diff options
Diffstat (limited to 'sound/soc')
41 files changed, 216 insertions, 573 deletions
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 <gwossum@acm.org> - * - * 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 <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/kernel.h> -#include <linux/errno.h> -#include <linux/clk.h> -#include <linux/timer.h> -#include <linux/interrupt.h> -#include <linux/platform_device.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include <mach/at32ap700x.h> -#include <mach/portmux.h> - -#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 <gwossum@acm.org>"); -MODULE_DESCRIPTION("ASoC machine driver for LRS PlayPaq"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 4584514d93d..fa787d45d74 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -33,7 +33,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CX20442 select SND_SOC_DA7210 if I2C select SND_SOC_DFBMCS320 - select SND_SOC_JZ4740_CODEC if SOC_JZ4740 + select SND_SOC_JZ4740_CODEC select SND_SOC_LM4857 if I2C select SND_SOC_MAX98088 if I2C select SND_SOC_MAX98095 if I2C 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) diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 1ccf8dd4757..45c63028b40 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -245,7 +245,7 @@ static const char *adau1373_bass_hpf_cutoff_text[] = { }; static const unsigned int adau1373_bass_tlv[] = { - TLV_DB_RANGE_HEAD(4), + TLV_DB_RANGE_HEAD(3), 0, 2, TLV_DB_SCALE_ITEM(-600, 600, 1), 3, 4, TLV_DB_SCALE_ITEM(950, 250, 0), 5, 7, TLV_DB_SCALE_ITEM(1400, 150, 0), 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); diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 23d1bd5dadd..69fde1506fe 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -434,7 +434,8 @@ static int cs4271_soc_suspend(struct snd_soc_codec *codec, pm_message_t mesg) { int ret; /* Set power-down bit */ - ret = snd_soc_update_bits(codec, CS4271_MODE2, 0, CS4271_MODE2_PDN); + ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, + CS4271_MODE2_PDN); if (ret < 0) return ret; return 0; @@ -501,8 +502,9 @@ static int cs4271_probe(struct snd_soc_codec *codec) return ret; } - ret = snd_soc_update_bits(codec, CS4271_MODE2, 0, - CS4271_MODE2_PDN | CS4271_MODE2_CPEN); + ret = snd_soc_update_bits(codec, CS4271_MODE2, + CS4271_MODE2_PDN | CS4271_MODE2_CPEN, + CS4271_MODE2_PDN | CS4271_MODE2_CPEN); if (ret < 0) return ret; ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); 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), }; diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c index e373f8f0690..3e1f4e172bf 100644 --- a/sound/soc/codecs/jz4740.c +++ b/sound/soc/codecs/jz4740.c @@ -15,6 +15,7 @@ #include <linux/module.h> #include <linux/platform_device.h> #include <linux/slab.h> +#include <linux/io.h> #include <linux/delay.h> 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); diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index 27a078cbb6e..4646e808b90 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -177,7 +177,7 @@ static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -95625, 375, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); /* {0, +20, +24, +30, +35, +40, +44, +50, +52}dB */ static unsigned int mic_bst_tlv[] = { - TLV_DB_RANGE_HEAD(6), + TLV_DB_RANGE_HEAD(7), 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), 2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0), diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index d15695d1c27..bbcf921166f 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -365,7 +365,7 @@ static const DECLARE_TLV_DB_SCALE(capture_6db_attenuate, -600, 600, 0); /* tlv for mic gain, 0db 20db 30db 40db */ static const unsigned int mic_gain_tlv[] = { - TLV_DB_RANGE_HEAD(4), + TLV_DB_RANGE_HEAD(2), 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), 1, 3, TLV_DB_SCALE_ITEM(2000, 1000, 0), }; diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index bb82408ab8e..d2f37152f94 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -76,6 +76,8 @@ struct sta32x_priv { unsigned int mclk; unsigned int format; + + u32 coef_shadow[STA32X_COEF_COUNT]; }; static const DECLARE_TLV_DB_SCALE(mvol_tlv, -12700, 50, 1); @@ -227,6 +229,7 @@ static int sta32x_coefficient_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); int numcoef = kcontrol->private_value >> 16; int index = kcontrol->private_value & 0xffff; unsigned int cfud; @@ -239,6 +242,11 @@ static int sta32x_coefficient_put(struct snd_kcontrol *kcontrol, snd_soc_write(codec, STA32X_CFUD, cfud); snd_soc_write(codec, STA32X_CFADDR2, index); + for (i = 0; i < numcoef && (index + i < STA32X_COEF_COUNT); i++) + sta32x->coef_shadow[index + i] = + (ucontrol->value.bytes.data[3 * i] << 16) + | (ucontrol->value.bytes.data[3 * i + 1] << 8) + | (ucontrol->value.bytes.data[3 * i + 2]); for (i = 0; i < 3 * numcoef; i++) snd_soc_write(codec, STA32X_B1CF1 + i, ucontrol->value.bytes.data[i]); @@ -252,6 +260,48 @@ static int sta32x_coefficient_put(struct snd_kcontrol *kcontrol, return 0; } +int sta32x_sync_coef_shadow(struct snd_soc_codec *codec) +{ + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + unsigned int cfud; + int i; + + /* preserve reserved bits in STA32X_CFUD */ + cfud = snd_soc_read(codec, STA32X_CFUD) & 0xf0; + + for (i = 0; i < STA32X_COEF_COUNT; i++) { + snd_soc_write(codec, STA32X_CFADDR2, i); + snd_soc_write(codec, STA32X_B1CF1, + (sta32x->coef_shadow[i] >> 16) & 0xff); + snd_soc_write(codec, STA32X_B1CF2, + (sta32x->coef_shadow[i] >> 8) & 0xff); + snd_soc_write(codec, STA32X_B1CF3, + (sta32x->coef_shadow[i]) & 0xff); + /* chip documentation does not say if the bits are + * self-clearing, so do it explicitly */ + snd_soc_write(codec, STA32X_CFUD, cfud); + snd_soc_write(codec, STA32X_CFUD, cfud | 0x01); + } + return 0; +} + +int sta32x_cache_sync(struct snd_soc_codec *codec) +{ + unsigned int mute; + int rc; + + if (!codec->cache_sync) + return 0; + + /* mute during register sync */ + mute = snd_soc_read(codec, STA32X_MMUTE); + snd_soc_write(codec, STA32X_MMUTE, mute | STA32X_MMUTE_MMUTE); + sta32x_sync_coef_shadow(codec); + rc = snd_soc_cache_sync(codec); + snd_soc_write(codec, STA32X_MMUTE, mute); + return rc; +} + #define SINGLE_COEF(xname, index) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = sta32x_coefficient_info, \ @@ -661,7 +711,7 @@ static int sta32x_set_bias_level(struct snd_soc_codec *codec, return ret; } - snd_soc_cache_sync(codec); + sta32x_cache_sync(codec); } /* Power up to mute */ @@ -790,6 +840,17 @@ static int sta32x_probe(struct snd_soc_codec *codec) STA32X_CxCFG_OM_MASK, 2 << STA32X_CxCFG_OM_SHIFT); + /* initialize coefficient shadow RAM with reset values */ + for (i = 4; i <= 49; i += 5) + sta32x->coef_shadow[i] = 0x400000; + for (i = 50; i <= 54; i++) + sta32x->coef_shadow[i] = 0x7fffff; + sta32x->coef_shadow[55] = 0x5a9df7; + sta32x->coef_shadow[56] = 0x7fffff; + sta32x->coef_shadow[59] = 0x7fffff; + sta32x->coef_shadow[60] = 0x400000; + sta32x->coef_shadow[61] = 0x400000; + sta32x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Bias level configuration will have done an extra enable */ regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); diff --git a/sound/soc/codecs/sta32x.h b/sound/soc/codecs/sta32x.h index b97ee5a7566..d8e32a6262e 100644 --- a/sound/soc/codecs/sta32x.h +++ b/sound/soc/codecs/sta32x.h @@ -19,6 +19,7 @@ /* STA326 register addresses */ #define STA32X_REGISTER_COUNT 0x2d +#define STA32X_COEF_COUNT 62 #define STA32X_CONFA 0x00 #define STA32X_CONFB 0x01 diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index c5ca8cfea60..0441893e270 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -863,13 +863,13 @@ static struct i2c_driver uda1380_i2c_driver = { static int __init uda1380_modinit(void) { - int ret; + int ret = 0; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) ret = i2c_add_driver(&uda1380_i2c_driver); if (ret != 0) pr_err("Failed to register UDA1380 I2C driver: %d\n", ret); #endif - return 0; + return ret; } module_init(uda1380_modinit); diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 7e5ec03f6f8..a7c9ae17fc7 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -453,6 +453,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, snd_soc_write(codec, WM8731_PWR, 0xffff); regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); + codec->cache_sync = 1; break; } codec->dapm.bias_level = level; diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index a9504710bb6..3a629d0d690 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -190,6 +190,9 @@ static int wm8753_set_dai(struct snd_kcontrol *kcontrol, struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); u16 ioctl; + if (wm8753->dai_func == ucontrol->value.integer.value[0]) + return 0; + if (codec->active) return -EBUSY; diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index 0293763debe..5a14d5c0e0e 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -60,6 +60,8 @@ static int wm8958_dsp2_fw(struct snd_soc_codec *codec, const char *name, } if (memcmp(fw->data, "WMFW", 4) != 0) { + memcpy(&data32, fw->data, sizeof(data32)); + data32 = be32_to_cpu(data32); dev_err(codec->dev, "%s: firmware has bad file magic %08x\n", name, data32); goto err; diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 91d3c6dbeba..53edd9a8c75 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -1973,7 +1973,7 @@ static int wm8962_reset(struct snd_soc_codec *codec) static const DECLARE_TLV_DB_SCALE(inpga_tlv, -2325, 75, 0); static const DECLARE_TLV_DB_SCALE(mixin_tlv, -1500, 300, 0); static const unsigned int mixinpga_tlv[] = { - TLV_DB_RANGE_HEAD(7), + TLV_DB_RANGE_HEAD(5), 0, 1, TLV_DB_SCALE_ITEM(0, 600, 0), 2, 2, TLV_DB_SCALE_ITEM(1300, 1300, 0), 3, 4, TLV_DB_SCALE_ITEM(1800, 200, 0), @@ -1988,7 +1988,7 @@ static const DECLARE_TLV_DB_SCALE(bypass_tlv, -1500, 300, 0); static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1); static const DECLARE_TLV_DB_SCALE(hp_tlv, -700, 100, 0); static const unsigned int classd_tlv[] = { - TLV_DB_RANGE_HEAD(7), + TLV_DB_RANGE_HEAD(2), 0, 6, TLV_DB_SCALE_ITEM(0, 150, 0), 7, 7, TLV_DB_SCALE_ITEM(1200, 0, 0), }; diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index eec8e143511..d1a142f48b0 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -512,7 +512,7 @@ static const DECLARE_TLV_DB_SCALE(drc_comp_threash, -4500, 75, 0); static const DECLARE_TLV_DB_SCALE(drc_comp_amp, -2250, 75, 0); static const DECLARE_TLV_DB_SCALE(drc_min_tlv, -1800, 600, 0); static const unsigned int drc_max_tlv[] = { - TLV_DB_RANGE_HEAD(4), + TLV_DB_RANGE_HEAD(2), 0, 2, TLV_DB_SCALE_ITEM(1200, 600, 0), 3, 3, TLV_DB_SCALE_ITEM(3600, 0, 0), }; diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 6b73efd2699..d0c545b73d7 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -56,7 +56,7 @@ static int wm8994_retune_mobile_base[] = { static int wm8994_readable(struct snd_soc_codec *codec, unsigned int reg) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - struct wm8994 *control = wm8994->control_data; + struct wm8994 *control = codec->control_data; switch (reg) { case WM8994_GPIO_1: @@ -1325,15 +1325,15 @@ SND_SOC_DAPM_DAC("DAC1R", NULL, WM8994_POWER_MANAGEMENT_5, 0, 0), }; static const struct snd_soc_dapm_widget wm8994_adc_revd_widgets[] = { -SND_SOC_DAPM_MUX_E("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux, - adc_mux_ev, SND_SOC_DAPM_PRE_PMU), -SND_SOC_DAPM_MUX_E("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux, - adc_mux_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_VIRT_MUX_E("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux, + adc_mux_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_VIRT_MUX_E("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux, + adc_mux_ev, SND_SOC_DAPM_PRE_PMU), }; static const struct snd_soc_dapm_widget wm8994_adc_widgets[] = { -SND_SOC_DAPM_MUX("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux), -SND_SOC_DAPM_MUX("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux), +SND_SOC_DAPM_VIRT_MUX("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux), +SND_SOC_DAPM_VIRT_MUX("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux), }; static const struct snd_soc_dapm_widget wm8994_dapm_widgets[] = { @@ -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); @@ -3030,19 +3035,34 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) { struct wm8994_priv *wm8994 = data; struct snd_soc_codec *codec = wm8994->codec; - int reg; + int reg, count; - reg = snd_soc_read(codec, WM8958_MIC_DETECT_3); - if (reg < 0) { - dev_err(codec->dev, "Failed to read mic detect status: %d\n", - reg); - return IRQ_NONE; - } + /* We may occasionally read a detection without an impedence + * range being provided - if that happens loop again. + */ + count = 10; + do { + reg = snd_soc_read(codec, WM8958_MIC_DETECT_3); + if (reg < 0) { + dev_err(codec->dev, + "Failed to read mic detect status: %d\n", + reg); + return IRQ_NONE; + } - if (!(reg & WM8958_MICD_VALID)) { - dev_dbg(codec->dev, "Mic detect data not valid\n"); - goto out; - } + if (!(reg & WM8958_MICD_VALID)) { + dev_dbg(codec->dev, "Mic detect data not valid\n"); + goto out; + } + + if (!(reg & WM8958_MICD_STS) || (reg & WM8958_MICD_LVL_MASK)) + break; + + msleep(1); + } while (count--); + + if (count == 0) + dev_warn(codec->dev, "No impedence range reported for jack\n"); #ifndef CONFIG_SND_SOC_WM8994_MODULE trace_snd_soc_jack_irq(dev_name(codec->dev)); @@ -3163,6 +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; @@ -3180,9 +3202,9 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994_request_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, wm8994_fifo_error, "FIFO error", codec); - wm8994_request_irq(wm8994->control_data, WM8994_IRQ_TEMP_WARN, + wm8994_request_irq(codec->control_data, WM8994_IRQ_TEMP_WARN, wm8994_temp_warn, "Thermal warning", codec); - wm8994_request_irq(wm8994->control_data, WM8994_IRQ_TEMP_SHUT, + wm8994_request_irq(codec->control_data, WM8994_IRQ_TEMP_SHUT, wm8994_temp_shut, "Thermal shutdown", codec); ret = wm8994_request_irq(codec->control_data, WM8994_IRQ_DCS_DONE, diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 645c980d6b8..a33b04d1719 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -1968,6 +1968,7 @@ static int wm8996_set_sysclk(struct snd_soc_dai *dai, break; case 24576000: ratediv = WM8996_SYSCLK_DIV; + wm8996->sysclk /= 2; case 12288000: snd_soc_update_bits(codec, WM8996_AIF_RATE, WM8996_SYSCLK_RATE, WM8996_SYSCLK_RATE); diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 3cd35a02c28..4a398c3bfe8 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -807,7 +807,6 @@ static int wm9081_set_bias_level(struct snd_soc_codec *codec, mdelay(100); /* Normal bias enable & soft start off */ - reg |= WM9081_BIAS_ENA; reg &= ~WM9081_VMID_RAMP; snd_soc_write(codec, WM9081_VMID_CONTROL, reg); @@ -818,7 +817,7 @@ static int wm9081_set_bias_level(struct snd_soc_codec *codec, } /* VMID 2*240k */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); + reg = snd_soc_read(codec, WM9081_VMID_CONTROL); reg &= ~WM9081_VMID_SEL_MASK; reg |= 0x04; snd_soc_write(codec, WM9081_VMID_CONTROL, reg); @@ -830,14 +829,15 @@ static int wm9081_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_OFF: - /* Startup bias source */ + /* Startup bias source and disable bias */ reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); reg |= WM9081_BIAS_SRC; + reg &= ~WM9081_BIAS_ENA; snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); - /* Disable VMID and biases with soft ramping */ + /* Disable VMID with soft ramping */ reg = snd_soc_read(codec, WM9081_VMID_CONTROL); - reg &= ~(WM9081_VMID_SEL_MASK | WM9081_BIAS_ENA); + reg &= ~WM9081_VMID_SEL_MASK; reg |= WM9081_VMID_RAMP; snd_soc_write(codec, WM9081_VMID_CONTROL, reg); diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index 2b5252c9e37..f94c06057c6 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -177,19 +177,19 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec) } static const unsigned int in_tlv[] = { - TLV_DB_RANGE_HEAD(6), + TLV_DB_RANGE_HEAD(3), 0, 0, TLV_DB_SCALE_ITEM(-600, 0, 0), 1, 3, TLV_DB_SCALE_ITEM(-350, 350, 0), 4, 6, TLV_DB_SCALE_ITEM(600, 600, 0), }; static const unsigned int mix_tlv[] = { - TLV_DB_RANGE_HEAD(4), + TLV_DB_RANGE_HEAD(2), 0, 2, TLV_DB_SCALE_ITEM(-1200, 300, 0), 3, 3, TLV_DB_SCALE_ITEM(0, 0, 0), }; static const DECLARE_TLV_DB_SCALE(out_tlv, -5700, 100, 0); static const unsigned int spkboost_tlv[] = { - TLV_DB_RANGE_HEAD(7), + TLV_DB_RANGE_HEAD(2), 0, 6, TLV_DB_SCALE_ITEM(0, 150, 0), 7, 7, TLV_DB_SCALE_ITEM(1200, 0, 0), }; diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 84f33d4ea2c..48e61e91240 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -40,7 +40,7 @@ static const DECLARE_TLV_DB_SCALE(outmix_tlv, -2100, 300, 0); static const DECLARE_TLV_DB_SCALE(spkmixout_tlv, -1800, 600, 1); static const DECLARE_TLV_DB_SCALE(outpga_tlv, -5700, 100, 0); static const unsigned int spkboost_tlv[] = { - TLV_DB_RANGE_HEAD(7), + TLV_DB_RANGE_HEAD(2), 0, 6, TLV_DB_SCALE_ITEM(0, 150, 0), 7, 7, TLV_DB_SCALE_ITEM(1200, 0, 0), }; diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 0268cf98973..83c4bd5b2dd 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -694,6 +694,7 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) /* Initialize the the device_attribute structure */ dev_attr = &ssi_private->dev_attr; + sysfs_attr_init(&dev_attr->attr); dev_attr->attr.name = "statistics"; dev_attr->attr.mode = S_IRUGO; dev_attr->show = fsl_sysfs_ssi_show; 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 { diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index b133bfcc584..738391757f2 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -28,7 +28,7 @@ config SND_MXC_SOC_WM1133_EV1 config SND_SOC_MX27VIS_AIC32X4 tristate "SoC audio support for Visstrim M10 boards" - depends on MACH_IMX27_VISSTRIM_M10 + depends on MACH_IMX27_VISSTRIM_M10 && I2C select SND_SOC_TLV320AIC32X4 select SND_MXC_SOC_MX2 help diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig index 8f49e165f4d..c62d715235e 100644 --- a/sound/soc/kirkwood/Kconfig +++ b/sound/soc/kirkwood/Kconfig @@ -12,6 +12,7 @@ config SND_KIRKWOOD_SOC_I2S config SND_KIRKWOOD_SOC_OPENRD tristate "SoC Audio support for Kirkwood Openrd Client" depends on SND_KIRKWOOD_SOC && (MACH_OPENRD_CLIENT || MACH_OPENRD_ULTIMATE) + depends on I2C select SND_KIRKWOOD_SOC_I2S select SND_SOC_CS42L51 help @@ -20,7 +21,7 @@ config SND_KIRKWOOD_SOC_OPENRD config SND_KIRKWOOD_SOC_T5325 tristate "SoC Audio support for HP t5325" - depends on SND_KIRKWOOD_SOC && MACH_T5325 + depends on SND_KIRKWOOD_SOC && MACH_T5325 && I2C select SND_KIRKWOOD_SOC_I2S select SND_SOC_ALC5623 help diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c index dea5aa4aa64..f39d7dd9fbc 100644 --- a/sound/soc/mxs/mxs-pcm.c +++ b/sound/soc/mxs/mxs-pcm.c @@ -357,3 +357,6 @@ static void __exit snd_mxs_pcm_exit(void) platform_driver_unregister(&mxs_pcm_driver); } module_exit(snd_mxs_pcm_exit); + +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:mxs-pcm-audio"); diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index 7fbeaec06eb..1c57f6630a4 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -171,3 +171,4 @@ module_exit(mxs_sgtl5000_exit); MODULE_AUTHOR("Freescale Semiconductor, Inc."); MODULE_DESCRIPTION("MXS ALSA SoC Machine driver"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:mxs-sgtl5000"); diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c index 9c0edad90d8..a4e3237956e 100644 --- a/sound/soc/nuc900/nuc900-ac97.c +++ b/sound/soc/nuc900/nuc900-ac97.c @@ -365,7 +365,8 @@ static int __devinit nuc900_ac97_drvprobe(struct platform_device *pdev) if (ret) goto out3; - mfp_set_groupg(nuc900_audio->dev); /* enbale ac97 multifunction pin*/ + /* enbale ac97 multifunction pin */ + mfp_set_groupg(nuc900_audio->dev, "nuc900-audio"); return 0; diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index ffd2242e305..a0f7d3cfa47 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -151,6 +151,7 @@ config SND_SOC_ZYLONITE config SND_SOC_RAUMFELD tristate "SoC Audio support Raumfeld audio adapter" depends on SND_PXA2XX_SOC && (MACH_RAUMFELD_SPEAKER || MACH_RAUMFELD_CONNECTOR) + depends on I2C && SPI_MASTER select SND_PXA_SOC_SSP select SND_SOC_CS4270 select SND_SOC_AK4104 @@ -159,7 +160,7 @@ config SND_SOC_RAUMFELD config SND_PXA2XX_SOC_HX4700 tristate "SoC Audio support for HP iPAQ hx4700" - depends on SND_PXA2XX_SOC && MACH_H4700 + depends on SND_PXA2XX_SOC && MACH_H4700 && I2C select SND_PXA2XX_SOC_I2S select SND_SOC_AK4641 help diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c index 65c124831a0..c664e33fb6d 100644 --- a/sound/soc/pxa/hx4700.c +++ b/sound/soc/pxa/hx4700.c @@ -209,9 +209,10 @@ static int __devinit hx4700_audio_probe(struct platform_device *pdev) snd_soc_card_hx4700.dev = &pdev->dev; ret = snd_soc_register_card(&snd_soc_card_hx4700); if (ret) - return ret; + gpio_free_array(hx4700_audio_gpios, + ARRAY_SIZE(hx4700_audio_gpios)); - return 0; + return ret; } static int __devexit hx4700_audio_remove(struct platform_device *pdev) diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c index 1826acf20f7..8e523fd9189 100644 --- a/sound/soc/samsung/jive_wm8750.c +++ b/sound/soc/samsung/jive_wm8750.c @@ -101,7 +101,6 @@ static int jive_wm8750_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - int err; /* These endpoints are not being used. */ snd_soc_dapm_nc_pin(dapm, "LINPUT2"); @@ -131,7 +130,7 @@ static struct snd_soc_card snd_soc_machine_jive = { .dai_link = &jive_dai, .num_links = 1, - .dapm_widgtets = wm8750_dapm_widgets, + .dapm_widgets = wm8750_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(wm8750_dapm_widgets), .dapm_routes = audio_map, .num_dapm_routes = ARRAY_SIZE(audio_map), diff --git a/sound/soc/samsung/smdk2443_wm9710.c b/sound/soc/samsung/smdk2443_wm9710.c index 3a0dbfc793f..8bd1dc5706b 100644 --- a/sound/soc/samsung/smdk2443_wm9710.c +++ b/sound/soc/samsung/smdk2443_wm9710.c @@ -12,6 +12,7 @@ * */ +#include <linux/module.h> #include <sound/soc.h> static struct snd_soc_card smdk2443; 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 <sound/pcm_params.h> +#include <linux/module.h> /* * Default CFG switch settings to use this driver: 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"); 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 diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 0c12b98484b..4220bb0f273 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -58,7 +58,36 @@ int snd_soc_params_to_bclk(struct snd_pcm_hw_params *params) } EXPORT_SYMBOL_GPL(snd_soc_params_to_bclk); -static struct snd_soc_platform_driver dummy_platform; +static const struct snd_pcm_hardware dummy_dma_hardware = { + .formats = 0xffffffff, + .channels_min = 1, + .channels_max = UINT_MAX, + + /* Random values to keep userspace happy when checking constraints */ + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER, + .buffer_bytes_max = 128*1024, + .period_bytes_min = PAGE_SIZE, + .period_bytes_max = PAGE_SIZE*2, + .periods_min = 2, + .periods_max = 128, +}; + +static int dummy_dma_open(struct snd_pcm_substream *substream) +{ + snd_soc_set_runtime_hwparams(substream, &dummy_dma_hardware); + + return 0; +} + +static struct snd_pcm_ops dummy_dma_ops = { + .open = dummy_dma_open, + .ioctl = snd_pcm_lib_ioctl, +}; + +static struct snd_soc_platform_driver dummy_platform = { + .ops = &dummy_dma_ops, +}; static __devinit int snd_soc_dummy_probe(struct platform_device *pdev) { |