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

[alsa-devel] [PATCH RFC 00/26] Kill set_fs() in ALSA codes
by Takashi Iwai 27 Mar '19

27 Mar '19

Hi, this is a slightly lengthy patchset, basically targeted for 4.13, thus it's still in RFC. The main purpose of this patchset is to get rid of get_fs() / set_fs() usages in ALSA codes. The patchset starts from a patch that looks fairly irrelevant with the goal (the conversion of HD-audio bind-mute code), but it's a step for the long trail. The biggest change in the patchset is the conversion of copy & silence PCM ops into the new copy_silence ops. It helped for reducing the code size and for handling the in-kernel buffer copies. The current patches are found in topic/kill-set_fs branch of my sound.git tree. thanks, Takashi === Takashi Iwai (26): ALSA: hda - Simplify bound-beep mute control for ALC268 ALSA: hda - Move bind-mixer switch codes to generic parser ALSA: hda - Remove the generic bind ctl helpers ALSA: hda - Remove the use of set_fs() ALSA: hda - Fix a typo in comment ALSA: hda - Remove superfluous header inclusions ALSA: opl3: Kill unused set_fs() ALSA: emu10k1: Get rid of set_fs() usage ALSA: pcm: Remove set_fs() in PCM core code ALSA: pcm: Introduce copy_silence PCM ops ALSA: Update document about copy_silence PCM ops ALSA: dummy: Convert to copy_silence ops ALSA: es1938: Convert to copy_silence ops ALSA: korg1212: Convert to copy_silence ops ALSA: nm256: Convert to copy_silence ops ALSA: rme32: Convert to copy_silence ops ALSA: rme96: Convert to copy_silence ops ALSA: rme9652: Convert to copy_silence ops ALSA: hdsp: Convert to copy_silence ops ALSA: gus: Convert to copy_silence ops ALSA: sb: Convert to copy_silence ops ALSA: sh: Convert to copy_silence ops ASoC: blackfin: Convert to copy_silence ops [media] solo6x10: Convert to copy_silence ops ALSA: pcm: Drop the old copy and silence ops ALSA: pcm: Kill set_fs() usage in OSS layer and USB gadget .../sound/kernel-api/writing-an-alsa-driver.rst | 110 +++++---- drivers/media/pci/solo6x10/solo6x10-g723.c | 11 +- drivers/usb/gadget/function/u_uac1.c | 8 +- include/sound/pcm.h | 19 +- sound/core/oss/pcm_oss.c | 62 ++--- sound/core/pcm_compat.c | 22 +- sound/core/pcm_lib.c | 174 +++++++++----- sound/core/pcm_native.c | 122 ++++++---- sound/drivers/dummy.c | 13 +- sound/drivers/opl3/opl3_oss.c | 14 -- sound/isa/gus/gus_pcm.c | 43 +--- sound/isa/sb/emu8000_pcm.c | 99 +++----- sound/pci/emu10k1/emufx.c | 127 ++++++----- sound/pci/es1938.c | 11 +- sound/pci/hda/hda_codec.c | 251 +++------------------ sound/pci/hda/hda_generic.c | 46 +++- sound/pci/hda/hda_local.h | 61 ----- sound/pci/hda/patch_realtek.c | 37 ++- sound/pci/korg1212/korg1212.c | 128 +++-------- sound/pci/nm256/nm256.c | 35 ++- sound/pci/rme32.c | 49 ++-- sound/pci/rme96.c | 52 ++--- sound/pci/rme9652/hdsp.c | 44 ++-- sound/pci/rme9652/rme9652.c | 46 ++-- sound/sh/sh_dac_audio.c | 40 +--- sound/soc/blackfin/bf5xx-ac97-pcm.c | 6 +- sound/soc/blackfin/bf5xx-ac97.c | 18 +- sound/soc/blackfin/bf5xx-i2s-pcm.c | 46 ++-- sound/soc/soc-pcm.c | 3 +- 29 files changed, 696 insertions(+), 1001 deletions(-) -- 2.12.2

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[alsa-devel] [PATCH v3 00/25] ASoC: qcom: Add support to QDSP based Audio
by srinivas.kandagatla＠linaro.org 21 May '18

21 May '18

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> This patchset aims to provide a basic version of QCOM DSP based audio support which is available in downstream andriod kernels. This patchset support audio playback on HDMI-RX, MI2S, SLIMBus 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, MI2S on DB410c on top of mainline, Also tested SLIMBus analog audio using wcd9355 with an additional patches. Here is my test branch incase somone want to try these patches https://git.linaro.org/people/srinivas.kandagatla/linux.git/log/?h=v4.16-rc… 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 v2 (https://lwn.net/Articles/741472/) - fixed locking issues with ASM and APR. - Added support to MI2S and SLIMBus ports - moved dma device to apr. - added dt bindings for ASM, ADM, AFE. - Fixed dt bindings for msm8996 sound card. - added compatible strings to ASM, ADM, AFE modules. - Cleaned up common code as suggested in review. - converted sound card driver to proper apr client driver. - Fixed various issues spotted by Banajit and Kasam. - Tested on DB410c, DB820c for both analog and digital playbacks. - Exported q6core functions as suggested by Rohit. - Fixed various issues with dsp carsh/recover usecase. - Added multiple ASM streams as requested in review. Srinivas Kandagatla (25): 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 dt-bindings: sound: qcom: Add bindings for q6afe ASoC: qcom: qdsp6: Add support to Q6AFE dt-bindings: sound: qcom: Add bindings for q6adm ASoC: qcom: qdsp6: Add support to Q6ADM dt-bindings: sound: qcom: Add bindings for q6asm 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 ASoC: qcom: q6afe: add SLIMBus port Support ASoC: qcom: q6afe-dai: add support to slim afe dais ASoC: qcom: q6routing: add support to all SLIMBus Mixers ASoC: qcom: q6afe: add support to MI2S ports ASoC: qcom: q6afe: add support to MI2S sysclks ASoC: qcom: q6afe-dai: add support to 4 MI2S ports ASoC: qcom: q6routing: add support to MI2S Mixers 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 | 83 ++ .../devicetree/bindings/sound/qcom,apq8096.txt | 89 ++ .../devicetree/bindings/sound/qcom,q6adm.txt | 31 + .../devicetree/bindings/sound/qcom,q6afe.txt | 38 + .../devicetree/bindings/sound/qcom,q6asm.txt | 38 + arch/arm64/boot/dts/qcom/apq8096-db820c.dtsi | 44 +- arch/arm64/boot/dts/qcom/msm8996.dtsi | 62 ++ drivers/soc/qcom/Kconfig | 9 + drivers/soc/qcom/Makefile | 1 + drivers/soc/qcom/apr.c | 381 +++++++ include/dt-bindings/soc/qcom,apr.h | 27 + include/dt-bindings/sound/qcom,q6afe.h | 33 + include/dt-bindings/sound/qcom,q6asm.h | 22 + include/linux/mod_devicetable.h | 11 + include/linux/soc/qcom/apr.h | 131 +++ sound/soc/qcom/Kconfig | 42 + sound/soc/qcom/Makefile | 5 + sound/soc/qcom/apq8096.c | 173 ++++ sound/soc/qcom/qdsp6/Makefile | 5 + sound/soc/qcom/qdsp6/q6adm.c | 634 ++++++++++++ sound/soc/qcom/qdsp6/q6adm.h | 29 + sound/soc/qcom/qdsp6/q6afe-dai.c | 645 ++++++++++++ sound/soc/qcom/qdsp6/q6afe.c | 875 ++++++++++++++++ sound/soc/qcom/qdsp6/q6afe.h | 74 ++ sound/soc/qcom/qdsp6/q6asm-dai.c | 621 ++++++++++++ sound/soc/qcom/qdsp6/q6asm.c | 1063 ++++++++++++++++++++ sound/soc/qcom/qdsp6/q6asm.h | 66 ++ sound/soc/qcom/qdsp6/q6core.c | 235 +++++ sound/soc/qcom/qdsp6/q6core.h | 9 + sound/soc/qcom/qdsp6/q6dsp-common.c | 67 ++ sound/soc/qcom/qdsp6/q6dsp-common.h | 24 + sound/soc/qcom/qdsp6/q6dsp-errno.h | 97 ++ sound/soc/qcom/qdsp6/q6routing.c | 758 ++++++++++++++ sound/soc/qcom/qdsp6/q6routing.h | 9 + 34 files changed, 6430 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/soc/qcom/qcom,apr.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,apq8096.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6adm.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6afe.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6asm.txt create mode 100644 drivers/soc/qcom/apr.c create mode 100644 include/dt-bindings/soc/qcom,apr.h create mode 100644 include/dt-bindings/sound/qcom,q6afe.h create mode 100644 include/dt-bindings/sound/qcom,q6asm.h 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/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/q6core.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-common.c create mode 100644 sound/soc/qcom/qdsp6/q6dsp-common.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-errno.h create mode 100644 sound/soc/qcom/qdsp6/q6routing.c create mode 100644 sound/soc/qcom/qdsp6/q6routing.h -- 2.15.1

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[alsa-devel] [PATCH] ASoC: Intel: Boards: Add CNL RT274 I2S machine driver
by Guneshwor Singh 17 May '18

17 May '18

Add CNL I2S machine driver using Realtek RT274 codec in I2S mode configured to ssp0. Signed-off-by: Guneshwor Singh <guneshwor.o.singh(a)intel.com> --- sound/soc/intel/boards/Kconfig | 13 ++ sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/cnl_rt274.c | 250 +++++++++++++++++++++++++++++++++++++ 3 files changed, 265 insertions(+) create mode 100644 sound/soc/intel/boards/cnl_rt274.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 6f754708a48c..29066f0b744f 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -262,4 +262,17 @@ config SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH Say Y if you have such a device. If unsure select "N". +config SND_SOC_INTEL_CNL_RT274_MACH + tristate "ASoC Audio driver for Cannonlake with RT274 I2S mode" + depends on X86 && ACPI && I2C + select SND_SOC_INTEL_SST + depends on SND_SOC_INTEL_SKYLAKE + select SND_SOC_RT274 + select SND_SOC_DMIC + help + This adds support for ASoC machine driver for Cannonlake platform + with RT274 I2S audio codec. + Say Y if you have such a device. + If unsure select "N". + endif diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 69d2dfaeb00c..a7a832c072ca 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -19,6 +19,7 @@ snd-soc-kbl_rt5663_rt5514_max98927-objs := kbl_rt5663_rt5514_max98927.o snd-soc-skl_rt286-objs := skl_rt286.o snd-skl_nau88l25_max98357a-objs := skl_nau88l25_max98357a.o snd-soc-skl_nau88l25_ssm4567-objs := skl_nau88l25_ssm4567.o +snd-soc-cnl-rt274-objs := cnl_rt274.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o @@ -40,3 +41,4 @@ obj-$(CONFIG_SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH) += snd-soc-kbl_rt566 obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH) += snd-skl_nau88l25_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH) += snd-soc-skl_nau88l25_ssm4567.o +obj-$(CONFIG_SND_SOC_INTEL_CNL_RT274_MACH) += snd-soc-cnl-rt274.o diff --git a/sound/soc/intel/boards/cnl_rt274.c b/sound/soc/intel/boards/cnl_rt274.c new file mode 100644 index 000000000000..19458fb50f55 --- /dev/null +++ b/sound/soc/intel/boards/cnl_rt274.c @@ -0,0 +1,250 @@ +/* + * cnl_rt274.c - ASOC Machine driver for CNL + * + * Copyright (C) 2016-17 Intel Corp + * Author: Guneshwor Singh <guneshwor.o.singh(a)intel.com> + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ +#include <linux/module.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include "../../codecs/rt274.h" + +#define CNL_FREQ_OUT 24000000 +#define CNL_BE_FIXUP_RATE 48000 +#define RT274_CODEC_DAI "rt274-aif1" + +static int cnl_rt274_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_dai *codec_dai = + snd_soc_card_get_codec_dai(card, RT274_CODEC_DAI); + int ret, ratio = 100; + + if (!codec_dai) + return -EINVAL; + + /* Codec needs clock for Jack detection and button press */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT274_SCLK_S_PLL2, + CNL_FREQ_OUT, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "set codec sysclk failed: %d\n", ret); + return ret; + } + + if (SND_SOC_DAPM_EVENT_ON(event)) { + ret = snd_soc_dai_set_bclk_ratio(codec_dai, ratio); + if (ret) { + dev_err(codec_dai->dev, + "set bclk ratio failed: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, 0, RT274_PLL2_S_BCLK, + CNL_BE_FIXUP_RATE * ratio, + CNL_FREQ_OUT); + if (ret) { + dev_err(codec_dai->dev, + "enable PLL2 failed: %d\n", ret); + return ret; + } + } + + return 0; +} + +static struct snd_soc_jack cnl_headset; + +/* Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin cnl_headset_pins[] = { + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, +}; + +static const struct snd_kcontrol_new cnl_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Mic Jack"), +}; + +static const struct snd_soc_dapm_widget cnl_rt274_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_MIC("SoC DMIC", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + cnl_rt274_clock_control, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +}; + +static int cnl_dmic_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *channels = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + + if (params_channels(params) == 2) + channels->min = channels->max = 2; + else + channels->min = channels->max = 4; + + return 0; +} + +static const struct snd_soc_dapm_route cnl_map[] = { + {"Headphone Jack", NULL, "HPO Pin"}, + {"MIC", NULL, "Mic Jack"}, + {"DMic", NULL, "SoC DMIC"}, + {"DMIC01 Rx", NULL, "Capture"}, + {"dmic01_hifi", NULL, "DMIC01 Rx"}, + + {"AIF1 Playback", NULL, "ssp0 Tx"}, + {"ssp0 Tx", NULL, "codec1_out"}, + {"ssp0 Tx", NULL, "codec0_out"}, + + {"ssp0 Rx", NULL, "AIF1 Capture"}, + {"codec0_in", NULL, "ssp0 Rx"}, + + {"Headphone Jack", NULL, "Platform Clock"}, + {"Mic Jack", NULL, "Platform Clock"}, +}; + +static int cnl_rt274_init(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_soc_codec *codec = runtime->codec; + struct snd_soc_card *card = runtime->card; + struct snd_soc_dai *codec_dai = runtime->codec_dai; + int ret; + + ret = snd_soc_card_jack_new(runtime->card, "Headset", + SND_JACK_HEADSET, &cnl_headset, + cnl_headset_pins, ARRAY_SIZE(cnl_headset_pins)); + if (ret) + return ret; + + ret = snd_soc_codec_set_jack(codec, &cnl_headset, NULL); + if (ret) + return ret; + + /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xf, 0xf, 4, 24); + if (ret < 0) { + dev_err(runtime->dev, "can't set codec pcm format %d\n", ret); + return ret; + } + + card->dapm.idle_bias_off = true; + + return 0; +} + +static int cnl_be_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + + rate->min = rate->max = CNL_BE_FIXUP_RATE; + channels->min = channels->max = 2; + snd_mask_none(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT)); + snd_mask_set(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), + SNDRV_PCM_FORMAT_S24_LE); + + return 0; +} + +static struct snd_soc_dai_link cnl_rt274_dailink[] = { + { + .name = "SSP0-Codec", + .cpu_dai_name = "SSP0 Pin", + .codec_name = "i2c-INT34C2:00", + .codec_dai_name = "rt274-aif1", + .platform_name = "0000:00:1f.3", + .be_hw_params_fixup = cnl_be_fixup, + .no_pcm = 1, + .ignore_pmdown_time = 1, + .dai_fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, + .dpcm_playback = 1, + .dpcm_capture = 1, + .init = cnl_rt274_init, + }, + { + .name = "dmic01", + .cpu_dai_name = "DMIC01 Pin", + .codec_name = "dmic-codec", + .codec_dai_name = "dmic-hifi", + .platform_name = "0000:00:1f.3", + .be_hw_params_fixup = cnl_dmic_fixup, + .no_pcm = 1, + .ignore_suspend = 1, + .dpcm_capture = 1, + }, +}; + +static int +cnl_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *link) +{ + link->platform_name = "0000:00:1f.3"; + link->nonatomic = 1; + + return 0; +} + +/* SoC card */ +static struct snd_soc_card snd_soc_card_cnl = { + .name = "cnl-audio", + .dai_link = cnl_rt274_dailink, + .num_links = ARRAY_SIZE(cnl_rt274_dailink), + .dapm_widgets = cnl_rt274_widgets, + .num_dapm_widgets = ARRAY_SIZE(cnl_rt274_widgets), + .dapm_routes = cnl_map, + .num_dapm_routes = ARRAY_SIZE(cnl_map), + .controls = cnl_controls, + .num_controls = ARRAY_SIZE(cnl_controls), + .add_dai_link = cnl_add_dai_link, + .fully_routed = true, +}; + +static int snd_cnl_rt274_probe(struct platform_device *pdev) +{ + snd_soc_card_cnl.dev = &pdev->dev; + + return devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cnl); +} + +static struct platform_driver snd_cnl_rt274_driver = { + .driver = { + .name = "cnl_rt274", + .pm = &snd_soc_pm_ops, + }, + .probe = snd_cnl_rt274_probe, +}; + +module_platform_driver(snd_cnl_rt274_driver); + +MODULE_AUTHOR("Guneshwor Singh <guneshwor.o.singh(a)intel.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:cnl_rt274"); -- 2.15.0

4 6

[alsa-devel] [PATCH 0/1] Handling of serializers
by Vishal Thanki 17 May '18

17 May '18

On a custom AM33xx based platform, we use 4 channels for audio output (2 * Left/Right) by making use of two TX serializers. The audio input path only uses 2 channels, hence one RX serializer. If the recording is started before the playback, the playback cannot make use of all 4 channels because of the imposed max number of channels by recording. And if playback is started before recording, the hw_params set during recording stream would disable an active TX serializer which causes audio distortion. I have tried to fix this unwanted behavior in following patch. Please comment if this should be done differently. Vishal Thanki (1): ASoC: davinci-mcasp: Only disable inactive serializer sound/soc/davinci/davinci-mcasp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) -- 2.14.3

3 6

[alsa-devel] [PATCH 43/45] sound/soc/zte: remove duplicate includes
by Pravin Shedge 17 May '18

17 May '18

These duplicate includes have been found with scripts/checkincludes.pl but they have been removed manually to avoid removing false positives. Signed-off-by: Pravin Shedge <pravin.shedge4linux(a)gmail.com> --- sound/soc/zte/zx-i2s.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/zte/zx-i2s.c b/sound/soc/zte/zx-i2s.c index 9a05659..93428e1 100644 --- a/sound/soc/zte/zx-i2s.c +++ b/sound/soc/zte/zx-i2s.c @@ -20,9 +20,6 @@ #include <sound/core.h> #include <sound/dmaengine_pcm.h> #include <sound/initval.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> #define ZX_I2S_PROCESS_CTRL 0x04 #define ZX_I2S_TIMING_CTRL 0x08 -- 2.7.4

2 2

[alsa-devel] KASAN: null-ptr-deref Write in linear_transfer
by syzbot 13 May '18

13 May '18

Hello, syzkaller hit the following crash on 30a7acd573899fd8b8ac39236eff6468b195ac7d git://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/master compiler: gcc (GCC) 7.1.1 20170620 .config is attached Raw console output is attached. C reproducer is attached syzkaller reproducer is attached. See https://goo.gl/kgGztJ for information about syzkaller reproducers IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+a8f5641f452c7e6abf03(a)syzkaller.appspotmail.com It will help syzbot understand when the bug is fixed. See footer for details. If you forward the report, please keep this part and the footer. ================================================================== BUG: KASAN: null-ptr-deref in memcpy include/linux/string.h:344 [inline] BUG: KASAN: null-ptr-deref in do_convert sound/core/oss/linear.c:52 [inline] BUG: KASAN: null-ptr-deref in convert sound/core/oss/linear.c:81 [inline] BUG: KASAN: null-ptr-deref in linear_transfer+0x634/0x900 sound/core/oss/linear.c:110 Write of size 2 at addr (null) by task syzkaller360172/7860 CPU: 0 PID: 7860 Comm: syzkaller360172 Not tainted 4.15.0-rc6+ #155 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:17 [inline] dump_stack+0x194/0x257 lib/dump_stack.c:53 kasan_report_error mm/kasan/report.c:349 [inline] kasan_report+0x13b/0x340 mm/kasan/report.c:409 check_memory_region_inline mm/kasan/kasan.c:260 [inline] check_memory_region+0x137/0x190 mm/kasan/kasan.c:267 memcpy+0x37/0x50 mm/kasan/kasan.c:303 memcpy include/linux/string.h:344 [inline] do_convert sound/core/oss/linear.c:52 [inline] convert sound/core/oss/linear.c:81 [inline] linear_transfer+0x634/0x900 sound/core/oss/linear.c:110 snd_pcm_plug_write_transfer+0x22d/0x420 sound/core/oss/pcm_plugin.c:611 snd_pcm_oss_write2+0x260/0x420 sound/core/oss/pcm_oss.c:1311 snd_pcm_oss_sync1+0x1cc/0x550 sound/core/oss/pcm_oss.c:1530 snd_pcm_oss_sync+0x5b6/0x830 sound/core/oss/pcm_oss.c:1604 snd_pcm_oss_release+0x20b/0x280 sound/core/oss/pcm_oss.c:2431 __fput+0x327/0x7e0 fs/file_table.c:210 ____fput+0x15/0x20 fs/file_table.c:244 task_work_run+0x199/0x270 kernel/task_work.c:113 exit_task_work include/linux/task_work.h:22 [inline] do_exit+0x9bb/0x1ad0 kernel/exit.c:865 do_group_exit+0x149/0x400 kernel/exit.c:968 get_signal+0x73f/0x16c0 kernel/signal.c:2335 do_signal+0x90/0x1eb0 arch/x86/kernel/signal.c:809 exit_to_usermode_loop+0x214/0x310 arch/x86/entry/common.c:158 prepare_exit_to_usermode arch/x86/entry/common.c:195 [inline] syscall_return_slowpath arch/x86/entry/common.c:264 [inline] do_syscall_32_irqs_on arch/x86/entry/common.c:333 [inline] do_fast_syscall_32+0xbfd/0xf9d arch/x86/entry/common.c:389 entry_SYSENTER_compat+0x54/0x63 arch/x86/entry/entry_64_compat.S:129 RIP: 0023:0xf7fcac79 RSP: 002b:00000000f7f8412c EFLAGS: 00000292 ORIG_RAX: 00000000000000f0 RAX: fffffffffffffe00 RBX: 00000000080fd024 RCX: 0000000000000000 RDX: 0000000000000000 RSI: 0000000000000000 RDI: 00000000080fd024 RBP: 0000000000000001 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: 0000000000000000 R13: 0000000000000000 R14: 0000000000000000 R15: 0000000000000000 ================================================================== Kernel panic - not syncing: panic_on_warn set ... CPU: 0 PID: 7860 Comm: syzkaller360172 Tainted: G B 4.15.0-rc6+ #155 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:17 [inline] dump_stack+0x194/0x257 lib/dump_stack.c:53 panic+0x1e4/0x41c kernel/panic.c:183 kasan_end_report+0x50/0x50 mm/kasan/report.c:176 kasan_report_error mm/kasan/report.c:356 [inline] kasan_report+0x144/0x340 mm/kasan/report.c:409 check_memory_region_inline mm/kasan/kasan.c:260 [inline] check_memory_region+0x137/0x190 mm/kasan/kasan.c:267 memcpy+0x37/0x50 mm/kasan/kasan.c:303 memcpy include/linux/string.h:344 [inline] do_convert sound/core/oss/linear.c:52 [inline] convert sound/core/oss/linear.c:81 [inline] linear_transfer+0x634/0x900 sound/core/oss/linear.c:110 snd_pcm_plug_write_transfer+0x22d/0x420 sound/core/oss/pcm_plugin.c:611 snd_pcm_oss_write2+0x260/0x420 sound/core/oss/pcm_oss.c:1311 snd_pcm_oss_sync1+0x1cc/0x550 sound/core/oss/pcm_oss.c:1530 snd_pcm_oss_sync+0x5b6/0x830 sound/core/oss/pcm_oss.c:1604 snd_pcm_oss_release+0x20b/0x280 sound/core/oss/pcm_oss.c:2431 __fput+0x327/0x7e0 fs/file_table.c:210 ____fput+0x15/0x20 fs/file_table.c:244 task_work_run+0x199/0x270 kernel/task_work.c:113 exit_task_work include/linux/task_work.h:22 [inline] do_exit+0x9bb/0x1ad0 kernel/exit.c:865 do_group_exit+0x149/0x400 kernel/exit.c:968 get_signal+0x73f/0x16c0 kernel/signal.c:2335 do_signal+0x90/0x1eb0 arch/x86/kernel/signal.c:809 exit_to_usermode_loop+0x214/0x310 arch/x86/entry/common.c:158 prepare_exit_to_usermode arch/x86/entry/common.c:195 [inline] syscall_return_slowpath arch/x86/entry/common.c:264 [inline] do_syscall_32_irqs_on arch/x86/entry/common.c:333 [inline] do_fast_syscall_32+0xbfd/0xf9d arch/x86/entry/common.c:389 entry_SYSENTER_compat+0x54/0x63 arch/x86/entry/entry_64_compat.S:129 RIP: 0023:0xf7fcac79 RSP: 002b:00000000f7f8412c EFLAGS: 00000292 ORIG_RAX: 00000000000000f0 RAX: fffffffffffffe00 RBX: 00000000080fd024 RCX: 0000000000000000 RDX: 0000000000000000 RSI: 0000000000000000 RDI: 00000000080fd024 RBP: 0000000000000001 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: 0000000000000000 R13: 0000000000000000 R14: 0000000000000000 R15: 0000000000000000 Dumping ftrace buffer: (ftrace buffer empty) Kernel Offset: disabled Rebooting in 86400 seconds.. --- This bug is generated by a dumb bot. It may contain errors. See https://goo.gl/tpsmEJ for details. Direct all questions to syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. If you forgot to add the Reported-by tag, once the fix for this bug is merged into any tree, please reply to this email with: #syz fix: exact-commit-title If you want to test a patch for this bug, please reply with: #syz test: git://repo/address.git branch and provide the patch inline or as an attachment. To mark this as a duplicate of another syzbot report, please reply with: #syz dup: exact-subject-of-another-report If it's a one-off invalid bug report, please reply with: #syz invalid Note: if the crash happens again, it will cause creation of a new bug report. Note: all commands must start from beginning of the line in the email body.

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[alsa-devel] Prepare to covert Codec/Platform into Component
by Kuninori Morimoto 09 May '18

09 May '18

Hi Mark, Lars-Peter I will post all prepare patches to convert Codec/Platform into Component. It has 3 steps. 1) add new necessary functions 2) remove/don't use codec hw_write/read 3) remove/don't use codec reg_cache If these are accepted, all prepares were completed. Next, I can post "convert Codec/Platform into Component" patch bomb. These are too many convert patches, to avoid review issue, I will post few of them as sample. 4) sample of don't use rtd->codec 5) sample of convert Platform into Component 6) sample of convert Codec into Component I will post some of "remove" patches too for each steps. I will re-post all "remove" patches if all are accepted. For your review, please consider below. If these are not good, all covert patch doesn't work. 1. use devm_snd_soc_register_component() and remove snd_soc_unregister_component() My patches is using devm_ version function, and removed unregister function. This is not a big deal 2. New flags. Because "Codec" and "CPU/Platform" are using different features, we need to use some new flags. .idle_bias_off/on .ignore_pmdown_time/pmdown_time These are because "CPU/Platform" and "Codec" are using different default value. "Codec" side convert patches are using these. .endianness .non_legacy_dai_naming These are because "Codec" only needs fixup format, and not use "legacy" DAI name. "Codec" side convert patches are using these I added its flags info on each patch. 3. rtd->platform rtd->platform will be removed. Now we can find all rtd connected component by using rtdcom_lookup. rtd->platform will be replaced to snd_soc_rtdcom_lookup(rtd, DRV_NAME); Here, DRV_NAME is platform's component driver name. 4. rtd->codec rtd->codec will be removed. In soc_bind_dai_link(), It is rtd->codec = rtd->codec_dai->codec; And now, DAI has codec and component pointer. So, rtd->codec will be replaced into rtd->codec_dai->component. Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH] ASoC: soc-core: remove legacy_dai_naming from snd_soc_register_dais()
by Kuninori Morimoto 09 May '18

09 May '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> We can get legacy dai name flag from component driver. Thus, there is no need to have its parameter on snd_soc_register_dais(). Let's remove unneeded parameter Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1faebc04..9c65477 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3146,8 +3146,7 @@ static struct snd_soc_dai *soc_add_dai(struct snd_soc_component *component, * parent's name. */ static int snd_soc_register_dais(struct snd_soc_component *component, - struct snd_soc_dai_driver *dai_drv, size_t count, - bool legacy_dai_naming) + struct snd_soc_dai_driver *dai_drv, size_t count) { struct device *dev = component->dev; struct snd_soc_dai *dai; @@ -3159,7 +3158,7 @@ static int snd_soc_register_dais(struct snd_soc_component *component, for (i = 0; i < count; i++) { dai = soc_add_dai(component, dai_drv + i, - count == 1 && legacy_dai_naming); + count == 1 && !component->driver->non_legacy_dai_naming); if (dai == NULL) { ret = -ENOMEM; goto err; @@ -3440,8 +3439,7 @@ int snd_soc_add_component(struct device *dev, } } - ret = snd_soc_register_dais(component, dai_drv, num_dai, - !component_driver->non_legacy_dai_naming); + ret = snd_soc_register_dais(component, dai_drv, num_dai); if (ret < 0) { dev_err(dev, "ASoC: Failed to register DAIs: %d\n", ret); goto err_cleanup; -- 1.9.1

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[alsa-devel] [FIX][PATCH] ASoC: topology: Fix logical inversion in set_link_hw_format()
by Xiuli Pan 16 Apr '18

16 Apr '18

From: Pan Xiuli <xiuli.pan(a)linux.intel.com> The master/slave conventions are wrt. the codec. The topology code contains a logical inversion, fix to follow ASoC usage. Signed-off-by: Pan Xiuli <xiuli.pan(a)linux.intel.com> --- sound/soc/soc-topology.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 561f74d..2931248 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -2030,11 +2030,11 @@ static void set_link_hw_format(struct snd_soc_dai_link *link, /* clock masters */ bclk_master = hw_config->bclk_master; fsync_master = hw_config->fsync_master; - if (!bclk_master && !fsync_master) + if (bclk_master && fsync_master) link->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; - else if (bclk_master && !fsync_master) - link->dai_fmt |= SND_SOC_DAIFMT_CBS_CFM; else if (!bclk_master && fsync_master) + link->dai_fmt |= SND_SOC_DAIFMT_CBS_CFM; + else if (bclk_master && !fsync_master) link->dai_fmt |= SND_SOC_DAIFMT_CBM_CFS; else link->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; -- 2.7.4

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[alsa-devel] [PATCH] ASoC: Intel: atom: fix ACPI/PCI Kconfig
by Pierre-Louis Bossart 12 Apr '18

12 Apr '18

The split between ACPI and PCI platforms generated issues with randconfig: with SND_SST_ATOM_HIFI2_PLATFORM_PCI=y and SND_SST_ATOM_HIFI2_PLATFORM=m, we get this module link failure: ERROR: "sst_context_init" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! ERROR: "sst_context_cleanup" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! ERROR: "sst_alloc_drv_context" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! ERROR: "intel_sst_pm" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! ERROR: "sst_configure_runtime_pm" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! To keep things simple, let's expose two configs for SND_SST_ATOM_HIFI2_PLATFORM_PCI and SND_SST_ATOM_HIFI2_PLATFORM_ACPI, which select a common SND_SST_ATOM_HIFI2_PLATFORM option. To avoid breaking existing solutions with the semantics change, SND_SST_ATOM_HIFI2_PLATFORM_ACPI uses "default ACPI" so that "make oldnoconfig" and "make olddefconfig" still work as expected. Also remove mentions of Medfield while we are at it since it was removed recently. Reported-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: 4772c16ede52 ("ASoC: Intel: Kconfig: Simplify-clarify ACPI/PCI dependencies") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/Kconfig | 22 +++++++++++++--------- 1 file changed, 13 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index f2c9e8c5970a..d7ac77b0197b 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -72,24 +72,28 @@ config SND_SOC_INTEL_BAYTRAIL for Baytrail Chromebooks but this option is now deprecated and is not recommended, use SND_SST_ATOM_HIFI2_PLATFORM instead. +config SND_SST_ATOM_HIFI2_PLATFORM + tristate + select SND_SOC_COMPRESS + config SND_SST_ATOM_HIFI2_PLATFORM_PCI - tristate "PCI HiFi2 (Medfield, Merrifield) Platforms" + tristate "PCI HiFi2 (Merrifield) Platforms" depends on X86 && PCI select SND_SST_IPC_PCI - select SND_SOC_COMPRESS + select SND_SST_ATOM_HIFI2_PLATFORM help - If you have a Intel Medfield or Merrifield/Edison platform, then + If you have a Intel Merrifield/Edison platform, then enable this option by saying Y or m. Distros will typically not - enable this option: Medfield devices are not available to - developers and while Merrifield/Edison can run a mainline kernel with - limited functionality it will require a firmware file which - is not in the standard firmware tree + enable this option: while Merrifield/Edison can run a mainline + kernel with limited functionality it will require a firmware file + which is not in the standard firmware tree -config SND_SST_ATOM_HIFI2_PLATFORM +config SND_SST_ATOM_HIFI2_PLATFORM_ACPI tristate "ACPI HiFi2 (Baytrail, Cherrytrail) Platforms" + default ACPI depends on X86 && ACPI select SND_SST_IPC_ACPI - select SND_SOC_COMPRESS + select SND_SST_ATOM_HIFI2_PLATFORM select SND_SOC_ACPI_INTEL_MATCH select IOSF_MBI help -- 2.14.1

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