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[alsa-devel] [PATCH] ASoC: max98090: sync regcache on entering STANDBY
by Dylan Reid 12 Feb '14

12 Feb '14

Sync regcache when entering STANDBY from OFF. ON isn't entered with OFF as the current state, so the registers were not being re-synced after suspend/resume. The 98088 and 98095 already call regcache_sync from STANDBY. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/soc/codecs/max98090.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 51f9b3d..149b57f 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1769,16 +1769,6 @@ static int max98090_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: - if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - ret = regcache_sync(max98090->regmap); - - if (ret != 0) { - dev_err(codec->dev, - "Failed to sync cache: %d\n", ret); - return ret; - } - } - if (max98090->jack_state == M98090_JACK_STATE_HEADSET) { /* * Set to normal bias level. @@ -1792,6 +1782,16 @@ static int max98090_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = regcache_sync(max98090->regmap); + if (ret != 0) { + dev_err(codec->dev, + "Failed to sync cache: %d\n", ret); + return ret; + } + } + break; + case SND_SOC_BIAS_OFF: /* Set internal pull-up to lowest power mode */ snd_soc_update_bits(codec, M98090_REG_JACK_DETECT, -- 1.8.1.3.605.g02339dd

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[alsa-devel] [BUG]3.14-rc2 [PATCH] Revert "ALSA: hda/realtek - Avoid invalid COEFs for ALC271X"
by Martin Kepplinger 12 Feb '14

12 Feb '14

This reverts commit d3c56568f43807135f2c2a09582a69f809f0d8b7. The reverted commit breaks audio through headphone line out on the Acer TravelMate B113 (Type1Sku0) Notebook, my main work machine. I don't know much about it but this fixes my problem. Bisected and tested. Tested-by: Martin Kepplinger <martink(a)posteo.de> Signed-off-by: Martin Kepplinger <martink(a)posteo.de> --- sound/pci/hda/patch_realtek.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d9693ca..0f5af34 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4434,9 +4434,6 @@ static void alc269_fill_coef(struct hda_codec *codec) if (spec->codec_variant != ALC269_TYPE_ALC269VB) return; - /* ALC271X doesn't seem to support these COEFs (bko#52181) */ - if (!strcmp(codec->chip_name, "ALC271X")) - return; if ((alc_get_coef0(codec) & 0x00ff) < 0x015) { alc_write_coef_idx(codec, 0xf, 0x960b); -- 1.7.10.4

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[alsa-devel] [PATCH v3 0/2] ASoC: atmel_ssc_dai: add option to choose clock
by Bo Shen 12 Feb '14

12 Feb '14

When SSC work in slave mode, the clock can come from TK pin and also can come from RK pin, this is hardware design decided. So, make it available to choose where the clock from. Changes in v3: - Move the property from card to ssc device Series-changes: 2 - using "-" replace "_" in binding document Bo Shen (2): ASoC: atmel_ssc_dai: make option to choose clock Binding: atmel-ssc: add option to choose clock Documentation/devicetree/bindings/misc/atmel-ssc.txt | 8 ++++++++ drivers/misc/atmel-ssc.c | 6 ++++++ include/linux/atmel-ssc.h | 1 + sound/soc/atmel/atmel_ssc_dai.c | 13 +++++++++---- 4 files changed, 24 insertions(+), 4 deletions(-) -- 1.8.5.2

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Re: [alsa-devel] linux-next: Tree for Feb 10 (snd_hda)
by Randy Dunlap 12 Feb '14

12 Feb '14

On 02/09/2014 09:17 PM, Stephen Rothwell wrote: > Hi all, > > This tree fails (more than usual) the powerpc allyesconfig build. > > Changes since 20140207: > on x86_64: sound/built-in.o: In function `snd_hda_codec_configure': (.text+0x520a2): undefined reference to `snd_hda_parse_hdmi_codec' sound/built-in.o: In function `snd_hda_codec_configure': (.text+0x520a9): undefined reference to `snd_hda_parse_hdmi_codec' Lots of builtin drivers, but CONFIG_SND_HDA_CODEC_HDMI=m Full randconfig file is attached. -- ~Randy

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[alsa-devel] [PATCH] ASoC: blackfin: Fix machine driver Kconfig dependencies
by Lars-Peter Clausen 12 Feb '14

12 Feb '14

Since the machine driver selects the CODEC driver we need to make sure that the machine driver is only selectable if the CODEC driver can be build. This avoids build errors under some configurations (which typically only result from randconfig). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/blackfin/Kconfig | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index 54f74f8..4544d8e 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -11,7 +11,7 @@ config SND_BF5XX_I2S config SND_BF5XX_SOC_SSM2602 tristate "SoC SSM2602 Audio Codec Add-On Card support" - depends on SND_BF5XX_I2S && (SPI_MASTER || I2C) + depends on SND_BF5XX_I2S && SND_SOC_I2C_AND_SPI select SND_BF5XX_SOC_I2S if !BF60x select SND_BF6XX_SOC_I2S if BF60x select SND_SOC_SSM2602 @@ -21,10 +21,9 @@ config SND_BF5XX_SOC_SSM2602 config SND_SOC_BFIN_EVAL_ADAU1701 tristate "Support for the EVAL-ADAU1701MINIZ board on Blackfin eval boards" - depends on SND_BF5XX_I2S + depends on SND_BF5XX_I2S && I2C select SND_BF5XX_SOC_I2S select SND_SOC_ADAU1701 - select I2C help Say Y if you want to add support for the Analog Devices EVAL-ADAU1701MINIZ board connected to one of the Blackfin evaluation boards like the @@ -45,7 +44,7 @@ config SND_SOC_BFIN_EVAL_ADAU1373 config SND_SOC_BFIN_EVAL_ADAV80X tristate "Support for the EVAL-ADAV80X boards on Blackfin eval boards" - depends on SND_BF5XX_I2S && (SPI_MASTER || I2C) + depends on SND_BF5XX_I2S && SND_SOC_I2C_AND_SPI select SND_BF5XX_SOC_I2S select SND_SOC_ADAV80X help @@ -58,7 +57,7 @@ config SND_SOC_BFIN_EVAL_ADAV80X config SND_BF5XX_SOC_AD1836 tristate "SoC AD1836 Audio support for BF5xx" - depends on SND_BF5XX_I2S + depends on SND_BF5XX_I2S && SPI_MASTER select SND_BF5XX_SOC_I2S select SND_SOC_AD1836 help @@ -66,7 +65,7 @@ config SND_BF5XX_SOC_AD1836 config SND_BF5XX_SOC_AD193X tristate "SoC AD193X Audio support for Blackfin" - depends on SND_BF5XX_I2S + depends on SND_BF5XX_I2S && SND_SOC_I2C_AND_SPI select SND_BF5XX_SOC_I2S select SND_SOC_AD193X help -- 1.8.0

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[alsa-devel] [PATCH] ASoC: dapm: Add locking to snd_soc_dapm_xxxx_pin functions
by Charles Keepax 12 Feb '14

12 Feb '14

snd_soc_dapm_disable_pin, snd_soc_dapm_enable_pin and snd_soc_dapm_force_enable_pin all require the dapm_mutex to be held when they are called as they edit the dirty list. There are 385 usages of these functions and only 7 hold the lock whilst calling. This patch moves the locking into snd_soc_dapm_set_pin and fixes up the places where the lock was held on the caller side. This saves on fixing up all the current users and also is much more consistant with the rest of the DAPM API which all handles the locking internally. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Hi, I have put the changes in a single patch to avoid bisect problems, but let me know if it would be better split into seperate patches. Thanks, Charles drivers/extcon/extcon-arizona.c | 11 ----------- drivers/input/misc/arizona-haptics.c | 19 ------------------- sound/soc/soc-dapm.c | 8 ++++---- 3 files changed, 4 insertions(+), 34 deletions(-) diff --git a/drivers/extcon/extcon-arizona.c b/drivers/extcon/extcon-arizona.c index c20602f..84167f4 100644 --- a/drivers/extcon/extcon-arizona.c +++ b/drivers/extcon/extcon-arizona.c @@ -222,26 +222,19 @@ static void arizona_extcon_pulse_micbias(struct arizona_extcon_info *info) struct snd_soc_dapm_context *dapm = arizona->dapm; int ret; - mutex_lock(&dapm->card->dapm_mutex); - ret = snd_soc_dapm_force_enable_pin(dapm, widget); if (ret != 0) dev_warn(arizona->dev, "Failed to enable %s: %d\n", widget, ret); - mutex_unlock(&dapm->card->dapm_mutex); - snd_soc_dapm_sync(dapm); if (!arizona->pdata.micd_force_micbias) { - mutex_lock(&dapm->card->dapm_mutex); - ret = snd_soc_dapm_disable_pin(arizona->dapm, widget); if (ret != 0) dev_warn(arizona->dev, "Failed to disable %s: %d\n", widget, ret); - mutex_unlock(&dapm->card->dapm_mutex); snd_soc_dapm_sync(dapm); } @@ -304,16 +297,12 @@ static void arizona_stop_mic(struct arizona_extcon_info *info) ARIZONA_MICD_ENA, 0, &change); - mutex_lock(&dapm->card->dapm_mutex); - ret = snd_soc_dapm_disable_pin(dapm, widget); if (ret != 0) dev_warn(arizona->dev, "Failed to disable %s: %d\n", widget, ret); - mutex_unlock(&dapm->card->dapm_mutex); - snd_soc_dapm_sync(dapm); if (info->micd_reva) { diff --git a/drivers/input/misc/arizona-haptics.c b/drivers/input/misc/arizona-haptics.c index 7a04f54..ef2e281 100644 --- a/drivers/input/misc/arizona-haptics.c +++ b/drivers/input/misc/arizona-haptics.c @@ -37,7 +37,6 @@ static void arizona_haptics_work(struct work_struct *work) struct arizona_haptics, work); struct arizona *arizona = haptics->arizona; - struct mutex *dapm_mutex = &arizona->dapm->card->dapm_mutex; int ret; if (!haptics->arizona->dapm) { @@ -67,13 +66,10 @@ static void arizona_haptics_work(struct work_struct *work) return; } - mutex_lock_nested(dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - ret = snd_soc_dapm_enable_pin(arizona->dapm, "HAPTICS"); if (ret != 0) { dev_err(arizona->dev, "Failed to start HAPTICS: %d\n", ret); - mutex_unlock(dapm_mutex); return; } @@ -81,21 +77,14 @@ static void arizona_haptics_work(struct work_struct *work) if (ret != 0) { dev_err(arizona->dev, "Failed to sync DAPM: %d\n", ret); - mutex_unlock(dapm_mutex); return; } - - mutex_unlock(dapm_mutex); - } else { /* This disable sequence will be a noop if already enabled */ - mutex_lock_nested(dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - ret = snd_soc_dapm_disable_pin(arizona->dapm, "HAPTICS"); if (ret != 0) { dev_err(arizona->dev, "Failed to disable HAPTICS: %d\n", ret); - mutex_unlock(dapm_mutex); return; } @@ -103,12 +92,9 @@ static void arizona_haptics_work(struct work_struct *work) if (ret != 0) { dev_err(arizona->dev, "Failed to sync DAPM: %d\n", ret); - mutex_unlock(dapm_mutex); return; } - mutex_unlock(dapm_mutex); - ret = regmap_update_bits(arizona->regmap, ARIZONA_HAPTICS_CONTROL_1, ARIZONA_HAP_CTRL_MASK, @@ -155,16 +141,11 @@ static int arizona_haptics_play(struct input_dev *input, void *data, static void arizona_haptics_close(struct input_dev *input) { struct arizona_haptics *haptics = input_get_drvdata(input); - struct mutex *dapm_mutex = &haptics->arizona->dapm->card->dapm_mutex; cancel_work_sync(&haptics->work); - mutex_lock_nested(dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - if (haptics->arizona->dapm) snd_soc_dapm_disable_pin(haptics->arizona->dapm, "HAPTICS"); - - mutex_unlock(dapm_mutex); } static int arizona_haptics_probe(struct platform_device *pdev) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index dc8ff13..a44b560 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2325,6 +2325,8 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, { struct snd_soc_dapm_widget *w = dapm_find_widget(dapm, pin, true); + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + if (!w) { dev_err(dapm->dev, "ASoC: DAPM unknown pin %s\n", pin); return -EINVAL; @@ -2337,6 +2339,8 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, if (status == 0) w->force = 0; + mutex_unlock(&dapm->card->dapm_mutex); + return 0; } @@ -3210,15 +3214,11 @@ int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol, struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); const char *pin = (const char *)kcontrol->private_value; - mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - if (ucontrol->value.integer.value[0]) snd_soc_dapm_enable_pin(&card->dapm, pin); else snd_soc_dapm_disable_pin(&card->dapm, pin); - mutex_unlock(&card->dapm_mutex); - snd_soc_dapm_sync(&card->dapm); return 0; } -- 1.7.2.5

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[alsa-devel] [PATCH v2 1/2] ASoC: fsl-spdif: big-endian support
by Xiubo Li 12 Feb '14

12 Feb '14

For most platforms, the CPU and SPDIF device is in the same endianess mode. While for the LS1 platform, the CPU is in LE mode and the SPDIF is in BE mode. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> Cc: Nicolin Chen <Guangyu.Chen(a)freescale.com> Acked-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- Fix some issues from Nicolin Chen's comments. Documentation/devicetree/bindings/sound/fsl,spdif.txt | 5 +++++ sound/soc/fsl/fsl_spdif.c | 5 ++++- 2 files changed, 9 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.txt b/Documentation/devicetree/bindings/sound/fsl,spdif.txt index f2ae335..3e9e82c8 100644 --- a/Documentation/devicetree/bindings/sound/fsl,spdif.txt +++ b/Documentation/devicetree/bindings/sound/fsl,spdif.txt @@ -29,6 +29,10 @@ Required properties: can also be referred to TxClk_Source bit of register SPDIF_STC. + - big-endian : If this property is absent, the native endian mode will + be in use as default, or the big endian mode will be in use for all the + device registers. + Example: spdif: spdif@02004000 { @@ -50,5 +54,6 @@ spdif: spdif@02004000 { "rxtx5", "rxtx6", "rxtx7"; + big-endian; status = "okay"; }; diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 8f36f49..6452ca8 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -985,7 +985,7 @@ static bool fsl_spdif_writeable_reg(struct device *dev, unsigned int reg) } } -static const struct regmap_config fsl_spdif_regmap_config = { +static struct regmap_config fsl_spdif_regmap_config = { .reg_bits = 32, .reg_stride = 4, .val_bits = 32, @@ -1105,6 +1105,9 @@ static int fsl_spdif_probe(struct platform_device *pdev) memcpy(&spdif_priv->cpu_dai_drv, &fsl_spdif_dai, sizeof(fsl_spdif_dai)); spdif_priv->cpu_dai_drv.name = spdif_priv->name; + if (of_property_read_bool(np, "big-endian")) + fsl_spdif_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG; + /* Get the addresses and IRQ */ res = platform_get_resource(pdev, IORESOURCE_MEM, 0); regs = devm_ioremap_resource(&pdev->dev, res); -- 1.8.4

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[alsa-devel] [PATCH] ASoC: Intel: Add a mfld prefix to Intel SST drivers.
by Liam Girdwood 12 Feb '14

12 Feb '14

Resent with correct email for Mark. In order to differentiate the different Intel SST audio core drivers we need to rename the current drivers with a mfld prefix. This also includes renaming in the Makefile and Kconfig Acked-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/Kconfig | 4 ++-- sound/soc/intel/Makefile | 4 ++-- sound/soc/intel/{sst_dsp.h => sst-mfld-dsp.h} | 8 ++++---- sound/soc/intel/{sst_platform.c => sst-mfld-platform.c} | 8 ++++---- sound/soc/intel/{sst_platform.h => sst-mfld-platform.h} | 4 ++-- 5 files changed, 14 insertions(+), 14 deletions(-) rename sound/soc/intel/{sst_dsp.h => sst-mfld-dsp.h} (96%) rename sound/soc/intel/{sst_platform.c => sst-mfld-platform.c} (99%) rename sound/soc/intel/{sst_platform.h => sst-mfld-platform.h} (97%) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 61c10bf..4d9d0a5 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -2,12 +2,12 @@ config SND_MFLD_MACHINE tristate "SOC Machine Audio driver for Intel Medfield MID platform" depends on INTEL_SCU_IPC select SND_SOC_SN95031 - select SND_SST_PLATFORM + select SND_SST_MFLD_PLATFORM help This adds support for ASoC machine driver for Intel(R) MID Medfield platform used as alsa device in audio substem in Intel(R) MID devices Say Y if you have such a device If unsure select "N". -config SND_SST_PLATFORM +config SND_SST_MFLD_PLATFORM tristate diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 6398833..eb899fc 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -1,5 +1,5 @@ -snd-soc-sst-platform-objs := sst_platform.o +snd-soc-sst-mfld-platform-objs := sst-mfld-platform.o snd-soc-mfld-machine-objs := mfld_machine.o -obj-$(CONFIG_SND_SST_PLATFORM) += snd-soc-sst-platform.o +obj-$(CONFIG_SND_SST_MFLD_PLATFORM) += snd-soc-sst-mfld-platform.o obj-$(CONFIG_SND_MFLD_MACHINE) += snd-soc-mfld-machine.o diff --git a/sound/soc/intel/sst_dsp.h b/sound/soc/intel/sst-mfld-dsp.h similarity index 96% rename from sound/soc/intel/sst_dsp.h rename to sound/soc/intel/sst-mfld-dsp.h index 0fce1de..3b63edc 100644 --- a/sound/soc/intel/sst_dsp.h +++ b/sound/soc/intel/sst-mfld-dsp.h @@ -1,7 +1,7 @@ -#ifndef __SST_DSP_H__ -#define __SST_DSP_H__ +#ifndef __SST_MFLD_DSP_H__ +#define __SST_MFLD_DSP_H__ /* - * sst_dsp.h - Intel SST Driver for audio engine + * sst_mfld_dsp.h - Intel SST Driver for audio engine * * Copyright (C) 2008-12 Intel Corporation * Authors: Vinod Koul <vinod.koul(a)linux.intel.com> @@ -131,4 +131,4 @@ struct snd_sst_params { struct snd_sst_alloc_params_ext aparams; }; -#endif /* __SST_DSP_H__ */ +#endif /* __SST_MFLD_DSP_H__ */ diff --git a/sound/soc/intel/sst_platform.c b/sound/soc/intel/sst-mfld-platform.c similarity index 99% rename from sound/soc/intel/sst_platform.c rename to sound/soc/intel/sst-mfld-platform.c index f465a81..840306c 100644 --- a/sound/soc/intel/sst_platform.c +++ b/sound/soc/intel/sst-mfld-platform.c @@ -1,5 +1,5 @@ /* - * sst_platform.c - Intel MID Platform driver + * sst_mfld_platform.c - Intel MID Platform driver * * Copyright (C) 2010-2013 Intel Corp * Author: Vinod Koul <vinod.koul(a)intel.com> @@ -33,7 +33,7 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/compress_driver.h> -#include "sst_platform.h" +#include "sst-mfld-platform.h" static struct sst_device *sst; static DEFINE_MUTEX(sst_lock); @@ -709,7 +709,7 @@ static int sst_platform_remove(struct platform_device *pdev) static struct platform_driver sst_platform_driver = { .driver = { - .name = "sst-platform", + .name = "sst-mfld-platform", .owner = THIS_MODULE, }, .probe = sst_platform_probe, @@ -722,4 +722,4 @@ MODULE_DESCRIPTION("ASoC Intel(R) MID Platform driver"); MODULE_AUTHOR("Vinod Koul <vinod.koul(a)intel.com>"); MODULE_AUTHOR("Harsha Priya <priya.harsha(a)intel.com>"); MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:sst-platform"); +MODULE_ALIAS("platform:sst-mfld-platform"); diff --git a/sound/soc/intel/sst_platform.h b/sound/soc/intel/sst-mfld-platform.h similarity index 97% rename from sound/soc/intel/sst_platform.h rename to sound/soc/intel/sst-mfld-platform.h index bee64fb..0c4e2dd 100644 --- a/sound/soc/intel/sst_platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -1,5 +1,5 @@ /* - * sst_platform.h - Intel MID Platform driver header file + * sst_mfld_platform.h - Intel MID Platform driver header file * * Copyright (C) 2010 Intel Corp * Author: Vinod Koul <vinod.koul(a)intel.com> @@ -27,7 +27,7 @@ #ifndef __SST_PLATFORMDRV_H__ #define __SST_PLATFORMDRV_H__ -#include "sst_dsp.h" +#include "sst-mfld-dsp.h" #define SST_MONO 1 #define SST_STEREO 2 -- 1.8.3.2

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[alsa-devel] [PATCH] ASoC: Intel: Add a mfld prefix to Intel SST drivers.
by Liam Girdwood 12 Feb '14

12 Feb '14

In order to differentiate the different Intel SST audio core drivers we need to rename the current drivers with a mfld prefix. This also includes renaming in the Makefile and Kconfig Acked-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/Kconfig | 4 ++-- sound/soc/intel/Makefile | 4 ++-- sound/soc/intel/{sst_dsp.h => sst-mfld-dsp.h} | 8 ++++---- sound/soc/intel/{sst_platform.c => sst-mfld-platform.c} | 8 ++++---- sound/soc/intel/{sst_platform.h => sst-mfld-platform.h} | 4 ++-- 5 files changed, 14 insertions(+), 14 deletions(-) rename sound/soc/intel/{sst_dsp.h => sst-mfld-dsp.h} (96%) rename sound/soc/intel/{sst_platform.c => sst-mfld-platform.c} (99%) rename sound/soc/intel/{sst_platform.h => sst-mfld-platform.h} (97%) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 61c10bf..4d9d0a5 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -2,12 +2,12 @@ config SND_MFLD_MACHINE tristate "SOC Machine Audio driver for Intel Medfield MID platform" depends on INTEL_SCU_IPC select SND_SOC_SN95031 - select SND_SST_PLATFORM + select SND_SST_MFLD_PLATFORM help This adds support for ASoC machine driver for Intel(R) MID Medfield platform used as alsa device in audio substem in Intel(R) MID devices Say Y if you have such a device If unsure select "N". -config SND_SST_PLATFORM +config SND_SST_MFLD_PLATFORM tristate diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 6398833..eb899fc 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -1,5 +1,5 @@ -snd-soc-sst-platform-objs := sst_platform.o +snd-soc-sst-mfld-platform-objs := sst-mfld-platform.o snd-soc-mfld-machine-objs := mfld_machine.o -obj-$(CONFIG_SND_SST_PLATFORM) += snd-soc-sst-platform.o +obj-$(CONFIG_SND_SST_MFLD_PLATFORM) += snd-soc-sst-mfld-platform.o obj-$(CONFIG_SND_MFLD_MACHINE) += snd-soc-mfld-machine.o diff --git a/sound/soc/intel/sst_dsp.h b/sound/soc/intel/sst-mfld-dsp.h similarity index 96% rename from sound/soc/intel/sst_dsp.h rename to sound/soc/intel/sst-mfld-dsp.h index 0fce1de..3b63edc 100644 --- a/sound/soc/intel/sst_dsp.h +++ b/sound/soc/intel/sst-mfld-dsp.h @@ -1,7 +1,7 @@ -#ifndef __SST_DSP_H__ -#define __SST_DSP_H__ +#ifndef __SST_MFLD_DSP_H__ +#define __SST_MFLD_DSP_H__ /* - * sst_dsp.h - Intel SST Driver for audio engine + * sst_mfld_dsp.h - Intel SST Driver for audio engine * * Copyright (C) 2008-12 Intel Corporation * Authors: Vinod Koul <vinod.koul(a)linux.intel.com> @@ -131,4 +131,4 @@ struct snd_sst_params { struct snd_sst_alloc_params_ext aparams; }; -#endif /* __SST_DSP_H__ */ +#endif /* __SST_MFLD_DSP_H__ */ diff --git a/sound/soc/intel/sst_platform.c b/sound/soc/intel/sst-mfld-platform.c similarity index 99% rename from sound/soc/intel/sst_platform.c rename to sound/soc/intel/sst-mfld-platform.c index f465a81..840306c 100644 --- a/sound/soc/intel/sst_platform.c +++ b/sound/soc/intel/sst-mfld-platform.c @@ -1,5 +1,5 @@ /* - * sst_platform.c - Intel MID Platform driver + * sst_mfld_platform.c - Intel MID Platform driver * * Copyright (C) 2010-2013 Intel Corp * Author: Vinod Koul <vinod.koul(a)intel.com> @@ -33,7 +33,7 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/compress_driver.h> -#include "sst_platform.h" +#include "sst-mfld-platform.h" static struct sst_device *sst; static DEFINE_MUTEX(sst_lock); @@ -709,7 +709,7 @@ static int sst_platform_remove(struct platform_device *pdev) static struct platform_driver sst_platform_driver = { .driver = { - .name = "sst-platform", + .name = "sst-mfld-platform", .owner = THIS_MODULE, }, .probe = sst_platform_probe, @@ -722,4 +722,4 @@ MODULE_DESCRIPTION("ASoC Intel(R) MID Platform driver"); MODULE_AUTHOR("Vinod Koul <vinod.koul(a)intel.com>"); MODULE_AUTHOR("Harsha Priya <priya.harsha(a)intel.com>"); MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:sst-platform"); +MODULE_ALIAS("platform:sst-mfld-platform"); diff --git a/sound/soc/intel/sst_platform.h b/sound/soc/intel/sst-mfld-platform.h similarity index 97% rename from sound/soc/intel/sst_platform.h rename to sound/soc/intel/sst-mfld-platform.h index bee64fb..0c4e2dd 100644 --- a/sound/soc/intel/sst_platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -1,5 +1,5 @@ /* - * sst_platform.h - Intel MID Platform driver header file + * sst_mfld_platform.h - Intel MID Platform driver header file * * Copyright (C) 2010 Intel Corp * Author: Vinod Koul <vinod.koul(a)intel.com> @@ -27,7 +27,7 @@ #ifndef __SST_PLATFORMDRV_H__ #define __SST_PLATFORMDRV_H__ -#include "sst_dsp.h" +#include "sst-mfld-dsp.h" #define SST_MONO 1 #define SST_STEREO 2 -- 1.8.3.2

1 0

[alsa-devel] [PATCH] ASoC: rsnd: print error if there is SRC settings mismatch
by Kuninori Morimoto 12 Feb '14

12 Feb '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> rsnd request clock master if SRC is used Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/scu.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/sh/rcar/scu.c b/sound/soc/sh/rcar/scu.c index 6e5c763..3984d4b 100644 --- a/sound/soc/sh/rcar/scu.c +++ b/sound/soc/sh/rcar/scu.c @@ -122,6 +122,14 @@ static int rsnd_scu_ssi_mode_init(struct rsnd_mod *mod, struct rsnd_priv *priv = rsnd_mod_to_priv(mod); struct rsnd_scu *scu = rsnd_mod_to_scu(mod); int id = rsnd_mod_id(mod); + u32 convert_rate = rsnd_scu_convert_rate(scu); + + if (convert_rate && !rsnd_dai_is_clk_master(rdai)) { + struct device *dev = rsnd_priv_to_dev(priv); + + dev_err(dev, "rsnd should be clk master when you rate convert\n"); + return -EINVAL; + } /* * SSI_MODE0 -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: rsnd: set DIV_EN register on rsnd_adg_set_convert_clk_gen2()
by Kuninori Morimoto 12 Feb '14

12 Feb '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> DIV_EN register enable bit is required when you use Gen2 SRC Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/adg.c | 16 +++++++++++++--- sound/soc/sh/rcar/gen.c | 1 + sound/soc/sh/rcar/rsnd.h | 1 + 3 files changed, 15 insertions(+), 3 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 821791e..fc1dce6 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -110,8 +110,8 @@ int rsnd_adg_set_convert_clk_gen2(struct rsnd_mod *mod, struct rsnd_priv *priv = rsnd_mod_to_priv(mod); struct rsnd_adg *adg = rsnd_priv_to_adg(priv); struct device *dev = rsnd_priv_to_dev(priv); - int idx, sel, div, step; - u32 val; + int idx, sel, div, step, ret; + u32 val, en; unsigned int min, diff; unsigned int sel_rate [] = { clk_get_rate(adg->clk[CLKA]), /* 0000: CLKA */ @@ -123,6 +123,7 @@ int rsnd_adg_set_convert_clk_gen2(struct rsnd_mod *mod, min = ~0; val = 0; + en = 0; for (sel = 0; sel < ARRAY_SIZE(sel_rate); sel++) { idx = 0; step = 2; @@ -135,6 +136,7 @@ int rsnd_adg_set_convert_clk_gen2(struct rsnd_mod *mod, if (min > diff) { val = (sel << 8) | idx; min = diff; + en = 1 << (sel + 1); /* fixme */ } /* @@ -156,7 +158,15 @@ int rsnd_adg_set_convert_clk_gen2(struct rsnd_mod *mod, return -EIO; } - return rsnd_adg_set_src_timsel_gen2(rdai, mod, io, val); + ret = rsnd_adg_set_src_timsel_gen2(rdai, mod, io, val); + if (ret < 0) { + dev_err(dev, "timsel error\n"); + return ret; + } + + rsnd_mod_bset(mod, DIV_EN, en, en); + + return 0; } int rsnd_adg_set_convert_timing_gen2(struct rsnd_mod *mod, diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 3e03a8b..0a43b90 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -253,6 +253,7 @@ static int rsnd_gen2_regmap_init(struct rsnd_priv *priv, struct rsnd_gen *gen) RSND_GEN2_S_REG(gen, ADG, AUDIO_CLK_SEL0, 0x0c), RSND_GEN2_S_REG(gen, ADG, AUDIO_CLK_SEL1, 0x10), RSND_GEN2_S_REG(gen, ADG, AUDIO_CLK_SEL2, 0x14), + RSND_GEN2_S_REG(gen, ADG, DIV_EN, 0x30), RSND_GEN2_S_REG(gen, ADG, SRCIN_TIMSEL0, 0x34), RSND_GEN2_S_REG(gen, ADG, SRCIN_TIMSEL1, 0x38), RSND_GEN2_S_REG(gen, ADG, SRCIN_TIMSEL2, 0x3c), diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 8b66dc1..5691bfc 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -67,6 +67,7 @@ enum rsnd_reg { RSND_REG_AUDIO_CLK_SEL3, /* for Gen1 */ RSND_REG_AUDIO_CLK_SEL4, /* for Gen1 */ RSND_REG_AUDIO_CLK_SEL5, /* for Gen1 */ + RSND_REG_DIV_EN, /* for Gen2 */ RSND_REG_SRCIN_TIMSEL0, /* for Gen2 */ RSND_REG_SRCIN_TIMSEL1, /* for Gen2 */ RSND_REG_SRCIN_TIMSEL2, /* for Gen2 */ -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: rsnd: tidyup original for_each_rsnd_xxx macro
by Kuninori Morimoto 12 Feb '14

12 Feb '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current for_each_rsnd_xxx macro will read out-of-array's memory after last loop operation. It was not good C language operation, and the binary which was compiled by (at least) gcc 4.8.1 is broken This patch tidyup these issues Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/adg.c | 7 ++++--- sound/soc/sh/rcar/rsnd.h | 7 ++++--- 2 files changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 8d3a82e..bc8961c 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -25,9 +25,10 @@ struct rsnd_adg { }; #define for_each_rsnd_clk(pos, adg, i) \ - for (i = 0, (pos) = adg->clk[i]; \ - i < CLKMAX; \ - i++, (pos) = adg->clk[i]) + for (i = 0; \ + (i < CLKMAX) && \ + ((pos) = adg->clk[i]); \ + i++) #define rsnd_priv_to_adg(priv) ((struct rsnd_adg *)(priv)->adg) diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 8b66dc1..9e4efb4 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -216,9 +216,10 @@ struct rsnd_dai { #define rsnd_dai_nr(priv) ((priv)->dai_nr) #define for_each_rsnd_dai(rdai, priv, i) \ - for (i = 0, (rdai) = rsnd_dai_get(priv, i); \ - i < rsnd_dai_nr(priv); \ - i++, (rdai) = rsnd_dai_get(priv, i)) + for (i = 0; \ + (i < rsnd_dai_nr(priv)) && \ + ((rdai) = rsnd_dai_get(priv, i)); \ + i++) struct rsnd_dai *rsnd_dai_get(struct rsnd_priv *priv, int id); int rsnd_dai_disconnect(struct rsnd_mod *mod); -- 1.7.9.5

2 1

[alsa-devel] [PATCH][RFC] alsa-lib: heavy pcm atomics cleanup
by riku.voipio＠linaro.org 12 Feb '14

12 Feb '14

From: Riku Voipio <riku.voipio(a)linaro.org> The following patch comes from the realization that at least ARM code for atomics is quite broken and nobody has cared for a decade. A quick dive shows that only snd_atomic_{read,write}_{begin,end} appear to be used widely. These are implemented using wmb/rmb. Only other use of atomic functions is in pcm_meter.c. The #SND_PCM_TYPE_METER plugin type appears rarely, if ever, used. I presume these days anyone who wants a meter/scope will do in pulseaudio layer instead of alsa. It would seem better fit to have pcm_meter in alsa-plugins instead of alsa-lib, but I guess that would be an ABI break... So instead, I'm proposing here 1. Removal of all hand-crafted atomics from iatomic.h apart from barriers, which are used in snd_atomic_{read,write}_{begin,end}. 2. Using __sync_synchronize as the default fallback for barriers. This has been available since gcc 4.1, so it shouldn't be a problem. 3. Defining the few atomics used by pcm_meter.c withing pcm_meter.c itself, using gcc atomic builtins[1]. 4. Since gcc atomic builtins are available only since gcc 4.7, add a check for that in gcc configure.in, and don't build pcm meter plugin if using older gcc. The last point has the impact, that if there actually is someone who 1) uses the meter plugin 2) wants to upgrade to 2014 alsa-lib 3) but does not want to use a 2012+ gcc - that someone will be inconvenienced. Finally remove the unneeded configure check for cpu type. We can trust the gcc to set right flags for us. [1] http://gcc.gnu.org/onlinedocs/gcc/_005f_005fatomic-Builtins.html Signed-off-by: Riku Voipio <riku.voipio(a)linaro.org> --- configure.in | 53 +-- include/iatomic.h | 1044 +-------------------------------------------------- src/pcm/pcm_meter.c | 9 +- 3 files changed, 33 insertions(+), 1073 deletions(-) diff --git a/configure.in b/configure.in index 35fd89f..bb56eb6 100644 --- a/configure.in +++ b/configure.in @@ -309,40 +309,6 @@ fi AC_SUBST(ALSA_DEPLIBS) -dnl Check for architecture -AC_MSG_CHECKING(for architecture) -case "$host" in -i?86*) - AC_MSG_RESULT(x86) - ;; -x86_64*) - AC_MSG_RESULT(x86) - ;; -alpha*) - AC_MSG_RESULT(alpha) - ;; -powerpc*|ppc*) - AC_MSG_RESULT(ppc) - CPPFLAGS="$CPPFLAGS -D__ppc__" - ;; -ia64*) - AC_MSG_RESULT(ia64) - CPPFLAGS="$CPPFLAGS -D__ia64__" - ;; -mips*) - AC_MSG_RESULT(mips) - CPPFLAGS="$CPPFLAGS -D__mips__" - ;; -arm*) - AC_MSG_RESULT(arm) - CPPFLAGS="$CPPFLAGS -D__arm__" - ;; -*) - AC_MSG_RESULT($host_cpu) - echo "No atomic operations supported.." - ;; -esac - dnl Check for wordexp.h AC_CHECK_HEADERS([wordexp.h]) @@ -488,6 +454,21 @@ else pcm_plugins="" fi +dnl check atomics for pcm_meter + +AC_MSG_CHECKING([whether GCC supports builtin atomic intrinsics]) +if test -z "$gcc_have_atomics"; then + gcc_have_atomics=no + AC_TRY_LINK([], + [int i; + __atomic_load_n(&i, __ATOMIC_SEQ_CST); + __atomic_add_fetch(&i, 0, __ATOMIC_SEQ_CST); + ], + [gcc_have_atomics=yes], + [gcc_have_atomics=no]) +fi +AC_MSG_RESULT($gcc_have_atomics) + PCM_PLUGIN_LIST="copy linear route mulaw alaw adpcm rate plug multi shm file null empty share meter hooks lfloat ladspa dmix dshare dsnoop asym iec958 softvol extplug ioplug mmap_emul" build_pcm_plugin="no" @@ -531,6 +512,10 @@ if test "$softfloat" = "yes"; then build_pcm_ladspa="no" fi +if test "$gcc_have_atomics" != "yes"; then + build_pcm_meter="no" +fi + AM_CONDITIONAL([BUILD_PCM_PLUGIN], [test x$build_pcm_plugin = xyes]) AM_CONDITIONAL([BUILD_PCM_PLUGIN_COPY], [test x$build_pcm_copy = xyes]) AM_CONDITIONAL([BUILD_PCM_PLUGIN_LINEAR], [test x$build_pcm_linear = xyes]) diff --git a/include/iatomic.h b/include/iatomic.h index 2393297..acdd3e2 100644 --- a/include/iatomic.h +++ b/include/iatomic.h @@ -1,349 +1,21 @@ #ifndef __ALSA_IATOMIC_H #define __ALSA_IATOMIC_H -#if defined(__i386__) || defined(__x86_64__) - -/* - * Atomic operations that C can't guarantee us. Useful for - * resource counting etc.. - */ - -#define ATOMIC_SMP_LOCK "lock ; " - -/* - * Make sure gcc doesn't try to be clever and move things around - * on us. We need to use _exactly_ the address the user gave us, - * not some alias that contains the same information. - */ -typedef struct { volatile int counter; } atomic_t; - -#define ATOMIC_INIT(i) { (i) } - -/** - * atomic_read - read atomic variable - * @v: pointer of type atomic_t - * - * Atomically reads the value of @v. Note that the guaranteed - * useful range of an atomic_t is only 24 bits. - */ -#define atomic_read(v) ((v)->counter) - -/** - * atomic_set - set atomic variable - * @v: pointer of type atomic_t - * @i: required value - * - * Atomically sets the value of @v to @i. Note that the guaranteed - * useful range of an atomic_t is only 24 bits. - */ -#define atomic_set(v,i) (((v)->counter) = (i)) - -/** - * atomic_add - add integer to atomic variable - * @i: integer value to add - * @v: pointer of type atomic_t - * - * Atomically adds @i to @v. Note that the guaranteed useful range - * of an atomic_t is only 24 bits. - */ -static __inline__ void atomic_add(int i, atomic_t *v) -{ - __asm__ __volatile__( - ATOMIC_SMP_LOCK "addl %1,%0" - :"=m" (v->counter) - :"ir" (i), "m" (v->counter)); -} - -/** - * atomic_sub - subtract the atomic variable - * @i: integer value to subtract - * @v: pointer of type atomic_t - * - * Atomically subtracts @i from @v. Note that the guaranteed - * useful range of an atomic_t is only 24 bits. - */ -static __inline__ void atomic_sub(int i, atomic_t *v) -{ - __asm__ __volatile__( - ATOMIC_SMP_LOCK "subl %1,%0" - :"=m" (v->counter) - :"ir" (i), "m" (v->counter)); -} - -/** - * atomic_sub_and_test - subtract value from variable and test result - * @i: integer value to subtract - * @v: pointer of type atomic_t - * - * Atomically subtracts @i from @v and returns - * true if the result is zero, or false for all - * other cases. Note that the guaranteed - * useful range of an atomic_t is only 24 bits. - */ -static __inline__ int atomic_sub_and_test(int i, atomic_t *v) -{ - unsigned char c; - - __asm__ __volatile__( - ATOMIC_SMP_LOCK "subl %2,%0; sete %1" - :"=m" (v->counter), "=qm" (c) - :"ir" (i), "m" (v->counter) : "memory"); - return c; -} - -/** - * atomic_inc - increment atomic variable - * @v: pointer of type atomic_t - * - * Atomically increments @v by 1. Note that the guaranteed - * useful range of an atomic_t is only 24 bits. - */ -static __inline__ void atomic_inc(atomic_t *v) -{ - __asm__ __volatile__( - ATOMIC_SMP_LOCK "incl %0" - :"=m" (v->counter) - :"m" (v->counter)); -} - -/** - * atomic_dec - decrement atomic variable - * @v: pointer of type atomic_t - * - * Atomically decrements @v by 1. Note that the guaranteed - * useful range of an atomic_t is only 24 bits. - */ -static __inline__ void atomic_dec(atomic_t *v) -{ - __asm__ __volatile__( - ATOMIC_SMP_LOCK "decl %0" - :"=m" (v->counter) - :"m" (v->counter)); -} - -/** - * atomic_dec_and_test - decrement and test - * @v: pointer of type atomic_t - * - * Atomically decrements @v by 1 and - * returns true if the result is 0, or false for all other - * cases. Note that the guaranteed - * useful range of an atomic_t is only 24 bits. - */ -static __inline__ int atomic_dec_and_test(atomic_t *v) -{ - unsigned char c; - - __asm__ __volatile__( - ATOMIC_SMP_LOCK "decl %0; sete %1" - :"=m" (v->counter), "=qm" (c) - :"m" (v->counter) : "memory"); - return c != 0; -} - -/** - * atomic_inc_and_test - increment and test - * @v: pointer of type atomic_t - * - * Atomically increments @v by 1 - * and returns true if the result is zero, or false for all - * other cases. Note that the guaranteed - * useful range of an atomic_t is only 24 bits. - */ -static __inline__ int atomic_inc_and_test(atomic_t *v) -{ - unsigned char c; - - __asm__ __volatile__( - ATOMIC_SMP_LOCK "incl %0; sete %1" - :"=m" (v->counter), "=qm" (c) - :"m" (v->counter) : "memory"); - return c != 0; -} - -/** - * atomic_add_negative - add and test if negative - * @v: pointer of type atomic_t - * @i: integer value to add - * - * Atomically adds @i to @v and returns true - * if the result is negative, or false when - * result is greater than or equal to zero. Note that the guaranteed - * useful range of an atomic_t is only 24 bits. - */ -static __inline__ int atomic_add_negative(int i, atomic_t *v) -{ - unsigned char c; - - __asm__ __volatile__( - ATOMIC_SMP_LOCK "addl %2,%0; sets %1" - :"=m" (v->counter), "=qm" (c) - :"ir" (i), "m" (v->counter) : "memory"); - return c; -} - -/* These are x86-specific, used by some header files */ -#define atomic_clear_mask(mask, addr) \ -__asm__ __volatile__(ATOMIC_SMP_LOCK "andl %0,%1" \ -: : "r" (~(mask)),"m" (*addr) : "memory") - -#define atomic_set_mask(mask, addr) \ -__asm__ __volatile__(ATOMIC_SMP_LOCK "orl %0,%1" \ -: : "r" (mask),"m" (*addr) : "memory") - -/* - * Force strict CPU ordering. - * And yes, this is required on UP too when we're talking - * to devices. - * - * For now, "wmb()" doesn't actually do anything, as all - * Intel CPU's follow what Intel calls a *Processor Order*, - * in which all writes are seen in the program order even - * outside the CPU. - * - * I expect future Intel CPU's to have a weaker ordering, - * but I'd also expect them to finally get their act together - * and add some real memory barriers if so. - */ - #ifdef __i386__ #define mb() __asm__ __volatile__ ("lock; addl $0,0(%%esp)": : :"memory") #define rmb() mb() #define wmb() __asm__ __volatile__ ("": : :"memory") -#else +#define IATOMIC_DEFINED 1 +#endif + +#ifdef __x86_64__ #define mb() asm volatile("mfence":::"memory") #define rmb() asm volatile("lfence":::"memory") #define wmb() asm volatile("sfence":::"memory") -#endif - -#undef ATOMIC_SMP_LOCK - #define IATOMIC_DEFINED 1 - -#endif /* __i386__ */ +#endif #ifdef __ia64__ - -/* - * On IA-64, counter must always be volatile to ensure that that the - * memory accesses are ordered. - */ -typedef struct { volatile int counter; } atomic_t; - -#define ATOMIC_INIT(i) ((atomic_t) { (i) }) - -#define atomic_read(v) ((v)->counter) -#define atomic_set(v,i) (((v)->counter) = (i)) - -/* stripped version - we need only 4byte version */ -#define ia64_cmpxchg(sem,ptr,old,new,size) \ -({ \ - __typeof__(ptr) _p_ = (ptr); \ - __typeof__(new) _n_ = (new); \ - unsigned long _o_, _r_; \ - _o_ = (unsigned int) (long) (old); \ - __asm__ __volatile__ ("mov ar.ccv=%0;;" :: "rO"(_o_)); \ - __asm__ __volatile__ ("cmpxchg4."sem" %0=[%1],%2,ar.ccv" \ - : "=r"(_r_) : "r"(_p_), "r"(_n_) : "memory"); \ - (__typeof__(old)) _r_; \ -}) - -static __inline__ int -ia64_atomic_add (int i, atomic_t *v) -{ - int old, new; - // CMPXCHG_BUGCHECK_DECL - - do { - // CMPXCHG_BUGCHECK(v); - old = atomic_read(v); - new = old + i; - } while (ia64_cmpxchg("acq", v, old, old + i, sizeof(atomic_t)) != old); - return new; -} - -static __inline__ int -ia64_atomic_sub (int i, atomic_t *v) -{ - int old, new; - // CMPXCHG_BUGCHECK_DECL - - do { - // CMPXCHG_BUGCHECK(v); - old = atomic_read(v); - new = old - i; - } while (ia64_cmpxchg("acq", v, old, new, sizeof(atomic_t)) != old); - return new; -} - -#define IA64_FETCHADD(tmp,v,n,sz) \ -({ \ - switch (sz) { \ - case 4: \ - __asm__ __volatile__ ("fetchadd4.rel %0=[%1],%2" \ - : "=r"(tmp) : "r"(v), "i"(n) : "memory"); \ - break; \ - \ - case 8: \ - __asm__ __volatile__ ("fetchadd8.rel %0=[%1],%2" \ - : "=r"(tmp) : "r"(v), "i"(n) : "memory"); \ - break; \ - } \ -}) - -#define ia64_fetch_and_add(i,v) \ -({ \ - unsigned long _tmp; \ - volatile __typeof__(*(v)) *_v = (v); \ - switch (i) { \ - case -16: IA64_FETCHADD(_tmp, _v, -16, sizeof(*(v))); break; \ - case -8: IA64_FETCHADD(_tmp, _v, -8, sizeof(*(v))); break; \ - case -4: IA64_FETCHADD(_tmp, _v, -4, sizeof(*(v))); break; \ - case -1: IA64_FETCHADD(_tmp, _v, -1, sizeof(*(v))); break; \ - case 1: IA64_FETCHADD(_tmp, _v, 1, sizeof(*(v))); break; \ - case 4: IA64_FETCHADD(_tmp, _v, 4, sizeof(*(v))); break; \ - case 8: IA64_FETCHADD(_tmp, _v, 8, sizeof(*(v))); break; \ - case 16: IA64_FETCHADD(_tmp, _v, 16, sizeof(*(v))); break; \ - } \ - (__typeof__(*v)) (_tmp + (i)); /* return new value */ \ -}) - -/* - * Atomically add I to V and return TRUE if the resulting value is - * negative. - */ -static __inline__ int -atomic_add_negative (int i, atomic_t *v) -{ - return ia64_atomic_add(i, v) < 0; -} - -#define atomic_add_return(i,v) \ - ((__builtin_constant_p(i) && \ - ( (i == 1) || (i == 4) || (i == 8) || (i == 16) \ - || (i == -1) || (i == -4) || (i == -8) || (i == -16))) \ - ? ia64_fetch_and_add(i, &(v)->counter) \ - : ia64_atomic_add(i, v)) - -#define atomic_sub_return(i,v) \ - ((__builtin_constant_p(i) && \ - ( (i == 1) || (i == 4) || (i == 8) || (i == 16) \ - || (i == -1) || (i == -4) || (i == -8) || (i == -16))) \ - ? ia64_fetch_and_add(-(i), &(v)->counter) \ - : ia64_atomic_sub(i, v)) - -#define atomic_dec_return(v) atomic_sub_return(1, (v)) -#define atomic_inc_return(v) atomic_add_return(1, (v)) - -#define atomic_sub_and_test(i,v) (atomic_sub_return((i), (v)) == 0) -#define atomic_dec_and_test(v) (atomic_sub_return(1, (v)) == 0) -#define atomic_inc_and_test(v) (atomic_add_return(1, (v)) != 0) - -#define atomic_add(i,v) atomic_add_return((i), (v)) -#define atomic_sub(i,v) atomic_sub_return((i), (v)) -#define atomic_inc(v) atomic_add(1, (v)) -#define atomic_dec(v) atomic_sub(1, (v)) - /* * Macros to force memory ordering. In these descriptions, "previous" * and "subsequent" refer to program order; "visible" means that all @@ -376,111 +48,6 @@ atomic_add_negative (int i, atomic_t *v) #ifdef __alpha__ -/* - * Atomic operations that C can't guarantee us. Useful for - * resource counting etc... - * - * But use these as seldom as possible since they are much slower - * than regular operations. - */ - - -/* - * Counter is volatile to make sure gcc doesn't try to be clever - * and move things around on us. We need to use _exactly_ the address - * the user gave us, not some alias that contains the same information. - */ -typedef struct { volatile int counter; } atomic_t; - -#define ATOMIC_INIT(i) ( (atomic_t) { (i) } ) - -#define atomic_read(v) ((v)->counter) -#define atomic_set(v,i) ((v)->counter = (i)) - -/* - * To get proper branch prediction for the main line, we must branch - * forward to code at the end of this object's .text section, then - * branch back to restart the operation. - */ - -static __inline__ void atomic_add(int i, atomic_t * v) -{ - unsigned long temp; - __asm__ __volatile__( - "1: ldl_l %0,%1\n" - " addl %0,%2,%0\n" - " stl_c %0,%1\n" - " beq %0,2f\n" - ".subsection 2\n" - "2: br 1b\n" - ".previous" - :"=&r" (temp), "=m" (v->counter) - :"Ir" (i), "m" (v->counter)); -} - -static __inline__ void atomic_sub(int i, atomic_t * v) -{ - unsigned long temp; - __asm__ __volatile__( - "1: ldl_l %0,%1\n" - " subl %0,%2,%0\n" - " stl_c %0,%1\n" - " beq %0,2f\n" - ".subsection 2\n" - "2: br 1b\n" - ".previous" - :"=&r" (temp), "=m" (v->counter) - :"Ir" (i), "m" (v->counter)); -} - -/* - * Same as above, but return the result value - */ -static __inline__ long atomic_add_return(int i, atomic_t * v) -{ - long temp, result; - __asm__ __volatile__( - "1: ldl_l %0,%1\n" - " addl %0,%3,%2\n" - " addl %0,%3,%0\n" - " stl_c %0,%1\n" - " beq %0,2f\n" - " mb\n" - ".subsection 2\n" - "2: br 1b\n" - ".previous" - :"=&r" (temp), "=m" (v->counter), "=&r" (result) - :"Ir" (i), "m" (v->counter) : "memory"); - return result; -} - -static __inline__ long atomic_sub_return(int i, atomic_t * v) -{ - long temp, result; - __asm__ __volatile__( - "1: ldl_l %0,%1\n" - " subl %0,%3,%2\n" - " subl %0,%3,%0\n" - " stl_c %0,%1\n" - " beq %0,2f\n" - " mb\n" - ".subsection 2\n" - "2: br 1b\n" - ".previous" - :"=&r" (temp), "=m" (v->counter), "=&r" (result) - :"Ir" (i), "m" (v->counter) : "memory"); - return result; -} - -#define atomic_dec_return(v) atomic_sub_return(1,(v)) -#define atomic_inc_return(v) atomic_add_return(1,(v)) - -#define atomic_sub_and_test(i,v) (atomic_sub_return((i), (v)) == 0) -#define atomic_dec_and_test(v) (atomic_sub_return(1, (v)) == 0) - -#define atomic_inc(v) atomic_add(1,(v)) -#define atomic_dec(v) atomic_sub(1,(v)) - #define mb() \ __asm__ __volatile__("mb": : :"memory") @@ -496,168 +63,6 @@ __asm__ __volatile__("wmb": : :"memory") #ifdef __powerpc__ -typedef struct { volatile int counter; } atomic_t; - -#define ATOMIC_INIT(i) { (i) } - -#define atomic_read(v) ((v)->counter) -#define atomic_set(v,i) (((v)->counter) = (i)) - -extern void atomic_clear_mask(unsigned long mask, unsigned long *addr); -extern void atomic_set_mask(unsigned long mask, unsigned long *addr); - -#define SMP_ISYNC "\n\tisync" - -static __inline__ void atomic_add(int a, atomic_t *v) -{ - int t; - - __asm__ __volatile__( -"1: lwarx %0,0,%3 # atomic_add\n\ - add %0,%2,%0\n\ - stwcx. %0,0,%3\n\ - bne- 1b" - : "=&r" (t), "=m" (v->counter) - : "r" (a), "r" (&v->counter), "m" (v->counter) - : "cc"); -} - -static __inline__ int atomic_add_return(int a, atomic_t *v) -{ - int t; - - __asm__ __volatile__( -"1: lwarx %0,0,%2 # atomic_add_return\n\ - add %0,%1,%0\n\ - stwcx. %0,0,%2\n\ - bne- 1b" - SMP_ISYNC - : "=&r" (t) - : "r" (a), "r" (&v->counter) - : "cc", "memory"); - - return t; -} - -static __inline__ void atomic_sub(int a, atomic_t *v) -{ - int t; - - __asm__ __volatile__( -"1: lwarx %0,0,%3 # atomic_sub\n\ - subf %0,%2,%0\n\ - stwcx. %0,0,%3\n\ - bne- 1b" - : "=&r" (t), "=m" (v->counter) - : "r" (a), "r" (&v->counter), "m" (v->counter) - : "cc"); -} - -static __inline__ int atomic_sub_return(int a, atomic_t *v) -{ - int t; - - __asm__ __volatile__( -"1: lwarx %0,0,%2 # atomic_sub_return\n\ - subf %0,%1,%0\n\ - stwcx. %0,0,%2\n\ - bne- 1b" - SMP_ISYNC - : "=&r" (t) - : "r" (a), "r" (&v->counter) - : "cc", "memory"); - - return t; -} - -static __inline__ void atomic_inc(atomic_t *v) -{ - int t; - - __asm__ __volatile__( -"1: lwarx %0,0,%2 # atomic_inc\n\ - addic %0,%0,1\n\ - stwcx. %0,0,%2\n\ - bne- 1b" - : "=&r" (t), "=m" (v->counter) - : "r" (&v->counter), "m" (v->counter) - : "cc"); -} - -static __inline__ int atomic_inc_return(atomic_t *v) -{ - int t; - - __asm__ __volatile__( -"1: lwarx %0,0,%1 # atomic_inc_return\n\ - addic %0,%0,1\n\ - stwcx. %0,0,%1\n\ - bne- 1b" - SMP_ISYNC - : "=&r" (t) - : "r" (&v->counter) - : "cc", "memory"); - - return t; -} - -static __inline__ void atomic_dec(atomic_t *v) -{ - int t; - - __asm__ __volatile__( -"1: lwarx %0,0,%2 # atomic_dec\n\ - addic %0,%0,-1\n\ - stwcx. %0,0,%2\n\ - bne- 1b" - : "=&r" (t), "=m" (v->counter) - : "r" (&v->counter), "m" (v->counter) - : "cc"); -} - -static __inline__ int atomic_dec_return(atomic_t *v) -{ - int t; - - __asm__ __volatile__( -"1: lwarx %0,0,%1 # atomic_dec_return\n\ - addic %0,%0,-1\n\ - stwcx. %0,0,%1\n\ - bne- 1b" - SMP_ISYNC - : "=&r" (t) - : "r" (&v->counter) - : "cc", "memory"); - - return t; -} - -#define atomic_sub_and_test(a, v) (atomic_sub_return((a), (v)) == 0) -#define atomic_dec_and_test(v) (atomic_dec_return((v)) == 0) - -/* - * Atomically test *v and decrement if it is greater than 0. - * The function returns the old value of *v minus 1. - */ -static __inline__ int atomic_dec_if_positive(atomic_t *v) -{ - int t; - - __asm__ __volatile__( -"1: lwarx %0,0,%1 # atomic_dec_if_positive\n\ - addic. %0,%0,-1\n\ - blt- 2f\n\ - stwcx. %0,0,%1\n\ - bne- 1b" - SMP_ISYNC - "\n\ -2:" : "=&r" (t) - : "r" (&v->counter) - : "cc", "memory"); - - return t; -} - /* * Memory barrier. * The sync instruction guarantees that all memory accesses initiated @@ -682,447 +87,14 @@ static __inline__ int atomic_dec_if_positive(atomic_t *v) #endif /* __powerpc__ */ -#ifdef __mips__ - -typedef struct { volatile int counter; } atomic_t; - -#define ATOMIC_INIT(i) { (i) } - -/* - * atomic_read - read atomic variable - * @v: pointer of type atomic_t - * - * Atomically reads the value of @v. Note that the guaranteed - * useful range of an atomic_t is only 24 bits. - */ -#define atomic_read(v) ((v)->counter) - -/* - * atomic_set - set atomic variable - * @v: pointer of type atomic_t - * @i: required value - * - * Atomically sets the value of @v to @i. Note that the guaranteed - * useful range of an atomic_t is only 24 bits. - */ -#define atomic_set(v,i) ((v)->counter = (i)) - -/* - * for MIPS II and better we can use ll/sc instruction, and kernel 2.4.3+ - * will emulate it on MIPS I. - */ - -/* - * atomic_add - add integer to atomic variable - * @i: integer value to add - * @v: pointer of type atomic_t - * - * Atomically adds @i to @v. Note that the guaranteed useful range - * of an atomic_t is only 24 bits. - */ -static __inline__ void atomic_add(int i, atomic_t * v) -{ - unsigned long temp; - - __asm__ __volatile__( - ".set push \n" - ".set mips2 \n" - "1: ll %0, %1 # atomic_add\n" - " addu %0, %2 \n" - " sc %0, %1 \n" - " beqz %0, 1b \n" - ".set pop \n" - : "=&r" (temp), "=m" (v->counter) - : "Ir" (i), "m" (v->counter)); -} - -/* - * atomic_sub - subtract the atomic variable - * @i: integer value to subtract - * @v: pointer of type atomic_t - * - * Atomically subtracts @i from @v. Note that the guaranteed - * useful range of an atomic_t is only 24 bits. - */ -static __inline__ void atomic_sub(int i, atomic_t * v) -{ - unsigned long temp; - - __asm__ __volatile__( - ".set push \n" - ".set mips2 \n" - "1: ll %0, %1 # atomic_sub\n" - " subu %0, %2 \n" - " sc %0, %1 \n" - " beqz %0, 1b \n" - ".set pop \n" - : "=&r" (temp), "=m" (v->counter) - : "Ir" (i), "m" (v->counter)); -} - -/* - * Same as above, but return the result value - */ -static __inline__ int atomic_add_return(int i, atomic_t * v) -{ - unsigned long temp, result; - - __asm__ __volatile__( - ".set push # atomic_add_return\n" - ".set noreorder \n" - ".set mips2 \n" - "1: ll %1, %2 \n" - " addu %0, %1, %3 \n" - " sc %0, %2 \n" - " beqz %0, 1b \n" - " addu %0, %1, %3 \n" - ".set pop \n" - : "=&r" (result), "=&r" (temp), "=m" (v->counter) - : "Ir" (i), "m" (v->counter) - : "memory"); - - return result; -} - -static __inline__ int atomic_sub_return(int i, atomic_t * v) -{ - unsigned long temp, result; - - __asm__ __volatile__( - ".set push \n" - ".set mips2 \n" - ".set noreorder # atomic_sub_return\n" - "1: ll %1, %2 \n" - " subu %0, %1, %3 \n" - " sc %0, %2 \n" - " beqz %0, 1b \n" - " subu %0, %1, %3 \n" - ".set pop \n" - : "=&r" (result), "=&r" (temp), "=m" (v->counter) - : "Ir" (i), "m" (v->counter) - : "memory"); - - return result; -} - -#define atomic_dec_return(v) atomic_sub_return(1,(v)) -#define atomic_inc_return(v) atomic_add_return(1,(v)) - -/* - * atomic_sub_and_test - subtract value from variable and test result - * @i: integer value to subtract - * @v: pointer of type atomic_t - * - * Atomically subtracts @i from @v and returns - * true if the result is zero, or false for all - * other cases. Note that the guaranteed - * useful range of an atomic_t is only 24 bits. - */ -#define atomic_sub_and_test(i,v) (atomic_sub_return((i), (v)) == 0) - -/* - * atomic_inc_and_test - increment and test - * @v: pointer of type atomic_t - * - * Atomically increments @v by 1 - * and returns true if the result is zero, or false for all - * other cases. Note that the guaranteed - * useful range of an atomic_t is only 24 bits. - */ -#define atomic_inc_and_test(v) (atomic_inc_return(1, (v)) == 0) - -/* - * atomic_dec_and_test - decrement by 1 and test - * @v: pointer of type atomic_t - * - * Atomically decrements @v by 1 and - * returns true if the result is 0, or false for all other - * cases. Note that the guaranteed - * useful range of an atomic_t is only 24 bits. - */ -#define atomic_dec_and_test(v) (atomic_sub_return(1, (v)) == 0) - -/* - * atomic_inc - increment atomic variable - * @v: pointer of type atomic_t - * - * Atomically increments @v by 1. Note that the guaranteed - * useful range of an atomic_t is only 24 bits. - */ -#define atomic_inc(v) atomic_add(1,(v)) - -/* - * atomic_dec - decrement and test - * @v: pointer of type atomic_t - * - * Atomically decrements @v by 1. Note that the guaranteed - * useful range of an atomic_t is only 24 bits. - */ -#define atomic_dec(v) atomic_sub(1,(v)) - -/* - * atomic_add_negative - add and test if negative - * @v: pointer of type atomic_t - * @i: integer value to add - * - * Atomically adds @i to @v and returns true - * if the result is negative, or false when - * result is greater than or equal to zero. Note that the guaranteed - * useful range of an atomic_t is only 24 bits. - * - * Currently not implemented for MIPS. - */ - -#define mb() \ -__asm__ __volatile__( \ - "# prevent instructions being moved around\n\t" \ - ".set\tnoreorder\n\t" \ - "# 8 nops to fool the R4400 pipeline\n\t" \ - "nop;nop;nop;nop;nop;nop;nop;nop\n\t" \ - ".set\treorder" \ - : /* no output */ \ - : /* no input */ \ - : "memory") -#define rmb() mb() -#define wmb() mb() - -#define IATOMIC_DEFINED 1 - -#endif /* __mips__ */ - -#ifdef __arm__ - -/* - * FIXME: bellow code is valid only for SA11xx - */ - -/* - * Save the current interrupt enable state & disable IRQs - */ -#define local_irq_save(x) \ - ({ \ - unsigned long temp; \ - __asm__ __volatile__( \ - "mrs %0, cpsr @ local_irq_save\n" \ -" orr %1, %0, #128\n" \ -" msr cpsr_c, %1" \ - : "=r" (x), "=r" (temp) \ - : \ - : "memory"); \ - }) - -/* - * restore saved IRQ & FIQ state - */ -#define local_irq_restore(x) \ - __asm__ __volatile__( \ - "msr cpsr_c, %0 @ local_irq_restore\n" \ - : \ - : "r" (x) \ - : "memory") - -#define __save_flags_cli(x) local_irq_save(x) -#define __restore_flags(x) local_irq_restore(x) - -typedef struct { volatile int counter; } atomic_t; - -#define ATOMIC_INIT(i) { (i) } - -#define atomic_read(v) ((v)->counter) -#define atomic_set(v,i) (((v)->counter) = (i)) - -static __inline__ void atomic_add(int i, volatile atomic_t *v) -{ - unsigned long flags; - - __save_flags_cli(flags); - v->counter += i; - __restore_flags(flags); -} - -static __inline__ void atomic_sub(int i, volatile atomic_t *v) -{ - unsigned long flags; - - __save_flags_cli(flags); - v->counter -= i; - __restore_flags(flags); -} - -static __inline__ void atomic_inc(volatile atomic_t *v) -{ - unsigned long flags; - - __save_flags_cli(flags); - v->counter += 1; - __restore_flags(flags); -} - -static __inline__ void atomic_dec(volatile atomic_t *v) -{ - unsigned long flags; - - __save_flags_cli(flags); - v->counter -= 1; - __restore_flags(flags); -} - -static __inline__ int atomic_dec_and_test(volatile atomic_t *v) -{ - unsigned long flags; - int result; - - __save_flags_cli(flags); - v->counter -= 1; - result = (v->counter == 0); - __restore_flags(flags); - - return result; -} - -static __inline__ int atomic_add_negative(int i, volatile atomic_t *v) -{ - unsigned long flags; - int result; - - __save_flags_cli(flags); - v->counter += i; - result = (v->counter < 0); - __restore_flags(flags); - - return result; -} - -static __inline__ void atomic_clear_mask(unsigned long mask, unsigned long *addr) -{ - unsigned long flags; - - __save_flags_cli(flags); - *addr &= ~mask; - __restore_flags(flags); -} - -#define mb() __asm__ __volatile__ ("" : : : "memory") -#define rmb() mb() -#define wmb() mb() - -#define IATOMIC_DEFINED 1 - -#endif /* __arm__ */ - -#ifdef __sh__ - -typedef struct { volatile int counter; } atomic_t; - -#define ATOMIC_INIT(i) { (i) } - -#define atomic_read(v) ((v)->counter) -#define atomic_set(v,i) (((v)->counter) = (i)) - -#define atomic_dec_return(v) atomic_sub_return(1,(v)) -#define atomic_inc_return(v) atomic_add_return(1,(v)) - -#define atomic_sub_and_test(i,v) (atomic_sub_return((i), (v)) == 0) -#define atomic_dec_and_test(v) (atomic_sub_return(1, (v)) == 0) -#define atomic_inc_and_test(v) (atomic_add_return(1, (v)) != 0) - -#define atomic_add(i,v) atomic_add_return((i),(v)) -#define atomic_sub(i,v) atomic_sub_return((i),(v)) -#define atomic_inc(v) atomic_add(1,(v)) -#define atomic_dec(v) atomic_sub(1,(v)) - -static __inline__ int atomic_add_return(int i, volatile atomic_t *v) -{ - int result; - - asm volatile ( - " .align 2\n" - " mova 99f, r0\n" - " mov r15, r1\n" - " mov #-6, r15\n" - " mov.l @%2, %0\n" - " add %1, %0\n" - " mov.l %0, @%2\n" - "99: mov r1, r15" - : "=&r"(result) - : "r"(i), "r"(v) - : "r0", "r1"); - - return result; -} - -static __inline__ int atomic_sub_return(int i, volatile atomic_t *v) -{ - int result; - - asm volatile ( - " .align 2\n" - " mova 99f, r0\n" - " mov r15, r1\n" - " mov #-6, r15\n" - " mov.l @%2, %0\n" - " sub %1, %0\n" - " mov.l %0, @%2\n" - "99: mov r1, r15" - : "=&r"(result) - : "r"(i), "r"(v) - : "r0", "r1"); - - return result; -} - -#define mb() __asm__ __volatile__ ("" : : : "memory") -#define rmb() mb() -#define wmb() mb() - -#define IATOMIC_DEFINED 1 - -#endif /* __sh__ */ - -#ifdef __bfin__ - -#include <bfin_fixed_code.h> - -typedef struct { volatile int counter; } atomic_t; - -#define ATOMIC_INIT(i) { (i) } +#ifndef IATOMIC_DEFINED -#define atomic_read(v) ((v)->counter) -#define atomic_set(v,i) (((v)->counter) = (i)) -#define atomic_add(i,v) bfin_atomic_add32(&(v)->counter, i) -#define atomic_sub(i,v) bfin_atomic_sub32(&(v)->counter, i) -#define atomic_inc(v) bfin_atomic_inc32(&(v)->counter); -#define atomic_dec(v) bfin_atomic_dec32(&(v)->counter); +/* Generic __sync_synchronize is available from gcc 4.1 */ -#define mb() __asm__ __volatile__ ("" : : : "memory") +#define mb() __sync_synchronize() #define rmb() mb() #define wmb() mb() -#define IATOMIC_DEFINED 1 - -#endif /* __bfin__ */ - -#ifndef IATOMIC_DEFINED -/* - * non supported architecture. - */ -#warning "Atomic operations are not supported on this architecture." - -typedef struct { volatile int counter; } atomic_t; - -#define ATOMIC_INIT(i) { (i) } - -#define atomic_read(v) ((v)->counter) -#define atomic_set(v,i) (((v)->counter) = (i)) -#define atomic_add(i,v) (((v)->counter) += (i)) -#define atomic_sub(i,v) (((v)->counter) -= (i)) -#define atomic_inc(v) (((v)->counter)++) -#define atomic_dec(v) (((v)->counter)--) - -#define mb() -#define rmb() -#define wmb() - #define IATOMIC_DEFINED 1 #endif /* IATOMIC_DEFINED */ diff --git a/src/pcm/pcm_meter.c b/src/pcm/pcm_meter.c index 573b6fe..676fbef 100644 --- a/src/pcm/pcm_meter.c +++ b/src/pcm/pcm_meter.c @@ -33,7 +33,10 @@ #include <dlfcn.h> #include "pcm_local.h" #include "pcm_plugin.h" -#include "iatomic.h" + +#define atomic_read(ptr) __atomic_load_n(ptr, __ATOMIC_SEQ_CST ) +#define atomic_add(ptr, n) __atomic_add_fetch(ptr, n, __ATOMIC_SEQ_CST) +#define atomic_dec(ptr) __atomic_sub_fetch(ptr, 1, __ATOMIC_SEQ_CST) #ifndef PIC /* entry for static linking */ @@ -61,7 +64,7 @@ typedef struct _snd_pcm_meter { struct list_head scopes; int closed; int running; - atomic_t reset; + int reset; pthread_t thread; pthread_mutex_t update_mutex; pthread_mutex_t running_mutex; @@ -288,7 +291,7 @@ static int snd_pcm_meter_prepare(snd_pcm_t *pcm) { snd_pcm_meter_t *meter = pcm->private_data; int err; - atomic_inc(&meter->reset); + atomic_add(&meter->reset, 1); err = snd_pcm_prepare(meter->gen.slave); if (err >= 0) { if (pcm->stream == SND_PCM_STREAM_PLAYBACK) -- 1.8.1.2

3 2

[alsa-devel] [PATCH 1/2] ASoC: ak4671: Convert to table based control init
by Mark Brown 11 Feb '14

11 Feb '14

From: Mark Brown <broonie(a)linaro.org> Saves code and adds error handling. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/ak4671.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index 25bdf6ad4a54..456bd0a065b1 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -628,9 +628,6 @@ static int ak4671_probe(struct snd_soc_codec *codec) return ret; } - snd_soc_add_codec_controls(codec, ak4671_snd_controls, - ARRAY_SIZE(ak4671_snd_controls)); - ak4671_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return ret; @@ -646,6 +643,8 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4671 = { .probe = ak4671_probe, .remove = ak4671_remove, .set_bias_level = ak4671_set_bias_level, + .controls = ak4671_snd_controls, + .num_controls = ARRAY_SIZE(ak4671_snd_controls), .reg_cache_size = AK4671_CACHEREGNUM, .reg_word_size = sizeof(u8), .reg_cache_default = ak4671_reg, -- 1.9.0.rc3

1 1

[alsa-devel] [PATCH 1/2] ASoC: cs42l51: Don't log if we fail to allocate memory
by Mark Brown 11 Feb '14

11 Feb '14

From: Mark Brown <broonie(a)linaro.org> The VM subsystem already logs quite loudly if we run out of memory so don't bother here. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/cs42l51.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 6e9ea8379a91..50c08c8fc9bc 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -574,10 +574,8 @@ static int cs42l51_i2c_probe(struct i2c_client *i2c_client, cs42l51 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs42l51_private), GFP_KERNEL); - if (!cs42l51) { - dev_err(&i2c_client->dev, "could not allocate codec\n"); + if (!cs42l51) return -ENOMEM; - } i2c_set_clientdata(i2c_client, cs42l51); cs42l51->control_type = SND_SOC_I2C; -- 1.9.0.rc3

2 3

[alsa-devel] [PATCH] ASoC: smdk_wm8994: Fix typo "wm8894"
by Paul Bolle 11 Feb '14

11 Feb '14

Signed-off-by: Paul Bolle <pebolle(a)tiscali.nl> --- 0) This typo was introduced in commit 0d47acc4ffaa ("ASoC: smdk_wm8994: Make driver name more unique"). 1) So currently there's a difference between platform_drive.driver.name and the (sub)string used in MODULE_ALIAS(). If I'd understand the platform_driver interface I might have been able to say whether that difference is only cosmetic or might actually break stuff. But I don't understand it so I can't say. I can't even say whether this typo shows up in some interface to user space. 2) Entirely untested. sound/soc/samsung/smdk_wm8994.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index d38ae98..682eb4f 100644 --- a/sound/soc/samsung/smdk_wm8994.c +++ b/sound/soc/samsung/smdk_wm8994.c @@ -202,7 +202,7 @@ static int smdk_audio_probe(struct platform_device *pdev) static struct platform_driver smdk_audio_driver = { .driver = { - .name = "smdk-audio-wm8894", + .name = "smdk-audio-wm8994", .owner = THIS_MODULE, .of_match_table = of_match_ptr(samsung_wm8994_of_match), .pm = &snd_soc_pm_ops, -- 1.8.5.3

2 1

[alsa-devel] [PATCH] ASoC: fsl_esai: Check the return value from clk_prepare_enable()
by Fabio Estevam 11 Feb '14

11 Feb '14

clk_prepare_enable() may fail, so let's check its return value and propagate it in the case of error. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_esai.c | 27 ++++++++++++++++++++++----- 1 file changed, 22 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index d0c72ed..e64115c 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -431,17 +431,26 @@ static int fsl_esai_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) static int fsl_esai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { + int ret; struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); /* * Some platforms might use the same bit to gate all three or two of * clocks, so keep all clocks open/close at the same time for safety */ - clk_prepare_enable(esai_priv->coreclk); - if (!IS_ERR(esai_priv->extalclk)) - clk_prepare_enable(esai_priv->extalclk); - if (!IS_ERR(esai_priv->fsysclk)) - clk_prepare_enable(esai_priv->fsysclk); + ret = clk_prepare_enable(esai_priv->coreclk); + if (ret) + return ret; + if (!IS_ERR(esai_priv->extalclk)) { + ret = clk_prepare_enable(esai_priv->extalclk); + if (ret) + goto err_extalck; + } + if (!IS_ERR(esai_priv->fsysclk)) { + ret = clk_prepare_enable(esai_priv->fsysclk); + if (ret) + goto err_fsysclk; + } if (!dai->active) { /* Reset Port C */ @@ -463,6 +472,14 @@ static int fsl_esai_startup(struct snd_pcm_substream *substream, } return 0; + +err_fsysclk: + if (!IS_ERR(esai_priv->extalclk)) + clk_disable_unprepare(esai_priv->extalclk); +err_extalck: + clk_disable_unprepare(esai_priv->coreclk); + + return ret; } static int fsl_esai_hw_params(struct snd_pcm_substream *substream, -- 1.8.1.2

2 1

[alsa-devel] [PATCH 0/7] Audio support for Armada 370 DB
by Thomas Petazzoni 11 Feb '14

11 Feb '14

Hello, This series of patches enable audio support on the Marvell Armada 370 Development Board. Since both the I2S controller on the SoC side and the I2C audio codec are already supported by the kernel, the amount of work is fairly limited. Also, since the DT bindings that allows to replace the ASoC board driver are still being ironed out, I'm proposing to add an old-style ASoC board driver for now. This set of patches only provide support for the analog output and input, I am waiting for the optical cable to arrive to get the digital input and output to work. Patches 1 to 3 are to be taken by the ASoC maintainer, while patches 4 to 7 are to be taken by the mvebu maintainers. Note that the audio support for Armada 370 also needs a fix to the CS42L51, which is being discussed with the author of the change that apparently introduced the problem (see discussion at http://mailman.alsa-project.org/pipermail/alsa-devel/2014-January/071916.ht…). Thanks, Thomas Thomas Petazzoni (7): sound: codec: add Device Tree binding to cs42l51 sound: soc: enable Kirkwood driver for mvebu platforms sound: soc: add ASoC board driver for Armada 370 DB ARM: mvebu: add audio I2S controller to Armada 370 Device Tree ARM: mvebu: add I2C0 muxing option for Armada 370 SoC ARM: mvebu: add audio support to Armada 370 DB ARM: mvebu: enable audio options in mvebu_defconfig .../devicetree/bindings/i2c/trivial-devices.txt | 1 + .../devicetree/bindings/sound/mvebu-audio.txt | 1 + arch/arm/boot/dts/armada-370-db.dts | 46 +++++++ arch/arm/boot/dts/armada-370.dtsi | 29 +++++ arch/arm/configs/mvebu_defconfig | 5 + sound/soc/codecs/cs42l51.c | 7 + sound/soc/kirkwood/Kconfig | 10 +- sound/soc/kirkwood/Makefile | 2 + sound/soc/kirkwood/armada-370-db.c | 144 +++++++++++++++++++++ sound/soc/kirkwood/kirkwood-i2s.c | 1 + 10 files changed, 245 insertions(+), 1 deletion(-) create mode 100644 sound/soc/kirkwood/armada-370-db.c -- 1.8.3.2

3 15

[alsa-devel] [PATCH] ASoC: wm8995: Remove unused pointer in hp_supply_event()
by Christian Engelmayer 11 Feb '14

11 Feb '14

Remove unused driver data pointer 'wm8995' in function hp_supply_event(). Detected by Coverity: CID 141181. Signed-off-by: Christian Engelmayer <cengelma(a)gmx.at> --- Applies against v3.14-rc1 as well as branch for-next in tree git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git. --- sound/soc/codecs/wm8995.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 4300caf..403e1db 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -561,10 +561,8 @@ static int hp_supply_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec; - struct wm8995_priv *wm8995; codec = w->codec; - wm8995 = snd_soc_codec_get_drvdata(codec); switch (event) { case SND_SOC_DAPM_PRE_PMU: -- 1.8.3.2

2 1

[alsa-devel] [PATCH 1/2] ASoC: fsl-spdif: big-endian support
by Xiubo Li 11 Feb '14

11 Feb '14

For most platforms, the CPU and SPDIF device is in the same endianess mode. While for the LS1 platform, the CPU is in LE mode and the SPDIF is in BE mode. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> Cc: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- Documentation/devicetree/bindings/sound/fsl,spdif.txt | 4 ++++ sound/soc/fsl/fsl_spdif.c | 8 +++++++- 2 files changed, 11 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.txt b/Documentation/devicetree/bindings/sound/fsl,spdif.txt index f2ae335..433ad00 100644 --- a/Documentation/devicetree/bindings/sound/fsl,spdif.txt +++ b/Documentation/devicetree/bindings/sound/fsl,spdif.txt @@ -29,6 +29,10 @@ Required properties: can also be referred to TxClk_Source bit of register SPDIF_STC. + - big-endian : If this property is absent, the native endian mode will + be in use as default, or the big endian mode will be in use for all the + device registers. + Example: spdif: spdif@02004000 { diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 8f36f49..2f8ad75 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -83,6 +83,8 @@ struct fsl_spdif_priv { struct snd_dmaengine_dai_dma_data dma_params_tx; struct snd_dmaengine_dai_dma_data dma_params_rx; + bool big_endian; + /* The name space will be allocated dynamically */ char name[0]; }; @@ -985,7 +987,7 @@ static bool fsl_spdif_writeable_reg(struct device *dev, unsigned int reg) } } -static const struct regmap_config fsl_spdif_regmap_config = { +static struct regmap_config fsl_spdif_regmap_config = { .reg_bits = 32, .reg_stride = 4, .val_bits = 32, @@ -1105,6 +1107,10 @@ static int fsl_spdif_probe(struct platform_device *pdev) memcpy(&spdif_priv->cpu_dai_drv, &fsl_spdif_dai, sizeof(fsl_spdif_dai)); spdif_priv->cpu_dai_drv.name = spdif_priv->name; + spdif_priv->big_endian = of_property_read_bool(np, "big-endian"); + if (spdif_priv->big_endian) + fsl_spdif_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG; + /* Get the addresses and IRQ */ res = platform_get_resource(pdev, IORESOURCE_MEM, 0); regs = devm_ioremap_resource(&pdev->dev, res); -- 1.8.4

3 7

[alsa-devel] Noise on right channel of WM9715
by Andreas Zeiler 11 Feb '14

11 Feb '14

Hi, we have a board with an iMX6 processor from freescale and a Wolfson WM9715. I can play music/sound files with it. We're on Linux 3.0.35. The problem is that there is a noise on the right channel and this noise changes with the frequency. I don't mean the sample rate with that. I'm playing a wav-file with a Sinus from 440Hz to 880Hz or higher. I can hear how the noise is changing. Then I played a silence file with nothing in it. When I see this file it has some small artefacts (in audacity). On the left channel I hear nothing as expected. On the right channel I hear some clicking. It's not periodic but I hear it very often. So I turned off everything I don't need in the WM9715. So just the DAC (PCM (Register 0x18)) and the Headphone Mixer (register 0x04) are turned on. Everything else is in default state or muted. AC97 communication is working. I have a good clock etc. I think that there is something wrong when my audio data gets into the ac97 slot. Maybe it gets shifted wrong or something else. Can someone help me with this? Does someone know what can be wrong here?? Mit freundlichen Grüßen / With kind regards Andreas Zeiler (zeiler(a)fs-net.de<mailto:zeiler@fs-net.de>) ________________________________ armStone: Pico-ITX ARM based Single Board Computer ________________________________ F&S Elektronik Systeme GmbH, Stuttgart, Germany Geschaeftsfuehrer: Holger Froelich, Guenther Scholz Handelsregister Stuttgart HRB 18493, VAT DE178964418

1 0

[alsa-devel] [PATCH v2] ASoC: fsl-sai: convert to use regmap API for Freeacale SAI
by Xiubo Li 11 Feb '14

11 Feb '14

Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> Cc: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- This version applies against sound-3.14-rc1. sound/soc/fsl/fsl_sai.c | 255 +++++++++++++++++++++++++++--------------------- sound/soc/fsl/fsl_sai.h | 47 +++++---- 2 files changed, 166 insertions(+), 136 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index cdd3fa8..faa65af 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -15,6 +15,7 @@ #include <linux/dmaengine.h> #include <linux/module.h> #include <linux/of_address.h> +#include <linux/regmap.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/dmaengine_pcm.h> @@ -22,34 +23,6 @@ #include "fsl_sai.h" -static inline u32 sai_readl(struct fsl_sai *sai, - const void __iomem *addr) -{ - u32 val; - - val = __raw_readl(addr); - - if (likely(sai->big_endian_regs)) - val = be32_to_cpu(val); - else - val = le32_to_cpu(val); - rmb(); - - return val; -} - -static inline void sai_writel(struct fsl_sai *sai, - u32 val, void __iomem *addr) -{ - wmb(); - if (likely(sai->big_endian_regs)) - val = cpu_to_be32(val); - else - val = cpu_to_le32(val); - - __raw_writel(val, addr); -} - static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int fsl_dir) { @@ -61,7 +34,8 @@ static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, else reg_cr2 = FSL_SAI_RCR2; - val_cr2 = sai_readl(sai, sai->base + reg_cr2); + regmap_read(sai->regmap, reg_cr2, &val_cr2); + val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK; switch (clk_id) { @@ -81,7 +55,7 @@ static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, return -EINVAL; } - sai_writel(sai, val_cr2, sai->base + reg_cr2); + regmap_write(sai->regmap, reg_cr2, val_cr2); return 0; } @@ -89,32 +63,22 @@ static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, static int fsl_sai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { - struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); int ret; if (dir == SND_SOC_CLOCK_IN) return 0; - ret = clk_prepare_enable(sai->clk); - if (ret) - return ret; - ret = fsl_sai_set_dai_sysclk_tr(cpu_dai, clk_id, freq, FSL_FMT_TRANSMITTER); if (ret) { dev_err(cpu_dai->dev, "Cannot set tx sysclk: %d\n", ret); - goto err_clk; + return ret; } ret = fsl_sai_set_dai_sysclk_tr(cpu_dai, clk_id, freq, FSL_FMT_RECEIVER); - if (ret) { + if (ret) dev_err(cpu_dai->dev, "Cannot set rx sysclk: %d\n", ret); - goto err_clk; - } - -err_clk: - clk_disable_unprepare(sai->clk); return ret; } @@ -133,8 +97,8 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, reg_cr4 = FSL_SAI_RCR4; } - val_cr2 = sai_readl(sai, sai->base + reg_cr2); - val_cr4 = sai_readl(sai, sai->base + reg_cr4); + regmap_read(sai->regmap, reg_cr2, &val_cr2); + regmap_read(sai->regmap, reg_cr4, &val_cr4); if (sai->big_endian_data) val_cr4 &= ~FSL_SAI_CR4_MF; @@ -183,35 +147,25 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, return -EINVAL; } - sai_writel(sai, val_cr2, sai->base + reg_cr2); - sai_writel(sai, val_cr4, sai->base + reg_cr4); + regmap_write(sai->regmap, reg_cr2, val_cr2); + regmap_write(sai->regmap, reg_cr4, val_cr4); return 0; } static int fsl_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { - struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); int ret; - ret = clk_prepare_enable(sai->clk); - if (ret) - return ret; - ret = fsl_sai_set_dai_fmt_tr(cpu_dai, fmt, FSL_FMT_TRANSMITTER); if (ret) { dev_err(cpu_dai->dev, "Cannot set tx format: %d\n", ret); - goto err_clk; + return ret; } ret = fsl_sai_set_dai_fmt_tr(cpu_dai, fmt, FSL_FMT_RECEIVER); - if (ret) { + if (ret) dev_err(cpu_dai->dev, "Cannot set rx format: %d\n", ret); - goto err_clk; - } - -err_clk: - clk_disable_unprepare(sai->clk); return ret; } @@ -235,11 +189,12 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, reg_mr = FSL_SAI_RMR; } - val_cr4 = sai_readl(sai, sai->base + reg_cr4); + regmap_read(sai->regmap, reg_cr4, &val_cr4); + regmap_read(sai->regmap, reg_cr4, &val_cr5); + val_cr4 &= ~FSL_SAI_CR4_SYWD_MASK; val_cr4 &= ~FSL_SAI_CR4_FRSZ_MASK; - val_cr5 = sai_readl(sai, sai->base + reg_cr5); val_cr5 &= ~FSL_SAI_CR5_WNW_MASK; val_cr5 &= ~FSL_SAI_CR5_W0W_MASK; val_cr5 &= ~FSL_SAI_CR5_FBT_MASK; @@ -257,9 +212,9 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, val_cr4 |= FSL_SAI_CR4_FRSZ(channels); val_mr = ~0UL - ((1 << channels) - 1); - sai_writel(sai, val_cr4, sai->base + reg_cr4); - sai_writel(sai, val_cr5, sai->base + reg_cr5); - sai_writel(sai, val_mr, sai->base + reg_mr); + regmap_write(sai->regmap, reg_cr4, val_cr4); + regmap_write(sai->regmap, reg_cr5, val_cr5); + regmap_write(sai->regmap, reg_mr, val_mr); return 0; } @@ -268,44 +223,34 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); - u32 tcsr, rcsr, val_cr2, val_cr3, reg_cr3; - - val_cr2 = sai_readl(sai, sai->base + FSL_SAI_TCR2); - val_cr2 &= ~FSL_SAI_CR2_SYNC; - sai_writel(sai, val_cr2, sai->base + FSL_SAI_TCR2); + u32 tcsr, rcsr; - val_cr2 = sai_readl(sai, sai->base + FSL_SAI_RCR2); - val_cr2 |= FSL_SAI_CR2_SYNC; - sai_writel(sai, val_cr2, sai->base + FSL_SAI_RCR2); + regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC, + ~FSL_SAI_CR2_SYNC); + regmap_update_bits(sai->regmap, FSL_SAI_RCR2, FSL_SAI_CR2_SYNC, + FSL_SAI_CR2_SYNC); - tcsr = sai_readl(sai, sai->base + FSL_SAI_TCSR); - rcsr = sai_readl(sai, sai->base + FSL_SAI_RCSR); + regmap_read(sai->regmap, FSL_SAI_TCSR, &tcsr); + regmap_read(sai->regmap, FSL_SAI_RCSR, &rcsr); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { tcsr |= FSL_SAI_CSR_FRDE; rcsr &= ~FSL_SAI_CSR_FRDE; - reg_cr3 = FSL_SAI_TCR3; } else { rcsr |= FSL_SAI_CSR_FRDE; tcsr &= ~FSL_SAI_CSR_FRDE; - reg_cr3 = FSL_SAI_RCR3; } - val_cr3 = sai_readl(sai, sai->base + reg_cr3); - switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: tcsr |= FSL_SAI_CSR_TERE; rcsr |= FSL_SAI_CSR_TERE; - val_cr3 |= FSL_SAI_CR3_TRCE; - sai_writel(sai, val_cr3, sai->base + reg_cr3); - sai_writel(sai, rcsr, sai->base + FSL_SAI_RCSR); - sai_writel(sai, tcsr, sai->base + FSL_SAI_TCSR); + regmap_write(sai->regmap, FSL_SAI_RCSR, rcsr); + regmap_write(sai->regmap, FSL_SAI_TCSR, tcsr); break; - case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: @@ -314,11 +259,8 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, rcsr &= ~FSL_SAI_CSR_TERE; } - val_cr3 &= ~FSL_SAI_CR3_TRCE; - - sai_writel(sai, tcsr, sai->base + FSL_SAI_TCSR); - sai_writel(sai, rcsr, sai->base + FSL_SAI_RCSR); - sai_writel(sai, val_cr3, sai->base + reg_cr3); + regmap_write(sai->regmap, FSL_SAI_TCSR, tcsr); + regmap_write(sai->regmap, FSL_SAI_RCSR, rcsr); break; default: return -EINVAL; @@ -331,16 +273,32 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + u32 reg; - return clk_prepare_enable(sai->clk); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + reg = FSL_SAI_TCR3; + else + reg = FSL_SAI_RCR3; + + regmap_update_bits(sai->regmap, reg, FSL_SAI_CR3_TRCE, + FSL_SAI_CR3_TRCE); + + return 0; } static void fsl_sai_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + u32 reg; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + reg = FSL_SAI_TCR3; + else + reg = FSL_SAI_RCR3; - clk_disable_unprepare(sai->clk); + regmap_update_bits(sai->regmap, reg, FSL_SAI_CR3_TRCE, + ~FSL_SAI_CR3_TRCE); } static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { @@ -355,18 +313,13 @@ static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev); - int ret; - - ret = clk_prepare_enable(sai->clk); - if (ret) - return ret; - sai_writel(sai, 0x0, sai->base + FSL_SAI_RCSR); - sai_writel(sai, 0x0, sai->base + FSL_SAI_TCSR); - sai_writel(sai, FSL_SAI_MAXBURST_TX * 2, sai->base + FSL_SAI_TCR1); - sai_writel(sai, FSL_SAI_MAXBURST_RX - 1, sai->base + FSL_SAI_RCR1); - - clk_disable_unprepare(sai->clk); + regmap_update_bits(sai->regmap, FSL_SAI_TCSR, 0xffffffff, 0x0); + regmap_update_bits(sai->regmap, FSL_SAI_RCSR, 0xffffffff, 0x0); + regmap_update_bits(sai->regmap, FSL_SAI_TCR1, FSL_SAI_CR1_RFW_MASK, + FSL_SAI_MAXBURST_TX * 2); + regmap_update_bits(sai->regmap, FSL_SAI_RCR1, FSL_SAI_CR1_RFW_MASK, + FSL_SAI_MAXBURST_RX - 1); snd_soc_dai_init_dma_data(cpu_dai, &sai->dma_params_tx, &sai->dma_params_rx); @@ -397,26 +350,109 @@ static const struct snd_soc_component_driver fsl_component = { .name = "fsl-sai", }; +static bool fsl_sai_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case FSL_SAI_TCSR: + case FSL_SAI_TCR1: + case FSL_SAI_TCR2: + case FSL_SAI_TCR3: + case FSL_SAI_TCR4: + case FSL_SAI_TCR5: + case FSL_SAI_TFR: + case FSL_SAI_TMR: + case FSL_SAI_RCSR: + case FSL_SAI_RCR1: + case FSL_SAI_RCR2: + case FSL_SAI_RCR3: + case FSL_SAI_RCR4: + case FSL_SAI_RCR5: + case FSL_SAI_RDR: + case FSL_SAI_RFR: + case FSL_SAI_RMR: + return true; + default: + return false; + } +} + +static bool fsl_sai_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case FSL_SAI_TFR: + case FSL_SAI_RFR: + case FSL_SAI_TDR: + case FSL_SAI_RDR: + return true; + default: + return false; + } + +} + +static bool fsl_sai_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case FSL_SAI_TCSR: + case FSL_SAI_TCR1: + case FSL_SAI_TCR2: + case FSL_SAI_TCR3: + case FSL_SAI_TCR4: + case FSL_SAI_TCR5: + case FSL_SAI_TDR: + case FSL_SAI_TMR: + case FSL_SAI_RCSR: + case FSL_SAI_RCR1: + case FSL_SAI_RCR2: + case FSL_SAI_RCR3: + case FSL_SAI_RCR4: + case FSL_SAI_RCR5: + case FSL_SAI_RMR: + return true; + default: + return false; + } +} + +static struct regmap_config fsl_sai_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + + .max_register = FSL_SAI_RMR, + .readable_reg = fsl_sai_readable_reg, + .volatile_reg = fsl_sai_volatile_reg, + .writeable_reg = fsl_sai_writeable_reg, +}; + static int fsl_sai_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; struct fsl_sai *sai; struct resource *res; + void __iomem *base; int ret; sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL); if (!sai) return -ENOMEM; + sai->big_endian_regs = of_property_read_bool(np, "big-endian-regs"); + if (sai->big_endian_regs) + fsl_sai_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG; + + sai->big_endian_data = of_property_read_bool(np, "big-endian-data"); + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - sai->base = devm_ioremap_resource(&pdev->dev, res); - if (IS_ERR(sai->base)) - return PTR_ERR(sai->base); - - sai->clk = devm_clk_get(&pdev->dev, "sai"); - if (IS_ERR(sai->clk)) { - dev_err(&pdev->dev, "Cannot get SAI's clock\n"); - return PTR_ERR(sai->clk); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) + return PTR_ERR(base); + + sai->regmap = devm_regmap_init_mmio_clk(&pdev->dev, + "sai", base, &fsl_sai_regmap_config); + if (IS_ERR(sai->regmap)) { + dev_err(&pdev->dev, "regmap init failed\n"); + return PTR_ERR(sai->regmap); } sai->dma_params_rx.addr = res->start + FSL_SAI_RDR; @@ -424,9 +460,6 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->dma_params_rx.maxburst = FSL_SAI_MAXBURST_RX; sai->dma_params_tx.maxburst = FSL_SAI_MAXBURST_TX; - sai->big_endian_regs = of_property_read_bool(np, "big-endian-regs"); - sai->big_endian_data = of_property_read_bool(np, "big-endian-data"); - platform_set_drvdata(pdev, sai); ret = devm_snd_soc_register_component(&pdev->dev, &fsl_component, diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 41bb62e..1571459 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -15,31 +15,36 @@ SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) +/* SAI Register Map Register */ +#define FSL_SAI_TCSR 0x00 /* SAI Transmit Control */ +#define FSL_SAI_TCR1 0x04 /* SAI Transmit Configuration 1 */ +#define FSL_SAI_TCR2 0x08 /* SAI Transmit Configuration 2 */ +#define FSL_SAI_TCR3 0x0c /* SAI Transmit Configuration 3 */ +#define FSL_SAI_TCR4 0x10 /* SAI Transmit Configuration 4 */ +#define FSL_SAI_TCR5 0x14 /* SAI Transmit Configuration 5 */ +#define FSL_SAI_TDR 0x20 /* SAI Transmit Data */ +#define FSL_SAI_TFR 0x40 /* SAI Transmit FIFO */ +#define FSL_SAI_TMR 0x60 /* SAI Transmit Mask */ +#define FSL_SAI_RCSR 0x80 /* SAI Receive Control */ +#define FSL_SAI_RCR1 0x84 /* SAI Receive Configuration 1 */ +#define FSL_SAI_RCR2 0x88 /* SAI Receive Configuration 2 */ +#define FSL_SAI_RCR3 0x8c /* SAI Receive Configuration 3 */ +#define FSL_SAI_RCR4 0x90 /* SAI Receive Configuration 4 */ +#define FSL_SAI_RCR5 0x94 /* SAI Receive Configuration 5 */ +#define FSL_SAI_RDR 0xa0 /* SAI Receive Data */ +#define FSL_SAI_RFR 0xc0 /* SAI Receive FIFO */ +#define FSL_SAI_RMR 0xe0 /* SAI Receive Mask */ + /* SAI Transmit/Recieve Control Register */ -#define FSL_SAI_TCSR 0x00 -#define FSL_SAI_RCSR 0x80 #define FSL_SAI_CSR_TERE BIT(31) #define FSL_SAI_CSR_FWF BIT(17) #define FSL_SAI_CSR_FRIE BIT(8) #define FSL_SAI_CSR_FRDE BIT(0) -/* SAI Transmit Data/FIFO/MASK Register */ -#define FSL_SAI_TDR 0x20 -#define FSL_SAI_TFR 0x40 -#define FSL_SAI_TMR 0x60 - -/* SAI Recieve Data/FIFO/MASK Register */ -#define FSL_SAI_RDR 0xa0 -#define FSL_SAI_RFR 0xc0 -#define FSL_SAI_RMR 0xe0 - /* SAI Transmit and Recieve Configuration 1 Register */ -#define FSL_SAI_TCR1 0x04 -#define FSL_SAI_RCR1 0x84 +#define FSL_SAI_CR1_RFW_MASK 0x1f /* SAI Transmit and Recieve Configuration 2 Register */ -#define FSL_SAI_TCR2 0x08 -#define FSL_SAI_RCR2 0x88 #define FSL_SAI_CR2_SYNC BIT(30) #define FSL_SAI_CR2_MSEL_MASK (0xff << 26) #define FSL_SAI_CR2_MSEL_BUS 0 @@ -50,15 +55,11 @@ #define FSL_SAI_CR2_BCD_MSTR BIT(24) /* SAI Transmit and Recieve Configuration 3 Register */ -#define FSL_SAI_TCR3 0x0c -#define FSL_SAI_RCR3 0x8c #define FSL_SAI_CR3_TRCE BIT(16) #define FSL_SAI_CR3_WDFL(x) (x) #define FSL_SAI_CR3_WDFL_MASK 0x1f /* SAI Transmit and Recieve Configuration 4 Register */ -#define FSL_SAI_TCR4 0x10 -#define FSL_SAI_RCR4 0x90 #define FSL_SAI_CR4_FRSZ(x) (((x) - 1) << 16) #define FSL_SAI_CR4_FRSZ_MASK (0x1f << 16) #define FSL_SAI_CR4_SYWD(x) (((x) - 1) << 8) @@ -69,8 +70,6 @@ #define FSL_SAI_CR4_FSD_MSTR BIT(0) /* SAI Transmit and Recieve Configuration 5 Register */ -#define FSL_SAI_TCR5 0x14 -#define FSL_SAI_RCR5 0x94 #define FSL_SAI_CR5_WNW(x) (((x) - 1) << 24) #define FSL_SAI_CR5_WNW_MASK (0x1f << 24) #define FSL_SAI_CR5_W0W(x) (((x) - 1) << 16) @@ -100,9 +99,7 @@ #define FSL_SAI_MAXBURST_RX 6 struct fsl_sai { - struct clk *clk; - - void __iomem *base; + struct regmap *regmap; bool big_endian_regs; bool big_endian_data; -- 1.8.4

3 2

[alsa-devel] Register wm8782 codec driver
by Ezequiel RG 10 Feb '14

10 Feb '14

Hi everyone, i'm trying to interface the wm8782 codec with my Beaglebone Black running 3.8.13 kernel. I'm getting this error when i load the device tree file: davinci_evm sound.11: ASoC: CODEC (null) not registered What i'm doing wrong?? The correponding source file are attached. Thanks for your help! wm8782.c /* * sound/soc/codecs/wm8782.c * simple, strap-pin configured 24bit 2ch ADC * * Copyright: 2011 Raumfeld GmbH * Author: Johannes Stezenbach <js(a)sig21.net> * * based on ad73311.c * Copyright:Analog Device Inc. * Author:Cliff Cai <cliff.cai(a)analog.com> * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. */ #include <linux/init.h> #include <linux/slab.h> #include <linux/module.h> #include <linux/kernel.h> #include <linux/device.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/ac97_codec.h> #include <sound/initval.h> #include <sound/soc.h> #include <linux/of_device.h> #include <linux/of_device.h> static const struct of_device_id wm8782_of_match[] = { { .compatible = "wlf,wm8782", }, { } }; MODULE_DEVICE_TABLE(of, wm8782_of_match); static const struct snd_soc_dapm_widget wm8782_dapm_widgets[] = { SND_SOC_DAPM_INPUT("AINL"), SND_SOC_DAPM_INPUT("AINR"), }; static const struct snd_soc_dapm_route wm8782_dapm_routes[] = { { "Capture", NULL, "AINL" }, { "Capture", NULL, "AINR" }, }; static struct snd_soc_dai_driver wm8782_dai = { .name = "wm8782-hifi", .capture = { .stream_name = "Capture", .channels_min = 2, .channels_max = 2, /* For configurations with FSAMPEN=0 */ .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, }, }; static struct snd_soc_codec_driver soc_codec_dev_wm8782 = { .dapm_widgets = wm8782_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(wm8782_dapm_widgets), .dapm_routes = wm8782_dapm_routes, .num_dapm_routes = ARRAY_SIZE(wm8782_dapm_routes), }; static int wm8782_probe(struct platform_device *pdev) { return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm8782, &wm8782_dai, 1); } static int wm8782_remove(struct platform_device *pdev) { snd_soc_unregister_codec(&pdev->dev); return 0; } static struct platform_driver wm8782_codec_driver = { .driver = { .name = "wm8782", .owner = THIS_MODULE, }, .probe = wm8782_probe, .remove = wm8782_remove, }; module_platform_driver(wm8782_codec_driver); MODULE_DESCRIPTION("ASoC WM8782 driver"); MODULE_AUTHOR("Johannes Stezenbach <js(a)sig21.net>"); MODULE_LICENSE("GPL"); davinci-evm.c /* * ASoC driver for TI DAVINCI EVM platform * * Author: Vladimir Barinov, <vbarinov(a)embeddedalley.com> * Copyright: (C) 2007 MontaVista Software, Inc., <source(a)mvista.com> * * 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. */ #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/timer.h> #include <linux/interrupt.h> #include <linux/platform_device.h> #include <linux/platform_data/edma.h> #include <linux/i2c.h> #include <linux/of_platform.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> #include <asm/dma.h> #include <asm/mach-types.h> #include <linux/edma.h> #include "davinci-pcm.h" #include "davinci-i2s.h" #include "davinci-mcasp.h" #include <linux/of_gpio.h> #define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \ SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF) static int evm_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 snd_soc_codec *codec = rtd->codec; struct snd_soc_card *soc_card = codec->card; struct device_node *np = soc_card->dev->of_node; int ret = 0; unsigned sysclk; if (np) { ret = of_property_read_u32(np, "ti,codec-clock-rate", &sysclk); if (ret < 0) return ret; } else { /* ASP1 on DM355 EVM is clocked by an external oscillator */ if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm() || machine_is_davinci_dm365_evm()) sysclk = 27000000; /* * ASP0 in DM6446 EVM is clocked by U55, as configured by * board-dm644x-evm.c using GPIOs from U18. There are six * options; here we "know" we use a 48 KHz sample rate. */ else if (machine_is_davinci_evm()) sysclk = 12288000; else if (machine_is_davinci_da830_evm() || machine_is_davinci_da850_evm()) sysclk = 24576000; else return -EINVAL; } /* set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, AUDIO_FORMAT); if (ret < 0) return ret; /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, AUDIO_FORMAT); if (ret < 0) return ret; /* set the codec system clock */ ret = snd_soc_dai_set_sysclk(codec_dai, 0, sysclk, SND_SOC_CLOCK_OUT); if (ret < 0) return ret; /* set the CPU system clock */ ret = snd_soc_dai_set_sysclk(cpu_dai, 0, sysclk, SND_SOC_CLOCK_OUT); if (ret < 0) return ret; return 0; } static int evm_spdif_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 *cpu_dai = rtd->cpu_dai; /* set cpu DAI configuration */ return snd_soc_dai_set_fmt(cpu_dai, AUDIO_FORMAT); } static struct snd_soc_ops evm_ops = { .hw_params = evm_hw_params, }; static struct snd_soc_ops evm_spdif_ops = { .hw_params = evm_spdif_hw_params, }; /* davinci-evm machine dapm widgets */ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_LINE("Line Out", NULL), SND_SOC_DAPM_MIC("Mic Jack", NULL), SND_SOC_DAPM_LINE("Line In", NULL), }; /* davinci-evm machine audio_mapnections to the codec pins */ static const struct snd_soc_dapm_route audio_map[] = { /* Headphone connected to HPLOUT, HPROUT */ {"Headphone Jack", NULL, "HPLOUT"}, {"Headphone Jack", NULL, "HPROUT"}, /* Line Out connected to LLOUT, RLOUT */ {"Line Out", NULL, "LLOUT"}, {"Line Out", NULL, "RLOUT"}, /* Mic connected to (MIC3L | MIC3R) */ {"MIC3L", NULL, "Mic Bias 2V"}, {"MIC3R", NULL, "Mic Bias 2V"}, {"Mic Bias 2V", NULL, "Mic Jack"}, /* Line In connected to (LINE1L | LINE2L), (LINE1R | LINE2R) */ {"LINE1L", NULL, "Line In"}, {"LINE2L", NULL, "Line In"}, {"LINE1R", NULL, "Line In"}, {"LINE2R", NULL, "Line In"}, }; /* Logic for a tda998x as connected on a davinci-evm */ static int evm_tda998x_init(struct snd_soc_pcm_runtime *rtd) { 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_clkdiv(cpu_dai, 0, 1); if (ret < 0) return ret; ret = snd_soc_dai_set_clkdiv(cpu_dai, 1, 8); if (ret < 0) return ret; ret = snd_soc_dai_set_sysclk(cpu_dai, 0, 0, SND_SOC_CLOCK_IN); if (ret < 0) return ret; ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_IB_NF); if (ret < 0) return ret; return 0; } /* Logic for a aic3x as connected on a davinci-evm */ static int evm_aic3x_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dapm_context *dapm = &codec->dapm; struct device_node *np = codec->card->dev->of_node; int ret; /* Add davinci-evm specific widgets */ snd_soc_dapm_new_controls(dapm, aic3x_dapm_widgets, ARRAY_SIZE(aic3x_dapm_widgets)); if (np) { ret = snd_soc_of_parse_audio_routing(codec->card, "ti,audio-routing"); if (ret) return ret; } else { /* Set up davinci-evm specific audio path audio_map */ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); } /* Divide McASP MCLK by 2 to provide 12MHz to codec */ ret = snd_soc_dai_set_clkdiv(cpu_dai, 0, 2); if (ret < 0) return ret; /* not connected */ snd_soc_dapm_disable_pin(dapm, "MONO_LOUT"); snd_soc_dapm_disable_pin(dapm, "HPLCOM"); snd_soc_dapm_disable_pin(dapm, "HPRCOM"); /* always connected */ snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); snd_soc_dapm_enable_pin(dapm, "Line Out"); snd_soc_dapm_enable_pin(dapm, "Mic Jack"); snd_soc_dapm_enable_pin(dapm, "Line In"); return 0; } /* davinci-evm digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link dm6446_evm_dai = { .name = "TLV320AIC3X", .stream_name = "AIC3X", .cpu_dai_name = "davinci-mcbsp", .codec_dai_name = "tlv320aic3x-hifi", .codec_name = "tlv320aic3x-codec.1-001b", .platform_name = "davinci-mcbsp", .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-mcbsp.1", .init = evm_aic3x_init, .ops = &evm_ops, }; static struct snd_soc_dai_link dm365_evm_dai = { #ifdef CONFIG_SND_DM365_AIC3X_CODEC .name = "TLV320AIC3X", .stream_name = "AIC3X", .cpu_dai_name = "davinci-mcbsp", .codec_dai_name = "tlv320aic3x-hifi", .init = evm_aic3x_init, .codec_name = "tlv320aic3x-codec.1-0018", .ops = &evm_ops, .platform_name = "davinci-mcbsp", #elif defined(CONFIG_SND_DM365_VOICE_CODEC) .name = "Voice Codec - CQ93VC", .stream_name = "CQ93", .cpu_dai_name = "davinci-vcif", .codec_dai_name = "cq93vc-hifi", .codec_name = "cq93vc-codec", .platform_name = "davinci-vcif", #endif }; static struct snd_soc_dai_link dm6467_evm_dai[] = { { .name = "TLV320AIC3X", .stream_name = "AIC3X", .cpu_dai_name= "davinci-mcasp.0", .codec_dai_name = "tlv320aic3x-hifi", .platform_name = "davinci-mcasp.0", .codec_name = "tlv320aic3x-codec.0-001a", .init = evm_aic3x_init, .ops = &evm_ops, }, { .name = "McASP", .stream_name = "spdif", .cpu_dai_name= "davinci-mcasp.1", .codec_dai_name = "dit-hifi", .codec_name = "spdif_dit", .platform_name = "davinci-mcasp.1", .ops = &evm_spdif_ops, }, }; static struct snd_soc_dai_link da830_evm_dai = { .name = "TLV320AIC3X", .stream_name = "AIC3X", .cpu_dai_name = "davinci-mcasp.1", .codec_dai_name = "tlv320aic3x-hifi", .codec_name = "tlv320aic3x-codec.1-0018", .platform_name = "davinci-mcasp.1", .init = evm_aic3x_init, .ops = &evm_ops, }; static struct snd_soc_dai_link da850_evm_dai = { .name = "TLV320AIC3X", .stream_name = "AIC3X", .cpu_dai_name= "davinci-mcasp.0", .codec_dai_name = "tlv320aic3x-hifi", .codec_name = "tlv320aic3x-codec.1-0018", .platform_name = "davinci-mcasp.0", .init = evm_aic3x_init, .ops = &evm_ops, }; /* davinci dm6446 evm audio machine driver */ static struct snd_soc_card dm6446_snd_soc_card_evm = { .name = "DaVinci DM6446 EVM", .owner = THIS_MODULE, .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", .owner = THIS_MODULE, .dai_link = &dm355_evm_dai, .num_links = 1, }; /* davinci dm365 evm audio machine driver */ static struct snd_soc_card dm365_snd_soc_card_evm = { .name = "DaVinci DM365 EVM", .owner = THIS_MODULE, .dai_link = &dm365_evm_dai, .num_links = 1, }; /* davinci dm6467 evm audio machine driver */ static struct snd_soc_card dm6467_snd_soc_card_evm = { .name = "DaVinci DM6467 EVM", .owner = THIS_MODULE, .dai_link = dm6467_evm_dai, .num_links = ARRAY_SIZE(dm6467_evm_dai), }; static struct snd_soc_card da830_snd_soc_card = { .name = "DA830/OMAP-L137 EVM", .owner = THIS_MODULE, .dai_link = &da830_evm_dai, .num_links = 1, }; static struct snd_soc_card da850_snd_soc_card = { .name = "DA850/OMAP-L138 EVM", .owner = THIS_MODULE, .dai_link = &da850_evm_dai, .num_links = 1, }; #if defined(CONFIG_OF) enum { MACHINE_VERSION_1 = 0,/* DM365 with Voice Codec */ MACHINE_VERSION_2,/* DM365/DA8xx/OMAPL1x/AM33xx */ MACHINE_VERSION_3,/* AM33xx BeagleBone Black */ }; static const struct of_device_id davinci_evm_dt_ids[] = { { .compatible = "ti,dm365-voice-codec-audio", .data = (void *)MACHINE_VERSION_1, }, { .compatible = "ti,da830-evm-audio", .data = (void *)MACHINE_VERSION_2, }, { .compatible = "ti,am33xx-beaglebone-black", .data = (void *)MACHINE_VERSION_3, }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, davinci_mcasp_dt_ids); /* * This struct is just used as place holder. It will be filled with * data from dt node */ static struct snd_soc_dai_link evm_dai = { .name= "WM8782", .stream_name= "AIC3X", .codec_dai_name= "wm8782-hifi", }; /* davinci evm audio machine driver */ static struct snd_soc_card evm_soc_card = { .owner = THIS_MODULE, .dai_link = &evm_dai, .num_links = 1, }; static int davinci_evm_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; const struct of_device_id *match = of_match_device(of_match_ptr(davinci_evm_dt_ids), &pdev->dev); u32 machine_ver, clk_gpio; int ret = 0; machine_ver = (u32)match->data; switch (machine_ver) { case MACHINE_VERSION_1: evm_dai.name= "Voice Codec - CQ93VC"; evm_dai.stream_name= "CQ93"; evm_dai.codec_dai_name= "cq93vc-hifi"; break; case MACHINE_VERSION_2: evm_dai.ops = &evm_ops; evm_dai.init = evm_aic3x_init; break; case MACHINE_VERSION_3: evm_dai.name= "NXP TDA HDMI Chip"; evm_dai.stream_name= "HDMI"; evm_dai.codec_dai_name= "nxp-hdmi-hifi"; evm_dai.init = evm_tda998x_init; /* * Move GPIO handling out of the probe, if probe gets * deferred, the gpio will have been claimed on previous * probe and will fail on the second and susequent probes */ clk_gpio = of_get_named_gpio(np, "mcasp_clock_enable", 0); if (clk_gpio < 0) { dev_err(&pdev->dev, "failed to find mcasp_clock enable GPIO!\n"); return -EINVAL; } ret = gpio_request_one(clk_gpio, GPIOF_OUT_INIT_HIGH, "McASP Clock Enable Pin"); if (ret < 0) { dev_err(&pdev->dev, "Failed to claim McASP Clock Enable pin\n"); return -EINVAL; } gpio_set_value(clk_gpio, 1); break; } evm_dai.codec_of_node = of_parse_phandle(np, "ti,audio-codec", 0); if (!evm_dai.codec_of_node) return -EINVAL; evm_dai.cpu_of_node = of_parse_phandle(np, "ti,mcasp-controller", 0); if (!evm_dai.cpu_of_node) return -EINVAL; evm_dai.platform_of_node = evm_dai.cpu_of_node; evm_soc_card.dev = &pdev->dev; ret = snd_soc_of_parse_card_name(&evm_soc_card, "ti,model"); if (ret) return ret; ret = snd_soc_register_card(&evm_soc_card); if (ret) dev_err(&pdev->dev, "snd_soc_register_card fuck failed (%d)\n", ret); return ret; } static int davinci_evm_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); snd_soc_unregister_card(card); return 0; } static struct platform_driver davinci_evm_driver = { .probe= davinci_evm_probe, .remove= davinci_evm_remove, .driver= { .name= "davinci_evm", .owner= THIS_MODULE, .of_match_table = of_match_ptr(davinci_evm_dt_ids), }, }; #endif static struct platform_device *evm_snd_device; static int __init evm_init(void) { struct snd_soc_card *evm_snd_dev_data; int index; int ret; #if defined(CONFIG_OF) /* * If dtb is there, the devices will be created dynamically. * Only register platfrom driver structure. */ if (of_have_populated_dt()) return platform_driver_register(&davinci_evm_driver); #endif if (machine_is_davinci_evm()) { evm_snd_dev_data = &dm6446_snd_soc_card_evm; index = 0; } else if (machine_is_davinci_dm355_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; index = 0; } else if (machine_is_davinci_dm6467_evm()) { evm_snd_dev_data = &dm6467_snd_soc_card_evm; index = 0; } else if (machine_is_davinci_da830_evm()) { evm_snd_dev_data = &da830_snd_soc_card; index = 1; } else if (machine_is_davinci_da850_evm()) { evm_snd_dev_data = &da850_snd_soc_card; index = 0; } else return -EINVAL; evm_snd_device = platform_device_alloc("soc-audio", index); if (!evm_snd_device) return -ENOMEM; platform_set_drvdata(evm_snd_device, evm_snd_dev_data); ret = platform_device_add(evm_snd_device); if (ret) platform_device_put(evm_snd_device); return ret; } static void __exit evm_exit(void) { #if defined(CONFIG_OF) if (of_have_populated_dt()) { platform_driver_unregister(&davinci_evm_driver); return; } #endif platform_device_unregister(evm_snd_device); } module_init(evm_init); module_exit(evm_exit); MODULE_AUTHOR("Vladimir Barinov"); MODULE_DESCRIPTION("TI DAVINCI EVM ASoC driver"); MODULE_LICENSE("GPL"); device tree file /* * Copyright (C) 2014 John Rhoades * * 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. */ /dts-v1/; /plugin/; / { compatible = "ti,beaglebone-black"; part-number = "jsr-cs4271"; version = "00A0"; /* state the resources this cape uses */ exclusive-use = /* the pin header uses */ "P9.17", /* cs4271 reset */ /* i2c2 */ "P9.19", /* i2c2: scl */ "P9.20", /* i2c2: sca */ /* mcasp0 */ "P9.31", /* mcasp0: aclkx 0x190 */ "P9.29", /* mcasp0: fsx 0x194 */ /* not used "P9.30", /* mcasp0: axr0 0x198 */ "P9.28", /* mcasp0: axr2 0x19c */ "P9.42", /* mcasp0: aclkr 0x1a0 */ "P9.27", /* mcasp0: fsr 0x1a4 */ /* not used "P9.41", /* mcasp0: axr1 0x1a8 */ "P9.25", /* mcasp0: axr3 0x1ac */ /* the hardware IP uses */ "gpio1_27", "snd-soc-wm8782", // needed to get module loaded "mcasp0"; fragment@0 { target = <&am33xx_pinmux>; __overlay__ { wm8782_i2c2_reset: wm8782_i2c2_reset { pinctrl-single,pins = < 0x078 0x0f /* gpio0_5, mode7 */ >; }; wm8782_i2c2_pins: wm8782_i2c2_pins { pinctrl-single,pins = < 0x178 0x73 /* I2C2_sda */ 0x17c 0x73 /* I2C2_scl */ >; }; mcasp0_pins: mcasp0_pins { pinctrl-single,pins = < 0x190 0x20 /* P9.31 mcasp0 0->aclkx */ 0x194 0x20 /* P9.29 mcasp0 0->fsx */ 0x198 0x20 /* P9.30 mcasp0 0->axr0 */ 0x19c 0x22 /* P9.28 mcasp0 0->ahclkr, 2->axr2 */ 0x1a0 0x20 /* P9.42 mcasp0 0->aclkr, 2->axr2 */ 0x1a4 0x20 /* P9.27 mcasp0 0->fsr, 2->axr3 */ 0x1a8 0x20 /* P9.41 mcasp0 0->axr1 */ 0x1ac 0x22 /* P9.25 mcasp0 0->ahclkx, 2->axr3 */ >; }; }; }; fragment@2 { target = <&i2c2>; __overlay__ { #address-cells = <1>; #size-cells = <0>; wm8782: wm8782@10 { compatible = ",wm8782"; reg = <0x10>; status = "okay"; }; }; }; fragment@3 { target = <&mcasp0>; __overlay__ { pinctrl-names = "default"; pinctrl-0 = <&mcasp0_pins>; status = "okay"; op-mode = <0>; /* MCASP_I2S_MODE */ tdm-slots = <2>; num-serializer = <16>; serial-dir = < /* 0: INACTIVE, 1: TX, 2: RX */ 0 0 2 1 /* axr2: capture data, axr3: play data */ 0 0 0 0 0 0 0 0 0 0 0 0 >; tx-num-evt = <1>; rx-num-evt = <1>; }; }; fragment@4 { target = <&ocp>; __overlay__ { sound { compatible = "ti,da830-evm-audio"; //"ti,am33xx-beaglebone-black"; ti,model = "DA830 EVM"; // "wm8782"; ti,audio-codec = <&wm8782>; ti,mcasp-controller = <&mcasp0>; ti,codec-clock-rate = <24576000>; #mcasp_clock_enable = <&gpio2 27 0>; /* BeagleBone Black Clk enable on GPIO1_27 */ ti,audio-routing = "Headphone Jack", "HPLOUT", "Headphone Jack", "HPROUT", "LINE1L", "Line In", "LINE1R", "Line In"; }; }; }; };

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Re: [alsa-devel] Fw: [Bug 1155202] [Intel DZ77SL-50K, Intel PantherPoint HDMI, Digital Out, HDMI] No sound at all
by niraj kulkarni 10 Feb '14

10 Feb '14

Removing these 3 usleep ranges is working now. Although I have a doubt for applying this patch on all chips. There have been a number of boards released after this particular board, but nobody else is facing this problem. Since spec does not specify a max on deasserting RST, some boards might actually take more than 100uS to reset. Will this patch work for those boards? @@ -1149,7 +1149,7 @@ timeout = jiffies + msecs_to_jiffies(100); while ((azx_readb(chip, GCTL) & ICH6_GCTL_RESET) && time_before(jiffies, timeout)) - usleep_range(500, 1000); + cpu_relax(); } /* exit link reset */ @@ -1162,7 +1162,7 @@ timeout = jiffies + msecs_to_jiffies(100); while (!azx_readb(chip, GCTL) && time_before(jiffies, timeout)) - usleep_range(500, 1000); + cpu_relax(); } /* reset codec link */ @@ -1180,7 +1180,7 @@ /* delay for >= 100us for codec PLL to settle per spec * Rev 0.9 section 5.5.1 */ - usleep_range(500, 1000); + //usleep_range(500, 1000); /* Bring controller out of reset */ azx_exit_link_reset(chip);

2 3

[alsa-devel] [PATCH] ALSA: hda - Disable static quirks for C-Media codecs
by Takashi Iwai 10 Feb '14

10 Feb '14

According to alsa-info.sh outputs, all three entries with static quirks have the correct pin configs, so it's safe to remove static quirks. For now, turn the static quirks off via ifdef. The dead codes will be removed in later release. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_cmedia.c | 31 +++++++++++++++++++++---------- 1 file changed, 21 insertions(+), 10 deletions(-) diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index 9c6ce73b03c5..139ef301c914 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -32,6 +32,9 @@ #include "hda_jack.h" #include "hda_generic.h" +#undef ENABLE_CMI_STATIC_QUIRKS + +#ifdef ENABLE_CMI_STATIC_QUIRKS #define NUM_PINS 11 @@ -45,10 +48,12 @@ enum { CMI_AUTO, /* let driver guess it */ CMI_MODELS }; +#endif /* ENABLE_CMI_STATIC_QUIRKS */ struct cmi_spec { struct hda_gen_spec gen; +#ifdef ENABLE_CMI_STATIC_QUIRKS /* below are only for static models */ int board_config; @@ -81,8 +86,10 @@ struct cmi_spec { /* multichannel pins */ struct hda_verb multi_init[9]; /* 2 verbs for each pin + terminator */ +#endif /* ENABLE_CMI_STATIC_QUIRKS */ }; +#ifdef ENABLE_CMI_STATIC_QUIRKS /* * input MUX */ @@ -566,6 +573,7 @@ static const struct hda_codec_ops cmi9880_patch_ops = { .init = cmi9880_init, .free = cmi9880_free, }; +#endif /* ENABLE_CMI_STATIC_QUIRKS */ /* * stuff for auto-parser @@ -588,15 +596,20 @@ static int cmi_parse_auto_config(struct hda_codec *codec) err = snd_hda_parse_pin_defcfg(codec, cfg, NULL, 0); if (err < 0) - return err; + goto error; err = snd_hda_gen_parse_auto_config(codec, cfg); if (err < 0) - return err; + goto error; codec->patch_ops = cmi_auto_patch_ops; return 0; + + error: + snd_hda_gen_free(codec); + return err; } + static int patch_cmi9880(struct hda_codec *codec) { struct cmi_spec *spec; @@ -606,6 +619,7 @@ static int patch_cmi9880(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; +#ifdef ENABLE_CMI_STATIC_QUIRKS spec->board_config = snd_hda_check_board_config(codec, CMI_MODELS, cmi9880_models, cmi9880_cfg_tbl); @@ -615,14 +629,8 @@ static int patch_cmi9880(struct hda_codec *codec) spec->board_config = CMI_AUTO; /* try everything */ } - if (spec->board_config == CMI_AUTO) { - int err = cmi_parse_auto_config(codec); - if (err < 0) { - snd_hda_gen_free(codec); - return err; - } - return 0; - } + if (spec->board_config == CMI_AUTO) + return cmi_parse_auto_config(codec); /* copy default DAC NIDs */ memcpy(spec->dac_nids, cmi9880_dac_nids, sizeof(spec->dac_nids)); @@ -669,6 +677,9 @@ static int patch_cmi9880(struct hda_codec *codec) codec->patch_ops = cmi9880_patch_ops; return 0; +#else + return cmi_parse_auto_config(codec); +#endif } /* -- 1.8.5.2

1 0

[alsa-devel] [PATCH v3 0/2] *** SUBJECT HERE ***
by Jean-Francois Moine 10 Feb '14

10 Feb '14

*** BLURB HERE *** Jean-Francois Moine (2): drivers/base: permit base components to omit the bind/unbind ops drivers/base: declare phandle DT nodes as components drivers/base/component.c | 21 +++++++++++++++++++-- drivers/base/core.c | 18 ++++++++++++++++++ include/linux/of.h | 2 ++ 3 files changed, 39 insertions(+), 2 deletions(-) -- 1.9.rc1

4 7

[alsa-devel] [PATCH v2 0/8] da9055: Driver initialisation fixes, add DT support
by Adam Thomson 10 Feb '14

10 Feb '14

This patch series provides the following updates for DA9055 drivers: - Fixes an issue with da9055 driver initialisation (conflicting device ids) of PMIC (MFD) and CODEC drivers which means they cannot be used together on the same I2C bus. - Add initial DT support for DA9055 related drivers, including binding documentation. - Remove use of regmap_irq_get_virq() in driver probes which was conflicting with use of platform_get_irq_byname(). platform_get_irq_byname() already returns the VIRQ number due to MFD core translation so using regmap_irq_get_virq() on that returned value results in an incorrect IRQ being requested. The driver probes then fail because of this. [Changes since v1]: - Regulator: * fixed coding style issue (lack of braces for if,else statement) * removal of 'ifdef' around DT related headers * Use of 'of_regulator_match()' for retrieving regulator related info from DT nodes. - General: * Instead of removing platform_get_irq_byname() usage and device resources to fix IRQ conflict in driver probes, now removed 'regmap_irq_get_virq()' as advised. This has been tested on target for both DT and non-DT setups and works as expected. Adam Thomson (8): ASoC: da9055: Fix device registration of PMIC and CODEC devices ASoC: da9055: Add DT support for CODEC mfd: da9055: Add DT support for PMIC regulator: da9055: Add DT support onkey: da9055: Remove use of regmap_irq_get_virq() hwmon: da9055: Remove use of regmap_irq_get_virq() rtc: da9055: Remove use of regmap_irq_get_virq() regulator: da9055: Remove use of regmap_irq_get_virq() Documentation/devicetree/bindings/mfd/da9055.txt | 73 ++++++++++++++++++++ Documentation/devicetree/bindings/sound/da9055.txt | 22 ++++++ drivers/hwmon/da9055-hwmon.c | 4 - drivers/input/misc/da9055_onkey.c | 1 - drivers/mfd/da9055-i2c.c | 20 +++++- drivers/regulator/da9055-regulator.c | 73 ++++++++++++++++++-- drivers/rtc/rtc-da9055.c | 4 +- sound/soc/codecs/da9055.c | 19 +++++- 8 files changed, 201 insertions(+), 15 deletions(-) create mode 100644 Documentation/devicetree/bindings/mfd/da9055.txt create mode 100644 Documentation/devicetree/bindings/sound/da9055.txt

5 24

[alsa-devel] [PATCH 0/3] Add i.MX6q Kosagi Novena support
by Sean Cross 10 Feb '14

10 Feb '14

The Kosagi Novena board contains an i.MX6 Quad processor. This patchset adds support for this platform, along with support for its own audio codec. The platform uses an ES8328-based codec, which is currently unsupported in the Linux kernel. The audio codec supports playback and recording, as well as jack detection support for the headset. Audio codec support is required as an uninitialized audio codec pulls I2C2 lines down, preventing the bus from working. There are two known issues with this patch, and I would like some feedback on them: * I'm unclear on how audio routing works. The ES8328 has five outputs: LOUT1, ROUT1, LOUT2, ROUT2, and OUT3. It also has four inputs. The board maps Headphone L/R to L/ROUT2, and Speaker to L/ROUT1. What is the proper way to describe this audio routing? * The snd-soc-imx-novena.ko module oopses when removing it. The backtrace shows that it's caused by "Unable to handle kernel paging request at virtual address", due to snd_jack_dev_free possibly getting called twice. What could be causing this? I never explicitly free the jack device: (snd_ctl_remove) from [<804c73d0>] (snd_jack_dev_free+0x48/0x68) (snd_jack_dev_free) from [<804c6ef8>] (snd_device_free+0xd0/0x148) (snd_device_free) from [<804c72b0>] (snd_device_free_all+0x90/0xb0) (snd_device_free_all) from [<804c20ac>] (snd_card_do_free+0x40/0xec) (snd_card_do_free) from [<804c22b4>] (snd_card_free+0x74/0x80) (snd_card_free) from [<804de444>] (soc_cleanup_card_resources+0xb4/0xbc) (soc_cleanup_card_resources) from [<804de460>] (snd_soc_unregister_card+0x14/0x1c) (snd_soc_unregister_card) from [<7f032014>] (imx_novena_remove+0x14/0x1c [snd_soc_imx_novena]) (imx_novena_remove [snd_soc_imx_novena]) from [<8035c258>] (__device_release_driver+0x64/0xb0) (__device_release_driver) from [<8035c81c>] (driver_detach+0xa4/0xcc) (driver_detach) from [<8035be88>] (bus_remove_driver+0x64/0x9c) (bus_remove_driver) from [<8007ea44>] (SyS_delete_module+0x110/0x188) (SyS_delete_module) from [<8000e080>] (ret_fast_syscall+0x0/0x30) Sean Cross (3): sound: soc: codecs: Add es8328 codec sound: soc: fsl: Add support for Novena onboard audio dts: imx: add kosagi novena imx6q dts file arch/arm/boot/dts/imx6q-novena.dts | 517 +++++++++++++++++++++++++++++++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/es8328.c | 611 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/es8328.h | 240 +++++++++++++++ sound/soc/fsl/Kconfig | 13 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-novena.c | 344 +++++++++++++++++++++ 8 files changed, 1733 insertions(+) create mode 100644 arch/arm/boot/dts/imx6q-novena.dts create mode 100644 sound/soc/codecs/es8328.c create mode 100644 sound/soc/codecs/es8328.h create mode 100644 sound/soc/fsl/imx-novena.c -- 1.8.3.2

3 13

[alsa-devel] [PATCH] ASoC: fsl-esai: fix ESAI TDM slot setting
by Xiubo Li 10 Feb '14

10 Feb '14

Cc: Nicolin Chen <Guangyu.Chen(a)freescale.com> Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/fsl/fsl_esai.c | 4 ++-- sound/soc/fsl/fsl_esai.h | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index d0c72ed..c84026c 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -326,7 +326,7 @@ static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMA, ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(tx_mask)); regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMB, - ESAI_xSMA_xS_MASK, ESAI_xSMB_xS(tx_mask)); + ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(tx_mask)); regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(slots)); @@ -334,7 +334,7 @@ static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMA, ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(rx_mask)); regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMB, - ESAI_xSMA_xS_MASK, ESAI_xSMB_xS(rx_mask)); + ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(rx_mask)); esai_priv->slot_width = slot_width; diff --git a/sound/soc/fsl/fsl_esai.h b/sound/soc/fsl/fsl_esai.h index 9c9f957..75e1403 100644 --- a/sound/soc/fsl/fsl_esai.h +++ b/sound/soc/fsl/fsl_esai.h @@ -322,7 +322,7 @@ #define ESAI_xSMB_xS_SHIFT 0 #define ESAI_xSMB_xS_WIDTH 16 #define ESAI_xSMB_xS_MASK (((1 << ESAI_xSMB_xS_WIDTH) - 1) << ESAI_xSMB_xS_SHIFT) -#define ESAI_xSMB_xS(v) (((v) >> ESAI_xSMA_xS_WIDTH) & ESAI_xSMA_xS_MASK) +#define ESAI_xSMB_xS(v) (((v) >> ESAI_xSMA_xS_WIDTH) & ESAI_xSMB_xS_MASK) /* Port C Direction Register -- REG_ESAI_PRRC 0xF8 */ #define ESAI_PRRC_PDC_SHIFT 0 -- 1.8.4

4 5

[alsa-devel] [PATCH v3 0/4] off-codec widgets
by Xiubo Li 10 Feb '14

10 Feb '14

This patch series has removed the code about tdm slot, and need to do more study. This adds "ASoC: binding: add widgets.txt" new patch. This also revises some others from Mark's comment. Xiubo Li (4): ASoC: add snd_soc_of_parse_audio_simple_widgets for DT ASoC: simple-card: add off-codec widgets supports. ASoC: binding: add widgets.txt ASoC: binding: for new properties documenting and usage .../devicetree/bindings/sound/simple-card.txt | 12 ++- .../devicetree/bindings/sound/widgets.txt | 20 +++++ include/sound/soc.h | 2 + sound/soc/generic/simple-card.c | 8 ++ sound/soc/soc-core.c | 87 ++++++++++++++++++++++ 5 files changed, 127 insertions(+), 2 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/widgets.txt -- 1.8.4

2 5

[alsa-devel] [PATCH] ASoC: fsl: fix pm support of machine drivers
by Shawn Guo 10 Feb '14

10 Feb '14

The commit 1abe729 (ASoC: fsl: Add missing pm to current machine drivers) enables pm support for a few IMX machine drivers. But it does not update dev drvdata to be the pointer to 'card'. This causes the kernel dump below in system suspend, because snd_soc_suspend() expects that the dev drvdata points to 'card', while it still points to the private data of machine driver. This patch fixes imx-sgtl5000 and imx-wm8962 by attaching 'card' to dev drvdata and private data to card drvdata. For imx-mc13783, I simply revert the pm change because it must be broken for the same reason and I don't have hardware to test pm enabling code. $ echo mem > /sys/power/state PM: Syncing filesystems ... done. PM: Preparing system for mem sleep mmc1: card e624 removed Freezing user space processes ... (elapsed 0.002 seconds) done. Freezing remaining freezable tasks ... (elapsed 0.002 seconds) done. PM: Entering mem sleep INFO: trying to register non-static key. the code is fine but needs lockdep annotation. turning off the locking correctness validator. CPU: 0 PID: 1861 Comm: bash Not tainted 3.14.0-rc1+ #1648 Backtrace: [<80012144>] (dump_backtrace) from [<800122e4>] (show_stack+0x18/0x1c) r6:8079c77c r5:00000c5a r4:00000000 r3:00000000 [<800122cc>] (show_stack) from [<80637ac0>] (dump_stack+0x78/0x94) [<80637a48>] (dump_stack) from [<80028918>] (warn_slowpath_common+0x6c/0x8c) r4:bdb21c38 r3:be62df00 [<800288ac>] (warn_slowpath_common) from [<800289dc>] (warn_slowpath_fmt+0x38/0x40) r8:be62e3a8 r7:bf122960 r6:00000005 r5:00000000 r4:00000000 [<800289a8>] (warn_slowpath_fmt) from [<8006518c>] (__lock_acquire+0x1ae0/0x1ce0) r3:8079d598 r2:80799e70 [<800636ac>] (__lock_acquire) from [<80065894>] (lock_acquire+0x68/0x7c) r10:bdb20000 r9:be62df00 r8:00000000 r7:00000000 r6:60000013 r5:bdb20000 r4:00000000 [<8006582c>] (lock_acquire) from [<8063c938>] (mutex_lock_nested+0x5c/0x3b8) r7:00000000 r6:80dfc78c r5:804be444 r4:bf122928 [<8063c8dc>] (mutex_lock_nested) from [<804be444>] (snd_soc_suspend+0x34/0x42c) r10:00000000 r9:00000000 r8:00000000 r7:bf1c4444 r6:bf1c4410 r5:be978150 r4:be978010 [<804be410>] (snd_soc_suspend) from [<8034392c>] (platform_pm_suspend+0x34/0x64) r10:00000000 r8:00000000 r7:bf1c4444 r6:bf1c4410 r5:803438f8 r4:bf1c4410 [<803438f8>] (platform_pm_suspend) from [<80348e18>] (dpm_run_callback.isra.7+0x34/0x6c) [<80348de4>] (dpm_run_callback.isra.7) from [<80349354>] (__device_suspend+0x10c/0x220) r9:808dd974 r8:808c4a5c r6:00000002 r5:80e5001c r4:bf1c4410 [<80349248>] (__device_suspend) from [<8034a338>] (dpm_suspend+0x60/0x220) r7:bf1c4410 r6:808dd90c r5:80e5001c r4:bf1c44c0 [<8034a2d8>] (dpm_suspend) from [<8034a790>] (dpm_suspend_start+0x60/0x68) r10:8079a818 r9:00000000 r8:00000004 r7:80dfbe90 r6:80641eec r5:00000000 r4:00000002 [<8034a730>] (dpm_suspend_start) from [<8006a788>] (suspend_devices_and_enter+0x74/0x318) r4:00000003 r3:80dfbe98 [<8006a714>] (suspend_devices_and_enter) from [<8006abd8>] (pm_suspend+0x1ac/0x244) r10:8079a818 r8:00000004 r7:00000003 r6:80641eec r5:00000000 r4:00000003 [<8006aa2c>] (pm_suspend) from [<80069a4c>] (state_store+0x70/0xc0) r5:00000003 r4:bd85ea40 [<800699dc>] (state_store) from [<80294034>] (kobj_attr_store+0x1c/0x28) r10:beb9fe08 r8:00000000 r7:bdb21f78 r6:bd85ea40 r5:00000004 r4:beb9fe00 [<80294018>] (kobj_attr_store) from [<80140f90>] (sysfs_kf_write+0x54/0x58) [<80140f3c>] (sysfs_kf_write) from [<8014474c>] (kernfs_fop_write+0xc4/0x160) r6:bd85ea40 r5:beb9fe00 r4:00000004 r3:80140f3c [<80144688>] (kernfs_fop_write) from [<800dfa14>] (vfs_write+0xbc/0x184) r10:00000000 r9:00000000 r8:00000000 r7:bdb21f78 r6:00500c08 r5:00000004 r4:be782600 [<800df958>] (vfs_write) from [<800dfe00>] (SyS_write+0x48/0x70) r10:00000000 r8:00000000 r7:00000004 r6:00500c08 r5:00000000 r4:be782600 [<800dfdb8>] (SyS_write) from [<8000e800>] (ret_fast_syscall+0x0/0x48) r9:bdb20000 r8:8000e9c4 r7:00000004 r6:00500c08 r5:00000004 r4:76eb65e0 Fixes: 1abe729 (ASoC: fsl: Add missing pm to current machine drivers) Cc: stable(a)vger.kernel.org Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- sound/soc/fsl/imx-mc13783.c | 1 - sound/soc/fsl/imx-sgtl5000.c | 10 ++++++---- sound/soc/fsl/imx-wm8962.c | 11 +++++++---- 3 files changed, 13 insertions(+), 9 deletions(-) diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c index 79cee78..a2fd732 100644 --- a/sound/soc/fsl/imx-mc13783.c +++ b/sound/soc/fsl/imx-mc13783.c @@ -160,7 +160,6 @@ static struct platform_driver imx_mc13783_audio_driver = { .driver = { .name = "imx_mc13783", .owner = THIS_MODULE, - .pm = &snd_soc_pm_ops, }, .probe = imx_mc13783_probe, .remove = imx_mc13783_remove diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index f2beae7..1cb22dd 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -33,8 +33,7 @@ struct imx_sgtl5000_data { static int imx_sgtl5000_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct imx_sgtl5000_data *data = container_of(rtd->card, - struct imx_sgtl5000_data, card); + struct imx_sgtl5000_data *data = snd_soc_card_get_drvdata(rtd->card); struct device *dev = rtd->card->dev; int ret; @@ -159,13 +158,15 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) data->card.dapm_widgets = imx_sgtl5000_dapm_widgets; data->card.num_dapm_widgets = ARRAY_SIZE(imx_sgtl5000_dapm_widgets); + platform_set_drvdata(pdev, &data->card); + snd_soc_card_set_drvdata(&data->card, data); + ret = devm_snd_soc_register_card(&pdev->dev, &data->card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); goto fail; } - platform_set_drvdata(pdev, data); of_node_put(ssi_np); of_node_put(codec_np); @@ -184,7 +185,8 @@ fail: static int imx_sgtl5000_remove(struct platform_device *pdev) { - struct imx_sgtl5000_data *data = platform_get_drvdata(pdev); + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct imx_sgtl5000_data *data = snd_soc_card_get_drvdata(card); clk_put(data->codec_clk); diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 3fd76bc..3a3d17c 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -71,7 +71,7 @@ static int imx_wm8962_set_bias_level(struct snd_soc_card *card, { struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; struct imx_priv *priv = &card_priv; - struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); struct device *dev = &priv->pdev->dev; unsigned int pll_out; int ret; @@ -137,7 +137,7 @@ static int imx_wm8962_late_probe(struct snd_soc_card *card) { struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; struct imx_priv *priv = &card_priv; - struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); struct device *dev = &priv->pdev->dev; int ret; @@ -264,13 +264,15 @@ static int imx_wm8962_probe(struct platform_device *pdev) data->card.late_probe = imx_wm8962_late_probe; data->card.set_bias_level = imx_wm8962_set_bias_level; + platform_set_drvdata(pdev, &data->card); + snd_soc_card_set_drvdata(&data->card, data); + ret = devm_snd_soc_register_card(&pdev->dev, &data->card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); goto clk_fail; } - platform_set_drvdata(pdev, data); of_node_put(ssi_np); of_node_put(codec_np); @@ -289,7 +291,8 @@ fail: static int imx_wm8962_remove(struct platform_device *pdev) { - struct imx_wm8962_data *data = platform_get_drvdata(pdev); + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); if (!IS_ERR(data->codec_clk)) clk_disable_unprepare(data->codec_clk); -- 1.7.9.5

3 2

[alsa-devel] [PATCH 0/8] New and Resend the old patches
by Xiubo Li 10 Feb '14

10 Feb '14

New and Resend the old patches basing the newest code version. And this patch series has been tested based the VF610 Tower board.

5 24

[alsa-devel] [PATCH] ALSA: hda - Fix undefined symbol due to builtin/module mixup
by Takashi Iwai 10 Feb '14

10 Feb '14

Even after the fix for leftover kconfig handling (commit f8f1becf), the current code still doesn't handle properly the builtin/module mixup case between the core snd-hda-codec and other codec drivers. For example, when CONFIG_SND_HDA_INTEL=y and CONFIG_SND_HDA_CODEC_HDMI=m, it'll end up with an unresolved symbol snd_hda_parse_hdmi_codec. This patch fixes the issue. Now codec->parser points to the parser object *only* when a module (either generic or HDMI parser) is loaded and bound. When a builtin symbol is used, codec->parser still points to NULL. This is the difference from the previous versions. Fixes: f8f1becfa4ac ('ALSA: hda - Fix leftover ifdef checks after modularization') Reported-by: Fengguang Wu <fengguang.wu(a)intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 30 +++++++++++++++--------------- 1 file changed, 15 insertions(+), 15 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 69acbf758bbb..dafcf82139e2 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1339,23 +1339,15 @@ get_hda_cvt_setup(struct hda_codec *codec, hda_nid_t nid) /* * Dynamic symbol binding for the codec parsers */ -#ifdef MODULE -#define load_parser_sym(sym) ((int (*)(struct hda_codec *))symbol_request(sym)) -#define unload_parser_addr(addr) symbol_put_addr(addr) -#else -#define load_parser_sym(sym) (sym) -#define unload_parser_addr(addr) do {} while (0) -#endif #define load_parser(codec, sym) \ - ((codec)->parser = load_parser_sym(sym)) + ((codec)->parser = (int (*)(struct hda_codec *))symbol_request(sym)) static void unload_parser(struct hda_codec *codec) { - if (codec->parser) { - unload_parser_addr(codec->parser); - codec->parser = NULL; - } + if (codec->parser) + symbol_put_addr(codec->parser); + codec->parser = NULL; } /* @@ -1620,12 +1612,20 @@ int snd_hda_codec_configure(struct hda_codec *codec) patch = codec->preset->patch; if (!patch) { unload_parser(codec); /* to be sure */ - if (is_likely_hdmi_codec(codec)) + if (is_likely_hdmi_codec(codec)) { +#if IS_MODULE(CONFIG_SND_HDA_CODEC_HDMI) patch = load_parser(codec, snd_hda_parse_hdmi_codec); -#if IS_ENABLED(CONFIG_SND_HDA_GENERIC) - if (!patch) +#elif IS_BUILTIN(CONFIG_SND_HDA_CODEC_HDMI) + patch = snd_hda_parse_hdmi_codec; +#endif + } + if (!patch) { +#if IS_MODULE(CONFIG_SND_HDA_GENERIC) patch = load_parser(codec, snd_hda_parse_generic_codec); +#elif IS_BUILTIN(CONFIG_SND_HDA_GENERIC) + patch = snd_hda_parse_generic_codec; #endif + } if (!patch) { printk(KERN_ERR "hda-codec: No codec parser is available\n"); return -ENODEV; -- 1.8.5.2

1 0

[alsa-devel] [PATCH v2] ALSA: Replace with IS_ENABLED()
by Takashi Iwai 10 Feb '14

10 Feb '14

Replace the lengthy #if defined(XXX) || defined(XXX_MODULE) with the new IS_ENABLED() macro. The patch still doesn't cover all ifdefs. For example, the dependency on CONFIG_GAMEPORT is still open-coded because this also has an extra dependency on MODULE. Similarly, an open-coded ifdef in pcm_oss.c and some sequencer-related stuff are left untouched. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: Remove conversions wrt sequencer that unsatisfy config dependency properly sound/core/control.c | 2 +- sound/core/info.c | 4 ++-- sound/core/init.c | 8 ++++---- sound/core/pcm.c | 10 +++++----- sound/core/pcm_native.c | 2 +- sound/core/rtctimer.c | 4 ++-- sound/core/sound_oss.c | 2 +- sound/core/timer.c | 4 ++-- sound/drivers/opl3/opl3_synth.c | 2 +- sound/pci/hda/hda_intel.c | 2 +- 10 files changed, 20 insertions(+), 20 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index d8aa206e8bde..c228f00e5701 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -151,7 +151,7 @@ void snd_ctl_notify(struct snd_card *card, unsigned int mask, if (snd_BUG_ON(!card || !id)) return; read_lock(&card->ctl_files_rwlock); -#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_MIXER_OSS) card->mixer_oss_change_count++; #endif list_for_each_entry(ctl, &card->ctl_files, list) { diff --git a/sound/core/info.c b/sound/core/info.c index e79baa11b60e..7916c07b9324 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -540,7 +540,7 @@ int __init snd_info_init(void) snd_oss_root = entry; } #endif -#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE) +#if IS_ENABLED(CONFIG_SND_SEQUENCER) { struct snd_info_entry *entry; if ((entry = snd_info_create_module_entry(THIS_MODULE, "seq", NULL)) == NULL) @@ -567,7 +567,7 @@ int __exit snd_info_done(void) snd_minor_info_done(); snd_info_version_done(); if (snd_proc_root) { -#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE) +#if IS_ENABLED(CONFIG_SND_SEQUENCER) snd_info_free_entry(snd_seq_root); #endif #ifdef CONFIG_SND_OSSEMUL diff --git a/sound/core/init.c b/sound/core/init.c index 0d42fcda0de2..a16d765cdf47 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -94,7 +94,7 @@ static int module_slot_match(struct module *module, int idx) return match; } -#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_MIXER_OSS) int (*snd_mixer_oss_notify_callback)(struct snd_card *card, int free_flag); EXPORT_SYMBOL(snd_mixer_oss_notify_callback); #endif @@ -394,7 +394,7 @@ int snd_card_disconnect(struct snd_card *card) /* phase 3: notify all connected devices about disconnection */ /* at this point, they cannot respond to any calls except release() */ -#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_MIXER_OSS) if (snd_mixer_oss_notify_callback) snd_mixer_oss_notify_callback(card, SND_MIXER_OSS_NOTIFY_DISCONNECT); #endif @@ -430,7 +430,7 @@ EXPORT_SYMBOL(snd_card_disconnect); */ static int snd_card_do_free(struct snd_card *card) { -#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_MIXER_OSS) if (snd_mixer_oss_notify_callback) snd_mixer_oss_notify_callback(card, SND_MIXER_OSS_NOTIFY_FREE); #endif @@ -723,7 +723,7 @@ int snd_card_register(struct snd_card *card) snd_cards[card->number] = card; mutex_unlock(&snd_card_mutex); init_info_for_card(card); -#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_MIXER_OSS) if (snd_mixer_oss_notify_callback) snd_mixer_oss_notify_callback(card, SND_MIXER_OSS_NOTIFY_REGISTER); #endif diff --git a/sound/core/pcm.c b/sound/core/pcm.c index e1e9e0c999fe..091a05c1d5b3 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -295,7 +295,7 @@ static const char *snd_pcm_state_name(snd_pcm_state_t state) return snd_pcm_state_names[(__force int)state]; } -#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_PCM_OSS) #include <linux/soundcard.h> static const char *snd_pcm_oss_format_name(int format) @@ -398,7 +398,7 @@ static void snd_pcm_substream_proc_hw_params_read(struct snd_info_entry *entry, snd_iprintf(buffer, "rate: %u (%u/%u)\n", runtime->rate, runtime->rate_num, runtime->rate_den); snd_iprintf(buffer, "period_size: %lu\n", runtime->period_size); snd_iprintf(buffer, "buffer_size: %lu\n", runtime->buffer_size); -#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_PCM_OSS) if (substream->oss.oss) { snd_iprintf(buffer, "OSS format: %s\n", snd_pcm_oss_format_name(runtime->oss.format)); snd_iprintf(buffer, "OSS channels: %u\n", runtime->oss.channels); @@ -651,7 +651,7 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) struct snd_pcm_str *pstr = &pcm->streams[stream]; struct snd_pcm_substream *substream, *prev; -#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_PCM_OSS) mutex_init(&pstr->oss.setup_mutex); #endif pstr->stream = stream; @@ -807,7 +807,7 @@ EXPORT_SYMBOL(snd_pcm_new_internal); static void snd_pcm_free_stream(struct snd_pcm_str * pstr) { struct snd_pcm_substream *substream, *substream_next; -#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_PCM_OSS) struct snd_pcm_oss_setup *setup, *setupn; #endif substream = pstr->substream; @@ -819,7 +819,7 @@ static void snd_pcm_free_stream(struct snd_pcm_str * pstr) substream = substream_next; } snd_pcm_stream_proc_done(pstr); -#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_PCM_OSS) for (setup = pstr->oss.setup_list; setup; setup = setupn) { setupn = setup->next; kfree(setup->task_name); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 10b5d1a9dec0..e3664116736b 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -399,7 +399,7 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, return -EBADFD; } snd_pcm_stream_unlock_irq(substream); -#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_PCM_OSS) if (!substream->oss.oss) #endif if (atomic_read(&substream->mmap_count)) diff --git a/sound/core/rtctimer.c b/sound/core/rtctimer.c index e85e72baff9e..b27223746b0e 100644 --- a/sound/core/rtctimer.c +++ b/sound/core/rtctimer.c @@ -27,7 +27,7 @@ #include <sound/core.h> #include <sound/timer.h> -#if defined(CONFIG_RTC) || defined(CONFIG_RTC_MODULE) +#if IS_ENABLED(CONFIG_RTC) #include <linux/mc146818rtc.h> @@ -185,4 +185,4 @@ MODULE_LICENSE("GPL"); MODULE_ALIAS("snd-timer-" __stringify(SNDRV_TIMER_GLOBAL_RTC)); -#endif /* CONFIG_RTC || CONFIG_RTC_MODULE */ +#endif /* IS_ENABLED(CONFIG_RTC) */ diff --git a/sound/core/sound_oss.c b/sound/core/sound_oss.c index 726a49ac9725..5391c5e139ef 100644 --- a/sound/core/sound_oss.c +++ b/sound/core/sound_oss.c @@ -21,7 +21,7 @@ #ifdef CONFIG_SND_OSSEMUL -#if !defined(CONFIG_SOUND) && !(defined(MODULE) && defined(CONFIG_SOUND_MODULE)) +#if !IS_ENABLED(CONFIG_SOUND) #error "Enable the OSS soundcore multiplexer (CONFIG_SOUND) in the kernel." #endif diff --git a/sound/core/timer.c b/sound/core/timer.c index 6ddcf06f52f9..cbec51464c5b 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -35,9 +35,9 @@ #include <sound/initval.h> #include <linux/kmod.h> -#if defined(CONFIG_SND_HRTIMER) || defined(CONFIG_SND_HRTIMER_MODULE) +#if IS_ENABLED(CONFIG_SND_HRTIMER) #define DEFAULT_TIMER_LIMIT 4 -#elif defined(CONFIG_SND_RTCTIMER) || defined(CONFIG_SND_RTCTIMER_MODULE) +#elif IS_ENABLED(CONFIG_SND_RTCTIMER) #define DEFAULT_TIMER_LIMIT 2 #else #define DEFAULT_TIMER_LIMIT 1 diff --git a/sound/drivers/opl3/opl3_synth.c b/sound/drivers/opl3/opl3_synth.c index 742a4b642fd9..ddcc1a325a61 100644 --- a/sound/drivers/opl3/opl3_synth.c +++ b/sound/drivers/opl3/opl3_synth.c @@ -24,7 +24,7 @@ #include <sound/opl3.h> #include <sound/asound_fm.h> -#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE) +#if IS_ENABLED(CONFIG_SND_SEQUENCER) #define OPL3_SUPPORT_SYNTH #endif diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e87eac552ec5..17c83fbc9f84 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -4137,7 +4137,7 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { .driver_data = AZX_DRIVER_CTHDA | AZX_DCAPS_PRESET_CTHDA }, { PCI_DEVICE(0x1102, 0x0012), .driver_data = AZX_DRIVER_CTHDA | AZX_DCAPS_PRESET_CTHDA }, -#if !defined(CONFIG_SND_CTXFI) && !defined(CONFIG_SND_CTXFI_MODULE) +#if !IS_ENABLED(CONFIG_SND_CTXFI) /* the following entry conflicts with snd-ctxfi driver, * as ctxfi driver mutates from HD-audio to native mode with * a special command sequence. -- 1.8.5.2

1 0

[alsa-devel] [PATCH] Refresh patch, .reset_resume added to struct usb_driver
by eliot＠blennerhassett.gen.nz 10 Feb '14

10 Feb '14

From: Eliot Blennerhassett <eliot(a)blennerhassett.gen.nz> Signed-off-by: Eliot Blennerhassett <eliot(a)blennerhassett.gen.nz> --- alsa/usb/card.patch | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/alsa/usb/card.patch b/alsa/usb/card.patch index 679bf15..a37c93d 100644 --- a/alsa/usb/card.patch +++ b/alsa/usb/card.patch @@ -1,5 +1,5 @@ ---- ../alsa-kernel/usb/card.c 2013-10-09 14:00:50.128683771 +0200 -+++ card.c 2013-10-15 15:45:31.779430219 +0200 +--- ../alsa-kernel/usb/card.c 2014-02-09 14:35:45.872517538 +1300 ++++ card.c 2014-02-09 16:45:45.116673978 +1300 @@ -1,3 +1,4 @@ +#include "card.inc" /* @@ -67,15 +67,15 @@ /* * new 2.5 USB kernel API */ -@@ -727,6 +742,7 @@ - #define usb_audio_suspend NULL +@@ -738,6 +753,7 @@ #define usb_audio_resume NULL + #define usb_audio_reset_resume NULL #endif /* CONFIG_PM */ +#endif /* OLD_USB */ static struct usb_device_id usb_audio_ids [] = { #include "quirks-table.h" -@@ -742,13 +758,25 @@ +@@ -753,14 +769,25 @@ */ static struct usb_driver usb_audio_driver = { @@ -88,6 +88,7 @@ +#ifndef OLD_USB .suspend = usb_audio_suspend, .resume = usb_audio_resume, +- .reset_resume = usb_audio_reset_resume, +#endif +#ifdef OLD_USB + .driver_list = LIST_HEAD_INIT(usb_audio_driver.driver_list), -- 1.7.10.4

3 2

[alsa-devel] [PATCH] [trivial] alsa: Fix typos in alsa-driver-api.xml
by Masanari Iida 10 Feb '14

10 Feb '14

This patch fixed 2 typos in DocBook/alsa-driver-api.xml. It is because this file is generated by make xmldocs, I have to fix typos within source files. Signed-off-by: Masanari Iida <standby24x7(a)gmail.com> --- sound/core/pcm_native.c | 2 +- sound/core/rawmidi.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 01a5e05..10b5d1a 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -954,7 +954,7 @@ static struct action_ops snd_pcm_action_stop = { * * The state of each stream is then changed to the given state unconditionally. * - * Return: Zero if succesful, or a negative error code. + * Return: Zero if successful, or a negative error code. */ int snd_pcm_stop(struct snd_pcm_substream *substream, snd_pcm_state_t state) { diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 7b596b5..f016be7 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -1101,7 +1101,7 @@ int snd_rawmidi_transmit_peek(struct snd_rawmidi_substream *substream, /** * snd_rawmidi_transmit_ack - acknowledge the transmission * @substream: the rawmidi substream - * @count: the tranferred count + * @count: the transferred count * * Advances the hardware pointer for the internal output buffer with * the given size and updates the condition. -- 1.9.rc1

2 1

[alsa-devel] [PATCH] ALSA: Replace with IS_ENABLED()
by Takashi Iwai 10 Feb '14

10 Feb '14

Replace the lengthy #if defined(XXX) || defined(XXX_MODULE) with the new IS_ENABLED() macro. The sequencer-related ifdefs in some places (e.g. rawmidi.c) are a bit tricky. Instead of adding the extra dependency on CONFIG_MODULE, use the another Kconfig such as CONFIG_SND_RAQMIDI_SEQ that already takes such a dependency into account. The patch still doesn't cover all ifdefs. For example, the dependency on CONFIG_GAMEPORT is still open-coded because this also has an extra dependency on CONFIG_MODULE. Similarly, an open-coded ifdef is still left in pcm_oss.c for OSS mixer ioctl. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/control.c | 2 +- sound/core/info.c | 4 ++-- sound/core/init.c | 8 ++++---- sound/core/pcm.c | 10 +++++----- sound/core/pcm_native.c | 2 +- sound/core/rawmidi.c | 4 ++-- sound/core/rtctimer.c | 4 ++-- sound/core/sound_oss.c | 2 +- sound/core/timer.c | 4 ++-- sound/drivers/opl3/opl3_lib.c | 2 +- sound/drivers/opl3/opl3_synth.c | 2 +- sound/drivers/opl4/opl4_lib.c | 4 ++-- sound/drivers/opl4/opl4_local.h | 2 +- sound/isa/sb/emu8000.c | 2 +- sound/isa/sb/sb16.c | 2 +- sound/pci/emu10k1/emu10k1.c | 2 +- sound/pci/hda/hda_intel.c | 2 +- 17 files changed, 29 insertions(+), 29 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index d8aa206e8bde..c228f00e5701 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -151,7 +151,7 @@ void snd_ctl_notify(struct snd_card *card, unsigned int mask, if (snd_BUG_ON(!card || !id)) return; read_lock(&card->ctl_files_rwlock); -#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_MIXER_OSS) card->mixer_oss_change_count++; #endif list_for_each_entry(ctl, &card->ctl_files, list) { diff --git a/sound/core/info.c b/sound/core/info.c index e79baa11b60e..7916c07b9324 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -540,7 +540,7 @@ int __init snd_info_init(void) snd_oss_root = entry; } #endif -#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE) +#if IS_ENABLED(CONFIG_SND_SEQUENCER) { struct snd_info_entry *entry; if ((entry = snd_info_create_module_entry(THIS_MODULE, "seq", NULL)) == NULL) @@ -567,7 +567,7 @@ int __exit snd_info_done(void) snd_minor_info_done(); snd_info_version_done(); if (snd_proc_root) { -#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE) +#if IS_ENABLED(CONFIG_SND_SEQUENCER) snd_info_free_entry(snd_seq_root); #endif #ifdef CONFIG_SND_OSSEMUL diff --git a/sound/core/init.c b/sound/core/init.c index 0d42fcda0de2..a16d765cdf47 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -94,7 +94,7 @@ static int module_slot_match(struct module *module, int idx) return match; } -#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_MIXER_OSS) int (*snd_mixer_oss_notify_callback)(struct snd_card *card, int free_flag); EXPORT_SYMBOL(snd_mixer_oss_notify_callback); #endif @@ -394,7 +394,7 @@ int snd_card_disconnect(struct snd_card *card) /* phase 3: notify all connected devices about disconnection */ /* at this point, they cannot respond to any calls except release() */ -#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_MIXER_OSS) if (snd_mixer_oss_notify_callback) snd_mixer_oss_notify_callback(card, SND_MIXER_OSS_NOTIFY_DISCONNECT); #endif @@ -430,7 +430,7 @@ EXPORT_SYMBOL(snd_card_disconnect); */ static int snd_card_do_free(struct snd_card *card) { -#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_MIXER_OSS) if (snd_mixer_oss_notify_callback) snd_mixer_oss_notify_callback(card, SND_MIXER_OSS_NOTIFY_FREE); #endif @@ -723,7 +723,7 @@ int snd_card_register(struct snd_card *card) snd_cards[card->number] = card; mutex_unlock(&snd_card_mutex); init_info_for_card(card); -#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_MIXER_OSS) if (snd_mixer_oss_notify_callback) snd_mixer_oss_notify_callback(card, SND_MIXER_OSS_NOTIFY_REGISTER); #endif diff --git a/sound/core/pcm.c b/sound/core/pcm.c index e1e9e0c999fe..091a05c1d5b3 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -295,7 +295,7 @@ static const char *snd_pcm_state_name(snd_pcm_state_t state) return snd_pcm_state_names[(__force int)state]; } -#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_PCM_OSS) #include <linux/soundcard.h> static const char *snd_pcm_oss_format_name(int format) @@ -398,7 +398,7 @@ static void snd_pcm_substream_proc_hw_params_read(struct snd_info_entry *entry, snd_iprintf(buffer, "rate: %u (%u/%u)\n", runtime->rate, runtime->rate_num, runtime->rate_den); snd_iprintf(buffer, "period_size: %lu\n", runtime->period_size); snd_iprintf(buffer, "buffer_size: %lu\n", runtime->buffer_size); -#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_PCM_OSS) if (substream->oss.oss) { snd_iprintf(buffer, "OSS format: %s\n", snd_pcm_oss_format_name(runtime->oss.format)); snd_iprintf(buffer, "OSS channels: %u\n", runtime->oss.channels); @@ -651,7 +651,7 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) struct snd_pcm_str *pstr = &pcm->streams[stream]; struct snd_pcm_substream *substream, *prev; -#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_PCM_OSS) mutex_init(&pstr->oss.setup_mutex); #endif pstr->stream = stream; @@ -807,7 +807,7 @@ EXPORT_SYMBOL(snd_pcm_new_internal); static void snd_pcm_free_stream(struct snd_pcm_str * pstr) { struct snd_pcm_substream *substream, *substream_next; -#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_PCM_OSS) struct snd_pcm_oss_setup *setup, *setupn; #endif substream = pstr->substream; @@ -819,7 +819,7 @@ static void snd_pcm_free_stream(struct snd_pcm_str * pstr) substream = substream_next; } snd_pcm_stream_proc_done(pstr); -#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_PCM_OSS) for (setup = pstr->oss.setup_list; setup; setup = setupn) { setupn = setup->next; kfree(setup->task_name); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 10b5d1a9dec0..e3664116736b 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -399,7 +399,7 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, return -EBADFD; } snd_pcm_stream_unlock_irq(substream); -#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_PCM_OSS) if (!substream->oss.oss) #endif if (atomic_read(&substream->mmap_count)) diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index f016be72b100..f15dbb8001f1 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -1530,7 +1530,7 @@ static int snd_rawmidi_dev_free(struct snd_device *device) return snd_rawmidi_free(rmidi); } -#if defined(CONFIG_SND_SEQUENCER) || (defined(MODULE) && defined(CONFIG_SND_SEQUENCER_MODULE)) +#if IS_ENABLED(CONFIG_SND_RAWMIDI_SEQ) static void snd_rawmidi_dev_seq_free(struct snd_seq_device *device) { struct snd_rawmidi *rmidi = device->private_data; @@ -1605,7 +1605,7 @@ static int snd_rawmidi_dev_register(struct snd_device *device) } } rmidi->proc_entry = entry; -#if defined(CONFIG_SND_SEQUENCER) || (defined(MODULE) && defined(CONFIG_SND_SEQUENCER_MODULE)) +#if IS_ENABLED(CONFIG_SND_RAWMIDI_SEQ) if (!rmidi->ops || !rmidi->ops->dev_register) { /* own registration mechanism */ if (snd_seq_device_new(rmidi->card, rmidi->device, SNDRV_SEQ_DEV_ID_MIDISYNTH, 0, &rmidi->seq_dev) >= 0) { rmidi->seq_dev->private_data = rmidi; diff --git a/sound/core/rtctimer.c b/sound/core/rtctimer.c index e85e72baff9e..b27223746b0e 100644 --- a/sound/core/rtctimer.c +++ b/sound/core/rtctimer.c @@ -27,7 +27,7 @@ #include <sound/core.h> #include <sound/timer.h> -#if defined(CONFIG_RTC) || defined(CONFIG_RTC_MODULE) +#if IS_ENABLED(CONFIG_RTC) #include <linux/mc146818rtc.h> @@ -185,4 +185,4 @@ MODULE_LICENSE("GPL"); MODULE_ALIAS("snd-timer-" __stringify(SNDRV_TIMER_GLOBAL_RTC)); -#endif /* CONFIG_RTC || CONFIG_RTC_MODULE */ +#endif /* IS_ENABLED(CONFIG_RTC) */ diff --git a/sound/core/sound_oss.c b/sound/core/sound_oss.c index 726a49ac9725..5391c5e139ef 100644 --- a/sound/core/sound_oss.c +++ b/sound/core/sound_oss.c @@ -21,7 +21,7 @@ #ifdef CONFIG_SND_OSSEMUL -#if !defined(CONFIG_SOUND) && !(defined(MODULE) && defined(CONFIG_SOUND_MODULE)) +#if !IS_ENABLED(CONFIG_SOUND) #error "Enable the OSS soundcore multiplexer (CONFIG_SOUND) in the kernel." #endif diff --git a/sound/core/timer.c b/sound/core/timer.c index 6ddcf06f52f9..cbec51464c5b 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -35,9 +35,9 @@ #include <sound/initval.h> #include <linux/kmod.h> -#if defined(CONFIG_SND_HRTIMER) || defined(CONFIG_SND_HRTIMER_MODULE) +#if IS_ENABLED(CONFIG_SND_HRTIMER) #define DEFAULT_TIMER_LIMIT 4 -#elif defined(CONFIG_SND_RTCTIMER) || defined(CONFIG_SND_RTCTIMER_MODULE) +#elif IS_ENABLED(CONFIG_SND_RTCTIMER) #define DEFAULT_TIMER_LIMIT 2 #else #define DEFAULT_TIMER_LIMIT 1 diff --git a/sound/drivers/opl3/opl3_lib.c b/sound/drivers/opl3/opl3_lib.c index 33d9a857a262..d7091e56d0d0 100644 --- a/sound/drivers/opl3/opl3_lib.c +++ b/sound/drivers/opl3/opl3_lib.c @@ -530,7 +530,7 @@ int snd_opl3_hwdep_new(struct snd_opl3 * opl3, opl3->hwdep = hw; opl3->seq_dev_num = seq_device; -#if defined(CONFIG_SND_SEQUENCER) || (defined(MODULE) && defined(CONFIG_SND_SEQUENCER_MODULE)) +#if IS_ENABLED(CONFIG_SND_OPL3_LIB_SEQ) if (snd_seq_device_new(card, seq_device, SNDRV_SEQ_DEV_ID_OPL3, sizeof(struct snd_opl3 *), &opl3->seq_dev) >= 0) { strcpy(opl3->seq_dev->name, hw->name); diff --git a/sound/drivers/opl3/opl3_synth.c b/sound/drivers/opl3/opl3_synth.c index 742a4b642fd9..ddcc1a325a61 100644 --- a/sound/drivers/opl3/opl3_synth.c +++ b/sound/drivers/opl3/opl3_synth.c @@ -24,7 +24,7 @@ #include <sound/opl3.h> #include <sound/asound_fm.h> -#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE) +#if IS_ENABLED(CONFIG_SND_SEQUENCER) #define OPL3_SUPPORT_SYNTH #endif diff --git a/sound/drivers/opl4/opl4_lib.c b/sound/drivers/opl4/opl4_lib.c index b953fb4aa298..c88679291091 100644 --- a/sound/drivers/opl4/opl4_lib.c +++ b/sound/drivers/opl4/opl4_lib.c @@ -153,7 +153,7 @@ static int snd_opl4_detect(struct snd_opl4 *opl4) return 0; } -#if defined(CONFIG_SND_SEQUENCER) || (defined(MODULE) && defined(CONFIG_SND_SEQUENCER_MODULE)) +#if IS_ENABLED(CONFIG_SND_OPL4_LIB_SEQ) static void snd_opl4_seq_dev_free(struct snd_seq_device *seq_dev) { struct snd_opl4 *opl4 = seq_dev->private_data; @@ -253,7 +253,7 @@ int snd_opl4_create(struct snd_card *card, snd_opl4_create_proc(opl4); #endif -#if defined(CONFIG_SND_SEQUENCER) || (defined(MODULE) && defined(CONFIG_SND_SEQUENCER_MODULE)) +#if IS_ENABLED(SND_OPL4_LIB_SEQ) opl4->seq_client = -1; if (opl4->hardware < OPL3_HW_OPL4_ML) snd_opl4_create_seq_dev(opl4, seq_device); diff --git a/sound/drivers/opl4/opl4_local.h b/sound/drivers/opl4/opl4_local.h index 470e5a758a02..be10b3981365 100644 --- a/sound/drivers/opl4/opl4_local.h +++ b/sound/drivers/opl4/opl4_local.h @@ -184,7 +184,7 @@ struct snd_opl4 { #endif struct mutex access_mutex; -#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE) +#if IS_ENABLED(SND_OPL4_LIB_SEQ) int used; int seq_dev_num; diff --git a/sound/isa/sb/emu8000.c b/sound/isa/sb/emu8000.c index 45fcdff611f9..83de44cd74a3 100644 --- a/sound/isa/sb/emu8000.c +++ b/sound/isa/sb/emu8000.c @@ -1139,7 +1139,7 @@ snd_emu8000_new(struct snd_card *card, int index, long port, int seq_ports, snd_emu8000_free(hw); return err; } -#if defined(CONFIG_SND_SEQUENCER) || (defined(MODULE) && defined(CONFIG_SND_SEQUENCER_MODULE)) +#if IS_ENABLED(CONFIG_SND_SBAWE_SEQ) if (snd_seq_device_new(card, index, SNDRV_SEQ_DEV_ID_EMU8000, sizeof(struct snd_emu8000*), &awe) >= 0) { strcpy(awe->name, "EMU-8000"); diff --git a/sound/isa/sb/sb16.c b/sound/isa/sb/sb16.c index a4130993955f..2bfbb3aaf51b 100644 --- a/sound/isa/sb/sb16.c +++ b/sound/isa/sb/sb16.c @@ -62,7 +62,7 @@ MODULE_SUPPORTED_DEVICE("{{Creative Labs,SB AWE 32}," #define SNDRV_DEBUG_IRQ #endif -#if defined(SNDRV_SBAWE) && (defined(CONFIG_SND_SEQUENCER) || (defined(MODULE) && defined(CONFIG_SND_SEQUENCER_MODULE))) +#if defined(SNDRV_SBAWE) && IS_ENABLED(CONFIG_SND_SBAWE_SEQ) #define SNDRV_SBAWE_EMU8000 #endif diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c index 9e1bd0c39a8c..887540ff5c62 100644 --- a/sound/pci/emu10k1/emu10k1.c +++ b/sound/pci/emu10k1/emu10k1.c @@ -37,7 +37,7 @@ MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Creative Labs,SB Live!/PCI512/E-mu APS}," "{Creative Labs,SB Audigy}}"); -#if defined(CONFIG_SND_SEQUENCER) || (defined(MODULE) && defined(CONFIG_SND_SEQUENCER_MODULE)) +#if IS_ENABLED(CONFIG_SND_EMU10K1_SEQ) #define ENABLE_SYNTH #include <sound/emu10k1_synth.h> #endif diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e87eac552ec5..17c83fbc9f84 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -4137,7 +4137,7 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { .driver_data = AZX_DRIVER_CTHDA | AZX_DCAPS_PRESET_CTHDA }, { PCI_DEVICE(0x1102, 0x0012), .driver_data = AZX_DRIVER_CTHDA | AZX_DCAPS_PRESET_CTHDA }, -#if !defined(CONFIG_SND_CTXFI) && !defined(CONFIG_SND_CTXFI_MODULE) +#if !IS_ENABLED(CONFIG_SND_CTXFI) /* the following entry conflicts with snd-ctxfi driver, * as ctxfi driver mutates from HD-audio to native mode with * a special command sequence. -- 1.8.5.2

1 0

[alsa-devel] [RFC v2][PATCH 00/38] Enhancement for support of some firewire devices
by Takashi Sakamoto 10 Feb '14

10 Feb '14

This series of patch is to update patches which I've posted below: [RFC][PATCH 00/17] Enhancement for firewire-lib http://mailman.alsa-project.org/pipermail/alsa-devel/2013-November/069163.h… [RFC][PATCH 00/13] a new driver for BridgeCo BeBoB based device http://mailman.alsa-project.org/pipermail/alsa-devel/2013-November/069183.h… [PATCH v2 0/8][RFC] a driver for Fireworks based devices http://mailman.alsa-project.org/pipermail/alsa-devel/2013-December/070043.h… In this time, I'll post all patches because now they have dependency each other. Each driver has lines in asound.h/firewire.h. Changes since previous RFC version of my patches: [firewire-lib] - support 32bit sample for capturing (remove 16bit sample support) - limit the number of MIDI conformant data channel by 1 - change the way to trigger MIDI messages - fix some bugs to playback/capture samples in 'dual wire' mode - fix wrong mappling of channels in 'dual wire' mode - add unit/subunit support for PLUG INFO command - add limit for dev_err() [fireworks] - improve return value for hwdep interface [bebob] - reconfirm supported devices - read root directory in config rom to get vendor name charactors - add a new way to detect clock source - remove model specific operations for PreSonus - split codes for M-Audio to three patches - remove control interface except for M-Audio special devices Best regards, Takashi Sakamoto Takashi Sakamoto (38): firewire-lib: Rename functions, structure, member for AMDTP firewire-lib: Add macros instead of fixed value for AMDTP firewire-lib: Add 'direction' member to 'amdtp_stream' structure firewire-lib: Split some codes into functions to reuse for both streams firewire-lib: Add support for AMDTP in-stream and PCM capture firewire-lib: Add support for MIDI capture/playback firewire-lib: Give syt value as parameter to handle_out_packet() firewire-lib: Add support for duplex streams synchronization in blocking mode firewire-lib: Add sort function for transmitted packet firewire-lib: Add transfer delay to synchronized duplex streams firewire-lib: Add support for channel mapping firewire-lib: Rename macros, variables and functions for CMP firewire-lib: Add 'direction' member to 'cmp_connection' structure firewire-lib: Add handling output connection by CMP firewire-lib: Add a new function to check others' connection firewire-lib: Add some AV/C general commands firewire-lib: Add quirks for Fireworks fireworks: Add skelton for Fireworks based devices fireworks: Add transaction and some commands fireworks: Add connection and stream management fireworks: Add proc interface for debugging purpose fireworks: Add MIDI interface fireworks: Add PCM interface fireworks: Add hwdep interface fireworks: Add command/response functionality into hwdep interface bebob: Add skelton for BeBoB based devices bebob: Add commands and connections/streams management bebob: Add proc interface for debugging purpose bebob: Add MIDI interface bebob: Add PCM interface bebob: Add hwdep interface bebob: Prepare for device specific operations bebob: Add support for Terratec PHASE, EWS series and Aureon bebob: Add support for Yamaha GO series bebob: Add support for Focusrite Saffire/SaffirePro series bebob: Add support for M-Audio usual Firewire series bebob: Add support for M-Audio special Firewire series bebob: Send a cue to load firmware for M-Audio Firewire series include/uapi/sound/asound.h | 4 +- include/uapi/sound/firewire.h | 20 + sound/firewire/Kconfig | 55 ++ sound/firewire/Makefile | 2 + sound/firewire/amdtp.c | 917 +++++++++++++++++------ sound/firewire/amdtp.h | 163 +++- sound/firewire/bebob/Makefile | 4 + sound/firewire/bebob/bebob.c | 430 +++++++++++ sound/firewire/bebob/bebob.h | 268 +++++++ sound/firewire/bebob/bebob_command.c | 352 +++++++++ sound/firewire/bebob/bebob_focusrite.c | 289 +++++++ sound/firewire/bebob/bebob_hwdep.c | 197 +++++ sound/firewire/bebob/bebob_maudio.c | 852 +++++++++++++++++++++ sound/firewire/bebob/bebob_midi.c | 156 ++++ sound/firewire/bebob/bebob_pcm.c | 430 +++++++++++ sound/firewire/bebob/bebob_proc.c | 185 +++++ sound/firewire/bebob/bebob_stream.c | 853 +++++++++++++++++++++ sound/firewire/bebob/bebob_terratec.c | 68 ++ sound/firewire/bebob/bebob_yamaha.c | 50 ++ sound/firewire/cmp.c | 217 ++++-- sound/firewire/cmp.h | 14 +- sound/firewire/dice.c | 46 +- sound/firewire/fcp.c | 151 ++++ sound/firewire/fcp.h | 21 + sound/firewire/fireworks/Makefile | 4 + sound/firewire/fireworks/fireworks.c | 330 ++++++++ sound/firewire/fireworks/fireworks.h | 222 ++++++ sound/firewire/fireworks/fireworks_command.c | 397 ++++++++++ sound/firewire/fireworks/fireworks_hwdep.c | 299 ++++++++ sound/firewire/fireworks/fireworks_midi.c | 166 ++++ sound/firewire/fireworks/fireworks_pcm.c | 449 +++++++++++ sound/firewire/fireworks/fireworks_proc.c | 206 +++++ sound/firewire/fireworks/fireworks_stream.c | 354 +++++++++ sound/firewire/fireworks/fireworks_transaction.c | 327 ++++++++ sound/firewire/speakers.c | 94 +-- 35 files changed, 8196 insertions(+), 396 deletions(-) create mode 100644 sound/firewire/bebob/Makefile create mode 100644 sound/firewire/bebob/bebob.c create mode 100644 sound/firewire/bebob/bebob.h create mode 100644 sound/firewire/bebob/bebob_command.c create mode 100644 sound/firewire/bebob/bebob_focusrite.c create mode 100644 sound/firewire/bebob/bebob_hwdep.c create mode 100644 sound/firewire/bebob/bebob_maudio.c create mode 100644 sound/firewire/bebob/bebob_midi.c create mode 100644 sound/firewire/bebob/bebob_pcm.c create mode 100644 sound/firewire/bebob/bebob_proc.c create mode 100644 sound/firewire/bebob/bebob_stream.c create mode 100644 sound/firewire/bebob/bebob_terratec.c create mode 100644 sound/firewire/bebob/bebob_yamaha.c create mode 100644 sound/firewire/fireworks/Makefile create mode 100644 sound/firewire/fireworks/fireworks.c create mode 100644 sound/firewire/fireworks/fireworks.h create mode 100644 sound/firewire/fireworks/fireworks_command.c create mode 100644 sound/firewire/fireworks/fireworks_hwdep.c create mode 100644 sound/firewire/fireworks/fireworks_midi.c create mode 100644 sound/firewire/fireworks/fireworks_pcm.c create mode 100644 sound/firewire/fireworks/fireworks_proc.c create mode 100644 sound/firewire/fireworks/fireworks_stream.c create mode 100644 sound/firewire/fireworks/fireworks_transaction.c -- 1.8.3.2

4 50

[alsa-devel] Workaround for "[AO_ALSA] Unable to set buffer time near: Invalid argument" error on HDMI 5.1 output
by Thilo Schulz 09 Feb '14

09 Feb '14

Hi, the purpose of this message is just to document my workaround solution I found to a bug (probably in ALSA) on a stable Debian system. This took me the better of a fine sunday and I'm posting this to spare anyone who might run into a similar problem the same trouble. And I'm posting this so the alsa devs may look into this (if it's not already fixed in recent alsa versions). Using mplayer, I recently tried outputting a 6 channel test wave file to the HDMI output on my Acer Aspire with a HDA Intel PCH card. 2 channels work fine, but trying to output 6 channels results in: $ mplayer test.wav -ao alsa:device=hdmi -channels 6 [...] [AO_ALSA] Unable to set buffer time near: Invalid argument Failed to initialize audio driver 'alsa:device=hdmi' Obviously, this code from mplayer's ao_alsa.c fails: unsigned int alsa_buffer_time = 500000; /* 0.5 s */ [...] if ((err = snd_pcm_hw_params_set_buffer_time_near(alsa_handler, \ alsa_hwparams, &alsa_buffer_time, NULL)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetBufferTimeNear, snd_strerror(err)); return 0; } A similar issue seems to have been discussed in this thread on alsa-devel: http://mailman.alsa-project.org/pipermail/alsa-devel/2011-March/037450.html And as a workaround it is suggested there to raise some memory preallocation parameter: $ cat /proc/asound/card0/pcm3p/sub0/prealloc 64 like so: $ echo 128 > /proc/asound/card0/pcm3p/sub0/prealloc this indeed fixes the problem and mplayer plays fine. Now here comes the curious part. _Lowering_ the parameter like this: $ echo 63 > /proc/asound/card0/pcm3p/sub0/prealloc makes the problem go away as well. Strange. For followup replies, please CC me, as I'm subscribed to neither mailing lists. To ALSA devs: I would be willing to help debug this issue if you'd give a few instructions (where to put a few printk messages in the module, etc) -- Beste regards, Thilo Schulz

1 0

[alsa-devel] [PATCH 25/28] Remove MACH_SMDKV310
by Richard Weinberger 09 Feb '14

09 Feb '14

The symbol is an orphan, get rid of it. Signed-off-by: Richard Weinberger <richard(a)nod.at> --- sound/soc/samsung/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 7ea576c..33fd372 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -145,7 +145,7 @@ config SND_SOC_SAMSUNG_RX1950_UDA1380 config SND_SOC_SAMSUNG_SMDK_WM9713 tristate "SoC AC97 Audio support for SMDK with WM9713" - depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110 || MACH_SMDKV310) + depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110) select SND_SOC_WM9713 select SND_SAMSUNG_AC97 help -- 1.8.4.2

1 0

[alsa-devel] [PATCH v2 0/7] simple card
by Xiubo Li 08 Feb '14

08 Feb '14

This patch series has been test on VF610 Tower board. Changed in V2: - Revise the problem from Jean-Francois Moine. - Add binding documentation patch. Xiubo Li (7): ASoC: simple-card: fix __asoc_simple_card_dai_init ASoC: simple-card: simplify the daifmt code ASoC: simple-card: Add snd_card's name parsing from DT node support ASoC: simple-card: add tdm slot supports ASoC: add snd_soc_of_parse_audio_simple_widgets for DT ASoC: simple-card: add off-codec widgets supports. ASoC: binding: for new properties documenting and usage .../devicetree/bindings/sound/simple-card.txt | 17 +++- include/sound/simple_card.h | 8 ++ include/sound/soc.h | 2 + sound/soc/generic/simple-card.c | 112 +++++++++++++++++---- sound/soc/soc-core.c | 87 ++++++++++++++++ 5 files changed, 204 insertions(+), 22 deletions(-) -- 1.8.4

3 16

[alsa-devel] v3.14-rc1 locking bug in system suspend
by Shawn Guo 08 Feb '14

08 Feb '14

I start seeing the following locking bug on IMX6 with suspend operation after moving to v3.14-rc1. Before I start looking into the issue, I would like to confirm if it's an IMX specific problem or one seen on other platforms. Please ask if you need more info. Shawn root@arm:~# echo mem > /sys/power/state PM: Syncing filesystems ... done. PM: Preparing system for mem sleep mmc1: card e624 removed Freezing user space processes ... (elapsed 0.002 seconds) done. Freezing remaining freezable tasks ... (elapsed 0.002 seconds) done. PM: Entering mem sleep INFO: trying to register non-static key. the code is fine but needs lockdep annotation. turning off the locking correctness validator. CPU: 0 PID: 1849 Comm: bash Not tainted 3.14.0-rc1+ #1648 Backtrace: [<80012144>] (dump_backtrace) from [<800122e4>] (show_stack+0x18/0x1c) r6:808c48a0 r5:00000000 r4:00000000 r3:00000000 [<800122cc>] (show_stack) from [<80637ac0>] (dump_stack+0x78/0x94) [<80637a48>] (dump_stack) from [<80064fa0>] (__lock_acquire+0x18f4/0x1ce0) r4:00000002 r3:bda3f200 [<800636ac>] (__lock_acquire) from [<80065894>] (lock_acquire+0x68/0x7c) r10:bd906000 r9:bda3f200 r8:00000000 r7:00000000 r6:60000013 r5:bd906000 r4:00000000 [<8006582c>] (lock_acquire) from [<8063c938>] (mutex_lock_nested+0x5c/0x3b8) r7:00000000 r6:80dfc78c r5:804be444 r4:bf122928 [<8063c8dc>] (mutex_lock_nested) from [<804be444>] (snd_soc_suspend+0x34/0x42c) r10:00000000 r9:00000000 r8:00000000 r7:bf1c4444 r6:bf1c4410 r5:be8fd150 r4:be8fd010 [<804be410>] (snd_soc_suspend) from [<8034392c>] (platform_pm_suspend+0x34/0x64) r10:00000000 r8:00000000 r7:bf1c4444 r6:bf1c4410 r5:803438f8 r4:bf1c4410 [<803438f8>] (platform_pm_suspend) from [<80348e18>] (dpm_run_callback.isra.7+0x34/0x6c) [<80348de4>] (dpm_run_callback.isra.7) from [<80349354>] (__device_suspend+0x10c/0x220) r9:808dd974 r8:808c4a5c r6:00000002 r5:80e5001c r4:bf1c4410 [<80349248>] (__device_suspend) from [<8034a338>] (dpm_suspend+0x60/0x220) r7:bf1c4410 r6:808dd90c r5:80e5001c r4:bf1c44c0 [<8034a2d8>] (dpm_suspend) from [<8034a790>] (dpm_suspend_start+0x60/0x68) r10:8079a818 r9:00000000 r8:00000004 r7:80dfbe90 r6:80641eec r5:00000000 r4:00000002 [<8034a730>] (dpm_suspend_start) from [<8006a788>] (suspend_devices_and_enter+0x74/0x318) r4:00000003 r3:80dfbe98 [<8006a714>] (suspend_devices_and_enter) from [<8006abd8>] (pm_suspend+0x1ac/0x244) r10:8079a818 r8:00000004 r7:00000003 r6:80641eec r5:00000000 r4:00000003 [<8006aa2c>] (pm_suspend) from [<80069a4c>] (state_store+0x70/0xc0) r5:00000003 r4:be9c7680 [<800699dc>] (state_store) from [<80294034>] (kobj_attr_store+0x1c/0x28) r10:bd99cb88 r8:00000000 r7:bd907f78 r6:be9c7680 r5:00000004 r4:bd99cb80 [<80294018>] (kobj_attr_store) from [<80140f90>] (sysfs_kf_write+0x54/0x58) [<80140f3c>] (sysfs_kf_write) from [<8014474c>] (kernfs_fop_write+0xc4/0x160) r6:be9c7680 r5:bd99cb80 r4:00000004 r3:80140f3c [<80144688>] (kernfs_fop_write) from [<800dfa14>] (vfs_write+0xbc/0x184) r10:00000000 r9:00000000 r8:00000000 r7:bd907f78 r6:00281c08 r5:00000004 r4:bda1bd00 [<800df958>] (vfs_write) from [<800dfe00>] (SyS_write+0x48/0x70) r10:00000000 r8:00000000 r7:00000004 r6:00281c08 r5:00000000 r4:bda1bd00 [<800dfdb8>] (SyS_write) from [<8000e800>] (ret_fast_syscall+0x0/0x48) r9:bd906000 r8:8000e9c4 r7:00000004 r6:00281c08 r5:00000004 r4:76f575e0 INFO: rcu_sched detected stalls on CPUs/tasks: { 0} (detected by 1, t=2104 jiffies, g=576, c=575, q=2) Task dump for CPU 0: bash R running 0 1849 1848 0x00000002 Backtrace: [<80065894>] (lock_acquire) from [<80061410>] (trace_hardirqs_off+0x14/0x18) Backtrace aborted due to bad frame pointer <bd907b3c> INFO: rcu_sched detected stalls on CPUs/tasks: { 0} (detected by 1, t=8424 jiffies, g=576, c=575, q=3) Task dump for CPU 0: bash R running 0 1849 1848 0x00000002 Backtrace: [<80065894>] (lock_acquire) from [<80061410>] (trace_hardirqs_off+0x14/0x18) Backtrace aborted due to bad frame pointer <bd907b3c>

1 2

[alsa-devel] [PATCH RFC 0/2] drivers/base: simplify simple DT-based components
by Jean-Francois Moine 08 Feb '14

08 Feb '14

This patch series tries to simplify the code of simple devices in case they are part of componentised subsystems, are declared in a DT, and are not using the component bin/unbind functions. Jean-Francois Moine (2): drivers/base: permit base components to omit the bind/unbind ops drivers/base: declare phandle DT nodes as components drivers/base/component.c | 21 +++++++++++++++++++-- drivers/base/core.c | 18 ++++++++++++++++++ include/linux/of.h | 2 ++ 3 files changed, 39 insertions(+), 2 deletions(-) -- 1.9.rc1

2 9

[alsa-devel] Error trying to compile driver
by Michael Weiner 08 Feb '14

08 Feb '14

I am trying to fix the sound on a laptop i am working on. Has CentOS 5.10 64bit installed and i am trying to use the alsa-compile.sh script but it fails with the following error: make -C /lib/modules/2.6.18-371.4.1.el5/build SUBDIRS=/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d CPP="gcc -E" CC="gcc" modules make[1]: Entering directory `/usr/src/kernels/2.6.18-371.4.1.el5-x86_64' CC [M] /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore/hwdep.o In file included from /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/include/adriver.h:1695, from /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore/hwdep.c:1: include/pcmcia/cistpl.h:557: error: expected specifier-qualifier-list before 'cisdata_t' include/pcmcia/cistpl.h:586: error: expected specifier-qualifier-list before 'cisdata_t' include/pcmcia/cistpl.h:590: warning: 'struct pcmcia_socket' declared inside parameter list include/pcmcia/cistpl.h:590: warning: its scope is only this definition or declaration, which is probably not what you want include/pcmcia/cistpl.h:593: warning: 'struct pcmcia_socket' declared inside parameter list include/pcmcia/cistpl.h:594: warning: 'struct pcmcia_socket' declared inside parameter list include/pcmcia/cistpl.h:595: warning: 'struct pcmcia_socket' declared inside parameter list include/pcmcia/cistpl.h:598: warning: 'struct pcmcia_socket' declared inside parameter list In file included from include/pcmcia/ds.h:23, from /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/include/adriver.h:1696, from /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore/hwdep.c:1: include/pcmcia/bulkmem.h:27: error: expected specifier-qualifier-list before 'memory_handle_t' include/pcmcia/bulkmem.h:38: warning: 'struct pcmcia_device' declared inside parameter list include/pcmcia/bulkmem.h:39: warning: 'struct pcmcia_device' declared inside parameter list In file included from /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/include/adriver.h:1696, from /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore/hwdep.c:1: include/pcmcia/ds.h:35: error: expected specifier-qualifier-list before 'win_req_t' include/pcmcia/ds.h:55: error: expected specifier-qualifier-list before 'adjust_t' make[3]: *** [/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore/hwdep.o] Error 1 make[2]: *** [/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore] Error 2 make[1]: *** [_module_/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d] Error 2 make[1]: Leaving directory `/usr/src/kernels/2.6.18-371.4.1.el5-x86_64' make: *** [compile] Error 2 Working directory /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d. Compilation of alsa-driver failed. Report this problem to the <alsa-devel(a)alsa-project.org> mailing list. Can anyone help me figure out why this is failing? Kernel is 2.6.18-371.4.1 Thank you in advance Michael Weiner --

1 0

[alsa-devel] [PATCH v3] ASoC: add RT286 CODEC driver
by bardliao＠realtek.com 07 Feb '14

07 Feb '14

From: Bard Liao <bardliao(a)realtek.com> This patch adds the ALC286 codec driver. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- * Initialize all cache registers in rt286_reg_cache, and remove rt286_reg_cache size define. --- include/sound/rt286.h | 19 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt286.c | 1373 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt286.h | 89 +++ 5 files changed, 1487 insertions(+) create mode 100644 include/sound/rt286.h create mode 100644 sound/soc/codecs/rt286.c create mode 100644 sound/soc/codecs/rt286.h diff --git a/include/sound/rt286.h b/include/sound/rt286.h new file mode 100644 index 0000000..6c0c1c9 --- /dev/null +++ b/include/sound/rt286.h @@ -0,0 +1,19 @@ +/* + * linux/sound/rt286.h -- Platform data for RT286 + * + * Copyright 2013 Realtek Microelectronics + * + * 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. + */ + +#ifndef __LINUX_SND_RT286_H +#define __LINUX_SND_RT286_H + +struct rt286_platform_data { + bool cbj_en; /*combo jack enable*/ + bool irq_en; /*IRQ enable*/ +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 983d087a..755a39d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -61,6 +61,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM3008 select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C + select SND_SOC_RT286 if I2C select SND_SOC_SGTL5000 if I2C select SND_SOC_SI476X if MFD_SI476X_CORE select SND_SOC_SN95031 if INTEL_SCU_IPC @@ -319,6 +320,9 @@ config SND_SOC_RT5631 config SND_SOC_RT5640 tristate +config SND_SOC_RT286 + tristate + #Freescale sgtl5000 codec config SND_SOC_SGTL5000 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index bc12676..b31615c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -48,6 +48,7 @@ snd-soc-pcm1792a-codec-objs := pcm1792a.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o +snd-soc-rt286-objs := rt286.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o snd-soc-alc5632-objs := alc5632.o @@ -181,6 +182,7 @@ obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o +obj-$(CONFIG_SND_SOC_RT286) += snd-soc-rt286.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o obj-$(CONFIG_SND_SOC_SI476X) += snd-soc-si476x.o diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c new file mode 100644 index 0000000..214e5fc --- /dev/null +++ b/sound/soc/codecs/rt286.c @@ -0,0 +1,1373 @@ +/* + * rt286.c -- RT286 ALSA SoC audio codec driver + * + * Copyright 2013 Realtek Semiconductor Corp. + * Author: Bard Liao <bardliao(a)realtek.com> + * + * 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. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <linux/acpi.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/rt286.h> + +#include "../../pci/hda/hda_codec.h" +#include "rt286.h" + +struct rt286_priv { + struct regmap *regmap; + struct rt286_platform_data pdata; + int sys_clk; +}; + +static unsigned int rt286_reg_cache[] = { + [NODE_ID_AUDIO_FUNCTION_GROUP] = 0x0000, + [NODE_ID_DAC_OUT1] = 0x7f7f, + [NODE_ID_DAC_OUT2] = 0x7f7f, + [NODE_ID_SPDIF] = 0x0000, + [NODE_ID_ADC_IN1] = 0x4343, + [NODE_ID_ADC_IN2] = 0x4343, + [NODE_ID_MIC1] = 0x0000, + [NODE_ID_MIXER_IN] = 0x000b, + [NODE_ID_MIXER_OUT1] = 0x0002, + [NODE_ID_MIXER_OUT2] = 0x0000, + [NODE_ID_DMIC1] = 0x0000, + [NODE_ID_DMIC2] = 0x0000, + [NODE_ID_LINE1] = 0x0000, + [NODE_ID_BEEP] = 0x0000, + [NODE_ID_VENDOR_REGISTERS] = 0x0000, + [NODE_ID_SPK_OUT] = 0x0000, + [NODE_ID_HP_OUT] = 0x0000, + [NODE_ID_MIXER_IN1] = 0x0000, + [NODE_ID_MIXER_IN2] = 0x0000, +}; + +static int rt286_hw_read(void *context, unsigned int reg, unsigned int *value) +{ + struct i2c_client *client = context; + struct i2c_msg xfer[2]; + int ret; + unsigned int buf = 0x0; + + if (reg <= 0xff) { /*read cache*/ + *value = rt286_reg_cache[reg]; + return 0; + } + reg = cpu_to_be32(reg); + /* Write register */ + xfer[0].addr = client->addr; + xfer[0].flags = 0; + xfer[0].len = 4; + xfer[0].buf = (u8 *)® + + /* Read data */ + xfer[1].addr = client->addr; + xfer[1].flags = I2C_M_RD; + xfer[1].len = 4; + xfer[1].buf = (u8 *)&buf; + + ret = i2c_transfer(client->adapter, xfer, 2); + if (ret < 0) + return ret; + else if (ret != 2) + return -EIO; + + *value = be32_to_cpu(buf); + + return 0; +} + +static int rt286_hw_write(void *context, unsigned int reg, unsigned int value) +{ + struct i2c_client *client = context; + u8 data[4]; + int ret; + + if (reg <= 0xff) { /*write cache*/ + rt286_reg_cache[reg] = value; + return 0; + } + data[0] = (reg >> 24) & 0xff; + data[1] = (reg >> 16) & 0xff; + data[2] = (reg >> 8) & 0xff; + data[3] = reg & 0xff; + + ret = i2c_master_send(client, data, 4); + if (ret == 4) + return 0; + if (ret < 0) + return ret; + else + return -EIO; +} + +static unsigned int rt286_read(struct snd_soc_codec *codec, + unsigned int vid, + unsigned int nid, + unsigned int data) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int verb; + unsigned int buf = 0x0; + + verb = (nid << 20) | (vid << 8) | data; + + regmap_read(rt286->regmap, verb, &buf); + return buf; +} + +static int rt286_write(struct snd_soc_codec *codec, + unsigned int vid, + unsigned int nid, + unsigned int value) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int reg; + + reg = (nid << 20) | (vid << 8) | value; + + return regmap_write(rt286->regmap, reg, 0); +} + +static int rt286_update_bits(struct snd_soc_codec *codec, unsigned int vid, + unsigned int nid, unsigned int data, + unsigned int mask, unsigned int value) +{ + unsigned int old, new; + int change, ret; + + old = rt286_read(codec, vid | 0x800, nid, data); + new = (old & ~mask) | (value & mask); + change = old != new; + + if (change) { + ret = rt286_write(codec, vid, nid, new); + if (ret < 0) { + dev_err(codec->dev, + "Failed to write private reg: %d\n", ret); + goto err; + } + } + return change; + +err: + return ret; +} + +static int rt286_index_write(struct snd_soc_codec *codec, + unsigned int wid, unsigned int index, unsigned int data) +{ + int ret; + + ret = rt286_write(codec, AC_VERB_SET_COEF_INDEX, + wid, index); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private addr: %d\n", ret); + goto err; + } + ret = rt286_write(codec, AC_VERB_SET_PROC_COEF, + wid, data); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private value: %d\n", ret); + goto err; + } + + return 0; +err: + return ret; +} + +static unsigned int rt286_index_read(struct snd_soc_codec *codec, + unsigned int wid, unsigned int index) +{ + int ret; + + ret = rt286_write(codec, AC_VERB_SET_COEF_INDEX, + wid, index); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private addr: %d\n", ret); + return ret; + } + return rt286_read(codec, AC_VERB_GET_PROC_COEF, + wid, index); +} + +static int rt286_index_update_bits(struct snd_soc_codec *codec, + unsigned int wid, unsigned int index, + unsigned int mask, unsigned int data) +{ + unsigned int old, new; + int change, ret; + + old = rt286_index_read(codec, wid, index); + new = (old & ~mask) | (data & mask); + change = old != new; + + if (change) { + ret = rt286_index_write(codec, wid, index, new); + if (ret < 0) { + dev_err(codec->dev, + "Failed to write private reg: %d\n", ret); + goto err; + } + } + return change; + +err: + return ret; +} + +static int rt286_support_power_controls[] = { + NODE_ID_DAC_OUT1, + NODE_ID_DAC_OUT2, + NODE_ID_ADC_IN1, + NODE_ID_ADC_IN2, + NODE_ID_MIC1, + NODE_ID_DMIC1, + NODE_ID_DMIC2, + NODE_ID_SPK_OUT, + NODE_ID_HP_OUT, +}; +#define RT286_POWER_REG_LEN ARRAY_SIZE(rt286_support_power_controls) + +static const struct reg_default rt286_reg[] = { + { NODE_ID_DAC_OUT1, 0x7f7f }, + { NODE_ID_DAC_OUT2, 0x7f7f }, + { NODE_ID_SPDIF, 0x0000 }, + { NODE_ID_ADC_IN1, 0x4343 }, + { NODE_ID_ADC_IN2, 0x4343 }, + { NODE_ID_MIC1, 0x0000 }, + { NODE_ID_MIXER_IN, 0x000b }, + { NODE_ID_MIXER_OUT1, 0x0002 }, + { NODE_ID_MIXER_OUT2, 0x0000 }, + { NODE_ID_SPK_OUT, 0x0000 }, + { NODE_ID_HP_OUT, 0x0000 }, + { NODE_ID_MIXER_IN1, 0x0005 }, + { NODE_ID_MIXER_IN2, 0x0005 }, +}; + +static bool rt286_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NODE_ID_DAC_OUT1: + case NODE_ID_DAC_OUT2: + case NODE_ID_ADC_IN1: + case NODE_ID_ADC_IN2: + case NODE_ID_MIC1: + case NODE_ID_SPDIF: + case NODE_ID_MIXER_IN: + case NODE_ID_MIXER_OUT1: + case NODE_ID_MIXER_OUT2: + case NODE_ID_DMIC1: + case NODE_ID_DMIC2: + case NODE_ID_SPK_OUT: + case NODE_ID_HP_OUT: + case NODE_ID_MIXER_IN1: + case NODE_ID_MIXER_IN2: + return true; + default: + return false; + } +} + +int rt286_jack_detect(struct snd_soc_codec *codec, bool *hp, bool *mic) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int val; + int i; + + *hp = false; + *mic = false; + + if (rt286->pdata.cbj_en) { + *hp = rt286_read(codec, AC_VERB_GET_PIN_SENSE, + NODE_ID_HP_OUT, 0); + if (*hp) { + /* power on HV,VERF */ + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x03, 0x1001, 0x0); + /* power LDO1 */ + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x08, 0x4, 0x4); + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_MIC1, 0x24); + rt286_write(codec, AC_VERB_SET_COEF_INDEX, + 0x20, 0x50); + val = rt286_read(codec, AC_VERB_GET_PROC_COEF, + 0x20, 0x50); + + msleep(200); + i = 40; + while (((val & 0x0800) == 0) && (i > 0)) { + rt286_write(codec, AC_VERB_SET_COEF_INDEX, + 0x20, 0x50); + val = rt286_read(codec, + AC_VERB_GET_PROC_COEF, 0x20, 0x50); + i--; + msleep(20); + } + + if (0x0400 == (val & 0x0700)) { + *mic = false; + + rt286_write(codec, + AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_MIC1, 0x20); + /* power off HV,VERF */ + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x03, 0x1001, 0x1001); + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x04, 0xC000, 0x0000); + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x4f, 0x0030, 0x0000); + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x02, 0xc000, 0x0000); + } else if ((0x0200 == (val & 0x0700)) || + (0x0100 == (val & 0x0700))) { + *mic = true; + } else { + *mic = false; + } + + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x20, 0x0060, 0x0000); + } else { + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x20, 0x0060, 0x0020); + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x04, 0xC000, 0x8000); + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x4f, 0x0030, 0x0020); + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x02, 0xc000, 0x8000); + + *mic = false; + } + } else { + *hp = rt286_read(codec, AC_VERB_GET_PIN_SENSE, + NODE_ID_HP_OUT, 0); + *mic = rt286_read(codec, AC_VERB_GET_PIN_SENSE, + NODE_ID_MIC1, 0); + } + + + /* Clear IRQ */ + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x33, 0x1, 0x1); + return 0; +} +EXPORT_SYMBOL_GPL(rt286_jack_detect); + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6350, 50, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); + +static int rt286_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int io = 0x8000; + + if (NODE_ID_ADC_IN1 == mc->reg) + io = 0; + ucontrol->value.integer.value[0] = + rt286_read(codec, AC_VERB_GET_AMP_GAIN_MUTE, + mc->reg, 0x2000 | io) & 0x7f; + ucontrol->value.integer.value[1] = + rt286_read(codec, AC_VERB_GET_AMP_GAIN_MUTE, + mc->reg, io) & 0x7f; + + return 0; +} + +static int rt286_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int vall, valr, io; + + if (NODE_ID_ADC_IN1 == mc->reg) + io = 0x4000; + else + io = 0x8000; + vall = ucontrol->value.integer.value[0]; + valr = ucontrol->value.integer.value[1]; + if (vall == valr) { + vall = vall | 0x3000 | io; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, vall); + } else { + vall = vall | 0x2000 | io; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, vall); + valr = valr | 0x1000 | io; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, valr); + } + + return 0; +} + +static int rt286_mic_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = + rt286_read(codec, AC_VERB_GET_AMP_GAIN_MUTE, + mc->reg, 0x0) & 0x3; + + return 0; +} + +static int rt286_mic_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int val; + + val = ucontrol->value.integer.value[0]; + val = val | 0x7000; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, val); + + return 0; +} + +static int rt286_playback_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + + ucontrol->value.integer.value[0] = + !(rt286_read(codec, AC_VERB_GET_AMP_GAIN_MUTE, + mc->reg, 0xa000) & 0x80); + ucontrol->value.integer.value[1] = + !(rt286_read(codec, AC_VERB_GET_AMP_GAIN_MUTE, + mc->reg, 0x8000) & 0x80); + + return 0; +} + +static int rt286_playback_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int vall, valr, val = 0; + + vall = (!ucontrol->value.integer.value[0] << 7); + valr = (!ucontrol->value.integer.value[1] << 7); + if (vall) + val |= 0x8000; + if (valr) + val |= 0x0080; + + if (vall == valr) { + vall = vall | 0xb000; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, vall); + } else { + vall = vall | 0xa000; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, vall); + valr = valr | 0x9000; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, valr); + } + snd_soc_update_bits(codec, mc->reg, 0x8080, val); + + return 0; +} + +static const struct snd_kcontrol_new rt286_snd_controls[] = { + SOC_DOUBLE_EXT_TLV("DAC0 Playback Volume", NODE_ID_DAC_OUT1, + 8, 0, 0x7f, 0, + rt286_vol_get, rt286_vol_put, + out_vol_tlv), + SOC_DOUBLE_EXT_TLV("ADC0 Capture Volume", NODE_ID_ADC_IN1, + 8, 0, 0x7f, 0, + rt286_vol_get, rt286_vol_put, + out_vol_tlv), + SOC_SINGLE_EXT_TLV("AMIC Volume", NODE_ID_MIC1, 0, 0x3, 0, + rt286_mic_gain_get, rt286_mic_gain_put, + mic_vol_tlv), + SOC_DOUBLE_EXT("Headphone Playback Switch", NODE_ID_HP_OUT, + 15, 8, 1, 1, rt286_playback_switch_get, + rt286_playback_switch_put), + SOC_DOUBLE_EXT("Speaker Playback Switch", NODE_ID_SPK_OUT, + 15, 8, 1, 1, rt286_playback_switch_get, + rt286_playback_switch_put), +}; + +static void hw_configure(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + int i; + + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_AUDIO_FUNCTION_GROUP, AC_PWRST_D3); + for (i = 0; i < RT286_POWER_REG_LEN; i++) { + rt286_write(codec, AC_VERB_SET_POWER_STATE, + rt286_support_power_controls[i], AC_PWRST_D3); + } + + if (rt286->pdata.irq_en) + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x33, 0x0002, 0x0002); + + if (!rt286->pdata.cbj_en) { + rt286_index_write(codec, + NODE_ID_VENDOR_REGISTERS, 0x50, 0x0000); + rt286_index_write(codec, + NODE_ID_VENDOR_REGISTERS, 0x19, 0x0816); + rt286_index_write(codec, + NODE_ID_VENDOR_REGISTERS, 0x20, 0x0000); + } + + mdelay(10); + + /* Set configuration default */ + rt286_write(codec, AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, + NODE_ID_MIC1, 0x00); + rt286_write(codec, AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, + NODE_ID_DMIC1, 0x00); + rt286_write(codec, AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, + NODE_ID_DMIC2, 0x00); + + mdelay(10); + + if (rt286->pdata.cbj_en) + rt286_index_write(codec, + NODE_ID_VENDOR_REGISTERS, 0x4f, 0x5029); + else + rt286_index_write(codec, + NODE_ID_VENDOR_REGISTERS, 0x4f, 0xb029); + + /*Power down LDO2*/ + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x08, 0x0008, 0x0000); +} + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt286_front_mix[] = { + SOC_DAPM_SINGLE("DAC Switch", NODE_ID_MIXER_OUT1, + RT286_M_FRONT_DAC_SFT, 1, 1), + SOC_DAPM_SINGLE("RECMIX Switch", NODE_ID_MIXER_OUT1, + RT286_M_FRONT_REC_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt286_rec_mix[] = { + SOC_DAPM_SINGLE("Beep Switch", NODE_ID_MIXER_IN, + RT286_M_REC_BEEP_SFT, 1, 1), + SOC_DAPM_SINGLE("Line1 Switch", NODE_ID_MIXER_IN, + RT286_M_REC_LINE1_SFT, 1, 1), + SOC_DAPM_SINGLE("Mic1 Switch", NODE_ID_MIXER_IN, + RT286_M_REC_MIC1_SFT, 1, 1), + SOC_DAPM_SINGLE("I2S Switch", NODE_ID_MIXER_IN, + RT286_M_REC_I2S_SFT, 1, 1), +}; + +/* ADC0 source */ +static const char * const rt286_adc_src[] = { + "Mic", "Dmic" +}; + +static int rt286_adc_values[] = { + 0, 5, +}; + +static const SOC_VALUE_ENUM_SINGLE_DECL( + rt286_adc0_enum, NODE_ID_MIXER_IN1, RT286_ADC_SEL_SFT, + RT286_ADC_SEL_MASK, rt286_adc_src, rt286_adc_values); + +static const struct snd_kcontrol_new rt286_adc0_mux = + SOC_DAPM_VALUE_ENUM("ADC 0 source", rt286_adc0_enum); + +static const SOC_VALUE_ENUM_SINGLE_DECL( + rt286_adc1_enum, NODE_ID_MIXER_IN2, RT286_ADC_SEL_SFT, + RT286_ADC_SEL_MASK, rt286_adc_src, rt286_adc_values); + +static const struct snd_kcontrol_new rt286_adc1_mux = + SOC_DAPM_VALUE_ENUM("ADC 1 source", rt286_adc1_enum); + +/* HP-OUT source */ +static const char * const rt286_hpo_src[] = { + "Front", "Surround" +}; + +static const SOC_ENUM_SINGLE_DECL(rt286_hpo_enum, NODE_ID_HP_OUT, + RT286_HP_SEL_SFT, rt286_hpo_src); + +static const struct snd_kcontrol_new rt286_hpo_mux = +SOC_DAPM_ENUM("HPO source", rt286_hpo_enum); + +/* SPK-OUT source */ +static const char * const rt286_spo_src[] = { + "Front", "Surround" +}; + +static const SOC_ENUM_SINGLE_DECL(rt286_spo_enum, NODE_ID_SPK_OUT, + RT286_SPK_SEL_SFT, rt286_spo_src); + +static const struct snd_kcontrol_new rt286_spo_mux = +SOC_DAPM_ENUM("SPO source", rt286_spo_enum); + +/* SPDIF source */ +static const char * const rt286_spdif_src[] = { + "PCM-IN 0", "PCM-IN 1", "SP-OUT", "PP" +}; + +static const SOC_ENUM_SINGLE_DECL(rt286_spdif_enum, NODE_ID_SPDIF, + RT286_SPDIF_SEL_SFT, rt286_spdif_src); + +static const struct snd_kcontrol_new rt286_spdif_mux = +SOC_DAPM_ENUM("SPDIF source", rt286_spdif_enum); + + +static int rt286_spk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_SPK_OUT, AC_PWRST_D0); + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_SPK_OUT, 0x40); + rt286_write(codec, AC_VERB_SET_EAPD_BTLENABLE, + NODE_ID_SPK_OUT, 0x0002); + break; + case SND_SOC_DAPM_PRE_PMD: + rt286_write(codec, AC_VERB_SET_EAPD_BTLENABLE, + NODE_ID_SPK_OUT, 0x0000); + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_SPK_OUT, 0x00); + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_SPK_OUT, AC_PWRST_D3); + break; + + default: + return 0; + } + + return 0; +} + +static int rt286_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + unsigned int val; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + val = snd_soc_read(codec, NODE_ID_HP_OUT) & 0x8080; + switch (val) { + case 0x0: + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, + NODE_ID_HP_OUT, 0xb000); + break; + case 0x8000: + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, + NODE_ID_HP_OUT, 0x9000); + break; + case 0x0080: + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, + NODE_ID_HP_OUT, 0xa000); + break; + } + if (val != 0x8080) + mdelay(100); + break; + + case SND_SOC_DAPM_PRE_PMD: + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, + NODE_ID_HP_OUT, 0xb080); + break; + + default: + return 0; + } + + return 0; +} + +static int rt286_set_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt286_write(codec, AC_VERB_SET_POWER_STATE, + w->reg, AC_PWRST_D0); + break; + case SND_SOC_DAPM_PRE_PMD: + rt286_write(codec, AC_VERB_SET_POWER_STATE, + w->reg, AC_PWRST_D3); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_set_dmic1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_DMIC1, 0x20); + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_DMIC1, AC_PWRST_D0); + break; + case SND_SOC_DAPM_PRE_PMD: + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_DMIC1, AC_PWRST_D3); + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_DMIC1, 0x0); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_hp_pow_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_HP_OUT, AC_PWRST_D0); + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_HP_OUT, 0x40); + break; + + case SND_SOC_DAPM_PRE_PMD: + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_HP_OUT, 0x00); + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_HP_OUT, AC_PWRST_D3); + break; + + default: + return 0; + } + + return 0; +} + +static int rt286_adc_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + struct soc_enum *e; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (rt286->pdata.cbj_en) { + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x04, 0xC000, 0x8000); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x4f, 0x0030, 0x0020); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x02, 0xc000, 0x8000); + } + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x0f, 0x0020, 0x0000); + rt286_write(codec, AC_VERB_SET_POWER_STATE, + w->reg, AC_PWRST_D0); + rt286_update_bits(codec, AC_VERB_SET_AMP_GAIN_MUTE, + w->reg, 0, 0x7080, 0x7000); + break; + case SND_SOC_DAPM_PRE_PMD: + rt286_update_bits(codec, AC_VERB_SET_AMP_GAIN_MUTE, + w->reg, 0, 0x7080, 0x7080); + rt286_write(codec, AC_VERB_SET_POWER_STATE, + w->reg, AC_PWRST_D3); + break; + case SND_SOC_DAPM_POST_REG: + e = (struct soc_enum *)kcontrol->private_value; + rt286_write(codec, AC_VERB_SET_CONNECT_SEL, e->reg, + snd_soc_read(codec, e->reg) & RT286_ADC_SEL_MASK); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_out_mux_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + + switch (event) { + case SND_SOC_DAPM_POST_REG: + rt286_write(codec, AC_VERB_SET_CONNECT_SEL, + e->reg, snd_soc_read(codec, e->reg) & 0x1); + break; + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt286_dapm_widgets[] = { + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC1 Pin"), + SND_SOC_DAPM_INPUT("DMIC2 Pin"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("LINE1"), + SND_SOC_DAPM_INPUT("Beep"), + + /* DMIC */ + SND_SOC_DAPM_PGA_E("DMIC1", NODE_ID_DMIC1, 15, 0, + NULL, 0, rt286_set_dmic1_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_E("DMIC2", NODE_ID_DMIC2, 15, 0, + NULL, 0, rt286_set_power_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SUPPLY("DMIC Receiver", SND_SOC_NOPM, + 0, 0, NULL, 0), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIX", SND_SOC_NOPM, 0, 0, + rt286_rec_mix, ARRAY_SIZE(rt286_rec_mix)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC 0", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC 1", NULL, SND_SOC_NOPM, 0, 0), + + /* ADC Mux */ + SND_SOC_DAPM_MUX_E("ADC 0 Mux", NODE_ID_ADC_IN1, 15, 0, + &rt286_adc0_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_REG), + SND_SOC_DAPM_MUX_E("ADC 1 Mux", NODE_ID_ADC_IN2, 15, 0, + &rt286_adc1_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_REG), + + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), + + + /* Output Side */ + + /* DACs */ + SND_SOC_DAPM_DAC_E("DAC 0", NULL, NODE_ID_DAC_OUT1, + 15, 0, rt286_set_power_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_DAC_E("DAC 1", NULL, NODE_ID_DAC_OUT2, + 15, 0, rt286_set_power_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + /* Output Mux */ + SND_SOC_DAPM_MUX_E("SPK Mux", SND_SOC_NOPM, 0, 0, + &rt286_spo_mux, + rt286_out_mux_event, + SND_SOC_DAPM_POST_REG), + + SND_SOC_DAPM_MUX_E("HPO Mux", SND_SOC_NOPM, 0, 0, + &rt286_hpo_mux, + rt286_out_mux_event, + SND_SOC_DAPM_POST_REG), + + SND_SOC_DAPM_SUPPLY("HP Power", SND_SOC_NOPM, + 0, 0, rt286_hp_pow_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + + SND_SOC_DAPM_MUX("SPDIF Mux", SND_SOC_NOPM, 0, 0, + &rt286_spdif_mux), + + /* Output Mixer */ + SND_SOC_DAPM_MIXER("Front", SND_SOC_NOPM, 0, 0, + rt286_front_mix, ARRAY_SIZE(rt286_front_mix)), + SND_SOC_DAPM_PGA("Surround", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Output Pga */ + SND_SOC_DAPM_PGA_E("SPO", SND_SOC_NOPM, 0, 0, + NULL, 0, rt286_spk_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_E("HPO", SND_SOC_NOPM, 0, 0, + NULL, 0, rt286_hp_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("SPOL"), + SND_SOC_DAPM_OUTPUT("SPOR"), + SND_SOC_DAPM_OUTPUT("HPO Pin"), + SND_SOC_DAPM_OUTPUT("SPDIF"), +}; + +static const struct snd_soc_dapm_route rt286_dapm_routes[] = { + {"DMIC1", NULL, "DMIC1 Pin"}, + {"DMIC2", NULL, "DMIC2 Pin"}, + {"DMIC1", NULL, "DMIC Receiver"}, + {"DMIC2", NULL, "DMIC Receiver"}, + + {"RECMIX", "Beep Switch", "Beep"}, + {"RECMIX", "Line1 Switch", "LINE1"}, + {"RECMIX", "Mic1 Switch", "MIC1"}, + + {"ADC 0 Mux", "Dmic", "DMIC1"}, + {"ADC 0 Mux", "Mic", "MIC1"}, + {"ADC 1 Mux", "Dmic", "DMIC2"}, + {"ADC 1 Mux", "Mic", "MIC1"}, + + {"ADC 0", NULL, "ADC 0 Mux"}, + {"ADC 1", NULL, "ADC 1 Mux"}, + + {"AIF1TX", NULL, "ADC 0"}, + {"AIF2TX", NULL, "ADC 1"}, + + {"DAC 0", NULL, "AIF1RX"}, + {"DAC 1", NULL, "AIF2RX"}, + + {"Front", "DAC Switch", "DAC 0"}, + {"Front", "RECMIX Switch", "RECMIX"}, + + {"Surround", NULL, "DAC 1"}, + + {"SPK Mux", "Front", "Front"}, + {"SPK Mux", "Surround", "Surround"}, + + {"HPO Mux", "Front", "Front"}, + {"HPO Mux", "Surround", "Surround"}, + + {"SPO", NULL, "SPK Mux"}, + {"HPO", NULL, "HPO Mux"}, + {"HPO", NULL, "HP Power"}, + + {"SPOL", NULL, "SPO"}, + {"SPOR", NULL, "SPO"}, + {"HPO Pin", NULL, "HPO"}, +}; + +static int rt286_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0; + int d_len_code; + + switch (params_rate(params)) { + /* bit 14 0:48K 1:44.1K */ + case 44100: + val |= 0x4000; + break; + case 48000: + break; + default: + dev_err(codec->dev, "Unsupported sample rate %d\n", + params_rate(params)); + return -EINVAL; + } + switch (rt286->sys_clk) { + case 12288000: + case 24576000: + if (params_rate(params) != 48000) { + dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n", + params_rate(params), rt286->sys_clk); + return -EINVAL; + } + break; + case 11289600: + case 22579200: + if (params_rate(params) != 44100) { + dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n", + params_rate(params), rt286->sys_clk); + return -EINVAL; + } + break; + } + + if (params_channels(params) <= 16) { + /* bit 3:0 Number of Channel */ + val |= (params_channels(params) - 1); + } else { + dev_err(codec->dev, "Unsupported channels %d\n", + params_channels(params)); + return -EINVAL; + } + + d_len_code = 0; + switch (params_format(params)) { + /* bit 6:4 Bits per Sample */ + case SNDRV_PCM_FORMAT_S16_LE: + d_len_code = 0; + val |= (0x1 << 4); + break; + case SNDRV_PCM_FORMAT_S32_LE: + d_len_code = 2; + val |= (0x4 << 4); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + d_len_code = 1; + val |= (0x2 << 4); + break; + case SNDRV_PCM_FORMAT_S24_LE: + d_len_code = 2; + val |= (0x3 << 4); + break; + case SNDRV_PCM_FORMAT_S8: + break; + default: + return -EINVAL; + } + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x0018, d_len_code << 3); + pr_debug("format val = 0x%lx\n", val); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + rt286_update_bits(codec, AC_VERB_SET_STREAM_FORMAT, + NODE_ID_DAC_OUT1, 0, 0x407f, val); + else + rt286_update_bits(codec, AC_VERB_SET_STREAM_FORMAT, + NODE_ID_ADC_IN1, 0, 0x407f, val); + + return 0; +} + +static int rt286_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x0800, 0x800); + break; + case SND_SOC_DAIFMT_CBS_CFS: + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x0800, 0x0); + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x0300, 0x0); + break; + case SND_SOC_DAIFMT_LEFT_J: + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x0300, 0x1 << 8); + break; + case SND_SOC_DAIFMT_DSP_A: + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x0300, 0x2 << 8); + break; + case SND_SOC_DAIFMT_DSP_B: + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x0300, 0x3 << 8); + break; + default: + return -EINVAL; + } + /* bit 15 Stream Type 0:PCM 1:Non-PCM */ + rt286_update_bits(codec, AC_VERB_SET_STREAM_FORMAT, + NODE_ID_DAC_OUT1, 0, 0x8000, 0x0); + rt286_update_bits(codec, AC_VERB_SET_STREAM_FORMAT, + NODE_ID_ADC_IN1, 0, 0x8000, 0x0); + + return 0; +} + +static int rt286_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + pr_debug("%s freq=%d\n", __func__, freq); + + if (RT286_SCLK_S_MCLK == clk_id) { + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x0a, 0x0100, 0x0); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x49, 0x0020, 0x0020); + } else { + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x0a, 0x0100, 0x0100); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x63, 0x4, 0x4); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x49, 0x20, 0x0); + } + + switch (freq) { + case 19200000: + case 24000000: + if (RT286_SCLK_S_MCLK == clk_id) { + dev_err(codec->dev, "Should not use MCLK\n"); + return -EINVAL; + } + break; + case 12288000: + case 11289600: + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x0a, 0x8, 0x0); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x0b, 0xfc1e, 0x0004); + break; + case 24576000: + case 22579200: + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x0a, 0x8, 0x8); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x0b, 0xfc1e, 0x5406); + break; + default: + dev_err(codec->dev, "Unsupported system clock\n"); + return -EINVAL; + } + + rt286->sys_clk = freq; + + return 0; +} + +static int rt286_set_dai_dfs(struct snd_soc_dai *dai, unsigned int fs) +{ + struct snd_soc_codec *codec = dai->codec; + + pr_debug("%s fs=%d\n", __func__, fs); + if (50 == fs) + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x1000, 0x1000); + else + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x1000, 0x0); + + + return 0; +} + +static int rt286_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_AUDIO_FUNCTION_GROUP, AC_PWRST_D0); + break; + + case SND_SOC_BIAS_OFF: + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_AUDIO_FUNCTION_GROUP, AC_PWRST_D3); + break; + + default: + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +static int rt286_probe(struct snd_soc_codec *codec) +{ + hw_configure(codec); + codec->dapm.bias_level = SND_SOC_BIAS_OFF; + + return 0; +} + +static int rt286_remove(struct snd_soc_codec *codec) +{ + rt286_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +#define RT286_STEREO_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +#define RT286_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +struct snd_soc_dai_ops rt286_aif_dai_ops = { + .hw_params = rt286_hw_params, + .set_fmt = rt286_set_dai_fmt, + .set_sysclk = rt286_set_dai_sysclk, + .set_dfs = rt286_set_dai_dfs, +}; + +struct snd_soc_dai_driver rt286_dai[] = { + { + .name = "rt286-aif1", + .id = RT286_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .ops = &rt286_aif_dai_ops, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt286 = { + .probe = rt286_probe, + .remove = rt286_remove, + .set_bias_level = rt286_set_bias_level, + .idle_bias_off = true, + .controls = rt286_snd_controls, + .num_controls = ARRAY_SIZE(rt286_snd_controls), + .dapm_widgets = rt286_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt286_dapm_widgets), + .dapm_routes = rt286_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt286_dapm_routes), +}; + +static const struct regmap_config rt286_regmap = { + .reg_bits = 32, + .val_bits = 32, + .reg_write = rt286_hw_write, + .reg_read = rt286_hw_read, + .readable_reg = rt286_readable_register, + .cache_type = REGCACHE_NONE, +}; + +static const struct i2c_device_id rt286_i2c_id[] = { + {"rt5640", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, rt286_i2c_id); + +static struct acpi_device_id rt286_acpi_match[] = { + { "INT33CA", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt286_acpi_match); + +static int rt286_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt286_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt286_priv *rt286; + struct device *dev = &i2c->dev; + int ret; + + rt286 = devm_kzalloc(&i2c->dev, + sizeof(struct rt286_priv), + GFP_KERNEL); + if (NULL == rt286) + return -ENOMEM; + + rt286->regmap = devm_regmap_init(dev, NULL, i2c, &rt286_regmap); + if (IS_ERR(rt286->regmap)) { + ret = PTR_ERR(rt286->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + i2c_set_clientdata(i2c, rt286); + + if (pdata) + rt286->pdata = *pdata; + + ret = devm_snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt286, + rt286_dai, ARRAY_SIZE(rt286_dai)); + if (ret < 0) + goto err; + + return 0; +err: + return ret; +} + +static int rt286_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + +struct i2c_driver rt286_i2c_driver = { + .driver = { + .name = "rt5640", + .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(rt286_acpi_match), + }, + .probe = rt286_i2c_probe, + .remove = rt286_i2c_remove, + .id_table = rt286_i2c_id, +}; + +static int __init rt286_modinit(void) +{ + return i2c_add_driver(&rt286_i2c_driver); +} + +module_init(rt286_modinit); + +static void __exit rt286_modexit(void) +{ + i2c_del_driver(&rt286_i2c_driver); +} + +module_exit(rt286_modexit); + +MODULE_DESCRIPTION("ASoC RT286 driver"); +MODULE_AUTHOR("Bard Liao <bardliao(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt286.h b/sound/soc/codecs/rt286.h new file mode 100644 index 0000000..c579c53 --- /dev/null +++ b/sound/soc/codecs/rt286.h @@ -0,0 +1,89 @@ +/* + * rt286.h -- RT286 ALSA SoC audio driver + * + * Copyright 2011 Realtek Microelectronics + * Author: Johnny Hsu <johnnyhsu(a)realtek.com> + * + * 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. + */ + +#ifndef __RT286_H__ +#define __RT286_H__ + +#define NODE_ID_AUDIO_FUNCTION_GROUP 0x01 +#define NODE_ID_DAC_OUT1 0x02 +#define NODE_ID_DAC_OUT2 0x03 +#define NODE_ID_ADC_IN1 0x09 +#define NODE_ID_ADC_IN2 0x08 +#define NODE_ID_MIXER_IN 0x0b +#define NODE_ID_MIXER_OUT1 0x0c +#define NODE_ID_MIXER_OUT2 0x0d +#define NODE_ID_DMIC1 0x12 +#define NODE_ID_DMIC2 0x13 +#define NODE_ID_SPK_OUT 0x14 +#define NODE_ID_MIC1 0x18 +#define NODE_ID_LINE1 0x1a +#define NODE_ID_BEEP 0x1d +#define NODE_ID_SPDIF 0x1e +#define NODE_ID_VENDOR_REGISTERS 0x20 +#define NODE_ID_HP_OUT 0x21 +#define NODE_ID_MIXER_IN1 0x22 +#define NODE_ID_MIXER_IN2 0x23 +#define TOTAL_NODE_ID 0x23 + +#define CONNECTION_INDEX_MIC1 0X0 +#define CONNECTION_INDEX_DMIC 0X5 + +/* SPDIF (0x06) */ +#define RT286_SPDIF_SEL_SFT 0 +#define RT286_SPDIF_SEL_PCM0 0 +#define RT286_SPDIF_SEL_PCM1 1 +#define RT286_SPDIF_SEL_SPOUT 2 +#define RT286_SPDIF_SEL_PP 3 + +/* RECMIX (0x0b) */ +#define RT286_M_REC_BEEP_SFT 0 +#define RT286_M_REC_LINE1_SFT 1 +#define RT286_M_REC_MIC1_SFT 2 +#define RT286_M_REC_I2S_SFT 3 + +/* Front (0x0c) */ +#define RT286_M_FRONT_DAC_SFT 0 +#define RT286_M_FRONT_REC_SFT 1 + +/* SPK-OUT (0x14) */ +#define RT286_SPK_SEL_MASK 0x1 +#define RT286_SPK_SEL_SFT 0 +#define RT286_SPK_SEL_F 0 +#define RT286_SPK_SEL_S 1 + +/* HP-OUT (0x21) */ +#define RT286_HP_SEL_MASK 0x1 +#define RT286_HP_SEL_SFT 0 +#define RT286_HP_SEL_F 0 +#define RT286_HP_SEL_S 1 + +/* ADC (0x22) (0x23) */ +#define RT286_ADC_SEL_MASK 0x7 +#define RT286_ADC_SEL_SFT 0 +#define RT286_ADC_SEL_SURR 0 +#define RT286_ADC_SEL_FRONT 1 +#define RT286_ADC_SEL_DMIC 2 +#define RT286_ADC_SEL_BEEP 4 +#define RT286_ADC_SEL_LINE1 5 +#define RT286_ADC_SEL_I2S 6 +#define RT286_ADC_SEL_MIC1 7 + +#define RT286_SCLK_S_MCLK 0 +#define RT286_SCLK_S_PLL 1 + +enum { + RT286_AIF1, + RT286_AIFS, +}; + +int rt286_jack_detect(struct snd_soc_codec *codec, bool *hp, bool *mic); + +#endif /* __RT286_H__ */ -- 1.8.1.1.439.g50a6b54

3 5

[alsa-devel] [PATCH] SoC: wm8991: Remove unused pointer in wm8991_probe()
by Christian Engelmayer 07 Feb '14

07 Feb '14

Remove unused pointer 'wm8991' in function wm8991_probe(). The last user vanished with a86652e5 (ASoC: wm8991: Convert to direct regmap API usage) Detected by Coverity: CID 1162831 Signed-off-by: Christian Engelmayer <cengelma(a)gmx.at> --- sound/soc/codecs/wm8991.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index dba0306..244eb09 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -1251,11 +1251,8 @@ static int wm8991_remove(struct snd_soc_codec *codec) static int wm8991_probe(struct snd_soc_codec *codec) { - struct wm8991_priv *wm8991; int ret; - wm8991 = snd_soc_codec_get_drvdata(codec); - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret); -- 1.8.3.2

2 3

[alsa-devel] [PATCH] ASoC: tobermory: Stop the FLL if we fail to switch SYSCLK to it
by Mark Brown 07 Feb '14

07 Feb '14

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/samsung/tobermory.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/samsung/tobermory.c b/sound/soc/samsung/tobermory.c index f21ff608a819..1807b75ccc12 100644 --- a/sound/soc/samsung/tobermory.c +++ b/sound/soc/samsung/tobermory.c @@ -44,6 +44,8 @@ static int tobermory_set_bias_level(struct snd_soc_card *card, SND_SOC_CLOCK_IN); if (ret < 0) { pr_err("Failed to set SYSCLK: %d\n", ret); + snd_soc_dai_set_pll(codec_dai, WM8962_FLL, + 0, 0, 0); return ret; } } -- 1.9.rc1

1 0

[alsa-devel] [PATCH 1/2] ASoC: pcm512x: Implement paging support
by Mark Brown 07 Feb '14

07 Feb '14

From: Mark Brown <broonie(a)linaro.org> The PCM512x devices use a paged register map covering the entire register range. Implement support for this, mapping pages in at addresses starting at 0x100 for ease of use (though since the pages are numbered from 0 there is going to be an off by one when looking at the first byte as a page number). Also mark the new registers as accessible with the exception of the coefficient RAM which is a bit fiddly and may benefit from some extra handling to linearise the blocks. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/pcm512x.c | 65 ++++++++++++++++------- sound/soc/codecs/pcm512x.h | 126 +++++++++++++++++++++++++-------------------- 2 files changed, 116 insertions(+), 75 deletions(-) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 1150381fc373..cdcb51e4c86f 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -66,22 +66,29 @@ PCM512x_REGULATOR_EVENT(1) PCM512x_REGULATOR_EVENT(2) static const struct reg_default pcm512x_reg_defaults[] = { - { PCM512x_RESET, 0x00 }, - { PCM512x_POWER, 0x00 }, - { PCM512x_MUTE, 0x00 }, - { PCM512x_DSP, 0x00 }, - { PCM512x_PLL_REF, 0x00 }, - { PCM512x_DAC_ROUTING, 0x11 }, - { PCM512x_DSP_PROGRAM, 0x01 }, - { PCM512x_CLKDET, 0x00 }, - { PCM512x_AUTO_MUTE, 0x00 }, - { PCM512x_ERROR_DETECT, 0x00 }, - { PCM512x_DIGITAL_VOLUME_1, 0x00 }, - { PCM512x_DIGITAL_VOLUME_2, 0x30 }, - { PCM512x_DIGITAL_VOLUME_3, 0x30 }, - { PCM512x_DIGITAL_MUTE_1, 0x22 }, - { PCM512x_DIGITAL_MUTE_2, 0x00 }, - { PCM512x_DIGITAL_MUTE_3, 0x07 }, + { PCM512x_RESET, 0x00 }, + { PCM512x_POWER, 0x00 }, + { PCM512x_MUTE, 0x00 }, + { PCM512x_DSP, 0x00 }, + { PCM512x_PLL_REF, 0x00 }, + { PCM512x_DAC_ROUTING, 0x11 }, + { PCM512x_DSP_PROGRAM, 0x01 }, + { PCM512x_CLKDET, 0x00 }, + { PCM512x_AUTO_MUTE, 0x00 }, + { PCM512x_ERROR_DETECT, 0x00 }, + { PCM512x_DIGITAL_VOLUME_1, 0x00 }, + { PCM512x_DIGITAL_VOLUME_2, 0x30 }, + { PCM512x_DIGITAL_VOLUME_3, 0x30 }, + { PCM512x_DIGITAL_MUTE_1, 0x22 }, + { PCM512x_DIGITAL_MUTE_2, 0x00 }, + { PCM512x_DIGITAL_MUTE_3, 0x07 }, + { PCM512x_OUTPUT_AMPLITUDE, 0x00 }, + { PCM512x_ANALOG_GAIN_CTRL, 0x00 }, + { PCM512x_UNDERVOLTAGE_PROT, 0x00 }, + { PCM512x_ANALOG_MUTE_CTRL, 0x00 }, + { PCM512x_ANALOG_GAIN_BOOST, 0x00 }, + { PCM512x_VCOM_CTRL_1, 0x00 }, + { PCM512x_VCOM_CTRL_2, 0x01 }, }; static bool pcm512x_readable(struct device *dev, unsigned int reg) @@ -141,9 +148,18 @@ static bool pcm512x_readable(struct device *dev, unsigned int reg) case PCM512x_ANALOG_MUTE_DET: case PCM512x_GPIN: case PCM512x_DIGITAL_MUTE_DET: + case PCM512x_OUTPUT_AMPLITUDE: + case PCM512x_ANALOG_GAIN_CTRL: + case PCM512x_UNDERVOLTAGE_PROT: + case PCM512x_ANALOG_MUTE_CTRL: + case PCM512x_ANALOG_GAIN_BOOST: + case PCM512x_VCOM_CTRL_1: + case PCM512x_VCOM_CTRL_2: + case PCM512x_CRAM_CTRL: return true; default: - return false; + /* There are 256 raw register addresses */ + return reg < 0xff; } } @@ -159,9 +175,11 @@ static bool pcm512x_volatile(struct device *dev, unsigned int reg) case PCM512x_ANALOG_MUTE_DET: case PCM512x_GPIN: case PCM512x_DIGITAL_MUTE_DET: + case PCM512x_CRAM_CTRL: return true; default: - return false; + /* There are 256 raw register addresses */ + return reg < 0xff; } } @@ -343,6 +361,14 @@ static struct snd_soc_codec_driver pcm512x_codec_driver = { .num_dapm_routes = ARRAY_SIZE(pcm512x_dapm_routes), }; +static const struct regmap_range_cfg pcm512x_range = { + .name = "Pages", .range_min = PCM512x_VIRT_BASE, + .range_max = PCM512x_MAX_REGISTER, + .selector_reg = PCM512x_PAGE, + .selector_mask = 0xff, + .window_start = 0, .window_len = 0x100, +}; + static const struct regmap_config pcm512x_regmap = { .reg_bits = 8, .val_bits = 8, @@ -350,6 +376,9 @@ static const struct regmap_config pcm512x_regmap = { .readable_reg = pcm512x_readable, .volatile_reg = pcm512x_volatile, + .ranges = &pcm512x_range, + .num_ranges = 1, + .max_register = PCM512x_MAX_REGISTER, .reg_defaults = pcm512x_reg_defaults, .num_reg_defaults = ARRAY_SIZE(pcm512x_reg_defaults), diff --git a/sound/soc/codecs/pcm512x.h b/sound/soc/codecs/pcm512x.h index b2f518ecb35c..e58743c965d6 100644 --- a/sound/soc/codecs/pcm512x.h +++ b/sound/soc/codecs/pcm512x.h @@ -17,66 +17,78 @@ #ifndef _SND_SOC_PCM512X #define _SND_SOC_PCM512X -#define PCM512x_PAGE_0_BASE 0 +#define PCM512x_VIRT_BASE 0x100 +#define PCM512x_PAGE_LEN 0x100 +#define PCM512x_PAGE_BASE(n) (PCM512x_VIRT_BASE + (PCM512x_PAGE_LEN * n)) #define PCM512x_PAGE 0 -#define PCM512x_RESET (PCM512x_PAGE_0_BASE + 1) -#define PCM512x_POWER (PCM512x_PAGE_0_BASE + 2) -#define PCM512x_MUTE (PCM512x_PAGE_0_BASE + 3) -#define PCM512x_PLL_EN (PCM512x_PAGE_0_BASE + 4) -#define PCM512x_SPI_MISO_FUNCTION (PCM512x_PAGE_0_BASE + 6) -#define PCM512x_DSP (PCM512x_PAGE_0_BASE + 7) -#define PCM512x_GPIO_EN (PCM512x_PAGE_0_BASE + 8) -#define PCM512x_BCLK_LRCLK_CFG (PCM512x_PAGE_0_BASE + 9) -#define PCM512x_DSP_GPIO_INPUT (PCM512x_PAGE_0_BASE + 10) -#define PCM512x_MASTER_MODE (PCM512x_PAGE_0_BASE + 12) -#define PCM512x_PLL_REF (PCM512x_PAGE_0_BASE + 13) -#define PCM512x_PLL_COEFF_0 (PCM512x_PAGE_0_BASE + 20) -#define PCM512x_PLL_COEFF_1 (PCM512x_PAGE_0_BASE + 21) -#define PCM512x_PLL_COEFF_2 (PCM512x_PAGE_0_BASE + 22) -#define PCM512x_PLL_COEFF_3 (PCM512x_PAGE_0_BASE + 23) -#define PCM512x_PLL_COEFF_4 (PCM512x_PAGE_0_BASE + 24) -#define PCM512x_DSP_CLKDIV (PCM512x_PAGE_0_BASE + 27) -#define PCM512x_DAC_CLKDIV (PCM512x_PAGE_0_BASE + 28) -#define PCM512x_NCP_CLKDIV (PCM512x_PAGE_0_BASE + 29) -#define PCM512x_OSR_CLKDIV (PCM512x_PAGE_0_BASE + 30) -#define PCM512x_MASTER_CLKDIV_1 (PCM512x_PAGE_0_BASE + 32) -#define PCM512x_MASTER_CLKDIV_2 (PCM512x_PAGE_0_BASE + 33) -#define PCM512x_FS_SPEED_MODE (PCM512x_PAGE_0_BASE + 34) -#define PCM512x_IDAC_1 (PCM512x_PAGE_0_BASE + 35) -#define PCM512x_IDAC_2 (PCM512x_PAGE_0_BASE + 36) -#define PCM512x_ERROR_DETECT (PCM512x_PAGE_0_BASE + 37) -#define PCM512x_I2S_1 (PCM512x_PAGE_0_BASE + 40) -#define PCM512x_I2S_2 (PCM512x_PAGE_0_BASE + 41) -#define PCM512x_DAC_ROUTING (PCM512x_PAGE_0_BASE + 42) -#define PCM512x_DSP_PROGRAM (PCM512x_PAGE_0_BASE + 43) -#define PCM512x_CLKDET (PCM512x_PAGE_0_BASE + 44) -#define PCM512x_AUTO_MUTE (PCM512x_PAGE_0_BASE + 59) -#define PCM512x_DIGITAL_VOLUME_1 (PCM512x_PAGE_0_BASE + 60) -#define PCM512x_DIGITAL_VOLUME_2 (PCM512x_PAGE_0_BASE + 61) -#define PCM512x_DIGITAL_VOLUME_3 (PCM512x_PAGE_0_BASE + 62) -#define PCM512x_DIGITAL_MUTE_1 (PCM512x_PAGE_0_BASE + 63) -#define PCM512x_DIGITAL_MUTE_2 (PCM512x_PAGE_0_BASE + 64) -#define PCM512x_DIGITAL_MUTE_3 (PCM512x_PAGE_0_BASE + 65) -#define PCM512x_GPIO_OUTPUT_1 (PCM512x_PAGE_0_BASE + 80) -#define PCM512x_GPIO_OUTPUT_2 (PCM512x_PAGE_0_BASE + 81) -#define PCM512x_GPIO_OUTPUT_3 (PCM512x_PAGE_0_BASE + 82) -#define PCM512x_GPIO_OUTPUT_4 (PCM512x_PAGE_0_BASE + 83) -#define PCM512x_GPIO_OUTPUT_5 (PCM512x_PAGE_0_BASE + 84) -#define PCM512x_GPIO_OUTPUT_6 (PCM512x_PAGE_0_BASE + 85) -#define PCM512x_GPIO_CONTROL_1 (PCM512x_PAGE_0_BASE + 86) -#define PCM512x_GPIO_CONTROL_2 (PCM512x_PAGE_0_BASE + 87) -#define PCM512x_OVERFLOW (PCM512x_PAGE_0_BASE + 90) -#define PCM512x_RATE_DET_1 (PCM512x_PAGE_0_BASE + 91) -#define PCM512x_RATE_DET_2 (PCM512x_PAGE_0_BASE + 92) -#define PCM512x_RATE_DET_3 (PCM512x_PAGE_0_BASE + 93) -#define PCM512x_RATE_DET_4 (PCM512x_PAGE_0_BASE + 94) -#define PCM512x_ANALOG_MUTE_DET (PCM512x_PAGE_0_BASE + 108) -#define PCM512x_GPIN (PCM512x_PAGE_0_BASE + 119) -#define PCM512x_DIGITAL_MUTE_DET (PCM512x_PAGE_0_BASE + 120) - -#define PCM512x_MAX_REGISTER (PCM512x_PAGE_0_BASE + 120) +#define PCM512x_RESET (PCM512x_PAGE_BASE(0) + 1) +#define PCM512x_POWER (PCM512x_PAGE_BASE(0) + 2) +#define PCM512x_MUTE (PCM512x_PAGE_BASE(0) + 3) +#define PCM512x_PLL_EN (PCM512x_PAGE_BASE(0) + 4) +#define PCM512x_SPI_MISO_FUNCTION (PCM512x_PAGE_BASE(0) + 6) +#define PCM512x_DSP (PCM512x_PAGE_BASE(0) + 7) +#define PCM512x_GPIO_EN (PCM512x_PAGE_BASE(0) + 8) +#define PCM512x_BCLK_LRCLK_CFG (PCM512x_PAGE_BASE(0) + 9) +#define PCM512x_DSP_GPIO_INPUT (PCM512x_PAGE_BASE(0) + 10) +#define PCM512x_MASTER_MODE (PCM512x_PAGE_BASE(0) + 12) +#define PCM512x_PLL_REF (PCM512x_PAGE_BASE(0) + 13) +#define PCM512x_PLL_COEFF_0 (PCM512x_PAGE_BASE(0) + 20) +#define PCM512x_PLL_COEFF_1 (PCM512x_PAGE_BASE(0) + 21) +#define PCM512x_PLL_COEFF_2 (PCM512x_PAGE_BASE(0) + 22) +#define PCM512x_PLL_COEFF_3 (PCM512x_PAGE_BASE(0) + 23) +#define PCM512x_PLL_COEFF_4 (PCM512x_PAGE_BASE(0) + 24) +#define PCM512x_DSP_CLKDIV (PCM512x_PAGE_BASE(0) + 27) +#define PCM512x_DAC_CLKDIV (PCM512x_PAGE_BASE(0) + 28) +#define PCM512x_NCP_CLKDIV (PCM512x_PAGE_BASE(0) + 29) +#define PCM512x_OSR_CLKDIV (PCM512x_PAGE_BASE(0) + 30) +#define PCM512x_MASTER_CLKDIV_1 (PCM512x_PAGE_BASE(0) + 32) +#define PCM512x_MASTER_CLKDIV_2 (PCM512x_PAGE_BASE(0) + 33) +#define PCM512x_FS_SPEED_MODE (PCM512x_PAGE_BASE(0) + 34) +#define PCM512x_IDAC_1 (PCM512x_PAGE_BASE(0) + 35) +#define PCM512x_IDAC_2 (PCM512x_PAGE_BASE(0) + 36) +#define PCM512x_ERROR_DETECT (PCM512x_PAGE_BASE(0) + 37) +#define PCM512x_I2S_1 (PCM512x_PAGE_BASE(0) + 40) +#define PCM512x_I2S_2 (PCM512x_PAGE_BASE(0) + 41) +#define PCM512x_DAC_ROUTING (PCM512x_PAGE_BASE(0) + 42) +#define PCM512x_DSP_PROGRAM (PCM512x_PAGE_BASE(0) + 43) +#define PCM512x_CLKDET (PCM512x_PAGE_BASE(0) + 44) +#define PCM512x_AUTO_MUTE (PCM512x_PAGE_BASE(0) + 59) +#define PCM512x_DIGITAL_VOLUME_1 (PCM512x_PAGE_BASE(0) + 60) +#define PCM512x_DIGITAL_VOLUME_2 (PCM512x_PAGE_BASE(0) + 61) +#define PCM512x_DIGITAL_VOLUME_3 (PCM512x_PAGE_BASE(0) + 62) +#define PCM512x_DIGITAL_MUTE_1 (PCM512x_PAGE_BASE(0) + 63) +#define PCM512x_DIGITAL_MUTE_2 (PCM512x_PAGE_BASE(0) + 64) +#define PCM512x_DIGITAL_MUTE_3 (PCM512x_PAGE_BASE(0) + 65) +#define PCM512x_GPIO_OUTPUT_1 (PCM512x_PAGE_BASE(0) + 80) +#define PCM512x_GPIO_OUTPUT_2 (PCM512x_PAGE_BASE(0) + 81) +#define PCM512x_GPIO_OUTPUT_3 (PCM512x_PAGE_BASE(0) + 82) +#define PCM512x_GPIO_OUTPUT_4 (PCM512x_PAGE_BASE(0) + 83) +#define PCM512x_GPIO_OUTPUT_5 (PCM512x_PAGE_BASE(0) + 84) +#define PCM512x_GPIO_OUTPUT_6 (PCM512x_PAGE_BASE(0) + 85) +#define PCM512x_GPIO_CONTROL_1 (PCM512x_PAGE_BASE(0) + 86) +#define PCM512x_GPIO_CONTROL_2 (PCM512x_PAGE_BASE(0) + 87) +#define PCM512x_OVERFLOW (PCM512x_PAGE_BASE(0) + 90) +#define PCM512x_RATE_DET_1 (PCM512x_PAGE_BASE(0) + 91) +#define PCM512x_RATE_DET_2 (PCM512x_PAGE_BASE(0) + 92) +#define PCM512x_RATE_DET_3 (PCM512x_PAGE_BASE(0) + 93) +#define PCM512x_RATE_DET_4 (PCM512x_PAGE_BASE(0) + 94) +#define PCM512x_ANALOG_MUTE_DET (PCM512x_PAGE_BASE(0) + 108) +#define PCM512x_GPIN (PCM512x_PAGE_BASE(0) + 119) +#define PCM512x_DIGITAL_MUTE_DET (PCM512x_PAGE_BASE(0) + 120) + +#define PCM512x_OUTPUT_AMPLITUDE (PCM512x_PAGE_BASE(1) + 1) +#define PCM512x_ANALOG_GAIN_CTRL (PCM512x_PAGE_BASE(1) + 2) +#define PCM512x_UNDERVOLTAGE_PROT (PCM512x_PAGE_BASE(1) + 5) +#define PCM512x_ANALOG_MUTE_CTRL (PCM512x_PAGE_BASE(1) + 6) +#define PCM512x_ANALOG_GAIN_BOOST (PCM512x_PAGE_BASE(1) + 7) +#define PCM512x_VCOM_CTRL_1 (PCM512x_PAGE_BASE(1) + 8) +#define PCM512x_VCOM_CTRL_2 (PCM512x_PAGE_BASE(1) + 9) + +#define PCM512x_CRAM_CTRL (PCM512x_PAGE_BASE(44) + 1) + +#define PCM512x_MAX_REGISTER (PCM512x_PAGE_BASE(44) + 1) /* Page 0, Register 1 - reset */ #define PCM512x_RSTR (1 << 0) -- 1.9.rc1

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[alsa-devel] Sony Vaio Pro svp112a1cm
by Hendrik-Jan Heins 07 Feb '14

07 Feb '14

Dear list, to report the Spny Vaio Pro svp112a1cm THe sound device in this notebook works, but the internal microphone not. This seeems to be the same device as in teh 13 inch version of this laptop, but they seem to have changed the ID of the device. Maybe the same fix can be applied to fix? All details of the machine are available here: http://www.alsa-project.org/db/?f=970d897015298925b86d4d2c68a106fba22eb184 thank you. best regards, Hendrik-Jan

2 4

[alsa-devel] ASoC driver for multi-codec sound device
by Christopher Seifert 07 Feb '14

07 Feb '14

Hi all, we are currently working on an ASoC driver for a multi-codec sound device. The kernel we use is version 3.10. Each codec has it's own I2C address and are connected via I2S to a single CPU DAI. We'd like to use the codecs in TDM mode later on, so we can use them as a single multi channel ALSA device. Currently we can use one codec in stereo mode with playback and capturing working using a device tree. We were able to get a second codec working (using the snd_soc_aux_dev structure in our driver) in so far we can see and use the codec's controls via alsamixer, however the codec seems to ignore the commands or does not receive any at all. What we'd like to know: 1) How to we achieve proper initialisation of all codecs? More specifically, which function (ALSA side) or callback (codec side) is used to set up the codecs clocks (via I2C)? 2) As far as we know the driver we are building upon does not implement own PCM or DMA drivers. Is it necessary to do this on our own to get the codecs working in TDM mode? Also when working in I2S (stereo) mode on ALSA side, is it possible to tell ALSA to use more than 2 channels of the sound buffer? We currently have interleaved audio samples from more than one codec in our sound buffer (transferred by DMA) and we'd like to select a subset of these depending on how the sound device was opened (for example "arecord -c 2" vs. "arecord -c 4" and so on). Which are the audio sample order ALSA supports or expects, specifically in the case of more than 2 audio channels? (Interleaved "L0, L1, L2, ..., Ln, R0, R1, ..., Rn" vs. "L0, R0, L1, R1, ..., Ln, Rn") 3) If there is no custom DMA driver implemented how does the ALSA framework use the DMA controller? In which case do we need to write our own DMA driver? Are there possibilities to use the ALSA framework's internal DMA handling for our purposes? 4) How are PCM and DMA correlated in ALSA framework? Specifically, do the implementations depend on each other or are they independent, so we can for example implement a DMA driver and use ALSA's default PCM? We're stuck with this, so we appreciate any kind of help pointing us in the right direction. If there are any required informations missing please don't hesitate to ask. Thanks, Achim and Christopher

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[alsa-devel] [PATCH] ASoC: rsnd: use device dependency clock
by Kuninori Morimoto 07 Feb '14

07 Feb '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current R-Car sound driver is using device independent audio clock, but it is not good design for DT (= common clock) support. This patch adds device dependent clock support. But, there are some platform which is using old style clock. Old style clock is still supported at this point, but it will be removed soon. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/adg.c | 35 ++++++++++++++++++++++++++++++----- 1 file changed, 30 insertions(+), 5 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index a53235c..e82ab75 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -259,8 +259,9 @@ int rsnd_adg_probe(struct platform_device *pdev, { struct rsnd_adg *adg; struct device *dev = rsnd_priv_to_dev(priv); - struct clk *clk; + struct clk *clk, *clk_orig; int i; + bool use_old_style = false; adg = devm_kzalloc(dev, sizeof(*adg), GFP_KERNEL); if (!adg) { @@ -268,10 +269,34 @@ int rsnd_adg_probe(struct platform_device *pdev, return -ENOMEM; } - adg->clk[CLKA] = clk_get(NULL, "audio_clk_a"); - adg->clk[CLKB] = clk_get(NULL, "audio_clk_b"); - adg->clk[CLKC] = clk_get(NULL, "audio_clk_c"); - adg->clk[CLKI] = clk_get(NULL, "audio_clk_internal"); + clk_orig = clk_get(dev, NULL); + adg->clk[CLKA] = clk_get(dev, "clk_a"); + adg->clk[CLKB] = clk_get(dev, "clk_b"); + adg->clk[CLKC] = clk_get(dev, "clk_c"); + adg->clk[CLKI] = clk_get(dev, "clk_i"); + + for_each_rsnd_clk(clk, adg, i) { + if (clk_orig == clk) { + dev_warn(dev, + "Audio clock doesn't support new style," + "use old style\n"); + use_old_style = true; + break; + } + } + + /* + * note: + * these exist in order to keep compatible with old style, + * but will be removed soon + */ + if (use_old_style) { + adg->clk[CLKA] = clk_get(NULL, "audio_clk_a"); + adg->clk[CLKB] = clk_get(NULL, "audio_clk_b"); + adg->clk[CLKC] = clk_get(NULL, "audio_clk_c"); + adg->clk[CLKI] = clk_get(NULL, "audio_clk_internal"); + } + for_each_rsnd_clk(clk, adg, i) { if (IS_ERR(clk)) { dev_err(dev, "Audio clock failed\n"); -- 1.7.9.5

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[alsa-devel] [PATCH 1/3] sound: Include appropriate header file in pcsp/pcsp_input.c
by Rashika Kheria 07 Feb '14

07 Feb '14

Include appropriate header file in drivers/pcsp/pcsp_input.c because it defines function whose prototype definitions are present in drivers/pcsp/pcsp_input.h. This eliminates the following warning in drivers/pcsp/pcsp_input.c: sound/drivers/pcsp/pcsp_input.c:42:6: warning: no previous prototype for ‘pcspkr_stop_sound’ [-Wmissing-prototypes] sound/drivers/pcsp/pcsp_input.c:80:5: warning: no previous prototype for ‘pcspkr_input_init’ [-Wmissing-prototypes] sound/drivers/pcsp/pcsp_input.c:110:5: warning: no previous prototype for ‘pcspkr_input_remove’ [-Wmissing-prototypes] Signed-off-by: Rashika Kheria <rashika.kheria(a)gmail.com> --- sound/drivers/pcsp/pcsp_input.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/drivers/pcsp/pcsp_input.c b/sound/drivers/pcsp/pcsp_input.c index b874b0a..0ecf8a4 100644 --- a/sound/drivers/pcsp/pcsp_input.c +++ b/sound/drivers/pcsp/pcsp_input.c @@ -16,6 +16,7 @@ #include <linux/input.h> #include <asm/io.h> #include "pcsp.h" +#include "pcsp_input.h" static void pcspkr_do_sound(unsigned int count) { -- 1.7.9.5

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[alsa-devel] [PATCH 1/2] ASoC: codecs: Put the CODEC drivers in a menu
by Mark Brown 07 Feb '14

07 Feb '14

From: Mark Brown <broonie(a)linaro.org> Now they're visible they get a bit noisy. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/Kconfig | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 28faa2d0eaef..6b67e86f3f2c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -8,6 +8,8 @@ config SND_SOC_I2C_AND_SPI default y if I2C=y default y if SPI_MASTER=y +menu "CODEC drivers" + config SND_SOC_ALL_CODECS tristate "Build all ASoC CODEC drivers" depends on COMPILE_TEST @@ -588,3 +590,5 @@ config SND_SOC_ML26124 config SND_SOC_TPA6130A2 tristate "Texas Instruments TPA6130A2 headphone amplifier" depends on I2C + +endmenu -- 1.9.rc1

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[alsa-devel] [PATCH] ASoC: rt5640: Add ACPI ID for Intel Baytrail
by Jarkko Nikula 07 Feb '14

07 Feb '14

Realtek RT5640 uses ACPI ID "10EC5640" for Intel Baytrail platforms. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> Cc: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5640.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index a3fb41179636..886924934aa5 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2093,6 +2093,7 @@ MODULE_DEVICE_TABLE(i2c, rt5640_i2c_id); #ifdef CONFIG_ACPI static struct acpi_device_id rt5640_acpi_match[] = { { "INT33CA", 0 }, + { "10EC5640", 0 }, { }, }; MODULE_DEVICE_TABLE(acpi, rt5640_acpi_match); -- 1.8.5.3

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Re: [alsa-devel] [PATCH RFC 26/46] drivers/base: provide an infrastructure for componentised subsystems
by Daniel Vetter 07 Feb '14

07 Feb '14

I've chatted a bit with Hans Verkuil about this topic at fosdem and apparently both v4l and alsa have something like this already in their helper libraries. Adding more people as fyi in case they want to switch to the new driver core stuff from Russell. -Daniel On Thu, Jan 2, 2014 at 10:27 PM, Russell King <rmk+kernel(a)arm.linux.org.uk> wrote: > Subsystems such as ALSA, DRM and others require a single card-level > device structure to represent a subsystem. However, firmware tends to > describe the individual devices and the connections between them. > > Therefore, we need a way to gather up the individual component devices > together, and indicate when we have all the component devices. > > We do this in DT by providing a "superdevice" node which specifies > the components, eg: > > imx-drm { > compatible = "fsl,drm"; > crtcs = <&ipu1>; > connectors = <&hdmi>; > }; > > The superdevice is declared into the component support, along with the > subcomponents. The superdevice receives callbacks to locate the > subcomponents, and identify when all components are present. At this > point, we bind the superdevice, which causes the appropriate subsystem > to be initialised in the conventional way. > > When any of the components or superdevice are removed from the system, > we unbind the superdevice, thereby taking the subsystem down. > > Signed-off-by: Russell King <rmk+kernel(a)arm.linux.org.uk> > --- > drivers/base/Makefile | 2 +- > drivers/base/component.c | 379 +++++++++++++++++++++++++++++++++++++++++++++ > include/linux/component.h | 31 ++++ > 3 files changed, 411 insertions(+), 1 deletions(-) > create mode 100644 drivers/base/component.c > create mode 100644 include/linux/component.h > > diff --git a/drivers/base/Makefile b/drivers/base/Makefile > index 94e8a80e87f8..870ecfd503af 100644 > --- a/drivers/base/Makefile > +++ b/drivers/base/Makefile > @@ -1,6 +1,6 @@ > # Makefile for the Linux device tree > > -obj-y := core.o bus.o dd.o syscore.o \ > +obj-y := component.o core.o bus.o dd.o syscore.o \ > driver.o class.o platform.o \ > cpu.o firmware.o init.o map.o devres.o \ > attribute_container.o transport_class.o \ > diff --git a/drivers/base/component.c b/drivers/base/component.c > new file mode 100644 > index 000000000000..5492cd8d2247 > --- /dev/null > +++ b/drivers/base/component.c > @@ -0,0 +1,379 @@ > +/* > + * Componentized device handling. > + * > + * 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 is work in progress. We gather up the component devices into a list, > + * and bind them when instructed. At the moment, we're specific to the DRM > + * subsystem, and only handles one master device, but this doesn't have to be > + * the case. > + */ > +#include <linux/component.h> > +#include <linux/device.h> > +#include <linux/kref.h> > +#include <linux/list.h> > +#include <linux/module.h> > +#include <linux/mutex.h> > +#include <linux/slab.h> > + > +struct master { > + struct list_head node; > + struct list_head components; > + bool bound; > + > + const struct component_master_ops *ops; > + struct device *dev; > +}; > + > +struct component { > + struct list_head node; > + struct list_head master_node; > + struct master *master; > + bool bound; > + > + const struct component_ops *ops; > + struct device *dev; > +}; > + > +static DEFINE_MUTEX(component_mutex); > +static LIST_HEAD(component_list); > +static LIST_HEAD(masters); > + > +static struct master *__master_find(struct device *dev, const struct component_master_ops *ops) > +{ > + struct master *m; > + > + list_for_each_entry(m, &masters, node) > + if (m->dev == dev && (!ops || m->ops == ops)) > + return m; > + > + return NULL; > +} > + > +/* Attach an unattached component to a master. */ > +static void component_attach_master(struct master *master, struct component *c) > +{ > + c->master = master; > + > + list_add_tail(&c->master_node, &master->components); > +} > + > +/* Detach a component from a master. */ > +static void component_detach_master(struct master *master, struct component *c) > +{ > + list_del(&c->master_node); > + > + c->master = NULL; > +} > + > +int component_master_add_child(struct master *master, > + int (*compare)(struct device *, void *), void *compare_data) > +{ > + struct component *c; > + int ret = -ENXIO; > + > + list_for_each_entry(c, &component_list, node) { > + if (c->master) > + continue; > + > + if (compare(c->dev, compare_data)) { > + component_attach_master(master, c); > + ret = 0; > + break; > + } > + } > + > + return ret; > +} > +EXPORT_SYMBOL_GPL(component_master_add_child); > + > +/* Detach all attached components from this master */ > +static void master_remove_components(struct master *master) > +{ > + while (!list_empty(&master->components)) { > + struct component *c = list_first_entry(&master->components, > + struct component, master_node); > + > + WARN_ON(c->master != master); > + > + component_detach_master(master, c); > + } > +} > + > +/* > + * Try to bring up a master. If component is NULL, we're interested in > + * this master, otherwise it's a component which must be present to try > + * and bring up the master. > + * > + * Returns 1 for successful bringup, 0 if not ready, or -ve errno. > + */ > +static int try_to_bring_up_master(struct master *master, > + struct component *component) > +{ > + int ret = 0; > + > + if (!master->bound) { > + /* > + * Search the list of components, looking for components that > + * belong to this master, and attach them to the master. > + */ > + if (master->ops->add_components(master->dev, master)) { > + /* Failed to find all components */ > + master_remove_components(master); > + ret = 0; > + goto out; > + } > + > + if (component && component->master != master) { > + master_remove_components(master); > + ret = 0; > + goto out; > + } > + > + /* Found all components */ > + ret = master->ops->bind(master->dev); > + if (ret < 0) { > + master_remove_components(master); > + goto out; > + } > + > + master->bound = true; > + ret = 1; > + } > +out: > + > + return ret; > +} > + > +static int try_to_bring_up_masters(struct component *component) > +{ > + struct master *m; > + int ret = 0; > + > + list_for_each_entry(m, &masters, node) { > + ret = try_to_bring_up_master(m, component); > + if (ret != 0) > + break; > + } > + > + return ret; > +} > + > +static void take_down_master(struct master *master) > +{ > + if (master->bound) { > + master->ops->unbind(master->dev); > + master->bound = false; > + } > + > + master_remove_components(master); > +} > + > +int component_master_add(struct device *dev, const struct component_master_ops *ops) > +{ > + struct master *master; > + int ret; > + > + master = kzalloc(sizeof(*master), GFP_KERNEL); > + if (!master) > + return -ENOMEM; > + > + master->dev = dev; > + master->ops = ops; > + INIT_LIST_HEAD(&master->components); > + > + /* Add to the list of available masters. */ > + mutex_lock(&component_mutex); > + list_add(&master->node, &masters); > + > + ret = try_to_bring_up_master(master, NULL); > + > + if (ret < 0) { > + /* Delete off the list if we weren't successful */ > + list_del(&master->node); > + kfree(master); > + } > + mutex_unlock(&component_mutex); > + > + return ret < 0 ? ret : 0; > +} > +EXPORT_SYMBOL_GPL(component_master_add); > + > +void component_master_del(struct device *dev, const struct component_master_ops *ops) > +{ > + struct master *master; > + > + mutex_lock(&component_mutex); > + master = __master_find(dev, ops); > + if (master) { > + take_down_master(master); > + > + list_del(&master->node); > + kfree(master); > + } > + mutex_unlock(&component_mutex); > +} > +EXPORT_SYMBOL_GPL(component_master_del); > + > +static void component_unbind(struct component *component, > + struct master *master, void *data) > +{ > + WARN_ON(!component->bound); > + > + component->ops->unbind(component->dev, master->dev, data); > + component->bound = false; > + > + /* Release all resources claimed in the binding of this component */ > + devres_release_group(component->dev, component); > +} > + > +void component_unbind_all(struct device *master_dev, void *data) > +{ > + struct master *master; > + struct component *c; > + > + WARN_ON(!mutex_is_locked(&component_mutex)); > + > + master = __master_find(master_dev, NULL); > + if (!master) > + return; > + > + list_for_each_entry_reverse(c, &master->components, master_node) > + component_unbind(c, master, data); > +} > +EXPORT_SYMBOL_GPL(component_unbind_all); > + > +static int component_bind(struct component *component, struct master *master, > + void *data) > +{ > + int ret; > + > + /* > + * Each component initialises inside its own devres group. > + * This allows us to roll-back a failed component without > + * affecting anything else. > + */ > + if (!devres_open_group(master->dev, NULL, GFP_KERNEL)) > + return -ENOMEM; > + > + /* > + * Also open a group for the device itself: this allows us > + * to release the resources claimed against the sub-device > + * at the appropriate moment. > + */ > + if (!devres_open_group(component->dev, component, GFP_KERNEL)) { > + devres_release_group(master->dev, NULL); > + return -ENOMEM; > + } > + > + dev_dbg(master->dev, "binding %s (ops %ps)\n", > + dev_name(component->dev), component->ops); > + > + ret = component->ops->bind(component->dev, master->dev, data); > + if (!ret) { > + component->bound = true; > + > + /* > + * Close the component device's group so that resources > + * allocated in the binding are encapsulated for removal > + * at unbind. Remove the group on the DRM device as we > + * can clean those resources up independently. > + */ > + devres_close_group(component->dev, NULL); > + devres_remove_group(master->dev, NULL); > + > + dev_info(master->dev, "bound %s (ops %ps)\n", > + dev_name(component->dev), component->ops); > + } else { > + devres_release_group(component->dev, NULL); > + devres_release_group(master->dev, NULL); > + > + dev_err(master->dev, "failed to bind %s (ops %ps): %d\n", > + dev_name(component->dev), component->ops, ret); > + } > + > + return ret; > +} > + > +int component_bind_all(struct device *master_dev, void *data) > +{ > + struct master *master; > + struct component *c; > + int ret = 0; > + > + WARN_ON(!mutex_is_locked(&component_mutex)); > + > + master = __master_find(master_dev, NULL); > + if (!master) > + return -EINVAL; > + > + list_for_each_entry(c, &master->components, master_node) { > + ret = component_bind(c, master, data); > + if (ret) > + break; > + } > + > + if (ret != 0) { > + list_for_each_entry_continue_reverse(c, &master->components, > + master_node) > + component_unbind(c, master, data); > + } > + > + return ret; > +} > +EXPORT_SYMBOL_GPL(component_bind_all); > + > +int component_add(struct device *dev, const struct component_ops *ops) > +{ > + struct component *component; > + int ret; > + > + component = kzalloc(sizeof(*component), GFP_KERNEL); > + if (!component) > + return -ENOMEM; > + > + component->ops = ops; > + component->dev = dev; > + > + dev_dbg(dev, "adding component (ops %ps)\n", ops); > + > + mutex_lock(&component_mutex); > + list_add_tail(&component->node, &component_list); > + > + ret = try_to_bring_up_masters(component); > + if (ret < 0) { > + list_del(&component->node); > + > + kfree(component); > + } > + mutex_unlock(&component_mutex); > + > + return ret < 0 ? ret : 0; > +} > +EXPORT_SYMBOL_GPL(component_add); > + > +void component_del(struct device *dev, const struct component_ops *ops) > +{ > + struct component *c, *component = NULL; > + > + mutex_lock(&component_mutex); > + list_for_each_entry(c, &component_list, node) > + if (c->dev == dev && c->ops == ops) { > + list_del(&c->node); > + component = c; > + break; > + } > + > + if (component && component->master) > + take_down_master(component->master); > + > + mutex_unlock(&component_mutex); > + > + WARN_ON(!component); > + kfree(component); > +} > +EXPORT_SYMBOL_GPL(component_del); > + > +MODULE_LICENSE("GPL v2"); > diff --git a/include/linux/component.h b/include/linux/component.h > new file mode 100644 > index 000000000000..73657636db0b > --- /dev/null > +++ b/include/linux/component.h > @@ -0,0 +1,31 @@ > +#ifndef COMPONENT_H > +#define COMPONENT_H > + > +struct device; > + > +struct component_ops { > + int (*bind)(struct device *, struct device *, void *); > + void (*unbind)(struct device *, struct device *, void *); > +}; > + > +int component_add(struct device *, const struct component_ops *); > +void component_del(struct device *, const struct component_ops *); > + > +int component_bind_all(struct device *, void *); > +void component_unbind_all(struct device *, void *); > + > +struct master; > + > +struct component_master_ops { > + int (*add_components)(struct device *, struct master *); > + int (*bind)(struct device *); > + void (*unbind)(struct device *); > +}; > + > +int component_master_add(struct device *, const struct component_master_ops *); > +void component_master_del(struct device *, const struct component_master_ops *); > + > +int component_master_add_child(struct master *master, > + int (*compare)(struct device *, void *), void *compare_data); > + > +#endif > -- > 1.7.4.4 > > _______________________________________________ > dri-devel mailing list > dri-devel(a)lists.freedesktop.org > http://lists.freedesktop.org/mailman/listinfo/dri-devel -- Daniel Vetter Software Engineer, Intel Corporation +41 (0) 79 365 57 48 - http://blog.ffwll.ch

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[alsa-devel] [PATCH] ALSA: hda - Fix leftover ifdef checks after modularization
by Takashi Iwai 07 Feb '14

07 Feb '14

Since the commit [595fe1b702c3: ALSA: hda - Make CONFIG_SND_HDA_CODEC_* tristate], the kconfig variables for the generic parser and codec drivers can be "m" instead of boolean, but some codes are left unchanged to check only #ifdef CONFIG_SND_HDA_CODEC_XXX, which is no longer true for modules. This patch fixes them by replacing with IS_ENABLED() macros. Fixes: 595fe1b702c3 ('ALSA: hda - Make CONFIG_SND_HDA_CODEC_* tristate') Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=70161 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 6 +++--- sound/pci/hda/hda_intel.c | 2 +- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index ec4536c8d8d4..69acbf758bbb 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -932,7 +932,7 @@ int snd_hda_bus_new(struct snd_card *card, } EXPORT_SYMBOL_GPL(snd_hda_bus_new); -#ifdef CONFIG_SND_HDA_GENERIC +#if IS_ENABLED(CONFIG_SND_HDA_GENERIC) #define is_generic_config(codec) \ (codec->modelname && !strcmp(codec->modelname, "generic")) #else @@ -1570,7 +1570,7 @@ int snd_hda_codec_update_widgets(struct hda_codec *codec) EXPORT_SYMBOL_GPL(snd_hda_codec_update_widgets); -#ifdef CONFIG_SND_HDA_CODEC_HDMI +#if IS_ENABLED(CONFIG_SND_HDA_CODEC_HDMI) /* if all audio out widgets are digital, let's assume the codec as a HDMI/DP */ static bool is_likely_hdmi_codec(struct hda_codec *codec) { @@ -1622,7 +1622,7 @@ int snd_hda_codec_configure(struct hda_codec *codec) unload_parser(codec); /* to be sure */ if (is_likely_hdmi_codec(codec)) patch = load_parser(codec, snd_hda_parse_hdmi_codec); -#ifdef CONFIG_SND_HDA_GENERIC +#if IS_ENABLED(CONFIG_SND_HDA_GENERIC) if (!patch) patch = load_parser(codec, snd_hda_parse_generic_codec); #endif diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index fa2879a21a50..e354ab1ec20f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -198,7 +198,7 @@ MODULE_DESCRIPTION("Intel HDA driver"); #endif #if defined(CONFIG_PM) && defined(CONFIG_VGA_SWITCHEROO) -#ifdef CONFIG_SND_HDA_CODEC_HDMI +#if IS_ENABLED(CONFIG_SND_HDA_CODEC_HDMI) #define SUPPORT_VGA_SWITCHEROO #endif #endif -- 1.8.5.2

1 0

[alsa-devel] [PATCH] ALSA: hda - Add a headset quirk for Dell XPS 13
by David Henningsson 07 Feb '14

07 Feb '14

This quirk is needed for the headset microphone to work. Alsa-info at http://www.alsa-project.org/db/?f=8c7dfe857ceff462ca2de133e67023c0f68de9cb Cc: stable(a)vger.kernel.org (3.10+) Reported-by: Po-Hsu Lin <po-hsu.lin(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bee4b8c..2cb6a39 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5146,6 +5146,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x060a, "Dell XPS 13", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0623, "Dell", ALC668_FIXUP_AUTO_MUTE), SND_PCI_QUIRK(0x1028, 0x0624, "Dell", ALC668_FIXUP_AUTO_MUTE), SND_PCI_QUIRK(0x1028, 0x0625, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda - Rename ASUS subwoofer quirks
by David Henningsson 07 Feb '14

07 Feb '14

Just a small refactoring to make the next patch slightly simpler. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bee4b8c..0295efe 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4925,9 +4925,9 @@ enum { ALC662_FIXUP_INV_DMIC, ALC668_FIXUP_DELL_MIC_NO_PRESENCE, ALC668_FIXUP_HEADSET_MODE, - ALC662_FIXUP_BASS_CHMAP, + ALC662_FIXUP_BASS_MODE4_CHMAP, ALC662_FIXUP_BASS_1A, - ALC662_FIXUP_BASS_1A_CHMAP, + ALC662_FIXUP_BASS_CHMAP, ALC668_FIXUP_AUTO_MUTE, }; @@ -5115,7 +5115,7 @@ static const struct hda_fixup alc662_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_mode_alc668, }, - [ALC662_FIXUP_BASS_CHMAP] = { + [ALC662_FIXUP_BASS_MODE4_CHMAP] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_bass_chmap, .chained = true, @@ -5127,12 +5127,12 @@ static const struct hda_fixup alc662_fixups[] = { {0x1a, 0x80106111}, /* bass speaker */ {} }, + .chained = true, + .chain_id = ALC662_FIXUP_BASS_CHMAP, }, - [ALC662_FIXUP_BASS_1A_CHMAP] = { + [ALC662_FIXUP_BASS_CHMAP] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_bass_chmap, - .chained = true, - .chain_id = ALC662_FIXUP_BASS_1A, }, }; @@ -5153,9 +5153,9 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0628, "Dell", ALC668_FIXUP_AUTO_MUTE), SND_PCI_QUIRK(0x1028, 0x064e, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), - SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_BASS_1A_CHMAP), - SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_CHMAP), - SND_PCI_QUIRK(0x1043, 0x1bf3, "ASUS N76VZ", ALC662_FIXUP_BASS_CHMAP), + SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_BASS_1A), + SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_MODE4_CHMAP), + SND_PCI_QUIRK(0x1043, 0x1bf3, "ASUS N76VZ", ALC662_FIXUP_BASS_MODE4_CHMAP), SND_PCI_QUIRK(0x1043, 0x8469, "ASUS mobo", ALC662_FIXUP_NO_JACK_DETECT), SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_FIXUP_ASUS_MODE2), SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD), -- 1.7.9.5

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[alsa-devel] [PATCH v2] MIDI driver for Behringer BCD2000 USB device
by Mario Kicherer 06 Feb '14

06 Feb '14

This patch adds initial support for the Behringer BCD2000 USB DJ controller. At the moment, only the MIDI part of the device is working, i.e. knobs, buttons and LEDs. I also plan to add support for the audio part, but I assume that this will require more effort than the rather simple MIDI interface. Progress can be tracked at https://github.com/anyc/snd-usb-bcd2000. Changes since v1: - fixed the various code style issues, thanks to Daniel Mack and checkpatch.pl. Signed-off-by: Mario Kicherer <dev(a)kicherer.org> --- sound/usb/Kconfig | 14 ++ sound/usb/Makefile | 2 +- sound/usb/bcd2000/Makefile | 3 + sound/usb/bcd2000/bcd2000.c | 553 ++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 571 insertions(+), 1 deletion(-) create mode 100644 sound/usb/bcd2000/Makefile create mode 100644 sound/usb/bcd2000/bcd2000.c diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index de9408b..2abafea 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -146,5 +146,19 @@ config SND_USB_HIFACE To compile this driver as a module, choose M here: the module will be called snd-usb-hiface. +config SND_USB_BCD2000 + tristate "Behringer BCD2000 MIDI driver" + select SND_HWDEP + select SND_RAWMIDI + help + Say Y here to include MIDI support for the Behringer BCD2000 DJ + controller. + + Audio support is still work-in-progress at + https://github.com/anyc/snd-usb-bcd2000 + + To compile this driver as a module, choose M here: the module + will be called snd-usb-bcd2000. + endif # SND_USB diff --git a/sound/usb/Makefile b/sound/usb/Makefile index abe668f..2b92f0d 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -23,4 +23,4 @@ obj-$(CONFIG_SND_USB_UA101) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_USX2Y) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_US122L) += snd-usbmidi-lib.o -obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ hiface/ +obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ hiface/ bcd2000/ diff --git a/sound/usb/bcd2000/Makefile b/sound/usb/bcd2000/Makefile new file mode 100644 index 0000000..6540136 --- /dev/null +++ b/sound/usb/bcd2000/Makefile @@ -0,0 +1,3 @@ +snd-usb-bcd2000-y := bcd2000.o + +obj-$(CONFIG_SND_USB_BCD2000) += snd-usb-bcd2000.o \ No newline at end of file diff --git a/sound/usb/bcd2000/bcd2000.c b/sound/usb/bcd2000/bcd2000.c new file mode 100644 index 0000000..d8c5525 --- /dev/null +++ b/sound/usb/bcd2000/bcd2000.c @@ -0,0 +1,553 @@ +/* + * Behringer BCD2000 driver + * + * Copyright (C) 2014 Mario Kicherer (dev(a)kicherer.org) + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#include <linux/kernel.h> +#include <linux/errno.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/module.h> +#include <linux/usb.h> +#include <linux/usb/audio.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/rawmidi.h> +#include "../usbaudio.h" +#include "../midi.h" + +#define DEVICE_NAME "Behringer BCD2000" +#define DEVICE_SHORTNAME "bcd2000" + +#define PREFIX DEVICE_SHORTNAME ": " +#define BUFSIZE 64 + +static struct usb_device_id id_table[] = { + { USB_DEVICE(0x1397, 0x00bd) }, + { }, +}; + +static int debug; +module_param(debug, int, S_IRUGO | S_IWUSR); +MODULE_PARM_DESC(debug, "enable debug output (default: 0)"); + +static unsigned char device_cmd_prefix[] = {0x03, 0x00}; + +static unsigned char bcd2000_init_sequence[] = { + 0x07, 0x00, 0x00, 0x00, 0x78, 0x48, 0x1c, 0x81, + 0xc4, 0x00, 0x00, 0x00, 0x5e, 0x53, 0x4a, 0xf7, + 0x18, 0xfa, 0x11, 0xff, 0x6c, 0xf3, 0x90, 0xff, + 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, + 0x18, 0xfa, 0x11, 0xff, 0x14, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0xf2, 0x34, 0x4a, 0xf7, + 0x18, 0xfa, 0x11, 0xff}; + +struct bcd2000 { + struct usb_device *dev; + struct snd_card *card; + struct usb_interface *intf; + int card_index; + struct snd_pcm *pcm; + struct list_head midi_list; + spinlock_t lock; + struct mutex mutex; + + int midi_out_active; + struct snd_rawmidi *rmidi; + struct snd_rawmidi_substream *midi_receive_substream; + struct snd_rawmidi_substream *midi_out_substream; + + unsigned char midi_in_buf[BUFSIZE]; + unsigned char midi_out_buf[BUFSIZE]; + + unsigned char midi_cmd_buf[3]; + unsigned char midi_cmd_offset; + + struct urb *midi_out_urb; + struct urb *midi_in_urb; +}; + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; + +static DEFINE_MUTEX(devices_mutex); +static unsigned int devices_used; +static struct usb_driver bcd2000_driver; + + + + + + +static int bcd2000_midi_input_open(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +static int bcd2000_midi_input_close(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +static void bcd2000_midi_input_trigger(struct snd_rawmidi_substream *substream, + int up) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + + if (!bcd2k) + return; + + bcd2k->midi_receive_substream = up ? substream : NULL; +} + +static void bcd2000_midi_handle_input(struct bcd2000 *bcd2k, + const unsigned char *buf, int len) +{ + unsigned char length, cmd, buf_offset, tocopy; + + if (!bcd2k->midi_receive_substream) + return; + + if (debug) { + print_hex_dump(KERN_DEBUG, "received from device: ", + DUMP_PREFIX_NONE, 16, 1, buf, len, false); + } + + if (len < 2) + return; + + /* + * The BCD2000 sends MIDI commands with 2 or 3 bytes length. In case of + * 2 bytes, the command is the same as before. + * + * Packet structure: mm nn oo (pp) + * mm: payload length + * nn: MIDI command or note + * oo: note or velocity + * pp: velocity + */ + + length = buf[0]; + + /* ignore packets without payload */ + if (length == 0) + return; + + /* + * The MIDI packets the BCD2000 sends can be arbitrarily truncated or + * concatenated. Therefore, this loop accumulates the bytes from the + * input buffer until a full valid MIDI packet is in the MIDI command + * buffer "midi_cmd_buf". + */ + + buf_offset = 1; + while (buf_offset <= length) { + cmd = buf[buf_offset]; + + if (bcd2k->midi_cmd_offset == 0 && cmd != 0x90 && cmd != 0xb0) { + /* + * this is a 2-byte midi packet -> reuse command byte + * from last midi packet + */ + bcd2k->midi_cmd_offset = 1; + } + + /* determine the number of bytes we want to copy this time */ + tocopy = min(3 - bcd2k->midi_cmd_offset, + length - (buf_offset - 1)); + + /* safety check */ + if (bcd2k->midi_cmd_offset + tocopy < BUFSIZE && + buf_offset + tocopy < len) { + memcpy(&bcd2k->midi_cmd_buf[bcd2k->midi_cmd_offset], + &buf[buf_offset], tocopy); + } else { + snd_printk(KERN_ERR PREFIX "access violation in %s\n", + __func__); + } + + bcd2k->midi_cmd_offset += tocopy; + buf_offset += tocopy; + + /* is our MIDI packet complete? */ + if (bcd2k->midi_cmd_offset == 3) { + if (debug) { + print_hex_dump(KERN_DEBUG, + "sending to userspace: ", + DUMP_PREFIX_NONE, 16, 1, + bcd2k->midi_cmd_buf, + bcd2k->midi_cmd_offset, false); + } + + /* send MIDI packet */ + snd_rawmidi_receive(bcd2k->midi_receive_substream, + bcd2k->midi_cmd_buf, 3); + + bcd2k->midi_cmd_offset = 0; + } + } +} + +static int bcd2000_midi_output_open(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +static int bcd2000_midi_output_close(struct snd_rawmidi_substream *substream) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + if (bcd2k->midi_out_active) { + usb_free_urb(bcd2k->midi_out_urb); + bcd2k->midi_out_active = 0; + } + return 0; +} + +static void bcd2000_midi_send(struct bcd2000 *bcd2k, + struct snd_rawmidi_substream *substream) +{ + int len, ret; + + BUILD_BUG_ON(sizeof(device_cmd_prefix) >= BUFSIZE); + /* copy the "set LED" command bytes */ + memcpy(bcd2k->midi_out_buf, device_cmd_prefix, + sizeof(device_cmd_prefix)); + + /* + * get MIDI packet and leave space for command prefix + * and payload length + */ + len = snd_rawmidi_transmit(substream, + bcd2k->midi_out_buf + 3, BUFSIZE - 3); + + if (len <= 0) + return; + + /* set payload length */ + bcd2k->midi_out_buf[2] = len; + bcd2k->midi_out_urb->transfer_buffer_length = BUFSIZE; + + if (debug) { + print_hex_dump(KERN_DEBUG, "sending to device: ", + DUMP_PREFIX_NONE, 16, 1, + bcd2k->midi_out_buf, len+3, false); + } + + /* send packet to the BCD2000 */ + ret = usb_submit_urb(bcd2k->midi_out_urb, GFP_ATOMIC); + if (ret < 0) { + dev_err(&bcd2k->dev->dev, PREFIX + "%s (%p): usb_submit_urb() failed" + "ret=%d, len=%d\n", __func__, substream, ret, len); + } else { + bcd2k->midi_out_active = 1; + } +} + +static void bcd2000_midi_output_done(struct urb *urb) +{ + struct bcd2000 *bcd2k = urb->context; + + bcd2k->midi_out_active = 0; + if (urb->status != 0) { + dev_err(&urb->dev->dev, PREFIX "urb status: %d\n", urb->status); + return; + } + + if (!bcd2k->midi_out_substream) + return; + + bcd2000_midi_send(bcd2k, bcd2k->midi_out_substream); +} + +static void bcd2000_command_reply_dispatch(struct urb *urb) +{ + int ret; + struct device *dev = &urb->dev->dev; + struct bcd2000 *bcd2k = urb->context; + + if (urb->status || !bcd2k) { + dev_warn(dev, "received urb->status = %i\n", urb->status); + return; + } + + if (urb->actual_length > 0) + bcd2000_midi_handle_input(bcd2k, urb->transfer_buffer, + urb->actual_length); + + bcd2k->midi_in_urb->actual_length = 0; + ret = usb_submit_urb(bcd2k->midi_in_urb, GFP_ATOMIC); + if (ret < 0) + dev_err(dev, "unable to submit urb. OOM!?\n"); +} + +static void bcd2000_midi_output_trigger(struct snd_rawmidi_substream *substream, + int up) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + + if (up) { + bcd2k->midi_out_substream = substream; + if (!bcd2k->midi_out_active) + bcd2000_midi_send(bcd2k, substream); + } else { + bcd2k->midi_out_substream = NULL; + } +} + +static struct snd_rawmidi_ops bcd2000_midi_output = { + .open = bcd2000_midi_output_open, + .close = bcd2000_midi_output_close, + .trigger = bcd2000_midi_output_trigger, +}; + +static struct snd_rawmidi_ops bcd2000_midi_input = { + .open = bcd2000_midi_input_open, + .close = bcd2000_midi_input_close, + .trigger = bcd2000_midi_input_trigger, +}; + +static void bcd2000_init_device(struct bcd2000 *bcd2k) +{ + int ret; + + /* copy init sequence into buffer */ + memcpy(bcd2k->midi_out_buf, bcd2000_init_sequence, 52); + bcd2k->midi_out_urb->transfer_buffer_length = 52; + + /* submit sequence */ + ret = usb_submit_urb(bcd2k->midi_out_urb, GFP_ATOMIC); + if (ret < 0) { + dev_err(&bcd2k->dev->dev, PREFIX + "%s: usb_submit_urb() failed," + "ret=%d: ", __func__, ret); + } else { + bcd2k->midi_out_active = 1; + } + + /* send empty packet to enable button and controller events */ + bcd2k->midi_in_urb->actual_length = 0; + ret = usb_submit_urb(bcd2k->midi_in_urb, GFP_ATOMIC); +} + +static void bcd2000_init_midi(struct bcd2000 *bcd2k) +{ + int ret; + struct snd_rawmidi *rmidi; + + ret = snd_rawmidi_new(bcd2k->card, bcd2k->card->shortname, 0, + 1, /* output */ + 1, /* input */ + &rmidi); + + if (ret < 0) + return; + + strlcpy(rmidi->name, bcd2k->card->shortname, sizeof(rmidi->name)); + + rmidi->info_flags = SNDRV_RAWMIDI_INFO_DUPLEX; + rmidi->private_data = bcd2k; + + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT; + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, + &bcd2000_midi_output); + + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT; + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, + &bcd2000_midi_input); + + bcd2k->rmidi = rmidi; + + bcd2k->midi_in_urb = usb_alloc_urb(0, GFP_KERNEL); + bcd2k->midi_out_urb = usb_alloc_urb(0, GFP_KERNEL); + + usb_fill_int_urb(bcd2k->midi_in_urb, bcd2k->dev, + usb_rcvbulkpipe(bcd2k->dev, 0x81), + bcd2k->midi_in_buf, BUFSIZE, + bcd2000_command_reply_dispatch, bcd2k, 1); + + usb_fill_int_urb(bcd2k->midi_out_urb, bcd2k->dev, + usb_sndbulkpipe(bcd2k->dev, 0x1), + bcd2k->midi_out_buf, BUFSIZE, + bcd2000_midi_output_done, bcd2k, 1); + + bcd2000_init_device(bcd2k); +} + +static void bcd2000_close_midi(struct bcd2000 *bcd2k) +{ + usb_free_urb(bcd2k->midi_out_urb); + usb_free_urb(bcd2k->midi_in_urb); +} + + + + + +/* + * general part + */ + +static void bcd2000_free_usb_related_resources(struct bcd2000 *bcd2k, + struct usb_interface *interface) +{ + struct usb_interface *intf; + + mutex_lock(&bcd2k->mutex); + intf = bcd2k->intf; + bcd2k->intf = NULL; + mutex_unlock(&bcd2k->mutex); + if (intf) { + usb_set_intfdata(intf, NULL); + if (intf != interface) + usb_driver_release_interface(&bcd2000_driver, + intf); + } +} + +static void bcd2000_card_free(struct snd_card *card) +{ + /* empty for now */ +} + +static int bcd2000_probe(struct usb_interface *interface, + const struct usb_device_id *usb_id) +{ + struct snd_card *card; + struct bcd2000 *bcd2k; + unsigned int card_index; + char usb_path[32]; + int err; + + + mutex_lock(&devices_mutex); + + for (card_index = 0; card_index < SNDRV_CARDS; ++card_index) + if (enable[card_index] && !(devices_used & (1 << card_index))) + break; + if (card_index >= SNDRV_CARDS) { + mutex_unlock(&devices_mutex); + return -ENOENT; + } + err = snd_card_create(index[card_index], id[card_index], THIS_MODULE, + sizeof(*bcd2k), &card); + if (err < 0) { + mutex_unlock(&devices_mutex); + return err; + } + card->private_free = bcd2000_card_free; + bcd2k = card->private_data; + bcd2k->dev = interface_to_usbdev(interface); + bcd2k->card = card; + bcd2k->card_index = card_index; + INIT_LIST_HEAD(&bcd2k->midi_list); + spin_lock_init(&bcd2k->lock); + mutex_init(&bcd2k->mutex); + + bcd2k->intf = interface; + + snd_card_set_dev(card, &interface->dev); + + strcpy(card->driver, DEVICE_NAME); + strcpy(card->shortname, DEVICE_SHORTNAME); + usb_make_path(bcd2k->dev, usb_path, sizeof(usb_path)); + snprintf(bcd2k->card->longname, sizeof(bcd2k->card->longname), + DEVICE_NAME ", at %s, %s speed", + usb_path, + bcd2k->dev->speed == USB_SPEED_HIGH ? "high" : "full"); + + dev_info(&bcd2k->dev->dev, PREFIX "%s", bcd2k->card->longname); + + + + bcd2000_init_midi(bcd2k); + + + err = snd_card_register(card); + if (err < 0) + goto probe_error; + + usb_set_intfdata(interface, bcd2k); + devices_used |= 1 << card_index; + + mutex_unlock(&devices_mutex); + return 0; + +probe_error: + bcd2000_free_usb_related_resources(bcd2k, interface); + snd_card_free(card); + mutex_unlock(&devices_mutex); + return err; +} + +static void bcd2000_disconnect(struct usb_interface *interface) +{ + struct bcd2000 *bcd2k = usb_get_intfdata(interface); + struct list_head *midi; + + if (!bcd2k) + return; + + bcd2000_close_midi(bcd2k); + + mutex_lock(&devices_mutex); + + /* make sure that userspace cannot create new requests */ + snd_card_disconnect(bcd2k->card); + + /* make sure that there are no pending USB requests */ + list_for_each(midi, &bcd2k->midi_list) + snd_usbmidi_disconnect(midi); + + bcd2000_free_usb_related_resources(bcd2k, interface); + + devices_used &= ~(1 << bcd2k->card_index); + + snd_card_free_when_closed(bcd2k->card); + + mutex_unlock(&devices_mutex); +} + +static struct usb_driver bcd2000_driver = { + .name = DEVICE_SHORTNAME, + .probe = bcd2000_probe, + .disconnect = bcd2000_disconnect, + .id_table = id_table, +}; + +static int __init bcd2000_init(void) +{ + int retval = 0; + + retval = usb_register(&bcd2000_driver); + if (retval) + snd_printk(KERN_INFO "usb_register failed. Error: %d", retval); + return retval; +} + +static void __exit bcd2000_exit(void) +{ + usb_deregister(&bcd2000_driver); +} + +module_init(bcd2000_init); +module_exit(bcd2000_exit); + +MODULE_DEVICE_TABLE(usb, id_table); +MODULE_AUTHOR("Mario Kicherer, dev(a)kicherer.org"); +MODULE_DESCRIPTION("Behringer BCD2000 driver"); +MODULE_LICENSE("GPL"); -- 1.8.3.2

4 7

[alsa-devel] [PATCH] ASoC: pcm512x: More constification
by Mark Brown 06 Feb '14

06 Feb '14

From: Mark Brown <broonie(a)linaro.org> Since the core now takes const strings for enums we should be constifying them (and the regulator supplies while we're at it). Reported-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/pcm512x.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 5ad3e9aa3cb4..1150381fc373 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -30,7 +30,7 @@ #include "pcm512x.h" #define PCM512x_NUM_SUPPLIES 3 -static const char *pcm512x_supply_names[PCM512x_NUM_SUPPLIES] = { +static const char * const pcm512x_supply_names[PCM512x_NUM_SUPPLIES] = { "AVDD", "DVDD", "CPVDD", @@ -167,7 +167,7 @@ static bool pcm512x_volatile(struct device *dev, unsigned int reg) static const DECLARE_TLV_DB_SCALE(digital_tlv, -10350, 50, 1); -static const char *pcm512x_dsp_program_texts[] = { +static const char * const pcm512x_dsp_program_texts[] = { "FIR interpolation with de-emphasis", "Low latency IIR with de-emphasis", "Fixed process flow", @@ -188,14 +188,14 @@ static const SOC_VALUE_ENUM_SINGLE_DECL(pcm512x_dsp_program, pcm512x_dsp_program_texts, pcm512x_dsp_program_values); -static const char *pcm512x_clk_missing_text[] = { +static const char * const pcm512x_clk_missing_text[] = { "1s", "2s", "3s", "4s", "5s", "6s", "7s", "8s" }; static const struct soc_enum pcm512x_clk_missing = SOC_ENUM_SINGLE(PCM512x_CLKDET, 0, 8, pcm512x_clk_missing_text); -static const char *pcm512x_autom_text[] = { +static const char * const pcm512x_autom_text[] = { "21ms", "106ms", "213ms", "533ms", "1.07s", "2.13s", "5.33s", "10.66s" }; @@ -207,7 +207,7 @@ static const struct soc_enum pcm512x_autom_r = SOC_ENUM_SINGLE(PCM512x_AUTO_MUTE, PCM512x_ATMR_SHIFT, 8, pcm512x_autom_text); -static const char *pcm512x_ramp_rate_text[] = { +static const char * const pcm512x_ramp_rate_text[] = { "1 sample/update", "2 samples/update", "4 samples/update", "Immediate" }; @@ -224,7 +224,7 @@ static const struct soc_enum pcm512x_vedf = SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_2, PCM512x_VEDF_SHIFT, 4, pcm512x_ramp_rate_text); -static const char *pcm512x_ramp_step_text[] = { +static const char * const pcm512x_ramp_step_text[] = { "4dB/step", "2dB/step", "1dB/step", "0.5dB/step" }; -- 1.9.rc1

1 0

[alsa-devel] [PATCH 0/8] da9055: Driver initialisation fixes, add DT support
by Adam Thomson 06 Feb '14

06 Feb '14

This patch series provides the following updates for DA9055 drivers: - Fixes an issue with da9055 driver initialisation (conflicting device ids) of PMIC (MFD) and CODEC drivers. - Add initial DT support for DA9055 related drivers, including binding documentation. - Remove conflicting use of platform_get_irq_byname() in driver probes. - Remove unnecessary resource structures for MFD cells as a result of removing use of platform_get_irq_byname(). Adam Thomson (8): ASoC: da9055: Fix device registration of PMIC and CODEC devices ASoC: da9055: Add DT support for CODEC mfd: da9055: Add DT support for PMIC regulator: da9055: Add DT support onkey: da9055: Remove use of platform_get_irq_byname() hwmon: da9055: Remove use of platform_get_irq_byname() rtc: da9055: Remove use of platform_get_irq_byname() mfd: da9055: Remove unused resource structures for mfd cells. Documentation/devicetree/bindings/mfd/da9055.txt | 73 ++++++++++++++++++++ Documentation/devicetree/bindings/sound/da9055.txt | 22 ++++++ drivers/hwmon/da9055-hwmon.c | 7 +-- drivers/input/misc/da9055_onkey.c | 9 +-- drivers/mfd/da9055-core.c | 46 ------------ drivers/mfd/da9055-i2c.c | 20 +++++- drivers/regulator/da9055-regulator.c | 59 ++++++++++++++-- drivers/rtc/rtc-da9055.c | 3 +- sound/soc/codecs/da9055.c | 19 +++++- 9 files changed, 188 insertions(+), 70 deletions(-) create mode 100644 Documentation/devicetree/bindings/mfd/da9055.txt create mode 100644 Documentation/devicetree/bindings/sound/da9055.txt

4 18

[alsa-devel] [PATCH] ALSA: hda - Fix inconsistent Mic mute LED
by Takashi Iwai 06 Feb '14

06 Feb '14

The current code for controlling mic mute LED in patch_sigmatel.c blindly assumes that there is a single capture switch. But, there can be multiple multiple ones, and each of them flips the state, ended up in an inconsistent state. For fixing this problem, this patch adds kcontrol to be passed to the hook function so that the callee can check which switch is being accessed. In stac_capture_led_hook(), the state is checked as a bitmask, and turns on the LED when all capture switches are off. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 8 ++++---- sound/pci/hda/hda_generic.h | 1 + sound/pci/hda/patch_conexant.c | 3 ++- sound/pci/hda/patch_realtek.c | 9 ++++++--- sound/pci/hda/patch_sigmatel.c | 27 +++++++++++++++++---------- sound/pci/hda/thinkpad_helper.c | 1 + 6 files changed, 31 insertions(+), 18 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 8321a97d5c05..d9a09bdd09db 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3269,7 +3269,7 @@ static int cap_put_caller(struct snd_kcontrol *kcontrol, mutex_unlock(&codec->control_mutex); snd_hda_codec_flush_cache(codec); /* flush the updates */ if (err >= 0 && spec->cap_sync_hook) - spec->cap_sync_hook(codec, ucontrol); + spec->cap_sync_hook(codec, kcontrol, ucontrol); return err; } @@ -3390,7 +3390,7 @@ static int cap_single_sw_put(struct snd_kcontrol *kcontrol, return ret; if (spec->cap_sync_hook) - spec->cap_sync_hook(codec, ucontrol); + spec->cap_sync_hook(codec, kcontrol, ucontrol); return ret; } @@ -3795,7 +3795,7 @@ static int mux_select(struct hda_codec *codec, unsigned int adc_idx, return 0; snd_hda_activate_path(codec, path, true, false); if (spec->cap_sync_hook) - spec->cap_sync_hook(codec, NULL); + spec->cap_sync_hook(codec, NULL, NULL); path_power_down_sync(codec, old_path); return 1; } @@ -5270,7 +5270,7 @@ static void init_input_src(struct hda_codec *codec) } if (spec->cap_sync_hook) - spec->cap_sync_hook(codec, NULL); + spec->cap_sync_hook(codec, NULL, NULL); } /* set right pin controls for digital I/O */ diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 07f767231c9f..c908afbe4d94 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -274,6 +274,7 @@ struct hda_gen_spec { void (*init_hook)(struct hda_codec *codec); void (*automute_hook)(struct hda_codec *codec); void (*cap_sync_hook)(struct hda_codec *codec, + struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); /* PCM hooks */ diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 4e0ec146553d..bcf91bea3317 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3291,7 +3291,8 @@ static void cxt_update_headset_mode(struct hda_codec *codec) } static void cxt_update_headset_mode_hook(struct hda_codec *codec, - struct snd_ctl_elem_value *ucontrol) + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { cxt_update_headset_mode(codec); } diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 56a8f1876603..6aabaa1c19f5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -708,7 +708,8 @@ static void alc_inv_dmic_sync(struct hda_codec *codec, bool force) } static void alc_inv_dmic_hook(struct hda_codec *codec, - struct snd_ctl_elem_value *ucontrol) + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { alc_inv_dmic_sync(codec, false); } @@ -3211,7 +3212,8 @@ static void alc269_fixup_hp_gpio_mute_hook(void *private_data, int enabled) /* turn on/off mic-mute LED per capture hook */ static void alc269_fixup_hp_gpio_mic_mute_hook(struct hda_codec *codec, - struct snd_ctl_elem_value *ucontrol) + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct alc_spec *spec = codec->spec; unsigned int oldval = spec->gpio_led; @@ -3521,7 +3523,8 @@ static void alc_update_headset_mode(struct hda_codec *codec) } static void alc_update_headset_mode_hook(struct hda_codec *codec, - struct snd_ctl_elem_value *ucontrol) + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { alc_update_headset_mode(codec); } diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 6998cf29b9bc..7311badf6a94 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -194,7 +194,7 @@ struct sigmatel_spec { int default_polarity; unsigned int mic_mute_led_gpio; /* capture mute LED GPIO */ - bool mic_mute_led_on; /* current mic mute state */ + unsigned int mic_enabled; /* current mic mute state (bitmask) */ /* stream */ unsigned int stream_delay; @@ -324,19 +324,26 @@ static void stac_gpio_set(struct hda_codec *codec, unsigned int mask, /* hook for controlling mic-mute LED GPIO */ static void stac_capture_led_hook(struct hda_codec *codec, - struct snd_ctl_elem_value *ucontrol) + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct sigmatel_spec *spec = codec->spec; - bool mute; + unsigned int mask; + bool cur_mute, prev_mute; - if (!ucontrol) + if (!kcontrol || !ucontrol) return; - mute = !(ucontrol->value.integer.value[0] || - ucontrol->value.integer.value[1]); - if (spec->mic_mute_led_on != mute) { - spec->mic_mute_led_on = mute; - if (mute) + mask = 1U << snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + prev_mute = !spec->mic_enabled; + if (ucontrol->value.integer.value[0] || + ucontrol->value.integer.value[1]) + spec->mic_enabled |= mask; + else + spec->mic_enabled &= ~mask; + cur_mute = !spec->mic_enabled; + if (cur_mute != prev_mute) { + if (cur_mute) spec->gpio_data |= spec->mic_mute_led_gpio; else spec->gpio_data &= ~spec->mic_mute_led_gpio; @@ -4462,7 +4469,7 @@ static void stac_setup_gpio(struct hda_codec *codec) if (spec->mic_mute_led_gpio) { spec->gpio_mask |= spec->mic_mute_led_gpio; spec->gpio_dir |= spec->mic_mute_led_gpio; - spec->mic_mute_led_on = true; + spec->mic_enabled = 0; spec->gpio_data |= spec->mic_mute_led_gpio; spec->gen.cap_sync_hook = stac_capture_led_hook; diff --git a/sound/pci/hda/thinkpad_helper.c b/sound/pci/hda/thinkpad_helper.c index 5799fbc24c28..8fe3b8c18ed4 100644 --- a/sound/pci/hda/thinkpad_helper.c +++ b/sound/pci/hda/thinkpad_helper.c @@ -39,6 +39,7 @@ static void update_tpacpi_mute_led(void *private_data, int enabled) } static void update_tpacpi_micmute_led(struct hda_codec *codec, + struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { if (!ucontrol || !led_set_func) -- 1.8.5.2

2 6

[alsa-devel] [alsa-plugins] [PATCH 2/2] Allow number of channels
by Corfu, ValentinX 06 Feb '14

06 Feb '14

>From 92cf8488d3d2e271fdb4b1fb41fbc006cc59677b Mon Sep 17 00:00:00 2001 From: Valentin Corfu <valentinx.corfu(a)intel.com> Date: Fri, 31 Jan 2014 08:08:51 -0500 Subject: [PATCH 2/2] Allow number of channels The modification allows to specify the number of channels as parameters in the configuration file The given number of channels are then created but not connected. Signed-off-by: Valentin Corfu <valentinx.corfu(a)intel.com> diff --git a/jack/pcm_jack.c b/jack/pcm_jack.c index 475c619..feef4a0 100644 --- a/jack/pcm_jack.c +++ b/jack/pcm_jack.c @@ -325,7 +325,7 @@ static int make_nonblock(int fd) static int snd_pcm_jack_open(snd_pcm_t **pcmp, const char *name, snd_config_t *playback_conf, snd_config_t *capture_conf, - snd_pcm_stream_t stream, int mode) + snd_pcm_stream_t stream, int mode, long num_channels) { snd_pcm_jack_t *jack; int err; @@ -341,11 +341,19 @@ static int snd_pcm_jack_open(snd_pcm_t **pcmp, const char *name, jack->fd = -1; jack->io.poll_fd = -1; - err = parse_ports(jack, stream == SND_PCM_STREAM_PLAYBACK ? - playback_conf : capture_conf); - if (err) { - snd_pcm_jack_free(jack); - return err; + if (num_channels > 0) { + jack->num_ports = num_channels; + jack->port_names = calloc(num_channels, sizeof(char*)); + if (jack->port_names == NULL) + return -ENOMEM; + } + else { + err = parse_ports(jack, stream == SND_PCM_STREAM_PLAYBACK ? + playback_conf : capture_conf); + if (err) { + snd_pcm_jack_free(jack); + return err; + } } jack->channels = jack->num_ports; @@ -422,6 +430,7 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) snd_config_t *playback_conf = NULL; snd_config_t *capture_conf = NULL; int err; + long num_channels = 0; snd_config_for_each(i, next, conf) { snd_config_t *n = snd_config_iterator_entry(i); @@ -434,6 +443,10 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) snd_config_get_string(n, &name); continue; } + if (strcmp(id, "nchan") == 0) { + snd_config_get_integer(n, &num_channels); + continue; + } if (strcmp(id, "playback_ports") == 0) { if (snd_config_get_type(n) != SND_CONFIG_TYPE_COMPOUND) { SNDERR("Invalid type for %s", id); @@ -454,7 +467,7 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) return -EINVAL; } - err = snd_pcm_jack_open(pcmp, name, playback_conf, capture_conf, stream, mode); + err = snd_pcm_jack_open(pcmp, name, playback_conf, capture_conf, stream, mode, num_channels); return err; } -- 1.8.1.4 -------------------------------------------------------------- Intel Shannon Limited Registered in Ireland Registered Office: Collinstown Industrial Park, Leixlip, County Kildare Registered Number: 308263 Business address: Dromore House, East Park, Shannon, Co. Clare This e-mail and any attachments may contain confidential material for the sole use of the intended recipient(s). Any review or distribution by others is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies.

2 1

[alsa-devel] [alsa-plugins] [PATCH 1/2] Allow to specify a client name as parameters
by Corfu, ValentinX 06 Feb '14

06 Feb '14

>From 1ac8268330df7564edf252a542a27f3cf4f4472f Mon Sep 17 00:00:00 2001 From: Valentin Corfu <valentinx.corfu(a)intel.com> Date: Fri, 31 Jan 2014 08:06:23 -0500 Subject: [PATCH 1/2] Allow to specify a client name as parameters The current jack client name contains the process id of the application providing the audio samples. This leads to unpredictable jack client names which makes handling of the connections by a controlling application very hard. This modification now, allows to specify a client name as parameters in the configuration file. The implementation is backward compatible and simply adds a new configuration option to the plugin. Signed-off-by: Valentin Corfu <valentinx.corfu(a)intel.com> diff --git a/jack/pcm_jack.c b/jack/pcm_jack.c index c51fe84..475c619 100644 --- a/jack/pcm_jack.c +++ b/jack/pcm_jack.c @@ -355,12 +355,19 @@ static int snd_pcm_jack_open(snd_pcm_t **pcmp, const char *name, snd_pcm_jack_free(jack); return -EINVAL; } - - if (snprintf(jack_client_name, sizeof(jack_client_name), "alsa-jack.%s%s.%d.%d", name, - stream == SND_PCM_STREAM_PLAYBACK ? "P" : "C", getpid(), num++) - >= (int)sizeof(jack_client_name)) { - fprintf(stderr, "%s: WARNING: JACK client name '%s' truncated to %d characters, might not be unique\n", - __func__, jack_client_name, (int)strlen(jack_client_name)); + if ( name == NULL ) { + if (snprintf(jack_client_name, sizeof(jack_client_name), "alsa-jack.%s%s.%d.%d", name, + stream == SND_PCM_STREAM_PLAYBACK ? "P" : "C", getpid(), num++) + >= (int)sizeof(jack_client_name)) { + fprintf(stderr, "%s: WARNING: JACK client name '%s' truncated to %d characters, might not be unique\n", + __func__, jack_client_name, (int)strlen(jack_client_name)); + } + } else { + if (snprintf(jack_client_name, sizeof(jack_client_name), "%s", name) + >= (int)sizeof(jack_client_name)) { + fprintf(stderr, "%s: WARNING: JACK client name '%s' truncated to %d characters, might not be unique\n", + __func__, jack_client_name, (int)strlen(jack_client_name)); + } } jack->client = jack_client_new(jack_client_name); @@ -423,6 +430,10 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) continue; if (strcmp(id, "comment") == 0 || strcmp(id, "type") == 0 || strcmp(id, "hint") == 0) continue; + if (strcmp(id, "name") == 0) { + snd_config_get_string(n, &name); + continue; + } if (strcmp(id, "playback_ports") == 0) { if (snd_config_get_type(n) != SND_CONFIG_TYPE_COMPOUND) { SNDERR("Invalid type for %s", id); -- 1.8.1.4 -------------------------------------------------------------- Intel Shannon Limited Registered in Ireland Registered Office: Collinstown Industrial Park, Leixlip, County Kildare Registered Number: 308263 Business address: Dromore House, East Park, Shannon, Co. Clare This e-mail and any attachments may contain confidential material for the sole use of the intended recipient(s). Any review or distribution by others is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies.

2 1

[alsa-devel] [PATCH] ALSA: pcm: export some constraint functions
by Jean-Francois Moine 06 Feb '14

06 Feb '14

The functions snd_pcm_hw_constraint_mask() and snd_pcm_hw_constraint_mask64() are needed to add audio constraints to substreams at open/startup time. This patch makes them available to ALSA modules. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/core/pcm_lib.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index a210467..aea4972 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1219,6 +1219,7 @@ int snd_pcm_hw_constraint_mask(struct snd_pcm_runtime *runtime, snd_pcm_hw_param return -EINVAL; return 0; } +EXPORT_SYMBOL(snd_pcm_hw_constraint_mask); /** * snd_pcm_hw_constraint_mask64 - apply the given bitmap mask constraint @@ -1242,6 +1243,7 @@ int snd_pcm_hw_constraint_mask64(struct snd_pcm_runtime *runtime, snd_pcm_hw_par return -EINVAL; return 0; } +EXPORT_SYMBOL(snd_pcm_hw_constraint_mask64); /** * snd_pcm_hw_constraint_integer - apply an integer constraint to an interval -- 1.9.rc1

2 1

[alsa-devel] [PATCH v3 0/5] add a TDA998x CODEC
by Jean-Francois Moine 05 Feb '14

05 Feb '14

The TDA998x HDMI transmitter accepts audio input from either I2S or S/PDIF. Theses inputs have different intrinsic constraints and these constraints may be modified by the audio parameters of the connected video device. The choice of I2S or S/PDIF may be the done by the user or by automatic processing (DPCM?) at each audio starting time. This asks for a dynamic audio input switch in the HDMI driver. This patch series implements the TDA998x specific CODEC. A simple function call mechanism is used for exchanges between the CODEC and the HDMI driver. Note: the changes in the TDA998x I2C driver are based on my previous patch series: http://lists.freedesktop.org/archives/dri-devel/2014-January/052837.html - v3 - move the DT description of the CODEC as a subnode of the TDA998x I2C (Mark Brown remark - is this the right way?) - change CODEC loading mechanism - simplify tda998x i2c reference search - CODEC DT documentation in drm/i2c/tda998x.txt - v2 - add ACLK setting and code optimization in patch 1 - from Mark Brown's remarks in patch 2: - don't compile the codec when CONFIG_ALL_CODECS - simplify the code about start/stop audio - fix coding style errors - add audio-port-names associated to audio-ports - add audio-port-names in patch 4 - add patch 5 'adjust the audio CTS_N pre-divider from audio format' for the Beaglebone-Black board (Jyri Sarha) Jean-Francois Moine (5): drm/i2c: tda998x: add a function for dynamic audio input switch ASoC: tda998x: add a codec driver for the TDA998x ASoC: tda998x: add DT documentation of the tda998x CODEC ASoC: tda998x: adjust the audio hw parameters from EDID ASoC: tda998x: adjust the audio CTS_N pre-divider from audio format .../devicetree/bindings/drm/i2c/tda998x.txt | 17 ++ drivers/gpu/drm/i2c/tda998x_drv.c | 86 +++++- include/drm/i2c/tda998x.h | 11 +- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tda998x.c | 303 +++++++++++++++++++++ 6 files changed, 419 insertions(+), 6 deletions(-) create mode 100644 sound/soc/codecs/tda998x.c -- 1.9.rc1

4 25

[alsa-devel] [PATCH] ASoC: cs42l73: Don't mix SNDRV_PCM_RATE_KNOT with specific rates
by Lars-Peter Clausen 05 Feb '14

05 Feb '14

SNDRV_PCM_RATE_KNOT means that the device can support rates that can not be expressed using the rate bits. The driver will provide a list of those rates specified through constraints. Any rate bits that are set in the rates mask will be ignored. So setting other rate bits besides SNDRV_PCM_RATE_KNOT wont have any effect, but might be confusing to the casual reader, so remove them. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/cs42l73.c | 15 ++++++--------- 1 file changed, 6 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 549d5d6..7cae046 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1255,9 +1255,6 @@ static int cs42l73_pcm_startup(struct snd_pcm_substream *substream, return 0; } -/* SNDRV_PCM_RATE_KNOT -> 12000, 24000 Hz, limit with constraint list */ -#define CS42L73_RATES (SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT) - #define CS42L73_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) @@ -1278,14 +1275,14 @@ static struct snd_soc_dai_driver cs42l73_dai[] = { .stream_name = "XSP Playback", .channels_min = 1, .channels_max = 2, - .rates = CS42L73_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = CS42L73_FORMATS, }, .capture = { .stream_name = "XSP Capture", .channels_min = 1, .channels_max = 2, - .rates = CS42L73_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = CS42L73_FORMATS, }, .ops = &cs42l73_ops, @@ -1298,14 +1295,14 @@ static struct snd_soc_dai_driver cs42l73_dai[] = { .stream_name = "ASP Playback", .channels_min = 2, .channels_max = 2, - .rates = CS42L73_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = CS42L73_FORMATS, }, .capture = { .stream_name = "ASP Capture", .channels_min = 2, .channels_max = 2, - .rates = CS42L73_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = CS42L73_FORMATS, }, .ops = &cs42l73_ops, @@ -1318,14 +1315,14 @@ static struct snd_soc_dai_driver cs42l73_dai[] = { .stream_name = "VSP Playback", .channels_min = 1, .channels_max = 2, - .rates = CS42L73_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = CS42L73_FORMATS, }, .capture = { .stream_name = "VSP Capture", .channels_min = 1, .channels_max = 2, - .rates = CS42L73_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = CS42L73_FORMATS, }, .ops = &cs42l73_ops, -- 1.8.0

5 10

[alsa-devel] [PATCH] MIDI driver for Behringer BCD2000 USB device
by Mario Kicherer 05 Feb '14

05 Feb '14

This patch adds initial support for the Behringer BCD2000 USB DJ controller. At the moment, only the MIDI part of the device is working, i.e. knobs, buttons and LEDs. I also plan to add support for the audio part, but I assume that this will require more effort than the rather simple MIDI interface. Progress can be tracked at https://github.com/anyc/snd-usb-bcd2000. Signed-off-by: Mario Kicherer <dev(a)kicherer.org> --- sound/usb/Kconfig | 14 + sound/usb/Makefile | 2 +- sound/usb/bcd2000/Makefile | 3 + sound/usb/bcd2000/bcd2000.c | 629 ++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 647 insertions(+), 1 deletion(-) create mode 100644 sound/usb/bcd2000/Makefile create mode 100644 sound/usb/bcd2000/bcd2000.c diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index de9408b..50af405 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -146,5 +146,19 @@ config SND_USB_HIFACE To compile this driver as a module, choose M here: the module will be called snd-usb-hiface. +config SND_USB_BCD2000 + tristate "Behringer BCD2000 MIDI driver" + select SND_HWDEP + select SND_RAWMIDI + help + Say Y here to include MIDI support for the Behringer BCD2000 DJ + controller. + + Audio support is still work-in-progress at + https://github.com/anyc/snd-usb-bcd2000 + + To compile this driver as a module, choose M here: the module + will be called snd-usb-bcd2000. + endif # SND_USB diff --git a/sound/usb/Makefile b/sound/usb/Makefile index abe668f..2b92f0d 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -23,4 +23,4 @@ obj-$(CONFIG_SND_USB_UA101) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_USX2Y) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_US122L) += snd-usbmidi-lib.o -obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ hiface/ +obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ hiface/ bcd2000/ diff --git a/sound/usb/bcd2000/Makefile b/sound/usb/bcd2000/Makefile new file mode 100644 index 0000000..6540136 --- /dev/null +++ b/sound/usb/bcd2000/Makefile @@ -0,0 +1,3 @@ +snd-usb-bcd2000-y := bcd2000.o + +obj-$(CONFIG_SND_USB_BCD2000) += snd-usb-bcd2000.o \ No newline at end of file diff --git a/sound/usb/bcd2000/bcd2000.c b/sound/usb/bcd2000/bcd2000.c new file mode 100644 index 0000000..100c39f --- /dev/null +++ b/sound/usb/bcd2000/bcd2000.c @@ -0,0 +1,629 @@ +/* + * Behringer BCD2000 driver + * + * Copyright (C) 2014 Mario Kicherer (dev(a)kicherer.org) + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifdef CONFIG_USB_DEBUG +#define DEBUG 1 +#else +#define DEBUG 0 +#endif +#include <linux/kernel.h> +#include <linux/errno.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/module.h> +#include <linux/usb.h> +#include <linux/usb/audio.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/rawmidi.h> +#include "../usbaudio.h" +#include "../midi.h" + +#define DEVICE_NAME "Behringer BCD2000" +#define DEVICE_SHORTNAME "bcd2000" + +#define PREFIX DEVICE_SHORTNAME ": " +#define BUFSIZE 64 + +static struct usb_device_id id_table [] = { + { USB_DEVICE(0x1397, 0x00bd) }, + { }, +}; +MODULE_DEVICE_TABLE (usb, id_table); +MODULE_AUTHOR("Mario Kicherer, dev(a)kicherer.org"); +MODULE_DESCRIPTION("Behringer BCD2000 driver"); +MODULE_LICENSE("GPL"); + +static int debug = 0; +module_param(debug, int, S_IRUGO | S_IWUSR); +MODULE_PARM_DESC(debug, "enable debug output (default: 0)"); + +static unsigned char set_led_prefix[] = {0x03, 0x00}; + +static unsigned char bcd2000_init_sequence[] = { + 0x07, 0x00, 0x00, 0x00, 0x78, 0x48, 0x1c, 0x81, + 0xc4, 0x00, 0x00, 0x00, 0x5e, 0x53, 0x4a, 0xf7, + 0x18, 0xfa, 0x11, 0xff, 0x6c, 0xf3, 0x90, 0xff, + 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, + 0x18, 0xfa, 0x11, 0xff, 0x14, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0xf2, 0x34, 0x4a, 0xf7, + 0x18, 0xfa, 0x11, 0xff}; + +struct bcd2000 { + struct usb_device *dev; + struct snd_card *card; + struct usb_interface *intf; + int card_index; + struct snd_pcm *pcm; + struct list_head midi_list; + spinlock_t lock; + struct mutex mutex; + + int midi_out_active; + struct snd_rawmidi *rmidi; + struct snd_rawmidi_substream *midi_receive_substream; + struct snd_rawmidi_substream *midi_out_substream; + + unsigned char midi_in_buf[BUFSIZE]; + unsigned char midi_out_buf[BUFSIZE]; + + unsigned char midi_cmd_buf[3]; + unsigned char midi_cmd_offset; + + struct urb * midi_out_urb; + struct urb * midi_in_urb; +}; + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; + +static DEFINE_MUTEX(devices_mutex); +static unsigned int devices_used; +static struct usb_driver bcd2000_driver; + + + + + + +static int bcd2000_midi_input_open(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +static int bcd2000_midi_input_close(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +static void bcd2000_midi_input_trigger(struct snd_rawmidi_substream *substream, int up) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + + if (!bcd2k) + return; + + bcd2k->midi_receive_substream = up ? substream : NULL; +} + +static void bcd2000_midi_handle_input(struct bcd2000 *bcd2k, + const unsigned char *buf, int len) +{ + unsigned char length, cmd, buf_offset, tocopy; + int ret; + + if (!bcd2k->midi_receive_substream) + return; + + if (DEBUG || debug) { + printk(KERN_DEBUG PREFIX "received (%d bytes): ", len); + for (ret=0; ret < len; ret++) { + printk("%x ", buf[ret]); + } + printk("\n"); + } + + if (len < 2) + return; + + /* + * The BCD2000 sends MIDI commands with 2 or 3 bytes length. In case of + * 2 bytes, the command is the same as before. + * + * Packet structure: mm nn oo (pp) + * mm: payload length + * nn: MIDI command or note + * oo: note or velocity + * pp: velocity + */ + + length = buf[0]; + + /* ignore packets without payload, should not happen */ + if (length == 0) + return; + + /* + * The MIDI packets the BCD2000 sends can be arbitrarily truncated or + * concatenated. Therefore, this loop accumulates the bytes from the + * input buffer until a full valid MIDI packet is in the MIDI command + * buffer "midi_cmd_buf". + */ + + buf_offset = 1; + while (buf_offset <= length) { + cmd = buf[buf_offset]; + + if (bcd2k->midi_cmd_offset == 0 && cmd != 0x90 && cmd != 0xb0) { + /* + * this is a 2-byte midi packet -> reuse command byte from + * last midi packet + */ + bcd2k->midi_cmd_offset = 1; + } + + /* determine the number of bytes we want to copy this time */ + tocopy = min( 3 - bcd2k->midi_cmd_offset, length - (buf_offset - 1)); + + if (tocopy > 0) { + memcpy(&bcd2k->midi_cmd_buf[bcd2k->midi_cmd_offset], + &buf[buf_offset], tocopy); + } else + printk(KERN_ERR PREFIX "error parsing\n"); + + bcd2k->midi_cmd_offset += tocopy; + buf_offset += tocopy; + + /* is our MIDI packet complete? */ + if (bcd2k->midi_cmd_offset == 3) { + if (DEBUG || debug) { + printk(KERN_DEBUG PREFIX "send midi packet (%d bytes): ", + bcd2k->midi_cmd_offset); + for (ret=0;ret< bcd2k->midi_cmd_offset;ret++) { + printk("%x ", bcd2k->midi_cmd_buf[ret]); + } + printk("\n"); + } + + /* send MIDI packet */ + snd_rawmidi_receive(bcd2k->midi_receive_substream, + bcd2k->midi_cmd_buf, 3); + + bcd2k->midi_cmd_offset = 0; + } + } +} + +static int bcd2000_midi_output_open(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +static int bcd2000_midi_output_close(struct snd_rawmidi_substream *substream) +{ + struct bcd2000 * bcd2k = substream->rmidi->private_data; + if (bcd2k->midi_out_active) { + usb_free_urb(bcd2k->midi_out_urb); + bcd2k->midi_out_active = 0; + } + return 0; +} + +static void bcd2000_midi_send(struct bcd2000 *bcd2k, + struct snd_rawmidi_substream *substream) +{ + int len, ret; + + /* copy the "set LED" command bytes */ + memcpy(bcd2k->midi_out_buf, set_led_prefix, sizeof(set_led_prefix)); + + /* get MIDI packet */ + len = snd_rawmidi_transmit(substream, bcd2k->midi_out_buf + 3, BUFSIZE - 3); + + if (len <= 0) + return; + + /* set payload length */ + bcd2k->midi_out_buf[2] = len; + bcd2k->midi_out_urb->transfer_buffer_length = BUFSIZE; + + if (DEBUG || debug) { + printk(KERN_DEBUG PREFIX "sending: "); + for (ret=0;ret< 3 + len;ret++) { + printk("%x ", bcd2k->midi_out_buf[ret]); + } + printk("\n"); + } + + /* send packet to the BCD2000 */ + ret = usb_submit_urb(bcd2k->midi_out_urb, GFP_ATOMIC); + if (ret < 0) { + printk(KERN_ERR PREFIX + "bcd2000_midi_send(%p): usb_submit_urb() failed," + "ret=%d, len=%d: ", substream, ret, len); + switch (ret) { + case -ENOMEM: printk("ENOMEM\n"); break; + case -ENODEV: printk("ENODEV\n"); break; + case -ENXIO: printk("ENXIO\n"); break; + case -EINVAL: printk("EINVAL\n"); break; + case -EAGAIN: printk("EAGAIN\n"); break; + case -EFBIG: printk("EFBIG\n"); break; + case -EPIPE: printk("EPIPE\n"); break; + case -EMSGSIZE: printk("EMSGSIZE\n"); break; + default: printk("\n"); + } + } else { + bcd2k->midi_out_active = 1; + } +} + +static void bcd2000_midi_output_done(struct urb* urb) +{ + struct bcd2000 * bcd2k = urb->context; + + bcd2k->midi_out_active = 0; + if (urb->status != 0) { + printk(KERN_ERR PREFIX "urb status: %d\n", urb->status); + return; + } + + if (!bcd2k->midi_out_substream) + return; + + bcd2000_midi_send(bcd2k, bcd2k->midi_out_substream); +} + +static void bcd2000_command_reply_dispatch(struct urb* urb) +{ + int ret; + struct device *dev = &urb->dev->dev; + struct bcd2000 *bcd2k = urb->context; + + if (urb->status || !bcd2k) { + dev_warn(dev, "received urb->status = %i\n", urb->status); + return; + } + + if (urb->actual_length > 0) + bcd2000_midi_handle_input(bcd2k, urb->transfer_buffer, urb->actual_length); + + bcd2k->midi_in_urb->actual_length = 0; + ret = usb_submit_urb(bcd2k->midi_in_urb, GFP_ATOMIC); + if (ret < 0) + dev_err(dev, "unable to submit urb. OOM!?\n"); +} + +static void bcd2000_midi_output_trigger(struct snd_rawmidi_substream *substream, int up) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + + if (up) { + bcd2k->midi_out_substream = substream; + if (!bcd2k->midi_out_active) + bcd2000_midi_send(bcd2k, substream); + } else { + bcd2k->midi_out_substream = NULL; + } +} + +static struct snd_rawmidi_ops bcd2000_midi_output = +{ + .open = bcd2000_midi_output_open, + .close = bcd2000_midi_output_close, + .trigger = bcd2000_midi_output_trigger, +}; + +static struct snd_rawmidi_ops bcd2000_midi_input = +{ + .open = bcd2000_midi_input_open, + .close = bcd2000_midi_input_close, + .trigger = bcd2000_midi_input_trigger, +}; + +static void bcd2000_init_device(struct bcd2000 * bcd2k) { + int ret; + + /* copy init sequence into buffer */ + memcpy(bcd2k->midi_out_buf, bcd2000_init_sequence, 52); + bcd2k->midi_out_urb->transfer_buffer_length = 52; + + /* submit sequence */ + ret = usb_submit_urb(bcd2k->midi_out_urb, GFP_ATOMIC); + if (ret < 0) { + printk(KERN_ERR PREFIX + "bcd2000_init_device: usb_submit_urb() failed," + "ret=%d: ", ret); + switch (ret) { + case -ENOMEM: printk("ENOMEM\n"); break; + case -ENODEV: printk("ENODEV\n"); break; + case -ENXIO: printk("ENXIO\n"); break; + case -EINVAL: printk("EINVAL\n"); break; + case -EAGAIN: printk("EAGAIN\n"); break; + case -EFBIG: printk("EFBIG\n"); break; + case -EPIPE: printk("EPIPE\n"); break; + case -EMSGSIZE: printk("EMSGSIZE\n"); break; + default: printk("\n"); + } + } else { + bcd2k->midi_out_active = 1; + } + + /* send empty packet to enable button and controller events */ + bcd2k->midi_in_urb->actual_length = 0; + ret = usb_submit_urb(bcd2k->midi_in_urb, GFP_ATOMIC); +} + +static void bcd2000_init_midi(struct bcd2000 * bcd2k) { + int ret; + struct snd_rawmidi *rmidi; + + ret = snd_rawmidi_new(bcd2k->card, bcd2k->card->shortname, 0, + 1, /* output */ + 1, /* input */ + &rmidi); + + if (ret < 0) + return; + + strlcpy(rmidi->name, bcd2k->card->shortname, sizeof(rmidi->name)); + + rmidi->info_flags = SNDRV_RAWMIDI_INFO_DUPLEX; + rmidi->private_data = bcd2k; + + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT; + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, + &bcd2000_midi_output); + + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT; + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, + &bcd2000_midi_input); + + bcd2k->rmidi = rmidi; + + bcd2k->midi_in_urb = usb_alloc_urb(0, GFP_KERNEL); + bcd2k->midi_out_urb = usb_alloc_urb(0, GFP_KERNEL); + + usb_fill_int_urb(bcd2k->midi_in_urb, bcd2k->dev, + usb_rcvbulkpipe(bcd2k->dev, 0x81), + bcd2k->midi_in_buf, BUFSIZE, + bcd2000_command_reply_dispatch, bcd2k, 1); + + usb_fill_int_urb(bcd2k->midi_out_urb, bcd2k->dev, + usb_sndbulkpipe(bcd2k->dev, 0x1), + bcd2k->midi_out_buf, BUFSIZE, + bcd2000_midi_output_done, bcd2k, 1); + + bcd2000_init_device(bcd2k); +} + +static void bcd2000_close_midi(struct bcd2000 * bcd2k) { + usb_free_urb(bcd2k->midi_out_urb); + usb_free_urb(bcd2k->midi_in_urb); +} + + + + + +/* + * general part + */ + +static void bcd2000_free_usb_related_resources(struct bcd2000 *bcd2k, + struct usb_interface *interface) +{ + unsigned int i; + struct usb_interface *intf; + +// mutex_lock(&ua->mutex); +// free_stream_urbs(&ua->capture); +// free_stream_urbs(&ua->playback); +// mutex_unlock(&ua->mutex); +// free_stream_buffers(ua, &ua->capture); +// free_stream_buffers(ua, &ua->playback); + + for (i = 0; i < 1; ++i) { + mutex_lock(&bcd2k->mutex); + intf = bcd2k->intf; + bcd2k->intf = NULL; + mutex_unlock(&bcd2k->mutex); + if (intf) { + usb_set_intfdata(intf, NULL); + if (intf != interface) + usb_driver_release_interface(&bcd2000_driver, + intf); + } + } +} + +static void bcd2000_card_free(struct snd_card *card) +{ +// struct bcd2000 *bcd2k = card->private_data; +// mutex_destroy(&ua->mutex); +} + +static int bcd2000_probe(struct usb_interface *interface, const struct usb_device_id *usb_id) +{ + struct snd_card *card; + struct bcd2000 * bcd2k; + unsigned int card_index; + char usb_path[32]; + int err; + + + mutex_lock(&devices_mutex); + + for (card_index = 0; card_index < SNDRV_CARDS; ++card_index) + if (enable[card_index] && !(devices_used & (1 << card_index))) + break; + if (card_index >= SNDRV_CARDS) { + mutex_unlock(&devices_mutex); + return -ENOENT; + } + err = snd_card_create(index[card_index], id[card_index], THIS_MODULE, + sizeof(*bcd2k), &card); + if (err < 0) { + mutex_unlock(&devices_mutex); + return err; + } + card->private_free = bcd2000_card_free; + bcd2k = card->private_data; + bcd2k->dev = interface_to_usbdev(interface); + bcd2k->card = card; + bcd2k->card_index = card_index; + INIT_LIST_HEAD(&bcd2k->midi_list); + spin_lock_init(&bcd2k->lock); + mutex_init(&bcd2k->mutex); + +// INIT_LIST_HEAD(&bcd2k->ready_playback_urbs); +// tasklet_init(&bcd2k->playback_tasklet, +// playback_tasklet, (unsigned long)bcd2k); +// init_waitqueue_head(&bcd2k->alsa_capture_wait); +// init_waitqueue_head(&bcd2k->rate_feedback_wait); +// init_waitqueue_head(&bcd2k->alsa_playback_wait); + + bcd2k->intf = interface; + + snd_card_set_dev(card, &interface->dev); + + strcpy(card->driver, DEVICE_NAME); + strcpy(card->shortname, DEVICE_SHORTNAME); + usb_make_path(bcd2k->dev, usb_path, sizeof(usb_path)); + snprintf(bcd2k->card->longname, sizeof(bcd2k->card->longname), + DEVICE_NAME ", at %s, %s speed", + usb_path, + bcd2k->dev->speed == USB_SPEED_HIGH ? "high" : "full"); + + printk(KERN_INFO PREFIX "%s", bcd2k->card->longname); + +// err = alloc_stream_buffers(bcd2k, &bcd2k->capture); +// if (err < 0) +// goto probe_error; +// err = alloc_stream_buffers(bcd2k, &bcd2k->playback); +// if (err < 0) +// goto probe_error; +// +// err = alloc_stream_urbs(bcd2k, &bcd2k->capture, capture_urb_complete); +// if (err < 0) +// goto probe_error; +// err = alloc_stream_urbs(bcd2k, &bcd2k->playback, playback_urb_complete); +// if (err < 0) +// goto probe_error; +// +// err = snd_pcm_new(card, "bcd2000", 0, 1, 1, &bcd2k->pcm); +// if (err < 0) +// goto probe_error; +// bcd2k->pcm->private_data = bcd2k; +// strcpy(bcd2k->pcm->name, "bcd2000"); +// snd_pcm_set_ops(bcd2k->pcm, SNDRV_PCM_STREAM_PLAYBACK, &playback_pcm_ops); +// snd_pcm_set_ops(bcd2k->pcm, SNDRV_PCM_STREAM_CAPTURE, &capture_pcm_ops); + +// err = snd_usbmidi_create(card, bcd2k->intf, +// &bcd2k->midi_list, &midi_quirk); +// if (err < 0) +// goto probe_error; + + + + bcd2000_init_midi(bcd2k); + + + err = snd_card_register(card); + if (err < 0) + goto probe_error; + + usb_set_intfdata(interface, bcd2k); + devices_used |= 1 << card_index; + + mutex_unlock(&devices_mutex); + return 0; + +probe_error: + bcd2000_free_usb_related_resources(bcd2k, interface); + snd_card_free(card); + mutex_unlock(&devices_mutex); + return err; +} + +static void bcd2000_disconnect(struct usb_interface *interface) +{ + struct bcd2000 *bcd2k = usb_get_intfdata(interface); + struct list_head *midi; + + if (!bcd2k) + return; + + bcd2000_close_midi(bcd2k); + + mutex_lock(&devices_mutex); + +// set_bit(DISCONNECTED, &bcd2k->states); +// wake_up(&bcd2k->rate_feedback_wait); + + /* make sure that userspace cannot create new requests */ + snd_card_disconnect(bcd2k->card); + + /* make sure that there are no pending USB requests */ + list_for_each(midi, &bcd2k->midi_list) + snd_usbmidi_disconnect(midi); + +// abort_alsa_playback(bcd2k); +// abort_alsa_capture(bcd2k); +// mutex_lock(&bcd2k->mutex); +// stop_usb_playback(bcd2k); +// stop_usb_capture(bcd2k); +// mutex_unlock(&bcd2k->mutex); + + bcd2000_free_usb_related_resources(bcd2k, interface); + + devices_used &= ~(1 << bcd2k->card_index); + + snd_card_free_when_closed(bcd2k->card); + + mutex_unlock(&devices_mutex); +} + +static struct usb_driver bcd2000_driver = { + .name = DEVICE_SHORTNAME, + .probe = bcd2000_probe, + .disconnect = bcd2000_disconnect, + .id_table = id_table, +}; + +static int __init bcd2000_init(void) +{ + int retval = 0; + + retval = usb_register(&bcd2000_driver); + if (retval) + printk(KERN_INFO "usb_register failed. Error number %d", retval); + return retval; +} + +static void __exit bcd2000_exit(void) +{ + usb_deregister(&bcd2000_driver); +} + +module_init (bcd2000_init); +module_exit (bcd2000_exit); -- 1.8.3.2

2 3

[alsa-devel] [PATCH] ASoC: cs4271: Remove outdated comment
by Lars-Peter Clausen 05 Feb '14

05 Feb '14

Commit 1b1861ead ("ASoC: cs4271: convert to direct regmap API usage") removed the bus_type field from the cs4271_private struct, but left the comment that described the field in there. This patch removes the comment. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/cs4271.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index ce05fd9..f7bbe6f 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -159,7 +159,6 @@ static bool cs4271_volatile_reg(struct device *dev, unsigned int reg) } struct cs4271_private { - /* SND_SOC_I2C or SND_SOC_SPI */ unsigned int mclk; bool master; bool deemph; -- 1.8.0

3 2

[alsa-devel] [PATCH 1/2] ALSA: hda - hdmi: introduce patch_nvhdmi()
by Stephen Warren 05 Feb '14

05 Feb '14

From: Anssi Hannula <anssi.hannula(a)iki.fi> (This is a backport of *part* of upstream 611885bc963a "ALSA: hda - hdmi: Disallow unsupported 2ch remapping on NVIDIA codecs" to stable 3.10 through 3.12. Later stable already contain all of the original patch.) Mainline commit 611885bc963a "ALSA: hda - hdmi: Disallow unsupported 2ch remapping on NVIDIA codecs" introduces function patch_nvhdmi(). That function is edited by 75fae117a5db "ALSA: hda/hdmi - allow PIN_OUT to be dynamically enabled". In order to backport the PIN_OUT patch, I am first back-porting just the addition of function patch_nvhdmi(), so that the conflicts applying the PIN_OUT patch are simplified. Ideally, one might backport all of 611885bc963a. However, that commit doesn't apply to stable kernels, since it relies on a chain of other patches which implement new features. Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> [swarren, extracted just a small part of the original patch] Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/pci/hda/patch_hdmi.c | 60 ++++++++++++++++++++++++++++------------------ 1 file changed, 37 insertions(+), 23 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index f26c42c92db7..aad73a1fc2cd 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2560,6 +2560,20 @@ static int patch_nvhdmi_8ch_7x(struct hda_codec *codec) return 0; } +static int patch_nvhdmi(struct hda_codec *codec) +{ + struct hdmi_spec *spec; + int err; + + err = patch_generic_hdmi(codec); + if (err) + return err; + + spec = codec->spec; + + return 0; +} + /* * ATI-specific implementations * @@ -2632,30 +2646,30 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x10de0005, .name = "MCP77/78 HDMI", .patch = patch_nvhdmi_8ch_7x }, { .id = 0x10de0006, .name = "MCP77/78 HDMI", .patch = patch_nvhdmi_8ch_7x }, { .id = 0x10de0007, .name = "MCP79/7A HDMI", .patch = patch_nvhdmi_8ch_7x }, -{ .id = 0x10de000a, .name = "GPU 0a HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de000b, .name = "GPU 0b HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de000c, .name = "MCP89 HDMI", .patch = patch_generic_hdmi }, -{ .id = 0x10de000d, .name = "GPU 0d HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0010, .name = "GPU 10 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0011, .name = "GPU 11 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0012, .name = "GPU 12 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0013, .name = "GPU 13 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0014, .name = "GPU 14 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0015, .name = "GPU 15 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0016, .name = "GPU 16 HDMI/DP", .patch = patch_generic_hdmi }, +{ .id = 0x10de000a, .name = "GPU 0a HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de000b, .name = "GPU 0b HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de000c, .name = "MCP89 HDMI", .patch = patch_nvhdmi }, +{ .id = 0x10de000d, .name = "GPU 0d HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0010, .name = "GPU 10 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0011, .name = "GPU 11 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0012, .name = "GPU 12 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0013, .name = "GPU 13 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0014, .name = "GPU 14 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0015, .name = "GPU 15 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0016, .name = "GPU 16 HDMI/DP", .patch = patch_nvhdmi }, /* 17 is known to be absent */ -{ .id = 0x10de0018, .name = "GPU 18 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0019, .name = "GPU 19 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de001a, .name = "GPU 1a HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de001b, .name = "GPU 1b HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de001c, .name = "GPU 1c HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0040, .name = "GPU 40 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0041, .name = "GPU 41 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0042, .name = "GPU 42 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0043, .name = "GPU 43 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0044, .name = "GPU 44 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0051, .name = "GPU 51 HDMI/DP", .patch = patch_generic_hdmi }, -{ .id = 0x10de0060, .name = "GPU 60 HDMI/DP", .patch = patch_generic_hdmi }, +{ .id = 0x10de0018, .name = "GPU 18 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0019, .name = "GPU 19 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de001a, .name = "GPU 1a HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de001b, .name = "GPU 1b HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de001c, .name = "GPU 1c HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0040, .name = "GPU 40 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0041, .name = "GPU 41 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0042, .name = "GPU 42 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0043, .name = "GPU 43 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0044, .name = "GPU 44 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0051, .name = "GPU 51 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0060, .name = "GPU 60 HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch }, { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch }, { .id = 0x11069f80, .name = "VX900 HDMI/DP", .patch = patch_via_hdmi }, -- 1.8.1.5

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Re: [alsa-devel] Strange sound of IDT 82hd206xx6 at font port in AC97 mode
by Randy Li 05 Feb '14

05 Feb '14

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 于 2014年02月04日 17:01, Raymond Yau 写道: > > > >> Check the front audio connector of your motherboard in the user >> manual >> >> Check the front audio panel specification of your computer >> chassis whether support Jack detection of headphone and mic >> >> Only a few multi channel AC97 2.3 codecs (e.g. ALC850) support >> jack detection and most AC97 front audio panel does not have >> jack detection circuit >> >> http://www.intel.com/support/motherboards/desktop/sb/cs-015851.htm > >> > >>> I can't hear the sound in some part of a film and I can hear >>> the stranger accent of sound in the other part of a film(or a >>> music). But all the 7.1 channel in the back ports can output >>> the sound match my mircophone(all are the same). >> >> The front audio panel headphone share the front channel signal >> only when playing multi channel audio >> >> You need to use applications（e.g mplayer） which can down mix >> multi channel to stereo or select the stereo track instead of >> DTS of movie DVD when you are using headphone >> > I find the other thing, this headphone is for android mobile which > combine controller, headphone and speaker. If you don't plug it > into font port fully, it works. I have tested a nornal headphone, > it works too. > > Maybe it cause by my hardware, it is not alsa's problem. > > > > Simple mixer control 'Swap Center/LFE',0 Capabilities: pswitch > pswitch-joined penum Playback channels: Mono Mono: Playback [off] > > > > Do this switch change the output of center and lfe when you run > > speaker-test -c 6 -t wav -D hw:0,0 > I only hear the font left and font right in both side of my headphone(the one for android) > seem that your are not using latest alsa driver The alsa driver? Well, shall I try the latest kernel? thank you ayaka -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.12 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iQEcBAEBAgAGBQJS8ORfAAoJEPb4VsMIzTzi/AYH/1UnTvhJACTMi2e9P+mqAnv5 tAH/bVBh7l8RDX6UnYO5O3V9af5NvTHxdYVHcnjjyP2bh95YUSCJansP2xViydh2 51u/fuXU4sx1TgZzvw1LZpa8cqjRXcIaFbUQWP23qzAV5yFwrFcEsrvW3KgHCHVj dwqs6d/tQzWjNvM2LwTIqZmX4u4dSUUaJ//3ETC19ubi1YJkrqFT6vWUUYwPxkzm Oz9SELfJho81xk/TAePWh9HPUYpEGJfJzbDTwbkrLLnt65GCpl7aeRbjCbRq7CPN bUrKd+VsvziNZyKUHaImDwRqnVGUcyPRdG8rzFEfMWHoRsN0NSRAR2dE/oTqOpY= =k4l9 -----END PGP SIGNATURE-----

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[alsa-devel] [PATCH] ALSA: hda - Improve loopback path lookups for AD1983
by Takashi Iwai 05 Feb '14

05 Feb '14

AD1983 has flexible loopback routes and the generic parser would take wrong path confusingly instead of taking individual paths via NID 0x0c and 0x0d. For avoiding it, limit the connections at these widgets so that the parser can think more straightforwardly. This fixes the regression of the missing line-in loopback on Dell machine. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=70011 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_analog.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 195cd62cdce5..df3652ad15ef 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -491,6 +491,8 @@ static int ad1983_add_spdif_mux_ctl(struct hda_codec *codec) static int patch_ad1983(struct hda_codec *codec) { struct ad198x_spec *spec; + static hda_nid_t conn_0c[] = { 0x08 }; + static hda_nid_t conn_0d[] = { 0x09 }; int err; err = alloc_ad_spec(codec); @@ -501,6 +503,11 @@ static int patch_ad1983(struct hda_codec *codec) spec->gen.mixer_nid = 0x0e; spec->gen.beep_nid = 0x10; set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); + + /* limit the loopback routes not to confuse the parser */ + snd_hda_override_conn_list(codec, 0x0c, ARRAY_SIZE(conn_0c), conn_0c); + snd_hda_override_conn_list(codec, 0x0d, ARRAY_SIZE(conn_0d), conn_0d); + err = ad198x_parse_auto_config(codec, false); if (err < 0) goto error; -- 1.8.5.2

1 0

[alsa-devel] [PATCH] ALSA: hda - Add missing mixer widget for AD1983
by Takashi Iwai 05 Feb '14

05 Feb '14

The mixer widget on AD1983 at NID 0x0e was missing in the commit [f2f8be43c5c9: ALSA: hda - Add aamix NID to AD codecs]. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=70011 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_analog.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 50b2427f19ca..195cd62cdce5 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -498,6 +498,7 @@ static int patch_ad1983(struct hda_codec *codec) return err; spec = codec->spec; + spec->gen.mixer_nid = 0x0e; spec->gen.beep_nid = 0x10; set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); err = ad198x_parse_auto_config(codec, false); -- 1.8.5.2

1 0

[alsa-devel] [PATCH] ALSA: hda/realtek - Avoid invalid COEFs for ALC271X
by Takashi Iwai 05 Feb '14

05 Feb '14

We've seen often problems after suspend/resume on Acer Aspire One AO725 with ALC271X codec as reported in kernel bugzilla, and it turned out that some COEFs doesn't work and triggers the codec communication stall. Since these magic COEF setups are specific to ALC269VB for some PLL configurations, the machine works even without these manual adjustment. So, let's simply avoid applying them for ALC271X. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=52181 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 56a8f1876603..e16a71e586f0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4427,6 +4427,9 @@ static void alc269_fill_coef(struct hda_codec *codec) if (spec->codec_variant != ALC269_TYPE_ALC269VB) return; + /* ALC271X doesn't seem to support these COEFs (bko#52181) */ + if (!strcmp(codec->chip_name, "ALC271X")) + return; if ((alc_get_coef0(codec) & 0x00ff) < 0x015) { alc_write_coef_idx(codec, 0xf, 0x960b); -- 1.8.5.2

1 0

[alsa-devel] DMA screams on MSP
by Linus Walleij 04 Feb '14

04 Feb '14

Hi Lee, I get this in my bootlog nowadays (v3.14-rc1): [ 2.713897] dma dma0chan29: [d40_validate_conf] Invalid device type (14) physical channel exeeds available physical channels [ 2.725158] dma dma0chan30: [d40_validate_conf] Invalid device type (14) physical channel exeeds available physical channels [ 2.737457] dma dma0chan31: [d40_validate_conf] Invalid device type (14) physical channel exeeds available physical channels [ 2.750793] dma dma0chan32: [d40_validate_conf] Invalid device type (14) physical channel exeeds available physical channels [ 2.762145] dma dma0chan33: [d40_validate_conf] Invalid device type (14) physical channel exeeds available physical channels [ 2.773376] dma dma0chan34: [d40_validate_conf] Invalid device type (14) physical channel exeeds available physical channels [ 2.784606] dma dma0chan35: [d40_validate_conf] Invalid device type (14) physical channel exeeds available physical channels [ 2.795837] dma dma0chan36: [d40_validate_conf] Invalid device type (14) physical channel exeeds available physical channels [ 2.807067] dma dma0chan37: [d40_validate_conf] Invalid device type (14) physical channel exeeds available physical channels [ 2.818298] dma dma0chan38: [d40_validate_conf] Invalid device type (14) physical channel exeeds available physical channels [ 2.829528] dma dma0chan39: [d40_validate_conf] Invalid device type (14) physical channel exeeds available physical channels [ 2.840728] dma dma0chan40: [d40_validate_conf] Invalid device type (14) physical channel exeeds available physical channels [ 2.851959] dma dma0chan41: [d40_validate_conf] Invalid device type (14) physical channel exeeds available physical channels [ 2.863189] dma dma0chan42: [d40_validate_conf] Invalid device type (14) physical channel exeeds available physical channels (...) [ 5.315887] dma dma0chan252: [d40_validate_conf] Invalid device type (14) physical channel exeeds available physical channels [ 5.327209] dma dma0chan253: [d40_validate_conf] Invalid device type (14) physical channel exeeds available physical channels [ 5.338531] dma dma0chan254: [d40_validate_conf] Invalid device type (14) physical channel exeeds available physical channels [ 5.349853] dma dma0chan255: [d40_validate_conf] Invalid device type (14) physical channel exeeds available physical channels (I patched dma40 to state the exact cause instead of just "invalid device type".) Like something (I guess ALSA SoC?) is trying real hard to get physical channel 14, but I can't seem to figure out what is going on. I will try to see if this also appears on the Snowball but I guess so? Any hints appreciated... Yours, Linus Walleij

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[alsa-devel] [PATCH 1/2] devicetree: Add devicetree bindings documentation for the ADAU1977
by Lars-Peter Clausen 04 Feb '14

04 Feb '14

The ADAU1977 is a four channel audio ADC. It can either be connected to an I2C or a SPI bus. Both bindings are described in this document. Other than the standard I2C and SPI bus properties a regulator for the AVDD supply needs to be specified in the bindings. Optionally, if present in the hardware design, a regulator for the DVDD supply and a GPIO connected to the chips reset pin can be specified. The bindings also allow to specify the microphone bias voltage that should be used with the hardware design. Cc: Rob Herring <robh+dt(a)kernel.org> Cc: Pawel Moll <pawel.moll(a)arm.com> Cc: Mark Rutland <mark.rutland(a)arm.com> Cc: Ian Campbell <ijc+devicetree(a)hellion.org.uk> Cc: Kumar Gala <galak(a)codeaurora.org> Cc: devicetree(a)vger.kernel.org Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- .../devicetree/bindings/sound/adi,adau1977.txt | 58 ++++++++++++++++++++++ 1 file changed, 58 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/adi,adau1977.txt diff --git a/Documentation/devicetree/bindings/sound/adi,adau1977.txt b/Documentation/devicetree/bindings/sound/adi,adau1977.txt new file mode 100644 index 0000000..3cf03d8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,adau1977.txt @@ -0,0 +1,58 @@ +Analog Devices ADAU1977, ADAU1978, ADAU1979 devicetree bindings + +The ADAU1977 is a four channel audio ADC. It can either be connected to an I2C +or a SPI bus. Both bindings are described in this document. Depending on the bus +type different properties must/can be specified. + +Shared required properties: + - compatible: Must be "adi,adau1977", "adi,adau1978" or adi,adau1979" + - AVDD-supply: A handle to the power supply connected to the AVDD pin + +Shared optional properties: + - DVDD-supply: A handle to the power supply connected to the DVDD pin. If + specified the internal LDO will be disabled. + - reset-gpios: GPIO chip handle and specifier that define which GPIO is + connected to the chips PD/RST pin. If not specified the reset pin is assumed + to be hardwired to VCC. + - adi,micbias: Configures the voltage setting for the MICBIAS pin. If not + specified the default value will be 8.5 Volts. This property is only valid + for the ADAU1977. + Valid values for this property are: + 0: 5.0V + 1: 5.5V + 2: 6.0V + 3: 6.5V + 4: 7.0V + 5: 7.5V + 6: 8.0V + 7: 8.5V + 8: 9.0V + +I2C required properties: + - reg: The I2C address of the chip + +SPI required properties: + - reg: The SPI chipselect signal of the SPI master associated with this chip + - spi-max-frequency: Maximum spi frequency to use. + +Examples: + +&i2c_bus { + adau1977: codec@11 { + compatible = "adi,adau1977"; + reg = <0x11>; + reset-gpios = <&gpio 5 0>; + AVDD-supply = <&codec_avdd_supply>; + DVDD-supply = <&codec_dvdd_supply>; + }; +}; + +&spi_bus { + adau1977: codec@1 { + compatible = "adi,adau1977"; + reg = <0x1>; + spi-max-frequency = <10000000>; + AVDD-supply = <&codec_avdd_supply>; + adi,micbias = <5>; + }; +}; -- 1.8.0

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[alsa-devel] [PATCH 1/2 v2] mfd: wm5102: Update register patch
by Charles Keepax 04 Feb '14

04 Feb '14

Update the register patch based on latest evaluation of the device. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Changed since v1: - Updated commit message on patch 2 Thanks, Charles drivers/mfd/wm5102-tables.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/drivers/mfd/wm5102-tables.c b/drivers/mfd/wm5102-tables.c index 1e9a4b2..f9b1e96 100644 --- a/drivers/mfd/wm5102-tables.c +++ b/drivers/mfd/wm5102-tables.c @@ -73,6 +73,7 @@ static const struct reg_default wm5102_revb_patch[] = { { 0x171, 0x0000 }, { 0x35E, 0x000C }, { 0x2D4, 0x0000 }, + { 0x4DC, 0x0900 }, { 0x80, 0x0000 }, }; -- 1.7.2.5

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[alsa-devel] Strange sound of IDT 82hd206xx6 at font port in AC97 mode
by Randy Li 04 Feb '14

04 Feb '14

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello My mainboard is ECS A780GM-A, the sound chip in it is IDT 82HD206XX6, it shows that below 00:14.2 Audio device: Advanced Micro Devices [AMD] nee ATI SBx00 Azalia (Intel HDA) Subsystem: Elitegroup Computer Systems Device 2816 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=slow >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 16 Region 0: Memory at f9ff0000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Kernel driver in use: snd_hda_intel In my bios I don't have a switch to make the font port mode into AC97 or HD Audio. In windows you need to install a driver and set the font port into AC97. In Linux, you can use the font port, but it seems that audio channel is wrong, I can't hear the sound in some part of a film and I can hear the stranger accent of sound in the other part of a film(or a music). But all the 7.1 channel in the back ports can output the sound match my mircophone(all are the same). How to fix the problem? Kernel is 3.4.77 ayaka -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.12 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iQEcBAEBAgAGBQJS7z6nAAoJEPb4VsMIzTziIY0H/1Um6EnDRGIJ+4DLl6h5AgCl Bzcg3haOtFPX4j473TWOOUVqFjesjYqlvWIQaRLorGTNVVwBYRLef/MG4R4DKeti pXa6gu92UrkAxOtCWUE71u/snCkEzfe6py5uRV6ut2fL+FpewujHQYrxpyIlBmro MaWEN6BjMK2qUFCgx3te7ssKa0HEamjzIPa2HWEDKryNB4jTrBLjYEWng3dXESu3 czZIFgMCOIHOsRiaB9P1Z/7NTx0xgHsBOspKMUNS/s4hmP5kQ1cev04qrDCB9c2N /mUMStViGuZ13r8rO527KEsvuXc41Yt3V9b34wLi+FoIGZoKfDn+Y6K/3u63fD0= =y/91 -----END PGP SIGNATURE-----

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[alsa-devel] [PATCH 1/2] mfd: wm5102: Update register patch
by Charles Keepax 03 Feb '14

03 Feb '14

Update the register patch based on latest evaluation of the device. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- drivers/mfd/wm5102-tables.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/drivers/mfd/wm5102-tables.c b/drivers/mfd/wm5102-tables.c index 1e9a4b2..f9b1e96 100644 --- a/drivers/mfd/wm5102-tables.c +++ b/drivers/mfd/wm5102-tables.c @@ -73,6 +73,7 @@ static const struct reg_default wm5102_revb_patch[] = { { 0x171, 0x0000 }, { 0x35E, 0x000C }, { 0x2D4, 0x0000 }, + { 0x4DC, 0x0900 }, { 0x80, 0x0000 }, }; -- 1.7.2.5

3 7

[alsa-devel] [PATCH 1/2] ASoC: h1940_uda1380: Don't modify runtime->hw
by Lars-Peter Clausen 03 Feb '14

03 Feb '14

Individual components are not supposed to modify the runtime->hw struct itself as it is manged by the ASoC core. If a component wants to limit the list of supported rates it should set a rate constraint. The h1940_uda1380 already does this, so it is safe to just remove the code that modifies runtime->hw. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/samsung/h1940_uda1380.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index ec5cbdd..7a85ce8 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -66,10 +66,6 @@ static int h1940_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - runtime->hw.rate_min = hw_rates.list[0]; - runtime->hw.rate_max = hw_rates.list[hw_rates.count - 1]; - runtime->hw.rates = SNDRV_PCM_RATE_KNOT; - return snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_rates); -- 1.8.0

3 3

[alsa-devel] [PATCH 0/3] ASoC: davinci-mcasp: Cleanup patches
by Peter Ujfalusi 03 Feb '14

03 Feb '14

Hi Mark, The first patch in this series is the one which did not applied cleanly from my previous series. The other two is just generic code cleanup patches. Regards, Peter --- Peter Ujfalusi (3): ASoC: davinci-mcasp: Code cleanup in davinci_mcasp_hw_params() ASoC: davinci-mcasp: Rename platform data struct ASoC: davinci-mcasp: Remove long lines from suspend/resume callbacks include/linux/platform_data/davinci_asp.h | 4 +- sound/soc/davinci/davinci-mcasp.c | 86 +++++++++++++++---------------- 2 files changed, 46 insertions(+), 44 deletions(-) -- 1.8.5.3

2 4

Re: [alsa-devel] linux-next: Tree for Feb 3 (sound/soc/soc-io.c)
by Randy Dunlap 03 Feb '14

03 Feb '14

On 02/02/2014 07:16 PM, Stephen Rothwell wrote: > Hi all, > > This tree fails (more than usual) the powerpc allyesconfig build. > > Changes since 20140131: > on i386: when CONFIG_REGMAP_I2C=m: sound/built-in.o: In function `snd_soc_codec_set_cache_io': (.text+0x29555): undefined reference to `regmap_init_i2c' Full randconfig file is attached. -- ~Randy

1 0

[alsa-devel] compress: adding support for hardware dependent params
by Banajit Goswami 03 Feb '14

03 Feb '14

Hi, We have a requirement, where we need to set a number of hardware specific parameters for compressed offload use-cases, before starting a stream. As these parameters need to be set once for the entire session, we would like to explore the best way to set these additional parameters somewhere before calling trigger() function. Below are few of the proposed solutions that we could think of at this point for getting this done- 1. Adding new IOCTL call: Add a new ioctl control in compress-offload.c file to handle hardware specific parameters setting. This new ioctl control will call into soc-compress.c and eventually call into platform driver ioctl to set those hardware specific parameters. In this case, we will add a new structure, let's say "snd_compr_hw_ctrl_params", with the same fields as "snd_compr_metadata" struct. 2. Adding a function hw_dependant_hw_params(), which will be called after hw_params() and before prepare() function. The new function will eventually call into the platform driver to get the hardware specific parameters set. Once implemented, this would look something like- hw_params(); hw_dependant_hw_params(); prepare(); 3. Add a function "hw_dependant_hw_params()", as _part_ of the hw_params() function. This new function will be called after the (existing) generic parameters are set. So once implemented, the proposed function calls would look like- hw_params() { generic_hw_params(); hw_dependant_hw_params(); } prepare(); We would like to get feedback on these proposed solutions and get a discussion going to see how best these requirements can be fitted in the current Compressed ALSA framework (for mainlining). Thanks and Regards, Banajit Goswami The Qualcomm Innovation Center, Inc. is a member of Code Aurora Forum, hosted by The Linux Foundation

2 1

[alsa-devel] [PATCH v3 0/6] ASoC: tlv320aic32x4: DT support
by Markus Pargmann 03 Feb '14

03 Feb '14

Hi, The series adds basic DT support for the TI tlv320aic32x4 codec. Patch 3 ("ASoC: tlv320aic32x4: Use signed int mixer controls") depends on series "[PATCH 0/2] ASoC: core: volume control using signed register values". I added the remaining two patches of the "ASoC: tlv320aic32x4 bugfixes" series to the beginning of this series to have all tlv320aic32x4 patches in one series. Regards, Markus Changes in v3: - Rename property gpio-reset to reset-gpios - Use negative constants as minimum volume instead of casting to negative integers Changes in v2: - Regulator support - Fix documentation for gpio property Markus Pargmann (6): ASoC: tlv320aic32x4: Fix mono playback ASoC: tlv320aic32x4: Fix MICPGA input configuration ASoC: tlv320aic32x4: Use signed int mixer controls ASoC: tlv320aic32x4: DT support ASoC: tlv320aic32x4: Support for master clock ASoC: tlv320aic32x4: Support for regulators .../devicetree/bindings/sound/tlv320aic32x4.txt | 28 ++++ sound/soc/codecs/tlv320aic32x4.c | 154 ++++++++++++++++++--- sound/soc/codecs/tlv320aic32x4.h | 3 + 3 files changed, 164 insertions(+), 21 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic32x4.txt -- 1.8.5.2

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[alsa-devel] [PATCH 0/4] ASoC: davinci-mcasp: Fix master mode capture (w/o playback)
by Peter Ujfalusi 03 Feb '14

03 Feb '14

Hi, When McASP is master the TX clocks are used for capture clocking as well. Other parts of the code already does the configuration for both tx and rx - for different bits in xxFMT/xxFMCT registers we should do the same in other places. Along with this fix I have added three cleanup patch which came along when I was debugging the issue. Regards, Peter --- Peter Ujfalusi (4): ASoC: davinci-mcasp: Harmonize the sub hw_params function names ASoC: davinci-mcasp: Configure xxTDM, xxFMT and xxFMCT registers synchronously ASoC: davinci-mcasp: Return value handling cleanup for mcasp_common_hw_param() ASoC: davinci-mcasp: Code cleanup in davinci_mcasp_hw_params() sound/soc/davinci/davinci-mcasp.c | 81 +++++++++++++++++++-------------------- 1 file changed, 39 insertions(+), 42 deletions(-) -- 1.8.5.3

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Re: [alsa-devel] Got headphone out working via hda-verb setting on MacPro1, 1
by Takashi Iwai 03 Feb '14

03 Feb '14

At Fri, 31 Jan 2014 18:50:11 +0100, Nicolai Beuermann wrote: > > Am 22.01.2014 11:29, schrieb Takashi Iwai: > > At Tue, 21 Jan 2014 01:13:03 +0100, > > Nicolai Beuermann wrote: > >> > >> Hello, > >> hda-analyzer told me to write a note to this list when settings are > >> successful. > >> > >> My setup: > >> MacPro 1,1 > >> Kernel 3.10.17 (gentoo sources) > >> ALSA k3.10.17 > >> > >> Had to set > >> /usr/bin/hda-verb /dev/snd/hwC0D0 0x18 SET_PIN_WIDGET_CONTROL 0xc1 > >> to get the headphone out at the front working. > >> > >> Wrote a bit (in german) about it: > >> http://www.nico-beuermann.de/blogging/archives/117-Hier-gibt-es-was-auf-die… > > > > Could you check whether 3.13 still has the same problem? > > If yes, please give alsa-info.sh output. Better to run it with > > --no-upload and attach the output. > > 3.13 still has the same problem. > alsa-infos's output has been attached. Thanks. Could you try the patch below with 3.14-rc1? 3.13.1 still doesn't contain the fix for MBA 1,1 (which will be merged to stable kernel soon later). Alternatively, you can cherry-pick the commit e7729a415315fcd9516912050d85d5aaebcededc ALSA: hda - Fix silent output on MacBook Air 1,1 from Linus tree, and apply the patch below on its top. Takashi -- 8< -- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: hda - Fix missing VREF setup for Mac Pro 1,1 Mac Pro 1,1 with ALC889A codec needs the VREF setup on NID 0x18 to VREF50, in order to make the speaker working. The same fixup was already needed for MacBook Air 1,1, so we can reuse it. Reported-by: Nicolai Beuermann <mail(a)nico-beuermann.de> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cbeb67d98852..af7a0c372292 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1821,6 +1821,7 @@ enum { ALC889_FIXUP_IMAC91_VREF, ALC889_FIXUP_MBA11_VREF, ALC889_FIXUP_MBA21_VREF, + ALC889_FIXUP_MP11_VREF, ALC882_FIXUP_INV_DMIC, ALC882_FIXUP_NO_PRIMARY_HP, ALC887_FIXUP_ASUS_BASS, @@ -2190,6 +2191,12 @@ static const struct hda_fixup alc882_fixups[] = { .chained = true, .chain_id = ALC889_FIXUP_MBP_VREF, }, + [ALC889_FIXUP_MP11_VREF] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc889_fixup_mba11_vref, + .chained = true, + .chain_id = ALC885_FIXUP_MACPRO_GPIO, + }, [ALC882_FIXUP_INV_DMIC] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_inv_dmic_0x12, @@ -2253,7 +2260,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC889_FIXUP_MBP_VREF), SND_PCI_QUIRK(0x106b, 0x00a1, "Macbook", ALC889_FIXUP_MBP_VREF), SND_PCI_QUIRK(0x106b, 0x00a4, "MacbookPro 4,1", ALC889_FIXUP_MBP_VREF), - SND_PCI_QUIRK(0x106b, 0x0c00, "Mac Pro", ALC885_FIXUP_MACPRO_GPIO), + SND_PCI_QUIRK(0x106b, 0x0c00, "Mac Pro", ALC889_FIXUP_MP11_VREF), SND_PCI_QUIRK(0x106b, 0x1000, "iMac 24", ALC885_FIXUP_MACPRO_GPIO), SND_PCI_QUIRK(0x106b, 0x2800, "AppleTV", ALC885_FIXUP_MACPRO_GPIO), SND_PCI_QUIRK(0x106b, 0x2c00, "MacbookPro rev3", ALC889_FIXUP_MBP_VREF), -- 1.8.5.2

1 0

[alsa-devel] mentoring for open sound hardware project?
by Hubert Denkmair 03 Feb '14

03 Feb '14

Hello list, I'm currently pushing a project for free audio hardware (basically a digital audio interface / mixer with attached linux system). The details are not final yet, but the design involves a FPGA for audio routing and a ARM SoC board for recording, playback etc. Plan is to support quite a few channels on the linux side, say 20 in, 20 out at 96kHz/24bit. Of course, this needs a Alsa driver. My question is: Is there a "best way" to do the FPGA<->Linux connection? Using the linux GPMC system for communication is the best thing we came up with until now. Are there already drivers doing something alike that we could build upon? Since there is no kernel developer in our team (yet), is someone on this list willing to mentor our driver development? Greetings, Hubert Denkmair

1 0

[alsa-devel] [RFC][PATCH 0/8] A new driver for OXFW970/971 based devices
by Takashi Sakamoto 03 Feb '14

03 Feb '14

This series of patch is based on my previous series below: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-December/070424.h… This series of patch add a new driver, snd-oxfw, for OXFW970/971 based devices. These chipsets are already supported by snd-firewire-speakers but this new driver aim to support recording equipment. Current supported devices: * Behringer F-Control Audio 202 * Mackie Onyx-i series (former model) * Mackie Onyx Satellite Devices possible to be supported if identifying IDs: * Mackie, d.2 pro * Mackie, d.4 pro * Mackie, U.420 * Mackie, U.420d * Mackie, Tapco Link.Firewire I worked with Behringer F-Control Audio 202. So I want someone to test with the other devices, especially about some quirks. Clemens, would you gime me your opinions about issues below? And if you know something about this chipset, would you share the information? 1.snd-firewire-speakers and snd-oxfw With this series of patches, ALSA has two drivers for the same chipsets. Merging these two drivers is a bit difficult because: - snd-firewire-speakers has control interface but snd-oxfw don't - snd-firewire-speakers don't support in-stream but snd-oxfw supports 2.currently snd-oxfw works with streams in SYT-Match Due to invalid sequence of 'presentation timestamp' in transmitted packets. In detail, please see 4th patch. 3.assumption of channnel formation Some commands may not be implemented. In detail, please see 3rd patch. 4.don't check source of clock All of supported devices (including can be) don't have the functionality to switch source of clock. So I want to omit this. For issue 3 and 4, Descriptor mechanism in 'Enhancements to the AV/C General Specification 3.0 Version 1.1' may be a best solution but it costs much to implement against the number of available devices. I want to avoid this huge mechanism if possible. Regards Takashi Sakamoto Takashi Sakamoto (8): oxfw: Add skelton for OXFW970/971 based devices oxfw: Read firmware version to name card oxfw: Add some AV/C commands for channel formation of AMDTP stream oxfw: Add connections and streams management oxfw: Add proc interface for debugging purpose oxfw: Add MIDI interface oxfw: Add PCM interface oxfw: Add hwdep interface include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 3 +- sound/firewire/Kconfig | 17 + sound/firewire/Makefile | 1 + sound/firewire/oxfw/Makefile | 3 + sound/firewire/oxfw/oxfw.c | 243 ++++++++++++++ sound/firewire/oxfw/oxfw.h | 140 ++++++++ sound/firewire/oxfw/oxfw_command.c | 161 ++++++++++ sound/firewire/oxfw/oxfw_hwdep.c | 197 ++++++++++++ sound/firewire/oxfw/oxfw_midi.c | 156 +++++++++ sound/firewire/oxfw/oxfw_pcm.c | 421 ++++++++++++++++++++++++ sound/firewire/oxfw/oxfw_proc.c | 61 ++++ sound/firewire/oxfw/oxfw_stream.c | 639 +++++++++++++++++++++++++++++++++++++ 13 files changed, 2043 insertions(+), 2 deletions(-) create mode 100644 sound/firewire/oxfw/Makefile create mode 100644 sound/firewire/oxfw/oxfw.c create mode 100644 sound/firewire/oxfw/oxfw.h create mode 100644 sound/firewire/oxfw/oxfw_command.c create mode 100644 sound/firewire/oxfw/oxfw_hwdep.c create mode 100644 sound/firewire/oxfw/oxfw_midi.c create mode 100644 sound/firewire/oxfw/oxfw_pcm.c create mode 100644 sound/firewire/oxfw/oxfw_proc.c create mode 100644 sound/firewire/oxfw/oxfw_stream.c -- 1.8.3.2

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[alsa-devel] Audio artefacts but no xruns
by Mark Hills 02 Feb '14

02 Feb '14

Hi, I have several applications for realtime processing of audio in/out, using small buffer sizes. I often find that I can get audible time-based distoritions, yet no xrun is reported, and the application is unaware. This problem is something I have, for a long time, taken for granted with ALSA. But it doesn't seem right and I would like to understand what's going on here. I am testing using different USB audio devices; snd-usb-audio and snd-usb-caiaq. Often using plughw or hw. Is this some fundamental limitation of USB, that it is unaware of problems nearer to the hardware? Or is this some bug where it's worth me narrowing down a specific test case and trying to fix it? Or some limitation of the use of intermediate processing like plughw? The effects range from occasional subtle clicks through ringing or noticable gaps in the audio, on record or playback. At very least I would like to guarantee a notification in userland that the integrity of the recording or playback is affected. This is particularly prevalent on, eg. Raspberry Pi. My other tests are on x86 platforms. Thanks -- Mark

1 0

Re: [alsa-devel] [FFADO-devel] Call for testing: ALSA driver for Fireworks/BeBoB based devices
by Maximilian Engelhardt 02 Feb '14

02 Feb '14

On Monday 14 October 2013 12:17:31 Takashi Sakamoto wrote: > Hi all, > > This is a call for testing my ALSA driver for Fireworks/BeBoB based devices. > > Please test 'snd-fireworks' for Fireworks and 'snd-bebob' for BeBoB if > you have some devices listed in the end of this mail. > > Status: > - still under development > - Without snd-dice and Clemens' development (I must do this later) > > Functionality: > - playback/capturing (full duplex) with PCM/MIDI interface > - hardware metering for some devices with CONTROL interface > - switching clock source/digital interface/digital mode with CONTROL > interface > - print hardware status with PROC interface > > Note: > - Don't use simultaneously 'ALSA PCM/MIDI playback/capture' and 'jackd > with Firewire (FFADO) backend'. Both of them try connecting to the > device when another is running. > - I add much modification into snd-firewire-lib for full duplex > synchronization of receive/transmit AMDTP stream. > > Requirement: > - Linux kernel 3.11 or later because of Juju (nickname of Firewire > stack) changing its API. > - Dynamic Kernel Module Support (DKMS) is reccomended for safely > installing/uninstalling > (I work with Ubuntu 13.10) > > Bug report: > - report with /proc/asound/cardX/#XXX > - please send your experiences to me with the output > > How to install (DKMS): > 1. $ git clone https://github.com/takaswie/snd-firewire-improve.git > 2. $ ln -s $(pwd)/snd-firewire-improve/ /usr/src/alsa-firewire-3.11 > (superuser) > 3. $ dkms install snd-firewire/3.11 (superuser) > > How to uninstall (DKMS): > 1. $ modprobe -r snd-bebob snd-fireworks snd-firewire-lib (superuser) > 2. $ dkms remove ans-firewire/3.11 --all (superuser) > 3. $ rm /usr/src/alsa-firewire-3.11 (superuser) > 4. $ rm snd-firewire-improve > > How to install (Manual): > 1. $ git clone https://github.com/takaswie/snd-firewire-improve.git > 2. $ cd snd-firewire-improve > 3. $ make > 4, backup system snd-firewire-lib/snd-firewire-speakers/snd-isight > (superuser) > 5. install > snd-firewire-lib/snd-firewire-speakers/snd-isight/snd-fireworks/snd-bebob > (superuser) 6. depmod -a (superuser) > > How to uninstall (Manual) > 1. modprobe -r snd-firewire-lib snd-firewire-speakers snd-isight > snd-fireworks snd-bebob (superuser) > 2. remove > snd-firewire-lib/snd-firewire-speakers/snd-isight/snd-fireworks/snd-bebob > (superuser) 3. recover snd-firewire-lib/snd-firewire-speakers/snd-isight > (superuser) 4. depmod -a (superuser) > > Confirmed to work: > - AudioFire4 > - AudioFirePre8 > - Ozonic > - Firewire Solo > - Firewire Audiophile > - Firewire 410 > > == Fireworks based devices > [Echo Audio] > AudioFire2 > AudioFire4 > AudioFirePre8 > AudioFire8 (till 2009) > AudioFire8 (since 2009) > AudioFire12 > > [Gibson] > RIP > > [Mackie] > Onyx 400F > Onyx 1200F > > == BeBoB based devices > [Yamaha] > GO44 > GO46 > > [M-Audio] > (to control mixer channels please use FFADO upstream) > Ozonic > Firewire 410 > Firewire Audiophile > Firewire Solo > NRV10 > ProFireLightbridge > > [Focusrite] > SaffirePro 26 I/O > SaffirePro 10 I/O > Saffire(LE) > > [Edirol] > FA-66 > FA-101 > > [TerraTecElectronic GmbH] > Phase88FW > PhaseX24FW > > [PreSonus] > FireBox > FirePod > > [Mackie] > OnyxFirewire > > [Tascam] > IF-FW/DM > > [Behringer] > X32 > > [ApogeeElectronics] > Rosetta200 > > [ESI] > Quatafire610 > > > Regards > > Takashi Sakamoto > o-takashi(a)sakamocchi.jp > > ---------------------------------------------------------------------------- > -- October Webinars: Code for Performance > Free Intel webinars can help you accelerate application performance. > Explore tips for MPI, OpenMP, advanced profiling, and more. Get the most > from the latest Intel processors and coprocessors. See abstracts and > register > > http://pubads.g.doubleclick.net/gampad/clk?id=60134071&iu=/4140/ostg.clktrk > _______________________________________________ > FFADO-devel mailing list > FFADO-devel(a)lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/ffado-devel Hi Takashi, Today I finally had some time to try your patches. I have a Terratec Phase 24FW Card. I did use the latest version (as of today) from https://github.com/takaswie/snd-firewire-improve.git on a Debian testing system. I did not use dkms but loaded the driver manually: $ cd sound/firewire $ modprobe -r snd-firewire-lib $ insmod ./snd-firewire-lib.ko $ insmod ./bebob/snd-bebob.ko First I have to say the driver does work and I can play music through the device, so thanks for your work. Unfortunately I managed to crash my system with it. I was using mplayer2 to play sound files and it seems that if I let mplayer2 play multiple files the system crashed somewhere between the end of the last file and the start of the new file (probably a race or something). I only did some quick testes because I hat to reboot after every crash (even sysrq doesn't work anymore), but I think the crash doesn't happen every time mplayer2 switches to the next file, but most of the time. If I let it play only one file I didn't observe any crashes. Here is the mplayer2 command I did use for testing: $ mplayer -ao alsa:device=hw=1 <some files> And here the output of /proc/asound/cardX/#XXX $ cat /proc/asound/card1/#clock Sampling rate: 48000 Clock Source: Internal $ cat /proc/asound/card1/#firmware Manufacturer: bridgeCo Protocol Ver: 1 Build Ver: 0 GUID: 0x000AAC04005F9C07 Model ID: 0x04 Model Rev: 1 Firmware Date: 20060420 Firmware Time: 105241 Firmware ID: 0x4 Firmware Ver: 18681119 Base Addr: 0x20080000 Max Size: 1572864 Loader Date: 20040719 Loader Time: 134002 $ cat /proc/asound/card1/#formation Output Stream from device: Rate PCM MIDI 22050 0 0 24000 0 0 32000 4 1 44100 4 1 48000 4 1 88200 4 1 96000 4 1 176400 0 0 192000 4 1 Input Stream to device: Rate PCM MIDI 22050 0 0 24000 0 0 32000 6 1 44100 6 1 48000 6 1 88200 6 1 96000 6 1 176400 0 0 192000 6 1 Attached is the crash log as far as I could obtain it. pulseaudio[4885]: [pulseaudio] module-alsa-card.c: Failed to find a working profile. pulseaudio[4885]: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="1" name="firewire-0x000aac04005f9c07" card_name="alsa_card.firewire-0x000aac04005f9c07" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes card_properties="module-udev-detect.discovered=1""): initialization failed. kernel: [ 334.172898] BUG: unable to handle kernel paging request at ffffffffffffffff kernel: [ 334.172934] IP: [<ffffffffa0154670>] fw_iso_context_queue_flush+0x10/0x10 [firewire_core] kernel: [ 334.172977] PGD 180f067 PUD 1811067 PMD 0 kernel: [ 334.172998] Oops: 0000 [#1] SMP kernel: [ 334.173015] Modules linked in: snd_bebob(O) snd_firewire_lib(O) sit ebtable_nat tunnel4 ip_tunnel ebtables cpufreq_stats cpufreq_userspace cpufreq_conservative parport_pc cpufreq_powersave ppdev lp ip6table_raw parport nf_conntrack_ipv6 nf_defrag_ipv6 ip6table_filter ip6_tables xt_REDIRECT iptable_nat nf_nat_ipv4 nf_nat xt_CT iptable_raw ipt_REJECT xt_LOG xt_limit xt_tcpudp nf_conntrack_ipv4 nf_defrag_ipv4 xt_conntrack nf_conntrack iptable_filter ip_tables x_tables binfmt_misc deflate ctr twofish_generic twofish_avx_x86_64 twofish_x86_64_3way twofish_x86_64 twofish_common camellia_generic camellia_x86_64 serpent_avx_x86_64 serpent_sse2_x86_64 serpent_generic xts blowfish_generic blowfish_x86_64 blowfish_common cast5_generic cast_common des_generic cbc cmac xcbc rmd160 sha512_generic sha256_generic crypto_null af_key xfrm_algo arc4 ecb md4 hmac nls_utf8 cifs rpcsec_gss_krb5 nfsv4 dns_resolver nfsd auth_rpcgss oid_registry nfs_acl nfs lockd fscache sunrpc w83627ehf hwmon_vid loop fuse ecryptfs isl6423 stv6110x stv090x snd_hda_codec_hdmi snd_hda_codec_realtek iTCO_wdt iTCO_vendor_support stv0299 rc_hauppauge mxm_wmi x86_pkg_temp_thermal intel_powerclamp coretemp kvm_intel kvm crc32_pclmul crc32c_intel ghash_clmulni_intel aesni_intel aes_x86_64 ablk_helper cryptd lrw gf128mul glue_helper psmouse snd_hda_intel snd_hda_codec snd_hwdep snd_pcm_oss snd_mixer_oss snd_pcm snd_page_alloc snd_seq_midi snd_seq_midi_event pcspkr snd_rawmidi budget budget_ci serio_raw budget_core ttpci_eeprom snd_seq i2c_i801 saa7146 snd_seq_device snd_timer rc_core dvb_core lpc_ich mfd_core snd soundcore evdev i915 drm_kms_helper wmi drm i2c_algo_bit i2c_core video button processor mei_me mei ext4 crc16 mbcache jbd2 btrfs xor raid6_pq crc32c libcrc32c dm_mod raid1 md_mod sg sd_mod sr_mod cdrom crc_t10dif crct10dif_common hid_generic usbhid hid ahci firewire_ohci libahci firewire_core crc_itu_t libata scsi_mod ehci_pci ehci_hcd e1000e xhci_hcd ptp pps_core usbcore usb_common thermal fan thermal_sys kernel: [ 334.173881] CPU: 0 PID: 5343 Comm: mplayer Tainted: G O 3.12-1-amd64 #1 Debian 3.12.6-2 kernel: [ 334.173913] Hardware name: /DZ77BH-55K, BIOS BHZ7710H.86A.0100.2013.0517.0942 05/17/2013 kernel: [ 334.173947] task: ffff88040da7a0c0 ti: ffff8803da2e4000 task.ti: ffff8803da2e4000 kernel: [ 334.173973] RIP: 0010:[<ffffffffa0154670>] [<ffffffffa0154670>] fw_iso_context_queue_flush+0x10/0x10 [firewire_core] kernel: [ 334.174015] RSP: 0018:ffff8803da2e5d70 EFLAGS: 00010002 kernel: [ 334.174035] RAX: ffffffffa0acd040 RBX: ffff8800c99c95c8 RCX: 0000000000001c2c kernel: [ 334.174060] RDX: ffff88040dcff000 RSI: 0000000000000000 RDI: ffffffffffffffff kernel: [ 334.174086] RBP: ffff8803da2e5de8 R08: 0000000000000000 R09: 00000000000031a1 kernel: [ 334.174111] R10: 0000000052ed7c8e R11: 0000000000000246 R12: 00007fffa996f808 kernel: [ 334.174137] R13: 00007fffa996f808 R14: 0000000000000000 R15: ffff8803f70a8c00 kernel: [ 334.174162] FS: 00007f286ae957c0(0000) GS:ffff88041f200000(0000) knlGS:0000000000000000 kernel: [ 334.174190] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 kernel: [ 334.174211] CR2: ffffffffffffffff CR3: 000000040c1d3000 CR4: 00000000001407f0 kernel: [ 334.174236] Stack: kernel: [ 334.174245] ffffffffa0abbca9 ffff8803da3a7000 ffffffffa0569bc4 0000000000000000 kernel: [ 334.174277] 0000000c00000000 0000000081019435 0000000000000097 ffffffffa03e321b kernel: [ 334.174309] ffff8803edca3558 ffff8803edca3530 ffff8803f70a8c00 00007fffa996f808 kernel: [ 334.174340] Call Trace: kernel: [ 334.174352] [<ffffffffa0abbca9>] ? amdtp_stream_pcm_pointer+0x29/0x40 [snd_firewire_lib] kernel: [ 334.174388] [<ffffffffa0569bc4>] ? snd_pcm_update_hw_ptr0+0x34/0x3a0 [snd_pcm] kernel: [ 334.174419] [<ffffffffa03e321b>] ? snd_timer_notify+0x6b/0x150 [snd_timer] kernel: [ 334.174448] [<ffffffffa056363a>] ? snd_pcm_delay+0x5a/0x140 [snd_pcm] kernel: [ 334.174476] [<ffffffffa0565623>] ? snd_pcm_common_ioctl1+0x5f3/0xc90 [snd_pcm] kernel: [ 334.174506] [<ffffffffa056a5f3>] ? snd_pcm_lib_write1+0x3a3/0x3f0 [snd_pcm] kernel: [ 334.174536] [<ffffffffa0567b80>] ? pcm_chmap_ctl_private_free+0x30/0x30 [snd_pcm] kernel: [ 334.174567] [<ffffffffa0565db0>] ? snd_pcm_playback_ioctl1+0xf0/0x240 [snd_pcm] kernel: [ 334.174597] [<ffffffffa0565f26>] ? snd_pcm_playback_ioctl+0x26/0x30 [snd_pcm] kernel: [ 334.174625] [<ffffffff81184d94>] ? do_vfs_ioctl+0x2d4/0x4b0 kernel: [ 334.174648] [<ffffffff81184ff0>] ? SyS_ioctl+0x80/0xa0 kernel: [ 334.174669] [<ffffffff81499b39>] ? system_call_fastpath+0x16/0x1b kernel: [ 334.174691] Code: 00 48 8b 40 60 ff e0 0f 1f 40 00 48 8b 07 48 8b 00 48 8b 40 70 ff e0 0f 1f 40 00 48 8b 07 48 8b 00 48 8b 40 78 ff e0 0f 1f 40 00 <48> 8b 07 48 8b 00 48 8b 80 80 00 00 00 ff e0 90 48 8b 07 48 8b kernel: [ 334.174857] RIP [<ffffffffa0154670>] fw_iso_context_queue_flush+0x10/0x10 [firewire_core] kernel: [ 334.174893] RSP <ffff8803da2e5d70> kernel: [ 334.174906] CR2: ffffffffffffffff kernel: [ 334.184706] ---[ end trace 4fd37a42001c8a5d ]--- kernel: [ 355.198207] INFO: rcu_sched detected stalls on CPUs/tasks: { 0} (detected by 2, t=5252 jiffies, g=3743, c=3742, q=2720) kernel: [ 355.198212] sending NMI to all CPUs: kernel: [ 355.198213] NMI backtrace for cpu 2 kernel: [ 355.198215] CPU: 2 PID: 3929 Comm: collectd Tainted: G D O 3.12-1-amd64 #1 Debian 3.12.6-2 kernel: [ 355.198216] Hardware name: /DZ77BH-55K, BIOS BHZ7710H.86A.0100.2013.0517.0942 05/17/2013 kernel: [ 355.198226] task: ffff88040bcad0c0 ti: ffff88040bf76000 task.ti: ffff88040bf76000 kernel: [ 355.198227] RIP: 0010:[<ffffffff81275bdd>] [<ffffffff81275bdd>] find_next_bit+0x4d/0xd0 kernel: [ 355.198231] RSP: 0018:ffff88041f303df0 EFLAGS: 00000046 kernel: [ 355.198231] RAX: 0000000000000000 RBX: ffff88041f30de00 RCX: 0000000000000001 kernel: [ 355.198232] RDX: ffff88041f30de38 RSI: 0000000000000080 RDI: 0000000000000180 kernel: [ 355.198233] RBP: ffff88041f20de40 R08: ffff88041f30de08 R09: 0000000000000008 kernel: [ 355.198234] R10: 0000000000000000 R11: ffff88041f303bb6 R12: 000000000000de80 kernel: [ 355.198234] R13: 0000000000080000 R14: 0000000000000055 R15: 0000000000000002 kernel: [ 355.198235] FS: 00007f8149847700(0000) GS:ffff88041f300000(0000) knlGS:0000000000000000 kernel: [ 355.198236] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 kernel: [ 355.198237] CR2: 00007f8157da9000 CR3: 000000040d165000 CR4: 00000000001407e0 kernel: [ 355.198238] Stack: kernel: [ 355.198239] ffffffff81043637 0000000000000008 000000000000de40 0000000200000002 kernel: [ 355.198241] 0000000000000082 0000000000002710 ffffffff818419c0 ffffffff818419c0 kernel: [ 355.198242] ffff88040bf76000 0000000000000aa0 ffff88041f30ed80 ffffffff8103f88e kernel: [ 355.198243] Call Trace: kernel: [ 355.198244] <IRQ> kernel: [ 355.198245] [<ffffffff81043637>] ? __x2apic_send_IPI_mask+0xb7/0x190 kernel: [ 355.198251] [<ffffffff8103f88e>] ? arch_trigger_all_cpu_backtrace+0x4e/0x80 kernel: [ 355.198253] [<ffffffff810e1b48>] ? rcu_check_callbacks+0x5b8/0x600 kernel: [ 355.198256] [<ffffffff810b4c30>] ? tick_sched_handle.isra.15+0x60/0x60 kernel: [ 355.198258] [<ffffffff8106881b>] ? update_process_times+0x3b/0x70 kernel: [ 355.198259] [<ffffffff810b4beb>] ? tick_sched_handle.isra.15+0x1b/0x60 kernel: [ 355.198260] [<ffffffff810b4c67>] ? tick_sched_timer+0x37/0x60 kernel: [ 355.198263] [<ffffffff8107ea22>] ? __run_hrtimer+0x62/0x1b0 kernel: [ 355.198264] [<ffffffff8107f1c9>] ? hrtimer_interrupt+0xe9/0x220 kernel: [ 355.198266] [<ffffffff8103deb6>] ? smp_apic_timer_interrupt+0x36/0x50 kernel: [ 355.198269] [<ffffffff8149a71d>] ? apic_timer_interrupt+0x6d/0x80 kernel: [ 355.198269] <EOI> kernel: [ 355.198270] [<ffffffff810ba548>] ? generic_exec_single+0x68/0x90 kernel: [ 355.198273] [<ffffffff810ba564>] ? generic_exec_single+0x84/0x90 kernel: [ 355.198275] [<ffffffff81287cb0>] ? ucs2_strncmp+0x60/0x60 kernel: [ 355.198277] [<ffffffff810ba655>] ? smp_call_function_single+0xe5/0x190 kernel: [ 355.198278] [<ffffffff81287cb0>] ? ucs2_strncmp+0x60/0x60 kernel: [ 355.198280] [<ffffffff81287dd6>] ? rdmsr_on_cpu+0x46/0x60 kernel: [ 355.198295] [<ffffffffa06561f3>] ? show_temp+0x93/0xd0 [coretemp] kernel: [ 355.198298] [<ffffffff81342157>] ? dev_attr_show+0x17/0x50 kernel: [ 355.198300] [<ffffffff811decc9>] ? sysfs_read_file+0x99/0x170 kernel: [ 355.198302] [<ffffffff81172f94>] ? vfs_read+0x94/0x160 kernel: [ 355.198304] [<ffffffff81173a83>] ? SyS_read+0x43/0xa0 kernel: [ 355.198306] [<ffffffff81499b39>] ? system_call_fastpath+0x16/0x1b kernel: [ 355.198306] Code: 3f 75 6b 48 f7 c6 c0 ff ff ff 0f 84 93 00 00 00 49 8b 00 49 8d 50 08 48 85 c0 74 0f eb 3f 48 83 c2 08 48 8b 42 f8 48 85 c0 75 32 <48> 83 ee 40 48 83 c7 40 48 f7 c6 c0 ff ff ff 75 e2 48 85 f6 48 kernel: [ 355.198321] NMI backtrace for cpu 0 kernel: [ 355.198324] CPU: 0 PID: 5343 Comm: mplayer Tainted: G D O 3.12-1-amd64 #1 Debian 3.12.6-2 kernel: [ 355.198324] Hardware name: /DZ77BH-55K, BIOS BHZ7710H.86A.0100.2013.0517.0942 05/17/2013 kernel: [ 355.198325] task: ffff88040da7a0c0 ti: ffff8803da2e4000 task.ti: ffff8803da2e4000 kernel: [ 355.198326] RIP: 0010:[<ffffffff81492083>] [<ffffffff81492083>] _raw_spin_lock+0x23/0x30 kernel: [ 355.198329] RSP: 0018:ffff8803da2e5a30 EFLAGS: 00000097 kernel: [ 355.198330] RAX: 0000000000001c2c RBX: ffff8803f70a8c00 RCX: 0000000000001c2d kernel: [ 355.198331] RDX: 0000000000001c2d RSI: ffff88040b1b5580 RDI: ffff8803f70a8d70 kernel: [ 355.198331] RBP: ffff8803da3a7000 R08: 0000000000000000 R09: 0000000000000000 kernel: [ 355.198332] R10: ffff88040b1b5590 R11: 0000000000000117 R12: ffff88040b1b5580 kernel: [ 355.198333] R13: ffff88040dca7e40 R14: ffff88040b029d98 R15: ffff8803d9156330 kernel: [ 355.198334] FS: 0000000000000000(0000) GS:ffff88041f200000(0000) knlGS:0000000000000000 kernel: [ 355.198335] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 kernel: [ 355.198335] CR2: ffffffffffffffff CR3: 000000000180c000 CR4: 00000000001407f0 kernel: [ 355.198336] Stack: kernel: [ 355.198337] ffffffffa0563cb8 ffff8803f70a8c00 ffff88040b029c00 ffffffffa0563d2a kernel: [ 355.198338] ffff8803f70a8c00 ffff88040b029c00 ffffffffa0563e68 ffff88040b1b5580 kernel: [ 355.198340] 0000000000000008 ffff8803d9156330 ffff8803d3d7e780 ffff88040e6656e0 kernel: [ 355.198341] Call Trace: kernel: [ 355.198351] [<ffffffffa0563cb8>] ? snd_pcm_drop+0x48/0xb0 [snd_pcm] kernel: [ 355.198355] [<ffffffffa0563d2a>] ? snd_pcm_release_substream.part.29+0xa/0x90 [snd_pcm] kernel: [ 355.198359] [<ffffffffa0563e68>] ? snd_pcm_release+0x98/0xb0 [snd_pcm] kernel: [ 355.198361] [<ffffffff81174a9a>] ? __fput+0xca/0x220 kernel: [ 355.198363] [<ffffffff81078904>] ? task_work_run+0xb4/0xd0 kernel: [ 355.198365] [<ffffffff8105d8c3>] ? do_exit+0x2b3/0xa40 kernel: [ 355.198367] [<ffffffff814932b8>] ? oops_end+0x98/0xe0 kernel: [ 355.198369] [<ffffffff81489540>] ? no_context+0x26a/0x276 kernel: [ 355.198371] [<ffffffff81495a0e>] ? __do_page_fault+0x2fe/0x500 kernel: [ 355.198373] [<ffffffff8111abea>] ? __alloc_pages_nodemask+0x14a/0xa30 kernel: [ 355.198376] [<ffffffff81127fcb>] ? shmem_find_get_pages_and_swap.part.18+0xcb/0x120 [some part missing...] Greetings, Maxi

1 0

[alsa-devel] unsubscribe
by animan9＠gmail.com 01 Feb '14

01 Feb '14

unsubscribe Sent from Windows Mail

1 0

[alsa-devel] [PATCH 1/4] ASoC: wm8962: Hold a runtime PM reference while handling interrupts
by Mark Brown 31 Jan '14

31 Jan '14

From: Mark Brown <broonie(a)linaro.org> If the device is runtime suspended then we can't interact with it as it may have been powered off and the register map will be in cache only mode. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/wm8962.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 97db3b45b411..aa7ae76d5e28 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3003,6 +3003,12 @@ static irqreturn_t wm8962_irq(int irq, void *data) unsigned int active; int reg, ret; + ret = pm_runtime_get_sync(dev); + if (ret < 0) { + dev_err(dev, "Failed to resume: %d\n", ret); + return ret; + } + ret = regmap_read(wm8962->regmap, WM8962_INTERRUPT_STATUS_2_MASK, &mask); if (ret != 0) { @@ -3019,8 +3025,10 @@ static irqreturn_t wm8962_irq(int irq, void *data) active &= ~mask; - if (!active) + if (!active) { + pm_runtime_put(dev); return IRQ_NONE; + } /* Acknowledge the interrupts */ ret = regmap_write(wm8962->regmap, WM8962_INTERRUPT_STATUS_2, active); @@ -3070,6 +3078,8 @@ static irqreturn_t wm8962_irq(int irq, void *data) msecs_to_jiffies(250)); } + pm_runtime_put(dev); + return IRQ_HANDLED; } -- 1.9.rc1

3 6

[alsa-devel] [PATCH 0/3] ASoC: davinci-mcasp/evm: PM related changes
by Peter Ujfalusi 31 Jan '14

31 Jan '14

Hi, In order to be able to benefit from the core's feature to handle the dai pinctrl default/sleep states the davinci-evm should set it's .pm section snd_soc_pm_ops. In the McASP driver consolidate the pm_runtime use to avoid unbalanced refcounting: trigger:start increased while trigger:stop did not decreased the pm counter. Move the pm callbacks from platfrom device to the soc_dai_driver. Now it is only a matter of dts update to enable the pin level power saving for McASP. Regards, Peter --- Peter Ujfalusi (3): ASoC: davinci-mcasp: Consolidate pm_runtime_get/put() use in the driver ASoC: davinci-evm: Add pm callbacks to platform driver ASoC: davinci-mcasp: Move pm callbacks from platform device to soc_dai_driver sound/soc/davinci/davinci-evm.c | 1 + sound/soc/davinci/davinci-mcasp.c | 97 +++++++++++++++++++-------------------- 2 files changed, 47 insertions(+), 51 deletions(-) -- 1.8.5.3

2 6

[alsa-devel] [PATCH] ALSA: hda/hdmi - allow PIN_OUT to be dynamically enabled
by Stephen Warren 31 Jan '14

31 Jan '14

From: Stephen Warren <swarren(a)nvidia.com> Commit 384a48d71520 "ALSA: hda: HDMI: Support codecs with fewer cvts than pins" dynamically enabled each pin widget's PIN_OUT only when the pin was actively in use. This was required on certain NVIDIA CODECs for correct operation. Specifically, if multiple pin widgets each had their mux input select the same audio converter widget and each pin widget had PIN_OUT enabled, then only one of the pin widgets would actually receive the audio, and often not the one the user wanted! However, this apparently broke some Intel systems, and commit 6169b673618b "ALSA: hda - Always turn on pins for HDMI/DP" reverted the dynamic setting of PIN_OUT. This in turn broke the afore-mentioned NVIDIA CODECs. This change supports either dynamic or static handling of PIN_OUT, selected by a flag set up during CODEC initialization. This flag is enabled for all recent NVIDIA GPUs. Reported-by: Uosis <uosisl(a)gmail.com> Cc: <stable(a)vger.kernel.org> # v3.13 Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- v3.12 or earlier stable require some manual back-porting effort, or perhaps just a variety of other commits to be backported. I'm investigating exactly which commits, and whether they're appropriate for stable. --- sound/pci/hda/patch_hdmi.c | 40 ++++++++++++++++++++++++++++++++++++---- 1 file changed, 36 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 64f0a5e73a25..5ef95034d041 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -132,6 +132,9 @@ struct hdmi_spec { struct hdmi_eld temp_eld; struct hdmi_ops ops; + + bool dyn_pin_out; + /* * Non-generic VIA/NVIDIA specific */ @@ -500,15 +503,25 @@ static void hdmi_write_dip_byte(struct hda_codec *codec, hda_nid_t pin_nid, static void hdmi_init_pin(struct hda_codec *codec, hda_nid_t pin_nid) { + struct hdmi_spec *spec = codec->spec; + int pin_out; + /* Unmute */ if (get_wcaps(codec, pin_nid) & AC_WCAP_OUT_AMP) snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); - /* Enable pin out: some machines with GM965 gets broken output when - * the pin is disabled or changed while using with HDMI - */ + + if (spec->dyn_pin_out) + /* Disable pin out until stream is active */ + pin_out = 0; + else + /* Enable pin out: some machines with GM965 gets broken output + * when the pin is disabled or changed while using with HDMI + */ + pin_out = PIN_OUT; + snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + AC_VERB_SET_PIN_WIDGET_CONTROL, pin_out); } static int hdmi_get_channel_count(struct hda_codec *codec, hda_nid_t cvt_nid) @@ -1735,6 +1748,7 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); hda_nid_t pin_nid = per_pin->pin_nid; bool non_pcm; + int pinctl; non_pcm = check_non_pcm_per_cvt(codec, cvt_nid); mutex_lock(&per_pin->lock); @@ -1744,6 +1758,14 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, hdmi_setup_audio_infoframe(codec, per_pin, non_pcm); mutex_unlock(&per_pin->lock); + if (spec->dyn_pin_out) { + pinctl = snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + pinctl | PIN_OUT); + } + return spec->ops.setup_stream(codec, cvt_nid, pin_nid, stream_tag, format); } @@ -1763,6 +1785,7 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, int cvt_idx, pin_idx; struct hdmi_spec_per_cvt *per_cvt; struct hdmi_spec_per_pin *per_pin; + int pinctl; if (hinfo->nid) { cvt_idx = cvt_nid_to_cvt_index(spec, hinfo->nid); @@ -1779,6 +1802,14 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, return -EINVAL; per_pin = get_pin(spec, pin_idx); + if (spec->dyn_pin_out) { + pinctl = snd_hda_codec_read(codec, per_pin->pin_nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + snd_hda_codec_write(codec, per_pin->pin_nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + pinctl & ~PIN_OUT); + } + snd_hda_spdif_ctls_unassign(codec, pin_idx); mutex_lock(&per_pin->lock); @@ -2840,6 +2871,7 @@ static int patch_nvhdmi(struct hda_codec *codec) return err; spec = codec->spec; + spec->dyn_pin_out = true; spec->ops.chmap_cea_alloc_validate_get_type = nvhdmi_chmap_cea_alloc_validate_get_type; -- 1.8.1.5

2 1

[alsa-devel] [PATCH v2 0/5] add a TDA998x CODEC
by Jean-Francois Moine 30 Jan '14

30 Jan '14

The TDA998x HDMI transmitter accepts audio input from either I2S or S/PDIF. Theses inputs have different intrinsic constraints and these constraints may be modified by the audio parameters of the connected video device. The choice of I2S or S/PDIF may be the done by the user or by automatic processing (DPCM?) at each audio starting time. This asks for a dynamic audio input switch in the HDMI driver. This patch series implements the TDA998x specific CODEC. A simple function call mechanism is used for exchanges between the CODEC and the HDMI driver. Note: the changes in the TDA998x I2C driver are based on my previous patch series: http://lists.freedesktop.org/archives/dri-devel/2014-January/052837.html - v2 - add ACLK setting and code optimization in patch 1 - from Mark Brown's remarks in patch 2: - don't compile the codec when CONFIG_ALL_CODECS - simplify the code about start/stop audio - fix coding style errors - add audio-port-names associated to audio-ports - add audio-port-names in patch 4 - add patch 5 'adjust the audio CTS_N pre-divider from audio format' for the Beaglebone-Black board (Jyri Sarha) Jean-Francois Moine (5): drm/i2c: tda998x: add a function for dynamic audio input switch ASoC: tda998x: add a codec driver for TDA998x ASoC: tda998x: add DT documentation ASoC: tda998x: adjust the audio hw parameters from EDID ASoC: tda998x: adjust the audio CTS_N pre-divider from audio format .../devicetree/bindings/sound/tda998x.txt | 16 + drivers/gpu/drm/i2c/tda998x_drv.c | 82 +++++- include/drm/i2c/tda998x.h | 11 +- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tda998x.c | 325 +++++++++++++++++++++ 6 files changed, 436 insertions(+), 6 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tda998x.txt create mode 100644 sound/soc/codecs/tda998x.c -- 1.9.rc1

2 6

[alsa-devel] [PATCH RFC v3 0/8] Beaglebone-Black HDMI audio
by Jyri Sarha 30 Jan '14

30 Jan '14

Changes since RFC v2 version of the patches: - Dropped out already applied: ASoC: hdmi-codec: Add devicetree binding with documentation - Addresses Mark's comments here: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-December/070605.h… - ASoC: davinci-evm: Add named clock reference to DT bindings - Get rid of unnecessary castings - Add mclk NULL checks - Use devm_clk_get() - Change clock name from "ti,codec-clock" to "mclk" - Address Mark's comments here: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-December/070606.h… - ASoC: davinci-evm: HDMI audio support for TDA998x trough McASP I2S bus - Get rid of unnecessary castings - Update commit message - Add: ASoC: davinci-mcasp: Set BCLK divider if McASP is BCLK master - Use snd_soc_params_to_bclk() Changes since the first RFC version of the patches: - Drop out already applied: ASoC: hdmi-codec: Add SNDRV_PCM_FMTBIT_32_LE playback format - Change sound node's compatible property form: "ti,am33xx-beaglebone-black" to "ti,am33xx-beaglebone-black-audio" - Some minor style issue fixes from TI internal review Jyri Sarha (8): clk: add gpio controlled clock ASoC: davinci-evm: Add named clock reference to DT bindings ASoC: davinci-mcasp: Set BCLK divider if McASP is BCLK master ASoC: davinci-evm: HDMI audio support for TDA998x trough McASP I2S bus ASoC: davinci: HDMI audio build for AM33XX and TDA998x drm/tilcdc: Add I2C HDMI audio config for tda998x ARM: OMAP2+: omap2plus_defconfig: Enable tilcdc and TDA998X HDMI support ARM: OMAP2+: omap2plus_defconfig: Enable BeagleBone Black HDMI audio support .../devicetree/bindings/clock/gpio-clock.txt | 21 ++ .../bindings/sound/davinci-evm-audio.txt | 13 +- arch/arm/configs/omap2plus_defconfig | 5 + drivers/clk/Makefile | 1 + drivers/clk/clk-gpio.c | 210 +++++++++++++++++++ drivers/gpu/drm/tilcdc/tilcdc_slave.c | 24 ++- include/linux/clk-provider.h | 25 +++ sound/soc/davinci/Kconfig | 12 ++ sound/soc/davinci/Makefile | 1 + sound/soc/davinci/davinci-evm.c | 211 +++++++++++++++++++- sound/soc/davinci/davinci-mcasp.c | 20 ++ 11 files changed, 534 insertions(+), 9 deletions(-) create mode 100644 Documentation/devicetree/bindings/clock/gpio-clock.txt create mode 100644 drivers/clk/clk-gpio.c -- 1.7.9.5

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[alsa-devel] [PATCH 00/11] HD-audio patches for 3.15
by Takashi Iwai 30 Jan '14

30 Jan '14

Hi, here is a series of patches for HD-audio for 3.15, i.e. no urgent fixes but mainly more enhancements and cleanups of static quirks. The patches are found in sound-unstable git tree topic/hda branch for now. They'll be merged to the main sound git tree once when the merge window is closed. Takashi

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[alsa-devel] [PATCH v4 01/19] ALSA: Oxygen: Add the separate SPI waiting function
by Roman Volkov 30 Jan '14

30 Jan '14

The oxygen_wait_spi() function now performs waiting when the SPI bus completes a transaction. Introduce the timeout error checking and increase timeout to 200 from 40. Signed-off-by: Roman Volkov <v1ron(a)mail.ru> --- sound/pci/oxygen/oxygen_io.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/pci/oxygen/oxygen_io.c b/sound/pci/oxygen/oxygen_io.c index 521eae4..471e64a 100644 --- a/sound/pci/oxygen/oxygen_io.c +++ b/sound/pci/oxygen/oxygen_io.c @@ -194,6 +194,21 @@ void oxygen_write_ac97_masked(struct oxygen *chip, unsigned int codec, } EXPORT_SYMBOL(oxygen_write_ac97_masked); +static int oxygen_wait_spi(struct oxygen *chip) +{ + /* Higher timeout to be sure: 200 us, + Actual transaction should not need more than 40 us */ + unsigned int count; + for (count = 50; count > 0; count--) { + udelay(4); + if ((oxygen_read8(chip, OXYGEN_SPI_CONTROL) & + OXYGEN_SPI_BUSY) == 0) + return 0; + } + snd_printk(KERN_ERR "Oxygen: SPI wait timeout\n"); + return -EIO; +} + void oxygen_write_spi(struct oxygen *chip, u8 control, unsigned int data) { unsigned int count; -- 1.7.10.4

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[alsa-devel] [PATCH 1/2] ASoC: Samsung: Remove invalid dependencies
by Sachin Kamat 30 Jan '14

30 Jan '14

These symbols got eliminated when non-DT support for Exynos was removed. Remove them. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/samsung/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 1f369c69469c..c006d4aafd5a 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -145,7 +145,7 @@ config SND_SOC_SAMSUNG_RX1950_UDA1380 config SND_SOC_SAMSUNG_SMDK_WM9713 tristate "SoC AC97 Audio support for SMDK with WM9713" - depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110 || MACH_SMDKV310 || MACH_SMDKC210) + depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110) select SND_SOC_WM9713 select SND_SAMSUNG_AC97 help -- 1.7.9.5

2 2

[alsa-devel] [PATCH] ALSA: hda - add headset mic detect quirks for another Dell laptop
by Hui Wang 30 Jan '14

30 Jan '14

When we plug a 3-ring headset on the Dell machine (Vendor ID: 0x10ec0255, Subsystem ID: 0x1028064d), the headset mic can't be detected, after apply this patch, the headset mic can work well. BugLink: https://bugs.launchpad.net/bugs/1260303 Cc: David Henningsson <david.henningsson(a)canonical.com> Tested-by: Doro Wu <fan-cheng.wu(a)canonical.com> Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d723c1f..56a8f18 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4294,6 +4294,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x063e, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x063f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0640, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x064d, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0651, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0652, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0653, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), -- 1.8.1.2

2 1

[alsa-devel] [PATCH 1/2] ASoC: fsl: imx-sgtl5000: No need to initialize local variable data
by Philipp Zabel 30 Jan '14

30 Jan '14

The first access to this variable is an assignment to the result of a kzalloc function call. Remove the superfluous initialisation. Signed-off-by: Philipp Zabel <p.zabel(a)pengutronix.de> --- sound/soc/fsl/imx-sgtl5000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index f2beae7..3e60b4c 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -62,7 +62,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) struct device_node *ssi_np, *codec_np; struct platform_device *ssi_pdev; struct i2c_client *codec_dev; - struct imx_sgtl5000_data *data = NULL; + struct imx_sgtl5000_data *data; int int_port, ext_port; int ret; -- 1.8.5.3

3 4

[alsa-devel] Regression due to 6169b673618b "ALSA: hda - Always turn on pins for HDMI/DP"
by Stephen Warren 30 Jan '14

30 Jan '14

Takashi, It looks like commit 6169b673618b "ALSA: hda - Always turn on pins for HDMI/DP" has caused a regression on NVIDIA graphics cards. It's been reported to us by at least two separate users. On NVIDIA GPUs, it appears that if multiple pin complex widgets have their input mux set to the same audio output widget and PIN_OUT, then only one of the pin complex widgets will actually receive the audio. That's the reason we need to dynamically toggle PIN_OUT on and off, so that only actually active pin complex widgets are enabled. Without this, audio will be routed to the same pin complex all the time, rather than the user-selected device. I'm not sure if this is the way HDA CODECs are meant to work, but it certainly is how they do work:-) So, I'd like to basically revert 6169b673618b "ALSA: hda - Always turn on pins for HDMI/DP", and return the code to the state it was after commit 384a48d71520 "ALSA: hda: HDMI: Support codecs with fewer cvts than pins". However, I obviously don't want to re-introduce the regression that your commit solved. I think we should solve this by including the code for both behaviours in the kernel, and having either a flag in struct hdmi_spec, or some extra ops in struct hdmi_ops, select between the two paths. Does that seem reasonable? If so, the only remaining question is: which behaviour should be default a/k/a should I blacklist the problematic GM965 (mentioned in https://bugzilla.kernel.org/show_bug.cgi?id=51421) against having the dynamic PIN_OUT behaviour, or whitelist NVIDIA GPUs as requiring the dynamic PIN_OUT behaviour? I know that my patch worked fine on my wife's Intel chipset, but I have no idea if the dynamic PIN_OUT behaviour was actually needed there, or simply tolerated, and I have no way to test much aside from NVIDIA GPUs. That all said, the dynamic PIN_OUT behaviour seems to make most sense to me, or some systems may end up with audio being sent to multiple outputs when only one was expected.

2 1

[alsa-devel] [PATCH v2 1/2] ASoC: imx-audmux: add RXFS/RXCLK defines for 6-wire connections
by Philipp Zabel 30 Jan '14

30 Jan '14

In asynchronous mode, a RxFS and RxClk connection needs to be made between two ports. Add a define for the bit to be set in the *SEL fields. Signed-off-by: Philipp Zabel <p.zabel(a)pengutronix.de> --- include/dt-bindings/sound/fsl-imx-audmux.h | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/include/dt-bindings/sound/fsl-imx-audmux.h b/include/dt-bindings/sound/fsl-imx-audmux.h index 50b09e9..63b48f8 100644 --- a/include/dt-bindings/sound/fsl-imx-audmux.h +++ b/include/dt-bindings/sound/fsl-imx-audmux.h @@ -24,6 +24,10 @@ #define MX51_AUDMUX_PORT6 5 #define MX51_AUDMUX_PORT7 6 +/* TFSEL/TCSEL/RFSEL/RCSEL can be sourced from Rx/Tx clocks on i.MX51/53/6Q */ +#define IMX_AUDMUX_RXFS 0x8 +#define IMX_AUDMUX_RXCLK 0x8 + /* Register definitions for the i.MX21/27 Digital Audio Multiplexer */ #define IMX_AUDMUX_V1_PCR_INMMASK(x) ((x) & 0xff) #define IMX_AUDMUX_V1_PCR_INMEN (1 << 8) -- 1.8.5.3

2 2

[alsa-devel] [PATCH 1/2] ASoC: imx-audmux: add RXFS/RXCLK defines for 6-wire connections
by Philipp Zabel 30 Jan '14

30 Jan '14

In asynchronous mode, a RxFS and RxClk connection needs to be made between two ports. Add a define for the bit to be set in the *SEL fields. Signed-off-by: Philipp Zabel <p.zabel(a)pengutronix.de> --- include/dt-bindings/sound/fsl-imx-audmux.h | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/include/dt-bindings/sound/fsl-imx-audmux.h b/include/dt-bindings/sound/fsl-imx-audmux.h index 50b09e9..63b48f8 100644 --- a/include/dt-bindings/sound/fsl-imx-audmux.h +++ b/include/dt-bindings/sound/fsl-imx-audmux.h @@ -24,6 +24,10 @@ #define MX51_AUDMUX_PORT6 5 #define MX51_AUDMUX_PORT7 6 +/* TFSEL/TCSEL/RFSEL/RCSEL can be sourced from Rx/Tx clocks on i.MX51/53/6Q */ +#define IMX_AUDMUX_RXFS 0x8 +#define IMX_AUDMUX_RXCLK 0x8 + /* Register definitions for the i.MX21/27 Digital Audio Multiplexer */ #define IMX_AUDMUX_V1_PCR_INMMASK(x) ((x) & 0xff) #define IMX_AUDMUX_V1_PCR_INMEN (1 << 8) -- 1.8.5.3

2 3

[alsa-devel] [PATCH 0/2] ASoC: core: volume control using signed register values
by Markus Pargmann 29 Jan '14

29 Jan '14

Hi, There are a lot of codecs using negative and positive volume control ranges as values in their registers. These two patches add support for arbitrary size signed integer values in registers. Regards, Markus Markus Pargmann (2): ASoC: core: Add signed register volume control logic ASoC: soc.h: Add SOC_DOUBLE_R_S_TLV include/sound/soc.h | 15 ++++++++++++ sound/soc/soc-core.c | 68 ++++++++++++++++++++++++++++++++++++++++++++++------ 2 files changed, 76 insertions(+), 7 deletions(-) -- 1.8.5.2

2 3

[alsa-devel] [RFC PATCH] ALSA: hda - Do not accept responses from non-existing codecs
by David Henningsson 29 Jan '14

29 Jan '14

While looking into some spurious responses, I found that the addr value was treated a bit inconsistent: values 8..0xf will be treated as codec 0 and values 0..7 will be treated as no error regardless of whether there is a codec there, or not. With this patch, all non-existing codecs will be treated equally. In addition, printing rp and wp could help figuring out if the wp value is reported wrongly from the controller or if something else is wrong. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_intel.c | 23 +++++++++-------------- 1 file changed, 9 insertions(+), 14 deletions(-) Note: this patch is untested (hence the RFC indication) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index fa2879a..e87eac5 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -834,18 +834,6 @@ static unsigned int azx_command_addr(u32 cmd) return addr; } -static unsigned int azx_response_addr(u32 res) -{ - unsigned int addr = res & 0xf; - - if (addr >= AZX_MAX_CODECS) { - snd_BUG(); - addr = 0; - } - - return addr; -} - /* send a command */ static int azx_corb_send_cmd(struct hda_bus *bus, u32 val) { @@ -907,8 +895,15 @@ static void azx_update_rirb(struct azx *chip) rp = chip->rirb.rp << 1; /* an RIRB entry is 8-bytes */ res_ex = le32_to_cpu(chip->rirb.buf[rp + 1]); res = le32_to_cpu(chip->rirb.buf[rp]); - addr = azx_response_addr(res_ex); - if (res_ex & ICH6_RIRB_EX_UNSOL_EV) + addr = res_ex & 0xf; + if ((addr >= AZX_MAX_CODECS) || !(chip->codec_mask & (1 << addr))) { + snd_printk(KERN_ERR SFX "%s: spurious response %#x:%#x, rp = %d, wp = %d", + pci_name(chip->pci), + res, res_ex, + chip->rirb.rp, wp); + snd_BUG(); + } + else if (res_ex & ICH6_RIRB_EX_UNSOL_EV) snd_hda_queue_unsol_event(chip->bus, res, res_ex); else if (chip->rirb.cmds[addr]) { chip->rirb.res[addr] = res; -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: hda - Add parameter for dumping processing coefficients
by David Henningsson 29 Jan '14

29 Jan '14

Processing coefficients are often a vital part of the codec's configuration, so dumping them can be important. However, because they are undocumented and secret, we do not want to enable this for all codecs by default. Therefore instead add this as a debugging parameter. I have prepared for codecs that want to enable this by default by the extra dump_coef bitfield, but unsure if we want to do that as long as the (unlikely, but still) race remains. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_codec.h | 1 + sound/pci/hda/hda_proc.c | 34 ++++++++++++++++++++++++++++++++-- 2 files changed, 33 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 2b5d19e..ab2a444 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -361,6 +361,7 @@ struct hda_codec { unsigned int epss:1; /* supporting EPSS? */ unsigned int cached_write:1; /* write only to caches */ unsigned int dp_mst:1; /* support DP1.2 Multi-stream transport */ + unsigned int dump_coef:1; /* dump processing coefs in codec proc file */ #ifdef CONFIG_PM unsigned int power_on :1; /* current (global) power-state */ unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index a8cb22e..ce5a6da 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -24,9 +24,14 @@ #include <linux/init.h> #include <linux/slab.h> #include <sound/core.h> +#include <linux/module.h> #include "hda_codec.h" #include "hda_local.h" +static int dump_coef = -1; +module_param(dump_coef, int, 0644); +MODULE_PARM_DESC(dump_coef, "Dump processing coefficients in codec proc file (-1=auto, 0=disable, 1=enable)"); + static char *bits_names(unsigned int bits, char *names[], int size) { int i, n; @@ -488,14 +493,39 @@ static void print_unsol_cap(struct snd_info_buffer *buffer, (unsol & AC_UNSOL_ENABLED) ? 1 : 0); } +static inline bool can_dump_coef(struct hda_codec *codec) +{ + switch (dump_coef) { + case 0: return false; + case 1: return true; + default: return codec->dump_coef; + } +} + static void print_proc_caps(struct snd_info_buffer *buffer, struct hda_codec *codec, hda_nid_t nid) { + unsigned int i, ncoeff, oldindex; unsigned int proc_caps = snd_hda_param_read(codec, nid, AC_PAR_PROC_CAP); + ncoeff = (proc_caps & AC_PCAP_NUM_COEF) >> AC_PCAP_NUM_COEF_SHIFT; snd_iprintf(buffer, " Processing caps: benign=%d, ncoeff=%d\n", - proc_caps & AC_PCAP_BENIGN, - (proc_caps & AC_PCAP_NUM_COEF) >> AC_PCAP_NUM_COEF_SHIFT); + proc_caps & AC_PCAP_BENIGN, ncoeff); + + if (!can_dump_coef(codec)) + return; + + /* Note: This is racy - another process could run in parallel and change + the coef index too. */ + oldindex = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_COEF_INDEX, 0); + for (i = 0; i < ncoeff; i++) { + unsigned int val; + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_COEF_INDEX, i); + val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PROC_COEF, + 0); + snd_iprintf(buffer, " Coeff 0x%02x: 0x%04x\n", i, val); + } + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_COEF_INDEX, oldindex); } static void print_conn_list(struct snd_info_buffer *buffer, -- 1.7.9.5

2 1

[alsa-devel] OSS virtual driver - dma buffer size problems
by Igor Kotrasiński 29 Jan '14

29 Jan '14

Resolved. Used snd_pcm_hw_constraint_minmax.

1 0

[alsa-devel] BockW 4th sound doesn't work on latest linus/master
by Kuninori Morimoto 29 Jan '14

29 Jan '14

Hi Simon, Mark I noticed that bockw 4th sound device doesn't work on latest linus/master. My git bisect said the trigger is this patch e14ee5deab24200e4b70fe31a8c806f0acd3d37c (ARM: shmobile: bockw: Do not enable CONFIG_DEVTMPFS defconfig) This patch removed below from defconfig -CONFIG_DEVTMPFS=y -CONFIG_DEVTMPFS_MOUNT=y But, why this has relationship to sound ?? #0 - #3 sound device have no problem. But #4 sound device will be broken by this patch Do you have any idea ? --- my kernel log ------- ... ALSA device list: #0: SSI56-AK4643 #1: SSI3-AK4554(playback) #2: SSI4-AK4554(capture) #3: SSI7-AK4554(playback) #4: SSI8-AK4554(capture) --- test result of 4th device (bad case) --- # /home/arecord -c 2 -t wav -d 10 -r 48000 -f S16_LE -D hw:4 /tmp/hoge ALSA lib pcm_hw.c:1433:(_snd_pcm_hw_open) Invalid value for card arecord: main:608: audio open error: No such file or directory --- test result of 4th device (good case) --- # /home/arecord -c 2 -t wav -d 10 -r 48000 -f S16_LE -D hw:4 /tmp/hoge Recording WAVE '/tmp/hoge' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo

2 4

[alsa-devel] ASoC updates for v3.14
by Mark Brown 29 Jan '14

29 Jan '14

The following changes since commit 7552f34a790069a008bd3e2ab4c0954b30c2f63b: Merge tag 'asoc-v3.14-3' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus (2014-01-20 15:48:55 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.14-rc1 for you to fetch changes up to e65523ca79fd087d9888598eadeb9554e2bd2102: Merge remote-tracking branches 'asoc/fix/arizona', 'asoc/fix/fsl', 'asoc/fix/omap', 'asoc/fix/samsung', 'asoc/fix/simple', 'asoc/fix/tlv320aic32x4' and 'asoc/fix/wm5100' into asoc-linus (2014-01-27 18:18:36 +0000) ---------------------------------------------------------------- ASoC: Fixes for v3.14 A few fixes, all in drivers. Nothing stands out particularly, the biggest set of fixes is some build coverage issues from Sachin. ---------------------------------------------------------------- Charles Keepax (1): ASoC: wm5110: Extend SYSCLK patch file for rev D Fabio Estevam (2): ASoC: fsl_ssi: We do support master mode now ASoC: fsl_ssi: Do not print 'baud clock' error message all the time Mark Brown (2): ASoC: omap: Make RX51 depend on GPIOLIB due to jack usage Merge remote-tracking branches 'asoc/fix/arizona', 'asoc/fix/fsl', 'asoc/fix/omap', 'asoc/fix/samsung', 'asoc/fix/simple', 'asoc/fix/tlv320aic32x4' and 'asoc/fix/wm5100' into asoc-linus Markus Pargmann (2): ASoC: tlv320aic32x4: Fix mono playback ASoC: tlv320aic32x4: Fix MICPGA input configuration Sachin Kamat (8): ASoC: samsung: Fix build regressions due to gpio re-org ASoC: Samsung: s3c-i2s-v2: Fix build error ASoC: samsung: smdk_wm8994: Fix build error ASoC: wm5100: Export wm5100_detect ASoC: samsung: Fix Kconfig dependency ASoC: Samsung: Fix build error due to missing dependency ASoC: samsung: Add NULL check in i2s.c ASoC: samsung: Remove dma.h inclusion Xiubo Li (1): ASoC: simple-card: fix simple card widgets routing property name usage .../devicetree/bindings/sound/simple-card.txt | 2 +- MAINTAINERS | 3 ++ Makefile | 2 +- arch/arm/kernel/devtree.c | 2 +- arch/arm/kernel/perf_event_cpu.c | 2 +- arch/arm/kernel/traps.c | 5 +- arch/arm/mach-highbank/highbank.c | 1 + arch/arm/mach-omap2/omap4-common.c | 1 + arch/arm/mm/init.c | 2 +- arch/arm/net/bpf_jit_32.c | 6 +-- arch/arm64/include/asm/io.h | 2 +- arch/mips/include/asm/cacheops.h | 2 +- arch/mips/include/asm/r4kcache.h | 51 +++++++++++---------- arch/mips/mm/c-r4k.c | 11 ++++- arch/parisc/include/uapi/asm/socket.h | 2 +- arch/powerpc/kernel/prom_init.c | 22 +++++---- arch/powerpc/net/bpf_jit_comp.c | 7 +-- arch/s390/net/bpf_jit_comp.c | 29 +++++++----- arch/sparc/net/bpf_jit_comp.c | 17 +++++-- arch/x86/include/asm/fpu-internal.h | 13 +++--- arch/x86/kernel/cpu/perf_event_amd_ibs.c | 53 ++++++++++++++++++---- arch/x86/kernel/entry_32.S | 4 +- arch/x86/kernel/entry_64.S | 2 +- arch/x86/kvm/lapic.c | 2 +- arch/x86/mm/fault.c | 18 ++++++++ arch/x86/net/bpf_jit_comp.c | 14 ++++-- arch/x86/vdso/vclock_gettime.c | 8 ++-- drivers/acpi/acpi_lpss.c | 1 - drivers/clocksource/cadence_ttc_timer.c | 21 +++++---- drivers/gpu/drm/drm_modes.c | 2 +- drivers/gpu/drm/i915/i915_irq.c | 2 + drivers/gpu/drm/i915/intel_ddi.c | 8 +++- drivers/gpu/drm/i915/intel_display.c | 4 +- drivers/gpu/drm/nouveau/core/include/subdev/i2c.h | 2 +- .../gpu/drm/nouveau/core/include/subdev/instmem.h | 7 +++ drivers/gpu/drm/nouveau/core/subdev/i2c/base.c | 4 +- drivers/gpu/drm/nouveau/core/subdev/mxm/nv50.c | 4 +- drivers/gpu/drm/nouveau/core/subdev/therm/ic.c | 10 ++-- drivers/gpu/drm/nouveau/dispnv04/dfp.c | 2 +- drivers/gpu/drm/nouveau/dispnv04/tvnv04.c | 2 +- drivers/hwmon/coretemp.c | 2 +- drivers/md/md.c | 18 ++++++-- drivers/md/md.h | 3 ++ drivers/md/raid1.c | 3 +- drivers/md/raid10.c | 12 ++--- drivers/md/raid5.c | 7 +-- drivers/net/ethernet/broadcom/bnx2x/bnx2x_main.c | 29 ++++++------ drivers/net/ethernet/chelsio/cxgb4/l2t.c | 2 +- drivers/net/ethernet/emulex/benet/be_main.c | 11 ++++- drivers/net/ethernet/intel/e1000e/netdev.c | 8 +--- drivers/net/ethernet/qlogic/qlge/qlge_main.c | 2 + drivers/net/ethernet/via/via-rhine.c | 1 + drivers/net/usb/dm9601.c | 12 +++++ drivers/net/usb/usbnet.c | 2 +- drivers/pinctrl/pinctrl-baytrail.c | 1 - fs/dcache.c | 7 ++- fs/fs-writeback.c | 15 +++--- fs/namespace.c | 2 +- fs/nilfs2/segment.c | 10 ++-- include/linux/crash_dump.h | 2 + include/linux/i2c.h | 2 +- include/linux/seqlock.h | 27 +++++++---- include/net/if_inet6.h | 1 - kernel/fork.c | 2 +- kernel/sched/fair.c | 2 +- kernel/time/sched_clock.c | 6 +-- lib/percpu_counter.c | 4 +- mm/util.c | 5 +- net/batman-adv/main.c | 2 +- net/core/filter.c | 30 +----------- net/ieee802154/nl-phy.c | 6 ++- net/ipv4/inet_diag.c | 5 +- net/ipv4/ipmr.c | 7 ++- net/ipv4/tcp_metrics.c | 51 +++++++++++++-------- net/ipv6/addrconf.c | 38 +++++++--------- net/ipv6/ip6mr.c | 7 ++- net/rds/ib_recv.c | 7 ++- sound/soc/codecs/tlv320aic32x4.c | 28 +++++++----- sound/soc/codecs/tlv320aic32x4.h | 3 ++ sound/soc/codecs/wm5100.c | 2 + sound/soc/codecs/wm5110.c | 48 ++++++++++++++++++++ sound/soc/fsl/fsl_ssi.c | 9 +--- sound/soc/omap/Kconfig | 1 + sound/soc/samsung/Kconfig | 4 +- sound/soc/samsung/h1940_uda1380.c | 1 + sound/soc/samsung/i2s.c | 5 +- sound/soc/samsung/neo1973_wm8753.c | 1 + sound/soc/samsung/rx1950_uda1380.c | 1 + sound/soc/samsung/s3c-i2s-v2.c | 6 +-- sound/soc/samsung/s3c2412-i2s.c | 2 + sound/soc/samsung/s3c24xx-i2s.c | 2 + sound/soc/samsung/smartq_wm8987.c | 1 + sound/soc/samsung/smdk_wm8994.c | 4 +- 93 files changed, 525 insertions(+), 296 deletions(-)

2 1

[alsa-devel] OSS virtual driver - dma buffer size problems
by Igor Kotrasiński 28 Jan '14

28 Jan '14

Hello, I'm trying to get the "cuckoo" OSS module (virtual ALSA driver) to work, and have some trouble properly setting up the dma buffer size. Relevant code (old type names are properly typedeffed from new ones) is: static snd_pcm_hardware_t snd_cuckoo_playback = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START), .channels_min = 1, .channels_max = 2, .buffer_bytes_max = (64 * 1024), .period_bytes_min = 64, .period_bytes_max = (32 * 1024), .periods_min = 2, .periods_max = 1024, (...) }; static snd_pcm_ops_t snd_cuckoo_playback_ops = { .open = snd_cuckoo_playback_open, .hw_params = snd_cuckoo_playback_hw_params, .prepare = snd_cuckoo_playback_prepare, (...) }; static int snd_cuckoo_playback_hw_params (snd_pcm_substream_t * substream, snd_pcm_hw_params_t * hw_params) { cuckoo_t *chip = snd_pcm_substream_chip (substream); snd_pcm_runtime_t *runtime = substream->runtime; dmap_t *dmap; //relevant buffer data here snd_dma_buffer_t *dmab; (...) if (runtime->dma_buffer_p) dmab = runtime->dma_buffer_p; else dmab = &(substream->dma_buffer); dmab->area = dmap->dmabuf; dmab->addr = dmap->dmabuf_phys; dmab->bytes = dmap -> buffsize; //supposed to be 4096 snd_pcm_set_runtime_buffer(substream, dmab); memset (dmap->dmabuf, 0, dmap->buffsize); return 0; } Inspecting values in snd_cuckoo_playback_prepare shows that runtime->dma_bytes is correctly set to 4096, but runtime->buffer_size is 16384 (with frame_bits == 32) and snd_pcm_lib_buffer_bytes(substream) returns 65536. If not aborted, it indeed overwrites the kernel. What is the correct way to pass a manually initialized dma buffer?

1 0

[alsa-devel] [PATCH 0/27]: ASoC: rsnd: add Renesas Gen2 DMAEngine support
by Kuninori Morimoto 28 Jan '14

28 Jan '14

Hi Mark These are Renesas Gen2 DMAEngine support patches. #1 - #26 are cleanup/prepare patches. #27 adds Gen2 DMAEngine support. These patches are based on asoc-v3.14-3 tag Kuninori Morimoto (27): 1) ASoC: rsnd: tidyup register naming of BUSIF_MODE 2) ASoC: rsnd: cleanup debug information method 3) ASoC: rsnd: remove meaningless rsnd_ssi_non() 4) ASoC: rsnd: control SCU ops in probe timing 5) ASoC: rsnd: add rsnd_scu_init(), and separate init/start 6) ASoC: rsnd: remove meaningless function parameter 7) ASoC: rsnd: merge rsnd_scu_start/stop() and rsnd_scu_transfer_start/stop() 8) ASoC: rsnd: rsnd_dai_is_clk_master() can be shared 9) ASoC: rsnd: remove pin sync option 10) ASoC: rsnd: SSI_MODE0/1 settings goes to scu.c 11) ASoC: rsnd: remove ssiu from ssi.c 12) ASoC: rsnd: rename rsnd_scu_convert_rate_ctrl() 13) ASoC: rsnd: explain SRC bypass mode settings in comment 14) ASoC: rsnd: remove duplicate *priv from rsnd_dma 15) ASoC: rsnd: non 0 is error on probe 16) ASoC: rsnd: fixup Gen2 module naming 17) ASoC: rsnd: don't use schedule_work() when rsnd_dma_start() 18) ASoC: rsnd: SCU should be called before SSI 19) ASoC: rsnd: clarify scu.c area 20) ASoC: rsnd: Merge macros in scu.c 21) ASoC: rsnd: rsnd_scu_hpbif_is_enable() become macro 22) ASoC: rsnd: merge SRC clock timing/setting 23) ASoC: rsnd: add rsnd_ssi_is_play() 24) ASoC: rsnd: extracts Gen1/Gen2 common parts 25) ASoC: rsnd: remove SSI dependent DMAEngine callback 26) ASoC: rsnd: rsnd_ssi_probe() goes forwarder than rsnd_scu_probe() 27) ASoC: rsnd: add Gen2 SRC and DMAEngine support include/sound/rcar_snd.h | 7 +- sound/soc/sh/rcar/adg.c | 158 ++++++++++-- sound/soc/sh/rcar/core.c | 70 ++++-- sound/soc/sh/rcar/gen.c | 38 ++- sound/soc/sh/rcar/rsnd.h | 66 +++-- sound/soc/sh/rcar/scu.c | 605 ++++++++++++++++++++++++++++++++++------------ sound/soc/sh/rcar/ssi.c | 222 +++++------------ 7 files changed, 779 insertions(+), 387 deletions(-)) Best regards --- Kuninori Morimoto

5 41

[alsa-devel] [PATCH 0/4] add a TDA998x CODEC
by Jean-Francois Moine 28 Jan '14

28 Jan '14

The TDA998x HDMI transmitter accepts audio input from either I2S or S/PDIF. Theses inputs have different intrinsic constraints and these constraints may be modified by the audio parameters of the connected video device. The choice of I2S or S/PDIF may be the done by the user or by automatic processing (DPCM?) at each audio starting time. This asks for a dynamic audio input switch in the HDMI driver. This patch series implements the TDA998x specific CODEC. A simple function call mechanism is used for exchanges between the CODEC and the HDMI driver. Jean-Francois Moine (4): drm/i2c: tda998x: add a function for dynamic audio input switch ASoC: tda998x: add a codec driver for TDA998x ASoC: tda998x: add DT documentation ASoC: tda998x: adjust the audio hw parameters from EDID .../devicetree/bindings/sound/tda998x.txt | 14 + drivers/gpu/drm/i2c/tda998x_drv.c | 58 +++- include/drm/i2c/tda998x.h | 8 +- sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tda998x.c | 304 +++++++++++++++++++++ 6 files changed, 390 insertions(+), 3 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tda998x.txt create mode 100644 sound/soc/codecs/tda998x.c -- 1.8.5.3

4 14

[alsa-devel] Fwd: [LAC-team] LAC 2014: the Linux Audio Conference - Deadline Extension
by Jörn Nettingsmeier 28 Jan '14

28 Jan '14

FYI -------- Original Message -------- Subject: [LAC-team] LAC 2014: the Linux Audio Conference - Deadline Extension Date: Mon, 27 Jan 2014 20:02:53 +0100 From: Götz Dipper <dipper(a)zkm.de> To: LAC-team(a)lists.linuxaudio.org sorry for >< please >> << Hi all, The Linux Audio Conference submissions deadline has been extended! It is now February 3rd, 2014 (23:59 HAST) So, if you were considering to submit a paper but couldn't make up your mind yet, here is your chance to become active! Never forget that this conference lives through the people participating in it. February 3rd is the new deadline for all submission types: papers, music, installations, workshop proposals. Check out the link below for more info: http://lac.linuxaudio.org/2014/participation Please spread this information to anyone who might be interested. If you have any questions, drop us a line at lac(a)linuxaudio.org We are looking forward to seeing you in Karlsruhe in May! Thanks, The LAC2014 organization team _______________________________________________ LAC-team mailing list LAC-team(a)lists.linuxaudio.org http://lists.linuxaudio.org/listinfo/lac-team

1 0

[alsa-devel] [PATCH 0/3] ASoC: atmel_ssc_dai: add option to choose clock
by Bo Shen 27 Jan '14

27 Jan '14

When SSC work in slave mode, the clock can come from TK pin and also can come from RK pin, this is hardware design decided. So, make it available to choose where the clock from. Bo Shen (3): ASoC: atmel_ssc_dai: make option to choose clock ASoC: atmel_wm8904: make it available to choose clock Binding: atmel-wm8904: add option to choose clock Documentation/devicetree/bindings/sound/atmel-wm8904.txt | 2 ++ sound/soc/atmel/atmel_ssc_dai.c | 16 ++++++++++++---- sound/soc/atmel/atmel_ssc_dai.h | 1 + sound/soc/atmel/atmel_wm8904.c | 10 ++++++++++ 4 files changed, 25 insertions(+), 4 deletions(-) -- 1.8.5.2

4 12

[alsa-devel] [PATCH] ASoC: fsl-sai: convert to use regmap API for Freeacale SAI
by Xiubo Li 27 Jan '14

27 Jan '14

Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> Cc: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 243 +++++++++++++++++++++++++----------------------- sound/soc/fsl/fsl_sai.h | 47 +++++----- 2 files changed, 150 insertions(+), 140 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 2858a6c..a80d709 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -15,6 +15,7 @@ #include <linux/dmaengine.h> #include <linux/module.h> #include <linux/of_address.h> +#include <linux/regmap.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/dmaengine_pcm.h> @@ -22,34 +23,6 @@ #include "fsl_sai.h" -static inline u32 sai_readl(struct fsl_sai *sai, - const void __iomem *addr) -{ - u32 val; - - val = __raw_readl(addr); - - if (likely(sai->big_endian_regs)) - val = be32_to_cpu(val); - else - val = le32_to_cpu(val); - rmb(); - - return val; -} - -static inline void sai_writel(struct fsl_sai *sai, - u32 val, void __iomem *addr) -{ - wmb(); - if (likely(sai->big_endian_regs)) - val = cpu_to_be32(val); - else - val = cpu_to_le32(val); - - __raw_writel(val, addr); -} - static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int fsl_dir) { @@ -61,7 +34,8 @@ static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, else reg_cr2 = FSL_SAI_RCR2; - val_cr2 = sai_readl(sai, sai->base + reg_cr2); + regmap_read(sai->regmap, reg_cr2, &val_cr2); + val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK; switch (clk_id) { @@ -81,40 +55,28 @@ static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, return -EINVAL; } - sai_writel(sai, val_cr2, sai->base + reg_cr2); - - return 0; + return regmap_write(sai->regmap, reg_cr2, val_cr2); } static int fsl_sai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { - struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); int ret; if (dir == SND_SOC_CLOCK_IN) return 0; - ret = clk_prepare_enable(sai->clk); - if (ret) - return ret; - ret = fsl_sai_set_dai_sysclk_tr(cpu_dai, clk_id, freq, FSL_FMT_TRANSMITTER); if (ret) { dev_err(cpu_dai->dev, "Cannot set tx sysclk: %d\n", ret); - goto err_clk; + return ret; } ret = fsl_sai_set_dai_sysclk_tr(cpu_dai, clk_id, freq, FSL_FMT_RECEIVER); - if (ret) { + if (ret) dev_err(cpu_dai->dev, "Cannot set rx sysclk: %d\n", ret); - goto err_clk; - } - -err_clk: - clk_disable_unprepare(sai->clk); return ret; } @@ -133,8 +95,8 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, reg_cr4 = FSL_SAI_RCR4; } - val_cr2 = sai_readl(sai, sai->base + reg_cr2); - val_cr4 = sai_readl(sai, sai->base + reg_cr4); + regmap_read(sai->regmap, reg_cr2, &val_cr2); + regmap_read(sai->regmap, reg_cr4, &val_cr4); if (sai->big_endian_data) val_cr4 &= ~FSL_SAI_CR4_MF; @@ -183,35 +145,25 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, return -EINVAL; } - sai_writel(sai, val_cr2, sai->base + reg_cr2); - sai_writel(sai, val_cr4, sai->base + reg_cr4); + regmap_write(sai->regmap, reg_cr2, val_cr2); + regmap_write(sai->regmap, reg_cr4, val_cr4); return 0; } static int fsl_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { - struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); int ret; - ret = clk_prepare_enable(sai->clk); - if (ret) - return ret; - ret = fsl_sai_set_dai_fmt_tr(cpu_dai, fmt, FSL_FMT_TRANSMITTER); if (ret) { dev_err(cpu_dai->dev, "Cannot set tx format: %d\n", ret); - goto err_clk; + return ret; } ret = fsl_sai_set_dai_fmt_tr(cpu_dai, fmt, FSL_FMT_RECEIVER); - if (ret) { + if (ret) dev_err(cpu_dai->dev, "Cannot set rx format: %d\n", ret); - goto err_clk; - } - -err_clk: - clk_disable_unprepare(sai->clk); return ret; } @@ -235,11 +187,12 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, reg_mr = FSL_SAI_RMR; } - val_cr4 = sai_readl(sai, sai->base + reg_cr4); + regmap_read(sai->regmap, reg_cr4, &val_cr4); + regmap_read(sai->regmap, reg_cr4, &val_cr5); + val_cr4 &= ~FSL_SAI_CR4_SYWD_MASK; val_cr4 &= ~FSL_SAI_CR4_FRSZ_MASK; - val_cr5 = sai_readl(sai, sai->base + reg_cr5); val_cr5 &= ~FSL_SAI_CR5_WNW_MASK; val_cr5 &= ~FSL_SAI_CR5_W0W_MASK; val_cr5 &= ~FSL_SAI_CR5_FBT_MASK; @@ -257,9 +210,9 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, val_cr4 |= FSL_SAI_CR4_FRSZ(channels); val_mr = ~0UL - ((1 << channels) - 1); - sai_writel(sai, val_cr4, sai->base + reg_cr4); - sai_writel(sai, val_cr5, sai->base + reg_cr5); - sai_writel(sai, val_mr, sai->base + reg_mr); + regmap_write(sai->regmap, reg_cr4, val_cr4); + regmap_write(sai->regmap, reg_cr5, val_cr5); + regmap_write(sai->regmap, reg_mr, val_mr); return 0; } @@ -268,18 +221,15 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); - u32 tcsr, rcsr, val_cr2; - - val_cr2 = sai_readl(sai, sai->base + FSL_SAI_TCR2); - val_cr2 &= ~FSL_SAI_CR2_SYNC; - sai_writel(sai, val_cr2, sai->base + FSL_SAI_TCR2); + u32 tcsr, rcsr; - val_cr2 = sai_readl(sai, sai->base + FSL_SAI_RCR2); - val_cr2 |= FSL_SAI_CR2_SYNC; - sai_writel(sai, val_cr2, sai->base + FSL_SAI_RCR2); + regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC, + ~FSL_SAI_CR2_SYNC); + regmap_update_bits(sai->regmap, FSL_SAI_RCR2, FSL_SAI_CR2_SYNC, + FSL_SAI_CR2_SYNC); - tcsr = sai_readl(sai, sai->base + FSL_SAI_TCSR); - rcsr = sai_readl(sai, sai->base + FSL_SAI_RCSR); + regmap_read(sai->regmap, FSL_SAI_TCSR, &tcsr); + regmap_read(sai->regmap, FSL_SAI_RCSR, &rcsr); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { tcsr |= FSL_SAI_CSR_FRDE; @@ -296,10 +246,9 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, tcsr |= FSL_SAI_CSR_TERE; rcsr |= FSL_SAI_CSR_TERE; - sai_writel(sai, rcsr, sai->base + FSL_SAI_RCSR); - sai_writel(sai, tcsr, sai->base + FSL_SAI_TCSR); + regmap_write(sai->regmap, FSL_SAI_RCSR, rcsr); + regmap_write(sai->regmap, FSL_SAI_TCSR, tcsr); break; - case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: @@ -308,8 +257,8 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, rcsr &= ~FSL_SAI_CSR_TERE; } - sai_writel(sai, tcsr, sai->base + FSL_SAI_TCSR); - sai_writel(sai, rcsr, sai->base + FSL_SAI_RCSR); + regmap_write(sai->regmap, FSL_SAI_TCSR, tcsr); + regmap_write(sai->regmap, FSL_SAI_RCSR, rcsr); break; default: return -EINVAL; @@ -322,41 +271,32 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); - u32 val, reg; - int ret; - - ret = clk_prepare_enable(sai->clk); - if (ret) - return ret; + u32 reg; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) reg = FSL_SAI_TCR3; else reg = FSL_SAI_RCR3; - val = sai_readl(sai, sai->base + reg); - val |= FSL_SAI_CR3_TRCE; - sai_writel(sai, val, sai->base + reg); + regmap_update_bits(sai->regmap, reg, FSL_SAI_CR3_TRCE, + FSL_SAI_CR3_TRCE); - return clk_prepare_enable(sai->clk); + return 0; } static void fsl_sai_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); - u32 val, reg; + u32 reg; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) reg = FSL_SAI_TCR3; else reg = FSL_SAI_RCR3; - val = sai_readl(sai, sai->base + reg); - val &= ~FSL_SAI_CR3_TRCE; - sai_writel(sai, val, sai->base + reg); - - clk_disable_unprepare(sai->clk); + regmap_update_bits(sai->regmap, reg, FSL_SAI_CR3_TRCE, + ~FSL_SAI_CR3_TRCE); } static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { @@ -371,18 +311,13 @@ static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev); - int ret; - - ret = clk_prepare_enable(sai->clk); - if (ret) - return ret; - sai_writel(sai, 0x0, sai->base + FSL_SAI_RCSR); - sai_writel(sai, 0x0, sai->base + FSL_SAI_TCSR); - sai_writel(sai, FSL_SAI_MAXBURST_TX * 2, sai->base + FSL_SAI_TCR1); - sai_writel(sai, FSL_SAI_MAXBURST_RX - 1, sai->base + FSL_SAI_RCR1); - - clk_disable_unprepare(sai->clk); + regmap_update_bits(sai->regmap, FSL_SAI_TCSR, 0xffffffff, 0x0); + regmap_update_bits(sai->regmap, FSL_SAI_RCSR, 0xffffffff, 0x0); + regmap_update_bits(sai->regmap, FSL_SAI_TCR1, FSL_SAI_CR1_RFW_MASK, + FSL_SAI_MAXBURST_TX * 2); + regmap_update_bits(sai->regmap, FSL_SAI_RCR1, FSL_SAI_CR1_RFW_MASK, + FSL_SAI_MAXBURST_RX - 1); snd_soc_dai_init_dma_data(cpu_dai, &sai->dma_params_tx, &sai->dma_params_rx); @@ -413,11 +348,88 @@ static const struct snd_soc_component_driver fsl_component = { .name = "fsl-sai", }; +/* Freescale SAI regmap */ +static bool fsl_sai_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case FSL_SAI_TCSR: + case FSL_SAI_TCR1: + case FSL_SAI_TCR2: + case FSL_SAI_TCR3: + case FSL_SAI_TCR4: + case FSL_SAI_TCR5: + case FSL_SAI_TFR: + case FSL_SAI_TMR: + case FSL_SAI_RCSR: + case FSL_SAI_RCR1: + case FSL_SAI_RCR2: + case FSL_SAI_RCR3: + case FSL_SAI_RCR4: + case FSL_SAI_RCR5: + case FSL_SAI_RDR: + case FSL_SAI_RFR: + case FSL_SAI_RMR: + return true; + default: + return false; + } +} + +static bool fsl_sai_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case FSL_SAI_TFR: + case FSL_SAI_RFR: + case FSL_SAI_TDR: + case FSL_SAI_RDR: + return true; + default: + return false; + } + +} + +static bool fsl_sai_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case FSL_SAI_TCSR: + case FSL_SAI_TCR1: + case FSL_SAI_TCR2: + case FSL_SAI_TCR3: + case FSL_SAI_TCR4: + case FSL_SAI_TCR5: + case FSL_SAI_TDR: + case FSL_SAI_TMR: + case FSL_SAI_RCSR: + case FSL_SAI_RCR1: + case FSL_SAI_RCR2: + case FSL_SAI_RCR3: + case FSL_SAI_RCR4: + case FSL_SAI_RCR5: + case FSL_SAI_RMR: + return true; + default: + return false; + } +} + +static const struct regmap_config fsl_sai_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + + .max_register = FSL_SAI_RMR, + .readable_reg = fsl_sai_readable_reg, + .volatile_reg = fsl_sai_volatile_reg, + .writeable_reg = fsl_sai_writeable_reg, +}; + static int fsl_sai_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; struct fsl_sai *sai; struct resource *res; + void __iomem *base; int ret; sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL); @@ -425,14 +437,15 @@ static int fsl_sai_probe(struct platform_device *pdev) return -ENOMEM; res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - sai->base = devm_ioremap_resource(&pdev->dev, res); - if (IS_ERR(sai->base)) - return PTR_ERR(sai->base); - - sai->clk = devm_clk_get(&pdev->dev, "sai"); - if (IS_ERR(sai->clk)) { - dev_err(&pdev->dev, "Cannot get SAI's clock\n"); - return PTR_ERR(sai->clk); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) + return PTR_ERR(base); + + sai->regmap = devm_regmap_init_mmio_clk(&pdev->dev, + "sai", base, &fsl_sai_regmap_config); + if (IS_ERR(sai->regmap)) { + dev_err(&pdev->dev, "regmap init failed\n"); + return PTR_ERR(sai->regmap); } sai->dma_params_rx.addr = res->start + FSL_SAI_RDR; diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 41bb62e..1571459 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -15,31 +15,36 @@ SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) +/* SAI Register Map Register */ +#define FSL_SAI_TCSR 0x00 /* SAI Transmit Control */ +#define FSL_SAI_TCR1 0x04 /* SAI Transmit Configuration 1 */ +#define FSL_SAI_TCR2 0x08 /* SAI Transmit Configuration 2 */ +#define FSL_SAI_TCR3 0x0c /* SAI Transmit Configuration 3 */ +#define FSL_SAI_TCR4 0x10 /* SAI Transmit Configuration 4 */ +#define FSL_SAI_TCR5 0x14 /* SAI Transmit Configuration 5 */ +#define FSL_SAI_TDR 0x20 /* SAI Transmit Data */ +#define FSL_SAI_TFR 0x40 /* SAI Transmit FIFO */ +#define FSL_SAI_TMR 0x60 /* SAI Transmit Mask */ +#define FSL_SAI_RCSR 0x80 /* SAI Receive Control */ +#define FSL_SAI_RCR1 0x84 /* SAI Receive Configuration 1 */ +#define FSL_SAI_RCR2 0x88 /* SAI Receive Configuration 2 */ +#define FSL_SAI_RCR3 0x8c /* SAI Receive Configuration 3 */ +#define FSL_SAI_RCR4 0x90 /* SAI Receive Configuration 4 */ +#define FSL_SAI_RCR5 0x94 /* SAI Receive Configuration 5 */ +#define FSL_SAI_RDR 0xa0 /* SAI Receive Data */ +#define FSL_SAI_RFR 0xc0 /* SAI Receive FIFO */ +#define FSL_SAI_RMR 0xe0 /* SAI Receive Mask */ + /* SAI Transmit/Recieve Control Register */ -#define FSL_SAI_TCSR 0x00 -#define FSL_SAI_RCSR 0x80 #define FSL_SAI_CSR_TERE BIT(31) #define FSL_SAI_CSR_FWF BIT(17) #define FSL_SAI_CSR_FRIE BIT(8) #define FSL_SAI_CSR_FRDE BIT(0) -/* SAI Transmit Data/FIFO/MASK Register */ -#define FSL_SAI_TDR 0x20 -#define FSL_SAI_TFR 0x40 -#define FSL_SAI_TMR 0x60 - -/* SAI Recieve Data/FIFO/MASK Register */ -#define FSL_SAI_RDR 0xa0 -#define FSL_SAI_RFR 0xc0 -#define FSL_SAI_RMR 0xe0 - /* SAI Transmit and Recieve Configuration 1 Register */ -#define FSL_SAI_TCR1 0x04 -#define FSL_SAI_RCR1 0x84 +#define FSL_SAI_CR1_RFW_MASK 0x1f /* SAI Transmit and Recieve Configuration 2 Register */ -#define FSL_SAI_TCR2 0x08 -#define FSL_SAI_RCR2 0x88 #define FSL_SAI_CR2_SYNC BIT(30) #define FSL_SAI_CR2_MSEL_MASK (0xff << 26) #define FSL_SAI_CR2_MSEL_BUS 0 @@ -50,15 +55,11 @@ #define FSL_SAI_CR2_BCD_MSTR BIT(24) /* SAI Transmit and Recieve Configuration 3 Register */ -#define FSL_SAI_TCR3 0x0c -#define FSL_SAI_RCR3 0x8c #define FSL_SAI_CR3_TRCE BIT(16) #define FSL_SAI_CR3_WDFL(x) (x) #define FSL_SAI_CR3_WDFL_MASK 0x1f /* SAI Transmit and Recieve Configuration 4 Register */ -#define FSL_SAI_TCR4 0x10 -#define FSL_SAI_RCR4 0x90 #define FSL_SAI_CR4_FRSZ(x) (((x) - 1) << 16) #define FSL_SAI_CR4_FRSZ_MASK (0x1f << 16) #define FSL_SAI_CR4_SYWD(x) (((x) - 1) << 8) @@ -69,8 +70,6 @@ #define FSL_SAI_CR4_FSD_MSTR BIT(0) /* SAI Transmit and Recieve Configuration 5 Register */ -#define FSL_SAI_TCR5 0x14 -#define FSL_SAI_RCR5 0x94 #define FSL_SAI_CR5_WNW(x) (((x) - 1) << 24) #define FSL_SAI_CR5_WNW_MASK (0x1f << 24) #define FSL_SAI_CR5_W0W(x) (((x) - 1) << 16) @@ -100,9 +99,7 @@ #define FSL_SAI_MAXBURST_RX 6 struct fsl_sai { - struct clk *clk; - - void __iomem *base; + struct regmap *regmap; bool big_endian_regs; bool big_endian_data; -- 1.8.4

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[alsa-devel] [PATCH v2 0/3] ASoC: atmel_ssc_dai: add option to choose clock
by Bo Shen 27 Jan '14

27 Jan '14

When SSC work in slave mode, the clock can come from TK pin and also can come from RK pin, this is hardware design decided. So, make it available to choose where the clock from. Changes in v2: - using "-" replace "_" in binding document Bo Shen (3): ASoC: atmel_ssc_dai: make option to choose clock ASoC: atmel_wm8904: make it available to choose clock Binding: atmel-wm8904: add option to choose clock Documentation/devicetree/bindings/sound/atmel-wm8904.txt | 6 ++++++ sound/soc/atmel/atmel_ssc_dai.c | 16 ++++++++++++---- sound/soc/atmel/atmel_ssc_dai.h | 1 + sound/soc/atmel/atmel_wm8904.c | 10 ++++++++++ 4 files changed, 29 insertions(+), 4 deletions(-) -- 1.8.5.2

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[alsa-devel] [PATCH] ASoC: arizona: Check clocking during hw_params rather than startup
by Charles Keepax 27 Jan '14

27 Jan '14

It is a fairly common usage to configure SYSCLK as part of the hw_params callback in the machine driver. The current driver applies PCM constraints based on the value of SYSCLK during the startup callback however because many systems will have not configured SYSCLK yet this will often use the SYSCLK value that was used for the last stream. The patch checks the clocking constraints as part of the hw_params callback, which will be after any clocking changes have been made by the machine driver. Change-Id: I48e16256e9ed4df2f4ef1350c356b8b331d288bf Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 79 +++++++------------------------------------ 1 files changed, 13 insertions(+), 66 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index e4295fe..ae5031b 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -990,29 +990,6 @@ static const int arizona_48k_bclk_rates[] = { 24576000, }; -static const unsigned int arizona_48k_rates[] = { - 12000, - 24000, - 48000, - 96000, - 192000, - 384000, - 768000, - 4000, - 8000, - 16000, - 32000, - 64000, - 128000, - 256000, - 512000, -}; - -static const struct snd_pcm_hw_constraint_list arizona_48k_constraint = { - .count = ARRAY_SIZE(arizona_48k_rates), - .list = arizona_48k_rates, -}; - static const int arizona_44k1_bclk_rates[] = { -1, 44100, @@ -1035,21 +1012,6 @@ static const int arizona_44k1_bclk_rates[] = { 22579200, }; -static const unsigned int arizona_44k1_rates[] = { - 11025, - 22050, - 44100, - 88200, - 176400, - 352800, - 705600, -}; - -static const struct snd_pcm_hw_constraint_list arizona_44k1_constraint = { - .count = ARRAY_SIZE(arizona_44k1_rates), - .list = arizona_44k1_rates, -}; - static int arizona_sr_vals[] = { 0, 12000, @@ -1077,13 +1039,15 @@ static int arizona_sr_vals[] = { 512000, }; -static int arizona_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) +static int arizona_hw_params_rate(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); struct arizona_dai_priv *dai_priv = &priv->dai[dai->id - 1]; - const struct snd_pcm_hw_constraint_list *constraint; + int base = dai->driver->base; + int i, sr_val; unsigned int base_rate; switch (dai_priv->clk) { @@ -1094,31 +1058,16 @@ static int arizona_startup(struct snd_pcm_substream *substream, base_rate = priv->asyncclk; break; default: - return 0; + arizona_aif_err(dai, "Unknown clock: %d\n", dai_priv->clk); + return -EINVAL; } - if (base_rate == 0) - return 0; - - if (base_rate % 8000) - constraint = &arizona_44k1_constraint; - else - constraint = &arizona_48k_constraint; - - return snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - constraint); -} - -static int arizona_hw_params_rate(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); - struct arizona_dai_priv *dai_priv = &priv->dai[dai->id - 1]; - int base = dai->driver->base; - int i, sr_val; + if (!!(base_rate % 8000) != !!(params_rate(params) % 8000)) { + arizona_aif_err(dai, + "Rate %dHz not supported off %dHz clock\n", + params_rate(params), base_rate); + return -EINVAL; + } /* * We will need to be more flexible than this in future, @@ -1308,7 +1257,6 @@ static int arizona_set_tristate(struct snd_soc_dai *dai, int tristate) } const struct snd_soc_dai_ops arizona_dai_ops = { - .startup = arizona_startup, .set_fmt = arizona_set_fmt, .hw_params = arizona_hw_params, .set_sysclk = arizona_dai_set_sysclk, @@ -1317,7 +1265,6 @@ const struct snd_soc_dai_ops arizona_dai_ops = { EXPORT_SYMBOL_GPL(arizona_dai_ops); const struct snd_soc_dai_ops arizona_simple_dai_ops = { - .startup = arizona_startup, .hw_params = arizona_hw_params_rate, .set_sysclk = arizona_dai_set_sysclk, }; -- 1.7.2.5

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[alsa-devel] Fwd: [PATCH] fix alsa-tools/hdspmixer for RME 9632 A
by Daniel Turing 27 Jan '14

27 Jan '14

Sorry, this should've been CC'd to the list, forwarding here: ---------- Forwarded message ---------- From: Daniel Turing <mail(a)danielturing.com> Date: Mon, Jan 27, 2014 at 10:52 AM Subject: Re: [PATCH] fix alsa-tools/hdspmixer for RME 9632 A To: Adrian Knoth <adi(a)drcomp.erfurt.thur.de> Adi, thanks for getting back to me on this. On Thu, Jan 23, 2014 at 8:05 PM, Adrian Knoth <adi(a)drcomp.erfurt.thur.de> wrote: > OTOH, I have to admit I don't fully understand the patch. What you're > doing is to increment the number of output channels in case you have an > additional input module. > > The input module is handled in HDSPMixerCard.cxx. Is it just a side > effect? Because semantically, it is wrong to create output destinations > just to get input channels rendered. > > Why is hdspmixer segfaulting? Do you happen to have a backtrace or > explanation? This should be part of the commit message. I admit i don't fully understand hdspmixer ;) Here's the backtrace of the segfault in the unpatched version: [dan@demo2 hdspmixer]$ gdb src/hdspmixer GNU gdb (GDB) 7.6.2 Copyright (C) 2013 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html> This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type "show copying" and "show warranty" for details. This GDB was configured as "x86_64-unknown-linux-gnu". For bug reporting instructions, please see: <http://www.gnu.org/software/gdb/bugs/>... Reading symbols from /home/dan/alsa-tools/alsa-tools-1.0.27/hdspmixer/src/hdspmixer...done. (gdb) r Starting program: /home/dan/alsa-tools/alsa-tools-1.0.27/hdspmixer/src/hdspmixer warning: no loadable sections found in added symbol-file system-supplied DSO at 0x7ffff7ffa000 warning: Could not load shared library symbols for linux-vdso.so.1. Do you need "set solib-search-path" or "set sysroot"? [Thread debugging using libthread_db enabled] Using host libthread_db library "/usr/lib/libthread_db.so.1". HDSPMixer 1.11 - Copyright (C) 2003 Thomas Charbonnel <thomas(a)undata.org> This program comes with ABSOLUTELY NO WARRANTY HDSPMixer is free software, see the file COPYING for details Looking for RME cards: Card 0: RME Hammerfall HDSP 9632 at 0xfebf0000, irq 17 HDSP 9632 found! 1 RME cards card found. Initializing default presets Program received signal SIGSEGV, Segmentation fault. 0x00007ffff71399eb in fl_draw(char const*, int, int, int, int, unsigned int, Fl_Image*, int) () from /usr/lib/libfltk.so.1.3 (gdb) bt #0 0x00007ffff71399eb in fl_draw(char const*, int, int, int, int, unsigned int, Fl_Image*, int) () from /usr/lib/libfltk.so.1.3 #1 0x000000000040aa73 in HDSPMixerSelector::draw (this=0x6a8a50) at HDSPMixerSelector.cxx:180 #2 0x00007ffff70f0709 in Fl_Group::draw_child(Fl_Widget&) const () from /usr/lib/libfltk.so.1.3 #3 0x000000000040a647 in HDSPMixerIOMixer::draw (this=0x6a2360) at HDSPMixerIOMixer.cxx:79 #4 0x00007ffff70f0709 in Fl_Group::draw_child(Fl_Widget&) const () from /usr/lib/libfltk.so.1.3 #5 0x00007ffff70f08fb in Fl_Group::draw_children() () from /usr/lib/libfltk.so.1.3 #6 0x00007ffff70f0709 in Fl_Group::draw_child(Fl_Widget&) const () from /usr/lib/libfltk.so.1.3 #7 0x00007ffff711108b in Fl_Scroll::draw_clip(void*, int, int, int, int) () from /usr/lib/libfltk.so.1.3 #8 0x00007ffff7111d29 in Fl_Scroll::draw() () from /usr/lib/libfltk.so.1.3 #9 0x00007ffff70f0709 in Fl_Group::draw_child(Fl_Widget&) const () from /usr/lib/libfltk.so.1.3 #10 0x00007ffff70f08fb in Fl_Group::draw_children() () from /usr/lib/libfltk.so.1.3 #11 0x00007ffff70e9707 in Fl_Double_Window::flush(int) () from /usr/lib/libfltk.so.1.3 #12 0x00007ffff70db362 in Fl::flush() () from /usr/lib/libfltk.so.1.3 #13 0x00007ffff70dc1c5 in Fl::wait(double) () from /usr/lib/libfltk.so.1.3 #14 0x00007ffff70dc2fd in Fl::run() () from /usr/lib/libfltk.so.1.3 #15 0x0000000000404433 in main (argc=1, argv=0x7fffffffeb88) at hdspmixer.cxx:122 (gdb) It crashes when it tries to render the mixer labels for the inputs (top row). Hdspmixer correctly shows 16 channels, but fails to initialize the right amount of labels (in HDSPMixerSelector::draw(), mvalue()->label() is NULL or some illegal low address value like 0x20 for the last four channels). So i dont think the labels initialized in HDSPMixerSelector::setLabels only relate to outputs, instead they seem to double for inputs and outputs. I'm sorry that i have only limited time and energy to work on this. If the patch would go thru as it is, i can try to jump thru the hoops for "Occasional Developers" though. Best, -daniel -- from the desk of Daniel Turing see also: http://danielturing.com -- from the desk of Daniel Turing see also: http://danielturing.com

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Re: [alsa-devel] [Bug 1155202] [Intel DZ77SL-50K, Intel PantherPoint HDMI, Digital Out, HDMI] No sound at all
by Niraj Kulkarni 27 Jan '14

27 Jan '14

>This is doubtful. Here we see the behavior of the PCI controller, and >there shouldn't be difference by CPU instructions. >The difference in the code between azx_reset() and your code are: >- Use of azx_writeb() and azx_writel() >- Limited loop time for GCTL_RESET bit check I tried both things. It did not work. So, I tried adding a delay between while loop and reset writel in my code. A few experiments with it shows that delay between reset state and on state determines codec detection. Only a delay of less than 100 uSec results in successful codec detection. Here is table of my observations: usleep        result 5-10             ok 50-100         ok 75-100         ok 75-150         fail 100-200       fail 250-500       fail 500-1000     fail Since link_reset_enter and exit have a lot of code in between (atleast greater than 100us) it always results in failure.

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix silent output on MacBook Air 1, 1
by Adrien Vergé 25 Jan '14

25 Jan '14

Similarly to other Apple products, MBA 1,1 needs a specific quirk. Pin 0x18 must be set to VREF_50 to have sound output. This was no longer done since commit 1a97b7f, resulting in a mute built-in speaker. This patch corrects the regression by creating a fixup for the MBA 1,1. Tested-by: Adrien Vergé <adrienverge(a)gmail.com> Signed-off-by: Adrien Vergé <adrienverge(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 19 ++++++++++++++++++- 1 file changed, 18 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c564694..723de28 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1780,6 +1780,7 @@ enum { ALC889_FIXUP_DAC_ROUTE, ALC889_FIXUP_MBP_VREF, ALC889_FIXUP_IMAC91_VREF, + ALC889_FIXUP_MBA11_VREF, ALC889_FIXUP_MBA21_VREF, ALC882_FIXUP_INV_DMIC, ALC882_FIXUP_NO_PRIMARY_HP, @@ -1910,6 +1911,16 @@ static void alc889_fixup_imac91_vref(struct hda_codec *codec, alc889_fixup_mac_pins(codec, nids, ARRAY_SIZE(nids)); } +/* Set VREF on speaker pins on mba11 */ +static void alc889_fixup_mba11_vref(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + static hda_nid_t nids[1] = { 0x18 }; + + if (action == HDA_FIXUP_ACT_INIT) + alc889_fixup_mac_pins(codec, nids, ARRAY_SIZE(nids)); +} + /* Set VREF on speaker pins on mba21 */ static void alc889_fixup_mba21_vref(struct hda_codec *codec, const struct hda_fixup *fix, int action) @@ -2119,6 +2130,12 @@ static const struct hda_fixup alc882_fixups[] = { .chained = true, .chain_id = ALC882_FIXUP_GPIO1, }, + [ALC889_FIXUP_MBA11_VREF] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc889_fixup_mba11_vref, + .chained = true, + .chain_id = ALC889_FIXUP_MBP_VREF, + }, [ALC889_FIXUP_MBA21_VREF] = { .type = HDA_FIXUP_FUNC, .v.func = alc889_fixup_mba21_vref, @@ -2194,7 +2211,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x106b, 0x2c00, "MacbookPro rev3", ALC889_FIXUP_MBP_VREF), SND_PCI_QUIRK(0x106b, 0x3000, "iMac", ALC889_FIXUP_MBP_VREF), SND_PCI_QUIRK(0x106b, 0x3200, "iMac 7,1 Aluminum", ALC882_FIXUP_EAPD), - SND_PCI_QUIRK(0x106b, 0x3400, "MacBookAir 1,1", ALC889_FIXUP_MBP_VREF), + SND_PCI_QUIRK(0x106b, 0x3400, "MacBookAir 1,1", ALC889_FIXUP_MBA11_VREF), SND_PCI_QUIRK(0x106b, 0x3500, "MacBookAir 2,1", ALC889_FIXUP_MBA21_VREF), SND_PCI_QUIRK(0x106b, 0x3600, "Macbook 3,1", ALC889_FIXUP_MBP_VREF), SND_PCI_QUIRK(0x106b, 0x3800, "MacbookPro 4,1", ALC889_FIXUP_MBP_VREF), -- 1.8.5.2

4 7

[alsa-devel] [PATCH 1/2] ASoC: pcm1681: Convert to params_width()
by Mark Brown 25 Jan '14

25 Jan '14

From: Mark Brown <broonie(a)linaro.org> This will help support future enhancements in the way we negotiate parameters in the core. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/pcm1681.c | 15 ++++++++++----- 1 file changed, 10 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c index 73f9c3630e2c..e427544183d7 100644 --- a/sound/soc/codecs/pcm1681.c +++ b/sound/soc/codecs/pcm1681.c @@ -172,16 +172,21 @@ static int pcm1681_hw_params(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); int val = 0, ret; - int pcm_format = params_format(params); priv->rate = params_rate(params); switch (priv->format & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_RIGHT_J: - if (pcm_format == SNDRV_PCM_FORMAT_S24_LE) - val = 0x00; - else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE) - val = 0x03; + switch (params_width(params)) { + case 24: + val = 0; + break; + case 16: + val = 3; + break; + default: + return -EINVAL; + } break; case SND_SOC_DAIFMT_I2S: val = 0x04; -- 1.8.5.3

2 2

[alsa-devel] Audio usb switch to alternate setting 0
by Konstantinos Georgantas 25 Jan '14

25 Jan '14

Hello, I am trying to control from the client side which Alternate Setting to use. Basically I would like to be able to switch from Alternate Setting 0 to 1 and vice versa so that I am not receiving any set interface requests in the meantime. From the audio class 2.0 documentation I understand that I should be able to do something like this by using the active alternate setting control, and here comes my main problem. When I generate an interrupt to the interrupt endpoint I should expect a GET request which I never see in the host side (hope I am right here). Here comes the interrupt: interrupt->bInfo = 0x00; interrupt->bAttribute = 0x01; interrupt->wValue = 0x0100; interrupt->wIndex = 0x0004; Could you please tell me if such a functionality is supported and how the interrupt should look like? Thank you, Konstantinos

2 3

[alsa-devel] [PATCH RFC 0/9] Beaglebone-Black HDMI audio
by Jyri Sarha 24 Jan '14

24 Jan '14

This set of patches implements PCM HDMI audio driver for Beaglebone-Black HDMI output. The patches have been rebased on top of: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound topic/davinci There is couple of earlier RFC versions of the clk patch floating around, but the one here is the latest version. I'll send the associated dts changes shortly as a separate patch. Best regards, Jyri Jyri Sarha (9): clk: add gpio controlled clock ASoC: davinci-evm: Add named clock reference to DT bindings ASoC: davinci-evm: HDMI audio support for TDA998x trough McASP I2S bus ASoC: hdmi-codec: Add devicetree binding with documentation ASoC: hdmi-codec: Add SNDRV_PCM_FMTBIT_32_LE playback format ASoC: davinci: HDMI audio build for AM33XX and TDA998x drm/tilcdc: Add I2C HDMI audio config for tda998x ARM: OMAP2+: omap2plus_defconfig: Enable tilcdc and TDA998X HDMI support ARM: OMAP2+: omap2plus_defconfig: Enable BeagleBone Black HDMI audio support .../devicetree/bindings/clock/gpio-clock.txt | 21 ++ .../bindings/sound/davinci-evm-audio.txt | 12 +- Documentation/devicetree/bindings/sound/hdmi.txt | 17 ++ arch/arm/configs/omap2plus_defconfig | 5 + drivers/clk/Makefile | 1 + drivers/clk/clk-gpio.c | 209 ++++++++++++++++++ drivers/gpu/drm/tilcdc/tilcdc_slave.c | 24 ++- include/linux/clk-provider.h | 25 +++ sound/soc/codecs/hdmi.c | 12 +- sound/soc/davinci/Kconfig | 12 ++ sound/soc/davinci/Makefile | 1 + sound/soc/davinci/davinci-evm.c | 227 +++++++++++++++++++- 12 files changed, 558 insertions(+), 8 deletions(-) create mode 100644 Documentation/devicetree/bindings/clock/gpio-clock.txt create mode 100644 Documentation/devicetree/bindings/sound/hdmi.txt create mode 100644 drivers/clk/clk-gpio.c -- 1.7.9.5

6 58

[alsa-devel] HD-audio runtime PM
by Takashi Iwai 24 Jan '14

24 Jan '14

Hi, after my previous fix, the runtime PM seems working stably finally. However, there seem still some glitches: 1. The wakeup via jack or HDMI/DP detection doesn't seem to work on my test machines. WAKEEN is set properly. And its value can be read correctly at the point of runtime resume, too. 2. We don't change the device to D3 in runtime suspend. So I guess we save little power as of now. Actually, setting to D3 and resuming to D0 works fine, as far as I tested -- except for HDMI/DP ELD read out on Haswell. Although ELD read doesn't give any errors, the received bytes are corrupted. In my case, the read starts from offset 0x1c. The HDMI can be still played even at that state, but then it falls back to the stereo 2ch mode, of course. The tests were performed on both Haswell (DP and LynxPoint-LP) and IvyBridge (PantherPoint) machines. The first problem has been present, and it's also the behavior of some old chips where the codec can't go to sleep with unsol wakeup. So, in this regard, it's no regression, per se. But certainly it's no good thing. I thought Mengdong once made WAKEEN working. Mengdong, do you remember what was the condition? Could you check whether the latest kernel (at best sound git tree for-linus branch) still works? About the second point: we should do D3 if we do care power, and that's the very right reason for runtime PM. But the ELD issue is a bad bug, and it makes me wonder whether it's a graphics driver side issue or a codec side. As a quick workaround, we can implement D3 on controllers but for Haswell HDMI/DP. In anyway, this is no urgent issue, clearly targeted for 3.14. The whole patch series are found in my sound-unstable git tree test/hda branch. git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git thanks, Takashi

4 28

[alsa-devel] [PATCH 1/3] ASoC: Samsung: Add NULL check in i2s.c
by Sachin Kamat 24 Jan '14

24 Jan '14

'res' could be NULL from one of the operations above (line 1243). Thus check 'res' for NULL before releasing the region to avoid null pointer dereference. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/samsung/i2s.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 92f64363427d..a9da24f34834 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1268,7 +1268,8 @@ static int samsung_i2s_probe(struct platform_device *pdev) return 0; err: - release_mem_region(regs_base, resource_size(res)); + if (res) + release_mem_region(regs_base, resource_size(res)); return ret; } -- 1.7.9.5

2 7

[alsa-devel] [PATCH 1/2] ASoC: wm8580: Fix build error
by Sachin Kamat 24 Jan '14

24 Jan '14

Re-organize the code a bit to fix the following build errors and warnings: sound/soc/codecs/wm8580.c:932:21: error: variable ‘wm8580_regmap’ has initializer but incomplete type static const struct regmap_config wm8580_regmap = { sound/soc/codecs/wm8580.c:233:20: warning: ‘wm8580_supply_names’ defined but not used [-Wunused-variable] sound/soc/codecs/wm8580.c:840:34: warning: ‘wm8580_dai’ defined but not used [-Wunused-variable] sound/soc/codecs/wm8580.c:914:36: warning: ‘soc_codec_dev_wm8580’ defined but not used [-Wunused-variable] Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/codecs/wm8580.c | 38 ++++++++++++++++++++------------------ 1 file changed, 20 insertions(+), 18 deletions(-) diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 318989acbbe5..7a9d6fa9f3b3 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -214,28 +214,12 @@ static const struct reg_default wm8580_reg_defaults[] = { { 52, 0x0000 }, }; -static bool wm8580_volatile(struct device *dev, unsigned int reg) -{ - switch (reg) { - case WM8580_RESET: - return true; - default: - return false; - } -} - struct pll_state { unsigned int in; unsigned int out; }; #define WM8580_NUM_SUPPLIES 3 -static const char *wm8580_supply_names[WM8580_NUM_SUPPLIES] = { - "AVDD", - "DVDD", - "PVDD", -}; - /* codec private data */ struct wm8580_priv { struct regmap *regmap; @@ -837,6 +821,7 @@ static const struct snd_soc_dai_ops wm8580_dai_ops_capture = { .set_pll = wm8580_set_dai_pll, }; +#if IS_ENABLED(CONFIG_I2C) static struct snd_soc_dai_driver wm8580_dai[] = { { .name = "wm8580-hifi-playback", @@ -863,6 +848,7 @@ static struct snd_soc_dai_driver wm8580_dai[] = { .ops = &wm8580_dai_ops_capture, }, }; +#endif static int wm8580_probe(struct snd_soc_codec *codec) { @@ -911,7 +897,7 @@ static int wm8580_remove(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_wm8580 = { +static struct snd_soc_codec_driver soc_codec_dev_wm8580 __maybe_unused = { .probe = wm8580_probe, .remove = wm8580_remove, .set_bias_level = wm8580_set_bias_level, @@ -929,6 +915,23 @@ static const struct of_device_id wm8580_of_match[] = { { }, }; +#if IS_ENABLED(CONFIG_I2C) +static const char *wm8580_supply_names[WM8580_NUM_SUPPLIES] = { + "AVDD", + "DVDD", + "PVDD", +}; + +static bool wm8580_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WM8580_RESET: + return true; + default: + return false; + } +} + static const struct regmap_config wm8580_regmap = { .reg_bits = 7, .val_bits = 9, @@ -941,7 +944,6 @@ static const struct regmap_config wm8580_regmap = { .volatile_reg = wm8580_volatile, }; -#if IS_ENABLED(CONFIG_I2C) static int wm8580_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { -- 1.7.9.5

2 5

Re: [alsa-devel] Fw: [Bug 1155202] [Intel DZ77SL-50K, Intel PantherPoint HDMI, Digital Out, HDMI] No sound at all
by Takashi Iwai 24 Jan '14

24 Jan '14

At 22 Jan 2014 14:20:23 -0000, Niraj Kulkarni wrote: > > Let me try to explain my findings: > > > > > How does this magic sequence come? > > By dumping Win7 driver's init sequence and reproducing it through ALSA driver. > > > > >why does it need to set UNSOL flag at first? > > No such need, I've put it after regular init so that only affected chipsets are reset again. Putting this anywhere around init code works (tried it). OK. > >Is the while loop guaranteed to quit for a reasonable time? > > Being household user, I can only confirm on my chipset. Only a thorough testing may ensure sanity. Then you must put a timeout not to lock up the machine. > >Can't it be simply calling azx_reset() twice? > > It is interesting hardware bug which kicks in only during first init after power-on. So if Win7 driver in qemu inits the board, and I simply plugin "unchanged" hda_intel afterwards, it still detects codecs. But multiple normal resets do not have any effect. Being no hardware expert, I can only speculate that tight spin loop after reset has the crux. A normal delay call there does not detect codec. This is doubtful. Here we see the behavior of the PCI controller, and there shouldn't be difference by CPU instructions. The difference in the code between azx_reset() and your code are: - Use of azx_writeb() and azx_writel() - Limited loop time for GCTL_RESET bit check For the first one, you can try the code change like below: -- 8< -- diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index fa2879a21a50..03525cff565a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1143,10 +1143,10 @@ static void azx_enter_link_reset(struct azx *chip) unsigned long timeout; /* reset controller */ - azx_writel(chip, GCTL, azx_readl(chip, GCTL) & ~ICH6_GCTL_RESET); + azx_writel(chip, GCTL, 0); timeout = jiffies + msecs_to_jiffies(100); - while ((azx_readb(chip, GCTL) & ICH6_GCTL_RESET) && + while (azx_readl(chip, GCTL) && time_before(jiffies, timeout)) usleep_range(500, 1000); } @@ -1156,10 +1156,10 @@ static void azx_exit_link_reset(struct azx *chip) { unsigned long timeout; - azx_writeb(chip, GCTL, azx_readb(chip, GCTL) | ICH6_GCTL_RESET); + azx_writel(chip, GCTL, ICH6_GCTL_RESET); timeout = jiffies + msecs_to_jiffies(100); - while (!azx_readb(chip, GCTL) && + while (!azx_readl(chip, GCTL) && time_before(jiffies, timeout)) usleep_range(500, 1000); } -- 8< -- For the second point, you can increase the timeout value e.g. timeout = jiffies + msecs_to_jiffies(500); > >And, yet another question comes up: doesn't S4 need the similar workaround? If S4 works, what's the difference? > > Haven't tried with sleep-wake or hibernate-restore sequence yet. Please check it. I guess S4 is broken with your patch. Takashi

1 0

Re: [alsa-devel] [PATCH Resend 7/8] ASoC: add snd_soc_of_parse_audio_simple_widgets for DeviceTree
by Mark Brown 24 Jan '14

24 Jan '14

On Fri, Jan 24, 2014 at 02:46:33AM +0000, Li.Xiubo(a)freescale.com wrote: > > > +static struct snd_soc_dapm_widget simple_widgets[] = { > > > + SND_SOC_DAPM_MIC("Mic", NULL), > > > + SND_SOC_DAPM_LINE("Line", NULL), > > > + SND_SOC_DAPM_HP("Hp", NULL), > > > + SND_SOC_DAPM_SPK("Spk", NULL), > > > +}; > This is the templates for "Microphone XXX", "Line XXX", "Headphone XXX" > and "Spk XXX" . > > Does this mean we're restricted to a particular set of names? That > > seems sad and won't work if there's a desire for more than one of a > > given widget - the main use case I can see is multiple microphones with > > separate microphone biases. How about having some templates that we > > copy and then replace the name with the one the user supplied? > This code has already supplied this I think. OK, that's fine - this is one of the things that the lack of binding documentation doesn't help with, it says what the code is supposed to do. When reviewing DT code without bindings I tend to just skim the code.

1 0

[alsa-devel] [PATCH] alsa-libs: use CLOCK_REALTIME for non-monotonic ts
by Olivier Langlois 24 Jan '14

24 Jan '14

gettimeofday() is the same than clock_gettime(CLOCK_REALTIME) except for the loss of precision and the double unnecessary conversion timespec <-> timeval. Signed-off-by: Olivier Langlois <olivier(a)trillion01.com> --- src/pcm/pcm_local.h | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/src/pcm/pcm_local.h b/src/pcm/pcm_local.h index e1c0baa..8a6c743 100644 --- a/src/pcm/pcm_local.h +++ b/src/pcm/pcm_local.h @@ -961,17 +961,21 @@ typedef union snd_tmp_double { /* get the current timestamp */ static inline void gettimestamp(snd_htimestamp_t *tstamp, int monotonic) { -#if defined(HAVE_CLOCK_GETTIME) && defined(CLOCK_MONOTONIC) +#if defined(HAVE_CLOCK_GETTIME) +#if defined(CLOCK_MONOTONIC) if (monotonic) { clock_gettime(CLOCK_MONOTONIC, tstamp); } else { #endif + clock_gettime(CLOCK_REALTIME, tstamp); +#else struct timeval tv; gettimeofday(&tv, 0); tstamp->tv_sec = tv.tv_sec; tstamp->tv_nsec = tv.tv_usec * 1000L; -#if defined(HAVE_CLOCK_GETTIME) && defined(CLOCK_MONOTONIC) +#endif +#if defined(HAVE_CLOCK_GETTIME) } #endif } -- 1.8.5.3

2 1

[alsa-devel] Kernel Patch: MOTU Micro Lite (5x5 MIDI USB Interface)
by Studio65 24 Jan '14

24 Jan '14

Hello Alsa Team, may I try to explain my/our problem as short as possible. We have been truing to hack a kernel patch to get a MOTU Micro Lite (5x5 MIDI USB Interface) to run within a Debian "wheezy" with RT Kernel. We did get the five channels appear in #qjackctl or #patchage. we tried to receive input from #vkeybd and get output to #gmidimonitor while on the Hardware we connected: MOTU Midi In 1 <--> Motu Midi Out 1 MOTU Midi In 2 <--> Motu Midi Out 2 MOTU Midi In 3 <--> Motu Midi Out 3 MOTU Midi In 4 <--> Motu Midi Out 4 MOTU Midi In 5 <--> Motu Midi Out 5 ######################################################################################################################## Here is some Logs: Output of #tail -f /var/log/messages http://pastebin.com/R6xppq7k ----------------------------------------------------------------------- Jan 5 02:31:40 debian kernel: [24513.648043] usb 2-2: new full-speed USB device number 8 using ohci_hcd Jan 5 02:31:40 debian kernel: [24513.869063] usb 2-2: New USB device found, idVendor=07fd, idProduct=0001 Jan 5 02:31:40 debian kernel: [24513.869068] usb 2-2: New USB device strings: Mfr=1, Product=2, SerialNumber=0 Jan 5 02:31:40 debian kernel: [24513.869072] usb 2-2: Product: micro lite Jan 5 02:31:40 debian kernel: [24513.869074] usb 2-2: Manufacturer: Mark of the Unicorn Jan 5 02:31:40 debian kernel: [24513.872194] created 16 output and 16 input ports Jan 5 02:38:04 debian kernel: [24897.727180] usb 2-2: USB disconnect, device number 8 Jan 5 02:38:14 debian kernel: [24907.724039] ALSA rawmidi.c:185 >rawmidi drain error (avail = 1678, buffer_size = 4096) Jan 5 02:38:24 debian kernel: [24917.724024] ALSA rawmidi.c:185 >rawmidi drain error (avail = 1678, buffer_size = 4096) Jan 5 02:38:34 debian kernel: [24927.724032] ALSA rawmidi.c:185 >rawmidi drain error (avail = 1678, buffer_size = 4096) Jan 5 02:38:44 debian kernel: [24937.724038] ALSA rawmidi.c:185 >rawmidi drain error (avail = 1678, buffer_size = 4096) Jan 5 02:38:54 debian kernel: [24947.724025] ALSA rawmidi.c:185 >rawmidi drain error (avail = 1678, buffer_size = 4096) Jan 5 02:39:04 debian kernel: [24957.724031] ALSA rawmidi.c:185 >rawmidi drain error (avail = 1678, buffer_size = 4096) Jan 5 02:39:14 debian kernel: [24967.724026] ALSA rawmidi.c:185 >rawmidi drain error (avail = 1678, buffer_size = 4096) Jan 5 02:39:24 debian kernel: [24977.724021] ALSA rawmidi.c:185 >rawmidi drain error (avail = 1678, buffer_size = 4096) Jan 5 02:39:34 debian kernel: [24987.724029] ALSA rawmidi.c:185 >rawmidi drain error (avail = 1678, buffer_size = 4096) Jan 5 02:39:44 debian kernel: [24997.724030] ALSA rawmidi.c:185 >rawmidi drain error (avail = 1678, buffer_size = 4096) Jan 5 02:39:54 debian kernel: [25007.724034] ALSA rawmidi.c:185 >rawmidi drain error (avail = 1678, buffer_size = 4096) Jan 5 02:40:04 debian kernel: [25017.724034] ALSA rawmidi.c:185 >rawmidi drain error (avail = 1678, buffer_size = 4096) Jan 5 02:40:14 debian kernel: [25027.724026] ALSA rawmidi.c:185 >rawmidi drain error (avail = 1678, buffer_size = 4096) Jan 5 02:40:24 debian kernel: [25037.724037] ALSA rawmidi.c:185 >rawmidi drain error (avail = 1678, buffer_size = 4096) Jan 5 02:40:34 debian kernel: [25047.724041] ALSA rawmidi.c:185 >rawmidi drain error (avail = 1678, buffer_size = 4096) Jan 5 02:40:44 debian kernel: [25057.724021] ALSA rawmidi.c:185 >rawmidi drain error (avail = 1678, buffer_size = 4096) ------------------------------------------------------------------------------------------------------------------------ ######################################################################################################################## Output of #lsusb -vv http://pastebin.com/Du0N6hCa ------------------------------------------------------------------------------------------------------------------------ nathanael@debian:~$ lsusb -vv Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Couldn't open device, some information will be missing Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 9 Hub bDeviceSubClass 0 Unused bDeviceProtocol 0 Full speed (or root) hub bMaxPacketSize0 64 idVendor 0x1d6b Linux Foundation idProduct 0x0002 2.0 root hub bcdDevice 3.02 iManufacturer 3 iProduct 2 iSerial 1 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 25 bNumInterfaces 1 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xe0 Self Powered Remote Wakeup MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 9 Hub bInterfaceSubClass 0 Unused bInterfaceProtocol 0 Full speed (or root) hub iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0004 1x 4 bytes bInterval 12 Bus 002 Device 001: ID 1d6b:0001 Linux Foundation 1.1 root hub Couldn't open device, some information will be missing Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.10 bDeviceClass 9 Hub bDeviceSubClass 0 Unused bDeviceProtocol 0 Full speed (or root) hub bMaxPacketSize0 64 idVendor 0x1d6b Linux Foundation idProduct 0x0001 1.1 root hub bcdDevice 3.02 iManufacturer 3 iProduct 2 iSerial 1 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 25 bNumInterfaces 1 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xe0 Self Powered Remote Wakeup MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 9 Hub bInterfaceSubClass 0 Unused bInterfaceProtocol 0 Full speed (or root) hub iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0002 1x 2 bytes bInterval 255 Bus 003 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Couldn't open device, some information will be missing Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 9 Hub bDeviceSubClass 0 Unused bDeviceProtocol 0 Full speed (or root) hub bMaxPacketSize0 64 idVendor 0x1d6b Linux Foundation idProduct 0x0002 2.0 root hub bcdDevice 3.02 iManufacturer 3 iProduct 2 iSerial 1 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 25 bNumInterfaces 1 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xe0 Self Powered Remote Wakeup MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 9 Hub bInterfaceSubClass 0 Unused bInterfaceProtocol 0 Full speed (or root) hub iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0004 1x 4 bytes bInterval 12 Bus 004 Device 001: ID 1d6b:0001 Linux Foundation 1.1 root hub Couldn't open device, some information will be missing Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.10 bDeviceClass 9 Hub bDeviceSubClass 0 Unused bDeviceProtocol 0 Full speed (or root) hub bMaxPacketSize0 64 idVendor 0x1d6b Linux Foundation idProduct 0x0001 1.1 root hub bcdDevice 3.02 iManufacturer 3 iProduct 2 iSerial 1 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 25 bNumInterfaces 1 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xe0 Self Powered Remote Wakeup MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 9 Hub bInterfaceSubClass 0 Unused bInterfaceProtocol 0 Full speed (or root) hub iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0002 1x 2 bytes bInterval 255 Bus 004 Device 002: ID 04b4:0033 Cypress Semiconductor Corp. Mouse Couldn't open device, some information will be missing Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.10 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 8 idVendor 0x04b4 Cypress Semiconductor Corp. idProduct 0x0033 Mouse bcdDevice 1.00 iManufacturer 0 iProduct 0 iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 34 bNumInterfaces 1 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xa0 (Bus Powered) Remote Wakeup MaxPower 100mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 3 Human Interface Device bInterfaceSubClass 1 Boot Interface Subclass bInterfaceProtocol 2 Mouse iInterface 0 HID Device Descriptor: bLength 9 bDescriptorType 33 bcdHID 1.00 bCountryCode 0 Not supported bNumDescriptors 1 bDescriptorType 34 Report wDescriptorLength 61 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0008 1x 8 bytes bInterval 10 Bus 002 Device 009: ID 07fd:0001 Mark of the Unicorn FastLane Quad MIDI Interface Couldn't open device, some information will be missing Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.00 bDeviceClass 255 Vendor Specific Class bDeviceSubClass 3 bDeviceProtocol 0 bMaxPacketSize0 8 idVendor 0x07fd Mark of the Unicorn idProduct 0x0001 FastLane Quad MIDI Interface bcdDevice 1.00 iManufacturer 1 iProduct 2 iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 107 bNumInterfaces 2 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 100mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 255 iInterface 0 ** UNRECOGNIZED: 0b 24 01 01 05 00 02 05 00 01 00 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 255 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 255 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0020 1x 32 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x02 EP 2 OUT bmAttributes 1 Transfer Type Isochronous Synch Type None Usage Type Data wMaxPacketSize 0x0020 1x 32 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 2 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 255 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0020 1x 32 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x02 EP 2 OUT bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0020 1x 32 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 3 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 255 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0020 1x 32 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x02 EP 2 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0020 1x 32 bytes bInterval 1 ------------------------------------------------------------------------------------------------------------------------ ######################################################################################################################## Greets, Regards and all the good stuff... Nathanael Studio65 _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

2 1

[alsa-devel] [PATCH] ASoC: simple-card: fix simple card widgets routing property name usage
by Xiubo Li 24 Jan '14

24 Jan '14

Fix the usage of simple card widgets routing property, and make it the same with simple card routing property name. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- Documentation/devicetree/bindings/sound/simple-card.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt index e9e20ec..19c84df 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ b/Documentation/devicetree/bindings/sound/simple-card.txt @@ -43,7 +43,7 @@ Example: sound { compatible = "simple-audio-card"; simple-audio-card,format = "left_j"; - simple-audio-routing = + simple-audio-card,routing = "MIC_IN", "Mic Jack", "Headphone Jack", "HP_OUT", "Ext Spk", "LINE_OUT"; -- 1.8.4

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Re: [alsa-devel] [ISSUE] rtl8192ce appears to interfere with ALSA playback
by Olivier Langlois 24 Jan '14

24 Jan '14

James, I think that I have found and fix the cause of the timing miscalculation [1] I have recompiled rtlwifi modules to have access to the debug option which I have set to 4. I have also compiled ALSA modules with the XRUN_DEBUG option that I have activated with echo 1 > /proc/asound/card0/pcm0p/xrun_debug I have reproduced the problems while collecting dmesg output and using 'watch -n1 cat /proc/interrupts' I have made a couple of observations. 1. there is a small burst of interrupts for rtl8192ce at a regular interval which seems to happen during rtl_op_sw_scan and the underruns appear to coincide with that. 2. I was expecting to see XRUN entries in dmesg but I didn't. The closest that I have is [ 1700.756017] PCM: Lost interrupts? [Q] (stream=0, delta=16108, new_hw_ptr=383756, old_hw_ptr=367648) 3. I didn't realise that the default ALSA device was a plugin using dmix. I have no underruns by using directly the ALSA HW device (hw:0,0) 4. I have seen this in dmesg logs: hda-intel: IRQ timing workaround is activated for card #0. Suggest a bigger bdl_pos_adj. here is the aplay output with the dmesg logs around this time: $ aplay -v ~/Music/sine.wav Playing WAVE '/home/lano1106/Music/sine.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Plug PCM: Rate conversion PCM (48000, sformat=S32_LE) Converter: linear-interpolation Protocol version: 10002 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 15052 period_size : 940 period_time : 21333 tstamp_mode : NONE period_step : 1 avail_min : 940 period_event : 0 start_threshold : 15052 stop_threshold : 15052 silence_threshold: 0 silence_size : 0 boundary : 986447872 Slave: Soft volume PCM Control: PCM Playback Volume min_dB: -51 max_dB: 0 resolution: 256 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 16384 period_size : 1024 period_time : 21333 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 16384 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Slave: Direct Stream Mixing PCM Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 16384 period_size : 1024 period_time : 21333 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 16384 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Hardware PCM card 0 'HDA Intel' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 16384 period_size : 1024 period_time : 21333 tstamp_mode : ENABLE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 1 stop_threshold : 1073741824 silence_threshold: 0 silence_size : 1073741824 boundary : 1073741824 appl_ptr : 0 hw_ptr : 0 underrun!!! (at least 7098.589 ms long) Status: state : XRUN trigger_time: 1692.774265154 tstamp : 1700.760145135 delay : -301 avail : 15353 avail_max : 15341 [ 1692.761247] hda-intel 0000:00:1b.0: azx_pcm_prepare: bufsize=0x20000, format=0x31 [ 1692.761268] hda_codec_setup_stream: NID=0x13, stream=0x8, channel=0, format=0x31 [ 1692.761275] hda_codec_setup_stream: NID=0x14, stream=0x8, channel=0, format=0x31 [ 1694.303968] rtl8192ce:_rtl92ce_phy_set_rf_power_state():<0-1> IPS Set eRf nic enable [ 1694.304021] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1694.304086] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1694.304200] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1694.304314] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1694.304427] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1694.304541] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1694.304655] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1694.304693] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1694.304693] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1694.304693] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1694.304693] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1694.304693] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:98053f15:10 [ 1694.304693] rtl8192ce:rtl92ce_sw_led_on():<0-1> LedAddr:4E ledpin=1 [ 1694.304693] rtl8192c_common:rtl92c_download_fw():<0-1> Firmware Version(49), Signature(0x88c1),Size(32) [ 1694.318113] rtl8192ce:rtl92ce_enable_hw_security_config():<0-1> PairwiseEncAlgorithm = 0 GroupEncAlgorithm = 0 [ 1694.318113] rtl8192ce:rtl92ce_enable_hw_security_config():<0-1> The SECR-value cc [ 1694.318113] rtl8192c_common:rtl92c_dm_check_txpower_tracking_thermal_meter():<0-1> Schedule TxPowerTracking direct call!! [ 1694.318113] rtl8192c_common:rtl92c_dm_txpower_tracking_callback_thermalmeter():<0-1> rtl92c_dm_txpower_tracking_callback_thermalmeter [ 1694.318113] rtl8192c_common:rtl92c_dm_txpower_tracking_callback_thermalmeter():<0-1> Readback Thermal Meter = 0xd pre thermal meter 0xf eeprom_thermalmeter 0xf [ 1694.318113] rtl8192c_common:rtl92c_dm_txpower_tracking_callback_thermalmeter():<0-1> Initial pathA ele_d reg0xc80 = 0x40000000, ofdm_index=0xc [ 1694.318113] rtl8192c_common:rtl92c_dm_txpower_tracking_callback_thermalmeter():<0-1> Initial reg0xa24 = 0x90e1317, cck_index=0xc, ch14 0 [ 1694.318113] rtl8192c_common:rtl92c_dm_txpower_tracking_callback_thermalmeter():<0-1> Readback Thermal Meter = 0xd pre thermal meter 0xf eeprom_thermalmeter 0xf delta 0x2 delta_lck 0x1 delta_iqk 0x1 [ 1694.318113] rtl8192c_common:rtl92c_dm_txpower_tracking_callback_thermalmeter():<0-1> <=== [ 1694.318113] rtl8192c_common:rtl92c_dm_initialize_txpower_tracking_thermalmeter():<0-1> pMgntInfo->txpower_tracking = 1 [ 1694.318113] rtl8192ce:rtl92ce_led_control():<0-1> ledaction 3 [ 1694.318113] rtl8192ce:rtl92ce_sw_led_on():<0-1> LedAddr:4E ledpin=1 [ 1694.640920] PCM: Lost interrupts? [Q] (stream=0, delta=14444, new_hw_ptr=90248, old_hw_ptr=75804) [ 1694.641440] rtl8192c_common:rtl92c_dm_dynamic_bb_powersaving():<0-0> Not connected to any [ 1694.641454] rtl8192c_common:rtl92c_dm_dynamic_bb_powersaving():<0-0> AP Ext Port PWDB = 0x0 [ 1694.641472] rtl8192c_common:rtl92c_dm_check_txpower_tracking_thermal_meter():<0-0> Trigger 92S Thermal Meter!! [ 1694.641517] rtlwifi:rtl_op_sw_scan_start():<0-0> [ 1694.641530] rtl8192ce:rtl92ce_led_control():<0-0> ledaction 6 [ 1694.641545] rtl8192c_common:rtl92c_dm_write_dig():<0-0> cur_igvalue = 0x20, pre_igvalue = 0x0, back_val = 10 [ 1694.654779] rtl8192c_common:rtl92c_phy_sw_chnl():<0-0> sw_chnl_inprogress false schedule workitem [ 1694.654786] rtl8192ce:rtl92ce_set_hw_reg():<0-0> HW_VAR_SLOT_TIME 14 [ 1694.724173] rtl8192c_common:rtl92c_phy_sw_chnl():<0-0> sw_chnl_inprogress false schedule workitem [ 1694.724179] rtl8192ce:rtl92ce_set_hw_reg():<0-0> HW_VAR_SLOT_TIME 14 [ 1694.794176] rtl8192c_common:rtl92c_phy_sw_chnl():<0-0> sw_chnl_inprogress false schedule workitem [ 1694.794181] rtl8192ce:rtl92ce_set_hw_reg():<0-0> HW_VAR_SLOT_TIME 14 [ 1694.864217] rtl8192c_common:rtl92c_phy_sw_chnl():<0-0> sw_chnl_inprogress false schedule workitem [ 1694.864226] rtl8192ce:rtl92ce_set_hw_reg():<0-0> HW_VAR_SLOT_TIME 14 [ 1694.934184] rtl8192c_common:rtl92c_phy_sw_chnl():<0-0> sw_chnl_inprogress false schedule workitem [ 1694.934193] rtl8192ce:rtl92ce_set_hw_reg():<0-0> HW_VAR_SLOT_TIME 14 [ 1695.004187] rtl8192c_common:rtl92c_phy_sw_chnl():<0-0> sw_chnl_inprogress false schedule workitem [ 1695.004196] rtl8192ce:rtl92ce_set_hw_reg():<0-0> HW_VAR_SLOT_TIME 14 [ 1695.074187] rtl8192c_common:rtl92c_phy_sw_chnl():<0-0> sw_chnl_inprogress false schedule workitem [ 1695.074196] rtl8192ce:rtl92ce_set_hw_reg():<0-0> HW_VAR_SLOT_TIME 14 [ 1695.144185] rtl8192c_common:rtl92c_phy_sw_chnl():<0-0> sw_chnl_inprogress false schedule workitem [ 1695.144194] rtl8192ce:rtl92ce_set_hw_reg():<0-0> HW_VAR_SLOT_TIME 14 [ 1695.214184] rtl8192c_common:rtl92c_phy_sw_chnl():<0-0> sw_chnl_inprogress false schedule workitem [ 1695.214196] rtl8192ce:rtl92ce_set_hw_reg():<0-0> HW_VAR_SLOT_TIME 14 [ 1695.284182] rtl8192c_common:rtl92c_phy_sw_chnl():<0-0> sw_chnl_inprogress false schedule workitem [ 1695.284192] rtl8192ce:rtl92ce_set_hw_reg():<0-0> HW_VAR_SLOT_TIME 14 [ 1695.354186] rtl8192c_common:rtl92c_phy_sw_chnl():<0-0> sw_chnl_inprogress false schedule workitem [ 1695.354195] rtl8192ce:rtl92ce_set_hw_reg():<0-0> HW_VAR_SLOT_TIME 14 [ 1695.424207] rtl8192c_common:rtl92c_phy_sw_chnl():<0-0> sw_chnl_inprogress false schedule workitem [ 1695.424217] rtl8192ce:rtl92ce_set_hw_reg():<0-0> HW_VAR_SLOT_TIME 14 [ 1695.424275] rtlwifi:rtl_op_sw_scan_complete():<0-0> [ 1695.424282] rtl8192c_common:rtl92c_dm_write_dig():<0-0> cur_igvalue = 0x22, pre_igvalue = 0x22, back_val = 10 [ 1695.524091] rtl8192ce:_rtl92ce_phy_set_rf_power_state():<0-0> IPS Set eRf nic disable [ 1695.524141] rtl8192ce:rtl92ce_led_control():<0-0> ledaction 2 [ 1695.524147] rtl8192ce:rtl92ce_sw_led_on():<0-0> LedAddr:4E ledpin=1 [ 1700.419054] rtl8192ce:_rtl92ce_phy_set_rf_power_state():<0-1> IPS Set eRf nic enable [ 1700.419106] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1700.419172] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1700.419286] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1700.419399] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1700.419513] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1700.419627] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1700.419745] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1700.419866] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1700.419984] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1700.420015] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1700.420015] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1700.420015] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:98053f15:10 [ 1700.420015] rtl8192ce:rtl92ce_sw_led_on():<0-1> LedAddr:4E ledpin=1 [ 1700.420015] rtl8192c_common:rtl92c_download_fw():<0-1> Firmware Version(49), Signature(0x88c1),Size(32) [ 1700.420015] rtl8192ce:rtl92ce_enable_hw_security_config():<0-1> PairwiseEncAlgorithm = 0 GroupEncAlgorithm = 0 [ 1700.420015] rtl8192ce:rtl92ce_enable_hw_security_config():<0-1> The SECR-value cc [ 1700.420015] rtl8192c_common:rtl92c_dm_check_txpower_tracking_thermal_meter():<0-1> Schedule TxPowerTracking direct call!! [ 1700.420015] rtl8192c_common:rtl92c_dm_txpower_tracking_callback_thermalmeter():<0-1> rtl92c_dm_txpower_tracking_callback_thermalmeter [ 1700.420015] rtl8192c_common:rtl92c_dm_txpower_tracking_callback_thermalmeter():<0-1> Readback Thermal Meter = 0xd pre thermal meter 0xf eeprom_thermalmeter 0xf [ 1700.420015] rtl8192c_common:rtl92c_dm_txpower_tracking_callback_thermalmeter():<0-1> Initial pathA ele_d reg0xc80 = 0x40000000, ofdm_index=0xc [ 1700.420015] rtl8192c_common:rtl92c_dm_txpower_tracking_callback_thermalmeter():<0-1> Initial reg0xa24 = 0x90e1317, cck_index=0xc, ch14 0 [ 1700.420015] rtl8192c_common:rtl92c_dm_txpower_tracking_callback_thermalmeter():<0-1> Readback Thermal Meter = 0xd pre thermal meter 0xf eeprom_thermalmeter 0xf delta 0x2 delta_lck 0x1 delta_iqk 0x1 [ 1700.420015] rtl8192c_common:rtl92c_dm_txpower_tracking_callback_thermalmeter():<0-1> <=== [ 1700.420015] rtl8192c_common:rtl92c_dm_initialize_txpower_tracking_thermalmeter():<0-1> pMgntInfo->txpower_tracking = 1 [ 1700.420015] rtl8192ce:rtl92ce_led_control():<0-1> ledaction 3 [ 1700.420015] rtl8192ce:rtl92ce_sw_led_on():<0-1> LedAddr:4E ledpin=1 [ 1700.756017] PCM: Lost interrupts? [Q] (stream=0, delta=16108, new_hw_ptr=383756, old_hw_ptr=367648) [ 1700.756513] rtlwifi:rtl_op_sw_scan_start():<0-0> [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-January/071521.ht… On Thu, 2014-01-16 at 16:55 +1100, James Cameron wrote: > On Thu, Jan 16, 2014 at 12:25:42AM -0500, Olivier Langlois wrote: > > On 01/15/2014 12:01 PM, Larry Finger wrote: > > >On 01/15/2014 12:37 AM, Olivier Langlois wrote: > > >>How to reproduce: > > >> > > >>1. Enable Wifi while not connecting to any AP. > > >>2. lano1106@hpmini ~/Music $ aplay -c1 sine.wav > > >>underrun!!! (at least 1856093977.967 ms long) > > Indeed, that timing is fishy. > > I've looked at the alsa-devel thread [1] and your problem description > there. > > At OLPC during development we found similar symptoms showing up in > ALSA playback that were contributed to other drivers, but the > underlying causes were in the ALSA driver for our codec chip, and the > other drivers were changing the appearance of the fault only because > of timing changes. > > But there's no need to conclude that interrupts were disabled or > delayed. There can be many other causes. > > The timing miscalculation by ALSA might be important. I can't figure > out how a wireless driver can easily cause that. > > 1. > http://mailman.alsa-project.org/pipermail/alsa-devel/2014-January/071142.ht… >

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[alsa-devel] [PATCH] fix alsa-tools/hdspmixer for RME 9632 A
by Daniel Turing 23 Jan '14

23 Jan '14

Dear alsa-devel, Jaroslav and Adi, since i never turned up here, let me just quickly congratulate and thank y'all for your outstanding work on linux audio. My world wouldn't be as great without you! I run a HDSP 9632 with the "AI4s-192 AIO" 4x Analog input extension board. It works great so far, but hdspmixer segfaults because it tries to show 16 channels with fltk, but doesnt care to initialize the channel labels for the input extension board. Since one can only add either the output (AO4S), or the input (AI4S) extension board, and hdspmixer already prepares channel labels for the output case, the fix is quite easy, and attached. I'd be happy if this patch could find its way into mainline alsa-tools. Thanks, -daniel -- from the desk of Daniel Turing see also: http://danielturing.com

2 1

[alsa-devel] [PATCH 1/3] ASoC: wm5102: Correct typo in EQ coefficient sizes
by Charles Keepax 23 Jan '14

23 Jan '14

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Sorry seems there was a small typo in my patches from yesterday. Let me know if you would rather I respun the patches from yesterday but for now here they are as an incremental fix. Thanks, Charles sound/soc/codecs/wm5102.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 5e03a83..ebffe81 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -685,7 +685,7 @@ ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), -SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 18), +SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 19), SOC_SINGLE("EQ1 Mode Switch", ARIZONA_EQ1_2, ARIZONA_EQ1_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, 24, 0, eq_tlv), @@ -698,7 +698,7 @@ SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 18), +SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 19), SOC_SINGLE("EQ2 Mode Switch", ARIZONA_EQ2_2, ARIZONA_EQ2_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, 24, 0, eq_tlv), @@ -711,7 +711,7 @@ SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 18), +SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 19), SOC_SINGLE("EQ3 Mode Switch", ARIZONA_EQ3_2, ARIZONA_EQ3_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, 24, 0, eq_tlv), @@ -724,7 +724,7 @@ SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 18), +SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 19), SOC_SINGLE("EQ4 Mode Switch", ARIZONA_EQ4_2, ARIZONA_EQ4_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, 24, 0, eq_tlv), -- 1.7.2.5

2 3

[alsa-devel] [TINYCOMPRESS PATCH 1/2 v3] crec: Initial version of a compressed capture utility
by Charles Keepax 23 Jan '14

23 Jan '14

This version only supports capture of PCM streams over a compressed device and saves these as WAV files. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Changes since v2: - Switched to using POSIX file IO to improve signal safety Thanks, Charles Android.mk | 10 ++ crec.c | 418 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++ makefile.linux | 20 ++- 3 files changed, 441 insertions(+), 7 deletions(-) create mode 100644 crec.c diff --git a/Android.mk b/Android.mk index 104dfef..3cdd3d0 100644 --- a/Android.mk +++ b/Android.mk @@ -20,3 +20,13 @@ LOCAL_MODULE_TAGS := optional include $(BUILD_EXECUTABLE) +include $(CLEAR_VARS) + +LOCAL_C_INCLUDES:= $(LOCAL_PATH)/include +LOCAL_SRC_FILES:= crec.c +LOCAL_MODULE := crec +LOCAL_SHARED_LIBRARIES:= libcutils libutils libtinycompress +LOCAL_MODULE_TAGS := optional + +include $(BUILD_EXECUTABLE) + diff --git a/crec.c b/crec.c new file mode 100644 index 0000000..df53e4e --- /dev/null +++ b/crec.c @@ -0,0 +1,418 @@ +/* + * BSD LICENSE + * + * crec command line recorder for compress audio record in alsa + * Copyright (c) 2011-2012, Intel Corporation + * Copyright (c) 2013-2014, Wolfson Microelectronic Ltd. + * All rights reserved. + * + * Author: Vinod Koul <vinod.koul(a)linux.intel.com> + * Author: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> + * + * Redistribution and use in source and binary forms, with or without + * modification, are permitted provided that the following conditions are met: + * + * Redistributions of source code must retain the above copyright notice, + * this list of conditions and the following disclaimer. + * Redistributions in binary form must reproduce the above copyright notice, + * this list of conditions and the following disclaimer in the documentation + * and/or other materials provided with the distribution. + * Neither the name of Intel Corporation nor the names of its contributors + * may be used to endorse or promote products derived from this software + * without specific prior written permission. + * + * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" + * AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE + * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE + * ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE + * LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR + * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF + * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS + * INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN + * CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) + * ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF + * THE POSSIBILITY OF SUCH DAMAGE. + * + * LGPL LICENSE + * + * crec command line recorder for compress audio record in alsa + * Copyright (c) 2011-2012, Intel Corporation + * Copyright (c) 2013-2014, Wolfson Microelectronic Ltd. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU Lesser General Public License, + * version 2.1, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU Lesser General Public + * License for more details. + * + * You should have received a copy of the GNU Lesser General Public License + * along with this program; if not, write to + * the Free Software Foundation, Inc., + * 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include <stdint.h> +#include <linux/types.h> +#include <fcntl.h> +#include <errno.h> +#include <unistd.h> +#include <stdio.h> +#include <stdlib.h> +#include <string.h> +#include <signal.h> +#include <stdbool.h> +#include <getopt.h> +#include <sys/time.h> +#define __force +#define __bitwise +#define __user +#include "sound/compress_params.h" +#include "sound/compress_offload.h" +#include "tinycompress/tinycompress.h" + +static int verbose; + +static const unsigned int DEFAULT_CHANNELS = 1; +static const unsigned int DEFAULT_RATE = 44100; +static const unsigned int DEFAULT_FORMAT = SNDRV_PCM_FORMAT_S16_LE; + +struct riff_chunk { + char desc[4]; + uint32_t size; +} __attribute__((__packed__)); + +struct wave_header { + struct { + struct riff_chunk chunk; + char format[4]; + } __attribute__((__packed__)) riff; + + struct { + struct riff_chunk chunk; + uint16_t type; + uint16_t channels; + uint32_t rate; + uint32_t byterate; + uint16_t blockalign; + uint16_t samplebits; + } __attribute__((__packed__)) fmt; + + struct { + struct riff_chunk chunk; + } __attribute__((__packed__)) data; +} __attribute__((__packed__)); + +const struct wave_header blank_wave_header = { + .riff = { + .chunk = { + .desc = "RIFF", + }, + .format = "WAVE", + }, + .fmt = { + .chunk = { + .desc = "fmt ", /* Note the space is important here */ + .size = sizeof(blank_wave_header.fmt) - + sizeof(blank_wave_header.fmt.chunk), + }, + .type = 0x01, /* PCM */ + }, + .data = { + .chunk = { + .desc = "data", + }, + }, +}; + +static void init_wave_header(struct wave_header *header, uint16_t channels, + uint32_t rate, uint16_t samplebits) +{ + memcpy(header, &blank_wave_header, sizeof(blank_wave_header)); + + header->fmt.channels = channels; + header->fmt.rate = rate; + header->fmt.byterate = channels * rate * (samplebits / 8); + header->fmt.blockalign = channels * (samplebits / 8); + header->fmt.samplebits = samplebits; +} + +static void size_wave_header(struct wave_header *header, uint32_t size) +{ + header->riff.chunk.size = sizeof(*header) - + sizeof(header->riff.chunk) + size; + header->data.chunk.size = size; +} + +static void usage(void) +{ + fprintf(stderr, "usage: crec [OPTIONS] filename\n" + "-c\tcard number\n" + "-d\tdevice node\n" + "-b\tbuffer size\n" + "-f\tfragments\n" + "-v\tverbose mode\n" + "-l\tlength of record in seconds\n" + "-h\tPrints this help list\n\n" + "-C\tSpecify the number of channels (default %u)\n" + "-R\tSpecify the sample rate (default %u)\n" + "-F\tSpecify the format: S16_LE, S32_LE (default S16_LE)\n\n" + "Example:\n" + "\tcrec -c 1 -d 2 test.wav\n" + "\tcrec -f 5 test.wav\n", + DEFAULT_CHANNELS, DEFAULT_RATE); + + exit(EXIT_FAILURE); +} + +static int print_time(struct compress *compress) +{ + unsigned int avail; + struct timespec tstamp; + + if (compress_get_hpointer(compress, &avail, &tstamp) != 0) { + fprintf(stderr, "Error querying timestamp\n"); + fprintf(stderr, "ERR: %s\n", compress_get_error(compress)); + return -1; + } else { + printf("DSP recorded %jd.%jd\n", + (intmax_t)tstamp.tv_sec, (intmax_t)tstamp.tv_nsec*1000); + } + return 0; +} + +void capture_samples(char *name, unsigned int card, unsigned int device, + unsigned long buffer_size, unsigned int frag, + unsigned int length, unsigned int rate, + unsigned int channels, unsigned int format) +{ + struct compr_config config; + struct snd_codec codec; + struct compress *compress; + struct wave_header header; + int file; + char *buffer; + size_t written; + int read, ret; + unsigned int size, total_read = 0; + unsigned int samplebits; + + switch (format) { + case SNDRV_PCM_FORMAT_S32_LE: + samplebits = 32; + break; + default: + samplebits = 16; + break; + } + + /* Convert length from seconds to bytes */ + length = length * rate * (samplebits / 8) * channels; + + if (verbose) + printf("%s: entry, reading %u bytes\n", __func__, length); + + file = open(name, O_WRONLY); + if (file == -1) { + fprintf(stderr, "Unable to open file '%s'\n", name); + exit(EXIT_FAILURE); + } + + /* Write a header, will update with size once record is complete */ + init_wave_header(&header, channels, rate, samplebits); + written = write(file, &header, sizeof(header)); + if (written != sizeof(header)) { + fprintf(stderr, "Error writing output file header: %s\n", + strerror(errno)); + goto file_exit; + } + + memset(&codec, 0, sizeof(codec)); + memset(&config, 0, sizeof(config)); + codec.id = SND_AUDIOCODEC_PCM; + codec.ch_in = channels; + codec.ch_out = channels; + codec.sample_rate = rate; + if (!codec.sample_rate) { + fprintf(stderr, "invalid sample rate %d\n", rate); + goto file_exit; + } + codec.format = format; + if ((buffer_size != 0) && (frag != 0)) { + config.fragment_size = buffer_size/frag; + config.fragments = frag; + } + config.codec = &codec; + + compress = compress_open(card, device, COMPRESS_OUT, &config); + if (!compress || !is_compress_ready(compress)) { + fprintf(stderr, "Unable to open Compress device %d:%d\n", + card, device); + fprintf(stderr, "ERR: %s\n", compress_get_error(compress)); + goto file_exit; + }; + + if (verbose) + printf("%s: Opened compress device\n", __func__); + + size = config.fragment_size; + buffer = malloc(size * config.fragments); + if (!buffer) { + fprintf(stderr, "Unable to allocate %d bytes\n", size); + goto comp_exit; + } + + printf("Recording file %s On Card %u device %u, with buffer of %lu bytes\n", + name, card, device, buffer_size); + printf("Format %u Channels %u, %u Hz\n", + codec.id, codec.ch_out, rate); + + compress_start(compress); + + if (verbose) + printf("%s: Capturing audio NOW!!!\n", __func__); + + do { + if (length && size > length - total_read) + size = length - total_read; + + read = compress_read(compress, buffer, size); + if (read < 0) { + fprintf(stderr, "Error reading sample\n"); + fprintf(stderr, "ERR: %s\n", compress_get_error(compress)); + goto buf_exit; + } + if ((unsigned int)read != size) { + fprintf(stderr, "We read %d, DSP sent %d\n", + size, read); + } + + if (read > 0) { + total_read += read; + + written = write(file, buffer, read); + if (written != (size_t)read) { + fprintf(stderr, "Error writing output file: %s\n", + strerror(errno)); + goto buf_exit; + } + if (verbose) { + print_time(compress); + printf("%s: read %d\n", __func__, read); + } + } + } while (!length || total_read < length); + + ret = compress_stop(compress); + if (ret < 0) { + fprintf(stderr, "Error closing stream\n"); + fprintf(stderr, "ERR: %s\n", compress_get_error(compress)); + } + + /* Update file header now we know file size */ + size_wave_header(&header, total_read); + ret = lseek(file, 0, SEEK_SET); + if (ret < 0) { + fprintf(stderr, "Error seeking: %s\n", strerror(errno)); + goto buf_exit; + } + written = write(file, &header, sizeof(header)); + if (written != sizeof(header)) { + fprintf(stderr, "Error updating output file header: %s\n", + strerror(errno)); + goto buf_exit; + } + + if (verbose) + printf("%s: exit success\n", __func__); + + free(buffer); + close(file); + + compress_close(compress); + + return; +buf_exit: + free(buffer); +comp_exit: + compress_close(compress); +file_exit: + close(file); + + if (verbose) + printf("%s: exit failure\n", __func__); + + exit(EXIT_FAILURE); +} + +int main(int argc, char **argv) +{ + char *file; + unsigned long buffer_size = 0; + int c; + unsigned int card = 0, device = 0, frag = 0, length = 0; + unsigned int rate = DEFAULT_RATE, channels = DEFAULT_CHANNELS; + unsigned int format = DEFAULT_FORMAT; + + if (argc < 2) + usage(); + + verbose = 0; + while ((c = getopt(argc, argv, "hvl:R:C:F:b:f:c:d:")) != -1) { + switch (c) { + case 'h': + usage(); + break; + case 'b': + buffer_size = strtol(optarg, NULL, 0); + break; + case 'f': + frag = strtol(optarg, NULL, 10); + break; + case 'c': + card = strtol(optarg, NULL, 10); + break; + case 'd': + device = strtol(optarg, NULL, 10); + break; + case 'v': + verbose = 1; + break; + case 'l': + length = strtol(optarg, NULL, 10); + break; + case 'R': + rate = strtol(optarg, NULL, 10); + break; + case 'C': + channels = strtol(optarg, NULL, 10); + break; + case 'F': + if (strcmp(optarg, "S16_LE") == 0) { + format = SNDRV_PCM_FORMAT_S16_LE; + } else if (strcmp(optarg, "S32_LE") == 0) { + format = SNDRV_PCM_FORMAT_S32_LE; + } else { + fprintf(stderr, "Unrecognised format: %s\n", + optarg); + usage(); + } + break; + default: + exit(EXIT_FAILURE); + } + } + if (optind >= argc) + usage(); + + file = argv[optind]; + + capture_samples(file, card, device, buffer_size, frag, length, + rate, channels, format); + + printf("Finish capturing... Close Normally\n"); + exit(EXIT_SUCCESS); +} + diff --git a/makefile.linux b/makefile.linux index 677878d..81782d0 100644 --- a/makefile.linux +++ b/makefile.linux @@ -1,10 +1,13 @@ LIB = libtinycompress -BIN = cplay VER = 0.1 LIBSRC = compress.c -SRC = cplay.c utils.c +PLAYBIN = cplay +RECBIN = crec +PLAYSRC = cplay.c utils.c +RECSRC = crec.c utils.c LIBOBJ = ${LIBSRC:.c=.o} -OBJ = ${SRC:.c=.o} +PLAYOBJ = ${PLAYSRC:.c=.o} +RECOBJ = ${RECSRC:.c=.o} CC = gcc CROSS_COMPILE = @@ -13,23 +16,26 @@ CFLAGS += -std=c99 -Wall -Wextra -Wunused -DVERSION=\"${VER}\" -I./include LDFLAGS += -L. -ltinycompress LIBLDFLAGS = -lasound -all: libtinycompress cplay +all: libtinycompress cplay crec $(LIB): ${LIBOBJ} @echo " LD "$@ @${CROSS_COMPILE}${CC} ${CFLAGS} -shared -Wl,-soname,$@.so -o $@.so ${LIBOBJ} ${LIBLDFLAGS} -$(BIN): ${OBJ} ${LIB} +$(PLAYBIN): ${PLAYOBJ} ${LIB} @echo " LD "$@ - @${CROSS_COMPILE}${CC} ${CFLAGS} -o $@ ${OBJ} ${LDFLAGS} + @${CROSS_COMPILE}${CC} ${CFLAGS} -o $@ ${PLAYOBJ} ${LDFLAGS} +$(RECBIN): ${RECOBJ} ${LIB} + @echo " LD "$@ + @${CROSS_COMPILE}${CC} ${CFLAGS} -o $@ ${RECOBJ} ${LDFLAGS} %.o: %.c @echo " CC "$< @${CROSS_COMPILE}${CC} ${CFLAGS} -c -fPIC -o $@ $< clean: - @rm -rf ${BIN} ${OBJ} ${LIB}.so ${LIBOBJ} + @rm -rf ${BIN} ${PLAYOBJ} ${RECOBJ} ${LIB}.so ${LIBOBJ} .PHONY: all clean -- 1.7.2.5

1 1

[alsa-devel] abdul, please add me to your LinkedIn network
by Putinei Ionut via LinkedIn 23 Jan '14

23 Jan '14

LinkedIn ------------ Putinei Ionut requested to add you as a connection on LinkedIn: ------------------------------------------ abdul, I'd like to add you to my professional network on LinkedIn. - Putinei Accept invitation from Putinei Ionut http://www.linkedin.com/e/jpxkk3-hqs72qer-24/uii-rtptP1H4ViCg9YiBUOvto1r4_5… View profile of Putinei Ionut http://www.linkedin.com/e/jpxkk3-hqs72qer-24/rso/227244299/bCFj/name/138036… ------------------------------------------ You are receiving Invitation emails. This email was intended for abdul moiz. Learn why this is included: http://www.linkedin.com/e/jpxkk3-hqs72qer-24/plh/http%3A%2F%2Fhelp%2Elinked… (c) 2012, LinkedIn Corporation. 2029 Stierlin Ct, Mountain View, CA 94043, USA.

1 0

[alsa-devel] [PATCH] ASoC: tlv320aic32x4: Use gpio_is_valid()
by Fabio Estevam 23 Jan '14

23 Jan '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> We could use gpio_is_valid() for doing the sanity check on the GPIO number. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/tlv320aic32x4.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 385dec1..02d2d3c 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -687,7 +687,7 @@ static int aic32x4_i2c_probe(struct i2c_client *i2c, aic32x4->rstn_gpio = -1; } - if (aic32x4->rstn_gpio >= 0) { + if (gpio_is_valid(aic32x4->rstn_gpio)) { ret = devm_gpio_request_one(&i2c->dev, aic32x4->rstn_gpio, GPIOF_OUT_INIT_LOW, "tlv320aic32x4 rstn"); if (ret != 0) -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ASoC: codecs: Make OF supported CODECs visible in Kconfig
by Mark Brown 23 Jan '14

23 Jan '14

From: Mark Brown <broonie(a)linaro.org> Now that we have a generic card driver we can't rely on the card driver selecting the CODECs for us so make the CODECs that can be enabled with OF directly selectable in Kconfig. For the platforms not using OF it's not clear that we don't still want to have some board specific selection since the kernel needs to contain code to register the devices; ACPI could provide this from firmware does not yet support any kind of generic card. It may also be desirable to hide these if OF is not enabled to reduce noise. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/Kconfig | 96 +++++++++++++++++++++++++++++++----------------- 1 file changed, 62 insertions(+), 34 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index fa989cb77fe4..e0c2febf3bc0 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -191,8 +191,9 @@ config SND_SOC_AD73311 tristate config SND_SOC_ADAU1701 + tristate "Analog Devices ADAU1701 CODEC" + depends on I2C select SND_SOC_SIGMADSP - tristate config SND_SOC_ADAU1373 tristate @@ -204,28 +205,31 @@ config SND_SOC_ADS117X tristate config SND_SOC_AK4104 - tristate + tristate "AKM AK4104 CODEC" + depends on SPI_MASTER config SND_SOC_AK4535 tristate config SND_SOC_AK4554 - tristate + tristate "AKM AK4554 CODEC" config SND_SOC_AK4641 tristate config SND_SOC_AK4642 - tristate + tristate "AKM AK4642 CODEC" + depends on I2C config SND_SOC_AK4671 tristate config SND_SOC_AK5386 - tristate + tristate "AKM AK5638 CODEC" config SND_SOC_ALC5623 tristate + config SND_SOC_ALC5632 tristate @@ -236,14 +240,17 @@ config SND_SOC_CS42L51 tristate config SND_SOC_CS42L52 - tristate + tristate "Cirrus Logic CS42L52 CODEC" + depends on I2C config SND_SOC_CS42L73 - tristate + tristate "Cirrus Logic CS42L73 CODEC" + depends on I2C # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 - tristate + tristate "Cirrus Logic CS4270 CODEC" + depends on I2C # Cirrus Logic CS4270 Codec VD = 3.3V Errata # Select if you are affected by the errata where the part will not function @@ -254,7 +261,7 @@ config SND_SOC_CS4270_VD33_ERRATA depends on SND_SOC_CS4270 config SND_SOC_CS4271 - tristate + tristate "Cirrus Logic CS4271 CODEC" config SND_SOC_CX20442 tristate @@ -285,6 +292,9 @@ config SND_SOC_BT_SCO config SND_SOC_DMIC tristate +config SND_SOC_HDMI_CODEC + tristate "HDMI stub CODEC" + config SND_SOC_ISABELLE tristate @@ -303,14 +313,13 @@ config SND_SOC_MAX98095 config SND_SOC_MAX9850 tristate -config SND_SOC_HDMI_CODEC - tristate - config SND_SOC_PCM1681 - tristate + tristate "Texas Instruments PCM1681 CODEC" + depends on I2C config SND_SOC_PCM1792A - tristate + tristate "Texas Instruments PCM1792A CODEC" + depends on SPI_MASTER config SND_SOC_PCM3008 tristate @@ -326,7 +335,8 @@ config SND_SOC_RT5640 #Freescale sgtl5000 codec config SND_SOC_SGTL5000 - tristate + tristate "Freescale SGTL5000 CODEC" + depends on I2C config SND_SOC_SI476X tristate @@ -339,7 +349,7 @@ config SND_SOC_SN95031 tristate config SND_SOC_SPDIF - tristate + tristate "S/PDIF CODEC" config SND_SOC_SSM2518 tristate @@ -357,7 +367,8 @@ config SND_SOC_STAC9766 tristate config SND_SOC_TAS5086 - tristate + tristate "Texas Instruments TAS5086 speaker amplifier" + depends on I2C config SND_SOC_TLV320AIC23 tristate @@ -370,7 +381,8 @@ config SND_SOC_TLV320AIC32X4 tristate config SND_SOC_TLV320AIC3X - tristate + tristate "Texas Instruments TLV320AIC3x CODECs" + depends on I2C config SND_SOC_TLV320DAC33 tristate @@ -419,55 +431,69 @@ config SND_SOC_WM8400 tristate config SND_SOC_WM8510 - tristate + tristate "Wolfson Microelectronics WM8510 CODEC" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8523 - tristate + tristate "Wolfson Microelectronics WM8523 DAC" + depends on I2C config SND_SOC_WM8580 - tristate + tristate "Wolfson Microelectronics WM8523 CODEC" + depends on I2C config SND_SOC_WM8711 - tristate + tristate "Wolfson Microelectronics WM8711 CODEC" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8727 tristate config SND_SOC_WM8728 - tristate + tristate "Wolfson Microelectronics WM8728 DAC" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8731 - tristate + tristate "Wolfson Microelectronics WM8731 CODEC" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8737 - tristate + tristate "Wolfson Microelectronics WM8737 ADC" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8741 - tristate + tristate "Wolfson Microelectronics WM8737 DAC" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8750 - tristate + tristate "Wolfson Microelectronics WM8750 CODEC" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8753 - tristate + tristate "Wolfson Microelectronics WM8753 CODEC" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8770 - tristate + tristate "Wolfson Microelectronics WM8770 CODEC" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8776 - tristate + tristate "Wolfson Microelectronics WM8776 CODEC" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8782 tristate config SND_SOC_WM8804 - tristate + tristate "Wolfson Microelectronics WM8804 S/PDIF transceiver" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8900 tristate config SND_SOC_WM8903 - tristate + tristate "Wolfson Microelectronics WM8903 CODEC" + depends on I2C config SND_SOC_WM8904 tristate @@ -485,7 +511,8 @@ config SND_SOC_WM8961 tristate config SND_SOC_WM8962 - tristate + tristate "Wolfson Microelectronics WM8962 CODEC" + depends on I2C config SND_SOC_WM8971 tristate @@ -558,4 +585,5 @@ config SND_SOC_ML26124 tristate config SND_SOC_TPA6130A2 - tristate + tristate "Texas Instruments TPA6130A2 headphone amplifier" + depends on I2C -- 1.8.5.3

2 2

[alsa-devel] [PATCH] ASoC: ak4554: Add to SND_SOC_ALL_CODECS
by Mark Brown 23 Jan '14

23 Jan '14

From: Mark Brown <broonie(a)linaro.org> For build coverage. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 5f74fc3f672c..fa989cb77fe4 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -25,6 +25,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C + select SND_SOC_AK4554 select SND_SOC_AK4641 if I2C select SND_SOC_AK4642 if I2C select SND_SOC_AK4671 if I2C -- 1.8.5.3

1 0

[alsa-devel] [PATCH 1/2] ASoC: Samsung: Fix Kconfig dependency
by Sachin Kamat 23 Jan '14

23 Jan '14

Select S3C24XX_DMA instead of S3C2410_DMA to avoid following dependency issues and build errors: warning: (CPU_S3C2410 && CPU_S3C2442 && SND_SOC_SAMSUNG && SND_S3C24XX_I2S && SND_S3C2412_SOC_I2S && SND_SOC_SAMSUNG_SMDK2443_WM9710 && SND_SOC_SAMSUNG_LN2440SBC_ALC650) selects S3C2410_DMA which has unmet direct dependencies (ARCH_S3C24XX && S3C24XX_DMA && (CPU_S3C2410 || CPU_S3C2442)) warning: (CPU_S3C2410 && CPU_S3C2442 && SND_SOC_SAMSUNG && SND_S3C24XX_I2S && SND_S3C2412_SOC_I2S && SND_SOC_SAMSUNG_SMDK2443_WM9710 && SND_SOC_SAMSUNG_LN2440SBC_ALC650) selects S3C2410_DMA which has unmet direct dependencies (ARCH_S3C24XX && S3C24XX_DMA && (CPU_S3C2410 || CPU_S3C2442)) arch/arm/mach-s3c24xx/built-in.o: In function `s3c2410_dma_add': arch/arm/mach-s3c24xx/dma-s3c2410.c:134: undefined reference to `s3c2410_dma_init' arch/arm/mach-s3c24xx/dma-s3c2410.c:135: undefined reference to `s3c24xx_dma_order_set' arch/arm/mach-s3c24xx/dma-s3c2410.c:136: undefined reference to `s3c24xx_dma_init_map' arch/arm/plat-samsung/include/plat/dma-ops.h:60: undefined reference to `s3c_dma_get_ops' sound/soc/samsung/s3c24xx-i2s.c:293: undefined reference to `s3c2410_dma_ctrl' arch/arm/plat-samsung/include/plat/dma-ops.h:60: undefined reference to `s3c_dma_get_ops' arch/arm/plat-samsung/include/plat/dma-ops.h:60: undefined reference to `s3c_dma_get_ops' sound/built-in.o: In function `s3c2412_i2s_trigger': sound/soc/samsung/s3c-i2s-v2.c:432: undefined reference to `s3c2410_dma_ctrl' Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/samsung/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 27930fc432dc..e8fbf4e5732f 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -19,7 +19,7 @@ config SND_S3C_DMA_LEGACY config SND_S3C24XX_I2S tristate - select S3C2410_DMA + select S3C24XX_DMA config SND_S3C_I2SV2_SOC tristate -- 1.7.9.5

2 1

[alsa-devel] [PATCH 2/2] ASoC: wm5100: Export wm5100_detect
by Sachin Kamat 23 Jan '14

23 Jan '14

Export the symbol so that it is accessible to modules. Fixes the following error: ERROR: "wm5100_detect" [sound/soc/samsung/snd-soc-lowland.ko] undefined! Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/codecs/wm5100.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 4cf91deabc02..4e3e31aaf509 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -14,6 +14,7 @@ #include <linux/moduleparam.h> #include <linux/init.h> #include <linux/delay.h> +#include <linux/export.h> #include <linux/pm.h> #include <linux/gcd.h> #include <linux/gpio.h> @@ -2141,6 +2142,7 @@ int wm5100_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) return 0; } +EXPORT_SYMBOL_GPL(wm5100_detect); static irqreturn_t wm5100_irq(int irq, void *data) { -- 1.7.9.5

3 2

[alsa-devel] [PATCH] ALSA: Refactor slot assignment code
by Takashi Iwai 23 Jan '14

23 Jan '14

There are two loops that are almost identical but only with different checks. Refactor them with a simple helper, and give a bit more comments what's doing there. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/init.c | 57 +++++++++++++++++++++++++++++-------------------------- 1 file changed, 30 insertions(+), 27 deletions(-) diff --git a/sound/core/init.c b/sound/core/init.c index e3c93cd77ee7..0d42fcda0de2 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -131,6 +131,31 @@ static inline int init_info_for_card(struct snd_card *card) #define init_info_for_card(card) #endif +static int check_empty_slot(struct module *module, int slot) +{ + return !slots[slot] || !*slots[slot]; +} + +/* return an empty slot number (>= 0) found in the given bitmask @mask. + * @mask == -1 == 0xffffffff means: take any free slot up to 32 + * when no slot is available, return the original @mask as is. + */ +static int get_slot_from_bitmask(int mask, int (*check)(struct module *, int), + struct module *module) +{ + int slot; + + for (slot = 0; slot < SNDRV_CARDS; slot++) { + if (slot < 32 && !(mask & (1U << slot))) + continue; + if (!test_bit(slot, snd_cards_lock)) { + if (check(module, slot)) + return slot; /* found */ + } + } + return mask; /* unchanged */ +} + /** * snd_card_create - create and initialize a soundcard structure * @idx: card index (address) [0 ... (SNDRV_CARDS-1)] @@ -152,7 +177,7 @@ int snd_card_create(int idx, const char *xid, struct snd_card **card_ret) { struct snd_card *card; - int err, idx2; + int err; if (snd_BUG_ON(!card_ret)) return -EINVAL; @@ -167,32 +192,10 @@ int snd_card_create(int idx, const char *xid, strlcpy(card->id, xid, sizeof(card->id)); err = 0; mutex_lock(&snd_card_mutex); - if (idx < 0) { - for (idx2 = 0; idx2 < SNDRV_CARDS; idx2++) { - /* idx == -1 == 0xffff means: take any free slot */ - if (idx2 < 32 && !(idx & (1U << idx2))) - continue; - if (!test_bit(idx2, snd_cards_lock)) { - if (module_slot_match(module, idx2)) { - idx = idx2; - break; - } - } - } - } - if (idx < 0) { - for (idx2 = 0; idx2 < SNDRV_CARDS; idx2++) { - /* idx == -1 == 0xffff means: take any free slot */ - if (idx2 < 32 && !(idx & (1U << idx2))) - continue; - if (!test_bit(idx2, snd_cards_lock)) { - if (!slots[idx2] || !*slots[idx2]) { - idx = idx2; - break; - } - } - } - } + if (idx < 0) /* first check the matching module-name slot */ + idx = get_slot_from_bitmask(idx, module_slot_match, module); + if (idx < 0) /* if not matched, assign an empty slot */ + idx = get_slot_from_bitmask(idx, check_empty_slot, module); if (idx < 0) err = -ENODEV; else if (idx < snd_ecards_limit) { -- 1.8.5.2

1 0

[alsa-devel] [patch] ALSA: bits vs bytes bug in snd_card_create()
by Dan Carpenter 23 Jan '14

23 Jan '14

The test here is intended intended to prevent shift wrapping bugs when we do "1U << idx2". We should consider the number of bits in a u32 instead of the number of bytes. Fixes: 7bb2491b35a2 ('ALSA: Add kconfig to specify the max card numbers') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- WARNING: Please check this one carefully because I might be wrong. diff --git a/sound/core/init.c b/sound/core/init.c index 1351f22f651c..e9e3db321e64 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -183,7 +183,7 @@ int snd_card_create(int idx, const char *xid, if (idx < 0) { for (idx2 = 0; idx2 < SNDRV_CARDS; idx2++) { /* idx == -1 == 0xffff means: take any free slot */ - if (idx2 < sizeof(int) && !(idx & (1U << idx2))) + if (idx2 < 32 && !(idx & (1U << idx2))) continue; if (!test_bit(idx2, snd_cards_lock)) { if (!slots[idx2] || !*slots[idx2]) {

3 5

[alsa-devel] [PATCH] alsa-libs: initialize pcm_dmix monotonic field
by Olivier Langlois 23 Jan '14

23 Jan '14

not doing so, leaves the pcm object in an inconsistent state since 'info' field is copied from the slave which is then used when snd_pcm_hw_params_is_monotonic() is called. For instance, when using dmix with aplay and an underrun is occuring, the following info is returned: underrun!!! (at least 1248687948.256 ms long) Status: state : XRUN trigger_time: 1390347762.628483000 tstamp : 1390347766.184350000 delay : -635 avail : 15687 avail_max : 15675 now is computed from CLOCK_MONOTONIC while pcm status tstamps are from gettimeofday(). After the fix, underruns are still occuring on my setup but at least the displayed info is correct: underrun!!! (at least 7630.409 ms long) Status: state : XRUN trigger_time: 7652.739201431 tstamp : 7660.369600636 delay : -624 avail : 15676 avail_max : 15664 Signed-off-by: Olivier Langlois <olivier(a)trillion01.com> --- src/pcm/pcm_direct.c | 2 ++ src/pcm/pcm_dmix.c | 2 +- 2 files changed, 3 insertions(+), 1 deletion(-) diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index 899c250..5416cf7 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -857,6 +857,7 @@ static void save_slave_setting(snd_pcm_direct_t *dmix, snd_pcm_t *spcm) COPY_SLAVE(buffer_time); COPY_SLAVE(sample_bits); COPY_SLAVE(frame_bits); + COPY_SLAVE(monotonic); } #undef COPY_SLAVE @@ -1220,6 +1221,7 @@ static void copy_slave_setting(snd_pcm_direct_t *dmix, snd_pcm_t *spcm) COPY_SLAVE(buffer_time); COPY_SLAVE(sample_bits); COPY_SLAVE(frame_bits); + COPY_SLAVE(monotonic); spcm->info &= ~SND_PCM_INFO_PAUSE; spcm->boundary = recalc_boundary_size(dmix->shmptr->s.boundary, spcm->buffer_size); diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index 2bd5d39..4aa6d4e 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -1104,7 +1104,7 @@ int snd_pcm_dmix_open(snd_pcm_t **pcmp, const char *name, pcm->poll_fd = dmix->poll_fd; pcm->poll_events = POLLIN; /* it's different than other plugins */ - + pcm->monotonic = spcm->monotonic; pcm->mmap_rw = 1; snd_pcm_set_hw_ptr(pcm, &dmix->hw_ptr, -1, 0); snd_pcm_set_appl_ptr(pcm, &dmix->appl_ptr, -1, 0); -- 1.8.5.3

2 1

[alsa-devel] [PATCH v2 8/8] ASoC: simple-card: simplify code
by Jean-Francois Moine 22 Jan '14

22 Jan '14

Rename the pointer to the private data structure to 'priv' to avoid confusion with the platform data pointer. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/soc/generic/simple-card.c | 32 +++++++++++----------- 1 file changed, 16 insertions(+), 16 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 0f11c28..6443c87 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -47,18 +47,18 @@ static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct simple_card_data *info = + struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *codec = rtd->codec_dai; struct snd_soc_dai *cpu = rtd->cpu_dai; - unsigned int daifmt = info->daifmt; + unsigned int daifmt = priv->daifmt; int ret; - ret = __asoc_simple_card_dai_init(codec, &info->codec_dai, daifmt); + ret = __asoc_simple_card_dai_init(codec, &priv->codec_dai, daifmt); if (ret < 0) return ret; - ret = __asoc_simple_card_dai_init(cpu, &info->cpu_dai, daifmt); + ret = __asoc_simple_card_dai_init(cpu, &priv->cpu_dai, daifmt); if (ret < 0) return ret; @@ -129,21 +129,21 @@ parse_error: } static int asoc_simple_card_parse_of(struct device_node *node, - struct simple_card_data *info, + struct simple_card_data *priv, struct device *dev) { - struct snd_soc_dai_link *dai_link = info->snd_card.dai_link; + struct snd_soc_dai_link *dai_link = priv->snd_card.dai_link; struct device_node *np; char *name; int ret; /* get CPU/CODEC common format via simple-audio-card,format */ - info->daifmt = snd_soc_of_parse_daifmt(node, "simple-audio-card,") & + priv->daifmt = snd_soc_of_parse_daifmt(node, "simple-audio-card,") & (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK); /* DAPM routes */ if (of_property_read_bool(node, "simple-audio-card,routing")) { - ret = snd_soc_of_parse_audio_routing(&info->snd_card, + ret = snd_soc_of_parse_audio_routing(&priv->snd_card, "simple-audio-card,routing"); if (ret) return ret; @@ -154,7 +154,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, np = of_get_child_by_name(node, "simple-audio-card,cpu"); if (np) ret = asoc_simple_card_sub_parse_of(np, - &info->cpu_dai, + &priv->cpu_dai, &dai_link->cpu_of_node, &dai_link->cpu_dai_name); if (ret < 0) @@ -165,7 +165,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, np = of_get_child_by_name(node, "simple-audio-card,codec"); if (np) ret = asoc_simple_card_sub_parse_of(np, - &info->codec_dai, + &priv->codec_dai, &dai_link->codec_of_node, &dai_link->codec_dai_name); if (ret < 0) @@ -181,22 +181,22 @@ static int asoc_simple_card_parse_of(struct device_node *node, GFP_KERNEL); sprintf(name, "%s-%s", dai_link->cpu_dai_name, dai_link->codec_dai_name); - info->snd_card.name = name; + priv->snd_card.name = name; dai_link->name = dai_link->stream_name = name; /* simple-card assumes platform == cpu */ dai_link->platform_of_node = dai_link->cpu_of_node; dev_dbg(dev, "card-name : %s\n", name); - dev_dbg(dev, "platform : %04x\n", info->daifmt); + dev_dbg(dev, "platform : %04x\n", priv->daifmt); dev_dbg(dev, "cpu : %s / %04x / %d\n", dai_link->cpu_dai_name, - info->cpu_dai.fmt, - info->cpu_dai.sysclk); + priv->cpu_dai.fmt, + priv->cpu_dai.sysclk); dev_dbg(dev, "codec : %s / %04x / %d\n", dai_link->codec_dai_name, - info->codec_dai.fmt, - info->codec_dai.sysclk); + priv->codec_dai.fmt, + priv->codec_dai.sysclk); return 0; } -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 1

[alsa-devel] [PATCH v2 7/8] ASoC: simple-card: simplify code
by Jean-Francois Moine 22 Jan '14

22 Jan '14

The platform data structure contains information which is used only by the driver, and the driver allocates platform information fields which are of no use. Move the driver specific data to a new private structure and cleanup the platform data structure. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- include/sound/simple_card.h | 4 -- sound/soc/generic/simple-card.c | 14 +++++++++++--- 2 files changed, 11 insertions(+), 7 deletions(-) diff --git a/include/sound/simple_card.h b/include/sound/simple_card.h index 6c74527..e1ac996 100644 --- a/include/sound/simple_card.h +++ b/include/sound/simple_card.h @@ -29,10 +29,6 @@ struct asoc_simple_card_info { unsigned int daifmt; struct asoc_simple_dai cpu_dai; struct asoc_simple_dai codec_dai; - - /* used in simple-card.c */ - struct snd_soc_dai_link snd_link; - struct snd_soc_card snd_card; }; #endif /* __SIMPLE_CARD_H */ diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 90c6fd5..0f11c28 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -15,6 +15,14 @@ #include <linux/string.h> #include <sound/simple_card.h> +struct simple_card_data { + struct snd_soc_card snd_card; + unsigned int daifmt; + struct asoc_simple_dai cpu_dai; + struct asoc_simple_dai codec_dai; + struct snd_soc_dai_link snd_link; +}; + static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, struct asoc_simple_dai *set, unsigned int daifmt) @@ -39,7 +47,7 @@ static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct asoc_simple_card_info *info = + struct simple_card_data *info = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *codec = rtd->codec_dai; struct snd_soc_dai *cpu = rtd->cpu_dai; @@ -121,7 +129,7 @@ parse_error: } static int asoc_simple_card_parse_of(struct device_node *node, - struct asoc_simple_card_info *info, + struct simple_card_data *info, struct device *dev) { struct snd_soc_dai_link *dai_link = info->snd_card.dai_link; @@ -195,7 +203,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, static int asoc_simple_card_probe(struct platform_device *pdev) { - struct asoc_simple_card_info *priv; + struct simple_card_data *priv; struct snd_soc_dai_link *dai_link; struct device_node *np = pdev->dev.of_node; struct device *dev = &pdev->dev; -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 1

[alsa-devel] [PATCH v2 6/8] ASoC: simple-card: simplify code
by Jean-Francois Moine 22 Jan '14

22 Jan '14

In the non-DT sequence, the platform data is copied as a whole to the dynamic card info and the same pointer 'cinfo' is used to copy the platform information to the card. Use 'priv' as the pointer to the dynamic card info and copy only the useful information from the platform data. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/soc/generic/simple-card.c | 34 +++++++++++++--------- 1 file changed, 20 insertions(+), 14 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 9068ab4..90c6fd5 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -195,40 +195,42 @@ static int asoc_simple_card_parse_of(struct device_node *node, static int asoc_simple_card_probe(struct platform_device *pdev) { - struct asoc_simple_card_info *cinfo; + struct asoc_simple_card_info *priv; struct snd_soc_dai_link *dai_link; struct device_node *np = pdev->dev.of_node; struct device *dev = &pdev->dev; int ret; - cinfo = devm_kzalloc(dev, sizeof(*cinfo), GFP_KERNEL); - if (!cinfo) + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) return -ENOMEM; /* * init snd_soc_card */ - cinfo->snd_card.owner = THIS_MODULE; - cinfo->snd_card.dev = dev; - dai_link = &cinfo->snd_link; - cinfo->snd_card.dai_link = dai_link; - cinfo->snd_card.num_links = 1; + priv->snd_card.owner = THIS_MODULE; + priv->snd_card.dev = dev; + dai_link = &priv->snd_link; + priv->snd_card.dai_link = dai_link; + priv->snd_card.num_links = 1; if (np && of_device_is_available(np)) { - ret = asoc_simple_card_parse_of(np, cinfo, dev); + ret = asoc_simple_card_parse_of(np, priv, dev); if (ret < 0) { if (ret != -EPROBE_DEFER) dev_err(dev, "parse error %d\n", ret); return ret; } } else { - if (!dev->platform_data) { + struct asoc_simple_card_info *cinfo; + + cinfo = dev->platform_data; + if (!cinfo) { dev_err(dev, "no info for asoc-simple-card\n"); return -EINVAL; } - memcpy(cinfo, dev->platform_data, sizeof(*cinfo)); if (!cinfo->name || !cinfo->card || !cinfo->codec_dai.name || @@ -239,13 +241,17 @@ static int asoc_simple_card_probe(struct platform_device *pdev) return -EINVAL; } - cinfo->snd_card.name = cinfo->card; + priv->snd_card.name = cinfo->card; dai_link->name = cinfo->name; dai_link->stream_name = cinfo->name; dai_link->platform_name = cinfo->platform; dai_link->codec_name = cinfo->codec; dai_link->cpu_dai_name = cinfo->cpu_dai.name; dai_link->codec_dai_name = cinfo->codec_dai.name; + memcpy(&priv->cpu_dai, &cinfo->cpu_dai, + sizeof(priv->cpu_dai)); + memcpy(&priv->codec_dai, &cinfo->codec_dai, + sizeof(priv->codec_dai)); } /* @@ -253,9 +259,9 @@ static int asoc_simple_card_probe(struct platform_device *pdev) */ dai_link->init = asoc_simple_card_dai_init; - snd_soc_card_set_drvdata(&cinfo->snd_card, cinfo); + snd_soc_card_set_drvdata(&priv->snd_card, priv); - return devm_snd_soc_register_card(&pdev->dev, &cinfo->snd_card); + return devm_snd_soc_register_card(&pdev->dev, &priv->snd_card); } static const struct of_device_id asoc_simple_of_match[] = { -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 1

[alsa-devel] [PATCH v2 5/8] ASoC: simple-card: simplify code
by Jean-Francois Moine 22 Jan '14

22 Jan '14

The CPU and CODEC DAI names are still copied to the user info structure. Put them directly in the DAI links. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/soc/generic/simple-card.c | 28 ++++++++++++---------- 1 file changed, 16 insertions(+), 12 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 797696f..9068ab4 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -60,7 +60,8 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) static int asoc_simple_card_sub_parse_of(struct device_node *np, struct asoc_simple_dai *dai, - const struct device_node **p_node) + const struct device_node **p_node, + const char **name) { struct device_node *node; struct clk *clk; @@ -76,7 +77,7 @@ asoc_simple_card_sub_parse_of(struct device_node *np, *p_node = node; /* get dai->name */ - ret = snd_soc_of_get_dai_name(np, &dai->name); + ret = snd_soc_of_get_dai_name(np, name); if (ret < 0) goto parse_error; @@ -146,7 +147,8 @@ static int asoc_simple_card_parse_of(struct device_node *node, if (np) ret = asoc_simple_card_sub_parse_of(np, &info->cpu_dai, - &dai_link->cpu_of_node); + &dai_link->cpu_of_node, + &dai_link->cpu_dai_name); if (ret < 0) return ret; @@ -156,19 +158,21 @@ static int asoc_simple_card_parse_of(struct device_node *node, if (np) ret = asoc_simple_card_sub_parse_of(np, &info->codec_dai, - &dai_link->codec_of_node); + &dai_link->codec_of_node, + &dai_link->codec_dai_name); if (ret < 0) return ret; - if (!info->cpu_dai.name || !info->codec_dai.name) + if (!dai_link->cpu_dai_name || !dai_link->codec_dai_name) return -EINVAL; /* card name is created from CPU/CODEC dai name */ name = devm_kzalloc(dev, - strlen(info->cpu_dai.name) + - strlen(info->codec_dai.name) + 2, + strlen(dai_link->cpu_dai_name) + + strlen(dai_link->codec_dai_name) + 2, GFP_KERNEL); - sprintf(name, "%s-%s", info->cpu_dai.name, info->codec_dai.name); + sprintf(name, "%s-%s", dai_link->cpu_dai_name, + dai_link->codec_dai_name); info->snd_card.name = name; dai_link->name = dai_link->stream_name = name; @@ -178,11 +182,11 @@ static int asoc_simple_card_parse_of(struct device_node *node, dev_dbg(dev, "card-name : %s\n", name); dev_dbg(dev, "platform : %04x\n", info->daifmt); dev_dbg(dev, "cpu : %s / %04x / %d\n", - info->cpu_dai.name, + dai_link->cpu_dai_name, info->cpu_dai.fmt, info->cpu_dai.sysclk); dev_dbg(dev, "codec : %s / %04x / %d\n", - info->codec_dai.name, + dai_link->codec_dai_name, info->codec_dai.fmt, info->codec_dai.sysclk); @@ -240,13 +244,13 @@ static int asoc_simple_card_probe(struct platform_device *pdev) dai_link->stream_name = cinfo->name; dai_link->platform_name = cinfo->platform; dai_link->codec_name = cinfo->codec; + dai_link->cpu_dai_name = cinfo->cpu_dai.name; + dai_link->codec_dai_name = cinfo->codec_dai.name; } /* * init snd_soc_dai_link */ - dai_link->cpu_dai_name = cinfo->cpu_dai.name; - dai_link->codec_dai_name = cinfo->codec_dai.name; dai_link->init = asoc_simple_card_dai_init; snd_soc_card_set_drvdata(&cinfo->snd_card, cinfo); -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 1

[alsa-devel] [PATCH v2 4/8] ASoC: simple-card: simplify code
by Jean-Francois Moine 22 Jan '14

22 Jan '14

Have a cleaner code using a DAI link pointer. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/soc/generic/simple-card.c | 29 ++++++++++++---------- 1 file changed, 16 insertions(+), 13 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 89f83b3..797696f 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -123,6 +123,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, struct asoc_simple_card_info *info, struct device *dev) { + struct snd_soc_dai_link *dai_link = info->snd_card.dai_link; struct device_node *np; char *name; int ret; @@ -145,7 +146,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, if (np) ret = asoc_simple_card_sub_parse_of(np, &info->cpu_dai, - &info->snd_link.cpu_of_node); + &dai_link->cpu_of_node); if (ret < 0) return ret; @@ -155,7 +156,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, if (np) ret = asoc_simple_card_sub_parse_of(np, &info->codec_dai, - &info->snd_link.codec_of_node); + &dai_link->codec_of_node); if (ret < 0) return ret; @@ -169,10 +170,10 @@ static int asoc_simple_card_parse_of(struct device_node *node, GFP_KERNEL); sprintf(name, "%s-%s", info->cpu_dai.name, info->codec_dai.name); info->snd_card.name = name; - info->snd_link.name = info->snd_link.stream_name = name; + dai_link->name = dai_link->stream_name = name; /* simple-card assumes platform == cpu */ - info->snd_link.platform_of_node = info->snd_link.cpu_of_node; + dai_link->platform_of_node = dai_link->cpu_of_node; dev_dbg(dev, "card-name : %s\n", name); dev_dbg(dev, "platform : %04x\n", info->daifmt); @@ -191,6 +192,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, static int asoc_simple_card_probe(struct platform_device *pdev) { struct asoc_simple_card_info *cinfo; + struct snd_soc_dai_link *dai_link; struct device_node *np = pdev->dev.of_node; struct device *dev = &pdev->dev; int ret; @@ -204,8 +206,9 @@ static int asoc_simple_card_probe(struct platform_device *pdev) */ cinfo->snd_card.owner = THIS_MODULE; cinfo->snd_card.dev = dev; - cinfo->snd_card.dai_link = &cinfo->snd_link; - cinfo->snd_card.num_links = 1; + dai_link = &cinfo->snd_link; + cinfo->snd_card.dai_link = dai_link; + cinfo->snd_card.num_links = 1; if (np && of_device_is_available(np)) { @@ -233,18 +236,18 @@ static int asoc_simple_card_probe(struct platform_device *pdev) } cinfo->snd_card.name = cinfo->card; - cinfo->snd_link.name = cinfo->name; - cinfo->snd_link.stream_name = cinfo->name; - cinfo->snd_link.platform_name = cinfo->platform; - cinfo->snd_link.codec_name = cinfo->codec; + dai_link->name = cinfo->name; + dai_link->stream_name = cinfo->name; + dai_link->platform_name = cinfo->platform; + dai_link->codec_name = cinfo->codec; } /* * init snd_soc_dai_link */ - cinfo->snd_link.cpu_dai_name = cinfo->cpu_dai.name; - cinfo->snd_link.codec_dai_name = cinfo->codec_dai.name; - cinfo->snd_link.init = asoc_simple_card_dai_init; + dai_link->cpu_dai_name = cinfo->cpu_dai.name; + dai_link->codec_dai_name = cinfo->codec_dai.name; + dai_link->init = asoc_simple_card_dai_init; snd_soc_card_set_drvdata(&cinfo->snd_card, cinfo); -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 1

[alsa-devel] [PATCH v2 3/8] ASoC: simple-card: simplify code
by Jean-Francois Moine 22 Jan '14

22 Jan '14

The OF pointers are put in the stack and then copied to the card descriptor. Put them directly at their right place. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/soc/generic/simple-card.c | 55 ++++++++-------------- 1 file changed, 20 insertions(+), 35 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 7c5dc73..89f83b3 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -60,8 +60,9 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) static int asoc_simple_card_sub_parse_of(struct device_node *np, struct asoc_simple_dai *dai, - struct device_node **node) + const struct device_node **p_node) { + struct device_node *node; struct clk *clk; int ret; @@ -69,9 +70,10 @@ asoc_simple_card_sub_parse_of(struct device_node *np, * get node via "sound-dai = <&phandle port>" * it will be used as xxx_of_node on soc_bind_dai_link() */ - *node = of_parse_phandle(np, "sound-dai", 0); - if (!*node) + node = of_parse_phandle(np, "sound-dai", 0); + if (!node) return -ENODEV; + *p_node = node; /* get dai->name */ ret = snd_soc_of_get_dai_name(np, &dai->name); @@ -104,7 +106,7 @@ asoc_simple_card_sub_parse_of(struct device_node *np, "system-clock-frequency", &dai->sysclk); } else { - clk = of_clk_get(*node, 0); + clk = of_clk_get(node, 0); if (!IS_ERR(clk)) dai->sysclk = clk_get_rate(clk); } @@ -112,17 +114,14 @@ asoc_simple_card_sub_parse_of(struct device_node *np, ret = 0; parse_error: - of_node_put(*node); + of_node_put(node); return ret; } static int asoc_simple_card_parse_of(struct device_node *node, struct asoc_simple_card_info *info, - struct device *dev, - struct device_node **of_cpu, - struct device_node **of_codec, - struct device_node **of_platform) + struct device *dev) { struct device_node *np; char *name; @@ -146,7 +145,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, if (np) ret = asoc_simple_card_sub_parse_of(np, &info->cpu_dai, - of_cpu); + &info->snd_link.cpu_of_node); if (ret < 0) return ret; @@ -156,7 +155,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, if (np) ret = asoc_simple_card_sub_parse_of(np, &info->codec_dai, - of_codec); + &info->snd_link.codec_of_node); if (ret < 0) return ret; @@ -173,7 +172,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, info->snd_link.name = info->snd_link.stream_name = name; /* simple-card assumes platform == cpu */ - *of_platform = *of_cpu; + info->snd_link.platform_of_node = info->snd_link.cpu_of_node; dev_dbg(dev, "card-name : %s\n", name); dev_dbg(dev, "platform : %04x\n", info->daifmt); @@ -193,34 +192,29 @@ static int asoc_simple_card_probe(struct platform_device *pdev) { struct asoc_simple_card_info *cinfo; struct device_node *np = pdev->dev.of_node; - struct device_node *of_cpu, *of_codec, *of_platform; struct device *dev = &pdev->dev; int ret; - cinfo = NULL; - of_cpu = NULL; - of_codec = NULL; - of_platform = NULL; - cinfo = devm_kzalloc(dev, sizeof(*cinfo), GFP_KERNEL); if (!cinfo) return -ENOMEM; + /* + * init snd_soc_card + */ + cinfo->snd_card.owner = THIS_MODULE; + cinfo->snd_card.dev = dev; + cinfo->snd_card.dai_link = &cinfo->snd_link; + cinfo->snd_card.num_links = 1; + if (np && of_device_is_available(np)) { - cinfo->snd_card.dev = dev; - ret = asoc_simple_card_parse_of(np, cinfo, dev, - &of_cpu, - &of_codec, - &of_platform); + ret = asoc_simple_card_parse_of(np, cinfo, dev); if (ret < 0) { if (ret != -EPROBE_DEFER) dev_err(dev, "parse error %d\n", ret); return ret; } - cinfo->snd_link.cpu_of_node = of_cpu; - cinfo->snd_link.codec_of_node = of_codec; - cinfo->snd_link.platform_of_node = of_platform; } else { if (!dev->platform_data) { dev_err(dev, "no info for asoc-simple-card\n"); @@ -228,8 +222,6 @@ static int asoc_simple_card_probe(struct platform_device *pdev) } memcpy(cinfo, dev->platform_data, sizeof(*cinfo)); - cinfo->snd_card.dev = dev; - if (!cinfo->name || !cinfo->card || !cinfo->codec_dai.name || @@ -254,13 +246,6 @@ static int asoc_simple_card_probe(struct platform_device *pdev) cinfo->snd_link.codec_dai_name = cinfo->codec_dai.name; cinfo->snd_link.init = asoc_simple_card_dai_init; - /* - * init snd_soc_card - */ - cinfo->snd_card.owner = THIS_MODULE; - cinfo->snd_card.dai_link = &cinfo->snd_link; - cinfo->snd_card.num_links = 1; - snd_soc_card_set_drvdata(&cinfo->snd_card, cinfo); return devm_snd_soc_register_card(&pdev->dev, &cinfo->snd_card); -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 1

[alsa-devel] [PATCH v2 2/8] ASoC: simple-card: simplify code
by Jean-Francois Moine 22 Jan '14

22 Jan '14

The DT values are copied to the non-DT structure before being moved to the card structure. Set directly the DT values in the card and move the non-DT copy to the non-DT sequence. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/soc/generic/simple-card.c | 22 ++++++++++++---------- 1 file changed, 12 insertions(+), 10 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index f0784ca..7c5dc73 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -169,12 +169,13 @@ static int asoc_simple_card_parse_of(struct device_node *node, strlen(info->codec_dai.name) + 2, GFP_KERNEL); sprintf(name, "%s-%s", info->cpu_dai.name, info->codec_dai.name); - info->name = info->card = name; + info->snd_card.name = name; + info->snd_link.name = info->snd_link.stream_name = name; /* simple-card assumes platform == cpu */ *of_platform = *of_cpu; - dev_dbg(dev, "card-name : %s\n", info->card); + dev_dbg(dev, "card-name : %s\n", name); dev_dbg(dev, "platform : %04x\n", info->daifmt); dev_dbg(dev, "cpu : %s / %04x / %d\n", info->cpu_dai.name, @@ -217,6 +218,9 @@ static int asoc_simple_card_probe(struct platform_device *pdev) dev_err(dev, "parse error %d\n", ret); return ret; } + cinfo->snd_link.cpu_of_node = of_cpu; + cinfo->snd_link.codec_of_node = of_codec; + cinfo->snd_link.platform_of_node = of_platform; } else { if (!dev->platform_data) { dev_err(dev, "no info for asoc-simple-card\n"); @@ -235,26 +239,24 @@ static int asoc_simple_card_probe(struct platform_device *pdev) dev_err(dev, "insufficient asoc_simple_card_info settings\n"); return -EINVAL; } + + cinfo->snd_card.name = cinfo->card; + cinfo->snd_link.name = cinfo->name; + cinfo->snd_link.stream_name = cinfo->name; + cinfo->snd_link.platform_name = cinfo->platform; + cinfo->snd_link.codec_name = cinfo->codec; } /* * init snd_soc_dai_link */ - cinfo->snd_link.name = cinfo->name; - cinfo->snd_link.stream_name = cinfo->name; cinfo->snd_link.cpu_dai_name = cinfo->cpu_dai.name; - cinfo->snd_link.platform_name = cinfo->platform; - cinfo->snd_link.codec_name = cinfo->codec; cinfo->snd_link.codec_dai_name = cinfo->codec_dai.name; - cinfo->snd_link.cpu_of_node = of_cpu; - cinfo->snd_link.codec_of_node = of_codec; - cinfo->snd_link.platform_of_node = of_platform; cinfo->snd_link.init = asoc_simple_card_dai_init; /* * init snd_soc_card */ - cinfo->snd_card.name = cinfo->card; cinfo->snd_card.owner = THIS_MODULE; cinfo->snd_card.dai_link = &cinfo->snd_link; cinfo->snd_card.num_links = 1; -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 1

[alsa-devel] [PATCH v2 1/8] ASoC: simple-card: simplify code
by Jean-Francois Moine 22 Jan '14

22 Jan '14

The check of the mandatory fields is done for DT in its specific sequence. Move the global check to the non-DT sequence. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/soc/generic/simple-card.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 2a1b1b5..f0784ca 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -225,16 +225,16 @@ static int asoc_simple_card_probe(struct platform_device *pdev) memcpy(cinfo, dev->platform_data, sizeof(*cinfo)); cinfo->snd_card.dev = dev; - } - if (!cinfo->name || - !cinfo->card || - !cinfo->codec_dai.name || - !(cinfo->codec || of_codec) || - !(cinfo->platform || of_platform) || - !(cinfo->cpu_dai.name || of_cpu)) { - dev_err(dev, "insufficient asoc_simple_card_info settings\n"); - return -EINVAL; + if (!cinfo->name || + !cinfo->card || + !cinfo->codec_dai.name || + !cinfo->codec || + !cinfo->platform || + !cinfo->cpu_dai.name) { + dev_err(dev, "insufficient asoc_simple_card_info settings\n"); + return -EINVAL; + } } /* -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 1

[alsa-devel] [PATCH v2 0/8] ASoC: simple-card: simplify code
by Jean-Francois Moine 22 Jan '14

22 Jan '14

This patch set simplifies the code and make easier a possible multi-DAI links extension. -- v2: - split the changes into 8 patches (Mark Brown request) - remove the dynamic DAI link array allocation - fix lack of init function when DT use -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 1

[alsa-devel] [PATCH v2 0/6] ASoC: Samsung: Fix build errors and warnings
by Sachin Kamat 22 Jan '14

22 Jan '14

This series attempts to fix several build errors and warnings that came up while building for different Samsung defconfigs. Some of these could possibly be applied to stable kernels as well (like patch 2 & 6). This series is based on latest linux-next and is build tested on various Samsung configs. Only change in v2 is that CONFIG_OF has been dropped in patch 6 as suggested by reviewers. This made earlier patch 7 redundant and has been dropped in this version of the series. Sachin Kamat (6): ASoC: Samsung: Fix build regressions due to gpio re-org ASoC: Samsung: s3c-i2s-v2: Fix build error ASoC: Samsung: h1940_uda1380: Remove unused variable ASoC: Samsung: h1940_uda1380: Fix build warning ASoC: Samsung: rx1950_uda1380: Remove unused variable ASoC: Samsung: smdk_wm8994: Fix build error sound/soc/samsung/h1940_uda1380.c | 4 ++-- sound/soc/samsung/neo1973_wm8753.c | 1 + sound/soc/samsung/rx1950_uda1380.c | 2 +- sound/soc/samsung/s3c-i2s-v2.c | 6 +++--- sound/soc/samsung/s3c2412-i2s.c | 2 ++ sound/soc/samsung/s3c24xx-i2s.c | 2 ++ sound/soc/samsung/smartq_wm8987.c | 1 + sound/soc/samsung/smdk_wm8994.c | 4 +--- 8 files changed, 13 insertions(+), 9 deletions(-) -- 1.7.9.5

2 11

[alsa-devel] Hello list! My first question!
by gianluca 22 Jan '14

22 Jan '14

Hello list! I am working on a custom embedded ARM based-board (iMX28) with a SGTL5000 audio codec (correctly seen by kernel 3.12.2) but the lineout lines (LINEOUT_L and LINEOUT_R) are inputs for a Texas Instruments' Volume Controller named LM48100Q i2c-bus based. I would like to know if there are any chances to add a ALSA Driver for this VOLUME CONTROLLER. I am looking at sounds/i2c/other/pt2258.c driver and it looks very similar to lm48100. Actually this lm48100 has 5 registers but only 3 are useful to me: MODE CONTROL (how many inputs it uses (1 or 2) or POWER ON/OFF), and one register for INPUT1 VOLUME SETTING and another one for INPUT2 VOLUME SETTING. Or I will be forced to use this in a userspace driver accessing the i2c, but with no integration in any linux-based audio mixer alsa tool. Can you help me on this? Regards, -- ,,, (o o) ======oOO==(_)==OOo====== Gianluca Renzi R&D phone: +39.0542.609120 fax: +39.0542.609212 .oooO Oooo. ======( )==( )======= \ ( ) / \_) (_/ =================== | I would like to | | fix the world, | | but they're not | | giving me the | \ source code! / --------------- ¯\_(".)_/¯

1 0

[alsa-devel] Fw: [Bug 1155202] [Intel DZ77SL-50K, Intel PantherPoint HDMI, Digital Out, HDMI] No sound at all
by Niraj Kulkarni 22 Jan '14

22 Jan '14

Let me try to explain my findings: > How does this magic sequence come? By dumping Win7 driver's init sequence and reproducing it through ALSA driver. >why does it need to set UNSOL flag at first? No such need, I've put it after regular init so that only affected chipsets are reset again. Putting this anywhere around init code works (tried it). >Is the while loop guaranteed to quit for a reasonable time? Being household user, I can only confirm on my chipset. Only a thorough testing may ensure sanity. >Can't it be simply calling azx_reset() twice? It is interesting hardware bug which kicks in only during first init after power-on. So if  Win7 driver in qemu inits the board, and I simply plugin "unchanged" hda_intel afterwards, it still detects codecs. But multiple normal resets do not have any effect. Being no hardware expert, I can only speculate that tight spin loop after reset has the crux. A normal delay call there does not detect codec. >And, yet another question comes up: doesn't S4 need the similar workaround? If S4 works, what's the difference? Haven't tried with sleep-wake or hibernate-restore sequence yet. -- You received this bug notification because you are subscribed to the bug report. https://bugs.launchpad.net/bugs/1155202 Title:  [Intel DZ77SL-50K, Intel PantherPoint HDMI, Digital Out, HDMI] No  sound at all Status in “alsa-driver” package in Ubuntu:  Incomplete Bug description:  I am using ubuntu 10.04  I expected sound  Got no sound at all  I am very new to ubuntu so I have no idea how to find the version of  the package I am using  ProblemType: Bug  DistroRelease: Ubuntu 12.04  Package: alsa-base 1.0.25+dfsg-0ubuntu1.1  ProcVersionSignature: Ubuntu 3.2.0-38.61-generic-pae 3.2.37  Uname: Linux 3.2.0-38-generic-pae i686  AlsaVersion: Advanced Linux Sound Architecture Driver Version 1.0.24.  AplayDevices:   **** List of PLAYBACK Hardware Devices ****   card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0]     Subdevices: 1/1     Subdevice #0: subdevice #0  ApportVersion: 2.0.1-0ubuntu17.1  Architecture: i386  ArecordDevices: **** List of CAPTURE Hardware Devices ****  AudioDevicesInUse:   USER        PID ACCESS COMMAND   /dev/snd/controlC0:  jan        1750 F.... pulseaudio  Card0.Amixer.info:   Card hw:0 'PCH'/'HDA Intel PCH at 0xf7c30000 irq 45'     Mixer name   : 'Intel PantherPoint HDMI'     Components   : 'HDA:80862806,80860101,00100000'     Controls      : 7     Simple ctrls  : 2  CurrentDmesg: [   24.291482] eth0: no IPv6 routers present  Date: Thu Mar 14 08:53:13 2013  InstallationMedia: Ubuntu 12.04.1 LTS "Precise Pangolin" - Release i386 (20120817.3)  MarkForUpload: True  PackageArchitecture: all  ProcEnviron:   TERM=xterm   PATH=(custom, no user)   LANG=en_US.UTF-8   SHELL=/bin/bash  SourcePackage: alsa-driver  Symptom: audio  Symptom_AlsaPlaybackTest: ALSA playback test through plughw:PCH failed  Symptom_Card: Built-in Audio - HDA Intel PCH  Symptom_DevicesInUse:   1750  jan       F.... pulseaudio       PID ACCESS COMMAND  Symptom_Jack: Digital Out, HDMI  Symptom_Type: No sound at all  Title: [, Intel PantherPoint HDMI, Digital Out, HDMI] No sound at all  UpgradeStatus: No upgrade log present (probably fresh install)  dmi.bios.date: 03/19/2012  dmi.bios.vendor: Intel Corporation  dmi.bios.version: SLZ7710H.86A.0055.2012.0319.2140  dmi.board.name: DZ77SL-50K  dmi.board.vendor: Intel Corporation  dmi.board.version: AAG55115-300  dmi.chassis.type: 3  dmi.modalias: dmi:bvnIntelCorporation:bvrSLZ7710H.86A.0055.2012.0319.2140:bd03/19/2012:svn:pn:pvr:rvnIntelCorporation:rnDZ77SL-50K:rvrAAG55115-300:cvn:ct3:cvr: To manage notifications about this bug go to: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1155202/+subscri…

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[alsa-devel] [RFC PATCH] ALSA: hda - Detect codecs on Intel DZ77SL-50K
by David Henningsson 22 Jan '14

22 Jan '14

From: Niraj <kulkarniniraj(a)rediffmail.com> Apparently this board seems to need some extra initialization before a proper codec probe value can be read. BugLink: https://bugs.launchpad.net/bugs/1155202 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_intel.c | 6 ++++++ 1 file changed, 6 insertions(+) Hi Takashi, Here's a patch from Niraj, which has found this code below to make audio work on the Intel DZ77SL-50K board. Without it, no codecs will be detected. I'm not sure if you like it this way or if you want me or Niraj to change anything, or debug further. Or maybe add a new quirk for this particular device? diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index fa2879a..2bf46e9 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1306,6 +1306,12 @@ static void azx_init_chip(struct azx *chip, int full_reset) /* reset controller */ azx_reset(chip, full_reset); + if (!chip->codec_mask) { + azx_writel(chip, GCTL, 0x100); + while(azx_readl(chip, GCTL) != 0x0); + azx_writel(chip, GCTL, 0x1); + azx_reset(chip, full_reset); + } /* initialize interrupts */ azx_int_clear(chip); -- 1.7.9.5

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[alsa-devel] [PATCH v2] ASoC: wm_adsp: Add debug print to note that the DSP has shutdown
by Charles Keepax 22 Jan '14

22 Jan '14

It can be useful for debugging purposes to see at what point the DSP has powered down, so add a message to inform us of this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 4fbcab6..90d7baf 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1637,6 +1637,8 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, list_del(&alg_region->list); kfree(alg_region); } + + adsp_dbg(dsp, "Shutdown complete\n"); break; default: -- 1.7.2.5

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[alsa-devel] [PATCH 0/7] ASoC: Samsung: Fix build errors and warnings
by Sachin Kamat 22 Jan '14

22 Jan '14

This series attempts to fix several build errors and warnings that came up while building for deifferent Samsung defconfigs. Some of these could possibly be applied to stable kernels as well (like patch 2 & 6). This series is based on latest linux-next and is build tested on various Samsung configs. Sachin Kamat (7): ASoC: Samsung: Fix build regressions due to gpio re-org ASoC: Samsung: s3c-i2s-v2: Fix build error ASoC: Samsung: h1940_uda1380: Remove unused variable ASoC: Samsung: h1940_uda1380: Fix build warning ASoC: Samsung: rx1950_uda1380: Remove unused variable ASoC: Samsung: smdk_wm8994: Fix build error ASoC: Samsung: smdk_wm8994: Fix unused variable warning sound/soc/samsung/h1940_uda1380.c | 4 ++-- sound/soc/samsung/neo1973_wm8753.c | 1 + sound/soc/samsung/rx1950_uda1380.c | 2 +- sound/soc/samsung/s3c-i2s-v2.c | 6 +++--- sound/soc/samsung/s3c2412-i2s.c | 2 ++ sound/soc/samsung/s3c24xx-i2s.c | 2 ++ sound/soc/samsung/smartq_wm8987.c | 1 + sound/soc/samsung/smdk_wm8994.c | 12 ++++++------ 8 files changed, 18 insertions(+), 12 deletions(-) -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: wm8804: Allow control of master clock divider in PLL generation
by Daniel Matuschek 22 Jan '14

22 Jan '14

WM8804 can run with PLL frequencies of 256xfs and 128xfs for most sample rates. At 192kHz only 128xfs is supported. The existing driver selects 128xfs automatically for some lower samples rates. By using an additional mclk_div divider, it is now possible to control the behaviour. This allows using 256xfs PLL frequency on all sample rates up to 96kHz. It should allow lower jitter and better signal quality. The behavior has to be controlled by the sound card driver, because some sample frequency share the same setting. e.g. 192kHz and 96kHz use 24.576MHz master clock. The only difference is the MCLK divider. Signed-off-by: Daniel Matuschek <daniel(a)matuschek.net> --- sound/soc/codecs/wm8804.c | 17 ++++++++++++++--- sound/soc/codecs/wm8804.h | 4 ++++ 2 files changed, 18 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 1704b1e..4619bf8 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -63,6 +63,7 @@ struct wm8804_priv { struct regmap *regmap; struct regulator_bulk_data supplies[WM8804_NUM_SUPPLIES]; struct notifier_block disable_nb[WM8804_NUM_SUPPLIES]; + int mclk_div; }; static int txsrc_get(struct snd_kcontrol *kcontrol, @@ -318,7 +319,7 @@ static struct { #define FIXED_PLL_SIZE ((1ULL << 22) * 10) static int pll_factors(struct pll_div *pll_div, unsigned int target, - unsigned int source) + unsigned int source, unsigned int mclk_div) { u64 Kpart; unsigned long int K, Ndiv, Nmod, tmp; @@ -330,7 +331,8 @@ static int pll_factors(struct pll_div *pll_div, unsigned int target, */ for (i = 0; i < ARRAY_SIZE(post_table); i++) { tmp = target * post_table[i].div; - if (tmp >= 90000000 && tmp <= 100000000) { + if ((tmp >= 90000000 && tmp <= 100000000) && + (mclk_div == post_table[i].mclkdiv)) { pll_div->freqmode = post_table[i].freqmode; pll_div->mclkdiv = post_table[i].mclkdiv; target *= post_table[i].div; @@ -387,8 +389,12 @@ static int wm8804_set_pll(struct snd_soc_dai *dai, int pll_id, } else { int ret; struct pll_div pll_div; + struct wm8804_priv *wm8804; - ret = pll_factors(&pll_div, freq_out, freq_in); + wm8804 = snd_soc_codec_get_drvdata(codec); + + ret = pll_factors(&pll_div, freq_out, freq_in, + wm8804->mclk_div); if (ret) return ret; @@ -452,6 +458,7 @@ static int wm8804_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) { struct snd_soc_codec *codec; + struct wm8804_priv *wm8804; codec = dai->codec; switch (div_id) { @@ -459,6 +466,10 @@ static int wm8804_set_clkdiv(struct snd_soc_dai *dai, snd_soc_update_bits(codec, WM8804_PLL5, 0x30, (div & 0x3) << 4); break; + case WM8804_MCLK_DIV: + wm8804 = snd_soc_codec_get_drvdata(codec); + wm8804->mclk_div = div; + break; default: dev_err(dai->dev, "Unknown clock divider: %d\n", div_id); return -EINVAL; diff --git a/sound/soc/codecs/wm8804.h b/sound/soc/codecs/wm8804.h index 8ec14f5..e72d4f4 100644 --- a/sound/soc/codecs/wm8804.h +++ b/sound/soc/codecs/wm8804.h @@ -57,5 +57,9 @@ #define WM8804_CLKOUT_SRC_OSCCLK 4 #define WM8804_CLKOUT_DIV 1 +#define WM8804_MCLK_DIV 2 + +#define WM8804_MCLKDIV_256FS 0 +#define WM8804_MCLKDIV_128FS 1 #endif /* _WM8804_H */ -- 1.7.9.5

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[alsa-devel] Got headphone out working via hda-verb setting on MacPro1, 1
by Nicolai Beuermann 22 Jan '14

22 Jan '14

Hello, hda-analyzer told me to write a note to this list when settings are successful. My setup: MacPro 1,1 Kernel 3.10.17 (gentoo sources) ALSA k3.10.17 Had to set /usr/bin/hda-verb /dev/snd/hwC0D0 0x18 SET_PIN_WIDGET_CONTROL 0xc1 to get the headphone out at the front working. Wrote a bit (in german) about it: http://www.nico-beuermann.de/blogging/archives/117-Hier-gibt-es-was-auf-die… regards Nicolai Beuermann -- mailto: mail(a)nico-beuermann.de online: http://www.nico-beuermann.de GnuPG Fingerabdruck: 56DA 4E32 3A4A 52AC B769 DFC2 BF3E 9805 09BB 4259

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[alsa-devel] [PATCH v2 0/4] ASoC: tlv320aic32x4: DT support
by Markus Pargmann 22 Jan '14

22 Jan '14

Hi, The series adds basic DT support for the TI tlv320aic32x4 codec. v2 adds a new patch to support regulators. The codec now decides on the given regulators if LDO for AVDD should be enabled. 'ti,ldo-enable' was removed. Regards, Markus Changes in v2: - Regulator support - Fix documentation for gpio property Markus Pargmann (4): ASoC: tlv320aic32x4: Use signed int mixer controls ASoC: tlv320aic32x4: DT support ASoC: tlv320aic32x4: Support for master clock ASoC: tlv320aic32x4: Support for regulators .../devicetree/bindings/sound/tlv320aic32x4.txt | 28 +++++ sound/soc/codecs/tlv320aic32x4.c | 126 +++++++++++++++++++-- 2 files changed, 144 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic32x4.txt -- 1.8.5.2

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[alsa-devel] [PATCH] ALSA: cs46xx: Fix memory leak at destructor
by Takashi Iwai 22 Jan '14

22 Jan '14

The release of module object itself was forgotten. Spotted by COVERIY CID 1162828. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/cs46xx/cs46xx_lib.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index f18e5878f58b..062398ec5335 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -369,6 +369,7 @@ static void free_module_desc(struct dsp_module_desc *module) kfree(module->segments[i].data); kfree(module->segments); } + kfree(module); } /* firmware binary format: -- 1.8.5.2

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[alsa-devel] [PATCH] alsa-utils: fix timespec to msec conversion
by Olivier Langlois 22 Jan '14

22 Jan '14

There is 1M nsec in a msec. Signed-off-by: Olivier Langlois <olivier(a)trillion01.com> --- aplay/aplay.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/aplay/aplay.c b/aplay/aplay.c index 69e8bda..33a8ed5 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -1509,7 +1509,7 @@ static void xrun(void) timermsub(&now, &tstamp, &diff); fprintf(stderr, _("%s!!! (at least %.3f ms long)\n"), stream == SND_PCM_STREAM_PLAYBACK ? _("underrun") : _("overrun"), - diff.tv_sec * 1000 + diff.tv_nsec / 10000000.0); + diff.tv_sec * 1000 + diff.tv_nsec / 1000000.0); #else fprintf(stderr, "%s !!!\n", _("underrun")); #endif -- 1.8.5.3

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[alsa-devel] [PATCH - my module 1/1] control_hw: fix potential memory leak
by vanyacpp＠gmail.com 22 Jan '14

22 Jan '14

From: Ivan Sorokin <vanyacpp(a)gmail.com> Signed-off-by: Ivan Sorokin <vanyacpp(a)gmail.com> diff --git a/src/control/control_hw.c b/src/control/control_hw.c index 148097f..dfc9dcd 100644 --- a/src/control/control_hw.c +++ b/src/control/control_hw.c @@ -240,8 +240,10 @@ static int snd_ctl_hw_elem_tlv(snd_ctl_t *handle, int op_flag, return -errno; } if (op_flag == 0) { - if (xtlv->tlv[1] + 2 * sizeof(unsigned int) > tlv_size) + if (xtlv->tlv[1] + 2 * sizeof(unsigned int) > tlv_size) { + free(xtlv); return -EFAULT; + } memcpy(tlv, xtlv->tlv, xtlv->tlv[1] + 2 * sizeof(unsigned int)); } free(xtlv); -- 1.8.1.2

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[alsa-devel] [ISSUE] crackles and clicks
by Eric F 22 Jan '14

22 Jan '14

Hello, I tried my DAC (Tenor TE8802 chipset) in USB on Archlinux (LTS 3.10.26). I can hear regularly crackles ans clicks. I've tried all available USB ports, other kernels (3.12.6, 3.12.7), other distro (ubuntu), without luck... The DAC is Teac UD-H01. here is lsusb verbose : http://ix.io/9Q4 Feel free to ask for more precisions, I'll give you all you need. I thank you very much. Eric

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[alsa-devel] HD audio problem (fwd)
by Russ Whitaker 22 Jan '14

22 Jan '14

Still not heard from you. WHY? ---------- Forwarded message ---------- Date: Mon, 30 Dec 2013 20:20:31 -0800 (PST) From: Russ Whitaker <russ(a)ashlandhome.net> To: alsa-devel(a)alsa-project.org Subject: HD audio problem (fwd) Have not heard back from you. What should I do? ---------- Forwarded message ---------- Date: Thu, 19 Dec 2013 23:48:19 -0800 (PST) From: Russ Whitaker <russ(a)ashlandhome.net> To: alsa-devel(a)alsa-project.org Subject: HD audio problem Hi Bought a new mini-itx mb: ECS KBN-I On board sound chip: VT1705CF Installed kernel linux-3.10.24 Otherwise distro is Slackware64 14.1 mplayer plays .mp3 files just fine. No sound from video clips that use Adobe flashplayer, and I get this error message: ALSA lib pcm_dmix.c:1022:(snd_pcm_dmix_open) unable to open slave I think the problem is: card0 and card1 are interchanged as card0 is listed as the HDMI video output. How to fixit, please. Thank you, Russ

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[alsa-devel] [PATCH 2/2] ASoC: fsl_ssi: Do not print 'baud clock' error message all the time
by Fabio Estevam 21 Jan '14

21 Jan '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> Currently everytime we get the following message on boot: fsl-ssi-dai 202c000.ssi: could not get baud clock: -2 This is not really useful information to get on every boot, so make it a debug message instead. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 7b1772d..3155611 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1407,7 +1407,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) */ ssi_private->baudclk = devm_clk_get(&pdev->dev, "baud"); if (IS_ERR(ssi_private->baudclk)) - dev_warn(&pdev->dev, "could not get baud clock: %ld\n", + dev_dbg(&pdev->dev, "could not get baud clock: %ld\n", PTR_ERR(ssi_private->baudclk)); else clk_prepare_enable(ssi_private->baudclk); -- 1.8.1.2

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[alsa-devel] [PATCH] ASoC: fsl: use snd_soc_dai_init_dma_data()
by Xiubo Li 21 Jan '14

21 Jan '14

Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/fsl/fsl_spdif.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 4d075f1..8f36f49 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -911,8 +911,8 @@ static int fsl_spdif_dai_probe(struct snd_soc_dai *dai) { struct fsl_spdif_priv *spdif_private = snd_soc_dai_get_drvdata(dai); - dai->playback_dma_data = &spdif_private->dma_params_tx; - dai->capture_dma_data = &spdif_private->dma_params_rx; + snd_soc_dai_init_dma_data(dai, &spdif_private->dma_params_tx, + &spdif_private->dma_params_rx); snd_soc_add_dai_controls(dai, fsl_spdif_ctrls, ARRAY_SIZE(fsl_spdif_ctrls)); -- 1.8.4

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[alsa-devel] [PATCH 1/3] ASoC: wm5102: Improve EQ coefficient controls
by Charles Keepax 21 Jan '14

21 Jan '14

The EQ coefficient binary controls overlapped with the volume controls for the B4 and B5 volumes, which were controllable from either the coefficient control or the volume control itself. This patch adds controls for the mode and moves the coefficient control to only cover the coefficients. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 17 ++++++++--------- 1 files changed, 8 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index a08e8bf..ac19f62 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -685,15 +685,8 @@ ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), -SND_SOC_BYTES_MASK("EQ1 Coefficients", ARIZONA_EQ1_1, 21, - ARIZONA_EQ1_ENA_MASK), -SND_SOC_BYTES_MASK("EQ2 Coefficients", ARIZONA_EQ2_1, 21, - ARIZONA_EQ2_ENA_MASK), -SND_SOC_BYTES_MASK("EQ3 Coefficients", ARIZONA_EQ3_1, 21, - ARIZONA_EQ3_ENA_MASK), -SND_SOC_BYTES_MASK("EQ4 Coefficients", ARIZONA_EQ4_1, 21, - ARIZONA_EQ4_ENA_MASK), - +SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 18), +SOC_SINGLE("EQ1 Mode Switch", ARIZONA_EQ1_2, ARIZONA_EQ1_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT, @@ -705,6 +698,8 @@ SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, 24, 0, eq_tlv), +SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 18), +SOC_SINGLE("EQ2 Mode Switch", ARIZONA_EQ2_2, ARIZONA_EQ2_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT, @@ -716,6 +711,8 @@ SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, 24, 0, eq_tlv), +SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 18), +SOC_SINGLE("EQ3 Mode Switch", ARIZONA_EQ3_2, ARIZONA_EQ3_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT, @@ -727,6 +724,8 @@ SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, 24, 0, eq_tlv), +SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 18), +SOC_SINGLE("EQ4 Mode Switch", ARIZONA_EQ4_2, ARIZONA_EQ4_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT, -- 1.7.2.5

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[alsa-devel] [PATCH] ASoC: wm5110: Extend SYSCLK patch file for rev D
by Charles Keepax 21 Jan '14

21 Jan '14

Latest evaluation of the the device has given some patch file additions for improved performance. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5110.c | 48 +++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 48 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 34fbb03..92f1b40 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -43,6 +43,54 @@ static const struct reg_default wm5110_sysclk_revd_patch[] = { { 0x3133, 0x1201 }, { 0x3183, 0x1501 }, { 0x31D3, 0x1401 }, + { 0x0049, 0x01ea }, + { 0x004a, 0x01f2 }, + { 0x0057, 0x01e7 }, + { 0x0058, 0x01fb }, + { 0x33ce, 0xc4f5 }, + { 0x33cf, 0x1361 }, + { 0x33d0, 0x0402 }, + { 0x33d1, 0x4700 }, + { 0x33d2, 0x026d }, + { 0x33d3, 0xff00 }, + { 0x33d4, 0x026d }, + { 0x33d5, 0x0101 }, + { 0x33d6, 0xc4f5 }, + { 0x33d7, 0x0361 }, + { 0x33d8, 0x0402 }, + { 0x33d9, 0x6701 }, + { 0x33da, 0xc4f5 }, + { 0x33db, 0x136f }, + { 0x33dc, 0xc4f5 }, + { 0x33dd, 0x134f }, + { 0x33de, 0xc4f5 }, + { 0x33df, 0x131f }, + { 0x33e0, 0x026d }, + { 0x33e1, 0x4f01 }, + { 0x33e2, 0x026d }, + { 0x33e3, 0xf100 }, + { 0x33e4, 0x026d }, + { 0x33e5, 0x0001 }, + { 0x33e6, 0xc4f5 }, + { 0x33e7, 0x0361 }, + { 0x33e8, 0x0402 }, + { 0x33e9, 0x6601 }, + { 0x33ea, 0xc4f5 }, + { 0x33eb, 0x136f }, + { 0x33ec, 0xc4f5 }, + { 0x33ed, 0x134f }, + { 0x33ee, 0xc4f5 }, + { 0x33ef, 0x131f }, + { 0x33f0, 0x026d }, + { 0x33f1, 0x4e01 }, + { 0x33f2, 0x026d }, + { 0x33f3, 0xf000 }, + { 0x33f6, 0xc4f5 }, + { 0x33f7, 0x1361 }, + { 0x33f8, 0x0402 }, + { 0x33f9, 0x4600 }, + { 0x33fa, 0x026d }, + { 0x33fb, 0xfe00 }, }; static int wm5110_sysclk_ev(struct snd_soc_dapm_widget *w, -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ASoC: wm_adsp: Add info print to note that the DSP has shutdown
by Charles Keepax 21 Jan '14

21 Jan '14

It can be useful for debugging purposes to see at what point the DSP has powered down, so add a message to inform us of this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 4fbcab6..a15f75b 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1637,6 +1637,8 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, list_del(&alg_region->list); kfree(alg_region); } + + adsp_info(dsp, "Shutdown complete\n"); break; default: -- 1.7.2.5

2 1

[alsa-devel] [alsa-plugins] [PATCH 2/2] Allow to specify a client name as parameters
by Corfu, ValentinX 21 Jan '14

21 Jan '14

0001-Modify-jack-client-name.patch is spitted in: 0001-Allow-number-of-channels.patch and 0002-Allow-to-specify-a-client-name-as-parameters-in-the-.patch >From 8d6d397306bdd09c6c3fae83543e124c89443e94 Mon Sep 17 00:00:00 2001 From: Valentin Corfu <valentinx.corfu(a)intel.com> Date: Tue, 21 Jan 2014 11:32:51 -0500 Subject: [PATCH 2/2] Allow to specify a client name as parameters The current jack client name contains the process id of the application providing the audio samples. This leads to unpredictable jack client names which makes handling of the connections by a controlling application very hard. This modification now, allows to specify a client name as parameters in the configuration file. The implementation is backward compatible and simply adds a new configuration option to the plugin. Signed-off-by: Valentin Corfu <valentinx.corfu(a)intel.com> diff --git a/jack/pcm_jack.c b/jack/pcm_jack.c index 4306a13..61ac197 100644 --- a/jack/pcm_jack.c +++ b/jack/pcm_jack.c @@ -32,6 +32,11 @@ typedef enum _jack_format { SND_PCM_JACK_FORMAT_RAW } snd_pcm_jack_format_t; +typedef enum _naming_scheme { + SND_PCM_JACK_STD_NAME, + SND_PCM_JACK_USER_DEFINED_NAME +} snd_pcm_jack_naming_t; + typedef struct { snd_pcm_ioplug_t io; @@ -325,7 +330,8 @@ static int make_nonblock(int fd) static int snd_pcm_jack_open(snd_pcm_t **pcmp, const char *name, snd_config_t *playback_conf, snd_config_t *capture_conf, - snd_pcm_stream_t stream, int mode) + snd_pcm_stream_t stream, int mode, + snd_pcm_jack_naming_t naming) { snd_pcm_jack_t *jack; int err; @@ -356,11 +362,19 @@ static int snd_pcm_jack_open(snd_pcm_t **pcmp, const char *name, return -EINVAL; } - if (snprintf(jack_client_name, sizeof(jack_client_name), "alsa-jack.%s%s.%d.%d", name, - stream == SND_PCM_STREAM_PLAYBACK ? "P" : "C", getpid(), num++) - >= (int)sizeof(jack_client_name)) { - fprintf(stderr, "%s: WARNING: JACK client name '%s' truncated to %d characters, might not be unique\n", - __func__, jack_client_name, (int)strlen(jack_client_name)); + if (naming == SND_PCM_JACK_STD_NAME ) { + if (snprintf(jack_client_name, sizeof(jack_client_name), "alsa-jack.%s%s.%d.%d", name, + stream == SND_PCM_STREAM_PLAYBACK ? "P" : "C", getpid(), num++) + >= (int)sizeof(jack_client_name)) { + fprintf(stderr, "%s: WARNING: JACK client name '%s' truncated to %d characters, might not be unique\n", + __func__, jack_client_name, (int)strlen(jack_client_name)); + } + } else { + if (snprintf(jack_client_name, sizeof(jack_client_name), "%s", name) + >= (int)sizeof(jack_client_name)) { + fprintf(stderr, "%s: WARNING: JACK client name '%s' truncated to %d characters, might not be unique\n", + __func__, jack_client_name, (int)strlen(jack_client_name)); + } } jack->client = jack_client_new(jack_client_name); @@ -417,6 +431,8 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) int err; long num_channels = 0; long chan; + const char *local_name = name; + snd_pcm_jack_naming_t naming = SND_PCM_JACK_STD_NAME; snd_config_for_each(i, next, conf) { snd_config_t *n = snd_config_iterator_entry(i); @@ -425,6 +441,11 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) continue; if (strcmp(id, "comment") == 0 || strcmp(id, "type") == 0 || strcmp(id, "hint") == 0) continue; + if (strcmp(id, "name") == 0) { + snd_config_get_string(n, &local_name); + naming = SND_PCM_JACK_USER_DEFINED_NAME; + continue; + } if (strcmp(id, "nchan") == 0) { snd_config_get_integer(n, &num_channels); continue; @@ -449,7 +470,10 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) return -EINVAL; } - if ( (num_channels != 0) && (playback_conf == NULL) && (capture_conf == NULL) ) { + if ( (naming == SND_PCM_JACK_USER_DEFINED_NAME) + && (num_channels != 0) + && (playback_conf == NULL) + && (capture_conf == NULL) ) { err = snd_config_make_compound(&playback_conf, "playback_ports", 0); if (err) { SNDERR("Error during snd_config_make_compound(p): %s\n", strerror(err)); @@ -486,7 +510,7 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) return err; } } - err = snd_pcm_jack_open(pcmp, name, playback_conf, capture_conf, stream, mode); + err = snd_pcm_jack_open(pcmp, local_name, playback_conf, capture_conf, stream, mode, naming); if (err) { SNDERR("Error during snd_pcm_jack_open: %s\n", strerror(err)); return err; @@ -513,7 +537,7 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) return err; } } else { - err = snd_pcm_jack_open(pcmp, name, playback_conf, capture_conf, stream, mode); + err = snd_pcm_jack_open(pcmp, local_name, playback_conf, capture_conf, stream, mode, naming); } return err; -- 1.8.1.4 -------------------------------------------------------------- Intel Shannon Limited Registered in Ireland Registered Office: Collinstown Industrial Park, Leixlip, County Kildare Registered Number: 308263 Business address: Dromore House, East Park, Shannon, Co. Clare This e-mail and any attachments may contain confidential material for the sole use of the intended recipient(s). Any review or distribution by others is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies.

2 1

[alsa-devel] [alsa-plugins] [PATCH 1/2] Allow number of channels
by Corfu, ValentinX 21 Jan '14

21 Jan '14

0001-Modify-jack-client-name.patch is spitted in: 0001-Allow-number-of-channels.patch and 0002-Allow-to-specify-a-client-name-as-parameters-in-the-.patch >From d247e4d675dd68c6f683a8c09ff792172396e870 Mon Sep 17 00:00:00 2001 From: Valentin Corfu <valentinx.corfu(a)intel.com> Date: Tue, 21 Jan 2014 10:10:52 -0500 Subject: [PATCH 1/2] Allow number of channels The modification allows to specify the number of channels as parameters in the configuration file The given number of channels are then created but not connected. Signed-off-by: Valentin Corfu <valentinx.corfu(a)intel.com> diff --git a/jack/pcm_jack.c b/jack/pcm_jack.c index c51fe84..4306a13 100644 --- a/jack/pcm_jack.c +++ b/jack/pcm_jack.c @@ -415,6 +415,8 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) snd_config_t *playback_conf = NULL; snd_config_t *capture_conf = NULL; int err; + long num_channels = 0; + long chan; snd_config_for_each(i, next, conf) { snd_config_t *n = snd_config_iterator_entry(i); @@ -423,6 +425,10 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) continue; if (strcmp(id, "comment") == 0 || strcmp(id, "type") == 0 || strcmp(id, "hint") == 0) continue; + if (strcmp(id, "nchan") == 0) { + snd_config_get_integer(n, &num_channels); + continue; + } if (strcmp(id, "playback_ports") == 0) { if (snd_config_get_type(n) != SND_CONFIG_TYPE_COMPOUND) { SNDERR("Invalid type for %s", id); @@ -443,7 +449,72 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) return -EINVAL; } - err = snd_pcm_jack_open(pcmp, name, playback_conf, capture_conf, stream, mode); + if ( (num_channels != 0) && (playback_conf == NULL) && (capture_conf == NULL) ) { + err = snd_config_make_compound(&playback_conf, "playback_ports", 0); + if (err) { + SNDERR("Error during snd_config_make_compound(p): %s\n", strerror(err)); + return err; + } + err = snd_config_make_compound(&capture_conf, "capture_ports", 0); + if (err) { + SNDERR("Error during snd_config_make_compound(c): %s\n", strerror(err)); + return err; + } + char value[4]; + for (chan=0; chan<num_channels; ++chan) { + snprintf(value, sizeof(value), "%d", chan); + snd_config_t *newLeafPlayback; + snd_config_t *newLeafCapture; + err = snd_config_imake_integer(&newLeafPlayback, value, chan); + if (err) { + SNDERR("Error during snd_config_imake_string(p): %s\n", strerror(err)); + return err; + } + err = snd_config_add(playback_conf, newLeafPlayback); + if (err) { + SNDERR("Error during snd_config_add(p): %s\n", strerror(err)); + return err; + } + err = snd_config_imake_integer(&newLeafCapture, value, chan); + if (err) { + SNDERR("Error during snd_config_imake_string(c): %s\n", strerror(err)); + return err; + } + err = snd_config_add(capture_conf, newLeafCapture); + if (err) { + SNDERR("Error during snd_config_add(c): %s\n", strerror(err)); + return err; + } + } + err = snd_pcm_jack_open(pcmp, name, playback_conf, capture_conf, stream, mode); + if (err) { + SNDERR("Error during snd_pcm_jack_open: %s\n", strerror(err)); + return err; + } + + err = snd_config_delete_compound_members(playback_conf); + if (err) { + SNDERR("Error during snd_config_delete_compound_members(p): %s\n", strerror(err)); + return err; + } + err = snd_config_delete_compound_members(capture_conf); + if (err) { + SNDERR("Error during snd_config_delete_compound_members(c): %s\n", strerror(err)); + return err; + } + err = snd_config_delete(playback_conf); + if (err) { + SNDERR("Error during snd_config_delete(p): %s\n", strerror(err)); + return err; + } + err = snd_config_delete(capture_conf); + if (err) { + SNDERR("Error during snd_config_delete(c): %s\n", strerror(err)); + return err; + } + } else { + err = snd_pcm_jack_open(pcmp, name, playback_conf, capture_conf, stream, mode); + } return err; } -- 1.8.1.4 -------------------------------------------------------------- Intel Shannon Limited Registered in Ireland Registered Office: Collinstown Industrial Park, Leixlip, County Kildare Registered Number: 308263 Business address: Dromore House, East Park, Shannon, Co. Clare This e-mail and any attachments may contain confidential material for the sole use of the intended recipient(s). Any review or distribution by others is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies.

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: Resume mixer values properly
by Takashi Iwai 21 Jan '14

21 Jan '14

Implement reset_resume callback so that the mixer values are properly restored. Still no boot quirks are called, so it might not work well on some devices. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/card.c | 18 ++++++++-- sound/usb/mixer.c | 99 ++++++++++++++++++++++++++++++++++++++++++++----------- sound/usb/mixer.h | 7 ++-- 3 files changed, 99 insertions(+), 25 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index d979050e6a6a..025224136129 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -691,12 +691,12 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) } list_for_each_entry(mixer, &chip->mixer_list, list) - snd_usb_mixer_inactivate(mixer); + snd_usb_mixer_suspend(mixer); return 0; } -static int usb_audio_resume(struct usb_interface *intf) +static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) { struct snd_usb_audio *chip = usb_get_intfdata(intf); struct usb_mixer_interface *mixer; @@ -711,7 +711,7 @@ static int usb_audio_resume(struct usb_interface *intf) * we just notify and restart the mixers */ list_for_each_entry(mixer, &chip->mixer_list, list) { - err = snd_usb_mixer_activate(mixer); + err = snd_usb_mixer_resume(mixer, reset_resume); if (err < 0) goto err_out; } @@ -723,9 +723,20 @@ static int usb_audio_resume(struct usb_interface *intf) err_out: return err; } + +static int usb_audio_resume(struct usb_interface *intf) +{ + return __usb_audio_resume(intf, false); +} + +static int usb_audio_reset_resume(struct usb_interface *intf) +{ + return __usb_audio_resume(intf, true); +} #else #define usb_audio_suspend NULL #define usb_audio_resume NULL +#define usb_audio_reset_resume NULL #endif /* CONFIG_PM */ static struct usb_device_id usb_audio_ids [] = { @@ -747,6 +758,7 @@ static struct usb_driver usb_audio_driver = { .disconnect = usb_audio_disconnect, .suspend = usb_audio_suspend, .resume = usb_audio_resume, + .reset_resume = usb_audio_reset_resume, .id_table = usb_audio_ids, .supports_autosuspend = 1, }; diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 44b0ba4feab3..aa9bc19aae68 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2299,26 +2299,6 @@ requeue: } } -/* stop any bus activity of a mixer */ -void snd_usb_mixer_inactivate(struct usb_mixer_interface *mixer) -{ - usb_kill_urb(mixer->urb); - usb_kill_urb(mixer->rc_urb); -} - -int snd_usb_mixer_activate(struct usb_mixer_interface *mixer) -{ - int err; - - if (mixer->urb) { - err = usb_submit_urb(mixer->urb, GFP_NOIO); - if (err < 0) - return err; - } - - return 0; -} - /* create the handler for the optional status interrupt endpoint */ static int snd_usb_mixer_status_create(struct usb_mixer_interface *mixer) { @@ -2417,3 +2397,82 @@ void snd_usb_mixer_disconnect(struct list_head *p) usb_kill_urb(mixer->urb); usb_kill_urb(mixer->rc_urb); } + +#ifdef CONFIG_PM +/* stop any bus activity of a mixer */ +static void snd_usb_mixer_inactivate(struct usb_mixer_interface *mixer) +{ + usb_kill_urb(mixer->urb); + usb_kill_urb(mixer->rc_urb); +} + +static int snd_usb_mixer_activate(struct usb_mixer_interface *mixer) +{ + int err; + + if (mixer->urb) { + err = usb_submit_urb(mixer->urb, GFP_NOIO); + if (err < 0) + return err; + } + + return 0; +} + +int snd_usb_mixer_suspend(struct usb_mixer_interface *mixer) +{ + snd_usb_mixer_inactivate(mixer); + return 0; +} + +static int restore_mixer_value(struct usb_mixer_elem_info *cval) +{ + int c, err, idx; + + if (cval->cmask) { + idx = 0; + for (c = 0; c < MAX_CHANNELS; c++) { + if (!(cval->cmask & (1 << c))) + continue; + if (cval->cached & (1 << c)) { + err = set_cur_mix_value(cval, c + 1, idx, + cval->cache_val[idx]); + if (err < 0) + return err; + } + idx++; + } + } else { + /* master */ + if (cval->cached) { + err = set_cur_mix_value(cval, 0, 0, *cval->cache_val); + if (err < 0) + return err; + } + } + + return 0; +} + +int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume) +{ + struct usb_mixer_elem_info *cval; + int id, err; + + /* FIXME: any mixer quirks? */ + + if (reset_resume) { + /* restore cached mixer values */ + for (id = 0; id < MAX_ID_ELEMS; id++) { + for (cval = mixer->id_elems[id]; cval; + cval = cval->next_id_elem) { + err = restore_mixer_value(cval); + if (err < 0) + return err; + } + } + } + + return snd_usb_mixer_activate(mixer); +} +#endif diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index aab80df201bd..73b1f649447b 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -63,8 +63,6 @@ void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid); int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, int request, int validx, int value_set); -void snd_usb_mixer_inactivate(struct usb_mixer_interface *mixer); -int snd_usb_mixer_activate(struct usb_mixer_interface *mixer); int snd_usb_mixer_add_control(struct usb_mixer_interface *mixer, struct snd_kcontrol *kctl); @@ -72,4 +70,9 @@ int snd_usb_mixer_add_control(struct usb_mixer_interface *mixer, int snd_usb_mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag, unsigned int size, unsigned int __user *_tlv); +#ifdef CONFIG_PM +int snd_usb_mixer_suspend(struct usb_mixer_interface *mixer); +int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume); +#endif + #endif /* __USBMIXER_H */ -- 1.8.5.2

1 0

Re: [alsa-devel] [PATCH] ASoC: da9055: Partially fix device
by Opensource [Adam Thomson] 21 Jan '14

21 Jan '14

On Wed, 8 Jan 2014 20:38:55 +0000, Mark Brown wrote: > It has never been possible to even load the da9055 driver since both the > MFD and CODEC drivers register the I2C device "da9055". Fix this by using > the usual pattern for MFDs with multiple I2C addresses and having the CODEC > driver reference an I2C dummy registered by the MFD. > Since I don't know which I2C address to use for the CODEC a FIXME has been > left in the MFD, this doesn't make anything any worse since the device has > never been able to load in the first place. What you're saying here is incorrect, as previously discussed at length in this thread: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-September/066027.… The two drivers did work together fine until someone in 3.12 changed the PMIC I2C Id back to just 'da9055' and broke it again. Also, I wanted to add a simple patch to tidy up the I2C Id names and avoid this but you would not agree. The codec part can be standalone and by doing what you've done here means that customers would need to now add additional, unnecessary code to be able to use just the codec driver. This is not right to me.

2 11

[alsa-devel] [PATCH] ASoC: omap: Make RX51 depend on GPIOLIB due to jack usage
by Mark Brown 21 Jan '14

21 Jan '14

From: Mark Brown <broonie(a)linaro.org> Since the GPIO jacks are only supported if gpiolib is built and fail to compile otherwise add a build depedency. This is unlikely to have any practical impact outside of coverage testing. Reported-by: Russell King <linux(a)arm.linux.org.uk> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/omap/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 4a07f7179690..22ad9c5654b5 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -30,6 +30,7 @@ config SND_OMAP_SOC_RX51 select SND_OMAP_SOC_MCBSP select SND_SOC_TLV320AIC3X select SND_SOC_TPA6130A2 + depends on GPIOLIB help Say Y if you want to add support for SoC audio on Nokia RX-51 hardware. This is also known as Nokia N900 product. -- 1.8.5.2

2 1

[alsa-devel] ALSA: hda - Fix possible races in HDMI driver - lockup on shutdown when radeon.audio=1 after using audacity
by Arthur Marsh 21 Jan '14

21 Jan '14

I have had reproducible lock-ups on shut-down (at the shutting down ALSA stage) of my AMD64 machine (Asus M3A78Pro motherboard, BIOS 1701 01/27/2011, CPU AMD Athlon(tm) II X4 640 Processor) running the 64 bit Linux kernel more recent than 3.12 when *both* radeon.audio=1 was set and I had been running audacity 2.0.5. (iommu=noaperture is also set). The problem was reproducible with the stock Debian kernel linux-image-3.13-rc6-amd64 version 3.13~rc6-1~exp1. The machine is using an ATI/AMD 3850HD video card with a DVI cable to a DVI input on my monitor, and the default audio device is the motherboard's on-board audio device: 00:14.2 Audio device: Advanced Micro Devices, Inc. [AMD/ATI] SBx00 Azalia (Intel HDA) 01:00.0 VGA compatible controller: Advanced Micro Devices, Inc. [AMD/ATI] RV670 [Radeon HD 3690/3850] 01:00.1 Audio device: Advanced Micro Devices, Inc. [AMD/ATI] RV670/680 HDMI Audio [Radeon HD 3690/3800 Series] $ git bisect bad cbbaa603a03cc46681e24d6b2804b62fde95a2af is the first bad commit commit cbbaa603a03cc46681e24d6b2804b62fde95a2af Author: Takashi Iwai <tiwai(a)suse.de> Date: Thu Oct 17 18:03:24 2013 +0200 ALSA: hda - Fix possible races in HDMI driver Some per_pin fields and ELD contents might be changed dynamically in multiple ways where the concurrent accesses are still opened in the current code. This patch fixes such possible races by using eld->lock in appropriate places. Reported-by: Anssi Hannula <anssi.hannula(a)iki.fi> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> :040000 040000 0c29281f82a3ebd9a704d481114f9cfcefea07c8 d71fd101125cd29a628cb5e66c7ee4f56e28329b M sound When running audacity from the command line there was the following output: ALSA lib pcm_dsnoop.c:618:(snd_pcm_dsnoop_open) unable to open slave ALSA lib pcm_dmix.c:1022:(snd_pcm_dmix_open) unable to open slave ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.rear ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.center_lfe ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.side ALSA lib pcm_dmix.c:1022:(snd_pcm_dmix_open) unable to open slave Expression 'stream->playback.pcm' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 4611 Expression 'stream->playback.pcm' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 4611 Expression 'stream->playback.pcm' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 4611 I am happy to supply further information or run further tests to help in isolating the problem and verifying a solution. Arthur.

2 5

[alsa-devel] [PATCH] ASoC: wm8804: Allow control of master clock divider in PLLgeneration
by Daniel Matuschek 20 Jan '14

20 Jan '14

WM8804 can run with PLL frequencies of 256xfs and 128xfs for most sample rates. At 192kHz only 128xfs is supported. The existing driver selects 128xfs automatically for some lower samples rates. By using an additional mclk_div divider, it is now possible to control the behaviour. This allows using 256xfs PLL frequency on all sample rates up to 96kHz. It should allow lower jitter and better signal quality. The behavior has to be controlled by the sound card driver, because some sample frequency share the same setting. e.g. 192kHz and 96kHz use 24.576MHz master clock. The only difference is the MCLK divider. Signed-off-by: Daniel Matuschek <daniel(a)matuschek.net> Acked-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Tested-by: Florian Meier <florian.meier(a)koalo.de> --- sound/soc/codecs/wm8804.c | 17 ++++++++++++++--- sound/soc/codecs/wm8804.h | 4 ++++ 2 files changed, 18 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 9bc8206..4163786 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -63,6 +63,7 @@ struct wm8804_priv { struct regmap *regmap; struct regulator_bulk_data supplies[WM8804_NUM_SUPPLIES]; struct notifier_block disable_nb[WM8804_NUM_SUPPLIES]; + int mclk_div; }; static int txsrc_get(struct snd_kcontrol *kcontrol, @@ -318,7 +319,7 @@ static struct { #define FIXED_PLL_SIZE ((1ULL << 22) * 10) static int pll_factors(struct pll_div *pll_div, unsigned int target, - unsigned int source) + unsigned int source, unsigned int mclk_div) { u64 Kpart; unsigned long int K, Ndiv, Nmod, tmp; @@ -330,7 +331,8 @@ static int pll_factors(struct pll_div *pll_div, unsigned int target, */ for (i = 0; i < ARRAY_SIZE(post_table); i++) { tmp = target * post_table[i].div; - if (tmp >= 90000000 && tmp <= 100000000) { + if ((tmp >= 90000000 && tmp <= 100000000) && + (mclk_div == post_table[i].mclkdiv)) { pll_div->freqmode = post_table[i].freqmode; pll_div->mclkdiv = post_table[i].mclkdiv; target *= post_table[i].div; @@ -387,8 +389,12 @@ static int wm8804_set_pll(struct snd_soc_dai *dai, int pll_id, } else { int ret; struct pll_div pll_div; + struct wm8804_priv *wm8804; - ret = pll_factors(&pll_div, freq_out, freq_in); + wm8804 = snd_soc_codec_get_drvdata(codec); + + ret = pll_factors(&pll_div, freq_out, freq_in, + wm8804->mclk_div); if (ret) return ret; @@ -452,6 +458,7 @@ static int wm8804_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) { struct snd_soc_codec *codec; + struct wm8804_priv *wm8804; codec = dai->codec; switch (div_id) { @@ -459,6 +466,10 @@ static int wm8804_set_clkdiv(struct snd_soc_dai *dai, snd_soc_update_bits(codec, WM8804_PLL5, 0x30, (div & 0x3) << 4); break; + case WM8804_MCLK_DIV: + wm8804 = snd_soc_codec_get_drvdata(codec); + wm8804->mclk_div = div; + break; default: dev_err(dai->dev, "Unknown clock divider: %d\n", div_id); return -EINVAL; diff --git a/sound/soc/codecs/wm8804.h b/sound/soc/codecs/wm8804.h index 8ec14f5..e72d4f4 100644 --- a/sound/soc/codecs/wm8804.h +++ b/sound/soc/codecs/wm8804.h @@ -57,5 +57,9 @@ #define WM8804_CLKOUT_SRC_OSCCLK 4 #define WM8804_CLKOUT_DIV 1 +#define WM8804_MCLK_DIV 2 + +#define WM8804_MCLKDIV_256FS 0 +#define WM8804_MCLKDIV_128FS 1 #endif /* _WM8804_H */ -- 1.8.3.2

1 0

[alsa-devel] ASoC updates for v3.14
by Mark Brown 20 Jan '14

20 Jan '14

The following changes since commit 258753361534a40ad7180c742da813fc659e427b: Merge branch 'for-next' into for-linus (2014-01-20 10:20:14 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.14-3 for you to fetch changes up to ac6d7c48e3a5e82e876114b9ca0b489e96997743: Merge remote-tracking branch 'asoc/topic/compress' into asoc-next (2014-01-20 11:50:41 +0000) ---------------------------------------------------------------- ASoC: Last updates for the merge window A couple more fixes plus some extensions to DPCM for use with compressed audio from Liam which arrived just after my previous pull request. ---------------------------------------------------------------- Arun Shamanna Lakshmi (1): ASoC: dapm: Fix double prefix addition Liam Girdwood (2): ASoC: DPCM: make some DPCM API calls non static for compressed usage ASoC: compress: Add suport for DPCM into compressed audio Mark Brown (3): Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/dma' into asoc-next Merge remote-tracking branch 'asoc/topic/compress' into asoc-next Xiubo Li (1): ASoC: core: Fix possible NULL pointer dereference of pcm->config include/sound/soc-dpcm.h | 22 +++ include/sound/soc.h | 1 + sound/soc/soc-compress.c | 301 +++++++++++++++++++++++++++++++++- sound/soc/soc-dapm.c | 11 +- sound/soc/soc-generic-dmaengine-pcm.c | 9 +- sound/soc/soc-pcm.c | 29 ++-- 6 files changed, 347 insertions(+), 26 deletions(-)

2 1

[alsa-devel] [PATCH v3 01/16] ALSA: Oxygen: Add the separate SPI waiting function
by Roman Volkov 20 Jan '14

20 Jan '14

This function performs waiting when the SPI bus completes a transaction. Timeout error checking introduced and the timeout increased to 400 from 40. Signed-off-by: Roman Volkov <v1ron(a)mail.ru> --- sound/pci/oxygen/oxygen_io.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/pci/oxygen/oxygen_io.c b/sound/pci/oxygen/oxygen_io.c index 521eae4..a917f37 100644 --- a/sound/pci/oxygen/oxygen_io.c +++ b/sound/pci/oxygen/oxygen_io.c @@ -194,6 +194,21 @@ void oxygen_write_ac97_masked(struct oxygen *chip, unsigned int codec, } EXPORT_SYMBOL(oxygen_write_ac97_masked); +static int oxygen_wait_spi(struct oxygen *chip) +{ + /* Higher timeout to be sure: 400 microseconds */ + unsigned int count; + for (count = 100; count > 0; count--) { + if ((oxygen_read8(chip, OXYGEN_SPI_CONTROL) & + OXYGEN_SPI_BUSY) == 0) + return 0; + udelay(4); + --count; + } + snd_printk(KERN_ERR "Oxygen: SPI wait timeout\n"); + return -EIO; +} + void oxygen_write_spi(struct oxygen *chip, u8 control, unsigned int data) { unsigned int count; -- 1.7.10.4

2 23

[alsa-devel] Thinkpad T520 (Low Volume)
by Raymond Yau 20 Jan '14

20 Jan '14

> 2014/1/16 Axel Braun <axel.braun(a)gmx.de> > same problem on a ThinkPad T520, sound volume is way too low - watching movie via loudspeakers is fairly impossible. > Any hint you could give me? > > post the output of alsa-info.sh , there is a bug in alsa-driver which treat > dock headphone and headphone to support surround 4.0 you get low volume because Amp Out vals are 0x3d instead of 0x4a but speaker volumes are at 100% (0dB) Simple mixer control 'Speaker',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 74 Mono: Front Left: Playback 74 [100%] [0.00dB] [on] Front Right: Playback 74 [100%] [0.00dB] [on] Node 0x10 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Control: name="Speaker Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Speaker Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="CX20590 Analog", type="Audio", device=0 Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-Out vals: [0x3d 0x3d] https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/?id=ca2… thinkpad t520 was supported with CXT5066_AUTO which use cxt5066_update_speaker which check port a and port d Do this mean thinkpad don't support surround 4.0 by dock headphone and headphone ? https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… ALSA: hda/conexant - Set up the missing docking-station pins [ 9.956636] ALSA hda_auto_parser.c:393 autoconfig: line_outs=1 (0x1f/0x0/0x0/0x0/0x0) type:speaker [ 9.956639] ALSA hda_auto_parser.c:397 speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 9.956641] ALSA hda_auto_parser.c:401 hp_outs=2 (0x1c/0x19/0x0/0x0/0x0) [ 9.956643] ALSA hda_auto_parser.c:402 mono: mono_out=0x0 [ 9.956644] ALSA hda_auto_parser.c:406 inputs: [ 9.956646] ALSA hda_auto_parser.c:410 Mic=0x1b [ 9.956647] ALSA hda_auto_parser.c:410 Internal Mic=0x23 [ 9.956649] ALSA hda_auto_parser.c:410 Dock Mic=0x1a the driver select this as best config ==> Best config: lo_type=1, wired=1, mio=1 multi_outs = 1f/0/0/0 : 10/0/0/0 (type SP) out path: depth=2 :10:1f hp_outs = 19/1c/0/0 : 11/11/0/0 hp path: depth=2 :11:19 hp path: depth=2 :11:1c https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… ALSA: hda - Fix multi-io channel mode management this patch change the the channel_count from 2 to 4 since there are two headphone + if (cfg->line_out_type != AUTO_PIN_HP_OUT) + spec->const_channel_count = max(spec->const_channel_count, + cfg->hp_outs * 2); if you don't have docking station , the workaround is to remove those pin fix of node 0x1a and 0x1c

3 5

[alsa-devel] valgrind error in snd_tlv_get_dB_range
by Ivan Sorokin 20 Jan '14

20 Jan '14

One program (qasmixer -- alsa mixer) on my system prints lots of valgrind errors. The first one is this: ==25716== Conditional jump or move depends on uninitialised value(s) ==25716== at 0x4E72307: snd_tlv_get_dB_range (tlv.c:170) ==25716== by 0x4E8A746: get_dB_range (simple_none.c:1162) ==25716== by 0x4E8A7E7: get_dB_range_ops (simple_none.c:1176) ==25716== by 0x4E85590: snd_mixer_selem_get_playback_dB_range (simple.c:298) ==25716== by 0x472B2B: QSnd::Mixer_Simple_Elem::set_snd_mixer_selem_id(_snd_mixer_selem_id*) (mixer_simple_elem.cpp:161) ==25716== by 0x4724B8: QSnd::Mixer_Simple_Elem::Mixer_Simple_Elem(QObject*, _snd_mixer*, _snd_mixer_selem_id*) (mixer_simple_elem.cpp:31) ==25716== by 0x479F17: QSnd::Mixer_Simple::create_mixer_elem(_snd_mixer_elem*) (mixer_simple.cpp:318) ==25716== by 0x4797B3: QSnd::Mixer_Simple::create_mixer_elems() (mixer_simple.cpp:217) ==25716== by 0x4795DF: QSnd::Mixer_Simple::open(QString const&) (mixer_simple.cpp:178) ==25716== by 0x4D4754: Tray_Mixer::load_mixer() (tray_mixer.cpp:222) ==25716== by 0x4D4505: Tray_Mixer::set_mdev_setup(Tray_Mixer_MDev_Setup*) (tray_mixer.cpp:137) ==25716== by 0x4DB863: Desktop_Items::tray_mixer_create() (desktop_items.cpp:504) ==25716== Uninitialised value was created by a heap allocation ==25716== at 0x4C2CD7B: malloc (in /usr/lib/valgrind/vgpreload_memcheck-amd64-linux.so) ==25716== by 0x4E8A572: init_db_range (simple_none.c:1114) ==25716== by 0x4E8A70D: get_dB_range (simple_none.c:1159) ==25716== by 0x4E8A7E7: get_dB_range_ops (simple_none.c:1176) ==25716== by 0x4E85590: snd_mixer_selem_get_playback_dB_range (simple.c:298) ==25716== by 0x472B2B: QSnd::Mixer_Simple_Elem::set_snd_mixer_selem_id(_snd_mixer_selem_id*) (mixer_simple_elem.cpp:161) ==25716== by 0x4724B8: QSnd::Mixer_Simple_Elem::Mixer_Simple_Elem(QObject*, _snd_mixer*, _snd_mixer_selem_id*) (mixer_simple_elem.cpp:31) ==25716== by 0x479F17: QSnd::Mixer_Simple::create_mixer_elem(_snd_mixer_elem*) (mixer_simple.cpp:318) ==25716== by 0x4797B3: QSnd::Mixer_Simple::create_mixer_elems() (mixer_simple.cpp:217) ==25716== by 0x4795DF: QSnd::Mixer_Simple::open(QString const&) (mixer_simple.cpp:178) ==25716== by 0x4D4754: Tray_Mixer::load_mixer() (tray_mixer.cpp:222) ==25716== by 0x4D4505: Tray_Mixer::set_mdev_setup(Tray_Mixer_MDev_Setup*) (tray_mixer.cpp:137) I compiled libasound2 from the latest git from git://gitorious.org/alsa/alsa-lib.git with -g -O0, so you can see line numbers. As I understand the problem is that init_db_range initialize invalid rec->db_info, because snd_hctl_elem_tlv_read failed to fully initialize tlv: > tlv = malloc(tlv_size); // line 1114 > if (! tlv) > return -ENOMEM; > if (snd_hctl_elem_tlv_read(ctl, tlv, tlv_size) < 0) > goto error; > db_size = snd_tlv_parse_dB_info(tlv, tlv_size, &dbrec); > if (db_size < 0) > goto error; > rec->db_info = malloc(db_size); > if (!rec->db_info) > goto error; > memcpy(rec->db_info, dbrec, db_size); > free(tlv); I tried to trace problem in snd_hctl_elem_tlv_read, but I failed to do so due to lack of experience with alsa-lib. Could someone more experienced with alsa-lib look at the problem?

3 3

[alsa-devel] Realtec ALC898 supported?
by Haug Bürger 20 Jan '14

20 Jan '14

Hi, I would like to know if the new Realtec 89x chips are supported. I found drivers on the Realtec site but the seem to be older than the alsa-project.org drivers and I'm not sure if they support the 89x chips. thanks Haug

2 1

[alsa-devel] [PATCH] ASoC: core: Use devm_kzalloc() instead kzalloc()
by Xiubo Li 20 Jan '14

20 Jan '14

Makes the code slightly shorter Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/soc-generic-dmaengine-pcm.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 5bace12..bfb012f 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -388,7 +388,7 @@ int snd_dmaengine_pcm_register(struct device *dev, struct dmaengine_pcm *pcm; int ret; - pcm = kzalloc(sizeof(*pcm), GFP_KERNEL); + pcm = devm_kzalloc(dev, sizeof(*pcm), GFP_KERNEL); if (!pcm) return -ENOMEM; @@ -408,7 +408,6 @@ int snd_dmaengine_pcm_register(struct device *dev, err_free_dma: dmaengine_pcm_release_chan(pcm); - kfree(pcm); return ret; } EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_register); @@ -433,7 +432,6 @@ void snd_dmaengine_pcm_unregister(struct device *dev) snd_soc_remove_platform(platform); dmaengine_pcm_release_chan(pcm); - kfree(pcm); } EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_unregister); -- 1.8.4

4 3

[alsa-devel] [PATCH] ASoC: dapm: Fix double prefix addition
by Arun Shamanna Lakshmi 17 Jan '14

17 Jan '14

The prefix for the codec driver can be used during dual identical codec usecases. However, dapm adds prefix twice for codec DAI widget in snd_soc_dapm_add_route API. This change is to avoid double prefix addition for codec DAI widget and is needed while using identical dual codecs. Signed-off-by: Songhee Baek <sbaek(a)nvidia.com> Signed-off-by: Arun Shamanna Lakshmi <aruns(a)nvidia.com> --- sound/soc/soc-dapm.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 2a44fe9..dc8ff13 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2476,7 +2476,8 @@ err: } static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, - const struct snd_soc_dapm_route *route) + const struct snd_soc_dapm_route *route, + unsigned int is_prefixed) { struct snd_soc_dapm_widget *wsource = NULL, *wsink = NULL, *w; struct snd_soc_dapm_widget *wtsource = NULL, *wtsink = NULL; @@ -2486,7 +2487,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, char prefixed_source[80]; int ret; - if (dapm->codec && dapm->codec->name_prefix) { + if (dapm->codec && dapm->codec->name_prefix && !is_prefixed) { snprintf(prefixed_sink, sizeof(prefixed_sink), "%s %s", dapm->codec->name_prefix, route->sink); sink = prefixed_sink; @@ -2614,7 +2615,7 @@ int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm, mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); for (i = 0; i < num; i++) { - r = snd_soc_dapm_add_route(dapm, route); + r = snd_soc_dapm_add_route(dapm, route, false); if (r < 0) { dev_err(dapm->dev, "ASoC: Failed to add route %s -> %s -> %s\n", route->source, @@ -3670,7 +3671,7 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card) cpu_dai->codec->name, r.source, codec_dai->platform->name, r.sink); - snd_soc_dapm_add_route(&card->dapm, &r); + snd_soc_dapm_add_route(&card->dapm, &r, true); } /* connect BE DAI capture if widgets are valid */ @@ -3681,7 +3682,7 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card) codec_dai->codec->name, r.source, cpu_dai->platform->name, r.sink); - snd_soc_dapm_add_route(&card->dapm, &r); + snd_soc_dapm_add_route(&card->dapm, &r, true); } } -- 1.7.9.5

2 1

Re: [alsa-devel] [PATCH] ASoC: wm8804: Allow control of master clock divider in PLL generation
by Florian Meier 17 Jan '14

17 Jan '14

On 01/17/2014 07:33 PM, Mark Brown wrote: > On Fri, Jan 17, 2014 at 07:06:24PM +0100, Florian Meier wrote: >> > Intentionally off-list? Oh no - I am sorry! >> If I remember correctly the error was >> "codec can not start from non-off bias with idle_bias_off==true" > >> I think the solution is just to set idle_bias_off = false and everything >> seems to be working with that. I just don't know if there might be any >> side effects. > > Setting it to false increases power consumption since the device is > kept more powered on when idle but reduces startup time from idle. For > digital only devices like the wm8804 there shouldn't be any reason to > keep it powered up when not in use, the startup time is generally > negligable anyway. > So a better solution would be to set SND_SOC_BIAS_OFF instead of SND_SOC_BIAS_STANDBY at the end of probe?

2 1

[alsa-devel] [PATCH 0/4] ASoC: tlv320aic32x4: DT support
by Markus Pargmann 17 Jan '14

17 Jan '14

Hi, Here are some more updates for the tlv320aic32x4 codec. This series fixes the mixer controls to operate on the full range instead of only the positive range. This depends on my other series: "ASoC: core: volume control using signed register values" Patch 3 adds DT support. Patch 4 adds support to control the master clock connected to the codec. Regards, Markus Markus Pargmann (4): ASoC: tlv320aic32x4: Use signed int mixer controls ASoC: tlv320aic32x4: Use gpio_is_valid ASoC: tlv320aic32x4: DT support ASoC: tlv320aic32x4: Support for master clock .../devicetree/bindings/sound/tlv320aic32x4.txt | 25 +++++++ sound/soc/codecs/tlv320aic32x4.c | 83 ++++++++++++++++++---- 2 files changed, 96 insertions(+), 12 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic32x4.txt -- 1.8.5.2

2 6

[alsa-devel] [PATCH v2 0/2] Dynamic routing for compressed streams.
by Liam Girdwood 17 Jan '14

17 Jan '14

Changes for V2 * We no longer copy the compressed params into the BE hw_params. This was bogus, but didn't show up on Haswell testing as the BE params were never used. The DSP ran the BE is a fixed configuration so all BE params were ignored. * The BE hw_params are now zeroed to force the machine driver to fixup the BE configuration. i.e. machine driver can query the DSP encodec/decoder for correct hw_params given the use case. * No longer need to allocate memory for hw_params. Regards Liam

2 1

[alsa-devel] [PATCH v2 1/2] ASoC: DPCM: make some DPCM API calls non static for compressed usage
by Liam Girdwood 17 Jan '14

17 Jan '14

The ASoC compressed code needs to call the internal DPCM APIs in order to dynamically route compressed data to different DAIs. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- include/sound/soc-dpcm.h | 22 ++++++++++++++++++++++ sound/soc/soc-pcm.c | 29 ++++++++++++----------------- 2 files changed, 34 insertions(+), 17 deletions(-) diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index 047d657..2883a7a 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -11,6 +11,7 @@ #ifndef __LINUX_SND_SOC_DPCM_H #define __LINUX_SND_SOC_DPCM_H +#include <linux/slab.h> #include <linux/list.h> #include <sound/pcm.h> @@ -135,4 +136,25 @@ int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute); int soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd); int soc_dpcm_runtime_update(struct snd_soc_card *); +int dpcm_path_get(struct snd_soc_pcm_runtime *fe, + int stream, struct snd_soc_dapm_widget_list **list_); +int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, + int stream, struct snd_soc_dapm_widget_list **list, int new); +int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream); +int dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream); +void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream); +void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream); +int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream); +int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int tream); +int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd); +int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream); +int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, + int event); + +static inline void dpcm_path_put(struct snd_soc_dapm_widget_list **list) +{ + kfree(*list); +} + + #endif diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 1534115..7023f7e 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -58,7 +58,7 @@ int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream, EXPORT_SYMBOL_GPL(snd_soc_set_runtime_hwparams); /* DPCM stream event, send event to FE and all active BEs. */ -static int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, +int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, int event) { struct snd_soc_dpcm *dpcm; @@ -877,7 +877,7 @@ static void dpcm_be_reparent(struct snd_soc_pcm_runtime *fe, } /* disconnect a BE and FE */ -static void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) +void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm, *d; @@ -973,7 +973,7 @@ static int widget_in_list(struct snd_soc_dapm_widget_list *list, return 0; } -static int dpcm_path_get(struct snd_soc_pcm_runtime *fe, +int dpcm_path_get(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_dapm_widget_list **list_) { struct snd_soc_dai *cpu_dai = fe->cpu_dai; @@ -995,11 +995,6 @@ static int dpcm_path_get(struct snd_soc_pcm_runtime *fe, return paths; } -static inline void dpcm_path_put(struct snd_soc_dapm_widget_list **list) -{ - kfree(*list); -} - static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_dapm_widget_list **list_) { @@ -1069,7 +1064,7 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, continue; /* don't connect if FE is not running */ - if (!fe->dpcm[stream].runtime) + if (!fe->dpcm[stream].runtime && !fe->fe_compr) continue; /* newly connected FE and BE */ @@ -1094,7 +1089,7 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, * Find the corresponding BE DAIs that source or sink audio to this * FE substream. */ -static int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, +int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_dapm_widget_list **list, int new) { if (new) @@ -1103,7 +1098,7 @@ static int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, return dpcm_prune_paths(fe, stream, list); } -static void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream) +void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; @@ -1141,7 +1136,7 @@ static void dpcm_be_dai_startup_unwind(struct snd_soc_pcm_runtime *fe, } } -static int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; int err, count = 0; @@ -1293,7 +1288,7 @@ be_err: return ret; } -static int dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; @@ -1354,7 +1349,7 @@ static int dpcm_fe_dai_shutdown(struct snd_pcm_substream *substream) return 0; } -static int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; @@ -1419,7 +1414,7 @@ static int dpcm_fe_dai_hw_free(struct snd_pcm_substream *substream) return 0; } -static int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; int ret; @@ -1549,7 +1544,7 @@ static int dpcm_do_trigger(struct snd_soc_dpcm *dpcm, return ret; } -static int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, +int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd) { struct snd_soc_dpcm *dpcm; @@ -1717,7 +1712,7 @@ out: return ret; } -static int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; int ret = 0; -- 1.8.3.2

1 1

[alsa-devel] ProVideo PV 918A - audi o device created but cannot capture sound
by siddharth 17 Jan '14

17 Jan '14

Hi, I have installed a ProVideo PV 981. I was able to get the sound device detected by putting modprobe snd-bt87x load_all=1 digital_rate=48000 enable=1 but am unable to capture sound. I am using Fedora 19 on an intel i5 machine. In dmesg i get the error: [ 235.145932] ALSA sound/pci/bt87x.c:860 unknown card 0x878-0xaa00:0x1460 [ 235.145935] ALSA sound/pci/bt87x.c:863 please mail id, board name, and, if it works, the correct digital_rate option to <alsa-devel(a)alsa-project.org> [ 235.146061] ALSA sound/pci/bt87x.c:931 bt87x0: Using board 0, analog, digital (rate 48000 Hz) Any help will be appreciated. regards Siddharth

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Re: [alsa-devel] [pulseaudio-discuss] Need help with audio gain on an Intel HDA equipt ASUS M2N-SLI Deluxe
by Raymond Yau 17 Jan '14

17 Jan '14

2014/1/16 Axel Braun <axel.braun(a)gmx.de> > Am Donnerstag, 16. Januar 2014, 16:23:19 schrieb Raymond Yau: > > Any hint you could give me? > > > > post the output of alsa-info.sh , there is a bug in alsa-driver which > treat > > dock headphone and headphone to support surround 4.0 > > /sys/class/sound/hwC0D0/init_pin_configs: 0x19 0x04211040 0x1a 0x61a19050 0x1b 0x04a11060 0x1c 0x6121401f 0x1d 0x40f001f0 0x1e 0x40f001f0 0x1f 0x90170110 0x20 0x40f001f0 0x22 0x40f001f0 0x23 0x90a60170 /sys/class/sound/hwC0D0/driver_pin_configs: 0x19 0x042110ff 0x1a 0x21a190f0 0x1c 0x212140ff https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… it was using auto but now the driver modify the dock headphone , dock mic and headphone pin default. there are also changes in the thinkpad_acpi module too. https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… do this thinkpad support headset or headphone ? it is strange that speaker volume is at max but amp out vals at node 0x11 is 0x3d instead of 0x4a (0dB) Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 74 Mono: Playback 61 [82%] [-13.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 74 Mono: Front Left: Playback 0 [0%] [-74.00dB] [off] Front Right: Playback 0 [0%] [-74.00dB] [off] Simple mixer control 'Speaker',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 74 Mono: Front Left: Playback 74 [100%] [0.00dB] [on] Front Right: Playback 74 [100%] [0.00dB] [on] Node 0x10 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Control: name="Speaker Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Speaker Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="CX20590 Analog", type="Audio", device=0 Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-Out vals: [0x3d 0x3d] Converter: stream=8, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x11 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-Out vals: [0x80 0x80] Converter: stream=8, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0

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[alsa-devel] [PATCH 1/2] ASoC: DPCM: make some DPCM API calls non static for compressed usage
by Liam Girdwood 17 Jan '14

17 Jan '14

The ASoC compressed code needs to call the internal DPCM APIs in order to dynamically route compressed data to different DAIs. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- include/sound/soc-dpcm.h | 22 ++++++++++++++++++++++ sound/soc/soc-pcm.c | 29 ++++++++++++----------------- 2 files changed, 34 insertions(+), 17 deletions(-) diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index 047d657..2883a7a 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -11,6 +11,7 @@ #ifndef __LINUX_SND_SOC_DPCM_H #define __LINUX_SND_SOC_DPCM_H +#include <linux/slab.h> #include <linux/list.h> #include <sound/pcm.h> @@ -135,4 +136,25 @@ int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute); int soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd); int soc_dpcm_runtime_update(struct snd_soc_card *); +int dpcm_path_get(struct snd_soc_pcm_runtime *fe, + int stream, struct snd_soc_dapm_widget_list **list_); +int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, + int stream, struct snd_soc_dapm_widget_list **list, int new); +int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream); +int dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream); +void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream); +void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream); +int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream); +int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int tream); +int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd); +int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream); +int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, + int event); + +static inline void dpcm_path_put(struct snd_soc_dapm_widget_list **list) +{ + kfree(*list); +} + + #endif diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 1534115..7023f7e 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -58,7 +58,7 @@ int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream, EXPORT_SYMBOL_GPL(snd_soc_set_runtime_hwparams); /* DPCM stream event, send event to FE and all active BEs. */ -static int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, +int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, int event) { struct snd_soc_dpcm *dpcm; @@ -877,7 +877,7 @@ static void dpcm_be_reparent(struct snd_soc_pcm_runtime *fe, } /* disconnect a BE and FE */ -static void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) +void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm, *d; @@ -973,7 +973,7 @@ static int widget_in_list(struct snd_soc_dapm_widget_list *list, return 0; } -static int dpcm_path_get(struct snd_soc_pcm_runtime *fe, +int dpcm_path_get(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_dapm_widget_list **list_) { struct snd_soc_dai *cpu_dai = fe->cpu_dai; @@ -995,11 +995,6 @@ static int dpcm_path_get(struct snd_soc_pcm_runtime *fe, return paths; } -static inline void dpcm_path_put(struct snd_soc_dapm_widget_list **list) -{ - kfree(*list); -} - static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_dapm_widget_list **list_) { @@ -1069,7 +1064,7 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, continue; /* don't connect if FE is not running */ - if (!fe->dpcm[stream].runtime) + if (!fe->dpcm[stream].runtime && !fe->fe_compr) continue; /* newly connected FE and BE */ @@ -1094,7 +1089,7 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, * Find the corresponding BE DAIs that source or sink audio to this * FE substream. */ -static int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, +int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_dapm_widget_list **list, int new) { if (new) @@ -1103,7 +1098,7 @@ static int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, return dpcm_prune_paths(fe, stream, list); } -static void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream) +void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; @@ -1141,7 +1136,7 @@ static void dpcm_be_dai_startup_unwind(struct snd_soc_pcm_runtime *fe, } } -static int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; int err, count = 0; @@ -1293,7 +1288,7 @@ be_err: return ret; } -static int dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; @@ -1354,7 +1349,7 @@ static int dpcm_fe_dai_shutdown(struct snd_pcm_substream *substream) return 0; } -static int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; @@ -1419,7 +1414,7 @@ static int dpcm_fe_dai_hw_free(struct snd_pcm_substream *substream) return 0; } -static int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; int ret; @@ -1549,7 +1544,7 @@ static int dpcm_do_trigger(struct snd_soc_dpcm *dpcm, return ret; } -static int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, +int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd) { struct snd_soc_dpcm *dpcm; @@ -1717,7 +1712,7 @@ out: return ret; } -static int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; int ret = 0; -- 1.8.3.2

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[alsa-devel] [PATCH v4 0/9] ASoC: fsl-ssi: offline/online configuration and cleanups
by Markus Pargmann 17 Jan '14

17 Jan '14

Hi, This patch series fixes some fsl-ssi code that does not act exactly as it is described in the reference manuals. The reference manuals before imx51 do mention that some register bits, including TDMAE/RDMAE, should not be changed while the SSI unit is enabled (SSIEN). At the same time the SDMA unit has a undefined request handling if there are DMA requests before the SDMA engine and its channel is configured. The reference manual states that SSI DMA requests are sent as soon as the FIFO and the DMAE bits are enabled. It does not mention a dependency to SCR's TE/RE bit. To avoid sending DMA requests before a channel is configured, we have to seperate the fsl-ssi behavior into offline and online configuration. SoCs before imx51 have to use offline configuration while later SoC versions can use online reconfiguration. This series adds support for online configuration and cleans up the configuration register code especially in fsl_ssi_trigger. At the end we have a seperation between enable/disable logic and the actual configuration register values used to enable/disable TX/RX. v4 is another rebase onto latest topic/fsl branch. It includes the DAI master mode changes from Nicolin. @Nicolin: Could you please test this series with your hardware to verify that your patch still works? I tested this series on mx53. The series is based on Mark Brown's topic/fsl branch. Regards, Markus Changes in v4: - Support for irqstats on all imx SoCs - Fix offline_config and use switch statement - Introduce ssi_private->irq_stats bool Changes in v3: - Rebase on latest topic/fsl - Added patch to cleanup probe function error handling - Fix interrupt stats for imx53 - Replace imx50-ssi by imx51-ssi Markus Pargmann (9): ASoC: fsl-ssi: Fix probe error handling ASoC: fsl-ssi: Move sysfs stats to debugfs ASoC: fsl-ssi: Add imx51-ssi and of_device_id matching ASoC: fsl-ssi: Fix interrupt stats for imx ASoC: fsl-ssi: Add offline_config flag ASoC: fsl-ssi: Add configuration helper functions ASoC: fsl-ssi: Move RX/TX configuration to seperate functions ASoC: fsl-ssi: Drop ac97 specific trigger function ARM: DTS: imx5* imx6*, use imx51-ssi arch/arm/boot/dts/imx53.dtsi | 10 +- arch/arm/boot/dts/imx6qdl.dtsi | 12 +- arch/arm/boot/dts/imx6sl.dtsi | 12 +- sound/soc/fsl/fsl_ssi.c | 567 ++++++++++++++++++++++++++++------------- 4 files changed, 411 insertions(+), 190 deletions(-) -- 1.8.5.1

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[alsa-devel] [PATCH] ASoC: core: Fix possible NULL pointer dereference of pcm->config
by Xiubo Li 16 Jan '14

16 Jan '14

Since the soc generic dmaengine pcm driver allows using the defualt settings, so the pcm->config maybe NULL. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/soc-generic-dmaengine-pcm.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 560a778..5bace12 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -176,17 +176,20 @@ static struct dma_chan *dmaengine_pcm_compat_request_channel( { struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); struct snd_dmaengine_dai_dma_data *dma_data; + dma_filter_fn fn = NULL; dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) && pcm->chan[0]) return pcm->chan[0]; - if (pcm->config->compat_request_channel) + if (pcm->config && pcm->config->compat_request_channel) return pcm->config->compat_request_channel(rtd, substream); - return snd_dmaengine_pcm_request_channel(pcm->config->compat_filter_fn, - dma_data->filter_data); + if (pcm->config) + fn = pcm->config->compat_filter_fn; + + return snd_dmaengine_pcm_request_channel(fn, dma_data->filter_data); } static bool dmaengine_pcm_can_report_residue(struct dma_chan *chan) -- 1.8.4

3 2

[alsa-devel] ASoC fixes for v3.13
by Mark Brown 16 Jan '14

16 Jan '14

The following changes since commit 7e22e91102c6b9df7c4ae2168910e19d2bb14cd6: Linux 3.13-rc8 (2014-01-12 17:04:18 +0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.13-rc8 for you to fetch changes up to c5281227c90578c7b7beffca56e8c3a68d4607de: Merge remote-tracking branches 'asoc/fix/adau1701', 'asoc/fix/fsl' and 'asoc/fix/tlv320aic32x4' into asoc-linus (2014-01-15 23:17:15 +0000) ---------------------------------------------------------------- ASoC: Fixes for v3.13 A few small fixes in drivers, nothing too remarkable here but all good to have - mainly these are fixes for things that were introduced in the last merge window but only just got useful testing. ---------------------------------------------------------------- Lars-Peter Clausen (1): ASoC: adau1701: Fix ADAU1701_SEROCTL_WORD_LEN_16 constant Mark Brown (1): Merge remote-tracking branches 'asoc/fix/adau1701', 'asoc/fix/fsl' and 'asoc/fix/tlv320aic32x4' into asoc-linus Markus Pargmann (2): ASoC: codec: tlv320aic32x4: Fix regmap range config ASoC: tlv320aic32x4: Fix regmap range_min Nicolin Chen (1): ASoC: imx-spdif: Use snd-soc-dummy CODEC driver to link card sound/soc/codecs/adau1701.c | 2 +- sound/soc/codecs/tlv320aic32x4.c | 4 +- sound/soc/fsl/Kconfig | 1 - sound/soc/fsl/imx-spdif.c | 84 ++++++++++------------------------------ 4 files changed, 24 insertions(+), 67 deletions(-)

2 4

[alsa-devel] [GIT PULL] ASoC updates for v3.13
by Mark Brown 16 Jan '14

16 Jan '14

The following changes since commit 7e22e91102c6b9df7c4ae2168910e19d2bb14cd6: Linux 3.13-rc8 (2014-01-12 17:04:18 +0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.13-rc8-2 for you to fetch changes up to c6affc0dba45bb2074ff56b21386d22eeaf682f8: Merge remote-tracking branches 'asoc/fix/adau1701' and 'asoc/fix/tlv320aic32x4' into asoc-linus (2014-01-16 14:02:52 +0000) ---------------------------------------------------------------- ASoC: Fixes for v3.13 A few small fixes in drivers, nothing too remarkable here but all good to have - mainly these are fixes for things that were introduced in the last merge window but only just got useful testing. ---------------------------------------------------------------- Lars-Peter Clausen (1): ASoC: adau1701: Fix ADAU1701_SEROCTL_WORD_LEN_16 constant Mark Brown (1): Merge remote-tracking branches 'asoc/fix/adau1701' and 'asoc/fix/tlv320aic32x4' into asoc-linus Markus Pargmann (2): ASoC: codec: tlv320aic32x4: Fix regmap range config ASoC: tlv320aic32x4: Fix regmap range_min sound/soc/codecs/adau1701.c | 2 +- sound/soc/codecs/tlv320aic32x4.c | 4 ++-- 2 files changed, 3 insertions(+), 3 deletions(-)

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[alsa-devel] [GIT PULL] ASoC updates for v3.14
by Mark Brown 16 Jan '14

16 Jan '14

The following changes since commit cdef2e5f358db5fd27eec4c72eb295c232287813: sound: oss: remove last sleep_on users (2014-01-14 16:12:07 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.14-2 for you to fetch changes up to 701caa51a2ce74182d39380ca11defeb163d98c1: Merge remote-tracking branches 'asoc/topic/adsp', 'asoc/topic/atmel', 'asoc/topic/bcm2835', 'asoc/topic/docs', 'asoc/topic/fsl', 'asoc/topic/generic', 'asoc/topic/kirkwood', 'asoc/topic/mc13783', 'asoc/topic/mxs', 'asoc/topic/nuc900', 'asoc/topic/sai', 'asoc/topic/sh', 'asoc/topic/ssm2602', 'asoc/topic/tlv320aic3x', 'asoc/topic/twl4030', 'asoc/topic/ux500', 'asoc/topic/width' and 'asoc/topic/x86' into for-tiwai (2014-01-16 12:44:01 +0000) ---------------------------------------------------------------- ASoC: More updates for v3.14 A few more updates for v3.14 since the last set, highlights include: - Lots of DMA updates from Lars-Peter - Improvements to the constraints handling code from Lars-Peter - A very helpful conversion of the TWL4030 driver to regmap from Peter - A new driver for the Freescale ESAI controller from Nicolin Chen - Conversion of some of the drivers to use params_width() ---------------------------------------------------------------- Alexander Shiyan (5): ASoC: fsl_ssi: Fix printing return code on clk error ASoC: mc13783: Use module_platform_driver_probe() ASoC: mc13783: Use core error messages if registration fails ASoC: mc13783: Drop fixed ADC & DAC ports usage ASoC: mc13783: trivial: Cleanup module Andreas Pretzsch (1): ASoC: ssm2602: add 16kHz sampling rate support Arun Shamanna Lakshmi (1): ASoC: dapm: Change prototype of soc_widget_read Charles Keepax (5): ASoC: wm_adsp: Factor out ADSP2 boot proceedure ASoC: wm_adsp: Start DSP booting earlier in the DAPM process ASoC: wm_adsp: Mark wm_adsp2_boot_work as static mfd: wm5110: Add registers for headphone short circuit control ASoC: wm5110: Add controls for headphone short circuit protection Fabio Estevam (1): ASoC: fsl: fsl_ssi: Use '%ld' to print 'long int' Jean-Francois Moine (1): ASoC: pcm: Fix lack of platform bespoke_trigger() call Lars-Peter Clausen (24): ASoC: atmel: Don't set unused struct snd_pcm_hardware fields ASoC: dpcm: Add helper function for initializing runtime pcm ASoC: dpcm: Allow PCMs to omit the set of supported formats ASoC: fsl: Don't set unused struct snd_pcm_hardware fields ASoC: mxs: Don't set unused struct snd_pcm_hardware fields ASoC: mxs: Remove SND_DMAENGINE_PCM_FLAG_NO_RESIDUE flag ASoC: adau1701: Fix ADAU1701_SEROCTL_WORD_LEN_16 constant ASoC: intel: Don't set unused struct snd_pcm_hardware fields ASoC: kirkwood: Don't set unused struct snd_pcm_hardware fields ASoC: nuc900: Don't set unused struct snd_pcm_hardware fields ASoC: sh: Don't set unused struct snd_pcm_hardware fields ASoC: ux500: Don't set unused struct snd_pcm_hardware fields ASoC: Allow PCMs to restrict the supported formats ASoC: pcm: Properly initialize hw->rate_max ASoC: fsl: Don't mix SNDRV_PCM_RATE_CONTINUOUS with specific rates ASoC: s6000: Don't mix SNDRV_PCM_RATE_CONTINUOUS with specific rates ALSA: Add helper function for intersecting two rate masks ASoC: pcm: Use snd_pcm_rate_mask_intersect() helper dma: Indicate residue granularity in dma_slave_caps dma: pl330: Set residue_granularity ASoC: generic-dmaengine-pcm: Check NO_RESIDUE flag at runtime ASoC: generic-dmaengine-pcm: Check DMA residue granularity ASoC: axi-{spdif,i2s}: Remove SND_DMAENGINE_PCM_FLAG_NO_RESIDUE flag ASoC: samsung: Remove SND_DMAENGINE_PCM_FLAG_NO_RESIDUE flag Lee Jones (10): ARM: ux500: Don't use enums for MSP IDs - for easy DT conversion ASoC: ux500: Provide better checking for Device Tree and/or Platform Data ASoC: Ux500: Match platform by device node when booting Device Tree ASoC: ux500_pcm: Stop pretending that we support varying address widths ASoC: ux500_pcm: Expect different saved DMA data when obtaining from DAI store ASoC: ux500_pcm: Extend Device Tree support to deal with DMA data ASoC: ux500: Store DMA data in the DAI differently in the pdata and DT case ASoC: ux500_pcm: Take out pointless dev_dbg() call ASoC: ux500_pcm: Differentiate between pdata and DT initialisation ASoC: ux500: Dramatically reduce the size of the DAI driver data struct Liam Girdwood (4): ASoC: docs: Update the Overview document ASoC: dpcm: Explicitly set BE DAI link supported stream directions ASoC: utils: Add internal call to determine if DAI is dummy. ASoC: sapm: Automatically connect DAI link widgets in DAPM graph. Mark Brown (30): Merge tag 'ib-asoc-3.14.2' of git://git.linaro.org/people/ljones/mfd into asoc-tlw4030 ASoC: ad1836: Use params_width() rather than explicit memory format ASoC: ad193x: Use params_width() rather than memory format ASoC: adau1373: Use params_width() rather than memory format ASoC: adau1701: Use params_width() rather than memory format ASoC: adav80x: Use params_width() rather than memory format Merge tag 'v3.13-rc3' into asoc-arizona ASoC: alc5623: Use params_width() rather than memory format ASoC: alc5632: Use params_width() rather than memory format ASoC: cs42l51: Use params_width() rather than memory format ASoC: da7210: Use params_width() rather than memory format ASoC: da7213: Use params_width() rather than memory format ASoC: da732x: Use params_width() rather than memory format ASoC: da9055: Use params_width() rather than memory format ASoC: isabelle: Use params_width() rather than memory format ASoC: max98088: Use params_width() rather than memory format ASoC: max98090: Use params_width() rather than memory format ASoC: max98095: Use params_width() rather than memory format ASoC: max9850: Use params_width() rather than memory format Merge tag 'v3.13-rc3' into asoc-pcm Merge branch 'topic/axi' of git://git.kernel.org/.../broonie/sound into asoc-dma Merge branch 'topic/samsung' of git://git.kernel.org/.../broonie/sound into asoc-dma ASoC: core: Return -ENOTSUPP from set_sysclk() if no operation provided Merge remote-tracking branches 'asoc/fix/adau1701' and 'asoc/fix/tlv320aic32x4' into for-tiwai Merge remote-tracking branch 'asoc/topic/core' into for-tiwai Merge remote-tracking branch 'asoc/topic/dapm' into for-tiwai Merge remote-tracking branch 'asoc/topic/dma' into for-tiwai Merge remote-tracking branch 'asoc/topic/pcm' into for-tiwai Merge remote-tracking branch 'asoc/topic/arizona' into for-tiwai Merge remote-tracking branches 'asoc/topic/adsp', 'asoc/topic/atmel', 'asoc/topic/bcm2835', 'asoc/topic/docs', 'asoc/topic/fsl', 'asoc/topic/generic', 'asoc/topic/kirkwood', 'asoc/topic/mc13783', 'asoc/topic/mxs', 'asoc/topic/nuc900', 'asoc/topic/sai', 'asoc/topic/sh', 'asoc/topic/ssm2602', 'asoc/topic/tlv320aic3x', 'asoc/topic/twl4030', 'asoc/topic/ux500', 'asoc/topic/width' and 'asoc/topic/x86' into for-tiwai Markus Pargmann (13): ASoC: fsl-ssi: Fix probe error handling ASoC: fsl-ssi: Move sysfs stats to debugfs ASoC: fsl-ssi: Add imx51-ssi and of_device_id matching ASoC: fsl-ssi: Fix interrupt stats for imx ASoC: fsl-ssi: Add offline_config flag ASoC: fsl-ssi: Add configuration helper functions ASoC: fsl-ssi: Move RX/TX configuration to seperate functions ASoC: fsl-ssi: Drop ac97 specific trigger function ASoC: fsl-ssi doc: Add list of supported compatibles ASoC: fsl-ssi: Fix stats compile warning ASoC: codec: tlv320aic32x4: Fix regmap range config ASoC: tlv320aic3x: Add tlv320aic32x4 as compatible ASoC: tlv320aic32x4: Fix regmap range_min Nicolin Chen (2): ASoC: fsl_ssi: Set default slot number for common cases ASoC: fsl_esai: Add ESAI CPU DAI driver Peter Ujfalusi (11): mfd: twl-core: Simplify IO wrapper functions by moving common code out mfd: twl-core: API to set the regcache bypass for a given regmap in twl mfd: twl-core: Enable regcache for audio registers ASoC: twl4030: Separate write condition checking from I/O function ASoC: twl4030: Remove check defaults functionality ASoC: twl4030: Remove reset registers functionality ASoC: twl4030: Introduce local ctl register cache ASoC: twl4030: Remove local reg cache ASoC: twl4030: Parameter alignment fixes (for code consistency) ASoC: twl4030: Move the ctl cache update local to twl4030_write() function ASoC: twl4030: Pass the twl4030_priv directly to twl4030_can_write_to_chip() Takashi Iwai (1): ASoC: bcm: Remove obsoleted Kconfig dependency Wei Yongjun (2): ASoC: ux500: Fix sparse non static symbol warning ASoC: fsl-ssi: Add missing clk_disable_unprepare() on error in fsl_ssi_probe() Xiubo Li (8): ASoC: simple-card: Fix the sysclk selection. ASoC: simple-card: fix the DAPM routes map parsing ASoC: simple-card: keep the property's name the same pattern ASoC: fsl-sai: Clean up the code ASoC: simple-card: fix a bug where cinfo will be NULL before using it ASoC: simple-card: fix the cinfo error check ASoC: simple-card: use snd_soc_card_set/get_drvdata ASoC: simple-card: fix one bug to writing to the platform data .../devicetree/bindings/sound/fsl,esai.txt | 50 ++ .../devicetree/bindings/sound/fsl,ssi.txt | 7 +- .../devicetree/bindings/sound/simple-card.txt | 2 +- .../devicetree/bindings/sound/tlv320aic3x.txt | 1 + Documentation/sound/alsa/soc/overview.txt | 27 +- arch/arm/mach-ux500/board-mop500-audio.c | 8 +- drivers/dma/pl330.c | 1 + drivers/mfd/twl-core.c | 190 ++++- drivers/mfd/wm5110-tables.c | 6 + include/linux/dmaengine.h | 28 + include/linux/i2c/twl.h | 5 +- include/linux/mfd/arizona/registers.h | 27 + include/linux/platform_data/asoc-ux500-msp.h | 9 +- include/sound/pcm.h | 2 + include/sound/soc-dai.h | 2 + include/sound/soc-dapm.h | 1 + include/sound/soc.h | 4 + sound/core/pcm_misc.c | 39 + sound/soc/adi/axi-i2s.c | 3 +- sound/soc/adi/axi-spdif.c | 3 +- sound/soc/atmel/atmel-pcm-dma.c | 1 - sound/soc/atmel/atmel-pcm-pdc.c | 1 - sound/soc/bcm/Kconfig | 1 - sound/soc/codecs/ad1836.c | 10 +- sound/soc/codecs/ad193x.c | 10 +- sound/soc/codecs/adau1373.c | 10 +- sound/soc/codecs/adau1701.c | 36 +- sound/soc/codecs/adav80x.c | 30 +- sound/soc/codecs/alc5623.c | 10 +- sound/soc/codecs/alc5632.c | 8 +- sound/soc/codecs/arizona.h | 17 +- sound/soc/codecs/cs42l51.c | 14 +- sound/soc/codecs/da7210.c | 10 +- sound/soc/codecs/da7213.c | 10 +- sound/soc/codecs/da732x.c | 10 +- sound/soc/codecs/da9055.c | 10 +- sound/soc/codecs/isabelle.c | 6 +- sound/soc/codecs/max98088.c | 6 +- sound/soc/codecs/max98090.c | 4 +- sound/soc/codecs/max98095.c | 6 +- sound/soc/codecs/max9850.c | 8 +- sound/soc/codecs/mc13783.c | 34 +- sound/soc/codecs/ssm2602.c | 14 +- sound/soc/codecs/tlv320aic32x4.c | 4 +- sound/soc/codecs/twl4030.c | 376 ++++------ sound/soc/codecs/wm5110.c | 7 + sound/soc/codecs/wm_adsp.c | 199 +++-- sound/soc/codecs/wm_adsp.h | 12 +- sound/soc/fsl/Kconfig | 3 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_dma.c | 7 - sound/soc/fsl/fsl_esai.c | 815 +++++++++++++++++++++ sound/soc/fsl/fsl_esai.h | 354 +++++++++ sound/soc/fsl/fsl_sai.c | 7 +- sound/soc/fsl/fsl_ssi.c | 586 ++++++++++----- sound/soc/fsl/imx-pcm-dma.c | 3 - sound/soc/fsl/imx-pcm-fiq.c | 3 - sound/soc/fsl/mpc5200_dma.c | 4 - sound/soc/fsl/mpc5200_psc_i2s.c | 3 +- sound/soc/generic/simple-card.c | 70 +- sound/soc/intel/sst_platform.c | 10 - sound/soc/intel/sst_platform.h | 4 - sound/soc/kirkwood/kirkwood-dma.c | 16 - sound/soc/mxs/mxs-pcm.c | 6 - sound/soc/nuc900/nuc900-pcm.c | 3 - sound/soc/s6000/s6000-i2s.c | 3 +- sound/soc/samsung/dmaengine.c | 1 - sound/soc/sh/dma-sh7760.c | 17 - sound/soc/sh/fsi.c | 6 - sound/soc/sh/rcar/core.c | 6 - sound/soc/soc-core.c | 5 +- sound/soc/soc-dapm.c | 75 +- sound/soc/soc-generic-dmaengine-pcm.c | 66 +- sound/soc/soc-pcm.c | 58 +- sound/soc/soc-utils.c | 7 + sound/soc/ux500/mop500.c | 2 + sound/soc/ux500/ux500_msp_dai.c | 146 ++-- sound/soc/ux500/ux500_msp_i2s.c | 56 +- sound/soc/ux500/ux500_msp_i2s.h | 2 +- sound/soc/ux500/ux500_pcm.c | 65 +- 80 files changed, 2656 insertions(+), 1034 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,esai.txt create mode 100644 sound/soc/fsl/fsl_esai.c create mode 100644 sound/soc/fsl/fsl_esai.h

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[alsa-devel] different sample rate in alsa.conf file, ALSA DMIX mode
by Pavel Botev 16 Jan '14

16 Jan '14

Hi ALSA support team, Can we set more sample rate for DMIX mode in the alsa.conf file than the default one (defaults.pcm.dmix.rate 48000)? Can we use one alsa.conf file, with many sample rates (8KHz, 11.025KHz, 16KHz, 22.05KHz, 32KHz, 44.1KHz, 48KHz, 64KHz, 88.2KHz and 96KHz), not with just one default sample rate? The problem is that when we use one sample rate in alsa.conf and other in the ALSA application, we have resampling executed which is very CPU intensive. Thus we search a way to use alsa.conf file with many sample rates, thus avoiding resampling. If not possible, can we change the algorithm of resampling, thus have low CPU usage when resampling is activated? Thanks for your time. Best regards, Pavel

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