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

pcm_old.h does not work with LTO
by GitHub issues - opened 08 Nov '22

08 Nov '22

alsa-project/alsa-lib issue #281 was opened from kkofler: Applications using `ALSA_PCM_OLD_HW_PARAMS_API` and/or `ALSA_PCM_OLD_SW_PARAMS_API` get miscompiled if they are compiled with `-flto` (GCC link-time optimization (LTO)). See https://bugzilla.redhat.com/show_bug.cgi?id=1910437 and in particular https://bugzilla.redhat.com/show_bug.cgi?id=1910437#c23 . The issue is that the `pcm_old.h` header uses constructs like: ```c asm(".symver snd_pcm_hw_params_set_rate_near,snd_pcm_hw_params_set_rate_near(a)ALSA_0.9"); ``` which do not seem to work with LTO enabled. Maybe this needs to use the `symver` attribute instead? In any case, what happens is that the caller in the application attempts to call the new function `snd_pcm_hw_params_set_rate_near(a)ALSA_0.9.0rc4` with the ABI of the old one `snd_pcm_hw_params_set_rate_near(a)ALSA_0.9`, so an integer is passed where ALSA expects a pointer, leading to a segfault (or worse). The workaround is to compile the application without LTO enabled (which makes it work fine), but I believe that this is an upstream ALSA issue and should be fixed in the ALSA header files. Issue URL : https://github.com/alsa-project/alsa-lib/issues/281 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH] ASoC: Correct the header guard of wm8904.h
by Wei Li 08 Nov '22

08 Nov '22

Rename the header guard of wm8904.h from __MFD_WM8994_PDATA_H__ to __WM8904_PDATA_H__ what corresponding with the file name. Fixes: a91eb199e4dc ("ASoC: Initial WM8904 CODEC driver") Signed-off-by: Wei Li <liwei391(a)huawei.com> --- include/sound/wm8904.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/wm8904.h b/include/sound/wm8904.h index 88ac1870510e..f74293a83a17 100644 --- a/include/sound/wm8904.h +++ b/include/sound/wm8904.h @@ -7,8 +7,8 @@ * Author: Mark Brown <broonie(a)opensource.wolfsonmicro.com> */ -#ifndef __MFD_WM8994_PDATA_H__ -#define __MFD_WM8994_PDATA_H__ +#ifndef __WM8904_PDATA_H__ +#define __WM8904_PDATA_H__ /* Used to enable configuration of a GPIO to all zeros */ #define WM8904_GPIO_NO_CONFIG 0x8000 -- 2.25.1

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[PATCH] ALSA: hda/hdmi - enable runtime pm for more AMD display audio
by Evan Quan 08 Nov '22

08 Nov '22

We are able to power down the GPU and audio via the GPU driver so flag these asics as supporting runtime pm. Signed-off-by: Evan Quan <evan.quan(a)amd.com> Change-Id: Ib7ea9bfe7b62b98afd63f96a2d98795504e7ae04 --- sound/pci/hda/hda_intel.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 0a83eb6b88b1..e6a33e34efb9 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2712,6 +2712,9 @@ static const struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x1002, 0xab28), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS | AZX_DCAPS_PM_RUNTIME }, + { PCI_DEVICE(0x1002, 0xab30), + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS | + AZX_DCAPS_PM_RUNTIME }, { PCI_DEVICE(0x1002, 0xab38), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS | AZX_DCAPS_PM_RUNTIME }, -- 2.34.1

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[PATCH 1/2] ALSA: usb-audio: Yet more regression for for the delayed card registration
by Takashi Iwai 08 Nov '22

08 Nov '22

Although we tried to fix the regression for the recent changes with the delayed card registration, it doesn't seem covering the all cases; e.g. on Roland EDIROL M-100FX, where the generic quirk for Roland devices is applied, it misses the card registration because the detection of the last interface (apparently for MIDI) fails. This patch is an attempt to recover from those failures by calling the card register also at the error path for the secondary interfaces. The card register condition is also extended to match with the old check in the previous patch, too (i.e. the simple check of the interface number) for catching the probe with errors. Fixes: 39efc9c8a973 ("ALSA: usb-audio: Fix last interface check for registration") Cc: <stable(a)vger.kernel.org> Link: https://bugzilla.suse.com/show_bug.cgi?id=1205111 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/card.c | 29 ++++++++++++++++++++--------- 1 file changed, 20 insertions(+), 9 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index a5ed11ea1145..26268ffb8274 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -742,6 +742,18 @@ get_alias_quirk(struct usb_device *dev, unsigned int id) return NULL; } +/* register card if we reach to the last interface or to the specified + * one given via option + */ +static int try_to_register_card(struct snd_usb_audio *chip, int ifnum) +{ + if (check_delayed_register_option(chip) == ifnum || + chip->last_iface == ifnum || + usb_interface_claimed(usb_ifnum_to_if(chip->dev, chip->last_iface))) + return snd_card_register(chip->card); + return 0; +} + /* * probe the active usb device * @@ -880,15 +892,9 @@ static int usb_audio_probe(struct usb_interface *intf, chip->need_delayed_register = false; /* clear again */ } - /* register card if we reach to the last interface or to the specified - * one given via option - */ - if (check_delayed_register_option(chip) == ifnum || - usb_interface_claimed(usb_ifnum_to_if(dev, chip->last_iface))) { - err = snd_card_register(chip->card); - if (err < 0) - goto __error; - } + err = try_to_register_card(chip, ifnum); + if (err < 0) + goto __error_no_register; if (chip->quirk_flags & QUIRK_FLAG_SHARE_MEDIA_DEVICE) { /* don't want to fail when snd_media_device_create() fails */ @@ -907,6 +913,11 @@ static int usb_audio_probe(struct usb_interface *intf, return 0; __error: + /* in the case of error in secondary interface, still try to register */ + if (chip) + try_to_register_card(chip, ifnum); + + __error_no_register: if (chip) { /* chip->active is inside the chip->card object, * decrement before memory is possibly returned. -- 2.35.3

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[PATCH] ASoC: Trival: Fix spelling error in Kconfig help.
by Maurycy Z 07 Nov '22

07 Nov '22

In a few places in Kconfig files, 'system' is spelled as 'ststem'. This patch changes them to the intended 'system'. Signed-off-by: Maurycy Zalweski <10maurycy10(a)gmail.com> --- sound/soc/fsl/Kconfig | 2 +- sound/soc/generic/Kconfig | 4 ++-- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 614eceda6..c7fe12fbd 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -366,7 +366,7 @@ config SND_SOC_IMX_CARD help This option enables audio sound card support for i.MX boards with OF-graph DT bindings. - It also support DPCM of single CPU multi Codec ststem. + It also support DPCM of single CPU multi Codec system. endif # SND_IMX_SOC diff --git a/sound/soc/generic/Kconfig b/sound/soc/generic/Kconfig index b6df4e26b..955ac8d3f 100644 --- a/sound/soc/generic/Kconfig +++ b/sound/soc/generic/Kconfig @@ -7,7 +7,7 @@ config SND_SIMPLE_CARD select SND_SIMPLE_CARD_UTILS help This option enables generic simple sound card support - It also support DPCM of multi CPU single Codec ststem. + It also support DPCM of multi CPU single Codec system. config SND_AUDIO_GRAPH_CARD tristate "ASoC Audio Graph sound card support" @@ -16,7 +16,7 @@ config SND_AUDIO_GRAPH_CARD help This option enables generic simple sound card support with OF-graph DT bindings. - It also support DPCM of multi CPU single Codec ststem. + It also support DPCM of multi CPU single Codec system. config SND_AUDIO_GRAPH_CARD2 tristate "ASoC Audio Graph sound card2 support" -- 2.38.1

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[PATCH] ALSA: usb-audio: Add quirks for MacroSilicon MS2100/MS2106 devices (4.9-5.4)
by John Veness 07 Nov '22

07 Nov '22

commit 6e2c9105e0b743c92a157389d40f00b81bdd09fe upstream. Treat the claimed 96kHz 1ch in the descriptors as 48kHz 2ch, so that the audio stream doesn't sound mono. Also fix initial stream alignment, so that left and right channels are in the correct order. Signed-off-by: John Veness <john-linux(a)pelago.org.uk> Link: https://lore.kernel.org/r/20220624140757.28758-1-john-linux@pelago.org.uk Cc: <stable(a)vger.kernel.org> # 5.4 Cc: <stable(a)vger.kernel.org> # 4.19 Cc: <stable(a)vger.kernel.org> # 4.14 Cc: <stable(a)vger.kernel.org> # 4.9 --- Backported to 4.9-5.4. sound/usb/quirks-table.h | 58 ++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks.c | 1 + 2 files changed, 59 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index c29ccdf9e8bc..5f44b142a8b0 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3759,6 +3759,64 @@ ALC1220_VB_DESKTOP(0x26ce, 0x0a01), /* Asrock TRX40 Creator */ } }, +/* + * MacroSilicon MS2100/MS2106 based AV capture cards + * + * These claim 96kHz 1ch in the descriptors, but are actually 48kHz 2ch. + * They also need QUIRK_AUDIO_ALIGN_TRANSFER, which makes one wonder if + * they pretend to be 96kHz mono as a workaround for stereo being broken + * by that... + * + * They also have an issue with initial stream alignment that causes the + * channels to be swapped and out of phase, which is dealt with in quirks.c. + */ +{ + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .idVendor = 0x534d, + .idProduct = 0x0021, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "MacroSilicon", + .product_name = "MS210x", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = &(const struct snd_usb_audio_quirk[]) { + { + .ifnum = 2, + .type = QUIRK_AUDIO_ALIGN_TRANSFER, + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_MIXER, + }, + { + .ifnum = 3, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels = 2, + .iface = 3, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0, + .endpoint = 0x82, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 48000, + .rate_max = 48000, + } + }, + { + .ifnum = -1 + } + } + } +}, + /* * MacroSilicon MS2109 based HDMI capture cards * diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 72223545abfd..d61f95dc1abf 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1440,6 +1440,7 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, case USB_ID(0x041e, 0x3f19): /* E-Mu 0204 USB */ set_format_emu_quirk(subs, fmt); break; + case USB_ID(0x534d, 0x0021): /* MacroSilicon MS2100/MS2106 */ case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */ subs->stream_offset_adj = 2; break; -- 2.34.1

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[PATCH] ASoC: SOF: topology: No need to assign core ID if token parsing failed
by Peter Ujfalusi 07 Nov '22

07 Nov '22

Move the return value check before attempting to assign the core ID to the swidget since we are going to fail the sof_widget_ready() and free up swidget anyways. Fixes: 909dadf21aae ("ASoC: SOF: topology: Make DAI widget parsing IPC agnostic") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/sof/topology.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 38855dd60617..6a0e7f3b5023 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -1344,16 +1344,6 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index, break; } - if (sof_debug_check_flag(SOF_DBG_DISABLE_MULTICORE)) { - swidget->core = SOF_DSP_PRIMARY_CORE; - } else { - int core = sof_get_token_value(SOF_TKN_COMP_CORE_ID, swidget->tuples, - swidget->num_tuples); - - if (core >= 0) - swidget->core = core; - } - /* check token parsing reply */ if (ret < 0) { dev_err(scomp->dev, @@ -1365,6 +1355,16 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index, return ret; } + if (sof_debug_check_flag(SOF_DBG_DISABLE_MULTICORE)) { + swidget->core = SOF_DSP_PRIMARY_CORE; + } else { + int core = sof_get_token_value(SOF_TKN_COMP_CORE_ID, swidget->tuples, + swidget->num_tuples); + + if (core >= 0) + swidget->core = core; + } + /* bind widget to external event */ if (tw->event_type) { if (widget_ops[w->id].bind_event) { -- 2.38.1

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[PATCH v1] ASoC: sof_es8336: reduce pop noise on speaker
by Zhu Ning 07 Nov '22

07 Nov '22

The Speaker GPIO needs to be turned on slightly behind the codec turned on. It also need to be turned off slightly before the codec turned down. Current code uses delay in DAPM_EVENT to do it but the mdelay delays the DAPM itself and thus has no effect. A delayed_work is added to turn on the speaker. The Speaker is turned off in .trigger since trigger is called slightly before the DAPM events. Signed-off-by: Zhu Ning <zhuning(a)everest-semi.com> ------------ v1: cancel delayed work while disabling speaker. --- sound/soc/intel/boards/sof_es8336.c | 60 ++++++++++++++++++++++------- 1 file changed, 46 insertions(+), 14 deletions(-) diff --git a/sound/soc/intel/boards/sof_es8336.c b/sound/soc/intel/boards/sof_es8336.c index fbb42e54947a..70713e4b07dc 100644 --- a/sound/soc/intel/boards/sof_es8336.c +++ b/sound/soc/intel/boards/sof_es8336.c @@ -63,6 +63,7 @@ struct sof_es8336_private { struct snd_soc_jack jack; struct list_head hdmi_pcm_list; bool speaker_en; + struct delayed_work pcm_pop_work; }; struct sof_hdmi_pcm { @@ -111,6 +112,46 @@ static void log_quirks(struct device *dev) dev_info(dev, "quirk headset at mic1 port enabled\n"); } +static void pcm_pop_work_events(struct work_struct *work) +{ + struct sof_es8336_private *priv = + container_of(work, struct sof_es8336_private, pcm_pop_work.work); + + gpiod_set_value_cansleep(priv->gpio_speakers, priv->speaker_en); + + if (quirk & SOF_ES8336_HEADPHONE_GPIO) + gpiod_set_value_cansleep(priv->gpio_headphone, priv->speaker_en); + +} + +static int sof_8336_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_card *card = rtd->card; + struct sof_es8336_private *priv = snd_soc_card_get_drvdata(card); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + break; + + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + if (priv->speaker_en == false) + if (substream->stream == 0) { + cancel_delayed_work(&priv->pcm_pop_work); + gpiod_set_value_cansleep(priv->gpio_speakers, true); + } + break; + default: + return -EINVAL; + } + + return 0; +} + static int sof_es8316_speaker_power_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -122,19 +163,7 @@ static int sof_es8316_speaker_power_event(struct snd_soc_dapm_widget *w, priv->speaker_en = !SND_SOC_DAPM_EVENT_ON(event); - if (SND_SOC_DAPM_EVENT_ON(event)) - msleep(70); - - gpiod_set_value_cansleep(priv->gpio_speakers, priv->speaker_en); - - if (!(quirk & SOF_ES8336_HEADPHONE_GPIO)) - return 0; - - if (SND_SOC_DAPM_EVENT_ON(event)) - msleep(70); - - gpiod_set_value_cansleep(priv->gpio_headphone, priv->speaker_en); - + queue_delayed_work(system_wq, &priv->pcm_pop_work, msecs_to_jiffies(70)); return 0; } @@ -344,6 +373,7 @@ static int sof_es8336_hw_params(struct snd_pcm_substream *substream, /* machine stream operations */ static struct snd_soc_ops sof_es8336_ops = { .hw_params = sof_es8336_hw_params, + .trigger = sof_8336_trigger, }; static struct snd_soc_dai_link_component platform_component[] = { @@ -723,7 +753,8 @@ static int sof_es8336_probe(struct platform_device *pdev) } INIT_LIST_HEAD(&priv->hdmi_pcm_list); - + INIT_DELAYED_WORK(&priv->pcm_pop_work, + pcm_pop_work_events); snd_soc_card_set_drvdata(card, priv); if (mach->mach_params.dmic_num > 0) { @@ -752,6 +783,7 @@ static int sof_es8336_remove(struct platform_device *pdev) struct snd_soc_card *card = platform_get_drvdata(pdev); struct sof_es8336_private *priv = snd_soc_card_get_drvdata(card); + cancel_delayed_work(&priv->pcm_pop_work); gpiod_put(priv->gpio_speakers); device_remove_software_node(priv->codec_dev); put_device(priv->codec_dev); -- 2.37.2

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[PATCH v4 0/3] Convert soundwire bindings to DT schema
by Srinivasa Rao Mandadapu 07 Nov '22

07 Nov '22

Convert soundwire bindings text file to DT schema and update device tree entries to follow strict dt-bindings. Changes Since V3: -- Remove subnode description and add appropriate pattern properties. -- Add interrput names in example. -- update some properties description. -- Revert minIteams change in previous version. -- Rebase to latest code base. -- Update dtsi node names, -- Remove redundant property in soundwire node. Srinivasa Rao Mandadapu (3): arm64: dts: qcom: Update soundwire slave node names arm64: dts: qcom: Remove redundant soundwire property dt-bindings: soundwire: Convert text bindings to DT Schema .../devicetree/bindings/soundwire/qcom,sdw.txt | 215 ------------------ .../bindings/soundwire/qcom,soundwire.yaml | 241 +++++++++++++++++++++ arch/arm64/boot/dts/qcom/qrb5165-rb5.dts | 4 +- arch/arm64/boot/dts/qcom/sc7280.dtsi | 1 - arch/arm64/boot/dts/qcom/sdm845-db845c.dts | 4 +- .../boot/dts/qcom/sdm850-lenovo-yoga-c630.dts | 4 +- arch/arm64/boot/dts/qcom/sdm850-samsung-w737.dts | 4 +- arch/arm64/boot/dts/qcom/sm8250-mtp.dts | 4 +- arch/arm64/boot/dts/qcom/sm8250.dtsi | 1 - 9 files changed, 251 insertions(+), 227 deletions(-) delete mode 100644 Documentation/devicetree/bindings/soundwire/qcom,sdw.txt create mode 100644 Documentation/devicetree/bindings/soundwire/qcom,soundwire.yaml -- 2.7.4

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ALC1220 DualCodec no optical out
by GitHub issues - opened 07 Nov '22

07 Nov '22

alsa-project/alsa-ucm-conf issue #239 was opened from Almoped: With current pipewire and alsa-ucm 1.2.8 I am unable to select profiles which has iec958 output, actually I am unable to select any card profile at all in pavucontrol. This is possibly because all the electrical output jacks are disconnected and optical out does not have jackdetect. Or maybe DualCodecs/HiFi.conf does not configure optical out. If I remove alsa/ucm2/HDA/DualCodecs/HiFi.conf it uses alsa-card-profile/mixer/profile-sets/default.conf, goes on a probing mission I get a working optical out and the possibility to select an array of other card profiles in pavucontrol. Can anyone recommend a better way to solve this? I've attached pactl list cards output with working and nonworking optical out. [pactl_no_optical.txt](https://github.com/alsa-project/alsa-ucm-conf/files/9… [pactl_working_optical.txt](https://github.com/alsa-project/alsa-ucm-conf/fi… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/239 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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