June 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH v3] ASoC: Intel: kbl_rt5663_rt5514_max98927: Split be_hw_params_fixup function
by Lukasz Majczak 16 Jul '20

16 Jul '20

Split be_hw_params_fixup function for different codecs as current common function, leads to crash while trying to get snd_soc_dpcm with container_of() macro in kabylake_ssp_fixup(). The crash call path looks as below: soc_pcm_hw_params() snd_soc_dai_hw_params(codec_dai, substream, &codec_params); rtd->dai_link->be_hw_params_fixup(rtd, params) kabylake_ssp_fixup() In this case, codec_params is just a copy of an internal structure and is not embedded into struct snd_soc_dpcm thus we cannot use container_of() on it. v1 -> v2: - Extract dmic from SSP0 as every BE should have own fixup function. v2 -> v3: - Restore naming in the dapm route table to not confuse with other drivers - Fixed indentations Signed-off-by: Lukasz Majczak <lma(a)semihalf.com> --- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 130 +++++++++++------- 1 file changed, 84 insertions(+), 46 deletions(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 1b1f8d7a4ea3..b9596cf3a6c4 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -328,42 +328,51 @@ static const struct snd_soc_ops kabylake_rt5663_fe_ops = { .startup = kbl_fe_startup, }; -static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) +static void kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params, snd_pcm_format_t pcm_fmt) { struct snd_interval *rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *chan = hw_param_interval(params, + struct snd_interval *channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - struct snd_soc_dpcm *dpcm = container_of( - params, struct snd_soc_dpcm, hw_params); - struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link; - struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link; /* * The ADSP will convert the FE rate to 48k, stereo, 24 bit */ - if (!strcmp(fe_dai_link->name, "Kbl Audio Port") || - !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") || - !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) { - rate->min = rate->max = 48000; - chan->min = chan->max = 2; - snd_mask_none(fmt); - snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); - } else if (!strcmp(fe_dai_link->name, "Kbl Audio DMIC cap")) { - if (params_channels(params) == 2 || - DMIC_CH(dmic_constraints) == 2) - chan->min = chan->max = 2; - else - chan->min = chan->max = 4; - } - /* - * The speaker on the SSP0 supports S16_LE and not S24_LE. - * thus changing the mask here - */ - if (!strcmp(be_dai_link->name, "SSP0-Codec")) - snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); + + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + snd_mask_none(fmt); + snd_mask_set_format(fmt, pcm_fmt); +} + +static int kabylake_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + kabylake_ssp_fixup(rtd, params, SNDRV_PCM_FORMAT_S16_LE); + return 0; +} + +static int kabylake_ssp1_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + + kabylake_ssp_fixup(rtd, params, SNDRV_PCM_FORMAT_S24_LE); + return 0; +} + +static int kabylake_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 || DMIC_CH(dmic_constraints) == 2) + channels->min = channels->max = 2; + else + channels->min = channels->max = 4; return 0; } @@ -400,20 +409,6 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, int ret = 0, j; for_each_rtd_codec_dais(rtd, j, codec_dai) { - if (!strcmp(codec_dai->component->name, RT5514_DEV_NAME)) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0, 8, 16); - if (ret < 0) { - dev_err(rtd->dev, "set TDM slot err:%d\n", ret); - return ret; - } - - ret = snd_soc_dai_set_sysclk(codec_dai, - RT5514_SCLK_S_MCLK, 24576000, SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(rtd->dev, "set sysclk err: %d\n", ret); - return ret; - } - } if (!strcmp(codec_dai->component->name, MAXIM_DEV0_NAME)) { ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x30, 3, 8, 16); if (ret < 0) { @@ -433,10 +428,36 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, return ret; } +static int kabylake_dmic01_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int ret = 0; + + ret = snd_soc_dai_set_tdm_slot(rtd->codec_dai, 0xF, 0, 8, 16); + if (ret < 0) { + dev_err(rtd->dev, "set TDM slot err:%d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(rtd->codec_dai, + RT5514_SCLK_S_MCLK, 24576000, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "set sysclk err: %d\n", ret); + return ret; + } + + return ret; +} + static struct snd_soc_ops kabylake_ssp0_ops = { .hw_params = kabylake_ssp0_hw_params, }; +static struct snd_soc_ops kabylake_dmic01_ops = { + .hw_params = kabylake_dmic01_hw_params, +}; + static const unsigned int channels_dmic[] = { 4, }; @@ -507,14 +528,19 @@ SND_SOC_DAILINK_DEF(ssp0_pin, SND_SOC_DAILINK_DEF(ssp0_codec, DAILINK_COMP_ARRAY( /* Left */ COMP_CODEC(MAXIM_DEV0_NAME, KBL_MAXIM_CODEC_DAI), - /* Right */COMP_CODEC(MAXIM_DEV1_NAME, KBL_MAXIM_CODEC_DAI), - /* dmic */ COMP_CODEC(RT5514_DEV_NAME, KBL_REALTEK_DMIC_CODEC_DAI))); + /* Right */COMP_CODEC(MAXIM_DEV1_NAME, KBL_MAXIM_CODEC_DAI))); SND_SOC_DAILINK_DEF(ssp1_pin, DAILINK_COMP_ARRAY(COMP_CPU("SSP1 Pin"))); SND_SOC_DAILINK_DEF(ssp1_codec, DAILINK_COMP_ARRAY(COMP_CODEC(RT5663_DEV_NAME, KBL_REALTEK_CODEC_DAI))); +SND_SOC_DAILINK_DEF(dmic01_pin, + DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin"))); +SND_SOC_DAILINK_DEF(dmic01_codec, + DAILINK_COMP_ARRAY( + COMP_CODEC(RT5514_DEV_NAME, KBL_REALTEK_DMIC_CODEC_DAI))); + SND_SOC_DAILINK_DEF(idisp1_pin, DAILINK_COMP_ARRAY(COMP_CPU("iDisp1 Pin"))); SND_SOC_DAILINK_DEF(idisp1_codec, @@ -618,9 +644,8 @@ static struct snd_soc_dai_link kabylake_dais[] = { SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .ignore_pmdown_time = 1, - .be_hw_params_fixup = kabylake_ssp_fixup, + .be_hw_params_fixup = kabylake_ssp0_fixup, .dpcm_playback = 1, - .dpcm_capture = 1, .ops = &kabylake_ssp0_ops, SND_SOC_DAILINK_REG(ssp0_pin, ssp0_codec, platform), }, @@ -632,12 +657,25 @@ static struct snd_soc_dai_link kabylake_dais[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .ignore_pmdown_time = 1, - .be_hw_params_fixup = kabylake_ssp_fixup, + .be_hw_params_fixup = kabylake_ssp1_fixup, .ops = &kabylake_rt5663_ops, .dpcm_playback = 1, .dpcm_capture = 1, SND_SOC_DAILINK_REG(ssp1_pin, ssp1_codec, platform), }, + { + .name = "dmic01", + .id = 2, + .no_pcm = 1, + .dai_fmt = SND_SOC_DAIFMT_DSP_B | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = kabylake_dmic_fixup, + .dpcm_capture = 1, + .ops = &kabylake_dmic01_ops, + SND_SOC_DAILINK_REG(dmic01_pin, dmic01_codec, platform), + }, { .name = "iDisp1", .id = 3, base-commit: a4f6fc98cd2fa1774bcaeb248c67156ef9402a56 -- 2.25.1

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[PATCH 0/9] soundwire: intel: revisit SHIM programming
by Bard Liao 15 Jul '20

15 Jul '20

This series does some cleanup, revisits SHIM programming sequences, and merges Soundwire interrupt handlers/threads. Bard Liao (2): soundwire: intel/cadence: merge Soundwire interrupt handlers/threads Soundwire: intel_init: save Slave(s) _ADR info in sdw_intel_ctx Pierre-Louis Bossart (6): soundwire: intel: reuse code for wait loops to set/clear bits soundwire: intel: revisit SHIM programming sequences. soundwire: intel: introduce a helper to arm link synchronization soundwire: intel: introduce helper for link synchronization soundwire: intel_init: add implementation of sdw_intel_enable_irq() soundwire: intel_init: use EXPORT_SYMBOL_NS Rander Wang (1): soundwire: intel: add wake interrupt support drivers/soundwire/cadence_master.c | 18 +- drivers/soundwire/cadence_master.h | 4 + drivers/soundwire/intel.c | 389 ++++++++++++++++++++++------ drivers/soundwire/intel.h | 9 + drivers/soundwire/intel_init.c | 101 +++++++- include/linux/soundwire/sdw_intel.h | 2 + 6 files changed, 425 insertions(+), 98 deletions(-) -- 2.17.1

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[PATCH v2 0/5] soundwire: handle stream at the dailink level
by Bard Liao 15 Jul '20

15 Jul '20

Currently, stream is handled at the dai level. But we have to handle stream at the dailink level in the multi-cpu dailink usage. changes in v2: - Add kernel doc - Use single line trace log Pierre-Louis Bossart (5): soundwire: intel: implement get_sdw_stream() operations soundwire: stream: add helper to startup/shutdown streams soundwire: intel: remove stream allocation/free soundwire: cadence: allocate/free dma_data in set_sdw_stream soundwire: intel: don't free dma_data in DAI shutdown Documentation/driver-api/soundwire/stream.rst | 11 ++- drivers/soundwire/cadence_master.c | 52 +++++++--- drivers/soundwire/intel.c | 82 ++++------------ drivers/soundwire/stream.c | 98 +++++++++++++++++++ include/linux/soundwire/sdw.h | 2 + 5 files changed, 167 insertions(+), 78 deletions(-) -- 2.17.1

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[PATCH v2] dt-bindings: ASoC: Convert UniPhier EVEA codec to json-schema
by Masahiro Yamada 14 Jul '20

14 Jul '20

Convert the UniPhier EVEA sound codec binding to DT schema format. Signed-off-by: Masahiro Yamada <yamada.masahiro(a)socionext.com> --- Changes in v2: - Add schema for subnode 'port' .../sound/socionext,uniphier-evea.yaml | 70 +++++++++++++++++++ .../bindings/sound/uniphier,evea.txt | 26 ------- 2 files changed, 70 insertions(+), 26 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/socionext,uniphier-evea.yaml delete mode 100644 Documentation/devicetree/bindings/sound/uniphier,evea.txt diff --git a/Documentation/devicetree/bindings/sound/socionext,uniphier-evea.yaml b/Documentation/devicetree/bindings/sound/socionext,uniphier-evea.yaml new file mode 100644 index 000000000000..228168f685cf --- /dev/null +++ b/Documentation/devicetree/bindings/sound/socionext,uniphier-evea.yaml @@ -0,0 +1,70 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/socionext,uniphier-evea.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: UniPhier EVEA SoC-internal sound codec + +maintainers: + - <alsa-devel(a)alsa-project.org> + +properties: + compatible: + const: socionext,uniphier-evea + + reg: + maxItems: 1 + + clock-names: + items: + - const: evea + - const: exiv + + clocks: + minItems: 2 + maxItems: 2 + + reset-names: + items: + - const: evea + - const: exiv + - const: adamv + + resets: + minItems: 3 + maxItems: 3 + + "#sound-dai-cells": + const: 1 + +patternProperties: + "^port@[0-9]$": + type: object + properties: + endpoint: true + required: + - endpoint + +additionalProperties: false + +required: + - compatible + - reg + - clock-names + - clocks + - reset-names + - resets + - "#sound-dai-cells" + +examples: + - | + codec@57900000 { + compatible = "socionext,uniphier-evea"; + reg = <0x57900000 0x1000>; + clock-names = "evea", "exiv"; + clocks = <&sys_clk 41>, <&sys_clk 42>; + reset-names = "evea", "exiv", "adamv"; + resets = <&sys_rst 41>, <&sys_rst 42>, <&adamv_rst 0>; + #sound-dai-cells = <1>; + }; diff --git a/Documentation/devicetree/bindings/sound/uniphier,evea.txt b/Documentation/devicetree/bindings/sound/uniphier,evea.txt deleted file mode 100644 index 3f31b235f18b..000000000000 --- a/Documentation/devicetree/bindings/sound/uniphier,evea.txt +++ /dev/null @@ -1,26 +0,0 @@ -Socionext EVEA - UniPhier SoC internal codec driver - -Required properties: -- compatible : should be "socionext,uniphier-evea". -- reg : offset and length of the register set for the device. -- clock-names : should include following entries: - "evea", "exiv" -- clocks : a list of phandle, should contain an entry for each - entries in clock-names. -- reset-names : should include following entries: - "evea", "exiv", "adamv" -- resets : a list of phandle, should contain reset entries of - reset-names. -- #sound-dai-cells: should be 1. - -Example: - - codec { - compatible = "socionext,uniphier-evea"; - reg = <0x57900000 0x1000>; - clock-names = "evea", "exiv"; - clocks = <&sys_clk 41>, <&sys_clk 42>; - reset-names = "evea", "exiv", "adamv"; - resets = <&sys_rst 41>, <&sys_rst 42>, <&adamv_rst 0>; - #sound-dai-cells = <1>; - }; -- 2.25.1

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[PATCH v2] dt-bindings: ASoC: Convert UniPhier AIO audio system to json-schema
by Masahiro Yamada 14 Jul '20

14 Jul '20

Convert the UniPhier AIO audio system binding to DT schema format. Signed-off-by: Masahiro Yamada <yamada.masahiro(a)socionext.com> --- Changes in v2: - Add schema for subnode 'port' .../sound/socionext,uniphier-aio.yaml | 81 +++++++++++++++++++ .../bindings/sound/uniphier,aio.txt | 45 ----------- 2 files changed, 81 insertions(+), 45 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/socionext,uniphier-aio.yaml delete mode 100644 Documentation/devicetree/bindings/sound/uniphier,aio.txt diff --git a/Documentation/devicetree/bindings/sound/socionext,uniphier-aio.yaml b/Documentation/devicetree/bindings/sound/socionext,uniphier-aio.yaml new file mode 100644 index 000000000000..4987eb91f2ab --- /dev/null +++ b/Documentation/devicetree/bindings/sound/socionext,uniphier-aio.yaml @@ -0,0 +1,81 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/socionext,uniphier-aio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: UniPhier AIO audio system + +maintainers: + - <alsa-devel(a)alsa-project.org> + +properties: + compatible: + enum: + - socionext,uniphier-ld11-aio + - socionext,uniphier-ld20-aio + - socionext,uniphier-pxs2-aio + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clock-names: + const: aio + + clocks: + maxItems: 1 + + reset-names: + const: aio + + resets: + maxItems: 1 + + socionext,syscon: + description: | + Specifies a phandle to soc-glue, which is used for changing mode of S/PDIF + signal pin to output from Hi-Z. This property is optional if you use I2S + signal pins only. + $ref: "/schemas/types.yaml#/definitions/phandle" + + "#sound-dai-cells": + const: 1 + +patternProperties: + "^port@[0-9]$": + type: object + properties: + endpoint: true + required: + - endpoint + +additionalProperties: false + +required: + - compatible + - reg + - interrupts + - clock-names + - clocks + - reset-names + - resets + - "#sound-dai-cells" + +examples: + - | + audio@56000000 { + compatible = "socionext,uniphier-ld20-aio"; + reg = <0x56000000 0x80000>; + interrupts = <0 144 4>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_aout>; + clock-names = "aio"; + clocks = <&sys_clk 40>; + reset-names = "aio"; + resets = <&sys_rst 40>; + #sound-dai-cells = <1>; + socionext,syscon = <&soc_glue>; + }; diff --git a/Documentation/devicetree/bindings/sound/uniphier,aio.txt b/Documentation/devicetree/bindings/sound/uniphier,aio.txt deleted file mode 100644 index 4ce68ed6f2f2..000000000000 --- a/Documentation/devicetree/bindings/sound/uniphier,aio.txt +++ /dev/null @@ -1,45 +0,0 @@ -Socionext UniPhier SoC audio driver - -The Socionext UniPhier audio subsystem consists of I2S and S/PDIF blocks in -the same register space. - -Required properties: -- compatible : should be one of the following: - "socionext,uniphier-ld11-aio" - "socionext,uniphier-ld20-aio" - "socionext,uniphier-pxs2-aio" -- reg : offset and length of the register set for the device. -- interrupts : should contain I2S or S/PDIF interrupt. -- pinctrl-names : should be "default". -- pinctrl-0 : defined I2S signal pins for an external codec chip. -- clock-names : should include following entries: - "aio" -- clocks : a list of phandle, should contain an entry for each - entry in clock-names. -- reset-names : should include following entries: - "aio" -- resets : a list of phandle, should contain an entry for each - entry in reset-names. -- #sound-dai-cells: should be 1. - -Optional properties: -- socionext,syscon: a phandle, should contain soc-glue. - The soc-glue is used for changing mode of S/PDIF signal pin - to Output from Hi-Z. This property is optional if you use - I2S signal pins only. - -Example: - audio { - compatible = "socionext,uniphier-ld20-aio"; - reg = <0x56000000 0x80000>; - interrupts = <0 144 4>; - pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_aout>; - clock-names = "aio"; - clocks = <&sys_clk 40>; - reset-names = "aio"; - resets = <&sys_rst 40>; - #sound-dai-cells = <1>; - - socionext,syscon = <&sg>; - }; -- 2.25.1

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[PATCH] ALSA: hda: Enable sync-write operation as default for all controllers
by Takashi Iwai 14 Jul '20

14 Jul '20

In the end we already enabled the sync-write mode for most of HD-audio controllers including Intel, and it's no big merit to keep the async write mode for the rest. Let's make it as default and drop the superfluous AZX_DCAPS_SYNC_WRITE bit flag. Also, avoid to set the allow_bus_reset flag, which is a quite unstable and hackish behavior that was needed only for some early platforms (decades ago). The straight fallback to the single cmd mode is more robust. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_controller.c | 11 ++--------- sound/pci/hda/hda_controller.h | 2 +- sound/pci/hda/hda_intel.c | 16 +++++++--------- 3 files changed, 10 insertions(+), 19 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 9765652a73d7..80016b7b6849 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1202,15 +1202,8 @@ int azx_bus_init(struct azx *chip, const char *model) if (chip->driver_caps & AZX_DCAPS_4K_BDLE_BOUNDARY) bus->core.align_bdle_4k = true; - /* AMD chipsets often cause the communication stalls upon certain - * sequence like the pin-detection. It seems that forcing the synced - * access works around the stall. Grrr... - */ - if (chip->driver_caps & AZX_DCAPS_SYNC_WRITE) { - dev_dbg(chip->card->dev, "Enable sync_write for stable communication\n"); - bus->core.sync_write = 1; - bus->allow_bus_reset = 1; - } + /* enable sync_write flag for stable communication as default */ + bus->core.sync_write = 1; return 0; } diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index 82e26442724b..fe171685492d 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -33,7 +33,7 @@ #define AZX_DCAPS_POSFIX_LPIB (1 << 16) /* Use LPIB as default */ #define AZX_DCAPS_AMD_WORKAROUND (1 << 17) /* AMD-specific workaround */ #define AZX_DCAPS_NO_64BIT (1 << 18) /* No 64bit address */ -#define AZX_DCAPS_SYNC_WRITE (1 << 19) /* sync each cmd write */ +/* 19 unused */ #define AZX_DCAPS_OLD_SSYNC (1 << 20) /* Old SSYNC reg for ICH */ #define AZX_DCAPS_NO_ALIGN_BUFSIZE (1 << 21) /* no buffer size alignment */ /* 22 unused */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 3565e2ab0965..b2bd21f5a52e 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -283,13 +283,12 @@ enum { /* quirks for old Intel chipsets */ #define AZX_DCAPS_INTEL_ICH \ - (AZX_DCAPS_OLD_SSYNC | AZX_DCAPS_NO_ALIGN_BUFSIZE |\ - AZX_DCAPS_SYNC_WRITE) + (AZX_DCAPS_OLD_SSYNC | AZX_DCAPS_NO_ALIGN_BUFSIZE) /* quirks for Intel PCH */ #define AZX_DCAPS_INTEL_PCH_BASE \ (AZX_DCAPS_NO_ALIGN_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY |\ - AZX_DCAPS_SNOOP_TYPE(SCH) | AZX_DCAPS_SYNC_WRITE) + AZX_DCAPS_SNOOP_TYPE(SCH)) /* PCH up to IVB; no runtime PM; bind with i915 gfx */ #define AZX_DCAPS_INTEL_PCH_NOPM \ @@ -304,13 +303,13 @@ enum { #define AZX_DCAPS_INTEL_HASWELL \ (/*AZX_DCAPS_ALIGN_BUFSIZE |*/ AZX_DCAPS_COUNT_LPIB_DELAY |\ AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_COMPONENT |\ - AZX_DCAPS_SNOOP_TYPE(SCH) | AZX_DCAPS_SYNC_WRITE) + AZX_DCAPS_SNOOP_TYPE(SCH)) /* Broadwell HDMI can't use position buffer reliably, force to use LPIB */ #define AZX_DCAPS_INTEL_BROADWELL \ (/*AZX_DCAPS_ALIGN_BUFSIZE |*/ AZX_DCAPS_POSFIX_LPIB |\ AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_COMPONENT |\ - AZX_DCAPS_SNOOP_TYPE(SCH) | AZX_DCAPS_SYNC_WRITE) + AZX_DCAPS_SNOOP_TYPE(SCH)) #define AZX_DCAPS_INTEL_BAYTRAIL \ (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_I915_COMPONENT) @@ -321,19 +320,18 @@ enum { #define AZX_DCAPS_INTEL_SKYLAKE \ (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME |\ - AZX_DCAPS_SYNC_WRITE |\ AZX_DCAPS_SEPARATE_STREAM_TAG | AZX_DCAPS_I915_COMPONENT) #define AZX_DCAPS_INTEL_BROXTON AZX_DCAPS_INTEL_SKYLAKE /* quirks for ATI SB / AMD Hudson */ #define AZX_DCAPS_PRESET_ATI_SB \ - (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_POSFIX_LPIB |\ + (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_POSFIX_LPIB |\ AZX_DCAPS_SNOOP_TYPE(ATI)) /* quirks for ATI/AMD HDMI */ #define AZX_DCAPS_PRESET_ATI_HDMI \ - (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_POSFIX_LPIB|\ + (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_POSFIX_LPIB|\ AZX_DCAPS_NO_MSI64) /* quirks for ATI HDMI with snoop off */ @@ -342,7 +340,7 @@ enum { /* quirks for AMD SB */ #define AZX_DCAPS_PRESET_AMD_SB \ - (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_AMD_WORKAROUND |\ + (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_AMD_WORKAROUND |\ AZX_DCAPS_SNOOP_TYPE(ATI) | AZX_DCAPS_PM_RUNTIME) /* quirks for Nvidia */ -- 2.25.0

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[PATCH v2] dt-bindings: sound: convert rk3328 codec binding to yaml
by Katsuhiro Suzuki 14 Jul '20

14 Jul '20

This patch converts Rockchip rk3328 audio codec binding to DT schema. And adds description about "mclk" clock and fixes some errors in original example. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- .../bindings/sound/rockchip,rk3328-codec.txt | 28 -------- .../bindings/sound/rockchip,rk3328-codec.yaml | 70 +++++++++++++++++++ 2 files changed, 70 insertions(+), 28 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.yaml diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt deleted file mode 100644 index 1ecd75d2032a..000000000000 --- a/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt +++ /dev/null @@ -1,28 +0,0 @@ -* Rockchip Rk3328 internal codec - -Required properties: - -- compatible: "rockchip,rk3328-codec" -- reg: physical base address of the controller and length of memory mapped - region. -- rockchip,grf: the phandle of the syscon node for GRF register. -- clocks: a list of phandle + clock-specifer pairs, one for each entry in clock-names. -- clock-names: should be "pclk". -- spk-depop-time-ms: speak depop time msec. - -Optional properties: - -- mute-gpios: GPIO specifier for external line driver control (typically the - dedicated GPIO_MUTE pin) - -Example for rk3328 internal codec: - -codec: codec@ff410000 { - compatible = "rockchip,rk3328-codec"; - reg = <0x0 0xff410000 0x0 0x1000>; - rockchip,grf = <&grf>; - clocks = <&cru PCLK_ACODEC>; - clock-names = "pclk"; - mute-gpios = <&grf_gpio 0 GPIO_ACTIVE_LOW>; - spk-depop-time-ms = 100; -}; diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.yaml b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.yaml new file mode 100644 index 000000000000..525b48c2f5de --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.yaml @@ -0,0 +1,70 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip,rk3328-codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip rk3328 internal codec + +maintainers: + - Heiko Stuebner <heiko(a)sntech.de> + +properties: + compatible: + const: rockchip,rk3328-codec + + reg: + maxItems: 1 + + clocks: + items: + - description: clock for audio codec + - description: clock for I2S master clock + + clock-names: + items: + - const: pclk + - const: mclk + + rockchip,grf: + $ref: /schemas/types.yaml#/definitions/phandle + description: + The phandle of the syscon node for the GRF register. + + spk-depop-time-ms: + $ref: /schemas/types.yaml#/definitions/uint32 + default: 200 + description: + Speaker depop time in msec. + + mute-gpios: + maxItems: 1 + description: + GPIO specifier for external line driver control (typically the + dedicated GPIO_MUTE pin) + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + - clocks + - clock-names + - rockchip,grf + - "#sound-dai-cells" + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/clock/rk3328-cru.h> + codec: codec@ff410000 { + compatible = "rockchip,rk3328-codec"; + reg = <0xff410000 0x1000>; + clocks = <&cru PCLK_ACODECPHY>, <&cru SCLK_I2S1>; + clock-names = "pclk", "mclk"; + rockchip,grf = <&grf>; + mute-gpios = <&grf_gpio 0 GPIO_ACTIVE_LOW>; + spk-depop-time-ms = <100>; + #sound-dai-cells = <0>; + }; -- 2.27.0

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ASoC: rt5670/cht_bsw_rt5672: Some bugfixes and cleanups
by Hans de Goede 10 Jul '20

10 Jul '20

Hi All, This series is mainly the result of me working on fixing the extra set of speakers in the Lenovo Miix 2 10 kbd dock not working with Linux (patches 1 and 3) while working on this I noticed some other bugs (patches 2 and 4) and I also ended up doing some generic cleanups (patches 5 and 6). For patches 1-3 or 1-4 it would be nice it they can go to a future 5.8-rc# 5-6 are mostly -next material. Regards, Hans

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Re: KASAN: use-after-free Read in line6_submit_audio_in_all_urbs
by Alan Stern 10 Jul '20

10 Jul '20

On Fri, Jun 26, 2020 at 10:18:12AM -0700, syzbot wrote: > Hello, > > syzbot found the following crash on: > > HEAD commit: fb574682 usbip: tools: fix module name in man page > git tree: https://git.kernel.org/pub/scm/linux/kernel/git/gregkh/usb.git usb-testing > console output: https://syzkaller.appspot.com/x/log.txt?x=156560b1100000 > kernel config: https://syzkaller.appspot.com/x/.config?x=63b40b2ae167bad6 > dashboard link: https://syzkaller.appspot.com/bug?extid=145012a46658ac00fc9e > compiler: gcc (GCC) 10.1.0-syz 20200507 > > Unfortunately, I don't have any reproducer for this crash yet. > > IMPORTANT: if you fix the bug, please add the following tag to the commit: > Reported-by: syzbot+145012a46658ac00fc9e(a)syzkaller.appspotmail.com > > snd_usb_toneport 5-1:0.0: URB in #0 submission failed (-19) > snd_usb_toneport 5-1:0.0: URB in #0 submission failed (-19) > snd_usb_toneport 5-1:0.0: URB in #0 submission failed (-19) > snd_usb_toneport 5-1:0.0: URB in #0 submission failed (-19) > ================================================================== > BUG: KASAN: use-after-free in line6_submit_audio_in_all_urbs+0x10b/0x120 sound/usb/line6/capture.c:72 > Read of size 8 at addr ffff8881cffb1800 by task kworker/1:5/3257 > > CPU: 1 PID: 3257 Comm: kworker/1:5 Not tainted 5.8.0-rc1-syzkaller #0 > Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 > Workqueue: events line6_startup_work > Call Trace: > __dump_stack lib/dump_stack.c:77 [inline] > dump_stack+0xf6/0x16e lib/dump_stack.c:118 > print_address_description.constprop.0+0x1a/0x210 mm/kasan/report.c:383 > __kasan_report mm/kasan/report.c:513 [inline] > kasan_report.cold+0x37/0x7c mm/kasan/report.c:530 > line6_submit_audio_in_all_urbs+0x10b/0x120 sound/usb/line6/capture.c:72 > line6_stream_start+0x207/0x230 sound/usb/line6/pcm.c:197 > line6_pcm_acquire+0x161/0x210 sound/usb/line6/pcm.c:318 > line6_startup_work+0x42/0x50 sound/usb/line6/driver.c:734 > process_one_work+0x94c/0x15f0 kernel/workqueue.c:2269 > worker_thread+0x64c/0x1120 kernel/workqueue.c:2415 > kthread+0x392/0x470 kernel/kthread.c:291 > ret_from_fork+0x1f/0x30 arch/x86/entry/entry_64.S:293 > > Allocated by task 76: > save_stack+0x1b/0x40 mm/kasan/common.c:48 > set_track mm/kasan/common.c:56 [inline] > __kasan_kmalloc.constprop.0+0xc2/0xd0 mm/kasan/common.c:494 > kmalloc include/linux/slab.h:555 [inline] > kzalloc include/linux/slab.h:669 [inline] > line6_init_pcm+0x2a7/0x9e0 sound/usb/line6/pcm.c:533 > toneport_init+0xdd/0x6b0 sound/usb/line6/toneport.c:419 > line6_probe+0xaa0/0x1330 sound/usb/line6/driver.c:809 It look like the cancel_delayed_work() in line6_disconnect() needs to be cancel_delayed_work_sync(). Unfortunately we can't test this until syzbot is able to reproduce the bug. Alan Stern

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echoaudio: Fix some long standing bugs
by Mark Hills 09 Jul '20

09 Jul '20

Here follows some patches for the echoaudio PCM handling. Tested on the Layla3G on x86-64. I do have a Darla somewhere, but it's not practical for me to test with that right now. -- Mark

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