From 836b31fe1a9791d84ed95783ed42ebd5719f7918 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 23 Apr 2014 14:59:12 +0900 Subject: ASoC: rsnd: call rsnd_dai_pointer_update() from outside of lock rsnd_soc_dai_trigger() will be called after rsnd_dai_pointer_update() function which is using rsnd_lock(). Thus, it should be called from outside of rsnd_lock(). Kernel will be hangup without this patch. Special thanks to Kataoka-san Reported-by: Ryo Kataoka Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 215b668166be..89424470a1f3 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -197,13 +197,12 @@ static void rsnd_dma_complete(void *data) * rsnd_dai_pointer_update() will be called twice, * ant it will breaks io->byte_pos */ - - rsnd_dai_pointer_update(io, io->byte_per_period); - if (dma->submit_loop) rsnd_dma_continue(dma); rsnd_unlock(priv, flags); + + rsnd_dai_pointer_update(io, io->byte_per_period); } static void __rsnd_dma_start(struct rsnd_dma *dma) -- cgit v1.2.3 From b8909783a22b4f169ade830e0aa6dafe313426f1 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 24 Apr 2014 08:25:56 +0200 Subject: ASoC: imx-audmux: Fix section mismatch audmux_debugfs_init() is marked as __init, but is called from imx_audmux_probe() which is not marked as __init. This creates a section mismatch and a potential runtime crash (if imx_audmux_probe() is called after the .init section was dropped). This patch removes the __init annotation from audmux_debugfs_init(), which fixes the following warning: WARNING: sound/soc/built-in.o(.text+0x86960): Section mismatch in reference from the function imx_audmux_probe() to the function .init.text:audmux_debugfs_init() Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/fsl/imx-audmux.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index ac869931d7f1..267717aa96c1 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -145,7 +145,7 @@ static const struct file_operations audmux_debugfs_fops = { .llseek = default_llseek, }; -static void __init audmux_debugfs_init(void) +static void audmux_debugfs_init(void) { int i; char buf[20]; -- cgit v1.2.3 From 2a1c23e3396d5a77d6c8e379d18cc657b6225779 Mon Sep 17 00:00:00 2001 From: Joe Perches Date: Thu, 24 Apr 2014 18:51:03 -0700 Subject: ASoC: tlv320aic31xx: Convert /n to \n Use a newline character appropriately. Signed-off-by: Joe Perches Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic31xx.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index fa158cfe9b32..d1929de641e2 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -376,7 +376,7 @@ static int aic31xx_dapm_power_event(struct snd_soc_dapm_widget *w, reg = AIC31XX_ADCFLAG; break; default: - dev_err(w->codec->dev, "Unknown widget '%s' calling %s/n", + dev_err(w->codec->dev, "Unknown widget '%s' calling %s\n", w->name, __func__); return -EINVAL; } -- cgit v1.2.3 From 89e47f62cf3eea7ad5e3d7d72ea846be37d6e352 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Tue, 6 May 2014 16:55:59 +0800 Subject: ASoC: fsl_esai: Fix incorrect condition within ratio range check for FP The range here from 1 to 16 is confined to FP divider only while the sck_div indicates if the calculation contains PSR and PM dividers. So for the case using PSR and PM since the sck_div is true, the range of ratio would simply become bigger than 16. So this patch fixes the condition here and adds one line comments to make the purpose here clear. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_esai.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index c8e5db1414d7..35c0b9295a80 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -307,7 +307,8 @@ static int fsl_esai_set_bclk(struct snd_soc_dai *dai, bool tx, u32 freq) return -EINVAL; } - if (esai_priv->sck_div[tx] && (ratio > 16 || ratio == 0)) { + /* The ratio should be contented by FP alone if bypassing PM and PSR */ + if (!esai_priv->sck_div[tx] && (ratio > 16 || ratio == 0)) { dev_err(dai->dev, "the ratio is out of range (1 ~ 16)\n"); return -EINVAL; } -- cgit v1.2.3 From 57ebbcafab0ce8cce4493c6a243ecdd7066e6ef1 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Tue, 6 May 2014 16:56:00 +0800 Subject: ASoC: fsl_esai: Only bypass sck_div for EXTAL source ESAI can only output EXTAL clock source directly. But for FSYS clock source, ESAI can not output it without getting through PSR PM dividers. So this patch adds an extra check in the code. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_esai.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 35c0b9295a80..c674fd99158b 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -258,10 +258,16 @@ static int fsl_esai_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, return -EINVAL; } - if (ratio == 1) { + /* Only EXTAL source can be output directly without using PSR and PM */ + if (ratio == 1 && clksrc == esai_priv->extalclk) { /* Bypass all the dividers if not being needed */ ecr |= tx ? ESAI_ECR_ETO : ESAI_ECR_ERO; goto out; + } else if (ratio < 2) { + /* The ratio should be no less than 2 if using other sources */ + dev_err(dai->dev, "failed to derive required HCK%c rate\n", + tx ? 'T' : 'R'); + return -EINVAL; } ret = fsl_esai_divisor_cal(dai, tx, ratio, false, 0); -- cgit v1.2.3 From 4f8210f66e5a200c63770ea9445ff913a28a3be2 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Tue, 6 May 2014 16:56:02 +0800 Subject: ASoC: fsl_esai: Set PCRC and PRRC registers at the end of hw_params() According to Reference Manual -- ESAI Initialization chapter, as the standard procedure of ESAI personal reset, the PCRC and PRRC registers should be remained in its reset value and then configured after T/RCCR and T/RCR configurations's done but before TE/RE's enabling. So this patch moves PCRC and PRRC settings to the end of hw_params(). Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_esai.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index c674fd99158b..496ce2eb2f1f 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -461,12 +461,6 @@ static int fsl_esai_startup(struct snd_pcm_substream *substream, } if (!dai->active) { - /* Reset Port C */ - regmap_update_bits(esai_priv->regmap, REG_ESAI_PRRC, - ESAI_PRRC_PDC_MASK, ESAI_PRRC_PDC(ESAI_GPIO)); - regmap_update_bits(esai_priv->regmap, REG_ESAI_PCRC, - ESAI_PCRC_PC_MASK, ESAI_PCRC_PC(ESAI_GPIO)); - /* Set synchronous mode */ regmap_update_bits(esai_priv->regmap, REG_ESAI_SAICR, ESAI_SAICR_SYNC, esai_priv->synchronous ? @@ -526,6 +520,11 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), mask, val); + /* Remove ESAI personal reset by configuring ESAI_PCRC and ESAI_PRRC */ + regmap_update_bits(esai_priv->regmap, REG_ESAI_PRRC, + ESAI_PRRC_PDC_MASK, ESAI_PRRC_PDC(ESAI_GPIO)); + regmap_update_bits(esai_priv->regmap, REG_ESAI_PCRC, + ESAI_PCRC_PC_MASK, ESAI_PCRC_PC(ESAI_GPIO)); return 0; } -- cgit v1.2.3 From 44330ab516c15dda8a1e660eeaf0003f84e43e3f Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 13 May 2014 13:45:15 +0100 Subject: ASoC: wm8962: Update register CLASS_D_CONTROL_1 to be non-volatile The register CLASS_D_CONTROL_1 is marked as volatile because it contains a bit, DAC_MUTE, which is also mirrored in the ADC_DAC_CONTROL_1 register. This causes problems for the "Speaker Switch" control, which will report an error if the CODEC is suspended because it relies on a volatile register. To resolve this issue mark CLASS_D_CONTROL_1 as non-volatile and manually keep the register cache in sync by updating both bits when changing the mute status. Reported-by: Shawn Guo Signed-off-by: Charles Keepax Tested-by: Shawn Guo Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm8962.c | 15 ++++++++++++--- sound/soc/codecs/wm8962.h | 4 ++++ 2 files changed, 16 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 5522d2566c67..ecd26dd2e442 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -154,6 +154,7 @@ static struct reg_default wm8962_reg[] = { { 40, 0x0000 }, /* R40 - SPKOUTL volume */ { 41, 0x0000 }, /* R41 - SPKOUTR volume */ + { 49, 0x0010 }, /* R49 - Class D Control 1 */ { 51, 0x0003 }, /* R51 - Class D Control 2 */ { 56, 0x0506 }, /* R56 - Clocking 4 */ @@ -795,7 +796,6 @@ static bool wm8962_volatile_register(struct device *dev, unsigned int reg) case WM8962_ALC2: case WM8962_THERMAL_SHUTDOWN_STATUS: case WM8962_ADDITIONAL_CONTROL_4: - case WM8962_CLASS_D_CONTROL_1: case WM8962_DC_SERVO_6: case WM8962_INTERRUPT_STATUS_1: case WM8962_INTERRUPT_STATUS_2: @@ -2929,13 +2929,22 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source, static int wm8962_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - int val; + int val, ret; if (mute) - val = WM8962_DAC_MUTE; + val = WM8962_DAC_MUTE | WM8962_DAC_MUTE_ALT; else val = 0; + /** + * The DAC mute bit is mirrored in two registers, update both to keep + * the register cache consistent. + */ + ret = snd_soc_update_bits(codec, WM8962_CLASS_D_CONTROL_1, + WM8962_DAC_MUTE_ALT, val); + if (ret < 0) + return ret; + return snd_soc_update_bits(codec, WM8962_ADC_DAC_CONTROL_1, WM8962_DAC_MUTE, val); } diff --git a/sound/soc/codecs/wm8962.h b/sound/soc/codecs/wm8962.h index a1a5d5294c19..910aafd09d21 100644 --- a/sound/soc/codecs/wm8962.h +++ b/sound/soc/codecs/wm8962.h @@ -1954,6 +1954,10 @@ #define WM8962_SPKOUTL_ENA_MASK 0x0040 /* SPKOUTL_ENA */ #define WM8962_SPKOUTL_ENA_SHIFT 6 /* SPKOUTL_ENA */ #define WM8962_SPKOUTL_ENA_WIDTH 1 /* SPKOUTL_ENA */ +#define WM8962_DAC_MUTE_ALT 0x0010 /* DAC_MUTE */ +#define WM8962_DAC_MUTE_ALT_MASK 0x0010 /* DAC_MUTE */ +#define WM8962_DAC_MUTE_ALT_SHIFT 4 /* DAC_MUTE */ +#define WM8962_DAC_MUTE_ALT_WIDTH 1 /* DAC_MUTE */ #define WM8962_SPKOUTL_PGA_MUTE 0x0002 /* SPKOUTL_PGA_MUTE */ #define WM8962_SPKOUTL_PGA_MUTE_MASK 0x0002 /* SPKOUTL_PGA_MUTE */ #define WM8962_SPKOUTL_PGA_MUTE_SHIFT 1 /* SPKOUTL_PGA_MUTE */ -- cgit v1.2.3