From 64895739835c431fbc62dfabbe2267fff552a2f2 Mon Sep 17 00:00:00 2001 From: Nenghua Cao Date: Mon, 3 Mar 2014 19:08:23 +0800 Subject: asoc: soc-core: fix coccinelle warnings sound/soc/soc-core.c:2708:6-13: WARNING: Assignment of bool to 0/1 sound/soc/soc-core.c:2726:3-10: WARNING: Assignment of bool to 0/1 More information about semantic patching is available at http://coccinelle.lip6.fr/ Signed-off-by: Nenghua Cao Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index fe1df50805a3..3477525247c3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2818,7 +2818,7 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; int err; - bool type_2r = 0; + bool type_2r = false; unsigned int val2 = 0; unsigned int val, val_mask; @@ -2836,7 +2836,7 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, val |= val2 << rshift; } else { val2 = val2 << shift; - type_2r = 1; + type_2r = true; } } err = snd_soc_update_bits_locked(codec, reg, val_mask, val); -- cgit v1.2.3 From 7f0af4ae86110c46b43fb57f2e0002aa81ca6a67 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 12 Mar 2014 15:27:32 +0100 Subject: ASoC: omap-abe-twl6040: Register machine level DMIC DAPM routes with the card Machine level DAPM widgets and routes should be registered in the card's DAPM context, rather than in the CODEC's context. Signed-off-by: Lars-Peter Clausen Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/omap-abe-twl6040.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index ebb13906b3a0..024dafc3e298 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -203,8 +203,7 @@ static const struct snd_soc_dapm_route dmic_audio_map[] = { static int omap_abe_dmic_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_dapm_context *dapm = &rtd->card->dapm; return snd_soc_dapm_add_routes(dapm, dmic_audio_map, ARRAY_SIZE(dmic_audio_map)); -- cgit v1.2.3 From d343a660d3c94423759382869b7f90c8657ebbf1 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 12 Mar 2014 15:27:35 +0100 Subject: ASoC: davinci-evm: Register machine level DAPM elements with the card Machine level DAPM widgets and routes should be registered in the card's DAPM context, rather than in the CODEC's context. While we are at it also drop the snd_soc_dapm_enable_pin() calls, since pins are enabled by default and also turn the snd_soc_dapm_disable_pin() calls into snd_soc_dapm_nc_pin() calls for unconnected pins. Signed-off-by: Lars-Peter Clausen Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-evm.c | 22 ++++++++-------------- 1 file changed, 8 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 621e9a997d4c..cab98a580053 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -123,35 +123,29 @@ static const struct snd_soc_dapm_route audio_map[] = { /* Logic for a aic3x as connected on a davinci-evm */ static int evm_aic3x_init(struct snd_soc_pcm_runtime *rtd) { + struct snd_soc_card *card = rtd->card; struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; struct device_node *np = codec->card->dev->of_node; int ret; /* Add davinci-evm specific widgets */ - snd_soc_dapm_new_controls(dapm, aic3x_dapm_widgets, + snd_soc_dapm_new_controls(&card->dapm, aic3x_dapm_widgets, ARRAY_SIZE(aic3x_dapm_widgets)); if (np) { - ret = snd_soc_of_parse_audio_routing(codec->card, - "ti,audio-routing"); + ret = snd_soc_of_parse_audio_routing(card, "ti,audio-routing"); if (ret) return ret; } else { /* Set up davinci-evm specific audio path audio_map */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_add_routes(&card->dapm, audio_map, + ARRAY_SIZE(audio_map)); } /* not connected */ - snd_soc_dapm_disable_pin(dapm, "MONO_LOUT"); - snd_soc_dapm_disable_pin(dapm, "HPLCOM"); - snd_soc_dapm_disable_pin(dapm, "HPRCOM"); - - /* always connected */ - snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); - snd_soc_dapm_enable_pin(dapm, "Line Out"); - snd_soc_dapm_enable_pin(dapm, "Mic Jack"); - snd_soc_dapm_enable_pin(dapm, "Line In"); + snd_soc_dapm_nc_pin(&codec->dapm, "MONO_LOUT"); + snd_soc_dapm_nc_pin(&codec->dapm, "HPLCOM"); + snd_soc_dapm_nc_pin(&codec->dapm, "HPRCOM"); return 0; } -- cgit v1.2.3 From e585ca342dbbfe7102985d9ed4eae3f9e1d77ced Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 14 Mar 2014 09:33:36 +0100 Subject: ASoC: max98090: Remove unused control_data field The driver assigns a value to the control_data field of the driver's state struct, but never reads it again. Which means it is unused and can be removed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/max98090.c | 1 - sound/soc/codecs/max98090.h | 1 - 2 files changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index c7b9e901bdac..361862d4fa65 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2350,7 +2350,6 @@ static int max98090_i2c_probe(struct i2c_client *i2c, max98090->devtype = id->driver_data; i2c_set_clientdata(i2c, max98090); - max98090->control_data = i2c; max98090->pdata = i2c->dev.platform_data; max98090->irq = i2c->irq; diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h index 7e103f249053..1a4e2334a7b2 100644 --- a/sound/soc/codecs/max98090.h +++ b/sound/soc/codecs/max98090.h @@ -1523,7 +1523,6 @@ struct max98090_priv { struct regmap *regmap; struct snd_soc_codec *codec; enum max98090_type devtype; - void *control_data; struct max98090_pdata *pdata; unsigned int sysclk; unsigned int bclk; -- cgit v1.2.3 From c7a507eea1db1430476289f525f9c853d5d485e8 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 13 Mar 2014 17:56:25 -0700 Subject: ASoC: fsi: fixup SND_SOC_DAIFMT_CBx_CFx flags SND_SOC_DAIFMT_CBx_CFx means "codec" side master/slave mode. Then, FSI will be master mode if it was SND_SOC_DAIFMT_CBS_CFS. This patch fixup platform settings too. Then, it tidyups SND_SOC_DAIFMT_INV settings. Acked-by: Simon Horman 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 | 4 +--- arch/arm/mach-shmobile/board-mackerel.c | 6 ++---- arch/sh/boards/mach-ecovec24/setup.c | 4 +--- arch/sh/boards/mach-se/7724/setup.c | 4 +--- sound/soc/sh/fsi.c | 2 +- 6 files changed, 7 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/arch/arm/mach-shmobile/board-armadillo800eva.c b/arch/arm/mach-shmobile/board-armadillo800eva.c index 93533e2710a8..9323854242ca 100644 --- a/arch/arm/mach-shmobile/board-armadillo800eva.c +++ b/arch/arm/mach-shmobile/board-armadillo800eva.c @@ -988,14 +988,12 @@ static struct asoc_simple_card_info fsi_wm8978_info = { .card = "FSI2A-WM8978", .codec = "wm8978.0-001a", .platform = "sh_fsi2", - .daifmt = SND_SOC_DAIFMT_I2S, + .daifmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, .cpu_dai = { .name = "fsia-dai", - .fmt = SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_IB_NF, }, .codec_dai = { .name = "wm8978-hifi", - .fmt = SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_NB_NF, .sysclk = 12288000, }, }; diff --git a/arch/arm/mach-shmobile/board-kzm9g.c b/arch/arm/mach-shmobile/board-kzm9g.c index bc40b853ffd3..03dc3ac84502 100644 --- a/arch/arm/mach-shmobile/board-kzm9g.c +++ b/arch/arm/mach-shmobile/board-kzm9g.c @@ -589,14 +589,12 @@ static struct asoc_simple_card_info fsi2_ak4648_info = { .card = "FSI2A-AK4648", .codec = "ak4642-codec.0-0012", .platform = "sh_fsi2", - .daifmt = SND_SOC_DAIFMT_LEFT_J, + .daifmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBM_CFM, .cpu_dai = { .name = "fsia-dai", - .fmt = SND_SOC_DAIFMT_CBS_CFS, }, .codec_dai = { .name = "ak4642-hifi", - .fmt = SND_SOC_DAIFMT_CBM_CFM, .sysclk = 11289600, }, }; diff --git a/arch/arm/mach-shmobile/board-mackerel.c b/arch/arm/mach-shmobile/board-mackerel.c index 3aba0372f630..8c4332182b42 100644 --- a/arch/arm/mach-shmobile/board-mackerel.c +++ b/arch/arm/mach-shmobile/board-mackerel.c @@ -509,9 +509,9 @@ static struct asoc_simple_card_info fsi2_hdmi_info = { .card = "FSI2B-HDMI", .codec = "sh-mobile-hdmi", .platform = "sh_fsi2", + .fmt = SND_SOC_DAIFMT_CBS_CFS, .cpu_dai = { .name = "fsib-dai", - .fmt = SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF, }, .codec_dai = { .name = "sh_mobile_hdmi-hifi", @@ -905,14 +905,12 @@ static struct asoc_simple_card_info fsi2_ak4643_info = { .card = "FSI2A-AK4643", .codec = "ak4642-codec.0-0013", .platform = "sh_fsi2", - .daifmt = SND_SOC_DAIFMT_LEFT_J, + .daifmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBM_CFM, .cpu_dai = { .name = "fsia-dai", - .fmt = SND_SOC_DAIFMT_CBS_CFS, }, .codec_dai = { .name = "ak4642-hifi", - .fmt = SND_SOC_DAIFMT_CBM_CFM, .sysclk = 11289600, }, }; diff --git a/arch/sh/boards/mach-ecovec24/setup.c b/arch/sh/boards/mach-ecovec24/setup.c index 5bc3a15465c7..85d5255d259f 100644 --- a/arch/sh/boards/mach-ecovec24/setup.c +++ b/arch/sh/boards/mach-ecovec24/setup.c @@ -861,14 +861,12 @@ static struct asoc_simple_card_info fsi_da7210_info = { .card = "FSIB-DA7210", .codec = "da7210.0-001a", .platform = "sh_fsi.0", - .daifmt = SND_SOC_DAIFMT_I2S, + .daifmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, .cpu_dai = { .name = "fsib-dai", - .fmt = SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_IB_NF, }, .codec_dai = { .name = "da7210-hifi", - .fmt = SND_SOC_DAIFMT_CBM_CFM, }, }; diff --git a/arch/sh/boards/mach-se/7724/setup.c b/arch/sh/boards/mach-se/7724/setup.c index 21e4230659a5..1162bc6945a3 100644 --- a/arch/sh/boards/mach-se/7724/setup.c +++ b/arch/sh/boards/mach-se/7724/setup.c @@ -304,14 +304,12 @@ static struct asoc_simple_card_info fsi_ak4642_info = { .card = "FSIA-AK4642", .codec = "ak4642-codec.0-0012", .platform = "sh_fsi.0", - .daifmt = SND_SOC_DAIFMT_LEFT_J, + .daifmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBM_CFM, .cpu_dai = { .name = "fsia-dai", - .fmt = SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_IB_NF, }, .codec_dai = { .name = "ak4642-hifi", - .fmt = SND_SOC_DAIFMT_CBM_CFM, .sysclk = 11289600, }, }; diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 1967f44e7cd4..710a079a7377 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1711,9 +1711,9 @@ static int fsi_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: - fsi->clk_master = 1; break; case SND_SOC_DAIFMT_CBS_CFS: + fsi->clk_master = 1; /* codec is slave, cpu is master */ break; default: return -EINVAL; -- cgit v1.2.3 From e1508289404ab6ca28e0dc931612600f0441c417 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 13 Mar 2014 17:56:43 -0700 Subject: ASoC: rcar: fixup SND_SOC_DAIFMT_CBx_CFx flags SND_SOC_DAIFMT_CBx_CFx means "codec" side master/slave mode. Then, rcar will be master mode if it was SND_SOC_DAIFMT_CBS_CFS. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 4 ++-- 1 file 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 743de5e3b1e1..3a4fe9d0d4f2 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -486,10 +486,10 @@ static int rsnd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: - rdai->clk_master = 1; + rdai->clk_master = 0; break; case SND_SOC_DAIFMT_CBS_CFS: - rdai->clk_master = 0; + rdai->clk_master = 1; /* codec is slave, cpu is master */ break; default: return -EINVAL; -- cgit v1.2.3 From e512e001dafa54e5ac7244416e340750a4356b41 Mon Sep 17 00:00:00 2001 From: Jean-Francois Moine Date: Tue, 11 Mar 2014 10:03:40 +0100 Subject: ASoC: simple-card: Fix the reference count of device nodes The reference count of some device nodes is not correctly reset at end of card probe. Signed-off-by: Jean-Francois Moine Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 50 ++++++++++++++++++++++++++++++----------- 1 file changed, 37 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 5dd47691ba41..dcf37fb69b35 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -105,12 +105,12 @@ asoc_simple_card_sub_parse_of(struct device_node *np, /* get dai->name */ ret = snd_soc_of_get_dai_name(np, name); if (ret < 0) - goto parse_error; + return ret; /* parse TDM slot */ ret = snd_soc_of_parse_tdm_slot(np, &dai->slots, &dai->slot_width); if (ret) - goto parse_error; + return ret; /* * bitclock-inversion, frame-inversion @@ -130,7 +130,7 @@ asoc_simple_card_sub_parse_of(struct device_node *np, clk = of_clk_get(np, 0); if (IS_ERR(clk)) { ret = PTR_ERR(clk); - goto parse_error; + return ret; } dai->sysclk = clk_get_rate(clk); @@ -144,12 +144,7 @@ asoc_simple_card_sub_parse_of(struct device_node *np, dai->sysclk = clk_get_rate(clk); } - ret = 0; - -parse_error: - of_node_put(node); - - return ret; + return 0; } static int asoc_simple_card_parse_of(struct device_node *node, @@ -187,22 +182,26 @@ static int asoc_simple_card_parse_of(struct device_node *node, /* CPU sub-node */ ret = -EINVAL; np = of_get_child_by_name(node, "simple-audio-card,cpu"); - if (np) + if (np) { ret = asoc_simple_card_sub_parse_of(np, priv->daifmt, &priv->cpu_dai, &dai_link->cpu_of_node, &dai_link->cpu_dai_name); + of_node_put(np); + } if (ret < 0) return ret; /* CODEC sub-node */ ret = -EINVAL; np = of_get_child_by_name(node, "simple-audio-card,codec"); - if (np) + if (np) { ret = asoc_simple_card_sub_parse_of(np, priv->daifmt, &priv->codec_dai, &dai_link->codec_of_node, &dai_link->codec_dai_name); + of_node_put(np); + } if (ret < 0) return ret; @@ -248,6 +247,27 @@ static int asoc_simple_card_parse_of(struct device_node *node, return 0; } +/* update the reference count of the devices nodes at end of probe */ +static int asoc_simple_card_unref(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct snd_soc_dai_link *dai_link; + struct device_node *np; + int num_links; + + for (num_links = 0, dai_link = card->dai_link; + num_links < card->num_links; + num_links++, dai_link++) { + np = (struct device_node *) dai_link->cpu_of_node; + if (np) + of_node_put(np); + np = (struct device_node *) dai_link->codec_of_node; + if (np) + of_node_put(np); + } + return 0; +} + static int asoc_simple_card_probe(struct platform_device *pdev) { struct simple_card_data *priv; @@ -275,7 +295,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) if (ret < 0) { if (ret != -EPROBE_DEFER) dev_err(dev, "parse error %d\n", ret); - return ret; + goto err; } } else { struct asoc_simple_card_info *cinfo; @@ -318,7 +338,11 @@ static int asoc_simple_card_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(&priv->snd_card, priv); - return devm_snd_soc_register_card(&pdev->dev, &priv->snd_card); + ret = devm_snd_soc_register_card(&pdev->dev, &priv->snd_card); + +err: + asoc_simple_card_unref(pdev); + return ret; } static const struct of_device_id asoc_simple_of_match[] = { -- cgit v1.2.3 From fd218aa3e5d4ee522cbfe88ad4dd83eb891096fb Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Mon, 17 Mar 2014 09:31:31 +0200 Subject: ASoC: tlv320aic31xx: Turn power off only once. Regulator code keep count of enables and disables. Double disable causes an ugly warning. Signed-off-by: Jyri Sarha Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic31xx.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index d3517a919776..1f243c2c98fd 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1020,7 +1020,8 @@ static int aic31xx_set_bias_level(struct snd_soc_codec *codec, } break; case SND_SOC_BIAS_OFF: - aic31xx_power_off(codec); + if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) + aic31xx_power_off(codec); break; } codec->dapm.bias_level = level; -- cgit v1.2.3 From 58726f58a6432495bd71d1fc7c9454cbc85f5a5c Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 14 Mar 2014 16:42:45 +0200 Subject: ASoC: davinci: Add edma dmaengine platform driver Platform driver glue for SoC using eDMA3 to use dmaengine PCM. The maximum number of periods need to be limited to 19 since the edma dmaengine driver limits the paRAM slot use for audio at in cyclic mode. Signed-off-by: Peter Ujfalusi Reviewed-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/davinci/edma-pcm.c | 57 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/davinci/edma-pcm.h | 25 +++++++++++++++++++ 2 files changed, 82 insertions(+) create mode 100644 sound/soc/davinci/edma-pcm.c create mode 100644 sound/soc/davinci/edma-pcm.h (limited to 'sound') diff --git a/sound/soc/davinci/edma-pcm.c b/sound/soc/davinci/edma-pcm.c new file mode 100644 index 000000000000..d38afb1c61ae --- /dev/null +++ b/sound/soc/davinci/edma-pcm.c @@ -0,0 +1,57 @@ +/* + * edma-pcm.c - eDMA PCM driver using dmaengine for AM3xxx, AM4xxx + * + * Copyright (C) 2014 Texas Instruments, Inc. + * + * Author: Peter Ujfalusi + * + * Based on: sound/soc/tegra/tegra_pcm.c + * + * 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 + +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_INTERLEAVED, + .buffer_bytes_max = 128 * 1024, + .period_bytes_min = 32, + .period_bytes_max = 64 * 1024, + .periods_min = 2, + .periods_max = 19, /* Limit by edma dmaengine driver */ +}; + +static const struct snd_dmaengine_pcm_config edma_dmaengine_pcm_config = { + .pcm_hardware = &edma_pcm_hardware, + .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, + .compat_filter_fn = edma_filter_fn, + .prealloc_buffer_size = 128 * 1024, +}; + +int edma_pcm_platform_register(struct device *dev) +{ + return devm_snd_dmaengine_pcm_register(dev, &edma_dmaengine_pcm_config, + SND_DMAENGINE_PCM_FLAG_COMPAT); +} +EXPORT_SYMBOL_GPL(edma_pcm_platform_register); + +MODULE_AUTHOR("Peter Ujfalusi "); +MODULE_DESCRIPTION("eDMA PCM ASoC platform driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/davinci/edma-pcm.h b/sound/soc/davinci/edma-pcm.h new file mode 100644 index 000000000000..894c378c0f74 --- /dev/null +++ b/sound/soc/davinci/edma-pcm.h @@ -0,0 +1,25 @@ +/* + * edma-pcm.h - eDMA PCM driver using dmaengine for AM3xxx, AM4xxx + * + * Copyright (C) 2014 Texas Instruments, Inc. + * + * Author: Peter Ujfalusi + * + * Based on: sound/soc/tegra/tegra_pcm.h + * + * 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 __EDMA_PCM_H__ +#define __EDMA_PCM_H__ + +int edma_pcm_platform_register(struct device *dev); + +#endif /* __EDMA_PCM_H__ */ -- cgit v1.2.3 From 8de131f2e6de83f23059527fc4732c1efe27f93d Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 14 Mar 2014 16:42:46 +0200 Subject: ASoC: davinci-mcasp: Provide correct filter_data for dmaengine for non-DT boot When we boot with non-DT mode the damengine will need the channel number and a filter function in order to get the channel. The filter_data is filled in the DAI driver while the filter_function will be provided by the edma-pcm driver. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 21 ++++++++++++++++----- 1 file changed, 16 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index b0ae0677f023..a01ae97c90aa 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1026,6 +1026,7 @@ nodata: static int davinci_mcasp_probe(struct platform_device *pdev) { struct davinci_pcm_dma_params *dma_params; + struct snd_dmaengine_dai_dma_data *dma_data; struct resource *mem, *ioarea, *res, *dat; struct davinci_mcasp_pdata *pdata; struct davinci_mcasp *mcasp; @@ -1095,6 +1096,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) mcasp->dat_port = true; dma_params = &mcasp->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; + dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK]; dma_params->asp_chan_q = pdata->asp_chan_q; dma_params->ram_chan_q = pdata->ram_chan_q; dma_params->sram_pool = pdata->sram_pool; @@ -1105,7 +1107,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dma_params->dma_addr = mem->start + pdata->tx_dma_offset; /* Unconditional dmaengine stuff */ - mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr = dma_params->dma_addr; + dma_data->addr = dma_params->dma_addr; res = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (res) @@ -1113,7 +1115,14 @@ static int davinci_mcasp_probe(struct platform_device *pdev) else dma_params->channel = pdata->tx_dma_channel; + /* dmaengine filter data for DT and non-DT boot */ + if (pdev->dev.of_node) + dma_data->filter_data = "tx"; + else + dma_data->filter_data = &dma_params->channel; + dma_params = &mcasp->dma_params[SNDRV_PCM_STREAM_CAPTURE]; + dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE]; dma_params->asp_chan_q = pdata->asp_chan_q; dma_params->ram_chan_q = pdata->ram_chan_q; dma_params->sram_pool = pdata->sram_pool; @@ -1124,7 +1133,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dma_params->dma_addr = mem->start + pdata->rx_dma_offset; /* Unconditional dmaengine stuff */ - mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE].addr = dma_params->dma_addr; + dma_data->addr = dma_params->dma_addr; if (mcasp->version < MCASP_VERSION_3) { mcasp->fifo_base = DAVINCI_MCASP_V2_AFIFO_BASE; @@ -1140,9 +1149,11 @@ static int davinci_mcasp_probe(struct platform_device *pdev) else dma_params->channel = pdata->rx_dma_channel; - /* Unconditional dmaengine stuff */ - mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK].filter_data = "tx"; - mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE].filter_data = "rx"; + /* dmaengine filter data for DT and non-DT boot */ + if (pdev->dev.of_node) + dma_data->filter_data = "rx"; + else + dma_data->filter_data = &dma_params->channel; dev_set_drvdata(&pdev->dev, mcasp); -- cgit v1.2.3 From c56c4d74c6f96d0ff605d8948e127099cf5e6681 Mon Sep 17 00:00:00 2001 From: Jean-Francois Moine Date: Sat, 15 Mar 2014 11:32:42 +0100 Subject: ASoC: simple-card: Simplify code The global DAI format is used only in the function asoc_simple_card_parse_of(). So, move it from the private data to the stack. Signed-off-by: Jean-Francois Moine Reviewed-by: Xiubo Li Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index dcf37fb69b35..ca7e63ef858a 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -20,7 +20,6 @@ struct simple_card_data { struct snd_soc_card snd_card; - unsigned int daifmt; struct asoc_simple_dai cpu_dai; struct asoc_simple_dai codec_dai; struct snd_soc_dai_link snd_link; @@ -154,13 +153,14 @@ static int asoc_simple_card_parse_of(struct device_node *node, struct snd_soc_dai_link *dai_link = priv->snd_card.dai_link; struct device_node *np; char *name; + unsigned int daifmt; int ret; /* parsing the card name from DT */ snd_soc_of_parse_card_name(&priv->snd_card, "simple-audio-card,name"); /* get CPU/CODEC common format via simple-audio-card,format */ - priv->daifmt = snd_soc_of_parse_daifmt(node, "simple-audio-card,") & + daifmt = snd_soc_of_parse_daifmt(node, "simple-audio-card,") & (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK); /* off-codec widgets */ @@ -183,7 +183,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, ret = -EINVAL; np = of_get_child_by_name(node, "simple-audio-card,cpu"); if (np) { - ret = asoc_simple_card_sub_parse_of(np, priv->daifmt, + ret = asoc_simple_card_sub_parse_of(np, daifmt, &priv->cpu_dai, &dai_link->cpu_of_node, &dai_link->cpu_dai_name); @@ -196,7 +196,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, ret = -EINVAL; np = of_get_child_by_name(node, "simple-audio-card,codec"); if (np) { - ret = asoc_simple_card_sub_parse_of(np, priv->daifmt, + ret = asoc_simple_card_sub_parse_of(np, daifmt, &priv->codec_dai, &dai_link->codec_of_node, &dai_link->codec_dai_name); @@ -223,7 +223,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, dai_link->platform_of_node = dai_link->cpu_of_node; dev_dbg(dev, "card-name : %s\n", name); - dev_dbg(dev, "platform : %04x\n", priv->daifmt); + dev_dbg(dev, "platform : %04x\n", daifmt); dev_dbg(dev, "cpu : %s / %04x / %d\n", dai_link->cpu_dai_name, priv->cpu_dai.fmt, -- cgit v1.2.3 From bfe723f6eae285e399615d99f297d1646a6253fe Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 14 Mar 2014 09:26:11 +0100 Subject: ASoC: sirf-audio-codec: Remove snd_soc_codec_set_cache_io() call There was a overlap between the snd_soc_codec_set_cache_io() cleanup and the addition of the sirf-audio-codec resulting in the sirf-audio-codec driver still using the old signature of snd_soc_codec_set_cache_io(), which will cause a compile error. Since the core is able to automatically setup IO for this driver we can just remove both the snd_soc_set_cache_io() call and the control_data assignment. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/sirf-audio-codec.c | 9 --------- 1 file changed, 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sirf-audio-codec.c b/sound/soc/codecs/sirf-audio-codec.c index 90e3a228bae4..58e7c1f23771 100644 --- a/sound/soc/codecs/sirf-audio-codec.c +++ b/sound/soc/codecs/sirf-audio-codec.c @@ -337,18 +337,9 @@ struct snd_soc_dai_driver sirf_audio_codec_dai = { static int sirf_audio_codec_probe(struct snd_soc_codec *codec) { - int ret; struct snd_soc_dapm_context *dapm = &codec->dapm; - struct sirf_audio_codec *sirf_audio_codec = snd_soc_codec_get_drvdata(codec); pm_runtime_enable(codec->dev); - codec->control_data = sirf_audio_codec->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } if (of_device_is_compatible(codec->dev->of_node, "sirf,prima2-audio-codec")) { snd_soc_dapm_new_controls(dapm, -- cgit v1.2.3 From dac7e40404a6b1e7442c01ef4c2e7e149b9627e5 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sun, 16 Mar 2014 23:06:25 +0800 Subject: ASoC: tlv320aic31xx: Don't call kfree for memory allocated by devm_kzalloc The kfree call is not necessary, but we need to call snd_soc_unregister_codec() in remove(). Signed-off-by: Axel Lin Acked-by: Jyri Sarha Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic31xx.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 1f243c2c98fd..e463ae7fe1f4 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1229,7 +1229,6 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, return -ENOMEM; aic31xx->regmap = devm_regmap_init_i2c(i2c, regmap_config); - if (IS_ERR(aic31xx->regmap)) { ret = PTR_ERR(aic31xx->regmap); dev_err(&i2c->dev, "Failed to allocate register map: %d\n", @@ -1242,18 +1241,14 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, aic31xx_device_init(aic31xx); - ret = snd_soc_register_codec(&i2c->dev, &soc_codec_driver_aic31xx, + return snd_soc_register_codec(&i2c->dev, &soc_codec_driver_aic31xx, aic31xx_dai_driver, ARRAY_SIZE(aic31xx_dai_driver)); - - return ret; } static int aic31xx_i2c_remove(struct i2c_client *i2c) { - struct aic31xx_priv *aic31xx = dev_get_drvdata(&i2c->dev); - - kfree(aic31xx); + snd_soc_unregister_codec(&i2c->dev); return 0; } @@ -1275,7 +1270,7 @@ static struct i2c_driver aic31xx_i2c_driver = { .of_match_table = of_match_ptr(tlv320aic31xx_of_match), }, .probe = aic31xx_i2c_probe, - .remove = (aic31xx_i2c_remove), + .remove = aic31xx_i2c_remove, .id_table = aic31xx_i2c_id, }; -- cgit v1.2.3 From ab64246cf8c31f70a390dcabd134097c3aec45ab Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 13 Mar 2014 21:24:54 +0100 Subject: ASoC: codecs: Replace instances of rtd->codec with dai->codec With CODEC to CODEC links rtd->codec does not necessarily point to the driver's CODEC. CODEC drivers should always use dai->codec and never even look at the PCM runtime. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Acked-by: Jyri Sarha Signed-off-by: Mark Brown --- sound/soc/codecs/isabelle.c | 3 +-- sound/soc/codecs/mc13783.c | 6 ++---- sound/soc/codecs/rt5640.c | 3 +-- sound/soc/codecs/sta529.c | 3 +-- sound/soc/codecs/tlv320aic31xx.c | 5 ++--- sound/soc/codecs/uda134x.c | 3 +-- sound/soc/codecs/uda1380.c | 3 +-- sound/soc/codecs/wm8580.c | 3 +-- 8 files changed, 10 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c index 5839048ec467..7c8ba02a6d1b 100644 --- a/sound/soc/codecs/isabelle.c +++ b/sound/soc/codecs/isabelle.c @@ -910,8 +910,7 @@ static int isabelle_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 snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 aif = 0; unsigned int fs_val = 0; diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 582c2bbd42cb..bebba7fb8819 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -106,8 +106,7 @@ static int mc13783_pcm_hw_params_dac(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 snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; unsigned int rate = params_rate(params); int i; @@ -126,8 +125,7 @@ static int mc13783_pcm_hw_params_codec(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 snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; unsigned int rate = params_rate(params); unsigned int val; diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index a3fb41179636..7877a5e14c54 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1601,8 +1601,7 @@ static int get_clk_info(int sclk, int rate) static int rt5640_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 snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); unsigned int val_len = 0, val_clk, mask_clk; int dai_sel, pre_div, bclk_ms, frame_size; diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c index 40c07be9b581..fa4b050fe6e6 100644 --- a/sound/soc/codecs/sta529.c +++ b/sound/soc/codecs/sta529.c @@ -193,8 +193,7 @@ static int sta529_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 snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; int pdata, play_freq_val, record_freq_val; int bclk_to_fs_ratio; diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index e463ae7fe1f4..fa158cfe9b32 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -753,10 +753,9 @@ static int aic31xx_setup_pll(struct snd_soc_codec *codec, static int aic31xx_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, - struct snd_soc_dai *tmp) + struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u8 data = 0; dev_dbg(codec->dev, "## %s: format %d width %d rate %d\n", diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index c94d4c1e3dac..edf27acc1d77 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -203,8 +203,7 @@ static int uda134x_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 snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct uda134x_priv *uda134x = snd_soc_codec_get_drvdata(codec); u8 hw_params; diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 726df6d43c2b..dc47f6f5cd83 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -564,8 +564,7 @@ static int uda1380_pcm_hw_params(struct snd_pcm_substream *substream, static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK); /* shut down WSPLL power if running from this clock */ diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 318989acbbe5..49c2869b681c 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -504,8 +504,7 @@ static int wm8580_paif_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 snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8580_priv *wm8580 = snd_soc_codec_get_drvdata(codec); u16 paifa = 0; u16 paifb = 0; -- cgit v1.2.3 From cece5656901f09db13fbb569ff04f627ec2e0ab6 Mon Sep 17 00:00:00 2001 From: Thomas Petazzoni Date: Wed, 12 Feb 2014 18:20:57 +0100 Subject: ASoC: add S/PDIF support to Armada 370 DB ASoC driver The Armada 370 DB board not only has analog audio input/output, but also S/PDIF input/output. This commit adds support for S/PDIF in the ASoC machine driver of the Armada 370 DB platform, and adjusts the Device Tree bindings documentation accordingly. Signed-off-by: Thomas Petazzoni Signed-off-by: Mark Brown --- .../bindings/sound/armada-370db-audio.txt | 9 ++++--- sound/soc/kirkwood/Kconfig | 1 + sound/soc/kirkwood/armada-370-db.c | 28 ++++++++++++++++++++++ 3 files changed, 35 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/armada-370db-audio.txt b/Documentation/devicetree/bindings/sound/armada-370db-audio.txt index 3893b4d15a20..bf984d238620 100644 --- a/Documentation/devicetree/bindings/sound/armada-370db-audio.txt +++ b/Documentation/devicetree/bindings/sound/armada-370db-audio.txt @@ -11,14 +11,17 @@ Mandatory properties: * marvell,audio-controller: a phandle that points to the audio controller of the Armada 370 SoC. - * marvell,audio-codec: a phandle that points to the analog audio - codec connected to the Armada 370 SoC. + * marvell,audio-codec: a set of three phandles that points to: + + 1/ the analog audio codec connected to the Armada 370 SoC + 2/ the S/PDIF transceiver + 3/ the S/PDIF receiver Example: sound { compatible = "marvell,a370db-audio"; marvell,audio-controller = <&audio_controller>; - marvell,audio-codec = <&audio_codec>; + marvell,audio-codec = <&audio_codec &spdif_out &spdif_in>; status = "okay"; }; diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig index 2dc3ecf34801..49f8437665de 100644 --- a/sound/soc/kirkwood/Kconfig +++ b/sound/soc/kirkwood/Kconfig @@ -10,6 +10,7 @@ config SND_KIRKWOOD_SOC_ARMADA370_DB tristate "SoC Audio support for Armada 370 DB" depends on SND_KIRKWOOD_SOC && (ARCH_MVEBU || COMPILE_TEST) && I2C select SND_SOC_CS42L51 + select SND_SOC_SPDIF help Say Y if you want to add support for SoC audio on the Armada 370 Development Board. diff --git a/sound/soc/kirkwood/armada-370-db.c b/sound/soc/kirkwood/armada-370-db.c index 977639b3ffde..c44333849259 100644 --- a/sound/soc/kirkwood/armada-370-db.c +++ b/sound/soc/kirkwood/armada-370-db.c @@ -67,6 +67,20 @@ static struct snd_soc_dai_link a370db_dai[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, .ops = &a370db_ops, }, +{ + .name = "S/PDIF out", + .stream_name = "spdif-out", + .cpu_dai_name = "spdif", + .codec_dai_name = "dit-hifi", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, +}, +{ + .name = "S/PDIF in", + .stream_name = "spdif-in", + .cpu_dai_name = "spdif", + .codec_dai_name = "dir-hifi", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, +}, }; static struct snd_soc_card a370db = { @@ -95,6 +109,20 @@ static int a370db_probe(struct platform_device *pdev) of_parse_phandle(pdev->dev.of_node, "marvell,audio-codec", 0); + a370db_dai[1].cpu_of_node = a370db_dai[0].cpu_of_node; + a370db_dai[1].platform_of_node = a370db_dai[0].cpu_of_node; + + a370db_dai[1].codec_of_node = + of_parse_phandle(pdev->dev.of_node, + "marvell,audio-codec", 1); + + a370db_dai[2].cpu_of_node = a370db_dai[0].cpu_of_node; + a370db_dai[2].platform_of_node = a370db_dai[0].cpu_of_node; + + a370db_dai[2].codec_of_node = + of_parse_phandle(pdev->dev.of_node, + "marvell,audio-codec", 2); + return devm_snd_soc_register_card(card->dev, card); } -- cgit v1.2.3 From 1555b652970e541fa1cb80c61ffc696bbfb92bb7 Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Tue, 18 Mar 2014 13:56:21 -0500 Subject: ASoC: cs42l73: Fix mask bits for SOC_VALUE_ENUM_SINGLE The mask bits values were wrong for the SOC_VALUE_ENUM_SINGLE for the mono mix controls. Reported-by: Takashi Iwai Signed-off-by: Brian Austin Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/cs42l73.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 549d5d6a3fef..7b95f7cbc515 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -319,7 +319,7 @@ static const char * const cs42l73_mono_mix_texts[] = { static const unsigned int cs42l73_mono_mix_values[] = { 0, 1, 2 }; static const struct soc_enum spk_asp_enum = - SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 6, 1, + SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 6, 3, ARRAY_SIZE(cs42l73_mono_mix_texts), cs42l73_mono_mix_texts, cs42l73_mono_mix_values); @@ -337,7 +337,7 @@ static const struct snd_kcontrol_new spk_xsp_mixer = SOC_DAPM_ENUM("Route", spk_xsp_enum); static const struct soc_enum esl_asp_enum = - SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 2, 5, + SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 2, 3, ARRAY_SIZE(cs42l73_mono_mix_texts), cs42l73_mono_mix_texts, cs42l73_mono_mix_values); @@ -346,7 +346,7 @@ static const struct snd_kcontrol_new esl_asp_mixer = SOC_DAPM_ENUM("Route", esl_asp_enum); static const struct soc_enum esl_xsp_enum = - SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 0, 7, + SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 0, 3, ARRAY_SIZE(cs42l73_mono_mix_texts), cs42l73_mono_mix_texts, cs42l73_mono_mix_values); -- cgit v1.2.3 From d31a33dd7792c7d6c11fda226a3b9e4fb7f86f95 Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Tue, 18 Mar 2014 14:01:47 -0500 Subject: ASoC: cs42l52: Fix mask bits for SOC_VALUE_ENUM_SINGLE The mask bits values were wrong for the SOC_VALUE_ENUM_SINGLE for the PCM/ADC Swap controls Reported-by: Takashi Iwai Signed-off-by: Brian Austin Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/cs42l52.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 0bac6d5a4ac8..1102ced9b20e 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -347,7 +347,7 @@ static const char * const right_swap_text[] = { static const unsigned int swap_values[] = { 0, 1, 3 }; static const struct soc_enum adca_swap_enum = - SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 2, 1, + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 2, 3, ARRAY_SIZE(left_swap_text), left_swap_text, swap_values); @@ -356,7 +356,7 @@ static const struct snd_kcontrol_new adca_mixer = SOC_DAPM_ENUM("Route", adca_swap_enum); static const struct soc_enum pcma_swap_enum = - SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 6, 1, + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 6, 3, ARRAY_SIZE(left_swap_text), left_swap_text, swap_values); @@ -365,7 +365,7 @@ static const struct snd_kcontrol_new pcma_mixer = SOC_DAPM_ENUM("Route", pcma_swap_enum); static const struct soc_enum adcb_swap_enum = - SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 0, 1, + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 0, 3, ARRAY_SIZE(right_swap_text), right_swap_text, swap_values); @@ -374,7 +374,7 @@ static const struct snd_kcontrol_new adcb_mixer = SOC_DAPM_ENUM("Route", adcb_swap_enum); static const struct soc_enum pcmb_swap_enum = - SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 4, 1, + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 4, 3, ARRAY_SIZE(right_swap_text), right_swap_text, swap_values); -- cgit v1.2.3 From 46c39cae292fd691f32e573e6c2c854e36614c93 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Wed, 12 Mar 2014 11:02:11 +0800 Subject: ASoC: simple-card: overwrite cpu_dai->fmt with codec_dai->fmt The current simple-card driver separates the daimft for cpu_dai and codec_dai. So we might get different values for them (0x4003 and 0x1003 for example): asoc-simple-card sound-cs42888.12: cpu : 2024000.esai / 4003 / 132000000 asoc-simple-card sound-cs42888.12: codec : cs42888 / 1003 / 24576000 asoc-simple-card sound-cs42888.12: cs42888 <-> 2024000.esai mapping ok This is not allowed at all as we need to keep the DAIFMT settings identical for both the ends of the link. Thus this patch fixes it by overwriting the cpu_dai->fmt with codec_dai->fmt since we defined the DAIFMT_MASTER basing on CODEC at the first place while the other bits are same. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/simple-card.txt | 6 ++++++ sound/soc/generic/simple-card.c | 20 ++++++++++++++------ 2 files changed, 20 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt index b30c222f9cd3..881914b139ca 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ b/Documentation/devicetree/bindings/sound/simple-card.txt @@ -43,6 +43,12 @@ Optional CPU/CODEC subnodes properties: clock node (= common clock), or "system-clock-frequency" (if system doens't support common clock) +Note: + * For 'format', 'frame-master', 'bitclock-master', 'bitclock-inversion' and + 'frame-inversion', the simple card will use the settings of CODEC for both + CPU and CODEC sides as we need to keep the settings identical for both ends + of the link. + Example: sound { diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index ca7e63ef858a..2ee8ed56bcf1 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -151,6 +151,8 @@ static int asoc_simple_card_parse_of(struct device_node *node, struct device *dev) { struct snd_soc_dai_link *dai_link = priv->snd_card.dai_link; + struct asoc_simple_dai *codec_dai = &priv->codec_dai; + struct asoc_simple_dai *cpu_dai = &priv->cpu_dai; struct device_node *np; char *name; unsigned int daifmt; @@ -184,7 +186,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, np = of_get_child_by_name(node, "simple-audio-card,cpu"); if (np) { ret = asoc_simple_card_sub_parse_of(np, daifmt, - &priv->cpu_dai, + cpu_dai, &dai_link->cpu_of_node, &dai_link->cpu_dai_name); of_node_put(np); @@ -197,7 +199,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, np = of_get_child_by_name(node, "simple-audio-card,codec"); if (np) { ret = asoc_simple_card_sub_parse_of(np, daifmt, - &priv->codec_dai, + codec_dai, &dai_link->codec_of_node, &dai_link->codec_dai_name); of_node_put(np); @@ -205,6 +207,12 @@ static int asoc_simple_card_parse_of(struct device_node *node, if (ret < 0) return ret; + /* + * overwrite cpu_dai->fmt as its DAIFMT_MASTER bit is based on CODEC + * while the other bits should be identical unless buggy SW/HW design. + */ + cpu_dai->fmt = codec_dai->fmt; + if (!dai_link->cpu_dai_name || !dai_link->codec_dai_name) return -EINVAL; @@ -226,12 +234,12 @@ static int asoc_simple_card_parse_of(struct device_node *node, dev_dbg(dev, "platform : %04x\n", daifmt); dev_dbg(dev, "cpu : %s / %04x / %d\n", dai_link->cpu_dai_name, - priv->cpu_dai.fmt, - priv->cpu_dai.sysclk); + cpu_dai->fmt, + cpu_dai->sysclk); dev_dbg(dev, "codec : %s / %04x / %d\n", dai_link->codec_dai_name, - priv->codec_dai.fmt, - priv->codec_dai.sysclk); + codec_dai->fmt, + codec_dai->sysclk); /* * soc_bind_dai_link() will check cpu name -- cgit v1.2.3 From 657254714ad2ba69b73fcb02f0b1db378b1f220e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 20 Feb 2014 09:08:43 +0900 Subject: ASoC: io: Remove support for ASoC cache in conjunction with regmap Since all regmap CODECs should be (and are) using the more advance regmap cache infrastructure remove the code which supports that and just proxy I/O straight through to regmap. Signed-off-by: Mark Brown --- sound/soc/soc-io.c | 35 ++++------------------------------- 1 file changed, 4 insertions(+), 31 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index 8aa086996866..260efc8466fc 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -23,21 +23,6 @@ static int hw_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { - int ret; - - if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size && - !codec->cache_bypass) { - ret = snd_soc_cache_write(codec, reg, value); - if (ret < 0) - return -1; - } - - if (codec->cache_only) { - codec->cache_sync = 1; - return 0; - } - return regmap_write(codec->control_data, reg, value); } @@ -46,23 +31,11 @@ static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg) int ret; unsigned int val; - if (reg >= codec->driver->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg) || - codec->cache_bypass) { - if (codec->cache_only) - return -1; - - ret = regmap_read(codec->control_data, reg, &val); - if (ret == 0) - return val; - else - return -1; - } - - ret = snd_soc_cache_read(codec, reg, &val); - if (ret < 0) + ret = regmap_read(codec->control_data, reg, &val); + if (ret == 0) + return val; + else return -1; - return val; } /** -- cgit v1.2.3 From 7272e051157ccd5871b5d939548d0ba5a94a2965 Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Wed, 19 Mar 2014 10:40:02 -0500 Subject: ASoC: cs42l51: Fix SOC_DOUBLE_R_SX_TLV shift values for ADC, PCM, and Analog kcontrols The shift values for the ADC,PCM, and Analog kcontrols were wrong causing wrong values for the SOC_DOUBLE_R_SX_TLV macros Fixed the TLV for aout_tlv to show -102dB correctly Fixes: 1d99f2436d (ASoC: core: Rework SOC_DOUBLE_R_SX_TLV add SOC_SINGLE_SX_TLV) Reported-by: Thomas Petazzoni Signed-off-by: Brian Austin Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/cs42l51.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 6e9ea8379a91..7a272fa90b39 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -124,9 +124,8 @@ static int cs42l51_set_chan_mix(struct snd_kcontrol *kcontrol, static const DECLARE_TLV_DB_SCALE(adc_pcm_tlv, -5150, 50, 0); static const DECLARE_TLV_DB_SCALE(tone_tlv, -1050, 150, 0); -/* This is a lie. after -102 db, it stays at -102 */ -/* maybe a range would be better */ -static const DECLARE_TLV_DB_SCALE(aout_tlv, -11550, 50, 0); + +static const DECLARE_TLV_DB_SCALE(aout_tlv, -10200, 50, 0); static const DECLARE_TLV_DB_SCALE(boost_tlv, 1600, 1600, 0); static const char *chan_mix[] = { @@ -141,7 +140,7 @@ static const struct soc_enum cs42l51_chan_mix = static const struct snd_kcontrol_new cs42l51_snd_controls[] = { SOC_DOUBLE_R_SX_TLV("PCM Playback Volume", CS42L51_PCMA_VOL, CS42L51_PCMB_VOL, - 6, 0x19, 0x7F, adc_pcm_tlv), + 0, 0x19, 0x7F, adc_pcm_tlv), SOC_DOUBLE_R("PCM Playback Switch", CS42L51_PCMA_VOL, CS42L51_PCMB_VOL, 7, 1, 1), SOC_DOUBLE_R_SX_TLV("Analog Playback Volume", @@ -149,7 +148,7 @@ static const struct snd_kcontrol_new cs42l51_snd_controls[] = { 0, 0x34, 0xE4, aout_tlv), SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L51_ADCA_VOL, CS42L51_ADCB_VOL, - 6, 0x19, 0x7F, adc_pcm_tlv), + 0, 0x19, 0x7F, adc_pcm_tlv), SOC_DOUBLE_R("ADC Mixer Switch", CS42L51_ADCA_VOL, CS42L51_ADCB_VOL, 7, 1, 1), SOC_SINGLE("Playback Deemphasis Switch", CS42L51_DAC_CTL, 3, 1, 0), -- cgit v1.2.3 From 591d14f00796a4250d800d316e3db1fea8a57e20 Mon Sep 17 00:00:00 2001 From: Dylan Reid Date: Mon, 17 Mar 2014 22:08:49 -0700 Subject: ASoC: tegra: Use flat regcache When using an rbtree cache, there can be allocations the first time a register is accessed. This can cause an attempt to schedule while atomic in the case that the regmap is using a spinlock. This could be fixed by either initializing all the registers or using a flat cache. The register maps for tegra30_ahub and tegra30_i2s are dense and don't save much from using a tree so convert them to flat. Tegra30 changes tested on Norrin, Tegra20 changes compile. Signed-off-by: Dylan Reid Tested-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra20_ac97.c | 2 +- sound/soc/tegra/tegra20_das.c | 2 +- sound/soc/tegra/tegra20_i2s.c | 2 +- sound/soc/tegra/tegra20_spdif.c | 2 +- sound/soc/tegra/tegra30_ahub.c | 4 ++-- sound/soc/tegra/tegra30_i2s.c | 2 +- 6 files changed, 7 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index cf5e1cfe818d..0a59e2383ef3 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -306,7 +306,7 @@ static const struct regmap_config tegra20_ac97_regmap_config = { .readable_reg = tegra20_ac97_wr_rd_reg, .volatile_reg = tegra20_ac97_volatile_reg, .precious_reg = tegra20_ac97_precious_reg, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_FLAT, }; static int tegra20_ac97_platform_probe(struct platform_device *pdev) diff --git a/sound/soc/tegra/tegra20_das.c b/sound/soc/tegra/tegra20_das.c index e72392927bd2..a634f13b3ffc 100644 --- a/sound/soc/tegra/tegra20_das.c +++ b/sound/soc/tegra/tegra20_das.c @@ -128,7 +128,7 @@ static const struct regmap_config tegra20_das_regmap_config = { .max_register = LAST_REG(DAC_INPUT_DATA_CLK_SEL), .writeable_reg = tegra20_das_wr_rd_reg, .readable_reg = tegra20_das_wr_rd_reg, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_FLAT, }; static int tegra20_das_probe(struct platform_device *pdev) diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c index 42c1f6bfaf2e..79a9932ffe6e 100644 --- a/sound/soc/tegra/tegra20_i2s.c +++ b/sound/soc/tegra/tegra20_i2s.c @@ -333,7 +333,7 @@ static const struct regmap_config tegra20_i2s_regmap_config = { .readable_reg = tegra20_i2s_wr_rd_reg, .volatile_reg = tegra20_i2s_volatile_reg, .precious_reg = tegra20_i2s_precious_reg, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_FLAT, }; static int tegra20_i2s_platform_probe(struct platform_device *pdev) diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index 8c7c1028e579..a0ce92400faf 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -259,7 +259,7 @@ static const struct regmap_config tegra20_spdif_regmap_config = { .readable_reg = tegra20_spdif_wr_rd_reg, .volatile_reg = tegra20_spdif_volatile_reg, .precious_reg = tegra20_spdif_precious_reg, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_FLAT, }; static int tegra20_spdif_platform_probe(struct platform_device *pdev) diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index d6f4c9940e0c..0db68f49f4d9 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -471,7 +471,7 @@ static const struct regmap_config tegra30_ahub_apbif_regmap_config = { .readable_reg = tegra30_ahub_apbif_wr_rd_reg, .volatile_reg = tegra30_ahub_apbif_volatile_reg, .precious_reg = tegra30_ahub_apbif_precious_reg, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_FLAT, }; static bool tegra30_ahub_ahub_wr_rd_reg(struct device *dev, unsigned int reg) @@ -490,7 +490,7 @@ static const struct regmap_config tegra30_ahub_ahub_regmap_config = { .max_register = LAST_REG(AUDIO_RX), .writeable_reg = tegra30_ahub_ahub_wr_rd_reg, .readable_reg = tegra30_ahub_ahub_wr_rd_reg, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_FLAT, }; static struct tegra30_ahub_soc_data soc_data_tegra30 = { diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index 49ad9366add8..f146c41dd3ec 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -357,7 +357,7 @@ static const struct regmap_config tegra30_i2s_regmap_config = { .writeable_reg = tegra30_i2s_wr_rd_reg, .readable_reg = tegra30_i2s_wr_rd_reg, .volatile_reg = tegra30_i2s_volatile_reg, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_FLAT, }; static const struct tegra30_i2s_soc_data tegra30_i2s_config = { -- cgit v1.2.3 From 0c516b4ff85c0be4cee5b30ae59c9565c7f91a00 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Thu, 20 Mar 2014 18:18:37 +0800 Subject: ASoC: cs42xx8: Add codec driver support for CS42448/CS42888 This patch adds support for the Cirrus Logic CS42448/CS42888 Audio CODEC that has six/four 24-bit AD and eight 24-bit DA converters. [ CS42448/CS42888 supports both I2C and SPI control ports. As initial patch, this patch only adds the support for I2C. ] Signed-off-by: Nicolin Chen Acked-by: Brian Austin Acked-by: Paul Handrigan Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/cs42xx8.txt | 28 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/cs42xx8-i2c.c | 64 +++ sound/soc/codecs/cs42xx8.c | 602 +++++++++++++++++++++ sound/soc/codecs/cs42xx8.h | 238 ++++++++ 6 files changed, 946 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs42xx8.txt create mode 100644 sound/soc/codecs/cs42xx8-i2c.c create mode 100644 sound/soc/codecs/cs42xx8.c create mode 100644 sound/soc/codecs/cs42xx8.h (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/cs42xx8.txt b/Documentation/devicetree/bindings/sound/cs42xx8.txt new file mode 100644 index 000000000000..f631fbca6284 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs42xx8.txt @@ -0,0 +1,28 @@ +CS42448/CS42888 audio CODEC + +Required properties: + + - compatible : must contain one of "cirrus,cs42448" and "cirrus,cs42888" + + - reg : the I2C address of the device for I2C + + - clocks : a list of phandles + clock-specifiers, one for each entry in + clock-names + + - clock-names : must contain "mclk" + + - VA-supply, VD-supply, VLS-supply, VLC-supply: power supplies for the device, + as covered in Documentation/devicetree/bindings/regulator/regulator.txt + +Example: + +codec: cs42888@48 { + compatible = "cirrus,cs42888"; + reg = <0x48>; + clocks = <&codec_mclk 0>; + clock-names = "mclk"; + VA-supply = <®_audio>; + VD-supply = <®_audio>; + VLS-supply = <®_audio>; + VLC-supply = <®_audio>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 983d087aa92a..a79c0d141f90 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -37,6 +37,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS42L73 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 select SND_SOC_CX20442 if TTY select SND_SOC_DA7210 if I2C select SND_SOC_DA7213 if I2C @@ -254,6 +255,15 @@ config SND_SOC_CS4270_VD33_ERRATA config SND_SOC_CS4271 tristate +config SND_SOC_CS42XX8 + tristate + +config SND_SOC_CS42XX8_I2C + tristate "Cirrus Logic CS42448/CS42888 CODEC (I2C)" + depends on I2C + select SND_SOC_CS42XX8 + select REGMAP_I2C + config SND_SOC_CX20442 tristate depends on TTY diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index bc126764a44d..cfe5d634c812 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -23,6 +23,8 @@ snd-soc-cs42l52-objs := cs42l52.o snd-soc-cs42l73-objs := cs42l73.o snd-soc-cs4270-objs := cs4270.o snd-soc-cs4271-objs := cs4271.o +snd-soc-cs42xx8-objs := cs42xx8.o +snd-soc-cs42xx8-i2c-objs := cs42xx8-i2c.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o @@ -156,6 +158,8 @@ obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o obj-$(CONFIG_SND_SOC_CS42L73) += snd-soc-cs42l73.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o +obj-$(CONFIG_SND_SOC_CS42XX8) += snd-soc-cs42xx8.o +obj-$(CONFIG_SND_SOC_CS42XX8_I2C) += snd-soc-cs42xx8-i2c.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o diff --git a/sound/soc/codecs/cs42xx8-i2c.c b/sound/soc/codecs/cs42xx8-i2c.c new file mode 100644 index 000000000000..657dce27eade --- /dev/null +++ b/sound/soc/codecs/cs42xx8-i2c.c @@ -0,0 +1,64 @@ +/* + * Cirrus Logic CS42448/CS42888 Audio CODEC DAI I2C 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 "cs42xx8.h" + +static int cs42xx8_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + u32 ret = cs42xx8_probe(&i2c->dev, + devm_regmap_init_i2c(i2c, &cs42xx8_regmap_config)); + if (ret) + return ret; + + pm_runtime_enable(&i2c->dev); + pm_request_idle(&i2c->dev); + + return 0; +} + +static int cs42xx8_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + pm_runtime_disable(&i2c->dev); + + return 0; +} + +static struct i2c_device_id cs42xx8_i2c_id[] = { + {"cs42448", (kernel_ulong_t)&cs42448_data}, + {"cs42888", (kernel_ulong_t)&cs42888_data}, + {} +}; +MODULE_DEVICE_TABLE(i2c, cs42xx8_i2c_id); + +static struct i2c_driver cs42xx8_i2c_driver = { + .driver = { + .name = "cs42xx8", + .owner = THIS_MODULE, + .pm = &cs42xx8_pm, + }, + .probe = cs42xx8_i2c_probe, + .remove = cs42xx8_i2c_remove, + .id_table = cs42xx8_i2c_id, +}; + +module_i2c_driver(cs42xx8_i2c_driver); + +MODULE_DESCRIPTION("Cirrus Logic CS42448/CS42888 ALSA SoC Codec I2C Driver"); +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c new file mode 100644 index 000000000000..082299a4e2fa --- /dev/null +++ b/sound/soc/codecs/cs42xx8.c @@ -0,0 +1,602 @@ +/* + * Cirrus Logic CS42448/CS42888 Audio CODEC 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 "cs42xx8.h" + +#define CS42XX8_NUM_SUPPLIES 4 +static const char *const cs42xx8_supply_names[CS42XX8_NUM_SUPPLIES] = { + "VA", + "VD", + "VLS", + "VLC", +}; + +#define CS42XX8_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/* codec private data */ +struct cs42xx8_priv { + struct regulator_bulk_data supplies[CS42XX8_NUM_SUPPLIES]; + const struct cs42xx8_driver_data *drvdata; + struct regmap *regmap; + struct clk *clk; + + bool slave_mode; + unsigned long sysclk; +}; + +/* -127.5dB to 0dB with step of 0.5dB */ +static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1); +/* -64dB to 24dB with step of 0.5dB */ +static const DECLARE_TLV_DB_SCALE(adc_tlv, -6400, 50, 0); + +static const char *const cs42xx8_adc_single[] = { "Differential", "Single-Ended" }; +static const char *const cs42xx8_szc[] = { "Immediate Change", "Zero Cross", + "Soft Ramp", "Soft Ramp on Zero Cross" }; + +static const struct soc_enum adc1_single_enum = + SOC_ENUM_SINGLE(CS42XX8_ADCCTL, 4, 2, cs42xx8_adc_single); +static const struct soc_enum adc2_single_enum = + SOC_ENUM_SINGLE(CS42XX8_ADCCTL, 3, 2, cs42xx8_adc_single); +static const struct soc_enum adc3_single_enum = + SOC_ENUM_SINGLE(CS42XX8_ADCCTL, 2, 2, cs42xx8_adc_single); +static const struct soc_enum dac_szc_enum = + SOC_ENUM_SINGLE(CS42XX8_TXCTL, 5, 4, cs42xx8_szc); +static const struct soc_enum adc_szc_enum = + SOC_ENUM_SINGLE(CS42XX8_TXCTL, 0, 4, cs42xx8_szc); + +static const struct snd_kcontrol_new cs42xx8_snd_controls[] = { + SOC_DOUBLE_R_TLV("DAC1 Playback Volume", CS42XX8_VOLAOUT1, + CS42XX8_VOLAOUT2, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_TLV("DAC2 Playback Volume", CS42XX8_VOLAOUT3, + CS42XX8_VOLAOUT4, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_TLV("DAC3 Playback Volume", CS42XX8_VOLAOUT5, + CS42XX8_VOLAOUT6, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_TLV("DAC4 Playback Volume", CS42XX8_VOLAOUT7, + CS42XX8_VOLAOUT8, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_S_TLV("ADC1 Capture Volume", CS42XX8_VOLAIN1, + CS42XX8_VOLAIN2, 0, -0x80, 0x30, 7, 0, adc_tlv), + SOC_DOUBLE_R_S_TLV("ADC2 Capture Volume", CS42XX8_VOLAIN3, + CS42XX8_VOLAIN4, 0, -0x80, 0x30, 7, 0, adc_tlv), + SOC_DOUBLE("DAC1 Invert Switch", CS42XX8_DACINV, 0, 1, 1, 0), + SOC_DOUBLE("DAC2 Invert Switch", CS42XX8_DACINV, 2, 3, 1, 0), + SOC_DOUBLE("DAC3 Invert Switch", CS42XX8_DACINV, 4, 5, 1, 0), + SOC_DOUBLE("DAC4 Invert Switch", CS42XX8_DACINV, 6, 7, 1, 0), + SOC_DOUBLE("ADC1 Invert Switch", CS42XX8_ADCINV, 0, 1, 1, 0), + SOC_DOUBLE("ADC2 Invert Switch", CS42XX8_ADCINV, 2, 3, 1, 0), + SOC_SINGLE("ADC High-Pass Filter Switch", CS42XX8_ADCCTL, 7, 1, 1), + SOC_SINGLE("DAC De-emphasis Switch", CS42XX8_ADCCTL, 5, 1, 0), + SOC_ENUM("ADC1 Single Ended Mode Switch", adc1_single_enum), + SOC_ENUM("ADC2 Single Ended Mode Switch", adc2_single_enum), + SOC_SINGLE("DAC Single Volume Control Switch", CS42XX8_TXCTL, 7, 1, 0), + SOC_ENUM("DAC Soft Ramp & Zero Cross Control Switch", dac_szc_enum), + SOC_SINGLE("DAC Auto Mute Switch", CS42XX8_TXCTL, 4, 1, 0), + SOC_SINGLE("Mute ADC Serial Port Switch", CS42XX8_TXCTL, 3, 1, 0), + SOC_SINGLE("ADC Single Volume Control Switch", CS42XX8_TXCTL, 2, 1, 0), + SOC_ENUM("ADC Soft Ramp & Zero Cross Control Switch", adc_szc_enum), +}; + +static const struct snd_kcontrol_new cs42xx8_adc3_snd_controls[] = { + SOC_DOUBLE_R_S_TLV("ADC3 Capture Volume", CS42XX8_VOLAIN5, + CS42XX8_VOLAIN6, 0, -0x80, 0x30, 7, 0, adc_tlv), + SOC_DOUBLE("ADC3 Invert Switch", CS42XX8_ADCINV, 4, 5, 1, 0), + SOC_ENUM("ADC3 Single Ended Mode Switch", adc3_single_enum), +}; + +static const struct snd_soc_dapm_widget cs42xx8_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC1", "Playback", CS42XX8_PWRCTL, 1, 1), + SND_SOC_DAPM_DAC("DAC2", "Playback", CS42XX8_PWRCTL, 2, 1), + SND_SOC_DAPM_DAC("DAC3", "Playback", CS42XX8_PWRCTL, 3, 1), + SND_SOC_DAPM_DAC("DAC4", "Playback", CS42XX8_PWRCTL, 4, 1), + + SND_SOC_DAPM_OUTPUT("AOUT1L"), + SND_SOC_DAPM_OUTPUT("AOUT1R"), + SND_SOC_DAPM_OUTPUT("AOUT2L"), + SND_SOC_DAPM_OUTPUT("AOUT2R"), + SND_SOC_DAPM_OUTPUT("AOUT3L"), + SND_SOC_DAPM_OUTPUT("AOUT3R"), + SND_SOC_DAPM_OUTPUT("AOUT4L"), + SND_SOC_DAPM_OUTPUT("AOUT4R"), + + SND_SOC_DAPM_ADC("ADC1", "Capture", CS42XX8_PWRCTL, 5, 1), + SND_SOC_DAPM_ADC("ADC2", "Capture", CS42XX8_PWRCTL, 6, 1), + + SND_SOC_DAPM_INPUT("AIN1L"), + SND_SOC_DAPM_INPUT("AIN1R"), + SND_SOC_DAPM_INPUT("AIN2L"), + SND_SOC_DAPM_INPUT("AIN2R"), + + SND_SOC_DAPM_SUPPLY("PWR", CS42XX8_PWRCTL, 0, 1, NULL, 0), +}; + +static const struct snd_soc_dapm_widget cs42xx8_adc3_dapm_widgets[] = { + SND_SOC_DAPM_ADC("ADC3", "Capture", CS42XX8_PWRCTL, 7, 1), + + SND_SOC_DAPM_INPUT("AIN3L"), + SND_SOC_DAPM_INPUT("AIN3R"), +}; + +static const struct snd_soc_dapm_route cs42xx8_dapm_routes[] = { + /* Playback */ + { "AOUT1L", NULL, "DAC1" }, + { "AOUT1R", NULL, "DAC1" }, + { "DAC1", NULL, "PWR" }, + + { "AOUT2L", NULL, "DAC2" }, + { "AOUT2R", NULL, "DAC2" }, + { "DAC2", NULL, "PWR" }, + + { "AOUT3L", NULL, "DAC3" }, + { "AOUT3R", NULL, "DAC3" }, + { "DAC3", NULL, "PWR" }, + + { "AOUT4L", NULL, "DAC4" }, + { "AOUT4R", NULL, "DAC4" }, + { "DAC4", NULL, "PWR" }, + + /* Capture */ + { "ADC1", NULL, "AIN1L" }, + { "ADC1", NULL, "AIN1R" }, + { "ADC1", NULL, "PWR" }, + + { "ADC2", NULL, "AIN2L" }, + { "ADC2", NULL, "AIN2R" }, + { "ADC2", NULL, "PWR" }, +}; + +static const struct snd_soc_dapm_route cs42xx8_adc3_dapm_routes[] = { + /* Capture */ + { "ADC3", NULL, "AIN3L" }, + { "ADC3", NULL, "AIN3R" }, + { "ADC3", NULL, "PWR" }, +}; + +struct cs42xx8_ratios { + unsigned int ratio; + unsigned char speed; + unsigned char mclk; +}; + +static const struct cs42xx8_ratios cs42xx8_ratios[] = { + { 64, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_256(4) }, + { 96, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_384(4) }, + { 128, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_512(4) }, + { 192, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_768(4) }, + { 256, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_256(1) }, + { 384, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_384(1) }, + { 512, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_512(1) }, + { 768, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_768(1) }, + { 1024, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_1024(1) } +}; + +static int cs42xx8_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec); + + cs42xx8->sysclk = freq; + + return 0; +} + +static int cs42xx8_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec); + u32 val; + + /* Set DAI format */ + switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_LEFT_J: + val = CS42XX8_INTF_DAC_DIF_LEFTJ | CS42XX8_INTF_ADC_DIF_LEFTJ; + break; + case SND_SOC_DAIFMT_I2S: + val = CS42XX8_INTF_DAC_DIF_I2S | CS42XX8_INTF_ADC_DIF_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + val = CS42XX8_INTF_DAC_DIF_RIGHTJ | CS42XX8_INTF_ADC_DIF_RIGHTJ; + break; + default: + dev_err(codec->dev, "unsupported dai format\n"); + return -EINVAL; + } + + regmap_update_bits(cs42xx8->regmap, CS42XX8_INTF, + CS42XX8_INTF_DAC_DIF_MASK | + CS42XX8_INTF_ADC_DIF_MASK, val); + + /* Set master/slave audio interface */ + switch (format & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + cs42xx8->slave_mode = true; + break; + case SND_SOC_DAIFMT_CBM_CFM: + cs42xx8->slave_mode = false; + break; + default: + dev_err(codec->dev, "unsupported master/slave mode\n"); + return -EINVAL; + } + + return 0; +} + +static int cs42xx8_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 snd_soc_codec *codec = rtd->codec; + struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + u32 ratio = cs42xx8->sysclk / params_rate(params); + u32 i, fm, val, mask; + + for (i = 0; i < ARRAY_SIZE(cs42xx8_ratios); i++) { + if (cs42xx8_ratios[i].ratio == ratio) + break; + } + + if (i == ARRAY_SIZE(cs42xx8_ratios)) { + dev_err(codec->dev, "unsupported sysclk ratio\n"); + return -EINVAL; + } + + mask = CS42XX8_FUNCMOD_MFREQ_MASK; + val = cs42xx8_ratios[i].mclk; + + fm = cs42xx8->slave_mode ? CS42XX8_FM_AUTO : cs42xx8_ratios[i].speed; + + regmap_update_bits(cs42xx8->regmap, CS42XX8_FUNCMOD, + CS42XX8_FUNCMOD_xC_FM_MASK(tx) | mask, + CS42XX8_FUNCMOD_xC_FM(tx, fm) | val); + + return 0; +} + +static int cs42xx8_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec); + + regmap_update_bits(cs42xx8->regmap, CS42XX8_DACMUTE, + CS42XX8_DACMUTE_ALL, mute ? CS42XX8_DACMUTE_ALL : 0); + + return 0; +} + +static const struct snd_soc_dai_ops cs42xx8_dai_ops = { + .set_fmt = cs42xx8_set_dai_fmt, + .set_sysclk = cs42xx8_set_dai_sysclk, + .hw_params = cs42xx8_hw_params, + .digital_mute = cs42xx8_digital_mute, +}; + +static struct snd_soc_dai_driver cs42xx8_dai = { + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = CS42XX8_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = CS42XX8_FORMATS, + }, + .ops = &cs42xx8_dai_ops, +}; + +static const struct reg_default cs42xx8_reg[] = { + { 0x01, 0x01 }, /* Chip I.D. and Revision Register */ + { 0x02, 0x00 }, /* Power Control */ + { 0x03, 0xF0 }, /* Functional Mode */ + { 0x04, 0x46 }, /* Interface Formats */ + { 0x05, 0x00 }, /* ADC Control & DAC De-Emphasis */ + { 0x06, 0x10 }, /* Transition Control */ + { 0x07, 0x00 }, /* DAC Channel Mute */ + { 0x08, 0x00 }, /* Volume Control AOUT1 */ + { 0x09, 0x00 }, /* Volume Control AOUT2 */ + { 0x0a, 0x00 }, /* Volume Control AOUT3 */ + { 0x0b, 0x00 }, /* Volume Control AOUT4 */ + { 0x0c, 0x00 }, /* Volume Control AOUT5 */ + { 0x0d, 0x00 }, /* Volume Control AOUT6 */ + { 0x0e, 0x00 }, /* Volume Control AOUT7 */ + { 0x0f, 0x00 }, /* Volume Control AOUT8 */ + { 0x10, 0x00 }, /* DAC Channel Invert */ + { 0x11, 0x00 }, /* Volume Control AIN1 */ + { 0x12, 0x00 }, /* Volume Control AIN2 */ + { 0x13, 0x00 }, /* Volume Control AIN3 */ + { 0x14, 0x00 }, /* Volume Control AIN4 */ + { 0x15, 0x00 }, /* Volume Control AIN5 */ + { 0x16, 0x00 }, /* Volume Control AIN6 */ + { 0x17, 0x00 }, /* ADC Channel Invert */ + { 0x18, 0x00 }, /* Status Control */ + { 0x1a, 0x00 }, /* Status Mask */ + { 0x1b, 0x00 }, /* MUTEC Pin Control */ +}; + +static bool cs42xx8_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42XX8_STATUS: + return true; + default: + return false; + } +} + +static bool cs42xx8_writeable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42XX8_CHIPID: + case CS42XX8_STATUS: + return false; + default: + return true; + } +} + +const struct regmap_config cs42xx8_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS42XX8_LASTREG, + .reg_defaults = cs42xx8_reg, + .num_reg_defaults = ARRAY_SIZE(cs42xx8_reg), + .volatile_reg = cs42xx8_volatile_register, + .writeable_reg = cs42xx8_writeable_register, + .cache_type = REGCACHE_RBTREE, +}; +EXPORT_SYMBOL_GPL(cs42xx8_regmap_config); + +static int cs42xx8_codec_probe(struct snd_soc_codec *codec) +{ + struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; + + switch (cs42xx8->drvdata->num_adcs) { + case 3: + snd_soc_add_codec_controls(codec, cs42xx8_adc3_snd_controls, + ARRAY_SIZE(cs42xx8_adc3_snd_controls)); + snd_soc_dapm_new_controls(dapm, cs42xx8_adc3_dapm_widgets, + ARRAY_SIZE(cs42xx8_adc3_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, cs42xx8_adc3_dapm_routes, + ARRAY_SIZE(cs42xx8_adc3_dapm_routes)); + break; + default: + break; + } + + /* Mute all DAC channels */ + regmap_write(cs42xx8->regmap, CS42XX8_DACMUTE, CS42XX8_DACMUTE_ALL); + + return 0; +} + +static const struct snd_soc_codec_driver cs42xx8_driver = { + .probe = cs42xx8_codec_probe, + .idle_bias_off = true, + + .controls = cs42xx8_snd_controls, + .num_controls = ARRAY_SIZE(cs42xx8_snd_controls), + .dapm_widgets = cs42xx8_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs42xx8_dapm_widgets), + .dapm_routes = cs42xx8_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(cs42xx8_dapm_routes), +}; + +const struct cs42xx8_driver_data cs42448_data = { + .name = "cs42448", + .num_adcs = 3, +}; +EXPORT_SYMBOL_GPL(cs42448_data); + +const struct cs42xx8_driver_data cs42888_data = { + .name = "cs42888", + .num_adcs = 2, +}; +EXPORT_SYMBOL_GPL(cs42888_data); + +const struct of_device_id cs42xx8_of_match[] = { + { .compatible = "cirrus,cs42448", .data = &cs42448_data, }, + { .compatible = "cirrus,cs42888", .data = &cs42888_data, }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, cs42xx8_of_match); +EXPORT_SYMBOL_GPL(cs42xx8_of_match); + +int cs42xx8_probe(struct device *dev, struct regmap *regmap) +{ + const struct of_device_id *of_id = of_match_device(cs42xx8_of_match, dev); + struct cs42xx8_priv *cs42xx8; + int ret, val, i; + + cs42xx8 = devm_kzalloc(dev, sizeof(*cs42xx8), GFP_KERNEL); + if (cs42xx8 == NULL) + return -ENOMEM; + + dev_set_drvdata(dev, cs42xx8); + + if (of_id) + cs42xx8->drvdata = of_id->data; + + if (!cs42xx8->drvdata) { + dev_err(dev, "failed to find driver data\n"); + return -EINVAL; + } + + cs42xx8->clk = devm_clk_get(dev, "mclk"); + if (IS_ERR(cs42xx8->clk)) { + dev_err(dev, "failed to get the clock: %ld\n", + PTR_ERR(cs42xx8->clk)); + return -EINVAL; + } + + cs42xx8->sysclk = clk_get_rate(cs42xx8->clk); + + for (i = 0; i < ARRAY_SIZE(cs42xx8->supplies); i++) + cs42xx8->supplies[i].supply = cs42xx8_supply_names[i]; + + ret = devm_regulator_bulk_get(dev, + ARRAY_SIZE(cs42xx8->supplies), cs42xx8->supplies); + if (ret) { + dev_err(dev, "failed to request supplies: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(cs42xx8->supplies), + cs42xx8->supplies); + if (ret) { + dev_err(dev, "failed to enable supplies: %d\n", ret); + return ret; + } + + /* Make sure hardware reset done */ + msleep(5); + + cs42xx8->regmap = regmap; + if (IS_ERR(cs42xx8->regmap)) { + ret = PTR_ERR(cs42xx8->regmap); + dev_err(dev, "failed to allocate regmap: %d\n", ret); + goto err_enable; + } + + /* + * We haven't marked the chip revision as volatile due to + * sharing a register with the right input volume; explicitly + * bypass the cache to read it. + */ + regcache_cache_bypass(cs42xx8->regmap, true); + + /* Validate the chip ID */ + regmap_read(cs42xx8->regmap, CS42XX8_CHIPID, &val); + if (val < 0) { + dev_err(dev, "failed to get device ID: %x", val); + ret = -EINVAL; + goto err_enable; + } + + /* The top four bits of the chip ID should be 0000 */ + if ((val & CS42XX8_CHIPID_CHIP_ID_MASK) != 0x00) { + dev_err(dev, "unmatched chip ID: %d\n", + val & CS42XX8_CHIPID_CHIP_ID_MASK); + ret = -EINVAL; + goto err_enable; + } + + dev_info(dev, "found device, revision %X\n", + val & CS42XX8_CHIPID_REV_ID_MASK); + + regcache_cache_bypass(cs42xx8->regmap, false); + + cs42xx8_dai.name = cs42xx8->drvdata->name; + + /* Each adc supports stereo input */ + cs42xx8_dai.capture.channels_max = cs42xx8->drvdata->num_adcs * 2; + + ret = snd_soc_register_codec(dev, &cs42xx8_driver, &cs42xx8_dai, 1); + if (ret) { + dev_err(dev, "failed to register codec:%d\n", ret); + goto err_enable; + } + + regcache_cache_only(cs42xx8->regmap, true); + +err_enable: + regulator_bulk_disable(ARRAY_SIZE(cs42xx8->supplies), + cs42xx8->supplies); + + return ret; +} +EXPORT_SYMBOL_GPL(cs42xx8_probe); + +#ifdef CONFIG_PM_RUNTIME +static int cs42xx8_runtime_resume(struct device *dev) +{ + struct cs42xx8_priv *cs42xx8 = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(cs42xx8->clk); + if (ret) { + dev_err(dev, "failed to enable mclk: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(cs42xx8->supplies), + cs42xx8->supplies); + if (ret) { + dev_err(dev, "failed to enable supplies: %d\n", ret); + goto err_clk; + } + + /* Make sure hardware reset done */ + msleep(5); + + regcache_cache_only(cs42xx8->regmap, false); + + ret = regcache_sync(cs42xx8->regmap); + if (ret) { + dev_err(dev, "failed to sync regmap: %d\n", ret); + goto err_bulk; + } + + return 0; + +err_bulk: + regulator_bulk_disable(ARRAY_SIZE(cs42xx8->supplies), + cs42xx8->supplies); +err_clk: + clk_disable_unprepare(cs42xx8->clk); + + return ret; +} + +static int cs42xx8_runtime_suspend(struct device *dev) +{ + struct cs42xx8_priv *cs42xx8 = dev_get_drvdata(dev); + + regcache_cache_only(cs42xx8->regmap, true); + + regulator_bulk_disable(ARRAY_SIZE(cs42xx8->supplies), + cs42xx8->supplies); + + clk_disable_unprepare(cs42xx8->clk); + + return 0; +} +#endif + +const struct dev_pm_ops cs42xx8_pm = { + SET_RUNTIME_PM_OPS(cs42xx8_runtime_suspend, cs42xx8_runtime_resume, NULL) +}; +EXPORT_SYMBOL_GPL(cs42xx8_pm); + +MODULE_DESCRIPTION("Cirrus Logic CS42448/CS42888 ALSA SoC Codec Driver"); +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42xx8.h b/sound/soc/codecs/cs42xx8.h new file mode 100644 index 000000000000..da0b94aee419 --- /dev/null +++ b/sound/soc/codecs/cs42xx8.h @@ -0,0 +1,238 @@ +/* + * cs42xx8.h - Cirrus Logic CS42448/CS42888 Audio CODEC driver 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 _CS42XX8_H +#define _CS42XX8_H + +struct cs42xx8_driver_data { + char name[32]; + int num_adcs; +}; + +extern const struct dev_pm_ops cs42xx8_pm; +extern const struct cs42xx8_driver_data cs42448_data; +extern const struct cs42xx8_driver_data cs42888_data; +extern const struct regmap_config cs42xx8_regmap_config; +int cs42xx8_probe(struct device *dev, struct regmap *regmap); + +/* CS42888 register map */ +#define CS42XX8_CHIPID 0x01 /* Chip ID */ +#define CS42XX8_PWRCTL 0x02 /* Power Control */ +#define CS42XX8_FUNCMOD 0x03 /* Functional Mode */ +#define CS42XX8_INTF 0x04 /* Interface Formats */ +#define CS42XX8_ADCCTL 0x05 /* ADC Control */ +#define CS42XX8_TXCTL 0x06 /* Transition Control */ +#define CS42XX8_DACMUTE 0x07 /* DAC Mute Control */ +#define CS42XX8_VOLAOUT1 0x08 /* Volume Control AOUT1 */ +#define CS42XX8_VOLAOUT2 0x09 /* Volume Control AOUT2 */ +#define CS42XX8_VOLAOUT3 0x0A /* Volume Control AOUT3 */ +#define CS42XX8_VOLAOUT4 0x0B /* Volume Control AOUT4 */ +#define CS42XX8_VOLAOUT5 0x0C /* Volume Control AOUT5 */ +#define CS42XX8_VOLAOUT6 0x0D /* Volume Control AOUT6 */ +#define CS42XX8_VOLAOUT7 0x0E /* Volume Control AOUT7 */ +#define CS42XX8_VOLAOUT8 0x0F /* Volume Control AOUT8 */ +#define CS42XX8_DACINV 0x10 /* DAC Channel Invert */ +#define CS42XX8_VOLAIN1 0x11 /* Volume Control AIN1 */ +#define CS42XX8_VOLAIN2 0x12 /* Volume Control AIN2 */ +#define CS42XX8_VOLAIN3 0x13 /* Volume Control AIN3 */ +#define CS42XX8_VOLAIN4 0x14 /* Volume Control AIN4 */ +#define CS42XX8_VOLAIN5 0x15 /* Volume Control AIN5 */ +#define CS42XX8_VOLAIN6 0x16 /* Volume Control AIN6 */ +#define CS42XX8_ADCINV 0x17 /* ADC Channel Invert */ +#define CS42XX8_STATUSCTL 0x18 /* Status Control */ +#define CS42XX8_STATUS 0x19 /* Status */ +#define CS42XX8_STATUSM 0x1A /* Status Mask */ +#define CS42XX8_MUTEC 0x1B /* MUTEC Pin Control */ + +#define CS42XX8_FIRSTREG CS42XX8_CHIPID +#define CS42XX8_LASTREG CS42XX8_MUTEC +#define CS42XX8_NUMREGS (CS42XX8_LASTREG - CS42XX8_FIRSTREG + 1) +#define CS42XX8_I2C_INCR 0x80 + +/* Chip I.D. and Revision Register (Address 01h) */ +#define CS42XX8_CHIPID_CHIP_ID_MASK 0xF0 +#define CS42XX8_CHIPID_REV_ID_MASK 0x0F + +/* Power Control (Address 02h) */ +#define CS42XX8_PWRCTL_PDN_ADC3_SHIFT 7 +#define CS42XX8_PWRCTL_PDN_ADC3_MASK (1 << CS42XX8_PWRCTL_PDN_ADC3_SHIFT) +#define CS42XX8_PWRCTL_PDN_ADC3 (1 << CS42XX8_PWRCTL_PDN_ADC3_SHIFT) +#define CS42XX8_PWRCTL_PDN_ADC2_SHIFT 6 +#define CS42XX8_PWRCTL_PDN_ADC2_MASK (1 << CS42XX8_PWRCTL_PDN_ADC2_SHIFT) +#define CS42XX8_PWRCTL_PDN_ADC2 (1 << CS42XX8_PWRCTL_PDN_ADC2_SHIFT) +#define CS42XX8_PWRCTL_PDN_ADC1_SHIFT 5 +#define CS42XX8_PWRCTL_PDN_ADC1_MASK (1 << CS42XX8_PWRCTL_PDN_ADC1_SHIFT) +#define CS42XX8_PWRCTL_PDN_ADC1 (1 << CS42XX8_PWRCTL_PDN_ADC1_SHIFT) +#define CS42XX8_PWRCTL_PDN_DAC4_SHIFT 4 +#define CS42XX8_PWRCTL_PDN_DAC4_MASK (1 << CS42XX8_PWRCTL_PDN_DAC4_SHIFT) +#define CS42XX8_PWRCTL_PDN_DAC4 (1 << CS42XX8_PWRCTL_PDN_DAC4_SHIFT) +#define CS42XX8_PWRCTL_PDN_DAC3_SHIFT 3 +#define CS42XX8_PWRCTL_PDN_DAC3_MASK (1 << CS42XX8_PWRCTL_PDN_DAC3_SHIFT) +#define CS42XX8_PWRCTL_PDN_DAC3 (1 << CS42XX8_PWRCTL_PDN_DAC3_SHIFT) +#define CS42XX8_PWRCTL_PDN_DAC2_SHIFT 2 +#define CS42XX8_PWRCTL_PDN_DAC2_MASK (1 << CS42XX8_PWRCTL_PDN_DAC2_SHIFT) +#define CS42XX8_PWRCTL_PDN_DAC2 (1 << CS42XX8_PWRCTL_PDN_DAC2_SHIFT) +#define CS42XX8_PWRCTL_PDN_DAC1_SHIFT 1 +#define CS42XX8_PWRCTL_PDN_DAC1_MASK (1 << CS42XX8_PWRCTL_PDN_DAC1_SHIFT) +#define CS42XX8_PWRCTL_PDN_DAC1 (1 << CS42XX8_PWRCTL_PDN_DAC1_SHIFT) +#define CS42XX8_PWRCTL_PDN_SHIFT 0 +#define CS42XX8_PWRCTL_PDN_MASK (1 << CS42XX8_PWRCTL_PDN_SHIFT) +#define CS42XX8_PWRCTL_PDN (1 << CS42XX8_PWRCTL_PDN_SHIFT) + +/* Functional Mode (Address 03h) */ +#define CS42XX8_FUNCMOD_DAC_FM_SHIFT 6 +#define CS42XX8_FUNCMOD_DAC_FM_WIDTH 2 +#define CS42XX8_FUNCMOD_DAC_FM_MASK (((1 << CS42XX8_FUNCMOD_DAC_FM_WIDTH) - 1) << CS42XX8_FUNCMOD_DAC_FM_SHIFT) +#define CS42XX8_FUNCMOD_DAC_FM(v) ((v) << CS42XX8_FUNCMOD_DAC_FM_SHIFT) +#define CS42XX8_FUNCMOD_ADC_FM_SHIFT 4 +#define CS42XX8_FUNCMOD_ADC_FM_WIDTH 2 +#define CS42XX8_FUNCMOD_ADC_FM_MASK (((1 << CS42XX8_FUNCMOD_ADC_FM_WIDTH) - 1) << CS42XX8_FUNCMOD_ADC_FM_SHIFT) +#define CS42XX8_FUNCMOD_ADC_FM(v) ((v) << CS42XX8_FUNCMOD_ADC_FM_SHIFT) +#define CS42XX8_FUNCMOD_xC_FM_MASK(x) ((x) ? CS42XX8_FUNCMOD_DAC_FM_MASK : CS42XX8_FUNCMOD_ADC_FM_MASK) +#define CS42XX8_FUNCMOD_xC_FM(x, v) ((x) ? CS42XX8_FUNCMOD_DAC_FM(v) : CS42XX8_FUNCMOD_ADC_FM(v)) +#define CS42XX8_FUNCMOD_MFREQ_SHIFT 1 +#define CS42XX8_FUNCMOD_MFREQ_WIDTH 3 +#define CS42XX8_FUNCMOD_MFREQ_MASK (((1 << CS42XX8_FUNCMOD_MFREQ_WIDTH) - 1) << CS42XX8_FUNCMOD_MFREQ_SHIFT) +#define CS42XX8_FUNCMOD_MFREQ_256(s) ((0 << CS42XX8_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42XX8_FUNCMOD_MFREQ_384(s) ((1 << CS42XX8_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42XX8_FUNCMOD_MFREQ_512(s) ((2 << CS42XX8_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42XX8_FUNCMOD_MFREQ_768(s) ((3 << CS42XX8_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42XX8_FUNCMOD_MFREQ_1024(s) ((4 << CS42XX8_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) + +#define CS42XX8_FM_SINGLE 0 +#define CS42XX8_FM_DOUBLE 1 +#define CS42XX8_FM_QUAD 2 +#define CS42XX8_FM_AUTO 3 + +/* Interface Formats (Address 04h) */ +#define CS42XX8_INTF_FREEZE_SHIFT 7 +#define CS42XX8_INTF_FREEZE_MASK (1 << CS42XX8_INTF_FREEZE_SHIFT) +#define CS42XX8_INTF_FREEZE (1 << CS42XX8_INTF_FREEZE_SHIFT) +#define CS42XX8_INTF_AUX_DIF_SHIFT 6 +#define CS42XX8_INTF_AUX_DIF_MASK (1 << CS42XX8_INTF_AUX_DIF_SHIFT) +#define CS42XX8_INTF_AUX_DIF (1 << CS42XX8_INTF_AUX_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_SHIFT 3 +#define CS42XX8_INTF_DAC_DIF_WIDTH 3 +#define CS42XX8_INTF_DAC_DIF_MASK (((1 << CS42XX8_INTF_DAC_DIF_WIDTH) - 1) << CS42XX8_INTF_DAC_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_LEFTJ (0 << CS42XX8_INTF_DAC_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_I2S (1 << CS42XX8_INTF_DAC_DIF_SHIFT) +#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_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) +#define CS42XX8_INTF_ADC_DIF_LEFTJ (0 << CS42XX8_INTF_ADC_DIF_SHIFT) +#define CS42XX8_INTF_ADC_DIF_I2S (1 << CS42XX8_INTF_ADC_DIF_SHIFT) +#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) + +/* ADC Control & DAC De-Emphasis (Address 05h) */ +#define CS42XX8_ADCCTL_ADC_HPF_FREEZE_SHIFT 7 +#define CS42XX8_ADCCTL_ADC_HPF_FREEZE_MASK (1 << CS42XX8_ADCCTL_ADC_HPF_FREEZE_SHIFT) +#define CS42XX8_ADCCTL_ADC_HPF_FREEZE (1 << CS42XX8_ADCCTL_ADC_HPF_FREEZE_SHIFT) +#define CS42XX8_ADCCTL_DAC_DEM_SHIFT 5 +#define CS42XX8_ADCCTL_DAC_DEM_MASK (1 << CS42XX8_ADCCTL_DAC_DEM_SHIFT) +#define CS42XX8_ADCCTL_DAC_DEM (1 << CS42XX8_ADCCTL_DAC_DEM_SHIFT) +#define CS42XX8_ADCCTL_ADC1_SINGLE_SHIFT 4 +#define CS42XX8_ADCCTL_ADC1_SINGLE_MASK (1 << CS42XX8_ADCCTL_ADC1_SINGLE_SHIFT) +#define CS42XX8_ADCCTL_ADC1_SINGLE (1 << CS42XX8_ADCCTL_ADC1_SINGLE_SHIFT) +#define CS42XX8_ADCCTL_ADC2_SINGLE_SHIFT 3 +#define CS42XX8_ADCCTL_ADC2_SINGLE_MASK (1 << CS42XX8_ADCCTL_ADC2_SINGLE_SHIFT) +#define CS42XX8_ADCCTL_ADC2_SINGLE (1 << CS42XX8_ADCCTL_ADC2_SINGLE_SHIFT) +#define CS42XX8_ADCCTL_ADC3_SINGLE_SHIFT 2 +#define CS42XX8_ADCCTL_ADC3_SINGLE_MASK (1 << CS42XX8_ADCCTL_ADC3_SINGLE_SHIFT) +#define CS42XX8_ADCCTL_ADC3_SINGLE (1 << CS42XX8_ADCCTL_ADC3_SINGLE_SHIFT) +#define CS42XX8_ADCCTL_AIN5_MUX_SHIFT 1 +#define CS42XX8_ADCCTL_AIN5_MUX_MASK (1 << CS42XX8_ADCCTL_AIN5_MUX_SHIFT) +#define CS42XX8_ADCCTL_AIN5_MUX (1 << CS42XX8_ADCCTL_AIN5_MUX_SHIFT) +#define CS42XX8_ADCCTL_AIN6_MUX_SHIFT 0 +#define CS42XX8_ADCCTL_AIN6_MUX_MASK (1 << CS42XX8_ADCCTL_AIN6_MUX_SHIFT) +#define CS42XX8_ADCCTL_AIN6_MUX (1 << CS42XX8_ADCCTL_AIN6_MUX_SHIFT) + +/* Transition Control (Address 06h) */ +#define CS42XX8_TXCTL_DAC_SNGVOL_SHIFT 7 +#define CS42XX8_TXCTL_DAC_SNGVOL_MASK (1 << CS42XX8_TXCTL_DAC_SNGVOL_SHIFT) +#define CS42XX8_TXCTL_DAC_SNGVOL (1 << CS42XX8_TXCTL_DAC_SNGVOL_SHIFT) +#define CS42XX8_TXCTL_DAC_SZC_SHIFT 5 +#define CS42XX8_TXCTL_DAC_SZC_WIDTH 2 +#define CS42XX8_TXCTL_DAC_SZC_MASK (((1 << CS42XX8_TXCTL_DAC_SZC_WIDTH) - 1) << CS42XX8_TXCTL_DAC_SZC_SHIFT) +#define CS42XX8_TXCTL_DAC_SZC_IC (0 << CS42XX8_TXCTL_DAC_SZC_SHIFT) +#define CS42XX8_TXCTL_DAC_SZC_ZC (1 << CS42XX8_TXCTL_DAC_SZC_SHIFT) +#define CS42XX8_TXCTL_DAC_SZC_SR (2 << CS42XX8_TXCTL_DAC_SZC_SHIFT) +#define CS42XX8_TXCTL_DAC_SZC_SRZC (3 << CS42XX8_TXCTL_DAC_SZC_SHIFT) +#define CS42XX8_TXCTL_AMUTE_SHIFT 4 +#define CS42XX8_TXCTL_AMUTE_MASK (1 << CS42XX8_TXCTL_AMUTE_SHIFT) +#define CS42XX8_TXCTL_AMUTE (1 << CS42XX8_TXCTL_AMUTE_SHIFT) +#define CS42XX8_TXCTL_MUTE_ADC_SP_SHIFT 3 +#define CS42XX8_TXCTL_MUTE_ADC_SP_MASK (1 << CS42XX8_TXCTL_MUTE_ADC_SP_SHIFT) +#define CS42XX8_TXCTL_MUTE_ADC_SP (1 << CS42XX8_TXCTL_MUTE_ADC_SP_SHIFT) +#define CS42XX8_TXCTL_ADC_SNGVOL_SHIFT 2 +#define CS42XX8_TXCTL_ADC_SNGVOL_MASK (1 << CS42XX8_TXCTL_ADC_SNGVOL_SHIFT) +#define CS42XX8_TXCTL_ADC_SNGVOL (1 << CS42XX8_TXCTL_ADC_SNGVOL_SHIFT) +#define CS42XX8_TXCTL_ADC_SZC_SHIFT 0 +#define CS42XX8_TXCTL_ADC_SZC_MASK (((1 << CS42XX8_TXCTL_ADC_SZC_WIDTH) - 1) << CS42XX8_TXCTL_ADC_SZC_SHIFT) +#define CS42XX8_TXCTL_ADC_SZC_IC (0 << CS42XX8_TXCTL_ADC_SZC_SHIFT) +#define CS42XX8_TXCTL_ADC_SZC_ZC (1 << CS42XX8_TXCTL_ADC_SZC_SHIFT) +#define CS42XX8_TXCTL_ADC_SZC_SR (2 << CS42XX8_TXCTL_ADC_SZC_SHIFT) +#define CS42XX8_TXCTL_ADC_SZC_SRZC (3 << CS42XX8_TXCTL_ADC_SZC_SHIFT) + +/* DAC Channel Mute (Address 07h) */ +#define CS42XX8_DACMUTE_AOUT(n) (0x1 << n) +#define CS42XX8_DACMUTE_ALL 0xff + +/* Status Control (Address 18h)*/ +#define CS42XX8_STATUSCTL_INI_SHIFT 2 +#define CS42XX8_STATUSCTL_INI_WIDTH 2 +#define CS42XX8_STATUSCTL_INI_MASK (((1 << CS42XX8_STATUSCTL_INI_WIDTH) - 1) << CS42XX8_STATUSCTL_INI_SHIFT) +#define CS42XX8_STATUSCTL_INT_ACTIVE_HIGH (0 << CS42XX8_STATUSCTL_INI_SHIFT) +#define CS42XX8_STATUSCTL_INT_ACTIVE_LOW (1 << CS42XX8_STATUSCTL_INI_SHIFT) +#define CS42XX8_STATUSCTL_INT_OPEN_DRAIN (2 << CS42XX8_STATUSCTL_INI_SHIFT) + +/* Status (Address 19h)*/ +#define CS42XX8_STATUS_DAC_CLK_ERR_SHIFT 4 +#define CS42XX8_STATUS_DAC_CLK_ERR_MASK (1 << CS42XX8_STATUS_DAC_CLK_ERR_SHIFT) +#define CS42XX8_STATUS_ADC_CLK_ERR_SHIFT 3 +#define CS42XX8_STATUS_ADC_CLK_ERR_MASK (1 << CS42XX8_STATUS_ADC_CLK_ERR_SHIFT) +#define CS42XX8_STATUS_ADC3_OVFL_SHIFT 2 +#define CS42XX8_STATUS_ADC3_OVFL_MASK (1 << CS42XX8_STATUS_ADC3_OVFL_SHIFT) +#define CS42XX8_STATUS_ADC2_OVFL_SHIFT 1 +#define CS42XX8_STATUS_ADC2_OVFL_MASK (1 << CS42XX8_STATUS_ADC2_OVFL_SHIFT) +#define CS42XX8_STATUS_ADC1_OVFL_SHIFT 0 +#define CS42XX8_STATUS_ADC1_OVFL_MASK (1 << CS42XX8_STATUS_ADC1_OVFL_SHIFT) + +/* Status Mask (Address 1Ah) */ +#define CS42XX8_STATUS_DAC_CLK_ERR_M_SHIFT 4 +#define CS42XX8_STATUS_DAC_CLK_ERR_M_MASK (1 << CS42XX8_STATUS_DAC_CLK_ERR_M_SHIFT) +#define CS42XX8_STATUS_ADC_CLK_ERR_M_SHIFT 3 +#define CS42XX8_STATUS_ADC_CLK_ERR_M_MASK (1 << CS42XX8_STATUS_ADC_CLK_ERR_M_SHIFT) +#define CS42XX8_STATUS_ADC3_OVFL_M_SHIFT 2 +#define CS42XX8_STATUS_ADC3_OVFL_M_MASK (1 << CS42XX8_STATUS_ADC3_OVFL_M_SHIFT) +#define CS42XX8_STATUS_ADC2_OVFL_M_SHIFT 1 +#define CS42XX8_STATUS_ADC2_OVFL_M_MASK (1 << CS42XX8_STATUS_ADC2_OVFL_M_SHIFT) +#define CS42XX8_STATUS_ADC1_OVFL_M_SHIFT 0 +#define CS42XX8_STATUS_ADC1_OVFL_M_MASK (1 << CS42XX8_STATUS_ADC1_OVFL_M_SHIFT) + +/* MUTEC Pin Control (Address 1Bh) */ +#define CS42XX8_MUTEC_MCPOLARITY_SHIFT 1 +#define CS42XX8_MUTEC_MCPOLARITY_MASK (1 << CS42XX8_MUTEC_MCPOLARITY_SHIFT) +#define CS42XX8_MUTEC_MCPOLARITY_ACTIVE_LOW (0 << CS42XX8_MUTEC_MCPOLARITY_SHIFT) +#define CS42XX8_MUTEC_MCPOLARITY_ACTIVE_HIGH (1 << CS42XX8_MUTEC_MCPOLARITY_SHIFT) +#define CS42XX8_MUTEC_MUTEC_ACTIVE_SHIFT 0 +#define CS42XX8_MUTEC_MUTEC_ACTIVE_MASK (1 << CS42XX8_MUTEC_MUTEC_ACTIVE_SHIFT) +#define CS42XX8_MUTEC_MUTEC_ACTIVE (1 << CS42XX8_MUTEC_MUTEC_ACTIVE_SHIFT) +#endif /* _CS42XX8_H */ -- cgit v1.2.3 From 43b956349de28884265e8237a5cd1f669fbaa485 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 12 Mar 2014 15:27:31 +0100 Subject: ASoC: mfld_machine: Convert to table based DAPM and control setup Use table based setup to register the controls and DAPM widgets and routes. This on one hand makes the code a bit cleaner and on the other hand the board level DAPM elements get registered in the card's DAPM context rather than in the CODEC's DAPM context. The mfld_machine driver is a bit special in that it directly writes to one of the CODEC registers from one of the control handlers. Previous to this patch it was able to get a pointer to the CODEC from the control, since the control was registered with the CODEC. This won't be possible anymore once the control is registered with the card. Since there are already global variables in the driver accessed in the same function the patch adds a global variable that holds a pointer to the CODEC and uses that. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/intel/mfld_machine.c | 49 +++++++++++++++++------------------------- 1 file changed, 20 insertions(+), 29 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/mfld_machine.c b/sound/soc/intel/mfld_machine.c index 0cef32e9d402..461694263c46 100644 --- a/sound/soc/intel/mfld_machine.c +++ b/sound/soc/intel/mfld_machine.c @@ -53,6 +53,7 @@ enum soc_mic_bias_zones { static unsigned int hs_switch; static unsigned int lo_dac; +static struct snd_soc_codec *mfld_codec; struct mfld_mc_private { void __iomem *int_base; @@ -100,8 +101,8 @@ static int headset_get_switch(struct snd_kcontrol *kcontrol, static int headset_set_switch(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_context *dapm = &codec->dapm; + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_context *dapm = &card->dapm; if (ucontrol->value.integer.value[0] == hs_switch) return 0; @@ -127,10 +128,8 @@ static int headset_set_switch(struct snd_kcontrol *kcontrol, return 0; } -static void lo_enable_out_pins(struct snd_soc_codec *codec) +static void lo_enable_out_pins(struct snd_soc_dapm_context *dapm) { - struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_enable_pin_unlocked(dapm, "IHFOUTL"); snd_soc_dapm_enable_pin_unlocked(dapm, "IHFOUTR"); snd_soc_dapm_enable_pin_unlocked(dapm, "LINEOUTL"); @@ -156,8 +155,8 @@ static int lo_get_switch(struct snd_kcontrol *kcontrol, static int lo_set_switch(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_context *dapm = &codec->dapm; + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_context *dapm = &card->dapm; if (ucontrol->value.integer.value[0] == lo_dac) return 0; @@ -167,35 +166,35 @@ static int lo_set_switch(struct snd_kcontrol *kcontrol, /* we dont want to work with last state of lineout so just enable all * pins and then disable pins not required */ - lo_enable_out_pins(codec); + lo_enable_out_pins(dapm); switch (ucontrol->value.integer.value[0]) { case 0: pr_debug("set vibra path\n"); snd_soc_dapm_disable_pin_unlocked(dapm, "VIB1OUT"); snd_soc_dapm_disable_pin_unlocked(dapm, "VIB2OUT"); - snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0); + snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0); break; case 1: pr_debug("set hs path\n"); snd_soc_dapm_disable_pin_unlocked(dapm, "Headphones"); snd_soc_dapm_disable_pin_unlocked(dapm, "EPOUT"); - snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x22); + snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0x22); break; case 2: pr_debug("set spkr path\n"); snd_soc_dapm_disable_pin_unlocked(dapm, "IHFOUTL"); snd_soc_dapm_disable_pin_unlocked(dapm, "IHFOUTR"); - snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x44); + snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0x44); break; case 3: pr_debug("set null path\n"); snd_soc_dapm_disable_pin_unlocked(dapm, "LINEOUTL"); snd_soc_dapm_disable_pin_unlocked(dapm, "LINEOUTR"); - snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x66); + snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0x66); break; } @@ -238,26 +237,11 @@ static void mfld_jack_check(unsigned int intr_status) static int mfld_init(struct snd_soc_pcm_runtime *runtime) { - struct snd_soc_codec *codec = runtime->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; + struct snd_soc_dapm_context *dapm = &runtime->card->dapm; int ret_val; - /* Add jack sense widgets */ - snd_soc_dapm_new_controls(dapm, mfld_widgets, ARRAY_SIZE(mfld_widgets)); - - /* Set up the map */ - snd_soc_dapm_add_routes(dapm, mfld_map, ARRAY_SIZE(mfld_map)); + mfld_codec = runtime->codec; - /* always connected */ - snd_soc_dapm_enable_pin(dapm, "Headphones"); - snd_soc_dapm_enable_pin(dapm, "Mic"); - - ret_val = snd_soc_add_codec_controls(codec, mfld_snd_controls, - ARRAY_SIZE(mfld_snd_controls)); - if (ret_val) { - pr_err("soc_add_controls failed %d", ret_val); - return ret_val; - } /* default is earpiece pin, userspace sets it explcitly */ snd_soc_dapm_disable_pin(dapm, "Headphones"); /* default is lineout NC, userspace sets it explcitly */ @@ -352,6 +336,13 @@ static struct snd_soc_card snd_soc_card_mfld = { .owner = THIS_MODULE, .dai_link = mfld_msic_dailink, .num_links = ARRAY_SIZE(mfld_msic_dailink), + + .controls = mfld_snd_controls, + .num_controls = ARRAY_SIZE(mfld_snd_controls), + .dapm_widgets = mfld_widgets, + .num_dapm_widgets = ARRAY_SIZE(mfld_widgets), + .dapm_routes = mfld_map, + .num_dapm_routes = ARRAY_SIZE(mfld_map), }; static irqreturn_t snd_mfld_jack_intr_handler(int irq, void *dev) -- cgit v1.2.3 From 90e8e50fce3585d6f9902701de08389b027dadc6 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 17 Mar 2014 19:29:55 -0700 Subject: ASoC: rsnd: add DeviceTree support Support for loading the Renesas R-Car sound driver via DeviceTree. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/renesas,rsnd.txt | 96 ++++++++++++++++ sound/soc/sh/rcar/adg.c | 1 + sound/soc/sh/rcar/core.c | 122 ++++++++++++++++++++- sound/soc/sh/rcar/gen.c | 15 +++ sound/soc/sh/rcar/rsnd.h | 11 ++ sound/soc/sh/rcar/src.c | 36 ++++++ sound/soc/sh/rcar/ssi.c | 56 ++++++++++ 7 files changed, 334 insertions(+), 3 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/renesas,rsnd.txt (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt new file mode 100644 index 000000000000..7c6d33f29796 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -0,0 +1,96 @@ +Renesas R-Car sound + +Required properties: +- compatible : "renesas,rcar_sound-gen1" if generation1 + "renesas,rcar_sound-gen2" if generation2 +- reg : Should contain the register physical address. + required register is + SRU/ADG/SSI if generation1 + SRU/ADG/SSIU/SSI if generation2 +- rcar_sound,ssi : SSI subnode +- rcar_sound,scu : SCU subnode +- rcar_sound,dai : DAI subnode + +SSI subnode properties: +- interrupts : Should contain SSI interrupt for PIO transfer +- shared-pin : if shared clock pin + +DAI subnode properties: +- playback : list of playback modules +- capture : list of capture modules + +Example: + +rcar_sound: rcar_sound@0xffd90000 { + #sound-dai-cells = <1>; + compatible = "renesas,rcar_sound-gen2"; + reg = <0 0xec500000 0 0x1000>, /* SCU */ + <0 0xec5a0000 0 0x100>, /* ADG */ + <0 0xec540000 0 0x1000>, /* SSIU */ + <0 0xec541000 0 0x1280>; /* SSI */ + + rcar_sound,src { + src0: src@0 { }; + src1: src@1 { }; + src2: src@2 { }; + src3: src@3 { }; + src4: src@4 { }; + src5: src@5 { }; + src6: src@6 { }; + src7: src@7 { }; + src8: src@8 { }; + src9: src@9 { }; + }; + + rcar_sound,ssi { + ssi0: ssi@0 { + interrupts = <0 370 IRQ_TYPE_LEVEL_HIGH>; + }; + ssi1: ssi@1 { + interrupts = <0 371 IRQ_TYPE_LEVEL_HIGH>; + }; + ssi2: ssi@2 { + interrupts = <0 372 IRQ_TYPE_LEVEL_HIGH>; + }; + ssi3: ssi@3 { + interrupts = <0 373 IRQ_TYPE_LEVEL_HIGH>; + }; + ssi4: ssi@4 { + interrupts = <0 374 IRQ_TYPE_LEVEL_HIGH>; + }; + ssi5: ssi@5 { + interrupts = <0 375 IRQ_TYPE_LEVEL_HIGH>; + }; + ssi6: ssi@6 { + interrupts = <0 376 IRQ_TYPE_LEVEL_HIGH>; + }; + ssi7: ssi@7 { + interrupts = <0 377 IRQ_TYPE_LEVEL_HIGH>; + }; + ssi8: ssi@8 { + interrupts = <0 378 IRQ_TYPE_LEVEL_HIGH>; + }; + ssi9: ssi@9 { + interrupts = <0 379 IRQ_TYPE_LEVEL_HIGH>; + }; + }; + + rcar_sound,dai { + dai0 { + playback = <&ssi5 &src5>; + capture = <&ssi6>; + }; + dai1 { + playback = <&ssi3>; + }; + dai2 { + capture = <&ssi4>; + }; + dai3 { + playback = <&ssi7>; + }; + dai4 { + capture = <&ssi8>; + }; + }; +}; diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 953f1cce982d..69c44269ebdb 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -392,6 +392,7 @@ static void rsnd_adg_ssi_clk_init(struct rsnd_priv *priv, struct rsnd_adg *adg) } int rsnd_adg_probe(struct platform_device *pdev, + const struct rsnd_of_data *of_data, struct rsnd_priv *priv) { struct rsnd_adg *adg; diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 6a1b45df8101..e77f7716f1d7 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -100,6 +100,21 @@ #define RSND_RATES SNDRV_PCM_RATE_8000_96000 #define RSND_FMTS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE) +static struct rsnd_of_data rsnd_of_data_gen1 = { + .flags = RSND_GEN1, +}; + +static struct rsnd_of_data rsnd_of_data_gen2 = { + .flags = RSND_GEN2, +}; + +static struct of_device_id rsnd_of_match[] = { + { .compatible = "renesas,rcar_sound-gen1", .data = &rsnd_of_data_gen1 }, + { .compatible = "renesas,rcar_sound-gen2", .data = &rsnd_of_data_gen2 }, + {}, +}; +MODULE_DEVICE_TABLE(of, rsnd_of_match); + /* * rsnd_platform functions */ @@ -620,7 +635,92 @@ static int rsnd_path_init(struct rsnd_priv *priv, return ret; } +static void rsnd_of_parse_dai(struct platform_device *pdev, + const struct rsnd_of_data *of_data, + struct rsnd_priv *priv) +{ + struct device_node *dai_node, *dai_np; + struct device_node *ssi_node, *ssi_np; + struct device_node *src_node, *src_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; + + if (!of_data) + return; + + dai_node = of_get_child_by_name(dev->of_node, "rcar_sound,dai"); + if (!dai_node) + return; + + nr = of_get_child_count(dai_node); + if (!nr) + return; + + dai_info = devm_kzalloc(dev, + sizeof(struct rsnd_dai_platform_info) * nr, + GFP_KERNEL); + if (!dai_info) { + dev_err(dev, "dai info allocation error\n"); + return; + } + + info->dai_info_nr = nr; + info->dai_info = dai_info; + + 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"); + +#define mod_parse(name) \ +if (name##_node) { \ + struct rsnd_##name##_platform_info *name##_info; \ + \ + name##_i = 0; \ + for_each_child_of_node(name##_node, name##_np) { \ + name##_info = info->name##_info + name##_i; \ + \ + if (name##_np == playback) \ + dai_info->playback.name = name##_info; \ + if (name##_np == capture) \ + dai_info->capture.name = name##_info; \ + \ + name##_i++; \ + } \ +} + + /* + * parse all dai + */ + dai_i = 0; + for_each_child_of_node(dai_node, dai_np) { + dai_info = info->dai_info + dai_i; + + for (i = 0;; i++) { + + playback = of_parse_phandle(dai_np, "playback", i); + capture = of_parse_phandle(dai_np, "capture", i); + + if (!playback && !capture) + break; + + mod_parse(ssi); + mod_parse(src); + + if (playback) + of_node_put(playback); + if (capture) + of_node_put(capture); + } + + dai_i++; + } +} + static int rsnd_dai_probe(struct platform_device *pdev, + const struct rsnd_of_data *of_data, struct rsnd_priv *priv) { struct snd_soc_dai_driver *drv; @@ -628,13 +728,16 @@ static int rsnd_dai_probe(struct platform_device *pdev, struct rsnd_dai *rdai; struct rsnd_mod *pmod, *cmod; struct device *dev = rsnd_priv_to_dev(priv); - int dai_nr = info->dai_info_nr; + int dai_nr; int i; + rsnd_of_parse_dai(pdev, of_data, priv); + /* * dai_nr should be set via dai_info_nr, * but allow it to keeping compatible */ + dai_nr = info->dai_info_nr; if (!dai_nr) { /* get max dai nr */ for (dai_nr = 0; dai_nr < 32; dai_nr++) { @@ -802,7 +905,10 @@ static int rsnd_probe(struct platform_device *pdev) struct rsnd_priv *priv; struct device *dev = &pdev->dev; struct rsnd_dai *rdai; + const struct of_device_id *of_id = of_match_device(rsnd_of_match, dev); + const struct rsnd_of_data *of_data; int (*probe_func[])(struct platform_device *pdev, + const struct rsnd_of_data *of_data, struct rsnd_priv *priv) = { rsnd_gen_probe, rsnd_ssi_probe, @@ -812,7 +918,16 @@ static int rsnd_probe(struct platform_device *pdev) }; int ret, i; - info = pdev->dev.platform_data; + info = NULL; + of_data = NULL; + if (of_id) { + info = devm_kzalloc(&pdev->dev, + sizeof(struct rcar_snd_info), GFP_KERNEL); + of_data = of_id->data; + } else { + info = pdev->dev.platform_data; + } + if (!info) { dev_err(dev, "driver needs R-Car sound information\n"); return -ENODEV; @@ -835,7 +950,7 @@ static int rsnd_probe(struct platform_device *pdev) * init each module */ for (i = 0; i < ARRAY_SIZE(probe_func); i++) { - ret = probe_func[i](pdev, priv); + ret = probe_func[i](pdev, of_data, priv); if (ret) return ret; } @@ -903,6 +1018,7 @@ static int rsnd_remove(struct platform_device *pdev) static struct platform_driver rsnd_driver = { .driver = { .name = "rcar_sound", + .of_match_table = rsnd_of_match, }, .probe = rsnd_probe, .remove = rsnd_remove, diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 9094970dbdfb..50a1ef3eb1c6 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -359,13 +359,28 @@ static int rsnd_gen1_probe(struct platform_device *pdev, /* * Gen */ +static void rsnd_of_parse_gen(struct platform_device *pdev, + const struct rsnd_of_data *of_data, + struct rsnd_priv *priv) +{ + struct rcar_snd_info *info = priv->info; + + if (!of_data) + return; + + info->flags = of_data->flags; +} + int rsnd_gen_probe(struct platform_device *pdev, + const struct rsnd_of_data *of_data, struct rsnd_priv *priv) { struct device *dev = rsnd_priv_to_dev(priv); struct rsnd_gen *gen; int ret; + rsnd_of_parse_gen(pdev, of_data, priv); + gen = devm_kzalloc(dev, sizeof(*gen), GFP_KERNEL); if (!gen) { dev_err(dev, "GEN allocate failed\n"); diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index c46e0afa54ae..619d198c7d2e 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -17,6 +17,8 @@ #include #include #include +#include +#include #include #include #include @@ -113,6 +115,7 @@ enum rsnd_reg { #define RSND_REG_SRCOUT_TIMSEL4 RSND_REG_SHARE18 #define RSND_REG_AUDIO_CLK_SEL2 RSND_REG_SHARE19 +struct rsnd_of_data; struct rsnd_priv; struct rsnd_mod; struct rsnd_dai; @@ -260,6 +263,7 @@ int rsnd_dai_pointer_offset(struct rsnd_dai_stream *io, int additional); * R-Car Gen1/Gen2 */ int rsnd_gen_probe(struct platform_device *pdev, + const struct rsnd_of_data *of_data, struct rsnd_priv *priv); void __iomem *rsnd_gen_reg_get(struct rsnd_priv *priv, struct rsnd_mod *mod, @@ -273,6 +277,7 @@ void __iomem *rsnd_gen_reg_get(struct rsnd_priv *priv, int rsnd_adg_ssi_clk_stop(struct rsnd_mod *mod); int rsnd_adg_ssi_clk_try_start(struct rsnd_mod *mod, unsigned int rate); int rsnd_adg_probe(struct platform_device *pdev, + const struct rsnd_of_data *of_data, struct rsnd_priv *priv); int rsnd_adg_set_convert_clk_gen1(struct rsnd_priv *priv, struct rsnd_mod *mod, @@ -290,6 +295,10 @@ int rsnd_adg_set_convert_timing_gen2(struct rsnd_mod *mod, /* * R-Car sound priv */ +struct rsnd_of_data { + u32 flags; +}; + struct rsnd_priv { struct device *dev; @@ -348,6 +357,7 @@ struct rsnd_priv { * R-Car SRC */ int rsnd_src_probe(struct platform_device *pdev, + const struct rsnd_of_data *of_data, struct rsnd_priv *priv); struct rsnd_mod *rsnd_src_mod_get(struct rsnd_priv *priv, int id); unsigned int rsnd_src_get_ssi_rate(struct rsnd_priv *priv, @@ -366,6 +376,7 @@ int rsnd_src_enable_ssi_irq(struct rsnd_mod *ssi_mod, * R-Car SSI */ int rsnd_ssi_probe(struct platform_device *pdev, + const struct rsnd_of_data *of_data, struct rsnd_priv *priv); struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id); struct rsnd_mod *rsnd_ssi_mod_get_frm_dai(struct rsnd_priv *priv, diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index ea6a214985d0..eee75ebf961c 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -628,7 +628,41 @@ struct rsnd_mod *rsnd_src_mod_get(struct rsnd_priv *priv, int id) return &((struct rsnd_src *)(priv->src) + id)->mod; } +static void rsnd_of_parse_src(struct platform_device *pdev, + const struct rsnd_of_data *of_data, + struct rsnd_priv *priv) +{ + struct device_node *src_node; + struct rcar_snd_info *info = rsnd_priv_to_info(priv); + struct rsnd_src_platform_info *src_info; + struct device *dev = &pdev->dev; + int nr; + + if (!of_data) + return; + + src_node = of_get_child_by_name(dev->of_node, "rcar_sound,src"); + if (!src_node) + return; + + nr = of_get_child_count(src_node); + if (!nr) + return; + + src_info = devm_kzalloc(dev, + sizeof(struct rsnd_src_platform_info) * nr, + GFP_KERNEL); + if (!src_info) { + dev_err(dev, "src info allocation error\n"); + return; + } + + info->src_info = src_info; + info->src_info_nr = nr; +} + int rsnd_src_probe(struct platform_device *pdev, + const struct rsnd_of_data *of_data, struct rsnd_priv *priv) { struct rcar_snd_info *info = rsnd_priv_to_info(priv); @@ -639,6 +673,8 @@ int rsnd_src_probe(struct platform_device *pdev, char name[RSND_SRC_NAME_SIZE]; int i, nr; + rsnd_of_parse_src(pdev, of_data, priv); + /* * init SRC */ diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 633b23d209b9..4b7e20603dd7 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -588,7 +588,61 @@ static void rsnd_ssi_parent_clk_setup(struct rsnd_priv *priv, struct rsnd_ssi *s } } + +static void rsnd_of_parse_ssi(struct platform_device *pdev, + const struct rsnd_of_data *of_data, + struct rsnd_priv *priv) +{ + struct device_node *node; + struct device_node *np; + struct rsnd_ssi_platform_info *ssi_info; + struct rcar_snd_info *info = rsnd_priv_to_info(priv); + struct device *dev = &pdev->dev; + int nr, i; + + if (!of_data) + return; + + node = of_get_child_by_name(dev->of_node, "rcar_sound,ssi"); + if (!node) + return; + + nr = of_get_child_count(node); + if (!nr) + return; + + ssi_info = devm_kzalloc(dev, + sizeof(struct rsnd_ssi_platform_info) * nr, + GFP_KERNEL); + if (!ssi_info) { + dev_err(dev, "ssi info allocation error\n"); + return; + } + + info->ssi_info = ssi_info; + info->ssi_info_nr = nr; + + i = -1; + for_each_child_of_node(node, np) { + i++; + + ssi_info = info->ssi_info + i; + + /* + * pin settings + */ + if (of_get_property(np, "shared-pin", NULL)) + ssi_info->flags |= RSND_SSI_CLK_PIN_SHARE; + + /* + * irq + */ + ssi_info->pio_irq = irq_of_parse_and_map(np, 0); + } +} + int rsnd_ssi_probe(struct platform_device *pdev, + const struct rsnd_of_data *of_data, struct rsnd_priv *priv) { struct rcar_snd_info *info = rsnd_priv_to_info(priv); @@ -600,6 +654,8 @@ int rsnd_ssi_probe(struct platform_device *pdev, char name[RSND_SSI_NAME_SIZE]; int i, nr; + rsnd_of_parse_ssi(pdev, of_data, priv); + /* * init SSI */ -- cgit v1.2.3 From 546d4dd3bb99466e4268256bad615a5ed6457a55 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 21 Mar 2014 20:23:51 +0100 Subject: ASoC: mfld_machine: Fix compile error Fixes: 115f3f8 ("ASoC: mfld_machine: Convert to table based DAPM and control setup") Reported-by: kbuild test robot Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/intel/mfld_machine.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/mfld_machine.c b/sound/soc/intel/mfld_machine.c index 461694263c46..031d78783fc8 100644 --- a/sound/soc/intel/mfld_machine.c +++ b/sound/soc/intel/mfld_machine.c @@ -254,7 +254,7 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) snd_soc_dapm_disable_pin(dapm, "LINEINR"); /* Headset and button jack detection */ - ret_val = snd_soc_jack_new(codec, "Intel(R) MID Audio Jack", + ret_val = snd_soc_jack_new(mfld_codec, "Intel(R) MID Audio Jack", SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1, &mfld_jack); if (ret_val) { -- cgit v1.2.3 From 017d9491ce203c620ad1377f46a3ce78d554b2de Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 22 Mar 2014 18:29:15 +0000 Subject: ASoC: pcm: Drop incorrect double/extra frees The changes in "ASoC: pcm: free path list before exiting from error conditions" actually introduced both double frees (in case where the path list was allocated but empty) and frees of unallocated memory (in cases where the error being handled was -ENOMEM. Drop the commit for now. Fixes: e4ad1accb (ASoC: pcm: free path list before exiting from error conditions) Reported-by: Ben Hutchings Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/soc-pcm.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 28522bd03b8e..47e1ce771e65 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1989,7 +1989,6 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card) paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list); if (paths < 0) { - dpcm_path_put(&list); dev_warn(fe->dev, "ASoC: %s no valid %s path\n", fe->dai_link->name, "playback"); mutex_unlock(&card->mutex); @@ -2019,7 +2018,6 @@ capture: paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list); if (paths < 0) { - dpcm_path_put(&list); dev_warn(fe->dev, "ASoC: %s no valid %s path\n", fe->dai_link->name, "capture"); mutex_unlock(&card->mutex); @@ -2084,7 +2082,6 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) fe->dpcm[stream].runtime = fe_substream->runtime; if (dpcm_path_get(fe, stream, &list) <= 0) { - dpcm_path_put(&list); dev_dbg(fe->dev, "ASoC: %s no valid %s route\n", fe->dai_link->name, stream ? "capture" : "playback"); } -- cgit v1.2.3