December 2022 - Alsa-devel - mailman.alsa-project.org

alsaloop: confusing line of code and easier example of the use of snd_pcm_poll_descriptors API
by GitHub issues - opened 23 Dec '22

23 Dec '22

alsa-project/alsa-utils issue #185 was opened from anthonio9: Hi! Recently I've been studying the `alsaloop` software, I'm trying to understand how should the `snd_pcm_poll_descriptors` api be used reliably. Reading through the `pcmjob.c` file I've noticed that in the `pcmjob_pollfds_handle` function in line 1964 there's a condition that does not exactly make sense to me. What's the purpose of variable `loopcount`? If the code is more or less ```cpp int loopcount = 0; do { // instructions loopcount++; } while ((ccount > 0 || pcount > 0) && loopcount > 10); // line 1964 ``` isn't it equal to running the loop just once without any loops? After all, the `loopcount` will never reach 10 and the conditional statement will always turn false finishing the loop after one iteration. Correct me if I'm wrong. I'd also appreciate a simpler example than `alsaloop` focusing just on the audio parts, on transferring audio from input to output reliably with the use of `snd_pcm_poll_descriptors`. Thanks Issue URL : https://github.com/alsa-project/alsa-utils/issues/185 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Re: [PATCH 0/2] drm: Add component_match_add_of and convert users of drm_of_component_match_add
by Sean Anderson 22 Dec '22

22 Dec '22

Hi Robin, On 12/16/22 12:41, Robin Murphy wrote: > On 2022-12-16 17:08, Sean Anderson wrote: >> On 11/3/22 14:22, Sean Anderson wrote: >>> This series adds a new function component_match_add_of to simplify the >>> common case of calling component_match_add_release with >>> component_release_of and component_compare_of. There is already >>> drm_of_component_match_add, which allows for a custom compare function. >>> However, all existing users just use component_compare_of (or an >>> equivalent). >>> >>> I can split the second commit up if that is easier to review. >>> >>> >>> Sean Anderson (2): >>> component: Add helper for device nodes >>> drm: Convert users of drm_of_component_match_add to >>> component_match_add_of >>> >>> .../gpu/drm/arm/display/komeda/komeda_drv.c | 6 ++-- >>> drivers/gpu/drm/arm/hdlcd_drv.c | 9 +----- >>> drivers/gpu/drm/arm/malidp_drv.c | 11 +------ >>> drivers/gpu/drm/armada/armada_drv.c | 10 ++++--- >>> drivers/gpu/drm/drm_of.c | 29 +++---------------- >>> drivers/gpu/drm/etnaviv/etnaviv_drv.c | 4 +-- >>> .../gpu/drm/hisilicon/kirin/kirin_drm_drv.c | 3 +- >>> drivers/gpu/drm/ingenic/ingenic-drm-drv.c | 3 +- >>> drivers/gpu/drm/mediatek/mtk_drm_drv.c | 4 +-- >>> drivers/gpu/drm/msm/msm_drv.c | 14 ++++----- >>> drivers/gpu/drm/sti/sti_drv.c | 3 +- >>> drivers/gpu/drm/sun4i/sun4i_drv.c | 3 +- >>> drivers/gpu/drm/tilcdc/tilcdc_external.c | 10 ++----- >>> drivers/iommu/mtk_iommu.c | 3 +- >>> drivers/iommu/mtk_iommu_v1.c | 3 +- >>> include/drm/drm_of.h | 12 -------- >>> include/linux/component.h | 9 ++++++ >>> sound/soc/codecs/wcd938x.c | 6 ++-- >>> 18 files changed, 46 insertions(+), 96 deletions(-) >>> >> >> ping? >> >> Should I send a v2 broken up like Mark suggested? > > FWIW you'll need to rebase the IOMMU changes on 6.2-rc1 anyway - mtk_iommu stops using component_match_add_release() at all. > > Thanks, > Robin. I am preparing v2 of this series, but I don't see these changes on drm-next, which seems to have been updated to at least 6.2-rc1. I tried searching for these changes on lore [1], but I don't see them there either. Do you have a commit hash/lore link for these changes? --Sean [1] https://lore.kernel.org/all/?q=dfn%3Adrivers%2Fiommu%2Fmtk_iommu.c+b%3Acomp…

1 0

ALC4080 z790 apex no sound
by GitHub issues - opened 22 Dec '22

22 Dec '22

alsa-project/alsa-ucm-conf issue #257 was opened from pipes80-collab: good evening everyone, I have a problem with the realtek alc4080 audio peripheral integrated in the Z790 apex, the peripheral connected to my yamaha 5.1 amplifier signals the pcm but no sound comes out, everything works normally under windows, I followed some solutions on the alsa-ucm configuration -conf for the device but to no avail. then invo output lsusb and aplay -l Bus 006 Device 001: ID 1d6b:0003 Linux Foundation 3.0 root hub Bus 005 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 004 Device 001: ID 1d6b:0003 Linux Foundation 3.0 root hub Bus 003 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 002 Device 002: ID 174c:3074 ASMedia Technology Inc. ASM1074 SuperSpeed hub Bus 002 Device 001: ID 1d6b:0003 Linux Foundation 3.0 root hub Bus 001 Device 005: ID 046d:c08b Logitech, Inc. G502 SE HERO Gaming Mouse Bus 001 Device 003: ID 054c:0ce6 Sony Corp. DualSense wireless controller (PS5) Bus 001 Device 006: ID 195d:2030 Itron Technology iONE Func KB-460 Gaming Keyboard Bus 001 Device 004: ID 05e3:0608 Genesys Logic, Inc. Hub Bus 001 Device 002: ID 174c:2074 ASMedia Technology Inc. ASM1074 High-Speed hub Bus 001 Device 009: ID 05e3:0608 Genesys Logic, Inc. Hub Bus 001 Device 008: ID 0b05:1a52 ASUSTek Computer, Inc. USB Audio Bus 001 Device 007: ID 0b05:18f3 ASUSTek Computer, Inc. AURA LED Controller Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub [filippo@pipes ~]$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Audio [USB Audio], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Audio [USB Audio], device 1: USB Audio [USB Audio #1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Audio [USB Audio], device 2: USB Audio [USB Audio #2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Audio [USB Audio], device 3: USB Audio [USB Audio #3] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Controller [Wireless Controller], device 0: USB Audio [USB Audio] Subdevices: 1/1 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/257 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: Intel: avs: Add support for RT5663 codec
by Alicja Michalska 22 Dec '22

22 Dec '22

This patch adds support for RT5663 codec on KBL platform. Such hardware configuration can be found in Google Pixelbook (Google/Eve). Reported-and-tested-by: CoolStar <coolstarorganization(a)gmail.com> Signed-off-by: Alicja Michalska <ahplka19(a)gmail.com> diff --git a/sound/soc/intel/avs/board_selection.c b/sound/soc/intel/avs/board_selection.c index b2823c2107f7..b167a641d1d5 100644 --- a/sound/soc/intel/avs/board_selection.c +++ b/sound/soc/intel/avs/board_selection.c @@ -159,6 +159,14 @@ static struct snd_soc_acpi_mach avs_kbl_i2s_machines[] = { }, .tplg_filename = "da7219-tplg.bin", }, + { + .id = "10EC5663", + .drv_name = "avs_rt5663", + .mach_params = { + .i2s_link_mask = AVS_SSP(1), + }, + .tplg_filename = "rt5663-tplg.bin", + }, {}, }; diff --git a/sound/soc/intel/avs/boards/Kconfig b/sound/soc/intel/avs/boards/Kconfig index e4c230efe8d7..04c090c3f9b5 100644 --- a/sound/soc/intel/avs/boards/Kconfig +++ b/sound/soc/intel/avs/boards/Kconfig @@ -125,6 +125,16 @@ config SND_SOC_INTEL_AVS_MACH_RT5682 Say Y or m if you have such a device. This is a recommended option. If unsure select "N". +config SND_SOC_INTEL_AVS_MACH_RT5663 + tristate "rt5663 in I2S mode" + depends on I2C + depends on MFD_INTEL_LPSS || COMPILE_TEST + select SND_SOC_RT5663_I2C + help + This adds support for ASoC machine driver with RT5663 I2S 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_AVS_MACH_SSM4567 tristate "ssm4567 I2S board" depends on I2C diff --git a/sound/soc/intel/avs/boards/Makefile b/sound/soc/intel/avs/boards/Makefile index b81343420370..3db863fc26a7 100644 --- a/sound/soc/intel/avs/boards/Makefile +++ b/sound/soc/intel/avs/boards/Makefile @@ -13,6 +13,7 @@ snd-soc-avs-rt274-objs := rt274.o snd-soc-avs-rt286-objs := rt286.o snd-soc-avs-rt298-objs := rt298.o snd-soc-avs-rt5682-objs := rt5682.o +snd-soc-avs-rt5663-objs := rt5663.o snd-soc-avs-ssm4567-objs := ssm4567.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_DA7219) += snd-soc-avs-da7219.o @@ -28,4 +29,5 @@ obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT274) += snd-soc-avs-rt274.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT286) += snd-soc-avs-rt286.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT298) += snd-soc-avs-rt298.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT5682) += snd-soc-avs-rt5682.o +obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT5663) += snd-soc-avs-rt5663.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_SSM4567) += snd-soc-avs-ssm4567.o diff --git a/sound/soc/intel/avs/boards/rt5663.c b/sound/soc/intel/avs/boards/rt5663.c new file mode 100644 index 000000000000..7d8f45267d27 --- /dev/null +++ b/sound/soc/intel/avs/boards/rt5663.c @@ -0,0 +1,249 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) CoolStar. All rights reserved. +// Based off da7219 module +// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// +// Author: CoolStar <coolstarorganization.com> +// Author: Cezary Rojewski <cezary.rojewski(a)intel.com> +// + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> +#include <sound/soc-dapm.h> +#include <uapi/linux/input-event-codes.h> +#include "../../../codecs/rt5663.h" + +#define RT5663_DAI_NAME "rt5663-aif" + +static const struct snd_kcontrol_new card_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), +}; + +static const struct snd_soc_dapm_widget card_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), +}; + +static const struct snd_soc_dapm_route card_base_routes[] = { + /* HP jack connectors - unknown if we have jack detection */ + {"Headphone Jack", NULL, "HPOL"}, + {"Headphone Jack", NULL, "HPOR"}, + + {"IN1P", NULL, "Headset Mic"}, + {"IN1N", NULL, "Headset Mic"}, +}; + +static int avs_rt5663_codec_init(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component; + struct snd_soc_card *card = runtime->card; + struct snd_soc_jack *jack; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(runtime, 0); + int ret; + + jack = snd_soc_card_get_drvdata(card); + + rt5663_sel_asrc_clk_src(component, + RT5663_DA_STEREO_FILTER | RT5663_AD_STEREO_FILTER, + RT5663_CLK_SEL_I2S1_ASRC); + + snd_soc_dai_set_sysclk(codec_dai, + RT5663_SCLK_S_MCLK, 24576000, SND_SOC_CLOCK_IN); + + /* + * Headset buttons map to the google Reference headset. + * These can be configured by userspace. + */ + ret = snd_soc_card_jack_new(card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1 | SND_JACK_BTN_2 | + SND_JACK_BTN_3 | SND_JACK_LINEOUT, jack); + if (ret) { + dev_err(card->dev, "Headset Jack creation failed: %d\n", ret); + return ret; + } + + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + + return 0; +} + +static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port, + struct snd_soc_dai_link **dai_link) +{ + struct snd_soc_dai_link_component *platform; + struct snd_soc_dai_link *dl; + + dl = devm_kzalloc(dev, sizeof(*dl), GFP_KERNEL); + platform = devm_kzalloc(dev, sizeof(*platform), GFP_KERNEL); + if (!dl || !platform) + return -ENOMEM; + + platform->name = platform_name; + + dl->name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", ssp_port); + dl->cpus = devm_kzalloc(dev, sizeof(*dl->cpus), GFP_KERNEL); + dl->codecs = devm_kzalloc(dev, sizeof(*dl->codecs), GFP_KERNEL); + if (!dl->name || !dl->cpus || !dl->codecs) + return -ENOMEM; + + dl->cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", ssp_port); + dl->codecs->name = devm_kasprintf(dev, GFP_KERNEL, "i2c-10EC5663:00"); + dl->codecs->dai_name = RT5663_DAI_NAME; + if (!dl->cpus->dai_name || !dl->codecs->name || !dl->codecs->dai_name) + return -ENOMEM; + + dl->num_cpus = 1; + dl->num_codecs = 1; + dl->platforms = platform; + dl->num_platforms = 1; + dl->id = 0; + dl->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS; + dl->init = avs_rt5663_codec_init; + dl->nonatomic = 1; + dl->no_pcm = 1; + dl->dpcm_capture = 1; + dl->dpcm_playback = 1; + + *dai_link = dl; + + return 0; +} + +static int avs_create_dapm_routes(struct device *dev, int ssp_port, + struct snd_soc_dapm_route **routes, int *num_routes) +{ + struct snd_soc_dapm_route *dr; + const int num_base = ARRAY_SIZE(card_base_routes); + const int num_dr = num_base + 2; + int idx; + + dr = devm_kcalloc(dev, num_dr, sizeof(*dr), GFP_KERNEL); + if (!dr) + return -ENOMEM; + + memcpy(dr, card_base_routes, num_base * sizeof(*dr)); + + idx = num_base; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "AIF Playback"); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Tx", ssp_port); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + idx++; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Rx", ssp_port); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "AIF Capture"); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + *routes = dr; + *num_routes = num_dr; + + return 0; +} + +static int avs_card_set_jack(struct snd_soc_card *card, struct snd_soc_jack *jack) +{ + struct snd_soc_component *component; + + for_each_card_components(card, component) + snd_soc_component_set_jack(component, jack, NULL); + return 0; +} + +static int avs_card_remove(struct snd_soc_card *card) +{ + return avs_card_set_jack(card, NULL); +} + +static int avs_card_suspend_pre(struct snd_soc_card *card) +{ + return avs_card_set_jack(card, NULL); +} + +static int avs_card_resume_post(struct snd_soc_card *card) +{ + struct snd_soc_jack *jack = snd_soc_card_get_drvdata(card); + + return avs_card_set_jack(card, jack); +} + +static int avs_rt5663_probe(struct platform_device *pdev) +{ + struct snd_soc_dapm_route *routes; + struct snd_soc_dai_link *dai_link; + struct snd_soc_acpi_mach *mach; + struct snd_soc_card *card; + struct snd_soc_jack *jack; + struct device *dev = &pdev->dev; + const char *pname; + int num_routes, ssp_port, ret; + + mach = dev_get_platdata(dev); + pname = mach->mach_params.platform; + ssp_port = __ffs(mach->mach_params.i2s_link_mask); + + ret = avs_create_dai_link(dev, pname, ssp_port, &dai_link); + if (ret) { + dev_err(dev, "Failed to create dai link: %d", ret); + return ret; + } + + ret = avs_create_dapm_routes(dev, ssp_port, &routes, &num_routes); + if (ret) { + dev_err(dev, "Failed to create dapm routes: %d", ret); + return ret; + } + + jack = devm_kzalloc(dev, sizeof(*jack), GFP_KERNEL); + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!jack || !card) + return -ENOMEM; + + card->name = "avs_rt5663"; + card->dev = dev; + card->owner = THIS_MODULE; + card->remove = avs_card_remove; + card->suspend_pre = avs_card_suspend_pre; + card->resume_post = avs_card_resume_post; + card->dai_link = dai_link; + card->num_links = 1; + card->controls = card_controls; + card->num_controls = ARRAY_SIZE(card_controls); + card->dapm_widgets = card_widgets; + card->num_dapm_widgets = ARRAY_SIZE(card_widgets); + card->dapm_routes = routes; + card->num_dapm_routes = num_routes; + card->fully_routed = true; + snd_soc_card_set_drvdata(card, jack); + + ret = snd_soc_fixup_dai_links_platform_name(card, pname); + if (ret) + return ret; + + return devm_snd_soc_register_card(dev, card); +} + +static struct platform_driver avs_rt5663_driver = { + .probe = avs_rt5663_probe, + .driver = { + .name = "avs_rt5663", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(avs_rt5663_driver); + +MODULE_AUTHOR("CoolStar <coolstarorganization(a)gmail.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:avs_rt5663");

3 4

[PATCH V7 0/5] ASoC: codecs: Add Awinic AW883XX audio amplifier driver
by wangweidong.a＠awinic.com 22 Dec '22

22 Dec '22

From: Weidong Wang <wangweidong.a(a)awinic.com> The Awinic AW883XX is an I2S/TDM input, high efficiency digital Smart K audio amplifier with an integrated 10.25V smart boost convert Add a DT schema for describing Awinic AW883xx audio amplifiers. They are controlled using I2C v6 -> v7: Change name-prefix.yaml to dai-common.yaml in awinic,aw883xx.yaml file Delete redundant header files Use EINVAL and so on instead of custom error return values Remove unnecessary comment No longer assign NULL to pointer Change the way the if statement is written Use devm_kcalloc instead of devm_kzalloc Use crc8 and crc32 that come with linux Weidong Wang (5): ASoC: codecs: Add i2c and codec registration for aw883xx and their associated operation functions ASoC: codecs: Aw883xx function for ACF file parse and check ASoC: codecs: Aw883xx common function for ALSA Audio Driver ASoC: codecs: Aw883xx chip register file, data type file and Kconfig Makefile ASoC: dt-bindings: Add schema for "awinic,aw883xx" .../bindings/sound/awinic,aw883xx.yaml | 49 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/aw883xx/aw883xx.c | 706 +++++++ sound/soc/codecs/aw883xx/aw883xx.h | 61 + sound/soc/codecs/aw883xx/aw883xx_bin_parse.c | 1138 ++++++++++ sound/soc/codecs/aw883xx/aw883xx_bin_parse.h | 123 ++ sound/soc/codecs/aw883xx/aw883xx_data_type.h | 143 ++ sound/soc/codecs/aw883xx/aw883xx_device.c | 1840 +++++++++++++++++ sound/soc/codecs/aw883xx/aw883xx_device.h | 201 ++ .../soc/codecs/aw883xx/aw883xx_pid_2049_reg.h | 490 +++++ 11 files changed, 4767 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/awinic,aw883xx.yaml create mode 100644 sound/soc/codecs/aw883xx/aw883xx.c create mode 100644 sound/soc/codecs/aw883xx/aw883xx.h create mode 100644 sound/soc/codecs/aw883xx/aw883xx_bin_parse.c create mode 100644 sound/soc/codecs/aw883xx/aw883xx_bin_parse.h create mode 100644 sound/soc/codecs/aw883xx/aw883xx_data_type.h create mode 100644 sound/soc/codecs/aw883xx/aw883xx_device.c create mode 100644 sound/soc/codecs/aw883xx/aw883xx_device.h create mode 100644 sound/soc/codecs/aw883xx/aw883xx_pid_2049_reg.h base-commit: 9d2f6060fe4c3b49d0cdc1dce1c99296f33379c8 -- 2.38.1

2 8

[PATCH 0/3] Add support for XCVR on i.MX93 platform
by Chancel Liu 22 Dec '22

22 Dec '22

This patchset supports XCVR on i.MX93 platform. Chancel Liu (3): ASoC: dt-bindings: fsl,xcvr: Add compatible string for i.MX93 platform ASoC: fsl_xcvr: Add support for i.MX93 platform ASoC: fsl_xcvr: Add constraints of period size while using eDMA .../devicetree/bindings/sound/fsl,xcvr.yaml | 1 + sound/soc/fsl/fsl_xcvr.c | 167 +++++++++++++----- sound/soc/fsl/fsl_xcvr.h | 7 + 3 files changed, 127 insertions(+), 48 deletions(-) -- 2.25.1

3 6

[PATCH V7 5/5] ASoC: dt-bindings: Add schema for "awinic,aw883xx"
by wangweidong.a＠awinic.com 22 Dec '22

22 Dec '22

From: Weidong Wang <wangweidong.a(a)awinic.com> Add a DT schema for describing Awinic AW883xx audio amplifiers. They are controlled using I2C. Signed-off-by: Weidong Wang <wangweidong.a(a)awinic.com> --- .../bindings/sound/awinic,aw883xx.yaml | 49 +++++++++++++++++++ 1 file changed, 49 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/awinic,aw883xx.yaml diff --git a/Documentation/devicetree/bindings/sound/awinic,aw883xx.yaml b/Documentation/devicetree/bindings/sound/awinic,aw883xx.yaml new file mode 100644 index 000000000000..af4e0e27f8f7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/awinic,aw883xx.yaml @@ -0,0 +1,49 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/awinic,aw883xx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Awinic AW883xx Smart Audio Amplifier + +maintainers: + - Stephan Gerhold <stephan(a)gerhold.net> + +description: + The Awinic AW883XX is an I2S/TDM input, high efficiency + digital Smart K audio amplifier with an integrated 10.25V + smart boost convert. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: awinic,aw883xx_smartpa + + reg: + maxItems: 1 + + reset-gpios: + maxItems: 1 + + sound-name-prefix: true + +required: + - compatible + - reg + - reset-gpios + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + audio-codec@34 { + compatible = "awinic,aw883xx_smartpa"; + reg = <0x34>; + reset-gpios = <&gpio 10 GPIO_ACTIVE_LOW>; + }; + }; -- 2.38.1

3 2

nhlt ssp aux data and blob ver 1.5
by GitHub pull_request - opened 22 Dec '22

22 Dec '22

alsa-project/alsa-utils pull request #184 was opened from juimonen: Add support in nhlt plugin for: - ssp blob ver 1.5 - ssp auxiliary data Request URL : https://github.com/alsa-project/alsa-utils/pull/184 Patch URL : https://github.com/alsa-project/alsa-utils/pull/184.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[GIT PULL] ASoC updates for v6.2-3
by Mark Brown 22 Dec '22

22 Dec '22

The following changes since commit 9472382db38452df15f9f2f74b1dff34848e56b2: Add SOF panic dump support for AMD platform. (2022-12-05 17:28:07 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v6.2-3 for you to fetch changes up to a12a383e59ce486abd719b6bda33c353a3b385e7: ASoC: lochnagar: Fix unused lochnagar_of_match warning (2022-12-15 13:53:00 +0000) ---------------------------------------------------------------- ASoC: Updates for v6.2 Some more small fixes and board quirks that came in since my last update, the main one being the fixes from Kai for issues around the attempts to get kexec working well on SOF based systems. ---------------------------------------------------------------- Bard Liao (1): ASoC: Intel: sof_realtek_common: set ret = 0 as initial value Brent Lu (1): ASoC: Intel: sof_rt5682: add jsl_rt5682 board config Cezary Rojewski (22): ALSA: hda: Allow for compress stream to hdac_ext_stream assignment ALSA: hda: Prepare for compress stream support ALSA: hda: Interrupt servicing and BDL setup for compress streams ASoC: Intel: avs: Introduce avs_log_buffer_status_locked() ASoC: Intel: avs: Drop fifo_lock ASoC: Intel: avs: Introduce debug-context aware helpers ASoC: Intel: avs: Make enable_logs() dependent on DEBUG_FS ASoC: Intel: avs: Drop usage of debug members in non-debug code ASoC: Intel: avs: Add data probing requests ASoC: Intel: avs: Probe compress operations ASoC: Intel: avs: Data probing soc-component ASoC: Intel: avs: Add probe machine board ASoC: Intel: avs: Probing and firmware tracing over debugfs ASoC: Intel: avs: Gather remaining logs on strace_release() ASoC: Intel: avs: Allow for dumping FW_REGS area ASoC: Intel: avs: Allow for dumping debug window snapshot ASoC: Intel: Skylake: Update pipe_config_idx before filling BE params ASoC: Intel: Skylake: Remove skl_tplg_is_multi_fmt() ASoC: Intel: Skylake: Drop pipe_config_idx ASoC: Intel: Skylake: Introduce single place for pipe-config selection ASoC: Intel: Skylake: Fix driver hang during shutdown ASoC: Intel: Skylake: Use SG allocation for SKL-based firmware load Chancel Liu (1): ASoC: soc-pcm.c: Clear DAIs parameters after stream_active is updated Gongjun Song (2): ASoC: Intel: sof_sdw: use common helpers for all Realtek amps ASoC: Intel: soc-acpi: update codec addr on 0C11/0C4F product Hans de Goede (2): ASoC: Intel: bytcr_rt5640: Add quirk for the Advantech MICA-071 tablet ASoC: rt5670: Remove unbalanced pm_runtime_put() Jiapeng Chong (1): ASoC: codecs: wcd-clsh: Remove the unused function Kai Vehmanen (2): ASoC: SOF: Intel: pci-tgl: unblock S5 entry if DMA stop has failed" ASoC: SOF: Revert: "core: unregister clients and machine drivers in .shutdown" Krzysztof Kozlowski (1): ASoC: lochnagar: Fix unused lochnagar_of_match warning Marek Szyprowski (1): ASoC: wm8994: Fix potential deadlock Marek Vasut (1): ASoC: dt-bindings: fsl-sai: Reinstate i.MX93 SAI compatible string Mark Brown (4): ASoC: Intel: avs: Data probing and fw logging ASoC: Intel: Skylake: Topology and shutdown fixes ASoC: Intel: boards: updates for SOF boards ASoC: SOF: remove unregister calls from shutdown Moises Cardona (1): ASoC: Intel: Add HP Stream 8 to bytcr_rt5640.c Wang Jingjin (2): ASoC: rockchip: pdm: Add missing clk_disable_unprepare() in rockchip_pdm_runtime_resume() ASoC: rockchip: spdif: Add missing clk_disable_unprepare() in rk_spdif_runtime_resume() Wang Yufen (3): ASoC: mediatek: mt8173-rt5650-rt5514: fix refcount leak in mt8173_rt5650_rt5514_dev_probe() ASoC: audio-graph-card: fix refcount leak of cpu_ep in __graph_for_each_link() ASoC: mediatek: mt8183: fix refcount leak in mt8183_mt6358_ts3a227_max98357_dev_probe() YC Hung (2): ASoC: mediatek: mt8195: add sof be ops to check audio active ASoC: SOF: mediatek: initialize panic_info to zero Yang Yingliang (1): ASoC: sof_es8336: fix possible use-after-free in sof_es8336_remove() ye xingchen (1): ASoC: imx-audmux: use sysfs_emit() to instead of scnprintf() .../devicetree/bindings/sound/fsl,sai.yaml | 1 + include/sound/hdaudio_ext.h | 2 + sound/hda/ext/hdac_ext_stream.c | 41 ++ sound/hda/hdac_controller.c | 4 +- sound/hda/hdac_stream.c | 51 ++- sound/soc/codecs/lochnagar-sc.c | 2 +- sound/soc/codecs/rt5670.c | 2 - sound/soc/codecs/wcd-clsh-v2.c | 6 - sound/soc/codecs/wm8994.c | 5 + sound/soc/fsl/imx-audmux.c | 3 +- sound/soc/generic/audio-graph-card.c | 4 +- sound/soc/intel/Kconfig | 1 + sound/soc/intel/avs/Makefile | 4 + sound/soc/intel/avs/apl.c | 24 +- sound/soc/intel/avs/avs.h | 80 +++- sound/soc/intel/avs/board_selection.c | 33 ++ sound/soc/intel/avs/boards/Kconfig | 8 + sound/soc/intel/avs/boards/Makefile | 2 + sound/soc/intel/avs/boards/probe.c | 64 +++ sound/soc/intel/avs/core.c | 2 + sound/soc/intel/avs/debugfs.c | 436 +++++++++++++++++++++ sound/soc/intel/avs/ipc.c | 2 +- sound/soc/intel/avs/messages.c | 104 ++++- sound/soc/intel/avs/messages.h | 53 +++ sound/soc/intel/avs/pcm.c | 6 +- sound/soc/intel/avs/probes.c | 313 +++++++++++++++ sound/soc/intel/avs/registers.h | 3 +- sound/soc/intel/avs/skl.c | 17 +- sound/soc/intel/avs/utils.c | 22 -- sound/soc/intel/boards/Makefile | 3 +- sound/soc/intel/boards/bytcr_rt5640.c | 25 ++ sound/soc/intel/boards/sof_es8336.c | 2 +- sound/soc/intel/boards/sof_realtek_common.c | 3 +- sound/soc/intel/boards/sof_rt5682.c | 6 + sound/soc/intel/boards/sof_sdw.c | 10 +- sound/soc/intel/boards/sof_sdw_common.h | 20 +- sound/soc/intel/boards/sof_sdw_rt1316.c | 239 ----------- sound/soc/intel/boards/sof_sdw_rt1318.c | 120 ------ .../boards/{sof_sdw_rt1308.c => sof_sdw_rt_amp.c} | 139 +++++-- sound/soc/intel/common/soc-acpi-intel-jsl-match.c | 5 + sound/soc/intel/common/soc-acpi-intel-rpl-match.c | 2 +- sound/soc/intel/skylake/skl-sst-cldma.c | 27 +- sound/soc/intel/skylake/skl-topology.c | 73 ++-- sound/soc/intel/skylake/skl-topology.h | 1 - sound/soc/intel/skylake/skl.c | 5 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 7 +- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 14 +- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 30 ++ sound/soc/rockchip/rockchip_pdm.c | 1 + sound/soc/rockchip/rockchip_spdif.c | 1 + sound/soc/soc-pcm.c | 20 +- sound/soc/sof/core.c | 9 - sound/soc/sof/intel/hda-dsp.c | 72 ++++ sound/soc/sof/intel/hda.h | 1 + sound/soc/sof/intel/tgl.c | 2 +- sound/soc/sof/mediatek/mtk-adsp-common.c | 2 +- 56 files changed, 1519 insertions(+), 615 deletions(-) create mode 100644 sound/soc/intel/avs/boards/probe.c create mode 100644 sound/soc/intel/avs/debugfs.c create mode 100644 sound/soc/intel/avs/probes.c delete mode 100644 sound/soc/intel/boards/sof_sdw_rt1316.c delete mode 100644 sound/soc/intel/boards/sof_sdw_rt1318.c rename sound/soc/intel/boards/{sof_sdw_rt1308.c => sof_sdw_rt_amp.c} (59%)

2 1

[PATCH] ALSA: usb-audio: Add new quirk FIXED_RATE for JBL Quantum810 Wireless
by Jaroslav Kysela 22 Dec '22

22 Dec '22

It seems that the firmware is broken and does not accept the UAC_EP_CS_ATTR_SAMPLE_RATE URB. There is only one rate (48000Hz) available in the descriptors for the output endpoint. Create a new quirk QUIRK_FLAG_FIXED_RATE to skip the rate setup when only one rate is available (fixed). BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=216798 Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/usb/card.h | 1 + sound/usb/endpoint.c | 16 ++++++++++------ sound/usb/endpoint.h | 3 ++- sound/usb/implicit.c | 6 +++++- sound/usb/implicit.h | 2 +- sound/usb/pcm.c | 36 +++++++++++++++++++++++++++++++++--- sound/usb/pcm.h | 2 ++ sound/usb/quirks.c | 2 ++ sound/usb/usbaudio.h | 4 ++++ 9 files changed, 60 insertions(+), 12 deletions(-) diff --git a/sound/usb/card.h b/sound/usb/card.h index 40061550105a..6ec95b2edf86 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -131,6 +131,7 @@ struct snd_usb_endpoint { bool lowlatency_playback; /* low-latency playback mode */ bool need_setup; /* (re-)need for hw_params? */ bool need_prepare; /* (re-)need for prepare? */ + bool fixed_rate; /* skip rate setup */ /* for hw constraints */ const struct audioformat *cur_audiofmt; diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 4aaf0784940b..419302e2057e 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -769,7 +769,8 @@ struct snd_usb_endpoint * snd_usb_endpoint_open(struct snd_usb_audio *chip, const struct audioformat *fp, const struct snd_pcm_hw_params *params, - bool is_sync_ep) + bool is_sync_ep, + bool fixed_rate) { struct snd_usb_endpoint *ep; int ep_num = is_sync_ep ? fp->sync_ep : fp->endpoint; @@ -825,6 +826,7 @@ snd_usb_endpoint_open(struct snd_usb_audio *chip, ep->implicit_fb_sync = fp->implicit_fb; ep->need_setup = true; ep->need_prepare = true; + ep->fixed_rate = fixed_rate; usb_audio_dbg(chip, " channels=%d, rate=%d, format=%s, period_bytes=%d, periods=%d, implicit_fb=%d\n", ep->cur_channels, ep->cur_rate, @@ -1413,11 +1415,13 @@ static int init_sample_rate(struct snd_usb_audio *chip, if (clock && !clock->need_setup) return 0; - err = snd_usb_init_sample_rate(chip, ep->cur_audiofmt, rate); - if (err < 0) { - if (clock) - clock->rate = 0; /* reset rate */ - return err; + if (!ep->fixed_rate) { + err = snd_usb_init_sample_rate(chip, ep->cur_audiofmt, rate); + if (err < 0) { + if (clock) + clock->rate = 0; /* reset rate */ + return err; + } } if (clock) diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index e67ea28faa54..924f4351588c 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -14,7 +14,8 @@ struct snd_usb_endpoint * snd_usb_endpoint_open(struct snd_usb_audio *chip, const struct audioformat *fp, const struct snd_pcm_hw_params *params, - bool is_sync_ep); + bool is_sync_ep, + bool fixed_rate); void snd_usb_endpoint_close(struct snd_usb_audio *chip, struct snd_usb_endpoint *ep); int snd_usb_endpoint_set_params(struct snd_usb_audio *chip, diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c index f3e8484b3d9c..41ac7185b42b 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -15,6 +15,7 @@ #include "usbaudio.h" #include "card.h" #include "helper.h" +#include "pcm.h" #include "implicit.h" enum { @@ -455,7 +456,8 @@ const struct audioformat * snd_usb_find_implicit_fb_sync_format(struct snd_usb_audio *chip, const struct audioformat *target, const struct snd_pcm_hw_params *params, - int stream) + int stream, + bool *fixed_rate) { struct snd_usb_substream *subs; const struct audioformat *fp, *sync_fmt = NULL; @@ -483,6 +485,8 @@ snd_usb_find_implicit_fb_sync_format(struct snd_usb_audio *chip, } } + if (fixed_rate) + *fixed_rate = snd_usb_pcm_has_fixed_rate(subs); return sync_fmt; } diff --git a/sound/usb/implicit.h b/sound/usb/implicit.h index ccb415a0ea86..7f1577b6c4d3 100644 --- a/sound/usb/implicit.h +++ b/sound/usb/implicit.h @@ -9,6 +9,6 @@ const struct audioformat * snd_usb_find_implicit_fb_sync_format(struct snd_usb_audio *chip, const struct audioformat *target, const struct snd_pcm_hw_params *params, - int stream); + int stream, bool *fixed_rate); #endif /* __USBAUDIO_IMPLICIT_H */ diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 9557bd4d1bbc..99a66d0ef5b2 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -157,6 +157,31 @@ find_substream_format(struct snd_usb_substream *subs, true, subs); } +bool snd_usb_pcm_has_fixed_rate(struct snd_usb_substream *subs) +{ + const struct audioformat *fp; + struct snd_usb_audio *chip = subs->stream->chip; + int rate = -1; + + if (!(chip->quirk_flags & QUIRK_FLAG_FIXED_RATE)) + return false; + list_for_each_entry(fp, &subs->fmt_list, list) { + if (fp->rates & SNDRV_PCM_RATE_CONTINUOUS) + return false; + if (fp->nr_rates < 1) + continue; + if (fp->nr_rates > 1) + return false; + if (rate < 0) { + rate = fp->rate_table[0]; + continue; + } + if (rate != fp->rate_table[0]) + return false; + } + return true; +} + static int init_pitch_v1(struct snd_usb_audio *chip, int ep) { struct usb_device *dev = chip->dev; @@ -450,12 +475,14 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, struct snd_usb_audio *chip = subs->stream->chip; const struct audioformat *fmt; const struct audioformat *sync_fmt; + bool fixed_rate, sync_fixed_rate; int ret; ret = snd_media_start_pipeline(subs); if (ret) return ret; + fixed_rate = snd_usb_pcm_has_fixed_rate(subs); fmt = find_substream_format(subs, hw_params); if (!fmt) { usb_audio_dbg(chip, @@ -469,7 +496,8 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, if (fmt->implicit_fb) { sync_fmt = snd_usb_find_implicit_fb_sync_format(chip, fmt, hw_params, - !substream->stream); + !substream->stream, + &sync_fixed_rate); if (!sync_fmt) { usb_audio_dbg(chip, "cannot find sync format: ep=0x%x, iface=%d:%d, format=%s, rate=%d, channels=%d\n", @@ -482,6 +510,7 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, } } else { sync_fmt = fmt; + sync_fixed_rate = fixed_rate; } ret = snd_usb_lock_shutdown(chip); @@ -499,7 +528,7 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, close_endpoints(chip, subs); } - subs->data_endpoint = snd_usb_endpoint_open(chip, fmt, hw_params, false); + subs->data_endpoint = snd_usb_endpoint_open(chip, fmt, hw_params, false, fixed_rate); if (!subs->data_endpoint) { ret = -EINVAL; goto unlock; @@ -508,7 +537,8 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, if (fmt->sync_ep) { subs->sync_endpoint = snd_usb_endpoint_open(chip, sync_fmt, hw_params, - fmt == sync_fmt); + fmt == sync_fmt, + sync_fixed_rate); if (!subs->sync_endpoint) { ret = -EINVAL; goto unlock; diff --git a/sound/usb/pcm.h b/sound/usb/pcm.h index 493a4e34d78d..388fe2ba346d 100644 --- a/sound/usb/pcm.h +++ b/sound/usb/pcm.h @@ -6,6 +6,8 @@ void snd_usb_set_pcm_ops(struct snd_pcm *pcm, int stream); int snd_usb_pcm_suspend(struct snd_usb_stream *as); int snd_usb_pcm_resume(struct snd_usb_stream *as); +bool snd_usb_pcm_has_fixed_rate(struct snd_usb_substream *as); + int snd_usb_init_pitch(struct snd_usb_audio *chip, const struct audioformat *fmt); void snd_usb_preallocate_buffer(struct snd_usb_substream *subs); diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 58b37bfc885c..3d13fdf7590c 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2152,6 +2152,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_GENERIC_IMPLICIT_FB), DEVICE_FLG(0x0525, 0xa4ad, /* Hamedal C20 usb camero */ QUIRK_FLAG_IFACE_SKIP_CLOSE), + DEVICE_FLG(0x0ecb, 0x2069, /* JBL Quantum810 Wireless */ + QUIRK_FLAG_FIXED_RATE), /* Vendor matches */ VENDOR_FLG(0x045e, /* MS Lifecam */ diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 2aba508a4831..f5a8dca66457 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -175,6 +175,9 @@ extern bool snd_usb_skip_validation; * QUIRK_FLAG_FORCE_IFACE_RESET * Force an interface reset whenever stopping & restarting a stream * (e.g. after xrun) + * QUIRK_FLAG_FIXED_RATE + * Do not set PCM rate (frequency) when only one rate is available + * for the given endpoint. */ #define QUIRK_FLAG_GET_SAMPLE_RATE (1U << 0) @@ -198,5 +201,6 @@ extern bool snd_usb_skip_validation; #define QUIRK_FLAG_SKIP_IMPLICIT_FB (1U << 18) #define QUIRK_FLAG_IFACE_SKIP_CLOSE (1U << 19) #define QUIRK_FLAG_FORCE_IFACE_RESET (1U << 20) +#define QUIRK_FLAG_FIXED_RATE (1U << 21) #endif /* __USBAUDIO_H */ -- 2.38.1

2 1