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

[alsa-devel] [PATCH v6 00/24] 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> Thankyou everyone for providing feedback and testing v5 patchset. 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. Patches are also tested on SDM845 by Rohit. Here is my test branch incase someone want to try these patches https://git.linaro.org/people/srinivas.kandagatla/linux.git/log/?h=v4.17-qd… Here is block diagram to give a quick overview of the components +---------+ +---------+ +---------+ | q6asm | |q6routing| | q6afe | | dais | <------> | mixers | <-----> | dais | +---------+ +---------+ +---------+ ^ ^ ^ | | | | +------------------+----------------+ | | | | | | 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 v5 (https://lkml.org/lkml/2018/4/18/553) - updated bindings as reviewed by Rob H. - added missing binding for q6core - fixed copp double count reported by Amit Pundir - Updated proper SPDX licence on headers - reorder patches to fix bisect errors reported by kbuild Srinivas Kandagatla (24): soc: qcom dt-bindings: Add APR bus bindings soc: qcom: Add APR bus driver ASoC: qdsp6: dt-bindings: Add q6core dt bindings ASoC: qdsp6: dt-bindings: Add q6afe dt bindings ASoC: qdsp6: dt-bindings: Add q6adm dt bindings ASoC: qdsp6: dt-bindings: Add q6asm dt bindings ASoC: qdsp6: q6common: Add qdsp6 helper functions ASoC: qdsp6: q6core: Add q6core driver ASoC: qdsp6: q6afe: Add q6afe driver ASoC: qdsp6: qdafe: Add SLIMBus port Support ASoC: qdsp6: q6afe: Add support to MI2S ports ASoC: qdsp6: q6afe: Add support to MI2S sysclks ASoC: qdsp6: q6adm: Add q6adm driver ASoC: qdsp6: q6asm: Add q6asm driver ASoC: qdsp6: q6asm: Add support to memory map and unmap ASoC: qdsp6: q6asm: Add support to audio stream apis ASoC: qdsp6: q6routing: Add q6routing driver ASoC: qdsp6: q6routing: Add support to all SLIMBus Mixers ASoC: qdsp6: q6routing: Add support to MI2S Mixers ASoC: qdsp6: q6afe: Add q6afe dai driver ASoC: qdsp6: q6asm: Add q6asm dai driver ASoC: qdsp6: dt-bindings: Add apq8096 machine bindings ASoC: qcom: apq8096: Add db820c machine driver MAINTAINERS: Add myself as co-maintainer of qcom audio .../devicetree/bindings/soc/qcom/qcom,apr.txt | 84 ++ .../devicetree/bindings/sound/qcom,apq8096.txt | 109 ++ .../devicetree/bindings/sound/qcom,q6adm.txt | 33 + .../devicetree/bindings/sound/qcom,q6afe.txt | 88 ++ .../devicetree/bindings/sound/qcom,q6asm.txt | 33 + .../devicetree/bindings/sound/qcom,q6core.txt | 21 + MAINTAINERS | 1 + drivers/soc/qcom/Kconfig | 9 + drivers/soc/qcom/Makefile | 1 + drivers/soc/qcom/apr.c | 384 ++++++ include/dt-bindings/soc/qcom,apr.h | 28 + include/dt-bindings/sound/qcom,q6afe.h | 31 + include/dt-bindings/sound/qcom,q6asm.h | 22 + include/linux/mod_devicetable.h | 11 + include/linux/soc/qcom/apr.h | 130 ++ sound/soc/qcom/Kconfig | 50 + sound/soc/qcom/Makefile | 5 + sound/soc/qcom/apq8096.c | 238 ++++ sound/soc/qcom/qdsp6/Makefile | 8 + sound/soc/qcom/qdsp6/q6adm.c | 635 ++++++++++ sound/soc/qcom/qdsp6/q6adm.h | 25 + sound/soc/qcom/qdsp6/q6afe-dai.c | 752 +++++++++++ sound/soc/qcom/qdsp6/q6afe.c | 1071 ++++++++++++++++ sound/soc/qcom/qdsp6/q6afe.h | 193 +++ sound/soc/qcom/qdsp6/q6asm-dai.c | 632 ++++++++++ sound/soc/qcom/qdsp6/q6asm.c | 1312 ++++++++++++++++++++ sound/soc/qcom/qdsp6/q6asm.h | 69 + sound/soc/qcom/qdsp6/q6core.c | 380 ++++++ sound/soc/qcom/qdsp6/q6core.h | 15 + sound/soc/qcom/qdsp6/q6dsp-common.c | 66 + sound/soc/qcom/qdsp6/q6dsp-common.h | 24 + sound/soc/qcom/qdsp6/q6dsp-errno.h | 51 + sound/soc/qcom/qdsp6/q6routing.c | 982 +++++++++++++++ sound/soc/qcom/qdsp6/q6routing.h | 9 + 34 files changed, 7502 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 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 Documentation/devicetree/bindings/sound/qcom,q6core.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 -- 2.16.2

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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

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[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

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[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] [PATCH 0/4] ALSA: usb: UAC3 new features.
by Jorge Sanjuan 15 May '18

15 May '18

Now that the UAC3 patch [1] has made it to linux-next I have some extra features to make a UAC3 device fully work in Linux. Including Jack insertion control that I have put on top of this other patch [2] for UAC2. Also adding support for the UAC3 Mixer Unit which is most likely to appear in most headset type devices. UAC3 devices also require to have a Basic Audio Device (BADD) in a separate config for which both Ruslan Bilovol and myself have submited different approaches[3][4] but I don't know what the final merge will be. Once there is official support for BADD, we'll need to test it with an actual UAC3 device to confirm it all wokrs. All this features are tested with an actual UAC3 device that is still in development. For this patch series, only the legacy config (#1. UAC1/UAC2) and the UAC3 config have been tested. The BADD config is only tested using and updated verison of [4]. [1]: https://patchwork.kernel.org/patch/10298179/ [2]: https://patchwork.kernel.org/patch/10305847/ [3]: https://patchwork.kernel.org/patch/10340851/ [4]: https://www.spinics.net/lists/alsa-devel/msg71617.html Based on linux-next tag: next-20180420 Jorge Sanjuan (3): ALSA: usb-audio: UAC3. Add support for mixer unit. ALSA: usb-audio: Use Class Specific EP for UAC3 devices. ALSA: usb-audio: UAC3 Add support for connector insertion. Michael Drake (1): ALSA: usb-audio: ADC3: Fix channel mapping conversion for ADC3. include/linux/usb/audio-v2.h | 7 ++ include/linux/usb/audio-v3.h | 14 ++++ include/uapi/linux/usb/audio.h | 13 ++- sound/usb/mixer.c | 175 ++++++++++++++++++++++++++++++++++++----- sound/usb/stream.c | 19 ++++- 5 files changed, 204 insertions(+), 24 deletions(-) -- 2.11.0

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[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] [PATCH v6 00/13] soundwire: Add stream support
by Vinod Koul 11 May '18

11 May '18

This series adds support in SoundWire subsystem for: - Documentation for stream support - stream management - data port management - DAI ops in cadence and Intel drivers - ASoC API to propagate SDW stream Updates in v6: - Fix return error as pointed by Sanyog - Update stream state enum documentation as pointed by Pierre and few other nits Updated in v5: - Add todo for device-device changes in future and some comment updates based on feedback from Pierre Updates in v4: - Add more comments for sdw_release_master_stream() - Make sure ports are freed upon sdw_stream_remove_slave() - Clarify snd_soc_dai_set_sdw_stream() and setting of NULL upon release. Updates in v3: - Remove stream state transition checks and redefine stream states - Update bansk switch routines - Other nitpicks pointed out by Pierre. Updates in v2: - Make ASoC API inlined - Make stream states as states and not action - Update the direction enum - Fix some typos and comment updates *** SUBJECT HERE *** *** BLURB HERE *** Sanyog Kale (7): Documentation: soundwire: Add more documentation soundwire: Add support for SoundWire stream management soundwire: Add support for port management soundwire: Add Master and Slave port programming soundwire: Add helpers for ports operations soundwire: Add bank switch routine soundwire: Add stream configuration APIs Shreyas NC (2): ASoC: Add SoundWire stream programming interface soundwire: Remove cdns_master_ops Vinod Koul (4): soundwire: cdns: Add port routines soundwire: cdns: Add stream routines soundwire: intel: Add stream initialization soundwire: intel: Add audio DAI ops .../driver-api/soundwire/error_handling.rst | 65 + Documentation/driver-api/soundwire/index.rst | 3 + Documentation/driver-api/soundwire/locking.rst | 106 ++ Documentation/driver-api/soundwire/stream.rst | 372 +++++ drivers/soundwire/Kconfig | 2 +- drivers/soundwire/Makefile | 2 +- drivers/soundwire/bus.c | 43 + drivers/soundwire/bus.h | 72 + drivers/soundwire/cadence_master.c | 451 +++++- drivers/soundwire/cadence_master.h | 151 ++ drivers/soundwire/intel.c | 525 ++++++- drivers/soundwire/intel.h | 4 + drivers/soundwire/intel_init.c | 3 + drivers/soundwire/stream.c | 1479 ++++++++++++++++++++ include/linux/soundwire/sdw.h | 332 ++++- include/linux/soundwire/sdw_intel.h | 14 + include/sound/soc-dai.h | 23 + 17 files changed, 3633 insertions(+), 14 deletions(-) create mode 100644 Documentation/driver-api/soundwire/error_handling.rst create mode 100644 Documentation/driver-api/soundwire/locking.rst create mode 100644 Documentation/driver-api/soundwire/stream.rst create mode 100644 drivers/soundwire/stream.c -- 2.7.4

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[alsa-devel] [PATCH 1/4] ASoC: compress: Only assign compr->ops->copy once
by Charles Keepax 11 May '18

11 May '18

There are only one set of ops on the compressed stream so no need to reassign the copy callback repeatedly, stop after copy is seen to be necessary. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/soc-compress.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index ba56f87f96d4c..62875c6a93a14 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -973,6 +973,7 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) continue; compr->ops->copy = soc_compr_copy; + break; } -- 2.11.0

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[alsa-devel] [PATCH 1/2] ASoC: rt5645: Add platform-data for Lenovo Ideapad Mixx 310
by Hans de Goede 11 May '18

11 May '18

The Lenovo Ideapad Mixx 310 has a differential internal analog mic, add platform-data for this. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/codecs/rt5645.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index bc8d829ce45b..70f351c61d41 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3652,6 +3652,11 @@ static const struct rt5645_platform_data asus_t100ha_platform_data = { .inv_jd1_1 = true, }; +static const struct rt5645_platform_data lenovo_ideapad_miix_310_pdata = { + .jd_mode = 3, + .in2_diff = true, +}; + static const struct rt5645_platform_data jd_mode3_platform_data = { .jd_mode = 3, }; @@ -3735,6 +3740,15 @@ static const struct dmi_system_id dmi_platform_data[] = { }, .driver_data = (void *)&jd_mode3_platform_data, }, + { + .ident = "Lenovo Ideapad Miix 310", + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "LENOVO"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "80SG"), + DMI_EXACT_MATCH(DMI_PRODUCT_VERSION, "MIIX 310-10ICR"), + }, + .driver_data = (void *)&lenovo_ideapad_miix_310_pdata, + }, { } }; -- 2.17.0

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[alsa-devel] [PATCH 0/5] ASoC: cirrus: Series of fixes for ep93xx-i2s
by Alexander Sverdlin 11 May '18

11 May '18

This series is a result of long-running debugging of rarely screwed I2S stream out of EP9302 chip. Logic analyser shows that the stream is shifted by one byte (out of 32 bits per sample) which leads to noise intead of original stream. DMA is verified to work fine (no underrun). The issue is only reproducible under high load (CPU or AMBA bus -- not yet clear). This stream corruption happens at the same time I2S controller reports internal FIFO underrun (even though according to documentation, it should handle an underrun gracefully). First two patches are fixing unrelated issues in I2S driver found during debugging of the stream shift. Third one is simplification of the driver to prepare for workaround. Both controller configuration options for TX FIFO underrun handling were tested, both do not work reliable. Hence, this ugly watchdog in the last patch. The whole series is tested on a board similar to EDB9302 (identical in sound part). Alexander Sverdlin (5): ASoC: cirrus: i2s: Fix LRCLK configuration ASoC: cirrus: i2s: Fix {TX|RX}LinCtrlData setup ASoC: cirrus: i2s: Stop enabling I2S2 and I2S3 FIFOs ARM: ep93xx: i2s: Add IRQ to platform device resources ASoC: cirrus: i2s: IRQ-based stream watchdog arch/arm/mach-ep93xx/core.c | 1 + sound/soc/cirrus/Kconfig | 17 +++++++ sound/soc/cirrus/edb93xx.c | 2 +- sound/soc/cirrus/ep93xx-i2s.c | 106 +++++++++++++++++++++++++++++++++-------- sound/soc/cirrus/snappercl15.c | 2 +- 5 files changed, 106 insertions(+), 22 deletions(-) -- 2.16.2

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