From c593b520cf70b0672680da04cc1e8c5f93bd739d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 27 Oct 2010 20:11:17 -0700 Subject: ASoC: Check return value of struct_strtoul() in pmdown_time_set() strict_strtoul() has just been made must check so do so. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 70d9a7394b2..805343fe903 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -165,8 +165,11 @@ static ssize_t pmdown_time_set(struct device *dev, { struct snd_soc_pcm_runtime *rtd = container_of(dev, struct snd_soc_pcm_runtime, dev); + int ret; - strict_strtol(buf, 10, &rtd->pmdown_time); + ret = strict_strtol(buf, 10, &rtd->pmdown_time); + if (ret) + return ret; return count; } -- cgit v1.2.3-70-g09d2 From 911a0f0bfc01750590e8ac6e7f9f4921f470b0d1 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 26 Oct 2010 11:45:59 +0300 Subject: ASoC: tlv320dac33: Error handling for broken chip Correct/Implement handling of broken chip. Fail the soc_prope if the communication with the chip fails (can not read chip ID). Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/codecs/tlv320dac33.c | 26 +++++++++++++++++++------- 1 file changed, 19 insertions(+), 7 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index d251ff54a2d..fed14582b49 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -200,7 +200,7 @@ static int dac33_read(struct snd_soc_codec *codec, unsigned int reg, u8 *value) { struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); - int val; + int val, ret = 0; *value = reg & 0xff; @@ -210,6 +210,7 @@ static int dac33_read(struct snd_soc_codec *codec, unsigned int reg, if (val < 0) { dev_err(codec->dev, "Read failed (%d)\n", val); value[0] = dac33_read_reg_cache(codec, reg); + ret = val; } else { value[0] = val; dac33_write_reg_cache(codec, reg, val); @@ -218,7 +219,7 @@ static int dac33_read(struct snd_soc_codec *codec, unsigned int reg, value[0] = dac33_read_reg_cache(codec, reg); } - return 0; + return ret; } static int dac33_write(struct snd_soc_codec *codec, unsigned int reg, @@ -329,13 +330,18 @@ static void dac33_init_chip(struct snd_soc_codec *codec) dac33_read_reg_cache(codec, DAC33_LINER_TO_RLO_VOL)); } -static inline void dac33_read_id(struct snd_soc_codec *codec) +static inline int dac33_read_id(struct snd_soc_codec *codec) { + int i, ret = 0; u8 reg; - dac33_read(codec, DAC33_DEVICE_ID_MSB, ®); - dac33_read(codec, DAC33_DEVICE_ID_LSB, ®); - dac33_read(codec, DAC33_DEVICE_REV_ID, ®); + for (i = 0; i < 3; i++) { + ret = dac33_read(codec, DAC33_DEVICE_ID_MSB + i, ®); + if (ret < 0) + break; + } + + return ret; } static inline void dac33_soft_power(struct snd_soc_codec *codec, int power) @@ -1414,9 +1420,15 @@ static int dac33_soc_probe(struct snd_soc_codec *codec) dev_err(codec->dev, "Failed to power up codec: %d\n", ret); goto err_power; } - dac33_read_id(codec); + ret = dac33_read_id(codec); dac33_hard_power(codec, 0); + if (ret < 0) { + dev_err(codec->dev, "Failed to read chip ID: %d\n", ret); + ret = -ENODEV; + goto err_power; + } + /* Check if the IRQ number is valid and request it */ if (dac33->irq >= 0) { ret = request_irq(dac33->irq, dac33_interrupt_handler, -- cgit v1.2.3-70-g09d2 From d54e1f4fdf4cf9754b7220ae4cb66dcae0fc1702 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 29 Oct 2010 14:07:25 +0300 Subject: ASoC: tlv320dac33: Limit the US_TO_SAMPLES macro Limit the time window to maximum 1s in the macro. The driver deals with much shorter times (<200ms). This will fix a rare division by zero bug in Mode1. This could happen, when the work is not executed in time (within mode1_latency) after the interrupt. In this case the DAC33 will not receive the needed nSample command in time, and enters to an unknown state, and won't recover. In such event the time window will increase, and eventually going to be bigger than 1s, resulting devision by zero. Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/codecs/tlv320dac33.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index fed14582b49..c47c20d21ea 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -58,7 +58,7 @@ (1000000000 / ((rate * 1000) / samples)) #define US_TO_SAMPLES(rate, us) \ - (rate / (1000000 / us)) + (rate / (1000000 / (us < 1000000 ? us : 1000000))) #define UTHR_FROM_PERIOD_SIZE(samples, playrate, burstrate) \ ((samples * 5000) / ((burstrate * 5000) / (burstrate - playrate))) -- cgit v1.2.3-70-g09d2 From 1bc13b2e3518ff7856924d7c2bdf06196f605260 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 29 Oct 2010 09:49:37 +0300 Subject: ASoC: tlv320dac33: Mode1 FIFO auto configuration fix Do not allow invalid (too big) nSample value, when FIFO Mode1 and automatic fifo configuration has been selected. Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/codecs/tlv320dac33.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index c47c20d21ea..c5ab8c80577 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1082,6 +1082,9 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream) /* Number of samples under i2c latency */ dac33->alarm_threshold = US_TO_SAMPLES(rate, dac33->mode1_latency); + nsample_limit = DAC33_BUFFER_SIZE_SAMPLES - + dac33->alarm_threshold; + if (dac33->auto_fifo_config) { if (period_size <= dac33->alarm_threshold) /* @@ -1092,6 +1095,8 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream) ((dac33->alarm_threshold / period_size) + (dac33->alarm_threshold % period_size ? 1 : 0)); + else if (period_size > nsample_limit) + dac33->nsample = nsample_limit; else dac33->nsample = period_size; } else { @@ -1103,8 +1108,7 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream) */ dac33->nsample_max = substream->runtime->buffer_size - period_size; - nsample_limit = DAC33_BUFFER_SIZE_SAMPLES - - dac33->alarm_threshold; + if (dac33->nsample_max > nsample_limit) dac33->nsample_max = nsample_limit; -- cgit v1.2.3-70-g09d2 From 63f7526f26f0a9291ac3f7a986aa18ebfb61ec19 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Thu, 28 Oct 2010 14:05:40 +0300 Subject: ASoC: tpa6130a2: Fix unbalanced regulator disables This driver has unbalanced regulator_disable when doing module loading and unloading. This is because tpa6130a2_probe followed by tpa6130a2_remove calls twice tpa6130a2_power(0). Fix this by implementing a state checking in tpa6130a2_power. Signed-off-by: Jarkko Nikula Cc: Peter Ujfalusi Acked-by: Mark Brown Acked-by: Peter Ujfalusi Signed-off-by: Liam Girdwood --- sound/soc/codecs/tpa6130a2.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 329acc1a207..83b5631b13a 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -125,7 +125,7 @@ static int tpa6130a2_power(int power) data = i2c_get_clientdata(tpa6130a2_client); mutex_lock(&data->mutex); - if (power) { + if (power && !data->power_state) { /* Power on */ if (data->power_gpio >= 0) gpio_set_value(data->power_gpio, 1); @@ -153,7 +153,7 @@ static int tpa6130a2_power(int power) val = tpa6130a2_read(TPA6130A2_REG_CONTROL); val &= ~TPA6130A2_SWS; tpa6130a2_i2c_write(TPA6130A2_REG_CONTROL, val); - } else { + } else if (!power && data->power_state) { /* set SWS */ val = tpa6130a2_read(TPA6130A2_REG_CONTROL); val |= TPA6130A2_SWS; -- cgit v1.2.3-70-g09d2 From 6d212d8e86fb4221bd91b9266b7567ee2b83bd01 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 29 Oct 2010 15:41:17 -0700 Subject: ASoC: Remove volatility from WM8900 POWER1 register Not all bits can be read back from POWER1 so avoid corruption when using a read/modify/write cycle by marking it non-volatile - the only thing we read back from it is the chip revision which has diagnostic value only. We can re-add later but that's a more invasive change than is suitable for a bugfix. Signed-off-by: Mark Brown Acked-by: Liam Girdwood Cc: stable@kernel.org --- sound/soc/codecs/wm8900.c | 6 ------ 1 file changed, 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index b4f11724a63..aca4b1ea10b 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -186,7 +186,6 @@ static int wm8900_volatile_register(unsigned int reg) { switch (reg) { case WM8900_REG_ID: - case WM8900_REG_POWER1: return 1; default: return 0; @@ -1200,11 +1199,6 @@ static int wm8900_probe(struct snd_soc_codec *codec) return -ENODEV; } - /* Read back from the chip */ - reg = snd_soc_read(codec, WM8900_REG_POWER1); - reg = (reg >> 12) & 0xf; - dev_info(codec->dev, "WM8900 revision %d\n", reg); - wm8900_reset(codec); /* Turn the chip on */ -- cgit v1.2.3-70-g09d2 From 703dde6219346bc3b7d41d4fa2c36846d728e52c Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 29 Oct 2010 16:47:44 +0300 Subject: ASoC: Fix SND_SOC_ALL_CODECS typo for jz4740 Include jz4740.c to SND_SOC_ALL_CODECS when the dependencies are met. Signed-off-by: Jarkko Nikula Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 94a9d06b902..02a9751bf14 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -26,7 +26,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS42L51 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_DA7210 if I2C - select SND_SOC_JZ4740 if SOC_JZ4740 + select SND_SOC_JZ4740_CODEC if SOC_JZ4740 select SND_SOC_MAX98088 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 -- cgit v1.2.3-70-g09d2 From 76a6106f124e375df0ea6ba6bcf204b8caff786a Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 29 Oct 2010 16:47:45 +0300 Subject: ASoC: Include cx20442 to SND_SOC_ALL_CODECS Signed-off-by: Jarkko Nikula Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 02a9751bf14..3b5690d28b8 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -25,6 +25,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS42L51 if I2C select SND_SOC_CS4270 if I2C + select SND_SOC_CX20442 select SND_SOC_DA7210 if I2C select SND_SOC_JZ4740_CODEC if SOC_JZ4740 select SND_SOC_MAX98088 if I2C -- cgit v1.2.3-70-g09d2 From 5a0b07433ddd808ecbb5f4287b61be6fa7af1b57 Mon Sep 17 00:00:00 2001 From: Joe Perches Date: Sat, 30 Oct 2010 14:08:56 -0700 Subject: ASoC: Update WARN uses in wm_hubs Add missing newlines. Signed-off-by: Joe Perches Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm_hubs.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 2cb81538cd9..19ca782ac97 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -123,7 +123,7 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) reg_r = reg & WM8993_DCS_DAC_WR_VAL_0_MASK; break; default: - WARN(1, "Unknown DCS readback method"); + WARN(1, "Unknown DCS readback method\n"); break; } -- cgit v1.2.3-70-g09d2 From cb9906229595941d632fc4022b05da4f9533856a Mon Sep 17 00:00:00 2001 From: Eric Miao Date: Tue, 2 Nov 2010 05:10:07 +0800 Subject: ASoC: fix the building issue of missing codec field in 'struct snd_soc_card' Signed-off-by: Mark Brown --- sound/soc/pxa/tosa.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index a3bfb2e8b70..73d0edd8ded 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -79,7 +79,7 @@ static void tosa_ext_control(struct snd_soc_codec *codec) static int tosa_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->card->codec; + struct snd_soc_codec *codec = rtd->codec; /* check the jack status at stream startup */ tosa_ext_control(codec); -- cgit v1.2.3-70-g09d2 From c46e0079cec40b49fbdb86a088cfd50b250fef47 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 3 Nov 2010 15:04:45 +0800 Subject: ASoC: Fix snd_soc_register_dais error handling kzalloc for dai may fail at any iteration of the for loop, thus properly unregister already registered DAIs before return error. The error handling code in snd_soc_register_dais() already ensure all the DAIs are unregistered before return error, we can remove the error handling code to unregister DAIs in snd_soc_register_codec(). Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 614a8b30d87..441285ade02 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3043,8 +3043,10 @@ int snd_soc_register_dais(struct device *dev, for (i = 0; i < count; i++) { dai = kzalloc(sizeof(struct snd_soc_dai), GFP_KERNEL); - if (dai == NULL) - return -ENOMEM; + if (dai == NULL) { + ret = -ENOMEM; + goto err; + } /* create DAI component name */ dai->name = fmt_multiple_name(dev, &dai_drv[i]); @@ -3263,9 +3265,6 @@ int snd_soc_register_codec(struct device *dev, return 0; error: - for (i--; i >= 0; i--) - snd_soc_unregister_dai(dev); - if (codec->reg_cache) kfree(codec->reg_cache); kfree(codec->name); -- cgit v1.2.3-70-g09d2 From 233538501f707b0176f09af7039fec1e3fcac6e7 Mon Sep 17 00:00:00 2001 From: Janusz Krzysztofik Date: Tue, 2 Nov 2010 15:50:32 +0100 Subject: ASoC: OMAP: fix OMAP1 compilation problem In the new code introduced with commit cf4c87abe238ec17cd0255b4e21abd949d7f811e, "OMAP: McBSP: implement McBSP CLKR and FSR signal muxing via mach-omap2/mcbsp.c", the way omap1 build is supposed to bypass omap2 specific functionality doesn't optimize out all omap2 specific stuff. This breaks linking phase for omap1 machines, giving "undefined reference to `omap2_mcbsp1_mux_clkr_src'" and "undefined reference to `omap2_mcbsp1_mux_fsr_src'" errors. Fix it. Created and tested against linux-2.6.37-rc1. Signed-off-by: Janusz Krzysztofik Acked-by: Mark Brown Acked-by: Paul Walmsley Acked-by: Jarkko Nikula Signed-off-by: Liam Girdwood --- sound/soc/omap/omap-mcbsp.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index d211c9fa5a9..7e84f24b9a8 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -644,15 +644,23 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, case OMAP_MCBSP_CLKR_SRC_CLKR: + if (cpu_class_is_omap1()) + break; omap2_mcbsp1_mux_clkr_src(CLKR_SRC_CLKR); break; case OMAP_MCBSP_CLKR_SRC_CLKX: + if (cpu_class_is_omap1()) + break; omap2_mcbsp1_mux_clkr_src(CLKR_SRC_CLKX); break; case OMAP_MCBSP_FSR_SRC_FSR: + if (cpu_class_is_omap1()) + break; omap2_mcbsp1_mux_fsr_src(FSR_SRC_FSR); break; case OMAP_MCBSP_FSR_SRC_FSX: + if (cpu_class_is_omap1()) + break; omap2_mcbsp1_mux_fsr_src(FSR_SRC_FSX); break; default: -- cgit v1.2.3-70-g09d2 From 75e3f3137cb570661c2ad3035a139dda671fbb63 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Wed, 3 Nov 2010 16:39:00 +0200 Subject: ASoC: tpa6130a2: Get rid of compile warning from tpa6130a2_power MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Patch "ASoC: tpa6130a2: Fix unbalanced regulator disables" introduced a compiler warning "‘ret’ may be used uninitialized in this function". Initialize ret to zero to get rid of it and making sure that the function does not return any random error code when the code is falling through. Signed-off-by: Jarkko Nikula Signed-off-by: Takashi Iwai --- sound/soc/codecs/tpa6130a2.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 83b5631b13a..ee4fb201de6 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -119,7 +119,7 @@ static int tpa6130a2_power(int power) { struct tpa6130a2_data *data; u8 val; - int ret; + int ret = 0; BUG_ON(tpa6130a2_client == NULL); data = i2c_get_clientdata(tpa6130a2_client); -- cgit v1.2.3-70-g09d2 From 74a557e27ff86a5a1f8d5f24c178c70b98367b12 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 3 Nov 2010 09:37:06 -0400 Subject: ASoC: Check return value of strict_strtoul() in WM8962 strict_strtoul() has been made __must_check so do so. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8962.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 894d0cd3aa9..e8092745a20 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3500,8 +3500,11 @@ static ssize_t wm8962_beep_set(struct device *dev, { struct wm8962_priv *wm8962 = dev_get_drvdata(dev); long int time; + int ret; - strict_strtol(buf, 10, &time); + ret = strict_strtol(buf, 10, &time); + if (ret != 0) + return ret; input_event(wm8962->beep, EV_SND, SND_TONE, time); -- cgit v1.2.3-70-g09d2 From add330ec29cb00b26cf45ffb4773bb9094a48368 Mon Sep 17 00:00:00 2001 From: Sascha Hauer Date: Thu, 4 Nov 2010 17:05:40 +0100 Subject: ASoC i.MX eukrea tlv320: Fix for multicomponent Signed-off-by: Sascha Hauer Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/imx/eukrea-tlv320.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/imx/eukrea-tlv320.c b/sound/soc/imx/eukrea-tlv320.c index b59675257ce..dd4fffdbd17 100644 --- a/sound/soc/imx/eukrea-tlv320.c +++ b/sound/soc/imx/eukrea-tlv320.c @@ -34,8 +34,8 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | @@ -79,10 +79,10 @@ static struct snd_soc_ops eukrea_tlv320_snd_ops = { static struct snd_soc_dai_link eukrea_tlv320_dai = { .name = "tlv320aic23", .stream_name = "TLV320AIC23", - .codec_dai = "tlv320aic23-hifi", + .codec_dai_name = "tlv320aic23-hifi", .platform_name = "imx-pcm-audio.0", .codec_name = "tlv320aic23-codec.0-001a", - .cpu_dai = "imx-ssi.0", + .cpu_dai_name = "imx-ssi.0", .ops = &eukrea_tlv320_snd_ops, }; -- cgit v1.2.3-70-g09d2 From bf0199b7a5085e8d1908d2b0a9c530ed8d142fb8 Mon Sep 17 00:00:00 2001 From: Sascha Hauer Date: Thu, 4 Nov 2010 17:05:41 +0100 Subject: ASoC i.MX phycore ac97: remove unnecessary includes Signed-off-by: Sascha Hauer Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/imx/phycore-ac97.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/imx/phycore-ac97.c b/sound/soc/imx/phycore-ac97.c index 6a65dd70551..cf46a17d692 100644 --- a/sound/soc/imx/phycore-ac97.c +++ b/sound/soc/imx/phycore-ac97.c @@ -20,9 +20,6 @@ #include #include -#include "../codecs/wm9712.h" -#include "imx-ssi.h" - static struct snd_soc_card imx_phycore; static struct snd_soc_ops imx_phycore_hifi_ops = { -- cgit v1.2.3-70-g09d2 From f562be51fe9021c913e661c46681cb5bae70f369 Mon Sep 17 00:00:00 2001 From: Sascha Hauer Date: Thu, 4 Nov 2010 17:05:42 +0100 Subject: ASoC i.MX: register dma audio device We have two different transfer methods on i.MX: FIQ and DMA. Since the merge of the ASoC multicomponent support the DMA device is lost. Add it again. Also, imx_ssi_dai_probe has to be called for !AC97 aswell. Signed-off-by: Sascha Hauer Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/imx/imx-ssi.c | 44 +++++++++++++++++++++++++++++--------------- sound/soc/imx/imx-ssi.h | 1 + 2 files changed, 30 insertions(+), 15 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index d4bd345b0a8..d2d98c75ee8 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -439,7 +439,22 @@ void imx_pcm_free(struct snd_pcm *pcm) } EXPORT_SYMBOL_GPL(imx_pcm_free); +static int imx_ssi_dai_probe(struct snd_soc_dai *dai) +{ + struct imx_ssi *ssi = dev_get_drvdata(dai->dev); + uint32_t val; + + snd_soc_dai_set_drvdata(dai, ssi); + + val = SSI_SFCSR_TFWM0(ssi->dma_params_tx.burstsize) | + SSI_SFCSR_RFWM0(ssi->dma_params_rx.burstsize); + writel(val, ssi->base + SSI_SFCSR); + + return 0; +} + static struct snd_soc_dai_driver imx_ssi_dai = { + .probe = imx_ssi_dai_probe, .playback = { .channels_min = 2, .channels_max = 2, @@ -455,20 +470,6 @@ static struct snd_soc_dai_driver imx_ssi_dai = { .ops = &imx_ssi_pcm_dai_ops, }; -static int imx_ssi_dai_probe(struct snd_soc_dai *dai) -{ - struct imx_ssi *ssi = dev_get_drvdata(dai->dev); - uint32_t val; - - snd_soc_dai_set_drvdata(dai, ssi); - - val = SSI_SFCSR_TFWM0(ssi->dma_params_tx.burstsize) | - SSI_SFCSR_RFWM0(ssi->dma_params_rx.burstsize); - writel(val, ssi->base + SSI_SFCSR); - - return 0; -} - static struct snd_soc_dai_driver imx_ac97_dai = { .probe = imx_ssi_dai_probe, .ac97_control = 1, @@ -677,7 +678,17 @@ static int imx_ssi_probe(struct platform_device *pdev) goto failed_register; } - ssi->soc_platform_pdev = platform_device_alloc("imx-fiq-pcm-audio", pdev->id); + ssi->soc_platform_pdev_fiq = platform_device_alloc("imx-fiq-pcm-audio", pdev->id); + if (!ssi->soc_platform_pdev_fiq) + goto failed_pdev_fiq_alloc; + platform_set_drvdata(ssi->soc_platform_pdev_fiq, ssi); + ret = platform_device_add(ssi->soc_platform_pdev_fiq); + if (ret) { + dev_err(&pdev->dev, "failed to add platform device\n"); + goto failed_pdev_fiq_add; + } + + ssi->soc_platform_pdev = platform_device_alloc("imx-pcm-audio", pdev->id); if (!ssi->soc_platform_pdev) goto failed_pdev_alloc; platform_set_drvdata(ssi->soc_platform_pdev, ssi); @@ -692,6 +703,9 @@ static int imx_ssi_probe(struct platform_device *pdev) failed_pdev_add: platform_device_put(ssi->soc_platform_pdev); failed_pdev_alloc: +failed_pdev_fiq_add: + platform_device_put(ssi->soc_platform_pdev_fiq); +failed_pdev_fiq_alloc: snd_soc_unregister_dai(&pdev->dev); failed_register: failed_ac97: diff --git a/sound/soc/imx/imx-ssi.h b/sound/soc/imx/imx-ssi.h index 53b780d9b2b..4fc17da1186 100644 --- a/sound/soc/imx/imx-ssi.h +++ b/sound/soc/imx/imx-ssi.h @@ -212,6 +212,7 @@ struct imx_ssi { int enabled; struct platform_device *soc_platform_pdev; + struct platform_device *soc_platform_pdev_fiq; }; struct snd_soc_platform *imx_ssi_fiq_init(struct platform_device *pdev, -- cgit v1.2.3-70-g09d2 From bf974a0d77a318a733a47c18a47fa6ff8960c361 Mon Sep 17 00:00:00 2001 From: Sascha Hauer Date: Thu, 4 Nov 2010 17:05:43 +0100 Subject: ASoC i.MX: switch to new DMA api Signed-off-by: Sascha Hauer Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/imx/imx-pcm-dma-mx2.c | 221 ++++++++++++++++++---------------------- sound/soc/imx/imx-ssi.h | 3 + 2 files changed, 101 insertions(+), 123 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c index fd493ee1428..671ef8dd524 100644 --- a/sound/soc/imx/imx-pcm-dma-mx2.c +++ b/sound/soc/imx/imx-pcm-dma-mx2.c @@ -20,6 +20,7 @@ #include #include #include +#include #include #include @@ -27,165 +28,146 @@ #include #include -#include +#include #include "imx-ssi.h" struct imx_pcm_runtime_data { - int sg_count; - struct scatterlist *sg_list; - int period; + int period_bytes; int periods; - unsigned long dma_addr; int dma; - struct snd_pcm_substream *substream; unsigned long offset; unsigned long size; - unsigned long period_cnt; void *buf; int period_time; + struct dma_async_tx_descriptor *desc; + struct dma_chan *dma_chan; + struct imx_dma_data dma_data; }; -/* Called by the DMA framework when a period has elapsed */ -static void imx_ssi_dma_progression(int channel, void *data, - struct scatterlist *sg) +static void audio_dma_irq(void *data) { - struct snd_pcm_substream *substream = data; + struct snd_pcm_substream *substream = (struct snd_pcm_substream *)data; struct snd_pcm_runtime *runtime = substream->runtime; struct imx_pcm_runtime_data *iprtd = runtime->private_data; - if (!sg) - return; - - runtime = iprtd->substream->runtime; + iprtd->offset += iprtd->period_bytes; + iprtd->offset %= iprtd->period_bytes * iprtd->periods; - iprtd->offset = sg->dma_address - runtime->dma_addr; - - snd_pcm_period_elapsed(iprtd->substream); + snd_pcm_period_elapsed(substream); } -static void imx_ssi_dma_callback(int channel, void *data) +static bool filter(struct dma_chan *chan, void *param) { - pr_err("%s shouldn't be called\n", __func__); -} + struct imx_pcm_runtime_data *iprtd = param; -static void snd_imx_dma_err_callback(int channel, void *data, int err) -{ - struct snd_pcm_substream *substream = data; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct imx_pcm_dma_params *dma_params = - snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); - struct snd_pcm_runtime *runtime = substream->runtime; - struct imx_pcm_runtime_data *iprtd = runtime->private_data; - int ret; + if (!imx_dma_is_general_purpose(chan)) + return false; - pr_err("DMA timeout on channel %d -%s%s%s%s\n", - channel, - err & IMX_DMA_ERR_BURST ? " burst" : "", - err & IMX_DMA_ERR_REQUEST ? " request" : "", - err & IMX_DMA_ERR_TRANSFER ? " transfer" : "", - err & IMX_DMA_ERR_BUFFER ? " buffer" : ""); + chan->private = &iprtd->dma_data; - imx_dma_disable(iprtd->dma); - ret = imx_dma_setup_sg(iprtd->dma, iprtd->sg_list, iprtd->sg_count, - IMX_DMA_LENGTH_LOOP, dma_params->dma_addr, - substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? - DMA_MODE_WRITE : DMA_MODE_READ); - if (!ret) - imx_dma_enable(iprtd->dma); + return true; } -static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream) +static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct imx_pcm_dma_params *dma_params; struct snd_pcm_runtime *runtime = substream->runtime; struct imx_pcm_runtime_data *iprtd = runtime->private_data; + struct dma_slave_config slave_config; + dma_cap_mask_t mask; + enum dma_slave_buswidth buswidth; int ret; dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - iprtd->dma = imx_dma_request_by_prio(DRV_NAME, DMA_PRIO_HIGH); - if (iprtd->dma < 0) { - pr_err("Failed to claim the audio DMA\n"); - return -ENODEV; - } + iprtd->dma_data.peripheral_type = IMX_DMATYPE_SSI; + iprtd->dma_data.priority = DMA_PRIO_HIGH; + iprtd->dma_data.dma_request = dma_params->dma; - ret = imx_dma_setup_handlers(iprtd->dma, - imx_ssi_dma_callback, - snd_imx_dma_err_callback, substream); - if (ret) - goto out; + /* Try to grab a DMA channel */ + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); + iprtd->dma_chan = dma_request_channel(mask, filter, iprtd); + if (!iprtd->dma_chan) + return -EINVAL; - ret = imx_dma_setup_progression_handler(iprtd->dma, - imx_ssi_dma_progression); - if (ret) { - pr_err("Failed to setup the DMA handler\n"); - goto out; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + buswidth = DMA_SLAVE_BUSWIDTH_2_BYTES; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + case SNDRV_PCM_FORMAT_S24_LE: + buswidth = DMA_SLAVE_BUSWIDTH_4_BYTES; + break; + default: + return 0; } - ret = imx_dma_config_channel(iprtd->dma, - IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, - IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, - dma_params->dma, 1); - if (ret < 0) { - pr_err("Cannot configure DMA channel: %d\n", ret); - goto out; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + slave_config.direction = DMA_TO_DEVICE; + slave_config.dst_addr = dma_params->dma_addr; + slave_config.dst_addr_width = buswidth; + slave_config.dst_maxburst = dma_params->burstsize; + } else { + slave_config.direction = DMA_FROM_DEVICE; + slave_config.src_addr = dma_params->dma_addr; + slave_config.src_addr_width = buswidth; + slave_config.src_maxburst = dma_params->burstsize; } - imx_dma_config_burstlen(iprtd->dma, dma_params->burstsize * 2); + ret = dmaengine_slave_config(iprtd->dma_chan, &slave_config); + if (ret) + return ret; return 0; -out: - imx_dma_free(iprtd->dma); - return ret; } static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { + struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_pcm_runtime *runtime = substream->runtime; struct imx_pcm_runtime_data *iprtd = runtime->private_data; - int i; unsigned long dma_addr; + struct dma_chan *chan; + struct imx_pcm_dma_params *dma_params; + int ret; - imx_ssi_dma_alloc(substream); + dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + ret = imx_ssi_dma_alloc(substream, params); + if (ret) + return ret; + chan = iprtd->dma_chan; iprtd->size = params_buffer_bytes(params); iprtd->periods = params_periods(params); - iprtd->period = params_period_bytes(params); + iprtd->period_bytes = params_period_bytes(params); iprtd->offset = 0; iprtd->period_time = HZ / (params_rate(params) / params_period_size(params)); snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - if (iprtd->sg_count != iprtd->periods) { - kfree(iprtd->sg_list); - - iprtd->sg_list = kcalloc(iprtd->periods + 1, - sizeof(struct scatterlist), GFP_KERNEL); - if (!iprtd->sg_list) - return -ENOMEM; - iprtd->sg_count = iprtd->periods + 1; - } - - sg_init_table(iprtd->sg_list, iprtd->sg_count); dma_addr = runtime->dma_addr; - for (i = 0; i < iprtd->periods; i++) { - iprtd->sg_list[i].page_link = 0; - iprtd->sg_list[i].offset = 0; - iprtd->sg_list[i].dma_address = dma_addr; - iprtd->sg_list[i].length = iprtd->period; - dma_addr += iprtd->period; + iprtd->buf = (unsigned int *)substream->dma_buffer.area; + + iprtd->desc = chan->device->device_prep_dma_cyclic(chan, dma_addr, + iprtd->period_bytes * iprtd->periods, + iprtd->period_bytes, + substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + DMA_TO_DEVICE : DMA_FROM_DEVICE); + if (!iprtd->desc) { + dev_err(&chan->dev->device, "cannot prepare slave dma\n"); + return -EINVAL; } - /* close the loop */ - iprtd->sg_list[iprtd->sg_count - 1].offset = 0; - iprtd->sg_list[iprtd->sg_count - 1].length = 0; - iprtd->sg_list[iprtd->sg_count - 1].page_link = - ((unsigned long) iprtd->sg_list | 0x01) & ~0x02; + iprtd->desc->callback = audio_dma_irq; + iprtd->desc->callback_param = substream; + return 0; } @@ -194,41 +176,21 @@ static int snd_imx_pcm_hw_free(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct imx_pcm_runtime_data *iprtd = runtime->private_data; - if (iprtd->dma >= 0) { - imx_dma_free(iprtd->dma); - iprtd->dma = -EINVAL; + if (iprtd->dma_chan) { + dma_release_channel(iprtd->dma_chan); + iprtd->dma_chan = NULL; } - kfree(iprtd->sg_list); - iprtd->sg_list = NULL; - return 0; } static int snd_imx_pcm_prepare(struct snd_pcm_substream *substream) { - struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct imx_pcm_dma_params *dma_params; - struct imx_pcm_runtime_data *iprtd = runtime->private_data; - int err; dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - iprtd->substream = substream; - iprtd->buf = (unsigned int *)substream->dma_buffer.area; - iprtd->period_cnt = 0; - - pr_debug("%s: buf: %p period: %d periods: %d\n", - __func__, iprtd->buf, iprtd->period, iprtd->periods); - - err = imx_dma_setup_sg(iprtd->dma, iprtd->sg_list, iprtd->sg_count, - IMX_DMA_LENGTH_LOOP, dma_params->dma_addr, - substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? - DMA_MODE_WRITE : DMA_MODE_READ); - if (err) - return err; - return 0; } @@ -241,14 +203,14 @@ static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - imx_dma_enable(iprtd->dma); + dmaengine_submit(iprtd->desc); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - imx_dma_disable(iprtd->dma); + dmaengine_terminate_all(iprtd->dma_chan); break; default: @@ -263,6 +225,9 @@ static snd_pcm_uframes_t snd_imx_pcm_pointer(struct snd_pcm_substream *substream struct snd_pcm_runtime *runtime = substream->runtime; struct imx_pcm_runtime_data *iprtd = runtime->private_data; + pr_debug("%s: %ld %ld\n", __func__, iprtd->offset, + bytes_to_frames(substream->runtime, iprtd->offset)); + return bytes_to_frames(substream->runtime, iprtd->offset); } @@ -279,7 +244,7 @@ static struct snd_pcm_hardware snd_imx_hardware = { .channels_max = 2, .buffer_bytes_max = IMX_SSI_DMABUF_SIZE, .period_bytes_min = 128, - .period_bytes_max = 16 * 1024, + .period_bytes_max = 65535, /* Limited by SDMA engine */ .periods_min = 2, .periods_max = 255, .fifo_size = 0, @@ -304,11 +269,23 @@ static int snd_imx_open(struct snd_pcm_substream *substream) } snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware); + + return 0; +} + +static int snd_imx_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + + kfree(iprtd); + return 0; } static struct snd_pcm_ops imx_pcm_ops = { .open = snd_imx_open, + .close = snd_imx_close, .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_imx_pcm_hw_params, .hw_free = snd_imx_pcm_hw_free, @@ -340,7 +317,6 @@ static struct platform_driver imx_pcm_driver = { .name = "imx-pcm-audio", .owner = THIS_MODULE, }, - .probe = imx_soc_platform_probe, .remove = __devexit_p(imx_soc_platform_remove), }; @@ -356,4 +332,3 @@ static void __exit snd_imx_pcm_exit(void) platform_driver_unregister(&imx_pcm_driver); } module_exit(snd_imx_pcm_exit); - diff --git a/sound/soc/imx/imx-ssi.h b/sound/soc/imx/imx-ssi.h index 4fc17da1186..a4406a13489 100644 --- a/sound/soc/imx/imx-ssi.h +++ b/sound/soc/imx/imx-ssi.h @@ -185,6 +185,9 @@ #define DRV_NAME "imx-ssi" +#include +#include + struct imx_pcm_dma_params { int dma; unsigned long dma_addr; -- cgit v1.2.3-70-g09d2 From 6424dca23e6b5a2f7a19a69cf7c0990b11717b00 Mon Sep 17 00:00:00 2001 From: Sascha Hauer Date: Thu, 4 Nov 2010 17:05:44 +0100 Subject: phycore-ac97: add ac97 to cardname We have different codecs on the pcm038 (ac97 wm9712 and mc13783). To make alsactl restore work correctly these should have different names. Signed-off-by: Sascha Hauer Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/imx/phycore-ac97.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/imx/phycore-ac97.c b/sound/soc/imx/phycore-ac97.c index cf46a17d692..39f23734781 100644 --- a/sound/soc/imx/phycore-ac97.c +++ b/sound/soc/imx/phycore-ac97.c @@ -38,7 +38,7 @@ static struct snd_soc_dai_link imx_phycore_dai_ac97[] = { }; static struct snd_soc_card imx_phycore = { - .name = "PhyCORE-audio", + .name = "PhyCORE-ac97-audio", .dai_link = imx_phycore_dai_ac97, .num_links = ARRAY_SIZE(imx_phycore_dai_ac97), }; -- cgit v1.2.3-70-g09d2 From 71a295602ed967fa22d96d57a2e38bb86de24db7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 5 Nov 2010 13:50:48 -0400 Subject: ASoC: Lock the CODEC in PXA external jack controls When doing anything with the system, especially DAPM, we need to hold the CODEC mutex. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/pxa/corgi.c | 5 +++++ sound/soc/pxa/magician.c | 4 ++++ sound/soc/pxa/poodle.c | 5 +++++ sound/soc/pxa/spitz.c | 5 +++++ sound/soc/pxa/tosa.c | 5 +++++ 5 files changed, 24 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 97e9423615c..f451acd4935 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -100,8 +100,13 @@ static int corgi_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; + mutex_lock(&codec->mutex); + /* check the jack status at stream startup */ corgi_ext_control(codec); + + mutex_unlock(&codec->mutex); + return 0; } diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index b8207ced407..5ef0526924b 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -72,9 +72,13 @@ static int magician_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; + mutex_lock(&codec->mutex); + /* check the jack status at stream startup */ magician_ext_control(codec); + mutex_unlock(&codec->mutex); + return 0; } diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index af84ee9c5e1..84edd0385a2 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -77,8 +77,13 @@ static int poodle_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; + mutex_lock(&codec->mutex); + /* check the jack status at stream startup */ poodle_ext_control(codec); + + mutex_unlock(&codec->mutex); + return 0; } diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index f470f360f4d..0b30d7de24e 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -108,8 +108,13 @@ static int spitz_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; + mutex_lock(&codec->mutex); + /* check the jack status at stream startup */ spitz_ext_control(codec); + + mutex_unlock(&codec->mutex); + return 0; } diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index 73d0edd8ded..7b983f93545 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -81,8 +81,13 @@ static int tosa_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; + mutex_lock(&codec->mutex); + /* check the jack status at stream startup */ tosa_ext_control(codec); + + mutex_unlock(&codec->mutex); + return 0; } -- cgit v1.2.3-70-g09d2 From 197ebd4053c42351e3737d83aebb33ed97ed2dd8 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Fri, 5 Nov 2010 10:36:24 +0000 Subject: ASoC: WM8776: Removed unneeded struct member The member reg_cache is not used at all and therefore it should be removed. This member was usually needed for older versions of ASoC that did not handle caching automatically and had to be done in the driver itself. Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm8776.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index 04182c464e3..0132a27140a 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -34,7 +34,6 @@ /* codec private data */ struct wm8776_priv { enum snd_soc_control_type control_type; - u16 reg_cache[WM8776_CACHEREGNUM]; int sysclk[2]; }; -- cgit v1.2.3-70-g09d2 From 1ebd0061ededeb8b495360a772d0b885dd3e036e Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 8 Nov 2010 13:24:58 +0800 Subject: ASoC: Return proper error if snd_soc_register_dais fails in psc_i2s_of_probe Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/fsl/mpc5200_psc_i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index 74ffed41340..9018fa5bf0d 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -160,7 +160,7 @@ static int __devinit psc_i2s_of_probe(struct platform_device *op, rc = snd_soc_register_dais(&op->dev, psc_i2s_dai, ARRAY_SIZE(psc_i2s_dai)); if (rc != 0) { pr_err("Failed to register DAI\n"); - return 0; + return rc; } psc_dma = dev_get_drvdata(&op->dev); -- cgit v1.2.3-70-g09d2 From b0fc7b840926654a3a6eaf0f41f3a4da33441d3d Mon Sep 17 00:00:00 2001 From: Marek Belisko Date: Mon, 8 Nov 2010 13:14:51 +0100 Subject: ASoC: s3c24xx: Fix compilation problem for mini2440 When make mini2440_defconfig compilation end with undefined references to DMA functions. There was missing selection for S3C2410_DMA when compile ASoC audio for S3C24xx CPU. Tested on mini2440 board. Signed-off-by: Marek Belisko Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/s3c24xx/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 8a6b53ccd20..d85bf8a0abb 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -2,6 +2,7 @@ config SND_S3C24XX_SOC tristate "SoC Audio for the Samsung S3CXXXX chips" depends on ARCH_S3C2410 || ARCH_S3C64XX || ARCH_S5PC100 || ARCH_S5PV210 select S3C64XX_DMA if ARCH_S3C64XX + select S3C2410_DMA if ARCH_S3C2410 help Say Y or M if you want to add support for codecs attached to the S3C24XX AC97 or I2S interfaces. You will also need to -- cgit v1.2.3-70-g09d2 From c28a9926f28e8c7c52603db58754a78008768ca1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 9 Nov 2010 12:00:11 +0000 Subject: ASoC: Remove broken WM8350 direction constants The WM8350 driver was using some custom constants to interpret the direction of the MCLK signal which had the opposite values to those used as standard by the ASoC core, causing confusion in machine drivers such as the 1133-EV1 board. Reported-by: Tommy Zhu Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- include/linux/mfd/wm8350/audio.h | 3 --- sound/soc/codecs/wm8350.c | 2 +- 2 files changed, 1 insertion(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/include/linux/mfd/wm8350/audio.h b/include/linux/mfd/wm8350/audio.h index a95141eafce..bd581c6fa08 100644 --- a/include/linux/mfd/wm8350/audio.h +++ b/include/linux/mfd/wm8350/audio.h @@ -522,9 +522,6 @@ #define WM8350_MCLK_SEL_PLL_32K 3 #define WM8350_MCLK_SEL_MCLK 5 -#define WM8350_MCLK_DIR_OUT 0 -#define WM8350_MCLK_DIR_IN 1 - /* clock divider id's */ #define WM8350_ADC_CLKDIV 0 #define WM8350_DAC_CLKDIV 1 diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index f4f1fba38eb..4f3e919a075 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -831,7 +831,7 @@ static int wm8350_set_dai_sysclk(struct snd_soc_dai *codec_dai, } /* MCLK direction */ - if (dir == WM8350_MCLK_DIR_OUT) + if (dir == SND_SOC_CLOCK_OUT) wm8350_set_bits(wm8350, WM8350_CLOCK_CONTROL_2, WM8350_MCLK_DIR); else -- cgit v1.2.3-70-g09d2 From 0049317edb76d17bfac736b658523c15935391a3 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 9 Nov 2010 14:38:58 +0000 Subject: ASoC: Ensure sane WM835x AIF configuration by default Ensure that whatever ran before us leaves the WM835x with a sane default audio interface configuration as we do not override the companding, loopback or tristate settings and do not reset the chip at startup (as it is a PMIC). Reported-by: Keiji Mitsuhisa Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8350.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 4f3e919a075..7611add7f8c 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1586,6 +1586,13 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec) wm8350_set_bits(wm8350, WM8350_ROUT2_VOLUME, WM8350_OUT2_VU | WM8350_OUT2R_MUTE); + /* Make sure AIF tristating is disabled by default */ + wm8350_clear_bits(wm8350, WM8350_AI_FORMATING, WM8350_AIF_TRI); + + /* Make sure we've got a sane companding setup too */ + wm8350_clear_bits(wm8350, WM8350_ADC_DAC_COMP, + WM8350_DAC_COMP | WM8350_LOOPBACK); + /* Make sure jack detect is disabled to start off with */ wm8350_clear_bits(wm8350, WM8350_JACK_DETECT, WM8350_JDL_ENA | WM8350_JDR_ENA); -- cgit v1.2.3-70-g09d2 From bbde7814cbc54d6b564d3f65b4b0e82eddef30a6 Mon Sep 17 00:00:00 2001 From: Ryan Mallon Date: Thu, 11 Nov 2010 09:02:30 +1300 Subject: Fix Atmel soc audio boards Kconfig dependency Add Kconfig dependency on AT91_PROGRAMMABLE_CLOCKS for the Atmel SoC audio SAM9G20-EK and PlayPaq boards. Fixes link errors on missing clk_set_parent and clk_set_rate when building without AT91_PROGRAMMABLE_CLOCKS. Signed-off-by: Ryan Mallon Acked-by: Geoffrey Wossum Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/atmel/Kconfig | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index e720d5e6f04..bee3c94f58b 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -16,7 +16,8 @@ config SND_ATMEL_SOC_SSC config SND_AT91_SOC_SAM9G20_WM8731 tristate "SoC Audio support for WM8731-based At91sam9g20 evaluation board" - depends on ATMEL_SSC && ARCH_AT91SAM9G20 && SND_ATMEL_SOC + depends on ATMEL_SSC && ARCH_AT91SAM9G20 && SND_ATMEL_SOC && \ + AT91_PROGRAMMABLE_CLOCKS select SND_ATMEL_SOC_SSC select SND_SOC_WM8731 help @@ -25,7 +26,7 @@ config SND_AT91_SOC_SAM9G20_WM8731 config SND_AT32_SOC_PLAYPAQ tristate "SoC Audio support for PlayPaq with WM8510" - depends on SND_ATMEL_SOC && BOARD_PLAYPAQ + depends on SND_ATMEL_SOC && BOARD_PLAYPAQ && AT91_PROGRAMMABLE_CLOCKS select SND_ATMEL_SOC_SSC select SND_SOC_WM8510 help -- cgit v1.2.3-70-g09d2 From ccb3b84fa0fb6fb7b46b461881fd60440f579696 Mon Sep 17 00:00:00 2001 From: Vasily Khoruzhick Date: Sat, 13 Nov 2010 14:53:41 +0200 Subject: ASoC: RX1950: Fix hw_params function Unfortunatelly, I misunderstood datasheet, and on s3c244x-iis when MPLLin source for master clock is selected, prescaler has no effect. Remove dividor calculation for 44100 rate; remove 88200 rate at all, rx1950 can't do it. Signed-off-by: Vasily Khoruzhick Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/s3c24xx/rx1950_uda1380.c | 20 +++----------------- 1 file changed, 3 insertions(+), 17 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/s3c24xx/rx1950_uda1380.c b/sound/soc/s3c24xx/rx1950_uda1380.c index ffd5cf2fb0a..468cc11fdf4 100644 --- a/sound/soc/s3c24xx/rx1950_uda1380.c +++ b/sound/soc/s3c24xx/rx1950_uda1380.c @@ -50,7 +50,6 @@ static unsigned int rates[] = { 16000, 44100, 48000, - 88200, }; static struct snd_pcm_hw_constraint_list hw_rates = { @@ -130,7 +129,6 @@ static const struct snd_soc_dapm_route audio_map[] = { }; static struct platform_device *s3c24xx_snd_device; -static struct clk *xtal; static int rx1950_startup(struct snd_pcm_substream *substream) { @@ -179,10 +177,8 @@ static int rx1950_hw_params(struct snd_pcm_substream *substream, case 44100: case 88200: clk_source = S3C24XX_CLKSRC_MPLL; - fs_mode = S3C2410_IISMOD_256FS; - div = clk_get_rate(xtal) / (256 * rate); - if (clk_get_rate(xtal) % (256 * rate) > (128 * rate)) - div++; + fs_mode = S3C2410_IISMOD_384FS; + div = 1; break; default: printk(KERN_ERR "%s: rate %d is not supported\n", @@ -210,7 +206,7 @@ static int rx1950_hw_params(struct snd_pcm_substream *substream, /* set MCLK division for sample rate */ ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, - S3C2410_IISMOD_384FS); + fs_mode); if (ret < 0) return ret; @@ -295,17 +291,8 @@ static int __init rx1950_init(void) goto err_plat_add; } - xtal = clk_get(&s3c24xx_snd_device->dev, "xtal"); - - if (IS_ERR(xtal)) { - ret = PTR_ERR(xtal); - platform_device_unregister(s3c24xx_snd_device); - goto err_clk; - } - return 0; -err_clk: err_plat_add: err_plat_alloc: err_gpio_conf: @@ -320,7 +307,6 @@ static void __exit rx1950_exit(void) platform_device_unregister(s3c24xx_snd_device); snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios), hp_jack_gpios); - clk_put(xtal); gpio_free(S3C2410_GPA(1)); } -- cgit v1.2.3-70-g09d2 From bcbb243396b82b0369465e9a547b7d5278cd26ad Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 12 Nov 2010 15:14:55 +0000 Subject: ASoC: Fix dapm_seq_compare() for multi-component Ensure that we keep all widget powerups in DAPM sequence by making the CODEC the last thing we compare on rather than the first thing. Also fix the fact that we're currently comparing the widget pointers rather than the CODEC pointers when we do the substraction so we won't get stable results. Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7d85c6496af..75ed6491222 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -683,12 +683,12 @@ static int dapm_seq_compare(struct snd_soc_dapm_widget *a, struct snd_soc_dapm_widget *b, int sort[]) { - if (a->codec != b->codec) - return (unsigned long)a - (unsigned long)b; if (sort[a->id] != sort[b->id]) return sort[a->id] - sort[b->id]; if (a->reg != b->reg) return a->reg - b->reg; + if (a->codec != b->codec) + return (unsigned long)a->codec - (unsigned long)b->codec; return 0; } -- cgit v1.2.3-70-g09d2 From 11e713a07e0c03e2202ad1e87cd91d45842ce3da Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 16 Nov 2010 18:39:19 +0000 Subject: ASoC: Fix register cache setup WM8994 for multi-component During the multi-component conversion the WM8994 register cache init got lost. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8994.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 0db59c3aa5d..830dfdd66c5 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3903,6 +3903,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) return -ENOMEM; snd_soc_codec_set_drvdata(codec, wm8994); + codec->reg_cache = &wm8994->reg_cache; + wm8994->pdata = dev_get_platdata(codec->dev->parent); wm8994->codec = codec; -- cgit v1.2.3-70-g09d2 From bedad0ca3fb2ba52c347b54a97b78d32e406dd96 Mon Sep 17 00:00:00 2001 From: Chris Paulson-Ellis Date: Tue, 16 Nov 2010 12:27:09 +0000 Subject: ASoC: davinci: fixes for multi-component Multi-component commit f0fba2ad broke a few things which this patch should fix. Tested on the DM355 EVM. I've been as careful as I can, but it would be good if those with access to other Davinci boards could test. -- The multi-component commit put the initialisation of snd_soc_dai.[capture|playback]_dma_data into snd_soc_dai_ops.hw_params of the McBSP, McASP & VCIF drivers (davinci-i2s.c, davinci-mcasp.c & davinci-vcif.c). The initialisation had to be moved from the probe function in these drivers because davinci_*_dai changed from snd_soc_dai to snd_soc_dai_driver. Unfortunately, the DMA params pointer is needed by davinci_pcm_open (in davinci-pcm.c) before hw_params is called. I have moved the initialisation to a new snd_soc_dai_ops.startup function in each of these drivers. This fix indicates that all platforms that use davinci-pcm must have been broken and need to test with this fix. -- The multi-component commit also changed the McBSP driver name from "davinci-asp" to "davinci-i2s" in davinci-i2s.c without updating the board level references to the driver name. This change is understandable, as there is a similarly named "davinci-mcasp" driver in davinci-mcasp.c. There is probably no 'correct' name for this driver. The DM6446 datasheet calls it the "ASP" and describes it as a "specialised McBSP". The DM355 datasheet calls it the "ASP" and describes it as a "specialised ASP". The DM365 datasheet calls it the "McBSP". Rather than fix this problem by reverting to "davinci-asp", I've elected to avoid future confusion with the "davinci-mcasp" driver by changing it to "davinci-mcbsp", which is also consistent with the names of the functions in the driver. There are other fixes required, so it was never going to be as simple as a revert anyway. -- The DM365 only has one McBSP port (of the McBSP platforms, only the DM355 has 2 ports), so I've changed the the id of the platform_device from 0 to -1. -- In davinci-evm.c, the DM6446 EVM can no longer share a snd_soc_dai_link structure with the DM355 EVM as they use different cpu DAI names (the DM355 has 2 ports and the EVM uses the second port, but the DM6446 only has 1 port). This also means that the 2 boards need different snd_soc_card structures. -- The codec_name entries in davinci-evm.c didn't match the i2c ids in the board files. I have only checked and fixed the details of the names used for the McBSP based platforms. Someone with a McASP based platform (eg DA8xx) should check the others. Signed-off-by: Chris Paulson-Ellis Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- arch/arm/mach-davinci/dm355.c | 6 +++--- arch/arm/mach-davinci/dm365.c | 6 +++--- arch/arm/mach-davinci/dm644x.c | 4 ++-- sound/soc/davinci/davinci-evm.c | 40 +++++++++++++++++++++++++++----------- sound/soc/davinci/davinci-i2s.c | 15 ++++++++++---- sound/soc/davinci/davinci-mcasp.c | 13 ++++++++++--- sound/soc/davinci/davinci-sffsdr.c | 2 +- sound/soc/davinci/davinci-vcif.c | 13 ++++++++++--- 8 files changed, 69 insertions(+), 30 deletions(-) (limited to 'sound/soc') diff --git a/arch/arm/mach-davinci/dm355.c b/arch/arm/mach-davinci/dm355.c index 9be261beae7..2652af124ac 100644 --- a/arch/arm/mach-davinci/dm355.c +++ b/arch/arm/mach-davinci/dm355.c @@ -359,8 +359,8 @@ static struct clk_lookup dm355_clks[] = { CLK(NULL, "uart1", &uart1_clk), CLK(NULL, "uart2", &uart2_clk), CLK("i2c_davinci.1", NULL, &i2c_clk), - CLK("davinci-asp.0", NULL, &asp0_clk), - CLK("davinci-asp.1", NULL, &asp1_clk), + CLK("davinci-mcbsp.0", NULL, &asp0_clk), + CLK("davinci-mcbsp.1", NULL, &asp1_clk), CLK("davinci_mmc.0", NULL, &mmcsd0_clk), CLK("davinci_mmc.1", NULL, &mmcsd1_clk), CLK("spi_davinci.0", NULL, &spi0_clk), @@ -664,7 +664,7 @@ static struct resource dm355_asp1_resources[] = { }; static struct platform_device dm355_asp1_device = { - .name = "davinci-asp", + .name = "davinci-mcbsp", .id = 1, .num_resources = ARRAY_SIZE(dm355_asp1_resources), .resource = dm355_asp1_resources, diff --git a/arch/arm/mach-davinci/dm365.c b/arch/arm/mach-davinci/dm365.c index a12065e8726..c466d710d3c 100644 --- a/arch/arm/mach-davinci/dm365.c +++ b/arch/arm/mach-davinci/dm365.c @@ -459,7 +459,7 @@ static struct clk_lookup dm365_clks[] = { CLK(NULL, "usb", &usb_clk), CLK("davinci_emac.1", NULL, &emac_clk), CLK("davinci_voicecodec", NULL, &voicecodec_clk), - CLK("davinci-asp.0", NULL, &asp0_clk), + CLK("davinci-mcbsp", NULL, &asp0_clk), CLK(NULL, "rto", &rto_clk), CLK(NULL, "mjcp", &mjcp_clk), CLK(NULL, NULL, NULL), @@ -922,8 +922,8 @@ static struct resource dm365_asp_resources[] = { }; static struct platform_device dm365_asp_device = { - .name = "davinci-asp", - .id = 0, + .name = "davinci-mcbsp", + .id = -1, .num_resources = ARRAY_SIZE(dm365_asp_resources), .resource = dm365_asp_resources, }; diff --git a/arch/arm/mach-davinci/dm644x.c b/arch/arm/mach-davinci/dm644x.c index 0608dd776a1..9a2376b3137 100644 --- a/arch/arm/mach-davinci/dm644x.c +++ b/arch/arm/mach-davinci/dm644x.c @@ -302,7 +302,7 @@ static struct clk_lookup dm644x_clks[] = { CLK("davinci_emac.1", NULL, &emac_clk), CLK("i2c_davinci.1", NULL, &i2c_clk), CLK("palm_bk3710", NULL, &ide_clk), - CLK("davinci-asp", NULL, &asp_clk), + CLK("davinci-mcbsp", NULL, &asp_clk), CLK("davinci_mmc.0", NULL, &mmcsd_clk), CLK(NULL, "spi", &spi_clk), CLK(NULL, "gpio", &gpio_clk), @@ -580,7 +580,7 @@ static struct resource dm644x_asp_resources[] = { }; static struct platform_device dm644x_asp_device = { - .name = "davinci-asp", + .name = "davinci-mcbsp", .id = -1, .num_resources = ARRAY_SIZE(dm644x_asp_resources), .resource = dm644x_asp_resources, diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 2b07b17a6b2..bc9e6b0b3f6 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -157,12 +157,23 @@ static int evm_aic3x_init(struct snd_soc_pcm_runtime *rtd) } /* davinci-evm digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link evm_dai = { +static struct snd_soc_dai_link dm6446_evm_dai = { .name = "TLV320AIC3X", .stream_name = "AIC3X", - .cpu_dai_name = "davinci-mcasp.0", + .cpu_dai_name = "davinci-mcbsp", .codec_dai_name = "tlv320aic3x-hifi", - .codec_name = "tlv320aic3x-codec.0-001a", + .codec_name = "tlv320aic3x-codec.1-001b", + .platform_name = "davinci-pcm-audio", + .init = evm_aic3x_init, + .ops = &evm_ops, +}; + +static struct snd_soc_dai_link dm355_evm_dai = { + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai_name = "davinci-mcbsp.1", + .codec_dai_name = "tlv320aic3x-hifi", + .codec_name = "tlv320aic3x-codec.1-001b", .platform_name = "davinci-pcm-audio", .init = evm_aic3x_init, .ops = &evm_ops, @@ -172,10 +183,10 @@ static struct snd_soc_dai_link dm365_evm_dai = { #ifdef CONFIG_SND_DM365_AIC3X_CODEC .name = "TLV320AIC3X", .stream_name = "AIC3X", - .cpu_dai_name = "davinci-i2s", + .cpu_dai_name = "davinci-mcbsp", .codec_dai_name = "tlv320aic3x-hifi", .init = evm_aic3x_init, - .codec_name = "tlv320aic3x-codec.0-001a", + .codec_name = "tlv320aic3x-codec.1-0018", .ops = &evm_ops, #elif defined(CONFIG_SND_DM365_VOICE_CODEC) .name = "Voice Codec - CQ93VC", @@ -219,10 +230,17 @@ static struct snd_soc_dai_link da8xx_evm_dai = { .ops = &evm_ops, }; -/* davinci dm6446, dm355 evm audio machine driver */ -static struct snd_soc_card snd_soc_card_evm = { - .name = "DaVinci EVM", - .dai_link = &evm_dai, +/* davinci dm6446 evm audio machine driver */ +static struct snd_soc_card dm6446_snd_soc_card_evm = { + .name = "DaVinci DM6446 EVM", + .dai_link = &dm6446_evm_dai, + .num_links = 1, +}; + +/* davinci dm355 evm audio machine driver */ +static struct snd_soc_card dm355_snd_soc_card_evm = { + .name = "DaVinci DM355 EVM", + .dai_link = &dm355_evm_dai, .num_links = 1, }; @@ -261,10 +279,10 @@ static int __init evm_init(void) int ret; if (machine_is_davinci_evm()) { - evm_snd_dev_data = &snd_soc_card_evm; + evm_snd_dev_data = &dm6446_snd_soc_card_evm; index = 0; } else if (machine_is_davinci_dm355_evm()) { - evm_snd_dev_data = &snd_soc_card_evm; + evm_snd_dev_data = &dm355_snd_soc_card_evm; index = 1; } else if (machine_is_davinci_dm365_evm()) { evm_snd_dev_data = &dm365_snd_soc_card_evm; diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index d46b545d41f..9e0e565e6ed 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -426,9 +426,6 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, snd_pcm_format_t fmt; unsigned element_cnt = 1; - dai->capture_dma_data = dev->dma_params; - dai->playback_dma_data = dev->dma_params; - /* general line settings */ spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { @@ -601,6 +598,15 @@ static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } +static int davinci_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_set_dma_data(dai, substream, dev->dma_params); + return 0; +} + static void davinci_i2s_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -612,6 +618,7 @@ static void davinci_i2s_shutdown(struct snd_pcm_substream *substream, #define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 static struct snd_soc_dai_ops davinci_i2s_dai_ops = { + .startup = davinci_i2s_startup, .shutdown = davinci_i2s_shutdown, .prepare = davinci_i2s_prepare, .trigger = davinci_i2s_trigger, @@ -749,7 +756,7 @@ static struct platform_driver davinci_mcbsp_driver = { .probe = davinci_i2s_probe, .remove = davinci_i2s_remove, .driver = { - .name = "davinci-i2s", + .name = "davinci-mcbsp", .owner = THIS_MODULE, }, }; diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 86918ee1241..fb55d2c5d70 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -715,9 +715,6 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, int word_length; u8 fifo_level; - cpu_dai->capture_dma_data = dev->dma_params; - cpu_dai->playback_dma_data = dev->dma_params; - davinci_hw_common_param(dev, substream->stream); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) fifo_level = dev->txnumevt; @@ -799,7 +796,17 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, return ret; } +static int davinci_mcasp_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct davinci_audio_dev *dev = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_set_dma_data(dai, substream, dev->dma_params); + return 0; +} + static struct snd_soc_dai_ops davinci_mcasp_dai_ops = { + .startup = davinci_mcasp_startup, .trigger = davinci_mcasp_trigger, .hw_params = davinci_mcasp_hw_params, .set_fmt = davinci_mcasp_set_dai_fmt, diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c index 009b6521a1b..6c6666a1f94 100644 --- a/sound/soc/davinci/davinci-sffsdr.c +++ b/sound/soc/davinci/davinci-sffsdr.c @@ -84,7 +84,7 @@ static struct snd_soc_ops sffsdr_ops = { static struct snd_soc_dai_link sffsdr_dai = { .name = "PCM3008", /* Codec name */ .stream_name = "PCM3008 HiFi", - .cpu_dai_name = "davinci-asp.0", + .cpu_dai_name = "davinci-mcbsp", .codec_dai_name = "pcm3008-hifi", .codec_name = "pcm3008-codec", .platform_name = "davinci-pcm-audio", diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c index ea232f6a2c2..fb4cc1edf33 100644 --- a/sound/soc/davinci/davinci-vcif.c +++ b/sound/soc/davinci/davinci-vcif.c @@ -97,9 +97,6 @@ static int davinci_vcif_hw_params(struct snd_pcm_substream *substream, &davinci_vcif_dev->dma_params[substream->stream]; u32 w; - dai->capture_dma_data = davinci_vcif_dev->dma_params; - dai->playback_dma_data = davinci_vcif_dev->dma_params; - /* Restart the codec before setup */ davinci_vcif_stop(substream); davinci_vcif_start(substream); @@ -174,9 +171,19 @@ static int davinci_vcif_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } +static int davinci_vcif_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct davinci_vcif_dev *dev = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_set_dma_data(dai, substream, dev->dma_params); + return 0; +} + #define DAVINCI_VCIF_RATES SNDRV_PCM_RATE_8000_48000 static struct snd_soc_dai_ops davinci_vcif_dai_ops = { + .startup = davinci_vcif_startup, .trigger = davinci_vcif_trigger, .hw_params = davinci_vcif_hw_params, }; -- cgit v1.2.3-70-g09d2 From fb762a5b37e74023f1793cdf64e40d4da38b30ec Mon Sep 17 00:00:00 2001 From: Jesse Marroquin Date: Wed, 17 Nov 2010 14:26:40 -0600 Subject: ASoC: Add support for MAX98089 CODEC This patch adds initial support for the MAX98089 CODEC. Signed-off-by: Jesse Marroquin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/max98088.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index bc22ee93a75..470cb93b1d1 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -28,6 +28,11 @@ #include #include "max98088.h" +enum max98088_type { + MAX98088, + MAX98089, +}; + struct max98088_cdata { unsigned int rate; unsigned int fmt; @@ -36,6 +41,7 @@ struct max98088_cdata { struct max98088_priv { u8 reg_cache[M98088_REG_CNT]; + enum max98088_type devtype; void *control_data; struct max98088_pdata *pdata; unsigned int sysclk; @@ -2040,6 +2046,8 @@ static int max98088_i2c_probe(struct i2c_client *i2c, if (max98088 == NULL) return -ENOMEM; + max98088->devtype = id->driver_data; + i2c_set_clientdata(i2c, max98088); max98088->control_data = i2c; max98088->pdata = i2c->dev.platform_data; @@ -2059,7 +2067,8 @@ static int __devexit max98088_i2c_remove(struct i2c_client *client) } static const struct i2c_device_id max98088_i2c_id[] = { - { "max98088", 0 }, + { "max98088", MAX98088 }, + { "max98089", MAX98089 }, { } }; MODULE_DEVICE_TABLE(i2c, max98088_i2c_id); -- cgit v1.2.3-70-g09d2 From 2811fe2beb7cb9f34eef4bc9627dcabb401bc05e Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 19 Nov 2010 15:48:06 +0800 Subject: ASoC: uda134x - set reg_cache_default to uda134x_reg After checking the code in 2.6.36, I found this is missing during multi-component conversion. Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/uda134x.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 7540a509a6f..464f0cfa4c7 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -597,6 +597,7 @@ static struct snd_soc_codec_driver soc_codec_dev_uda134x = { .resume = uda134x_soc_resume, .reg_cache_size = sizeof(uda134x_reg), .reg_word_size = sizeof(u8), + .reg_cache_default = uda134x_reg, .reg_cache_step = 1, .read = uda134x_read_reg_cache, .write = uda134x_write, -- cgit v1.2.3-70-g09d2 From 8575d93386d6ce9a3d4961134018d4e6c6bed618 Mon Sep 17 00:00:00 2001 From: Vasiliy Kulikov Date: Sun, 21 Nov 2010 20:40:21 +0300 Subject: ASoC: s3c24xx: test wrong variable After clk_get() mclk is checked three times instead of mout_epll and sclk_spdif checks. Signed-off-by: Vasiliy Kulikov Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/s3c24xx/smdk_spdif.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/s3c24xx/smdk_spdif.c b/sound/soc/s3c24xx/smdk_spdif.c index f31d22ad7c8..c8bd90488a8 100644 --- a/sound/soc/s3c24xx/smdk_spdif.c +++ b/sound/soc/s3c24xx/smdk_spdif.c @@ -38,7 +38,7 @@ static int set_audio_clock_heirachy(struct platform_device *pdev) } mout_epll = clk_get(NULL, "mout_epll"); - if (IS_ERR(fout_epll)) { + if (IS_ERR(mout_epll)) { printk(KERN_WARNING "%s: Cannot find mout_epll.\n", __func__); ret = -EINVAL; @@ -54,7 +54,7 @@ static int set_audio_clock_heirachy(struct platform_device *pdev) } sclk_spdif = clk_get(NULL, "sclk_spdif"); - if (IS_ERR(fout_epll)) { + if (IS_ERR(sclk_spdif)) { printk(KERN_WARNING "%s: Cannot find sclk_spdif.\n", __func__); ret = -EINVAL; -- cgit v1.2.3-70-g09d2 From 13a2e06c5898d27aadabfdb9830169101b21432f Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 22 Nov 2010 08:20:54 +0800 Subject: ASoC: stac9766 - set reg_cache_default to stac9766_reg Looks like this is missing during multi-component conversion. Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/stac9766.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 00d67cc8e20..061f9e5a497 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -383,6 +383,7 @@ static struct snd_soc_codec_driver soc_codec_dev_stac9766 = { .reg_cache_size = sizeof(stac9766_reg), .reg_word_size = sizeof(u16), .reg_cache_step = 2, + .reg_cache_default = stac9766_reg, }; static __devinit int stac9766_probe(struct platform_device *pdev) -- cgit v1.2.3-70-g09d2 From b3915d1fb6557dda206f4644ba9aa96ffd9a99d2 Mon Sep 17 00:00:00 2001 From: Vasiliy Kulikov Date: Mon, 22 Nov 2010 18:59:13 +0300 Subject: ASoC: atmel: test wrong variable After clk_get() mclk is checked second time instead of pllb check. In patch v1 Jarkko Nikula noticed that PTR_ERR() is also has wrong argument. Signed-off-by: Vasiliy Kulikov Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/atmel/sam9g20_wm8731.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 293569dfd0e..032e17dd8fd 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -222,9 +222,9 @@ static int __init at91sam9g20ek_init(void) } pllb = clk_get(NULL, "pllb"); - if (IS_ERR(mclk)) { + if (IS_ERR(pllb)) { printk(KERN_ERR "ASoC: Failed to get PLLB\n"); - ret = PTR_ERR(mclk); + ret = PTR_ERR(pllb); goto err_mclk; } ret = clk_set_parent(mclk, pllb); -- cgit v1.2.3-70-g09d2 From f71a4734b1ad7edccbfd9bd395df328ebbd94287 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 22 Nov 2010 19:11:48 +0000 Subject: ASoC: Fix multi-component mismerge in WM8523 Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8523.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 712ef7c76f9..9a433a5396c 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -146,7 +146,6 @@ static int wm8523_startup(struct snd_pcm_substream *substream, return -EINVAL; } - return 0; snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &wm8523->rate_constraint); -- cgit v1.2.3-70-g09d2 From eba19fdd818dfec3782ff095591e51c9bd617403 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 19 Nov 2010 16:09:15 +0000 Subject: ASoC: Restore WM8994 volatile and readable register operations They went AWOL during the multi-component merge. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8994.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 830dfdd66c5..ea3ee9fde2b 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -4073,6 +4073,8 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8994 = { .resume = wm8994_resume, .read = wm8994_read, .write = wm8994_write, + .readable_register = wm8994_readable, + .volatile_register = wm8994_volatile, .set_bias_level = wm8994_set_bias_level, }; -- cgit v1.2.3-70-g09d2 From 92a5288501685bf05fc348ee2a3115a9bd9ae36f Mon Sep 17 00:00:00 2001 From: Jesper Juhl Date: Mon, 22 Nov 2010 22:54:03 +0100 Subject: ASoC: MPC5200: Eliminate duplicate include of of_device.h Eliminate duplicate #include from sound/soc/fsl/mpc5200_dma.c Signed-off-by: Jesper Juhl Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/fsl/mpc5200_dma.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index dce6b551cd7..f92dca07cd3 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -9,7 +9,6 @@ #include #include #include -#include #include #include -- cgit v1.2.3-70-g09d2 From 7a479b02843c8d78ef51a64d1168592258440c97 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 23 Nov 2010 14:14:07 +0800 Subject: ASoC: Do not update the cache if write to hardware failed Signed-off-by: Axel Lin Acked-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/tpa6130a2.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index ee4fb201de6..d2c24309567 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -78,8 +78,10 @@ static int tpa6130a2_i2c_write(int reg, u8 value) if (data->power_state) { val = i2c_smbus_write_byte_data(tpa6130a2_client, reg, value); - if (val < 0) + if (val < 0) { dev_err(&tpa6130a2_client->dev, "Write failed\n"); + return val; + } } /* Either powered on or off, we save the context */ -- cgit v1.2.3-70-g09d2 From bc5954f00e80c55140f546c80f34a8660bdd2c5f Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 23 Nov 2010 15:56:21 +0800 Subject: ASoC: max98088 - fix a memory leak Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/max98088.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 470cb93b1d1..d63e28773eb 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -2019,7 +2019,10 @@ err_access: static int max98088_remove(struct snd_soc_codec *codec) { + struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); + max98088_set_bias_level(codec, SND_SOC_BIAS_OFF); + kfree(max98088->eq_texts); return 0; } -- cgit v1.2.3-70-g09d2 From cd70978cb59fd20dccdfc790ea8bb308c2dfd1d6 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 23 Nov 2010 15:57:49 +0800 Subject: ASoC: wm8904 - fix memory leaks Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm8904.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 33be84e506e..fca60a0b57b 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2498,6 +2498,8 @@ static int wm8904_remove(struct snd_soc_codec *codec) wm8904_set_bias_level(codec, SND_SOC_BIAS_OFF); regulator_bulk_free(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); + kfree(wm8904->retune_mobile_texts); + kfree(wm8904->drc_texts); return 0; } -- cgit v1.2.3-70-g09d2 From 24fb2b1174ddc1f844e2008eb5b3105832860395 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 23 Nov 2010 15:58:39 +0800 Subject: ASoC: wm8994 - fix memory leaks Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index ea3ee9fde2b..4d3e6f1ac58 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -4061,6 +4061,8 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC2_DET, wm8994); wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_SHRT, wm8994); wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_DET, wm8994); + kfree(wm8994->retune_mobile_texts); + kfree(wm8994->drc_texts); kfree(wm8994); return 0; -- cgit v1.2.3-70-g09d2 From d4bc99b977e3a1dd10a84a01ebe59ac2ccebf0cd Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 24 Nov 2010 02:44:06 +0000 Subject: ARM: mach-shmobile: ap4evb: FSI clock use proper process for HDMI Current AP4 FSI set_rate function used bogus clock process which didn't care enable/disable and clk->usecound. To solve this issue, this patch also modify FSI driver to call set_rate with enough options. This patch modify it. Signed-off-by: Kuninori Morimoto Signed-off-by: Paul Mundt --- arch/arm/mach-shmobile/board-ap4evb.c | 58 +++++++++++++++++++++++++++-------- arch/arm/mach-shmobile/clock-sh7372.c | 2 +- include/sound/sh_fsi.h | 6 ++-- sound/soc/sh/fsi.c | 19 ++++++++++-- 4 files changed, 67 insertions(+), 18 deletions(-) (limited to 'sound/soc') diff --git a/arch/arm/mach-shmobile/board-ap4evb.c b/arch/arm/mach-shmobile/board-ap4evb.c index d3260542b94..61c1068198e 100644 --- a/arch/arm/mach-shmobile/board-ap4evb.c +++ b/arch/arm/mach-shmobile/board-ap4evb.c @@ -567,40 +567,72 @@ static struct platform_device *qhd_devices[] __initdata = { /* FSI */ #define IRQ_FSI evt2irq(0x1840) +static int __fsi_set_rate(struct clk *clk, long rate, int enable) +{ + int ret = 0; + + if (rate <= 0) + return ret; -static int fsi_set_rate(int is_porta, int rate) + if (enable) { + ret = clk_set_rate(clk, clk_round_rate(clk, rate)); + if (0 == ret) + ret = clk_enable(clk); + } else { + clk_disable(clk); + } + + return ret; +} + +static int fsi_set_rate(struct device *dev, int is_porta, int rate, int enable) { struct clk *fsib_clk; struct clk *fdiv_clk = &sh7372_fsidivb_clk; + long fsib_rate = 0; + long fdiv_rate = 0; + int ackmd_bpfmd; int ret; /* set_rate is not needed if port A */ if (is_porta) return 0; - fsib_clk = clk_get(NULL, "fsib_clk"); - if (IS_ERR(fsib_clk)) - return -EINVAL; - switch (rate) { case 44100: - clk_set_rate(fsib_clk, clk_round_rate(fsib_clk, 11283000)); - ret = SH_FSI_ACKMD_256 | SH_FSI_BPFMD_64; + fsib_rate = rate * 256; + ackmd_bpfmd = SH_FSI_ACKMD_256 | SH_FSI_BPFMD_64; break; case 48000: - clk_set_rate(fsib_clk, clk_round_rate(fsib_clk, 85428000)); - clk_set_rate(fdiv_clk, clk_round_rate(fdiv_clk, 12204000)); - ret = SH_FSI_ACKMD_256 | SH_FSI_BPFMD_64; + fsib_rate = 85428000; /* around 48kHz x 256 x 7 */ + fdiv_rate = rate * 256; + ackmd_bpfmd = SH_FSI_ACKMD_256 | SH_FSI_BPFMD_64; break; default: pr_err("unsupported rate in FSI2 port B\n"); - ret = -EINVAL; - break; + return -EINVAL; } + /* FSI B setting */ + fsib_clk = clk_get(dev, "ickb"); + if (IS_ERR(fsib_clk)) + return -EIO; + + ret = __fsi_set_rate(fsib_clk, fsib_rate, enable); clk_put(fsib_clk); + if (ret < 0) + return ret; - return ret; + /* FSI DIV setting */ + ret = __fsi_set_rate(fdiv_clk, fdiv_rate, enable); + if (ret < 0) { + /* disable FSI B */ + if (enable) + __fsi_set_rate(fsib_clk, fsib_rate, 0); + return ret; + } + + return ackmd_bpfmd; } static struct sh_fsi_platform_info fsi_info = { diff --git a/arch/arm/mach-shmobile/clock-sh7372.c b/arch/arm/mach-shmobile/clock-sh7372.c index 13226323e4e..4191e292112 100644 --- a/arch/arm/mach-shmobile/clock-sh7372.c +++ b/arch/arm/mach-shmobile/clock-sh7372.c @@ -471,7 +471,7 @@ static int fsidiv_set_rate(struct clk *clk, return -ENOENT; __raw_writel(idx << 16, clk->mapping->base); - return fsidiv_enable(clk); + return 0; } static struct clk_ops fsidiv_clk_ops = { diff --git a/include/sound/sh_fsi.h b/include/sound/sh_fsi.h index fa60cbda90a..d79894192ae 100644 --- a/include/sound/sh_fsi.h +++ b/include/sound/sh_fsi.h @@ -85,7 +85,9 @@ * ACK_MD (FSI2) * CKG1 (FSI) * - * err: return value < 0 + * err : return value < 0 + * no change : return value == 0 + * change xMD : return value > 0 * * 0x-00000AB * @@ -111,7 +113,7 @@ struct sh_fsi_platform_info { unsigned long porta_flags; unsigned long portb_flags; - int (*set_rate)(int is_porta, int rate); /* for master mode */ + int (*set_rate)(struct device *dev, int is_porta, int rate, int enable); }; #endif /* __SOUND_FSI_H */ diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 507e709f280..136414f163e 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -132,6 +132,8 @@ struct fsi_priv { struct fsi_stream playback; struct fsi_stream capture; + long rate; + u32 mst_ctrl; }; @@ -854,10 +856,17 @@ static void fsi_dai_shutdown(struct snd_pcm_substream *substream, { struct fsi_priv *fsi = fsi_get_priv(substream); int is_play = fsi_is_play(substream); + struct fsi_master *master = fsi_get_master(fsi); + int (*set_rate)(struct device *dev, int is_porta, int rate, int enable); fsi_irq_disable(fsi, is_play); fsi_clk_ctrl(fsi, 0); + set_rate = master->info->set_rate; + if (set_rate && fsi->rate) + set_rate(dai->dev, fsi_is_port_a(fsi), fsi->rate, 0); + fsi->rate = 0; + pm_runtime_put_sync(dai->dev); } @@ -891,9 +900,10 @@ static int fsi_dai_hw_params(struct snd_pcm_substream *substream, { struct fsi_priv *fsi = fsi_get_priv(substream); struct fsi_master *master = fsi_get_master(fsi); - int (*set_rate)(int is_porta, int rate) = master->info->set_rate; + int (*set_rate)(struct device *dev, int is_porta, int rate, int enable); int fsi_ver = master->core->ver; int is_play = fsi_is_play(substream); + long rate = params_rate(params); int ret; /* if slave mode, set_rate is not needed */ @@ -901,10 +911,15 @@ static int fsi_dai_hw_params(struct snd_pcm_substream *substream, return 0; /* it is error if no set_rate */ + set_rate = master->info->set_rate; if (!set_rate) return -EIO; - ret = set_rate(fsi_is_port_a(fsi), params_rate(params)); + ret = set_rate(dai->dev, fsi_is_port_a(fsi), rate, 1); + if (ret < 0) /* error */ + return ret; + + fsi->rate = rate; if (ret > 0) { u32 data = 0; -- cgit v1.2.3-70-g09d2 From 22de4e1fe446794acaebdf19dcaff4256d659972 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 19 Nov 2010 07:23:17 +0000 Subject: ARM: mach-shmobile: ap4evb: FSI clock use proper process for ak4642 Current AP4 FSI didn't use set_rate for ak4642, and used dummy rate when init. And FSI driver was modified to always call set_rate. The user which are using FSI set_rate is only AP4 now. Signed-off-by: Kuninori Morimoto Signed-off-by: Paul Mundt --- arch/arm/mach-shmobile/board-ap4evb.c | 96 ++++++++++++++++++++++++----------- sound/soc/sh/fsi.c | 8 +-- 2 files changed, 68 insertions(+), 36 deletions(-) (limited to 'sound/soc') diff --git a/arch/arm/mach-shmobile/board-ap4evb.c b/arch/arm/mach-shmobile/board-ap4evb.c index 61c1068198e..e084b423146 100644 --- a/arch/arm/mach-shmobile/board-ap4evb.c +++ b/arch/arm/mach-shmobile/board-ap4evb.c @@ -575,7 +575,7 @@ static int __fsi_set_rate(struct clk *clk, long rate, int enable) return ret; if (enable) { - ret = clk_set_rate(clk, clk_round_rate(clk, rate)); + ret = clk_set_rate(clk, rate); if (0 == ret) ret = clk_enable(clk); } else { @@ -585,7 +585,56 @@ static int __fsi_set_rate(struct clk *clk, long rate, int enable) return ret; } -static int fsi_set_rate(struct device *dev, int is_porta, int rate, int enable) +static int __fsi_set_round_rate(struct clk *clk, long rate, int enable) +{ + return __fsi_set_rate(clk, clk_round_rate(clk, rate), enable); +} + +static int fsi_ak4642_set_rate(struct device *dev, int rate, int enable) +{ + struct clk *fsia_ick; + struct clk *fsiack; + int ret = -EIO; + + fsia_ick = clk_get(dev, "icka"); + if (IS_ERR(fsia_ick)) + return PTR_ERR(fsia_ick); + + /* + * FSIACK is connected to AK4642, + * and use external clock pin from it. + * it is parent of fsia_ick now. + */ + fsiack = clk_get_parent(fsia_ick); + if (!fsiack) + goto fsia_ick_out; + + /* + * we get 1/1 divided clock by setting same rate to fsiack and fsia_ick + * + ** FIXME ** + * Because the freq_table of external clk (fsiack) are all 0, + * the return value of clk_round_rate became 0. + * So, it use __fsi_set_rate here. + */ + ret = __fsi_set_rate(fsiack, rate, enable); + if (ret < 0) + goto fsiack_out; + + ret = __fsi_set_round_rate(fsia_ick, rate, enable); + if ((ret < 0) && enable) + __fsi_set_round_rate(fsiack, rate, 0); /* disable FSI ACK */ + +fsiack_out: + clk_put(fsiack); + +fsia_ick_out: + clk_put(fsia_ick); + + return 0; +} + +static int fsi_hdmi_set_rate(struct device *dev, int rate, int enable) { struct clk *fsib_clk; struct clk *fdiv_clk = &sh7372_fsidivb_clk; @@ -594,10 +643,6 @@ static int fsi_set_rate(struct device *dev, int is_porta, int rate, int enable) int ackmd_bpfmd; int ret; - /* set_rate is not needed if port A */ - if (is_porta) - return 0; - switch (rate) { case 44100: fsib_rate = rate * 256; @@ -618,23 +663,35 @@ static int fsi_set_rate(struct device *dev, int is_porta, int rate, int enable) if (IS_ERR(fsib_clk)) return -EIO; - ret = __fsi_set_rate(fsib_clk, fsib_rate, enable); + ret = __fsi_set_round_rate(fsib_clk, fsib_rate, enable); clk_put(fsib_clk); if (ret < 0) return ret; /* FSI DIV setting */ - ret = __fsi_set_rate(fdiv_clk, fdiv_rate, enable); + ret = __fsi_set_round_rate(fdiv_clk, fdiv_rate, enable); if (ret < 0) { /* disable FSI B */ if (enable) - __fsi_set_rate(fsib_clk, fsib_rate, 0); + __fsi_set_round_rate(fsib_clk, fsib_rate, 0); return ret; } return ackmd_bpfmd; } +static int fsi_set_rate(struct device *dev, int is_porta, int rate, int enable) +{ + int ret; + + if (is_porta) + ret = fsi_ak4642_set_rate(dev, rate, enable); + else + ret = fsi_hdmi_set_rate(dev, rate, enable); + + return ret; +} + static struct sh_fsi_platform_info fsi_info = { .porta_flags = SH_FSI_BRS_INV | SH_FSI_OUT_SLAVE_MODE | @@ -928,23 +985,11 @@ out: device_initcall(hdmi_init_pm_clock); -#define FSIACK_DUMMY_RATE 48000 static int __init fsi_init_pm_clock(void) { struct clk *fsia_ick; int ret; - /* - * FSIACK is connected to AK4642, - * and the rate is depend on playing sound rate. - * So, set dummy rate (= 48k) here - */ - ret = clk_set_rate(&sh7372_fsiack_clk, FSIACK_DUMMY_RATE); - if (ret < 0) { - pr_err("Cannot set FSIACK dummy rate: %d\n", ret); - return ret; - } - fsia_ick = clk_get(&fsi_device.dev, "icka"); if (IS_ERR(fsia_ick)) { ret = PTR_ERR(fsia_ick); @@ -953,16 +998,9 @@ static int __init fsi_init_pm_clock(void) } ret = clk_set_parent(fsia_ick, &sh7372_fsiack_clk); - if (ret < 0) { - pr_err("Cannot set FSI-A parent: %d\n", ret); - goto out; - } - - ret = clk_set_rate(fsia_ick, FSIACK_DUMMY_RATE); if (ret < 0) - pr_err("Cannot set FSI-A rate: %d\n", ret); + pr_err("Cannot set FSI-A parent: %d\n", ret); -out: clk_put(fsia_ick); return ret; diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 136414f163e..4c2404b1b86 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -902,18 +902,12 @@ static int fsi_dai_hw_params(struct snd_pcm_substream *substream, struct fsi_master *master = fsi_get_master(fsi); int (*set_rate)(struct device *dev, int is_porta, int rate, int enable); int fsi_ver = master->core->ver; - int is_play = fsi_is_play(substream); long rate = params_rate(params); int ret; - /* if slave mode, set_rate is not needed */ - if (!fsi_is_master_mode(fsi, is_play)) - return 0; - - /* it is error if no set_rate */ set_rate = master->info->set_rate; if (!set_rate) - return -EIO; + return 0; ret = set_rate(dai->dev, fsi_is_port_a(fsi), rate, 1); if (ret < 0) /* error */ -- cgit v1.2.3-70-g09d2 From 08b1a38465cab8c2224a5202c7a3b5e5f5630894 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 24 Nov 2010 10:20:33 +0800 Subject: ASoC: wm8961 - clear WM8961_DACSLOPE bit for normal mode DACSLOPE bit of Register 06h ADC and DAC Control 2: 0: Normal mode 1: Sloping stop-band mode Thus in the case of normal mode, we should clear DACSLOPE bit. Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/wm8961.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 4f326f60410..93fecf5f94c 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -711,7 +711,7 @@ static int wm8961_hw_params(struct snd_pcm_substream *substream, if (fs <= 24000) reg |= WM8961_DACSLOPE; else - reg &= WM8961_DACSLOPE; + reg &= ~WM8961_DACSLOPE; snd_soc_write(codec, WM8961_ADC_DAC_CONTROL_2, reg); return 0; -- cgit v1.2.3-70-g09d2 From 2f7dceeda4708f470fd927adb3861bd8ebbe2310 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 24 Nov 2010 10:21:54 +0800 Subject: ASoC: wm8961 - clear WM8961_MCLKDIV bit for freq <= 16500000 MCLKDIV bit of Register 04h Clocking1: 0 : Divide by 1 1 : Divide by 2 Thus in the case of freq <= 16500000, we should clear MCLKDIV bit. Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/wm8961.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 93fecf5f94c..8340485c985 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -736,7 +736,7 @@ static int wm8961_set_sysclk(struct snd_soc_dai *dai, int clk_id, freq /= 2; } else { dev_dbg(codec->dev, "Using MCLK/1 for %dHz MCLK\n", freq); - reg &= WM8961_MCLKDIV; + reg &= ~WM8961_MCLKDIV; } snd_soc_write(codec, WM8961_CLOCKING1, reg); -- cgit v1.2.3-70-g09d2 From 5c12d20145ce30f9f8b7415d36dace5fb4dcc4f0 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 24 Nov 2010 15:20:48 +0800 Subject: ASoC: Return proper error for omap3pandora_soc_init Return PTR_ERR(omap3pandora_dac_reg) instead of 0 if regulator_get failed. Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/omap/omap3pandora.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index dbd9d96b5f9..4ee33ce2cb9 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -306,6 +306,7 @@ static int __init omap3pandora_soc_init(void) pr_err(PREFIX "Failed to get DAC regulator from %s: %ld\n", dev_name(&omap3pandora_snd_device->dev), PTR_ERR(omap3pandora_dac_reg)); + ret = PTR_ERR(omap3pandora_dac_reg); goto fail3; } -- cgit v1.2.3-70-g09d2 From d6f443ae4c1d54379ad5953d7bcb89a63387184d Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 24 Nov 2010 16:44:23 +0800 Subject: ASoC: nuc900-ac97: fix a memory leak Signed-off-by: Axel Lin Acked-by: Liam Girdwood Acked-by: Wan ZongShun Signed-off-by: Mark Brown --- sound/soc/nuc900/nuc900-ac97.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c index 293dc748797..e00e39dd657 100644 --- a/sound/soc/nuc900/nuc900-ac97.c +++ b/sound/soc/nuc900/nuc900-ac97.c @@ -384,7 +384,6 @@ out0: static int __devexit nuc900_ac97_drvremove(struct platform_device *pdev) { - snd_soc_unregister_dai(&pdev->dev); clk_put(nuc900_ac97_data->clk); @@ -392,6 +391,7 @@ static int __devexit nuc900_ac97_drvremove(struct platform_device *pdev) release_mem_region(nuc900_ac97_data->res->start, resource_size(nuc900_ac97_data->res)); + kfree(nuc900_ac97_data); nuc900_ac97_data = NULL; return 0; -- cgit v1.2.3-70-g09d2 From 846172dfe33c7ee07638e04f94dd90e21dfdc5ba Mon Sep 17 00:00:00 2001 From: Dmitry Artamonow Date: Thu, 25 Nov 2010 00:46:15 +0300 Subject: ASoC: fix SND_PXA2XX_LIB Kconfig warning Fix following warning observed when SND_PXA2XX_SOC is set and SND_ARM isn't: warning: (SND_PXA2XX_AC97 && SOUND && !M68K && SND && SND_ARM && ARCH_PXA || SND_PXA2XX_SOC && SOUND && !M68K && SND && SND_SOC && ARCH_PXA) selects SND_PXA2XX_LIB which has unmet direct dependencies (SOUND && !M68K && SND && SND_ARM) Signed-off-by: Dmitry Artamonow Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/pxa/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 37f191bbfdd..580f4857130 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -1,6 +1,7 @@ config SND_PXA2XX_SOC tristate "SoC Audio for the Intel PXA2xx chip" depends on ARCH_PXA + select SND_ARM select SND_PXA2XX_LIB help Say Y or M if you want to add support for codecs attached to -- cgit v1.2.3-70-g09d2 From 3b6bc354cb22b1069f88acdc7673d3476fbadfca Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 25 Nov 2010 17:23:55 +0800 Subject: ASoC: Call snd_soc_unregister_dais instead of snd_soc_unregister_dai in sh4_soc_dai_remove We call snd_soc_register_dais() in sh4_soc_dai_probe(), thus we should call snd_soc_unregister_dais() in sh4_soc_dai_remove(). Otherwise, we got "too many arguments to function 'snd_soc_unregister_dai'" error message. Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/sh/ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c index 40bbdf1591d..05192d97b37 100644 --- a/sound/soc/sh/ssi.c +++ b/sound/soc/sh/ssi.c @@ -387,7 +387,7 @@ static int __devinit sh4_soc_dai_probe(struct platform_device *pdev) static int __devexit sh4_soc_dai_remove(struct platform_device *pdev) { - snd_soc_unregister_dai(&pdev->dev, ARRAY_SIZE(sh4_ssi_dai)); + snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(sh4_ssi_dai)); return 0; } -- cgit v1.2.3-70-g09d2 From 4e1f86509732ccc39938974db0612d14afbca953 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 25 Nov 2010 15:07:25 +0800 Subject: ASoC: efika-audio-fabric: fix resource leak in efika_fabric_init error path Add missing platform_device_put() if platform_device_add() failed. Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/fsl/efika-audio-fabric.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/fsl/efika-audio-fabric.c b/sound/soc/fsl/efika-audio-fabric.c index 53251e6b5bd..108b5d8bd0e 100644 --- a/sound/soc/fsl/efika-audio-fabric.c +++ b/sound/soc/fsl/efika-audio-fabric.c @@ -76,6 +76,7 @@ static __init int efika_fabric_init(void) rc = platform_device_add(pdev); if (rc) { pr_err("efika_fabric_init: platform_device_add() failed\n"); + platform_device_put(pdev); return -ENODEV; } return 0; -- cgit v1.2.3-70-g09d2 From 917dac0ff1754776b86967b0ec1750022d9c4265 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 25 Nov 2010 15:08:31 +0800 Subject: ASoC: pcm030-audio-fabric: fix resource leak in pcm030_fabric_init error path Add missing platform_device_put() if platform_device_add() failed. Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/fsl/pcm030-audio-fabric.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index 25f27ec1dd6..ba4d85e317e 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -76,6 +76,7 @@ static __init int pcm030_fabric_init(void) rc = platform_device_add(pdev); if (rc) { pr_err("pcm030_fabric_init: platform_device_add() failed\n"); + platform_device_put(pdev); return -ENODEV; } return 0; -- cgit v1.2.3-70-g09d2 From b193deead8637291138a8c1c49753ee686fa5b17 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 25 Nov 2010 10:44:59 +0800 Subject: ASoC: snd-soc-afeb9260: remove unneeded platform_device_del in error path Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/atmel/snd-soc-afeb9260.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c index e3d283561c1..86e0f8586dc 100644 --- a/sound/soc/atmel/snd-soc-afeb9260.c +++ b/sound/soc/atmel/snd-soc-afeb9260.c @@ -167,7 +167,6 @@ static int __init afeb9260_soc_init(void) return 0; err1: - platform_device_del(afeb9260_snd_device); platform_device_put(afeb9260_snd_device); return err; } -- cgit v1.2.3-70-g09d2 From c7a734e58ed237ecac2608a70eb31ba64e21c768 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 25 Nov 2010 15:11:03 +0800 Subject: ASoC: sam9g20_wm8731: fix resource leak in at91sam9g20ek_init error path Fix the error path to properly free allocated resources. Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/atmel/sam9g20_wm8731.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 032e17dd8fd..e521ada8054 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -240,6 +240,7 @@ static int __init at91sam9g20ek_init(void) if (!at91sam9g20ek_snd_device) { printk(KERN_ERR "ASoC: Platform device allocation failed\n"); ret = -ENOMEM; + goto err_mclk; } platform_set_drvdata(at91sam9g20ek_snd_device, @@ -248,11 +249,13 @@ static int __init at91sam9g20ek_init(void) ret = platform_device_add(at91sam9g20ek_snd_device); if (ret) { printk(KERN_ERR "ASoC: Platform device allocation failed\n"); - platform_device_put(at91sam9g20ek_snd_device); + goto err_device_add; } return ret; +err_device_add: + platform_device_put(at91sam9g20ek_snd_device); err_mclk: clk_put(mclk); mclk = NULL; -- cgit v1.2.3-70-g09d2 From 14abca3dfc51c0a4f798183f131d63bfd6552bd4 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 25 Nov 2010 15:12:30 +0800 Subject: ASoC: simone: fix resource leak in simone_init error path Fix the error path to properly free allocated resources. Signed-off-by: Axel Lin Acked-by: Mika Westerberg Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/ep93xx/simone.c | 18 ++++++++++-------- 1 file changed, 10 insertions(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/ep93xx/simone.c b/sound/soc/ep93xx/simone.c index 4b0d1991372..286817946c5 100644 --- a/sound/soc/ep93xx/simone.c +++ b/sound/soc/ep93xx/simone.c @@ -54,24 +54,26 @@ static int __init simone_init(void) ret = platform_device_add(simone_snd_ac97_device); if (ret) - goto fail; + goto fail1; simone_snd_device = platform_device_alloc("soc-audio", -1); if (!simone_snd_device) { ret = -ENOMEM; - goto fail; + goto fail2; } platform_set_drvdata(simone_snd_device, &snd_soc_simone); ret = platform_device_add(simone_snd_device); - if (ret) { - platform_device_put(simone_snd_device); - goto fail; - } + if (ret) + goto fail3; - return ret; + return 0; -fail: +fail3: + platform_device_put(simone_snd_device); +fail2: + platform_device_del(simone_snd_ac97_device); +fail1: platform_device_put(simone_snd_ac97_device); return ret; } -- cgit v1.2.3-70-g09d2 From ac8f924af555573e29b126ac5cef4fdd122ae517 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 25 Nov 2010 15:13:09 +0800 Subject: ASoC: imx-ssi: fix resource leak Fix imx_ssi_probe() error path and imx_ssi_remove() to properly free allocated resources. Signed-off-by: Axel Lin Acked-by: Sascha Hauer Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/imx/imx-ssi.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index d2d98c75ee8..390b6ffc265 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -679,8 +679,11 @@ static int imx_ssi_probe(struct platform_device *pdev) } ssi->soc_platform_pdev_fiq = platform_device_alloc("imx-fiq-pcm-audio", pdev->id); - if (!ssi->soc_platform_pdev_fiq) + if (!ssi->soc_platform_pdev_fiq) { + ret = -ENOMEM; goto failed_pdev_fiq_alloc; + } + platform_set_drvdata(ssi->soc_platform_pdev_fiq, ssi); ret = platform_device_add(ssi->soc_platform_pdev_fiq); if (ret) { @@ -689,8 +692,11 @@ static int imx_ssi_probe(struct platform_device *pdev) } ssi->soc_platform_pdev = platform_device_alloc("imx-pcm-audio", pdev->id); - if (!ssi->soc_platform_pdev) + if (!ssi->soc_platform_pdev) { + ret = -ENOMEM; goto failed_pdev_alloc; + } + platform_set_drvdata(ssi->soc_platform_pdev, ssi); ret = platform_device_add(ssi->soc_platform_pdev); if (ret) { @@ -703,6 +709,7 @@ static int imx_ssi_probe(struct platform_device *pdev) failed_pdev_add: platform_device_put(ssi->soc_platform_pdev); failed_pdev_alloc: + platform_device_del(ssi->soc_platform_pdev_fiq); failed_pdev_fiq_add: platform_device_put(ssi->soc_platform_pdev_fiq); failed_pdev_fiq_alloc: @@ -726,8 +733,8 @@ static int __devexit imx_ssi_remove(struct platform_device *pdev) struct resource *res = platform_get_resource(pdev, IORESOURCE_MEM, 0); struct imx_ssi *ssi = platform_get_drvdata(pdev); - platform_device_del(ssi->soc_platform_pdev); - platform_device_put(ssi->soc_platform_pdev); + platform_device_unregister(ssi->soc_platform_pdev); + platform_device_unregister(ssi->soc_platform_pdev_fiq); snd_soc_unregister_dai(&pdev->dev); -- cgit v1.2.3-70-g09d2 From 09de9533348632fbbf32ce618f669882aa718817 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 25 Nov 2010 15:14:03 +0800 Subject: ASoC: phycore-ac97: fix resource leak Fix imx_phycore_init() error path and imx_phycore_exit() to properly free allocated resources. Signed-off-by: Axel Lin Acked-by: Sascha Hauer Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/imx/phycore-ac97.c | 28 +++++++++++++++++++++------- 1 file changed, 21 insertions(+), 7 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/imx/phycore-ac97.c b/sound/soc/imx/phycore-ac97.c index 39f23734781..9eabc28667e 100644 --- a/sound/soc/imx/phycore-ac97.c +++ b/sound/soc/imx/phycore-ac97.c @@ -43,6 +43,7 @@ static struct snd_soc_card imx_phycore = { .num_links = ARRAY_SIZE(imx_phycore_dai_ac97), }; +static struct platform_device *imx_phycore_snd_ac97_device; static struct platform_device *imx_phycore_snd_device; static int __init imx_phycore_init(void) @@ -53,29 +54,42 @@ static int __init imx_phycore_init(void) /* return happy. We might run on a totally different machine */ return 0; - imx_phycore_snd_device = platform_device_alloc("soc-audio", -1); - if (!imx_phycore_snd_device) + imx_phycore_snd_ac97_device = platform_device_alloc("soc-audio", -1); + if (!imx_phycore_snd_ac97_device) return -ENOMEM; - platform_set_drvdata(imx_phycore_snd_device, &imx_phycore); - ret = platform_device_add(imx_phycore_snd_device); + platform_set_drvdata(imx_phycore_snd_ac97_device, &imx_phycore); + ret = platform_device_add(imx_phycore_snd_ac97_device); + if (ret) + goto fail1; imx_phycore_snd_device = platform_device_alloc("wm9712-codec", -1); - if (!imx_phycore_snd_device) - return -ENOMEM; + if (!imx_phycore_snd_device) { + ret = -ENOMEM; + goto fail2; + } ret = platform_device_add(imx_phycore_snd_device); if (ret) { printk(KERN_ERR "ASoC: Platform device allocation failed\n"); - platform_device_put(imx_phycore_snd_device); + goto fail3; } + return 0; + +fail3: + platform_device_put(imx_phycore_snd_device); +fail2: + platform_device_del(imx_phycore_snd_ac97_device); +fail1: + platform_device_put(imx_phycore_snd_ac97_device); return ret; } static void __exit imx_phycore_exit(void) { platform_device_unregister(imx_phycore_snd_device); + platform_device_unregister(imx_phycore_snd_ac97_device); } late_initcall(imx_phycore_init); -- cgit v1.2.3-70-g09d2 From 8b6b30ab665d3bbb23180c39f6215e6f64516ed0 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 25 Nov 2010 11:33:14 +0800 Subject: ASoC: davinci-vcif - fix a memory leak Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-vcif.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c index fb4cc1edf33..9d2afccc3a2 100644 --- a/sound/soc/davinci/davinci-vcif.c +++ b/sound/soc/davinci/davinci-vcif.c @@ -247,7 +247,10 @@ fail: static int davinci_vcif_remove(struct platform_device *pdev) { + struct davinci_vcif_dev *davinci_vcif_dev = dev_get_drvdata(&pdev->dev); + snd_soc_unregister_dai(&pdev->dev); + kfree(davinci_vcif_dev); return 0; } -- cgit v1.2.3-70-g09d2 From fe99b55994f08d321cc5f621c3634b1de4961d01 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 24 Nov 2010 22:40:59 +0800 Subject: ASoC: tlv320aic3x - fix variable may be used uninitialized warning If aic3x_read failed , val is used uninitialized. Fix it by initializing val to 0. This patch fixes below compile warning: sound/soc/codecs/tlv320aic3x.c: In function 'aic3x_get_gpio': sound/soc/codecs/tlv320aic3x.c:1183: warning: 'val' may be used uninitialized in this function sound/soc/codecs/tlv320aic3x.c: In function 'aic3x_headset_detected': sound/soc/codecs/tlv320aic3x.c:1211: warning: 'val' may be used uninitialized in this function sound/soc/codecs/tlv320aic3x.c: In function 'aic3x_button_pressed': sound/soc/codecs/tlv320aic3x.c:1219: warning: 'val' may be used uninitialized in this function Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index fc687790188..77b8f9ae29b 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1176,7 +1176,7 @@ EXPORT_SYMBOL_GPL(aic3x_set_gpio); int aic3x_get_gpio(struct snd_soc_codec *codec, int gpio) { u8 reg = gpio ? AIC3X_GPIO2_REG : AIC3X_GPIO1_REG; - u8 val, bit = gpio ? 2: 1; + u8 val = 0, bit = gpio ? 2 : 1; aic3x_read(codec, reg, &val); return (val >> bit) & 1; @@ -1204,7 +1204,7 @@ EXPORT_SYMBOL_GPL(aic3x_set_headset_detection); int aic3x_headset_detected(struct snd_soc_codec *codec) { - u8 val; + u8 val = 0; aic3x_read(codec, AIC3X_HEADSET_DETECT_CTRL_B, &val); return (val >> 4) & 1; } @@ -1212,7 +1212,7 @@ EXPORT_SYMBOL_GPL(aic3x_headset_detected); int aic3x_button_pressed(struct snd_soc_codec *codec) { - u8 val; + u8 val = 0; aic3x_read(codec, AIC3X_HEADSET_DETECT_CTRL_B, &val); return (val >> 5) & 1; } -- cgit v1.2.3-70-g09d2 From 25436180ee8bed6740f29d92c2030c759885c147 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 24 Nov 2010 22:24:01 +0800 Subject: ASoC: Fix resource reclaim for osk5912 In current implementation, there are resources leak in the error path. This patch properly reclaims the allocated resources in the error path. Also adds a missing clk_put in osk_soc_exit. Signed-off-by: Axel Lin Acked-by: Jarkko Nikula Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/omap/osk5912.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c index f0e66255642..65ae00e976e 100644 --- a/sound/soc/omap/osk5912.c +++ b/sound/soc/omap/osk5912.c @@ -177,7 +177,8 @@ static int __init osk_soc_init(void) tlv320aic23_mclk = clk_get(dev, "mclk"); if (IS_ERR(tlv320aic23_mclk)) { printk(KERN_ERR "Could not get mclk clock\n"); - return -ENODEV; + err = PTR_ERR(tlv320aic23_mclk); + goto err2; } /* @@ -188,7 +189,7 @@ static int __init osk_soc_init(void) if (clk_set_rate(tlv320aic23_mclk, CODEC_CLOCK)) { printk(KERN_ERR "Cannot set MCLK for AIC23 CODEC\n"); err = -ECANCELED; - goto err1; + goto err3; } } @@ -196,9 +197,12 @@ static int __init osk_soc_init(void) (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK); return 0; -err1: + +err3: clk_put(tlv320aic23_mclk); +err2: platform_device_del(osk_snd_device); +err1: platform_device_put(osk_snd_device); return err; @@ -207,6 +211,7 @@ err1: static void __exit osk_soc_exit(void) { + clk_put(tlv320aic23_mclk); platform_device_unregister(osk_snd_device); } -- cgit v1.2.3-70-g09d2 From 5a8f1d4701a50bc2a1e112f6c8e7d30f63597eae Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 29 Nov 2010 17:39:10 +0800 Subject: ASoC: Fix compile error for nuc900-ac97.c Fix below compile error by add a missing ';'. CC sound/soc/nuc900/nuc900-ac97.o sound/soc/nuc900/nuc900-ac97.c:300: warning: initialization from incompatible pointer type sound/soc/nuc900/nuc900-ac97.c:301: warning: initialization from incompatible pointer type sound/soc/nuc900/nuc900-ac97.c:318: error: expected ',' or ';' before 'static' sound/soc/nuc900/nuc900-ac97.c:405: error: 'nuc900_ac97_drvprobe' undeclared here (not in a function) make[3]: *** [sound/soc/nuc900/nuc900-ac97.o] Error 1 make[2]: *** [sound/soc/nuc900] Error 2 make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/nuc900/nuc900-ac97.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c index e00e39dd657..4f056b4a102 100644 --- a/sound/soc/nuc900/nuc900-ac97.c +++ b/sound/soc/nuc900/nuc900-ac97.c @@ -313,7 +313,7 @@ static struct snd_soc_dai_driver nuc900_ac97_dai = { .channels_max = 2, }, .ops = &nuc900_ac97_dai_ops, -} +}; static int __devinit nuc900_ac97_drvprobe(struct platform_device *pdev) { -- cgit v1.2.3-70-g09d2 From e3edefbd4a9071daf388978355f69c37fbeae261 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 29 Nov 2010 17:40:05 +0800 Subject: ASoC: Fix prototype for nuc900_ac97_probe and nuc900_ac97_remove This patch fixes below compile warning: CC sound/soc/nuc900/nuc900-ac97.o sound/soc/nuc900/nuc900-ac97.c:300: warning: initialization from incompatible pointer type sound/soc/nuc900/nuc900-ac97.c:301: warning: initialization from incompatible pointer type Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/nuc900/nuc900-ac97.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c index 4f056b4a102..3d9d8b1636b 100644 --- a/sound/soc/nuc900/nuc900-ac97.c +++ b/sound/soc/nuc900/nuc900-ac97.c @@ -263,8 +263,7 @@ static int nuc900_ac97_trigger(struct snd_pcm_substream *substream, return ret; } -static int nuc900_ac97_probe(struct platform_device *pdev, - struct snd_soc_dai *dai) +static int nuc900_ac97_probe(struct snd_soc_dai *dai) { struct nuc900_audio *nuc900_audio = nuc900_ac97_data; unsigned long val; @@ -284,12 +283,12 @@ static int nuc900_ac97_probe(struct platform_device *pdev, return 0; } -static void nuc900_ac97_remove(struct platform_device *pdev, - struct snd_soc_dai *dai) +static int nuc900_ac97_remove(struct snd_soc_dai *dai) { struct nuc900_audio *nuc900_audio = nuc900_ac97_data; clk_disable(nuc900_audio->clk); + return 0; } static struct snd_soc_dai_ops nuc900_ac97_dai_ops = { -- cgit v1.2.3-70-g09d2 From a7a9820bae19775df1d6cc70d2571ee26e099413 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 29 Nov 2010 17:40:53 +0800 Subject: ASoC: Fix compile error for nuc900-pcm.c This patch fixes below error: CC sound/soc/nuc900/nuc900-pcm.o sound/soc/nuc900/nuc900-pcm.c: In function 'nuc900_dma_open': sound/soc/nuc900/nuc900-pcm.c:267: error: 'nuc900_ac97_data' undeclared (first use in this function) sound/soc/nuc900/nuc900-pcm.c:267: error: (Each undeclared identifier is reported only once sound/soc/nuc900/nuc900-pcm.c:267: error: for each function it appears in.) sound/soc/nuc900/nuc900-pcm.c: At top level: sound/soc/nuc900/nuc900-pcm.c:337: error: expected ',' or ';' before 'static' sound/soc/nuc900/nuc900-pcm.c:354: error: 'nuc900_soc_platform_probe' undeclared here (not in a function) make[3]: *** [sound/soc/nuc900/nuc900-pcm.o] Error 1 make[2]: *** [sound/soc/nuc900] Error 2 make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/nuc900/nuc900-audio.h | 2 ++ sound/soc/nuc900/nuc900-pcm.c | 2 +- 2 files changed, 3 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/nuc900/nuc900-audio.h b/sound/soc/nuc900/nuc900-audio.h index aeed8ead2b2..59f7e8ed1a6 100644 --- a/sound/soc/nuc900/nuc900-audio.h +++ b/sound/soc/nuc900/nuc900-audio.h @@ -110,4 +110,6 @@ struct nuc900_audio { }; +extern struct nuc900_audio *nuc900_ac97_data; + #endif /*end _NUC900_AUDIO_H */ diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c index 195d1ac9477..2245f8b8edc 100644 --- a/sound/soc/nuc900/nuc900-pcm.c +++ b/sound/soc/nuc900/nuc900-pcm.c @@ -332,7 +332,7 @@ static struct snd_soc_platform_driver nuc900_soc_platform = { .ops = &nuc900_dma_ops, .pcm_new = nuc900_dma_new, .pcm_free = nuc900_dma_free_dma_buffers, -} +}; static int __devinit nuc900_soc_platform_probe(struct platform_device *pdev) { -- cgit v1.2.3-70-g09d2 From 3f90e5028a03be4496a04e4599b16f4420ff1304 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 29 Nov 2010 17:43:39 +0800 Subject: ASoC: Remove unneeded !! operations while checking return value of nuc900_checkready I think this unneededd !! operations just reduce the readability. Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/nuc900/nuc900-ac97.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c index 3d9d8b1636b..dac6732da96 100644 --- a/sound/soc/nuc900/nuc900-ac97.c +++ b/sound/soc/nuc900/nuc900-ac97.c @@ -49,7 +49,7 @@ static unsigned short nuc900_ac97_read(struct snd_ac97 *ac97, mutex_lock(&ac97_mutex); val = nuc900_checkready(); - if (!!val) { + if (val) { dev_err(nuc900_audio->dev, "AC97 codec is not ready\n"); goto out; } @@ -102,7 +102,7 @@ static void nuc900_ac97_write(struct snd_ac97 *ac97, unsigned short reg, mutex_lock(&ac97_mutex); tmp = nuc900_checkready(); - if (!!tmp) + if (tmp) dev_err(nuc900_audio->dev, "AC97 codec is not ready\n"); /* clear the R_WB bit and write register index */ @@ -149,7 +149,7 @@ static void nuc900_ac97_warm_reset(struct snd_ac97 *ac97) udelay(100); val = nuc900_checkready(); - if (!!val) + if (val) dev_err(nuc900_audio->dev, "AC97 codec is not ready\n"); mutex_unlock(&ac97_mutex); -- cgit v1.2.3-70-g09d2 From 67bd489aa309a680b1462ad635df29e8825152d2 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 29 Nov 2010 14:54:58 +0800 Subject: ASoC: Add missing dev_set_drvdata in mpc8610_hpcd_probe Otherwise, calling dev_get_drvdata in mpc8610_hpcd_remove returns NULL. Signed-off-by: Axel Lin Acked-by: Timur Tabi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/fsl/mpc8610_hpcd.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 0d7dcf1e486..7d7847a1e66 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -498,6 +498,7 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) dev_err(&pdev->dev, "platform device add failed\n"); goto error; } + dev_set_drvdata(&pdev->dev, sound_device); of_node_put(codec_np); -- cgit v1.2.3-70-g09d2 From 39a545559f8d5f13e8a4a7dfddcaad0e2ba9bcfb Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 29 Nov 2010 14:55:58 +0800 Subject: ASoC: Add missing dev_set_drvdata in p1022_ds_probe Otherwise, calling dev_get_drvdata in p1022_ds_remove returns NULL. Signed-off-by: Axel Lin Acked-by: Timur Tabi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/fsl/p1022_ds.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index 63b9eaa1ebc..026b756961e 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -498,6 +498,7 @@ static int p1022_ds_probe(struct platform_device *pdev) dev_err(&pdev->dev, "platform device add failed\n"); goto error; } + dev_set_drvdata(&pdev->dev, sound_device); of_node_put(codec_np); -- cgit v1.2.3-70-g09d2 From 3f1af9d26fb02a99a60a045b8ae93ccc6fe50b97 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 29 Nov 2010 17:42:47 +0800 Subject: ASoC: Fix missing spin_unlock_irqrestore In nuc900_dma_hw_params(), if snd_pcm_lib_malloc_pages failed it returns without calling spin_unlock_irqrestore(). Since snd_pcm_lib_malloc_pages() does not touch struct nuc900_audio, we don't need to hold the lock while calling snd_pcm_lib_malloc_pages(). Fix it by moving spin_lock_irqsave() down to after snd_pcm_lib_malloc_pages(). In nuc900_dma_prepare(), spin_unlock_irqrestore() is missing in the error path. Fix it by removing the return in default case. Signed-off-by: Axel Lin Acked-by: Wan ZongShun Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/nuc900/nuc900-pcm.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c index 2245f8b8edc..8263f56dc66 100644 --- a/sound/soc/nuc900/nuc900-pcm.c +++ b/sound/soc/nuc900/nuc900-pcm.c @@ -50,12 +50,12 @@ static int nuc900_dma_hw_params(struct snd_pcm_substream *substream, unsigned long flags; int ret = 0; - spin_lock_irqsave(&nuc900_audio->lock, flags); - ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); if (ret < 0) return ret; + spin_lock_irqsave(&nuc900_audio->lock, flags); + nuc900_audio->substream = substream; nuc900_audio->dma_addr[substream->stream] = runtime->dma_addr; nuc900_audio->buffersize[substream->stream] = @@ -169,6 +169,7 @@ static int nuc900_dma_prepare(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct nuc900_audio *nuc900_audio = runtime->private_data; unsigned long flags, val; + int ret = 0; spin_lock_irqsave(&nuc900_audio->lock, flags); @@ -197,10 +198,10 @@ static int nuc900_dma_prepare(struct snd_pcm_substream *substream) AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val); break; default: - return -EINVAL; + ret = -EINVAL; } spin_unlock_irqrestore(&nuc900_audio->lock, flags); - return 0; + return ret; } static int nuc900_dma_trigger(struct snd_pcm_substream *substream, int cmd) -- cgit v1.2.3-70-g09d2 From b1d36b1c3573fd5adecbd313d30a8bdc8d7fbc5e Mon Sep 17 00:00:00 2001 From: Daniel Glöckner Date: Tue, 30 Nov 2010 01:00:16 +0100 Subject: s6000-i2s: fix compilation MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit A semicolon was missing. Signed-off-by: Daniel Glöckner Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/s6000/s6000-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c index 8778faa174a..3052f64b240 100644 --- a/sound/soc/s6000/s6000-i2s.c +++ b/sound/soc/s6000/s6000-i2s.c @@ -434,7 +434,7 @@ static struct snd_soc_dai_driver s6000_i2s_dai = { .rate_max = 1562500, }, .ops = &s6000_i2s_dai_ops, -} +}; static int __devinit s6000_i2s_probe(struct platform_device *pdev) { -- cgit v1.2.3-70-g09d2 From 9e4ea718d3c53f9f2a65ddddf95ffd7743be458e Mon Sep 17 00:00:00 2001 From: Daniel Glöckner Date: Tue, 30 Nov 2010 01:00:17 +0100 Subject: s6000-pcm: fix compilation MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit s6000_soc_platform has lost its forward declaration and there no longer is a name element in it, so use a string constant when calling request_irq. Signed-off-by: Daniel Glöckner Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/s6000/s6000-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c index 271fd222bf1..ab3ccaec72d 100644 --- a/sound/soc/s6000/s6000-pcm.c +++ b/sound/soc/s6000/s6000-pcm.c @@ -473,7 +473,7 @@ static int s6000_pcm_new(struct snd_card *card, } res = request_irq(params->irq, s6000_pcm_irq, IRQF_SHARED, - s6000_soc_platform.name, pcm); + "s6000-audio", pcm); if (res) { printk(KERN_ERR "s6000-pcm couldn't get IRQ\n"); return res; -- cgit v1.2.3-70-g09d2 From b76fb39d49f67a484a6adc8f041d9ad833f6860e Mon Sep 17 00:00:00 2001 From: Daniel Glöckner Date: Tue, 30 Nov 2010 01:00:18 +0100 Subject: s6105-ipcam: fix compilation MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit When the s6105-ipcam ASoC driver had been converted to the multi-component API, a single reference to a former structure element remained, blocking successful compilation. Signed-off-by: Daniel Glöckner Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/s6000/s6105-ipcam.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/s6000/s6105-ipcam.c b/sound/soc/s6000/s6105-ipcam.c index 96c05e13753..c1244c5bc73 100644 --- a/sound/soc/s6000/s6105-ipcam.c +++ b/sound/soc/s6000/s6105-ipcam.c @@ -167,7 +167,7 @@ static int s6105_aic3x_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_sync(codec); - snd_ctl_add(codec->snd_card, snd_ctl_new1(&audio_out_mux, codec)); + snd_ctl_add(codec->card->snd_card, snd_ctl_new1(&audio_out_mux, codec)); return 0; } -- cgit v1.2.3-70-g09d2 From 2062ea522bb58bb2aeee86d051b37136491ccd65 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Wed, 1 Dec 2010 09:38:55 +0000 Subject: ASoC: WM8731: Fix incorrect mask for bypass path disable According to the datasheet the bypass path enable/disable is bit 3 therefore we need 0x8 and not 0x4. Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm8731.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 631385802eb..e725c09a3e7 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -526,7 +526,7 @@ static int wm8731_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8731_RINVOL, 0x100, 0); /* Disable bypass path by default */ - snd_soc_update_bits(codec, WM8731_APANA, 0x4, 0); + snd_soc_update_bits(codec, WM8731_APANA, 0x8, 0); snd_soc_add_controls(codec, wm8731_snd_controls, ARRAY_SIZE(wm8731_snd_controls)); -- cgit v1.2.3-70-g09d2 From 0ffd22b694b739b3dc3f80bc93726b581e8e8af5 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Wed, 1 Dec 2010 11:01:20 +0200 Subject: ASoC: omap: N810: Don't select CONFIG_OMAP_MUX but make it as dependency Not all omap boards use kernel based pin multiplexing so CONFIG_SND_OMAP_SOC_N810 should not select it by default as it can make harm to other boards in multi-board kernels. Therefore put CONFIG_OMAP_MUX as a dependency to N810 ASoC machine driver. Thanks to Tony Lindgren for noticing. Signed-off-by: Jarkko Nikula Cc: Tony Lindgren Acked-by: Mark Brown Acked-by: Tony Lindgren Signed-off-by: Liam Girdwood --- sound/soc/omap/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index d542ea2ff6b..a088db6d509 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -12,8 +12,8 @@ config SND_OMAP_SOC_MCPDM config SND_OMAP_SOC_N810 tristate "SoC Audio support for Nokia N810" depends on SND_OMAP_SOC && MACH_NOKIA_N810 && I2C + depends on OMAP_MUX select SND_OMAP_SOC_MCBSP - select OMAP_MUX select SND_SOC_TLV320AIC3X help Say Y if you want to add support for SoC audio on Nokia N810. -- cgit v1.2.3-70-g09d2 From ed8cc471d75365f8590c76f580def899d58028c0 Mon Sep 17 00:00:00 2001 From: Uk Kim Date: Sun, 5 Dec 2010 17:26:07 +0900 Subject: ASoC: Fix swap of left and right channels for WM8993/4 speaker boost gain SPKOUTL_BOOST start from third bit, SPKOUTLR_BOOST start from 0 bit. Signed-off-by: Uk Kim Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/wm_hubs.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 19ca782ac97..0e24092722c 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -293,7 +293,7 @@ SOC_DOUBLE_R("Speaker Switch", SOC_DOUBLE_R("Speaker ZC Switch", WM8993_SPEAKER_VOLUME_LEFT, WM8993_SPEAKER_VOLUME_RIGHT, 7, 1, 0), -SOC_DOUBLE_TLV("Speaker Boost Volume", WM8993_SPKOUT_BOOST, 0, 3, 7, 0, +SOC_DOUBLE_TLV("Speaker Boost Volume", WM8993_SPKOUT_BOOST, 3, 0, 7, 0, spkboost_tlv), SOC_ENUM("Speaker Reference", speaker_ref), SOC_ENUM("Speaker Mode", speaker_mode), -- cgit v1.2.3-70-g09d2 From 1dcb4f38e5bc28dfce0f8c7eef184a090b03bfc7 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 6 Dec 2010 16:48:03 +0800 Subject: ASoC: Hold client_mutex while calling snd_soc_instantiate_cards() As the comments of snd_soc_instantiate_cards() said, snd_soc_instantiate_cards() must be called with client_mutex. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 441285ade02..02ae7bea3b5 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3072,7 +3072,9 @@ int snd_soc_register_dais(struct device *dev, pr_debug("Registered DAI '%s'\n", dai->name); } + mutex_lock(&client_mutex); snd_soc_instantiate_cards(); + mutex_unlock(&client_mutex); return 0; err: -- cgit v1.2.3-70-g09d2 From 681e36924788aeea2717c07cc42a21c9c86d7559 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 7 Dec 2010 20:56:30 +0800 Subject: ASoC: Fix resource leak if soc_register_ac97_dai_link failed Properly free the resources in the case of soc_register_ac97_dai_link failure. Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 02ae7bea3b5..85b7d548f16 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1619,12 +1619,14 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) #ifdef CONFIG_SND_SOC_AC97_BUS /* register any AC97 codecs */ for (i = 0; i < card->num_rtd; i++) { - ret = soc_register_ac97_dai_link(&card->rtd[i]); - if (ret < 0) { - printk(KERN_ERR "asoc: failed to register AC97 %s\n", card->name); - goto probe_dai_err; - } + ret = soc_register_ac97_dai_link(&card->rtd[i]); + if (ret < 0) { + printk(KERN_ERR "asoc: failed to register AC97 %s\n", card->name); + while (--i >= 0) + soc_unregister_ac97_dai_link(&card->rtd[i]); + goto probe_dai_err; } + } #endif card->instantiated = 1; -- cgit v1.2.3-70-g09d2 From 6b464321d276e448d478c99202c19d83f2bd25f4 Mon Sep 17 00:00:00 2001 From: Jassi Brar Date: Tue, 7 Dec 2010 19:23:07 +0900 Subject: ASoC: WM8580: Debug BCLK and sample size In case of SNDRV_PCM_FORMAT_S32_LE, we need to set WM8580_AIF_LENGTH_32, rather than WM8580_AIF_LENGTH_24. Also, the BCLK has to be 64fs, for sample size of 20, 24 and 32 bits. Signed-off-by: Jassi Brar Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm8580.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index a2e0ed59b37..879dff2714d 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -491,16 +491,16 @@ static int wm8580_paif_hw_params(struct snd_pcm_substream *substream, paifa |= 0x8; break; case SNDRV_PCM_FORMAT_S20_3LE: - paifa |= 0x10; + paifa |= 0x0; paifb |= WM8580_AIF_LENGTH_20; break; case SNDRV_PCM_FORMAT_S24_LE: - paifa |= 0x10; + paifa |= 0x0; paifb |= WM8580_AIF_LENGTH_24; break; case SNDRV_PCM_FORMAT_S32_LE: - paifa |= 0x10; - paifb |= WM8580_AIF_LENGTH_24; + paifa |= 0x0; + paifb |= WM8580_AIF_LENGTH_32; break; default: return -EINVAL; -- cgit v1.2.3-70-g09d2 From 2a7b1a00206895cfa444fd83477dca67a88a9d25 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 7 Dec 2010 15:32:38 +0000 Subject: ASoC: Correct WM8962 interrupt mask register read Fix mismerge from the out of tree BSP where this support was developed. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index e8092745a20..1304ca91a11 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3339,7 +3339,7 @@ static irqreturn_t wm8962_irq(int irq, void *data) int mask; int active; - mask = snd_soc_read(codec, WM8962_INTERRUPT_STATUS_2); + mask = snd_soc_read(codec, WM8962_INTERRUPT_STATUS_2_MASK); active = snd_soc_read(codec, WM8962_INTERRUPT_STATUS_2); active &= ~mask; -- cgit v1.2.3-70-g09d2 From 3f343f8512c7882a3637d9aea4ec6b3801cbcdc5 Mon Sep 17 00:00:00 2001 From: Dmitry Artamonow Date: Wed, 8 Dec 2010 23:36:17 +0300 Subject: ASoC: fix deemphasis control in wm8904/55/60 codecs Deemphasis control's .get callback should update control's value instead of returning it - return value of callback function is used for indicating error or success of operation. Signed-off-by: Dmitry Artamonow Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/wm8904.c | 3 ++- sound/soc/codecs/wm8955.c | 3 ++- sound/soc/codecs/wm8960.c | 3 ++- 3 files changed, 6 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index fca60a0b57b..9001cc48ba1 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -818,7 +818,8 @@ static int wm8904_get_deemph(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec); - return wm8904->deemph; + ucontrol->value.enumerated.item[0] = wm8904->deemph; + return 0; } static int wm8904_put_deemph(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index f89ad6c9a80..9cbab8e1de0 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -380,7 +380,8 @@ static int wm8955_get_deemph(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct wm8955_priv *wm8955 = snd_soc_codec_get_drvdata(codec); - return wm8955->deemph; + ucontrol->value.enumerated.item[0] = wm8955->deemph; + return 0; } static int wm8955_put_deemph(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 8d5efb333c3..21986c42272 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -138,7 +138,8 @@ static int wm8960_get_deemph(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); - return wm8960->deemph; + ucontrol->value.enumerated.item[0] = wm8960->deemph; + return 0; } static int wm8960_put_deemph(struct snd_kcontrol *kcontrol, -- cgit v1.2.3-70-g09d2 From a0968628097380be52db8b4664da98fc425546a5 Mon Sep 17 00:00:00 2001 From: Seungwhan Youn Date: Thu, 9 Dec 2010 18:07:52 +0900 Subject: ASoC: WM8580: Fix R8 initial value Acc to WM8580 manual, the default value for R8 is 0x10, not 0x1c. Signed-off-by: Seungwhan Youn Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/wm8580.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 879dff2714d..8725d4e7543 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -161,7 +161,7 @@ static const u16 wm8580_reg[] = { 0x0121, 0x017e, 0x007d, 0x0014, /*R3*/ 0x0121, 0x017e, 0x007d, 0x0194, /*R7*/ - 0x001c, 0x0002, 0x0002, 0x00c2, /*R11*/ + 0x0010, 0x0002, 0x0002, 0x00c2, /*R11*/ 0x0182, 0x0082, 0x000a, 0x0024, /*R15*/ 0x0009, 0x0000, 0x00ff, 0x0000, /*R19*/ 0x00ff, 0x00ff, 0x00ff, 0x00ff, /*R23*/ -- cgit v1.2.3-70-g09d2 From 862af8adbe6b9ccb7c00c13717b1f92465f79aa2 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 10 Dec 2010 20:53:55 +0200 Subject: ASoC: Fix bias power down of non-DAPM codec Currently bias of non-DAPM codec will be powered down (standby/off) whenever there is a stream stop. This is wrong in simultaneous playback/capture since the bias is put down immediately after stopping the first stream. Fix this by using the codec->active count when figuring out the needed bias level after stream stop. Signed-off-by: Jarkko Nikula Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 75ed6491222..c721502833b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -944,6 +944,9 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) case SND_SOC_DAPM_STREAM_RESUME: sys_power = 1; break; + case SND_SOC_DAPM_STREAM_STOP: + sys_power = !!codec->active; + break; case SND_SOC_DAPM_STREAM_SUSPEND: sys_power = 0; break; -- cgit v1.2.3-70-g09d2