June 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] ASoC updates for 3.5
by Mark Brown 18 Jun '12

18 Jun '12

The following changes since commit 69c5b7530651c8e375d2fbecd157a9a9a20c4cc9: ASoC: tegra: add MODULE_DEVICE_TABLE to tegra30_ahub (2012-06-08 08:49:30 +0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.5 for you to fetch changes up to 3419ae781f1592b3d367107db6500090495490cd: ASoC: tegra+wm8903: turn of mic detect when card is removed (2012-06-12 11:38:48 +0800) ---------------------------------------------------------------- ASoC: Updates for 3.5 All driver specific and fairly small. The pxa-ssp changes are larger than I'd like but they're build failures and are pretty clear to inspection. ---------------------------------------------------------------- Mark Brown (5): ASoC: wm8904: Fix GPIO and MICBIAS initialisation for regmap conversion ASoC: wm8904: Fix cache only management ASoC: wm8996: Remove spurious regulator_bulk_free() ASoC: wm8996: Move reset before the initial regulator disable ASoC: wm8996: Mark the CODEC as cache only when powering off on boot Qiao Zhou (2): ARM: MMP: add pxa910-ssp into ssp_id_table ASoC: fix pxa-ssp compiling issue under mach-mmp Stephen Warren (1): ASoC: tegra+wm8903: turn of mic detect when card is removed arch/arm/plat-pxa/ssp.c | 1 + include/linux/pxa2xx_ssp.h | 2 ++ include/linux/spi/pxa2xx_spi.h | 2 +- sound/soc/codecs/wm8904.c | 26 ++++++++++++-------------- sound/soc/codecs/wm8996.c | 8 +++----- sound/soc/pxa/pxa-ssp.c | 38 +++++++++++--------------------------- sound/soc/tegra/tegra_wm8903.c | 13 +++++++++++++ 7 files changed, 43 insertions(+), 47 deletions(-)

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[alsa-devel] [patch] ASoC: Ux500: unlock on an error path
by Dan Carpenter 17 Jun '12

17 Jun '12

There is a missing mutex_unlock() here. The cleanup path also has more debug output. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 389dd66..3c79592 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -1235,7 +1235,8 @@ static int anc_status_control_put(struct snd_kcontrol *kcontrol, req != ANC_APPLY_IIR) { dev_err(dev, "%s: ERROR: Unsupported status to set '%s'!\n", __func__, enum_anc_state[req]); - return -EINVAL; + status = -EINVAL; + goto cleanup; } apply_fir = req == ANC_APPLY_FIR || req == ANC_APPLY_FIR_IIR; apply_iir = req == ANC_APPLY_IIR || req == ANC_APPLY_FIR_IIR;

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[alsa-devel] [PATCH] ASoC: Ux500: Add machine-driver
by Ola Lilja 17 Jun '12

17 Jun '12

Add machine-driver for ST-Ericsson U8500 platform, including support for the AB8500-codec. Signed-off-by: Ola Lilja <ola.o.lilja(a)stericsson.com> --- sound/soc/ux500/Kconfig | 11 + sound/soc/ux500/Makefile | 3 + sound/soc/ux500/mop500.c | 113 ++++++++++ sound/soc/ux500/mop500_ab8500.c | 431 +++++++++++++++++++++++++++++++++++++++ sound/soc/ux500/mop500_ab8500.h | 22 ++ 5 files changed, 580 insertions(+), 0 deletions(-) create mode 100644 sound/soc/ux500/mop500.c create mode 100644 sound/soc/ux500/mop500_ab8500.c create mode 100644 sound/soc/ux500/mop500_ab8500.h diff --git a/sound/soc/ux500/Kconfig b/sound/soc/ux500/Kconfig index 1d38515..069330d 100644 --- a/sound/soc/ux500/Kconfig +++ b/sound/soc/ux500/Kconfig @@ -19,3 +19,14 @@ config SND_SOC_UX500_PLAT_DMA select SND_SOC_DMAENGINE_PCM help Say Y if you want to enable the Ux500 platform-driver. + ++config SND_SOC_UX500_MACH_MOP500 ++ tristate "Machine - MOP500 (Ux500 + AB8500)" + depends on AB8500_CORE && AB8500_GPADC && SND_SOC_UX500 + select SND_SOC_AB8500_CODEC + select SND_SOC_UX500_PLAT_MSP_I2S + select SND_SOC_UX500_PLAT_DMA + help + Select this to enable the MOP500 machine-driver. + This will enable platform-drivers for: Ux500 + This will enable codec-drivers for: AB8500 diff --git a/sound/soc/ux500/Makefile b/sound/soc/ux500/Makefile index 4634bf0..cce0c11 100644 --- a/sound/soc/ux500/Makefile +++ b/sound/soc/ux500/Makefile @@ -5,3 +5,6 @@ obj-$(CONFIG_SND_SOC_UX500_PLAT_MSP_I2S) += snd-soc-ux500-plat-msp-i2s.o snd-soc-ux500-plat-dma-objs := ux500_pcm.o obj-$(CONFIG_SND_SOC_UX500_PLAT_DMA) += snd-soc-ux500-plat-dma.o + +snd-soc-ux500-mach-mop500-objs := mop500.o mop500_ab8500.o +obj-$(CONFIG_SND_SOC_UX500_MACH_MOP500) += snd-soc-ux500-mach-mop500.o diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c new file mode 100644 index 0000000..31c4d26 --- /dev/null +++ b/sound/soc/ux500/mop500.c @@ -0,0 +1,113 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja (ola.o.lilja(a)stericsson.com) + * for ST-Ericsson. + * + * License terms: + * + * 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 <asm/mach-types.h> + +#include <linux/module.h> +#include <linux/io.h> +#include <linux/spi/spi.h> + +#include <sound/soc.h> +#include <sound/initval.h> + +#include "ux500_pcm.h" +#include "ux500_msp_dai.h" + +#include <mop500_ab8500.h> + +/* Define the whole MOP500 soundcard, linking platform to the codec-drivers */ +struct snd_soc_dai_link mop500_dai_links[] = { + { + .name = "ab8500_0", + .stream_name = "ab8500_0", + .cpu_dai_name = "ux500-msp-i2s.1", + .codec_dai_name = "ab8500-codec-dai.0", + .platform_name = "ux500-pcm.0", + .codec_name = "ab8500-codec.0", + .init = mop500_ab8500_machine_init, + .ops = mop500_ab8500_ops, + }, + { + .name = "ab8500_1", + .stream_name = "ab8500_1", + .cpu_dai_name = "ux500-msp-i2s.3", + .codec_dai_name = "ab8500-codec-dai.1", + .platform_name = "ux500-pcm.0", + .codec_name = "ab8500-codec.0", + .init = NULL, + .ops = mop500_ab8500_ops, + }, +}; + +static struct snd_soc_card mop500_card = { + .name = "MOP500-card", + .probe = NULL, + .dai_link = mop500_dai_links, + .num_links = ARRAY_SIZE(mop500_dai_links), +}; + +static int __devinit mop500_probe(struct platform_device *pdev) +{ + int ret; + + pr_debug("%s: Enter.\n", __func__); + + dev_dbg(&pdev->dev, "%s: Enter.\n", __func__); + + mop500_card.dev = &pdev->dev; + + dev_dbg(&pdev->dev, "%s: Card %s: Set platform drvdata.\n", + __func__, mop500_card.name); + platform_set_drvdata(pdev, &mop500_card); + + snd_soc_card_set_drvdata(&mop500_card, NULL); + + dev_dbg(&pdev->dev, "%s: Card %s: num_links = %d\n", + __func__, mop500_card.name, mop500_card.num_links); + dev_dbg(&pdev->dev, "%s: Card %s: DAI-link 0: name = %s\n", + __func__, mop500_card.name, mop500_card.dai_link[0].name); + dev_dbg(&pdev->dev, "%s: Card %s: DAI-link 0: stream_name = %s\n", + __func__, mop500_card.name, + mop500_card.dai_link[0].stream_name); + + ret = snd_soc_register_card(&mop500_card); + if (ret) + dev_err(&pdev->dev, + "Error: snd_soc_register_card failed (%d)!\n", + ret); + + return ret; +} + +static int __devexit mop500_remove(struct platform_device *pdev) +{ + struct snd_soc_card *mop500_card = platform_get_drvdata(pdev); + + pr_debug("%s: Enter.\n", __func__); + + snd_soc_unregister_card(mop500_card); + mop500_ab8500_remove(mop500_card); + + return 0; +} + +static struct platform_driver snd_soc_mop500_driver = { + .driver = { + .owner = THIS_MODULE, + .name = "snd-soc-mop500", + }, + .probe = mop500_probe, + .remove = __devexit_p(mop500_remove), +}; + +module_platform_driver(snd_soc_mop500_driver); diff --git a/sound/soc/ux500/mop500_ab8500.c b/sound/soc/ux500/mop500_ab8500.c new file mode 100644 index 0000000..78cce23 --- /dev/null +++ b/sound/soc/ux500/mop500_ab8500.c @@ -0,0 +1,431 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, + * Kristoffer Karlsson <kristoffer.karlsson(a)stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * 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/device.h> +#include <linux/io.h> +#include <linux/clk.h> + +#include <mach/hardware.h> + +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> + +#include "ux500_pcm.h" +#include "ux500_msp_dai.h" +#include "../codecs/ab8500-codec.h" + +#define TX_SLOT_MONO 0x0008 +#define TX_SLOT_STEREO 0x000a +#define RX_SLOT_MONO 0x0001 +#define RX_SLOT_STEREO 0x0003 +#define TX_SLOT_8CH 0x00FF +#define RX_SLOT_8CH 0x00FF + +#define DEF_TX_SLOTS TX_SLOT_STEREO +#define DEF_RX_SLOTS RX_SLOT_MONO + +#define DRIVERMODE_NORMAL 0 +#define DRIVERMODE_CODEC_ONLY 1 + +/* Slot configuration */ +static unsigned int tx_slots = DEF_TX_SLOTS; +static unsigned int rx_slots = DEF_RX_SLOTS; + +/* Clocks */ +static const char * const enum_mclk[] = { + "SYSCLK", + "ULPCLK" +}; +enum mclk { + MCLK_SYSCLK, + MCLK_ULPCLK, +}; + +static SOC_ENUM_SINGLE_EXT_DECL(soc_enum_mclk, enum_mclk); + +/* Private data for machine-part MOP500<->AB8500 */ +struct mop500_ab8500_drvdata { + /* Clocks */ + enum mclk mclk_sel; + struct clk *clk_ptr_intclk; + struct clk *clk_ptr_sysclk; + struct clk *clk_ptr_ulpclk; +}; + +static inline const char *get_mclk_str(enum mclk mclk_sel) +{ + switch (mclk_sel) { + case MCLK_SYSCLK: + return "SYSCLK"; + case MCLK_ULPCLK: + return "ULPCLK"; + default: + return "Unknown"; + } +} + +static int mop500_ab8500_set_mclk(struct device *dev, + struct mop500_ab8500_drvdata *drvdata) +{ + int status; + struct clk *clk_ptr; + + if (IS_ERR(drvdata->clk_ptr_intclk)) { + dev_err(dev, + "%s: ERROR: intclk not initialized!\n", __func__); + return -EIO; + } + + switch (drvdata->mclk_sel) { + case MCLK_SYSCLK: + clk_ptr = drvdata->clk_ptr_sysclk; + break; + case MCLK_ULPCLK: + clk_ptr = drvdata->clk_ptr_ulpclk; + break; + default: + return -EINVAL; + } + + if (IS_ERR(clk_ptr)) { + dev_err(dev, "%s: ERROR: %s not initialized!\n", __func__, + get_mclk_str(drvdata->mclk_sel)); + return -EIO; + } + + status = clk_set_parent(drvdata->clk_ptr_intclk, clk_ptr); + if (status) + dev_err(dev, + "%s: ERROR: Setting intclk parent to %s failed (ret = %d)!", + __func__, get_mclk_str(drvdata->mclk_sel), status); + else + dev_dbg(dev, + "%s: intclk parent changed to %s.\n", + __func__, get_mclk_str(drvdata->mclk_sel)); + + return status; +} + +/* + * Control-events + */ + +static int mclk_input_control_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct mop500_ab8500_drvdata *drvdata = + snd_soc_card_get_drvdata(codec->card); + + ucontrol->value.enumerated.item[0] = drvdata->mclk_sel; + + return 0; +} + +static int mclk_input_control_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct mop500_ab8500_drvdata *drvdata = + snd_soc_card_get_drvdata(codec->card); + unsigned int val = ucontrol->value.enumerated.item[0]; + + if (val > (unsigned int)MCLK_ULPCLK) + return -EINVAL; + if (drvdata->mclk_sel == val) + return 0; + + drvdata->mclk_sel = val; + + return 1; +} + +/* + * Controls + */ + +static struct snd_kcontrol_new mop500_ab8500_ctrls[] = { + SOC_ENUM_EXT("Master Clock Select", + soc_enum_mclk, + mclk_input_control_get, mclk_input_control_put), + /* Digital interface - Clocks */ + SOC_SINGLE("Digital Interface Master Generator Switch", + AB8500_DIGIFCONF1, AB8500_DIGIFCONF1_ENMASTGEN, + 1, 0), + SOC_SINGLE("Digital Interface 0 Bit-clock Switch", + AB8500_DIGIFCONF1, AB8500_DIGIFCONF1_ENFSBITCLK0, + 1, 0), + SOC_SINGLE("Digital Interface 1 Bit-clock Switch", + AB8500_DIGIFCONF1, AB8500_DIGIFCONF1_ENFSBITCLK1, + 1, 0), + SOC_DAPM_PIN_SWITCH("Headset Left"), + SOC_DAPM_PIN_SWITCH("Headset Right"), + SOC_DAPM_PIN_SWITCH("Earpiece"), + SOC_DAPM_PIN_SWITCH("Speaker Left"), + SOC_DAPM_PIN_SWITCH("Speaker Right"), + SOC_DAPM_PIN_SWITCH("LineOut Left"), + SOC_DAPM_PIN_SWITCH("LineOut Right"), + SOC_DAPM_PIN_SWITCH("Vibra 1"), + SOC_DAPM_PIN_SWITCH("Vibra 2"), + SOC_DAPM_PIN_SWITCH("Mic 1"), + SOC_DAPM_PIN_SWITCH("Mic 2"), + SOC_DAPM_PIN_SWITCH("LineIn Left"), + SOC_DAPM_PIN_SWITCH("LineIn Right"), + SOC_DAPM_PIN_SWITCH("DMic 1"), + SOC_DAPM_PIN_SWITCH("DMic 2"), + SOC_DAPM_PIN_SWITCH("DMic 3"), + SOC_DAPM_PIN_SWITCH("DMic 4"), + SOC_DAPM_PIN_SWITCH("DMic 5"), + SOC_DAPM_PIN_SWITCH("DMic 6"), +}; + +/* ASoC */ + +int mop500_ab8500_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + /* Set audio-clock source */ + return mop500_ab8500_set_mclk(rtd->card->dev, + snd_soc_card_get_drvdata(rtd->card)); +} + +void mop500_ab8500_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct device *dev = rtd->card->dev; + + dev_dbg(dev, "%s: Enter\n", __func__); + + /* Reset slots configuration to default(s) */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + tx_slots = DEF_TX_SLOTS; + else + rx_slots = DEF_RX_SLOTS; +} + +int mop500_ab8500_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 device *dev = rtd->card->dev; + unsigned int fmt; + int channels, ret = 0, driver_mode, slots; + unsigned int sw_codec, sw_cpu; + bool is_playback; + + dev_dbg(dev, "%s: Enter\n", __func__); + + dev_dbg(dev, "%s: substream->pcm->name = %s\n" + "substream->pcm->id = %s.\n" + "substream->name = %s.\n" + "substream->number = %d.\n", + __func__, + substream->pcm->name, + substream->pcm->id, + substream->name, + substream->number); + + channels = params_channels(params); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S32_LE: + sw_cpu = 32; + break; + + case SNDRV_PCM_FORMAT_S16_LE: + sw_cpu = 16; + break; + + default: + return -EINVAL; + } + + /* Setup codec depending on driver-mode */ + if (channels == 8) + driver_mode = DRIVERMODE_CODEC_ONLY; + else + driver_mode = DRIVERMODE_NORMAL; + dev_dbg(dev, "%s: Driver-mode: %s.\n", __func__, + (driver_mode == DRIVERMODE_NORMAL) ? "NORMAL" : "CODEC_ONLY"); + + /* Setup format */ + + if (driver_mode == DRIVERMODE_NORMAL) { + fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CONT; + } else { + fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_GATED; + } + + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + dev_err(dev, + "%s: ERROR: snd_soc_dai_set_fmt failed for codec_dai (ret = %d)!\n", + __func__, ret); + return ret; + } + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) { + dev_err(dev, + "%s: ERROR: snd_soc_dai_set_fmt failed for cpu_dai (ret = %d)!\n", + __func__, ret); + return ret; + } + + /* Setup TDM-slots */ + + is_playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + switch (channels) { + case 1: + slots = 16; + tx_slots = (is_playback) ? TX_SLOT_MONO : 0; + rx_slots = (is_playback) ? 0 : RX_SLOT_MONO; + break; + case 2: + slots = 16; + tx_slots = (is_playback) ? TX_SLOT_STEREO : 0; + rx_slots = (is_playback) ? 0 : RX_SLOT_STEREO; + break; + case 8: + slots = 16; + tx_slots = (is_playback) ? TX_SLOT_8CH : 0; + rx_slots = (is_playback) ? 0 : RX_SLOT_8CH; + break; + default: + return -EINVAL; + } + + if (driver_mode == DRIVERMODE_NORMAL) + sw_codec = sw_cpu; + else + sw_codec = 20; + + dev_dbg(dev, "%s: CPU-DAI TDM: TX=0x%04X RX=0x%04x\n", __func__, + tx_slots, rx_slots); + ret = snd_soc_dai_set_tdm_slot(cpu_dai, tx_slots, rx_slots, slots, + sw_cpu); + if (ret) + return ret; + + dev_dbg(dev, "%s: CODEC-DAI TDM: TX=0x%04X RX=0x%04x\n", __func__, + tx_slots, rx_slots); + ret = snd_soc_dai_set_tdm_slot(codec_dai, tx_slots, rx_slots, slots, + sw_codec); + if (ret) + return ret; + + return 0; +} + +struct snd_soc_ops mop500_ab8500_ops[] = { + { + .hw_params = mop500_ab8500_hw_params, + .startup = mop500_ab8500_startup, + .shutdown = mop500_ab8500_shutdown, + } +}; + +int mop500_ab8500_machine_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct device *dev = rtd->card->dev; + struct mop500_ab8500_drvdata *drvdata; + int ret; + + dev_dbg(dev, "%s Enter.\n", __func__); + + /* Create driver private-data struct */ + drvdata = devm_kzalloc(dev, sizeof(struct mop500_ab8500_drvdata), + GFP_KERNEL); + snd_soc_card_set_drvdata(rtd->card, drvdata); + + /* Setup clocks */ + + drvdata->clk_ptr_sysclk = clk_get(dev, "sysclk"); + if (IS_ERR(drvdata->clk_ptr_sysclk)) + dev_warn(dev, "%s: WARNING: clk_get failed for 'sysclk'!\n", + __func__); + drvdata->clk_ptr_ulpclk = clk_get(dev, "ulpclk"); + if (IS_ERR(drvdata->clk_ptr_ulpclk)) + dev_warn(dev, "%s: WARNING: clk_get failed for 'ulpclk'!\n", + __func__); + drvdata->clk_ptr_intclk = clk_get(dev, "intclk"); + if (IS_ERR(drvdata->clk_ptr_intclk)) + dev_warn(dev, "%s: WARNING: clk_get failed for 'intclk'!\n", + __func__); + + /* Set intclk default parent to ulpclk */ + drvdata->mclk_sel = MCLK_ULPCLK; + ret = mop500_ab8500_set_mclk(dev, drvdata); + if (ret < 0) + dev_warn(dev, "%s: WARNING: mop500_ab8500_set_mclk!\n", + __func__); + + drvdata->mclk_sel = MCLK_ULPCLK; + + /* Add controls */ + ret = snd_soc_add_codec_controls(codec, mop500_ab8500_ctrls, + ARRAY_SIZE(mop500_ab8500_ctrls)); + if (ret < 0) { + pr_err("%s: Failed to add machine-controls (%d)!\n", + __func__, ret); + return ret; + } + + ret = snd_soc_dapm_disable_pin(&codec->dapm, "Earpiece"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Speaker Left"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Speaker Right"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineOut Left"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineOut Right"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Vibra 1"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Vibra 2"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Mic 1"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Mic 2"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineIn Left"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineIn Right"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 1"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 2"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 3"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 4"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 5"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 6"); + + return ret; +} + +void mop500_ab8500_remove(struct snd_soc_card *card) +{ + struct mop500_ab8500_drvdata *drvdata = snd_soc_card_get_drvdata(card); + + if (drvdata->clk_ptr_sysclk != NULL) + clk_put(drvdata->clk_ptr_sysclk); + if (drvdata->clk_ptr_ulpclk != NULL) + clk_put(drvdata->clk_ptr_ulpclk); + if (drvdata->clk_ptr_intclk != NULL) + clk_put(drvdata->clk_ptr_intclk); + + snd_soc_card_set_drvdata(card, drvdata); +} diff --git a/sound/soc/ux500/mop500_ab8500.h b/sound/soc/ux500/mop500_ab8500.h new file mode 100644 index 0000000..cca5b33 --- /dev/null +++ b/sound/soc/ux500/mop500_ab8500.h @@ -0,0 +1,22 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja(a)stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * 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 MOP500_AB8500_H +#define MOP500_AB8500_H + +extern struct snd_soc_ops mop500_ab8500_ops[]; + +int mop500_ab8500_machine_init(struct snd_soc_pcm_runtime *runtime); +void mop500_ab8500_remove(struct snd_soc_card *card); + +#endif -- 1.7.8.3

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[alsa-devel] [PATCH] USB: fix gathering of interface associations
by Daniel Mack 17 Jun '12

17 Jun '12

TEAC's UD-H01 (and probably other devices) have a gap in the interface number allocation of their descriptors: Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 220 bNumInterfaces 3 [...] Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 [...] Interface Association: bLength 8 bDescriptorType 11 bFirstInterface 2 bInterfaceCount 2 bFunctionClass 1 Audio bFunctionSubClass 0 bFunctionProtocol 32 iFunction 4 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 [...] Once a configuration is selected, usb_set_configuration() walks the known interfaces of a given configuration and calls find_iad() on each of them to set the interface association pointer the interface is included in. The problem here is that the loop variable is taken for the interface number in the comparison logic that gathers the association. Which is fine as long as the descriptors are sane. In the case above, however, the logic gets out of sync and the interface association fields of all interfaces beyond the interface number gap are wrong. Fix this by passing the interface's bInterfaceNumber to find_iad() instead. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Reported-by: bEN <ml_all(a)circa.be> Reported-by: Ivan Perrone <ivanperrone(a)hotmail.com> --- This patch has been tested on a virtual USB device emulated in QEMU. I'm still waiting for test results based on real hardware, but the bug is rather obvious is and the fix really simple. Daniel drivers/usb/core/message.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/drivers/usb/core/message.c b/drivers/usb/core/message.c index b548cf1..bdd1c67 100644 --- a/drivers/usb/core/message.c +++ b/drivers/usb/core/message.c @@ -1838,7 +1838,6 @@ free_interfaces: intfc = cp->intf_cache[i]; intf->altsetting = intfc->altsetting; intf->num_altsetting = intfc->num_altsetting; - intf->intf_assoc = find_iad(dev, cp, i); kref_get(&intfc->ref); alt = usb_altnum_to_altsetting(intf, 0); @@ -1851,6 +1850,8 @@ free_interfaces: if (!alt) alt = &intf->altsetting[0]; + intf->intf_assoc = + find_iad(dev, cp, alt->desc.bInterfaceNumber); intf->cur_altsetting = alt; usb_enable_interface(dev, intf, true); intf->dev.parent = &dev->dev; -- 1.7.10.2

2 3

[alsa-devel] [PATCH 0/2] RFC: support for audio wall clock
by Pierre-Louis Bossart 15 Jun '12

15 Jun '12

Attached a set of patches to show how the audio wall clock (eg. the HDAudio one) can be used to provide precise audio timestamps and track the drift between audio time and system time. On my laptop I am able to track a 7ppm delta. I will explain in more details how this can be used during LPC in San Diego, but with the summer coming I'd like to get feedback as there are changes to the core/alsa-lib that haven't been modified in years. Pierre-Louis Bossart (2): ALSA: core: add hooks for audio timestamps read from WALLCLOCK ALSA: hda: support for wallclock timestamps include/sound/asound.h | 7 +++- include/sound/pcm.h | 2 + sound/core/pcm_lib.c | 15 ++++++- sound/core/pcm_native.c | 2 + sound/pci/hda/hda_intel.c | 106 ++++++++++++++++++++++++++++++++++++++++++++- 5 files changed, 128 insertions(+), 4 deletions(-) -- 1.7.6.5

4 11

[alsa-devel] [PATCH] pcm: support for audio wallclock
by Pierre-Louis Bossart 15 Jun '12

15 Jun '12

add new snd_pcm_status_get_htstamp_wallclk() routine to query the audio timestamps read from the wallclock. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- include/pcm.h | 1 + include/sound/asound.h | 7 ++++++- src/pcm/pcm.c | 16 ++++++++++++++++ 3 files changed, 23 insertions(+), 1 deletions(-) diff --git a/include/pcm.h b/include/pcm.h index 4997557..325c0ea 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -858,6 +858,7 @@ void snd_pcm_status_get_trigger_tstamp(const snd_pcm_status_t *obj, snd_timestam void snd_pcm_status_get_trigger_htstamp(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr); void snd_pcm_status_get_tstamp(const snd_pcm_status_t *obj, snd_timestamp_t *ptr); void snd_pcm_status_get_htstamp(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr); +void snd_pcm_status_get_htstamp_wallclk(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr); snd_pcm_sframes_t snd_pcm_status_get_delay(const snd_pcm_status_t *obj); snd_pcm_uframes_t snd_pcm_status_get_avail(const snd_pcm_status_t *obj); snd_pcm_uframes_t snd_pcm_status_get_avail_max(const snd_pcm_status_t *obj); diff --git a/include/sound/asound.h b/include/sound/asound.h index 07c03fa..c186fd6 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -285,6 +285,7 @@ enum sndrv_pcm_subformat { #define SNDRV_PCM_INFO_JOINT_DUPLEX 0x00200000 /* playback and capture stream are somewhat correlated */ #define SNDRV_PCM_INFO_SYNC_START 0x00400000 /* pcm support some kind of sync go */ #define SNDRV_PCM_INFO_NO_PERIOD_WAKEUP 0x00800000 /* period wakeup can be disabled */ +#define SNDRV_PCM_INFO_HAS_WALL_CLOCK 0x01000000 /* has audio wall clock for audio/system time sync */ enum sndrv_pcm_state { SNDRV_PCM_STATE_OPEN = 0, /* stream is open */ @@ -426,7 +427,10 @@ struct sndrv_pcm_status { sndrv_pcm_uframes_t avail_max; /* max frames available on hw since last status */ sndrv_pcm_uframes_t overrange; /* count of ADC (capture) overrange detections from last status */ int suspended_state; /* suspended stream state */ - unsigned char reserved[60]; /* must be filled with zero */ + union { + unsigned char reserved[60]; /* must be filled with zero */ + struct timespec audio_tstamp; /* audio wall clock timestamp */ + } ext; }; struct sndrv_pcm_mmap_status { @@ -434,6 +438,7 @@ struct sndrv_pcm_mmap_status { int pad1; /* Needed for 64 bit alignment */ sndrv_pcm_uframes_t hw_ptr; /* RO: hw ptr (0...boundary-1) */ struct timespec tstamp; /* Timestamp */ + struct timespec audio_tstamp; /* audio wall clock timestamp */ int suspended_state; /* RO: suspended stream state */ }; diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 18b43b3..2221865 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -6213,6 +6213,22 @@ void snd_pcm_status_get_htstamp(const snd_pcm_status_t *obj, snd_htimestamp_t *p use_default_symbol_version(__snd_pcm_status_get_htstamp, snd_pcm_status_get_htstamp, ALSA_0.9.0rc8); /** + * \brief Get "now" hi-res audio wallclock timestamp from a PCM status container + * \param obj pointer to #snd_pcm_status_t + * \param ptr Pointer to returned timestamp + */ +#ifndef DOXYGEN +void INTERNAL(snd_pcm_status_get_htstamp_wallclk)(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr) +#else +void snd_pcm_status_get_htstamp_wallclk(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr) +#endif +{ + assert(obj && ptr); + *ptr = obj->ext.audio_tstamp; +} +use_default_symbol_version(__snd_pcm_status_get_htstamp_wallclk, snd_pcm_status_get_htstamp_wallclk, ALSA_0.9.0rc8); /* FIXME, no idea why this is needed */ + +/** * \brief Get delay from a PCM status container (see #snd_pcm_delay) * \return Delay in frames * -- 1.7.6.5

2 2

[alsa-devel] [PATCH] ASoC: omap-mcpdm: Convert to use devm_*
by Peter Ujfalusi 15 Jun '12

15 Jun '12

Switch to use devm_* te make the probe/remove code more cleaner. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/omap-mcpdm.c | 52 ++++++++++-------------------------------- 1 files changed, 13 insertions(+), 39 deletions(-) diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 59d47ab..b5e23af 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -445,9 +445,8 @@ static __devinit int asoc_mcpdm_probe(struct platform_device *pdev) { struct omap_mcpdm *mcpdm; struct resource *res; - int ret = 0; - mcpdm = kzalloc(sizeof(struct omap_mcpdm), GFP_KERNEL); + mcpdm = devm_kzalloc(&pdev->dev, sizeof(struct omap_mcpdm), GFP_KERNEL); if (!mcpdm) return -ENOMEM; @@ -456,55 +455,30 @@ static __devinit int asoc_mcpdm_probe(struct platform_device *pdev) mutex_init(&mcpdm->mutex); res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (res == NULL) { - dev_err(&pdev->dev, "no resource\n"); - goto err_res; - } + if (res == NULL) + return -ENOMEM; - if (!request_mem_region(res->start, resource_size(res), "McPDM")) { - ret = -EBUSY; - goto err_res; - } + if (!devm_request_mem_region(&pdev->dev, res->start, + resource_size(res), "McPDM")) + return -EBUSY; - mcpdm->io_base = ioremap(res->start, resource_size(res)); - if (!mcpdm->io_base) { - ret = -ENOMEM; - goto err_iomap; - } + mcpdm->io_base = devm_ioremap(&pdev->dev, res->start, + resource_size(res)); + if (!mcpdm->io_base) + return -ENOMEM; mcpdm->irq = platform_get_irq(pdev, 0); - if (mcpdm->irq < 0) { - ret = mcpdm->irq; - goto err_irq; - } + if (mcpdm->irq < 0) + return mcpdm->irq; mcpdm->dev = &pdev->dev; - ret = snd_soc_register_dai(&pdev->dev, &omap_mcpdm_dai); - if (!ret) - return 0; - -err_irq: - iounmap(mcpdm->io_base); -err_iomap: - release_mem_region(res->start, resource_size(res)); -err_res: - kfree(mcpdm); - return ret; + return snd_soc_register_dai(&pdev->dev, &omap_mcpdm_dai); } static int __devexit asoc_mcpdm_remove(struct platform_device *pdev) { - struct omap_mcpdm *mcpdm = platform_get_drvdata(pdev); - struct resource *res; - snd_soc_unregister_dai(&pdev->dev); - - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - iounmap(mcpdm->io_base); - release_mem_region(res->start, resource_size(res)); - - kfree(mcpdm); return 0; } -- 1.7.8.6

2 1

Re: [alsa-devel] [PATCH 1/1] hda_intel: activate COMBO mode for Intel client chipsets
by Takashi Iwai 15 Jun '12

15 Jun '12

At Thu, 14 Jun 2012 14:23:53 -0700, Seth Heasley wrote: > > This patch activates the COMBO position_fix for recent Intel client chipsets. COMBO mode is the recommended setting for Intel chipsets and eliminates HD audio warnings in dmesg. This patch has been tested on Lynx Point, Panther Point, and Cougar Pont. > > Signed-off-by: Seth Heasley <seth.heasley(a)intel.com> Applied now. Thanks. Takashi > --- > sound/pci/hda/hda_intel.c | 11 ++++++++--- > 1 files changed, 8 insertions(+), 3 deletions(-) > > diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > index 2b6392b..466a88c 100644 > --- a/sound/pci/hda/hda_intel.c > +++ b/sound/pci/hda/hda_intel.c > @@ -535,6 +535,7 @@ enum { > #define AZX_DCAPS_BUFSIZE (1 << 21) /* no buffer size alignment */ > #define AZX_DCAPS_ALIGN_BUFSIZE (1 << 22) /* buffer size alignment */ > #define AZX_DCAPS_4K_BDLE_BOUNDARY (1 << 23) /* BDLE in 4k boundary */ > +#define AZX_DCAPS_POSFIX_COMBO (1 << 24) /* Use COMBO as default */ > > /* quirks for ATI SB / AMD Hudson */ > #define AZX_DCAPS_PRESET_ATI_SB \ > @@ -2728,6 +2729,10 @@ static int __devinit check_position_fix(struct azx *chip, int fix) > snd_printd(SFX "Using LPIB position fix\n"); > return POS_FIX_LPIB; > } > + if (chip->driver_caps & AZX_DCAPS_POSFIX_COMBO) { > + snd_printd(SFX "Using COMBO position fix\n"); > + return POS_FIX_COMBO; > + } > return POS_FIX_AUTO; > } > > @@ -3240,7 +3245,7 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { > /* CPT */ > { PCI_DEVICE(0x8086, 0x1c20), > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | > - AZX_DCAPS_BUFSIZE }, > + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, > /* PBG */ > { PCI_DEVICE(0x8086, 0x1d20), > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | > @@ -3248,11 +3253,11 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { > /* Panther Point */ > { PCI_DEVICE(0x8086, 0x1e20), > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | > - AZX_DCAPS_BUFSIZE}, > + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, > /* Lynx Point */ > { PCI_DEVICE(0x8086, 0x8c20), > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | > - AZX_DCAPS_BUFSIZE}, > + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, > /* SCH */ > { PCI_DEVICE(0x8086, 0x811b), > .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | > -- > 1.7.4.4 >

1 0

[alsa-devel] McBSP in TDM for audio codec
by learnom learnom 15 Jun '12

15 Jun '12

Hi All, I have to support 4 channels in I2S format in TDM mode. Processor: OMAP 3703 If any one have any idea on how to go about this. Please let me know. Thanks, learnom

2 3

[alsa-devel] Anyone using alsactl preinit / postinit scripts?
by Takashi Iwai 14 Jun '12

14 Jun '12

Hi, is anyone already using preinit / postinit scripts of alsactl init? I'm thinking of appliny the patch below, but before that, I'd like to make sure that *.conf is the appropriate extension. thanks, Takashi --- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] alsactl: Read only *.conf files when a directory is passed via INCLUDE When alsactl init is invoked and a directory path is passed to INCLUDE command in the config file, read only *.conf files in that directory. This will avoid reading backup files or invalid files that have been created accidentally. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- alsactl/init_parse.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/alsactl/init_parse.c b/alsactl/init_parse.c index 51b515c..83d12ba 100644 --- a/alsactl/init_parse.c +++ b/alsactl/init_parse.c @@ -1502,12 +1502,16 @@ static int parse_line(struct space *space, char *line, size_t linesize) if (dir) { count = strlen(string); while ((dirent = readdir(dir)) != NULL) { - if (strcmp(dirent->d_name, ".") == 0 || - strcmp(dirent->d_name, "..") == 0) + const char *name = dirent->d_name; + const char *ext; + if (*name == '.') + continue; + ext = strrchr(name, ','); + if (!ext || strcmp(ext, ".conf")) continue; string[count] = '\0'; strlcat(string, "/", sizeof(string)); - strlcat(string, dirent->d_name, sizeof(string)); + strlcat(string, name, sizeof(string)); space->go_to = NULL; space->rootdir = new_root_dir(string); if (space->rootdir) { -- 1.7.10.3

2 2

[alsa-devel] [PATCH] ALSA: HDA: Create phantom jacks for fixed inputs and outputs
by David Henningsson 14 Jun '12

14 Jun '12

PulseAudio sometimes have difficulties knowing that there is a "Speaker" or "Internal Mic", if they have no individual volume controls or selectors. As a result, only e g "Headphone" might be created for a laptop, but no "Speaker". To help out, create phantom jacks (that are always present, at least for now) for "Speaker", "Internal Mic" etc, in case we detect them. The naming convention is e g "Speaker Phantom Jack". In order not to pollute the /dev/input namespace with even more devices, these are added to the kcontrols only, not the input devices. An implementation note: The possibility to override how to actually do jack sensing opens up for more advanced usage of the jack infrastructure in the future (e g letting the headphone jack detection set presence status for both headphone and mic in case of a headset jack, among other things). Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_jack.c | 64 ++++++++++++++++++++++++++++++++++------------ sound/pci/hda/hda_jack.h | 5 ++++ 2 files changed, 52 insertions(+), 17 deletions(-) diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index 2dd1c11..c5569f1 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -133,6 +133,13 @@ static void jack_detect_update(struct hda_codec *codec, } } +static void jack_detect_always_present(struct hda_codec *codec, + struct hda_jack_tbl *jack) +{ + jack->pin_sense = AC_PINSENSE_PRESENCE; + jack->jack_dirty = 0; +} + /** * snd_hda_set_dirty_all - Mark all the cached as dirty * @@ -162,7 +169,8 @@ u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid) { struct hda_jack_tbl *jack = snd_hda_jack_tbl_get(codec, nid); if (jack) { - jack_detect_update(codec, jack); + if (jack->update_func) + jack->update_func(codec, jack); return jack->pin_sense; } return read_pin_sense(codec, nid); @@ -215,7 +223,8 @@ void snd_hda_jack_report_sync(struct hda_codec *codec) for (i = 0; i < codec->jacktbl.used; i++, jack++) if (jack->nid) { - jack_detect_update(codec, jack); + if (jack->update_func) + jack->update_func(codec, jack); if (!jack->kctl) continue; state = get_jack_plug_state(jack->pin_sense); @@ -264,8 +273,8 @@ static void hda_free_jack_priv(struct snd_jack *jack) * This assigns a jack-detection kctl to the given pin. The kcontrol * will have the given name and index. */ -int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, - const char *name, int idx) +static int __snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, + const char *name, int idx, bool phantom_jack) { struct hda_jack_tbl *jack; struct snd_kcontrol *kctl; @@ -283,19 +292,33 @@ int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, if (err < 0) return err; jack->kctl = kctl; + if (phantom_jack) + jack->update_func = jack_detect_always_present; + else + jack->update_func = jack_detect_update; + state = snd_hda_jack_detect(codec, nid); snd_kctl_jack_report(codec->bus->card, kctl, state); #ifdef CONFIG_SND_HDA_INPUT_JACK - jack->type = get_input_jack_type(codec, nid); - err = snd_jack_new(codec->bus->card, name, jack->type, &jack->jack); - if (err < 0) - return err; - jack->jack->private_data = jack; - jack->jack->private_free = hda_free_jack_priv; - snd_jack_report(jack->jack, state ? jack->type : 0); + if (!phantom_jack) { + jack->type = get_input_jack_type(codec, nid); + err = snd_jack_new(codec->bus->card, name, jack->type, + &jack->jack); + if (err < 0) + return err; + jack->jack->private_data = jack; + jack->jack->private_free = hda_free_jack_priv; + snd_jack_report(jack->jack, state ? jack->type : 0); + } #endif return 0; } + +int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, + const char *name, int idx) +{ + return __snd_hda_jack_add_kctl(codec, nid, name, idx, false); +} EXPORT_SYMBOL_HDA(snd_hda_jack_add_kctl); static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, @@ -305,25 +328,32 @@ static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, unsigned int def_conf, conn; char name[44]; int idx, err; + bool phantom_jack; if (!nid) return 0; - if (!is_jack_detectable(codec, nid)) - return 0; def_conf = snd_hda_codec_get_pincfg(codec, nid); conn = get_defcfg_connect(def_conf); - if (conn != AC_JACK_PORT_COMPLEX) + if (conn == AC_JACK_PORT_NONE) return 0; + phantom_jack = (conn != AC_JACK_PORT_COMPLEX) || + !is_jack_detectable(codec, nid); snd_hda_get_pin_label(codec, nid, cfg, name, sizeof(name), &idx); + if (phantom_jack) + /* Example final name: "Internal Mic Phantom Jack" */ + strncat(name, " Phantom", sizeof(name) - strlen(name) - 1); if (!strcmp(name, lastname) && idx == *lastidx) idx++; - strncpy(lastname, name, 44); + strncpy(lastname, name, sizeof(name)); *lastidx = idx; - err = snd_hda_jack_add_kctl(codec, nid, name, idx); + err = __snd_hda_jack_add_kctl(codec, nid, name, idx, phantom_jack); if (err < 0) return err; - return snd_hda_jack_detect_enable(codec, nid, 0); + + if (!phantom_jack) + return snd_hda_jack_detect_enable(codec, nid, 0); + return 0; } /** diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h index 8ae5246..4055236 100644 --- a/sound/pci/hda/hda_jack.h +++ b/sound/pci/hda/hda_jack.h @@ -13,6 +13,10 @@ #define __SOUND_HDA_JACK_H struct auto_pin_cfg; +struct hda_jack_tbl; + +typedef void (*snd_hda_jack_detect_update)(struct hda_codec *codec, + struct hda_jack_tbl *jack); struct hda_jack_tbl { hda_nid_t nid; @@ -24,6 +28,7 @@ struct hda_jack_tbl { unsigned int jack_detect:1; /* capable of jack-detection? */ unsigned int jack_dirty:1; /* needs to update? */ struct snd_kcontrol *kctl; /* assigned kctl for jack-detection */ + snd_hda_jack_detect_update update_func; /* function to call to perform jack detection */ #ifdef CONFIG_SND_HDA_INPUT_JACK int type; struct snd_jack *jack; -- 1.7.9.5

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Re: [alsa-devel] [PATCH 1/1] hda_intel: switch Intel chipsets to COMBO position_fix
by Takashi Iwai 14 Jun '12

14 Jun '12

At Wed, 13 Jun 2012 17:58:57 +0000, Heasley, Seth wrote: > > >> This patch implements the COMBO position_fix for recent Intel client > >chipsets. > >> > >> Signed-off-by: Seth Heasley <seth.heasley(a)intel.com> > > > >Thanks for the patch. > >But could you give a bit more context why this change is needed? > > The COMBO method was recommended for Intel chipsets by a certain audio maintainer. ;) It was also tested on the most recent Intel chips because we were seeing warning messages when using DMA positioning ("Invalid position buffer, using LPIB read method instead."). Could you resend the patch with such some reasonable comments in the changelog? thanks, Takashi > >> --- > >> sound/pci/hda/hda_intel.c | 11 ++++++++--- > >> 1 files changed, 8 insertions(+), 3 deletions(-) > >> > >> diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > >> index 2b6392b..466a88c 100644 > >> --- a/sound/pci/hda/hda_intel.c > >> +++ b/sound/pci/hda/hda_intel.c > >> @@ -535,6 +535,7 @@ enum { > >> #define AZX_DCAPS_BUFSIZE (1 << 21) /* no buffer size > >alignment */ > >> #define AZX_DCAPS_ALIGN_BUFSIZE (1 << 22) /* buffer size > >alignment */ > >> #define AZX_DCAPS_4K_BDLE_BOUNDARY (1 << 23) /* BDLE in 4k boundary > >*/ > >> +#define AZX_DCAPS_POSFIX_COMBO (1 << 24) /* Use COMBO as > >default */ > >> > >> /* quirks for ATI SB / AMD Hudson */ > >> #define AZX_DCAPS_PRESET_ATI_SB \ > >> @@ -2728,6 +2729,10 @@ static int __devinit check_position_fix(struct azx > >*chip, int fix) > >> snd_printd(SFX "Using LPIB position fix\n"); > >> return POS_FIX_LPIB; > >> } > >> + if (chip->driver_caps & AZX_DCAPS_POSFIX_COMBO) { > >> + snd_printd(SFX "Using COMBO position fix\n"); > >> + return POS_FIX_COMBO; > >> + } > >> return POS_FIX_AUTO; > >> } > >> > >> @@ -3240,7 +3245,7 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { > >> /* CPT */ > >> { PCI_DEVICE(0x8086, 0x1c20), > >> .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | > >> - AZX_DCAPS_BUFSIZE }, > >> + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, > >> /* PBG */ > >> { PCI_DEVICE(0x8086, 0x1d20), > >> .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | @@ - > >3248,11 > >> +3253,11 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { > >> /* Panther Point */ > >> { PCI_DEVICE(0x8086, 0x1e20), > >> .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | > >> - AZX_DCAPS_BUFSIZE}, > >> + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, > >> /* Lynx Point */ > >> { PCI_DEVICE(0x8086, 0x8c20), > >> .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | > >> - AZX_DCAPS_BUFSIZE}, > >> + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, > >> /* SCH */ > >> { PCI_DEVICE(0x8086, 0x811b), > >> .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | > >> -- > >> 1.7.4.4 > >> >

1 0

[alsa-devel] Does amixer have commands to turn on / off Codec TWL4030 analog microphone bias voltage output?
by Philip Chu 14 Jun '12

14 Jun '12

I am using Linux kernel 3.0 and a Logic PD DM3530 SOM LV, looking for the amixer commands to turn on / off analog microphone bias supply. Following lines do not have the bias needed for Headset MIC input: amixer set 'Analog Left AUXL' nocap amixer set 'Analog Right AUXR' nocap amixer set 'Analog Left Main Mic' nocap amixer set 'Analog Left Headset Mic' cap Anybody can help? Thanks, Philip Chu principal software engineer LogicPD www.logicpd.com 5 Clock Tower Place, Suite 215 Maynard, MA 01754 Phone: 978-243-2450 Fax: 978-461-1598

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Re: [alsa-devel] [PATCH v2 0/5] mmp audio support
by Arnd Bergmann 14 Jun '12

14 Jun '12

On Monday 11 June 2012, Zhangfei Gao wrote: > > v1->v2: > modify accroingly with suggestion from Vinod and Mark > remove deprecated chan->private and dma.h > remove prep_slave_sg > remove module_init and move operation to probe > misc change It seems that this driver is missing DT bindings, which really need to get added. The brownstone platform is currently supported using both DT and ATAG probing, but we want to remove the ATAG variant eventually, so adding a new driver for that platform without support for DT booting is a step in the wrong direction. Arnd -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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Re: [alsa-devel] Questions regarding adding regmap support for TLV320AIC3262
by Mark Brown 13 Jun '12

13 Jun '12

On Tue, Jun 12, 2012 at 08:45:51AM +0000, Navada Kanyana, Mukund wrote: Fix your mailer to word wrap within paragraphs. See email-clients.txt in the kernel. > In order to support such a regmap, I would like to push the > following patch for regmap.c to add support for 24-bit register > addressing. This looks fine, please submit as covered in SubmittingPatches in the kernel source. You might also want to have a look in the list archives for the patches that some of the guys at Dialog have been submitting for devices with paged register maps, it sounds like these might be useful for this device. Please always CC the appropriate mailing list for the subsytem, in this case linux-kernel. There's nothing ASoC specific about this.

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[alsa-devel] [PATCH 1/5] ASoC: wm8903: Move pin configuration into I2C probe() function
by Mark Brown 13 Jun '12

13 Jun '12

Ensure that the device pins are configured as soon as possible by moving the pin configration (including MICBIAS) into the I2C probe() function. This had been done in the CODEC probe() function when we were relying on the ASoC register I/O code. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- This series is untested but should deal with the WM8903 side of what I was saying earlier. sound/soc/codecs/wm8903.c | 92 +++++++++++++++++++++++---------------------- 1 file changed, 47 insertions(+), 45 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index e76a66d..7200088 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1,7 +1,7 @@ /* * wm8903.c -- WM8903 ALSA SoC Audio driver * - * Copyright 2008-11 Wolfson Microelectronics + * Copyright 2008-12 Wolfson Microelectronics * Copyright 2011-2012 NVIDIA, Inc. * * Author: Mark Brown <broonie(a)opensource.wolfsonmicro.com> @@ -2081,10 +2081,9 @@ static int wm8903_probe(struct snd_soc_codec *codec) { struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); struct wm8903_platform_data *pdata = wm8903->pdata; - int ret, i; + int ret; int trigger, irq_pol; u16 val; - bool mic_gpio = false; wm8903->codec = codec; codec->control_data = wm8903->regmap; @@ -2095,47 +2094,6 @@ static int wm8903_probe(struct snd_soc_codec *codec) return ret; } - /* Set up GPIOs, detect if any are MIC detect outputs */ - for (i = 0; i < ARRAY_SIZE(pdata->gpio_cfg); i++) { - if ((!pdata->gpio_cfg[i]) || - (pdata->gpio_cfg[i] > WM8903_GPIO_CONFIG_ZERO)) - continue; - - snd_soc_write(codec, WM8903_GPIO_CONTROL_1 + i, - pdata->gpio_cfg[i] & 0x7fff); - - val = (pdata->gpio_cfg[i] & WM8903_GP1_FN_MASK) - >> WM8903_GP1_FN_SHIFT; - - switch (val) { - case WM8903_GPn_FN_MICBIAS_CURRENT_DETECT: - case WM8903_GPn_FN_MICBIAS_SHORT_DETECT: - mic_gpio = true; - break; - default: - break; - } - } - - /* Set up microphone detection */ - snd_soc_write(codec, WM8903_MIC_BIAS_CONTROL_0, - pdata->micdet_cfg); - - /* Microphone detection needs the WSEQ clock */ - if (pdata->micdet_cfg) - snd_soc_update_bits(codec, WM8903_WRITE_SEQUENCER_0, - WM8903_WSEQ_ENA, WM8903_WSEQ_ENA); - - /* If microphone detection is enabled by pdata but - * detected via IRQ then interrupts can be lost before - * the machine driver has set up microphone detection - * IRQs as the IRQs are clear on read. The detection - * will be enabled when the machine driver configures. - */ - WARN_ON(!mic_gpio && (pdata->micdet_cfg & WM8903_MICDET_ENA)); - - wm8903->mic_delay = pdata->micdet_delay; - if (wm8903->irq) { if (pdata->irq_active_low) { trigger = IRQF_TRIGGER_LOW; @@ -2316,8 +2274,9 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, { struct wm8903_platform_data *pdata = dev_get_platdata(&i2c->dev); struct wm8903_priv *wm8903; + bool mic_gpio = false; unsigned int val; - int ret; + int ret, i; wm8903 = devm_kzalloc(&i2c->dev, sizeof(struct wm8903_priv), GFP_KERNEL); @@ -2361,6 +2320,8 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, } } + pdata = wm8903->pdata; + ret = regmap_read(wm8903->regmap, WM8903_SW_RESET_AND_ID, &val); if (ret != 0) { dev_err(&i2c->dev, "Failed to read chip ID: %d\n", ret); @@ -2385,6 +2346,47 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, wm8903_init_gpio(wm8903); + /* Set up GPIO pin state, detect if any are MIC detect outputs */ + for (i = 0; i < ARRAY_SIZE(pdata->gpio_cfg); i++) { + if ((!pdata->gpio_cfg[i]) || + (pdata->gpio_cfg[i] > WM8903_GPIO_CONFIG_ZERO)) + continue; + + regmap_write(wm8903->regmap, WM8903_GPIO_CONTROL_1 + i, + pdata->gpio_cfg[i] & 0x7fff); + + val = (pdata->gpio_cfg[i] & WM8903_GP1_FN_MASK) + >> WM8903_GP1_FN_SHIFT; + + switch (val) { + case WM8903_GPn_FN_MICBIAS_CURRENT_DETECT: + case WM8903_GPn_FN_MICBIAS_SHORT_DETECT: + mic_gpio = true; + break; + default: + break; + } + } + + /* Set up microphone detection */ + regmap_write(wm8903->regmap, WM8903_MIC_BIAS_CONTROL_0, + pdata->micdet_cfg); + + /* Microphone detection needs the WSEQ clock */ + if (pdata->micdet_cfg) + regmap_update_bits(wm8903->regmap, WM8903_WRITE_SEQUENCER_0, + WM8903_WSEQ_ENA, WM8903_WSEQ_ENA); + + /* If microphone detection is enabled by pdata but + * detected via IRQ then interrupts can be lost before + * the machine driver has set up microphone detection + * IRQs as the IRQs are clear on read. The detection + * will be enabled when the machine driver configures. + */ + WARN_ON(!mic_gpio && (pdata->micdet_cfg & WM8903_MICDET_ENA)); + + wm8903->mic_delay = pdata->micdet_delay; + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8903, &wm8903_dai, 1); if (ret != 0) -- 1.7.10

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Re: [alsa-devel] [PATCH v2 0/5] mmp audio support
by Mark Brown 13 Jun '12

13 Jun '12

On Mon, Jun 11, 2012 at 06:04:36PM +0800, Zhangfei Gao wrote: > These patches provide mmp audio support under alsa via dmaengine > Support platfrom pxa688 and pxa910 > mmp_tdma.c and mmp-pcm.c is shared for pxa688 and pxa910 > pxa688 (mmp2) use mmp2-sspa.c > pxa910 directly use pxa-ssp.c > mmp_tdma.c is under dmaengine framework Applied 2-4. I'm not seeing a direct code dependency between this and the dmaengine driver in patch 1 (or review of it from Vinod yet) so I'm expecting that to go via the dmaengine tree.

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[alsa-devel] Possible message repeat
by Brent Weatherall 13 Jun '12

13 Jun '12

Hello. I sent this message previously before I solidified my membership to the list and it seems my message never posted. Here it is again. My apologies if did go through and I missed it. I am new to audio development in Linux. I have worked with user space development with respect to UVC code and I have worked with specifying an XU in UVC. I wrote a user space driver/application to talk to a device supporting the UVC XU in Linux. I am trying to specify an extension to the UAC specification and then write a user space driver/application to use the extension when a device that supports it is available in Linux. Pretty much the same situation as I have already done with UVC XU. Windows and Mac code will follow, but right now I am only concerned with Linux. Can anyone assist me with the best path forward for writing user space code to use the XU I am going to define? Meaning generically speaking how to use UAC XU controls in ALSA. I was looking for code samples that use the audio.h header file interface, but after a day and a half I am coming to conclusion that only kernel space uses audio.h and that I will need to go through ALSA to accomplish my task. Again, this will be user space code I want distribute easily, with minimal dependencies. Our code will be apache licensed afterwards if possible, just like our UVC code was. So any work I do should be freely available to the community afterwards. **EDIT FROM PREVIOUS MESSAGE** After more searching, it appears that any XU controls would be processed through usbmixer.c, but I don't see how to use the code in usbmixer from user space. And so far I can't tell what source usbmixer resides in. I downloaded the alsa-lib source in Fedora, but I don't see that file. Is this a Linux kernel file? So far I am having lots of trouble finding anything useful regarding audio XUs in Linux. My apologies if this should be simple. Thank you in advance for any assistance. Best Regards, Brent

2 1

[alsa-devel] [PATCH 1/4] ASoC: when initializing CPU DAI, don't duplicate any CODEC init
by Stephen Warren 13 Jun '12

13 Jun '12

From: Stephen Warren <swarren(a)nvidia.com> If the CPU-side of a DAI link is a CODEC rather than a standalone DAI, the codec initialization will call try_module_get() and create the DAI widgets. Ensure that this isn't duplicated when the CPU DAI itself is probed, if the CPU DAI is part of a CODEC. Note that this is not a complete fix on its own, since there's no guarantee that the CODEC itself will be initialized - currently that only happens if the CODEC is also used as the CODEC-side of a DAI link, and that initialization may happen before or after the DAIs within the CODEC are initialized. However, such a scenario doesn't necessarily currently work, and I don't think this change alone makes it any worse. This is fixed in a couple patches time. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/soc-core.c | 14 +++++++++----- 1 files changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3d803f3..448d4a7 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -983,7 +983,9 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order) } cpu_dai->probed = 0; list_del(&cpu_dai->card_list); - module_put(cpu_dai->dev->driver->owner); + + if (!cpu_dai->codec) + module_put(cpu_dai->dev->driver->owner); } } @@ -1257,11 +1259,13 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order) /* probe the cpu_dai */ if (!cpu_dai->probed && cpu_dai->driver->probe_order == order) { - cpu_dai->dapm.card = card; - if (!try_module_get(cpu_dai->dev->driver->owner)) - return -ENODEV; + if (!cpu_dai->codec) { + cpu_dai->dapm.card = card; + if (!try_module_get(cpu_dai->dev->driver->owner)) + return -ENODEV; - snd_soc_dapm_new_dai_widgets(&cpu_dai->dapm, cpu_dai); + snd_soc_dapm_new_dai_widgets(&cpu_dai->dapm, cpu_dai); + } if (cpu_dai->driver->probe) { ret = cpu_dai->driver->probe(cpu_dai); -- 1.7.0.4

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Re: [alsa-devel] [PATCH v3] sound/soc/lapis: add platform driver for ML7213
by Mark Brown 13 Jun '12

13 Jun '12

On Tue, Mar 06, 2012 at 02:48:05PM +0900, Tomoya MORINAGA wrote: > 2012年3月3日1:39 Mark Brown <broonie(a)opensource.wolfsonmicro.com>: > > I just merged Lars-Peter's dmaengine library code which has been on the > > list for a week or so - this should be updated to use that next time > > it's posted. That should save a lot of code from the driver and make > > sure it's following best practices for dmaengine use. > Sorry, we can't use this library. This sort of response really isn't on. Do you have some reason for saying this? Flat out refusing to do things with no reason is not useful. > >> +static struct ioh_i2s_data *i2s_data; > >> +static struct ioh_i2s_dma dmadata[MAX_I2S_CH]; > > Why are these needed, aren't they dynamically allocated by the driver? > You mean ASoC driver can't use global variable ? In general no Linux driver should be using a global variable except for things like this. > >> + case SNDRV_PCM_FORMAT_S32_LE: > >> + byte = 24; > >> + break; > > That looks wrong... are you sure you don't support S24_LE or something? > This is correct. > Because maximum transmit size of ML7213's I2S hw is 24bit. Note we often lay out 24 bit audio in 32 bit blocks. > >> +static struct platform_driver ioh_i2s_driver_plat = { > > > >> +static struct platform_driver ioh_dai_driver_plat = { > > > >> +static int ioh_i2s_pci_probe(struct pci_dev *pdev, > >> + const struct pci_device_id *id) > > Why are you creating these platform devices? I don't understand the > > function they serve. The code handling them looks to have quite a few > > problems but I'm not clear they should be there in the first place. > if these platform devices aren't used, device detection doesn't work correctly. > So, I added these. You've not actually mentioned the problem you were seeing... > >> + rv = request_irq(pdev->irq, ioh_i2s_irq, IRQF_SHARED, "ml7213_ioh", > >> + pdev); > >> + if (rv != 0) { > >> + printk(KERN_ERR "Failed to allocate irq\n"); > >> + goto out_irq; > >> + } > > Are you *sure* you're ready to handle interrupts at this point? > This is just registering interrupt handler. > As long as interrupt register is not enabled, the interrupt handler is > not called. > This is common request_irq description. > What's is your concern ? Apart from anything else you've got the interrupt requested as IRQF_SHARED so the interrupt could get called at any time. It's also not clear that you've got the hardware in a known good state.

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[alsa-devel] [PATCH] ASoC: ml26124: Convert to devm_regmap_init_i2c
by Axel Lin 13 Jun '12

13 Jun '12

This fixes a leak if snd_soc_register_codec fails. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/ml26124.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c index 22cb5bf..96aa5fa 100644 --- a/sound/soc/codecs/ml26124.c +++ b/sound/soc/codecs/ml26124.c @@ -638,7 +638,7 @@ static __devinit int ml26124_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, priv); - priv->regmap = regmap_init_i2c(i2c, &ml26124_i2c_regmap); + priv->regmap = devm_regmap_init_i2c(i2c, &ml26124_i2c_regmap); if (IS_ERR(priv->regmap)) { ret = PTR_ERR(priv->regmap); dev_err(&i2c->dev, "regmap_init_i2c() failed: %d\n", ret); @@ -651,10 +651,7 @@ static __devinit int ml26124_i2c_probe(struct i2c_client *i2c, static __devexit int ml26124_i2c_remove(struct i2c_client *client) { - struct ml26124_priv *priv = i2c_get_clientdata(client); - snd_soc_unregister_codec(&client->dev); - regmap_exit(priv->regmap); return 0; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: Ux500: Correct license strings
by Ola Lilja 13 Jun '12

13 Jun '12

GPLv2 -> GPL v2 Signed-off-by: Ola Lilja <ola.o.lilja(a)stericsson.com> --- sound/soc/codecs/ab8500-codec.c | 2 +- sound/soc/ux500/ux500_msp_dai.c | 2 +- sound/soc/ux500/ux500_msp_i2s.c | 2 +- sound/soc/ux500/ux500_pcm.c | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index dba246a..dd419c8 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2510,4 +2510,4 @@ static struct platform_driver ab8500_codec_platform_driver = { }; module_platform_driver(ab8500_codec_platform_driver); -MODULE_LICENSE("GPLv2"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index 93c6c40..62ac028 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -840,4 +840,4 @@ static struct platform_driver msp_i2s_driver = { }; module_platform_driver(msp_i2s_driver); -MODULE_LICENSE("GPLv2"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c index 496dec1..ee14d2d 100644 --- a/sound/soc/ux500/ux500_msp_i2s.c +++ b/sound/soc/ux500/ux500_msp_i2s.c @@ -739,4 +739,4 @@ void ux500_msp_i2s_cleanup_msp(struct platform_device *pdev, devm_kfree(&pdev->dev, msp); } -MODULE_LICENSE("GPLv2"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c index 66b080e..97d8e4d 100644 --- a/sound/soc/ux500/ux500_pcm.c +++ b/sound/soc/ux500/ux500_pcm.c @@ -315,4 +315,4 @@ static struct platform_driver ux500_pcm_driver = { }; module_platform_driver(ux500_pcm_driver); -MODULE_LICENSE("GPLv2"); +MODULE_LICENSE("GPL v2"); -- 1.7.8.3

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Re: [alsa-devel] [PATCH 1/1] hda_intel: switch Intel chipsets to COMBO position_fix
by Takashi Iwai 13 Jun '12

13 Jun '12

At Tue, 12 Jun 2012 16:30:00 -0700, Seth Heasley wrote: > > This patch implements the COMBO position_fix for recent Intel client chipsets. > > Signed-off-by: Seth Heasley <seth.heasley(a)intel.com> Thanks for the patch. But could you give a bit more context why this change is needed? Takashi > --- > sound/pci/hda/hda_intel.c | 11 ++++++++--- > 1 files changed, 8 insertions(+), 3 deletions(-) > > diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > index 2b6392b..466a88c 100644 > --- a/sound/pci/hda/hda_intel.c > +++ b/sound/pci/hda/hda_intel.c > @@ -535,6 +535,7 @@ enum { > #define AZX_DCAPS_BUFSIZE (1 << 21) /* no buffer size alignment */ > #define AZX_DCAPS_ALIGN_BUFSIZE (1 << 22) /* buffer size alignment */ > #define AZX_DCAPS_4K_BDLE_BOUNDARY (1 << 23) /* BDLE in 4k boundary */ > +#define AZX_DCAPS_POSFIX_COMBO (1 << 24) /* Use COMBO as default */ > > /* quirks for ATI SB / AMD Hudson */ > #define AZX_DCAPS_PRESET_ATI_SB \ > @@ -2728,6 +2729,10 @@ static int __devinit check_position_fix(struct azx *chip, int fix) > snd_printd(SFX "Using LPIB position fix\n"); > return POS_FIX_LPIB; > } > + if (chip->driver_caps & AZX_DCAPS_POSFIX_COMBO) { > + snd_printd(SFX "Using COMBO position fix\n"); > + return POS_FIX_COMBO; > + } > return POS_FIX_AUTO; > } > > @@ -3240,7 +3245,7 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { > /* CPT */ > { PCI_DEVICE(0x8086, 0x1c20), > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | > - AZX_DCAPS_BUFSIZE }, > + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, > /* PBG */ > { PCI_DEVICE(0x8086, 0x1d20), > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | > @@ -3248,11 +3253,11 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { > /* Panther Point */ > { PCI_DEVICE(0x8086, 0x1e20), > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | > - AZX_DCAPS_BUFSIZE}, > + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, > /* Lynx Point */ > { PCI_DEVICE(0x8086, 0x8c20), > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | > - AZX_DCAPS_BUFSIZE}, > + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, > /* SCH */ > { PCI_DEVICE(0x8086, 0x811b), > .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | > -- > 1.7.4.4 >

1 0

[alsa-devel] [PATCH] ALSA: HDA: Pin fixup for Zotac Z68 motherboard
by David Henningsson 12 Jun '12

12 Jun '12

Pin 0x1b was connected to the front panel connector, which according to the HDA standard should contain a mic and a headphone. In this case, the headphone was listed as "line out" by BIOS. Cc: stable(a)kernel.org BugLink: https://bugs.launchpad.net/bugs/993162 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 224410e..b896681 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6439,6 +6439,7 @@ enum { ALC662_FIXUP_ASUS_MODE7, ALC662_FIXUP_ASUS_MODE8, ALC662_FIXUP_NO_JACK_DETECT, + ALC662_FIXUP_ZOTAC_Z68, }; static const struct alc_fixup alc662_fixups[] = { @@ -6588,6 +6589,13 @@ static const struct alc_fixup alc662_fixups[] = { .type = ALC_FIXUP_FUNC, .v.func = alc_fixup_no_jack_detect, }, + [ALC662_FIXUP_ZOTAC_Z68] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x1b, 0x02214020 }, /* Front HP */ + { } + } + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -6601,6 +6609,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD), + SND_PCI_QUIRK(0x19da, 0xa130, "Zotac Z68", ALC662_FIXUP_ZOTAC_Z68), SND_PCI_QUIRK(0x1b35, 0x2206, "CZC P10T", ALC662_FIXUP_CZC_P10T), #if 0 -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/2] alsa: compress_core: don't wake up on pause
by Vinod Koul 12 Jun '12

12 Jun '12

during pause the core should maintain the status-quo on the device and pointers and not wake up. If app needs it should call DROP explcitly. Signed-off-by: Namarta Kohli <namartax.kohli(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)linux.intel.com> --- sound/core/compress_offload.c | 4 +--- 1 files changed, 1 insertions(+), 3 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index a68aed7..375f7a0 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -502,10 +502,8 @@ static int snd_compr_pause(struct snd_compr_stream *stream) if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) return -EPERM; retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_PUSH); - if (!retval) { + if (!retval) stream->runtime->state = SNDRV_PCM_STATE_PAUSED; - wake_up(&stream->runtime->sleep); - } return retval; } -- 1.7.0.4

2 2