From 72531c9434fa884d20cb3c36fcec83752f32fdf4 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 22 Nov 2011 09:46:51 +0800 Subject: ASoC: Fix wrong define for AD1836_ADC_WORD_OFFSET According to the datasheet: The BIT[5:4] of ADC Control Register 2 is to control the word width. 00 = 25 Bits 01 = 20 Bits 10 = 16 Bits 11 = Invalid Thus, the AD1836_ADC_WORD_OFFSET should be defined as 4. Signed-off-by: Axel Lin Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/ad1836.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/ad1836.h b/sound/soc/codecs/ad1836.h index 444747f0db2..dd7be0dbbc5 100644 --- a/sound/soc/codecs/ad1836.h +++ b/sound/soc/codecs/ad1836.h @@ -34,7 +34,7 @@ #define AD1836_ADC_CTRL2 13 #define AD1836_ADC_WORD_LEN_MASK 0x30 -#define AD1836_ADC_WORD_OFFSET 5 +#define AD1836_ADC_WORD_OFFSET 4 #define AD1836_ADC_SERFMT_MASK (7 << 6) #define AD1836_ADC_SERFMT_PCK256 (0x4 << 6) #define AD1836_ADC_SERFMT_PCK128 (0x5 << 6) -- cgit v1.2.3-70-g09d2 From 5c4b2aa3fd1dc30af098de5dec766a817621ace2 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 22 Nov 2011 14:47:44 +0800 Subject: ASoC: max9877: Update register if either val or val2 is changed In the case of ((max9877_regs[reg] >> shift) & mask) != val but ((max9877_regs[reg2] >> shift) & mask) == val2, current code does not update the registers. Fix the logic to update registers if either val or val2 is changed. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/max9877.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c index 9e7e964a5fa..dcf6f2a1600 100644 --- a/sound/soc/codecs/max9877.c +++ b/sound/soc/codecs/max9877.c @@ -106,13 +106,13 @@ static int max9877_set_2reg(struct snd_kcontrol *kcontrol, unsigned int mask = mc->max; unsigned int val = (ucontrol->value.integer.value[0] & mask); unsigned int val2 = (ucontrol->value.integer.value[1] & mask); - unsigned int change = 1; + unsigned int change = 0; - if (((max9877_regs[reg] >> shift) & mask) == val) - change = 0; + if (((max9877_regs[reg] >> shift) & mask) != val) + change = 1; - if (((max9877_regs[reg2] >> shift) & mask) == val2) - change = 0; + if (((max9877_regs[reg2] >> shift) & mask) != val2) + change = 1; if (change) { max9877_regs[reg] &= ~(mask << shift); -- cgit v1.2.3-70-g09d2 From 3d94a2a53a3979c30620e3adea10f20bef8267b3 Mon Sep 17 00:00:00 2001 From: Boojin Kim Date: Tue, 22 Nov 2011 11:03:22 +0900 Subject: ASoC: SAMSUNG: Fix build error This patch adds to fix following build errors. sound/soc/codecs/wm8994.c: In function 'wm8994_readable': sound/soc/codecs/wm8994.c:58: warning: unused variable 'wm8994' sound/soc/samsung/smdk_wm8994.c:176: error: expected declaration specifiers or '...' before string constant sound/soc/samsung/smdk_wm8994.c:176: warning: data definition has no type or storage class sound/soc/samsung/smdk_wm8994.c:176: warning: type defaults to 'int' in declaration of 'MODULE_DESCRIPTION' sound/soc/samsung/smdk_wm8994.c:176: warning: function declaration isn't a prototype sound/soc/samsung/smdk_wm8994.c:177: error: expected declaration specifiers or '...' before string constant Signed-off-by: Boojin Kim Signed-off-by: Mark Brown --- sound/soc/samsung/smdk_wm8994.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index f75e43997d5..ad9ac42522e 100644 --- a/sound/soc/samsung/smdk_wm8994.c +++ b/sound/soc/samsung/smdk_wm8994.c @@ -9,6 +9,7 @@ #include "../codecs/wm8994.h" #include +#include /* * Default CFG switch settings to use this driver: -- cgit v1.2.3-70-g09d2 From d66b8537b30fbaf79e0f467fa6b7e1a2191cba83 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Tue, 22 Nov 2011 17:17:23 +0100 Subject: ASoC: cs4720: use snd_soc_cache_sync() Replace the manual register restore mechanism in cs4270.c and call the generic snd_soc_cache_sync() handler instead. This factors code out in favour of core facilities and also fixes a bus confusion that is most probably caused by intermixing i2c-regmap functions and i2c_smbus_* accessors. Signed-off-by: Daniel Mack Reported-and-tested-by: Sven Neumann Acked-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/codecs/cs4270.c | 10 +--------- 1 file changed, 1 insertion(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index f1f237ecec2..73f46eb459f 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -601,7 +601,6 @@ static int cs4270_soc_suspend(struct snd_soc_codec *codec, pm_message_t mesg) static int cs4270_soc_resume(struct snd_soc_codec *codec) { struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec); - struct i2c_client *i2c_client = to_i2c_client(codec->dev); int reg; regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies), @@ -612,14 +611,7 @@ static int cs4270_soc_resume(struct snd_soc_codec *codec) ndelay(500); /* first restore the entire register cache ... */ - for (reg = CS4270_FIRSTREG; reg <= CS4270_LASTREG; reg++) { - u8 val = snd_soc_read(codec, reg); - - if (i2c_smbus_write_byte_data(i2c_client, reg, val)) { - dev_err(codec->dev, "i2c write failed\n"); - return -EIO; - } - } + snd_soc_cache_sync(codec); /* ... then disable the power-down bits */ reg = snd_soc_read(codec, CS4270_PWRCTL); -- cgit v1.2.3-70-g09d2 From 380c88303812951f6c838241366a66a03fb5c897 Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Tue, 22 Nov 2011 14:38:59 -0600 Subject: ASoC: mpc8610: tell the CS4270 codec that it's the master Commit ac601555 ("ASoC: Return early with -EINVAL if invalid dai format is detected") requires the machine driver to tell the CS4270 codec driver whether the CS4270 should be configured for master or slave operation. Signed-off-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/fsl/mpc8610_hpcd.c | 24 ++++++++++++++++-------- 1 file changed, 16 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 31af405bda8..ae49f1c78c6 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -392,7 +392,8 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) } if (strcasecmp(sprop, "i2s-slave") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_I2S; + machine_data->dai_format = + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM; machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; @@ -409,31 +410,38 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) } machine_data->clk_frequency = be32_to_cpup(iprop); } else if (strcasecmp(sprop, "i2s-master") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_I2S; + machine_data->dai_format = + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS; machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; } else if (strcasecmp(sprop, "lj-slave") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_LEFT_J; + machine_data->dai_format = + SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBM_CFM; machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; } else if (strcasecmp(sprop, "lj-master") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_LEFT_J; + machine_data->dai_format = + SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBS_CFS; machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; } else if (strcasecmp(sprop, "rj-slave") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_RIGHT_J; + machine_data->dai_format = + SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBM_CFM; machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; } else if (strcasecmp(sprop, "rj-master") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_RIGHT_J; + machine_data->dai_format = + SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBS_CFS; machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; } else if (strcasecmp(sprop, "ac97-slave") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_AC97; + machine_data->dai_format = + SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBM_CFM; machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; } else if (strcasecmp(sprop, "ac97-master") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_AC97; + machine_data->dai_format = + SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBS_CFS; machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; } else { -- cgit v1.2.3-70-g09d2 From 4ca8af579c9748376db537575f7a811c179fe50a Mon Sep 17 00:00:00 2001 From: Paul Bolle Date: Wed, 23 Nov 2011 10:39:10 +0100 Subject: ASoC: drop support for PlayPaq with WM8510 SoC Audio support for PlayPaq with WM8510 got added in commit 9aaca9683b ("[ALSA] Revised AT32 ASoC Patch"). That support depends on BOARD_PLAYPAQ. That Kconfig symbol didn't exist when that support got added in v2.6.27. It still doesn't. It has never been possible to even build this driver. Drop it. Signed-off-by: Paul Bolle Signed-off-by: Mark Brown --- sound/soc/atmel/Kconfig | 21 +- sound/soc/atmel/Makefile | 4 - sound/soc/atmel/playpaq_wm8510.c | 473 --------------------------------------- 3 files changed, 1 insertion(+), 497 deletions(-) delete mode 100644 sound/soc/atmel/playpaq_wm8510.c (limited to 'sound') diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index bee3c94f58b..d1fcc816ce9 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -1,6 +1,6 @@ config SND_ATMEL_SOC tristate "SoC Audio for the Atmel System-on-Chip" - depends on ARCH_AT91 || AVR32 + depends on ARCH_AT91 help Say Y or M if you want to add support for codecs attached to the ATMEL SSC interface. You will also need @@ -24,25 +24,6 @@ config SND_AT91_SOC_SAM9G20_WM8731 Say Y if you want to add support for SoC audio on WM8731-based AT91sam9g20 evaluation board. -config SND_AT32_SOC_PLAYPAQ - tristate "SoC Audio support for PlayPaq with WM8510" - depends on SND_ATMEL_SOC && BOARD_PLAYPAQ && AT91_PROGRAMMABLE_CLOCKS - select SND_ATMEL_SOC_SSC - select SND_SOC_WM8510 - help - Say Y or M here if you want to add support for SoC audio - on the LRS PlayPaq. - -config SND_AT32_SOC_PLAYPAQ_SLAVE - bool "Run CODEC on PlayPaq in slave mode" - depends on SND_AT32_SOC_PLAYPAQ - default n - help - Say Y if you want to run with the AT32 SSC generating the BCLK - and FRAME signals on the PlayPaq. Unless you want to play - with the AT32 as the SSC master, you probably want to say N here, - as this will give you better sound quality. - config SND_AT91_SOC_AFEB9260 tristate "SoC Audio support for AFEB9260 board" depends on ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile index e7ea56bd5f8..a5c0bf19da7 100644 --- a/sound/soc/atmel/Makefile +++ b/sound/soc/atmel/Makefile @@ -8,9 +8,5 @@ obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o # AT91 Machine Support snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o -# AT32 Machine Support -snd-soc-playpaq-objs := playpaq_wm8510.o - obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o -obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o obj-$(CONFIG_SND_AT91_SOC_AFEB9260) += snd-soc-afeb9260.o diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c deleted file mode 100644 index 73ae99ad457..00000000000 --- a/sound/soc/atmel/playpaq_wm8510.c +++ /dev/null @@ -1,473 +0,0 @@ -/* sound/soc/at32/playpaq_wm8510.c - * ASoC machine driver for PlayPaq using WM8510 codec - * - * Copyright (C) 2008 Long Range Systems - * Geoffrey Wossum - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * - * This code is largely inspired by sound/soc/at91/eti_b1_wm8731.c - * - * NOTE: If you don't have the AT32 enhanced portmux configured (which - * isn't currently in the mainline or Atmel patched kernel), you will - * need to set the MCLK pin (PA30) to peripheral A in your board initialization - * code. Something like: - * at32_select_periph(GPIO_PIN_PA(30), GPIO_PERIPH_A, 0); - * - */ - -/* #define DEBUG */ - -#include -#include -#include -#include -#include -#include -#include -#include - -#include -#include -#include -#include - -#include -#include - -#include "../codecs/wm8510.h" -#include "atmel-pcm.h" -#include "atmel_ssc_dai.h" - - -/*-------------------------------------------------------------------------*\ - * constants -\*-------------------------------------------------------------------------*/ -#define MCLK_PIN GPIO_PIN_PA(30) -#define MCLK_PERIPH GPIO_PERIPH_A - - -/*-------------------------------------------------------------------------*\ - * data types -\*-------------------------------------------------------------------------*/ -/* SSC clocking data */ -struct ssc_clock_data { - /* CMR div */ - unsigned int cmr_div; - - /* Frame period (as needed by xCMR.PERIOD) */ - unsigned int period; - - /* The SSC clock rate these settings where calculated for */ - unsigned long ssc_rate; -}; - - -/*-------------------------------------------------------------------------*\ - * module data -\*-------------------------------------------------------------------------*/ -static struct clk *_gclk0; -static struct clk *_pll0; - -#define CODEC_CLK (_gclk0) - - -/*-------------------------------------------------------------------------*\ - * Sound SOC operations -\*-------------------------------------------------------------------------*/ -#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE -static struct ssc_clock_data playpaq_wm8510_calc_ssc_clock( - struct snd_pcm_hw_params *params, - struct snd_soc_dai *cpu_dai) -{ - struct at32_ssc_info *ssc_p = snd_soc_dai_get_drvdata(cpu_dai); - struct ssc_device *ssc = ssc_p->ssc; - struct ssc_clock_data cd; - unsigned int rate, width_bits, channels; - unsigned int bitrate, ssc_div; - unsigned actual_rate; - - - /* - * Figure out required bitrate - */ - rate = params_rate(params); - channels = params_channels(params); - width_bits = snd_pcm_format_physical_width(params_format(params)); - bitrate = rate * width_bits * channels; - - - /* - * Figure out required SSC divider and period for required bitrate - */ - cd.ssc_rate = clk_get_rate(ssc->clk); - ssc_div = cd.ssc_rate / bitrate; - cd.cmr_div = ssc_div / 2; - if (ssc_div & 1) { - /* round cmr_div up */ - cd.cmr_div++; - } - cd.period = width_bits - 1; - - - /* - * Find actual rate, compare to requested rate - */ - actual_rate = (cd.ssc_rate / (cd.cmr_div * 2)) / (2 * (cd.period + 1)); - pr_debug("playpaq_wm8510: Request rate = %u, actual rate = %u\n", - rate, actual_rate); - - - return cd; -} -#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ - - - -static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct at32_ssc_info *ssc_p = snd_soc_dai_get_drvdata(cpu_dai); - struct ssc_device *ssc = ssc_p->ssc; - unsigned int pll_out = 0, bclk = 0, mclk_div = 0; - int ret; - - - /* Due to difficulties with getting the correct clocks from the AT32's - * PLL0, we're going to let the CODEC be in charge of all the clocks - */ -#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE - const unsigned int fmt = (SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); -#else - struct ssc_clock_data cd; - const unsigned int fmt = (SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); -#endif - - if (ssc == NULL) { - pr_warning("playpaq_wm8510_hw_params: ssc is NULL!\n"); - return -EINVAL; - } - - - /* - * Figure out PLL and BCLK dividers for WM8510 - */ - switch (params_rate(params)) { - case 48000: - pll_out = 24576000; - mclk_div = WM8510_MCLKDIV_2; - bclk = WM8510_BCLKDIV_8; - break; - - case 44100: - pll_out = 22579200; - mclk_div = WM8510_MCLKDIV_2; - bclk = WM8510_BCLKDIV_8; - break; - - case 22050: - pll_out = 22579200; - mclk_div = WM8510_MCLKDIV_4; - bclk = WM8510_BCLKDIV_8; - break; - - case 16000: - pll_out = 24576000; - mclk_div = WM8510_MCLKDIV_6; - bclk = WM8510_BCLKDIV_8; - break; - - case 11025: - pll_out = 22579200; - mclk_div = WM8510_MCLKDIV_8; - bclk = WM8510_BCLKDIV_8; - break; - - case 8000: - pll_out = 24576000; - mclk_div = WM8510_MCLKDIV_12; - bclk = WM8510_BCLKDIV_8; - break; - - default: - pr_warning("playpaq_wm8510: Unsupported sample rate %d\n", - params_rate(params)); - return -EINVAL; - } - - - /* - * set CPU and CODEC DAI configuration - */ - ret = snd_soc_dai_set_fmt(codec_dai, fmt); - if (ret < 0) { - pr_warning("playpaq_wm8510: " - "Failed to set CODEC DAI format (%d)\n", - ret); - return ret; - } - ret = snd_soc_dai_set_fmt(cpu_dai, fmt); - if (ret < 0) { - pr_warning("playpaq_wm8510: " - "Failed to set CPU DAI format (%d)\n", - ret); - return ret; - } - - - /* - * Set CPU clock configuration - */ -#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE - cd = playpaq_wm8510_calc_ssc_clock(params, cpu_dai); - pr_debug("playpaq_wm8510: cmr_div = %d, period = %d\n", - cd.cmr_div, cd.period); - ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_CMR_DIV, cd.cmr_div); - if (ret < 0) { - pr_warning("playpaq_wm8510: Failed to set CPU CMR_DIV (%d)\n", - ret); - return ret; - } - ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_TCMR_PERIOD, - cd.period); - if (ret < 0) { - pr_warning("playpaq_wm8510: " - "Failed to set CPU transmit period (%d)\n", - ret); - return ret; - } -#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ - - - /* - * Set CODEC clock configuration - */ - pr_debug("playpaq_wm8510: " - "pll_in = %ld, pll_out = %u, bclk = %x, mclk = %x\n", - clk_get_rate(CODEC_CLK), pll_out, bclk, mclk_div); - - -#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_BCLKDIV, bclk); - if (ret < 0) { - pr_warning - ("playpaq_wm8510: Failed to set CODEC DAI BCLKDIV (%d)\n", - ret); - return ret; - } -#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ - - - ret = snd_soc_dai_set_pll(codec_dai, 0, 0, - clk_get_rate(CODEC_CLK), pll_out); - if (ret < 0) { - pr_warning("playpaq_wm8510: Failed to set CODEC DAI PLL (%d)\n", - ret); - return ret; - } - - - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_MCLKDIV, mclk_div); - if (ret < 0) { - pr_warning("playpaq_wm8510: Failed to set CODEC MCLKDIV (%d)\n", - ret); - return ret; - } - - - return 0; -} - - - -static struct snd_soc_ops playpaq_wm8510_ops = { - .hw_params = playpaq_wm8510_hw_params, -}; - - - -static const struct snd_soc_dapm_widget playpaq_dapm_widgets[] = { - SND_SOC_DAPM_MIC("Int Mic", NULL), - SND_SOC_DAPM_SPK("Ext Spk", NULL), -}; - - - -static const struct snd_soc_dapm_route intercon[] = { - /* speaker connected to SPKOUT */ - {"Ext Spk", NULL, "SPKOUTP"}, - {"Ext Spk", NULL, "SPKOUTN"}, - - {"Mic Bias", NULL, "Int Mic"}, - {"MICN", NULL, "Mic Bias"}, - {"MICP", NULL, "Mic Bias"}, -}; - - - -static int playpaq_wm8510_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - int i; - - /* - * Add DAPM widgets - */ - for (i = 0; i < ARRAY_SIZE(playpaq_dapm_widgets); i++) - snd_soc_dapm_new_control(dapm, &playpaq_dapm_widgets[i]); - - - - /* - * Setup audio path interconnects - */ - snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); - - - - /* always connected pins */ - snd_soc_dapm_enable_pin(dapm, "Int Mic"); - snd_soc_dapm_enable_pin(dapm, "Ext Spk"); - - - - /* Make CSB show PLL rate */ - snd_soc_dai_set_clkdiv(rtd->codec_dai, WM8510_OPCLKDIV, - WM8510_OPCLKDIV_1 | 4); - - return 0; -} - - - -static struct snd_soc_dai_link playpaq_wm8510_dai = { - .name = "WM8510", - .stream_name = "WM8510 PCM", - .cpu_dai_name= "atmel-ssc-dai.0", - .platform_name = "atmel-pcm-audio", - .codec_name = "wm8510-codec.0-0x1a", - .codec_dai_name = "wm8510-hifi", - .init = playpaq_wm8510_init, - .ops = &playpaq_wm8510_ops, -}; - - - -static struct snd_soc_card snd_soc_playpaq = { - .name = "LRS_PlayPaq_WM8510", - .dai_link = &playpaq_wm8510_dai, - .num_links = 1, -}; - -static struct platform_device *playpaq_snd_device; - - -static int __init playpaq_asoc_init(void) -{ - int ret = 0; - - /* - * Configure MCLK for WM8510 - */ - _gclk0 = clk_get(NULL, "gclk0"); - if (IS_ERR(_gclk0)) { - _gclk0 = NULL; - ret = PTR_ERR(_gclk0); - goto err_gclk0; - } - _pll0 = clk_get(NULL, "pll0"); - if (IS_ERR(_pll0)) { - _pll0 = NULL; - ret = PTR_ERR(_pll0); - goto err_pll0; - } - ret = clk_set_parent(_gclk0, _pll0); - if (ret) { - pr_warning("snd-soc-playpaq: " - "Failed to set PLL0 as parent for DAC clock\n"); - goto err_set_clk; - } - clk_set_rate(CODEC_CLK, 12000000); - clk_enable(CODEC_CLK); - -#if defined CONFIG_AT32_ENHANCED_PORTMUX - at32_select_periph(MCLK_PIN, MCLK_PERIPH, 0); -#endif - - - /* - * Create and register platform device - */ - playpaq_snd_device = platform_device_alloc("soc-audio", 0); - if (playpaq_snd_device == NULL) { - ret = -ENOMEM; - goto err_device_alloc; - } - - platform_set_drvdata(playpaq_snd_device, &snd_soc_playpaq); - - ret = platform_device_add(playpaq_snd_device); - if (ret) { - pr_warning("playpaq_wm8510: platform_device_add failed (%d)\n", - ret); - goto err_device_add; - } - - return 0; - - -err_device_add: - if (playpaq_snd_device != NULL) { - platform_device_put(playpaq_snd_device); - playpaq_snd_device = NULL; - } -err_device_alloc: -err_set_clk: - if (_pll0 != NULL) { - clk_put(_pll0); - _pll0 = NULL; - } -err_pll0: - if (_gclk0 != NULL) { - clk_put(_gclk0); - _gclk0 = NULL; - } - return ret; -} - - -static void __exit playpaq_asoc_exit(void) -{ - if (_gclk0 != NULL) { - clk_put(_gclk0); - _gclk0 = NULL; - } - if (_pll0 != NULL) { - clk_put(_pll0); - _pll0 = NULL; - } - -#if defined CONFIG_AT32_ENHANCED_PORTMUX - at32_free_pin(MCLK_PIN); -#endif - - platform_device_unregister(playpaq_snd_device); - playpaq_snd_device = NULL; -} - -module_init(playpaq_asoc_init); -module_exit(playpaq_asoc_exit); - -MODULE_AUTHOR("Geoffrey Wossum "); -MODULE_DESCRIPTION("ASoC machine driver for LRS PlayPaq"); -MODULE_LICENSE("GPL"); -- cgit v1.2.3-70-g09d2 From b284362b6b45150d171ff5bed92bc416b040aead Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 23 Nov 2011 12:46:11 +0800 Subject: ASoC: cs42l51: Fix off-by-one for reg_cache_size Just checking the code in cs42l51_fill_cache(): The cache pointer points to codec->reg_cache + 1. I think it is because CS42L51_FIRSTREG is 0x01, so codec->reg_cache[0] is not used here. Then we read CS42L51_NUMREGS bytes to cache. So we need reg_cache_size to be CS42L51_NUMREGS + 1. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l51.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 8c3c8205d19..1ee66361f61 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -555,7 +555,7 @@ static int cs42l51_probe(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_device_cs42l51 = { .probe = cs42l51_probe, - .reg_cache_size = CS42L51_NUMREGS, + .reg_cache_size = CS42L51_NUMREGS + 1, .reg_word_size = sizeof(u8), }; -- cgit v1.2.3-70-g09d2 From 5ff1ddf22b2584d00d7e0ba5a8eab07b5338bd84 Mon Sep 17 00:00:00 2001 From: Eric Miao Date: Wed, 23 Nov 2011 22:37:00 +0800 Subject: ASoC: skip resume of soc-audio devices without codecs There are cases where there is no working codec on the soc-audio devices, and snd_soc_suspend() will skip such device when suspending. Yet its counterpart snd_soc_resume() does not check this, causing complaints about spinlock lockup: [ 176.726087] BUG: spinlock lockup on CPU#0, kworker/0:2/1067, d8ab82a8 [ 176.732539] [<80014a14>] (unwind_backtrace+0x0/0xec) from [<805b3fc8>] (dump_stack+0x20/0x24) [ 176.741082] [<805b3fc8>] (dump_stack+0x20/0x24) from [<80322208>] (do_raw_spin_lock+0x118/0x158) [ 176.749882] [<80322208>] (do_raw_spin_lock+0x118/0x158) from [<805b7874>] (_raw_spin_lock_irqsave+0x5c/0x68) [ 176.759723] [<805b7874>] (_raw_spin_lock_irqsave+0x5c/0x68) from [<8002a020>] (__wake_up+0x2c/0x5c) [ 176.768781] [<8002a020>] (__wake_up+0x2c/0x5c) from [<804a6de8>] (soc_resume_deferred+0x3c/0x2b0) [ 176.777666] [<804a6de8>] (soc_resume_deferred+0x3c/0x2b0) from [<8004ee20>] (process_one_work+0x2e8/0x50c) [ 176.787334] [<8004ee20>] (process_one_work+0x2e8/0x50c) from [<8004fd08>] (worker_thread+0x1c8/0x2e0) [ 176.796566] [<8004fd08>] (worker_thread+0x1c8/0x2e0) from [<80053ec8>] (kthread+0xa4/0xb0) [ 176.804843] [<80053ec8>] (kthread+0xa4/0xb0) from [<8000ea70>] (kernel_thread_exit+0x0/0x8) Signed-off-by: Eric Miao Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a5d3685a5d3..a25fa63ce9a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -709,6 +709,12 @@ int snd_soc_resume(struct device *dev) struct snd_soc_card *card = dev_get_drvdata(dev); int i, ac97_control = 0; + /* If the initialization of this soc device failed, there is no codec + * associated with it. Just bail out in this case. + */ + if (list_empty(&card->codec_dev_list)) + return 0; + /* AC97 devices might have other drivers hanging off them so * need to resume immediately. Other drivers don't have that * problem and may take a substantial amount of time to resume -- cgit v1.2.3-70-g09d2 From 5b895eec118ab5fec7b69102d73c1b04a86140b9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 27 Nov 2011 15:54:49 +0000 Subject: ASoC: Correct name of Speyside Main Speaker widget Signed-off-by: Mark Brown --- sound/soc/samsung/speyside.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index 85bf541a771..4b8e35410eb 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -191,7 +191,7 @@ static int speyside_late_probe(struct snd_soc_card *card) snd_soc_dapm_ignore_suspend(&card->dapm, "Headset Mic"); snd_soc_dapm_ignore_suspend(&card->dapm, "Main AMIC"); snd_soc_dapm_ignore_suspend(&card->dapm, "Main DMIC"); - snd_soc_dapm_ignore_suspend(&card->dapm, "Speaker"); + snd_soc_dapm_ignore_suspend(&card->dapm, "Main Speaker"); snd_soc_dapm_ignore_suspend(&card->dapm, "WM1250 Output"); snd_soc_dapm_ignore_suspend(&card->dapm, "WM1250 Input"); -- cgit v1.2.3-70-g09d2 From fc07ecd851bd082265b52838eff12f50b88f6114 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 28 Nov 2011 21:16:56 +0000 Subject: ASoC: Error out if we can't generate a LRCLK at all for WM8994 Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 9c982e47eb9..36ba1edfff8 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2357,6 +2357,11 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, bclk |= best << WM8994_AIF1_BCLK_DIV_SHIFT; lrclk = bclk_rate / params_rate(params); + if (!lrclk) { + dev_err(dai->dev, "Unable to generate LRCLK from %dHz BCLK\n", + bclk_rate); + return -EINVAL; + } dev_dbg(dai->dev, "Using LRCLK rate %d for actual LRCLK %dHz\n", lrclk, bclk_rate / lrclk); -- cgit v1.2.3-70-g09d2 From fc8e6e8668e74fbf8e00d6e143d7f43b20f73f32 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 28 Nov 2011 18:48:46 +0000 Subject: ASoC: Supply dcs_codes for newer WM1811 revisions Based on initial data. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 36ba1edfff8..6c298854900 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3183,6 +3183,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) switch (wm8994->revision) { case 0: case 1: + case 2: + case 3: wm8994->hubs.dcs_codes_l = -9; wm8994->hubs.dcs_codes_r = -5; break; -- cgit v1.2.3-70-g09d2 From ae7cc709f2ec11b49fc31b20cd8c943794ae9576 Mon Sep 17 00:00:00 2001 From: John F Leach Date: Mon, 28 Nov 2011 19:41:27 -0500 Subject: ALSA: usb-audio - Support for Roland GAIA SH-01 Synthesizer Added table quirks entry for Roland GAIA SH-01 Synthesizer based upon Roland SH-201 table entry as template. USB MIDI and audio was tested with Muse and Audacity. Signed-off-by: John F Leach Signed-off-by: Takashi Iwai --- sound/usb/quirks-table.h | 31 +++++++++++++++++++++++++++++++ 1 file changed, 31 insertions(+) (limited to 'sound') diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index b61945f3af9..32d2a21f2e3 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1632,6 +1632,37 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + /* Roland GAIA SH-01 */ + USB_DEVICE(0x0582, 0x0111), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Roland", + .product_name = "GAIA", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = &(const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0003, + .in_cables = 0x0003 + } + }, + { + .ifnum = -1 + } + } + } +}, { USB_DEVICE(0x0582, 0x0113), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { -- cgit v1.2.3-70-g09d2 From 542c9a0a2fa351149c4a3467589a54cafcf0a1dd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 29 Nov 2011 13:01:30 +0100 Subject: ALSA: hda - Avoid touching mute-VREF pin for IDT codecs Some HP laptops use a pin VREF for controlling the mute LED, and such a pin shouldn't be powered off. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index f3658658548..f4f4ebeed9e 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4441,7 +4441,9 @@ static int stac92xx_init(struct hda_codec *codec) int pinctl, def_conf; /* power on when no jack detection is available */ - if (!spec->hp_detect) { + /* or when the VREF is used for controlling LED */ + if (!spec->hp_detect || + (spec->gpio_led > 8 && spec->gpio_led == nid)) { stac_toggle_power_map(codec, nid, 1); continue; } -- cgit v1.2.3-70-g09d2 From 4f8b6c7dc80ac9619db033c7f6fc355eab9514f5 Mon Sep 17 00:00:00 2001 From: Marc Vertes Date: Tue, 29 Nov 2011 12:21:17 +0100 Subject: ALSA: hda_intel - revert a quirk that affect VIA chipsets This quirk sould be reverted. It has the following probems: 1) The quirk was intended to "ASUS MV2-MX SE" motherboards only, but the ID used matches a much broader range, potentially all boards containing a VIA chipset model in the family of vendor VIA 0x1106 and audio device ID 0x3288, which encompasses VIA-VT82xx, VIA-VT1xx and VIA-VT20xx chipsets. 2) VIA chipsets rely on azx_via_get_position() to handle correctly dma transfers during capture. Using POS_FIX_LPIB instead of POS_FIX_VIACOMBO leads to partially corrupted input buffers during capture. The effects of this bug are not immediately visible, it took strong DSP expertise, some expensive signal generator and a spectrum analyzer to identify it and verify correct behaviour using original default. 3) It's almost certain that the quirk did not fix the real problem, if there was one. Refer to original submission: http://mailman.alsa-project.org/pipermail/alsa-devel/2010-February/025109.html Signed-of-by: Marc Vertes Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 096507d2ca9..7d98240def0 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2508,7 +2508,6 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB), SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB), - SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB), SND_PCI_QUIRK(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB), SND_PCI_QUIRK(0x1458, 0xa022, "ga-ma770-ud3", POS_FIX_LPIB), SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB), -- cgit v1.2.3-70-g09d2 From 88d686027bb43f585914c77dd363f6e817b42c2a Mon Sep 17 00:00:00 2001 From: Charles Chin Date: Thu, 1 Dec 2011 11:21:00 +0100 Subject: ALSA: hda - Fix S3/S4 problem on machines with VREF-pin mute-LED The verb command in stac92xx_post_suspend caused the audio to stop working after resuming from S3 mode on HP laptops with the VREF-pin mute-LED control. Removing relevant post_suspend registering. Although removing D3 on AFG is no optimal solution, the impact should be small in comparison with the broken S3/S4. Signed-off-by: Charles Chin Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 18 ------------------ 1 file changed, 18 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index f4f4ebeed9e..d8d2f9dccd9 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5057,20 +5057,6 @@ static int stac92xx_pre_resume(struct hda_codec *codec) return 0; } -static int stac92xx_post_suspend(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - if (spec->gpio_led > 8) { - /* with vref-out pin used for mute led control - * codec AFG is prevented from D3 state, but on - * system suspend it can (and should) be used - */ - snd_hda_codec_read(codec, codec->afg, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - } - return 0; -} - static void stac92xx_set_power_state(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state) { @@ -5670,8 +5656,6 @@ again: } else { codec->patch_ops.set_power_state = stac92xx_set_power_state; - codec->patch_ops.post_suspend = - stac92xx_post_suspend; } codec->patch_ops.pre_resume = stac92xx_pre_resume; codec->patch_ops.check_power_status = @@ -5985,8 +5969,6 @@ again: } else { codec->patch_ops.set_power_state = stac92xx_set_power_state; - codec->patch_ops.post_suspend = - stac92xx_post_suspend; } codec->patch_ops.pre_resume = stac92xx_pre_resume; codec->patch_ops.check_power_status = -- cgit v1.2.3-70-g09d2