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

[PATCH] [v2] ASoC: Intel: sof_rt5682: Add support for max98390 speaker amp
by Mark Hsieh 24 Aug '21

24 Aug '21

Configure adl_max98390_rt5682 to support the rt5682 headset codec and max98390 speaker BUG=b:191811888 TEST=emerge-brya chromeos-kernel-5_10 Signed-off-by: Mark Hsieh <mark_hsieh(a)wistron.corp-partner.google.com> --- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/sof_maxim_common.c | 59 +++++++++++++++++++ sound/soc/intel/boards/sof_maxim_common.h | 12 ++++ sound/soc/intel/boards/sof_rt5682.c | 20 ++++++- .../intel/common/soc-acpi-intel-adl-match.c | 13 ++++ 5 files changed, 104 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 7e29b0d911e2..0017c08c5a62 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -470,6 +470,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH select SND_SOC_RT1015 select SND_SOC_RT1015P select SND_SOC_RT5682_I2C + select SND_SOC_MAX98390 select SND_SOC_DMIC select SND_SOC_HDAC_HDMI select SND_SOC_INTEL_HDA_DSP_COMMON diff --git a/sound/soc/intel/boards/sof_maxim_common.c b/sound/soc/intel/boards/sof_maxim_common.c index e9c52f8b6428..477883d571ab 100644 --- a/sound/soc/intel/boards/sof_maxim_common.c +++ b/sound/soc/intel/boards/sof_maxim_common.c @@ -133,6 +133,65 @@ void max_98373_set_codec_conf(struct snd_soc_card *card) } EXPORT_SYMBOL_NS(max_98373_set_codec_conf, SND_SOC_INTEL_SOF_MAXIM_COMMON); +/* + * Maxim MAX98390 + */ + +static struct snd_soc_codec_conf max_98390_codec_conf[] = { + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV0_NAME), + .name_prefix = "Right", + }, + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV1_NAME), + .name_prefix = "Left", + }, +}; + +struct snd_soc_dai_link_component max_98390_components[] = { + { /* For Right */ + .name = MAX_98390_DEV0_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, + { /* For Left */ + .name = MAX_98390_DEV1_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, +}; +EXPORT_SYMBOL_NS(max_98390_components, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +static int max_98390_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai; + int j; + + for_each_rtd_codec_dais(rtd, j, codec_dai) { + if (!strcmp(codec_dai->component->name, MAX_98390_DEV0_NAME)) { + /* DEV0 tdm slot configuration */ + snd_soc_dai_set_tdm_slot(codec_dai, 0x03, 3, 2, 16); + } + if (!strcmp(codec_dai->component->name, MAX_98390_DEV1_NAME)) { + /* DEV1 tdm slot configuration */ + snd_soc_dai_set_tdm_slot(codec_dai, 0x0C, 3, 2, 16); + } + } + return 0; +} + +struct snd_soc_ops max_98390_ops = { + .hw_params = max_98390_hw_params, +}; +EXPORT_SYMBOL_NS(max_98390_ops, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +void max_98390_set_codec_conf(struct snd_soc_card *card) +{ + card->codec_conf = max_98390_codec_conf; + card->num_configs = ARRAY_SIZE(max_98390_codec_conf); +} +EXPORT_SYMBOL_NS(max_98390_set_codec_conf, SND_SOC_INTEL_SOF_MAXIM_COMMON); + /* * Maxim MAX98357A */ diff --git a/sound/soc/intel/boards/sof_maxim_common.h b/sound/soc/intel/boards/sof_maxim_common.h index 2674f1e373ef..2458758866f3 100644 --- a/sound/soc/intel/boards/sof_maxim_common.h +++ b/sound/soc/intel/boards/sof_maxim_common.h @@ -24,6 +24,18 @@ int max_98373_spk_codec_init(struct snd_soc_pcm_runtime *rtd); void max_98373_set_codec_conf(struct snd_soc_card *card); int max_98373_trigger(struct snd_pcm_substream *substream, int cmd); +/* + * Maxim MAX98390 + */ +#define MAX_98390_CODEC_DAI "max98390-aif1" +#define MAX_98390_DEV0_NAME "i2c-MX98390:00" +#define MAX_98390_DEV1_NAME "i2c-MX98390:01" + +extern struct snd_soc_dai_link_component max_98390_components[2]; +extern struct snd_soc_ops max_98390_ops; + +void max_98390_set_codec_conf(struct snd_soc_card *card); + /* * Maxim MAX98357A */ diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 39217223d50c..dc4966056b7d 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -49,6 +49,7 @@ #define SOF_RT1015P_SPEAKER_AMP_PRESENT BIT(16) #define SOF_MAX98373_SPEAKER_AMP_PRESENT BIT(17) #define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(18) +#define SOF_MAX98390_SPEAKER_AMP_PRESENT BIT(23) /* BT audio offload: reserve 3 bits for future */ #define SOF_BT_OFFLOAD_SSP_SHIFT 19 @@ -781,6 +782,13 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, } else if (sof_rt5682_quirk & SOF_RT1011_SPEAKER_AMP_PRESENT) { sof_rt1011_dai_link(&links[id]); + } else if (sof_rt5682_quirk & + SOF_MAX98390_SPEAKER_AMP_PRESENT) { + links[id].codecs = max_98390_components; + links[id].num_codecs = ARRAY_SIZE(max_98390_components); + links[id].init = max_98373_spk_codec_init; + links[id].ops = &max_98390_ops; + links[id].dpcm_capture = 1; } else { max_98357a_dai_link(&links[id]); } @@ -917,7 +925,8 @@ static int sof_audio_probe(struct platform_device *pdev) sof_rt1011_codec_conf(&sof_audio_card_rt5682); else if (sof_rt5682_quirk & SOF_RT1015P_SPEAKER_AMP_PRESENT) sof_rt1015p_codec_conf(&sof_audio_card_rt5682); - + else if (sof_rt5682_quirk & SOF_MAX98390_SPEAKER_AMP_PRESENT) + max_98390_set_codec_conf(&sof_audio_card_rt5682); if (sof_rt5682_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) sof_audio_card_rt5682.num_links++; @@ -1043,6 +1052,15 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(2) | SOF_RT5682_NUM_HDMIDEV(4)), }, + { + .name = "adl_max98390_rt5682", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98390_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(2) | + SOF_RT5682_NUM_HDMIDEV(4)), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index a0f6a69c7038..2db152998e4a 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -280,6 +280,11 @@ static const struct snd_soc_acpi_codecs adl_max98357a_amp = { .codecs = {"MX98357A"} }; +static const struct snd_soc_acpi_codecs adl_max98390_amp = { + .num_codecs = 1, + .codecs = {"MX98390"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { { .id = "10EC5682", @@ -297,6 +302,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .sof_fw_filename = "sof-adl.ri", .sof_tplg_filename = "sof-adl-max98357a-rt5682.tplg", }, + { + .id = "10EC5682", + .drv_name = "adl_max98390_rt5682", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98390_amp, + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-max98390-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_adl_machines); -- 2.17.1

4 3

alsactl: flush stdout
by GitHub pull_request - opened 24 Aug '21

24 Aug '21

alsa-project/alsa-utils pull request #109 was opened from Hello71: Signed-off-by: Alex Xu (Hello71) <alex_y_xu(a)yahoo.ca> Request URL : https://github.com/alsa-project/alsa-utils/pull/109 Patch URL : https://github.com/alsa-project/alsa-utils/pull/109.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Pinebook Pro's audio does not work
by GitHub issues - edited 24 Aug '21

24 Aug '21

alsa-project/alsa-ucm-conf issue #110 was edited from starnight: I have a [Pinebook Pro](https://www.pine64.org/pinebook-pro/) whose audio (including the internal speaker, headphone and internal MIC) does not work by default with general Linux. Here is the [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/7037916/… I notice Manjaro ARM has [asound.state](https://gitlab.manjaro.org/manjaro-arm/packages/community/pin… in [pinebookpro-post-install](https://gitlab.manjaro.org/manjaro-arm/packages/c… package. If I install the [asound.state](https://gitlab.manjaro.org/manjaro-arm/packages/community/pin… into `/var/lib/alsa/`, then `alsactl restore`, all of the audio works! However, I guess that **asound.state** is produced as a result, not the original configuration. If this relates to alsa-ucm-conf project, how should I configure the ucm for Pinebook Pro's audio? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/110 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Pinebook Pro's audio does not work
by GitHub issues - opened 24 Aug '21

24 Aug '21

alsa-project/alsa-ucm-conf issue #110 was opened from starnight: I have a Pinebook Pro whose audio (including the internal speaker, headphone and internal MIC) does not work by default with general Linux. Here is the [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/7037916/… I notice Manjaro ARM has [asound.state](https://gitlab.manjaro.org/manjaro-arm/packages/community/pin… in [pinebookpro-post-install](https://gitlab.manjaro.org/manjaro-arm/packages/c… package. If I install the [asound.state](https://gitlab.manjaro.org/manjaro-arm/packages/community/pin… into `/var/lib/alsa/`, then `alsactl restore`, all of the audio works! However, I guess that **asound.state** is produced as a result, not the original configuration. If this relates to alsa-ucm-conf project, how should I configure the ucm for Pinebook Pro's audio? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/110 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Timestamp
by GitHub pull_request - opened 24 Aug '21

24 Aug '21

alsa-project/alsa-utils pull request #108 was opened from folkertvanheusden: Added -T / --timestamp (with parameter "realtime", "monotonic" or "raw") which adds a timestamp before each received message when using --dump. Request URL : https://github.com/alsa-project/alsa-utils/pull/108 Patch URL : https://github.com/alsa-project/alsa-utils/pull/108.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

arecord is failing with -V stereo
by sujith kumar reddy 24 Aug '21

24 Aug '21

Hi All, arecord is failing vumeter option -V stereo only. localhost ~ # arecord -Dhw:1,2 -r48000 -c2 -fS32_LE /tmp/test_record.wav -M -d 1 -V stereo Recording WAVE '/tmp/test_record.wav' : Signed 32 bit Little Endian, Rate 48000 Hz, Stereo *** buffer overflow detected ***: terminated Aborted by signal Aborted... Please provide pointers to debug this option. Thanks Sujith

3 16

[PATCH alsa-utils 0/5] aplay: Fix bugs in peak calculations
by Takashi Iwai 24 Aug '21

24 Aug '21

The code in aplay for sample peak calculations and VU-meter representation seems to have a lot of bugs. Let's fix them. Takashi === Takashi Iwai (5): aplay: Fix conversion of unsigned samples in peak calculation aplay: Handle 16bit sample negative overflow in peak calculations aplay: Don't pass most negative integer to abs() in peak calculations aplay: Handle upper bound in peak calculations aplay: Fix out-of-bound access in stereo VU meter drawing aplay/aplay.c | 28 +++++++++++++++++++--------- 1 file changed, 19 insertions(+), 9 deletions(-) -- 2.26.2

1 5

rawmidi: hide the frame structure using a decoder function
by GitHub pull_request - opened 24 Aug '21

24 Aug '21

alsa-project/alsa-lib pull request #173 was opened from perexg: The frame structure should be hidden. Remove it and add snd_rawmidi_decode_frame0() which returns the timestamp and the midi chunk pointer to the application. BugLink: https://github.com/alsa-project/alsa-lib/issues/172 Request URL : https://github.com/alsa-project/alsa-lib/pull/173 Patch URL : https://github.com/alsa-project/alsa-lib/pull/173.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH for-5.15] ALSA: usb-audio: Move set-interface-first workaround into common quirk
by Takashi Iwai 24 Aug '21

24 Aug '21

The recent quirk for WALKMAN (commit 7af5a14371c1: "ALSA: usb-audio: Fix regression on Sony WALKMAN NW-A45 DAC") may be required for other devices and is worth to be put into the common quirk flags. This patch adds a new quirk flag bit QUIRK_FLAG_SET_IFACE_FIRST and a quirk table entry for the device. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Documentation/sound/alsa-configuration.rst | 1 + sound/usb/endpoint.c | 6 ++---- sound/usb/quirks.c | 2 ++ sound/usb/usbaudio.h | 3 +++ 4 files changed, 8 insertions(+), 4 deletions(-) diff --git a/Documentation/sound/alsa-configuration.rst b/Documentation/sound/alsa-configuration.rst index e3c33b8d6d78..65f61695f561 100644 --- a/Documentation/sound/alsa-configuration.rst +++ b/Documentation/sound/alsa-configuration.rst @@ -2278,6 +2278,7 @@ quirk_flags * bit 13: Disable runtime PM autosuspend * bit 14: Ignore errors for mixer access * bit 15: Support generic DSD raw U32_BE format + * bit 16: Set up the interface at first like UAC1 This module supports multiple devices, autoprobe and hotplugging. diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index e6d58d7674a0..79a8c569c62b 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -1293,10 +1293,8 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip, * to be set up before parameter setups */ iface_first = ep->cur_audiofmt->protocol == UAC_VERSION_1; - /* Workaround for Sony WALKMAN NW-A45 DAC; - * it requires the interface setup at first like UAC1 - */ - if (chip->usb_id == USB_ID(0x054c, 0x0b8c)) + /* Workaround for devices that require the interface setup at first like UAC1 */ + if (chip->quirk_flags & QUIRK_FLAG_SET_IFACE_FIRST) iface_first = true; if (iface_first) { err = endpoint_set_interface(chip, ep, true); diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 670abc6318f2..4479a590194f 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1790,6 +1790,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x04e8, 0xa051, /* Samsung USBC Headset (AKG) */ QUIRK_FLAG_SKIP_CLOCK_SELECTOR | QUIRK_FLAG_CTL_MSG_DELAY_5M), + DEVICE_FLG(0x054c, 0x0b8c, /* Sony WALKMAN NW-A45 DAC */ + QUIRK_FLAG_SET_IFACE_FIRST), DEVICE_FLG(0x0556, 0x0014, /* Phoenix Audio TMX320VC */ QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x05a3, 0x9420, /* ELP HD USB Camera */ diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 4e93668a2a48..94261d19cceb 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -161,6 +161,8 @@ extern bool snd_usb_skip_validation; * Ignore errors for mixer access * QUIRK_FLAG_DSD_RAW: * Support generic DSD raw U32_BE format + * QUIRK_FLAG_SET_IFACE_FIRST: + * Set up the interface at first like UAC1 */ #define QUIRK_FLAG_GET_SAMPLE_RATE (1U << 0) @@ -179,5 +181,6 @@ extern bool snd_usb_skip_validation; #define QUIRK_FLAG_DISABLE_AUTOSUSPEND (1U << 13) #define QUIRK_FLAG_IGNORE_CTL_ERROR (1U << 14) #define QUIRK_FLAG_DSD_RAW (1U << 15) +#define QUIRK_FLAG_SET_IFACE_FIRST (1U << 16) #endif /* __USBAUDIO_H */ -- 2.31.1

1 0

[PATCH] ALSA: usb-audio: Fix regression on Sony WALKMAN NW-A45 DAC
by Takashi Iwai 24 Aug '21

24 Aug '21

We've got a regression report for USB-audio with Sony WALKMAN NW-A45 DAC device where no sound is audible on recent kernel. The bisection resulted in the code change wrt endpoint management, and the further debug session revealed that it was caused by the order of the USB audio interface. In the earlier code, we always set up the USB interface at first before other setups, but it was changed to be done at the last for UAC2/3, which is more standard way, while keeping the old way for UAC1. OTOH, this device seems requiring the setup of the interface at first just like UAC1. This patch works around the regression by applying the interface setup specifically for the WALKMAN at the beginning of the endpoint setup procedure. This change is written straightforwardly to be easily backported in old kernels. A further cleanup to move the workaround into a generic quirk section will follow in a later patch. Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management") Cc: <stable(a)vger.kernel.org> BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=214105 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/endpoint.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index d33685874a6b..e6d58d7674a0 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -1293,6 +1293,11 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip, * to be set up before parameter setups */ iface_first = ep->cur_audiofmt->protocol == UAC_VERSION_1; + /* Workaround for Sony WALKMAN NW-A45 DAC; + * it requires the interface setup at first like UAC1 + */ + if (chip->usb_id == USB_ID(0x054c, 0x0b8c)) + iface_first = true; if (iface_first) { err = endpoint_set_interface(chip, ep, true); if (err < 0) -- 2.31.1

1 0