diff options
Diffstat (limited to 'sound/soc')
28 files changed, 115 insertions, 162 deletions
diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c index da2208e06b0d..5e4d499d8434 100644 --- a/sound/soc/atmel/snd-soc-afeb9260.c +++ b/sound/soc/atmel/snd-soc-afeb9260.c @@ -129,7 +129,7 @@ static struct snd_soc_dai_link afeb9260_dai = { .cpu_dai_name = "atmel-ssc-dai.0", .codec_dai_name = "tlv320aic23-hifi", .platform_name = "atmel_pcm-audio", - .codec_name = "tlv320aic23-codec.0-0x1a", + .codec_name = "tlv320aic23-codec.0-001a", .init = afeb9260_tlv320aic23_init, .ops = &afeb9260_ops, }; diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index 3abeeddc67d3..ae403597fd31 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -1,6 +1,7 @@ config SND_BF5XX_I2S tristate "SoC I2S Audio for the ADI BF5xx chip" depends on BLACKFIN + select SND_BF5XX_SOC_SPORT help Say Y or M if you want to add support for codecs attached to the Blackfin SPORT (synchronous serial ports) interface in I2S @@ -35,6 +36,7 @@ config SND_BFIN_AD73311_SE config SND_BF5XX_TDM tristate "SoC I2S(TDM mode) Audio for the ADI BF5xx chip" depends on (BLACKFIN && SND_SOC) + select SND_BF5XX_SOC_SPORT help Say Y or M if you want to add support for codecs attached to the Blackfin SPORT (synchronous serial ports) interface in TDM @@ -61,6 +63,10 @@ config SND_BF5XX_SOC_AD193X config SND_BF5XX_AC97 tristate "SoC AC97 Audio for the ADI BF5xx chip" depends on BLACKFIN + select AC97_BUS + select SND_SOC_AC97_BUS + select SND_BF5XX_SOC_SPORT + select SND_BF5XX_SOC_AC97 help Say Y or M if you want to add support for codecs attached to the Blackfin SPORT (synchronous serial ports) interface in slot 16 @@ -122,17 +128,12 @@ config SND_BF5XX_SOC_SPORT config SND_BF5XX_SOC_I2S tristate - select SND_BF5XX_SOC_SPORT config SND_BF5XX_SOC_TDM tristate - select SND_BF5XX_SOC_SPORT config SND_BF5XX_SOC_AC97 tristate - select AC97_BUS - select SND_SOC_AC97_BUS - select SND_BF5XX_SOC_SPORT config SND_BF5XX_SPORT_NUM int "Set a SPORT for Sound chip" diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index c5f856ec27ca..ffbac26b9bce 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -260,9 +260,9 @@ static int bf5xx_ac97_suspend(struct snd_soc_dai *dai) pr_debug("%s : sport %d\n", __func__, dai->id); if (!dai->active) return 0; - if (dai->capture.active) + if (dai->capture_active) sport_rx_stop(sport); - if (dai->playback.active) + if (dai->playback_active) sport_tx_stop(sport); return 0; } diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c index e902b24c1856..ad28663f5bbd 100644 --- a/sound/soc/blackfin/bf5xx-ssm2602.c +++ b/sound/soc/blackfin/bf5xx-ssm2602.c @@ -119,7 +119,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai = { .cpu_dai_name = "bf5xx-i2s", .codec_dai_name = "ssm2602-hifi", .platform_name = "bf5xx-pcm-audio", - .codec_name = "ssm2602-codec.0-0x1b", + .codec_name = "ssm2602-codec.0-001b", .ops = &bf5xx_ssm2602_ops, }; diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c index 125123929f16..5515ac9e05c7 100644 --- a/sound/soc/blackfin/bf5xx-tdm.c +++ b/sound/soc/blackfin/bf5xx-tdm.c @@ -210,7 +210,7 @@ static int bf5xx_tdm_set_channel_map(struct snd_soc_dai *dai, #ifdef CONFIG_PM static int bf5xx_tdm_suspend(struct snd_soc_dai *dai) { - struct sport_device *sport = dai->private_data; + struct sport_device *sport = snd_soc_dai_get_drvdata(dai); if (!dai->active) return 0; @@ -235,13 +235,13 @@ static int bf5xx_tdm_resume(struct snd_soc_dai *dai) ret = -EBUSY; } - ret = sport_config_rx(sport, IRFS, 0x1F, 0, 0); + ret = sport_config_rx(sport, 0, 0x1F, 0, 0); if (ret) { pr_err("SPORT is busy!\n"); ret = -EBUSY; } - ret = sport_config_tx(sport, ITFS, 0x1F, 0, 0); + ret = sport_config_tx(sport, 0, 0x1F, 0, 0); if (ret) { pr_err("SPORT is busy!\n"); ret = -EBUSY; @@ -303,14 +303,14 @@ static int __devinit bfin_tdm_probe(struct platform_device *pdev) goto sport_config_err; } - ret = sport_config_rx(sport_handle, IRFS, 0x1F, 0, 0); + ret = sport_config_rx(sport_handle, 0, 0x1F, 0, 0); if (ret) { pr_err("SPORT is busy!\n"); ret = -EBUSY; goto sport_config_err; } - ret = sport_config_tx(sport_handle, ITFS, 0x1F, 0, 0); + ret = sport_config_tx(sport_handle, 0, 0x1F, 0, 0); if (ret) { pr_err("SPORT is busy!\n"); ret = -EBUSY; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 883a312bb293..c48b23c1d4fc 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -44,7 +44,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TWL6040 if TWL4030_CORE select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C - select SND_SOC_WL1273 if WL1273_CORE + select SND_SOC_WL1273 if RADIO_WL1273 select SND_SOC_WM2000 if I2C select SND_SOC_WM8350 if MFD_WM8350 select SND_SOC_WM8400 if MFD_WM8400 diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index 46dbfd067f79..347a567b01e1 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -153,7 +153,7 @@ static int cq93vc_resume(struct snd_soc_codec *codec) static int cq93vc_probe(struct snd_soc_codec *codec) { - struct davinci_vc *davinci_vc = codec->dev->platform_data; + struct davinci_vc *davinci_vc = snd_soc_codec_get_drvdata(codec); davinci_vc->cq93vc.codec = codec; codec->control_data = davinci_vc; diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index 03d1e860d229..bb4bf65b9e7e 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -367,9 +367,12 @@ static int cx20442_codec_remove(struct snd_soc_codec *codec) return 0; } +static const u8 cx20442_reg = CX20442_TELOUT | CX20442_MIC; + static struct snd_soc_codec_driver cx20442_codec_dev = { .probe = cx20442_codec_probe, .remove = cx20442_codec_remove, + .reg_cache_default = &cx20442_reg, .reg_cache_size = 1, .reg_word_size = sizeof(u8), .read = cx20442_read_reg_cache, diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index d3ffa2f0122a..861b28f543d2 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -42,7 +42,7 @@ struct wl1273_priv { static int snd_wl1273_fm_set_i2s_mode(struct wl1273_core *core, int rate, int width) { - struct device *dev = &core->i2c_dev->dev; + struct device *dev = &core->client->dev; int r = 0; u16 mode; @@ -123,13 +123,13 @@ static int snd_wl1273_fm_set_i2s_mode(struct wl1273_core *core, dev_dbg(dev, "mode: 0x%04x\n", mode); if (core->i2s_mode != mode) { - r = wl1273_fm_write_cmd(core, WL1273_I2S_MODE_CONFIG_SET, mode); + r = core->write(core, WL1273_I2S_MODE_CONFIG_SET, mode); if (r) goto out; core->i2s_mode = mode; - r = wl1273_fm_write_cmd(core, WL1273_AUDIO_ENABLE, - WL1273_AUDIO_ENABLE_I2S); + r = core->write(core, WL1273_AUDIO_ENABLE, + WL1273_AUDIO_ENABLE_I2S); if (r) goto out; } @@ -142,8 +142,7 @@ out: static int snd_wl1273_fm_set_channel_number(struct wl1273_core *core, int channel_number) { - struct i2c_client *client = core->i2c_dev; - struct device *dev = &client->dev; + struct device *dev = &core->client->dev; int r = 0; dev_dbg(dev, "%s\n", __func__); @@ -154,17 +153,13 @@ static int snd_wl1273_fm_set_channel_number(struct wl1273_core *core, goto out; if (channel_number == 1 && core->mode == WL1273_MODE_RX) - r = wl1273_fm_write_cmd(core, WL1273_MOST_MODE_SET, - WL1273_RX_MONO); + r = core->write(core, WL1273_MOST_MODE_SET, WL1273_RX_MONO); else if (channel_number == 1 && core->mode == WL1273_MODE_TX) - r = wl1273_fm_write_cmd(core, WL1273_MONO_SET, - WL1273_TX_MONO); + r = core->write(core, WL1273_MONO_SET, WL1273_TX_MONO); else if (channel_number == 2 && core->mode == WL1273_MODE_RX) - r = wl1273_fm_write_cmd(core, WL1273_MOST_MODE_SET, - WL1273_RX_STEREO); + r = core->write(core, WL1273_MOST_MODE_SET, WL1273_RX_STEREO); else if (channel_number == 2 && core->mode == WL1273_MODE_TX) - r = wl1273_fm_write_cmd(core, WL1273_MONO_SET, - WL1273_TX_STEREO); + r = core->write(core, WL1273_MONO_SET, WL1273_TX_STEREO); else r = -EINVAL; out: @@ -237,7 +232,7 @@ static int snd_wl1273_fm_audio_put(struct snd_kcontrol *kcontrol, if (wl1273->core->audio_mode == val) return 0; - r = wl1273_fm_set_audio(wl1273->core, val); + r = wl1273->core->set_audio(wl1273->core, val); if (r < 0) return r; @@ -272,8 +267,8 @@ static int snd_wl1273_fm_volume_put(struct snd_kcontrol *kcontrol, dev_dbg(codec->dev, "%s: enter.\n", __func__); - r = wl1273_fm_set_volume(wl1273->core, - ucontrol->value.integer.value[0]); + r = wl1273->core->set_volume(wl1273->core, + ucontrol->value.integer.value[0]); if (r) return r; diff --git a/sound/soc/codecs/wl1273.h b/sound/soc/codecs/wl1273.h index 14ed027fdcfc..43ec7e668c51 100644 --- a/sound/soc/codecs/wl1273.h +++ b/sound/soc/codecs/wl1273.h @@ -25,77 +25,6 @@ #ifndef __WL1273_CODEC_H__ #define __WL1273_CODEC_H__ -/* I2S protocol, left channel first, data width 16 bits */ -#define WL1273_PCM_DEF_MODE 0x00 - -/* Rx */ -#define WL1273_AUDIO_ENABLE_I2S (1 << 0) -#define WL1273_AUDIO_ENABLE_ANALOG (1 << 1) - -/* Tx */ -#define WL1273_AUDIO_IO_SET_ANALOG 0 -#define WL1273_AUDIO_IO_SET_I2S 1 - -#define WL1273_POWER_SET_OFF 0 -#define WL1273_POWER_SET_FM (1 << 0) -#define WL1273_POWER_SET_RDS (1 << 1) -#define WL1273_POWER_SET_RETENTION (1 << 4) - -#define WL1273_PUPD_SET_OFF 0x00 -#define WL1273_PUPD_SET_ON 0x01 -#define WL1273_PUPD_SET_RETENTION 0x10 - -/* I2S mode */ -#define WL1273_IS2_WIDTH_32 0x0 -#define WL1273_IS2_WIDTH_40 0x1 -#define WL1273_IS2_WIDTH_22_23 0x2 -#define WL1273_IS2_WIDTH_23_22 0x3 -#define WL1273_IS2_WIDTH_48 0x4 -#define WL1273_IS2_WIDTH_50 0x5 -#define WL1273_IS2_WIDTH_60 0x6 -#define WL1273_IS2_WIDTH_64 0x7 -#define WL1273_IS2_WIDTH_80 0x8 -#define WL1273_IS2_WIDTH_96 0x9 -#define WL1273_IS2_WIDTH_128 0xa -#define WL1273_IS2_WIDTH 0xf - -#define WL1273_IS2_FORMAT_STD (0x0 << 4) -#define WL1273_IS2_FORMAT_LEFT (0x1 << 4) -#define WL1273_IS2_FORMAT_RIGHT (0x2 << 4) -#define WL1273_IS2_FORMAT_USER (0x3 << 4) - -#define WL1273_IS2_MASTER (0x0 << 6) -#define WL1273_IS2_SLAVEW (0x1 << 6) - -#define WL1273_IS2_TRI_AFTER_SENDING (0x0 << 7) -#define WL1273_IS2_TRI_ALWAYS_ACTIVE (0x1 << 7) - -#define WL1273_IS2_SDOWS_RR (0x0 << 8) -#define WL1273_IS2_SDOWS_RF (0x1 << 8) -#define WL1273_IS2_SDOWS_FR (0x2 << 8) -#define WL1273_IS2_SDOWS_FF (0x3 << 8) - -#define WL1273_IS2_TRI_OPT (0x0 << 10) -#define WL1273_IS2_TRI_ALWAYS (0x1 << 10) - -#define WL1273_IS2_RATE_48K (0x0 << 12) -#define WL1273_IS2_RATE_44_1K (0x1 << 12) -#define WL1273_IS2_RATE_32K (0x2 << 12) -#define WL1273_IS2_RATE_22_05K (0x4 << 12) -#define WL1273_IS2_RATE_16K (0x5 << 12) -#define WL1273_IS2_RATE_12K (0x8 << 12) -#define WL1273_IS2_RATE_11_025 (0x9 << 12) -#define WL1273_IS2_RATE_8K (0xa << 12) -#define WL1273_IS2_RATE (0xf << 12) - -#define WL1273_I2S_DEF_MODE (WL1273_IS2_WIDTH_32 | \ - WL1273_IS2_FORMAT_STD | \ - WL1273_IS2_MASTER | \ - WL1273_IS2_TRI_AFTER_SENDING | \ - WL1273_IS2_SDOWS_RR | \ - WL1273_IS2_TRI_OPT | \ - WL1273_IS2_RATE_48K) - int wl1273_get_format(struct snd_soc_codec *codec, unsigned int *fmt); #endif /* End of __WL1273_CODEC_H__ */ diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 5c87a634fc04..100aeee5ba96 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1183,7 +1183,7 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, WM8990_VMIDTOG); /* Delay to allow output caps to discharge */ - msleep(msecs_to_jiffies(300)); + msleep(300); /* Disable VMIDTOG */ snd_soc_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST | @@ -1195,17 +1195,17 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, /* Enable outputs */ snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1b00); - msleep(msecs_to_jiffies(50)); + msleep(50); /* Enable VMID at 2x50k */ snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f02); - msleep(msecs_to_jiffies(100)); + msleep(100); /* Enable VREF */ snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f03); - msleep(msecs_to_jiffies(600)); + msleep(600); /* Enable BUFIOEN */ snd_soc_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST | @@ -1250,7 +1250,7 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, /* Disable VMID */ snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f01); - msleep(msecs_to_jiffies(300)); + msleep(300); /* Enable all output discharge bits */ snd_soc_write(codec, WM8990_ANTIPOP1, WM8990_DIS_LLINE | diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 247a6a99feb8..37b8aa8a680f 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1287,9 +1287,9 @@ SND_SOC_DAPM_SUPPLY("DSPINTCLK", WM8994_CLOCKING_1, 1, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0), -SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", "AIF1 Capture", +SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", NULL, 0, WM8994_POWER_MANAGEMENT_4, 9, 0), -SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", "AIF1 Capture", +SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", NULL, 0, WM8994_POWER_MANAGEMENT_4, 8, 0), SND_SOC_DAPM_AIF_IN_E("AIF1DAC1L", NULL, 0, WM8994_POWER_MANAGEMENT_5, 9, 0, wm8958_aif_ev, @@ -1298,9 +1298,9 @@ SND_SOC_DAPM_AIF_IN_E("AIF1DAC1R", NULL, 0, WM8994_POWER_MANAGEMENT_5, 8, 0, wm8958_aif_ev, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), -SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", "AIF1 Capture", +SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", NULL, 0, WM8994_POWER_MANAGEMENT_4, 11, 0), -SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", "AIF1 Capture", +SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", NULL, 0, WM8994_POWER_MANAGEMENT_4, 10, 0), SND_SOC_DAPM_AIF_IN_E("AIF1DAC2L", NULL, 0, WM8994_POWER_MANAGEMENT_5, 11, 0, wm8958_aif_ev, @@ -1345,6 +1345,7 @@ SND_SOC_DAPM_AIF_IN_E("AIF2DACR", NULL, 0, SND_SOC_DAPM_AIF_IN("AIF1DACDAT", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("AIF2DACDAT", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_OUT("AIF1ADCDAT", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("AIF2ADCDAT", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_MUX("AIF1DAC Mux", SND_SOC_NOPM, 0, 0, &aif1dac_mux), @@ -1546,6 +1547,11 @@ static const struct snd_soc_dapm_route intercon[] = { { "AIF2DAC2R Mixer", "Left Sidetone Switch", "Left Sidetone" }, { "AIF2DAC2R Mixer", "Right Sidetone Switch", "Right Sidetone" }, + { "AIF1ADCDAT", NULL, "AIF1ADC1L" }, + { "AIF1ADCDAT", NULL, "AIF1ADC1R" }, + { "AIF1ADCDAT", NULL, "AIF1ADC2L" }, + { "AIF1ADCDAT", NULL, "AIF1ADC2R" }, + { "AIF2ADCDAT", NULL, "AIF2ADC Mux" }, /* AIF3 output */ @@ -1578,6 +1584,13 @@ static const struct snd_soc_dapm_route intercon[] = { { "Right Headphone Mux", "DAC", "DAC1R" }, }; +static const struct snd_soc_dapm_route wm8994_revd_intercon[] = { + { "AIF1DACDAT", NULL, "AIF2DACDAT" }, + { "AIF2DACDAT", NULL, "AIF1DACDAT" }, + { "AIF1ADCDAT", NULL, "AIF2ADCDAT" }, + { "AIF2ADCDAT", NULL, "AIF1ADCDAT" }, +}; + static const struct snd_soc_dapm_route wm8994_intercon[] = { { "AIF2DACL", NULL, "AIF2DAC Mux" }, { "AIF2DACR", NULL, "AIF2DAC Mux" }, @@ -2386,7 +2399,7 @@ static int wm8994_set_tristate(struct snd_soc_dai *codec_dai, int tristate) else val = 0; - return snd_soc_update_bits(codec, reg, mask, reg); + return snd_soc_update_bits(codec, reg, mask, val); } #define WM8994_RATES SNDRV_PCM_RATE_8000_96000 @@ -3129,6 +3142,11 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) case WM8994: snd_soc_dapm_add_routes(dapm, wm8994_intercon, ARRAY_SIZE(wm8994_intercon)); + + if (wm8994->revision < 4) + snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon, + ARRAY_SIZE(wm8994_revd_intercon)); + break; case WM8958: snd_soc_dapm_add_routes(dapm, wm8958_intercon, diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 6045cbde492b..608c84c5aa8e 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -1223,7 +1223,7 @@ static int wm8995_set_tristate(struct snd_soc_dai *codec_dai, int tristate) else val = 0; - return snd_soc_update_bits(codec, reg, mask, reg); + return snd_soc_update_bits(codec, reg, mask, val); } /* The size in bits of the FLL divide multiplied by 10 diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index c466982eed23..613df5db0b32 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -91,6 +91,7 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op) static void calibrate_dc_servo(struct snd_soc_codec *codec) { struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); + s8 offset; u16 reg, reg_l, reg_r, dcs_cfg; /* If we're using a digital only path and have a previously @@ -149,16 +150,14 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) hubs->dcs_codes); /* HPOUT1L */ - if (reg_l + hubs->dcs_codes > 0 && - reg_l + hubs->dcs_codes < 0xff) - reg_l += hubs->dcs_codes; - dcs_cfg = reg_l << WM8993_DCS_DAC_WR_VAL_1_SHIFT; + offset = reg_l; + offset += hubs->dcs_codes; + dcs_cfg = (u8)offset << WM8993_DCS_DAC_WR_VAL_1_SHIFT; /* HPOUT1R */ - if (reg_r + hubs->dcs_codes > 0 && - reg_r + hubs->dcs_codes < 0xff) - reg_r += hubs->dcs_codes; - dcs_cfg |= reg_r; + offset = reg_r; + offset += hubs->dcs_codes; + dcs_cfg |= (u8)offset; dev_dbg(codec->dev, "DCS result: %x\n", dcs_cfg); diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 0c2d6bacc681..fe7984221eb9 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -218,12 +218,24 @@ static struct snd_soc_dai_link dm6467_evm_dai[] = { .ops = &evm_spdif_ops, }, }; -static struct snd_soc_dai_link da8xx_evm_dai = { + +static struct snd_soc_dai_link da830_evm_dai = { + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai_name = "davinci-mcasp.1", + .codec_dai_name = "tlv320aic3x-hifi", + .codec_name = "tlv320aic3x-codec.1-0018", + .platform_name = "davinci-pcm-audio", + .init = evm_aic3x_init, + .ops = &evm_ops, +}; + +static struct snd_soc_dai_link da850_evm_dai = { .name = "TLV320AIC3X", .stream_name = "AIC3X", .cpu_dai_name= "davinci-mcasp.0", .codec_dai_name = "tlv320aic3x-hifi", - .codec_name = "tlv320aic3x-codec.0-001a", + .codec_name = "tlv320aic3x-codec.1-0018", .platform_name = "davinci-pcm-audio", .init = evm_aic3x_init, .ops = &evm_ops, @@ -259,13 +271,13 @@ static struct snd_soc_card dm6467_snd_soc_card_evm = { static struct snd_soc_card da830_snd_soc_card = { .name = "DA830/OMAP-L137 EVM", - .dai_link = &da8xx_evm_dai, + .dai_link = &da830_evm_dai, .num_links = 1, }; static struct snd_soc_card da850_snd_soc_card = { .name = "DA850/OMAP-L138 EVM", - .dai_link = &da8xx_evm_dai, + .dai_link = &da850_evm_dai, .num_links = 1, }; diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c index 9ac93f6b4f85..fff579a1c134 100644 --- a/sound/soc/ep93xx/ep93xx-i2s.c +++ b/sound/soc/ep93xx/ep93xx-i2s.c @@ -267,14 +267,16 @@ static int ep93xx_i2s_hw_params(struct snd_pcm_substream *substream, ep93xx_i2s_write_reg(info, EP93XX_I2S_RXWRDLEN, word_len); /* - * Calculate the sdiv (bit clock) and lrdiv (left/right clock) values. - * If the lrclk is pulse length is larger than the word size, then the - * bit clock will be gated for the unused bits. + * EP93xx I2S module can be setup so SCLK / LRCLK value can be + * 32, 64, 128. MCLK / SCLK value can be 2 and 4. + * We set LRCLK equal to `rate' and minimum SCLK / LRCLK + * value is 64, because our sample size is 32 bit * 2 channels. + * I2S standard permits us to transmit more bits than + * the codec uses. */ - div = (clk_get_rate(info->mclk) / params_rate(params)) * - params_channels(params); + div = clk_get_rate(info->mclk) / params_rate(params); for (sdiv = 2; sdiv <= 4; sdiv += 2) - for (lrdiv = 32; lrdiv <= 128; lrdiv <<= 1) + for (lrdiv = 64; lrdiv <= 128; lrdiv <<= 1) if (sdiv * lrdiv == div) { found = 1; goto out; @@ -341,9 +343,7 @@ static struct snd_soc_dai_ops ep93xx_i2s_dai_ops = { .set_fmt = ep93xx_i2s_set_dai_fmt, }; -#define EP93XX_I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ - SNDRV_PCM_FMTBIT_S24_LE | \ - SNDRV_PCM_FMTBIT_S32_LE) +#define EP93XX_I2S_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) static struct snd_soc_dai_driver ep93xx_i2s_dai = { .symmetric_rates= 1, diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 2101bdcee21f..3167be689621 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -507,8 +507,6 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd) /* Set up digital mute if not provided by the codec */ if (!codec_dai->driver->ops) { codec_dai->driver->ops = &ams_delta_dai_ops; - } else if (!codec_dai->driver->ops->digital_mute) { - codec_dai->driver->ops->digital_mute = ams_delta_digital_mute; } else { ams_delta_ops.startup = ams_delta_startup; ams_delta_ops.shutdown = ams_delta_shutdown; diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index fc592f0d5fc7..784cff5f67e8 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -307,10 +307,10 @@ static int corgi_wm8731_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link corgi_dai = { .name = "WM8731", .stream_name = "WM8731", - .cpu_dai_name = "pxa-is2-dai", + .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8731-hifi", .platform_name = "pxa-pcm-audio", - .codec_name = "wm8731-codec-0.001a", + .codec_name = "wm8731-codec-0.001b", .init = corgi_wm8731_init, .ops = &corgi_ops, }; diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 6298ee115e27..a7d4999f9b24 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -276,7 +276,7 @@ static struct snd_soc_dai_link poodle_dai = { .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8731-hifi", .platform_name = "pxa-pcm-audio", - .codec_name = "wm8731-codec.0-001a", + .codec_name = "wm8731-codec.0-001b", .init = poodle_wm8731_init, .ops = &poodle_ops, }; diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index c2acb69b957a..8e1571350630 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -315,10 +315,10 @@ static int spitz_wm8750_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link spitz_dai = { .name = "wm8750", .stream_name = "WM8750", - .cpu_dai_name = "pxa-is2", + .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8750-hifi", .platform_name = "pxa-pcm-audio", - .codec_name = "wm8750-codec.0-001a", + .codec_name = "wm8750-codec.0-001b", .init = spitz_wm8750_init, .ops = &spitz_ops, }; diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c index 2d4f896d7fec..3ceaef68e01d 100644 --- a/sound/soc/pxa/z2.c +++ b/sound/soc/pxa/z2.c @@ -104,6 +104,7 @@ static struct snd_soc_jack_gpio hs_jack_gpios[] = { .name = "hsdet-gpio", .report = SND_JACK_HEADSET, .debounce_time = 200, + .invert = 1, }, }; @@ -192,7 +193,7 @@ static struct snd_soc_dai_link z2_dai = { .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8750-hifi", .platform_name = "pxa-pcm-audio", - .codec_name = "wm8750-codec.0-001a", + .codec_name = "wm8750-codec.0-001b", .init = z2_wm8750_init, .ops = &z2_ops, }; diff --git a/sound/soc/samsung/neo1973_gta02_wm8753.c b/sound/soc/samsung/neo1973_gta02_wm8753.c index 3eec610c10f9..0d0ae2b9eef6 100644 --- a/sound/soc/samsung/neo1973_gta02_wm8753.c +++ b/sound/soc/samsung/neo1973_gta02_wm8753.c @@ -397,11 +397,11 @@ static struct snd_soc_dai_link neo1973_gta02_dai[] = { { /* Hifi Playback - for similatious use with voice below */ .name = "WM8753", .stream_name = "WM8753 HiFi", - .cpu_dai_name = "s3c24xx-i2s", + .cpu_dai_name = "s3c24xx-iis", .codec_dai_name = "wm8753-hifi", .init = neo1973_gta02_wm8753_init, .platform_name = "samsung-audio", - .codec_name = "wm8753-codec.0-0x1a", + .codec_name = "wm8753-codec.0-001a", .ops = &neo1973_gta02_hifi_ops, }, { /* Voice via BT */ @@ -410,7 +410,7 @@ static struct snd_soc_dai_link neo1973_gta02_dai[] = { .cpu_dai_name = "bluetooth-dai", .codec_dai_name = "wm8753-voice", .ops = &neo1973_gta02_voice_ops, - .codec_name = "wm8753-codec.0-0x1a", + .codec_name = "wm8753-codec.0-001a", .platform_name = "samsung-audio", }, }; diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index c7a24514beb5..d20815d5ab2e 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -559,9 +559,9 @@ static struct snd_soc_dai_link neo1973_dai[] = { .name = "WM8753", .stream_name = "WM8753 HiFi", .platform_name = "samsung-audio", - .cpu_dai_name = "s3c24xx-i2s", + .cpu_dai_name = "s3c24xx-iis", .codec_dai_name = "wm8753-hifi", - .codec_name = "wm8753-codec.0-0x1a", + .codec_name = "wm8753-codec.0-001a", .init = neo1973_wm8753_init, .ops = &neo1973_hifi_ops, }, @@ -571,7 +571,7 @@ static struct snd_soc_dai_link neo1973_dai[] = { .platform_name = "samsung-audio", .cpu_dai_name = "bluetooth-dai", .codec_dai_name = "wm8753-voice", - .codec_name = "wm8753-codec.0-0x1a", + .codec_name = "wm8753-codec.0-001a", .ops = &neo1973_voice_ops, }, }; diff --git a/sound/soc/samsung/s3c24xx_simtec_hermes.c b/sound/soc/samsung/s3c24xx_simtec_hermes.c index bb4292e3596c..08fcaaa66907 100644 --- a/sound/soc/samsung/s3c24xx_simtec_hermes.c +++ b/sound/soc/samsung/s3c24xx_simtec_hermes.c @@ -94,8 +94,8 @@ static int simtec_hermes_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link simtec_dai_aic33 = { .name = "tlv320aic33", .stream_name = "TLV320AIC33", - .codec_name = "tlv320aic3x-codec.0-0x1a", - .cpu_dai_name = "s3c24xx-i2s", + .codec_name = "tlv320aic3x-codec.0-001a", + .cpu_dai_name = "s3c24xx-iis", .codec_dai_name = "tlv320aic3x-hifi", .platform_name = "samsung-audio", .init = simtec_hermes_init, diff --git a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c index fbba4e367729..116e3e670167 100644 --- a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c +++ b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c @@ -85,8 +85,8 @@ static int simtec_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link simtec_dai_aic23 = { .name = "tlv320aic23", .stream_name = "TLV320AIC23", - .codec_name = "tlv320aic3x-codec.0-0x1a", - .cpu_dai_name = "s3c24xx-i2s", + .codec_name = "tlv320aic3x-codec.0-001a", + .cpu_dai_name = "s3c24xx-iis", .codec_dai_name = "tlv320aic3x-hifi", .platform_name = "samsung-audio", .init = simtec_tlv320aic23_init, diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c index cdc8ecbcb8ef..2c09e93dd566 100644 --- a/sound/soc/samsung/s3c24xx_uda134x.c +++ b/sound/soc/samsung/s3c24xx_uda134x.c @@ -228,7 +228,7 @@ static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = { .stream_name = "UDA134X", .codec_name = "uda134x-hifi", .codec_dai_name = "uda134x-hifi", - .cpu_dai_name = "s3c24xx-i2s", + .cpu_dai_name = "s3c24xx-iis", .ops = &s3c24xx_uda134x_ops, .platform_name = "samsung-audio", }; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index bac7291b6ff6..c3f6f1e72790 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1449,6 +1449,7 @@ static int soc_post_component_init(struct snd_soc_card *card, rtd = &card->rtd_aux[num]; name = aux_dev->name; } + rtd->card = card; /* machine controls, routes and widgets are not prefixed */ temp = codec->name_prefix; @@ -1471,7 +1472,6 @@ static int soc_post_component_init(struct snd_soc_card *card, /* register the rtd device */ rtd->codec = codec; - rtd->card = card; rtd->dev.parent = card->dev; rtd->dev.release = rtd_release; rtd->dev.init_name = name; @@ -1664,9 +1664,6 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num) goto out; found: - if (!try_module_get(codec->dev->driver->owner)) - return -ENODEV; - ret = soc_probe_codec(card, codec); if (ret < 0) return ret; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 499730ab5638..8194f150bab7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1742,7 +1742,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; - unsigned int val, val_mask; + unsigned int val; int connect, change; struct snd_soc_dapm_update update; @@ -1750,13 +1750,13 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, if (invert) val = max - val; - val_mask = mask << shift; + mask = mask << shift; val = val << shift; mutex_lock(&widget->codec->mutex); widget->value = val; - change = snd_soc_test_bits(widget->codec, reg, val_mask, val); + change = snd_soc_test_bits(widget->codec, reg, mask, val); if (change) { if (val) /* new connection */ |