, + RT5670_PDM1_R_SFT, rt5670_pdm_src); + +static const struct snd_kcontrol_new rt5670_pdm1_r_mux = + SOC_DAPM_ENUM("PDM1 R source", rt5670_pdm1_r_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_pdm2_l_enum, RT5670_PDM_OUT_CTRL, + RT5670_PDM2_L_SFT, rt5670_pdm_src); + +static const struct snd_kcontrol_new rt5670_pdm2_l_mux = + SOC_DAPM_ENUM("PDM2 L source", rt5670_pdm2_l_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_pdm2_r_enum, RT5670_PDM_OUT_CTRL, + RT5670_PDM2_R_SFT, rt5670_pdm_src); + +static const struct snd_kcontrol_new rt5670_pdm2_r_mux = + SOC_DAPM_ENUM("PDM2 R source", rt5670_pdm2_r_enum); + +/* MX-FA [12] */ +static const char * const rt5670_if1_adc1_in1_src[] = { + "IF_ADC1", "IF1_ADC3" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_if1_adc1_in1_enum, RT5670_DIG_MISC, + RT5670_IF1_ADC1_IN1_SFT, rt5670_if1_adc1_in1_src); + +static const struct snd_kcontrol_new rt5670_if1_adc1_in1_mux = + SOC_DAPM_ENUM("IF1 ADC1 IN1 source", rt5670_if1_adc1_in1_enum); + +/* MX-FA [11] */ +static const char * const rt5670_if1_adc1_in2_src[] = { + "IF1_ADC1_IN1", "IF1_ADC4" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_if1_adc1_in2_enum, RT5670_DIG_MISC, + RT5670_IF1_ADC1_IN2_SFT, rt5670_if1_adc1_in2_src); + +static const struct snd_kcontrol_new rt5670_if1_adc1_in2_mux = + SOC_DAPM_ENUM("IF1 ADC1 IN2 source", rt5670_if1_adc1_in2_enum); + +/* MX-FA [10] */ +static const char * const rt5670_if1_adc2_in1_src[] = { + "IF1_ADC2_IN", "IF1_ADC4" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_if1_adc2_in1_enum, RT5670_DIG_MISC, + RT5670_IF1_ADC2_IN1_SFT, rt5670_if1_adc2_in1_src); + +static const struct snd_kcontrol_new rt5670_if1_adc2_in1_mux = + SOC_DAPM_ENUM("IF1 ADC2 IN1 source", rt5670_if1_adc2_in1_enum); + +/* MX-9D [9:8] */ +static const char * const rt5670_vad_adc_src[] = { + "Sto1 ADC L", "Mono ADC L", "Mono ADC R", "Sto2 ADC L" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_vad_adc_enum, RT5670_VAD_CTRL4, + RT5670_VAD_SEL_SFT, rt5670_vad_adc_src); + +static const struct snd_kcontrol_new rt5670_vad_adc_mux = + SOC_DAPM_ENUM("VAD ADC source", rt5670_vad_adc_enum); + +static int rt5670_hp_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(rt5670->regmap, RT5670_CHARGE_PUMP, + RT5670_PM_HP_MASK, RT5670_PM_HP_HV); + regmap_update_bits(rt5670->regmap, RT5670_GEN_CTRL2, + 0x0400, 0x0400); + /* headphone amp power on */ + regmap_update_bits(rt5670->regmap, RT5670_PWR_ANLG1, + RT5670_PWR_HA | RT5670_PWR_FV1 | + RT5670_PWR_FV2, RT5670_PWR_HA | + RT5670_PWR_FV1 | RT5670_PWR_FV2); + /* depop parameters */ + regmap_write(rt5670->regmap, RT5670_DEPOP_M2, 0x3100); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x8009); + regmap_write(rt5670->regmap, RT5670_PR_BASE + + RT5670_HP_DCC_INT1, 0x9f00); + mdelay(20); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x8019); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x0004); + msleep(30); + break; + default: + return 0; + } + + return 0; +} + +static int rt5670_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* headphone unmute sequence */ + regmap_write(rt5670->regmap, RT5670_PR_BASE + + RT5670_MAMP_INT_REG2, 0xb400); + regmap_write(rt5670->regmap, RT5670_DEPOP_M3, 0x0772); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x805d); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x831d); + regmap_update_bits(rt5670->regmap, RT5670_GEN_CTRL2, + 0x0300, 0x0300); + regmap_update_bits(rt5670->regmap, RT5670_HP_VOL, + RT5670_L_MUTE | RT5670_R_MUTE, 0); + msleep(80); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x8019); + break; + + case SND_SOC_DAPM_PRE_PMD: + /* headphone mute sequence */ + regmap_write(rt5670->regmap, RT5670_PR_BASE + + RT5670_MAMP_INT_REG2, 0xb400); + regmap_write(rt5670->regmap, RT5670_DEPOP_M3, 0x0772); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x803d); + mdelay(10); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x831d); + mdelay(10); + regmap_update_bits(rt5670->regmap, RT5670_HP_VOL, + RT5670_L_MUTE | RT5670_R_MUTE, + RT5670_L_MUTE | RT5670_R_MUTE); + msleep(20); + regmap_update_bits(rt5670->regmap, + RT5670_GEN_CTRL2, 0x0300, 0x0); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x8019); + regmap_write(rt5670->regmap, RT5670_DEPOP_M3, 0x0707); + regmap_write(rt5670->regmap, RT5670_PR_BASE + + RT5670_MAMP_INT_REG2, 0xfc00); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5670_bst1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5670_PWR_ANLG2, + RT5670_PWR_BST1_P, RT5670_PWR_BST1_P); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5670_PWR_ANLG2, + RT5670_PWR_BST1_P, 0); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5670_bst2_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5670_PWR_ANLG2, + RT5670_PWR_BST2_P, RT5670_PWR_BST2_P); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5670_PWR_ANLG2, + RT5670_PWR_BST2_P, 0); + break; + + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt5670_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("PLL1", RT5670_PWR_ANLG2, + RT5670_PWR_PLL_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S DSP", RT5670_PWR_DIG2, + RT5670_PWR_I2S_DSP_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Det Power", RT5670_PWR_VOL, + RT5670_PWR_MIC_DET_BIT, 0, NULL, 0), + + /* ASRC */ + SND_SOC_DAPM_SUPPLY_S("I2S1 ASRC", 1, RT5670_ASRC_1, + 11, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("I2S2 ASRC", 1, RT5670_ASRC_1, + 12, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DAC STO ASRC", 1, RT5670_ASRC_1, + 10, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DAC MONO L ASRC", 1, RT5670_ASRC_1, + 9, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DAC MONO R ASRC", 1, RT5670_ASRC_1, + 8, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC STO1 ASRC", 1, RT5670_ASRC_1, + 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC STO2 ASRC", 1, RT5670_ASRC_1, + 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC MONO L ASRC", 1, RT5670_ASRC_1, + 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC MONO R ASRC", 1, RT5670_ASRC_1, + 0, 0, NULL, 0), + + /* Input Side */ + /* micbias */ + SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5670_PWR_ANLG2, + RT5670_PWR_MB1_BIT, 0, NULL, 0), + + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC L1"), + SND_SOC_DAPM_INPUT("DMIC R1"), + SND_SOC_DAPM_INPUT("DMIC L2"), + SND_SOC_DAPM_INPUT("DMIC R2"), + SND_SOC_DAPM_INPUT("DMIC L3"), + SND_SOC_DAPM_INPUT("DMIC R3"), + + SND_SOC_DAPM_INPUT("IN1P"), + SND_SOC_DAPM_INPUT("IN1N"), + SND_SOC_DAPM_INPUT("IN2P"), + SND_SOC_DAPM_INPUT("IN2N"), + + SND_SOC_DAPM_PGA("DMIC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DMIC2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DMIC3", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0, + set_dmic_clk, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY("DMIC1 Power", RT5670_DMIC_CTRL1, + RT5670_DMIC_1_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC2 Power", RT5670_DMIC_CTRL1, + RT5670_DMIC_2_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC3 Power", RT5670_DMIC_CTRL1, + RT5670_DMIC_3_EN_SFT, 0, NULL, 0), + /* Boost */ + SND_SOC_DAPM_PGA_E("BST1", RT5670_PWR_ANLG2, RT5670_PWR_BST1_BIT, + 0, NULL, 0, rt5670_bst1_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_E("BST2", RT5670_PWR_ANLG2, RT5670_PWR_BST2_BIT, + 0, NULL, 0, rt5670_bst2_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + /* Input Volume */ + SND_SOC_DAPM_PGA("INL VOL", RT5670_PWR_VOL, + RT5670_PWR_IN_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("INR VOL", RT5670_PWR_VOL, + RT5670_PWR_IN_R_BIT, 0, NULL, 0), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIXL", RT5670_PWR_MIXER, RT5670_PWR_RM_L_BIT, 0, + rt5670_rec_l_mix, ARRAY_SIZE(rt5670_rec_l_mix)), + SND_SOC_DAPM_MIXER("RECMIXR", RT5670_PWR_MIXER, RT5670_PWR_RM_R_BIT, 0, + rt5670_rec_r_mix, ARRAY_SIZE(rt5670_rec_r_mix)), + /* ADCs */ + SND_SOC_DAPM_ADC("ADC 1", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC 2", NULL, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_PGA("ADC 1_2", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("ADC 1 power", RT5670_PWR_DIG1, + RT5670_PWR_ADC_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC 2 power", RT5670_PWR_DIG1, + RT5670_PWR_ADC_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC clock", RT5670_PR_BASE + + RT5670_CHOP_DAC_ADC, 12, 0, NULL, 0), + /* ADC Mux */ + SND_SOC_DAPM_MUX("Stereo1 DMIC Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto1_dmic_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto_adc_l2_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto_adc_r2_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto_adc_l1_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto_adc_r1_mux), + SND_SOC_DAPM_MUX("Stereo2 DMIC Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto2_dmic_mux), + SND_SOC_DAPM_MUX("Stereo2 ADC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto2_adc_l2_mux), + SND_SOC_DAPM_MUX("Stereo2 ADC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto2_adc_r2_mux), + SND_SOC_DAPM_MUX("Stereo2 ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto2_adc_l1_mux), + SND_SOC_DAPM_MUX("Stereo2 ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto2_adc_r1_mux), + SND_SOC_DAPM_MUX("Stereo2 ADC LR Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto2_adc_lr_mux), + SND_SOC_DAPM_MUX("Mono DMIC L Mux", SND_SOC_NOPM, 0, 0, + &rt5670_mono_dmic_l_mux), + SND_SOC_DAPM_MUX("Mono DMIC R Mux", SND_SOC_NOPM, 0, 0, + &rt5670_mono_dmic_r_mux), + SND_SOC_DAPM_MUX("Mono ADC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_mono_adc_l2_mux), + SND_SOC_DAPM_MUX("Mono ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_mono_adc_l1_mux), + SND_SOC_DAPM_MUX("Mono ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_mono_adc_r1_mux), + SND_SOC_DAPM_MUX("Mono ADC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_mono_adc_r2_mux), + /* ADC Mixer */ + SND_SOC_DAPM_SUPPLY("ADC Stereo1 Filter", RT5670_PWR_DIG2, + RT5670_PWR_ADC_S1F_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC Stereo2 Filter", RT5670_PWR_DIG2, + RT5670_PWR_ADC_S2F_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Sto1 ADC MIXL", RT5670_STO1_ADC_DIG_VOL, + RT5670_L_MUTE_SFT, 1, rt5670_sto1_adc_l_mix, + ARRAY_SIZE(rt5670_sto1_adc_l_mix)), + SND_SOC_DAPM_MIXER("Sto1 ADC MIXR", RT5670_STO1_ADC_DIG_VOL, + RT5670_R_MUTE_SFT, 1, rt5670_sto1_adc_r_mix, + ARRAY_SIZE(rt5670_sto1_adc_r_mix)), + SND_SOC_DAPM_MIXER("Sto2 ADC MIXL", SND_SOC_NOPM, 0, 0, + rt5670_sto2_adc_l_mix, + ARRAY_SIZE(rt5670_sto2_adc_l_mix)), + SND_SOC_DAPM_MIXER("Sto2 ADC MIXR", SND_SOC_NOPM, 0, 0, + rt5670_sto2_adc_r_mix, + ARRAY_SIZE(rt5670_sto2_adc_r_mix)), + SND_SOC_DAPM_SUPPLY("ADC Mono Left Filter", RT5670_PWR_DIG2, + RT5670_PWR_ADC_MF_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Mono ADC MIXL", RT5670_MONO_ADC_DIG_VOL, + RT5670_L_MUTE_SFT, 1, rt5670_mono_adc_l_mix, + ARRAY_SIZE(rt5670_mono_adc_l_mix)), + SND_SOC_DAPM_SUPPLY("ADC Mono Right Filter", RT5670_PWR_DIG2, + RT5670_PWR_ADC_MF_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Mono ADC MIXR", RT5670_MONO_ADC_DIG_VOL, + RT5670_R_MUTE_SFT, 1, rt5670_mono_adc_r_mix, + ARRAY_SIZE(rt5670_mono_adc_r_mix)), + + /* ADC PGA */ + SND_SOC_DAPM_PGA("Stereo1 ADC MIXL", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo1 ADC MIXR", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo2 ADC MIXL", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo2 ADC MIXR", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Sto2 ADC LR MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo1 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo2 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mono ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("VAD_ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF_ADC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF_ADC2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF_ADC3", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1_ADC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1_ADC2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1_ADC3", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1_ADC4", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* DSP */ + SND_SOC_DAPM_PGA("TxDP_ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("TxDP_ADC_L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("TxDP_ADC_R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("TxDC_DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_MUX("TDM Data Mux", SND_SOC_NOPM, 0, 0, + &rt5670_txdp_slot_mux), + + SND_SOC_DAPM_MUX("DSP UL Mux", SND_SOC_NOPM, 0, 0, + &rt5670_dsp_ul_mux), + SND_SOC_DAPM_MUX("DSP DL Mux", SND_SOC_NOPM, 0, 0, + &rt5670_dsp_dl_mux), + + SND_SOC_DAPM_MUX("RxDP Mux", SND_SOC_NOPM, 0, 0, + &rt5670_rxdp_mux), + + /* IF2 Mux */ + SND_SOC_DAPM_MUX("IF2 ADC Mux", SND_SOC_NOPM, 0, 0, + &rt5670_if2_adc_in_mux), + + /* Digital Interface */ + SND_SOC_DAPM_SUPPLY("I2S1", RT5670_PWR_DIG1, + RT5670_PWR_I2S1_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1 L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1 R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC2 L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC2 R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S2", RT5670_PWR_DIG1, + RT5670_PWR_I2S2_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 ADC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 ADC R", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Digital Interface Select */ + SND_SOC_DAPM_MUX("IF1 ADC1 IN1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_if1_adc1_in1_mux), + SND_SOC_DAPM_MUX("IF1 ADC1 IN2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_if1_adc1_in2_mux), + SND_SOC_DAPM_MUX("IF1 ADC2 IN Mux", SND_SOC_NOPM, 0, 0, + &rt5670_if1_adc2_in_mux), + SND_SOC_DAPM_MUX("IF1 ADC2 IN1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_if1_adc2_in1_mux), + SND_SOC_DAPM_MUX("VAD ADC Mux", SND_SOC_NOPM, 0, 0, + &rt5670_vad_adc_mux), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, + RT5670_GPIO_CTRL1, RT5670_I2S2_PIN_SFT, 1), + + /* Audio DSP */ + SND_SOC_DAPM_PGA("Audio DSP", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Output Side */ + /* DAC mixer before sound effect */ + SND_SOC_DAPM_MIXER("DAC1 MIXL", SND_SOC_NOPM, 0, 0, + rt5670_dac_l_mix, ARRAY_SIZE(rt5670_dac_l_mix)), + SND_SOC_DAPM_MIXER("DAC1 MIXR", SND_SOC_NOPM, 0, 0, + rt5670_dac_r_mix, ARRAY_SIZE(rt5670_dac_r_mix)), + SND_SOC_DAPM_PGA("DAC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* DAC2 channel Mux */ + SND_SOC_DAPM_MUX("DAC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_dac_l2_mux), + SND_SOC_DAPM_MUX("DAC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_dac_r2_mux), + SND_SOC_DAPM_PGA("DAC L2 Volume", RT5670_PWR_DIG1, + RT5670_PWR_DAC_L2_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("DAC R2 Volume", RT5670_PWR_DIG1, + RT5670_PWR_DAC_R2_BIT, 0, NULL, 0), + + SND_SOC_DAPM_MUX("DAC1 L Mux", SND_SOC_NOPM, 0, 0, &rt5670_dac1l_mux), + SND_SOC_DAPM_MUX("DAC1 R Mux", SND_SOC_NOPM, 0, 0, &rt5670_dac1r_mux), + + /* DAC Mixer */ + SND_SOC_DAPM_SUPPLY("DAC Stereo1 Filter", RT5670_PWR_DIG2, + RT5670_PWR_DAC_S1F_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC Mono Left Filter", RT5670_PWR_DIG2, + RT5670_PWR_DAC_MF_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC Mono Right Filter", RT5670_PWR_DIG2, + RT5670_PWR_DAC_MF_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5670_sto_dac_l_mix, + ARRAY_SIZE(rt5670_sto_dac_l_mix)), + SND_SOC_DAPM_MIXER("Stereo DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5670_sto_dac_r_mix, + ARRAY_SIZE(rt5670_sto_dac_r_mix)), + SND_SOC_DAPM_MIXER("Mono DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5670_mono_dac_l_mix, + ARRAY_SIZE(rt5670_mono_dac_l_mix)), + SND_SOC_DAPM_MIXER("Mono DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5670_mono_dac_r_mix, + ARRAY_SIZE(rt5670_mono_dac_r_mix)), + SND_SOC_DAPM_MIXER("DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5670_dig_l_mix, + ARRAY_SIZE(rt5670_dig_l_mix)), + SND_SOC_DAPM_MIXER("DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5670_dig_r_mix, + ARRAY_SIZE(rt5670_dig_r_mix)), + + /* DACs */ + SND_SOC_DAPM_SUPPLY("DAC L1 Power", RT5670_PWR_DIG1, + RT5670_PWR_DAC_L1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC R1 Power", RT5670_PWR_DIG1, + RT5670_PWR_DAC_R1_BIT, 0, NULL, 0), + SND_SOC_DAPM_DAC("DAC L1", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC R1", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC L2", NULL, RT5670_PWR_DIG1, + RT5670_PWR_DAC_L2_BIT, 0), + + SND_SOC_DAPM_DAC("DAC R2", NULL, RT5670_PWR_DIG1, + RT5670_PWR_DAC_R2_BIT, 0), + /* OUT Mixer */ + + SND_SOC_DAPM_MIXER("OUT MIXL", RT5670_PWR_MIXER, RT5670_PWR_OM_L_BIT, + 0, rt5670_out_l_mix, ARRAY_SIZE(rt5670_out_l_mix)), + SND_SOC_DAPM_MIXER("OUT MIXR", RT5670_PWR_MIXER, RT5670_PWR_OM_R_BIT, + 0, rt5670_out_r_mix, ARRAY_SIZE(rt5670_out_r_mix)), + /* Ouput Volume */ + SND_SOC_DAPM_MIXER("HPOVOL MIXL", RT5670_PWR_VOL, + RT5670_PWR_HV_L_BIT, 0, + rt5670_hpvoll_mix, ARRAY_SIZE(rt5670_hpvoll_mix)), + SND_SOC_DAPM_MIXER("HPOVOL MIXR", RT5670_PWR_VOL, + RT5670_PWR_HV_R_BIT, 0, + rt5670_hpvolr_mix, ARRAY_SIZE(rt5670_hpvolr_mix)), + SND_SOC_DAPM_PGA("DAC 1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DAC 2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("HPOVOL", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* HPO/LOUT/Mono Mixer */ + SND_SOC_DAPM_MIXER("HPO MIX", SND_SOC_NOPM, 0, 0, + rt5670_hpo_mix, ARRAY_SIZE(rt5670_hpo_mix)), + SND_SOC_DAPM_MIXER("LOUT MIX", RT5670_PWR_ANLG1, RT5670_PWR_LM_BIT, + 0, rt5670_lout_mix, ARRAY_SIZE(rt5670_lout_mix)), + SND_SOC_DAPM_SUPPLY_S("Improve HP Amp Drv", 1, SND_SOC_NOPM, 0, 0, + rt5670_hp_power_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_SUPPLY("HP L Amp", RT5670_PWR_ANLG1, + RT5670_PWR_HP_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("HP R Amp", RT5670_PWR_ANLG1, + RT5670_PWR_HP_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, + rt5670_hp_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SWITCH("LOUT L Playback", SND_SOC_NOPM, 0, 0, + &lout_l_enable_control), + SND_SOC_DAPM_SWITCH("LOUT R Playback", SND_SOC_NOPM, 0, 0, + &lout_r_enable_control), + SND_SOC_DAPM_PGA("LOUT Amp", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* PDM */ + SND_SOC_DAPM_SUPPLY("PDM1 Power", RT5670_PWR_DIG2, + RT5670_PWR_PDM1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PDM2 Power", RT5670_PWR_DIG2, + RT5670_PWR_PDM2_BIT, 0, NULL, 0), + + SND_SOC_DAPM_MUX("PDM1 L Mux", RT5670_PDM_OUT_CTRL, + RT5670_M_PDM1_L_SFT, 1, &rt5670_pdm1_l_mux), + SND_SOC_DAPM_MUX("PDM1 R Mux", RT5670_PDM_OUT_CTRL, + RT5670_M_PDM1_R_SFT, 1, &rt5670_pdm1_r_mux), + SND_SOC_DAPM_MUX("PDM2 L Mux", RT5670_PDM_OUT_CTRL, + RT5670_M_PDM2_L_SFT, 1, &rt5670_pdm2_l_mux), + SND_SOC_DAPM_MUX("PDM2 R Mux", RT5670_PDM_OUT_CTRL, + RT5670_M_PDM2_R_SFT, 1, &rt5670_pdm2_r_mux), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), + SND_SOC_DAPM_OUTPUT("LOUTL"), + SND_SOC_DAPM_OUTPUT("LOUTR"), + SND_SOC_DAPM_OUTPUT("PDM1L"), + SND_SOC_DAPM_OUTPUT("PDM1R"), + SND_SOC_DAPM_OUTPUT("PDM2L"), + SND_SOC_DAPM_OUTPUT("PDM2R"), +}; + +static const struct snd_soc_dapm_route rt5670_dapm_routes[] = { + { "ADC Stereo1 Filter", NULL, "ADC STO1 ASRC", is_using_asrc }, + { "ADC Stereo2 Filter", NULL, "ADC STO2 ASRC", is_using_asrc }, + { "ADC Mono Left Filter", NULL, "ADC MONO L ASRC", is_using_asrc }, + { "ADC Mono Right Filter", NULL, "ADC MONO R ASRC", is_using_asrc }, + { "DAC Mono Left Filter", NULL, "DAC MONO L ASRC", is_using_asrc }, + { "DAC Mono Right Filter", NULL, "DAC MONO R ASRC", is_using_asrc }, + { "DAC Stereo1 Filter", NULL, "DAC STO ASRC", is_using_asrc }, + + { "I2S1", NULL, "I2S1 ASRC" }, + { "I2S2", NULL, "I2S2 ASRC" }, + + { "DMIC1", NULL, "DMIC L1" }, + { "DMIC1", NULL, "DMIC R1" }, + { "DMIC2", NULL, "DMIC L2" }, + { "DMIC2", NULL, "DMIC R2" }, + { "DMIC3", NULL, "DMIC L3" }, + { "DMIC3", NULL, "DMIC R3" }, + + { "BST1", NULL, "IN1P" }, + { "BST1", NULL, "IN1N" }, + { "BST1", NULL, "Mic Det Power" }, + { "BST2", NULL, "IN2P" }, + { "BST2", NULL, "IN2N" }, + + { "INL VOL", NULL, "IN2P" }, + { "INR VOL", NULL, "IN2N" }, + + { "RECMIXL", "INL Switch", "INL VOL" }, + { "RECMIXL", "BST2 Switch", "BST2" }, + { "RECMIXL", "BST1 Switch", "BST1" }, + + { "RECMIXR", "INR Switch", "INR VOL" }, + { "RECMIXR", "BST2 Switch", "BST2" }, + { "RECMIXR", "BST1 Switch", "BST1" }, + + { "ADC 1", NULL, "RECMIXL" }, + { "ADC 1", NULL, "ADC 1 power" }, + { "ADC 1", NULL, "ADC clock" }, + { "ADC 2", NULL, "RECMIXR" }, + { "ADC 2", NULL, "ADC 2 power" }, + { "ADC 2", NULL, "ADC clock" }, + + { "DMIC L1", NULL, "DMIC CLK" }, + { "DMIC L1", NULL, "DMIC1 Power" }, + { "DMIC R1", NULL, "DMIC CLK" }, + { "DMIC R1", NULL, "DMIC1 Power" }, + { "DMIC L2", NULL, "DMIC CLK" }, + { "DMIC L2", NULL, "DMIC2 Power" }, + { "DMIC R2", NULL, "DMIC CLK" }, + { "DMIC R2", NULL, "DMIC2 Power" }, + { "DMIC L3", NULL, "DMIC CLK" }, + { "DMIC L3", NULL, "DMIC3 Power" }, + { "DMIC R3", NULL, "DMIC CLK" }, + { "DMIC R3", NULL, "DMIC3 Power" }, + + { "Stereo1 DMIC Mux", "DMIC1", "DMIC1" }, + { "Stereo1 DMIC Mux", "DMIC2", "DMIC2" }, + { "Stereo1 DMIC Mux", "DMIC3", "DMIC3" }, + + { "Stereo2 DMIC Mux", "DMIC1", "DMIC1" }, + { "Stereo2 DMIC Mux", "DMIC2", "DMIC2" }, + { "Stereo2 DMIC Mux", "DMIC3", "DMIC3" }, + + { "Mono DMIC L Mux", "DMIC1", "DMIC L1" }, + { "Mono DMIC L Mux", "DMIC2", "DMIC L2" }, + { "Mono DMIC L Mux", "DMIC3", "DMIC L3" }, + + { "Mono DMIC R Mux", "DMIC1", "DMIC R1" }, + { "Mono DMIC R Mux", "DMIC2", "DMIC R2" }, + { "Mono DMIC R Mux", "DMIC3", "DMIC R3" }, + + { "ADC 1_2", NULL, "ADC 1" }, + { "ADC 1_2", NULL, "ADC 2" }, + + { "Stereo1 ADC L2 Mux", "DMIC", "Stereo1 DMIC Mux" }, + { "Stereo1 ADC L2 Mux", "DAC MIX", "DAC MIXL" }, + { "Stereo1 ADC L1 Mux", "ADC", "ADC 1_2" }, + { "Stereo1 ADC L1 Mux", "DAC MIX", "DAC MIXL" }, + + { "Stereo1 ADC R1 Mux", "ADC", "ADC 1_2" }, + { "Stereo1 ADC R1 Mux", "DAC MIX", "DAC MIXR" }, + { "Stereo1 ADC R2 Mux", "DMIC", "Stereo1 DMIC Mux" }, + { "Stereo1 ADC R2 Mux", "DAC MIX", "DAC MIXR" }, + + { "Mono ADC L2 Mux", "DMIC", "Mono DMIC L Mux" }, + { "Mono ADC L2 Mux", "Mono DAC MIXL", "Mono DAC MIXL" }, + { "Mono ADC L1 Mux", "Mono DAC MIXL", "Mono DAC MIXL" }, + { "Mono ADC L1 Mux", "ADC1", "ADC 1" }, + + { "Mono ADC R1 Mux", "Mono DAC MIXR", "Mono DAC MIXR" }, + { "Mono ADC R1 Mux", "ADC2", "ADC 2" }, + { "Mono ADC R2 Mux", "DMIC", "Mono DMIC R Mux" }, + { "Mono ADC R2 Mux", "Mono DAC MIXR", "Mono DAC MIXR" }, + + { "Sto1 ADC MIXL", "ADC1 Switch", "Stereo1 ADC L1 Mux" }, + { "Sto1 ADC MIXL", "ADC2 Switch", "Stereo1 ADC L2 Mux" }, + { "Sto1 ADC MIXR", "ADC1 Switch", "Stereo1 ADC R1 Mux" }, + { "Sto1 ADC MIXR", "ADC2 Switch", "Stereo1 ADC R2 Mux" }, + + { "Stereo1 ADC MIXL", NULL, "Sto1 ADC MIXL" }, + { "Stereo1 ADC MIXL", NULL, "ADC Stereo1 Filter" }, + { "ADC Stereo1 Filter", NULL, "PLL1", is_sys_clk_from_pll }, + + { "Stereo1 ADC MIXR", NULL, "Sto1 ADC MIXR" }, + { "Stereo1 ADC MIXR", NULL, "ADC Stereo1 Filter" }, + { "ADC Stereo1 Filter", NULL, "PLL1", is_sys_clk_from_pll }, + + { "Mono ADC MIXL", "ADC1 Switch", "Mono ADC L1 Mux" }, + { "Mono ADC MIXL", "ADC2 Switch", "Mono ADC L2 Mux" }, + { "Mono ADC MIXL", NULL, "ADC Mono Left Filter" }, + { "ADC Mono Left Filter", NULL, "PLL1", is_sys_clk_from_pll }, + + { "Mono ADC MIXR", "ADC1 Switch", "Mono ADC R1 Mux" }, + { "Mono ADC MIXR", "ADC2 Switch", "Mono ADC R2 Mux" }, + { "Mono ADC MIXR", NULL, "ADC Mono Right Filter" }, + { "ADC Mono Right Filter", NULL, "PLL1", is_sys_clk_from_pll }, + + { "Stereo2 ADC L2 Mux", "DMIC", "Stereo2 DMIC Mux" }, + { "Stereo2 ADC L2 Mux", "DAC MIX", "DAC MIXL" }, + { "Stereo2 ADC L1 Mux", "ADC", "ADC 1_2" }, + { "Stereo2 ADC L1 Mux", "DAC MIX", "DAC MIXL" }, + + { "Stereo2 ADC R1 Mux", "ADC", "ADC 1_2" }, + { "Stereo2 ADC R1 Mux", "DAC MIX", "DAC MIXR" }, + { "Stereo2 ADC R2 Mux", "DMIC", "Stereo2 DMIC Mux" }, + { "Stereo2 ADC R2 Mux", "DAC MIX", "DAC MIXR" }, + + { "Sto2 ADC MIXL", "ADC1 Switch", "Stereo2 ADC L1 Mux" }, + { "Sto2 ADC MIXL", "ADC2 Switch", "Stereo2 ADC L2 Mux" }, + { "Sto2 ADC MIXR", "ADC1 Switch", "Stereo2 ADC R1 Mux" }, + { "Sto2 ADC MIXR", "ADC2 Switch", "Stereo2 ADC R2 Mux" }, + + { "Sto2 ADC LR MIX", NULL, "Sto2 ADC MIXL" }, + { "Sto2 ADC LR MIX", NULL, "Sto2 ADC MIXR" }, + + { "Stereo2 ADC LR Mux", "L", "Sto2 ADC MIXL" }, + { "Stereo2 ADC LR Mux", "LR", "Sto2 ADC LR MIX" }, + + { "Stereo2 ADC MIXL", NULL, "Stereo2 ADC LR Mux" }, + { "Stereo2 ADC MIXL", NULL, "ADC Stereo2 Filter" }, + { "ADC Stereo2 Filter", NULL, "PLL1", is_sys_clk_from_pll }, + + { "Stereo2 ADC MIXR", NULL, "Sto2 ADC MIXR" }, + { "Stereo2 ADC MIXR", NULL, "ADC Stereo2 Filter" }, + { "ADC Stereo2 Filter", NULL, "PLL1", is_sys_clk_from_pll }, + + { "VAD ADC Mux", "Sto1 ADC L", "Stereo1 ADC MIXL" }, + { "VAD ADC Mux", "Mono ADC L", "Mono ADC MIXL" }, + { "VAD ADC Mux", "Mono ADC R", "Mono ADC MIXR" }, + { "VAD ADC Mux", "Sto2 ADC L", "Sto2 ADC MIXL" }, + + { "VAD_ADC", NULL, "VAD ADC Mux" }, + + { "IF_ADC1", NULL, "Stereo1 ADC MIXL" }, + { "IF_ADC1", NULL, "Stereo1 ADC MIXR" }, + { "IF_ADC2", NULL, "Mono ADC MIXL" }, + { "IF_ADC2", NULL, "Mono ADC MIXR" }, + { "IF_ADC3", NULL, "Stereo2 ADC MIXL" }, + { "IF_ADC3", NULL, "Stereo2 ADC MIXR" }, + + { "IF1 ADC1 IN1 Mux", "IF_ADC1", "IF_ADC1" }, + { "IF1 ADC1 IN1 Mux", "IF1_ADC3", "IF1_ADC3" }, + + { "IF1 ADC1 IN2 Mux", "IF1_ADC1_IN1", "IF1 ADC1 IN1 Mux" }, + { "IF1 ADC1 IN2 Mux", "IF1_ADC4", "IF1_ADC4" }, + + { "IF1 ADC2 IN Mux", "IF_ADC2", "IF_ADC2" }, + { "IF1 ADC2 IN Mux", "VAD_ADC", "VAD_ADC" }, + + { "IF1 ADC2 IN1 Mux", "IF1_ADC2_IN", "IF1 ADC2 IN Mux" }, + { "IF1 ADC2 IN1 Mux", "IF1_ADC4", "IF1_ADC4" }, + + { "IF1_ADC1" , NULL, "IF1 ADC1 IN2 Mux" }, + { "IF1_ADC2" , NULL, "IF1 ADC2 IN1 Mux" }, + + { "Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXL" }, + { "Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXR" }, + { "Stereo2 ADC MIX", NULL, "Sto2 ADC MIXL" }, + { "Stereo2 ADC MIX", NULL, "Sto2 ADC MIXR" }, + { "Mono ADC MIX", NULL, "Mono ADC MIXL" }, + { "Mono ADC MIX", NULL, "Mono ADC MIXR" }, + + { "RxDP Mux", "IF2 DAC", "IF2 DAC" }, + { "RxDP Mux", "IF1 DAC", "IF1 DAC2" }, + { "RxDP Mux", "STO1 ADC Mixer", "Stereo1 ADC MIX" }, + { "RxDP Mux", "STO2 ADC Mixer", "Stereo2 ADC MIX" }, + { "RxDP Mux", "Mono ADC Mixer L", "Mono ADC MIXL" }, + { "RxDP Mux", "Mono ADC Mixer R", "Mono ADC MIXR" }, + { "RxDP Mux", "DAC1", "DAC MIX" }, + + { "TDM Data Mux", "Slot 0-1", "Stereo1 ADC MIX" }, + { "TDM Data Mux", "Slot 2-3", "Mono ADC MIX" }, + { "TDM Data Mux", "Slot 4-5", "Stereo2 ADC MIX" }, + { "TDM Data Mux", "Slot 6-7", "IF2 DAC" }, + + { "DSP UL Mux", "Bypass", "TDM Data Mux" }, + { "DSP UL Mux", NULL, "I2S DSP" }, + { "DSP DL Mux", "Bypass", "RxDP Mux" }, + { "DSP DL Mux", NULL, "I2S DSP" }, + + { "TxDP_ADC_L", NULL, "DSP UL Mux" }, + { "TxDP_ADC_R", NULL, "DSP UL Mux" }, + { "TxDC_DAC", NULL, "DSP DL Mux" }, + + { "TxDP_ADC", NULL, "TxDP_ADC_L" }, + { "TxDP_ADC", NULL, "TxDP_ADC_R" }, + + { "IF1 ADC", NULL, "I2S1" }, + { "IF1 ADC", NULL, "IF1_ADC1" }, + { "IF1 ADC", NULL, "IF1_ADC2" }, + { "IF1 ADC", NULL, "IF_ADC3" }, + { "IF1 ADC", NULL, "TxDP_ADC" }, + + { "IF2 ADC Mux", "IF_ADC1", "IF_ADC1" }, + { "IF2 ADC Mux", "IF_ADC2", "IF_ADC2" }, + { "IF2 ADC Mux", "IF_ADC3", "IF_ADC3" }, + { "IF2 ADC Mux", "TxDC_DAC", "TxDC_DAC" }, + { "IF2 ADC Mux", "TxDP_ADC", "TxDP_ADC" }, + { "IF2 ADC Mux", "VAD_ADC", "VAD_ADC" }, + + { "IF2 ADC L", NULL, "IF2 ADC Mux" }, + { "IF2 ADC R", NULL, "IF2 ADC Mux" }, + + { "IF2 ADC", NULL, "I2S2" }, + { "IF2 ADC", NULL, "IF2 ADC L" }, + { "IF2 ADC", NULL, "IF2 ADC R" }, + + { "AIF1TX", NULL, "IF1 ADC" }, + { "AIF2TX", NULL, "IF2 ADC" }, + + { "IF1 DAC1", NULL, "AIF1RX" }, + { "IF1 DAC2", NULL, "AIF1RX" }, + { "IF2 DAC", NULL, "AIF2RX" }, + + { "IF1 DAC1", NULL, "I2S1" }, + { "IF1 DAC2", NULL, "I2S1" }, + { "IF2 DAC", NULL, "I2S2" }, + + { "IF1 DAC2 L", NULL, "IF1 DAC2" }, + { "IF1 DAC2 R", NULL, "IF1 DAC2" }, + { "IF1 DAC1 L", NULL, "IF1 DAC1" }, + { "IF1 DAC1 R", NULL, "IF1 DAC1" }, + { "IF2 DAC L", NULL, "IF2 DAC" }, + { "IF2 DAC R", NULL, "IF2 DAC" }, + + { "DAC1 L Mux", "IF1 DAC", "IF1 DAC1 L" }, + { "DAC1 L Mux", "IF2 DAC", "IF2 DAC L" }, + + { "DAC1 R Mux", "IF1 DAC", "IF1 DAC1 R" }, + { "DAC1 R Mux", "IF2 DAC", "IF2 DAC R" }, + + { "DAC1 MIXL", "Stereo ADC Switch", "Stereo1 ADC MIXL" }, + { "DAC1 MIXL", "DAC1 Switch", "DAC1 L Mux" }, + { "DAC1 MIXL", NULL, "DAC Stereo1 Filter" }, + { "DAC1 MIXR", "Stereo ADC Switch", "Stereo1 ADC MIXR" }, + { "DAC1 MIXR", "DAC1 Switch", "DAC1 R Mux" }, + { "DAC1 MIXR", NULL, "DAC Stereo1 Filter" }, + + { "DAC MIX", NULL, "DAC1 MIXL" }, + { "DAC MIX", NULL, "DAC1 MIXR" }, + + { "Audio DSP", NULL, "DAC1 MIXL" }, + { "Audio DSP", NULL, "DAC1 MIXR" }, + + { "DAC L2 Mux", "IF1 DAC", "IF1 DAC2 L" }, + { "DAC L2 Mux", "IF2 DAC", "IF2 DAC L" }, + { "DAC L2 Mux", "TxDC DAC", "TxDC_DAC" }, + { "DAC L2 Mux", "VAD_ADC", "VAD_ADC" }, + { "DAC L2 Volume", NULL, "DAC L2 Mux" }, + { "DAC L2 Volume", NULL, "DAC Mono Left Filter" }, + + { "DAC R2 Mux", "IF1 DAC", "IF1 DAC2 R" }, + { "DAC R2 Mux", "IF2 DAC", "IF2 DAC R" }, + { "DAC R2 Mux", "TxDC DAC", "TxDC_DAC" }, + { "DAC R2 Mux", "TxDP ADC", "TxDP_ADC" }, + { "DAC R2 Volume", NULL, "DAC R2 Mux" }, + { "DAC R2 Volume", NULL, "DAC Mono Right Filter" }, + + { "Stereo DAC MIXL", "DAC L1 Switch", "DAC1 MIXL" }, + { "Stereo DAC MIXL", "DAC R1 Switch", "DAC1 MIXR" }, + { "Stereo DAC MIXL", "DAC L2 Switch", "DAC L2 Volume" }, + { "Stereo DAC MIXL", NULL, "DAC Stereo1 Filter" }, + { "Stereo DAC MIXL", NULL, "DAC L1 Power" }, + { "Stereo DAC MIXR", "DAC R1 Switch", "DAC1 MIXR" }, + { "Stereo DAC MIXR", "DAC L1 Switch", "DAC1 MIXL" }, + { "Stereo DAC MIXR", "DAC R2 Switch", "DAC R2 Volume" }, + { "Stereo DAC MIXR", NULL, "DAC Stereo1 Filter" }, + { "Stereo DAC MIXR", NULL, "DAC R1 Power" }, + + { "Mono DAC MIXL", "DAC L1 Switch", "DAC1 MIXL" }, + { "Mono DAC MIXL", "DAC L2 Switch", "DAC L2 Volume" }, + { "Mono DAC MIXL", "DAC R2 Switch", "DAC R2 Volume" }, + { "Mono DAC MIXL", NULL, "DAC Mono Left Filter" }, + { "Mono DAC MIXR", "DAC R1 Switch", "DAC1 MIXR" }, + { "Mono DAC MIXR", "DAC R2 Switch", "DAC R2 Volume" }, + { "Mono DAC MIXR", "DAC L2 Switch", "DAC L2 Volume" }, + { "Mono DAC MIXR", NULL, "DAC Mono Right Filter" }, + + { "DAC MIXL", "Sto DAC Mix L Switch", "Stereo DAC MIXL" }, + { "DAC MIXL", "DAC L2 Switch", "DAC L2 Volume" }, + { "DAC MIXL", "DAC R2 Switch", "DAC R2 Volume" }, + { "DAC MIXR", "Sto DAC Mix R Switch", "Stereo DAC MIXR" }, + { "DAC MIXR", "DAC R2 Switch", "DAC R2 Volume" }, + { "DAC MIXR", "DAC L2 Switch", "DAC L2 Volume" }, + + { "DAC L1", NULL, "DAC L1 Power" }, + { "DAC L1", NULL, "Stereo DAC MIXL" }, + { "DAC L1", NULL, "PLL1", is_sys_clk_from_pll }, + { "DAC R1", NULL, "DAC R1 Power" }, + { "DAC R1", NULL, "Stereo DAC MIXR" }, + { "DAC R1", NULL, "PLL1", is_sys_clk_from_pll }, + { "DAC L2", NULL, "Mono DAC MIXL" }, + { "DAC L2", NULL, "PLL1", is_sys_clk_from_pll }, + { "DAC R2", NULL, "Mono DAC MIXR" }, + { "DAC R2", NULL, "PLL1", is_sys_clk_from_pll }, + + { "OUT MIXL", "BST1 Switch", "BST1" }, + { "OUT MIXL", "INL Switch", "INL VOL" }, + { "OUT MIXL", "DAC L2 Switch", "DAC L2" }, + { "OUT MIXL", "DAC L1 Switch", "DAC L1" }, + + { "OUT MIXR", "BST2 Switch", "BST2" }, + { "OUT MIXR", "INR Switch", "INR VOL" }, + { "OUT MIXR", "DAC R2 Switch", "DAC R2" }, + { "OUT MIXR", "DAC R1 Switch", "DAC R1" }, + + { "HPOVOL MIXL", "DAC1 Switch", "DAC L1" }, + { "HPOVOL MIXL", "INL Switch", "INL VOL" }, + { "HPOVOL MIXR", "DAC1 Switch", "DAC R1" }, + { "HPOVOL MIXR", "INR Switch", "INR VOL" }, + + { "DAC 2", NULL, "DAC L2" }, + { "DAC 2", NULL, "DAC R2" }, + { "DAC 1", NULL, "DAC L1" }, + { "DAC 1", NULL, "DAC R1" }, + { "HPOVOL", NULL, "HPOVOL MIXL" }, + { "HPOVOL", NULL, "HPOVOL MIXR" }, + { "HPO MIX", "DAC1 Switch", "DAC 1" }, + { "HPO MIX", "HPVOL Switch", "HPOVOL" }, + + { "LOUT MIX", "DAC L1 Switch", "DAC L1" }, + { "LOUT MIX", "DAC R1 Switch", "DAC R1" }, + { "LOUT MIX", "OUTMIX L Switch", "OUT MIXL" }, + { "LOUT MIX", "OUTMIX R Switch", "OUT MIXR" }, + + { "PDM1 L Mux", "Stereo DAC", "Stereo DAC MIXL" }, + { "PDM1 L Mux", "Mono DAC", "Mono DAC MIXL" }, + { "PDM1 L Mux", NULL, "PDM1 Power" }, + { "PDM1 R Mux", "Stereo DAC", "Stereo DAC MIXR" }, + { "PDM1 R Mux", "Mono DAC", "Mono DAC MIXR" }, + { "PDM1 R Mux", NULL, "PDM1 Power" }, + { "PDM2 L Mux", "Stereo DAC", "Stereo DAC MIXL" }, + { "PDM2 L Mux", "Mono DAC", "Mono DAC MIXL" }, + { "PDM2 L Mux", NULL, "PDM2 Power" }, + { "PDM2 R Mux", "Stereo DAC", "Stereo DAC MIXR" }, + { "PDM2 R Mux", "Mono DAC", "Mono DAC MIXR" }, + { "PDM2 R Mux", NULL, "PDM2 Power" }, + + { "HP Amp", NULL, "HPO MIX" }, + { "HP Amp", NULL, "Mic Det Power" }, + { "HPOL", NULL, "HP Amp" }, + { "HPOL", NULL, "HP L Amp" }, + { "HPOL", NULL, "Improve HP Amp Drv" }, + { "HPOR", NULL, "HP Amp" }, + { "HPOR", NULL, "HP R Amp" }, + { "HPOR", NULL, "Improve HP Amp Drv" }, + + { "LOUT Amp", NULL, "LOUT MIX" }, + { "LOUT L Playback", "Switch", "LOUT Amp" }, + { "LOUT R Playback", "Switch", "LOUT Amp" }, + { "LOUTL", NULL, "LOUT L Playback" }, + { "LOUTR", NULL, "LOUT R Playback" }, + { "LOUTL", NULL, "Improve HP Amp Drv" }, + { "LOUTR", NULL, "Improve HP Amp Drv" }, + + { "PDM1L", NULL, "PDM1 L Mux" }, + { "PDM1R", NULL, "PDM1 R Mux" }, + { "PDM2L", NULL, "PDM2 L Mux" }, + { "PDM2R", NULL, "PDM2 R Mux" }, +}; + +static int rt5670_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + unsigned int val_len = 0, val_clk, mask_clk; + int pre_div, bclk_ms, frame_size; + + rt5670->lrck[dai->id] = params_rate(params); + pre_div = rl6231_get_clk_info(rt5670->sysclk, rt5670->lrck[dai->id]); + if (pre_div < 0) { + dev_err(codec->dev, "Unsupported clock setting %d for DAI %d\n", + rt5670->lrck[dai->id], dai->id); + return -EINVAL; + } + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) { + dev_err(codec->dev, "Unsupported frame size: %d\n", frame_size); + return -EINVAL; + } + bclk_ms = frame_size > 32; + rt5670->bclk[dai->id] = rt5670->lrck[dai->id] * (32 << bclk_ms); + + dev_dbg(dai->dev, "bclk is %dHz and lrck is %dHz\n", + rt5670->bclk[dai->id], rt5670->lrck[dai->id]); + dev_dbg(dai->dev, "bclk_ms is %d and pre_div is %d for iis %d\n", + bclk_ms, pre_div, dai->id); + + switch (params_width(params)) { + case 16: + break; + case 20: + val_len |= RT5670_I2S_DL_20; + break; + case 24: + val_len |= RT5670_I2S_DL_24; + break; + case 8: + val_len |= RT5670_I2S_DL_8; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5670_AIF1: + mask_clk = RT5670_I2S_BCLK_MS1_MASK | RT5670_I2S_PD1_MASK; + val_clk = bclk_ms << RT5670_I2S_BCLK_MS1_SFT | + pre_div << RT5670_I2S_PD1_SFT; + snd_soc_update_bits(codec, RT5670_I2S1_SDP, + RT5670_I2S_DL_MASK, val_len); + snd_soc_update_bits(codec, RT5670_ADDA_CLK1, mask_clk, val_clk); + break; + case RT5670_AIF2: + mask_clk = RT5670_I2S_BCLK_MS2_MASK | RT5670_I2S_PD2_MASK; + val_clk = bclk_ms << RT5670_I2S_BCLK_MS2_SFT | + pre_div << RT5670_I2S_PD2_SFT; + snd_soc_update_bits(codec, RT5670_I2S2_SDP, + RT5670_I2S_DL_MASK, val_len); + snd_soc_update_bits(codec, RT5670_ADDA_CLK1, mask_clk, val_clk); + break; + default: + dev_err(codec->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + + return 0; +} + +static int rt5670_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + rt5670->master[dai->id] = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + reg_val |= RT5670_I2S_MS_S; + rt5670->master[dai->id] = 0; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + reg_val |= RT5670_I2S_BP_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_LEFT_J: + reg_val |= RT5670_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + reg_val |= RT5670_I2S_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + reg_val |= RT5670_I2S_DF_PCM_B; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5670_AIF1: + snd_soc_update_bits(codec, RT5670_I2S1_SDP, + RT5670_I2S_MS_MASK | RT5670_I2S_BP_MASK | + RT5670_I2S_DF_MASK, reg_val); + break; + case RT5670_AIF2: + snd_soc_update_bits(codec, RT5670_I2S2_SDP, + RT5670_I2S_MS_MASK | RT5670_I2S_BP_MASK | + RT5670_I2S_DF_MASK, reg_val); + break; + default: + dev_err(codec->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + return 0; +} + +static int rt5670_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + + if (freq == rt5670->sysclk && clk_id == rt5670->sysclk_src) + return 0; + + switch (clk_id) { + case RT5670_SCLK_S_MCLK: + reg_val |= RT5670_SCLK_SRC_MCLK; + break; + case RT5670_SCLK_S_PLL1: + reg_val |= RT5670_SCLK_SRC_PLL1; + break; + case RT5670_SCLK_S_RCCLK: + reg_val |= RT5670_SCLK_SRC_RCCLK; + break; + default: + dev_err(codec->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + snd_soc_update_bits(codec, RT5670_GLB_CLK, + RT5670_SCLK_SRC_MASK, reg_val); + rt5670->sysclk = freq; + rt5670->sysclk_src = clk_id; + + dev_dbg(dai->dev, "Sysclk is %dHz and clock id is %d\n", freq, clk_id); + + return 0; +} + +static int rt5670_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + struct rl6231_pll_code pll_code; + int ret; + + if (source == rt5670->pll_src && freq_in == rt5670->pll_in && + freq_out == rt5670->pll_out) + return 0; + + if (!freq_in || !freq_out) { + dev_dbg(codec->dev, "PLL disabled\n"); + + rt5670->pll_in = 0; + rt5670->pll_out = 0; + snd_soc_update_bits(codec, RT5670_GLB_CLK, + RT5670_SCLK_SRC_MASK, RT5670_SCLK_SRC_MCLK); + return 0; + } + + switch (source) { + case RT5670_PLL1_S_MCLK: + snd_soc_update_bits(codec, RT5670_GLB_CLK, + RT5670_PLL1_SRC_MASK, RT5670_PLL1_SRC_MCLK); + break; + case RT5670_PLL1_S_BCLK1: + case RT5670_PLL1_S_BCLK2: + case RT5670_PLL1_S_BCLK3: + case RT5670_PLL1_S_BCLK4: + switch (dai->id) { + case RT5670_AIF1: + snd_soc_update_bits(codec, RT5670_GLB_CLK, + RT5670_PLL1_SRC_MASK, RT5670_PLL1_SRC_BCLK1); + break; + case RT5670_AIF2: + snd_soc_update_bits(codec, RT5670_GLB_CLK, + RT5670_PLL1_SRC_MASK, RT5670_PLL1_SRC_BCLK2); + break; + default: + dev_err(codec->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + break; + default: + dev_err(codec->dev, "Unknown PLL source %d\n", source); + return -EINVAL; + } + + ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); + if (ret < 0) { + dev_err(codec->dev, "Unsupport input clock %d\n", freq_in); + return ret; + } + + dev_dbg(codec->dev, "bypass=%d m=%d n=%d k=%d\n", + pll_code.m_bp, (pll_code.m_bp ? 0 : pll_code.m_code), + pll_code.n_code, pll_code.k_code); + + snd_soc_write(codec, RT5670_PLL_CTRL1, + pll_code.n_code << RT5670_PLL_N_SFT | pll_code.k_code); + snd_soc_write(codec, RT5670_PLL_CTRL2, + (pll_code.m_bp ? 0 : pll_code.m_code) << RT5670_PLL_M_SFT | + pll_code.m_bp << RT5670_PLL_M_BP_SFT); + + rt5670->pll_in = freq_in; + rt5670->pll_out = freq_out; + rt5670->pll_src = source; + + return 0; +} + +static int rt5670_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int val = 0; + + if (rx_mask || tx_mask) + val |= (1 << 14); + + switch (slots) { + case 4: + val |= (1 << 12); + break; + case 6: + val |= (2 << 12); + break; + case 8: + val |= (3 << 12); + break; + case 2: + break; + default: + return -EINVAL; + } + + switch (slot_width) { + case 20: + val |= (1 << 10); + break; + case 24: + val |= (2 << 10); + break; + case 32: + val |= (3 << 10); + break; + case 16: + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, RT5670_TDM_CTRL_1, 0x7c00, val); + + return 0; +} + +static int rt5670_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) { + snd_soc_update_bits(codec, RT5670_PWR_ANLG1, + RT5670_PWR_VREF1 | RT5670_PWR_MB | + RT5670_PWR_BG | RT5670_PWR_VREF2, + RT5670_PWR_VREF1 | RT5670_PWR_MB | + RT5670_PWR_BG | RT5670_PWR_VREF2); + mdelay(10); + snd_soc_update_bits(codec, RT5670_PWR_ANLG1, + RT5670_PWR_FV1 | RT5670_PWR_FV2, + RT5670_PWR_FV1 | RT5670_PWR_FV2); + snd_soc_update_bits(codec, RT5670_CHARGE_PUMP, + RT5670_OSW_L_MASK | RT5670_OSW_R_MASK, + RT5670_OSW_L_DIS | RT5670_OSW_R_DIS); + snd_soc_update_bits(codec, RT5670_DIG_MISC, 0x1, 0x1); + snd_soc_update_bits(codec, RT5670_PWR_ANLG1, + RT5670_LDO_SEL_MASK, 0x3); + } + break; + case SND_SOC_BIAS_STANDBY: + snd_soc_write(codec, RT5670_PWR_DIG1, 0x0000); + snd_soc_write(codec, RT5670_PWR_DIG2, 0x0001); + snd_soc_write(codec, RT5670_PWR_VOL, 0x0000); + snd_soc_write(codec, RT5670_PWR_MIXER, 0x0001); + snd_soc_write(codec, RT5670_PWR_ANLG1, 0x2800); + snd_soc_write(codec, RT5670_PWR_ANLG2, 0x0004); + snd_soc_update_bits(codec, RT5670_DIG_MISC, 0x1, 0x0); + snd_soc_update_bits(codec, RT5670_PWR_ANLG1, + RT5670_LDO_SEL_MASK, 0x1); + break; + + default: + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +static int rt5670_probe(struct snd_soc_codec *codec) +{ + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + rt5670->codec = codec; + + return 0; +} + +static int rt5670_remove(struct snd_soc_codec *codec) +{ + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + regmap_write(rt5670->regmap, RT5670_RESET, 0); + return 0; +} + +#ifdef CONFIG_PM +static int rt5670_suspend(struct snd_soc_codec *codec) +{ + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt5670->regmap, true); + regcache_mark_dirty(rt5670->regmap); + return 0; +} + +static int rt5670_resume(struct snd_soc_codec *codec) +{ + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt5670->regmap, false); + regcache_sync(rt5670->regmap); + + return 0; +} +#else +#define rt5670_suspend NULL +#define rt5670_resume NULL +#endif + +#define RT5670_STEREO_RATES SNDRV_PCM_RATE_8000_96000 +#define RT5670_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +struct snd_soc_dai_ops rt5670_aif_dai_ops = { + .hw_params = rt5670_hw_params, + .set_fmt = rt5670_set_dai_fmt, + .set_sysclk = rt5670_set_dai_sysclk, + .set_tdm_slot = rt5670_set_tdm_slot, + .set_pll = rt5670_set_dai_pll, +}; + +struct snd_soc_dai_driver rt5670_dai[] = { + { + .name = "rt5670-aif1", + .id = RT5670_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5670_STEREO_RATES, + .formats = RT5670_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5670_STEREO_RATES, + .formats = RT5670_FORMATS, + }, + .ops = &rt5670_aif_dai_ops, + }, + { + .name = "rt5670-aif2", + .id = RT5670_AIF2, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5670_STEREO_RATES, + .formats = RT5670_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5670_STEREO_RATES, + .formats = RT5670_FORMATS, + }, + .ops = &rt5670_aif_dai_ops, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt5670 = { + .probe = rt5670_probe, + .remove = rt5670_remove, + .suspend = rt5670_suspend, + .resume = rt5670_resume, + .set_bias_level = rt5670_set_bias_level, + .idle_bias_off = true, + .controls = rt5670_snd_controls, + .num_controls = ARRAY_SIZE(rt5670_snd_controls), + .dapm_widgets = rt5670_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt5670_dapm_widgets), + .dapm_routes = rt5670_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt5670_dapm_routes), +}; + +static const struct regmap_config rt5670_regmap = { + .reg_bits = 8, + .val_bits = 16, + .max_register = RT5670_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5670_ranges) * + RT5670_PR_SPACING), + .volatile_reg = rt5670_volatile_register, + .readable_reg = rt5670_readable_register, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt5670_reg, + .num_reg_defaults = ARRAY_SIZE(rt5670_reg), + .ranges = rt5670_ranges, + .num_ranges = ARRAY_SIZE(rt5670_ranges), +}; + +static const struct i2c_device_id rt5670_i2c_id[] = { + { "rt5670", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, rt5670_i2c_id); + +static int rt5670_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt5670_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt5670_priv *rt5670; + int ret; + unsigned int val; + + rt5670 = devm_kzalloc(&i2c->dev, + sizeof(struct rt5670_priv), + GFP_KERNEL); + if (NULL == rt5670) + return -ENOMEM; + + i2c_set_clientdata(i2c, rt5670); + + if (pdata) + rt5670->pdata = *pdata; + + rt5670->regmap = devm_regmap_init_i2c(i2c, &rt5670_regmap); + if (IS_ERR(rt5670->regmap)) { + ret = PTR_ERR(rt5670->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + regmap_read(rt5670->regmap, RT5670_VENDOR_ID2, &val); + if (val != RT5670_DEVICE_ID) { + dev_err(&i2c->dev, + "Device with ID register %x is not rt5670/72\n", val); + return -ENODEV; + } + + regmap_write(rt5670->regmap, RT5670_RESET, 0); + regmap_update_bits(rt5670->regmap, RT5670_PWR_ANLG1, + RT5670_PWR_HP_L | RT5670_PWR_HP_R | + RT5670_PWR_VREF2, RT5670_PWR_VREF2); + msleep(100); + + regmap_write(rt5670->regmap, RT5670_RESET, 0); + + ret = regmap_register_patch(rt5670->regmap, init_list, + ARRAY_SIZE(init_list)); + if (ret != 0) + dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); + + if (rt5670->pdata.in2_diff) + regmap_update_bits(rt5670->regmap, RT5670_IN2, + RT5670_IN_DF2, RT5670_IN_DF2); + + if (i2c->irq) { + regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1, + RT5670_GP1_PIN_MASK, RT5670_GP1_PIN_IRQ); + regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL2, + RT5670_GP1_PF_MASK, RT5670_GP1_PF_OUT); + + } + + if (rt5670->pdata.jd_mode) { + regmap_update_bits(rt5670->regmap, RT5670_PWR_ANLG1, + RT5670_PWR_MB, RT5670_PWR_MB); + regmap_update_bits(rt5670->regmap, RT5670_PWR_ANLG2, + RT5670_PWR_JD1, RT5670_PWR_JD1); + regmap_update_bits(rt5670->regmap, RT5670_IRQ_CTRL1, + RT5670_JD1_1_EN_MASK, RT5670_JD1_1_EN); + regmap_update_bits(rt5670->regmap, RT5670_JD_CTRL3, + RT5670_JD_TRI_CBJ_SEL_MASK | + RT5670_JD_TRI_HPO_SEL_MASK, + RT5670_JD_CBJ_JD1_1 | RT5670_JD_HPO_JD1_1); + switch (rt5670->pdata.jd_mode) { + case 1: + regmap_update_bits(rt5670->regmap, RT5670_A_JD_CTRL1, + RT5670_JD1_MODE_MASK, + RT5670_JD1_MODE_0); + break; + case 2: + regmap_update_bits(rt5670->regmap, RT5670_A_JD_CTRL1, + RT5670_JD1_MODE_MASK, + RT5670_JD1_MODE_1); + break; + case 3: + regmap_update_bits(rt5670->regmap, RT5670_A_JD_CTRL1, + RT5670_JD1_MODE_MASK, + RT5670_JD1_MODE_2); + break; + default: + break; + } + } + + if (rt5670->pdata.dmic_en) { + regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1, + RT5670_GP2_PIN_MASK, + RT5670_GP2_PIN_DMIC1_SCL); + + switch (rt5670->pdata.dmic1_data_pin) { + case RT5670_DMIC_DATA_IN2P: + regmap_update_bits(rt5670->regmap, RT5670_DMIC_CTRL1, + RT5670_DMIC_1_DP_MASK, + RT5670_DMIC_1_DP_IN2P); + break; + + case RT5670_DMIC_DATA_GPIO6: + regmap_update_bits(rt5670->regmap, RT5670_DMIC_CTRL1, + RT5670_DMIC_1_DP_MASK, + RT5670_DMIC_1_DP_GPIO6); + regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1, + RT5670_GP6_PIN_MASK, + RT5670_GP6_PIN_DMIC1_SDA); + break; + + case RT5670_DMIC_DATA_GPIO7: + regmap_update_bits(rt5670->regmap, RT5670_DMIC_CTRL1, + RT5670_DMIC_1_DP_MASK, + RT5670_DMIC_1_DP_GPIO7); + regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1, + RT5670_GP7_PIN_MASK, + RT5670_GP7_PIN_DMIC1_SDA); + break; + + default: + break; + } + + switch (rt5670->pdata.dmic2_data_pin) { + case RT5670_DMIC_DATA_IN3N: + regmap_update_bits(rt5670->regmap, RT5670_DMIC_CTRL1, + RT5670_DMIC_2_DP_MASK, + RT5670_DMIC_2_DP_IN3N); + break; + + case RT5670_DMIC_DATA_GPIO8: + regmap_update_bits(rt5670->regmap, RT5670_DMIC_CTRL1, + RT5670_DMIC_2_DP_MASK, + RT5670_DMIC_2_DP_GPIO8); + regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1, + RT5670_GP8_PIN_MASK, + RT5670_GP8_PIN_DMIC2_SDA); + break; + + default: + break; + } + + switch (rt5670->pdata.dmic3_data_pin) { + case RT5670_DMIC_DATA_GPIO5: + regmap_update_bits(rt5670->regmap, RT5670_DMIC_CTRL2, + RT5670_DMIC_3_DP_MASK, + RT5670_DMIC_3_DP_GPIO5); + regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1, + RT5670_GP5_PIN_MASK, + RT5670_GP5_PIN_DMIC3_SDA); + break; + + case RT5670_DMIC_DATA_GPIO9: + case RT5670_DMIC_DATA_GPIO10: + dev_err(&i2c->dev, + "Always use GPIO5 as DMIC3 data pin\n"); + break; + + default: + break; + } + + } + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5670, + rt5670_dai, ARRAY_SIZE(rt5670_dai)); + if (ret < 0) + goto err; + + return 0; +err: + return ret; +} + +static int rt5670_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + +struct i2c_driver rt5670_i2c_driver = { + .driver = { + .name = "rt5670", + .owner = THIS_MODULE, + }, + .probe = rt5670_i2c_probe, + .remove = rt5670_i2c_remove, + .id_table = rt5670_i2c_id, +}; + +module_i2c_driver(rt5670_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT5670 driver"); +MODULE_AUTHOR("Bard Liao "); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h new file mode 100644 index 000000000000..a0b5c855b492 --- /dev/null +++ b/sound/soc/codecs/rt5670.h @@ -0,0 +1,2000 @@ +/* + * rt5670.h -- RT5670 ALSA SoC audio driver + * + * Copyright 2014 Realtek Microelectronics + * Author: Bard Liao + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __RT5670_H__ +#define __RT5670_H__ + +#include + +/* Info */ +#define RT5670_RESET 0x00 +#define RT5670_VENDOR_ID 0xfd +#define RT5670_VENDOR_ID1 0xfe +#define RT5670_VENDOR_ID2 0xff +/* I/O - Output */ +#define RT5670_HP_VOL 0x02 +#define RT5670_LOUT1 0x03 +/* I/O - Input */ +#define RT5670_CJ_CTRL1 0x0a +#define RT5670_CJ_CTRL2 0x0b +#define RT5670_CJ_CTRL3 0x0c +#define RT5670_IN2 0x0e +#define RT5670_INL1_INR1_VOL 0x0f +/* I/O - ADC/DAC/DMIC */ +#define RT5670_DAC1_DIG_VOL 0x19 +#define RT5670_DAC2_DIG_VOL 0x1a +#define RT5670_DAC_CTRL 0x1b +#define RT5670_STO1_ADC_DIG_VOL 0x1c +#define RT5670_MONO_ADC_DIG_VOL 0x1d +#define RT5670_ADC_BST_VOL1 0x1e +#define RT5670_STO2_ADC_DIG_VOL 0x1f +/* Mixer - D-D */ +#define RT5670_ADC_BST_VOL2 0x20 +#define RT5670_STO2_ADC_MIXER 0x26 +#define RT5670_STO1_ADC_MIXER 0x27 +#define RT5670_MONO_ADC_MIXER 0x28 +#define RT5670_AD_DA_MIXER 0x29 +#define RT5670_STO_DAC_MIXER 0x2a +#define RT5670_DD_MIXER 0x2b +#define RT5670_DIG_MIXER 0x2c +#define RT5670_DSP_PATH1 0x2d +#define RT5670_DSP_PATH2 0x2e +#define RT5670_DIG_INF1_DATA 0x2f +#define RT5670_DIG_INF2_DATA 0x30 +/* Mixer - PDM */ +#define RT5670_PDM_OUT_CTRL 0x31 +#define RT5670_PDM_DATA_CTRL1 0x32 +#define RT5670_PDM1_DATA_CTRL2 0x33 +#define RT5670_PDM1_DATA_CTRL3 0x34 +#define RT5670_PDM1_DATA_CTRL4 0x35 +#define RT5670_PDM2_DATA_CTRL2 0x36 +#define RT5670_PDM2_DATA_CTRL3 0x37 +#define RT5670_PDM2_DATA_CTRL4 0x38 +/* Mixer - ADC */ +#define RT5670_REC_L1_MIXER 0x3b +#define RT5670_REC_L2_MIXER 0x3c +#define RT5670_REC_R1_MIXER 0x3d +#define RT5670_REC_R2_MIXER 0x3e +/* Mixer - DAC */ +#define RT5670_HPO_MIXER 0x45 +#define RT5670_MONO_MIXER 0x4c +#define RT5670_OUT_L1_MIXER 0x4f +#define RT5670_OUT_R1_MIXER 0x52 +#define RT5670_LOUT_MIXER 0x53 +/* Power */ +#define RT5670_PWR_DIG1 0x61 +#define RT5670_PWR_DIG2 0x62 +#define RT5670_PWR_ANLG1 0x63 +#define RT5670_PWR_ANLG2 0x64 +#define RT5670_PWR_MIXER 0x65 +#define RT5670_PWR_VOL 0x66 +/* Private Register Control */ +#define RT5670_PRIV_INDEX 0x6a +#define RT5670_PRIV_DATA 0x6c +/* Format - ADC/DAC */ +#define RT5670_I2S4_SDP 0x6f +#define RT5670_I2S1_SDP 0x70 +#define RT5670_I2S2_SDP 0x71 +#define RT5670_I2S3_SDP 0x72 +#define RT5670_ADDA_CLK1 0x73 +#define RT5670_ADDA_CLK2 0x74 +#define RT5670_DMIC_CTRL1 0x75 +#define RT5670_DMIC_CTRL2 0x76 +/* Format - TDM Control */ +#define RT5670_TDM_CTRL_1 0x77 +#define RT5670_TDM_CTRL_2 0x78 +#define RT5670_TDM_CTRL_3 0x79 + +/* Function - Analog */ +#define RT5670_DSP_CLK 0x7f +#define RT5670_GLB_CLK 0x80 +#define RT5670_PLL_CTRL1 0x81 +#define RT5670_PLL_CTRL2 0x82 +#define RT5670_ASRC_1 0x83 +#define RT5670_ASRC_2 0x84 +#define RT5670_ASRC_3 0x85 +#define RT5670_ASRC_4 0x86 +#define RT5670_ASRC_5 0x87 +#define RT5670_ASRC_7 0x89 +#define RT5670_ASRC_8 0x8a +#define RT5670_ASRC_9 0x8b +#define RT5670_ASRC_10 0x8c +#define RT5670_ASRC_11 0x8d +#define RT5670_DEPOP_M1 0x8e +#define RT5670_DEPOP_M2 0x8f +#define RT5670_DEPOP_M3 0x90 +#define RT5670_CHARGE_PUMP 0x91 +#define RT5670_MICBIAS 0x93 +#define RT5670_A_JD_CTRL1 0x94 +#define RT5670_A_JD_CTRL2 0x95 +#define RT5670_ASRC_12 0x97 +#define RT5670_ASRC_13 0x98 +#define RT5670_ASRC_14 0x99 +#define RT5670_VAD_CTRL1 0x9a +#define RT5670_VAD_CTRL2 0x9b +#define RT5670_VAD_CTRL3 0x9c +#define RT5670_VAD_CTRL4 0x9d +#define RT5670_VAD_CTRL5 0x9e +/* Function - Digital */ +#define RT5670_ADC_EQ_CTRL1 0xae +#define RT5670_ADC_EQ_CTRL2 0xaf +#define RT5670_EQ_CTRL1 0xb0 +#define RT5670_EQ_CTRL2 0xb1 +#define RT5670_ALC_DRC_CTRL1 0xb2 +#define RT5670_ALC_DRC_CTRL2 0xb3 +#define RT5670_ALC_CTRL_1 0xb4 +#define RT5670_ALC_CTRL_2 0xb5 +#define RT5670_ALC_CTRL_3 0xb6 +#define RT5670_ALC_CTRL_4 0xb7 +#define RT5670_JD_CTRL 0xbb +#define RT5670_IRQ_CTRL1 0xbd +#define RT5670_IRQ_CTRL2 0xbe +#define RT5670_INT_IRQ_ST 0xbf +#define RT5670_GPIO_CTRL1 0xc0 +#define RT5670_GPIO_CTRL2 0xc1 +#define RT5670_GPIO_CTRL3 0xc2 +#define RT5670_SCRABBLE_FUN 0xcd +#define RT5670_SCRABBLE_CTRL 0xce +#define RT5670_BASE_BACK 0xcf +#define RT5670_MP3_PLUS1 0xd0 +#define RT5670_MP3_PLUS2 0xd1 +#define RT5670_ADJ_HPF1 0xd3 +#define RT5670_ADJ_HPF2 0xd4 +#define RT5670_HP_CALIB_AMP_DET 0xd6 +#define RT5670_SV_ZCD1 0xd9 +#define RT5670_SV_ZCD2 0xda +#define RT5670_IL_CMD 0xdb +#define RT5670_IL_CMD2 0xdc +#define RT5670_IL_CMD3 0xdd +#define RT5670_DRC_HL_CTRL1 0xe6 +#define RT5670_DRC_HL_CTRL2 0xe7 +#define RT5670_ADC_MONO_HP_CTRL1 0xec +#define RT5670_ADC_MONO_HP_CTRL2 0xed +#define RT5670_ADC_STO2_HP_CTRL1 0xee +#define RT5670_ADC_STO2_HP_CTRL2 0xef +#define RT5670_JD_CTRL3 0xf8 +#define RT5670_JD_CTRL4 0xf9 +/* General Control */ +#define RT5670_DIG_MISC 0xfa +#define RT5670_GEN_CTRL2 0xfb +#define RT5670_GEN_CTRL3 0xfc + + +/* Index of Codec Private Register definition */ +#define RT5670_DIG_VOL 0x00 +#define RT5670_PR_ALC_CTRL_1 0x01 +#define RT5670_PR_ALC_CTRL_2 0x02 +#define RT5670_PR_ALC_CTRL_3 0x03 +#define RT5670_PR_ALC_CTRL_4 0x04 +#define RT5670_PR_ALC_CTRL_5 0x05 +#define RT5670_PR_ALC_CTRL_6 0x06 +#define RT5670_BIAS_CUR1 0x12 +#define RT5670_BIAS_CUR3 0x14 +#define RT5670_CLSD_INT_REG1 0x1c +#define RT5670_MAMP_INT_REG2 0x37 +#define RT5670_CHOP_DAC_ADC 0x3d +#define RT5670_MIXER_INT_REG 0x3f +#define RT5670_3D_SPK 0x63 +#define RT5670_WND_1 0x6c +#define RT5670_WND_2 0x6d +#define RT5670_WND_3 0x6e +#define RT5670_WND_4 0x6f +#define RT5670_WND_5 0x70 +#define RT5670_WND_8 0x73 +#define RT5670_DIP_SPK_INF 0x75 +#define RT5670_HP_DCC_INT1 0x77 +#define RT5670_EQ_BW_LOP 0xa0 +#define RT5670_EQ_GN_LOP 0xa1 +#define RT5670_EQ_FC_BP1 0xa2 +#define RT5670_EQ_BW_BP1 0xa3 +#define RT5670_EQ_GN_BP1 0xa4 +#define RT5670_EQ_FC_BP2 0xa5 +#define RT5670_EQ_BW_BP2 0xa6 +#define RT5670_EQ_GN_BP2 0xa7 +#define RT5670_EQ_FC_BP3 0xa8 +#define RT5670_EQ_BW_BP3 0xa9 +#define RT5670_EQ_GN_BP3 0xaa +#define RT5670_EQ_FC_BP4 0xab +#define RT5670_EQ_BW_BP4 0xac +#define RT5670_EQ_GN_BP4 0xad +#define RT5670_EQ_FC_HIP1 0xae +#define RT5670_EQ_GN_HIP1 0xaf +#define RT5670_EQ_FC_HIP2 0xb0 +#define RT5670_EQ_BW_HIP2 0xb1 +#define RT5670_EQ_GN_HIP2 0xb2 +#define RT5670_EQ_PRE_VOL 0xb3 +#define RT5670_EQ_PST_VOL 0xb4 + + +/* global definition */ +#define RT5670_L_MUTE (0x1 << 15) +#define RT5670_L_MUTE_SFT 15 +#define RT5670_VOL_L_MUTE (0x1 << 14) +#define RT5670_VOL_L_SFT 14 +#define RT5670_R_MUTE (0x1 << 7) +#define RT5670_R_MUTE_SFT 7 +#define RT5670_VOL_R_MUTE (0x1 << 6) +#define RT5670_VOL_R_SFT 6 +#define RT5670_L_VOL_MASK (0x3f << 8) +#define RT5670_L_VOL_SFT 8 +#define RT5670_R_VOL_MASK (0x3f) +#define RT5670_R_VOL_SFT 0 + +/* Combo Jack Control 1 (0x0a) */ +#define RT5670_CBJ_BST1_MASK (0xf << 12) +#define RT5670_CBJ_BST1_SFT (12) +#define RT5670_CBJ_JD_HP_EN (0x1 << 9) +#define RT5670_CBJ_JD_MIC_EN (0x1 << 8) +#define RT5670_CBJ_BST1_EN (0x1 << 2) + +/* Combo Jack Control 1 (0x0b) */ +#define RT5670_CBJ_MN_JD (0x1 << 12) +#define RT5670_CAPLESS_EN (0x1 << 11) +#define RT5670_CBJ_DET_MODE (0x1 << 7) + +/* IN2 Control (0x0e) */ +#define RT5670_BST_MASK1 (0xf<<12) +#define RT5670_BST_SFT1 12 +#define RT5670_BST_MASK2 (0xf<<8) +#define RT5670_BST_SFT2 8 +#define RT5670_IN_DF1 (0x1 << 7) +#define RT5670_IN_SFT1 7 +#define RT5670_IN_DF2 (0x1 << 6) +#define RT5670_IN_SFT2 6 + +/* INL and INR Volume Control (0x0f) */ +#define RT5670_INL_SEL_MASK (0x1 << 15) +#define RT5670_INL_SEL_SFT 15 +#define RT5670_INL_SEL_IN4P (0x0 << 15) +#define RT5670_INL_SEL_MONOP (0x1 << 15) +#define RT5670_INL_VOL_MASK (0x1f << 8) +#define RT5670_INL_VOL_SFT 8 +#define RT5670_INR_SEL_MASK (0x1 << 7) +#define RT5670_INR_SEL_SFT 7 +#define RT5670_INR_SEL_IN4N (0x0 << 7) +#define RT5670_INR_SEL_MONON (0x1 << 7) +#define RT5670_INR_VOL_MASK (0x1f) +#define RT5670_INR_VOL_SFT 0 + +/* Sidetone Control (0x18) */ +#define RT5670_ST_SEL_MASK (0x7 << 9) +#define RT5670_ST_SEL_SFT 9 +#define RT5670_M_ST_DACR2 (0x1 << 8) +#define RT5670_M_ST_DACR2_SFT 8 +#define RT5670_M_ST_DACL2 (0x1 << 7) +#define RT5670_M_ST_DACL2_SFT 7 +#define RT5670_ST_EN (0x1 << 6) +#define RT5670_ST_EN_SFT 6 + +/* DAC1 Digital Volume (0x19) */ +#define RT5670_DAC_L1_VOL_MASK (0xff << 8) +#define RT5670_DAC_L1_VOL_SFT 8 +#define RT5670_DAC_R1_VOL_MASK (0xff) +#define RT5670_DAC_R1_VOL_SFT 0 + +/* DAC2 Digital Volume (0x1a) */ +#define RT5670_DAC_L2_VOL_MASK (0xff << 8) +#define RT5670_DAC_L2_VOL_SFT 8 +#define RT5670_DAC_R2_VOL_MASK (0xff) +#define RT5670_DAC_R2_VOL_SFT 0 + +/* DAC2 Control (0x1b) */ +#define RT5670_M_DAC_L2_VOL (0x1 << 13) +#define RT5670_M_DAC_L2_VOL_SFT 13 +#define RT5670_M_DAC_R2_VOL (0x1 << 12) +#define RT5670_M_DAC_R2_VOL_SFT 12 +#define RT5670_DAC2_L_SEL_MASK (0x7 << 4) +#define RT5670_DAC2_L_SEL_SFT 4 +#define RT5670_DAC2_R_SEL_MASK (0x7 << 0) +#define RT5670_DAC2_R_SEL_SFT 0 + +/* ADC Digital Volume Control (0x1c) */ +#define RT5670_ADC_L_VOL_MASK (0x7f << 8) +#define RT5670_ADC_L_VOL_SFT 8 +#define RT5670_ADC_R_VOL_MASK (0x7f) +#define RT5670_ADC_R_VOL_SFT 0 + +/* Mono ADC Digital Volume Control (0x1d) */ +#define RT5670_MONO_ADC_L_VOL_MASK (0x7f << 8) +#define RT5670_MONO_ADC_L_VOL_SFT 8 +#define RT5670_MONO_ADC_R_VOL_MASK (0x7f) +#define RT5670_MONO_ADC_R_VOL_SFT 0 + +/* ADC Boost Volume Control (0x1e) */ +#define RT5670_STO1_ADC_L_BST_MASK (0x3 << 14) +#define RT5670_STO1_ADC_L_BST_SFT 14 +#define RT5670_STO1_ADC_R_BST_MASK (0x3 << 12) +#define RT5670_STO1_ADC_R_BST_SFT 12 +#define RT5670_STO1_ADC_COMP_MASK (0x3 << 10) +#define RT5670_STO1_ADC_COMP_SFT 10 +#define RT5670_STO2_ADC_L_BST_MASK (0x3 << 8) +#define RT5670_STO2_ADC_L_BST_SFT 8 +#define RT5670_STO2_ADC_R_BST_MASK (0x3 << 6) +#define RT5670_STO2_ADC_R_BST_SFT 6 +#define RT5670_STO2_ADC_COMP_MASK (0x3 << 4) +#define RT5670_STO2_ADC_COMP_SFT 4 + +/* Stereo2 ADC Mixer Control (0x26) */ +#define RT5670_STO2_ADC_SRC_MASK (0x1 << 15) +#define RT5670_STO2_ADC_SRC_SFT 15 + +/* Stereo ADC Mixer Control (0x26 0x27) */ +#define RT5670_M_ADC_L1 (0x1 << 14) +#define RT5670_M_ADC_L1_SFT 14 +#define RT5670_M_ADC_L2 (0x1 << 13) +#define RT5670_M_ADC_L2_SFT 13 +#define RT5670_ADC_1_SRC_MASK (0x1 << 12) +#define RT5670_ADC_1_SRC_SFT 12 +#define RT5670_ADC_1_SRC_ADC (0x1 << 12) +#define RT5670_ADC_1_SRC_DACMIX (0x0 << 12) +#define RT5670_ADC_2_SRC_MASK (0x1 << 11) +#define RT5670_ADC_2_SRC_SFT 11 +#define RT5670_ADC_SRC_MASK (0x1 << 10) +#define RT5670_ADC_SRC_SFT 10 +#define RT5670_DMIC_SRC_MASK (0x3 << 8) +#define RT5670_DMIC_SRC_SFT 8 +#define RT5670_M_ADC_R1 (0x1 << 6) +#define RT5670_M_ADC_R1_SFT 6 +#define RT5670_M_ADC_R2 (0x1 << 5) +#define RT5670_M_ADC_R2_SFT 5 +#define RT5670_DMIC3_SRC_MASK (0x1 << 1) +#define RT5670_DMIC3_SRC_SFT 0 + +/* Mono ADC Mixer Control (0x28) */ +#define RT5670_M_MONO_ADC_L1 (0x1 << 14) +#define RT5670_M_MONO_ADC_L1_SFT 14 +#define RT5670_M_MONO_ADC_L2 (0x1 << 13) +#define RT5670_M_MONO_ADC_L2_SFT 13 +#define RT5670_MONO_ADC_L1_SRC_MASK (0x1 << 12) +#define RT5670_MONO_ADC_L1_SRC_SFT 12 +#define RT5670_MONO_ADC_L1_SRC_DACMIXL (0x0 << 12) +#define RT5670_MONO_ADC_L1_SRC_ADCL (0x1 << 12) +#define RT5670_MONO_ADC_L2_SRC_MASK (0x1 << 11) +#define RT5670_MONO_ADC_L2_SRC_SFT 11 +#define RT5670_MONO_ADC_L_SRC_MASK (0x1 << 10) +#define RT5670_MONO_ADC_L_SRC_SFT 10 +#define RT5670_MONO_DMIC_L_SRC_MASK (0x3 << 8) +#define RT5670_MONO_DMIC_L_SRC_SFT 8 +#define RT5670_M_MONO_ADC_R1 (0x1 << 6) +#define RT5670_M_MONO_ADC_R1_SFT 6 +#define RT5670_M_MONO_ADC_R2 (0x1 << 5) +#define RT5670_M_MONO_ADC_R2_SFT 5 +#define RT5670_MONO_ADC_R1_SRC_MASK (0x1 << 4) +#define RT5670_MONO_ADC_R1_SRC_SFT 4 +#define RT5670_MONO_ADC_R1_SRC_ADCR (0x1 << 4) +#define RT5670_MONO_ADC_R1_SRC_DACMIXR (0x0 << 4) +#define RT5670_MONO_ADC_R2_SRC_MASK (0x1 << 3) +#define RT5670_MONO_ADC_R2_SRC_SFT 3 +#define RT5670_MONO_DMIC_R_SRC_MASK (0x3) +#define RT5670_MONO_DMIC_R_SRC_SFT 0 + +/* ADC Mixer to DAC Mixer Control (0x29) */ +#define RT5670_M_ADCMIX_L (0x1 << 15) +#define RT5670_M_ADCMIX_L_SFT 15 +#define RT5670_M_DAC1_L (0x1 << 14) +#define RT5670_M_DAC1_L_SFT 14 +#define RT5670_DAC1_R_SEL_MASK (0x3 << 10) +#define RT5670_DAC1_R_SEL_SFT 10 +#define RT5670_DAC1_R_SEL_IF1 (0x0 << 10) +#define RT5670_DAC1_R_SEL_IF2 (0x1 << 10) +#define RT5670_DAC1_R_SEL_IF3 (0x2 << 10) +#define RT5670_DAC1_R_SEL_IF4 (0x3 << 10) +#define RT5670_DAC1_L_SEL_MASK (0x3 << 8) +#define RT5670_DAC1_L_SEL_SFT 8 +#define RT5670_DAC1_L_SEL_IF1 (0x0 << 8) +#define RT5670_DAC1_L_SEL_IF2 (0x1 << 8) +#define RT5670_DAC1_L_SEL_IF3 (0x2 << 8) +#define RT5670_DAC1_L_SEL_IF4 (0x3 << 8) +#define RT5670_M_ADCMIX_R (0x1 << 7) +#define RT5670_M_ADCMIX_R_SFT 7 +#define RT5670_M_DAC1_R (0x1 << 6) +#define RT5670_M_DAC1_R_SFT 6 + +/* Stereo DAC Mixer Control (0x2a) */ +#define RT5670_M_DAC_L1 (0x1 << 14) +#define RT5670_M_DAC_L1_SFT 14 +#define RT5670_DAC_L1_STO_L_VOL_MASK (0x1 << 13) +#define RT5670_DAC_L1_STO_L_VOL_SFT 13 +#define RT5670_M_DAC_L2 (0x1 << 12) +#define RT5670_M_DAC_L2_SFT 12 +#define RT5670_DAC_L2_STO_L_VOL_MASK (0x1 << 11) +#define RT5670_DAC_L2_STO_L_VOL_SFT 11 +#define RT5670_M_DAC_R1_STO_L (0x1 << 9) +#define RT5670_M_DAC_R1_STO_L_SFT 9 +#define RT5670_DAC_R1_STO_L_VOL_MASK (0x1 << 8) +#define RT5670_DAC_R1_STO_L_VOL_SFT 8 +#define RT5670_M_DAC_R1 (0x1 << 6) +#define RT5670_M_DAC_R1_SFT 6 +#define RT5670_DAC_R1_STO_R_VOL_MASK (0x1 << 5) +#define RT5670_DAC_R1_STO_R_VOL_SFT 5 +#define RT5670_M_DAC_R2 (0x1 << 4) +#define RT5670_M_DAC_R2_SFT 4 +#define RT5670_DAC_R2_STO_R_VOL_MASK (0x1 << 3) +#define RT5670_DAC_R2_STO_R_VOL_SFT 3 +#define RT5670_M_DAC_L1_STO_R (0x1 << 1) +#define RT5670_M_DAC_L1_STO_R_SFT 1 +#define RT5670_DAC_L1_STO_R_VOL_MASK (0x1) +#define RT5670_DAC_L1_STO_R_VOL_SFT 0 + +/* Mono DAC Mixer Control (0x2b) */ +#define RT5670_M_DAC_L1_MONO_L (0x1 << 14) +#define RT5670_M_DAC_L1_MONO_L_SFT 14 +#define RT5670_DAC_L1_MONO_L_VOL_MASK (0x1 << 13) +#define RT5670_DAC_L1_MONO_L_VOL_SFT 13 +#define RT5670_M_DAC_L2_MONO_L (0x1 << 12) +#define RT5670_M_DAC_L2_MONO_L_SFT 12 +#define RT5670_DAC_L2_MONO_L_VOL_MASK (0x1 << 11) +#define RT5670_DAC_L2_MONO_L_VOL_SFT 11 +#define RT5670_M_DAC_R2_MONO_L (0x1 << 10) +#define RT5670_M_DAC_R2_MONO_L_SFT 10 +#define RT5670_DAC_R2_MONO_L_VOL_MASK (0x1 << 9) +#define RT5670_DAC_R2_MONO_L_VOL_SFT 9 +#define RT5670_M_DAC_R1_MONO_R (0x1 << 6) +#define RT5670_M_DAC_R1_MONO_R_SFT 6 +#define RT5670_DAC_R1_MONO_R_VOL_MASK (0x1 << 5) +#define RT5670_DAC_R1_MONO_R_VOL_SFT 5 +#define RT5670_M_DAC_R2_MONO_R (0x1 << 4) +#define RT5670_M_DAC_R2_MONO_R_SFT 4 +#define RT5670_DAC_R2_MONO_R_VOL_MASK (0x1 << 3) +#define RT5670_DAC_R2_MONO_R_VOL_SFT 3 +#define RT5670_M_DAC_L2_MONO_R (0x1 << 2) +#define RT5670_M_DAC_L2_MONO_R_SFT 2 +#define RT5670_DAC_L2_MONO_R_VOL_MASK (0x1 << 1) +#define RT5670_DAC_L2_MONO_R_VOL_SFT 1 + +/* Digital Mixer Control (0x2c) */ +#define RT5670_M_STO_L_DAC_L (0x1 << 15) +#define RT5670_M_STO_L_DAC_L_SFT 15 +#define RT5670_STO_L_DAC_L_VOL_MASK (0x1 << 14) +#define RT5670_STO_L_DAC_L_VOL_SFT 14 +#define RT5670_M_DAC_L2_DAC_L (0x1 << 13) +#define RT5670_M_DAC_L2_DAC_L_SFT 13 +#define RT5670_DAC_L2_DAC_L_VOL_MASK (0x1 << 12) +#define RT5670_DAC_L2_DAC_L_VOL_SFT 12 +#define RT5670_M_STO_R_DAC_R (0x1 << 11) +#define RT5670_M_STO_R_DAC_R_SFT 11 +#define RT5670_STO_R_DAC_R_VOL_MASK (0x1 << 10) +#define RT5670_STO_R_DAC_R_VOL_SFT 10 +#define RT5670_M_DAC_R2_DAC_R (0x1 << 9) +#define RT5670_M_DAC_R2_DAC_R_SFT 9 +#define RT5670_DAC_R2_DAC_R_VOL_MASK (0x1 << 8) +#define RT5670_DAC_R2_DAC_R_VOL_SFT 8 +#define RT5670_M_DAC_R2_DAC_L (0x1 << 7) +#define RT5670_M_DAC_R2_DAC_L_SFT 7 +#define RT5670_DAC_R2_DAC_L_VOL_MASK (0x1 << 6) +#define RT5670_DAC_R2_DAC_L_VOL_SFT 6 +#define RT5670_M_DAC_L2_DAC_R (0x1 << 5) +#define RT5670_M_DAC_L2_DAC_R_SFT 5 +#define RT5670_DAC_L2_DAC_R_VOL_MASK (0x1 << 4) +#define RT5670_DAC_L2_DAC_R_VOL_SFT 4 + +/* DSP Path Control 1 (0x2d) */ +#define RT5670_RXDP_SEL_MASK (0x7 << 13) +#define RT5670_RXDP_SEL_SFT 13 +#define RT5670_RXDP_SRC_MASK (0x3 << 11) +#define RT5670_RXDP_SRC_SFT 11 +#define RT5670_RXDP_SRC_NOR (0x0 << 11) +#define RT5670_RXDP_SRC_DIV2 (0x1 << 11) +#define RT5670_RXDP_SRC_DIV3 (0x2 << 11) +#define RT5670_TXDP_SRC_MASK (0x3 << 4) +#define RT5670_TXDP_SRC_SFT 4 +#define RT5670_TXDP_SRC_NOR (0x0 << 4) +#define RT5670_TXDP_SRC_DIV2 (0x1 << 4) +#define RT5670_TXDP_SRC_DIV3 (0x2 << 4) +#define RT5670_TXDP_SLOT_SEL_MASK (0x3 << 2) +#define RT5670_TXDP_SLOT_SEL_SFT 2 +#define RT5670_DSP_UL_SEL (0x1 << 1) +#define RT5670_DSP_UL_SFT 1 +#define RT5670_DSP_DL_SEL 0x1 +#define RT5670_DSP_DL_SFT 0 + +/* DSP Path Control 2 (0x2e) */ +#define RT5670_TXDP_L_VOL_MASK (0x7f << 8) +#define RT5670_TXDP_L_VOL_SFT 8 +#define RT5670_TXDP_R_VOL_MASK (0x7f) +#define RT5670_TXDP_R_VOL_SFT 0 + +/* Digital Interface Data Control (0x2f) */ +#define RT5670_IF1_ADC2_IN_SEL (0x1 << 15) +#define RT5670_IF1_ADC2_IN_SFT 15 +#define RT5670_IF2_ADC_IN_MASK (0x7 << 12) +#define RT5670_IF2_ADC_IN_SFT 12 +#define RT5670_IF2_DAC_SEL_MASK (0x3 << 10) +#define RT5670_IF2_DAC_SEL_SFT 10 +#define RT5670_IF2_ADC_SEL_MASK (0x3 << 8) +#define RT5670_IF2_ADC_SEL_SFT 8 + +/* Digital Interface Data Control (0x30) */ +#define RT5670_IF4_ADC_IN_MASK (0x3 << 4) +#define RT5670_IF4_ADC_IN_SFT 4 + +/* PDM Output Control (0x31) */ +#define RT5670_PDM1_L_MASK (0x1 << 15) +#define RT5670_PDM1_L_SFT 15 +#define RT5670_M_PDM1_L (0x1 << 14) +#define RT5670_M_PDM1_L_SFT 14 +#define RT5670_PDM1_R_MASK (0x1 << 13) +#define RT5670_PDM1_R_SFT 13 +#define RT5670_M_PDM1_R (0x1 << 12) +#define RT5670_M_PDM1_R_SFT 12 +#define RT5670_PDM2_L_MASK (0x1 << 11) +#define RT5670_PDM2_L_SFT 11 +#define RT5670_M_PDM2_L (0x1 << 10) +#define RT5670_M_PDM2_L_SFT 10 +#define RT5670_PDM2_R_MASK (0x1 << 9) +#define RT5670_PDM2_R_SFT 9 +#define RT5670_M_PDM2_R (0x1 << 8) +#define RT5670_M_PDM2_R_SFT 8 +#define RT5670_PDM2_BUSY (0x1 << 7) +#define RT5670_PDM1_BUSY (0x1 << 6) +#define RT5670_PDM_PATTERN (0x1 << 5) +#define RT5670_PDM_GAIN (0x1 << 4) +#define RT5670_PDM_DIV_MASK (0x3) + +/* REC Left Mixer Control 1 (0x3b) */ +#define RT5670_G_HP_L_RM_L_MASK (0x7 << 13) +#define RT5670_G_HP_L_RM_L_SFT 13 +#define RT5670_G_IN_L_RM_L_MASK (0x7 << 10) +#define RT5670_G_IN_L_RM_L_SFT 10 +#define RT5670_G_BST4_RM_L_MASK (0x7 << 7) +#define RT5670_G_BST4_RM_L_SFT 7 +#define RT5670_G_BST3_RM_L_MASK (0x7 << 4) +#define RT5670_G_BST3_RM_L_SFT 4 +#define RT5670_G_BST2_RM_L_MASK (0x7 << 1) +#define RT5670_G_BST2_RM_L_SFT 1 + +/* REC Left Mixer Control 2 (0x3c) */ +#define RT5670_G_BST1_RM_L_MASK (0x7 << 13) +#define RT5670_G_BST1_RM_L_SFT 13 +#define RT5670_M_IN_L_RM_L (0x1 << 5) +#define RT5670_M_IN_L_RM_L_SFT 5 +#define RT5670_M_BST2_RM_L (0x1 << 3) +#define RT5670_M_BST2_RM_L_SFT 3 +#define RT5670_M_BST1_RM_L (0x1 << 1) +#define RT5670_M_BST1_RM_L_SFT 1 + +/* REC Right Mixer Control 1 (0x3d) */ +#define RT5670_G_HP_R_RM_R_MASK (0x7 << 13) +#define RT5670_G_HP_R_RM_R_SFT 13 +#define RT5670_G_IN_R_RM_R_MASK (0x7 << 10) +#define RT5670_G_IN_R_RM_R_SFT 10 +#define RT5670_G_BST4_RM_R_MASK (0x7 << 7) +#define RT5670_G_BST4_RM_R_SFT 7 +#define RT5670_G_BST3_RM_R_MASK (0x7 << 4) +#define RT5670_G_BST3_RM_R_SFT 4 +#define RT5670_G_BST2_RM_R_MASK (0x7 << 1) +#define RT5670_G_BST2_RM_R_SFT 1 + +/* REC Right Mixer Control 2 (0x3e) */ +#define RT5670_G_BST1_RM_R_MASK (0x7 << 13) +#define RT5670_G_BST1_RM_R_SFT 13 +#define RT5670_M_IN_R_RM_R (0x1 << 5) +#define RT5670_M_IN_R_RM_R_SFT 5 +#define RT5670_M_BST2_RM_R (0x1 << 3) +#define RT5670_M_BST2_RM_R_SFT 3 +#define RT5670_M_BST1_RM_R (0x1 << 1) +#define RT5670_M_BST1_RM_R_SFT 1 + +/* HPMIX Control (0x45) */ +#define RT5670_M_DAC2_HM (0x1 << 15) +#define RT5670_M_DAC2_HM_SFT 15 +#define RT5670_M_HPVOL_HM (0x1 << 14) +#define RT5670_M_HPVOL_HM_SFT 14 +#define RT5670_M_DAC1_HM (0x1 << 13) +#define RT5670_M_DAC1_HM_SFT 13 +#define RT5670_G_HPOMIX_MASK (0x1 << 12) +#define RT5670_G_HPOMIX_SFT 12 +#define RT5670_M_INR1_HMR (0x1 << 3) +#define RT5670_M_INR1_HMR_SFT 3 +#define RT5670_M_DACR1_HMR (0x1 << 2) +#define RT5670_M_DACR1_HMR_SFT 2 +#define RT5670_M_INL1_HML (0x1 << 1) +#define RT5670_M_INL1_HML_SFT 1 +#define RT5670_M_DACL1_HML (0x1) +#define RT5670_M_DACL1_HML_SFT 0 + +/* Mono Output Mixer Control (0x4c) */ +#define RT5670_M_DAC_R2_MA (0x1 << 15) +#define RT5670_M_DAC_R2_MA_SFT 15 +#define RT5670_M_DAC_L2_MA (0x1 << 14) +#define RT5670_M_DAC_L2_MA_SFT 14 +#define RT5670_M_OV_R_MM (0x1 << 13) +#define RT5670_M_OV_R_MM_SFT 13 +#define RT5670_M_OV_L_MM (0x1 << 12) +#define RT5670_M_OV_L_MM_SFT 12 +#define RT5670_G_MONOMIX_MASK (0x1 << 10) +#define RT5670_G_MONOMIX_SFT 10 +#define RT5670_M_DAC_R2_MM (0x1 << 9) +#define RT5670_M_DAC_R2_MM_SFT 9 +#define RT5670_M_DAC_L2_MM (0x1 << 8) +#define RT5670_M_DAC_L2_MM_SFT 8 +#define RT5670_M_BST4_MM (0x1 << 7) +#define RT5670_M_BST4_MM_SFT 7 + +/* Output Left Mixer Control 1 (0x4d) */ +#define RT5670_G_BST3_OM_L_MASK (0x7 << 13) +#define RT5670_G_BST3_OM_L_SFT 13 +#define RT5670_G_BST2_OM_L_MASK (0x7 << 10) +#define RT5670_G_BST2_OM_L_SFT 10 +#define RT5670_G_BST1_OM_L_MASK (0x7 << 7) +#define RT5670_G_BST1_OM_L_SFT 7 +#define RT5670_G_IN_L_OM_L_MASK (0x7 << 4) +#define RT5670_G_IN_L_OM_L_SFT 4 +#define RT5670_G_RM_L_OM_L_MASK (0x7 << 1) +#define RT5670_G_RM_L_OM_L_SFT 1 + +/* Output Left Mixer Control 2 (0x4e) */ +#define RT5670_G_DAC_R2_OM_L_MASK (0x7 << 13) +#define RT5670_G_DAC_R2_OM_L_SFT 13 +#define RT5670_G_DAC_L2_OM_L_MASK (0x7 << 10) +#define RT5670_G_DAC_L2_OM_L_SFT 10 +#define RT5670_G_DAC_L1_OM_L_MASK (0x7 << 7) +#define RT5670_G_DAC_L1_OM_L_SFT 7 + +/* Output Left Mixer Control 3 (0x4f) */ +#define RT5670_M_BST1_OM_L (0x1 << 5) +#define RT5670_M_BST1_OM_L_SFT 5 +#define RT5670_M_IN_L_OM_L (0x1 << 4) +#define RT5670_M_IN_L_OM_L_SFT 4 +#define RT5670_M_DAC_L2_OM_L (0x1 << 1) +#define RT5670_M_DAC_L2_OM_L_SFT 1 +#define RT5670_M_DAC_L1_OM_L (0x1) +#define RT5670_M_DAC_L1_OM_L_SFT 0 + +/* Output Right Mixer Control 1 (0x50) */ +#define RT5670_G_BST4_OM_R_MASK (0x7 << 13) +#define RT5670_G_BST4_OM_R_SFT 13 +#define RT5670_G_BST2_OM_R_MASK (0x7 << 10) +#define RT5670_G_BST2_OM_R_SFT 10 +#define RT5670_G_BST1_OM_R_MASK (0x7 << 7) +#define RT5670_G_BST1_OM_R_SFT 7 +#define RT5670_G_IN_R_OM_R_MASK (0x7 << 4) +#define RT5670_G_IN_R_OM_R_SFT 4 +#define RT5670_G_RM_R_OM_R_MASK (0x7 << 1) +#define RT5670_G_RM_R_OM_R_SFT 1 + +/* Output Right Mixer Control 2 (0x51) */ +#define RT5670_G_DAC_L2_OM_R_MASK (0x7 << 13) +#define RT5670_G_DAC_L2_OM_R_SFT 13 +#define RT5670_G_DAC_R2_OM_R_MASK (0x7 << 10) +#define RT5670_G_DAC_R2_OM_R_SFT 10 +#define RT5670_G_DAC_R1_OM_R_MASK (0x7 << 7) +#define RT5670_G_DAC_R1_OM_R_SFT 7 + +/* Output Right Mixer Control 3 (0x52) */ +#define RT5670_M_BST2_OM_R (0x1 << 6) +#define RT5670_M_BST2_OM_R_SFT 6 +#define RT5670_M_IN_R_OM_R (0x1 << 4) +#define RT5670_M_IN_R_OM_R_SFT 4 +#define RT5670_M_DAC_R2_OM_R (0x1 << 1) +#define RT5670_M_DAC_R2_OM_R_SFT 1 +#define RT5670_M_DAC_R1_OM_R (0x1) +#define RT5670_M_DAC_R1_OM_R_SFT 0 + +/* LOUT Mixer Control (0x53) */ +#define RT5670_M_DAC_L1_LM (0x1 << 15) +#define RT5670_M_DAC_L1_LM_SFT 15 +#define RT5670_M_DAC_R1_LM (0x1 << 14) +#define RT5670_M_DAC_R1_LM_SFT 14 +#define RT5670_M_OV_L_LM (0x1 << 13) +#define RT5670_M_OV_L_LM_SFT 13 +#define RT5670_M_OV_R_LM (0x1 << 12) +#define RT5670_M_OV_R_LM_SFT 12 +#define RT5670_G_LOUTMIX_MASK (0x1 << 11) +#define RT5670_G_LOUTMIX_SFT 11 + +/* Power Management for Digital 1 (0x61) */ +#define RT5670_PWR_I2S1 (0x1 << 15) +#define RT5670_PWR_I2S1_BIT 15 +#define RT5670_PWR_I2S2 (0x1 << 14) +#define RT5670_PWR_I2S2_BIT 14 +#define RT5670_PWR_DAC_L1 (0x1 << 12) +#define RT5670_PWR_DAC_L1_BIT 12 +#define RT5670_PWR_DAC_R1 (0x1 << 11) +#define RT5670_PWR_DAC_R1_BIT 11 +#define RT5670_PWR_DAC_L2 (0x1 << 7) +#define RT5670_PWR_DAC_L2_BIT 7 +#define RT5670_PWR_DAC_R2 (0x1 << 6) +#define RT5670_PWR_DAC_R2_BIT 6 +#define RT5670_PWR_ADC_L (0x1 << 2) +#define RT5670_PWR_ADC_L_BIT 2 +#define RT5670_PWR_ADC_R (0x1 << 1) +#define RT5670_PWR_ADC_R_BIT 1 +#define RT5670_PWR_CLS_D (0x1) +#define RT5670_PWR_CLS_D_BIT 0 + +/* Power Management for Digital 2 (0x62) */ +#define RT5670_PWR_ADC_S1F (0x1 << 15) +#define RT5670_PWR_ADC_S1F_BIT 15 +#define RT5670_PWR_ADC_MF_L (0x1 << 14) +#define RT5670_PWR_ADC_MF_L_BIT 14 +#define RT5670_PWR_ADC_MF_R (0x1 << 13) +#define RT5670_PWR_ADC_MF_R_BIT 13 +#define RT5670_PWR_I2S_DSP (0x1 << 12) +#define RT5670_PWR_I2S_DSP_BIT 12 +#define RT5670_PWR_DAC_S1F (0x1 << 11) +#define RT5670_PWR_DAC_S1F_BIT 11 +#define RT5670_PWR_DAC_MF_L (0x1 << 10) +#define RT5670_PWR_DAC_MF_L_BIT 10 +#define RT5670_PWR_DAC_MF_R (0x1 << 9) +#define RT5670_PWR_DAC_MF_R_BIT 9 +#define RT5670_PWR_ADC_S2F (0x1 << 8) +#define RT5670_PWR_ADC_S2F_BIT 8 +#define RT5670_PWR_PDM1 (0x1 << 7) +#define RT5670_PWR_PDM1_BIT 7 +#define RT5670_PWR_PDM2 (0x1 << 6) +#define RT5670_PWR_PDM2_BIT 6 + +/* Power Management for Analog 1 (0x63) */ +#define RT5670_PWR_VREF1 (0x1 << 15) +#define RT5670_PWR_VREF1_BIT 15 +#define RT5670_PWR_FV1 (0x1 << 14) +#define RT5670_PWR_FV1_BIT 14 +#define RT5670_PWR_MB (0x1 << 13) +#define RT5670_PWR_MB_BIT 13 +#define RT5670_PWR_LM (0x1 << 12) +#define RT5670_PWR_LM_BIT 12 +#define RT5670_PWR_BG (0x1 << 11) +#define RT5670_PWR_BG_BIT 11 +#define RT5670_PWR_HP_L (0x1 << 7) +#define RT5670_PWR_HP_L_BIT 7 +#define RT5670_PWR_HP_R (0x1 << 6) +#define RT5670_PWR_HP_R_BIT 6 +#define RT5670_PWR_HA (0x1 << 5) +#define RT5670_PWR_HA_BIT 5 +#define RT5670_PWR_VREF2 (0x1 << 4) +#define RT5670_PWR_VREF2_BIT 4 +#define RT5670_PWR_FV2 (0x1 << 3) +#define RT5670_PWR_FV2_BIT 3 +#define RT5670_LDO_SEL_MASK (0x3) +#define RT5670_LDO_SEL_SFT 0 + +/* Power Management for Analog 2 (0x64) */ +#define RT5670_PWR_BST1 (0x1 << 15) +#define RT5670_PWR_BST1_BIT 15 +#define RT5670_PWR_BST2 (0x1 << 13) +#define RT5670_PWR_BST2_BIT 13 +#define RT5670_PWR_MB1 (0x1 << 11) +#define RT5670_PWR_MB1_BIT 11 +#define RT5670_PWR_MB2 (0x1 << 10) +#define RT5670_PWR_MB2_BIT 10 +#define RT5670_PWR_PLL (0x1 << 9) +#define RT5670_PWR_PLL_BIT 9 +#define RT5670_PWR_BST1_P (0x1 << 6) +#define RT5670_PWR_BST1_P_BIT 6 +#define RT5670_PWR_BST2_P (0x1 << 4) +#define RT5670_PWR_BST2_P_BIT 4 +#define RT5670_PWR_JD1 (0x1 << 2) +#define RT5670_PWR_JD1_BIT 2 +#define RT5670_PWR_JD (0x1 << 1) +#define RT5670_PWR_JD_BIT 1 + +/* Power Management for Mixer (0x65) */ +#define RT5670_PWR_OM_L (0x1 << 15) +#define RT5670_PWR_OM_L_BIT 15 +#define RT5670_PWR_OM_R (0x1 << 14) +#define RT5670_PWR_OM_R_BIT 14 +#define RT5670_PWR_RM_L (0x1 << 11) +#define RT5670_PWR_RM_L_BIT 11 +#define RT5670_PWR_RM_R (0x1 << 10) +#define RT5670_PWR_RM_R_BIT 10 + +/* Power Management for Volume (0x66) */ +#define RT5670_PWR_HV_L (0x1 << 11) +#define RT5670_PWR_HV_L_BIT 11 +#define RT5670_PWR_HV_R (0x1 << 10) +#define RT5670_PWR_HV_R_BIT 10 +#define RT5670_PWR_IN_L (0x1 << 9) +#define RT5670_PWR_IN_L_BIT 9 +#define RT5670_PWR_IN_R (0x1 << 8) +#define RT5670_PWR_IN_R_BIT 8 +#define RT5670_PWR_MIC_DET (0x1 << 5) +#define RT5670_PWR_MIC_DET_BIT 5 + +/* I2S1/2/3 Audio Serial Data Port Control (0x70 0x71 0x72) */ +#define RT5670_I2S_MS_MASK (0x1 << 15) +#define RT5670_I2S_MS_SFT 15 +#define RT5670_I2S_MS_M (0x0 << 15) +#define RT5670_I2S_MS_S (0x1 << 15) +#define RT5670_I2S_IF_MASK (0x7 << 12) +#define RT5670_I2S_IF_SFT 12 +#define RT5670_I2S_O_CP_MASK (0x3 << 10) +#define RT5670_I2S_O_CP_SFT 10 +#define RT5670_I2S_O_CP_OFF (0x0 << 10) +#define RT5670_I2S_O_CP_U_LAW (0x1 << 10) +#define RT5670_I2S_O_CP_A_LAW (0x2 << 10) +#define RT5670_I2S_I_CP_MASK (0x3 << 8) +#define RT5670_I2S_I_CP_SFT 8 +#define RT5670_I2S_I_CP_OFF (0x0 << 8) +#define RT5670_I2S_I_CP_U_LAW (0x1 << 8) +#define RT5670_I2S_I_CP_A_LAW (0x2 << 8) +#define RT5670_I2S_BP_MASK (0x1 << 7) +#define RT5670_I2S_BP_SFT 7 +#define RT5670_I2S_BP_NOR (0x0 << 7) +#define RT5670_I2S_BP_INV (0x1 << 7) +#define RT5670_I2S_DL_MASK (0x3 << 2) +#define RT5670_I2S_DL_SFT 2 +#define RT5670_I2S_DL_16 (0x0 << 2) +#define RT5670_I2S_DL_20 (0x1 << 2) +#define RT5670_I2S_DL_24 (0x2 << 2) +#define RT5670_I2S_DL_8 (0x3 << 2) +#define RT5670_I2S_DF_MASK (0x3) +#define RT5670_I2S_DF_SFT 0 +#define RT5670_I2S_DF_I2S (0x0) +#define RT5670_I2S_DF_LEFT (0x1) +#define RT5670_I2S_DF_PCM_A (0x2) +#define RT5670_I2S_DF_PCM_B (0x3) + +/* I2S2 Audio Serial Data Port Control (0x71) */ +#define RT5670_I2S2_SDI_MASK (0x1 << 6) +#define RT5670_I2S2_SDI_SFT 6 +#define RT5670_I2S2_SDI_I2S1 (0x0 << 6) +#define RT5670_I2S2_SDI_I2S2 (0x1 << 6) + +/* ADC/DAC Clock Control 1 (0x73) */ +#define RT5670_I2S_BCLK_MS1_MASK (0x1 << 15) +#define RT5670_I2S_BCLK_MS1_SFT 15 +#define RT5670_I2S_BCLK_MS1_32 (0x0 << 15) +#define RT5670_I2S_BCLK_MS1_64 (0x1 << 15) +#define RT5670_I2S_PD1_MASK (0x7 << 12) +#define RT5670_I2S_PD1_SFT 12 +#define RT5670_I2S_PD1_1 (0x0 << 12) +#define RT5670_I2S_PD1_2 (0x1 << 12) +#define RT5670_I2S_PD1_3 (0x2 << 12) +#define RT5670_I2S_PD1_4 (0x3 << 12) +#define RT5670_I2S_PD1_6 (0x4 << 12) +#define RT5670_I2S_PD1_8 (0x5 << 12) +#define RT5670_I2S_PD1_12 (0x6 << 12) +#define RT5670_I2S_PD1_16 (0x7 << 12) +#define RT5670_I2S_BCLK_MS2_MASK (0x1 << 11) +#define RT5670_I2S_BCLK_MS2_SFT 11 +#define RT5670_I2S_BCLK_MS2_32 (0x0 << 11) +#define RT5670_I2S_BCLK_MS2_64 (0x1 << 11) +#define RT5670_I2S_PD2_MASK (0x7 << 8) +#define RT5670_I2S_PD2_SFT 8 +#define RT5670_I2S_PD2_1 (0x0 << 8) +#define RT5670_I2S_PD2_2 (0x1 << 8) +#define RT5670_I2S_PD2_3 (0x2 << 8) +#define RT5670_I2S_PD2_4 (0x3 << 8) +#define RT5670_I2S_PD2_6 (0x4 << 8) +#define RT5670_I2S_PD2_8 (0x5 << 8) +#define RT5670_I2S_PD2_12 (0x6 << 8) +#define RT5670_I2S_PD2_16 (0x7 << 8) +#define RT5670_I2S_BCLK_MS3_MASK (0x1 << 7) +#define RT5670_I2S_BCLK_MS3_SFT 7 +#define RT5670_I2S_BCLK_MS3_32 (0x0 << 7) +#define RT5670_I2S_BCLK_MS3_64 (0x1 << 7) +#define RT5670_I2S_PD3_MASK (0x7 << 4) +#define RT5670_I2S_PD3_SFT 4 +#define RT5670_I2S_PD3_1 (0x0 << 4) +#define RT5670_I2S_PD3_2 (0x1 << 4) +#define RT5670_I2S_PD3_3 (0x2 << 4) +#define RT5670_I2S_PD3_4 (0x3 << 4) +#define RT5670_I2S_PD3_6 (0x4 << 4) +#define RT5670_I2S_PD3_8 (0x5 << 4) +#define RT5670_I2S_PD3_12 (0x6 << 4) +#define RT5670_I2S_PD3_16 (0x7 << 4) +#define RT5670_DAC_OSR_MASK (0x3 << 2) +#define RT5670_DAC_OSR_SFT 2 +#define RT5670_DAC_OSR_128 (0x0 << 2) +#define RT5670_DAC_OSR_64 (0x1 << 2) +#define RT5670_DAC_OSR_32 (0x2 << 2) +#define RT5670_DAC_OSR_16 (0x3 << 2) +#define RT5670_ADC_OSR_MASK (0x3) +#define RT5670_ADC_OSR_SFT 0 +#define RT5670_ADC_OSR_128 (0x0) +#define RT5670_ADC_OSR_64 (0x1) +#define RT5670_ADC_OSR_32 (0x2) +#define RT5670_ADC_OSR_16 (0x3) + +/* ADC/DAC Clock Control 2 (0x74) */ +#define RT5670_DAC_L_OSR_MASK (0x3 << 14) +#define RT5670_DAC_L_OSR_SFT 14 +#define RT5670_DAC_L_OSR_128 (0x0 << 14) +#define RT5670_DAC_L_OSR_64 (0x1 << 14) +#define RT5670_DAC_L_OSR_32 (0x2 << 14) +#define RT5670_DAC_L_OSR_16 (0x3 << 14) +#define RT5670_ADC_R_OSR_MASK (0x3 << 12) +#define RT5670_ADC_R_OSR_SFT 12 +#define RT5670_ADC_R_OSR_128 (0x0 << 12) +#define RT5670_ADC_R_OSR_64 (0x1 << 12) +#define RT5670_ADC_R_OSR_32 (0x2 << 12) +#define RT5670_ADC_R_OSR_16 (0x3 << 12) +#define RT5670_DAHPF_EN (0x1 << 11) +#define RT5670_DAHPF_EN_SFT 11 +#define RT5670_ADHPF_EN (0x1 << 10) +#define RT5670_ADHPF_EN_SFT 10 + +/* Digital Microphone Control (0x75) */ +#define RT5670_DMIC_1_EN_MASK (0x1 << 15) +#define RT5670_DMIC_1_EN_SFT 15 +#define RT5670_DMIC_1_DIS (0x0 << 15) +#define RT5670_DMIC_1_EN (0x1 << 15) +#define RT5670_DMIC_2_EN_MASK (0x1 << 14) +#define RT5670_DMIC_2_EN_SFT 14 +#define RT5670_DMIC_2_DIS (0x0 << 14) +#define RT5670_DMIC_2_EN (0x1 << 14) +#define RT5670_DMIC_1L_LH_MASK (0x1 << 13) +#define RT5670_DMIC_1L_LH_SFT 13 +#define RT5670_DMIC_1L_LH_FALLING (0x0 << 13) +#define RT5670_DMIC_1L_LH_RISING (0x1 << 13) +#define RT5670_DMIC_1R_LH_MASK (0x1 << 12) +#define RT5670_DMIC_1R_LH_SFT 12 +#define RT5670_DMIC_1R_LH_FALLING (0x0 << 12) +#define RT5670_DMIC_1R_LH_RISING (0x1 << 12) +#define RT5670_DMIC_2_DP_MASK (0x1 << 10) +#define RT5670_DMIC_2_DP_SFT 10 +#define RT5670_DMIC_2_DP_GPIO8 (0x0 << 10) +#define RT5670_DMIC_2_DP_IN3N (0x1 << 10) +#define RT5670_DMIC_2L_LH_MASK (0x1 << 9) +#define RT5670_DMIC_2L_LH_SFT 9 +#define RT5670_DMIC_2L_LH_FALLING (0x0 << 9) +#define RT5670_DMIC_2L_LH_RISING (0x1 << 9) +#define RT5670_DMIC_2R_LH_MASK (0x1 << 8) +#define RT5670_DMIC_2R_LH_SFT 8 +#define RT5670_DMIC_2R_LH_FALLING (0x0 << 8) +#define RT5670_DMIC_2R_LH_RISING (0x1 << 8) +#define RT5670_DMIC_CLK_MASK (0x7 << 5) +#define RT5670_DMIC_CLK_SFT 5 +#define RT5670_DMIC_3_EN_MASK (0x1 << 4) +#define RT5670_DMIC_3_EN_SFT 4 +#define RT5670_DMIC_3_DIS (0x0 << 4) +#define RT5670_DMIC_3_EN (0x1 << 4) +#define RT5670_DMIC_1_DP_MASK (0x3 << 0) +#define RT5670_DMIC_1_DP_SFT 0 +#define RT5670_DMIC_1_DP_GPIO6 (0x0 << 0) +#define RT5670_DMIC_1_DP_IN2P (0x1 << 0) +#define RT5670_DMIC_1_DP_GPIO7 (0x2 << 0) + +/* Digital Microphone Control2 (0x76) */ +#define RT5670_DMIC_3_DP_MASK (0x3 << 6) +#define RT5670_DMIC_3_DP_SFT 6 +#define RT5670_DMIC_3_DP_GPIO9 (0x0 << 6) +#define RT5670_DMIC_3_DP_GPIO10 (0x1 << 6) +#define RT5670_DMIC_3_DP_GPIO5 (0x2 << 6) + +/* Global Clock Control (0x80) */ +#define RT5670_SCLK_SRC_MASK (0x3 << 14) +#define RT5670_SCLK_SRC_SFT 14 +#define RT5670_SCLK_SRC_MCLK (0x0 << 14) +#define RT5670_SCLK_SRC_PLL1 (0x1 << 14) +#define RT5670_SCLK_SRC_RCCLK (0x2 << 14) /* 15MHz */ +#define RT5670_PLL1_SRC_MASK (0x3 << 12) +#define RT5670_PLL1_SRC_SFT 12 +#define RT5670_PLL1_SRC_MCLK (0x0 << 12) +#define RT5670_PLL1_SRC_BCLK1 (0x1 << 12) +#define RT5670_PLL1_SRC_BCLK2 (0x2 << 12) +#define RT5670_PLL1_SRC_BCLK3 (0x3 << 12) +#define RT5670_PLL1_PD_MASK (0x1 << 3) +#define RT5670_PLL1_PD_SFT 3 +#define RT5670_PLL1_PD_1 (0x0 << 3) +#define RT5670_PLL1_PD_2 (0x1 << 3) + +#define RT5670_PLL_INP_MAX 40000000 +#define RT5670_PLL_INP_MIN 256000 +/* PLL M/N/K Code Control 1 (0x81) */ +#define RT5670_PLL_N_MAX 0x1ff +#define RT5670_PLL_N_MASK (RT5670_PLL_N_MAX << 7) +#define RT5670_PLL_N_SFT 7 +#define RT5670_PLL_K_MAX 0x1f +#define RT5670_PLL_K_MASK (RT5670_PLL_K_MAX) +#define RT5670_PLL_K_SFT 0 + +/* PLL M/N/K Code Control 2 (0x82) */ +#define RT5670_PLL_M_MAX 0xf +#define RT5670_PLL_M_MASK (RT5670_PLL_M_MAX << 12) +#define RT5670_PLL_M_SFT 12 +#define RT5670_PLL_M_BP (0x1 << 11) +#define RT5670_PLL_M_BP_SFT 11 + +/* ASRC Control 1 (0x83) */ +#define RT5670_STO_T_MASK (0x1 << 15) +#define RT5670_STO_T_SFT 15 +#define RT5670_STO_T_SCLK (0x0 << 15) +#define RT5670_STO_T_LRCK1 (0x1 << 15) +#define RT5670_M1_T_MASK (0x1 << 14) +#define RT5670_M1_T_SFT 14 +#define RT5670_M1_T_I2S2 (0x0 << 14) +#define RT5670_M1_T_I2S2_D3 (0x1 << 14) +#define RT5670_I2S2_F_MASK (0x1 << 12) +#define RT5670_I2S2_F_SFT 12 +#define RT5670_I2S2_F_I2S2_D2 (0x0 << 12) +#define RT5670_I2S2_F_I2S1_TCLK (0x1 << 12) +#define RT5670_DMIC_1_M_MASK (0x1 << 9) +#define RT5670_DMIC_1_M_SFT 9 +#define RT5670_DMIC_1_M_NOR (0x0 << 9) +#define RT5670_DMIC_1_M_ASYN (0x1 << 9) +#define RT5670_DMIC_2_M_MASK (0x1 << 8) +#define RT5670_DMIC_2_M_SFT 8 +#define RT5670_DMIC_2_M_NOR (0x0 << 8) +#define RT5670_DMIC_2_M_ASYN (0x1 << 8) + +/* ASRC Control 2 (0x84) */ +#define RT5670_MDA_L_M_MASK (0x1 << 15) +#define RT5670_MDA_L_M_SFT 15 +#define RT5670_MDA_L_M_NOR (0x0 << 15) +#define RT5670_MDA_L_M_ASYN (0x1 << 15) +#define RT5670_MDA_R_M_MASK (0x1 << 14) +#define RT5670_MDA_R_M_SFT 14 +#define RT5670_MDA_R_M_NOR (0x0 << 14) +#define RT5670_MDA_R_M_ASYN (0x1 << 14) +#define RT5670_MAD_L_M_MASK (0x1 << 13) +#define RT5670_MAD_L_M_SFT 13 +#define RT5670_MAD_L_M_NOR (0x0 << 13) +#define RT5670_MAD_L_M_ASYN (0x1 << 13) +#define RT5670_MAD_R_M_MASK (0x1 << 12) +#define RT5670_MAD_R_M_SFT 12 +#define RT5670_MAD_R_M_NOR (0x0 << 12) +#define RT5670_MAD_R_M_ASYN (0x1 << 12) +#define RT5670_ADC_M_MASK (0x1 << 11) +#define RT5670_ADC_M_SFT 11 +#define RT5670_ADC_M_NOR (0x0 << 11) +#define RT5670_ADC_M_ASYN (0x1 << 11) +#define RT5670_STO_DAC_M_MASK (0x1 << 5) +#define RT5670_STO_DAC_M_SFT 5 +#define RT5670_STO_DAC_M_NOR (0x0 << 5) +#define RT5670_STO_DAC_M_ASYN (0x1 << 5) +#define RT5670_I2S1_R_D_MASK (0x1 << 4) +#define RT5670_I2S1_R_D_SFT 4 +#define RT5670_I2S1_R_D_DIS (0x0 << 4) +#define RT5670_I2S1_R_D_EN (0x1 << 4) +#define RT5670_I2S2_R_D_MASK (0x1 << 3) +#define RT5670_I2S2_R_D_SFT 3 +#define RT5670_I2S2_R_D_DIS (0x0 << 3) +#define RT5670_I2S2_R_D_EN (0x1 << 3) +#define RT5670_PRE_SCLK_MASK (0x3) +#define RT5670_PRE_SCLK_SFT 0 +#define RT5670_PRE_SCLK_512 (0x0) +#define RT5670_PRE_SCLK_1024 (0x1) +#define RT5670_PRE_SCLK_2048 (0x2) + +/* ASRC Control 3 (0x85) */ +#define RT5670_I2S1_RATE_MASK (0xf << 12) +#define RT5670_I2S1_RATE_SFT 12 +#define RT5670_I2S2_RATE_MASK (0xf << 8) +#define RT5670_I2S2_RATE_SFT 8 + +/* ASRC Control 4 (0x89) */ +#define RT5670_I2S1_PD_MASK (0x7 << 12) +#define RT5670_I2S1_PD_SFT 12 +#define RT5670_I2S2_PD_MASK (0x7 << 8) +#define RT5670_I2S2_PD_SFT 8 + +/* HPOUT Over Current Detection (0x8b) */ +#define RT5670_HP_OVCD_MASK (0x1 << 10) +#define RT5670_HP_OVCD_SFT 10 +#define RT5670_HP_OVCD_DIS (0x0 << 10) +#define RT5670_HP_OVCD_EN (0x1 << 10) +#define RT5670_HP_OC_TH_MASK (0x3 << 8) +#define RT5670_HP_OC_TH_SFT 8 +#define RT5670_HP_OC_TH_90 (0x0 << 8) +#define RT5670_HP_OC_TH_105 (0x1 << 8) +#define RT5670_HP_OC_TH_120 (0x2 << 8) +#define RT5670_HP_OC_TH_135 (0x3 << 8) + +/* Class D Over Current Control (0x8c) */ +#define RT5670_CLSD_OC_MASK (0x1 << 9) +#define RT5670_CLSD_OC_SFT 9 +#define RT5670_CLSD_OC_PU (0x0 << 9) +#define RT5670_CLSD_OC_PD (0x1 << 9) +#define RT5670_AUTO_PD_MASK (0x1 << 8) +#define RT5670_AUTO_PD_SFT 8 +#define RT5670_AUTO_PD_DIS (0x0 << 8) +#define RT5670_AUTO_PD_EN (0x1 << 8) +#define RT5670_CLSD_OC_TH_MASK (0x3f) +#define RT5670_CLSD_OC_TH_SFT 0 + +/* Class D Output Control (0x8d) */ +#define RT5670_CLSD_RATIO_MASK (0xf << 12) +#define RT5670_CLSD_RATIO_SFT 12 +#define RT5670_CLSD_OM_MASK (0x1 << 11) +#define RT5670_CLSD_OM_SFT 11 +#define RT5670_CLSD_OM_MONO (0x0 << 11) +#define RT5670_CLSD_OM_STO (0x1 << 11) +#define RT5670_CLSD_SCH_MASK (0x1 << 10) +#define RT5670_CLSD_SCH_SFT 10 +#define RT5670_CLSD_SCH_L (0x0 << 10) +#define RT5670_CLSD_SCH_S (0x1 << 10) + +/* Depop Mode Control 1 (0x8e) */ +#define RT5670_SMT_TRIG_MASK (0x1 << 15) +#define RT5670_SMT_TRIG_SFT 15 +#define RT5670_SMT_TRIG_DIS (0x0 << 15) +#define RT5670_SMT_TRIG_EN (0x1 << 15) +#define RT5670_HP_L_SMT_MASK (0x1 << 9) +#define RT5670_HP_L_SMT_SFT 9 +#define RT5670_HP_L_SMT_DIS (0x0 << 9) +#define RT5670_HP_L_SMT_EN (0x1 << 9) +#define RT5670_HP_R_SMT_MASK (0x1 << 8) +#define RT5670_HP_R_SMT_SFT 8 +#define RT5670_HP_R_SMT_DIS (0x0 << 8) +#define RT5670_HP_R_SMT_EN (0x1 << 8) +#define RT5670_HP_CD_PD_MASK (0x1 << 7) +#define RT5670_HP_CD_PD_SFT 7 +#define RT5670_HP_CD_PD_DIS (0x0 << 7) +#define RT5670_HP_CD_PD_EN (0x1 << 7) +#define RT5670_RSTN_MASK (0x1 << 6) +#define RT5670_RSTN_SFT 6 +#define RT5670_RSTN_DIS (0x0 << 6) +#define RT5670_RSTN_EN (0x1 << 6) +#define RT5670_RSTP_MASK (0x1 << 5) +#define RT5670_RSTP_SFT 5 +#define RT5670_RSTP_DIS (0x0 << 5) +#define RT5670_RSTP_EN (0x1 << 5) +#define RT5670_HP_CO_MASK (0x1 << 4) +#define RT5670_HP_CO_SFT 4 +#define RT5670_HP_CO_DIS (0x0 << 4) +#define RT5670_HP_CO_EN (0x1 << 4) +#define RT5670_HP_CP_MASK (0x1 << 3) +#define RT5670_HP_CP_SFT 3 +#define RT5670_HP_CP_PD (0x0 << 3) +#define RT5670_HP_CP_PU (0x1 << 3) +#define RT5670_HP_SG_MASK (0x1 << 2) +#define RT5670_HP_SG_SFT 2 +#define RT5670_HP_SG_DIS (0x0 << 2) +#define RT5670_HP_SG_EN (0x1 << 2) +#define RT5670_HP_DP_MASK (0x1 << 1) +#define RT5670_HP_DP_SFT 1 +#define RT5670_HP_DP_PD (0x0 << 1) +#define RT5670_HP_DP_PU (0x1 << 1) +#define RT5670_HP_CB_MASK (0x1) +#define RT5670_HP_CB_SFT 0 +#define RT5670_HP_CB_PD (0x0) +#define RT5670_HP_CB_PU (0x1) + +/* Depop Mode Control 2 (0x8f) */ +#define RT5670_DEPOP_MASK (0x1 << 13) +#define RT5670_DEPOP_SFT 13 +#define RT5670_DEPOP_AUTO (0x0 << 13) +#define RT5670_DEPOP_MAN (0x1 << 13) +#define RT5670_RAMP_MASK (0x1 << 12) +#define RT5670_RAMP_SFT 12 +#define RT5670_RAMP_DIS (0x0 << 12) +#define RT5670_RAMP_EN (0x1 << 12) +#define RT5670_BPS_MASK (0x1 << 11) +#define RT5670_BPS_SFT 11 +#define RT5670_BPS_DIS (0x0 << 11) +#define RT5670_BPS_EN (0x1 << 11) +#define RT5670_FAST_UPDN_MASK (0x1 << 10) +#define RT5670_FAST_UPDN_SFT 10 +#define RT5670_FAST_UPDN_DIS (0x0 << 10) +#define RT5670_FAST_UPDN_EN (0x1 << 10) +#define RT5670_MRES_MASK (0x3 << 8) +#define RT5670_MRES_SFT 8 +#define RT5670_MRES_15MO (0x0 << 8) +#define RT5670_MRES_25MO (0x1 << 8) +#define RT5670_MRES_35MO (0x2 << 8) +#define RT5670_MRES_45MO (0x3 << 8) +#define RT5670_VLO_MASK (0x1 << 7) +#define RT5670_VLO_SFT 7 +#define RT5670_VLO_3V (0x0 << 7) +#define RT5670_VLO_32V (0x1 << 7) +#define RT5670_DIG_DP_MASK (0x1 << 6) +#define RT5670_DIG_DP_SFT 6 +#define RT5670_DIG_DP_DIS (0x0 << 6) +#define RT5670_DIG_DP_EN (0x1 << 6) +#define RT5670_DP_TH_MASK (0x3 << 4) +#define RT5670_DP_TH_SFT 4 + +/* Depop Mode Control 3 (0x90) */ +#define RT5670_CP_SYS_MASK (0x7 << 12) +#define RT5670_CP_SYS_SFT 12 +#define RT5670_CP_FQ1_MASK (0x7 << 8) +#define RT5670_CP_FQ1_SFT 8 +#define RT5670_CP_FQ2_MASK (0x7 << 4) +#define RT5670_CP_FQ2_SFT 4 +#define RT5670_CP_FQ3_MASK (0x7) +#define RT5670_CP_FQ3_SFT 0 +#define RT5670_CP_FQ_1_5_KHZ 0 +#define RT5670_CP_FQ_3_KHZ 1 +#define RT5670_CP_FQ_6_KHZ 2 +#define RT5670_CP_FQ_12_KHZ 3 +#define RT5670_CP_FQ_24_KHZ 4 +#define RT5670_CP_FQ_48_KHZ 5 +#define RT5670_CP_FQ_96_KHZ 6 +#define RT5670_CP_FQ_192_KHZ 7 + +/* HPOUT charge pump (0x91) */ +#define RT5670_OSW_L_MASK (0x1 << 11) +#define RT5670_OSW_L_SFT 11 +#define RT5670_OSW_L_DIS (0x0 << 11) +#define RT5670_OSW_L_EN (0x1 << 11) +#define RT5670_OSW_R_MASK (0x1 << 10) +#define RT5670_OSW_R_SFT 10 +#define RT5670_OSW_R_DIS (0x0 << 10) +#define RT5670_OSW_R_EN (0x1 << 10) +#define RT5670_PM_HP_MASK (0x3 << 8) +#define RT5670_PM_HP_SFT 8 +#define RT5670_PM_HP_LV (0x0 << 8) +#define RT5670_PM_HP_MV (0x1 << 8) +#define RT5670_PM_HP_HV (0x2 << 8) +#define RT5670_IB_HP_MASK (0x3 << 6) +#define RT5670_IB_HP_SFT 6 +#define RT5670_IB_HP_125IL (0x0 << 6) +#define RT5670_IB_HP_25IL (0x1 << 6) +#define RT5670_IB_HP_5IL (0x2 << 6) +#define RT5670_IB_HP_1IL (0x3 << 6) + +/* PV detection and SPK gain control (0x92) */ +#define RT5670_PVDD_DET_MASK (0x1 << 15) +#define RT5670_PVDD_DET_SFT 15 +#define RT5670_PVDD_DET_DIS (0x0 << 15) +#define RT5670_PVDD_DET_EN (0x1 << 15) +#define RT5670_SPK_AG_MASK (0x1 << 14) +#define RT5670_SPK_AG_SFT 14 +#define RT5670_SPK_AG_DIS (0x0 << 14) +#define RT5670_SPK_AG_EN (0x1 << 14) + +/* Micbias Control (0x93) */ +#define RT5670_MIC1_BS_MASK (0x1 << 15) +#define RT5670_MIC1_BS_SFT 15 +#define RT5670_MIC1_BS_9AV (0x0 << 15) +#define RT5670_MIC1_BS_75AV (0x1 << 15) +#define RT5670_MIC2_BS_MASK (0x1 << 14) +#define RT5670_MIC2_BS_SFT 14 +#define RT5670_MIC2_BS_9AV (0x0 << 14) +#define RT5670_MIC2_BS_75AV (0x1 << 14) +#define RT5670_MIC1_CLK_MASK (0x1 << 13) +#define RT5670_MIC1_CLK_SFT 13 +#define RT5670_MIC1_CLK_DIS (0x0 << 13) +#define RT5670_MIC1_CLK_EN (0x1 << 13) +#define RT5670_MIC2_CLK_MASK (0x1 << 12) +#define RT5670_MIC2_CLK_SFT 12 +#define RT5670_MIC2_CLK_DIS (0x0 << 12) +#define RT5670_MIC2_CLK_EN (0x1 << 12) +#define RT5670_MIC1_OVCD_MASK (0x1 << 11) +#define RT5670_MIC1_OVCD_SFT 11 +#define RT5670_MIC1_OVCD_DIS (0x0 << 11) +#define RT5670_MIC1_OVCD_EN (0x1 << 11) +#define RT5670_MIC1_OVTH_MASK (0x3 << 9) +#define RT5670_MIC1_OVTH_SFT 9 +#define RT5670_MIC1_OVTH_600UA (0x0 << 9) +#define RT5670_MIC1_OVTH_1500UA (0x1 << 9) +#define RT5670_MIC1_OVTH_2000UA (0x2 << 9) +#define RT5670_MIC2_OVCD_MASK (0x1 << 8) +#define RT5670_MIC2_OVCD_SFT 8 +#define RT5670_MIC2_OVCD_DIS (0x0 << 8) +#define RT5670_MIC2_OVCD_EN (0x1 << 8) +#define RT5670_MIC2_OVTH_MASK (0x3 << 6) +#define RT5670_MIC2_OVTH_SFT 6 +#define RT5670_MIC2_OVTH_600UA (0x0 << 6) +#define RT5670_MIC2_OVTH_1500UA (0x1 << 6) +#define RT5670_MIC2_OVTH_2000UA (0x2 << 6) +#define RT5670_PWR_MB_MASK (0x1 << 5) +#define RT5670_PWR_MB_SFT 5 +#define RT5670_PWR_MB_PD (0x0 << 5) +#define RT5670_PWR_MB_PU (0x1 << 5) +#define RT5670_PWR_CLK25M_MASK (0x1 << 4) +#define RT5670_PWR_CLK25M_SFT 4 +#define RT5670_PWR_CLK25M_PD (0x0 << 4) +#define RT5670_PWR_CLK25M_PU (0x1 << 4) + +/* Analog JD Control 1 (0x94) */ +#define RT5670_JD1_MODE_MASK (0x3 << 0) +#define RT5670_JD1_MODE_0 (0x0 << 0) +#define RT5670_JD1_MODE_1 (0x1 << 0) +#define RT5670_JD1_MODE_2 (0x2 << 0) + +/* VAD Control 4 (0x9d) */ +#define RT5670_VAD_SEL_MASK (0x3 << 8) +#define RT5670_VAD_SEL_SFT 8 + +/* EQ Control 1 (0xb0) */ +#define RT5670_EQ_SRC_MASK (0x1 << 15) +#define RT5670_EQ_SRC_SFT 15 +#define RT5670_EQ_SRC_DAC (0x0 << 15) +#define RT5670_EQ_SRC_ADC (0x1 << 15) +#define RT5670_EQ_UPD (0x1 << 14) +#define RT5670_EQ_UPD_BIT 14 +#define RT5670_EQ_CD_MASK (0x1 << 13) +#define RT5670_EQ_CD_SFT 13 +#define RT5670_EQ_CD_DIS (0x0 << 13) +#define RT5670_EQ_CD_EN (0x1 << 13) +#define RT5670_EQ_DITH_MASK (0x3 << 8) +#define RT5670_EQ_DITH_SFT 8 +#define RT5670_EQ_DITH_NOR (0x0 << 8) +#define RT5670_EQ_DITH_LSB (0x1 << 8) +#define RT5670_EQ_DITH_LSB_1 (0x2 << 8) +#define RT5670_EQ_DITH_LSB_2 (0x3 << 8) + +/* EQ Control 2 (0xb1) */ +#define RT5670_EQ_HPF1_M_MASK (0x1 << 8) +#define RT5670_EQ_HPF1_M_SFT 8 +#define RT5670_EQ_HPF1_M_HI (0x0 << 8) +#define RT5670_EQ_HPF1_M_1ST (0x1 << 8) +#define RT5670_EQ_LPF1_M_MASK (0x1 << 7) +#define RT5670_EQ_LPF1_M_SFT 7 +#define RT5670_EQ_LPF1_M_LO (0x0 << 7) +#define RT5670_EQ_LPF1_M_1ST (0x1 << 7) +#define RT5670_EQ_HPF2_MASK (0x1 << 6) +#define RT5670_EQ_HPF2_SFT 6 +#define RT5670_EQ_HPF2_DIS (0x0 << 6) +#define RT5670_EQ_HPF2_EN (0x1 << 6) +#define RT5670_EQ_HPF1_MASK (0x1 << 5) +#define RT5670_EQ_HPF1_SFT 5 +#define RT5670_EQ_HPF1_DIS (0x0 << 5) +#define RT5670_EQ_HPF1_EN (0x1 << 5) +#define RT5670_EQ_BPF4_MASK (0x1 << 4) +#define RT5670_EQ_BPF4_SFT 4 +#define RT5670_EQ_BPF4_DIS (0x0 << 4) +#define RT5670_EQ_BPF4_EN (0x1 << 4) +#define RT5670_EQ_BPF3_MASK (0x1 << 3) +#define RT5670_EQ_BPF3_SFT 3 +#define RT5670_EQ_BPF3_DIS (0x0 << 3) +#define RT5670_EQ_BPF3_EN (0x1 << 3) +#define RT5670_EQ_BPF2_MASK (0x1 << 2) +#define RT5670_EQ_BPF2_SFT 2 +#define RT5670_EQ_BPF2_DIS (0x0 << 2) +#define RT5670_EQ_BPF2_EN (0x1 << 2) +#define RT5670_EQ_BPF1_MASK (0x1 << 1) +#define RT5670_EQ_BPF1_SFT 1 +#define RT5670_EQ_BPF1_DIS (0x0 << 1) +#define RT5670_EQ_BPF1_EN (0x1 << 1) +#define RT5670_EQ_LPF_MASK (0x1) +#define RT5670_EQ_LPF_SFT 0 +#define RT5670_EQ_LPF_DIS (0x0) +#define RT5670_EQ_LPF_EN (0x1) +#define RT5670_EQ_CTRL_MASK (0x7f) + +/* Memory Test (0xb2) */ +#define RT5670_MT_MASK (0x1 << 15) +#define RT5670_MT_SFT 15 +#define RT5670_MT_DIS (0x0 << 15) +#define RT5670_MT_EN (0x1 << 15) + +/* DRC/AGC Control 1 (0xb4) */ +#define RT5670_DRC_AGC_P_MASK (0x1 << 15) +#define RT5670_DRC_AGC_P_SFT 15 +#define RT5670_DRC_AGC_P_DAC (0x0 << 15) +#define RT5670_DRC_AGC_P_ADC (0x1 << 15) +#define RT5670_DRC_AGC_MASK (0x1 << 14) +#define RT5670_DRC_AGC_SFT 14 +#define RT5670_DRC_AGC_DIS (0x0 << 14) +#define RT5670_DRC_AGC_EN (0x1 << 14) +#define RT5670_DRC_AGC_UPD (0x1 << 13) +#define RT5670_DRC_AGC_UPD_BIT 13 +#define RT5670_DRC_AGC_AR_MASK (0x1f << 8) +#define RT5670_DRC_AGC_AR_SFT 8 +#define RT5670_DRC_AGC_R_MASK (0x7 << 5) +#define RT5670_DRC_AGC_R_SFT 5 +#define RT5670_DRC_AGC_R_48K (0x1 << 5) +#define RT5670_DRC_AGC_R_96K (0x2 << 5) +#define RT5670_DRC_AGC_R_192K (0x3 << 5) +#define RT5670_DRC_AGC_R_441K (0x5 << 5) +#define RT5670_DRC_AGC_R_882K (0x6 << 5) +#define RT5670_DRC_AGC_R_1764K (0x7 << 5) +#define RT5670_DRC_AGC_RC_MASK (0x1f) +#define RT5670_DRC_AGC_RC_SFT 0 + +/* DRC/AGC Control 2 (0xb5) */ +#define RT5670_DRC_AGC_POB_MASK (0x3f << 8) +#define RT5670_DRC_AGC_POB_SFT 8 +#define RT5670_DRC_AGC_CP_MASK (0x1 << 7) +#define RT5670_DRC_AGC_CP_SFT 7 +#define RT5670_DRC_AGC_CP_DIS (0x0 << 7) +#define RT5670_DRC_AGC_CP_EN (0x1 << 7) +#define RT5670_DRC_AGC_CPR_MASK (0x3 << 5) +#define RT5670_DRC_AGC_CPR_SFT 5 +#define RT5670_DRC_AGC_CPR_1_1 (0x0 << 5) +#define RT5670_DRC_AGC_CPR_1_2 (0x1 << 5) +#define RT5670_DRC_AGC_CPR_1_3 (0x2 << 5) +#define RT5670_DRC_AGC_CPR_1_4 (0x3 << 5) +#define RT5670_DRC_AGC_PRB_MASK (0x1f) +#define RT5670_DRC_AGC_PRB_SFT 0 + +/* DRC/AGC Control 3 (0xb6) */ +#define RT5670_DRC_AGC_NGB_MASK (0xf << 12) +#define RT5670_DRC_AGC_NGB_SFT 12 +#define RT5670_DRC_AGC_TAR_MASK (0x1f << 7) +#define RT5670_DRC_AGC_TAR_SFT 7 +#define RT5670_DRC_AGC_NG_MASK (0x1 << 6) +#define RT5670_DRC_AGC_NG_SFT 6 +#define RT5670_DRC_AGC_NG_DIS (0x0 << 6) +#define RT5670_DRC_AGC_NG_EN (0x1 << 6) +#define RT5670_DRC_AGC_NGH_MASK (0x1 << 5) +#define RT5670_DRC_AGC_NGH_SFT 5 +#define RT5670_DRC_AGC_NGH_DIS (0x0 << 5) +#define RT5670_DRC_AGC_NGH_EN (0x1 << 5) +#define RT5670_DRC_AGC_NGT_MASK (0x1f) +#define RT5670_DRC_AGC_NGT_SFT 0 + +/* Jack Detect Control (0xbb) */ +#define RT5670_JD_MASK (0x7 << 13) +#define RT5670_JD_SFT 13 +#define RT5670_JD_DIS (0x0 << 13) +#define RT5670_JD_GPIO1 (0x1 << 13) +#define RT5670_JD_JD1_IN4P (0x2 << 13) +#define RT5670_JD_JD2_IN4N (0x3 << 13) +#define RT5670_JD_GPIO2 (0x4 << 13) +#define RT5670_JD_GPIO3 (0x5 << 13) +#define RT5670_JD_GPIO4 (0x6 << 13) +#define RT5670_JD_HP_MASK (0x1 << 11) +#define RT5670_JD_HP_SFT 11 +#define RT5670_JD_HP_DIS (0x0 << 11) +#define RT5670_JD_HP_EN (0x1 << 11) +#define RT5670_JD_HP_TRG_MASK (0x1 << 10) +#define RT5670_JD_HP_TRG_SFT 10 +#define RT5670_JD_HP_TRG_LO (0x0 << 10) +#define RT5670_JD_HP_TRG_HI (0x1 << 10) +#define RT5670_JD_SPL_MASK (0x1 << 9) +#define RT5670_JD_SPL_SFT 9 +#define RT5670_JD_SPL_DIS (0x0 << 9) +#define RT5670_JD_SPL_EN (0x1 << 9) +#define RT5670_JD_SPL_TRG_MASK (0x1 << 8) +#define RT5670_JD_SPL_TRG_SFT 8 +#define RT5670_JD_SPL_TRG_LO (0x0 << 8) +#define RT5670_JD_SPL_TRG_HI (0x1 << 8) +#define RT5670_JD_SPR_MASK (0x1 << 7) +#define RT5670_JD_SPR_SFT 7 +#define RT5670_JD_SPR_DIS (0x0 << 7) +#define RT5670_JD_SPR_EN (0x1 << 7) +#define RT5670_JD_SPR_TRG_MASK (0x1 << 6) +#define RT5670_JD_SPR_TRG_SFT 6 +#define RT5670_JD_SPR_TRG_LO (0x0 << 6) +#define RT5670_JD_SPR_TRG_HI (0x1 << 6) +#define RT5670_JD_MO_MASK (0x1 << 5) +#define RT5670_JD_MO_SFT 5 +#define RT5670_JD_MO_DIS (0x0 << 5) +#define RT5670_JD_MO_EN (0x1 << 5) +#define RT5670_JD_MO_TRG_MASK (0x1 << 4) +#define RT5670_JD_MO_TRG_SFT 4 +#define RT5670_JD_MO_TRG_LO (0x0 << 4) +#define RT5670_JD_MO_TRG_HI (0x1 << 4) +#define RT5670_JD_LO_MASK (0x1 << 3) +#define RT5670_JD_LO_SFT 3 +#define RT5670_JD_LO_DIS (0x0 << 3) +#define RT5670_JD_LO_EN (0x1 << 3) +#define RT5670_JD_LO_TRG_MASK (0x1 << 2) +#define RT5670_JD_LO_TRG_SFT 2 +#define RT5670_JD_LO_TRG_LO (0x0 << 2) +#define RT5670_JD_LO_TRG_HI (0x1 << 2) +#define RT5670_JD1_IN4P_MASK (0x1 << 1) +#define RT5670_JD1_IN4P_SFT 1 +#define RT5670_JD1_IN4P_DIS (0x0 << 1) +#define RT5670_JD1_IN4P_EN (0x1 << 1) +#define RT5670_JD2_IN4N_MASK (0x1) +#define RT5670_JD2_IN4N_SFT 0 +#define RT5670_JD2_IN4N_DIS (0x0) +#define RT5670_JD2_IN4N_EN (0x1) + +/* IRQ Control 1 (0xbd) */ +#define RT5670_IRQ_JD_MASK (0x1 << 15) +#define RT5670_IRQ_JD_SFT 15 +#define RT5670_IRQ_JD_BP (0x0 << 15) +#define RT5670_IRQ_JD_NOR (0x1 << 15) +#define RT5670_IRQ_OT_MASK (0x1 << 14) +#define RT5670_IRQ_OT_SFT 14 +#define RT5670_IRQ_OT_BP (0x0 << 14) +#define RT5670_IRQ_OT_NOR (0x1 << 14) +#define RT5670_JD_STKY_MASK (0x1 << 13) +#define RT5670_JD_STKY_SFT 13 +#define RT5670_JD_STKY_DIS (0x0 << 13) +#define RT5670_JD_STKY_EN (0x1 << 13) +#define RT5670_OT_STKY_MASK (0x1 << 12) +#define RT5670_OT_STKY_SFT 12 +#define RT5670_OT_STKY_DIS (0x0 << 12) +#define RT5670_OT_STKY_EN (0x1 << 12) +#define RT5670_JD_P_MASK (0x1 << 11) +#define RT5670_JD_P_SFT 11 +#define RT5670_JD_P_NOR (0x0 << 11) +#define RT5670_JD_P_INV (0x1 << 11) +#define RT5670_OT_P_MASK (0x1 << 10) +#define RT5670_OT_P_SFT 10 +#define RT5670_OT_P_NOR (0x0 << 10) +#define RT5670_OT_P_INV (0x1 << 10) +#define RT5670_JD1_1_EN_MASK (0x1 << 9) +#define RT5670_JD1_1_EN_SFT 9 +#define RT5670_JD1_1_DIS (0x0 << 9) +#define RT5670_JD1_1_EN (0x1 << 9) + +/* IRQ Control 2 (0xbe) */ +#define RT5670_IRQ_MB1_OC_MASK (0x1 << 15) +#define RT5670_IRQ_MB1_OC_SFT 15 +#define RT5670_IRQ_MB1_OC_BP (0x0 << 15) +#define RT5670_IRQ_MB1_OC_NOR (0x1 << 15) +#define RT5670_IRQ_MB2_OC_MASK (0x1 << 14) +#define RT5670_IRQ_MB2_OC_SFT 14 +#define RT5670_IRQ_MB2_OC_BP (0x0 << 14) +#define RT5670_IRQ_MB2_OC_NOR (0x1 << 14) +#define RT5670_MB1_OC_STKY_MASK (0x1 << 11) +#define RT5670_MB1_OC_STKY_SFT 11 +#define RT5670_MB1_OC_STKY_DIS (0x0 << 11) +#define RT5670_MB1_OC_STKY_EN (0x1 << 11) +#define RT5670_MB2_OC_STKY_MASK (0x1 << 10) +#define RT5670_MB2_OC_STKY_SFT 10 +#define RT5670_MB2_OC_STKY_DIS (0x0 << 10) +#define RT5670_MB2_OC_STKY_EN (0x1 << 10) +#define RT5670_MB1_OC_P_MASK (0x1 << 7) +#define RT5670_MB1_OC_P_SFT 7 +#define RT5670_MB1_OC_P_NOR (0x0 << 7) +#define RT5670_MB1_OC_P_INV (0x1 << 7) +#define RT5670_MB2_OC_P_MASK (0x1 << 6) +#define RT5670_MB2_OC_P_SFT 6 +#define RT5670_MB2_OC_P_NOR (0x0 << 6) +#define RT5670_MB2_OC_P_INV (0x1 << 6) +#define RT5670_MB1_OC_CLR (0x1 << 3) +#define RT5670_MB1_OC_CLR_SFT 3 +#define RT5670_MB2_OC_CLR (0x1 << 2) +#define RT5670_MB2_OC_CLR_SFT 2 + +/* GPIO Control 1 (0xc0) */ +#define RT5670_GP1_PIN_MASK (0x1 << 15) +#define RT5670_GP1_PIN_SFT 15 +#define RT5670_GP1_PIN_GPIO1 (0x0 << 15) +#define RT5670_GP1_PIN_IRQ (0x1 << 15) +#define RT5670_GP2_PIN_MASK (0x1 << 14) +#define RT5670_GP2_PIN_SFT 14 +#define RT5670_GP2_PIN_GPIO2 (0x0 << 14) +#define RT5670_GP2_PIN_DMIC1_SCL (0x1 << 14) +#define RT5670_GP3_PIN_MASK (0x3 << 12) +#define RT5670_GP3_PIN_SFT 12 +#define RT5670_GP3_PIN_GPIO3 (0x0 << 12) +#define RT5670_GP3_PIN_DMIC1_SDA (0x1 << 12) +#define RT5670_GP3_PIN_IRQ (0x2 << 12) +#define RT5670_GP4_PIN_MASK (0x1 << 11) +#define RT5670_GP4_PIN_SFT 11 +#define RT5670_GP4_PIN_GPIO4 (0x0 << 11) +#define RT5670_GP4_PIN_DMIC2_SDA (0x1 << 11) +#define RT5670_DP_SIG_MASK (0x1 << 10) +#define RT5670_DP_SIG_SFT 10 +#define RT5670_DP_SIG_TEST (0x0 << 10) +#define RT5670_DP_SIG_AP (0x1 << 10) +#define RT5670_GPIO_M_MASK (0x1 << 9) +#define RT5670_GPIO_M_SFT 9 +#define RT5670_GPIO_M_FLT (0x0 << 9) +#define RT5670_GPIO_M_PH (0x1 << 9) +#define RT5670_I2S2_PIN_MASK (0x1 << 8) +#define RT5670_I2S2_PIN_SFT 8 +#define RT5670_I2S2_PIN_I2S (0x0 << 8) +#define RT5670_I2S2_PIN_GPIO (0x1 << 8) +#define RT5670_GP5_PIN_MASK (0x1 << 7) +#define RT5670_GP5_PIN_SFT 7 +#define RT5670_GP5_PIN_GPIO5 (0x0 << 7) +#define RT5670_GP5_PIN_DMIC3_SDA (0x1 << 7) +#define RT5670_GP6_PIN_MASK (0x1 << 6) +#define RT5670_GP6_PIN_SFT 6 +#define RT5670_GP6_PIN_GPIO6 (0x0 << 6) +#define RT5670_GP6_PIN_DMIC1_SDA (0x1 << 6) +#define RT5670_GP7_PIN_MASK (0x3 << 4) +#define RT5670_GP7_PIN_SFT 4 +#define RT5670_GP7_PIN_GPIO7 (0x0 << 4) +#define RT5670_GP7_PIN_DMIC1_SDA (0x1 << 4) +#define RT5670_GP7_PIN_PDM_SCL2 (0x2 << 4) +#define RT5670_GP8_PIN_MASK (0x1 << 3) +#define RT5670_GP8_PIN_SFT 3 +#define RT5670_GP8_PIN_GPIO8 (0x0 << 3) +#define RT5670_GP8_PIN_DMIC2_SDA (0x1 << 3) +#define RT5670_GP9_PIN_MASK (0x1 << 2) +#define RT5670_GP9_PIN_SFT 2 +#define RT5670_GP9_PIN_GPIO9 (0x0 << 2) +#define RT5670_GP9_PIN_DMIC3_SDA (0x1 << 2) +#define RT5670_GP10_PIN_MASK (0x3) +#define RT5670_GP10_PIN_SFT 0 +#define RT5670_GP10_PIN_GPIO9 (0x0) +#define RT5670_GP10_PIN_DMIC3_SDA (0x1) +#define RT5670_GP10_PIN_PDM_ADT2 (0x2) + +/* GPIO Control 2 (0xc1) */ +#define RT5670_GP4_PF_MASK (0x1 << 11) +#define RT5670_GP4_PF_SFT 11 +#define RT5670_GP4_PF_IN (0x0 << 11) +#define RT5670_GP4_PF_OUT (0x1 << 11) +#define RT5670_GP4_OUT_MASK (0x1 << 10) +#define RT5670_GP4_OUT_SFT 10 +#define RT5670_GP4_OUT_LO (0x0 << 10) +#define RT5670_GP4_OUT_HI (0x1 << 10) +#define RT5670_GP4_P_MASK (0x1 << 9) +#define RT5670_GP4_P_SFT 9 +#define RT5670_GP4_P_NOR (0x0 << 9) +#define RT5670_GP4_P_INV (0x1 << 9) +#define RT5670_GP3_PF_MASK (0x1 << 8) +#define RT5670_GP3_PF_SFT 8 +#define RT5670_GP3_PF_IN (0x0 << 8) +#define RT5670_GP3_PF_OUT (0x1 << 8) +#define RT5670_GP3_OUT_MASK (0x1 << 7) +#define RT5670_GP3_OUT_SFT 7 +#define RT5670_GP3_OUT_LO (0x0 << 7) +#define RT5670_GP3_OUT_HI (0x1 << 7) +#define RT5670_GP3_P_MASK (0x1 << 6) +#define RT5670_GP3_P_SFT 6 +#define RT5670_GP3_P_NOR (0x0 << 6) +#define RT5670_GP3_P_INV (0x1 << 6) +#define RT5670_GP2_PF_MASK (0x1 << 5) +#define RT5670_GP2_PF_SFT 5 +#define RT5670_GP2_PF_IN (0x0 << 5) +#define RT5670_GP2_PF_OUT (0x1 << 5) +#define RT5670_GP2_OUT_MASK (0x1 << 4) +#define RT5670_GP2_OUT_SFT 4 +#define RT5670_GP2_OUT_LO (0x0 << 4) +#define RT5670_GP2_OUT_HI (0x1 << 4) +#define RT5670_GP2_P_MASK (0x1 << 3) +#define RT5670_GP2_P_SFT 3 +#define RT5670_GP2_P_NOR (0x0 << 3) +#define RT5670_GP2_P_INV (0x1 << 3) +#define RT5670_GP1_PF_MASK (0x1 << 2) +#define RT5670_GP1_PF_SFT 2 +#define RT5670_GP1_PF_IN (0x0 << 2) +#define RT5670_GP1_PF_OUT (0x1 << 2) +#define RT5670_GP1_OUT_MASK (0x1 << 1) +#define RT5670_GP1_OUT_SFT 1 +#define RT5670_GP1_OUT_LO (0x0 << 1) +#define RT5670_GP1_OUT_HI (0x1 << 1) +#define RT5670_GP1_P_MASK (0x1) +#define RT5670_GP1_P_SFT 0 +#define RT5670_GP1_P_NOR (0x0) +#define RT5670_GP1_P_INV (0x1) + +/* Scramble Function (0xcd) */ +#define RT5670_SCB_KEY_MASK (0xff) +#define RT5670_SCB_KEY_SFT 0 + +/* Scramble Control (0xce) */ +#define RT5670_SCB_SWAP_MASK (0x1 << 15) +#define RT5670_SCB_SWAP_SFT 15 +#define RT5670_SCB_SWAP_DIS (0x0 << 15) +#define RT5670_SCB_SWAP_EN (0x1 << 15) +#define RT5670_SCB_MASK (0x1 << 14) +#define RT5670_SCB_SFT 14 +#define RT5670_SCB_DIS (0x0 << 14) +#define RT5670_SCB_EN (0x1 << 14) + +/* Baseback Control (0xcf) */ +#define RT5670_BB_MASK (0x1 << 15) +#define RT5670_BB_SFT 15 +#define RT5670_BB_DIS (0x0 << 15) +#define RT5670_BB_EN (0x1 << 15) +#define RT5670_BB_CT_MASK (0x7 << 12) +#define RT5670_BB_CT_SFT 12 +#define RT5670_BB_CT_A (0x0 << 12) +#define RT5670_BB_CT_B (0x1 << 12) +#define RT5670_BB_CT_C (0x2 << 12) +#define RT5670_BB_CT_D (0x3 << 12) +#define RT5670_M_BB_L_MASK (0x1 << 9) +#define RT5670_M_BB_L_SFT 9 +#define RT5670_M_BB_R_MASK (0x1 << 8) +#define RT5670_M_BB_R_SFT 8 +#define RT5670_M_BB_HPF_L_MASK (0x1 << 7) +#define RT5670_M_BB_HPF_L_SFT 7 +#define RT5670_M_BB_HPF_R_MASK (0x1 << 6) +#define RT5670_M_BB_HPF_R_SFT 6 +#define RT5670_G_BB_BST_MASK (0x3f) +#define RT5670_G_BB_BST_SFT 0 + +/* MP3 Plus Control 1 (0xd0) */ +#define RT5670_M_MP3_L_MASK (0x1 << 15) +#define RT5670_M_MP3_L_SFT 15 +#define RT5670_M_MP3_R_MASK (0x1 << 14) +#define RT5670_M_MP3_R_SFT 14 +#define RT5670_M_MP3_MASK (0x1 << 13) +#define RT5670_M_MP3_SFT 13 +#define RT5670_M_MP3_DIS (0x0 << 13) +#define RT5670_M_MP3_EN (0x1 << 13) +#define RT5670_EG_MP3_MASK (0x1f << 8) +#define RT5670_EG_MP3_SFT 8 +#define RT5670_MP3_HLP_MASK (0x1 << 7) +#define RT5670_MP3_HLP_SFT 7 +#define RT5670_MP3_HLP_DIS (0x0 << 7) +#define RT5670_MP3_HLP_EN (0x1 << 7) +#define RT5670_M_MP3_ORG_L_MASK (0x1 << 6) +#define RT5670_M_MP3_ORG_L_SFT 6 +#define RT5670_M_MP3_ORG_R_MASK (0x1 << 5) +#define RT5670_M_MP3_ORG_R_SFT 5 + +/* MP3 Plus Control 2 (0xd1) */ +#define RT5670_MP3_WT_MASK (0x1 << 13) +#define RT5670_MP3_WT_SFT 13 +#define RT5670_MP3_WT_1_4 (0x0 << 13) +#define RT5670_MP3_WT_1_2 (0x1 << 13) +#define RT5670_OG_MP3_MASK (0x1f << 8) +#define RT5670_OG_MP3_SFT 8 +#define RT5670_HG_MP3_MASK (0x3f) +#define RT5670_HG_MP3_SFT 0 + +/* 3D HP Control 1 (0xd2) */ +#define RT5670_3D_CF_MASK (0x1 << 15) +#define RT5670_3D_CF_SFT 15 +#define RT5670_3D_CF_DIS (0x0 << 15) +#define RT5670_3D_CF_EN (0x1 << 15) +#define RT5670_3D_HP_MASK (0x1 << 14) +#define RT5670_3D_HP_SFT 14 +#define RT5670_3D_HP_DIS (0x0 << 14) +#define RT5670_3D_HP_EN (0x1 << 14) +#define RT5670_3D_BT_MASK (0x1 << 13) +#define RT5670_3D_BT_SFT 13 +#define RT5670_3D_BT_DIS (0x0 << 13) +#define RT5670_3D_BT_EN (0x1 << 13) +#define RT5670_3D_1F_MIX_MASK (0x3 << 11) +#define RT5670_3D_1F_MIX_SFT 11 +#define RT5670_3D_HP_M_MASK (0x1 << 10) +#define RT5670_3D_HP_M_SFT 10 +#define RT5670_3D_HP_M_SUR (0x0 << 10) +#define RT5670_3D_HP_M_FRO (0x1 << 10) +#define RT5670_M_3D_HRTF_MASK (0x1 << 9) +#define RT5670_M_3D_HRTF_SFT 9 +#define RT5670_M_3D_D2H_MASK (0x1 << 8) +#define RT5670_M_3D_D2H_SFT 8 +#define RT5670_M_3D_D2R_MASK (0x1 << 7) +#define RT5670_M_3D_D2R_SFT 7 +#define RT5670_M_3D_REVB_MASK (0x1 << 6) +#define RT5670_M_3D_REVB_SFT 6 + +/* Adjustable high pass filter control 1 (0xd3) */ +#define RT5670_2ND_HPF_MASK (0x1 << 15) +#define RT5670_2ND_HPF_SFT 15 +#define RT5670_2ND_HPF_DIS (0x0 << 15) +#define RT5670_2ND_HPF_EN (0x1 << 15) +#define RT5670_HPF_CF_L_MASK (0x7 << 12) +#define RT5670_HPF_CF_L_SFT 12 +#define RT5670_1ST_HPF_MASK (0x1 << 11) +#define RT5670_1ST_HPF_SFT 11 +#define RT5670_1ST_HPF_DIS (0x0 << 11) +#define RT5670_1ST_HPF_EN (0x1 << 11) +#define RT5670_HPF_CF_R_MASK (0x7 << 8) +#define RT5670_HPF_CF_R_SFT 8 +#define RT5670_ZD_T_MASK (0x3 << 6) +#define RT5670_ZD_T_SFT 6 +#define RT5670_ZD_F_MASK (0x3 << 4) +#define RT5670_ZD_F_SFT 4 +#define RT5670_ZD_F_IM (0x0 << 4) +#define RT5670_ZD_F_ZC_IM (0x1 << 4) +#define RT5670_ZD_F_ZC_IOD (0x2 << 4) +#define RT5670_ZD_F_UN (0x3 << 4) + +/* HP calibration control and Amp detection (0xd6) */ +#define RT5670_SI_DAC_MASK (0x1 << 11) +#define RT5670_SI_DAC_SFT 11 +#define RT5670_SI_DAC_AUTO (0x0 << 11) +#define RT5670_SI_DAC_TEST (0x1 << 11) +#define RT5670_DC_CAL_M_MASK (0x1 << 10) +#define RT5670_DC_CAL_M_SFT 10 +#define RT5670_DC_CAL_M_CAL (0x0 << 10) +#define RT5670_DC_CAL_M_NOR (0x1 << 10) +#define RT5670_DC_CAL_MASK (0x1 << 9) +#define RT5670_DC_CAL_SFT 9 +#define RT5670_DC_CAL_DIS (0x0 << 9) +#define RT5670_DC_CAL_EN (0x1 << 9) +#define RT5670_HPD_RCV_MASK (0x7 << 6) +#define RT5670_HPD_RCV_SFT 6 +#define RT5670_HPD_PS_MASK (0x1 << 5) +#define RT5670_HPD_PS_SFT 5 +#define RT5670_HPD_PS_DIS (0x0 << 5) +#define RT5670_HPD_PS_EN (0x1 << 5) +#define RT5670_CAL_M_MASK (0x1 << 4) +#define RT5670_CAL_M_SFT 4 +#define RT5670_CAL_M_DEP (0x0 << 4) +#define RT5670_CAL_M_CAL (0x1 << 4) +#define RT5670_CAL_MASK (0x1 << 3) +#define RT5670_CAL_SFT 3 +#define RT5670_CAL_DIS (0x0 << 3) +#define RT5670_CAL_EN (0x1 << 3) +#define RT5670_CAL_TEST_MASK (0x1 << 2) +#define RT5670_CAL_TEST_SFT 2 +#define RT5670_CAL_TEST_DIS (0x0 << 2) +#define RT5670_CAL_TEST_EN (0x1 << 2) +#define RT5670_CAL_P_MASK (0x3) +#define RT5670_CAL_P_SFT 0 +#define RT5670_CAL_P_NONE (0x0) +#define RT5670_CAL_P_CAL (0x1) +#define RT5670_CAL_P_DAC_CAL (0x2) + +/* Soft volume and zero cross control 1 (0xd9) */ +#define RT5670_SV_MASK (0x1 << 15) +#define RT5670_SV_SFT 15 +#define RT5670_SV_DIS (0x0 << 15) +#define RT5670_SV_EN (0x1 << 15) +#define RT5670_SPO_SV_MASK (0x1 << 14) +#define RT5670_SPO_SV_SFT 14 +#define RT5670_SPO_SV_DIS (0x0 << 14) +#define RT5670_SPO_SV_EN (0x1 << 14) +#define RT5670_OUT_SV_MASK (0x1 << 13) +#define RT5670_OUT_SV_SFT 13 +#define RT5670_OUT_SV_DIS (0x0 << 13) +#define RT5670_OUT_SV_EN (0x1 << 13) +#define RT5670_HP_SV_MASK (0x1 << 12) +#define RT5670_HP_SV_SFT 12 +#define RT5670_HP_SV_DIS (0x0 << 12) +#define RT5670_HP_SV_EN (0x1 << 12) +#define RT5670_ZCD_DIG_MASK (0x1 << 11) +#define RT5670_ZCD_DIG_SFT 11 +#define RT5670_ZCD_DIG_DIS (0x0 << 11) +#define RT5670_ZCD_DIG_EN (0x1 << 11) +#define RT5670_ZCD_MASK (0x1 << 10) +#define RT5670_ZCD_SFT 10 +#define RT5670_ZCD_PD (0x0 << 10) +#define RT5670_ZCD_PU (0x1 << 10) +#define RT5670_M_ZCD_MASK (0x3f << 4) +#define RT5670_M_ZCD_SFT 4 +#define RT5670_M_ZCD_RM_L (0x1 << 9) +#define RT5670_M_ZCD_RM_R (0x1 << 8) +#define RT5670_M_ZCD_SM_L (0x1 << 7) +#define RT5670_M_ZCD_SM_R (0x1 << 6) +#define RT5670_M_ZCD_OM_L (0x1 << 5) +#define RT5670_M_ZCD_OM_R (0x1 << 4) +#define RT5670_SV_DLY_MASK (0xf) +#define RT5670_SV_DLY_SFT 0 + +/* Soft volume and zero cross control 2 (0xda) */ +#define RT5670_ZCD_HP_MASK (0x1 << 15) +#define RT5670_ZCD_HP_SFT 15 +#define RT5670_ZCD_HP_DIS (0x0 << 15) +#define RT5670_ZCD_HP_EN (0x1 << 15) + + +/* Codec Private Register definition */ +/* 3D Speaker Control (0x63) */ +#define RT5670_3D_SPK_MASK (0x1 << 15) +#define RT5670_3D_SPK_SFT 15 +#define RT5670_3D_SPK_DIS (0x0 << 15) +#define RT5670_3D_SPK_EN (0x1 << 15) +#define RT5670_3D_SPK_M_MASK (0x3 << 13) +#define RT5670_3D_SPK_M_SFT 13 +#define RT5670_3D_SPK_CG_MASK (0x1f << 8) +#define RT5670_3D_SPK_CG_SFT 8 +#define RT5670_3D_SPK_SG_MASK (0x1f) +#define RT5670_3D_SPK_SG_SFT 0 + +/* Wind Noise Detection Control 1 (0x6c) */ +#define RT5670_WND_MASK (0x1 << 15) +#define RT5670_WND_SFT 15 +#define RT5670_WND_DIS (0x0 << 15) +#define RT5670_WND_EN (0x1 << 15) + +/* Wind Noise Detection Control 2 (0x6d) */ +#define RT5670_WND_FC_NW_MASK (0x3f << 10) +#define RT5670_WND_FC_NW_SFT 10 +#define RT5670_WND_FC_WK_MASK (0x3f << 4) +#define RT5670_WND_FC_WK_SFT 4 + +/* Wind Noise Detection Control 3 (0x6e) */ +#define RT5670_HPF_FC_MASK (0x3f << 6) +#define RT5670_HPF_FC_SFT 6 +#define RT5670_WND_FC_ST_MASK (0x3f) +#define RT5670_WND_FC_ST_SFT 0 + +/* Wind Noise Detection Control 4 (0x6f) */ +#define RT5670_WND_TH_LO_MASK (0x3ff) +#define RT5670_WND_TH_LO_SFT 0 + +/* Wind Noise Detection Control 5 (0x70) */ +#define RT5670_WND_TH_HI_MASK (0x3ff) +#define RT5670_WND_TH_HI_SFT 0 + +/* Wind Noise Detection Control 8 (0x73) */ +#define RT5670_WND_WIND_MASK (0x1 << 13) /* Read-Only */ +#define RT5670_WND_WIND_SFT 13 +#define RT5670_WND_STRONG_MASK (0x1 << 12) /* Read-Only */ +#define RT5670_WND_STRONG_SFT 12 +enum { + RT5670_NO_WIND, + RT5670_BREEZE, + RT5670_STORM, +}; + +/* Dipole Speaker Interface (0x75) */ +#define RT5670_DP_ATT_MASK (0x3 << 14) +#define RT5670_DP_ATT_SFT 14 +#define RT5670_DP_SPK_MASK (0x1 << 10) +#define RT5670_DP_SPK_SFT 10 +#define RT5670_DP_SPK_DIS (0x0 << 10) +#define RT5670_DP_SPK_EN (0x1 << 10) + +/* EQ Pre Volume Control (0xb3) */ +#define RT5670_EQ_PRE_VOL_MASK (0xffff) +#define RT5670_EQ_PRE_VOL_SFT 0 + +/* EQ Post Volume Control (0xb4) */ +#define RT5670_EQ_PST_VOL_MASK (0xffff) +#define RT5670_EQ_PST_VOL_SFT 0 + +/* Jack Detect Control 3 (0xf8) */ +#define RT5670_CMP_MIC_IN_DET_MASK (0x7 << 12) +#define RT5670_JD_CBJ_EN (0x1 << 7) +#define RT5670_JD_CBJ_POL (0x1 << 6) +#define RT5670_JD_TRI_CBJ_SEL_MASK (0x7 << 3) +#define RT5670_JD_TRI_CBJ_SEL_SFT (3) +#define RT5670_JD_CBJ_GPIO_JD1 (0x0 << 3) +#define RT5670_JD_CBJ_JD1_1 (0x1 << 3) +#define RT5670_JD_CBJ_JD1_2 (0x2 << 3) +#define RT5670_JD_CBJ_JD2 (0x3 << 3) +#define RT5670_JD_CBJ_JD3 (0x4 << 3) +#define RT5670_JD_CBJ_GPIO_JD2 (0x5 << 3) +#define RT5670_JD_CBJ_MX0B_12 (0x6 << 3) +#define RT5670_JD_TRI_HPO_SEL_MASK (0x7 << 3) +#define RT5670_JD_TRI_HPO_SEL_SFT (0) +#define RT5670_JD_HPO_GPIO_JD1 (0x0) +#define RT5670_JD_HPO_JD1_1 (0x1) +#define RT5670_JD_HPO_JD1_2 (0x2) +#define RT5670_JD_HPO_JD2 (0x3) +#define RT5670_JD_HPO_JD3 (0x4) +#define RT5670_JD_HPO_GPIO_JD2 (0x5) +#define RT5670_JD_HPO_MX0B_12 (0x6) + +/* Digital Misc Control (0xfa) */ +#define RT5670_RST_DSP (0x1 << 13) +#define RT5670_IF1_ADC1_IN1_SEL (0x1 << 12) +#define RT5670_IF1_ADC1_IN1_SFT 12 +#define RT5670_IF1_ADC1_IN2_SEL (0x1 << 11) +#define RT5670_IF1_ADC1_IN2_SFT 11 +#define RT5670_IF1_ADC2_IN1_SEL (0x1 << 10) +#define RT5670_IF1_ADC2_IN1_SFT 10 + +/* General Control2 (0xfb) */ +#define RT5670_RXDC_SRC_MASK (0x1 << 7) +#define RT5670_RXDC_SRC_STO (0x0 << 7) +#define RT5670_RXDC_SRC_MONO (0x1 << 7) +#define RT5670_RXDC_SRC_SFT (7) +#define RT5670_RXDP2_SEL_MASK (0x1 << 3) +#define RT5670_RXDP2_SEL_IF2 (0x0 << 3) +#define RT5670_RXDP2_SEL_ADC (0x1 << 3) +#define RT5670_RXDP2_SEL_SFT (3) + +/* System Clock Source */ +enum { + RT5670_SCLK_S_MCLK, + RT5670_SCLK_S_PLL1, + RT5670_SCLK_S_RCCLK, +}; + +/* PLL1 Source */ +enum { + RT5670_PLL1_S_MCLK, + RT5670_PLL1_S_BCLK1, + RT5670_PLL1_S_BCLK2, + RT5670_PLL1_S_BCLK3, + RT5670_PLL1_S_BCLK4, +}; + +enum { + RT5670_AIF1, + RT5670_AIF2, + RT5670_AIF3, + RT5670_AIF4, + RT5670_AIFS, +}; + +enum { + RT5670_DMIC_DATA_GPIO6, + RT5670_DMIC_DATA_IN2P, + RT5670_DMIC_DATA_GPIO7, +}; + +enum { + RT5670_DMIC_DATA_GPIO8, + RT5670_DMIC_DATA_IN3N, +}; + +enum { + RT5670_DMIC_DATA_GPIO9, + RT5670_DMIC_DATA_GPIO10, + RT5670_DMIC_DATA_GPIO5, +}; + +struct rt5670_priv { + struct snd_soc_codec *codec; + struct rt5670_platform_data pdata; + struct regmap *regmap; + + int sysclk; + int sysclk_src; + int lrck[RT5670_AIFS]; + int bclk[RT5670_AIFS]; + int master[RT5670_AIFS]; + + int pll_src; + int pll_in; + int pll_out; + + int dsp_sw; /* expected parameter setting */ + int dsp_rate; + int jack_type; +}; + +#endif /* __RT5670_H__ */ -- cgit v1.2.3 From b8257be5fff447681af8c8b72f1380e8a9c04286 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 1 Jul 2014 22:13:45 +0200 Subject: ASoC: core: Remove duplicated rtd->codec initialization rtd->codec is already initialized in soc_bind_dai_link(), so there is no need to do it again in soc_dai_link_init(). Removing the rtd->codec initialization from soc_dai_link_init() also removes the need for soc_dai_link_init() to know about the CODEC at all. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 72d4a2b009fa..0cd36b781f4d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1308,9 +1308,7 @@ static int soc_aux_dev_init(struct snd_soc_card *card, return 0; } -static int soc_dai_link_init(struct snd_soc_card *card, - struct snd_soc_codec *codec, - int num) +static int soc_dai_link_init(struct snd_soc_card *card, int num) { struct snd_soc_dai_link *dai_link = &card->dai_link[num]; struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; @@ -1325,8 +1323,6 @@ static int soc_dai_link_init(struct snd_soc_card *card, return ret; } - rtd->codec = codec; - return 0; } @@ -1344,7 +1340,7 @@ static int soc_post_component_init(struct snd_soc_card *card, dai_link = &card->dai_link[num]; rtd = &card->rtd[num]; name = dai_link->name; - ret = soc_dai_link_init(card, codec, num); + ret = soc_dai_link_init(card, num); } else { aux_dev = &card->aux_dev[num]; rtd = &card->rtd_aux[num]; -- cgit v1.2.3 From 48f466d112c9ca735ee765d8f5148d07a8c7bb1d Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 1 Jul 2014 22:13:46 +0200 Subject: ASoC: core: Replace soc_find_matching_codec() with soc_find_codec() soc_find_matching_codec() works in the same way as soc_find_codec() except that it only works for auxdevs. It can easily be replaced by the generic soc_find_codec(). Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 26 ++++---------------------- 1 file changed, 4 insertions(+), 22 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0cd36b781f4d..c8bdac2db377 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1649,32 +1649,13 @@ static void soc_unregister_ac97_dai_link(struct snd_soc_pcm_runtime *rtd) } #endif -static struct snd_soc_codec *soc_find_matching_codec(struct snd_soc_card *card, - int num) -{ - struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; - struct snd_soc_codec *codec; - - /* find CODEC from registered CODECs */ - list_for_each_entry(codec, &codec_list, list) { - if (aux_dev->codec_of_node && - (codec->dev->of_node != aux_dev->codec_of_node)) - continue; - if (aux_dev->codec_name && - strcmp(codec->component.name, aux_dev->codec_name)) - continue; - return codec; - } - - return NULL; -} - static int soc_check_aux_dev(struct snd_soc_card *card, int num) { struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; const char *codecname = aux_dev->codec_name; - struct snd_soc_codec *codec = soc_find_matching_codec(card, num); + struct snd_soc_codec *codec; + codec = soc_find_codec(aux_dev->codec_of_node, aux_dev->codec_name); if (codec) return 0; if (aux_dev->codec_of_node) @@ -1689,8 +1670,9 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num) struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; const char *codecname = aux_dev->codec_name; int ret = -ENODEV; - struct snd_soc_codec *codec = soc_find_matching_codec(card, num); + struct snd_soc_codec *codec; + codec = soc_find_codec(aux_dev->codec_of_node, aux_dev->codec_name); if (!codec) { if (aux_dev->codec_of_node) codecname = of_node_full_name(aux_dev->codec_of_node); -- cgit v1.2.3 From 44c69bb13905c3b2281a920c0b44059e88819993 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 1 Jul 2014 22:13:47 +0200 Subject: ASoC: core: Bind aux devs early Currently in snd_soc_instantiate_card() we only check if the aux dev exists, but do not yet assign it to its rtd. This means that we need to lookup the aux dev again in soc_probe_aux_dev(). This patch changes the behavior to assign the aux dev to the rtd in soc_check_aux_dev() (and renames it to soc_bind_aux_dev()). This simplifies the implementation a bit and also removes the need for soc_post_component_init() to know about the specific CODEC that was assigned to the rtd. The later is necessary for componentization as the code should work for all types of components not just CODECs. This new behavior is also more in sync with how soc_bind_dai_link()/soc_probe_link_dais() works. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 67 ++++++++++++++++++++-------------------------------- 1 file changed, 25 insertions(+), 42 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c8bdac2db377..b1600cdcdc14 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1286,9 +1286,7 @@ static void rtd_release(struct device *dev) kfree(dev); } -static int soc_aux_dev_init(struct snd_soc_card *card, - struct snd_soc_codec *codec, - int num) +static int soc_aux_dev_init(struct snd_soc_card *card, int num) { struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num]; @@ -1298,13 +1296,11 @@ static int soc_aux_dev_init(struct snd_soc_card *card, /* do machine specific initialization */ if (aux_dev->init) { - ret = aux_dev->init(&codec->dapm); + ret = aux_dev->init(&rtd->codec->dapm); if (ret < 0) return ret; } - rtd->codec = codec; - return 0; } @@ -1327,7 +1323,6 @@ static int soc_dai_link_init(struct snd_soc_card *card, int num) } static int soc_post_component_init(struct snd_soc_card *card, - struct snd_soc_codec *codec, int num, int dailess) { struct snd_soc_dai_link *dai_link = NULL; @@ -1345,7 +1340,7 @@ static int soc_post_component_init(struct snd_soc_card *card, aux_dev = &card->aux_dev[num]; rtd = &card->rtd_aux[num]; name = aux_dev->name; - ret = soc_aux_dev_init(card, codec, num); + ret = soc_aux_dev_init(card, num); } if (ret < 0) { @@ -1380,13 +1375,13 @@ static int soc_post_component_init(struct snd_soc_card *card, /* add DAPM sysfs entries for this codec */ ret = snd_soc_dapm_sys_add(rtd->dev); if (ret < 0) - dev_err(codec->dev, + dev_err(rtd->dev, "ASoC: failed to add codec dapm sysfs entries: %d\n", ret); /* add codec sysfs entries */ ret = device_create_file(rtd->dev, &dev_attr_codec_reg); if (ret < 0) - dev_err(codec->dev, + dev_err(rtd->dev, "ASoC: failed to add codec sysfs files: %d\n", ret); #ifdef CONFIG_DEBUG_FS @@ -1551,7 +1546,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) if (order != SND_SOC_COMP_ORDER_LAST) return 0; - ret = soc_post_component_init(card, codec, num, 0); + ret = soc_post_component_init(card, num, 0); if (ret) return ret; @@ -1649,51 +1644,39 @@ static void soc_unregister_ac97_dai_link(struct snd_soc_pcm_runtime *rtd) } #endif -static int soc_check_aux_dev(struct snd_soc_card *card, int num) -{ - struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; - const char *codecname = aux_dev->codec_name; - struct snd_soc_codec *codec; - - codec = soc_find_codec(aux_dev->codec_of_node, aux_dev->codec_name); - if (codec) - return 0; - if (aux_dev->codec_of_node) - codecname = of_node_full_name(aux_dev->codec_of_node); - - dev_err(card->dev, "ASoC: %s not registered\n", codecname); - return -EPROBE_DEFER; -} - -static int soc_probe_aux_dev(struct snd_soc_card *card, int num) +static int soc_bind_aux_dev(struct snd_soc_card *card, int num) { + struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num]; struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; const char *codecname = aux_dev->codec_name; - int ret = -ENODEV; - struct snd_soc_codec *codec; - codec = soc_find_codec(aux_dev->codec_of_node, aux_dev->codec_name); - if (!codec) { + rtd->codec = soc_find_codec(aux_dev->codec_of_node, codecname); + if (!rtd->codec) { if (aux_dev->codec_of_node) codecname = of_node_full_name(aux_dev->codec_of_node); - /* codec not found */ - dev_err(card->dev, "ASoC: codec %s not found", codecname); + dev_err(card->dev, "ASoC: %s not registered\n", codecname); return -EPROBE_DEFER; } - if (codec->probed) { - dev_err(codec->dev, "ASoC: codec already probed"); + return 0; +} + +static int soc_probe_aux_dev(struct snd_soc_card *card, int num) +{ + struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num]; + int ret; + + if (rtd->codec->probed) { + dev_err(rtd->codec->dev, "ASoC: codec already probed\n"); return -EBUSY; } - ret = soc_probe_codec(card, codec); + ret = soc_probe_codec(card, rtd->codec); if (ret < 0) return ret; - ret = soc_post_component_init(card, codec, num, 1); - - return ret; + return soc_post_component_init(card, num, 1); } static void soc_remove_aux_dev(struct snd_soc_card *card, int num) @@ -1745,9 +1728,9 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) goto base_error; } - /* check aux_devs too */ + /* bind aux_devs too */ for (i = 0; i < card->num_aux_devs; i++) { - ret = soc_check_aux_dev(card, i); + ret = soc_bind_aux_dev(card, i); if (ret != 0) goto base_error; } -- cgit v1.2.3 From 5f3484ac8775a183f855fd213907c4221d2fb11f Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 1 Jul 2014 22:13:48 +0200 Subject: ASoC: core: Move non-shared code paths out of snd_soc_post_component_init() There are two call sites for snd_soc_post_component_init(), one passes 0 and the other 1 for the 'dailess' parameter of snd_soc_post_component_init(). Depending on whether 'dailess' is 0 or 1 snd_soc_post_component_init() runs different code at the beginning and the end of the function. The patch moves this conditional code out of snd_soc_post_component_init() and into the call sites. This removes the need for snd_soc_post_component_init() to know whether it is called for a DAI link or a aux dev. Also do the initialization of rtd->card when the rtd struct is allocated. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 121 ++++++++++++++++++++------------------------------- 1 file changed, 46 insertions(+), 75 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b1600cdcdc14..c4db07f01d10 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1286,74 +1286,17 @@ static void rtd_release(struct device *dev) kfree(dev); } -static int soc_aux_dev_init(struct snd_soc_card *card, int num) +static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd, + const char *name) { - struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; - struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num]; - int ret; - - rtd->card = card; - - /* do machine specific initialization */ - if (aux_dev->init) { - ret = aux_dev->init(&rtd->codec->dapm); - if (ret < 0) - return ret; - } - - return 0; -} - -static int soc_dai_link_init(struct snd_soc_card *card, int num) -{ - struct snd_soc_dai_link *dai_link = &card->dai_link[num]; - struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; - int ret; - - rtd->card = card; - - /* do machine specific initialization */ - if (dai_link->init) { - ret = dai_link->init(rtd); - if (ret < 0) - return ret; - } - - return 0; -} - -static int soc_post_component_init(struct snd_soc_card *card, - int num, int dailess) -{ - struct snd_soc_dai_link *dai_link = NULL; - struct snd_soc_aux_dev *aux_dev = NULL; - struct snd_soc_pcm_runtime *rtd; - const char *name; int ret = 0; - if (!dailess) { - dai_link = &card->dai_link[num]; - rtd = &card->rtd[num]; - name = dai_link->name; - ret = soc_dai_link_init(card, num); - } else { - aux_dev = &card->aux_dev[num]; - rtd = &card->rtd_aux[num]; - name = aux_dev->name; - ret = soc_aux_dev_init(card, num); - } - - if (ret < 0) { - dev_err(card->dev, "ASoC: failed to init %s: %d\n", name, ret); - return ret; - } - /* register the rtd device */ rtd->dev = kzalloc(sizeof(struct device), GFP_KERNEL); if (!rtd->dev) return -ENOMEM; device_initialize(rtd->dev); - rtd->dev->parent = card->dev; + rtd->dev->parent = rtd->card->dev; rtd->dev->release = rtd_release; rtd->dev->init_name = name; dev_set_drvdata(rtd->dev, rtd); @@ -1366,7 +1309,7 @@ static int soc_post_component_init(struct snd_soc_card *card, if (ret < 0) { /* calling put_device() here to free the rtd->dev */ put_device(rtd->dev); - dev_err(card->dev, + dev_err(rtd->card->dev, "ASoC: failed to register runtime device: %d\n", ret); return ret; } @@ -1384,17 +1327,6 @@ static int soc_post_component_init(struct snd_soc_card *card, dev_err(rtd->dev, "ASoC: failed to add codec sysfs files: %d\n", ret); -#ifdef CONFIG_DEBUG_FS - /* add DPCM sysfs entries */ - if (!dailess && !dai_link->dynamic) - goto out; - - ret = soc_dpcm_debugfs_add(rtd); - if (ret < 0) - dev_err(rtd->dev, "ASoC: failed to add dpcm sysfs entries: %d\n", ret); - -out: -#endif return 0; } @@ -1546,10 +1478,33 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) if (order != SND_SOC_COMP_ORDER_LAST) return 0; - ret = soc_post_component_init(card, num, 0); + /* do machine specific initialization */ + if (dai_link->init) { + ret = dai_link->init(rtd); + if (ret < 0) { + dev_err(card->dev, "ASoC: failed to init %s: %d\n", + dai_link->name, ret); + return ret; + } + } + + ret = soc_post_component_init(rtd, dai_link->name); if (ret) return ret; +#ifdef CONFIG_DEBUG_FS + /* add DPCM sysfs entries */ + if (dai_link->dynamic) { + ret = soc_dpcm_debugfs_add(rtd); + if (ret < 0) { + dev_err(rtd->dev, + "ASoC: failed to add dpcm sysfs entries: %d\n", + ret); + return ret; + } + } +#endif + ret = device_create_file(rtd->dev, &dev_attr_pmdown_time); if (ret < 0) dev_warn(rtd->dev, "ASoC: failed to add pmdown_time sysfs: %d\n", @@ -1665,6 +1620,7 @@ static int soc_bind_aux_dev(struct snd_soc_card *card, int num) static int soc_probe_aux_dev(struct snd_soc_card *card, int num) { struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num]; + struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; int ret; if (rtd->codec->probed) { @@ -1676,7 +1632,17 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num) if (ret < 0) return ret; - return soc_post_component_init(card, num, 1); + /* do machine specific initialization */ + if (aux_dev->init) { + ret = aux_dev->init(&rtd->codec->dapm); + if (ret < 0) { + dev_err(card->dev, "ASoC: failed to init %s: %d\n", + aux_dev->name, ret); + return ret; + } + } + + return soc_post_component_init(rtd, aux_dev->name); } static void soc_remove_aux_dev(struct snd_soc_card *card, int num) @@ -3775,8 +3741,13 @@ int snd_soc_register_card(struct snd_soc_card *card) card->num_rtd = 0; card->rtd_aux = &card->rtd[card->num_links]; - for (i = 0; i < card->num_links; i++) + for (i = 0; i < card->num_links; i++) { + card->rtd[i].card = card; card->rtd[i].dai_link = &card->dai_link[i]; + } + + for (i = 0; i < card->num_aux_devs; i++) + card->rtd_aux[i].card = card; INIT_LIST_HEAD(&card->dapm_dirty); card->instantiated = 0; -- cgit v1.2.3 From c46af3124b05ccb37d1311e442f191bb5c1d888e Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 3 Jul 2014 16:56:43 +0200 Subject: ASoC: tas5086: add regulator consumer support The TAS5086 has two power domains, DVDD and AVDD. Enable them both as long as the codec is in use. Also, switch on the power to identify the chip at device probe level, and switch it off again afterwards. The codec level will take care for power handling later. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/ti,tas5086.txt | 5 ++ sound/soc/codecs/tas5086.c | 67 +++++++++++++++++++--- 2 files changed, 63 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/ti,tas5086.txt b/Documentation/devicetree/bindings/sound/ti,tas5086.txt index d2866a0d6a26..234dad296da7 100644 --- a/Documentation/devicetree/bindings/sound/ti,tas5086.txt +++ b/Documentation/devicetree/bindings/sound/ti,tas5086.txt @@ -31,6 +31,9 @@ Optional properties: Most systems should not set any of these properties. + - avdd-supply: Power supply for AVDD, providing 3.3V + - dvdd-supply: Power supply for DVDD, providing 3.3V + Examples: i2c_bus { @@ -39,5 +42,7 @@ Examples: reg = <0x1b>; reset-gpio = <&gpio 23 0>; ti,charge-period = <156000>; + avdd-supply = <&vdd_3v3_reg>; + dvdd-supply = <&vdd_3v3_reg>; }; }; diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index d48491a4a19d..be7194b43b7a 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -36,6 +36,7 @@ #include #include #include +#include #include #include #include @@ -240,6 +241,10 @@ static int tas5086_reg_read(void *context, unsigned int reg, return 0; } +static const char * const supply_names[] = { + "dvdd", "avdd" +}; + struct tas5086_private { struct regmap *regmap; unsigned int mclk, sclk; @@ -251,6 +256,7 @@ struct tas5086_private { int rate; /* GPIO driving Reset pin, if any */ int gpio_nreset; + struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)]; }; static int tas5086_deemph[] = { 0, 32000, 44100, 48000 }; @@ -773,6 +779,8 @@ static int tas5086_soc_suspend(struct snd_soc_codec *codec) if (ret < 0) return ret; + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); + return 0; } @@ -781,6 +789,10 @@ static int tas5086_soc_resume(struct snd_soc_codec *codec) struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); int ret; + ret = regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies); + if (ret < 0) + return ret; + tas5086_reset(priv); regcache_mark_dirty(priv->regmap); @@ -812,6 +824,12 @@ static int tas5086_probe(struct snd_soc_codec *codec) struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); int i, ret; + ret = regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies); + if (ret < 0) { + dev_err(codec->dev, "Failed to enable regulators: %d\n", ret); + return ret; + } + priv->pwm_start_mid_z = 0; priv->charge_period = 1300000; /* hardware default is 1300 ms */ @@ -832,16 +850,22 @@ static int tas5086_probe(struct snd_soc_codec *codec) } } + tas5086_reset(priv); ret = tas5086_init(codec->dev, priv); if (ret < 0) - return ret; + goto exit_disable_regulators; /* set master volume to 0 dB */ ret = regmap_write(priv->regmap, TAS5086_MASTER_VOL, 0x30); if (ret < 0) - return ret; + goto exit_disable_regulators; return 0; + +exit_disable_regulators: + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); + + return ret; } static int tas5086_remove(struct snd_soc_codec *codec) @@ -852,6 +876,8 @@ static int tas5086_remove(struct snd_soc_codec *codec) /* Set codec to the reset state */ gpio_set_value(priv->gpio_nreset, 0); + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); + return 0; }; @@ -900,6 +926,16 @@ static int tas5086_i2c_probe(struct i2c_client *i2c, if (!priv) return -ENOMEM; + for (i = 0; i < ARRAY_SIZE(supply_names); i++) + priv->supplies[i].supply = supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(priv->supplies), + priv->supplies); + if (ret < 0) { + dev_err(dev, "Failed to get regulators: %d\n", ret); + return ret; + } + priv->regmap = devm_regmap_init(dev, NULL, i2c, &tas5086_regmap); if (IS_ERR(priv->regmap)) { ret = PTR_ERR(priv->regmap); @@ -919,21 +955,34 @@ static int tas5086_i2c_probe(struct i2c_client *i2c, gpio_nreset = -EINVAL; priv->gpio_nreset = gpio_nreset; + + ret = regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies); + if (ret < 0) { + dev_err(dev, "Failed to enable regulators: %d\n", ret); + return ret; + } + tas5086_reset(priv); /* The TAS5086 always returns 0x03 in its TAS5086_DEV_ID register */ ret = regmap_read(priv->regmap, TAS5086_DEV_ID, &i); - if (ret < 0) - return ret; - - if (i != 0x3) { + if (ret == 0 && i != 0x3) { dev_err(dev, "Failed to identify TAS5086 codec (got %02x)\n", i); - return -ENODEV; + ret = -ENODEV; } - return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_tas5086, - &tas5086_dai, 1); + /* + * The chip has been identified, so we can turn off the power + * again until the dai link is set up. + */ + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); + + if (ret == 0) + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_tas5086, + &tas5086_dai, 1); + + return ret; } static int tas5086_i2c_remove(struct i2c_client *i2c) -- cgit v1.2.3 From 2bd8d1d5cf8905c7652fa90a8ec952ba52f820ac Mon Sep 17 00:00:00 2001 From: Rongjun Ying Date: Wed, 2 Jul 2014 10:45:41 +0800 Subject: ASoC: sirf: Add audio usp interface driver This patch adds ASoC support for SiRF SoCs USP interface. Features include: 1. Only support slave mode. 2. Support I2S and DSP_A mode. 3. Support S16_LE, S24_LE and S24_3LE formats. 4. Support stereo and mono mode. 5. The biggest Support is 192Khz sample rate. Signed-off-by: Rongjun Ying Signed-off-by: Mark Brown --- sound/soc/sirf/Kconfig | 6 + sound/soc/sirf/Makefile | 2 + sound/soc/sirf/sirf-usp.c | 415 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sirf/sirf-usp.h | 293 ++++++++++++++++++++++++++++++++ 4 files changed, 716 insertions(+) create mode 100644 sound/soc/sirf/sirf-usp.c create mode 100644 sound/soc/sirf/sirf-usp.h (limited to 'sound') diff --git a/sound/soc/sirf/Kconfig b/sound/soc/sirf/Kconfig index 89e89429b04a..840058dcad09 100644 --- a/sound/soc/sirf/Kconfig +++ b/sound/soc/sirf/Kconfig @@ -12,3 +12,9 @@ config SND_SOC_SIRF_AUDIO config SND_SOC_SIRF_AUDIO_PORT select REGMAP_MMIO tristate + +config SND_SOC_SIRF_USP + tristate "SoC Audio (I2S protocol) for SiRF SoC USP interface" + depends on SND_SOC_SIRF + select REGMAP_MMIO + tristate diff --git a/sound/soc/sirf/Makefile b/sound/soc/sirf/Makefile index 913b93231d4e..dd917f20f12f 100644 --- a/sound/soc/sirf/Makefile +++ b/sound/soc/sirf/Makefile @@ -1,5 +1,7 @@ snd-soc-sirf-audio-objs := sirf-audio.o snd-soc-sirf-audio-port-objs := sirf-audio-port.o +snd-soc-sirf-usp-objs := sirf-usp.o obj-$(CONFIG_SND_SOC_SIRF_AUDIO) += snd-soc-sirf-audio.o obj-$(CONFIG_SND_SOC_SIRF_AUDIO_PORT) += snd-soc-sirf-audio-port.o +obj-$(CONFIG_SND_SOC_SIRF_USP) += snd-soc-sirf-usp.o diff --git a/sound/soc/sirf/sirf-usp.c b/sound/soc/sirf/sirf-usp.c new file mode 100644 index 000000000000..bdf6aae3e6d0 --- /dev/null +++ b/sound/soc/sirf/sirf-usp.c @@ -0,0 +1,415 @@ +/* + * SiRF USP in I2S/DSP mode + * + * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company. + * + * Licensed under GPLv2 or later. + */ +#include +#include +#include +#include +#include +#include +#include +#include + +#include "sirf-usp.h" + +struct sirf_usp { + struct regmap *regmap; + struct clk *clk; + u32 mode1_reg; + u32 mode2_reg; + int daifmt_format; + struct snd_dmaengine_dai_dma_data playback_dma_data; + struct snd_dmaengine_dai_dma_data capture_dma_data; +}; + +static void sirf_usp_tx_enable(struct sirf_usp *usp) +{ + regmap_update_bits(usp->regmap, USP_TX_FIFO_OP, + USP_TX_FIFO_RESET, USP_TX_FIFO_RESET); + regmap_write(usp->regmap, USP_TX_FIFO_OP, 0); + + regmap_update_bits(usp->regmap, USP_TX_FIFO_OP, + USP_TX_FIFO_START, USP_TX_FIFO_START); + + regmap_update_bits(usp->regmap, USP_TX_RX_ENABLE, + USP_TX_ENA, USP_TX_ENA); +} + +static void sirf_usp_tx_disable(struct sirf_usp *usp) +{ + regmap_update_bits(usp->regmap, USP_TX_RX_ENABLE, + USP_TX_ENA, ~USP_TX_ENA); + /* FIFO stop */ + regmap_write(usp->regmap, USP_TX_FIFO_OP, 0); +} + +static void sirf_usp_rx_enable(struct sirf_usp *usp) +{ + regmap_update_bits(usp->regmap, USP_RX_FIFO_OP, + USP_RX_FIFO_RESET, USP_RX_FIFO_RESET); + regmap_write(usp->regmap, USP_RX_FIFO_OP, 0); + + regmap_update_bits(usp->regmap, USP_RX_FIFO_OP, + USP_RX_FIFO_START, USP_RX_FIFO_START); + + regmap_update_bits(usp->regmap, USP_TX_RX_ENABLE, + USP_RX_ENA, USP_RX_ENA); +} + +static void sirf_usp_rx_disable(struct sirf_usp *usp) +{ + regmap_update_bits(usp->regmap, USP_TX_RX_ENABLE, + USP_RX_ENA, ~USP_RX_ENA); + /* FIFO stop */ + regmap_write(usp->regmap, USP_RX_FIFO_OP, 0); +} + +static int sirf_usp_pcm_dai_probe(struct snd_soc_dai *dai) +{ + struct sirf_usp *usp = snd_soc_dai_get_drvdata(dai); + snd_soc_dai_init_dma_data(dai, &usp->playback_dma_data, + &usp->capture_dma_data); + return 0; +} + +static int sirf_usp_pcm_set_dai_fmt(struct snd_soc_dai *dai, + unsigned int fmt) +{ + struct sirf_usp *usp = snd_soc_dai_get_drvdata(dai); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + break; + default: + dev_err(dai->dev, "Only CBM and CFM supported\n"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_DSP_A: + usp->daifmt_format = (fmt & SND_SOC_DAIFMT_FORMAT_MASK); + break; + default: + dev_err(dai->dev, "Only I2S and DSP_A format supported\n"); + return -EINVAL; + } + + return 0; +} + +static int sirf_usp_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct sirf_usp *usp = snd_soc_dai_get_drvdata(dai); + + /* Configure RISC mode */ + regmap_update_bits(usp->regmap, USP_RISC_DSP_MODE, + USP_RISC_DSP_SEL, ~USP_RISC_DSP_SEL); + + /* + * Configure DMA IO Length register + * Set no limit, USP can receive data continuously until it is diabled + */ + regmap_write(usp->regmap, USP_TX_DMA_IO_LEN, 0); + regmap_write(usp->regmap, USP_RX_DMA_IO_LEN, 0); + + regmap_write(usp->regmap, USP_RX_FRAME_CTRL, USP_SINGLE_SYNC_MODE); + + regmap_write(usp->regmap, USP_TX_FRAME_CTRL, USP_TXC_SLAVE_CLK_SAMPLE); + + /* Configure Mode2 register */ + regmap_write(usp->regmap, USP_MODE2, (1 << USP_RXD_DELAY_LEN_OFFSET) | + (0 << USP_TXD_DELAY_LEN_OFFSET)); + + /* Configure Mode1 register */ + regmap_write(usp->regmap, USP_MODE1, + USP_SYNC_MODE | USP_EN | USP_TXD_ACT_EDGE_FALLING | + USP_RFS_ACT_LEVEL_LOGIC1 | USP_TFS_ACT_LEVEL_LOGIC1 | + USP_TX_UFLOW_REPEAT_ZERO); + + /* Configure RX DMA IO Control register */ + regmap_write(usp->regmap, USP_RX_DMA_IO_CTRL, 0); + + /* Congiure RX FIFO Control register */ + regmap_write(usp->regmap, USP_RX_FIFO_CTRL, + (USP_RX_FIFO_THRESHOLD << USP_RX_FIFO_THD_OFFSET) | + (USP_TX_RX_FIFO_WIDTH_DWORD << USP_RX_FIFO_WIDTH_OFFSET)); + + /* Congiure RX FIFO Level Check register */ + regmap_write(usp->regmap, USP_RX_FIFO_LEVEL_CHK, + RX_FIFO_SC(0x04) | RX_FIFO_LC(0x0E) | RX_FIFO_HC(0x1B)); + + /* Configure TX DMA IO Control register*/ + regmap_write(usp->regmap, USP_TX_DMA_IO_CTRL, 0); + + /* Configure TX FIFO Control register */ + regmap_write(usp->regmap, USP_TX_FIFO_CTRL, + (USP_TX_FIFO_THRESHOLD << USP_TX_FIFO_THD_OFFSET) | + (USP_TX_RX_FIFO_WIDTH_DWORD << USP_TX_FIFO_WIDTH_OFFSET)); + /* Congiure TX FIFO Level Check register */ + regmap_write(usp->regmap, USP_TX_FIFO_LEVEL_CHK, + TX_FIFO_SC(0x1B) | TX_FIFO_LC(0x0E) | TX_FIFO_HC(0x04)); + + return 0; +} + +static int sirf_usp_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct sirf_usp *usp = snd_soc_dai_get_drvdata(dai); + u32 data_len, frame_len, shifter_len; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + data_len = 16; + frame_len = 16; + break; + case SNDRV_PCM_FORMAT_S24_LE: + data_len = 24; + frame_len = 32; + break; + case SNDRV_PCM_FORMAT_S24_3LE: + data_len = 24; + frame_len = 24; + break; + default: + dev_err(dai->dev, "Format unsupported\n"); + return -EINVAL; + } + + shifter_len = data_len; + + switch (usp->daifmt_format) { + case SND_SOC_DAIFMT_I2S: + regmap_update_bits(usp->regmap, USP_RX_FRAME_CTRL, + USP_I2S_SYNC_CHG, USP_I2S_SYNC_CHG); + break; + case SND_SOC_DAIFMT_DSP_A: + regmap_update_bits(usp->regmap, USP_RX_FRAME_CTRL, + USP_I2S_SYNC_CHG, 0); + frame_len = data_len * params_channels(params); + data_len = frame_len; + break; + default: + dev_err(dai->dev, "Only support I2S and DSP_A mode\n"); + return -EINVAL; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + regmap_update_bits(usp->regmap, USP_TX_FRAME_CTRL, + USP_TXC_DATA_LEN_MASK | USP_TXC_FRAME_LEN_MASK + | USP_TXC_SHIFTER_LEN_MASK, + ((data_len - 1) << USP_TXC_DATA_LEN_OFFSET) + | ((frame_len - 1) << USP_TXC_FRAME_LEN_OFFSET) + | ((shifter_len - 1) << USP_TXC_SHIFTER_LEN_OFFSET)); + else + regmap_update_bits(usp->regmap, USP_RX_FRAME_CTRL, + USP_RXC_DATA_LEN_MASK | USP_RXC_FRAME_LEN_MASK + | USP_RXC_SHIFTER_LEN_MASK, + ((data_len - 1) << USP_RXC_DATA_LEN_OFFSET) + | ((frame_len - 1) << USP_RXC_FRAME_LEN_OFFSET) + | ((shifter_len - 1) << USP_RXC_SHIFTER_LEN_OFFSET)); + + regmap_update_bits(usp->regmap, USP_MODE1, + USP_CLOCK_MODE_SLAVE, USP_CLOCK_MODE_SLAVE); + regmap_update_bits(usp->regmap, USP_MODE2, + USP_TFS_CLK_SLAVE_MODE | USP_RFS_CLK_SLAVE_MODE, + USP_TFS_CLK_SLAVE_MODE | USP_RFS_CLK_SLAVE_MODE); + + return 0; +} + +static int sirf_usp_pcm_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct sirf_usp *usp = snd_soc_dai_get_drvdata(dai); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + sirf_usp_tx_enable(usp); + else + sirf_usp_rx_enable(usp); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + sirf_usp_tx_disable(usp); + else + sirf_usp_rx_disable(usp); + break; + } + + return 0; +} + +static const struct snd_soc_dai_ops sirf_usp_pcm_dai_ops = { + .startup = sirf_usp_i2s_startup, + .trigger = sirf_usp_pcm_trigger, + .set_fmt = sirf_usp_pcm_set_dai_fmt, + .hw_params = sirf_usp_pcm_hw_params, +}; + +static struct snd_soc_dai_driver sirf_usp_pcm_dai = { + .probe = sirf_usp_pcm_dai_probe, + .name = "sirf-usp-pcm", + .id = 0, + .playback = { + .stream_name = "SiRF USP PCM Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S24_3LE, + }, + .capture = { + .stream_name = "SiRF USP PCM Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S24_3LE, + }, + .ops = &sirf_usp_pcm_dai_ops, +}; + +#ifdef CONFIG_PM_RUNTIME +static int sirf_usp_pcm_runtime_suspend(struct device *dev) +{ + struct sirf_usp *usp = dev_get_drvdata(dev); + clk_disable_unprepare(usp->clk); + return 0; +} + +static int sirf_usp_pcm_runtime_resume(struct device *dev) +{ + struct sirf_usp *usp = dev_get_drvdata(dev); + return clk_prepare_enable(usp->clk); +} +#endif + +#ifdef CONFIG_PM_SLEEP +static int sirf_usp_pcm_suspend(struct device *dev) +{ + struct sirf_usp *usp = dev_get_drvdata(dev); + + if (!pm_runtime_status_suspended(dev)) { + regmap_read(usp->regmap, USP_MODE1, &usp->mode1_reg); + regmap_read(usp->regmap, USP_MODE2, &usp->mode2_reg); + sirf_usp_pcm_runtime_suspend(dev); + } + return 0; +} + +static int sirf_usp_pcm_resume(struct device *dev) +{ + struct sirf_usp *usp = dev_get_drvdata(dev); + int ret; + + if (!pm_runtime_status_suspended(dev)) { + ret = sirf_usp_pcm_runtime_resume(dev); + if (ret) + return ret; + regmap_write(usp->regmap, USP_MODE1, usp->mode1_reg); + regmap_write(usp->regmap, USP_MODE2, usp->mode2_reg); + } + return 0; +} +#endif + +static const struct snd_soc_component_driver sirf_usp_component = { + .name = "sirf-usp", +}; + +static const struct regmap_config sirf_usp_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = USP_RX_FIFO_DATA, + .cache_type = REGCACHE_NONE, +}; + +static int sirf_usp_pcm_probe(struct platform_device *pdev) +{ + int ret; + struct sirf_usp *usp; + void __iomem *base; + struct resource *mem_res; + + usp = devm_kzalloc(&pdev->dev, sizeof(struct sirf_usp), + GFP_KERNEL); + if (!usp) + return -ENOMEM; + + platform_set_drvdata(pdev, usp); + + mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap(&pdev->dev, mem_res->start, + resource_size(mem_res)); + if (base == NULL) + return -ENOMEM; + usp->regmap = devm_regmap_init_mmio(&pdev->dev, base, + &sirf_usp_regmap_config); + if (IS_ERR(usp->regmap)) + return PTR_ERR(usp->regmap); + + usp->clk = devm_clk_get(&pdev->dev, NULL); + if (IS_ERR(usp->clk)) { + dev_err(&pdev->dev, "Get clock failed.\n"); + return PTR_ERR(usp->clk); + } + + pm_runtime_enable(&pdev->dev); + + ret = devm_snd_soc_register_component(&pdev->dev, &sirf_usp_component, + &sirf_usp_pcm_dai, 1); + if (ret) { + dev_err(&pdev->dev, "Register Audio SoC dai failed.\n"); + return ret; + } + return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); +} + +static int sirf_usp_pcm_remove(struct platform_device *pdev) +{ + pm_runtime_disable(&pdev->dev); + return 0; +} + +static const struct of_device_id sirf_usp_pcm_of_match[] = { + { .compatible = "sirf,prima2-usp-pcm", }, + {} +}; +MODULE_DEVICE_TABLE(of, sirf_usp_pcm_of_match); + +static const struct dev_pm_ops sirf_usp_pcm_pm_ops = { + SET_RUNTIME_PM_OPS(sirf_usp_pcm_runtime_suspend, + sirf_usp_pcm_runtime_resume, NULL) + SET_SYSTEM_SLEEP_PM_OPS(sirf_usp_pcm_suspend, sirf_usp_pcm_resume) +}; + +static struct platform_driver sirf_usp_pcm_driver = { + .driver = { + .name = "sirf-usp-pcm", + .owner = THIS_MODULE, + .of_match_table = sirf_usp_pcm_of_match, + .pm = &sirf_usp_pcm_pm_ops, + }, + .probe = sirf_usp_pcm_probe, + .remove = sirf_usp_pcm_remove, +}; + +module_platform_driver(sirf_usp_pcm_driver); + +MODULE_DESCRIPTION("SiRF SoC USP PCM bus driver"); +MODULE_AUTHOR("RongJun Ying "); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/sirf/sirf-usp.h b/sound/soc/sirf/sirf-usp.h new file mode 100644 index 000000000000..bf0201cb15bc --- /dev/null +++ b/sound/soc/sirf/sirf-usp.h @@ -0,0 +1,293 @@ +/* + * arch/arm/mach-prima2/include/mach/sirfsoc_usp.h + * + * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company. + * + * Licensed under GPLv2 or later. + */ + +#ifndef _SIRF_USP_H +#define _SIRF_USP_H + +/* USP Registers */ +#define USP_MODE1 0x00 +#define USP_MODE2 0x04 +#define USP_TX_FRAME_CTRL 0x08 +#define USP_RX_FRAME_CTRL 0x0C +#define USP_TX_RX_ENABLE 0x10 +#define USP_INT_ENABLE 0x14 +#define USP_INT_STATUS 0x18 +#define USP_PIN_IO_DATA 0x1C +#define USP_RISC_DSP_MODE 0x20 +#define USP_AYSNC_PARAM_REG 0x24 +#define USP_IRDA_X_MODE_DIV 0x28 +#define USP_SM_CFG 0x2C +#define USP_TX_DMA_IO_CTRL 0x100 +#define USP_TX_DMA_IO_LEN 0x104 +#define USP_TX_FIFO_CTRL 0x108 +#define USP_TX_FIFO_LEVEL_CHK 0x10C +#define USP_TX_FIFO_OP 0x110 +#define USP_TX_FIFO_STATUS 0x114 +#define USP_TX_FIFO_DATA 0x118 +#define USP_RX_DMA_IO_CTRL 0x120 +#define USP_RX_DMA_IO_LEN 0x124 +#define USP_RX_FIFO_CTRL 0x128 +#define USP_RX_FIFO_LEVEL_CHK 0x12C +#define USP_RX_FIFO_OP 0x130 +#define USP_RX_FIFO_STATUS 0x134 +#define USP_RX_FIFO_DATA 0x138 + +/* USP MODE register-1 */ +#define USP_SYNC_MODE 0x00000001 +#define USP_CLOCK_MODE_SLAVE 0x00000002 +#define USP_LOOP_BACK_EN 0x00000004 +#define USP_HPSIR_EN 0x00000008 +#define USP_ENDIAN_CTRL_LSBF 0x00000010 +#define USP_EN 0x00000020 +#define USP_RXD_ACT_EDGE_FALLING 0x00000040 +#define USP_TXD_ACT_EDGE_FALLING 0x00000080 +#define USP_RFS_ACT_LEVEL_LOGIC1 0x00000100 +#define USP_TFS_ACT_LEVEL_LOGIC1 0x00000200 +#define USP_SCLK_IDLE_MODE_TOGGLE 0x00000400 +#define USP_SCLK_IDLE_LEVEL_LOGIC1 0x00000800 +#define USP_SCLK_PIN_MODE_IO 0x00001000 +#define USP_RFS_PIN_MODE_IO 0x00002000 +#define USP_TFS_PIN_MODE_IO 0x00004000 +#define USP_RXD_PIN_MODE_IO 0x00008000 +#define USP_TXD_PIN_MODE_IO 0x00010000 +#define USP_SCLK_IO_MODE_INPUT 0x00020000 +#define USP_RFS_IO_MODE_INPUT 0x00040000 +#define USP_TFS_IO_MODE_INPUT 0x00080000 +#define USP_RXD_IO_MODE_INPUT 0x00100000 +#define USP_TXD_IO_MODE_INPUT 0x00200000 +#define USP_IRDA_WIDTH_DIV_MASK 0x3FC00000 +#define USP_IRDA_WIDTH_DIV_OFFSET 0 +#define USP_IRDA_IDLE_LEVEL_HIGH 0x40000000 +#define USP_TX_UFLOW_REPEAT_ZERO 0x80000000 +#define USP_TX_ENDIAN_MODE 0x00000020 +#define USP_RX_ENDIAN_MODE 0x00000020 + +/* USP Mode Register-2 */ +#define USP_RXD_DELAY_LEN_MASK 0x000000FF +#define USP_RXD_DELAY_LEN_OFFSET 0 + +#define USP_TXD_DELAY_LEN_MASK 0x0000FF00 +#define USP_TXD_DELAY_LEN_OFFSET 8 + +#define USP_ENA_CTRL_MODE 0x00010000 +#define USP_FRAME_CTRL_MODE 0x00020000 +#define USP_TFS_SOURCE_MODE 0x00040000 +#define USP_TFS_MS_MODE 0x00080000 +#define USP_CLK_DIVISOR_MASK 0x7FE00000 +#define USP_CLK_DIVISOR_OFFSET 21 + +#define USP_TFS_CLK_SLAVE_MODE (1<<20) +#define USP_RFS_CLK_SLAVE_MODE (1<<19) + +#define USP_IRDA_DATA_WIDTH 0x80000000 + +/* USP Transmit Frame Control Register */ + +#define USP_TXC_DATA_LEN_MASK 0x000000FF +#define USP_TXC_DATA_LEN_OFFSET 0 + +#define USP_TXC_SYNC_LEN_MASK 0x0000FF00 +#define USP_TXC_SYNC_LEN_OFFSET 8 + +#define USP_TXC_FRAME_LEN_MASK 0x00FF0000 +#define USP_TXC_FRAME_LEN_OFFSET 16 + +#define USP_TXC_SHIFTER_LEN_MASK 0x1F000000 +#define USP_TXC_SHIFTER_LEN_OFFSET 24 + +#define USP_TXC_SLAVE_CLK_SAMPLE 0x20000000 + +#define USP_TXC_CLK_DIVISOR_MASK 0xC0000000 +#define USP_TXC_CLK_DIVISOR_OFFSET 30 + +/* USP Receive Frame Control Register */ + +#define USP_RXC_DATA_LEN_MASK 0x000000FF +#define USP_RXC_DATA_LEN_OFFSET 0 + +#define USP_RXC_FRAME_LEN_MASK 0x0000FF00 +#define USP_RXC_FRAME_LEN_OFFSET 8 + +#define USP_RXC_SHIFTER_LEN_MASK 0x001F0000 +#define USP_RXC_SHIFTER_LEN_OFFSET 16 + +#define USP_START_EDGE_MODE 0x00800000 +#define USP_I2S_SYNC_CHG 0x00200000 + +#define USP_RXC_CLK_DIVISOR_MASK 0x0F000000 +#define USP_RXC_CLK_DIVISOR_OFFSET 24 +#define USP_SINGLE_SYNC_MODE 0x00400000 + +/* Tx - RX Enable Register */ + +#define USP_RX_ENA 0x00000001 +#define USP_TX_ENA 0x00000002 + +/* USP Interrupt Enable and status Register */ +#define USP_RX_DONE_INT 0x00000001 +#define USP_TX_DONE_INT 0x00000002 +#define USP_RX_OFLOW_INT 0x00000004 +#define USP_TX_UFLOW_INT 0x00000008 +#define USP_RX_IO_DMA_INT 0x00000010 +#define USP_TX_IO_DMA_INT 0x00000020 +#define USP_RXFIFO_FULL_INT 0x00000040 +#define USP_TXFIFO_EMPTY_INT 0x00000080 +#define USP_RXFIFO_THD_INT 0x00000100 +#define USP_TXFIFO_THD_INT 0x00000200 +#define USP_UART_FRM_ERR_INT 0x00000400 +#define USP_RX_TIMEOUT_INT 0x00000800 +#define USP_TX_ALLOUT_INT 0x00001000 +#define USP_RXD_BREAK_INT 0x00008000 + +/* All possible TX interruots */ +#define USP_TX_INTERRUPT (USP_TX_DONE_INT|USP_TX_UFLOW_INT|\ + USP_TX_IO_DMA_INT|\ + USP_TXFIFO_EMPTY_INT|\ + USP_TXFIFO_THD_INT) +/* All possible RX interruots */ +#define USP_RX_INTERRUPT (USP_RX_DONE_INT|USP_RX_OFLOW_INT|\ + USP_RX_IO_DMA_INT|\ + USP_RXFIFO_FULL_INT|\ + USP_RXFIFO_THD_INT|\ + USP_RXFIFO_THD_INT|USP_RX_TIMEOUT_INT) + +#define USP_INT_ALL 0x1FFF + +/* USP Pin I/O Data Register */ + +#define USP_RFS_PIN_VALUE_MASK 0x00000001 +#define USP_TFS_PIN_VALUE_MASK 0x00000002 +#define USP_RXD_PIN_VALUE_MASK 0x00000004 +#define USP_TXD_PIN_VALUE_MASK 0x00000008 +#define USP_SCLK_PIN_VALUE_MASK 0x00000010 + +/* USP RISC/DSP Mode Register */ +#define USP_RISC_DSP_SEL 0x00000001 + +/* USP ASYNC PARAMETER Register*/ + +#define USP_ASYNC_TIMEOUT_MASK 0x0000FFFF +#define USP_ASYNC_TIMEOUT_OFFSET 0 +#define USP_ASYNC_TIMEOUT(x) (((x)&USP_ASYNC_TIMEOUT_MASK) \ + < Date: Thu, 3 Jul 2014 16:51:36 +0200 Subject: ASoC: adau1701: fix adau1701_reg_read() Fix a long standing bug in the read register routing of adau1701. The bytes arrive in the buffer in big-endian, so the result has to be shifted before and-ing the bytes in the loop. Signed-off-by: Daniel Mack Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/adau1701.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index d71c59cf7bdd..370b742117ef 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -230,8 +230,10 @@ static int adau1701_reg_read(void *context, unsigned int reg, *value = 0; - for (i = 0; i < size; i++) - *value |= recv_buf[i] << (i * 8); + for (i = 0; i < size; i++) { + *value <<= 8; + *value |= recv_buf[i]; + } return 0; } -- cgit v1.2.3 From bb17bc78885b6b2e53d46041605a7ed08c5274c2 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Tue, 1 Jul 2014 09:59:31 +0530 Subject: ASoC: twl4030: Remove unused variable 'status' is not used in the function. Remove it. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 69e12a311ba2..6ab157065353 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -344,17 +344,16 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) static void twl4030_apll_enable(struct snd_soc_codec *codec, int enable) { struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); - int status = -1; if (enable) { twl4030->apll_enabled++; if (twl4030->apll_enabled == 1) - status = twl4030_audio_enable_resource( + twl4030_audio_enable_resource( TWL4030_AUDIO_RES_APLL); } else { twl4030->apll_enabled--; if (!twl4030->apll_enabled) - status = twl4030_audio_disable_resource( + twl4030_audio_disable_resource( TWL4030_AUDIO_RES_APLL); } } -- cgit v1.2.3 From d1498b13ae8cdd0353a405089609825d306b8036 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Tue, 1 Jul 2014 09:59:32 +0530 Subject: ASoC: wm8350: Remove unused variable 'irq' is not used in the function. Remove it. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/wm8350.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 392285edb595..d9e634c55e81 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1341,21 +1341,18 @@ int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which, { struct wm8350_data *priv = snd_soc_codec_get_drvdata(codec); struct wm8350 *wm8350 = priv->wm8350; - int irq; int ena; switch (which) { case WM8350_JDL: priv->hpl.jack = jack; priv->hpl.report = report; - irq = WM8350_IRQ_CODEC_JCK_DET_L; ena = WM8350_JDL_ENA; break; case WM8350_JDR: priv->hpr.jack = jack; priv->hpr.report = report; - irq = WM8350_IRQ_CODEC_JCK_DET_R; ena = WM8350_JDR_ENA; break; -- cgit v1.2.3 From 63d36f8814e2837fe8658542bc724659b152324d Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Tue, 1 Jul 2014 09:59:33 +0530 Subject: ASoC: wm8996: Remove unused variable 'ret' is not used in the function. Remove it. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/wm8996.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 69266332760e..622f92565030 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -620,15 +620,12 @@ static int bg_event(struct snd_soc_dapm_widget *w, static int cp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - int ret = 0; - switch (event) { case SND_SOC_DAPM_POST_PMU: msleep(5); break; default: WARN(1, "Invalid event %d\n", event); - ret = -EINVAL; } return 0; -- cgit v1.2.3 From 0a37c6efec4a2fdc2563c5a8faa472b814deee80 Mon Sep 17 00:00:00 2001 From: Praveen Diwakar Date: Fri, 4 Jul 2014 11:17:41 +0530 Subject: ASoC: wm_adsp: Add missing MODULE_LICENSE Since MODULE_LICENSE is missing the module load fails, so add this for module. Signed-off-by: Praveen Diwakar Signed-off-by: Vinod Koul Reviewed-by: Charles Keepax Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm_adsp.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 060027182dcb..2537725dd53f 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1758,3 +1758,5 @@ int wm_adsp2_init(struct wm_adsp *adsp, bool dvfs) return 0; } EXPORT_SYMBOL_GPL(wm_adsp2_init); + +MODULE_LICENSE("GPL v2"); -- cgit v1.2.3 From 07cf7cbadb4d97a78be61119a406de8fe446467e Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Fri, 20 Jun 2014 14:41:13 +0800 Subject: ASoC: add RT286 CODEC driver This patch adds Realtek ALC286 codec driver. ALC286 is a dual mode codec, which can run as HD-A or I2S mode. It is controlled by HD-A verb commands via I2C protocol. The following is the I/O difference between ALC286 and general I2S codecs. 1. A HD-A verb command contains three parts, NID, VID, and PID. And an I2S command contains only two parts: address and data. 2. Not only the register address is written, but the read command also includes the entire write command. 3. rt286 uses different registers for read and write the same bits. We map verb command to regmap structure. However, we read most registers from cache to prevent the asymmetry read/write issue in rt286. Signed-off-by: Bard Liao Signed-off-by: Gustaw Lewandowski Signed-off-by: Mark Brown --- include/sound/rt286.h | 19 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt286.c | 1208 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt286.h | 193 ++++++++ 5 files changed, 1426 insertions(+) create mode 100644 include/sound/rt286.h create mode 100644 sound/soc/codecs/rt286.c create mode 100644 sound/soc/codecs/rt286.h (limited to 'sound') diff --git a/include/sound/rt286.h b/include/sound/rt286.h new file mode 100644 index 000000000000..eb773d1485f2 --- /dev/null +++ b/include/sound/rt286.h @@ -0,0 +1,19 @@ +/* + * linux/sound/rt286.h -- Platform data for RT286 + * + * Copyright 2013 Realtek Microelectronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __LINUX_SND_RT286_H +#define __LINUX_SND_RT286_H + +struct rt286_platform_data { + bool cbj_en; /*combo jack enable*/ + bool gpio2_en; /*GPIO2 enable*/ +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index cbfa1e18f651..115e5de7b7eb 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -74,6 +74,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM3008 select SND_SOC_PCM512x_I2C if I2C select SND_SOC_PCM512x_SPI if SPI_MASTER + select SND_SOC_RT286 if I2C select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C select SND_SOC_RT5645 if I2C @@ -449,6 +450,9 @@ config SND_SOC_RL6231 default m if SND_SOC_RT5645=m default m if SND_SOC_RT5651=m +config SND_SOC_RT286 + tristate + config SND_SOC_RT5631 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index be3377b8d73f..c39449a862b3 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -68,6 +68,7 @@ snd-soc-pcm512x-objs := pcm512x.o snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o snd-soc-pcm512x-spi-objs := pcm512x-spi.o snd-soc-rl6231-objs := rl6231.o +snd-soc-rt286-objs := rt286.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o snd-soc-rt5645-objs := rt5645.o @@ -233,6 +234,7 @@ obj-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o obj-$(CONFIG_SND_SOC_RL6231) += snd-soc-rl6231.o +obj-$(CONFIG_SND_SOC_RT286) += snd-soc-rt286.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o obj-$(CONFIG_SND_SOC_RT5645) += snd-soc-rt5645.o diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c new file mode 100644 index 000000000000..acfba9c74c52 --- /dev/null +++ b/sound/soc/codecs/rt286.c @@ -0,0 +1,1208 @@ +/* + * rt286.c -- RT286 ALSA SoC audio codec driver + * + * Copyright 2013 Realtek Semiconductor Corp. + * Author: Bard Liao + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "rt286.h" + +#define RT286_VENDOR_ID 0x10ec0286 + +struct rt286_priv { + struct regmap *regmap; + struct snd_soc_codec *codec; + struct rt286_platform_data pdata; + struct i2c_client *i2c; + struct snd_soc_jack *jack; + struct delayed_work jack_detect_work; + int sys_clk; + struct reg_default *index_cache; +}; + +static struct reg_default rt286_index_def[] = { + { 0x01, 0xaaaa }, + { 0x02, 0x8aaa }, + { 0x03, 0x0002 }, + { 0x04, 0xaf01 }, + { 0x08, 0x000d }, + { 0x09, 0xd810 }, + { 0x0a, 0x0060 }, + { 0x0b, 0x0000 }, + { 0x0f, 0x0000 }, + { 0x19, 0x0a17 }, + { 0x20, 0x0020 }, + { 0x33, 0x0208 }, + { 0x49, 0x0004 }, + { 0x4f, 0x50e9 }, + { 0x50, 0x2c00 }, + { 0x63, 0x2902 }, +}; +#define INDEX_CACHE_SIZE ARRAY_SIZE(rt286_index_def) + +static const struct reg_default rt286_reg[] = { + { 0x00170500, 0x00000400 }, + { 0x00220000, 0x00000031 }, + { 0x00239000, 0x0000007f }, + { 0x0023a000, 0x0000007f }, + { 0x00270500, 0x00000400 }, + { 0x00370500, 0x00000400 }, + { 0x00870500, 0x00000400 }, + { 0x00920000, 0x00000031 }, + { 0x00935000, 0x000000c3 }, + { 0x00936000, 0x000000c3 }, + { 0x00970500, 0x00000400 }, + { 0x00b37000, 0x00000097 }, + { 0x00b37200, 0x00000097 }, + { 0x00b37300, 0x00000097 }, + { 0x00c37000, 0x00000000 }, + { 0x00c37100, 0x00000080 }, + { 0x01270500, 0x00000400 }, + { 0x01370500, 0x00000400 }, + { 0x01371f00, 0x411111f0 }, + { 0x01439000, 0x00000080 }, + { 0x0143a000, 0x00000080 }, + { 0x01470700, 0x00000000 }, + { 0x01470500, 0x00000400 }, + { 0x01470c00, 0x00000000 }, + { 0x01470100, 0x00000000 }, + { 0x01837000, 0x00000000 }, + { 0x01870500, 0x00000400 }, + { 0x02050000, 0x00000000 }, + { 0x02139000, 0x00000080 }, + { 0x0213a000, 0x00000080 }, + { 0x02170100, 0x00000000 }, + { 0x02170500, 0x00000400 }, + { 0x02170700, 0x00000000 }, + { 0x02270100, 0x00000000 }, + { 0x02370100, 0x00000000 }, + { 0x02040000, 0x00004002 }, + { 0x01870700, 0x00000020 }, + { 0x00830000, 0x000000c3 }, + { 0x00930000, 0x000000c3 }, + { 0x01270700, 0x00000000 }, +}; + +static bool rt286_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0 ... 0xff: + case RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID): + case RT286_GET_HP_SENSE: + case RT286_GET_MIC1_SENSE: + case RT286_PROC_COEF: + return true; + default: + return false; + } + + +} + +static bool rt286_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0 ... 0xff: + case RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID): + case RT286_GET_HP_SENSE: + case RT286_GET_MIC1_SENSE: + case RT286_SET_AUDIO_POWER: + case RT286_SET_HPO_POWER: + case RT286_SET_SPK_POWER: + case RT286_SET_DMIC1_POWER: + case RT286_SPK_MUX: + case RT286_HPO_MUX: + case RT286_ADC0_MUX: + case RT286_ADC1_MUX: + case RT286_SET_MIC1: + case RT286_SET_PIN_HPO: + case RT286_SET_PIN_SPK: + case RT286_SET_PIN_DMIC1: + case RT286_SPK_EAPD: + case RT286_SET_AMP_GAIN_HPO: + case RT286_SET_DMIC2_DEFAULT: + case RT286_DACL_GAIN: + case RT286_DACR_GAIN: + case RT286_ADCL_GAIN: + case RT286_ADCR_GAIN: + case RT286_MIC_GAIN: + case RT286_SPOL_GAIN: + case RT286_SPOR_GAIN: + case RT286_HPOL_GAIN: + case RT286_HPOR_GAIN: + case RT286_F_DAC_SWITCH: + case RT286_F_RECMIX_SWITCH: + case RT286_REC_MIC_SWITCH: + case RT286_REC_I2S_SWITCH: + case RT286_REC_LINE_SWITCH: + case RT286_REC_BEEP_SWITCH: + case RT286_DAC_FORMAT: + case RT286_ADC_FORMAT: + case RT286_COEF_INDEX: + case RT286_PROC_COEF: + case RT286_SET_AMP_GAIN_ADC_IN1: + case RT286_SET_AMP_GAIN_ADC_IN2: + case RT286_SET_POWER(RT286_DAC_OUT1): + case RT286_SET_POWER(RT286_DAC_OUT2): + case RT286_SET_POWER(RT286_ADC_IN1): + case RT286_SET_POWER(RT286_ADC_IN2): + case RT286_SET_POWER(RT286_DMIC2): + case RT286_SET_POWER(RT286_MIC1): + return true; + default: + return false; + } +} + +static int rt286_hw_write(void *context, unsigned int reg, unsigned int value) +{ + struct i2c_client *client = context; + struct rt286_priv *rt286 = i2c_get_clientdata(client); + u8 data[4]; + int ret, i; + + /*handle index registers*/ + if (reg <= 0xff) { + rt286_hw_write(client, RT286_COEF_INDEX, reg); + reg = RT286_PROC_COEF; + for (i = 0; i < INDEX_CACHE_SIZE; i++) { + if (reg == rt286->index_cache[i].reg) { + rt286->index_cache[i].def = value; + break; + } + + } + } + + data[0] = (reg >> 24) & 0xff; + data[1] = (reg >> 16) & 0xff; + /* + * 4 bit VID: reg should be 0 + * 12 bit VID: value should be 0 + * So we use an OR operator to handle it rather than use if condition. + */ + data[2] = ((reg >> 8) & 0xff) | ((value >> 8) & 0xff); + data[3] = value & 0xff; + + ret = i2c_master_send(client, data, 4); + + if (ret == 4) + return 0; + else + pr_err("ret=%d\n", ret); + if (ret < 0) + return ret; + else + return -EIO; +} + +static int rt286_hw_read(void *context, unsigned int reg, unsigned int *value) +{ + struct i2c_client *client = context; + struct i2c_msg xfer[2]; + int ret; + __be32 be_reg; + unsigned int index, vid, buf = 0x0; + + /*handle index registers*/ + if (reg <= 0xff) { + rt286_hw_write(client, RT286_COEF_INDEX, reg); + reg = RT286_PROC_COEF; + } + + reg = reg | 0x80000; + vid = (reg >> 8) & 0xfff; + + if (AC_VERB_GET_AMP_GAIN_MUTE == (vid & 0xf00)) { + index = (reg >> 8) & 0xf; + reg = (reg & ~0xf0f) | index; + } + be_reg = cpu_to_be32(reg); + + /* Write register */ + xfer[0].addr = client->addr; + xfer[0].flags = 0; + xfer[0].len = 4; + xfer[0].buf = (u8 *)&be_reg; + + /* Read data */ + xfer[1].addr = client->addr; + xfer[1].flags = I2C_M_RD; + xfer[1].len = 4; + xfer[1].buf = (u8 *)&buf; + + ret = i2c_transfer(client->adapter, xfer, 2); + if (ret < 0) + return ret; + else if (ret != 2) + return -EIO; + + *value = be32_to_cpu(buf); + + return 0; +} + +static void rt286_index_sync(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + int i; + + for (i = 0; i < INDEX_CACHE_SIZE; i++) { + snd_soc_write(codec, rt286->index_cache[i].reg, + rt286->index_cache[i].def); + } +} + +static int rt286_support_power_controls[] = { + RT286_DAC_OUT1, + RT286_DAC_OUT2, + RT286_ADC_IN1, + RT286_ADC_IN2, + RT286_MIC1, + RT286_DMIC1, + RT286_DMIC2, + RT286_SPK_OUT, + RT286_HP_OUT, +}; +#define RT286_POWER_REG_LEN ARRAY_SIZE(rt286_support_power_controls) + +static int rt286_jack_detect(struct snd_soc_codec *codec, bool *hp, bool *mic) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int val, buf; + int i; + + *hp = false; + *mic = false; + + if (rt286->pdata.cbj_en) { + buf = snd_soc_read(codec, RT286_GET_HP_SENSE); + *hp = buf & 0x80000000; + if (*hp) { + /* power on HV,VERF */ + snd_soc_update_bits(codec, + RT286_POWER_CTRL1, 0x1001, 0x0); + /* power LDO1 */ + snd_soc_update_bits(codec, + RT286_POWER_CTRL2, 0x4, 0x4); + snd_soc_write(codec, RT286_SET_MIC1, 0x24); + val = snd_soc_read(codec, RT286_CBJ_CTRL2); + + msleep(200); + i = 40; + while (((val & 0x0800) == 0) && (i > 0)) { + val = snd_soc_read(codec, + RT286_CBJ_CTRL2); + i--; + msleep(20); + } + + if (0x0400 == (val & 0x0700)) { + *mic = false; + + snd_soc_write(codec, + RT286_SET_MIC1, 0x20); + /* power off HV,VERF */ + snd_soc_update_bits(codec, + RT286_POWER_CTRL1, 0x1001, 0x1001); + snd_soc_update_bits(codec, + RT286_A_BIAS_CTRL3, 0xc000, 0x0000); + snd_soc_update_bits(codec, + RT286_CBJ_CTRL1, 0x0030, 0x0000); + snd_soc_update_bits(codec, + RT286_A_BIAS_CTRL2, 0xc000, 0x0000); + } else if ((0x0200 == (val & 0x0700)) || + (0x0100 == (val & 0x0700))) { + *mic = true; + snd_soc_update_bits(codec, + RT286_A_BIAS_CTRL3, 0xc000, 0x8000); + snd_soc_update_bits(codec, + RT286_CBJ_CTRL1, 0x0030, 0x0020); + snd_soc_update_bits(codec, + RT286_A_BIAS_CTRL2, 0xc000, 0x8000); + } else { + *mic = false; + } + + snd_soc_update_bits(codec, + RT286_MISC_CTRL1, + 0x0060, 0x0000); + } else { + snd_soc_update_bits(codec, + RT286_MISC_CTRL1, + 0x0060, 0x0020); + snd_soc_update_bits(codec, + RT286_A_BIAS_CTRL3, + 0xc000, 0x8000); + snd_soc_update_bits(codec, + RT286_CBJ_CTRL1, + 0x0030, 0x0020); + snd_soc_update_bits(codec, + RT286_A_BIAS_CTRL2, + 0xc000, 0x8000); + + *mic = false; + } + } else { + buf = snd_soc_read(codec, RT286_GET_HP_SENSE); + *hp = buf & 0x80000000; + buf = snd_soc_read(codec, RT286_GET_MIC1_SENSE); + *mic = buf & 0x80000000; + } + + return 0; +} + +static void rt286_jack_detect_work(struct work_struct *work) +{ + struct rt286_priv *rt286 = + container_of(work, struct rt286_priv, jack_detect_work.work); + int status = 0; + bool hp = false; + bool mic = false; + + rt286_jack_detect(rt286->codec, &hp, &mic); + + if (hp == true) + status |= SND_JACK_HEADPHONE; + + if (mic == true) + status |= SND_JACK_MICROPHONE; + + snd_soc_jack_report(rt286->jack, status, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); +} + +int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + rt286->jack = jack; + + /* Send an initial empty report */ + snd_soc_jack_report(rt286->jack, 0, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); + + return 0; +} +EXPORT_SYMBOL_GPL(rt286_mic_detect); + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6350, 50, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); + +static const struct snd_kcontrol_new rt286_snd_controls[] = { + SOC_DOUBLE_R_TLV("DAC0 Playback Volume", RT286_DACL_GAIN, + RT286_DACR_GAIN, 0, 0x7f, 0, out_vol_tlv), + SOC_DOUBLE_R_TLV("ADC0 Capture Volume", RT286_ADCL_GAIN, + RT286_ADCR_GAIN, 0, 0x7f, 0, out_vol_tlv), + SOC_SINGLE_TLV("AMIC Volume", RT286_MIC_GAIN, + 0, 0x3, 0, mic_vol_tlv), + SOC_DOUBLE_R("Speaker Playback Switch", RT286_SPOL_GAIN, + RT286_SPOR_GAIN, RT286_MUTE_SFT, 1, 1), +}; + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt286_front_mix[] = { + SOC_DAPM_SINGLE("DAC Switch", RT286_F_DAC_SWITCH, + RT286_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("RECMIX Switch", RT286_F_RECMIX_SWITCH, + RT286_MUTE_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt286_rec_mix[] = { + SOC_DAPM_SINGLE("Mic1 Switch", RT286_REC_MIC_SWITCH, + RT286_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("I2S Switch", RT286_REC_I2S_SWITCH, + RT286_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("Line1 Switch", RT286_REC_LINE_SWITCH, + RT286_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("Beep Switch", RT286_REC_BEEP_SWITCH, + RT286_MUTE_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new spo_enable_control = + SOC_DAPM_SINGLE("Switch", RT286_SET_PIN_SPK, + RT286_SET_PIN_SFT, 1, 0); + +static const struct snd_kcontrol_new hpol_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT286_HPOL_GAIN, + RT286_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new hpor_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT286_HPOR_GAIN, + RT286_MUTE_SFT, 1, 1); + +/* ADC0 source */ +static const char * const rt286_adc_src[] = { + "Mic", "RECMIX", "Dmic" +}; + +static const int rt286_adc_values[] = { + 0, 4, 5, +}; + +static SOC_VALUE_ENUM_SINGLE_DECL( + rt286_adc0_enum, RT286_ADC0_MUX, RT286_ADC_SEL_SFT, + RT286_ADC_SEL_MASK, rt286_adc_src, rt286_adc_values); + +static const struct snd_kcontrol_new rt286_adc0_mux = + SOC_DAPM_ENUM("ADC 0 source", rt286_adc0_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL( + rt286_adc1_enum, RT286_ADC1_MUX, RT286_ADC_SEL_SFT, + RT286_ADC_SEL_MASK, rt286_adc_src, rt286_adc_values); + +static const struct snd_kcontrol_new rt286_adc1_mux = + SOC_DAPM_ENUM("ADC 1 source", rt286_adc1_enum); + +static const char * const rt286_dac_src[] = { + "Front", "Surround" +}; +/* HP-OUT source */ +static SOC_ENUM_SINGLE_DECL(rt286_hpo_enum, RT286_HPO_MUX, + 0, rt286_dac_src); + +static const struct snd_kcontrol_new rt286_hpo_mux = +SOC_DAPM_ENUM("HPO source", rt286_hpo_enum); + +/* SPK-OUT source */ +static SOC_ENUM_SINGLE_DECL(rt286_spo_enum, RT286_SPK_MUX, + 0, rt286_dac_src); + +static const struct snd_kcontrol_new rt286_spo_mux = +SOC_DAPM_ENUM("SPO source", rt286_spo_enum); + +static int rt286_spk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_write(codec, + RT286_SPK_EAPD, RT286_SET_EAPD_HIGH); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_write(codec, + RT286_SPK_EAPD, RT286_SET_EAPD_LOW); + break; + + default: + return 0; + } + + return 0; +} + +static int rt286_set_dmic1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_write(codec, RT286_SET_PIN_DMIC1, 0x20); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_write(codec, RT286_SET_PIN_DMIC1, 0); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_adc_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + unsigned int nid; + + nid = (w->reg >> 20) & 0xff; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0), + 0x7080, 0x7000); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0), + 0x7080, 0x7080); + break; + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt286_dapm_widgets[] = { + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC1 Pin"), + SND_SOC_DAPM_INPUT("DMIC2 Pin"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("LINE1"), + SND_SOC_DAPM_INPUT("Beep"), + + /* DMIC */ + SND_SOC_DAPM_PGA_E("DMIC1", RT286_SET_POWER(RT286_DMIC1), 0, 1, + NULL, 0, rt286_set_dmic1_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA("DMIC2", RT286_SET_POWER(RT286_DMIC2), 0, 1, + NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC Receiver", SND_SOC_NOPM, + 0, 0, NULL, 0), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIX", SND_SOC_NOPM, 0, 0, + rt286_rec_mix, ARRAY_SIZE(rt286_rec_mix)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC 0", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC 1", NULL, SND_SOC_NOPM, 0, 0), + + /* ADC Mux */ + SND_SOC_DAPM_MUX_E("ADC 0 Mux", RT286_SET_POWER(RT286_ADC_IN1), 0, 1, + &rt286_adc0_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_MUX_E("ADC 1 Mux", RT286_SET_POWER(RT286_ADC_IN2), 0, 1, + &rt286_adc1_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), + + /* Output Side */ + /* DACs */ + SND_SOC_DAPM_DAC("DAC 0", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC 1", NULL, SND_SOC_NOPM, 0, 0), + + /* Output Mux */ + SND_SOC_DAPM_MUX("SPK Mux", SND_SOC_NOPM, 0, 0, &rt286_spo_mux), + SND_SOC_DAPM_MUX("HPO Mux", SND_SOC_NOPM, 0, 0, &rt286_hpo_mux), + + SND_SOC_DAPM_SUPPLY("HP Power", RT286_SET_PIN_HPO, + RT286_SET_PIN_SFT, 0, NULL, 0), + + /* Output Mixer */ + SND_SOC_DAPM_MIXER("Front", RT286_SET_POWER(RT286_DAC_OUT1), 0, 1, + rt286_front_mix, ARRAY_SIZE(rt286_front_mix)), + SND_SOC_DAPM_PGA("Surround", RT286_SET_POWER(RT286_DAC_OUT2), 0, 1, + NULL, 0), + + /* Output Pga */ + SND_SOC_DAPM_SWITCH_E("SPO", SND_SOC_NOPM, 0, 0, + &spo_enable_control, rt286_spk_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SWITCH("HPO L", SND_SOC_NOPM, 0, 0, + &hpol_enable_control), + SND_SOC_DAPM_SWITCH("HPO R", SND_SOC_NOPM, 0, 0, + &hpor_enable_control), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("SPOL"), + SND_SOC_DAPM_OUTPUT("SPOR"), + SND_SOC_DAPM_OUTPUT("HPO Pin"), + SND_SOC_DAPM_OUTPUT("SPDIF"), +}; + +static const struct snd_soc_dapm_route rt286_dapm_routes[] = { + {"DMIC1", NULL, "DMIC1 Pin"}, + {"DMIC2", NULL, "DMIC2 Pin"}, + {"DMIC1", NULL, "DMIC Receiver"}, + {"DMIC2", NULL, "DMIC Receiver"}, + + {"RECMIX", "Beep Switch", "Beep"}, + {"RECMIX", "Line1 Switch", "LINE1"}, + {"RECMIX", "Mic1 Switch", "MIC1"}, + + {"ADC 0 Mux", "Dmic", "DMIC1"}, + {"ADC 0 Mux", "RECMIX", "RECMIX"}, + {"ADC 0 Mux", "Mic", "MIC1"}, + {"ADC 1 Mux", "Dmic", "DMIC2"}, + {"ADC 1 Mux", "RECMIX", "RECMIX"}, + {"ADC 1 Mux", "Mic", "MIC1"}, + + {"ADC 0", NULL, "ADC 0 Mux"}, + {"ADC 1", NULL, "ADC 1 Mux"}, + + {"AIF1TX", NULL, "ADC 0"}, + {"AIF2TX", NULL, "ADC 1"}, + + {"DAC 0", NULL, "AIF1RX"}, + {"DAC 1", NULL, "AIF2RX"}, + + {"Front", "DAC Switch", "DAC 0"}, + {"Front", "RECMIX Switch", "RECMIX"}, + + {"Surround", NULL, "DAC 1"}, + + {"SPK Mux", "Front", "Front"}, + {"SPK Mux", "Surround", "Surround"}, + + {"HPO Mux", "Front", "Front"}, + {"HPO Mux", "Surround", "Surround"}, + + {"SPO", "Switch", "SPK Mux"}, + {"HPO L", "Switch", "HPO Mux"}, + {"HPO R", "Switch", "HPO Mux"}, + {"HPO L", NULL, "HP Power"}, + {"HPO R", NULL, "HP Power"}, + + {"SPOL", NULL, "SPO"}, + {"SPOR", NULL, "SPO"}, + {"HPO Pin", NULL, "HPO L"}, + {"HPO Pin", NULL, "HPO R"}, +}; + +static int rt286_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0; + int d_len_code; + + switch (params_rate(params)) { + /* bit 14 0:48K 1:44.1K */ + case 44100: + val |= 0x4000; + break; + case 48000: + break; + default: + dev_err(codec->dev, "Unsupported sample rate %d\n", + params_rate(params)); + return -EINVAL; + } + switch (rt286->sys_clk) { + case 12288000: + case 24576000: + if (params_rate(params) != 48000) { + dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n", + params_rate(params), rt286->sys_clk); + return -EINVAL; + } + break; + case 11289600: + case 22579200: + if (params_rate(params) != 44100) { + dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n", + params_rate(params), rt286->sys_clk); + return -EINVAL; + } + break; + } + + if (params_channels(params) <= 16) { + /* bit 3:0 Number of Channel */ + val |= (params_channels(params) - 1); + } else { + dev_err(codec->dev, "Unsupported channels %d\n", + params_channels(params)); + return -EINVAL; + } + + d_len_code = 0; + switch (params_width(params)) { + /* bit 6:4 Bits per Sample */ + case 16: + d_len_code = 0; + val |= (0x1 << 4); + break; + case 32: + d_len_code = 2; + val |= (0x4 << 4); + break; + case 20: + d_len_code = 1; + val |= (0x2 << 4); + break; + case 24: + d_len_code = 2; + val |= (0x3 << 4); + break; + case 8: + d_len_code = 3; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x0018, d_len_code << 3); + dev_dbg(codec->dev, "format val = 0x%x\n", val); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_update_bits(codec, RT286_DAC_FORMAT, 0x407f, val); + else + snd_soc_update_bits(codec, RT286_ADC_FORMAT, 0x407f, val); + + return 0; +} + +static int rt286_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x800, 0x800); + break; + case SND_SOC_DAIFMT_CBS_CFS: + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x800, 0x0); + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x300, 0x0); + break; + case SND_SOC_DAIFMT_LEFT_J: + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x300, 0x1 << 8); + break; + case SND_SOC_DAIFMT_DSP_A: + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x300, 0x2 << 8); + break; + case SND_SOC_DAIFMT_DSP_B: + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x300, 0x3 << 8); + break; + default: + return -EINVAL; + } + /* bit 15 Stream Type 0:PCM 1:Non-PCM */ + snd_soc_update_bits(codec, RT286_DAC_FORMAT, 0x8000, 0); + snd_soc_update_bits(codec, RT286_ADC_FORMAT, 0x8000, 0); + + return 0; +} + +static int rt286_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "%s freq=%d\n", __func__, freq); + + if (RT286_SCLK_S_MCLK == clk_id) { + snd_soc_update_bits(codec, + RT286_I2S_CTRL2, 0x0100, 0x0); + snd_soc_update_bits(codec, + RT286_PLL_CTRL1, 0x20, 0x20); + } else { + snd_soc_update_bits(codec, + RT286_I2S_CTRL2, 0x0100, 0x0100); + snd_soc_update_bits(codec, + RT286_PLL_CTRL, 0x4, 0x4); + snd_soc_update_bits(codec, + RT286_PLL_CTRL1, 0x20, 0x0); + } + + switch (freq) { + case 19200000: + if (RT286_SCLK_S_MCLK == clk_id) { + dev_err(codec->dev, "Should not use MCLK\n"); + return -EINVAL; + } + snd_soc_update_bits(codec, + RT286_I2S_CTRL2, 0x40, 0x40); + break; + case 24000000: + if (RT286_SCLK_S_MCLK == clk_id) { + dev_err(codec->dev, "Should not use MCLK\n"); + return -EINVAL; + } + snd_soc_update_bits(codec, + RT286_I2S_CTRL2, 0x40, 0x0); + break; + case 12288000: + case 11289600: + snd_soc_update_bits(codec, + RT286_I2S_CTRL2, 0x8, 0x0); + snd_soc_update_bits(codec, + RT286_CLK_DIV, 0xfc1e, 0x0004); + break; + case 24576000: + case 22579200: + snd_soc_update_bits(codec, + RT286_I2S_CTRL2, 0x8, 0x8); + snd_soc_update_bits(codec, + RT286_CLK_DIV, 0xfc1e, 0x5406); + break; + default: + dev_err(codec->dev, "Unsupported system clock\n"); + return -EINVAL; + } + + rt286->sys_clk = freq; + + return 0; +} + +static int rt286_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_codec *codec = dai->codec; + + dev_dbg(codec->dev, "%s ratio=%d\n", __func__, ratio); + if (50 == ratio) + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x1000, 0x1000); + else + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x1000, 0x0); + + + return 0; +} + +static int rt286_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) + snd_soc_write(codec, + RT286_SET_AUDIO_POWER, AC_PWRST_D0); + break; + + case SND_SOC_BIAS_STANDBY: + snd_soc_write(codec, + RT286_SET_AUDIO_POWER, AC_PWRST_D3); + break; + + default: + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +static irqreturn_t rt286_irq(int irq, void *data) +{ + struct rt286_priv *rt286 = data; + bool hp = false; + bool mic = false; + int status = 0; + + rt286_jack_detect(rt286->codec, &hp, &mic); + + /* Clear IRQ */ + snd_soc_update_bits(rt286->codec, + RT286_IRQ_CTRL, 0x1, 0x1); + + if (hp == true) + status |= SND_JACK_HEADPHONE; + + if (mic == true) + status |= SND_JACK_MICROPHONE; + + snd_soc_jack_report(rt286->jack, status, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); + + pm_wakeup_event(&rt286->i2c->dev, 300); + + return IRQ_HANDLED; +} + +static int rt286_probe(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + int i, ret; + + ret = snd_soc_read(codec, + RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID)); + if (ret != RT286_VENDOR_ID) { + dev_err(codec->dev, + "Device with ID register %x is not rt286\n", ret); + return -ENODEV; + } + + snd_soc_write(codec, RT286_SET_AUDIO_POWER, AC_PWRST_D3); + + for (i = 0; i < RT286_POWER_REG_LEN; i++) + snd_soc_write(codec, + RT286_SET_POWER(rt286_support_power_controls[i]), + AC_PWRST_D1); + + if (!rt286->pdata.cbj_en) { + snd_soc_write(codec, RT286_CBJ_CTRL2, 0x0000); + snd_soc_write(codec, RT286_MIC1_DET_CTRL, 0x0816); + snd_soc_write(codec, RT286_MISC_CTRL1, 0x0000); + snd_soc_update_bits(codec, + RT286_CBJ_CTRL1, 0xf000, 0xb000); + } else { + snd_soc_update_bits(codec, + RT286_CBJ_CTRL1, 0xf000, 0x5000); + } + + mdelay(10); + + if (!rt286->pdata.gpio2_en) + snd_soc_write(codec, RT286_SET_DMIC2_DEFAULT, 0x4000); + else + snd_soc_write(codec, RT286_SET_DMIC2_DEFAULT, 0); + + mdelay(10); + + /*Power down LDO2*/ + snd_soc_update_bits(codec, RT286_POWER_CTRL2, 0x8, 0x0); + + codec->dapm.bias_level = SND_SOC_BIAS_OFF; + rt286->codec = codec; + + rt286->i2c->irq = 0; + if (rt286->i2c->irq) { + snd_soc_update_bits(codec, + RT286_IRQ_CTRL, 0x2, 0x2); + + INIT_DELAYED_WORK(&rt286->jack_detect_work, + rt286_jack_detect_work); + schedule_delayed_work(&rt286->jack_detect_work, + msecs_to_jiffies(1250)); + + ret = request_threaded_irq(rt286->i2c->irq, NULL, rt286_irq, + IRQF_TRIGGER_HIGH | IRQF_ONESHOT, "rt286", rt286); + if (ret != 0) { + dev_err(codec->dev, + "Failed to reguest IRQ: %d\n", ret); + return ret; + } + } + + return 0; +} + +static int rt286_remove(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + cancel_delayed_work_sync(&rt286->jack_detect_work); + + return 0; +} + +#ifdef CONFIG_PM +static int rt286_suspend(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt286->regmap, true); + regcache_mark_dirty(rt286->regmap); + + return 0; +} + +static int rt286_resume(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt286->regmap, false); + rt286_index_sync(codec); + regcache_sync(rt286->regmap); + + return 0; +} +#else +#define rt286_suspend NULL +#define rt286_resume NULL +#endif + +#define RT286_STEREO_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +#define RT286_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +static const struct snd_soc_dai_ops rt286_aif_dai_ops = { + .hw_params = rt286_hw_params, + .set_fmt = rt286_set_dai_fmt, + .set_sysclk = rt286_set_dai_sysclk, + .set_bclk_ratio = rt286_set_bclk_ratio, +}; + +static struct snd_soc_dai_driver rt286_dai[] = { + { + .name = "rt286-aif1", + .id = RT286_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .ops = &rt286_aif_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "rt286-aif2", + .id = RT286_AIF2, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .ops = &rt286_aif_dai_ops, + .symmetric_rates = 1, + }, + +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt286 = { + .probe = rt286_probe, + .remove = rt286_remove, + .suspend = rt286_suspend, + .resume = rt286_resume, + .set_bias_level = rt286_set_bias_level, + .idle_bias_off = true, + .controls = rt286_snd_controls, + .num_controls = ARRAY_SIZE(rt286_snd_controls), + .dapm_widgets = rt286_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt286_dapm_widgets), + .dapm_routes = rt286_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt286_dapm_routes), +}; + +static const struct regmap_config rt286_regmap = { + .reg_bits = 32, + .val_bits = 32, + .max_register = 0x02370100, + .volatile_reg = rt286_volatile_register, + .readable_reg = rt286_readable_register, + .reg_write = rt286_hw_write, + .reg_read = rt286_hw_read, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt286_reg, + .num_reg_defaults = ARRAY_SIZE(rt286_reg), +}; + +static const struct i2c_device_id rt286_i2c_id[] = { + {"rt286", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, rt286_i2c_id); + +static const struct acpi_device_id rt286_acpi_match[] = { + { "INT343A", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt286_acpi_match); + +static int rt286_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt286_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt286_priv *rt286; + int ret; + + rt286 = devm_kzalloc(&i2c->dev, sizeof(*rt286), + GFP_KERNEL); + if (NULL == rt286) + return -ENOMEM; + + rt286->regmap = devm_regmap_init(&i2c->dev, NULL, i2c, &rt286_regmap); + if (IS_ERR(rt286->regmap)) { + ret = PTR_ERR(rt286->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + rt286->index_cache = rt286_index_def; + rt286->i2c = i2c; + i2c_set_clientdata(i2c, rt286); + + if (pdata) + rt286->pdata = *pdata; + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt286, + rt286_dai, ARRAY_SIZE(rt286_dai)); + + return ret; +} + +static int rt286_i2c_remove(struct i2c_client *i2c) +{ + struct rt286_priv *rt286 = i2c_get_clientdata(i2c); + + if (i2c->irq) + free_irq(i2c->irq, rt286); + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + + +struct i2c_driver rt286_i2c_driver = { + .driver = { + .name = "rt286", + .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(rt286_acpi_match), + }, + .probe = rt286_i2c_probe, + .remove = rt286_i2c_remove, + .id_table = rt286_i2c_id, +}; + +module_i2c_driver(rt286_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT286 driver"); +MODULE_AUTHOR("Bard Liao "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt286.h b/sound/soc/codecs/rt286.h new file mode 100644 index 000000000000..21c570f88e9b --- /dev/null +++ b/sound/soc/codecs/rt286.h @@ -0,0 +1,193 @@ +/* + * rt286.h -- RT286 ALSA SoC audio driver + * + * Copyright 2011 Realtek Microelectronics + * Author: Johnny Hsu + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __RT286_H__ +#define __RT286_H__ + +#define VERB_CMD(V, N, D) ((N << 20) | (V << 8) | D) + +#define RT286_AUDIO_FUNCTION_GROUP 0x01 +#define RT286_DAC_OUT1 0x02 +#define RT286_DAC_OUT2 0x03 +#define RT286_ADC_IN1 0x09 +#define RT286_ADC_IN2 0x08 +#define RT286_MIXER_IN 0x0b +#define RT286_MIXER_OUT1 0x0c +#define RT286_MIXER_OUT2 0x0d +#define RT286_DMIC1 0x12 +#define RT286_DMIC2 0x13 +#define RT286_SPK_OUT 0x14 +#define RT286_MIC1 0x18 +#define RT286_LINE1 0x1a +#define RT286_BEEP 0x1d +#define RT286_SPDIF 0x1e +#define RT286_VENDOR_REGISTERS 0x20 +#define RT286_HP_OUT 0x21 +#define RT286_MIXER_IN1 0x22 +#define RT286_MIXER_IN2 0x23 + +#define RT286_SET_PIN_SFT 6 +#define RT286_SET_PIN_ENABLE 0x40 +#define RT286_SET_PIN_DISABLE 0 +#define RT286_SET_EAPD_HIGH 0x2 +#define RT286_SET_EAPD_LOW 0 + +#define RT286_MUTE_SFT 7 + +/* Verb commands */ +#define RT286_GET_PARAM(NID, PARAM) VERB_CMD(AC_VERB_PARAMETERS, NID, PARAM) +#define RT286_SET_POWER(NID) VERB_CMD(AC_VERB_SET_POWER_STATE, NID, 0) +#define RT286_SET_AUDIO_POWER RT286_SET_POWER(RT286_AUDIO_FUNCTION_GROUP) +#define RT286_SET_HPO_POWER RT286_SET_POWER(RT286_HP_OUT) +#define RT286_SET_SPK_POWER RT286_SET_POWER(RT286_SPK_OUT) +#define RT286_SET_DMIC1_POWER RT286_SET_POWER(RT286_DMIC1) +#define RT286_SPK_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_SPK_OUT, 0) +#define RT286_HPO_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_HP_OUT, 0) +#define RT286_ADC0_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_MIXER_IN1, 0) +#define RT286_ADC1_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_MIXER_IN2, 0) +#define RT286_SET_MIC1\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_MIC1, 0) +#define RT286_SET_PIN_HPO\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_HP_OUT, 0) +#define RT286_SET_PIN_SPK\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_SPK_OUT, 0) +#define RT286_SET_PIN_DMIC1\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_DMIC1, 0) +#define RT286_SPK_EAPD\ + VERB_CMD(AC_VERB_SET_EAPD_BTLENABLE, RT286_SPK_OUT, 0) +#define RT286_SET_AMP_GAIN_HPO\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0) +#define RT286_SET_AMP_GAIN_ADC_IN1\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN1, 0) +#define RT286_SET_AMP_GAIN_ADC_IN2\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN2, 0) +#define RT286_GET_HP_SENSE\ + VERB_CMD(AC_VERB_GET_PIN_SENSE, RT286_HP_OUT, 0) +#define RT286_GET_MIC1_SENSE\ + VERB_CMD(AC_VERB_GET_PIN_SENSE, RT286_MIC1, 0) +#define RT286_SET_DMIC2_DEFAULT\ + VERB_CMD(AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, RT286_DMIC2, 0) +#define RT286_DACL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_DAC_OUT1, 0xa000) +#define RT286_DACR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_DAC_OUT1, 0x9000) +#define RT286_ADCL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN1, 0x6000) +#define RT286_ADCR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN1, 0x5000) +#define RT286_MIC_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIC1, 0x7000) +#define RT286_SPOL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_SPK_OUT, 0xa000) +#define RT286_SPOR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_SPK_OUT, 0x9000) +#define RT286_HPOL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0xa000) +#define RT286_HPOR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0x9000) +#define RT286_F_DAC_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_OUT1, 0x7000) +#define RT286_F_RECMIX_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_OUT1, 0x7100) +#define RT286_REC_MIC_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7000) +#define RT286_REC_I2S_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7100) +#define RT286_REC_LINE_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7200) +#define RT286_REC_BEEP_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7300) +#define RT286_DAC_FORMAT\ + VERB_CMD(AC_VERB_SET_STREAM_FORMAT, RT286_DAC_OUT1, 0) +#define RT286_ADC_FORMAT\ + VERB_CMD(AC_VERB_SET_STREAM_FORMAT, RT286_ADC_IN1, 0) +#define RT286_COEF_INDEX\ + VERB_CMD(AC_VERB_SET_COEF_INDEX, RT286_VENDOR_REGISTERS, 0) +#define RT286_PROC_COEF\ + VERB_CMD(AC_VERB_SET_PROC_COEF, RT286_VENDOR_REGISTERS, 0) + +/* Index registers */ +#define RT286_A_BIAS_CTRL1 0x01 +#define RT286_A_BIAS_CTRL2 0x02 +#define RT286_POWER_CTRL1 0x03 +#define RT286_A_BIAS_CTRL3 0x04 +#define RT286_POWER_CTRL2 0x08 +#define RT286_I2S_CTRL1 0x09 +#define RT286_I2S_CTRL2 0x0a +#define RT286_CLK_DIV 0x0b +#define RT286_POWER_CTRL3 0x0f +#define RT286_MIC1_DET_CTRL 0x19 +#define RT286_MISC_CTRL1 0x20 +#define RT286_IRQ_CTRL 0x33 +#define RT286_PLL_CTRL1 0x49 +#define RT286_CBJ_CTRL1 0x4f +#define RT286_CBJ_CTRL2 0x50 +#define RT286_PLL_CTRL 0x63 + +/* SPDIF (0x06) */ +#define RT286_SPDIF_SEL_SFT 0 +#define RT286_SPDIF_SEL_PCM0 0 +#define RT286_SPDIF_SEL_PCM1 1 +#define RT286_SPDIF_SEL_SPOUT 2 +#define RT286_SPDIF_SEL_PP 3 + +/* RECMIX (0x0b) */ +#define RT286_M_REC_BEEP_SFT 0 +#define RT286_M_REC_LINE1_SFT 1 +#define RT286_M_REC_MIC1_SFT 2 +#define RT286_M_REC_I2S_SFT 3 + +/* Front (0x0c) */ +#define RT286_M_FRONT_DAC_SFT 0 +#define RT286_M_FRONT_REC_SFT 1 + +/* SPK-OUT (0x14) */ +#define RT286_M_SPK_MUX_SFT 14 +#define RT286_SPK_SEL_MASK 0x1 +#define RT286_SPK_SEL_SFT 0 +#define RT286_SPK_SEL_F 0 +#define RT286_SPK_SEL_S 1 + +/* HP-OUT (0x21) */ +#define RT286_M_HP_MUX_SFT 14 +#define RT286_HP_SEL_MASK 0x1 +#define RT286_HP_SEL_SFT 0 +#define RT286_HP_SEL_F 0 +#define RT286_HP_SEL_S 1 + +/* ADC (0x22) (0x23) */ +#define RT286_ADC_SEL_MASK 0x7 +#define RT286_ADC_SEL_SFT 0 +#define RT286_ADC_SEL_SURR 0 +#define RT286_ADC_SEL_FRONT 1 +#define RT286_ADC_SEL_DMIC 2 +#define RT286_ADC_SEL_BEEP 4 +#define RT286_ADC_SEL_LINE1 5 +#define RT286_ADC_SEL_I2S 6 +#define RT286_ADC_SEL_MIC1 7 + +#define RT286_SCLK_S_MCLK 0 +#define RT286_SCLK_S_PLL 1 + +enum { + RT286_AIF1, + RT286_AIF2, + RT286_AIFS, +}; + +int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack); + +#endif /* __RT286_H__ */ + -- cgit v1.2.3 From 732814c8ffedd33a6b54f149b07ddec13c49fce9 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 3 Jul 2014 07:51:53 +0300 Subject: ALSA: pcm_dmaengine: Use the available wrapper to get physical width params_physical_width() is available via pcm_params.h Signed-off-by: Peter Ujfalusi Acked-by: Lars-Peter Clausen Acked-by: Vinod Koul Acked-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/core/pcm_dmaengine.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 76cbb9ec953a..d5611ec80381 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -65,7 +65,7 @@ int snd_hwparams_to_dma_slave_config(const struct snd_pcm_substream *substream, enum dma_slave_buswidth buswidth; int bits; - bits = snd_pcm_format_physical_width(params_format(params)); + bits = params_physical_width(params); if (bits < 8 || bits > 64) return -EINVAL; else if (bits == 8) -- cgit v1.2.3 From 2d38df12832186e592c1330dc5c363a778800cf1 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 3 Jul 2014 07:51:54 +0300 Subject: ASoC: generic-dmaengine-pcm: Prepare formats mask for valid physical sample sizes Based on the dma_slave_caps's addr_widths queried from the dma driver prepare the hw.formats mask to include only formats which is supported by the DMA engine. In case the dma driver does not implement the slave_caps the default assumption is that it supports 1, 2 and 4 bytes widths. Signed-off-by: Peter Ujfalusi Acked-by: Vinod Koul Acked-by: Takashi Iwai Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-generic-dmaengine-pcm.c | 37 ++++++++++++++++++++++++++++++++++- 1 file changed, 36 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 5bace124ef43..6307f85e871b 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -119,7 +119,10 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea struct snd_dmaengine_dai_dma_data *dma_data; struct dma_slave_caps dma_caps; struct snd_pcm_hardware hw; - int ret; + u32 addr_widths = BIT(DMA_SLAVE_BUSWIDTH_1_BYTE) | + BIT(DMA_SLAVE_BUSWIDTH_2_BYTES) | + BIT(DMA_SLAVE_BUSWIDTH_4_BYTES); + int i, ret; if (pcm->config && pcm->config->pcm_hardware) return snd_soc_set_runtime_hwparams(substream, @@ -146,6 +149,38 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea hw.info |= SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME; if (dma_caps.residue_granularity <= DMA_RESIDUE_GRANULARITY_SEGMENT) hw.info |= SNDRV_PCM_INFO_BATCH; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + addr_widths = dma_caps.dstn_addr_widths; + else + addr_widths = dma_caps.src_addr_widths; + } + + /* + * Prepare formats mask for valid/allowed sample types. If the dma does + * not have support for the given physical word size, it needs to be + * masked out so user space can not use the format which produces + * corrupted audio. + * In case the dma driver does not implement the slave_caps the default + * assumption is that it supports 1, 2 and 4 bytes widths. + */ + for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) { + int bits = snd_pcm_format_physical_width(i); + + /* Enable only samples with DMA supported physical widths */ + switch (bits) { + case 8: + case 16: + case 24: + case 32: + case 64: + if (addr_widths & (1 << (bits / 8))) + hw.formats |= (1LL << i); + break; + default: + /* Unsupported types */ + break; + } } return snd_soc_set_runtime_hwparams(substream, &hw); -- cgit v1.2.3 From 75f850fc229e090d3073708f55ca628db650b64e Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 3 Jul 2014 07:51:55 +0300 Subject: ALSA: pcm_dmaengine: Correct support for 24bits physical sample widths In case of _3LE/_3BE formats the samples are stored in 3 consecutive bytes without padding it to 4 bytes. This means that the DMA needs to be able to support 3 bytes word length in order to read/write the samples from memory correctly. Originally the code treated 24 bits physical length samples as they were 32 bits which leads to corruption when playing or recording audio. The hw.formats field has already been prepared to exclude formats not supported by the DMA engine in use, which means that only on platforms where 3 bytes is supported by the DMA will be able to use this format. Signed-off-by: Peter Ujfalusi Acked-by: Vinod Koul Acked-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/core/pcm_dmaengine.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index d5611ec80381..6542c4083594 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -72,6 +72,8 @@ int snd_hwparams_to_dma_slave_config(const struct snd_pcm_substream *substream, buswidth = DMA_SLAVE_BUSWIDTH_1_BYTE; else if (bits == 16) buswidth = DMA_SLAVE_BUSWIDTH_2_BYTES; + else if (bits == 24) + buswidth = DMA_SLAVE_BUSWIDTH_3_BYTES; else if (bits <= 32) buswidth = DMA_SLAVE_BUSWIDTH_4_BYTES; else -- cgit v1.2.3 From 001eaa25ca8cf4bfdb76fa8f2287b8d34287b9b6 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 30 Jun 2014 10:01:37 +0200 Subject: ASoC: s6000-i2s: Fix s6000_i2s_remove() return type The platform_driver remove callback return type is int not void. Fixes the following warning: sound/soc/s6000/s6000-i2s.c:604:19: warning: incorrect type in initializer (different base types) sound/soc/s6000/s6000-i2s.c:604:19: expected int ( *remove )( ... ) sound/soc/s6000/s6000-i2s.c:604:19: got void ( static [toplevel] *)( ... ) Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/s6000/s6000-i2s.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c index 7eba7979b9af..1c8d01166e5b 100644 --- a/sound/soc/s6000/s6000-i2s.c +++ b/sound/soc/s6000/s6000-i2s.c @@ -570,7 +570,7 @@ err_release_none: return ret; } -static void s6000_i2s_remove(struct platform_device *pdev) +static int s6000_i2s_remove(struct platform_device *pdev) { struct s6000_i2s_dev *dev = dev_get_drvdata(&pdev->dev); struct resource *region; @@ -597,6 +597,8 @@ static void s6000_i2s_remove(struct platform_device *pdev) iounmap(mmio); region = platform_get_resource(pdev, IORESOURCE_IO, 0); release_mem_region(region->start, resource_size(region)); + + return 0; } static struct platform_driver s6000_i2s_driver = { -- cgit v1.2.3 From b5dd60fd263147f723aeebc424dab7cd6613d3be Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 30 Jun 2014 10:01:38 +0200 Subject: ASoC: s6000: Allow to build when COMPILE_TEST is enabled Most of the ASoC s6000 code is architecture independent. This patch makes it possible to select the platform when COMPILE_TEST is enabled. The only architecture dependent code is the PCM driver which will still only be selected if XTENSA_VARIANT_S6000 is enabled. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/s6000/Kconfig | 10 ++++++++-- sound/soc/s6000/Makefile | 2 +- sound/soc/s6000/s6105-ipcam.c | 2 -- 3 files changed, 9 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/s6000/Kconfig b/sound/soc/s6000/Kconfig index c74eb3d4a47c..4823e1e4daef 100644 --- a/sound/soc/s6000/Kconfig +++ b/sound/soc/s6000/Kconfig @@ -1,17 +1,23 @@ config SND_S6000_SOC tristate "SoC Audio for the Stretch s6000 family" - depends on XTENSA_VARIANT_S6000 + depends on XTENSA_VARIANT_S6000 || COMPILE_TEST + depends on HAS_IOMEM + select SND_S6000_SOC_PCM if XTENSA_VARIANT_S6000 help Say Y or M if you want to add support for codecs attached to s6000 family chips. You will also need to select the platform to support below. +config SND_S6000_SOC_PCM + tristate + config SND_S6000_SOC_I2S tristate config SND_S6000_SOC_S6IPCAM tristate "SoC Audio support for Stretch 6105 IP Camera" - depends on SND_S6000_SOC && XTENSA_PLATFORM_S6105 + depends on SND_S6000_SOC + depends on XTENSA_PLATFORM_S6105 || COMPILE_TEST select SND_S6000_SOC_I2S select SND_SOC_TLV320AIC3X help diff --git a/sound/soc/s6000/Makefile b/sound/soc/s6000/Makefile index 7a613612e010..0f0ae2a012aa 100644 --- a/sound/soc/s6000/Makefile +++ b/sound/soc/s6000/Makefile @@ -2,7 +2,7 @@ snd-soc-s6000-objs := s6000-pcm.o snd-soc-s6000-i2s-objs := s6000-i2s.o -obj-$(CONFIG_SND_S6000_SOC) += snd-soc-s6000.o +obj-$(CONFIG_SND_S6000_SOC_PCM) += snd-soc-s6000.o obj-$(CONFIG_SND_S6000_SOC_I2S) += snd-soc-s6000-i2s.o # s6105 Machine Support diff --git a/sound/soc/s6000/s6105-ipcam.c b/sound/soc/s6000/s6105-ipcam.c index 0b21d1dc80c1..50875e66a0a3 100644 --- a/sound/soc/s6000/s6105-ipcam.c +++ b/sound/soc/s6000/s6105-ipcam.c @@ -19,8 +19,6 @@ #include #include -#include - #include "s6000-pcm.h" #include "s6000-i2s.h" -- cgit v1.2.3 From 13447cdc76a8c0a57c9b3395d896b01feb82f34b Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 30 Jun 2014 10:01:39 +0200 Subject: ASoC: s6105-ipcam: Automatically disconnect non-connected pins All CODEC input and output widgets are either in the DAPM routing table or manually marked as non-connected. This means the card is fully routed and we can let the core take care of disconnecting non-connected pins. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/s6000/s6105-ipcam.c | 15 +-------------- 1 file changed, 1 insertion(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/s6000/s6105-ipcam.c b/sound/soc/s6000/s6105-ipcam.c index 50875e66a0a3..3510c01f8a6a 100644 --- a/sound/soc/s6000/s6105-ipcam.c +++ b/sound/soc/s6000/s6105-ipcam.c @@ -133,22 +133,8 @@ static const struct snd_kcontrol_new audio_out_mux = { /* Logic for a aic3x as connected on the s6105 ip camera ref design */ static int s6105_aic3x_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; struct snd_soc_card *card = rtd->card; - /* not present */ - snd_soc_dapm_nc_pin(dapm, "MONO_LOUT"); - snd_soc_dapm_nc_pin(dapm, "LINE2L"); - snd_soc_dapm_nc_pin(dapm, "LINE2R"); - - /* not connected */ - snd_soc_dapm_nc_pin(dapm, "MIC3L"); /* LINE2L on this chip */ - snd_soc_dapm_nc_pin(dapm, "MIC3R"); /* LINE2R on this chip */ - snd_soc_dapm_nc_pin(dapm, "LLOUT"); - snd_soc_dapm_nc_pin(dapm, "RLOUT"); - snd_soc_dapm_nc_pin(dapm, "HPRCOM"); - /* must correspond to audio_out_mux.private_value initializer */ snd_soc_dapm_disable_pin(&card->dapm, "Audio Out Differential"); @@ -180,6 +166,7 @@ static struct snd_soc_card snd_soc_card_s6105 = { .num_dapm_widgets = ARRAY_SIZE(aic3x_dapm_widgets), .dapm_routes = audio_map, .num_dapm_routes = ARRAY_SIZE(audio_map), + .fully_routed = true, }; static struct s6000_snd_platform_data s6105_snd_data __initdata = { -- cgit v1.2.3 From 90eb1ab9e40c2d7596d9160151f6df4b5a71645f Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Fri, 4 Jul 2014 17:07:41 +0200 Subject: ASoC: sirf: provide pm-runtime functions when needed The newly introduced sirf-usp driver defines sirf_usp_pcm_{suspend,resume} functions only when PM_RUNTIME is enabled, but also uses them when that is disabled and only PM_SLEEP is turned on, resulting in this error: ../sound/soc/sirf/sirf-usp.c: In function 'sirf_usp_pcm_suspend': ../sound/soc/sirf/sirf-usp.c:308:3: error: implicit declaration of function 'sirf_usp_pcm_runtime_suspend' [-Werror=implicit-function-declaration] sirf_usp_pcm_runtime_suspend(dev); ^ ../sound/soc/sirf/sirf-usp.c: In function 'sirf_usp_pcm_resume': ../sound/soc/sirf/sirf-usp.c:319:3: error: implicit declaration of function 'sirf_usp_pcm_runtime_resume' [-Werror=implicit-function-declaration] ret = sirf_usp_pcm_runtime_resume(dev); ^ cc1: some warnings being treated as errors To fix that, this patch changes the #ifdef to CONFIG_PM, which is enabled when at least one of PM_SLEEP or PM_RUNTIME are enabled. Signed-off-by: Arnd Bergmann Signed-off-by: Mark Brown --- sound/soc/sirf/sirf-usp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sirf/sirf-usp.c b/sound/soc/sirf/sirf-usp.c index bdf6aae3e6d0..9693bc2a796d 100644 --- a/sound/soc/sirf/sirf-usp.c +++ b/sound/soc/sirf/sirf-usp.c @@ -282,7 +282,7 @@ static struct snd_soc_dai_driver sirf_usp_pcm_dai = { .ops = &sirf_usp_pcm_dai_ops, }; -#ifdef CONFIG_PM_RUNTIME +#ifdef CONFIG_PM static int sirf_usp_pcm_runtime_suspend(struct device *dev) { struct sirf_usp *usp = dev_get_drvdata(dev); -- cgit v1.2.3 From d3d4e5247b013008a39e4d5f69ce4c60ed57f997 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Fri, 4 Jul 2014 16:05:45 +0200 Subject: ASoC: samsung: Correct I2S DAI suspend/resume ops We should save/restore relevant I2S registers regardless of the dai->active flag, otherwise some settings are being lost after system suspend/resume cycle. E.g. I2S slave mode set only during dai initialization is not preserved and the device ends up in master mode after system resume. Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/samsung/i2s.c | 16 ++++++---------- 1 file changed, 6 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 2ac76fa3e742..5f9b255a8b38 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -920,11 +920,9 @@ static int i2s_suspend(struct snd_soc_dai *dai) { struct i2s_dai *i2s = to_info(dai); - if (dai->active) { - i2s->suspend_i2smod = readl(i2s->addr + I2SMOD); - i2s->suspend_i2scon = readl(i2s->addr + I2SCON); - i2s->suspend_i2spsr = readl(i2s->addr + I2SPSR); - } + i2s->suspend_i2smod = readl(i2s->addr + I2SMOD); + i2s->suspend_i2scon = readl(i2s->addr + I2SCON); + i2s->suspend_i2spsr = readl(i2s->addr + I2SPSR); return 0; } @@ -933,11 +931,9 @@ static int i2s_resume(struct snd_soc_dai *dai) { struct i2s_dai *i2s = to_info(dai); - if (dai->active) { - writel(i2s->suspend_i2scon, i2s->addr + I2SCON); - writel(i2s->suspend_i2smod, i2s->addr + I2SMOD); - writel(i2s->suspend_i2spsr, i2s->addr + I2SPSR); - } + writel(i2s->suspend_i2scon, i2s->addr + I2SCON); + writel(i2s->suspend_i2smod, i2s->addr + I2SMOD); + writel(i2s->suspend_i2spsr, i2s->addr + I2SPSR); return 0; } -- cgit v1.2.3 From 978b641f9563019a24032d5dee8a75963cd248ff Mon Sep 17 00:00:00 2001 From: Tushar Behera Date: Fri, 4 Jul 2014 14:42:16 +0530 Subject: ASoC: max98090: Add check for CODEC type CODEC type (MAX98090/MAX98091) can be specified from device-tree file, it can also be obtained from the CODEC during runtime. Add an explicit check to figure out if both are matching, else print a message warning about the same. Signed-off-by: Tushar Behera Signed-off-by: Mark Brown --- sound/soc/codecs/max98090.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index c00b36872dfe..2c2c5b22f60f 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2215,6 +2215,7 @@ static int max98090_probe(struct snd_soc_codec *codec) { struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); struct max98090_cdata *cdata; + enum max98090_type devtype; int ret = 0; dev_dbg(codec->dev, "max98090_probe\n"); @@ -2250,16 +2251,21 @@ static int max98090_probe(struct snd_soc_codec *codec) } if ((ret >= M98090_REVA) && (ret <= M98090_REVA + 0x0f)) { - max98090->devtype = MAX98090; + devtype = MAX98090; dev_info(codec->dev, "MAX98090 REVID=0x%02x\n", ret); } else if ((ret >= M98091_REVA) && (ret <= M98091_REVA + 0x0f)) { - max98090->devtype = MAX98091; + devtype = MAX98091; dev_info(codec->dev, "MAX98091 REVID=0x%02x\n", ret); } else { - max98090->devtype = MAX98090; + devtype = MAX98090; dev_err(codec->dev, "Unrecognized revision 0x%02x\n", ret); } + if (max98090->devtype != devtype) { + dev_warn(codec->dev, "Mismatch in DT specified CODEC type.\n"); + max98090->devtype = devtype; + } + max98090->jack_state = M98090_JACK_STATE_NO_HEADSET; INIT_DELAYED_WORK(&max98090->jack_work, max98090_jack_work); -- cgit v1.2.3 From eba843201a8e5824c5e6e539db6cd1a6ba84f145 Mon Sep 17 00:00:00 2001 From: Tushar Behera Date: Fri, 4 Jul 2014 14:42:17 +0530 Subject: ASoC: max98090: Remove redundant max98090_handle_pdata() max98090_handle_pdata() is not doing anything other than printing a message if pdata is not valid. This can be removed. Signed-off-by: Tushar Behera Signed-off-by: Mark Brown --- sound/soc/codecs/max98090.c | 13 ------------- 1 file changed, 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 2c2c5b22f60f..0e59e5117e43 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2200,17 +2200,6 @@ static struct snd_soc_dai_driver max98090_dai[] = { } }; -static void max98090_handle_pdata(struct snd_soc_codec *codec) -{ - struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); - struct max98090_pdata *pdata = max98090->pdata; - - if (!pdata) { - dev_err(codec->dev, "No platform data\n"); - return; - } -} - static int max98090_probe(struct snd_soc_codec *codec) { struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); @@ -2310,8 +2299,6 @@ static int max98090_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, M98090_REG_MIC_BIAS_VOLTAGE, M98090_MBVSEL_MASK, M98090_MBVSEL_2V8); - max98090_handle_pdata(codec); - max98090_add_widgets(codec); err_access: -- cgit v1.2.3 From 00ad93e26375d974801886070738d0c7cf187fdf Mon Sep 17 00:00:00 2001 From: Tushar Behera Date: Fri, 4 Jul 2014 14:22:59 +0530 Subject: ASoC: samsung: Make card name for Snow configurable Snow sound-card driver supports multiple boards with different audio codecs. Updating the sound card name per board basis would provide some more information to the end-user. Signed-off-by: Tushar Behera Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/snow.txt | 4 ++++ sound/soc/samsung/snow.c | 3 +++ 2 files changed, 7 insertions(+) (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/snow.txt b/Documentation/devicetree/bindings/sound/snow.txt index e0b7a8207f5f..6df74f15687f 100644 --- a/Documentation/devicetree/bindings/sound/snow.txt +++ b/Documentation/devicetree/bindings/sound/snow.txt @@ -8,11 +8,15 @@ Required properties: - samsung,i2s-controller: The phandle of the Samsung I2S controller - samsung,audio-codec: The phandle of the audio codec +Optional: +- samsung,model: The name of the sound-card + Example: sound { compatible = "google,snow-audio-max98095"; + samsung,model = "Snow-I2S-MAX98095"; samsung,i2s-controller = <&i2s0>; samsung,audio-codec = <&max98095>; }; diff --git a/sound/soc/samsung/snow.c b/sound/soc/samsung/snow.c index 8bbd348358dd..0acf5d0eed53 100644 --- a/sound/soc/samsung/snow.c +++ b/sound/soc/samsung/snow.c @@ -92,6 +92,9 @@ static int snow_probe(struct platform_device *pdev) card->dev = &pdev->dev; + /* Update card-name if provided through DT, else use default name */ + snd_soc_of_parse_card_name(card, "samsung,model"); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); -- cgit v1.2.3 From 1bb60b3a5ae526287fc6d0c55391f5664cc1e672 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 5 Jul 2014 17:54:24 +0100 Subject: ASoC: s6105-ipcam: Fix build for I2C dependencies The s6105-ipcam driver calls i2c_register_board_info() which means that not only does it need I2C to be enabled to be built but since that symbol is not exported it means that the driver also can't be linked as a module. Use a bool instead of a tristate and add a dependency on I2C to fix these issues. Commit b5dd60fd263 (ASoC: s6000: Allow to build when COMPILE_TEST is enabled) made this problem more visible since it enabled build of the driver on more common architectures but this has always been an issue. Signed-off-by: Mark Brown --- sound/soc/s6000/Kconfig | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/s6000/Kconfig b/sound/soc/s6000/Kconfig index 4823e1e4daef..f244a2566f20 100644 --- a/sound/soc/s6000/Kconfig +++ b/sound/soc/s6000/Kconfig @@ -15,8 +15,9 @@ config SND_S6000_SOC_I2S tristate config SND_S6000_SOC_S6IPCAM - tristate "SoC Audio support for Stretch 6105 IP Camera" - depends on SND_S6000_SOC + bool "SoC Audio support for Stretch 6105 IP Camera" + depends on SND_S6000_SOC=y + depends on I2C=y depends on XTENSA_PLATFORM_S6105 || COMPILE_TEST select SND_S6000_SOC_I2S select SND_SOC_TLV320AIC3X -- cgit v1.2.3 From cc6b0ae96239ac77fac569e871704a4f1bf3c595 Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Mon, 7 Jul 2014 15:36:59 +0800 Subject: ASoC: rt5677: Remove the redundant definition in head file The patch removes the redundant definition in head file Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.h | 7 ------- 1 file changed, 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index 08252e7c1ab4..863393e62096 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1393,13 +1393,6 @@ #define RT5677_DSP_IB_9_L (0x1 << 1) #define RT5677_DSP_IB_9_L_SFT 1 -/* Debug String Length */ -#define RT5677_REG_DISP_LEN 23 - -#define RT5677_NO_JACK BIT(0) -#define RT5677_HEADSET_DET BIT(1) -#define RT5677_HEADPHO_DET BIT(2) - /* System Clock Source */ enum { RT5677_SCLK_S_MCLK, -- cgit v1.2.3 From f18803a3758a55feb376490a76c351ab40c3cf03 Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Mon, 7 Jul 2014 15:37:00 +0800 Subject: ASoC: rt5677: Modify the voltage level in the BIAS OFF stage The patch modifies the voltage level in the BIAS OFF stage Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 98370ae3b38d..67f14556462f 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -3124,7 +3124,7 @@ static int rt5677_set_bias_level(struct snd_soc_codec *codec, regmap_update_bits(rt5677->regmap, RT5677_DIG_MISC, 0x1, 0x0); regmap_write(rt5677->regmap, RT5677_PWR_DIG1, 0x0000); regmap_write(rt5677->regmap, RT5677_PWR_DIG2, 0x0000); - regmap_write(rt5677->regmap, RT5677_PWR_ANLG1, 0x0000); + regmap_write(rt5677->regmap, RT5677_PWR_ANLG1, 0x0022); regmap_write(rt5677->regmap, RT5677_PWR_ANLG2, 0x0000); regmap_update_bits(rt5677->regmap, RT5677_PR_BASE + RT5677_BIAS_CUR4, 0x0f00, 0x0000); -- cgit v1.2.3 From 305b8d8782c3b4aa572d496769b93cc3db2ae892 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Mon, 7 Jul 2014 16:48:36 +0800 Subject: ASoC: RT286: remove test code Remove test code. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt286.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index acfba9c74c52..7c5f9d0f0af2 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -993,7 +993,6 @@ static int rt286_probe(struct snd_soc_codec *codec) codec->dapm.bias_level = SND_SOC_BIAS_OFF; rt286->codec = codec; - rt286->i2c->irq = 0; if (rt286->i2c->irq) { snd_soc_update_bits(codec, RT286_IRQ_CTRL, 0x2, 0x2); -- cgit v1.2.3 From 4b21768a95d68fe26a6a9f08ca93a7c59c13fcac Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Mon, 7 Jul 2014 16:48:37 +0800 Subject: ASoC: RT286: check ID in i2c level Move ID check from asoc level to i2c level. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt286.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 7c5f9d0f0af2..53eb7f37bb73 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -952,14 +952,6 @@ static int rt286_probe(struct snd_soc_codec *codec) struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); int i, ret; - ret = snd_soc_read(codec, - RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID)); - if (ret != RT286_VENDOR_ID) { - dev_err(codec->dev, - "Device with ID register %x is not rt286\n", ret); - return -ENODEV; - } - snd_soc_write(codec, RT286_SET_AUDIO_POWER, AC_PWRST_D3); for (i = 0; i < RT286_POWER_REG_LEN; i++) @@ -1164,6 +1156,14 @@ static int rt286_i2c_probe(struct i2c_client *i2c, return ret; } + regmap_read(rt286->regmap, + RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &ret); + if (ret != RT286_VENDOR_ID) { + dev_err(&i2c->dev, + "Device with ID register %x is not rt286\n", ret); + return -ENODEV; + } + rt286->index_cache = rt286_index_def; rt286->i2c = i2c; i2c_set_clientdata(i2c, rt286); -- cgit v1.2.3 From 61a414c412886bdb98c8842c00b2f0a3d4436b12 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Mon, 7 Jul 2014 16:48:38 +0800 Subject: ASoC: RT286: move initial settings to _i2c_probe Move codec initial settings from asoc probe to i2c probe. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt286.c | 99 ++++++++++++++++++++++++------------------------ 1 file changed, 49 insertions(+), 50 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 53eb7f37bb73..e6f33ab78954 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -950,59 +950,10 @@ static irqreturn_t rt286_irq(int irq, void *data) static int rt286_probe(struct snd_soc_codec *codec) { struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); - int i, ret; - - snd_soc_write(codec, RT286_SET_AUDIO_POWER, AC_PWRST_D3); - - for (i = 0; i < RT286_POWER_REG_LEN; i++) - snd_soc_write(codec, - RT286_SET_POWER(rt286_support_power_controls[i]), - AC_PWRST_D1); - - if (!rt286->pdata.cbj_en) { - snd_soc_write(codec, RT286_CBJ_CTRL2, 0x0000); - snd_soc_write(codec, RT286_MIC1_DET_CTRL, 0x0816); - snd_soc_write(codec, RT286_MISC_CTRL1, 0x0000); - snd_soc_update_bits(codec, - RT286_CBJ_CTRL1, 0xf000, 0xb000); - } else { - snd_soc_update_bits(codec, - RT286_CBJ_CTRL1, 0xf000, 0x5000); - } - - mdelay(10); - - if (!rt286->pdata.gpio2_en) - snd_soc_write(codec, RT286_SET_DMIC2_DEFAULT, 0x4000); - else - snd_soc_write(codec, RT286_SET_DMIC2_DEFAULT, 0); - - mdelay(10); - - /*Power down LDO2*/ - snd_soc_update_bits(codec, RT286_POWER_CTRL2, 0x8, 0x0); codec->dapm.bias_level = SND_SOC_BIAS_OFF; rt286->codec = codec; - if (rt286->i2c->irq) { - snd_soc_update_bits(codec, - RT286_IRQ_CTRL, 0x2, 0x2); - - INIT_DELAYED_WORK(&rt286->jack_detect_work, - rt286_jack_detect_work); - schedule_delayed_work(&rt286->jack_detect_work, - msecs_to_jiffies(1250)); - - ret = request_threaded_irq(rt286->i2c->irq, NULL, rt286_irq, - IRQF_TRIGGER_HIGH | IRQF_ONESHOT, "rt286", rt286); - if (ret != 0) { - dev_err(codec->dev, - "Failed to reguest IRQ: %d\n", ret); - return ret; - } - } - return 0; } @@ -1141,7 +1092,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, { struct rt286_platform_data *pdata = dev_get_platdata(&i2c->dev); struct rt286_priv *rt286; - int ret; + int i, ret; rt286 = devm_kzalloc(&i2c->dev, sizeof(*rt286), GFP_KERNEL); @@ -1171,6 +1122,54 @@ static int rt286_i2c_probe(struct i2c_client *i2c, if (pdata) rt286->pdata = *pdata; + regmap_write(rt286->regmap, RT286_SET_AUDIO_POWER, AC_PWRST_D3); + + for (i = 0; i < RT286_POWER_REG_LEN; i++) + regmap_write(rt286->regmap, + RT286_SET_POWER(rt286_support_power_controls[i]), + AC_PWRST_D1); + + if (!rt286->pdata.cbj_en) { + regmap_write(rt286->regmap, RT286_CBJ_CTRL2, 0x0000); + regmap_write(rt286->regmap, RT286_MIC1_DET_CTRL, 0x0816); + regmap_write(rt286->regmap, RT286_MISC_CTRL1, 0x0000); + regmap_update_bits(rt286->regmap, + RT286_CBJ_CTRL1, 0xf000, 0xb000); + } else { + regmap_update_bits(rt286->regmap, + RT286_CBJ_CTRL1, 0xf000, 0x5000); + } + + mdelay(10); + + if (!rt286->pdata.gpio2_en) + regmap_write(rt286->regmap, RT286_SET_DMIC2_DEFAULT, 0x4000); + else + regmap_write(rt286->regmap, RT286_SET_DMIC2_DEFAULT, 0); + + mdelay(10); + + /*Power down LDO2*/ + regmap_update_bits(rt286->regmap, RT286_POWER_CTRL2, 0x8, 0x0); + + if (rt286->i2c->irq) { + regmap_update_bits(rt286->regmap, + RT286_IRQ_CTRL, 0x2, 0x2); + + INIT_DELAYED_WORK(&rt286->jack_detect_work, + rt286_jack_detect_work); + schedule_delayed_work(&rt286->jack_detect_work, + msecs_to_jiffies(1250)); + + ret = request_threaded_irq(rt286->i2c->irq, NULL, rt286_irq, + IRQF_TRIGGER_HIGH | IRQF_ONESHOT, "rt286", rt286); + if (ret != 0) { + dev_err(&i2c->dev, + "Failed to reguest IRQ: %d\n", ret); + return ret; + } + } + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt286, rt286_dai, ARRAY_SIZE(rt286_dai)); -- cgit v1.2.3 From bc6c4e455af9037bae619340bc95bf569806ba8b Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Mon, 7 Jul 2014 19:15:30 +0800 Subject: ASoC: RT286: Fix silent at the beginning of stream This patch fix the issue that the output is almost silent at the beginning of starting a playback. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt286.c | 20 +++++++++++++++++++- sound/soc/codecs/rt286.h | 5 +++++ 2 files changed, 24 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index e6f33ab78954..81033154a412 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -54,6 +54,7 @@ static struct reg_default rt286_index_def[] = { { 0x09, 0xd810 }, { 0x0a, 0x0060 }, { 0x0b, 0x0000 }, + { 0x0d, 0x2800 }, { 0x0f, 0x0000 }, { 0x19, 0x0a17 }, { 0x20, 0x0020 }, @@ -62,6 +63,9 @@ static struct reg_default rt286_index_def[] = { { 0x4f, 0x50e9 }, { 0x50, 0x2c00 }, { 0x63, 0x2902 }, + { 0x67, 0x1111 }, + { 0x68, 0x1016 }, + { 0x69, 0x273f }, }; #define INDEX_CACHE_SIZE ARRAY_SIZE(rt286_index_def) @@ -902,14 +906,23 @@ static int rt286_set_bias_level(struct snd_soc_codec *codec, { switch (level) { case SND_SOC_BIAS_PREPARE: - if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) + if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) { snd_soc_write(codec, RT286_SET_AUDIO_POWER, AC_PWRST_D0); + snd_soc_update_bits(codec, + RT286_DC_GAIN, 0x200, 0x200); + } + break; + + case SND_SOC_BIAS_ON: + mdelay(10); break; case SND_SOC_BIAS_STANDBY: snd_soc_write(codec, RT286_SET_AUDIO_POWER, AC_PWRST_D3); + snd_soc_update_bits(codec, + RT286_DC_GAIN, 0x200, 0x0); break; default: @@ -1152,6 +1165,11 @@ static int rt286_i2c_probe(struct i2c_client *i2c, /*Power down LDO2*/ regmap_update_bits(rt286->regmap, RT286_POWER_CTRL2, 0x8, 0x0); + /*Set depop parameter*/ + regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL2, 0x403a, 0x401a); + regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL3, 0xf777, 0x4737); + regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL4, 0x00ff, 0x003f); + if (rt286->i2c->irq) { regmap_update_bits(rt286->regmap, RT286_IRQ_CTRL, 0x2, 0x2); diff --git a/sound/soc/codecs/rt286.h b/sound/soc/codecs/rt286.h index 21c570f88e9b..b539b7320a79 100644 --- a/sound/soc/codecs/rt286.h +++ b/sound/soc/codecs/rt286.h @@ -127,6 +127,7 @@ #define RT286_I2S_CTRL1 0x09 #define RT286_I2S_CTRL2 0x0a #define RT286_CLK_DIV 0x0b +#define RT286_DC_GAIN 0x0d #define RT286_POWER_CTRL3 0x0f #define RT286_MIC1_DET_CTRL 0x19 #define RT286_MISC_CTRL1 0x20 @@ -135,6 +136,10 @@ #define RT286_CBJ_CTRL1 0x4f #define RT286_CBJ_CTRL2 0x50 #define RT286_PLL_CTRL 0x63 +#define RT286_DEPOP_CTRL1 0x66 +#define RT286_DEPOP_CTRL2 0x67 +#define RT286_DEPOP_CTRL3 0x68 +#define RT286_DEPOP_CTRL4 0x69 /* SPDIF (0x06) */ #define RT286_SPDIF_SEL_SFT 0 -- cgit v1.2.3 From 121eb444135c25701051eb849e7ccf0dd412382b Mon Sep 17 00:00:00 2001 From: Thierry Reding Date: Mon, 7 Jul 2014 15:18:19 +0200 Subject: ASoC: max98090: Fix build warning The max98090_{suspend,resume}() functions are used for system sleep and therefore need to be guarded by CONFIG_PM_SLEEP rather than CONFIG_PM. Signed-off-by: Thierry Reding Signed-off-by: Mark Brown --- sound/soc/codecs/max98090.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 0e59e5117e43..6816578ea8f7 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2408,7 +2408,7 @@ static int max98090_runtime_suspend(struct device *dev) } #endif -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int max98090_resume(struct device *dev) { struct max98090_priv *max98090 = dev_get_drvdata(dev); -- cgit v1.2.3 From c4324bfa54d2a59b4920239c1b10b3e5f3a851d3 Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Tue, 8 Jul 2014 09:56:35 -0500 Subject: ASoC: cs42l56: Move ADC/PCM Swap to DAPM The Swap controls for ADC/PCM paths should be in the DAPM domain. Signed-off-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l56.c | 64 ++++++++++++++++++++++++++++++++-------------- 1 file changed, 45 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index fdc4bd27b0df..e5ef223be86c 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -318,24 +318,32 @@ static const struct soc_enum adca_swap_enum = ARRAY_SIZE(left_swap_text), left_swap_text, swap_values); +static const struct snd_kcontrol_new adca_swap_mux = + SOC_DAPM_ENUM("Route", adca_swap_enum); static const struct soc_enum pcma_swap_enum = SOC_VALUE_ENUM_SINGLE(CS42L56_CHAN_MIX_SWAP, 4, 3, ARRAY_SIZE(left_swap_text), left_swap_text, swap_values); +static const struct snd_kcontrol_new pcma_swap_mux = + SOC_DAPM_ENUM("Route", pcma_swap_enum); static const struct soc_enum adcb_swap_enum = SOC_VALUE_ENUM_SINGLE(CS42L56_CHAN_MIX_SWAP, 2, 3, ARRAY_SIZE(right_swap_text), right_swap_text, swap_values); +static const struct snd_kcontrol_new adcb_swap_mux = + SOC_DAPM_ENUM("Route", adcb_swap_enum); static const struct soc_enum pcmb_swap_enum = SOC_VALUE_ENUM_SINGLE(CS42L56_CHAN_MIX_SWAP, 6, 3, ARRAY_SIZE(right_swap_text), right_swap_text, swap_values); +static const struct snd_kcontrol_new pcmb_swap_mux = + SOC_DAPM_ENUM("Route", pcmb_swap_enum); static const struct snd_kcontrol_new hpa_switch = SOC_DAPM_SINGLE("Switch", CS42L56_PWRCTL_2, 6, 1, 1); @@ -467,11 +475,6 @@ static const struct snd_kcontrol_new cs42l56_snd_controls[] = { SOC_SINGLE("ADCA Invert", CS42L56_MISC_ADC_CTL, 2, 1, 1), SOC_SINGLE("ADCB Invert", CS42L56_MISC_ADC_CTL, 3, 1, 1), - SOC_ENUM("PCMA Swap", pcma_swap_enum), - SOC_ENUM("PCMB Swap", pcmb_swap_enum), - SOC_ENUM("ADCA Swap", adca_swap_enum), - SOC_ENUM("ADCB Swap", adcb_swap_enum), - SOC_DOUBLE("HPF Switch", CS42L56_HPF_CTL, 5, 7, 1, 1), SOC_DOUBLE("HPF Freeze Switch", CS42L56_HPF_CTL, 4, 6, 1, 1), SOC_ENUM("HPFA Corner Freq", hpfa_freq_enum), @@ -570,6 +573,16 @@ static const struct snd_soc_dapm_widget cs42l56_dapm_widgets[] = { SND_SOC_DAPM_ADC("ADCA", NULL, CS42L56_PWRCTL_1, 1, 1), SND_SOC_DAPM_ADC("ADCB", NULL, CS42L56_PWRCTL_1, 2, 1), + SND_SOC_DAPM_MUX("ADCA Swap Mux", SND_SOC_NOPM, 0, 0, + &adca_swap_mux), + SND_SOC_DAPM_MUX("ADCB Swap Mux", SND_SOC_NOPM, 0, 0, + &adcb_swap_mux), + + SND_SOC_DAPM_MUX("PCMA Swap Mux", SND_SOC_NOPM, 0, 0, + &pcma_swap_mux), + SND_SOC_DAPM_MUX("PCMB Swap Mux", SND_SOC_NOPM, 0, 0, + &pcmb_swap_mux), + SND_SOC_DAPM_DAC("DACA", NULL, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC("DACB", NULL, SND_SOC_NOPM, 0, 0), @@ -607,8 +620,19 @@ static const struct snd_soc_dapm_route cs42l56_audio_map[] = { {"Digital Output Mux", NULL, "ADCA"}, {"Digital Output Mux", NULL, "ADCB"}, - {"ADCB", NULL, "ADCB Mux"}, - {"ADCA", NULL, "ADCA Mux"}, + {"ADCB", NULL, "ADCB Swap Mux"}, + {"ADCA", NULL, "ADCA Swap Mux"}, + + {"ADCA Swap Mux", NULL, "ADCA"}, + {"ADCB Swap Mux", NULL, "ADCB"}, + + {"DACA", "Left", "ADCA Swap Mux"}, + {"DACA", "LR 2", "ADCA Swap Mux"}, + {"DACA", "Right", "ADCA Swap Mux"}, + + {"DACB", "Left", "ADCB Swap Mux"}, + {"DACB", "LR 2", "ADCB Swap Mux"}, + {"DACB", "Right", "ADCB Swap Mux"}, {"ADCA Mux", NULL, "AIN3A"}, {"ADCA Mux", NULL, "AIN2A"}, @@ -633,30 +657,32 @@ static const struct snd_soc_dapm_route cs42l56_audio_map[] = { {"PGAB Input Mux", NULL, "AIN2B"}, {"PGAB Input Mux", NULL, "AIN3B"}, - {"LOB", NULL, "Lineout Right"}, - {"LOA", NULL, "Lineout Left"}, - - {"Lineout Right", "Switch", "LINEOUTB Input Mux"}, - {"Lineout Left", "Switch", "LINEOUTA Input Mux"}, + {"LOB", "Switch", "LINEOUTB Input Mux"}, + {"LOA", "Switch", "LINEOUTA Input Mux"}, {"LINEOUTA Input Mux", "PGAA", "PGAA"}, {"LINEOUTB Input Mux", "PGAB", "PGAB"}, {"LINEOUTA Input Mux", "DACA", "DACA"}, {"LINEOUTB Input Mux", "DACB", "DACB"}, - {"HPA", NULL, "Headphone Left"}, - {"HPB", NULL, "Headphone Right"}, - - {"Headphone Right", "Switch", "HPB Input Mux"}, - {"Headphone Left", "Switch", "HPA Input Mux"}, + {"HPA", "Switch", "HPB Input Mux"}, + {"HPB", "Switch", "HPA Input Mux"}, {"HPA Input Mux", "PGAA", "PGAA"}, {"HPB Input Mux", "PGAB", "PGAB"}, {"HPA Input Mux", "DACA", "DACA"}, {"HPB Input Mux", "DACB", "DACB"}, - {"DACB", NULL, "HiFi Playback"}, - {"DACA", NULL, "HiFi Playback"}, + {"DACA", NULL, "PCMA Swap Mux"}, + {"DACB", NULL, "PCMB Swap Mux"}, + + {"PCMB Swap Mux", "Left", "HiFi Playback"}, + {"PCMB Swap Mux", "LR 2", "HiFi Playback"}, + {"PCMB Swap Mux", "Right", "HiFi Playback"}, + + {"PCMA Swap Mux", "Left", "HiFi Playback"}, + {"PCMA Swap Mux", "LR 2", "HiFi Playback"}, + {"PCMA Swap Mux", "Right", "HiFi Playback"}, }; -- cgit v1.2.3 From 4495c89fcf2624d542a27f1ecd70aa3524c54195 Mon Sep 17 00:00:00 2001 From: Jianqun Xu Date: Sat, 5 Jul 2014 19:13:03 +0800 Subject: ASoC: add driver for Rockchip RK3xxx I2S controller Add driver for i2s controller found on rk3066, rk3168 and rk3288 processors from rockchip. Tested on the RK3288 SDK board. Signed-off-by: Jianqun Xu Signed-off-by: Mark Brown --- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/rockchip/Kconfig | 12 + sound/soc/rockchip/Makefile | 4 + sound/soc/rockchip/rockchip_i2s.c | 530 ++++++++++++++++++++++++++++++++++++++ sound/soc/rockchip/rockchip_i2s.h | 223 ++++++++++++++++ 6 files changed, 771 insertions(+) create mode 100644 sound/soc/rockchip/Kconfig create mode 100644 sound/soc/rockchip/Makefile create mode 100644 sound/soc/rockchip/rockchip_i2s.c create mode 100644 sound/soc/rockchip/rockchip_i2s.h (limited to 'sound') diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 0060b31cc3f3..0e9623368ab0 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -47,6 +47,7 @@ source "sound/soc/kirkwood/Kconfig" source "sound/soc/intel/Kconfig" source "sound/soc/mxs/Kconfig" source "sound/soc/pxa/Kconfig" +source "sound/soc/rockchip/Kconfig" source "sound/soc/samsung/Kconfig" source "sound/soc/s6000/Kconfig" source "sound/soc/sh/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 5f1df02984f8..534714a1ca44 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -24,6 +24,7 @@ obj-$(CONFIG_SND_SOC) += nuc900/ obj-$(CONFIG_SND_SOC) += omap/ obj-$(CONFIG_SND_SOC) += kirkwood/ obj-$(CONFIG_SND_SOC) += pxa/ +obj-$(CONFIG_SND_SOC) += rockchip/ obj-$(CONFIG_SND_SOC) += samsung/ obj-$(CONFIG_SND_SOC) += s6000/ obj-$(CONFIG_SND_SOC) += sh/ diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig new file mode 100644 index 000000000000..c196a466eef6 --- /dev/null +++ b/sound/soc/rockchip/Kconfig @@ -0,0 +1,12 @@ +config SND_SOC_ROCKCHIP + tristate "ASoC support for Rockchip" + depends on COMPILE_TEST || ARCH_ROCKCHIP + select SND_SOC_GENERIC_DMAENGINE_PCM + select SND_ROCKCHIP_I2S + help + Say Y or M if you want to add support for codecs attached to + the Rockchip SoCs' Audio interfaces. You will also need to + select the audio interfaces to support below. + +config SND_ROCKCHIP_I2S + tristate diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile new file mode 100644 index 000000000000..1006418e1394 --- /dev/null +++ b/sound/soc/rockchip/Makefile @@ -0,0 +1,4 @@ +# ROCKCHIP Platform Support +snd-soc-i2s-objs := rockchip_i2s.o + +obj-$(CONFIG_SND_ROCKCHIP_I2S) += snd-soc-i2s.o diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c new file mode 100644 index 000000000000..af1d8eb94f04 --- /dev/null +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -0,0 +1,530 @@ +/* sound/soc/rockchip/rockchip_i2s.c + * + * ALSA SoC Audio Layer - Rockchip I2S Controller driver + * + * Copyright (c) 2014 Rockchip Electronics Co. Ltd. + * Author: Jianqun + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include + +#include "rockchip_i2s.h" + +#define DRV_NAME "rockchip-i2s" + +struct rk_i2s_dev { + struct device *dev; + + struct clk *hclk; + struct clk *mclk; + + struct snd_dmaengine_dai_dma_data capture_dma_data; + struct snd_dmaengine_dai_dma_data playback_dma_data; + + struct regmap *regmap; + +/* + * Used to indicate the tx/rx status. + * I2S controller hopes to start the tx and rx together, + * also to stop them when they are both try to stop. +*/ + bool tx_start; + bool rx_start; +}; + +static int i2s_runtime_suspend(struct device *dev) +{ + struct rk_i2s_dev *i2s = dev_get_drvdata(dev); + + clk_disable_unprepare(i2s->mclk); + + return 0; +} + +static int i2s_runtime_resume(struct device *dev) +{ + struct rk_i2s_dev *i2s = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(i2s->mclk); + if (ret) { + dev_err(i2s->dev, "clock enable failed %d\n", ret); + return ret; + } + + return 0; +} + +static inline struct rk_i2s_dev *to_info(struct snd_soc_dai *dai) +{ + return snd_soc_dai_get_drvdata(dai); +} + +static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) +{ + unsigned int val = 0; + int retry = 10; + + if (on) { + regmap_update_bits(i2s->regmap, I2S_DMACR, + I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_ENABLE); + + regmap_update_bits(i2s->regmap, I2S_XFER, + I2S_XFER_TXS_START | I2S_XFER_RXS_START, + I2S_XFER_TXS_START | I2S_XFER_RXS_START); + + i2s->tx_start = true; + } else { + i2s->tx_start = false; + + regmap_update_bits(i2s->regmap, I2S_DMACR, + I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_ENABLE); + + if (!i2s->rx_start) { + regmap_update_bits(i2s->regmap, I2S_XFER, + I2S_XFER_TXS_START | + I2S_XFER_RXS_START, + I2S_XFER_TXS_STOP | + I2S_XFER_RXS_STOP); + + regmap_update_bits(i2s->regmap, I2S_CLR, + I2S_CLR_TXC | I2S_CLR_TXC, + I2S_CLR_TXC | I2S_CLR_TXC); + + regmap_read(i2s->regmap, I2S_CLR, &val); + + /* Should wait for clear operation to finish */ + while (val) { + regmap_read(i2s->regmap, I2S_CLR, &val); + retry--; + if (!retry) + dev_warn(i2s->dev, "fail to clear\n"); + } + } + } +} + +static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) +{ + unsigned int val = 0; + int retry = 10; + + if (on) { + regmap_update_bits(i2s->regmap, I2S_DMACR, + I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_ENABLE); + + regmap_update_bits(i2s->regmap, I2S_XFER, + I2S_XFER_TXS_START | I2S_XFER_RXS_START, + I2S_XFER_TXS_START | I2S_XFER_RXS_START); + + i2s->rx_start = true; + } else { + i2s->rx_start = false; + + regmap_update_bits(i2s->regmap, I2S_DMACR, + I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_DISABLE); + + if (!i2s->tx_start) { + regmap_update_bits(i2s->regmap, I2S_XFER, + I2S_XFER_TXS_START | + I2S_XFER_RXS_START, + I2S_XFER_TXS_STOP | + I2S_XFER_RXS_STOP); + + regmap_update_bits(i2s->regmap, I2S_CLR, + I2S_CLR_TXC | I2S_CLR_TXC, + I2S_CLR_TXC | I2S_CLR_TXC); + + regmap_read(i2s->regmap, I2S_CLR, &val); + + /* Should wait for clear operation to finish */ + while (val) { + regmap_read(i2s->regmap, I2S_CLR, &val); + retry--; + if (!retry) + dev_warn(i2s->dev, "fail to clear\n"); + } + } + } +} + +static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct rk_i2s_dev *i2s = to_info(cpu_dai); + unsigned int mask = 0, val = 0; + + mask = I2S_CKR_MSS_SLAVE; + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + val = I2S_CKR_MSS_SLAVE; + break; + case SND_SOC_DAIFMT_CBM_CFM: + val = I2S_CKR_MSS_MASTER; + break; + default: + return -EINVAL; + } + + regmap_update_bits(i2s->regmap, I2S_CKR, mask, val); + + mask = I2S_TXCR_IBM_MASK; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + val = I2S_TXCR_IBM_RSJM; + break; + case SND_SOC_DAIFMT_LEFT_J: + val = I2S_TXCR_IBM_LSJM; + break; + case SND_SOC_DAIFMT_I2S: + val = I2S_TXCR_IBM_NORMAL; + break; + default: + return -EINVAL; + } + + regmap_update_bits(i2s->regmap, I2S_TXCR, mask, val); + + mask = I2S_RXCR_IBM_MASK; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + val = I2S_RXCR_IBM_RSJM; + break; + case SND_SOC_DAIFMT_LEFT_J: + val = I2S_RXCR_IBM_LSJM; + break; + case SND_SOC_DAIFMT_I2S: + val = I2S_RXCR_IBM_NORMAL; + break; + default: + return -EINVAL; + } + + regmap_update_bits(i2s->regmap, I2S_RXCR, mask, val); + + return 0; +} + +static int rockchip_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct rk_i2s_dev *i2s = to_info(dai); + unsigned int val = 0; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + val |= I2S_TXCR_VDW(8); + break; + case SNDRV_PCM_FORMAT_S16_LE: + val |= I2S_TXCR_VDW(16); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val |= I2S_TXCR_VDW(20); + break; + case SNDRV_PCM_FORMAT_S24_LE: + val |= I2S_TXCR_VDW(24); + break; + default: + return -EINVAL; + } + + regmap_update_bits(i2s->regmap, I2S_TXCR, I2S_TXCR_VDW_MASK, val); + regmap_update_bits(i2s->regmap, I2S_RXCR, I2S_RXCR_VDW_MASK, val); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + dai->playback_dma_data = &i2s->playback_dma_data; + regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_TDL_MASK, + I2S_DMACR_TDL(1) | I2S_DMACR_TDE_ENABLE); + } else { + dai->capture_dma_data = &i2s->capture_dma_data; + regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_RDL_MASK, + I2S_DMACR_RDL(1) | I2S_DMACR_RDE_ENABLE); + } + + return 0; +} + +static int rockchip_i2s_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct rk_i2s_dev *i2s = to_info(dai); + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + rockchip_snd_rxctrl(i2s, 1); + else + rockchip_snd_txctrl(i2s, 1); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + rockchip_snd_rxctrl(i2s, 0); + else + rockchip_snd_txctrl(i2s, 0); + break; + default: + ret = -EINVAL; + break; + } + + return ret; +} + +static int rockchip_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, + unsigned int freq, int dir) +{ + struct rk_i2s_dev *i2s = to_info(cpu_dai); + int ret; + + ret = clk_set_rate(i2s->mclk, freq); + if (ret) + dev_err(i2s->dev, "Fail to set mclk %d\n", ret); + + return ret; +} + +static const struct snd_soc_dai_ops rockchip_i2s_dai_ops = { + .hw_params = rockchip_i2s_hw_params, + .set_sysclk = rockchip_i2s_set_sysclk, + .set_fmt = rockchip_i2s_set_fmt, + .trigger = rockchip_i2s_trigger, +}; + +static struct snd_soc_dai_driver rockchip_i2s_dai = { + .playback = { + .channels_min = 2, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = (SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | + SNDRV_PCM_FMTBIT_S24_LE), + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = (SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | + SNDRV_PCM_FMTBIT_S24_LE), + }, + .ops = &rockchip_i2s_dai_ops, +}; + +static const struct snd_soc_component_driver rockchip_i2s_component = { + .name = DRV_NAME, +}; + +static bool rockchip_i2s_wr_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case I2S_TXCR: + case I2S_RXCR: + case I2S_CKR: + case I2S_DMACR: + case I2S_INTCR: + case I2S_XFER: + case I2S_CLR: + case I2S_TXDR: + return true; + default: + return false; + } +} + +static bool rockchip_i2s_rd_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case I2S_TXCR: + case I2S_RXCR: + case I2S_CKR: + case I2S_DMACR: + case I2S_INTCR: + case I2S_XFER: + case I2S_CLR: + case I2S_RXDR: + return true; + default: + return false; + } +} + +static bool rockchip_i2s_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case I2S_FIFOLR: + case I2S_INTSR: + return true; + default: + return false; + } +} + +static bool rockchip_i2s_precious_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case I2S_FIFOLR: + return true; + default: + return false; + } +} + +static const struct regmap_config rockchip_i2s_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = I2S_RXDR, + .writeable_reg = rockchip_i2s_wr_reg, + .readable_reg = rockchip_i2s_rd_reg, + .volatile_reg = rockchip_i2s_volatile_reg, + .precious_reg = rockchip_i2s_precious_reg, + .cache_type = REGCACHE_FLAT, +}; + +static int rockchip_i2s_probe(struct platform_device *pdev) +{ + struct rk_i2s_dev *i2s; + struct resource *res; + void __iomem *regs; + int ret; + + i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL); + if (!i2s) { + dev_err(&pdev->dev, "Can't allocate rk_i2s_dev\n"); + return -ENOMEM; + } + + /* try to prepare related clocks */ + i2s->hclk = devm_clk_get(&pdev->dev, "i2s_hclk"); + if (IS_ERR(i2s->hclk)) { + dev_err(&pdev->dev, "Can't retrieve i2s bus clock\n"); + return PTR_ERR(i2s->hclk); + } + + i2s->mclk = devm_clk_get(&pdev->dev, "i2s_clk"); + if (IS_ERR(i2s->mclk)) { + dev_err(&pdev->dev, "Can't retrieve i2s master clock\n"); + return PTR_ERR(i2s->mclk); + } + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + regs = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(regs)) { + dev_err(&pdev->dev, "No memory resource\n"); + return PTR_ERR(regs); + } + + i2s->regmap = devm_regmap_init_mmio(&pdev->dev, regs, + &rockchip_i2s_regmap_config); + if (IS_ERR(i2s->regmap)) { + dev_err(&pdev->dev, + "Failed to initialise managed register map\n"); + return PTR_ERR(i2s->regmap); + } + + i2s->playback_dma_data.addr = res->start + I2S_TXDR; + i2s->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + i2s->playback_dma_data.maxburst = 16; + + i2s->capture_dma_data.addr = res->start + I2S_RXDR; + i2s->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + i2s->capture_dma_data.maxburst = 16; + + i2s->dev = &pdev->dev; + dev_set_drvdata(&pdev->dev, i2s); + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = i2s_runtime_resume(&pdev->dev); + if (ret) + goto err_pm_disable; + } + + ret = devm_snd_soc_register_component(&pdev->dev, + &rockchip_i2s_component, + &rockchip_i2s_dai, 1); + if (ret) { + dev_err(&pdev->dev, "Could not register DAI\n"); + goto err_suspend; + } + + ret = snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) { + dev_err(&pdev->dev, "Could not register PCM\n"); + goto err_pcm_register; + } + + return 0; + +err_pcm_register: + snd_dmaengine_pcm_unregister(&pdev->dev); +err_suspend: + if (!pm_runtime_status_suspended(&pdev->dev)) + i2s_runtime_suspend(&pdev->dev); +err_pm_disable: + pm_runtime_disable(&pdev->dev); + + return ret; +} + +static int rockchip_i2s_remove(struct platform_device *pdev) +{ + struct rk_i2s_dev *i2s = dev_get_drvdata(&pdev->dev); + + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + i2s_runtime_suspend(&pdev->dev); + + clk_disable_unprepare(i2s->mclk); + clk_disable_unprepare(i2s->hclk); + snd_dmaengine_pcm_unregister(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); + + return 0; +} + +static const struct of_device_id rockchip_i2s_match[] = { + { .compatible = "rockchip,rk3066-i2s", }, + {}, +}; + +static const struct dev_pm_ops rockchip_i2s_pm_ops = { + SET_RUNTIME_PM_OPS(i2s_runtime_suspend, i2s_runtime_resume, + NULL) +}; + +static struct platform_driver rockchip_i2s_driver = { + .probe = rockchip_i2s_probe, + .remove = rockchip_i2s_remove, + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(rockchip_i2s_match), + .pm = &rockchip_i2s_pm_ops, + }, +}; +module_platform_driver(rockchip_i2s_driver); + +MODULE_DESCRIPTION("ROCKCHIP IIS ASoC Interface"); +MODULE_AUTHOR("jianqun "); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, rockchip_i2s_match); diff --git a/sound/soc/rockchip/rockchip_i2s.h b/sound/soc/rockchip/rockchip_i2s.h new file mode 100644 index 000000000000..89a5d8bc6ee7 --- /dev/null +++ b/sound/soc/rockchip/rockchip_i2s.h @@ -0,0 +1,223 @@ +/* + * sound/soc/rockchip/rockchip_i2s.h + * + * ALSA SoC Audio Layer - Rockchip I2S Controller driver + * + * Copyright (c) 2014 Rockchip Electronics Co. Ltd. + * Author: Jianqun xu + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _ROCKCHIP_IIS_H +#define _ROCKCHIP_IIS_H + +/* + * TXCR + * transmit operation control register +*/ +#define I2S_TXCR_RCNT_SHIFT 17 +#define I2S_TXCR_RCNT_MASK (0x3f << I2S_TXCR_RCNT_SHIFT) +#define I2S_TXCR_CSR_SHIFT 15 +#define I2S_TXCR_CSR(x) (x << I2S_TXCR_CSR_SHIFT) +#define I2S_TXCR_CSR_MASK (3 << I2S_TXCR_CSR_SHIFT) +#define I2S_TXCR_HWT BIT(14) +#define I2S_TXCR_SJM_SHIFT 12 +#define I2S_TXCR_SJM_R (0 << I2S_TXCR_SJM_SHIFT) +#define I2S_TXCR_SJM_L (1 << I2S_TXCR_SJM_SHIFT) +#define I2S_TXCR_FBM_SHIFT 11 +#define I2S_TXCR_FBM_MSB (0 << I2S_TXCR_FBM_SHIFT) +#define I2S_TXCR_FBM_LSB (1 << I2S_TXCR_FBM_SHIFT) +#define I2S_TXCR_IBM_SHIFT 9 +#define I2S_TXCR_IBM_NORMAL (0 << I2S_TXCR_IBM_SHIFT) +#define I2S_TXCR_IBM_LSJM (1 << I2S_TXCR_IBM_SHIFT) +#define I2S_TXCR_IBM_RSJM (2 << I2S_TXCR_IBM_SHIFT) +#define I2S_TXCR_IBM_MASK (3 << I2S_TXCR_IBM_SHIFT) +#define I2S_TXCR_PBM_SHIFT 7 +#define I2S_TXCR_PBM_MODE(x) (x << I2S_TXCR_PBM_SHIFT) +#define I2S_TXCR_PBM_MASK (3 << I2S_TXCR_PBM_SHIFT) +#define I2S_TXCR_TFS_SHIFT 5 +#define I2S_TXCR_TFS_I2S (0 << I2S_TXCR_TFS_SHIFT) +#define I2S_TXCR_TFS_PCM (1 << I2S_TXCR_TFS_SHIFT) +#define I2S_TXCR_VDW_SHIFT 0 +#define I2S_TXCR_VDW(x) ((x - 1) << I2S_TXCR_VDW_SHIFT) +#define I2S_TXCR_VDW_MASK (0x1f << I2S_TXCR_VDW_SHIFT) + +/* + * RXCR + * receive operation control register +*/ +#define I2S_RXCR_HWT BIT(14) +#define I2S_RXCR_SJM_SHIFT 12 +#define I2S_RXCR_SJM_R (0 << I2S_RXCR_SJM_SHIFT) +#define I2S_RXCR_SJM_L (1 << I2S_RXCR_SJM_SHIFT) +#define I2S_RXCR_FBM_SHIFT 11 +#define I2S_RXCR_FBM_MSB (0 << I2S_RXCR_FBM_SHIFT) +#define I2S_RXCR_FBM_LSB (1 << I2S_RXCR_FBM_SHIFT) +#define I2S_RXCR_IBM_SHIFT 9 +#define I2S_RXCR_IBM_NORMAL (0 << I2S_RXCR_IBM_SHIFT) +#define I2S_RXCR_IBM_LSJM (1 << I2S_RXCR_IBM_SHIFT) +#define I2S_RXCR_IBM_RSJM (2 << I2S_RXCR_IBM_SHIFT) +#define I2S_RXCR_IBM_MASK (3 << I2S_RXCR_IBM_SHIFT) +#define I2S_RXCR_PBM_SHIFT 7 +#define I2S_RXCR_PBM_MODE(x) (x << I2S_RXCR_PBM_SHIFT) +#define I2S_RXCR_PBM_MASK (3 << I2S_RXCR_PBM_SHIFT) +#define I2S_RXCR_TFS_SHIFT 5 +#define I2S_RXCR_TFS_I2S (0 << I2S_RXCR_TFS_SHIFT) +#define I2S_RXCR_TFS_PCM (1 << I2S_RXCR_TFS_SHIFT) +#define I2S_RXCR_VDW_SHIFT 0 +#define I2S_RXCR_VDW(x) ((x - 1) << I2S_RXCR_VDW_SHIFT) +#define I2S_RXCR_VDW_MASK (0x1f << I2S_RXCR_VDW_SHIFT) + +/* + * CKR + * clock generation register +*/ +#define I2S_CKR_MSS_SHIFT 27 +#define I2S_CKR_MSS_MASTER (0 << I2S_CKR_MSS_SHIFT) +#define I2S_CKR_MSS_SLAVE (1 << I2S_CKR_MSS_SHIFT) +#define I2S_CKR_MSS_MASK (1 << I2S_CKR_MSS_SHIFT) +#define I2S_CKR_CKP_SHIFT 26 +#define I2S_CKR_CKP_NEG (0 << I2S_CKR_CKP_SHIFT) +#define I2S_CKR_CKP_POS (1 << I2S_CKR_CKP_SHIFT) +#define I2S_CKR_RLP_SHIFT 25 +#define I2S_CKR_RLP_NORMAL (0 << I2S_CKR_RLP_SHIFT) +#define I2S_CKR_RLP_OPPSITE (1 << I2S_CKR_RLP_SHIFT) +#define I2S_CKR_TLP_SHIFT 24 +#define I2S_CKR_TLP_NORMAL (0 << I2S_CKR_TLP_SHIFT) +#define I2S_CKR_TLP_OPPSITE (1 << I2S_CKR_TLP_SHIFT) +#define I2S_CKR_MDIV_SHIFT 16 +#define I2S_CKR_MDIV(x) ((x - 1) << I2S_CKR_MDIV_SHIFT) +#define I2S_CKR_MDIV_MASK (0xff << I2S_CKR_MDIV_SHIFT) +#define I2S_CKR_RSD_SHIFT 8 +#define I2S_CKR_RSD(x) ((x - 1) << I2S_CKR_RSD_SHIFT) +#define I2S_CKR_RSD_MASK (0xff << I2S_CKR_RSD_SHIFT) +#define I2S_CKR_TSD_SHIFT 0 +#define I2S_CKR_TSD(x) ((x - 1) << I2S_CKR_TSD_SHIFT) +#define I2S_CKR_TSD_MASK (0xff << I2S_CKR_TSD_SHIFT) + +/* + * FIFOLR + * FIFO level register +*/ +#define I2S_FIFOLR_RFL_SHIFT 24 +#define I2S_FIFOLR_RFL_MASK (0x3f << I2S_FIFOLR_RFL_SHIFT) +#define I2S_FIFOLR_TFL3_SHIFT 18 +#define I2S_FIFOLR_TFL3_MASK (0x3f << I2S_FIFOLR_TFL3_SHIFT) +#define I2S_FIFOLR_TFL2_SHIFT 12 +#define I2S_FIFOLR_TFL2_MASK (0x3f << I2S_FIFOLR_TFL2_SHIFT) +#define I2S_FIFOLR_TFL1_SHIFT 6 +#define I2S_FIFOLR_TFL1_MASK (0x3f << I2S_FIFOLR_TFL1_SHIFT) +#define I2S_FIFOLR_TFL0_SHIFT 0 +#define I2S_FIFOLR_TFL0_MASK (0x3f << I2S_FIFOLR_TFL0_SHIFT) + +/* + * DMACR + * DMA control register +*/ +#define I2S_DMACR_RDE_SHIFT 24 +#define I2S_DMACR_RDE_DISABLE (0 << I2S_DMACR_RDE_SHIFT) +#define I2S_DMACR_RDE_ENABLE (1 << I2S_DMACR_RDE_SHIFT) +#define I2S_DMACR_RDL_SHIFT 16 +#define I2S_DMACR_RDL(x) ((x - 1) << I2S_DMACR_RDL_SHIFT) +#define I2S_DMACR_RDL_MASK (0x1f << I2S_DMACR_RDL_SHIFT) +#define I2S_DMACR_TDE_SHIFT 8 +#define I2S_DMACR_TDE_DISABLE (0 << I2S_DMACR_TDE_SHIFT) +#define I2S_DMACR_TDE_ENABLE (1 << I2S_DMACR_TDE_SHIFT) +#define I2S_DMACR_TDL_SHIFT 0 +#define I2S_DMACR_TDL(x) ((x - 1) << I2S_DMACR_TDL_SHIFT) +#define I2S_DMACR_TDL_MASK (0x1f << I2S_DMACR_TDL_SHIFT) + +/* + * INTCR + * interrupt control register +*/ +#define I2S_INTCR_RFT_SHIFT 20 +#define I2S_INTCR_RFT(x) ((x - 1) << I2S_INTCR_RFT_SHIFT) +#define I2S_INTCR_RXOIC BIT(18) +#define I2S_INTCR_RXOIE_SHIFT 17 +#define I2S_INTCR_RXOIE_DISABLE (0 << I2S_INTCR_RXOIE_SHIFT) +#define I2S_INTCR_RXOIE_ENABLE (1 << I2S_INTCR_RXOIE_SHIFT) +#define I2S_INTCR_RXFIE_SHIFT 16 +#define I2S_INTCR_RXFIE_DISABLE (0 << I2S_INTCR_RXFIE_SHIFT) +#define I2S_INTCR_RXFIE_ENABLE (1 << I2S_INTCR_RXFIE_SHIFT) +#define I2S_INTCR_TFT_SHIFT 4 +#define I2S_INTCR_TFT(x) ((x - 1) << I2S_INTCR_TFT_SHIFT) +#define I2S_INTCR_TFT_MASK (0x1f << I2S_INTCR_TFT_SHIFT) +#define I2S_INTCR_TXUIC BIT(2) +#define I2S_INTCR_TXUIE_SHIFT 1 +#define I2S_INTCR_TXUIE_DISABLE (0 << I2S_INTCR_TXUIE_SHIFT) +#define I2S_INTCR_TXUIE_ENABLE (1 << I2S_INTCR_TXUIE_SHIFT) + +/* + * INTSR + * interrupt status register +*/ +#define I2S_INTSR_TXEIE_SHIFT 0 +#define I2S_INTSR_TXEIE_DISABLE (0 << I2S_INTSR_TXEIE_SHIFT) +#define I2S_INTSR_TXEIE_ENABLE (1 << I2S_INTSR_TXEIE_SHIFT) +#define I2S_INTSR_RXOI_SHIFT 17 +#define I2S_INTSR_RXOI_INA (0 << I2S_INTSR_RXOI_SHIFT) +#define I2S_INTSR_RXOI_ACT (1 << I2S_INTSR_RXOI_SHIFT) +#define I2S_INTSR_RXFI_SHIFT 16 +#define I2S_INTSR_RXFI_INA (0 << I2S_INTSR_RXFI_SHIFT) +#define I2S_INTSR_RXFI_ACT (1 << I2S_INTSR_RXFI_SHIFT) +#define I2S_INTSR_TXUI_SHIFT 1 +#define I2S_INTSR_TXUI_INA (0 << I2S_INTSR_TXUI_SHIFT) +#define I2S_INTSR_TXUI_ACT (1 << I2S_INTSR_TXUI_SHIFT) +#define I2S_INTSR_TXEI_SHIFT 0 +#define I2S_INTSR_TXEI_INA (0 << I2S_INTSR_TXEI_SHIFT) +#define I2S_INTSR_TXEI_ACT (1 << I2S_INTSR_TXEI_SHIFT) + +/* + * XFER + * Transfer start register +*/ +#define I2S_XFER_RXS_SHIFT 1 +#define I2S_XFER_RXS_STOP (0 << I2S_XFER_RXS_SHIFT) +#define I2S_XFER_RXS_START (1 << I2S_XFER_RXS_SHIFT) +#define I2S_XFER_TXS_SHIFT 0 +#define I2S_XFER_TXS_STOP (0 << I2S_XFER_TXS_SHIFT) +#define I2S_XFER_TXS_START (1 << I2S_XFER_TXS_SHIFT) + +/* + * CLR + * clear SCLK domain logic register +*/ +#define I2S_CLR_RXC BIT(1) +#define I2S_CLR_TXC BIT(0) + +/* + * TXDR + * Transimt FIFO data register, write only. +*/ +#define I2S_TXDR_MASK (0xff) + +/* + * RXDR + * Receive FIFO data register, write only. +*/ +#define I2S_RXDR_MASK (0xff) + +/* Clock divider id */ +enum { + ROCKCHIP_DIV_MCLK = 0, + ROCKCHIP_DIV_BCLK, +}; + +/* I2S REGS */ +#define I2S_TXCR (0x0000) +#define I2S_RXCR (0x0004) +#define I2S_CKR (0x0008) +#define I2S_FIFOLR (0x000c) +#define I2S_DMACR (0x0010) +#define I2S_INTCR (0x0014) +#define I2S_INTSR (0x0018) +#define I2S_XFER (0x001c) +#define I2S_CLR (0x0020) +#define I2S_TXDR (0x0024) +#define I2S_RXDR (0x0028) + +#endif /* _ROCKCHIP_IIS_H */ -- cgit v1.2.3 From 7e07e7c0854b32181b314f5ba43007629594663a Mon Sep 17 00:00:00 2001 From: Tushar Behera Date: Fri, 4 Jul 2014 14:23:00 +0530 Subject: ASoC: core: Fix possible NULL pointer dereference snd_soc_of_parse_card_name() may be called before card->dev has been set, which results in a kernel panic. Unable to handle kernel NULL pointer dereference at virtual address 00000210 PC is at snd_soc_of_parse_card_name+0x18/0x54 LR is at snow_probe+0x5c/0xd4 Add an error check in snd_soc_of_parse_card_name() to take care of this case and print out a message in case of error. Signed-off-by: Tushar Behera Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b87d7d882e6d..63d11071fdba 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4420,9 +4420,16 @@ EXPORT_SYMBOL_GPL(snd_soc_unregister_codec); int snd_soc_of_parse_card_name(struct snd_soc_card *card, const char *propname) { - struct device_node *np = card->dev->of_node; + struct device_node *np; int ret; + if (!card->dev) { + pr_err("card->dev is not set before calling %s\n", __func__); + return -EINVAL; + } + + np = card->dev->of_node; + ret = of_property_read_string_index(np, propname, 0, &card->name); /* * EINVAL means the property does not exist. This is fine providing -- cgit v1.2.3 From e42be7e1420389349400e990a3b13d23b9c60d39 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Mon, 7 Jul 2014 14:26:48 -0300 Subject: ASoC: sgtl5000: Fix driver unbound Using the sgtl5000 codec driver as a module and trying to remove it causes the followig kernel oops: root@freescale /$ rmmod snd-soc-imx-sgtl5000 [ 117.122920] ------------[ cut here ]------------ [ 117.127609] WARNING: CPU: 0 PID: 631 at drivers/regulator/core.c:3604 regula) [ 117.137046] Modules linked in: snd_soc_imx_sgtl5000(-) snd_soc_sgtl5000 evbug [ 117.144315] CPU: 0 PID: 631 Comm: rmmod Not tainted 3.16.0-rc3-next-201407043 [ 117.153366] Backtrace: [ 117.155865] [<80011e5c>] (dump_backtrace) from [<80011ff8>] (show_stack+0x18) [ 117.163484] r6:802fcc48 r5:00000000 r4:00000000 r3:00000000 [ 117.169228] [<80011fe0>] (show_stack) from [<80668cc0>] (dump_stack+0x88/0xa) [ 117.176508] [<80668c38>] (dump_stack) from [<80029a38>] (warn_slowpath_commo) [ 117.184696] r5:00000009 r4:00000000 [ 117.188322] [<800299c8>] (warn_slowpath_common) from [<80029a80>] (warn_slow) [ 117.197150] r8:dd60d600 r7:ddfa6d00 r6:dd5d9064 r5:dd5e0f90 r4:dd5d9400 [ 117.203983] [<80029a5c>] (warn_slowpath_null) from [<802fcc48>] (regulator_u) [ 117.212828] [<802fcb74>] (regulator_unregister) from [<7f0047c4>] (ldo_regul) [ 117.223475] r4:dd59e300 r3:dd5e0f90 [ 117.227100] [<7f00479c>] (ldo_regulator_remove [snd_soc_sgtl5000]) from [<7f) [ 117.238959] r4:dd5d8000 r3:ddd51420 [ 117.242623] [<7f0047dc>] (sgtl5000_remove [snd_soc_sgtl5000]) from [<804e5b1) [ 117.252489] r5:00000000 r4:dd5d8000 [ 117.256111] [<804e5af8>] (soc_remove_codec) from [<804e5ed4>] (soc_remove_da) [ 117.264933] r4:ddfb640c r3:00000000 [ 117.268555] [<804e5c10>] (soc_remove_dai_links) from [<804e5fbc>] (snd_soc_u) [ 117.277810] r10:80359e48 r9:dd684000 r8:dd5ca800 r7:dd685e60 r6:00000001 r5c [ 117.285761] r4:dd5d9064 [ 117.288329] [<804e5f28>] (snd_soc_unregister_card) from [<804f29f0>] (devm_c) [ 117.297324] r6:00000004 r5:ddd0fc10 r4:dd5a7980 r3:804f29dc [ 117.303095] [<804f29dc>] (devm_card_release) from [<8035a418>] (release_node) [ 117.311369] [<8035a2a8>] (release_nodes) from [<8035aab4>] (devres_release_a) [ 117.319583] r10:00000000 r9:dd684000 r8:8000ed64 r7:00000081 r6:ddd0fc44 r54 [ 117.327543] r4:ddd0fc10 [ 117.330108] [<8035aa7c>] (devres_release_all) from [<803571c8>] (__device_re) [ 117.339214] r4:ddd0fc10 r3:dd5d9010 [ 117.342989] [<80357148>] (__device_release_driver) from [<80357a48>] (driver) [ 117.351607] r5:7f00c9e4 r4:ddd0fc10 [ 117.355285] [<8035798c>] (driver_detach) from [<80357030>] (bus_remove_drive) [ 117.363454] r6:00000880 r5:00000000 r4:7f00c9e4 r3:dd6c3a80 [ 117.369195] [<80356fdc>] (bus_remove_driver) from [<803580b8>] (driver_unreg) [ 117.377683] r4:7f00c9e4 r3:dd60d480 [ 117.381305] [<80358088>] (driver_unregister) from [<80358bb4>] (platform_dri) [ 117.390642] r4:7f00ca28 r3:7f00c35c [ 117.394309] [<80358ba0>] (platform_driver_unregister) from [<7f00c370>] (imx) [ 117.406188] [<7f00c35c>] (imx_sgtl5000_driver_exit [snd_soc_imx_sgtl5000]) f) [ 117.417452] [<8008c2b8>] (SyS_delete_module) from [<8000eba0>] (ret_fast_sys) [ 117.425753] r6:5f636f73 r5:5f646e73 r4:00016f40 [ 117.430428] ---[ end trace 8fd8a5cb39e46d0e ]--- This problem is well explained by Russell King: "The sgtl5000 uses a two-stage initialisation process. The first stage is when the platform driver is probed, where some resources are found and initialised. The second stage is via the codec driver's probe function, where regulators are found and initialised using the managed resource support. The problem here is that this works fine until the codec driver is unbound. When this occurs, sgtl5000_remove() is called which is a no-op as far as the managed resource code is concerned. The regulators remain allocated, and their pointers in sgtl5000_priv remain valid. If the codec is now re-probed, it will again try and find the regulators, which will now be busy. This will fail. That's not the only problem - if using the LDO regulator, the regulator is unregistered from the regulator core code, but it still has a user. When the user is cleaned up (eg, by removing the module) it hits the free'd regulator, and this can oops the kernel. This bug was originally introduced by 63e54cd9caa3ce ("ASoC: sgtl5000: Use devm_regulator_bulk_get()")." This reverts commit 63e54cd9caa3ce. Tested on a imx53-qsb board. Reported-by: Russell King Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 3d39f0b5b4a8..8f4c73d17c87 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1277,7 +1277,7 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) return ret; } - ret = devm_regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies), + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); if (ret) goto err_ldo_remove; @@ -1285,13 +1285,16 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); if (ret) - goto err_ldo_remove; + goto err_regulator_free; /* wait for all power rails bring up */ udelay(10); return 0; +err_regulator_free: + regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); err_ldo_remove: if (!external_vddd) ldo_regulator_remove(codec); @@ -1361,6 +1364,8 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) err: regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); + regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); ldo_regulator_remove(codec); return ret; @@ -1374,6 +1379,8 @@ static int sgtl5000_remove(struct snd_soc_codec *codec) regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); + regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); ldo_regulator_remove(codec); return 0; -- cgit v1.2.3 From 36765c9c0623e87ab5ac6cfc07f66072b83edfdc Mon Sep 17 00:00:00 2001 From: Manish Badarkhe Date: Tue, 8 Jul 2014 21:55:23 +0530 Subject: ASoC: omap-mcbsp: Use devm_snd_soc_register_component Replaced snd_soc_register_component with its devres equivalent, devm_snd_soc_register_component. Signed-off-by: Manish Badarkhe Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/omap-mcbsp.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index efe2cd699b77..bd3ef2a88be0 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -805,8 +805,9 @@ static int asoc_mcbsp_probe(struct platform_device *pdev) if (ret) return ret; - ret = snd_soc_register_component(&pdev->dev, &omap_mcbsp_component, - &omap_mcbsp_dai, 1); + ret = devm_snd_soc_register_component(&pdev->dev, + &omap_mcbsp_component, + &omap_mcbsp_dai, 1); if (ret) return ret; @@ -817,8 +818,6 @@ static int asoc_mcbsp_remove(struct platform_device *pdev) { struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev); - snd_soc_unregister_component(&pdev->dev); - if (mcbsp->pdata->ops && mcbsp->pdata->ops->free) mcbsp->pdata->ops->free(mcbsp->id); -- cgit v1.2.3 From 159baadf59416d9ddf860bac9f39e19a04b21603 Mon Sep 17 00:00:00 2001 From: Manish Badarkhe Date: Tue, 8 Jul 2014 22:23:17 +0530 Subject: ASoC: omap-dmic: Use devm_snd_soc_register_component Replaced snd_soc_register_component with its devres equivalent, devm_snd_soc_register_component. Signed-off-by: Manish Badarkhe Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/omap-dmic.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c index 6925d7141215..f125eb95d8c8 100644 --- a/sound/soc/omap/omap-dmic.c +++ b/sound/soc/omap/omap-dmic.c @@ -490,8 +490,9 @@ static int asoc_dmic_probe(struct platform_device *pdev) } - ret = snd_soc_register_component(&pdev->dev, &omap_dmic_component, - &omap_dmic_dai, 1); + ret = devm_snd_soc_register_component(&pdev->dev, + &omap_dmic_component, + &omap_dmic_dai, 1); if (ret) goto err_put_clk; @@ -510,7 +511,6 @@ static int asoc_dmic_remove(struct platform_device *pdev) { struct omap_dmic *dmic = platform_get_drvdata(pdev); - snd_soc_unregister_component(&pdev->dev); clk_put(dmic->fclk); return 0; -- cgit v1.2.3 From a40712a3e604bd59c7d3bdbabc9aec6208102783 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Fri, 4 Jul 2014 15:13:45 +0200 Subject: ASoC: samsung: Add machine driver for Odroid X2/U3 This patch adds the sound subsystem driver for Odroid-X2 and Odroid-U3 boards. The codec works in I2S master mode; there are two separate audio routing paths defined, as there are differences in the signal routing between the X2 and U3 boards, i.e. U3 uses single jack for headphones and microphone. Signed-off-by: Chen Zhen Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown --- sound/soc/samsung/Kconfig | 8 ++ sound/soc/samsung/Makefile | 2 + sound/soc/samsung/odroidx2_max98090.c | 177 ++++++++++++++++++++++++++++++++++ 3 files changed, 187 insertions(+) create mode 100644 sound/soc/samsung/odroidx2_max98090.c (limited to 'sound') diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 3be49cd2e984..333d645cea87 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -237,3 +237,11 @@ config SND_SOC_SNOW help Say Y if you want to add audio support for various Snow boards based on Exynos5 series of SoCs. + +config SND_SOC_ODROIDX2 + tristate "Audio support for Odroid-X2 and Odroid-U3" + depends on SND_SOC_SAMSUNG + select SND_SOC_MAX98090 + select SND_SAMSUNG_I2S + help + Say Y here to enable audio support for the Odroid-X2/U3. diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 6469199acea5..e8d9ccdc41fd 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -44,6 +44,7 @@ snd-soc-tobermory-objs := tobermory.o snd-soc-lowland-objs := lowland.o snd-soc-littlemill-objs := littlemill.o snd-soc-bells-objs := bells.o +snd-soc-odroidx2-max98090-objs := odroidx2_max98090.o obj-$(CONFIG_SND_SOC_SAMSUNG_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_SOC_SAMSUNG_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -69,3 +70,4 @@ obj-$(CONFIG_SND_SOC_TOBERMORY) += snd-soc-tobermory.o obj-$(CONFIG_SND_SOC_LOWLAND) += snd-soc-lowland.o obj-$(CONFIG_SND_SOC_LITTLEMILL) += snd-soc-littlemill.o obj-$(CONFIG_SND_SOC_BELLS) += snd-soc-bells.o +obj-$(CONFIG_SND_SOC_ODROIDX2) += snd-soc-odroidx2-max98090.o diff --git a/sound/soc/samsung/odroidx2_max98090.c b/sound/soc/samsung/odroidx2_max98090.c new file mode 100644 index 000000000000..278edf9e2a87 --- /dev/null +++ b/sound/soc/samsung/odroidx2_max98090.c @@ -0,0 +1,177 @@ +/* + * Copyright (C) 2014 Samsung Electronics Co., Ltd. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include +#include +#include +#include +#include "i2s.h" + +struct odroidx2_drv_data { + const struct snd_soc_dapm_widget *dapm_widgets; + unsigned int num_dapm_widgets; +}; + +/* The I2S CDCLK output clock frequency for the MAX98090 codec */ +#define MAX98090_MCLK 19200000 + +static int odroidx2_late_probe(struct snd_soc_card *card) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct snd_soc_dai *cpu_dai = card->rtd[0].cpu_dai; + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, MAX98090_MCLK, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* Set the cpu DAI configuration in order to use CDCLK */ + return snd_soc_dai_set_sysclk(cpu_dai, SAMSUNG_I2S_CDCLK, + 0, SND_SOC_CLOCK_OUT); +} + +static const struct snd_soc_dapm_widget odroidx2_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_MIC("DMIC", NULL), +}; + +static const struct snd_soc_dapm_widget odroidu3_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Speakers", NULL), +}; + +static struct snd_soc_dai_link odroidx2_dai[] = { + { + .name = "MAX98090", + .stream_name = "MAX98090 PCM", + .codec_dai_name = "HiFi", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, + } +}; + +static struct snd_soc_card odroidx2 = { + .owner = THIS_MODULE, + .dai_link = odroidx2_dai, + .num_links = ARRAY_SIZE(odroidx2_dai), + .fully_routed = true, + .late_probe = odroidx2_late_probe, +}; + +struct odroidx2_drv_data odroidx2_drvdata = { + .dapm_widgets = odroidx2_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(odroidx2_dapm_widgets), +}; + +struct odroidx2_drv_data odroidu3_drvdata = { + .dapm_widgets = odroidu3_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(odroidu3_dapm_widgets), +}; + +static const struct of_device_id odroidx2_audio_of_match[] = { + { + .compatible = "samsung,odroidx2-audio", + .data = &odroidx2_drvdata, + }, { + .compatible = "samsung,odroidu3-audio", + .data = &odroidu3_drvdata, + }, + { }, +}; +MODULE_DEVICE_TABLE(of, odroidx2_audio_of_match); + +static int odroidx2_audio_probe(struct platform_device *pdev) +{ + struct device_node *snd_node = pdev->dev.of_node; + struct snd_soc_card *card = &odroidx2; + struct device_node *i2s_node, *codec_node; + struct odroidx2_drv_data *dd; + const struct of_device_id *of_id; + int ret; + + of_id = of_match_node(odroidx2_audio_of_match, snd_node); + dd = (struct odroidx2_drv_data *)of_id->data; + + card->num_dapm_widgets = dd->num_dapm_widgets; + card->dapm_widgets = dd->dapm_widgets; + + card->dev = &pdev->dev; + + ret = snd_soc_of_parse_card_name(card, "samsung,model"); + if (ret < 0) + return ret; + + ret = snd_soc_of_parse_audio_routing(card, "samsung,audio-routing"); + if (ret < 0) + return ret; + + codec_node = of_parse_phandle(snd_node, "samsung,audio-codec", 0); + if (!codec_node) { + dev_err(&pdev->dev, + "Failed parsing samsung,i2s-codec property\n"); + return -EINVAL; + } + + i2s_node = of_parse_phandle(snd_node, "samsung,i2s-controller", 0); + if (!i2s_node) { + dev_err(&pdev->dev, + "Failed parsing samsung,i2s-controller property\n"); + ret = -EINVAL; + goto err_put_codec_n; + } + + odroidx2_dai[0].codec_of_node = codec_node; + odroidx2_dai[0].cpu_of_node = i2s_node; + odroidx2_dai[0].platform_of_node = i2s_node; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + goto err_put_i2s_n; + } + return 0; + +err_put_i2s_n: + of_node_put(i2s_node); +err_put_codec_n: + of_node_put(codec_node); + return ret; +} + +static int odroidx2_audio_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + + of_node_put((struct device_node *)odroidx2_dai[0].cpu_of_node); + of_node_put((struct device_node *)odroidx2_dai[0].codec_of_node); + + return 0; +} + +static struct platform_driver odroidx2_audio_driver = { + .driver = { + .name = "odroidx2-audio", + .owner = THIS_MODULE, + .of_match_table = odroidx2_audio_of_match, + .pm = &snd_soc_pm_ops, + }, + .probe = odroidx2_audio_probe, + .remove = odroidx2_audio_remove, +}; +module_platform_driver(odroidx2_audio_driver); + +MODULE_AUTHOR("Chen Zhen "); +MODULE_AUTHOR("Sylwester Nawrocki "); +MODULE_DESCRIPTION("ALSA SoC Odroid X2/U3 Audio Support"); +MODULE_LICENSE("GPL v2"); -- cgit v1.2.3 From 4714bc015de147c2da5248762b2772c6dbf160eb Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 9 Jul 2014 17:41:43 +0100 Subject: ASoC: arizona: Do not test ratio zero as it is not a valid setting Zero is not a valid FRATIO for the FLL, as such we shouldn't test it whilst refining the FRATIO. This patch does just that. Reported-by: Ryo Tsutsui Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 41b56ee6ff51..b03974eb541e 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1585,7 +1585,7 @@ static int arizona_calc_fratio(struct arizona_fll *fll, } } - for (ratio = init_ratio - 1; ratio >= 0; ratio--) { + for (ratio = init_ratio - 1; ratio > 0; ratio--) { if (ARIZONA_FLL_VCO_CORNER / (fll->vco_mult * ratio) < Fref) break; -- cgit v1.2.3 From 29fee829408117624efcd478f6f228d77039b969 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 9 Jul 2014 17:41:44 +0100 Subject: ASoC: arizona: Correct checking of FLL ratio limitations The check to ensure the Fref frequency is within the bounds for the current ratio, was placed in the wrong loop. The initial configuration will always be valid and the loop lowering the ratio will only reinforce this validity. The check should be on the loop increasing the ratio. This could on occasion cause an invalid ratio/Fref combination to be selected. Reported-by: Ryo Tsutsui Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index b03974eb541e..4c4727f0837c 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1578,6 +1578,10 @@ static int arizona_calc_fratio(struct arizona_fll *fll, while (div <= ARIZONA_FLL_MAX_REFDIV) { for (ratio = init_ratio; ratio <= ARIZONA_FLL_MAX_FRATIO; ratio++) { + if (ARIZONA_FLL_VCO_CORNER / (fll->vco_mult * ratio) < + Fref) + break; + if (target % (ratio * Fref)) { cfg->refdiv = refdiv; cfg->fratio = ratio - 1; @@ -1586,10 +1590,6 @@ static int arizona_calc_fratio(struct arizona_fll *fll, } for (ratio = init_ratio - 1; ratio > 0; ratio--) { - if (ARIZONA_FLL_VCO_CORNER / (fll->vco_mult * ratio) < - Fref) - break; - if (target % (ratio * Fref)) { cfg->refdiv = refdiv; cfg->fratio = ratio - 1; -- cgit v1.2.3 From 35a730a0790b2a6535e0b4a964c6a8b2d27fa019 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 9 Jul 2014 17:41:45 +0100 Subject: ASoC: arizona: Correct relationship between VCO corner and Fref When configuring the FLL we must ensure that the reference clock passed to the FLL is under a certain limit. This limit was specified incorrectly in the current code, this patch corrects this. Although the error will only be encountered in some edge cases. Reported-by: Ryo Tsutsui Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 4c4727f0837c..6084af76f337 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1578,8 +1578,8 @@ static int arizona_calc_fratio(struct arizona_fll *fll, while (div <= ARIZONA_FLL_MAX_REFDIV) { for (ratio = init_ratio; ratio <= ARIZONA_FLL_MAX_FRATIO; ratio++) { - if (ARIZONA_FLL_VCO_CORNER / (fll->vco_mult * ratio) < - Fref) + if ((ARIZONA_FLL_VCO_CORNER / 2) / + (fll->vco_mult * ratio) < Fref) break; if (target % (ratio * Fref)) { -- cgit v1.2.3 From 613124ce644c38857ef5780a96e6d400d6b11abd Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 9 Jul 2014 17:41:46 +0100 Subject: ASoC: arizona: Coding standards, remove unneeded brackets Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 6084af76f337..0a417e8bcadd 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1868,11 +1868,10 @@ int arizona_set_fll(struct arizona_fll *fll, int source, fll->sync_freq = Fref; fll->fout = Fout; - if (Fout) { + if (Fout) arizona_enable_fll(fll); - } else { + else arizona_disable_fll(fll); - } return 0; } -- cgit v1.2.3 From c393aca94f26e90290cd861743efbfbcd97c0316 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 9 Jul 2014 17:41:47 +0100 Subject: ASoC: arizona: Correct return value of arizona_is_enabled_fll arizona_is_enabled_fll currently returns a bool, but can throw an error. The error will be basically ignored and we will treat the FLL as already on. This patch changes the return to be an int and adds error code to propagate the error up to the callback. Reported-by: Anil Kumar Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 26 ++++++++++++++++---------- 1 file changed, 16 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 0a417e8bcadd..a6f09bb0a2e9 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1716,7 +1716,7 @@ static void arizona_apply_fll(struct arizona *arizona, unsigned int base, ARIZONA_FLL1_CTRL_UPD | cfg->n); } -static bool arizona_is_enabled_fll(struct arizona_fll *fll) +static int arizona_is_enabled_fll(struct arizona_fll *fll) { struct arizona *arizona = fll->arizona; unsigned int reg; @@ -1732,13 +1732,17 @@ static bool arizona_is_enabled_fll(struct arizona_fll *fll) return reg & ARIZONA_FLL1_ENA; } -static void arizona_enable_fll(struct arizona_fll *fll) +static int arizona_enable_fll(struct arizona_fll *fll) { struct arizona *arizona = fll->arizona; int ret; bool use_sync = false; + int already_enabled = arizona_is_enabled_fll(fll); struct arizona_fll_cfg cfg; + if (already_enabled < 0) + return already_enabled; + /* * If we have both REFCLK and SYNCCLK then enable both, * otherwise apply the SYNCCLK settings to REFCLK. @@ -1766,7 +1770,7 @@ static void arizona_enable_fll(struct arizona_fll *fll) ARIZONA_FLL1_SYNC_ENA, 0); } else { arizona_fll_err(fll, "No clocks provided\n"); - return; + return -EINVAL; } /* @@ -1781,7 +1785,7 @@ static void arizona_enable_fll(struct arizona_fll *fll) ARIZONA_FLL1_SYNC_BW, ARIZONA_FLL1_SYNC_BW); - if (!arizona_is_enabled_fll(fll)) + if (!already_enabled) pm_runtime_get(arizona->dev); /* Clear any pending completions */ @@ -1800,6 +1804,8 @@ static void arizona_enable_fll(struct arizona_fll *fll) msecs_to_jiffies(250)); if (ret == 0) arizona_fll_warn(fll, "Timed out waiting for lock\n"); + + return 0; } static void arizona_disable_fll(struct arizona_fll *fll) @@ -1821,7 +1827,7 @@ static void arizona_disable_fll(struct arizona_fll *fll) int arizona_set_fll_refclk(struct arizona_fll *fll, int source, unsigned int Fref, unsigned int Fout) { - int ret; + int ret = 0; if (fll->ref_src == source && fll->ref_freq == Fref) return 0; @@ -1836,17 +1842,17 @@ int arizona_set_fll_refclk(struct arizona_fll *fll, int source, fll->ref_freq = Fref; if (fll->fout && Fref > 0) { - arizona_enable_fll(fll); + ret = arizona_enable_fll(fll); } - return 0; + return ret; } EXPORT_SYMBOL_GPL(arizona_set_fll_refclk); int arizona_set_fll(struct arizona_fll *fll, int source, unsigned int Fref, unsigned int Fout) { - int ret; + int ret = 0; if (fll->sync_src == source && fll->sync_freq == Fref && fll->fout == Fout) @@ -1869,11 +1875,11 @@ int arizona_set_fll(struct arizona_fll *fll, int source, fll->fout = Fout; if (Fout) - arizona_enable_fll(fll); + ret = arizona_enable_fll(fll); else arizona_disable_fll(fll); - return 0; + return ret; } EXPORT_SYMBOL_GPL(arizona_set_fll); -- cgit v1.2.3 From 5e39a50bf8c644f2ef348037d3e3efdc03fa65b2 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 9 Jul 2014 17:41:48 +0100 Subject: ASoC: arizona: FLL freerun only required whilst disabling The FLL freerun is only required whilst we disable the FLL not the entire time the FLL is disabled. This patch moves the FLL freerun disable from the enable sequence to the disable sequence. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index a6f09bb0a2e9..7c362cb36b21 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1791,8 +1791,6 @@ static int arizona_enable_fll(struct arizona_fll *fll) /* Clear any pending completions */ try_wait_for_completion(&fll->ok); - regmap_update_bits_async(arizona->regmap, fll->base + 1, - ARIZONA_FLL1_FREERUN, 0); regmap_update_bits_async(arizona->regmap, fll->base + 1, ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA); if (use_sync) @@ -1819,6 +1817,8 @@ static void arizona_disable_fll(struct arizona_fll *fll) ARIZONA_FLL1_ENA, 0, &change); regmap_update_bits(arizona->regmap, fll->base + 0x11, ARIZONA_FLL1_SYNC_ENA, 0); + regmap_update_bits_async(arizona->regmap, fll->base + 1, + ARIZONA_FLL1_FREERUN, 0); if (change) pm_runtime_put_autosuspend(arizona->dev); -- cgit v1.2.3 From c8badda8eea6e9d142c8794519f6a57aa3ea3123 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 9 Jul 2014 17:41:49 +0100 Subject: ASoC: arizona: Update handling for input change on an active FLL Currently, the driver places no restrictions on changes that can be applied to an active FLL. However, it is only possible to change the input for an active FLL, to change the output the FLL should be stopped and then recofigured. This patch disallows changes in output frequency and adds some additional handling to ensure the output remains consistent across an input transition. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 7c362cb36b21..be3657ac52a4 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1500,6 +1500,12 @@ static int arizona_validate_fll(struct arizona_fll *fll, { unsigned int Fvco_min; + if (fll->fout && Fout != fll->fout) { + arizona_fll_err(fll, + "Can't change output on active FLL\n"); + return -EINVAL; + } + if (Fref / ARIZONA_FLL_MAX_REFDIV > ARIZONA_FLL_MAX_FREF) { arizona_fll_err(fll, "Can't scale %dMHz in to <=13.5MHz\n", @@ -1743,6 +1749,15 @@ static int arizona_enable_fll(struct arizona_fll *fll) if (already_enabled < 0) return already_enabled; + if (already_enabled) { + /* Facilitate smooth refclk across the transition */ + regmap_update_bits_async(fll->arizona->regmap, fll->base + 0x7, + ARIZONA_FLL1_GAIN_MASK, 0); + regmap_update_bits_async(fll->arizona->regmap, fll->base + 1, + ARIZONA_FLL1_FREERUN, + ARIZONA_FLL1_FREERUN); + } + /* * If we have both REFCLK and SYNCCLK then enable both, * otherwise apply the SYNCCLK settings to REFCLK. @@ -1798,6 +1813,10 @@ static int arizona_enable_fll(struct arizona_fll *fll) ARIZONA_FLL1_SYNC_ENA, ARIZONA_FLL1_SYNC_ENA); + if (already_enabled) + regmap_update_bits_async(arizona->regmap, fll->base + 1, + ARIZONA_FLL1_FREERUN, 0); + ret = wait_for_completion_timeout(&fll->ok, msecs_to_jiffies(250)); if (ret == 0) -- cgit v1.2.3 From 1b21572f8fd791f5b54e5989bc8e0cc5308d2e6c Mon Sep 17 00:00:00 2001 From: xujianqun Date: Fri, 11 Jul 2014 19:40:05 +0800 Subject: ASoC: rockchip: add missing module.h include MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Missing module.h in linux directory will cause compile error with randconfig, like as: sound/soc/rockchip/rockchip_i2s.c:526:20: error: expected declaration specifiers or ‘...’ before string constant MODULE_DESCRIPTION("ROCKCHIP IIS ASoC Interface"); Signed-off-by: xujianqun Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_i2s.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index af1d8eb94f04..663b1ed348db 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -10,6 +10,7 @@ * published by the Free Software Foundation. */ +#include #include #include #include -- cgit v1.2.3 From b97c60abf9a561f86ae71bd741add02673cc1a08 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Thu, 10 Jul 2014 18:11:13 +0200 Subject: ASoC: samsung-i2s: Maintain CDCLK settings across i2s_{shutdown/startup} Currently configuration of the CDCLK pad is being overwritten in the i2s_shutdown() callback in order to gate the SoC output clock. However if an ASoC machine driver doesn't restore that clock settings each time after opening the sound device this results in the CDCLK pin being permanently configured into input mode. I.e. the output clock will always stay disabled. Fix that by saving the CDCLKCON bit state in i2s_shutdown() and and restoring it in the i2s_startup() callback. Signed-off-by: Chen Zhen Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown --- sound/soc/samsung/i2s.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 5f9b255a8b38..d2533dbc8399 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -68,6 +68,8 @@ struct i2s_dai { #define DAI_OPENED (1 << 0) /* Dai is opened */ #define DAI_MANAGER (1 << 1) /* Dai is the manager */ unsigned mode; + /* CDCLK pin direction: 0 - input, 1 - output */ + unsigned int cdclk_out:1; /* Driver for this DAI */ struct snd_soc_dai_driver i2s_dai_drv; /* DMA parameters */ @@ -737,6 +739,9 @@ static int i2s_startup(struct snd_pcm_substream *substream, spin_unlock_irqrestore(&lock, flags); + if (!is_opened(other) && i2s->cdclk_out) + i2s_set_sysclk(dai, SAMSUNG_I2S_CDCLK, + 0, SND_SOC_CLOCK_OUT); return 0; } @@ -752,9 +757,13 @@ static void i2s_shutdown(struct snd_pcm_substream *substream, i2s->mode &= ~DAI_OPENED; i2s->mode &= ~DAI_MANAGER; - if (is_opened(other)) + if (is_opened(other)) { other->mode |= DAI_MANAGER; - + } else { + u32 mod = readl(i2s->addr + I2SMOD); + i2s->cdclk_out = !(mod & MOD_CDCLKCON); + other->cdclk_out = i2s->cdclk_out; + } /* Reset any constraint on RFS and BFS */ i2s->rfs = 0; i2s->bfs = 0; -- cgit v1.2.3 From 4c5258acfe7115382808a6697596b5efb91e2380 Mon Sep 17 00:00:00 2001 From: xujianqun Date: Sat, 12 Jul 2014 09:02:13 +0800 Subject: ASoC: rockchip: fix duplicated argument to && or || This resulted from an unfortunate copy/paste to similar DEFINE name. Detected automagically by kbuild. Signed-off-by: xujianqun Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_i2s.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 663b1ed348db..4203088518f0 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -89,7 +89,7 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) i2s->tx_start = false; regmap_update_bits(i2s->regmap, I2S_DMACR, - I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_ENABLE); + I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_DISABLE); if (!i2s->rx_start) { regmap_update_bits(i2s->regmap, I2S_XFER, @@ -99,8 +99,8 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) I2S_XFER_RXS_STOP); regmap_update_bits(i2s->regmap, I2S_CLR, - I2S_CLR_TXC | I2S_CLR_TXC, - I2S_CLR_TXC | I2S_CLR_TXC); + I2S_CLR_TXC | I2S_CLR_RXC, + I2S_CLR_TXC | I2S_CLR_RXC); regmap_read(i2s->regmap, I2S_CLR, &val); @@ -143,8 +143,8 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) I2S_XFER_RXS_STOP); regmap_update_bits(i2s->regmap, I2S_CLR, - I2S_CLR_TXC | I2S_CLR_TXC, - I2S_CLR_TXC | I2S_CLR_TXC); + I2S_CLR_TXC | I2S_CLR_RXC, + I2S_CLR_TXC | I2S_CLR_RXC); regmap_read(i2s->regmap, I2S_CLR, &val); -- cgit v1.2.3 From 9cb0fe9b0ee178474245a0a402eb628a3fb9a64a Mon Sep 17 00:00:00 2001 From: Himangi Saraogi Date: Fri, 11 Jul 2014 09:44:06 +0530 Subject: ASoC: omap-dmic: Use devm_clk_get This patch introduces the use of managed interfaces like devm_clk_get and does away with the clk_puts in the probe and remove functions. A label is also done away with. Signed-off-by: Himangi Saraogi Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/omap-dmic.c | 29 ++++++----------------------- 1 file changed, 6 insertions(+), 23 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c index f125eb95d8c8..0f34e28a3d55 100644 --- a/sound/soc/omap/omap-dmic.c +++ b/sound/soc/omap/omap-dmic.c @@ -466,7 +466,7 @@ static int asoc_dmic_probe(struct platform_device *pdev) mutex_init(&dmic->mutex); - dmic->fclk = clk_get(dmic->dev, "fck"); + dmic->fclk = devm_clk_get(dmic->dev, "fck"); if (IS_ERR(dmic->fclk)) { dev_err(dmic->dev, "cant get fck\n"); return -ENODEV; @@ -475,8 +475,7 @@ static int asoc_dmic_probe(struct platform_device *pdev) res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "dma"); if (!res) { dev_err(dmic->dev, "invalid dma memory resource\n"); - ret = -ENODEV; - goto err_put_clk; + return -ENODEV; } dmic->dma_data.addr = res->start + OMAP_DMIC_DATA_REG; @@ -484,34 +483,19 @@ static int asoc_dmic_probe(struct platform_device *pdev) res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu"); dmic->io_base = devm_ioremap_resource(&pdev->dev, res); - if (IS_ERR(dmic->io_base)) { - ret = PTR_ERR(dmic->io_base); - goto err_put_clk; - } + if (IS_ERR(dmic->io_base)) + return PTR_ERR(dmic->io_base); ret = devm_snd_soc_register_component(&pdev->dev, &omap_dmic_component, &omap_dmic_dai, 1); if (ret) - goto err_put_clk; + return ret; ret = omap_pcm_platform_register(&pdev->dev); if (ret) - goto err_put_clk; - - return 0; - -err_put_clk: - clk_put(dmic->fclk); - return ret; -} - -static int asoc_dmic_remove(struct platform_device *pdev) -{ - struct omap_dmic *dmic = platform_get_drvdata(pdev); - - clk_put(dmic->fclk); + return ret; return 0; } @@ -529,7 +513,6 @@ static struct platform_driver asoc_dmic_driver = { .of_match_table = omap_dmic_of_match, }, .probe = asoc_dmic_probe, - .remove = asoc_dmic_remove, }; module_platform_driver(asoc_dmic_driver); -- cgit v1.2.3 From d21e1b160561d2628af555aaf803c3d0ad28ed1a Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Fri, 11 Jul 2014 15:45:04 +0200 Subject: ASoC: samsung: add explicit i2c/spi dependencies I got another build error from SND_SOC_SMARTQ selecting SND_SOC_WM8750 without ensuring that I2C is enabled, in this example, i2c is a loadable module: sound/built-in.o: In function `wm8750_i2c_probe': :(.text+0x3e6c0): undefined reference to `devm_regmap_init_i2c' sound/built-in.o: In function `wm8750_exit': :(.exit.text+0x3f4): undefined reference to `i2c_del_driver' sound/built-in.o: In function `wm8750_modinit': :(.init.text+0x13d0): undefined reference to `i2c_register_driver' This changes the samsund ASoC Kconfig to have explicit I2C dependencies for each driver that uses an i2c bus and SPI dependencies for the ones using those. Signed-off-by: Arnd Bergmann Signed-off-by: Mark Brown --- sound/soc/samsung/Kconfig | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 333d645cea87..2c3a04bbc092 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -49,7 +49,7 @@ config SND_SOC_SAMSUNG_NEO1973_WM8753 config SND_SOC_SAMSUNG_JIVE_WM8750 tristate "SoC I2S Audio support for Jive" - depends on SND_SOC_SAMSUNG && MACH_JIVE + depends on SND_SOC_SAMSUNG && MACH_JIVE && I2C select SND_SOC_WM8750 select SND_S3C2412_SOC_I2S help @@ -148,7 +148,7 @@ config SND_SOC_SAMSUNG_SMDK_WM9713 config SND_SOC_SMARTQ tristate "SoC I2S Audio support for SmartQ board" - depends on SND_SOC_SAMSUNG && MACH_SMARTQ + depends on SND_SOC_SAMSUNG && MACH_SMARTQ && I2C select SND_SAMSUNG_I2S select SND_SOC_WM8750 @@ -200,7 +200,7 @@ config SND_SOC_SPEYSIDE config SND_SOC_TOBERMORY tristate "Audio support for Wolfson Tobermory" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && INPUT + depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && INPUT && I2C select SND_SAMSUNG_I2S select SND_SOC_WM8962 @@ -216,7 +216,7 @@ config SND_SOC_BELLS config SND_SOC_LOWLAND tristate "Audio support for Wolfson Lowland" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 + depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && I2C select SND_SAMSUNG_I2S select SND_SOC_WM5100 select SND_SOC_WM9081 -- cgit v1.2.3 From 3a549fbffcd141b27afa7b160d67993dd40a596f Mon Sep 17 00:00:00 2001 From: Kukjin Kim Date: Thu, 3 Jul 2014 07:40:17 +0900 Subject: ASoC: samsung: no more support for S5P6440 and S5P6450 SoCs This patch removes s5p64x0 related WM8580 because of removing support for s5p64x0 SoCs. Signed-off-by: Kukjin Kim [pebolle@tiscali.nl: reported missing dependency for smdk6450] Reported-by: Paul Bolle Signed-off-by: Mark Brown --- sound/soc/samsung/Kconfig | 4 ++-- sound/soc/samsung/smdk_wm8580pcm.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 2c3a04bbc092..91ee5023668b 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -57,7 +57,7 @@ config SND_SOC_SAMSUNG_JIVE_WM8750 config SND_SOC_SAMSUNG_SMDK_WM8580 tristate "SoC I2S Audio support for WM8580 on SMDK" - depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDK6440 || MACH_SMDK6450 || MACH_SMDKV210 || MACH_SMDKC110) + depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110) depends on REGMAP_I2C select SND_SOC_WM8580 select SND_SAMSUNG_I2S @@ -172,7 +172,7 @@ config SND_SOC_SAMSUNG_SMDK_SPDIF config SND_SOC_SMDK_WM8580_PCM tristate "SoC PCM Audio support for WM8580 on SMDK" - depends on SND_SOC_SAMSUNG && (MACH_SMDK6450 || MACH_SMDKV210 || MACH_SMDKC110) + depends on SND_SOC_SAMSUNG && (MACH_SMDKV210 || MACH_SMDKC110) depends on REGMAP_I2C select SND_SOC_WM8580 select SND_SAMSUNG_PCM diff --git a/sound/soc/samsung/smdk_wm8580pcm.c b/sound/soc/samsung/smdk_wm8580pcm.c index e119aaa91c28..63d079303561 100644 --- a/sound/soc/samsung/smdk_wm8580pcm.c +++ b/sound/soc/samsung/smdk_wm8580pcm.c @@ -25,7 +25,7 @@ * o '0' means 'OFF' * o 'X' means 'Don't care' * - * SMDK6410, SMDK6440, SMDK6450 Base B/D: CFG1-0000, CFG2-1111 + * SMDK6410 Base B/D: CFG1-0000, CFG2-1111 * SMDKC110, SMDKV210: CFGB11-100100, CFGB12-0000 */ -- cgit v1.2.3 From 8e89761876611f06ef4be865b4780b4361caf4af Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Mon, 14 Jul 2014 17:37:36 +0800 Subject: ASoC: Intel: Cleanup HSW pcm format support This change removes unsupported formats from System, Capture and Loopback FE DAIs. Also it fixes S24_LE support on all DAIs. While at this fix 24 bit flag for BYT as well. Signed-off-by: Jie Yang Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/intel/sst-baytrail-pcm.c | 2 +- sound/soc/intel/sst-haswell-pcm.c | 27 ++++++++++++++++++--------- 2 files changed, 19 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-baytrail-pcm.c b/sound/soc/intel/sst-baytrail-pcm.c index 8eab97368ea7..599401c0c655 100644 --- a/sound/soc/intel/sst-baytrail-pcm.c +++ b/sound/soc/intel/sst-baytrail-pcm.c @@ -32,7 +32,7 @@ static const struct snd_pcm_hardware sst_byt_pcm_hardware = { SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FORMAT_S24_LE, + SNDRV_PCM_FMTBIT_S24_LE, .period_bytes_min = 384, .period_bytes_max = 48000, .periods_min = 2, diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 058efb17c568..61bf6da4bb02 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -80,7 +80,7 @@ static const struct snd_pcm_hardware hsw_pcm_hardware = { SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE | + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE, .period_bytes_min = PAGE_SIZE, .period_bytes_max = (HSW_PCM_PERIODS_MAX / HSW_PCM_PERIODS_MIN) * PAGE_SIZE, @@ -400,7 +400,15 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream, sst_hsw_stream_set_valid(hsw, pcm_data->stream, 16); break; case SNDRV_PCM_FORMAT_S24_LE: - bits = SST_HSW_DEPTH_24BIT; + bits = SST_HSW_DEPTH_32BIT; + sst_hsw_stream_set_valid(hsw, pcm_data->stream, 24); + break; + case SNDRV_PCM_FORMAT_S8: + bits = SST_HSW_DEPTH_8BIT; + sst_hsw_stream_set_valid(hsw, pcm_data->stream, 8); + break; + case SNDRV_PCM_FORMAT_S32_LE: + bits = SST_HSW_DEPTH_32BIT; sst_hsw_stream_set_valid(hsw, pcm_data->stream, 32); break; default: @@ -685,8 +693,9 @@ static int hsw_pcm_new(struct snd_soc_pcm_runtime *rtd) } #define HSW_FORMATS \ - (SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S16_LE |\ - SNDRV_PCM_FMTBIT_S32_LE) + (SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S8) static struct snd_soc_dai_driver hsw_dais[] = { { @@ -696,7 +705,7 @@ static struct snd_soc_dai_driver hsw_dais[] = { .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, }, }, { @@ -727,8 +736,8 @@ static struct snd_soc_dai_driver hsw_dais[] = { .stream_name = "Loopback Capture", .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, - .formats = HSW_FORMATS, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, }, }, { @@ -737,8 +746,8 @@ static struct snd_soc_dai_driver hsw_dais[] = { .stream_name = "Analog Capture", .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, - .formats = HSW_FORMATS, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, }, }, }; -- cgit v1.2.3 From c761b58306b26ea78fa91488cbb04c61b35d583d Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Mon, 14 Jul 2014 17:11:11 +0800 Subject: ASoC: Intel: Add dummy read for SRAM block enable Add dummy read after each block enable, to workaround SRAM write missing bytes issue. Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-dsp.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c index 535f517629fd..a33b931181dc 100644 --- a/sound/soc/intel/sst-haswell-dsp.c +++ b/sound/soc/intel/sst-haswell-dsp.c @@ -359,6 +359,17 @@ static u32 hsw_block_get_bit(struct sst_mem_block *block) return bit; } +/*dummy read a SRAM block.*/ +static void sst_mem_block_dummy_read(struct sst_mem_block *block) +{ + u32 size; + u8 tmp_buf[4]; + struct sst_dsp *sst = block->dsp; + + size = block->size > 4 ? 4 : block->size; + memcpy_fromio(tmp_buf, sst->addr.lpe + block->offset, size); +} + /* enable 32kB memory block - locks held by caller */ static int hsw_block_enable(struct sst_mem_block *block) { @@ -378,6 +389,8 @@ static int hsw_block_enable(struct sst_mem_block *block) /* wait 18 DSP clock ticks */ udelay(10); + /*add a dummy read before the SRAM block is written, otherwise the writing may miss bytes sometimes.*/ + sst_mem_block_dummy_read(block); return 0; } -- cgit v1.2.3 From 15446c0b8dc79f5dfabfb689879609023713f421 Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Mon, 14 Jul 2014 17:11:09 +0800 Subject: ASoC: Intel: Merge wild cat point ADSP DRAM regions Merge D-SRAM0 D-SRAM1 D-SRAM2 to D-SRAM, for wild cat point ADSP mem regions. Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-dsp.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c index 535f517629fd..4720382260b0 100644 --- a/sound/soc/intel/sst-haswell-dsp.c +++ b/sound/soc/intel/sst-haswell-dsp.c @@ -313,9 +313,7 @@ static const struct sst_adsp_memregion lp_region[] = { /* wild cat point ADSP mem regions */ static const struct sst_adsp_memregion wpt_region[] = { - {0x00000, 0x40000, 8, SST_MEM_DRAM}, /* D-SRAM0 - 8 * 32kB */ - {0x40000, 0x80000, 8, SST_MEM_DRAM}, /* D-SRAM1 - 8 * 32kB */ - {0x80000, 0xA0000, 4, SST_MEM_DRAM}, /* D-SRAM2 - 4 * 32kB */ + {0x00000, 0xA0000, 20, SST_MEM_DRAM}, /* D-SRAM0,D-SRAM1,D-SRAM2 - 20 * 32kB */ {0xA0000, 0xF0000, 10, SST_MEM_IRAM}, /* I-SRAM - 10 * 32kB */ }; -- cgit v1.2.3 From 548793232fd29cfa1553bb45247aa5963632405c Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Mon, 14 Jul 2014 17:11:10 +0800 Subject: ASoC: Intel: Use a table for ADSP SRAM shift Use a table for ADSP IRAM/DRAM bit shift. Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-dsp.c | 39 +++++++++++++++++++++++++++++---------- 1 file changed, 29 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c index 4720382260b0..40bb0205d5c0 100644 --- a/sound/soc/intel/sst-haswell-dsp.c +++ b/sound/soc/intel/sst-haswell-dsp.c @@ -337,21 +337,40 @@ static int hsw_acpi_resource_map(struct sst_dsp *sst, struct sst_pdata *pdata) return 0; } +struct sst_sram_shift { + u32 dev_id; /* SST Device IDs */ + u32 iram_shift; + u32 dram_shift; +}; + +static const struct sst_sram_shift sram_shift[] = { + {SST_DEV_ID_LYNX_POINT, 6, 16}, /* lp */ + {SST_DEV_ID_WILDCAT_POINT, 2, 12}, /* wpt */ +}; static u32 hsw_block_get_bit(struct sst_mem_block *block) { - u32 bit = 0, shift = 0; + u32 bit = 0, shift = 0, index; + struct sst_dsp *sst = block->dsp; - switch (block->type) { - case SST_MEM_DRAM: - shift = 16; - break; - case SST_MEM_IRAM: - shift = 6; - break; - default: - return 0; + for (index = 0; index < ARRAY_SIZE(sram_shift); index++) { + if (sram_shift[index].dev_id == sst->id) + break; } + if (index < ARRAY_SIZE(sram_shift)) { + switch (block->type) { + case SST_MEM_DRAM: + shift = sram_shift[index].dram_shift; + break; + case SST_MEM_IRAM: + shift = sram_shift[index].iram_shift; + break; + default: + shift = 0; + } + } else + shift = 0; + bit = 1 << (block->index + shift); return bit; -- cgit v1.2.3 From 85e63007bbef7abc7145c807ed59d01738e09d39 Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Mon, 14 Jul 2014 17:11:12 +0800 Subject: ASoC: Intel: Start with all memory banks disabled All required banks are enabled during boot procedure. Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-dsp.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c index 40bb0205d5c0..977e29779d11 100644 --- a/sound/soc/intel/sst-haswell-dsp.c +++ b/sound/soc/intel/sst-haswell-dsp.c @@ -505,8 +505,9 @@ static int hsw_init(struct sst_dsp *sst, struct sst_pdata *pdata) } } - /* set default power gating mask */ - writel(0x0, sst->addr.pci_cfg + SST_VDRTCTL0); + /* set default power gating control, enable power gating control for all blocks. that is, + can't be accessed, please enable each block before accessing. */ + writel(0xffffffff, sst->addr.pci_cfg + SST_VDRTCTL0); return 0; } -- cgit v1.2.3 From 23c4fd5c9719e8fc60d589b9f9c7451120f4f3e9 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Mon, 14 Jul 2014 10:18:04 +0800 Subject: ASoC: rt286: make rt286_i2c_driver static Signed-off-by: Fengguang Wu Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt286.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 81033154a412..218f86efd196 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1206,7 +1206,7 @@ static int rt286_i2c_remove(struct i2c_client *i2c) } -struct i2c_driver rt286_i2c_driver = { +static struct i2c_driver rt286_i2c_driver = { .driver = { .name = "rt286", .owner = THIS_MODULE, -- cgit v1.2.3 From 1ad0e33060a64121c1c7acb7ec21a4fdef4aaed6 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Wed, 9 Jul 2014 14:57:49 +0530 Subject: ASoC: Intel: add sst shim register start-end variables the shim registers start and end can be useful while parsing the shim addresses, so add these Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-dsp.h | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h index e44423be66c4..967fb32c981d 100644 --- a/sound/soc/intel/sst-dsp.h +++ b/sound/soc/intel/sst-dsp.h @@ -53,6 +53,10 @@ #define SST_CSR2 0x80 #define SST_LTRC 0xE0 #define SST_HDMC 0xE8 + +#define SST_SHIM_BEGIN SST_CSR +#define SST_SHIM_END SST_HDMC + #define SST_DBGO 0xF0 #define SST_SHIM_SIZE 0x100 -- cgit v1.2.3 From 8813e66db73bfac940bfd08c31592365f0a56c74 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Wed, 9 Jul 2014 14:57:50 +0530 Subject: ASoC: Intel: mfld: add dsp error codes DSP returns error codes for IPC return so add them in driver Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-dsp.h | 15 +++++++++++++++ 1 file changed, 15 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/sst-mfld-dsp.h b/sound/soc/intel/sst-mfld-dsp.h index 2c887855e7d8..4257263157cd 100644 --- a/sound/soc/intel/sst-mfld-dsp.h +++ b/sound/soc/intel/sst-mfld-dsp.h @@ -171,6 +171,21 @@ enum stream_type { SST_STREAM_TYPE_MUSIC = 1, }; +enum sst_error_codes { + /* Error code,response to msgId: Description */ + /* Common error codes */ + SST_SUCCESS = 0, /* Success */ + SST_ERR_INVALID_STREAM_ID = 1, + SST_ERR_INVALID_MSG_ID = 2, + SST_ERR_INVALID_STREAM_OP = 3, + SST_ERR_INVALID_PARAMS = 4, + SST_ERR_INVALID_CODEC = 5, + SST_ERR_INVALID_MEDIA_TYPE = 6, + SST_ERR_STREAM_ERR = 7, + + SST_ERR_STREAM_IN_USE = 15, +}; + struct ipc_dsp_hdr { u16 mod_index_id:8; /*!< DSP Command ID specific to tasks */ u16 pipe_id:8; /*!< instance of the module in the pipeline */ -- cgit v1.2.3 From e310fb9141436672db680923228fba5aab206062 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Wed, 9 Jul 2014 14:57:51 +0530 Subject: ASoC: Intel: mfld: add generic parameter interface This interface will be used by subsequent patches to set/get parameters from DSP Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-platform.h | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index 9dc962ff1e1d..6c6a42c08e24 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -63,7 +63,9 @@ enum sst_controls { SST_SND_BUFFER_POINTER = 0x05, SST_SND_STREAM_INIT = 0x06, SST_SND_START = 0x07, - SST_MAX_CONTROLS = 0x07, + SST_SET_BYTE_STREAM = 0x100A, + SST_GET_BYTE_STREAM = 0x100B, + SST_MAX_CONTROLS = SST_GET_BYTE_STREAM, }; enum sst_stream_ops { @@ -127,6 +129,7 @@ struct compress_sst_ops { struct sst_ops { int (*open) (struct snd_sst_params *str_param); int (*device_control) (int cmd, void *arg); + int (*set_generic_params)(enum sst_controls cmd, void *arg); int (*close) (unsigned int str_id); }; -- cgit v1.2.3 From 8f501c7b4414d46a3ea6650f962b0fe9a69d9787 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Fri, 11 Jul 2014 15:45:07 +0200 Subject: ASoC: samsung: s3c24xx dmaengine follow-up Commit ae602456e83c92 ("ASoC: samsung: drop support for legacy S3C24XX DMA API") removed the old code for the samsung specific DMA interfaces, now that everybody can use dmaengine. This picks up the few remaining pieces left over by that patch: The most important one is the removal of the dma_data->ops->started() calls in ac97. My understanding is that these are only required for drivers that do not support cyclic transfers, which the new dma engine driver now does, so we can simply remove them. This would also fix at least one bug in the ac97 driver on newer machines, which currently gives us a NULL pointer dereference from trying to call dma_data->ops->started(). Further, we must no longer 'select' S3C2410_DMA, which conflicts with the dmaengine driver. The SND_S3C_DMA symbol is now useless, because it is always selected, so we can remove it and build the dmaengine support unconditionally. Finally, we should not 'select' S3C24XX_DMAC or S3C64XX_PL080, which may have additional dependencies. This replaces it with 'depends on', to be more conservative. Signed-off-by: Arnd Bergmann Signed-off-by: Mark Brown --- sound/soc/samsung/Kconfig | 10 ++-------- sound/soc/samsung/Makefile | 2 +- sound/soc/samsung/ac97.c | 17 ----------------- 3 files changed, 3 insertions(+), 26 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 91ee5023668b..55a38697443d 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -1,18 +1,14 @@ config SND_SOC_SAMSUNG tristate "ASoC support for Samsung" depends on PLAT_SAMSUNG - select S3C24XX_DMAC if ARCH_S3C24XX - select S3C64XX_PL080 if ARCH_S3C64XX - select SND_S3C_DMA + depends on S3C64XX_PL080 || !ARCH_S3C64XX + depends on S3C24XX_DMAC || !ARCH_S3C24XX select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for codecs attached to the Samsung SoCs' Audio interfaces. You will also need to select the audio interfaces to support below. -config SND_S3C_DMA - tristate - config SND_S3C24XX_I2S tristate @@ -77,7 +73,6 @@ config SND_SOC_SAMSUNG_SMDK_WM8994 config SND_SOC_SAMSUNG_SMDK2443_WM9710 tristate "SoC AC97 Audio support for SMDK2443 - WM9710" depends on SND_SOC_SAMSUNG && MACH_SMDK2443 - select S3C2410_DMA select AC97_BUS select SND_SOC_AC97_CODEC select SND_SAMSUNG_AC97 @@ -88,7 +83,6 @@ config SND_SOC_SAMSUNG_SMDK2443_WM9710 config SND_SOC_SAMSUNG_LN2440SBC_ALC650 tristate "SoC AC97 Audio support for LN2440SBC - ALC650" depends on SND_SOC_SAMSUNG && ARCH_S3C24XX - select S3C2410_DMA select AC97_BUS select SND_SOC_AC97_CODEC select SND_SAMSUNG_AC97 diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index e8d9ccdc41fd..91505ddaaf95 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -9,7 +9,7 @@ snd-soc-samsung-spdif-objs := spdif.o snd-soc-pcm-objs := pcm.o snd-soc-i2s-objs := i2s.o -obj-$(CONFIG_SND_S3C_DMA) += snd-soc-s3c-dma.o +obj-$(CONFIG_SND_SOC_SAMSUNG) += snd-soc-s3c-dma.o obj-$(CONFIG_SND_S3C24XX_I2S) += snd-soc-s3c24xx-i2s.o obj-$(CONFIG_SND_SAMSUNG_AC97) += snd-soc-ac97.o obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 68d9303047e8..2aa24d052a4a 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -19,7 +19,6 @@ #include -#include #include "regs-ac97.h" #include @@ -225,9 +224,6 @@ static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { u32 ac_glbctrl; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c_dma_params *dma_data = - snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) @@ -253,11 +249,6 @@ static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - if (!dma_data->ops) - dma_data->ops = samsung_dma_get_ops(); - - dma_data->ops->started(dma_data->channel); - return 0; } @@ -265,9 +256,6 @@ static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { u32 ac_glbctrl; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c_dma_params *dma_data = - snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); ac_glbctrl &= ~S3C_AC97_GLBCTRL_MICINTM_MASK; @@ -287,11 +275,6 @@ static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - if (!dma_data->ops) - dma_data->ops = samsung_dma_get_ops(); - - dma_data->ops->started(dma_data->channel); - return 0; } -- cgit v1.2.3 From dd2380e31cca0099aa49846e16103936aa9d818a Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Fri, 11 Jul 2014 15:45:08 +0200 Subject: ASoC: samsung: remove unused DMA data The s3c_dma_client structures and the 'ch' and 'ops' members in s3c_dma_params were only used by the legacy DMA driver and serve no function any more. This removes any reference to them. Signed-off-by: Arnd Bergmann Signed-off-by: Mark Brown --- sound/soc/samsung/ac97.c | 15 --------------- sound/soc/samsung/dma.h | 7 ------- sound/soc/samsung/i2s.c | 6 ------ sound/soc/samsung/pcm.c | 12 ------------ sound/soc/samsung/s3c-i2s-v2.c | 2 -- sound/soc/samsung/s3c2412-i2s.c | 4 ---- sound/soc/samsung/s3c24xx-i2s.c | 4 ---- sound/soc/samsung/spdif.c | 5 ----- 8 files changed, 55 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 2aa24d052a4a..e1615113fd84 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -38,30 +38,15 @@ struct s3c_ac97_info { }; static struct s3c_ac97_info s3c_ac97; -static struct s3c_dma_client s3c_dma_client_out = { - .name = "AC97 PCMOut" -}; - -static struct s3c_dma_client s3c_dma_client_in = { - .name = "AC97 PCMIn" -}; - -static struct s3c_dma_client s3c_dma_client_micin = { - .name = "AC97 MicIn" -}; - static struct s3c_dma_params s3c_ac97_pcm_out = { - .client = &s3c_dma_client_out, .dma_size = 4, }; static struct s3c_dma_params s3c_ac97_pcm_in = { - .client = &s3c_dma_client_in, .dma_size = 4, }; static struct s3c_dma_params s3c_ac97_mic_in = { - .client = &s3c_dma_client_micin, .dma_size = 4, }; diff --git a/sound/soc/samsung/dma.h b/sound/soc/samsung/dma.h index 070ab0f09609..0e85dcfec023 100644 --- a/sound/soc/samsung/dma.h +++ b/sound/soc/samsung/dma.h @@ -14,17 +14,10 @@ #include -struct s3c_dma_client { - char *name; -}; - struct s3c_dma_params { - struct s3c_dma_client *client; /* stream identifier */ int channel; /* Channel ID */ dma_addr_t dma_addr; int dma_size; /* Size of the DMA transfer */ - unsigned ch; - struct samsung_dma_ops *ops; char *ch_name; struct snd_dmaengine_dai_dma_data dma_data; }; diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 2ac76fa3e742..6b9de7429884 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1216,11 +1216,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) pri_dai->dma_playback.dma_addr = regs_base + I2STXD; pri_dai->dma_capture.dma_addr = regs_base + I2SRXD; - pri_dai->dma_playback.client = - (struct s3c_dma_client *)&pri_dai->dma_playback; pri_dai->dma_playback.ch_name = "tx"; - pri_dai->dma_capture.client = - (struct s3c_dma_client *)&pri_dai->dma_capture; pri_dai->dma_capture.ch_name = "rx"; pri_dai->dma_playback.dma_size = 4; pri_dai->dma_capture.dma_size = 4; @@ -1238,8 +1234,6 @@ static int samsung_i2s_probe(struct platform_device *pdev) goto err; } sec_dai->dma_playback.dma_addr = regs_base + I2STXDS; - sec_dai->dma_playback.client = - (struct s3c_dma_client *)&sec_dai->dma_playback; sec_dai->dma_playback.ch_name = "tx-sec"; if (!np) { diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index 4c5f97fe45c8..bac034b15a27 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -131,32 +131,20 @@ struct s3c_pcm_info { struct s3c_dma_params *dma_capture; }; -static struct s3c_dma_client s3c_pcm_dma_client_out = { - .name = "PCM Stereo out" -}; - -static struct s3c_dma_client s3c_pcm_dma_client_in = { - .name = "PCM Stereo in" -}; - static struct s3c_dma_params s3c_pcm_stereo_out[] = { [0] = { - .client = &s3c_pcm_dma_client_out, .dma_size = 4, }, [1] = { - .client = &s3c_pcm_dma_client_out, .dma_size = 4, }, }; static struct s3c_dma_params s3c_pcm_stereo_in[] = { [0] = { - .client = &s3c_pcm_dma_client_in, .dma_size = 4, }, [1] = { - .client = &s3c_pcm_dma_client_in, .dma_size = 4, }, }; diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index de6c321b8b68..df65c5b494b1 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -22,8 +22,6 @@ #include #include -#include - #include "regs-i2s-v2.h" #include "s3c-i2s-v2.h" #include "dma.h" diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index d6bc6dc0aafb..9180310e862a 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -34,16 +34,12 @@ #include "s3c2412-i2s.h" static struct s3c_dma_params s3c2412_i2s_pcm_stereo_out = { - .client = - (struct s3c_dma_client *)&s3c2412_i2s_pcm_stereo_out, .channel = DMACH_I2S_OUT, .ch_name = "tx", .dma_size = 4, }; static struct s3c_dma_params s3c2412_i2s_pcm_stereo_in = { - .client = - (struct s3c_dma_client *)&s3c2412_i2s_pcm_stereo_in, .channel = DMACH_I2S_IN, .ch_name = "rx", .dma_size = 4, diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index e8b98528e356..e87d9a2053b8 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -32,16 +32,12 @@ #include "s3c24xx-i2s.h" static struct s3c_dma_params s3c24xx_i2s_pcm_stereo_out = { - .client = - (struct s3c_dma_client *)&s3c24xx_i2s_pcm_stereo_out, .channel = DMACH_I2S_OUT, .ch_name = "tx", .dma_size = 2, }; static struct s3c_dma_params s3c24xx_i2s_pcm_stereo_in = { - .client = - (struct s3c_dma_client *)&s3c24xx_i2s_pcm_stereo_in, .channel = DMACH_I2S_IN, .ch_name = "rx", .dma_size = 2, diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c index d9ffc48fce5e..d7d2e208f486 100644 --- a/sound/soc/samsung/spdif.c +++ b/sound/soc/samsung/spdif.c @@ -93,10 +93,6 @@ struct samsung_spdif_info { struct s3c_dma_params *dma_playback; }; -static struct s3c_dma_client spdif_dma_client_out = { - .name = "S/PDIF Stereo out", -}; - static struct s3c_dma_params spdif_stereo_out; static struct samsung_spdif_info spdif_info; @@ -435,7 +431,6 @@ static int spdif_probe(struct platform_device *pdev) } spdif_stereo_out.dma_size = 2; - spdif_stereo_out.client = &spdif_dma_client_out; spdif_stereo_out.dma_addr = mem_res->start + DATA_OUTBUF; spdif_stereo_out.channel = dma_res->start; -- cgit v1.2.3 From 0cea76f3393782d67ccea8f07e9abf341bc4f60e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 15 Jul 2014 16:31:01 +0200 Subject: ALSA: control: Define SNDRV_CTL_TLV_OP_* constants Instead of hard-coded magic numbers, define constants for op_flag to tlv callbacks. Signed-off-by: Takashi Iwai --- include/sound/control.h | 7 ++++++- sound/core/control.c | 6 +++--- 2 files changed, 9 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/include/sound/control.h b/include/sound/control.h index 5358892b1b39..042613938a1d 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -31,10 +31,15 @@ typedef int (snd_kcontrol_info_t) (struct snd_kcontrol * kcontrol, struct snd_ct typedef int (snd_kcontrol_get_t) (struct snd_kcontrol * kcontrol, struct snd_ctl_elem_value * ucontrol); typedef int (snd_kcontrol_put_t) (struct snd_kcontrol * kcontrol, struct snd_ctl_elem_value * ucontrol); typedef int (snd_kcontrol_tlv_rw_t)(struct snd_kcontrol *kcontrol, - int op_flag, /* 0=read,1=write,-1=command */ + int op_flag, /* SNDRV_CTL_TLV_OP_XXX */ unsigned int size, unsigned int __user *tlv); +enum { + SNDRV_CTL_TLV_OP_READ = 0, + SNDRV_CTL_TLV_OP_WRITE = 1, + SNDRV_CTL_TLV_OP_CMD = -1, +}; struct snd_kcontrol_new { snd_ctl_elem_iface_t iface; /* interface identifier */ diff --git a/sound/core/control.c b/sound/core/control.c index f0b0e14497a5..b9611344ff9e 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1406,11 +1406,11 @@ static long snd_ctl_ioctl(struct file *file, unsigned int cmd, unsigned long arg case SNDRV_CTL_IOCTL_SUBSCRIBE_EVENTS: return snd_ctl_subscribe_events(ctl, ip); case SNDRV_CTL_IOCTL_TLV_READ: - return snd_ctl_tlv_ioctl(ctl, argp, 0); + return snd_ctl_tlv_ioctl(ctl, argp, SNDRV_CTL_TLV_OP_READ); case SNDRV_CTL_IOCTL_TLV_WRITE: - return snd_ctl_tlv_ioctl(ctl, argp, 1); + return snd_ctl_tlv_ioctl(ctl, argp, SNDRV_CTL_TLV_OP_WRITE); case SNDRV_CTL_IOCTL_TLV_COMMAND: - return snd_ctl_tlv_ioctl(ctl, argp, -1); + return snd_ctl_tlv_ioctl(ctl, argp, SNDRV_CTL_TLV_OP_CMD); case SNDRV_CTL_IOCTL_POWER: return -ENOPROTOOPT; case SNDRV_CTL_IOCTL_POWER_STATE: -- cgit v1.2.3 From 1d34f3ef4b6cd33c74b414df74b41a4606d1a306 Mon Sep 17 00:00:00 2001 From: Lv Zheng Date: Wed, 16 Jul 2014 16:59:02 +0800 Subject: ASoC: intel: Cleanup useless ACPI inclusion. The sst-haswell-dsp.c is an ACPI independent file, this patch removes ACPI header files for it. Signed-off-by: Lv Zheng Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-dsp.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c index 977e29779d11..7b8ad9923fe4 100644 --- a/sound/soc/intel/sst-haswell-dsp.c +++ b/sound/soc/intel/sst-haswell-dsp.c @@ -28,9 +28,6 @@ #include #include -#include -#include - #include "sst-dsp.h" #include "sst-dsp-priv.h" #include "sst-haswell-ipc.h" -- cgit v1.2.3 From 24600b54724ca8c7acc2d5b33b3178bb3f7a5c29 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 16 Jul 2014 15:12:02 +0300 Subject: ASoC: edma-pcm: Remove PCM_INFO_BATCH and add PCM_INFO_NO_PERIOD_WAKEUP flag The SNDRV_PCM_INFO_BATCH will be added back based on the dmaengine driver's capabilities. Patches for edma dmaengine driver has been prepared to suppress the interrupts. We can add this flag right away and have the benefit of not having interrupts during audio activity when the edma and ASoC patches are in the same tree. Signed-off-by: Peter Ujfalusi Tested-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/davinci/edma-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/davinci/edma-pcm.c b/sound/soc/davinci/edma-pcm.c index d38afb1c61ae..605e643133db 100644 --- a/sound/soc/davinci/edma-pcm.c +++ b/sound/soc/davinci/edma-pcm.c @@ -28,8 +28,8 @@ static const struct snd_pcm_hardware edma_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_BATCH | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_NO_PERIOD_WAKEUP | SNDRV_PCM_INFO_INTERLEAVED, .buffer_bytes_max = 128 * 1024, .period_bytes_min = 32, -- cgit v1.2.3 From e8ffacee0adacc1a1312a14acb123544cd4094b9 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 16 Jul 2014 15:12:03 +0300 Subject: ASoC: edma-pcm: Add empty functions for !CONFIG_SND_EDMA_SOC builds So drivers mixing with other platform drivers, like davinci-mcasp do not need to fiddle with CONFIG_SND_EDMA_SOC in their code. Signed-off-by: Peter Ujfalusi Tested-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/davinci/edma-pcm.h | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/soc/davinci/edma-pcm.h b/sound/soc/davinci/edma-pcm.h index 894c378c0f74..b0957744851c 100644 --- a/sound/soc/davinci/edma-pcm.h +++ b/sound/soc/davinci/edma-pcm.h @@ -20,6 +20,13 @@ #ifndef __EDMA_PCM_H__ #define __EDMA_PCM_H__ +#if IS_ENABLED(CONFIG_SND_EDMA_SOC) int edma_pcm_platform_register(struct device *dev); +#else +static inline int edma_pcm_platform_register(struct device *dev) +{ + return 0; +} +#endif /* CONFIG_SND_EDMA_SOC */ #endif /* __EDMA_PCM_H__ */ -- cgit v1.2.3 From f3f9cfa8a1a78a3bc0dc6b7d5c26baeb07a75499 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 16 Jul 2014 15:12:04 +0300 Subject: ASoC: davinci-mcasp: Use dmaengine based platform driver for AM335x/447x Use the edma-pcm with AM335x and AM437x SoCs. Keep using the davinci-pcm for daVinci devices, they can be switched to use the dmaengine based driver later when they are verified to work correctly. Signed-off-by: Peter Ujfalusi Tested-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/davinci/Kconfig | 18 ++++++++++++++---- sound/soc/davinci/Makefile | 2 ++ sound/soc/davinci/davinci-mcasp.c | 11 +++++++++-- 3 files changed, 25 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index fdbb16fffd30..b310dd3489ac 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -1,12 +1,22 @@ config SND_DAVINCI_SOC - tristate "SoC Audio for TI DAVINCI or AM33XX/AM43XX chips" - depends on ARCH_DAVINCI || SOC_AM33XX || SOC_AM43XX + tristate "SoC Audio for TI DAVINCI" + depends on ARCH_DAVINCI + +config SND_EDMA_SOC + tristate "SoC Audio for Texas Instruments chips using eDMA (AM33XX/43XX)" + depends on SOC_AM33XX || SOC_AM43XX + select SND_SOC_GENERIC_DMAENGINE_PCM + help + Say Y or M here if you want audio support for TI SoC which uses eDMA. + The following line of SoCs are supported by this platform driver: + - AM335x + - AM437x/AM438x config SND_DAVINCI_SOC_I2S tristate config SND_DAVINCI_SOC_MCASP - depends on SND_DAVINCI_SOC || SND_OMAP_SOC + depends on SND_DAVINCI_SOC || SND_OMAP_SOC || SND_EDMA_SOC tristate config SND_DAVINCI_SOC_VCIF @@ -19,7 +29,7 @@ config SND_DAVINCI_SOC_GENERIC_EVM config SND_AM33XX_SOC_EVM tristate "SoC Audio for the AM33XX chip based boards" - depends on SND_DAVINCI_SOC && SOC_AM33XX && I2C + depends on SND_EDMA_SOC && SOC_AM33XX && I2C select SND_DAVINCI_SOC_GENERIC_EVM help Say Y or M if you want to add support for SoC audio on AM33XX diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index 744d4d9a0184..09bf2ba92d38 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -1,10 +1,12 @@ # DAVINCI Platform Support snd-soc-davinci-objs := davinci-pcm.o +snd-soc-edma-objs := edma-pcm.o snd-soc-davinci-i2s-objs := davinci-i2s.o snd-soc-davinci-mcasp-objs:= davinci-mcasp.o snd-soc-davinci-vcif-objs:= davinci-vcif.o obj-$(CONFIG_SND_DAVINCI_SOC) += snd-soc-davinci.o +obj-$(CONFIG_SND_EDMA_SOC) += snd-soc-edma.o obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o obj-$(CONFIG_SND_DAVINCI_SOC_MCASP) += snd-soc-davinci-mcasp.o obj-$(CONFIG_SND_DAVINCI_SOC_VCIF) += snd-soc-davinci-vcif.o diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index f7dc538679b1..02421d4275f5 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -37,6 +37,7 @@ #include #include "davinci-pcm.h" +#include "edma-pcm.h" #include "davinci-mcasp.h" #define MCASP_MAX_AFIFO_DEPTH 64 @@ -831,7 +832,7 @@ static int davinci_mcasp_dai_probe(struct snd_soc_dai *dai) { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); - if (mcasp->version == MCASP_VERSION_4) { + if (mcasp->version >= MCASP_VERSION_3) { /* Using dmaengine PCM */ dai->playback_dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK]; @@ -1281,10 +1282,16 @@ static int davinci_mcasp_probe(struct platform_device *pdev) IS_MODULE(CONFIG_SND_DAVINCI_SOC)) case MCASP_VERSION_1: case MCASP_VERSION_2: - case MCASP_VERSION_3: ret = davinci_soc_platform_register(&pdev->dev); break; #endif +#if IS_BUILTIN(CONFIG_SND_EDMA_SOC) || \ + (IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \ + IS_MODULE(CONFIG_SND_EDMA_SOC)) + case MCASP_VERSION_3: + ret = edma_pcm_platform_register(&pdev->dev); + break; +#endif #if IS_BUILTIN(CONFIG_SND_OMAP_SOC) || \ (IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \ IS_MODULE(CONFIG_SND_OMAP_SOC)) -- cgit v1.2.3 From 8267525c99f7a8ddb71a6f3d56cf17d4073a9973 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 16 Jul 2014 14:04:41 +0200 Subject: ASoC: mcasp: don't override bclk divider if it was provided by the machine If a machine driver provides an BCLK divider to the McASP driver, skip the automatic calculation. This fixes machines on which the physical sample transport always works in 32 bits, even though not all of them are actually used. snd_soc_params_to_bclk() will fail to address such cases properly. Signed-off-by: Daniel Mack Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 02421d4275f5..c28508da34cf 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -65,6 +65,7 @@ struct davinci_mcasp { u8 num_serializer; u8 *serial_dir; u8 version; + u8 bclk_div; u16 bclk_lrclk_ratio; int streams; @@ -419,6 +420,7 @@ static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div ACLKXDIV(div - 1), ACLKXDIV_MASK); mcasp_mod_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRDIV(div - 1), ACLKRDIV_MASK); + mcasp->bclk_div = div; break; case 2: /* BCLK/LRCLK ratio */ @@ -721,8 +723,11 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, int period_size = params_period_size(params); int ret; - /* If mcasp is BCLK master we need to set BCLK divider */ - if (mcasp->bclk_master && mcasp->sysclk_freq) { + /* + * If mcasp is BCLK master, and a BCLK divider was not provided by + * the machine driver, we need to calculate the ratio. + */ + if (mcasp->bclk_master && mcasp->bclk_div == 0 && mcasp->sysclk_freq) { unsigned int bclk_freq = snd_soc_params_to_bclk(params); unsigned int div = mcasp->sysclk_freq / bclk_freq; if (mcasp->sysclk_freq % bclk_freq != 0) { -- cgit v1.2.3 From 249adddb1a3155718876c8473ef57717d5208e37 Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Tue, 15 Jul 2014 08:51:12 +0800 Subject: ASoC: Intel: Update FW version readback Update FW version readback. IPC_GLB_GET_FW_VERSION reads back the ABI version whilst the release version is in the mailbox. Update to use mailbox version for info logging. Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-ipc.c | 20 +++++++++++++++++--- 1 file changed, 17 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 434236343ddf..96373ab46f8c 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -183,7 +183,7 @@ struct sst_hsw_ipc_fw_ready { u32 inbox_size; u32 outbox_size; u32 fw_info_size; - u8 fw_info[1]; + u8 fw_info[IPC_MAX_MAILBOX_BYTES - 5 * sizeof(u32)]; } __attribute__((packed)); struct ipc_message { @@ -569,6 +569,9 @@ static void hsw_fw_ready(struct sst_hsw *hsw, u32 header) { struct sst_hsw_ipc_fw_ready fw_ready; u32 offset; + u8 fw_info[IPC_MAX_MAILBOX_BYTES - 5 * sizeof(u32)]; + char *tmp[5], *pinfo; + int i = 0; offset = (header & 0x1FFFFFFF) << 3; @@ -589,6 +592,19 @@ static void hsw_fw_ready(struct sst_hsw *hsw, u32 header) fw_ready.inbox_offset, fw_ready.inbox_size); dev_dbg(hsw->dev, " mailbox downstream 0x%x - size 0x%x\n", fw_ready.outbox_offset, fw_ready.outbox_size); + if (fw_ready.fw_info_size < sizeof(fw_ready.fw_info)) { + fw_ready.fw_info[fw_ready.fw_info_size] = 0; + dev_dbg(hsw->dev, " Firmware info: %s \n", fw_ready.fw_info); + + /* log the FW version info got from the mailbox here. */ + memcpy(fw_info, fw_ready.fw_info, fw_ready.fw_info_size); + pinfo = &fw_info[0]; + for (i = 0; i < sizeof(tmp) / sizeof(char *); i++) + tmp[i] = strsep(&pinfo, " "); + dev_info(hsw->dev, "FW loaded, mailbox readback FW info: type %s, - " + "version: %s.%s, build %s, source commit id: %s\n", + tmp[0], tmp[1], tmp[2], tmp[3], tmp[4]); + } } static void hsw_notification_work(struct work_struct *work) @@ -1775,8 +1791,6 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) /* get the FW version */ sst_hsw_fw_get_version(hsw, &version); - dev_info(hsw->dev, "FW loaded: type %d - version: %d.%d build %d\n", - version.type, version.major, version.minor, version.build); /* get the globalmixer */ ret = sst_hsw_mixer_get_info(hsw); -- cgit v1.2.3 From 7523a271682fc1f35355da344249ce8f8c8b8cb9 Mon Sep 17 00:00:00 2001 From: Omair Mohammed Abdullah Date: Tue, 15 Jul 2014 21:34:48 +0530 Subject: ASoC: core: add a helper for extended byte controls using TLV ALSA supports arbitrary length TLVs for each kcontrol that can be used to pass metadata about the control (e.g. volumes, enum information). The same transport mechanism is now used for arbitrary length data by defining a new helper. Signed-off-by: Omair Mohammed Abdullah Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- include/sound/soc.h | 14 +++++++++++++- sound/soc/soc-core.c | 21 +++++++++++++++++++++ 2 files changed, 34 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index ed9e2d7e5fdc..688fa667dee5 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -270,7 +270,14 @@ .get = xhandler_get, .put = xhandler_put, \ .private_value = (unsigned long)&(struct soc_bytes_ext) \ {.max = xcount} } - +#define SND_SOC_BYTES_TLV(xname, xcount, xhandler_get, xhandler_put) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE | \ + SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \ + .tlv.c = (snd_soc_bytes_tlv_callback), \ + .info = snd_soc_info_bytes_ext, \ + .private_value = (unsigned long)&(struct soc_bytes_ext) \ + {.max = xcount, .get = xhandler_get, .put = xhandler_put, } } #define SOC_SINGLE_XR_SX(xname, xregbase, xregcount, xnbits, \ xmin, xmax, xinvert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ @@ -552,6 +559,8 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_bytes_info_ext(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *ucontrol); +int snd_soc_bytes_tlv_callback(struct snd_kcontrol *kcontrol, int op_flag, + unsigned int size, unsigned int __user *tlv); int snd_soc_info_xr_sx(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_soc_get_xr_sx(struct snd_kcontrol *kcontrol, @@ -1119,6 +1128,9 @@ struct soc_bytes { struct soc_bytes_ext { int max; + /* used for TLV byte control */ + int (*get)(unsigned int __user *bytes, unsigned int size); + int (*put)(const unsigned int __user *bytes, unsigned int size); }; /* multi register control */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b87d7d882e6d..a9ce63f634b2 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3267,6 +3267,27 @@ int snd_soc_bytes_info_ext(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_GPL(snd_soc_bytes_info_ext); +int snd_soc_bytes_tlv_callback(struct snd_kcontrol *kcontrol, int op_flag, + unsigned int size, unsigned int __user *tlv) +{ + struct soc_bytes_ext *params = (void *)kcontrol->private_value; + unsigned int count = size < params->max ? size : params->max; + int ret = -ENXIO; + + switch (op_flag) { + case SNDRV_CTL_TLV_OP_READ: + if (params->get) + ret = params->get(tlv, count); + break; + case SNDRV_CTL_TLV_OP_WRITE: + if (params->put) + ret = params->put(tlv, count); + break; + } + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_bytes_tlv_callback); + /** * snd_soc_info_xr_sx - signed multi register info callback * @kcontrol: mreg control -- cgit v1.2.3 From 88bd870f02dff5c9445286e185f21873f25a977f Mon Sep 17 00:00:00 2001 From: Benoit Cousson Date: Tue, 8 Jul 2014 23:19:34 +0200 Subject: ASoC: core: Add initial support for DAI multicodec DAI link assumes a one to one mapping between CPU DAI and CODEC. In some cases, the same CPU DAI can be connected to several codecs. This is the case for example, if you connect two mono codecs to the same I2S link in order to have a stereo card. The current ASoC implementation does not allow such setup. Add support for DAI links composed of a single CPU DAI and multiple CODECs. Sound cards have to pass the CODECs array in the corresponding DAI link through a new 'snd_soc_dai_link_component' struct. Each CODEC in this array is described in the same manner single CODEC DAIs are (either DT/OF node or codec_name). Multi-codec links are not supported in the case of CODEC to CODEC links. Just print a warning if it happens. Based on an original code done by Misael. Signed-off-by: Benoit Cousson Signed-off-by: Misael Lopez Cruz Signed-off-by: Fabien Parent Tested-by: Lars-Peter Clausen Reviewed-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 4 + include/sound/soc.h | 13 +++ sound/soc/soc-core.c | 293 +++++++++++++++++++++++++++++++++--------------- 3 files changed, 220 insertions(+), 90 deletions(-) (limited to 'sound') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 031be2ab75d0..e8b3080d196a 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -272,6 +272,10 @@ struct snd_soc_dai { struct snd_soc_codec *codec; struct snd_soc_component *component; + /* CODEC TDM slot masks and params (for fixup) */ + unsigned int tx_mask; + unsigned int rx_mask; + struct snd_soc_card *card; struct list_head list; diff --git a/include/sound/soc.h b/include/sound/soc.h index 9a5b4f6fe847..f2142cf3f243 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -846,6 +846,12 @@ struct snd_soc_platform_driver { int (*bespoke_trigger)(struct snd_pcm_substream *, int); }; +struct snd_soc_dai_link_component { + const char *name; + const struct device_node *of_node; + const char *dai_name; +}; + struct snd_soc_platform { struct device *dev; const struct snd_soc_platform_driver *driver; @@ -891,6 +897,10 @@ struct snd_soc_dai_link { const struct device_node *codec_of_node; /* You MUST specify the DAI name within the codec */ const char *codec_dai_name; + + struct snd_soc_dai_link_component *codecs; + unsigned int num_codecs; + /* * You MAY specify the link's platform/PCM/DMA driver, either by * device name, or by DT/OF node, but not both. Some forms of link @@ -1089,6 +1099,9 @@ struct snd_soc_pcm_runtime { struct snd_soc_dai *codec_dai; struct snd_soc_dai *cpu_dai; + struct snd_soc_dai **codec_dais; + unsigned int num_codecs; + struct delayed_work delayed_work; #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_dpcm_root; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c32d8399e770..6e47610f73a0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -576,7 +576,7 @@ int snd_soc_suspend(struct device *dev) { struct snd_soc_card *card = dev_get_drvdata(dev); struct snd_soc_codec *codec; - int i; + int i, j; /* If the initialization of this soc device failed, there is no codec * associated with it. Just bail out in this case. @@ -596,14 +596,17 @@ int snd_soc_suspend(struct device *dev) /* mute any active DACs */ for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_dai *dai = card->rtd[i].codec_dai; - struct snd_soc_dai_driver *drv = dai->driver; if (card->rtd[i].dai_link->ignore_suspend) continue; - if (drv->ops->digital_mute && dai->playback_active) - drv->ops->digital_mute(dai, 1); + for (j = 0; j < card->rtd[i].num_codecs; j++) { + struct snd_soc_dai *dai = card->rtd[i].codec_dais[j]; + struct snd_soc_dai_driver *drv = dai->driver; + + if (drv->ops->digital_mute && dai->playback_active) + drv->ops->digital_mute(dai, 1); + } } /* suspend all pcms */ @@ -634,8 +637,12 @@ int snd_soc_suspend(struct device *dev) /* close any waiting streams and save state */ for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_dai **codec_dais = card->rtd[i].codec_dais; flush_delayed_work(&card->rtd[i].delayed_work); - card->rtd[i].codec->dapm.suspend_bias_level = card->rtd[i].codec->dapm.bias_level; + for (j = 0; j < card->rtd[i].num_codecs; j++) { + codec_dais[j]->codec->dapm.suspend_bias_level = + codec_dais[j]->codec->dapm.bias_level; + } } for (i = 0; i < card->num_rtd; i++) { @@ -719,7 +726,7 @@ static void soc_resume_deferred(struct work_struct *work) struct snd_soc_card *card = container_of(work, struct snd_soc_card, deferred_resume_work); struct snd_soc_codec *codec; - int i; + int i, j; /* our power state is still SNDRV_CTL_POWER_D3hot from suspend time, * so userspace apps are blocked from touching us @@ -780,14 +787,17 @@ static void soc_resume_deferred(struct work_struct *work) /* unmute any active DACs */ for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_dai *dai = card->rtd[i].codec_dai; - struct snd_soc_dai_driver *drv = dai->driver; if (card->rtd[i].dai_link->ignore_suspend) continue; - if (drv->ops->digital_mute && dai->playback_active) - drv->ops->digital_mute(dai, 0); + for (j = 0; j < card->rtd[i].num_codecs; j++) { + struct snd_soc_dai *dai = card->rtd[i].codec_dais[j]; + struct snd_soc_dai_driver *drv = dai->driver; + + if (drv->ops->digital_mute && dai->playback_active) + drv->ops->digital_mute(dai, 0); + } } for (i = 0; i < card->num_rtd; i++) { @@ -832,12 +842,19 @@ int snd_soc_resume(struct device *dev) /* activate pins from sleep state */ for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; - struct snd_soc_dai *codec_dai = card->rtd[i].codec_dai; + struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; + struct snd_soc_dai **codec_dais = rtd->codec_dais; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int j; + if (cpu_dai->active) pinctrl_pm_select_default_state(cpu_dai->dev); - if (codec_dai->active) - pinctrl_pm_select_default_state(codec_dai->dev); + + for (j = 0; j < rtd->num_codecs; j++) { + struct snd_soc_dai *codec_dai = codec_dais[j]; + if (codec_dai->active) + pinctrl_pm_select_default_state(codec_dai->dev); + } } /* AC97 devices might have other drivers hanging off them so @@ -869,8 +886,9 @@ EXPORT_SYMBOL_GPL(snd_soc_resume); static const struct snd_soc_dai_ops null_dai_ops = { }; -static struct snd_soc_codec *soc_find_codec(const struct device_node *codec_of_node, - const char *codec_name) +static struct snd_soc_codec *soc_find_codec( + const struct device_node *codec_of_node, + const char *codec_name) { struct snd_soc_codec *codec; @@ -908,9 +926,12 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) struct snd_soc_dai_link *dai_link = &card->dai_link[num]; struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; struct snd_soc_component *component; + struct snd_soc_dai_link_component *codecs = dai_link->codecs; + struct snd_soc_dai **codec_dais = rtd->codec_dais; struct snd_soc_platform *platform; struct snd_soc_dai *cpu_dai; const char *platform_name; + int i; dev_dbg(card->dev, "ASoC: binding %s at idx %d\n", dai_link->name, num); @@ -937,24 +958,30 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) return -EPROBE_DEFER; } - /* Find CODEC from registered list */ - rtd->codec = soc_find_codec(dai_link->codec_of_node, - dai_link->codec_name); - if (!rtd->codec) { - dev_err(card->dev, "ASoC: CODEC %s not registered\n", - dai_link->codec_name); - return -EPROBE_DEFER; - } + rtd->num_codecs = dai_link->num_codecs; - /* Find CODEC DAI from registered list */ - rtd->codec_dai = soc_find_codec_dai(rtd->codec, - dai_link->codec_dai_name); - if (!rtd->codec_dai) { - dev_err(card->dev, "ASoC: CODEC DAI %s not registered\n", - dai_link->codec_dai_name); - return -EPROBE_DEFER; + /* Find CODEC from registered CODECs */ + for (i = 0; i < rtd->num_codecs; i++) { + struct snd_soc_codec *codec; + codec = soc_find_codec(codecs[i].of_node, codecs[i].name); + if (!codec) { + dev_err(card->dev, "ASoC: CODEC %s not registered\n", + codecs[i].name); + return -EPROBE_DEFER; + } + + codec_dais[i] = soc_find_codec_dai(codec, codecs[i].dai_name); + if (!codec_dais[i]) { + dev_err(card->dev, "ASoC: CODEC DAI %s not registered\n", + codecs[i].dai_name); + return -EPROBE_DEFER; + } } + /* Single codec links expect codec and codec_dai in runtime data */ + rtd->codec_dai = codec_dais[0]; + rtd->codec = rtd->codec_dai->codec; + /* if there's no platform we match on the empty platform */ platform_name = dai_link->platform_name; if (!platform_name && !dai_link->platform_of_node) @@ -1045,8 +1072,8 @@ static void soc_remove_codec_dai(struct snd_soc_dai *codec_dai, int order) static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order) { struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; - struct snd_soc_dai *codec_dai = rtd->codec_dai, *cpu_dai = rtd->cpu_dai; - int err; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int i, err; /* unregister the rtd device */ if (rtd->dev_registered) { @@ -1057,7 +1084,8 @@ static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order) } /* remove the CODEC DAI */ - soc_remove_codec_dai(codec_dai, order); + for (i = 0; i < rtd->num_codecs; i++) + soc_remove_codec_dai(rtd->codec_dais[i], order); /* remove the cpu_dai */ if (cpu_dai && cpu_dai->probed && @@ -1080,9 +1108,9 @@ static void soc_remove_link_components(struct snd_soc_card *card, int num, { struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_platform *platform = rtd->platform; struct snd_soc_codec *codec; + int i; /* remove the platform */ if (platform && platform->probed && @@ -1091,8 +1119,8 @@ static void soc_remove_link_components(struct snd_soc_card *card, int num, } /* remove the CODEC-side CODEC */ - if (codec_dai) { - codec = codec_dai->codec; + for (i = 0; i < rtd->num_codecs; i++) { + codec = rtd->codec_dais[i]->codec; if (codec && codec->probed && codec->driver->remove_order == order) soc_remove_codec(codec); @@ -1335,9 +1363,8 @@ static int soc_probe_link_components(struct snd_soc_card *card, int num, { struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_platform *platform = rtd->platform; - int ret; + int i, ret; /* probe the CPU-side component, if it is a CODEC */ if (cpu_dai->codec && @@ -1348,12 +1375,14 @@ static int soc_probe_link_components(struct snd_soc_card *card, int num, return ret; } - /* probe the CODEC-side component */ - if (!codec_dai->codec->probed && - codec_dai->codec->driver->probe_order == order) { - ret = soc_probe_codec(card, codec_dai->codec); - if (ret < 0) - return ret; + /* probe the CODEC-side components */ + for (i = 0; i < rtd->num_codecs; i++) { + if (!rtd->codec_dais[i]->codec->probed && + rtd->codec_dais[i]->codec->driver->probe_order == order) { + ret = soc_probe_codec(card, rtd->codec_dais[i]->codec); + if (ret < 0) + return ret; + } } /* probe the platform */ @@ -1400,6 +1429,9 @@ static int soc_link_dai_widgets(struct snd_soc_card *card, struct snd_soc_dapm_widget *play_w, *capture_w; int ret; + if (rtd->num_codecs > 1) + dev_warn(card->dev, "ASoC: Multiple codecs not supported yet\n"); + /* link the DAI widgets */ play_w = codec_dai->playback_widget; capture_w = cpu_dai->capture_widget; @@ -1432,19 +1464,18 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) { struct snd_soc_dai_link *dai_link = &card->dai_link[num]; struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; - struct snd_soc_codec *codec = rtd->codec; struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int ret; + int i, ret; dev_dbg(card->dev, "ASoC: probe %s dai link %d late %d\n", card->name, num, order); /* config components */ cpu_dai->platform = platform; - codec_dai->card = card; cpu_dai->card = card; + for (i = 0; i < rtd->num_codecs; i++) + rtd->codec_dais[i]->card = card; /* set default power off timeout */ rtd->pmdown_time = pmdown_time; @@ -1471,9 +1502,11 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) } /* probe the CODEC DAI */ - ret = soc_probe_codec_dai(card, codec_dai, order); - if (ret) - return ret; + for (i = 0; i < rtd->num_codecs; i++) { + ret = soc_probe_codec_dai(card, rtd->codec_dais[i], order); + if (ret) + return ret; + } /* complete DAI probe during last probe */ if (order != SND_SOC_COMP_ORDER_LAST) @@ -1541,8 +1574,11 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) } /* add platform data for AC97 devices */ - if (rtd->codec_dai->driver->ac97_control) - snd_ac97_dev_add_pdata(codec->ac97, rtd->cpu_dai->ac97_pdata); + for (i = 0; i < rtd->num_codecs; i++) { + if (rtd->codec_dais[i]->driver->ac97_control) + snd_ac97_dev_add_pdata(rtd->codec_dais[i]->codec->ac97, + rtd->cpu_dai->ac97_pdata); + } return 0; } @@ -1580,11 +1616,6 @@ static int soc_register_ac97_codec(struct snd_soc_codec *codec, return 0; } -static int soc_register_ac97_dai_link(struct snd_soc_pcm_runtime *rtd) -{ - return soc_register_ac97_codec(rtd->codec, rtd->codec_dai); -} - static void soc_unregister_ac97_codec(struct snd_soc_codec *codec) { if (codec->ac97_registered) { @@ -1593,9 +1624,30 @@ static void soc_unregister_ac97_codec(struct snd_soc_codec *codec) } } +static int soc_register_ac97_dai_link(struct snd_soc_pcm_runtime *rtd) +{ + int i, ret; + + for (i = 0; i < rtd->num_codecs; i++) { + struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; + + ret = soc_register_ac97_codec(codec_dai->codec, codec_dai); + if (ret) { + while (--i >= 0) + soc_unregister_ac97_codec(codec_dai->codec); + return ret; + } + } + + return 0; +} + static void soc_unregister_ac97_dai_link(struct snd_soc_pcm_runtime *rtd) { - soc_unregister_ac97_codec(rtd->codec); + int i; + + for (i = 0; i < rtd->num_codecs; i++) + soc_unregister_ac97_codec(rtd->codec_dais[i]->codec); } #endif @@ -1794,16 +1846,23 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) card->num_dapm_routes); for (i = 0; i < card->num_links; i++) { + struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; dai_link = &card->dai_link[i]; dai_fmt = dai_link->dai_fmt; if (dai_fmt) { - ret = snd_soc_dai_set_fmt(card->rtd[i].codec_dai, - dai_fmt); - if (ret != 0 && ret != -ENOTSUPP) - dev_warn(card->rtd[i].codec_dai->dev, - "ASoC: Failed to set DAI format: %d\n", - ret); + struct snd_soc_dai **codec_dais = rtd->codec_dais; + int j; + + for (j = 0; j < rtd->num_codecs; j++) { + struct snd_soc_dai *codec_dai = codec_dais[j]; + + ret = snd_soc_dai_set_fmt(codec_dai, dai_fmt); + if (ret != 0 && ret != -ENOTSUPP) + dev_warn(codec_dai->dev, + "ASoC: Failed to set DAI format: %d\n", + ret); + } } /* If this is a regular CPU link there will be a platform */ @@ -2002,10 +2061,15 @@ int snd_soc_poweroff(struct device *dev) /* deactivate pins to sleep state */ for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; - struct snd_soc_dai *codec_dai = card->rtd[i].codec_dai; - pinctrl_pm_select_sleep_state(codec_dai->dev); + struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int j; + pinctrl_pm_select_sleep_state(cpu_dai->dev); + for (j = 0; j < rtd->num_codecs; j++) { + struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; + pinctrl_pm_select_sleep_state(codec_dai->dev); + } } return 0; @@ -3585,6 +3649,9 @@ int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, else snd_soc_xlate_tdm_slot_mask(slots, &tx_mask, &rx_mask); + dai->tx_mask = tx_mask; + dai->rx_mask = rx_mask; + if (dai->driver && dai->driver->ops->set_tdm_slot) return dai->driver->ops->set_tdm_slot(dai, tx_mask, rx_mask, slots, slot_width); @@ -3657,6 +3724,33 @@ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, } EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); +static int snd_soc_init_multicodec(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_link) +{ + /* Legacy codec/codec_dai link is a single entry in multicodec */ + if (dai_link->codec_name || dai_link->codec_of_node || + dai_link->codec_dai_name) { + dai_link->num_codecs = 1; + + dai_link->codecs = devm_kzalloc(card->dev, + sizeof(struct snd_soc_dai_link_component), + GFP_KERNEL); + if (!dai_link->codecs) + return -ENOMEM; + + dai_link->codecs[0].name = dai_link->codec_name; + dai_link->codecs[0].of_node = dai_link->codec_of_node; + dai_link->codecs[0].dai_name = dai_link->codec_dai_name; + } + + if (!dai_link->codecs) { + dev_err(card->dev, "ASoC: DAI link has no CODECs\n"); + return -EINVAL; + } + + return 0; +} + /** * snd_soc_register_card - Register a card with the ASoC core * @@ -3665,7 +3759,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); */ int snd_soc_register_card(struct snd_soc_card *card) { - int i, ret; + int i, j, ret; if (!card->name || !card->dev) return -EINVAL; @@ -3673,22 +3767,29 @@ int snd_soc_register_card(struct snd_soc_card *card) for (i = 0; i < card->num_links; i++) { struct snd_soc_dai_link *link = &card->dai_link[i]; - /* - * Codec must be specified by 1 of name or OF node, - * not both or neither. - */ - if (!!link->codec_name == !!link->codec_of_node) { - dev_err(card->dev, - "ASoC: Neither/both codec name/of_node are set for %s\n", - link->name); - return -EINVAL; + ret = snd_soc_init_multicodec(card, link); + if (ret) { + dev_err(card->dev, "ASoC: failed to init multicodec\n"); + return ret; } - /* Codec DAI name must be specified */ - if (!link->codec_dai_name) { - dev_err(card->dev, - "ASoC: codec_dai_name not set for %s\n", - link->name); - return -EINVAL; + + for (j = 0; j < link->num_codecs; j++) { + /* + * Codec must be specified by 1 of name or OF node, + * not both or neither. + */ + if (!!link->codecs[j].name == + !!link->codecs[j].of_node) { + dev_err(card->dev, "ASoC: Neither/both codec name/of_node are set for %s\n", + link->name); + return -EINVAL; + } + /* Codec DAI name must be specified */ + if (!link->codecs[j].dai_name) { + dev_err(card->dev, "ASoC: codec_dai_name not set for %s\n", + link->name); + return -EINVAL; + } } /* @@ -3744,6 +3845,12 @@ int snd_soc_register_card(struct snd_soc_card *card) for (i = 0; i < card->num_links; i++) { card->rtd[i].card = card; card->rtd[i].dai_link = &card->dai_link[i]; + card->rtd[i].codec_dais = devm_kzalloc(card->dev, + sizeof(struct snd_soc_dai *) * + (card->rtd[i].dai_link->num_codecs), + GFP_KERNEL); + if (card->rtd[i].codec_dais == NULL) + return -ENOMEM; } for (i = 0; i < card->num_aux_devs; i++) @@ -3760,10 +3867,16 @@ int snd_soc_register_card(struct snd_soc_card *card) /* deactivate pins to sleep state */ for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; - struct snd_soc_dai *codec_dai = card->rtd[i].codec_dai; - if (!codec_dai->active) - pinctrl_pm_select_sleep_state(codec_dai->dev); + struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int j; + + for (j = 0; j < rtd->num_codecs; j++) { + struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; + if (!codec_dai->active) + pinctrl_pm_select_sleep_state(codec_dai->dev); + } + if (!cpu_dai->active) pinctrl_pm_select_sleep_state(cpu_dai->dev); } -- cgit v1.2.3 From 2e5894d73789ee60d4d406fd3342a9a5152ec23c Mon Sep 17 00:00:00 2001 From: Benoit Cousson Date: Tue, 8 Jul 2014 23:19:35 +0200 Subject: ASoC: pcm: Add support for DAI multicodec Add multicodec support in soc-pcm.c Signed-off-by: Benoit Cousson Signed-off-by: Misael Lopez Cruz Signed-off-by: Fabien Parent Tested-by: Lars-Peter Clausen Reviewed-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 534 ++++++++++++++++++++++++++++++++++++---------------- 1 file changed, 368 insertions(+), 166 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 3bf0355ca30f..ec56d1831a86 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -47,22 +47,26 @@ void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream) { struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + int i; lockdep_assert_held(&rtd->pcm_mutex); if (stream == SNDRV_PCM_STREAM_PLAYBACK) { cpu_dai->playback_active++; - codec_dai->playback_active++; + for (i = 0; i < rtd->num_codecs; i++) + rtd->codec_dais[i]->playback_active++; } else { cpu_dai->capture_active++; - codec_dai->capture_active++; + for (i = 0; i < rtd->num_codecs; i++) + rtd->codec_dais[i]->capture_active++; } cpu_dai->active++; - codec_dai->active++; cpu_dai->component->active++; - codec_dai->component->active++; + for (i = 0; i < rtd->num_codecs; i++) { + rtd->codec_dais[i]->active++; + rtd->codec_dais[i]->component->active++; + } } /** @@ -78,22 +82,26 @@ void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream) void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream) { struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + int i; lockdep_assert_held(&rtd->pcm_mutex); if (stream == SNDRV_PCM_STREAM_PLAYBACK) { cpu_dai->playback_active--; - codec_dai->playback_active--; + for (i = 0; i < rtd->num_codecs; i++) + rtd->codec_dais[i]->playback_active--; } else { cpu_dai->capture_active--; - codec_dai->capture_active--; + for (i = 0; i < rtd->num_codecs; i++) + rtd->codec_dais[i]->capture_active--; } cpu_dai->active--; - codec_dai->active--; cpu_dai->component->active--; - codec_dai->component->active--; + for (i = 0; i < rtd->num_codecs; i++) { + rtd->codec_dais[i]->component->active--; + rtd->codec_dais[i]->active--; + } } /** @@ -107,11 +115,16 @@ void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream) */ bool snd_soc_runtime_ignore_pmdown_time(struct snd_soc_pcm_runtime *rtd) { + int i; + bool ignore = true; + if (!rtd->pmdown_time || rtd->dai_link->ignore_pmdown_time) return true; - return rtd->cpu_dai->component->ignore_pmdown_time && - rtd->codec_dai->component->ignore_pmdown_time; + for (i = 0; i < rtd->num_codecs; i++) + ignore &= rtd->codec_dais[i]->component->ignore_pmdown_time; + + return rtd->cpu_dai->component->ignore_pmdown_time && ignore; } /** @@ -222,8 +235,7 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - unsigned int rate, channels, sample_bits, symmetry; + unsigned int rate, channels, sample_bits, symmetry, i; rate = params_rate(params); channels = params_channels(params); @@ -231,8 +243,11 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, /* reject unmatched parameters when applying symmetry */ symmetry = cpu_dai->driver->symmetric_rates || - codec_dai->driver->symmetric_rates || rtd->dai_link->symmetric_rates; + + for (i = 0; i < rtd->num_codecs; i++) + symmetry |= rtd->codec_dais[i]->driver->symmetric_rates; + if (symmetry && cpu_dai->rate && cpu_dai->rate != rate) { dev_err(rtd->dev, "ASoC: unmatched rate symmetry: %d - %d\n", cpu_dai->rate, rate); @@ -240,8 +255,11 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, } symmetry = cpu_dai->driver->symmetric_channels || - codec_dai->driver->symmetric_channels || rtd->dai_link->symmetric_channels; + + for (i = 0; i < rtd->num_codecs; i++) + symmetry |= rtd->codec_dais[i]->driver->symmetric_channels; + if (symmetry && cpu_dai->channels && cpu_dai->channels != channels) { dev_err(rtd->dev, "ASoC: unmatched channel symmetry: %d - %d\n", cpu_dai->channels, channels); @@ -249,8 +267,11 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, } symmetry = cpu_dai->driver->symmetric_samplebits || - codec_dai->driver->symmetric_samplebits || rtd->dai_link->symmetric_samplebits; + + for (i = 0; i < rtd->num_codecs; i++) + symmetry |= rtd->codec_dais[i]->driver->symmetric_samplebits; + if (symmetry && cpu_dai->sample_bits && cpu_dai->sample_bits != sample_bits) { dev_err(rtd->dev, "ASoC: unmatched sample bits symmetry: %d - %d\n", cpu_dai->sample_bits, sample_bits); @@ -264,15 +285,20 @@ static bool soc_pcm_has_symmetry(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai_driver *cpu_driver = rtd->cpu_dai->driver; - struct snd_soc_dai_driver *codec_driver = rtd->codec_dai->driver; struct snd_soc_dai_link *link = rtd->dai_link; + unsigned int symmetry, i; + + symmetry = cpu_driver->symmetric_rates || link->symmetric_rates || + cpu_driver->symmetric_channels || link->symmetric_channels || + cpu_driver->symmetric_samplebits || link->symmetric_samplebits; - return cpu_driver->symmetric_rates || codec_driver->symmetric_rates || - link->symmetric_rates || cpu_driver->symmetric_channels || - codec_driver->symmetric_channels || link->symmetric_channels || - cpu_driver->symmetric_samplebits || - codec_driver->symmetric_samplebits || - link->symmetric_samplebits; + for (i = 0; i < rtd->num_codecs; i++) + symmetry = symmetry || + rtd->codec_dais[i]->driver->symmetric_rates || + rtd->codec_dais[i]->driver->symmetric_channels || + rtd->codec_dais[i]->driver->symmetric_samplebits; + + return symmetry; } /* @@ -284,9 +310,9 @@ static int sample_sizes[] = { 24, 32, }; -static void soc_pcm_set_msb(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai, int bits) +static void soc_pcm_set_msb(struct snd_pcm_substream *substream, int bits) { + struct snd_soc_pcm_runtime *rtd = substream->private_data; int ret, i; if (!bits) @@ -299,7 +325,7 @@ static void soc_pcm_set_msb(struct snd_pcm_substream *substream, ret = snd_pcm_hw_constraint_msbits(substream->runtime, 0, sample_sizes[i], bits); if (ret != 0) - dev_warn(dai->dev, + dev_warn(rtd->dev, "ASoC: Failed to set MSB %d/%d: %d\n", bits, sample_sizes[i], ret); } @@ -309,47 +335,95 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai; + int i; unsigned int bits = 0, cpu_bits; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - bits = codec_dai->driver->playback.sig_bits; + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = rtd->codec_dais[i]; + if (codec_dai->driver->playback.sig_bits == 0) { + bits = 0; + break; + } + bits = max(codec_dai->driver->playback.sig_bits, bits); + } cpu_bits = cpu_dai->driver->playback.sig_bits; } else { - bits = codec_dai->driver->capture.sig_bits; + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = rtd->codec_dais[i]; + if (codec_dai->driver->playback.sig_bits == 0) { + bits = 0; + break; + } + bits = max(codec_dai->driver->capture.sig_bits, bits); + } cpu_bits = cpu_dai->driver->capture.sig_bits; } - soc_pcm_set_msb(substream, codec_dai, bits); - soc_pcm_set_msb(substream, cpu_dai, cpu_bits); + soc_pcm_set_msb(substream, bits); + soc_pcm_set_msb(substream, cpu_bits); } -static void soc_pcm_init_runtime_hw(struct snd_pcm_runtime *runtime, - struct snd_soc_pcm_stream *codec_stream, - struct snd_soc_pcm_stream *cpu_stream) +static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) { + struct snd_pcm_runtime *runtime = substream->runtime; struct snd_pcm_hardware *hw = &runtime->hw; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai_driver *cpu_dai_drv = rtd->cpu_dai->driver; + struct snd_soc_dai_driver *codec_dai_drv; + struct snd_soc_pcm_stream *codec_stream; + struct snd_soc_pcm_stream *cpu_stream; + unsigned int chan_min = 0, chan_max = UINT_MAX; + unsigned int rate_min = 0, rate_max = UINT_MAX; + unsigned int rates = UINT_MAX; + u64 formats = ULLONG_MAX; + int i; - hw->channels_min = max(codec_stream->channels_min, - cpu_stream->channels_min); - hw->channels_max = min(codec_stream->channels_max, - cpu_stream->channels_max); - if (hw->formats) - hw->formats &= codec_stream->formats & cpu_stream->formats; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + cpu_stream = &cpu_dai_drv->playback; else - hw->formats = codec_stream->formats & cpu_stream->formats; - hw->rates = snd_pcm_rate_mask_intersect(codec_stream->rates, - cpu_stream->rates); + cpu_stream = &cpu_dai_drv->capture; - hw->rate_min = 0; - hw->rate_max = UINT_MAX; + /* first calculate min/max only for CODECs in the DAI link */ + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai_drv = rtd->codec_dais[i]->driver; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + codec_stream = &codec_dai_drv->playback; + else + codec_stream = &codec_dai_drv->capture; + chan_min = max(chan_min, codec_stream->channels_min); + chan_max = min(chan_max, codec_stream->channels_max); + rate_min = max(rate_min, codec_stream->rate_min); + rate_max = min_not_zero(rate_max, codec_stream->rate_max); + formats &= codec_stream->formats; + rates = snd_pcm_rate_mask_intersect(codec_stream->rates, rates); + } + + /* + * chan min/max cannot be enforced if there are multiple CODEC DAIs + * connected to a single CPU DAI, use CPU DAI's directly and let + * channel allocation be fixed up later + */ + if (rtd->num_codecs > 1) { + chan_min = cpu_stream->channels_min; + chan_max = cpu_stream->channels_max; + } + + hw->channels_min = max(chan_min, cpu_stream->channels_min); + hw->channels_max = min(chan_max, cpu_stream->channels_max); + if (hw->formats) + hw->formats &= formats & cpu_stream->formats; + else + hw->formats = formats & cpu_stream->formats; + hw->rates = snd_pcm_rate_mask_intersect(rates, cpu_stream->rates); snd_pcm_limit_hw_rates(runtime); hw->rate_min = max(hw->rate_min, cpu_stream->rate_min); - hw->rate_min = max(hw->rate_min, codec_stream->rate_min); + hw->rate_min = max(hw->rate_min, rate_min); hw->rate_max = min_not_zero(hw->rate_max, cpu_stream->rate_max); - hw->rate_max = min_not_zero(hw->rate_max, codec_stream->rate_max); + hw->rate_max = min_not_zero(hw->rate_max, rate_max); } /* @@ -363,15 +437,16 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver; - struct snd_soc_dai_driver *codec_dai_drv = codec_dai->driver; - int ret = 0; + struct snd_soc_dai *codec_dai; + const char *codec_dai_name = "multicodec"; + int i, ret = 0; pinctrl_pm_select_default_state(cpu_dai->dev); - pinctrl_pm_select_default_state(codec_dai->dev); + for (i = 0; i < rtd->num_codecs; i++) + pinctrl_pm_select_default_state(rtd->codec_dais[i]->dev); pm_runtime_get_sync(cpu_dai->dev); - pm_runtime_get_sync(codec_dai->dev); + for (i = 0; i < rtd->num_codecs; i++) + pm_runtime_get_sync(rtd->codec_dais[i]->dev); pm_runtime_get_sync(platform->dev); mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); @@ -395,13 +470,23 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) } } - if (codec_dai->driver->ops && codec_dai->driver->ops->startup) { - ret = codec_dai->driver->ops->startup(substream, codec_dai); - if (ret < 0) { - dev_err(codec_dai->dev, "ASoC: can't open codec" - " %s: %d\n", codec_dai->name, ret); - goto codec_dai_err; + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = rtd->codec_dais[i]; + if (codec_dai->driver->ops && codec_dai->driver->ops->startup) { + ret = codec_dai->driver->ops->startup(substream, + codec_dai); + if (ret < 0) { + dev_err(codec_dai->dev, + "ASoC: can't open codec %s: %d\n", + codec_dai->name, ret); + goto codec_dai_err; + } } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + codec_dai->tx_mask = 0; + else + codec_dai->rx_mask = 0; } if (rtd->dai_link->ops && rtd->dai_link->ops->startup) { @@ -418,13 +503,10 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) goto dynamic; /* Check that the codec and cpu DAIs are compatible */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - soc_pcm_init_runtime_hw(runtime, &codec_dai_drv->playback, - &cpu_dai_drv->playback); - } else { - soc_pcm_init_runtime_hw(runtime, &codec_dai_drv->capture, - &cpu_dai_drv->capture); - } + soc_pcm_init_runtime_hw(substream); + + if (rtd->num_codecs == 1) + codec_dai_name = rtd->codec_dai->name; if (soc_pcm_has_symmetry(substream)) runtime->hw.info |= SNDRV_PCM_INFO_JOINT_DUPLEX; @@ -432,18 +514,18 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) ret = -EINVAL; if (!runtime->hw.rates) { printk(KERN_ERR "ASoC: %s <-> %s No matching rates\n", - codec_dai->name, cpu_dai->name); + codec_dai_name, cpu_dai->name); goto config_err; } if (!runtime->hw.formats) { printk(KERN_ERR "ASoC: %s <-> %s No matching formats\n", - codec_dai->name, cpu_dai->name); + codec_dai_name, cpu_dai->name); goto config_err; } if (!runtime->hw.channels_min || !runtime->hw.channels_max || runtime->hw.channels_min > runtime->hw.channels_max) { printk(KERN_ERR "ASoC: %s <-> %s No matching channels\n", - codec_dai->name, cpu_dai->name); + codec_dai_name, cpu_dai->name); goto config_err; } @@ -456,14 +538,17 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) goto config_err; } - if (codec_dai->active) { - ret = soc_pcm_apply_symmetry(substream, codec_dai); - if (ret != 0) - goto config_err; + for (i = 0; i < rtd->num_codecs; i++) { + if (rtd->codec_dais[i]->active) { + ret = soc_pcm_apply_symmetry(substream, + rtd->codec_dais[i]); + if (ret != 0) + goto config_err; + } } pr_debug("ASoC: %s <-> %s info:\n", - codec_dai->name, cpu_dai->name); + codec_dai_name, cpu_dai->name); pr_debug("ASoC: rate mask 0x%x\n", runtime->hw.rates); pr_debug("ASoC: min ch %d max ch %d\n", runtime->hw.channels_min, runtime->hw.channels_max); @@ -482,10 +567,15 @@ config_err: rtd->dai_link->ops->shutdown(substream); machine_err: - if (codec_dai->driver->ops->shutdown) - codec_dai->driver->ops->shutdown(substream, codec_dai); + i = rtd->num_codecs; codec_dai_err: + while (--i >= 0) { + codec_dai = rtd->codec_dais[i]; + if (codec_dai->driver->ops->shutdown) + codec_dai->driver->ops->shutdown(substream, codec_dai); + } + if (platform->driver->ops && platform->driver->ops->close) platform->driver->ops->close(substream); @@ -496,10 +586,13 @@ out: mutex_unlock(&rtd->pcm_mutex); pm_runtime_put(platform->dev); - pm_runtime_put(codec_dai->dev); + for (i = 0; i < rtd->num_codecs; i++) + pm_runtime_put(rtd->codec_dais[i]->dev); pm_runtime_put(cpu_dai->dev); - if (!codec_dai->active) - pinctrl_pm_select_sleep_state(codec_dai->dev); + for (i = 0; i < rtd->num_codecs; i++) { + if (!rtd->codec_dais[i]->active) + pinctrl_pm_select_sleep_state(rtd->codec_dais[i]->dev); + } if (!cpu_dai->active) pinctrl_pm_select_sleep_state(cpu_dai->dev); @@ -515,7 +608,7 @@ static void close_delayed_work(struct work_struct *work) { struct snd_soc_pcm_runtime *rtd = container_of(work, struct snd_soc_pcm_runtime, delayed_work.work); - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dais[0]; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); @@ -544,7 +637,8 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai; + int i; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); @@ -554,14 +648,20 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) if (!cpu_dai->active) cpu_dai->rate = 0; - if (!codec_dai->active) - codec_dai->rate = 0; + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = rtd->codec_dais[i]; + if (!codec_dai->active) + codec_dai->rate = 0; + } if (cpu_dai->driver->ops->shutdown) cpu_dai->driver->ops->shutdown(substream, cpu_dai); - if (codec_dai->driver->ops->shutdown) - codec_dai->driver->ops->shutdown(substream, codec_dai); + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = rtd->codec_dais[i]; + if (codec_dai->driver->ops->shutdown) + codec_dai->driver->ops->shutdown(substream, codec_dai); + } if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown) rtd->dai_link->ops->shutdown(substream); @@ -591,10 +691,13 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) mutex_unlock(&rtd->pcm_mutex); pm_runtime_put(platform->dev); - pm_runtime_put(codec_dai->dev); + for (i = 0; i < rtd->num_codecs; i++) + pm_runtime_put(rtd->codec_dais[i]->dev); pm_runtime_put(cpu_dai->dev); - if (!codec_dai->active) - pinctrl_pm_select_sleep_state(codec_dai->dev); + for (i = 0; i < rtd->num_codecs; i++) { + if (!rtd->codec_dais[i]->active) + pinctrl_pm_select_sleep_state(rtd->codec_dais[i]->dev); + } if (!cpu_dai->active) pinctrl_pm_select_sleep_state(cpu_dai->dev); @@ -611,8 +714,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret = 0; + struct snd_soc_dai *codec_dai; + int i, ret = 0; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); @@ -634,12 +737,16 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) } } - if (codec_dai->driver->ops && codec_dai->driver->ops->prepare) { - ret = codec_dai->driver->ops->prepare(substream, codec_dai); - if (ret < 0) { - dev_err(codec_dai->dev, "ASoC: DAI prepare error: %d\n", - ret); - goto out; + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = rtd->codec_dais[i]; + if (codec_dai->driver->ops && codec_dai->driver->ops->prepare) { + ret = codec_dai->driver->ops->prepare(substream, + codec_dai); + if (ret < 0) { + dev_err(codec_dai->dev, + "ASoC: DAI prepare error: %d\n", ret); + goto out; + } } } @@ -662,13 +769,26 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) snd_soc_dapm_stream_event(rtd, substream->stream, SND_SOC_DAPM_STREAM_START); - snd_soc_dai_digital_mute(codec_dai, 0, substream->stream); + for (i = 0; i < rtd->num_codecs; i++) + snd_soc_dai_digital_mute(rtd->codec_dais[i], 0, + substream->stream); out: mutex_unlock(&rtd->pcm_mutex); return ret; } +static void soc_pcm_codec_params_fixup(struct snd_pcm_hw_params *params, + unsigned int mask) +{ + struct snd_interval *interval; + int channels = hweight_long(mask); + + interval = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + interval->min = channels; + interval->max = channels; +} + /* * Called by ALSA when the hardware params are set by application. This * function can also be called multiple times and can allocate buffers @@ -680,8 +800,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret = 0; + int i, ret = 0; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); @@ -698,13 +817,37 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } } - if (codec_dai->driver->ops && codec_dai->driver->ops->hw_params) { - ret = codec_dai->driver->ops->hw_params(substream, params, codec_dai); - if (ret < 0) { - dev_err(codec_dai->dev, "ASoC: can't set %s hw params:" - " %d\n", codec_dai->name, ret); - goto codec_err; + for (i = 0; i < rtd->num_codecs; i++) { + struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; + struct snd_pcm_hw_params codec_params; + + /* copy params for each codec */ + codec_params = *params; + + /* fixup params based on TDM slot masks */ + if (codec_dai->tx_mask) + soc_pcm_codec_params_fixup(&codec_params, + codec_dai->tx_mask); + if (codec_dai->rx_mask) + soc_pcm_codec_params_fixup(&codec_params, + codec_dai->rx_mask); + + if (codec_dai->driver->ops && + codec_dai->driver->ops->hw_params) { + ret = codec_dai->driver->ops->hw_params(substream, + &codec_params, codec_dai); + if (ret < 0) { + dev_err(codec_dai->dev, + "ASoC: can't set %s hw params: %d\n", + codec_dai->name, ret); + goto codec_err; + } } + + codec_dai->rate = params_rate(&codec_params); + codec_dai->channels = params_channels(&codec_params); + codec_dai->sample_bits = snd_pcm_format_physical_width( + params_format(&codec_params)); } if (cpu_dai->driver->ops && cpu_dai->driver->ops->hw_params) { @@ -731,11 +874,6 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, cpu_dai->sample_bits = snd_pcm_format_physical_width(params_format(params)); - codec_dai->rate = params_rate(params); - codec_dai->channels = params_channels(params); - codec_dai->sample_bits = - snd_pcm_format_physical_width(params_format(params)); - out: mutex_unlock(&rtd->pcm_mutex); return ret; @@ -745,10 +883,16 @@ platform_err: cpu_dai->driver->ops->hw_free(substream, cpu_dai); interface_err: - if (codec_dai->driver->ops && codec_dai->driver->ops->hw_free) - codec_dai->driver->ops->hw_free(substream, codec_dai); + i = rtd->num_codecs; codec_err: + while (--i >= 0) { + struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; + if (codec_dai->driver->ops && codec_dai->driver->ops->hw_free) + codec_dai->driver->ops->hw_free(substream, codec_dai); + codec_dai->rate = 0; + } + if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) rtd->dai_link->ops->hw_free(substream); @@ -764,8 +908,9 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai; bool playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + int i; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); @@ -776,16 +921,22 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) cpu_dai->sample_bits = 0; } - if (codec_dai->active == 1) { - codec_dai->rate = 0; - codec_dai->channels = 0; - codec_dai->sample_bits = 0; + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = rtd->codec_dais[i]; + if (codec_dai->active == 1) { + codec_dai->rate = 0; + codec_dai->channels = 0; + codec_dai->sample_bits = 0; + } } /* apply codec digital mute */ - if ((playback && codec_dai->playback_active == 1) || - (!playback && codec_dai->capture_active == 1)) - snd_soc_dai_digital_mute(codec_dai, 1, substream->stream); + for (i = 0; i < rtd->num_codecs; i++) { + if ((playback && rtd->codec_dais[i]->playback_active == 1) || + (!playback && rtd->codec_dais[i]->capture_active == 1)) + snd_soc_dai_digital_mute(rtd->codec_dais[i], 1, + substream->stream); + } /* free any machine hw params */ if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) @@ -796,8 +947,11 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) platform->driver->ops->hw_free(substream); /* now free hw params for the DAIs */ - if (codec_dai->driver->ops && codec_dai->driver->ops->hw_free) - codec_dai->driver->ops->hw_free(substream, codec_dai); + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = rtd->codec_dais[i]; + if (codec_dai->driver->ops && codec_dai->driver->ops->hw_free) + codec_dai->driver->ops->hw_free(substream, codec_dai); + } if (cpu_dai->driver->ops && cpu_dai->driver->ops->hw_free) cpu_dai->driver->ops->hw_free(substream, cpu_dai); @@ -811,13 +965,17 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - if (codec_dai->driver->ops && codec_dai->driver->ops->trigger) { - ret = codec_dai->driver->ops->trigger(substream, cmd, codec_dai); - if (ret < 0) - return ret; + struct snd_soc_dai *codec_dai; + int i, ret; + + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = rtd->codec_dais[i]; + if (codec_dai->driver->ops && codec_dai->driver->ops->trigger) { + ret = codec_dai->driver->ops->trigger(substream, + cmd, codec_dai); + if (ret < 0) + return ret; + } } if (platform->driver->ops && platform->driver->ops->trigger) { @@ -847,14 +1005,18 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - if (codec_dai->driver->ops && - codec_dai->driver->ops->bespoke_trigger) { - ret = codec_dai->driver->ops->bespoke_trigger(substream, cmd, codec_dai); - if (ret < 0) - return ret; + struct snd_soc_dai *codec_dai; + int i, ret; + + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = rtd->codec_dais[i]; + if (codec_dai->driver->ops && + codec_dai->driver->ops->bespoke_trigger) { + ret = codec_dai->driver->ops->bespoke_trigger(substream, + cmd, codec_dai); + if (ret < 0) + return ret; + } } if (platform->driver->bespoke_trigger) { @@ -880,10 +1042,12 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai; struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_uframes_t offset = 0; snd_pcm_sframes_t delay = 0; + snd_pcm_sframes_t codec_delay = 0; + int i; if (platform->driver->ops && platform->driver->ops->pointer) offset = platform->driver->ops->pointer(substream); @@ -891,11 +1055,21 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) if (cpu_dai->driver->ops && cpu_dai->driver->ops->delay) delay += cpu_dai->driver->ops->delay(substream, cpu_dai); - if (codec_dai->driver->ops && codec_dai->driver->ops->delay) - delay += codec_dai->driver->ops->delay(substream, codec_dai); + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = rtd->codec_dais[i]; + if (codec_dai->driver->ops && codec_dai->driver->ops->delay) + codec_delay = max(codec_delay, + codec_dai->driver->ops->delay(substream, + codec_dai)); + } + delay += codec_delay; + /* + * None of the existing platform drivers implement delay(), so + * for now the codec_dai of first multicodec entry is used + */ if (platform->driver->delay) - delay += platform->driver->delay(substream, codec_dai); + delay += platform->driver->delay(substream, rtd->codec_dais[0]); runtime->delay = delay; @@ -998,7 +1172,7 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, struct snd_soc_dapm_widget *widget, int stream) { struct snd_soc_pcm_runtime *be; - int i; + int i, j; if (stream == SNDRV_PCM_STREAM_PLAYBACK) { for (i = 0; i < card->num_links; i++) { @@ -1007,9 +1181,14 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, if (!be->dai_link->no_pcm) continue; - if (be->cpu_dai->playback_widget == widget || - be->codec_dai->playback_widget == widget) + if (be->cpu_dai->playback_widget == widget) return be; + + for (j = 0; j < be->num_codecs; j++) { + struct snd_soc_dai *dai = be->codec_dais[j]; + if (dai->playback_widget == widget) + return be; + } } } else { @@ -1019,9 +1198,14 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, if (!be->dai_link->no_pcm) continue; - if (be->cpu_dai->capture_widget == widget || - be->codec_dai->capture_widget == widget) + if (be->cpu_dai->capture_widget == widget) return be; + + for (j = 0; j < be->num_codecs; j++) { + struct snd_soc_dai *dai = be->codec_dais[j]; + if (dai->capture_widget == widget) + return be; + } } } @@ -1084,6 +1268,7 @@ static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream, /* Destroy any old FE <--> BE connections */ list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + unsigned int i; /* is there a valid CPU DAI widget for this BE */ widget = dai_get_widget(dpcm->be->cpu_dai, stream); @@ -1093,11 +1278,14 @@ static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream, continue; /* is there a valid CODEC DAI widget for this BE */ - widget = dai_get_widget(dpcm->be->codec_dai, stream); + for (i = 0; i < dpcm->be->num_codecs; i++) { + struct snd_soc_dai *dai = dpcm->be->codec_dais[i]; + widget = dai_get_widget(dai, stream); - /* prune the BE if it's no longer in our active list */ - if (widget && widget_in_list(list, widget)) - continue; + /* prune the BE if it's no longer in our active list */ + if (widget && widget_in_list(list, widget)) + continue; + } dev_dbg(fe->dev, "ASoC: pruning %s BE %s for %s\n", stream ? "capture" : "playback", @@ -2126,16 +2314,22 @@ int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute) list_for_each_entry(dpcm, clients, list_be) { struct snd_soc_pcm_runtime *be = dpcm->be; - struct snd_soc_dai *dai = be->codec_dai; - struct snd_soc_dai_driver *drv = dai->driver; + int i; if (be->dai_link->ignore_suspend) continue; - dev_dbg(be->dev, "ASoC: BE digital mute %s\n", be->dai_link->name); + for (i = 0; i < be->num_codecs; i++) { + struct snd_soc_dai *dai = be->codec_dais[i]; + struct snd_soc_dai_driver *drv = dai->driver; + + dev_dbg(be->dev, "ASoC: BE digital mute %s\n", + be->dai_link->name); - if (drv->ops && drv->ops->digital_mute && dai->playback_active) - drv->ops->digital_mute(dai, mute); + if (drv->ops && drv->ops->digital_mute && + dai->playback_active) + drv->ops->digital_mute(dai, mute); + } } return 0; @@ -2200,22 +2394,27 @@ static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream) int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) { struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_pcm *pcm; char new_name[64]; int ret = 0, playback = 0, capture = 0; + int i; if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm) { playback = rtd->dai_link->dpcm_playback; capture = rtd->dai_link->dpcm_capture; } else { - if (codec_dai->driver->playback.channels_min && - cpu_dai->driver->playback.channels_min) - playback = 1; - if (codec_dai->driver->capture.channels_min && - cpu_dai->driver->capture.channels_min) - capture = 1; + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = rtd->codec_dais[i]; + if (codec_dai->driver->playback.channels_min) + playback = 1; + if (codec_dai->driver->capture.channels_min) + capture = 1; + } + + capture = capture && cpu_dai->driver->capture.channels_min; + playback = playback && cpu_dai->driver->playback.channels_min; } if (rtd->dai_link->playback_only) { @@ -2241,7 +2440,9 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) rtd->dai_link->stream_name); else snprintf(new_name, sizeof(new_name), "%s %s-%d", - rtd->dai_link->stream_name, codec_dai->name, num); + rtd->dai_link->stream_name, + (rtd->num_codecs > 1) ? + "multicodec" : rtd->codec_dai->name, num); ret = snd_pcm_new(rtd->card->snd_card, new_name, num, playback, capture, &pcm); @@ -2314,8 +2515,9 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) pcm->private_free = platform->driver->pcm_free; out: - dev_info(rtd->card->dev, "%s <-> %s mapping ok\n", codec_dai->name, - cpu_dai->name); + dev_info(rtd->card->dev, "%s <-> %s mapping ok\n", + (rtd->num_codecs > 1) ? "multicodec" : rtd->codec_dai->name, + cpu_dai->name); return ret; } -- cgit v1.2.3 From 44ba2641b77ab4037b79873ea5afdac3329b1ea3 Mon Sep 17 00:00:00 2001 From: Benoit Cousson Date: Tue, 8 Jul 2014 23:19:36 +0200 Subject: ASoC: dapm: Add support for DAI multicodec Add multicodec support in soc-dapm.c Signed-off-by: Benoit Cousson Signed-off-by: Misael Lopez Cruz Signed-off-by: Fabien Parent Tested-by: Lars-Peter Clausen Reviewed-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 67 ++++++++++++++++++++++++++++++++++++---------------- 1 file changed, 47 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 4bf08cffd1f1..5c63c3b49f3f 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2084,12 +2084,8 @@ int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_context *dapm, } EXPORT_SYMBOL_GPL(snd_soc_dapm_mixer_update_power); -/* show dapm widget status in sys fs */ -static ssize_t dapm_widget_show(struct device *dev, - struct device_attribute *attr, char *buf) +static ssize_t dapm_widget_show_codec(struct snd_soc_codec *codec, char *buf) { - struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); - struct snd_soc_codec *codec =rtd->codec; struct snd_soc_dapm_widget *w; int count = 0; char *state = "not set"; @@ -2142,6 +2138,21 @@ static ssize_t dapm_widget_show(struct device *dev, return count; } +/* show dapm widget status in sys fs */ +static ssize_t dapm_widget_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); + int i, count = 0; + + for (i = 0; i < rtd->num_codecs; i++) { + struct snd_soc_codec *codec = rtd->codec_dais[i]->codec; + count += dapm_widget_show_codec(codec, buf + count); + } + + return count; +} + static DEVICE_ATTR(dapm_widget, 0444, dapm_widget_show, NULL); int snd_soc_dapm_sys_add(struct device *dev) @@ -3395,25 +3406,18 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) return 0; } -void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card) +static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, + struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_pcm_runtime *rtd = card->rtd; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dapm_widget *sink, *source; - struct snd_soc_dai *cpu_dai, *codec_dai; + struct snd_soc_dapm_route r; int i; - /* for each BE DAI link... */ - for (i = 0; i < card->num_rtd; i++) { - rtd = &card->rtd[i]; - cpu_dai = rtd->cpu_dai; - codec_dai = rtd->codec_dai; + memset(&r, 0, sizeof(r)); - /* - * dynamic FE links have no fixed DAI mapping. - * CODEC<->CODEC links have no direct connection. - */ - if (rtd->dai_link->dynamic || rtd->dai_link->params) - continue; + for (i = 0; i < rtd->num_codecs; i++) { + struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; /* there is no point in connecting BE DAI links with dummies */ if (snd_soc_dai_is_dummy(codec_dai) || @@ -3475,11 +3479,34 @@ static void soc_dapm_dai_stream_event(struct snd_soc_dai *dai, int stream, } } +void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card) +{ + struct snd_soc_pcm_runtime *rtd = card->rtd; + int i; + + /* for each BE DAI link... */ + for (i = 0; i < card->num_rtd; i++) { + rtd = &card->rtd[i]; + + /* + * dynamic FE links have no fixed DAI mapping. + * CODEC<->CODEC links have no direct connection. + */ + if (rtd->dai_link->dynamic || rtd->dai_link->params) + continue; + + dapm_connect_dai_link_widgets(card, rtd); + } +} + static void soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, int event) { + int i; + soc_dapm_dai_stream_event(rtd->cpu_dai, stream, event); - soc_dapm_dai_stream_event(rtd->codec_dai, stream, event); + for (i = 0; i < rtd->num_codecs; i++) + soc_dapm_dai_stream_event(rtd->codec_dais[i], stream, event); dapm_power_widgets(rtd->card, event); } -- cgit v1.2.3 From 8151d5e60232d31f35e04f4e49a5b0d98b00a737 Mon Sep 17 00:00:00 2001 From: Benoit Cousson Date: Tue, 8 Jul 2014 23:19:37 +0200 Subject: ASoC: compress: Prevent multicodec for compressed stream Multiple codecs does not make sense in the case of compressed stream. Exit with error if it happens. Signed-off-by: Benoit Cousson Tested-by: Lars-Peter Clausen Reviewed-by: Lars-Peter Clausen Acked-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/soc-compress.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index f96fb96b2678..27c06acce205 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -629,6 +629,11 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) char new_name[64]; int ret = 0, direction = 0; + if (rtd->num_codecs > 1) { + dev_err(rtd->card->dev, "Multicodec not supported for compressed stream\n"); + return -EINVAL; + } + /* check client and interface hw capabilities */ snprintf(new_name, sizeof(new_name), "%s %s-%d", rtd->dai_link->stream_name, codec_dai->name, num); -- cgit v1.2.3 From 93e6958a3674d2fa42e2c24ad5156e65da1d8621 Mon Sep 17 00:00:00 2001 From: Benoit Cousson Date: Tue, 8 Jul 2014 23:19:38 +0200 Subject: ASoC: pcm: Add soc_dai_hw_params helper Add a function helper to factorize the hw_params code. Suggested by Lars-Peter Clausen Signed-off-by: Benoit Cousson Tested-by: Lars-Peter Clausen Reviewed-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 4 ++++ sound/soc/soc-dapm.c | 28 ++++++++-------------------- sound/soc/soc-pcm.c | 43 ++++++++++++++++++++++++------------------- 3 files changed, 36 insertions(+), 39 deletions(-) (limited to 'sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index f2142cf3f243..98555f833ab4 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -436,6 +436,10 @@ int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream, int snd_soc_platform_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_platform *platform); +int soc_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai); + /* Jack reporting */ int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, int type, struct snd_soc_jack *jack); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 5c63c3b49f3f..9a047f360993 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3214,27 +3214,15 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: - if (source->driver->ops && source->driver->ops->hw_params) { - substream.stream = SNDRV_PCM_STREAM_CAPTURE; - ret = source->driver->ops->hw_params(&substream, - params, source); - if (ret != 0) { - dev_err(source->dev, - "ASoC: hw_params() failed: %d\n", ret); - goto out; - } - } + substream.stream = SNDRV_PCM_STREAM_CAPTURE; + ret = soc_dai_hw_params(&substream, params, source); + if (ret < 0) + goto out; - if (sink->driver->ops && sink->driver->ops->hw_params) { - substream.stream = SNDRV_PCM_STREAM_PLAYBACK; - ret = sink->driver->ops->hw_params(&substream, params, - sink); - if (ret != 0) { - dev_err(sink->dev, - "ASoC: hw_params() failed: %d\n", ret); - goto out; - } - } + substream.stream = SNDRV_PCM_STREAM_PLAYBACK; + ret = soc_dai_hw_params(&substream, params, sink); + if (ret < 0) + goto out; break; case SND_SOC_DAPM_POST_PMU: diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index ec56d1831a86..3ec87d8aa181 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -789,6 +789,24 @@ static void soc_pcm_codec_params_fixup(struct snd_pcm_hw_params *params, interval->max = channels; } +int soc_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + int ret; + + if (dai->driver->ops && dai->driver->ops->hw_params) { + ret = dai->driver->ops->hw_params(substream, params, dai); + if (ret < 0) { + dev_err(dai->dev, "ASoC: can't set %s hw params: %d\n", + dai->name, ret); + return ret; + } + } + + return 0; +} + /* * Called by ALSA when the hardware params are set by application. This * function can also be called multiple times and can allocate buffers @@ -832,17 +850,9 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, soc_pcm_codec_params_fixup(&codec_params, codec_dai->rx_mask); - if (codec_dai->driver->ops && - codec_dai->driver->ops->hw_params) { - ret = codec_dai->driver->ops->hw_params(substream, - &codec_params, codec_dai); - if (ret < 0) { - dev_err(codec_dai->dev, - "ASoC: can't set %s hw params: %d\n", - codec_dai->name, ret); - goto codec_err; - } - } + ret = soc_dai_hw_params(substream, &codec_params, codec_dai); + if(ret < 0) + goto codec_err; codec_dai->rate = params_rate(&codec_params); codec_dai->channels = params_channels(&codec_params); @@ -850,14 +860,9 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, params_format(&codec_params)); } - if (cpu_dai->driver->ops && cpu_dai->driver->ops->hw_params) { - ret = cpu_dai->driver->ops->hw_params(substream, params, cpu_dai); - if (ret < 0) { - dev_err(cpu_dai->dev, "ASoC: %s hw params failed: %d\n", - cpu_dai->name, ret); - goto interface_err; - } - } + ret = soc_dai_hw_params(substream, params, cpu_dai); + if (ret < 0) + goto interface_err; if (platform->driver->ops && platform->driver->ops->hw_params) { ret = platform->driver->ops->hw_params(substream, params); -- cgit v1.2.3 From afdb74fd708fb4330485212f76a70b91967b1f70 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 14 Jul 2014 10:35:40 +0800 Subject: ASoC: Intel: Add Broadwell Machine support Add support for Broadwell based machines with SST DSP audio. Signed-off-by: Liam Girdwood Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/Kconfig | 12 +++ sound/soc/intel/Makefile | 2 + sound/soc/intel/broadwell.c | 251 ++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 265 insertions(+) create mode 100644 sound/soc/intel/broadwell.c (limited to 'sound') diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index c30fedb3e149..0b305f9da3db 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -58,3 +58,15 @@ config SND_SOC_INTEL_BYT_MAX98090_MACH help This adds audio driver for Intel Baytrail platform based boards with the MAX98090 audio codec. + +config SND_SOC_INTEL_BROADWELL_MACH + tristate "ASoC Audio DSP support for Intel Broadwell Wildcatpoint" + depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS + select SND_SOC_INTEL_HASWELL + select SND_COMPRESS_OFFLOAD + select SND_SOC_RT286 + help + This adds support for the Wilcatpoint Audio DSP on Intel(R) Broadwell + Ultrabook platforms. + Say Y if you have such a device + If unsure select "N". diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 4bfca79a42ba..7acbfc43a0c6 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -24,7 +24,9 @@ obj-$(CONFIG_SND_SOC_INTEL_BAYTRAIL) += snd-soc-sst-baytrail-pcm.o snd-soc-sst-haswell-objs := haswell.o snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o +snd-soc-sst-broadwell-objs := broadwell.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o +obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o diff --git a/sound/soc/intel/broadwell.c b/sound/soc/intel/broadwell.c new file mode 100644 index 000000000000..0e550f14028f --- /dev/null +++ b/sound/soc/intel/broadwell.c @@ -0,0 +1,251 @@ +/* + * Intel Broadwell Wildcatpoint SST Audio + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License version + * 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + */ + +#include +#include +#include +#include +#include +#include + +#include "sst-dsp.h" +#include "sst-haswell-ipc.h" + +#include "../codecs/rt286.h" + +static const struct snd_soc_dapm_widget broadwell_widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_MIC("DMIC1", NULL), + SND_SOC_DAPM_MIC("DMIC2", NULL), + SND_SOC_DAPM_LINE("Line Jack", NULL), +}; + +static const struct snd_soc_dapm_route broadwell_rt286_map[] = { + + /* speaker */ + {"Speaker", NULL, "SPOR"}, + {"Speaker", NULL, "SPOL"}, + + /* HP jack connectors - unknown if we have jack deteck */ + {"Headphones", NULL, "HPO Pin"}, + + /* other jacks */ + {"MIC1", NULL, "Mic Jack"}, + {"LINE1", NULL, "Line Jack"}, + + /* digital mics */ + {"DMIC1 Pin", NULL, "DMIC1"}, + {"DMIC2 Pin", NULL, "DMIC2"}, + + /* CODEC BE connections */ + {"SSP0 CODEC IN", NULL, "AIF1 Capture"}, + {"AIF1 Playback", NULL, "SSP0 CODEC OUT"}, +}; + +static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* The ADSP will covert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP0 to 16 bit */ + snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - + SNDRV_PCM_HW_PARAM_FIRST_MASK], + SNDRV_PCM_FORMAT_S16_LE); + return 0; +} + +static int broadwell_rt286_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, RT286_SCLK_S_PLL, 24000000, + SND_SOC_CLOCK_IN); + + if (ret < 0) { + dev_err(rtd->dev, "can't set codec sysclk configuration\n"); + return ret; + } + + return ret; +} + +static struct snd_soc_ops broadwell_rt286_ops = { + .hw_params = broadwell_rt286_hw_params, +}; + +static int broadwell_rtd_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + struct sst_pdata *pdata = dev_get_platdata(rtd->platform->dev); + struct sst_hsw *broadwell = pdata->dsp; + int ret; + + /* Set ADSP SSP port settings */ + ret = sst_hsw_device_set_config(broadwell, SST_HSW_DEVICE_SSP_0, + SST_HSW_DEVICE_MCLK_FREQ_24_MHZ, + SST_HSW_DEVICE_CLOCK_MASTER, 9); + if (ret < 0) { + dev_err(rtd->dev, "error: failed to set device config\n"); + return ret; + } + + /* always connected - check HP for jack detect */ + snd_soc_dapm_enable_pin(dapm, "Headphones"); + snd_soc_dapm_enable_pin(dapm, "Speaker"); + snd_soc_dapm_enable_pin(dapm, "Mic Jack"); + snd_soc_dapm_enable_pin(dapm, "Line Jack"); + snd_soc_dapm_enable_pin(dapm, "DMIC1"); + snd_soc_dapm_enable_pin(dapm, "DMIC2"); + + return 0; +} + +/* broadwell digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link broadwell_rt286_dais[] = { + /* Front End DAI links */ + { + .name = "System PCM", + .stream_name = "System Playback", + .cpu_dai_name = "System Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .init = broadwell_rtd_init, + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + { + .name = "Offload0", + .stream_name = "Offload0 Playback", + .cpu_dai_name = "Offload0 Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + { + .name = "Offload1", + .stream_name = "Offload1 Playback", + .cpu_dai_name = "Offload1 Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + { + .name = "Loopback PCM", + .stream_name = "Loopback", + .cpu_dai_name = "Loopback Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 0, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + }, + { + .name = "Capture PCM", + .stream_name = "Capture", + .cpu_dai_name = "Capture Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + }, + + /* Back End DAI links */ + { + /* SSP0 - Codec */ + .name = "Codec", + .be_id = 0, + .cpu_dai_name = "snd-soc-dummy-dai", + .platform_name = "snd-soc-dummy", + .no_pcm = 1, + .codec_name = "i2c-INT343A:00", + .codec_dai_name = "rt286-aif1", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_suspend = 1, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = broadwell_ssp0_fixup, + .ops = &broadwell_rt286_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, +}; + +/* broadwell audio machine driver for WPT + RT286S */ +static struct snd_soc_card broadwell_rt286 = { + .name = "broadwell-rt286", + .owner = THIS_MODULE, + .dai_link = broadwell_rt286_dais, + .num_links = ARRAY_SIZE(broadwell_rt286_dais), + .dapm_widgets = broadwell_widgets, + .num_dapm_widgets = ARRAY_SIZE(broadwell_widgets), + .dapm_routes = broadwell_rt286_map, + .num_dapm_routes = ARRAY_SIZE(broadwell_rt286_map), + .fully_routed = true, +}; + +static int broadwell_audio_probe(struct platform_device *pdev) +{ + broadwell_rt286.dev = &pdev->dev; + + return snd_soc_register_card(&broadwell_rt286); +} + +static int broadwell_audio_remove(struct platform_device *pdev) +{ + snd_soc_unregister_card(&broadwell_rt286); + return 0; +} + +static struct platform_driver broadwell_audio = { + .probe = broadwell_audio_probe, + .remove = broadwell_audio_remove, + .driver = { + .name = "broadwell-audio", + .owner = THIS_MODULE, + }, +}; + +module_platform_driver(broadwell_audio) + +/* Module information */ +MODULE_AUTHOR("Liam Girdwood, Xingchao Wang"); +MODULE_DESCRIPTION("Intel SST Audio for WPT/Broadwell"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:broadwell-audio"); -- cgit v1.2.3 From 3d9501aff3032f41a0767906ebb5bd06a3753bde Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 17 Jul 2014 09:15:34 +0200 Subject: ASoC: Remove unused 'r' variable from dapm_connect_dai_link_widgets() It was accidentally added in commit 44ba2641 ("ASoC: dapm: Add support for DAI multicodec"). Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9a047f360993..8cb68a38ad19 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3399,11 +3399,8 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, { struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dapm_widget *sink, *source; - struct snd_soc_dapm_route r; int i; - memset(&r, 0, sizeof(r)); - for (i = 0; i < rtd->num_codecs; i++) { struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; -- cgit v1.2.3 From 30443408fd7201fd1911b09daccf92fae3cc700d Mon Sep 17 00:00:00 2001 From: Scott Jiang Date: Fri, 18 Jul 2014 16:14:57 +0800 Subject: ASoC: blackfin: use samples to set silence The third parameter for snd_pcm_format_set_silence needs the number of samples instead of sample bytes. Signed-off-by: Scott Jiang Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/blackfin/bf5xx-i2s-pcm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index a3881c4381c9..bcf591373a7a 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -290,19 +290,19 @@ static int bf5xx_pcm_silence(struct snd_pcm_substream *substream, unsigned int sample_size = runtime->sample_bits / 8; void *buf = runtime->dma_area; struct bf5xx_i2s_pcm_data *dma_data; - unsigned int offset, size; + unsigned int offset, samples; dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); if (dma_data->tdm_mode) { offset = pos * 8 * sample_size; - size = count * 8 * sample_size; + samples = count * 8; } else { offset = frames_to_bytes(runtime, pos); - size = frames_to_bytes(runtime, count); + samples = count * runtime->channels; } - snd_pcm_format_set_silence(runtime->format, buf + offset, size); + snd_pcm_format_set_silence(runtime->format, buf + offset, samples); return 0; } -- cgit v1.2.3 From 5df7f71d5cdfbcbfd7e1b68df9994609d33f7e58 Mon Sep 17 00:00:00 2001 From: Dan Murphy Date: Mon, 14 Jul 2014 15:10:45 -0500 Subject: ASoC: tas2552: Support TI TAS2552 Amplifier Support the TI TAS2552 Class D amplifier. The TAS2552 is a high efficiency Class-D audio power amplifier with advanced battery current management and an integrated Class-G boost The device constantly measures the current and voltage across the load and provides a digital stream of this information. Signed-off-by: Dan Murphy Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/tas2552.txt | 26 + include/sound/tas2552-plat.h | 25 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2552.c | 540 +++++++++++++++++++++ sound/soc/codecs/tas2552.h | 129 +++++ 6 files changed, 727 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas2552.txt create mode 100644 include/sound/tas2552-plat.h create mode 100644 sound/soc/codecs/tas2552.c create mode 100644 sound/soc/codecs/tas2552.h (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/tas2552.txt b/Documentation/devicetree/bindings/sound/tas2552.txt new file mode 100644 index 000000000000..55e2a0af5645 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2552.txt @@ -0,0 +1,26 @@ +Texas Instruments - tas2552 Codec module + +The tas2552 serial control bus communicates through I2C protocols + +Required properties: + - compatible - One of: + "ti,tas2552" - TAS2552 + - reg - I2C slave address + - supply-*: Required supply regulators are: + "vbat" battery voltage + "iovdd" I/O Voltage + "avdd" Analog DAC Voltage + +Optional properties: + - enable-gpio - gpio pin to enable/disable the device + +Example: + +tas2552: tas2552@41 { + compatible = "ti,tas2552"; + reg = <0x41>; + enable-gpio = <&gpio4 2 GPIO_ACTIVE_HIGH>; +}; + +For more product information please see the link below: +http://www.ti.com/product/TAS2552 diff --git a/include/sound/tas2552-plat.h b/include/sound/tas2552-plat.h new file mode 100644 index 000000000000..65e7627ba38e --- /dev/null +++ b/include/sound/tas2552-plat.h @@ -0,0 +1,25 @@ +/* + * TAS2552 driver platform header + * + * Copyright (C) 2014 Texas Instruments Inc. + * + * Author: Dan Murphy + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#ifndef TAS2552_PLAT_H +#define TAS2552_PLAT_H + +struct tas2552_platform_data { + int enable_gpio; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index cbfa1e18f651..480881747ec5 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -91,6 +91,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_STA350 if I2C select SND_SOC_STA529 if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS + select SND_SOC_TAS2552 if I2C select SND_SOC_TAS5086 if I2C select SND_SOC_TLV320AIC23_I2C if I2C select SND_SOC_TLV320AIC23_SPI if SPI_MASTER @@ -513,6 +514,10 @@ config SND_SOC_STA529 config SND_SOC_STAC9766 tristate +config SND_SOC_TAS2552 + tristate "Texas Instruments TAS2552 Mono Audio amplifier" + depends on I2C + config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index be3377b8d73f..d79de052ef8e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -160,6 +160,7 @@ snd-soc-wm-hubs-objs := wm_hubs.o # Amp snd-soc-max9877-objs := max9877.o snd-soc-tpa6130a2-objs := tpa6130a2.o +snd-soc-tas2552-objs := tas2552.o obj-$(CONFIG_SND_SOC_88PM860X) += snd-soc-88pm860x.o obj-$(CONFIG_SND_SOC_AB8500_CODEC) += snd-soc-ab8500-codec.o @@ -251,6 +252,7 @@ obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o obj-$(CONFIG_SND_SOC_STA350) += snd-soc-sta350.o obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o +obj-$(CONFIG_SND_SOC_TAS2552) += snd-soc-tas2552.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC23_I2C) += snd-soc-tlv320aic23-i2c.o diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c new file mode 100644 index 000000000000..f0760af5a21e --- /dev/null +++ b/sound/soc/codecs/tas2552.c @@ -0,0 +1,540 @@ +/* + * tas2552.c - ALSA SoC Texas Instruments TAS2552 Mono Audio Amplifier + * + * Copyright (C) 2014 Texas Instruments Incorporated - http://www.ti.com + * + * Author: Dan Murphy + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include + +#include +#include +#include +#include +#include +#include + +#include "tas2552.h" + +static struct reg_default tas2552_reg_defs[] = { + {TAS2552_CFG_1, 0x22}, + {TAS2552_CFG_3, 0x80}, + {TAS2552_DOUT, 0x00}, + {TAS2552_OUTPUT_DATA, 0xc0}, + {TAS2552_PDM_CFG, 0x01}, + {TAS2552_PGA_GAIN, 0x00}, + {TAS2552_BOOST_PT_CTRL, 0x0f}, + {TAS2552_RESERVED_0D, 0x00}, + {TAS2552_LIMIT_RATE_HYS, 0x08}, + {TAS2552_CFG_2, 0xef}, + {TAS2552_SER_CTRL_1, 0x00}, + {TAS2552_SER_CTRL_2, 0x00}, + {TAS2552_PLL_CTRL_1, 0x10}, + {TAS2552_PLL_CTRL_2, 0x00}, + {TAS2552_PLL_CTRL_3, 0x00}, + {TAS2552_BTIP, 0x8f}, + {TAS2552_BTS_CTRL, 0x80}, + {TAS2552_LIMIT_RELEASE, 0x04}, + {TAS2552_LIMIT_INT_COUNT, 0x00}, + {TAS2552_EDGE_RATE_CTRL, 0x40}, + {TAS2552_VBAT_DATA, 0x00}, +}; + +#define TAS2552_NUM_SUPPLIES 3 +static const char *tas2552_supply_names[TAS2552_NUM_SUPPLIES] = { + "vbat", /* vbat voltage */ + "iovdd", /* I/O Voltage */ + "avdd", /* Analog DAC Voltage */ +}; + +struct tas2552_data { + struct snd_soc_codec *codec; + struct regmap *regmap; + struct i2c_client *tas2552_client; + struct regulator_bulk_data supplies[TAS2552_NUM_SUPPLIES]; + struct gpio_desc *enable_gpio; + unsigned char regs[TAS2552_VBAT_DATA]; + unsigned int mclk; +}; + +static void tas2552_sw_shutdown(struct tas2552_data *tas_data, int sw_shutdown) +{ + u8 cfg1_reg; + + if (sw_shutdown) + cfg1_reg = 0; + else + cfg1_reg = TAS2552_SWS_MASK; + + snd_soc_update_bits(tas_data->codec, TAS2552_CFG_1, + TAS2552_SWS_MASK, cfg1_reg); +} + +static int tas2552_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct tas2552_data *tas2552 = dev_get_drvdata(codec->dev); + int sample_rate, pll_clk; + int d; + u8 p, j; + + /* Turn on Class D amplifier */ + snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_CLASSD_EN_MASK, + TAS2552_CLASSD_EN); + + if (!tas2552->mclk) + return -EINVAL; + + snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_PLL_ENABLE, 0); + + if (tas2552->mclk == TAS2552_245MHZ_CLK || + tas2552->mclk == TAS2552_225MHZ_CLK) { + /* By pass the PLL configuration */ + snd_soc_update_bits(codec, TAS2552_PLL_CTRL_2, + TAS2552_PLL_BYPASS_MASK, + TAS2552_PLL_BYPASS); + } else { + /* Fill in the PLL control registers for J & D + * PLL_CLK = (.5 * freq * J.D) / 2^p + * Need to fill in J and D here based on incoming freq + */ + p = snd_soc_read(codec, TAS2552_PLL_CTRL_1); + p = (p >> 7); + sample_rate = params_rate(params); + + if (sample_rate == 48000) + pll_clk = TAS2552_245MHZ_CLK; + else if (sample_rate == 44100) + pll_clk = TAS2552_225MHZ_CLK; + else { + dev_vdbg(codec->dev, "Substream sample rate is not found %i\n", + params_rate(params)); + return -EINVAL; + } + + j = (pll_clk * 2 * (1 << p)) / tas2552->mclk; + d = (pll_clk * 2 * (1 << p)) % tas2552->mclk; + + snd_soc_update_bits(codec, TAS2552_PLL_CTRL_1, + TAS2552_PLL_J_MASK, j); + snd_soc_write(codec, TAS2552_PLL_CTRL_2, + (d >> 7) & TAS2552_PLL_D_UPPER_MASK); + snd_soc_write(codec, TAS2552_PLL_CTRL_3, + d & TAS2552_PLL_D_LOWER_MASK); + + } + + snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_PLL_ENABLE, + TAS2552_PLL_ENABLE); + + return 0; +} + +static int tas2552_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + u8 serial_format; + u8 serial_control_mask; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + serial_format = 0x00; + break; + case SND_SOC_DAIFMT_CBS_CFM: + serial_format = TAS2552_WORD_CLK_MASK; + break; + case SND_SOC_DAIFMT_CBM_CFS: + serial_format = TAS2552_BIT_CLK_MASK; + break; + case SND_SOC_DAIFMT_CBM_CFM: + serial_format = (TAS2552_BIT_CLK_MASK | TAS2552_WORD_CLK_MASK); + break; + default: + dev_vdbg(codec->dev, "DAI Format master is not found\n"); + return -EINVAL; + } + + serial_control_mask = TAS2552_BIT_CLK_MASK | TAS2552_WORD_CLK_MASK; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + serial_format &= TAS2552_DAIFMT_I2S_MASK; + break; + case SND_SOC_DAIFMT_DSP_A: + serial_format |= TAS2552_DAIFMT_DSP; + break; + case SND_SOC_DAIFMT_RIGHT_J: + serial_format |= TAS2552_DAIFMT_RIGHT_J; + break; + case SND_SOC_DAIFMT_LEFT_J: + serial_format |= TAS2552_DAIFMT_LEFT_J; + break; + default: + dev_vdbg(codec->dev, "DAI Format is not found\n"); + return -EINVAL; + } + + if (fmt & SND_SOC_DAIFMT_FORMAT_MASK) + serial_control_mask |= TAS2552_DATA_FORMAT_MASK; + + snd_soc_update_bits(codec, TAS2552_SER_CTRL_1, serial_control_mask, + serial_format); + + return 0; +} + +static int tas2552_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct tas2552_data *tas2552 = dev_get_drvdata(codec->dev); + + tas2552->mclk = freq; + + return 0; +} + +static int tas2552_mute(struct snd_soc_dai *dai, int mute) +{ + u8 cfg1_reg; + struct snd_soc_codec *codec = dai->codec; + + if (mute) + cfg1_reg = TAS2552_MUTE_MASK; + else + cfg1_reg = ~TAS2552_MUTE_MASK; + + snd_soc_update_bits(codec, TAS2552_CFG_1, TAS2552_MUTE_MASK, cfg1_reg); + + return 0; +} + +#ifdef CONFIG_PM_RUNTIME +static int tas2552_runtime_suspend(struct device *dev) +{ + struct tas2552_data *tas2552 = dev_get_drvdata(dev); + + tas2552_sw_shutdown(tas2552, 0); + + if (tas2552->enable_gpio) + gpiod_set_value(tas2552->enable_gpio, 0); + + regcache_cache_only(tas2552->regmap, true); + regcache_mark_dirty(tas2552->regmap); + + return 0; +} + +static int tas2552_runtime_resume(struct device *dev) +{ + struct tas2552_data *tas2552 = dev_get_drvdata(dev); + + if (tas2552->enable_gpio) + gpiod_set_value(tas2552->enable_gpio, 1); + + tas2552_sw_shutdown(tas2552, 1); + + regcache_cache_only(tas2552->regmap, false); + regcache_sync(tas2552->regmap); + + return 0; +} +#endif + +static const struct dev_pm_ops tas2552_pm = { + SET_RUNTIME_PM_OPS(tas2552_runtime_suspend, tas2552_runtime_resume, + NULL) +}; + +static void tas2552_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + + snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_PLL_ENABLE, 0); +} + +static struct snd_soc_dai_ops tas2552_speaker_dai_ops = { + .hw_params = tas2552_hw_params, + .set_sysclk = tas2552_set_dai_sysclk, + .set_fmt = tas2552_set_dai_fmt, + .shutdown = tas2552_shutdown, + .digital_mute = tas2552_mute, +}; + +/* Formats supported by TAS2552 driver. */ +#define TAS2552_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +/* TAS2552 dai structure. */ +static struct snd_soc_dai_driver tas2552_dai[] = { + { + .name = "tas2552-amplifier", + .playback = { + .stream_name = "Speaker", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = TAS2552_FORMATS, + }, + .ops = &tas2552_speaker_dai_ops, + }, +}; + +/* + * DAC digital volumes. From -7 to 24 dB in 1 dB steps + */ +static DECLARE_TLV_DB_SCALE(dac_tlv, -7, 100, 24); + +static const struct snd_kcontrol_new tas2552_snd_controls[] = { + SOC_SINGLE_TLV("Speaker Driver Playback Volume", + TAS2552_PGA_GAIN, 0, 0x1f, 1, dac_tlv), +}; + +static const struct reg_default tas2552_init_regs[] = { + { TAS2552_RESERVED_0D, 0xc0 }, +}; + +static int tas2552_codec_probe(struct snd_soc_codec *codec) +{ + struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); + int ret; + + tas2552->codec = codec; + + ret = regulator_bulk_enable(ARRAY_SIZE(tas2552->supplies), + tas2552->supplies); + + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", + ret); + return ret; + } + + if (tas2552->enable_gpio) + gpiod_set_value(tas2552->enable_gpio, 1); + + ret = pm_runtime_get_sync(codec->dev); + if (ret < 0) { + dev_err(codec->dev, "Enabling device failed: %d\n", + ret); + goto probe_fail; + } + + snd_soc_write(codec, TAS2552_CFG_1, TAS2552_MUTE_MASK | + TAS2552_PLL_SRC_BCLK); + snd_soc_write(codec, TAS2552_CFG_3, TAS2552_I2S_OUT_SEL | + TAS2552_DIN_SRC_SEL_AVG_L_R | TAS2552_88_96KHZ); + snd_soc_write(codec, TAS2552_DOUT, TAS2552_PDM_DATA_I); + snd_soc_write(codec, TAS2552_OUTPUT_DATA, TAS2552_PDM_DATA_V_I | 0x8); + snd_soc_write(codec, TAS2552_PDM_CFG, TAS2552_PDM_BCLK_SEL); + snd_soc_write(codec, TAS2552_BOOST_PT_CTRL, TAS2552_APT_DELAY_200 | + TAS2552_APT_THRESH_2_1_7); + + ret = regmap_register_patch(tas2552->regmap, tas2552_init_regs, + ARRAY_SIZE(tas2552_init_regs)); + if (ret != 0) { + dev_err(codec->dev, "Failed to write init registers: %d\n", + ret); + goto patch_fail; + } + + snd_soc_write(codec, TAS2552_CFG_2, TAS2552_CLASSD_EN | + TAS2552_BOOST_EN | TAS2552_APT_EN | + TAS2552_LIM_EN); + return 0; + +patch_fail: + pm_runtime_put(codec->dev); +probe_fail: + if (tas2552->enable_gpio) + gpiod_set_value(tas2552->enable_gpio, 0); + + regulator_bulk_disable(ARRAY_SIZE(tas2552->supplies), + tas2552->supplies); + return -EIO; +} + +static int tas2552_codec_remove(struct snd_soc_codec *codec) +{ + struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); + + if (tas2552->enable_gpio) + gpiod_set_value(tas2552->enable_gpio, 0); + + return 0; +}; + +#ifdef CONFIG_PM +static int tas2552_suspend(struct snd_soc_codec *codec) +{ + struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = regulator_bulk_disable(ARRAY_SIZE(tas2552->supplies), + tas2552->supplies); + + if (ret != 0) + dev_err(codec->dev, "Failed to disable supplies: %d\n", + ret); + return 0; +} + +static int tas2552_resume(struct snd_soc_codec *codec) +{ + struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = regulator_bulk_enable(ARRAY_SIZE(tas2552->supplies), + tas2552->supplies); + + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", + ret); + } + + return 0; +} +#else +#define tas2552_suspend NULL +#define tas2552_resume NULL +#endif + +static struct snd_soc_codec_driver soc_codec_dev_tas2552 = { + .probe = tas2552_codec_probe, + .remove = tas2552_codec_remove, + .suspend = tas2552_suspend, + .resume = tas2552_resume, + .controls = tas2552_snd_controls, + .num_controls = ARRAY_SIZE(tas2552_snd_controls), +}; + +static const struct regmap_config tas2552_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = TAS2552_MAX_REG, + .reg_defaults = tas2552_reg_defs, + .num_reg_defaults = ARRAY_SIZE(tas2552_reg_defs), + .cache_type = REGCACHE_RBTREE, +}; + +static int tas2552_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct device *dev; + struct tas2552_data *data; + int ret; + int i; + + dev = &client->dev; + data = devm_kzalloc(&client->dev, sizeof(*data), GFP_KERNEL); + if (data == NULL) + return -ENOMEM; + + data->enable_gpio = devm_gpiod_get(dev, "enable"); + if (IS_ERR(data->enable_gpio)) { + ret = PTR_ERR(data->enable_gpio); + if (ret != -ENOENT && ret != -ENOSYS) + return ret; + + data->enable_gpio = NULL; + } else { + gpiod_direction_output(data->enable_gpio, 0); + } + + data->tas2552_client = client; + data->regmap = devm_regmap_init_i2c(client, &tas2552_regmap_config); + if (IS_ERR(data->regmap)) { + ret = PTR_ERR(data->regmap); + dev_err(&client->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + for (i = 0; i < ARRAY_SIZE(data->supplies); i++) + data->supplies[i].supply = tas2552_supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(data->supplies), + data->supplies); + if (ret != 0) + dev_err(dev, "Failed to request supplies: %d\n", ret); + + pm_runtime_set_active(&client->dev); + pm_runtime_set_autosuspend_delay(&client->dev, 1000); + pm_runtime_use_autosuspend(&client->dev); + pm_runtime_enable(&client->dev); + pm_runtime_mark_last_busy(&client->dev); + pm_runtime_put_sync_autosuspend(&client->dev); + + dev_set_drvdata(&client->dev, data); + + ret = snd_soc_register_codec(&client->dev, + &soc_codec_dev_tas2552, + tas2552_dai, ARRAY_SIZE(tas2552_dai)); + if (ret < 0) + dev_err(&client->dev, "Failed to register codec: %d\n", ret); + + return 0; +} + +static int tas2552_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id tas2552_id[] = { + { "tas2552", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, tas2552_id); + +#if IS_ENABLED(CONFIG_OF) +static const struct of_device_id tas2552_of_match[] = { + { .compatible = "ti,tas2552", }, + {}, +}; +MODULE_DEVICE_TABLE(of, tas2552_of_match); +#endif + +static struct i2c_driver tas2552_i2c_driver = { + .driver = { + .name = "tas2552", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(tas2552_of_match), + .pm = &tas2552_pm, + }, + .probe = tas2552_probe, + .remove = tas2552_i2c_remove, + .id_table = tas2552_id, +}; + +module_i2c_driver(tas2552_i2c_driver); + +MODULE_AUTHOR("Dan Muprhy "); +MODULE_DESCRIPTION("TAS2552 Audio amplifier driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2552.h b/sound/soc/codecs/tas2552.h new file mode 100644 index 000000000000..6cea8f31bf88 --- /dev/null +++ b/sound/soc/codecs/tas2552.h @@ -0,0 +1,129 @@ +/* + * tas2552.h - ALSA SoC Texas Instruments TAS2552 Mono Audio Amplifier + * + * Copyright (C) 2014 Texas Instruments Incorporated - http://www.ti.com + * + * Author: Dan Murphy + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#ifndef __TAS2552_H__ +#define __TAS2552_H__ + +/* Register Address Map */ +#define TAS2552_DEVICE_STATUS 0x00 +#define TAS2552_CFG_1 0x01 +#define TAS2552_CFG_2 0x02 +#define TAS2552_CFG_3 0x03 +#define TAS2552_DOUT 0x04 +#define TAS2552_SER_CTRL_1 0x05 +#define TAS2552_SER_CTRL_2 0x06 +#define TAS2552_OUTPUT_DATA 0x07 +#define TAS2552_PLL_CTRL_1 0x08 +#define TAS2552_PLL_CTRL_2 0x09 +#define TAS2552_PLL_CTRL_3 0x0a +#define TAS2552_BTIP 0x0b +#define TAS2552_BTS_CTRL 0x0c +#define TAS2552_RESERVED_0D 0x0d +#define TAS2552_LIMIT_RATE_HYS 0x0e +#define TAS2552_LIMIT_RELEASE 0x0f +#define TAS2552_LIMIT_INT_COUNT 0x10 +#define TAS2552_PDM_CFG 0x11 +#define TAS2552_PGA_GAIN 0x12 +#define TAS2552_EDGE_RATE_CTRL 0x13 +#define TAS2552_BOOST_PT_CTRL 0x14 +#define TAS2552_VER_NUM 0x16 +#define TAS2552_VBAT_DATA 0x19 +#define TAS2552_MAX_REG 0x20 + +/* CFG1 Register Masks */ +#define TAS2552_MUTE_MASK (1 << 2) +#define TAS2552_SWS_MASK (1 << 1) +#define TAS2552_WCLK_MASK 0x07 +#define TAS2552_CLASSD_EN_MASK (1 << 7) + +/* CFG2 Register Masks */ +#define TAS2552_CLASSD_EN (1 << 7) +#define TAS2552_BOOST_EN (1 << 6) +#define TAS2552_APT_EN (1 << 5) +#define TAS2552_PLL_ENABLE (1 << 3) +#define TAS2552_LIM_EN (1 << 2) +#define TAS2552_IVSENSE_EN (1 << 1) + +/* CFG3 Register Masks */ +#define TAS2552_WORD_CLK_MASK (1 << 7) +#define TAS2552_BIT_CLK_MASK (1 << 6) +#define TAS2552_DATA_FORMAT_MASK (0x11 << 2) + +#define TAS2552_DAIFMT_I2S_MASK 0xf3 +#define TAS2552_DAIFMT_DSP (1 << 3) +#define TAS2552_DAIFMT_RIGHT_J (1 << 4) +#define TAS2552_DAIFMT_LEFT_J (0x11 << 3) + +#define TAS2552_PLL_SRC_MCLK 0x00 +#define TAS2552_PLL_SRC_BCLK (1 << 3) +#define TAS2552_PLL_SRC_IVCLKIN (1 << 4) +#define TAS2552_PLL_SRC_1_8_FIXED (0x11 << 3) + +#define TAS2552_DIN_SRC_SEL_MUTED 0x00 +#define TAS2552_DIN_SRC_SEL_LEFT (1 << 4) +#define TAS2552_DIN_SRC_SEL_RIGHT (1 << 5) +#define TAS2552_DIN_SRC_SEL_AVG_L_R (0x11 << 4) + +#define TAS2552_PDM_IN_SEL (1 << 5) +#define TAS2552_I2S_OUT_SEL (1 << 6) +#define TAS2552_ANALOG_IN_SEL (1 << 7) + +/* CFG3 WCLK Dividers */ +#define TAS2552_8KHZ 0x00 +#define TAS2552_11_12KHZ (1 << 1) +#define TAS2552_16KHZ (1 << 2) +#define TAS2552_22_24KHZ (1 << 3) +#define TAS2552_32KHZ (1 << 4) +#define TAS2552_44_48KHZ (1 << 5) +#define TAS2552_88_96KHZ (1 << 6) +#define TAS2552_176_192KHZ (1 << 7) + +/* OUTPUT_DATA register */ +#define TAS2552_PDM_DATA_I 0x00 +#define TAS2552_PDM_DATA_V (1 << 6) +#define TAS2552_PDM_DATA_I_V (1 << 7) +#define TAS2552_PDM_DATA_V_I (0x11 << 6) + +/* PDM CFG Register */ +#define TAS2552_PDM_DATA_ES_RISE 0x4 + +#define TAS2552_PDM_PLL_CLK_SEL 0x00 +#define TAS2552_PDM_IV_CLK_SEL (1 << 1) +#define TAS2552_PDM_BCLK_SEL (1 << 2) +#define TAS2552_PDM_MCLK_SEL (1 << 3) + +/* Boost pass-through register */ +#define TAS2552_APT_DELAY_50 0x00 +#define TAS2552_APT_DELAY_75 (1 << 1) +#define TAS2552_APT_DELAY_125 (1 << 2) +#define TAS2552_APT_DELAY_200 (1 << 3) + +#define TAS2552_APT_THRESH_2_5 0x00 +#define TAS2552_APT_THRESH_1_7 (1 << 3) +#define TAS2552_APT_THRESH_1_4_1_1 (1 << 4) +#define TAS2552_APT_THRESH_2_1_7 (0x11 << 2) + +/* PLL Control Register */ +#define TAS2552_245MHZ_CLK 24576000 +#define TAS2552_225MHZ_CLK 22579200 +#define TAS2552_PLL_J_MASK 0x7f +#define TAS2552_PLL_D_UPPER_MASK 0x3f +#define TAS2552_PLL_D_LOWER_MASK 0xff +#define TAS2552_PLL_BYPASS_MASK 0x80 +#define TAS2552_PLL_BYPASS 0x80 + +#endif -- cgit v1.2.3 From eff952b733d4c1ff3a6b35accce940b223372978 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Thu, 17 Jul 2014 21:21:37 +0800 Subject: ASoC: fsl_sai: Reset FIFOs after disabling TE/RE SAI will not clear their FIFOs after disabling TE/RE. Therfore, the driver should take care the task so as not to let useless data remain in the FIFO. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index c5a0e8af8226..b10dbd835d5c 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -371,10 +371,13 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, /* Check if the opposite FRDE is also disabled */ if (!(tx ? rcsr & FSL_SAI_CSR_FRDE : tcsr & FSL_SAI_CSR_FRDE)) { + /* Disable both directions and reset their FIFOs */ regmap_update_bits(sai->regmap, FSL_SAI_TCSR, - FSL_SAI_CSR_TERE, 0); + FSL_SAI_CSR_TERE | FSL_SAI_CSR_FR, + FSL_SAI_CSR_FR); regmap_update_bits(sai->regmap, FSL_SAI_RCSR, - FSL_SAI_CSR_TERE, 0); + FSL_SAI_CSR_TERE | FSL_SAI_CSR_FR, + FSL_SAI_CSR_FR); } break; default: -- cgit v1.2.3 From 4800f88b615f194ae3c1577038a7ccd871c907c9 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Thu, 17 Jul 2014 21:21:38 +0800 Subject: ASoC: fsl_sai: Fix incorrect register writing in fsl_sai_isr() In the rx irq handling part, we should clear the flags in RCSR not TCSR. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index b10dbd835d5c..1b6ee2ce849f 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -106,7 +106,7 @@ irq_rx: xcsr &= ~FSL_SAI_CSR_xF_MASK; if (flags) - regmap_write(sai->regmap, FSL_SAI_TCSR, flags | xcsr); + regmap_write(sai->regmap, FSL_SAI_RCSR, flags | xcsr); out: if (irq_none) -- cgit v1.2.3 From 95b47f8de787214f7db88b26759d7edc7c64d74a Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Thu, 17 Jul 2014 13:16:54 -0500 Subject: ASoC: cs42l56: Fix stereo channel register assignment for Headphone and LineOut volume mixers Stereo Headphone and LineOut volume mixers are now attached to HPA+HPB, LOA+LOB. Reported-by: Ryan Harvey Signed-off-by: Ryan Harvey Signed-off-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l56.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index fdc4bd27b0df..8e68ef5de849 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -445,9 +445,9 @@ static const struct snd_kcontrol_new cs42l56_snd_controls[] = { SOC_DOUBLE("ADC Boost Switch", CS42L56_GAIN_BIAS_CTL, 3, 2, 1, 1), SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L56_HPA_VOLUME, - CS42L56_HPA_VOLUME, 0, 0x44, 0x55, hl_tlv), + CS42L56_HPB_VOLUME, 0, 0x44, 0x55, hl_tlv), SOC_DOUBLE_R_SX_TLV("LineOut Volume", CS42L56_LOA_VOLUME, - CS42L56_LOA_VOLUME, 0, 0x44, 0x55, hl_tlv), + CS42L56_LOB_VOLUME, 0, 0x44, 0x55, hl_tlv), SOC_SINGLE_TLV("Bass Shelving Volume", CS42L56_TONE_CTL, 0, 0x00, 1, tone_tlv), -- cgit v1.2.3 From a046558719770f94d8660f77654ce9cc712ffe54 Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Thu, 17 Jul 2014 13:16:55 -0500 Subject: ASoC: Fix SOC_DOUBLE_R_SX_TLV volume mixer arguments Remove unnecessary bit shifts. Correct min value to match datasheet. Num steps = number of steps between min and max. Reported-by: Ryan Harvey Signed-off-by: Ryan Harvey Signed-off-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l52.c | 14 +++++++------- sound/soc/codecs/cs42l56.c | 12 ++++++------ sound/soc/codecs/cs42l73.c | 2 +- 3 files changed, 14 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 071fc77f2f06..969167d8b71e 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -399,15 +399,15 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_MASTERB_VOL, 0, 0x34, 0xE4, hl_tlv), SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L52_HPA_VOL, - CS42L52_HPB_VOL, 0, 0x34, 0xCC, hpd_tlv), + CS42L52_HPB_VOL, 0, 0x34, 0xC0, hpd_tlv), SOC_ENUM("Headphone Analog Gain", hp_gain_enum), SOC_DOUBLE_R_SX_TLV("Speaker Volume", CS42L52_SPKA_VOL, - CS42L52_SPKB_VOL, 0, 0x1, 0xff, hl_tlv), + CS42L52_SPKB_VOL, 0, 0x40, 0xC0, hl_tlv), SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL, - CS42L52_PASSTHRUB_VOL, 6, 0x18, 0x90, pga_tlv), + CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pga_tlv), SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0), @@ -417,10 +417,10 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { SOC_ENUM("MIC Bias Level", mic_bias_level_enum), SOC_DOUBLE_R_SX_TLV("ADC Volume", CS42L52_ADCA_VOL, - CS42L52_ADCB_VOL, 7, 0x80, 0xA0, ipd_tlv), + CS42L52_ADCB_VOL, 0, 0xA0, 0x78, ipd_tlv), SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L52_ADCA_MIXER_VOL, CS42L52_ADCB_MIXER_VOL, - 6, 0x7f, 0x19, ipd_tlv), + 0, 0x19, 0x7F, ipd_tlv), SOC_DOUBLE("ADC Switch", CS42L52_ADC_MISC_CTL, 0, 1, 1, 0), @@ -428,11 +428,11 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_ADCB_MIXER_VOL, 7, 1, 1), SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L52_PGAA_CTL, - CS42L52_PGAB_CTL, 0, 0x28, 0x30, pga_tlv), + CS42L52_PGAB_CTL, 0, 0x28, 0x24, pga_tlv), SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume", CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, - 0, 0x7f, 0x19, mix_tlv), + 0, 0x19, 0x7f, mix_tlv), SOC_DOUBLE_R("PCM Mixer Switch", CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, 7, 1, 1), diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 8e68ef5de849..24fbffee09ea 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -421,15 +421,15 @@ static const struct soc_enum ng_delay_enum = static const struct snd_kcontrol_new cs42l56_snd_controls[] = { SOC_DOUBLE_R_SX_TLV("Master Volume", CS42L56_MASTER_A_VOLUME, - CS42L56_MASTER_B_VOLUME, 0, 0x34, 0xfd, adv_tlv), + CS42L56_MASTER_B_VOLUME, 0, 0x34, 0xE4, adv_tlv), SOC_DOUBLE("Master Mute Switch", CS42L56_DSP_MUTE_CTL, 0, 1, 1, 1), SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L56_ADCA_MIX_VOLUME, - CS42L56_ADCB_MIX_VOLUME, 0, 0x88, 0xa9, hl_tlv), + CS42L56_ADCB_MIX_VOLUME, 0, 0x88, 0x90, hl_tlv), SOC_DOUBLE("ADC Mixer Mute Switch", CS42L56_DSP_MUTE_CTL, 6, 7, 1, 1), SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume", CS42L56_PCMA_MIX_VOLUME, - CS42L56_PCMB_MIX_VOLUME, 0, 0x88, 0xa9, hl_tlv), + CS42L56_PCMB_MIX_VOLUME, 0, 0x88, 0x90, hl_tlv), SOC_DOUBLE("PCM Mixer Mute Switch", CS42L56_DSP_MUTE_CTL, 4, 5, 1, 1), SOC_SINGLE_TLV("Analog Advisory Volume", @@ -438,16 +438,16 @@ static const struct snd_kcontrol_new cs42l56_snd_controls[] = { CS42L56_DIGINPUT_ADV_VOLUME, 0, 0x00, 1, adv_tlv), SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L56_PGAA_MUX_VOLUME, - CS42L56_PGAB_MUX_VOLUME, 0, 0x34, 0xfd, pga_tlv), + CS42L56_PGAB_MUX_VOLUME, 0, 0x34, 0x24, pga_tlv), SOC_DOUBLE_R_TLV("ADC Volume", CS42L56_ADCA_ATTENUATOR, CS42L56_ADCB_ATTENUATOR, 0, 0x00, 1, adc_tlv), SOC_DOUBLE("ADC Mute Switch", CS42L56_MISC_ADC_CTL, 2, 3, 1, 1), SOC_DOUBLE("ADC Boost Switch", CS42L56_GAIN_BIAS_CTL, 3, 2, 1, 1), SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L56_HPA_VOLUME, - CS42L56_HPB_VOLUME, 0, 0x44, 0x55, hl_tlv), + CS42L56_HPB_VOLUME, 0, 0x84, 0x48, hl_tlv), SOC_DOUBLE_R_SX_TLV("LineOut Volume", CS42L56_LOA_VOLUME, - CS42L56_LOB_VOLUME, 0, 0x44, 0x55, hl_tlv), + CS42L56_LOB_VOLUME, 0, 0x84, 0x48, hl_tlv), SOC_SINGLE_TLV("Bass Shelving Volume", CS42L56_TONE_CTL, 0, 0x00, 1, tone_tlv), diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index ae3717992d56..8658194f50bf 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -401,7 +401,7 @@ static const struct snd_kcontrol_new cs42l73_snd_controls[] = { CS42L73_LOBAVOL, 0, 0x41, 0x4B, hpaloa_tlv), SOC_DOUBLE_R_SX_TLV("Input PGA Analog Volume", CS42L73_MICAPREPGAAVOL, - CS42L73_MICBPREPGABVOL, 5, 0x34, + CS42L73_MICBPREPGABVOL, 0, 0x34, 0x24, micpga_tlv), SOC_DOUBLE_R("MIC Preamp Switch", CS42L73_MICAPREPGAAVOL, -- cgit v1.2.3 From e295a4a43e90d46d3b16374dc908533bb40a3936 Mon Sep 17 00:00:00 2001 From: Dan Murphy Date: Fri, 18 Jul 2014 12:31:07 -0500 Subject: ASoC: tas2552: Fix PM sequencing In the pm suspend/resume it is better to disable the GPIO after the regmap_cache setting calls so that if the call is interrupted the new reg values will be cached and set on resume. Also add pm_runtime_put in the remove call. Signed-off-by: Dan Murphy Signed-off-by: Mark Brown --- sound/soc/codecs/tas2552.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index f0760af5a21e..a3ae39448a48 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -239,12 +239,12 @@ static int tas2552_runtime_suspend(struct device *dev) tas2552_sw_shutdown(tas2552, 0); - if (tas2552->enable_gpio) - gpiod_set_value(tas2552->enable_gpio, 0); - regcache_cache_only(tas2552->regmap, true); regcache_mark_dirty(tas2552->regmap); + if (tas2552->enable_gpio) + gpiod_set_value(tas2552->enable_gpio, 0); + return 0; } @@ -382,6 +382,8 @@ static int tas2552_codec_remove(struct snd_soc_codec *codec) { struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); + pm_runtime_put(codec->dev); + if (tas2552->enable_gpio) gpiod_set_value(tas2552->enable_gpio, 0); -- cgit v1.2.3 From 9898e1ccf5af70ad2d03d5b77c591fb243c0f021 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 17 Jul 2014 22:01:05 +0200 Subject: ASoC: Remove per card platform list The platform_dev_list was added in commit f0fba2ad1b ("ASoC: multi-component - ASoC Multi-Component Support") and while platforms are added and remove from that list it is otherwise unused. This patch removes it again. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 3 --- sound/soc/soc-core.c | 2 -- 2 files changed, 5 deletions(-) (limited to 'sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index 98555f833ab4..1f5b4901415e 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -865,7 +865,6 @@ struct snd_soc_platform { struct snd_soc_card *card; struct list_head list; - struct list_head card_list; struct snd_soc_component component; @@ -1056,7 +1055,6 @@ struct snd_soc_card { /* lists of probed devices belonging to this card */ struct list_head codec_dev_list; - struct list_head platform_dev_list; struct list_head widgets; struct list_head paths; @@ -1299,7 +1297,6 @@ static inline void *snd_soc_pcm_get_drvdata(struct snd_soc_pcm_runtime *rtd) static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card) { INIT_LIST_HEAD(&card->codec_dev_list); - INIT_LIST_HEAD(&card->platform_dev_list); INIT_LIST_HEAD(&card->widgets); INIT_LIST_HEAD(&card->paths); INIT_LIST_HEAD(&card->dapm_list); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6e47610f73a0..a5edb31ddfff 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1027,7 +1027,6 @@ static int soc_remove_platform(struct snd_soc_platform *platform) soc_cleanup_platform_debugfs(platform); platform->probed = 0; - list_del(&platform->card_list); module_put(platform->dev->driver->owner); return 0; @@ -1297,7 +1296,6 @@ static int soc_probe_platform(struct snd_soc_card *card, /* mark platform as probed and add to card platform list */ platform->probed = 1; - list_add(&platform->card_list, &card->platform_dev_list); list_add(&platform->component.dapm.list, &card->dapm_list); return 0; -- cgit v1.2.3 From 093c4e5c3cc6dd320f6cfec54c5490d1567fb05c Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 17 Jul 2014 22:01:06 +0200 Subject: ASoC: tegra: Replace instances of rtd->codec->card with rtd->card No need to go via the CODEC to get a pointer to the card. This will help to eventually remove the card field from the snd_soc_codec struct. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_alc5632.c | 5 ++--- sound/soc/tegra/tegra_max98090.c | 5 ++--- sound/soc/tegra/tegra_rt5640.c | 5 ++--- sound/soc/tegra/tegra_wm8753.c | 3 +-- sound/soc/tegra/tegra_wm8903.c | 5 ++--- sound/soc/tegra/trimslice.c | 3 +-- 6 files changed, 10 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 02734bd4f09b..a83aff09dce2 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -41,8 +41,7 @@ static int tegra_alc5632_asoc_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = codec_dai->codec; - struct snd_soc_card *card = codec->card; + struct snd_soc_card *card = rtd->card; struct tegra_alc5632 *alc5632 = snd_soc_card_get_drvdata(card); int srate, mclk; int err; @@ -105,7 +104,7 @@ static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_codec *codec = codec_dai->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(codec->card); + struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(rtd->card); snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, &tegra_alc5632_hs_jack); diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c index ce73e1f62c4b..b86cd9936ef1 100644 --- a/sound/soc/tegra/tegra_max98090.c +++ b/sound/soc/tegra/tegra_max98090.c @@ -49,8 +49,7 @@ static int tegra_max98090_asoc_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = codec_dai->codec; - struct snd_soc_card *card = codec->card; + struct snd_soc_card *card = rtd->card; struct tegra_max98090 *machine = snd_soc_card_get_drvdata(card); int srate, mclk; int err; @@ -127,7 +126,7 @@ static int tegra_max98090_asoc_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_codec *codec = codec_dai->codec; - struct tegra_max98090 *machine = snd_soc_card_get_drvdata(codec->card); + struct tegra_max98090 *machine = snd_soc_card_get_drvdata(rtd->card); if (gpio_is_valid(machine->gpio_hp_det)) { snd_soc_jack_new(codec, "Headphones", SND_JACK_HEADPHONE, diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c index 4feb16a99e02..a6898831fb9f 100644 --- a/sound/soc/tegra/tegra_rt5640.c +++ b/sound/soc/tegra/tegra_rt5640.c @@ -51,8 +51,7 @@ static int tegra_rt5640_asoc_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = codec_dai->codec; - struct snd_soc_card *card = codec->card; + struct snd_soc_card *card = rtd->card; struct tegra_rt5640 *machine = snd_soc_card_get_drvdata(card); int srate, mclk; int err; @@ -110,7 +109,7 @@ static int tegra_rt5640_asoc_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_codec *codec = codec_dai->codec; - struct tegra_rt5640 *machine = snd_soc_card_get_drvdata(codec->card); + struct tegra_rt5640 *machine = snd_soc_card_get_drvdata(rtd->card); snd_soc_jack_new(codec, "Headphones", SND_JACK_HEADPHONE, &tegra_rt5640_hp_jack); diff --git a/sound/soc/tegra/tegra_wm8753.c b/sound/soc/tegra/tegra_wm8753.c index 8e774d1a243c..769e28f6642e 100644 --- a/sound/soc/tegra/tegra_wm8753.c +++ b/sound/soc/tegra/tegra_wm8753.c @@ -55,8 +55,7 @@ static int tegra_wm8753_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = codec_dai->codec; - struct snd_soc_card *card = codec->card; + struct snd_soc_card *card = rtd->card; struct tegra_wm8753 *machine = snd_soc_card_get_drvdata(card); int srate, mclk; int err; diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 0939661df60b..86e05e938585 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -60,8 +60,7 @@ static int tegra_wm8903_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = codec_dai->codec; - struct snd_soc_card *card = codec->card; + struct snd_soc_card *card = rtd->card; struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); int srate, mclk; int err; @@ -173,7 +172,7 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_codec *codec = codec_dai->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - struct snd_soc_card *card = codec->card; + struct snd_soc_card *card = rtd->card; struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); if (gpio_is_valid(machine->gpio_hp_det)) { diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c index 734bfcd21148..589d2d9b553a 100644 --- a/sound/soc/tegra/trimslice.c +++ b/sound/soc/tegra/trimslice.c @@ -50,8 +50,7 @@ static int trimslice_asoc_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = codec_dai->codec; - struct snd_soc_card *card = codec->card; + struct snd_soc_card *card = rtd->card; struct tegra_trimslice *trimslice = snd_soc_card_get_drvdata(card); int srate, mclk; int err; -- cgit v1.2.3 From 00200107a296cad3a950049a5ad7134a0d962b8d Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 17 Jul 2014 22:01:07 +0200 Subject: ASoC: Move card field form platform/codec to component Both the snd_soc_codec and snd_soc_platform struct do have a pointer to the parent card and both handle this pointer in mostly the same way. This patch moves the card field to the component level which will allow further code consolidation between platforms and CODECS. Since there are only a handful of users of the snd_soc_codec struct's card field (and none of the snd_soc_platform's) these are update in this patch as well, which allows it to be removed from the snd_soc_codec struct. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 3 +-- sound/soc/codecs/ac97.c | 4 ++-- sound/soc/codecs/cx20442.c | 6 +++--- sound/soc/codecs/uda134x.c | 2 +- sound/soc/codecs/wm8960.c | 2 +- sound/soc/codecs/wm_adsp.c | 4 ++-- sound/soc/omap/ams-delta.c | 2 +- sound/soc/soc-core.c | 19 ++++++++++--------- sound/soc/soc-dapm.c | 2 +- sound/soc/soc-jack.c | 4 ++-- 10 files changed, 24 insertions(+), 24 deletions(-) (limited to 'sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index 1f5b4901415e..5ee6ddde4831 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -691,6 +691,7 @@ struct snd_soc_component { int id; const char *name_prefix; struct device *dev; + struct snd_soc_card *card; unsigned int active; @@ -725,7 +726,6 @@ struct snd_soc_codec { const struct snd_soc_codec_driver *driver; struct mutex mutex; - struct snd_soc_card *card; struct list_head list; struct list_head card_list; @@ -863,7 +863,6 @@ struct snd_soc_platform { unsigned int suspended:1; /* platform is suspended */ unsigned int probed:1; - struct snd_soc_card *card; struct list_head list; struct snd_soc_component component; diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 8d9ba4ba4bfe..e889e1b84192 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -89,8 +89,8 @@ static int ac97_soc_probe(struct snd_soc_codec *codec) int ret; /* add codec as bus device for standard ac97 */ - ret = snd_ac97_bus(codec->card->snd_card, 0, soc_ac97_ops, NULL, - &ac97_bus); + ret = snd_ac97_bus(codec->component.card->snd_card, 0, soc_ac97_ops, + NULL, &ac97_bus); if (ret < 0) return ret; diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index d5fd00a64748..4ba60eb2cde1 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -253,7 +253,7 @@ static void v253_close(struct tty_struct *tty) /* Prevent the codec driver from further accessing the modem */ codec->hw_write = NULL; cx20442->control_data = NULL; - codec->card->pop_time = 0; + codec->component.card->pop_time = 0; } /* Line discipline .hangup() */ @@ -281,7 +281,7 @@ static void v253_receive(struct tty_struct *tty, /* Set up codec driver access to modem controls */ cx20442->control_data = tty; codec->hw_write = (hw_write_t)tty->ops->write; - codec->card->pop_time = 1; + codec->component.card->pop_time = 1; } } @@ -372,7 +372,7 @@ static int cx20442_codec_probe(struct snd_soc_codec *codec) snd_soc_codec_set_drvdata(codec, cx20442); codec->hw_write = NULL; - codec->card->pop_time = 0; + codec->component.card->pop_time = 0; return 0; } diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index edf27acc1d77..12fc0aed7503 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -479,7 +479,7 @@ static struct snd_soc_dai_driver uda134x_dai = { static int uda134x_soc_probe(struct snd_soc_codec *codec) { struct uda134x_priv *uda134x; - struct uda134x_platform_data *pd = codec->card->dev->platform_data; + struct uda134x_platform_data *pd = codec->component.card->dev->platform_data; const struct snd_soc_dapm_widget *widgets; unsigned num_widgets; diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index a145d0431b63..e96349b04ba6 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -472,7 +472,7 @@ static int wm8960_add_widgets(struct snd_soc_codec *codec) * list each time to find the desired power state do so now * and save the result. */ - list_for_each_entry(w, &codec->card->widgets, list) { + list_for_each_entry(w, &codec->component.card->widgets, list) { if (w->dapm != &codec->dapm) continue; if (strcmp(w->name, "LOUT1 PGA") == 0) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 060027182dcb..fb86f072fa67 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1382,7 +1382,7 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, int ret; int val; - dsp->card = codec->card; + dsp->card = codec->component.card; switch (event) { case SND_SOC_DAPM_POST_PMU: @@ -1617,7 +1617,7 @@ int wm_adsp2_early_event(struct snd_soc_dapm_widget *w, struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); struct wm_adsp *dsp = &dsps[w->shift]; - dsp->card = codec->card; + dsp->card = codec->component.card; switch (event) { case SND_SOC_DAPM_PRE_PMU: diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 0cc41f94de4e..8c9cc64a9dfb 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -301,7 +301,7 @@ static int cx81801_open(struct tty_struct *tty) static void cx81801_close(struct tty_struct *tty) { struct snd_soc_codec *codec = tty->disc_data; - struct snd_soc_dapm_context *dapm = &codec->card->dapm; + struct snd_soc_dapm_context *dapm = &codec->component.card->dapm; del_timer_sync(&cx81801_timer); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a5edb31ddfff..a0ae3d2263d1 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -292,7 +292,7 @@ static struct dentry *soc_debugfs_create_dir(struct dentry *parent, static void soc_init_codec_debugfs(struct snd_soc_codec *codec) { - struct dentry *debugfs_card_root = codec->card->debugfs_card_root; + struct dentry *debugfs_card_root = codec->component.card->debugfs_card_root; codec->debugfs_codec_root = soc_debugfs_create_dir(debugfs_card_root, "codec:%s", @@ -325,7 +325,7 @@ static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec) static void soc_init_platform_debugfs(struct snd_soc_platform *platform) { - struct dentry *debugfs_card_root = platform->card->debugfs_card_root; + struct dentry *debugfs_card_root = platform->component.card->debugfs_card_root; platform->debugfs_platform_root = soc_debugfs_create_dir(debugfs_card_root, "platform:%s", @@ -546,11 +546,12 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec) int err; codec->ac97->dev.bus = &ac97_bus_type; - codec->ac97->dev.parent = codec->card->dev; + codec->ac97->dev.parent = codec->component.card->dev; codec->ac97->dev.release = soc_ac97_device_release; dev_set_name(&codec->ac97->dev, "%d-%d:%s", - codec->card->snd_card->number, 0, codec->component.name); + codec->component.card->snd_card->number, 0, + codec->component.name); err = device_register(&codec->ac97->dev); if (err < 0) { dev_err(codec->dev, "ASoC: Can't register ac97 bus\n"); @@ -1179,7 +1180,7 @@ static int soc_probe_codec(struct snd_soc_card *card, const struct snd_soc_codec_driver *driver = codec->driver; struct snd_soc_dai *dai; - codec->card = card; + codec->component.card = card; codec->dapm.card = card; soc_set_name_prefix(card, &codec->component); @@ -1255,7 +1256,7 @@ static int soc_probe_platform(struct snd_soc_card *card, struct snd_soc_component *component; struct snd_soc_dai *dai; - platform->card = card; + platform->component.card = card; platform->component.dapm.card = card; if (!try_module_get(platform->dev->driver->owner)) @@ -2406,7 +2407,7 @@ EXPORT_SYMBOL_GPL(snd_soc_card_get_kcontrol); int snd_soc_add_codec_controls(struct snd_soc_codec *codec, const struct snd_kcontrol_new *controls, int num_controls) { - struct snd_card *card = codec->card->snd_card; + struct snd_card *card = codec->component.card->snd_card; return snd_soc_add_controls(card, codec->dev, controls, num_controls, codec->component.name_prefix, &codec->component); @@ -2426,7 +2427,7 @@ EXPORT_SYMBOL_GPL(snd_soc_add_codec_controls); int snd_soc_add_platform_controls(struct snd_soc_platform *platform, const struct snd_kcontrol_new *controls, int num_controls) { - struct snd_card *card = platform->card->snd_card; + struct snd_card *card = platform->component.card->snd_card; return snd_soc_add_controls(card, platform->dev, controls, num_controls, NULL, &platform->component); @@ -3101,7 +3102,7 @@ EXPORT_SYMBOL_GPL(snd_soc_get_volsw_range); int snd_soc_limit_volume(struct snd_soc_codec *codec, const char *name, int max) { - struct snd_card *card = codec->card->snd_card; + struct snd_card *card = codec->component.card->snd_card; struct snd_kcontrol *kctl; struct soc_mixer_control *mc; int found = 0; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8cb68a38ad19..8348352dc2c6 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2090,7 +2090,7 @@ static ssize_t dapm_widget_show_codec(struct snd_soc_codec *codec, char *buf) int count = 0; char *state = "not set"; - list_for_each_entry(w, &codec->card->widgets, list) { + list_for_each_entry(w, &codec->component.card->widgets, list) { if (w->dapm != &codec->dapm) continue; diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index d0d98810af91..ab47fea997a3 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -43,7 +43,7 @@ int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, int type, INIT_LIST_HEAD(&jack->jack_zones); BLOCKING_INIT_NOTIFIER_HEAD(&jack->notifier); - return snd_jack_new(codec->card->snd_card, id, type, &jack->jack); + return snd_jack_new(codec->component.card->snd_card, id, type, &jack->jack); } EXPORT_SYMBOL_GPL(snd_soc_jack_new); @@ -260,7 +260,7 @@ static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) static irqreturn_t gpio_handler(int irq, void *data) { struct snd_soc_jack_gpio *gpio = data; - struct device *dev = gpio->jack->codec->card->dev; + struct device *dev = gpio->jack->codec->component.card->dev; trace_snd_soc_jack_irq(gpio->name); -- cgit v1.2.3 From 0f2780ad4c2a398528c7bb1572158d6e894e5dd2 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 17 Jul 2014 22:01:08 +0200 Subject: ASoC: Add function to register component controls We have now everything in place to actual let a component register controls. Add a function which allows to do so. Also update snd_soc_add_codec_controls() and snd_soc_platform_controls() to use this new function internally. And while we are at it also change the num_controls parameter of those two functions from int to unsigned int. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 6 ++++-- sound/soc/soc-core.c | 35 +++++++++++++++++++++++++---------- 2 files changed, 29 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index 5ee6ddde4831..50ff6f3459aa 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -507,10 +507,12 @@ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, const char *prefix); struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card, const char *name); +int snd_soc_add_component_controls(struct snd_soc_component *component, + const struct snd_kcontrol_new *controls, unsigned int num_controls); int snd_soc_add_codec_controls(struct snd_soc_codec *codec, - const struct snd_kcontrol_new *controls, int num_controls); + const struct snd_kcontrol_new *controls, unsigned int num_controls); int snd_soc_add_platform_controls(struct snd_soc_platform *platform, - const struct snd_kcontrol_new *controls, int num_controls); + const struct snd_kcontrol_new *controls, unsigned int num_controls); int snd_soc_add_card_controls(struct snd_soc_card *soc_card, const struct snd_kcontrol_new *controls, int num_controls); int snd_soc_add_dai_controls(struct snd_soc_dai *dai, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a0ae3d2263d1..28caa63ae526 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2393,6 +2393,25 @@ struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card, } EXPORT_SYMBOL_GPL(snd_soc_card_get_kcontrol); +/** + * snd_soc_add_component_controls - Add an array of controls to a component. + * + * @component: Component to add controls to + * @controls: Array of controls to add + * @num_controls: Number of elements in the array + * + * Return: 0 for success, else error. + */ +int snd_soc_add_component_controls(struct snd_soc_component *component, + const struct snd_kcontrol_new *controls, unsigned int num_controls) +{ + struct snd_card *card = component->card->snd_card; + + return snd_soc_add_controls(card, component->dev, controls, + num_controls, component->name_prefix, component); +} +EXPORT_SYMBOL_GPL(snd_soc_add_component_controls); + /** * snd_soc_add_codec_controls - add an array of controls to a codec. * Convenience function to add a list of controls. Many codecs were @@ -2405,12 +2424,10 @@ EXPORT_SYMBOL_GPL(snd_soc_card_get_kcontrol); * Return 0 for success, else error. */ int snd_soc_add_codec_controls(struct snd_soc_codec *codec, - const struct snd_kcontrol_new *controls, int num_controls) + const struct snd_kcontrol_new *controls, unsigned int num_controls) { - struct snd_card *card = codec->component.card->snd_card; - - return snd_soc_add_controls(card, codec->dev, controls, num_controls, - codec->component.name_prefix, &codec->component); + return snd_soc_add_component_controls(&codec->component, controls, + num_controls); } EXPORT_SYMBOL_GPL(snd_soc_add_codec_controls); @@ -2425,12 +2442,10 @@ EXPORT_SYMBOL_GPL(snd_soc_add_codec_controls); * Return 0 for success, else error. */ int snd_soc_add_platform_controls(struct snd_soc_platform *platform, - const struct snd_kcontrol_new *controls, int num_controls) + const struct snd_kcontrol_new *controls, unsigned int num_controls) { - struct snd_card *card = platform->component.card->snd_card; - - return snd_soc_add_controls(card, platform->dev, controls, num_controls, - NULL, &platform->component); + return snd_soc_add_component_controls(&platform->component, controls, + num_controls); } EXPORT_SYMBOL_GPL(snd_soc_add_platform_controls); -- cgit v1.2.3 From 2bf865ba30405056e7922b960a1687bb4d5a4999 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 16 Jul 2014 23:18:44 -0700 Subject: ASoC: rsnd: add missing src/dst_addr_width for DMAEngine Renesas new DMAEngine driver requests src/dst_addr_width Signed-off-by: Kuninori Morimoto Acked-by: Laurent Pinchart Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 907d4802fd5c..c48d999a3fce 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -338,6 +338,8 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, cfg.direction = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; cfg.src_addr = rsnd_gen_dma_addr(priv, mod_from, is_play, 1); cfg.dst_addr = rsnd_gen_dma_addr(priv, mod_to, is_play, 0); + cfg.src_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + cfg.dst_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; dev_dbg(dev, "dma : %s %pad -> %pad\n", dma_name, &cfg.src_addr, &cfg.dst_addr); -- cgit v1.2.3 From 3d02cc748fad80f6900283d3497562f5f8a94069 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Tue, 22 Jul 2014 11:50:22 +0100 Subject: ASoC: wm5110: Set symmetric samplebits on AIFs Different playback and capture bits-per-sample are not supported on the AIFs Signed-off-by: Richard Fitzgerald Signed-off-by: Mark Brown --- sound/soc/codecs/wm5110.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 62ef54456499..2f2ec26d831c 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -1485,6 +1485,7 @@ static struct snd_soc_dai_driver wm5110_dai[] = { }, .ops = &arizona_dai_ops, .symmetric_rates = 1, + .symmetric_samplebits = 1, }, { .name = "wm5110-aif2", @@ -1506,6 +1507,7 @@ static struct snd_soc_dai_driver wm5110_dai[] = { }, .ops = &arizona_dai_ops, .symmetric_rates = 1, + .symmetric_samplebits = 1, }, { .name = "wm5110-aif3", @@ -1527,6 +1529,7 @@ static struct snd_soc_dai_driver wm5110_dai[] = { }, .ops = &arizona_dai_ops, .symmetric_rates = 1, + .symmetric_samplebits = 1, }, { .name = "wm5110-slim1", -- cgit v1.2.3 From 11ecf2b9ace74f0a4737076af6f39425c99339f1 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Tue, 22 Jul 2014 11:51:57 +0100 Subject: ASoC: wm5102: Set symmetric samplebits on AIFs Different playback and capture bits-per-sample are not supported on the AIFs Signed-off-by: Richard Fitzgerald Signed-off-by: Mark Brown --- sound/soc/codecs/wm5102.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 289b64d89abd..2e96d6e0e13e 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1653,6 +1653,7 @@ static struct snd_soc_dai_driver wm5102_dai[] = { }, .ops = &arizona_dai_ops, .symmetric_rates = 1, + .symmetric_samplebits = 1, }, { .name = "wm5102-aif2", @@ -1674,6 +1675,7 @@ static struct snd_soc_dai_driver wm5102_dai[] = { }, .ops = &arizona_dai_ops, .symmetric_rates = 1, + .symmetric_samplebits = 1, }, { .name = "wm5102-aif3", @@ -1695,6 +1697,7 @@ static struct snd_soc_dai_driver wm5102_dai[] = { }, .ops = &arizona_dai_ops, .symmetric_rates = 1, + .symmetric_samplebits = 1, }, { .name = "wm5102-slim1", -- cgit v1.2.3 From 8a349451ea2aad521e028b139fdfe6779e73851c Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Tue, 22 Jul 2014 11:52:56 +0100 Subject: ASoC: wm8997: Set symmetric samplebits on AIFs Different playback and capture bits-per-sample are not supported on the AIFs Signed-off-by: Richard Fitzgerald Signed-off-by: Mark Brown --- sound/soc/codecs/wm8997.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index bb9b47b956aa..ab33fe596519 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -967,6 +967,7 @@ static struct snd_soc_dai_driver wm8997_dai[] = { }, .ops = &arizona_dai_ops, .symmetric_rates = 1, + .symmetric_samplebits = 1, }, { .name = "wm8997-aif2", @@ -988,6 +989,7 @@ static struct snd_soc_dai_driver wm8997_dai[] = { }, .ops = &arizona_dai_ops, .symmetric_rates = 1, + .symmetric_samplebits = 1, }, { .name = "wm8997-slim1", -- cgit v1.2.3 From bedd4b1993ecb7228f8e5d6673e25d4482bd302b Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Tue, 22 Jul 2014 11:42:06 +0100 Subject: ASoC: arizona: Disable AIF TX/RX before configuring it Changes to the AIF configuration registers only take effect when the AIF is disabled. If the configuration is being changed from the previous setup, temporarily disable the AIF. Signed-off-by: Dimitris Papastamos Signed-off-by: Richard Fitzgerald Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 87 ++++++++++++++++++++++++++++++++++++---------- 1 file changed, 69 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index be3657ac52a4..9a730689a0ac 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1209,6 +1209,27 @@ static int arizona_hw_params_rate(struct snd_pcm_substream *substream, return 0; } +static bool arizona_aif_cfg_changed(struct snd_soc_codec *codec, + int base, int bclk, int lrclk, int frame) +{ + int val; + + val = snd_soc_read(codec, base + ARIZONA_AIF_BCLK_CTRL); + if (bclk != (val & ARIZONA_AIF1_BCLK_FREQ_MASK)) + return true; + + val = snd_soc_read(codec, base + ARIZONA_AIF_TX_BCLK_RATE); + if (lrclk != (val & ARIZONA_AIF1TX_BCPF_MASK)) + return true; + + val = snd_soc_read(codec, base + ARIZONA_AIF_FRAME_CTRL_1); + if (frame != (val & (ARIZONA_AIF1TX_WL_MASK | + ARIZONA_AIF1TX_SLOT_LEN_MASK))) + return true; + + return false; +} + static int arizona_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -1224,6 +1245,8 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, int tdm_width = arizona->tdm_width[dai->id - 1]; int tdm_slots = arizona->tdm_slots[dai->id - 1]; int bclk, lrclk, wl, frame, bclk_target; + bool reconfig; + unsigned int aif_tx_state, aif_rx_state; if (params_rate(params) % 8000) rates = &arizona_44k1_bclk_rates[0]; @@ -1274,28 +1297,56 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, wl = snd_pcm_format_width(params_format(params)); frame = wl << ARIZONA_AIF1TX_WL_SHIFT | wl; + reconfig = arizona_aif_cfg_changed(codec, base, bclk, lrclk, frame); + + if (reconfig) { + /* Save AIF TX/RX state */ + aif_tx_state = snd_soc_read(codec, + base + ARIZONA_AIF_TX_ENABLES); + aif_rx_state = snd_soc_read(codec, + base + ARIZONA_AIF_RX_ENABLES); + /* Disable AIF TX/RX before reconfiguring it */ + regmap_update_bits_async(arizona->regmap, + base + ARIZONA_AIF_TX_ENABLES, 0xff, 0x0); + regmap_update_bits(arizona->regmap, + base + ARIZONA_AIF_RX_ENABLES, 0xff, 0x0); + } + ret = arizona_hw_params_rate(substream, params, dai); if (ret != 0) - return ret; + goto restore_aif; - regmap_update_bits_async(arizona->regmap, - base + ARIZONA_AIF_BCLK_CTRL, - ARIZONA_AIF1_BCLK_FREQ_MASK, bclk); - regmap_update_bits_async(arizona->regmap, - base + ARIZONA_AIF_TX_BCLK_RATE, - ARIZONA_AIF1TX_BCPF_MASK, lrclk); - regmap_update_bits_async(arizona->regmap, - base + ARIZONA_AIF_RX_BCLK_RATE, - ARIZONA_AIF1RX_BCPF_MASK, lrclk); - regmap_update_bits_async(arizona->regmap, - base + ARIZONA_AIF_FRAME_CTRL_1, - ARIZONA_AIF1TX_WL_MASK | - ARIZONA_AIF1TX_SLOT_LEN_MASK, frame); - regmap_update_bits(arizona->regmap, base + ARIZONA_AIF_FRAME_CTRL_2, - ARIZONA_AIF1RX_WL_MASK | - ARIZONA_AIF1RX_SLOT_LEN_MASK, frame); + if (reconfig) { + regmap_update_bits_async(arizona->regmap, + base + ARIZONA_AIF_BCLK_CTRL, + ARIZONA_AIF1_BCLK_FREQ_MASK, bclk); + regmap_update_bits_async(arizona->regmap, + base + ARIZONA_AIF_TX_BCLK_RATE, + ARIZONA_AIF1TX_BCPF_MASK, lrclk); + regmap_update_bits_async(arizona->regmap, + base + ARIZONA_AIF_RX_BCLK_RATE, + ARIZONA_AIF1RX_BCPF_MASK, lrclk); + regmap_update_bits_async(arizona->regmap, + base + ARIZONA_AIF_FRAME_CTRL_1, + ARIZONA_AIF1TX_WL_MASK | + ARIZONA_AIF1TX_SLOT_LEN_MASK, frame); + regmap_update_bits(arizona->regmap, + base + ARIZONA_AIF_FRAME_CTRL_2, + ARIZONA_AIF1RX_WL_MASK | + ARIZONA_AIF1RX_SLOT_LEN_MASK, frame); + } - return 0; +restore_aif: + if (reconfig) { + /* Restore AIF TX/RX state */ + regmap_update_bits_async(arizona->regmap, + base + ARIZONA_AIF_TX_ENABLES, + 0xff, aif_tx_state); + regmap_update_bits(arizona->regmap, + base + ARIZONA_AIF_RX_ENABLES, + 0xff, aif_rx_state); + } + return ret; } static const char *arizona_dai_clk_str(int clk_id) -- cgit v1.2.3 From a9ef83f287b63ec8a753a3a7c6eac124f708544b Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 25 Jul 2014 15:07:47 +0300 Subject: ASoC: tlv320aic31xx: Add dependency for I2C and select REGMAP_I2C in Kconfig The codec can be configured via I2C and using regmap_i2c. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index cbfa1e18f651..9115ab83c656 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -534,6 +534,8 @@ config SND_SOC_TLV320AIC26 config SND_SOC_TLV320AIC31XX tristate + depends on I2C + select REGMAP_I2C config SND_SOC_TLV320AIC32X4 tristate -- cgit v1.2.3 From 96665e312940a0915e74a98cf75938a57d12c723 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 25 Jul 2014 15:07:48 +0300 Subject: ASoC: tlv320aic31xx: Add text in Kconfig so the codec can be used with simple-card In order to show the option under 'CODEC drivers' we need to have text in Kconfig. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9115ab83c656..0ad292f41913 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -533,7 +533,7 @@ config SND_SOC_TLV320AIC26 depends on SPI config SND_SOC_TLV320AIC31XX - tristate + tristate "Texas Instruments TLV320AIC31xx CODECs" depends on I2C select REGMAP_I2C -- cgit v1.2.3 From d62a3dcd4d75b1713d12697afdbffaf9a9da8f43 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 24 Jul 2014 01:51:31 -0700 Subject: ASoC: rsnd: fixup dai remove callback operation rsnd driver is using SSI/SRC/DVC which are using "mod" base operation. These "mod" are supporting "probe" and "remove" callbacks. Current rsnd_probe should call "remove" if "probe" was failed, since "probe" might be having DMAEngine handle. Some mod's "remove" callback might be called without calling "probe", but it is no problem. because "remove" do nothing in such case. So, all mod's "remove" should be called when error case of rsnd_probe() and rsnd_remove(). Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index c48d999a3fce..f07742f1eb11 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1043,11 +1043,11 @@ static int rsnd_probe(struct platform_device *pdev) for_each_rsnd_dai(rdai, priv, i) { ret = rsnd_dai_call(probe, &rdai->playback, rdai); if (ret) - return ret; + goto exit_snd_probe; ret = rsnd_dai_call(probe, &rdai->capture, rdai); if (ret) - return ret; + goto exit_snd_probe; } /* @@ -1075,6 +1075,11 @@ static int rsnd_probe(struct platform_device *pdev) exit_snd_soc: snd_soc_unregister_platform(dev); +exit_snd_probe: + for_each_rsnd_dai(rdai, priv, i) { + rsnd_dai_call(remove, &rdai->playback, rdai); + rsnd_dai_call(remove, &rdai->capture, rdai); + } return ret; } @@ -1083,21 +1088,16 @@ static int rsnd_remove(struct platform_device *pdev) { struct rsnd_priv *priv = dev_get_drvdata(&pdev->dev); struct rsnd_dai *rdai; - int ret, i; + int ret = 0, i; pm_runtime_disable(&pdev->dev); for_each_rsnd_dai(rdai, priv, i) { - ret = rsnd_dai_call(remove, &rdai->playback, rdai); - if (ret) - return ret; - - ret = rsnd_dai_call(remove, &rdai->capture, rdai); - if (ret) - return ret; + ret |= rsnd_dai_call(remove, &rdai->playback, rdai); + ret |= rsnd_dai_call(remove, &rdai->capture, rdai); } - return 0; + return ret; } static struct platform_driver rsnd_driver = { -- cgit v1.2.3 From f4075a8f452aff5465c6522c92da9db71ed11b7f Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Wed, 23 Jul 2014 19:23:38 +0800 Subject: ASoC: fsl_sai: Reduce race condition during TE/RE enabling For trigger start, we don't need to check if it's the first time to enable TE/RE or second time. It doesn't hurt to enable them any way, which in the meantime can reduce race condition for TE/RE enabling. For trigger stop, we will definitely clear FRDE of current direction. Thus the driver only needs to read the opposite one's. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 18 +++++++----------- 1 file changed, 7 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 1b6ee2ce849f..a437899d2d11 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -327,7 +327,7 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; - u32 tcsr, rcsr; + u32 xcsr; /* * The transmitter bit clock and frame sync are to be @@ -338,9 +338,6 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, regmap_update_bits(sai->regmap, FSL_SAI_RCR2, FSL_SAI_CR2_SYNC, FSL_SAI_CR2_SYNC); - regmap_read(sai->regmap, FSL_SAI_TCSR, &tcsr); - regmap_read(sai->regmap, FSL_SAI_RCSR, &rcsr); - /* * It is recommended that the transmitter is the last enabled * and the first disabled. @@ -349,12 +346,10 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (!(tcsr & FSL_SAI_CSR_FRDE || rcsr & FSL_SAI_CSR_FRDE)) { - regmap_update_bits(sai->regmap, FSL_SAI_RCSR, - FSL_SAI_CSR_TERE, FSL_SAI_CSR_TERE); - regmap_update_bits(sai->regmap, FSL_SAI_TCSR, - FSL_SAI_CSR_TERE, FSL_SAI_CSR_TERE); - } + regmap_update_bits(sai->regmap, FSL_SAI_RCSR, + FSL_SAI_CSR_TERE, FSL_SAI_CSR_TERE); + regmap_update_bits(sai->regmap, FSL_SAI_TCSR, + FSL_SAI_CSR_TERE, FSL_SAI_CSR_TERE); regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), FSL_SAI_CSR_xIE_MASK, FSL_SAI_FLAGS); @@ -370,7 +365,8 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, FSL_SAI_CSR_xIE_MASK, 0); /* Check if the opposite FRDE is also disabled */ - if (!(tx ? rcsr & FSL_SAI_CSR_FRDE : tcsr & FSL_SAI_CSR_FRDE)) { + regmap_read(sai->regmap, FSL_SAI_xCSR(!tx), &xcsr); + if (!(xcsr & FSL_SAI_CSR_FRDE)) { /* Disable both directions and reset their FIFOs */ regmap_update_bits(sai->regmap, FSL_SAI_TCSR, FSL_SAI_CSR_TERE | FSL_SAI_CSR_FR, -- cgit v1.2.3 From c44b56af9ca3a6f135d8f22b9a240f53909b371e Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Wed, 23 Jul 2014 19:23:39 +0800 Subject: ASoC: fsl_sai: Don't reset FIFO until TE/RE bit is unset TE/RE bit of T/RCSR will remain set untill the current frame is physically finished. The FIFO reset operation should wait this bit's totally cleared rather than ignoring its status which might cause TE/RE disabling failed. This patch adds delay and timeout to wait for its completion before FIFO reset. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 19 ++++++++++++++----- 1 file changed, 14 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index a437899d2d11..a79a9b0c08fd 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -327,7 +327,7 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; - u32 xcsr; + u32 xcsr, count = 100; /* * The transmitter bit clock and frame sync are to be @@ -369,11 +369,20 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, if (!(xcsr & FSL_SAI_CSR_FRDE)) { /* Disable both directions and reset their FIFOs */ regmap_update_bits(sai->regmap, FSL_SAI_TCSR, - FSL_SAI_CSR_TERE | FSL_SAI_CSR_FR, - FSL_SAI_CSR_FR); + FSL_SAI_CSR_TERE, 0); regmap_update_bits(sai->regmap, FSL_SAI_RCSR, - FSL_SAI_CSR_TERE | FSL_SAI_CSR_FR, - FSL_SAI_CSR_FR); + FSL_SAI_CSR_TERE, 0); + + /* TERE will remain set till the end of current frame */ + do { + udelay(10); + regmap_read(sai->regmap, FSL_SAI_xCSR(tx), &xcsr); + } while (--count && xcsr & FSL_SAI_CSR_TERE); + + regmap_update_bits(sai->regmap, FSL_SAI_TCSR, + FSL_SAI_CSR_FR, FSL_SAI_CSR_FR); + regmap_update_bits(sai->regmap, FSL_SAI_RCSR, + FSL_SAI_CSR_FR, FSL_SAI_CSR_FR); } break; default: -- cgit v1.2.3 From a3fdc6749edf4dcb07df3a10bbdd9850ed5fd01a Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Wed, 23 Jul 2014 19:23:40 +0800 Subject: ASoC: fsl_sai: Improve enable flow in fsl_sai_trigger() The previous enable flow: 1, Enable TE&RE (SAI starts to consume tx FIFO and feed rx FIFO) 2, Mask IRQ of Tx/Rx to enable its interrupt. 3, Enable DMA request of Tx/Rx. As this flow would enable DMA request later than TERE, the Tx FIFO would be easily emptied into underrun while Rx FIFO would be easily stuffed into overrun due to the delayed DMA transfering. This issue happened merely occational before the patch 'ASoC: fsl_sai: Reset FIFOs after disabling TE/RE' because there were useless data remaining in the FIFO for the gap. However, it manifested after FIFO reset's implemented. After this patch, the new flow: 1, Enable DMA request of Tx/Rx. 2, Enable TE&RE (SAI starts to consume tx FIFO and feed rx FIFO) 3, Mask IRQ of Tx/Rx to enable its interrupt. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index a79a9b0c08fd..364410be3c4e 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -346,6 +346,9 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), + FSL_SAI_CSR_FRDE, FSL_SAI_CSR_FRDE); + regmap_update_bits(sai->regmap, FSL_SAI_RCSR, FSL_SAI_CSR_TERE, FSL_SAI_CSR_TERE); regmap_update_bits(sai->regmap, FSL_SAI_TCSR, @@ -353,8 +356,6 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), FSL_SAI_CSR_xIE_MASK, FSL_SAI_FLAGS); - regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), - FSL_SAI_CSR_FRDE, FSL_SAI_CSR_FRDE); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: -- cgit v1.2.3 From c62f9d8f7fcefe87e362b75c1c6a4333fc1019ac Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 23 Jul 2014 16:42:21 +0800 Subject: ASoC: tas2552: Return proper error for probe error paths Return error if devm_regulator_bulk_get() or snd_soc_register_codec() fails. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/tas2552.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index a3ae39448a48..23b32960ff1d 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -482,8 +482,10 @@ static int tas2552_probe(struct i2c_client *client, ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(data->supplies), data->supplies); - if (ret != 0) + if (ret != 0) { dev_err(dev, "Failed to request supplies: %d\n", ret); + return ret; + } pm_runtime_set_active(&client->dev); pm_runtime_set_autosuspend_delay(&client->dev, 1000); @@ -500,7 +502,7 @@ static int tas2552_probe(struct i2c_client *client, if (ret < 0) dev_err(&client->dev, "Failed to register codec: %d\n", ret); - return 0; + return ret; } static int tas2552_i2c_remove(struct i2c_client *client) -- cgit v1.2.3 From 9779caf9faec1a366deb0b5877bfbc551917f03b Mon Sep 17 00:00:00 2001 From: Rongjun Ying Date: Wed, 23 Jul 2014 13:19:32 +0800 Subject: ASoC: sirf-usp: Fixed a bug for playback and capture work at the same time 1. The startup function invoked when the playback and capture. If start playback when capturing, the registers are re-initinitialised. That cause the playback fail. So move the startup code into runtime resume. 2. Modified: If non RUNTIME_PM support, the probe need enable clock and initinitialise registers. 3. Refine code. Signed-off-by: Rongjun Ying Signed-off-by: Mark Brown --- sound/soc/sirf/sirf-usp.c | 54 +++++++++++++++++++++++------------------------ 1 file changed, 27 insertions(+), 27 deletions(-) (limited to 'sound') diff --git a/sound/soc/sirf/sirf-usp.c b/sound/soc/sirf/sirf-usp.c index 9693bc2a796d..3a730374e259 100644 --- a/sound/soc/sirf/sirf-usp.c +++ b/sound/soc/sirf/sirf-usp.c @@ -103,11 +103,8 @@ static int sirf_usp_pcm_set_dai_fmt(struct snd_soc_dai *dai, return 0; } -static int sirf_usp_i2s_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) +static void sirf_usp_i2s_init(struct sirf_usp *usp) { - struct sirf_usp *usp = snd_soc_dai_get_drvdata(dai); - /* Configure RISC mode */ regmap_update_bits(usp->regmap, USP_RISC_DSP_MODE, USP_RISC_DSP_SEL, ~USP_RISC_DSP_SEL); @@ -119,19 +116,16 @@ static int sirf_usp_i2s_startup(struct snd_pcm_substream *substream, regmap_write(usp->regmap, USP_TX_DMA_IO_LEN, 0); regmap_write(usp->regmap, USP_RX_DMA_IO_LEN, 0); - regmap_write(usp->regmap, USP_RX_FRAME_CTRL, USP_SINGLE_SYNC_MODE); - - regmap_write(usp->regmap, USP_TX_FRAME_CTRL, USP_TXC_SLAVE_CLK_SAMPLE); - /* Configure Mode2 register */ regmap_write(usp->regmap, USP_MODE2, (1 << USP_RXD_DELAY_LEN_OFFSET) | - (0 << USP_TXD_DELAY_LEN_OFFSET)); + (0 << USP_TXD_DELAY_LEN_OFFSET) | + USP_TFS_CLK_SLAVE_MODE | USP_RFS_CLK_SLAVE_MODE); /* Configure Mode1 register */ regmap_write(usp->regmap, USP_MODE1, USP_SYNC_MODE | USP_EN | USP_TXD_ACT_EDGE_FALLING | USP_RFS_ACT_LEVEL_LOGIC1 | USP_TFS_ACT_LEVEL_LOGIC1 | - USP_TX_UFLOW_REPEAT_ZERO); + USP_TX_UFLOW_REPEAT_ZERO | USP_CLOCK_MODE_SLAVE); /* Configure RX DMA IO Control register */ regmap_write(usp->regmap, USP_RX_DMA_IO_CTRL, 0); @@ -155,8 +149,6 @@ static int sirf_usp_i2s_startup(struct snd_pcm_substream *substream, /* Congiure TX FIFO Level Check register */ regmap_write(usp->regmap, USP_TX_FIFO_LEVEL_CHK, TX_FIFO_SC(0x1B) | TX_FIFO_LC(0x0E) | TX_FIFO_HC(0x04)); - - return 0; } static int sirf_usp_pcm_hw_params(struct snd_pcm_substream *substream, @@ -204,23 +196,19 @@ static int sirf_usp_pcm_hw_params(struct snd_pcm_substream *substream, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) regmap_update_bits(usp->regmap, USP_TX_FRAME_CTRL, USP_TXC_DATA_LEN_MASK | USP_TXC_FRAME_LEN_MASK - | USP_TXC_SHIFTER_LEN_MASK, + | USP_TXC_SHIFTER_LEN_MASK | USP_TXC_SLAVE_CLK_SAMPLE, ((data_len - 1) << USP_TXC_DATA_LEN_OFFSET) | ((frame_len - 1) << USP_TXC_FRAME_LEN_OFFSET) - | ((shifter_len - 1) << USP_TXC_SHIFTER_LEN_OFFSET)); + | ((shifter_len - 1) << USP_TXC_SHIFTER_LEN_OFFSET) + | USP_TXC_SLAVE_CLK_SAMPLE); else regmap_update_bits(usp->regmap, USP_RX_FRAME_CTRL, USP_RXC_DATA_LEN_MASK | USP_RXC_FRAME_LEN_MASK - | USP_RXC_SHIFTER_LEN_MASK, + | USP_RXC_SHIFTER_LEN_MASK | USP_SINGLE_SYNC_MODE, ((data_len - 1) << USP_RXC_DATA_LEN_OFFSET) | ((frame_len - 1) << USP_RXC_FRAME_LEN_OFFSET) - | ((shifter_len - 1) << USP_RXC_SHIFTER_LEN_OFFSET)); - - regmap_update_bits(usp->regmap, USP_MODE1, - USP_CLOCK_MODE_SLAVE, USP_CLOCK_MODE_SLAVE); - regmap_update_bits(usp->regmap, USP_MODE2, - USP_TFS_CLK_SLAVE_MODE | USP_RFS_CLK_SLAVE_MODE, - USP_TFS_CLK_SLAVE_MODE | USP_RFS_CLK_SLAVE_MODE); + | ((shifter_len - 1) << USP_RXC_SHIFTER_LEN_OFFSET) + | USP_SINGLE_SYNC_MODE); return 0; } @@ -253,7 +241,6 @@ static int sirf_usp_pcm_trigger(struct snd_pcm_substream *substream, int cmd, } static const struct snd_soc_dai_ops sirf_usp_pcm_dai_ops = { - .startup = sirf_usp_i2s_startup, .trigger = sirf_usp_pcm_trigger, .set_fmt = sirf_usp_pcm_set_dai_fmt, .hw_params = sirf_usp_pcm_hw_params, @@ -282,7 +269,6 @@ static struct snd_soc_dai_driver sirf_usp_pcm_dai = { .ops = &sirf_usp_pcm_dai_ops, }; -#ifdef CONFIG_PM static int sirf_usp_pcm_runtime_suspend(struct device *dev) { struct sirf_usp *usp = dev_get_drvdata(dev); @@ -293,9 +279,15 @@ static int sirf_usp_pcm_runtime_suspend(struct device *dev) static int sirf_usp_pcm_runtime_resume(struct device *dev) { struct sirf_usp *usp = dev_get_drvdata(dev); - return clk_prepare_enable(usp->clk); + int ret; + ret = clk_prepare_enable(usp->clk); + if (ret) { + dev_err(dev, "clk_enable failed: %d\n", ret); + return ret; + } + sirf_usp_i2s_init(usp); + return 0; } -#endif #ifdef CONFIG_PM_SLEEP static int sirf_usp_pcm_suspend(struct device *dev) @@ -369,6 +361,11 @@ static int sirf_usp_pcm_probe(struct platform_device *pdev) } pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = sirf_usp_pcm_runtime_resume(&pdev->dev); + if (ret) + return ret; + } ret = devm_snd_soc_register_component(&pdev->dev, &sirf_usp_component, &sirf_usp_pcm_dai, 1); @@ -381,7 +378,10 @@ static int sirf_usp_pcm_probe(struct platform_device *pdev) static int sirf_usp_pcm_remove(struct platform_device *pdev) { - pm_runtime_disable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) + sirf_usp_pcm_runtime_suspend(&pdev->dev); + else + pm_runtime_disable(&pdev->dev); return 0; } -- cgit v1.2.3 From a72d2abbe5752f3a773c4d8b7b41ae41f617f772 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 25 Jul 2014 13:12:54 +0300 Subject: ASoC: tlv320aic31xx: Do not ignore errors in aic31xx_device_init() We need to return the error codes from aic31xx_device_init() and return from the i2c_probe with the error code. We will have kernel panic (NULL pointer dereference) in regulator_register_notifier() in case the devm_regulator_bulk_get() fails (with -EPROBE_DEFER for example). Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/tlv320aic31xx.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 23419109ecac..1cdae8ccc61b 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1178,7 +1178,7 @@ static void aic31xx_pdata_from_of(struct aic31xx_priv *aic31xx) } #endif /* CONFIG_OF */ -static void aic31xx_device_init(struct aic31xx_priv *aic31xx) +static int aic31xx_device_init(struct aic31xx_priv *aic31xx) { int ret, i; @@ -1197,7 +1197,7 @@ static void aic31xx_device_init(struct aic31xx_priv *aic31xx) "aic31xx-reset-pin"); if (ret < 0) { dev_err(aic31xx->dev, "not able to acquire gpio\n"); - return; + return ret; } } @@ -1210,6 +1210,7 @@ static void aic31xx_device_init(struct aic31xx_priv *aic31xx) if (ret != 0) dev_err(aic31xx->dev, "Failed to request supplies: %d\n", ret); + return ret; } static int aic31xx_i2c_probe(struct i2c_client *i2c, @@ -1239,7 +1240,9 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, aic31xx->pdata.codec_type = id->driver_data; - aic31xx_device_init(aic31xx); + ret = aic31xx_device_init(aic31xx); + if (ret) + return ret; return snd_soc_register_codec(&i2c->dev, &soc_codec_driver_aic31xx, aic31xx_dai_driver, -- cgit v1.2.3 From 55b219440b255f9b9064c621550d29e5f4f8c20f Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Mon, 28 Jul 2014 21:21:00 +0800 Subject: ASoC: rockchip: remove redundant dev_err call in rockchip_i2s_probe() There is a error message within devm_ioremap_resource already, so remove the dev_err call to avoid redundant error message. Signed-off-by: Wei Yongjun Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_i2s.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 4203088518f0..8d8e4b59049f 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -428,10 +428,8 @@ static int rockchip_i2s_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); regs = devm_ioremap_resource(&pdev->dev, res); - if (IS_ERR(regs)) { - dev_err(&pdev->dev, "No memory resource\n"); + if (IS_ERR(regs)) return PTR_ERR(regs); - } i2s->regmap = devm_regmap_init_mmio(&pdev->dev, regs, &rockchip_i2s_regmap_config); -- cgit v1.2.3 From 4efd1fc746f54f7ff0d95744d8ec6fa17441d332 Mon Sep 17 00:00:00 2001 From: Michael Trimarchi Date: Mon, 28 Jul 2014 14:26:48 +0200 Subject: ASoC: pcm1792a: Add 88200 frequency Signed-off-by: Michael Trimarchi Signed-off-by: Mark Brown --- sound/soc/codecs/pcm1792a.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/pcm1792a.h b/sound/soc/codecs/pcm1792a.h index 7a83d1fc102a..51d5470fee16 100644 --- a/sound/soc/codecs/pcm1792a.h +++ b/sound/soc/codecs/pcm1792a.h @@ -18,7 +18,8 @@ #define __PCM1792A_H__ #define PCM1792A_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_8000_48000 | \ - SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000) + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_192000) #define PCM1792A_FORMATS (SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S16_LE) -- cgit v1.2.3 From 3aae27991957156b2903e8058dbda616085a9734 Mon Sep 17 00:00:00 2001 From: Michael Trimarchi Date: Mon, 28 Jul 2014 14:26:49 +0200 Subject: ASoC: pcm1792a: Add controls for output invert and rolloff switch [s/output/Output/ -- broonie] Signed-off-by: Michael Trimarchi Signed-off-by: Mark Brown --- sound/soc/codecs/pcm1792a.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/pcm1792a.c b/sound/soc/codecs/pcm1792a.c index 3a80ba4452df..57b0c94a710b 100644 --- a/sound/soc/codecs/pcm1792a.c +++ b/sound/soc/codecs/pcm1792a.c @@ -36,6 +36,7 @@ #define PCM1792A_DAC_VOL_LEFT 0x10 #define PCM1792A_DAC_VOL_RIGHT 0x11 #define PCM1792A_FMT_CONTROL 0x12 +#define PCM1792A_MODE_CONTROL 0x13 #define PCM1792A_SOFT_MUTE PCM1792A_FMT_CONTROL #define PCM1792A_FMT_MASK 0x70 @@ -164,6 +165,8 @@ static const struct snd_kcontrol_new pcm1792a_controls[] = { SOC_DOUBLE_R_RANGE_TLV("DAC Playback Volume", PCM1792A_DAC_VOL_LEFT, PCM1792A_DAC_VOL_RIGHT, 0, 0xf, 0xff, 0, pcm1792a_dac_tlv), + SOC_SINGLE("DAC Invert Output Switch", PCM1792A_MODE_CONTROL, 7, 1, 0), + SOC_SINGLE("DAC Rolloff Filter Switch", PCM1792A_MODE_CONTROL, 1, 1, 0), }; static const struct snd_soc_dapm_widget pcm1792a_dapm_widgets[] = { -- cgit v1.2.3 From 27204ca8b981160c3b65cdc226c9070f76cbbd36 Mon Sep 17 00:00:00 2001 From: Kiran Padwal Date: Mon, 28 Jul 2014 11:41:04 +0530 Subject: ASoC: ak4642: Make of_device_id array const Make of_device_id array const, because all OF functions handle it as const Signed-off-by: Kiran Padwal Signed-off-by: Mark Brown --- sound/soc/codecs/ak4642.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 3ba4c0f11418..041712592e29 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -547,7 +547,7 @@ static const struct ak4642_drvdata ak4648_drvdata = { .extended_frequencies = 1, }; -static struct of_device_id ak4642_of_match[]; +static const struct of_device_id ak4642_of_match[]; static int ak4642_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -593,7 +593,7 @@ static int ak4642_i2c_remove(struct i2c_client *client) return 0; } -static struct of_device_id ak4642_of_match[] = { +static const struct of_device_id ak4642_of_match[] = { { .compatible = "asahi-kasei,ak4642", .data = &ak4642_drvdata}, { .compatible = "asahi-kasei,ak4643", .data = &ak4643_drvdata}, { .compatible = "asahi-kasei,ak4648", .data = &ak4648_drvdata}, -- cgit v1.2.3 From 90f601efc886ee5881594c5d931b76775975e155 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 29 Jul 2014 13:50:57 +0800 Subject: ASoC: rt286: Fix null pointer issue To make the interrupt safe if it happens to be called before the card is ready, we use regmap read/write in the interrupt handler. Also, we try to prevent the interrupt happen before the card is ready by enabling codec's IRQ in the ASoC probe. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt286.c | 74 +++++++++++++++++++++++------------------------- 1 file changed, 36 insertions(+), 38 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 218f86efd196..e4f6102efc1a 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -36,7 +36,6 @@ struct rt286_priv { struct regmap *regmap; - struct snd_soc_codec *codec; struct rt286_platform_data pdata; struct i2c_client *i2c; struct snd_soc_jack *jack; @@ -295,9 +294,8 @@ static int rt286_support_power_controls[] = { }; #define RT286_POWER_REG_LEN ARRAY_SIZE(rt286_support_power_controls) -static int rt286_jack_detect(struct snd_soc_codec *codec, bool *hp, bool *mic) +static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic) { - struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); unsigned int val, buf; int i; @@ -305,23 +303,23 @@ static int rt286_jack_detect(struct snd_soc_codec *codec, bool *hp, bool *mic) *mic = false; if (rt286->pdata.cbj_en) { - buf = snd_soc_read(codec, RT286_GET_HP_SENSE); + regmap_read(rt286->regmap, RT286_GET_HP_SENSE, &buf); *hp = buf & 0x80000000; if (*hp) { /* power on HV,VERF */ - snd_soc_update_bits(codec, + regmap_update_bits(rt286->regmap, RT286_POWER_CTRL1, 0x1001, 0x0); /* power LDO1 */ - snd_soc_update_bits(codec, + regmap_update_bits(rt286->regmap, RT286_POWER_CTRL2, 0x4, 0x4); - snd_soc_write(codec, RT286_SET_MIC1, 0x24); - val = snd_soc_read(codec, RT286_CBJ_CTRL2); + regmap_write(rt286->regmap, RT286_SET_MIC1, 0x24); + regmap_read(rt286->regmap, RT286_CBJ_CTRL2, &val); msleep(200); i = 40; while (((val & 0x0800) == 0) && (i > 0)) { - val = snd_soc_read(codec, - RT286_CBJ_CTRL2); + regmap_read(rt286->regmap, + RT286_CBJ_CTRL2, &val); i--; msleep(20); } @@ -329,53 +327,53 @@ static int rt286_jack_detect(struct snd_soc_codec *codec, bool *hp, bool *mic) if (0x0400 == (val & 0x0700)) { *mic = false; - snd_soc_write(codec, + regmap_write(rt286->regmap, RT286_SET_MIC1, 0x20); /* power off HV,VERF */ - snd_soc_update_bits(codec, + regmap_update_bits(rt286->regmap, RT286_POWER_CTRL1, 0x1001, 0x1001); - snd_soc_update_bits(codec, + regmap_update_bits(rt286->regmap, RT286_A_BIAS_CTRL3, 0xc000, 0x0000); - snd_soc_update_bits(codec, + regmap_update_bits(rt286->regmap, RT286_CBJ_CTRL1, 0x0030, 0x0000); - snd_soc_update_bits(codec, + regmap_update_bits(rt286->regmap, RT286_A_BIAS_CTRL2, 0xc000, 0x0000); } else if ((0x0200 == (val & 0x0700)) || (0x0100 == (val & 0x0700))) { *mic = true; - snd_soc_update_bits(codec, + regmap_update_bits(rt286->regmap, RT286_A_BIAS_CTRL3, 0xc000, 0x8000); - snd_soc_update_bits(codec, + regmap_update_bits(rt286->regmap, RT286_CBJ_CTRL1, 0x0030, 0x0020); - snd_soc_update_bits(codec, + regmap_update_bits(rt286->regmap, RT286_A_BIAS_CTRL2, 0xc000, 0x8000); } else { *mic = false; } - snd_soc_update_bits(codec, + regmap_update_bits(rt286->regmap, RT286_MISC_CTRL1, 0x0060, 0x0000); } else { - snd_soc_update_bits(codec, + regmap_update_bits(rt286->regmap, RT286_MISC_CTRL1, 0x0060, 0x0020); - snd_soc_update_bits(codec, + regmap_update_bits(rt286->regmap, RT286_A_BIAS_CTRL3, 0xc000, 0x8000); - snd_soc_update_bits(codec, + regmap_update_bits(rt286->regmap, RT286_CBJ_CTRL1, 0x0030, 0x0020); - snd_soc_update_bits(codec, + regmap_update_bits(rt286->regmap, RT286_A_BIAS_CTRL2, 0xc000, 0x8000); *mic = false; } } else { - buf = snd_soc_read(codec, RT286_GET_HP_SENSE); + regmap_read(rt286->regmap, RT286_GET_HP_SENSE, &buf); *hp = buf & 0x80000000; - buf = snd_soc_read(codec, RT286_GET_MIC1_SENSE); + regmap_read(rt286->regmap, RT286_GET_MIC1_SENSE, &buf); *mic = buf & 0x80000000; } @@ -390,7 +388,7 @@ static void rt286_jack_detect_work(struct work_struct *work) bool hp = false; bool mic = false; - rt286_jack_detect(rt286->codec, &hp, &mic); + rt286_jack_detect(rt286, &hp, &mic); if (hp == true) status |= SND_JACK_HEADPHONE; @@ -940,11 +938,10 @@ static irqreturn_t rt286_irq(int irq, void *data) bool mic = false; int status = 0; - rt286_jack_detect(rt286->codec, &hp, &mic); + rt286_jack_detect(rt286, &hp, &mic); /* Clear IRQ */ - snd_soc_update_bits(rt286->codec, - RT286_IRQ_CTRL, 0x1, 0x1); + regmap_update_bits(rt286->regmap, RT286_IRQ_CTRL, 0x1, 0x1); if (hp == true) status |= SND_JACK_HEADPHONE; @@ -965,7 +962,16 @@ static int rt286_probe(struct snd_soc_codec *codec) struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); codec->dapm.bias_level = SND_SOC_BIAS_OFF; - rt286->codec = codec; + + if (rt286->i2c->irq) { + regmap_update_bits(rt286->regmap, + RT286_IRQ_CTRL, 0x2, 0x2); + + INIT_DELAYED_WORK(&rt286->jack_detect_work, + rt286_jack_detect_work); + schedule_delayed_work(&rt286->jack_detect_work, + msecs_to_jiffies(1250)); + } return 0; } @@ -1171,14 +1177,6 @@ static int rt286_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL4, 0x00ff, 0x003f); if (rt286->i2c->irq) { - regmap_update_bits(rt286->regmap, - RT286_IRQ_CTRL, 0x2, 0x2); - - INIT_DELAYED_WORK(&rt286->jack_detect_work, - rt286_jack_detect_work); - schedule_delayed_work(&rt286->jack_detect_work, - msecs_to_jiffies(1250)); - ret = request_threaded_irq(rt286->i2c->irq, NULL, rt286_irq, IRQF_TRIGGER_HIGH | IRQF_ONESHOT, "rt286", rt286); if (ret != 0) { -- cgit v1.2.3 From 689dc643859953651ffb7111fdbcff2eb0f02841 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Tue, 29 Jul 2014 18:39:37 +0800 Subject: ASoC: cs42xx8: Add SND_SOC_DAIFMT_DSP_A support According to the spec, the definition of TDM and ONELINE_24 for CS42XX8_INTF_DAC and CS42XX8_INTF_ADC is wrong. correct them and enable SND_SOC_DAIFMT_DSP_A support. Signed-off-by: Shengjiu Wang Acked-by: Brian Austin Acked-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/codecs/cs42xx8.c | 3 +++ sound/soc/codecs/cs42xx8.h | 8 ++++---- 2 files changed, 7 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c index ec53ffc4d8ce..02b1520ae0bc 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -219,6 +219,9 @@ static int cs42xx8_set_dai_fmt(struct snd_soc_dai *codec_dai, case SND_SOC_DAIFMT_RIGHT_J: val = CS42XX8_INTF_DAC_DIF_RIGHTJ | CS42XX8_INTF_ADC_DIF_RIGHTJ; break; + case SND_SOC_DAIFMT_DSP_A: + val = CS42XX8_INTF_DAC_DIF_TDM | CS42XX8_INTF_ADC_DIF_TDM; + break; default: dev_err(codec->dev, "unsupported dai format\n"); return -EINVAL; diff --git a/sound/soc/codecs/cs42xx8.h b/sound/soc/codecs/cs42xx8.h index da0b94aee419..b2c10e537ef6 100644 --- a/sound/soc/codecs/cs42xx8.h +++ b/sound/soc/codecs/cs42xx8.h @@ -128,8 +128,8 @@ int cs42xx8_probe(struct device *dev, struct regmap *regmap); #define CS42XX8_INTF_DAC_DIF_RIGHTJ (2 << CS42XX8_INTF_DAC_DIF_SHIFT) #define CS42XX8_INTF_DAC_DIF_RIGHTJ_16 (3 << CS42XX8_INTF_DAC_DIF_SHIFT) #define CS42XX8_INTF_DAC_DIF_ONELINE_20 (4 << CS42XX8_INTF_DAC_DIF_SHIFT) -#define CS42XX8_INTF_DAC_DIF_ONELINE_24 (6 << CS42XX8_INTF_DAC_DIF_SHIFT) -#define CS42XX8_INTF_DAC_DIF_TDM (7 << CS42XX8_INTF_DAC_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_ONELINE_24 (5 << CS42XX8_INTF_DAC_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_TDM (6 << CS42XX8_INTF_DAC_DIF_SHIFT) #define CS42XX8_INTF_ADC_DIF_SHIFT 0 #define CS42XX8_INTF_ADC_DIF_WIDTH 3 #define CS42XX8_INTF_ADC_DIF_MASK (((1 << CS42XX8_INTF_ADC_DIF_WIDTH) - 1) << CS42XX8_INTF_ADC_DIF_SHIFT) @@ -138,8 +138,8 @@ int cs42xx8_probe(struct device *dev, struct regmap *regmap); #define CS42XX8_INTF_ADC_DIF_RIGHTJ (2 << CS42XX8_INTF_ADC_DIF_SHIFT) #define CS42XX8_INTF_ADC_DIF_RIGHTJ_16 (3 << CS42XX8_INTF_ADC_DIF_SHIFT) #define CS42XX8_INTF_ADC_DIF_ONELINE_20 (4 << CS42XX8_INTF_ADC_DIF_SHIFT) -#define CS42XX8_INTF_ADC_DIF_ONELINE_24 (6 << CS42XX8_INTF_ADC_DIF_SHIFT) -#define CS42XX8_INTF_ADC_DIF_TDM (7 << CS42XX8_INTF_ADC_DIF_SHIFT) +#define CS42XX8_INTF_ADC_DIF_ONELINE_24 (5 << CS42XX8_INTF_ADC_DIF_SHIFT) +#define CS42XX8_INTF_ADC_DIF_TDM (6 << CS42XX8_INTF_ADC_DIF_SHIFT) /* ADC Control & DAC De-Emphasis (Address 05h) */ #define CS42XX8_ADCCTL_ADC_HPF_FREEZE_SHIFT 7 -- cgit v1.2.3 From b8c637864a6904a9ba8e0df556d5bdf9f26b2c54 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 29 Jul 2014 00:37:31 -0700 Subject: ASoC: rsnd: use regmap_mmio instead of original regmap bus Current rsnd driver is using regmap and regmap_field. It used original regmap bus which is single regmap instance for multi register mapping. This patch modifies it to use regmap_mmio bus, and tidyuped probe method Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/Kconfig | 2 +- sound/soc/sh/rcar/gen.c | 439 +++++++++++++++++++++--------------------------- 2 files changed, 189 insertions(+), 252 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index b43fdf0d08af..80245b6eebd6 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -37,7 +37,7 @@ config SND_SOC_SH4_SIU config SND_SOC_RCAR tristate "R-Car series SRU/SCU/SSIU/SSI support" select SND_SIMPLE_CARD - select REGMAP + select REGMAP_MMIO help This option enables R-Car SUR/SCU/SSIU/SSI sound support diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 73ce4c90efda..5f9e0722abcf 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -15,63 +15,35 @@ struct rsnd_gen { struct rsnd_gen_ops *ops; - struct regmap *regmap; + struct regmap *regmap[RSND_BASE_MAX]; struct regmap_field *regs[RSND_REG_MAX]; }; #define rsnd_priv_to_gen(p) ((struct rsnd_gen *)(p)->gen) -#define RSND_REG_SET(gen, id, reg_id, offset, _id_offset, _id_size) \ - [id] = { \ - .reg = (unsigned int)gen->base[reg_id] + offset, \ - .lsb = 0, \ - .msb = 31, \ - .id_size = _id_size, \ - .id_offset = _id_offset, \ - } - -/* - * basic function - */ -static int rsnd_regmap_write32(void *context, const void *_data, size_t count) -{ - struct rsnd_priv *priv = context; - struct device *dev = rsnd_priv_to_dev(priv); - u32 *data = (u32 *)_data; - u32 val = data[1]; - void __iomem *reg = (void *)data[0]; - - iowrite32(val, reg); - - dev_dbg(dev, "w %p : %08x\n", reg, val); - - return 0; -} - -static int rsnd_regmap_read32(void *context, - const void *_data, size_t reg_size, - void *_val, size_t val_size) -{ - struct rsnd_priv *priv = context; - struct device *dev = rsnd_priv_to_dev(priv); - u32 *data = (u32 *)_data; - u32 *val = (u32 *)_val; - void __iomem *reg = (void *)data[0]; - - *val = ioread32(reg); - - dev_dbg(dev, "r %p : %08x\n", reg, *val); +struct rsnd_regmap_field_conf { + int idx; + unsigned int reg_offset; + unsigned int id_offset; +}; - return 0; +#define RSND_REG_SET(id, offset, _id_offset) \ +{ \ + .idx = id, \ + .reg_offset = offset, \ + .id_offset = _id_offset, \ } +/* single address mapping */ +#define RSND_GEN_S_REG(id, offset) \ + RSND_REG_SET(RSND_REG_##id, offset, 0) -static struct regmap_bus rsnd_regmap_bus = { - .write = rsnd_regmap_write32, - .read = rsnd_regmap_read32, - .reg_format_endian_default = REGMAP_ENDIAN_NATIVE, - .val_format_endian_default = REGMAP_ENDIAN_NATIVE, -}; +/* multi address mapping */ +#define RSND_GEN_M_REG(id, offset, _id_offset) \ + RSND_REG_SET(RSND_REG_##id, offset, _id_offset) +/* + * basic function + */ static int rsnd_is_accessible_reg(struct rsnd_priv *priv, struct rsnd_gen *gen, enum rsnd_reg reg) { @@ -88,6 +60,7 @@ static int rsnd_is_accessible_reg(struct rsnd_priv *priv, u32 rsnd_read(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg) { + struct device *dev = rsnd_priv_to_dev(priv); struct rsnd_gen *gen = rsnd_priv_to_gen(priv); u32 val; @@ -96,6 +69,8 @@ u32 rsnd_read(struct rsnd_priv *priv, regmap_fields_read(gen->regs[reg], rsnd_mod_id(mod), &val); + dev_dbg(dev, "r %s - 0x%04d : %08x\n", rsnd_mod_name(mod), reg, val); + return val; } @@ -103,12 +78,15 @@ void rsnd_write(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg, u32 data) { + struct device *dev = rsnd_priv_to_dev(priv); struct rsnd_gen *gen = rsnd_priv_to_gen(priv); if (!rsnd_is_accessible_reg(priv, gen, reg)) return; regmap_fields_write(gen->regs[reg], rsnd_mod_id(mod), data); + + dev_dbg(dev, "w %s - 0x%04d : %08x\n", rsnd_mod_name(mod), reg, data); } void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, @@ -123,33 +101,58 @@ void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, mask, data); } -static int rsnd_gen_regmap_init(struct rsnd_priv *priv, - struct rsnd_gen *gen, - struct reg_field *regf) +#define rsnd_gen_regmap_init(priv, id_size, reg_id, conf) \ + _rsnd_gen_regmap_init(priv, id_size, reg_id, conf, ARRAY_SIZE(conf)) +static int _rsnd_gen_regmap_init(struct rsnd_priv *priv, + int id_size, + int reg_id, + struct rsnd_regmap_field_conf *conf, + int conf_size) { - int i; + struct platform_device *pdev = rsnd_priv_to_pdev(priv); + struct rsnd_gen *gen = rsnd_priv_to_gen(priv); struct device *dev = rsnd_priv_to_dev(priv); + struct resource *res; struct regmap_config regc; + struct regmap_field *regs; + struct regmap *regmap; + struct reg_field regf; + void __iomem *base; + int i; memset(®c, 0, sizeof(regc)); regc.reg_bits = 32; regc.val_bits = 32; + regc.reg_stride = 4; - gen->regmap = devm_regmap_init(dev, &rsnd_regmap_bus, priv, ®c); - if (IS_ERR(gen->regmap)) { - dev_err(dev, "regmap error %ld\n", PTR_ERR(gen->regmap)); - return PTR_ERR(gen->regmap); - } + res = platform_get_resource(pdev, IORESOURCE_MEM, reg_id); + if (!res) + return -ENODEV; + + base = devm_ioremap_resource(dev, res); + if (IS_ERR(base)) + return PTR_ERR(base); - for (i = 0; i < RSND_REG_MAX; i++) { - gen->regs[i] = NULL; - if (!regf[i].reg) - continue; + regmap = devm_regmap_init_mmio(dev, base, ®c); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); - gen->regs[i] = devm_regmap_field_alloc(dev, gen->regmap, regf[i]); - if (IS_ERR(gen->regs[i])) - return PTR_ERR(gen->regs[i]); + gen->base[reg_id] = base; + gen->regmap[reg_id] = regmap; + for (i = 0; i < conf_size; i++) { + + regf.reg = conf[i].reg_offset; + regf.id_offset = conf[i].id_offset; + regf.lsb = 0; + regf.msb = 31; + regf.id_size = id_size; + + regs = devm_regmap_field_alloc(dev, regmap, regf); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + gen->regs[conf[i].idx] = regs; } return 0; @@ -271,119 +274,85 @@ dma_addr_t rsnd_gen_dma_addr(struct rsnd_priv *priv, /* * Gen2 */ - -/* single address mapping */ -#define RSND_GEN2_S_REG(gen, reg, id, offset) \ - RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN2_##reg, offset, 0, 10) - -/* multi address mapping */ -#define RSND_GEN2_M_REG(gen, reg, id, offset, _id_offset) \ - RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN2_##reg, offset, _id_offset, 10) - -static int rsnd_gen2_regmap_init(struct rsnd_priv *priv, struct rsnd_gen *gen) -{ - struct reg_field regf[RSND_REG_MAX] = { - RSND_GEN2_S_REG(gen, SSIU, SSI_MODE0, 0x800), - RSND_GEN2_S_REG(gen, SSIU, SSI_MODE1, 0x804), - /* FIXME: it needs SSI_MODE2/3 in the future */ - RSND_GEN2_M_REG(gen, SSIU, SSI_BUSIF_MODE, 0x0, 0x80), - RSND_GEN2_M_REG(gen, SSIU, SSI_BUSIF_ADINR,0x4, 0x80), - RSND_GEN2_M_REG(gen, SSIU, SSI_CTRL, 0x10, 0x80), - RSND_GEN2_M_REG(gen, SSIU, INT_ENABLE, 0x18, 0x80), - - RSND_GEN2_M_REG(gen, SCU, SRC_BUSIF_MODE, 0x0, 0x20), - RSND_GEN2_M_REG(gen, SCU, SRC_ROUTE_MODE0,0xc, 0x20), - RSND_GEN2_M_REG(gen, SCU, SRC_CTRL, 0x10, 0x20), - RSND_GEN2_M_REG(gen, SCU, CMD_ROUTE_SLCT, 0x18c, 0x20), - RSND_GEN2_M_REG(gen, SCU, CMD_CTRL, 0x190, 0x20), - RSND_GEN2_M_REG(gen, SCU, SRC_SWRSR, 0x200, 0x40), - RSND_GEN2_M_REG(gen, SCU, SRC_SRCIR, 0x204, 0x40), - RSND_GEN2_M_REG(gen, SCU, SRC_ADINR, 0x214, 0x40), - RSND_GEN2_M_REG(gen, SCU, SRC_IFSCR, 0x21c, 0x40), - RSND_GEN2_M_REG(gen, SCU, SRC_IFSVR, 0x220, 0x40), - RSND_GEN2_M_REG(gen, SCU, SRC_SRCCR, 0x224, 0x40), - RSND_GEN2_M_REG(gen, SCU, SRC_BSDSR, 0x22c, 0x40), - RSND_GEN2_M_REG(gen, SCU, SRC_BSISR, 0x238, 0x40), - RSND_GEN2_M_REG(gen, SCU, DVC_SWRSR, 0xe00, 0x100), - RSND_GEN2_M_REG(gen, SCU, DVC_DVUIR, 0xe04, 0x100), - RSND_GEN2_M_REG(gen, SCU, DVC_ADINR, 0xe08, 0x100), - RSND_GEN2_M_REG(gen, SCU, DVC_DVUCR, 0xe10, 0x100), - RSND_GEN2_M_REG(gen, SCU, DVC_ZCMCR, 0xe14, 0x100), - RSND_GEN2_M_REG(gen, SCU, DVC_VOL0R, 0xe28, 0x100), - RSND_GEN2_M_REG(gen, SCU, DVC_VOL1R, 0xe2c, 0x100), - RSND_GEN2_M_REG(gen, SCU, DVC_DVUER, 0xe48, 0x100), - - RSND_GEN2_S_REG(gen, ADG, BRRA, 0x00), - RSND_GEN2_S_REG(gen, ADG, BRRB, 0x04), - RSND_GEN2_S_REG(gen, ADG, SSICKR, 0x08), - RSND_GEN2_S_REG(gen, ADG, AUDIO_CLK_SEL0, 0x0c), - RSND_GEN2_S_REG(gen, ADG, AUDIO_CLK_SEL1, 0x10), - RSND_GEN2_S_REG(gen, ADG, AUDIO_CLK_SEL2, 0x14), - RSND_GEN2_S_REG(gen, ADG, DIV_EN, 0x30), - RSND_GEN2_S_REG(gen, ADG, SRCIN_TIMSEL0, 0x34), - RSND_GEN2_S_REG(gen, ADG, SRCIN_TIMSEL1, 0x38), - RSND_GEN2_S_REG(gen, ADG, SRCIN_TIMSEL2, 0x3c), - RSND_GEN2_S_REG(gen, ADG, SRCIN_TIMSEL3, 0x40), - RSND_GEN2_S_REG(gen, ADG, SRCIN_TIMSEL4, 0x44), - RSND_GEN2_S_REG(gen, ADG, SRCOUT_TIMSEL0, 0x48), - RSND_GEN2_S_REG(gen, ADG, SRCOUT_TIMSEL1, 0x4c), - RSND_GEN2_S_REG(gen, ADG, SRCOUT_TIMSEL2, 0x50), - RSND_GEN2_S_REG(gen, ADG, SRCOUT_TIMSEL3, 0x54), - RSND_GEN2_S_REG(gen, ADG, SRCOUT_TIMSEL4, 0x58), - RSND_GEN2_S_REG(gen, ADG, CMDOUT_TIMSEL, 0x5c), - - RSND_GEN2_M_REG(gen, SSI, SSICR, 0x00, 0x40), - RSND_GEN2_M_REG(gen, SSI, SSISR, 0x04, 0x40), - RSND_GEN2_M_REG(gen, SSI, SSITDR, 0x08, 0x40), - RSND_GEN2_M_REG(gen, SSI, SSIRDR, 0x0c, 0x40), - RSND_GEN2_M_REG(gen, SSI, SSIWSR, 0x20, 0x40), - }; - - return rsnd_gen_regmap_init(priv, gen, regf); -} - static int rsnd_gen2_probe(struct platform_device *pdev, struct rsnd_priv *priv) { struct device *dev = rsnd_priv_to_dev(priv); - struct rsnd_gen *gen = rsnd_priv_to_gen(priv); - struct resource *scu_res; - struct resource *adg_res; - struct resource *ssiu_res; - struct resource *ssi_res; - int ret; - - /* - * map address - */ - scu_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN2_SCU); - adg_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN2_ADG); - ssiu_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN2_SSIU); - ssi_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN2_SSI); - - gen->base[RSND_GEN2_SCU] = devm_ioremap_resource(dev, scu_res); - gen->base[RSND_GEN2_ADG] = devm_ioremap_resource(dev, adg_res); - gen->base[RSND_GEN2_SSIU] = devm_ioremap_resource(dev, ssiu_res); - gen->base[RSND_GEN2_SSI] = devm_ioremap_resource(dev, ssi_res); - if (IS_ERR(gen->base[RSND_GEN2_SCU]) || - IS_ERR(gen->base[RSND_GEN2_ADG]) || - IS_ERR(gen->base[RSND_GEN2_SSIU]) || - IS_ERR(gen->base[RSND_GEN2_SSI])) - return -ENODEV; - - ret = rsnd_gen2_regmap_init(priv, gen); - if (ret < 0) - return ret; - - dev_dbg(dev, "Gen2 device probed\n"); - dev_dbg(dev, "SCU : %pap => %p\n", &scu_res->start, - gen->base[RSND_GEN2_SCU]); - dev_dbg(dev, "ADG : %pap => %p\n", &adg_res->start, - gen->base[RSND_GEN2_ADG]); - dev_dbg(dev, "SSIU : %pap => %p\n", &ssiu_res->start, - gen->base[RSND_GEN2_SSIU]); - dev_dbg(dev, "SSI : %pap => %p\n", &ssi_res->start, - gen->base[RSND_GEN2_SSI]); + struct rsnd_regmap_field_conf conf_ssiu[] = { + RSND_GEN_S_REG(SSI_MODE0, 0x800), + RSND_GEN_S_REG(SSI_MODE1, 0x804), + /* FIXME: it needs SSI_MODE2/3 in the future */ + RSND_GEN_M_REG(SSI_BUSIF_MODE, 0x0, 0x80), + RSND_GEN_M_REG(SSI_BUSIF_ADINR, 0x4, 0x80), + RSND_GEN_M_REG(SSI_CTRL, 0x10, 0x80), + RSND_GEN_M_REG(INT_ENABLE, 0x18, 0x80), + }; + struct rsnd_regmap_field_conf conf_scu[] = { + RSND_GEN_M_REG(SRC_BUSIF_MODE, 0x0, 0x20), + RSND_GEN_M_REG(SRC_ROUTE_MODE0, 0xc, 0x20), + RSND_GEN_M_REG(SRC_CTRL, 0x10, 0x20), + RSND_GEN_M_REG(CMD_ROUTE_SLCT, 0x18c, 0x20), + RSND_GEN_M_REG(CMD_CTRL, 0x190, 0x20), + RSND_GEN_M_REG(SRC_SWRSR, 0x200, 0x40), + RSND_GEN_M_REG(SRC_SRCIR, 0x204, 0x40), + RSND_GEN_M_REG(SRC_ADINR, 0x214, 0x40), + RSND_GEN_M_REG(SRC_IFSCR, 0x21c, 0x40), + RSND_GEN_M_REG(SRC_IFSVR, 0x220, 0x40), + RSND_GEN_M_REG(SRC_SRCCR, 0x224, 0x40), + RSND_GEN_M_REG(SRC_BSDSR, 0x22c, 0x40), + RSND_GEN_M_REG(SRC_BSISR, 0x238, 0x40), + RSND_GEN_M_REG(DVC_SWRSR, 0xe00, 0x100), + RSND_GEN_M_REG(DVC_DVUIR, 0xe04, 0x100), + RSND_GEN_M_REG(DVC_ADINR, 0xe08, 0x100), + RSND_GEN_M_REG(DVC_DVUCR, 0xe10, 0x100), + RSND_GEN_M_REG(DVC_ZCMCR, 0xe14, 0x100), + RSND_GEN_M_REG(DVC_VOL0R, 0xe28, 0x100), + RSND_GEN_M_REG(DVC_VOL1R, 0xe2c, 0x100), + RSND_GEN_M_REG(DVC_DVUER, 0xe48, 0x100), + }; + struct rsnd_regmap_field_conf conf_adg[] = { + RSND_GEN_S_REG(BRRA, 0x00), + RSND_GEN_S_REG(BRRB, 0x04), + RSND_GEN_S_REG(SSICKR, 0x08), + RSND_GEN_S_REG(AUDIO_CLK_SEL0, 0x0c), + RSND_GEN_S_REG(AUDIO_CLK_SEL1, 0x10), + RSND_GEN_S_REG(AUDIO_CLK_SEL2, 0x14), + RSND_GEN_S_REG(DIV_EN, 0x30), + RSND_GEN_S_REG(SRCIN_TIMSEL0, 0x34), + RSND_GEN_S_REG(SRCIN_TIMSEL1, 0x38), + RSND_GEN_S_REG(SRCIN_TIMSEL2, 0x3c), + RSND_GEN_S_REG(SRCIN_TIMSEL3, 0x40), + RSND_GEN_S_REG(SRCIN_TIMSEL4, 0x44), + RSND_GEN_S_REG(SRCOUT_TIMSEL0, 0x48), + RSND_GEN_S_REG(SRCOUT_TIMSEL1, 0x4c), + RSND_GEN_S_REG(SRCOUT_TIMSEL2, 0x50), + RSND_GEN_S_REG(SRCOUT_TIMSEL3, 0x54), + RSND_GEN_S_REG(SRCOUT_TIMSEL4, 0x58), + RSND_GEN_S_REG(CMDOUT_TIMSEL, 0x5c), + }; + struct rsnd_regmap_field_conf conf_ssi[] = { + RSND_GEN_M_REG(SSICR, 0x00, 0x40), + RSND_GEN_M_REG(SSISR, 0x04, 0x40), + RSND_GEN_M_REG(SSITDR, 0x08, 0x40), + RSND_GEN_M_REG(SSIRDR, 0x0c, 0x40), + RSND_GEN_M_REG(SSIWSR, 0x20, 0x40), + }; + int ret_ssiu; + int ret_scu; + int ret_adg; + int ret_ssi; + + ret_ssiu = rsnd_gen_regmap_init(priv, 10, RSND_GEN2_SSIU, conf_ssiu); + ret_scu = rsnd_gen_regmap_init(priv, 10, RSND_GEN2_SCU, conf_scu); + ret_adg = rsnd_gen_regmap_init(priv, 10, RSND_GEN2_ADG, conf_adg); + ret_ssi = rsnd_gen_regmap_init(priv, 10, RSND_GEN2_SSI, conf_ssi); + if (ret_ssiu < 0 || + ret_scu < 0 || + ret_adg < 0 || + ret_ssi < 0) + return ret_ssiu | ret_scu | ret_adg | ret_ssi; + + dev_dbg(dev, "Gen2 is probed\n"); return 0; } @@ -392,92 +361,60 @@ static int rsnd_gen2_probe(struct platform_device *pdev, * Gen1 */ -/* single address mapping */ -#define RSND_GEN1_S_REG(gen, reg, id, offset) \ - RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN1_##reg, offset, 0, 9) - -/* multi address mapping */ -#define RSND_GEN1_M_REG(gen, reg, id, offset, _id_offset) \ - RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN1_##reg, offset, _id_offset, 9) - -static int rsnd_gen1_regmap_init(struct rsnd_priv *priv, struct rsnd_gen *gen) -{ - struct reg_field regf[RSND_REG_MAX] = { - RSND_GEN1_S_REG(gen, SRU, SRC_ROUTE_SEL, 0x00), - RSND_GEN1_S_REG(gen, SRU, SRC_TMG_SEL0, 0x08), - RSND_GEN1_S_REG(gen, SRU, SRC_TMG_SEL1, 0x0c), - RSND_GEN1_S_REG(gen, SRU, SRC_TMG_SEL2, 0x10), - RSND_GEN1_S_REG(gen, SRU, SRC_ROUTE_CTRL, 0xc0), - RSND_GEN1_S_REG(gen, SRU, SSI_MODE0, 0xD0), - RSND_GEN1_S_REG(gen, SRU, SSI_MODE1, 0xD4), - RSND_GEN1_M_REG(gen, SRU, SRC_BUSIF_MODE, 0x20, 0x4), - RSND_GEN1_M_REG(gen, SRU, SRC_ROUTE_MODE0,0x50, 0x8), - RSND_GEN1_M_REG(gen, SRU, SRC_SWRSR, 0x200, 0x40), - RSND_GEN1_M_REG(gen, SRU, SRC_SRCIR, 0x204, 0x40), - RSND_GEN1_M_REG(gen, SRU, SRC_ADINR, 0x214, 0x40), - RSND_GEN1_M_REG(gen, SRU, SRC_IFSCR, 0x21c, 0x40), - RSND_GEN1_M_REG(gen, SRU, SRC_IFSVR, 0x220, 0x40), - RSND_GEN1_M_REG(gen, SRU, SRC_SRCCR, 0x224, 0x40), - RSND_GEN1_M_REG(gen, SRU, SRC_MNFSR, 0x228, 0x40), - - RSND_GEN1_S_REG(gen, ADG, BRRA, 0x00), - RSND_GEN1_S_REG(gen, ADG, BRRB, 0x04), - RSND_GEN1_S_REG(gen, ADG, SSICKR, 0x08), - RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL0, 0x0c), - RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL1, 0x10), - RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL3, 0x18), - RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL4, 0x1c), - RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL5, 0x20), - - RSND_GEN1_M_REG(gen, SSI, SSICR, 0x00, 0x40), - RSND_GEN1_M_REG(gen, SSI, SSISR, 0x04, 0x40), - RSND_GEN1_M_REG(gen, SSI, SSITDR, 0x08, 0x40), - RSND_GEN1_M_REG(gen, SSI, SSIRDR, 0x0c, 0x40), - RSND_GEN1_M_REG(gen, SSI, SSIWSR, 0x20, 0x40), - }; - - return rsnd_gen_regmap_init(priv, gen, regf); -} - static int rsnd_gen1_probe(struct platform_device *pdev, struct rsnd_priv *priv) { struct device *dev = rsnd_priv_to_dev(priv); - struct rsnd_gen *gen = rsnd_priv_to_gen(priv); - struct resource *sru_res; - struct resource *adg_res; - struct resource *ssi_res; - int ret; - - /* - * map address - */ - sru_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN1_SRU); - adg_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN1_ADG); - ssi_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN1_SSI); - - gen->base[RSND_GEN1_SRU] = devm_ioremap_resource(dev, sru_res); - gen->base[RSND_GEN1_ADG] = devm_ioremap_resource(dev, adg_res); - gen->base[RSND_GEN1_SSI] = devm_ioremap_resource(dev, ssi_res); - if (IS_ERR(gen->base[RSND_GEN1_SRU]) || - IS_ERR(gen->base[RSND_GEN1_ADG]) || - IS_ERR(gen->base[RSND_GEN1_SSI])) - return -ENODEV; + struct rsnd_regmap_field_conf conf_sru[] = { + RSND_GEN_S_REG(SRC_ROUTE_SEL, 0x00), + RSND_GEN_S_REG(SRC_TMG_SEL0, 0x08), + RSND_GEN_S_REG(SRC_TMG_SEL1, 0x0c), + RSND_GEN_S_REG(SRC_TMG_SEL2, 0x10), + RSND_GEN_S_REG(SRC_ROUTE_CTRL, 0xc0), + RSND_GEN_S_REG(SSI_MODE0, 0xD0), + RSND_GEN_S_REG(SSI_MODE1, 0xD4), + RSND_GEN_M_REG(SRC_BUSIF_MODE, 0x20, 0x4), + RSND_GEN_M_REG(SRC_ROUTE_MODE0, 0x50, 0x8), + RSND_GEN_M_REG(SRC_SWRSR, 0x200, 0x40), + RSND_GEN_M_REG(SRC_SRCIR, 0x204, 0x40), + RSND_GEN_M_REG(SRC_ADINR, 0x214, 0x40), + RSND_GEN_M_REG(SRC_IFSCR, 0x21c, 0x40), + RSND_GEN_M_REG(SRC_IFSVR, 0x220, 0x40), + RSND_GEN_M_REG(SRC_SRCCR, 0x224, 0x40), + RSND_GEN_M_REG(SRC_MNFSR, 0x228, 0x40), + }; + struct rsnd_regmap_field_conf conf_adg[] = { + RSND_GEN_S_REG(BRRA, 0x00), + RSND_GEN_S_REG(BRRB, 0x04), + RSND_GEN_S_REG(SSICKR, 0x08), + RSND_GEN_S_REG(AUDIO_CLK_SEL0, 0x0c), + RSND_GEN_S_REG(AUDIO_CLK_SEL1, 0x10), + RSND_GEN_S_REG(AUDIO_CLK_SEL3, 0x18), + RSND_GEN_S_REG(AUDIO_CLK_SEL4, 0x1c), + RSND_GEN_S_REG(AUDIO_CLK_SEL5, 0x20), + }; + struct rsnd_regmap_field_conf conf_ssi[] = { + RSND_GEN_M_REG(SSICR, 0x00, 0x40), + RSND_GEN_M_REG(SSISR, 0x04, 0x40), + RSND_GEN_M_REG(SSITDR, 0x08, 0x40), + RSND_GEN_M_REG(SSIRDR, 0x0c, 0x40), + RSND_GEN_M_REG(SSIWSR, 0x20, 0x40), + }; + int ret_sru; + int ret_adg; + int ret_ssi; - ret = rsnd_gen1_regmap_init(priv, gen); - if (ret < 0) - return ret; + ret_sru = rsnd_gen_regmap_init(priv, 9, RSND_GEN1_SRU, conf_sru); + ret_adg = rsnd_gen_regmap_init(priv, 9, RSND_GEN1_ADG, conf_adg); + ret_ssi = rsnd_gen_regmap_init(priv, 9, RSND_GEN1_SSI, conf_ssi); + if (ret_sru < 0 || + ret_adg < 0 || + ret_ssi < 0) + return ret_sru | ret_adg | ret_ssi; - dev_dbg(dev, "Gen1 device probed\n"); - dev_dbg(dev, "SRU : %pap => %p\n", &sru_res->start, - gen->base[RSND_GEN1_SRU]); - dev_dbg(dev, "ADG : %pap => %p\n", &adg_res->start, - gen->base[RSND_GEN1_ADG]); - dev_dbg(dev, "SSI : %pap => %p\n", &ssi_res->start, - gen->base[RSND_GEN1_SSI]); + dev_dbg(dev, "Gen1 is probed\n"); return 0; - } /* -- cgit v1.2.3 From 3117bb3109dc223e186302f5dc8ce9ed04adca90 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Tue, 29 Jul 2014 18:08:53 +0800 Subject: ASoC: fsl_asrc: Add ASRC ASoC CPU DAI and platform drivers The Asynchronous Sample Rate Converter (ASRC) converts the sampling rate of a signal associated with an input clock into a signal associated with a different output clock. The driver currently works as a Front End of DPCM with other Back Ends DAI links such as ESAI<->CS42888 and SSI<->WM8962 and SAI. It converts the original sample rate to a common rate supported by Back Ends for playback while converts the common rate of Back Ends to a desired rate for capture. It has 3 pairs to support three different substreams within totally 10 channels. Signed-off-by: Nicolin Chen Reviewed-by: Varka Bhadram Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/fsl,asrc.txt | 60 ++ sound/soc/fsl/Kconfig | 9 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_asrc.c | 992 +++++++++++++++++++++ sound/soc/fsl/fsl_asrc.h | 461 ++++++++++ sound/soc/fsl/fsl_asrc_dma.c | 386 ++++++++ 6 files changed, 1910 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,asrc.txt create mode 100644 sound/soc/fsl/fsl_asrc.c create mode 100644 sound/soc/fsl/fsl_asrc.h create mode 100644 sound/soc/fsl/fsl_asrc_dma.c (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/fsl,asrc.txt b/Documentation/devicetree/bindings/sound/fsl,asrc.txt new file mode 100644 index 000000000000..b93362a570be --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,asrc.txt @@ -0,0 +1,60 @@ +Freescale Asynchronous Sample Rate Converter (ASRC) Controller + +The Asynchronous Sample Rate Converter (ASRC) converts the sampling rate of a +signal associated with an input clock into a signal associated with a different +output clock. The driver currently works as a Front End of DPCM with other Back +Ends Audio controller such as ESAI, SSI and SAI. It has three pairs to support +three substreams within totally 10 channels. + +Required properties: + + - compatible : Contains "fsl,imx35-asrc" or "fsl,imx53-asrc". + + - reg : Offset and length of the register set for the device. + + - interrupts : Contains the spdif interrupt. + + - dmas : Generic dma devicetree binding as described in + Documentation/devicetree/bindings/dma/dma.txt. + + - dma-names : Contains "rxa", "rxb", "rxc", "txa", "txb" and "txc". + + - clocks : Contains an entry for each entry in clock-names. + + - clock-names : Contains the following entries + "mem" Peripheral access clock to access registers. + "ipg" Peripheral clock to driver module. + "asrck_<0-f>" Clock sources for input and output clock. + + - big-endian : If this property is absent, the little endian mode + will be in use as default. Otherwise, the big endian + mode will be in use for all the device registers. + + - fsl,asrc-rate : Defines a mutual sample rate used by DPCM Back Ends. + + - fsl,asrc-width : Defines a mutual sample width used by DPCM Back Ends. + +Example: + +asrc: asrc@02034000 { + compatible = "fsl,imx53-asrc"; + reg = <0x02034000 0x4000>; + interrupts = <0 50 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&clks 107>, <&clks 107>, <&clks 0>, + <&clks 0>, <&clks 0>, <&clks 0>, <&clks 0>, + <&clks 0>, <&clks 0>, <&clks 0>, <&clks 0>, + <&clks 0>, <&clks 0>, <&clks 0>, <&clks 0>, + <&clks 107>, <&clks 0>, <&clks 0>; + clock-names = "mem", "ipg", "asrck0", + "asrck_1", "asrck_2", "asrck_3", "asrck_4", + "asrck_5", "asrck_6", "asrck_7", "asrck_8", + "asrck_9", "asrck_a", "asrck_b", "asrck_c", + "asrck_d", "asrck_e", "asrck_f"; + dmas = <&sdma 17 23 1>, <&sdma 18 23 1>, <&sdma 19 23 1>, + <&sdma 20 23 1>, <&sdma 21 23 1>, <&sdma 22 23 1>; + dma-names = "rxa", "rxb", "rxc", + "txa", "txb", "txc"; + fsl,asrc-rate = <48000>; + fsl,asrc-width = <16>; + status = "okay"; +}; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 37933629cbed..56fd32b3a881 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -2,6 +2,15 @@ menu "SoC Audio for Freescale CPUs" comment "Common SoC Audio options for Freescale CPUs:" +config SND_SOC_FSL_ASRC + tristate "Asynchronous Sample Rate Converter (ASRC) module support" + select REGMAP_MMIO + help + Say Y if you want to add Asynchronous Sample Rate Converter (ASRC) + support for the Freescale CPUs. + This option is only useful for out-of-tree drivers since + in-tree drivers select it automatically. + config SND_SOC_FSL_SAI tristate "Synchronous Audio Interface (SAI) module support" select REGMAP_MMIO diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index db254e358c18..9ff59267eac9 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -11,6 +11,7 @@ snd-soc-p1022-rdk-objs := p1022_rdk.o obj-$(CONFIG_SND_SOC_P1022_RDK) += snd-soc-p1022-rdk.o # Freescale SSI/DMA/SAI/SPDIF Support +snd-soc-fsl-asrc-objs := fsl_asrc.o fsl_asrc_dma.o snd-soc-fsl-sai-objs := fsl_sai.o snd-soc-fsl-ssi-y := fsl_ssi.o snd-soc-fsl-ssi-$(CONFIG_DEBUG_FS) += fsl_ssi_dbg.o @@ -18,6 +19,7 @@ snd-soc-fsl-spdif-objs := fsl_spdif.o snd-soc-fsl-esai-objs := fsl_esai.o snd-soc-fsl-utils-objs := fsl_utils.o snd-soc-fsl-dma-objs := fsl_dma.o +obj-$(CONFIG_SND_SOC_FSL_ASRC) += snd-soc-fsl-asrc.o obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o obj-$(CONFIG_SND_SOC_FSL_SSI) += snd-soc-fsl-ssi.o obj-$(CONFIG_SND_SOC_FSL_SPDIF) += snd-soc-fsl-spdif.o diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c new file mode 100644 index 000000000000..27a4a7084343 --- /dev/null +++ b/sound/soc/fsl/fsl_asrc.c @@ -0,0 +1,992 @@ +/* + * Freescale ASRC ALSA SoC Digital Audio Interface (DAI) driver + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "fsl_asrc.h" + +#define IDEAL_RATIO_DECIMAL_DEPTH 26 + +#define pair_err(fmt, ...) \ + dev_err(&asrc_priv->pdev->dev, "Pair %c: " fmt, 'A' + index, ##__VA_ARGS__) + +#define pair_dbg(fmt, ...) \ + dev_dbg(&asrc_priv->pdev->dev, "Pair %c: " fmt, 'A' + index, ##__VA_ARGS__) + +/* Sample rates are aligned with that defined in pcm.h file */ +static const u8 process_option[][8][2] = { + /* 32kHz 44.1kHz 48kHz 64kHz 88.2kHz 96kHz 176kHz 192kHz */ + {{0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 5512Hz */ + {{0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 8kHz */ + {{0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 11025Hz */ + {{0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 16kHz */ + {{0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 22050Hz */ + {{0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0},}, /* 32kHz */ + {{0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0},}, /* 44.1kHz */ + {{0, 2}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0},}, /* 48kHz */ + {{1, 2}, {0, 2}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0},}, /* 64kHz */ + {{1, 2}, {1, 2}, {1, 2}, {1, 1}, {1, 1}, {1, 1}, {1, 1}, {1, 1},}, /* 88.2kHz */ + {{1, 2}, {1, 2}, {1, 2}, {1, 1}, {1, 1}, {1, 1}, {1, 1}, {1, 1},}, /* 96kHz */ + {{2, 2}, {2, 2}, {2, 2}, {2, 1}, {2, 1}, {2, 1}, {2, 1}, {2, 1},}, /* 176kHz */ + {{2, 2}, {2, 2}, {2, 2}, {2, 1}, {2, 1}, {2, 1}, {2, 1}, {2, 1},}, /* 192kHz */ +}; + +/* Corresponding to process_option */ +static int supported_input_rate[] = { + 5512, 8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000, 88200, + 96000, 176400, 192000, +}; + +static int supported_asrc_rate[] = { + 32000, 44100, 48000, 64000, 88200, 96000, 176400, 192000, +}; + +/** + * The following tables map the relationship between asrc_inclk/asrc_outclk in + * fsl_asrc.h and the registers of ASRCSR + */ +static unsigned char input_clk_map_imx35[] = { + 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 0xa, 0xb, 0xc, 0xd, 0xe, 0xf, +}; + +static unsigned char output_clk_map_imx35[] = { + 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 0xa, 0xb, 0xc, 0xd, 0xe, 0xf, +}; + +/* i.MX53 uses the same map for input and output */ +static unsigned char input_clk_map_imx53[] = { +/* 0x0 0x1 0x2 0x3 0x4 0x5 0x6 0x7 0x8 0x9 0xa 0xb 0xc 0xd 0xe 0xf */ + 0x0, 0x1, 0x2, 0x7, 0x4, 0x5, 0x6, 0x3, 0x8, 0x9, 0xa, 0xb, 0xc, 0xf, 0xe, 0xd, +}; + +static unsigned char output_clk_map_imx53[] = { +/* 0x0 0x1 0x2 0x3 0x4 0x5 0x6 0x7 0x8 0x9 0xa 0xb 0xc 0xd 0xe 0xf */ + 0x8, 0x9, 0xa, 0x7, 0xc, 0x5, 0x6, 0xb, 0x0, 0x1, 0x2, 0x3, 0x4, 0xf, 0xe, 0xd, +}; + +static unsigned char *clk_map[2]; + +/** + * Request ASRC pair + * + * It assigns pair by the order of A->C->B because allocation of pair B, + * within range [ANCA, ANCA+ANCB-1], depends on the channels of pair A + * while pair A and pair C are comparatively independent. + */ +static int fsl_asrc_request_pair(int channels, struct fsl_asrc_pair *pair) +{ + enum asrc_pair_index index = ASRC_INVALID_PAIR; + struct fsl_asrc *asrc_priv = pair->asrc_priv; + struct device *dev = &asrc_priv->pdev->dev; + unsigned long lock_flags; + int i, ret = 0; + + spin_lock_irqsave(&asrc_priv->lock, lock_flags); + + for (i = ASRC_PAIR_A; i < ASRC_PAIR_MAX_NUM; i++) { + if (asrc_priv->pair[i] != NULL) + continue; + + index = i; + + if (i != ASRC_PAIR_B) + break; + } + + if (index == ASRC_INVALID_PAIR) { + dev_err(dev, "all pairs are busy now\n"); + ret = -EBUSY; + } else if (asrc_priv->channel_avail < channels) { + dev_err(dev, "can't afford required channels: %d\n", channels); + ret = -EINVAL; + } else { + asrc_priv->channel_avail -= channels; + asrc_priv->pair[index] = pair; + pair->channels = channels; + pair->index = index; + } + + spin_unlock_irqrestore(&asrc_priv->lock, lock_flags); + + return ret; +} + +/** + * Release ASRC pair + * + * It clears the resource from asrc_priv and releases the occupied channels. + */ +static void fsl_asrc_release_pair(struct fsl_asrc_pair *pair) +{ + struct fsl_asrc *asrc_priv = pair->asrc_priv; + enum asrc_pair_index index = pair->index; + unsigned long lock_flags; + + /* Make sure the pair is disabled */ + regmap_update_bits(asrc_priv->regmap, REG_ASRCTR, + ASRCTR_ASRCEi_MASK(index), 0); + + spin_lock_irqsave(&asrc_priv->lock, lock_flags); + + asrc_priv->channel_avail += pair->channels; + asrc_priv->pair[index] = NULL; + pair->error = 0; + + spin_unlock_irqrestore(&asrc_priv->lock, lock_flags); +} + +/** + * Configure input and output thresholds + */ +static void fsl_asrc_set_watermarks(struct fsl_asrc_pair *pair, u32 in, u32 out) +{ + struct fsl_asrc *asrc_priv = pair->asrc_priv; + enum asrc_pair_index index = pair->index; + + regmap_update_bits(asrc_priv->regmap, REG_ASRMCR(index), + ASRMCRi_EXTTHRSHi_MASK | + ASRMCRi_INFIFO_THRESHOLD_MASK | + ASRMCRi_OUTFIFO_THRESHOLD_MASK, + ASRMCRi_EXTTHRSHi | + ASRMCRi_INFIFO_THRESHOLD(in) | + ASRMCRi_OUTFIFO_THRESHOLD(out)); +} + +/** + * Calculate the total divisor between asrck clock rate and sample rate + * + * It follows the formula clk_rate = samplerate * (2 ^ prescaler) * divider + */ +static u32 fsl_asrc_cal_asrck_divisor(struct fsl_asrc_pair *pair, u32 div) +{ + u32 ps; + + /* Calculate the divisors: prescaler [2^0, 2^7], divder [1, 8] */ + for (ps = 0; div > 8; ps++) + div >>= 1; + + return ((div - 1) << ASRCDRi_AxCPi_WIDTH) | ps; +} + +/** + * Calculate and set the ratio for Ideal Ratio mode only + * + * The ratio is a 32-bit fixed point value with 26 fractional bits. + */ +static int fsl_asrc_set_ideal_ratio(struct fsl_asrc_pair *pair, + int inrate, int outrate) +{ + struct fsl_asrc *asrc_priv = pair->asrc_priv; + enum asrc_pair_index index = pair->index; + unsigned long ratio; + int i; + + if (!outrate) { + pair_err("output rate should not be zero\n"); + return -EINVAL; + } + + /* Calculate the intergal part of the ratio */ + ratio = (inrate / outrate) << IDEAL_RATIO_DECIMAL_DEPTH; + + /* ... and then the 26 depth decimal part */ + inrate %= outrate; + + for (i = 1; i <= IDEAL_RATIO_DECIMAL_DEPTH; i++) { + inrate <<= 1; + + if (inrate < outrate) + continue; + + ratio |= 1 << (IDEAL_RATIO_DECIMAL_DEPTH - i); + inrate -= outrate; + + if (!inrate) + break; + } + + regmap_write(asrc_priv->regmap, REG_ASRIDRL(index), ratio); + regmap_write(asrc_priv->regmap, REG_ASRIDRH(index), ratio >> 24); + + return 0; +} + +/** + * Configure the assigned ASRC pair + * + * It configures those ASRC registers according to a configuration instance + * of struct asrc_config which includes in/output sample rate, width, channel + * and clock settings. + */ +static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) +{ + struct asrc_config *config = pair->config; + struct fsl_asrc *asrc_priv = pair->asrc_priv; + enum asrc_pair_index index = pair->index; + u32 inrate = config->input_sample_rate, indiv; + u32 outrate = config->output_sample_rate, outdiv; + bool ideal = config->inclk == INCLK_NONE; + u32 clk_index[2], div[2]; + int in, out, channels; + struct clk *clk; + + if (!config) { + pair_err("invalid pair config\n"); + return -EINVAL; + } + + /* Validate channels */ + if (config->channel_num < 1 || config->channel_num > 10) { + pair_err("does not support %d channels\n", config->channel_num); + return -EINVAL; + } + + /* Validate output width */ + if (config->output_word_width == ASRC_WIDTH_8_BIT) { + pair_err("does not support 8bit width output\n"); + return -EINVAL; + } + + /* Validate input and output sample rates */ + for (in = 0; in < ARRAY_SIZE(supported_input_rate); in++) + if (inrate == supported_input_rate[in]) + break; + + if (in == ARRAY_SIZE(supported_input_rate)) { + pair_err("unsupported input sample rate: %dHz\n", inrate); + return -EINVAL; + } + + for (out = 0; out < ARRAY_SIZE(supported_asrc_rate); out++) + if (outrate == supported_asrc_rate[out]) + break; + + if (out == ARRAY_SIZE(supported_asrc_rate)) { + pair_err("unsupported output sample rate: %dHz\n", outrate); + return -EINVAL; + } + + /* Validate input and output clock sources */ + clk_index[IN] = clk_map[IN][config->inclk]; + clk_index[OUT] = clk_map[OUT][config->outclk]; + + /* We only have output clock for ideal ratio mode */ + clk = asrc_priv->asrck_clk[clk_index[ideal ? OUT : IN]]; + + div[IN] = clk_get_rate(clk) / inrate; + if (div[IN] == 0) { + pair_err("failed to support input sample rate %dHz by asrck_%x\n", + inrate, clk_index[ideal ? OUT : IN]); + return -EINVAL; + } + + clk = asrc_priv->asrck_clk[clk_index[OUT]]; + + /* Use fixed output rate for Ideal Ratio mode (INCLK_NONE) */ + if (ideal) + div[OUT] = clk_get_rate(clk) / IDEAL_RATIO_RATE; + else + div[OUT] = clk_get_rate(clk) / outrate; + + if (div[OUT] == 0) { + pair_err("failed to support output sample rate %dHz by asrck_%x\n", + outrate, clk_index[OUT]); + return -EINVAL; + } + + /* Set the channel number */ + channels = config->channel_num; + + if (asrc_priv->channel_bits < 4) + channels /= 2; + + /* Update channels for current pair */ + regmap_update_bits(asrc_priv->regmap, REG_ASRCNCR, + ASRCNCR_ANCi_MASK(index, asrc_priv->channel_bits), + ASRCNCR_ANCi(index, channels, asrc_priv->channel_bits)); + + /* Default setting: Automatic selection for processing mode */ + regmap_update_bits(asrc_priv->regmap, REG_ASRCTR, + ASRCTR_ATSi_MASK(index), ASRCTR_ATS(index)); + regmap_update_bits(asrc_priv->regmap, REG_ASRCTR, + ASRCTR_USRi_MASK(index), 0); + + /* Set the input and output clock sources */ + regmap_update_bits(asrc_priv->regmap, REG_ASRCSR, + ASRCSR_AICSi_MASK(index) | ASRCSR_AOCSi_MASK(index), + ASRCSR_AICS(index, clk_index[IN]) | + ASRCSR_AOCS(index, clk_index[OUT])); + + /* Calculate the input clock divisors */ + indiv = fsl_asrc_cal_asrck_divisor(pair, div[IN]); + outdiv = fsl_asrc_cal_asrck_divisor(pair, div[OUT]); + + /* Suppose indiv and outdiv includes prescaler, so add its MASK too */ + regmap_update_bits(asrc_priv->regmap, REG_ASRCDR(index), + ASRCDRi_AOCPi_MASK(index) | ASRCDRi_AICPi_MASK(index) | + ASRCDRi_AOCDi_MASK(index) | ASRCDRi_AICDi_MASK(index), + ASRCDRi_AOCP(index, outdiv) | ASRCDRi_AICP(index, indiv)); + + /* Implement word_width configurations */ + regmap_update_bits(asrc_priv->regmap, REG_ASRMCR1(index), + ASRMCR1i_OW16_MASK | ASRMCR1i_IWD_MASK, + ASRMCR1i_OW16(config->output_word_width) | + ASRMCR1i_IWD(config->input_word_width)); + + /* Enable BUFFER STALL */ + regmap_update_bits(asrc_priv->regmap, REG_ASRMCR(index), + ASRMCRi_BUFSTALLi_MASK, ASRMCRi_BUFSTALLi); + + /* Set default thresholds for input and output FIFO */ + fsl_asrc_set_watermarks(pair, ASRC_INPUTFIFO_THRESHOLD, + ASRC_INPUTFIFO_THRESHOLD); + + /* Configure the followings only for Ideal Ratio mode */ + if (!ideal) + return 0; + + /* Clear ASTSx bit to use Ideal Ratio mode */ + regmap_update_bits(asrc_priv->regmap, REG_ASRCTR, + ASRCTR_ATSi_MASK(index), 0); + + /* Enable Ideal Ratio mode */ + regmap_update_bits(asrc_priv->regmap, REG_ASRCTR, + ASRCTR_IDRi_MASK(index) | ASRCTR_USRi_MASK(index), + ASRCTR_IDR(index) | ASRCTR_USR(index)); + + /* Apply configurations for pre- and post-processing */ + regmap_update_bits(asrc_priv->regmap, REG_ASRCFG, + ASRCFG_PREMODi_MASK(index) | ASRCFG_POSTMODi_MASK(index), + ASRCFG_PREMOD(index, process_option[in][out][0]) | + ASRCFG_POSTMOD(index, process_option[in][out][1])); + + return fsl_asrc_set_ideal_ratio(pair, inrate, outrate); +} + +/** + * Start the assigned ASRC pair + * + * It enables the assigned pair and makes it stopped at the stall level. + */ +static void fsl_asrc_start_pair(struct fsl_asrc_pair *pair) +{ + struct fsl_asrc *asrc_priv = pair->asrc_priv; + enum asrc_pair_index index = pair->index; + int reg, retry = 10, i; + + /* Enable the current pair */ + regmap_update_bits(asrc_priv->regmap, REG_ASRCTR, + ASRCTR_ASRCEi_MASK(index), ASRCTR_ASRCE(index)); + + /* Wait for status of initialization */ + do { + udelay(5); + regmap_read(asrc_priv->regmap, REG_ASRCFG, ®); + reg &= ASRCFG_INIRQi_MASK(index); + } while (!reg && --retry); + + /* Make the input fifo to ASRC STALL level */ + regmap_read(asrc_priv->regmap, REG_ASRCNCR, ®); + for (i = 0; i < pair->channels * 4; i++) + regmap_write(asrc_priv->regmap, REG_ASRDI(index), 0); + + /* Enable overload interrupt */ + regmap_write(asrc_priv->regmap, REG_ASRIER, ASRIER_AOLIE); +} + +/** + * Stop the assigned ASRC pair + */ +static void fsl_asrc_stop_pair(struct fsl_asrc_pair *pair) +{ + struct fsl_asrc *asrc_priv = pair->asrc_priv; + enum asrc_pair_index index = pair->index; + + /* Stop the current pair */ + regmap_update_bits(asrc_priv->regmap, REG_ASRCTR, + ASRCTR_ASRCEi_MASK(index), 0); +} + +/** + * Get DMA channel according to the pair and direction. + */ +struct dma_chan *fsl_asrc_get_dma_channel(struct fsl_asrc_pair *pair, bool dir) +{ + struct fsl_asrc *asrc_priv = pair->asrc_priv; + enum asrc_pair_index index = pair->index; + char name[4]; + + sprintf(name, "%cx%c", dir == IN ? 'r' : 't', index + 'a'); + + return dma_request_slave_channel(&asrc_priv->pdev->dev, name); +} +EXPORT_SYMBOL_GPL(fsl_asrc_get_dma_channel); + +static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct fsl_asrc *asrc_priv = snd_soc_dai_get_drvdata(dai); + int width = snd_pcm_format_width(params_format(params)); + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_asrc_pair *pair = runtime->private_data; + unsigned int channels = params_channels(params); + unsigned int rate = params_rate(params); + struct asrc_config config; + int word_width, ret; + + ret = fsl_asrc_request_pair(channels, pair); + if (ret) { + dev_err(dai->dev, "fail to request asrc pair\n"); + return ret; + } + + pair->config = &config; + + if (width == 16) + width = ASRC_WIDTH_16_BIT; + else + width = ASRC_WIDTH_24_BIT; + + if (asrc_priv->asrc_width == 16) + word_width = ASRC_WIDTH_16_BIT; + else + word_width = ASRC_WIDTH_24_BIT; + + config.pair = pair->index; + config.channel_num = channels; + config.inclk = INCLK_NONE; + config.outclk = OUTCLK_ASRCK1_CLK; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + config.input_word_width = width; + config.output_word_width = word_width; + config.input_sample_rate = rate; + config.output_sample_rate = asrc_priv->asrc_rate; + } else { + config.input_word_width = word_width; + config.output_word_width = width; + config.input_sample_rate = asrc_priv->asrc_rate; + config.output_sample_rate = rate; + } + + ret = fsl_asrc_config_pair(pair); + if (ret) { + dev_err(dai->dev, "fail to config asrc pair\n"); + return ret; + } + + return 0; +} + +static int fsl_asrc_dai_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_asrc_pair *pair = runtime->private_data; + + if (pair) + fsl_asrc_release_pair(pair); + + return 0; +} + +static int fsl_asrc_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_asrc_pair *pair = runtime->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + fsl_asrc_start_pair(pair); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + fsl_asrc_stop_pair(pair); + break; + default: + return -EINVAL; + } + + return 0; +} + +static struct snd_soc_dai_ops fsl_asrc_dai_ops = { + .hw_params = fsl_asrc_dai_hw_params, + .hw_free = fsl_asrc_dai_hw_free, + .trigger = fsl_asrc_dai_trigger, +}; + +static int fsl_asrc_dai_probe(struct snd_soc_dai *dai) +{ + struct fsl_asrc *asrc_priv = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, &asrc_priv->dma_params_tx, + &asrc_priv->dma_params_rx); + + return 0; +} + +#define FSL_ASRC_RATES SNDRV_PCM_RATE_8000_192000 +#define FSL_ASRC_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FORMAT_S20_3LE) + +static struct snd_soc_dai_driver fsl_asrc_dai = { + .probe = fsl_asrc_dai_probe, + .playback = { + .stream_name = "ASRC-Playback", + .channels_min = 1, + .channels_max = 10, + .rates = FSL_ASRC_RATES, + .formats = FSL_ASRC_FORMATS, + }, + .capture = { + .stream_name = "ASRC-Capture", + .channels_min = 1, + .channels_max = 10, + .rates = FSL_ASRC_RATES, + .formats = FSL_ASRC_FORMATS, + }, + .ops = &fsl_asrc_dai_ops, +}; + +static const struct snd_soc_component_driver fsl_asrc_component = { + .name = "fsl-asrc-dai", +}; + +static bool fsl_asrc_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case REG_ASRCTR: + case REG_ASRIER: + case REG_ASRCNCR: + case REG_ASRCFG: + case REG_ASRCSR: + case REG_ASRCDR1: + case REG_ASRCDR2: + case REG_ASRSTR: + case REG_ASRPM1: + case REG_ASRPM2: + case REG_ASRPM3: + case REG_ASRPM4: + case REG_ASRPM5: + case REG_ASRTFR1: + case REG_ASRCCR: + case REG_ASRDOA: + case REG_ASRDOB: + case REG_ASRDOC: + case REG_ASRIDRHA: + case REG_ASRIDRLA: + case REG_ASRIDRHB: + case REG_ASRIDRLB: + case REG_ASRIDRHC: + case REG_ASRIDRLC: + case REG_ASR76K: + case REG_ASR56K: + case REG_ASRMCRA: + case REG_ASRFSTA: + case REG_ASRMCRB: + case REG_ASRFSTB: + case REG_ASRMCRC: + case REG_ASRFSTC: + case REG_ASRMCR1A: + case REG_ASRMCR1B: + case REG_ASRMCR1C: + return true; + default: + return false; + } +} + +static bool fsl_asrc_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case REG_ASRSTR: + case REG_ASRDIA: + case REG_ASRDIB: + case REG_ASRDIC: + case REG_ASRDOA: + case REG_ASRDOB: + case REG_ASRDOC: + case REG_ASRFSTA: + case REG_ASRFSTB: + case REG_ASRFSTC: + case REG_ASRCFG: + return true; + default: + return false; + } +} + +static bool fsl_asrc_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case REG_ASRCTR: + case REG_ASRIER: + case REG_ASRCNCR: + case REG_ASRCFG: + case REG_ASRCSR: + case REG_ASRCDR1: + case REG_ASRCDR2: + case REG_ASRSTR: + case REG_ASRPM1: + case REG_ASRPM2: + case REG_ASRPM3: + case REG_ASRPM4: + case REG_ASRPM5: + case REG_ASRTFR1: + case REG_ASRCCR: + case REG_ASRDIA: + case REG_ASRDIB: + case REG_ASRDIC: + case REG_ASRIDRHA: + case REG_ASRIDRLA: + case REG_ASRIDRHB: + case REG_ASRIDRLB: + case REG_ASRIDRHC: + case REG_ASRIDRLC: + case REG_ASR76K: + case REG_ASR56K: + case REG_ASRMCRA: + case REG_ASRMCRB: + case REG_ASRMCRC: + case REG_ASRMCR1A: + case REG_ASRMCR1B: + case REG_ASRMCR1C: + return true; + default: + return false; + } +} + +static struct regmap_config fsl_asrc_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + + .max_register = REG_ASRMCR1C, + .readable_reg = fsl_asrc_readable_reg, + .volatile_reg = fsl_asrc_volatile_reg, + .writeable_reg = fsl_asrc_writeable_reg, + .cache_type = REGCACHE_RBTREE, +}; + +/** + * Initialize ASRC registers with a default configurations + */ +static int fsl_asrc_init(struct fsl_asrc *asrc_priv) +{ + /* Halt ASRC internal FP when input FIFO needs data for pair A, B, C */ + regmap_write(asrc_priv->regmap, REG_ASRCTR, ASRCTR_ASRCEN); + + /* Disable interrupt by default */ + regmap_write(asrc_priv->regmap, REG_ASRIER, 0x0); + + /* Apply recommended settings for parameters from Reference Manual */ + regmap_write(asrc_priv->regmap, REG_ASRPM1, 0x7fffff); + regmap_write(asrc_priv->regmap, REG_ASRPM2, 0x255555); + regmap_write(asrc_priv->regmap, REG_ASRPM3, 0xff7280); + regmap_write(asrc_priv->regmap, REG_ASRPM4, 0xff7280); + regmap_write(asrc_priv->regmap, REG_ASRPM5, 0xff7280); + + /* Base address for task queue FIFO. Set to 0x7C */ + regmap_update_bits(asrc_priv->regmap, REG_ASRTFR1, + ASRTFR1_TF_BASE_MASK, ASRTFR1_TF_BASE(0xfc)); + + /* Set the processing clock for 76KHz to 133M */ + regmap_write(asrc_priv->regmap, REG_ASR76K, 0x06D6); + + /* Set the processing clock for 56KHz to 133M */ + return regmap_write(asrc_priv->regmap, REG_ASR56K, 0x0947); +} + +/** + * Interrupt handler for ASRC + */ +static irqreturn_t fsl_asrc_isr(int irq, void *dev_id) +{ + struct fsl_asrc *asrc_priv = (struct fsl_asrc *)dev_id; + struct device *dev = &asrc_priv->pdev->dev; + enum asrc_pair_index index; + u32 status; + + regmap_read(asrc_priv->regmap, REG_ASRSTR, &status); + + /* Clean overload error */ + regmap_write(asrc_priv->regmap, REG_ASRSTR, ASRSTR_AOLE); + + /* + * We here use dev_dbg() for all exceptions because ASRC itself does + * not care if FIFO overflowed or underrun while a warning in the + * interrupt would result a ridged conversion. + */ + for (index = ASRC_PAIR_A; index < ASRC_PAIR_MAX_NUM; index++) { + if (!asrc_priv->pair[index]) + continue; + + if (status & ASRSTR_ATQOL) { + asrc_priv->pair[index]->error |= ASRC_TASK_Q_OVERLOAD; + dev_dbg(dev, "ASRC Task Queue FIFO overload\n"); + } + + if (status & ASRSTR_AOOL(index)) { + asrc_priv->pair[index]->error |= ASRC_OUTPUT_TASK_OVERLOAD; + pair_dbg("Output Task Overload\n"); + } + + if (status & ASRSTR_AIOL(index)) { + asrc_priv->pair[index]->error |= ASRC_INPUT_TASK_OVERLOAD; + pair_dbg("Input Task Overload\n"); + } + + if (status & ASRSTR_AODO(index)) { + asrc_priv->pair[index]->error |= ASRC_OUTPUT_BUFFER_OVERFLOW; + pair_dbg("Output Data Buffer has overflowed\n"); + } + + if (status & ASRSTR_AIDU(index)) { + asrc_priv->pair[index]->error |= ASRC_INPUT_BUFFER_UNDERRUN; + pair_dbg("Input Data Buffer has underflowed\n"); + } + } + + return IRQ_HANDLED; +} + +static int fsl_asrc_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct fsl_asrc *asrc_priv; + struct resource *res; + void __iomem *regs; + int irq, ret, i; + char tmp[16]; + + asrc_priv = devm_kzalloc(&pdev->dev, sizeof(*asrc_priv), GFP_KERNEL); + if (!asrc_priv) + return -ENOMEM; + + asrc_priv->pdev = pdev; + strcpy(asrc_priv->name, np->name); + + /* Get the addresses and IRQ */ + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + regs = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + asrc_priv->paddr = res->start; + + /* Register regmap and let it prepare core clock */ + if (of_property_read_bool(np, "big-endian")) + fsl_asrc_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG; + + asrc_priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "mem", regs, + &fsl_asrc_regmap_config); + if (IS_ERR(asrc_priv->regmap)) { + dev_err(&pdev->dev, "failed to init regmap\n"); + return PTR_ERR(asrc_priv->regmap); + } + + irq = platform_get_irq(pdev, 0); + if (irq < 0) { + dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); + return irq; + } + + ret = devm_request_irq(&pdev->dev, irq, fsl_asrc_isr, 0, + asrc_priv->name, asrc_priv); + if (ret) { + dev_err(&pdev->dev, "failed to claim irq %u: %d\n", irq, ret); + return ret; + } + + asrc_priv->mem_clk = devm_clk_get(&pdev->dev, "mem"); + if (IS_ERR(asrc_priv->mem_clk)) { + dev_err(&pdev->dev, "failed to get mem clock\n"); + return PTR_ERR(asrc_priv->ipg_clk); + } + + asrc_priv->ipg_clk = devm_clk_get(&pdev->dev, "ipg"); + if (IS_ERR(asrc_priv->ipg_clk)) { + dev_err(&pdev->dev, "failed to get ipg clock\n"); + return PTR_ERR(asrc_priv->ipg_clk); + } + + for (i = 0; i < ASRC_CLK_MAX_NUM; i++) { + sprintf(tmp, "asrck_%x", i); + asrc_priv->asrck_clk[i] = devm_clk_get(&pdev->dev, tmp); + if (IS_ERR(asrc_priv->asrck_clk[i])) { + dev_err(&pdev->dev, "failed to get %s clock\n", tmp); + return PTR_ERR(asrc_priv->asrck_clk[i]); + } + } + + if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx35-asrc")) { + asrc_priv->channel_bits = 3; + clk_map[IN] = input_clk_map_imx35; + clk_map[OUT] = output_clk_map_imx35; + } else { + asrc_priv->channel_bits = 4; + clk_map[IN] = input_clk_map_imx53; + clk_map[OUT] = output_clk_map_imx53; + } + + ret = fsl_asrc_init(asrc_priv); + if (ret) { + dev_err(&pdev->dev, "failed to init asrc %d\n", ret); + return -EINVAL; + } + + asrc_priv->channel_avail = 10; + + ret = of_property_read_u32(np, "fsl,asrc-rate", + &asrc_priv->asrc_rate); + if (ret) { + dev_err(&pdev->dev, "failed to get output rate\n"); + return -EINVAL; + } + + ret = of_property_read_u32(np, "fsl,asrc-width", + &asrc_priv->asrc_width); + if (ret) { + dev_err(&pdev->dev, "failed to get output width\n"); + return -EINVAL; + } + + if (asrc_priv->asrc_width != 16 && asrc_priv->asrc_width != 24) { + dev_warn(&pdev->dev, "unsupported width, switching to 24bit\n"); + asrc_priv->asrc_width = 24; + } + + platform_set_drvdata(pdev, asrc_priv); + pm_runtime_enable(&pdev->dev); + spin_lock_init(&asrc_priv->lock); + + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_asrc_component, + &fsl_asrc_dai, 1); + if (ret) { + dev_err(&pdev->dev, "failed to register ASoC DAI\n"); + return ret; + } + + ret = devm_snd_soc_register_platform(&pdev->dev, &fsl_asrc_platform); + if (ret) { + dev_err(&pdev->dev, "failed to register ASoC platform\n"); + return ret; + } + + dev_info(&pdev->dev, "driver registered\n"); + + return 0; +} + +#if CONFIG_PM_RUNTIME +static int fsl_asrc_runtime_resume(struct device *dev) +{ + struct fsl_asrc *asrc_priv = dev_get_drvdata(dev); + int i; + + clk_prepare_enable(asrc_priv->mem_clk); + clk_prepare_enable(asrc_priv->ipg_clk); + for (i = 0; i < ASRC_CLK_MAX_NUM; i++) + clk_prepare_enable(asrc_priv->asrck_clk[i]); + + return 0; +} + +static int fsl_asrc_runtime_suspend(struct device *dev) +{ + struct fsl_asrc *asrc_priv = dev_get_drvdata(dev); + int i; + + for (i = 0; i < ASRC_CLK_MAX_NUM; i++) + clk_disable_unprepare(asrc_priv->asrck_clk[i]); + clk_disable_unprepare(asrc_priv->ipg_clk); + clk_disable_unprepare(asrc_priv->mem_clk); + + return 0; +} +#endif /* CONFIG_PM_RUNTIME */ + +#if CONFIG_PM_SLEEP +static int fsl_asrc_suspend(struct device *dev) +{ + struct fsl_asrc *asrc_priv = dev_get_drvdata(dev); + + regcache_cache_only(asrc_priv->regmap, true); + regcache_mark_dirty(asrc_priv->regmap); + + return 0; +} + +static int fsl_asrc_resume(struct device *dev) +{ + struct fsl_asrc *asrc_priv = dev_get_drvdata(dev); + u32 asrctr; + + /* Stop all pairs provisionally */ + regmap_read(asrc_priv->regmap, REG_ASRCTR, &asrctr); + regmap_update_bits(asrc_priv->regmap, REG_ASRCTR, + ASRCTR_ASRCEi_ALL_MASK, 0); + + /* Restore all registers */ + regcache_cache_only(asrc_priv->regmap, false); + regcache_sync(asrc_priv->regmap); + + /* Restart enabled pairs */ + regmap_update_bits(asrc_priv->regmap, REG_ASRCTR, + ASRCTR_ASRCEi_ALL_MASK, asrctr); + + return 0; +} +#endif /* CONFIG_PM_SLEEP */ + +static const struct dev_pm_ops fsl_asrc_pm = { + SET_RUNTIME_PM_OPS(fsl_asrc_runtime_suspend, fsl_asrc_runtime_resume, NULL) + SET_SYSTEM_SLEEP_PM_OPS(fsl_asrc_suspend, fsl_asrc_resume) +}; + +static const struct of_device_id fsl_asrc_ids[] = { + { .compatible = "fsl,imx35-asrc", }, + { .compatible = "fsl,imx53-asrc", }, + {} +}; +MODULE_DEVICE_TABLE(of, fsl_asrc_ids); + +static struct platform_driver fsl_asrc_driver = { + .probe = fsl_asrc_probe, + .driver = { + .name = "fsl-asrc", + .of_match_table = fsl_asrc_ids, + .pm = &fsl_asrc_pm, + }, +}; +module_platform_driver(fsl_asrc_driver); + +MODULE_DESCRIPTION("Freescale ASRC ASoC driver"); +MODULE_AUTHOR("Nicolin Chen "); +MODULE_ALIAS("platform:fsl-asrc"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/fsl/fsl_asrc.h b/sound/soc/fsl/fsl_asrc.h new file mode 100644 index 000000000000..a3f211f53c23 --- /dev/null +++ b/sound/soc/fsl/fsl_asrc.h @@ -0,0 +1,461 @@ +/* + * fsl_asrc.h - Freescale ASRC ALSA SoC header file + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#ifndef _FSL_ASRC_H +#define _FSL_ASRC_H + +#define IN 0 +#define OUT 1 + +#define ASRC_DMA_BUFFER_NUM 2 +#define ASRC_INPUTFIFO_THRESHOLD 32 +#define ASRC_OUTPUTFIFO_THRESHOLD 32 +#define ASRC_FIFO_THRESHOLD_MIN 0 +#define ASRC_FIFO_THRESHOLD_MAX 63 +#define ASRC_DMA_BUFFER_SIZE (1024 * 48 * 4) +#define ASRC_MAX_BUFFER_SIZE (1024 * 48) +#define ASRC_OUTPUT_LAST_SAMPLE 8 + +#define IDEAL_RATIO_RATE 1000000 + +#define REG_ASRCTR 0x00 +#define REG_ASRIER 0x04 +#define REG_ASRCNCR 0x0C +#define REG_ASRCFG 0x10 +#define REG_ASRCSR 0x14 + +#define REG_ASRCDR1 0x18 +#define REG_ASRCDR2 0x1C +#define REG_ASRCDR(i) ((i < 2) ? REG_ASRCDR1 : REG_ASRCDR2) + +#define REG_ASRSTR 0x20 +#define REG_ASRRA 0x24 +#define REG_ASRRB 0x28 +#define REG_ASRRC 0x2C +#define REG_ASRPM1 0x40 +#define REG_ASRPM2 0x44 +#define REG_ASRPM3 0x48 +#define REG_ASRPM4 0x4C +#define REG_ASRPM5 0x50 +#define REG_ASRTFR1 0x54 +#define REG_ASRCCR 0x5C + +#define REG_ASRDIA 0x60 +#define REG_ASRDOA 0x64 +#define REG_ASRDIB 0x68 +#define REG_ASRDOB 0x6C +#define REG_ASRDIC 0x70 +#define REG_ASRDOC 0x74 +#define REG_ASRDI(i) (REG_ASRDIA + (i << 3)) +#define REG_ASRDO(i) (REG_ASRDOA + (i << 3)) +#define REG_ASRDx(x, i) (x == IN ? REG_ASRDI(i) : REG_ASRDO(i)) + +#define REG_ASRIDRHA 0x80 +#define REG_ASRIDRLA 0x84 +#define REG_ASRIDRHB 0x88 +#define REG_ASRIDRLB 0x8C +#define REG_ASRIDRHC 0x90 +#define REG_ASRIDRLC 0x94 +#define REG_ASRIDRH(i) (REG_ASRIDRHA + (i << 3)) +#define REG_ASRIDRL(i) (REG_ASRIDRLA + (i << 3)) + +#define REG_ASR76K 0x98 +#define REG_ASR56K 0x9C + +#define REG_ASRMCRA 0xA0 +#define REG_ASRFSTA 0xA4 +#define REG_ASRMCRB 0xA8 +#define REG_ASRFSTB 0xAC +#define REG_ASRMCRC 0xB0 +#define REG_ASRFSTC 0xB4 +#define REG_ASRMCR(i) (REG_ASRMCRA + (i << 3)) +#define REG_ASRFST(i) (REG_ASRFSTA + (i << 3)) + +#define REG_ASRMCR1A 0xC0 +#define REG_ASRMCR1B 0xC4 +#define REG_ASRMCR1C 0xC8 +#define REG_ASRMCR1(i) (REG_ASRMCR1A + (i << 2)) + + +/* REG0 0x00 REG_ASRCTR */ +#define ASRCTR_ATSi_SHIFT(i) (20 + i) +#define ASRCTR_ATSi_MASK(i) (1 << ASRCTR_ATSi_SHIFT(i)) +#define ASRCTR_ATS(i) (1 << ASRCTR_ATSi_SHIFT(i)) +#define ASRCTR_USRi_SHIFT(i) (14 + (i << 1)) +#define ASRCTR_USRi_MASK(i) (1 << ASRCTR_USRi_SHIFT(i)) +#define ASRCTR_USR(i) (1 << ASRCTR_USRi_SHIFT(i)) +#define ASRCTR_IDRi_SHIFT(i) (13 + (i << 1)) +#define ASRCTR_IDRi_MASK(i) (1 << ASRCTR_IDRi_SHIFT(i)) +#define ASRCTR_IDR(i) (1 << ASRCTR_IDRi_SHIFT(i)) +#define ASRCTR_SRST_SHIFT 4 +#define ASRCTR_SRST_MASK (1 << ASRCTR_SRST_SHIFT) +#define ASRCTR_SRST (1 << ASRCTR_SRST_SHIFT) +#define ASRCTR_ASRCEi_SHIFT(i) (1 + i) +#define ASRCTR_ASRCEi_MASK(i) (1 << ASRCTR_ASRCEi_SHIFT(i)) +#define ASRCTR_ASRCE(i) (1 << ASRCTR_ASRCEi_SHIFT(i)) +#define ASRCTR_ASRCEi_ALL_MASK (0x7 << ASRCTR_ASRCEi_SHIFT(0)) +#define ASRCTR_ASRCEN_SHIFT 0 +#define ASRCTR_ASRCEN_MASK (1 << ASRCTR_ASRCEN_SHIFT) +#define ASRCTR_ASRCEN (1 << ASRCTR_ASRCEN_SHIFT) + +/* REG1 0x04 REG_ASRIER */ +#define ASRIER_AFPWE_SHIFT 7 +#define ASRIER_AFPWE_MASK (1 << ASRIER_AFPWE_SHIFT) +#define ASRIER_AFPWE (1 << ASRIER_AFPWE_SHIFT) +#define ASRIER_AOLIE_SHIFT 6 +#define ASRIER_AOLIE_MASK (1 << ASRIER_AOLIE_SHIFT) +#define ASRIER_AOLIE (1 << ASRIER_AOLIE_SHIFT) +#define ASRIER_ADOEi_SHIFT(i) (3 + i) +#define ASRIER_ADOEi_MASK(i) (1 << ASRIER_ADOEi_SHIFT(i)) +#define ASRIER_ADOE(i) (1 << ASRIER_ADOEi_SHIFT(i)) +#define ASRIER_ADIEi_SHIFT(i) (0 + i) +#define ASRIER_ADIEi_MASK(i) (1 << ASRIER_ADIEi_SHIFT(i)) +#define ASRIER_ADIE(i) (1 << ASRIER_ADIEi_SHIFT(i)) + +/* REG2 0x0C REG_ASRCNCR */ +#define ASRCNCR_ANCi_SHIFT(i, b) (b * i) +#define ASRCNCR_ANCi_MASK(i, b) (((1 << b) - 1) << ASRCNCR_ANCi_SHIFT(i, b)) +#define ASRCNCR_ANCi(i, v, b) ((v << ASRCNCR_ANCi_SHIFT(i, b)) & ASRCNCR_ANCi_MASK(i, b)) + +/* REG3 0x10 REG_ASRCFG */ +#define ASRCFG_INIRQi_SHIFT(i) (21 + i) +#define ASRCFG_INIRQi_MASK(i) (1 << ASRCFG_INIRQi_SHIFT(i)) +#define ASRCFG_INIRQi (1 << ASRCFG_INIRQi_SHIFT(i)) +#define ASRCFG_NDPRi_SHIFT(i) (18 + i) +#define ASRCFG_NDPRi_MASK(i) (1 << ASRCFG_NDPRi_SHIFT(i)) +#define ASRCFG_NDPRi (1 << ASRCFG_NDPRi_SHIFT(i)) +#define ASRCFG_POSTMODi_SHIFT(i) (8 + (i << 2)) +#define ASRCFG_POSTMODi_WIDTH 2 +#define ASRCFG_POSTMODi_MASK(i) (((1 << ASRCFG_POSTMODi_WIDTH) - 1) << ASRCFG_POSTMODi_SHIFT(i)) +#define ASRCFG_POSTMOD(i, v) ((v) << ASRCFG_POSTMODi_SHIFT(i)) +#define ASRCFG_POSTMODi_UP(i) (0 << ASRCFG_POSTMODi_SHIFT(i)) +#define ASRCFG_POSTMODi_DCON(i) (1 << ASRCFG_POSTMODi_SHIFT(i)) +#define ASRCFG_POSTMODi_DOWN(i) (2 << ASRCFG_POSTMODi_SHIFT(i)) +#define ASRCFG_PREMODi_SHIFT(i) (6 + (i << 2)) +#define ASRCFG_PREMODi_WIDTH 2 +#define ASRCFG_PREMODi_MASK(i) (((1 << ASRCFG_PREMODi_WIDTH) - 1) << ASRCFG_PREMODi_SHIFT(i)) +#define ASRCFG_PREMOD(i, v) ((v) << ASRCFG_PREMODi_SHIFT(i)) +#define ASRCFG_PREMODi_UP(i) (0 << ASRCFG_PREMODi_SHIFT(i)) +#define ASRCFG_PREMODi_DCON(i) (1 << ASRCFG_PREMODi_SHIFT(i)) +#define ASRCFG_PREMODi_DOWN(i) (2 << ASRCFG_PREMODi_SHIFT(i)) +#define ASRCFG_PREMODi_BYPASS(i) (3 << ASRCFG_PREMODi_SHIFT(i)) + +/* REG4 0x14 REG_ASRCSR */ +#define ASRCSR_AxCSi_WIDTH 4 +#define ASRCSR_AxCSi_MASK ((1 << ASRCSR_AxCSi_WIDTH) - 1) +#define ASRCSR_AOCSi_SHIFT(i) (12 + (i << 2)) +#define ASRCSR_AOCSi_MASK(i) (((1 << ASRCSR_AxCSi_WIDTH) - 1) << ASRCSR_AOCSi_SHIFT(i)) +#define ASRCSR_AOCS(i, v) ((v) << ASRCSR_AOCSi_SHIFT(i)) +#define ASRCSR_AICSi_SHIFT(i) (i << 2) +#define ASRCSR_AICSi_MASK(i) (((1 << ASRCSR_AxCSi_WIDTH) - 1) << ASRCSR_AICSi_SHIFT(i)) +#define ASRCSR_AICS(i, v) ((v) << ASRCSR_AICSi_SHIFT(i)) + +/* REG5&6 0x18 & 0x1C REG_ASRCDR1 & ASRCDR2 */ +#define ASRCDRi_AxCPi_WIDTH 3 +#define ASRCDRi_AICPi_SHIFT(i) (0 + (i % 2) * 6) +#define ASRCDRi_AICPi_MASK(i) (((1 << ASRCDRi_AxCPi_WIDTH) - 1) << ASRCDRi_AICPi_SHIFT(i)) +#define ASRCDRi_AICP(i, v) ((v) << ASRCDRi_AICPi_SHIFT(i)) +#define ASRCDRi_AICDi_SHIFT(i) (3 + (i % 2) * 6) +#define ASRCDRi_AICDi_MASK(i) (((1 << ASRCDRi_AxCPi_WIDTH) - 1) << ASRCDRi_AICDi_SHIFT(i)) +#define ASRCDRi_AICD(i, v) ((v) << ASRCDRi_AICDi_SHIFT(i)) +#define ASRCDRi_AOCPi_SHIFT(i) ((i < 2) ? 12 + i * 6 : 6) +#define ASRCDRi_AOCPi_MASK(i) (((1 << ASRCDRi_AxCPi_WIDTH) - 1) << ASRCDRi_AOCPi_SHIFT(i)) +#define ASRCDRi_AOCP(i, v) ((v) << ASRCDRi_AOCPi_SHIFT(i)) +#define ASRCDRi_AOCDi_SHIFT(i) ((i < 2) ? 15 + i * 6 : 9) +#define ASRCDRi_AOCDi_MASK(i) (((1 << ASRCDRi_AxCPi_WIDTH) - 1) << ASRCDRi_AOCDi_SHIFT(i)) +#define ASRCDRi_AOCD(i, v) ((v) << ASRCDRi_AOCDi_SHIFT(i)) + +/* REG7 0x20 REG_ASRSTR */ +#define ASRSTR_DSLCNT_SHIFT 21 +#define ASRSTR_DSLCNT_MASK (1 << ASRSTR_DSLCNT_SHIFT) +#define ASRSTR_DSLCNT (1 << ASRSTR_DSLCNT_SHIFT) +#define ASRSTR_ATQOL_SHIFT 20 +#define ASRSTR_ATQOL_MASK (1 << ASRSTR_ATQOL_SHIFT) +#define ASRSTR_ATQOL (1 << ASRSTR_ATQOL_SHIFT) +#define ASRSTR_AOOLi_SHIFT(i) (17 + i) +#define ASRSTR_AOOLi_MASK(i) (1 << ASRSTR_AOOLi_SHIFT(i)) +#define ASRSTR_AOOL(i) (1 << ASRSTR_AOOLi_SHIFT(i)) +#define ASRSTR_AIOLi_SHIFT(i) (14 + i) +#define ASRSTR_AIOLi_MASK(i) (1 << ASRSTR_AIOLi_SHIFT(i)) +#define ASRSTR_AIOL(i) (1 << ASRSTR_AIOLi_SHIFT(i)) +#define ASRSTR_AODOi_SHIFT(i) (11 + i) +#define ASRSTR_AODOi_MASK(i) (1 << ASRSTR_AODOi_SHIFT(i)) +#define ASRSTR_AODO(i) (1 << ASRSTR_AODOi_SHIFT(i)) +#define ASRSTR_AIDUi_SHIFT(i) (8 + i) +#define ASRSTR_AIDUi_MASK(i) (1 << ASRSTR_AIDUi_SHIFT(i)) +#define ASRSTR_AIDU(i) (1 << ASRSTR_AIDUi_SHIFT(i)) +#define ASRSTR_FPWT_SHIFT 7 +#define ASRSTR_FPWT_MASK (1 << ASRSTR_FPWT_SHIFT) +#define ASRSTR_FPWT (1 << ASRSTR_FPWT_SHIFT) +#define ASRSTR_AOLE_SHIFT 6 +#define ASRSTR_AOLE_MASK (1 << ASRSTR_AOLE_SHIFT) +#define ASRSTR_AOLE (1 << ASRSTR_AOLE_SHIFT) +#define ASRSTR_AODEi_SHIFT(i) (3 + i) +#define ASRSTR_AODFi_MASK(i) (1 << ASRSTR_AODEi_SHIFT(i)) +#define ASRSTR_AODF(i) (1 << ASRSTR_AODEi_SHIFT(i)) +#define ASRSTR_AIDEi_SHIFT(i) (0 + i) +#define ASRSTR_AIDEi_MASK(i) (1 << ASRSTR_AIDEi_SHIFT(i)) +#define ASRSTR_AIDE(i) (1 << ASRSTR_AIDEi_SHIFT(i)) + +/* REG10 0x54 REG_ASRTFR1 */ +#define ASRTFR1_TF_BASE_WIDTH 7 +#define ASRTFR1_TF_BASE_SHIFT 6 +#define ASRTFR1_TF_BASE_MASK (((1 << ASRTFR1_TF_BASE_WIDTH) - 1) << ASRTFR1_TF_BASE_SHIFT) +#define ASRTFR1_TF_BASE(i) ((i) << ASRTFR1_TF_BASE_SHIFT) + +/* + * REG22 0xA0 REG_ASRMCRA + * REG24 0xA8 REG_ASRMCRB + * REG26 0xB0 REG_ASRMCRC + */ +#define ASRMCRi_ZEROBUFi_SHIFT 23 +#define ASRMCRi_ZEROBUFi_MASK (1 << ASRMCRi_ZEROBUFi_SHIFT) +#define ASRMCRi_ZEROBUFi (1 << ASRMCRi_ZEROBUFi_SHIFT) +#define ASRMCRi_EXTTHRSHi_SHIFT 22 +#define ASRMCRi_EXTTHRSHi_MASK (1 << ASRMCRi_EXTTHRSHi_SHIFT) +#define ASRMCRi_EXTTHRSHi (1 << ASRMCRi_EXTTHRSHi_SHIFT) +#define ASRMCRi_BUFSTALLi_SHIFT 21 +#define ASRMCRi_BUFSTALLi_MASK (1 << ASRMCRi_BUFSTALLi_SHIFT) +#define ASRMCRi_BUFSTALLi (1 << ASRMCRi_BUFSTALLi_SHIFT) +#define ASRMCRi_BYPASSPOLYi_SHIFT 20 +#define ASRMCRi_BYPASSPOLYi_MASK (1 << ASRMCRi_BYPASSPOLYi_SHIFT) +#define ASRMCRi_BYPASSPOLYi (1 << ASRMCRi_BYPASSPOLYi_SHIFT) +#define ASRMCRi_OUTFIFO_THRESHOLD_WIDTH 6 +#define ASRMCRi_OUTFIFO_THRESHOLD_SHIFT 12 +#define ASRMCRi_OUTFIFO_THRESHOLD_MASK (((1 << ASRMCRi_OUTFIFO_THRESHOLD_WIDTH) - 1) << ASRMCRi_OUTFIFO_THRESHOLD_SHIFT) +#define ASRMCRi_OUTFIFO_THRESHOLD(v) (((v) << ASRMCRi_OUTFIFO_THRESHOLD_SHIFT) & ASRMCRi_OUTFIFO_THRESHOLD_MASK) +#define ASRMCRi_RSYNIFi_SHIFT 11 +#define ASRMCRi_RSYNIFi_MASK (1 << ASRMCRi_RSYNIFi_SHIFT) +#define ASRMCRi_RSYNIFi (1 << ASRMCRi_RSYNIFi_SHIFT) +#define ASRMCRi_RSYNOFi_SHIFT 10 +#define ASRMCRi_RSYNOFi_MASK (1 << ASRMCRi_RSYNOFi_SHIFT) +#define ASRMCRi_RSYNOFi (1 << ASRMCRi_RSYNOFi_SHIFT) +#define ASRMCRi_INFIFO_THRESHOLD_WIDTH 6 +#define ASRMCRi_INFIFO_THRESHOLD_SHIFT 0 +#define ASRMCRi_INFIFO_THRESHOLD_MASK (((1 << ASRMCRi_INFIFO_THRESHOLD_WIDTH) - 1) << ASRMCRi_INFIFO_THRESHOLD_SHIFT) +#define ASRMCRi_INFIFO_THRESHOLD(v) (((v) << ASRMCRi_INFIFO_THRESHOLD_SHIFT) & ASRMCRi_INFIFO_THRESHOLD_MASK) + +/* + * REG23 0xA4 REG_ASRFSTA + * REG25 0xAC REG_ASRFSTB + * REG27 0xB4 REG_ASRFSTC + */ +#define ASRFSTi_OAFi_SHIFT 23 +#define ASRFSTi_OAFi_MASK (1 << ASRFSTi_OAFi_SHIFT) +#define ASRFSTi_OAFi (1 << ASRFSTi_OAFi_SHIFT) +#define ASRFSTi_OUTPUT_FIFO_WIDTH 7 +#define ASRFSTi_OUTPUT_FIFO_SHIFT 12 +#define ASRFSTi_OUTPUT_FIFO_MASK (((1 << ASRFSTi_OUTPUT_FIFO_WIDTH) - 1) << ASRFSTi_OUTPUT_FIFO_SHIFT) +#define ASRFSTi_IAEi_SHIFT 11 +#define ASRFSTi_IAEi_MASK (1 << ASRFSTi_OAFi_SHIFT) +#define ASRFSTi_IAEi (1 << ASRFSTi_OAFi_SHIFT) +#define ASRFSTi_INPUT_FIFO_WIDTH 7 +#define ASRFSTi_INPUT_FIFO_SHIFT 0 +#define ASRFSTi_INPUT_FIFO_MASK ((1 << ASRFSTi_INPUT_FIFO_WIDTH) - 1) + +/* REG28 0xC0 & 0xC4 & 0xC8 REG_ASRMCR1i */ +#define ASRMCR1i_IWD_WIDTH 3 +#define ASRMCR1i_IWD_SHIFT 9 +#define ASRMCR1i_IWD_MASK (((1 << ASRMCR1i_IWD_WIDTH) - 1) << ASRMCR1i_IWD_SHIFT) +#define ASRMCR1i_IWD(v) ((v) << ASRMCR1i_IWD_SHIFT) +#define ASRMCR1i_IMSB_SHIFT 8 +#define ASRMCR1i_IMSB_MASK (1 << ASRMCR1i_IMSB_SHIFT) +#define ASRMCR1i_IMSB_MSB (1 << ASRMCR1i_IMSB_SHIFT) +#define ASRMCR1i_IMSB_LSB (0 << ASRMCR1i_IMSB_SHIFT) +#define ASRMCR1i_OMSB_SHIFT 2 +#define ASRMCR1i_OMSB_MASK (1 << ASRMCR1i_OMSB_SHIFT) +#define ASRMCR1i_OMSB_MSB (1 << ASRMCR1i_OMSB_SHIFT) +#define ASRMCR1i_OMSB_LSB (0 << ASRMCR1i_OMSB_SHIFT) +#define ASRMCR1i_OSGN_SHIFT 1 +#define ASRMCR1i_OSGN_MASK (1 << ASRMCR1i_OSGN_SHIFT) +#define ASRMCR1i_OSGN (1 << ASRMCR1i_OSGN_SHIFT) +#define ASRMCR1i_OW16_SHIFT 0 +#define ASRMCR1i_OW16_MASK (1 << ASRMCR1i_OW16_SHIFT) +#define ASRMCR1i_OW16(v) ((v) << ASRMCR1i_OW16_SHIFT) + + +enum asrc_pair_index { + ASRC_INVALID_PAIR = -1, + ASRC_PAIR_A = 0, + ASRC_PAIR_B = 1, + ASRC_PAIR_C = 2, +}; + +#define ASRC_PAIR_MAX_NUM (ASRC_PAIR_C + 1) + +enum asrc_inclk { + INCLK_NONE = 0x03, + INCLK_ESAI_RX = 0x00, + INCLK_SSI1_RX = 0x01, + INCLK_SSI2_RX = 0x02, + INCLK_SSI3_RX = 0x07, + INCLK_SPDIF_RX = 0x04, + INCLK_MLB_CLK = 0x05, + INCLK_PAD = 0x06, + INCLK_ESAI_TX = 0x08, + INCLK_SSI1_TX = 0x09, + INCLK_SSI2_TX = 0x0a, + INCLK_SSI3_TX = 0x0b, + INCLK_SPDIF_TX = 0x0c, + INCLK_ASRCK1_CLK = 0x0f, +}; + +enum asrc_outclk { + OUTCLK_NONE = 0x03, + OUTCLK_ESAI_TX = 0x00, + OUTCLK_SSI1_TX = 0x01, + OUTCLK_SSI2_TX = 0x02, + OUTCLK_SSI3_TX = 0x07, + OUTCLK_SPDIF_TX = 0x04, + OUTCLK_MLB_CLK = 0x05, + OUTCLK_PAD = 0x06, + OUTCLK_ESAI_RX = 0x08, + OUTCLK_SSI1_RX = 0x09, + OUTCLK_SSI2_RX = 0x0a, + OUTCLK_SSI3_RX = 0x0b, + OUTCLK_SPDIF_RX = 0x0c, + OUTCLK_ASRCK1_CLK = 0x0f, +}; + +#define ASRC_CLK_MAX_NUM 16 + +enum asrc_word_width { + ASRC_WIDTH_24_BIT = 0, + ASRC_WIDTH_16_BIT = 1, + ASRC_WIDTH_8_BIT = 2, +}; + +struct asrc_config { + enum asrc_pair_index pair; + unsigned int channel_num; + unsigned int buffer_num; + unsigned int dma_buffer_size; + unsigned int input_sample_rate; + unsigned int output_sample_rate; + enum asrc_word_width input_word_width; + enum asrc_word_width output_word_width; + enum asrc_inclk inclk; + enum asrc_outclk outclk; +}; + +struct asrc_req { + unsigned int chn_num; + enum asrc_pair_index index; +}; + +struct asrc_querybuf { + unsigned int buffer_index; + unsigned int input_length; + unsigned int output_length; + unsigned long input_offset; + unsigned long output_offset; +}; + +struct asrc_convert_buffer { + void *input_buffer_vaddr; + void *output_buffer_vaddr; + unsigned int input_buffer_length; + unsigned int output_buffer_length; +}; + +struct asrc_status_flags { + enum asrc_pair_index index; + unsigned int overload_error; +}; + +enum asrc_error_status { + ASRC_TASK_Q_OVERLOAD = 0x01, + ASRC_OUTPUT_TASK_OVERLOAD = 0x02, + ASRC_INPUT_TASK_OVERLOAD = 0x04, + ASRC_OUTPUT_BUFFER_OVERFLOW = 0x08, + ASRC_INPUT_BUFFER_UNDERRUN = 0x10, +}; + +struct dma_block { + dma_addr_t dma_paddr; + void *dma_vaddr; + unsigned int length; +}; + +/** + * fsl_asrc_pair: ASRC Pair private data + * + * @asrc_priv: pointer to its parent module + * @config: configuration profile + * @error: error record + * @index: pair index (ASRC_PAIR_A, ASRC_PAIR_B, ASRC_PAIR_C) + * @channels: occupied channel number + * @desc: input and output dma descriptors + * @dma_chan: inputer and output DMA channels + * @dma_data: private dma data + * @pos: hardware pointer position + * @private: pair private area + */ +struct fsl_asrc_pair { + struct fsl_asrc *asrc_priv; + struct asrc_config *config; + unsigned int error; + + enum asrc_pair_index index; + unsigned int channels; + + struct dma_async_tx_descriptor *desc[2]; + struct dma_chan *dma_chan[2]; + struct imx_dma_data dma_data; + unsigned int pos; + + void *private; +}; + +/** + * fsl_asrc_pair: ASRC private data + * + * @dma_params_rx: DMA parameters for receive channel + * @dma_params_tx: DMA parameters for transmit channel + * @pdev: platform device pointer + * @regmap: regmap handler + * @paddr: physical address to the base address of registers + * @mem_clk: clock source to access register + * @ipg_clk: clock source to drive peripheral + * @asrck_clk: clock sources to driver ASRC internal logic + * @lock: spin lock for resource protection + * @pair: pair pointers + * @channel_bits: width of ASRCNCR register for each pair + * @channel_avail: non-occupied channel numbers + * @asrc_rate: default sample rate for ASoC Back-Ends + * @asrc_width: default sample width for ASoC Back-Ends + * @name: driver name + */ +struct fsl_asrc { + struct snd_dmaengine_dai_dma_data dma_params_rx; + struct snd_dmaengine_dai_dma_data dma_params_tx; + struct platform_device *pdev; + struct regmap *regmap; + unsigned long paddr; + struct clk *mem_clk; + struct clk *ipg_clk; + struct clk *asrck_clk[ASRC_CLK_MAX_NUM]; + spinlock_t lock; + + struct fsl_asrc_pair *pair[ASRC_PAIR_MAX_NUM]; + unsigned int channel_bits; + unsigned int channel_avail; + + int asrc_rate; + int asrc_width; + + char name[32]; +}; + +extern struct snd_soc_platform_driver fsl_asrc_platform; +struct dma_chan *fsl_asrc_get_dma_channel(struct fsl_asrc_pair *pair, bool dir); +#endif /* _FSL_ASRC_H */ diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c new file mode 100644 index 000000000000..5b1e73e97817 --- /dev/null +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -0,0 +1,386 @@ +/* + * Freescale ASRC ALSA SoC Platform (DMA) driver + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#include +#include +#include +#include +#include + +#include "fsl_asrc.h" + +#define FSL_ASRC_DMABUF_SIZE (256 * 1024) + +static struct snd_pcm_hardware snd_imx_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .buffer_bytes_max = FSL_ASRC_DMABUF_SIZE, + .period_bytes_min = 128, + .period_bytes_max = 65535, /* Limited by SDMA engine */ + .periods_min = 2, + .periods_max = 255, + .fifo_size = 0, +}; + +static bool filter(struct dma_chan *chan, void *param) +{ + if (!imx_dma_is_general_purpose(chan)) + return false; + + chan->private = param; + + return true; +} + +static void fsl_asrc_dma_complete(void *arg) +{ + struct snd_pcm_substream *substream = arg; + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_asrc_pair *pair = runtime->private_data; + + pair->pos += snd_pcm_lib_period_bytes(substream); + if (pair->pos >= snd_pcm_lib_buffer_bytes(substream)) + pair->pos = 0; + + snd_pcm_period_elapsed(substream); +} + +static int fsl_asrc_dma_prepare_and_submit(struct snd_pcm_substream *substream) +{ + u8 dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? OUT : IN; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_asrc_pair *pair = runtime->private_data; + struct device *dev = rtd->platform->dev; + unsigned long flags = DMA_CTRL_ACK; + + /* Prepare and submit Front-End DMA channel */ + if (!substream->runtime->no_period_wakeup) + flags |= DMA_PREP_INTERRUPT; + + pair->pos = 0; + pair->desc[!dir] = dmaengine_prep_dma_cyclic( + pair->dma_chan[!dir], runtime->dma_addr, + snd_pcm_lib_buffer_bytes(substream), + snd_pcm_lib_period_bytes(substream), + dir == OUT ? DMA_TO_DEVICE : DMA_FROM_DEVICE, flags); + if (!pair->desc[!dir]) { + dev_err(dev, "failed to prepare slave DMA for Front-End\n"); + return -ENOMEM; + } + + pair->desc[!dir]->callback = fsl_asrc_dma_complete; + pair->desc[!dir]->callback_param = substream; + + dmaengine_submit(pair->desc[!dir]); + + /* Prepare and submit Back-End DMA channel */ + pair->desc[dir] = dmaengine_prep_dma_cyclic( + pair->dma_chan[dir], 0xffff, 64, 64, DMA_DEV_TO_DEV, 0); + if (!pair->desc[dir]) { + dev_err(dev, "failed to prepare slave DMA for Back-End\n"); + return -ENOMEM; + } + + dmaengine_submit(pair->desc[dir]); + + return 0; +} + +static int fsl_asrc_dma_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_asrc_pair *pair = runtime->private_data; + int ret; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ret = fsl_asrc_dma_prepare_and_submit(substream); + if (ret) + return ret; + dma_async_issue_pending(pair->dma_chan[IN]); + dma_async_issue_pending(pair->dma_chan[OUT]); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + dmaengine_terminate_all(pair->dma_chan[OUT]); + dmaengine_terminate_all(pair->dma_chan[IN]); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int fsl_asrc_dma_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + enum dma_slave_buswidth buswidth = DMA_SLAVE_BUSWIDTH_2_BYTES; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct snd_dmaengine_dai_dma_data *dma_params_fe = NULL; + struct snd_dmaengine_dai_dma_data *dma_params_be = NULL; + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_asrc_pair *pair = runtime->private_data; + struct fsl_asrc *asrc_priv = pair->asrc_priv; + struct dma_slave_config config_fe, config_be; + enum asrc_pair_index index = pair->index; + struct device *dev = rtd->platform->dev; + int stream = substream->stream; + struct imx_dma_data *tmp_data; + struct snd_soc_dpcm *dpcm; + struct dma_chan *tmp_chan; + struct device *dev_be; + u8 dir = tx ? OUT : IN; + dma_cap_mask_t mask; + int ret; + + /* Fetch the Back-End dma_data from DPCM */ + list_for_each_entry(dpcm, &rtd->dpcm[stream].be_clients, list_be) { + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_pcm_substream *substream_be; + struct snd_soc_dai *dai = be->cpu_dai; + + if (dpcm->fe != rtd) + continue; + + substream_be = snd_soc_dpcm_get_substream(be, stream); + dma_params_be = snd_soc_dai_get_dma_data(dai, substream_be); + dev_be = dai->dev; + break; + } + + if (!dma_params_be) { + dev_err(dev, "failed to get the substream of Back-End\n"); + return -EINVAL; + } + + /* Override dma_data of the Front-End and config its dmaengine */ + dma_params_fe = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + dma_params_fe->addr = asrc_priv->paddr + REG_ASRDx(!dir, index); + dma_params_fe->maxburst = dma_params_be->maxburst; + + pair->dma_chan[!dir] = fsl_asrc_get_dma_channel(pair, !dir); + if (!pair->dma_chan[!dir]) { + dev_err(dev, "failed to request DMA channel\n"); + return -EINVAL; + } + + memset(&config_fe, 0, sizeof(config_fe)); + ret = snd_dmaengine_pcm_prepare_slave_config(substream, params, &config_fe); + if (ret) { + dev_err(dev, "failed to prepare DMA config for Front-End\n"); + return ret; + } + + ret = dmaengine_slave_config(pair->dma_chan[!dir], &config_fe); + if (ret) { + dev_err(dev, "failed to config DMA channel for Front-End\n"); + return ret; + } + + /* Request and config DMA channel for Back-End */ + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); + dma_cap_set(DMA_CYCLIC, mask); + + /* Get DMA request of Back-End */ + tmp_chan = dma_request_slave_channel(dev_be, tx ? "tx" : "rx"); + tmp_data = tmp_chan->private; + pair->dma_data.dma_request = tmp_data->dma_request; + dma_release_channel(tmp_chan); + + /* Get DMA request of Front-End */ + tmp_chan = fsl_asrc_get_dma_channel(pair, dir); + tmp_data = tmp_chan->private; + pair->dma_data.dma_request2 = tmp_data->dma_request; + pair->dma_data.peripheral_type = tmp_data->peripheral_type; + pair->dma_data.priority = tmp_data->priority; + dma_release_channel(tmp_chan); + + pair->dma_chan[dir] = dma_request_channel(mask, filter, &pair->dma_data); + if (!pair->dma_chan[dir]) { + dev_err(dev, "failed to request DMA channel for Back-End\n"); + return -EINVAL; + } + + if (asrc_priv->asrc_width == 16) + buswidth = DMA_SLAVE_BUSWIDTH_2_BYTES; + else + buswidth = DMA_SLAVE_BUSWIDTH_4_BYTES; + + config_be.direction = DMA_DEV_TO_DEV; + config_be.src_addr_width = buswidth; + config_be.src_maxburst = dma_params_be->maxburst; + config_be.dst_addr_width = buswidth; + config_be.dst_maxburst = dma_params_be->maxburst; + + if (tx) { + config_be.src_addr = asrc_priv->paddr + REG_ASRDO(index); + config_be.dst_addr = dma_params_be->addr; + } else { + config_be.dst_addr = asrc_priv->paddr + REG_ASRDI(index); + config_be.src_addr = dma_params_be->addr; + } + + ret = dmaengine_slave_config(pair->dma_chan[dir], &config_be); + if (ret) { + dev_err(dev, "failed to config DMA channel for Back-End\n"); + return ret; + } + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + + return 0; +} + +static int fsl_asrc_dma_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_asrc_pair *pair = runtime->private_data; + + snd_pcm_set_runtime_buffer(substream, NULL); + + if (pair->dma_chan[IN]) + dma_release_channel(pair->dma_chan[IN]); + + if (pair->dma_chan[OUT]) + dma_release_channel(pair->dma_chan[OUT]); + + pair->dma_chan[IN] = NULL; + pair->dma_chan[OUT] = NULL; + + return 0; +} + +static int fsl_asrc_dma_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct device *dev = rtd->platform->dev; + struct fsl_asrc *asrc_priv = dev_get_drvdata(dev); + struct fsl_asrc_pair *pair; + + pair = kzalloc(sizeof(struct fsl_asrc_pair), GFP_KERNEL); + if (!pair) { + dev_err(dev, "failed to allocate pair\n"); + return -ENOMEM; + } + + pair->asrc_priv = asrc_priv; + + runtime->private_data = pair; + + snd_pcm_hw_constraint_integer(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware); + + return 0; +} + +static int fsl_asrc_dma_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_asrc_pair *pair = runtime->private_data; + struct fsl_asrc *asrc_priv = pair->asrc_priv; + + if (pair && asrc_priv->pair[pair->index] == pair) + asrc_priv->pair[pair->index] = NULL; + + kfree(pair); + + return 0; +} + +static snd_pcm_uframes_t fsl_asrc_dma_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsl_asrc_pair *pair = runtime->private_data; + + return bytes_to_frames(substream->runtime, pair->pos); +} + +static struct snd_pcm_ops fsl_asrc_dma_pcm_ops = { + .ioctl = snd_pcm_lib_ioctl, + .hw_params = fsl_asrc_dma_hw_params, + .hw_free = fsl_asrc_dma_hw_free, + .trigger = fsl_asrc_dma_trigger, + .open = fsl_asrc_dma_startup, + .close = fsl_asrc_dma_shutdown, + .pointer = fsl_asrc_dma_pcm_pointer, +}; + +static int fsl_asrc_dma_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + struct snd_pcm_substream *substream; + struct snd_pcm *pcm = rtd->pcm; + int ret, i; + + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); + if (ret) { + dev_err(card->dev, "failed to set DMA mask\n"); + return ret; + } + + for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_LAST; i++) { + substream = pcm->streams[i].substream; + if (!substream) + continue; + + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->card->dev, + FSL_ASRC_DMABUF_SIZE, &substream->dma_buffer); + if (ret) { + dev_err(card->dev, "failed to allocate DMA buffer\n"); + goto err; + } + } + + return 0; + +err: + if (--i == 0 && pcm->streams[i].substream) + snd_dma_free_pages(&pcm->streams[i].substream->dma_buffer); + + return ret; +} + +static void fsl_asrc_dma_pcm_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + int i; + + for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_LAST; i++) { + substream = pcm->streams[i].substream; + if (!substream) + continue; + + snd_dma_free_pages(&substream->dma_buffer); + substream->dma_buffer.area = NULL; + substream->dma_buffer.addr = 0; + } +} + +struct snd_soc_platform_driver fsl_asrc_platform = { + .ops = &fsl_asrc_dma_pcm_ops, + .pcm_new = fsl_asrc_dma_pcm_new, + .pcm_free = fsl_asrc_dma_pcm_free, +}; +EXPORT_SYMBOL_GPL(fsl_asrc_platform); -- cgit v1.2.3 From 71864b22f50ceab0025ae2a1cd81b956878ded2b Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Sun, 20 Jul 2014 11:43:07 +0800 Subject: ASoC: samsung: Fix return value check in s3c2412_iis_dev_probe() In case of error, the function devm_ioremap_resource() returns ERR_PTR() and never returns NULL. The NULL test in the return value check should be replaced with IS_ERR(). Also remove redundant return value check of platform_get_resource(). Signed-off-by: Wei Yongjun Signed-off-by: Mark Brown --- sound/soc/samsung/s3c2412-i2s.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 9180310e862a..27b339c6580e 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -154,13 +154,9 @@ static int s3c2412_iis_dev_probe(struct platform_device *pdev) struct resource *res; res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) { - dev_err(&pdev->dev, "Can't get IO resource.\n"); - return -ENOENT; - } s3c2412_i2s.regs = devm_ioremap_resource(&pdev->dev, res); - if (s3c2412_i2s.regs == NULL) - return -ENXIO; + if (IS_ERR(s3c2412_i2s.regs)) + return PTR_ERR(s3c2412_i2s.regs); s3c2412_i2s_pcm_stereo_out.dma_addr = res->start + S3C2412_IISTXD; s3c2412_i2s_pcm_stereo_in.dma_addr = res->start + S3C2412_IISRXD; -- cgit v1.2.3 From d3dacda9390b936a1c341d868f548944cc1c70de Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Tue, 29 Jul 2014 16:10:16 -0300 Subject: ASoC: fsl_asrc: Use 'ifdef' for config options Fix the following build errors that were observed by building with make ARCH=microblaze allyesconfig: >> sound/soc/fsl/fsl_asrc.c:906:5: warning: "CONFIG_PM_RUNTIME" is not defined [-Wundef] #if CONFIG_PM_RUNTIME ^ >> sound/soc/fsl/fsl_asrc.c:934:5: warning: "CONFIG_PM_SLEEP" is not defined [-Wundef] #if CONFIG_PM_SLEEP ^ >> sound/soc/fsl/fsl_asrc.c:906:5: warning: "CONFIG_PM_RUNTIME" is not defined [-Wundef] #if CONFIG_PM_RUNTIME ^ >> sound/soc/fsl/fsl_asrc.c:934:5: warning: "CONFIG_PM_SLEEP" is not defined [-Wundef] #if CONFIG_PM_SLEEP Reported-by: kbuild test robot Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_asrc.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 27a4a7084343..b9a288877267 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -903,7 +903,7 @@ static int fsl_asrc_probe(struct platform_device *pdev) return 0; } -#if CONFIG_PM_RUNTIME +#ifdef CONFIG_PM_RUNTIME static int fsl_asrc_runtime_resume(struct device *dev) { struct fsl_asrc *asrc_priv = dev_get_drvdata(dev); @@ -931,7 +931,7 @@ static int fsl_asrc_runtime_suspend(struct device *dev) } #endif /* CONFIG_PM_RUNTIME */ -#if CONFIG_PM_SLEEP +#ifdef CONFIG_PM_SLEEP static int fsl_asrc_suspend(struct device *dev) { struct fsl_asrc *asrc_priv = dev_get_drvdata(dev); -- cgit v1.2.3 From 08074dc1ae8fe5e9b40b6dc26cbee4b04337750a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 29 Jul 2014 20:13:00 +0100 Subject: ASoC: tlv320aic23: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic23.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 686b8b85b956..d67167920c2f 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -364,16 +364,16 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, iface_reg = snd_soc_read(codec, TLV320AIC23_DIGT_FMT) & ~(0x03 << 2); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: iface_reg |= (0x01 << 2); break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: iface_reg |= (0x02 << 2); break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: iface_reg |= (0x03 << 2); break; } -- cgit v1.2.3 From 270a85c80c8645368f9768b120bd9eac4db23590 Mon Sep 17 00:00:00 2001 From: Andrew Lunn Date: Thu, 10 Jul 2014 23:36:23 +0200 Subject: ASoC: kirkwood: Remove unused drivers Both kirkwood-openrd and kirkwood-t5325 drivers have been replaced with DT based simple-card equivelents. So remove these drivers. Signed-off-by: Andrew Lunn Acked-by: Jason Cooper Signed-off-by: Mark Brown --- sound/soc/kirkwood/Kconfig | 17 ----- sound/soc/kirkwood/Makefile | 4 -- sound/soc/kirkwood/kirkwood-openrd.c | 109 -------------------------------- sound/soc/kirkwood/kirkwood-t5325.c | 116 ----------------------------------- 4 files changed, 246 deletions(-) delete mode 100644 sound/soc/kirkwood/kirkwood-openrd.c delete mode 100644 sound/soc/kirkwood/kirkwood-t5325.c (limited to 'sound') diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig index 06f4e8aa93ae..1f7c7ee3527a 100644 --- a/sound/soc/kirkwood/Kconfig +++ b/sound/soc/kirkwood/Kconfig @@ -15,20 +15,3 @@ config SND_KIRKWOOD_SOC_ARMADA370_DB Say Y if you want to add support for SoC audio on the Armada 370 Development Board. -config SND_KIRKWOOD_SOC_OPENRD - tristate "SoC Audio support for Kirkwood Openrd Client" - depends on SND_KIRKWOOD_SOC && (MACH_OPENRD_CLIENT || MACH_OPENRD_ULTIMATE || COMPILE_TEST) - depends on I2C - select SND_SOC_CS42L51 - help - Say Y if you want to add support for SoC audio on - Openrd Client. - -config SND_KIRKWOOD_SOC_T5325 - tristate "SoC Audio support for HP t5325" - depends on SND_KIRKWOOD_SOC && (MACH_T5325 || COMPILE_TEST) && I2C - select SND_SOC_ALC5623 - help - Say Y if you want to add support for SoC audio on - the HP t5325 thin client. - diff --git a/sound/soc/kirkwood/Makefile b/sound/soc/kirkwood/Makefile index 7c1d8fe09e6b..c36b03d8006c 100644 --- a/sound/soc/kirkwood/Makefile +++ b/sound/soc/kirkwood/Makefile @@ -2,10 +2,6 @@ snd-soc-kirkwood-objs := kirkwood-dma.o kirkwood-i2s.o obj-$(CONFIG_SND_KIRKWOOD_SOC) += snd-soc-kirkwood.o -snd-soc-openrd-objs := kirkwood-openrd.o -snd-soc-t5325-objs := kirkwood-t5325.o snd-soc-armada-370-db-objs := armada-370-db.o obj-$(CONFIG_SND_KIRKWOOD_SOC_ARMADA370_DB) += snd-soc-armada-370-db.o -obj-$(CONFIG_SND_KIRKWOOD_SOC_OPENRD) += snd-soc-openrd.o -obj-$(CONFIG_SND_KIRKWOOD_SOC_T5325) += snd-soc-t5325.o diff --git a/sound/soc/kirkwood/kirkwood-openrd.c b/sound/soc/kirkwood/kirkwood-openrd.c deleted file mode 100644 index 65f2a5b9ec3b..000000000000 --- a/sound/soc/kirkwood/kirkwood-openrd.c +++ /dev/null @@ -1,109 +0,0 @@ -/* - * kirkwood-openrd.c - * - * (c) 2010 Arnaud Patard - * (c) 2010 Arnaud Patard - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ - -#include -#include -#include -#include -#include -#include -#include -#include "../codecs/cs42l51.h" - -static int openrd_client_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - unsigned int freq; - - switch (params_rate(params)) { - default: - case 44100: - freq = 11289600; - break; - case 48000: - freq = 12288000; - break; - case 96000: - freq = 24576000; - break; - } - - return snd_soc_dai_set_sysclk(codec_dai, 0, freq, SND_SOC_CLOCK_IN); - -} - -static struct snd_soc_ops openrd_client_ops = { - .hw_params = openrd_client_hw_params, -}; - - -static struct snd_soc_dai_link openrd_client_dai[] = { -{ - .name = "CS42L51", - .stream_name = "CS42L51 HiFi", - .cpu_dai_name = "i2s", - .platform_name = "mvebu-audio", - .codec_dai_name = "cs42l51-hifi", - .codec_name = "cs42l51-codec.0-004a", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, - .ops = &openrd_client_ops, -}, -}; - - -static struct snd_soc_card openrd_client = { - .name = "OpenRD Client", - .owner = THIS_MODULE, - .dai_link = openrd_client_dai, - .num_links = ARRAY_SIZE(openrd_client_dai), -}; - -static int openrd_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &openrd_client; - int ret; - - card->dev = &pdev->dev; - - ret = snd_soc_register_card(card); - if (ret) - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", - ret); - return ret; -} - -static int openrd_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - return 0; -} - -static struct platform_driver openrd_driver = { - .driver = { - .name = "openrd-client-audio", - .owner = THIS_MODULE, - }, - .probe = openrd_probe, - .remove = openrd_remove, -}; - -module_platform_driver(openrd_driver); - -/* Module information */ -MODULE_AUTHOR("Arnaud Patard "); -MODULE_DESCRIPTION("ALSA SoC OpenRD Client"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:openrd-client-audio"); diff --git a/sound/soc/kirkwood/kirkwood-t5325.c b/sound/soc/kirkwood/kirkwood-t5325.c deleted file mode 100644 index 844b8415a011..000000000000 --- a/sound/soc/kirkwood/kirkwood-t5325.c +++ /dev/null @@ -1,116 +0,0 @@ -/* - * kirkwood-t5325.c - * - * (c) 2010 Arnaud Patard - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ - -#include -#include -#include -#include -#include -#include -#include -#include "../codecs/alc5623.h" - -static int t5325_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - unsigned int freq; - - freq = params_rate(params) * 256; - - return snd_soc_dai_set_sysclk(codec_dai, 0, freq, SND_SOC_CLOCK_IN); - -} - -static struct snd_soc_ops t5325_ops = { - .hw_params = t5325_hw_params, -}; - -static const struct snd_soc_dapm_widget t5325_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_SPK("Speaker", NULL), - SND_SOC_DAPM_MIC("Mic Jack", NULL), -}; - -static const struct snd_soc_dapm_route t5325_route[] = { - { "Headphone Jack", NULL, "HPL" }, - { "Headphone Jack", NULL, "HPR" }, - - {"Speaker", NULL, "SPKOUT"}, - {"Speaker", NULL, "SPKOUTN"}, - - { "MIC1", NULL, "Mic Jack" }, - { "MIC2", NULL, "Mic Jack" }, -}; - -static struct snd_soc_dai_link t5325_dai[] = { -{ - .name = "ALC5621", - .stream_name = "ALC5621 HiFi", - .cpu_dai_name = "i2s", - .platform_name = "mvebu-audio", - .codec_dai_name = "alc5621-hifi", - .codec_name = "alc562x-codec.0-001a", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, - .ops = &t5325_ops, -}, -}; - -static struct snd_soc_card t5325 = { - .name = "t5325", - .owner = THIS_MODULE, - .dai_link = t5325_dai, - .num_links = ARRAY_SIZE(t5325_dai), - - .dapm_widgets = t5325_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(t5325_dapm_widgets), - .dapm_routes = t5325_route, - .num_dapm_routes = ARRAY_SIZE(t5325_route), -}; - -static int t5325_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &t5325; - int ret; - - card->dev = &pdev->dev; - - ret = snd_soc_register_card(card); - if (ret) - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", - ret); - return ret; -} - -static int t5325_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - return 0; -} - -static struct platform_driver t5325_driver = { - .driver = { - .name = "t5325-audio", - .owner = THIS_MODULE, - }, - .probe = t5325_probe, - .remove = t5325_remove, -}; - -module_platform_driver(t5325_driver); - -MODULE_AUTHOR("Arnaud Patard "); -MODULE_DESCRIPTION("ALSA SoC t5325 audio client"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:t5325-audio"); -- cgit v1.2.3 From 6145dfc6d96a350f50b939d8b5b5301e17f4c2c5 Mon Sep 17 00:00:00 2001 From: Andrew Lunn Date: Thu, 10 Jul 2014 23:36:24 +0200 Subject: ASoC: kirkwood: Remove ARCH_KIRKWOOD dependency mach-kirkwood has been removed, now that kirkwood lives in mach-mvebu. Remove ARCH_KIRKWOOD since ARCH_MVEBU is sufficient. Signed-off-by: Andrew Lunn Acked-by: Jason Cooper Signed-off-by: Mark Brown --- sound/soc/kirkwood/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig index 1f7c7ee3527a..132bb83f8e99 100644 --- a/sound/soc/kirkwood/Kconfig +++ b/sound/soc/kirkwood/Kconfig @@ -1,6 +1,6 @@ config SND_KIRKWOOD_SOC tristate "SoC Audio for the Marvell Kirkwood and Dove chips" - depends on ARCH_KIRKWOOD || ARCH_DOVE || ARCH_MVEBU || MACH_KIRKWOOD || COMPILE_TEST + depends on ARCH_DOVE || ARCH_MVEBU || COMPILE_TEST help Say Y or M if you want to add support for codecs attached to the Kirkwood I2S interface. You will also need to select the -- cgit v1.2.3 From ee4a6ce6cd74a9eace247656c5b109f31c73ab8d Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Wed, 30 Jul 2014 20:05:44 +0800 Subject: ASoC: Intel: Fix naming of HMDC register macros. HMDC is the correct naming for this register. Signed-off-by: Liam Girdwood Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/sst-dsp.h | 8 ++++---- sound/soc/intel/sst-haswell-dsp.c | 6 +++--- 2 files changed, 7 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h index 967fb32c981d..21a85eb196f1 100644 --- a/sound/soc/intel/sst-dsp.h +++ b/sound/soc/intel/sst-dsp.h @@ -52,7 +52,7 @@ #define SST_CLKCTL 0x78 #define SST_CSR2 0x80 #define SST_LTRC 0xE0 -#define SST_HDMC 0xE8 +#define SST_HMDC 0xE8 #define SST_SHIM_BEGIN SST_CSR #define SST_SHIM_END SST_HDMC @@ -122,9 +122,9 @@ /* LTRC */ #define SST_LTRC_VAL(x) (x << 0) -/* HDMC */ -#define SST_HDMC_HDDA0(x) (x << 0) -#define SST_HDMC_HDDA1(x) (x << 7) +/* HMDC */ +#define SST_HMDC_HDDA0(x) (x << 0) +#define SST_HMDC_HDDA1(x) (x << 7) /* SST Vendor Defined Registers and bits */ diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c index 7b8ad9923fe4..0e1dde8c35e6 100644 --- a/sound/soc/intel/sst-haswell-dsp.c +++ b/sound/soc/intel/sst-haswell-dsp.c @@ -269,9 +269,9 @@ static void hsw_boot(struct sst_dsp *sst) SST_CSR2_SDFD_SSP1); /* enable DMA engine 0,1 all channels to access host memory */ - sst_dsp_shim_update_bits_unlocked(sst, SST_HDMC, - SST_HDMC_HDDA1(0xff) | SST_HDMC_HDDA0(0xff), - SST_HDMC_HDDA1(0xff) | SST_HDMC_HDDA0(0xff)); + sst_dsp_shim_update_bits_unlocked(sst, SST_HMDC, + SST_HMDC_HDDA1(0xff) | SST_HMDC_HDDA0(0xff), + SST_HMDC_HDDA1(0xff) | SST_HMDC_HDDA0(0xff)); /* disable all clock gating */ writel(0x0, sst->addr.pci_cfg + SST_VDRTCTL2); -- cgit v1.2.3 From d7d7d1eda0a2baaa4f6e02f0d58e81ea71dcbea2 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Wed, 30 Jul 2014 20:08:18 +0800 Subject: ASoC: Intel: Add macros for SST shim register bits. Add some register definitions for other shim register bits. Signed-off-by: Liam Girdwood Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/sst-dsp.h | 27 +++++++++++++++++++++++++++ 1 file changed, 27 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h index 21a85eb196f1..3165dfa97408 100644 --- a/sound/soc/intel/sst-dsp.h +++ b/sound/soc/intel/sst-dsp.h @@ -77,6 +77,8 @@ #define SST_CSR_S0IOCS (0x1 << 21) #define SST_CSR_S1IOCS (0x1 << 23) #define SST_CSR_LPCS (0x1 << 31) +#define SST_CSR_24MHZ_LPCS (SST_CSR_SBCS0 | SST_CSR_SBCS1 | SST_CSR_LPCS) +#define SST_CSR_24MHZ_NO_LPCS (SST_CSR_SBCS0 | SST_CSR_SBCS1) #define SST_BYT_CSR_RST (0x1 << 0) #define SST_BYT_CSR_VECTOR_SEL (0x1 << 1) #define SST_BYT_CSR_STALL (0x1 << 2) @@ -96,6 +98,14 @@ #define SST_IMRX_DONE (0x1 << 0) #define SST_BYT_IMRX_REQUEST (0x1 << 1) +/* IMRD / IMD */ +#define SST_IMRD_DONE (0x1 << 0) +#define SST_IMRD_BUSY (0x1 << 1) +#define SST_IMRD_SSP0 (0x1 << 16) +#define SST_IMRD_DMAC0 (0x1 << 21) +#define SST_IMRD_DMAC1 (0x1 << 22) +#define SST_IMRD_DMAC (SST_IMRD_DMAC0 | SST_IMRD_DMAC1) + /* IPCX / IPCC */ #define SST_IPCX_DONE (0x1 << 30) #define SST_IPCX_BUSY (0x1 << 31) @@ -125,6 +135,18 @@ /* HMDC */ #define SST_HMDC_HDDA0(x) (x << 0) #define SST_HMDC_HDDA1(x) (x << 7) +#define SST_HMDC_HDDA_E0_CH0 1 +#define SST_HMDC_HDDA_E0_CH1 2 +#define SST_HMDC_HDDA_E0_CH2 4 +#define SST_HMDC_HDDA_E0_CH3 8 +#define SST_HMDC_HDDA_E1_CH0 SST_HMDC_HDDA1(SST_HMDC_HDDA_E0_CH0) +#define SST_HMDC_HDDA_E1_CH1 SST_HMDC_HDDA1(SST_HMDC_HDDA_E0_CH1) +#define SST_HMDC_HDDA_E1_CH2 SST_HMDC_HDDA1(SST_HMDC_HDDA_E0_CH2) +#define SST_HMDC_HDDA_E1_CH3 SST_HMDC_HDDA1(SST_HMDC_HDDA_E0_CH3) +#define SST_HMDC_HDDA_E0_ALLCH (SST_HMDC_HDDA_E0_CH0 | SST_HMDC_HDDA_E0_CH1 | \ + SST_HMDC_HDDA_E0_CH2 | SST_HMDC_HDDA_E0_CH3) +#define SST_HMDC_HDDA_E1_ALLCH (SST_HMDC_HDDA_E1_CH0 | SST_HMDC_HDDA_E1_CH1 | \ + SST_HMDC_HDDA_E1_CH2 | SST_HMDC_HDDA_E1_CH3) /* SST Vendor Defined Registers and bits */ @@ -134,11 +156,16 @@ #define SST_VDRTCTL3 0xaC /* VDRTCTL0 */ +#define SST_VDRTCL0_APLLSE_MASK 1 #define SST_VDRTCL0_DSRAMPGE_SHIFT 16 #define SST_VDRTCL0_DSRAMPGE_MASK (0xffff << SST_VDRTCL0_DSRAMPGE_SHIFT) #define SST_VDRTCL0_ISRAMPGE_SHIFT 6 #define SST_VDRTCL0_ISRAMPGE_MASK (0x3ff << SST_VDRTCL0_ISRAMPGE_SHIFT) +/* PMCS */ +#define SST_PMCS 0x84 +#define SST_PMCS_PS_MASK 0x3 + struct sst_dsp; /* -- cgit v1.2.3 From 81552612501f436f3824f056f95fdc04b8a60e1f Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Wed, 30 Jul 2014 20:09:47 +0800 Subject: ASoC: Intel: Add notification trace for reset. Add trace notification of IPC stream reset. Signed-off-by: Liam Girdwood Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-ipc.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 96373ab46f8c..436c2fa23c80 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -687,7 +687,9 @@ static void hsw_stream_update(struct sst_hsw *hsw, struct ipc_message *msg) switch (stream_msg) { case IPC_STR_STAGE_MESSAGE: case IPC_STR_NOTIFICATION: + break; case IPC_STR_RESET: + trace_ipc_notification("stream reset", stream->reply.stream_hw_id); break; case IPC_STR_PAUSE: stream->running = false; -- cgit v1.2.3 From 543ec637e00a9000772c315a8c98fa6ede563c5b Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Wed, 30 Jul 2014 20:11:26 +0800 Subject: ASoC: Intel: Add debug to set DX state Add some debugging info to help with Dx state debug. Signed-off-by: Liam Girdwood Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-ipc.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 436c2fa23c80..9825d195b8c9 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -1646,7 +1646,7 @@ int sst_hsw_dx_set_state(struct sst_hsw *hsw, enum sst_hsw_dx_state state, struct sst_hsw_ipc_dx_reply *dx) { u32 header, state_; - int ret; + int ret, item; header = IPC_GLB_TYPE(IPC_GLB_ENTER_DX_STATE); state_ = state; @@ -1660,6 +1660,13 @@ int sst_hsw_dx_set_state(struct sst_hsw *hsw, return ret; } + for (item = 0; item < dx->entries_no; item++) { + dev_dbg(hsw->dev, + "Item[%d] offset[%x] - size[%x] - source[%x]\n", + item, dx->mem_info[item].offset, + dx->mem_info[item].size, + dx->mem_info[item].source); + } dev_dbg(hsw->dev, "ipc: got %d entry numbers for state %d\n", dx->entries_no, state); -- cgit v1.2.3 From 30bba67c43c0dd36696d1209c19afc3f25f2f3f3 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Wed, 30 Jul 2014 20:18:32 +0800 Subject: ASoC: Intel: Check ops before we derefference pointers. Check ops pointer members before we can derefference them. Signed-off-by: Liam Girdwood Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/sst-dsp.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-dsp.c b/sound/soc/intel/sst-dsp.c index 0b715b20a2d7..cd23060a0d86 100644 --- a/sound/soc/intel/sst-dsp.c +++ b/sound/soc/intel/sst-dsp.c @@ -224,19 +224,23 @@ EXPORT_SYMBOL_GPL(sst_dsp_shim_update_bits64); void sst_dsp_dump(struct sst_dsp *sst) { - sst->ops->dump(sst); + if (sst->ops->dump) + sst->ops->dump(sst); } EXPORT_SYMBOL_GPL(sst_dsp_dump); void sst_dsp_reset(struct sst_dsp *sst) { - sst->ops->reset(sst); + if (sst->ops->reset) + sst->ops->reset(sst); } EXPORT_SYMBOL_GPL(sst_dsp_reset); int sst_dsp_boot(struct sst_dsp *sst) { - sst->ops->boot(sst); + if (sst->ops->boot) + sst->ops->boot(sst); + return 0; } EXPORT_SYMBOL_GPL(sst_dsp_boot); -- cgit v1.2.3 From bdb9eb49671566afb9ca2025752f57d0e1a6b2a3 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 30 Jul 2014 11:27:06 -0300 Subject: ASoC: fsl: fsl_asrc: Select SND_SOC_GENERIC_DMAENGINE_PCM Building a kernel with SND_SOC_GENERIC_DMAENGINE_PCM=n leads to the following error: ERROR: "snd_dmaengine_pcm_prepare_slave_config" [sound/soc/fsl/snd-soc-fsl-asrc.ko] undefined! Let SND_SOC_FSL_ASRC select SND_SOC_GENERIC_DMAENGINE_PCM in order to fix such error. Reported-by: kbuild test robot Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 56fd32b3a881..0ec1b7a42910 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -5,6 +5,7 @@ comment "Common SoC Audio options for Freescale CPUs:" config SND_SOC_FSL_ASRC tristate "Asynchronous Sample Rate Converter (ASRC) module support" select REGMAP_MMIO + select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y if you want to add Asynchronous Sample Rate Converter (ASRC) support for the Freescale CPUs. -- cgit v1.2.3 From 0caf3eb7213a14b44977d1ca4b69d7f3d7fd474a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:28:13 +0100 Subject: ASoC: 88pm860x: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown --- sound/soc/codecs/88pm860x-codec.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 3c4b10ff48c1..922006dd0583 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -945,11 +945,11 @@ static int pm860x_pcm_hw_params(struct snd_pcm_substream *substream, unsigned char inf = 0, mask = 0; /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: inf &= ~PCM_INF2_18WL; break; - case SNDRV_PCM_FORMAT_S18_3LE: + case 18: inf |= PCM_INF2_18WL; break; default: @@ -1044,11 +1044,11 @@ static int pm860x_i2s_hw_params(struct snd_pcm_substream *substream, unsigned char inf; /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: inf = 0; break; - case SNDRV_PCM_FORMAT_S18_3LE: + case 18: inf = PCM_INF2_18WL; break; default: -- cgit v1.2.3 From d2a9b1c1f4a339b8f4baf04669e63b094a255bed Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:28:24 +0100 Subject: ASoC: adau17x1: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown Acked-by: Lars-Peter Clausen --- sound/soc/codecs/adau17x1.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 2961fae9670a..0b659704e60c 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -359,14 +359,14 @@ static int adau17x1_hw_params(struct snd_pcm_substream *substream, if (adau->dai_fmt != SND_SOC_DAIFMT_RIGHT_J) return 0; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: val = ADAU17X1_SERIAL_PORT1_DELAY16; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: val = ADAU17X1_SERIAL_PORT1_DELAY8; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: val = ADAU17X1_SERIAL_PORT1_DELAY0; break; default: -- cgit v1.2.3 From b05e16d21251e1ba45af58d4f662b0460c1bdc8c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:31:44 +0100 Subject: ASoC: max98088: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown --- sound/soc/codecs/max98088.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 9134982807b5..2cd3e5427441 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1299,12 +1299,12 @@ static int max98088_dai2_hw_params(struct snd_pcm_substream *substream, rate = params_rate(params); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: snd_soc_update_bits(codec, M98088_REG_1C_DAI2_FORMAT, M98088_DAI_WS, 0); break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: snd_soc_update_bits(codec, M98088_REG_1C_DAI2_FORMAT, M98088_DAI_WS, M98088_DAI_WS); break; -- cgit v1.2.3 From 1ae1f3a20016c202f0570306ca73942d7b1871d6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:31:56 +0100 Subject: ASoC: max98095: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown --- sound/soc/codecs/max98095.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 89ec00424880..0ee6797d5083 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -1280,12 +1280,12 @@ static int max98095_dai2_hw_params(struct snd_pcm_substream *substream, rate = params_rate(params); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: snd_soc_update_bits(codec, M98095_034_DAI2_FORMAT, M98095_DAI_WS, 0); break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: snd_soc_update_bits(codec, M98095_034_DAI2_FORMAT, M98095_DAI_WS, M98095_DAI_WS); break; @@ -1341,12 +1341,12 @@ static int max98095_dai3_hw_params(struct snd_pcm_substream *substream, rate = params_rate(params); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: snd_soc_update_bits(codec, M98095_03E_DAI3_FORMAT, M98095_DAI_WS, 0); break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: snd_soc_update_bits(codec, M98095_03E_DAI3_FORMAT, M98095_DAI_WS, M98095_DAI_WS); break; -- cgit v1.2.3 From e6777ead100d92db92d1e00b5e5ea4a688391719 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:32:12 +0100 Subject: ASoC: rt5631: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown --- sound/soc/codecs/rt5631.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index 30e234708579..1ba27db660a6 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -1370,16 +1370,16 @@ static int rt5631_hifi_pcm_params(struct snd_pcm_substream *substream, return coeff; } - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: iface |= RT5631_SDP_I2S_DL_20; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: iface |= RT5631_SDP_I2S_DL_24; break; - case SNDRV_PCM_FORMAT_S8: + case 8: iface |= RT5631_SDP_I2S_DL_8; break; default: -- cgit v1.2.3 From 794f33d2fb9823adbcd99d002ea3f9eb4379ae4d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:33:41 +0100 Subject: ASoC: rt5651: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown --- sound/soc/codecs/rt5651.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index ea4b1c652a26..bb0a3ab5416c 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -1366,16 +1366,16 @@ static int rt5651_hw_params(struct snd_pcm_substream *substream, dev_dbg(dai->dev, "bclk_ms is %d and pre_div is %d for iis %d\n", bclk_ms, pre_div, dai->id); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: val_len |= RT5651_I2S_DL_20; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: val_len |= RT5651_I2S_DL_24; break; - case SNDRV_PCM_FORMAT_S8: + case 8: val_len |= RT5651_I2S_DL_8; break; default: -- cgit v1.2.3 From dacc2aefcc8bf019af0d559561339dfef8829a10 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:46:05 +0100 Subject: ASoC: sgtl5000: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 3d39f0b5b4a8..a0e1f9c67ab2 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -724,25 +724,25 @@ static int sgtl5000_pcm_hw_params(struct snd_pcm_substream *substream, return ret; /* set i2s data format */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: if (sgtl5000->fmt == SND_SOC_DAIFMT_RIGHT_J) return -EINVAL; i2s_ctl |= SGTL5000_I2S_DLEN_16 << SGTL5000_I2S_DLEN_SHIFT; i2s_ctl |= SGTL5000_I2S_SCLKFREQ_32FS << SGTL5000_I2S_SCLKFREQ_SHIFT; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: i2s_ctl |= SGTL5000_I2S_DLEN_20 << SGTL5000_I2S_DLEN_SHIFT; i2s_ctl |= SGTL5000_I2S_SCLKFREQ_64FS << SGTL5000_I2S_SCLKFREQ_SHIFT; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: i2s_ctl |= SGTL5000_I2S_DLEN_24 << SGTL5000_I2S_DLEN_SHIFT; i2s_ctl |= SGTL5000_I2S_SCLKFREQ_64FS << SGTL5000_I2S_SCLKFREQ_SHIFT; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: if (sgtl5000->fmt == SND_SOC_DAIFMT_RIGHT_J) return -EINVAL; i2s_ctl |= SGTL5000_I2S_DLEN_32 << SGTL5000_I2S_DLEN_SHIFT; -- cgit v1.2.3 From 0a49f706be520e01cfb4dd17134358cd603d4d8a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:46:36 +0100 Subject: ASoC: si476x: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown --- sound/soc/codecs/si476x.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index f26befb0c297..cdf882fa7716 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -167,17 +167,17 @@ static int si476x_codec_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: + switch (params_width(params)) { + case 8: width = SI476X_PCM_FORMAT_S8; break; - case SNDRV_PCM_FORMAT_S16_LE: + case 16: width = SI476X_PCM_FORMAT_S16_LE; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: width = SI476X_PCM_FORMAT_S20_3LE; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: width = SI476X_PCM_FORMAT_S24_LE; break; default: -- cgit v1.2.3 From 9519dd4c1f6386943cb84ef71009b8d372edb3ac Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:46:50 +0100 Subject: ASoC: sn95031: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown --- sound/soc/codecs/sn95031.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 42dff26b3a2a..cf8fa40662f0 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -661,12 +661,12 @@ static int sn95031_pcm_hw_params(struct snd_pcm_substream *substream, { unsigned int format, rate; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: format = BIT(4)|BIT(5); break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: format = 0; break; default: -- cgit v1.2.3 From 737e0f89ed830cc48115140f8e494f5fe9dd7abe Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:47:24 +0100 Subject: ASoC: sta32x: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown --- sound/soc/codecs/sta32x.c | 19 ++++++------------- 1 file changed, 6 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 0579d187135b..48740855566d 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -678,15 +678,11 @@ static int sta32x_hw_params(struct snd_pcm_substream *substream, confb = snd_soc_read(codec, STA32X_CONFB); confb &= ~(STA32X_CONFB_SAI_MASK | STA32X_CONFB_SAIFB); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S24_LE: - case SNDRV_PCM_FORMAT_S24_BE: - case SNDRV_PCM_FORMAT_S24_3LE: - case SNDRV_PCM_FORMAT_S24_3BE: + switch (params_width(params)) { + case 24: pr_debug("24bit\n"); /* fall through */ - case SNDRV_PCM_FORMAT_S32_LE: - case SNDRV_PCM_FORMAT_S32_BE: + case 32: pr_debug("24bit or 32bit\n"); switch (sta32x->format) { case SND_SOC_DAIFMT_I2S: @@ -701,8 +697,7 @@ static int sta32x_hw_params(struct snd_pcm_substream *substream, } break; - case SNDRV_PCM_FORMAT_S20_3LE: - case SNDRV_PCM_FORMAT_S20_3BE: + case 20: pr_debug("20bit\n"); switch (sta32x->format) { case SND_SOC_DAIFMT_I2S: @@ -717,8 +712,7 @@ static int sta32x_hw_params(struct snd_pcm_substream *substream, } break; - case SNDRV_PCM_FORMAT_S18_3LE: - case SNDRV_PCM_FORMAT_S18_3BE: + case 18: pr_debug("18bit\n"); switch (sta32x->format) { case SND_SOC_DAIFMT_I2S: @@ -733,8 +727,7 @@ static int sta32x_hw_params(struct snd_pcm_substream *substream, } break; - case SNDRV_PCM_FORMAT_S16_LE: - case SNDRV_PCM_FORMAT_S16_BE: + case 16: pr_debug("16bit\n"); switch (sta32x->format) { case SND_SOC_DAIFMT_I2S: -- cgit v1.2.3 From 1f9ffcb0be08f83006f2d3c7e841d8ba7cdd1a9b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:47:32 +0100 Subject: ASoC: sta529: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown --- sound/soc/codecs/sta529.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c index a40c4b0196a3..7ece89b24674 100644 --- a/sound/soc/codecs/sta529.c +++ b/sound/soc/codecs/sta529.c @@ -197,16 +197,16 @@ static int sta529_hw_params(struct snd_pcm_substream *substream, int pdata, play_freq_val, record_freq_val; int bclk_to_fs_ratio; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: pdata = 1; bclk_to_fs_ratio = 0; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: pdata = 2; bclk_to_fs_ratio = 1; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: pdata = 3; bclk_to_fs_ratio = 2; break; -- cgit v1.2.3 From 560cfb14c6d7f219904ba4b81c0a581ffd7870ea Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:47:05 +0100 Subject: ASoC: ssm2518: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown Acked-by: Lars-Peter Clausen --- sound/soc/codecs/ssm2518.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c index 56adb3e2def9..e8680bea5f86 100644 --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -361,11 +361,11 @@ static int ssm2518_hw_params(struct snd_pcm_substream *substream, return -EINVAL; if (ssm2518->right_j) { - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: ctrl1 |= SSM2518_SAI_CTRL1_FMT_RJ_16BIT; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: ctrl1 |= SSM2518_SAI_CTRL1_FMT_RJ_24BIT; break; default: -- cgit v1.2.3 From 00a37032a099572302ae89c893bd5c90b924c537 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:47:13 +0100 Subject: ASoC: ssm2602: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown Acked-by: Lars-Peter Clausen --- sound/soc/codecs/ssm2602.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 97b0454eb346..484b3bbe8624 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -275,17 +275,17 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream, regmap_write(ssm2602->regmap, SSM2602_SRATE, srate); /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: iface = 0x0; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: iface = 0x4; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: iface = 0x8; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: iface = 0xc; break; default: -- cgit v1.2.3 From c665330c1951618d824b7789434b4de843681f07 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:47:45 +0100 Subject: ASoC: tas5086: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown --- sound/soc/codecs/tas5086.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index d48491a4a19d..9871cf374055 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -419,14 +419,14 @@ static int tas5086_hw_params(struct snd_pcm_substream *substream, } /* ... then add the offset for the sample bit depth. */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: val += 0; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: val += 1; break; - case SNDRV_PCM_FORMAT_S24_3LE: + case 24: val += 2; break; default: -- cgit v1.2.3 From 93d0ad8f374c59aa9f72493b8c5fcae303ca09ca Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:48:04 +0100 Subject: ASoC: tlv320aic26: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic26.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 43069de3d56a..620ab9ea1ef0 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -71,8 +71,8 @@ static int aic26_hw_params(struct snd_pcm_substream *substream, dev_dbg(&aic26->spi->dev, "aic26_hw_params(substream=%p, params=%p)\n", substream, params); - dev_dbg(&aic26->spi->dev, "rate=%i format=%i\n", params_rate(params), - params_format(params)); + dev_dbg(&aic26->spi->dev, "rate=%i width=%d\n", params_rate(params), + params_width(params)); switch (params_rate(params)) { case 8000: fsref = 48000; divisor = AIC26_DIV_6; break; @@ -89,11 +89,11 @@ static int aic26_hw_params(struct snd_pcm_substream *substream, } /* select data word length */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: wlen = AIC26_WLEN_16; break; - case SNDRV_PCM_FORMAT_S16_BE: wlen = AIC26_WLEN_16; break; - case SNDRV_PCM_FORMAT_S24_BE: wlen = AIC26_WLEN_24; break; - case SNDRV_PCM_FORMAT_S32_BE: wlen = AIC26_WLEN_32; break; + switch (params_width(params)) { + case 8: wlen = AIC26_WLEN_16; break; + case 16: wlen = AIC26_WLEN_16; break; + case 24: wlen = AIC26_WLEN_24; break; + case 32: wlen = AIC26_WLEN_32; break; default: dev_dbg(&aic26->spi->dev, "bad format\n"); return -EINVAL; } -- cgit v1.2.3 From 88be681b46216ef0eef9b82c35571185b95b6568 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:48:13 +0100 Subject: ASoC: tlv320aic31xx: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic31xx.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 23419109ecac..42f23f382c5d 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -759,8 +759,8 @@ static int aic31xx_hw_params(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; u8 data = 0; - dev_dbg(codec->dev, "## %s: format %d width %d rate %d\n", - __func__, params_format(params), params_width(params), + dev_dbg(codec->dev, "## %s: width %d rate %d\n", + __func__, params_width(params), params_rate(params)); switch (params_width(params)) { @@ -779,8 +779,8 @@ static int aic31xx_hw_params(struct snd_pcm_substream *substream, AIC31XX_IFACE1_DATALEN_SHIFT); break; default: - dev_err(codec->dev, "%s: Unsupported format %d\n", - __func__, params_format(params)); + dev_err(codec->dev, "%s: Unsupported width %d\n", + __func__, params_width(params)); return -EINVAL; } -- cgit v1.2.3 From bd8a571163368627013e9763b9b6d572ba4083b5 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:48:27 +0100 Subject: ASoC: tlv320aic32x4: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 1d9b117345a3..5d618e50cbf1 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -450,16 +450,16 @@ static int aic32x4_hw_params(struct snd_pcm_substream *substream, data = snd_soc_read(codec, AIC32X4_IFACE1); data = data & ~(3 << 4); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: data |= (AIC32X4_WORD_LEN_20BITS << AIC32X4_DOSRMSB_SHIFT); break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: data |= (AIC32X4_WORD_LEN_24BITS << AIC32X4_DOSRMSB_SHIFT); break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: data |= (AIC32X4_WORD_LEN_32BITS << AIC32X4_DOSRMSB_SHIFT); break; } -- cgit v1.2.3 From c60f23cb0a4b12091d92c24d365234968025a08e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:48:44 +0100 Subject: ASoC: tlv320dac33: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown Acked-by: Peter Ujfalusi --- sound/soc/codecs/tlv320dac33.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index df3a7506c023..34f8ff886c33 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -832,18 +832,18 @@ static int dac33_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: dac33->fifo_size = DAC33_FIFO_SIZE_16BIT; dac33->burst_rate = CALC_BURST_RATE(dac33->burst_bclkdiv, 32); break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: dac33->fifo_size = DAC33_FIFO_SIZE_24BIT; dac33->burst_rate = CALC_BURST_RATE(dac33->burst_bclkdiv, 64); break; default: - dev_err(codec->dev, "unsupported format %d\n", - params_format(params)); + dev_err(codec->dev, "unsupported width %d\n", + params_width(params)); return -EINVAL; } -- cgit v1.2.3 From 04f630d845d1deb2e1d6f39f7788ff30b686a389 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:49:12 +0100 Subject: ASoC: twl4030: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown Acked-by: Peter Ujfalusi --- sound/soc/codecs/twl4030.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 69e12a311ba2..6164a7e43896 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1764,16 +1764,16 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, old_format = twl4030_read(codec, TWL4030_REG_AUDIO_IF); format = old_format; format &= ~TWL4030_DATA_WIDTH; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: format |= TWL4030_DATA_WIDTH_16S_16W; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: format |= TWL4030_DATA_WIDTH_32S_24W; break; default: - dev_err(codec->dev, "%s: unknown format %d\n", __func__, - params_format(params)); + dev_err(codec->dev, "%s: unsupported bits/sample %d\n", + __func__, params_width(params)); return -EINVAL; } -- cgit v1.2.3 From aa9ffad68a0a5aa00e7c475e28c9951432d701c4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:49:26 +0100 Subject: ASoC: uda134x: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown --- sound/soc/codecs/uda134x.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index edf27acc1d77..ad145ba85cd5 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -243,14 +243,14 @@ static int uda134x_hw_params(struct snd_pcm_substream *substream, case SND_SOC_DAIFMT_I2S: break; case SND_SOC_DAIFMT_RIGHT_J: - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: hw_params |= (1<<1); break; - case SNDRV_PCM_FORMAT_S18_3LE: + case 18: hw_params |= (1<<2); break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: hw_params |= ((1<<2) | (1<<1)); break; default: -- cgit v1.2.3 From 9630181aac6e709ff248e68474701f763b286c98 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:49:36 +0100 Subject: ASoC: wl1273: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown --- sound/soc/codecs/wl1273.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index 4ead0dc02b87..34f64dc281ec 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -341,8 +341,9 @@ static int wl1273_hw_params(struct snd_pcm_substream *substream, struct wl1273_core *core = wl1273->core; unsigned int rate, width, r; - if (params_format(params) != SNDRV_PCM_FORMAT_S16_LE) { - pr_err("Only SNDRV_PCM_FORMAT_S16_LE supported.\n"); + if (params_width(params) != 16) { + dev_err(dai->dev, "%d bits/sample not supported\n", + params_width(params)); return -EINVAL; } -- cgit v1.2.3 From 12efd9f4b7518081315bdc32ab91291698739047 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:28:37 +0100 Subject: ASoC: cs4265: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown Acked-by: Paul Handrigan --- sound/soc/codecs/cs4265.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index 4c4bf216d51a..a20b30ca52c0 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -455,7 +455,7 @@ static int cs4265_pcm_hw_params(struct snd_pcm_substream *substream, CS4265_SPDIF_CTL2_DIF, (1 << 6)); break; case SND_SOC_DAIFMT_RIGHT_J: - if (params_format(params) & SNDRV_PCM_FORMAT_S16_LE) { + if (params_width(params) == 16) { snd_soc_update_bits(codec, CS4265_DAC_CTL, CS4265_DAC_CTL_DIF, (1 << 5)); snd_soc_update_bits(codec, CS4265_ADC_CTL, -- cgit v1.2.3 From 1e6453acda6718992373237a9f8fb36024d30040 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:49:48 +0100 Subject: ASoC: wm8350: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8350.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 392285edb595..b3053a98db82 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -918,16 +918,16 @@ static int wm8350_pcm_hw_params(struct snd_pcm_substream *substream, ~WM8350_AIF_WL_MASK; /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: iface |= 0x1 << 10; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: iface |= 0x2 << 10; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: iface |= 0x3 << 10; break; } -- cgit v1.2.3 From 6fe12c2fbbb3bf66431fdf75fbdbdc57088f760c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:49:53 +0100 Subject: ASoC: wm8400: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8400.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 06e913d3fea1..72471bef2e9a 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -1095,16 +1095,16 @@ static int wm8400_hw_params(struct snd_pcm_substream *substream, audio1 &= ~WM8400_AIF_WL_MASK; /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: audio1 |= WM8400_AIF_WL_20BITS; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: audio1 |= WM8400_AIF_WL_24BITS; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: audio1 |= WM8400_AIF_WL_32BITS; break; } -- cgit v1.2.3 From 19e4b7886bdf997746e53f7ae80a7e5dc9e04c4d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:50:04 +0100 Subject: ASoC: wm8510: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8510.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 1c1e328feeb8..e11127f9069e 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -449,16 +449,16 @@ static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream, u16 adn = snd_soc_read(codec, WM8510_ADD) & 0x1f1; /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: iface |= 0x0020; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: iface |= 0x0040; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: iface |= 0x0060; break; } -- cgit v1.2.3 From 3fafcc8e61f55ab281dfbdb5d0121f090ecd3320 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:51:02 +0100 Subject: ASoC: wm8523: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8523.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 601ee8178af1..ec1f5740dbd0 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -163,16 +163,16 @@ static int wm8523_hw_params(struct snd_pcm_substream *substream, aifctrl2 |= lrclk_ratios[i].value; aifctrl1 &= ~WM8523_WL_MASK; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: aifctrl1 |= 0x8; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: aifctrl1 |= 0x10; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: aifctrl1 |= 0x18; break; } -- cgit v1.2.3 From 993ba8e5954bc6aeb9d7bc0e4341e4e25edd9a32 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:51:10 +0100 Subject: ASoC: wm8580: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8580.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 7665ff6aea6d..911605ee25b0 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -511,19 +511,19 @@ static int wm8580_paif_hw_params(struct snd_pcm_substream *substream, int i, ratio, osr; /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: paifa |= 0x8; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: paifa |= 0x0; paifb |= WM8580_AIF_LENGTH_20; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: paifa |= 0x0; paifb |= WM8580_AIF_LENGTH_24; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: paifa |= 0x0; paifb |= WM8580_AIF_LENGTH_32; break; -- cgit v1.2.3 From b9abb5485667f05bf9590a31b0ff4d35bf0d91c1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:51:18 +0100 Subject: ASoC: wm8711: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8711.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index b0fbcb377baf..32187e739b4f 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -169,13 +169,13 @@ static int wm8711_hw_params(struct snd_pcm_substream *substream, snd_soc_write(codec, WM8711_SRATE, srate); /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: iface |= 0x0004; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: iface |= 0x0008; break; } -- cgit v1.2.3 From 9fbad31a9e3b461dc87b099efdbb7c90036e2260 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:51:26 +0100 Subject: ASoC: wm8728: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8728.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index bac7fc28fe71..38ff826f589a 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -94,13 +94,13 @@ static int wm8728_hw_params(struct snd_pcm_substream *substream, dac &= ~0x18; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: dac |= 0x10; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: dac |= 0x08; break; default: -- cgit v1.2.3 From dfb6778e64c6aef0a763f42b93487ea578670008 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:51:32 +0100 Subject: ASoC: wm8731: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8731.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 5ada61611324..eebb3280bfad 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -348,13 +348,13 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream, snd_soc_write(codec, WM8731_SRATE, srate); /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: iface |= 0x0004; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: iface |= 0x0008; break; } -- cgit v1.2.3 From a5d0de23b203cc40df7196858b7a6ad546bb8408 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:51:39 +0100 Subject: ASoC: wm8737: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8737.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index b27f26cdc049..744a422ecb05 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -367,16 +367,16 @@ static int wm8737_hw_params(struct snd_pcm_substream *substream, clocking |= coeff_div[i].usb | (coeff_div[i].sr << WM8737_SR_SHIFT); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: af |= 0x8; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: af |= 0x10; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: af |= 0x18; break; default: -- cgit v1.2.3 From 34967ad2426257761cb158128f8e819d3fb0c713 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:51:45 +0100 Subject: ASoC: wm8741: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8741.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index b33542a04607..a237f1627f61 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -241,26 +241,26 @@ static int wm8741_hw_params(struct snd_pcm_substream *substream, } /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: iface |= 0x0001; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: iface |= 0x0002; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: iface |= 0x0003; break; default: dev_dbg(codec->dev, "wm8741_hw_params: Unsupported bit size param = %d", - params_format(params)); + params_width(params)); return -EINVAL; } dev_dbg(codec->dev, "wm8741_hw_params: bit size param = %d", - params_format(params)); + params_width(params)); snd_soc_write(codec, WM8741_FORMAT_CONTROL, iface); return 0; -- cgit v1.2.3 From 7e322dff367bcdb7fe9d85d5afd470ce3c751d1b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:52:02 +0100 Subject: ASoC: wm8750: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8750.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 33990b63d214..67653a2db223 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -586,16 +586,16 @@ static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream, int coeff = get_coeff(wm8750->sysclk, params_rate(params)); /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: iface |= 0x0004; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: iface |= 0x0008; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: iface |= 0x000c; break; } -- cgit v1.2.3 From f21b66043d52dde17e9ebb498df9b4b3883e3d5e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:52:19 +0100 Subject: ASoC: wm8753: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8753.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 53e57b4049a8..e54e097f4fcb 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -937,16 +937,16 @@ static int wm8753_pcm_hw_params(struct snd_pcm_substream *substream, u16 srate = snd_soc_read(codec, WM8753_SRATE1) & 0x017f; /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: voice |= 0x0004; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: voice |= 0x0008; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: voice |= 0x000c; break; } @@ -1176,16 +1176,16 @@ static int wm8753_i2s_hw_params(struct snd_pcm_substream *substream, coeff_div[coeff].usb); /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: hifi |= 0x0004; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: hifi |= 0x0008; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: hifi |= 0x000c; break; } -- cgit v1.2.3 From 18ce1da9da69b5fa53f896754a66a91b1ec30908 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:52:26 +0100 Subject: ASoC: wm8770: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8770.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c index c61aeb38efb8..180e7a098726 100644 --- a/sound/soc/codecs/wm8770.c +++ b/sound/soc/codecs/wm8770.c @@ -426,16 +426,16 @@ static int wm8770_hw_params(struct snd_pcm_substream *substream, wm8770 = snd_soc_codec_get_drvdata(codec); iface = 0; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: iface |= 0x10; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: iface |= 0x20; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: iface |= 0x30; break; } -- cgit v1.2.3 From 16cfd485401ab4dcf8a06a56c14b9a1a5544a1c9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:52:37 +0100 Subject: ASoC: wm8804: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8804.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index d96e5963ee35..0ea01dfcb6e1 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -270,19 +270,19 @@ static int wm8804_hw_params(struct snd_pcm_substream *substream, codec = dai->codec; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: blen = 0x0; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: blen = 0x1; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: blen = 0x2; break; default: dev_err(dai->dev, "Unsupported word length: %u\n", - params_format(params)); + params_width(params)); return -EINVAL; } -- cgit v1.2.3 From 0cd44891584628904108a6dcc4f44ee9425b47b7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:52:44 +0100 Subject: ASoC: wm8900: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8900.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index d09fdce57f5a..44a5f1511f0f 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -640,16 +640,16 @@ static int wm8900_hw_params(struct snd_pcm_substream *substream, reg = snd_soc_read(codec, WM8900_REG_AUDIO1) & ~0x60; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: reg |= 0x20; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: reg |= 0x40; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: reg |= 0x60; break; default: -- cgit v1.2.3 From 6139ea270d64225f7820284da544ea6a59e42fad Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:52:53 +0100 Subject: ASoC: wm8903: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8903.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index b84940c359a1..fa4bccf601e6 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1477,19 +1477,19 @@ static int wm8903_hw_params(struct snd_pcm_substream *substream, aif1 &= ~WM8903_AIF_WL_MASK; bclk = 2 * fs; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: bclk *= 16; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: bclk *= 20; aif1 |= 0x4; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: bclk *= 24; aif1 |= 0x8; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: bclk *= 32; aif1 |= 0xc; break; -- cgit v1.2.3 From 2f44b0437aae1c5badb63a18fda0d95e51591e23 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:53:00 +0100 Subject: ASoC: wm8904: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8904.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index f7c549949c54..79b303a1f2a7 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1290,16 +1290,16 @@ static int wm8904_hw_params(struct snd_pcm_substream *substream, wm8904->bclk = snd_soc_params_to_bclk(params); } - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: aif1 |= 0x40; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: aif1 |= 0x80; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: aif1 |= 0xc0; break; default: -- cgit v1.2.3 From 74b24c3866f4cbd129a21651a1e2dfe81a10b268 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:53:08 +0100 Subject: ASoC: wm8940: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8940.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index fc6eec9ad66b..52011043e54c 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -430,19 +430,19 @@ static int wm8940_i2s_hw_params(struct snd_pcm_substream *substream, if (ret) goto error_ret; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: + switch (params_width(params)) { + case 8: companding = companding | (1 << 5); break; - case SNDRV_PCM_FORMAT_S16_LE: + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: iface |= (1 << 5); break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: iface |= (2 << 5); break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: iface |= (3 << 5); break; } -- cgit v1.2.3 From 1df93ca37501f752add410f5994e1259ae714e17 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:53:16 +0100 Subject: ASoC: wm8955: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8955.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index 2a35108f233d..09d91d9dc4ee 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -597,17 +597,17 @@ static int wm8955_hw_params(struct snd_pcm_substream *substream, int ret; int wl; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: wl = 0; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: wl = 0x4; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: wl = 0x8; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: wl = 0xc; break; default: -- cgit v1.2.3 From 39e9cc46469e1d56522e6de45a6b2468d4d7eb60 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:53:23 +0100 Subject: ASoC: wm8960: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8960.c | 15 ++++++--------- 1 file changed, 6 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index a145d0431b63..1916ac74fdc4 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -567,24 +567,21 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, WM8960_IFACE1) & 0xfff3; - snd_pcm_format_t format = params_format(params); int i; /* bit size */ - switch (format) { - case SNDRV_PCM_FORMAT_S16_LE: - case SNDRV_PCM_FORMAT_S16_BE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: - case SNDRV_PCM_FORMAT_S20_3BE: + case 20: iface |= 0x0004; break; - case SNDRV_PCM_FORMAT_S24_LE: - case SNDRV_PCM_FORMAT_S24_BE: + case 24: iface |= 0x0008; break; default: - dev_err(codec->dev, "unsupported format %i\n", format); + dev_err(codec->dev, "unsupported width %d\n", + params_width(params)); return -EINVAL; } -- cgit v1.2.3 From 5d3aef91a49a6c2fb8d7714ac814c6befc9ae763 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:53:30 +0100 Subject: ASoC: wm8961: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8961.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 9c88f04442b3..41d23e920ad5 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -565,16 +565,16 @@ static int wm8961_hw_params(struct snd_pcm_substream *substream, reg = snd_soc_read(codec, WM8961_AUDIO_INTERFACE_0); reg &= ~WM8961_WL_MASK; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: reg |= 1 << WM8961_WL_SHIFT; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: reg |= 2 << WM8961_WL_SHIFT; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: reg |= 3 << WM8961_WL_SHIFT; break; default: -- cgit v1.2.3 From ec4dc01eb41a7eb03675516975f704b355fe2fc1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:53:36 +0100 Subject: ASoC: wm8962: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8962.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index ca2fda9d72be..4be6f2afdbc4 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2586,16 +2586,16 @@ static int wm8962_hw_params(struct snd_pcm_substream *substream, if (wm8962->lrclk % 8000 == 0) adctl3 |= WM8962_SAMPLE_RATE_INT_MODE; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: aif0 |= 0x4; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: aif0 |= 0x8; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: aif0 |= 0xc; break; default: -- cgit v1.2.3 From d835e99a89b9e02b7e4837debff5ea049e293ba1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:53:43 +0100 Subject: ASoC: wm8971: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8971.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 09b7b4200221..0499cd4cfb71 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -517,16 +517,16 @@ static int wm8971_pcm_hw_params(struct snd_pcm_substream *substream, int coeff = get_coeff(wm8971->sysclk, params_rate(params)); /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: iface |= 0x0004; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: iface |= 0x0008; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: iface |= 0x000c; break; } -- cgit v1.2.3 From 6afdc9a0e7062a804a47c4a21d775e29b864d367 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:53:50 +0100 Subject: ASoC: wm8974: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8974.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 0627c56fa44e..682e9eda1019 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -445,16 +445,16 @@ static int wm8974_pcm_hw_params(struct snd_pcm_substream *substream, u16 adn = snd_soc_read(codec, WM8974_ADD) & 0x1f1; /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: iface |= 0x0020; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: iface |= 0x0040; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: iface |= 0x0060; break; } -- cgit v1.2.3 From a4c8261d0b1a4f0d95de9fb3b56d201ad941e7f1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:53:57 +0100 Subject: ASoC: wm8978: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8978.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 28ef46c91f62..ee2ba574952b 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -736,16 +736,16 @@ static int wm8978_hw_params(struct snd_pcm_substream *substream, return -EINVAL; /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: iface_ctl |= 0x20; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: iface_ctl |= 0x40; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: iface_ctl |= 0x60; break; } @@ -817,8 +817,8 @@ static int wm8978_hw_params(struct snd_pcm_substream *substream, wm8978->sysclk == WM8978_MCLK ? ", consider using PLL" : ""); - dev_dbg(codec->dev, "%s: fmt %d, rate %u, MCLK divisor #%d\n", __func__, - params_format(params), params_rate(params), best); + dev_dbg(codec->dev, "%s: width %d, rate %u, MCLK divisor #%d\n", __func__, + params_width(params), params_rate(params), best); /* MCLK divisor mask = 0xe0 */ snd_soc_update_bits(codec, WM8978_CLOCKING, 0xe0, best << 5); -- cgit v1.2.3 From af8ff1469e66437954dd016713ae630f94e0d38c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:54:09 +0100 Subject: ASoC: wm8983: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8983.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c index 19d5baa38f5c..ac5defda8824 100644 --- a/sound/soc/codecs/wm8983.c +++ b/sound/soc/codecs/wm8983.c @@ -719,22 +719,22 @@ static int wm8983_hw_params(struct snd_pcm_substream *substream, wm8983->bclk = ret; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: blen = 0x0; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: blen = 0x1; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: blen = 0x2; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: blen = 0x3; break; default: dev_err(dai->dev, "Unsupported word length %u\n", - params_format(params)); + params_width(params)); return -EINVAL; } -- cgit v1.2.3 From c37642c728c21b37de76885d2203fcace256b011 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:54:16 +0100 Subject: ASoC: wm8985: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8985.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c index 0f5780c09f3a..b0f643458e0a 100644 --- a/sound/soc/codecs/wm8985.c +++ b/sound/soc/codecs/wm8985.c @@ -698,22 +698,22 @@ static int wm8985_hw_params(struct snd_pcm_substream *substream, if ((int)wm8985->bclk < 0) return wm8985->bclk; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: blen = 0x0; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: blen = 0x1; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: blen = 0x2; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: blen = 0x3; break; default: dev_err(dai->dev, "Unsupported word length %u\n", - params_format(params)); + params_width(params)); return -EINVAL; } -- cgit v1.2.3 From 8cd2ba092c71d06b0d8f10649022f663ff5d9f77 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:54:23 +0100 Subject: ASoC: wm8988: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8988.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index d3fea46d58e8..a5130d965146 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -687,16 +687,16 @@ static int wm8988_pcm_hw_params(struct snd_pcm_substream *substream, } /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: iface |= 0x0004; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: iface |= 0x0008; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: iface |= 0x000c; break; } -- cgit v1.2.3 From a351901d166ed17195c375f5de2edf48c5e9f327 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:54:30 +0100 Subject: ASoC: wm8990: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8990.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index b5c1f0f07058..03e43e3f395e 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1073,16 +1073,16 @@ static int wm8990_hw_params(struct snd_pcm_substream *substream, audio1 &= ~WM8990_AIF_WL_MASK; /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: audio1 |= WM8990_AIF_WL_20BITS; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: audio1 |= WM8990_AIF_WL_24BITS; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: audio1 |= WM8990_AIF_WL_32BITS; break; } -- cgit v1.2.3 From 20a77bbd27ff7c04166327cc463437c17c2185f8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:54:38 +0100 Subject: ASoC: wm8991: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8991.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index b8fd284fc0c0..d0be89731cdb 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -1081,16 +1081,16 @@ static int wm8991_hw_params(struct snd_pcm_substream *substream, audio1 &= ~WM8991_AIF_WL_MASK; /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: audio1 |= WM8991_AIF_WL_20BITS; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: audio1 |= WM8991_AIF_WL_24BITS; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: audio1 |= WM8991_AIF_WL_32BITS; break; } -- cgit v1.2.3 From ae62ba67b39f9f6db3e85dde357540e87edfc45b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:54:45 +0100 Subject: ASoC: wm8993: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8993.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index f825dc04ebe1..93b14eda355a 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1214,19 +1214,19 @@ static int wm8993_hw_params(struct snd_pcm_substream *substream, wm8993->tdm_slots, wm8993->tdm_width); wm8993->bclk *= wm8993->tdm_width * wm8993->tdm_slots; } else { - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: wm8993->bclk *= 16; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: wm8993->bclk *= 20; aif1 |= 0x8; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: wm8993->bclk *= 24; aif1 |= 0x10; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: wm8993->bclk *= 32; aif1 |= 0x18; break; -- cgit v1.2.3 From e29fe49699ddd9998dc46696331097941c1ae79b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:54:58 +0100 Subject: ASoC: wm8994: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8994.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 247b39013fba..e7dcaf3645db 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2815,19 +2815,19 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, } bclk_rate = params_rate(params); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: bclk_rate *= 16; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: bclk_rate *= 20; aif1 |= 0x20; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: bclk_rate *= 24; aif1 |= 0x40; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: bclk_rate *= 32; aif1 |= 0x60; break; @@ -2966,16 +2966,16 @@ static int wm8994_aif3_hw_params(struct snd_pcm_substream *substream, return 0; } - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: aif1 |= 0x20; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: aif1 |= 0x40; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: aif1 |= 0x60; break; default: -- cgit v1.2.3 From f882728c978c8e6f78b4b63efe0f7880b8139327 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:55:03 +0100 Subject: ASoC: wm8995: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8995.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 863a2c38bcb5..cae4ac5a5730 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -1597,21 +1597,21 @@ static int wm8995_hw_params(struct snd_pcm_substream *substream, return bclk_rate; aif1 = 0; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: aif1 |= (0x1 << WM8995_AIF1_WL_SHIFT); break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: aif1 |= (0x2 << WM8995_AIF1_WL_SHIFT); break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: aif1 |= (0x3 << WM8995_AIF1_WL_SHIFT); break; default: dev_err(dai->dev, "Unsupported word length %u\n", - params_format(params)); + params_width(params)); return -EINVAL; } -- cgit v1.2.3 From 9572696ddb3f258d64cad9bc4ac3468d1285fdd1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:55:10 +0100 Subject: ASoC: wm9081: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm9081.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 185eb97769e7..0cdc9e2184ab 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1029,19 +1029,19 @@ static int wm9081_hw_params(struct snd_pcm_substream *substream, /* Otherwise work out a BCLK from the sample size */ wm9081->bclk = 2 * wm9081->fs; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: wm9081->bclk *= 16; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: wm9081->bclk *= 20; aif2 |= 0x4; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: wm9081->bclk *= 24; aif2 |= 0x8; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: wm9081->bclk *= 32; aif2 |= 0xc; break; -- cgit v1.2.3 From 563fe71fe76260e32aefc1a2c03f4634c4d5edf9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:55:23 +0100 Subject: ASoC: wm9713: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm9713.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 2a9c6d11330c..bddee30a4bc7 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -953,16 +953,16 @@ static int wm9713_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; u16 reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0xfff3; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: reg |= 0x0004; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: reg |= 0x0008; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: reg |= 0x000c; break; } -- cgit v1.2.3 From 19a23a5d76e59f84caafea7a3299c23894ecad63 Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Wed, 30 Jul 2014 18:36:00 +0530 Subject: ASoC: Intel: mfld-pcm: Fix to use correct sst_data pointer Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-platform-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 7de87887d9f8..47df05ed3ac3 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -573,7 +573,7 @@ static int sst_platform_probe(struct platform_device *pdev) struct sst_platform_data *pdata = pdev->dev.platform_data; drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL); - if (sst == NULL) { + if (drv == NULL) { pr_err("kzalloc failed\n"); return -ENOMEM; } -- cgit v1.2.3 From 2741d43a1edd13c81a50ceb63f4edbf5fedb53ce Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Wed, 30 Jul 2014 18:39:05 +0530 Subject: ASoC: Intel: mfld-pcm: Allocate platform data Platform data may be null during platform_device_add. Allocate platform data before using. Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-platform-pcm.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 47df05ed3ac3..706212a6a68c 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -570,7 +570,7 @@ static int sst_platform_probe(struct platform_device *pdev) { struct sst_data *drv; int ret; - struct sst_platform_data *pdata = pdev->dev.platform_data; + struct sst_platform_data *pdata; drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL); if (drv == NULL) { @@ -578,6 +578,12 @@ static int sst_platform_probe(struct platform_device *pdev) return -ENOMEM; } + pdata = devm_kzalloc(&pdev->dev, sizeof(*pdata), GFP_KERNEL); + if (pdata == NULL) { + pr_err("kzalloc failed for pdata\n"); + return -ENOMEM; + } + pdata->pdev_strm_map = dpcm_strm_map; pdata->strm_map_size = ARRAY_SIZE(dpcm_strm_map); drv->pdata = pdata; -- cgit v1.2.3 From d526416c4fb23a48ed2547138c43e96fa3901124 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Thu, 31 Jul 2014 12:07:40 +0800 Subject: ASoC: fsl_asrc: Fix sparse warnings in FSL_ASRC_FORMATS due to typo reproduce: make C=1 CF=-D__CHECK_ENDIAN__ sparse warnings: (new ones prefixed by >>) >> sound/soc/fsl/fsl_asrc.c:563:28: sparse: restricted snd_pcm_format_t degrades to integer >> sound/soc/fsl/fsl_asrc.c:570:28: sparse: restricted snd_pcm_format_t degrades to integer vim +563 sound/soc/fsl/fsl_asrc.c 557 .probe = fsl_asrc_dai_probe, 558 .playback = { 559 .stream_name = "ASRC-Playback", 560 .channels_min = 1, 561 .channels_max = 10, 562 .rates = FSL_ASRC_RATES, > 563 .formats = FSL_ASRC_FORMATS, 564 }, 565 .capture = { 566 .stream_name = "ASRC-Capture", 567 .channels_min = 1, 568 .channels_max = 10, 569 .rates = FSL_ASRC_RATES, > 570 .formats = FSL_ASRC_FORMATS, 571 }, 572 .ops = &fsl_asrc_dai_ops, 573 }; Reported-by: kbuild test robot Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_asrc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index b9a288877267..910525b5e5fc 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -551,7 +551,7 @@ static int fsl_asrc_dai_probe(struct snd_soc_dai *dai) #define FSL_ASRC_RATES SNDRV_PCM_RATE_8000_192000 #define FSL_ASRC_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S16_LE | \ - SNDRV_PCM_FORMAT_S20_3LE) + SNDRV_PCM_FMTBIT_S20_3LE) static struct snd_soc_dai_driver fsl_asrc_dai = { .probe = fsl_asrc_dai_probe, -- cgit v1.2.3 From d387dd08e444b22f844475780fe12a1ad1c6fffd Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Thu, 31 Jul 2014 12:32:09 +0300 Subject: ASoC: fsl_asrc: fix an error code in fsl_asrc_probe() There is a cut and paste bug so it returns success instead of the error code. Signed-off-by: Dan Carpenter Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_asrc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 910525b5e5fc..424b23c38d9a 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -826,7 +826,7 @@ static int fsl_asrc_probe(struct platform_device *pdev) asrc_priv->mem_clk = devm_clk_get(&pdev->dev, "mem"); if (IS_ERR(asrc_priv->mem_clk)) { dev_err(&pdev->dev, "failed to get mem clock\n"); - return PTR_ERR(asrc_priv->ipg_clk); + return PTR_ERR(asrc_priv->mem_clk); } asrc_priv->ipg_clk = devm_clk_get(&pdev->dev, "ipg"); -- cgit v1.2.3 From 4548728981de259d7d37d0ae968a777b09794168 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Thu, 31 Jul 2014 15:57:51 +0300 Subject: ASoC: pxa: pxa-ssp: small leak in probe() There is a small memory leak if probe() fails. Fixes: 2023c90c3a2c ('ASoC: pxa: pxa-ssp: add DT bindings') Signed-off-by: Dan Carpenter Signed-off-by: Mark Brown --- sound/soc/pxa/pxa-ssp.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 199a8b377553..0109f6c2334e 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -723,7 +723,8 @@ static int pxa_ssp_probe(struct snd_soc_dai *dai) ssp_handle = of_parse_phandle(dev->of_node, "port", 0); if (!ssp_handle) { dev_err(dev, "unable to get 'port' phandle\n"); - return -ENODEV; + ret = -ENODEV; + goto err_priv; } priv->ssp = pxa_ssp_request_of(ssp_handle, "SoC audio"); -- cgit v1.2.3 From 054cd7f4b986a941b56372b743602f7df6569c98 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 30 Jul 2014 23:52:04 -0700 Subject: ASoC: rsnd: care detail of SRC_BSDSR Driver should care more detail of SRC_BSDSR settings. The sound includes noise without this patch if it used SRC. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/src.c | 32 +++++++++++++++++++++++++++++++- 1 file changed, 31 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 4d39505c21cf..1ef811a26bd7 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -483,15 +483,45 @@ static struct rsnd_mod_ops rsnd_src_gen1_ops = { static int rsnd_src_set_convert_rate_gen2(struct rsnd_mod *mod, struct rsnd_dai *rdai) { + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct device *dev = rsnd_priv_to_dev(priv); + struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); + struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + struct rsnd_src *src = rsnd_mod_to_src(mod); + uint ratio; int ret; + /* 6 - 1/6 are very enough ratio for SRC_BSDSR */ + if (!rsnd_src_convert_rate(src)) + ratio = 0; + else if (rsnd_src_convert_rate(src) > runtime->rate) + ratio = 100 * rsnd_src_convert_rate(src) / runtime->rate; + else + ratio = 100 * runtime->rate / rsnd_src_convert_rate(src); + + if (ratio > 600) { + dev_err(dev, "FSO/FSI ratio error\n"); + return -EINVAL; + } + ret = rsnd_src_set_convert_rate(mod, rdai); if (ret < 0) return ret; rsnd_mod_write(mod, SRC_SRCCR, 0x00011110); - rsnd_mod_write(mod, SRC_BSDSR, 0x01800000); + switch (rsnd_mod_id(mod)) { + case 5: + case 6: + case 7: + case 8: + rsnd_mod_write(mod, SRC_BSDSR, 0x02400000); + break; + default: + rsnd_mod_write(mod, SRC_BSDSR, 0x01800000); + break; + } + rsnd_mod_write(mod, SRC_BSISR, 0x00100060); return 0; -- cgit v1.2.3 From 1cc7195929501b96fccce42646f1ad0ffe2598a6 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 30 Jul 2014 23:52:26 -0700 Subject: ASoC: rsnd: care audio local bus data format consistency R-Car sound uses Audio Local Bus which uses Lch/Rch format. This bus is used if driver uses BUSIF. But sound data is written as Rch/Lch format in register. This means Rch <-> Lch will be inverted. SSIU :: BUSIF_DALIGN is used to controlling data format. Reported-by: Jun Watanabe Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/gen.c | 1 + sound/soc/sh/rcar/rsnd.h | 2 ++ sound/soc/sh/rcar/src.c | 19 +++++++++++++++++++ 3 files changed, 22 insertions(+) (limited to 'sound') diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 5f9e0722abcf..5db055750991 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -284,6 +284,7 @@ static int rsnd_gen2_probe(struct platform_device *pdev, /* FIXME: it needs SSI_MODE2/3 in the future */ RSND_GEN_M_REG(SSI_BUSIF_MODE, 0x0, 0x80), RSND_GEN_M_REG(SSI_BUSIF_ADINR, 0x4, 0x80), + RSND_GEN_M_REG(BUSIF_DALIGN, 0x8, 0x80), RSND_GEN_M_REG(SSI_CTRL, 0x10, 0x80), RSND_GEN_M_REG(INT_ENABLE, 0x18, 0x80), }; diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 631b149df08f..d119adf97c9c 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -90,6 +90,7 @@ enum rsnd_reg { RSND_REG_SHARE19, RSND_REG_SHARE20, RSND_REG_SHARE21, + RSND_REG_SHARE22, RSND_REG_MAX, }; @@ -127,6 +128,7 @@ enum rsnd_reg { #define RSND_REG_AUDIO_CLK_SEL2 RSND_REG_SHARE19 #define RSND_REG_CMD_CTRL RSND_REG_SHARE20 #define RSND_REG_CMDOUT_TIMSEL RSND_REG_SHARE21 +#define RSND_REG_BUSIF_DALIGN RSND_REG_SHARE22 struct rsnd_of_data; struct rsnd_priv; diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 1ef811a26bd7..9183e0145503 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -110,6 +110,8 @@ int rsnd_src_ssiu_start(struct rsnd_mod *ssi_mod, struct rsnd_dai *rdai, int use_busif) { + struct rsnd_dai_stream *io = rsnd_mod_to_io(ssi_mod); + struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); int ssi_id = rsnd_mod_id(ssi_mod); /* @@ -146,10 +148,27 @@ int rsnd_src_ssiu_start(struct rsnd_mod *ssi_mod, * DMA settings for SSIU */ if (use_busif) { + u32 val = 0x76543210; + u32 mask = ~0; + rsnd_mod_write(ssi_mod, SSI_BUSIF_ADINR, rsnd_get_adinr(ssi_mod)); rsnd_mod_write(ssi_mod, SSI_BUSIF_MODE, 1); rsnd_mod_write(ssi_mod, SSI_CTRL, 0x1); + + mask <<= runtime->channels * 4; + val = val & mask; + + switch (runtime->sample_bits) { + case 16: + val |= 0x67452301 & ~mask; + break; + case 32: + val |= 0x76543210 & ~mask; + break; + } + rsnd_mod_write(ssi_mod, BUSIF_DALIGN, val); + } return 0; -- cgit v1.2.3 From 4f8f86aa41232cc4595136f18013324482fdf749 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 30 Jul 2014 23:52:50 -0700 Subject: ASoC: rsnd: add missing dev_dbg() in rsnd_bset() b8c637864a6904a9ba8e0df556d5bdf9f26b2c54 (ASoC: rsnd: use regmap_mmio instead of original regmap bus) used regmap_mmio and modified dev_dbg() for rsnd_read/write(). But rsnd_bset() is missing it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/gen.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 5db055750991..3fdf3be7b99a 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -92,6 +92,7 @@ void rsnd_write(struct rsnd_priv *priv, void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg, u32 mask, u32 data) { + struct device *dev = rsnd_priv_to_dev(priv); struct rsnd_gen *gen = rsnd_priv_to_gen(priv); if (!rsnd_is_accessible_reg(priv, gen, reg)) @@ -99,6 +100,9 @@ void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, regmap_fields_update_bits(gen->regs[reg], rsnd_mod_id(mod), mask, data); + + dev_dbg(dev, "b %s - 0x%04d : %08x/%08x\n", + rsnd_mod_name(mod), reg, data, mask); } #define rsnd_gen_regmap_init(priv, id_size, reg_id, conf) \ -- cgit v1.2.3 From 7c63f3c02f5a998621758239abed43c5bd454b70 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 30 Jul 2014 23:53:03 -0700 Subject: ASoC: rsnd: fixup pcm_new callback method bff58ea4f43d9b4a9fd6fb05fabc8f50f68131f5 (ASoC: rsnd: add DVC support) added DVC support, and it added pcm_new callback feature for it. Then it called all DAI's pcm_new callback, and it was wrong. This patch fixup it and call correct callback. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 20 +++++++++----------- 1 file changed, 9 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index f07742f1eb11..d7c669790dd4 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -944,19 +944,17 @@ static struct snd_pcm_ops rsnd_pcm_ops = { static int rsnd_pcm_new(struct snd_soc_pcm_runtime *rtd) { - struct rsnd_priv *priv = snd_soc_dai_get_drvdata(rtd->cpu_dai); - struct rsnd_dai *rdai; - int i, ret; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); + int ret; - for_each_rsnd_dai(rdai, priv, i) { - ret = rsnd_dai_call(pcm_new, &rdai->playback, rdai, rtd); - if (ret) - return ret; + ret = rsnd_dai_call(pcm_new, &rdai->playback, rdai, rtd); + if (ret) + return ret; - ret = rsnd_dai_call(pcm_new, &rdai->capture, rdai, rtd); - if (ret) - return ret; - } + ret = rsnd_dai_call(pcm_new, &rdai->capture, rdai, rtd); + if (ret) + return ret; return snd_pcm_lib_preallocate_pages_for_all( rtd->pcm, -- cgit v1.2.3 From d7821953cfe9803c593a682320468ce2de862803 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Tue, 29 Jul 2014 18:38:39 +0800 Subject: ASoC: wm8962: Let CODEC driver enable and disable its own MCLK snd_soc_open() will trigger pm_runtime resume() which will then enable the regulator and initialization. So we should make sure the MCLK is enabled before this resume(). Previously we let the machine driver get the clock and enable it in its probe(). However, considering about power saving, it'll be better to enable it when it's going to be used and disable it after using. So this patch just simply adds clk_get() and clk_enable() to WM8962 driver. Meanwhile, it marks clock pointer to NULL if no clock assigned to it so it will not break any current function. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- include/sound/wm8962.h | 1 + sound/soc/codecs/wm8962.c | 19 +++++++++++++++++++ 2 files changed, 20 insertions(+) (limited to 'sound') diff --git a/include/sound/wm8962.h b/include/sound/wm8962.h index 79e6d427b858..0af7c1674cbf 100644 --- a/include/sound/wm8962.h +++ b/include/sound/wm8962.h @@ -37,6 +37,7 @@ #define WM8962_GPIO_FN_MICSCD 22 struct wm8962_pdata { + struct clk *mclk; int gpio_base; u32 gpio_init[WM8962_MAX_GPIO]; diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index ca2fda9d72be..eac29031ae26 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -14,6 +14,7 @@ #include #include #include +#include #include #include #include @@ -3541,6 +3542,8 @@ static int wm8962_set_pdata_from_of(struct i2c_client *i2c, pdata->gpio_init[i] = 0x0; } + pdata->mclk = devm_clk_get(&i2c->dev, NULL); + return 0; } @@ -3572,6 +3575,14 @@ static int wm8962_i2c_probe(struct i2c_client *i2c, return ret; } + /* Mark the mclk pointer to NULL if no mclk assigned */ + if (IS_ERR(wm8962->pdata.mclk)) { + /* But do not ignore the request for probe defer */ + if (PTR_ERR(wm8962->pdata.mclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + wm8962->pdata.mclk = NULL; + } + for (i = 0; i < ARRAY_SIZE(wm8962->supplies); i++) wm8962->supplies[i].supply = wm8962_supply_names[i]; @@ -3780,6 +3791,12 @@ static int wm8962_runtime_resume(struct device *dev) struct wm8962_priv *wm8962 = dev_get_drvdata(dev); int ret; + ret = clk_prepare_enable(wm8962->pdata.mclk); + if (ret) { + dev_err(dev, "Failed to enable MCLK: %d\n", ret); + return ret; + } + ret = regulator_bulk_enable(ARRAY_SIZE(wm8962->supplies), wm8962->supplies); if (ret != 0) { @@ -3839,6 +3856,8 @@ static int wm8962_runtime_suspend(struct device *dev) regulator_bulk_disable(ARRAY_SIZE(wm8962->supplies), wm8962->supplies); + clk_disable_unprepare(wm8962->pdata.mclk); + return 0; } #endif -- cgit v1.2.3 From 74ccb27c35c799a14933c282c4e3c864886fc429 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Wed, 30 Jul 2014 11:10:26 +0800 Subject: ASoC: fsl_esai: Add stream names for DPCM usage DPCM needs extra dapm routes in the machine driver to route audio between Front-End and Back-End. In order to differ the stream names in the route map from CODECs, we here add specific stream names to ESAI driver so that we can implement ASRC via DPCM to it. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_esai.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index d719caf26dc2..72d154e7dd03 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -624,12 +624,14 @@ static int fsl_esai_dai_probe(struct snd_soc_dai *dai) static struct snd_soc_dai_driver fsl_esai_dai = { .probe = fsl_esai_dai_probe, .playback = { + .stream_name = "CPU-Playback", .channels_min = 1, .channels_max = 12, .rates = FSL_ESAI_RATES, .formats = FSL_ESAI_FORMATS, }, .capture = { + .stream_name = "CPU-Capture", .channels_min = 1, .channels_max = 8, .rates = FSL_ESAI_RATES, -- cgit v1.2.3 From 20d5b76fb2c7070c70fc91b666f5395e5d16e197 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Wed, 30 Jul 2014 11:10:27 +0800 Subject: ASoC: fsl_sai: Add stream names for DPCM usage DPCM needs extra dapm routes in the machine driver to route audio between Front-End and Back-End. In order to differ the stream names in the route map from CODECs, we here add specific stream names to SAI driver so that we can implement ASRC via DPCM to it. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index c5a0e8af8226..575515d804b4 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -446,12 +446,14 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) static struct snd_soc_dai_driver fsl_sai_dai = { .probe = fsl_sai_dai_probe, .playback = { + .stream_name = "CPU-Playback", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = FSL_SAI_FORMATS, }, .capture = { + .stream_name = "CPU-Capture", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, -- cgit v1.2.3 From 756409320bcb366aa5954b4162612aa4be7e37a4 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Wed, 30 Jul 2014 11:10:28 +0800 Subject: ASoC: fsl_spdif: Add stream names for DPCM usage DPCM needs extra dapm routes in the machine driver to route audio between Front-End and Back-End. In order to differ the stream names in the route map from CODECs, we here add specific stream names to SPDIF driver so that we can implement ASRC via DPCM to it. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_spdif.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index b912d45a2a4c..5fbcd9993bd0 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -969,12 +969,14 @@ static int fsl_spdif_dai_probe(struct snd_soc_dai *dai) static struct snd_soc_dai_driver fsl_spdif_dai = { .probe = &fsl_spdif_dai_probe, .playback = { + .stream_name = "CPU-Playback", .channels_min = 2, .channels_max = 2, .rates = FSL_SPDIF_RATES_PLAYBACK, .formats = FSL_SPDIF_FORMATS_PLAYBACK, }, .capture = { + .stream_name = "CPU-Capture", .channels_min = 2, .channels_max = 2, .rates = FSL_SPDIF_RATES_CAPTURE, -- cgit v1.2.3 From e365500459095276d19a920d5be1a65d0ef9999c Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Wed, 30 Jul 2014 11:10:29 +0800 Subject: ASoC: fsl_ssi: Add stream names for DPCM usage DPCM needs extra dapm routes in the machine driver to route audio between Front-End and Back-End. In order to differ the stream names in the route map from CODECs, we here add specific stream names to SSI driver so that we can implement ASRC via DPCM to it. Signed-off-by: Nicolin Chen Acked-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 9bfef55d77d1..00660acdb805 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1032,12 +1032,14 @@ static const struct snd_soc_dai_ops fsl_ssi_dai_ops = { static struct snd_soc_dai_driver fsl_ssi_dai_template = { .probe = fsl_ssi_dai_probe, .playback = { + .stream_name = "CPU-Playback", .channels_min = 1, .channels_max = 2, .rates = FSLSSI_I2S_RATES, .formats = FSLSSI_I2S_FORMATS, }, .capture = { + .stream_name = "CPU-Capture", .channels_min = 1, .channels_max = 2, .rates = FSLSSI_I2S_RATES, -- cgit v1.2.3 From c8448051a5e7fda464486eaf5a4ef0c5b6d96673 Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Wed, 30 Jul 2014 08:04:12 +0800 Subject: ASoC: sirf-audio-codec: Fix return value check in sirf_audio_codec_driver_probe() In case of error, the function devm_ioremap_resource() returns ERR_PTR() and never returns NULL. The NULL test in the return value check should be replaced with IS_ERR(). Signed-off-by: Wei Yongjun Signed-off-by: Mark Brown --- sound/soc/codecs/sirf-audio-codec.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sirf-audio-codec.c b/sound/soc/codecs/sirf-audio-codec.c index d90cb0fafcb2..06ba4923fd5a 100644 --- a/sound/soc/codecs/sirf-audio-codec.c +++ b/sound/soc/codecs/sirf-audio-codec.c @@ -471,8 +471,8 @@ static int sirf_audio_codec_driver_probe(struct platform_device *pdev) mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); base = devm_ioremap_resource(&pdev->dev, mem_res); - if (base == NULL) - return -ENOMEM; + if (IS_ERR(base)) + return PTR_ERR(base); sirf_audio_codec->regmap = devm_regmap_init_mmio(&pdev->dev, base, &sirf_audio_codec_regmap_config); -- cgit v1.2.3 From 1a7889ca8aba333d7c74fad543d692c31bc7f280 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 31 Jul 2014 18:08:18 -0700 Subject: ASoC: rsnd: fixup SND_SOC_DAIFMT_xB_xF behavior In current R-Car rsnd driver, the SND_SOC_DAIFMT_xB_xF flags are used to HW default behavior, but, it should be used to specific format. The waveforms of LEFT_J/RIGHT_J format with SND_SOC_DAIFMT_NB_NF flag will be started from "falling edge" without this patch. But, it should be started from "rising edge". Reported-by: Jun Watanabe Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 43 ++++++++++++++++++++++--------------------- 1 file changed, 22 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index d7c669790dd4..19f78963e8b9 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -624,40 +624,41 @@ static int rsnd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - /* set clock inversion */ - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_IF: - rdai->bit_clk_inv = 0; - rdai->frm_clk_inv = 1; - break; - case SND_SOC_DAIFMT_IB_NF: - rdai->bit_clk_inv = 1; - rdai->frm_clk_inv = 0; - break; - case SND_SOC_DAIFMT_IB_IF: - rdai->bit_clk_inv = 1; - rdai->frm_clk_inv = 1; - break; - case SND_SOC_DAIFMT_NB_NF: - default: - rdai->bit_clk_inv = 0; - rdai->frm_clk_inv = 0; - break; - } - /* set format */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: rdai->sys_delay = 0; rdai->data_alignment = 0; + rdai->frm_clk_inv = 0; break; case SND_SOC_DAIFMT_LEFT_J: rdai->sys_delay = 1; rdai->data_alignment = 0; + rdai->frm_clk_inv = 1; break; case SND_SOC_DAIFMT_RIGHT_J: rdai->sys_delay = 1; rdai->data_alignment = 1; + rdai->frm_clk_inv = 1; + break; + } + + /* set clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_IF: + rdai->bit_clk_inv = rdai->bit_clk_inv; + rdai->frm_clk_inv = !rdai->frm_clk_inv; + break; + case SND_SOC_DAIFMT_IB_NF: + rdai->bit_clk_inv = !rdai->bit_clk_inv; + rdai->frm_clk_inv = rdai->frm_clk_inv; + break; + case SND_SOC_DAIFMT_IB_IF: + rdai->bit_clk_inv = !rdai->bit_clk_inv; + rdai->frm_clk_inv = !rdai->frm_clk_inv; + break; + case SND_SOC_DAIFMT_NB_NF: + default: break; } -- cgit v1.2.3 From 3e3e2922c12f29b177b18acc15655186466a92f2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 Jul 2014 12:48:36 +0100 Subject: ASoC: tlv320aic3x: Convert to params_width() The CODEC doesn't care how data is laid out in memory. Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index d275890a6827..00b127fe2b77 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -873,17 +873,16 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, /* select data word length */ data = snd_soc_read(codec, AIC3X_ASD_INTF_CTRLB) & (~(0x3 << 4)); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: data |= (0x01 << 4); break; - case SNDRV_PCM_FORMAT_S24_3LE: - case SNDRV_PCM_FORMAT_S24_LE: + case 24: data |= (0x02 << 4); break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: data |= (0x03 << 4); break; } -- cgit v1.2.3 From aaed2a62c2890c098113878c66396403f36c8ea9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 4 Mar 2014 17:20:56 +0800 Subject: ASoC: cx20442: Fix strange indentation Signed-off-by: Mark Brown --- sound/soc/codecs/cx20442.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index d5fd00a64748..1dd250f16a4e 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -383,8 +383,8 @@ static int cx20442_codec_remove(struct snd_soc_codec *codec) struct cx20442_priv *cx20442 = snd_soc_codec_get_drvdata(codec); if (cx20442->control_data) { - struct tty_struct *tty = cx20442->control_data; - tty_hangup(tty); + struct tty_struct *tty = cx20442->control_data; + tty_hangup(tty); } if (!IS_ERR(cx20442->por)) { -- cgit v1.2.3 From 486b09c750e58777976ad74a37de7b4252630332 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 1 Aug 2014 03:10:47 -0700 Subject: ASoC: rsnd: tidyup DVC control method DVC can use Volume and Mute control, and these control methods doesn't have much difference. This patch cleanup current method, and it will be used for Mute control. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/dvc.c | 59 ++++++++++++++++++++++++++++--------------------- 1 file changed, 34 insertions(+), 25 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index 9096fb03d001..12a0a2068d34 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -20,7 +20,7 @@ struct rsnd_dvc { struct rsnd_dvc_platform_info *info; /* rcar_snd.h */ struct rsnd_mod mod; struct clk *clk; - long volume[RSND_DVC_VOLUME_NUM]; + u8 volume[RSND_DVC_VOLUME_NUM]; }; #define rsnd_mod_to_dvc(_mod) \ @@ -151,12 +151,11 @@ static int rsnd_dvc_volume_info(struct snd_kcontrol *kctrl, static int rsnd_dvc_volume_get(struct snd_kcontrol *kctrl, struct snd_ctl_elem_value *ucontrol) { - struct rsnd_mod *mod = snd_kcontrol_chip(kctrl); - struct rsnd_dvc *dvc = rsnd_mod_to_dvc(mod); + u8 *val = (u8 *)kctrl->private_value; int i; for (i = 0; i < RSND_DVC_VOLUME_NUM; i++) - ucontrol->value.integer.value[i] = dvc->volume[i]; + ucontrol->value.integer.value[i] = val[i]; return 0; } @@ -165,47 +164,38 @@ static int rsnd_dvc_volume_put(struct snd_kcontrol *kctrl, struct snd_ctl_elem_value *ucontrol) { struct rsnd_mod *mod = snd_kcontrol_chip(kctrl); - struct rsnd_dvc *dvc = rsnd_mod_to_dvc(mod); + u8 *val = (u8 *)kctrl->private_value; int i, change = 0; for (i = 0; i < RSND_DVC_VOLUME_NUM; i++) { - if (ucontrol->value.integer.value[i] < 0 || - ucontrol->value.integer.value[i] > RSND_DVC_VOLUME_MAX) - return -EINVAL; - - change |= (ucontrol->value.integer.value[i] != dvc->volume[i]); + change |= (ucontrol->value.integer.value[i] != val[i]); + val[i] = ucontrol->value.integer.value[i]; } - if (change) { - for (i = 0; i < RSND_DVC_VOLUME_NUM; i++) - dvc->volume[i] = ucontrol->value.integer.value[i]; - + if (change) rsnd_dvc_volume_update(mod); - } return change; } -static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, - struct rsnd_dai *rdai, - struct snd_soc_pcm_runtime *rtd) +static int __rsnd_dvc_pcm_new(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct snd_soc_pcm_runtime *rtd, + const unsigned char *name, + u8 *private) { - struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); struct snd_card *card = rtd->card->snd_card; struct snd_kcontrol *kctrl; - static struct snd_kcontrol_new knew = { + struct snd_kcontrol_new knew = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = name, .info = rsnd_dvc_volume_info, .get = rsnd_dvc_volume_get, .put = rsnd_dvc_volume_put, + .private_value = (unsigned long)private, }; int ret; - if (rsnd_dai_is_play(rdai, io)) - knew.name = "Playback Volume"; - else - knew.name = "Capture Volume"; - kctrl = snd_ctl_new1(&knew, mod); if (!kctrl) return -ENOMEM; @@ -217,6 +207,25 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, return 0; } +static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct snd_soc_pcm_runtime *rtd) +{ + struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); + struct rsnd_dvc *dvc = rsnd_mod_to_dvc(mod); + int ret; + + /* Volume */ + ret = __rsnd_dvc_pcm_new(mod, rdai, rtd, + rsnd_dai_is_play(rdai, io) ? + "DVC Out Playback Volume" : "DVC In Capture Volume", + dvc->volume); + if (ret < 0) + return ret; + + return 0; +} + static struct rsnd_mod_ops rsnd_dvc_ops = { .name = DVC_NAME, .probe = rsnd_dvc_probe_gen2, -- cgit v1.2.3 From cd2b65741e72da64508957cd1cde85116102d8dd Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 1 Aug 2014 03:10:55 -0700 Subject: ASoC: rsnd: enable Mute control on DVC DVC can control Mute. This patch supports it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/dvc.c | 34 +++++++++++++++++++++++++++++----- 1 file changed, 29 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index 12a0a2068d34..3f443930c2b1 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -21,6 +21,7 @@ struct rsnd_dvc { struct rsnd_mod mod; struct clk *clk; u8 volume[RSND_DVC_VOLUME_NUM]; + u8 mute[RSND_DVC_VOLUME_NUM]; }; #define rsnd_mod_to_dvc(_mod) \ @@ -37,13 +38,18 @@ static void rsnd_dvc_volume_update(struct rsnd_mod *mod) struct rsnd_dvc *dvc = rsnd_mod_to_dvc(mod); u32 max = (0x00800000 - 1); u32 vol[RSND_DVC_VOLUME_NUM]; + u32 mute = 0; int i; - for (i = 0; i < RSND_DVC_VOLUME_NUM; i++) + for (i = 0; i < RSND_DVC_VOLUME_NUM; i++) { vol[i] = max / RSND_DVC_VOLUME_MAX * dvc->volume[i]; + mute |= (!!dvc->mute[i]) << i; + } rsnd_mod_write(mod, DVC_VOL0R, vol[0]); rsnd_mod_write(mod, DVC_VOL1R, vol[1]); + + rsnd_mod_write(mod, DVC_ZCMCR, mute); } static int rsnd_dvc_probe_gen2(struct rsnd_mod *mod, @@ -96,8 +102,8 @@ static int rsnd_dvc_init(struct rsnd_mod *dvc_mod, rsnd_mod_write(dvc_mod, DVC_ADINR, rsnd_get_adinr(dvc_mod)); - /* enable Volume */ - rsnd_mod_write(dvc_mod, DVC_DVUCR, 0x100); + /* enable Volume / Mute */ + rsnd_mod_write(dvc_mod, DVC_DVUCR, 0x101); /* ch0/ch1 Volume */ rsnd_dvc_volume_update(dvc_mod); @@ -140,10 +146,20 @@ static int rsnd_dvc_stop(struct rsnd_mod *mod, static int rsnd_dvc_volume_info(struct snd_kcontrol *kctrl, struct snd_ctl_elem_info *uinfo) { - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + struct rsnd_mod *mod = snd_kcontrol_chip(kctrl); + struct rsnd_dvc *dvc = rsnd_mod_to_dvc(mod); + u8 *val = (u8 *)kctrl->private_value; + uinfo->count = RSND_DVC_VOLUME_NUM; uinfo->value.integer.min = 0; - uinfo->value.integer.max = RSND_DVC_VOLUME_MAX; + + if (val == dvc->volume) { + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->value.integer.max = RSND_DVC_VOLUME_MAX; + } else { + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->value.integer.max = 1; + } return 0; } @@ -223,6 +239,14 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, if (ret < 0) return ret; + /* Mute */ + ret = __rsnd_dvc_pcm_new(mod, rdai, rtd, + rsnd_dai_is_play(rdai, io) ? + "DVC Out Mute Switch" : "DVC In Mute Switch", + dvc->mute); + if (ret < 0) + return ret; + return 0; } -- cgit v1.2.3 From 4ebd599e3570f392987df62f361d1742cc62f774 Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Fri, 1 Aug 2014 22:54:19 +0800 Subject: ASoC: Intel: Add dependency to DW_DMAC for BDW platform Add dependency to DW_DMAC for broadwell machine, which have built in DW dma engines. Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 0b305f9da3db..f5b4a9c79cdf 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -61,7 +61,7 @@ config SND_SOC_INTEL_BYT_MAX98090_MACH config SND_SOC_INTEL_BROADWELL_MACH tristate "ASoC Audio DSP support for Intel Broadwell Wildcatpoint" - depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS + depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && DW_DMAC select SND_SOC_INTEL_HASWELL select SND_COMPRESS_OFFLOAD select SND_SOC_RT286 -- cgit v1.2.3 From 97cfc751e1f2c300e093a9d2840aeee075db68d4 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 1 Aug 2014 23:08:38 +0800 Subject: ASoC: Intel: Delete message when IPC timeout occurs This fixes a bug where we dont delete the current message when an IPC message timeout occurs. Signed-off-by: Liam Girdwood Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-ipc.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 9825d195b8c9..1ca71a283761 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -502,6 +502,7 @@ static int tx_wait_done(struct sst_hsw *hsw, struct ipc_message *msg, ipc_shim_dbg(hsw, "message timeout"); trace_ipc_error("error message timeout for", msg->header); + list_del(&msg->list); ret = -ETIMEDOUT; } else { -- cgit v1.2.3 From 94ce33456dbada5cb6ae1e10bd8895f034de731d Mon Sep 17 00:00:00 2001 From: Paweł Piskorski Date: Fri, 1 Aug 2014 23:09:44 +0800 Subject: ASoC: Intel: Don't issue ipc when processing response MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Make sure we dont issue IPC when we are processing a response. Signed-off-by: Paweł Piskorski Signed-off-by: Liam Girdwood Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-ipc.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 1ca71a283761..ae204a6e316b 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -457,9 +457,10 @@ static void ipc_tx_msgs(struct kthread_work *work) return; } - /* if the DSP is busy we will TX messages after IRQ */ + /* if the DSP is busy, we will TX messages after IRQ. + * also postpone if we are in the middle of procesing completion irq*/ ipcx = sst_dsp_shim_read_unlocked(hsw->dsp, SST_IPCX); - if (ipcx & SST_IPCX_BUSY) { + if (ipcx & (SST_IPCX_BUSY | SST_IPCX_DONE)) { spin_unlock_irqrestore(&hsw->dsp->spinlock, flags); return; } -- cgit v1.2.3 From d6e08617cb0fd7e5cd9effa6ba51dd00b06a0cf1 Mon Sep 17 00:00:00 2001 From: Paweł Piskorski Date: Fri, 1 Aug 2014 23:10:43 +0800 Subject: ASoC: Intel: update stream only on stream IPC msgs MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Only update the stream when the IPC message type matches stream type. Signed-off-by: Paweł Piskorski Signed-off-by: Liam Girdwood Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-ipc.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index ae204a6e316b..b6291516dbbf 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -782,7 +782,8 @@ static int hsw_process_reply(struct sst_hsw *hsw, u32 header) } /* update any stream states */ - hsw_stream_update(hsw, msg); + if (msg_get_global_type(header) == IPC_GLB_STREAM_MESSAGE) + hsw_stream_update(hsw, msg); /* wake up and return the error if we have waiters on this message ? */ list_del(&msg->list); -- cgit v1.2.3 From ff62b95894b88622dd6373730d57a8a7e126f687 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 1 Aug 2014 17:22:19 +0100 Subject: ASoC: rt5670: Staticise non-exported symbols Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 879d42e1a1bd..31f664fb832f 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2407,7 +2407,7 @@ static int rt5670_resume(struct snd_soc_codec *codec) #define RT5670_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) -struct snd_soc_dai_ops rt5670_aif_dai_ops = { +static struct snd_soc_dai_ops rt5670_aif_dai_ops = { .hw_params = rt5670_hw_params, .set_fmt = rt5670_set_dai_fmt, .set_sysclk = rt5670_set_dai_sysclk, @@ -2415,7 +2415,7 @@ struct snd_soc_dai_ops rt5670_aif_dai_ops = { .set_pll = rt5670_set_dai_pll, }; -struct snd_soc_dai_driver rt5670_dai[] = { +static struct snd_soc_dai_driver rt5670_dai[] = { { .name = "rt5670-aif1", .id = RT5670_AIF1, @@ -2675,7 +2675,7 @@ static int rt5670_i2c_remove(struct i2c_client *i2c) return 0; } -struct i2c_driver rt5670_i2c_driver = { +static struct i2c_driver rt5670_i2c_driver = { .driver = { .name = "rt5670", .owner = THIS_MODULE, -- cgit v1.2.3 From 019575726e6d93933ed6823475d09b932be2a5fc Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 1 Aug 2014 17:30:38 +0100 Subject: ASoC: rt5670: Fix duplicate const warnings Since SOC_ENUM_SINGLE_DECL() includes a const as part of the macro adding an explicit const in the source is duplication and causes sparse to warn. Remove the extra consts. Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.c | 109 ++++++++++++++++------------------------------ 1 file changed, 37 insertions(+), 72 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 31f664fb832f..ba9d9b4d4857 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -424,10 +424,10 @@ static const char * const rt5670_data_select[] = { "Normal", "Swap", "left copy to right", "right copy to left" }; -static const SOC_ENUM_SINGLE_DECL(rt5670_if2_dac_enum, RT5670_DIG_INF1_DATA, +static SOC_ENUM_SINGLE_DECL(rt5670_if2_dac_enum, RT5670_DIG_INF1_DATA, RT5670_IF2_DAC_SEL_SFT, rt5670_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5670_if2_adc_enum, RT5670_DIG_INF1_DATA, +static SOC_ENUM_SINGLE_DECL(rt5670_if2_adc_enum, RT5670_DIG_INF1_DATA, RT5670_IF2_ADC_SEL_SFT, rt5670_data_select); static const struct snd_kcontrol_new rt5670_snd_controls[] = { @@ -786,15 +786,13 @@ static const char * const rt5670_dac1_src[] = { "IF1 DAC", "IF2 DAC" }; -static const SOC_ENUM_SINGLE_DECL( - rt5670_dac1l_enum, RT5670_AD_DA_MIXER, +static SOC_ENUM_SINGLE_DECL(rt5670_dac1l_enum, RT5670_AD_DA_MIXER, RT5670_DAC1_L_SEL_SFT, rt5670_dac1_src); static const struct snd_kcontrol_new rt5670_dac1l_mux = SOC_DAPM_ENUM("DAC1 L source", rt5670_dac1l_enum); -static const SOC_ENUM_SINGLE_DECL( - rt5670_dac1r_enum, RT5670_AD_DA_MIXER, +static SOC_ENUM_SINGLE_DECL(rt5670_dac1r_enum, RT5670_AD_DA_MIXER, RT5670_DAC1_R_SEL_SFT, rt5670_dac1_src); static const struct snd_kcontrol_new rt5670_dac1r_mux = @@ -807,8 +805,7 @@ static const char * const rt5670_dac12_src[] = { "Bass", "VAD_ADC", "IF4 DAC" }; -static const SOC_ENUM_SINGLE_DECL( - rt5670_dac2l_enum, RT5670_DAC_CTRL, +static SOC_ENUM_SINGLE_DECL(rt5670_dac2l_enum, RT5670_DAC_CTRL, RT5670_DAC2_L_SEL_SFT, rt5670_dac12_src); static const struct snd_kcontrol_new rt5670_dac_l2_mux = @@ -818,8 +815,7 @@ static const char * const rt5670_dacr2_src[] = { "IF1 DAC", "IF2 DAC", "IF3 DAC", "TxDC DAC", "TxDP ADC", "IF4 DAC" }; -static const SOC_ENUM_SINGLE_DECL( - rt5670_dac2r_enum, RT5670_DAC_CTRL, +static SOC_ENUM_SINGLE_DECL(rt5670_dac2r_enum, RT5670_DAC_CTRL, RT5670_DAC2_R_SEL_SFT, rt5670_dacr2_src); static const struct snd_kcontrol_new rt5670_dac_r2_mux = @@ -831,8 +827,7 @@ static const char * const rt5670_rxdp_src[] = { "Mono ADC Mixer L", "Mono ADC Mixer R", "DAC1" }; -static const SOC_ENUM_SINGLE_DECL( - rt5670_rxdp_enum, RT5670_DSP_PATH1, +static SOC_ENUM_SINGLE_DECL(rt5670_rxdp_enum, RT5670_DSP_PATH1, RT5670_RXDP_SEL_SFT, rt5670_rxdp_src); static const struct snd_kcontrol_new rt5670_rxdp_mux = @@ -843,15 +838,13 @@ static const char * const rt5670_dsp_bypass_src[] = { "DSP", "Bypass" }; -static const SOC_ENUM_SINGLE_DECL( - rt5670_dsp_ul_enum, RT5670_DSP_PATH1, +static SOC_ENUM_SINGLE_DECL(rt5670_dsp_ul_enum, RT5670_DSP_PATH1, RT5670_DSP_UL_SFT, rt5670_dsp_bypass_src); static const struct snd_kcontrol_new rt5670_dsp_ul_mux = SOC_DAPM_ENUM("DSP UL source", rt5670_dsp_ul_enum); -static const SOC_ENUM_SINGLE_DECL( - rt5670_dsp_dl_enum, RT5670_DSP_PATH1, +static SOC_ENUM_SINGLE_DECL(rt5670_dsp_dl_enum, RT5670_DSP_PATH1, RT5670_DSP_DL_SFT, rt5670_dsp_bypass_src); static const struct snd_kcontrol_new rt5670_dsp_dl_mux = @@ -863,8 +856,7 @@ static const char * const rt5670_stereo2_adc_lr_src[] = { "L", "LR" }; -static const SOC_ENUM_SINGLE_DECL( - rt5670_stereo2_adc_lr_enum, RT5670_STO2_ADC_MIXER, +static SOC_ENUM_SINGLE_DECL(rt5670_stereo2_adc_lr_enum, RT5670_STO2_ADC_MIXER, RT5670_STO2_ADC_SRC_SFT, rt5670_stereo2_adc_lr_src); static const struct snd_kcontrol_new rt5670_sto2_adc_lr_mux = @@ -876,8 +868,7 @@ static const char * const rt5670_stereo_adc1_src[] = { "DAC MIX", "ADC" }; -static const SOC_ENUM_SINGLE_DECL( - rt5670_stereo1_adc1_enum, RT5670_STO1_ADC_MIXER, +static SOC_ENUM_SINGLE_DECL(rt5670_stereo1_adc1_enum, RT5670_STO1_ADC_MIXER, RT5670_ADC_1_SRC_SFT, rt5670_stereo_adc1_src); static const struct snd_kcontrol_new rt5670_sto_adc_l1_mux = @@ -886,8 +877,7 @@ static const struct snd_kcontrol_new rt5670_sto_adc_l1_mux = static const struct snd_kcontrol_new rt5670_sto_adc_r1_mux = SOC_DAPM_ENUM("Stereo1 ADC R1 source", rt5670_stereo1_adc1_enum); -static const SOC_ENUM_SINGLE_DECL( - rt5670_stereo2_adc1_enum, RT5670_STO2_ADC_MIXER, +static SOC_ENUM_SINGLE_DECL(rt5670_stereo2_adc1_enum, RT5670_STO2_ADC_MIXER, RT5670_ADC_1_SRC_SFT, rt5670_stereo_adc1_src); static const struct snd_kcontrol_new rt5670_sto2_adc_l1_mux = @@ -901,8 +891,7 @@ static const char * const rt5670_stereo_adc2_src[] = { "DAC MIX", "DMIC" }; -static const SOC_ENUM_SINGLE_DECL( - rt5670_stereo1_adc2_enum, RT5670_STO1_ADC_MIXER, +static SOC_ENUM_SINGLE_DECL(rt5670_stereo1_adc2_enum, RT5670_STO1_ADC_MIXER, RT5670_ADC_2_SRC_SFT, rt5670_stereo_adc2_src); static const struct snd_kcontrol_new rt5670_sto_adc_l2_mux = @@ -911,8 +900,7 @@ static const struct snd_kcontrol_new rt5670_sto_adc_l2_mux = static const struct snd_kcontrol_new rt5670_sto_adc_r2_mux = SOC_DAPM_ENUM("Stereo1 ADC R2 source", rt5670_stereo1_adc2_enum); -static const SOC_ENUM_SINGLE_DECL( - rt5670_stereo2_adc2_enum, RT5670_STO2_ADC_MIXER, +static SOC_ENUM_SINGLE_DECL(rt5670_stereo2_adc2_enum, RT5670_STO2_ADC_MIXER, RT5670_ADC_2_SRC_SFT, rt5670_stereo_adc2_src); static const struct snd_kcontrol_new rt5670_sto2_adc_l2_mux = @@ -926,15 +914,13 @@ static const char * const rt5670_stereo_adc_src[] = { "ADC1L ADC2R", "ADC3" }; -static const SOC_ENUM_SINGLE_DECL( - rt5670_stereo1_adc_enum, RT5670_STO1_ADC_MIXER, +static SOC_ENUM_SINGLE_DECL(rt5670_stereo1_adc_enum, RT5670_STO1_ADC_MIXER, RT5670_ADC_SRC_SFT, rt5670_stereo_adc_src); static const struct snd_kcontrol_new rt5670_sto_adc_mux = SOC_DAPM_ENUM("Stereo1 ADC source", rt5670_stereo1_adc_enum); -static const SOC_ENUM_SINGLE_DECL( - rt5670_stereo2_adc_enum, RT5670_STO2_ADC_MIXER, +static SOC_ENUM_SINGLE_DECL(rt5670_stereo2_adc_enum, RT5670_STO2_ADC_MIXER, RT5670_ADC_SRC_SFT, rt5670_stereo_adc_src); static const struct snd_kcontrol_new rt5670_sto2_adc_mux = @@ -945,15 +931,13 @@ static const char * const rt5670_stereo_dmic_src[] = { "DMIC1", "DMIC2", "DMIC3" }; -static const SOC_ENUM_SINGLE_DECL( - rt5670_stereo1_dmic_enum, RT5670_STO1_ADC_MIXER, +static SOC_ENUM_SINGLE_DECL(rt5670_stereo1_dmic_enum, RT5670_STO1_ADC_MIXER, RT5670_DMIC_SRC_SFT, rt5670_stereo_dmic_src); static const struct snd_kcontrol_new rt5670_sto1_dmic_mux = SOC_DAPM_ENUM("Stereo1 DMIC source", rt5670_stereo1_dmic_enum); -static const SOC_ENUM_SINGLE_DECL( - rt5670_stereo2_dmic_enum, RT5670_STO2_ADC_MIXER, +static SOC_ENUM_SINGLE_DECL(rt5670_stereo2_dmic_enum, RT5670_STO2_ADC_MIXER, RT5670_DMIC_SRC_SFT, rt5670_stereo_dmic_src); static const struct snd_kcontrol_new rt5670_sto2_dmic_mux = @@ -964,8 +948,7 @@ static const char * const rt5670_stereo_dmic3_src[] = { "DMIC3", "PDM ADC" }; -static const SOC_ENUM_SINGLE_DECL( - rt5670_stereo_dmic3_enum, RT5670_STO1_ADC_MIXER, +static SOC_ENUM_SINGLE_DECL(rt5670_stereo_dmic3_enum, RT5670_STO1_ADC_MIXER, RT5670_DMIC3_SRC_SFT, rt5670_stereo_dmic3_src); static const struct snd_kcontrol_new rt5670_sto_dmic3_mux = @@ -977,8 +960,7 @@ static const char * const rt5670_mono_adc_l1_src[] = { "Mono DAC MIXL", "ADC1" }; -static const SOC_ENUM_SINGLE_DECL( - rt5670_mono_adc_l1_enum, RT5670_MONO_ADC_MIXER, +static SOC_ENUM_SINGLE_DECL(rt5670_mono_adc_l1_enum, RT5670_MONO_ADC_MIXER, RT5670_MONO_ADC_L1_SRC_SFT, rt5670_mono_adc_l1_src); static const struct snd_kcontrol_new rt5670_mono_adc_l1_mux = @@ -988,8 +970,7 @@ static const char * const rt5670_mono_adc_l2_src[] = { "Mono DAC MIXL", "DMIC" }; -static const SOC_ENUM_SINGLE_DECL( - rt5670_mono_adc_l2_enum, RT5670_MONO_ADC_MIXER, +static SOC_ENUM_SINGLE_DECL(rt5670_mono_adc_l2_enum, RT5670_MONO_ADC_MIXER, RT5670_MONO_ADC_L2_SRC_SFT, rt5670_mono_adc_l2_src); static const struct snd_kcontrol_new rt5670_mono_adc_l2_mux = @@ -1000,15 +981,13 @@ static const char * const rt5670_mono_dmic_src[] = { "DMIC1", "DMIC2", "DMIC3" }; -static const SOC_ENUM_SINGLE_DECL( - rt5670_mono_dmic_l_enum, RT5670_MONO_ADC_MIXER, +static SOC_ENUM_SINGLE_DECL(rt5670_mono_dmic_l_enum, RT5670_MONO_ADC_MIXER, RT5670_MONO_DMIC_L_SRC_SFT, rt5670_mono_dmic_src); static const struct snd_kcontrol_new rt5670_mono_dmic_l_mux = SOC_DAPM_ENUM("Mono DMIC left source", rt5670_mono_dmic_l_enum); /* MX-28 [1:0] */ -static const SOC_ENUM_SINGLE_DECL( - rt5670_mono_dmic_r_enum, RT5670_MONO_ADC_MIXER, +static SOC_ENUM_SINGLE_DECL(rt5670_mono_dmic_r_enum, RT5670_MONO_ADC_MIXER, RT5670_MONO_DMIC_R_SRC_SFT, rt5670_mono_dmic_src); static const struct snd_kcontrol_new rt5670_mono_dmic_r_mux = @@ -1018,8 +997,7 @@ static const char * const rt5670_mono_adc_r1_src[] = { "Mono DAC MIXR", "ADC2" }; -static const SOC_ENUM_SINGLE_DECL( - rt5670_mono_adc_r1_enum, RT5670_MONO_ADC_MIXER, +static SOC_ENUM_SINGLE_DECL(rt5670_mono_adc_r1_enum, RT5670_MONO_ADC_MIXER, RT5670_MONO_ADC_R1_SRC_SFT, rt5670_mono_adc_r1_src); static const struct snd_kcontrol_new rt5670_mono_adc_r1_mux = @@ -1029,8 +1007,7 @@ static const char * const rt5670_mono_adc_r2_src[] = { "Mono DAC MIXR", "DMIC" }; -static const SOC_ENUM_SINGLE_DECL( - rt5670_mono_adc_r2_enum, RT5670_MONO_ADC_MIXER, +static SOC_ENUM_SINGLE_DECL(rt5670_mono_adc_r2_enum, RT5670_MONO_ADC_MIXER, RT5670_MONO_ADC_R2_SRC_SFT, rt5670_mono_adc_r2_src); static const struct snd_kcontrol_new rt5670_mono_adc_r2_mux = @@ -1041,8 +1018,7 @@ static const char * const rt5670_txdp_slot_src[] = { "Slot 0-1", "Slot 2-3", "Slot 4-5", "Slot 6-7" }; -static const SOC_ENUM_SINGLE_DECL( - rt5670_txdp_slot_enum, RT5670_DSP_PATH1, +static SOC_ENUM_SINGLE_DECL(rt5670_txdp_slot_enum, RT5670_DSP_PATH1, RT5670_TXDP_SLOT_SEL_SFT, rt5670_txdp_slot_src); static const struct snd_kcontrol_new rt5670_txdp_slot_mux = @@ -1053,8 +1029,7 @@ static const char * const rt5670_if1_adc2_in_src[] = { "IF_ADC2", "VAD_ADC" }; -static const SOC_ENUM_SINGLE_DECL( - rt5670_if1_adc2_in_enum, RT5670_DIG_INF1_DATA, +static SOC_ENUM_SINGLE_DECL(rt5670_if1_adc2_in_enum, RT5670_DIG_INF1_DATA, RT5670_IF1_ADC2_IN_SFT, rt5670_if1_adc2_in_src); static const struct snd_kcontrol_new rt5670_if1_adc2_in_mux = @@ -1065,8 +1040,7 @@ static const char * const rt5670_if2_adc_in_src[] = { "IF_ADC1", "IF_ADC2", "IF_ADC3", "TxDC_DAC", "TxDP_ADC", "VAD_ADC" }; -static const SOC_ENUM_SINGLE_DECL( - rt5670_if2_adc_in_enum, RT5670_DIG_INF1_DATA, +static SOC_ENUM_SINGLE_DECL(rt5670_if2_adc_in_enum, RT5670_DIG_INF1_DATA, RT5670_IF2_ADC_IN_SFT, rt5670_if2_adc_in_src); static const struct snd_kcontrol_new rt5670_if2_adc_in_mux = @@ -1077,8 +1051,7 @@ static const char * const rt5670_if4_adc_in_src[] = { "IF_ADC1", "IF_ADC2", "IF_ADC3" }; -static const SOC_ENUM_SINGLE_DECL( - rt5670_if4_adc_in_enum, RT5670_DIG_INF2_DATA, +static SOC_ENUM_SINGLE_DECL(rt5670_if4_adc_in_enum, RT5670_DIG_INF2_DATA, RT5670_IF4_ADC_IN_SFT, rt5670_if4_adc_in_src); static const struct snd_kcontrol_new rt5670_if4_adc_in_mux = @@ -1089,29 +1062,25 @@ static const char * const rt5670_pdm_src[] = { "Mono DAC", "Stereo DAC" }; -static const SOC_ENUM_SINGLE_DECL( - rt5670_pdm1_l_enum, RT5670_PDM_OUT_CTRL, +static SOC_ENUM_SINGLE_DECL(rt5670_pdm1_l_enum, RT5670_PDM_OUT_CTRL, RT5670_PDM1_L_SFT, rt5670_pdm_src); static const struct snd_kcontrol_new rt5670_pdm1_l_mux = SOC_DAPM_ENUM("PDM1 L source", rt5670_pdm1_l_enum); -static const SOC_ENUM_SINGLE_DECL( - rt5670_pdm1_r_enum, RT5670_PDM_OUT_CTRL, +static SOC_ENUM_SINGLE_DECL(rt5670_pdm1_r_enum, RT5670_PDM_OUT_CTRL, RT5670_PDM1_R_SFT, rt5670_pdm_src); static const struct snd_kcontrol_new rt5670_pdm1_r_mux = SOC_DAPM_ENUM("PDM1 R source", rt5670_pdm1_r_enum); -static const SOC_ENUM_SINGLE_DECL( - rt5670_pdm2_l_enum, RT5670_PDM_OUT_CTRL, +static SOC_ENUM_SINGLE_DECL(rt5670_pdm2_l_enum, RT5670_PDM_OUT_CTRL, RT5670_PDM2_L_SFT, rt5670_pdm_src); static const struct snd_kcontrol_new rt5670_pdm2_l_mux = SOC_DAPM_ENUM("PDM2 L source", rt5670_pdm2_l_enum); -static const SOC_ENUM_SINGLE_DECL( - rt5670_pdm2_r_enum, RT5670_PDM_OUT_CTRL, +static SOC_ENUM_SINGLE_DECL(rt5670_pdm2_r_enum, RT5670_PDM_OUT_CTRL, RT5670_PDM2_R_SFT, rt5670_pdm_src); static const struct snd_kcontrol_new rt5670_pdm2_r_mux = @@ -1122,8 +1091,7 @@ static const char * const rt5670_if1_adc1_in1_src[] = { "IF_ADC1", "IF1_ADC3" }; -static const SOC_ENUM_SINGLE_DECL( - rt5670_if1_adc1_in1_enum, RT5670_DIG_MISC, +static SOC_ENUM_SINGLE_DECL(rt5670_if1_adc1_in1_enum, RT5670_DIG_MISC, RT5670_IF1_ADC1_IN1_SFT, rt5670_if1_adc1_in1_src); static const struct snd_kcontrol_new rt5670_if1_adc1_in1_mux = @@ -1134,8 +1102,7 @@ static const char * const rt5670_if1_adc1_in2_src[] = { "IF1_ADC1_IN1", "IF1_ADC4" }; -static const SOC_ENUM_SINGLE_DECL( - rt5670_if1_adc1_in2_enum, RT5670_DIG_MISC, +static SOC_ENUM_SINGLE_DECL(rt5670_if1_adc1_in2_enum, RT5670_DIG_MISC, RT5670_IF1_ADC1_IN2_SFT, rt5670_if1_adc1_in2_src); static const struct snd_kcontrol_new rt5670_if1_adc1_in2_mux = @@ -1146,8 +1113,7 @@ static const char * const rt5670_if1_adc2_in1_src[] = { "IF1_ADC2_IN", "IF1_ADC4" }; -static const SOC_ENUM_SINGLE_DECL( - rt5670_if1_adc2_in1_enum, RT5670_DIG_MISC, +static SOC_ENUM_SINGLE_DECL(rt5670_if1_adc2_in1_enum, RT5670_DIG_MISC, RT5670_IF1_ADC2_IN1_SFT, rt5670_if1_adc2_in1_src); static const struct snd_kcontrol_new rt5670_if1_adc2_in1_mux = @@ -1158,8 +1124,7 @@ static const char * const rt5670_vad_adc_src[] = { "Sto1 ADC L", "Mono ADC L", "Mono ADC R", "Sto2 ADC L" }; -static const SOC_ENUM_SINGLE_DECL( - rt5670_vad_adc_enum, RT5670_VAD_CTRL4, +static SOC_ENUM_SINGLE_DECL(rt5670_vad_adc_enum, RT5670_VAD_CTRL4, RT5670_VAD_SEL_SFT, rt5670_vad_adc_src); static const struct snd_kcontrol_new rt5670_vad_adc_mux = -- cgit v1.2.3 From d8df26bb57d2a86365de46a5421b97417401e39a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 1 Aug 2014 17:19:00 +0100 Subject: ASoC: adau1977: Fix truncation warning on 64 bit architectures Negating ADAU1977_BLOCK_POWER_SAI_LDO_EN creates an unsigned long constant with all bits set which on 64 bit architectures needs to be truncated to an unsigned int, generating a warning. Add an explicit cast since we know this is OK. Signed-off-by: Mark Brown Acked-by: Lars-Peter Clausen --- sound/soc/codecs/adau1977.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/adau1977.c b/sound/soc/codecs/adau1977.c index fd55da7cb9d4..70ab35744aba 100644 --- a/sound/soc/codecs/adau1977.c +++ b/sound/soc/codecs/adau1977.c @@ -968,7 +968,7 @@ int adau1977_probe(struct device *dev, struct regmap *regmap, if (adau1977->dvdd_reg) power_off_mask = ~0; else - power_off_mask = ~ADAU1977_BLOCK_POWER_SAI_LDO_EN; + power_off_mask = (unsigned int)~ADAU1977_BLOCK_POWER_SAI_LDO_EN; ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_BLOCK_POWER_SAI, power_off_mask, 0x00); -- cgit v1.2.3 From 6ccf62c7bea561cca7ffbd50839f883327080800 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Mon, 4 Aug 2014 12:19:48 +0800 Subject: ASoC: fsl_sarc_dma: Check pair before using it The patch 3117bb3109dc: "ASoC: fsl_asrc: Add ASRC ASoC CPU DAI and platform drivers" from Jul 29, 2014, leads to the following Smatch complaint: sound/soc/fsl/fsl_asrc_dma.c:304 fsl_asrc_dma_shutdown() warn: variable dereferenced before check 'pair' (see line 302) sound/soc/fsl/fsl_asrc_dma.c 301 struct fsl_asrc_pair *pair = runtime->private_data; 302 struct fsl_asrc *asrc_priv = pair->asrc_priv; ^^^^^^^^^^^^^^^ Dereference. 303 304 if (pair && asrc_priv->pair[pair->index] == pair) ^^^^ Check. 305 asrc_priv->pair[pair->index] = NULL; 306 So we just let the driver check pair before using it. Reported-by: Dan Carpenter Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_asrc_dma.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index 5b1e73e97817..ffc000bc1f15 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -299,9 +299,14 @@ static int fsl_asrc_dma_shutdown(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_asrc_pair *pair = runtime->private_data; - struct fsl_asrc *asrc_priv = pair->asrc_priv; + struct fsl_asrc *asrc_priv; + + if (!pair) + return 0; + + asrc_priv = pair->asrc_priv; - if (pair && asrc_priv->pair[pair->index] == pair) + if (asrc_priv->pair[pair->index] == pair) asrc_priv->pair[pair->index] = NULL; kfree(pair); -- cgit v1.2.3 From 4e13eb722153a5ad66edd80bc26c3028d96a7b93 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Mon, 4 Aug 2014 12:19:49 +0800 Subject: ASoC: fsl_asrc: Don't access members of config before checking it sound/soc/fsl/fsl_asrc.c:250 fsl_asrc_config_pair() warn: variable dereferenced before check 'config' (see line 243) git remote add next git://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git git remote update next git checkout 3117bb3109dc223e186302f5dc8ce9ed04adca90 vim +/config +250 sound/soc/fsl/fsl_asrc.c 237 */ 238 static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) 239 { 240 struct asrc_config *config = pair->config; 241 struct fsl_asrc *asrc_priv = pair->asrc_priv; 242 enum asrc_pair_index index = pair->index; @243 u32 inrate = config->input_sample_rate, indiv; 244 u32 outrate = config->output_sample_rate, outdiv; 245 bool ideal = config->inclk == INCLK_NONE; 246 u32 clk_index[2], div[2]; 247 int in, out, channels; 248 struct clk *clk; 249 @250 if (!config) { 251 pair_err("invalid pair config\n"); 252 return -EINVAL; 253 } Reported-by: Dan Carpenter Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_asrc.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 424b23c38d9a..822110420b71 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -240,12 +240,11 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) struct asrc_config *config = pair->config; struct fsl_asrc *asrc_priv = pair->asrc_priv; enum asrc_pair_index index = pair->index; - u32 inrate = config->input_sample_rate, indiv; - u32 outrate = config->output_sample_rate, outdiv; - bool ideal = config->inclk == INCLK_NONE; + u32 inrate, outrate, indiv, outdiv; u32 clk_index[2], div[2]; int in, out, channels; struct clk *clk; + bool ideal; if (!config) { pair_err("invalid pair config\n"); @@ -264,6 +263,10 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) return -EINVAL; } + inrate = config->input_sample_rate; + outrate = config->output_sample_rate; + ideal = config->inclk == INCLK_NONE; + /* Validate input and output sample rates */ for (in = 0; in < ARRAY_SIZE(supported_input_rate); in++) if (inrate == supported_input_rate[in]) -- cgit v1.2.3 From 37119dd791f3195ab35cb3b9cccec94bdc709c57 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 4 Aug 2014 08:56:27 +0300 Subject: ASoC: davinci: Enable menuconfig entry for McASP In order to be able to use simple-card with McASP on TI SoC based boards we need to be able to select the McASP via menuconfig. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/Kconfig | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index b310dd3489ac..d69510c53239 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -16,8 +16,14 @@ config SND_DAVINCI_SOC_I2S tristate config SND_DAVINCI_SOC_MCASP + tristate "Multichannel Audio Serial Port (McASP) support" depends on SND_DAVINCI_SOC || SND_OMAP_SOC || SND_EDMA_SOC - tristate + help + Say Y or M here if you want to have support for McASP IP found in + various Texas Instruments SoCs like: + - daVinci devices + - Sitara line of SoCs (AM335x, AM438x, etc) + - DRA7x devices config SND_DAVINCI_SOC_VCIF tristate -- cgit v1.2.3 From e5f89768e9bc1f441d18e2299518a2907e5017c9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 1 Aug 2014 17:55:55 +0100 Subject: ASoC: imx-audmux: Use uintptr_t for port numbers Since we pass the port number through file private data for debugfs we cast it to and from a pointer so use uintptr_t in order to ensure that the types are compatible, avoiding warnings on 64 bit platforms where pointers are 64 bit and unsigned integers 32 bit. Signed-off-by: Mark Brown --- sound/soc/fsl/imx-audmux.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 267717aa96c1..46f9beb6b273 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -67,7 +67,7 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, { ssize_t ret; char *buf; - int port = (int)file->private_data; + uintptr_t port = (uintptr_t)file->private_data; u32 pdcr, ptcr; if (audmux_clk) { @@ -147,7 +147,7 @@ static const struct file_operations audmux_debugfs_fops = { static void audmux_debugfs_init(void) { - int i; + uintptr_t i; char buf[20]; audmux_debugfs_root = debugfs_create_dir("audmux", NULL); @@ -157,10 +157,10 @@ static void audmux_debugfs_init(void) } for (i = 0; i < MX31_AUDMUX_PORT7_SSI_PINS_7 + 1; i++) { - snprintf(buf, sizeof(buf), "ssi%d", i); + snprintf(buf, sizeof(buf), "ssi%lu", i); if (!debugfs_create_file(buf, 0444, audmux_debugfs_root, (void *)i, &audmux_debugfs_fops)) - pr_warning("Failed to create AUDMUX port %d debugfs file\n", + pr_warning("Failed to create AUDMUX port %lu debugfs file\n", i); } } -- cgit v1.2.3