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ucm2: Qualcomm: sc8280xp: fix device numbers
by GitHub pull_request - opened 28 Jun '23

28 Jun '23

alsa-project/alsa-ucm-conf pull request #331 was opened from Srinivas-Kandagatla: Due to a bug in the Qualcomm ASoC drivers, we ended up with card device numbers starting at some number that is not dai-link id. This bug is now fixed, so update the ucm files inline with this. Fixes: https://github.com/alsa-project/alsa-ucm-conf/pull/284 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/331 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/331.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

topology: pre-processor: Add support for object arrays
by GitHub pull_request - edited 27 Jun '23

27 Jun '23

alsa-project/alsa-utils pull request #216 was edited from ranj063: Introduce a new keyword, "CombineArrays" to instantiate multiple nodes of the type specified. For example: CombineArrays.Object.Base.input_audio_format [ { in_rate [ 8000 16000 48000 ] in_bit_depth [ 32 ] in_valid_bit_depth [ 24 32 ] } ] This would be expanded into 6 objects with the rate, bit_depth and valid_bit_depth combinations of the arrays of values above. Object.Base.input_audio_format [ { in_rate 8000 in_bit_depth 32 in_valid_bit_depth 32 } { in_rate 16000 in_bit_depth 32 in_valid_bit_depth 32 } { in_rate 48000 in_bit_depth 32 in_valid_bit_depth 32 } { in_rate 8000 in_bit_depth 32 in_valid_bit_depth 24 } { in_rate 16000 in_bit_depth 32 in_valid_bit_depth 24 } { in_rate 48000 in_bit_depth 32 in_valid_bit_depth 24 } ] The CombineArrays definition is an array so that multiple combinations can be specified in order to deal with only valid combinations. For example, 16-bit bit_depth is only valid with 16-bit valid_bit_depth. So if we also want to add those combinations with the multiple rates, the definition would look like: CombineArrays.Object.Base.input_audio_format [ { in_rate [ 8000 16000 48000 ] in_bit_depth [ 32 ] in_valid_bit_depth [ 24 32 ] } { in_rate [ 8000 16000 48000 ] in_bit_depth [ 16 ] in_valid_bit_depth [ 16 ] } ] Request URL : https://github.com/alsa-project/alsa-utils/pull/216 Patch URL : https://github.com/alsa-project/alsa-utils/pull/216.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Intel/sof-essx8336: Fix HiFi.conf
by GitHub pull_request - edited 27 Jun '23

27 Jun '23

alsa-project/alsa-ucm-conf pull request #328 was edited from xry111: 1. <del>Enable Left/Right Headphone Mixer Left/Right DAC Switch in BootSequence. Despite they seem enabled in sof-essx8336.conf, if I don't add this I'll need to enable them with alsamixer. I guess they are killed by disdevall.</del> 2. Switch Headphone on/off in Headphones enable/disable sequence. This should be obvious. 3. "Headset Switch" is not recognized and triggers errors running "alsaucm -c hw:0 set _verb Hifi". Change it to "Headset Mic Switch". And in the disable sequence we should turn it off, not on. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/328 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/328.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ALSA: sb: fix potential deadlock on &chip->mixer_lock
by Chengfeng Ye 27 Jun '23

27 Jun '23

As &chip->mixer_lock is also acquired by the irq snd_sb8dsp_interrupt() which executes under hard-irq context, code executing under process context should disable irq before acquiring the lock, otherwise deadlock could happen if the process context hold the lock then preempt by the interruption. As the ALSA Driver document described, PCM prepare callbacks are not executed with irq disabled by default, thus the acquiring of &chip->mixer_lock should be irq disabled. Possible deadlock scenario: snd_sb8_playback_prepare -> spin_lock(&chip->mixer_lock); <irq interrupt> -> snd_sb8dsp_interrupt() -> snd_sb8_capture_trigger() -> spin_lock(&chip->mixer_lock); (deadlock here) This flaw was found using an experimental static analysis tool we are developing for irq-related deadlock. The tentative patch fix the potential deadlock by spin_lock_irqsave(). Signed-off-by: Chengfeng Ye <dg573847474(a)gmail.com> --- sound/isa/sb/sb8_main.c | 16 ++++++++++------ 1 file changed, 10 insertions(+), 6 deletions(-) diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c index 2ed176a5a574..81af8dcddcd2 100644 --- a/sound/isa/sb/sb8_main.c +++ b/sound/isa/sb/sb8_main.c @@ -148,10 +148,12 @@ static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream) snd_sbdsp_command(chip, format); else if (stereo) { /* set playback stereo mode */ - spin_lock(&chip->mixer_lock); + unsigned long flags; + + spin_lock_irqsave(&chip->mixer_lock, flags); mixreg = snd_sbmixer_read(chip, SB_DSP_STEREO_SW); snd_sbmixer_write(chip, SB_DSP_STEREO_SW, mixreg | 0x02); - spin_unlock(&chip->mixer_lock); + spin_unlock_irqrestore(&chip->mixer_lock, flags); /* Soundblaster hardware programming reference guide, 3-23 */ snd_sbdsp_command(chip, SB_DSP_DMA8_EXIT); @@ -164,12 +166,13 @@ static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream) } snd_sbdsp_command(chip, SB_DSP_SAMPLE_RATE); if (stereo) { + unsigned long flags; snd_sbdsp_command(chip, 256 - runtime->rate_den / 2); - spin_lock(&chip->mixer_lock); + spin_lock_irqsave(&chip->mixer_lock, flags); /* save output filter status and turn it off */ mixreg = snd_sbmixer_read(chip, SB_DSP_PLAYBACK_FILT); snd_sbmixer_write(chip, SB_DSP_PLAYBACK_FILT, mixreg | 0x20); - spin_unlock(&chip->mixer_lock); + spin_unlock_irqrestore(&chip->mixer_lock, flags); /* just use force_mode16 for temporary storate... */ chip->force_mode16 = mixreg; } else { @@ -289,12 +292,13 @@ static int snd_sb8_capture_prepare(struct snd_pcm_substream *substream) snd_sbdsp_command(chip, SB_DSP_STEREO_8BIT); snd_sbdsp_command(chip, SB_DSP_SAMPLE_RATE); if (stereo) { + unsigned long flags; snd_sbdsp_command(chip, 256 - runtime->rate_den / 2); - spin_lock(&chip->mixer_lock); + spin_lock_irqsave(&chip->mixer_lock, flags); /* save input filter status and turn it off */ mixreg = snd_sbmixer_read(chip, SB_DSP_CAPTURE_FILT); snd_sbmixer_write(chip, SB_DSP_CAPTURE_FILT, mixreg | 0x20); - spin_unlock(&chip->mixer_lock); + spin_lock_irqsave(&chip->mixer_lock, flags); /* just use force_mode16 for temporary storate... */ chip->force_mode16 = mixreg; } else { -- 2.17.1

3 2

[PATCH 0/8] ASoC: Use the maple tree register cache for RealTek SoundWire CODECs
by Mark Brown 27 Jun '23

27 Jun '23

The RealTek SoundWire CODECs don't support bulk register operations so they gain no advantage from using the rbtree cache, convert them to use the maple tree cache instead - it is more modern. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Mark Brown (8): ASoC: rt700: Use maple tree register cache ASoC: rt711: Use maple tree register cache ASoC: rt712: Use maple tree register cache ASoC: rt715: Use maple tree register cache ASoC: rt722: Use maple tree register cache ASoC: rt1308: Use maple tree register cache ASoC: rt1316: Use maple tree register cache ASoC: rt1318: Use maple tree register cache sound/soc/codecs/rt1308-sdw.c | 2 +- sound/soc/codecs/rt1316-sdw.c | 2 +- sound/soc/codecs/rt1318-sdw.c | 2 +- sound/soc/codecs/rt700-sdw.c | 2 +- sound/soc/codecs/rt711-sdca-sdw.c | 4 ++-- sound/soc/codecs/rt711-sdw.c | 2 +- sound/soc/codecs/rt712-sdca-dmic.c | 4 ++-- sound/soc/codecs/rt712-sdca-sdw.c | 4 ++-- sound/soc/codecs/rt715-sdca-sdw.c | 4 ++-- sound/soc/codecs/rt722-sdca-sdw.c | 4 ++-- 10 files changed, 15 insertions(+), 15 deletions(-) --- base-commit: 08e6c4bb17087584261c4aff555d32fc1c620b81 change-id: 20230609-asoc-rt-sdw-maple-fa6714ac4318 Best regards, -- Mark Brown <broonie(a)kernel.org>

3 14

[PATCH AUTOSEL 4.14 5/5] ALSA: seq: oss: Fix racy open/close of MIDI devices
by Sasha Levin 27 Jun '23

27 Jun '23

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 297224fc0922e7385573a30c29ffdabb67f27b7d ] Although snd_seq_oss_midi_open() and snd_seq_oss_midi_close() can be called concurrently from different code paths, we have no proper data protection against races. Introduce open_mutex to each seq_oss_midi object for avoiding the races. Reported-by: "Gong, Sishuai" <sishuai(a)purdue.edu> Closes: https://lore.kernel.org/r/7DC9AF71-F481-4ABA-955F-76C535661E33@purdue.edu Link: https://lore.kernel.org/r/20230612125533.27461-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/seq/oss/seq_oss_midi.c | 35 +++++++++++++++++++------------ 1 file changed, 22 insertions(+), 13 deletions(-) diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c index cc8f06638edca..7226c03f15934 100644 --- a/sound/core/seq/oss/seq_oss_midi.c +++ b/sound/core/seq/oss/seq_oss_midi.c @@ -50,6 +50,7 @@ struct seq_oss_midi { struct snd_midi_event *coder; /* MIDI event coder */ struct seq_oss_devinfo *devinfo; /* assigned OSSseq device */ snd_use_lock_t use_lock; + struct mutex open_mutex; }; @@ -184,6 +185,7 @@ snd_seq_oss_midi_check_new_port(struct snd_seq_port_info *pinfo) mdev->flags = pinfo->capability; mdev->opened = 0; snd_use_lock_init(&mdev->use_lock); + mutex_init(&mdev->open_mutex); /* copy and truncate the name of synth device */ strlcpy(mdev->name, pinfo->name, sizeof(mdev->name)); @@ -332,14 +334,16 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) int perm; struct seq_oss_midi *mdev; struct snd_seq_port_subscribe subs; + int err; if ((mdev = get_mididev(dp, dev)) == NULL) return -ENODEV; + mutex_lock(&mdev->open_mutex); /* already used? */ if (mdev->opened && mdev->devinfo != dp) { - snd_use_lock_free(&mdev->use_lock); - return -EBUSY; + err = -EBUSY; + goto unlock; } perm = 0; @@ -349,14 +353,14 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) perm |= PERM_READ; perm &= mdev->flags; if (perm == 0) { - snd_use_lock_free(&mdev->use_lock); - return -ENXIO; + err = -ENXIO; + goto unlock; } /* already opened? */ if ((mdev->opened & perm) == perm) { - snd_use_lock_free(&mdev->use_lock); - return 0; + err = 0; + goto unlock; } perm &= ~mdev->opened; @@ -381,13 +385,17 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) } if (! mdev->opened) { - snd_use_lock_free(&mdev->use_lock); - return -ENXIO; + err = -ENXIO; + goto unlock; } mdev->devinfo = dp; + err = 0; + + unlock: + mutex_unlock(&mdev->open_mutex); snd_use_lock_free(&mdev->use_lock); - return 0; + return err; } /* @@ -401,10 +409,9 @@ snd_seq_oss_midi_close(struct seq_oss_devinfo *dp, int dev) if ((mdev = get_mididev(dp, dev)) == NULL) return -ENODEV; - if (! mdev->opened || mdev->devinfo != dp) { - snd_use_lock_free(&mdev->use_lock); - return 0; - } + mutex_lock(&mdev->open_mutex); + if (!mdev->opened || mdev->devinfo != dp) + goto unlock; memset(&subs, 0, sizeof(subs)); if (mdev->opened & PERM_WRITE) { @@ -423,6 +430,8 @@ snd_seq_oss_midi_close(struct seq_oss_devinfo *dp, int dev) mdev->opened = 0; mdev->devinfo = NULL; + unlock: + mutex_unlock(&mdev->open_mutex); snd_use_lock_free(&mdev->use_lock); return 0; } -- 2.39.2

1 0

[PATCH AUTOSEL 4.19 5/5] ALSA: seq: oss: Fix racy open/close of MIDI devices
by Sasha Levin 27 Jun '23

27 Jun '23

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 297224fc0922e7385573a30c29ffdabb67f27b7d ] Although snd_seq_oss_midi_open() and snd_seq_oss_midi_close() can be called concurrently from different code paths, we have no proper data protection against races. Introduce open_mutex to each seq_oss_midi object for avoiding the races. Reported-by: "Gong, Sishuai" <sishuai(a)purdue.edu> Closes: https://lore.kernel.org/r/7DC9AF71-F481-4ABA-955F-76C535661E33@purdue.edu Link: https://lore.kernel.org/r/20230612125533.27461-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/seq/oss/seq_oss_midi.c | 35 +++++++++++++++++++------------ 1 file changed, 22 insertions(+), 13 deletions(-) diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c index 838c3c8b403cb..2ddfd6fed122e 100644 --- a/sound/core/seq/oss/seq_oss_midi.c +++ b/sound/core/seq/oss/seq_oss_midi.c @@ -50,6 +50,7 @@ struct seq_oss_midi { struct snd_midi_event *coder; /* MIDI event coder */ struct seq_oss_devinfo *devinfo; /* assigned OSSseq device */ snd_use_lock_t use_lock; + struct mutex open_mutex; }; @@ -184,6 +185,7 @@ snd_seq_oss_midi_check_new_port(struct snd_seq_port_info *pinfo) mdev->flags = pinfo->capability; mdev->opened = 0; snd_use_lock_init(&mdev->use_lock); + mutex_init(&mdev->open_mutex); /* copy and truncate the name of synth device */ strlcpy(mdev->name, pinfo->name, sizeof(mdev->name)); @@ -332,14 +334,16 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) int perm; struct seq_oss_midi *mdev; struct snd_seq_port_subscribe subs; + int err; if ((mdev = get_mididev(dp, dev)) == NULL) return -ENODEV; + mutex_lock(&mdev->open_mutex); /* already used? */ if (mdev->opened && mdev->devinfo != dp) { - snd_use_lock_free(&mdev->use_lock); - return -EBUSY; + err = -EBUSY; + goto unlock; } perm = 0; @@ -349,14 +353,14 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) perm |= PERM_READ; perm &= mdev->flags; if (perm == 0) { - snd_use_lock_free(&mdev->use_lock); - return -ENXIO; + err = -ENXIO; + goto unlock; } /* already opened? */ if ((mdev->opened & perm) == perm) { - snd_use_lock_free(&mdev->use_lock); - return 0; + err = 0; + goto unlock; } perm &= ~mdev->opened; @@ -381,13 +385,17 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) } if (! mdev->opened) { - snd_use_lock_free(&mdev->use_lock); - return -ENXIO; + err = -ENXIO; + goto unlock; } mdev->devinfo = dp; + err = 0; + + unlock: + mutex_unlock(&mdev->open_mutex); snd_use_lock_free(&mdev->use_lock); - return 0; + return err; } /* @@ -401,10 +409,9 @@ snd_seq_oss_midi_close(struct seq_oss_devinfo *dp, int dev) if ((mdev = get_mididev(dp, dev)) == NULL) return -ENODEV; - if (! mdev->opened || mdev->devinfo != dp) { - snd_use_lock_free(&mdev->use_lock); - return 0; - } + mutex_lock(&mdev->open_mutex); + if (!mdev->opened || mdev->devinfo != dp) + goto unlock; memset(&subs, 0, sizeof(subs)); if (mdev->opened & PERM_WRITE) { @@ -423,6 +430,8 @@ snd_seq_oss_midi_close(struct seq_oss_devinfo *dp, int dev) mdev->opened = 0; mdev->devinfo = NULL; + unlock: + mutex_unlock(&mdev->open_mutex); snd_use_lock_free(&mdev->use_lock); return 0; } -- 2.39.2

1 0

[PATCH AUTOSEL 5.4 6/6] ALSA: seq: oss: Fix racy open/close of MIDI devices
by Sasha Levin 27 Jun '23

27 Jun '23

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 297224fc0922e7385573a30c29ffdabb67f27b7d ] Although snd_seq_oss_midi_open() and snd_seq_oss_midi_close() can be called concurrently from different code paths, we have no proper data protection against races. Introduce open_mutex to each seq_oss_midi object for avoiding the races. Reported-by: "Gong, Sishuai" <sishuai(a)purdue.edu> Closes: https://lore.kernel.org/r/7DC9AF71-F481-4ABA-955F-76C535661E33@purdue.edu Link: https://lore.kernel.org/r/20230612125533.27461-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/seq/oss/seq_oss_midi.c | 35 +++++++++++++++++++------------ 1 file changed, 22 insertions(+), 13 deletions(-) diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c index f73ee0798aeab..be80ce72e0c72 100644 --- a/sound/core/seq/oss/seq_oss_midi.c +++ b/sound/core/seq/oss/seq_oss_midi.c @@ -37,6 +37,7 @@ struct seq_oss_midi { struct snd_midi_event *coder; /* MIDI event coder */ struct seq_oss_devinfo *devinfo; /* assigned OSSseq device */ snd_use_lock_t use_lock; + struct mutex open_mutex; }; @@ -171,6 +172,7 @@ snd_seq_oss_midi_check_new_port(struct snd_seq_port_info *pinfo) mdev->flags = pinfo->capability; mdev->opened = 0; snd_use_lock_init(&mdev->use_lock); + mutex_init(&mdev->open_mutex); /* copy and truncate the name of synth device */ strlcpy(mdev->name, pinfo->name, sizeof(mdev->name)); @@ -319,14 +321,16 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) int perm; struct seq_oss_midi *mdev; struct snd_seq_port_subscribe subs; + int err; if ((mdev = get_mididev(dp, dev)) == NULL) return -ENODEV; + mutex_lock(&mdev->open_mutex); /* already used? */ if (mdev->opened && mdev->devinfo != dp) { - snd_use_lock_free(&mdev->use_lock); - return -EBUSY; + err = -EBUSY; + goto unlock; } perm = 0; @@ -336,14 +340,14 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) perm |= PERM_READ; perm &= mdev->flags; if (perm == 0) { - snd_use_lock_free(&mdev->use_lock); - return -ENXIO; + err = -ENXIO; + goto unlock; } /* already opened? */ if ((mdev->opened & perm) == perm) { - snd_use_lock_free(&mdev->use_lock); - return 0; + err = 0; + goto unlock; } perm &= ~mdev->opened; @@ -368,13 +372,17 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) } if (! mdev->opened) { - snd_use_lock_free(&mdev->use_lock); - return -ENXIO; + err = -ENXIO; + goto unlock; } mdev->devinfo = dp; + err = 0; + + unlock: + mutex_unlock(&mdev->open_mutex); snd_use_lock_free(&mdev->use_lock); - return 0; + return err; } /* @@ -388,10 +396,9 @@ snd_seq_oss_midi_close(struct seq_oss_devinfo *dp, int dev) if ((mdev = get_mididev(dp, dev)) == NULL) return -ENODEV; - if (! mdev->opened || mdev->devinfo != dp) { - snd_use_lock_free(&mdev->use_lock); - return 0; - } + mutex_lock(&mdev->open_mutex); + if (!mdev->opened || mdev->devinfo != dp) + goto unlock; memset(&subs, 0, sizeof(subs)); if (mdev->opened & PERM_WRITE) { @@ -410,6 +417,8 @@ snd_seq_oss_midi_close(struct seq_oss_devinfo *dp, int dev) mdev->opened = 0; mdev->devinfo = NULL; + unlock: + mutex_unlock(&mdev->open_mutex); snd_use_lock_free(&mdev->use_lock); return 0; } -- 2.39.2

1 0

[PATCH AUTOSEL 5.10 7/7] ALSA: seq: oss: Fix racy open/close of MIDI devices
by Sasha Levin 27 Jun '23

27 Jun '23

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 297224fc0922e7385573a30c29ffdabb67f27b7d ] Although snd_seq_oss_midi_open() and snd_seq_oss_midi_close() can be called concurrently from different code paths, we have no proper data protection against races. Introduce open_mutex to each seq_oss_midi object for avoiding the races. Reported-by: "Gong, Sishuai" <sishuai(a)purdue.edu> Closes: https://lore.kernel.org/r/7DC9AF71-F481-4ABA-955F-76C535661E33@purdue.edu Link: https://lore.kernel.org/r/20230612125533.27461-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/seq/oss/seq_oss_midi.c | 35 +++++++++++++++++++------------ 1 file changed, 22 insertions(+), 13 deletions(-) diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c index f73ee0798aeab..be80ce72e0c72 100644 --- a/sound/core/seq/oss/seq_oss_midi.c +++ b/sound/core/seq/oss/seq_oss_midi.c @@ -37,6 +37,7 @@ struct seq_oss_midi { struct snd_midi_event *coder; /* MIDI event coder */ struct seq_oss_devinfo *devinfo; /* assigned OSSseq device */ snd_use_lock_t use_lock; + struct mutex open_mutex; }; @@ -171,6 +172,7 @@ snd_seq_oss_midi_check_new_port(struct snd_seq_port_info *pinfo) mdev->flags = pinfo->capability; mdev->opened = 0; snd_use_lock_init(&mdev->use_lock); + mutex_init(&mdev->open_mutex); /* copy and truncate the name of synth device */ strlcpy(mdev->name, pinfo->name, sizeof(mdev->name)); @@ -319,14 +321,16 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) int perm; struct seq_oss_midi *mdev; struct snd_seq_port_subscribe subs; + int err; if ((mdev = get_mididev(dp, dev)) == NULL) return -ENODEV; + mutex_lock(&mdev->open_mutex); /* already used? */ if (mdev->opened && mdev->devinfo != dp) { - snd_use_lock_free(&mdev->use_lock); - return -EBUSY; + err = -EBUSY; + goto unlock; } perm = 0; @@ -336,14 +340,14 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) perm |= PERM_READ; perm &= mdev->flags; if (perm == 0) { - snd_use_lock_free(&mdev->use_lock); - return -ENXIO; + err = -ENXIO; + goto unlock; } /* already opened? */ if ((mdev->opened & perm) == perm) { - snd_use_lock_free(&mdev->use_lock); - return 0; + err = 0; + goto unlock; } perm &= ~mdev->opened; @@ -368,13 +372,17 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) } if (! mdev->opened) { - snd_use_lock_free(&mdev->use_lock); - return -ENXIO; + err = -ENXIO; + goto unlock; } mdev->devinfo = dp; + err = 0; + + unlock: + mutex_unlock(&mdev->open_mutex); snd_use_lock_free(&mdev->use_lock); - return 0; + return err; } /* @@ -388,10 +396,9 @@ snd_seq_oss_midi_close(struct seq_oss_devinfo *dp, int dev) if ((mdev = get_mididev(dp, dev)) == NULL) return -ENODEV; - if (! mdev->opened || mdev->devinfo != dp) { - snd_use_lock_free(&mdev->use_lock); - return 0; - } + mutex_lock(&mdev->open_mutex); + if (!mdev->opened || mdev->devinfo != dp) + goto unlock; memset(&subs, 0, sizeof(subs)); if (mdev->opened & PERM_WRITE) { @@ -410,6 +417,8 @@ snd_seq_oss_midi_close(struct seq_oss_devinfo *dp, int dev) mdev->opened = 0; mdev->devinfo = NULL; + unlock: + mutex_unlock(&mdev->open_mutex); snd_use_lock_free(&mdev->use_lock); return 0; } -- 2.39.2

1 0

[PATCH AUTOSEL 5.15 9/9] ALSA: seq: oss: Fix racy open/close of MIDI devices
by Sasha Levin 27 Jun '23

27 Jun '23

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 297224fc0922e7385573a30c29ffdabb67f27b7d ] Although snd_seq_oss_midi_open() and snd_seq_oss_midi_close() can be called concurrently from different code paths, we have no proper data protection against races. Introduce open_mutex to each seq_oss_midi object for avoiding the races. Reported-by: "Gong, Sishuai" <sishuai(a)purdue.edu> Closes: https://lore.kernel.org/r/7DC9AF71-F481-4ABA-955F-76C535661E33@purdue.edu Link: https://lore.kernel.org/r/20230612125533.27461-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/seq/oss/seq_oss_midi.c | 35 +++++++++++++++++++------------ 1 file changed, 22 insertions(+), 13 deletions(-) diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c index 07efb38f58ac1..f2940b29595f0 100644 --- a/sound/core/seq/oss/seq_oss_midi.c +++ b/sound/core/seq/oss/seq_oss_midi.c @@ -37,6 +37,7 @@ struct seq_oss_midi { struct snd_midi_event *coder; /* MIDI event coder */ struct seq_oss_devinfo *devinfo; /* assigned OSSseq device */ snd_use_lock_t use_lock; + struct mutex open_mutex; }; @@ -172,6 +173,7 @@ snd_seq_oss_midi_check_new_port(struct snd_seq_port_info *pinfo) mdev->flags = pinfo->capability; mdev->opened = 0; snd_use_lock_init(&mdev->use_lock); + mutex_init(&mdev->open_mutex); /* copy and truncate the name of synth device */ strscpy(mdev->name, pinfo->name, sizeof(mdev->name)); @@ -322,15 +324,17 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) int perm; struct seq_oss_midi *mdev; struct snd_seq_port_subscribe subs; + int err; mdev = get_mididev(dp, dev); if (!mdev) return -ENODEV; + mutex_lock(&mdev->open_mutex); /* already used? */ if (mdev->opened && mdev->devinfo != dp) { - snd_use_lock_free(&mdev->use_lock); - return -EBUSY; + err = -EBUSY; + goto unlock; } perm = 0; @@ -340,14 +344,14 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) perm |= PERM_READ; perm &= mdev->flags; if (perm == 0) { - snd_use_lock_free(&mdev->use_lock); - return -ENXIO; + err = -ENXIO; + goto unlock; } /* already opened? */ if ((mdev->opened & perm) == perm) { - snd_use_lock_free(&mdev->use_lock); - return 0; + err = 0; + goto unlock; } perm &= ~mdev->opened; @@ -372,13 +376,17 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) } if (! mdev->opened) { - snd_use_lock_free(&mdev->use_lock); - return -ENXIO; + err = -ENXIO; + goto unlock; } mdev->devinfo = dp; + err = 0; + + unlock: + mutex_unlock(&mdev->open_mutex); snd_use_lock_free(&mdev->use_lock); - return 0; + return err; } /* @@ -393,10 +401,9 @@ snd_seq_oss_midi_close(struct seq_oss_devinfo *dp, int dev) mdev = get_mididev(dp, dev); if (!mdev) return -ENODEV; - if (! mdev->opened || mdev->devinfo != dp) { - snd_use_lock_free(&mdev->use_lock); - return 0; - } + mutex_lock(&mdev->open_mutex); + if (!mdev->opened || mdev->devinfo != dp) + goto unlock; memset(&subs, 0, sizeof(subs)); if (mdev->opened & PERM_WRITE) { @@ -415,6 +422,8 @@ snd_seq_oss_midi_close(struct seq_oss_devinfo *dp, int dev) mdev->opened = 0; mdev->devinfo = NULL; + unlock: + mutex_unlock(&mdev->open_mutex); snd_use_lock_free(&mdev->use_lock); return 0; } -- 2.39.2

1 0

[PATCH AUTOSEL 6.1 15/15] ALSA: seq: oss: Fix racy open/close of MIDI devices
by Sasha Levin 27 Jun '23

27 Jun '23

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 297224fc0922e7385573a30c29ffdabb67f27b7d ] Although snd_seq_oss_midi_open() and snd_seq_oss_midi_close() can be called concurrently from different code paths, we have no proper data protection against races. Introduce open_mutex to each seq_oss_midi object for avoiding the races. Reported-by: "Gong, Sishuai" <sishuai(a)purdue.edu> Closes: https://lore.kernel.org/r/7DC9AF71-F481-4ABA-955F-76C535661E33@purdue.edu Link: https://lore.kernel.org/r/20230612125533.27461-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/seq/oss/seq_oss_midi.c | 35 +++++++++++++++++++------------ 1 file changed, 22 insertions(+), 13 deletions(-) diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c index 07efb38f58ac1..f2940b29595f0 100644 --- a/sound/core/seq/oss/seq_oss_midi.c +++ b/sound/core/seq/oss/seq_oss_midi.c @@ -37,6 +37,7 @@ struct seq_oss_midi { struct snd_midi_event *coder; /* MIDI event coder */ struct seq_oss_devinfo *devinfo; /* assigned OSSseq device */ snd_use_lock_t use_lock; + struct mutex open_mutex; }; @@ -172,6 +173,7 @@ snd_seq_oss_midi_check_new_port(struct snd_seq_port_info *pinfo) mdev->flags = pinfo->capability; mdev->opened = 0; snd_use_lock_init(&mdev->use_lock); + mutex_init(&mdev->open_mutex); /* copy and truncate the name of synth device */ strscpy(mdev->name, pinfo->name, sizeof(mdev->name)); @@ -322,15 +324,17 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) int perm; struct seq_oss_midi *mdev; struct snd_seq_port_subscribe subs; + int err; mdev = get_mididev(dp, dev); if (!mdev) return -ENODEV; + mutex_lock(&mdev->open_mutex); /* already used? */ if (mdev->opened && mdev->devinfo != dp) { - snd_use_lock_free(&mdev->use_lock); - return -EBUSY; + err = -EBUSY; + goto unlock; } perm = 0; @@ -340,14 +344,14 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) perm |= PERM_READ; perm &= mdev->flags; if (perm == 0) { - snd_use_lock_free(&mdev->use_lock); - return -ENXIO; + err = -ENXIO; + goto unlock; } /* already opened? */ if ((mdev->opened & perm) == perm) { - snd_use_lock_free(&mdev->use_lock); - return 0; + err = 0; + goto unlock; } perm &= ~mdev->opened; @@ -372,13 +376,17 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) } if (! mdev->opened) { - snd_use_lock_free(&mdev->use_lock); - return -ENXIO; + err = -ENXIO; + goto unlock; } mdev->devinfo = dp; + err = 0; + + unlock: + mutex_unlock(&mdev->open_mutex); snd_use_lock_free(&mdev->use_lock); - return 0; + return err; } /* @@ -393,10 +401,9 @@ snd_seq_oss_midi_close(struct seq_oss_devinfo *dp, int dev) mdev = get_mididev(dp, dev); if (!mdev) return -ENODEV; - if (! mdev->opened || mdev->devinfo != dp) { - snd_use_lock_free(&mdev->use_lock); - return 0; - } + mutex_lock(&mdev->open_mutex); + if (!mdev->opened || mdev->devinfo != dp) + goto unlock; memset(&subs, 0, sizeof(subs)); if (mdev->opened & PERM_WRITE) { @@ -415,6 +422,8 @@ snd_seq_oss_midi_close(struct seq_oss_devinfo *dp, int dev) mdev->opened = 0; mdev->devinfo = NULL; + unlock: + mutex_unlock(&mdev->open_mutex); snd_use_lock_free(&mdev->use_lock); return 0; } -- 2.39.2

1 0

[PATCH AUTOSEL 6.3 16/16] ALSA: seq: oss: Fix racy open/close of MIDI devices
by Sasha Levin 27 Jun '23

27 Jun '23

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 297224fc0922e7385573a30c29ffdabb67f27b7d ] Although snd_seq_oss_midi_open() and snd_seq_oss_midi_close() can be called concurrently from different code paths, we have no proper data protection against races. Introduce open_mutex to each seq_oss_midi object for avoiding the races. Reported-by: "Gong, Sishuai" <sishuai(a)purdue.edu> Closes: https://lore.kernel.org/r/7DC9AF71-F481-4ABA-955F-76C535661E33@purdue.edu Link: https://lore.kernel.org/r/20230612125533.27461-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/seq/oss/seq_oss_midi.c | 35 +++++++++++++++++++------------ 1 file changed, 22 insertions(+), 13 deletions(-) diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c index 07efb38f58ac1..f2940b29595f0 100644 --- a/sound/core/seq/oss/seq_oss_midi.c +++ b/sound/core/seq/oss/seq_oss_midi.c @@ -37,6 +37,7 @@ struct seq_oss_midi { struct snd_midi_event *coder; /* MIDI event coder */ struct seq_oss_devinfo *devinfo; /* assigned OSSseq device */ snd_use_lock_t use_lock; + struct mutex open_mutex; }; @@ -172,6 +173,7 @@ snd_seq_oss_midi_check_new_port(struct snd_seq_port_info *pinfo) mdev->flags = pinfo->capability; mdev->opened = 0; snd_use_lock_init(&mdev->use_lock); + mutex_init(&mdev->open_mutex); /* copy and truncate the name of synth device */ strscpy(mdev->name, pinfo->name, sizeof(mdev->name)); @@ -322,15 +324,17 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) int perm; struct seq_oss_midi *mdev; struct snd_seq_port_subscribe subs; + int err; mdev = get_mididev(dp, dev); if (!mdev) return -ENODEV; + mutex_lock(&mdev->open_mutex); /* already used? */ if (mdev->opened && mdev->devinfo != dp) { - snd_use_lock_free(&mdev->use_lock); - return -EBUSY; + err = -EBUSY; + goto unlock; } perm = 0; @@ -340,14 +344,14 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) perm |= PERM_READ; perm &= mdev->flags; if (perm == 0) { - snd_use_lock_free(&mdev->use_lock); - return -ENXIO; + err = -ENXIO; + goto unlock; } /* already opened? */ if ((mdev->opened & perm) == perm) { - snd_use_lock_free(&mdev->use_lock); - return 0; + err = 0; + goto unlock; } perm &= ~mdev->opened; @@ -372,13 +376,17 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) } if (! mdev->opened) { - snd_use_lock_free(&mdev->use_lock); - return -ENXIO; + err = -ENXIO; + goto unlock; } mdev->devinfo = dp; + err = 0; + + unlock: + mutex_unlock(&mdev->open_mutex); snd_use_lock_free(&mdev->use_lock); - return 0; + return err; } /* @@ -393,10 +401,9 @@ snd_seq_oss_midi_close(struct seq_oss_devinfo *dp, int dev) mdev = get_mididev(dp, dev); if (!mdev) return -ENODEV; - if (! mdev->opened || mdev->devinfo != dp) { - snd_use_lock_free(&mdev->use_lock); - return 0; - } + mutex_lock(&mdev->open_mutex); + if (!mdev->opened || mdev->devinfo != dp) + goto unlock; memset(&subs, 0, sizeof(subs)); if (mdev->opened & PERM_WRITE) { @@ -415,6 +422,8 @@ snd_seq_oss_midi_close(struct seq_oss_devinfo *dp, int dev) mdev->opened = 0; mdev->devinfo = NULL; + unlock: + mutex_unlock(&mdev->open_mutex); snd_use_lock_free(&mdev->use_lock); return 0; } -- 2.39.2

1 0

Focusrite Scarlett 2i2 3rd Gen suffers from high output latency with the 1.2.9-1 update
by GitHub issues - edited 26 Jun '23

26 Jun '23

alsa-project/alsa-ucm-conf issue #329 was edited from IsaacLambat: Hello, I have recently upgraded from `alsa-ucm-conf 1.2.8-1` to `1.2.9-1` and this has caused some latency of around 1/2 seconds for outputting audio via the `Focusrite Scarlett 2i2 3rd Gen` which is connected to the computer via USB. No latency is introduced when plugging wired headphones directly in to the motherboard so issue seems to be isolated to the device. I noticed the change logs indicate better support for this device was added in this version (which is great!) so these changes may have affected the existing settings of the headphones. I have downgraded only `alsa-ucm-conf` to `1.2.8-1` and the output has no latency so I am using this as a workaround. I use PulseAudio with ALSA on Arch Linux. Versions: ``` alsa-card-profiles - 0.3.71 alsa-lib - 1.2.9 alsa-ucm-conf - 1.2.9 alsa-utils - 1.2.9 pulseaudio - 16.1 Arch Linux - 6.3.9-arch1-1 ``` dmesg detects the product as `Scarlett 2i2 USB` Please let me know if any more information is needed (such as logs or config files) and if there is anything I should try. Many thanks :D Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/329 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[GIT PULL] ASoC updates for v6.5
by Mark Brown 26 Jun '23

26 Jun '23

The following changes since commit 1f1ef7e5bbe2de125da413e238915f6047ea4ba2: ASoC: intel: sof_sdw: Fixup typo in device link checking (2023-06-19 12:58:22 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v6.5 for you to fetch changes up to 2d0cad0473bd1ffbc5842be0b9f2546265acb011: ASoC: core: Always store of_node when getting DAI link component (2023-06-23 22:09:00 +0100) ---------------------------------------------------------------- ASoC: Updates for v6.5 A fairly quiet release from a core and framework point of view, but a very big one from the point of view of new drivers: - More refectoring from Morimoto-san, this time mainly around DAI links and how we control the ordering of trigger() callbacks. - Convert a lot of drivers to use maple tree based caches. - Lots of work on the x86 driver stack. - Compressed audio support for Qualcomm. - Support for AMD SoundWire, Analog Devices SSM3515, Google Chameleon, Ingenic X1000, Intel systems with various CODECs, Longsoon platforms, Maxim MAX98388, Mediatek MT8188, Nuvoton NAU8825C, NXP platforms with NAU8822, Qualcomm WSA884x, StarFive JH7110, Texas Instruments TAS2781. ---------------------------------------------------------------- Aidan MacDonald (2): ASoC: jz4740-i2s: Add support for X1000 SoC ASoC: ingenic: Add compatible string for X1000 SoC Alexander Stein (1): ASoC: dt-bindings: tlv320aic32x4: convert to DT schema format Amadeusz Sławiński (5): ASoC: topology: Log control load errors in soc_tplg_control_load() ASoC: topology: Remove redundant logs ASoC: topology: Do not split message string on multiple lines ASoC: topology: Remove redundant log ASoC: topology: Remove redundant log AngeloGioacchino Del Regno (10): ASoC: SOF: mediatek: mt8195: Use snd_sof_ipc_process_reply() helper ASoC: SOF: mediatek: mt8186: Use snd_sof_ipc_process_reply() helper ASoC: mediatek: mt8195-afe-pcm: Clean up unnecessary functions ASoC: mediatek: mt8195-afe-pcm: Simplify runtime PM during probe ASoC: mediatek: mt8195-afe-pcm: Simplify with dev_err_probe() ASoC: mediatek: mt8192-mt6359: Remove " Jack" from Headphone pin name ASoC: mediatek: mt8188-mt6359: Compress of_device_id entries ASoC: mediatek: mt8188-mt6359: Cleanup return 0 disguised as return ret ASoC: mediatek: mt8188-mt6359: Clean up log levels ASoC: mediatek: mt8188-mt6359: Use bitfield macros for registers Anup Sharma (2): ASoC: dt-bindings: nau8540: Convert to dtschema ASoC: dt-bindings: rt1016: Convert to dtschema Arnd Bergmann (9): ASoC: amd: vangogh: select CONFIG_SND_AMD_ACP_CONFIG ASoC: amd: vangogh: select CONFIG_SND_AMD_ACP_CONFIG ASoC: mediatek: mt8188-mt6359: add i2c dependency ASoC: max98388: fix unused function warnings ASoC: loongson: fix unused PM function warning ASoC: loongson: add PCI dependency ASoC: loongson: fix compile testing on 32-bit ASoC: loongson: fix address space confusion ASoC: tas2781: fix Kconfig dependencies Arun Gopal Kondaveeti (1): ASoC: amd: update pm_runtime enable sequence Balamurugan C (3): ASoC: Intel: ADL: Enable HDMI-In capture feature support for non-I2S codec boards. ASoC: Intel: ADL: Moving amp only boards into end of the table. ASoC: Intel: Sof_ssp_amp: Correcting author name. Bard Liao (21): ASoC: Intel: sof_sdw: remove SOF_SDW_TGL_HDMI for MeteorLake devices ASoC: Intel: sof_sdw: add quirk for MTL RVP ASoC: Intel: sof_sdw: start set codec init function with an adr index ASoC: Intel: sof_sdw: add missing exit callback ASoC: Intel: sof_sdw: add dai info ASoC: Intel: sof_sdw: use predefine dailink id ASoC: Intel: sof_sdw: add codec_info pointer ASoC: Intel: sdw_sof: append dai_type and remove codec_type ASoC: Intel: sof_sdw: add multi dailink support for a codec ASoC: Intel: sof_sdw_rt_sdca_jack_common: test SOF_JACK_JDSRC in _exit ASoC: Intel: sof_sdw: rename SOF_RT711_JDSRC to SOF_JACK_JDSRC ASoC: Intel: sof_sdw: make rt711_sdca be generic ASoC: Intel: sof_sdw: add rt712 support ASoC: Intel: soc-acpi-intel-tgl-match: add rt712 ID ASoC: Intel: soc-acpi-intel-mtl-match: add rt712 ID ASoC: Intel: sof_sdw: add rt713 support ASoC: Intel: sof_sdw: increase sdw pin index for each sdw link ASoC: add N cpus to M codecs dai link support ASoC: Intel: sof_sdw: add dai_link_codec_ch_map ASoC: SOF: sof-audio: add is_virtual_widget helper ASoC: SOF: sof-audio: test virtual widget in sof_walk_widgets_in_order Brent Lu (12): ASoC: Intel: avs-da7219: remove redundant dapm routes ASoC: Intel: avs-dmic: remove redundant dapm routes ASoC: Intel: avs-hdaudio: remove redundant dapm routes ASoC: Intel: avs-max98357a: remove redundant dapm routes ASoC: Intel: avs-max98373: remove redundant dapm routes ASoC: Intel: avs-max98927: remove redundant dapm routes ASoC: Intel: avs-nau8825: remove redundant dapm routes ASoC: Intel: avs-rt274: remove redundant dapm routes ASoC: Intel: avs-rt286: remove redundant dapm routes ASoC: Intel: avs-rt298: remove redundant dapm routes ASoC: Intel: avs-rt5682: remove redundant dapm routes ASoC: Intel: avs-ssm4567: remove redundant dapm routes Carlos Bilbao (1): ASoC: amd: ps: Update copyright notice Chao Song (1): ASoC: Intel: soc-acpi: add support for MTL SDCA boards Charles Keepax (4): ASoC: soc-component: Add notify control helper function ASoC: ak4118: Update to use new component control notify helper ASoC: wm_adsp: Update to use new component control notify helepr ASoC: cs35l45: Relicense to GPL only Christophe JAILLET (8): ASoC: adau1761: Use the devm_clk_get_optional() helper ASoC: atmel: sam9g20_wm8731: Remove the unneeded include <linux/i2c.h> ASoC: cs42l51: Use the devm_clk_get_optional() helper ASoC: rt5659: Use the devm_clk_get_optional() helper ASoC: stm32: sai: Use the devm_clk_get_optional() helper ASoC: cs53l30: Use the devm_clk_get_optional() helper ASoC: rt5682s: Use the devm_clk_get_optional() helper ASoC: tegra: Simplify code around clk_get_rate() handling Claudiu Beznea (3): ASoC: do not include pm_runtime.h if not used ASoC: use pm.h instead of runtime_pm.h ASoC: imx-audmix: check return value of devm_kasprintf() Colin Ian King (1): ASoC: tas2781: Fix spelling mistake "calibraiton" -> "calibration" Cristian Ciocaltea (2): ASoC: es8316: Increment max value for ALC Capture Target Volume control ASoC: es8316: Do not set rate constraints for unsupported MCLKs Curtis Malainey (2): ASoC: SOF: Refactor rx function for fuzzing ASoC: SOF: Add IPC3 Kernel Injector Dan Carpenter (4): ASoC: mediatek: mt8188-mt6359: clean up a return in codec_init ASoC: loongson: fix error codes in loongson_card_parse_acpi() ASoC: max98388: fix error code in probe() ASoC: tas2781: Fix error code in tas2781_load_calibration() David Lin (9): ASoC: dt-bindings: nau8825: Convert to dtschema ASoC: dt-bindings: nau8824: Convert to dtschema ASoC: dt-bindings: nau8315: Convert to dtschema ASoC: dt-bindings: nau8810: Convert to dtschema ASoC: nau8825: Add pre-charge actions for input ASoC: nau8825: Add the management of headset detection for power saving ASoC: nau8825: Add registers patch for NAU8825C ASoC: nau8825: Update the calculation of FLL for NAU8825C ASoC: nau8825: Update output control for NAU8825C David Rau (3): ASoC: dt-bindings: dialog,da7219: convert to dtschema ASoC: dt-bindings: da7219: Add jack-ins-det-pty property ASoC: da7219: Add Jack insertion detection polarity Fabio Estevam (1): ASoC: fsl-asoc-card: Allow passing the number of slots in use Harshit Mogalapalli (1): ASoC: SOF: Intel: hda-dai: Fix locking in hda_ipc4_pre_trigger() Hui Wang (2): ASoC: fsl-asoc-card: add nau8822 support ASoC: bindings: fsl-asoc-card: add compatible string for nau8822 codec Jack Yu (1): ASoC: rt722-sdca: Add RT722 SDCA driver Jiapeng Chong (2): ASoC: SOF: Simplify the calculation of variables ASoC: max98088: clean up some inconsistent indenting Juerg Haefliger (3): ASoC: siu: Add MODULE_FIRMWARE macro ASoC: rt5677: Add MODULE_FIRMWARE macro ASoC: codecs: wm0010: Add MODULE_FIRMWARE macros Krzysztof Kozlowski (17): ASoC: dt-bindings: google,sc7180-trogdor: allow up to four codec DAIs ASoC: codecs: rt1308: do not store status in state container ASoC: codecs: rt1316: do not store status in state container ASoC: codecs: rt1318: do not store status in state container ASoC: codecs: rt5682: do not store status in state container ASoC: codecs: rt700: do not store status in state container ASoC: codecs: rt711-sdca: do not store status in state container ASoC: codecs: rt711: do not store status in state container ASoC: codecs: rt712-sdca-dmic: do not store status in state container ASoC: codecs: rt712-sdca: do not store status in state container ASoC: codecs: rt715-sdca: do not store status in state container ASoC: codecs: rt715: do not store status in state container ASoC: codecs: rt722-sdca: do not store status in state container ASoC: codecs: wsa883x: use existing define instead of raw value ASoC: dt-bindings: cirrus,cs35l45: drop unneeded quotes ASoC: dt-bindings: qcom,wsa8840: Add WSA884x family of speakers ASoC: codecs: wsa884x: Add WSA884x family of speakers Kuninori Morimoto (38): ASoC: soc-utils.c: add asoc_dummy_dlc ASoC: ti: use asoc_dummy_dlc ASoC: sof: use asoc_dummy_dlc ASoC: amd: use asoc_dummy_dlc ASoC: fsl: use asoc_dummy_dlc ASoC: qcom: use asoc_dummy_dlc ASoC: atmel: use asoc_dummy_dlc ASoC: meson: use asoc_dummy_dlc ASoC: intel: avs: use asoc_dummy_dlc ASoC: intel: sof: use asoc_dummy_dlc ASoC: intel: skylake: use asoc_dummy_dlc ASoC: simple_card_utils.c: use asoc_dummy_dlc ASoC: soc-topology.c: add comment for Platform/Codec ASoC: soc-pcm.c: indicate error if stream has no playback no capture ASoC: soc-pcm.c: use dai_link on soc_get_playback_capture() ASoC: soc-pcm.c: cleanup soc_get_playback_capture() error ASoC: soc-pcm.c: use temporary variable at soc_get_playback_capture() ASoC: soc-pcm.c: tidyup playback/capture_only at soc_get_playback_capture() ASoC: soc.h: remove snd_soc_compr_ops :: trigger ASoC: add snd_soc_get_stream_cpu() ASoC: audio-graph-card2-custom-sample: add missing CPU:Codec = 1:N sample ASoC: audio-graph-card2-custom-sample.dtsi: remove DT warning ASoC: add new trigger ordering method ASoC: amd: use use new trigger ordering method ASoC: atmel: use use new trigger ordering method ASoC: starfive: use use new trigger ordering method ASoC: remove old trigger ordering method ASoC: simple-card-utils.c: share asoc_graph_parse_dai() ASoC: simple_card_utils: remove unused cpus/codecs/platforms from props ASoC: soc-core.c: add snd_soc_{of_}get_dlc() ASoC: soc-core.c: add index on snd_soc_of_get_dai_name() ASoC: fsl: use snd_soc_{of_}get_dlc() ASoC: qcom: use snd_soc_{of_}get_dlc() ASoC: meson: use snd_soc_{of_}get_dlc() ASoC: samsung: use snd_soc_{of_}get_dlc() ASoC: loongson: use snd_soc_{of_}get_dlc() ASoC: soc-core.c: use snd_soc_{of_}get_dlc() ASoC: simple-card.c: use snd_soc_{of_}get_dlc() Mac Chiang (1): ASoC: Intel: sof_rt5682: Add mtl support RT1019P speaker Marek Vasut (2): ASoC: dt-bindings: ssm2518: Convert to dtschema ASoC: dt-bindings: audio-graph-card: Expand 'widgets' documentation Mark Brown (89): ASoC: rt715: Use maple tree register cache ASoC: rt5682: Use a maple tree based register cache ASoC: MediaTek MT8195/86 Cleanups ASoC: add and use asoc_dummy_dlc ASoC: qcom: add display port module support. ASoC SSM3515 codec driver ASoC: Merge up fixes for CI ASoC: Intel: machine driver updates ASoC: SOF: Intel: hda-dai: prepare LNL support ASoC: SOF: Intel/ipc4: Do not reset BE DAI pipeline ASoC: Factor out control notification support ASoC: mediatek: mt8188: revise AFE driver ASoC: SOF: ipc4-topology: Improve the audio format ASoC: codecs: do not store status in state containe Add Chameleon v3 ASoC audio Merge tag 'v6.4-rc2' into asoc-6.5 to get fixes for CI ASoC: cs35l56: Bugfixes and efficiency improvement ASoC: SOF: ipc4: Querry CPC value from firmware's ASoC: topology: Clean up error messages handling ASoC: SOF: Intel: mtl: Enable multicore support ASoC: do not include runtime_pm.h if not needed ASoC: mt6359: kselftest fix and driver extension Improve support for sof_ipc{3|4}_bytes_ext_put ASoC: da7219: Add jack insertion detection polarity ASoC: SOF: add mt8188 audio support ASoC: topology: allow for partial dailink stream_name ES8316 audio codec fixes on Rock5B ASoC: minor cleanup for soc_get_playback_capture() Add TDM audio on StarFive JH7110 Add NAU8825C support ASoC: SOF: Intel: LunarLake preparation patches ASoC: mt8188: add new board support ASoC: Intel: machine driver updates for 6.5 ASoC: Intel: HDA: two cleanups ASoC: stm32: fix dtbs_check warnings ASoC: mt8188-mt6359: Cleanups Fix error check and cleanup for JH7110 TDM ASoC: mediatek: mt8173: Fix error paths ASoC: Intel: avs: remove redundant dapm routes ASoC: cs35l32: Use maple tree register cache ASoC: cs35l33: Use maple tree register cache ASoC: cs35l34: Use maple tree register cache ASoC: cs35l35: Use maple tree register cache ASoC: cs4234: Use maple tree register cache ASoC: cs42l42: Use maple tree register cache ASoC: cs42l73: Use maple tree register cache ASoC: cs42l83: Use maple tree register cache ASoC: cs43130: Use maple tree register cache ASoC: cs35l30: Use maple tree register cache ASoC: Use maple tree for Cirrus Logic devices ASoC: add N cpus to M codecs dai link support ASoC: add new trigger ordering method ASoC: rt700: Use maple tree register cache ASoC: rt711: Use maple tree register cache ASoC: rt712: Use maple tree register cache ASoC: rt715: Use maple tree register cache ASoC: rt722: Use maple tree register cache ASoC: rt1308: Use maple tree register cache ASoC: rt1316: Use maple tree register cache ASoC: rt1318: Use maple tree register cache ASoC: Use the maple tree register cache for RealTek ASoC: Merge fixes due to dependencies ASoC: SOF: misc updates for 6.5 ASoC: es8316: Use maple tree register cache ASoC: es8328: Use maple tree register cache ASoC: rt1011: Use maple tree register cache ASoC: rt1019: Use maple tree register cache ASoC: rt1305: Use maple tree register cache ASoC: rt1308: Use maple tree register cache ASoC: rt5514: Use maple tree register cache ASoC: rt5616: Use maple tree register cache ASoC: rt5631: Use maple tree register cache ASoC: rt5640: Use maple tree register cache ASoC: rt5645: Use maple tree register cache ASoC: rt5651: Use maple tree register cache ASoC: rt5660: Use maple tree register cache ASoC: rt5665: Use maple tree register cache ASoC: rt5665: Use maple tree register cache ASoC: rt5668: Use maple tree register cache ASoC: rt5670: Use maple tree register cache ASoC: rt5682: Use maple tree register cache ASoC: max98363: Remove cache defaults for volatile registers ASoC: Convert Realtek I2C drivers to use maple tree ASoC: Use maple tree register cache for Everest Semi ASoC: qcom: audioreach: add compress offload add snd_soc_{of_}get_dlc() ASoC: amd: ps: add SoundWire support ASoC: Merge up fixes for CI ASoC: core: Always store of_node when getting DAI link component Martin Povišer (2): ASoC: dt-bindings: Add adi,ssm3515 amp schema ASoC: ssm3515: Add new amp driver Maxim Kochetkov (6): ASoC: dwc: add optional reset support ASoC: dt-bindings: designware-i2s: add optional resets ASoC: dwc: extend supported formats ASoC: dwc: add DMA handshake control ASoC: dwc: don't assign addr_width for dt configs ASoC: codecs: max98090: Allow dsp_a mode Min-Hua Chen (2): ASoC: tegra: tegra210_adx: fix snd_pcm_format_t type ASoC: ti: davinci-mcasp: Use pcm_for_each_format() macro Mohammad Rafi Shaik (4): ASoC: qcom: SC7280: audioreach: Add sc7280 hardware param fixup callback ASoC: q6dsp: q6apm: add end of stream events ASoC: q6dsp: audioreach: Add support to set compress format params ASoC: q6dsp: audioreach: Add gapless feature support Nícolas F. R. A. Prado (1): ASoC: mediatek: mt8192-mt6359: Go back to old headphone pin name Olivier Moysan (2): ASoC: dt-bindings: stm32: document audio of graph port for i2s ASoC: dt-bindings: document audio of graph port for cs42l51 Paul Olaru (2): ASoC: sof: Improve sof_ipc3_bytes_ext_put function ASoC: sof: Improve sof_ipc4_bytes_ext_put function Paweł Anikiel (4): ASoC: Add Google Chameleon v3 i2s driver ASoC: Add Google Chameleon v3 codec driver ASoC: dt-bindings: Add Google Chameleon v3 audio codec ASoC: dt-bindings: Add Google Chameleon v3 i2s device Peter Ujfalusi (12): ASoC: Intel: sof_sdw: add quirk for LNL RVP ASoC: SOF: ipc4-topology: Use set_get_data() to send LARGE_CONFIG message ASoC: SOF: ipc4-loader: Drop unused bss_size from struct sof_ipc4_fw_module ASoC: SOF: ipc4-loader: Save a pointer to fm_config in sof_ipc4_fw_module ASoC: SOF: ipc4-topology: Rename sof_ipc4_update_pipeline_mem_usage() to be generic ASoC: SOF: ipc4-topology: Do not use the CPC value from topology ASoC: SOF: ipc4-loader/topology: Query the CPC value from manifest ASoC: SOF: core: Free the firmware trace before calling snd_sof_shutdown() ASoC: SOF: Add new sof_debug flag to request message payload dump ASoC: SOF: ipc3: Dump IPC message payload ASoC: SOF: ipc4: Switch to use the sof_debug:bit11 to dump message payload ASoC: SOF: pm: Remove duplicated code in sof_suspend Pierre-Louis Bossart (26): ASoC: Intel: sof_sdw: add new mapping for HP Spectre x360 ASoC: Intel: soc-acpi: add tables for LunarLake ASoC: SOF: Intel: hda-dai: simplify .prepare callback ASoC: SOF: Intel: hda-dai: remove use of cpu_dai->component drvdata ASoC: SOF: Intel: fix DAI number mismatch ASoC: SOF: Intel: clarify initialization when HDA_AUDIO_CODEC is not used ASoC: SOF: Intel: Kconfig: move selection of PROBE_WORK_QUEUE ASoC: SOF: Intel: hda-dai: move hda_dai_prepare() code ASoC: SOF: Intel: hda-dai: mark functions as __maybe_unused ASoC: SOF: Intel: hda-dai: use HDA_LINK instead of HDA_AUDIO_CODEC ASoC: SOF: Intel: remove mutual exclusion between NOCODEC and HDA_LINK ASoC: Intel: soc-acpi: add table for RPL Dell SKU 0BDA ASoC: Intel: sof_sdw: add quick for Dell SKU 0BDA ASoC: Intel: soc-acpi: add tables for Dell SKU 0B34 ASoC: Intel: sof-sdw: add Dell SKU 0B34 ASoC: SOF: Intel: hda-dai: add error checks to prevent static analysis warnings ASoC: SOF: Intel: hda-dai: add codec_dai_set_stream callback ASoC: SOF: Intel: hda-dai: add calc_stream_format callback ASoC: SOF: Intel: hda-dai: add get_hlink callback ASoC: SOF: ipc4-topology: extend ALH-specific data structure ASoC: SOF: ipc4-topology: introduce DMA config TLV ASoC: SOF: ipc4-topology: add DMA config TLV to IPC data ASoC: SOF: Intel: mtl: prepare for code reuse ASoC: SOF: Intel: hda: add helper to extract SoundWire link count ASoC: SOF: Intel: fix SoundWire/HDaudio mutual exclusion ASoC: SOF: Intel: hda-pcm: remove kernel parameter init Rander Wang (3): ASoC: SOF: Intel: mtl: add core_get & put support on MeterLake platforms ASoC: SOF: Intel: tgl: unify core_put on IPC3 & IPC4 path ASoC: SOF: Intel: mtl: setup primary core info on MeteorLake platform Ranjani Sridharan (14): ASoC: SOF: ipc4-topology: Handle input/output audio format special case ASoC: SOF: ipc4-topology: Add a helper function for output format selection ASoC: SOF: ipc4-topology: Move the call to init output format ASoC: SOF: ipc4-topology: Rename sof_ipc4_init_audio_fmt() ASoC: SOF: ipc4-topology: Handle output format special case ASoC: SOF: ipc4-topology: Add a new helper function to get the valid bits ASoC: SOF: ipc4-topology: Modify the output format selection logic ASoC: SOF: ipc4-topology: New helper to check if all output formats are the same ASoC: SOF: ipc4-topology: Modify input audio format selection logic ASoC: SOF: Intel: hda-dai-ops: Split the get_hext_stream() op for IPC4 ASoC: SOF: ipc4-pcm: reset all pipelines during FE DAI hw_free ASoC: topology: Allow partial matching when finding DAI link ASoC: SOF: topology: Use partial match for connecting DAI link and DAI widget ASoC: SOF: Intel: HDA: Limit the number of dai drivers for nocodec mode Ricardo Ribalda Delgado (2): ASoC: mediatek: mt8173: Fix snd_soc_component_initialize error path ASoC: mediatek: mt8173: Fix irq error path Richard Fitzgerald (1): firmware: cs_dsp: Log that a bin file was loaded Rob Herring (3): ASoC: dt-bindings: More dropping unneeded quotes ASoC: dt-bindings: ti,tlv320aic3x: Add missing type for "gpio-reset" ASoC: dt-bindings: microchip,sama7g5-pdmc: Simplify "microchip,mic-pos" constraints Rsplwe (1): ASoC: amd: yc: Add MECHREVO Jiaolong Series MRID6 into DMI table Ryan Lee (2): ASoC: dt-bindings: max98388: add amplifier driver ASoC: max98388: add amplifier driver Sameer Pujar (2): ASoC: tegra: Use normal system sleep for ASRC ASoC: tegra: Remove stale comments in AHUB Shenghao Ding (4): ASoC: tas2781: Add Header file for tas2781 driver ASoC: tas2781: firmware lib ASoC: tas2781: Add tas2781 driver ASoC: dt-bindings: Add tas2781 amplifier Shengjiu Wang (1): ASoC: fsl_sai: MCLK bind with TX/RX enable bit Simon Trimmer (3): ASoC: cs35l56: Move DSP part string generation so that it is done only once ASoC: cs35l56: sdw_write_no_pm() should be performed under a pm_runtime request ASoC: cs35l56: In secure mode skip SHUTDOWN and RESET around fw download Srinivas Kandagatla (12): ASoC: qcom: q6dsp-common: move channel allocation to common ASoC: qcom: audioreach: add support for DISPLAY PORT SINK module ASoC: qcom: q6dsp: add support to more display ports ASoC: qcom: q6apm: add support to display ports in lpass dais ASoC: q6dsp: audioreach: add helper function to set u32 param ASoC: q6dsp: audioreach: Add placeholder decoder for compress playback ASoC: q6dsp: q6apm-dai: Add open/free compress DAI callbacks ASoC: q6dsp: q6apm-dai: Add compress DAI and codec caps get callbacks ASoC: q6dsp: q6apm-dai: Add trigger/pointer compress DAI callbacks ASoC: q6dsp: q6apm-dai: Add compress set params and metadata DAI callbacks ASoC: q6dsp: q6apm-dai: Add mmap and copy compress DAI callbacks ASoC: qcom: common: add default jack dapm pins Syed Saba Kareem (4): ASoC: amd: ps: remove the register read and write wrappers. ASoC: amd: ps: refactor acp power on and reset functions. ASoC: amd: acp: clear pdm dma interrupt mask ASoC: amd: acp: remove acp poweroff function Terry Cheong (3): ASoC: Intel: Add rpl_max98373_8825 driver ASoC: Intel: Add rpl_rt1019_rt5682 driver ASoC: Intel: sof_rt5682: reorder quirk table Tom Rix (1): ASoC: max98388: set variable soc_codec_dev_max98388 storage-class-specifier to static Trevor Wu (22): ASoC: mediatek: mt8188: remove supply AUDIO_HIRES ASoC: mediatek: mt8188: complete set_tdm_slot function ASoC: mediatek: mt8188: revise ETDM control flow ASoC: mediatek: mt8188: refine APLL control ASoC: mediatek: mt8188: combine afe component registration ASoC: mediatek: mt8188: add bus protection ASoC: mediatek: mt8188: add required clocks ASoC: dt-bindings: mediatek,mt8188-afe: correct clock name ASoC: dt-bindings: mediatek,mt8188-afe: add audio properties ASoC: mediatek: mt6359: add supply for MTKAIF ASoC: mediatek: mt6359: fix kselftest error of playback gain ASoC: mediatek: mt6359: add mtkaif gpio setting ASoC: mediatek: mt6359: update route for lineout mux ASoC: SOF: mediatek: add mt8188 audio support ASoC: SOF: mediatek: add adsp debug dump ASoC: mediatek: mt8188: separate ADDA playback dai from capture dai ASoC: dt-bindings: mediatek,mt8188-mt6359: remove ADDA_BE from link-name ASoC: mediatek: mt8188-mt6359: register hdmi/dp jack pins ASoC: mediatek: common: soundcard driver add dai_fmt support ASoC: soc-dapm.c: clean up debugfs for freed widget ASoC: mediatek: mt8188-mt6359: support new board with nau88255 ASoC: dt-bindings: mediatek,mt8188-mt6359: add NAU8825 support Uday M Bhat (6): ASoC: Intel: sof_sdw: Add helper function for cs42l42 codec ASoC: Intel: sof_sdw: Rename sof_sdw_max98373.c file to sof_sdw_maxim.c ASoC: Intel: sof_sdw: Modify maxim helper functions and structure names ASoC: Intel: sof_sdw: Add support for MAX98363 codec ASoC: Intel: sof_sdw: Add support for Rex soundwire ASoC: Intel: soc-acpi: add Rex CS42l42 and MAX98363 SoundWire entries Uwe Kleine-König (2): ASoC: Switch i2c drivers back to use .probe() ASoC: Switch two more i2c drivers back to use .probe() Venkata Prasad Potturu (3): ASoC: SOF: amd: Add pci revision id check ASoC: amd: vangogh: Add check for acp config flags in vangogh platform ASoC: SOF: amd: Add support for IPC with a reply_size set to zero Vijendar Mukunda (9): ASoC: amd: ps: create platform devices based on acp config ASoC: amd: ps: handle SoundWire interrupts in acp pci driver ASoC: amd: ps: add SoundWire dma driver ASoC: amd: ps: add SoundWire dma driver dma ops ASoC: amd: ps: add support for SoundWire DMA interrupts ASoC: amd: ps: add pm ops support for SoundWire dma driver ASoC: amd: ps: enable SoundWire dma driver build ASoC: amd: update comments in Kconfig file ASoC: amd: ps: add acp_reset flag check in acp pci driver pm ops. Vitaly Rodionov (1): ASoC: cs42l42: Add PLL ratio table values Walker Chen (4): ASoC: dt-bindings: Add TDM controller bindings for StarFive JH7110 ASoC: starfive: Add JH7110 TDM driver ASoC: starfive: Fix an error check in jh7110_tdm_clk_reset_get() ASoC: starfive: Remove some unused macros Yang Li (2): ASoC: codecs: wcd938x: Remove unneeded semicolon ASoC: SOF: ipc4-topology: Fix an unsigned comparison which can never be negative YingKun Meng (1): ASoC: loongson: change the type of variable irq to int Yingkun Meng (3): ASoC: Add support for Loongson I2S controller ASoC: loongson: Add Loongson ASoC Sound Card Support ASoC: dt-bindings: Add support for Loongson audio card .mailmap | 5 + CREDITS | 4 + .../admin-guide/quickly-build-trimmed-linux.rst | 49 +- Documentation/block/index.rst | 1 - Documentation/block/request.rst | 99 - Documentation/cdrom/index.rst | 6 +- .../devicetree/bindings/ata/ceva,ahci-1v84.yaml | 2 +- .../bindings/display/msm/dsi-controller-main.yaml | 12 + 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.../devicetree/bindings/sound/qcom,wsa8840.yaml | 66 + .../devicetree/bindings/sound/realtek,rt1016.yaml | 40 + Documentation/devicetree/bindings/sound/rt1016.txt | 17 - .../bindings/sound/snps,designware-i2s.yaml | 3 +- .../devicetree/bindings/sound/ssm2518.txt | 20 - .../devicetree/bindings/sound/st,stm32-i2s.yaml | 11 + .../bindings/sound/starfive,jh7110-tdm.yaml | 98 + .../devicetree/bindings/sound/tas2562.yaml | 4 +- .../devicetree/bindings/sound/tas2770.yaml | 4 +- .../devicetree/bindings/sound/tas27xx.yaml | 4 +- .../devicetree/bindings/sound/ti,tas2781.yaml | 74 + .../bindings/sound/ti,tlv320aic32x4.yaml | 101 + .../devicetree/bindings/sound/ti,tlv320aic3x.yaml | 1 + .../devicetree/bindings/sound/tlv320aic32x4.txt | 42 - .../devicetree/bindings/sound/wlf,wm8903.yaml | 4 +- .../filesystems/ramfs-rootfs-initramfs.rst | 3 +- Documentation/filesystems/sharedsubtree.rst | 4 + Documentation/fpga/index.rst | 2 +- Documentation/locking/index.rst | 2 +- Documentation/networking/bonding.rst | 9 +- Documentation/networking/index.rst | 2 +- Documentation/networking/x25-iface.rst | 3 +- Documentation/pcmcia/index.rst | 2 +- Documentation/s390/vfio-ap.rst | 1 - Documentation/staging/crc32.rst | 2 +- Documentation/timers/index.rst | 2 +- MAINTAINERS | 2327 +++++++++---------- Makefile | 2 +- arch/arm/boot/dts/stm32f429.dtsi | 1 + arch/arm/boot/dts/stm32f7-pinctrl.dtsi | 82 + arch/arm/include/asm/arm_pmuv3.h | 6 +- arch/arm/kernel/unwind.c | 25 +- arch/arm/mach-sa1100/jornada720_ssp.c | 5 +- arch/arm/vfp/entry.S | 7 +- arch/arm/vfp/vfphw.S | 6 +- arch/arm64/include/asm/arm_pmuv3.h | 6 +- arch/arm64/include/asm/cputype.h | 8 + arch/arm64/include/asm/kvm_pgtable.h | 1 + arch/arm64/kernel/mte.c | 7 +- arch/arm64/kernel/vdso.c | 2 +- arch/arm64/kvm/fpsimd.c | 26 +- arch/arm64/kvm/hyp/include/hyp/switch.h | 12 +- arch/arm64/kvm/hyp/pgtable.c | 41 +- arch/arm64/kvm/inject_fault.c | 2 +- arch/arm64/kvm/vgic/vgic-v3.c | 4 + arch/arm64/kvm/vmid.c | 7 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fs/ext4/namei.c | 53 +- fs/ext4/super.c | 37 +- fs/ext4/xattr.c | 5 +- fs/gfs2/super.c | 8 + fs/ksmbd/connection.c | 3 +- fs/ksmbd/oplock.c | 5 +- fs/ksmbd/oplock.h | 2 +- fs/ksmbd/smb2misc.c | 5 +- fs/ksmbd/smb2pdu.c | 19 +- fs/lockd/svc.c | 2 +- fs/nfs/dir.c | 2 +- fs/nfsd/nfsctl.c | 25 +- fs/nfsd/trace.h | 6 +- fs/nilfs2/inode.c | 18 + fs/notify/inotify/inotify_fsnotify.c | 11 +- fs/pipe.c | 6 +- fs/statfs.c | 4 +- fs/xfs/libxfs/xfs_ag.c | 19 +- fs/xfs/libxfs/xfs_bmap.c | 5 +- fs/xfs/scrub/bmap.c | 4 +- fs/xfs/scrub/common.c | 26 - fs/xfs/scrub/common.h | 2 - fs/xfs/scrub/fscounters.c | 13 +- fs/xfs/scrub/scrub.c | 2 - fs/xfs/scrub/scrub.h | 1 - fs/xfs/scrub/trace.h | 1 - fs/xfs/xfs_bmap_util.c | 4 +- fs/xfs/xfs_icache.c | 40 +- fs/xfs/xfs_iomap.c | 5 +- fs/xfs/xfs_mount.h | 3 + fs/xfs/xfs_super.c | 3 + include/drm/display/drm_dp.h | 5 +- include/drm/display/drm_dp_helper.h | 5 +- include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h | 8 + include/linux/blkdev.h | 2 - include/linux/compiler.h | 5 +- include/linux/device/class.h | 1 + include/linux/dim.h | 3 +- include/linux/if_vlan.h | 17 + include/linux/io_uring.h | 10 +- include/linux/pci_ids.h | 1 + include/linux/phy.h | 2 - include/linux/shrinker.h | 13 +- include/linux/sunrpc/svc_rdma.h | 2 +- include/linux/sunrpc/svc_xprt.h | 2 +- include/linux/tpm.h | 1 + include/linux/usb/composite.h | 2 +- include/media/dvb_frontend.h | 6 +- include/media/dvb_net.h | 4 + include/media/dvbdev.h | 15 + include/net/bonding.h | 2 +- include/net/nexthop.h | 23 - include/net/sock.h | 2 +- include/sound/cs35l56.h | 1 + include/sound/da7219-aad.h | 6 + include/sound/simple_card_utils.h | 7 +- include/sound/soc-acpi-intel-match.h | 2 + include/sound/soc-component.h | 15 +- include/sound/soc.h | 40 +- include/sound/tas2781-dsp.h | 183 ++ include/sound/tas2781-tlv.h | 21 + include/sound/tas2781.h | 164 ++ include/ufs/ufshcd.h | 2 +- kernel/events/core.c | 14 +- kernel/locking/rwsem.c | 8 +- kernel/sched/core.c | 4 +- kernel/time/tick-broadcast.c | 120 +- kernel/trace/fprobe.c | 73 +- kernel/trace/rethook.c | 4 +- lib/dim/dim.c | 5 +- lib/dim/net_dim.c | 3 +- lib/dim/rdma_dim.c | 3 +- lib/maple_tree.c | 12 +- mm/kfence/kfence.h | 2 +- mm/shrinker_debug.c | 15 +- mm/vmscan.c | 5 +- mm/zsmalloc.c | 36 +- mm/zswap.c | 16 + net/8021q/vlan_dev.c | 4 +- net/atm/resources.c | 2 + net/bridge/br_forward.c | 2 +- net/bridge/br_private_tunnel.h | 8 +- net/can/isotp.c | 2 +- net/can/j1939/socket.c | 2 +- net/core/datagram.c | 15 +- net/core/dev.c | 2 +- net/core/skbuff.c | 4 +- net/core/stream.c | 12 +- net/devlink/core.c | 16 +- net/devlink/devl_internal.h | 1 - net/devlink/leftover.c | 5 +- net/ipv4/af_inet.c | 2 +- net/ipv4/tcp.c | 14 +- net/ipv4/tcp_bpf.c | 2 +- net/ipv4/tcp_input.c | 4 +- net/ipv4/tcp_ipv4.c | 5 +- net/ipv6/ip6_fib.c | 16 +- net/ipv6/ip6_gre.c | 13 +- net/key/af_key.c | 12 +- net/llc/af_llc.c | 8 +- net/mac80211/cfg.c | 7 +- net/mac80211/chan.c | 75 +- net/mac80211/ieee80211_i.h | 3 +- net/mac80211/trace.h | 2 +- net/mac80211/tx.c | 5 +- net/mac80211/util.c | 2 +- net/netfilter/core.c | 6 +- net/netfilter/nf_conntrack_netlink.c | 4 + net/netfilter/nf_conntrack_standalone.c | 3 +- net/netfilter/nf_tables_api.c | 4 +- net/netfilter/nft_chain_filter.c | 9 +- net/netfilter/nft_set_rbtree.c | 20 +- net/netlink/af_netlink.c | 8 +- net/nsh/nsh.c | 8 +- net/packet/af_packet.c | 6 +- net/smc/smc_close.c | 4 +- net/smc/smc_rx.c | 4 +- net/smc/smc_tx.c | 4 +- net/socket.c | 2 +- net/sunrpc/auth_gss/gss_krb5_crypto.c | 10 + net/sunrpc/svc.c | 19 +- net/sunrpc/svc_xprt.c | 24 +- net/sunrpc/svcsock.c | 46 +- net/sunrpc/xprtrdma/svc_rdma_recvfrom.c | 11 +- net/sunrpc/xprtrdma/svc_rdma_transport.c | 2 +- net/tipc/bearer.c | 17 +- net/tipc/bearer.h | 3 + net/tipc/link.c | 9 +- net/tipc/socket.c | 4 +- net/tipc/udp_media.c | 5 +- net/tls/tls_main.c | 3 +- net/unix/af_unix.c | 22 +- net/vmw_vsock/af_vsock.c | 2 +- net/wireless/scan.c | 6 +- net/xfrm/xfrm_device.c | 2 +- net/xfrm/xfrm_interface_core.c | 54 +- net/xfrm/xfrm_policy.c | 20 +- net/xfrm/xfrm_user.c | 15 +- sound/core/oss/pcm_plugin.h | 16 +- sound/firewire/digi00x/digi00x-stream.c | 4 +- sound/hda/hdac_regmap.c | 1 - sound/pci/cs46xx/cs46xx_lib.c | 2 +- sound/pci/hda/hda_bind.c | 1 - sound/pci/hda/hda_generic.c | 7 +- sound/pci/hda/patch_hdmi.c | 5 + sound/pci/hda/patch_realtek.c | 34 +- sound/soc/Kconfig | 3 + sound/soc/Makefile | 3 + sound/soc/amd/Kconfig | 5 +- sound/soc/amd/acp-da7219-max98357a.c | 20 +- sound/soc/amd/acp-es8336.c | 2 +- sound/soc/amd/acp/acp-mach-common.c | 43 +- sound/soc/amd/acp/acp-pci.c | 1 - sound/soc/amd/acp/acp-pdm.c | 2 +- sound/soc/amd/acp/acp-platform.c | 1 - sound/soc/amd/acp/acp-rembrandt.c | 25 - sound/soc/amd/acp/acp-renoir.c | 17 - sound/soc/amd/ps/Makefile | 2 + sound/soc/amd/ps/acp63.h | 183 +- sound/soc/amd/ps/pci-ps.c | 484 +++- sound/soc/amd/ps/ps-pdm-dma.c | 66 +- sound/soc/amd/ps/ps-sdw-dma.c | 555 +++++ sound/soc/amd/raven/acp3x-pcm-dma.c | 3 +- sound/soc/amd/renoir/acp3x-pdm-dma.c | 3 +- sound/soc/amd/vangogh/acp5x-pcm-dma.c | 4 +- sound/soc/amd/vangogh/acp5x.h | 2 + sound/soc/amd/vangogh/pci-acp5x.c | 7 +- sound/soc/amd/yc/acp6x-mach.c | 7 + sound/soc/amd/yc/acp6x-pdm-dma.c | 3 +- sound/soc/atmel/atmel-classd.c | 8 +- sound/soc/atmel/atmel-pdmic.c | 8 +- sound/soc/atmel/mchp-pdmc.c | 2 +- sound/soc/atmel/sam9g20_wm8731.c | 1 - sound/soc/codecs/Kconfig | 67 + sound/soc/codecs/Makefile | 16 + sound/soc/codecs/ad193x-i2c.c | 2 +- sound/soc/codecs/adau1372-i2c.c | 2 +- sound/soc/codecs/adau1373.c | 2 +- sound/soc/codecs/adau1701.c | 2 +- sound/soc/codecs/adau1761-i2c.c | 2 +- sound/soc/codecs/adau1781-i2c.c | 2 +- sound/soc/codecs/adau17x1.c | 13 +- sound/soc/codecs/adau1977-i2c.c | 2 +- sound/soc/codecs/adau7118-i2c.c | 2 +- sound/soc/codecs/adav803.c | 2 +- sound/soc/codecs/ak4118.c | 13 +- sound/soc/codecs/ak4375.c | 2 +- sound/soc/codecs/ak4458.c | 2 +- sound/soc/codecs/ak4535.c | 2 +- sound/soc/codecs/ak4613.c | 2 +- sound/soc/codecs/ak4641.c | 2 +- sound/soc/codecs/ak4642.c | 2 +- sound/soc/codecs/ak4671.c | 2 +- sound/soc/codecs/ak5558.c | 2 +- sound/soc/codecs/alc5623.c | 2 +- sound/soc/codecs/alc5632.c | 2 +- sound/soc/codecs/aw88395/aw88395.c | 2 +- sound/soc/codecs/chv3-codec.c | 41 + sound/soc/codecs/cs35l32.c | 4 +- sound/soc/codecs/cs35l33.c | 4 +- sound/soc/codecs/cs35l34.c | 4 +- sound/soc/codecs/cs35l35.c | 4 +- sound/soc/codecs/cs35l36.c | 2 +- sound/soc/codecs/cs35l41-i2c.c | 2 +- sound/soc/codecs/cs35l45-i2c.c | 6 +- sound/soc/codecs/cs35l45-spi.c | 4 +- sound/soc/codecs/cs35l45-tables.c | 2 +- sound/soc/codecs/cs35l45.c | 4 +- sound/soc/codecs/cs35l45.h | 2 +- sound/soc/codecs/cs35l56-i2c.c | 2 +- sound/soc/codecs/cs35l56.c | 65 +- sound/soc/codecs/cs4234.c | 4 +- sound/soc/codecs/cs4265.c | 2 +- sound/soc/codecs/cs4270.c | 2 +- sound/soc/codecs/cs4271-i2c.c | 2 +- sound/soc/codecs/cs42l42-i2c.c | 2 +- sound/soc/codecs/cs42l42.c | 9 +- sound/soc/codecs/cs42l51-i2c.c | 2 +- sound/soc/codecs/cs42l51.c | 9 +- sound/soc/codecs/cs42l52.c | 2 +- sound/soc/codecs/cs42l56.c | 2 +- sound/soc/codecs/cs42l73.c | 4 +- sound/soc/codecs/cs42l83-i2c.c | 4 +- sound/soc/codecs/cs42xx8-i2c.c | 2 +- sound/soc/codecs/cs43130.c | 4 +- sound/soc/codecs/cs4341.c | 2 +- sound/soc/codecs/cs4349.c | 2 +- sound/soc/codecs/cs53l30.c | 14 +- sound/soc/codecs/cx2072x.c | 2 +- sound/soc/codecs/da7210.c | 2 +- sound/soc/codecs/da7213.c | 2 +- sound/soc/codecs/da7218.c | 2 +- sound/soc/codecs/da7219-aad.c | 42 +- sound/soc/codecs/da7219.c | 2 +- sound/soc/codecs/da732x.c | 2 +- sound/soc/codecs/da9055.c | 2 +- sound/soc/codecs/es8316.c | 27 +- sound/soc/codecs/es8326.c | 2 +- sound/soc/codecs/es8328-i2c.c | 2 +- sound/soc/codecs/es8328.c | 2 +- sound/soc/codecs/isabelle.c | 2 +- sound/soc/codecs/lm4857.c | 2 +- sound/soc/codecs/lm49453.c | 2 +- sound/soc/codecs/max9768.c | 2 +- sound/soc/codecs/max98088.c | 24 +- sound/soc/codecs/max98090.c | 56 +- sound/soc/codecs/max98090.h | 3 +- sound/soc/codecs/max98095.c | 2 +- sound/soc/codecs/max98363.c | 6 - sound/soc/codecs/max98371.c | 2 +- sound/soc/codecs/max98373-i2c.c | 4 +- sound/soc/codecs/max98388.c | 1013 ++++++++ sound/soc/codecs/max98388.h | 234 ++ sound/soc/codecs/max98390.c | 2 +- sound/soc/codecs/max98396.c | 2 +- sound/soc/codecs/max9850.c | 2 +- sound/soc/codecs/max98504.c | 2 +- sound/soc/codecs/max98520.c | 2 +- sound/soc/codecs/max9860.c | 2 +- sound/soc/codecs/max9867.c | 2 +- sound/soc/codecs/max9877.c | 2 +- sound/soc/codecs/max98925.c | 2 +- sound/soc/codecs/max98926.c | 2 +- sound/soc/codecs/max98927.c | 2 +- sound/soc/codecs/ml26124.c | 2 +- sound/soc/codecs/mt6359.c | 152 +- sound/soc/codecs/mt6660.c | 2 +- sound/soc/codecs/nau8540.c | 2 +- sound/soc/codecs/nau8810.c | 2 +- sound/soc/codecs/nau8821.c | 2 +- sound/soc/codecs/nau8822.c | 2 +- sound/soc/codecs/nau8824.c | 2 +- sound/soc/codecs/nau8825.c | 150 +- sound/soc/codecs/nau8825.h | 11 + sound/soc/codecs/pcm1681.c | 2 +- sound/soc/codecs/pcm1789-i2c.c | 2 +- sound/soc/codecs/pcm179x-i2c.c | 2 +- sound/soc/codecs/pcm186x-i2c.c | 2 +- sound/soc/codecs/pcm186x.c | 1 - sound/soc/codecs/pcm3060-i2c.c | 2 +- sound/soc/codecs/pcm3168a-i2c.c | 2 +- sound/soc/codecs/pcm512x-i2c.c | 2 +- sound/soc/codecs/rk3328_codec.c | 1 - sound/soc/codecs/rt1011.c | 4 +- sound/soc/codecs/rt1015.c | 2 +- sound/soc/codecs/rt1016.c | 2 +- sound/soc/codecs/rt1019.c | 4 +- sound/soc/codecs/rt1305.c | 4 +- sound/soc/codecs/rt1308-sdw.c | 7 +- sound/soc/codecs/rt1308-sdw.h | 1 - sound/soc/codecs/rt1308.c | 4 +- sound/soc/codecs/rt1316-sdw.c | 7 +- sound/soc/codecs/rt1316-sdw.h | 1 - sound/soc/codecs/rt1318-sdw.c | 7 +- sound/soc/codecs/rt1318-sdw.h | 1 - sound/soc/codecs/rt274.c | 2 +- sound/soc/codecs/rt286.c | 2 +- sound/soc/codecs/rt298.c | 2 +- sound/soc/codecs/rt5514.c | 4 +- sound/soc/codecs/rt5616.c | 4 +- sound/soc/codecs/rt5631.c | 4 +- sound/soc/codecs/rt5640.c | 4 +- sound/soc/codecs/rt5645.c | 6 +- sound/soc/codecs/rt5651.c | 4 +- sound/soc/codecs/rt5659.c | 12 +- sound/soc/codecs/rt5660.c | 4 +- sound/soc/codecs/rt5663.c | 6 +- sound/soc/codecs/rt5665.c | 4 +- sound/soc/codecs/rt5668.c | 4 +- sound/soc/codecs/rt5670.c | 4 +- sound/soc/codecs/rt5677.c | 4 +- sound/soc/codecs/rt5682-i2c.c | 5 +- sound/soc/codecs/rt5682-sdw.c | 7 +- sound/soc/codecs/rt5682.h | 1 - sound/soc/codecs/rt5682s.c | 16 +- sound/soc/codecs/rt700-sdw.c | 7 +- sound/soc/codecs/rt700.h | 1 - sound/soc/codecs/rt711-sdca-sdw.c | 9 +- sound/soc/codecs/rt711-sdca.h | 1 - sound/soc/codecs/rt711-sdw.c | 7 +- sound/soc/codecs/rt711.h | 1 - sound/soc/codecs/rt712-sdca-dmic.c | 9 +- sound/soc/codecs/rt712-sdca-dmic.h | 1 - sound/soc/codecs/rt712-sdca-sdw.c | 9 +- sound/soc/codecs/rt712-sdca.h | 1 - sound/soc/codecs/rt715-sdca-sdw.c | 9 +- sound/soc/codecs/rt715-sdca.h | 1 - sound/soc/codecs/rt715-sdw.c | 6 +- sound/soc/codecs/rt715.h | 1 - sound/soc/codecs/rt722-sdca-sdw.c | 507 ++++ sound/soc/codecs/rt722-sdca-sdw.h | 124 + sound/soc/codecs/rt722-sdca.c | 1555 +++++++++++++ sound/soc/codecs/rt722-sdca.h | 237 ++ sound/soc/codecs/rt9120.c | 2 +- sound/soc/codecs/sgtl5000.c | 2 +- sound/soc/codecs/sma1303.c | 2 +- sound/soc/codecs/src4xxx-i2c.c | 2 +- sound/soc/codecs/ssm2518.c | 2 +- sound/soc/codecs/ssm2602-i2c.c | 2 +- sound/soc/codecs/ssm3515.c | 448 ++++ sound/soc/codecs/ssm4567.c | 2 +- sound/soc/codecs/sta32x.c | 2 +- sound/soc/codecs/sta350.c | 2 +- sound/soc/codecs/sta529.c | 2 +- sound/soc/codecs/tas2552.c | 2 +- sound/soc/codecs/tas2562.c | 3 +- sound/soc/codecs/tas2764.c | 2 +- sound/soc/codecs/tas2770.c | 2 +- sound/soc/codecs/tas2780.c | 2 +- sound/soc/codecs/tas2781-comlib.c | 534 +++++ sound/soc/codecs/tas2781-fmwlib.c | 2428 ++++++++++++++++++++ sound/soc/codecs/tas2781-i2c.c | 763 ++++++ sound/soc/codecs/tas5086.c | 2 +- sound/soc/codecs/tas571x.c | 2 +- sound/soc/codecs/tas5720.c | 3 +- sound/soc/codecs/tas5805m.c | 2 +- sound/soc/codecs/tas6424.c | 3 +- sound/soc/codecs/tda7419.c | 2 +- sound/soc/codecs/tfa9879.c | 2 +- sound/soc/codecs/tfa989x.c | 2 +- sound/soc/codecs/tlv320adc3xxx.c | 2 +- sound/soc/codecs/tlv320adcx140.c | 2 +- sound/soc/codecs/tlv320aic23-i2c.c | 2 +- sound/soc/codecs/tlv320aic31xx.c | 2 +- sound/soc/codecs/tlv320aic32x4-i2c.c | 2 +- sound/soc/codecs/tlv320aic3x-i2c.c | 2 +- sound/soc/codecs/tlv320dac33.c | 2 +- sound/soc/codecs/tpa6130a2.c | 2 +- sound/soc/codecs/ts3a227e.c | 2 +- sound/soc/codecs/tscs42xx.c | 2 +- sound/soc/codecs/tscs454.c | 2 +- sound/soc/codecs/uda1380.c | 2 +- sound/soc/codecs/wcd938x-sdw.c | 2 +- sound/soc/codecs/wm0010.c | 3 + sound/soc/codecs/wm1250-ev1.c | 2 +- sound/soc/codecs/wm2000.c | 2 +- sound/soc/codecs/wm2200.c | 2 +- sound/soc/codecs/wm5100.c | 2 +- sound/soc/codecs/wm8510.c | 2 +- sound/soc/codecs/wm8523.c | 2 +- sound/soc/codecs/wm8580.c | 2 +- sound/soc/codecs/wm8711.c | 2 +- sound/soc/codecs/wm8728.c | 2 +- sound/soc/codecs/wm8731-i2c.c | 2 +- sound/soc/codecs/wm8737.c | 2 +- sound/soc/codecs/wm8741.c | 2 +- sound/soc/codecs/wm8750.c | 2 +- sound/soc/codecs/wm8753.c | 2 +- sound/soc/codecs/wm8776.c | 2 +- sound/soc/codecs/wm8804-i2c.c | 2 +- sound/soc/codecs/wm8900.c | 2 +- sound/soc/codecs/wm8903.c | 2 +- sound/soc/codecs/wm8904.c | 2 +- sound/soc/codecs/wm8940.c | 2 +- sound/soc/codecs/wm8955.c | 2 +- sound/soc/codecs/wm8960.c | 2 +- sound/soc/codecs/wm8961.c | 2 +- sound/soc/codecs/wm8962.c | 2 +- sound/soc/codecs/wm8971.c | 2 +- sound/soc/codecs/wm8974.c | 2 +- sound/soc/codecs/wm8978.c | 2 +- sound/soc/codecs/wm8983.c | 2 +- sound/soc/codecs/wm8985.c | 2 +- sound/soc/codecs/wm8988.c | 2 +- sound/soc/codecs/wm8990.c | 2 +- sound/soc/codecs/wm8991.c | 2 +- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm8995.c | 2 +- sound/soc/codecs/wm8996.c | 2 +- sound/soc/codecs/wm9081.c | 2 +- sound/soc/codecs/wm9090.c | 2 +- sound/soc/codecs/wm_adsp.c | 21 +- sound/soc/codecs/wsa883x.c | 3 +- sound/soc/codecs/wsa884x.c | 1936 ++++++++++++++++ sound/soc/dwc/dwc-i2s.c | 78 +- sound/soc/dwc/local.h | 7 + sound/soc/fsl/fsl-asoc-card.c | 22 +- sound/soc/fsl/fsl_sai.c | 24 +- sound/soc/fsl/fsl_sai.h | 2 + sound/soc/fsl/imx-audmix.c | 24 +- sound/soc/fsl/imx-card.c | 25 +- sound/soc/fsl/imx-rpmsg.c | 6 +- sound/soc/fsl/imx-spdif.c | 8 +- sound/soc/generic/audio-graph-card.c | 107 +- .../generic/audio-graph-card2-custom-sample.dtsi | 165 +- sound/soc/generic/audio-graph-card2.c | 107 +- sound/soc/generic/simple-card-utils.c | 118 +- sound/soc/generic/simple-card.c | 4 +- sound/soc/google/Kconfig | 6 + sound/soc/google/Makefile | 2 + sound/soc/google/chv3-i2s.c | 338 +++ sound/soc/intel/atom/sst/sst_acpi.c | 1 - sound/soc/intel/atom/sst/sst_ipc.c | 1 - sound/soc/intel/atom/sst/sst_loader.c | 1 - sound/soc/intel/atom/sst/sst_pci.c | 1 - sound/soc/intel/atom/sst/sst_stream.c | 1 - sound/soc/intel/avs/boards/da7219.c | 45 +- sound/soc/intel/avs/boards/dmic.c | 2 - sound/soc/intel/avs/boards/hdaudio.c | 65 +- sound/soc/intel/avs/boards/i2s_test.c | 6 +- sound/soc/intel/avs/boards/max98357a.c | 39 +- sound/soc/intel/avs/boards/max98373.c | 45 +- sound/soc/intel/avs/boards/max98927.c | 45 +- sound/soc/intel/avs/boards/nau8825.c | 45 +- sound/soc/intel/avs/boards/rt274.c | 45 +- sound/soc/intel/avs/boards/rt286.c | 45 +- sound/soc/intel/avs/boards/rt298.c | 45 +- sound/soc/intel/avs/boards/rt5682.c | 45 +- sound/soc/intel/avs/boards/ssm4567.c | 57 +- sound/soc/intel/boards/Kconfig | 4 + sound/soc/intel/boards/Makefile | 10 +- sound/soc/intel/boards/ehl_rt5660.c | 8 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 8 +- sound/soc/intel/boards/sof_cs42l42.c | 11 +- sound/soc/intel/boards/sof_es8336.c | 11 +- sound/soc/intel/boards/sof_nau8825.c | 21 +- sound/soc/intel/boards/sof_pcm512x.c | 3 +- sound/soc/intel/boards/sof_rt5682.c | 94 +- sound/soc/intel/boards/sof_sdw.c | 669 ++++-- sound/soc/intel/boards/sof_sdw_common.h | 86 +- sound/soc/intel/boards/sof_sdw_cs42l42.c | 131 ++ .../boards/{sof_sdw_max98373.c => sof_sdw_maxim.c} | 59 +- sound/soc/intel/boards/sof_sdw_rt711.c | 4 +- sound/soc/intel/boards/sof_sdw_rt712_sdca.c | 102 + ..._rt711_sdca.c => sof_sdw_rt_sdca_jack_common.c} | 73 +- sound/soc/intel/boards/sof_ssp_amp.c | 29 +- sound/soc/intel/common/Makefile | 1 + sound/soc/intel/common/soc-acpi-intel-adl-match.c | 46 +- sound/soc/intel/common/soc-acpi-intel-lnl-match.c | 72 + sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 167 ++ sound/soc/intel/common/soc-acpi-intel-rpl-match.c | 54 + sound/soc/intel/common/soc-acpi-intel-tgl-match.c | 53 + sound/soc/jz4740/jz4740-i2s.c | 9 + sound/soc/loongson/Kconfig | 27 + sound/soc/loongson/Makefile | 8 + sound/soc/loongson/loongson_card.c | 218 ++ sound/soc/loongson/loongson_dma.c | 350 +++ sound/soc/loongson/loongson_dma.h | 16 + sound/soc/loongson/loongson_i2s.c | 269 +++ sound/soc/loongson/loongson_i2s.h | 71 + sound/soc/loongson/loongson_i2s_pci.c | 171 ++ sound/soc/mediatek/Kconfig | 5 + sound/soc/mediatek/common/mtk-soundcard-driver.c | 53 +- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 13 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 2 +- sound/soc/mediatek/mt8186/mt8186-afe-control.c | 1 - .../mt8186/mt8186-mt6366-da7219-max98357.c | 1 - .../mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c | 1 - sound/soc/mediatek/mt8188/mt8188-afe-clk.c | 142 +- sound/soc/mediatek/mt8188/mt8188-afe-clk.h | 15 + sound/soc/mediatek/mt8188/mt8188-afe-common.h | 3 +- sound/soc/mediatek/mt8188/mt8188-afe-pcm.c | 113 +- sound/soc/mediatek/mt8188/mt8188-dai-adda.c | 110 +- sound/soc/mediatek/mt8188/mt8188-dai-etdm.c | 922 ++++---- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 453 +++- sound/soc/mediatek/mt8188/mt8188-reg.h | 2 + sound/soc/mediatek/mt8192/mt8192-afe-control.c | 2 - sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 99 +- sound/soc/meson/axg-card.c | 11 +- sound/soc/meson/gx-card.c | 3 +- sound/soc/meson/meson-card-utils.c | 26 +- sound/soc/meson/meson-card.h | 3 +- sound/soc/qcom/common.c | 34 +- sound/soc/qcom/lpass-sc7180.c | 2 +- sound/soc/qcom/lpass-sc7280.c | 2 +- sound/soc/qcom/qdsp6/audioreach.c | 325 ++- sound/soc/qcom/qdsp6/audioreach.h | 63 + sound/soc/qcom/qdsp6/q6afe-dai.c | 34 +- sound/soc/qcom/qdsp6/q6apm-dai.c | 445 ++++ sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 39 + sound/soc/qcom/qdsp6/q6apm.c | 68 + sound/soc/qcom/qdsp6/q6apm.h | 6 + sound/soc/qcom/qdsp6/q6dsp-common.c | 35 + sound/soc/qcom/qdsp6/q6dsp-common.h | 1 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 43 +- sound/soc/qcom/sc7280.c | 23 +- sound/soc/samsung/odroid.c | 16 +- sound/soc/sh/siu_dai.c | 2 + sound/soc/soc-component.c | 22 + sound/soc/soc-compress.c | 1 - sound/soc/soc-core.c | 112 +- sound/soc/soc-dapm.c | 75 +- sound/soc/soc-pcm.c | 209 +- sound/soc/soc-topology.c | 91 +- sound/soc/soc-utils.c | 7 + sound/soc/sof/Kconfig | 11 + sound/soc/sof/Makefile | 2 + sound/soc/sof/amd/acp-ipc.c | 7 + sound/soc/sof/amd/acp.h | 3 + sound/soc/sof/amd/pci-rmb.c | 3 + sound/soc/sof/amd/pci-rn.c | 3 + sound/soc/sof/core.c | 4 +- sound/soc/sof/intel/Kconfig | 3 +- sound/soc/sof/intel/hda-bus.c | 11 +- sound/soc/sof/intel/hda-dai-ops.c | 137 +- sound/soc/sof/intel/hda-dai.c | 168 +- sound/soc/sof/intel/hda-loader-skl.c | 1 - sound/soc/sof/intel/hda-pcm.c | 2 +- sound/soc/sof/intel/hda-stream.c | 1 - sound/soc/sof/intel/hda.c | 43 +- sound/soc/sof/intel/hda.h | 25 +- sound/soc/sof/intel/mtl.c | 64 +- sound/soc/sof/intel/mtl.h | 7 + sound/soc/sof/intel/skl.c | 1 - sound/soc/sof/intel/tgl.c | 10 +- sound/soc/sof/ipc3-control.c | 54 +- sound/soc/sof/ipc3-priv.h | 2 + sound/soc/sof/ipc3.c | 102 +- sound/soc/sof/ipc4-control.c | 39 +- sound/soc/sof/ipc4-loader.c | 72 +- sound/soc/sof/ipc4-pcm.c | 4 +- sound/soc/sof/ipc4-priv.h | 10 +- sound/soc/sof/ipc4-topology.c | 459 ++-- sound/soc/sof/ipc4-topology.h | 74 +- sound/soc/sof/ipc4.c | 44 +- sound/soc/sof/mediatek/mt8186/mt8186-clk.c | 1 - sound/soc/sof/mediatek/mt8186/mt8186.c | 119 +- sound/soc/sof/mediatek/mt8186/mt8186.h | 5 + sound/soc/sof/mediatek/mt8195/mt8195-clk.c | 1 - sound/soc/sof/mediatek/mt8195/mt8195.c | 36 +- sound/soc/sof/nocodec.c | 8 +- sound/soc/sof/pcm.c | 2 +- sound/soc/sof/pm.c | 12 +- sound/soc/sof/sof-audio.c | 47 +- sound/soc/sof/sof-audio.h | 1 + sound/soc/sof/sof-client-ipc-kernel-injector.c | 162 ++ sound/soc/sof/sof-client.c | 52 + sound/soc/sof/sof-client.h | 1 + sound/soc/sof/sof-priv.h | 3 + sound/soc/sof/topology.c | 2 +- sound/soc/starfive/Kconfig | 15 + sound/soc/starfive/Makefile | 2 + sound/soc/starfive/jh7110_tdm.c | 670 ++++++ sound/soc/stm/stm32_sai_sub.c | 9 +- sound/soc/tegra/tegra186_asrc.c | 4 +- sound/soc/tegra/tegra20_ac97.c | 1 - sound/soc/tegra/tegra20_i2s.c | 9 +- sound/soc/tegra/tegra20_spdif.c | 9 +- sound/soc/tegra/tegra210_adx.c | 2 +- sound/soc/tegra/tegra210_ahub.c | 10 - sound/soc/ti/davinci-mcasp.c | 27 +- sound/soc/ti/omap-hdmi.c | 8 +- sound/soc/ti/omap-mcbsp-st.c | 1 - sound/usb/format.c | 1 + tools/arch/arm64/include/uapi/asm/kvm.h | 36 + tools/arch/x86/include/asm/cpufeatures.h | 26 +- tools/arch/x86/include/asm/disabled-features.h | 8 +- tools/arch/x86/include/asm/msr-index.h | 2 + tools/arch/x86/include/uapi/asm/kvm.h | 3 + tools/arch/x86/include/uapi/asm/prctl.h | 8 + tools/arch/x86/include/uapi/asm/unistd_32.h | 3 + tools/arch/x86/lib/memcpy_64.S | 34 +- tools/arch/x86/lib/memset_64.S | 47 +- tools/include/asm/alternative.h | 3 +- tools/include/uapi/drm/drm.h | 57 +- tools/include/uapi/drm/i915_drm.h | 25 +- tools/include/uapi/linux/const.h | 2 +- tools/include/uapi/linux/in.h | 1 + tools/include/uapi/linux/kvm.h | 12 +- tools/include/uapi/linux/prctl.h | 2 + tools/include/uapi/sound/asound.h | 14 +- tools/perf/Makefile.config | 6 + tools/perf/Makefile.perf | 22 +- tools/perf/arch/arm/util/cs-etm.c | 9 +- tools/perf/arch/arm64/util/header.c | 4 +- tools/perf/arch/arm64/util/pmu.c | 2 +- tools/perf/arch/s390/entry/syscalls/syscall.tbl | 2 +- tools/perf/bench/mem-memcpy-x86-64-asm-def.h | 4 - tools/perf/bench/mem-memcpy-x86-64-asm.S | 2 +- tools/perf/bench/mem-memset-x86-64-asm-def.h | 4 - tools/perf/bench/mem-memset-x86-64-asm.S | 2 +- tools/perf/builtin-script.c | 7 + tools/perf/builtin-stat.c | 38 +- .../pmu-events/arch/x86/alderlake/adl-metrics.json | 26 + .../arch/x86/alderlaken/adln-metrics.json | 14 + .../pmu-events/arch/x86/broadwell/bdw-metrics.json | 12 + .../arch/x86/broadwellde/bdwde-metrics.json | 12 + .../arch/x86/broadwellx/bdx-metrics.json | 12 + .../arch/x86/cascadelakex/clx-metrics.json | 12 + .../pmu-events/arch/x86/haswell/hsw-metrics.json | 12 + .../pmu-events/arch/x86/haswellx/hsx-metrics.json | 12 + .../pmu-events/arch/x86/icelake/icl-metrics.json | 12 + .../pmu-events/arch/x86/icelakex/icx-metrics.json | 12 + .../pmu-events/arch/x86/ivybridge/ivb-metrics.json | 12 + .../pmu-events/arch/x86/ivytown/ivt-metrics.json | 12 + .../pmu-events/arch/x86/jaketown/jkt-metrics.json | 12 + .../arch/x86/sandybridge/snb-metrics.json | 12 + .../arch/x86/sapphirerapids/spr-metrics.json | 12 + .../pmu-events/arch/x86/skylake/skl-metrics.json | 12 + .../pmu-events/arch/x86/skylakex/skx-metrics.json | 12 + .../pmu-events/arch/x86/tigerlake/tgl-metrics.json | 12 + tools/perf/pmu-events/jevents.py | 4 +- tools/perf/pmu-events/pmu-events.h | 1 + tools/perf/tests/attr.py | 6 +- tools/perf/tests/attr/base-stat | 2 +- tools/perf/tests/attr/test-stat-default | 80 +- tools/perf/tests/attr/test-stat-detailed-1 | 95 +- tools/perf/tests/attr/test-stat-detailed-2 | 119 +- tools/perf/tests/attr/test-stat-detailed-3 | 127 +- tools/perf/tests/expr.c | 3 +- tools/perf/tests/parse-metric.c | 1 + tools/perf/tests/shell/stat.sh | 13 + tools/perf/tests/shell/test_intel_pt.sh | 7 + tools/perf/tests/shell/test_java_symbol.sh | 2 +- tools/perf/trace/beauty/arch_prctl.c | 2 + tools/perf/trace/beauty/x86_arch_prctl.sh | 1 + tools/perf/util/bpf_skel/lock_contention.bpf.c | 2 + tools/perf/util/bpf_skel/vmlinux.h | 1 + tools/perf/util/evsel.c | 37 +- tools/perf/util/evsel.h | 1 + tools/perf/util/expr.y | 6 +- tools/perf/util/metricgroup.c | 10 +- tools/perf/util/parse-events.c | 23 +- tools/perf/util/stat-display.c | 2 +- tools/perf/util/stat-shadow.c | 25 +- tools/power/cpupower/lib/powercap.c | 23 +- .../cpupower/utils/idle_monitor/mperf_monitor.c | 31 +- tools/testing/cxl/test/mock.c | 2 +- .../selftests/drivers/net/bonding/bond_options.sh | 50 + .../drivers/net/bonding/bond_topo_3d1c.sh | 2 + tools/testing/selftests/ftrace/Makefile | 3 +- tools/testing/selftests/ftrace/ftracetest | 63 +- tools/testing/selftests/ftrace/ftracetest-ktap | 8 + tools/testing/selftests/net/fib_nexthops.sh | 2 +- tools/testing/selftests/net/forwarding/lib.sh | 3 +- .../selftests/net/srv6_end_dt4_l3vpn_test.sh | 17 +- tools/testing/selftests/netfilter/nft_flowtable.sh | 145 +- tools/testing/selftests/sgx/Makefile | 1 + virt/kvm/kvm_main.c | 59 +- 1093 files changed, 28206 insertions(+), 7728 deletions(-) delete mode 100644 Documentation/block/request.rst create mode 100644 Documentation/devicetree/bindings/sound/adi,max98388.yaml create mode 100644 Documentation/devicetree/bindings/sound/adi,ssm2518.yaml create mode 100644 Documentation/devicetree/bindings/sound/adi,ssm3515.yaml delete mode 100644 Documentation/devicetree/bindings/sound/da7219.txt create mode 100644 Documentation/devicetree/bindings/sound/dialog,da7219.yaml create mode 100644 Documentation/devicetree/bindings/sound/google,chv3-codec.yaml create mode 100644 Documentation/devicetree/bindings/sound/google,chv3-i2s.yaml create mode 100644 Documentation/devicetree/bindings/sound/loongson,ls-audio-card.yaml delete mode 100644 Documentation/devicetree/bindings/sound/nau8315.txt delete mode 100644 Documentation/devicetree/bindings/sound/nau8540.txt delete mode 100644 Documentation/devicetree/bindings/sound/nau8810.txt delete mode 100644 Documentation/devicetree/bindings/sound/nau8824.txt delete mode 100644 Documentation/devicetree/bindings/sound/nau8825.txt create mode 100644 Documentation/devicetree/bindings/sound/nuvoton,nau8315.yaml create mode 100644 Documentation/devicetree/bindings/sound/nuvoton,nau8540.yaml create mode 100644 Documentation/devicetree/bindings/sound/nuvoton,nau8810.yaml create mode 100644 Documentation/devicetree/bindings/sound/nuvoton,nau8824.yaml create mode 100644 Documentation/devicetree/bindings/sound/nuvoton,nau8825.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml create mode 100644 Documentation/devicetree/bindings/sound/realtek,rt1016.yaml delete mode 100644 Documentation/devicetree/bindings/sound/rt1016.txt delete mode 100644 Documentation/devicetree/bindings/sound/ssm2518.txt create mode 100644 Documentation/devicetree/bindings/sound/starfive,jh7110-tdm.yaml create mode 100644 Documentation/devicetree/bindings/sound/ti,tas2781.yaml create mode 100644 Documentation/devicetree/bindings/sound/ti,tlv320aic32x4.yaml delete mode 100644 Documentation/devicetree/bindings/sound/tlv320aic32x4.txt create mode 100644 arch/um/drivers/harddog.h create mode 100644 arch/um/drivers/harddog_user_exp.c create mode 100644 include/sound/tas2781-dsp.h create mode 100644 include/sound/tas2781-tlv.h create mode 100644 include/sound/tas2781.h create mode 100644 sound/soc/amd/ps/ps-sdw-dma.c create mode 100644 sound/soc/codecs/chv3-codec.c create mode 100644 sound/soc/codecs/max98388.c create mode 100644 sound/soc/codecs/max98388.h create mode 100644 sound/soc/codecs/rt722-sdca-sdw.c create mode 100644 sound/soc/codecs/rt722-sdca-sdw.h create mode 100644 sound/soc/codecs/rt722-sdca.c create mode 100644 sound/soc/codecs/rt722-sdca.h create mode 100644 sound/soc/codecs/ssm3515.c create mode 100644 sound/soc/codecs/tas2781-comlib.c create mode 100644 sound/soc/codecs/tas2781-fmwlib.c create mode 100644 sound/soc/codecs/tas2781-i2c.c create mode 100644 sound/soc/codecs/wsa884x.c create mode 100644 sound/soc/google/Kconfig create mode 100644 sound/soc/google/Makefile create mode 100644 sound/soc/google/chv3-i2s.c create mode 100644 sound/soc/intel/boards/sof_sdw_cs42l42.c rename sound/soc/intel/boards/{sof_sdw_max98373.c => sof_sdw_maxim.c} (65%) create mode 100644 sound/soc/intel/boards/sof_sdw_rt712_sdca.c rename sound/soc/intel/boards/{sof_sdw_rt711_sdca.c => sof_sdw_rt_sdca_jack_common.c} (61%) create mode 100644 sound/soc/intel/common/soc-acpi-intel-lnl-match.c create mode 100644 sound/soc/loongson/Kconfig create mode 100644 sound/soc/loongson/Makefile create mode 100644 sound/soc/loongson/loongson_card.c create mode 100644 sound/soc/loongson/loongson_dma.c create mode 100644 sound/soc/loongson/loongson_dma.h create mode 100644 sound/soc/loongson/loongson_i2s.c create mode 100644 sound/soc/loongson/loongson_i2s.h create mode 100644 sound/soc/loongson/loongson_i2s_pci.c create mode 100644 sound/soc/sof/sof-client-ipc-kernel-injector.c create mode 100644 sound/soc/starfive/Kconfig create mode 100644 sound/soc/starfive/Makefile create mode 100644 sound/soc/starfive/jh7110_tdm.c create mode 100755 tools/testing/selftests/ftrace/ftracetest-ktap

2 1

[BUG] ALSA: core: pcm_memory: a possible data race in do_alloc_pages()
by Tuo Li 26 Jun '23

26 Jun '23

Hello, Our static analysis tool finds a possible data race in ALSA in Linux 6.4.0. In some functions, the field snd_card.total_pcm_alloc_bytes is accessed with holding the lock snd_card.memory_mutex. Here is an example: do_free_pages() --> Line 57 mutex_lock(&card->memory_mutex); --> Line 61 (Lock card->memory_mutex) card->total_pcm_alloc_bytes -= dmab->bytes; --> Line 63 (Access card->total_pcm_alloc_bytes) However, in the function do_alloc_pages(): if (max_alloc_per_card && card->total_pcm_alloc_bytes + size > max_alloc_per_card) --> Line 41 the variable card->total_pcm_alloc_bytes is accessed without holding the lock card->memory_mutex, and thus a data race can occur. In my opinion, this data race may be harmful, because the value of card->total_pcm_alloc_bytes may be changed by another thread after the if check. Therefore, its value may be too large after Line 51 and can cause memory bugs such as buffer overflow: card->total_pcm_alloc_bytes += dmab->bytes; --> Line 51 I am not quite sure whether this possible data race is real and how to fix it if it is real. Any feedback would be appreciated, thanks! Reported-by: BassCheck <bass(a)buaa.edu.cn> Best wishes, Tuo Li

3 11

[PATCH] ALSA: hda/realtek: Enable mute/micmute LEDs and limit mic boost on EliteBook
by Andy Chi 26 Jun '23

26 Jun '23

On HP EliteBook 835/845/845W G10, the audio LEDs can be enabled by ALC285_FIXUP_HP_MUTE_LED. So use it accordingly. Signed-off-by: Andy Chi <andy.chi(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ea09de384c91..42d39ad84117 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9493,9 +9493,9 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8b63, "HP Elite Dragonfly 13.5 inch G4", ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b65, "HP ProBook 455 15.6 inch G10 Notebook PC", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x8b66, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), - SND_PCI_QUIRK(0x103c, 0x8b70, "HP EliteBook 835 G10", ALC287_FIXUP_CS35L41_I2C_2), - SND_PCI_QUIRK(0x103c, 0x8b72, "HP EliteBook 845 G10", ALC287_FIXUP_CS35L41_I2C_2), - SND_PCI_QUIRK(0x103c, 0x8b74, "HP EliteBook 845W G10", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8b70, "HP EliteBook 835 G10", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8b72, "HP EliteBook 845 G10", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8b74, "HP EliteBook 845W G10", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b77, "HP ElieBook 865 G10", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8b7a, "HP", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b7d, "HP", ALC236_FIXUP_HP_GPIO_LED), -- 2.34.1

2 1

udev attributes to detect compress-offload devices
by Carlos Rafael Giani 26 Jun '23

26 Jun '23

Are there attributes in udev that allow for unambiguously detect a compress-offload device? Currently, I have the problem that I can't autodetect compress-offload devices reliably, because I see nothing in udev that identifies a device as such. The next best thing thus far has to be to rely on the /dev/snd/comprC*D* name. But, these device nodes can be renamed, can't they?

1 0

[PATCH 0/8] Few audio fixes on Tegra platforms
by Sameer Pujar 26 Jun '23

26 Jun '23

This series fixes some of the issues which were observed during an attempt to enhance automated test coverage on Jetson AGX Orin. Below is a short summary of the issues and fixes: * Sample rate coversion failures above 48kHz. * AMX and ADX test cases failures due to incorrect byte mask. * Atomic sleep in RT5640 codec which is present on Jetson AGX Orin. * AHUB clock fixes on Tegra234 and previous chips. * Minor cleanups in ASRC and AHUB driver. Sameer Pujar (4): ASoC: rt5640: Fix sleep in atomic context ASoC: tegra: Use normal system sleep for ASRC ASoC: tegra: Remove stale comments in AHUB arm64: tegra: Update AHUB clock parent and rate Sheetal (4): ASoC: tegra: Fix SFC conversion for few rates ASoC: tegra: Fix AMX byte map ASoC: tegra: Fix ADX byte map arm64: tegra: Update AHUB clock parent and rate on Tegra234 arch/arm64/boot/dts/nvidia/tegra186.dtsi | 3 ++- arch/arm64/boot/dts/nvidia/tegra194.dtsi | 3 ++- arch/arm64/boot/dts/nvidia/tegra210.dtsi | 3 ++- arch/arm64/boot/dts/nvidia/tegra234.dtsi | 3 ++- sound/soc/codecs/rt5640.c | 20 ++++++++++++-------- sound/soc/tegra/tegra186_asrc.c | 4 ++-- sound/soc/tegra/tegra210_adx.c | 24 ++++++++++++------------ sound/soc/tegra/tegra210_ahub.c | 10 ---------- sound/soc/tegra/tegra210_amx.c | 30 ++++++++++++------------------ sound/soc/tegra/tegra210_sfc.c | 31 ++++++++++++++++++++++++++++++- sound/soc/tegra/tegra210_sfc.h | 4 ++-- 11 files changed, 78 insertions(+), 57 deletions(-) -- 2.7.4

2 17

[PATCH] ASoC: loongson: fix address space confusion
by Arnd Bergmann 26 Jun '23

26 Jun '23

From: Arnd Bergmann <arnd(a)arndb.de> The i2s driver uses the mapped __iomem address of the FIFO as the DMA address for the device. This apparently works on loongarch because of the way it handles __iomem pointers as aliases of physical addresses, but this is not portable to other architectures and causes a compiler warning when dma addresses are not the same size as pointers: sound/soc/loongson/loongson_i2s_pci.c: In function 'loongson_i2s_pci_probe': sound/soc/loongson/loongson_i2s_pci.c:110:29: error: cast from pointer to integer of different size [-Werror=pointer-to-int-cast] 110 | tx_data->dev_addr = (dma_addr_t)i2s->reg_base + LS_I2S_TX_DATA; | ^ sound/soc/loongson/loongson_i2s_pci.c:113:29: error: cast from pointer to integer of different size [-Werror=pointer-to-int-cast] 113 | rx_data->dev_addr = (dma_addr_t)i2s->reg_base + LS_I2S_RX_DATA; | ^ Change the driver to instead use the physical address as stored in the PCI BAR resource directly. Since 'dev_addr' is a 32-bit address, I think this results in the same truncated address on loongarch but is otherwise closer to portable code and avoids the warning. Fixes: d84881e06836d ("ASoC: Add support for Loongson I2S controller") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- This is only build tested and I don't understand the loongarch specifics that well, please review or test for actual hardware requirements. --- sound/soc/loongson/loongson_i2s_pci.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/loongson/loongson_i2s_pci.c b/sound/soc/loongson/loongson_i2s_pci.c index 6dcfb17d3276d..fa90361865c6c 100644 --- a/sound/soc/loongson/loongson_i2s_pci.c +++ b/sound/soc/loongson/loongson_i2s_pci.c @@ -107,10 +107,10 @@ static int loongson_i2s_pci_probe(struct pci_dev *pdev, tx_data = &i2s->tx_dma_data; rx_data = &i2s->rx_dma_data; - tx_data->dev_addr = (dma_addr_t)i2s->reg_base + LS_I2S_TX_DATA; + tx_data->dev_addr = pci_resource_start(pdev, 0) + LS_I2S_TX_DATA; tx_data->order_addr = i2s->reg_base + LS_I2S_TX_ORDER; - rx_data->dev_addr = (dma_addr_t)i2s->reg_base + LS_I2S_RX_DATA; + rx_data->dev_addr = pci_resource_start(pdev, 0) + LS_I2S_RX_DATA; rx_data->order_addr = i2s->reg_base + LS_I2S_RX_ORDER; tx_data->irq = fwnode_irq_get_byname(fwnode, "tx"); -- 2.39.2

3 2

[PATCH v2] ASoC: amd: update pm_runtime enable sequence
by Arun Gopal Kondaveeti 26 Jun '23

26 Jun '23

pm_runtime_allow() is not needed for ACP child platform devices. Replace pm_runtime_allow() with pm_runtime_mark_last_busy() & pm_runtime_set_active() in pm_runtime enable sequence for ACP child platform drivers. Signed-off-by: Arun Gopal <arungopal.kondaveeti(a)amd.com> --- Changes since v1: -- Updated commit message --- sound/soc/amd/ps/ps-pdm-dma.c | 3 ++- sound/soc/amd/raven/acp3x-pcm-dma.c | 3 ++- sound/soc/amd/renoir/acp3x-pdm-dma.c | 3 ++- sound/soc/amd/vangogh/acp5x-pcm-dma.c | 4 ++-- sound/soc/amd/yc/acp6x-pdm-dma.c | 3 ++- 5 files changed, 10 insertions(+), 6 deletions(-) diff --git a/sound/soc/amd/ps/ps-pdm-dma.c b/sound/soc/amd/ps/ps-pdm-dma.c index bdbbb797c74d..d48f7c5af289 100644 --- a/sound/soc/amd/ps/ps-pdm-dma.c +++ b/sound/soc/amd/ps/ps-pdm-dma.c @@ -391,8 +391,9 @@ static int acp63_pdm_audio_probe(struct platform_device *pdev) } pm_runtime_set_autosuspend_delay(&pdev->dev, ACP_SUSPEND_DELAY_MS); pm_runtime_use_autosuspend(&pdev->dev); + pm_runtime_mark_last_busy(&pdev->dev); + pm_runtime_set_active(&pdev->dev); pm_runtime_enable(&pdev->dev); - pm_runtime_allow(&pdev->dev); return 0; } diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 7362dd15ad30..9538f3ffc5d9 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -416,8 +416,9 @@ static int acp3x_audio_probe(struct platform_device *pdev) pm_runtime_set_autosuspend_delay(&pdev->dev, 2000); pm_runtime_use_autosuspend(&pdev->dev); + pm_runtime_mark_last_busy(&pdev->dev); + pm_runtime_set_active(&pdev->dev); pm_runtime_enable(&pdev->dev); - pm_runtime_allow(&pdev->dev); return 0; } diff --git a/sound/soc/amd/renoir/acp3x-pdm-dma.c b/sound/soc/amd/renoir/acp3x-pdm-dma.c index 4e299f96521f..c3b47e9bd239 100644 --- a/sound/soc/amd/renoir/acp3x-pdm-dma.c +++ b/sound/soc/amd/renoir/acp3x-pdm-dma.c @@ -430,8 +430,9 @@ static int acp_pdm_audio_probe(struct platform_device *pdev) } pm_runtime_set_autosuspend_delay(&pdev->dev, ACP_SUSPEND_DELAY_MS); pm_runtime_use_autosuspend(&pdev->dev); + pm_runtime_mark_last_busy(&pdev->dev); + pm_runtime_set_active(&pdev->dev); pm_runtime_enable(&pdev->dev); - pm_runtime_allow(&pdev->dev); return 0; } diff --git a/sound/soc/amd/vangogh/acp5x-pcm-dma.c b/sound/soc/amd/vangogh/acp5x-pcm-dma.c index 29901ee4bfe3..587dec5bb33d 100644 --- a/sound/soc/amd/vangogh/acp5x-pcm-dma.c +++ b/sound/soc/amd/vangogh/acp5x-pcm-dma.c @@ -409,9 +409,9 @@ static int acp5x_audio_probe(struct platform_device *pdev) } pm_runtime_set_autosuspend_delay(&pdev->dev, 2000); pm_runtime_use_autosuspend(&pdev->dev); + pm_runtime_mark_last_busy(&pdev->dev); + pm_runtime_set_active(&pdev->dev); pm_runtime_enable(&pdev->dev); - pm_runtime_allow(&pdev->dev); - return 0; } diff --git a/sound/soc/amd/yc/acp6x-pdm-dma.c b/sound/soc/amd/yc/acp6x-pdm-dma.c index d818eba48546..72c4591e451b 100644 --- a/sound/soc/amd/yc/acp6x-pdm-dma.c +++ b/sound/soc/amd/yc/acp6x-pdm-dma.c @@ -383,8 +383,9 @@ static int acp6x_pdm_audio_probe(struct platform_device *pdev) } pm_runtime_set_autosuspend_delay(&pdev->dev, ACP_SUSPEND_DELAY_MS); pm_runtime_use_autosuspend(&pdev->dev); + pm_runtime_mark_last_busy(&pdev->dev); + pm_runtime_set_active(&pdev->dev); pm_runtime_enable(&pdev->dev); - pm_runtime_allow(&pdev->dev); return 0; } -- 2.25.1

4 5

ALC287 problem
by ollierazdow＠protonmail.com 26 Jun '23

26 Jun '23

I have a problem with the ALC287 codec that might indicate a missing fix in the Kernel. There is digital noise in the output unless I run the command: sudo hda-verb /dev/snd/hwC0D0 0x1d SET_PIN_WIDGET_CONTROL 0x0 I looked through the file /sound/pci/hda/patch_realtek.c and found ALC285_FIXUP_LENOVO_PC_BEEP_IN_NOISE but it does not affect my hardware. The latest Kernel I tried was 6.1.32. I'm not familiar enough with how the file is organized to write a patch myself, but If anyone has advice about how to address it I could try. I wanted to bring alert to this though because it seems like a missing fix. I also posted here with a bit more detail: https://bugzilla.kernel.org/show_bug.cgi?id=217580

1 0

USB-Audio: Add Steinberg UR44C
by GitHub pull_request - opened 25 Jun '23

25 Jun '23

alsa-project/alsa-ucm-conf pull request #330 was opened from cyberchrime: Adds support for the Steinberg UR44C. In fact, it is similar to the Steinberg UR44, but uses USB-C with USB 3.1 and has a different USB devnum (1730 instead of 1700). Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/330 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/330.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

MIC capture stops due to playback
by Arjav Parikh 25 Jun '23

25 Jun '23

Hi I am using alsa api's in my C Program to implement two-way talk i.e capturing data from mic and playing sound on speaker received from Cloud. I am using a Microsemi ZL38050 Audio DSP connected to an Ambarella CV25 SOC via I2S/SPI. Randomly I have observed that If I stop and resume receiving data from Cloud to play it on speaker using snd_pcm_writei() then my capture path stops. I do not hear any sound captured from MIC. Can someone help me to debug this issue because when this issue occurs there are no errors reported from Alsa nor do I see anything in dmesg as well. Thanks & Regards, Arjav Parikh

1 0

[PATCH] ALSA: aloop: Fix &cable->lock deadlock issues
by YE Chengfeng 25 Jun '23

25 Jun '23

The timer loopback_jiffies_timer_function is executed under bottom-half softirq context and require a spinlock, thus other process context code requiring the same lock (i.e., loopback_trigger, loopback_pointer) can deadlock with the timer if it is preempted while holding the lock. Deadlock scenario: loopback_trigger -> spin_lock(&cable->lock); <timer interrupt> -> loopback_jiffies_timer_function -> spin_lock_irqsave(&dpcm->cable->lock, flags); Fix the potential deadlock by using spin_lock_irqsave. Signed-off-by: Chengfeng Ye <cyeaa(a)connect.ust.hk> --- sound/drivers/aloop.c | 22 ++++++++++++---------- 1 file changed, 12 insertions(+), 10 deletions(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index a38e602b4fc6..8ee93f8581b4 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -379,6 +379,7 @@ static int loopback_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_pcm_runtime *runtime = substream->runtime; struct loopback_pcm *dpcm = runtime->private_data; struct loopback_cable *cable = dpcm->cable; + unsigned long flags; int err = 0, stream = 1 << substream->stream; switch (cmd) { @@ -389,39 +390,39 @@ static int loopback_trigger(struct snd_pcm_substream *substream, int cmd) dpcm->last_jiffies = jiffies; dpcm->pcm_rate_shift = 0; dpcm->last_drift = 0; - spin_lock(&cable->lock); + spin_lock_irqsave(&cable->lock, flags); cable->running |= stream; cable->pause &= ~stream; err = cable->ops->start(dpcm); - spin_unlock(&cable->lock); + spin_unlock_irqrestore(&cable->lock, flags); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) loopback_active_notify(dpcm); break; case SNDRV_PCM_TRIGGER_STOP: - spin_lock(&cable->lock); + spin_lock_irqsave(&cable->lock, flags); cable->running &= ~stream; cable->pause &= ~stream; err = cable->ops->stop(dpcm); - spin_unlock(&cable->lock); + spin_unlock_irqrestore(&cable->lock, flags); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) loopback_active_notify(dpcm); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: - spin_lock(&cable->lock); + spin_lock_irqsave(&cable->lock, flags); cable->pause |= stream; err = cable->ops->stop(dpcm); - spin_unlock(&cable->lock); + spin_unlock_irqrestore(&cable->lock, flags); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) loopback_active_notify(dpcm); break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: - spin_lock(&cable->lock); + spin_lock_irqsave(&cable->lock, flags); dpcm->last_jiffies = jiffies; cable->pause &= ~stream; err = cable->ops->start(dpcm); - spin_unlock(&cable->lock); + spin_unlock_irqrestore(&cable->lock, flags); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) loopback_active_notify(dpcm); break; @@ -865,12 +866,13 @@ static snd_pcm_uframes_t loopback_pointer(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct loopback_pcm *dpcm = runtime->private_data; snd_pcm_uframes_t pos; + unsigned long flags; - spin_lock(&dpcm->cable->lock); + spin_lock_irqsave(&dpcm->cable->lock, flags); if (dpcm->cable->ops->pos_update) dpcm->cable->ops->pos_update(dpcm->cable); pos = dpcm->buf_pos; - spin_unlock(&dpcm->cable->lock); + spin_unlock_irqrestore(&dpcm->cable->lock, flags); return bytes_to_frames(runtime, pos); } -- 2.17.1

2 1

[PATCH] ALSA: pcm: fix ELD constraints for (E)AC3, DTS(-HD) and MLP formats
by Matthias Reichl 25 Jun '23

25 Jun '23

The SADs of compressed formats contain the channel and sample rate info of the audio data inside the compressed stream, but when building constraints we must use the rates and channels used to transport the compressed streams. eg 48kHz 6ch EAC3 needs to be transmitted as a 2ch 192kHz stream. This patch fixes the constraints for the common AC3 and DTS formats, the constraints for the less common MPEG, DSD etc formats are copied directly from the info in the SADs as before as I don't have the specs and equipment to test those. Signed-off-by: Matthias Reichl <hias(a)horus.com> --- sound/core/pcm_drm_eld.c | 73 ++++++++++++++++++++++++++++++++++++++-- 1 file changed, 70 insertions(+), 3 deletions(-) diff --git a/sound/core/pcm_drm_eld.c b/sound/core/pcm_drm_eld.c index 4b5faae5d16e5..07075071972dd 100644 --- a/sound/core/pcm_drm_eld.c +++ b/sound/core/pcm_drm_eld.c @@ -2,11 +2,25 @@ /* * PCM DRM helpers */ +#include <linux/bitfield.h> #include <linux/export.h> +#include <linux/hdmi.h> #include <drm/drm_edid.h> #include <sound/pcm.h> #include <sound/pcm_drm_eld.h> +#define SAD0_CHANNELS_MASK GENMASK(2, 0) /* max number of channels - 1 */ +#define SAD0_FORMAT_MASK GENMASK(6, 3) /* audio format */ + +#define SAD1_RATE_MASK GENMASK(6, 0) /* bitfield of supported rates */ +#define SAD1_RATE_32000_MASK BIT(0) +#define SAD1_RATE_44100_MASK BIT(1) +#define SAD1_RATE_48000_MASK BIT(2) +#define SAD1_RATE_88200_MASK BIT(3) +#define SAD1_RATE_96000_MASK BIT(4) +#define SAD1_RATE_176400_MASK BIT(5) +#define SAD1_RATE_192000_MASK BIT(6) + static const unsigned int eld_rates[] = { 32000, 44100, @@ -17,9 +31,62 @@ static const unsigned int eld_rates[] = { 192000, }; +static unsigned int map_rate_families(const u8 *sad, + unsigned int mask_32000, + unsigned int mask_44100, + unsigned int mask_48000) +{ + unsigned int rate_mask = 0; + + if (sad[1] & SAD1_RATE_32000_MASK) + rate_mask |= mask_32000; + if (sad[1] & (SAD1_RATE_44100_MASK | SAD1_RATE_88200_MASK | SAD1_RATE_176400_MASK)) + rate_mask |= mask_44100; + if (sad[1] & (SAD1_RATE_48000_MASK | SAD1_RATE_96000_MASK | SAD1_RATE_192000_MASK)) + rate_mask |= mask_48000; + return rate_mask; +} + +static unsigned int sad_rate_mask(const u8 *sad) +{ + switch (FIELD_GET(SAD0_FORMAT_MASK, sad[0])) { + case HDMI_AUDIO_CODING_TYPE_PCM: + return sad[1] & SAD1_RATE_MASK; + case HDMI_AUDIO_CODING_TYPE_AC3: + case HDMI_AUDIO_CODING_TYPE_DTS: + return map_rate_families(sad, + SAD1_RATE_32000_MASK, + SAD1_RATE_44100_MASK, + SAD1_RATE_48000_MASK); + case HDMI_AUDIO_CODING_TYPE_EAC3: + case HDMI_AUDIO_CODING_TYPE_DTS_HD: + case HDMI_AUDIO_CODING_TYPE_MLP: + return map_rate_families(sad, + 0, + SAD1_RATE_176400_MASK, + SAD1_RATE_192000_MASK); + default: + /* TODO adjust for other compressed formats as well */ + return sad[1] & SAD1_RATE_MASK; + } +} + static unsigned int sad_max_channels(const u8 *sad) { - return 1 + (sad[0] & 7); + switch (FIELD_GET(SAD0_FORMAT_MASK, sad[0])) { + case HDMI_AUDIO_CODING_TYPE_PCM: + return 1 + FIELD_GET(SAD0_CHANNELS_MASK, sad[0]); + case HDMI_AUDIO_CODING_TYPE_AC3: + case HDMI_AUDIO_CODING_TYPE_DTS: + case HDMI_AUDIO_CODING_TYPE_EAC3: + return 2; + case HDMI_AUDIO_CODING_TYPE_DTS_HD: + case HDMI_AUDIO_CODING_TYPE_MLP: + return 8; + default: + /* TODO adjust for other compressed formats as well */ + return 1 + FIELD_GET(SAD0_CHANNELS_MASK, sad[0]); + } } static int eld_limit_rates(struct snd_pcm_hw_params *params, @@ -42,7 +109,7 @@ static int eld_limit_rates(struct snd_pcm_hw_params *params, * requested number of channels. */ if (c->min <= max_channels) - rate_mask |= sad[1]; + rate_mask |= sad_rate_mask(sad); } } @@ -70,7 +137,7 @@ static int eld_limit_channels(struct snd_pcm_hw_params *params, rate_mask |= BIT(i); for (i = drm_eld_sad_count(eld); i > 0; i--, sad += 3) - if (rate_mask & sad[1]) + if (rate_mask & sad_rate_mask(sad)) t.max = max(t.max, sad_max_channels(sad)); } -- 2.39.2

3 4

[PATCH v2 RESEND] ASoC: fsl MPC52xx drivers require PPC_BESTCOMM
by Randy Dunlap 25 Jun '23

25 Jun '23

Both SND_MPC52xx_SOC_PCM030 and SND_MPC52xx_SOC_EFIKA select SND_SOC_MPC5200_AC97. The latter symbol depends on PPC_BESTCOMM, so the 2 former symbols should also depend on PPC_BESTCOMM since "select" does not follow any dependency chains. This prevents a kconfig warning and build errors: WARNING: unmet direct dependencies detected for SND_SOC_MPC5200_AC97 Depends on [n]: SOUND [=y] && !UML && SND [=m] && SND_SOC [=m] && SND_POWERPC_SOC [=m] && PPC_MPC52xx [=y] && PPC_BESTCOMM [=n] Selected by [m]: - SND_MPC52xx_SOC_PCM030 [=m] && SOUND [=y] && !UML && SND [=m] && SND_SOC [=m] && SND_POWERPC_SOC [=m] && PPC_MPC5200_SIMPLE [=y] - SND_MPC52xx_SOC_EFIKA [=m] && SOUND [=y] && !UML && SND [=m] && SND_SOC [=m] && SND_POWERPC_SOC [=m] && PPC_EFIKA [=y] ERROR: modpost: "mpc5200_audio_dma_destroy" [sound/soc/fsl/mpc5200_psc_ac97.ko] undefined! ERROR: modpost: "mpc5200_audio_dma_create" [sound/soc/fsl/mpc5200_psc_ac97.ko] undefined! Fixes: 40d9ec14e7e1 ("ASoC: remove BROKEN from Efika and pcm030 fabric drivers") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Grant Likely <grant.likely(a)secretlab.ca> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Shengjiu Wang <shengjiu.wang(a)gmail.com> Cc: Xiubo Li <Xiubo.Lee(a)gmail.com> Cc: alsa-devel(a)alsa-project.org Cc: linuxppc-dev(a)lists.ozlabs.org Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> --- v2: use correct email address for Mark Brown. sound/soc/fsl/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff -- a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -243,7 +243,7 @@ config SND_SOC_MPC5200_AC97 config SND_MPC52xx_SOC_PCM030 tristate "SoC AC97 Audio support for Phytec pcm030 and WM9712" - depends on PPC_MPC5200_SIMPLE + depends on PPC_MPC5200_SIMPLE && PPC_BESTCOMM select SND_SOC_MPC5200_AC97 select SND_SOC_WM9712 help @@ -252,7 +252,7 @@ config SND_MPC52xx_SOC_PCM030 config SND_MPC52xx_SOC_EFIKA tristate "SoC AC97 Audio support for bbplan Efika and STAC9766" - depends on PPC_EFIKA + depends on PPC_EFIKA && PPC_BESTCOMM select SND_SOC_MPC5200_AC97 select SND_SOC_STAC9766 help

2 3

Focusrite Scarlett 2i2 3rd Gen suffers from high output latency with the 1.2.9-1 update
by GitHub issues - edited 25 Jun '23

25 Jun '23

alsa-project/alsa-ucm-conf issue #329 was edited from IsaacLambat: Hello, I have recently upgraded from `alsa-ucm-conf 1.2.8-1` to `1.2.9-1` and this has caused some latency of around 1/2 seconds for outputting audio via the `Focusrite Scarlett 2i2 3rd Gen` with is connected the computer via USB. No latency is introduced when plugging wired headphones directly in to the motherboard so this seems to be isolated to the device. I noticed the change logs indicate better support for this device was added in this version (which is great!) so these changes may have affected the existing settings of the headphones. I have downgraded only `alsa-ucm-conf` to `1.2.8-1` and the output has no latency so I am using this as a workaround. I use PulseAudio with ALSA on Arch Linux. Versions: ``` alsa-card-profiles - 0.3.71 alsa-lib - 1.2.9 alsa-ucm-conf - 1.2.9 alsa-utils - 1.2.9 pulseaudio - 16.1 Arch Linux - 6.3.9-arch1-1 ``` dmesg detects the product as `Scarlett 2i2 USB` Please let me know if any more information is needed (such as logs or config files) and if there is anything I should try. Many thanks :D Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/329 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2 00/45] Add support for sam9x7 SoC family
by Varshini Rajendran 24 Jun '23

24 Jun '23

This patch series adds support for the new SoC family - sam9x7. - The device tree, configs and drivers are added - Clock driver for sam9x7 is added - Support for basic peripherals is added - Target board SAM9X75 Curiosity is added Changes in v2: -------------- - Added sam9x7 specific compatibles in DT with fallbacks - Documented all the newly added DT compatible strings - Added device tree for the target board sam9x75 curiosity and documented the same in the DT bindings documentation - Removed the dt nodes that are not supported at the moment - Removed the configs added by previous version that are not supported at the moment - Fixed all the corrections in the commit message - Changed all the instances of copyright year to 2023 - Added sam9x7 flag in PIT64B configuration - Moved macro definitions to header file - Added another divider in mck characteristics in the pmc driver - Fixed the memory leak in the pmc driver - Dropped patches that are no longer needed - Picked up Acked-by and Reviewed-by tags Hari Prasath (1): irqchip/atmel-aic5: Add support for sam9x7 aic Varshini Rajendran (44): dt-bindings: microchip: atmel,at91rm9200-tcb: add sam9x60, sam9x7 compatible dt-bindings: usb: ehci: Add atmel at91sam9g45-ehci compatible dt-bindings: usb: generic-ehci: Document clock-names property dt-bindings: net: cdns,macb: add documentation for sam9x7 ethernet interface ARM: at91: pm: add support for sam9x7 SoC family ARM: at91: pm: add sam9x7 SoC init config ARM: at91: add support in SoC driver for new sam9x7 clk: at91: clk-sam9x60-pll: re-factor to support individual core freq outputs clk: at91: sam9x7: add support for HW PLL freq dividers clk: at91: sama7g5: move mux table macros to header file dt-bindings: clk: at91: add bindings for SAM9X7's clock controller dt-bindings: reset: atmel,at91sam9260-reset: add sam9x7 binding dt-bindings: atmel-sysreg: add bindings for sam9x7 dt-bindings: crypto: add bindings for sam9x7 in Atmel AES dt-bindings: crypto: add bindings for sam9x7 in Atmel SHA dt-bindings: crypto: add bindings for sam9x7 in Atmel TDES dt-bindings: dmaengine: at_xdmac: add compatible with microchip,sam9x7 dt-bindings: i2c: at91: Add SAM9X7 compatible string dt-bindings: mfd: at91: Add SAM9X7 compatible string dt-bindings: atmel-gpbr: add microchip,sam9x7-gpbr dt-bindings: atmel-matrix: add microchip,sam9x7-matrix dt-bindings: atmel-smc: add microchip,sam9x7-smc dt-bindings: atmel-ssc: add microchip,sam9x7-ssc dt-bindings: sdhci-of-at91: add microchip,sam9x7-sdhci dt-bindings: atmel-nand: add microchip,sam9x7-pmecc dt-bindings: pinctrl: at91: add bindings for SAM9X7 dt-bindings: rng: atmel,at91-trng: document sam9x7 TRNG dt-bindings: rtc: at91rm9200: add sam9x7 compatible dt-bindings: rtt: at91rm9260: add sam9x7 compatible dt-bindings: serial: atmel,at91-usart: add compatible for sam9x7 dt-bindings: atmel-classd: add sam9x7 compatible spi: dt-bindings: atmel,at91rm9200-spi: add sam9x7 compatible dt-bindings: usb: atmel: Update DT bindings documentation for sam9x7 dt-bindings: watchdog: sama5d4-wdt: add compatible for sam9x7-wdt dt-bindings: irqchip/atmel-aic5: Add support for sam9x7 aic clk: at91: sam9x7: add sam9x7 pmc driver power: reset: at91-poweroff: lookup for proper pmc dt node for sam9x7 power: reset: at91-reset: add reset support for sam9x7 SoC power: reset: at91-reset: add sdhwc support for sam9x7 SoC ARM: at91: Kconfig: add config flag for SAM9X7 SoC ARM: configs: at91: enable config flags for sam9x7 SoC family ARM: dts: at91: sam9x7: add device tree for SoC dt-bindings: arm: add sam9x75 curiosity board ARM: dts: at91: sam9x75_curiosity: add device tree for sam9x75 curiosity board .../devicetree/bindings/arm/atmel-at91.yaml | 6 + .../devicetree/bindings/arm/atmel-sysregs.txt | 7 +- .../devicetree/bindings/clock/at91-clock.txt | 7 +- .../crypto/atmel,at91sam9g46-aes.yaml | 5 +- .../crypto/atmel,at91sam9g46-sha.yaml | 5 +- .../crypto/atmel,at91sam9g46-tdes.yaml | 5 +- .../devicetree/bindings/dma/atmel-xdma.txt | 4 +- .../bindings/i2c/atmel,at91sam-i2c.yaml | 3 + .../interrupt-controller/atmel,aic.txt | 2 +- .../devicetree/bindings/mfd/atmel-flexcom.txt | 2 +- .../devicetree/bindings/mfd/atmel-gpbr.txt | 1 + .../devicetree/bindings/mfd/atmel-matrix.txt | 1 + .../devicetree/bindings/mfd/atmel-smc.txt | 1 + .../devicetree/bindings/misc/atmel-ssc.txt | 1 + .../devicetree/bindings/mmc/sdhci-atmel.txt | 4 +- .../devicetree/bindings/mtd/atmel-nand.txt | 1 + .../devicetree/bindings/net/cdns,macb.yaml | 1 + .../bindings/pinctrl/atmel,at91-pinctrl.txt | 3 +- .../reset/atmel,at91sam9260-reset.yaml | 1 + .../bindings/rng/atmel,at91-trng.yaml | 1 + .../bindings/rtc/atmel,at91rm9200-rtc.yaml | 1 + .../bindings/rtc/atmel,at91sam9260-rtt.yaml | 1 + .../bindings/serial/atmel,at91-usart.yaml | 3 + .../soc/microchip/atmel,at91rm9200-tcb.yaml | 2 + .../bindings/sound/atmel,sama5d2-classd.yaml | 5 +- .../bindings/spi/atmel,at91rm9200-spi.yaml | 1 + .../devicetree/bindings/usb/atmel-usb.txt | 9 +- .../devicetree/bindings/usb/generic-ehci.yaml | 5 + .../bindings/watchdog/atmel,sama5d4-wdt.yaml | 1 + arch/arm/boot/dts/Makefile | 2 + arch/arm/boot/dts/at91-sam9x75_curiosity.dts | 336 +++++ arch/arm/boot/dts/sam9x7.dtsi | 1237 +++++++++++++++++ arch/arm/configs/at91_dt_defconfig | 1 + arch/arm/mach-at91/Kconfig | 23 +- arch/arm/mach-at91/Makefile | 1 + arch/arm/mach-at91/generic.h | 2 + arch/arm/mach-at91/pm.c | 35 + arch/arm/mach-at91/sam9x7.c | 34 + drivers/clk/at91/Makefile | 1 + drivers/clk/at91/clk-sam9x60-pll.c | 50 +- drivers/clk/at91/pmc.h | 18 + drivers/clk/at91/sam9x60.c | 7 + drivers/clk/at91/sam9x7.c | 942 +++++++++++++ drivers/clk/at91/sama7g5.c | 42 +- drivers/irqchip/irq-atmel-aic5.c | 10 + drivers/power/reset/Kconfig | 4 +- drivers/power/reset/at91-sama5d2_shdwc.c | 1 + drivers/soc/atmel/soc.c | 23 + drivers/soc/atmel/soc.h | 9 + 49 files changed, 2806 insertions(+), 61 deletions(-) create mode 100644 arch/arm/boot/dts/at91-sam9x75_curiosity.dts create mode 100644 arch/arm/boot/dts/sam9x7.dtsi create mode 100644 arch/arm/mach-at91/sam9x7.c create mode 100644 drivers/clk/at91/sam9x7.c -- 2.25.1

4 82

[PATCH] ASoC: tas2781: Fix error code in tas2781_load_calibration()
by Dan Carpenter 24 Jun '23

24 Jun '23

Return -EINVAL instead of success on this error path. Fixes: 915f5eadebd2 ("ASoC: tas2781: firmware lib") Signed-off-by: Dan Carpenter <dan.carpenter(a)linaro.org> --- sound/soc/codecs/tas2781-fmwlib.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index cbf0aef2c001..eb55abae0d7b 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -1924,6 +1924,7 @@ int tas2781_load_calibration(void *context, char *file_name, if (!fw_entry->size) { dev_err(tas_priv->dev, "%s: file read error: size = %lu\n", __func__, (unsigned long)fw_entry->size); + ret = -EINVAL; goto out; } fmw.size = fw_entry->size; -- 2.39.2

2 1

Focusrite Scarlett 2i2 3rd Gen suffers from high output latency with the 1.2.9-1 update
by GitHub issues - edited 23 Jun '23

23 Jun '23

alsa-project/alsa-ucm-conf issue #329 was edited from IsaacLambat: Hello, I have recently upgraded from `alsa-ucm-conf 1.2.8-1` to `1.2.9-1` and this has caused some latency of around 1 second for outputting audio via the `Focusrite Scarlett 2i2 3rd Gen` with is connected the computer via USB. No latency is introduced when plugging wired headphones directly in to the motherboard so this seems to be isolated to the device. I noticed the change logs indicate better support for this device was added in this version (which is great!) so these changes may have affected the existing settings of the headphones. I have downgraded only `alsa-ucm-conf` to `1.2.8-1` and the output has no latency so I am using this as a workaround. I use PulseAudio with ALSA on Arch Linux. Versions: ``` alsa-card-profiles - 0.3.71 alsa-lib - 1.2.9 alsa-ucm-conf - 1.2.9 alsa-utils - 1.2.9 pulseaudio - 16.1 Arch Linux - 6.3.9-arch1-1 ``` dmesg detects the product as `Scarlett 2i2 USB` Please let me know if any more information is needed (such as logs or config files) and if there is anything I should try. Many thanks :D Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/329 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Focusrite Scarlett 2i2 3rd Gen suffers from high output latency with the 1.2.9-1 update
by GitHub issues - opened 23 Jun '23

23 Jun '23

alsa-project/alsa-ucm-conf issue #329 was opened from IsaacLambat: Hello, I have recently upgraded from `alsa-ucm-conf 1.2.8-1` to `1.2.9-1` and this has caused some latency of around 1 second for outputting audio via the `Focusrite Scarlett 2i2 3rd Gen` with is connected the computer via USB. No latency is introduced when plugging wired headphones directly in to the motherboard so this seems to be isolated to the device. I noticed the change logs indicate better support for this device was added in this version (which is great!) so these changes may have affected the existing settings of the headphones. I have downgraded only `alsa-ucm-conf` to `1.2.8-1` and the output has no latency so I am using this as a workaround. I use PulseAudio with ALSA on Arch Linux. Versions: ``` alsa-card-profiles - 0.3.71 alsa-lib - 1.2.9 alsa-ucm-conf - 1.2.9 alsa-utils - 1.2.9 pulseaudio - 16.1 Arch Linux - 6.3.9-arch1-1 ``` Please let me know if any more information is needed (such as logs or config files) and if there is anything I should try. Many thanks :D Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/329 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 01/12] ASoC: amd: acp: clear pdm dma interrupt mask
by Syed Saba Kareem 23 Jun '23

23 Jun '23

Clear pdm dma interrupt mask in acp_dmic_shutdown(). 'Fixes: c32bd332ce5c9 ("ASoC: amd: acp: Add generic support for PDM controller on ACP")' Signed-off-by: Syed Saba Kareem <Syed.SabaKareem(a)amd.com> --- sound/soc/amd/acp/acp-pdm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp/acp-pdm.c b/sound/soc/amd/acp/acp-pdm.c index 66ec6b6a5972..f8030b79ac17 100644 --- a/sound/soc/amd/acp/acp-pdm.c +++ b/sound/soc/amd/acp/acp-pdm.c @@ -176,7 +176,7 @@ static void acp_dmic_dai_shutdown(struct snd_pcm_substream *substream, /* Disable DMIC interrupts */ ext_int_ctrl = readl(ACP_EXTERNAL_INTR_CNTL(adata, 0)); - ext_int_ctrl |= ~PDM_DMA_INTR_MASK; + ext_int_ctrl &= ~PDM_DMA_INTR_MASK; writel(ext_int_ctrl, ACP_EXTERNAL_INTR_CNTL(adata, 0)); } -- 2.25.1

4 27

[PATCH] ALSA: ump: Export MIDI1 / UMP conversion helpers
by Takashi Iwai 23 Jun '23

23 Jun '23

Yet more preliminary work for the upcoming USB gadget support. Now export the helpers to convert between legacy MIDI1 and UMP data for handling the MIDI 1.0 USB interface. The header file is moved to include/sound. The API functions are slightly changed, so that they can be used without the direct access to snd_ump object. The allocation is done in ump.c itself as it's a simple kcalloc(). Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/ump_convert.h | 46 +++++++++++++++++++++ sound/core/ump.c | 18 ++++----- sound/core/ump_convert.c | 80 ++++++++++++++++--------------------- sound/core/ump_convert.h | 43 -------------------- 4 files changed, 89 insertions(+), 98 deletions(-) create mode 100644 include/sound/ump_convert.h delete mode 100644 sound/core/ump_convert.h diff --git a/include/sound/ump_convert.h b/include/sound/ump_convert.h new file mode 100644 index 000000000000..28c364c63245 --- /dev/null +++ b/include/sound/ump_convert.h @@ -0,0 +1,46 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +#ifndef __SOUND_UMP_CONVERT_H +#define __SOUND_UMP_CONVERT_H + +#include <sound/ump_msg.h> + +/* context for converting from legacy control messages to UMP packet */ +struct ump_cvt_to_ump_bank { + bool rpn_set; + bool nrpn_set; + bool bank_set; + unsigned char cc_rpn_msb, cc_rpn_lsb; + unsigned char cc_nrpn_msb, cc_nrpn_lsb; + unsigned char cc_data_msb, cc_data_lsb; + unsigned char cc_bank_msb, cc_bank_lsb; +}; + +/* context for converting from MIDI1 byte stream to UMP packet */ +struct ump_cvt_to_ump { + /* MIDI1 intermediate buffer */ + unsigned char buf[4]; + int len; + int cmd_bytes; + + /* UMP output packet */ + u32 ump[4]; + int ump_bytes; + + /* various status */ + unsigned int in_sysex; + struct ump_cvt_to_ump_bank bank[16]; /* per channel */ +}; + +int snd_ump_convert_from_ump(const u32 *data, unsigned char *dst, + unsigned char *group_ret); +void snd_ump_convert_to_ump(struct ump_cvt_to_ump *cvt, unsigned char group, + unsigned int protocol, unsigned char c); + +/* reset the converter context, called at each open to ump */ +static inline void snd_ump_convert_reset(struct ump_cvt_to_ump *ctx) +{ + memset(ctx, 0, sizeof(*ctx)); + +} + +#endif /* __SOUND_UMP_CONVERT_H */ diff --git a/sound/core/ump.c b/sound/core/ump.c index 5e17351ca984..246348766ec1 100644 --- a/sound/core/ump.c +++ b/sound/core/ump.c @@ -11,7 +11,7 @@ #include <sound/core.h> #include <sound/rawmidi.h> #include <sound/ump.h> -#include "ump_convert.h" +#include <sound/ump_convert.h> #define ump_err(ump, fmt, args...) dev_err(&(ump)->core.dev, fmt, ##args) #define ump_warn(ump, fmt, args...) dev_warn(&(ump)->core.dev, fmt, ##args) @@ -87,7 +87,7 @@ static void snd_ump_endpoint_free(struct snd_rawmidi *rmidi) ump->private_free(ump); #if IS_ENABLED(CONFIG_SND_UMP_LEGACY_RAWMIDI) - snd_ump_convert_free(ump); + kfree(ump->out_cvts); #endif } @@ -1002,7 +1002,7 @@ static int snd_ump_legacy_open(struct snd_rawmidi_substream *substream) goto unlock; } ump->legacy_out_opens++; - snd_ump_reset_convert_to_ump(ump, group); + snd_ump_convert_reset(&ump->out_cvts[group]); } spin_lock_irq(&ump->legacy_locks[dir]); ump->legacy_substreams[dir][group] = substream; @@ -1091,7 +1091,7 @@ static int process_legacy_output(struct snd_ump_endpoint *ump, ctx = &ump->out_cvts[group]; while (!ctx->ump_bytes && snd_rawmidi_transmit(substream, &c, 1) > 0) - snd_ump_convert_to_ump(ump, group, c); + snd_ump_convert_to_ump(ctx, group, ump->info.protocol, c); if (ctx->ump_bytes && ctx->ump_bytes <= count) { size = ctx->ump_bytes; memcpy(buffer, ctx->ump, size); @@ -1113,7 +1113,7 @@ static void process_legacy_input(struct snd_ump_endpoint *ump, const u32 *src, const int dir = SNDRV_RAWMIDI_STREAM_INPUT; int size; - size = snd_ump_convert_from_ump(ump, src, buf, &group); + size = snd_ump_convert_from_ump(src, buf, &group); if (size <= 0) return; spin_lock_irqsave(&ump->legacy_locks[dir], flags); @@ -1130,9 +1130,9 @@ int snd_ump_attach_legacy_rawmidi(struct snd_ump_endpoint *ump, bool input, output; int err; - err = snd_ump_convert_init(ump); - if (err < 0) - return err; + ump->out_cvts = kcalloc(16, sizeof(*ump->out_cvts), GFP_KERNEL); + if (!ump->out_cvts) + return -ENOMEM; input = ump->core.info_flags & SNDRV_RAWMIDI_INFO_INPUT; output = ump->core.info_flags & SNDRV_RAWMIDI_INFO_OUTPUT; @@ -1140,7 +1140,7 @@ int snd_ump_attach_legacy_rawmidi(struct snd_ump_endpoint *ump, output ? 16 : 0, input ? 16 : 0, &rmidi); if (err < 0) { - snd_ump_convert_free(ump); + kfree(ump->out_cvts); return err; } diff --git a/sound/core/ump_convert.c b/sound/core/ump_convert.c index 48ab3e1bd62e..fb61df424a87 100644 --- a/sound/core/ump_convert.c +++ b/sound/core/ump_convert.c @@ -8,7 +8,7 @@ #include <sound/core.h> #include <sound/asound.h> #include <sound/ump.h> -#include "ump_convert.h" +#include <sound/ump_convert.h> /* * Upgrade / downgrade value bits @@ -205,12 +205,18 @@ static int cvt_ump_sysex7_to_legacy(const u32 *data, unsigned char *buf) return size; } -/* convert from a UMP packet @data to MIDI 1.0 bytes at @buf; - * the target group is stored at @group_ret, - * returns the number of bytes of MIDI 1.0 stream +/** + * snd_ump_convert_from_ump - convert from UMP to legacy MIDI + * @data: UMP packet + * @buf: buffer to store legacy MIDI data + * @group_ret: pointer to store the target group + * + * Convert from a UMP packet @data to MIDI 1.0 bytes at @buf. + * The target group is stored at @group_ret. + * + * The function returns the number of bytes of MIDI 1.0 stream. */ -int snd_ump_convert_from_ump(struct snd_ump_endpoint *ump, - const u32 *data, +int snd_ump_convert_from_ump(const u32 *data, unsigned char *buf, unsigned char *group_ret) { @@ -230,6 +236,7 @@ int snd_ump_convert_from_ump(struct snd_ump_endpoint *ump, return 0; } +EXPORT_SYMBOL_GPL(snd_ump_convert_from_ump); /* * MIDI 1 byte stream -> UMP conversion @@ -302,10 +309,10 @@ static void fill_rpn(struct ump_cvt_to_ump_bank *cc, } /* convert to a MIDI 1.0 Channel Voice message */ -static int cvt_legacy_cmd_to_ump(struct snd_ump_endpoint *ump, - struct ump_cvt_to_ump *cvt, - unsigned char group, u32 *data, - unsigned char bytes) +static int cvt_legacy_cmd_to_ump(struct ump_cvt_to_ump *cvt, + unsigned char group, + unsigned int protocol, + u32 *data, unsigned char bytes) { const unsigned char *buf = cvt->buf; struct ump_cvt_to_ump_bank *cc; @@ -316,7 +323,7 @@ static int cvt_legacy_cmd_to_ump(struct snd_ump_endpoint *ump, BUILD_BUG_ON(sizeof(union snd_ump_midi2_msg) != 8); /* for MIDI 1.0 UMP, it's easy, just pack it into UMP */ - if (ump->info.protocol & SNDRV_UMP_EP_INFO_PROTO_MIDI1) { + if (protocol & SNDRV_UMP_EP_INFO_PROTO_MIDI1) { data[0] = ump_compose(UMP_MSG_TYPE_MIDI1_CHANNEL_VOICE, group, 0, buf[0]); data[0] |= buf[1] << 8; @@ -413,8 +420,8 @@ static int cvt_legacy_cmd_to_ump(struct snd_ump_endpoint *ump, return 8; } -static int do_convert_to_ump(struct snd_ump_endpoint *ump, - unsigned char group, unsigned char c, u32 *data) +static int do_convert_to_ump(struct ump_cvt_to_ump *cvt, unsigned char group, + unsigned int protocol, unsigned char c, u32 *data) { /* bytes for 0x80-0xf0 */ static unsigned char cmd_bytes[8] = { @@ -424,7 +431,6 @@ static int do_convert_to_ump(struct snd_ump_endpoint *ump, static unsigned char system_bytes[16] = { 0, 2, 3, 2, 0, 0, 1, 0, 1, 1, 1, 1, 0, 0, 1, 1 }; - struct ump_cvt_to_ump *cvt = &ump->out_cvts[group]; unsigned char bytes; if (c == UMP_MIDI1_MSG_SYSEX_START) { @@ -478,40 +484,22 @@ static int do_convert_to_ump(struct snd_ump_endpoint *ump, cvt->len = 1; if ((cvt->buf[0] & 0xf0) == UMP_MIDI1_MSG_REALTIME) return cvt_legacy_system_to_ump(cvt, group, data); - return cvt_legacy_cmd_to_ump(ump, cvt, group, data, cvt->cmd_bytes); + return cvt_legacy_cmd_to_ump(cvt, group, protocol, data, cvt->cmd_bytes); } -/* feed a MIDI 1.0 byte @c and convert to a UMP packet; - * the target group is @group, - * the result is stored in out_cvts[group].ump[] and out_cvts[group].ump_bytes +/** + * snd_ump_convert_to_ump - convert legacy MIDI byte to UMP packet + * @cvt: converter context + * @group: target UMP group + * @protocol: target UMP protocol + * @c: MIDI 1.0 byte data + * + * Feed a MIDI 1.0 byte @c and convert to a UMP packet if completed. + * The result is stored in the buffer in @cvt. */ -void snd_ump_convert_to_ump(struct snd_ump_endpoint *ump, - unsigned char group, unsigned char c) +void snd_ump_convert_to_ump(struct ump_cvt_to_ump *cvt, unsigned char group, + unsigned int protocol, unsigned char c) { - struct ump_cvt_to_ump *cvt = &ump->out_cvts[group]; - - cvt->ump_bytes = do_convert_to_ump(ump, group, c, cvt->ump); -} - -/* reset the converter context, called at each open */ -void snd_ump_reset_convert_to_ump(struct snd_ump_endpoint *ump, - unsigned char group) -{ - memset(&ump->out_cvts[group], 0, sizeof(*ump->out_cvts)); -} - -/* initialize converters */ -int snd_ump_convert_init(struct snd_ump_endpoint *ump) -{ - ump->out_cvts = kcalloc(16, sizeof(*ump->out_cvts), GFP_KERNEL); - if (!ump->out_cvts) - return -ENOMEM; - return 0; -} - -/* release resources */ -void snd_ump_convert_free(struct snd_ump_endpoint *ump) -{ - kfree(ump->out_cvts); - ump->out_cvts = NULL; + cvt->ump_bytes = do_convert_to_ump(cvt, group, protocol, c, cvt->ump); } +EXPORT_SYMBOL_GPL(snd_ump_convert_to_ump); diff --git a/sound/core/ump_convert.h b/sound/core/ump_convert.h deleted file mode 100644 index bbfe96084779..000000000000 --- a/sound/core/ump_convert.h +++ /dev/null @@ -1,43 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-or-later -#ifndef __UMP_CONVERT_H -#define __UMP_CONVERT_H - -#include <sound/ump_msg.h> - -/* context for converting from legacy control messages to UMP packet */ -struct ump_cvt_to_ump_bank { - bool rpn_set; - bool nrpn_set; - bool bank_set; - unsigned char cc_rpn_msb, cc_rpn_lsb; - unsigned char cc_nrpn_msb, cc_nrpn_lsb; - unsigned char cc_data_msb, cc_data_lsb; - unsigned char cc_bank_msb, cc_bank_lsb; -}; - -/* context for converting from MIDI1 byte stream to UMP packet */ -struct ump_cvt_to_ump { - /* MIDI1 intermediate buffer */ - unsigned char buf[4]; - int len; - int cmd_bytes; - - /* UMP output packet */ - u32 ump[4]; - int ump_bytes; - - /* various status */ - unsigned int in_sysex; - struct ump_cvt_to_ump_bank bank[16]; /* per channel */ -}; - -int snd_ump_convert_init(struct snd_ump_endpoint *ump); -void snd_ump_convert_free(struct snd_ump_endpoint *ump); -int snd_ump_convert_from_ump(struct snd_ump_endpoint *ump, - const u32 *data, unsigned char *dst, - unsigned char *group_ret); -void snd_ump_convert_to_ump(struct snd_ump_endpoint *ump, - unsigned char group, unsigned char c); -void snd_ump_reset_convert_to_ump(struct snd_ump_endpoint *ump, - unsigned char group); -#endif /* __UMP_CONVERT_H */ -- 2.35.3

1 0

[bug report] ASoC: tas2781: Add tas2781 driver
by Dan Carpenter 23 Jun '23

23 Jun '23

Hello Shenghao Ding, The patch ef3bcde75d06: "ASoC: tas2781: Add tas2781 driver" from Jun 18, 2023, leads to the following Smatch static checker warning: sound/soc/codecs/tas2781-i2c.c:651 tasdevice_parse_dt() warn: assigning signed to unsigned: 'tas_priv->ndev = ndev' 's32min-s32max' sound/soc/codecs/tas2781-i2c.c 602 static void tasdevice_parse_dt(struct tasdevice_priv *tas_priv) 603 { 604 struct i2c_client *client = (struct i2c_client *)tas_priv->client; 605 unsigned int dev_addrs[TASDEVICE_MAX_CHANNELS]; 606 int rc, i, ndev = 0; 607 608 if (tas_priv->isacpi) { 609 ndev = device_property_read_u32_array(&client->dev, 610 "ti,audio-slots", NULL, 0); When we pass NULL to device_property_read_u32_array() then it returns then number of items. 611 if (ndev <= 0) { 612 ndev = 1; 613 dev_addrs[0] = client->addr; 614 } else { 615 ndev = (ndev < ARRAY_SIZE(dev_addrs)) 616 ? ndev : ARRAY_SIZE(dev_addrs); 617 ndev = device_property_read_u32_array(&client->dev, ^^^^^^^ Smatch is concerned that this value can be negative. But actually this sets it to zero, doesn't it? Is that intentional? It feels like we should leave ndev as the number of items. Or if we want ndev to be zero do "ndev = 0;" on the next line. 618 "ti,audio-slots", dev_addrs, ndev); 619 } 620 621 tas_priv->irq_info.irq_gpio = 622 acpi_dev_gpio_irq_get(ACPI_COMPANION(&client->dev), 0); 623 } else { 624 struct device_node *np = tas_priv->dev->of_node; 625 #ifdef CONFIG_OF 626 const __be32 *reg, *reg_end; 627 int len, sw, aw; 628 629 aw = of_n_addr_cells(np); 630 sw = of_n_size_cells(np); 631 if (sw == 0) { 632 reg = (const __be32 *)of_get_property(np, 633 "reg", &len); 634 reg_end = reg + len/sizeof(*reg); 635 ndev = 0; 636 do { 637 dev_addrs[ndev] = of_read_number(reg, aw); 638 reg += aw; 639 ndev++; 640 } while (reg < reg_end); 641 } else { 642 ndev = 1; 643 dev_addrs[0] = client->addr; 644 } 645 #else 646 ndev = 1; 647 dev_addrs[0] = client->addr; 648 #endif 649 tas_priv->irq_info.irq_gpio = of_irq_get(np, 0); 650 } --> 651 tas_priv->ndev = ndev; ^^^^^^^^^^^^^^^^^^^^^ Warning 652 for (i = 0; i < ndev; i++) 653 tas_priv->tasdevice[i].dev_addr = dev_addrs[i]; 654 655 tas_priv->reset = devm_gpiod_get_optional(&client->dev, 656 "reset-gpios", GPIOD_OUT_HIGH); regards, dan carpenter

2 1

[PATCH v2 1/1] ASoC: codecs: max98090: Allow dsp_a mode
by Maxim Kochetkov 23 Jun '23

23 Jun '23

TDM mode for max98090 is dsp_a compatible with such limitations: 1) Up to four timeslots supported. 2) Only 16 bits timeslots supported. 3) Only 2 active timeslots (L/R) supported. We want to setup TDM mode only when dsp_a mode is selected. So move M98090_REG_TDM_FORMAT/M98090_REG_TDM_CONTROL registers setup from max98090_set_tdm_slot() to the max98090_dai_set_fmt(). Also extend max98090_set_tdm_slot() with all TDM limitations check. Signed-off-by: Maxim Kochetkov <fido_max(a)inbox.ru> --- sound/soc/codecs/max98090.c | 53 ++++++++++++++++++++----------------- sound/soc/codecs/max98090.h | 3 ++- 2 files changed, 30 insertions(+), 26 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 7bc463910d4f..2adf744c6526 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1581,7 +1581,7 @@ static int max98090_dai_set_fmt(struct snd_soc_dai *codec_dai, struct snd_soc_component *component = codec_dai->component; struct max98090_priv *max98090 = snd_soc_component_get_drvdata(component); struct max98090_cdata *cdata; - u8 regval; + u8 regval, tdm_regval; max98090->dai_fmt = fmt; cdata = &max98090->dai[0]; @@ -1590,6 +1590,7 @@ static int max98090_dai_set_fmt(struct snd_soc_dai *codec_dai, cdata->fmt = fmt; regval = 0; + tdm_regval = 0; switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { case SND_SOC_DAIFMT_CBC_CFC: /* Set to consumer mode PLL - MAS mode off */ @@ -1635,7 +1636,8 @@ static int max98090_dai_set_fmt(struct snd_soc_dai *codec_dai, regval |= M98090_RJ_MASK; break; case SND_SOC_DAIFMT_DSP_A: - /* Not supported mode */ + tdm_regval |= M98090_TDM_MASK; + break; default: dev_err(component->dev, "DAI format unsupported"); return -EINVAL; @@ -1664,11 +1666,20 @@ static int max98090_dai_set_fmt(struct snd_soc_dai *codec_dai, * seen for the case of TDM mode. The remaining cases have * normal logic. */ - if (max98090->tdm_slots > 1) + if (tdm_regval) regval ^= M98090_BCI_MASK; snd_soc_component_write(component, M98090_REG_INTERFACE_FORMAT, regval); + + regval = 0; + if (tdm_regval) + regval = max98090->tdm_lslot << M98090_TDM_SLOTL_SHIFT | + max98090->tdm_rslot << M98090_TDM_SLOTR_SHIFT | + 0 << M98090_TDM_SLOTDLY_SHIFT; + + snd_soc_component_write(component, M98090_REG_TDM_FORMAT, regval); + snd_soc_component_write(component, M98090_REG_TDM_CONTROL, tdm_regval); } return 0; @@ -1679,33 +1690,22 @@ static int max98090_set_tdm_slot(struct snd_soc_dai *codec_dai, { struct snd_soc_component *component = codec_dai->component; struct max98090_priv *max98090 = snd_soc_component_get_drvdata(component); - struct max98090_cdata *cdata; - cdata = &max98090->dai[0]; if (slots < 0 || slots > 4) return -EINVAL; - max98090->tdm_slots = slots; - max98090->tdm_width = slot_width; - - if (max98090->tdm_slots > 1) { - /* SLOTL SLOTR SLOTDLY */ - snd_soc_component_write(component, M98090_REG_TDM_FORMAT, - 0 << M98090_TDM_SLOTL_SHIFT | - 1 << M98090_TDM_SLOTR_SHIFT | - 0 << M98090_TDM_SLOTDLY_SHIFT); - - /* FSW TDM */ - snd_soc_component_update_bits(component, M98090_REG_TDM_CONTROL, - M98090_TDM_MASK, - M98090_TDM_MASK); - } + if (slot_width != 16) + return -EINVAL; - /* - * Normally advisable to set TDM first, but this permits either order - */ - cdata->fmt = 0; - max98090_dai_set_fmt(codec_dai, max98090->dai_fmt); + if (rx_mask != tx_mask) + return -EINVAL; + + if (!rx_mask) + return -EINVAL; + + max98090->tdm_slots = slots; + max98090->tdm_lslot = ffs(rx_mask) - 1; + max98090->tdm_rslot = fls(rx_mask) - 1; return 0; } @@ -2408,6 +2408,9 @@ static int max98090_probe(struct snd_soc_component *component) max98090->pa1en = 0; max98090->pa2en = 0; + max98090->tdm_lslot = 0; + max98090->tdm_rslot = 1; + ret = snd_soc_component_read(component, M98090_REG_REVISION_ID); if (ret < 0) { dev_err(component->dev, "Failed to read device revision: %d\n", diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h index a197114b0dad..6ce8dd176e48 100644 --- a/sound/soc/codecs/max98090.h +++ b/sound/soc/codecs/max98090.h @@ -1533,7 +1533,8 @@ struct max98090_priv { struct snd_soc_jack *jack; unsigned int dai_fmt; int tdm_slots; - int tdm_width; + int tdm_lslot; + int tdm_rslot; u8 lin_state; unsigned int pa1en; unsigned int pa2en; -- 2.40.1

2 1

[PATCH] ASoC: tas2781: fix Kconfig dependencies
by Arnd Bergmann 23 Jun '23

23 Jun '23

From: Arnd Bergmann <arnd(a)arndb.de> The new driver has two modules that both get enabled for build testing when all codecs are selected. The comlib part has an i2c dependency, so this remains disabled on builds without i2c, but then the other one fails to link: ERROR: modpost: "tasdevice_dev_bulk_write" [sound/soc/codecs/snd-soc-tas2781-fmwlib.ko] undefined! ERROR: modpost: "tasdevice_dev_update_bits" [sound/soc/codecs/snd-soc-tas2781-fmwlib.ko] undefined! ERROR: modpost: "tasdevice_dev_bulk_read" [sound/soc/codecs/snd-soc-tas2781-fmwlib.ko] undefined! ERROR: modpost: "tasdevice_dev_read" [sound/soc/codecs/snd-soc-tas2781-fmwlib.ko] undefined! ERROR: modpost: "tasdevice_dev_write" [sound/soc/codecs/snd-soc-tas2781-fmwlib.ko] undefined! There are many ways to address this, adding an explicit dependency seems to be the clearest method that keeps the structure of the driver otherwise unchanged. Fixes: ef3bcde75d06d ("ASoC: tas2781: Add tas2781 driver") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f481f02a35710..2a62dbd5339e4 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1741,6 +1741,7 @@ config SND_SOC_TAS2781_COMLIB tristate config SND_SOC_TAS2781_FMWLIB + depends on SND_SOC_TAS2781_COMLIB tristate default n -- 2.39.2

2 1

Re: [PATCH 6/7] soundwire: intel: remove unnecessary init
by K Smart 23 Jun '23

23 Jun '23

REMOVE NOW

1 0

[PATCH] ASoC: qcom: common: add default jack dapm pins
by Srinivas Kandagatla 23 Jun '23

23 Jun '23

If the soundcard does not specify the dapm pins, let the common code add these pins for jack. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/common.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c index c1f24af17506..555feb845c41 100644 --- a/sound/soc/qcom/common.c +++ b/sound/soc/qcom/common.c @@ -8,6 +8,11 @@ #include "qdsp6/q6afe.h" #include "common.h" +static const struct snd_soc_dapm_widget qcom_jack_snd_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), +}; + int qcom_snd_parse_of(struct snd_soc_card *card) { struct device_node *np; @@ -169,6 +174,11 @@ int qcom_snd_parse_of(struct snd_soc_card *card) of_node_put(platform); } + if (!card->dapm_widgets) { + card->dapm_widgets = qcom_jack_snd_widgets; + card->num_dapm_widgets = ARRAY_SIZE(qcom_jack_snd_widgets); + } + return 0; err: of_node_put(cpu); -- 2.21.0

3 2

[PATCH] ASoC: dt-bindings: microchip,sama7g5-pdmc: Simplify "microchip,mic-pos" constraints
by Rob Herring 23 Jun '23

23 Jun '23

"enum" values should be integers or strings, not arrays (though json-schema does allow arrays, we do not). In this case, all possible combinations are allowed anyways, so there's little point in expressing as an array. Signed-off-by: Rob Herring <robh(a)kernel.org> --- .../devicetree/bindings/sound/microchip,sama7g5-pdmc.yaml | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/microchip,sama7g5-pdmc.yaml b/Documentation/devicetree/bindings/sound/microchip,sama7g5-pdmc.yaml index 9b40268537cb..9aa65c975c4e 100644 --- a/Documentation/devicetree/bindings/sound/microchip,sama7g5-pdmc.yaml +++ b/Documentation/devicetree/bindings/sound/microchip,sama7g5-pdmc.yaml @@ -56,13 +56,9 @@ properties: items: items: - description: value for DS line + enum: [0, 1] - description: value for sampling edge - anyOf: - - enum: - - [0, 0] - - [0, 1] - - [1, 0] - - [1, 1] + enum: [0, 1] minItems: 1 maxItems: 4 uniqueItems: true -- 2.40.1

4 3

[PATCH 1/2] ASoC: fsl-asoc-card: add nau8822 support
by Hui Wang 23 Jun '23

23 Jun '23

This is for an imx6sx EVB which has a nau8822 codec connects to the SSI2 interface, so add the nau8822 support in this machine driver. Because the codec driver nau8822.c doesn't handle mclk enabling, here adding a codec_priv->mclk for nau8822 and similar codecs which need to enable the mclk in the machine driver, and enable the mclk in the card_late_probe() conditionally. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/soc/fsl/fsl-asoc-card.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index d9a0d4768c4d..89504865cb1d 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -27,6 +27,7 @@ #include "../codecs/wm8960.h" #include "../codecs/wm8994.h" #include "../codecs/tlv320aic31xx.h" +#include "../codecs/nau8822.h" #define CS427x_SYSCLK_MCLK 0 @@ -45,6 +46,7 @@ * @pll_id: PLL id for set_pll() */ struct codec_priv { + struct clk *mclk; unsigned long mclk_freq; unsigned long free_freq; u32 mclk_id; @@ -522,6 +524,9 @@ static int fsl_asoc_card_late_probe(struct snd_soc_card *card) return ret; } + if (!IS_ERR_OR_NULL(codec_priv->mclk)) + clk_prepare_enable(codec_priv->mclk); + return 0; } @@ -682,6 +687,14 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->codec_priv.free_freq = priv->codec_priv.mclk_freq; priv->card.dapm_routes = NULL; priv->card.num_dapm_routes = 0; + } else if (of_device_is_compatible(np, "fsl,imx-audio-nau8822")) { + codec_dai_name = "nau8822-hifi"; + priv->codec_priv.mclk_id = NAU8822_CLK_MCLK; + priv->codec_priv.fll_id = NAU8822_CLK_PLL; + priv->codec_priv.pll_id = NAU8822_CLK_PLL; + priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + if (codec_dev) + priv->codec_priv.mclk = devm_clk_get(codec_dev, NULL); } else { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); ret = -EINVAL; @@ -907,6 +920,7 @@ static const struct of_device_id fsl_asoc_card_dt_ids[] = { { .compatible = "fsl,imx-audio-wm8524", }, { .compatible = "fsl,imx-audio-si476x", }, { .compatible = "fsl,imx-audio-wm8958", }, + { .compatible = "fsl,imx-audio-nau8822", }, {} }; MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids); -- 2.25.1

2 2

[PATCH] ASoC: max98388: set variable soc_codec_dev_max98388 storage-class-specifier to static
by Tom Rix 23 Jun '23

23 Jun '23

smatch reports sound/soc/codecs/max98388.c:890:39: warning: symbol 'soc_codec_dev_max98388' was not declared. Should it be static? This variable is only used in its defining file, so it should be static. Signed-off-by: Tom Rix <trix(a)redhat.com> --- sound/soc/codecs/max98388.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98388.c b/sound/soc/codecs/max98388.c index 8062a7115007..b7d1d5c7b72f 100644 --- a/sound/soc/codecs/max98388.c +++ b/sound/soc/codecs/max98388.c @@ -887,7 +887,7 @@ static const struct regmap_config max98388_regmap = { .cache_type = REGCACHE_RBTREE, }; -const struct snd_soc_component_driver soc_codec_dev_max98388 = { +static const struct snd_soc_component_driver soc_codec_dev_max98388 = { .probe = max98388_probe, .controls = max98388_snd_controls, .num_controls = ARRAY_SIZE(max98388_snd_controls), -- 2.27.0

2 1

Intel/sof-essx8336: Fix HiFi.conf
by GitHub pull_request - opened 23 Jun '23

23 Jun '23

alsa-project/alsa-ucm-conf pull request #328 was opened from xry111: 1. Enable Left/Right Headphone Mixer Left/Right DAC Switch in BootSequence. Despite they seem enabled in sof-essx8336.conf, if I don't add this I'll need to enable them with alsamixer. I guess they are killed by disdevall. 2. Switch Headphone on/off in Headphones enable/disable sequence. This should be obvious. 3. "Headset Switch" is not recognized and triggers errors running "alsaucm -c hw:0 set _verb Hifi". Change it to "Headset Mic Switch". And in the disable sequence we should turn it off, not on. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/328 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/328.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v5 00/13] Add support for IIO devices in ASoC
by Herve Codina 22 Jun '23

22 Jun '23

Several weeks ago, I sent a series [1] for adding a potentiometer as an auxiliary device in ASoC. The feedback was that the potentiometer should be directly handled in IIO (as other potentiometers) and something more generic should be present in ASoC in order to have a binding to import some IIO devices into sound cards. The series related to the IIO potentiometer device is already applied. This series introduces audio-iio-aux. Its goal is to offer the binding between IIO and ASoC. It exposes attached IIO devices as ASoC auxiliary devices and allows to control them through mixer controls. On my system, the IIO device is a potentiometer and it is present in an amplifier design present in the audio path. Compare to the previous iteration https://lore.kernel.org/linux-kernel/20230614074904.29085-1-herve.codina@bo… This v5 series mainly: - Fixes {min,max}_array macros Best regards, Hervé [1] https://lore.kernel.org/linux-kernel/20230203111422.142479-1-herve.codina@b… [2] https://lore.kernel.org/linux-kernel/20230421085245.302169-1-herve.codina@b… Changes v4 -> v5 - Patches 1, 2, 3, 4, 5, 9, 10, 11, 12, 13 No changes. - Patch 6 Fix commit log. Add 'Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com>' - Patch 7 Fix the macros to be able to use them with: - an array defined as int *buff; - an array defined as int buff[N]; - Rework the way to "unconstify" the temporary variable to avoid issues due to integer promotion. Add 'Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com>' - Patch 8 Add 'Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com>' Changes v3 -> v4 - Patches 1, 2 No changes. - Patches 3, 4, 5 Add 'Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com>'. - Patch 6 (new in v4) Fix headers inclusion order. - Patch 7 (patch 6 in v3) Add a comment related to __must_be_array() Use __array[0] of *__array - Patch 8 (patch 7 in v3) Fix minmax.h inclusion order. Add 'Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com>'. - Patch 9 (patch 8 in v3) Add 'Suggested-by: Jonathan Cameron <Jonathan.Cameron(a)huawei.com>'. Add 'Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com>'. - Patch 10 (patch 9 in v3) Add 'Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com>'. - Patch 11 (patch 10 in v3) Fix a typo. Add 'Suggested-by: Andy Shevchenko <andy.shevchenko(a)gmail.com>'. Add 'Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com>'. - Patch 12 (patch 11 in v3) Fix typos in the commit log. Fix headers inclusion order. Removed unneeded variable initialization. Replace {0} by {}. Use struct device *dev in probe(). Check an error on the snd-control-invert-range property read. - Patch 13 (patch12 in v3) No changes. Changes v2 -> v3 - Patches 1, 2 No changes. - Patch 3, 4 Add 'Acked-by: Jonathan Cameron <Jonathan.Cameron(a)huawei.com>'. - Patch 5 (new in v3) Removed the 'unused' variable and check the null pointer when used. - Patch 6 (new in v3) Introduce {min,max}_array(). - Patch 7 (new in v3) Use max_array() in iio_channel_read_max(). - Patch 8 (new in v3) Replace a FIXME comment by a TODO one. - Patch 9 (patch 5 in v2) Removed the 'unused' variable and check the null pointer when used. Use min_array(). Remplace a FIXME comment by a TODO one. - Patch 10 (patch 6 in v2) Convert existing macros to return a compound litteral instead of adding a new helper. - Patch 11 (patch 7 in v2) Remove the file name from the C file header. Use directly converted DAPM macros. Replace <linux/module.h> by <linux/mod_devicetable.h>. Add <linux/platform_device.h>. Be sure that min <= max. Swap values if it is not the case. Move the bool structure member after the int ones. Remove unneeded assignements. Use dev_err_probe() when relevant. Use str_on_off(). Use static_assert() instead of BUILD_BUG_ON(). Remove unneeded comma and blank line. Use device_property_*() instead of the OF API. - patch 8 available in v2 removed as already applied - Patch 12 (patch 9 in v2) Use devm_add_action_or_reset(). Call simple_populate_aux() from simple_parse_of(). Changes v1 -> v2 - Patch 1 Rename simple-iio-aux to audio-iio-aux Rename invert to snd-control-invert-range Remove the /schemas/iio/iio-consumer.yaml reference Remove the unneeded '|' after description - Patch 2 (new in v2) Introduce the simple-audio-card additional-devs subnode - Patch 3 (new in v2) Check err before switch() in iio_channel_read_max() - Patch 4 (new in v2) Fix raw reads and raw writes documentation - Patch 5 (patch 2 in v1) Check err before switch() in iio_channel_read_min() Fix documentation - Patch 6 (path 3 in v1) No changes - Patch 7 (patch 4 in v1) Rename simple-iio-aux to audio-iio-aux Rename invert to snd-control-invert-range Remove the mask usage from audio_iio_aux_{get,put}_volsw helpers Use directly PTR_ERR() in dev_err_probe() parameter Remove the '!!' construction Remove of_match_ptr() - Patch 8 (new in v2) Add a missing of_node_put() in the simple-card driver - Patch 9 (new in v2) Handle additional-devs in the simple-card driver Herve Codina (13): ASoC: dt-bindings: Add audio-iio-aux ASoC: dt-bindings: simple-card: Add additional-devs subnode iio: inkern: Check error explicitly in iio_channel_read_max() iio: consumer.h: Fix raw values documentation notes iio: inkern: Remove the 'unused' variable usage in iio_channel_read_max() iio: inkern: Fix headers inclusion order minmax: Introduce {min,max}_array() iio: inkern: Use max_array() to get the maximum value from an array iio: inkern: Replace a FIXME comment by a TODO one iio: inkern: Add a helper to query an available minimum raw value ASoC: soc-dapm.h: Convert macros to return a compound literal ASoC: codecs: Add support for the generic IIO auxiliary devices ASoC: simple-card: Handle additional devices .../bindings/sound/audio-iio-aux.yaml | 64 ++++ .../bindings/sound/simple-card.yaml | 53 +++ drivers/iio/inkern.c | 86 ++++- include/linux/iio/consumer.h | 37 +- include/linux/minmax.h | 64 ++++ include/sound/soc-dapm.h | 138 ++++--- sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/audio-iio-aux.c | 338 ++++++++++++++++++ sound/soc/generic/simple-card.c | 46 ++- 10 files changed, 769 insertions(+), 71 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/audio-iio-aux.yaml create mode 100644 sound/soc/codecs/audio-iio-aux.c -- 2.40.1

6 29

[PATCH] ASoC: dt-bindings: nau8821: Convert to dtschema
by Seven Lee 22 Jun '23

22 Jun '23

Convert the NAU8821 audio CODEC bindings to DT schema. Signed-off-by: Seven Lee <wtli(a)nuvoton.com> --- .../devicetree/bindings/sound/nau8821.txt | 55 -------- .../bindings/sound/nuvoton,nau8821.yaml | 120 ++++++++++++++++++ 2 files changed, 120 insertions(+), 55 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/nau8821.txt create mode 100644 Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml diff --git a/Documentation/devicetree/bindings/sound/nau8821.txt b/Documentation/devicetree/bindings/sound/nau8821.txt deleted file mode 100644 index 7c84e7c7327a..000000000000 --- a/Documentation/devicetree/bindings/sound/nau8821.txt +++ /dev/null @@ -1,55 +0,0 @@ -Nuvoton NAU88L21 audio codec - -This device supports I2C only. - -Required properties: - - compatible : Must be "nuvoton,nau8821" - - - reg : the I2C address of the device. This is either 0x1B (CSB=0) or 0x54 (CSB=1). - -Optional properties: - - nuvoton,jkdet-enable: Enable jack detection via JKDET pin. - - nuvoton,jkdet-pull-enable: Enable JKDET pin pull. If set - pin pull enabled, - otherwise pin in high impedance state. - - nuvoton,jkdet-pull-up: Pull-up JKDET pin. If set then JKDET pin is pull up, otherwise pull down. - - nuvoton,jkdet-polarity: JKDET pin polarity. 0 - active high, 1 - active low. - - - nuvoton,vref-impedance: VREF Impedance selection - 0 - Open - 1 - 25 kOhm - 2 - 125 kOhm - 3 - 2.5 kOhm - - - nuvoton,micbias-voltage: Micbias voltage level. - 0 - VDDA - 1 - VDDA - 2 - VDDA * 1.1 - 3 - VDDA * 1.2 - 4 - VDDA * 1.3 - 5 - VDDA * 1.4 - 6 - VDDA * 1.53 - 7 - VDDA * 1.53 - - - nuvoton,jack-insert-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms - - nuvoton,jack-eject-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms - - - nuvoton,dmic-clk-threshold: the ADC threshold of DMIC clock. - - nuvoton,key_enable: Headset button detection switch. - -Example: - - headset: nau8821@1b { - compatible = "nuvoton,nau8821"; - reg = <0x1b>; - interrupt-parent = <&gpio>; - interrupts = <23 IRQ_TYPE_LEVEL_LOW>; - nuvoton,jkdet-enable; - nuvoton,jkdet-pull-enable; - nuvoton,jkdet-pull-up; - nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>; - nuvoton,vref-impedance = <2>; - nuvoton,micbias-voltage = <6>; - nuvoton,jack-insert-debounce = <7>; - nuvoton,jack-eject-debounce = <7>; - nuvoton,dmic-clk-threshold = 3072000; - }; diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml new file mode 100644 index 000000000000..728223d20876 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml @@ -0,0 +1,120 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nuvoton,nau8821.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NAU88L21 audio codec + +maintainers: + - Seven Lee <wtli(a)nuvoton.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: nuvoton,nau8821 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + nuvoton,jkdet-enable: + description: Enable jack detection via JKDET pin. + type: boolean + + nuvoton,jkdet-pull-enable: + description: Enable JKDET pin pull. If set - pin pull enabled, + otherwise pin in high impedance state. + type: boolean + + nuvoton,jkdet-pull-up: + description: Pull-up JKDET pin. If set then JKDET pin is pull up, + otherwise pull down. + type: boolean + + nuvoton,jkdet-polarity: + description: JKDET pin polarity. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # active high + - 1 # active low + default: 1 + + nuvoton,micbias-voltage: + description: MICBIAS output level select. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # VDDA + - 1 # VDDA * 1 + - 2 # VDDA * 1.1 + - 3 # VDDA * 1.2 + - 4 # VDDA * 1.3 + - 5 # VDDA * 1.4 + - 6 # VDDA * 1.53 + - 7 # VDDA * 1.53 + default: 6 + + nuvoton,vref-impedance: + description: VMID Tie-off impedance select. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # open(default) + - 1 # 25KOhms + - 2 # 125KOhms + - 3 # 2.5KOhms + default: 2 + + nuvoton,jack-insert-debounce: + description: number from 0 to 7 that sets debounce time to 2^(n+2)ms. + $ref: /schemas/types.yaml#/definitions/uint32 + maximum: 7 + default: 7 + + nuvoton,jack-eject-debounce: + description: number from 0 to 7 that sets debounce time to 2^(n+2)ms. + $ref: /schemas/types.yaml#/definitions/uint32 + maximum: 7 + default: 0 + + nuvoton,dmic-clk-threshold: + description: DMIC clock speed expected value. Unit is Hz. + $ref: /schemas/types.yaml#/definitions/uint32 + default: 3072000 + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/interrupt-controller/irq.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@1b { + #sound-dai-cells = <0>; + compatible = "nuvoton,nau8821"; + reg = <0x1b>; + interrupt-parent = <&gpio>; + interrupts = <23 IRQ_TYPE_LEVEL_LOW>; + nuvoton,jkdet-enable; + nuvoton,jkdet-pull-enable; + nuvoton,jkdet-pull-up; + nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>; + nuvoton,micbias-voltage = <6>; + nuvoton,vref-impedance = <2>; + nuvoton,jack-insert-debounce = <7>; + nuvoton,jack-eject-debounce = <0>; + nuvoton,dmic-clk-threshold = <3072000>; + }; + }; -- 2.25.1

3 2

[PATCH 0/8] ALSA: emu10k1: add support for high-bitrate modes of E-MU cards
by Oswald Buddenhagen 22 Jun '23

22 Jun '23

This series is what all the work was about: support the "dual-/quad-pumped" modes of the E-MU cards. Oswald Buddenhagen (8): ALSA: emu10k1: introduce alternative E-MU D.A.S. mode ALSA: emu10k1: improve mixer control naming in E-MU D.A.S. mode ALSA: emu10k1: set the "no filtering" bits on PCM voices ALSA: emu10k1: make playback in E-MU D.A.S. mode 32-bit ALSA: add snd_ctl_add_locked() ALSA: emu10k1: add support for 2x/4x word clocks in E-MU D.A.S. mode ALSA: emu10k1: add high-rate capture in E-MU D.A.S. mode ALSA: emu10k1: add high-rate playback in E-MU D.A.S. mode include/sound/control.h | 1 + include/sound/emu10k1.h | 11 + sound/core/control.c | 31 ++ sound/pci/emu10k1/emu10k1.c | 29 +- sound/pci/emu10k1/emu10k1_main.c | 19 +- sound/pci/emu10k1/emufx.c | 109 +++- sound/pci/emu10k1/emumixer.c | 901 +++++++++++++++++++++++++++---- sound/pci/emu10k1/emupcm.c | 422 +++++++++++++-- sound/pci/emu10k1/emuproc.c | 5 + sound/pci/emu10k1/io.c | 30 +- sound/pci/emu10k1/voice.c | 6 + 11 files changed, 1383 insertions(+), 181 deletions(-) -- 2.40.0.152.g15d061e6df

3 21

[PATCH 1/1] ASoC: codecs: max98090: Allow dsp_a mode
by Maxim Kochetkov 22 Jun '23

22 Jun '23

TDM mode for max98090 is dsp_a compatible. So allow it. Signed-off-by: Maxim Kochetkov <fido_max(a)inbox.ru> --- sound/soc/codecs/max98090.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 7bc463910d4f..403926254c84 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1635,7 +1635,7 @@ static int max98090_dai_set_fmt(struct snd_soc_dai *codec_dai, regval |= M98090_RJ_MASK; break; case SND_SOC_DAIFMT_DSP_A: - /* Not supported mode */ + break; default: dev_err(component->dev, "DAI format unsupported"); return -EINVAL; -- 2.40.1

2 10

topology: plugins: nhlt: set dmic stereo mode only in hw version 1
by GitHub pull_request - opened 22 Jun '23

22 Jun '23

alsa-project/alsa-utils pull request #222 was opened from juimonen: Dmic stereo mode should be set only in hw version 1. In later hw versions this bit is reserved. Request URL : https://github.com/alsa-project/alsa-utils/pull/222 Patch URL : https://github.com/alsa-project/alsa-utils/pull/222.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] ASoC: soc-compress: Fix deadlock in soc_compr_open_fe
by yixuanjiang 21 Jun '23

21 Jun '23

Modify the error handling flow by release lock. The require mutex will keep holding if open fail. Fixes: aa9ff6a4955f ("ASoC: soc-compress: Reposition and add pcm_mutex") Signed-off-by: yixuanjiang <yixuanjiang(a)google.com> --- sound/soc/soc-compress.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index d8715db5e415..2117fd61cf8f 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -194,6 +194,7 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) snd_soc_dai_compr_shutdown(cpu_dai, cstream, 1); out: dpcm_path_put(&list); + snd_soc_dpcm_mutex_unlock(fe); be_err: fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; snd_soc_card_mutex_unlock(fe->card); -- 2.41.0.162.gfafddb0af9-goog

2 1

[PATCH v2 1/2] ASoC: Intel: sof_rt5682: reorder quirk table
by Mac Chiang 21 Jun '23

21 Jun '23

From: Terry Cheong <htcheong(a)chromium.org> Reorder the entries in quirk table to group entries with same platform. Signed-off-by: Terry Cheong <htcheong(a)chromium.org> Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- For the purpose of rebasing PATCH 2/2, pick up from link:https://github.com/thesofproject/linux/pull/4419 Mainly pick up mtl_rt1019_rt5682 driver for Chrome projects support. Link:https://github.com/thesofproject/linux/pull/4417 Changes v2: - missing Signed-off-by in PATCH 1/2, added it. --- sound/soc/intel/boards/sof_rt5682.c | 52 ++++++++++++++--------------- 1 file changed, 26 insertions(+), 26 deletions(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 86bbc1fea6ff..be34d7595537 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -981,14 +981,13 @@ static const struct platform_device_id board_ids[] = { .name = "sof_rt5682", }, { - .name = "tgl_mx98357_rt5682", + .name = "cml_rt1015_rt5682", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_MCLK_24MHZ | SOF_RT5682_SSP_CODEC(0) | SOF_SPEAKER_AMP_PRESENT | - SOF_RT5682_SSP_AMP(1) | - SOF_RT5682_NUM_HDMIDEV(4) | - SOF_BT_OFFLOAD_SSP(2) | - SOF_SSP_BT_OFFLOAD_PRESENT), + SOF_RT1015_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1)), }, { .name = "jsl_rt5682_rt1015", @@ -1000,33 +999,38 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(1)), }, { - .name = "tgl_mx98373_rt5682", + .name = "jsl_rt5682_mx98360", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_MCLK_24MHZ | SOF_RT5682_SSP_CODEC(0) | SOF_SPEAKER_AMP_PRESENT | - SOF_MAX98373_SPEAKER_AMP_PRESENT | - SOF_RT5682_SSP_AMP(1) | - SOF_RT5682_NUM_HDMIDEV(4) | - SOF_BT_OFFLOAD_SSP(2) | - SOF_SSP_BT_OFFLOAD_PRESENT), + SOF_MAX98360A_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1)), }, { - .name = "jsl_rt5682_mx98360", + .name = "jsl_rt5682_rt1015p", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | SOF_RT5682_MCLK_24MHZ | SOF_RT5682_SSP_CODEC(0) | SOF_SPEAKER_AMP_PRESENT | - SOF_MAX98360A_SPEAKER_AMP_PRESENT | + SOF_RT1015P_SPEAKER_AMP_PRESENT | SOF_RT5682_SSP_AMP(1)), }, { - .name = "cml_rt1015_rt5682", + .name = "jsl_rt5682", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | SOF_RT5682_MCLK_24MHZ | + SOF_RT5682_SSP_CODEC(0)), + }, + { + .name = "tgl_mx98357_rt5682", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | SOF_RT5682_SSP_CODEC(0) | SOF_SPEAKER_AMP_PRESENT | - SOF_RT1015_SPEAKER_AMP_PRESENT | - SOF_RT5682_SSP_AMP(1)), + SOF_RT5682_SSP_AMP(1) | + SOF_RT5682_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), }, { .name = "tgl_rt1011_rt5682", @@ -1040,13 +1044,15 @@ static const struct platform_device_id board_ids[] = { SOF_SSP_BT_OFFLOAD_PRESENT), }, { - .name = "jsl_rt5682_rt1015p", + .name = "tgl_mx98373_rt5682", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_RT5682_MCLK_24MHZ | SOF_RT5682_SSP_CODEC(0) | SOF_SPEAKER_AMP_PRESENT | - SOF_RT1015P_SPEAKER_AMP_PRESENT | - SOF_RT5682_SSP_AMP(1)), + SOF_MAX98373_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1) | + SOF_RT5682_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), }, { .name = "adl_mx98373_rt5682", @@ -1149,12 +1155,6 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(1) | SOF_RT5682_NUM_HDMIDEV(4)), }, - { - .name = "jsl_rt5682", - .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_RT5682_MCLK_24MHZ | - SOF_RT5682_SSP_CODEC(0)), - }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); -- 2.20.1

2 2

[PATCH v4 1/2] ASoC: dt-bindings: qcom,wsa8840: Add WSA884x family of speakers
by Krzysztof Kozlowski 21 Jun '23

21 Jun '23

Add binding for WSA8840/WSA8845/WSA8845H smart speaker amplifiers used in Qualcomm QRD8550 board with SM8550 SoC. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Changes in v4: 1. None. Changes in v3: 1. None. Changes in v2: 1. Correct compatible (sdw version 1 -> 2). Cc: Patrick Lai <quic_plai(a)quicinc.com> Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- .../bindings/sound/qcom,wsa8840.yaml | 66 +++++++++++++++++++ 1 file changed, 66 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml diff --git a/Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml b/Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml new file mode 100644 index 000000000000..e6723c9e312a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml @@ -0,0 +1,66 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,wsa8840.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm WSA8840/WSA8845/WSA8845H smart speaker amplifier + +maintainers: + - Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> + - Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> + +description: + WSA884X is a family of Qualcomm Aqstic smart speaker amplifiers using + SoundWire digital audio interface. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: sdw20217020400 + + reg: + maxItems: 1 + + powerdown-gpios: + description: Powerdown/Shutdown line to use (pin SD_N) + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + vdd-1p8-supply: true + vdd-io-supply: true + +required: + - compatible + - reg + - powerdown-gpios + - '#sound-dai-cells' + - vdd-1p8-supply + - vdd-io-supply + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + soundwire-controller { + #address-cells = <2>; + #size-cells = <0>; + + speaker@0,1 { + compatible = "sdw20217020400"; + reg = <0 1>; + pinctrl-names = "default"; + pinctrl-0 = <&spkr_2_sd_n_active>; + powerdown-gpios = <&lpass_tlmm 18 GPIO_ACTIVE_LOW>; + #sound-dai-cells = <0>; + sound-name-prefix = "SpkrRight"; + vdd-1p8-supply = <&vreg_l15b_1p8>; + vdd-io-supply = <&vreg_l3g_1p2>; + }; + }; -- 2.34.1

3 3

[PATCH] ASoC: fsl-asoc-card: Allow passing the number of slots in use
by Fabio Estevam 21 Jun '23

21 Jun '23

From: Fabio Estevam <festevam(a)denx.de> Currently, fsl-asoc-card supports passing the width of the TDM slot, but not the number of slots in use, as it harcodes it as two slots. Add support for passing the number of slots in use. Signed-off-by: Fabio Estevam <festevam(a)denx.de> --- sound/soc/fsl/fsl-asoc-card.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 40870668ee24..98137c33f877 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -60,6 +60,7 @@ struct codec_priv { * @sysclk_dir: SYSCLK directions for set_sysclk() * @sysclk_id: SYSCLK ids for set_sysclk() * @slot_width: Slot width of each frame + * @slot_num: Number of slots of each frame * * Note: [1] for tx and [0] for rx */ @@ -68,6 +69,7 @@ struct cpu_priv { u32 sysclk_dir[2]; u32 sysclk_id[2]; u32 slot_width; + u32 slot_num; }; /** @@ -189,7 +191,11 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, } if (cpu_priv->slot_width) { - ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2, + if (!cpu_priv->slot_num) + cpu_priv->slot_num = 2; + + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, + cpu_priv->slot_num, cpu_priv->slot_width); if (ret && ret != -ENOTSUPP) { dev_err(dev, "failed to set TDM slot for cpu dai\n"); -- 2.34.1

2 1

[GIT PULL] ASoC fixes for v6.4-rc7
by Mark Brown 21 Jun '23

21 Jun '23

The following changes since commit f9fd804aa0a36f15a35ca070ec4c52650876cc29: ASoC: tegra: Fix Master Volume Control (2023-06-13 12:10:56 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.4-rc7 for you to fetch changes up to 1f1ef7e5bbe2de125da413e238915f6047ea4ba2: ASoC: intel: sof_sdw: Fixup typo in device link checking (2023-06-19 12:58:22 +0100) ---------------------------------------------------------------- ASoC: Fix for v6.4 A fix for a typoed iterator in the Intel Soundwire driver, fairly simple on inspection though not reviewed by Intel. ---------------------------------------------------------------- Charles Keepax (1): ASoC: intel: sof_sdw: Fixup typo in device link checking sound/soc/intel/boards/sof_sdw.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)

2 1

[PATCH] ALSA: hda/realtek: Add quirk for ASUS ROG GV601V
by Luke D. Jones 21 Jun '23

21 Jun '23

Adds the required quirk to enable the Cirrus amp and correct pins on the ASUS ROG GV601V series. While this works if the related _DSD properties are made available, these aren't included in the ACPI of these laptops (yet). Signed-off-by: Luke D. Jones <luke(a)ljones.dev> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 17d5bdd4be6f..dabfdecece26 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9527,6 +9527,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), SND_PCI_QUIRK(0x1043, 0x1473, "ASUS GU604V", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1483, "ASUS GU603V", ALC285_FIXUP_ASUS_HEADSET_MIC), + SND_PCI_QUIRK(0x1043, 0x1493, "ASUS GV601V", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), SND_PCI_QUIRK(0x1043, 0x1662, "ASUS GV301QH", ALC294_FIXUP_ASUS_DUAL_SPK), SND_PCI_QUIRK(0x1043, 0x1683, "ASUS UM3402YAR", ALC287_FIXUP_CS35L41_I2C_2), -- 2.41.0

2 1

[PATCH 0/4] soundwire: allow for more than 8 devices, keep IDA for wake-capable devices
by Bard Liao 21 Jun '23

21 Jun '23

This series suggests a hybrid strategy for device number allocation, where only wake-capable devices use a system-unique Device Number which will be used on LunarLake to handle wake-ups using the HDaudio WAKEEN and WAKESTS. Pierre-Louis Bossart (4): soundwire: add enum to control device number allocation soundwire: introduce SDW_DEV_NUM_ALLOC_IDA_WAKE_ONLY soundwire: extend parameters of new_peripheral_assigned() callback soundwire: intel_auxdevice: use SDW_DEV_NUM_ALLOC_IDA_WAKE_ONLY drivers/soundwire/bus.c | 28 ++++++++++++++++++++++------ drivers/soundwire/intel_auxdevice.c | 26 ++++++++++++++++++++++---- include/linux/soundwire/sdw.h | 24 ++++++++++++++++++++++-- 3 files changed, 66 insertions(+), 12 deletions(-) -- 2.25.1

3 13

[PATCH] soundwire: qcom: fix storing port config out-of-bounds
by Krzysztof Kozlowski 21 Jun '23

21 Jun '23

The 'qcom_swrm_ctrl->pconfig' has size of QCOM_SDW_MAX_PORTS (14), however we index it starting from 1, not 0, to match real port numbers. This can lead to writing port config past 'pconfig' bounds and overwriting next member of 'qcom_swrm_ctrl' struct. Reported also by smatch: drivers/soundwire/qcom.c:1269 qcom_swrm_get_port_config() error: buffer overflow 'ctrl->pconfig' 14 <= 14 Fixes: 9916c02ccd74 ("soundwire: qcom: cleanup internal port config indexing") Cc: <stable(a)vger.kernel.org> Reported-by: kernel test robot <lkp(a)intel.com> Reported-by: Dan Carpenter <error27(a)gmail.com> Link: https://lore.kernel.org/r/202305201301.sCJ8UDKV-lkp@intel.com/ Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- drivers/soundwire/qcom.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 7cb1b7eba814..88a772075907 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -202,7 +202,8 @@ struct qcom_swrm_ctrl { u32 intr_mask; u8 rcmd_id; u8 wcmd_id; - struct qcom_swrm_port_config pconfig[QCOM_SDW_MAX_PORTS]; + /* Port numbers are 1 - 14 */ + struct qcom_swrm_port_config pconfig[QCOM_SDW_MAX_PORTS + 1]; struct sdw_stream_runtime *sruntime[SWRM_MAX_DAIS]; enum sdw_slave_status status[SDW_MAX_DEVICES + 1]; int (*reg_read)(struct qcom_swrm_ctrl *ctrl, int reg, u32 *val); -- 2.34.1

4 5

[PATCH 0/3] ALSA: ump: Preliminary changes for gadget driver
by Takashi Iwai 21 Jun '23

21 Jun '23

Hi, this is a small patch set to change the UMP core for the upcoming gadget driver support. Basically exporting a couple of helper functions and adding a flag to suppress the internal UMP handling. No functional changes by those alone. Takashi === Takashi Iwai (3): ALSA: ump: Add helper to change MIDI protocol ALSA: ump: Add no_process_stream flag ALSA: ump: Export snd_ump_receive_ump_val() include/sound/ump.h | 4 ++++ sound/core/ump.c | 46 +++++++++++++++++++++++++++++++++++---------- 2 files changed, 40 insertions(+), 10 deletions(-) -- 2.35.3

1 3

[PATCH 1/2] soundwire: bus: Prevent lockdep asserts when stream has multiple buses
by Richard Fitzgerald 21 Jun '23

21 Jun '23

Give the bus_lock and msg_lock of each bus a different unique key so that it is possible to acquire the locks of multiple buses without lockdep asserting a possible deadlock. Using mutex_init() to initialize a mutex gives all those mutexes the same lock class. Lockdep checking treats it as an error to attempt to take a mutex while already holding a mutex of the same class. This causes a lockdep assert when sdw_acquire_bus_lock() attempts to lock multiple buses, and when do_bank_switch() takes multiple msg_lock. [ 138.697350] WARNING: possible recursive locking detected [ 138.697366] 6.3.0-test #1 Tainted: G E [ 138.697380] -------------------------------------------- [ 138.697394] play/903 is trying to acquire lock: [ 138.697409] ffff99b8c41aa8c8 (&bus->bus_lock){+.+.}-{3:3}, at: sdw_prepare_stream+0x52/0x2e0 [ 138.697443] but task is already holding lock: [ 138.697468] ffff99b8c41af8c8 (&bus->bus_lock){+.+.}-{3:3}, at: sdw_prepare_stream+0x52/0x2e0 [ 138.697493] other info that might help us debug this: [ 138.697521] Possible unsafe locking scenario: [ 138.697540] CPU0 [ 138.697550] ---- [ 138.697559] lock(&bus->bus_lock); [ 138.697570] lock(&bus->bus_lock); [ 138.697581] *** DEADLOCK *** Giving each mutex a unique key allows multiple to be held without triggering a lockdep assert. But note that it does not allow them to be taken in one order then a different order. If two mutexes are taken in the order A, B then they must always be taken in that order otherwise they could deadlock. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- drivers/soundwire/bus.c | 15 +++++++++++++-- include/linux/soundwire/sdw.h | 3 +++ 2 files changed, 16 insertions(+), 2 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index b44f8d0affa6..dba920ec88f6 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -69,8 +69,17 @@ int sdw_bus_master_add(struct sdw_bus *bus, struct device *parent, return -EINVAL; } - mutex_init(&bus->msg_lock); - mutex_init(&bus->bus_lock); + /* + * Give each bus_lock and msg_lock a unique key so that lockdep won't + * trigger a deadlock warning when the locks of several buses are + * grabbed during configuration of a multi-bus stream. + */ + lockdep_register_key(&bus->msg_lock_key); + __mutex_init(&bus->msg_lock, "msg_lock", &bus->msg_lock_key); + + lockdep_register_key(&bus->bus_lock_key); + __mutex_init(&bus->bus_lock, "bus_lock", &bus->bus_lock_key); + INIT_LIST_HEAD(&bus->slaves); INIT_LIST_HEAD(&bus->m_rt_list); @@ -181,6 +190,8 @@ void sdw_bus_master_delete(struct sdw_bus *bus) sdw_master_device_del(bus); sdw_bus_debugfs_exit(bus); + lockdep_unregister_key(&bus->bus_lock_key); + lockdep_unregister_key(&bus->msg_lock_key); ida_free(&sdw_bus_ida, bus->id); } EXPORT_SYMBOL(sdw_bus_master_delete); diff --git a/include/linux/soundwire/sdw.h b/include/linux/soundwire/sdw.h index c076a3f879b3..f523ceabd059 100644 --- a/include/linux/soundwire/sdw.h +++ b/include/linux/soundwire/sdw.h @@ -5,6 +5,7 @@ #define __SOUNDWIRE_H #include <linux/bug.h> +#include <linux/lockdep_types.h> #include <linux/mod_devicetable.h> #include <linux/bitfield.h> @@ -907,7 +908,9 @@ struct sdw_bus { struct list_head slaves; DECLARE_BITMAP(assigned, SDW_MAX_DEVICES); struct mutex bus_lock; + struct lock_class_key bus_lock_key; struct mutex msg_lock; + struct lock_class_key msg_lock_key; int (*compute_params)(struct sdw_bus *bus); const struct sdw_master_ops *ops; const struct sdw_master_port_ops *port_ops; -- 2.30.2

3 4

[PATCH] soundwire: intel_ace2x: fix SND_SOC_SOF_HDA_MLINK dependency
by Arnd Bergmann 21 Jun '23

21 Jun '23

From: Arnd Bergmann <arnd(a)arndb.de> The ace2x driver can be build with or without mlink support, but when SND_SOC_SOF_HDA_MLINK is set to =m and soundwire is built-in, it fails with a link error: ld.lld: error: undefined symbol: hdac_bus_eml_sdw_wait_syncpu_unlocked >>> referenced by intel_ace2x.c >>> drivers/soundwire/intel_ace2x.o:(intel_link_power_up) in archive vmlinux.a ld.lld: error: undefined symbol: hdac_bus_eml_sdw_sync_arm_unlocked >>> referenced by intel_ace2x.c >>> drivers/soundwire/intel_ace2x.o:(intel_sync_arm) in archive vmlinux.a Add a Kconfig dependency that prevents that broken configuration but still allows soundwire to be a loadable module instead. Fixes: 4d1e2464a1104 ("soundwire: intel_ace2x: add sync_arm/sync_go helpers") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- drivers/soundwire/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/soundwire/Kconfig b/drivers/soundwire/Kconfig index fa71c9a36df7a..4d8f3b7024ae5 100644 --- a/drivers/soundwire/Kconfig +++ b/drivers/soundwire/Kconfig @@ -37,6 +37,7 @@ config SOUNDWIRE_INTEL select SOUNDWIRE_GENERIC_ALLOCATION select AUXILIARY_BUS depends on ACPI && SND_SOC + depends on SND_SOC_SOF_HDA_MLINK || !SND_SOC_SOF_HDA_MLINK help SoundWire Intel Master driver. If you have an Intel platform which has a SoundWire Master then -- 2.39.2

2 1

ALSA PM issues
by Rizwan Mubarak Chikodi 21 Jun '23

21 Jun '23

Hi ALSA team, We are using the standard ALSA driver framework for capture and playback in our Linux platform. In our platform arecord and aplay work as expected using I2S-CODEC, In one of our applications we have to do Suspend to RAM (S2R) operation. When Suspend to RAM triggers while playing audio through I2S-CODEC, The playback fails to resume-back once the system is resumed. We have thoroughly verified the drivers of the I2S controller and CODEC, But have not observed any issues, We are suspecting that the issue might be in the ASOC layer. Following are the details: * kernel version: 5.10.41 * Platform: RisCV64 We are playing audio in one terminal using " aplay (using standard alsa-utils package)" command and running S2R command " echo mem > /sys/power/state " in other terminals to suspend and resume the system. We are facing input/error once the system resumed back as mentioned below, --------------------------------------------------------------------------------------------- root@XX-SOC-BSP-31JAN23-REL12-v1:~# aplay -Dhw:2,0 3.wav Playing WAVE '3.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo aplay: pcm_write:2061: write error: Input/output error ---------------------------------------------------------------------------------------------- And we found that in " pcm_lib.c " the tout time is 2.5 seconds, once we do Suspend to resume we observe the tout is zero. if (runtime->no_period_wakeup) wait_time = MAX_SCHEDULE_TIMEOUT; else { /* use wait time from substream if available */ if (substream->wait_time) { wait_time = substream->wait_time; } else { wait_time = 10; if (runtime->rate) { long t = runtime->period_size * 2 / runtime->rate; wait_time = max(t, wait_time); } wait_time = msecs_to_jiffies(wait_time * 1000); printk("******sound/core/pcm_lib.c ::: tout = %ld ****",wait_time); } } ....... tout = schedule_timeout(wait_time); Thanks & regards Rizwan Chikodi

1 0

acp63: add initial support for AMD Pink Sardine - ACP63
by GitHub pull_request - opened 21 Jun '23

21 Jun '23

alsa-project/alsa-ucm-conf pull request #327 was opened from pma1: Link: #326 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/327 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/327.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 0/2] AMD Vangogh support for NAU8821/MAX98388
by Cristian Ciocaltea 21 Jun '23

21 Jun '23

This patch series extends the Vangogh machine driver to support a variant based on the Nuvoton NAU88L21 Codec and the Analog Devices MAX98388 Speaker Amplifier. Cristian Ciocaltea (2): ASoC: amd: vangogh: Add ACPI probe support ASoC: amd: vangogh: Add support for NAU8821/MAX98388 variant sound/soc/amd/Kconfig | 5 +- sound/soc/amd/vangogh/acp5x-mach.c | 155 ++++++++++++++++++++++++----- sound/soc/amd/vangogh/acp5x.h | 2 +- sound/soc/amd/vangogh/pci-acp5x.c | 3 - 4 files changed, 134 insertions(+), 31 deletions(-) -- 2.41.0

1 3

[PATCH] sound: make all 'class' structures const
by Greg Kroah-Hartman 21 Jun '23

21 Jun '23

From: Ivan Orlov <ivan.orlov0322(a)gmail.com> Now that the driver core allows for struct class to be in read-only memory, making all 'class' structures to be declared at build time placing them into read-only memory, instead of having to be dynamically allocated at load time. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Ivan Orlov <ivan.orlov0322(a)gmail.com> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: Geoff Levand <geoff(a)infradead.org> Cc: Thierry Reding <treding(a)nvidia.com> Cc: "Uwe Kleine-König" <u.kleine-koenig(a)pengutronix.de> Cc: alsa-devel(a)alsa-project.org Suggested-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Ivan Orlov <ivan.orlov0322(a)gmail.com> Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- include/sound/core.h | 2 +- sound/core/control_led.c | 2 +- sound/core/init.c | 4 ++-- sound/sound_core.c | 23 ++++++++++++----------- 4 files changed, 16 insertions(+), 15 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index 3edc4ab08774..bbcecfff723d 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -232,7 +232,7 @@ static inline struct device *snd_card_get_device_link(struct snd_card *card) extern int snd_major; extern int snd_ecards_limit; -extern struct class *sound_class; +extern const struct class sound_class; #ifdef CONFIG_SND_DEBUG extern struct dentry *sound_debugfs_root; #endif diff --git a/sound/core/control_led.c b/sound/core/control_led.c index 3cadd40100f3..ee77547bf8dc 100644 --- a/sound/core/control_led.c +++ b/sound/core/control_led.c @@ -737,7 +737,7 @@ static int __init snd_ctl_led_init(void) unsigned int group; device_initialize(&snd_ctl_led_dev); - snd_ctl_led_dev.class = sound_class; + snd_ctl_led_dev.class = &sound_class; snd_ctl_led_dev.release = snd_ctl_led_dev_release; dev_set_name(&snd_ctl_led_dev, "ctl-led"); if (device_add(&snd_ctl_led_dev)) { diff --git a/sound/core/init.c b/sound/core/init.c index df0c22480375..baef2688d0cf 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -129,7 +129,7 @@ void snd_device_initialize(struct device *dev, struct snd_card *card) device_initialize(dev); if (card) dev->parent = &card->card_dev; - dev->class = sound_class; + dev->class = &sound_class; dev->release = default_release; } EXPORT_SYMBOL_GPL(snd_device_initialize); @@ -331,7 +331,7 @@ static int snd_card_init(struct snd_card *card, struct device *parent, device_initialize(&card->card_dev); card->card_dev.parent = parent; - card->card_dev.class = sound_class; + card->card_dev.class = &sound_class; card->card_dev.release = release_card_device; card->card_dev.groups = card->dev_groups; card->dev_groups[0] = &card_dev_attr_group; diff --git a/sound/sound_core.c b/sound/sound_core.c index 4f6911274d56..d81fed1c1226 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -23,9 +23,6 @@ static inline int init_oss_soundcore(void) { return 0; } static inline void cleanup_oss_soundcore(void) { } #endif -struct class *sound_class; -EXPORT_SYMBOL(sound_class); - MODULE_DESCRIPTION("Core sound module"); MODULE_AUTHOR("Alan Cox"); MODULE_LICENSE("GPL"); @@ -37,6 +34,12 @@ static char *sound_devnode(const struct device *dev, umode_t *mode) return kasprintf(GFP_KERNEL, "snd/%s", dev_name(dev)); } +const struct class sound_class = { + .name = "sound", + .devnode = sound_devnode, +}; +EXPORT_SYMBOL(sound_class); + static int __init init_soundcore(void) { int rc; @@ -45,21 +48,19 @@ static int __init init_soundcore(void) if (rc) return rc; - sound_class = class_create("sound"); - if (IS_ERR(sound_class)) { + rc = class_register(&sound_class); + if (rc) { cleanup_oss_soundcore(); - return PTR_ERR(sound_class); + return rc; } - sound_class->devnode = sound_devnode; - return 0; } static void __exit cleanup_soundcore(void) { cleanup_oss_soundcore(); - class_destroy(sound_class); + class_unregister(&sound_class); } subsys_initcall(init_soundcore); @@ -276,7 +277,7 @@ static int sound_insert_unit(struct sound_unit **list, const struct file_operati } } - device_create(sound_class, dev, MKDEV(SOUND_MAJOR, s->unit_minor), + device_create(&sound_class, dev, MKDEV(SOUND_MAJOR, s->unit_minor), NULL, "%s", s->name+6); return s->unit_minor; @@ -302,7 +303,7 @@ static void sound_remove_unit(struct sound_unit **list, int unit) if (!preclaim_oss) __unregister_chrdev(SOUND_MAJOR, p->unit_minor, 1, p->name); - device_destroy(sound_class, MKDEV(SOUND_MAJOR, p->unit_minor)); + device_destroy(&sound_class, MKDEV(SOUND_MAJOR, p->unit_minor)); kfree(p); } } -- 2.41.0

2 1

[PATCH v3] ALSA: compress: allow setting codec params after next track
by Srinivas Kandagatla 21 Jun '23

21 Jun '23

For gapless playback it is possible that each track can have different codec profile with same decoder, for example we have WMA album, we may have different tracks as WMA v9, WMA v10 and so on Or if DSP's like QDSP have abililty to switch decoders on single stream for each track, then this call could be used to set new codec parameters. Existing code does not allow to change this profile while doing gapless playback. Reuse existing SNDRV_COMPRESS_SET_PARAMS to set this new track params along some additional checks to enforce proper state machine. With this new changes now the user can call SNDRV_COMPRESS_SET_PARAMS anytime after setting next track and additional check in write should also ensure that params are set before writing new data. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Changes since v2: simplified the logic to allow set_params and udated the documentation accordingly Documentation/sound/designs/compress-offload.rst | 11 ++++++----- sound/core/compress_offload.c | 5 ++++- 2 files changed, 10 insertions(+), 6 deletions(-) diff --git a/Documentation/sound/designs/compress-offload.rst b/Documentation/sound/designs/compress-offload.rst index 935f325dbc77..655624f77092 100644 --- a/Documentation/sound/designs/compress-offload.rst +++ b/Documentation/sound/designs/compress-offload.rst @@ -268,11 +268,12 @@ with setting of meta_data and signalling for next track :: | | | V | +----------+ - | | | - | |NEXT_TRACK| - | | | - | +----------+ - | | + | compr_set_params() | | + | +-----------|NEXT_TRACK| + | | | | + | | +--+-------+ + | | | | + | +--------------+ | | | | | compr_partial_drain() | | diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 243acad89fd3..30f73097447b 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -589,7 +589,7 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg) struct snd_compr_params *params; int retval; - if (stream->runtime->state == SNDRV_PCM_STATE_OPEN) { + if (stream->runtime->state == SNDRV_PCM_STATE_OPEN || stream->next_track) { /* * we should allow parameter change only when stream has been * opened not in other cases @@ -612,6 +612,9 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg) if (retval) goto out; + if (stream->next_track) + goto out; + stream->metadata_set = false; stream->next_track = false; -- 2.21.0

3 2

Thinkpad Z16 Gen2 Mic mute LED not switching on when press F4
by GitHub issues - opened 21 Jun '23

21 Jun '23

alsa-project/alsa-ucm-conf issue #326 was opened from pma1: The mic and sound output works fine. alsamixer can switch on/off the F4/Mic-mute LED automatically. But when press Fn+F4, the mic-mute led is not working. /sys/class/sound/ctl-led/mic/card1# cat list 129 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/326 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

use-case: Add new LatencyOffsetMs field
by GitHub pull_request - opened 21 Jun '23

21 Jun '23

alsa-project/alsa-lib pull request #332 was opened from afq984: Some drivers do not report accurate latencies. Add a LatencyOffsetMs UCM Value allow adding a static offset to the reported latency. Request URL : https://github.com/alsa-project/alsa-lib/pull/332 Patch URL : https://github.com/alsa-project/alsa-lib/pull/332.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH V4 0/9] ASoC: amd: ps: add SoundWire support
by Vijendar Mukunda 21 Jun '23

21 Jun '23

This patch series add support for - Platform device creation for SoundWire Manager instances and PDM controller. - SoundWire DMA driver. - Interrupt handling for SoundWire manager related interrupts, SoundWire DMA interrupts and ACP error interrupts. - ACP PCI driver PM ops modification with respect to SoundWire Power modes. Changes since v3: - use pdev_config instead of pdev_mask in the code. - define platform device configuration macros rather than enums - add comments for MIPI DisCo version - refactor SoundWire DMA start/stop sequence using single function - refactor "acp_reset" flag related functionality. Changes since v2: - add comments in irq handler. - remove workqueue for SoundWire DMA interrupts and use thread implementation for DMA interrupt handling. - add error checks in sdw_amd_scan_controller() - remove passing "acp_lock" as platform resource for SoundWire DMA driver and PDM driver. - retrieve "acp_lock" reference from parent driver directly and use the reference in SoundWire DMA driver. - add handling for acp pci driver probe even when no ACP PDM or SoundWire manager platform devices created. - Fix indentation for acp_sdw_dma_count structure variables. - Use macro instead of hard coded values for FIFO offset and PTE offset. - Change pm_runtime enable sequence in SoundWire DMA driver probe function. - Refactor system level resume callback in SoundWire DMA Changes since v1: - update "soundwire" as "SoundWire" in code. - add comments for platform device mask and platform device count - remove unused variables in acp pci driver private data structure - refactor dma enable register structures in SoundWire DMA driver - add TODO comments in IRQ handler - update IRQ mask values using bit macros - Fix build error reported in Makefile - rename "sdw_dma_stream_instance" structure name as "acp_sdw_dma_stream" Vijendar Mukunda (9): ASoC: amd: ps: create platform devices based on acp config ASoC: amd: ps: handle SoundWire interrupts in acp pci driver ASoC: amd: ps: add SoundWire dma driver ASoC: amd: ps: add SoundWire dma driver dma ops ASoC: amd: ps: add support for SoundWire DMA interrupts ASoC: amd: ps: add pm ops support for SoundWire dma driver ASoC: amd: ps: enable SoundWire dma driver build ASoC: amd: update comments in Kconfig file ASoC: amd: ps: add acp_reset flag check in acp pci driver pm ops. sound/soc/amd/Kconfig | 3 +- sound/soc/amd/ps/Makefile | 2 + sound/soc/amd/ps/acp63.h | 172 ++++++++++- sound/soc/amd/ps/pci-ps.c | 419 +++++++++++++++++++++++-- sound/soc/amd/ps/ps-sdw-dma.c | 555 ++++++++++++++++++++++++++++++++++ 5 files changed, 1115 insertions(+), 36 deletions(-) create mode 100644 sound/soc/amd/ps/ps-sdw-dma.c -- 2.34.1

4 24

[PATCH v3 0/9] add snd_soc_{of_}get_dlc()
by Kuninori Morimoto 21 Jun '23

21 Jun '23

Hi Mark Current soc-core.c has snd_soc_{of_}get_dai_name() to get DAI name for dlc (snd_soc_dai_link_component). It gets .dai_name, but we need .of_node too. Therefor user need to arrange. It will be more useful if it gets both .dai_name and .of_node. This patch adds snd_soc_{of_}get_dlc() for it, and convert to use it. v2 -> v3 - care loongson on [2/9] v1 -> v2 - add patch for loongson Link: https://lore.kernel.org/r/87cz1rdmp0.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87mt12hp75.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (9): ASoC: soc-core.c: add snd_soc_{of_}get_dlc() ASoC: soc-core.c: add index on snd_soc_of_get_dai_name() ASoC: fsl: use snd_soc_{of_}get_dlc() ASoC: qcom: use snd_soc_{of_}get_dlc() ASoC: meson: use snd_soc_{of_}get_dlc() ASoC: samsung: use snd_soc_{of_}get_dlc() ASoC: loongson: use snd_soc_{of_}get_dlc() ASoC: soc-core.c: use snd_soc_{of_}get_dlc() ASoC: simple-card.c: use snd_soc_{of_}get_dlc() include/sound/soc.h | 8 ++- sound/soc/fsl/imx-card.c | 14 +---- sound/soc/fsl/imx-rpmsg.c | 3 +- sound/soc/generic/simple-card-utils.c | 4 +- sound/soc/generic/simple-card.c | 4 +- sound/soc/loongson/loongson_card.c | 34 ++++------ sound/soc/mediatek/mt8173/mt8173-rt5650.c | 2 +- sound/soc/meson/axg-card.c | 3 +- sound/soc/meson/gx-card.c | 3 +- sound/soc/meson/meson-card-utils.c | 16 ++--- sound/soc/meson/meson-card.h | 3 +- sound/soc/qcom/common.c | 13 +--- sound/soc/samsung/odroid.c | 16 +---- sound/soc/soc-core.c | 77 ++++++++++++----------- 14 files changed, 79 insertions(+), 121 deletions(-) -- 2.25.1

2 10

[PATCH v2 0/9] add snd_soc_{of_}get_dlc()
by Kuninori Morimoto 21 Jun '23

21 Jun '23

Hi Mark Current soc-core.c has snd_soc_{of_}get_dai_name() to get DAI name for dlc (snd_soc_dai_link_component). It gets .dai_name, but we need .of_node too. Therefor user need to arrange. It will be more useful if it gets both .dai_name and .of_node. This patch adds snd_soc_{of_}get_dlc() for it, and convert to use it. v1 -> v2 - care loongson Link: https://lore.kernel.org/r/87mt12hp75.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (9): ASoC: soc-core.c: add snd_soc_{of_}get_dlc() ASoC: soc-core.c: add index on snd_soc_of_get_dai_name() ASoC: fsl: use snd_soc_{of_}get_dlc() ASoC: qcom: use snd_soc_{of_}get_dlc() ASoC: meson: use snd_soc_{of_}get_dlc() ASoC: samsung: use snd_soc_{of_}get_dlc() ASoC: loongson: use snd_soc_{of_}get_dlc() ASoC: soc-core.c: use snd_soc_{of_}get_dlc() ASoC: simple-card.c: use snd_soc_{of_}get_dlc() include/sound/soc.h | 8 ++- sound/soc/fsl/imx-card.c | 14 +---- sound/soc/fsl/imx-rpmsg.c | 3 +- sound/soc/generic/simple-card-utils.c | 4 +- sound/soc/generic/simple-card.c | 4 +- sound/soc/loongson/loongson_card.c | 34 ++++------ sound/soc/mediatek/mt8173/mt8173-rt5650.c | 2 +- sound/soc/meson/axg-card.c | 3 +- sound/soc/meson/gx-card.c | 3 +- sound/soc/meson/meson-card-utils.c | 16 ++--- sound/soc/meson/meson-card.h | 3 +- sound/soc/qcom/common.c | 13 +--- sound/soc/samsung/odroid.c | 16 +---- sound/soc/soc-core.c | 77 ++++++++++++----------- 14 files changed, 79 insertions(+), 121 deletions(-) -- 2.25.1

2 11

[PATCH] ASoC: intel: sof_sdw: Fixup typo in device link checking
by Charles Keepax 21 Jun '23

21 Jun '23

The loop checking for multiple different devices on a single sdw link contains a typo accidentally using i twice instead of j. Correct to the correct index variable. Fixes: dc5a3e60a4b5 ("ASoC: Intel: sof_sdw: append codec type to dai link name") Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/intel/boards/sof_sdw.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 6faf4a43eaf5a..144f082c63fda 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1347,7 +1347,7 @@ static int sof_card_dai_links_create(struct device *dev, if ((SDW_PART_ID(adr_link->adr_d[i].adr) != SDW_PART_ID(adr_link->adr_d[j].adr)) || (SDW_MFG_ID(adr_link->adr_d[i].adr) != - SDW_MFG_ID(adr_link->adr_d[i].adr))) { + SDW_MFG_ID(adr_link->adr_d[j].adr))) { append_codec_type = true; goto out; } -- 2.30.2

2 1

[PATCH 0/8] add snd_soc_{of_}get_dlc()
by Kuninori Morimoto 21 Jun '23

21 Jun '23

Hi Mark Current soc-core.c has snd_soc_{of_}get_dai_name() to get DAI name for dlc (snd_soc_dai_link_component). It gets .dai_name, but we need .of_node too. Therefor user need to arrange. It will be more useful if it gets both .dai_name and .of_node. This patch adds snd_soc_{of_}get_dlc() for it, and convert to use it. Kuninori Morimoto (8): ASoC: soc-core.c: add snd_soc_{of_}get_dlc() ASoC: soc-core.c: add index on snd_soc_of_get_dai_name() ASoC: fsl: use snd_soc_{of_}get_dlc() ASoC: qcom: use snd_soc_{of_}get_dlc() ASoC: meson: use snd_soc_{of_}get_dlc() ASoC: samsung: use snd_soc_{of_}get_dlc() ASoC: soc-core.c: use snd_soc_{of_}get_dlc() ASoC: simple-card.c: use snd_soc_{of_}get_dlc() include/sound/soc.h | 8 ++- sound/soc/fsl/imx-card.c | 14 +---- sound/soc/fsl/imx-rpmsg.c | 3 +- sound/soc/generic/simple-card-utils.c | 4 +- sound/soc/generic/simple-card.c | 4 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 2 +- sound/soc/meson/axg-card.c | 3 +- sound/soc/meson/gx-card.c | 3 +- sound/soc/meson/meson-card-utils.c | 16 ++--- sound/soc/meson/meson-card.h | 3 +- sound/soc/qcom/common.c | 13 +--- sound/soc/samsung/odroid.c | 16 +---- sound/soc/soc-core.c | 77 ++++++++++++----------- 13 files changed, 68 insertions(+), 98 deletions(-) -- 2.25.1

2 10

[PATCH][next] ASoC: tas2781: Fix spelling mistake "calibraiton" -> "calibration"
by Colin Ian King 20 Jun '23

20 Jun '23

There is a spelling mistake in a dev_err message. Fix it. Also fix grammar and add space between last word and (%d)". Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- sound/soc/codecs/tas2781-fmwlib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index 432b19ccec8c..cbf0aef2c001 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -1863,7 +1863,7 @@ static int fw_parse_calibration_data(struct tasdevice_priv *tas_priv, if (tas_fmw->nr_calibrations != 1) { dev_err(tas_priv->dev, - "%s: only support one calibraiton(%d)!\n", + "%s: only supports one calibration (%d)!\n", __func__, tas_fmw->nr_calibrations); goto out; } -- 2.39.2

2 1

[PATCH 1/1] ASoC: dwc: add DMA handshake control
by Maxim Kochetkov 20 Jun '23

20 Jun '23

DMA mode uses hardware handshake signals. DMACR register is used to enable the DMA Controller interface operation. So add DMA enable/disable to i2s_start()/i2s_stop() functions if using DMA mode. Signed-off-by: Maxim Kochetkov <fido_max(a)inbox.ru> --- sound/soc/dwc/dwc-i2s.c | 39 +++++++++++++++++++++++++++++++++++++-- sound/soc/dwc/local.h | 6 ++++++ 2 files changed, 43 insertions(+), 2 deletions(-) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 9e7065dd854c..02b9894e99a7 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -150,19 +150,51 @@ static irqreturn_t i2s_irq_handler(int irq, void *dev_id) return IRQ_NONE; } +static void i2s_enable_dma(struct dw_i2s_dev *dev, u32 stream) +{ + u32 dma_reg = i2s_read_reg(dev->i2s_base, I2S_DMACR); + + /* Enable DMA handshake for stream */ + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + dma_reg |= I2S_DMAEN_TXBLOCK; + else + dma_reg |= I2S_DMAEN_RXBLOCK; + + i2s_write_reg(dev->i2s_base, I2S_DMACR, dma_reg); +} + +static void i2s_disable_dma(struct dw_i2s_dev *dev, u32 stream) +{ + u32 dma_reg = i2s_read_reg(dev->i2s_base, I2S_DMACR); + + /* Disable DMA handshake for stream */ + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + dma_reg &= ~I2S_DMAEN_TXBLOCK; + i2s_write_reg(dev->i2s_base, I2S_RTXDMA, 1); + } else { + dma_reg &= ~I2S_DMAEN_RXBLOCK; + i2s_write_reg(dev->i2s_base, I2S_RRXDMA, 1); + } + i2s_write_reg(dev->i2s_base, I2S_DMACR, dma_reg); +} + static void i2s_start(struct dw_i2s_dev *dev, struct snd_pcm_substream *substream) { struct i2s_clk_config_data *config = &dev->config; i2s_write_reg(dev->i2s_base, IER, 1); - i2s_enable_irqs(dev, substream->stream, config->chan_nr); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) i2s_write_reg(dev->i2s_base, ITER, 1); else i2s_write_reg(dev->i2s_base, IRER, 1); + if (dev->use_pio) + i2s_enable_irqs(dev, substream->stream, config->chan_nr); + else + i2s_enable_dma(dev, substream->stream); + i2s_write_reg(dev->i2s_base, CER, 1); } @@ -176,7 +208,10 @@ static void i2s_stop(struct dw_i2s_dev *dev, else i2s_write_reg(dev->i2s_base, IRER, 0); - i2s_disable_irqs(dev, substream->stream, 8); + if (dev->use_pio) + i2s_disable_irqs(dev, substream->stream, 8); + else + i2s_disable_dma(dev, substream->stream); if (!dev->active) { i2s_write_reg(dev->i2s_base, CER, 0); diff --git a/sound/soc/dwc/local.h b/sound/soc/dwc/local.h index d64bd4f8fd34..ba4e397099be 100644 --- a/sound/soc/dwc/local.h +++ b/sound/soc/dwc/local.h @@ -53,6 +53,12 @@ #define I2S_COMP_VERSION 0x01F8 #define I2S_COMP_TYPE 0x01FC +#define I2S_RRXDMA 0x01C4 +#define I2S_RTXDMA 0x01CC +#define I2S_DMACR 0x0200 +#define I2S_DMAEN_RXBLOCK (1 << 16) +#define I2S_DMAEN_TXBLOCK (1 << 17) + /* * Component parameter register fields - define the I2S block's * configuration. -- 2.40.1

2 1

[PATCH 1/1] ASoC: dwc: don't assign addr_width for dt configs
by Maxim Kochetkov 20 Jun '23

20 Jun '23

For proper DMA operation addr_width must corresponds with audio format (S16, S24, S32, etc). Proper bus width calculations is performed by snd_hwparams_to_dma_slave_config(). So drop wrong addr_width asignment for dt configs and let snd_hwparams_to_dma_slave_config() do the job. Signed-off-by: Maxim Kochetkov <fido_max(a)inbox.ru> --- sound/soc/dwc/dwc-i2s.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 9e7065dd854c..21398ee924e4 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -558,13 +558,9 @@ static int dw_configure_dai_by_dt(struct dw_i2s_dev *dev, u32 comp1 = i2s_read_reg(dev->i2s_base, I2S_COMP_PARAM_1); u32 comp2 = i2s_read_reg(dev->i2s_base, I2S_COMP_PARAM_2); u32 fifo_depth = 1 << (1 + COMP1_FIFO_DEPTH_GLOBAL(comp1)); - u32 idx = COMP1_APB_DATA_WIDTH(comp1); u32 idx2; int ret; - if (WARN_ON(idx >= ARRAY_SIZE(bus_widths))) - return -EINVAL; - ret = dw_configure_dai(dev, dw_i2s_dai, SNDRV_PCM_RATE_8000_192000); if (ret < 0) return ret; @@ -574,7 +570,6 @@ static int dw_configure_dai_by_dt(struct dw_i2s_dev *dev, dev->capability |= DWC_I2S_PLAY; dev->play_dma_data.dt.addr = res->start + I2S_TXDMA; - dev->play_dma_data.dt.addr_width = bus_widths[idx]; dev->play_dma_data.dt.fifo_size = fifo_depth * (fifo_width[idx2]) >> 8; dev->play_dma_data.dt.maxburst = 16; @@ -584,7 +579,6 @@ static int dw_configure_dai_by_dt(struct dw_i2s_dev *dev, dev->capability |= DWC_I2S_RECORD; dev->capture_dma_data.dt.addr = res->start + I2S_RXDMA; - dev->capture_dma_data.dt.addr_width = bus_widths[idx]; dev->capture_dma_data.dt.fifo_size = fifo_depth * (fifo_width[idx2] >> 8); dev->capture_dma_data.dt.maxburst = 16; -- 2.40.1

2 1

[PATCH v3 00/11] ASoC: qcom: audioreach: add compress offload support
by Srinivas Kandagatla 20 Jun '23

20 Jun '23

This patchset adds compressed offload support to Qualcomm audioreach drivers. Currently it supports AAC, MP3 and FALC along with gapless. Tested this on SM8450 and sc7280. thanks, srini Changes since v2: - fixed order of sign-off - added placeholder decoder and encoder to audioreach_set_media_format Mohammad Rafi Shaik (4): ASoC: qcom: SC7280: audioreach: Add sc7280 hardware param fixup callback ASoC: q6dsp: q6apm: add end of stream events ASoC: q6dsp: audioreach: Add support to set compress format params ASoC: q6dsp: audioreach: Add gapless feature support Srinivas Kandagatla (7): ASoC: q6dsp: audioreach: add helper function to set u32 param ASoC: q6dsp: audioreach: Add placeholder decoder for compress playback ASoC: q6dsp: q6apm-dai: Add open/free compress DAI callbacks ASoC: q6dsp: q6apm-dai: Add compress DAI and codec caps get callbacks ASoC: q6dsp: q6apm-dai: Add trigger/pointer compress DAI callbacks ASoC: q6dsp: q6apm-dai: Add compress set params and metadata DAI callbacks ASoC: q6dsp: q6apm-dai: Add mmap and copy compress DAI callbacks sound/soc/qcom/qdsp6/audioreach.c | 250 ++++++++++------- sound/soc/qcom/qdsp6/audioreach.h | 51 ++++ sound/soc/qcom/qdsp6/q6apm-dai.c | 445 ++++++++++++++++++++++++++++++ sound/soc/qcom/qdsp6/q6apm.c | 68 +++++ sound/soc/qcom/qdsp6/q6apm.h | 6 + sound/soc/qcom/sc7280.c | 23 +- 6 files changed, 747 insertions(+), 96 deletions(-) -- 2.21.0

2 12

[PATCH 0/2] ASoC: Use maple tree register cache for Everest Semi CODECs
by Mark Brown 20 Jun '23

20 Jun '23

Several of the Everest Semi CODECs only support single register read and write operations and therefore do not benefit from using the rbtree cache over the maple tree cache, convert them to the more modern maple tree cache. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Mark Brown (2): ASoC: es8316: Use maple tree register cache ASoC: es8328: Use maple tree register cache sound/soc/codecs/es8316.c | 2 +- sound/soc/codecs/es8328.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) --- base-commit: 9561de3a55bed6bdd44a12820ba81ec416e705a7 change-id: 20230609-asoc-es-maple-3b696631e609 Best regards, -- Mark Brown <broonie(a)kernel.org>

1 3

[regression] Bug 216818 - The microphone mute led not working after linux 6
by Thorsten Leemhuis 20 Jun '23

20 Jun '23

Hi, this is your Linux kernel regression tracker speaking. I noticed a regression report in bugzilla.kernel.org. As many (most?) kernel developer don't keep an eye on it, I decided to forward it by mail. Quoting from https://bugzilla.kernel.org/show_bug.cgi?id=216818 : > sonic82003(a)gmail.com 2022-12-18 08:52:32 UTC > > The mic mute led of my ThinkPad X1 Carbon Gen 9 doesn't work anymore after updating linux to version 6. > I can still turn it on by running > > echo 1 > /sys/class/leds/platform::micmute/brightness > > With linux-lts it still works fine. See the ticket for more details. Note, I found a similar report that (despite my attempts to prevent things like this from happening) fell through the cracks here: https://bugzilla.kernel.org/show_bug.cgi?id=216355 > plum 2022-08-13 02:11:01 UTC > > I upgrade to kernel 5.19.1 but found my thinkpad x1 carbon 2021's mute led stop working. > > Function is okay but LED won't light up. > > Back to kernel 5.18 and it's normal and working again. > > Fedora 36 64 bit > Gnome-shell 42 >From a quick research it looks to me like this is an issue for the sounds maintainers, as the LED itself apparently works. If that is something for the platform people instead please speak up. BTW, let me use this mail to also add the report to the list of tracked regressions to ensure it's doesn't fall through the cracks: #regzbot introduced: v5.18..v5.19 https://bugzilla.kernel.org/show_bug.cgi?id=216818 #regzbot ignore-activity Ciao, Thorsten (wearing his 'the Linux kernel's regression tracker' hat) P.S.: As the Linux kernel's regression tracker I deal with a lot of reports and sometimes miss something important when writing mails like this. If that's the case here, don't hesitate to tell me in a public reply, it's in everyone's interest to set the public record straight.

5 5

[bug report] ASoC: loongson: Add Loongson ASoC Sound Card Support
by Dan Carpenter 20 Jun '23

20 Jun '23

Hello Yingkun Meng, The patch d24028606e76: "ASoC: loongson: Add Loongson ASoC Sound Card Support" from Jun 14, 2023, leads to the following Smatch static checker warning: sound/soc/loongson/loongson_card.c:157 loongson_card_parse_of() error: uninitialized symbol 'cpu_dai_name'. sound/soc/loongson/loongson_card.c:158 loongson_card_parse_of() error: uninitialized symbol 'codec_dai_name'. sound/soc/loongson/loongson_card.c 115 static int loongson_card_parse_of(struct loongson_card_data *data) 116 { 117 const char *cpu_dai_name, *codec_dai_name; 118 struct device_node *cpu, *codec; 119 struct snd_soc_card *card = &data->snd_card; 120 struct device *dev = card->dev; 121 struct of_phandle_args args; 122 int ret, i; 123 124 cpu = of_get_child_by_name(dev->of_node, "cpu"); 125 if (!cpu) { 126 dev_err(dev, "platform property missing or invalid\n"); 127 return -EINVAL; 128 } 129 codec = of_get_child_by_name(dev->of_node, "codec"); 130 if (!codec) { 131 dev_err(dev, "audio-codec property missing or invalid\n"); 132 ret = -EINVAL; 133 goto err; 134 } 135 136 ret = of_parse_phandle_with_args(cpu, "sound-dai", 137 "#sound-dai-cells", 0, &args); 138 if (ret) { 139 dev_err(dev, "codec node missing #sound-dai-cells\n"); 140 goto err; 141 } 142 for (i = 0; i < card->num_links; i++) 143 loongson_dai_links[i].cpus->of_node = args.np; 144 145 ret = of_parse_phandle_with_args(codec, "sound-dai", 146 "#sound-dai-cells", 0, &args); 147 if (ret) { 148 dev_err(dev, "codec node missing #sound-dai-cells\n"); 149 goto err; 150 } 151 for (i = 0; i < card->num_links; i++) 152 loongson_dai_links[i].codecs->of_node = args.np; 153 154 snd_soc_of_get_dai_name(cpu, &cpu_dai_name); 155 snd_soc_of_get_dai_name(codec, &codec_dai_name); It looks like snd_soc_of_get_dai_name() can fail but I can add an exception for that if it never fails in real life. 156 for (i = 0; i < card->num_links; i++) { --> 157 loongson_dai_links[i].cpus->dai_name = cpu_dai_name; 158 loongson_dai_links[i].codecs->dai_name = codec_dai_name; 159 } 160 of_node_put(cpu); 161 of_node_put(codec); 162 163 return 0; 164 165 err: 166 of_node_put(cpu); 167 of_node_put(codec); 168 return ret; 169 } regards, dan carpenter

2 1

[PATCH v6 1/4] ASoC: tas2781: Add Header file for tas2781 driver
by Shenghao Ding 20 Jun '23

20 Jun '23

Create Header file for tas2781 driver. Signed-off-by: Shenghao Ding <13916275206(a)139.com> --- Changes in v6: - remove /* __TAS2781_LIB_H__ */ at the end of the file - remove #include <linux/kernel.h> in the tas2781.h - remove enum channel in tas2781-dsp.h - Add TASDEVICE_MAX_CHANNELS in tas2781-dsp.h - remove global addr related definition in tas2781.h - reorder the member in tasdevice_priv - fix the firmware version compatibility issue --- include/sound/tas2781-dsp.h | 183 ++++++++++++++++++++++++++++++++++++ include/sound/tas2781-tlv.h | 21 +++++ include/sound/tas2781.h | 164 ++++++++++++++++++++++++++++++++ 3 files changed, 368 insertions(+) create mode 100644 include/sound/tas2781-dsp.h create mode 100644 include/sound/tas2781-tlv.h create mode 100644 include/sound/tas2781.h diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h new file mode 100644 index 000000000000..bd1b72bf47a5 --- /dev/null +++ b/include/sound/tas2781-dsp.h @@ -0,0 +1,183 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier +// +// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2781 driver implements a flexible and configurable +// algo coefficient setting for one, two, or even multiple +// TAS2781 chips. +// +// Author: Shenghao Ding <shenghao-ding(a)ti.com> +// Author: Kevin Lu <kevin-lu(a)ti.com> +// + +#ifndef __TASDEVICE_DSP_H__ +#define __TASDEVICE_DSP_H__ + +#define MAIN_ALL_DEVICES 0x0d +#define MAIN_DEVICE_A 0x01 +#define MAIN_DEVICE_B 0x08 +#define MAIN_DEVICE_C 0x10 +#define MAIN_DEVICE_D 0x14 +#define COEFF_DEVICE_A 0x03 +#define COEFF_DEVICE_B 0x0a +#define COEFF_DEVICE_C 0x11 +#define COEFF_DEVICE_D 0x15 +#define PRE_DEVICE_A 0x04 +#define PRE_DEVICE_B 0x0b +#define PRE_DEVICE_C 0x12 +#define PRE_DEVICE_D 0x16 + +#define PPC3_VERSION 0x4100 +#define PPC3_VERSION_TAS2781 0x14600 +#define TASDEVICE_DEVICE_SUM 8 +#define TASDEVICE_CONFIG_SUM 64 + +#define TASDEVICE_MAX_CHANNELS 8 + +enum tasdevice_dsp_dev_idx { + TASDEVICE_DSP_TAS_2555 = 0, + TASDEVICE_DSP_TAS_2555_STEREO, + TASDEVICE_DSP_TAS_2557_MONO, + TASDEVICE_DSP_TAS_2557_DUAL_MONO, + TASDEVICE_DSP_TAS_2559, + TASDEVICE_DSP_TAS_2563, + TASDEVICE_DSP_TAS_2563_DUAL_MONO = 7, + TASDEVICE_DSP_TAS_2563_QUAD, + TASDEVICE_DSP_TAS_2563_21, + TASDEVICE_DSP_TAS_2781, + TASDEVICE_DSP_TAS_2781_DUAL_MONO, + TASDEVICE_DSP_TAS_2781_21, + TASDEVICE_DSP_TAS_2781_QUAD, + TASDEVICE_DSP_TAS_MAX_DEVICE +}; + +struct tasdevice_fw_fixed_hdr { + unsigned int fwsize; + unsigned int ppcver; + unsigned int drv_ver; +}; + +struct tasdevice_dspfw_hdr { + struct tasdevice_fw_fixed_hdr fixed_hdr; + unsigned short device_family; + unsigned short device; + unsigned char ndev; +}; + +struct tasdev_blk { + int nr_retry; + unsigned int type; + unsigned char is_pchksum_present; + unsigned char pchksum; + unsigned char is_ychksum_present; + unsigned char ychksum; + unsigned int nr_cmds; + unsigned int blk_size; + unsigned int nr_subblocks; + unsigned char *data; +}; + +struct tasdevice_data { + char name[64]; + unsigned int nr_blk; + struct tasdev_blk *dev_blks; +}; + +struct tasdevice_prog { + unsigned int prog_size; + struct tasdevice_data dev_data; +}; + +struct tasdevice_config { + unsigned int cfg_size; + char name[64]; + struct tasdevice_data dev_data; +}; + +struct tasdevice_calibration { + struct tasdevice_data dev_data; +}; + +struct tasdevice_fw { + struct tasdevice_dspfw_hdr fw_hdr; + unsigned short nr_programs; + struct tasdevice_prog *programs; + unsigned short nr_configurations; + struct tasdevice_config *configs; + unsigned short nr_calibrations; + struct tasdevice_calibration *calibrations; + struct device *dev; +}; + +enum tasdevice_dsp_fw_state { + TASDEVICE_DSP_FW_NONE = 0, + TASDEVICE_DSP_FW_PENDING, + TASDEVICE_DSP_FW_FAIL, + TASDEVICE_DSP_FW_ALL_OK, +}; + +enum tasdevice_bin_blk_type { + TASDEVICE_BIN_BLK_COEFF = 1, + TASDEVICE_BIN_BLK_POST_POWER_UP, + TASDEVICE_BIN_BLK_PRE_SHUTDOWN, + TASDEVICE_BIN_BLK_PRE_POWER_UP, + TASDEVICE_BIN_BLK_POST_SHUTDOWN +}; + +struct tasdevice_rca_hdr { + unsigned int img_sz; + unsigned int checksum; + unsigned int binary_version_num; + unsigned int drv_fw_version; + unsigned char plat_type; + unsigned char dev_family; + unsigned char reserve; + unsigned char ndev; + unsigned char devs[TASDEVICE_DEVICE_SUM]; + unsigned int nconfig; + unsigned int config_size[TASDEVICE_CONFIG_SUM]; +}; + +struct tasdev_blk_data { + unsigned char dev_idx; + unsigned char block_type; + unsigned short yram_checksum; + unsigned int block_size; + unsigned int n_subblks; + unsigned char *regdata; +}; + +struct tasdevice_config_info { + unsigned int nblocks; + unsigned int real_nblocks; + unsigned char active_dev; + struct tasdev_blk_data **blk_data; +}; + +struct tasdevice_rca { + struct tasdevice_rca_hdr fw_hdr; + int ncfgs; + struct tasdevice_config_info **cfg_info; + int profile_cfg_id; +}; + +void tasdevice_select_cfg_blk(void *context, int conf_no, + unsigned char block_type); +void tasdevice_config_info_remove(void *context); +void tasdevice_dsp_remove(void *context); +int tasdevice_dsp_parser(void *context); +int tasdevice_rca_parser(void *context, const struct firmware *fmw); +void tasdevice_dsp_remove(void *context); +void tasdevice_calbin_remove(void *context); +int tasdevice_select_tuningprm_cfg(void *context, int prm, + int cfg_no, int rca_conf_no); +int tasdevice_prmg_load(void *context, int prm_no); +int tasdevice_prmg_calibdata_load(void *context, int prm_no); +void tasdevice_tuning_switch(void *context, int state); +int tas2781_load_calibration(void *context, char *file_name, + unsigned short i); + +#endif diff --git a/include/sound/tas2781-tlv.h b/include/sound/tas2781-tlv.h new file mode 100644 index 000000000000..4038dd421150 --- /dev/null +++ b/include/sound/tas2781-tlv.h @@ -0,0 +1,21 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier +// +// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2781 driver implements a flexible and configurable +// algo coefficient setting for one, two, or even multiple +// TAS2781 chips. +// +// Author: Shenghao Ding <shenghao-ding(a)ti.com> +// + +#ifndef __TAS2781_TLV_H__ +#define __TAS2781_TLV_H__ + +static const DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 100, 0); +static const DECLARE_TLV_DB_SCALE(amp_vol_tlv, 1100, 50, 0); + +#endif diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h new file mode 100644 index 000000000000..9de5c57c148c --- /dev/null +++ b/include/sound/tas2781.h @@ -0,0 +1,164 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier +// +// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2781 driver implements a flexible and configurable +// algo coefficient setting for one, two, or even multiple +// TAS2781 chips. +// +// Author: Shenghao Ding <shenghao-ding(a)ti.com> +// Author: Kevin Lu <kevin-lu(a)ti.com> +// + +#ifndef __TAS2781_H__ +#define __TAS2781_H__ + +#include "tas2781-dsp.h" + +/* version number */ +#define TAS2781_DRV_VER 1 +#define SMARTAMP_MODULE_NAME "tas2781" +#define TAS2781_GLOBAL_ADDR 0x40 +#define TASDEVICE_RATES (SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 |\ + SNDRV_PCM_RATE_88200) + +#define TASDEVICE_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/*PAGE Control Register (available in page0 of each book) */ +#define TASDEVICE_PAGE_SELECT 0x00 +#define TASDEVICE_BOOKCTL_PAGE 0x00 +#define TASDEVICE_BOOKCTL_REG 127 +#define TASDEVICE_BOOK_ID(reg) (reg / (256 * 128)) +#define TASDEVICE_PAGE_ID(reg) ((reg % (256 * 128)) / 128) +#define TASDEVICE_PAGE_REG(reg) ((reg % (256 * 128)) % 128) +#define TASDEVICE_PGRG(reg) (reg % (256 * 128)) +#define TASDEVICE_REG(book, page, reg) (((book * 256 * 128) + \ + (page * 128)) + reg) + +/*Software Reset */ +#define TAS2781_REG_SWRESET TASDEVICE_REG(0x0, 0X0, 0x01) +#define TAS2781_REG_SWRESET_RESET BIT(0) + +/*I2C Checksum */ +#define TASDEVICE_I2CChecksum TASDEVICE_REG(0x0, 0x0, 0x7E) + +/* Volume control */ +#define TAS2781_DVC_LVL TASDEVICE_REG(0x0, 0x0, 0x1A) +#define TAS2781_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) +#define TAS2781_AMP_LEVEL_MASK GENMASK(5, 1) + +#define TASDEVICE_CMD_SING_W 0x1 +#define TASDEVICE_CMD_BURST 0x2 +#define TASDEVICE_CMD_DELAY 0x3 +#define TASDEVICE_CMD_FIELD_W 0x4 + +enum audio_device { + TAS2781 = 0, +}; + +enum device_catlog_id { + LENOVO = 0, + OTHERS +}; + +struct tasdevice { + struct tasdevice_fw *cali_data_fmw; + unsigned int dev_addr; + unsigned int err_code; + unsigned char cur_book; + short cur_prog; + short cur_conf; + bool is_loading; + bool is_loaderr; +}; + +struct tasdevice_irqinfo { + int irq_gpio; + int irq; +}; + +struct calidata { + unsigned char *data; + unsigned long total_sz; +}; + +struct tasdevice_priv { + struct tasdevice tasdevice[TASDEVICE_MAX_CHANNELS]; + struct tasdevice_irqinfo irq_info; + struct tasdevice_rca rcabin; + struct calidata cali_data; + struct tasdevice_fw *fmw; + struct gpio_desc *reset; + struct mutex codec_lock; + struct regmap *regmap; + struct device *dev; + struct tm tm; + + enum device_catlog_id catlog_id; + const char *acpi_subsystem_id; + unsigned char cal_binaryname[TASDEVICE_MAX_CHANNELS][64]; + unsigned char crc8_lkp_tbl[CRC8_TABLE_SIZE]; + unsigned char coef_binaryname[64]; + unsigned char rca_binaryname[64]; + unsigned char dev_name[32]; + unsigned char ndev; + unsigned int magic_num; + unsigned int chip_id; + unsigned int sysclk; + + int cur_prog; + int cur_conf; + int fw_state; + int index; + void *client; + void *codec; + bool force_fwload_status; + bool playback_started; + bool isacpi; + int (*fw_parse_variable_header)(struct tasdevice_priv *tas_priv, + const struct firmware *fmw, int offset); + int (*fw_parse_program_data)(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset); + int (*fw_parse_configuration_data)(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset); + int (*tasdevice_load_block)(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block); +}; + +void tas2781_reset(struct tasdevice_priv *tas_dev); +int tascodec_init(struct tasdevice_priv *tas_priv, void *codec, + void (*cont)(const struct firmware *fw, void *context)); +struct tasdevice_priv *tasdevice_kzalloc(struct i2c_client *i2c); +int tasdevice_init(struct tasdevice_priv *tas_priv); +void tasdevice_remove(struct tasdevice_priv *tas_priv); +int tasdevice_dev_read(struct tasdevice_priv *tas_priv, + unsigned short chn, unsigned int reg, unsigned int *value); +int tasdevice_dev_write(struct tasdevice_priv *tas_priv, + unsigned short chn, unsigned int reg, unsigned int value); +int tasdevice_dev_bulk_write( + struct tasdevice_priv *tas_priv, unsigned short chn, + unsigned int reg, unsigned char *p_data, unsigned int n_length); +int tasdevice_dev_bulk_read(struct tasdevice_priv *tas_priv, + unsigned short chn, unsigned int reg, unsigned char *p_data, + unsigned int n_length); +int tasdevice_dev_update_bits( + struct tasdevice_priv *tasdevice, unsigned short chn, + unsigned int reg, unsigned int mask, unsigned int value); +int tasdevice_amp_putvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc); +int tasdevice_amp_getvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc); +int tasdevice_digital_putvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc); +int tasdevice_digital_getvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc); + +#endif /* __TAS2781_H__ */ -- 2.34.1

3 5

[PATCH] ASoC: max98388: fix error code in probe()
by Dan Carpenter 20 Jun '23

20 Jun '23

This seems like a copy and paste bug. Return the correct variable. It should be "ret" instead of PTR_ERR(max98388->regmap). Fixes: 6a8e1d46f062 ("ASoC: max98388: add amplifier driver") Signed-off-by: Dan Carpenter <dan.carpenter(a)linaro.org> --- sound/soc/codecs/max98388.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98388.c b/sound/soc/codecs/max98388.c index 8062a7115007..74fbaf7d7c39 100644 --- a/sound/soc/codecs/max98388.c +++ b/sound/soc/codecs/max98388.c @@ -960,7 +960,7 @@ static int max98388_i2c_probe(struct i2c_client *i2c) ret = regmap_read(max98388->regmap, MAX98388_R22FF_REV_ID, &reg); if (ret < 0) - return dev_err_probe(&i2c->dev, PTR_ERR(max98388->regmap), + return dev_err_probe(&i2c->dev, ret, "Failed to read the revision ID\n"); dev_info(&i2c->dev, "MAX98388 revisionID: 0x%02X\n", reg); -- 2.39.2

2 1

[PATCH] ASoC: loongson: fix error codes in loongson_card_parse_acpi()
by Dan Carpenter 20 Jun '23

20 Jun '23

The acpi_node_get_property_reference() function returns kernel error codes and not ACPI error codes. So, although it does not affect the compiled code, using the ACPI_FAILURE() macro is wrong. Secondly, if the is_acpi_device_node() function returns false, then we should return -ENOENT instead of returning success. Fixes: d24028606e76 ("ASoC: loongson: Add Loongson ASoC Sound Card Support") Signed-off-by: Dan Carpenter <dan.carpenter(a)linaro.org> --- sound/soc/loongson/loongson_card.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/loongson/loongson_card.c b/sound/soc/loongson/loongson_card.c index 965eaf4e9109..08df05cb4328 100644 --- a/sound/soc/loongson/loongson_card.c +++ b/sound/soc/loongson/loongson_card.c @@ -81,9 +81,9 @@ static int loongson_card_parse_acpi(struct loongson_card_data *data) /* fixup platform name based on reference node */ memset(&args, 0, sizeof(args)); ret = acpi_node_get_property_reference(fwnode, "cpu", 0, &args); - if (ACPI_FAILURE(ret) || !is_acpi_device_node(args.fwnode)) { + if (ret || !is_acpi_device_node(args.fwnode)) { dev_err(card->dev, "No matching phy in ACPI table\n"); - return ret; + return ret ?: -ENOENT; } adev = to_acpi_device_node(args.fwnode); phy_dev = acpi_get_first_physical_node(adev); @@ -95,9 +95,9 @@ static int loongson_card_parse_acpi(struct loongson_card_data *data) /* fixup codec name based on reference node */ memset(&args, 0, sizeof(args)); ret = acpi_node_get_property_reference(fwnode, "codec", 0, &args); - if (ACPI_FAILURE(ret) || !is_acpi_device_node(args.fwnode)) { + if (ret || !is_acpi_device_node(args.fwnode)) { dev_err(card->dev, "No matching phy in ACPI table\n"); - return ret; + return ret ?: -ENOENT; } adev = to_acpi_device_node(args.fwnode); snprintf(codec_name, sizeof(codec_name), "i2c-%s", acpi_dev_name(adev)); -- 2.39.2

2 1

[PATCH v1] ASoC: loongson: change the type of variable irq to int
by YingKun Meng 20 Jun '23

20 Jun '23

We use variable 'irq' to store the return value of fwnode_get_irq_byname(). A negative value indicates that the operation failed. If the type of 'irq' is unsigned int, we never know if the operation failed. Reported-by: Harshit Mogalapalli <harshit.m.mogalapalli(a)oracle.com> Closes: https://lore.kernel.org/loongarch/325dd825-6fa5-0ebc-4b7e-7acf2d2840e4@loon… Signed-off-by: YingKun Meng <mengyingkun(a)loongson.cn> --- sound/soc/loongson/loongson_i2s.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/loongson/loongson_i2s.h b/sound/soc/loongson/loongson_i2s.h index 52788f6a94ad..89492eebf834 100644 --- a/sound/soc/loongson/loongson_i2s.h +++ b/sound/soc/loongson/loongson_i2s.h @@ -45,7 +45,7 @@ struct loongson_dma_data { dma_addr_t dev_addr; /* device physical address for DMA */ void __iomem *order_addr; /* DMA order register */ - u32 irq; /* DMA irq */ + int irq; /* DMA irq */ }; struct loongson_i2s { -- 2.33.0

2 1

[PATCH] ASoC: imx-audmix: check return value of devm_kasprintf()
by Claudiu Beznea 20 Jun '23

20 Jun '23

devm_kasprintf() returns a pointer to dynamically allocated memory. Pointer could be NULL in case allocation fails. Check pointer validity. Identified with coccinelle (kmerr.cocci script). Fixes: b86ef5367761 ("ASoC: fsl: Add Audio Mixer machine driver") Signed-off-by: Claudiu Beznea <claudiu.beznea(a)microchip.com> --- Hi, This has been addressed using kmerr.cocci script proposed for update at [1]. Thank you, Claudiu Beznea [1] https://lore.kernel.org/all/20230530074044.1603426-1-claudiu.beznea@microch… sound/soc/fsl/imx-audmix.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c index 2c57fe9d2d08..af06268ee57b 100644 --- a/sound/soc/fsl/imx-audmix.c +++ b/sound/soc/fsl/imx-audmix.c @@ -228,6 +228,8 @@ static int imx_audmix_probe(struct platform_device *pdev) dai_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s%s", fe_name_pref, args.np->full_name + 1); + if (!dai_name) + return -ENOMEM; dev_info(pdev->dev.parent, "DAI FE name:%s\n", dai_name); @@ -236,6 +238,8 @@ static int imx_audmix_probe(struct platform_device *pdev) capture_dai_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s %s", dai_name, "CPU-Capture"); + if (!capture_dai_name) + return -ENOMEM; } priv->dai[i].cpus = &dlc[0]; @@ -263,6 +267,8 @@ static int imx_audmix_probe(struct platform_device *pdev) "AUDMIX-Playback-%d", i); be_cp = devm_kasprintf(&pdev->dev, GFP_KERNEL, "AUDMIX-Capture-%d", i); + if (!be_name || !be_pb || !be_cp) + return -ENOMEM; priv->dai[num_dai + i].cpus = &dlc[2]; priv->dai[num_dai + i].codecs = &dlc[3]; @@ -287,6 +293,9 @@ static int imx_audmix_probe(struct platform_device *pdev) priv->dapm_routes[i].source = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s %s", dai_name, "CPU-Playback"); + if (!priv->dapm_routes[i].source) + return -ENOMEM; + priv->dapm_routes[i].sink = be_pb; priv->dapm_routes[num_dai + i].source = be_pb; priv->dapm_routes[num_dai + i].sink = be_cp; -- 2.34.1

3 2

[PATCH] ASoC: max98363: Remove cache defaults for volatile registers
by Mark Brown 20 Jun '23

20 Jun '23

The max98363 driver provides cache defaults for a number of volatile registers. This is not meaningful, the cache values will never be used so at best they will just consume memory and at worst they will be used in preference to real values from the device, remove them. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/max98363.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/codecs/max98363.c b/sound/soc/codecs/max98363.c index dcce06bff756..747941d9fd00 100644 --- a/sound/soc/codecs/max98363.c +++ b/sound/soc/codecs/max98363.c @@ -15,11 +15,6 @@ #include "max98363.h" static struct reg_default max98363_reg[] = { - {MAX98363_R2001_INTR_RAW, 0x0}, - {MAX98363_R2003_INTR_STATE, 0x0}, - {MAX98363_R2005_INTR_FALG, 0x0}, - {MAX98363_R2007_INTR_EN, 0x0}, - {MAX98363_R2009_INTR_CLR, 0x0}, {MAX98363_R2021_ERR_MON_CTRL, 0x0}, {MAX98363_R2022_SPK_MON_THRESH, 0x0}, {MAX98363_R2023_SPK_MON_DURATION, 0x0}, @@ -28,7 +23,6 @@ static struct reg_default max98363_reg[] = { {MAX98363_R2040_AMP_VOL, 0x0}, {MAX98363_R2041_AMP_GAIN, 0x5}, {MAX98363_R2042_DSP_CFG, 0x0}, - {MAX98363_R21FF_REV_ID, 0x0}, }; static bool max98363_readable_register(struct device *dev, unsigned int reg) --- base-commit: 44c026a73be8038f03dbdeef028b642880cf1511 change-id: 20230609-asoc-mx98363-volatile-8d88f2d7cfd8 Best regards, -- Mark Brown <broonie(a)kernel.org>

1 1

[PATCH] ALSA: hda/realtek: Add quirk for ASUS ROG G634Z
by Luke D. Jones 19 Jun '23

19 Jun '23

Adds the required quirk to enable the Cirrus amp and correct pins on the ASUS ROG G634Z series. While this works if the related _DSD properties are made available, these aren't included in the ACPI of these laptops (yet). Signed-off-by: Luke D. Jones <luke(a)ljones.dev> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a5d55a7063d3..437cf13d10d2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9552,6 +9552,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x1c62, "ASUS GU603", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1c92, "ASUS ROG Strix G15", ALC285_FIXUP_ASUS_G533Z_PINS), + SND_PCI_QUIRK(0x1043, 0x1caf, "ASUS G634JYR/JZR", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1d42, "ASUS Zephyrus G14 2022", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1d4e, "ASUS TM420", ALC256_FIXUP_ASUS_HPE), -- 2.40.1

2 1

v1.2.7: Regression detecting builtin microphone on Lenovo T14G1 (Intel Comet Lake)
by GitHub issues - opened 19 Jun '23

19 Jun '23

alsa-project/alsa-ucm-conf issue #325 was opened from myrkr: I'm currently testing an update of alsa-ucm-conf from version v1.2.5.1 to version v1.2.9 and noticed a regression. Kernel: v6.1.34 alsa-lib: v1.2.6.1 pulseaudio: 14.2 I use a Lenovo T14G1 with an HDMI monitor connected via USB-C (but the monitor is not in use by Xorg). I want to use the built-in speaker and microphone of the device. Before the update I got the following output from "pactl list": [pactl-before.log](https://github.com/alsa-project/alsa-ucm-conf/files/11788… ``` Card #0 Name: alsa_card.pci-0000_00_1f.3-platform-skl_hda_dsp_generic Driver: module-alsa-card.c Owner Module: 1 Properties: alsa.card = "0" alsa.card_name = "sof-hda-dsp" alsa.long_card_name = "LENOVO-20S1S19N00-ThinkPadT14Gen1" device.bus_path = "pci-0000:00:1f.3-platform-skl_hda_dsp_generic" sysfs.path = "/devices/pci0000:00/0000:00:1f.3/skl_hda_dsp_generic/sound/card0" device.bus = "pci" device.vendor.id = "8086" device.vendor.name = "Intel Corporation" device.product.id = "02c8" device.product.name = "Comet Lake PCH-LP cAVS" device.string = "0" device.description = "Comet Lake PCH-LP cAVS" module-udev-detect.discovered = "1" device.icon_name = "audio-card-pci" Profiles: HiFi: Play HiFi quality Music (sinks: 4, sources: 2, priority: 8000, available: yes) off: Off (sinks: 0, sources: 0, priority: 0, available: yes) Active Profile: HiFi Ports: [Out] HDMI3: HDMI / DisplayPort 3 Output (type: HDMI, priority: 700, latency offset: 0 usec, not available) Part of profile(s): HiFi [Out] HDMI2: HDMI / DisplayPort 2 Output (type: HDMI, priority: 600, latency offset: 0 usec, not available) Part of profile(s): HiFi [Out] HDMI1: HDMI / DisplayPort 1 Output (type: HDMI, priority: 500, latency offset: 0 usec, not available) Part of profile(s): HiFi [Out] Speaker: Speaker (type: Speaker, priority: 100, latency offset: 0 usec, availability unknown) Part of profile(s): HiFi [Out] Headphones: Headphones (type: Headphones, priority: 200, latency offset: 0 usec, not available) Part of profile(s): HiFi [In] Mic2: Headphones Stereo Microphone (type: Mic, priority: 200, latency offset: 0 usec, not available) Part of profile(s): HiFi [In] Mic1: Digital Microphone (type: Mic, priority: 100, latency offset: 0 usec, availability unknown) Part of profile(s): HiFi ``` On this cards the ports `Speaker` and `Mic1` are currently selected and usable. After the update to version v1.2.9 the output shows: [pactl-after.log](https://github.com/alsa-project/alsa-ucm-conf/files/117881… ``` Card #0 Name: alsa_card.pci-0000_00_1f.3-platform-skl_hda_dsp_generic Driver: module-alsa-card.c Owner Module: 1 Properties: alsa.card = "0" alsa.card_name = "sof-hda-dsp" alsa.long_card_name = "LENOVO-20S1S19N00-ThinkPadT14Gen1" device.bus_path = "pci-0000:00:1f.3-platform-skl_hda_dsp_generic" sysfs.path = "/devices/pci0000:00/0000:00:1f.3/skl_hda_dsp_generic/sound/card0" device.bus = "pci" device.vendor.id = "8086" device.vendor.name = "Intel Corporation" device.product.id = "02c8" device.product.name = "Comet Lake PCH-LP cAVS" device.string = "0" device.description = "Comet Lake PCH-LP cAVS" module-udev-detect.discovered = "1" device.icon_name = "audio-card-pci" Profiles: input:stereo-fallback: Stereo Input (sinks: 0, sources: 1, priority: 51, available: no) input:multichannel-input: Multichannel Input (sinks: 0, sources: 1, priority: 1, available: yes) output:stereo-fallback: Stereo Output (sinks: 1, sources: 0, priority: 5100, available: yes) output:stereo-fallback+input:stereo-fallback: Stereo Output + Stereo Input (sinks: 1, sources: 1, priority: 5151, available: no) output:stereo-fallback+input:multichannel-input: Stereo Output + Multichannel Input (sinks: 1, sources: 1, priority: 5101, available: yes) output:multichannel-output: Multichannel Output (sinks: 1, sources: 0, priority: 100, available: yes) output:multichannel-output+input:stereo-fallback: Multichannel Output + Stereo Input (sinks: 1, sources: 1, priority: 151, available: no) output:multichannel-output+input:multichannel-input: Multichannel Duplex (sinks: 1, sources: 1, priority: 101, available: yes) off: Off (sinks: 0, sources: 0, priority: 0, available: yes) Active Profile: output:stereo-fallback+input:multichannel-input Ports: analog-input-mic: Microphone (type: Mic, priority: 8700, latency offset: 0 usec, not available) Part of profile(s): input:stereo-fallback, output:stereo-fallback+input:stereo-fallback, output:multichannel-output+input:stereo-fallback analog-output-speaker: Speakers (type: Speaker, priority: 10000, latency offset: 0 usec, availability unknown) Properties: device.icon_name = "audio-speakers" Part of profile(s): output:stereo-fallback, output:stereo-fallback+input:stereo-fallback, output:stereo-fallback+input:multichannel-input analog-output-headphones: Headphones (type: Headphones, priority: 9900, latency offset: 0 usec, not available) Properties: device.icon_name = "audio-headphones" Part of profile(s): output:stereo-fallback, output:stereo-fallback+input:stereo-fallback, output:stereo-fallback+input:multichannel-input ``` The `Speakers` are selected for output but there is no port I can select for input. I did a `git bisect` and found out that reverting the following commit from v1.2.7 solves the problem: ``` commit e2cfda5b53f4718f1128fb4df75ec8fb7274d36b Author: Jaroslav Kysela <perex(a)perex.cz> Date: Mon May 23 13:33:24 2022 +0200 HDMI - use new Macro syntax Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/325 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH v4 1/3] ASoC: Add support for Loongson I2S controller
by YingKun Meng 19 Jun '23

19 Jun '23

From: Yingkun Meng <mengyingkun(a)loongson.cn> Loongson I2S controller is found on 7axxx/2kxxx chips from loongson, it is a PCI device with two private DMA controllers, one for playback, the other for capture. The driver supports the use of DTS or ACPI to describe device resources. Signed-off-by: Yingkun Meng <mengyingkun(a)loongson.cn> --- changes v3 -> v4 Add __packed attribute for loongson_dma_desc struct. sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/loongson/Kconfig | 16 ++ sound/soc/loongson/Makefile | 4 + sound/soc/loongson/loongson_dma.c | 350 ++++++++++++++++++++++++++ sound/soc/loongson/loongson_dma.h | 16 ++ sound/soc/loongson/loongson_i2s.c | 269 ++++++++++++++++++++ sound/soc/loongson/loongson_i2s.h | 71 ++++++ sound/soc/loongson/loongson_i2s_pci.c | 171 +++++++++++++ 9 files changed, 899 insertions(+) create mode 100644 sound/soc/loongson/Kconfig create mode 100644 sound/soc/loongson/Makefile create mode 100644 sound/soc/loongson/loongson_dma.c create mode 100644 sound/soc/loongson/loongson_dma.h create mode 100644 sound/soc/loongson/loongson_i2s.c create mode 100644 sound/soc/loongson/loongson_i2s.h create mode 100644 sound/soc/loongson/loongson_i2s_pci.c diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 4b6e5a802880..bfa9622e1ab1 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -79,6 +79,7 @@ source "sound/soc/google/Kconfig" source "sound/soc/hisilicon/Kconfig" source "sound/soc/jz4740/Kconfig" source "sound/soc/kirkwood/Kconfig" +source "sound/soc/loongson/Kconfig" source "sound/soc/img/Kconfig" source "sound/soc/intel/Kconfig" source "sound/soc/mediatek/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 9d9b228e4508..8376fdb217ed 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -46,6 +46,7 @@ obj-$(CONFIG_SND_SOC) += fsl/ obj-$(CONFIG_SND_SOC) += google/ obj-$(CONFIG_SND_SOC) += hisilicon/ obj-$(CONFIG_SND_SOC) += jz4740/ +obj-$(CONFIG_SND_SOC) += loongson/ obj-$(CONFIG_SND_SOC) += img/ obj-$(CONFIG_SND_SOC) += intel/ obj-$(CONFIG_SND_SOC) += mediatek/ diff --git a/sound/soc/loongson/Kconfig b/sound/soc/loongson/Kconfig new file mode 100644 index 000000000000..4478ac91e402 --- /dev/null +++ b/sound/soc/loongson/Kconfig @@ -0,0 +1,16 @@ +# SPDX-License-Identifier: GPL-2.0 +menu "SoC Audio for Loongson CPUs" + depends on LOONGARCH || COMPILE_TEST + +config SND_SOC_LOONGSON_I2S_PCI + tristate "Loongson I2S-PCI Device Driver" + select REGMAP_MMIO + depends on PCI + help + Say Y or M if you want to add support for I2S driver for + Loongson I2S controller. + + The controller is found in loongson bridge chips or SoCs, + and work as a PCI device. + +endmenu diff --git a/sound/soc/loongson/Makefile b/sound/soc/loongson/Makefile new file mode 100644 index 000000000000..099af7989103 --- /dev/null +++ b/sound/soc/loongson/Makefile @@ -0,0 +1,4 @@ +# SPDX-License-Identifier: GPL-2.0 +#Platform Support +snd-soc-loongson-i2s-pci-objs := loongson_i2s_pci.o loongson_i2s.o loongson_dma.o +obj-$(CONFIG_SND_SOC_LOONGSON_I2S_PCI) += snd-soc-loongson-i2s-pci.o diff --git a/sound/soc/loongson/loongson_dma.c b/sound/soc/loongson/loongson_dma.c new file mode 100644 index 000000000000..65b6719e61c5 --- /dev/null +++ b/sound/soc/loongson/loongson_dma.c @@ -0,0 +1,350 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Loongson ALSA SoC Platform (DMA) driver +// +// Copyright (C) 2023 Loongson Technology Corporation Limited +// Author: Yingkun Meng <mengyingkun(a)loongson.cn> +// + +#include <linux/module.h> +#include <linux/io-64-nonatomic-lo-hi.h> +#include <linux/delay.h> +#include <linux/pm_runtime.h> +#include <linux/dma-mapping.h> +#include <sound/soc.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include "loongson_i2s.h" + +/* DMA dma_order Register */ +#define DMA_ORDER_STOP (1 << 4) /* DMA stop */ +#define DMA_ORDER_START (1 << 3) /* DMA start */ +#define DMA_ORDER_ASK_VALID (1 << 2) /* DMA ask valid flag */ +#define DMA_ORDER_AXI_UNCO (1 << 1) /* Uncache access */ +#define DMA_ORDER_ADDR_64 (1 << 0) /* 64bits address support */ + +#define DMA_ORDER_ASK_MASK (~0x1fUL) /* Ask addr mask */ +#define DMA_ORDER_CTRL_MASK (0x0fUL) /* Control mask */ + +/* + * DMA registers descriptor. + */ +struct loongson_dma_desc { + u32 order; /* Next descriptor address register */ + u32 saddr; /* Source address register */ + u32 daddr; /* Device address register */ + u32 length; /* Total length register */ + u32 step_length; /* Memory stride register */ + u32 step_times; /* Repeat time register */ + u32 cmd; /* Command register */ + u32 stats; /* Status register */ + u32 order_hi; /* Next descriptor high address register */ + u32 saddr_hi; /* High source address register */ + u32 res[6]; /* Reserved */ +} __packed; + +struct loongson_runtime_data { + struct loongson_dma_data *dma_data; + + struct loongson_dma_desc *dma_desc_arr; + dma_addr_t dma_desc_arr_phy; + int dma_desc_arr_size; + + struct loongson_dma_desc *dma_pos_desc; + dma_addr_t dma_pos_desc_phy; +}; + +static const struct snd_pcm_hardware ls_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_PAUSE, + .formats = (SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | + SNDRV_PCM_FMTBIT_S24_LE), + .period_bytes_min = 128, + .period_bytes_max = 128 * 1024, + .periods_min = 1, + .periods_max = PAGE_SIZE / sizeof(struct loongson_dma_desc), + .buffer_bytes_max = 1024 * 1024, +}; + +static struct +loongson_dma_desc *dma_desc_save(struct loongson_runtime_data *prtd) +{ + void __iomem *order_reg = prtd->dma_data->order_addr; + u64 val; + + val = (u64)prtd->dma_pos_desc_phy & DMA_ORDER_ASK_MASK; + val |= (readq(order_reg) & DMA_ORDER_CTRL_MASK); + val |= DMA_ORDER_ASK_VALID; + writeq(val, order_reg); + + while (readl(order_reg) & DMA_ORDER_ASK_VALID) + udelay(2); + + return prtd->dma_pos_desc; +} + +static int loongson_pcm_trigger(struct snd_soc_component *component, + struct snd_pcm_substream *substream, int cmd) +{ + struct loongson_runtime_data *prtd = substream->runtime->private_data; + struct device *dev = substream->pcm->card->dev; + void __iomem *order_reg = prtd->dma_data->order_addr; + u64 val; + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + val = prtd->dma_pos_desc_phy & DMA_ORDER_ASK_MASK; + if (dev->coherent_dma_mask == DMA_BIT_MASK(64)) + val |= DMA_ORDER_ADDR_64; + else + val &= ~DMA_ORDER_ADDR_64; + val |= (readq(order_reg) & DMA_ORDER_CTRL_MASK); + val |= DMA_ORDER_START; + writeq(val, order_reg); + + while ((readl(order_reg) & DMA_ORDER_START)) + udelay(2); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + dma_desc_save(prtd); + + /* dma stop */ + val = readq(order_reg) | DMA_ORDER_STOP; + writeq(val, order_reg); + udelay(1000); + + break; + default: + dev_err(dev, "Invalid pcm trigger operation\n"); + return -EINVAL; + } + + return ret; +} + +static int loongson_pcm_hw_params(struct snd_soc_component *component, + struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct device *dev = substream->pcm->card->dev; + struct loongson_runtime_data *prtd = runtime->private_data; + size_t buf_len = params_buffer_bytes(params); + size_t period_len = params_period_bytes(params); + dma_addr_t order_addr, mem_addr; + struct loongson_dma_desc *desc; + u32 num_periods; + int i; + + if (buf_len % period_len) { + dev_err(dev, "buf len not multiply of period len\n"); + return -EINVAL; + } + + num_periods = buf_len / period_len; + if (!num_periods || num_periods > prtd->dma_desc_arr_size) { + dev_err(dev, "dma data too small or too big\n"); + return -EINVAL; + } + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = buf_len; + + /* initialize dma descriptor array */ + mem_addr = runtime->dma_addr; + order_addr = prtd->dma_desc_arr_phy; + for (i = 0; i < num_periods; i++) { + desc = &prtd->dma_desc_arr[i]; + + /* next descriptor physical address */ + order_addr += sizeof(*desc); + desc->order = lower_32_bits(order_addr | BIT(0)); + desc->order_hi = upper_32_bits(order_addr); + + desc->saddr = lower_32_bits(mem_addr); + desc->saddr_hi = upper_32_bits(mem_addr); + desc->daddr = prtd->dma_data->dev_addr; + + desc->cmd = BIT(0); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + desc->cmd |= BIT(12); + + desc->length = period_len >> 2; + desc->step_length = 0; + desc->step_times = 1; + + mem_addr += period_len; + } + desc = &prtd->dma_desc_arr[num_periods - 1]; + desc->order = lower_32_bits(prtd->dma_desc_arr_phy | BIT(0)); + desc->order_hi = upper_32_bits(prtd->dma_desc_arr_phy); + + /* init position descriptor */ + *prtd->dma_pos_desc = *prtd->dma_desc_arr; + + return 0; +} + +static snd_pcm_uframes_t +loongson_pcm_pointer(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct loongson_runtime_data *prtd = runtime->private_data; + struct loongson_dma_desc *desc; + snd_pcm_uframes_t x; + u64 addr; + + desc = dma_desc_save(prtd); + addr = ((u64)desc->saddr_hi << 32) | desc->saddr; + + x = bytes_to_frames(runtime, addr - runtime->dma_addr); + if (x == runtime->buffer_size) + x = 0; + return x; +} + +static irqreturn_t loongson_pcm_dma_irq(int irq, void *devid) +{ + struct snd_pcm_substream *substream = devid; + + snd_pcm_period_elapsed(substream); + return IRQ_HANDLED; +} + +static int loongson_pcm_open(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_card *card = substream->pcm->card; + struct loongson_runtime_data *prtd; + struct loongson_dma_data *dma_data; + int ret; + + /* + * For mysterious reasons (and despite what the manual says) + * playback samples are lost if the DMA count is not a multiple + * of the DMA burst size. Let's add a rule to enforce that. + */ + snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 128); + snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 128); + snd_pcm_hw_constraint_integer(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + snd_soc_set_runtime_hwparams(substream, &ls_pcm_hardware); + + prtd = kzalloc(sizeof(*prtd), GFP_KERNEL); + if (!prtd) + return -ENOMEM; + + prtd->dma_desc_arr = dma_alloc_coherent(card->dev, PAGE_SIZE, + &prtd->dma_desc_arr_phy, + GFP_KERNEL); + if (!prtd->dma_desc_arr) { + ret = -ENOMEM; + goto desc_err; + } + prtd->dma_desc_arr_size = PAGE_SIZE / sizeof(*prtd->dma_desc_arr); + + prtd->dma_pos_desc = dma_alloc_coherent(card->dev, + sizeof(*prtd->dma_pos_desc), + &prtd->dma_pos_desc_phy, + GFP_KERNEL); + if (!prtd->dma_pos_desc) { + ret = -ENOMEM; + goto pos_err; + } + + dma_data = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); + prtd->dma_data = dma_data; + + substream->runtime->private_data = prtd; + + return 0; +pos_err: + dma_free_coherent(card->dev, PAGE_SIZE, prtd->dma_desc_arr, + prtd->dma_desc_arr_phy); +desc_err: + kfree(prtd); + + return ret; +} + +static int loongson_pcm_close(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct snd_card *card = substream->pcm->card; + struct loongson_runtime_data *prtd = substream->runtime->private_data; + + dma_free_coherent(card->dev, PAGE_SIZE, prtd->dma_desc_arr, + prtd->dma_desc_arr_phy); + + dma_free_coherent(card->dev, sizeof(*prtd->dma_pos_desc), + prtd->dma_pos_desc, prtd->dma_pos_desc_phy); + + kfree(prtd); + return 0; +} + +static int loongson_pcm_mmap(struct snd_soc_component *component, + struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + return remap_pfn_range(vma, vma->vm_start, + substream->dma_buffer.addr >> PAGE_SHIFT, + vma->vm_end - vma->vm_start, vma->vm_page_prot); +} + +static int loongson_pcm_new(struct snd_soc_component *component, + struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + struct snd_pcm_substream *substream; + struct loongson_dma_data *dma_data; + unsigned int i; + int ret; + + for_each_pcm_streams(i) { + substream = rtd->pcm->streams[i].substream; + if (!substream) + continue; + + dma_data = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), + substream); + ret = devm_request_irq(card->dev, dma_data->irq, + loongson_pcm_dma_irq, + IRQF_TRIGGER_HIGH, LS_I2S_DRVNAME, + substream); + if (ret < 0) { + dev_err(card->dev, "request irq for DMA failed\n"); + return ret; + } + } + + return snd_pcm_set_fixed_buffer_all(rtd->pcm, SNDRV_DMA_TYPE_DEV, + card->dev, + ls_pcm_hardware.buffer_bytes_max); +} + +const struct snd_soc_component_driver loongson_i2s_component = { + .name = LS_I2S_DRVNAME, + .open = loongson_pcm_open, + .close = loongson_pcm_close, + .hw_params = loongson_pcm_hw_params, + .trigger = loongson_pcm_trigger, + .pointer = loongson_pcm_pointer, + .mmap = loongson_pcm_mmap, + .pcm_construct = loongson_pcm_new, +}; diff --git a/sound/soc/loongson/loongson_dma.h b/sound/soc/loongson/loongson_dma.h new file mode 100644 index 000000000000..073ee8c0c046 --- /dev/null +++ b/sound/soc/loongson/loongson_dma.h @@ -0,0 +1,16 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * ALSA ASoC interface for the Loongson platform + * + * Copyright (C) 2023 Loongson Technology Corporation Limited + * Author: Yingkun Meng <mengyingkun(a)loongson.cn> + */ + +#ifndef _LOONGSON_DMA_H +#define _LOONGSON_DMA_H + +#include <sound/soc.h> + +extern const struct snd_soc_component_driver loongson_i2s_component; + +#endif diff --git a/sound/soc/loongson/loongson_i2s.c b/sound/soc/loongson/loongson_i2s.c new file mode 100644 index 000000000000..35d34568be79 --- /dev/null +++ b/sound/soc/loongson/loongson_i2s.c @@ -0,0 +1,269 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Common functions for loongson I2S controller driver +// +// Copyright (C) 2023 Loongson Technology Corporation Limited. +// Author: Yingkun Meng <mengyingkun(a)loongson.cn> +// + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/delay.h> +#include <linux/pm_runtime.h> +#include <linux/dma-mapping.h> +#include <sound/soc.h> +#include <linux/regmap.h> +#include <sound/pcm_params.h> +#include "loongson_i2s.h" + +#define LOONGSON_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +static int loongson_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct loongson_i2s *i2s = snd_soc_dai_get_drvdata(dai); + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + regmap_update_bits(i2s->regmap, LS_I2S_CTRL, + I2S_CTRL_TX_EN | I2S_CTRL_TX_DMA_EN, + I2S_CTRL_TX_EN | I2S_CTRL_TX_DMA_EN); + else + regmap_update_bits(i2s->regmap, LS_I2S_CTRL, + I2S_CTRL_RX_EN | I2S_CTRL_RX_DMA_EN, + I2S_CTRL_RX_EN | I2S_CTRL_RX_DMA_EN); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + regmap_update_bits(i2s->regmap, LS_I2S_CTRL, + I2S_CTRL_TX_EN | I2S_CTRL_TX_DMA_EN, 0); + else + regmap_update_bits(i2s->regmap, LS_I2S_CTRL, + I2S_CTRL_RX_EN | I2S_CTRL_RX_DMA_EN, 0); + break; + default: + ret = -EINVAL; + } + + return ret; +} + +static int loongson_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct loongson_i2s *i2s = snd_soc_dai_get_drvdata(dai); + u32 clk_rate = i2s->clk_rate; + u32 sysclk = i2s->sysclk; + u32 bits = params_width(params); + u32 chans = params_channels(params); + u32 fs = params_rate(params); + u32 bclk_ratio, mclk_ratio; + u32 mclk_ratio_frac; + u32 val = 0; + + switch (i2s->rev_id) { + case 0: + bclk_ratio = DIV_ROUND_CLOSEST(clk_rate, + (bits * chans * fs * 2)) - 1; + mclk_ratio = DIV_ROUND_CLOSEST(clk_rate, (sysclk * 2)) - 1; + + /* According to 2k1000LA user manual, set bits == depth */ + val |= (bits << 24); + val |= (bits << 16); + val |= (bclk_ratio << 8); + val |= mclk_ratio; + regmap_write(i2s->regmap, LS_I2S_CFG, val); + + break; + case 1: + bclk_ratio = DIV_ROUND_CLOSEST(sysclk, + (bits * chans * fs * 2)) - 1; + mclk_ratio = clk_rate / sysclk; + mclk_ratio_frac = DIV_ROUND_CLOSEST(((u64)clk_rate << 16), + sysclk) - (mclk_ratio << 16); + + regmap_read(i2s->regmap, LS_I2S_CFG, &val); + val |= (bits << 24); + val |= (bclk_ratio << 8); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + val |= (bits << 16); + else + val |= bits; + regmap_write(i2s->regmap, LS_I2S_CFG, val); + + val = (mclk_ratio_frac << 16) | mclk_ratio; + regmap_write(i2s->regmap, LS_I2S_CFG1, val); + + break; + default: + dev_err(i2s->dev, "I2S revision invalid\n"); + return -EINVAL; + } + + return 0; +} + +static int loongson_i2s_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct loongson_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + i2s->sysclk = freq; + + return 0; +} + +static int loongson_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct loongson_i2s *i2s = snd_soc_dai_get_drvdata(dai); + u32 val; + int ret; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_RIGHT_J: + regmap_update_bits(i2s->regmap, LS_I2S_CTRL, I2S_CTRL_MSB, + I2S_CTRL_MSB); + break; + default: + return -EINVAL; + } + + + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BC_FC: + break; + case SND_SOC_DAIFMT_BP_FC: + /* Enable master mode */ + regmap_update_bits(i2s->regmap, LS_I2S_CTRL, I2S_CTRL_MASTER, + I2S_CTRL_MASTER); + if (i2s->rev_id == 1) { + ret = regmap_read_poll_timeout_atomic(i2s->regmap, + LS_I2S_CTRL, val, + val & I2S_CTRL_CLK_READY, + 10, 500000); + if (ret < 0) + dev_warn(dai->dev, "wait BCLK ready timeout\n"); + } + break; + case SND_SOC_DAIFMT_BC_FP: + /* Enable MCLK */ + if (i2s->rev_id == 1) { + regmap_update_bits(i2s->regmap, LS_I2S_CTRL, + I2S_CTRL_MCLK_EN, + I2S_CTRL_MCLK_EN); + ret = regmap_read_poll_timeout_atomic(i2s->regmap, + LS_I2S_CTRL, val, + val & I2S_CTRL_MCLK_READY, + 10, 500000); + if (ret < 0) + dev_warn(dai->dev, "wait MCLK ready timeout\n"); + } + break; + case SND_SOC_DAIFMT_BP_FP: + /* Enable MCLK */ + if (i2s->rev_id == 1) { + regmap_update_bits(i2s->regmap, LS_I2S_CTRL, + I2S_CTRL_MCLK_EN, + I2S_CTRL_MCLK_EN); + ret = regmap_read_poll_timeout_atomic(i2s->regmap, + LS_I2S_CTRL, val, + val & I2S_CTRL_MCLK_READY, + 10, 500000); + if (ret < 0) + dev_warn(dai->dev, "wait MCLK ready timeout\n"); + } + + /* Enable master mode */ + regmap_update_bits(i2s->regmap, LS_I2S_CTRL, I2S_CTRL_MASTER, + I2S_CTRL_MASTER); + if (i2s->rev_id == 1) { + ret = regmap_read_poll_timeout_atomic(i2s->regmap, + LS_I2S_CTRL, val, + val & I2S_CTRL_CLK_READY, + 10, 500000); + if (ret < 0) + dev_warn(dai->dev, "wait BCLK ready timeout\n"); + } + break; + default: + return -EINVAL; + } + + return 0; +} + +static const struct snd_soc_dai_ops loongson_i2s_dai_ops = { + .trigger = loongson_i2s_trigger, + .hw_params = loongson_i2s_hw_params, + .set_sysclk = loongson_i2s_set_dai_sysclk, + .set_fmt = loongson_i2s_set_fmt, +}; + +static int loongson_i2s_dai_probe(struct snd_soc_dai *cpu_dai) +{ + struct loongson_i2s *i2s = dev_get_drvdata(cpu_dai->dev); + + snd_soc_dai_init_dma_data(cpu_dai, &i2s->playback_dma_data, + &i2s->capture_dma_data); + snd_soc_dai_set_drvdata(cpu_dai, i2s); + + return 0; +} + +struct snd_soc_dai_driver loongson_i2s_dai = { + .name = "loongson-i2s", + .probe = loongson_i2s_dai_probe, + .playback = { + .stream_name = "CPU-Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = LOONGSON_I2S_FORMATS, + }, + .capture = { + .stream_name = "CPU-Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = LOONGSON_I2S_FORMATS, + }, + .ops = &loongson_i2s_dai_ops, + .symmetric_rate = 1, +}; + +static int i2s_suspend(struct device *dev) +{ + struct loongson_i2s *i2s = dev_get_drvdata(dev); + + regcache_cache_only(i2s->regmap, true); + + return 0; +} + +static int i2s_resume(struct device *dev) +{ + struct loongson_i2s *i2s = dev_get_drvdata(dev); + int ret; + + regcache_cache_only(i2s->regmap, false); + regcache_mark_dirty(i2s->regmap); + ret = regcache_sync(i2s->regmap); + + return ret; +} + +const struct dev_pm_ops loongson_i2s_pm = { + SET_SYSTEM_SLEEP_PM_OPS(i2s_suspend, i2s_resume) +}; diff --git a/sound/soc/loongson/loongson_i2s.h b/sound/soc/loongson/loongson_i2s.h new file mode 100644 index 000000000000..52788f6a94ad --- /dev/null +++ b/sound/soc/loongson/loongson_i2s.h @@ -0,0 +1,71 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * ALSA I2S interface for the Loongson platform + * + * Copyright (C) 2023 Loongson Technology Corporation Limited + * Author: Yingkun Meng <mengyingkun(a)loongson.cn> + */ + +#ifndef _LOONGSON_I2S_H +#define _LOONGSON_I2S_H + +#include <linux/regmap.h> +#include <sound/dmaengine_pcm.h> + +/* I2S Common Registers */ +#define LS_I2S_VER 0x00 /* I2S Version */ +#define LS_I2S_CFG 0x04 /* I2S Config */ +#define LS_I2S_CTRL 0x08 /* I2S Control */ +#define LS_I2S_RX_DATA 0x0C /* I2S DMA RX Address */ +#define LS_I2S_TX_DATA 0x10 /* I2S DMA TX Address */ + +/* 2K2000 I2S Specify Registers */ +#define LS_I2S_CFG1 0x14 /* I2S Config1 */ + +/* 7A2000 I2S Specify Registers */ +#define LS_I2S_TX_ORDER 0x100 /* TX DMA Order */ +#define LS_I2S_RX_ORDER 0x110 /* RX DMA Order */ + +/* Loongson I2S Control Register */ +#define I2S_CTRL_MCLK_READY (1 << 16) /* MCLK ready */ +#define I2S_CTRL_MASTER (1 << 15) /* Master mode */ +#define I2S_CTRL_MSB (1 << 14) /* MSB bit order */ +#define I2S_CTRL_RX_EN (1 << 13) /* RX enable */ +#define I2S_CTRL_TX_EN (1 << 12) /* TX enable */ +#define I2S_CTRL_RX_DMA_EN (1 << 11) /* DMA RX enable */ +#define I2S_CTRL_CLK_READY (1 << 8) /* BCLK ready */ +#define I2S_CTRL_TX_DMA_EN (1 << 7) /* DMA TX enable */ +#define I2S_CTRL_RESET (1 << 4) /* Controller soft reset */ +#define I2S_CTRL_MCLK_EN (1 << 3) /* Enable MCLK */ +#define I2S_CTRL_RX_INT_EN (1 << 1) /* RX interrupt enable */ +#define I2S_CTRL_TX_INT_EN (1 << 0) /* TX interrupt enable */ + +#define LS_I2S_DRVNAME "loongson-i2s" + +struct loongson_dma_data { + dma_addr_t dev_addr; /* device physical address for DMA */ + void __iomem *order_addr; /* DMA order register */ + u32 irq; /* DMA irq */ +}; + +struct loongson_i2s { + struct device *dev; + union { + struct snd_dmaengine_dai_dma_data playback_dma_data; + struct loongson_dma_data tx_dma_data; + }; + union { + struct snd_dmaengine_dai_dma_data capture_dma_data; + struct loongson_dma_data rx_dma_data; + }; + struct regmap *regmap; + void __iomem *reg_base; + u32 rev_id; + u32 clk_rate; + u32 sysclk; +}; + +extern const struct dev_pm_ops loongson_i2s_pm; +extern struct snd_soc_dai_driver loongson_i2s_dai; + +#endif diff --git a/sound/soc/loongson/loongson_i2s_pci.c b/sound/soc/loongson/loongson_i2s_pci.c new file mode 100644 index 000000000000..6dcfb17d3276 --- /dev/null +++ b/sound/soc/loongson/loongson_i2s_pci.c @@ -0,0 +1,171 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// loongson_i2s_pci.c -- Loongson I2S controller driver +// +// Copyright (C) 2023 Loongson Technology Corporation Limited +// Author: Yingkun Meng <mengyingkun(a)loongson.cn> +// + +#include <linux/module.h> +#include <linux/delay.h> +#include <linux/pm_runtime.h> +#include <linux/dma-mapping.h> +#include <linux/acpi.h> +#include <linux/pci.h> +#include <sound/soc.h> +#include "loongson_i2s.h" +#include "loongson_dma.h" + +static bool loongson_i2s_wr_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case LS_I2S_CFG: + case LS_I2S_CTRL: + case LS_I2S_RX_DATA: + case LS_I2S_TX_DATA: + case LS_I2S_CFG1: + return true; + default: + return false; + }; +} + +static bool loongson_i2s_rd_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case LS_I2S_VER: + case LS_I2S_CFG: + case LS_I2S_CTRL: + case LS_I2S_RX_DATA: + case LS_I2S_TX_DATA: + case LS_I2S_CFG1: + return true; + default: + return false; + }; +} + +static bool loongson_i2s_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case LS_I2S_CFG: + case LS_I2S_CTRL: + case LS_I2S_RX_DATA: + case LS_I2S_TX_DATA: + case LS_I2S_CFG1: + return true; + default: + return false; + }; +} + +static const struct regmap_config loongson_i2s_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = LS_I2S_CFG1, + .writeable_reg = loongson_i2s_wr_reg, + .readable_reg = loongson_i2s_rd_reg, + .volatile_reg = loongson_i2s_volatile_reg, + .cache_type = REGCACHE_FLAT, +}; + +static int loongson_i2s_pci_probe(struct pci_dev *pdev, + const struct pci_device_id *pid) +{ + const struct fwnode_handle *fwnode = pdev->dev.fwnode; + struct loongson_dma_data *tx_data, *rx_data; + struct loongson_i2s *i2s; + int ret; + + if (pcim_enable_device(pdev)) { + dev_err(&pdev->dev, "pci_enable_device failed\n"); + return -ENODEV; + } + + i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL); + if (!i2s) + return -ENOMEM; + + i2s->rev_id = pdev->revision; + i2s->dev = &pdev->dev; + pci_set_drvdata(pdev, i2s); + + ret = pcim_iomap_regions(pdev, 1 << 0, dev_name(&pdev->dev)); + if (ret < 0) { + dev_err(&pdev->dev, "iomap_regions failed\n"); + return ret; + } + i2s->reg_base = pcim_iomap_table(pdev)[0]; + i2s->regmap = devm_regmap_init_mmio(&pdev->dev, i2s->reg_base, + &loongson_i2s_regmap_config); + if (IS_ERR(i2s->regmap)) { + dev_err(&pdev->dev, "regmap_init_mmio failed\n"); + return PTR_ERR(i2s->regmap); + } + + tx_data = &i2s->tx_dma_data; + rx_data = &i2s->rx_dma_data; + + tx_data->dev_addr = (dma_addr_t)i2s->reg_base + LS_I2S_TX_DATA; + tx_data->order_addr = i2s->reg_base + LS_I2S_TX_ORDER; + + rx_data->dev_addr = (dma_addr_t)i2s->reg_base + LS_I2S_RX_DATA; + rx_data->order_addr = i2s->reg_base + LS_I2S_RX_ORDER; + + tx_data->irq = fwnode_irq_get_byname(fwnode, "tx"); + if (tx_data->irq < 0) { + dev_err(&pdev->dev, "dma tx irq invalid\n"); + return tx_data->irq; + } + + rx_data->irq = fwnode_irq_get_byname(fwnode, "rx"); + if (rx_data->irq < 0) { + dev_err(&pdev->dev, "dma rx irq invalid\n"); + return rx_data->irq; + } + + device_property_read_u32(&pdev->dev, "clock-frequency", &i2s->clk_rate); + if (!i2s->clk_rate) { + dev_err(&pdev->dev, "clock-frequency property invalid\n"); + return -EINVAL; + } + + dma_set_mask_and_coherent(&pdev->dev, DMA_BIT_MASK(64)); + + if (i2s->rev_id == 1) { + regmap_write(i2s->regmap, LS_I2S_CTRL, I2S_CTRL_RESET); + udelay(200); + } + + ret = devm_snd_soc_register_component(&pdev->dev, + &loongson_i2s_component, + &loongson_i2s_dai, 1); + if (ret) { + dev_err(&pdev->dev, "register DAI failed %d\n", ret); + return ret; + } + + return 0; +} + +static const struct pci_device_id loongson_i2s_ids[] = { + { PCI_DEVICE(PCI_VENDOR_ID_LOONGSON, 0x7a27) }, + { }, +}; +MODULE_DEVICE_TABLE(pci, loongson_i2s_ids); + +static struct pci_driver loongson_i2s_driver = { + .name = "loongson-i2s-pci", + .id_table = loongson_i2s_ids, + .probe = loongson_i2s_pci_probe, + .driver = { + .owner = THIS_MODULE, + .pm = pm_sleep_ptr(&loongson_i2s_pm), + }, +}; +module_pci_driver(loongson_i2s_driver); + +MODULE_DESCRIPTION("Loongson I2S Master Mode ASoC Driver"); +MODULE_AUTHOR("Loongson Technology Corporation Limited"); +MODULE_LICENSE("GPL"); -- 2.33.0

4 9

[PATCH] sound: Fix traditional comparison using max/min method
by Li Dong 19 Jun '23

19 Jun '23

It would be better to replace the traditional ternary conditional operator with max()/min() Signed-off-by: Li Dong <lidong(a)vivo.com> --- sound/soc/starfive/jh7110_tdm.c | 11 ++--------- 1 file changed, 2 insertions(+), 9 deletions(-) diff --git a/sound/soc/starfive/jh7110_tdm.c b/sound/soc/starfive/jh7110_tdm.c index 5f5a6ca7dbda..a17437dd8c3e 100644 --- a/sound/soc/starfive/jh7110_tdm.c +++ b/sound/soc/starfive/jh7110_tdm.c @@ -187,15 +187,8 @@ static int jh7110_tdm_syncdiv(struct jh7110_tdm_dev *tdm) { u32 sl, sscale, syncdiv; - if (tdm->rx.sl >= tdm->tx.sl) - sl = tdm->rx.sl; - else - sl = tdm->tx.sl; - - if (tdm->rx.sscale >= tdm->tx.sscale) - sscale = tdm->rx.sscale; - else - sscale = tdm->tx.sscale; + sl = max(tdm->rx.sl, tdm->tx.sl); + sscale = max(tdm->rx.sscale, tdm->tx.sscale); syncdiv = tdm->pcmclk / tdm->samplerate - 1; -- 2.31.1.windows.1

1 0

[PATCH v3 0/6] Add cs42l43 PC focused SoundWire CODEC
by Charles Keepax 19 Jun '23

19 Jun '23

This patch chain adds support for the Cirrus Logic cs42l43 PC focused SoundWire CODEC. The chain is currently based of Lee's for-mfd-next branch. Thanks, Charles Charles Keepax (4): dt-bindings: mfd: cirrus,cs42l43: Add initial DT binding mfd: cs42l43: Add support for cs42l43 core driver pinctrl: cs42l43: Add support for the cs42l43 ASoC: cs42l43: Add support for the cs42l43 Lucas Tanure (2): soundwire: bus: Allow SoundWire peripherals to register IRQ handlers spi: cs42l43: Add SPI controller support .../bindings/sound/cirrus,cs42l43.yaml | 313 +++ MAINTAINERS | 4 + drivers/mfd/Kconfig | 23 + drivers/mfd/Makefile | 3 + drivers/mfd/cs42l43-i2c.c | 86 + drivers/mfd/cs42l43-sdw.c | 213 ++ drivers/mfd/cs42l43.c | 1141 +++++++++ drivers/mfd/cs42l43.h | 23 + drivers/pinctrl/cirrus/Kconfig | 11 + drivers/pinctrl/cirrus/Makefile | 2 + drivers/pinctrl/cirrus/pinctrl-cs42l43.c | 609 +++++ drivers/soundwire/bus.c | 31 + drivers/soundwire/bus_type.c | 12 + drivers/spi/Kconfig | 7 + drivers/spi/Makefile | 1 + drivers/spi/spi-cs42l43.c | 281 ++ include/linux/mfd/cs42l43-regs.h | 1172 +++++++++ include/linux/mfd/cs42l43.h | 102 + include/linux/soundwire/sdw.h | 9 + include/sound/cs42l43.h | 17 + sound/soc/codecs/Kconfig | 16 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/cs42l43-jack.c | 967 +++++++ sound/soc/codecs/cs42l43-sdw.c | 74 + sound/soc/codecs/cs42l43.c | 2278 +++++++++++++++++ sound/soc/codecs/cs42l43.h | 131 + 26 files changed, 7530 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cirrus,cs42l43.yaml create mode 100644 drivers/mfd/cs42l43-i2c.c create mode 100644 drivers/mfd/cs42l43-sdw.c create mode 100644 drivers/mfd/cs42l43.c create mode 100644 drivers/mfd/cs42l43.h create mode 100644 drivers/pinctrl/cirrus/pinctrl-cs42l43.c create mode 100644 drivers/spi/spi-cs42l43.c create mode 100644 include/linux/mfd/cs42l43-regs.h create mode 100644 include/linux/mfd/cs42l43.h create mode 100644 include/sound/cs42l43.h create mode 100644 sound/soc/codecs/cs42l43-jack.c create mode 100644 sound/soc/codecs/cs42l43-sdw.c create mode 100644 sound/soc/codecs/cs42l43.c create mode 100644 sound/soc/codecs/cs42l43.h -- 2.30.2

4 14

[PATCH v4 0/6] Add cs42l43 PC focused SoundWire CODEC
by Charles Keepax 19 Jun '23

19 Jun '23

This patch chain adds support for the Cirrus Logic cs42l43 PC focused SoundWire CODEC. The chain is currently based of Lee's for-mfd-next branch. Thanks, Charles Charles Keepax (4): dt-bindings: mfd: cirrus,cs42l43: Add initial DT binding mfd: cs42l43: Add support for cs42l43 core driver pinctrl: cs42l43: Add support for the cs42l43 ASoC: cs42l43: Add support for the cs42l43 Lucas Tanure (2): soundwire: bus: Allow SoundWire peripherals to register IRQ handlers spi: cs42l43: Add SPI controller support .../bindings/sound/cirrus,cs42l43.yaml | 313 +++ MAINTAINERS | 4 + drivers/mfd/Kconfig | 23 + drivers/mfd/Makefile | 3 + drivers/mfd/cs42l43-i2c.c | 87 + drivers/mfd/cs42l43-sdw.c | 222 ++ drivers/mfd/cs42l43.c | 1262 +++++++++ drivers/mfd/cs42l43.h | 23 + drivers/pinctrl/cirrus/Kconfig | 11 + drivers/pinctrl/cirrus/Makefile | 2 + drivers/pinctrl/cirrus/pinctrl-cs42l43.c | 609 +++++ drivers/soundwire/bus.c | 32 + drivers/soundwire/bus_type.c | 12 + drivers/spi/Kconfig | 7 + drivers/spi/Makefile | 1 + drivers/spi/spi-cs42l43.c | 281 ++ include/linux/mfd/cs42l43-regs.h | 1184 +++++++++ include/linux/mfd/cs42l43.h | 102 + include/linux/soundwire/sdw.h | 9 + include/sound/cs42l43.h | 17 + sound/soc/codecs/Kconfig | 16 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/cs42l43-jack.c | 969 +++++++ sound/soc/codecs/cs42l43-sdw.c | 74 + sound/soc/codecs/cs42l43.c | 2278 +++++++++++++++++ sound/soc/codecs/cs42l43.h | 131 + 26 files changed, 7676 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cirrus,cs42l43.yaml create mode 100644 drivers/mfd/cs42l43-i2c.c create mode 100644 drivers/mfd/cs42l43-sdw.c create mode 100644 drivers/mfd/cs42l43.c create mode 100644 drivers/mfd/cs42l43.h create mode 100644 drivers/pinctrl/cirrus/pinctrl-cs42l43.c create mode 100644 drivers/spi/spi-cs42l43.c create mode 100644 include/linux/mfd/cs42l43-regs.h create mode 100644 include/linux/mfd/cs42l43.h create mode 100644 include/sound/cs42l43.h create mode 100644 sound/soc/codecs/cs42l43-jack.c create mode 100644 sound/soc/codecs/cs42l43-sdw.c create mode 100644 sound/soc/codecs/cs42l43.c create mode 100644 sound/soc/codecs/cs42l43.h -- 2.30.2

1 6

ucm2: PinePhone: use "Mix Mono" routing for earpiece
by GitHub pull_request - opened 18 Jun '23

18 Jun '23

alsa-project/alsa-ucm-conf pull request #324 was opened from a-wai: The earpiece speaker is a mono device, using only a single channel (in our case, the left one) from the DA0 output. This causes loss of information as the right channel is completely discarded when playing stereo audio. In order to avoid this issue, set `AIF1 DA0 Stereo Playback Route` to `Mix Mono` when using the "Earpiece" output port (and only in this case). Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/324 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/324.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

fix three compile warnings
by GitHub pull_request - opened 18 Jun '23

18 Jun '23

alsa-project/alsa-utils pull request #221 was opened from critkitten: fix three compile warnings Request URL : https://github.com/alsa-project/alsa-utils/pull/221 Patch URL : https://github.com/alsa-project/alsa-utils/pull/221.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

topology: nhlt: intel: support m/n divider and xtal/cardinal/pll clock source
by GitHub pull_request - edited 18 Jun '23

18 Jun '23

alsa-project/alsa-utils pull request #220 was edited from brentlu: Current alsatplg supports only cardinal clock source without m/n divider. Code in SOF ssp driver is ported here to implement the divider function. I tested the code on ADL brya device with following setting: MCLK: 19.2MHz BCLK: 3.072Mz (m 24 n 25) PR of topology is here: https://github.com/thesofproject/sof/pull/7826 Request URL : https://github.com/alsa-project/alsa-utils/pull/220 Patch URL : https://github.com/alsa-project/alsa-utils/pull/220.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Mixer settings faulty on M-Audio C400 (usb) after suspend until they're touched in alsamixer
by GitHub issues - opened 17 Jun '23

17 Jun '23

alsa-project/alsa-lib issue #331 was opened from ri0t: I don't know if this is the right place to report it, but: * I do a suspend and resume on my machine * Now, my mixer settings for my M-Audio C400 usb-soundcard are set to output all signals on the first pcm out * alsamixer doesn't reflect the true settings of the mixer, instead it shows the last used (correct, or wanted) settings * Once i touch one of the mixer controls in alsamixer, the settings get set to the correct values (+ that change i did) * This repeats every suspend/resume-cycle I've observed this on Debian Bullseye and now on NixOS 23.05. Thanks in advance, even if you point me to a better place to post this. I can supply more debug info, once i know what you need, but the upload from alsa-info to www.alsa-project.org failed somehow without providing me a url. So, here's some basics: ``` !!Kernel Information !!------------------ Kernel release: #1-NixOS SMP PREEMPT_DYNAMIC Tue May 30 13:03:33 UTC 2023 Operating System: GNU/Linux Architecture: x86_64 Processor: unknown SMP Enabled: Yes !!ALSA Version !!------------ Driver version: k6.1.31 Library version: Utilities version: 1.2.9 !!Loaded ALSA modules !!------------------- snd_hda_intel (card 0) snd_usb_audio (card 1) snd_hda_intel (card 2) !!Sound Servers on this system !!---------------------------- PipeWire: Installed - Yes (/run/current-system/sw/bin/pipewire) Running - Yes Jack: Installed - Yes (/run/current-system/sw/bin/jackd) Running - No Jack2: Installed - Yes (/run/current-system/sw/bin/jackdbus) Running - Yes !!Soundcards recognised by ALSA !!----------------------------- 0 [NVidia ]: HDA-Intel - HDA NVidia HDA NVidia at 0xfc080000 irq 119 1 [C400 ]: USB-Audio - Fast Track C400 M-Audio Fast Track C400 at usb-0000:0c:00.3-1, high speed 2 [Generic ]: HDA-Intel - HD-Audio Generic HD-Audio Generic at 0xfcb00000 irq 125 !!USB Stream information !!---------------------- --startcollapse-- M-Audio Fast Track C400 at usb-0000:0c:00.3-1, high speed : USB Audio Playback: Status: Running Interface = 2 Altset = 1 Packet Size = 162 Momentary freq = 48000 Hz (0x6.0000) Interface 2 Altset 1 Format: S24_3LE Channels: 6 Endpoint: 0x01 (1 OUT) (ADAPTIVE) Rates: 44100, 48000, 88200, 96000 Data packet interval: 125 us Bits: 0 Sync Endpoint: 0x81 (1 IN) Sync EP Interface: 3 Sync EP Altset: 1 Implicit Feedback Mode: Yes Capture: Status: Stop Interface 3 Altset 1 Format: S24_3LE Channels: 4 Endpoint: 0x81 (1 IN) (ASYNC) Rates: 44100, 48000, 88200, 96000 Data packet interval: 125 us Bits: 0 --endcollapse-- !!Modprobe options (Sound related) !!-------------------------------- snd_pcsp: index=-2 snd_usb_audio: index=-2 snd_atiixp_modem: index=-2 snd_intel8x0m: index=-2 snd_via82xx_modem: index=-2 !!Loaded sound module options !!--------------------------- !!Module: snd_usb_audio autoclock : Y delayed_register : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) device_setup : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) ignore_ctl_error : N implicit_fb : N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N index : -2,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 lowlatency : Y pid : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 quirk_alias : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) quirk_flags : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 skip_validation : N use_vmalloc : Y vid : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 !!USB Mixer information !!--------------------- --startcollapse-- USB Mixer: usb_id=0x07632030, ctrlif=1, ctlerr=0 Card: M-Audio Fast Track C400 at usb-0000:0c:00.3-1, high speed Unit: 64 Control: name="Effect Return 6", index=0 Info: id=64, control=1, cmask=0x400, channels=1, type="S16" Volume: min=46854, max=65403, dBmin=-7297, dBmax=-51 Unit: 64 Control: name="Effect Return 5", index=0 Info: id=64, control=1, cmask=0x8, channels=1, type="S16" Volume: min=46854, max=65403, dBmin=-7297, dBmax=-51 Unit: 64 Control: name="Effect Return 4", index=0 Info: id=64, control=1, cmask=0x100, channels=1, type="S16" Volume: min=46854, max=65403, dBmin=-7297, dBmax=-51 Unit: 64 Control: name="Effect Return 3", index=0 Info: id=64, control=1, cmask=0x2, channels=1, type="S16" Volume: min=46854, max=65403, dBmin=-7297, dBmax=-51 Unit: 64 Control: name="Effect Return 2", index=0 Info: id=64, control=1, cmask=0x40, channels=1, type="S16" Volume: min=46854, max=65403, dBmin=-7297, dBmax=-51 Unit: 64 Control: name="Effect Return 1", index=0 Info: id=64, control=1, cmask=0x0, channels=1, type="S16" Volume: min=46854, max=65403, dBmin=-7297, dBmax=-51 Unit: 64 Control: name="In4-Out6 Playback Volume", index=0 Info: id=64, control=1, cmask=0x10, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="In4-Out5 Playback Volume", index=0 Info: id=64, control=1, cmask=0x8, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="In4-Out4 Playback Volume", index=0 Info: id=64, control=1, cmask=0x4, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="In4-Out3 Playback Volume", index=0 Info: id=64, control=1, cmask=0x2, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="In4-Out2 Playback Volume", index=0 Info: id=64, control=1, cmask=0x1, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="In4-Out1 Playback Volume", index=0 Info: id=64, control=1, cmask=0x0, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="In3-Out6 Playback Volume", index=0 Info: id=64, control=1, cmask=0x10, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="In3-Out5 Playback Volume", index=0 Info: id=64, control=1, cmask=0x8, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="In3-Out4 Playback Volume", index=0 Info: id=64, control=1, cmask=0x4, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="In3-Out3 Playback Volume", index=0 Info: id=64, control=1, cmask=0x2, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="In3-Out2 Playback Volume", index=0 Info: id=64, control=1, cmask=0x1, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="In3-Out1 Playback Volume", index=0 Info: id=64, control=1, cmask=0x0, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="In2-Out6 Playback Volume", index=0 Info: id=64, control=1, cmask=0x10, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="In2-Out5 Playback Volume", index=0 Info: id=64, control=1, cmask=0x8, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="In2-Out4 Playback Volume", index=0 Info: id=64, control=1, cmask=0x4, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="In2-Out3 Playback Volume", index=0 Info: id=64, control=1, cmask=0x2, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="In2-Out2 Playback Volume", index=0 Info: id=64, control=1, cmask=0x1, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="In2-Out1 Playback Volume", index=0 Info: id=64, control=1, cmask=0x0, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="In1-Out6 Playback Volume", index=0 Info: id=64, control=1, cmask=0x10, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="In1-Out5 Playback Volume", index=0 Info: id=64, control=1, cmask=0x8, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="In1-Out4 Playback Volume", index=0 Info: id=64, control=1, cmask=0x4, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="In1-Out3 Playback Volume", index=0 Info: id=64, control=1, cmask=0x2, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="In1-Out2 Playback Volume", index=0 Info: id=64, control=1, cmask=0x1, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="In1-Out1 Playback Volume", index=0 Info: id=64, control=1, cmask=0x0, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="PCM6-Out6 Playback Volume", index=0 Info: id=64, control=1, cmask=0x10, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="PCM6-Out5 Playback Volume", index=0 Info: id=64, control=1, cmask=0x8, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="PCM6-Out4 Playback Volume", index=0 Info: id=64, control=1, cmask=0x4, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="PCM6-Out3 Playback Volume", index=0 Info: id=64, control=1, cmask=0x2, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="PCM6-Out2 Playback Volume", index=0 Info: id=64, control=1, cmask=0x1, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="PCM6-Out1 Playback Volume", index=0 Info: id=64, control=1, cmask=0x0, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="PCM5-Out6 Playback Volume", index=0 Info: id=64, control=1, cmask=0x10, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="PCM5-Out5 Playback Volume", index=0 Info: id=64, control=1, cmask=0x8, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="PCM5-Out4 Playback Volume", index=0 Info: id=64, control=1, cmask=0x4, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="PCM5-Out3 Playback Volume", index=0 Info: id=64, control=1, cmask=0x2, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="PCM5-Out2 Playback Volume", index=0 Info: id=64, control=1, cmask=0x1, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="PCM5-Out1 Playback Volume", index=0 Info: id=64, control=1, cmask=0x0, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="PCM4-Out6 Playback Volume", index=0 Info: id=64, control=1, cmask=0x10, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="PCM4-Out5 Playback Volume", index=0 Info: id=64, control=1, cmask=0x8, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="PCM4-Out4 Playback Volume", index=0 Info: id=64, control=1, cmask=0x4, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="PCM4-Out3 Playback Volume", index=0 Info: id=64, control=1, cmask=0x2, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="PCM4-Out2 Playback Volume", index=0 Info: id=64, control=1, cmask=0x1, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="PCM4-Out1 Playback Volume", index=0 Info: id=64, control=1, cmask=0x0, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="PCM3-Out6 Playback Volume", index=0 Info: id=64, control=1, cmask=0x10, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="PCM3-Out5 Playback Volume", index=0 Info: id=64, control=1, cmask=0x8, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="PCM3-Out4 Playback Volume", index=0 Info: id=64, control=1, cmask=0x4, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="PCM3-Out3 Playback Volume", index=0 Info: id=64, control=1, cmask=0x2, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="PCM3-Out2 Playback Volume", index=0 Info: id=64, control=1, cmask=0x1, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="PCM3-Out1 Playback Volume", index=0 Info: id=64, control=1, cmask=0x0, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="PCM2-Out6 Playback Volume", index=0 Info: id=64, control=1, cmask=0x10, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="PCM2-Out5 Playback Volume", index=0 Info: id=64, control=1, cmask=0x8, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="PCM2-Out4 Playback Volume", index=0 Info: id=64, control=1, cmask=0x4, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="PCM2-Out3 Playback Volume", index=0 Info: id=64, control=1, cmask=0x2, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="PCM2-Out2 Playback Volume", index=0 Info: id=64, control=1, cmask=0x1, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="PCM2-Out1 Playback Volume", index=0 Info: id=64, control=1, cmask=0x0, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="PCM1-Out6 Playback Volume", index=0 Info: id=64, control=1, cmask=0x10, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="PCM1-Out5 Playback Volume", index=0 Info: id=64, control=1, cmask=0x8, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="PCM1-Out4 Playback Volume", index=0 Info: id=64, control=1, cmask=0x4, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="PCM1-Out3 Playback Volume", index=0 Info: id=64, control=1, cmask=0x2, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="PCM1-Out2 Playback Volume", index=0 Info: id=64, control=1, cmask=0x1, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="PCM1-Out1 Playback Volume", index=0 Info: id=64, control=1, cmask=0x0, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 64 Control: name="PCM Volume", index=4 Info: id=64, control=10, cmask=0xf, channels=4, type="S16" Volume: min=0, max=0, dBmin=0, dBmax=0 Unit: 64 Control: name="PCM Volume", index=3 Info: id=64, control=8, cmask=0xf, channels=4, type="S16" Volume: min=0, max=0, dBmin=0, dBmax=0 Unit: 64 Control: name="PCM Volume", index=2 Info: id=64, control=7, cmask=0xf, channels=4, type="S16" Volume: min=0, max=0, dBmin=0, dBmax=0 Unit: 64 Control: name="PCM Volume", index=1 Info: id=64, control=6, cmask=0xf, channels=4, type="S16" Volume: min=0, max=0, dBmin=0, dBmax=0 Unit: 64 Control: name="PCM Volume", index=0 Info: id=64, control=5, cmask=0xf, channels=4, type="S16" Volume: min=0, max=0, dBmin=0, dBmax=0 Unit: 64 Control: name="Line Volume", index=3 Info: id=64, control=4, cmask=0xf, channels=4, type="S16" Volume: min=0, max=0, dBmin=0, dBmax=0 Unit: 64 Control: name="Line Volume", index=2 Info: id=64, control=3, cmask=0xf, channels=4, type="S16" Volume: min=0, max=0, dBmin=0, dBmax=0 Unit: 64 Control: name="Line Volume", index=1 Info: id=64, control=2, cmask=0xf, channels=4, type="S16" Volume: min=0, max=0, dBmin=0, dBmax=0 Unit: 64 Control: name="Line Volume", index=0 Info: id=64, control=1, cmask=0xf, channels=4, type="S16" Volume: min=0, max=0, dBmin=0, dBmax=0 Unit: 66 Control: name="Effect Send AIn4", index=0 Info: id=66, control=1, cmask=0x100, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 66 Control: name="Effect Send AIn3", index=0 Info: id=66, control=1, cmask=0x80, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 66 Control: name="Effect Send AIn2", index=0 Info: id=66, control=1, cmask=0x40, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 66 Control: name="Effect Send AIn1", index=0 Info: id=66, control=1, cmask=0x20, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 66 Control: name="Effect Send DOut6", index=0 Info: id=66, control=1, cmask=0x10, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 66 Control: name="Effect Send DOut5", index=0 Info: id=66, control=1, cmask=0x8, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 66 Control: name="Effect Send DOut4", index=0 Info: id=66, control=1, cmask=0x4, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 66 Control: name="Effect Send DOut3", index=0 Info: id=66, control=1, cmask=0x2, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 66 Control: name="Effect Send DOut2", index=0 Info: id=66, control=1, cmask=0x1, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 66 Control: name="Effect Send DOut1", index=0 Info: id=66, control=1, cmask=0x0, channels=1, type="S16" Volume: min=46587, max=64766, dBmin=-7401, dBmax=-300 Unit: 67 Control: name="Effect Feedback Volume", index=0 Info: id=67, control=5, cmask=0x0, channels=1, type="U8" Volume: min=0, max=255, dBmin=0, dBmax=99 Unit: 67 Control: name="Effect Duration", index=0 Info: id=67, control=4, cmask=0x0, channels=1, type="S16" Volume: min=0, max=65535, dBmin=0, dBmax=0 Unit: 67 Control: name="Effect Volume", index=0 Info: id=67, control=3, cmask=0x0, channels=1, type="U8" Volume: min=0, max=255, dBmin=0, dBmax=99 Unit: 67 Control: name="Effect Program Switch", index=0 Unit: 128 Control: name="PCM Clock Source", index=0 Info: id=128, control=1, cmask=0x0, channels=1, type="U8" Volume: min=1, max=2, dBmin=0, dBmax=0 Unit: 130 Control: name="Clock Source 130 Validity", index=0 Info: id=130, control=2, cmask=0x0, channels=1, type="BOOLEAN" Volume: min=0, max=1, dBmin=0, dBmax=0 --endcollapse-- !!-------Mixer controls for card C400 Card sysdefault:1 'C400'/'M-Audio Fast Track C400 at usb-0000:0c:00.3-1, high speed' Mixer name : 'USB Mixer' Components : 'USB0763:2030' Controls : 91 Simple ctrls : 90 Simple mixer control 'PCM',0 Capabilities: volume Playback channels: Front Left - Front Right - Rear Left - Rear Right Capture channels: Front Left - Front Right - Rear Left - Rear Right Limits: 0 - 0 Front Left: 0 [0%] Front Right: 0 [0%] Rear Left: 0 [0%] Rear Right: 0 [0%] Simple mixer control 'PCM Clock Source',0 Capabilities: enum Items: 'Internal' 'SPDIF' Item0: 'Internal' Simple mixer control 'PCM1-Out1',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 148 [93%] [-7.92dB] Simple mixer control 'PCM1-Out2',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'PCM1-Out3',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'PCM1-Out4',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'PCM1-Out5',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'PCM1-Out6',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'PCM2-Out1',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'PCM2-Out2',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 148 [93%] [-7.92dB] Simple mixer control 'PCM2-Out3',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'PCM2-Out4',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'PCM2-Out5',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'PCM2-Out6',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'PCM3-Out1',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'PCM3-Out2',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'PCM3-Out3',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'PCM3-Out4',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'PCM3-Out5',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'PCM3-Out6',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'PCM4-Out1',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'PCM4-Out2',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'PCM4-Out3',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'PCM4-Out4',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'PCM4-Out5',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'PCM4-Out6',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'PCM5-Out1',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'PCM5-Out2',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'PCM5-Out3',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'PCM5-Out4',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'PCM5-Out5',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'PCM5-Out6',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'PCM6-Out1',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'PCM6-Out2',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'PCM6-Out3',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'PCM6-Out4',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'PCM6-Out5',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'PCM6-Out6',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'PCM',1 Capabilities: volume Playback channels: Front Left - Front Right - Rear Left - Rear Right Capture channels: Front Left - Front Right - Rear Left - Rear Right Limits: 0 - 0 Front Left: 0 [0%] Front Right: 0 [0%] Rear Left: 0 [0%] Rear Right: 0 [0%] Simple mixer control 'PCM',2 Capabilities: volume Playback channels: Front Left - Front Right - Rear Left - Rear Right Capture channels: Front Left - Front Right - Rear Left - Rear Right Limits: 0 - 0 Front Left: 0 [0%] Front Right: 0 [0%] Rear Left: 0 [0%] Rear Right: 0 [0%] Simple mixer control 'PCM',3 Capabilities: volume Playback channels: Front Left - Front Right - Rear Left - Rear Right Capture channels: Front Left - Front Right - Rear Left - Rear Right Limits: 0 - 0 Front Left: 0 [0%] Front Right: 0 [0%] Rear Left: 0 [0%] Rear Right: 0 [0%] Simple mixer control 'PCM',4 Capabilities: volume Playback channels: Front Left - Front Right - Rear Left - Rear Right Capture channels: Front Left - Front Right - Rear Left - Rear Right Limits: 0 - 0 Front Left: 0 [0%] Front Right: 0 [0%] Rear Left: 0 [0%] Rear Right: 0 [0%] Simple mixer control 'Line',0 Capabilities: volume Playback channels: Front Left - Front Right - Rear Left - Rear Right Capture channels: Front Left - Front Right - Rear Left - Rear Right Limits: 0 - 0 Front Left: 0 [0%] Front Right: 0 [0%] Rear Left: 0 [0%] Rear Right: 0 [0%] Simple mixer control 'Line',1 Capabilities: volume Playback channels: Front Left - Front Right - Rear Left - Rear Right Capture channels: Front Left - Front Right - Rear Left - Rear Right Limits: 0 - 0 Front Left: 0 [0%] Front Right: 0 [0%] Rear Left: 0 [0%] Rear Right: 0 [0%] Simple mixer control 'Line',2 Capabilities: volume Playback channels: Front Left - Front Right - Rear Left - Rear Right Capture channels: Front Left - Front Right - Rear Left - Rear Right Limits: 0 - 0 Front Left: 0 [0%] Front Right: 0 [0%] Rear Left: 0 [0%] Rear Right: 0 [0%] Simple mixer control 'Line',3 Capabilities: volume Playback channels: Front Left - Front Right - Rear Left - Rear Right Capture channels: Front Left - Front Right - Rear Left - Rear Right Limits: 0 - 0 Front Left: 0 [0%] Front Right: 0 [0%] Rear Left: 0 [0%] Rear Right: 0 [0%] Simple mixer control 'Effect',0 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 255 Mono: 0 [0%] [0.00dB] Simple mixer control 'Effect Duration',0 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 285 Mono: 0 [0%] Simple mixer control 'Effect Feedback',0 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 255 Mono: 0 [0%] Simple mixer control 'Effect Program',0 Capabilities: enum Items: 'Room 1' 'Room 2' 'Room 3' 'Hall 1' 'Hall 2' 'Plate' 'Delay' 'Echo' Item0: 'Room 1' Simple mixer control 'Effect Return 1',0 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 162 Mono: 0 [0%] Simple mixer control 'Effect Return 2',0 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 162 Mono: 0 [0%] Simple mixer control 'Effect Return 3',0 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 162 Mono: 0 [0%] Simple mixer control 'Effect Return 4',0 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 162 Mono: 0 [0%] Simple mixer control 'Effect Return 5',0 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 162 Mono: 0 [0%] Simple mixer control 'Effect Return 6',0 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 162 Mono: 0 [0%] Simple mixer control 'Effect Send AIn1',0 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 159 Mono: 0 [0%] Simple mixer control 'Effect Send AIn2',0 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 159 Mono: 0 [0%] Simple mixer control 'Effect Send AIn3',0 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 159 Mono: 0 [0%] Simple mixer control 'Effect Send AIn4',0 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 159 Mono: 0 [0%] Simple mixer control 'Effect Send DOut1',0 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 159 Mono: 0 [0%] Simple mixer control 'Effect Send DOut2',0 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 159 Mono: 0 [0%] Simple mixer control 'Effect Send DOut3',0 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 159 Mono: 0 [0%] Simple mixer control 'Effect Send DOut4',0 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 159 Mono: 0 [0%] Simple mixer control 'Effect Send DOut5',0 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 159 Mono: 0 [0%] Simple mixer control 'Effect Send DOut6',0 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 159 Mono: 0 [0%] Simple mixer control 'In1-Out1',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'In1-Out2',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'In1-Out3',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'In1-Out4',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'In1-Out5',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'In1-Out6',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'In2-Out1',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'In2-Out2',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'In2-Out3',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'In2-Out4',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'In2-Out5',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'In2-Out6',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'In3-Out1',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'In3-Out2',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'In3-Out3',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'In3-Out4',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'In3-Out5',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'In3-Out6',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'In4-Out1',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'In4-Out2',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'In4-Out3',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'In4-Out4',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'In4-Out5',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] Simple mixer control 'In4-Out6',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 159 Mono: Playback 0 [0%] [-74.01dB] !!-------Mixer controls for card Generic Card sysdefault:2 'Generic'/'HD-Audio Generic at 0xfcb00000 irq 125' Mixer name : 'Realtek ALC1220' Components : 'HDA:10ec1220,1458a0c3,00100101' Controls : 47 Simple ctrls : 21 Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 87 Mono: Playback 61 [70%] [-19.50dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 87 Mono: Front Left: Playback 87 [100%] [0.00dB] [on] Front Right: Playback 87 [100%] [0.00dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [0.00dB] Front Right: Playback 255 [100%] [0.00dB] Simple mixer control 'Front',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 87 Mono: Front Left: Playback 87 [100%] [0.00dB] [on] Front Right: Playback 87 [100%] [0.00dB] [on] Simple mixer control 'Front Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Front Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] Simple mixer control 'Surround',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 87 Mono: Front Left: Playback 87 [100%] [0.00dB] [on] Front Right: Playback 87 [100%] [0.00dB] [on] Simple mixer control 'Center',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 87 Mono: Playback 87 [100%] [0.00dB] [on] Simple mixer control 'LFE',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 87 Mono: Playback 87 [100%] [0.00dB] [on] Simple mixer control 'Line',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Line Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958 Default PCM',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 63 Front Left: Capture 39 [62%] [12.00dB] [on] Front Right: Capture 39 [62%] [12.00dB] [on] Simple mixer control 'Capture',1 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 63 Front Left: Capture 0 [0%] [-17.25dB] [off] Front Right: Capture 0 [0%] [-17.25dB] [off] Simple mixer control 'Auto-Mute Mode',0 Capabilities: enum Items: 'Disabled' 'Enabled' Item0: 'Enabled' Simple mixer control 'Input Source',0 Capabilities: cenum Items: 'Front Mic' 'Rear Mic' 'Line' Item0: 'Front Mic' Simple mixer control 'Input Source',1 Capabilities: cenum Items: 'Front Mic' 'Rear Mic' 'Line' Item0: 'Front Mic' Simple mixer control 'Loopback Mixing',0 Capabilities: enum Items: 'Disabled' 'Enabled' Item0: 'Disabled' Simple mixer control 'Rear Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Rear Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/331 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 1/4] ASoC: max98388: fix unused function warnings
by Arnd Bergmann 17 Jun '23

17 Jun '23

From: Arnd Bergmann <arnd(a)arndb.de> The PM functions are never referenced when CONFIG_PM_SLEEP is disabled: sound/soc/codecs/max98388.c:854:12: error: unused function 'max98388_suspend' [-Werror,-Wunused-function] static int max98388_suspend(struct device *dev) ^ sound/soc/codecs/max98388.c:864:12: error: unused function 'max98388_resume' [-Werror,-Wunused-function] static int max98388_resume(struct device *dev) Fix this by using the modern SYSTEM_SLEEP_PM_OPS() macro in place of the deprecated SET_SYSTEM_SLEEP_PM_OPS() version, and use pm_sleep_ptr() to hide the entire structure as well. On a related note, the of_match_ptr() and ACPI_PTR() macros have the same problem and would cause the device id table to be unused when the driver is built-in and the respective subsystems are disabled. This does not cause warnings unless -Wunused-const-variable is passed to the compiler, but it's better to just not use the macros at all here. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/max98388.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/max98388.c b/sound/soc/codecs/max98388.c index 8062a71150074..3d03c4bac6c55 100644 --- a/sound/soc/codecs/max98388.c +++ b/sound/soc/codecs/max98388.c @@ -873,7 +873,7 @@ static int max98388_resume(struct device *dev) } static const struct dev_pm_ops max98388_pm = { - SET_SYSTEM_SLEEP_PM_OPS(max98388_suspend, max98388_resume) + SYSTEM_SLEEP_PM_OPS(max98388_suspend, max98388_resume) }; static const struct regmap_config max98388_regmap = { @@ -998,9 +998,9 @@ MODULE_DEVICE_TABLE(acpi, max98388_acpi_match); static struct i2c_driver max98388_i2c_driver = { .driver = { .name = "max98388", - .of_match_table = of_match_ptr(max98388_of_match), - .acpi_match_table = ACPI_PTR(max98388_acpi_match), - .pm = &max98388_pm, + .of_match_table = max98388_of_match, + .acpi_match_table = max98388_acpi_match, + .pm = pm_sleep_ptr(&max98388_pm), }, .probe = max98388_i2c_probe, .id_table = max98388_i2c_id, -- 2.39.2

3 8

imx8mn: Trying to get EASRC to work
by Fabio Estevam 16 Jun '23

16 Jun '23

Hi, I am working on an imx8mn-based board with an ES9080 codec and I am trying to get EASCR to work. (The ES9080 driver is not upstreamed yet). Audio playback works fine via simple-audio-card and here are my changes to get EASRC to work: - sound-ess { - compatible = "simple-audio-card"; - simple-audio-card,name = "ess-audio"; - simple-audio-card,format = "left_j"; - simple-audio-card,frame-master = <&s2cpu>; - simple-audio-card,bitclock-master = <&s2cpu>; - - s2cpu: simple-audio-card,cpu { - sound-dai = <&sai2>; - dai-tdm-slot-num = <8>; - dai-tdm-slot-width = <32>; - }; - - simple-audio-card,codec { - sound-dai = <&es9080>; - }; + sound-es9080 { + compatible = "fsl,imx-audio-es9080"; + model = "es9080-with-easrc"; + audio-cpu = <&sai2>; + audio-codec = <&es9080>; + audio-asrc = <&easrc>; }; }; diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 8d14b5593658..27b8f2c0cd65 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -624,6 +624,14 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) codec_dai_name = "sgtl5000"; priv->codec_priv.mclk_id = SGTL5000_SYSCLK; priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; + } else if (of_device_is_compatible(np, "fsl,imx-audio-es9080")) { + codec_dai_name = "es9080"; + priv->dai_fmt |= SND_SOC_DAIFMT_CBC_CFC; + priv->dai_link[1].dpcm_capture = 0; + priv->dai_link[2].dpcm_capture = 0; + priv->cpu_priv.slot_width = 32; + priv->card.dapm_routes = audio_map_tx; + priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx); } else if (of_device_is_compatible(np, "fsl,imx-audio-tlv320aic32x4")) { codec_dai_name = "tlv320aic32x4-hifi"; priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; @@ -902,6 +910,7 @@ static const struct of_device_id fsl_asoc_card_dt_ids[] = { { .compatible = "fsl,imx-audio-tlv320aic32x4", }, { .compatible = "fsl,imx-audio-tlv320aic31xx", }, { .compatible = "fsl,imx-audio-sgtl5000", }, + { .compatible = "fsl,imx-audio-es9080", }, { .compatible = "fsl,imx-audio-wm8962", }, { .compatible = "fsl,imx-audio-wm8960", }, { .compatible = "fsl,imx-audio-mqs", }, The es9080-with-easrc card is registered, but after running 'speaker-test' no audio output is present. Is there a way to use the EASRC driver with the simple-audio-card without touching fsl-asoc-card.c? That would be preferable. Would you have any suggestions as to what may be missing to get EASRC to work? Thanks, Fabio Estevam

1 1

[PATCH AUTOSEL 6.1 20/33] ASoC: Intel: avs: Account for UID of ACPI device
by Sasha Levin 16 Jun '23

16 Jun '23

From: Cezary Rojewski <cezary.rojewski(a)intel.com> [ Upstream commit 836855100b87b4dd7a82546131779dc255c18b67 ] Configurations with multiple codecs attached to the platform are supported but only if each from the set is different. Add new field representing the 'Unique ID' so that codecs that share Vendor and Part IDs can be differentiated and thus enabling support for such configurations. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Link: https://lore.kernel.org/r/20230519201711.4073845-6-amadeuszx.slawinski@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/soc-acpi.h | 1 + sound/soc/intel/avs/board_selection.c | 2 +- 2 files changed, 2 insertions(+), 1 deletion(-) diff --git a/include/sound/soc-acpi.h b/include/sound/soc-acpi.h index b38fd25c57295..528279056b3ab 100644 --- a/include/sound/soc-acpi.h +++ b/include/sound/soc-acpi.h @@ -170,6 +170,7 @@ struct snd_soc_acpi_link_adr { /* Descriptor for SST ASoC machine driver */ struct snd_soc_acpi_mach { u8 id[ACPI_ID_LEN]; + const char *uid; const struct snd_soc_acpi_codecs *comp_ids; const u32 link_mask; const struct snd_soc_acpi_link_adr *links; diff --git a/sound/soc/intel/avs/board_selection.c b/sound/soc/intel/avs/board_selection.c index 87f9c18be238d..87353b4b0cd73 100644 --- a/sound/soc/intel/avs/board_selection.c +++ b/sound/soc/intel/avs/board_selection.c @@ -394,7 +394,7 @@ static int avs_register_i2s_boards(struct avs_dev *adev) } for (mach = boards->machs; mach->id[0]; mach++) { - if (!acpi_dev_present(mach->id, NULL, -1)) + if (!acpi_dev_present(mach->id, mach->uid, -1)) continue; if (mach->machine_quirk) -- 2.39.2

2 1

[PATCH AUTOSEL 5.10 13/21] ASoC: soc-pcm: test if a BE can be prepared
by Sasha Levin 16 Jun '23

16 Jun '23

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> [ Upstream commit e123036be377ddf628226a7c6d4f9af5efd113d3 ] In the BE hw_params configuration, the existing code checks if any of the existing FEs are prepared, running, paused or suspended - and skips the configuration in those cases. This allows multiple calls of hw_params which the ALSA state machine supports. This check is not handled for the prepare stage, which can lead to the same BE being prepared multiple times. This patch adds a check similar to that of the hw_params, with the main difference being that the suspended state is allowed: the ALSA state machine allows a transition from suspended to prepared with hw_params skipped. This problem was detected on Intel IPC4/SoundWire devices, where the BE dailink .prepare stage is used to configure the SoundWire stream with a bank switch. Multiple .prepare calls lead to conflicts with the .trigger operation with IPC4 configurations. This problem was not detected earlier on Intel devices, HDaudio BE dailinks detect that the link is already prepared and skip the configuration, and for IPC3 devices there is no BE trigger. Link: https://github.com/thesofproject/sof/issues/7596 Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com Link: https://lore.kernel.org/r/20230517185731.487124-1-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/soc-dpcm.h | 4 ++++ sound/soc/soc-pcm.c | 20 ++++++++++++++++++++ 2 files changed, 24 insertions(+) diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index 0f6c50b17bba8..bd8795198a7d6 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -121,6 +121,10 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream); +/* can this BE perform prepare */ +int snd_soc_dpcm_can_be_prepared(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream); + /* is the current PCM operation for this FE ? */ int snd_soc_dpcm_fe_can_update(struct snd_soc_pcm_runtime *fe, int stream); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index fb874f924bbe3..e52c030bd17a2 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2332,6 +2332,9 @@ int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) if (!snd_soc_dpcm_be_can_update(fe, be, stream)) continue; + if (!snd_soc_dpcm_can_be_prepared(fe, be, stream)) + continue; + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND) && @@ -2972,3 +2975,20 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); + +/* + * We can only prepare a BE DAI if any of it's FE are not prepared, + * running or paused for the specified stream direction. + */ +int snd_soc_dpcm_can_be_prepared(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + const enum snd_soc_dpcm_state state[] = { + SND_SOC_DPCM_STATE_START, + SND_SOC_DPCM_STATE_PAUSED, + SND_SOC_DPCM_STATE_PREPARE, + }; + + return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); +} +EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_prepared); -- 2.39.2

2 1

[PATCH AUTOSEL 6.1 28/57] ASoC: SOF: sof-client-probes: fix pm_runtime imbalance in error handling
by Sasha Levin 16 Jun '23

16 Jun '23

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit bc424273c74c1565c459c8f2a6ed95caee368d0a ] When an error occurs, we need to make sure the device can pm_runtime suspend instead of keeping it active. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com Reviewed-by: Daniel Baluta <daniel.baluta(a)nxp.com Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com Link: https://lore.kernel.org/r/20230512103315.8921-4-peter.ujfalusi@linux.intel.… Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/sof-client-probes.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/sof/sof-client-probes.c b/sound/soc/sof/sof-client-probes.c index ddeabbb5580e1..1e521a9e34d27 100644 --- a/sound/soc/sof/sof-client-probes.c +++ b/sound/soc/sof/sof-client-probes.c @@ -441,12 +441,7 @@ static ssize_t sof_probes_dfs_points_read(struct file *file, char __user *to, ret = sof_probes_points_info(cdev, &desc, &num_desc); if (ret < 0) - goto exit; - - pm_runtime_mark_last_busy(dev); - err = pm_runtime_put_autosuspend(dev); - if (err < 0) - dev_err_ratelimited(dev, "debugfs read failed to idle %d\n", err); + goto pm_error; for (i = 0; i < num_desc; i++) { offset = strlen(buf); @@ -464,6 +459,13 @@ static ssize_t sof_probes_dfs_points_read(struct file *file, char __user *to, ret = simple_read_from_buffer(to, count, ppos, buf, strlen(buf)); kfree(desc); + +pm_error: + pm_runtime_mark_last_busy(dev); + err = pm_runtime_put_autosuspend(dev); + if (err < 0) + dev_err_ratelimited(dev, "debugfs read failed to idle %d\n", err); + exit: kfree(buf); return ret; -- 2.39.2

2 1

topology: nhlt: intel: support m/n divider and xtal/cardinal/pll clock source
by GitHub pull_request - edited 16 Jun '23

16 Jun '23

alsa-project/alsa-utils pull request #220 was edited from brentlu: Current alsatplg supports only cardinal clock source without m/n divider. Code in SOF ssp driver is ported here to implement the divider function. I tested the code on ADL brya device with following setting: MCLK: 19.2MHz BCLK: 3.072Mz (m 24 n 25) Request URL : https://github.com/alsa-project/alsa-utils/pull/220 Patch URL : https://github.com/alsa-project/alsa-utils/pull/220.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

topology: nhlt: intel: support m/n divider and xtal/cardinal/pll clock source
by GitHub pull_request - opened 16 Jun '23

16 Jun '23

alsa-project/alsa-utils pull request #220 was opened from brentlu: Current alsatplg supports only cardinal clock source without m/n divider. Code in SOF ssp driver is ported here to implement the clock source selection and divider function. I tested the code on ADL brya device with following setting: MCLK: 19.2MHz BCLK: 3.072Mz (m 24 n 25) Request URL : https://github.com/alsa-project/alsa-utils/pull/220 Patch URL : https://github.com/alsa-project/alsa-utils/pull/220.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH v2 1/1] ASoC: dt-bindings: tlv320aic32x4: convert to DT schema format
by Alexander Stein 16 Jun '23

16 Jun '23

Convert the binding to DT schema format. Since commit 514b044cba667 ("ASoC: tlv320aic32x4: Model PLL in CCF") clocks & clock-names = "mclk" is mandatory, it has been added to required properties as well. '#sound-dai-cells' is added for reference from simple-audio-card. Signed-off-by: Alexander Stein <alexander.stein(a)ew.tq-group.com> --- Changes in v2: * Rebased onto for-next of [1] * Add vendor prefix 'ti' to file name * Simplify 'reg' property * Removed description from 'reset-gpios' * Added reference to dai-common.yaml * Fixed node name in example There is no in-tree user for aic32x4-gpio-func, so I just converted the bindings to YAML and skipped introducing a dt-binding header for defining the constants. [1] https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git/log/ .../bindings/sound/ti,tlv320aic32x4.yaml | 101 ++++++++++++++++++ .../bindings/sound/tlv320aic32x4.txt | 42 -------- 2 files changed, 101 insertions(+), 42 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/ti,tlv320aic32x4.yaml delete mode 100644 Documentation/devicetree/bindings/sound/tlv320aic32x4.txt diff --git a/Documentation/devicetree/bindings/sound/ti,tlv320aic32x4.yaml b/Documentation/devicetree/bindings/sound/ti,tlv320aic32x4.yaml new file mode 100644 index 0000000000000..a7cc9aa344684 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tlv320aic32x4.yaml @@ -0,0 +1,101 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2019 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tlv320aic32x4.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TLV320AIC32x4 Stereo Audio codec + +maintainers: + - Alexander Stein <alexander.stein(a)ew.tq-group.com> + +description: | + The TLV320AIC32x4 audio codec can be accessed using I2C or SPI + +properties: + compatible: + enum: + - ti,tas2505 + - ti,tlv320aic32x4 + - ti,tlv320aic32x6 + + reg: + maxItems: 1 + + clocks: + items: + - description: Master clock + + clock-names: + items: + - const: mclk + + av-supply: + description: Analog core power supply + + dv-supply: + description: Digital core power supply + + iov-supply: + description: Digital IO power supply + + ldoin-supply: + description: LDO power supply + + reset-gpios: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + aic32x4-gpio-func: + description: | + GPIO function configuration for pins MFP1-MFP5. + Types are defined in include/sound/tlv320aic32x4.h + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 5 + maxItems: 5 + +required: + - compatible + - reg + - clocks + - clock-names + - iov-supply + +allOf: + - $ref: dai-common.yaml# + - if: + not: + required: + - ldoin-supply + then: + required: + - av-supply + - dv-supply + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + audio-codec@18 { + compatible = "ti,tlv320aic32x4"; + reg = <0x18>; + iov-supply = <&reg_3v3>; + ldoin-supply = <&reg_3v3>; + clocks = <&clks 201>; + clock-names = "mclk"; + aic32x4-gpio-func= < + 0xff /* AIC32X4_MFPX_DEFAULT_VALUE */ + 0xff /* AIC32X4_MFPX_DEFAULT_VALUE */ + 0x04 /* MFP3 AIC32X4_MFP3_GPIO_ENABLED */ + 0xff /* AIC32X4_MFPX_DEFAULT_VALUE */ + 0x08 /* MFP5 AIC32X4_MFP5_GPIO_INPUT */ + >; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt b/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt deleted file mode 100644 index 0b4e21bde5bc1..0000000000000 --- a/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt +++ /dev/null @@ -1,42 +0,0 @@ -Texas Instruments - tlv320aic32x4 Codec module - -The tlv320aic32x4 serial control bus communicates through I2C protocols - -Required properties: - - compatible - "string" - One of: - "ti,tlv320aic32x4" TLV320AIC3204 - "ti,tlv320aic32x6" TLV320AIC3206, TLV320AIC3256 - "ti,tas2505" TAS2505, TAS2521 - - reg: I2C slave address - - *-supply: Required supply regulators are: - "iov" - digital IO power supply - "ldoin" - LDO power supply - "dv" - Digital core power supply - "av" - Analog core power supply - If you supply ldoin, dv and av are optional. Otherwise they are required - See regulator/regulator.txt for more information about the detailed binding - format. - -Optional properties: - - reset-gpios: Reset-GPIO phandle with args as described in gpio/gpio.txt - - clocks/clock-names: Clock named 'mclk' for the master clock of the codec. - See clock/clock-bindings.txt for information about the detailed format. - - aic32x4-gpio-func - <array of 5 int> - - Types are defined in include/sound/tlv320aic32x4.h - - -Example: - -codec: tlv320aic32x4@18 { - compatible = "ti,tlv320aic32x4"; - reg = <0x18>; - clocks = <&clks 201>; - clock-names = "mclk"; - aic32x4-gpio-func= < - 0xff /* AIC32X4_MFPX_DEFAULT_VALUE */ - 0xff /* AIC32X4_MFPX_DEFAULT_VALUE */ - 0x04 /* MFP3 AIC32X4_MFP3_GPIO_ENABLED */ - 0xff /* AIC32X4_MFPX_DEFAULT_VALUE */ - 0x08 /* MFP5 AIC32X4_MFP5_GPIO_INPUT */ - >; -}; -- 2.34.1

3 9

[PATCH] ASoC: codecs: wm0010: Add MODULE_FIRMWARE macros
by Juerg Haefliger 16 Jun '23

16 Jun '23

The module loads firmware so add MODULE_FIRMWARE macros to provide that information via modinfo. Signed-off-by: Juerg Haefliger <juerg.haefliger(a)canonical.com> --- sound/soc/codecs/wm0010.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 034a4e858c7e..1d4259433f47 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -994,3 +994,6 @@ module_spi_driver(wm0010_spi_driver); MODULE_DESCRIPTION("ASoC WM0010 driver"); MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); MODULE_LICENSE("GPL"); + +MODULE_FIRMWARE("wm0010.dfw"); +MODULE_FIRMWARE("wm0010_stage2.bin"); -- 2.37.2

3 2

[PATCH] ASoC: rt5677: Add MODULE_FIRMWARE macro
by Juerg Haefliger 16 Jun '23

16 Jun '23

The module loads firmware so add a MODULE_FIRMWARE macro to provide that information via modinfo. Signed-off-by: Juerg Haefliger <juerg.haefliger(a)canonical.com> --- sound/soc/codecs/rt5677.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 3bf019b3f700..c8fb98e3406f 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -5712,3 +5712,5 @@ module_i2c_driver(rt5677_i2c_driver); MODULE_DESCRIPTION("ASoC RT5677 driver"); MODULE_AUTHOR("Oder Chiou <oder_chiou(a)realtek.com>"); MODULE_LICENSE("GPL v2"); + +MODULE_FIRMWARE("rt5677_elf_vad"); -- 2.37.2

2 1

[PATCH 0/8] ASoC: SOF: misc updates for 6.5
by Pierre-Louis Bossart 16 Jun '23

16 Jun '23

Couple of improvements on virtual_widget support, firmware trace free, IPC payload dump, duplicated code in suspend and MeteorLake primary code support. Bard Liao (2): ASoC: SOF: sof-audio: add is_virtual_widget helper ASoC: SOF: sof-audio: test virtual widget in sof_walk_widgets_in_order Peter Ujfalusi (5): ASoC: SOF: core: Free the firmware trace before calling snd_sof_shutdown() ASoC: SOF: Add new sof_debug flag to request message payload dump ASoC: SOF: ipc3: Dump IPC message payload ASoC: SOF: ipc4: Switch to use the sof_debug:bit11 to dump message payload ASoC: SOF: pm: Remove duplicated code in sof_suspend Rander Wang (1): ASoC: SOF: Intel: mtl: setup primary core info on MeteorLake platform sound/soc/sof/core.c | 4 +++- sound/soc/sof/intel/mtl.c | 19 ++++++++++++---- sound/soc/sof/ipc3.c | 39 ++++++++++++++++++++++++++++++++ sound/soc/sof/ipc4.c | 46 ++++++++++++++++++-------------------- sound/soc/sof/pm.c | 12 ++++------ sound/soc/sof/sof-audio.c | 47 ++++++++++++++++++++++++++------------- sound/soc/sof/sof-priv.h | 3 +++ 7 files changed, 117 insertions(+), 53 deletions(-) -- 2.37.2

2 9

[PATCH 1/1] ASoC: dt-bindings: tlv320aic32x4: convert to DT schema format
by Alexander Stein 16 Jun '23

16 Jun '23

Convert the binding to DT schema format. Since commit 514b044cba667 ("ASoC: tlv320aic32x4: Model PLL in CCF") clocks & clock-names = "mclk" is mandatory, it has been added to required properties as well. '#sound-dai-cells' is added for reference from simple-audio-card. Signed-off-by: Alexander Stein <alexander.stein(a)ew.tq-group.com> --- There is no in-tree user for aic32x4-gpio-func, so I just converted the bindings to YAML and skipped introducing a dt-binding header for defining the constants. .../bindings/sound/tlv320aic32x4.txt | 42 ------- .../bindings/sound/tlv320aic32x4.yaml | 103 ++++++++++++++++++ 2 files changed, 103 insertions(+), 42 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/tlv320aic32x4.txt create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic32x4.yaml diff --git a/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt b/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt deleted file mode 100644 index 0b4e21bde5bc1..0000000000000 --- a/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt +++ /dev/null @@ -1,42 +0,0 @@ -Texas Instruments - tlv320aic32x4 Codec module - -The tlv320aic32x4 serial control bus communicates through I2C protocols - -Required properties: - - compatible - "string" - One of: - "ti,tlv320aic32x4" TLV320AIC3204 - "ti,tlv320aic32x6" TLV320AIC3206, TLV320AIC3256 - "ti,tas2505" TAS2505, TAS2521 - - reg: I2C slave address - - *-supply: Required supply regulators are: - "iov" - digital IO power supply - "ldoin" - LDO power supply - "dv" - Digital core power supply - "av" - Analog core power supply - If you supply ldoin, dv and av are optional. Otherwise they are required - See regulator/regulator.txt for more information about the detailed binding - format. - -Optional properties: - - reset-gpios: Reset-GPIO phandle with args as described in gpio/gpio.txt - - clocks/clock-names: Clock named 'mclk' for the master clock of the codec. - See clock/clock-bindings.txt for information about the detailed format. - - aic32x4-gpio-func - <array of 5 int> - - Types are defined in include/sound/tlv320aic32x4.h - - -Example: - -codec: tlv320aic32x4@18 { - compatible = "ti,tlv320aic32x4"; - reg = <0x18>; - clocks = <&clks 201>; - clock-names = "mclk"; - aic32x4-gpio-func= < - 0xff /* AIC32X4_MFPX_DEFAULT_VALUE */ - 0xff /* AIC32X4_MFPX_DEFAULT_VALUE */ - 0x04 /* MFP3 AIC32X4_MFP3_GPIO_ENABLED */ - 0xff /* AIC32X4_MFPX_DEFAULT_VALUE */ - 0x08 /* MFP5 AIC32X4_MFP5_GPIO_INPUT */ - >; -}; diff --git a/Documentation/devicetree/bindings/sound/tlv320aic32x4.yaml b/Documentation/devicetree/bindings/sound/tlv320aic32x4.yaml new file mode 100644 index 0000000000000..ebf84cdcb68d5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tlv320aic32x4.yaml @@ -0,0 +1,103 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2019 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/tlv320aic32x4.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TLV320AIC32x4 Stereo Audio codec + +maintainers: + - Alexander Stein <alexander.stein(a)ew.tq-group.com> + +description: | + The TLV320AIC32x4 audio codec can be accessed using I2C or SPI + +properties: + compatible: + enum: + - ti,tas2505 + - ti,tlv320aic32x4 + - ti,tlv320aic32x6 + + reg: + description: | + I2C address or SPI chip select of the device + + clocks: + items: + - description: Master clock + + clock-names: + items: + - const: mclk + + av-supply: + description: Analog core power supply + + dv-supply: + description: Digital core power supply + + iov-supply: + description: Digital IO power supply + + ldoin-supply: + description: LDO power supply + + reset-gpios: + maxItems: 1 + description: | + GPIO used for hardware reset + + '#sound-dai-cells': + const: 0 + + aic32x4-gpio-func: + description: | + GPIO function configuration for pins MFP1-MFP5. + Types are defined in include/sound/tlv320aic32x4.h + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 5 + maxItems: 5 + +required: + - compatible + - reg + - clocks + - clock-names + - iov-supply + +additionalProperties: false + +allOf: + - if: + not: + required: + - ldoin-supply + then: + required: + - av-supply + - dv-supply + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec: tlv320aic32x4@18 { + compatible = "ti,tlv320aic32x4"; + reg = <0x18>; + iov-supply = <&reg_3v3>; + ldoin-supply = <&reg_3v3>; + clocks = <&clks 201>; + clock-names = "mclk"; + aic32x4-gpio-func= < + 0xff /* AIC32X4_MFPX_DEFAULT_VALUE */ + 0xff /* AIC32X4_MFPX_DEFAULT_VALUE */ + 0x04 /* MFP3 AIC32X4_MFP3_GPIO_ENABLED */ + 0xff /* AIC32X4_MFPX_DEFAULT_VALUE */ + 0x08 /* MFP5 AIC32X4_MFP5_GPIO_INPUT */ + >; + }; + }; -- 2.34.1

3 3

[PATCH] ASoC: amd: correct pm_runtime enable sequence
by Arun Gopal Kondaveeti 16 Jun '23

16 Jun '23

Replace pm_runtime_allow() with pm_runtime_mark_last_busy() & pm_runtime_set_active() in pm_runtime enable sequence for dma drivers. Signed-off-by: Arun Gopal <arungopal.kondaveeti(a)amd.com> --- sound/soc/amd/ps/ps-pdm-dma.c | 3 ++- sound/soc/amd/raven/acp3x-pcm-dma.c | 3 ++- sound/soc/amd/renoir/acp3x-pdm-dma.c | 3 ++- sound/soc/amd/vangogh/acp5x-pcm-dma.c | 4 ++-- sound/soc/amd/yc/acp6x-pdm-dma.c | 3 ++- 5 files changed, 10 insertions(+), 6 deletions(-) diff --git a/sound/soc/amd/ps/ps-pdm-dma.c b/sound/soc/amd/ps/ps-pdm-dma.c index bdbbb797c74d..d48f7c5af289 100644 --- a/sound/soc/amd/ps/ps-pdm-dma.c +++ b/sound/soc/amd/ps/ps-pdm-dma.c @@ -391,8 +391,9 @@ static int acp63_pdm_audio_probe(struct platform_device *pdev) } pm_runtime_set_autosuspend_delay(&pdev->dev, ACP_SUSPEND_DELAY_MS); pm_runtime_use_autosuspend(&pdev->dev); + pm_runtime_mark_last_busy(&pdev->dev); + pm_runtime_set_active(&pdev->dev); pm_runtime_enable(&pdev->dev); - pm_runtime_allow(&pdev->dev); return 0; } diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 7362dd15ad30..9538f3ffc5d9 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -416,8 +416,9 @@ static int acp3x_audio_probe(struct platform_device *pdev) pm_runtime_set_autosuspend_delay(&pdev->dev, 2000); pm_runtime_use_autosuspend(&pdev->dev); + pm_runtime_mark_last_busy(&pdev->dev); + pm_runtime_set_active(&pdev->dev); pm_runtime_enable(&pdev->dev); - pm_runtime_allow(&pdev->dev); return 0; } diff --git a/sound/soc/amd/renoir/acp3x-pdm-dma.c b/sound/soc/amd/renoir/acp3x-pdm-dma.c index 4e299f96521f..c3b47e9bd239 100644 --- a/sound/soc/amd/renoir/acp3x-pdm-dma.c +++ b/sound/soc/amd/renoir/acp3x-pdm-dma.c @@ -430,8 +430,9 @@ static int acp_pdm_audio_probe(struct platform_device *pdev) } pm_runtime_set_autosuspend_delay(&pdev->dev, ACP_SUSPEND_DELAY_MS); pm_runtime_use_autosuspend(&pdev->dev); + pm_runtime_mark_last_busy(&pdev->dev); + pm_runtime_set_active(&pdev->dev); pm_runtime_enable(&pdev->dev); - pm_runtime_allow(&pdev->dev); return 0; } diff --git a/sound/soc/amd/vangogh/acp5x-pcm-dma.c b/sound/soc/amd/vangogh/acp5x-pcm-dma.c index 29901ee4bfe3..587dec5bb33d 100644 --- a/sound/soc/amd/vangogh/acp5x-pcm-dma.c +++ b/sound/soc/amd/vangogh/acp5x-pcm-dma.c @@ -409,9 +409,9 @@ static int acp5x_audio_probe(struct platform_device *pdev) } pm_runtime_set_autosuspend_delay(&pdev->dev, 2000); pm_runtime_use_autosuspend(&pdev->dev); + pm_runtime_mark_last_busy(&pdev->dev); + pm_runtime_set_active(&pdev->dev); pm_runtime_enable(&pdev->dev); - pm_runtime_allow(&pdev->dev); - return 0; } diff --git a/sound/soc/amd/yc/acp6x-pdm-dma.c b/sound/soc/amd/yc/acp6x-pdm-dma.c index d818eba48546..72c4591e451b 100644 --- a/sound/soc/amd/yc/acp6x-pdm-dma.c +++ b/sound/soc/amd/yc/acp6x-pdm-dma.c @@ -383,8 +383,9 @@ static int acp6x_pdm_audio_probe(struct platform_device *pdev) } pm_runtime_set_autosuspend_delay(&pdev->dev, ACP_SUSPEND_DELAY_MS); pm_runtime_use_autosuspend(&pdev->dev); + pm_runtime_mark_last_busy(&pdev->dev); + pm_runtime_set_active(&pdev->dev); pm_runtime_enable(&pdev->dev); - pm_runtime_allow(&pdev->dev); return 0; } -- 2.25.1

3 3

[PATCH v4 00/13] Add support for IIO devices in ASoC
by Herve Codina 16 Jun '23

16 Jun '23

Several weeks ago, I sent a series [1] for adding a potentiometer as an auxiliary device in ASoC. The feedback was that the potentiometer should be directly handled in IIO (as other potentiometers) and something more generic should be present in ASoC in order to have a binding to import some IIO devices into sound cards. The series related to the IIO potentiometer device is already applied. This series introduces audio-iio-aux. Its goal is to offer the binding between IIO and ASoC. It exposes attached IIO devices as ASoC auxiliary devices and allows to control them through mixer controls. On my system, the IIO device is a potentiometer and it is present in an amplifier design present in the audio path. Compare to the previous iteration https://lore.kernel.org/linux-kernel/20230612122926.107333-1-herve.codina@b… This v4 series mainly: - Fixes headers inclusion - Removes unneeded variable initialization - Adds a missing error check - Fixes typos Best regards, Hervé [1] https://lore.kernel.org/linux-kernel/20230203111422.142479-1-herve.codina@b… [2] https://lore.kernel.org/linux-kernel/20230421085245.302169-1-herve.codina@b… Changes v3 -> v4 - Patches 1, 2 No changes. - Patches 3, 4, 5 Add 'Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com>'. - Patch 6 (new in v4) Fix headers inclusion order. - Patch 7 (patch 6 in v3) Add a comment related to __must_be_array() Use __array[0] of *__array - Patch 8 (patch 7 in v3) Fix minmax.h inclusion order. Add 'Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com>'. - Patch 9 (patch 8 in v3) Add 'Suggested-by: Jonathan Cameron <Jonathan.Cameron(a)huawei.com>'. Add 'Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com>'. - Patch 10 (patch 9 in v3) Add 'Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com>'. - Patch 11 (patch 10 in v3) Fix a typo. Add 'Suggested-by: Andy Shevchenko <andy.shevchenko(a)gmail.com>'. Add 'Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com>'. - Patch 12 (patch 11 in v3) Fix typos in the commit log. Fix headers inclusion order. Removed unneeded variable initialization. Replace {0} by {}. Use struct device *dev in probe(). Check an error on the snd-control-invert-range property read. - Patch 13 (patch12 in v3) No changes. Changes v2 -> v3 - Patches 1, 2 No changes. - Patch 3, 4 Add 'Acked-by: Jonathan Cameron <Jonathan.Cameron(a)huawei.com>'. - Patch 5 (new in v3) Removed the 'unused' variable and check the null pointer when used. - Patch 6 (new in v3) Introduce {min,max}_array(). - Patch 7 (new in v3) Use max_array() in iio_channel_read_max(). - Patch 8 (new in v3) Replace a FIXME comment by a TODO one. - Patch 9 (patch 5 in v2) Removed the 'unused' variable and check the null pointer when used. Use min_array(). Remplace a FIXME comment by a TODO one. - Patch 10 (patch 6 in v2) Convert existing macros to return a compound litteral instead of adding a new helper. - Patch 11 (patch 7 in v2) Remove the file name from the C file header. Use directly converted DAPM macros. Replace <linux/module.h> by <linux/mod_devicetable.h>. Add <linux/platform_device.h>. Be sure that min <= max. Swap values if it is not the case. Move the bool structure member after the int ones. Remove unneeded assignements. Use dev_err_probe() when relevant. Use str_on_off(). Use static_assert() instead of BUILD_BUG_ON(). Remove unneeded comma and blank line. Use device_property_*() instead of the OF API. - patch 8 available in v2 removed as already applied - Patch 12 (patch 9 in v2) Use devm_add_action_or_reset(). Call simple_populate_aux() from simple_parse_of(). Changes v1 -> v2 - Patch 1 Rename simple-iio-aux to audio-iio-aux Rename invert to snd-control-invert-range Remove the /schemas/iio/iio-consumer.yaml reference Remove the unneeded '|' after description - Patch 2 (new in v2) Introduce the simple-audio-card additional-devs subnode - Patch 3 (new in v2) Check err before switch() in iio_channel_read_max() - Patch 4 (new in v2) Fix raw reads and raw writes documentation - Patch 5 (patch 2 in v1) Check err before switch() in iio_channel_read_min() Fix documentation - Patch 6 (path 3 in v1) No changes - Patch 7 (patch 4 in v1) Rename simple-iio-aux to audio-iio-aux Rename invert to snd-control-invert-range Remove the mask usage from audio_iio_aux_{get,put}_volsw helpers Use directly PTR_ERR() in dev_err_probe() parameter Remove the '!!' construction Remove of_match_ptr() - Patch 8 (new in v2) Add a missing of_node_put() in the simple-card driver - Patch 9 (new in v2) Handle additional-devs in the simple-card driver Herve Codina (13): ASoC: dt-bindings: Add audio-iio-aux ASoC: dt-bindings: simple-card: Add additional-devs subnode iio: inkern: Check error explicitly in iio_channel_read_max() iio: consumer.h: Fix raw values documentation notes iio: inkern: Remove the 'unused' variable usage in iio_channel_read_max() iio: inkern: Fix headers inclusion order minmax: Introduce {min,max}_array() iio: inkern: Use max_array() to get the maximum value from an array iio: inkern: Replace a FIXME comment by a TODO one iio: inkern: Add a helper to query an available minimum raw value ASoC: soc-dapm.h: Convert macros to return a compound literal ASoC: codecs: Add support for the generic IIO auxiliary devices ASoC: simple-card: Handle additional devices .../bindings/sound/audio-iio-aux.yaml | 64 ++++ .../bindings/sound/simple-card.yaml | 53 +++ drivers/iio/inkern.c | 86 ++++- include/linux/iio/consumer.h | 37 +- include/linux/minmax.h | 36 ++ include/sound/soc-dapm.h | 138 ++++--- sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/audio-iio-aux.c | 338 ++++++++++++++++++ sound/soc/generic/simple-card.c | 46 ++- 10 files changed, 741 insertions(+), 71 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/audio-iio-aux.yaml create mode 100644 sound/soc/codecs/audio-iio-aux.c -- 2.40.1

3 25

Re: sound/soc/soc-pcm.c:2089:5: warning: stack frame size (2064) exceeds limit (2048) in 'dpcm_be_dai_trigger'
by Pierre-Louis Bossart 16 Jun '23

16 Jun '23

On 6/15/23 20:34, kernel test robot wrote: > Hi Pierre-Louis, > > First bad commit (maybe != root cause): > > tree: https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git master > head: b6dad5178ceaf23f369c3711062ce1f2afc33644 > commit: 9995c1d096c8ab1b5f1edc4141257719f6a53524 ASoC: soc-pcm: improve BE transition for PAUSE_RELEASE > date: 1 year, 2 months ago > config: riscv-buildonly-randconfig-r001-20230615 (https://download.01.org/0day-ci/archive/20230616/202306160240.ahGjvPqw-lkp@…) > compiler: clang version 14.0.6 (https://github.com/llvm/llvm-project.git f28c006a5895fc0e329fe15fead81e37457cb1d1) > reproduce (this is a W=1 build): > mkdir -p ~/bin > wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross > chmod +x ~/bin/make.cross > # install riscv cross compiling tool for clang build > # apt-get install binutils-riscv64-linux-gnu > # https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/commit/?… > git remote add linus https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git > git fetch --no-tags linus master > git checkout 9995c1d096c8ab1b5f1edc4141257719f6a53524 > # save the config file > mkdir build_dir && cp config build_dir/.config > COMPILER_INSTALL_PATH=$HOME/0day COMPILER=clang ~/bin/make.cross W=1 O=build_dir ARCH=riscv olddefconfig > COMPILER_INSTALL_PATH=$HOME/0day COMPILER=clang ~/bin/make.cross W=1 O=build_dir ARCH=riscv SHELL=/bin/bash ./ drivers/ fs/ lib/zstd/ mm/ sound/pci/hda/ sound/soc/ > > If you fix the issue in a separate patch/commit (i.e. not just a new version of > the same patch/commit), kindly add following tags > | Reported-by: kernel test robot <lkp(a)intel.com> > | Closes: https://lore.kernel.org/oe-kbuild-all/202306160240.ahGjvPqw-lkp@intel.com/ > > All warnings (new ones prefixed by >>): > >>> sound/soc/soc-pcm.c:2089:5: warning: stack frame size (2064) exceeds limit (2048) in 'dpcm_be_dai_trigger' [-Wframe-larger-than] > int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, Not sure what causes this warning to show up, the code hasn't changed in 1.2 years, nor what to do about it.... > ^ > 1 warning generated. > > > vim +/dpcm_be_dai_trigger +2089 sound/soc/soc-pcm.c > > 01d7584cd2e5a93 Liam Girdwood 2012-04-25 2088 > 23607025303af6e Liam Girdwood 2014-01-17 @2089 int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, > 45c0a188ca59e7e Mark Brown 2012-05-09 2090 int cmd) > 01d7584cd2e5a93 Liam Girdwood 2012-04-25 2091 { > db3aa39c9106842 Kuninori Morimoto 2021-03-15 2092 struct snd_soc_pcm_runtime *be; > 9995c1d096c8ab1 Pierre-Louis Bossart 2022-04-06 2093 bool pause_stop_transition; > 01d7584cd2e5a93 Liam Girdwood 2012-04-25 2094 struct snd_soc_dpcm *dpcm; > b2ae80663008a76 Takashi Iwai 2021-12-07 2095 unsigned long flags; > 01d7584cd2e5a93 Liam Girdwood 2012-04-25 2096 int ret = 0; > 01d7584cd2e5a93 Liam Girdwood 2012-04-25 2097 > 8d6258a4dd26783 Kuninori Morimoto 2018-09-18 2098 for_each_dpcm_be(fe, stream, dpcm) { > db3aa39c9106842 Kuninori Morimoto 2021-03-15 2099 struct snd_pcm_substream *be_substream; > 01d7584cd2e5a93 Liam Girdwood 2012-04-25 2100 > db3aa39c9106842 Kuninori Morimoto 2021-03-15 2101 be = dpcm->be; > db3aa39c9106842 Kuninori Morimoto 2021-03-15 2102 be_substream = snd_soc_dpcm_get_substream(be, stream); > 01d7584cd2e5a93 Liam Girdwood 2012-04-25 2103 > 3c75c0ea5da749b Takashi Iwai 2022-01-19 2104 snd_soc_dpcm_stream_lock_irqsave_nested(be, stream, flags); > b2ae80663008a76 Takashi Iwai 2021-12-07 2105 > 01d7584cd2e5a93 Liam Girdwood 2012-04-25 2106 /* is this op for this BE ? */ > 01d7584cd2e5a93 Liam Girdwood 2012-04-25 2107 if (!snd_soc_dpcm_be_can_update(fe, be, stream)) > b2ae80663008a76 Takashi Iwai 2021-12-07 2108 goto next; > 01d7584cd2e5a93 Liam Girdwood 2012-04-25 2109 > a9faca15a644ff7 Kuninori Morimoto 2020-12-01 2110 dev_dbg(be->dev, "ASoC: trigger BE %s cmd %d\n", > a9faca15a644ff7 Kuninori Morimoto 2020-12-01 2111 be->dai_link->name, cmd); > a9faca15a644ff7 Kuninori Morimoto 2020-12-01 2112 > 01d7584cd2e5a93 Liam Girdwood 2012-04-25 2113 switch (cmd) { > 01d7584cd2e5a93 Liam Girdwood 2012-04-25 2114 case SNDRV_PCM_TRIGGER_START: > 848aedfdc6ba25a Pierre-Louis Bossart 2021-12-07 2115 if (!be->dpcm[stream].be_start && > 848aedfdc6ba25a Pierre-Louis Bossart 2021-12-07 2116 (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && > 21fca8bdbb64df1 이경택 2020-04-01 2117 (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && > 3202e2f5fac0032 Mark Brown 2021-08-30 2118 (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) > b2ae80663008a76 Takashi Iwai 2021-12-07 2119 goto next; > 6479f7588651cbc Pierre-Louis Bossart 2021-08-17 2120 > 848aedfdc6ba25a Pierre-Louis Bossart 2021-12-07 2121 be->dpcm[stream].be_start++; > 848aedfdc6ba25a Pierre-Louis Bossart 2021-12-07 2122 if (be->dpcm[stream].be_start != 1) > 848aedfdc6ba25a Pierre-Louis Bossart 2021-12-07 2123 goto next; > 848aedfdc6ba25a Pierre-Louis Bossart 2021-12-07 2124 > a9faca15a644ff7 Kuninori Morimoto 2020-12-01 2125 ret = soc_pcm_trigger(be_substream, cmd); > 848aedfdc6ba25a Pierre-Louis Bossart 2021-12-07 2126 if (ret) { > 848aedfdc6ba25a Pierre-Louis Bossart 2021-12-07 2127 be->dpcm[stream].be_start--; > b2ae80663008a76 Takashi Iwai 2021-12-07 2128 goto next; > 848aedfdc6ba25a Pierre-Louis Bossart 2021-12-07 2129 } > 01d7584cd2e5a93 Liam Girdwood 2012-04-25 2130 > 3202e2f5fac0032 Mark Brown 2021-08-30 2131 be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; > 01d7584cd2e5a93 Liam Girdwood 2012-04-25 2132 break; > 01d7584cd2e5a93 Liam Girdwood 2012-04-25 2133 case SNDRV_PCM_TRIGGER_RESUME: > 3202e2f5fac0032 Mark Brown 2021-08-30 2134 if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND)) > b2ae80663008a76 Takashi Iwai 2021-12-07 2135 goto next; > 6479f7588651cbc Pierre-Louis Bossart 2021-08-17 2136 > 848aedfdc6ba25a Pierre-Louis Bossart 2021-12-07 2137 be->dpcm[stream].be_start++; > 848aedfdc6ba25a Pierre-Louis Bossart 2021-12-07 2138 if (be->dpcm[stream].be_start != 1) > 848aedfdc6ba25a Pierre-Louis Bossart 2021-12-07 2139 goto next; > 848aedfdc6ba25a Pierre-Louis Bossart 2021-12-07 2140 > a9faca15a644ff7 Kuninori Morimoto 2020-12-01 2141 ret = soc_pcm_trigger(be_substream, cmd); > 848aedfdc6ba25a Pierre-Louis Bossart 2021-12-07 2142 if (ret) { > 848aedfdc6ba25a Pierre-Louis Bossart 2021-12-07 2143 be->dpcm[stream].be_start--; > b2ae80663008a76 Takashi Iwai 2021-12-07 2144 goto next; > 848aedfdc6ba25a Pierre-Louis Bossart 2021-12-07 2145 } > 01d7584cd2e5a93 Liam Girdwood 2012-04-25 2146 > 3202e2f5fac0032 Mark Brown 2021-08-30 2147 be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; > 01d7584cd2e5a93 Liam Girdwood 2012-04-25 2148 break; > 01d7584cd2e5a93 Liam Girdwood 2012-04-25 2149 case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: > 3aa1e96a2b95e2e Pierre-Louis Bossart 2021-12-07 2150 if (!be->dpcm[stream].be_start && > 3aa1e96a2b95e2e Pierre-Louis Bossart 2021-12-07 2151 (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) && > 3aa1e96a2b95e2e Pierre-Louis Bossart 2021-12-07 2152 (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) > b2ae80663008a76 Takashi Iwai 2021-12-07 2153 goto next; > 6479f7588651cbc Pierre-Louis Bossart 2021-08-17 2154 > 9995c1d096c8ab1 Pierre-Louis Bossart 2022-04-06 2155 fe->dpcm[stream].fe_pause = false; > 9995c1d096c8ab1 Pierre-Louis Bossart 2022-04-06 2156 be->dpcm[stream].be_pause--; > 9995c1d096c8ab1 Pierre-Louis Bossart 2022-04-06 2157 > 848aedfdc6ba25a Pierre-Louis Bossart 2021-12-07 2158 be->dpcm[stream].be_start++; > 848aedfdc6ba25a Pierre-Louis Bossart 2021-12-07 2159 if (be->dpcm[stream].be_start != 1) > 848aedfdc6ba25a Pierre-Louis Bossart 2021-12-07 2160 goto next; > 848aedfdc6ba25a Pierre-Louis Bossart 2021-12-07 2161 > a9faca15a644ff7 Kuninori Morimoto 2020-12-01 2162 ret = soc_pcm_trigger(be_substream, cmd); > 848aedfdc6ba25a Pierre-Louis Bossart 2021-12-07 2163 if (ret) { > 848aedfdc6ba25a Pierre-Louis Bossart 2021-12-07 2164 be->dpcm[stream].be_start--; > b2ae80663008a76 Takashi Iwai 2021-12-07 2165 goto next; > 848aedfdc6ba25a Pierre-Louis Bossart 2021-12-07 2166 } > 01d7584cd2e5a93 Liam Girdwood 2012-04-25 2167 > 3202e2f5fac0032 Mark Brown 2021-08-30 2168 be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; > 01d7584cd2e5a93 Liam Girdwood 2012-04-25 2169 break; > 01d7584cd2e5a93 Liam Girdwood 2012-04-25 2170 case SNDRV_PCM_TRIGGER_STOP: > 21fca8bdbb64df1 이경택 2020-04-01 2171 if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) && > 3202e2f5fac0032 Mark Brown 2021-08-30 2172 (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) > b2ae80663008a76 Takashi Iwai 2021-12-07 2173 goto next; > 01d7584cd2e5a93 Liam Girdwood 2012-04-25 2174 > 848aedfdc6ba25a Pierre-Louis Bossart 2021-12-07 2175 if (be->dpcm[stream].state == SND_SOC_DPCM_STATE_START) > 848aedfdc6ba25a Pierre-Louis Bossart 2021-12-07 2176 be->dpcm[stream].be_start--; > 848aedfdc6ba25a Pierre-Louis Bossart 2021-12-07 2177 > 848aedfdc6ba25a Pierre-Louis Bossart 2021-12-07 2178 if (be->dpcm[stream].be_start != 0) > b2ae80663008a76 Takashi Iwai 2021-12-07 2179 goto next; > 01d7584cd2e5a93 Liam Girdwood 2012-04-25 2180 > 9995c1d096c8ab1 Pierre-Louis Bossart 2022-04-06 2181 pause_stop_transition = false; > 9995c1d096c8ab1 Pierre-Louis Bossart 2022-04-06 2182 if (fe->dpcm[stream].fe_pause) { > 9995c1d096c8ab1 Pierre-Louis Bossart 2022-04-06 2183 pause_stop_transition = true; > 9995c1d096c8ab1 Pierre-Louis Bossart 2022-04-06 2184 fe->dpcm[stream].fe_pause = false; > 9995c1d096c8ab1 Pierre-Louis Bossart 2022-04-06 2185 be->dpcm[stream].be_pause--; > 9995c1d096c8ab1 Pierre-Louis Bossart 2022-04-06 2186 } > 9995c1d096c8ab1 Pierre-Louis Bossart 2022-04-06 2187 > 9995c1d096c8ab1 Pierre-Louis Bossart 2022-04-06 2188 if (be->dpcm[stream].be_pause != 0) > 9995c1d096c8ab1 Pierre-Louis Bossart 2022-04-06 2189 ret = soc_pcm_trigger(be_substream, SNDRV_PCM_TRIGGER_PAUSE_PUSH); > 9995c1d096c8ab1 Pierre-Louis Bossart 2022-04-06 2190 else > 9995c1d096c8ab1 Pierre-Louis Bossart 2022-04-06 2191 ret = soc_pcm_trigger(be_substream, SNDRV_PCM_TRIGGER_STOP); > 9995c1d096c8ab1 Pierre-Louis Bossart 2022-04-06 2192 > 848aedfdc6ba25a Pierre-Louis Bossart 2021-12-07 2193 if (ret) { > 848aedfdc6ba25a Pierre-Louis Bossart 2021-12-07 2194 if (be->dpcm[stream].state == SND_SOC_DPCM_STATE_START) > 848aedfdc6ba25a Pierre-Louis Bossart 2021-12-07 2195 be->dpcm[stream].be_start++; > 9995c1d096c8ab1 Pierre-Louis Bossart 2022-04-06 2196 if (pause_stop_transition) { > 9995c1d096c8ab1 Pierre-Louis Bossart 2022-04-06 2197 fe->dpcm[stream].fe_pause = true; > 9995c1d096c8ab1 Pierre-Louis Bossart 2022-04-06 2198 be->dpcm[stream].be_pause++; > 9995c1d096c8ab1 Pierre-Louis Bossart 2022-04-06 2199 } > b2ae80663008a76 Takashi Iwai 2021-12-07 2200 goto next; > 848aedfdc6ba25a Pierre-Louis Bossart 2021-12-07 2201 } > 01d7584cd2e5a93 Liam Girdwood 2012-04-25 2202 > 9995c1d096c8ab1 Pierre-Louis Bossart 2022-04-06 2203 if (be->dpcm[stream].be_pause != 0) > 9995c1d096c8ab1 Pierre-Louis Bossart 2022-04-06 2204 be->dpcm[stream].state = SND_SOC_DPCM_STATE_PAUSED; > 9995c1d096c8ab1 Pierre-Louis Bossart 2022-04-06 2205 else > 3202e2f5fac0032 Mark Brown 2021-08-30 2206 be->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP; > 9995c1d096c8ab1 Pierre-Louis Bossart 2022-04-06 2207 > 01d7584cd2e5a93 Liam Girdwood 2012-04-25 2208 break; > 01d7584cd2e5a93 Liam Girdwood 2012-04-25 2209 case SNDRV_PCM_TRIGGER_SUSPEND: > 3202e2f5fac0032 Mark Brown 2021-08-30 2210 if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) > b2ae80663008a76 Takashi Iwai 2021-12-07 2211 goto next; > 01d7584cd2e5a93 Liam Girdwood 2012-04-25 2212 > 848aedfdc6ba25a Pierre-Louis Bossart 2021-12-07 2213 be->dpcm[stream].be_start--; > 848aedfdc6ba25a Pierre-Louis Bossart 2021-12-07 2214 if (be->dpcm[stream].be_start != 0) > b2ae80663008a76 Takashi Iwai 2021-12-07 2215 goto next; > 01d7584cd2e5a93 Liam Girdwood 2012-04-25 2216 > a9faca15a644ff7 Kuninori Morimoto 2020-12-01 2217 ret = soc_pcm_trigger(be_substream, cmd); > 848aedfdc6ba25a Pierre-Louis Bossart 2021-12-07 2218 if (ret) { > 848aedfdc6ba25a Pierre-Louis Bossart 2021-12-07 2219 be->dpcm[stream].be_start++; > b2ae80663008a76 Takashi Iwai 2021-12-07 2220 goto next; > 848aedfdc6ba25a Pierre-Louis Bossart 2021-12-07 2221 } > 01d7584cd2e5a93 Liam Girdwood 2012-04-25 2222 > 3202e2f5fac0032 Mark Brown 2021-08-30 2223 be->dpcm[stream].state = SND_SOC_DPCM_STATE_SUSPEND; > 01d7584cd2e5a93 Liam Girdwood 2012-04-25 2224 break; > 01d7584cd2e5a93 Liam Girdwood 2012-04-25 2225 case SNDRV_PCM_TRIGGER_PAUSE_PUSH: > 3202e2f5fac0032 Mark Brown 2021-08-30 2226 if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) > b2ae80663008a76 Takashi Iwai 2021-12-07 2227 goto next; > 01d7584cd2e5a93 Liam Girdwood 2012-04-25 2228 > 9995c1d096c8ab1 Pierre-Louis Bossart 2022-04-06 2229 fe->dpcm[stream].fe_pause = true; > 9995c1d096c8ab1 Pierre-Louis Bossart 2022-04-06 2230 be->dpcm[stream].be_pause++; > 9995c1d096c8ab1 Pierre-Louis Bossart 2022-04-06 2231 > 848aedfdc6ba25a Pierre-Louis Bossart 2021-12-07 2232 be->dpcm[stream].be_start--; > 848aedfdc6ba25a Pierre-Louis Bossart 2021-12-07 2233 if (be->dpcm[stream].be_start != 0) > b2ae80663008a76 Takashi Iwai 2021-12-07 2234 goto next; > 01d7584cd2e5a93 Liam Girdwood 2012-04-25 2235 > a9faca15a644ff7 Kuninori Morimoto 2020-12-01 2236 ret = soc_pcm_trigger(be_substream, cmd); > 848aedfdc6ba25a Pierre-Louis Bossart 2021-12-07 2237 if (ret) { > 848aedfdc6ba25a Pierre-Louis Bossart 2021-12-07 2238 be->dpcm[stream].be_start++; > b2ae80663008a76 Takashi Iwai 2021-12-07 2239 goto next; > 848aedfdc6ba25a Pierre-Louis Bossart 2021-12-07 2240 } > 01d7584cd2e5a93 Liam Girdwood 2012-04-25 2241 > 3202e2f5fac0032 Mark Brown 2021-08-30 2242 be->dpcm[stream].state = SND_SOC_DPCM_STATE_PAUSED; > 01d7584cd2e5a93 Liam Girdwood 2012-04-25 2243 break; > 01d7584cd2e5a93 Liam Girdwood 2012-04-25 2244 } > b2ae80663008a76 Takashi Iwai 2021-12-07 2245 next: > b2ae80663008a76 Takashi Iwai 2021-12-07 2246 snd_soc_dpcm_stream_unlock_irqrestore(be, stream, flags); > b2ae80663008a76 Takashi Iwai 2021-12-07 2247 if (ret) > b2ae80663008a76 Takashi Iwai 2021-12-07 2248 break; > 01d7584cd2e5a93 Liam Girdwood 2012-04-25 2249 } > db3aa39c9106842 Kuninori Morimoto 2021-03-15 2250 if (ret < 0) > db3aa39c9106842 Kuninori Morimoto 2021-03-15 2251 dev_err(fe->dev, "ASoC: %s() failed at %s (%d)\n", > db3aa39c9106842 Kuninori Morimoto 2021-03-15 2252 __func__, be->dai_link->name, ret); > 01d7584cd2e5a93 Liam Girdwood 2012-04-25 2253 return ret; > 01d7584cd2e5a93 Liam Girdwood 2012-04-25 2254 } > 01d7584cd2e5a93 Liam Girdwood 2012-04-25 2255 EXPORT_SYMBOL_GPL(dpcm_be_dai_trigger); > 01d7584cd2e5a93 Liam Girdwood 2012-04-25 2256 > > :::::: The code at line 2089 was first introduced by commit > :::::: 23607025303af6e84bc2cd4cabe89c21f6a22a3f ASoC: DPCM: make some DPCM API calls non static for compressed usage > > :::::: TO: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> > :::::: CC: Mark Brown <broonie(a)linaro.org> >

2 1

[PATCH AUTOSEL 4.19 2/6] ASoC: nau8824: Add quirk to active-high jack-detect
by Sasha Levin 16 Jun '23

16 Jun '23

From: Edson Juliano Drosdeck <edson.drosdeck(a)gmail.com> [ Upstream commit e384dba03e3294ce7ea69e4da558e9bf8f0e8946 ] Add entries for Positivo laptops: CW14Q01P, K1424G, N14ZP74G to the DMI table, so that active-high jack-detect will work properly on these laptops. Signed-off-by: Edson Juliano Drosdeck <edson.drosdeck(a)gmail.com> Link: https://lore.kernel.org/r/20230529181911.632851-1-edson.drosdeck@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/nau8824.c | 24 ++++++++++++++++++++++++ 1 file changed, 24 insertions(+) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 4f18bb272e929..0ecea65a80b46 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -1899,6 +1899,30 @@ static const struct dmi_system_id nau8824_quirk_table[] = { }, .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH), }, + { + /* Positivo CW14Q01P */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Positivo Tecnologia SA"), + DMI_MATCH(DMI_BOARD_NAME, "CW14Q01P"), + }, + .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH), + }, + { + /* Positivo K1424G */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Positivo Tecnologia SA"), + DMI_MATCH(DMI_BOARD_NAME, "K1424G"), + }, + .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH), + }, + { + /* Positivo N14ZP74G */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Positivo Tecnologia SA"), + DMI_MATCH(DMI_BOARD_NAME, "N14ZP74G"), + }, + .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH), + }, {} }; -- 2.39.2

1 0

[PATCH AUTOSEL 5.4 3/8] ASoC: nau8824: Add quirk to active-high jack-detect
by Sasha Levin 16 Jun '23

16 Jun '23

From: Edson Juliano Drosdeck <edson.drosdeck(a)gmail.com> [ Upstream commit e384dba03e3294ce7ea69e4da558e9bf8f0e8946 ] Add entries for Positivo laptops: CW14Q01P, K1424G, N14ZP74G to the DMI table, so that active-high jack-detect will work properly on these laptops. Signed-off-by: Edson Juliano Drosdeck <edson.drosdeck(a)gmail.com> Link: https://lore.kernel.org/r/20230529181911.632851-1-edson.drosdeck@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/nau8824.c | 24 ++++++++++++++++++++++++ 1 file changed, 24 insertions(+) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index a95fe3fff1db8..9b22219a76937 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -1896,6 +1896,30 @@ static const struct dmi_system_id nau8824_quirk_table[] = { }, .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH), }, + { + /* Positivo CW14Q01P */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Positivo Tecnologia SA"), + DMI_MATCH(DMI_BOARD_NAME, "CW14Q01P"), + }, + .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH), + }, + { + /* Positivo K1424G */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Positivo Tecnologia SA"), + DMI_MATCH(DMI_BOARD_NAME, "K1424G"), + }, + .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH), + }, + { + /* Positivo N14ZP74G */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Positivo Tecnologia SA"), + DMI_MATCH(DMI_BOARD_NAME, "N14ZP74G"), + }, + .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH), + }, {} }; -- 2.39.2

1 0

[PATCH AUTOSEL 5.4 2/8] ASoC: simple-card: Add missing of_node_put() in case of error
by Sasha Levin 16 Jun '23

16 Jun '23

From: Herve Codina <herve.codina(a)bootlin.com> [ Upstream commit 8938f75a5e35c597a647c28984a0304da7a33d63 ] In the error path, a of_node_put() for platform is missing. Just add it. Signed-off-by: Herve Codina <herve.codina(a)bootlin.com> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Link: https://lore.kernel.org/r/20230523151223.109551-9-herve.codina@bootlin.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/generic/simple-card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 4484c40c7a39c..653181d7eaa34 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -404,6 +404,7 @@ static int simple_for_each_link(struct asoc_simple_priv *priv, if (ret < 0) { of_node_put(codec); + of_node_put(plat); of_node_put(np); goto error; } -- 2.39.2

1 0

[PATCH AUTOSEL 5.10 06/14] ASoC: nau8824: Add quirk to active-high jack-detect
by Sasha Levin 16 Jun '23

16 Jun '23

From: Edson Juliano Drosdeck <edson.drosdeck(a)gmail.com> [ Upstream commit e384dba03e3294ce7ea69e4da558e9bf8f0e8946 ] Add entries for Positivo laptops: CW14Q01P, K1424G, N14ZP74G to the DMI table, so that active-high jack-detect will work properly on these laptops. Signed-off-by: Edson Juliano Drosdeck <edson.drosdeck(a)gmail.com> Link: https://lore.kernel.org/r/20230529181911.632851-1-edson.drosdeck@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/nau8824.c | 24 ++++++++++++++++++++++++ 1 file changed, 24 insertions(+) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index a95fe3fff1db8..9b22219a76937 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -1896,6 +1896,30 @@ static const struct dmi_system_id nau8824_quirk_table[] = { }, .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH), }, + { + /* Positivo CW14Q01P */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Positivo Tecnologia SA"), + DMI_MATCH(DMI_BOARD_NAME, "CW14Q01P"), + }, + .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH), + }, + { + /* Positivo K1424G */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Positivo Tecnologia SA"), + DMI_MATCH(DMI_BOARD_NAME, "K1424G"), + }, + .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH), + }, + { + /* Positivo N14ZP74G */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Positivo Tecnologia SA"), + DMI_MATCH(DMI_BOARD_NAME, "N14ZP74G"), + }, + .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH), + }, {} }; -- 2.39.2

1 0

[PATCH AUTOSEL 5.10 05/14] soundwire: qcom: add proper error paths in qcom_swrm_startup()
by Sasha Levin 16 Jun '23

16 Jun '23

From: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> [ Upstream commit 99e09b9c0ab43346c52f2787ca4e5c4b1798362e ] Reverse actions in qcom_swrm_startup() error paths to avoid leaking stream memory and keeping runtime PM unbalanced. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20230517163736.997553-1-krzysztof.kozlowski@linar… Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/qcom.c | 17 +++++++++++++---- 1 file changed, 13 insertions(+), 4 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index ac73258792e6c..68ec7fde48cba 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -643,8 +643,10 @@ static int qcom_swrm_startup(struct snd_pcm_substream *substream, int ret, i; sruntime = sdw_alloc_stream(dai->name); - if (!sruntime) - return -ENOMEM; + if (!sruntime) { + ret = -ENOMEM; + goto err_alloc; + } ctrl->sruntime[dai->id] = sruntime; @@ -654,12 +656,19 @@ static int qcom_swrm_startup(struct snd_pcm_substream *substream, if (ret < 0 && ret != -ENOTSUPP) { dev_err(dai->dev, "Failed to set sdw stream on %s", codec_dai->name); - sdw_release_stream(sruntime); - return ret; + goto err_set_stream; } } return 0; + +err_set_stream: + sdw_release_stream(sruntime); +err_alloc: + pm_runtime_mark_last_busy(ctrl->dev); + pm_runtime_put_autosuspend(ctrl->dev); + + return ret; } static void qcom_swrm_shutdown(struct snd_pcm_substream *substream, -- 2.39.2

1 0

[PATCH AUTOSEL 5.10 04/14] ASoC: simple-card: Add missing of_node_put() in case of error
by Sasha Levin 16 Jun '23

16 Jun '23

From: Herve Codina <herve.codina(a)bootlin.com> [ Upstream commit 8938f75a5e35c597a647c28984a0304da7a33d63 ] In the error path, a of_node_put() for platform is missing. Just add it. Signed-off-by: Herve Codina <herve.codina(a)bootlin.com> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Link: https://lore.kernel.org/r/20230523151223.109551-9-herve.codina@bootlin.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/generic/simple-card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index d916ec69c24ff..ac97e8b7978c7 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -410,6 +410,7 @@ static int simple_for_each_link(struct asoc_simple_priv *priv, if (ret < 0) { of_node_put(codec); + of_node_put(plat); of_node_put(np); goto error; } -- 2.39.2

1 0

[PATCH AUTOSEL 5.15 07/16] ASoC: nau8824: Add quirk to active-high jack-detect
by Sasha Levin 16 Jun '23

16 Jun '23

From: Edson Juliano Drosdeck <edson.drosdeck(a)gmail.com> [ Upstream commit e384dba03e3294ce7ea69e4da558e9bf8f0e8946 ] Add entries for Positivo laptops: CW14Q01P, K1424G, N14ZP74G to the DMI table, so that active-high jack-detect will work properly on these laptops. Signed-off-by: Edson Juliano Drosdeck <edson.drosdeck(a)gmail.com> Link: https://lore.kernel.org/r/20230529181911.632851-1-edson.drosdeck@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/nau8824.c | 24 ++++++++++++++++++++++++ 1 file changed, 24 insertions(+) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 27589900f4fbf..f7ff130a9485d 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -1866,6 +1866,30 @@ static const struct dmi_system_id nau8824_quirk_table[] = { }, .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH), }, + { + /* Positivo CW14Q01P */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Positivo Tecnologia SA"), + DMI_MATCH(DMI_BOARD_NAME, "CW14Q01P"), + }, + .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH), + }, + { + /* Positivo K1424G */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Positivo Tecnologia SA"), + DMI_MATCH(DMI_BOARD_NAME, "K1424G"), + }, + .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH), + }, + { + /* Positivo N14ZP74G */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Positivo Tecnologia SA"), + DMI_MATCH(DMI_BOARD_NAME, "N14ZP74G"), + }, + .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH), + }, {} }; -- 2.39.2

1 0

[PATCH AUTOSEL 5.15 06/16] soundwire: qcom: add proper error paths in qcom_swrm_startup()
by Sasha Levin 16 Jun '23

16 Jun '23

From: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> [ Upstream commit 99e09b9c0ab43346c52f2787ca4e5c4b1798362e ] Reverse actions in qcom_swrm_startup() error paths to avoid leaking stream memory and keeping runtime PM unbalanced. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20230517163736.997553-1-krzysztof.kozlowski@linar… Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/qcom.c | 17 +++++++++++++---- 1 file changed, 13 insertions(+), 4 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 2ba0911f5d0cc..64b157f5f80db 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -1026,8 +1026,10 @@ static int qcom_swrm_startup(struct snd_pcm_substream *substream, int ret, i; sruntime = sdw_alloc_stream(dai->name); - if (!sruntime) - return -ENOMEM; + if (!sruntime) { + ret = -ENOMEM; + goto err_alloc; + } ctrl->sruntime[dai->id] = sruntime; @@ -1037,12 +1039,19 @@ static int qcom_swrm_startup(struct snd_pcm_substream *substream, if (ret < 0 && ret != -ENOTSUPP) { dev_err(dai->dev, "Failed to set sdw stream on %s\n", codec_dai->name); - sdw_release_stream(sruntime); - return ret; + goto err_set_stream; } } return 0; + +err_set_stream: + sdw_release_stream(sruntime); +err_alloc: + pm_runtime_mark_last_busy(ctrl->dev); + pm_runtime_put_autosuspend(ctrl->dev); + + return ret; } static void qcom_swrm_shutdown(struct snd_pcm_substream *substream, -- 2.39.2

1 0

[PATCH AUTOSEL 5.15 05/16] soundwire: dmi-quirks: add new mapping for HP Spectre x360
by Sasha Levin 16 Jun '23

16 Jun '23

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 700581ede41d029403feec935df4616309696fd7 ] A BIOS/DMI update seems to have broken some devices, let's add a new mapping. Link: https://github.com/thesofproject/linux/issues/4323 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://lore.kernel.org/r/20230515074859.3097-1-yung-chuan.liao@linux.intel… Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/dmi-quirks.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/drivers/soundwire/dmi-quirks.c b/drivers/soundwire/dmi-quirks.c index 2bf534632f644..39f0cc2a5b333 100644 --- a/drivers/soundwire/dmi-quirks.c +++ b/drivers/soundwire/dmi-quirks.c @@ -63,6 +63,13 @@ static const struct dmi_system_id adr_remap_quirk_table[] = { }, .driver_data = (void *)intel_tgl_bios, }, + { + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "HP"), + DMI_MATCH(DMI_BOARD_NAME, "8709"), + }, + .driver_data = (void *)intel_tgl_bios, + }, { /* quirk used for NUC15 'Bishop County' LAPBC510 and LAPBC710 skews */ .matches = { -- 2.39.2

1 0

[PATCH AUTOSEL 5.15 04/16] ASoC: simple-card: Add missing of_node_put() in case of error
by Sasha Levin 16 Jun '23

16 Jun '23

From: Herve Codina <herve.codina(a)bootlin.com> [ Upstream commit 8938f75a5e35c597a647c28984a0304da7a33d63 ] In the error path, a of_node_put() for platform is missing. Just add it. Signed-off-by: Herve Codina <herve.codina(a)bootlin.com> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Link: https://lore.kernel.org/r/20230523151223.109551-9-herve.codina@bootlin.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/generic/simple-card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index bc3e24c6a28a8..283aa21879aa5 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -417,6 +417,7 @@ static int __simple_for_each_link(struct asoc_simple_priv *priv, if (ret < 0) { of_node_put(codec); + of_node_put(plat); of_node_put(np); goto error; } -- 2.39.2

1 0

[PATCH AUTOSEL 6.1 16/26] ALSA: hda/realtek: Add "Intel Reference board" and "NUC 13" SSID in the ALC256
by Sasha Levin 16 Jun '23

16 Jun '23

From: "Sayed, Karimuddin" <karimuddin.sayed(a)intel.com> [ Upstream commit 1a93f10c5b12bd766a537b24a50fca5373467303 ] Add "Intel Reference boad" and "Intel NUC 13" SSID in the alc256. Enable jack headset volume buttons Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Sayed, Karimuddin <karimuddin.sayed(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20230602193812.66768-1-pierre-louis.bossart@linux… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7b5f194513c7b..42bd8ae5892b6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9588,6 +9588,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x10ec, 0x124c, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x10ec, 0x1252, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x10ec, 0x1254, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), + SND_PCI_QUIRK(0x10ec, 0x12cc, "Intel Reference board", ALC225_FIXUP_HEADSET_JACK), SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-SZ6", ALC269_FIXUP_HEADSET_MODE), SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_AMP), @@ -9807,6 +9808,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x8086, 0x2074, "Intel NUC 8", ALC233_FIXUP_INTEL_NUC8_DMIC), SND_PCI_QUIRK(0x8086, 0x2080, "Intel NUC 8 Rugged", ALC256_FIXUP_INTEL_NUC8_RUGGED), SND_PCI_QUIRK(0x8086, 0x2081, "Intel NUC 10", ALC256_FIXUP_INTEL_NUC10), + SND_PCI_QUIRK(0x8086, 0x3038, "Intel NUC 13", ALC225_FIXUP_HEADSET_JACK), SND_PCI_QUIRK(0xf111, 0x0001, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), #if 0 -- 2.39.2

1 0

[PATCH AUTOSEL 6.1 15/26] ASoC: fsl_sai: Enable BCI bit if SAI works on synchronous mode with BYP asserted
by Sasha Levin 16 Jun '23

16 Jun '23

From: Chancel Liu <chancel.liu(a)nxp.com> [ Upstream commit 32cf0046a652116d6a216d575f3049a9ff9dd80d ] There's an issue on SAI synchronous mode that TX/RX side can't get BCLK from RX/TX it sync with if BYP bit is asserted. It's a workaround to fix it that enable SION of IOMUX pad control and assert BCI. For example if TX sync with RX which means both TX and RX are using clk form RX and BYP=1. TX can get BCLK only if the following two conditions are valid: 1. SION of RX BCLK IOMUX pad is set to 1 2. BCI of TX is set to 1 Signed-off-by: Chancel Liu <chancel.liu(a)nxp.com> Acked-by: Shengjiu Wang <shengjiu.wang(a)gmail.com> Link: https://lore.kernel.org/r/20230530103012.3448838-1-chancel.liu@nxp.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl_sai.c | 11 +++++++++-- sound/soc/fsl/fsl_sai.h | 1 + 2 files changed, 10 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 6d88af5b287fe..b33104715c7ba 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -491,14 +491,21 @@ static int fsl_sai_set_bclk(struct snd_soc_dai *dai, bool tx, u32 freq) regmap_update_bits(sai->regmap, reg, FSL_SAI_CR2_MSEL_MASK, FSL_SAI_CR2_MSEL(sai->mclk_id[tx])); - if (savediv == 1) + if (savediv == 1) { regmap_update_bits(sai->regmap, reg, FSL_SAI_CR2_DIV_MASK | FSL_SAI_CR2_BYP, FSL_SAI_CR2_BYP); - else + if (fsl_sai_dir_is_synced(sai, adir)) + regmap_update_bits(sai->regmap, FSL_SAI_xCR2(tx, ofs), + FSL_SAI_CR2_BCI, FSL_SAI_CR2_BCI); + else + regmap_update_bits(sai->regmap, FSL_SAI_xCR2(tx, ofs), + FSL_SAI_CR2_BCI, 0); + } else { regmap_update_bits(sai->regmap, reg, FSL_SAI_CR2_DIV_MASK | FSL_SAI_CR2_BYP, savediv / 2 - 1); + } if (sai->soc_data->max_register >= FSL_SAI_MCTL) { /* SAI is in master mode at this point, so enable MCLK */ diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 697f6690068c8..c5423f81e4560 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -116,6 +116,7 @@ /* SAI Transmit and Receive Configuration 2 Register */ #define FSL_SAI_CR2_SYNC BIT(30) +#define FSL_SAI_CR2_BCI BIT(28) #define FSL_SAI_CR2_MSEL_MASK (0x3 << 26) #define FSL_SAI_CR2_MSEL_BUS 0 #define FSL_SAI_CR2_MSEL_MCLK1 BIT(26) -- 2.39.2

1 0

[PATCH AUTOSEL 6.1 11/26] ASoC: amd: yc: Add Thinkpad Neo14 to quirks list for acp6x
by Sasha Levin 16 Jun '23

16 Jun '23

From: Sicong Jiang <kevin.jiangsc(a)gmail.com> [ Upstream commit 57d1e8900495cf1751cec74db16fe1a0fe47efbb ] Thinkpad Neo14 Ryzen Edition uses Ryzen 6800H processor, and adding to quirks list for acp6x will enable internal mic. Signed-off-by: Sicong Jiang <kevin.jiangsc(a)gmail.com> Link: https://lore.kernel.org/r/20230531090635.89565-1-kevin.jiangsc@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- 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 84b401b685f7f..c1ca3ceac5f2f 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -171,6 +171,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "21CL"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "21EF"), + } + }, { .driver_data = &acp6x_card, .matches = { -- 2.39.2

1 0

[PATCH AUTOSEL 6.1 10/26] ASoC: nau8824: Add quirk to active-high jack-detect
by Sasha Levin 16 Jun '23

16 Jun '23

From: Edson Juliano Drosdeck <edson.drosdeck(a)gmail.com> [ Upstream commit e384dba03e3294ce7ea69e4da558e9bf8f0e8946 ] Add entries for Positivo laptops: CW14Q01P, K1424G, N14ZP74G to the DMI table, so that active-high jack-detect will work properly on these laptops. Signed-off-by: Edson Juliano Drosdeck <edson.drosdeck(a)gmail.com> Link: https://lore.kernel.org/r/20230529181911.632851-1-edson.drosdeck@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/nau8824.c | 24 ++++++++++++++++++++++++ 1 file changed, 24 insertions(+) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 4f19fd9b65d11..5a4db8944d06a 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -1903,6 +1903,30 @@ static const struct dmi_system_id nau8824_quirk_table[] = { }, .driver_data = (void *)(NAU8824_MONO_SPEAKER), }, + { + /* Positivo CW14Q01P */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Positivo Tecnologia SA"), + DMI_MATCH(DMI_BOARD_NAME, "CW14Q01P"), + }, + .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH), + }, + { + /* Positivo K1424G */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Positivo Tecnologia SA"), + DMI_MATCH(DMI_BOARD_NAME, "K1424G"), + }, + .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH), + }, + { + /* Positivo N14ZP74G */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Positivo Tecnologia SA"), + DMI_MATCH(DMI_BOARD_NAME, "N14ZP74G"), + }, + .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH), + }, {} }; -- 2.39.2

1 0

[PATCH AUTOSEL 6.1 09/26] soundwire: qcom: add proper error paths in qcom_swrm_startup()
by Sasha Levin 16 Jun '23

16 Jun '23

From: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> [ Upstream commit 99e09b9c0ab43346c52f2787ca4e5c4b1798362e ] Reverse actions in qcom_swrm_startup() error paths to avoid leaking stream memory and keeping runtime PM unbalanced. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20230517163736.997553-1-krzysztof.kozlowski@linar… Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/qcom.c | 17 +++++++++++++---- 1 file changed, 13 insertions(+), 4 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 21c50972047f5..b2eb3090f4b46 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -1090,8 +1090,10 @@ static int qcom_swrm_startup(struct snd_pcm_substream *substream, } sruntime = sdw_alloc_stream(dai->name); - if (!sruntime) - return -ENOMEM; + if (!sruntime) { + ret = -ENOMEM; + goto err_alloc; + } ctrl->sruntime[dai->id] = sruntime; @@ -1101,12 +1103,19 @@ static int qcom_swrm_startup(struct snd_pcm_substream *substream, if (ret < 0 && ret != -ENOTSUPP) { dev_err(dai->dev, "Failed to set sdw stream on %s\n", codec_dai->name); - sdw_release_stream(sruntime); - return ret; + goto err_set_stream; } } return 0; + +err_set_stream: + sdw_release_stream(sruntime); +err_alloc: + pm_runtime_mark_last_busy(ctrl->dev); + pm_runtime_put_autosuspend(ctrl->dev); + + return ret; } static void qcom_swrm_shutdown(struct snd_pcm_substream *substream, -- 2.39.2

1 0

[PATCH AUTOSEL 6.1 08/26] soundwire: dmi-quirks: add new mapping for HP Spectre x360
by Sasha Levin 16 Jun '23

16 Jun '23

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 700581ede41d029403feec935df4616309696fd7 ] A BIOS/DMI update seems to have broken some devices, let's add a new mapping. Link: https://github.com/thesofproject/linux/issues/4323 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://lore.kernel.org/r/20230515074859.3097-1-yung-chuan.liao@linux.intel… Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/dmi-quirks.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/drivers/soundwire/dmi-quirks.c b/drivers/soundwire/dmi-quirks.c index 58ea013fa918a..2a1096dab63d3 100644 --- a/drivers/soundwire/dmi-quirks.c +++ b/drivers/soundwire/dmi-quirks.c @@ -99,6 +99,13 @@ static const struct dmi_system_id adr_remap_quirk_table[] = { }, .driver_data = (void *)intel_tgl_bios, }, + { + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "HP"), + DMI_MATCH(DMI_BOARD_NAME, "8709"), + }, + .driver_data = (void *)intel_tgl_bios, + }, { /* quirk used for NUC15 'Bishop County' LAPBC510 and LAPBC710 skews */ .matches = { -- 2.39.2

1 0

[PATCH AUTOSEL 6.1 07/26] ASoC: simple-card: Add missing of_node_put() in case of error
by Sasha Levin 16 Jun '23

16 Jun '23

From: Herve Codina <herve.codina(a)bootlin.com> [ Upstream commit 8938f75a5e35c597a647c28984a0304da7a33d63 ] In the error path, a of_node_put() for platform is missing. Just add it. Signed-off-by: Herve Codina <herve.codina(a)bootlin.com> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Link: https://lore.kernel.org/r/20230523151223.109551-9-herve.codina@bootlin.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/generic/simple-card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index feb55b66239b8..fbb682747f598 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -416,6 +416,7 @@ static int __simple_for_each_link(struct asoc_simple_priv *priv, if (ret < 0) { of_node_put(codec); + of_node_put(plat); of_node_put(np); goto error; } -- 2.39.2

1 0

[PATCH AUTOSEL 6.1 06/26] ASoC: codecs: wcd938x-sdw: do not set can_multi_write flag
by Sasha Levin 16 Jun '23

16 Jun '23

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> [ Upstream commit 2d7c2f9272de6347a9cec0fc07708913692c0ae3 ] regmap-sdw does not support multi register writes, so there is no point in setting this flag. This also leads to incorrect programming of WSA codecs with regmap_multi_reg_write() call. This invalid configuration should have been rejected by regmap-sdw. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Link: https://lore.kernel.org/r/20230523165414.14560-1-srinivas.kandagatla@linaro… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wcd938x-sdw.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wcd938x-sdw.c b/sound/soc/codecs/wcd938x-sdw.c index 402286dfaea44..9c10200ff34b2 100644 --- a/sound/soc/codecs/wcd938x-sdw.c +++ b/sound/soc/codecs/wcd938x-sdw.c @@ -1190,7 +1190,6 @@ static const struct regmap_config wcd938x_regmap_config = { .readable_reg = wcd938x_readable_register, .writeable_reg = wcd938x_writeable_register, .volatile_reg = wcd938x_volatile_register, - .can_multi_write = true, }; static const struct sdw_slave_ops wcd9380_slave_ops = { -- 2.39.2

1 0

[PATCH AUTOSEL 6.3 18/30] ALSA: hda/realtek: Add "Intel Reference board" and "NUC 13" SSID in the ALC256
by Sasha Levin 16 Jun '23

16 Jun '23

From: "Sayed, Karimuddin" <karimuddin.sayed(a)intel.com> [ Upstream commit 1a93f10c5b12bd766a537b24a50fca5373467303 ] Add "Intel Reference boad" and "Intel NUC 13" SSID in the alc256. Enable jack headset volume buttons Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Sayed, Karimuddin <karimuddin.sayed(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20230602193812.66768-1-pierre-louis.bossart@linux… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7b5f194513c7b..42bd8ae5892b6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9588,6 +9588,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x10ec, 0x124c, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x10ec, 0x1252, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x10ec, 0x1254, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), + SND_PCI_QUIRK(0x10ec, 0x12cc, "Intel Reference board", ALC225_FIXUP_HEADSET_JACK), SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-SZ6", ALC269_FIXUP_HEADSET_MODE), SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_AMP), @@ -9807,6 +9808,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x8086, 0x2074, "Intel NUC 8", ALC233_FIXUP_INTEL_NUC8_DMIC), SND_PCI_QUIRK(0x8086, 0x2080, "Intel NUC 8 Rugged", ALC256_FIXUP_INTEL_NUC8_RUGGED), SND_PCI_QUIRK(0x8086, 0x2081, "Intel NUC 10", ALC256_FIXUP_INTEL_NUC10), + SND_PCI_QUIRK(0x8086, 0x3038, "Intel NUC 13", ALC225_FIXUP_HEADSET_JACK), SND_PCI_QUIRK(0xf111, 0x0001, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), #if 0 -- 2.39.2

1 0

[PATCH AUTOSEL 6.3 16/30] ASoC: fsl_sai: Enable BCI bit if SAI works on synchronous mode with BYP asserted
by Sasha Levin 16 Jun '23

16 Jun '23

From: Chancel Liu <chancel.liu(a)nxp.com> [ Upstream commit 32cf0046a652116d6a216d575f3049a9ff9dd80d ] There's an issue on SAI synchronous mode that TX/RX side can't get BCLK from RX/TX it sync with if BYP bit is asserted. It's a workaround to fix it that enable SION of IOMUX pad control and assert BCI. For example if TX sync with RX which means both TX and RX are using clk form RX and BYP=1. TX can get BCLK only if the following two conditions are valid: 1. SION of RX BCLK IOMUX pad is set to 1 2. BCI of TX is set to 1 Signed-off-by: Chancel Liu <chancel.liu(a)nxp.com> Acked-by: Shengjiu Wang <shengjiu.wang(a)gmail.com> Link: https://lore.kernel.org/r/20230530103012.3448838-1-chancel.liu@nxp.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl_sai.c | 11 +++++++++-- sound/soc/fsl/fsl_sai.h | 1 + 2 files changed, 10 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 990bba0be1fb1..8a64f3c1d1556 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -491,14 +491,21 @@ static int fsl_sai_set_bclk(struct snd_soc_dai *dai, bool tx, u32 freq) regmap_update_bits(sai->regmap, reg, FSL_SAI_CR2_MSEL_MASK, FSL_SAI_CR2_MSEL(sai->mclk_id[tx])); - if (savediv == 1) + if (savediv == 1) { regmap_update_bits(sai->regmap, reg, FSL_SAI_CR2_DIV_MASK | FSL_SAI_CR2_BYP, FSL_SAI_CR2_BYP); - else + if (fsl_sai_dir_is_synced(sai, adir)) + regmap_update_bits(sai->regmap, FSL_SAI_xCR2(tx, ofs), + FSL_SAI_CR2_BCI, FSL_SAI_CR2_BCI); + else + regmap_update_bits(sai->regmap, FSL_SAI_xCR2(tx, ofs), + FSL_SAI_CR2_BCI, 0); + } else { regmap_update_bits(sai->regmap, reg, FSL_SAI_CR2_DIV_MASK | FSL_SAI_CR2_BYP, savediv / 2 - 1); + } if (sai->soc_data->max_register >= FSL_SAI_MCTL) { /* SAI is in master mode at this point, so enable MCLK */ diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 197748a888d5f..a53c4f0e25faf 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -116,6 +116,7 @@ /* SAI Transmit and Receive Configuration 2 Register */ #define FSL_SAI_CR2_SYNC BIT(30) +#define FSL_SAI_CR2_BCI BIT(28) #define FSL_SAI_CR2_MSEL_MASK (0x3 << 26) #define FSL_SAI_CR2_MSEL_BUS 0 #define FSL_SAI_CR2_MSEL_MCLK1 BIT(26) -- 2.39.2

1 0

[PATCH AUTOSEL 6.3 12/30] ASoC: amd: yc: Add Thinkpad Neo14 to quirks list for acp6x
by Sasha Levin 16 Jun '23

16 Jun '23

From: Sicong Jiang <kevin.jiangsc(a)gmail.com> [ Upstream commit 57d1e8900495cf1751cec74db16fe1a0fe47efbb ] Thinkpad Neo14 Ryzen Edition uses Ryzen 6800H processor, and adding to quirks list for acp6x will enable internal mic. Signed-off-by: Sicong Jiang <kevin.jiangsc(a)gmail.com> Link: https://lore.kernel.org/r/20230531090635.89565-1-kevin.jiangsc@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- 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 84b401b685f7f..c1ca3ceac5f2f 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -171,6 +171,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "21CL"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "21EF"), + } + }, { .driver_data = &acp6x_card, .matches = { -- 2.39.2

1 0

[PATCH AUTOSEL 6.3 11/30] ASoC: nau8824: Add quirk to active-high jack-detect
by Sasha Levin 16 Jun '23

16 Jun '23

From: Edson Juliano Drosdeck <edson.drosdeck(a)gmail.com> [ Upstream commit e384dba03e3294ce7ea69e4da558e9bf8f0e8946 ] Add entries for Positivo laptops: CW14Q01P, K1424G, N14ZP74G to the DMI table, so that active-high jack-detect will work properly on these laptops. Signed-off-by: Edson Juliano Drosdeck <edson.drosdeck(a)gmail.com> Link: https://lore.kernel.org/r/20230529181911.632851-1-edson.drosdeck@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/nau8824.c | 24 ++++++++++++++++++++++++ 1 file changed, 24 insertions(+) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 4f19fd9b65d11..5a4db8944d06a 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -1903,6 +1903,30 @@ static const struct dmi_system_id nau8824_quirk_table[] = { }, .driver_data = (void *)(NAU8824_MONO_SPEAKER), }, + { + /* Positivo CW14Q01P */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Positivo Tecnologia SA"), + DMI_MATCH(DMI_BOARD_NAME, "CW14Q01P"), + }, + .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH), + }, + { + /* Positivo K1424G */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Positivo Tecnologia SA"), + DMI_MATCH(DMI_BOARD_NAME, "K1424G"), + }, + .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH), + }, + { + /* Positivo N14ZP74G */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Positivo Tecnologia SA"), + DMI_MATCH(DMI_BOARD_NAME, "N14ZP74G"), + }, + .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH), + }, {} }; -- 2.39.2

1 0

[PATCH AUTOSEL 6.3 09/30] soundwire: qcom: add proper error paths in qcom_swrm_startup()
by Sasha Levin 16 Jun '23

16 Jun '23

From: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> [ Upstream commit 99e09b9c0ab43346c52f2787ca4e5c4b1798362e ] Reverse actions in qcom_swrm_startup() error paths to avoid leaking stream memory and keeping runtime PM unbalanced. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20230517163736.997553-1-krzysztof.kozlowski@linar… Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/qcom.c | 17 +++++++++++++---- 1 file changed, 13 insertions(+), 4 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 30575ed20947e..0dcdbd4e1ec3a 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -1098,8 +1098,10 @@ static int qcom_swrm_startup(struct snd_pcm_substream *substream, } sruntime = sdw_alloc_stream(dai->name); - if (!sruntime) - return -ENOMEM; + if (!sruntime) { + ret = -ENOMEM; + goto err_alloc; + } ctrl->sruntime[dai->id] = sruntime; @@ -1109,12 +1111,19 @@ static int qcom_swrm_startup(struct snd_pcm_substream *substream, if (ret < 0 && ret != -ENOTSUPP) { dev_err(dai->dev, "Failed to set sdw stream on %s\n", codec_dai->name); - sdw_release_stream(sruntime); - return ret; + goto err_set_stream; } } return 0; + +err_set_stream: + sdw_release_stream(sruntime); +err_alloc: + pm_runtime_mark_last_busy(ctrl->dev); + pm_runtime_put_autosuspend(ctrl->dev); + + return ret; } static void qcom_swrm_shutdown(struct snd_pcm_substream *substream, -- 2.39.2

1 0

[PATCH AUTOSEL 6.3 08/30] soundwire: dmi-quirks: add new mapping for HP Spectre x360
by Sasha Levin 16 Jun '23

16 Jun '23

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 700581ede41d029403feec935df4616309696fd7 ] A BIOS/DMI update seems to have broken some devices, let's add a new mapping. Link: https://github.com/thesofproject/linux/issues/4323 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://lore.kernel.org/r/20230515074859.3097-1-yung-chuan.liao@linux.intel… Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/dmi-quirks.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/drivers/soundwire/dmi-quirks.c b/drivers/soundwire/dmi-quirks.c index 58ea013fa918a..2a1096dab63d3 100644 --- a/drivers/soundwire/dmi-quirks.c +++ b/drivers/soundwire/dmi-quirks.c @@ -99,6 +99,13 @@ static const struct dmi_system_id adr_remap_quirk_table[] = { }, .driver_data = (void *)intel_tgl_bios, }, + { + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "HP"), + DMI_MATCH(DMI_BOARD_NAME, "8709"), + }, + .driver_data = (void *)intel_tgl_bios, + }, { /* quirk used for NUC15 'Bishop County' LAPBC510 and LAPBC710 skews */ .matches = { -- 2.39.2

1 0

[PATCH AUTOSEL 6.3 07/30] ASoC: simple-card: Add missing of_node_put() in case of error
by Sasha Levin 16 Jun '23

16 Jun '23

From: Herve Codina <herve.codina(a)bootlin.com> [ Upstream commit 8938f75a5e35c597a647c28984a0304da7a33d63 ] In the error path, a of_node_put() for platform is missing. Just add it. Signed-off-by: Herve Codina <herve.codina(a)bootlin.com> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Link: https://lore.kernel.org/r/20230523151223.109551-9-herve.codina@bootlin.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/generic/simple-card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index e98932c167542..5f8468ff36562 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -416,6 +416,7 @@ static int __simple_for_each_link(struct asoc_simple_priv *priv, if (ret < 0) { of_node_put(codec); + of_node_put(plat); of_node_put(np); goto error; } -- 2.39.2

1 0

[PATCH AUTOSEL 6.3 06/30] ASoC: codecs: wcd938x-sdw: do not set can_multi_write flag
by Sasha Levin 16 Jun '23

16 Jun '23

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> [ Upstream commit 2d7c2f9272de6347a9cec0fc07708913692c0ae3 ] regmap-sdw does not support multi register writes, so there is no point in setting this flag. This also leads to incorrect programming of WSA codecs with regmap_multi_reg_write() call. This invalid configuration should have been rejected by regmap-sdw. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Link: https://lore.kernel.org/r/20230523165414.14560-1-srinivas.kandagatla@linaro… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wcd938x-sdw.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wcd938x-sdw.c b/sound/soc/codecs/wcd938x-sdw.c index 402286dfaea44..9c10200ff34b2 100644 --- a/sound/soc/codecs/wcd938x-sdw.c +++ b/sound/soc/codecs/wcd938x-sdw.c @@ -1190,7 +1190,6 @@ static const struct regmap_config wcd938x_regmap_config = { .readable_reg = wcd938x_readable_register, .writeable_reg = wcd938x_writeable_register, .volatile_reg = wcd938x_volatile_register, - .can_multi_write = true, }; static const struct sdw_slave_ops wcd9380_slave_ops = { -- 2.39.2

1 0

[PATCH] kselftest/alsa: Increase kselftest timeout
by Nícolas F. R. A. Prado 16 Jun '23

16 Jun '23

The default timeout for kselftests is 45 seconds, but pcm-test can take longer than that to run depending on the number of PCMs present on a device. As a data point, running pcm-test on mt8192-asurada-spherion takes about 1m15s. Set the timeout to 10 minutes, which should give enough slack to run the test even on devices with many PCMs. Signed-off-by: Nícolas F. R. A. Prado <nfraprado(a)collabora.com> --- tools/testing/selftests/alsa/settings | 1 + 1 file changed, 1 insertion(+) create mode 100644 tools/testing/selftests/alsa/settings diff --git a/tools/testing/selftests/alsa/settings b/tools/testing/selftests/alsa/settings new file mode 100644 index 000000000000..a62d2fa1275c --- /dev/null +++ b/tools/testing/selftests/alsa/settings @@ -0,0 +1 @@ +timeout=600 -- 2.39.0

4 5

[GIT PULL] ASoC fixes for v6.4-rc6-2
by Mark Brown 15 Jun '23

15 Jun '23

The following changes since commit 32cf0046a652116d6a216d575f3049a9ff9dd80d: ASoC: fsl_sai: Enable BCI bit if SAI works on synchronous mode with BYP asserted (2023-06-02 12:54:07 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.4-rc6-2 for you to fetch changes up to f9fd804aa0a36f15a35ca070ec4c52650876cc29: ASoC: tegra: Fix Master Volume Control (2023-06-13 12:10:56 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.4 A couple more fixes for v6.4, one fixing a misleading error log and another stopping us seeing spurious failures setting the master volume on some Tegra systems introduced by a change to how we calculate delay times. ---------------------------------------------------------------- Jon Hunter (1): ASoC: tegra: Fix Master Volume Control Richard Fitzgerald (1): firmware: cs_dsp: Log correct region name in bin error messages drivers/firmware/cirrus/cs_dsp.c | 5 +++-- sound/soc/tegra/tegra_pcm.c | 3 +++ 2 files changed, 6 insertions(+), 2 deletions(-)

2 1

[PATCH] ASoC: soc-compress: Fix deadlock in soc_compr_open_fe
by yixuanjiang 15 Jun '23

15 Jun '23

Modify the error handling flow by release lock. The require pcm_mutex will keep holding if open fail. Fixes: aa9ff6a4955f ("ASoC: soc-compress: Reposition and add pcm_mutex") Signed-off-by: yixuanjiang <yixuanjiang(a)google.com> Cc: stable(a)vger.kernel.org # v5.15+ --- sound/soc/soc-compress.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 256e45001f85..b6727b91dd85 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -166,6 +166,7 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) snd_soc_dai_compr_shutdown(cpu_dai, cstream, 1); out: dpcm_path_put(&list); + mutex_unlock(&fe->card->pcm_mutex); be_err: fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; mutex_unlock(&fe->card->mutex); -- 2.41.0.162.gfafddb0af9-goog

3 2

[PATCH V3 1/2] ASoC: dt-bindings: max98388: add amplifier driver
by “Ryan 15 Jun '23

15 Jun '23

From: Ryan Lee <ryans.lee(a)analog.com> Add dt-bindings information for Analog Devices MAX98388 I2S Amplifier Signed-off-by: Ryan Lee <ryans.lee(a)analog.com> Reviewed-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Changes from v1: Removed unnecessary blank line and description. Modified quotes. Changes from v2: No change. .../bindings/sound/adi,max98388.yaml | 79 +++++++++++++++++++ 1 file changed, 79 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/adi,max98388.yaml diff --git a/Documentation/devicetree/bindings/sound/adi,max98388.yaml b/Documentation/devicetree/bindings/sound/adi,max98388.yaml new file mode 100644 index 000000000000..93ccd5905736 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,max98388.yaml @@ -0,0 +1,79 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/adi,max98388.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Analog Devices MAX98388 Speaker Amplifier + +maintainers: + - Ryan Lee <ryans.lee(a)analog.com> + +description: + The MAX98388 is a mono Class-D speaker amplifier with I/V feedback. + The device provides a PCM interface for audio data and a standard + I2C interface for control data communication. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - adi,max98388 + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + adi,vmon-slot-no: + description: slot number of the voltage feedback monitor + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 15 + default: 0 + + adi,imon-slot-no: + description: slot number of the current feedback monitor + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 15 + default: 1 + + adi,interleave-mode: + description: + For cases where a single combined channel for the I/V feedback data + is not sufficient, the device can also be configured to share + a single data output channel on alternating frames. + In this configuration, the current and voltage data will be frame + interleaved on a single output channel. + type: boolean + + reset-gpios: + maxItems: 1 + +required: + - compatible + - reg + - '#sound-dai-cells' + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + max98388: amplifier@39 { + compatible = "adi,max98388"; + reg = <0x39>; + #sound-dai-cells = <0>; + adi,vmon-slot-no = <0>; + adi,imon-slot-no = <1>; + adi,interleave-mode; + reset-gpios = <&gpio 4 GPIO_ACTIVE_LOW>; + }; + }; base-commit: 53ab6975c12d1ad86c599a8927e8c698b144d669 -- 2.34.1

2 2

[PATCH v3 3/3] ASoC: dt-bindings: Add support for Loongson audio card
by YingKun Meng 15 Jun '23

15 Jun '23

From: Yingkun Meng <mengyingkun(a)loongson.cn> The audio card uses loongson I2S controller present in 7axxx/2kxxx chips to transfer audio data. On loongson platform, the chip has only one I2S controller. Signed-off-by: Yingkun Meng <mengyingkun(a)loongson.cn> --- Changes v2 -> v3 No change. Changes v1 -> v2 Add chip model restriction for title and descrition. Add 'required' option for sound-dai property. .../sound/loongson,ls-audio-card.yaml | 70 +++++++++++++++++++ 1 file changed, 70 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/loongson,ls-audio-card.yaml diff --git a/Documentation/devicetree/bindings/sound/loongson,ls-audio-card.yaml b/Documentation/devicetree/bindings/sound/loongson,ls-audio-card.yaml new file mode 100644 index 000000000000..61e8babed402 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/loongson,ls-audio-card.yaml @@ -0,0 +1,70 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/loongson,ls-audio-card.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Loongson 7axxx/2kxxx ASoC audio sound card driver + +maintainers: + - Yingkun Meng <mengyingkun(a)loongson.cn> + +description: + The binding describes the sound card present in loongson + 7axxx/2kxxx platform. The sound card is an ASoC component + which uses Loongson I2S controller to transfer the audio data. + +properties: + compatible: + const: loongson,ls-audio-card + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User specified audio sound card name + + mclk-fs: + $ref: simple-card.yaml#/definitions/mclk-fs + + cpu: + description: Holds subnode which indicates cpu dai. + type: object + additionalProperties: false + properties: + sound-dai: + maxItems: 1 + required: + - sound-dai + + codec: + description: Holds subnode which indicates codec dai. + type: object + additionalProperties: false + properties: + sound-dai: + maxItems: 1 + required: + - sound-dai + +required: + - compatible + - model + - mclk-fs + - cpu + - codec + +additionalProperties: false + +examples: + - | + sound { + compatible = "loongson,ls-audio-card"; + model = "loongson-audio"; + mclk-fs = <512>; + + cpu { + sound-dai = <&i2s>; + }; + codec { + sound-dai = <&es8323>; + }; + }; -- 2.33.0

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[PATCH v3 2/3] ASoC: loongson: Add Loongson ASoC Sound Card Support
by YingKun Meng 15 Jun '23

15 Jun '23

From: Yingkun Meng <mengyingkun(a)loongson.cn> The Loongson ASoC Sound Card is a general ASoC DAI Link driver that can be used for Loongson CPU DAI drivers and external CODECs. The driver supports the use of ACPI table to describe device resources. On loongson 7axxx platforms, the audio device is an ACPI device. Signed-off-by: Yingkun Meng <mengyingkun(a)loongson.cn> --- sound/soc/loongson/Kconfig | 10 ++ sound/soc/loongson/Makefile | 4 + sound/soc/loongson/loongson_card.c | 230 +++++++++++++++++++++++++++++ 3 files changed, 244 insertions(+) create mode 100644 sound/soc/loongson/loongson_card.c diff --git a/sound/soc/loongson/Kconfig b/sound/soc/loongson/Kconfig index 4478ac91e402..c175f9de19a8 100644 --- a/sound/soc/loongson/Kconfig +++ b/sound/soc/loongson/Kconfig @@ -13,4 +13,14 @@ config SND_SOC_LOONGSON_I2S_PCI The controller is found in loongson bridge chips or SoCs, and work as a PCI device. +config SND_SOC_LOONGSON_CARD + tristate "Loongson Sound Card Driver" + select SND_SOC_LOONGSON_I2S_PCI + help + Say Y or M if you want to add support for SoC audio using + loongson I2S controller. + + The driver add support for ALSA SoC Audio support using + loongson I2S controller. + endmenu diff --git a/sound/soc/loongson/Makefile b/sound/soc/loongson/Makefile index 099af7989103..601a905a4860 100644 --- a/sound/soc/loongson/Makefile +++ b/sound/soc/loongson/Makefile @@ -2,3 +2,7 @@ #Platform Support snd-soc-loongson-i2s-pci-objs := loongson_i2s_pci.o loongson_i2s.o loongson_dma.o obj-$(CONFIG_SND_SOC_LOONGSON_I2S_PCI) += snd-soc-loongson-i2s-pci.o + +#Machine Support +snd-soc-loongson-card-objs := loongson_card.o +obj-$(CONFIG_SND_SOC_LOONGSON_CARD) += snd-soc-loongson-card.o diff --git a/sound/soc/loongson/loongson_card.c b/sound/soc/loongson/loongson_card.c new file mode 100644 index 000000000000..965eaf4e9109 --- /dev/null +++ b/sound/soc/loongson/loongson_card.c @@ -0,0 +1,230 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Loongson ASoC Audio Machine driver +// +// Copyright (C) 2023 Loongson Technology Corporation Limited +// Author: Yingkun Meng <mengyingkun(a)loongson.cn> +// + +#include <linux/module.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> +#include <linux/acpi.h> +#include <linux/pci.h> +#include <sound/pcm_params.h> + +static char codec_name[SND_ACPI_I2C_ID_LEN]; + +struct loongson_card_data { + struct snd_soc_card snd_card; + unsigned int mclk_fs; +}; + +static int loongson_card_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct loongson_card_data *ls_card = snd_soc_card_get_drvdata(rtd->card); + int ret, mclk; + + if (ls_card->mclk_fs) { + mclk = ls_card->mclk_fs * params_rate(params); + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, + SND_SOC_CLOCK_OUT); + if (ret < 0) { + dev_err(codec_dai->dev, "cpu_dai clock not set\n"); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "codec_dai clock not set\n"); + return ret; + } + } + return 0; +} + +static const struct snd_soc_ops loongson_ops = { + .hw_params = loongson_card_hw_params, +}; + +SND_SOC_DAILINK_DEFS(analog, + DAILINK_COMP_ARRAY(COMP_CPU("loongson-i2s")), + DAILINK_COMP_ARRAY(COMP_EMPTY()), + DAILINK_COMP_ARRAY(COMP_EMPTY())); + +static struct snd_soc_dai_link loongson_dai_links[] = { + { + .name = "Loongson Audio Port", + .stream_name = "Loongson Audio", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_IB_NF + | SND_SOC_DAIFMT_CBC_CFC, + SND_SOC_DAILINK_REG(analog), + .ops = &loongson_ops, + }, +}; + +static int loongson_card_parse_acpi(struct loongson_card_data *data) +{ + struct snd_soc_card *card = &data->snd_card; + struct fwnode_handle *fwnode = card->dev->fwnode; + struct fwnode_reference_args args; + const char *codec_dai_name; + struct acpi_device *adev; + struct device *phy_dev; + int ret, i; + + /* fixup platform name based on reference node */ + memset(&args, 0, sizeof(args)); + ret = acpi_node_get_property_reference(fwnode, "cpu", 0, &args); + if (ACPI_FAILURE(ret) || !is_acpi_device_node(args.fwnode)) { + dev_err(card->dev, "No matching phy in ACPI table\n"); + return ret; + } + adev = to_acpi_device_node(args.fwnode); + phy_dev = acpi_get_first_physical_node(adev); + if (!phy_dev) + return -EPROBE_DEFER; + for (i = 0; i < card->num_links; i++) + loongson_dai_links[i].platforms->name = dev_name(phy_dev); + + /* fixup codec name based on reference node */ + memset(&args, 0, sizeof(args)); + ret = acpi_node_get_property_reference(fwnode, "codec", 0, &args); + if (ACPI_FAILURE(ret) || !is_acpi_device_node(args.fwnode)) { + dev_err(card->dev, "No matching phy in ACPI table\n"); + return ret; + } + adev = to_acpi_device_node(args.fwnode); + snprintf(codec_name, sizeof(codec_name), "i2c-%s", acpi_dev_name(adev)); + for (i = 0; i < card->num_links; i++) + loongson_dai_links[i].codecs->name = codec_name; + + device_property_read_string(card->dev, "codec-dai-name", + &codec_dai_name); + for (i = 0; i < card->num_links; i++) + loongson_dai_links[i].codecs->dai_name = codec_dai_name; + + return 0; +} + +static int loongson_card_parse_of(struct loongson_card_data *data) +{ + const char *cpu_dai_name, *codec_dai_name; + struct device_node *cpu, *codec; + struct snd_soc_card *card = &data->snd_card; + struct device *dev = card->dev; + struct of_phandle_args args; + int ret, i; + + cpu = of_get_child_by_name(dev->of_node, "cpu"); + if (!cpu) { + dev_err(dev, "platform property missing or invalid\n"); + return -EINVAL; + } + codec = of_get_child_by_name(dev->of_node, "codec"); + if (!codec) { + dev_err(dev, "audio-codec property missing or invalid\n"); + ret = -EINVAL; + goto err; + } + + ret = of_parse_phandle_with_args(cpu, "sound-dai", + "#sound-dai-cells", 0, &args); + if (ret) { + dev_err(dev, "codec node missing #sound-dai-cells\n"); + goto err; + } + for (i = 0; i < card->num_links; i++) + loongson_dai_links[i].cpus->of_node = args.np; + + ret = of_parse_phandle_with_args(codec, "sound-dai", + "#sound-dai-cells", 0, &args); + if (ret) { + dev_err(dev, "codec node missing #sound-dai-cells\n"); + goto err; + } + for (i = 0; i < card->num_links; i++) + loongson_dai_links[i].codecs->of_node = args.np; + + snd_soc_of_get_dai_name(cpu, &cpu_dai_name); + snd_soc_of_get_dai_name(codec, &codec_dai_name); + for (i = 0; i < card->num_links; i++) { + loongson_dai_links[i].cpus->dai_name = cpu_dai_name; + loongson_dai_links[i].codecs->dai_name = codec_dai_name; + } + of_node_put(cpu); + of_node_put(codec); + + return 0; + +err: + of_node_put(cpu); + of_node_put(codec); + return ret; +} + +static int loongson_asoc_card_probe(struct platform_device *pdev) +{ + struct loongson_card_data *ls_priv; + struct snd_soc_card *card; + int ret; + + ls_priv = devm_kzalloc(&pdev->dev, sizeof(*ls_priv), GFP_KERNEL); + if (!ls_priv) + return -ENOMEM; + + card = &ls_priv->snd_card; + + card->dev = &pdev->dev; + card->owner = THIS_MODULE; + card->dai_link = loongson_dai_links; + card->num_links = ARRAY_SIZE(loongson_dai_links); + snd_soc_card_set_drvdata(card, ls_priv); + + ret = device_property_read_string(&pdev->dev, "model", &card->name); + if (ret) { + dev_err(&pdev->dev, "Error parsing card name: %d\n", ret); + return ret; + } + ret = device_property_read_u32(&pdev->dev, "mclk-fs", &ls_priv->mclk_fs); + if (ret) { + dev_err(&pdev->dev, "Error parsing mclk-fs: %d\n", ret); + return ret; + } + + if (has_acpi_companion(&pdev->dev)) + ret = loongson_card_parse_acpi(ls_priv); + else + ret = loongson_card_parse_of(ls_priv); + if (ret < 0) + return ret; + + ret = devm_snd_soc_register_card(&pdev->dev, card); + + return ret; +} + +static const struct of_device_id loongson_asoc_dt_ids[] = { + { .compatible = "loongson,ls-audio-card" }, + { /* sentinel */ }, +}; +MODULE_DEVICE_TABLE(of, loongson_asoc_dt_ids); + +static struct platform_driver loongson_audio_driver = { + .probe = loongson_asoc_card_probe, + .driver = { + .name = "loongson-asoc-card", + .pm = &snd_soc_pm_ops, + .of_match_table = of_match_ptr(loongson_asoc_dt_ids), + }, +}; +module_platform_driver(loongson_audio_driver); + +MODULE_DESCRIPTION("Loongson ASoc Sound Card driver"); +MODULE_AUTHOR("Loongson Technology Corporation Limited"); +MODULE_LICENSE("GPL"); -- 2.33.0

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[PATCH] ASoC: dt-bindings: audio-graph-card: Expand 'widgets' documentation
by Marek Vasut 15 Jun '23

15 Jun '23

Document the encoding of 'widgets' property to avoid confusion. Signed-off-by: Marek Vasut <marex(a)denx.de> --- Cc: Krzysztof Kozlowski <krzysztof.kozlowski+dt(a)linaro.org> Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Rob Herring <robh+dt(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Cc: devicetree(a)vger.kernel.org --- Documentation/devicetree/bindings/sound/audio-graph.yaml | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/audio-graph.yaml b/Documentation/devicetree/bindings/sound/audio-graph.yaml index c87eb91de1593..ed31e04ff6a68 100644 --- a/Documentation/devicetree/bindings/sound/audio-graph.yaml +++ b/Documentation/devicetree/bindings/sound/audio-graph.yaml @@ -24,7 +24,11 @@ properties: connection's sink, the second being the connection's source. $ref: /schemas/types.yaml#/definitions/non-unique-string-array widgets: - description: User specified audio sound widgets. + description: | + User specified audio sound widgets. + Each entry is a pair of strings, the first being the type of + widget ("Microphone", "Line", "Headphone", "Speaker"), the + second being the machine specific name for the widget. $ref: /schemas/types.yaml#/definitions/non-unique-string-array convert-rate: $ref: /schemas/sound/dai-params.yaml#/$defs/dai-sample-rate -- 2.39.2

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[PATCH] ASoC: simple_card_utils: remove unused cpus/codecs/platforms from props
by Kuninori Morimoto 15 Jun '23

15 Jun '23

simple_dai_props has cpus/codecs/platforms. These pointer were used for dai_link before, but are allocated today since commit 050c7950fd70 ("ASoC: simple-card-utils: alloc dai_link information for CPU/Codec/Platform"). We don't need to keep it anymore. This patch removes these. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/simple_card_utils.h | 3 --- sound/soc/generic/simple-card-utils.c | 6 ------ 2 files changed, 9 deletions(-) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 9daef37fe9a8..b450d5873227 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -59,9 +59,6 @@ struct asoc_simple_priv { struct simple_dai_props { struct asoc_simple_dai *cpu_dai; struct asoc_simple_dai *codec_dai; - struct snd_soc_dai_link_component *cpus; - struct snd_soc_dai_link_component *codecs; - struct snd_soc_dai_link_component *platforms; struct asoc_simple_data adata; struct snd_soc_codec_conf *codec_conf; struct prop_nums num; diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 6a3c9e4e1cfe..f94c48aa126c 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -903,7 +903,6 @@ int asoc_simple_init_priv(struct asoc_simple_priv *priv, for (i = 0; i < li->link; i++) { if (li->num[i].cpus) { /* Normal CPU */ - dai_props[i].cpus = dai_link[i].cpus = dlcs; dai_props[i].num.cpus = dai_link[i].num_cpus = li->num[i].cpus; @@ -913,7 +912,6 @@ int asoc_simple_init_priv(struct asoc_simple_priv *priv, dais += li->num[i].cpus; } else { /* DPCM Be's CPU = dummy */ - dai_props[i].cpus = dai_link[i].cpus = &asoc_dummy_dlc; dai_props[i].num.cpus = dai_link[i].num_cpus = 1; @@ -921,7 +919,6 @@ int asoc_simple_init_priv(struct asoc_simple_priv *priv, if (li->num[i].codecs) { /* Normal Codec */ - dai_props[i].codecs = dai_link[i].codecs = dlcs; dai_props[i].num.codecs = dai_link[i].num_codecs = li->num[i].codecs; @@ -937,7 +934,6 @@ int asoc_simple_init_priv(struct asoc_simple_priv *priv, } } else { /* DPCM Fe's Codec = dummy */ - dai_props[i].codecs = dai_link[i].codecs = &asoc_dummy_dlc; dai_props[i].num.codecs = dai_link[i].num_codecs = 1; @@ -945,7 +941,6 @@ int asoc_simple_init_priv(struct asoc_simple_priv *priv, if (li->num[i].platforms) { /* Have Platform */ - dai_props[i].platforms = dai_link[i].platforms = dlcs; dai_props[i].num.platforms = dai_link[i].num_platforms = li->num[i].platforms; @@ -953,7 +948,6 @@ int asoc_simple_init_priv(struct asoc_simple_priv *priv, dlcs += li->num[i].platforms; } else { /* Doesn't have Platform */ - dai_props[i].platforms = dai_link[i].platforms = NULL; dai_props[i].num.platforms = dai_link[i].num_platforms = 0; -- 2.25.1

2 1

[PATCH 00/16] ASoC: Convert Realtek I2C drivers to use maple tree register cache
by Mark Brown 15 Jun '23

15 Jun '23

Many of the Realtek I2C/SPI devices only support single register read and write operations so don't benefit from using the rbtree cache instead of the more modern maple tree cache, convert them to maple tree. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Mark Brown (16): ASoC: rt1011: Use maple tree register cache ASoC: rt1019: Use maple tree register cache ASoC: rt1305: Use maple tree register cache ASoC: rt1308: Use maple tree register cache ASoC: rt5514: Use maple tree register cache ASoC: rt5616: Use maple tree register cache ASoC: rt5631: Use maple tree register cache ASoC: rt5640: Use maple tree register cache ASoC: rt5645: Use maple tree register cache ASoC: rt5651: Use maple tree register cache ASoC: rt5660: Use maple tree register cache ASoC: rt5665: Use maple tree register cache ASoC: rt5665: Use maple tree register cache ASoC: rt5668: Use maple tree register cache ASoC: rt5670: Use maple tree register cache ASoC: rt5682: Use maple tree register cache sound/soc/codecs/rt1011.c | 2 +- sound/soc/codecs/rt1019.c | 2 +- sound/soc/codecs/rt1305.c | 2 +- sound/soc/codecs/rt1308.c | 2 +- sound/soc/codecs/rt5514.c | 2 +- sound/soc/codecs/rt5616.c | 2 +- sound/soc/codecs/rt5631.c | 2 +- sound/soc/codecs/rt5640.c | 2 +- sound/soc/codecs/rt5645.c | 4 ++-- sound/soc/codecs/rt5651.c | 2 +- sound/soc/codecs/rt5660.c | 2 +- sound/soc/codecs/rt5663.c | 4 ++-- sound/soc/codecs/rt5665.c | 2 +- sound/soc/codecs/rt5668.c | 2 +- sound/soc/codecs/rt5670.c | 2 +- sound/soc/codecs/rt5682-i2c.c | 2 +- 16 files changed, 18 insertions(+), 18 deletions(-) --- base-commit: 9561de3a55bed6bdd44a12820ba81ec416e705a7 change-id: 20230609-asoc-rt-maple-8a75e2da9555 Best regards, -- Mark Brown <broonie(a)kernel.org>

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[PATCH v3 1/3] ASoC: Add support for Loongson I2S controller
by YingKun Meng 15 Jun '23

15 Jun '23

From: Yingkun Meng <mengyingkun(a)loongson.cn> Loongson I2S controller is found on 7axxx/2kxxx chips from loongson, it is a PCI device with two private DMA controllers, one for playback, the other for capture. The driver supports the use of DTS or ACPI to describe device resources. Signed-off-by: Yingkun Meng <mengyingkun(a)loongson.cn> --- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/loongson/Kconfig | 16 ++ sound/soc/loongson/Makefile | 4 + sound/soc/loongson/loongson_dma.c | 350 ++++++++++++++++++++++++++ sound/soc/loongson/loongson_dma.h | 16 ++ sound/soc/loongson/loongson_i2s.c | 269 ++++++++++++++++++++ sound/soc/loongson/loongson_i2s.h | 71 ++++++ sound/soc/loongson/loongson_i2s_pci.c | 171 +++++++++++++ 9 files changed, 899 insertions(+) create mode 100644 sound/soc/loongson/Kconfig create mode 100644 sound/soc/loongson/Makefile create mode 100644 sound/soc/loongson/loongson_dma.c create mode 100644 sound/soc/loongson/loongson_dma.h create mode 100644 sound/soc/loongson/loongson_i2s.c create mode 100644 sound/soc/loongson/loongson_i2s.h create mode 100644 sound/soc/loongson/loongson_i2s_pci.c diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 4b6e5a802880..bfa9622e1ab1 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -79,6 +79,7 @@ source "sound/soc/google/Kconfig" source "sound/soc/hisilicon/Kconfig" source "sound/soc/jz4740/Kconfig" source "sound/soc/kirkwood/Kconfig" +source "sound/soc/loongson/Kconfig" source "sound/soc/img/Kconfig" source "sound/soc/intel/Kconfig" source "sound/soc/mediatek/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 9d9b228e4508..8376fdb217ed 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -46,6 +46,7 @@ obj-$(CONFIG_SND_SOC) += fsl/ obj-$(CONFIG_SND_SOC) += google/ obj-$(CONFIG_SND_SOC) += hisilicon/ obj-$(CONFIG_SND_SOC) += jz4740/ +obj-$(CONFIG_SND_SOC) += loongson/ obj-$(CONFIG_SND_SOC) += img/ obj-$(CONFIG_SND_SOC) += intel/ obj-$(CONFIG_SND_SOC) += mediatek/ diff --git a/sound/soc/loongson/Kconfig b/sound/soc/loongson/Kconfig new file mode 100644 index 000000000000..4478ac91e402 --- /dev/null +++ b/sound/soc/loongson/Kconfig @@ -0,0 +1,16 @@ +# SPDX-License-Identifier: GPL-2.0 +menu "SoC Audio for Loongson CPUs" + depends on LOONGARCH || COMPILE_TEST + +config SND_SOC_LOONGSON_I2S_PCI + tristate "Loongson I2S-PCI Device Driver" + select REGMAP_MMIO + depends on PCI + help + Say Y or M if you want to add support for I2S driver for + Loongson I2S controller. + + The controller is found in loongson bridge chips or SoCs, + and work as a PCI device. + +endmenu diff --git a/sound/soc/loongson/Makefile b/sound/soc/loongson/Makefile new file mode 100644 index 000000000000..099af7989103 --- /dev/null +++ b/sound/soc/loongson/Makefile @@ -0,0 +1,4 @@ +# SPDX-License-Identifier: GPL-2.0 +#Platform Support +snd-soc-loongson-i2s-pci-objs := loongson_i2s_pci.o loongson_i2s.o loongson_dma.o +obj-$(CONFIG_SND_SOC_LOONGSON_I2S_PCI) += snd-soc-loongson-i2s-pci.o diff --git a/sound/soc/loongson/loongson_dma.c b/sound/soc/loongson/loongson_dma.c new file mode 100644 index 000000000000..ff494df2aa0f --- /dev/null +++ b/sound/soc/loongson/loongson_dma.c @@ -0,0 +1,350 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Loongson ALSA SoC Platform (DMA) driver +// +// Copyright (C) 2023 Loongson Technology Corporation Limited +// Author: Yingkun Meng <mengyingkun(a)loongson.cn> +// + +#include <linux/module.h> +#include <linux/io-64-nonatomic-lo-hi.h> +#include <linux/delay.h> +#include <linux/pm_runtime.h> +#include <linux/dma-mapping.h> +#include <sound/soc.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include "loongson_i2s.h" + +/* DMA dma_order Register */ +#define DMA_ORDER_STOP (1 << 4) /* DMA stop */ +#define DMA_ORDER_START (1 << 3) /* DMA start */ +#define DMA_ORDER_ASK_VALID (1 << 2) /* DMA ask valid flag */ +#define DMA_ORDER_AXI_UNCO (1 << 1) /* Uncache access */ +#define DMA_ORDER_ADDR_64 (1 << 0) /* 64bits address support */ + +#define DMA_ORDER_ASK_MASK (~0x1fUL) /* Ask addr mask */ +#define DMA_ORDER_CTRL_MASK (0x0fUL) /* Control mask */ + +/* + * DMA registers descriptor. + */ +struct loongson_dma_desc { + u32 order; /* Next descriptor address register */ + u32 saddr; /* Source address register */ + u32 daddr; /* Device address register */ + u32 length; /* Total length register */ + u32 step_length; /* Memory stride register */ + u32 step_times; /* Repeat time register */ + u32 cmd; /* Command register */ + u32 stats; /* Status register */ + u32 order_hi; /* Next descriptor high address register */ + u32 saddr_hi; /* High source address register */ + u32 res[6]; /* Reserved */ +}; + +struct loongson_runtime_data { + struct loongson_dma_data *dma_data; + + struct loongson_dma_desc *dma_desc_arr; + dma_addr_t dma_desc_arr_phy; + int dma_desc_arr_size; + + struct loongson_dma_desc *dma_pos_desc; + dma_addr_t dma_pos_desc_phy; +}; + +static const struct snd_pcm_hardware ls_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_PAUSE, + .formats = (SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | + SNDRV_PCM_FMTBIT_S24_LE), + .period_bytes_min = 128, + .period_bytes_max = 128 * 1024, + .periods_min = 1, + .periods_max = PAGE_SIZE / sizeof(struct loongson_dma_desc), + .buffer_bytes_max = 1024 * 1024, +}; + +static struct +loongson_dma_desc *dma_desc_save(struct loongson_runtime_data *prtd) +{ + void __iomem *order_reg = prtd->dma_data->order_addr; + u64 val; + + val = (u64)prtd->dma_pos_desc_phy & DMA_ORDER_ASK_MASK; + val |= (readq(order_reg) & DMA_ORDER_CTRL_MASK); + val |= DMA_ORDER_ASK_VALID; + writeq(val, order_reg); + + while (readl(order_reg) & DMA_ORDER_ASK_VALID) + udelay(2); + + return prtd->dma_pos_desc; +} + +static int loongson_pcm_trigger(struct snd_soc_component *component, + struct snd_pcm_substream *substream, int cmd) +{ + struct loongson_runtime_data *prtd = substream->runtime->private_data; + struct device *dev = substream->pcm->card->dev; + void __iomem *order_reg = prtd->dma_data->order_addr; + u64 val; + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + val = prtd->dma_pos_desc_phy & DMA_ORDER_ASK_MASK; + if (dev->coherent_dma_mask == DMA_BIT_MASK(64)) + val |= DMA_ORDER_ADDR_64; + else + val &= ~DMA_ORDER_ADDR_64; + val |= (readq(order_reg) & DMA_ORDER_CTRL_MASK); + val |= DMA_ORDER_START; + writeq(val, order_reg); + + while ((readl(order_reg) & DMA_ORDER_START)) + udelay(2); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + dma_desc_save(prtd); + + /* dma stop */ + val = readq(order_reg) | DMA_ORDER_STOP; + writeq(val, order_reg); + udelay(1000); + + break; + default: + dev_err(dev, "Invalid pcm trigger operation\n"); + return -EINVAL; + } + + return ret; +} + +static int loongson_pcm_hw_params(struct snd_soc_component *component, + struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct device *dev = substream->pcm->card->dev; + struct loongson_runtime_data *prtd = runtime->private_data; + size_t buf_len = params_buffer_bytes(params); + size_t period_len = params_period_bytes(params); + dma_addr_t order_addr, mem_addr; + struct loongson_dma_desc *desc; + u32 num_periods; + int i; + + if (buf_len % period_len) { + dev_err(dev, "buf len not multiply of period len\n"); + return -EINVAL; + } + + num_periods = buf_len / period_len; + if (!num_periods || num_periods > prtd->dma_desc_arr_size) { + dev_err(dev, "dma data too small or too big\n"); + return -EINVAL; + } + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = buf_len; + + /* initialize dma descriptor array */ + mem_addr = runtime->dma_addr; + order_addr = prtd->dma_desc_arr_phy; + for (i = 0; i < num_periods; i++) { + desc = &prtd->dma_desc_arr[i]; + + /* next descriptor physical address */ + order_addr += sizeof(*desc); + desc->order = lower_32_bits(order_addr | BIT(0)); + desc->order_hi = upper_32_bits(order_addr); + + desc->saddr = lower_32_bits(mem_addr); + desc->saddr_hi = upper_32_bits(mem_addr); + desc->daddr = prtd->dma_data->dev_addr; + + desc->cmd = BIT(0); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + desc->cmd |= BIT(12); + + desc->length = period_len >> 2; + desc->step_length = 0; + desc->step_times = 1; + + mem_addr += period_len; + } + desc = &prtd->dma_desc_arr[num_periods - 1]; + desc->order = lower_32_bits(prtd->dma_desc_arr_phy | BIT(0)); + desc->order_hi = upper_32_bits(prtd->dma_desc_arr_phy); + + /* init position descriptor */ + *prtd->dma_pos_desc = *prtd->dma_desc_arr; + + return 0; +} + +static snd_pcm_uframes_t +loongson_pcm_pointer(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct loongson_runtime_data *prtd = runtime->private_data; + struct loongson_dma_desc *desc; + snd_pcm_uframes_t x; + u64 addr; + + desc = dma_desc_save(prtd); + addr = ((u64)desc->saddr_hi << 32) | desc->saddr; + + x = bytes_to_frames(runtime, addr - runtime->dma_addr); + if (x == runtime->buffer_size) + x = 0; + return x; +} + +static irqreturn_t loongson_pcm_dma_irq(int irq, void *devid) +{ + struct snd_pcm_substream *substream = devid; + + snd_pcm_period_elapsed(substream); + return IRQ_HANDLED; +} + +static int loongson_pcm_open(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_card *card = substream->pcm->card; + struct loongson_runtime_data *prtd; + struct loongson_dma_data *dma_data; + int ret; + + /* + * For mysterious reasons (and despite what the manual says) + * playback samples are lost if the DMA count is not a multiple + * of the DMA burst size. Let's add a rule to enforce that. + */ + snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 128); + snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 128); + snd_pcm_hw_constraint_integer(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + snd_soc_set_runtime_hwparams(substream, &ls_pcm_hardware); + + prtd = kzalloc(sizeof(*prtd), GFP_KERNEL); + if (!prtd) + return -ENOMEM; + + prtd->dma_desc_arr = dma_alloc_coherent(card->dev, PAGE_SIZE, + &prtd->dma_desc_arr_phy, + GFP_KERNEL); + if (!prtd->dma_desc_arr) { + ret = -ENOMEM; + goto desc_err; + } + prtd->dma_desc_arr_size = PAGE_SIZE / sizeof(*prtd->dma_desc_arr); + + prtd->dma_pos_desc = dma_alloc_coherent(card->dev, + sizeof(*prtd->dma_pos_desc), + &prtd->dma_pos_desc_phy, + GFP_KERNEL); + if (!prtd->dma_pos_desc) { + ret = -ENOMEM; + goto pos_err; + } + + dma_data = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); + prtd->dma_data = dma_data; + + substream->runtime->private_data = prtd; + + return 0; +pos_err: + dma_free_coherent(card->dev, PAGE_SIZE, prtd->dma_desc_arr, + prtd->dma_desc_arr_phy); +desc_err: + kfree(prtd); + + return ret; +} + +static int loongson_pcm_close(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct snd_card *card = substream->pcm->card; + struct loongson_runtime_data *prtd = substream->runtime->private_data; + + dma_free_coherent(card->dev, PAGE_SIZE, prtd->dma_desc_arr, + prtd->dma_desc_arr_phy); + + dma_free_coherent(card->dev, sizeof(*prtd->dma_pos_desc), + prtd->dma_pos_desc, prtd->dma_pos_desc_phy); + + kfree(prtd); + return 0; +} + +static int loongson_pcm_mmap(struct snd_soc_component *component, + struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + return remap_pfn_range(vma, vma->vm_start, + substream->dma_buffer.addr >> PAGE_SHIFT, + vma->vm_end - vma->vm_start, vma->vm_page_prot); +} + +static int loongson_pcm_new(struct snd_soc_component *component, + struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + struct snd_pcm_substream *substream; + struct loongson_dma_data *dma_data; + unsigned int i; + int ret; + + for_each_pcm_streams(i) { + substream = rtd->pcm->streams[i].substream; + if (!substream) + continue; + + dma_data = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), + substream); + ret = devm_request_irq(card->dev, dma_data->irq, + loongson_pcm_dma_irq, + IRQF_TRIGGER_HIGH, LS_I2S_DRVNAME, + substream); + if (ret < 0) { + dev_err(card->dev, "request irq for DMA failed\n"); + return ret; + } + } + + return snd_pcm_set_fixed_buffer_all(rtd->pcm, SNDRV_DMA_TYPE_DEV, + card->dev, + ls_pcm_hardware.buffer_bytes_max); +} + +const struct snd_soc_component_driver loongson_i2s_component = { + .name = LS_I2S_DRVNAME, + .open = loongson_pcm_open, + .close = loongson_pcm_close, + .hw_params = loongson_pcm_hw_params, + .trigger = loongson_pcm_trigger, + .pointer = loongson_pcm_pointer, + .mmap = loongson_pcm_mmap, + .pcm_construct = loongson_pcm_new, +}; diff --git a/sound/soc/loongson/loongson_dma.h b/sound/soc/loongson/loongson_dma.h new file mode 100644 index 000000000000..073ee8c0c046 --- /dev/null +++ b/sound/soc/loongson/loongson_dma.h @@ -0,0 +1,16 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * ALSA ASoC interface for the Loongson platform + * + * Copyright (C) 2023 Loongson Technology Corporation Limited + * Author: Yingkun Meng <mengyingkun(a)loongson.cn> + */ + +#ifndef _LOONGSON_DMA_H +#define _LOONGSON_DMA_H + +#include <sound/soc.h> + +extern const struct snd_soc_component_driver loongson_i2s_component; + +#endif diff --git a/sound/soc/loongson/loongson_i2s.c b/sound/soc/loongson/loongson_i2s.c new file mode 100644 index 000000000000..35d34568be79 --- /dev/null +++ b/sound/soc/loongson/loongson_i2s.c @@ -0,0 +1,269 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Common functions for loongson I2S controller driver +// +// Copyright (C) 2023 Loongson Technology Corporation Limited. +// Author: Yingkun Meng <mengyingkun(a)loongson.cn> +// + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/delay.h> +#include <linux/pm_runtime.h> +#include <linux/dma-mapping.h> +#include <sound/soc.h> +#include <linux/regmap.h> +#include <sound/pcm_params.h> +#include "loongson_i2s.h" + +#define LOONGSON_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +static int loongson_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct loongson_i2s *i2s = snd_soc_dai_get_drvdata(dai); + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + regmap_update_bits(i2s->regmap, LS_I2S_CTRL, + I2S_CTRL_TX_EN | I2S_CTRL_TX_DMA_EN, + I2S_CTRL_TX_EN | I2S_CTRL_TX_DMA_EN); + else + regmap_update_bits(i2s->regmap, LS_I2S_CTRL, + I2S_CTRL_RX_EN | I2S_CTRL_RX_DMA_EN, + I2S_CTRL_RX_EN | I2S_CTRL_RX_DMA_EN); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + regmap_update_bits(i2s->regmap, LS_I2S_CTRL, + I2S_CTRL_TX_EN | I2S_CTRL_TX_DMA_EN, 0); + else + regmap_update_bits(i2s->regmap, LS_I2S_CTRL, + I2S_CTRL_RX_EN | I2S_CTRL_RX_DMA_EN, 0); + break; + default: + ret = -EINVAL; + } + + return ret; +} + +static int loongson_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct loongson_i2s *i2s = snd_soc_dai_get_drvdata(dai); + u32 clk_rate = i2s->clk_rate; + u32 sysclk = i2s->sysclk; + u32 bits = params_width(params); + u32 chans = params_channels(params); + u32 fs = params_rate(params); + u32 bclk_ratio, mclk_ratio; + u32 mclk_ratio_frac; + u32 val = 0; + + switch (i2s->rev_id) { + case 0: + bclk_ratio = DIV_ROUND_CLOSEST(clk_rate, + (bits * chans * fs * 2)) - 1; + mclk_ratio = DIV_ROUND_CLOSEST(clk_rate, (sysclk * 2)) - 1; + + /* According to 2k1000LA user manual, set bits == depth */ + val |= (bits << 24); + val |= (bits << 16); + val |= (bclk_ratio << 8); + val |= mclk_ratio; + regmap_write(i2s->regmap, LS_I2S_CFG, val); + + break; + case 1: + bclk_ratio = DIV_ROUND_CLOSEST(sysclk, + (bits * chans * fs * 2)) - 1; + mclk_ratio = clk_rate / sysclk; + mclk_ratio_frac = DIV_ROUND_CLOSEST(((u64)clk_rate << 16), + sysclk) - (mclk_ratio << 16); + + regmap_read(i2s->regmap, LS_I2S_CFG, &val); + val |= (bits << 24); + val |= (bclk_ratio << 8); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + val |= (bits << 16); + else + val |= bits; + regmap_write(i2s->regmap, LS_I2S_CFG, val); + + val = (mclk_ratio_frac << 16) | mclk_ratio; + regmap_write(i2s->regmap, LS_I2S_CFG1, val); + + break; + default: + dev_err(i2s->dev, "I2S revision invalid\n"); + return -EINVAL; + } + + return 0; +} + +static int loongson_i2s_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct loongson_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + i2s->sysclk = freq; + + return 0; +} + +static int loongson_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct loongson_i2s *i2s = snd_soc_dai_get_drvdata(dai); + u32 val; + int ret; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_RIGHT_J: + regmap_update_bits(i2s->regmap, LS_I2S_CTRL, I2S_CTRL_MSB, + I2S_CTRL_MSB); + break; + default: + return -EINVAL; + } + + + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BC_FC: + break; + case SND_SOC_DAIFMT_BP_FC: + /* Enable master mode */ + regmap_update_bits(i2s->regmap, LS_I2S_CTRL, I2S_CTRL_MASTER, + I2S_CTRL_MASTER); + if (i2s->rev_id == 1) { + ret = regmap_read_poll_timeout_atomic(i2s->regmap, + LS_I2S_CTRL, val, + val & I2S_CTRL_CLK_READY, + 10, 500000); + if (ret < 0) + dev_warn(dai->dev, "wait BCLK ready timeout\n"); + } + break; + case SND_SOC_DAIFMT_BC_FP: + /* Enable MCLK */ + if (i2s->rev_id == 1) { + regmap_update_bits(i2s->regmap, LS_I2S_CTRL, + I2S_CTRL_MCLK_EN, + I2S_CTRL_MCLK_EN); + ret = regmap_read_poll_timeout_atomic(i2s->regmap, + LS_I2S_CTRL, val, + val & I2S_CTRL_MCLK_READY, + 10, 500000); + if (ret < 0) + dev_warn(dai->dev, "wait MCLK ready timeout\n"); + } + break; + case SND_SOC_DAIFMT_BP_FP: + /* Enable MCLK */ + if (i2s->rev_id == 1) { + regmap_update_bits(i2s->regmap, LS_I2S_CTRL, + I2S_CTRL_MCLK_EN, + I2S_CTRL_MCLK_EN); + ret = regmap_read_poll_timeout_atomic(i2s->regmap, + LS_I2S_CTRL, val, + val & I2S_CTRL_MCLK_READY, + 10, 500000); + if (ret < 0) + dev_warn(dai->dev, "wait MCLK ready timeout\n"); + } + + /* Enable master mode */ + regmap_update_bits(i2s->regmap, LS_I2S_CTRL, I2S_CTRL_MASTER, + I2S_CTRL_MASTER); + if (i2s->rev_id == 1) { + ret = regmap_read_poll_timeout_atomic(i2s->regmap, + LS_I2S_CTRL, val, + val & I2S_CTRL_CLK_READY, + 10, 500000); + if (ret < 0) + dev_warn(dai->dev, "wait BCLK ready timeout\n"); + } + break; + default: + return -EINVAL; + } + + return 0; +} + +static const struct snd_soc_dai_ops loongson_i2s_dai_ops = { + .trigger = loongson_i2s_trigger, + .hw_params = loongson_i2s_hw_params, + .set_sysclk = loongson_i2s_set_dai_sysclk, + .set_fmt = loongson_i2s_set_fmt, +}; + +static int loongson_i2s_dai_probe(struct snd_soc_dai *cpu_dai) +{ + struct loongson_i2s *i2s = dev_get_drvdata(cpu_dai->dev); + + snd_soc_dai_init_dma_data(cpu_dai, &i2s->playback_dma_data, + &i2s->capture_dma_data); + snd_soc_dai_set_drvdata(cpu_dai, i2s); + + return 0; +} + +struct snd_soc_dai_driver loongson_i2s_dai = { + .name = "loongson-i2s", + .probe = loongson_i2s_dai_probe, + .playback = { + .stream_name = "CPU-Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = LOONGSON_I2S_FORMATS, + }, + .capture = { + .stream_name = "CPU-Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = LOONGSON_I2S_FORMATS, + }, + .ops = &loongson_i2s_dai_ops, + .symmetric_rate = 1, +}; + +static int i2s_suspend(struct device *dev) +{ + struct loongson_i2s *i2s = dev_get_drvdata(dev); + + regcache_cache_only(i2s->regmap, true); + + return 0; +} + +static int i2s_resume(struct device *dev) +{ + struct loongson_i2s *i2s = dev_get_drvdata(dev); + int ret; + + regcache_cache_only(i2s->regmap, false); + regcache_mark_dirty(i2s->regmap); + ret = regcache_sync(i2s->regmap); + + return ret; +} + +const struct dev_pm_ops loongson_i2s_pm = { + SET_SYSTEM_SLEEP_PM_OPS(i2s_suspend, i2s_resume) +}; diff --git a/sound/soc/loongson/loongson_i2s.h b/sound/soc/loongson/loongson_i2s.h new file mode 100644 index 000000000000..52788f6a94ad --- /dev/null +++ b/sound/soc/loongson/loongson_i2s.h @@ -0,0 +1,71 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * ALSA I2S interface for the Loongson platform + * + * Copyright (C) 2023 Loongson Technology Corporation Limited + * Author: Yingkun Meng <mengyingkun(a)loongson.cn> + */ + +#ifndef _LOONGSON_I2S_H +#define _LOONGSON_I2S_H + +#include <linux/regmap.h> +#include <sound/dmaengine_pcm.h> + +/* I2S Common Registers */ +#define LS_I2S_VER 0x00 /* I2S Version */ +#define LS_I2S_CFG 0x04 /* I2S Config */ +#define LS_I2S_CTRL 0x08 /* I2S Control */ +#define LS_I2S_RX_DATA 0x0C /* I2S DMA RX Address */ +#define LS_I2S_TX_DATA 0x10 /* I2S DMA TX Address */ + +/* 2K2000 I2S Specify Registers */ +#define LS_I2S_CFG1 0x14 /* I2S Config1 */ + +/* 7A2000 I2S Specify Registers */ +#define LS_I2S_TX_ORDER 0x100 /* TX DMA Order */ +#define LS_I2S_RX_ORDER 0x110 /* RX DMA Order */ + +/* Loongson I2S Control Register */ +#define I2S_CTRL_MCLK_READY (1 << 16) /* MCLK ready */ +#define I2S_CTRL_MASTER (1 << 15) /* Master mode */ +#define I2S_CTRL_MSB (1 << 14) /* MSB bit order */ +#define I2S_CTRL_RX_EN (1 << 13) /* RX enable */ +#define I2S_CTRL_TX_EN (1 << 12) /* TX enable */ +#define I2S_CTRL_RX_DMA_EN (1 << 11) /* DMA RX enable */ +#define I2S_CTRL_CLK_READY (1 << 8) /* BCLK ready */ +#define I2S_CTRL_TX_DMA_EN (1 << 7) /* DMA TX enable */ +#define I2S_CTRL_RESET (1 << 4) /* Controller soft reset */ +#define I2S_CTRL_MCLK_EN (1 << 3) /* Enable MCLK */ +#define I2S_CTRL_RX_INT_EN (1 << 1) /* RX interrupt enable */ +#define I2S_CTRL_TX_INT_EN (1 << 0) /* TX interrupt enable */ + +#define LS_I2S_DRVNAME "loongson-i2s" + +struct loongson_dma_data { + dma_addr_t dev_addr; /* device physical address for DMA */ + void __iomem *order_addr; /* DMA order register */ + u32 irq; /* DMA irq */ +}; + +struct loongson_i2s { + struct device *dev; + union { + struct snd_dmaengine_dai_dma_data playback_dma_data; + struct loongson_dma_data tx_dma_data; + }; + union { + struct snd_dmaengine_dai_dma_data capture_dma_data; + struct loongson_dma_data rx_dma_data; + }; + struct regmap *regmap; + void __iomem *reg_base; + u32 rev_id; + u32 clk_rate; + u32 sysclk; +}; + +extern const struct dev_pm_ops loongson_i2s_pm; +extern struct snd_soc_dai_driver loongson_i2s_dai; + +#endif diff --git a/sound/soc/loongson/loongson_i2s_pci.c b/sound/soc/loongson/loongson_i2s_pci.c new file mode 100644 index 000000000000..6dcfb17d3276 --- /dev/null +++ b/sound/soc/loongson/loongson_i2s_pci.c @@ -0,0 +1,171 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// loongson_i2s_pci.c -- Loongson I2S controller driver +// +// Copyright (C) 2023 Loongson Technology Corporation Limited +// Author: Yingkun Meng <mengyingkun(a)loongson.cn> +// + +#include <linux/module.h> +#include <linux/delay.h> +#include <linux/pm_runtime.h> +#include <linux/dma-mapping.h> +#include <linux/acpi.h> +#include <linux/pci.h> +#include <sound/soc.h> +#include "loongson_i2s.h" +#include "loongson_dma.h" + +static bool loongson_i2s_wr_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case LS_I2S_CFG: + case LS_I2S_CTRL: + case LS_I2S_RX_DATA: + case LS_I2S_TX_DATA: + case LS_I2S_CFG1: + return true; + default: + return false; + }; +} + +static bool loongson_i2s_rd_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case LS_I2S_VER: + case LS_I2S_CFG: + case LS_I2S_CTRL: + case LS_I2S_RX_DATA: + case LS_I2S_TX_DATA: + case LS_I2S_CFG1: + return true; + default: + return false; + }; +} + +static bool loongson_i2s_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case LS_I2S_CFG: + case LS_I2S_CTRL: + case LS_I2S_RX_DATA: + case LS_I2S_TX_DATA: + case LS_I2S_CFG1: + return true; + default: + return false; + }; +} + +static const struct regmap_config loongson_i2s_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = LS_I2S_CFG1, + .writeable_reg = loongson_i2s_wr_reg, + .readable_reg = loongson_i2s_rd_reg, + .volatile_reg = loongson_i2s_volatile_reg, + .cache_type = REGCACHE_FLAT, +}; + +static int loongson_i2s_pci_probe(struct pci_dev *pdev, + const struct pci_device_id *pid) +{ + const struct fwnode_handle *fwnode = pdev->dev.fwnode; + struct loongson_dma_data *tx_data, *rx_data; + struct loongson_i2s *i2s; + int ret; + + if (pcim_enable_device(pdev)) { + dev_err(&pdev->dev, "pci_enable_device failed\n"); + return -ENODEV; + } + + i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL); + if (!i2s) + return -ENOMEM; + + i2s->rev_id = pdev->revision; + i2s->dev = &pdev->dev; + pci_set_drvdata(pdev, i2s); + + ret = pcim_iomap_regions(pdev, 1 << 0, dev_name(&pdev->dev)); + if (ret < 0) { + dev_err(&pdev->dev, "iomap_regions failed\n"); + return ret; + } + i2s->reg_base = pcim_iomap_table(pdev)[0]; + i2s->regmap = devm_regmap_init_mmio(&pdev->dev, i2s->reg_base, + &loongson_i2s_regmap_config); + if (IS_ERR(i2s->regmap)) { + dev_err(&pdev->dev, "regmap_init_mmio failed\n"); + return PTR_ERR(i2s->regmap); + } + + tx_data = &i2s->tx_dma_data; + rx_data = &i2s->rx_dma_data; + + tx_data->dev_addr = (dma_addr_t)i2s->reg_base + LS_I2S_TX_DATA; + tx_data->order_addr = i2s->reg_base + LS_I2S_TX_ORDER; + + rx_data->dev_addr = (dma_addr_t)i2s->reg_base + LS_I2S_RX_DATA; + rx_data->order_addr = i2s->reg_base + LS_I2S_RX_ORDER; + + tx_data->irq = fwnode_irq_get_byname(fwnode, "tx"); + if (tx_data->irq < 0) { + dev_err(&pdev->dev, "dma tx irq invalid\n"); + return tx_data->irq; + } + + rx_data->irq = fwnode_irq_get_byname(fwnode, "rx"); + if (rx_data->irq < 0) { + dev_err(&pdev->dev, "dma rx irq invalid\n"); + return rx_data->irq; + } + + device_property_read_u32(&pdev->dev, "clock-frequency", &i2s->clk_rate); + if (!i2s->clk_rate) { + dev_err(&pdev->dev, "clock-frequency property invalid\n"); + return -EINVAL; + } + + dma_set_mask_and_coherent(&pdev->dev, DMA_BIT_MASK(64)); + + if (i2s->rev_id == 1) { + regmap_write(i2s->regmap, LS_I2S_CTRL, I2S_CTRL_RESET); + udelay(200); + } + + ret = devm_snd_soc_register_component(&pdev->dev, + &loongson_i2s_component, + &loongson_i2s_dai, 1); + if (ret) { + dev_err(&pdev->dev, "register DAI failed %d\n", ret); + return ret; + } + + return 0; +} + +static const struct pci_device_id loongson_i2s_ids[] = { + { PCI_DEVICE(PCI_VENDOR_ID_LOONGSON, 0x7a27) }, + { }, +}; +MODULE_DEVICE_TABLE(pci, loongson_i2s_ids); + +static struct pci_driver loongson_i2s_driver = { + .name = "loongson-i2s-pci", + .id_table = loongson_i2s_ids, + .probe = loongson_i2s_pci_probe, + .driver = { + .owner = THIS_MODULE, + .pm = pm_sleep_ptr(&loongson_i2s_pm), + }, +}; +module_pci_driver(loongson_i2s_driver); + +MODULE_DESCRIPTION("Loongson I2S Master Mode ASoC Driver"); +MODULE_AUTHOR("Loongson Technology Corporation Limited"); +MODULE_LICENSE("GPL"); -- 2.33.0

2 3

[PATCH] ASoC: Intel: sof_rt5682: reorder quirk table
by Mac Chiang 15 Jun '23

15 Jun '23

From: Terry Cheong <htcheong(a)chromium.org> Reorder the entries in quirk table to group entries with same platform. Signed-off-by: Terry Cheong <htcheong(a)chromium.org> --- sound/soc/intel/boards/sof_rt5682.c | 52 ++++++++++++++--------------- 1 file changed, 26 insertions(+), 26 deletions(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 86bbc1fea6ff..be34d7595537 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -981,14 +981,13 @@ static const struct platform_device_id board_ids[] = { .name = "sof_rt5682", }, { - .name = "tgl_mx98357_rt5682", + .name = "cml_rt1015_rt5682", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_MCLK_24MHZ | SOF_RT5682_SSP_CODEC(0) | SOF_SPEAKER_AMP_PRESENT | - SOF_RT5682_SSP_AMP(1) | - SOF_RT5682_NUM_HDMIDEV(4) | - SOF_BT_OFFLOAD_SSP(2) | - SOF_SSP_BT_OFFLOAD_PRESENT), + SOF_RT1015_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1)), }, { .name = "jsl_rt5682_rt1015", @@ -1000,33 +999,38 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(1)), }, { - .name = "tgl_mx98373_rt5682", + .name = "jsl_rt5682_mx98360", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_MCLK_24MHZ | SOF_RT5682_SSP_CODEC(0) | SOF_SPEAKER_AMP_PRESENT | - SOF_MAX98373_SPEAKER_AMP_PRESENT | - SOF_RT5682_SSP_AMP(1) | - SOF_RT5682_NUM_HDMIDEV(4) | - SOF_BT_OFFLOAD_SSP(2) | - SOF_SSP_BT_OFFLOAD_PRESENT), + SOF_MAX98360A_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1)), }, { - .name = "jsl_rt5682_mx98360", + .name = "jsl_rt5682_rt1015p", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | SOF_RT5682_MCLK_24MHZ | SOF_RT5682_SSP_CODEC(0) | SOF_SPEAKER_AMP_PRESENT | - SOF_MAX98360A_SPEAKER_AMP_PRESENT | + SOF_RT1015P_SPEAKER_AMP_PRESENT | SOF_RT5682_SSP_AMP(1)), }, { - .name = "cml_rt1015_rt5682", + .name = "jsl_rt5682", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | SOF_RT5682_MCLK_24MHZ | + SOF_RT5682_SSP_CODEC(0)), + }, + { + .name = "tgl_mx98357_rt5682", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | SOF_RT5682_SSP_CODEC(0) | SOF_SPEAKER_AMP_PRESENT | - SOF_RT1015_SPEAKER_AMP_PRESENT | - SOF_RT5682_SSP_AMP(1)), + SOF_RT5682_SSP_AMP(1) | + SOF_RT5682_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), }, { .name = "tgl_rt1011_rt5682", @@ -1040,13 +1044,15 @@ static const struct platform_device_id board_ids[] = { SOF_SSP_BT_OFFLOAD_PRESENT), }, { - .name = "jsl_rt5682_rt1015p", + .name = "tgl_mx98373_rt5682", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_RT5682_MCLK_24MHZ | SOF_RT5682_SSP_CODEC(0) | SOF_SPEAKER_AMP_PRESENT | - SOF_RT1015P_SPEAKER_AMP_PRESENT | - SOF_RT5682_SSP_AMP(1)), + SOF_MAX98373_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1) | + SOF_RT5682_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), }, { .name = "adl_mx98373_rt5682", @@ -1149,12 +1155,6 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(1) | SOF_RT5682_NUM_HDMIDEV(4)), }, - { - .name = "jsl_rt5682", - .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_RT5682_MCLK_24MHZ | - SOF_RT5682_SSP_CODEC(0)), - }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); -- 2.20.1

2 2

[PATCH v3 1/2] ASoC: dt-bindings: qcom,wsa8840: Add WSA884x family of speakers
by Krzysztof Kozlowski 15 Jun '23

15 Jun '23

Add binding for WSA8840/WSA8845/WSA8845H smart speaker amplifiers used in Qualcomm QRD8550 board with SM8550 SoC. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Changes in v3: 1. None. Changes in v2: 1. Correct compatible (sdw version 1 -> 2). Cc: Patrick Lai <quic_plai(a)quicinc.com> Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- .../bindings/sound/qcom,wsa8840.yaml | 66 +++++++++++++++++++ 1 file changed, 66 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml diff --git a/Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml b/Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml new file mode 100644 index 000000000000..e6723c9e312a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml @@ -0,0 +1,66 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,wsa8840.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm WSA8840/WSA8845/WSA8845H smart speaker amplifier + +maintainers: + - Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> + - Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> + +description: + WSA884X is a family of Qualcomm Aqstic smart speaker amplifiers using + SoundWire digital audio interface. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: sdw20217020400 + + reg: + maxItems: 1 + + powerdown-gpios: + description: Powerdown/Shutdown line to use (pin SD_N) + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + vdd-1p8-supply: true + vdd-io-supply: true + +required: + - compatible + - reg + - powerdown-gpios + - '#sound-dai-cells' + - vdd-1p8-supply + - vdd-io-supply + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + soundwire-controller { + #address-cells = <2>; + #size-cells = <0>; + + speaker@0,1 { + compatible = "sdw20217020400"; + reg = <0 1>; + pinctrl-names = "default"; + pinctrl-0 = <&spkr_2_sd_n_active>; + powerdown-gpios = <&lpass_tlmm 18 GPIO_ACTIVE_LOW>; + #sound-dai-cells = <0>; + sound-name-prefix = "SpkrRight"; + vdd-1p8-supply = <&vreg_l15b_1p8>; + vdd-io-supply = <&vreg_l3g_1p2>; + }; + }; -- 2.34.1

2 3

pcm: fix minor bug in ioctl
by GitHub pull_request - opened 15 Jun '23

15 Jun '23

alsa-project/alsa-lib pull request #330 was opened from dancrossnyc: Commit 2115cdb added a new call to the `SNDRV_PCM_IOCTL_SW_PARAMS` ioctl on line 675 of src/pcm/pcm_hw.c, but passed the `sw_params` argument by value; this should be passed by pointer. I ran across this in the context of the direwolf software modem for amateur radio; debugging details are in https://groups.io/g/direwolf/message/8286 Fixes #329 Request URL : https://github.com/alsa-project/alsa-lib/pull/330 Patch URL : https://github.com/alsa-project/alsa-lib/pull/330.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

params argument to `SNDRV_PCM_IOCTL_SW_PARAMS` should be pointer
by GitHub issues - opened 15 Jun '23

15 Jun '23

alsa-project/alsa-lib issue #329 was opened from dancrossnyc: Commit 2115cdb4d added a call to the `SNDRV_PCM_IOCTL_SW_PARAMS` ioctl on line 675 of src/pcm/pcm_hw.c, but passes the `sw_params` argument by value; this should be passed by pointer. I ran across this in the context of the direwolf software modem for amateur radio; debugging details are in https://groups.io/g/direwolf/message/8286. The patch is trivial; I will send a PR. Thanks! Issue URL : https://github.com/alsa-project/alsa-lib/issues/329 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: simple-card-utils.c: share asoc_graph_parse_dai()
by Kuninori Morimoto 14 Jun '23

14 Jun '23

Current Audio Graph Card/Card2 implements asoc_simple_parse_dai() on each driver, but these are same function. This patch share it as asoc_graph_parse_dai(). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/simple_card_utils.h | 3 + sound/soc/generic/audio-graph-card.c | 107 +------------------------- sound/soc/generic/audio-graph-card2.c | 107 +------------------------- sound/soc/generic/simple-card-utils.c | 105 +++++++++++++++++++++++++ 4 files changed, 110 insertions(+), 212 deletions(-) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 0e46f985eeda..9daef37fe9a8 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -195,6 +195,9 @@ int asoc_simple_remove(struct platform_device *pdev); int asoc_graph_card_probe(struct snd_soc_card *card); int asoc_graph_is_ports0(struct device_node *port); +int asoc_graph_parse_dai(struct device_node *ep, + struct snd_soc_dai_link_component *dlc, + int *is_single_link); #ifdef DEBUG static inline void asoc_simple_debug_dai(struct asoc_simple_priv *priv, diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 4e85536a1b26..c6e0f9132193 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -55,60 +55,6 @@ static const struct snd_soc_ops graph_ops = { .hw_params = asoc_simple_hw_params, }; -static int graph_get_dai_id(struct device_node *ep) -{ - struct device_node *node; - struct device_node *endpoint; - struct of_endpoint info; - int i, id; - const u32 *reg; - int ret; - - /* use driver specified DAI ID if exist */ - ret = snd_soc_get_dai_id(ep); - if (ret != -ENOTSUPP) - return ret; - - /* use endpoint/port reg if exist */ - ret = of_graph_parse_endpoint(ep, &info); - if (ret == 0) { - /* - * Because it will count port/endpoint if it doesn't have "reg". - * But, we can't judge whether it has "no reg", or "reg = <0>" - * only of_graph_parse_endpoint(). - * We need to check "reg" property - */ - if (of_property_present(ep, "reg")) - return info.id; - - node = of_get_parent(ep); - reg = of_get_property(node, "reg", NULL); - of_node_put(node); - if (reg) - return info.port; - } - node = of_graph_get_port_parent(ep); - - /* - * Non HDMI sound case, counting port/endpoint on its DT - * is enough. Let's count it. - */ - i = 0; - id = -1; - for_each_endpoint_of_node(node, endpoint) { - if (endpoint == ep) - id = i; - i++; - } - - of_node_put(node); - - if (id < 0) - return -ENODEV; - - return id; -} - static bool soc_component_is_pcm(struct snd_soc_dai_link_component *dlc) { struct snd_soc_dai *dai = snd_soc_find_dai_with_mutex(dlc); @@ -120,57 +66,6 @@ static bool soc_component_is_pcm(struct snd_soc_dai_link_component *dlc) return false; } -static int asoc_simple_parse_dai(struct device_node *ep, - struct snd_soc_dai_link_component *dlc, - int *is_single_link) -{ - struct device_node *node; - struct of_phandle_args args; - int ret; - - if (!ep) - return 0; - - node = of_graph_get_port_parent(ep); - - /* Get dai->name */ - args.np = node; - args.args[0] = graph_get_dai_id(ep); - args.args_count = (of_graph_get_endpoint_count(node) > 1); - - /* - * FIXME - * - * Here, dlc->dai_name is pointer to CPU/Codec DAI name. - * If user unbinded CPU or Codec driver, but not for Sound Card, - * dlc->dai_name is keeping unbinded CPU or Codec - * driver's pointer. - * - * If user re-bind CPU or Codec driver again, ALSA SoC will try - * to rebind Card via snd_soc_try_rebind_card(), but because of - * above reason, it might can't bind Sound Card. - * Because Sound Card is pointing to released dai_name pointer. - * - * To avoid this rebind Card issue, - * 1) It needs to alloc memory to keep dai_name eventhough - * CPU or Codec driver was unbinded, or - * 2) user need to rebind Sound Card everytime - * if he unbinded CPU or Codec. - */ - ret = snd_soc_get_dai_name(&args, &dlc->dai_name); - if (ret < 0) { - of_node_put(node); - return ret; - } - - dlc->of_node = node; - - if (is_single_link) - *is_single_link = of_graph_get_endpoint_count(node) == 1; - - return 0; -} - static void graph_parse_convert(struct device *dev, struct device_node *ep, struct asoc_simple_data *adata) @@ -231,7 +126,7 @@ static int graph_parse_node(struct asoc_simple_priv *priv, graph_parse_mclk_fs(top, ep, dai_props); - ret = asoc_simple_parse_dai(ep, dlc, cpu); + ret = asoc_graph_parse_dai(ep, dlc, cpu); if (ret < 0) return ret; diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index 25aa79dd55b3..542c4a114940 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -353,111 +353,6 @@ static const struct snd_soc_ops graph_ops = { .hw_params = asoc_simple_hw_params, }; -static int graph_get_dai_id(struct device_node *ep) -{ - struct device_node *node; - struct device_node *endpoint; - struct of_endpoint info; - int i, id; - const u32 *reg; - int ret; - - /* use driver specified DAI ID if exist */ - ret = snd_soc_get_dai_id(ep); - if (ret != -ENOTSUPP) - return ret; - - /* use endpoint/port reg if exist */ - ret = of_graph_parse_endpoint(ep, &info); - if (ret == 0) { - /* - * Because it will count port/endpoint if it doesn't have "reg". - * But, we can't judge whether it has "no reg", or "reg = <0>" - * only of_graph_parse_endpoint(). - * We need to check "reg" property - */ - if (of_property_present(ep, "reg")) - return info.id; - - node = of_get_parent(ep); - reg = of_get_property(node, "reg", NULL); - of_node_put(node); - if (reg) - return info.port; - } - node = of_graph_get_port_parent(ep); - - /* - * Non HDMI sound case, counting port/endpoint on its DT - * is enough. Let's count it. - */ - i = 0; - id = -1; - for_each_endpoint_of_node(node, endpoint) { - if (endpoint == ep) - id = i; - i++; - } - - of_node_put(node); - - if (id < 0) - return -ENODEV; - - return id; -} - -static int asoc_simple_parse_dai(struct device_node *ep, - struct snd_soc_dai_link_component *dlc, - int *is_single_link) -{ - struct device_node *node; - struct of_phandle_args args; - int ret; - - if (!ep) - return 0; - - node = of_graph_get_port_parent(ep); - - /* Get dai->name */ - args.np = node; - args.args[0] = graph_get_dai_id(ep); - args.args_count = (of_graph_get_endpoint_count(node) > 1); - - /* - * FIXME - * - * Here, dlc->dai_name is pointer to CPU/Codec DAI name. - * If user unbinded CPU or Codec driver, but not for Sound Card, - * dlc->dai_name is keeping unbinded CPU or Codec - * driver's pointer. - * - * If user re-bind CPU or Codec driver again, ALSA SoC will try - * to rebind Card via snd_soc_try_rebind_card(), but because of - * above reason, it might can't bind Sound Card. - * Because Sound Card is pointing to released dai_name pointer. - * - * To avoid this rebind Card issue, - * 1) It needs to alloc memory to keep dai_name eventhough - * CPU or Codec driver was unbinded, or - * 2) user need to rebind Sound Card everytime - * if he unbinded CPU or Codec. - */ - ret = snd_soc_get_dai_name(&args, &dlc->dai_name); - if (ret < 0) { - of_node_put(node); - return ret; - } - - dlc->of_node = node; - - if (is_single_link) - *is_single_link = of_graph_get_endpoint_count(node) == 1; - - return 0; -} - static void graph_parse_convert(struct device_node *ep, struct simple_dai_props *props) { @@ -512,7 +407,7 @@ static int __graph_parse_node(struct asoc_simple_priv *priv, graph_parse_mclk_fs(ep, dai_props); - ret = asoc_simple_parse_dai(ep, dlc, &is_single_links); + ret = asoc_graph_parse_dai(ep, dlc, &is_single_links); if (ret < 0) return ret; diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index b5ac0f0d5e8e..6a3c9e4e1cfe 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -1019,6 +1019,111 @@ int asoc_graph_is_ports0(struct device_node *np) } EXPORT_SYMBOL_GPL(asoc_graph_is_ports0); +static int graph_get_dai_id(struct device_node *ep) +{ + struct device_node *node; + struct device_node *endpoint; + struct of_endpoint info; + int i, id; + int ret; + + /* use driver specified DAI ID if exist */ + ret = snd_soc_get_dai_id(ep); + if (ret != -ENOTSUPP) + return ret; + + /* use endpoint/port reg if exist */ + ret = of_graph_parse_endpoint(ep, &info); + if (ret == 0) { + /* + * Because it will count port/endpoint if it doesn't have "reg". + * But, we can't judge whether it has "no reg", or "reg = <0>" + * only of_graph_parse_endpoint(). + * We need to check "reg" property + */ + if (of_property_present(ep, "reg")) + return info.id; + + node = of_get_parent(ep); + ret = of_property_present(node, "reg"); + of_node_put(node); + if (ret) + return info.port; + } + node = of_graph_get_port_parent(ep); + + /* + * Non HDMI sound case, counting port/endpoint on its DT + * is enough. Let's count it. + */ + i = 0; + id = -1; + for_each_endpoint_of_node(node, endpoint) { + if (endpoint == ep) + id = i; + i++; + } + + of_node_put(node); + + if (id < 0) + return -ENODEV; + + return id; +} + +int asoc_graph_parse_dai(struct device_node *ep, + struct snd_soc_dai_link_component *dlc, + int *is_single_link) +{ + struct device_node *node; + struct of_phandle_args args = {}; + int ret; + + if (!ep) + return 0; + + node = of_graph_get_port_parent(ep); + + /* Get dai->name */ + args.np = node; + args.args[0] = graph_get_dai_id(ep); + args.args_count = (of_graph_get_endpoint_count(node) > 1); + + /* + * FIXME + * + * Here, dlc->dai_name is pointer to CPU/Codec DAI name. + * If user unbinded CPU or Codec driver, but not for Sound Card, + * dlc->dai_name is keeping unbinded CPU or Codec + * driver's pointer. + * + * If user re-bind CPU or Codec driver again, ALSA SoC will try + * to rebind Card via snd_soc_try_rebind_card(), but because of + * above reason, it might can't bind Sound Card. + * Because Sound Card is pointing to released dai_name pointer. + * + * To avoid this rebind Card issue, + * 1) It needs to alloc memory to keep dai_name eventhough + * CPU or Codec driver was unbinded, or + * 2) user need to rebind Sound Card everytime + * if he unbinded CPU or Codec. + */ + ret = snd_soc_get_dai_name(&args, &dlc->dai_name); + if (ret < 0) { + of_node_put(node); + return ret; + } + + dlc->of_node = node; + + if (is_single_link) + *is_single_link = of_graph_get_endpoint_count(node) == 1; + + return 0; +} +EXPORT_SYMBOL_GPL(asoc_graph_parse_dai); + /* Module information */ MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>"); MODULE_DESCRIPTION("ALSA SoC Simple Card Utils"); -- 2.25.1

2 1

[PATCH v2] sound: usb: quirks: add flag for HEM devices to enable native DSD playback
by Lukasz Tyl 14 Jun '23

14 Jun '23

This commit adds new DEVICE_FLG with QUIRK_FLAG_DSD_RAW and Vendor Id for HEM devices which supports native DSD. Prior to this change Linux kernel was not enabling native DSD playback for HEM devices, and as a result, DSD audio was being converted to PCM "on the fly". HEM devices, when connected to the system, would only play audio in PCM format, even if the source material was in DSD format. With the addition of new VENDOR_FLG in the quircks.c file, the devices are now correctly recognized, and raw DSD data is transmitted to the device, allowing for native DSD playback. Signed-off-by: Lukasz Tyl <ltyl(a)hem-e.com> --- Changes in v2: - correct entry position (entry sorted by vendor id). --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 3ecd1ba7fd4b..6cf55b7f7a04 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2191,6 +2191,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x2ab6, /* T+A devices */ QUIRK_FLAG_DSD_RAW), + VENDOR_FLG(0x3336, /* HEM devices */ + QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x3353, /* Khadas devices */ QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x3842, /* EVGA */ -- 2.34.1

2 1

[PATCH] ASoC: SOF: amd: Add support for IPC with a reply_size set to zero
by Venkata Prasad Potturu 14 Jun '23

14 Jun '23

Add support for IPC tx_message with a reply_size set to zero, return zero when message reply_size is zero at acp_dsp_ipc_get_reply(). Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- sound/soc/sof/amd/acp-ipc.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/sof/amd/acp-ipc.c b/sound/soc/sof/amd/acp-ipc.c index 749e856dc601..8a0fc635a997 100644 --- a/sound/soc/sof/amd/acp-ipc.c +++ b/sound/soc/sof/amd/acp-ipc.c @@ -130,6 +130,13 @@ static void acp_dsp_ipc_get_reply(struct snd_sof_dev *sdev) memcpy(msg->reply_data, &reply, sizeof(reply)); ret = reply.error; } else { + /* + * To support an IPC tx_message with a + * reply_size set to zero. + */ + if (!msg->reply_size) + goto out; + /* reply correct size ? */ if (reply.hdr.size != msg->reply_size && !(reply.hdr.cmd & SOF_IPC_GLB_PROBE)) { -- 2.25.1

2 1

[PATCH] ASoC: dt-bindings: ti,tlv320aic3x: Add missing type for "gpio-reset"
by Rob Herring 14 Jun '23

14 Jun '23

"gpio-reset" may be deprecated, but it still needs a type. Signed-off-by: Rob Herring <robh(a)kernel.org> --- Documentation/devicetree/bindings/sound/ti,tlv320aic3x.yaml | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/ti,tlv320aic3x.yaml b/Documentation/devicetree/bindings/sound/ti,tlv320aic3x.yaml index e8ca9f3369f8..206f6d61e362 100644 --- a/Documentation/devicetree/bindings/sound/ti,tlv320aic3x.yaml +++ b/Documentation/devicetree/bindings/sound/ti,tlv320aic3x.yaml @@ -61,6 +61,7 @@ properties: GPIO specification for the active low RESET input. gpio-reset: + $ref: /schemas/types.yaml#/definitions/uint32-matrix maxItems: 1 description: Deprecated, please use reset-gpios instead. -- 2.39.2

3 2

[PATCH v2] ALSA: compress: allow setting codec params after next track
by Srinivas Kandagatla 14 Jun '23

14 Jun '23

For gapless playback it is possible that each track can have different codec profile with same decoder, for example we have WMA album, we may have different tracks as WMA v9, WMA v10 and so on Or if DSP's like QDSP have abililty to switch decoders on single stream for each track, then this call could be used to set new codec parameters. Existing code does not allow to change this profile while doing gapless playback. Reuse existing SNDRV_COMPRESS_SET_PARAMS to set this new track params along some additional checks to enforce proper state machine. With this new changes now the user can call SNDRV_COMPRESS_SET_PARAMS anytime after setting next track and additional check in write should also ensure that params are set before writing new data. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Changes since v1/RFC: - removed introduction of new IOCTL, as suggested. - update the state-machine doc. .../sound/designs/compress-offload.rst | 52 +++++++++---------- sound/core/compress_offload.c | 10 ++-- 2 files changed, 33 insertions(+), 29 deletions(-) diff --git a/Documentation/sound/designs/compress-offload.rst b/Documentation/sound/designs/compress-offload.rst index 935f325dbc77..205cadcabe70 100644 --- a/Documentation/sound/designs/compress-offload.rst +++ b/Documentation/sound/designs/compress-offload.rst @@ -256,32 +256,32 @@ Gapless Playback SM For Gapless, we move from running state to partial drain and back, along with setting of meta_data and signalling for next track :: - - +----------+ - compr_drain_notify() | | - +------------------------>| RUNNING | - | | | - | +----------+ - | | - | | - | | compr_next_track() - | | - | V - | +----------+ - | | | - | |NEXT_TRACK| - | | | - | +----------+ - | | - | | - | | compr_partial_drain() - | | - | V - | +----------+ - | | | - +------------------------ | PARTIAL_ | - | DRAIN | - +----------+ + +----------+ + compr_drain_notify() | | compr_set_params() iff next-track set. + +------------------------>| RUNNING |----------------------+ + | | | | + | +----------+ | + | | | + | | | + | | compr_next_track() | + | | V + | V | + | +----------+ | + | | | | + | |NEXT_TRACK| | + | | | | + | +----------+ | + | | | + | +----------------------------+ + | | + | | compr_partial_drain() + | | + | V + | +----------+ + | | | + +------------------------ | PARTIAL_ | + | DRAIN | + +----------+ Not supported ============= diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 243acad89fd3..fe67228e74b3 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -294,6 +294,9 @@ static ssize_t snd_compr_write(struct file *f, const char __user *buf, case SNDRV_PCM_STATE_SETUP: case SNDRV_PCM_STATE_PREPARED: case SNDRV_PCM_STATE_RUNNING: + /* Make sure next track params are set before writing new data */ + if (stream->next_track) + return -EPERM; break; default: mutex_unlock(&stream->device->lock); @@ -589,7 +592,8 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg) struct snd_compr_params *params; int retval; - if (stream->runtime->state == SNDRV_PCM_STATE_OPEN) { + if (stream->runtime->state == SNDRV_PCM_STATE_OPEN || + (stream->runtime->state == SNDRV_PCM_STATE_RUNNING && stream->next_track)) { /* * we should allow parameter change only when stream has been * opened not in other cases @@ -615,7 +619,8 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg) stream->metadata_set = false; stream->next_track = false; - stream->runtime->state = SNDRV_PCM_STATE_SETUP; + if (stream->runtime->state == SNDRV_PCM_STATE_OPEN) + stream->runtime->state = SNDRV_PCM_STATE_SETUP; } else { return -EPERM; } @@ -956,7 +961,6 @@ static int snd_compr_partial_drain(struct snd_compr_stream *stream) return retval; } - stream->next_track = false; return snd_compress_wait_for_drain(stream); } -- 2.21.0

3 3

[PATCH 0/7] ASoC: mediatek: Add support for MT79xx SoC
by Maso Hunag 14 Jun '23

14 Jun '23

From: Maso Huang <maso.huang(a)mediatek.com> This series of patches adds support for MediaTek AFE of MT79xx SoC. Patches are based on broonie tree "for-next" branch. Maso Huang (7): ASoC: mediatek: mt79xx: add common header ASoC: mediatek: mt79xx: support audio clock control ASoC: mediatek: mt79xx: add platform driver ASoC: mediatek: mt79xx: support etdm in platform driver ASoC: mediatek: mt79xx: add machine driver with wm8960 ASoC: dt-bindings: mediatek,mt79xx-wm8960: add mt79xx-wm8960 document ASoC: dt-bindings: mediatek,mt79xx-afe: add audio afe document .../bindings/sound/mediatek,mt79xx-afe.yaml | 102 +++ .../sound/mediatek,mt79xx-wm8960.yaml | 53 ++ sound/soc/mediatek/Kconfig | 20 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/mt79xx/Makefile | 10 + sound/soc/mediatek/mt79xx/mt79xx-afe-clk.c | 123 ++++ sound/soc/mediatek/mt79xx/mt79xx-afe-clk.h | 18 + sound/soc/mediatek/mt79xx/mt79xx-afe-common.h | 49 ++ sound/soc/mediatek/mt79xx/mt79xx-afe-pcm.c | 608 ++++++++++++++++++ sound/soc/mediatek/mt79xx/mt79xx-dai-etdm.c | 421 ++++++++++++ sound/soc/mediatek/mt79xx/mt79xx-reg.h | 206 ++++++ sound/soc/mediatek/mt79xx/mt79xx-wm8960.c | 185 ++++++ 12 files changed, 1796 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mediatek,mt79xx-afe.yaml create mode 100644 Documentation/devicetree/bindings/sound/mediatek,mt79xx-wm8960.yaml create mode 100644 sound/soc/mediatek/mt79xx/Makefile create mode 100644 sound/soc/mediatek/mt79xx/mt79xx-afe-clk.c create mode 100644 sound/soc/mediatek/mt79xx/mt79xx-afe-clk.h create mode 100644 sound/soc/mediatek/mt79xx/mt79xx-afe-common.h create mode 100644 sound/soc/mediatek/mt79xx/mt79xx-afe-pcm.c create mode 100644 sound/soc/mediatek/mt79xx/mt79xx-dai-etdm.c create mode 100644 sound/soc/mediatek/mt79xx/mt79xx-reg.h create mode 100644 sound/soc/mediatek/mt79xx/mt79xx-wm8960.c -- 2.18.0

5 13

[PATCH v3 0/3] Add Loongson I2S controller support
by YingKun Meng 14 Jun '23

14 Jun '23

Hi all, This patchset adds support for Loongson I2S controller, and introduce a ASoC machine driver for loongson platform. The Loongson I2S controller is available on Loongson 7a2000/2k2000 chips, works as a PCI device. It has two private DMA controllers used to playback and capture. Each DMA controller has one channel. The ASoC machine driver adds support for audio device which using loongson I2S controller to tranfser the audio data. The audio device uses "PRP0001" as its ACPI device ID, which provides a means to use the existing DT-compatible device identification in ACPI. Thanks. --- Changes v2 -> v3 - Patch 1 Split the driver into PCI driver and component driver. Move interrupt requesting from pcm_open to pcm_create. Use pcim_xxx() API to get iomap of I2S register. Add condition checking for the number of periods. Add set_fmt() callback for DAI ops. Add volatile registers for regmap of I2S. Remove redundant condition checking. Update comment style. - Patch 2 Remove redundant SoC DAI link. Update comment style. - Patch 3 No change. Changes v1 -> v2 - Patch 1 Replace pci_xxx() APIs with pcim_xxx() APIs. Fix build errors from lkp(a)intel.com. Minor changes in log printing. - Patch 2 Use ACPI bindings to reference I2S and codec nodes. - Patch 3 Add chip model restriction for title and description. Add 'required' option for sound-dai property. Yingkun Meng (3): ASoC: Add support for Loongson I2S controller ASoC: loongson: Add Loongson ASoC Sound Card Support ASoC: dt-bindings: Add support for Loongson audio card .../sound/loongson,ls-audio-card.yaml | 70 ++++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/loongson/Kconfig | 26 ++ sound/soc/loongson/Makefile | 8 + sound/soc/loongson/loongson_card.c | 230 ++++++++++++ sound/soc/loongson/loongson_dma.c | 350 ++++++++++++++++++ sound/soc/loongson/loongson_dma.h | 16 + sound/soc/loongson/loongson_i2s.c | 269 ++++++++++++++ sound/soc/loongson/loongson_i2s.h | 71 ++++ sound/soc/loongson/loongson_i2s_pci.c | 171 +++++++++ 11 files changed, 1213 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/loongson,ls-audio-card.yaml create mode 100644 sound/soc/loongson/Kconfig create mode 100644 sound/soc/loongson/Makefile create mode 100644 sound/soc/loongson/loongson_card.c create mode 100644 sound/soc/loongson/loongson_dma.c create mode 100644 sound/soc/loongson/loongson_dma.h create mode 100644 sound/soc/loongson/loongson_i2s.c create mode 100644 sound/soc/loongson/loongson_i2s.h create mode 100644 sound/soc/loongson/loongson_i2s_pci.c base-commit: a11e6515b019da62266b731ff20bc6863f00df4d -- 2.33.0

1 0

[PATCH] sound: usb: quirks: add flag for HEM devices to enable native DSD playback
by Lukasz Tyl 14 Jun '23

14 Jun '23

This commit adds new DEVICE_FLG with QUIRK_FLAG_DSD_RAW and Vendor Id for HEM devices which supports native DSD. Prior to this change Linux kernel was not enabling native DSD playback for HEM devices, and as a result, DSD audio was being converted to PCM "on the fly". HEM devices, when connected to the system, would only play audio in PCM format, even if the source material was in DSD format. With the addition of new VENDOR_FLG in the quircks.c file, the devices are now correctly recognized, and raw DSD data is transmitted to the device, allowing for native DSD playback. Signed-off-by: Lukasz Tyl <ltyl(a)hem-e.com> --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 3ecd1ba7fd4b..1414bf15ac21 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2197,6 +2197,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0xc502, /* HiBy devices */ QUIRK_FLAG_DSD_RAW), + VENDOR_FLG(0x3336, /* HEM devices */ + QUIRK_FLAG_DSD_RAW), {} /* terminator */ }; -- 2.34.1

2 1

[ PATCH RESEND v2 3/3] ASoC: dt-bindings: Add support for Loongson audio card
by YingKun Meng 14 Jun '23

14 Jun '23

From: Yingkun Meng <mengyingkun(a)loongson.cn> The audio card uses loongson I2S controller present in 7axxx/2kxxx chips to transfer audio data. On loongson platform, the chip has only one I2S controller. Signed-off-by: Yingkun Meng <mengyingkun(a)loongson.cn> --- v1 -> v2: * Add chip model restriction for title and description. * Add 'required' restriction for sound-dai property. .../sound/loongson,ls-audio-card.yaml | 70 +++++++++++++++++++ 1 file changed, 70 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/loongson,ls-audio-card.yaml diff --git a/Documentation/devicetree/bindings/sound/loongson,ls-audio-card.yaml b/Documentation/devicetree/bindings/sound/loongson,ls-audio-card.yaml new file mode 100644 index 000000000000..61e8babed402 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/loongson,ls-audio-card.yaml @@ -0,0 +1,70 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/loongson,ls-audio-card.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Loongson 7axxx/2kxxx ASoC audio sound card driver + +maintainers: + - Yingkun Meng <mengyingkun(a)loongson.cn> + +description: + The binding describes the sound card present in loongson + 7axxx/2kxxx platform. The sound card is an ASoC component + which uses Loongson I2S controller to transfer the audio data. + +properties: + compatible: + const: loongson,ls-audio-card + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User specified audio sound card name + + mclk-fs: + $ref: simple-card.yaml#/definitions/mclk-fs + + cpu: + description: Holds subnode which indicates cpu dai. + type: object + additionalProperties: false + properties: + sound-dai: + maxItems: 1 + required: + - sound-dai + + codec: + description: Holds subnode which indicates codec dai. + type: object + additionalProperties: false + properties: + sound-dai: + maxItems: 1 + required: + - sound-dai + +required: + - compatible + - model + - mclk-fs + - cpu + - codec + +additionalProperties: false + +examples: + - | + sound { + compatible = "loongson,ls-audio-card"; + model = "loongson-audio"; + mclk-fs = <512>; + + cpu { + sound-dai = <&i2s>; + }; + codec { + sound-dai = <&es8323>; + }; + }; -- 2.33.0

1 0

Re: [PATCH 7/7] ASoC: dt-bindings: mediatek,mt79xx-afe: add audio afe document
by Krzysztof Kozlowski 14 Jun '23

14 Jun '23

On 14/06/2023 11:19, Maso Huang (黃加竹) wrote: > On Wed, 2023-06-14 at 10:21 +0200, Krzysztof Kozlowski wrote: >> >> External email : Please do not click links or open attachments until >> you have verified the sender or the content. >> On 14/06/2023 09:37, Maso Huang (黃加竹) wrote: >>>>>> I already saw AFE, why it cannot be part of existing bindings? >>>> >>>> Can you answer this? >>>> >>> >>> Did you mean mtk-afe-pcm.txt? >>> If yes, I'll modify mtk-afe-pcm.txt to yaml format, and add mt7986 >> to >>> its compatible list. >>> >> >> No, I meant mediatek,mt8188-afe.yaml. >> >> Aren't you working on some old tree? If so, please don't... >> >> Best regards, >> Krzysztof >> > > Hi Krzysztof, > AFE is common name for our audio hardware, and the design might be > different for soc, like clock. > > And the design is the same for mt7981/mt7986/mt7988. > Is it better to create a new dtbinding file mediatk,mt7986-afe.yaml? Is it different? That was my question whether it can be part of existing bindings. Best regards, Krzysztof

1 0

[ PATCH v2 3/3] ASoC: dt-bindings: Add support for Loongson audio card
by YingKun Meng 14 Jun '23

14 Jun '23

From: Yingkun Meng <mengyingkun(a)loongson.cn> The audio card uses loongson I2S controller present in 7axxx/2kxxx chips to transfer audio data. On loongson platform, the chip has only one I2S controller. Signed-off-by: Yingkun Meng <mengyingkun(a)loongson.cn> --- .../sound/loongson,ls-audio-card.yaml | 70 +++++++++++++++++++ 1 file changed, 70 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/loongson,ls-audio-card.yaml diff --git a/Documentation/devicetree/bindings/sound/loongson,ls-audio-card.yaml b/Documentation/devicetree/bindings/sound/loongson,ls-audio-card.yaml new file mode 100644 index 000000000000..61e8babed402 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/loongson,ls-audio-card.yaml @@ -0,0 +1,70 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/loongson,ls-audio-card.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Loongson 7axxx/2kxxx ASoC audio sound card driver + +maintainers: + - Yingkun Meng <mengyingkun(a)loongson.cn> + +description: + The binding describes the sound card present in loongson + 7axxx/2kxxx platform. The sound card is an ASoC component + which uses Loongson I2S controller to transfer the audio data. + +properties: + compatible: + const: loongson,ls-audio-card + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User specified audio sound card name + + mclk-fs: + $ref: simple-card.yaml#/definitions/mclk-fs + + cpu: + description: Holds subnode which indicates cpu dai. + type: object + additionalProperties: false + properties: + sound-dai: + maxItems: 1 + required: + - sound-dai + + codec: + description: Holds subnode which indicates codec dai. + type: object + additionalProperties: false + properties: + sound-dai: + maxItems: 1 + required: + - sound-dai + +required: + - compatible + - model + - mclk-fs + - cpu + - codec + +additionalProperties: false + +examples: + - | + sound { + compatible = "loongson,ls-audio-card"; + model = "loongson-audio"; + mclk-fs = <512>; + + cpu { + sound-dai = <&i2s>; + }; + codec { + sound-dai = <&es8323>; + }; + }; -- 2.33.0

2 4

[ PATCH v2 2/3] ASoC: loongson: Add Loongson ASoC Sound Card Support
by YingKun Meng 14 Jun '23

14 Jun '23

From: Yingkun Meng <mengyingkun(a)loongson.cn> The Loongson ASoC Sound Card is a general ASoC DAI Link driver that can be used for Loongson CPU DAI drivers and external CODECs. The driver supports the use of ACPI table to describe device resources. On loongson 7axxx platforms, the audio device is an ACPI device. Signed-off-by: Yingkun Meng <mengyingkun(a)loongson.cn> --- sound/soc/loongson/Kconfig | 10 ++ sound/soc/loongson/Makefile | 4 + sound/soc/loongson/loongson_card.c | 237 +++++++++++++++++++++++++++++ 3 files changed, 251 insertions(+) create mode 100644 sound/soc/loongson/loongson_card.c diff --git a/sound/soc/loongson/Kconfig b/sound/soc/loongson/Kconfig index 4478ac91e402..c175f9de19a8 100644 --- a/sound/soc/loongson/Kconfig +++ b/sound/soc/loongson/Kconfig @@ -13,4 +13,14 @@ config SND_SOC_LOONGSON_I2S_PCI The controller is found in loongson bridge chips or SoCs, and work as a PCI device. +config SND_SOC_LOONGSON_CARD + tristate "Loongson Sound Card Driver" + select SND_SOC_LOONGSON_I2S_PCI + help + Say Y or M if you want to add support for SoC audio using + loongson I2S controller. + + The driver add support for ALSA SoC Audio support using + loongson I2S controller. + endmenu diff --git a/sound/soc/loongson/Makefile b/sound/soc/loongson/Makefile index cfd0de1b1b22..a173a0fe17fe 100644 --- a/sound/soc/loongson/Makefile +++ b/sound/soc/loongson/Makefile @@ -2,3 +2,7 @@ #Platform Support snd-soc-loongson-i2s-pci-objs := loongson_i2s_pci.o loongson_i2s.o obj-$(CONFIG_SND_SOC_LOONGSON_I2S_PCI) += snd-soc-loongson-i2s-pci.o + +#Machine Support +snd-soc-loongson-card-objs := loongson_card.o +obj-$(CONFIG_SND_SOC_LOONGSON_CARD) += snd-soc-loongson-card.o diff --git a/sound/soc/loongson/loongson_card.c b/sound/soc/loongson/loongson_card.c new file mode 100644 index 000000000000..7f0ce8957a4f --- /dev/null +++ b/sound/soc/loongson/loongson_card.c @@ -0,0 +1,237 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Loongson ASoC Audio driver + * + * Copyright (C) 2022 Loongson Technology Corporation Limited + */ + +#include <linux/module.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> +#include <linux/acpi.h> +#include <linux/pci.h> +#include <sound/pcm_params.h> + +static char codec_name[SND_ACPI_I2C_ID_LEN]; + +struct loongson_card_data { + struct snd_soc_card snd_card; + unsigned int mclk_fs; +}; + +static int loongson_card_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct loongson_card_data *ls_card = snd_soc_card_get_drvdata(rtd->card); + int ret, mclk; + + if (ls_card->mclk_fs) { + mclk = ls_card->mclk_fs * params_rate(params); + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, + SND_SOC_CLOCK_OUT); + if (ret < 0) { + dev_err(codec_dai->dev, "cpu_dai clock not set\n"); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "codec_dai clock not set\n"); + return ret; + } + } + return 0; +} + +static const struct snd_soc_ops loongson_ops = { + .hw_params = loongson_card_hw_params, +}; + +SND_SOC_DAILINK_DEFS(analog, + DAILINK_COMP_ARRAY(COMP_CPU("loongson-i2s")), + DAILINK_COMP_ARRAY(COMP_EMPTY()), + DAILINK_COMP_ARRAY(COMP_EMPTY())); + +static struct snd_soc_dai_link loongson_dai_links[] = { + { + .name = "HiFi PAIF TX", + .stream_name = "Playback", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_IB_NF + | SND_SOC_DAIFMT_CBC_CFC, + SND_SOC_DAILINK_REG(analog), + .ops = &loongson_ops, + }, + { + .name = "HiFi PAIF RX", + .stream_name = "Capture", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_IB_NF + | SND_SOC_DAIFMT_CBC_CFC, + SND_SOC_DAILINK_REG(analog), + .ops = &loongson_ops, + }, +}; + +static int loongson_card_parse_acpi(struct loongson_card_data *data) +{ + struct snd_soc_card *card = &data->snd_card; + struct fwnode_handle *fwnode = card->dev->fwnode; + struct fwnode_reference_args args; + const char *codec_dai_name; + struct acpi_device *adev; + struct device *phy_dev; + int ret, i; + + /* fixup platform name based on reference node */ + memset(&args, 0, sizeof(args)); + ret = acpi_node_get_property_reference(fwnode, "cpu", 0, &args); + if (ACPI_FAILURE(ret) || !is_acpi_device_node(args.fwnode)) { + dev_err(card->dev, "No matching phy in ACPI table\n"); + return ret; + } + adev = to_acpi_device_node(args.fwnode); + phy_dev = acpi_get_first_physical_node(adev); + if (!phy_dev) + return -EPROBE_DEFER; + for (i = 0; i < card->num_links; i++) + loongson_dai_links[i].platforms->name = dev_name(phy_dev); + + /* fixup codec name based on reference node */ + memset(&args, 0, sizeof(args)); + ret = acpi_node_get_property_reference(fwnode, "codec", 0, &args); + if (ACPI_FAILURE(ret) || !is_acpi_device_node(args.fwnode)) { + dev_err(card->dev, "No matching phy in ACPI table\n"); + return ret; + } + adev = to_acpi_device_node(args.fwnode); + snprintf(codec_name, sizeof(codec_name), "i2c-%s", acpi_dev_name(adev)); + for (i = 0; i < card->num_links; i++) + loongson_dai_links[i].codecs->name = codec_name; + + device_property_read_string(card->dev, "codec-dai-name", + &codec_dai_name); + for (i = 0; i < card->num_links; i++) + loongson_dai_links[i].codecs->dai_name = codec_dai_name; + + return 0; +} + +static int loongson_card_parse_of(struct loongson_card_data *data) +{ + const char *cpu_dai_name, *codec_dai_name; + struct device_node *cpu, *codec; + struct snd_soc_card *card = &data->snd_card; + struct device *dev = card->dev; + struct of_phandle_args args; + int ret, i; + + cpu = of_get_child_by_name(dev->of_node, "cpu"); + if (!cpu) { + dev_err(dev, "platform property missing or invalid\n"); + return -EINVAL; + } + codec = of_get_child_by_name(dev->of_node, "codec"); + if (!codec) { + dev_err(dev, "audio-codec property missing or invalid\n"); + ret = -EINVAL; + goto err; + } + + ret = of_parse_phandle_with_args(cpu, "sound-dai", + "#sound-dai-cells", 0, &args); + if (ret) { + dev_err(dev, "codec node missing #sound-dai-cells\n"); + goto err; + } + for (i = 0; i < card->num_links; i++) + loongson_dai_links[i].cpus->of_node = args.np; + + ret = of_parse_phandle_with_args(codec, "sound-dai", + "#sound-dai-cells", 0, &args); + if (ret) { + dev_err(dev, "codec node missing #sound-dai-cells\n"); + goto err; + } + for (i = 0; i < card->num_links; i++) + loongson_dai_links[i].codecs->of_node = args.np; + + snd_soc_of_get_dai_name(cpu, &cpu_dai_name); + snd_soc_of_get_dai_name(codec, &codec_dai_name); + for (i = 0; i < card->num_links; i++) { + loongson_dai_links[i].cpus->dai_name = cpu_dai_name; + loongson_dai_links[i].codecs->dai_name = codec_dai_name; + } + of_node_put(cpu); + of_node_put(codec); + + return 0; + +err: + of_node_put(cpu); + of_node_put(codec); + return ret; +} + +static int loongson_asoc_card_probe(struct platform_device *pdev) +{ + struct loongson_card_data *ls_priv; + struct snd_soc_card *card; + int ret; + + ls_priv = devm_kzalloc(&pdev->dev, sizeof(*ls_priv), GFP_KERNEL); + if (!ls_priv) + return -ENOMEM; + + card = &ls_priv->snd_card; + + card->dev = &pdev->dev; + card->owner = THIS_MODULE; + card->dai_link = loongson_dai_links; + card->num_links = ARRAY_SIZE(loongson_dai_links); + snd_soc_card_set_drvdata(card, ls_priv); + + ret = device_property_read_string(&pdev->dev, "model", &card->name); + if (ret) { + dev_err(&pdev->dev, "Error parsing card name: %d\n", ret); + return ret; + } + ret = device_property_read_u32(&pdev->dev, "mclk-fs", &ls_priv->mclk_fs); + if (ret) { + dev_err(&pdev->dev, "Error parsing mclk-fs: %d\n", ret); + return ret; + } + + if (has_acpi_companion(&pdev->dev)) + ret = loongson_card_parse_acpi(ls_priv); + else + ret = loongson_card_parse_of(ls_priv); + if (ret < 0) + return ret; + + ret = devm_snd_soc_register_card(&pdev->dev, card); + + return ret; +} + +static const struct of_device_id loongson_asoc_dt_ids[] = { + { .compatible = "loongson,ls-audio-card" }, + { /* sentinel */ }, +}; +MODULE_DEVICE_TABLE(of, loongson_asoc_dt_ids); + +static struct platform_driver loongson_audio_driver = { + .probe = loongson_asoc_card_probe, + .driver = { + .name = "loongson-asoc-card", + .pm = &snd_soc_pm_ops, + .of_match_table = of_match_ptr(loongson_asoc_dt_ids), + }, +}; +module_platform_driver(loongson_audio_driver); + +MODULE_DESCRIPTION("Loongson ASoc Sound Card driver"); +MODULE_AUTHOR("Loongson Technology Corporation Limited"); +MODULE_LICENSE("GPL"); -- 2.33.0

2 2

[ PATCH v2 1/3] ASoC: Add support for Loongson I2S controller
by YingKun Meng 14 Jun '23

14 Jun '23

From: Yingkun Meng <mengyingkun(a)loongson.cn> Loongson I2S controller is found on 7axxx/2kxxx chips from loongson, it is a PCI device with two private DMA controllers, one for playback, the other for capture. The driver supports the use of DTS or ACPI to describe device resources. Signed-off-by: Yingkun Meng <mengyingkun(a)loongson.cn> Reported-by: kernel test robot <lkp(a)intel.com> Closes: https://lore.kernel.org/oe-kbuild-all/202306060223.9hdivLrx-lkp@intel.com/ Closes: https://lore.kernel.org/oe-kbuild-all/202306060320.Sphw0ihy-lkp@intel.com/ --- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/loongson/Kconfig | 16 + sound/soc/loongson/Makefile | 4 + sound/soc/loongson/loongson_i2s.c | 213 +++++++++++ sound/soc/loongson/loongson_i2s.h | 70 ++++ sound/soc/loongson/loongson_i2s_pci.c | 500 ++++++++++++++++++++++++++ 7 files changed, 805 insertions(+) create mode 100644 sound/soc/loongson/Kconfig create mode 100644 sound/soc/loongson/Makefile create mode 100644 sound/soc/loongson/loongson_i2s.c create mode 100644 sound/soc/loongson/loongson_i2s.h create mode 100644 sound/soc/loongson/loongson_i2s_pci.c diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 4b6e5a802880..bfa9622e1ab1 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -79,6 +79,7 @@ source "sound/soc/google/Kconfig" source "sound/soc/hisilicon/Kconfig" source "sound/soc/jz4740/Kconfig" source "sound/soc/kirkwood/Kconfig" +source "sound/soc/loongson/Kconfig" source "sound/soc/img/Kconfig" source "sound/soc/intel/Kconfig" source "sound/soc/mediatek/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 9d9b228e4508..8376fdb217ed 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -46,6 +46,7 @@ obj-$(CONFIG_SND_SOC) += fsl/ obj-$(CONFIG_SND_SOC) += google/ obj-$(CONFIG_SND_SOC) += hisilicon/ obj-$(CONFIG_SND_SOC) += jz4740/ +obj-$(CONFIG_SND_SOC) += loongson/ obj-$(CONFIG_SND_SOC) += img/ obj-$(CONFIG_SND_SOC) += intel/ obj-$(CONFIG_SND_SOC) += mediatek/ diff --git a/sound/soc/loongson/Kconfig b/sound/soc/loongson/Kconfig new file mode 100644 index 000000000000..4478ac91e402 --- /dev/null +++ b/sound/soc/loongson/Kconfig @@ -0,0 +1,16 @@ +# SPDX-License-Identifier: GPL-2.0 +menu "SoC Audio for Loongson CPUs" + depends on LOONGARCH || COMPILE_TEST + +config SND_SOC_LOONGSON_I2S_PCI + tristate "Loongson I2S-PCI Device Driver" + select REGMAP_MMIO + depends on PCI + help + Say Y or M if you want to add support for I2S driver for + Loongson I2S controller. + + The controller is found in loongson bridge chips or SoCs, + and work as a PCI device. + +endmenu diff --git a/sound/soc/loongson/Makefile b/sound/soc/loongson/Makefile new file mode 100644 index 000000000000..cfd0de1b1b22 --- /dev/null +++ b/sound/soc/loongson/Makefile @@ -0,0 +1,4 @@ +# SPDX-License-Identifier: GPL-2.0 +#Platform Support +snd-soc-loongson-i2s-pci-objs := loongson_i2s_pci.o loongson_i2s.o +obj-$(CONFIG_SND_SOC_LOONGSON_I2S_PCI) += snd-soc-loongson-i2s-pci.o diff --git a/sound/soc/loongson/loongson_i2s.c b/sound/soc/loongson/loongson_i2s.c new file mode 100644 index 000000000000..31e7cf93ee6e --- /dev/null +++ b/sound/soc/loongson/loongson_i2s.c @@ -0,0 +1,213 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Loongson-2K I2S master mode driver + * + * Copyright (C) 2022 Loongson Technology Corporation Limited + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/delay.h> +#include <linux/pm_runtime.h> +#include <linux/dma-mapping.h> +#include <sound/soc.h> +#include <linux/regmap.h> +#include <sound/pcm_params.h> +#include "loongson_i2s.h" + +#define LOONGSON_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +static int loongson_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct loongson_i2s *i2s = snd_soc_dai_get_drvdata(dai); + u32 val; + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + /* Enable MCLK */ + if (i2s->rev_id == 1) { + regmap_update_bits(i2s->regmap, LS_I2S_CTRL, + I2S_CTRL_MCLK_EN, + I2S_CTRL_MCLK_EN); + ret = regmap_read_poll_timeout_atomic(i2s->regmap, + LS_I2S_CTRL, val, + val & I2S_CTRL_MCLK_READY, + 10, 2000); + if (ret < 0) + dev_warn(dai->dev, "wait MCLK ready timeout\n"); + } + + /* Enable master mode */ + regmap_update_bits(i2s->regmap, LS_I2S_CTRL, I2S_CTRL_MASTER, + I2S_CTRL_MASTER); + if (i2s->rev_id == 1) { + ret = regmap_read_poll_timeout_atomic(i2s->regmap, + LS_I2S_CTRL, val, + val & I2S_CTRL_CLK_READY, + 10, 2000); + if (ret < 0) + dev_warn(dai->dev, "wait BCLK ready timeout\n"); + } + fallthrough; + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + regmap_update_bits(i2s->regmap, LS_I2S_CTRL, + I2S_CTRL_TX_EN | I2S_CTRL_TX_DMA_EN, + I2S_CTRL_TX_EN | I2S_CTRL_TX_DMA_EN); + else + regmap_update_bits(i2s->regmap, LS_I2S_CTRL, + I2S_CTRL_RX_EN | I2S_CTRL_RX_DMA_EN, + I2S_CTRL_RX_EN | I2S_CTRL_RX_DMA_EN); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + regmap_update_bits(i2s->regmap, LS_I2S_CTRL, + I2S_CTRL_TX_EN | I2S_CTRL_TX_DMA_EN, 0); + else + regmap_update_bits(i2s->regmap, LS_I2S_CTRL, + I2S_CTRL_RX_EN | I2S_CTRL_RX_DMA_EN, 0); + break; + default: + ret = -EINVAL; + } + + return ret; +} + +static int loongson_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct loongson_i2s *i2s = snd_soc_dai_get_drvdata(dai); + u32 clk_rate = i2s->clk_rate; + u32 sysclk = i2s->sysclk; + u32 bits = params_width(params); + u32 chans = params_channels(params); + u32 fs = params_rate(params); + u32 bclk_ratio, mclk_ratio; + u32 mclk_ratio_frac; + u32 val = 0; + + if (i2s->rev_id == 0) { + bclk_ratio = DIV_ROUND_CLOSEST(clk_rate, + (bits * chans * fs * 2)) - 1; + mclk_ratio = DIV_ROUND_CLOSEST(clk_rate, (sysclk * 2)) - 1; + + /* According to 2k1000LA user manual, set bits == depth */ + val |= (bits << 24); + val |= (bits << 16); + val |= (bclk_ratio << 8); + val |= mclk_ratio; + regmap_write(i2s->regmap, LS_I2S_CFG, val); + } else if (i2s->rev_id == 1) { + bclk_ratio = DIV_ROUND_CLOSEST(sysclk, + (bits * chans * fs * 2)) - 1; + mclk_ratio = clk_rate / sysclk; + mclk_ratio_frac = DIV_ROUND_CLOSEST(((u64)clk_rate << 16), + sysclk) - (mclk_ratio << 16); + + regmap_read(i2s->regmap, LS_I2S_CFG, &val); + val |= (bits << 24); + val |= (bclk_ratio << 8); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + val |= (bits << 16); + else + val |= bits; + regmap_write(i2s->regmap, LS_I2S_CFG, val); + + val = (mclk_ratio_frac << 16) | mclk_ratio; + regmap_write(i2s->regmap, LS_I2S_CFG1, val); + } else + dev_err(i2s->dev, "I2S revision invalid\n"); + + return 0; +} + +static int loongson_i2s_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct loongson_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + i2s->sysclk = freq; + + return 0; +} + +static const struct snd_soc_dai_ops loongson_i2s_dai_ops = { + .trigger = loongson_i2s_trigger, + .hw_params = loongson_i2s_hw_params, + .set_sysclk = loongson_i2s_set_dai_sysclk, +}; + +static int loongson_i2s_dai_probe(struct snd_soc_dai *cpu_dai) +{ + struct loongson_i2s *i2s = dev_get_drvdata(cpu_dai->dev); + + snd_soc_dai_init_dma_data(cpu_dai, &i2s->playback_dma_data, + &i2s->capture_dma_data); + snd_soc_dai_set_drvdata(cpu_dai, i2s); + + return 0; +} + +struct snd_soc_dai_driver loongson_i2s_dai = { + .name = "loongson-i2s", + .probe = loongson_i2s_dai_probe, + .playback = { + .stream_name = "CPU-Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = LOONGSON_I2S_FORMATS, + }, + .capture = { + .stream_name = "CPU-Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = LOONGSON_I2S_FORMATS, + }, + .ops = &loongson_i2s_dai_ops, + .symmetric_rate = 1, +}; + +static int i2s_suspend(struct device *dev) +{ + struct loongson_i2s *i2s = dev_get_drvdata(dev); + + regcache_cache_only(i2s->regmap, true); + + return 0; +} + +static int i2s_resume(struct device *dev) +{ + struct loongson_i2s *i2s = dev_get_drvdata(dev); + int ret; + + regcache_cache_only(i2s->regmap, false); + regcache_mark_dirty(i2s->regmap); + ret = regcache_sync(i2s->regmap); + + return ret; +} + +const struct dev_pm_ops loongson_i2s_pm = { + SET_SYSTEM_SLEEP_PM_OPS(i2s_suspend, i2s_resume) +}; + +void loongson_i2s_init(struct loongson_i2s *i2s) +{ + if (i2s->rev_id == 1) { + regmap_write(i2s->regmap, LS_I2S_CTRL, I2S_CTRL_RESET); + udelay(200); + } +} diff --git a/sound/soc/loongson/loongson_i2s.h b/sound/soc/loongson/loongson_i2s.h new file mode 100644 index 000000000000..7735eadb3bf3 --- /dev/null +++ b/sound/soc/loongson/loongson_i2s.h @@ -0,0 +1,70 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * ALSA I2S interface for the Loongson chip + * + */ +#ifndef _LOONGSON_I2S_H +#define _LOONGSON_I2S_H + +#include <linux/regmap.h> +#include <sound/dmaengine_pcm.h> + +/* I2S Common Registers */ +#define LS_I2S_VER 0x00 /* I2S Version */ +#define LS_I2S_CFG 0x04 /* I2S Config */ +#define LS_I2S_CTRL 0x08 /* I2S Control */ +#define LS_I2S_RX_DATA 0x0C /* I2S DMA RX Address */ +#define LS_I2S_TX_DATA 0x10 /* I2S DMA TX Address */ + +/* 2K2000 I2S Specify Registers */ +#define LS_I2S_CFG1 0x14 /* I2S Config1 */ + +/* 7A2000 I2S Specify Registers */ +#define LS_I2S_TX_ORDER 0x100 /* TX DMA Order */ +#define LS_I2S_RX_ORDER 0x110 /* RX DMA Order */ + +/* Loongson I2S Control Register */ +#define I2S_CTRL_MCLK_READY (1 << 16) /* MCLK ready */ +#define I2S_CTRL_MASTER (1 << 15) /* Master mode */ +#define I2S_CTRL_MSB (1 << 14) /* MSB bit order */ +#define I2S_CTRL_RX_EN (1 << 13) /* RX enable */ +#define I2S_CTRL_TX_EN (1 << 12) /* TX enable */ +#define I2S_CTRL_RX_DMA_EN (1 << 11) /* DMA RX enable */ +#define I2S_CTRL_CLK_READY (1 << 8) /* BCLK ready */ +#define I2S_CTRL_TX_DMA_EN (1 << 7) /* DMA TX enable */ +#define I2S_CTRL_RESET (1 << 4) /* Controller soft reset */ +#define I2S_CTRL_MCLK_EN (1 << 3) /* Enable MCLK */ +#define I2S_CTRL_RX_INT_EN (1 << 1) /* RX interrupt enable */ +#define I2S_CTRL_TX_INT_EN (1 << 0) /* TX interrupt enable */ + +#define LS_I2S_DRVNAME "loongson-i2s" + +struct loongson_dma_data { + dma_addr_t dev_addr; /* device physical address for DMA */ + void __iomem *order_addr; /* DMA order register */ + u32 irq; /* DMA irq */ +}; + +struct loongson_i2s { + struct device *dev; + union { + struct snd_dmaengine_dai_dma_data playback_dma_data; + struct loongson_dma_data tx_dma_data; + }; + union { + struct snd_dmaengine_dai_dma_data capture_dma_data; + struct loongson_dma_data rx_dma_data; + }; + struct regmap *regmap; + void __iomem *reg_base; + u32 rev_id; + u32 clk_rate; + u32 sysclk; +}; + +extern const struct dev_pm_ops loongson_i2s_pm; +extern struct snd_soc_dai_driver loongson_i2s_dai; + +void loongson_i2s_init(struct loongson_i2s *i2s); + +#endif diff --git a/sound/soc/loongson/loongson_i2s_pci.c b/sound/soc/loongson/loongson_i2s_pci.c new file mode 100644 index 000000000000..f09713b560e9 --- /dev/null +++ b/sound/soc/loongson/loongson_i2s_pci.c @@ -0,0 +1,500 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Loongson-2K I2S master mode driver + * + * Copyright (C) 2022 Loongson Technology Corporation Limited + */ + +#include <linux/module.h> +#include <linux/io-64-nonatomic-lo-hi.h> +#include <linux/delay.h> +#include <linux/pm_runtime.h> +#include <linux/dma-mapping.h> +#include <linux/acpi.h> +#include <linux/pci.h> +#include <sound/soc.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include "loongson_i2s.h" + +/* DMA dma_order Register */ +#define DMA_ORDER_STOP (1 << 4) /* DMA stop */ +#define DMA_ORDER_START (1 << 3) /* DMA start */ +#define DMA_ORDER_ASK_VALID (1 << 2) /* DMA ask valid flag */ +#define DMA_ORDER_AXI_UNCO (1 << 1) /* Uncache access */ +#define DMA_ORDER_ADDR_64 (1 << 0) /* 64bits address support */ + +#define DMA_ORDER_ASK_MASK (~0x1fUL) /* Ask addr mask */ +#define DMA_ORDER_CTRL_MASK (0x0fUL) /* Control mask */ + +#define BAR_NUM 0 + +/* + * DMA registers descriptor. + */ +struct loongson_dma_desc { + u32 order; /* Next descriptor address register */ + u32 saddr; /* Source address register */ + u32 daddr; /* Device address register */ + u32 length; /* Total length register */ + u32 step_length; /* Memory stride register */ + u32 step_times; /* Repeat time register */ + u32 cmd; /* Command register */ + u32 stats; /* Status register */ + u32 order_hi; /* Next descriptor high address register */ + u32 saddr_hi; /* High source address register */ + u32 res[6]; /* Reserved */ +}; + +struct loongson_runtime_data { + struct loongson_dma_data *dma_data; + + struct loongson_dma_desc *dma_desc_arr; + dma_addr_t dma_desc_arr_phy; + + struct loongson_dma_desc *dma_pos_desc; + dma_addr_t dma_pos_desc_phy; +}; + +static const struct snd_pcm_hardware ls_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_PAUSE, + .formats = (SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | + SNDRV_PCM_FMTBIT_S24_LE), + .period_bytes_min = 128, + .period_bytes_max = 128 * 1024, + .periods_min = 1, + .periods_max = PAGE_SIZE / sizeof(struct loongson_dma_desc), + .buffer_bytes_max = 1024 * 1024, +}; + +static struct +loongson_dma_desc *dma_desc_save(struct loongson_runtime_data *prtd) +{ + void __iomem *order_reg = prtd->dma_data->order_addr; + u64 val; + + val = (u64)prtd->dma_pos_desc_phy & DMA_ORDER_ASK_MASK; + val |= (readq(order_reg) & DMA_ORDER_CTRL_MASK); + val |= DMA_ORDER_ASK_VALID; + writeq(val, order_reg); + + while (readl(order_reg) & DMA_ORDER_ASK_VALID) + udelay(2); + + return prtd->dma_pos_desc; +} + +static int loongson_pcm_trigger(struct snd_soc_component *component, + struct snd_pcm_substream *substream, int cmd) +{ + struct loongson_runtime_data *prtd = substream->runtime->private_data; + struct device *dev = substream->pcm->card->dev; + void __iomem *order_reg = prtd->dma_data->order_addr; + u64 val; + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + val = prtd->dma_pos_desc_phy & DMA_ORDER_ASK_MASK; + if (dev->coherent_dma_mask == DMA_BIT_MASK(64)) + val |= DMA_ORDER_ADDR_64; + else + val &= ~DMA_ORDER_ADDR_64; + val |= (readq(order_reg) & DMA_ORDER_CTRL_MASK); + val |= DMA_ORDER_START; + writeq(val, order_reg); + + while ((readl(order_reg) & DMA_ORDER_START)) + udelay(2); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + dma_desc_save(prtd); + + /* dma stop */ + val = readq(order_reg) | DMA_ORDER_STOP; + writeq(val, order_reg); + udelay(1000); + + break; + default: + ret = -EINVAL; + } + + return ret; +} + +static int loongson_pcm_hw_params(struct snd_soc_component *component, + struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct device *dev = component->dev; + struct loongson_runtime_data *prtd = runtime->private_data; + size_t buf_len = params_buffer_bytes(params); + size_t period_len = params_period_bytes(params); + dma_addr_t order_addr, mem_addr; + struct loongson_dma_desc *desc; + u32 num_periods; + int i; + + if (buf_len % period_len) { + dev_err(dev, "buf len not multiply of period len\n"); + return -EINVAL; + } + + num_periods = buf_len / period_len; + if (!num_periods) { + dev_err(dev, "dma data too small\n"); + return -EINVAL; + } + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = buf_len; + + /* initialize dma descriptor array */ + mem_addr = runtime->dma_addr; + order_addr = prtd->dma_desc_arr_phy; + for (i = 0; i < num_periods; i++) { + desc = &prtd->dma_desc_arr[i]; + + /* next descriptor physical address */ + order_addr += sizeof(*desc); + desc->order = lower_32_bits(order_addr | BIT(0)); + desc->order_hi = upper_32_bits(order_addr); + + desc->saddr = lower_32_bits(mem_addr); + desc->saddr_hi = upper_32_bits(mem_addr); + desc->daddr = prtd->dma_data->dev_addr; + + desc->cmd = BIT(0); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + desc->cmd |= BIT(12); + + desc->length = period_len >> 2; + desc->step_length = 0; + desc->step_times = 1; + + mem_addr += period_len; + } + if (num_periods > 0) { + desc = &prtd->dma_desc_arr[num_periods - 1]; + desc->order = lower_32_bits(prtd->dma_desc_arr_phy | BIT(0)); + desc->order_hi = upper_32_bits(prtd->dma_desc_arr_phy); + } + + /* init position descriptor */ + *prtd->dma_pos_desc = *prtd->dma_desc_arr; + + return 0; +} + +static snd_pcm_uframes_t +loongson_pcm_pointer(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct loongson_runtime_data *prtd = runtime->private_data; + struct loongson_dma_desc *desc; + snd_pcm_uframes_t x; + u64 addr; + + desc = dma_desc_save(prtd); + addr = ((u64)desc->saddr_hi << 32) | desc->saddr; + + x = bytes_to_frames(runtime, addr - runtime->dma_addr); + if (x == runtime->buffer_size) + x = 0; + return x; +} + +static irqreturn_t loongson_pcm_dma_irq(int irq, void *devid) +{ + struct snd_pcm_substream *substream = devid; + + snd_pcm_period_elapsed(substream); + return IRQ_HANDLED; +} + +static int loongson_pcm_open(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_card *card = substream->pcm->card; + struct loongson_runtime_data *prtd; + struct loongson_dma_data *dma_data; + int ret; + + /* + * For mysterious reasons (and despite what the manual says) + * playback samples are lost if the DMA count is not a multiple + * of the DMA burst size. Let's add a rule to enforce that. + */ + snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 128); + snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 128); + snd_pcm_hw_constraint_integer(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + snd_soc_set_runtime_hwparams(substream, &ls_pcm_hardware); + + prtd = kzalloc(sizeof(*prtd), GFP_KERNEL); + if (!prtd) + return -ENOMEM; + + prtd->dma_desc_arr = dma_alloc_coherent(card->dev, PAGE_SIZE, + &prtd->dma_desc_arr_phy, + GFP_KERNEL); + if (!prtd->dma_desc_arr) { + ret = -ENOMEM; + goto desc_err; + } + + prtd->dma_pos_desc = dma_alloc_coherent(card->dev, + sizeof(*prtd->dma_pos_desc), + &prtd->dma_pos_desc_phy, + GFP_KERNEL); + if (!prtd->dma_pos_desc) { + ret = -ENOMEM; + goto pos_err; + } + + dma_data = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); + ret = request_irq(dma_data->irq, loongson_pcm_dma_irq, + IRQF_TRIGGER_HIGH, LS_I2S_DRVNAME, substream); + if (ret < 0) + goto irq_err; + + prtd->dma_data = dma_data; + substream->runtime->private_data = prtd; + + return 0; +irq_err: + dma_free_coherent(card->dev, sizeof(*prtd->dma_pos_desc), + prtd->dma_pos_desc, prtd->dma_pos_desc_phy); +pos_err: + dma_free_coherent(card->dev, PAGE_SIZE, prtd->dma_desc_arr, + prtd->dma_desc_arr_phy); +desc_err: + kfree(prtd); + + return ret; +} + +static int loongson_pcm_close(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_card *card = substream->pcm->card; + struct loongson_runtime_data *prtd = substream->runtime->private_data; + struct loongson_dma_data *dma_data; + + dma_free_coherent(card->dev, PAGE_SIZE, prtd->dma_desc_arr, + prtd->dma_desc_arr_phy); + + dma_free_coherent(card->dev, sizeof(*prtd->dma_pos_desc), + prtd->dma_pos_desc, prtd->dma_pos_desc_phy); + + dma_data = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); + free_irq(dma_data->irq, substream); + + kfree(prtd); + return 0; +} + +static int loongson_pcm_mmap(struct snd_soc_component *component, + struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + return remap_pfn_range(vma, vma->vm_start, + substream->dma_buffer.addr >> PAGE_SHIFT, + vma->vm_end - vma->vm_start, vma->vm_page_prot); +} + +static int loongson_pcm_new(struct snd_soc_component *component, + struct snd_soc_pcm_runtime *rtd) +{ + struct snd_pcm *pcm = rtd->pcm; + + return snd_pcm_set_fixed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV, + pcm->card->dev, + ls_pcm_hardware.buffer_bytes_max); +} + +static const struct snd_soc_component_driver loongson_i2s_component = { + .name = LS_I2S_DRVNAME, + .open = loongson_pcm_open, + .close = loongson_pcm_close, + .hw_params = loongson_pcm_hw_params, + .trigger = loongson_pcm_trigger, + .pointer = loongson_pcm_pointer, + .mmap = loongson_pcm_mmap, + .pcm_construct = loongson_pcm_new, +}; + +static bool loongson_i2s_wr_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case LS_I2S_CFG: + case LS_I2S_CTRL: + case LS_I2S_RX_DATA: + case LS_I2S_TX_DATA: + case LS_I2S_CFG1: + return true; + default: + return false; + }; +} + +static bool loongson_i2s_rd_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case LS_I2S_VER: + case LS_I2S_CFG: + case LS_I2S_CTRL: + case LS_I2S_RX_DATA: + case LS_I2S_TX_DATA: + case LS_I2S_CFG1: + return true; + default: + return false; + }; +} + +static const struct regmap_config loongson_i2s_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = LS_I2S_CFG1, + .writeable_reg = loongson_i2s_wr_reg, + .readable_reg = loongson_i2s_rd_reg, + .cache_type = REGCACHE_FLAT, +}; + +static int loongson_i2s_pci_probe(struct pci_dev *pdev, + const struct pci_device_id *pid) +{ + const struct fwnode_handle *fwnode = pdev->dev.fwnode; + struct loongson_dma_data *tx_data, *rx_data; + struct loongson_i2s *i2s; + int ret; + + if (pcim_enable_device(pdev)) { + dev_err(&pdev->dev, "pci_enable_device failed\n"); + return -ENODEV; + } + + ret = pci_request_region(pdev, BAR_NUM, LS_I2S_DRVNAME); + if (ret) { + dev_err(&pdev->dev, "request regions failed %d\n", ret); + return ret; + } + + i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL); + if (!i2s) { + ret = -ENOMEM; + goto err_release; + } + i2s->rev_id = pdev->revision; + i2s->dev = &pdev->dev; + pci_set_drvdata(pdev, i2s); + + i2s->reg_base = pcim_iomap(pdev, BAR_NUM, 0); + if (!i2s->reg_base) { + dev_err(&pdev->dev, "pci_iomap_error\n"); + ret = -EIO; + goto err_release; + } + + i2s->regmap = devm_regmap_init_mmio(&pdev->dev, i2s->reg_base, + &loongson_i2s_regmap_config); + if (IS_ERR(i2s->regmap)) { + dev_err(&pdev->dev, "Failed to initialize register map"); + ret = PTR_ERR(i2s->regmap); + goto err_release; + } + + tx_data = &i2s->tx_dma_data; + rx_data = &i2s->rx_dma_data; + + tx_data->dev_addr = (dma_addr_t)i2s->reg_base + LS_I2S_TX_DATA; + tx_data->order_addr = i2s->reg_base + LS_I2S_TX_ORDER; + + rx_data->dev_addr = (dma_addr_t)i2s->reg_base + LS_I2S_RX_DATA; + rx_data->order_addr = i2s->reg_base + LS_I2S_RX_ORDER; + + tx_data->irq = fwnode_irq_get_byname(fwnode, "tx"); + if (tx_data->irq < 0) { + dev_err(&pdev->dev, "dma tx irq invalid\n"); + ret = tx_data->irq; + goto err_release; + } + + rx_data->irq = fwnode_irq_get_byname(fwnode, "rx"); + if (rx_data->irq < 0) { + dev_err(&pdev->dev, "dma rx irq invalid\n"); + ret = rx_data->irq; + goto err_release; + } + + device_property_read_u32(&pdev->dev, "clock-frequency", &i2s->clk_rate); + if (!i2s->clk_rate) { + dev_err(&pdev->dev, "clock-frequency property invalid\n"); + ret = -EINVAL; + goto err_release; + } + + dma_set_mask_and_coherent(&pdev->dev, DMA_BIT_MASK(64)); + + loongson_i2s_init(i2s); + + ret = devm_snd_soc_register_component(&pdev->dev, + &loongson_i2s_component, + &loongson_i2s_dai, 1); + if (ret) { + dev_err(&pdev->dev, "register DAI failed %d\n", ret); + goto err_release; + } + + return 0; +err_release: + pci_release_region(pdev, BAR_NUM); + return ret; +} + +static void loongson_i2s_pci_remove(struct pci_dev *pdev) +{ + pci_release_region(pdev, BAR_NUM); +} + +static const struct pci_device_id loongson_i2s_ids[] = { + { PCI_DEVICE(PCI_VENDOR_ID_LOONGSON, 0x7a27) }, + { }, +}; +MODULE_DEVICE_TABLE(pci, loongson_i2s_ids); + +static struct pci_driver loongson_i2s_driver = { + .name = "loongson-i2s-pci", + .id_table = loongson_i2s_ids, + .probe = loongson_i2s_pci_probe, + .remove = loongson_i2s_pci_remove, + .driver = { + .owner = THIS_MODULE, + .pm = pm_sleep_ptr(&loongson_i2s_pm), + }, +}; +module_pci_driver(loongson_i2s_driver); + +MODULE_DESCRIPTION("Loongson I2S Master Mode ASoC Driver"); +MODULE_AUTHOR("Loongson Technology Corporation Limited"); +MODULE_LICENSE("GPL"); -- 2.33.0

2 2

Re: [PATCH 6/7] ASoC: dt-bindings: mediatek,mt79xx-wm8960: add mt79xx-wm8960 document
by Krzysztof Kozlowski 14 Jun '23

14 Jun '23

On 14/06/2023 09:40, Maso Huang (黃加竹) wrote: >>> Signed-off-by: Maso Huang <maso.huang(a)mediatek.com> >>> --- >>> .../sound/mediatek,mt79xx-wm8960.yaml | 53 >> +++++++++++++++++++ >>> 1 file changed, 53 insertions(+) >>> create mode 100644 >> Documentation/devicetree/bindings/sound/mediatek,mt79xx-wm8960.yaml >>> >>> diff --git >> a/Documentation/devicetree/bindings/sound/mediatek,mt79xx-wm8960.yaml >> b/Documentation/devicetree/bindings/sound/mediatek,mt79xx-wm8960.yaml >>> new file mode 100644 >>> index 000000000000..26b38bb629da >>> --- /dev/null >>> +++ b/Documentation/devicetree/bindings/sound/mediatek,mt79xx- >> wm8960.yaml >>> @@ -0,0 +1,53 @@ >>> +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) >>> +%YAML 1.2 >>> +--- >>> +$id: >> http://devicetree.org/schemas/sound/mediatek,mt79xx-wm8960.yaml# >>> +$schema: http://devicetree.org/meta-schemas/core.yaml# >>> + >>> +title: MediaTek MT79xx ASOC sound card with WM8960 codec >> >> What is a MT79xx ASOC? Is it some specific SoC name? What does "A" >> stands for in SoC? XX also looks odd, I thought Mediatek uses only >> numbers. >> > > I'll use mt7986 instead of mt79xx in v2 patch. > And ASoC means ALSA SoC. ALSA is Linux stuff, so does not belong to bindings. I have no clue what is "ALSA SoC" (as SoC means System on Chip). Please describe real hardware not Linux or your configuration or some SW. Best regards, Krzysztof

1 0

Re: [PATCH 7/7] ASoC: dt-bindings: mediatek,mt79xx-afe: add audio afe document
by Krzysztof Kozlowski 14 Jun '23

14 Jun '23

On 14/06/2023 09:37, Maso Huang (黃加竹) wrote: >>>> I already saw AFE, why it cannot be part of existing bindings? >> >> Can you answer this? >> > > Did you mean mtk-afe-pcm.txt? > If yes, I'll modify mtk-afe-pcm.txt to yaml format, and add mt7986 to > its compatible list. > No, I meant mediatek,mt8188-afe.yaml. Aren't you working on some old tree? If so, please don't... Best regards, Krzysztof

1 0

Re: [PATCH 7/7] ASoC: dt-bindings: mediatek,mt79xx-afe: add audio afe document
by Krzysztof Kozlowski 14 Jun '23

14 Jun '23

On 14/06/2023 05:17, Maso Huang (黃加竹) wrote: > On Tue, 2023-06-13 at 10:51 +0200, Krzysztof Kozlowski wrote: >> >> External email : Please do not click links or open attachments until >> you have verified the sender or the content. >> >> On 12/06/2023 12:52, Maso Hunag wrote: >>> From: Maso Huang <maso.huang(a)mediatek.com> >>> >>> Add mt79xx audio afe document. >> >> Please use scripts/get_maintainers.pl to get a list of necessary >> people >> and lists to CC. It might happen, that command when run on an older >> kernel, gives you outdated entries. Therefore please be sure you >> base >> your patches on recent Linux kernel. >> > > Hi Krzysztif, > > Thanks for your review. And sorry for missing some necessary > maintainers. > What's your suggestion, resend these patches again with them, or add > them back in v2 patch? You need to fix the patch anyway, so use get_maintainers.pl in v2. I don't understand why you Cc here many unrelated people but not the actual maintainers which get_maintainers.pl asks you to Cc! > >>> >>> Signed-off-by: Maso Huang <maso.huang(a)mediatek.com> >>> --- >>> .../bindings/sound/mediatek,mt79xx-afe.yaml | 102 >> ++++++++++++++++++ >>> 1 file changed, 102 insertions(+) >>> create mode 100644 >> Documentation/devicetree/bindings/sound/mediatek,mt79xx-afe.yaml >>> >>> diff --git >> a/Documentation/devicetree/bindings/sound/mediatek,mt79xx-afe.yaml >> b/Documentation/devicetree/bindings/sound/mediatek,mt79xx-afe.yaml >>> new file mode 100644 >>> index 000000000000..11ef1cfdf49b >>> --- /dev/null >>> +++ b/Documentation/devicetree/bindings/sound/mediatek,mt79xx- >> afe.yaml >>> @@ -0,0 +1,102 @@ >>> +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) >>> +%YAML 1.2 >>> +--- >>> +$id: http://devicetree.org/schemas/sound/mediatek,mt79xx-afe.yaml# >>> +$schema: http://devicetree.org/meta-schemas/core.yaml# >>> + >>> +title: MediaTek AFE PCM controller for MT79xx >> >> 79XX sounds weird. Are you sure you are not using wildcards (which >> are >> not allowed)? >> > > We would like to use mt79xx for mt7986/mt7981/mt7988 series. > Or is it better to just use mt7986 for this series? You cannot use wildcard. Get some internal review of your patches prior to submission to mailing list. https://elixir.bootlin.com/linux/v6.1-rc1/source/Documentation/devicetree/b… > >>> + >>> +maintainers: >>> + - Maso Huang <maso.huang(a)mediatek.com> >>> + >>> +properties: >>> + compatible: >>> + oneOf: >>> + - const: mediatek,mt79xx-afe >>> + - items: >>> + - enum: >>> + - mediatek,mt7981-afe >>> + - mediatek,mt7986-afe >>> + - mediatek,mt7988-afe >>> + - const: mediatek,mt79xx-afe >> >> I already saw AFE, why it cannot be part of existing bindings? Can you answer this? >> >> This list is odd. 79xx, 7981? So it is wildcard? >> > > Yes, it is wildcard for mt7986/mt7981/mt7988 series. > Is it better to just use mt7986 for this series? No wildcards. Best regards, Krzysztof

1 0

[PATCH] ALSA: hda: Use maple tree register cache
by Mark Brown 14 Jun '23

14 Jun '23

HDA can only support single register read and write operations so does not benefit from block writes. This means it gets no benefit from using the rbtree register cache over the maple tree register cache so convert it to use maple trees instead, it is more modern. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/hda/hdac_regmap.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index fe3587547cfe..2caa1f9b858e 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -358,7 +358,7 @@ static const struct regmap_config hda_regmap_cfg = { .writeable_reg = hda_writeable_reg, .readable_reg = hda_readable_reg, .volatile_reg = hda_volatile_reg, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_MAPLE, .reg_read = hda_reg_read, .reg_write = hda_reg_write, .use_single_read = true, --- base-commit: 9561de3a55bed6bdd44a12820ba81ec416e705a7 change-id: 20230609-alsa-hda-maple-02f75ebb067e Best regards, -- Mark Brown <broonie(a)kernel.org>

2 10

bug in snd_usb_audio alsa driver, likely cause for audio freezes
by GitHub issues - opened 13 Jun '23

13 Jun '23

alsa-project/alsa-lib issue #328 was opened from VMichailidis: For the past few days I've been encountering this bug where during media playback audio will stop, a few seconds later the video will stop and allow the audio section to catch up. Also during voice calls and video games audio will occasionally stop. Initially I was using pipewire and jounalctl wasn't reporting any issues, eventually I switched to pulseaudio. The switch didn't fix anything, however journalctl has been reporting the following message: ``` Jun 13 17:18:25 melchior pulseaudio[1347]: ALSA woke us up to write new data to the device, but there was actually nothing to write. Jun 13 17:18:25 melchior pulseaudio[1347]: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Jun 13 17:18:25 melchior pulseaudio[1347]: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. ``` So I resolved to raise the issue here. Here is the rest of the relevant journalctl output ``` journalctl | grep ALSA Jun 07 22:12:14 melchior pulseaudio[44796]: ALSA woke us up to write new data to the device, but there was actually nothing to write. Jun 07 22:12:14 melchior pulseaudio[44796]: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Jun 07 23:41:12 melchior pulseaudio[1273]: ALSA woke us up to write new data to the device, but there was actually nothing to write. Jun 07 23:41:12 melchior pulseaudio[1273]: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Jun 08 01:39:23 melchior pulseaudio[946]: ALSA woke us up to write new data to the device, but there was actually nothing to write. Jun 08 01:39:23 melchior pulseaudio[946]: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Jun 08 18:00:28 melchior pulseaudio[1495]: ALSA woke us up to write new data to the device, but there was actually nothing to write. Jun 08 18:00:28 melchior pulseaudio[1495]: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Jun 09 23:18:45 melchior pulseaudio[1692]: ALSA woke us up to write new data to the device, but there was actually nothing to write. Jun 09 23:18:45 melchior pulseaudio[1692]: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Jun 10 00:10:17 melchior pulseaudio[1516]: ALSA woke us up to write new data to the device, but there was actually nothing to write. Jun 10 00:10:17 melchior pulseaudio[1516]: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Jun 10 15:00:28 melchior pulseaudio[1491]: ALSA woke us up to write new data to the device, but there was actually nothing to write. Jun 10 15:00:28 melchior pulseaudio[1491]: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Jun 11 01:16:33 melchior pulseaudio[1346]: ALSA woke us up to write new data to the device, but there was actually nothing to write. Jun 11 01:16:33 melchior pulseaudio[1346]: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Jun 11 13:30:54 melchior pulseaudio[1334]: ALSA woke us up to write new data to the device, but there was actually nothing to write. Jun 11 13:30:54 melchior pulseaudio[1334]: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Jun 11 17:13:20 melchior pulseaudio[1644]: ALSA woke us up to write new data to the device, but there was actually nothing to write. Jun 11 17:13:20 melchior pulseaudio[1644]: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Jun 12 19:53:14 melchior pulseaudio[1168]: ALSA woke us up to write new data to the device, but there was actually nothing to write. Jun 12 19:53:14 melchior pulseaudio[1168]: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Jun 13 17:18:25 melchior pulseaudio[1347]: ALSA woke us up to write new data to the device, but there was actually nothing to write. Jun 13 17:18:25 melchior pulseaudio[1347]: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. ``` ``` journalctl | grep pulseaudio Jun 07 21:51:03 melchior dbus-daemon[515]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.57' (uid=1000 pid=44796 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 07 21:51:03 melchior pulseaudio[44796]: Failed to open cookie file '/home/basil/.config/pulse/cookie': No such file or directory Jun 07 21:51:03 melchior pulseaudio[44796]: Failed to load authentication key '/home/basil/.config/pulse/cookie': No such file or directory Jun 07 21:51:03 melchior pulseaudio[44796]: stat('/etc/pulse/default.pa.d'): No such file or directory Jun 07 22:12:14 melchior pulseaudio[44796]: ALSA woke us up to write new data to the device, but there was actually nothing to write. Jun 07 22:12:14 melchior pulseaudio[44796]: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Jun 07 22:12:14 melchior pulseaudio[44796]: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Jun 07 22:48:10 melchior dbus-daemon[515]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.143' (uid=1000 pid=44796 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 07 22:48:10 melchior dbus-daemon[515]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.144' (uid=1000 pid=44796 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 07 22:48:10 melchior dbus-daemon[515]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.145' (uid=1000 pid=44796 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 07 22:48:10 melchior dbus-daemon[515]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.146' (uid=1000 pid=44796 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 07 22:48:10 melchior dbus-daemon[515]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.147' (uid=1000 pid=44796 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 07 22:48:10 melchior pulseaudio[44796]: Error opening PCM device iec958:3: No such file or directory Jun 07 22:48:10 melchior systemd[563]: pulseaudio.service: Consumed 14.257s CPU time. Jun 07 22:48:44 melchior dbus-daemon[563]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.18' (uid=1000 pid=1273 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 07 22:48:44 melchior pulseaudio[1273]: Failed to open cookie file '/home/basil/.config/pulse/cookie': No such file or directory Jun 07 22:48:44 melchior pulseaudio[1273]: Failed to load authentication key '/home/basil/.config/pulse/cookie': No such file or directory Jun 07 22:48:44 melchior pulseaudio[1273]: stat('/etc/pulse/default.pa.d'): No such file or directory Jun 07 23:05:43 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_output.pci-0000_03_00.1.hdmi-stereo-extra3" Jun 07 23:16:17 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_input.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.mono-fallback" Jun 07 23:16:17 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_output.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.iec958-stereo.monitor" Jun 07 23:16:40 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_output.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.iec958-stereo.monitor" Jun 07 23:25:05 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_output.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.iec958-stereo" Jun 07 23:25:05 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_output.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.iec958-stereo.monitor" Jun 07 23:25:05 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_output.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.analog-stereo" Jun 07 23:25:05 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_output.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.analog-stereo.monitor" Jun 07 23:25:05 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_output.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.analog-stereo.monitor" Jun 07 23:25:07 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_output.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.analog-stereo" Jun 07 23:25:07 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_output.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.analog-stereo.monitor" Jun 07 23:25:07 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_output.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.iec958-stereo" Jun 07 23:25:07 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_output.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.iec958-stereo.monitor" Jun 07 23:25:07 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_output.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.iec958-stereo.monitor" Jun 07 23:25:10 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_output.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.iec958-stereo" Jun 07 23:25:10 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_output.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.iec958-stereo.monitor" Jun 07 23:25:10 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_output.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.analog-stereo" Jun 07 23:25:10 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_output.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.analog-stereo.monitor" Jun 07 23:25:10 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_output.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.analog-stereo.monitor" Jun 07 23:41:12 melchior pulseaudio[1273]: ALSA woke us up to write new data to the device, but there was actually nothing to write. Jun 07 23:41:12 melchior pulseaudio[1273]: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Jun 07 23:41:12 melchior pulseaudio[1273]: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Jun 07 23:49:49 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_output.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.analog-stereo.monitor" Jun 08 00:05:37 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_output.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.analog-stereo.monitor" Jun 08 00:22:52 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_output.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.analog-stereo" Jun 08 00:22:52 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_output.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.analog-stereo.monitor" Jun 08 00:22:52 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_output.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.iec958-stereo" Jun 08 00:22:52 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_output.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.iec958-stereo.monitor" Jun 08 00:22:52 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_output.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.iec958-stereo.monitor" Jun 08 00:31:33 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_output.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.iec958-stereo" Jun 08 00:31:33 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_output.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.iec958-stereo.monitor" Jun 08 00:31:33 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_output.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.analog-stereo" Jun 08 00:31:33 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_input.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.mono-fallback" Jun 08 00:31:33 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_input.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.mono-fallback" Jun 08 00:31:36 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_input.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.mono-fallback" Jun 08 00:31:36 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_output.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.analog-stereo.monitor" Jun 08 00:34:26 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_output.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.analog-stereo" Jun 08 00:34:26 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_output.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.analog-stereo.monitor" Jun 08 00:34:26 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_output.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.iec958-stereo" Jun 08 00:34:26 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_output.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.iec958-stereo.monitor" Jun 08 00:34:26 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_output.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.iec958-stereo.monitor" Jun 08 01:08:06 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_output.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.iec958-stereo.monitor" Jun 08 01:14:21 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_input.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.mono-fallback" Jun 08 01:14:21 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_output.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.iec958-stereo.monitor" Jun 08 01:14:41 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_output.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.iec958-stereo.monitor" Jun 08 01:38:21 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_output.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.iec958-stereo" Jun 08 01:38:21 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_output.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.iec958-stereo" Jun 08 01:38:21 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_output.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.iec958-stereo" Jun 08 01:38:21 melchior kdeconnectd[824]: org.kde.pulseaudio: No object for name "alsa_input.usb-Elgato_Systems_Elgato_Wave_1_AS31J1A08137-00.mono-fallback" Jun 08 01:38:21 melchior pulseaudio[1273]: Error opening PCM device hw:3: No such file or directory Jun 08 01:38:21 melchior systemd[638]: pulseaudio.service: Consumed 1min 30.529s CPU time. Jun 08 01:38:53 melchior dbus-daemon[491]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.14' (uid=1000 pid=946 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 08 01:38:53 melchior pulseaudio[946]: Failed to open cookie file '/home/basil/.config/pulse/cookie': No such file or directory Jun 08 01:38:53 melchior pulseaudio[946]: Failed to load authentication key '/home/basil/.config/pulse/cookie': No such file or directory Jun 08 01:38:53 melchior pulseaudio[946]: stat('/etc/pulse/default.pa.d'): No such file or directory Jun 08 01:39:23 melchior pulseaudio[946]: ALSA woke us up to write new data to the device, but there was actually nothing to write. Jun 08 01:39:23 melchior pulseaudio[946]: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Jun 08 01:39:23 melchior pulseaudio[946]: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Jun 08 02:01:24 melchior dbus-daemon[491]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.37' (uid=1000 pid=946 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 08 02:01:24 melchior dbus-daemon[491]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.38' (uid=1000 pid=946 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 08 02:01:24 melchior dbus-daemon[491]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.39' (uid=1000 pid=946 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 08 02:01:24 melchior dbus-daemon[491]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.40' (uid=1000 pid=946 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 08 02:01:24 melchior dbus-daemon[491]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.41' (uid=1000 pid=946 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 08 02:01:24 melchior systemd[610]: pulseaudio.service: Consumed 12.550s CPU time. Jun 08 11:45:49 melchior dbus-daemon[516]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.18' (uid=1000 pid=1235 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 08 11:45:50 melchior pulseaudio[1235]: Failed to open cookie file '/home/basil/.config/pulse/cookie': No such file or directory Jun 08 11:45:50 melchior pulseaudio[1235]: Failed to load authentication key '/home/basil/.config/pulse/cookie': No such file or directory Jun 08 11:45:50 melchior pulseaudio[1235]: stat('/etc/pulse/default.pa.d'): No such file or directory Jun 08 12:32:43 melchior dbus-daemon[516]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.36' (uid=1000 pid=1235 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 08 12:32:43 melchior dbus-daemon[516]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.37' (uid=1000 pid=1235 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 08 12:32:43 melchior dbus-daemon[516]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.38' (uid=1000 pid=1235 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 08 12:32:43 melchior dbus-daemon[516]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.39' (uid=1000 pid=1235 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 08 12:32:43 melchior dbus-daemon[516]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.40' (uid=1000 pid=1235 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 08 12:32:43 melchior systemd[606]: pulseaudio.service: Consumed 12.826s CPU time. Jun 08 17:56:45 melchior dbus-daemon[528]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.18' (uid=1000 pid=1495 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 08 17:56:46 melchior pulseaudio[1495]: Failed to open cookie file '/home/basil/.config/pulse/cookie': No such file or directory Jun 08 17:56:46 melchior pulseaudio[1495]: Failed to load authentication key '/home/basil/.config/pulse/cookie': No such file or directory Jun 08 17:56:46 melchior pulseaudio[1495]: stat('/etc/pulse/default.pa.d'): No such file or directory Jun 08 18:00:28 melchior pulseaudio[1495]: ALSA woke us up to write new data to the device, but there was actually nothing to write. Jun 08 18:00:28 melchior pulseaudio[1495]: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Jun 08 18:00:28 melchior pulseaudio[1495]: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Jun 09 02:20:10 melchior dbus-daemon[528]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.111' (uid=1000 pid=1495 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 09 02:20:10 melchior dbus-daemon[528]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.112' (uid=1000 pid=1495 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 09 02:20:10 melchior dbus-daemon[528]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.113' (uid=1000 pid=1495 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 09 02:20:10 melchior dbus-daemon[528]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.114' (uid=1000 pid=1495 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 09 02:20:10 melchior dbus-daemon[528]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.115' (uid=1000 pid=1495 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 09 02:20:10 melchior systemd[616]: pulseaudio.service: Consumed 1min 809ms CPU time. Jun 09 19:31:43 melchior dbus-daemon[496]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.23' (uid=1000 pid=1692 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 09 19:31:44 melchior pulseaudio[1692]: Failed to open cookie file '/home/basil/.config/pulse/cookie': No such file or directory Jun 09 19:31:44 melchior pulseaudio[1692]: Failed to load authentication key '/home/basil/.config/pulse/cookie': No such file or directory Jun 09 19:31:44 melchior pulseaudio[1692]: stat('/etc/pulse/default.pa.d'): No such file or directory Jun 09 23:18:45 melchior pulseaudio[1692]: ALSA woke us up to write new data to the device, but there was actually nothing to write. Jun 09 23:18:45 melchior pulseaudio[1692]: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Jun 09 23:18:45 melchior pulseaudio[1692]: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Jun 10 00:08:24 melchior dbus-daemon[496]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.53' (uid=1000 pid=1692 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 10 00:08:24 melchior dbus-daemon[496]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.54' (uid=1000 pid=1692 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 10 00:08:24 melchior dbus-daemon[496]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.55' (uid=1000 pid=1692 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 10 00:08:24 melchior dbus-daemon[496]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.56' (uid=1000 pid=1692 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 10 00:08:24 melchior dbus-daemon[496]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.57' (uid=1000 pid=1692 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 10 00:08:24 melchior pulseaudio[1692]: Error opening PCM device hw:3: No such file or directory Jun 10 00:08:24 melchior systemd[615]: pulseaudio.service: Consumed 21.103s CPU time. Jun 10 00:09:16 melchior dbus-daemon[505]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.18' (uid=1000 pid=1516 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 10 00:09:16 melchior pulseaudio[1516]: Failed to open cookie file '/home/basil/.config/pulse/cookie': No such file or directory Jun 10 00:09:16 melchior pulseaudio[1516]: Failed to load authentication key '/home/basil/.config/pulse/cookie': No such file or directory Jun 10 00:09:16 melchior pulseaudio[1516]: stat('/etc/pulse/default.pa.d'): No such file or directory Jun 10 00:10:17 melchior pulseaudio[1516]: ALSA woke us up to write new data to the device, but there was actually nothing to write. Jun 10 00:10:17 melchior pulseaudio[1516]: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Jun 10 00:10:17 melchior pulseaudio[1516]: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Jun 10 00:41:02 melchior dbus-daemon[505]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.28' (uid=1000 pid=1516 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 10 00:41:02 melchior dbus-daemon[505]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.29' (uid=1000 pid=1516 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 10 00:41:02 melchior dbus-daemon[505]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.30' (uid=1000 pid=1516 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 10 00:41:02 melchior dbus-daemon[505]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.31' (uid=1000 pid=1516 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 10 00:41:02 melchior dbus-daemon[505]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.32' (uid=1000 pid=1516 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 10 00:41:02 melchior systemd[627]: pulseaudio.service: Consumed 12.940s CPU time. Jun 10 14:47:13 melchior dbus-daemon[472]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.18' (uid=1000 pid=1491 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 10 14:47:13 melchior pulseaudio[1491]: Failed to open cookie file '/home/basil/.config/pulse/cookie': No such file or directory Jun 10 14:47:13 melchior pulseaudio[1491]: Failed to load authentication key '/home/basil/.config/pulse/cookie': No such file or directory Jun 10 14:47:13 melchior pulseaudio[1491]: stat('/etc/pulse/default.pa.d'): No such file or directory Jun 10 15:00:28 melchior pulseaudio[1491]: ALSA woke us up to write new data to the device, but there was actually nothing to write. Jun 10 15:00:28 melchior pulseaudio[1491]: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Jun 10 15:00:28 melchior pulseaudio[1491]: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Jun 10 15:27:53 melchior dbus-daemon[472]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.35' (uid=1000 pid=1491 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 10 15:27:53 melchior dbus-daemon[472]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.36' (uid=1000 pid=1491 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 10 15:27:53 melchior dbus-daemon[472]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.37' (uid=1000 pid=1491 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 10 15:27:53 melchior dbus-daemon[472]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.38' (uid=1000 pid=1491 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 10 15:27:53 melchior dbus-daemon[472]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.39' (uid=1000 pid=1491 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 10 15:27:53 melchior pulseaudio[1491]: Error opening PCM device iec958:3: No such file or directory Jun 10 15:27:53 melchior systemd[618]: pulseaudio.service: Consumed 7.926s CPU time. Jun 10 18:38:08 melchior dbus-daemon[492]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.18' (uid=1000 pid=1304 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 10 18:38:08 melchior pulseaudio[1304]: Failed to open cookie file '/home/basil/.config/pulse/cookie': No such file or directory Jun 10 18:38:08 melchior pulseaudio[1304]: Failed to load authentication key '/home/basil/.config/pulse/cookie': No such file or directory Jun 10 18:38:08 melchior pulseaudio[1304]: stat('/etc/pulse/default.pa.d'): No such file or directory Jun 10 22:55:47 melchior dbus-daemon[492]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.90' (uid=1000 pid=1304 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 10 22:55:47 melchior dbus-daemon[492]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.91' (uid=1000 pid=1304 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 10 22:55:47 melchior dbus-daemon[492]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.92' (uid=1000 pid=1304 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 10 22:55:47 melchior dbus-daemon[492]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.93' (uid=1000 pid=1304 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 10 22:55:47 melchior dbus-daemon[492]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.94' (uid=1000 pid=1304 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 10 22:55:47 melchior systemd[612]: pulseaudio.service: Consumed 20.163s CPU time. Jun 11 00:59:13 melchior dbus-daemon[513]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.18' (uid=1000 pid=1346 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 11 00:59:14 melchior pulseaudio[1346]: Failed to open cookie file '/home/basil/.config/pulse/cookie': No such file or directory Jun 11 00:59:14 melchior pulseaudio[1346]: Failed to load authentication key '/home/basil/.config/pulse/cookie': No such file or directory Jun 11 00:59:14 melchior pulseaudio[1346]: stat('/etc/pulse/default.pa.d'): No such file or directory Jun 11 01:16:33 melchior pulseaudio[1346]: ALSA woke us up to write new data to the device, but there was actually nothing to write. Jun 11 01:16:33 melchior pulseaudio[1346]: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Jun 11 01:16:33 melchior pulseaudio[1346]: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Jun 11 01:29:42 melchior dbus-daemon[513]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.39' (uid=1000 pid=1346 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 11 01:29:42 melchior dbus-daemon[513]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.40' (uid=1000 pid=1346 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 11 01:29:42 melchior dbus-daemon[513]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.41' (uid=1000 pid=1346 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 11 01:29:42 melchior dbus-daemon[513]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.42' (uid=1000 pid=1346 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 11 01:29:42 melchior dbus-daemon[513]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.43' (uid=1000 pid=1346 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 11 01:29:42 melchior pulseaudio[1346]: Error opening PCM device hw:3: No such file or directory Jun 11 01:29:42 melchior systemd[638]: pulseaudio.service: Consumed 7.162s CPU time. Jun 11 11:37:46 melchior dbus-daemon[468]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.18' (uid=1000 pid=1334 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 11 11:37:46 melchior pulseaudio[1334]: Failed to open cookie file '/home/basil/.config/pulse/cookie': No such file or directory Jun 11 11:37:46 melchior pulseaudio[1334]: Failed to load authentication key '/home/basil/.config/pulse/cookie': No such file or directory Jun 11 11:37:46 melchior pulseaudio[1334]: stat('/etc/pulse/default.pa.d'): No such file or directory Jun 11 13:30:54 melchior pulseaudio[1334]: ALSA woke us up to write new data to the device, but there was actually nothing to write. Jun 11 13:30:54 melchior pulseaudio[1334]: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Jun 11 13:30:54 melchior pulseaudio[1334]: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Jun 11 13:51:22 melchior dbus-daemon[468]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.57' (uid=1000 pid=1334 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 11 13:51:22 melchior dbus-daemon[468]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.58' (uid=1000 pid=1334 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 11 13:51:22 melchior dbus-daemon[468]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.59' (uid=1000 pid=1334 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 11 13:51:22 melchior dbus-daemon[468]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.60' (uid=1000 pid=1334 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 11 13:51:22 melchior dbus-daemon[468]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.61' (uid=1000 pid=1334 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 11 13:51:22 melchior pulseaudio[1334]: Error opening PCM device hw:3: No such file or directory Jun 11 13:51:22 melchior systemd[618]: pulseaudio.service: Consumed 11.444s CPU time. Jun 11 15:53:44 melchior dbus-daemon[536]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.18' (uid=1000 pid=1644 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 11 15:53:44 melchior pulseaudio[1644]: Failed to open cookie file '/home/basil/.config/pulse/cookie': No such file or directory Jun 11 15:53:44 melchior pulseaudio[1644]: Failed to load authentication key '/home/basil/.config/pulse/cookie': No such file or directory Jun 11 15:53:44 melchior pulseaudio[1644]: stat('/etc/pulse/default.pa.d'): No such file or directory Jun 11 17:13:20 melchior pulseaudio[1644]: ALSA woke us up to write new data to the device, but there was actually nothing to write. Jun 11 17:13:20 melchior pulseaudio[1644]: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Jun 11 17:13:20 melchior pulseaudio[1644]: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Jun 11 21:20:39 melchior dbus-daemon[536]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.63' (uid=1000 pid=1644 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 11 21:20:39 melchior dbus-daemon[536]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.64' (uid=1000 pid=1644 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 11 21:20:39 melchior dbus-daemon[536]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.65' (uid=1000 pid=1644 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 11 21:20:39 melchior dbus-daemon[536]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.66' (uid=1000 pid=1644 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 11 21:20:39 melchior dbus-daemon[536]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.67' (uid=1000 pid=1644 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 11 21:20:39 melchior systemd[621]: pulseaudio.service: Consumed 13.392s CPU time. Jun 12 14:01:16 melchior dbus-daemon[472]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.16' (uid=1000 pid=1168 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 12 14:01:17 melchior pulseaudio[1168]: Failed to open cookie file '/home/basil/.config/pulse/cookie': No such file or directory Jun 12 14:01:17 melchior pulseaudio[1168]: Failed to load authentication key '/home/basil/.config/pulse/cookie': No such file or directory Jun 12 14:01:17 melchior pulseaudio[1168]: stat('/etc/pulse/default.pa.d'): No such file or directory Jun 12 19:53:14 melchior pulseaudio[1168]: ALSA woke us up to write new data to the device, but there was actually nothing to write. Jun 12 19:53:14 melchior pulseaudio[1168]: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Jun 12 19:53:14 melchior pulseaudio[1168]: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Jun 12 23:14:09 melchior dbus-daemon[472]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.71' (uid=1000 pid=1168 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 12 23:14:09 melchior dbus-daemon[472]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.72' (uid=1000 pid=1168 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 12 23:14:09 melchior dbus-daemon[472]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.73' (uid=1000 pid=1168 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 12 23:14:09 melchior dbus-daemon[472]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.74' (uid=1000 pid=1168 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 12 23:14:09 melchior dbus-daemon[472]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.75' (uid=1000 pid=1168 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 12 23:14:09 melchior systemd[652]: pulseaudio.service: Consumed 4min 4.210s CPU time. Jun 12 23:14:41 melchior dbus-daemon[510]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.23' (uid=1000 pid=1890 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 12 23:14:42 melchior pulseaudio[1890]: Failed to open cookie file '/home/basil/.config/pulse/cookie': No such file or directory Jun 12 23:14:42 melchior pulseaudio[1890]: Failed to load authentication key '/home/basil/.config/pulse/cookie': No such file or directory Jun 12 23:14:42 melchior pulseaudio[1890]: stat('/etc/pulse/default.pa.d'): No such file or directory Jun 12 23:22:19 melchior dbus-daemon[510]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.47' (uid=1000 pid=1890 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 12 23:22:19 melchior dbus-daemon[510]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.48' (uid=1000 pid=1890 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 12 23:22:19 melchior dbus-daemon[510]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.49' (uid=1000 pid=1890 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 12 23:22:19 melchior dbus-daemon[510]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.50' (uid=1000 pid=1890 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 12 23:22:19 melchior dbus-daemon[510]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.51' (uid=1000 pid=1890 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 12 23:22:19 melchior pulseaudio[1890]: Error opening PCM device hw:3: No such file or directory Jun 12 23:22:19 melchior systemd[623]: pulseaudio.service: Consumed 2.341s CPU time. Jun 13 15:42:21 melchior dbus-daemon[498]: [system] Activating via systemd: service name='org.freedesktop.RealtimeKit1' unit='rtkit-daemon.service' requested by ':1.18' (uid=1000 pid=1347 comm="/usr/bin/pulseaudio --daemonize=no --log-target=jo") Jun 13 15:42:21 melchior pulseaudio[1347]: Failed to open cookie file '/home/basil/.config/pulse/cookie': No such file or directory Jun 13 15:42:21 melchior pulseaudio[1347]: Failed to load authentication key '/home/basil/.config/pulse/cookie': No such file or directory Jun 13 15:42:21 melchior pulseaudio[1347]: stat('/etc/pulse/default.pa.d'): No such file or directory Jun 13 17:18:25 melchior pulseaudio[1347]: ALSA woke us up to write new data to the device, but there was actually nothing to write. Jun 13 17:18:25 melchior pulseaudio[1347]: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Jun 13 17:18:25 melchior pulseaudio[1347]: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/328 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v3 00/12] Add support for IIO devices in ASoC
by Herve Codina 13 Jun '23

13 Jun '23

Several weeks ago, I sent a series [1] for adding a potentiometer as an auxiliary device in ASoC. The feedback was that the potentiometer should be directly handled in IIO (as other potentiometers) and something more generic should be present in ASoC in order to have a binding to import some IIO devices into sound cards. The series related to the IIO potentiometer device is already applied. This series introduces audio-iio-aux. Its goal is to offer the binding between IIO and ASoC. It exposes attached IIO devices as ASoC auxiliary devices and allows to control them through mixer controls. On my system, the IIO device is a potentiometer and it is present in an amplifier design present in the audio path. Compare to the previous iteration https://lore.kernel.org/linux-kernel/20230523151223.109551-1-herve.codina@b… This v3 series mainly: - Reworks/Simplifies some IIO code. - Reworks/Simplified the audio-iio-aux driver. - Removes an already applied patch. Best regards, Hervé [1] https://lore.kernel.org/linux-kernel/20230203111422.142479-1-herve.codina@b… [2] https://lore.kernel.org/linux-kernel/20230421085245.302169-1-herve.codina@b… Changes v2 -> v3 - Patches 1, 2 No changes. - Patch 3, 4 Add 'Acked-by: Jonathan Cameron <Jonathan.Cameron(a)huawei.com>'. - Patch 5 (new in v3) Removed the 'unused' variable and check the null pointer when used. - Patch 6 (new in v3) Introduce {min,max}_array(). - Patch 7 (new in v3) Use max_array() in iio_channel_read_max(). - Patch 8 (new in v3) Replace a FIXME comment by a TODO one. - Patch 9 (patch 5 in v2) Removed the 'unused' variable and check the null pointer when used. Use min_array(). Remplace a FIXME comment by a TODO one. - Patch 10 (patch 6 in v2) Convert existing macros to return a compound litteral instead of adding a new helper. - Patch 11 (patch 7 in v2) Remove the file name from the C file header. Use directly converted DAPM macros. Replace <linux/module.h> by <linux/mod_devicetable.h>. Add <linux/platform_device.h>. Be sure that min <= max. Swap values if it is not the case. Move the bool structure member after the int ones. Remove unneeded assignements. Use dev_err_probe() when relevant. Use str_on_off(). Use static_assert() instead of BUILD_BUG_ON(). Remove unneeded comma and blank line. Use device_property_*() instead of the OF API. - patch 8 available in v2 removed as already applied - Patch 12 (patch 9 in v2) Use devm_add_action_or_reset(). Call simple_populate_aux() from simple_parse_of(). Changes v1 -> v2 - Patch 1 Rename simple-iio-aux to audio-iio-aux Rename invert to snd-control-invert-range Remove the /schemas/iio/iio-consumer.yaml reference Remove the unneeded '|' after description - Patch 2 (new in v2) Introduce the simple-audio-card additional-devs subnode - Patch 3 (new in v2) Check err before switch() in iio_channel_read_max() - Patch 4 (new in v2) Fix raw reads and raw writes documentation - Patch 5 (patch 2 in v1) Check err before switch() in iio_channel_read_min() Fix documentation - Patch 6 (path 3 in v1) No changes - Patch 7 (patch 4 in v1) Rename simple-iio-aux to audio-iio-aux Rename invert to snd-control-invert-range Remove the mask usage from audio_iio_aux_{get,put}_volsw helpers Use directly PTR_ERR() in dev_err_probe() parameter Remove the '!!' construction Remove of_match_ptr() - Patch 8 (new in v2) Add a missing of_node_put() in the simple-card driver - Patch 9 (new in v2) Handle additional-devs in the simple-card driver Herve Codina (12): ASoC: dt-bindings: Add audio-iio-aux ASoC: dt-bindings: simple-card: Add additional-devs subnode iio: inkern: Check error explicitly in iio_channel_read_max() iio: consumer.h: Fix raw values documentation notes iio: inkern: Remove the 'unused' variable usage in iio_channel_read_max() minmax: Introduce {min,max}_array() iio: inkern: Use max_array() to get the maximum value from an array iio: inkern: Replace a FIXME comment by a TODO one iio: inkern: Add a helper to query an available minimum raw value ASoC: soc-dapm.h: Convert macros to return a compound litteral ASoC: codecs: Add support for the generic IIO auxiliary devices ASoC: simple-card: Handle additional devices .../bindings/sound/audio-iio-aux.yaml | 64 ++++ .../bindings/sound/simple-card.yaml | 53 +++ drivers/iio/inkern.c | 84 ++++- include/linux/iio/consumer.h | 37 +- include/linux/minmax.h | 26 ++ include/sound/soc-dapm.h | 138 +++++--- sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/audio-iio-aux.c | 334 ++++++++++++++++++ sound/soc/generic/simple-card.c | 46 ++- 10 files changed, 726 insertions(+), 70 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/audio-iio-aux.yaml create mode 100644 sound/soc/codecs/audio-iio-aux.c -- 2.40.1

2 28

[PATCH 00/10] ASoC: Use maple tree for Cirrus Logic devices
by Mark Brown 13 Jun '23

13 Jun '23

A lot of the Cirrus Logic devices only support single register read/write operations so they get no benefit from using the rbtree cache over the more modern maple tree cache, convert them to use maple tree. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Mark Brown (10): ASoC: cs35l32: Use maple tree register cache ASoC: cs35l33: Use maple tree register cache ASoC: cs35l34: Use maple tree register cache ASoC: cs35l35: Use maple tree register cache ASoC: cs4234: Use maple tree register cache ASoC: cs42l42: Use maple tree register cache ASoC: cs42l73: Use maple tree register cache ASoC: cs42l83: Use maple tree register cache ASoC: cs43130: Use maple tree register cache ASoC: cs35l30: Use maple tree register cache sound/soc/codecs/cs35l32.c | 2 +- sound/soc/codecs/cs35l33.c | 2 +- sound/soc/codecs/cs35l34.c | 2 +- sound/soc/codecs/cs35l35.c | 2 +- sound/soc/codecs/cs4234.c | 2 +- sound/soc/codecs/cs42l42.c | 2 +- sound/soc/codecs/cs42l73.c | 2 +- sound/soc/codecs/cs42l83-i2c.c | 2 +- sound/soc/codecs/cs43130.c | 2 +- sound/soc/codecs/cs53l30.c | 2 +- 10 files changed, 10 insertions(+), 10 deletions(-) --- base-commit: 9561de3a55bed6bdd44a12820ba81ec416e705a7 change-id: 20230609-asoc-cirrus-maple-0aba1f5c18b8 Best regards, -- Mark Brown <broonie(a)kernel.org>

3 13

[PATCH v2 0/5] ASoC: add new trigger ordering method
by Kuninori Morimoto 13 Jun '23

13 Jun '23

Hi Mark This patch-set adds new "trigger" starting/stopping method. Link: https://lore.kernel.org/r/874jnihm8y.wl-kuninori.morimoto.gx@renesas.com v1 -> v2 - tidyup const vs static - care missing starfive Kuninori Morimoto (5): ASoC: add new trigger ordering method ASoC: amd: use use new trigger ordering method ASoC: atmel: use use new trigger ordering method ASoC: starfive: use use new trigger ordering method ASoC: remove old trigger ordering method include/sound/soc-component.h | 11 ++- include/sound/soc.h | 23 ++++-- sound/soc/amd/acp-da7219-max98357a.c | 20 +++--- sound/soc/amd/acp-es8336.c | 2 +- sound/soc/atmel/mchp-pdmc.c | 2 +- sound/soc/soc-pcm.c | 104 ++++++++++++++------------- sound/soc/starfive/jh7110_tdm.c | 2 +- 7 files changed, 93 insertions(+), 71 deletions(-) -- 2.25.1

3 7

[PATCH] ASoC: tegra: Fix Master Volume Control
by Jon Hunter 13 Jun '23

13 Jun '23

Commit 3ed2b549b39f ("ALSA: pcm: fix wait_time calculations") corrected the PCM wait_time calculations and in doing so reduced the calculated wait_time. This exposed an issue with the Tegra Master Volume Control (MVC) device where the reduced wait_time caused the MVC to fail. For now fix this by setting the default wait_time for Tegra to be 500ms. Fixes: 3ed2b549b39f ("ALSA: pcm: fix wait_time calculations") Signed-off-by: Jon Hunter <jonathanh(a)nvidia.com> --- sound/soc/tegra/tegra_pcm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index 468c8e77de21..0b69cebc9a33 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -117,6 +117,9 @@ int tegra_pcm_open(struct snd_soc_component *component, return ret; } + /* Set wait time to 500ms by default */ + substream->wait_time = 500; + return 0; } EXPORT_SYMBOL_GPL(tegra_pcm_open); -- 2.34.1

4 5

[PATCH] ASoC: siu: Add MODULE_FIRMWARE macro
by Juerg Haefliger 13 Jun '23

13 Jun '23

The module loads firmware so add a MODULE_FIRMWARE macro to provide that information via modinfo. Signed-off-by: Juerg Haefliger <juerg.haefliger(a)canonical.com> --- sound/soc/sh/siu_dai.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sh/siu_dai.c b/sound/soc/sh/siu_dai.c index 84e1b14e68e4..d0b5c543fd2f 100644 --- a/sound/soc/sh/siu_dai.c +++ b/sound/soc/sh/siu_dai.c @@ -796,3 +796,5 @@ module_platform_driver(siu_driver); MODULE_AUTHOR("Carlos Munoz <carlos(a)kenati.com>"); MODULE_DESCRIPTION("ALSA SoC SH7722 SIU driver"); MODULE_LICENSE("GPL"); + +MODULE_FIRMWARE("siu_spb.bin"); -- 2.37.2

2 1

[PATCH 0/2] ASoC: add N cpus to M codecs dai link support
by Bard Liao 13 Jun '23

13 Jun '23

Currently, ASoC supports dailinks with the following mappings: 1 cpu DAI to N codec DAIs N cpu DAIs to N codec DAIs But the mapping between N cpu DAIs and M codec DAIs is not supported. The reason is that we didn't have a mechanism to map cpu and codec DAIs This series suggests a new snd_soc_dai_link_codec_ch_map struct in struct snd_soc_dai_link{} which provides codec DAI to cpu DAI mapping information used to implement N cpu DAIs to M codec DAIs support. And add the codec_ch_maps to SOF SoundWire machine driver. Bard Liao (2): ASoC: add N cpus to M codecs dai link support ASoC: Intel: sof_sdw: add dai_link_codec_ch_map include/sound/soc.h | 6 +++ sound/soc/intel/boards/sof_sdw.c | 69 +++++++++++++++++++++++++ sound/soc/intel/boards/sof_sdw_common.h | 2 + sound/soc/intel/boards/sof_sdw_maxim.c | 1 + sound/soc/soc-dapm.c | 24 ++++++++- sound/soc/soc-pcm.c | 44 ++++++++++++++-- 6 files changed, 141 insertions(+), 5 deletions(-) -- 2.25.1

7 13

[PATCH v2] regmap: regcache: Don't sync read-only registers
by Takashi Iwai 13 Jun '23

13 Jun '23

regcache_maple_sync() tries to sync all cached values no matter whether it's writable or not. OTOH, regache_sync_val() does care the wrtability and returns -EIO for a read-only register. This results in an error message like: snd_hda_codec_realtek hdaudioC0D0: Unable to sync register 0x2f0009. -5 and the sync loop is aborted incompletely. This patch adds the writable register check to regcache_sync_val() for addressing the bug above. Note that, although we may add the check in the caller side (regcache_maple_sync()), here we put in regcache_sync_val(), so that a similar case like this can be avoided in future. Fixes: f033c26de5a5 ("regmap: Add maple tree based register cache") Link: https://lore.kernel.org/r/877cs7g6f1.wl-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- drivers/base/regmap/regcache.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/drivers/base/regmap/regcache.c b/drivers/base/regmap/regcache.c index 029564695dbb..97c681fcf9f6 100644 --- a/drivers/base/regmap/regcache.c +++ b/drivers/base/regmap/regcache.c @@ -284,6 +284,9 @@ static bool regcache_reg_needs_sync(struct regmap *map, unsigned int reg, { int ret; + if (!regmap_writeable(map, reg)) + return false; + /* If we don't know the chip just got reset, then sync everything. */ if (!map->no_sync_defaults) return true; -- 2.35.3

2 1

[ PATCH v2 3/3] ASoC: dt-bindings: Add support for Loongson audio card
by YingKun Meng 13 Jun '23

13 Jun '23

From: Yingkun Meng <mengyingkun(a)loongson.cn> The audio card uses loongson I2S controller present in 7axxx/2kxxx chips to transfer audio data. On loongson platform, the chip has only one I2S controller. Signed-off-by: Yingkun Meng <mengyingkun(a)loongson.cn> --- .../sound/loongson,ls-audio-card.yaml | 70 +++++++++++++++++++ 1 file changed, 70 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/loongson,ls-audio-card.yaml diff --git a/Documentation/devicetree/bindings/sound/loongson,ls-audio-card.yaml b/Documentation/devicetree/bindings/sound/loongson,ls-audio-card.yaml new file mode 100644 index 000000000000..61e8babed402 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/loongson,ls-audio-card.yaml @@ -0,0 +1,70 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/loongson,ls-audio-card.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Loongson 7axxx/2kxxx ASoC audio sound card driver + +maintainers: + - Yingkun Meng <mengyingkun(a)loongson.cn> + +description: + The binding describes the sound card present in loongson + 7axxx/2kxxx platform. The sound card is an ASoC component + which uses Loongson I2S controller to transfer the audio data. + +properties: + compatible: + const: loongson,ls-audio-card + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User specified audio sound card name + + mclk-fs: + $ref: simple-card.yaml#/definitions/mclk-fs + + cpu: + description: Holds subnode which indicates cpu dai. + type: object + additionalProperties: false + properties: + sound-dai: + maxItems: 1 + required: + - sound-dai + + codec: + description: Holds subnode which indicates codec dai. + type: object + additionalProperties: false + properties: + sound-dai: + maxItems: 1 + required: + - sound-dai + +required: + - compatible + - model + - mclk-fs + - cpu + - codec + +additionalProperties: false + +examples: + - | + sound { + compatible = "loongson,ls-audio-card"; + model = "loongson-audio"; + mclk-fs = <512>; + + cpu { + sound-dai = <&i2s>; + }; + codec { + sound-dai = <&es8323>; + }; + }; -- 2.33.0

4 9

[ PATCH v2 0/3] Add Loongson I2S controller support
by YingKun Meng 13 Jun '23

13 Jun '23

Hi all, This patchset adds support for Loongson I2S controller, and introduce a ASoC machine driver for loongson platform. The Loongson I2S controller is available on Loongson 7a2000/2k2000 chips, works as a PCI device. It has two private DMA controllers used to playback and capture. Each DMA controller has one channel. The ASoC machine driver adds support for audio device which using loongson I2S controller to tranfser the audio data. The audio device uses "PRP0001" as its ACPI device ID, which provides a means to use the existing DT-compatible device identification in ACPI. Thanks. --- v2: * Use ACPI bindings to reference I2S and codec nodes. * Add 'required' restriction for sound-dai property in dt-bindings. * Fix build errors from lkp(a)intel.com. * Replace pci_xxx() APIs with pcim_xxx() APIs. * mirror changes in log printing. v1: * Add support for Loongson I2S controller. Yingkun Meng (3): ASoC: Add support for Loongson I2S controller ASoC: loongson: Add Loongson ASoC Sound Card Support ASoC: dt-bindings: Add support for Loongson audio card .../sound/loongson,ls-audio-card.yaml | 70 +++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/loongson/Kconfig | 26 + sound/soc/loongson/Makefile | 8 + sound/soc/loongson/loongson_card.c | 237 +++++++++ sound/soc/loongson/loongson_i2s.c | 213 ++++++++ sound/soc/loongson/loongson_i2s.h | 70 +++ sound/soc/loongson/loongson_i2s_pci.c | 500 ++++++++++++++++++ 9 files changed, 1126 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/loongson,ls-audio-card.yaml create mode 100644 sound/soc/loongson/Kconfig create mode 100644 sound/soc/loongson/Makefile create mode 100644 sound/soc/loongson/loongson_card.c create mode 100644 sound/soc/loongson/loongson_i2s.c create mode 100644 sound/soc/loongson/loongson_i2s.h create mode 100644 sound/soc/loongson/loongson_i2s_pci.c base-commit: 62a97bea5cce5317d6d7630f7bcf0cdf5333e269 -- 2.33.0

2 2

Alsa working, but spits out messages each time
by GitHub issues - opened 13 Jun '23

13 Jun '23

alsa-project/alsa-lib issue #327 was opened from melihbudak99: I am using Armbian system on Orange pi and I am printing my voice commands to the screen. When I used the same system and packages a few months ago on the Armbian system, I used to get these messages once, but now it prints these messages every time I speak; `ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.front ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.rear ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.center_lfe ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.side ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.surround21 ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.surround21 ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.surround40 ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.surround41 ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.surround50 ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.surround51 ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.surround71 ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.iec958 ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.iec958 ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.iec958 ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.modem ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.modem ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.phoneline ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.phoneline` Issue URL : https://github.com/alsa-project/alsa-lib/issues/327 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] regmap: maple: Don't sync read-only registers
by Takashi Iwai 13 Jun '23

13 Jun '23

regcache_maple_sync() tries to sync all cached values no matter whether it's writable or not. OTOH, regache_sync_val() does care the wrtability and returns -EIO for a read-only register. This results in an error message like: snd_hda_codec_realtek hdaudioC0D0: Unable to sync register 0x2f0009. -5 and the sync loop is aborted incompletely. This patch adds the writable register check in the loop for addressing the bug. Fixes: f033c26de5a5 ("regmap: Add maple tree based register cache") Link: https://lore.kernel.org/r/877cs7g6f1.wl-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- drivers/base/regmap/regcache-maple.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/drivers/base/regmap/regcache-maple.c b/drivers/base/regmap/regcache-maple.c index c2e3a0f6c218..d2dad32dd399 100644 --- a/drivers/base/regmap/regcache-maple.c +++ b/drivers/base/regmap/regcache-maple.c @@ -203,6 +203,8 @@ static int regcache_maple_sync(struct regmap *map, unsigned int min, mas_for_each(&mas, entry, max) { for (r = max(mas.index, lmin); r <= min(mas.last, lmax); r++) { + if (!regmap_writeable(map, r)) + continue; mas_pause(&mas); rcu_read_unlock(); ret = regcache_sync_val(map, r, entry[r - mas.index]); -- 2.35.3

2 4

[PATCH V2 1/2] ASoC: dt-bindings: max98388: add amplifier driver
by “Ryan 13 Jun '23

13 Jun '23

From: Ryan Lee <ryans.lee(a)analog.com> Add dt-bindings information for Analog Devices MAX98388 I2S Amplifier Signed-off-by: Ryan Lee <ryans.lee(a)analog.com> --- Changes from v1: Removed unnecessary blank line and description. Modified quotes. .../bindings/sound/adi,max98388.yaml | 79 +++++++++++++++++++ 1 file changed, 79 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/adi,max98388.yaml diff --git a/Documentation/devicetree/bindings/sound/adi,max98388.yaml b/Documentation/devicetree/bindings/sound/adi,max98388.yaml new file mode 100644 index 000000000000..93ccd5905736 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,max98388.yaml @@ -0,0 +1,79 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/adi,max98388.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Analog Devices MAX98388 Speaker Amplifier + +maintainers: + - Ryan Lee <ryans.lee(a)analog.com> + +description: + The MAX98388 is a mono Class-D speaker amplifier with I/V feedback. + The device provides a PCM interface for audio data and a standard + I2C interface for control data communication. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - adi,max98388 + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + adi,vmon-slot-no: + description: slot number of the voltage feedback monitor + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 15 + default: 0 + + adi,imon-slot-no: + description: slot number of the current feedback monitor + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 15 + default: 1 + + adi,interleave-mode: + description: + For cases where a single combined channel for the I/V feedback data + is not sufficient, the device can also be configured to share + a single data output channel on alternating frames. + In this configuration, the current and voltage data will be frame + interleaved on a single output channel. + type: boolean + + reset-gpios: + maxItems: 1 + +required: + - compatible + - reg + - '#sound-dai-cells' + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + max98388: amplifier@39 { + compatible = "adi,max98388"; + reg = <0x39>; + #sound-dai-cells = <0>; + adi,vmon-slot-no = <0>; + adi,imon-slot-no = <1>; + adi,interleave-mode; + reset-gpios = <&gpio 4 GPIO_ACTIVE_LOW>; + }; + }; base-commit: 53ab6975c12d1ad86c599a8927e8c698b144d669 -- 2.34.1

3 4

[PATCH 00/36] ALSA: Add MIDI 2.0 support
by Takashi Iwai 13 Jun '23

13 Jun '23

Hi, this is a (largish) patch set for adding the support of MIDI 2.0 functionality, mainly targeted for USB devices. MIDI 2.0 is a complete overhaul of the 40-years old MIDI 1.0. Unlike MIDI 1.0 byte stream, MIDI 2.0 uses packets in 32bit words for Universal MIDI Packet (UMP) protocol. It supports both MIDI 1.0 commands for compatibility and the extended MIDI 2.0 commands for higher resolutions and more functions. For supporting the UMP, the patch set extends the existing ALSA rawmidi and sequencer interfaces, and adds the USB MIDI 2.0 support to the standard USB-audio driver. The rawmidi for UMP has a different device name (/dev/snd/umpC*D*) and it reads/writes UMP packet data in 32bit CPU-native endianness. For the old MIDI 1.0 applications, the legacy rawmidi interface is provided, too. As default, USB-audio driver will take the alternate setting for MIDI 2.0 interface, and the compatibility with MIDI 1.0 is provided via the rawmidi common layer. However, user may let the driver falling back to the old MIDI 1.0 interface by a module option, too. A UMP-capable rawmidi device can create the corresponding ALSA sequencer client(s) to support the UMP Endpoint and UMP Group connections. As a nature of ALSA sequencer, arbitrary connections between clients/ports are allowed, and the ALSA sequencer core performs the automatic conversions for the connections between a new UMP sequencer client and a legacy MIDI 1.0 sequencer client. It allows the existing application to use MIDI 2.0 devices without changes. The MIDI-CI, which is another major extension in MIDI 2.0, isn't covered by this patch set. It would be implemented rather in user-space. Roughly speaking, the first half of this patch set is for extending the rawmidi and USB-audio, and the second half is for extending the ALSA sequencer interface. The patch set is based on 6.4-rc2 kernel, but all patches can be cleanly applicable on 6.2 and 6.3 kernels, too (while 6.1 and older kernels would need minor adjustment for uapi header changes). The updates for alsa-lib and alsa-utils will follow shortly later. The author thanks members of MIDI Association OS/API Working Group, especially Andrew Mee, for great helps for the initial design and debugging / testing the drivers. Takashi --- Takashi Iwai (36): ALSA: rawmidi: Pass rawmidi directly to snd_rawmidi_kernel_open() ALSA: rawmidi: Add ioctl callback to snd_rawmidi_global_ops ALSA: rawmidi: UMP support ALSA: rawmidi: Skip UMP devices at SNDRV_CTL_IOCTL_RAWMIDI_NEXT_DEVICE ALSA: ump: Additional proc output ALSA: usb-audio: Manage number of rawmidis globally ALSA: usb-audio: Define USB MIDI 2.0 specs ALSA: usb-audio: USB MIDI 2.0 UMP support ALSA: usb-audio: Get UMP EP name string from USB interface ALSA: usb-audio: Trim superfluous "MIDI" suffix from UMP EP name ALSA: usb-audio: Create UMP blocks from USB MIDI GTBs ALSA: ump: Redirect rawmidi substream access via own helpers ALSA: ump: Add legacy raw MIDI support ALSA: usb-audio: Enable the legacy raw MIDI support ALSA: usb-audio: Inform inconsistent protocols in GTBs ALSA: seq: Clear padded bytes at expanding events ALSA: seq: Add snd_seq_expand_var_event_at() helper ALSA: seq: Treat snd_seq_client object directly in client drivers ALSA: seq: Drop dead code for the old broadcast support ALSA: seq: Check the conflicting port at port creation ALSA: seq: Check validity before creating a port object ALSA: seq: Prohibit creating ports with special numbers ALSA: seq: Introduce SNDRV_SEQ_IOCTL_USER_PVERSION ioctl ALSA: seq: Add UMP support ALSA: seq: Add port inactive flag ALSA: seq: Support MIDI 2.0 UMP Endpoint port ALSA: seq: Add port direction to snd_seq_port_info ALSA: seq: Add UMP group number to snd_seq_port_info ALSA: seq: Automatic conversion of UMP events ALSA: seq: Allow suppressing UMP conversions ALSA: seq: Bind UMP device ALSA: seq: ump: Create UMP Endpoint port for broadcast ALSA: seq: Add ioctls for client UMP info query and setup ALSA: seq: Print UMP Endpoint and Block information in proc outputs ALSA: seq: Add UMP group filter ALSA: docs: Add MIDI 2.0 documentation Documentation/sound/designs/index.rst | 1 + Documentation/sound/designs/midi-2.0.rst | 342 ++++++ include/linux/usb/midi-v2.h | 94 ++ include/sound/asequencer.h | 4 + include/sound/rawmidi.h | 16 +- include/sound/seq_device.h | 1 + include/sound/seq_kernel.h | 10 + include/sound/ump.h | 175 ++++ include/sound/ump_msg.h | 540 ++++++++++ include/uapi/sound/asequencer.h | 83 +- include/uapi/sound/asound.h | 58 +- sound/core/Kconfig | 13 + sound/core/Makefile | 3 + sound/core/rawmidi.c | 233 +++-- sound/core/rawmidi_compat.c | 4 + sound/core/seq/Kconfig | 14 + sound/core/seq/Makefile | 3 + sound/core/seq/seq_clientmgr.c | 577 +++++++---- sound/core/seq/seq_clientmgr.h | 27 +- sound/core/seq/seq_compat.c | 3 + sound/core/seq/seq_dummy.c | 9 + sound/core/seq/seq_memory.c | 98 +- sound/core/seq/seq_memory.h | 19 +- sound/core/seq/seq_midi.c | 12 +- sound/core/seq/seq_ports.c | 46 +- sound/core/seq/seq_ports.h | 23 +- sound/core/seq/seq_ump_client.c | 464 +++++++++ sound/core/seq/seq_ump_convert.c | 1203 ++++++++++++++++++++++ sound/core/seq/seq_ump_convert.h | 22 + sound/core/seq/seq_virmidi.c | 1 + sound/core/ump.c | 677 ++++++++++++ sound/core/ump_convert.c | 520 ++++++++++ sound/core/ump_convert.h | 43 + sound/usb/Kconfig | 11 + sound/usb/Makefile | 1 + sound/usb/card.c | 12 +- sound/usb/midi.c | 7 +- sound/usb/midi.h | 5 +- sound/usb/midi2.c | 1189 +++++++++++++++++++++ sound/usb/midi2.h | 33 + sound/usb/quirks.c | 8 +- sound/usb/usbaudio.h | 2 + 42 files changed, 6274 insertions(+), 332 deletions(-) create mode 100644 Documentation/sound/designs/midi-2.0.rst create mode 100644 include/linux/usb/midi-v2.h create mode 100644 include/sound/ump.h create mode 100644 include/sound/ump_msg.h create mode 100644 sound/core/seq/seq_ump_client.c create mode 100644 sound/core/seq/seq_ump_convert.c create mode 100644 sound/core/seq/seq_ump_convert.h create mode 100644 sound/core/ump.c create mode 100644 sound/core/ump_convert.c create mode 100644 sound/core/ump_convert.h create mode 100644 sound/usb/midi2.c create mode 100644 sound/usb/midi2.h -- 2.35.3

4 85

[PATCH 0/9] ALSA: emu10k1: improvements related to the switchable word clock of E-MU cards
by Oswald Buddenhagen 13 Jun '23

13 Jun '23

Oswald Buddenhagen (9): ALSA: emu10k1: split off E-MU fallback clock from clock source ALSA: emu10k1: make available E-MU clock sources card-specific ALSA: emu10k1: query rate of external clock sources on E-MU cards ALSA: emu10k1: fix sample rates for E-MU cards at 44.1 kHz word clock ALSA: emu10k1: fix synthesizer pitch for E-MU cards at 44.1 kHz ALSA: timer: minimize open-coded access to hw.resolution ALSA: emu10k1: fix timer for E-MU cards at 44.1 kHz word clock ALSA: emu10k1: add support for 12 kHz capture on Audigy ALSA: emu10k1: actually show some S/PDIF status in /proc for E-MU cards include/sound/emu10k1.h | 11 +- include/sound/emux_synth.h | 2 +- sound/core/timer.c | 18 +- sound/pci/emu10k1/emu10k1_callback.c | 10 ++ sound/pci/emu10k1/emu10k1_main.c | 7 +- sound/pci/emu10k1/emu10k1_synth.c | 1 - sound/pci/emu10k1/emumixer.c | 242 ++++++++++++++++----------- sound/pci/emu10k1/emupcm.c | 125 +++++++------- sound/pci/emu10k1/emuproc.c | 51 +++--- sound/pci/emu10k1/io.c | 72 ++++++++ sound/pci/emu10k1/timer.c | 20 ++- sound/synth/emux/emux_synth.c | 3 +- 12 files changed, 374 insertions(+), 188 deletions(-) -- 2.40.0.152.g15d061e6df

2 10

[PATCH v5 4/4] ASoC: dt-bindings: Add tas2781 amplifier
by Shenghao Ding 13 Jun '23

13 Jun '23

Create tas2781.yaml for tas2781 driver. Signed-off-by: Shenghao Ding <13916275206(a)139.com> --- Changes in v5: - remove ti,broadcast-addr - remove address-cells - remove size-cells - put compatible item first in properties - change the maxItems of reg from 4 to 8 - remove white space around <> - correct the reg format to <0x38>, <0x3a> etc - remove '\t' in the file - correct a comment in the example --- .../devicetree/bindings/sound/ti,tas2781.yaml | 73 +++++++++++++++++++ 1 file changed, 73 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,tas2781.yaml diff --git a/Documentation/devicetree/bindings/sound/ti,tas2781.yaml b/Documentation/devicetree/bindings/sound/ti,tas2781.yaml new file mode 100644 index 000000000000..61db14a39630 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tas2781.yaml @@ -0,0 +1,73 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2022 - 2023 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tas2781.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TAS2781 SmartAMP + +maintainers: + - Shenghao Ding <shenghao-ding(a)ti.com> + +description: + The TAS2781 is a mono, digital input Class-D audio amplifier + optimized for efficiently driving high peak power into small + loudspeakers. Integrated an on-chip DSP supports Texas Instruments + Smart Amp speaker protection algorithm. The integrated speaker + voltage and current sense provides for real time + monitoring of loudspeaker behavior. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - ti,tas2781 + + reg: + description: + I2C address, in multiple tas2781s case, all the i2c address + aggreate as one Audio Device to support multiple audio slots. + maxItems: 8 + items: + minimum: 0x38 + maximum: 0x3f + + reset-gpios: + maxItems: 1 + + interrupts: + maxItems: 1 + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + /* example with quad tas2781s, such as tablet or pad device */ + #address-cells = <1>; + #size-cells = <0>; + quad: codec@38 { + compatible = "ti,tas2781"; + reg = <0x38>, /* Audio slot 0 */ + <0x3a>, /* Audio slot 1 */ + <0x39>, /* Audio slot 2 */ + <0x3b>; /* Audio slot 3 */ + + #sound-dai-cells = <1>; + reset-gpios = <&gpio1 10 GPIO_ACTIVE_HIGH>; + interrupt-parent = <&gpio1>; + interrupts = <15>; + }; + }; +... -- 2.34.1

3 3

Alsa without support to usb headphone
by GitHub issues - opened 12 Jun '23

12 Jun '23

alsa-project/alsa-lib issue #326 was opened from wellinton-capixaba: Hello. O recently built a media player (mplayer) with alsa support. However, telling mplayer to play multimedia files with alsa, specifying the usb headphone as the output device, didn't work. However, latter I built another mplayer, this time using the system provided Also. Then I redid the test, and it worked as I originally expected. The only difference I am aware between both also as is their version. The one I built it is the 1.0.17, while the system uses a 1.0.21. Can be this the cause of the problem. Or is something else? Issue URL : https://github.com/alsa-project/alsa-lib/issues/326 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 00/12] ASoC: Intel: avs: remove redundant dapm routes
by Brent Lu 12 Jun '23

12 Jun '23

This patch series remove redundant dapm routes declared in multiple machine drivers. These routes will be created by snd_soc_dapm_connect_dai_link_widgets() during soundcard initialization. Following is the kernel log from a KBL chromebook nocturne. dmic: [ 13.773455] avs_dmic avs_dmic: connected DAI link dmic-codec:Capture -> dmic-platform:DMIC Rx [ 13.773460] avs_dmic avs_dmic: connected DAI link dmic-codec:Capture -> dmic-platform:DMIC WoV Rx max98373: [ 14.079536] avs_max98373 avs_max98373.1: connected DAI link avs_max98373.1-platform:ssp0 Tx -> i2c-MX98373:00:Right HiFi Playback [ 14.079545] avs_max98373 avs_max98373.1: connected DAI link i2c-MX98373:00:Right HiFi Capture -> avs_max98373.1-platform:ssp0 Rx [ 14.079550] avs_max98373 avs_max98373.1: connected DAI link avs_max98373.1-platform:ssp0 Tx -> i2c-MX98373:01:Left HiFi Playback [ 14.079554] avs_max98373 avs_max98373.1: connected DAI link i2c-MX98373:01:Left HiFi Capture -> avs_max98373.1-platform:ssp0 Rx hdaudio: [ 14.094818] avs_hdaudio avs_hdaudio.2: connected DAI link hdaudioB0D2-platform:hdaudioB0D2-cpu0 Tx -> hdaudioB0D2:HDMI 0 Playback [ 14.094824] avs_hdaudio avs_hdaudio.2: connected DAI link hdaudioB0D2-platform:hdaudioB0D2-cpu1 Tx -> hdaudioB0D2:HDMI 1 Playback [ 14.094828] avs_hdaudio avs_hdaudio.2: connected DAI link hdaudioB0D2-platform:hdaudioB0D2-cpu2 Tx -> hdaudioB0D2:HDMI 2 Playback Brent Lu (12): ASoC: Intel: avs-da7219: remove redundant dapm routes ASoC: Intel: avs-dmic: remove redundant dapm routes ASoC: Intel: avs-hdaudio: remove redundant dapm routes ASoC: Intel: avs-max98357a: remove redundant dapm routes ASoC: Intel: avs-max98373: remove redundant dapm routes ASoC: Intel: avs-max98927: remove redundant dapm routes ASoC: Intel: avs-nau8825: remove redundant dapm routes ASoC: Intel: avs-rt274: remove redundant dapm routes ASoC: Intel: avs-rt286: remove redundant dapm routes ASoC: Intel: avs-rt298: remove redundant dapm routes ASoC: Intel: avs-rt5682: remove redundant dapm routes ASoC: Intel: avs-ssm4567: remove redundant dapm routes sound/soc/intel/avs/boards/da7219.c | 45 ++---------------- sound/soc/intel/avs/boards/dmic.c | 2 - sound/soc/intel/avs/boards/hdaudio.c | 65 +------------------------- sound/soc/intel/avs/boards/max98357a.c | 39 ++-------------- sound/soc/intel/avs/boards/max98373.c | 45 ++---------------- sound/soc/intel/avs/boards/max98927.c | 45 ++---------------- sound/soc/intel/avs/boards/nau8825.c | 45 ++---------------- sound/soc/intel/avs/boards/rt274.c | 45 ++---------------- sound/soc/intel/avs/boards/rt286.c | 45 ++---------------- sound/soc/intel/avs/boards/rt298.c | 45 ++---------------- sound/soc/intel/avs/boards/rt5682.c | 45 ++---------------- sound/soc/intel/avs/boards/ssm4567.c | 57 ++-------------------- 12 files changed, 31 insertions(+), 492 deletions(-) -- 2.34.1

2 13

[PATCH] ASoC: audio-graph-card2-custom-sample.dtsi: remove DT warning
by Kuninori Morimoto 12 Jun '23

12 Jun '23

Current audio-graph-card2-custom-sample.dtsi is missing address-cells / size-cells / reg. Thus it get too many DT warnings. This patch solved it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- .../audio-graph-card2-custom-sample.dtsi | 151 ++++++++++++------ 1 file changed, 101 insertions(+), 50 deletions(-) diff --git a/sound/soc/generic/audio-graph-card2-custom-sample.dtsi b/sound/soc/generic/audio-graph-card2-custom-sample.dtsi index f5e7d669da45..2ac0de3c21da 100644 --- a/sound/soc/generic/audio-graph-card2-custom-sample.dtsi +++ b/sound/soc/generic/audio-graph-card2-custom-sample.dtsi @@ -172,83 +172,127 @@ &c2c_m >; multi { + #address-cells = <1>; + #size-cells = <0>; + ports@0 { + reg = <0>; + #address-cells = <1>; + #size-cells = <0>; /* [Multi-CPU] */ - mcpu0: port@0 { mcpu0_ep: endpoint { remote-endpoint = <&mcodec0_ep>; }; }; - port@1 { mcpu1_ep: endpoint { remote-endpoint = <&cpu1_ep>; }; }; - port@2 { mcpu2_ep: endpoint { remote-endpoint = <&cpu2_ep>; }; }; + mcpu0: port@0 { reg = <0>; mcpu0_ep: endpoint { remote-endpoint = <&mcodec0_ep>; }; }; + port@1 { reg = <1>; mcpu1_ep: endpoint { remote-endpoint = <&cpu1_ep>; }; }; + port@2 { reg = <2>; mcpu2_ep: endpoint { remote-endpoint = <&cpu2_ep>; }; }; }; /* [Multi-Codec] */ ports@1 { - port@0 { mcodec0_ep: endpoint { remote-endpoint = <&mcpu0_ep>; }; }; - port@1 { mcodec1_ep: endpoint { remote-endpoint = <&codec1_ep>; }; }; - port@2 { mcodec2_ep: endpoint { remote-endpoint = <&codec2_ep>; }; }; + reg = <1>; + #address-cells = <1>; + #size-cells = <0>; + port@0 { reg = <0>; mcodec0_ep: endpoint { remote-endpoint = <&mcpu0_ep>; }; }; + port@1 { reg = <1>; mcodec1_ep: endpoint { remote-endpoint = <&codec1_ep>; }; }; + port@2 { reg = <2>; mcodec2_ep: endpoint { remote-endpoint = <&codec2_ep>; }; }; }; /* [DPCM-Multi]::BE */ ports@2 { - port@0 { mbe_ep: endpoint { remote-endpoint = <&be10_ep>; }; }; - port@1 { mbe1_ep: endpoint { remote-endpoint = <&codec4_ep>; }; }; - port@2 { mbe2_ep: endpoint { remote-endpoint = <&codec5_ep>; }; }; + reg = <2>; + #address-cells = <1>; + #size-cells = <0>; + port@0 { reg = <0>; mbe_ep: endpoint { remote-endpoint = <&be10_ep>; }; }; + port@1 { reg = <1>; mbe1_ep: endpoint { remote-endpoint = <&codec4_ep>; }; }; + port@2 { reg = <2>; mbe2_ep: endpoint { remote-endpoint = <&codec5_ep>; }; }; }; /* [Codec2Codec-Multi]::CPU */ ports@3 { - port@0 { mc2c0_ep: endpoint { remote-endpoint = <&c2cmf_ep>; }; }; - port@1 { mc2c00_ep: endpoint { remote-endpoint = <&codec8_ep>; }; }; - port@2 { mc2c01_ep: endpoint { remote-endpoint = <&codec9_ep>; }; }; + reg = <3>; + #address-cells = <1>; + #size-cells = <0>; + port@0 { reg = <0>; mc2c0_ep: endpoint { remote-endpoint = <&c2cmf_ep>; }; }; + port@1 { reg = <1>; mc2c00_ep: endpoint { remote-endpoint = <&codec8_ep>; }; }; + port@2 { reg = <2>; mc2c01_ep: endpoint { remote-endpoint = <&codec9_ep>; }; }; }; /* [Codec2Codec-Multi]::Codec */ ports@4 { - port@0 { mc2c1_ep: endpoint { remote-endpoint = <&c2cmb_ep>; }; }; - port@1 { mc2c10_ep: endpoint { remote-endpoint = <&codec10_ep>; }; }; - port@2 { mc2c11_ep: endpoint { remote-endpoint = <&codec11_ep>; }; }; + reg = <4>; + #address-cells = <1>; + #size-cells = <0>; + port@0 { reg = <0>; mc2c1_ep: endpoint { remote-endpoint = <&c2cmb_ep>; }; }; + port@1 { reg = <1>; mc2c10_ep: endpoint { remote-endpoint = <&codec10_ep>; }; }; + port@2 { reg = <2>; mc2c11_ep: endpoint { remote-endpoint = <&codec11_ep>; }; }; }; /* [Semi-Multi] */ ports@5 { - port@0 { smcodec0_ep: endpoint { remote-endpoint = <&cpu7_ep>; }; }; - port@1 { smcodec1_ep: endpoint { remote-endpoint = <&codec12_ep>; }; }; - port@2 { smcodec2_ep: endpoint { remote-endpoint = <&codec13_ep>; }; }; + reg = <5>; + #address-cells = <1>; + #size-cells = <0>; + port@0 { reg = <0>; smcodec0_ep: endpoint { remote-endpoint = <&cpu7_ep>; }; }; + port@1 { reg = <1>; smcodec1_ep: endpoint { remote-endpoint = <&codec12_ep>; }; }; + port@2 { reg = <2>; smcodec2_ep: endpoint { remote-endpoint = <&codec13_ep>; }; }; }; }; dpcm { + #address-cells = <1>; + #size-cells = <0>; + ports@0 { + reg = <0>; + + #address-cells = <1>; + #size-cells = <0>; /* [DPCM]::FE */ - fe00: port@0 { fe00_ep: endpoint { remote-endpoint = <&cpu3_ep>; }; }; - fe01: port@1 { fe01_ep: endpoint { remote-endpoint = <&cpu4_ep>; }; }; + fe00: port@0 { reg = <0>; fe00_ep: endpoint { remote-endpoint = <&cpu3_ep>; }; }; + fe01: port@1 { reg = <1>; fe01_ep: endpoint { remote-endpoint = <&cpu4_ep>; }; }; /* [DPCM-Multi]::FE */ - fe10: port@2 { fe10_ep: endpoint { remote-endpoint = <&cpu5_ep>; }; }; - fe11: port@3 { fe11_ep: endpoint { remote-endpoint = <&cpu6_ep>; }; }; + fe10: port@2 { reg = <2>; fe10_ep: endpoint { remote-endpoint = <&cpu5_ep>; }; }; + fe11: port@3 { reg = <3>; fe11_ep: endpoint { remote-endpoint = <&cpu6_ep>; }; }; }; ports@1 { + reg = <1>; + + #address-cells = <1>; + #size-cells = <0>; /* [DPCM]::BE */ - be0: port@0 { be00_ep: endpoint { remote-endpoint = <&codec3_ep>; }; }; + be0: port@0 { reg = <0>; be00_ep: endpoint { remote-endpoint = <&codec3_ep>; }; }; /* [DPCM-Multi]::BE */ - be1: port@1 { be10_ep: endpoint { remote-endpoint = <&mbe_ep>; }; }; + be1: port@1 { reg = <1>; be10_ep: endpoint { remote-endpoint = <&mbe_ep>; }; }; }; }; codec2codec { + #address-cells = <1>; + #size-cells = <0>; /* [Codec2Codec] */ ports@0 { + reg = <0>; + + #address-cells = <1>; + #size-cells = <0>; + /* use default settings */ - c2c: port@0 { c2cf_ep: endpoint { remote-endpoint = <&codec6_ep>; }; }; - port@1 { c2cb_ep: endpoint { remote-endpoint = <&codec7_ep>; }; }; + c2c: port@0 { reg = <0>; c2cf_ep: endpoint { remote-endpoint = <&codec6_ep>; }; }; + port@1 { reg = <1>; c2cb_ep: endpoint { remote-endpoint = <&codec7_ep>; }; }; }; /* [Codec2Codec-Multi] */ ports@1 { + reg = <1>; + + #address-cells = <1>; + #size-cells = <0>; + /* use original settings */ rate = <48000>; - c2c_m: port@0 { c2cmf_ep: endpoint { remote-endpoint = <&mc2c0_ep>; }; }; - port@1 { c2cmb_ep: endpoint { remote-endpoint = <&mc2c1_ep>; }; }; + c2c_m: port@0 { reg = <0>; c2cmf_ep: endpoint { remote-endpoint = <&mc2c0_ep>; }; }; + port@1 { reg = <1>; c2cmb_ep: endpoint { remote-endpoint = <&mc2c1_ep>; }; }; }; }; }; @@ -264,25 +308,28 @@ test_cpu { */ compatible = "test-cpu"; ports { + #address-cells = <1>; + #size-cells = <0>; + bitclock-master; frame-master; /* [Normal] */ - cpu0: port@0 { cpu0_ep: endpoint { remote-endpoint = <&codec0_ep>; }; }; + cpu0: port@0 { reg = <0>; cpu0_ep: endpoint { remote-endpoint = <&codec0_ep>; }; }; /* [Multi-CPU] */ - port@1 { cpu1_ep: endpoint { remote-endpoint = <&mcpu1_ep>; }; }; - port@2 { cpu2_ep: endpoint { remote-endpoint = <&mcpu2_ep>; }; }; + port@1 { reg = <1>; cpu1_ep: endpoint { remote-endpoint = <&mcpu1_ep>; }; }; + port@2 { reg = <2>; cpu2_ep: endpoint { remote-endpoint = <&mcpu2_ep>; }; }; /* [DPCM]::FE */ - port@3 { cpu3_ep: endpoint { remote-endpoint = <&fe00_ep>; }; }; - port@4 { cpu4_ep: endpoint { remote-endpoint = <&fe01_ep>; }; }; + port@3 { reg = <3>; cpu3_ep: endpoint { remote-endpoint = <&fe00_ep>; }; }; + port@4 { reg = <4>; cpu4_ep: endpoint { remote-endpoint = <&fe01_ep>; }; }; /* [DPCM-Multi]::FE */ - port@5 { cpu5_ep: endpoint { remote-endpoint = <&fe10_ep>; }; }; - port@6 { cpu6_ep: endpoint { remote-endpoint = <&fe11_ep>; }; }; + port@5 { reg = <5>; cpu5_ep: endpoint { remote-endpoint = <&fe10_ep>; }; }; + port@6 { reg = <6>; cpu6_ep: endpoint { remote-endpoint = <&fe11_ep>; }; }; /* [Semi-Multi] */ - sm0: port@7 { cpu7_ep: endpoint { remote-endpoint = <&smcodec0_ep>; }; }; + sm0: port@7 { reg = <7>; cpu7_ep: endpoint { remote-endpoint = <&smcodec0_ep>; }; }; }; }; @@ -297,6 +344,9 @@ test_codec { */ compatible = "test-codec"; ports { + #address-cells = <1>; + #size-cells = <0>; + /* * prefix can be added to *component*, * see audio-graph-card2::routing @@ -304,39 +354,40 @@ ports { prefix = "TC"; /* [Normal] */ - port@0 { codec0_ep: endpoint { remote-endpoint = <&cpu0_ep>; }; }; + port@0 { reg = <0>; codec0_ep: endpoint { remote-endpoint = <&cpu0_ep>; }; }; /* [Multi-Codec] */ - port@1 { codec1_ep: endpoint { remote-endpoint = <&mcodec1_ep>; }; }; - port@2 { codec2_ep: endpoint { remote-endpoint = <&mcodec2_ep>; }; }; + port@1 { reg = <1>; codec1_ep: endpoint { remote-endpoint = <&mcodec1_ep>; }; }; + port@2 { reg = <2>; codec2_ep: endpoint { remote-endpoint = <&mcodec2_ep>; }; }; /* [DPCM]::BE */ port@3 { convert-rate = <44100>; - codec3_ep: endpoint { remote-endpoint = <&be00_ep>; }; + reg = <3>; codec3_ep: endpoint { remote-endpoint = <&be00_ep>; }; }; /* [DPCM-Multi]::BE */ - port@4 { codec4_ep: endpoint { remote-endpoint = <&mbe1_ep>; }; }; - port@5 { codec5_ep: endpoint { remote-endpoint = <&mbe2_ep>; }; }; + port@4 { reg = <4>; codec4_ep: endpoint { remote-endpoint = <&mbe1_ep>; }; }; + port@5 { reg = <5>; codec5_ep: endpoint { remote-endpoint = <&mbe2_ep>; }; }; /* [Codec2Codec] */ port@6 { bitclock-master; frame-master; - codec6_ep: endpoint { remote-endpoint = <&c2cf_ep>; }; }; - port@7 { codec7_ep: endpoint { remote-endpoint = <&c2cb_ep>; }; }; + reg = <6>; codec6_ep: endpoint { remote-endpoint = <&c2cf_ep>; }; }; + port@7 { reg = <7>; codec7_ep: endpoint { remote-endpoint = <&c2cb_ep>; }; }; /* [Codec2Codec-Multi] */ port@8 { bitclock-master; frame-master; - codec8_ep: endpoint { remote-endpoint = <&mc2c00_ep>; }; }; - port@9 { codec9_ep: endpoint { remote-endpoint = <&mc2c01_ep>; }; }; - port@10 { codec10_ep: endpoint { remote-endpoint = <&mc2c10_ep>; }; }; - port@11 { codec11_ep: endpoint { remote-endpoint = <&mc2c11_ep>; }; }; + reg = <8>; codec8_ep: endpoint { remote-endpoint = <&mc2c00_ep>; }; }; + port@9 { reg = <9>; codec9_ep: endpoint { remote-endpoint = <&mc2c01_ep>; }; }; + port@a { reg = <10>; codec10_ep: endpoint { remote-endpoint = <&mc2c10_ep>; }; }; + port@b { reg = <11>; codec11_ep: endpoint { remote-endpoint = <&mc2c11_ep>; }; }; /* [Semi-Multi] */ - port@12 { codec12_ep: endpoint { remote-endpoint = <&smcodec1_ep>; }; }; - port@13 { codec13_ep: endpoint { remote-endpoint = <&smcodec2_ep>; }; }; + port@c { reg = <12>; codec12_ep: endpoint { remote-endpoint = <&smcodec1_ep>; }; }; + port@d { reg = <13>; codec13_ep: endpoint { remote-endpoint = <&smcodec2_ep>; }; }; + }; }; }; -- 2.25.1

2 1

[PATCH] ASoC: Switch two more i2c drivers back to use .probe()
by Uwe Kleine-König 12 Jun '23

12 Jun '23

The previous conversion back to .probe() applied in commit 9abcd24002bf ("ASoC: Switch i2c drivers back to use .probe()") was created based on v6.3. Since then two more drivers were added which need to be convert back in the same way before eventually .probe_new() can be dropped from struct i2c_driver. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- Hello, cs35l56 was already in v6.4-rc1, ssm3515 is only in next. This patch is based on 4ac690bbae02 ("ASoC: ssm3515: Add new amp driver") which is the oldest commit in next that has both drivers. Best regards Uwe sound/soc/codecs/cs35l56-i2c.c | 2 +- sound/soc/codecs/ssm3515.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs35l56-i2c.c b/sound/soc/codecs/cs35l56-i2c.c index 295caad26224..ed2a41943d97 100644 --- a/sound/soc/codecs/cs35l56-i2c.c +++ b/sound/soc/codecs/cs35l56-i2c.c @@ -68,7 +68,7 @@ static struct i2c_driver cs35l56_i2c_driver = { .pm = &cs35l56_pm_ops_i2c_spi, }, .id_table = cs35l56_id_i2c, - .probe_new = cs35l56_i2c_probe, + .probe = cs35l56_i2c_probe, .remove = cs35l56_i2c_remove, }; diff --git a/sound/soc/codecs/ssm3515.c b/sound/soc/codecs/ssm3515.c index 784e890031a4..008cb3eb5758 100644 --- a/sound/soc/codecs/ssm3515.c +++ b/sound/soc/codecs/ssm3515.c @@ -439,7 +439,7 @@ static struct i2c_driver ssm3515_i2c_driver = { .name = "ssm3515", .of_match_table = of_match_ptr(ssm3515_of_match), }, - .probe_new = ssm3515_i2c_probe, + .probe = ssm3515_i2c_probe, }; module_i2c_driver(ssm3515_i2c_driver); base-commit: 4ac690bbae02e26e36e133becd86babb657126ef -- 2.39.2

2 1

[PATCH] ASoC: dt-bindings: cirrus,cs35l45: drop unneeded quotes
by Krzysztof Kozlowski 12 Jun '23

12 Jun '23

Cleanup bindings dropping unneeded quotes. Once all these are fixed, checking for this can be enabled in yamllint. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- .../devicetree/bindings/sound/cirrus,cs35l45.yaml | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/cirrus,cs35l45.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs35l45.yaml index 2ab74f995685..4c9acb8d4c4c 100644 --- a/Documentation/devicetree/bindings/sound/cirrus,cs35l45.yaml +++ b/Documentation/devicetree/bindings/sound/cirrus,cs35l45.yaml @@ -62,7 +62,7 @@ patternProperties: GPIO pin direction. Valid only when 'gpio-ctrl' is 1 0 = Output 1 = Input - $ref: "/schemas/types.yaml#/definitions/uint32" + $ref: /schemas/types.yaml#/definitions/uint32 minimum: 0 maximum: 1 default: 1 @@ -71,7 +71,7 @@ patternProperties: GPIO level. Valid only when 'gpio-ctrl' is 1 and 'gpio-dir' is 0 0 = Low 1 = High - $ref: "/schemas/types.yaml#/definitions/uint32" + $ref: /schemas/types.yaml#/definitions/uint32 minimum: 0 maximum: 1 default: 0 @@ -80,7 +80,7 @@ patternProperties: GPIO level. Valid only when 'gpio-ctrl' is 1 and 'gpio-dir' is 0 0 = CMOS 1 = Open Drain - $ref: "/schemas/types.yaml#/definitions/uint32" + $ref: /schemas/types.yaml#/definitions/uint32 minimum: 0 maximum: 1 default: 0 @@ -90,7 +90,7 @@ patternProperties: and 'gpio-dir' is 0 0 = Non-inverted, Active High 1 = Inverted, Active Low - $ref: "/schemas/types.yaml#/definitions/uint32" + $ref: /schemas/types.yaml#/definitions/uint32 minimum: 0 maximum: 1 default: 0 @@ -114,7 +114,7 @@ patternProperties: 0 = High impedance input 1 = Pin acts as a GPIO, direction controlled by 'gpio-dir' 2-7 = Reserved - $ref: "/schemas/types.yaml#/definitions/uint32" + $ref: /schemas/types.yaml#/definitions/uint32 minimum: 0 maximum: 7 default: 0 -- 2.34.1

3 2

[PATCH v2 0/2] ASoC: mediatek: mt8173: Fix error paths
by Ricardo Ribalda Delgado 12 Jun '23

12 Jun '23

ASoC: mediatek: mt8173, presented a couple of error paths errors, lets fix them. Signed-off-by: Ricardo Ribalda Delgado <ribalda(a)chromium.org> --- Changes in v2: - Fix other error paths bugs, Thanks Dan! - Link to v1: https://lore.kernel.org/r/20230612-mt8173-fixup-v1-1-81f5c90316d1@chromium.… --- Ricardo Ribalda Delgado (2): ASoC: mediatek: mt8173: Fix snd_soc_component_initialize error path ASoC: mediatek: mt8173: Fix irq error path sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) --- base-commit: 858fd168a95c5b9669aac8db6c14a9aeab446375 change-id: 20230612-mt8173-fixup-bc602a96dbad Best regards, -- Ricardo Ribalda Delgado <ribalda(a)chromium.org>

3 5

[PATCH] ASoC: SOF: ipc4/intel: Add missing mutex_unlock()
by Jyri Sarha 12 Jun '23

12 Jun '23

There was a missing mutex_unlock() in sof_ipc4_widget_free() use_chain_dma if-branch that caused a static analysis error. The branch should not be used in a normal working configuration and if its used its an indication of a bad topology. Add missing mutex_unlock() and a warning print if the if-branch is taken, and another warning print to a symmetric place in sof_ipc4_widget_setup(). Fixes: ca5ce0caa67fa9 ("ASoC: SOF: ipc4/intel: Add support for chained DMA") Reported-by: kernel test robot <lkp(a)intel.com> Reported-by: Dan Carpenter <error27(a)gmail.com> Link: https://lore.kernel.org/r/202303222050.dCw0fPCW-lkp@intel.com/ Signed-off-by: Jyri Sarha <jyri.sarha(a)linux.intel.com> --- sound/soc/sof/ipc4-topology.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index f1e1aed94da4..12775fcb6b54 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -1989,8 +1989,11 @@ static int sof_ipc4_widget_setup(struct snd_sof_dev *sdev, struct snd_sof_widget case snd_soc_dapm_scheduler: pipeline = swidget->private; - if (pipeline->use_chain_dma) + if (pipeline->use_chain_dma) { + dev_warn(sdev->dev, "use_chain_dma set for schduler %s", + swidget->widget->name); return 0; + } dev_dbg(sdev->dev, "pipeline: %d memory pages: %d\n", swidget->pipeline_id, pipeline->mem_usage); @@ -2145,8 +2148,12 @@ static int sof_ipc4_widget_free(struct snd_sof_dev *sdev, struct snd_sof_widget struct sof_ipc4_msg msg = {{ 0 }}; u32 header; - if (pipeline->use_chain_dma) + if (pipeline->use_chain_dma) { + dev_warn(sdev->dev, "use_chain_dma set for schduler %s", + swidget->widget->name); + mutex_unlock(&ipc4_data->pipeline_state_mutex); return 0; + } header = SOF_IPC4_GLB_PIPE_INSTANCE_ID(swidget->instance_id); header |= SOF_IPC4_MSG_TYPE_SET(SOF_IPC4_GLB_DELETE_PIPELINE); -- 2.34.1

4 3

[PATCH] ALSA: pcm: fix wait_time calculations
by Oswald Buddenhagen 12 Jun '23

12 Jun '23

... in wait_for_avail() and snd_pcm_drain(). t was calculated in seconds, so it would be pretty much always zero, to be subsequently de-facto ignored due to being max(t, 10)'d. And then it (i.e., 10) would be treated as secs, which doesn't seem right. However, fixing it to properly calculate msecs would potentially cause timeouts when using twice the period size for the default timeout (which seems reasonable to me), so instead use the buffer size plus 10 percent to be on the safe side ... but that still seems insufficient, presumably because the hardware typically needs a moment to fire up. To compensate for this, we up the minimal timeout to 100ms, which is still two orders of magnitude less than the bogus minimum. substream->wait_time was also misinterpreted as jiffies, despite being documented as being in msecs. Only the soc/sof driver sets it - to 500, which looks very much like msecs were intended. Speaking of which, shouldn't snd_pcm_drain() also use substream-> wait_time? As a drive-by, make the debug messages on timeout less confusing. Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- sound/core/pcm_lib.c | 11 +++++------ sound/core/pcm_native.c | 8 ++++---- 2 files changed, 9 insertions(+), 10 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 02fd65993e7e..af1eb136feb0 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1878,15 +1878,14 @@ static int wait_for_avail(struct snd_pcm_substream *substream, if (substream->wait_time) { wait_time = substream->wait_time; } else { - wait_time = 10; + wait_time = 100; if (runtime->rate) { - long t = runtime->period_size * 2 / - runtime->rate; + long t = runtime->buffer_size * 1100 / runtime->rate; wait_time = max(t, wait_time); } - wait_time = msecs_to_jiffies(wait_time * 1000); } + wait_time = msecs_to_jiffies(wait_time); } for (;;) { @@ -1934,8 +1933,8 @@ static int wait_for_avail(struct snd_pcm_substream *substream, } if (!tout) { pcm_dbg(substream->pcm, - "%s write error (DMA or IRQ trouble?)\n", - is_playback ? "playback" : "capture"); + "%s timeout (DMA or IRQ trouble?)\n", + is_playback ? "playback write" : "capture read"); err = -EIO; break; } diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 331380c2438b..94185267a7b9 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2159,12 +2159,12 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, if (runtime->no_period_wakeup) tout = MAX_SCHEDULE_TIMEOUT; else { - tout = 10; + tout = 100; if (runtime->rate) { - long t = runtime->period_size * 2 / runtime->rate; + long t = runtime->buffer_size * 1100 / runtime->rate; tout = max(t, tout); } - tout = msecs_to_jiffies(tout * 1000); + tout = msecs_to_jiffies(tout); } tout = schedule_timeout(tout); @@ -2187,7 +2187,7 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, result = -ESTRPIPE; else { dev_dbg(substream->pcm->card->dev, - "playback drain error (DMA or IRQ trouble?)\n"); + "playback drain timeout (DMA or IRQ trouble?)\n"); snd_pcm_stop(substream, SNDRV_PCM_STATE_SETUP); result = -EIO; } -- 2.40.0.152.g15d061e6df

3 5

[PATCH v2 1/2] ASoC: dt-bindings: qcom,wsa8840: Add WSA884x family of speakers
by Krzysztof Kozlowski 12 Jun '23

12 Jun '23

Add binding for WSA8840/WSA8845/WSA8845H smart speaker amplifiers used in Qualcomm QRD8550 board with SM8550 SoC. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Changes in v2: 1. Correct compatible (sdw version 1 -> 2). Cc: Patrick Lai <quic_plai(a)quicinc.com> Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- .../bindings/sound/qcom,wsa8840.yaml | 66 +++++++++++++++++++ 1 file changed, 66 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml diff --git a/Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml b/Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml new file mode 100644 index 000000000000..e6723c9e312a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml @@ -0,0 +1,66 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,wsa8840.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm WSA8840/WSA8845/WSA8845H smart speaker amplifier + +maintainers: + - Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> + - Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> + +description: + WSA884X is a family of Qualcomm Aqstic smart speaker amplifiers using + SoundWire digital audio interface. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: sdw20217020400 + + reg: + maxItems: 1 + + powerdown-gpios: + description: Powerdown/Shutdown line to use (pin SD_N) + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + vdd-1p8-supply: true + vdd-io-supply: true + +required: + - compatible + - reg + - powerdown-gpios + - '#sound-dai-cells' + - vdd-1p8-supply + - vdd-io-supply + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + soundwire-controller { + #address-cells = <2>; + #size-cells = <0>; + + speaker@0,1 { + compatible = "sdw20217020400"; + reg = <0 1>; + pinctrl-names = "default"; + pinctrl-0 = <&spkr_2_sd_n_active>; + powerdown-gpios = <&lpass_tlmm 18 GPIO_ACTIVE_LOW>; + #sound-dai-cells = <0>; + sound-name-prefix = "SpkrRight"; + vdd-1p8-supply = <&vreg_l15b_1p8>; + vdd-io-supply = <&vreg_l3g_1p2>; + }; + }; -- 2.34.1

2 4

[PATCH] ALSA: usb-audio: Fix broken resume due to UAC3 power state
by Takashi Iwai 12 Jun '23

12 Jun '23

As reported in the bugzilla below, the PM resume of a UAC3 device may fail due to the incomplete power state change, stuck at D1. The reason is that the driver expects the full D0 power state change only at hw_params, while the normal PCM resume procedure doesn't call hw_params. For fixing the bug, we add the same power state update to D0 at the prepare callback, which is certainly called by the resume procedure. Note that, with this change, the power state change in the hw_params becomes almost redundant, since snd_usb_hw_params() doesn't touch the parameters (at least it tires so). But dropping it is still a bit risky (e.g. we have the media-driver binding), so I leave the D0 power state change in snd_usb_hw_params() as is for now. Fixes: a0a4959eb4e9 ("ALSA: usb-audio: Operate UAC3 Power Domains in PCM callbacks") Cc: <stable(a)vger.kernel.org> Link: https://bugzilla.kernel.org/show_bug.cgi?id=217539 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/pcm.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index eec5232f9fb2..08bf535ed163 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -650,6 +650,10 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) goto unlock; } + ret = snd_usb_pcm_change_state(subs, UAC3_PD_STATE_D0); + if (ret < 0) + goto unlock; + again: if (subs->sync_endpoint) { ret = snd_usb_endpoint_prepare(chip, subs->sync_endpoint); -- 2.35.3

1 0

[PATCH] ALSA: seq: oss: Fix racy open/close of MIDI devices
by Takashi Iwai 12 Jun '23

12 Jun '23

Although snd_seq_oss_midi_open() and snd_seq_oss_midi_close() can be called concurrently from different code paths, we have no proper data protection against races. Introduce open_mutex to each seq_oss_midi object for avoiding the races. Reported-by: "Gong, Sishuai" <sishuai(a)purdue.edu> Closes: https://lore.kernel.org/r/7DC9AF71-F481-4ABA-955F-76C535661E33@purdue.edu Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/oss/seq_oss_midi.c | 35 +++++++++++++++++++------------ 1 file changed, 22 insertions(+), 13 deletions(-) diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c index 07efb38f58ac..f2940b29595f 100644 --- a/sound/core/seq/oss/seq_oss_midi.c +++ b/sound/core/seq/oss/seq_oss_midi.c @@ -37,6 +37,7 @@ struct seq_oss_midi { struct snd_midi_event *coder; /* MIDI event coder */ struct seq_oss_devinfo *devinfo; /* assigned OSSseq device */ snd_use_lock_t use_lock; + struct mutex open_mutex; }; @@ -172,6 +173,7 @@ snd_seq_oss_midi_check_new_port(struct snd_seq_port_info *pinfo) mdev->flags = pinfo->capability; mdev->opened = 0; snd_use_lock_init(&mdev->use_lock); + mutex_init(&mdev->open_mutex); /* copy and truncate the name of synth device */ strscpy(mdev->name, pinfo->name, sizeof(mdev->name)); @@ -322,15 +324,17 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) int perm; struct seq_oss_midi *mdev; struct snd_seq_port_subscribe subs; + int err; mdev = get_mididev(dp, dev); if (!mdev) return -ENODEV; + mutex_lock(&mdev->open_mutex); /* already used? */ if (mdev->opened && mdev->devinfo != dp) { - snd_use_lock_free(&mdev->use_lock); - return -EBUSY; + err = -EBUSY; + goto unlock; } perm = 0; @@ -340,14 +344,14 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) perm |= PERM_READ; perm &= mdev->flags; if (perm == 0) { - snd_use_lock_free(&mdev->use_lock); - return -ENXIO; + err = -ENXIO; + goto unlock; } /* already opened? */ if ((mdev->opened & perm) == perm) { - snd_use_lock_free(&mdev->use_lock); - return 0; + err = 0; + goto unlock; } perm &= ~mdev->opened; @@ -372,13 +376,17 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) } if (! mdev->opened) { - snd_use_lock_free(&mdev->use_lock); - return -ENXIO; + err = -ENXIO; + goto unlock; } mdev->devinfo = dp; + err = 0; + + unlock: + mutex_unlock(&mdev->open_mutex); snd_use_lock_free(&mdev->use_lock); - return 0; + return err; } /* @@ -393,10 +401,9 @@ snd_seq_oss_midi_close(struct seq_oss_devinfo *dp, int dev) mdev = get_mididev(dp, dev); if (!mdev) return -ENODEV; - if (! mdev->opened || mdev->devinfo != dp) { - snd_use_lock_free(&mdev->use_lock); - return 0; - } + mutex_lock(&mdev->open_mutex); + if (!mdev->opened || mdev->devinfo != dp) + goto unlock; memset(&subs, 0, sizeof(subs)); if (mdev->opened & PERM_WRITE) { @@ -415,6 +422,8 @@ snd_seq_oss_midi_close(struct seq_oss_devinfo *dp, int dev) mdev->opened = 0; mdev->devinfo = NULL; + unlock: + mutex_unlock(&mdev->open_mutex); snd_use_lock_free(&mdev->use_lock); return 0; } -- 2.35.3

1 0

[PATCH 0/2] ALSA: firewire: use 'GPL' string for module license
by Takashi Sakamoto 12 Jun '23

12 Jun '23

Hi, Nowadays, the use of "GPL" string in MODULE_LICENSE macro seems to obsolete "GPL v2" string. This series includes two patches to use it for modules in ALSA firewire stack contributed by two developers. This change does not pose any issue for modules contributed by Takashi Sakamoto, the author of this series, while it is advisable to have careful consideration for modules contributed by Clemens Ladisch, since the code related to license will be modified by a developer who is not the copyright holder. A commit bf7fbeeae6db ("module: Cure the MODULE_LICENSE "GPL" vs. "GPL v2" bogosity") adds detail documentation about the macro itself and clarifies the difference between "GPL" and "GPL v2". The macro is designed to provide sufficient information to kernel module loader and user space tools. Both strings have the same meanings, but the latter is legacy. In conclusion, any change made to the usage of the macro neither result in significant alterations to license nor copyright. Takashi Sakamoto (2): ALSA: firewire: use 'GPL' string for module license contributed by Takashi Sakamoto ALSA: firewire: use 'GPL' string for module license contributed by Clemens Ladisch sound/firewire/bebob/bebob.c | 2 +- sound/firewire/dice/dice.c | 2 +- sound/firewire/digi00x/digi00x.c | 2 +- sound/firewire/fireface/ff.c | 2 +- sound/firewire/fireworks/fireworks.c | 2 +- sound/firewire/isight.c | 2 +- sound/firewire/lib.c | 2 +- sound/firewire/motu/motu.c | 2 +- sound/firewire/oxfw/oxfw.c | 2 +- sound/firewire/tascam/tascam.c | 2 +- 10 files changed, 10 insertions(+), 10 deletions(-) -- 2.39.2

2 3

[PATCH v2 00/11] ASoC: qcom: audioreach: add compress offload support
by Srinivas Kandagatla 12 Jun '23

12 Jun '23

This patchset adds compressed offload support to Qualcomm audioreach drivers. Currently it supports AAC, MP3 and FALC along with gapless. Tested this on SM8450 and sc7280. thanks, srini Changes since v1: - removed lots of code duplication - moved ALSA patch out of this series. Mohammad Rafi Shaik (4): ASoC: qcom: SC7280: audioreach: Add sc7280 hardware param fixup callback ASoC: q6dsp: q6apm: add end of stream events ASoC: q6dsp: audioreach: Add support to set compress format params ASoC: q6dsp: audioreach: Add gapless feature support Srinivas Kandagatla (7): ASoC: q6dsp: audioreach: add helper function to set u32 param ASoC: q6dsp: audioreach: Add placeholder decoder for compress playback ASoC: q6dsp: q6apm-dai: Add open/free compress DAI callbacks ASoC: q6dsp: q6apm-dai: Add compress DAI and codec caps get callbacks ASoC: q6dsp: q6apm-dai: Add trigger/pointer compress DAI callbacks ASoC: q6dsp: q6apm-dai: Add compress set params and metadata DAI callbacks ASoC: q6dsp: q6apm-dai: Add mmap and copy compress DAI callbacks sound/soc/qcom/qdsp6/audioreach.c | 248 ++++++++++------- sound/soc/qcom/qdsp6/audioreach.h | 51 ++++ sound/soc/qcom/qdsp6/q6apm-dai.c | 445 ++++++++++++++++++++++++++++++ sound/soc/qcom/qdsp6/q6apm.c | 68 +++++ sound/soc/qcom/qdsp6/q6apm.h | 6 + sound/soc/qcom/sc7280.c | 23 +- 6 files changed, 745 insertions(+), 96 deletions(-) -- 2.21.0

2 13

[ PATCH v2 2/3] ASoC: loongson: Add Loongson ASoC Sound Card Support
by YingKun Meng 12 Jun '23

12 Jun '23

From: Yingkun Meng <mengyingkun(a)loongson.cn> The Loongson ASoC Sound Card is a general ASoC DAI Link driver that can be used for Loongson CPU DAI drivers and external CODECs. The driver supports the use of ACPI table to describe device resources. On loongson 7axxx platforms, the audio device is an ACPI device. Signed-off-by: Yingkun Meng <mengyingkun(a)loongson.cn> --- sound/soc/loongson/Kconfig | 10 ++ sound/soc/loongson/Makefile | 4 + sound/soc/loongson/loongson_card.c | 237 +++++++++++++++++++++++++++++ 3 files changed, 251 insertions(+) create mode 100644 sound/soc/loongson/loongson_card.c diff --git a/sound/soc/loongson/Kconfig b/sound/soc/loongson/Kconfig index 4478ac91e402..c175f9de19a8 100644 --- a/sound/soc/loongson/Kconfig +++ b/sound/soc/loongson/Kconfig @@ -13,4 +13,14 @@ config SND_SOC_LOONGSON_I2S_PCI The controller is found in loongson bridge chips or SoCs, and work as a PCI device. +config SND_SOC_LOONGSON_CARD + tristate "Loongson Sound Card Driver" + select SND_SOC_LOONGSON_I2S_PCI + help + Say Y or M if you want to add support for SoC audio using + loongson I2S controller. + + The driver add support for ALSA SoC Audio support using + loongson I2S controller. + endmenu diff --git a/sound/soc/loongson/Makefile b/sound/soc/loongson/Makefile index cfd0de1b1b22..a173a0fe17fe 100644 --- a/sound/soc/loongson/Makefile +++ b/sound/soc/loongson/Makefile @@ -2,3 +2,7 @@ #Platform Support snd-soc-loongson-i2s-pci-objs := loongson_i2s_pci.o loongson_i2s.o obj-$(CONFIG_SND_SOC_LOONGSON_I2S_PCI) += snd-soc-loongson-i2s-pci.o + +#Machine Support +snd-soc-loongson-card-objs := loongson_card.o +obj-$(CONFIG_SND_SOC_LOONGSON_CARD) += snd-soc-loongson-card.o diff --git a/sound/soc/loongson/loongson_card.c b/sound/soc/loongson/loongson_card.c new file mode 100644 index 000000000000..7f0ce8957a4f --- /dev/null +++ b/sound/soc/loongson/loongson_card.c @@ -0,0 +1,237 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Loongson ASoC Audio driver + * + * Copyright (C) 2022 Loongson Technology Corporation Limited + */ + +#include <linux/module.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> +#include <linux/acpi.h> +#include <linux/pci.h> +#include <sound/pcm_params.h> + +static char codec_name[SND_ACPI_I2C_ID_LEN]; + +struct loongson_card_data { + struct snd_soc_card snd_card; + unsigned int mclk_fs; +}; + +static int loongson_card_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct loongson_card_data *ls_card = snd_soc_card_get_drvdata(rtd->card); + int ret, mclk; + + if (ls_card->mclk_fs) { + mclk = ls_card->mclk_fs * params_rate(params); + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, + SND_SOC_CLOCK_OUT); + if (ret < 0) { + dev_err(codec_dai->dev, "cpu_dai clock not set\n"); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "codec_dai clock not set\n"); + return ret; + } + } + return 0; +} + +static const struct snd_soc_ops loongson_ops = { + .hw_params = loongson_card_hw_params, +}; + +SND_SOC_DAILINK_DEFS(analog, + DAILINK_COMP_ARRAY(COMP_CPU("loongson-i2s")), + DAILINK_COMP_ARRAY(COMP_EMPTY()), + DAILINK_COMP_ARRAY(COMP_EMPTY())); + +static struct snd_soc_dai_link loongson_dai_links[] = { + { + .name = "HiFi PAIF TX", + .stream_name = "Playback", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_IB_NF + | SND_SOC_DAIFMT_CBC_CFC, + SND_SOC_DAILINK_REG(analog), + .ops = &loongson_ops, + }, + { + .name = "HiFi PAIF RX", + .stream_name = "Capture", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_IB_NF + | SND_SOC_DAIFMT_CBC_CFC, + SND_SOC_DAILINK_REG(analog), + .ops = &loongson_ops, + }, +}; + +static int loongson_card_parse_acpi(struct loongson_card_data *data) +{ + struct snd_soc_card *card = &data->snd_card; + struct fwnode_handle *fwnode = card->dev->fwnode; + struct fwnode_reference_args args; + const char *codec_dai_name; + struct acpi_device *adev; + struct device *phy_dev; + int ret, i; + + /* fixup platform name based on reference node */ + memset(&args, 0, sizeof(args)); + ret = acpi_node_get_property_reference(fwnode, "cpu", 0, &args); + if (ACPI_FAILURE(ret) || !is_acpi_device_node(args.fwnode)) { + dev_err(card->dev, "No matching phy in ACPI table\n"); + return ret; + } + adev = to_acpi_device_node(args.fwnode); + phy_dev = acpi_get_first_physical_node(adev); + if (!phy_dev) + return -EPROBE_DEFER; + for (i = 0; i < card->num_links; i++) + loongson_dai_links[i].platforms->name = dev_name(phy_dev); + + /* fixup codec name based on reference node */ + memset(&args, 0, sizeof(args)); + ret = acpi_node_get_property_reference(fwnode, "codec", 0, &args); + if (ACPI_FAILURE(ret) || !is_acpi_device_node(args.fwnode)) { + dev_err(card->dev, "No matching phy in ACPI table\n"); + return ret; + } + adev = to_acpi_device_node(args.fwnode); + snprintf(codec_name, sizeof(codec_name), "i2c-%s", acpi_dev_name(adev)); + for (i = 0; i < card->num_links; i++) + loongson_dai_links[i].codecs->name = codec_name; + + device_property_read_string(card->dev, "codec-dai-name", + &codec_dai_name); + for (i = 0; i < card->num_links; i++) + loongson_dai_links[i].codecs->dai_name = codec_dai_name; + + return 0; +} + +static int loongson_card_parse_of(struct loongson_card_data *data) +{ + const char *cpu_dai_name, *codec_dai_name; + struct device_node *cpu, *codec; + struct snd_soc_card *card = &data->snd_card; + struct device *dev = card->dev; + struct of_phandle_args args; + int ret, i; + + cpu = of_get_child_by_name(dev->of_node, "cpu"); + if (!cpu) { + dev_err(dev, "platform property missing or invalid\n"); + return -EINVAL; + } + codec = of_get_child_by_name(dev->of_node, "codec"); + if (!codec) { + dev_err(dev, "audio-codec property missing or invalid\n"); + ret = -EINVAL; + goto err; + } + + ret = of_parse_phandle_with_args(cpu, "sound-dai", + "#sound-dai-cells", 0, &args); + if (ret) { + dev_err(dev, "codec node missing #sound-dai-cells\n"); + goto err; + } + for (i = 0; i < card->num_links; i++) + loongson_dai_links[i].cpus->of_node = args.np; + + ret = of_parse_phandle_with_args(codec, "sound-dai", + "#sound-dai-cells", 0, &args); + if (ret) { + dev_err(dev, "codec node missing #sound-dai-cells\n"); + goto err; + } + for (i = 0; i < card->num_links; i++) + loongson_dai_links[i].codecs->of_node = args.np; + + snd_soc_of_get_dai_name(cpu, &cpu_dai_name); + snd_soc_of_get_dai_name(codec, &codec_dai_name); + for (i = 0; i < card->num_links; i++) { + loongson_dai_links[i].cpus->dai_name = cpu_dai_name; + loongson_dai_links[i].codecs->dai_name = codec_dai_name; + } + of_node_put(cpu); + of_node_put(codec); + + return 0; + +err: + of_node_put(cpu); + of_node_put(codec); + return ret; +} + +static int loongson_asoc_card_probe(struct platform_device *pdev) +{ + struct loongson_card_data *ls_priv; + struct snd_soc_card *card; + int ret; + + ls_priv = devm_kzalloc(&pdev->dev, sizeof(*ls_priv), GFP_KERNEL); + if (!ls_priv) + return -ENOMEM; + + card = &ls_priv->snd_card; + + card->dev = &pdev->dev; + card->owner = THIS_MODULE; + card->dai_link = loongson_dai_links; + card->num_links = ARRAY_SIZE(loongson_dai_links); + snd_soc_card_set_drvdata(card, ls_priv); + + ret = device_property_read_string(&pdev->dev, "model", &card->name); + if (ret) { + dev_err(&pdev->dev, "Error parsing card name: %d\n", ret); + return ret; + } + ret = device_property_read_u32(&pdev->dev, "mclk-fs", &ls_priv->mclk_fs); + if (ret) { + dev_err(&pdev->dev, "Error parsing mclk-fs: %d\n", ret); + return ret; + } + + if (has_acpi_companion(&pdev->dev)) + ret = loongson_card_parse_acpi(ls_priv); + else + ret = loongson_card_parse_of(ls_priv); + if (ret < 0) + return ret; + + ret = devm_snd_soc_register_card(&pdev->dev, card); + + return ret; +} + +static const struct of_device_id loongson_asoc_dt_ids[] = { + { .compatible = "loongson,ls-audio-card" }, + { /* sentinel */ }, +}; +MODULE_DEVICE_TABLE(of, loongson_asoc_dt_ids); + +static struct platform_driver loongson_audio_driver = { + .probe = loongson_asoc_card_probe, + .driver = { + .name = "loongson-asoc-card", + .pm = &snd_soc_pm_ops, + .of_match_table = of_match_ptr(loongson_asoc_dt_ids), + }, +}; +module_platform_driver(loongson_audio_driver); + +MODULE_DESCRIPTION("Loongson ASoc Sound Card driver"); +MODULE_AUTHOR("Loongson Technology Corporation Limited"); +MODULE_LICENSE("GPL"); -- 2.33.0

1 0

[PATCH alsa-lib 0/3] ALSA-lib updates for UMP 1.1
by Takashi Iwai 12 Jun '23

12 Jun '23

Hi, this is a patch set for updating alsa-lib to catch up the changes for MIDI 2.0 UMP v1.1 features. The basic functionality remains, while the uapi header is updated a few API functions for the new features are added. Takashi === Takashi Iwai (3): uapi: Update rawmidi API to 2.0.4 ump: Add UMP 1.1 features seq: Add UMP 1.1 features include/seq.h | 3 ++ include/sound/uapi/asequencer.h | 5 ++- include/sound/uapi/asound.h | 21 ++++++++-- include/ump.h | 22 +++++++++++ include/ump_msg.h | 67 +++++++++++++++++++++++++++++++ src/Versions.in | 2 + src/rawmidi/ump.c | 70 +++++++++++++++++++++++++++++++++ src/seq/seq.c | 32 +++++++++++++++ 8 files changed, 218 insertions(+), 4 deletions(-) -- 2.35.3

1 3

[PATCH] ASoC: mediatek: mt8173: Fix error path
by Ricardo Ribalda Delgado 12 Jun '23

12 Jun '23

After reordering the irq probe, the error path was not properly done. Lets fix it. Reported-by: Dan Carpenter <dan.carpenter(a)linaro.org> Cc: stable(a)kernel.org Fixes: 4cbb264d4e91 ("ASoC: mediatek: mt8173: Enable IRQ when pdata is ready") Signed-off-by: Ricardo Ribalda Delgado <ribalda(a)chromium.org> --- 4cbb264d4e91 ("ASoC: mediatek: mt8173: Enable IRQ when pdata is ready") introduced a bug on the probe error path. Lets fix it. --- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c index f93c2ec8beb7..f2fdcc74069a 100644 --- a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c +++ b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c @@ -1070,6 +1070,10 @@ static int mt8173_afe_pcm_dev_probe(struct platform_device *pdev) afe->dev = &pdev->dev; + irq_id = platform_get_irq(pdev, 0); + if (irq_id <= 0) + return irq_id < 0 ? irq_id : -ENXIO; + afe->base_addr = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(afe->base_addr)) return PTR_ERR(afe->base_addr); @@ -1175,14 +1179,11 @@ static int mt8173_afe_pcm_dev_probe(struct platform_device *pdev) if (ret) goto err_cleanup_components; - irq_id = platform_get_irq(pdev, 0); - if (irq_id <= 0) - return irq_id < 0 ? irq_id : -ENXIO; ret = devm_request_irq(afe->dev, irq_id, mt8173_afe_irq_handler, 0, "Afe_ISR_Handle", (void *)afe); if (ret) { dev_err(afe->dev, "could not request_irq\n"); - goto err_pm_disable; + goto err_cleanup_components; } dev_info(&pdev->dev, "MT8173 AFE driver initialized.\n"); --- base-commit: 858fd168a95c5b9669aac8db6c14a9aeab446375 change-id: 20230612-mt8173-fixup-bc602a96dbad Best regards, -- Ricardo Ribalda Delgado <ribalda(a)chromium.org>

2 1

[PATCH 00/10] ALSA: Catch up MIDI 2.0 updates for UMP 1.1
by Takashi Iwai 12 Jun '23

12 Jun '23

Hi, as the updated MIDI 2.0 spec has been published freshly [*], this is a catch up to add the support for new specs, especially UMP v1.1 features, on Linux kernel. The new UMP v1.1 introduced the concept of Function Blocks (FB), which is a kind of superset of USB MIDI 2.0 Group Terminal Blocks (GTB). The patch set adds the support for FB as the primary information source while keeping the parse of GTB as fallback. Also UMP v1.1 supports the groupless messages, the protocol switch, static FBs, and other new fundamental features, and those are supported as well. Takashi [*] https://www.midi.org/midi-articles/details-about-midi-2-0-midi-ci-profiles-… === Takashi Iwai (10): ALSA: ump: Add more attributes to UMP EP and FB info ALSA: ump: Support UMP Endpoint and Function Block parsing ALSA: usb-audio: Parse UMP Endpoint and Function Blocks at first ALSA: usb-audio: Add midi2_ump_probe option ALSA: seq: ump: Handle groupless messages ALSA: seq: ump: Handle FB info update ALSA: seq: ump: Notify port changes to system port ALSA: seq: ump: Notify UMP protocol change to sequencer ALSA: ump: Add info flag bit for static blocks ALSA: docs: Update MIDI 2.0 documentation for UMP 1.1 enhancement Documentation/sound/designs/midi-2.0.rst | 39 +- include/sound/ump.h | 89 +++++ include/sound/ump_msg.h | 225 +++++++++++ include/uapi/sound/asequencer.h | 5 +- include/uapi/sound/asound.h | 21 +- sound/core/seq/seq_system.c | 1 + sound/core/seq/seq_ump_client.c | 79 +++- sound/core/seq/seq_ump_convert.c | 3 + sound/core/ump.c | 470 +++++++++++++++++++++++ sound/usb/midi2.c | 42 +- 10 files changed, 962 insertions(+), 12 deletions(-) -- 2.35.3

1 10

[PATCH 1/3] ASoC: Add support for Loongson I2S controller
by YingKun Meng 12 Jun '23

12 Jun '23

From: Yingkun Meng <mengyingkun(a)loongson.cn> Loongson I2S controller is found on 7axxx/2kxxx chips from loongson, it is a PCI device with two private DMA controllers, one for playback, the other for capture. The driver supports the use of DTS or ACPI to describe device resources. Signed-off-by: Yingkun Meng <mengyingkun(a)loongson.cn> --- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/loongson/Kconfig | 15 + sound/soc/loongson/Makefile | 4 + sound/soc/loongson/loongson_i2s.c | 209 +++++++++++ sound/soc/loongson/loongson_i2s.h | 70 ++++ sound/soc/loongson/loongson_i2s_pci.c | 478 ++++++++++++++++++++++++++ 7 files changed, 778 insertions(+) create mode 100644 sound/soc/loongson/Kconfig create mode 100644 sound/soc/loongson/Makefile create mode 100644 sound/soc/loongson/loongson_i2s.c create mode 100644 sound/soc/loongson/loongson_i2s.h create mode 100644 sound/soc/loongson/loongson_i2s_pci.c diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 4b6e5a802880..bfa9622e1ab1 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -79,6 +79,7 @@ source "sound/soc/google/Kconfig" source "sound/soc/hisilicon/Kconfig" source "sound/soc/jz4740/Kconfig" source "sound/soc/kirkwood/Kconfig" +source "sound/soc/loongson/Kconfig" source "sound/soc/img/Kconfig" source "sound/soc/intel/Kconfig" source "sound/soc/mediatek/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 9d9b228e4508..8376fdb217ed 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -46,6 +46,7 @@ obj-$(CONFIG_SND_SOC) += fsl/ obj-$(CONFIG_SND_SOC) += google/ obj-$(CONFIG_SND_SOC) += hisilicon/ obj-$(CONFIG_SND_SOC) += jz4740/ +obj-$(CONFIG_SND_SOC) += loongson/ obj-$(CONFIG_SND_SOC) += img/ obj-$(CONFIG_SND_SOC) += intel/ obj-$(CONFIG_SND_SOC) += mediatek/ diff --git a/sound/soc/loongson/Kconfig b/sound/soc/loongson/Kconfig new file mode 100644 index 000000000000..fd477e7da6e0 --- /dev/null +++ b/sound/soc/loongson/Kconfig @@ -0,0 +1,15 @@ +# SPDX-License-Identifier: GPL-2.0 +menu "SoC Audio for Loongson CPUs" + depends on LOONGARCH || COMPILE_TEST + +config SND_SOC_LOONGSON_I2S_PCI + tristate "Loongson I2S-PCI Device Driver" + select REGMAP_MMIO + help + Say Y or M if you want to add support for I2S driver for + Loongson I2S controller. + + The controller is found in loongson bridge chips or SoCs, + and work as a PCI device. + +endmenu diff --git a/sound/soc/loongson/Makefile b/sound/soc/loongson/Makefile new file mode 100644 index 000000000000..cfd0de1b1b22 --- /dev/null +++ b/sound/soc/loongson/Makefile @@ -0,0 +1,4 @@ +# SPDX-License-Identifier: GPL-2.0 +#Platform Support +snd-soc-loongson-i2s-pci-objs := loongson_i2s_pci.o loongson_i2s.o +obj-$(CONFIG_SND_SOC_LOONGSON_I2S_PCI) += snd-soc-loongson-i2s-pci.o diff --git a/sound/soc/loongson/loongson_i2s.c b/sound/soc/loongson/loongson_i2s.c new file mode 100644 index 000000000000..00793964237f --- /dev/null +++ b/sound/soc/loongson/loongson_i2s.c @@ -0,0 +1,209 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Loongson-2K I2S master mode driver + * + * Copyright (C) 2022 Loongson Technology Corporation Limited + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/delay.h> +#include <linux/pm_runtime.h> +#include <linux/dma-mapping.h> +#include <sound/soc.h> +#include <linux/regmap.h> +#include <sound/pcm_params.h> +#include "loongson_i2s.h" + +#define LOONGSON_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +static int loongson_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct loongson_i2s *i2s = snd_soc_dai_get_drvdata(dai); + u32 val; + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + /* Enable MCLK */ + if (i2s->rev_id == 1) { + regmap_update_bits(i2s->regmap, LS_I2S_CTRL, + I2S_CTRL_MCLK_EN, + I2S_CTRL_MCLK_EN); + regmap_read_poll_timeout_atomic(i2s->regmap, + LS_I2S_CTRL, val, + !(val & I2S_CTRL_MCLK_READY), + 10, 2000); + } + + /* Enable master mode */ + regmap_update_bits(i2s->regmap, LS_I2S_CTRL, I2S_CTRL_MASTER, + I2S_CTRL_MASTER); + if (i2s->rev_id == 1) { + regmap_read_poll_timeout_atomic(i2s->regmap, + LS_I2S_CTRL, val, + !(val & I2S_CTRL_CLK_READY), + 10, 2000); + } + fallthrough; + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + regmap_update_bits(i2s->regmap, LS_I2S_CTRL, + I2S_CTRL_TX_EN | I2S_CTRL_TX_DMA_EN, + I2S_CTRL_TX_EN | I2S_CTRL_TX_DMA_EN); + else + regmap_update_bits(i2s->regmap, LS_I2S_CTRL, + I2S_CTRL_RX_EN | I2S_CTRL_RX_DMA_EN, + I2S_CTRL_RX_EN | I2S_CTRL_RX_DMA_EN); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + regmap_update_bits(i2s->regmap, LS_I2S_CTRL, + I2S_CTRL_TX_EN | I2S_CTRL_TX_DMA_EN, 0); + else + regmap_update_bits(i2s->regmap, LS_I2S_CTRL, + I2S_CTRL_RX_EN | I2S_CTRL_RX_DMA_EN, 0); + break; + default: + ret = -EINVAL; + } + + return ret; +} + +static int loongson_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct loongson_i2s *i2s = snd_soc_dai_get_drvdata(dai); + u32 clk_rate = i2s->clk_rate; + u32 sysclk = i2s->sysclk; + u32 bits = params_width(params); + u32 chans = params_channels(params); + u32 fs = params_rate(params); + u32 bclk_ratio, mclk_ratio; + u32 mclk_ratio_frac; + u32 val = 0; + + if (i2s->rev_id == 0) { + bclk_ratio = DIV_ROUND_CLOSEST(clk_rate, + (bits * chans * fs * 2)) - 1; + mclk_ratio = DIV_ROUND_CLOSEST(clk_rate, (sysclk * 2)) - 1; + + /* According to 2k1000LA user manual, set bits == depth */ + val |= (bits << 24); + val |= (bits << 16); + val |= (bclk_ratio << 8); + val |= mclk_ratio; + regmap_write(i2s->regmap, LS_I2S_CFG, val); + } else if (i2s->rev_id == 1) { + bclk_ratio = DIV_ROUND_CLOSEST(sysclk, + (bits * chans * fs * 2)) - 1; + mclk_ratio = clk_rate / sysclk; + mclk_ratio_frac = DIV_ROUND_CLOSEST(((u64)clk_rate << 16), + sysclk) - (mclk_ratio << 16); + + regmap_read(i2s->regmap, LS_I2S_CFG, &val); + val |= (bits << 24); + val |= (bclk_ratio << 8); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + val |= (bits << 16); + else + val |= bits; + regmap_write(i2s->regmap, LS_I2S_CFG, val); + + val = (mclk_ratio_frac << 16) | mclk_ratio; + regmap_write(i2s->regmap, LS_I2S_CFG1, val); + } else + dev_err(i2s->dev, "I2S revision invalid\n"); + + return 0; +} + +static int loongson_i2s_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct loongson_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + i2s->sysclk = freq; + + return 0; +} + +static const struct snd_soc_dai_ops loongson_i2s_dai_ops = { + .trigger = loongson_i2s_trigger, + .hw_params = loongson_i2s_hw_params, + .set_sysclk = loongson_i2s_set_dai_sysclk, +}; + +static int loongson_i2s_dai_probe(struct snd_soc_dai *cpu_dai) +{ + struct loongson_i2s *i2s = dev_get_drvdata(cpu_dai->dev); + + snd_soc_dai_init_dma_data(cpu_dai, &i2s->playback_dma_data, + &i2s->capture_dma_data); + snd_soc_dai_set_drvdata(cpu_dai, i2s); + + return 0; +} + +struct snd_soc_dai_driver loongson_i2s_dai = { + .name = "loongson-i2s", + .probe = loongson_i2s_dai_probe, + .playback = { + .stream_name = "CPU-Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = LOONGSON_I2S_FORMATS, + }, + .capture = { + .stream_name = "CPU-Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = LOONGSON_I2S_FORMATS, + }, + .ops = &loongson_i2s_dai_ops, + .symmetric_rate = 1, +}; + +static int i2s_suspend(struct device *dev) +{ + struct loongson_i2s *i2s = dev_get_drvdata(dev); + + regcache_cache_only(i2s->regmap, true); + + return 0; +} + +static int i2s_resume(struct device *dev) +{ + struct loongson_i2s *i2s = dev_get_drvdata(dev); + int ret; + + regcache_cache_only(i2s->regmap, false); + regcache_mark_dirty(i2s->regmap); + ret = regcache_sync(i2s->regmap); + + return ret; +} + +const struct dev_pm_ops loongson_i2s_pm = { + SET_SYSTEM_SLEEP_PM_OPS(i2s_suspend, i2s_resume) +}; + +void loongson_i2s_init(struct loongson_i2s *i2s) +{ + if (i2s->rev_id == 1) { + regmap_write(i2s->regmap, LS_I2S_CTRL, I2S_CTRL_RESET); + udelay(200); + } +} diff --git a/sound/soc/loongson/loongson_i2s.h b/sound/soc/loongson/loongson_i2s.h new file mode 100644 index 000000000000..a743c34b8968 --- /dev/null +++ b/sound/soc/loongson/loongson_i2s.h @@ -0,0 +1,70 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * ALSA I2S interface for the Loongson chip + * + */ +#ifndef _LOONGSON_I2S_H +#define _LOONGSON_I2S_H + +#include <linux/regmap.h> +#include <sound/dmaengine_pcm.h> + +/* I2S Common Registers */ +#define LS_I2S_VER 0x00 /* I2S Version */ +#define LS_I2S_CFG 0x04 /* I2S Config */ +#define LS_I2S_CTRL 0x08 /* I2S Control */ +#define LS_I2S_RX_DATA 0x0C /* I2S DMA RX Address */ +#define LS_I2S_TX_DATA 0x10 /* I2S DMA TX Address */ + +/* 2K2000 I2S Specify Registers */ +#define LS_I2S_CFG1 0x14 /* I2S Config1 */ + +/* 7A2000 I2S Specify Registers */ +#define LS_I2S_TX_ORDER 0x100 /* TX DMA Order */ +#define LS_I2S_RX_ORDER 0x110 /* RX DMA Order */ + +/* Loongson I2S Control Register */ +#define I2S_CTRL_MCLK_READY (1 << 16) /* MCLK ready */ +#define I2S_CTRL_MASTER (1 << 15) /* Master mode */ +#define I2S_CTRL_MSB (1 << 14) /* MSB bit order */ +#define I2S_CTRL_RX_EN (1 << 13) /* RX enable */ +#define I2S_CTRL_TX_EN (1 << 12) /* TX enable */ +#define I2S_CTRL_RX_DMA_EN (1 << 11) /* DMA RX enable */ +#define I2S_CTRL_CLK_READY (1 << 8) /* BCLK ready */ +#define I2S_CTRL_TX_DMA_EN (1 << 7) /* DMA TX enable */ +#define I2S_CTRL_RESET (1 << 4) /* Controller soft reset */ +#define I2S_CTRL_MCLK_EN (1 << 3) /* Enable MCLK */ +#define I2S_CTRL_RX_INT_EN (1 << 1) /* RX interrupt enable */ +#define I2S_CTRL_TX_INT_EN (1 << 0) /* TX interrupt enable */ + +#define LS_I2S_DRVNAME "loongson-i2s" + +struct loongson_dma_data { + u64 dev_addr; /* device physical address for DMA */ + void __iomem *order_addr; /* DMA order register */ + u32 irq; /* DMA irq */ +}; + +struct loongson_i2s { + struct device *dev; + union { + struct snd_dmaengine_dai_dma_data playback_dma_data; + struct loongson_dma_data tx_dma_data; + }; + union { + struct snd_dmaengine_dai_dma_data capture_dma_data; + struct loongson_dma_data rx_dma_data; + }; + struct regmap *regmap; + void __iomem *reg_base; + u32 rev_id; + u32 clk_rate; + u32 sysclk; +}; + +extern const struct dev_pm_ops loongson_i2s_pm; +extern struct snd_soc_dai_driver loongson_i2s_dai; + +void loongson_i2s_init(struct loongson_i2s *i2s); + +#endif diff --git a/sound/soc/loongson/loongson_i2s_pci.c b/sound/soc/loongson/loongson_i2s_pci.c new file mode 100644 index 000000000000..e4b8c654b12c --- /dev/null +++ b/sound/soc/loongson/loongson_i2s_pci.c @@ -0,0 +1,478 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Loongson-2K I2S master mode driver + * + * Copyright (C) 2022 Loongson Technology Corporation Limited + */ + +#include <linux/module.h> +#include <linux/delay.h> +#include <linux/pm_runtime.h> +#include <linux/dma-mapping.h> +#include <linux/acpi.h> +#include <linux/pci.h> +#include <sound/soc.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include "loongson_i2s.h" + +/* DMA dma_order Register */ +#define DMA_ORDER_STOP (1 << 4) /* DMA stop */ +#define DMA_ORDER_START (1 << 3) /* DMA start */ +#define DMA_ORDER_ASK_VALID (1 << 2) /* DMA ask valid flag */ +#define DMA_ORDER_AXI_UNCO (1 << 1) /* Uncache access */ +#define DMA_ORDER_ADDR_64 (1 << 0) /* 64bits address support */ + +#define DMA_ORDER_ASK_MASK (~0x1fUL) /* Ask addr mask */ +#define DMA_ORDER_CTRL_MASK (0x0fUL) /* Control mask */ + +#define BAR_NUM 0 + +/* + * DMA registers descriptor. + */ +struct loongson_dma_desc { + u32 order; /* Next descriptor address register */ + u32 saddr; /* Source address register */ + u32 daddr; /* Device address register */ + u32 length; /* Total length register */ + u32 step_length; /* Memory stride register */ + u32 step_times; /* Repeat time register */ + u32 cmd; /* Command register */ + u32 stats; /* Status register */ + u32 order_hi; /* Next descriptor high address register */ + u32 saddr_hi; /* High source address register */ + u32 res[6]; /* Reserved */ +}; + +struct loongson_runtime_data { + struct loongson_dma_data *dma_data; + + struct loongson_dma_desc *dma_desc_arr; + dma_addr_t dma_desc_arr_phy; + + struct loongson_dma_desc *dma_pos_desc; + dma_addr_t dma_pos_desc_phy; +}; + +static const struct snd_pcm_hardware ls_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_PAUSE, + .formats = (SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | + SNDRV_PCM_FMTBIT_S24_LE), + .period_bytes_min = 128, + .period_bytes_max = 128 * 1024, + .periods_min = 1, + .periods_max = PAGE_SIZE / sizeof(struct loongson_dma_desc), + .buffer_bytes_max = 1024 * 1024, +}; + +static struct +loongson_dma_desc *dma_desc_save(struct loongson_runtime_data *prtd) +{ + void __iomem *order_reg = prtd->dma_data->order_addr; + u64 val; + + val = (u64)prtd->dma_pos_desc_phy & DMA_ORDER_ASK_MASK; + val |= (readq(order_reg) & DMA_ORDER_CTRL_MASK); + val |= DMA_ORDER_ASK_VALID; + writeq(val, order_reg); + + while (readl(order_reg) & DMA_ORDER_ASK_VALID) + udelay(2); + + return prtd->dma_pos_desc; +} + +static int loongson_pcm_trigger(struct snd_soc_component *component, + struct snd_pcm_substream *substream, int cmd) +{ + struct loongson_runtime_data *prtd = substream->runtime->private_data; + struct device *dev = substream->pcm->card->dev; + void __iomem *order_reg = prtd->dma_data->order_addr; + u64 val; + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + val = prtd->dma_pos_desc_phy & DMA_ORDER_ASK_MASK; + if (dev->coherent_dma_mask == DMA_BIT_MASK(64)) + val |= DMA_ORDER_ADDR_64; + else + val &= ~DMA_ORDER_ADDR_64; + val |= (readq(order_reg) & DMA_ORDER_CTRL_MASK); + val |= DMA_ORDER_START; + writeq(val, order_reg); + + while ((readl(order_reg) & DMA_ORDER_START)) + udelay(2); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + dma_desc_save(prtd); + + /* dma stop */ + val = readq(order_reg) | DMA_ORDER_STOP; + writeq(val, order_reg); + udelay(1000); + + break; + default: + ret = -EINVAL; + } + + return ret; +} + +static int loongson_pcm_hw_params(struct snd_soc_component *component, + struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct loongson_runtime_data *prtd = runtime->private_data; + size_t buf_len = params_buffer_bytes(params); + size_t period_len = params_period_bytes(params); + dma_addr_t order_addr, mem_addr; + struct loongson_dma_desc *desc; + u32 num_periods; + int i; + + if (buf_len % period_len) { + pr_err("buf len not multiply of period len\n"); + return -EINVAL; + } + + num_periods = buf_len / period_len; + if (!num_periods) { + pr_err("dma data too small\n"); + return -EINVAL; + } + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = buf_len; + + /* initialize dma descriptor array */ + mem_addr = runtime->dma_addr; + order_addr = prtd->dma_desc_arr_phy; + for (i = 0; i < num_periods; i++) { + desc = &prtd->dma_desc_arr[i]; + + /* next descriptor physical address */ + order_addr += sizeof(*desc); + desc->order = order_addr | BIT(0); + desc->order_hi = order_addr >> 32; + + desc->saddr = mem_addr; + desc->saddr_hi = mem_addr >> 32; + desc->daddr = prtd->dma_data->dev_addr; + + desc->cmd = BIT(0); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + desc->cmd |= BIT(12); + + desc->length = period_len >> 2; + desc->step_length = 0; + desc->step_times = 1; + + mem_addr += period_len; + } + if (num_periods > 0) { + desc = &prtd->dma_desc_arr[num_periods - 1]; + desc->order = prtd->dma_desc_arr_phy | BIT(0); + desc->order_hi = prtd->dma_desc_arr_phy >> 32; + } + + /* init position descriptor */ + *prtd->dma_pos_desc = *prtd->dma_desc_arr; + + return 0; +} + +static snd_pcm_uframes_t +loongson_pcm_pointer(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct loongson_runtime_data *prtd = runtime->private_data; + struct loongson_dma_desc *desc; + snd_pcm_uframes_t x; + u64 addr; + + desc = dma_desc_save(prtd); + addr = ((u64)desc->saddr_hi << 32) | desc->saddr; + + x = bytes_to_frames(runtime, addr - runtime->dma_addr); + if (x == runtime->buffer_size) + x = 0; + return x; +} + +static irqreturn_t loongson_pcm_dma_irq(int irq, void *devid) +{ + struct snd_pcm_substream *substream = devid; + + snd_pcm_period_elapsed(substream); + return IRQ_HANDLED; +} + +static int loongson_pcm_open(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_card *card = substream->pcm->card; + struct loongson_runtime_data *prtd; + struct loongson_dma_data *dma_data; + int ret; + + /* + * For mysterious reasons (and despite what the manual says) + * playback samples are lost if the DMA count is not a multiple + * of the DMA burst size. Let's add a rule to enforce that. + */ + snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 128); + snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 128); + snd_pcm_hw_constraint_integer(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + snd_soc_set_runtime_hwparams(substream, &ls_pcm_hardware); + + prtd = kzalloc(sizeof(*prtd), GFP_KERNEL); + if (!prtd) + return -ENOMEM; + + prtd->dma_desc_arr = dma_alloc_coherent(card->dev, PAGE_SIZE, + &prtd->dma_desc_arr_phy, + GFP_KERNEL); + if (!prtd->dma_desc_arr) { + ret = -ENOMEM; + goto desc_err; + } + + prtd->dma_pos_desc = dma_alloc_coherent(card->dev, + sizeof(*prtd->dma_pos_desc), + &prtd->dma_pos_desc_phy, + GFP_KERNEL); + if (!prtd->dma_pos_desc) { + ret = -ENOMEM; + goto pos_err; + } + + dma_data = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); + ret = request_irq(dma_data->irq, loongson_pcm_dma_irq, + IRQF_TRIGGER_RISING, LS_I2S_DRVNAME, substream); + if (ret < 0) + goto irq_err; + + prtd->dma_data = dma_data; + substream->runtime->private_data = prtd; + + return 0; +irq_err: + dma_free_coherent(card->dev, sizeof(*prtd->dma_pos_desc), + prtd->dma_pos_desc, prtd->dma_pos_desc_phy); +pos_err: + dma_free_coherent(card->dev, PAGE_SIZE, prtd->dma_desc_arr, + prtd->dma_desc_arr_phy); +desc_err: + kfree(prtd); + + return ret; +} + +static int loongson_pcm_close(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_card *card = substream->pcm->card; + struct loongson_runtime_data *prtd = substream->runtime->private_data; + struct loongson_dma_data *dma_data; + + dma_free_coherent(card->dev, PAGE_SIZE, prtd->dma_desc_arr, + prtd->dma_desc_arr_phy); + + dma_free_coherent(card->dev, sizeof(*prtd->dma_pos_desc), + prtd->dma_pos_desc, prtd->dma_pos_desc_phy); + + dma_data = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); + free_irq(dma_data->irq, substream); + + kfree(prtd); + return 0; +} + +static int loongson_pcm_mmap(struct snd_soc_component *component, + struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + return remap_pfn_range(vma, vma->vm_start, + substream->dma_buffer.addr >> PAGE_SHIFT, + vma->vm_end - vma->vm_start, vma->vm_page_prot); +} + +static int loongson_pcm_new(struct snd_soc_component *component, + struct snd_soc_pcm_runtime *rtd) +{ + struct snd_pcm *pcm = rtd->pcm; + + return snd_pcm_set_fixed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV, + pcm->card->dev, + ls_pcm_hardware.buffer_bytes_max); +} + +static const struct snd_soc_component_driver loongson_i2s_component = { + .name = LS_I2S_DRVNAME, + .open = loongson_pcm_open, + .close = loongson_pcm_close, + .hw_params = loongson_pcm_hw_params, + .trigger = loongson_pcm_trigger, + .pointer = loongson_pcm_pointer, + .mmap = loongson_pcm_mmap, + .pcm_construct = loongson_pcm_new, +}; + +static const struct regmap_config loongson_i2s_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = 0x110, + .cache_type = REGCACHE_FLAT, +}; + +static int loongson_i2s_pci_probe(struct pci_dev *pdev, + const struct pci_device_id *pid) +{ + const struct fwnode_handle *fwnode = pdev->dev.fwnode; + struct loongson_dma_data *tx_data, *rx_data; + struct loongson_i2s *i2s; + int ret; + + if (pci_enable_device(pdev)) { + dev_err(&pdev->dev, "pci_enable_device failed\n"); + return -ENODEV; + } + + ret = pci_request_region(pdev, BAR_NUM, LS_I2S_DRVNAME); + if (ret) { + dev_err(&pdev->dev, "request regions failed %d\n", ret); + goto err_disable; + } + + i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL); + if (!i2s) { + ret = -ENOMEM; + goto err_release; + } + i2s->rev_id = pdev->revision; + i2s->dev = &pdev->dev; + pci_set_drvdata(pdev, i2s); + + i2s->reg_base = pci_iomap(pdev, BAR_NUM, 0); + if (!i2s->reg_base) { + dev_err(&pdev->dev, "pci_iomap_error\n"); + ret = -EIO; + goto err_release; + } + + i2s->regmap = devm_regmap_init_mmio(&pdev->dev, i2s->reg_base, + &loongson_i2s_regmap_config); + if (IS_ERR(i2s->regmap)) { + dev_err(&pdev->dev, "Failed to initialize register map"); + ret = PTR_ERR(i2s->regmap); + goto err_unmap; + } + + tx_data = &i2s->tx_dma_data; + rx_data = &i2s->rx_dma_data; + + tx_data->dev_addr = (u64)i2s->reg_base + LS_I2S_TX_DATA; + tx_data->order_addr = i2s->reg_base + LS_I2S_TX_ORDER; + + rx_data->dev_addr = (u64)i2s->reg_base + LS_I2S_RX_DATA; + rx_data->order_addr = i2s->reg_base + LS_I2S_RX_ORDER; + + tx_data->irq = fwnode_irq_get_byname(fwnode, "tx"); + if (tx_data->irq < 0) { + dev_err(&pdev->dev, "dma tx irq invalid\n"); + ret = tx_data->irq; + goto err_unmap; + } + + rx_data->irq = fwnode_irq_get_byname(fwnode, "rx"); + if (rx_data->irq < 0) { + dev_err(&pdev->dev, "dma rx irq invalid\n"); + ret = rx_data->irq; + goto err_unmap; + } + + device_property_read_u32(&pdev->dev, "clock-frequency", &i2s->clk_rate); + if (!i2s->clk_rate) { + dev_err(&pdev->dev, "clock-frequency property invalid\n"); + ret = -EINVAL; + goto err_unmap; + } + + dma_set_mask_and_coherent(&pdev->dev, DMA_BIT_MASK(64)); + + dev_set_name(&pdev->dev, "%s", loongson_i2s_dai.name); + + loongson_i2s_init(i2s); + + ret = devm_snd_soc_register_component(&pdev->dev, + &loongson_i2s_component, + &loongson_i2s_dai, 1); + if (ret) { + dev_err(&pdev->dev, "register DAI failed %d\n", ret); + goto err_unmap; + } + + return 0; + +err_unmap: + pci_iounmap(pdev, i2s->reg_base); +err_release: + pci_release_region(pdev, BAR_NUM); +err_disable: + pci_disable_device(pdev); + return ret; +} + +static void loongson_i2s_pci_remove(struct pci_dev *pdev) +{ + struct loongson_i2s *i2s = dev_get_drvdata(&pdev->dev); + + pci_iounmap(pdev, i2s->reg_base); + pci_release_region(pdev, BAR_NUM); + pci_disable_device(pdev); +} + +static const struct pci_device_id loongson_i2s_ids[] = { + { PCI_DEVICE(PCI_VENDOR_ID_LOONGSON, 0x7a27) }, + { }, +}; +MODULE_DEVICE_TABLE(pci, loongson_i2s_ids); + +static struct pci_driver loongson_i2s_driver = { + .name = "loongson-i2s-pci", + .id_table = loongson_i2s_ids, + .probe = loongson_i2s_pci_probe, + .remove = loongson_i2s_pci_remove, + .driver = { + .owner = THIS_MODULE, + .pm = pm_sleep_ptr(&loongson_i2s_pm), + }, +}; +module_pci_driver(loongson_i2s_driver); + +MODULE_DESCRIPTION("Loongson I2S Master Mode ASoC Driver"); +MODULE_AUTHOR("Loongson Technology Corporation Limited"); +MODULE_LICENSE("GPL"); base-commit: 1fbcc5ab1c7a172ef1159b154c296fe1e9ce209b -- 2.33.0

4 12

[tiwai-sound:for-next 14/74] include/sound/ump_msg.h:196:13: error: duplicate member 'reserved'
by kernel test robot 11 Jun '23

11 Jun '23

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 15253079ca300160c92c9c0ee2541836463043f6 commit: 0b5288f5fe63eab687c14e5940b9e0d532b129f2 [14/74] ALSA: ump: Add legacy raw MIDI support config: parisc-allyesconfig (https://download.01.org/0day-ci/archive/20230611/202306112301.dMok4XSA-lkp@…) compiler: hppa-linux-gcc (GCC) 12.3.0 reproduce (this is a W=1 build): mkdir -p ~/bin wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross # https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/commit/?id=… git remote add tiwai-sound https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git git fetch --no-tags tiwai-sound for-next git checkout 0b5288f5fe63eab687c14e5940b9e0d532b129f2 # save the config file mkdir build_dir && cp config build_dir/.config COMPILER_INSTALL_PATH=$HOME/0day COMPILER=gcc-12.3.0 ~/bin/make.cross W=1 O=build_dir ARCH=parisc olddefconfig COMPILER_INSTALL_PATH=$HOME/0day COMPILER=gcc-12.3.0 ~/bin/make.cross W=1 O=build_dir ARCH=parisc SHELL=/bin/bash If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202306112301.dMok4XSA-lkp@intel.com/ Note: the tiwai-sound/for-next HEAD 15253079ca300160c92c9c0ee2541836463043f6 builds fine. It only hurts bisectability. All errors (new ones prefixed by >>): In file included from sound/core/ump_convert.h:5, from sound/core/ump.c:14: >> include/sound/ump_msg.h:196:13: error: duplicate member 'reserved' 196 | u32 reserved:8; | ^~~~~~~~ >> include/sound/ump_msg.h:197:13: error: duplicate member 'program' 197 | u32 program:8; | ^~~~~~~ >> include/sound/ump_msg.h:198:13: error: duplicate member 'channel' 198 | u32 channel:4; | ^~~~~~~ >> include/sound/ump_msg.h:199:13: error: duplicate member 'status' 199 | u32 status:4; | ^~~~~~ >> include/sound/ump_msg.h:200:13: error: duplicate member 'group' 200 | u32 group:4; | ^~~~~ >> include/sound/ump_msg.h:201:13: error: duplicate member 'type' 201 | u32 type:4; | ^~~~ vim +/reserved +196 include/sound/ump_msg.h 184 185 /* MIDI 1.0 Program Change (32bit) */ 186 struct snd_ump_midi1_msg_program { 187 #ifdef __BIG_ENDIAN_BITFIELD 188 u32 type:4; 189 u32 group:4; 190 u32 status:4; 191 u32 channel:4; 192 u32 program:8; 193 u32 reserved:8; 194 #else 195 #endif > 196 u32 reserved:8; > 197 u32 program:8; > 198 u32 channel:4; > 199 u32 status:4; > 200 u32 group:4; > 201 u32 type:4; 202 } __packed; 203 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

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