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[PATCH v4] ALSA: core: Fix deadlock when shutdown a frozen userspace
by Ricardo Ribalda 30 Nov '22

30 Nov '22

During kexec(), the userspace is frozen. Therefore we cannot wait for it to complete. Avoid running snd_sof_machine_unregister during shutdown. This fixes: [ 84.943749] Freezing user space processes ... (elapsed 0.111 seconds) done. [ 246.784446] INFO: task kexec-lite:5123 blocked for more than 122 seconds. [ 246.819035] Call Trace: [ 246.821782] <TASK> [ 246.824186] __schedule+0x5f9/0x1263 [ 246.828231] schedule+0x87/0xc5 [ 246.831779] snd_card_disconnect_sync+0xb5/0x127 ... [ 246.889249] snd_sof_device_shutdown+0xb4/0x150 [ 246.899317] pci_device_shutdown+0x37/0x61 [ 246.903990] device_shutdown+0x14c/0x1d6 [ 246.908391] kernel_kexec+0x45/0xb9 And: [ 246.893222] INFO: task kexec-lite:4891 blocked for more than 122 seconds. [ 246.927709] Call Trace: [ 246.930461] <TASK> [ 246.932819] __schedule+0x5f9/0x1263 [ 246.936855] ? fsnotify_grab_connector+0x5c/0x70 [ 246.942045] schedule+0x87/0xc5 [ 246.945567] schedule_timeout+0x49/0xf3 [ 246.949877] wait_for_completion+0x86/0xe8 [ 246.954463] snd_card_free+0x68/0x89 ... [ 247.001080] platform_device_unregister+0x12/0x35 Cc: stable(a)vger.kernel.org Fixes: 83bfc7e793b5 ("ASoC: SOF: core: unregister clients and machine drivers in .shutdown") Signed-off-by: Ricardo Ribalda <ribalda(a)chromium.org> --- To: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> To: Liam Girdwood <lgirdwood(a)gmail.com> To: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> To: Bard Liao <yung-chuan.liao(a)linux.intel.com> To: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> To: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> To: Daniel Baluta <daniel.baluta(a)nxp.com> To: Mark Brown <broonie(a)kernel.org> To: Jaroslav Kysela <perex(a)perex.cz> To: Takashi Iwai <tiwai(a)suse.com> Cc: sound-open-firmware(a)alsa-project.org Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org --- Changes in v4: - Do not call snd_sof_machine_unregister from shutdown. - Link to v3: https://lore.kernel.org/r/20221127-snd-freeze-v3-0-a2eda731ca14@chromium.org Changes in v3: - Wrap pm_freezing in a function - Link to v2: https://lore.kernel.org/r/20221127-snd-freeze-v2-0-d8a425ea9663@chromium.org Changes in v2: - Only use pm_freezing if CONFIG_FREEZER - Link to v1: https://lore.kernel.org/r/20221127-snd-freeze-v1-0-57461a366ec2@chromium.org --- sound/soc/sof/core.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 3e6141d03770..9616ba607ded 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -475,19 +475,16 @@ EXPORT_SYMBOL(snd_sof_device_remove); int snd_sof_device_shutdown(struct device *dev) { struct snd_sof_dev *sdev = dev_get_drvdata(dev); - struct snd_sof_pdata *pdata = sdev->pdata; if (IS_ENABLED(CONFIG_SND_SOC_SOF_PROBE_WORK_QUEUE)) cancel_work_sync(&sdev->probe_work); /* - * make sure clients and machine driver(s) are unregistered to force - * all userspace devices to be closed prior to the DSP shutdown sequence + * make sure clients are unregistered prior to the DSP shutdown + * sequence. */ sof_unregister_clients(sdev); - snd_sof_machine_unregister(sdev, pdata); - if (sdev->fw_state == SOF_FW_BOOT_COMPLETE) return snd_sof_shutdown(sdev); --- base-commit: 4312098baf37ee17a8350725e6e0d0e8590252d4 change-id: 20221127-snd-freeze-1ee143228326 Best regards, -- Ricardo Ribalda <ribalda(a)chromium.org>

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[PATCH] ALSA: dice: add support for Focusrite Saffire Pro 40 with TCD3070 ASIC
by Takashi Sakamoto 30 Nov '22

30 Nov '22

TC Applied Technologies (TCAT) produces TCD3070 as final DICE ASIC for communication in IEEE 1394 bus for IEC 61883-1/6 protocol. As long as I know, latter model of Focusrite Saffire Pro 40 is an application of the ASIC and only in the market for consumers. This patchset adds support for the device. The device has several remarkable points. 1. No support for extended synchronization information section in TCAT general protocol. The value of GLOBAL_EXTENDED_STATUS register is always zero. Additionally, NOTIFY_EXT_STATUS message is never emitted. 2. No support for TCAT protocol extension. Hard coding is required for format of CIP payload. 3. During several seconds after changing sampling rate, the block to process PCM frames is under disfunction. When starting packet streaming during the state, the block is never function till configuring different sampling rate and several seconds. This commit adds support for the device. The item 1 and 2 can be adaptable, while item 3 is not. It's not preferable that user process is forced to sleep during the disfunction in the call of ioctl(2) with SNDRV_PCM_IOCTL_HW_PARAMS or SNDRV_PCM_IOCTL_PREPARE request. It's inconvenient but let user configure preferable sampling rate in advance of starting PCM substream. The content of configuration ROM in the device I used is available at: * https://github.com/takaswie/am-config-roms/ I note that any mixer control operation is implemented by unique transaction. The frame of request consists of 16 bytes header followed by payload. header (4 quadlets): 1st: the type of request, prefixed with 0x8000 2nd: counter at 2 bytes in MSB side, the length of data at 2 bytes in LSB side 3rd: parameter 0 4th: parameter 1 payload (variable length if need): 5th-: data according to parameters The request frame is sent by block write request to 0x'ffff'e040'01c0. The frame of response is similar to the frame of request, but it is header only, thus fixed to 16 bytes. The response frame is sent to the address which is registered by lock transaction to 0x'ffff'e040'0008. If the operation results in batch of data, the 2nd quadlet of header includes the length of data like request. The data is itself readable by read block request to 0x'ffff'e040'0030, which includes both header and payload for data, thus the length to read should be the length of data plus 16 bytes for header The actual value of request, parameter 0, parameter 1, and data is unclear yet. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/dice/Makefile | 2 +- sound/firewire/dice/dice-focusrite.c | 25 +++++++++++++++++++++++++ sound/firewire/dice/dice.c | 10 ++++++++++ sound/firewire/dice/dice.h | 1 + 4 files changed, 37 insertions(+), 1 deletion(-) create mode 100644 sound/firewire/dice/dice-focusrite.c diff --git a/sound/firewire/dice/Makefile b/sound/firewire/dice/Makefile index 9bf7b960a720..a5f3fbf28b8c 100644 --- a/sound/firewire/dice/Makefile +++ b/sound/firewire/dice/Makefile @@ -2,5 +2,5 @@ snd-dice-objs := dice-transaction.o dice-stream.o dice-proc.o dice-midi.o \ dice-pcm.o dice-hwdep.o dice.o dice-tcelectronic.o \ dice-alesis.o dice-extension.o dice-mytek.o dice-presonus.o \ - dice-harman.o + dice-harman.o dice-focusrite.o obj-$(CONFIG_SND_DICE) += snd-dice.o diff --git a/sound/firewire/dice/dice-focusrite.c b/sound/firewire/dice/dice-focusrite.c new file mode 100644 index 000000000000..caf9f620c5d0 --- /dev/null +++ b/sound/firewire/dice/dice-focusrite.c @@ -0,0 +1,25 @@ +// SPDX-License-Identifier: GPL-2.0 +// dice-focusrite.c - a part of driver for DICE based devices +// +// Copyright (c) 2022 Takashi Sakamoto +// +// Licensed under the terms of the GNU General Public License, version 2. + +#include "dice.h" + +int snd_dice_detect_focusrite_pro40_tcd3070_formats(struct snd_dice *dice) +{ + // Focusrite shipped several variants of Saffire Pro 40. One of them is based on TCD3070-CH + // apart from the others with TCD2220. It doesn't support TCAT protocol extension. + dice->tx_pcm_chs[0][0] = 20; + dice->tx_midi_ports[0] = 1; + dice->rx_pcm_chs[0][0] = 20; + dice->rx_midi_ports[0] = 1; + + dice->tx_pcm_chs[0][1] = 16; + dice->tx_midi_ports[1] = 1; + dice->rx_pcm_chs[0][1] = 16; + dice->rx_midi_ports[1] = 1; + + return 0; +} diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index f75902bc8e74..6036a5edbcb8 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -382,6 +382,16 @@ static const struct ieee1394_device_id dice_id_table[] = { .model_id = 0x000001, .driver_data = (kernel_ulong_t)snd_dice_detect_harman_formats, }, + // Focusrite Saffire Pro 40 with TCD3070-CH. + // The model has quirk in its GUID, in which model field is 0x000013 and different from + // model ID (0x0000de) in its root/unit directory. + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_FOCUSRITE, + .model_id = 0x0000de, + .driver_data = (kernel_ulong_t)snd_dice_detect_focusrite_pro40_tcd3070_formats, + }, { .match_flags = IEEE1394_MATCH_VERSION, .version = DICE_INTERFACE, diff --git a/sound/firewire/dice/dice.h b/sound/firewire/dice/dice.h index fd440cc625f9..674f7d552c2e 100644 --- a/sound/firewire/dice/dice.h +++ b/sound/firewire/dice/dice.h @@ -231,5 +231,6 @@ int snd_dice_detect_extension_formats(struct snd_dice *dice); int snd_dice_detect_mytek_formats(struct snd_dice *dice); int snd_dice_detect_presonus_formats(struct snd_dice *dice); int snd_dice_detect_harman_formats(struct snd_dice *dice); +int snd_dice_detect_focusrite_pro40_tcd3070_formats(struct snd_dice *dice); #endif -- 2.37.2

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[PATCH] ALSA: dice: fix regression for Lexicon I-ONIX FW810S
by Takashi Sakamoto 30 Nov '22

30 Nov '22

For Lexicon I-ONIX FW810S, the call of ioctl(2) with SNDRV_PCM_IOCTL_HW_PARAMS can returns -ETIMEDOUT. This is a regression due to the commit 41319eb56e19 ("ALSA: dice: wait just for NOTIFY_CLOCK_ACCEPTED after GLOBAL_CLOCK_SELECT operation"). The device does not emit NOTIFY_CLOCK_ACCEPTED notification when accepting GLOBAL_CLOCK_SELECT operation with the same parameters as current ones. This commit fixes the regression. When receiving no notification, return -ETIMEDOUT as long as operating for any change. Fixes: 41319eb56e19 ("ALSA: dice: wait just for NOTIFY_CLOCK_ACCEPTED after GLOBAL_CLOCK_SELECT operation") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/dice/dice-stream.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c index f99e00083141..4c677c8546c7 100644 --- a/sound/firewire/dice/dice-stream.c +++ b/sound/firewire/dice/dice-stream.c @@ -59,7 +59,7 @@ int snd_dice_stream_get_rate_mode(struct snd_dice *dice, unsigned int rate, static int select_clock(struct snd_dice *dice, unsigned int rate) { - __be32 reg; + __be32 reg, new; u32 data; int i; int err; @@ -83,15 +83,17 @@ static int select_clock(struct snd_dice *dice, unsigned int rate) if (completion_done(&dice->clock_accepted)) reinit_completion(&dice->clock_accepted); - reg = cpu_to_be32(data); + new = cpu_to_be32(data); err = snd_dice_transaction_write_global(dice, GLOBAL_CLOCK_SELECT, - &reg, sizeof(reg)); + &new, sizeof(new)); if (err < 0) return err; if (wait_for_completion_timeout(&dice->clock_accepted, - msecs_to_jiffies(NOTIFICATION_TIMEOUT_MS)) == 0) - return -ETIMEDOUT; + msecs_to_jiffies(NOTIFICATION_TIMEOUT_MS)) == 0) { + if (reg != new) + return -ETIMEDOUT; + } return 0; } -- 2.37.2

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[RESEND PATCH -next] ASoC: amd: yc: Add Xiaomi Redmi Book Pro 14 2022 into DMI table
by Artem Lukyanov 30 Nov '22

30 Nov '22

This model requires an additional detection quirk to enable the internal microphone - BIOS doesn't seem to support AcpDmicConnected (nothing in acpidump output). Signed-off-by: Artem Lukyanov <dukzcry(a)ya.ru> --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index d9715bea965e..1f0b5527c594 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -213,6 +213,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Alienware m17 R5 AMD"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "TIMI"), + DMI_MATCH(DMI_PRODUCT_NAME, "Redmi Book Pro 14 2022"), + } + }, {} }; -- 2.36.2

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[PATCH] ASoC: SOF: mediatek: add shutdown callback
by Ricardo Ribalda 30 Nov '22

30 Nov '22

If we do not shutdown the peripheral properly at shutdown, the whole system crashes after kexec() on the first io access. Let's implement the appropriate callback. Signed-off-by: Ricardo Ribalda <ribalda(a)chromium.org> --- To: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> To: Liam Girdwood <lgirdwood(a)gmail.com> To: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> To: Bard Liao <yung-chuan.liao(a)linux.intel.com> To: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> To: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> To: Daniel Baluta <daniel.baluta(a)nxp.com> To: Mark Brown <broonie(a)kernel.org> To: Jaroslav Kysela <perex(a)perex.cz> To: Takashi Iwai <tiwai(a)suse.com> To: Matthias Brugger <matthias.bgg(a)gmail.com> Cc: sound-open-firmware(a)alsa-project.org Cc: alsa-devel(a)alsa-project.org Cc: linux-arm-kernel(a)lists.infradead.org Cc: linux-mediatek(a)lists.infradead.org Cc: linux-kernel(a)vger.kernel.org --- sound/soc/sof/mediatek/mt8186/mt8186.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/sof/mediatek/mt8186/mt8186.c b/sound/soc/sof/mediatek/mt8186/mt8186.c index 181189e00e02..79da25725987 100644 --- a/sound/soc/sof/mediatek/mt8186/mt8186.c +++ b/sound/soc/sof/mediatek/mt8186/mt8186.c @@ -430,6 +430,11 @@ static int mt8186_dsp_remove(struct snd_sof_dev *sdev) return 0; } +static int mt8186_dsp_shutdown(struct snd_sof_dev *sdev) +{ + return snd_sof_suspend(sdev->dev); +} + static int mt8186_dsp_suspend(struct snd_sof_dev *sdev, u32 target_state) { mt8186_sof_hifixdsp_shutdown(sdev); @@ -538,6 +543,7 @@ static struct snd_sof_dsp_ops sof_mt8186_ops = { /* probe and remove */ .probe = mt8186_dsp_probe, .remove = mt8186_dsp_remove, + .shutdown = mt8186_dsp_shutdown, /* DSP core boot */ .run = mt8186_run, @@ -629,6 +635,7 @@ MODULE_DEVICE_TABLE(of, sof_of_mt8186_ids); static struct platform_driver snd_sof_of_mt8186_driver = { .probe = sof_of_probe, .remove = sof_of_remove, + .shutdown = sof_of_shutdown, .driver = { .name = "sof-audio-of-mt8186", .pm = &sof_of_pm, --- base-commit: 4312098baf37ee17a8350725e6e0d0e8590252d4 change-id: 20221127-mtk-snd-e0abf36be4c0 Best regards, -- Ricardo Ribalda <ribalda(a)chromium.org>

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[PATCH] ASoC: amd: yc: Add Xiaomi Redmi Book Pro 14 2022 into DMI table
by Artem Lukyanov 30 Nov '22

30 Nov '22

This model requires an additional detection quirk to enable the internal microphone - BIOS doesn't seem to support AcpDmicConnected (nothing in acpidump output). Signed-off-by: Artem Lukyanov <dukzcry(a)ya.ru> --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index d9715bea9..1f0b5527c 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -213,6 +213,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Alienware m17 R5 AMD"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "TIMI"), + DMI_MATCH(DMI_PRODUCT_NAME, "Redmi Book Pro 14 2022"), + } + }, {} }; -- 2.36.2

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arecord leaves doubled audio at end after a SIGTERM
by GitHub issues - edited 30 Nov '22

30 Nov '22

alsa-project/alsa-utils issue #179 was edited from chconnor: Hello: I issue a command like this: `arecord -t wav -c 1 -f S16_LE -r44100 -D hw:2,0 test.wav ` It works fine, but then I hit ctrl-C to terminate. This works, but the file has "junk" at the end that repeats the last fraction of a second. E.g. in this demo file I ctrl-C when I say the word "one", and the recording is like "three, two, on-on": https://caseyconnor.org/pub/audio/arecord_bug.wav It's as if it is mistakenly writing a buffer an extra time? (In my total ignorance and total lack of qualifications, I traced the code to [here](https://github.com/alsa-project/alsa-lib/blob/d28e8cb29485cc93f741b01…, but I get lost and can't see if anything is awry...) (Edit: version 1.2.6 on Lubuntu 22.04 with a Shure X2U interface) Issue URL : https://github.com/alsa-project/alsa-utils/issues/179 Repository URL: https://github.com/alsa-project/alsa-utils

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arecord leaves doubled audio at end after a SIGTERM
by GitHub issues - edited 30 Nov '22

30 Nov '22

alsa-project/alsa-utils issue #179 was edited from chconnor: Hello: I issue a command like this: `arecord -t wav -c 1 -f S16_LE -r44100 -D hw:2,0 test.wav ` It works fine, but then I hit ctrl-C to terminate. This works, but the file has "junk" at the end that repeats the last fraction of a second. E.g. in this demo file I ctrl-C when I say the word "one", and the recording is like "three, two, on-on": https://caseyconnor.org/pub/audio/arecord_bug.wav It's as if it is mistakenly writing a buffer an extra time? (In my total ignorance and total lack of qualifications, I traced the code to [here](https://github.com/alsa-project/alsa-lib/blob/d28e8cb29485cc93f741b01…, but I get lost and can't see if anything is awry...) Issue URL : https://github.com/alsa-project/alsa-utils/issues/179 Repository URL: https://github.com/alsa-project/alsa-utils

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arecord leaves doubled audio at end after a SIGTERM
by GitHub issues - opened 30 Nov '22

30 Nov '22

alsa-project/alsa-utils issue #179 was opened from chconnor: Hello: I issue a command like this: `arecord -t wav -c 1 -f S16_LE -r44100 -D hw:2,0 test.wav ` It works fine, but then I hit ctrl-C to terminate. This works, but the file has "junk" at the end that repeats the last fraction of a second. E.g. in this demo file I ctrl-C when I say the word "one", and the recording is like "three, two, on-on": http://caseyconnor.org/pub/audio/arecord_bug.wav It's as if it is mistakenly writing a buffer an extra time? (In my total ignorance and total lack of qualifications, I traced the code to [here](https://github.com/alsa-project/alsa-lib/blob/d28e8cb29485cc93f741b01…, but I get lost and can't see if anything is awry...) Issue URL : https://github.com/alsa-project/alsa-utils/issues/179 Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH v2 0/2] ASoC: Intel: avs: rt5682: Refactor jack handling
by Cezary Rojewski 29 Nov '22

29 Nov '22

Leftover from recent series [1]. Following changes are proposed for the rt5682 sound card driver: 1) Move jack unassignment from platform_device->remove() to dai_link->exit(). This is done to make jack init and deinit flows symmetric 2) Remove platform_device->remove() function 3) Simplify card->suspend_pre() and card->resume_post() by making use of snd_soc_card_get_codec_dai() helper Changes in v2: - fixed compilation of the first patch in the series by moving offending pieces to the second patch [1]: https://lore.kernel.org/alsa-devel/20221125184032.2565979-1-cezary.rojewski… Amadeusz Sławiński (2): ASoC: Intel: avs: rt5682: Add define for codec DAI name ASoC: Intel: avs: rt5682: Refactor jack handling sound/soc/intel/avs/boards/rt5682.c | 31 ++++++++++++----------------- 1 file changed, 13 insertions(+), 18 deletions(-) -- 2.25.1

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