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

[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

[PATCH] ALSA: azt3328: Remove the unused function snd_azf3328_codec_outl()
by Jiapeng Chong 22 Dec '22

22 Dec '22

The function snd_azf3328_codec_outl is defined in the azt3328.c file, but not called elsewhere, so remove this unused function. sound/pci/azt3328.c:367:1: warning: unused function 'snd_azf3328_codec_outl'. Link: https://bugzilla.openanolis.cn/show_bug.cgi?id=3432 Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Jiapeng Chong <jiapeng.chong(a)linux.alibaba.com> --- sound/pci/azt3328.c | 9 --------- 1 file changed, 9 deletions(-) diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 7f329dfc5404..0c6754bf9455 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -363,15 +363,6 @@ snd_azf3328_codec_inw(const struct snd_azf3328_codec_data *codec, unsigned reg) return inw(codec->io_base + reg); } -static inline void -snd_azf3328_codec_outl(const struct snd_azf3328_codec_data *codec, - unsigned reg, - u32 value -) -{ - outl(value, codec->io_base + reg); -} - static inline void snd_azf3328_codec_outl_multi(const struct snd_azf3328_codec_data *codec, unsigned reg, const void *buffer, int count -- 2.20.1.7.g153144c

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Re: Re: [PATCH] ASoC: Intel: avs: Add support for RT5663 codec
by CoolStar 21 Dec '22

21 Dec '22

This is a question that Cezary would need to answer, as setting the format was simply brought over from the da7219 AVS machine driver.

1 0

Re: Re: [PATCH] ASoC: Intel: avs: Add support for RT5663 codec
by CoolStar 21 Dec '22

21 Dec '22

1 0

ALC4080 broken on MSI MPG X570S Carbon Max Wifi
by GitHub issues - opened 21 Dec '22

21 Dec '22

alsa-project/alsa-ucm-conf issue #256 was opened from lordhoto: With the current ALC4080 UCM2 configuration, sound is completely broken on my MSI MPG X570S Carbon Max Wifi Mainboard with an ALC4080. Regardless of whether I use the "HiFi 2.0 channels", "HiFi 5.1 channels" or "HiFi 7.1 channels" configuration on the front headphone output, the audio is sometimes "stuck" and not playing at all. This causes all sorts of issues: when you play a YouTube video it might not even start, after a while sound might play for a bit and then suddenly stop again. Even in Gnome's audio settings dialog when testing left/right the sound sometimes just does not play at all. I'm using pulseaudio 16.1. When I insert my old ALSA UCM2 configuration, the sound works flawlessly. I suppose there is something wrong with the current configuration, but I can't figure out what. Are there any logs or similar I might look into? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/256 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH V2 1/4] ASoC: amd: ps: implement api to retrieve acp device config
by Vijendar Mukunda 21 Dec '22

21 Dec '22

Implement API to retrieve acp device config and calculate platform device count and dev mask for platform device node creation. Currently for DMIC configuration, mask and dev count are calculated. Same api will be used to extend support for different ACP device configurations. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- v1->v2: - used ACP_DMIC_DEV macro instead of hard coded value - changed is_dmic_dev variable initialization - changed debug level sound/soc/amd/ps/acp63.h | 7 +++++++ sound/soc/amd/ps/pci-ps.c | 37 +++++++++++++++++++++++++++++++++++-- 2 files changed, 42 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/ps/acp63.h b/sound/soc/amd/ps/acp63.h index 5e7f9c1c1b0e..b015e845e5fa 100644 --- a/sound/soc/amd/ps/acp63.h +++ b/sound/soc/amd/ps/acp63.h @@ -54,6 +54,11 @@ /* time in ms for runtime suspend delay */ #define ACP_SUSPEND_DELAY_MS 2000 +#define ACP63_DMIC_ADDR 2 +#define ACP63_PDM_MODE_DEVS 3 +#define ACP63_PDM_DEV_MASK 1 +#define ACP_DMIC_DEV 2 + enum acp_config { ACP_CONFIG_0 = 0, ACP_CONFIG_1, @@ -102,4 +107,6 @@ struct acp63_dev_data { struct resource *res; bool acp63_audio_mode; struct platform_device *pdev[ACP63_DEVS]; + u16 pdev_mask; + u16 pdev_count; }; diff --git a/sound/soc/amd/ps/pci-ps.c b/sound/soc/amd/ps/pci-ps.c index 489f01a20699..9066a625e501 100644 --- a/sound/soc/amd/ps/pci-ps.c +++ b/sound/soc/amd/ps/pci-ps.c @@ -132,6 +132,38 @@ static irqreturn_t acp63_irq_handler(int irq, void *dev_id) return IRQ_NONE; } +void get_acp63_device_config(u32 config, struct pci_dev *pci, struct acp63_dev_data *acp_data) +{ + struct acpi_device *dmic_dev; + const union acpi_object *obj; + bool is_dmic_dev = false; + + dmic_dev = acpi_find_child_device(ACPI_COMPANION(&pci->dev), ACP63_DMIC_ADDR, 0); + if (dmic_dev) { + if (!acpi_dev_get_property(dmic_dev, "acp-audio-device-type", + ACPI_TYPE_INTEGER, &obj) && + obj->integer.value == ACP_DMIC_DEV) + is_dmic_dev = true; + } + + switch (config) { + case ACP_CONFIG_0: + case ACP_CONFIG_1: + case ACP_CONFIG_2: + case ACP_CONFIG_3: + case ACP_CONFIG_9: + case ACP_CONFIG_15: + dev_dbg(&pci->dev, "Audio Mode %d\n", config); + break; + default: + if (is_dmic_dev) { + acp_data->pdev_mask = ACP63_PDM_DEV_MASK; + acp_data->pdev_count = ACP63_PDM_MODE_DEVS; + } + break; + } +} + static int snd_acp63_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { @@ -183,6 +215,7 @@ static int snd_acp63_probe(struct pci_dev *pci, if (ret) goto release_regions; val = acp63_readl(adata->acp63_base + ACP_PIN_CONFIG); + get_acp63_device_config(val, pci, adata); switch (val) { case ACP_CONFIG_0: case ACP_CONFIG_1: @@ -195,14 +228,14 @@ static int snd_acp63_probe(struct pci_dev *pci, default: /* Checking DMIC hardware*/ - adev = acpi_find_child_device(ACPI_COMPANION(&pci->dev), 0x02, 0); + adev = acpi_find_child_device(ACPI_COMPANION(&pci->dev), ACP_DMIC_ADDR, 0); if (!adev) break; if (!acpi_dev_get_property(adev, "acp-audio-device-type", ACPI_TYPE_INTEGER, &obj) && - obj->integer.value == 2) { + obj->integer.value == ACP_DMIC_DEV) { adata->res = devm_kzalloc(&pci->dev, sizeof(struct resource), GFP_KERNEL); if (!adata->res) { ret = -ENOMEM; -- 2.34.1

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