July 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH v2 00/16] ASoC: qcom: Add AudioReach support
by Srinivas Kandagatla 30 Jul '21

30 Jul '21

This patchset adds ASoC driver support to configure signal processing framework ("AudioReach") which is integral part of Qualcomm next generation audio SDK and will be deployed on upcoming Qualcomm chipsets. It makes use of ASoC Topology to load graphs on to the DSP which is then managed by APM (Audio Processing Manager) service to prepare/start/stop. Here is simpified high-level block diagram of AudioReach: ___________________________________________________________ | CPU (Application Processor) | | +---------+ +---------+ +---------+ | | | q6apm | | q6apm | | q6afe | | | | dais | <------> | | <-----> | bedais | | | +---------+ +---------+ +---------+ | | ^ ^ | | | | +---------+ | | +---------+ v +---------->|topology | | | | q6prm | +---------+ | | | | | |<-------->| GPR | +---------+ | | +---------+ +---------+ | | ^ | |____________________________|______________________________| | | RPMSG (IPC over GLINK) ____________________________|______________________________ | | | | +-----------------------+ | | | | | | v v q6 (Audio DSP) | |+-----+ +----------------------------------+ | || PRM | | APM (Audio Processing Manager) | | |+-----+ | . Graph Management | | | | . Command Handing | | | | . Event Management | | | | ... | | | +----------------------------------+ | | ^ | |____________________________|______________________________| | | LPASS AIF ____________________________|______________________________ | | Audio I/O | | v | | +--------------------------------------------------+ | | | Audio devices | | | | CODEC | HDMI-TX | PCM | SLIMBUS | I2S |MI2S |...| | | | | | | +--------------------------------------------------+ | |___________________________________________________________| AudioReach has constructs of sub-graph, container and modules. Each sub-graph can have N containers and each Container can have N Modules and connections between them can be linear or non-linear. An audio function can be realized with one or many connected sub-graphs. There are also control/event paths between modules that can be wired up while building graph to achieve various control mechanism between modules. These concepts of Sub-Graph, Containers and Modules are represented in ASoC topology. Here is simple I2S graph with a Write Shared Memory and a Volume control module within a single Subgraph (1) with one Container (1) and 5 modules. ____________________________________________________________ | Sub-Graph [1] | | _______________________________________________________ | | | Container [1] | | | | [WR_SH] -> [PCM DEC] -> [PCM CONV] -> [VOL]-> [I2S-EP]| | | |_______________________________________________________| | |____________________________________________________________| For now this graph is split into two subgraphs to achieve dpcm like below: ________________________________________________ _________________ | Sub-Graph [1] | | Sub-Graph [2] | | ____________________________________________ | | _____________ | | | Container [1] | | | |Container [2]| | | | [WR_SH] -> [PCM DEC] -> [PCM CONV] -> [VOL]| | | | [I2S-EP] | | | |____________________________________________| | | |_____________| | |________________________________________________| |_________________| _________________ | Sub-Graph [3] | | _____________ | | |Container [3]| | | | [DMA-EP] | | | |_____________| | |_________________| This patchset adds very minimal support for AudioReach which includes supporting sub-graphs containing CODEC DMA ports and simple PCM Decoder/Encoder and Logger Modules. Additional capabilities will be built over time to expose features offered by AudioReach. This patchset is Tested on SM8250 SoC based Qualcomm Robotics Platform RB5 and SM9250 MTP with WSA881X Smart Speaker Amplifiers, DMICs connected via VA Macro and WCD938x Codec connected via TX and RX Macro. Sample WIP ASoC graphs are available at https://git.linaro.org/people/srinivas.kandagatla/audioreach-topology.git/ Thanks, srini Changes since RFC: - moved GPR support into APR driver to avoid code duplication. - Fixed various dt-bindings-check warnings and errors. - Moved include/dt-bindings headers to dt-bindings patch. - added PGA widget support - Fixed few ordering issues while testing with ASoC Topology 2.0 - Removed duplicate defines in various headers Srinivas Kandagatla (16): soc: dt-bindings: qcom: add gpr bindings soc: qcom: apr: make code more reuseable soc: qcom: apr: Add GPR support ASoC: qcom: dt-bindings: add bindings Audio Processing manager ASoC: qcom: audioreach: add basic pkt alloc support ASoC: qcom: audioreach: add q6apm support ASoC: qcom: audioreach: add module configuration command helpers ASoC: qcom: audioreach: add topology support ASoC: qcom: audioreach: add q6apm-dai support ASoC: qcom: audioreach: add bedai support ASoC: qcom: dt-bindings: add bindings for Proxy Resource Manager ASoC: qcom: audioreach: add q6prm support ASoC: qcom: dt-bindings: add audioreach soundcard compatibles ASoC: qcom: audioreach: add volume ctrl module support ASoC: qcom: audioreach: topology add dapm pga support ASoC: qcom: sm8250: Add audioreach support .../bindings/soc/qcom/qcom,gpr.yaml | 83 ++ .../devicetree/bindings/sound/qcom,q6apm.yaml | 87 ++ .../devicetree/bindings/sound/qcom,q6prm.yaml | 43 + .../bindings/sound/qcom,sm8250.yaml | 43 + drivers/soc/qcom/Kconfig | 8 + drivers/soc/qcom/apr.c | 270 +++- include/dt-bindings/soc/qcom,gpr.h | 18 + include/dt-bindings/sound/qcom,q6apm.h | 8 + include/linux/soc/qcom/apr.h | 69 +- include/uapi/sound/snd_ar_tokens.h | 198 +++ sound/soc/qcom/Kconfig | 19 + sound/soc/qcom/Makefile | 1 + sound/soc/qcom/audioreach/Makefile | 12 + sound/soc/qcom/audioreach/audioreach.c | 1162 +++++++++++++++++ sound/soc/qcom/audioreach/audioreach.h | 676 ++++++++++ sound/soc/qcom/audioreach/q6apm-bedai.c | 323 +++++ sound/soc/qcom/audioreach/q6apm-dai.c | 494 +++++++ sound/soc/qcom/audioreach/q6apm.c | 963 ++++++++++++++ sound/soc/qcom/audioreach/q6apm.h | 174 +++ sound/soc/qcom/audioreach/q6prm.c | 414 ++++++ sound/soc/qcom/audioreach/topology.c | 1075 +++++++++++++++ sound/soc/qcom/sm8250.c | 144 +- 22 files changed, 6236 insertions(+), 48 deletions(-) create mode 100644 Documentation/devicetree/bindings/soc/qcom/qcom,gpr.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6apm.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6prm.yaml create mode 100644 include/dt-bindings/soc/qcom,gpr.h create mode 100644 include/dt-bindings/sound/qcom,q6apm.h create mode 100644 include/uapi/sound/snd_ar_tokens.h create mode 100644 sound/soc/qcom/audioreach/Makefile create mode 100644 sound/soc/qcom/audioreach/audioreach.c create mode 100644 sound/soc/qcom/audioreach/audioreach.h create mode 100644 sound/soc/qcom/audioreach/q6apm-bedai.c create mode 100644 sound/soc/qcom/audioreach/q6apm-dai.c create mode 100644 sound/soc/qcom/audioreach/q6apm.c create mode 100644 sound/soc/qcom/audioreach/q6apm.h create mode 100644 sound/soc/qcom/audioreach/q6prm.c create mode 100644 sound/soc/qcom/audioreach/topology.c -- 2.21.0

5 39

[PATCH] ALSA: pcm: Fix mmap capability check
by Takashi Iwai 30 Jul '21

30 Jul '21

The hw_support_mmap() doesn't cover all memory allocation types and might use a wrong device pointer for checking the capability. Check the all memory allocation types more completely. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_native.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index c88c4316c417..6919d2943b9d 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -246,12 +246,18 @@ static bool hw_support_mmap(struct snd_pcm_substream *substream) if (!(substream->runtime->hw.info & SNDRV_PCM_INFO_MMAP)) return false; - if (substream->ops->mmap || - (substream->dma_buffer.dev.type != SNDRV_DMA_TYPE_DEV && - substream->dma_buffer.dev.type != SNDRV_DMA_TYPE_DEV_UC)) + if (substream->ops->mmap) return true; - return dma_can_mmap(substream->dma_buffer.dev.dev); + switch (substream->dma_buffer.dev.type) { + case SNDRV_DMA_TYPE_UNKNOWN: + return false; + case SNDRV_DMA_TYPE_CONTINUOUS: + case SNDRV_DMA_TYPE_VMALLOC: + return true; + default: + return dma_can_mmap(substream->dma_buffer.dev.dev); + } } static int constrain_mask_params(struct snd_pcm_substream *substream, -- 2.26.2

2 2

[PATCH 1/3] ASoC: cs42l42: Correct definition of ADC Volume control
by Richard Fitzgerald 29 Jul '21

29 Jul '21

The ADC volume is a signed 8-bit number with range -97 to +12, with -97 being mute. Use a SOC_SINGLE_S8_TLV() to define this and fix the DECLARE_TLV_DB_SCALE() to have the correct start and mute flag. Fixes: 2c394ca79604 ("ASoC: Add support for CS42L42 codec") Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/cs42l42.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index eff013f295be..2fd20511f246 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -405,7 +405,7 @@ static const struct regmap_config cs42l42_regmap = { .use_single_write = true, }; -static DECLARE_TLV_DB_SCALE(adc_tlv, -9600, 100, false); +static DECLARE_TLV_DB_SCALE(adc_tlv, -9700, 100, true); static DECLARE_TLV_DB_SCALE(mixer_tlv, -6300, 100, true); static const char * const cs42l42_hpf_freq_text[] = { @@ -444,8 +444,7 @@ static const struct snd_kcontrol_new cs42l42_snd_controls[] = { CS42L42_ADC_INV_SHIFT, true, false), SOC_SINGLE("ADC Boost Switch", CS42L42_ADC_CTL, CS42L42_ADC_DIG_BOOST_SHIFT, true, false), - SOC_SINGLE_SX_TLV("ADC Volume", CS42L42_ADC_VOLUME, - CS42L42_ADC_VOL_SHIFT, 0xA0, 0x6C, adc_tlv), + SOC_SINGLE_S8_TLV("ADC Volume", CS42L42_ADC_VOLUME, -97, 12, adc_tlv), SOC_SINGLE("ADC WNF Switch", CS42L42_ADC_WNF_HPF_CTL, CS42L42_ADC_WNF_EN_SHIFT, true, false), SOC_SINGLE("ADC HPF Switch", CS42L42_ADC_WNF_HPF_CTL, -- 2.11.0

2 4

[PATCH] ALSA: usb-audio: Avoid unnecessary or invalid connector selection at resume
by Takashi Iwai 29 Jul '21

29 Jul '21

The recent fix for the resume on Lenovo machines seems causing a regression on others. It's because the change always triggers the connector selection no matter which widget node type is. This patch addresses the regression by setting the resume callback selectively only for the connector widget. Fixes: 44609fc01f28 ("ALSA: usb-audio: Check connector value on resume") Cc: <stable(a)vger.kernel.org> BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=213897 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer.c | 35 ++++++++++++++++++++--------------- 1 file changed, 20 insertions(+), 15 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index f4cdaf1ba44a..9b713b4a5ec4 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1816,6 +1816,15 @@ static void get_connector_control_name(struct usb_mixer_interface *mixer, strlcat(name, " - Output Jack", name_size); } +/* get connector value to "wake up" the USB audio */ +static int connector_mixer_resume(struct usb_mixer_elem_list *list) +{ + struct usb_mixer_elem_info *cval = mixer_elem_list_to_info(list); + + get_connector_value(cval, NULL, NULL); + return 0; +} + /* Build a mixer control for a UAC connector control (jack-detect) */ static void build_connector_control(struct usb_mixer_interface *mixer, const struct usbmix_name_map *imap, @@ -1833,6 +1842,10 @@ static void build_connector_control(struct usb_mixer_interface *mixer, if (!cval) return; snd_usb_mixer_elem_init_std(&cval->head, mixer, term->id); + + /* set up a specific resume callback */ + cval->head.resume = connector_mixer_resume; + /* * UAC2: The first byte from reading the UAC2_TE_CONNECTOR control returns the * number of channels connected. @@ -3642,23 +3655,15 @@ static int restore_mixer_value(struct usb_mixer_elem_list *list) return 0; } -static int default_mixer_resume(struct usb_mixer_elem_list *list) -{ - struct usb_mixer_elem_info *cval = mixer_elem_list_to_info(list); - - /* get connector value to "wake up" the USB audio */ - if (cval->val_type == USB_MIXER_BOOLEAN && cval->channels == 1) - get_connector_value(cval, NULL, NULL); - - return 0; -} - static int default_mixer_reset_resume(struct usb_mixer_elem_list *list) { - int err = default_mixer_resume(list); + int err; - if (err < 0) - return err; + if (list->resume) { + err = list->resume(list); + if (err < 0) + return err; + } return restore_mixer_value(list); } @@ -3697,7 +3702,7 @@ void snd_usb_mixer_elem_init_std(struct usb_mixer_elem_list *list, list->id = unitid; list->dump = snd_usb_mixer_dump_cval; #ifdef CONFIG_PM - list->resume = default_mixer_resume; + list->resume = NULL; list->reset_resume = default_mixer_reset_resume; #endif } -- 2.26.2

1 0

[PATCH] ASoC: samsung: Constify static snd_soc_ops
by Rikard Falkeborn 29 Jul '21

29 Jul '21

These are only assigned to the ops field in the snd_soc_dai_link struct which is a pointer to const struct snd_soc_ops. Make them const to allow the compiler to put them in read-only memory. Signed-off-by: Rikard Falkeborn <rikard.falkeborn(a)gmail.com> --- sound/soc/samsung/aries_wm8994.c | 2 +- sound/soc/samsung/arndale.c | 4 ++-- sound/soc/samsung/h1940_uda1380.c | 2 +- sound/soc/samsung/littlemill.c | 2 +- sound/soc/samsung/midas_wm1811.c | 2 +- sound/soc/samsung/neo1973_wm8753.c | 4 ++-- sound/soc/samsung/rx1950_uda1380.c | 2 +- sound/soc/samsung/smartq_wm8987.c | 2 +- sound/soc/samsung/smdk_wm8580.c | 2 +- sound/soc/samsung/smdk_wm8994.c | 2 +- sound/soc/samsung/smdk_wm8994pcm.c | 2 +- sound/soc/samsung/tm2_wm5110.c | 6 +++--- sound/soc/samsung/tobermory.c | 2 +- 13 files changed, 17 insertions(+), 17 deletions(-) diff --git a/sound/soc/samsung/aries_wm8994.c b/sound/soc/samsung/aries_wm8994.c index 0ac5956ba270..313ab650f8d9 100644 --- a/sound/soc/samsung/aries_wm8994.c +++ b/sound/soc/samsung/aries_wm8994.c @@ -310,7 +310,7 @@ static int aries_hw_free(struct snd_pcm_substream *substream) /* * Main DAI operations */ -static struct snd_soc_ops aries_ops = { +static const struct snd_soc_ops aries_ops = { .hw_params = aries_hw_params, .hw_free = aries_hw_free, }; diff --git a/sound/soc/samsung/arndale.c b/sound/soc/samsung/arndale.c index 28587375813a..606ac5e33a8e 100644 --- a/sound/soc/samsung/arndale.c +++ b/sound/soc/samsung/arndale.c @@ -48,7 +48,7 @@ static int arndale_rt5631_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops arndale_rt5631_ops = { +static const struct snd_soc_ops arndale_rt5631_ops = { .hw_params = arndale_rt5631_hw_params, }; @@ -80,7 +80,7 @@ static int arndale_wm1811_hw_params(struct snd_pcm_substream *substream, rclk + 1, SND_SOC_CLOCK_IN); } -static struct snd_soc_ops arndale_wm1811_ops = { +static const struct snd_soc_ops arndale_wm1811_ops = { .hw_params = arndale_wm1811_hw_params, }; diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index 8aa78ff640f5..c994e67d1eaf 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -112,7 +112,7 @@ static int h1940_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops h1940_ops = { +static const struct snd_soc_ops h1940_ops = { .startup = h1940_startup, .hw_params = h1940_hw_params, }; diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c index a1ff1400857e..390f2dd735ad 100644 --- a/sound/soc/samsung/littlemill.c +++ b/sound/soc/samsung/littlemill.c @@ -130,7 +130,7 @@ static int littlemill_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops littlemill_ops = { +static const struct snd_soc_ops littlemill_ops = { .hw_params = littlemill_hw_params, }; diff --git a/sound/soc/samsung/midas_wm1811.c b/sound/soc/samsung/midas_wm1811.c index 1f9a553edf19..a2019535a0b1 100644 --- a/sound/soc/samsung/midas_wm1811.c +++ b/sound/soc/samsung/midas_wm1811.c @@ -129,7 +129,7 @@ static int midas_aif1_hw_params(struct snd_pcm_substream *substream, return midas_start_fll1(rtd, pll_out); } -static struct snd_soc_ops midas_aif1_ops = { +static const struct snd_soc_ops midas_aif1_ops = { .hw_params = midas_aif1_hw_params, }; diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index 9266070e0181..c98b68567a89 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -106,7 +106,7 @@ static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream) /* * Neo1973 WM8753 HiFi DAI opserations. */ -static struct snd_soc_ops neo1973_hifi_ops = { +static const struct snd_soc_ops neo1973_hifi_ops = { .hw_params = neo1973_hifi_hw_params, .hw_free = neo1973_hifi_hw_free, }; @@ -158,7 +158,7 @@ static int neo1973_voice_hw_free(struct snd_pcm_substream *substream) return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0, 0); } -static struct snd_soc_ops neo1973_voice_ops = { +static const struct snd_soc_ops neo1973_voice_ops = { .hw_params = neo1973_voice_hw_params, .hw_free = neo1973_voice_hw_free, }; diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c index 400a7f77c711..6ea1c8cc9167 100644 --- a/sound/soc/samsung/rx1950_uda1380.c +++ b/sound/soc/samsung/rx1950_uda1380.c @@ -62,7 +62,7 @@ static struct snd_soc_jack_gpio hp_jack_gpios[] = { }, }; -static struct snd_soc_ops rx1950_ops = { +static const struct snd_soc_ops rx1950_ops = { .startup = rx1950_startup, .hw_params = rx1950_hw_params, }; diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c index c95629becbc3..cee39ad16667 100644 --- a/sound/soc/samsung/smartq_wm8987.c +++ b/sound/soc/samsung/smartq_wm8987.c @@ -70,7 +70,7 @@ static int smartq_hifi_hw_params(struct snd_pcm_substream *substream, /* * SmartQ WM8987 HiFi DAI operations. */ -static struct snd_soc_ops smartq_hifi_ops = { +static const struct snd_soc_ops smartq_hifi_ops = { .hw_params = smartq_hifi_hw_params, }; diff --git a/sound/soc/samsung/smdk_wm8580.c b/sound/soc/samsung/smdk_wm8580.c index ed753a2f202e..78703d095a6f 100644 --- a/sound/soc/samsung/smdk_wm8580.c +++ b/sound/soc/samsung/smdk_wm8580.c @@ -86,7 +86,7 @@ static int smdk_hw_params(struct snd_pcm_substream *substream, /* * SMDK WM8580 DAI operations. */ -static struct snd_soc_ops smdk_ops = { +static const struct snd_soc_ops smdk_ops = { .hw_params = smdk_hw_params, }; diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index 39a7a449f554..7661b637946d 100644 --- a/sound/soc/samsung/smdk_wm8994.c +++ b/sound/soc/samsung/smdk_wm8994.c @@ -73,7 +73,7 @@ static int smdk_hw_params(struct snd_pcm_substream *substream, /* * SMDK WM8994 DAI operations. */ -static struct snd_soc_ops smdk_ops = { +static const struct snd_soc_ops smdk_ops = { .hw_params = smdk_hw_params, }; diff --git a/sound/soc/samsung/smdk_wm8994pcm.c b/sound/soc/samsung/smdk_wm8994pcm.c index a01640576f71..029448f5bedb 100644 --- a/sound/soc/samsung/smdk_wm8994pcm.c +++ b/sound/soc/samsung/smdk_wm8994pcm.c @@ -85,7 +85,7 @@ static int smdk_wm8994_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops smdk_wm8994_pcm_ops = { +static const struct snd_soc_ops smdk_wm8994_pcm_ops = { .hw_params = smdk_wm8994_pcm_hw_params, }; diff --git a/sound/soc/samsung/tm2_wm5110.c b/sound/soc/samsung/tm2_wm5110.c index 84c2c63d5a87..a2c77e6defec 100644 --- a/sound/soc/samsung/tm2_wm5110.c +++ b/sound/soc/samsung/tm2_wm5110.c @@ -126,7 +126,7 @@ static int tm2_aif1_hw_params(struct snd_pcm_substream *substream, return tm2_start_sysclk(rtd->card); } -static struct snd_soc_ops tm2_aif1_ops = { +static const struct snd_soc_ops tm2_aif1_ops = { .hw_params = tm2_aif1_hw_params, }; @@ -200,7 +200,7 @@ static int tm2_aif2_hw_free(struct snd_pcm_substream *substream) return ret; } -static struct snd_soc_ops tm2_aif2_ops = { +static const struct snd_soc_ops tm2_aif2_ops = { .hw_params = tm2_aif2_hw_params, .hw_free = tm2_aif2_hw_free, }; @@ -254,7 +254,7 @@ static int tm2_hdmi_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops tm2_hdmi_ops = { +static const struct snd_soc_ops tm2_hdmi_ops = { .hw_params = tm2_hdmi_hw_params, }; diff --git a/sound/soc/samsung/tobermory.c b/sound/soc/samsung/tobermory.c index c962d2c2a7f7..15223d860cb7 100644 --- a/sound/soc/samsung/tobermory.c +++ b/sound/soc/samsung/tobermory.c @@ -105,7 +105,7 @@ static int tobermory_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops tobermory_ops = { +static const struct snd_soc_ops tobermory_ops = { .hw_params = tobermory_hw_params, }; -- 2.32.0

3 2

[PATCH 1/2] cx20442: tty_ldisc_ops::write_wakeup is optional
by Jiri Slaby 29 Jul '21

29 Jul '21

TTY layer does nothing if tty_ldisc_ops::write_wakeup is NULL, so there is no need to implement an empty one in cx20442. Drop it. Signed-off-by: Jiri Slaby <jslaby(a)suse.cz> 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: alsa-devel(a)alsa-project.org --- sound/soc/codecs/cx20442.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index ec8d6e74b467..65c6b92d7b5f 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -279,11 +279,6 @@ static void v253_receive(struct tty_struct *tty, const unsigned char *cp, } } -/* Line discipline .write_wakeup() */ -static void v253_wakeup(struct tty_struct *tty) -{ -} - struct tty_ldisc_ops v253_ops = { .name = "cx20442", .owner = THIS_MODULE, @@ -291,7 +286,6 @@ struct tty_ldisc_ops v253_ops = { .close = v253_close, .hangup = v253_hangup, .receive_buf = v253_receive, - .write_wakeup = v253_wakeup, }; EXPORT_SYMBOL_GPL(v253_ops); -- 2.32.0

2 3

[PATCH 0/7] ASoC: soc-core: cleanup cppcheck warning
by Kuninori Morimoto 29 Jul '21

29 Jul '21

Hi Mark Now I'm posting audio-graph-card2 patch-set, but it seems it needs longer discussion. Thus I want to post more easy patch first, and reduce my local patches. These are cppcheck warning cleanup patches for soc-core. Kuninori Morimoto (7): ASoC: soc-core: cleanup cppcheck warning at snd_soc_set_dmi_name() ASoC: soc-core: cleanup cppcheck warning at snd_soc_get_dai_name() ASoC: soc-core: cleanup cppcheck warning at snd_soc_daifmt_parse_format() ASoC: soc-core: cleanup cppcheck warning at snd_soc_unregister_component() ASoC: soc-core: cleanup cppcheck warning at snd_soc_add_controls() ASoC: soc-core: cleanup cppcheck warning at snd_soc_of_parse_audio_simple_widgets() ASoC: soc-core: cleanup cppcheck warning at snd_soc_of_parse_audio_routing() sound/soc/soc-core.c | 36 ++++++++++++++++++------------------ 1 file changed, 18 insertions(+), 18 deletions(-) -- 2.25.1

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[PATCH v2 00/27] ALSA: hda/cirrus: Split generic cirrus HDA codecs and CS8490 bridge into separate modules.
by Vitaly Rodionov 29 Jul '21

29 Jul '21

This series of patches splits generic cirrus HDA codecs and CS8490 bridge into separate modules, adds support for multiple companion codecs connected to CS8409, and also adds support for new DELL HW platform. CS8409 part is not really a HDA codec, it is a HDA bridge where companion codecs (up to 16) can be attached. With growing number of supported configurations and platforms, patch_cirrus is getting less and less transparent and maintainable. So, the logical step is to separate generic Cirrus HDA codecs support and Cirrus HDA bridge support. Lots of improvements to existing functionality, code clean-up and refactoring, remove duplicated/redundant code, improve I2C functions etc. Add support for new DELL HW platform with 2 CS42L42 codecs for front and rear jacks. Lucas Tanure (12): ALSA: hda/cirrus: Move CS8409 HDA bridge to separate module ALSA: hda/cs8409: Move arrays of configuration to a new file ALSA: hda/cs8409: Disable unsolicited response for the first boot ALSA: hda/cs8409: Prevent I2C access during suspend time ALSA: hda/cs8409: Generalize volume controls ALSA: hda/cs8409: Dont disable I2C clock between consecutive accesses ALSA: hda/cs8409: Avoid setting the same I2C address for every access ALSA: hda/cs8409: Avoid re-setting the same page as the last access ALSA: hda/cs8409: Support i2c bulk read/write functions ALSA: hda/cs8409: Separate CS8409, CS42L42 and project functions ALSA: hda/cs8409: Move codec properties to its own struct ALSA: hda/cs8409: Add support for dolphin Stefan Binding (15): ALSA: hda/cs8409: Use enums for register names and coefficients ALSA: hda/cs8409: Mask all CS42L42 interrupts on initialization ALSA: hda/cs8409: Reduce HS pops/clicks for Cyborg ALSA: hda/cs8409: Disable unnecessary Ring Sense for Cyborg/Warlock/Bullseye ALSA: hda/cs8409: Disable unsolicited responses during suspend ALSA: hda/cs8409: Mask CS42L42 wake events ALSA: hda/cs8409: Simplify CS42L42 jack detect. ALSA: hda/cs8409: Support multiple sub_codecs for Suspend/Resume/Unsol events ALSA: hda/cs8409: Add Support to disable jack type detection for CS42L42 ALSA: hda/cs8409: Enable Full Scale Volume for Line Out Codec on Dolphin ALSA: hda/cs8409: Set fixed sample rate of 48kHz for CS42L42 ALSA: hda/cs8409: Use timeout rather than retries for I2C transaction waits ALSA: hda/cs8409: Remove unnecessary delays ALSA: hda/cs8409: Follow correct CS42L42 power down sequence for suspend ALSA: hda/cs8409: Unmute/Mute codec when stream starts/stops sound/pci/hda/Kconfig | 10 + sound/pci/hda/Makefile | 2 + sound/pci/hda/patch_cirrus.c | 1074 ----------------------- sound/pci/hda/patch_cs8409-tables.c | 560 ++++++++++++ sound/pci/hda/patch_cs8409.c | 1245 +++++++++++++++++++++++++++ sound/pci/hda/patch_cs8409.h | 369 ++++++++ 6 files changed, 2186 insertions(+), 1074 deletions(-) create mode 100644 sound/pci/hda/patch_cs8409-tables.c create mode 100644 sound/pci/hda/patch_cs8409.c create mode 100644 sound/pci/hda/patch_cs8409.h -- 2.25.1

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[PATCH 00/15] ALSA: usb-audio: Introduce and convert to quirk_flags
by Takashi Iwai 29 Jul '21

29 Jul '21

As more and more device-specific workarounds came up and gathered in various places, it becomes harder to manage. Now it's time to clean up and collect workarounds more consistently and make them more easily applicable. This patch set introduces a new bitmap flags, quirk_flags, to contain various device-specific quirks. Those are separate one from the quirks in quirks-table.h; the quirks-table.h entries are for more intrusive stuff that needs the descriptor override, while the new quirk_flags is for easier ones that are tied with the vendor:product IDs. Takashi === Takashi Iwai (15): ALSA: usb-audio: Introduce quirk_flags field ALSA: usb-audio: Move media-controller API quirk into quirk_flags ALSA: usb-audio: Move txfr_quirk handling to quirk_flags ALSA: usb-audio: Move tx_length quirk handling to quirk_flags ALSA: usb-audio: Move playback_first flag into quirk_flags ALSA: usb-audio: Move clock setup quirk into quirk_flags ALSA: usb-audio: Move ITF-USB DSD quirk handling into quirk_flags ALSA: usb-audio: Move control message delay quirk into quirk_flags ALSA: usb-audio: Move interface setup delay into quirk_flags ALSA: usb-audio: Move rate validation quirk into quirk_flags ALSA: usb-audio: Move autosuspend quirk into quirk_flags ALSA: usb-audio: Move ignore_ctl_error check into quirk_flags ALSA: usb-audio: Move generic DSD raw detection into quirk_flags ALSA: usb-audio: Add quirk_flags module option ALSA: doc: Add the description of quirk_flags option for snd-usb-audio Documentation/sound/alsa-configuration.rst | 26 +- sound/usb/card.c | 42 ++- sound/usb/clock.c | 15 +- sound/usb/endpoint.c | 14 +- sound/usb/format.c | 6 +- sound/usb/implicit.c | 6 +- sound/usb/mixer.c | 6 +- sound/usb/mixer.h | 3 +- sound/usb/mixer_maps.c | 22 -- sound/usb/quirks-table.h | 70 +--- sound/usb/quirks.c | 359 +++++++++++---------- sound/usb/quirks.h | 5 +- sound/usb/stream.c | 4 +- sound/usb/usbaudio.h | 65 +++- 14 files changed, 341 insertions(+), 302 deletions(-) -- 2.26.2

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[PATCH] ASoC: dapm: Revert "use component prefix when checking widget names"
by Richard Fitzgerald 29 Jul '21

29 Jul '21

This reverts commit ae4fc532244b ("ASoC: dapm: use component prefix when checking widget names") That commit breaks all users of the snd_soc_component_*_pin() functions because it results in the prefix being added twice. It also breaks code that correctly uses the snd_soc_dapm_*_pin() functions. Use the snd_soc_component_*_pin() functions if you want the component prefix to be prepended automatically. Use the raw snd_soc_dapm_*_pin() functions if the caller has the full name that should be matched exactly. Fixes: commit ae4fc532244b ("ASoC: dapm: use component prefix when checking widget names") Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/soc-dapm.c | 13 +------------ 1 file changed, 1 insertion(+), 12 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 91bf939d5233..1369a3fea911 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2528,20 +2528,9 @@ static struct snd_soc_dapm_widget *dapm_find_widget( { struct snd_soc_dapm_widget *w; struct snd_soc_dapm_widget *fallback = NULL; - char prefixed_pin[80]; - const char *pin_name; - const char *prefix = soc_dapm_prefix(dapm); - - if (prefix) { - snprintf(prefixed_pin, sizeof(prefixed_pin), "%s %s", - prefix, pin); - pin_name = prefixed_pin; - } else { - pin_name = pin; - } for_each_card_widgets(dapm->card, w) { - if (!strcmp(w->name, pin_name)) { + if (!strcmp(w->name, pin)) { if (w->dapm == dapm) return w; else -- 2.11.0

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