February 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] .gitignore: add test-driver
by Tanu Kaskinen 16 Feb '15

16 Feb '15

Signed-off-by: Tanu Kaskinen <tanu.kaskinen(a)linux.intel.com> --- .gitignore | 1 + 1 file changed, 1 insertion(+) diff --git a/.gitignore b/.gitignore index 05a2459..e1cc163 100644 --- a/.gitignore +++ b/.gitignore @@ -11,6 +11,7 @@ Makefile libtool depcomp ltmain.sh +test-driver version Makefile.in aclocal.m4 -- 1.9.3

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[alsa-devel] [PATCH 0/6] Modernize sequecner device binding
by Takashi Iwai 16 Feb '15

16 Feb '15

Hi, this is a series of patches for 3.21 (or whatever the next-to-next) to sanitze and modernize the device binding code of ALSA sequencer core. Takashi Takashi Iwai (6): ALSA: seq: Move EXPORT_SYMBOL() after each function ALSA: seq: Don't compile snd_seq_device_load_drivers() for built-in ALSA: seq: Rewrite sequencer device binding with standard bus ALSA: seq: Clean up device and driver structs ALSA: seq: Define driver object in each driver ALSA: seq: Drop snd_seq_autoload_lock() and _unlock() include/sound/seq_device.h | 46 +-- include/sound/seq_kernel.h | 6 +- sound/core/seq/oss/seq_oss.c | 22 +- sound/core/seq/oss/seq_oss_synth.c | 6 +- sound/core/seq/oss/seq_oss_synth.h | 4 +- sound/core/seq/seq_device.c | 569 +++++++++---------------------------- sound/core/seq/seq_dummy.c | 6 +- sound/core/seq/seq_midi.c | 36 +-- sound/drivers/opl3/opl3_seq.c | 34 +-- sound/drivers/opl4/opl4_seq.c | 33 +-- sound/isa/sb/emu8000_synth.c | 35 +-- sound/pci/emu10k1/emu10k1_synth.c | 35 +-- 12 files changed, 256 insertions(+), 576 deletions(-) -- 2.3.0

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[alsa-devel] [PATCH] MS Lifecam HD-5000: Don't attempt to get the sample rate
by Joe Turner 14 Feb '15

14 Feb '15

Hi Folks, This is my first time doing anything kernel-related, so apologies in advance for any faux-pas which are to follow. I hope this is at least going to the correct place! Currently after setting the sample rate of USB devices the driver tries to get the sample rate back from the device, to check that the two match up. The Lifecam doesn't support getting the sample rate, and so this operation times-out, and we continue regardless. Because the USB timeout is pretty long this causes issues with some apps, which begin to doubt the ability of the device to produce audio before the timeout is up and misbehave in various ways. See https://www.google.co.uk/search?q=%22cannot+get+freq+at+ep+0x82%22 for a range of reports relating to this issue. This patch adds a quirk so that rather than check the sample rate and timeout, we don't bother checking at all. Hope this all looks okay. Cheers, Joe

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[alsa-devel] [PATCH] ALSA: hda/tegra check correct return value from ioremap_resource
by eliot＠blennerhassett.gen.nz 14 Feb '15

14 Feb '15

From: Eliot Blennerhassett <eliot(a)blennerhassett.gen.nz> Signed-off-by: Eliot Blennerhassett <eliot(a)blennerhassett.gen.nz> --- sound/pci/hda/hda_tegra.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 227990b..375e94f 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -329,8 +329,8 @@ static int hda_tegra_init_chip(struct azx *chip, struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); hda->regs = devm_ioremap_resource(dev, res); - if (IS_ERR(chip->remap_addr)) - return PTR_ERR(chip->remap_addr); + if (IS_ERR(hda->regs)) + return PTR_ERR(hda->regs); chip->remap_addr = hda->regs + HDA_BAR0; chip->addr = res->start + HDA_BAR0; -- 1.9.1

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[alsa-devel] [PATCH] ASoC: fsl-asoc-card: Add snd_soc_of_parse_audio_routing()
by Nicolin Chen 14 Feb '15

14 Feb '15

This patch adds snd_soc_of_parse_audio_routing() to get dapm routes via Device Tree. Because snd_soc_of_parse_audio_routing() overrides dpam_routes, we use snd_soc_dapm_add_routes() to append the original build-in DPCM routes instead of linking them to dpam_routes directly. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/fsl/fsl-asoc-card.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 3f6959c..bdc9671 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -388,6 +388,8 @@ static int fsl_asoc_card_late_probe(struct snd_soc_card *card) struct device *dev = card->dev; int ret; + snd_soc_dapm_add_routes(&card->dapm, audio_map, ARRAY_SIZE(audio_map)); + ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, codec_priv->mclk_freq, SND_SOC_CLOCK_IN); if (ret) { @@ -503,15 +505,19 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->card.dev = &pdev->dev; priv->card.name = priv->name; priv->card.dai_link = priv->dai_link; - priv->card.dapm_routes = audio_map; priv->card.late_probe = fsl_asoc_card_late_probe; - priv->card.num_dapm_routes = ARRAY_SIZE(audio_map); priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets; priv->card.num_dapm_widgets = ARRAY_SIZE(fsl_asoc_card_dapm_widgets); memcpy(priv->dai_link, fsl_asoc_card_dai, sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link)); + ret = snd_soc_of_parse_audio_routing(&priv->card, "audio-routing"); + if (ret) { + dev_err(&pdev->dev, "failed to parse audio-routing: %d\n", ret); + goto asrc_fail; + } + /* Normal DAI Link */ priv->dai_link[0].cpu_of_node = cpu_np; priv->dai_link[0].codec_of_node = codec_np; -- 1.9.1

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[alsa-devel] [Patch V4 00/10] ASoC: QCOM: Add support for ipq806x SOC
by Kenneth Westfield 14 Feb '15

14 Feb '15

From: Kenneth Westfield <kwestfie(a)codeaurora.org> This patch series adds support for I2S audio playback on the Qualcomm Technologies ipq806x SOC. The ipq806x SOC has audio-related hardware blocks in its low-power audio subsystem (or LPASS). One of the relevant blocks in the LPASS is its low-power audio interface (or LPAIF). This contains an MI2S port, which is what these drivers are configured to use. The LPAIF also contains a DMA engine that is dedicated to moving audio samples into the transmit FIFO of the MI2S port. In addition, there is also low-power memory (LPM) within the audio subsystem, which is used for buffering the audio samples. The development board being used for testing contains the ipq806x SOC and a Maxim max98357a DAC/amp. One bus from the MI2S port of the SOC is connected to the DAC/amp for stereo playback. This bus is configured so that the SOC is bus master and consists of DATA, LRCLK, and BCLK. The DAC/amp does not need MCLK to operate. In addition, a single GPIO pin from the SOC is connected to the same DAC/amp, which gives enable/disable control over the DAC/amp. The specific drivers added are: * a codec DAI driver for controlling the DAC/amp * a CPU DAI driver for controlling the MI2S port * a platform driver for controlling the LPAIF DMA engine These drivers, together, are tied into simple-audio-card to complete the audio implementation. Corresponding additions to the device tree for the ipq806x SOC and its documentation has also been added. Also, as this is a new directory, the MAINTAINERS file has been updated as well. The LPASS also contains clocks that need to be controlled. Those drivers have been submitted as a separate patch series: [PATCH v3 0/8] qcom audio clock control drivers http://lkml.org/lkml/2015/1/19/656 = Changes since V3 [Patch V3 00/10] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2014-December/085694.h… * Placed the content of the inline functions into the callbacks. * Replaced use of readl/writel register access functions with regmap access functions. Notable exception is the ISR, which uses ioread32/iowrite32. * Rearranged the sequencing of the hardware block enables to fit within the ASoC framework callbacks, while remaining functional. REQ 1: The hardware requires the enable sequence to be: LPAIF-DMA[enable],then LPAIF-MI2S[enable], then DAC-GPIO[enable] REQ 2: The hardware requires the disable sequence to be: DAC-GPIO[disable], then LPAIF-MI2S[disable] * Corrected the implementation of the pointer callback. * Utilize the LPM to buffer audio samples, rather than memory external to LPASS. * Corrected the interrupt clearing in the ISR. * Implemented a default system clock (defined by the simple-card DT node), and optional LPASS DT node modifiers that can alter the system clock in order to expand the range of available bit clock frequencies. * Addressed all of the remaining issues raised by Mark Brown. * General code cleanup. = Changes since V2 [Patch v2 00/11] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2014-December/085186.h… * Removed the PCM platform driver from the DTS platform and tied it to the CPU DAI driver. * Changed I2S pinctrl to use generic naming convention and moved control to CPU DAI driver. It should be controlled now by soc-core's pinctrl_pm_* functionality. * Added stub DAPM support in codec driver. As the DAC GPIO needs to be enabled last when starting playback, and disabled first when stopping playback, it seems as though the trigger function may be the place for this. Suggestions are welcome for a better place to put this. * Removed machine driver and tied DAI drivers to simple-audio-card. * Packaged the build files and Maxim codec files together in one change. * Removed QCOM as vendor from Maxim code and documentation. * Separated the SOC and board definitions into the correct DTS files. * Update device tree documentation to reflect changes. * General code cleanup. = Changes since V1 [PATCH 0/9] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2014-November/084322.h… * Remove the native LPAIF driver, and move its functionality to the CPU DAI driver. * Add a codec driver to manage the pins going to the external DAC (previously managed by the machine driver). * Use devm_* and dev_* where possible. * ISR only handles relevant DMA channel now. * Update device tree documentation to reflect changes. * General code cleanup. Kenneth Westfield (10): MAINTAINERS: Add QCOM audio ASoC maintainer ASoC: max98357a: Document MAX98357A bindings ASoC: qcom: Document LPASS CPU bindings ASoC: codec: Add MAX98357A codec driver ASoC: ipq806x: add LPASS header files ASoC: ipq806x: Add LPASS CPU DAI driver ASoC: ipq806x: Add LPASS platform driver ASoC: qcom: Add ability to build QCOM drivers ASoC: Allow for building QCOM drivers ARM: dts: Model IPQ LPASS audio hardware .../devicetree/bindings/sound/max98357a.txt | 14 + .../devicetree/bindings/sound/qcom,lpass-cpu.txt | 66 +++ MAINTAINERS | 7 + arch/arm/boot/dts/qcom-ipq8064.dtsi | 23 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max98357a.c | 138 +++++ sound/soc/qcom/Kconfig | 27 + sound/soc/qcom/Makefile | 6 + sound/soc/qcom/lpass-cpu.c | 557 +++++++++++++++++++++ sound/soc/qcom/lpass-lpaif-ipq806x.h | 170 +++++++ sound/soc/qcom/lpass-platform.c | 514 +++++++++++++++++++ sound/soc/qcom/lpass.h | 64 +++ 15 files changed, 1594 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98357a.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt create mode 100644 sound/soc/codecs/max98357a.c create mode 100644 sound/soc/qcom/Kconfig create mode 100644 sound/soc/qcom/Makefile create mode 100644 sound/soc/qcom/lpass-cpu.c create mode 100644 sound/soc/qcom/lpass-lpaif-ipq806x.h create mode 100644 sound/soc/qcom/lpass-platform.c create mode 100644 sound/soc/qcom/lpass.h -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

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[alsa-devel] [PATCH] ASoC: Intel: set initial runtime PM status to active for ACPI-enumerated ADSP
by mengdong.lin＠intel.com 14 Feb '15

14 Feb '15

From: Mengdong Lin <mengdong.lin(a)intel.com> The ADSP on Braswell/Baytrail is an ACPI device. This patch sets its initial runtime PM status to active. Otherwise, its initial status is suspended and runtime_suspend ops will not be called after probe and thus cannot further trigger ACPI _PS3 (D3) method to put the device into low power D3cold state. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c index 8a8d56a..d6ea800 100644 --- a/sound/soc/intel/sst/sst.c +++ b/sound/soc/intel/sst/sst.c @@ -379,6 +379,10 @@ void sst_configure_runtime_pm(struct intel_sst_drv *ctx) * initially active. So change the state to active before * enabling the pm */ + + if (!acpi_disabled) + pm_runtime_set_active(ctx->dev); + pm_runtime_enable(ctx->dev); if (acpi_disabled) -- 1.9.1

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[alsa-devel] [Patch] ASoC: codec: Add missing deps/hdrs to MAX98357A
by Kenneth Westfield 14 Feb '15

14 Feb '15

From: Kenneth Westfield <kwestfie(a)codeaurora.org> For the max98357a codec driver: - Add missing build dependancy to GPIOLIB in Kconfig. - Add header files to avoid implicit declarations and indirect inclusions. - Remove use of DRV_NAME constant. Signed-off-by: Kenneth Westfield <kwestfie(a)codeaurora.org> --- for topic/max98357a branch sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/max98357a.c | 23 +++++++++++++++-------- 2 files changed, 16 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6ecac1e4428eafc491d6cfc25110f6b3dbf699ed..0500b3ed5fa5ac4d45663ad659da9db3003bc5a3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -69,7 +69,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98088 if I2C select SND_SOC_MAX98090 if I2C select SND_SOC_MAX98095 if I2C - select SND_SOC_MAX98357A + select SND_SOC_MAX98357A if GPIOLIB select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9768 if I2C select SND_SOC_MAX9877 if I2C diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index 1806333ea29e5a6b49e2a0adb8c58257d34b7508..4ee23fbc4e1237268ae104319a9cfe4fc369934e 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -12,11 +12,19 @@ * max98357a.c -- MAX98357A ALSA SoC Codec driver */ -#include <linux/module.h> +#include <linux/device.h> +#include <linux/err.h> #include <linux/gpio.h> +#include <linux/gpio/consumer.h> +#include <linux/kernel.h> +#include <linux/mod_devicetable.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/platform_device.h> +#include <sound/pcm.h> #include <sound/soc.h> - -#define DRV_NAME "max98357a" +#include <sound/soc-dai.h> +#include <sound/soc-dapm.h> static int max98357a_daiops_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) @@ -77,9 +85,9 @@ static struct snd_soc_dai_ops max98357a_dai_ops = { }; static struct snd_soc_dai_driver max98357a_dai_driver = { - .name = DRV_NAME, + .name = "max98357a", .playback = { - .stream_name = DRV_NAME "-playback", + .stream_name = "max98357a-playback", .formats = SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_S32, @@ -117,7 +125,7 @@ static int max98357a_platform_remove(struct platform_device *pdev) #ifdef CONFIG_OF static const struct of_device_id max98357a_device_id[] = { - { .compatible = "maxim," DRV_NAME, }, + { .compatible = "maxim,max98357a" }, {} }; MODULE_DEVICE_TABLE(of, max98357a_device_id); @@ -125,7 +133,7 @@ MODULE_DEVICE_TABLE(of, max98357a_device_id); static struct platform_driver max98357a_platform_driver = { .driver = { - .name = DRV_NAME, + .name = "max98357a", .of_match_table = of_match_ptr(max98357a_device_id), }, .probe = max98357a_platform_probe, @@ -135,4 +143,3 @@ module_platform_driver(max98357a_platform_driver); MODULE_DESCRIPTION("Maxim MAX98357A Codec Driver"); MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:" DRV_NAME); -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

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[alsa-devel] [PATCH] ASoC: omap-hdmi-audio: Add missing error check for devm_kzalloc
by Kiran Padwal 14 Feb '15

14 Feb '15

This patch add a missing check on the return value of devm_kzalloc, which would cause a NULL pointer dereference in a OOM situation. Signed-off-by: Kiran Padwal <kiran.padwal(a)smartplayin.com> --- sound/soc/omap/omap-hdmi-audio.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/omap/omap-hdmi-audio.c b/sound/soc/omap/omap-hdmi-audio.c index ccfb41c..f7eb42a 100644 --- a/sound/soc/omap/omap-hdmi-audio.c +++ b/sound/soc/omap/omap-hdmi-audio.c @@ -352,6 +352,9 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev) return ret; card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!card) + return -ENOMEM; + card->name = devm_kasprintf(dev, GFP_KERNEL, "HDMI %s", dev_name(ad->dssdev)); card->owner = THIS_MODULE; -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: davinci-mcasp: Add support for CBS_CFM mode
by Peter Ujfalusi 14 Feb '15

14 Feb '15

Support for setups where codec is bitclock slave and frame master. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 750124030576..c0e485cba9eb 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -441,6 +441,18 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AFSX | AFSR); mcasp->bclk_master = 1; break; + case SND_SOC_DAIFMT_CBS_CFM: + /* codec is clock slave and frame master */ + mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + + mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + + mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, ACLKX | ACLKR); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AFSX | AFSR); + mcasp->bclk_master = 1; + break; case SND_SOC_DAIFMT_CBM_CFS: /* codec is clock master and frame slave */ mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); -- 2.3.0

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