October 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Is this bug ? dpcm_prune_paths()
by Kuninori Morimoto 09 Oct '19

09 Oct '19

Hi ALSA ML. I'm checking soc-pcm.c, and I noticed strange code at dpcm_prune_paths(). static int dpcm_prune_paths(...) { ... /* Destroy any old FE <--> BE connections */ (1) for_each_dpcm_be(fe, stream, dpcm) { unsigned int i; - /* is there a valid CPU DAI widget for this BE */ | widget = dai_get_widget(dpcm->be->cpu_dai, stream); | (A) /* prune the BE if it's no longer in our active list */ | if (widget && widget_in_list(list, widget)) |(a) continue; - | /* is there a valid CODEC DAI widget for this BE */ |(2) for_each_rtd_codec_dai(dpcm->be, i, dai) { | widget = dai_get_widget(dai, stream); (B) | /* prune the BE if it's no longer in our active list */ | if (widget && widget_in_list(list, widget)) |(b) continue; - } - ... | dpcm->state = ... (C) dpcm->be->dpcm[stream].runtime_update = ... | prune++; - } ... } In my understanding, (A) part is for CPU, and (B) part is for Codec. Guessing from (a), I think it want to skip setup (C) if CPU widget exist and matches to list. If so, I guess (b) is maybe bug ? This continue is for (2) loop, thus, it is totally do nothing now. But maybe it want to be continue for (1) loop ? Thank you for your help !! Best regards --- Kuninori Morimoto

2 2

[alsa-devel] [PATCH -next] ASoc: tas2770: Fix build error without GPIOLIB
by YueHaibing 08 Oct '19

08 Oct '19

If GPIOLIB is not set, building fails: sound/soc/codecs/tas2770.c: In function tas2770_reset: sound/soc/codecs/tas2770.c:38:3: error: implicit declaration of function gpiod_set_value_cansleep; did you mean gpio_set_value_cansleep? [-Werror=implicit-function-declaration] gpiod_set_value_cansleep(tas2770->reset_gpio, 0); ^~~~~~~~~~~~~~~~~~~~~~~~ gpio_set_value_cansleep sound/soc/codecs/tas2770.c: In function tas2770_i2c_probe: sound/soc/codecs/tas2770.c:749:24: error: implicit declaration of function devm_gpiod_get_optional; did you mean devm_regulator_get_optional? [-Werror=implicit-function-declaration] tas2770->reset_gpio = devm_gpiod_get_optional(tas2770->dev, ^~~~~~~~~~~~~~~~~~~~~~~ devm_regulator_get_optional sound/soc/codecs/tas2770.c:751:13: error: GPIOD_OUT_HIGH undeclared (first use in this function); did you mean GPIOF_INIT_HIGH? GPIOD_OUT_HIGH); ^~~~~~~~~~~~~~ GPIOF_INIT_HIGH Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: 1a476abc723e ("tas2770: add tas2770 smart PA kernel driver") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index bcac957..d639f17 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1108,6 +1108,7 @@ config SND_SOC_TAS2552 config SND_SOC_TAS2770 tristate "Texas Instruments TAS2770 speaker amplifier" depends on I2C + select GPIOLIB config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" -- 2.7.4

4 8

[alsa-devel] [PATCH v2 0/4] ASoC: mt8183: fix audio playback slowly after playback
by Jiaxin Yu 08 Oct '19

08 Oct '19

This series patches add reset controller for MT8183, and audio will use it in machine driver during bootup, they depend on the following patch: 1. this series add support reset-controller in infra [v5,2/2] clk: reset: Modify reset-controller driver https://patchwork.kernel.org/patch/11060419/ v2 changes: 1. remove "WIP" that in the title of patches 2. add hyper link for the patch that depends on Jiaxin Yu (2): dt-bindings: medaitek: mt8183: add property "resets" && "reset-names" ASoC: mt8183: fix audio playback slowly after playback during bootup yong.liang (2): dt-bindings: mediatek: mt8183: Add #reset-cells watchdog: mtk_wdt: mt8183: Add reset controller .../bindings/sound/mt8183-afe-pcm.txt | 6 + .../devicetree/bindings/watchdog/mtk-wdt.txt | 9 +- drivers/watchdog/Kconfig | 1 + drivers/watchdog/mtk_wdt.c | 110 +++++++++++++++++- .../reset-controller/mt8183-resets.h | 13 +++ sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 15 +++ 6 files changed, 150 insertions(+), 4 deletions(-) -- 2.18.0

5 15

[alsa-devel] [PATCH 00/44] ASoC: don't use snd_pcm_ops
by Kuninori Morimoto 08 Oct '19

08 Oct '19

Hi Mark Current snd_soc_component_driver is using snd_pcm_ops. But, it is mainly for ALSA, thus, it not good much to ALSA SoC. For example .open case (1)(2), each callback can't know it is called for which component. Thus, it need to use using snd_soc_rtdcom_lookup() (3). --- ALSA SoC --- ... if (component->driver->ops && component->driver->ops->open) (1) return component->driver->ops->open(substream); ... --- driver --- (2) static int xxx_open(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; (3) struct snd_soc_component *component = snd_soc_rtdcom_lookup(..); ... } It works today, but, will not work in the future if we support multi CPU/Codec/Platform, because 1 rtd might have multiple components which have same driver name. In other words snd_soc_rtdcom_lookup() doesn't work correctly in the future. To solve this issue, each callback needs to be called with component. This patch copies each snd_pcm_ops member under component driver, and having "component" as parameter. --- ALSA SoC --- ... if (component->driver->open) => return component->driver->open(component, substream); ... --- driver --- => static int xxx_open(struct snd_soc_component *component, struct snd_pcm_substream *substream) { ... } Kuninori Morimoto (44): ASoC: soc-core: merge snd_pcm_ops member to component driver ASoC: soc-core: add new pcm_construct/pcm_destruct ASoC: soc-core: add snd_soc_pcm_lib_ioctl() ASoC: intel: sst-haswell-pcm: remove snd_pcm_ops ASoC: intel: skl-pcm: remove snd_pcm_ops ASoC: intel: sst-mfld-platform-pcm: remove snd_pcm_ops ASoC: intel: sst-baytrail-pcm: remove snd_pcm_ops ASoC: amd: acp3x-pcm-dma: remove snd_pcm_ops ASoC: amd: acp-pcm-dma: remove snd_pcm_ops ASoC: atmel: atmel-pcm-pdc: remove snd_pcm_ops ASoC: au1x: dbdma2: remove snd_pcm_ops ASoC: au1x: dma: remove snd_pcm_ops ASoC: xtensa: xtfpga-i2s: remove snd_pcm_ops ASoC: xilinx: xlnx_formatter_pcm: remove snd_pcm_ops ASoC: uniphier: aio-dma: remove snd_pcm_ops ASoC: txx9: txx9aclc: remove snd_pcm_ops ASoC: stm: stm32_adfsdm: remove snd_pcm_ops ASoC: sprd: sprd-pcm-dma: remove snd_pcm_ops ASoC: sof: pcm: remove snd_pcm_ops ASoC: sh: rsnd: remove snd_pcm_ops ASoC: sh: fsi: remove snd_pcm_ops ASoC: sh: dma-sh7760: remove snd_pcm_ops ASoC: sh: siu_pcm: remove snd_pcm_ops ASoC: samsung: idma: remove snd_pcm_ops ASoC: qcom: lpass-platform: remove snd_pcm_ops ASoC: qcom: q6routing: remove snd_pcm_ops ASoC: qcom: q6asm-dai: remove snd_pcm_ops ASoC: pxa: mmp-pcm: remove snd_pcm_ops ASoC: pxa: remove snd_pcm_ops ASoC: meson: remove snd_pcm_ops ASoC: mediatek: mtk-btcvsd: remove snd_pcm_ops ASoC: mediatek: remove snd_pcm_ops ASoC: kirkwood: kirkwood-dma: remove snd_pcm_ops ASoC: dwc: dwc-pcm: remove snd_pcm_ops ASoC: rt5514-spi: remove snd_pcm_ops ASoC: bcm: cygnus-pcm: remove snd_pcm_ops ASoC: fsl: imx-pcm-fiq: remove snd_pcm_ops ASoC: fsl: fsl_dma: remove snd_pcm_ops ASoC: fsl: mpc5200_dma: remove snd_pcm_ops ASoC: fsl: fsl_asrc_dma: remove snd_pcm_ops ASoC: fsl: fsl_dma: don't use snd_soc_rtdcom_lookup() ASoC: soc-generic-dmaengine-pcm: remove snd_pcm_ops ASoC: soc-utils: remove snd_pcm_ops ASoC: soc-component: remove snd_pcm_ops from component driver include/sound/pxa2xx-lib.h | 26 +++++- include/sound/soc-component.h | 41 ++++++++- include/sound/soc.h | 5 + sound/arm/pxa2xx-pcm-lib.c | 80 +++++++++++++--- sound/soc/amd/acp-pcm-dma.c | 63 ++++++------- sound/soc/amd/raven/acp3x-pcm-dma.c | 54 +++++------ sound/soc/atmel/atmel-pcm-pdc.c | 48 +++++----- sound/soc/au1x/dbdma2.c | 64 +++++++------ sound/soc/au1x/dma.c | 63 +++++++------ sound/soc/bcm/cygnus-pcm.c | 56 +++++------ sound/soc/codecs/rt5514-spi.c | 41 +++++---- sound/soc/dwc/dwc-pcm.c | 48 +++++----- sound/soc/fsl/fsl_asrc_dma.c | 56 +++++------ sound/soc/fsl/fsl_dma.c | 53 +++++------ sound/soc/fsl/imx-pcm-fiq.c | 56 ++++++----- sound/soc/fsl/mpc5200_dma.c | 51 ++++++----- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 30 +++--- sound/soc/intel/baytrail/sst-baytrail-pcm.c | 52 +++++------ sound/soc/intel/haswell/sst-haswell-pcm.c | 62 +++++++------ sound/soc/intel/skylake/skl-pcm.c | 67 +++++++++----- sound/soc/kirkwood/kirkwood-dma.c | 50 +++++----- .../soc/mediatek/common/mtk-afe-platform-driver.c | 28 +++--- .../soc/mediatek/common/mtk-afe-platform-driver.h | 10 +- sound/soc/mediatek/common/mtk-btcvsd.c | 76 ++++++--------- sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 11 ++- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 11 ++- sound/soc/meson/axg-fifo.c | 56 +++++------ sound/soc/meson/axg-fifo.h | 20 +++- sound/soc/meson/axg-frddr.c | 24 ++++- sound/soc/meson/axg-toddr.c | 24 ++++- sound/soc/pxa/mmp-pcm.c | 62 ++++++++----- sound/soc/pxa/pxa-ssp.c | 14 ++- sound/soc/pxa/pxa2xx-ac97.c | 14 ++- sound/soc/pxa/pxa2xx-i2s.c | 14 ++- sound/soc/pxa/pxa2xx-pcm.c | 14 ++- sound/soc/qcom/lpass-platform.c | 70 +++++++------- sound/soc/qcom/qdsp6/q6asm-dai.c | 71 +++++++------- sound/soc/qcom/qdsp6/q6routing.c | 14 +-- sound/soc/samsung/idma.c | 58 ++++++------ sound/soc/sh/dma-sh7760.c | 46 +++++----- sound/soc/sh/fsi.c | 31 ++++--- sound/soc/sh/rcar/core.c | 23 +++-- sound/soc/sh/siu_pcm.c | 44 +++++---- sound/soc/soc-component.c | 76 +++++++-------- sound/soc/soc-core.c | 12 +++ sound/soc/soc-generic-dmaengine-pcm.c | 102 ++++++++++----------- sound/soc/soc-pcm.c | 11 +-- sound/soc/soc-utils.c | 11 +-- sound/soc/sof/pcm.c | 88 +++++++++--------- sound/soc/sprd/sprd-pcm-dma.c | 74 +++++++-------- sound/soc/stm/stm32_adfsdm.c | 42 +++++---- sound/soc/txx9/txx9aclc.c | 48 +++++----- sound/soc/uniphier/aio-dma.c | 51 ++++++----- sound/soc/xilinx/xlnx_formatter_pcm.c | 54 +++++------ sound/soc/xtensa/xtfpga-i2s.c | 30 +++--- 55 files changed, 1311 insertions(+), 1089 deletions(-) -- 2.7.4

2 89

[alsa-devel] [PATCH v2] firmware: vpd: Add an interface to read VPD value
by Cheng-Yi Chiang 08 Oct '19

08 Oct '19

Add an interface for other driver to query VPD value. This will be used for ASoC machine driver to query calibration data stored in VPD for smart amplifier speaker resistor calibration. The example usage in ASoC machine driver is like: #define DSM_CALIB_KEY "dsm_calib" static int load_calibration_data(struct cml_card_private *ctx) { char *data = NULL; int ret; u32 value_len; /* Read calibration data from VPD. */ ret = vpd_attribute_read(1, DSM_CALIB_KEY, (u8 **)&data, &value_len); /* Parsing of this string...*/ } Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- Change from v1 to v2: - Use kmemdup to copy data. - Set value_len according to bin_attr.size. - Check return value of kmemdup. - Rename the function to vpd_attribute_read. - Add docstrings for the function. - Returns -ENOENT when the key is not found. - Use EXPORT_SYMBOL_GPL. Note: The user of this API is in ASoC machine driver cml_rt1011_rt5682. It is pending on the initial machine driver change https://patchwork.kernel.org/patch/11161145/ and the codec driver change to provide API to do calibration. https://patchwork.kernel.org/patch/11179237/ The draft patch of machine driver change which uses this API is at https://chromium-review.googlesource.com/c/chromiumos/third_party/kernel/+/… drivers/firmware/google/vpd.c | 31 ++++++++++++++++++++++ include/linux/firmware/google/google_vpd.h | 18 +++++++++++++ 2 files changed, 49 insertions(+) create mode 100644 include/linux/firmware/google/google_vpd.h diff --git a/drivers/firmware/google/vpd.c b/drivers/firmware/google/vpd.c index db0812263d46..c2be0e756402 100644 --- a/drivers/firmware/google/vpd.c +++ b/drivers/firmware/google/vpd.c @@ -65,6 +65,37 @@ static ssize_t vpd_attrib_read(struct file *filp, struct kobject *kobp, info->bin_attr.size); } +/** + * vpd_attribute_read - Read VPD value for a key. + * @ro: True for RO section. False for RW section. + * @key: A string for key. + * @value: Where to write the VPD value on success. The caller + * must free the memory. + * @value_len: The length of value in bytes. + * + * Returns 0 on success, -ENOENT when the key is not found, and + * -ENOMEM when failed to allocate memory for value. + */ +int vpd_attribute_read(bool ro, const char *key, + u8 **value, u32 *value_len) +{ + struct vpd_attrib_info *info; + struct vpd_section *sec = ro ? &ro_vpd : &rw_vpd; + + list_for_each_entry(info, &sec->attribs, list) { + if (strcmp(info->key, key) == 0) { + *value = kmemdup(info->value, info->bin_attr.size, + GFP_KERNEL); + if (!*value) + return -ENOMEM; + *value_len = info->bin_attr.size; + return 0; + } + } + return -ENOENT; +} +EXPORT_SYMBOL_GPL(vpd_attribute_read); + /* * vpd_section_check_key_name() * diff --git a/include/linux/firmware/google/google_vpd.h b/include/linux/firmware/google/google_vpd.h new file mode 100644 index 000000000000..4364eaa4e1e3 --- /dev/null +++ b/include/linux/firmware/google/google_vpd.h @@ -0,0 +1,18 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Google VPD interface. + * + * Copyright 2019 Google Inc. + */ + +/* Interface for reading VPD value on Chrome platform. */ + +#ifndef __GOOGLE_VPD_H +#define __GOOGLE_VPD_H + +#include <linux/types.h> + +int vpd_attribute_read(bool ro, const char *key, + u8 **value, u32 *value_len); + +#endif /* __GOOGLE_VPD_H */ -- 2.23.0.581.g78d2f28ef7-goog

2 2

[alsa-devel] [RFC PATCH 00/12] soundwire/SOF: updated interfaces, functional integration
by Pierre-Louis Bossart 08 Oct '19

08 Oct '19

This series builds on the 'soundwire: add Master device support, GreyBus style' RFC'. It provides enhancements to the stream callbacks, a split initialization. Most of the SOF patches were already submitted in an earlier RFC, and feedback on the parameters was taken into account. The main change here are the API changes with a split between ACPI scan, probe, startup steps. Known limits: a) Power management (regular suspend/resume and pm_runtime) is not supported for now as we need to run additional checks on hardware. This will be provided as a separate series. b) during validation checks on CML/ICL, initialization and playback/capture worked fine, but we observed a reproducible system freeze while doing load/unload tests, so likely an initialization missing or a leak to be fixed. Comments and feedback welcome. Pierre-Louis Bossart (7): ASoC: soc-acpi: add link_mask field ASoC: SOF: support alternate list of machines ASoC: SOF: Intel: add SoundWire configuration interface ASoC: SOF: Intel: add build support for SoundWire ASoC: SOF: IPC: dai-intel: move ALH declarations in header file ASoC: SOF: Intel: hda: add SoundWire stream config/free callbacks ASoC: SOF: Intel: hda: initial SoundWire machine driver autodetect Rander Wang (5): soundwire: intel: update stream callbacks for hwparams/free stream operations soundwire: intel: add prepare support in sdw dai driver soundwire: intel: add trigger support in sdw dai driver soundwire: intel: add sdw_stream_setup helper for .startup callback soundwire: intel: free all resources on hw_free() drivers/soundwire/intel.c | 181 ++++++++++++++++++- drivers/soundwire/intel_init.c | 2 +- include/linux/soundwire/sdw_intel.h | 40 ++++- include/sound/soc-acpi.h | 2 + include/sound/sof.h | 3 + include/sound/sof/dai-intel.h | 18 +- sound/soc/sof/intel/Kconfig | 23 +++ sound/soc/sof/intel/hda-loader.c | 9 + sound/soc/sof/intel/hda.c | 261 +++++++++++++++++++++++++++- sound/soc/sof/intel/hda.h | 36 ++++ 10 files changed, 546 insertions(+), 29 deletions(-) -- 2.20.1

2 14

[alsa-devel] [PATCH v2 1/3] ASoC: tas2770: Fix snd_soc_update_bits error handling
by Dan Murphy 07 Oct '19

07 Oct '19

According the documentation for snd_soc_update_bits the API will return a 1 if the update was successful with a value change, a 0 if the update was successful with no value change or a negative if the command just failed. So the value of return in the driver needs to be checked for being less then 0 or the caller may indicate failure when the value actually changed. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- v2 - Missed one update_bit return issue. sound/soc/codecs/tas2770.c | 46 +++++++++++++++++++++++++------------- 1 file changed, 30 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c index dbbb21fe0548..361d0bba72b3 100644 --- a/sound/soc/codecs/tas2770.c +++ b/sound/soc/codecs/tas2770.c @@ -82,7 +82,8 @@ static int tas2770_codec_suspend(struct snd_soc_component *component) TAS2770_PWR_CTRL, TAS2770_PWR_CTRL_MASK, TAS2770_PWR_CTRL_SHUTDOWN); - if (ret) + + if (ret < 0) return ret; return 0; @@ -96,8 +97,9 @@ static int tas2770_codec_resume(struct snd_soc_component *component) TAS2770_PWR_CTRL, TAS2770_PWR_CTRL_MASK, TAS2770_PWR_CTRL_ACTIVE); - if (ret) - return -EINVAL; + + if (ret < 0) + return ret; return 0; } @@ -149,7 +151,10 @@ static int tas2770_dac_event(struct snd_soc_dapm_widget *w, } end: - return ret; + if (ret < 0) + return ret; + + return 0; } static const struct snd_kcontrol_new isense_switch = @@ -199,7 +204,10 @@ static int tas2770_mute(struct snd_soc_dai *dai, int mute) TAS2770_PWR_CTRL_MASK, TAS2770_PWR_CTRL_ACTIVE); - return ret; + if (ret < 0) + return ret; + + return 0; } static int tas2770_set_bitwidth(struct tas2770_priv *tas2770, int bitwidth) @@ -252,7 +260,10 @@ static int tas2770_set_bitwidth(struct tas2770_priv *tas2770, int bitwidth) tas2770->i_sense_slot); end: - return ret; + if (ret < 0) + return ret; + + return 0; } static int tas2770_set_samplerate(struct tas2770_priv *tas2770, int samplerate) @@ -344,9 +355,11 @@ static int tas2770_set_samplerate(struct tas2770_priv *tas2770, int samplerate) } end: - if (!ret) - tas2770->sampling_rate = samplerate; - return ret; + if (ret < 0) + return ret; + + tas2770->sampling_rate = samplerate; + return 0; } static int tas2770_hw_params(struct snd_pcm_substream *substream, @@ -401,7 +414,7 @@ static int tas2770_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG1, TAS2770_TDM_CFG_REG1_RX_MASK, asi_cfg_1); - if (ret) + if (ret < 0) return ret; switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { @@ -426,7 +439,7 @@ static int tas2770_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG1, TAS2770_TDM_CFG_REG1_MASK, (tdm_rx_start_slot << TAS2770_TDM_CFG_REG1_51_SHIFT)); - if (ret) + if (ret < 0) return ret; value = snd_soc_component_read32(component, TAS2770_TDM_CFG_REG3); @@ -472,12 +485,12 @@ static int tas2770_set_dai_tdm_slot(struct snd_soc_dai *dai, ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG3, TAS2770_TDM_CFG_REG3_30_MASK, (left_slot << TAS2770_TDM_CFG_REG3_30_SHIFT)); - if (ret) + if (ret < 0) return ret; ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG3, TAS2770_TDM_CFG_REG3_RXS_MASK, (right_slot << TAS2770_TDM_CFG_REG3_RXS_SHIFT)); - if (ret) + if (ret < 0) return ret; switch (slot_width) { @@ -511,10 +524,11 @@ static int tas2770_set_dai_tdm_slot(struct snd_soc_dai *dai, ret = -EINVAL; } - if (!ret) - tas2770->slot_width = slot_width; + if (ret < 0) + return ret; - return ret; + tas2770->slot_width = slot_width; + return 0; } static struct snd_soc_dai_ops tas2770_dai_ops = { -- 2.22.0.214.g8dca754b1e

2 5

[alsa-devel] [PATCH 0/2] [RFC] ASoC: topology: fix access beyond data
by Guennadi Liakhovetski 07 Oct '19

07 Oct '19

This is likely incomplete, I haven't reviewed the whole of the topology parsing code, and I haven't tested this yet - neither for actually being a problem without these patches nor for fixing them. This is a preview to give everybody a chance to showt out if I misunderstood something and there isn't actually a problem to be fixed. Thanks Guennadi Guennadi Liakhovetski (2): ASoC: topology: protect against accessing beyond loaded topology data ASoC: topology: don't access beyond topology data sound/soc/soc-topology.c | 37 +++++++++++++++++++++++++++++-------- 1 file changed, 29 insertions(+), 8 deletions(-) -- 1.9.3

2 5

[alsa-devel] what's the kernel policy WRT firmware parsing security?
by Guennadi Liakhovetski 07 Oct '19

07 Oct '19

Hi, I decided to have a look at whether the ALSA topology parsing is bullet proof against malformed topology files. It seems not quite to be the case. At least I seem to have found a possibility of crashing the kernel by a malformed topology file. I haven't tested it, so, maybe I'm wrong. In principle, firmware files can only be written by root, and if you have root access to the system, it's anyway doomed. Is this the approach and we aren't really trying to make topology parsing 100% safe, or do we want to fix any such possible parsing issues? Thanks Guennadi

2 2

[alsa-devel] [PATCH 0/1] ALSA: hda: add fallback to polling to hdac_bus_get_response()
by Kai Vehmanen 07 Oct '19

07 Oct '19

Hey all, while debugging issues with some Intel platforms related to display audio codec probe (see https://lists.freedesktop.org/archives/intel-gfx/2019-October/214621.html ), I found a discrepancy in behaviour between snd-hda-intel and SOF, despite using the same snd-hda-codec-hdmi as the codec driver. The specific problem I was debugging appears in a stress test (designed to uncover the above display driver issue) where driver-unload, s3-suspend, resume and driver-reload is done in a loop and repeated for hundreds of iterations. When using SOF, I would get occasional probe fail due to a missing HDA irq. The AZX snd_hda_intel driver nicely survives this test. The explanation seems to be differences in the hdac get_response() implementation. While the specific issue could be solved with other means, the git history shows a number of rare issues with HDA codecs where polling has helped. It would seem best to align the logic with the AZX driver implementation that has seen much more usage over the years. This will benefit SOF and any other users of the HDAC library. Kai Vehmanen (1): ALSA: hda: add fallback to polling to hdac_bus_get_response() sound/hda/hdac_controller.c | 23 ++++++++++++++++++++++- 1 file changed, 22 insertions(+), 1 deletion(-) -- 2.17.1

2 3

[alsa-devel] [PATCH] ASoC: core: Fix pcm code debugfs error
by Hans de Goede 07 Oct '19

07 Oct '19

We can have 2 dcpm-s with the same backend and frontend name (capture + playback pair), this causes the following debugfs error on Intel Bay Trail systems: [ 298.969049] debugfs: Directory 'SSP2-Codec' with parent 'Baytrail Audio Port' already present! This commit adds a ":playback" or ":capture" postfix to the debugfs dir name fixing this. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/soc-pcm.c | 13 ++++++++++--- 1 file changed, 10 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index e163dde5eab1..2e723012eb52 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1146,6 +1146,7 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, { struct snd_soc_dpcm *dpcm; unsigned long flags; + char *name; /* only add new dpcms */ for_each_dpcm_be(fe, stream, dpcm) { @@ -1171,9 +1172,15 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, stream ? "<-" : "->", be->dai_link->name); #ifdef CONFIG_DEBUG_FS - dpcm->debugfs_state = debugfs_create_dir(be->dai_link->name, - fe->debugfs_dpcm_root); - debugfs_create_u32("state", 0644, dpcm->debugfs_state, &dpcm->state); + name = kasprintf(GFP_KERNEL, "%s:%s", be->dai_link->name, + stream ? "capture" : "playback"); + if (name) { + dpcm->debugfs_state = debugfs_create_dir(name, + fe->debugfs_dpcm_root); + debugfs_create_u32("state", 0644, dpcm->debugfs_state, + &dpcm->state); + kfree(name); + } #endif return 1; } -- 2.23.0

2 1

[alsa-devel] [PATCH 07/10] ASoC: jz4740: Use of_device_get_match_data()
by Stephen Boyd 07 Oct '19

07 Oct '19

This probe function is only called if the device is backed by a DT node, so switch this call to of_device_get_match_data() to reduce code size and simplify a bit. This also avoids needing to reference a potentially undefined variable because of_device_get_match_data() doesn't need to know anything beyond the struct device to find the match table. Cc: Arnd Bergmann <arnd(a)arndb.de> Cc: Geert Uytterhoeven <geert(a)linux-m68k.org> Cc: Paul Cercueil <paul(a)crapouillou.net> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Rob Herring <robh+dt(a)kernel.org> Cc: Frank Rowand <frowand.list(a)gmail.com> Cc: <alsa-devel(a)alsa-project.org> Signed-off-by: Stephen Boyd <swboyd(a)chromium.org> --- Please ack or pick for immediate merge so the last patch can be merged. sound/soc/jz4740/jz4740-i2s.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 13408de34055..d2dab4d24b87 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -503,9 +503,8 @@ static int jz4740_i2s_dev_probe(struct platform_device *pdev) if (!i2s) return -ENOMEM; - match = of_match_device(jz4740_of_matches, &pdev->dev); - if (match) - i2s->version = (enum jz47xx_i2s_version)match->data; + i2s->version = + (enum jz47xx_i2s_version)of_device_get_match_data(&pdev->dev); mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); i2s->base = devm_ioremap_resource(&pdev->dev, mem); -- Sent by a computer through tubes

2 1

[alsa-devel] [PATCH v2 RESEND] drm/bridge: dw-hdmi: Restore audio when setting a mode
by Cheng-Yi Chiang 07 Oct '19

07 Oct '19

From: Daniel Kurtz <djkurtz(a)chromium.org> When setting a new display mode, dw_hdmi_setup() calls dw_hdmi_enable_video_path(), which disables all hdmi clocks, including the audio clock. We should only (re-)enable the audio clock if audio was already enabled when setting the new mode. Without this patch, on RK3288, there will be HDMI audio on some monitors if i2s was played to headphone when the monitor was plugged. ACER H277HU and ASUS PB278 are two of the monitors showing this issue. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> Signed-off-by: Daniel Kurtz <djkurtz(a)chromium.org> Signed-off-by: Yakir Yang <ykk(a)rock-chips.com> --- Change from v1 to v2: - Use audio_lock to protect audio clock. - Fix the patch title. drivers/gpu/drm/bridge/synopsys/dw-hdmi.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c index aa7efd4da1c8..749d8e4c535b 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c @@ -1982,6 +1982,17 @@ static void hdmi_disable_overflow_interrupts(struct dw_hdmi *hdmi) HDMI_IH_MUTE_FC_STAT2); } +static void dw_hdmi_audio_restore(struct dw_hdmi *hdmi) +{ + unsigned long flags; + + spin_lock_irqsave(&hdmi->audio_lock, flags); + + hdmi_enable_audio_clk(hdmi, hdmi->audio_enable); + + spin_unlock_irqrestore(&hdmi->audio_lock, flags); +} + static int dw_hdmi_setup(struct dw_hdmi *hdmi, struct drm_display_mode *mode) { int ret; @@ -2045,7 +2056,7 @@ static int dw_hdmi_setup(struct dw_hdmi *hdmi, struct drm_display_mode *mode) /* HDMI Initialization Step E - Configure audio */ hdmi_clk_regenerator_update_pixel_clock(hdmi); - hdmi_enable_audio_clk(hdmi, true); + dw_hdmi_audio_restore(hdmi); } /* not for DVI mode */ -- 2.23.0.444.g18eeb5a265-goog

2 2

[alsa-devel] [PATCH -next] ASoC: fsl_mqs: remove set but not used variable 'bclk'
by YueHaibing 07 Oct '19

07 Oct '19

Fixes gcc '-Wunused-but-set-variable' warning: sound/soc/fsl/fsl_mqs.c: In function fsl_mqs_hw_params: sound/soc/fsl/fsl_mqs.c:54:6: warning: variable bclk set but not used [-Wunused-but-set-variable] It is never used, so can be removed. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/fsl/fsl_mqs.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_mqs.c b/sound/soc/fsl/fsl_mqs.c index c1619a5..7b9cab3 100644 --- a/sound/soc/fsl/fsl_mqs.c +++ b/sound/soc/fsl/fsl_mqs.c @@ -51,10 +51,9 @@ static int fsl_mqs_hw_params(struct snd_pcm_substream *substream, struct fsl_mqs *mqs_priv = snd_soc_component_get_drvdata(component); unsigned long mclk_rate; int div, res; - int bclk, lrclk; + int lrclk; mclk_rate = clk_get_rate(mqs_priv->mclk); - bclk = snd_soc_params_to_bclk(params); lrclk = params_rate(params); /* -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: fsl_mqs: Fix error handling in probe
by Dan Carpenter 07 Oct '19

07 Oct '19

There are several problems in the error handling in fsl_mqs_probe(). 1) "ret" isn't initialized on some paths. GCC has a feature which warns about uninitialized variables but the code initializes "ret" to zero at the start of the function so the checking is turned off. 2) "gpr_np" is a pointer so initializing it to zero is confusing and generates a Sparse warning. 3) of_parse_phandle() doesn't return error pointers on error, it returns NULL. 4) If devm_snd_soc_register_component() fails then the function should free the "gpr_np". Fixes: 9e28f6532c61 ("ASoC: fsl_mqs: Add MQS component driver") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/fsl/fsl_mqs.c | 27 +++++++++++++++------------ 1 file changed, 15 insertions(+), 12 deletions(-) diff --git a/sound/soc/fsl/fsl_mqs.c b/sound/soc/fsl/fsl_mqs.c index c1619a553514..f5f2f659bcbf 100644 --- a/sound/soc/fsl/fsl_mqs.c +++ b/sound/soc/fsl/fsl_mqs.c @@ -179,10 +179,10 @@ static const struct regmap_config fsl_mqs_regmap_config = { static int fsl_mqs_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; - struct device_node *gpr_np = 0; + struct device_node *gpr_np = NULL; struct fsl_mqs *mqs_priv; void __iomem *regs; - int ret = 0; + int ret; mqs_priv = devm_kzalloc(&pdev->dev, sizeof(*mqs_priv), GFP_KERNEL); if (!mqs_priv) @@ -199,17 +199,16 @@ static int fsl_mqs_probe(struct platform_device *pdev) if (mqs_priv->use_gpr) { gpr_np = of_parse_phandle(np, "gpr", 0); - if (IS_ERR(gpr_np)) { + if (!gpr_np) { dev_err(&pdev->dev, "failed to get gpr node by phandle\n"); - ret = PTR_ERR(gpr_np); - goto out; + return -EINVAL; } mqs_priv->regmap = syscon_node_to_regmap(gpr_np); if (IS_ERR(mqs_priv->regmap)) { dev_err(&pdev->dev, "failed to get gpr regmap\n"); ret = PTR_ERR(mqs_priv->regmap); - goto out; + goto err_free_gpr_np; } } else { regs = devm_platform_ioremap_resource(pdev, 0); @@ -230,7 +229,7 @@ static int fsl_mqs_probe(struct platform_device *pdev) if (IS_ERR(mqs_priv->ipg)) { dev_err(&pdev->dev, "failed to get the clock: %ld\n", PTR_ERR(mqs_priv->ipg)); - goto out; + return PTR_ERR(mqs_priv->ipg); } } @@ -238,17 +237,21 @@ static int fsl_mqs_probe(struct platform_device *pdev) if (IS_ERR(mqs_priv->mclk)) { dev_err(&pdev->dev, "failed to get the clock: %ld\n", PTR_ERR(mqs_priv->mclk)); - goto out; + ret = PTR_ERR(mqs_priv->mclk); + goto err_free_gpr_np; } dev_set_drvdata(&pdev->dev, mqs_priv); pm_runtime_enable(&pdev->dev); - return devm_snd_soc_register_component(&pdev->dev, &soc_codec_fsl_mqs, + ret = devm_snd_soc_register_component(&pdev->dev, &soc_codec_fsl_mqs, &fsl_mqs_dai, 1); -out: - if (!IS_ERR(gpr_np)) - of_node_put(gpr_np); + if (ret) + goto err_free_gpr_np; + return 0; + +err_free_gpr_np: + of_node_put(gpr_np); return ret; } -- 2.20.1

2 1

[alsa-devel] tegra30 tdm audio support
by Ben Dooks 07 Oct '19

07 Oct '19

This adds support for tegra30 tdm audio and fixes a bug in the fifo handling on rx when non-32 bit aligned data is used. v2: - moved edge-control and data-offset to the set_fmt callbacks - fixed dev_dbg in set_tdm callback - fixed dev_dbg message contents in set_tdm callback - changed fsync width to be permanently 1 clock v3: - cleanup fsync patch - fix rebase issue with tdm patch

3 19

[alsa-devel] [PATCH] ALSA: hdac: clear link output stream mapping
by Pierre-Louis Bossart 07 Oct '19

07 Oct '19

From: Rander Wang <rander.wang(a)linux.intel.com> Fix potential DMA hang upon starting playback on devices in HDA mode on Intel platforms (Gemini Lake/Whiskey Lake/Comet Lake/Ice Lake). It doesn't affect platforms before Gemini Lake or any Intel device in non-HDA mode. The reset value for the LOSDIV register is all output streams valid. Clear this register to invalidate non-existent streams when the bus is powered up. Signed-off-by: Rander Wang <rander.wang(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- include/sound/hda_register.h | 3 +++ sound/hda/ext/hdac_ext_controller.c | 5 +++++ 2 files changed, 8 insertions(+) diff --git a/include/sound/hda_register.h b/include/sound/hda_register.h index 0fd39295b426..057d2a2d0bd0 100644 --- a/include/sound/hda_register.h +++ b/include/sound/hda_register.h @@ -264,6 +264,9 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define AZX_REG_ML_LOUTPAY 0x20 #define AZX_REG_ML_LINPAY 0x30 +/* bit0 is reserved, with BIT(1) mapping to stream1 */ +#define ML_LOSIDV_STREAM_MASK 0xFFFE + #define ML_LCTL_SCF_MASK 0xF #define AZX_MLCTL_SPA (0x1 << 16) #define AZX_MLCTL_CPA (0x1 << 23) diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index 211ca85acd8c..cfab60d88c92 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -270,6 +270,11 @@ int snd_hdac_ext_bus_link_get(struct hdac_bus *bus, ret = snd_hdac_ext_bus_link_power_up(link); + /* + * clear the register to invalidate all the output streams + */ + snd_hdac_updatew(link->ml_addr, AZX_REG_ML_LOSIDV, + ML_LOSIDV_STREAM_MASK, 0); /* * wait for 521usec for codec to report status * HDA spec section 4.3 - Codec Discovery -- 2.20.1

2 1

[alsa-devel] [PATCH] ALSA: hda/realtek: Reduce the Headphone static noise on XPS 9350/9360
by Kai-Heng Feng 07 Oct '19

07 Oct '19

Headphone on XPS 9350/9360 produces a background white noise. The The noise level somehow correlates with "Headphone Mic Boost", when it sets to 1 the noise disappears. However, doing this has a side effect, which also decreases the overall headphone volume so I didn't send the patch upstream. The noise was bearable back then, but after commit 717f43d81afc ("ALSA: hda/realtek - Update headset mode for ALC256") the noise exacerbates to a point it starts hurting ears. So let's use the workaround to set "Headphone Mic Boost" to 1 and lock it so it's not touchable by userspace. Fixes: 717f43d81afc ("ALSA: hda/realtek - Update headset mode for ALC256") BugLink: https://bugs.launchpad.net/bugs/1654448 BugLink: https://bugs.launchpad.net/bugs/1845810 Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 24 +++++++++++++++++++++--- 1 file changed, 21 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b000b36ac3c6..b5c225a56b98 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5358,6 +5358,17 @@ static void alc271_hp_gate_mic_jack(struct hda_codec *codec, } } +static void alc256_fixup_dell_xps_13_headphone_noise2(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + + snd_hda_codec_amp_stereo(codec, 0x1a, HDA_INPUT, 0, HDA_AMP_VOLMASK, 1); + snd_hda_override_wcaps(codec, 0x1a, get_wcaps(codec, 0x1a) & ~AC_WCAP_IN_AMP); +} + static void alc269_fixup_limit_int_mic_boost(struct hda_codec *codec, const struct hda_fixup *fix, int action) @@ -5822,6 +5833,7 @@ enum { ALC298_FIXUP_DELL_AIO_MIC_NO_PRESENCE, ALC275_FIXUP_DELL_XPS, ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE, + ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2, ALC293_FIXUP_LENOVO_SPK_NOISE, ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, ALC255_FIXUP_DELL_SPK_NOISE, @@ -6558,6 +6570,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE }, + [ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc256_fixup_dell_xps_13_headphone_noise2, + .chained = true, + .chain_id = ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE + }, [ALC293_FIXUP_LENOVO_SPK_NOISE] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_disable_aamix, @@ -7001,17 +7019,17 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), - SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13 9350", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), + SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13 9350", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2), SND_PCI_QUIRK(0x1028, 0x0706, "Dell Inspiron 7559", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER), SND_PCI_QUIRK(0x1028, 0x0725, "Dell Inspiron 3162", ALC255_FIXUP_DELL_SPK_NOISE), SND_PCI_QUIRK(0x1028, 0x0738, "Dell Precision 5820", ALC269_FIXUP_NO_SHUTUP), - SND_PCI_QUIRK(0x1028, 0x075b, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), + SND_PCI_QUIRK(0x1028, 0x075b, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2), SND_PCI_QUIRK(0x1028, 0x075c, "Dell XPS 27 7760", ALC298_FIXUP_SPK_VOLUME), SND_PCI_QUIRK(0x1028, 0x075d, "Dell AIO", ALC298_FIXUP_SPK_VOLUME), SND_PCI_QUIRK(0x1028, 0x07b0, "Dell Precision 7520", ALC295_FIXUP_DISABLE_DAC3), SND_PCI_QUIRK(0x1028, 0x0798, "Dell Inspiron 17 7000 Gaming", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER), SND_PCI_QUIRK(0x1028, 0x080c, "Dell WYSE", ALC225_FIXUP_DELL_WYSE_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x082a, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), + SND_PCI_QUIRK(0x1028, 0x082a, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2), SND_PCI_QUIRK(0x1028, 0x084b, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB), SND_PCI_QUIRK(0x1028, 0x084e, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB), SND_PCI_QUIRK(0x1028, 0x0871, "Dell Precision 3630", ALC255_FIXUP_DELL_HEADSET_MIC), -- 2.17.1

2 1

[alsa-devel] [RESEND TRIVIAL] sound: Fix Kconfig indentation
by Krzysztof Kozlowski 07 Oct '19

07 Oct '19

Adjust indentation from spaces to tab (+optional two spaces) as in coding style with command like: $ sed -e 's/^ /\t/' -i */Kconfig Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> Acked-by: Mark Brown <broonie(a)kernel.org> --- sound/core/Kconfig | 28 ++++++++++---------- sound/drivers/Kconfig | 20 +++++++------- sound/firewire/Kconfig | 6 ++--- sound/isa/Kconfig | 18 ++++++------- sound/mips/Kconfig | 12 ++++----- sound/pci/Kconfig | 2 +- sound/soc/cirrus/Kconfig | 14 +++++----- sound/soc/codecs/Kconfig | 32 +++++++++++------------ sound/soc/intel/Kconfig | 2 +- sound/soc/intel/boards/Kconfig | 48 +++++++++++++++++----------------- sound/soc/pxa/Kconfig | 16 ++++++------ sound/soc/qcom/Kconfig | 20 +++++++------- sound/soc/samsung/Kconfig | 8 +++--- sound/soc/sof/imx/Kconfig | 12 ++++----- sound/soc/sof/intel/Kconfig | 22 ++++++++-------- sound/soc/xilinx/Kconfig | 20 +++++++------- sound/soc/zte/Kconfig | 12 ++++----- sound/usb/Kconfig | 32 +++++++++++------------ 18 files changed, 162 insertions(+), 162 deletions(-) diff --git a/sound/core/Kconfig b/sound/core/Kconfig index 4ee79ad6ae22..4044c42d8595 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -72,11 +72,11 @@ config SND_PCM_OSS config SND_PCM_OSS_PLUGINS bool "OSS PCM (digital audio) API - Include plugin system" depends on SND_PCM_OSS - default y + default y help - If you disable this option, the ALSA's OSS PCM API will not - support conversion of channels, formats and rates. It will - behave like most of new OSS/Free drivers in 2.4/2.6 kernels. + If you disable this option, the ALSA's OSS PCM API will not + support conversion of channels, formats and rates. It will + behave like most of new OSS/Free drivers in 2.4/2.6 kernels. config SND_PCM_TIMER bool "PCM timer interface" if EXPERT @@ -128,13 +128,13 @@ config SND_SUPPORT_OLD_API or older). config SND_PROC_FS - bool "Sound Proc FS Support" if EXPERT - depends on PROC_FS - default y - help - Say 'N' to disable Sound proc FS, which may reduce code size about - 9KB on x86_64 platform. - If unsure say Y. + bool "Sound Proc FS Support" if EXPERT + depends on PROC_FS + default y + help + Say 'N' to disable Sound proc FS, which may reduce code size about + 9KB on x86_64 platform. + If unsure say Y. config SND_VERBOSE_PROCFS bool "Verbose procfs contents" @@ -142,8 +142,8 @@ config SND_VERBOSE_PROCFS default y help Say Y here to include code for verbose procfs contents (provides - useful information to developers when a problem occurs). On the - other side, it makes the ALSA subsystem larger. + useful information to developers when a problem occurs). On the + other side, it makes the ALSA subsystem larger. config SND_VERBOSE_PRINTK bool "Verbose printk" @@ -164,7 +164,7 @@ config SND_DEBUG_VERBOSE depends on SND_DEBUG help Say Y here to enable extra-verbose debugging messages. - + Let me repeat: it enables EXTRA-VERBOSE DEBUGGING messages. So, say Y only if you are ready to be annoyed. diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig index 09932cc98e9d..15d6d46acf9c 100644 --- a/sound/drivers/Kconfig +++ b/sound/drivers/Kconfig @@ -1,7 +1,7 @@ # SPDX-License-Identifier: GPL-2.0-only config SND_MPU401_UART - tristate - select SND_RAWMIDI + tristate + select SND_RAWMIDI config SND_OPL3_LIB tristate @@ -90,16 +90,16 @@ config SND_DUMMY will be called snd-dummy. config SND_ALOOP - tristate "Generic loopback driver (PCM)" - select SND_PCM - help - Say 'Y' or 'M' to include support for the PCM loopback device. + tristate "Generic loopback driver (PCM)" + select SND_PCM + help + Say 'Y' or 'M' to include support for the PCM loopback device. This module returns played samples back to the user space using the standard ALSA PCM device. The devices are routed 0->1 and - 1->0, where first number is the playback PCM device and second + 1->0, where first number is the playback PCM device and second number is the capture device. Module creates two PCM devices and configured number of substreams (see the pcm_substreams module - parameter). + parameter). The loopback device allows time sychronization with an external timing source using the time shift universal control (+-20% @@ -142,12 +142,12 @@ config SND_MTS64 select SND_RAWMIDI help The ESI Miditerminal 4140 is a 4 In 4 Out MIDI Interface with - additional SMPTE Timecode capabilities for the parallel port. + additional SMPTE Timecode capabilities for the parallel port. Say 'Y' to include support for this device. To compile this driver as a module, chose 'M' here: the module - will be called snd-mts64. + will be called snd-mts64. config SND_SERIAL_U16550 tristate "UART16550 serial MIDI driver" diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index b0a904cdb932..995c2cefc222 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -77,7 +77,7 @@ config SND_BEBOB tristate "BridgeCo DM1000/DM1100/DM1500 with BeBoB firmware" select SND_FIREWIRE_LIB select SND_HWDEP - help + help Say Y here to include support for FireWire devices based on BridgeCo DM1000/DM1100/DM1500 with BeBoB firmware: * Edirol FA-66/FA-101 @@ -111,8 +111,8 @@ config SND_BEBOB * M-Audio FireWire 1814/ProjectMix IO * Digidesign Mbox 2 Pro - To compile this driver as a module, choose M here: the module - will be called snd-bebob. + To compile this driver as a module, choose M here: the module + will be called snd-bebob. config SND_FIREWIRE_DIGI00X tristate "Digidesign Digi 002/003 family support" diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index b690ed937cbe..6ffa48dd5983 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -2,22 +2,22 @@ # ALSA ISA drivers config SND_WSS_LIB - tristate - select SND_PCM + tristate + select SND_PCM select SND_TIMER config SND_SB_COMMON - tristate + tristate config SND_SB8_DSP - tristate - select SND_PCM - select SND_SB_COMMON + tristate + select SND_PCM + select SND_SB_COMMON config SND_SB16_DSP - tristate - select SND_PCM - select SND_SB_COMMON + tristate + select SND_PCM + select SND_SB_COMMON menuconfig SND_ISA bool "ISA sound devices" diff --git a/sound/mips/Kconfig b/sound/mips/Kconfig index 8a33402fd415..b497b803c834 100644 --- a/sound/mips/Kconfig +++ b/sound/mips/Kconfig @@ -14,15 +14,15 @@ config SND_SGI_O2 tristate "SGI O2 Audio" depends on SGI_IP32 select SND_PCM - help - Sound support for the SGI O2 Workstation. + help + Sound support for the SGI O2 Workstation. config SND_SGI_HAL2 - tristate "SGI HAL2 Audio" - depends on SGI_HAS_HAL2 + tristate "SGI HAL2 Audio" + depends on SGI_HAS_HAL2 select SND_PCM - help - Sound support for the SGI Indy and Indigo2 Workstation. + help + Sound support for the SGI Indy and Indigo2 Workstation. endif # SND_MIPS diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 7630f808d087..93bc9bef7641 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -217,7 +217,7 @@ config SND_CMIPCI will be called snd-cmipci. config SND_OXYGEN_LIB - tristate + tristate config SND_OXYGEN tristate "C-Media 8786, 8787, 8788 (Oxygen)" diff --git a/sound/soc/cirrus/Kconfig b/sound/soc/cirrus/Kconfig index 2333efac758a..8039a8febefa 100644 --- a/sound/soc/cirrus/Kconfig +++ b/sound/soc/cirrus/Kconfig @@ -33,13 +33,13 @@ config SND_EP93XX_SOC_AC97 select SND_SOC_AC97_BUS config SND_EP93XX_SOC_SNAPPERCL15 - tristate "SoC Audio support for Bluewater Systems Snapper CL15 module" - depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15 && I2C - select SND_EP93XX_SOC_I2S - select SND_SOC_TLV320AIC23_I2C - help - Say Y or M here if you want to add support for I2S audio on the - Bluewater Systems Snapper CL15 module. + tristate "SoC Audio support for Bluewater Systems Snapper CL15 module" + depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15 && I2C + select SND_EP93XX_SOC_I2S + select SND_SOC_TLV320AIC23_I2C + help + Say Y or M here if you want to add support for I2S audio on the + Bluewater Systems Snapper CL15 module. config SND_EP93XX_SOC_SIMONE tristate "SoC Audio support for Simplemachines Sim.One board" diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index bcac95785493..ed8ba18d93d3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -258,16 +258,16 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM9705 if (SND_SOC_AC97_BUS || SND_SOC_AC97_BUS_NEW) select SND_SOC_WM9712 if (SND_SOC_AC97_BUS || SND_SOC_AC97_BUS_NEW) select SND_SOC_WM9713 if (SND_SOC_AC97_BUS || SND_SOC_AC97_BUS_NEW) - help - Normally ASoC codec drivers are only built if a machine driver which - uses them is also built since they are only usable with a machine - driver. Selecting this option will allow these drivers to be built - without an explicit machine driver for test and development purposes. + help + Normally ASoC codec drivers are only built if a machine driver which + uses them is also built since they are only usable with a machine + driver. Selecting this option will allow these drivers to be built + without an explicit machine driver for test and development purposes. Support for the bus types used to access the codecs to be built must be selected separately. - If unsure select "N". + If unsure select "N". config SND_SOC_88PM860X tristate @@ -571,8 +571,8 @@ config SND_SOC_CS42XX8_I2C # Cirrus Logic CS43130 HiFi DAC config SND_SOC_CS43130 - tristate "Cirrus Logic CS43130 CODEC" - depends on I2C + tristate "Cirrus Logic CS43130 CODEC" + depends on I2C config SND_SOC_CS4341 tristate "Cirrus Logic CS4341 CODEC" @@ -644,19 +644,19 @@ config SND_SOC_L3 tristate config SND_SOC_DA7210 - tristate + tristate config SND_SOC_DA7213 - tristate + tristate config SND_SOC_DA7218 tristate config SND_SOC_DA7219 - tristate + tristate config SND_SOC_DA732X - tristate + tristate config SND_SOC_DA9055 tristate @@ -718,7 +718,7 @@ config SND_SOC_INNO_RK3036 select REGMAP_MMIO config SND_SOC_ISABELLE - tristate + tristate config SND_SOC_LM49453 tristate @@ -989,7 +989,7 @@ config SND_SOC_RT5640 tristate config SND_SOC_RT5645 - tristate + tristate config SND_SOC_RT5651 tristate @@ -1225,7 +1225,7 @@ config SND_SOC_UDA134X tristate config SND_SOC_UDA1380 - tristate + tristate depends on I2C config SND_SOC_WCD9335 @@ -1353,7 +1353,7 @@ config SND_SOC_WM8904 depends on I2C config SND_SOC_WM8940 - tristate + tristate config SND_SOC_WM8955 tristate diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 01c99750212a..597516c0c192 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -113,7 +113,7 @@ config SND_SOC_INTEL_SKYLAKE select SND_SOC_INTEL_CNL select SND_SOC_INTEL_CFL help - This is a backwards-compatible option to select all devices + This is a backwards-compatible option to select all devices supported by the Intel SST/Skylake driver. This option is no longer recommended and will be deprecated when the SOF driver is introduced. Distributions should explicitly diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 5c27f7ab4a5f..882ff36a7c9c 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -3,13 +3,13 @@ menuconfig SND_SOC_INTEL_MACH bool "Intel Machine drivers" depends on SND_SOC_INTEL_SST_TOPLEVEL || SND_SOC_SOF_INTEL_TOPLEVEL help - Intel ASoC Machine Drivers. If you have a Intel machine that - has an audio controller with a DSP and I2S or DMIC port, then - enable this option by saying Y + Intel ASoC Machine Drivers. If you have a Intel machine that + has an audio controller with a DSP and I2S or DMIC port, then + enable this option by saying Y - Note that the answer to this question doesn't directly affect the - kernel: saying N will just cause the configurator to skip all - the questions about Intel ASoC machine drivers. + Note that the answer to this question doesn't directly affect the + kernel: saying N will just cause the configurator to skip all + the questions about Intel ASoC machine drivers. if SND_SOC_INTEL_MACH @@ -114,11 +114,11 @@ config SND_SOC_INTEL_CHT_BSW_RT5672_MACH depends on X86_INTEL_LPSS || COMPILE_TEST select SND_SOC_ACPI select SND_SOC_RT5670 - help - This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell - platforms with RT5672 audio codec. - Say Y or m if you have such a device. This is a recommended option. - If unsure select "N". + help + This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell + platforms with RT5672 audio codec. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". config SND_SOC_INTEL_CHT_BSW_RT5645_MACH tristate "Cherrytrail & Braswell with RT5645/5650 codec" @@ -311,20 +311,20 @@ config SND_SOC_INTEL_KBL_RT5663_MAX98927_MACH If unsure select "N". config SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH - tristate "KBL with RT5663, RT5514 and MAX98927 in I2S Mode" + tristate "KBL with RT5663, RT5514 and MAX98927 in I2S Mode" depends on I2C && ACPI depends on MFD_INTEL_LPSS || COMPILE_TEST - depends on SPI - select SND_SOC_RT5663 - select SND_SOC_RT5514 - select SND_SOC_RT5514_SPI - select SND_SOC_MAX98927 - select SND_SOC_HDAC_HDMI - help - This adds support for ASoC Onboard Codec I2S machine driver. This will - create an alsa sound card for RT5663 + RT5514 + MAX98927. - Say Y or m if you have such a device. This is a recommended option. - If unsure select "N". + depends on SPI + select SND_SOC_RT5663 + select SND_SOC_RT5514 + select SND_SOC_RT5514_SPI + select SND_SOC_MAX98927 + select SND_SOC_HDAC_HDMI + help + This adds support for ASoC Onboard Codec I2S machine driver. This will + create an alsa sound card for RT5663 + RT5514 + MAX98927. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH tristate "KBL with DA7219 and MAX98357A in I2S Mode" @@ -393,7 +393,7 @@ config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH help This adds support for ASoC machine driver for Intel platforms SKL/KBL/BXT/APL with iDisp, HDA audio codecs. - Say Y or m if you have such a device. This is a recommended option. + Say Y or m if you have such a device. This is a recommended option. If unsure select "N". endif ## SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC || SND_SOC_SOF_HDA_AUDIO_CODEC diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 213d4dab0346..295cfffa4646 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -190,14 +190,14 @@ config SND_PXA2XX_SOC_MAGICIAN HTC Magician. config SND_PXA2XX_SOC_MIOA701 - tristate "SoC Audio support for MIO A701" - depends on SND_PXA2XX_SOC && MACH_MIOA701 + tristate "SoC Audio support for MIO A701" + depends on SND_PXA2XX_SOC && MACH_MIOA701 depends on AC97_BUS=n - select SND_PXA2XX_SOC_AC97 - select SND_SOC_WM9713 - help - Say Y if you want to add support for SoC audio on the - MIO A701. + select SND_PXA2XX_SOC_AC97 + select SND_SOC_WM9713 + help + Say Y if you want to add support for SoC audio on the + MIO A701. config SND_PXA2XX_SOC_IMOTE2 tristate "SoC Audio support for IMote 2" @@ -205,7 +205,7 @@ config SND_PXA2XX_SOC_IMOTE2 select SND_PXA2XX_SOC_I2S select SND_SOC_WM8940 help - Say Y if you want to add support for SoC audio on the + Say Y if you want to add support for SoC audio on the IMote 2. config SND_MMP_SOC_BROWNSTONE diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 60086858e920..6530d2462a9e 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -3,8 +3,8 @@ config SND_SOC_QCOM tristate "ASoC support for QCOM platforms" depends on ARCH_QCOM || COMPILE_TEST help - Say Y or M if you want to add support to use audio devices - in Qualcomm Technologies SOC-based platforms. + Say Y or M if you want to add support to use audio devices + in Qualcomm Technologies SOC-based platforms. config SND_SOC_LPASS_CPU tristate @@ -30,17 +30,17 @@ config SND_SOC_STORM select SND_SOC_LPASS_IPQ806X select SND_SOC_MAX98357A help - Say Y or M if you want add support for SoC audio on the - Qualcomm Technologies IPQ806X-based Storm board. + Say Y or M if you want add support for SoC audio on the + Qualcomm Technologies IPQ806X-based Storm board. config SND_SOC_APQ8016_SBC tristate "SoC Audio support for APQ8016 SBC platforms" depends on SND_SOC_QCOM select SND_SOC_LPASS_APQ8016 help - Support for Qualcomm Technologies LPASS audio block in - APQ8016 SOC-based systems. - Say Y if you want to use audio devices on MI2S. + Support for Qualcomm Technologies LPASS audio block in + APQ8016 SOC-based systems. + Say Y if you want to use audio devices on MI2S. config SND_SOC_QCOM_COMMON tristate @@ -93,9 +93,9 @@ config SND_SOC_MSM8996 select SND_SOC_QDSP6 select SND_SOC_QCOM_COMMON help - Support for Qualcomm Technologies LPASS audio block in - APQ8096 SoC-based systems. - Say Y if you want to use audio device on this SoCs + Support for Qualcomm Technologies LPASS audio block in + APQ8096 SoC-based systems. + Say Y if you want to use audio device on this SoCs config SND_SOC_SDM845 tristate "SoC Machine driver for SDM845 boards" diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 638983123d8f..9304177de78a 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -195,10 +195,10 @@ config SND_SOC_ODROID Say Y here to enable audio support for the Odroid XU3/XU4. config SND_SOC_ARNDALE_RT5631_ALC5631 - tristate "Audio support for RT5631(ALC5631) on Arndale Board" - depends on I2C - select SND_SAMSUNG_I2S - select SND_SOC_RT5631 + tristate "Audio support for RT5631(ALC5631) on Arndale Board" + depends on I2C + select SND_SAMSUNG_I2S + select SND_SOC_RT5631 config SND_SOC_SAMSUNG_TM2_WM5110 tristate "SoC I2S Audio support for WM5110 on TM2 board" diff --git a/sound/soc/sof/imx/Kconfig b/sound/soc/sof/imx/Kconfig index 5acae75f5750..6315fba8ce71 100644 --- a/sound/soc/sof/imx/Kconfig +++ b/sound/soc/sof/imx/Kconfig @@ -5,9 +5,9 @@ config SND_SOC_SOF_IMX_TOPLEVEL depends on ARM64|| COMPILE_TEST depends on SND_SOC_SOF_OF help - This adds support for Sound Open Firmware for NXP i.MX platforms. - Say Y if you have such a device. - If unsure select "N". + This adds support for Sound Open Firmware for NXP i.MX platforms. + Say Y if you have such a device. + If unsure select "N". if SND_SOC_SOF_IMX_TOPLEVEL @@ -16,8 +16,8 @@ config SND_SOC_SOF_IMX8 depends on IMX_SCU depends on IMX_DSP help - This adds support for Sound Open Firmware for NXP i.MX8 platforms - Say Y if you have such a device. - If unsure select "N". + This adds support for Sound Open Firmware for NXP i.MX8 platforms + Say Y if you have such a device. + If unsure select "N". endif ## SND_SOC_SOF_IMX_IMX_TOPLEVEL diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index d62f51d33be1..56f16e4543c0 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -36,7 +36,7 @@ config SND_SOC_SOF_INTEL_PCI config SND_SOC_SOF_INTEL_HIFI_EP_IPC tristate help - This option is not user-selectable but automagically handled by + This option is not user-selectable but automagically handled by 'select' statements at a higher level config SND_SOC_SOF_INTEL_ATOM_HIFI_EP @@ -217,31 +217,31 @@ config SND_SOC_SOF_COMETLAKE_H_SUPPORT config SND_SOC_SOF_TIGERLAKE_SUPPORT bool "SOF support for Tigerlake" help - This adds support for Sound Open Firmware for Intel(R) platforms - using the Tigerlake processors. - Say Y if you have such a device. - If unsure select "N". + This adds support for Sound Open Firmware for Intel(R) platforms + using the Tigerlake processors. + Say Y if you have such a device. + If unsure select "N". config SND_SOC_SOF_TIGERLAKE tristate select SND_SOC_SOF_HDA_COMMON help - This option is not user-selectable but automagically handled by + This option is not user-selectable but automagically handled by 'select' statements at a higher level config SND_SOC_SOF_ELKHARTLAKE_SUPPORT bool "SOF support for ElkhartLake" help - This adds support for Sound Open Firmware for Intel(R) platforms - using the ElkhartLake processors. - Say Y if you have such a device. - If unsure select "N". + This adds support for Sound Open Firmware for Intel(R) platforms + using the ElkhartLake processors. + Say Y if you have such a device. + If unsure select "N". config SND_SOC_SOF_ELKHARTLAKE tristate select SND_SOC_SOF_HDA_COMMON help - This option is not user-selectable but automagically handled by + This option is not user-selectable but automagically handled by 'select' statements at a higher level config SND_SOC_SOF_HDA_COMMON diff --git a/sound/soc/xilinx/Kconfig b/sound/soc/xilinx/Kconfig index 69973179ef15..1d3586b68db7 100644 --- a/sound/soc/xilinx/Kconfig +++ b/sound/soc/xilinx/Kconfig @@ -9,15 +9,15 @@ config SND_SOC_XILINX_I2S encapsulates PCM in AES format and sends AES data. config SND_SOC_XILINX_AUDIO_FORMATTER - tristate "Audio support for the the Xilinx audio formatter" - help - Select this option to enable Xilinx audio formatter - support. This provides DMA platform device support for - audio functionality. + tristate "Audio support for the the Xilinx audio formatter" + help + Select this option to enable Xilinx audio formatter + support. This provides DMA platform device support for + audio functionality. config SND_SOC_XILINX_SPDIF - tristate "Audio support for the the Xilinx SPDIF" - help - Select this option to enable Xilinx SPDIF Audio. - This provides playback and capture of SPDIF audio in - AES format. + tristate "Audio support for the the Xilinx SPDIF" + help + Select this option to enable Xilinx SPDIF Audio. + This provides playback and capture of SPDIF audio in + AES format. diff --git a/sound/soc/zte/Kconfig b/sound/soc/zte/Kconfig index a7842e4b791c..a23d4f13ca19 100644 --- a/sound/soc/zte/Kconfig +++ b/sound/soc/zte/Kconfig @@ -18,9 +18,9 @@ config ZX_I2S ZTE ZX I2S interface config ZX_TDM - tristate "ZTE ZX TDM Driver Support" - depends on COMMON_CLK - select SND_SOC_GENERIC_DMAENGINE_PCM - help - Say Y or M if you want to add support for codecs attached to the - ZTE ZX TDM interface + tristate "ZTE ZX TDM Driver Support" + depends on COMMON_CLK + select SND_SOC_GENERIC_DMAENGINE_PCM + help + Say Y or M if you want to add support for codecs attached to the + ZTE ZX TDM interface diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index e2c53a0841da..059242f15d75 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -107,24 +107,24 @@ config SND_USB_US122L will be called snd-usb-us122l. config SND_USB_6FIRE - tristate "TerraTec DMX 6Fire USB" - select FW_LOADER - select BITREVERSE - select SND_RAWMIDI - select SND_PCM - select SND_VMASTER - help - Say Y here to include support for TerraTec 6fire DMX USB interface. - - You will need firmware files in order to be able to use the device - after it has been coldstarted. An install script for the firmware - and further help can be found at - http://sixfireusb.sourceforge.net + tristate "TerraTec DMX 6Fire USB" + select FW_LOADER + select BITREVERSE + select SND_RAWMIDI + select SND_PCM + select SND_VMASTER + help + Say Y here to include support for TerraTec 6fire DMX USB interface. + + You will need firmware files in order to be able to use the device + after it has been coldstarted. An install script for the firmware + and further help can be found at + http://sixfireusb.sourceforge.net config SND_USB_HIFACE - tristate "M2Tech hiFace USB-SPDIF driver" - select SND_PCM - help + tristate "M2Tech hiFace USB-SPDIF driver" + select SND_PCM + help Select this option to include support for M2Tech hiFace USB-SPDIF interface. -- 2.17.1

2 1

[alsa-devel] [PATCH v2] ALSA: hda: add Intel DSP configuration / probe code
by Jaroslav Kysela 06 Oct '19

06 Oct '19

For distributions, we need one place where we can decide which driver will be activated for the auto-configation of the Intel's HDA hardware with DSP. Actually, we cover three drivers: * Legacy HDA * Intel SST * Intel Sound Open Firmware (SOF) All those drivers registers similar PCI IDs, so the first driver probed from the PCI stack can win. But... it is not guaranteed that the correct driver wins. This commit changes Intel's NHLT ACPI module to a common DSP probe module for the Intel's hardware. All above sound drivers calls this code. The user can force another behaviour using the module parameter 'dsp_driver' located in the 'snd-intel-dspcfg' module. This change allows to add specific dmi checks for the specific systems like proposed in the pull request: https://github.com/thesofproject/linux/pull/927 Tested on Lenovo Carbon X1 7th gen. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Cezary Rojewski <cezary.rojewski(a)intel.com> --- include/sound/intel-dsp-config.h | 34 ++++++++++ sound/hda/Kconfig | 10 ++- sound/hda/Makefile | 7 ++- sound/hda/intel-dsp-config.c | 105 +++++++++++++++++++++++++++++++ sound/hda/intel-nhlt.c | 3 - sound/pci/hda/Kconfig | 11 +--- sound/pci/hda/hda_intel.c | 48 +++++--------- sound/soc/intel/Kconfig | 2 +- sound/soc/intel/skylake/skl.c | 19 ++---- sound/soc/sof/intel/Kconfig | 2 +- sound/soc/sof/sof-pci-dev.c | 6 ++ 11 files changed, 179 insertions(+), 68 deletions(-) create mode 100644 include/sound/intel-dsp-config.h create mode 100644 sound/hda/intel-dsp-config.c diff --git a/include/sound/intel-dsp-config.h b/include/sound/intel-dsp-config.h new file mode 100644 index 000000000000..c36622bee3f8 --- /dev/null +++ b/include/sound/intel-dsp-config.h @@ -0,0 +1,34 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * intel-dsp-config.h - Intel DSP config + * + * Copyright (c) 2019 Jaroslav Kysela <perex(a)perex.cz> + */ + +#ifndef __INTEL_DSP_CONFIG_H__ +#define __INTEL_DSP_CONFIG_H__ + +struct pci_dev; + +enum { + SND_INTEL_DSP_DRIVER_ANY = 0, + SND_INTEL_DSP_DRIVER_LEGACY, + SND_INTEL_DSP_DRIVER_SST, + SND_INTEL_DSP_DRIVER_SOF, + SND_INTEL_DSP_DRIVER_LAST = SND_INTEL_DSP_DRIVER_SOF +}; + +#if IS_ENABLED(CONFIG_SND_INTEL_DSP_CONFIG) + +int snd_intel_dsp_driver_probe(struct pci_dev *pci); + +#else + +static inline int snd_intel_dsp_driver_probe(struct pci_dev *pci) +{ + return SND_INTEL_DSP_DRIVER_ANY; +} + +#endif + +#endif diff --git a/sound/hda/Kconfig b/sound/hda/Kconfig index 3d33fc1757ba..b0c88fe040ee 100644 --- a/sound/hda/Kconfig +++ b/sound/hda/Kconfig @@ -34,6 +34,12 @@ config SND_HDA_PREALLOC_SIZE via a proc file (/proc/asound/card*/pcm*/sub*/prealloc), too. config SND_INTEL_NHLT - tristate + bool # this config should be selected only for Intel ACPI platforms. - # A fallback is provided so that the code compiles in all cases. \ No newline at end of file + # A fallback is provided so that the code compiles in all cases. + +config SND_INTEL_DSP_CONFIG + tristate + select SND_INTEL_NHLT if ACPI + # this config should be selected only for Intel DSP platforms. + # A fallback is provided so that the code compiles in all cases. diff --git a/sound/hda/Makefile b/sound/hda/Makefile index 8560f6ef1b19..c593253c9829 100644 --- a/sound/hda/Makefile +++ b/sound/hda/Makefile @@ -14,5 +14,8 @@ obj-$(CONFIG_SND_HDA_CORE) += snd-hda-core.o #extended hda obj-$(CONFIG_SND_HDA_EXT_CORE) += ext/ -snd-intel-nhlt-objs := intel-nhlt.o -obj-$(CONFIG_SND_INTEL_NHLT) += snd-intel-nhlt.o +snd-intel-dspcfg-objs := intel-dsp-config.o +ifneq ($(CONFIG_ACPI),) + snd-intel-dspcfg-objs += intel-nhlt.o +endif +obj-$(CONFIG_SND_INTEL_DSP_CONFIG) += snd-intel-dspcfg.o diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c new file mode 100644 index 000000000000..f331e2216fe0 --- /dev/null +++ b/sound/hda/intel-dsp-config.c @@ -0,0 +1,105 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2019 Jaroslav Kysela <perex(a)perex.cz> + +#include <sound/core.h> +#include <linux/module.h> +#include <linux/pci.h> +#include <sound/intel-nhlt.h> +#include <sound/intel-dsp-config.h> + +static int dsp_driver; + +module_param(dsp_driver, int, 0444); +MODULE_PARM_DESC(dsp_driver, "Force the DSP driver for Intel DSP (0=auto, 1=noDSP, 2=legacy, 3=SST, 4=SOF)"); + +static const u16 sof_skl_table[] = { +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_LP) + 0x02c8, /* Cometlake-LP */ +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_H) + 0x06c8, /* Cometlake-H */ +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_GEMINILAKE) + 0x3198, /* Geminilake */ +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_CANNONLAKE) + 0x9dc8, /* Cannonlake */ +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COFFEELAKE) + 0xa348, /* Coffelake */ +#endif +}; + +static int snd_intel_dsp_check_device(u16 device, const u16 *table, u32 len) +{ + for (; len > 0; len--, table++) { + if (*table == device) + return 1; + } + return 0; +} + +static int snd_intel_dsp_check_dmic(struct pci_dev *pci) +{ + struct nhlt_acpi_table *nhlt; + int ret = 0; + + if (snd_intel_dsp_check_device(pci->device, sof_skl_table, ARRAY_SIZE(sof_skl_table))) { + nhlt = intel_nhlt_init(&pci->dev); + if (nhlt) { + if (intel_nhlt_get_dmic_geo(&pci->dev, nhlt)) + ret = 1; + intel_nhlt_free(nhlt); + } + } + return ret; +} + +int snd_intel_dsp_driver_probe(struct pci_dev *pci) +{ + if (dsp_driver > 0 && dsp_driver <= SND_INTEL_DSP_DRIVER_LAST) + return dsp_driver; + + /* Intel vendor only */ + if (snd_BUG_ON(pci->vendor != 0x8086)) + return SND_INTEL_DSP_DRIVER_ANY; + + /* + * detect DSP by checking class/subclass/prog-id information + * class=04 subclass 03 prog-if 00: no DSP, use legacy driver + * class=04 subclass 01 prog-if 00: DSP is present + * (and may be required e.g. for DMIC or SSP support) + * class=04 subclass 03 prog-if 80: use DSP or legacy mode + */ + if (pci->class == 0x040300) + return SND_INTEL_DSP_DRIVER_LEGACY; + if (pci->class != 0x040100 && pci->class != 0x040380) { + dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, selecting HDA legacy driver\n", pci->class); + return SND_INTEL_DSP_DRIVER_LEGACY; + } + + dev_info(&pci->dev, "DSP detected with PCI class/subclass/prog-if info 0x%06x\n", pci->class); + + /* Prefer SST driver for Broxton-P (Appololake) */ +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_APL) || IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE) + if (pci->device == 0x5a98) +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_APL) + return SND_INTEL_DSP_DRIVER_SST; +#else + return SND_INTEL_DSP_DRIVER_SOF; +#endif +#endif + + /* DMIC check for Skylake+ */ + if (snd_intel_dsp_check_dmic(pci)) { + dev_info(&pci->dev, "Digital mics found on Skylake+ platform, using SOF driver\n"); + return SND_INTEL_DSP_DRIVER_SOF; + } + + return SND_INTEL_DSP_DRIVER_ANY; +} + +EXPORT_SYMBOL_GPL(snd_intel_dsp_driver_probe); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel DSP config driver"); diff --git a/sound/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c index daede96f28ee..097ff6c10099 100644 --- a/sound/hda/intel-nhlt.c +++ b/sound/hda/intel-nhlt.c @@ -102,6 +102,3 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt) return dmic_geo; } EXPORT_SYMBOL_GPL(intel_nhlt_get_dmic_geo); - -MODULE_LICENSE("GPL v2"); -MODULE_DESCRIPTION("Intel NHLT driver"); diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index dae47a45b2b8..bd48335d09d7 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -12,7 +12,7 @@ config SND_HDA_INTEL tristate "HD Audio PCI" depends on SND_PCI select SND_HDA - select SND_INTEL_NHLT if ACPI + select SND_INTEL_DSP_CONFIG help Say Y here to include support for Intel "High Definition Audio" (Azalia) and its compatible devices. @@ -23,15 +23,6 @@ config SND_HDA_INTEL To compile this driver as a module, choose M here: the module will be called snd-hda-intel. -config SND_HDA_INTEL_DETECT_DMIC - bool "DMIC detection and probe abort" - depends on SND_HDA_INTEL - help - Say Y to detect digital microphones on SKL+ devices. DMICs - cannot be handled by the HDaudio legacy driver and are - currently only supported by the SOF driver. - If unsure say N. - config SND_HDA_TEGRA tristate "NVIDIA Tegra HD Audio" depends on ARCH_TEGRA diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 240f4ca76391..7e262678abeb 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -46,7 +46,7 @@ #include <sound/initval.h> #include <sound/hdaudio.h> #include <sound/hda_i915.h> -#include <sound/intel-nhlt.h> +#include <sound/intel-dsp-config.h> #include <linux/vgaarb.h> #include <linux/vga_switcheroo.h> #include <linux/firmware.h> @@ -124,7 +124,7 @@ static char *patch[SNDRV_CARDS]; static bool beep_mode[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = CONFIG_SND_HDA_INPUT_BEEP_MODE}; #endif -static bool dmic_detect = IS_ENABLED(CONFIG_SND_HDA_INTEL_DETECT_DMIC); +static bool no_dsp_driver; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for Intel HD audio interface."); @@ -159,8 +159,8 @@ module_param_array(beep_mode, bool, NULL, 0444); MODULE_PARM_DESC(beep_mode, "Select HDA Beep registration mode " "(0=off, 1=on) (default=1)."); #endif -module_param(dmic_detect, bool, 0444); -MODULE_PARM_DESC(dmic_detect, "DMIC detect on SKL+ platforms"); +module_param(no_dsp_driver, bool, 0444); +MODULE_PARM_DESC(no_dsp_driver, "Force this driver to be used and bypass the DSP driver selection"); #ifdef CONFIG_PM static int param_set_xint(const char *val, const struct kernel_param *kp); @@ -2020,25 +2020,6 @@ static const struct hda_controller_ops pci_hda_ops = { .position_check = azx_position_check, }; -static int azx_check_dmic(struct pci_dev *pci, struct azx *chip) -{ - struct nhlt_acpi_table *nhlt; - int ret = 0; - - if (chip->driver_type == AZX_DRIVER_SKL && - pci->class != 0x040300) { - nhlt = intel_nhlt_init(&pci->dev); - if (nhlt) { - if (intel_nhlt_get_dmic_geo(&pci->dev, nhlt)) { - ret = -ENODEV; - dev_info(&pci->dev, "Digital mics found on Skylake+ platform, aborting probe\n"); - } - intel_nhlt_free(nhlt); - } - } - return ret; -} - static int azx_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { @@ -2056,6 +2037,16 @@ static int azx_probe(struct pci_dev *pci, return -ENOENT; } + /* + * stop probe if another Intel's DSP driver should be activated + */ + if (!no_dsp_driver) { + err = snd_intel_dsp_driver_probe(pci); + if (err != SND_INTEL_DSP_DRIVER_ANY && + err != SND_INTEL_DSP_DRIVER_LEGACY) + return -ENODEV; + } + err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, 0, &card); if (err < 0) { @@ -2069,17 +2060,6 @@ static int azx_probe(struct pci_dev *pci, card->private_data = chip; hda = container_of(chip, struct hda_intel, chip); - /* - * stop probe if digital microphones detected on Skylake+ platform - * with the DSP enabled. This is an opt-in behavior defined at build - * time or at run-time with a module parameter - */ - if (dmic_detect) { - err = azx_check_dmic(pci, chip); - if (err < 0) - goto out_free; - } - pci_set_drvdata(pci, card); err = register_vga_switcheroo(chip); diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 01c99750212a..9ad89d56092b 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -215,7 +215,7 @@ config SND_SOC_INTEL_SKYLAKE_COMMON select SND_SOC_INTEL_SST select SND_SOC_HDAC_HDA if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC select SND_SOC_ACPI_INTEL_MATCH - select SND_INTEL_NHLT if ACPI + select SND_INTEL_DSP_CONFIG help If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/ GeminiLake or CannonLake platform with the DSP enabled in the BIOS diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 141dbbf975ac..58ba3e9469ba 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -27,6 +27,7 @@ #include <sound/hda_i915.h> #include <sound/hda_codec.h> #include <sound/intel-nhlt.h> +#include <sound/intel-dsp-config.h> #include "skl.h" #include "skl-sst-dsp.h" #include "skl-sst-ipc.h" @@ -987,22 +988,10 @@ static int skl_probe(struct pci_dev *pci, switch (skl_pci_binding) { case SND_SKL_PCI_BIND_AUTO: - /* - * detect DSP by checking class/subclass/prog-id information - * class=04 subclass 03 prog-if 00: no DSP, use legacy driver - * class=04 subclass 01 prog-if 00: DSP is present - * (and may be required e.g. for DMIC or SSP support) - * class=04 subclass 03 prog-if 80: use DSP or legacy mode - */ - if (pci->class == 0x040300) { - dev_info(&pci->dev, "The DSP is not enabled on this platform, aborting probe\n"); + err = snd_intel_dsp_driver_probe(pci); + if (err != SND_INTEL_DSP_DRIVER_ANY && + err != SND_INTEL_DSP_DRIVER_SST) return -ENODEV; - } - if (pci->class != 0x040100 && pci->class != 0x040380) { - dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, aborting probe\n", pci->class); - return -ENODEV; - } - dev_info(&pci->dev, "DSP detected with PCI class/subclass/prog-if info 0x%06x\n", pci->class); break; case SND_SKL_PCI_BIND_LEGACY: dev_info(&pci->dev, "Module parameter forced binding with HDaudio legacy, aborting probe\n"); diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index 479ba249e219..8a5d5c0f95f2 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -286,7 +286,7 @@ config SND_SOC_SOF_HDA tristate select SND_HDA_EXT_CORE if SND_SOC_SOF_HDA_LINK select SND_SOC_HDAC_HDA if SND_SOC_SOF_HDA_AUDIO_CODEC - select SND_INTEL_NHLT if ACPI + select SND_INTEL_DSP_CONFIG help This option is not user-selectable but automagically handled by 'select' statements at a higher level diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index d66412a77873..3a9e0e2a150d 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -12,6 +12,7 @@ #include <linux/module.h> #include <linux/pci.h> #include <linux/pm_runtime.h> +#include <sound/intel-dsp-config.h> #include <sound/soc-acpi.h> #include <sound/soc-acpi-intel-match.h> #include <sound/sof.h> @@ -277,6 +278,11 @@ static int sof_pci_probe(struct pci_dev *pci, const struct snd_sof_dsp_ops *ops; int ret; + ret = snd_intel_dsp_driver_probe(pci); + if (ret != SND_INTEL_DSP_DRIVER_ANY && + ret != SND_INTEL_DSP_DRIVER_SOF) + return -ENODEV; + dev_dbg(&pci->dev, "PCI DSP detected"); /* get ops for platform */ -- 2.20.1

2 2

[alsa-devel] [PATCH -next] ASoC: tas2770: fix platform_no_drv_owner.cocci warnings
by YueHaibing 06 Oct '19

06 Oct '19

Remove .owner field if calls are used which set it automatically Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/tas2770.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c index 9da88ccb1d51..dbbb21fe0548 100644 --- a/sound/soc/codecs/tas2770.c +++ b/sound/soc/codecs/tas2770.c @@ -793,7 +793,6 @@ MODULE_DEVICE_TABLE(of, tas2770_of_match); static struct i2c_driver tas2770_i2c_driver = { .driver = { .name = "tas2770", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(tas2770_of_match), }, .probe = tas2770_i2c_probe,

1 0

[alsa-devel] [PATCH 00/10] ASoC: mediatek: mt8183-mt6358-ts3a227-max98357: support WoV
by Tzung-Bi Shih 05 Oct '19

05 Oct '19

This series makes mt6358, cros_ec_codec, and mt8183-mt6358-ts3a227-max98357 support WoV (wake on voice). The first 3 commits are some cleanups and refactors. It looks like breaking the existing interface. But please be noticed that, the cros_ec_codec has not used by any real device yet. The refactor is very necessary to keep the style consistent and for easier to further extend and maintain. platform/chrome: cros_ec: remove unused EC feature ASoC: cros_ec_codec: refactor I2S RX ASoC: cros_ec_codec: extract DMIC EC command from I2S RX The 4th commit extends the feature offered from EC codec. platform/chrome: cros_ec: add common commands for EC codec The 5th commit changes the behavior of setting and getting DMIC gains. ASoC: cros_ec_codec: read max DMIC gain from EC codec The 6th and 7th commit make cros_ec_codec support WoV. ASoC: dt-bindings: cros_ec_codec: add SHM bindings ASoC: cros_ec_codec: support WoV The 8th commit sets necessary registers on mt6358 to support WoV. ASoC: mediatek: mt6358: support WoV The last 2 commit make machine driver mt8183-mt6358-ts3a227-max98357 support WoV if ec-codec is in DTS. ASoC: dt-bindings: mt8183: add ec-codec ASoC: mediatek: mt8183: support WoV Tzung-Bi Shih (10): platform/chrome: cros_ec: remove unused EC feature ASoC: cros_ec_codec: refactor I2S RX ASoC: cros_ec_codec: extract DMIC EC command from I2S RX platform/chrome: cros_ec: add common commands for EC codec ASoC: cros_ec_codec: read max DMIC gain from EC codec ASoC: dt-bindings: cros_ec_codec: add SHM bindings ASoC: cros_ec_codec: support WoV ASoC: mediatek: mt6358: support WoV ASoC: dt-bindings: mt8183: add ec-codec ASoC: mediatek: mt8183: support WoV .../bindings/sound/google,cros-ec-codec.txt | 24 +- .../sound/mt8183-mt6358-ts3a227-max98357.txt | 3 + drivers/platform/chrome/cros_ec_trace.c | 5 +- .../linux/platform_data/cros_ec_commands.h | 285 ++++- sound/soc/codecs/cros_ec_codec.c | 1125 +++++++++++++---- sound/soc/codecs/mt6358.c | 105 ++ sound/soc/mediatek/Kconfig | 1 + .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 70 +- 8 files changed, 1292 insertions(+), 326 deletions(-) -- 2.23.0.581.g78d2f28ef7-goog

2 11

[alsa-devel] alsa compliance test: H340 (USB audio) playback / capture rate asymmetry bug?
by Hin-Tak Leung 05 Oct '19

05 Oct '19

I am using Takashi Iwai's tree grafted onto mainline as DKMS ( as in https://github.com/HinTak/sound-usb-dkms ). Running the alsa compliance test commit f6167eb77d038b5b7a0d39645e7f2ae7fee6fdc0 (origin/stabilize-12464.B, origin/release-R78-12499.B) Capture all passed, but play back failed a couple, regarding the sample rate. It is a small head set with stereo head phone and a mic. ID 046d:0a38 Logitech, Inc. Headset H340 The question is, for such a cheap headset, why would the playback rate and the capture rate to be different? For many applications/ usages, with a device that's most capture and playback capable, we would like the rates to agree - both pass, or both fail in the same direction? Is this a hardware or software issue? Or, somebody suggested, I haven't looked, issue with the alsa compliance test itself, possibly regarding frame counts of usb devices? detail below. 5 passed, 0 failed Device Information Name: hw:CARD=H340 Stream: CAPTURE Format: ['S16_LE'] Channels range: [2, 2] Rate: [44100] Period_size range: [45, 131072] Buffer_size range: [90, 262144] Test Params Set channels 2: pass Set format S16_LE: pass Set rate 44100: pass Test Rates Set rate 44100: pass Test All Pairs Set channels 2, format S16_LE, rate 44100: pass 3 passed, 2 failed Device Information Name: hw:CARD=H340 Stream: PLAYBACK Format: ['S16_LE'] Channels range: [2, 2] Rate: [44100] Period_size range: [45, 131072] Buffer_size range: [90, 262144] Test Params Set channels 2: pass Set format S16_LE: pass Set rate 44100: pass Test Rates Set rate 44100: fail - Expected rate is 44100.000000, measure 44092.918362, difference 7.081638 > threshold 4.410000 Test All Pairs Set channels 2, format S16_LE, rate 44100: fail - Expected rate is 44100.000000, measure 44093.049192, difference 6.950808 > threshold 4.410000

4 7

[alsa-devel] [RFC PATCH 0/2] Alter the trigger order for FE/BE DAI's based on command
by Ranjani Sridharan 04 Oct '19

04 Oct '19

Currently, the trigger orders SND_SOC_DPCM_TRIGGER_PRE/POST determine the order in which FE DAI and BE DAI are triggered. In the case of SND_SOC_DPCM_TRIGGER_PRE, the FE DAI is triggered before the BE DAI and in the case of SND_SOC_DPCM_TRIGGER_POST, the BE DAI is triggered before the FE DAI. And this order remains the same irrespective of the trigger command. In the case of the SOF driver, during playback, the FW expects the BE DAI to be triggered before the FE DAI during the START trigger. The BE DAI trigger handles the starting of Link DMA and so it must be started before the FE DAI is started to prevent xruns during pause/release. This can be addressed by setting the trigger order for the FE dai link to SND_SOC_DPCM_TRIGGER_POST. But during the STOP trigger, the FW expects the FE DAI to be triggered before the BE DAI. Retaining the same order during the START and STOP commands, results in FW error as the DAI component in the FW is still active. The issue can be fixed by mirroring the trigger order of FE and BE DAI's during the START and STOP trigger. So, with the trigger order set to SND_SOC_DPCM_TRIGGER_PRE, the FE DAI will be trigger first during SNDRV_PCM_TRIGGER_START/STOP/RESUME and the BE DAI will be triggered first during the STOP/SUSPEND/PAUSE commands. Conversely, with the trigger order set to SND_SOC_DPCM_TRIGGER_POST, the BE DAI will be triggered first during the SNDRV_PCM_TRIGGER_START/STOP/RESUME commands and the FE DAI will be triggered first during the SNDRV_PCM_TRIGGER_STOP/SUSPEND/PAUSE commands. My first thought was to use a BESPOKE trigger for the SOF driver but looking more into the implementation of the bespoke trigger in soc_pcm_bespoke_trigger() didnt indicate it had much to do with the ordering of the FE/BE DAI's but rather just use the bespoke trigger callbacks in the DAI driver. More details on the SOF issue can be found in: Github Issue: https://github.com/thesofproject/linux/issues/1160 Ranjani Sridharan (2): ASoC: pcm: update FE/BE trigger order based on the command ASoC: SOF: topology: set trigger order for FE DAI link sound/soc/soc-pcm.c | 99 +++++++++++++++++++++++++++++----------- sound/soc/sof/topology.c | 4 ++ 2 files changed, 76 insertions(+), 27 deletions(-) -- 2.17.1

3 5

[alsa-devel] [PATCH] ASoC: tas2770: Fix snd_soc_update_bits error handling
by Dan Murphy 04 Oct '19

04 Oct '19

According the documentation for snd_soc_update_bits the API will return a 1 if the update was successful with a value change, a 0 if the update was successful with no value change or a negative if the command just failed. So the value of return in the driver needs to be checked for being less then 0 or the caller may indicate failure when the value actually changed. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- arch/arm/configs/omap2plus_defconfig | 1 + sound/soc/codecs/tas2770.c | 44 ++++++++++++++++++---------- 2 files changed, 30 insertions(+), 15 deletions(-) diff --git a/arch/arm/configs/omap2plus_defconfig b/arch/arm/configs/omap2plus_defconfig index 64eb896907bf..8a57185112a2 100644 --- a/arch/arm/configs/omap2plus_defconfig +++ b/arch/arm/configs/omap2plus_defconfig @@ -395,6 +395,7 @@ CONFIG_SND_SOC_OMAP_ABE_TWL6040=m CONFIG_SND_SOC_OMAP_HDMI=m CONFIG_SND_SOC_CPCAP=m CONFIG_SND_SOC_TLV320AIC23_I2C=m +CONFIG_SND_SOC_TAS2770=m CONFIG_SND_SIMPLE_CARD=m CONFIG_SND_AUDIO_GRAPH_CARD=m CONFIG_HID_GENERIC=m diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c index dbbb21fe0548..774c8f2833c8 100644 --- a/sound/soc/codecs/tas2770.c +++ b/sound/soc/codecs/tas2770.c @@ -82,7 +82,8 @@ static int tas2770_codec_suspend(struct snd_soc_component *component) TAS2770_PWR_CTRL, TAS2770_PWR_CTRL_MASK, TAS2770_PWR_CTRL_SHUTDOWN); - if (ret) + + if (ret < 0) return ret; return 0; @@ -96,8 +97,9 @@ static int tas2770_codec_resume(struct snd_soc_component *component) TAS2770_PWR_CTRL, TAS2770_PWR_CTRL_MASK, TAS2770_PWR_CTRL_ACTIVE); - if (ret) - return -EINVAL; + + if (ret < 0) + return ret; return 0; } @@ -149,7 +151,10 @@ static int tas2770_dac_event(struct snd_soc_dapm_widget *w, } end: - return ret; + if (ret < 0) + return ret; + + return 0; } static const struct snd_kcontrol_new isense_switch = @@ -199,7 +204,10 @@ static int tas2770_mute(struct snd_soc_dai *dai, int mute) TAS2770_PWR_CTRL_MASK, TAS2770_PWR_CTRL_ACTIVE); - return ret; + if (ret < 0) + return ret; + + return 0; } static int tas2770_set_bitwidth(struct tas2770_priv *tas2770, int bitwidth) @@ -252,7 +260,10 @@ static int tas2770_set_bitwidth(struct tas2770_priv *tas2770, int bitwidth) tas2770->i_sense_slot); end: - return ret; + if (ret < 0) + return ret; + + return 0; } static int tas2770_set_samplerate(struct tas2770_priv *tas2770, int samplerate) @@ -344,9 +355,11 @@ static int tas2770_set_samplerate(struct tas2770_priv *tas2770, int samplerate) } end: - if (!ret) - tas2770->sampling_rate = samplerate; - return ret; + if (ret < 0) + return ret; + + tas2770->sampling_rate = samplerate; + return 0; } static int tas2770_hw_params(struct snd_pcm_substream *substream, @@ -426,7 +439,7 @@ static int tas2770_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG1, TAS2770_TDM_CFG_REG1_MASK, (tdm_rx_start_slot << TAS2770_TDM_CFG_REG1_51_SHIFT)); - if (ret) + if (ret < 0) return ret; value = snd_soc_component_read32(component, TAS2770_TDM_CFG_REG3); @@ -472,12 +485,12 @@ static int tas2770_set_dai_tdm_slot(struct snd_soc_dai *dai, ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG3, TAS2770_TDM_CFG_REG3_30_MASK, (left_slot << TAS2770_TDM_CFG_REG3_30_SHIFT)); - if (ret) + if (ret < 0) return ret; ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG3, TAS2770_TDM_CFG_REG3_RXS_MASK, (right_slot << TAS2770_TDM_CFG_REG3_RXS_SHIFT)); - if (ret) + if (ret < 0) return ret; switch (slot_width) { @@ -511,10 +524,11 @@ static int tas2770_set_dai_tdm_slot(struct snd_soc_dai *dai, ret = -EINVAL; } - if (!ret) - tas2770->slot_width = slot_width; + if (ret < 0) + return ret; - return ret; + tas2770->slot_width = slot_width; + return 0; } static struct snd_soc_dai_ops tas2770_dai_ops = { -- 2.22.0.214.g8dca754b1e

2 2