January 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/x][RFC] ASoC: replace codec/platform to component
by Kuninori Morimoto 12 Feb '18

12 Feb '18

Hi Mark Finally, all prepare patch-set were accepted. These are [RFC] of codec/platform replace to component patch-set. It will be almost 200 patches in total. Thus, to avoid patch bomb (in case of rejection), I pickuped few patches and post these as [RFC]. Main is replacing codec/platform to component xxx_codec_xxx() -> xxx_component_xxx() xxx_platform_xxx() -> xxx_component_xxx() Note1 is that it is using devm_ version register function, and removed unregister from .remove (sometimes, removes .remove function itself) - snd_soc_register_xxx() + devm_snd_soc_register_component() ... - snd_soc_unregister_xxx() Note2 is that rtd->platform will be removed, thus it will be replaced to snd_soc_rtdcom_lookup() with driver name - rtd->platform + snd_soc_rtdcom_lookup(rtd, DRV_NAME) Note3 is that codec driver needs to have some flags which is needed for Codec -static const struct snd_soc_codec_driver xxx_codec; +static const struct snd_soc_component_driver xxx_codec = { + .idle_bias_on = 1, + .pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; here, .idle_bias_on was originally .idle_bias_off .pmdown_time was originally .ignore_pmdown_time .endianness is needed for Codec endianness check .non_legacy_dai_naming is needed for Codec DAI name matching Note4 is that some driver is controlling CPU/Platform in one driver as 2 component. Now we can merge these into one component. - snd_soc_register_platform(xxx) // for Platform - snd_soc_register_component(xxx) // for CPU + snd_soc_register_component(xxx) // for CPU/Platform Note4 is that [1/x] patch which removes (!rtd->platform) check from soc-core is needed for all platforms. This check is no longer needed for component version. If these [RFC] were OK, I will post all replacement patch-set. I think good timing to posting is around next -rc1 or merge-window About removing all unneeded codec/platform patch-set, when is good timing ? +1 version: includes replacement patch-set +2 version: includes remove patch-set or +1 version: includes both replacement/remove patch-set Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH 00/38] ASoC: replace platform to component
by Kuninori Morimoto 12 Feb '18

12 Feb '18

Hi Mark codec/platform can be replaced to component now. These patches do it for platform. One none is that all platform should be based [01/38] patch. These are based on "asoc-v4.16" tag Current existing "platform" related codes will be no longer needed by these patches. I can post it, but I don't know when is good timing Kuninori Morimoto (38): ASoC: remove rtd->platform checck ASoC: soc-utils: replace platform to component ASoC: soc-generic-dmaengine-pcm: replace platform to component ASoC: intel: atom: replace platform to component ASoC: intel: skylake: replace platform to component ASoC: intel: baytrail: replace platform to component ASoC: intel: haswell: replace platform to component ASoC: bcm: cygnus: replace platform to component ASoC: sh: rsnd: replace platform to component ASoC: sh: dma-sh7760: replace platform to component ASoC: sh: fsi: replace platform to component ASoC: sh: siu: replace platform to component ASoC: atmel: replace platform to component ASoC: amd: replace platform to component ASoC: cs47l24: replace platform to component ASoC: rt5514-spi: replace platform to component ASoC: wm5110: replace platform to component ASoC: wm5102: replace platform to component ASoC: davinci-i2s: replace platform to component ASoC: fsl: dma: replace platform to component ASoC: fsl: asrc: replace platform to component ASoC: fsl: imx: replace platform to component ASoC: fsl: mpc5200: replace platform to component ASoC: dwc-pcm: replace platform to component ASoC: kirkwood: replace platform to component ASoC: pxa: mmp: replace platform to component ASoC: pxa: pxa2xx: replace platform to component ASoC: qcom: lpass-platform: replace platform to component ASoC: samsung: idma: replace platform to component ASoC: xtfpga-i2s: replace platform to component ASoC: blackfin: bf5xx-ac97-pcm: replace platform to component ASoC: blackfin: bf5xx-i2s-pcm: replace platform to component ASoC: omap-pcm: replace platform to component ASoC: au1x: dbdma2: replace platform to component ASoC: au1x: dma: replace platform to component ASoC: nuc900-pcm: replace platform to component ASoC: txx9aclc: replace platform to component ASoC: mt2701/mt8173: replace platform to component include/sound/dmaengine_pcm.h | 2 + sound/soc/amd/acp-pcm-dma.c | 29 +++-- sound/soc/atmel/atmel-classd.c | 4 +- sound/soc/atmel/atmel-pcm-pdc.c | 6 +- sound/soc/atmel/atmel-pdmic.c | 4 +- sound/soc/au1x/dbdma2.c | 13 ++- sound/soc/au1x/dma.c | 12 ++- sound/soc/bcm/cygnus-pcm.c | 7 +- sound/soc/blackfin/bf5xx-ac97-pcm.c | 6 +- sound/soc/blackfin/bf5xx-i2s-pcm.c | 6 +- sound/soc/codecs/cs47l24.c | 42 +++----- sound/soc/codecs/rt5514-spi.c | 32 +++--- sound/soc/codecs/wm5102.c | 42 +++----- sound/soc/codecs/wm5110.c | 42 +++----- sound/soc/davinci/davinci-i2s.c | 13 +-- sound/soc/dwc/dwc-pcm.c | 5 +- sound/soc/fsl/fsl_asrc.c | 10 -- sound/soc/fsl/fsl_asrc.h | 3 +- sound/soc/fsl/fsl_asrc_dma.c | 14 ++- sound/soc/fsl/fsl_dma.c | 27 +++-- sound/soc/fsl/imx-pcm-fiq.c | 6 +- sound/soc/fsl/mpc5200_dma.c | 16 +-- sound/soc/intel/atom/sst-atom-controls.c | 36 +++---- sound/soc/intel/atom/sst-mfld-platform-compress.c | 4 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 32 ++---- sound/soc/intel/atom/sst-mfld-platform.h | 4 +- sound/soc/intel/baytrail/sst-baytrail-pcm.c | 66 +++++------- sound/soc/intel/boards/bdw-rt5677.c | 3 +- sound/soc/intel/boards/broadwell.c | 3 +- sound/soc/intel/boards/haswell.c | 3 +- sound/soc/intel/haswell/sst-haswell-ipc.h | 2 + sound/soc/intel/haswell/sst-haswell-pcm.c | 118 +++++++++------------ sound/soc/intel/skylake/skl-debug.c | 2 +- sound/soc/intel/skylake/skl-pcm.c | 52 ++++----- sound/soc/intel/skylake/skl-topology.c | 26 ++--- sound/soc/intel/skylake/skl-topology.h | 2 +- sound/soc/intel/skylake/skl.h | 2 +- sound/soc/kirkwood/kirkwood-dma.c | 3 +- sound/soc/kirkwood/kirkwood-i2s.c | 20 +--- sound/soc/kirkwood/kirkwood.h | 4 +- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 16 ++- .../soc/mediatek/common/mtk-afe-platform-driver.c | 9 +- .../soc/mediatek/common/mtk-afe-platform-driver.h | 3 +- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 50 +++++---- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 42 ++++---- sound/soc/nuc900/nuc900-pcm.c | 5 +- sound/soc/omap/omap-pcm.c | 5 +- sound/soc/pxa/mmp-pcm.c | 11 +- sound/soc/pxa/pxa2xx-pcm.c | 5 +- sound/soc/qcom/lpass-platform.c | 42 ++++---- sound/soc/samsung/idma.c | 5 +- sound/soc/sh/dma-sh7760.c | 5 +- sound/soc/sh/fsi.c | 24 +---- sound/soc/sh/rcar/core.c | 22 +--- sound/soc/sh/siu.h | 2 +- sound/soc/sh/siu_dai.c | 10 +- sound/soc/sh/siu_pcm.c | 9 +- sound/soc/soc-core.c | 5 - sound/soc/soc-generic-dmaengine-pcm.c | 55 ++++++---- sound/soc/soc-utils.c | 6 +- sound/soc/txx9/txx9aclc.c | 26 +++-- sound/soc/xtensa/xtfpga-i2s.c | 17 +-- 62 files changed, 518 insertions(+), 579 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: rt5665: remove rt5668 support
by Bard Liao 12 Feb '18

12 Feb '18

Remove rt5668 support from rt5665 codec driver since rt5668 will not go into mass production. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- Documentation/devicetree/bindings/sound/rt5665.txt | 4 ++-- sound/soc/codecs/rt5665.c | 5 ----- sound/soc/codecs/rt5665.h | 1 - 3 files changed, 2 insertions(+), 8 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/rt5665.txt b/Documentation/devicetree/bindings/sound/rt5665.txt index 419c892..8df1705 100644 --- a/Documentation/devicetree/bindings/sound/rt5665.txt +++ b/Documentation/devicetree/bindings/sound/rt5665.txt @@ -1,10 +1,10 @@ -RT5665/RT5666/RT5668 audio CODEC +RT5665/RT5666 audio CODEC This device supports I2C only. Required properties: -- compatible : One of "realtek,rt5665", "realtek,rt5666" or "realtek,rt5668". +- compatible : One of "realtek,rt5665", "realtek,rt5666". - reg : The I2C address of the device. diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index f05d362..e1b558b 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -4828,9 +4828,6 @@ static int rt5665_i2c_probe(struct i2c_client *i2c, case 0x0: rt5665->id = CODEC_5666; break; - case 0x6: - rt5665->id = CODEC_5668; - break; case 0x3: default: rt5665->id = CODEC_5665; @@ -4963,7 +4960,6 @@ static void rt5665_i2c_shutdown(struct i2c_client *client) static const struct of_device_id rt5665_of_match[] = { {.compatible = "realtek,rt5665"}, {.compatible = "realtek,rt5666"}, - {.compatible = "realtek,rt5668"}, {}, }; MODULE_DEVICE_TABLE(of, rt5665_of_match); @@ -4973,7 +4969,6 @@ MODULE_DEVICE_TABLE(of, rt5665_of_match); static const struct acpi_device_id rt5665_acpi_match[] = { {"10EC5665", 0,}, {"10EC5666", 0,}, - {"10EC5668", 0,}, {}, }; MODULE_DEVICE_TABLE(acpi, rt5665_acpi_match); diff --git a/sound/soc/codecs/rt5665.h b/sound/soc/codecs/rt5665.h index 5ddebd6..5abff0f 100644 --- a/sound/soc/codecs/rt5665.h +++ b/sound/soc/codecs/rt5665.h @@ -1978,7 +1978,6 @@ enum { enum { CODEC_5665, CODEC_5666, - CODEC_5668, }; /* filter mask */ -- 2.7.4

2 1

[alsa-devel] [PATCH 0/2] ASoC: stm32: spdifrx: change dai name
by Olivier Moysan 12 Feb '18

12 Feb '18

This patchset changes spdifrx dai name and fixes a typo issue. Olivier Moysan (2): ASoC: stm32: spdifrx: fix typo in function name. ASoC: stm32: spdifrx: Use default dai name sound/soc/stm/stm32_spdifrx.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) -- 1.9.1

2 4

[alsa-devel] [PATCH RESEND] sound: omap-twl4030: do not complain on -EPROBE_DEFER when initializing
by Ladislav Michl 11 Feb '18

11 Feb '18

Don't complain on -EPROBE_DEFER when when initializing, the driver probe will be retried later. Signed-off-by: Ladislav Michl <ladis(a)linux-mips.org> --- Note: Just a resend of patch from 9 Nov 2016 as there were no reply by then. sound/soc/omap/omap-twl4030.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c index cccc316743fa..7c1f71884768 100644 --- a/sound/soc/omap/omap-twl4030.c +++ b/sound/soc/omap/omap-twl4030.c @@ -322,8 +322,10 @@ static int omap_twl4030_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(card, priv); ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { - dev_err(&pdev->dev, "devm_snd_soc_register_card() failed: %d\n", - ret); + if (ret != -EPROBE_DEFER) + dev_err(&pdev->dev, + "devm_snd_soc_register_card() failed: %d\n", + ret); return ret; } -- 2.15.1

2 1

[alsa-devel] [PATCH v2] ASoC: use seq_file to dump the contents of dai_list, platform_list and codec_list
by Donglin Peng 08 Feb '18

08 Feb '18

Now the debugfs files dais/platforms/codecs have a size limit PAGE_SIZE and the user can not see the whole contents of dai_list/platform_list/codec_list when they are larger than this limit. This patch uses seq_file instead to make sure dais/platforms/codecs show the full contents of dai_list/platform_list/codec_list. Signed-off-by: Donglin Peng <dolinux.peng(a)gmail.com> --- Changelog v2 -fix commit log error sound/soc/soc-core.c | 111 +++++++++++++++++---------------------------------- 1 file changed, 37 insertions(+), 74 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c0edac80df34..7b582112e3fc 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -349,120 +349,84 @@ static void soc_init_codec_debugfs(struct snd_soc_component *component) "ASoC: Failed to create codec register debugfs file\n"); } -static ssize_t codec_list_read_file(struct file *file, char __user *user_buf, - size_t count, loff_t *ppos) +static int codec_list_seq_show(struct seq_file *m, void *v) { - char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL); - ssize_t len, ret = 0; struct snd_soc_codec *codec; - if (!buf) - return -ENOMEM; - mutex_lock(&client_mutex); - list_for_each_entry(codec, &codec_list, list) { - len = snprintf(buf + ret, PAGE_SIZE - ret, "%s\n", - codec->component.name); - if (len >= 0) - ret += len; - if (ret > PAGE_SIZE) { - ret = PAGE_SIZE; - break; - } - } + list_for_each_entry(codec, &codec_list, list) + seq_printf(m, "%s\n", codec->component.name); mutex_unlock(&client_mutex); - if (ret >= 0) - ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret); - - kfree(buf); + return 0; +} - return ret; +static int codec_list_seq_open(struct inode *inode, struct file *file) +{ + return single_open(file, codec_list_seq_show, NULL); } static const struct file_operations codec_list_fops = { - .read = codec_list_read_file, - .llseek = default_llseek,/* read accesses f_pos */ + .open = codec_list_seq_open, + .read = seq_read, + .llseek = seq_lseek, + .release = single_release, }; -static ssize_t dai_list_read_file(struct file *file, char __user *user_buf, - size_t count, loff_t *ppos) +static int dai_list_seq_show(struct seq_file *m, void *v) { - char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL); - ssize_t len, ret = 0; struct snd_soc_component *component; struct snd_soc_dai *dai; - if (!buf) - return -ENOMEM; - mutex_lock(&client_mutex); - list_for_each_entry(component, &component_list, list) { - list_for_each_entry(dai, &component->dai_list, list) { - len = snprintf(buf + ret, PAGE_SIZE - ret, "%s\n", - dai->name); - if (len >= 0) - ret += len; - if (ret > PAGE_SIZE) { - ret = PAGE_SIZE; - break; - } - } - } + list_for_each_entry(component, &component_list, list) + list_for_each_entry(dai, &component->dai_list, list) + seq_printf(m, "%s\n", dai->name); mutex_unlock(&client_mutex); - ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret); - - kfree(buf); + return 0; +} - return ret; +static int dai_list_seq_open(struct inode *inode, struct file *file) +{ + return single_open(file, dai_list_seq_show, NULL); } static const struct file_operations dai_list_fops = { - .read = dai_list_read_file, - .llseek = default_llseek,/* read accesses f_pos */ + .open = dai_list_seq_open, + .read = seq_read, + .llseek = seq_lseek, + .release = single_release, }; -static ssize_t platform_list_read_file(struct file *file, - char __user *user_buf, - size_t count, loff_t *ppos) +static int platform_list_seq_show(struct seq_file *m, void *v) { - char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL); - ssize_t len, ret = 0; struct snd_soc_platform *platform; - if (!buf) - return -ENOMEM; - mutex_lock(&client_mutex); - list_for_each_entry(platform, &platform_list, list) { - len = snprintf(buf + ret, PAGE_SIZE - ret, "%s\n", - platform->component.name); - if (len >= 0) - ret += len; - if (ret > PAGE_SIZE) { - ret = PAGE_SIZE; - break; - } - } + list_for_each_entry(platform, &platform_list, list) + seq_printf(m, "%s\n", platform->component.name); mutex_unlock(&client_mutex); - ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret); - - kfree(buf); + return 0; +} - return ret; +static int platform_list_seq_open(struct inode *inode, struct file *file) +{ + return single_open(file, platform_list_seq_show, NULL); } static const struct file_operations platform_list_fops = { - .read = platform_list_read_file, - .llseek = default_llseek,/* read accesses f_pos */ + .open = platform_list_seq_open, + .read = seq_read, + .llseek = seq_lseek, + .release = single_release, }; static void soc_init_card_debugfs(struct snd_soc_card *card) @@ -491,7 +455,6 @@ static void soc_cleanup_card_debugfs(struct snd_soc_card *card) debugfs_remove_recursive(card->debugfs_card_root); } - static void snd_soc_debugfs_init(void) { snd_soc_debugfs_root = debugfs_create_dir("asoc", NULL); -- 2.15.1

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[alsa-devel] [PATCH - JACK PCM plugin] jack: Write only valid values to the internal hw_ptr
by twischer＠de.adit-jv.com 08 Feb '18

08 Feb '18

From: Timo Wischer <twischer(a)de.adit-jv.com> Otherwise the ALSA thread could for example read a hw_ptr > buffer_size Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> diff --git a/jack/pcm_jack.c b/jack/pcm_jack.c index 4468ede..3aed332 100644 --- a/jack/pcm_jack.c +++ b/jack/pcm_jack.c @@ -137,6 +137,7 @@ static int snd_pcm_jack_process_cb(jack_nframes_t nframes, snd_pcm_ioplug_t *io) { snd_pcm_jack_t *jack = io->private_data; + snd_pcm_uframes_t hw_ptr; const snd_pcm_channel_area_t *areas; snd_pcm_uframes_t xfer = 0; unsigned int channel; @@ -155,12 +156,13 @@ snd_pcm_jack_process_cb(jack_nframes_t nframes, snd_pcm_ioplug_t *io) return 0; } } - + + hw_ptr = jack->hw_ptr; areas = snd_pcm_ioplug_mmap_areas(io); while (xfer < nframes) { snd_pcm_uframes_t frames = nframes - xfer; - snd_pcm_uframes_t offset = jack->hw_ptr; + snd_pcm_uframes_t offset = hw_ptr; snd_pcm_uframes_t cont = io->buffer_size - offset; if (cont < frames) @@ -173,10 +175,11 @@ snd_pcm_jack_process_cb(jack_nframes_t nframes, snd_pcm_ioplug_t *io) snd_pcm_area_copy(&areas[channel], offset, &jack->areas[channel], xfer, frames, io->format); } - jack->hw_ptr += frames; - jack->hw_ptr %= io->buffer_size; + hw_ptr += frames; + hw_ptr %= io->buffer_size; xfer += frames; } + jack->hw_ptr = hw_ptr; pcm_poll_unblock_check(io); /* unblock socket for polling if needed */ -- 2.7.4

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[alsa-devel] [RESEND PATCH v2 00/15] ASoC: qcom: Add support to QDSP6 based audio
by srinivas.kandagatla＠linaro.org 07 Feb '18

07 Feb '18

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> This patchset aims to provide a very basic version of QCOM DSP based audio support which is available in downstream andriod kernels. This patchset only support Digital audio based for HDMI-RX and will add support to other features as we move on. QDSP has both static and dynamic modules. static modules like AFE (Audio FrontEnd), ADM (Audio Device Manager), ASM(Audio Stream Manager) and CORE to provide this audio services. All these services use APR (Asynchronous Packet Router) protocol via smd/glink transport to communicate with Application processor. More details on each module is availble in there respective patch. This patchset is tested on DB820c, with HDMI audio playback on top of mainline, also tested slimbus analog audio using wcd9355 with an additional small patch. Here is block diagram to give a quick overview of the components +---------+ +---------+ +---------+ | q6asm | |q6routing| | q6afe | | fedai | <------> | mixers | <-----> | bedai | +---------+ +---------+ +---------+ ^ ^ ^ | | | | +------------------+----------------+ | | | | | | v v v v v +---------+ +---------+ +---------+ | q6ASM | | q6ADM | | q6AFE | +---------+ +---------+ +---------+ ^ ^ ^ ^ | | | CPU Side | ------+---------------------+-------------------+-------- | | | | |APR(smd/glink) | | | | | +------------------+----------------+ | | | | | | +-----+--+-----------------------------------+--+------- | | | | | QDSP Side | v v v v v v +---------+ +---------+ +---------+ | ASM | <------> | ADM | <-----> | AFE | +---------+ +---------+ +---------+ ^ | +-------------------+ | ---------------------------+-------------------------- | Audio I/O | v v +--------------------------------------------------+ | Audio devices | | CODEC | HDMI-TX | PCM | SLIMBUS | I2S |MI2S |...| | | +--------------------------------------------------+ Changes since RFC: - Converted APR driver to proper bus driver - Added API version info to apr devices - Added q6core driver. - Fixed various issues spotted by Banajit and Kasam. - Added kernel doc to exported symbols. - renamed qdsp6v2 to qdsp6 as api version info can be used from apr device. - removed unnecessary dt bindings for apr devices. Srinivas Kandagatla (15): dt-bindings: soc: qcom: Add bindings for APR bus soc: qcom: add support to APR bus driver ASoC: qcom: qdsp6: Add common qdsp6 helper functions ASoC: qcom: qdsp6: Add support to Q6AFE ASoC: qcom: qdsp6: Add support to Q6ADM ASoC: qcom: qdsp6: Add support to Q6ASM ASoC: qcom: q6asm: Add support to memory map and unmap ASoC: qcom: q6asm: add support to audio stream apis ASoC: qcom: qdsp6: Add support to Q6CORE ASoC: qcom: qdsp6: Add support to q6routing driver ASoC: qcom: qdsp6: Add support to q6afe dai driver ASoC: qcom: qdsp6: Add support to q6asm dai driver dt-bindings: sound: qcom: Add devicetree bindings for apq8096 ASoC: qcom: apq8096: Add db820c machine driver arm64: dts: msm8996: db820c: Add sound card support .../devicetree/bindings/soc/qcom/qcom,apr.txt | 28 + .../devicetree/bindings/sound/qcom,apq8096.txt | 22 + arch/arm64/boot/dts/qcom/apq8096-db820c.dtsi | 5 + arch/arm64/boot/dts/qcom/msm8996.dtsi | 33 + drivers/soc/qcom/Kconfig | 8 + drivers/soc/qcom/Makefile | 1 + drivers/soc/qcom/apr.c | 395 +++++++ include/linux/mod_devicetable.h | 13 + include/linux/soc/qcom/apr.h | 174 +++ sound/soc/qcom/Kconfig | 51 + sound/soc/qcom/Makefile | 5 + sound/soc/qcom/apq8096.c | 124 +++ sound/soc/qcom/qdsp6/Makefile | 7 + sound/soc/qcom/qdsp6/common.h | 225 ++++ sound/soc/qcom/qdsp6/q6adm.c | 602 +++++++++++ sound/soc/qcom/qdsp6/q6adm.h | 24 + sound/soc/qcom/qdsp6/q6afe-dai.c | 241 +++++ sound/soc/qcom/qdsp6/q6afe.c | 503 +++++++++ sound/soc/qcom/qdsp6/q6afe.h | 30 + sound/soc/qcom/qdsp6/q6asm-dai.c | 534 ++++++++++ sound/soc/qcom/qdsp6/q6asm.c | 1119 ++++++++++++++++++++ sound/soc/qcom/qdsp6/q6asm.h | 61 ++ sound/soc/qcom/qdsp6/q6core.c | 227 ++++ sound/soc/qcom/qdsp6/q6routing.c | 386 +++++++ sound/soc/qcom/qdsp6/q6routing.h | 9 + 25 files changed, 4827 insertions(+) create mode 100644 Documentation/devicetree/bindings/soc/qcom/qcom,apr.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,apq8096.txt create mode 100644 drivers/soc/qcom/apr.c create mode 100644 include/linux/soc/qcom/apr.h create mode 100644 sound/soc/qcom/apq8096.c create mode 100644 sound/soc/qcom/qdsp6/Makefile create mode 100644 sound/soc/qcom/qdsp6/common.h create mode 100644 sound/soc/qcom/qdsp6/q6adm.c create mode 100644 sound/soc/qcom/qdsp6/q6adm.h create mode 100644 sound/soc/qcom/qdsp6/q6afe-dai.c create mode 100644 sound/soc/qcom/qdsp6/q6afe.c create mode 100644 sound/soc/qcom/qdsp6/q6afe.h create mode 100644 sound/soc/qcom/qdsp6/q6asm-dai.c create mode 100644 sound/soc/qcom/qdsp6/q6asm.c create mode 100644 sound/soc/qcom/qdsp6/q6asm.h create mode 100644 sound/soc/qcom/qdsp6/q6core.c create mode 100644 sound/soc/qcom/qdsp6/q6routing.c create mode 100644 sound/soc/qcom/qdsp6/q6routing.h -- 2.15.0

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[alsa-devel] [PATCH] ALSA: hda - Fix headset mic detection problem for two Dell machines
by Hui Wang 07 Feb '18

07 Feb '18

One of them has the codec of alc256 and the other one has the codec of alc289. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(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 2347588..b791efe 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6735,6 +6735,11 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x14, 0x90170110}, {0x21, 0x02211020}), SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0x90a60130}, + {0x14, 0x90170110}, + {0x14, 0x01011020}, + {0x21, 0x0221101f}), + SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, ALC256_STANDARD_PINS), SND_HDA_PIN_QUIRK(0x10ec0256, 0x1043, "ASUS", ALC256_FIXUP_ASUS_MIC, {0x14, 0x90170110}, @@ -6803,6 +6808,10 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x12, 0x90a60120}, {0x14, 0x90170110}, {0x21, 0x0321101f}), + SND_HDA_PIN_QUIRK(0x10ec0289, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0xb7a60130}, + {0x14, 0x90170110}, + {0x21, 0x04211020}), SND_HDA_PIN_QUIRK(0x10ec0290, 0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1, ALC290_STANDARD_PINS, {0x15, 0x04211040}, -- 2.7.4

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[alsa-devel] [alsa-utils] [PATCH] Add support for no-rewind flag in aplay utility
by Sriram Periyasamy 07 Feb '18

07 Feb '18

From: Ramesh Babu <ramesh.babu(a)intel.com> The "no-rewind" flag is introduced in alsa. Application can use this flag to assure kernel driver that rewind won't be exercised. Driver could use this flag to alter it's buffering scheme to optimize power and performance. Signed-off-by: Ramesh Babu <ramesh.babu(a)intel.com> Signed-off-by: Sriram Periyasamy <sriramx.periyasamy(a)intel.com> --- aplay/aplay.c | 14 +++++++++++++- 1 file changed, 13 insertions(+), 1 deletion(-) diff --git a/aplay/aplay.c b/aplay/aplay.c index 6b740c281f8e..b50602a59024 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -108,6 +108,7 @@ static snd_pcm_stream_t stream = SND_PCM_STREAM_PLAYBACK; static int mmap_flag = 0; static int interleaved = 1; static int nonblock = 0; +static int no_rewind = 0; static volatile sig_atomic_t in_aborting = 0; static u_char *audiobuf = NULL; static snd_pcm_uframes_t chunk_size = 0; @@ -229,6 +230,7 @@ _("Usage: %s [OPTION]... [FILE]...\n" "-I, --separate-channels one file for each channel\n" "-i, --interactive allow interactive operation from stdin\n" "-m, --chmap=ch1,ch2,.. Give the channel map to override or follow\n" +"-w, --no-rewind enables no rewind mode\n" " --disable-resample disable automatic rate resample\n" " --disable-channels disable automatic channel conversions\n" " --disable-format disable automatic format conversions\n" @@ -470,7 +472,7 @@ int main(int argc, char *argv[]) { int duration_or_sample = 0; int option_index; - static const char short_options[] = "hnlLD:qt:c:f:r:d:s:MNF:A:R:T:B:vV:IPCi" + static const char short_options[] = "hnlLD:qt:c:f:r:d:s:w:MNF:A:R:T:B:vV:IPCi" #ifdef CONFIG_SUPPORT_CHMAP "m:" #endif @@ -514,6 +516,7 @@ int main(int argc, char *argv[]) {"process-id-file", 1, 0, OPT_PROCESS_ID_FILE}, {"use-strftime", 0, 0, OPT_USE_STRFTIME}, {"interactive", 0, 0, 'i'}, + {"no-rewind", 0, 0, 'w'}, {"dump-hw-params", 0, 0, OPT_DUMP_HWPARAMS}, {"fatal-errors", 0, 0, OPT_FATAL_ERRORS}, #ifdef CONFIG_SUPPORT_CHMAP @@ -753,6 +756,9 @@ int main(int argc, char *argv[]) case 'i': interactive = 1; break; + case 'w': + no_rewind = 1; + break; case OPT_DISABLE_RESAMPLE: open_mode |= SND_PCM_NO_AUTO_RESAMPLE; break; @@ -1396,6 +1402,12 @@ static void set_params(void) &buffer_frames); } assert(err >= 0); + + if (no_rewind) { + err = snd_pcm_hw_params_set_no_rewind(handle, params, 1); + assert(err >= 0); + } + monotonic = snd_pcm_hw_params_is_monotonic(params); can_pause = snd_pcm_hw_params_can_pause(params); err = snd_pcm_hw_params(handle, params); -- 2.7.4

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