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

[PATCH] ALSA: hda: Balance runtime/system PM if direct-complete is disabled
by Kai-Heng Feng 18 Jan '21

18 Jan '21

HDA controller can't be runtime-suspended after commit 215a22ed31a1 ("ALSA: hda: Refactor codjc PM to use direct-complete optimization"), which enables direct-complete for HDA codec. The HDA codec driver didn't expect direct-complete will be disabled after it returns a positive value from prepare() callback. However, there are some places that PM core can disable direct-complete. For instance, system hibernation or when codec has subordinates like LEDs. So if a device is prepared for direct-complete but PM core still calls codec's suspend or freeze callback, resume the device to keep PM operations balanced. Reported-by: Kenneth R. Crudup <kenny(a)panix.com> Fixes: 215a22ed31a1 ("ALSA: hda: Refactor codec PM to use direct-complete optimization") Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/hda_codec.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 687216e74526..0afbced979df 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2997,6 +2997,9 @@ static void hda_codec_pm_complete(struct device *dev) static int hda_codec_pm_suspend(struct device *dev) { + if (pm_runtime_status_suspended(dev)) + pm_runtime_resume(dev); + dev->power.power_state = PMSG_SUSPEND; return hda_codec_suspend(dev); } @@ -3009,6 +3012,9 @@ static int hda_codec_pm_resume(struct device *dev) static int hda_codec_pm_freeze(struct device *dev) { + if (pm_runtime_status_suspended(dev)) + pm_runtime_resume(dev); + dev->power.power_state = PMSG_FREEZE; return hda_codec_suspend(dev); } -- 2.29.2

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[PATCH] ALSA: usb-audio: Add support for Pioneer DJM-750
by Olivia Mackintosh 18 Jan '21

18 Jan '21

This adds the Pioneer DJ DJM-750 to the quirks table and ensures skip_pioneer_sync_ep() is (also) called: this device uses the vendor ID of 0x08e4 (I'm not sure why they use multiple vendor IDs but many just like to be awkward it seems). Playback on all 8 channels works. I'll likely keep this working in the future and submit futher patches and improvements as necessary. Signed-off-by: Olivia Mackintosh <livvy(a)base.nu> --- sound/usb/implicit.c | 3 +- sound/usb/quirks-table.h | 60 ++++++++++++++++++++++++++++++++++++++++ 2 files changed, 62 insertions(+), 1 deletion(-) diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c index 521cc846d9d9..e7216d0b860d 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -302,7 +302,8 @@ static int audioformat_implicit_fb_quirk(struct snd_usb_audio *chip, /* Pioneer devices with vendor spec class */ if (attr == USB_ENDPOINT_SYNC_ASYNC && alts->desc.bInterfaceClass == USB_CLASS_VENDOR_SPEC && - USB_ID_VENDOR(chip->usb_id) == 0x2b73 /* Pioneer */) { + (USB_ID_VENDOR(chip->usb_id) == 0x2b73 || /* Pioneer */ + USB_ID_VENDOR(chip->usb_id) == 0x08e4 /* Pioneer */)) { if (skip_pioneer_sync_ep(chip, fmt, alts)) return 1; } diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index c8a4bdf18207..98b31c3160ad 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3757,6 +3757,66 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } } }, +{ + /* + * Pioneer DJ DJM-750 + * 8 channels playback & 8 channels capture @ 44.1/48/96kHz S24LE + */ + USB_DEVICE_VENDOR_SPEC(0x08e4, 0x017f), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x05, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_44100| + SNDRV_PCM_RATE_48000| + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 3, + .rate_table = (unsigned int[]) { 44100, 48000, 96000 } + } + }, + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x86, + .ep_idx = 1, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC| + USB_ENDPOINT_USAGE_IMPLICIT_FB, + .rates = SNDRV_PCM_RATE_44100| + SNDRV_PCM_RATE_48000| + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 3, + .rate_table = (unsigned int[]) { 44100, 48000, 96000 } + } + }, + { + .ifnum = -1 + } + } + } +}, #undef USB_DEVICE_VENDOR_SPEC #undef USB_AUDIO_DEVICE -- 2.30.0

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[PATCH] dt-bindings: sound: intel, keembay-i2s: Fix dt binding errors
by Michael Sit Wei Hong 18 Jan '21

18 Jan '21

Fix devicetree binding errors caused by newly added parameters Signed-off-by: Michael Sit Wei Hong <michael.wei.hong.sit(a)intel.com> --- .../devicetree/bindings/sound/intel,keembay-i2s.yaml | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml b/Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml index e0658f122cbb..dba25c33f0b0 100644 --- a/Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml +++ b/Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml @@ -47,13 +47,11 @@ properties: dmas: items: - - description: DMA controller phandle and DMA channel - for TX and RX + - description: DMA TX channel + - description: DMA RX channel dma-names: items: - - description: "tx" for the transmit channel - "rx" for the receive channel - const: tx - const: rx -- 2.17.1

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[RESEND PATCH] ASoC: audio-graph-card: Drop remote-endpoint as required property
by Sameer Pujar 18 Jan '21

18 Jan '21

The remote-endpoint may not be available if it is part of some pluggable module. One such example would be an audio card, the Codec endpoint will not be available until it is plugged in. Hence drop 'remote-endpoint' as a required property. Cc: Rob Herring <robh(a)kernel.org> Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> --- Documentation/devicetree/bindings/sound/audio-graph-port.yaml | 3 --- 1 file changed, 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml index 2005014..766e910 100644 --- a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml +++ b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml @@ -71,9 +71,6 @@ properties: description: CPU to Codec rate channels. $ref: /schemas/types.yaml#/definitions/uint32 - required: - - remote-endpoint - ports: description: multi OF-Graph subnode type: object -- 2.7.4

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[PATCH] ASoC: dt-bindings: mt8192-mt6359: Fix indentation
by Jonathan Neuschäfer 18 Jan '21

18 Jan '21

The items of the 'maintainers' list are indented with three spaces. Use the usual two spaces instead, for consistency and to silence yamllint. Signed-off-by: Jonathan Neuschäfer <j.neuschaefer(a)gmx.net> --- .../bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml b/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml index bf8c8ba25009d..54650823b29a4 100644 --- a/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml +++ b/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml @@ -7,8 +7,8 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: Mediatek MT8192 with MT6359, RT1015 and RT5682 ASoC sound card driver maintainers: - - Jiaxin Yu <jiaxin.yu(a)mediatek.com> - - Shane Chien <shane.chien(a)mediatek.com> + - Jiaxin Yu <jiaxin.yu(a)mediatek.com> + - Shane Chien <shane.chien(a)mediatek.com> description: This binding describes the MT8192 sound card. -- 2.29.2

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[PATCH] ASoC: wm_adsp: Fix control name parsing for multi-fw
by James Schulman 18 Jan '21

18 Jan '21

When switching between firmware types, the wrong control can be selected when requesting control in kernel API. Use the currently selected DSP firwmare type to select the proper mixer control. Signed-off-by: James Schulman <james.schulman(a)cirrus.com> --- sound/soc/codecs/wm_adsp.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 1fc7bc1970ea..262c2db4f81c 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2033,11 +2033,14 @@ static struct wm_coeff_ctl *wm_adsp_get_ctl(struct wm_adsp *dsp, unsigned int alg) { struct wm_coeff_ctl *pos, *rslt = NULL; + const char *fw_txt = wm_adsp_fw_text[dsp->fw]; list_for_each_entry(pos, &dsp->ctl_list, list) { if (!pos->subname) continue; if (strncmp(pos->subname, name, pos->subname_len) == 0 && + strncmp(pos->fw_name, fw_txt, + SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == 0 && pos->alg_region.alg == alg && pos->alg_region.type == type) { rslt = pos; -- 2.17.1

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[PATCH v2] ASoC: SOF: Intel: initial support to AlderLake-P
by Kai Vehmanen 18 Jan '21

18 Jan '21

Add PCI id for the AlderLake-P. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> --- sound/soc/sof/sof-pci-dev.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) V2 changes: - fix compile error if CONFIG_SND_SOC_SOF_TIGERLAKE=n and CONFIG_SND_SOC_SOF_ALDERLAKE is enabled diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index 63b989e3ec40..ebd58d49d570 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -213,7 +213,7 @@ static const struct sof_dev_desc icl_desc = { }; #endif -#if IS_ENABLED(CONFIG_SND_SOC_SOF_TIGERLAKE) +#if IS_ENABLED(CONFIG_SND_SOC_SOF_TIGERLAKE) || IS_ENABLED(CONFIG_SND_SOC_SOF_ALDERLAKE) static const struct sof_dev_desc tgl_desc = { .machines = snd_soc_acpi_intel_tgl_machines, .alt_machines = snd_soc_acpi_intel_tgl_sdw_machines, @@ -230,7 +230,9 @@ static const struct sof_dev_desc tgl_desc = { .nocodec_tplg_filename = "sof-tgl-nocodec.tplg", .ops = &sof_tgl_ops, }; +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_TIGERLAKE) static const struct sof_dev_desc tglh_desc = { .machines = snd_soc_acpi_intel_tgl_machines, .alt_machines = snd_soc_acpi_intel_tgl_sdw_machines, @@ -512,6 +514,8 @@ static const struct pci_device_id sof_pci_ids[] = { #if IS_ENABLED(CONFIG_SND_SOC_SOF_ALDERLAKE) { PCI_DEVICE(0x8086, 0x7ad0), .driver_data = (unsigned long)&adls_desc}, + { PCI_DEVICE(0x8086, 0x51c8), + .driver_data = (unsigned long)&tgl_desc}, #endif { 0, } }; -- 2.29.2

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[PATCH] ASoC: SOF: Intel: initial support to AlderLake-P
by Kai Vehmanen 18 Jan '21

18 Jan '21

Add PCI id for the AlderLake-P. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> --- sound/soc/sof/sof-pci-dev.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index 63b989e3ec40..9c3cb719e268 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -512,6 +512,8 @@ static const struct pci_device_id sof_pci_ids[] = { #if IS_ENABLED(CONFIG_SND_SOC_SOF_ALDERLAKE) { PCI_DEVICE(0x8086, 0x7ad0), .driver_data = (unsigned long)&adls_desc}, + { PCI_DEVICE(0x8086, 0x51c8), + .driver_data = (unsigned long)&tgl_desc}, #endif { 0, } }; -- 2.29.2

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chtnau8824: Add support for using the SOF driver
by GitHub pull_request - opened 18 Jan '21

18 Jan '21

alsa-project/alsa-ucm-conf pull request #74 was opened from perexg: The old (and currently the default) SST driver uses TDM 4 slots 24 bit as wire format to the codec. Where as the new SOF driver uses standard I2S 2 channel 24 bit. Normally this should not impact the UCM settings, but on the NAU8824 the "DAC Right Channel Source" mixer setting must be set to 1 when using TDM 4 slots and to 0 when using I2S 2 channel mode. Getting this wrong (in either case) results in the right channel not outputting any sound. This commit introduces a RightOutputChannel variable which gets set to 0/1 depending on the driver and then uses that for the "DAC Right Channel Source" mixer setting so that we do the right thing depending on the driver. This has been tested on the following devices: Medion E2215T: Stereo speakers, analog mic Medion E2228T: Stereo speakers, stereo digital mics Cube iWork 8 Air: Mono speaker, analog mic Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/74 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/74.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH 0/3] ALSA: usb-audio: Yet more fixes for Pioneer devices
by Takashi Iwai 18 Jan '21

18 Jan '21

Hi, the Pioneer UAC1 devices still didn't work with the recent changes, and after debugging sessions, a few issues were discovered. Below are the patches to cover those errors. Takashi === Takashi Iwai (3): ALSA: usb-audio: Fix UAC1 rate setup for secondary endpoints ALSA: usb-audio: Set sample rate for all sharing EPs on UAC1 ALSA: usb-audio: Avoid implicit feedback on Pioneer devices sound/usb/clock.c | 21 ++++++--------------- sound/usb/endpoint.c | 9 +++++++++ sound/usb/implicit.c | 17 +++++++++-------- sound/usb/quirks.c | 28 ---------------------------- 4 files changed, 24 insertions(+), 51 deletions(-) -- 2.26.2

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