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[PATCH] ASoC: amd: yc: Add VivoBook Pro 15 to quirks list for acp6x
by BrenoRCBrito 21 Aug '23

21 Aug '23

VivoBook Pro 15 Ryzen Edition uses Ryzen 6800H processor, and adding to quirks list for acp6x will enable internal mic. Signed-off-by: BrenoRCBrito <brenorcbrito(a)gmail.com> --- 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 a2fe3bd4f9a1..a2661dd637a5 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -248,6 +248,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "M3402RA"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "ASUSTeK COMPUTER INC."), + DMI_MATCH(DMI_PRODUCT_NAME, "M6500RC"), + } + }, { .driver_data = &acp6x_card, .matches = { -- 2.40.1

2 1

[PATCH 0/4] ASoC: amd: acp: Add sound support for a line of HUAWEI laptops
by Marian Postevca 21 Aug '23

21 Aug '23

This series adds support for a line of HUAWEI laptops with AMD CPUs that connect using the ACP3x module to a ES8336 codec. The codec driver must be extended to support the S32 LE format and the MCLK div by 2 option. MCLK div by 2 is needed for one specific SKU which uses a 48Mhz MCLK which seems to be too high of a frequency for the codec and must be divided by 2. The acp legacy driver must also be extended by using callbacks so that the more complicated handling for this specific codec can be moved outside the more generic ACP code. The last patch tries to avoid anoying pop sounds when the speaker/headphones are enabled/disabled by delaying the handling of the GPIOs and using a mutex to avoid race conditions between the speaker power event callback and the trigger callback. Marian Postevca (4): ASoC: es8316: Enable support for S32 LE format and MCLK div by 2 ASoC: amd: acp: Add support for splitting the codec specific code from the ACP driver ASoC: amd: acp: Add machine driver that enables sound for systems with a ES8336 codec ASoC: amd: acp: Improve support for speaker power events sound/soc/amd/acp-config.c | 70 ++ sound/soc/amd/acp/Makefile | 2 +- sound/soc/amd/acp/acp-legacy-mach.c | 105 ++- sound/soc/amd/acp/acp-mach-common.c | 8 + sound/soc/amd/acp/acp-mach.h | 67 ++ sound/soc/amd/acp/acp-renoir.c | 4 + sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c | 615 ++++++++++++++++++ sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.h | 12 + sound/soc/codecs/es8316.c | 21 +- sound/soc/codecs/es8316.h | 3 + 10 files changed, 886 insertions(+), 21 deletions(-) create mode 100644 sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c create mode 100644 sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.h -- 2.39.1

4 25

[PATCH] ALSA: usb-audio: Fix init call orders for UAC1
by Takashi Iwai 21 Aug '23

21 Aug '23

There have been reports of USB-audio driver spewing errors at the probe time on a few devices like Jabra and Logitech. The suggested fix there couldn't be applied as is, unfortunately, because it'll likely break other devices. But, the patch suggested an interesting point: looking at the current init code in stream.c, one may notice that it does initialize differently from the device setup in endpoint.c. Namely, for UAC1, we should call snd_usb_init_pitch() and snd_usb_init_sample_rate() after setting the interface, while the init sequence at parsing calls them before setting the interface blindly. This patch changes the init sequence at parsing for UAC1 (and other devices that need a similar behavior) to be aligned with the rest of the code, setting the interface at first. And, this fixes the long-standing problems on a few UAC1 devices like Jabra / Logitech, as reported, too. Reported-and-tested-by: Joakim Tjernlund <joakim.tjernlund(a)infinera.com> Closes: https://lore.kernel.org/r/202bbbc0f51522e8545783c4c5577d12a8e2d56d.camel@in… Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/stream.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/usb/stream.c b/sound/usb/stream.c index f10f4e6d3fb8..3d4add94e367 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -1093,6 +1093,7 @@ static int __snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int i, altno, err, stream; struct audioformat *fp = NULL; struct snd_usb_power_domain *pd = NULL; + bool set_iface_first; int num, protocol; dev = chip->dev; @@ -1223,11 +1224,19 @@ static int __snd_usb_parse_audio_interface(struct snd_usb_audio *chip, return err; } + set_iface_first = false; + if (protocol == UAC_VERSION_1 || + (chip->quirk_flags & QUIRK_FLAG_SET_IFACE_FIRST)) + set_iface_first = true; + /* try to set the interface... */ usb_set_interface(chip->dev, iface_no, 0); + if (set_iface_first) + usb_set_interface(chip->dev, iface_no, altno); snd_usb_init_pitch(chip, fp); snd_usb_init_sample_rate(chip, fp, fp->rate_max); - usb_set_interface(chip->dev, iface_no, altno); + if (!set_iface_first) + usb_set_interface(chip->dev, iface_no, altno); } return 0; } -- 2.35.3

1 0

[PATCH][next] ALSA: hda/realtek: Fix spelling mistake "powe" -> "power"
by Colin Ian King 21 Aug '23

21 Aug '23

There is a spelling mistake in a quirk entry. Fix it. Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6707db55f9c5..54e17791c6a8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9967,7 +9967,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3869, "Lenovo Yoga7 14IAL7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), SND_PCI_QUIRK(0x17aa, 0x387d, "Yoga S780-16 pro Quad AAC", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x387e, "Yoga S780-16 pro Quad YC", ALC287_FIXUP_TAS2781_I2C), - SND_PCI_QUIRK(0x17aa, 0x3881, "YB9 dual powe mode2 YC", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x3881, "YB9 dual power mode2 YC", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x3884, "Y780 YG DUAL", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x3886, "Y780 VECO DUAL", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38a7, "Y780P AMD YG dual", ALC287_FIXUP_TAS2781_I2C), -- 2.39.2

2 1

How to determine which files were read/parsed
by GitHub issues - opened 21 Aug '23

21 Aug '23

alsa-project/alsa-lib issue #345 was opened from bmariuszb: I would like to know which files were read by alsa doing `alsactl restore` or `alsactl init`. So far I know that there is a folder `/usr/share/alsa` and it has `init/00main`, and this is probably read first, because when I deleted whole alsa dir I got error about this file, then I don't know what is read. Also there is a file `/var/lib/alsa/asound.state` which from my understanding contains values for different controls that are visible in alsamixer. There is also "user" config file in `/etc/asound.conf` and optionally any user can create `~/.asound.conf`. Am I missing any other files? getting the list of files in order they are read for each card or all cards would be helpfull in debugging. For example by default I found out that my card HDA Intel PCH has two devices ALC1150 Analog and ALC1150 Digital. Digital is probably one of the six jacks in the back of my PC that are unplugged so I don't care about them right now. However Analog in my PC is connected to speakers and also from motherboard comes another cable to front jack panel that I plugged my headphones. I have a few problems with the default configuration: 1. Headphone slider in alsamixer, known as control doesn't work. 2. There are too many controls in Playback that are useless in my opinion. 3. I haven't found yet how headphone and speakers jacks where merged into one Analog device. I wonder if it's a software merge or a hardware thing. I would like to combine card 2: NVidia [HDA NVidia], device 3: HDMI 0 [X2483_2481] with the merged card 1 device 0 (headphones and speakers) and I was wondering if it would be better to merge all 3 devices at once. Correct me if I'm wrong, but from my understanding you have to connect different devices to be able to dynamically switch between them, because normally every process, e.g browser connects to a pcm device at startup and you can't change it later. So the actual audio switching is unmuting one thing and muting rest of them. All "things" are connected to one pcm, right? Please correct me if I'm wrong or use wrong terminology. ![image](https://github.com/alsa-project/alsa-lib/assets/57236506/2d73bb21-e9ee-4172-8e8e-04785cf0a887) Issue URL : https://github.com/alsa-project/alsa-lib/issues/345 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 1/2] ALSA: hda/tas2781: Fix acpi device refcount leak at tas2781_read_acpi()
by Takashi Iwai 20 Aug '23

20 Aug '23

The error path at tas2781_read_acpi() doesn't release the acpi_device adev but releases another device physdev instead. This results in a refcount leak. Fix it by replacing with the right object. Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver") Reported-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Closes: https://lore.kernel.org/r/9f910785-e856-1539-e3e4-c9817af5fe67@linux.intel.… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/tas2781_hda_i2c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 35dafc4aec4f..0968ae915fd0 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -118,7 +118,7 @@ static int tas2781_read_acpi(struct tasdevice_priv *p, const char *hid) err: dev_err(p->dev, "read acpi error, ret: %d\n", ret); - put_device(physdev); + acpi_dev_put(adev); return ret; } -- 2.35.3

1 1

A potential Integer Overflow bug found in aplay/aplay.c
by GitHub issues - edited 20 Aug '23

20 Aug '23

alsa-project/alsa-utils issue #231 was edited from cascades-sjtu: Hi, I'm currently trying to use the static analysis tool [Infer](https://fbinfer.com) to find uncatched API-misuse bugs in OpenWrt packages, and I find a potential Integer Overflow in your project, version 1.2.9. The bug located in aplay/aplay.c. Firstly, the program tries to write the remains bytes in `audiobuf` with the length of `loaded` in line 2865, and `loaded` is later used as the parameter for `safe_read` in the loop, then `loaded` is used as the 2nd argument of `pct_write()` and finally after a multiply operation, it is used as the size of Malloc in `remap_data()`, as shown in the following code: ```cpp static void playback_go(int fd, size_t loaded, off_t count, int rtype, char *name) { int l, r; off_t written = 0; off_t c; header(rtype, name); set_params(); while (loaded > chunk_bytes && written < count && !in_aborting) { if (pcm_write(audiobuf + written, chunk_size) <= 0) return; written += chunk_bytes; loaded -= chunk_bytes; } if (written > 0 && loaded > 0) memmove(audiobuf, audiobuf + written, loaded); l = loaded; while (written < count && !in_aborting) { do { c = count - written; if (c > chunk_bytes) c = chunk_bytes; /* c < l, there is more data loaded * then we actually need to write */ if (c < l) l = c; c -= l; if (c == 0) break; r = safe_read(fd, audiobuf + l, c); if (r < 0) { perror(name); prg_exit(EXIT_FAILURE); } fdcount += r; if (r == 0) break; l += r; } while ((size_t)l < chunk_bytes); l = l * 8 / bits_per_frame; r = pcm_write(audiobuf, l); if (r != l) break; r = r * bits_per_frame / 8; written += r; l = 0; } if (!in_aborting) { snd_pcm_nonblock(handle, 0); snd_pcm_drain(handle); snd_pcm_nonblock(handle, nonblock); } } # in remap_data() chunk_bytes = count * bits_per_frame / 8; if (tmp_size < chunk_bytes) { free(tmp); tmp = malloc(chunk_bytes); if (!tmp) { error(_("not enough memory")); exit(1); } tmp_size = count; } ``` The parameter passed to Malloc may be overflowed so that the actual allocated memory is small. I also attached the analysis trace given by Infer FYI: ```json "trace": [ { "file": "aplay/aplay.c", "line": 940, "col": 14, "feature": [ "Input", "read" ] }, { "file": "aplay/aplay.c", "line": 940, "col": 14, "feature": [ "Input", "read" ] }, { "file": "aplay/aplay.c", "line": 2865, "col": 21, "feature": [ "Prune", [ "UnOp", "!", [ "BinOp", ">", [ "Var" ], [ "Cast", [ "Unsupported" ], [ "Const", [ "Cint", 0 ] ] ] ] ] ] }, { "file": "aplay/aplay.c", "line": 2868, "col": 2, "feature": [ "Store", [ "Var" ], [ "Var" ] ] }, { "file": "aplay/aplay.c", "line": 2878, "col": 8, "feature": [ "Prune", [ "UnOp", "!", [ "BinOp", "<", [ "Var" ], [ "Var" ] ] ] ] }, { "file": "aplay/aplay.c", "line": 2871, "col": 4, "feature": [ "Store", [ "Var" ], [ "BinOp", "-", [ "Var" ], [ "Var" ] ] ] }, { "file": "aplay/aplay.c", "line": 2872, "col": 8, "feature": [ "Prune", [ "UnOp", "!", [ "BinOp", ">", [ "Cast", [ "Unsupported" ], [ "Var" ] ], [ "Var" ] ] ] ] }, { "file": "aplay/aplay.c", "line": 2878, "col": 8, "feature": [ "Prune", [ "BinOp", "<", [ "Var" ], [ "Var" ] ] ] }, { "file": "aplay/aplay.c", "line": 2879, "col": 5, "feature": [ "Store", [ "Var" ], [ "Var" ] ] }, { "file": "aplay/aplay.c", "line": 2899, "col": 3, "feature": [ "Store", [ "Var" ], [ "BinOp", "/", [ "BinOp", "*", [ "Cast", [ "Unsupported" ], [ "Var" ] ], [ "Var" ] ], [ "Cast", [ "Unsupported" ], [ "Const", [ "Cint", 8 ] ] ] ] ] }, { "file": "aplay/aplay.c", "line": 2900, "col": 3, "feature": [ "Store", [ "Var" ], [ "BinOp", "+", [ "Var" ], [ "Var" ] ] ] }, { "file": "aplay/aplay.c", "line": 2871, "col": 4, "feature": [ "Store", [ "Var" ], [ "BinOp", "-", [ "Var" ], [ "Var" ] ] ] }, { "file": "aplay/aplay.c", "line": 2872, "col": 8, "feature": [ "Prune", [ "UnOp", "!", [ "BinOp", ">", [ "Cast", [ "Unsupported" ], [ "Var" ] ], [ "Var" ] ] ] ] }, { "file": "aplay/aplay.c", "line": 2878, "col": 8, "feature": [ "Prune", [ "BinOp", "<", [ "Var" ], [ "Var" ] ] ] }, { "file": "aplay/aplay.c", "line": 2879, "col": 5, "feature": [ "Store", [ "Var" ], [ "Var" ] ] }, { "file": "aplay/aplay.c", "line": 2895, "col": 3, "feature": [ "Store", [ "Var" ], [ "BinOp", "/", [ "Cast", [ "Unsupported" ], [ "BinOp", "*", [ "Var" ], [ "Const", [ "Cint", 8 ] ] ] ], [ "Var" ] ] ] }, { "file": "aplay/aplay.c", "line": 2896, "col": 7, "feature": [ "Call", "pcm_write" ] }, { "file": "aplay/aplay.c", "line": 2126, "col": 6, "feature": [ "Prune", [ "UnOp", "!", [ "BinOp", "<", [ "Var" ], [ "Var" ] ] ] ] }, { "file": "aplay/aplay.c", "line": 2130, "col": 9, "feature": [ "Call", "remap_data" ] }, { "file": "aplay/aplay.c", "line": 2069, "col": 2, "feature": [ "Store", [ "Var" ], [ "BinOp", "/", [ "BinOp", "*", [ "Var" ], [ "Var" ] ], [ "Cast", [ "Unsupported" ], [ "Const", [ "Cint", 8 ] ] ] ] ] }, { "file": "aplay/aplay.c", "line": 2070, "col": 6, "feature": [ "Prune", [ "BinOp", "<", [ "Var" ], [ "Var" ] ] ] }, { "file": "aplay/aplay.c", "line": 2072, "col": 9, "feature": [ "IntOverflow", "malloc", [ "Var" ] ] } ], ``` Issue URL : https://github.com/alsa-project/alsa-utils/issues/231 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

A potential Integer Overflow bug found in aplay/aplay.c
by GitHub issues - opened 20 Aug '23

20 Aug '23

alsa-project/alsa-utils issue #231 was opened from cascades-sjtu: Hi, I'm currently trying to use the static analysis tool [Infer](https://fbinfer.com) to find uncatched API-misuse bugs in OpenWrt packages, and I find a potential Integer Overflow in your project, version 1.2.9. The bug located in aplay/aplay.c. Firstly, the program tries to write the remains bytes in `audiobuf` with the length of `loaded` in line 2865, and `loaded` is later used as the parameter for `safe_read` in the loop, then `loaded` is used as the 2nd argument of `pct_write()` and finally after a multiply operation, it is used as the size of Malloc in `remap_data()`, as shown in the following code: ```cpp static void playback_go(int fd, size_t loaded, off_t count, int rtype, char *name) { int l, r; off_t written = 0; off_t c; header(rtype, name); set_params(); while (loaded > chunk_bytes && written < count && !in_aborting) { if (pcm_write(audiobuf + written, chunk_size) <= 0) return; written += chunk_bytes; loaded -= chunk_bytes; } if (written > 0 && loaded > 0) memmove(audiobuf, audiobuf + written, loaded); l = loaded; while (written < count && !in_aborting) { do { c = count - written; if (c > chunk_bytes) c = chunk_bytes; /* c < l, there is more data loaded * then we actually need to write */ if (c < l) l = c; c -= l; if (c == 0) break; r = safe_read(fd, audiobuf + l, c); if (r < 0) { perror(name); prg_exit(EXIT_FAILURE); } fdcount += r; if (r == 0) break; l += r; } while ((size_t)l < chunk_bytes); l = l * 8 / bits_per_frame; r = pcm_write(audiobuf, l); if (r != l) break; r = r * bits_per_frame / 8; written += r; l = 0; } if (!in_aborting) { snd_pcm_nonblock(handle, 0); snd_pcm_drain(handle); snd_pcm_nonblock(handle, nonblock); } } # in remap_data() chunk_bytes = count * bits_per_frame / 8; if (tmp_size < chunk_bytes) { free(tmp); tmp = malloc(chunk_bytes); if (!tmp) { error(_("not enough memory")); exit(1); } tmp_size = count; } ``` The parameter passed to Malloc may be overflowed so that the actual allocated memory is small. I also attached the analysis trace given by Infer FYI: ```json "trace": [ { "file": "aplay/aplay.c", "line": 940, "col": 14, "feature": [ "Input", "read" ] }, { "file": "aplay/aplay.c", "line": 940, "col": 14, "feature": [ "Input", "read" ] }, { "file": "aplay/aplay.c", "line": 2865, "col": 21, "feature": [ "Prune", [ "UnOp", "!", [ "BinOp", ">", [ "Var" ], [ "Cast", [ "Unsupported" ], [ "Const", [ "Cint", 0 ] ] ] ] ] ] }, { "file": "aplay/aplay.c", "line": 2868, "col": 2, "feature": [ "Store", [ "Var" ], [ "Var" ] ] }, { "file": "aplay/aplay.c", "line": 2878, "col": 8, "feature": [ "Prune", [ "UnOp", "!", [ "BinOp", "<", [ "Var" ], [ "Var" ] ] ] ] }, { "file": "aplay/aplay.c", "line": 2871, "col": 4, "feature": [ "Store", [ "Var" ], [ "BinOp", "-", [ "Var" ], [ "Var" ] ] ] }, { "file": "aplay/aplay.c", "line": 2872, "col": 8, "feature": [ "Prune", [ "UnOp", "!", [ "BinOp", ">", [ "Cast", [ "Unsupported" ], [ "Var" ] ], [ "Var" ] ] ] ] }, { "file": "aplay/aplay.c", "line": 2878, "col": 8, "feature": [ "Prune", [ "BinOp", "<", [ "Var" ], [ "Var" ] ] ] }, { "file": "aplay/aplay.c", "line": 2879, "col": 5, "feature": [ "Store", [ "Var" ], [ "Var" ] ] }, { "file": "aplay/aplay.c", "line": 2899, "col": 3, "feature": [ "Store", [ "Var" ], [ "BinOp", "/", [ "BinOp", "*", [ "Cast", [ "Unsupported" ], [ "Var" ] ], [ "Var" ] ], [ "Cast", [ "Unsupported" ], [ "Const", [ "Cint", 8 ] ] ] ] ] }, { "file": "aplay/aplay.c", "line": 2900, "col": 3, "feature": [ "Store", [ "Var" ], [ "BinOp", "+", [ "Var" ], [ "Var" ] ] ] }, { "file": "aplay/aplay.c", "line": 2871, "col": 4, "feature": [ "Store", [ "Var" ], [ "BinOp", "-", [ "Var" ], [ "Var" ] ] ] }, { "file": "aplay/aplay.c", "line": 2872, "col": 8, "feature": [ "Prune", [ "UnOp", "!", [ "BinOp", ">", [ "Cast", [ "Unsupported" ], [ "Var" ] ], [ "Var" ] ] ] ] }, { "file": "aplay/aplay.c", "line": 2878, "col": 8, "feature": [ "Prune", [ "BinOp", "<", [ "Var" ], [ "Var" ] ] ] }, { "file": "aplay/aplay.c", "line": 2879, "col": 5, "feature": [ "Store", [ "Var" ], [ "Var" ] ] }, { "file": "aplay/aplay.c", "line": 2895, "col": 3, "feature": [ "Store", [ "Var" ], [ "BinOp", "/", [ "Cast", [ "Unsupported" ], [ "BinOp", "*", [ "Var" ], [ "Const", [ "Cint", 8 ] ] ] ], [ "Var" ] ] ] }, { "file": "aplay/aplay.c", "line": 2896, "col": 7, "feature": [ "Call", "pcm_write" ] }, { "file": "aplay/aplay.c", "line": 2126, "col": 6, "feature": [ "Prune", [ "UnOp", "!", [ "BinOp", "<", [ "Var" ], [ "Var" ] ] ] ] }, { "file": "aplay/aplay.c", "line": 2130, "col": 9, "feature": [ "Call", "remap_data" ] }, { "file": "aplay/aplay.c", "line": 2069, "col": 2, "feature": [ "Store", [ "Var" ], [ "BinOp", "/", [ "BinOp", "*", [ "Var" ], [ "Var" ] ], [ "Cast", [ "Unsupported" ], [ "Const", [ "Cint", 8 ] ] ] ] ] }, { "file": "aplay/aplay.c", "line": 2070, "col": 6, "feature": [ "Prune", [ "BinOp", "<", [ "Var" ], [ "Var" ] ] ] }, { "file": "aplay/aplay.c", "line": 2072, "col": 9, "feature": [ "IntOverflow", "malloc", [ "Var" ] ] } ], ``` Issue URL : https://github.com/alsa-project/alsa-utils/issues/231 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

raw midi open fails results in verbose message to stdout
by GitHub issues - opened 20 Aug '23

20 Aug '23

alsa-project/alsa-lib issue #344 was opened from symdeb: ALSA lib rawmidi_hw.c:384:(snd_rawmidi_hw_open) open /dev/snd/umpC5D0 failed: No such file or directory Issue URL : https://github.com/alsa-project/alsa-lib/issues/344 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

ucm2 profile for Universal Audio Volt 2
by GitHub pull_request - edited 19 Aug '23

19 Aug '23

alsa-project/alsa-ucm-conf pull request #340 was edited from z411: This adds a profile for the (relatively recent) Universal Audio Volt 2 USB audio interface. It has a Stereo output and 2 mono inputs that are (like most budget interfaces) combined into a stereo input. This profile effectively splits these inputs into two Line inputs. These inputs can be used interchangeably for XLR microphones, Line level signals or Hi-Z instruments. The only weird thing is that in my machine the Mono Input 2 appears first in order but pretty sure I defined the prorities in the configuration file just fine. Everything works well on my machine. [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/12385352… The other Volt interfaces are similar so they surely could be accommodated in this profile as well if someone has them. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/340 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/340.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2 profile for Universal Audio Volt 2
by GitHub pull_request - edited 19 Aug '23

19 Aug '23

alsa-project/alsa-ucm-conf pull request #340 was edited from z411: This adds a profile for the (relatively recent) Universal Audio Volt 2 USB audio interface. It has a Stereo output and 2 mono inputs that are (like most budget interfaces) combined into a stereo input. This profile effectively splits these inputs into two Line inputs. These inputs can be used interchangeably for XLR microphones, Line level signals or Hi-Z instruments. The only weird thing is that in my machine the Mono Input 2 appears first in order but pretty sure I defined the prorities in the configuration file just fine. Everything works well on my machine. [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/12385352… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/340 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/340.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2 profile for Universal Audio Volt 2
by GitHub pull_request - edited 19 Aug '23

19 Aug '23

alsa-project/alsa-ucm-conf pull request #340 was edited from z411: This adds a profile for the (relatively recent) Universal Audio Volt 2 USB audio interface. It has a Stereo output and 2 mono inputs that are (like most budget interfaces) combined into a stereo input. This profile effectively splits these inputs into two Line inputs. These inputs can be used for XLR microphones, Line level signals or Hi-Z instruments. The only weird thing is that in my machine the Mono Input 2 appears first in order but pretty sure I defined the prorities in the configuration file just fine. Everything works well on my machine. [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/12385352… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/340 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/340.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2 profile for Universal Audio Volt 2
by GitHub pull_request - edited 19 Aug '23

19 Aug '23

alsa-project/alsa-ucm-conf pull request #340 was edited from z411: This adds a profile for the (relatively recent) Universal Audio Volt 2 USB audio interface. It has a Stereo output and 2 mono inputs that are (like most budget interfaces) combined into a stereo input. This profile effectively splits these inputs into two Line inputs. These inputs can be used for XLR microphones, Line level inputs or Hi-Z inputs. The only weird thing is that in my machine the Mono Input 2 appears first in order but pretty sure I defined the prorities in the configuration file just fine. Everything works well on my machine. [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/12385352… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/340 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/340.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2 profile for Universal Audio Volt 2
by GitHub pull_request - opened 19 Aug '23

19 Aug '23

alsa-project/alsa-ucm-conf pull request #340 was opened from z411: This adds a profile for the (relatively recent) Universal Audio Vol 2 USB aduio interface. It has a Stereo output and 2 mono inputs that are (like most budget interfaces) combined into a stereo input. This profile effectively splits these inputs into two Line inputs. These inputs can be used for XLR microphones, Line level inputs or Hi-Z inputs. The only weird thing is that in my machine the Mono Input 2 appears first in order but pretty sure I defined the prorities in the configuration file just fine. Everything works well on my machine. [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/12385352… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/340 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/340.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v4 1/2] ALSA: hda/tas2781: Add tas2781 HDA driver
by Shenghao Ding 18 Aug '23

18 Aug '23

Integrate tas2781 configs for Lenovo Laptops. All of the tas2781s in the laptop will be aggregated as one audio device. The code support realtek as the primary codec. Rename "struct cs35l41_dev_name" to "struct scodec_dev_name" for all other side codecs instead of the certain one. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- Changes in v4: - | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: | https://lore.kernel.org/oe-kbuild-all/202308172137.SthCPFbA-lkp@intel. | com/ - remove workaround code for 0x17aa38be, laptop vendor will fix it in acpi. - rename comp_match_tas2781_dev_name to avoid indentation - simplify the check of vendor id with Lenovo - ThinkPad is one of Lenovo's brands, I was suggested me to use ALC269_FIXUP_THINKPAD_ACPI. - Add comments on ACARD_SINGLE_RANGE_EXT_TLV - Add the range check for tas_priv->tasdevice[] in tas2781_acpi_get_i2c_res. - remove acpi_subsystem_id - Issue in Laptop 0x17aa38be ACPI talbe caused codec->bus->pci->subsystem_device is not equal to (codec->core.subsystem_id & 0xffff) in snd_hda_pick_fixup. The former is 0x3802 and the latter is 0x38be leads to getting the wrong fixup_id and enter into the wrong entry. Although, this issue has been raised to the laptop manufacturer, but the ACPI table is locked, cannot be changed any more. Correct the wrong entry in the code. - Rename "struct cs35l41_dev_name" to "struct scodec_dev_name" for all other side codecs instead of one certain codec. - Ignore the checkpatch complaints in alc269_fixup_tbl - Drop the hunk which is irrelevant with my code in alc_fixup_headset_mode_alc255_no_hp_mic - Add tiwai(a)suse.de into Cc list - remove useless index - combine ALC287_FIXUP_TAS2781_I2C_2 and ALC287_FIXUP_TAS2781_I2C_4 together as ALC287_FIXUP_TAS2781_I2C, The code view all the tas2781s in the laptop as one instance. - delete the white space at the end of the line in alc_fixup_headset_mode_alc255_no_hp_mic --- sound/pci/hda/patch_realtek.c | 88 +++++++++++++++++++++++++++++++++-- 1 file changed, 85 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 44fccfb93cff..ba1b02ed184a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6721,7 +6721,7 @@ static void comp_generic_playback_hook(struct hda_pcm_stream *hinfo, struct hda_ } } -struct cs35l41_dev_name { +struct scodec_dev_name { const char *bus; const char *hid; int index; @@ -6730,7 +6730,7 @@ struct cs35l41_dev_name { /* match the device name in a slightly relaxed manner */ static int comp_match_cs35l41_dev_name(struct device *dev, void *data) { - struct cs35l41_dev_name *p = data; + struct scodec_dev_name *p = data; const char *d = dev_name(dev); int n = strlen(p->bus); char tmp[32]; @@ -6746,12 +6746,32 @@ static int comp_match_cs35l41_dev_name(struct device *dev, void *data) return !strcmp(d + n, tmp); } +static int comp_match_tas2781_dev_name(struct device *dev, + void *data) +{ + struct scodec_dev_name *p = data; + const char *d = dev_name(dev); + int n = strlen(p->bus); + char tmp[32]; + + /* check the bus name */ + if (strncmp(d, p->bus, n)) + return 0; + /* skip the bus number */ + if (isdigit(d[n])) + n++; + /* the rest must be exact matching */ + snprintf(tmp, sizeof(tmp), "-%s:00", p->hid); + + return !strcmp(d + n, tmp); +} + static void cs35l41_generic_fixup(struct hda_codec *cdc, int action, const char *bus, const char *hid, int count) { struct device *dev = hda_codec_dev(cdc); struct alc_spec *spec = cdc->spec; - struct cs35l41_dev_name *rec; + struct scodec_dev_name *rec; int ret, i; switch (action) { @@ -6779,6 +6799,41 @@ static void cs35l41_generic_fixup(struct hda_codec *cdc, int action, const char } } +static void tas2781_generic_fixup(struct hda_codec *cdc, int action, + const char *bus, const char *hid) +{ + struct device *dev = hda_codec_dev(cdc); + struct alc_spec *spec = cdc->spec; + struct scodec_dev_name *rec; + int ret; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + rec = devm_kmalloc(dev, sizeof(*rec), GFP_KERNEL); + if (!rec) + return; + rec->bus = bus; + rec->hid = hid; + rec->index = 0; + spec->comps[0].codec = cdc; + component_match_add(dev, &spec->match, + comp_match_tas2781_dev_name, rec); + ret = component_master_add_with_match(dev, &comp_master_ops, + spec->match); + if (ret) + codec_err(cdc, + "Fail to register component aggregator %d\n", + ret); + else + spec->gen.pcm_playback_hook = + comp_generic_playback_hook; + break; + case HDA_FIXUP_ACT_FREE: + component_master_del(dev, &comp_master_ops); + break; + } +} + static void cs35l41_fixup_i2c_two(struct hda_codec *cdc, const struct hda_fixup *fix, int action) { cs35l41_generic_fixup(cdc, action, "i2c", "CSC3551", 2); @@ -6806,6 +6861,12 @@ static void alc287_fixup_legion_16ithg6_speakers(struct hda_codec *cdc, const st cs35l41_generic_fixup(cdc, action, "i2c", "CLSA0101", 2); } +static void tas2781_fixup_i2c(struct hda_codec *cdc, + const struct hda_fixup *fix, int action) +{ + tas2781_generic_fixup(cdc, action, "i2c", "TIAS2781"); +} + /* for alc295_fixup_hp_top_speakers */ #include "hp_x360_helper.c" @@ -7231,6 +7292,7 @@ enum { ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS, ALC236_FIXUP_DELL_DUAL_CODECS, ALC287_FIXUP_CS35L41_I2C_2_THINKPAD_ACPI, + ALC287_FIXUP_TAS2781_I2C, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -9309,6 +9371,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_THINKPAD_ACPI, }, + [ALC287_FIXUP_TAS2781_I2C] = { + .type = HDA_FIXUP_FUNC, + .v.func = tas2781_fixup_i2c, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -9884,6 +9952,20 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3853, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3855, "Legion 7 16ITHG6", ALC287_FIXUP_LEGION_16ITHG6), SND_PCI_QUIRK(0x17aa, 0x3869, "Lenovo Yoga7 14IAL7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), + SND_PCI_QUIRK(0x17aa, 0x387d, "Yoga S780-16 pro Quad AAC", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x387e, "Yoga S780-16 pro Quad YC", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x3881, "YB9 dual powe mode2 YC", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x3884, "Y780 YG DUAL", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x3886, "Y780 VECO DUAL", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38a7, "Y780P AMD YG dual", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38a8, "Y780P AMD VECO dual", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38ba, "Yoga S780-14.5 Air AMD quad YC", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38bb, "Yoga S780-14.5 Air AMD quad AAC", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38be, "Yoga S980-14.5 proX YC Dual", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38bf, "Yoga S980-14.5 proX LX Dual", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38c3, "Y980 DUAL", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38cb, "Y790 YG DUAL", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38cd, "Y790 VECO DUAL", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), -- 2.34.1

3 5

[PATCH] ASoC: amd: acp: Add kcontrols and widgets per-codec in common code
by Alper Nebi Yasak 18 Aug '23

18 Aug '23

Commit 7ac3404c2e21 ("ASoC: amd: acp: Map missing jack kcontrols") adds card kcontrols and DAPM widgets corresponding to jacks for Headphone and Headset Mic. But these were already being added in acp-legacy-mach.c and acp-sof-mach.c, so this causes a probe error: sof_mach rt5682s-hs-rt1019: control 2:0:0:Headphone Jack Switch:0 is already present sof_mach rt5682s-hs-rt1019: ASoC: Failed to add Headphone Jack Switch: -16 sof_mach rt5682s-hs-rt1019: devm_snd_soc_register_card(sof-rt5682s-hs-rt1019) failed: -16 sof_mach: probe of rt5682s-hs-rt1019 failed with error -16 Removing the new duplicates from the common code might be enough to fix the issue. But some of the codecs use different kcontrols and widgets. NAU8821 alone has an "Int Mic". MAX98360A has a single "Spk", while RT1019 has "Left Spk" and "Right Spk" (and two codec instances with name prefixes). Since there are per-codec differences, add the kcontrols and widgets in the common code as each dai-link is being initialized, instead of in the two machine drivers' probe(). Also, MAX98388 has Left and Right instances, but uses a single "SPK" widget. That seems weird, so normalize it to use two widgets (one per each instance like RT1019) while we're adding controls. Fixes: 7ac3404c2e21 ("ASoC: amd: acp: Map missing jack kcontrols") Signed-off-by: Alper Nebi Yasak <alpernebiyasak(a)gmail.com> --- Applies onto linux-next. sound/soc/amd/acp/acp-legacy-mach.c | 22 +--------- sound/soc/amd/acp/acp-mach-common.c | 68 +++++++++++++++++++++++++++-- sound/soc/amd/acp/acp-sof-mach.c | 21 +-------- 3 files changed, 67 insertions(+), 44 deletions(-) diff --git a/sound/soc/amd/acp/acp-legacy-mach.c b/sound/soc/amd/acp/acp-legacy-mach.c index 676ad50638d0..6d57d17ddfd7 100644 --- a/sound/soc/amd/acp/acp-legacy-mach.c +++ b/sound/soc/amd/acp/acp-legacy-mach.c @@ -75,23 +75,6 @@ static struct acp_card_drvdata rt5682s_rt1019_rmb_data = { .tdm_mode = false, }; -static const struct snd_kcontrol_new acp_controls[] = { - SOC_DAPM_PIN_SWITCH("Headphone Jack"), - SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("Spk"), - SOC_DAPM_PIN_SWITCH("Left Spk"), - SOC_DAPM_PIN_SWITCH("Right Spk"), - -}; - -static const struct snd_soc_dapm_widget acp_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_SPK("Spk", NULL), - SND_SOC_DAPM_SPK("Left Spk", NULL), - SND_SOC_DAPM_SPK("Right Spk", NULL), -}; - static int acp_asoc_probe(struct platform_device *pdev) { struct snd_soc_card *card = NULL; @@ -110,11 +93,8 @@ static int acp_asoc_probe(struct platform_device *pdev) card->dev = dev; card->owner = THIS_MODULE; card->name = pdev->id_entry->name; - card->dapm_widgets = acp_widgets; - card->num_dapm_widgets = ARRAY_SIZE(acp_widgets); - card->controls = acp_controls; - card->num_controls = ARRAY_SIZE(acp_controls); card->drvdata = (struct acp_card_drvdata *)pdev->id_entry->driver_data; + /* Widgets and controls added per-codec in acp-mach-common.c */ acp_card_drvdata = card->drvdata; dmi_id = dmi_first_match(acp_quirk_table); diff --git a/sound/soc/amd/acp/acp-mach-common.c b/sound/soc/amd/acp/acp-mach-common.c index f3abaa182fbb..a06af82b8056 100644 --- a/sound/soc/amd/acp/acp-mach-common.c +++ b/sound/soc/amd/acp/acp-mach-common.c @@ -513,6 +513,16 @@ SND_SOC_DAILINK_DEF(rt1019, DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC1019:00", "rt1019-aif"), COMP_CODEC("i2c-10EC1019:01", "rt1019-aif"))); +static const struct snd_kcontrol_new rt1019_controls[] = { + SOC_DAPM_PIN_SWITCH("Left Spk"), + SOC_DAPM_PIN_SWITCH("Right Spk"), +}; + +static const struct snd_soc_dapm_widget rt1019_widgets[] = { + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), +}; + static const struct snd_soc_dapm_route rt1019_map_lr[] = { { "Left Spk", NULL, "Left SPO" }, { "Right Spk", NULL, "Right SPO" }, @@ -533,10 +543,25 @@ static int acp_card_rt1019_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; struct acp_card_drvdata *drvdata = card->drvdata; + int ret; if (drvdata->amp_codec_id != RT1019) return -EINVAL; + ret = snd_soc_dapm_new_controls(&card->dapm, rt1019_widgets, + ARRAY_SIZE(rt1019_widgets)); + if (ret) { + dev_err(rtd->dev, "unable to add widget dapm controls, ret %d\n", ret); + return ret; + } + + ret = snd_soc_add_card_controls(card, rt1019_controls, + ARRAY_SIZE(rt1019_controls)); + if (ret) { + dev_err(rtd->dev, "unable to add card controls, ret %d\n", ret); + return ret; + } + return snd_soc_dapm_add_routes(&rtd->card->dapm, rt1019_map_lr, ARRAY_SIZE(rt1019_map_lr)); } @@ -667,6 +692,14 @@ static const struct snd_soc_ops acp_card_rt1019_ops = { SND_SOC_DAILINK_DEF(max98360a, DAILINK_COMP_ARRAY(COMP_CODEC("MX98360A:00", "HiFi"))); +static const struct snd_kcontrol_new max98360a_controls[] = { + SOC_DAPM_PIN_SWITCH("Spk"), +}; + +static const struct snd_soc_dapm_widget max98360a_widgets[] = { + SND_SOC_DAPM_SPK("Spk", NULL), +}; + static const struct snd_soc_dapm_route max98360a_map[] = { {"Spk", NULL, "Speaker"}, }; @@ -675,10 +708,25 @@ static int acp_card_maxim_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; struct acp_card_drvdata *drvdata = card->drvdata; + int ret; if (drvdata->amp_codec_id != MAX98360A) return -EINVAL; + ret = snd_soc_dapm_new_controls(&card->dapm, max98360a_widgets, + ARRAY_SIZE(max98360a_widgets)); + if (ret) { + dev_err(rtd->dev, "unable to add widget dapm controls, ret %d\n", ret); + return ret; + } + + ret = snd_soc_add_card_controls(card, max98360a_controls, + ARRAY_SIZE(max98360a_controls)); + if (ret) { + dev_err(rtd->dev, "unable to add card controls, ret %d\n", ret); + return ret; + } + return snd_soc_dapm_add_routes(&rtd->card->dapm, max98360a_map, ARRAY_SIZE(max98360a_map)); } @@ -744,13 +792,19 @@ SND_SOC_DAILINK_DEF(max98388, DAILINK_COMP_ARRAY(COMP_CODEC("i2c-ADS8388:00", "max98388-aif1"), COMP_CODEC("i2c-ADS8388:01", "max98388-aif1"))); +static const struct snd_kcontrol_new max98388_controls[] = { + SOC_DAPM_PIN_SWITCH("Left Spk"), + SOC_DAPM_PIN_SWITCH("Right Spk"), +}; + static const struct snd_soc_dapm_widget max98388_widgets[] = { - SND_SOC_DAPM_SPK("SPK", NULL), + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), }; static const struct snd_soc_dapm_route max98388_map[] = { - { "SPK", NULL, "Left BE_OUT" }, - { "SPK", NULL, "Right BE_OUT" }, + { "Left Spk", NULL, "Left BE_OUT" }, + { "Right Spk", NULL, "Right BE_OUT" }, }; static struct snd_soc_codec_conf max98388_conf[] = { @@ -803,6 +857,14 @@ static int acp_card_max98388_init(struct snd_soc_pcm_runtime *rtd) /* Don't need to add routes if widget addition failed */ return ret; } + + ret = snd_soc_add_card_controls(card, max98388_controls, + ARRAY_SIZE(max98388_controls)); + if (ret) { + dev_err(rtd->dev, "unable to add card controls, ret %d\n", ret); + return ret; + } + return snd_soc_dapm_add_routes(&rtd->card->dapm, max98388_map, ARRAY_SIZE(max98388_map)); } diff --git a/sound/soc/amd/acp/acp-sof-mach.c b/sound/soc/amd/acp/acp-sof-mach.c index a1c893f33f74..5223033a122f 100644 --- a/sound/soc/amd/acp/acp-sof-mach.c +++ b/sound/soc/amd/acp/acp-sof-mach.c @@ -94,22 +94,6 @@ static struct acp_card_drvdata sof_nau8821_max98388_data = { .tdm_mode = false, }; -static const struct snd_kcontrol_new acp_controls[] = { - SOC_DAPM_PIN_SWITCH("Headphone Jack"), - SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("Spk"), - SOC_DAPM_PIN_SWITCH("Left Spk"), - SOC_DAPM_PIN_SWITCH("Right Spk"), -}; - -static const struct snd_soc_dapm_widget acp_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_SPK("Spk", NULL), - SND_SOC_DAPM_SPK("Left Spk", NULL), - SND_SOC_DAPM_SPK("Right Spk", NULL), -}; - static int acp_sof_probe(struct platform_device *pdev) { struct snd_soc_card *card = NULL; @@ -128,11 +112,8 @@ static int acp_sof_probe(struct platform_device *pdev) card->dev = dev; card->owner = THIS_MODULE; card->name = pdev->id_entry->name; - card->dapm_widgets = acp_widgets; - card->num_dapm_widgets = ARRAY_SIZE(acp_widgets); - card->controls = acp_controls; - card->num_controls = ARRAY_SIZE(acp_controls); card->drvdata = (struct acp_card_drvdata *)pdev->id_entry->driver_data; + /* Widgets and controls added per-codec in acp-mach-common.c */ acp_card_drvdata = card->drvdata; dmi_id = dmi_first_match(acp_quirk_table); base-commit: 47762f08697484cf0c2f2904b8c52375ed26c8cb -- 2.40.1

2 1

[PATCH] ASoC: rt5645: improve the depop sequences of CBJ detection
by shumingf＠realtek.com 18 Aug '23

18 Aug '23

From: Shuming Fan <shumingf(a)realtek.com> This patch will improve the depop function of CBJ detection before the headphone playback when the system resumes. We also want to complete the CBJ detection in the system-level resume function. It could avoid triggering the detection at the playing state of headphones. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5645.c | 18 ++++++++++++------ 1 file changed, 12 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index e49aa28280e0..038d93e20883 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -80,6 +80,7 @@ static const struct reg_sequence init_list[] = { static const struct reg_sequence rt5650_init_list[] = { {0xf6, 0x0100}, + {RT5645_PWR_ANLG1, 0x02}, }; static const struct reg_default rt5645_reg[] = { @@ -1696,6 +1697,9 @@ static void hp_amp_power(struct snd_soc_component *component, int on) regmap_write(rt5645->regmap, RT5645_PR_BASE + RT5645_MAMP_INT_REG2, 0xfc00); snd_soc_component_write(component, RT5645_DEPOP_M2, 0x1140); + snd_soc_component_update_bits(component, RT5645_PWR_ANLG1, + RT5645_PWR_HP_L | RT5645_PWR_HP_R, + RT5645_PWR_HP_L | RT5645_PWR_HP_R); msleep(90); } else { /* depop parameters */ @@ -1743,7 +1747,8 @@ static void hp_amp_power(struct snd_soc_component *component, int on) snd_soc_component_write(component, RT5645_DEPOP_M2, 0x1140); msleep(100); snd_soc_component_write(component, RT5645_DEPOP_M1, 0x0001); - + snd_soc_component_update_bits(component, RT5645_PWR_ANLG1, + RT5645_PWR_HP_L | RT5645_PWR_HP_R, 0); } else { snd_soc_component_update_bits(component, RT5645_DEPOP_M1, RT5645_HP_SG_MASK | @@ -3150,7 +3155,7 @@ static int rt5645_jack_detect(struct snd_soc_component *component, int jack_inse unsigned int val; if (jack_insert) { - regmap_write(rt5645->regmap, RT5645_CHARGE_PUMP, 0x0e06); + regmap_write(rt5645->regmap, RT5645_CHARGE_PUMP, 0x0206); /* for jack type detect */ snd_soc_dapm_force_enable_pin(dapm, "LDO2"); @@ -3195,6 +3200,8 @@ static int rt5645_jack_detect(struct snd_soc_component *component, int jack_inse if (rt5645->pdata.level_trigger_irq) regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, RT5645_JD_1_1_MASK, RT5645_JD_1_1_NOR); + + regmap_write(rt5645->regmap, RT5645_CHARGE_PUMP, 0x0e06); } else { /* jack out */ rt5645->jack_type = 0; @@ -4003,13 +4010,13 @@ static int rt5645_i2c_probe(struct i2c_client *i2c) regmap_write(rt5645->regmap, RT5645_AD_DA_MIXER, 0x8080); - ret = regmap_register_patch(rt5645->regmap, init_list, + ret = regmap_multi_reg_write(rt5645->regmap, init_list, ARRAY_SIZE(init_list)); if (ret != 0) dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); if (rt5645->codec_type == CODEC_TYPE_RT5650) { - ret = regmap_register_patch(rt5645->regmap, rt5650_init_list, + ret = regmap_multi_reg_write(rt5645->regmap, rt5650_init_list, ARRAY_SIZE(rt5650_init_list)); if (ret != 0) dev_warn(&i2c->dev, "Apply rt5650 patch failed: %d\n", @@ -4220,8 +4227,7 @@ static int __maybe_unused rt5645_sys_resume(struct device *dev) if (rt5645->hp_jack) { rt5645->jack_type = 0; - queue_delayed_work(system_power_efficient_wq, - &rt5645->jack_detect_work, msecs_to_jiffies(0)); + rt5645_jack_detect_work(&rt5645->jack_detect_work.work); } return 0; } -- 2.34.1

2 1

[PATCH] ASoC: SOF: intel: hda: Clean up link DMA for IPC3 during stop
by Peter Ujfalusi 18 Aug '23

18 Aug '23

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> With IPC3, we reset hw_params during the stop trigger, so we should also clean up the link DMA during the stop trigger. Fixes: 1bf83fa6654c ("ASoC: SOF: Intel: hda-dai: Do not perform DMA cleanup during stop") Closes: https://github.com/thesofproject/linux/issues/4455 Closes: https://github.com/thesofproject/linux/issues/4482 Closes: https://bugzilla.kernel.org/show_bug.cgi?id=217673 Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/intel/hda-dai-ops.c | 11 ++++++++++- sound/soc/sof/intel/hda-dai.c | 5 ++--- sound/soc/sof/intel/hda.h | 2 ++ 3 files changed, 14 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/intel/hda-dai-ops.c b/sound/soc/sof/intel/hda-dai-ops.c index f3513796c189..f33051eac1c0 100644 --- a/sound/soc/sof/intel/hda-dai-ops.c +++ b/sound/soc/sof/intel/hda-dai-ops.c @@ -372,6 +372,7 @@ static const struct hda_dai_widget_dma_ops hda_ipc4_chain_dma_ops = { static int hda_ipc3_post_trigger(struct snd_sof_dev *sdev, struct snd_soc_dai *cpu_dai, struct snd_pcm_substream *substream, int cmd) { + struct hdac_ext_stream *hext_stream = hda_get_hext_stream(sdev, cpu_dai, substream); struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(cpu_dai, substream->stream); switch (cmd) { @@ -379,9 +380,17 @@ static int hda_ipc3_post_trigger(struct snd_sof_dev *sdev, struct snd_soc_dai *c case SNDRV_PCM_TRIGGER_STOP: { struct snd_sof_dai_config_data data = { 0 }; + int ret; data.dai_data = DMA_CHAN_INVALID; - return hda_dai_config(w, SOF_DAI_CONFIG_FLAGS_HW_FREE, &data); + ret = hda_dai_config(w, SOF_DAI_CONFIG_FLAGS_HW_FREE, &data); + if (ret < 0) + return ret; + + if (cmd == SNDRV_PCM_TRIGGER_STOP) + return hda_link_dma_cleanup(substream, hext_stream, cpu_dai); + + break; } case SNDRV_PCM_TRIGGER_PAUSE_PUSH: return hda_dai_config(w, SOF_DAI_CONFIG_FLAGS_PAUSE, NULL); diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 3297dea493aa..863865f3d77e 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -107,9 +107,8 @@ hda_dai_get_ops(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai return sdai->platform_private; } -static int hda_link_dma_cleanup(struct snd_pcm_substream *substream, - struct hdac_ext_stream *hext_stream, - struct snd_soc_dai *cpu_dai) +int hda_link_dma_cleanup(struct snd_pcm_substream *substream, struct hdac_ext_stream *hext_stream, + struct snd_soc_dai *cpu_dai) { const struct hda_dai_widget_dma_ops *ops = hda_dai_get_ops(substream, cpu_dai); struct sof_intel_hda_stream *hda_stream; diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 3f7c6fb05e5d..5b9e4ebcc18b 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -963,5 +963,7 @@ const struct hda_dai_widget_dma_ops * hda_select_dai_widget_ops(struct snd_sof_dev *sdev, struct snd_sof_widget *swidget); int hda_dai_config(struct snd_soc_dapm_widget *w, unsigned int flags, struct snd_sof_dai_config_data *data); +int hda_link_dma_cleanup(struct snd_pcm_substream *substream, struct hdac_ext_stream *hext_stream, + struct snd_soc_dai *cpu_dai); #endif -- 2.41.0

4 5

[PATCH -next] ALSA: asihpi: Remove unused declarations
by Yue Haibing 18 Aug '23

18 Aug '23

These are not implemented, so can remove them. Signed-off-by: Yue Haibing <yuehaibing(a)huawei.com> --- sound/pci/asihpi/hpi.h | 16 ---------------- sound/pci/asihpi/hpi_internal.h | 5 ----- 2 files changed, 21 deletions(-) diff --git a/sound/pci/asihpi/hpi.h b/sound/pci/asihpi/hpi.h index 3aebec763fb8..04a5cf6572cd 100644 --- a/sound/pci/asihpi/hpi.h +++ b/sound/pci/asihpi/hpi.h @@ -1191,19 +1191,6 @@ u16 hpi_adapter_set_mode_ex(u16 adapter_index, u32 adapter_mode, u16 hpi_adapter_get_mode(u16 adapter_index, u32 *padapter_mode); -u16 hpi_adapter_get_assert2(u16 adapter_index, u16 *p_assert_count, - char *psz_assert, u32 *p_param1, u32 *p_param2, - u32 *p_dsp_string_addr, u16 *p_processor_id); - -u16 hpi_adapter_test_assert(u16 adapter_index, u16 assert_id); - -u16 hpi_adapter_enable_capability(u16 adapter_index, u16 capability, u32 key); - -u16 hpi_adapter_self_test(u16 adapter_index); - -u16 hpi_adapter_debug_read(u16 adapter_index, u32 dsp_address, char *p_bytes, - int *count_bytes); - u16 hpi_adapter_set_property(u16 adapter_index, u16 property, u16 paramter1, u16 paramter2); @@ -1488,9 +1475,6 @@ u16 hpi_pad_get_program_type(u32 h_control, u32 *ppTY); u16 hpi_pad_get_rdsPI(u32 h_control, u32 *ppI); -u16 hpi_pad_get_program_type_string(u32 h_control, const u32 data_type, - const u32 pTY, char *psz_string, const u32 string_length); - /****************************/ /* AES/EBU Receiver control */ /****************************/ diff --git a/sound/pci/asihpi/hpi_internal.h b/sound/pci/asihpi/hpi_internal.h index 6859d51389f5..e569e3b33b8e 100644 --- a/sound/pci/asihpi/hpi_internal.h +++ b/sound/pci/asihpi/hpi_internal.h @@ -1394,17 +1394,12 @@ u32 hpi_indexes_to_handle(const char c_object, const u16 adapter_index, void hpi_send_recv(struct hpi_message *phm, struct hpi_response *phr); /* used in PnP OS/driver */ -u16 hpi_subsys_create_adapter(const struct hpi_resource *p_resource, - u16 *pw_adapter_index); - u16 hpi_outstream_host_buffer_get_info(u32 h_outstream, u8 **pp_buffer, struct hpi_hostbuffer_status **pp_status); u16 hpi_instream_host_buffer_get_info(u32 h_instream, u8 **pp_buffer, struct hpi_hostbuffer_status **pp_status); -u16 hpi_adapter_restart(u16 adapter_index); - /* The following 3 functions were last declared in header files for driver 3.10. HPI_ControlQuery() used to be the recommended way -- 2.34.1

2 1

[PATCH v3 0/9] sound: Use -EPROBE_DEFER instead of i915 module loading.
by Maarten Lankhorst 18 Aug '23

18 Aug '23

From: Maarten Lankhorst <dev(a)lankhorst.se> Explicitly loading i915 becomes a problem when upstreaming the new intel driver for Tiger Lake and higher graphics (xe). By loading i915, it doesn't wait for driver load of xe, and will fail completely before it loads. -EPROBE_DEFER has to be returned before any device is created in probe(), otherwise the removal of the device will cause EPROBE_DEFER to try again in an infinite loop. The conversion is done in gradual steps. First I add an argument to snd_hdac_i915_init to allow for -EPROBE_DEFER so I can convert each driver separately. Then I convert each driver to move snd_hdac_i915_init out of the workqueue. Finally I drop the ability to choose modprobe behavior after the last user is converted. I suspect the avs and skylake drivers used snd_hdac_i915_init purely for the modprobe, but I don't have the hardware to test if it can be safely removed. It can still be done easily in a followup patch to simplify probing. --- Changes since previous version: - Keep the workqueue for soc/sof/intel, instead only move out snd_hdac_i915_init. (Patch 8/9) Rest of patches unchanged. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Cezary Rojewski <cezary.rojewski(a)intel.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Cc: Bard Liao <yung-chuan.liao(a)linux.intel.com> Cc: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Cc: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Daniel Baluta <daniel.baluta(a)nxp.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Cc: sound-open-firmware(a)alsa-project.org Maarten Lankhorst (9): ALSA: hda/intel: Fix error handling in azx_probe() ALSA: hda/i915: Allow override of gpu binding. ALSA: hda/i915: Add an allow_modprobe argument to snd_hdac_i915_init ALSA: hda/i915: Allow xe as match for i915_component_master_match ASoC: Intel: avs: Move snd_hdac_i915_init to before probe_work. ASoC: Intel: Skylake: Move snd_hdac_i915_init to before probe_work. ALSA: hda/intel: Move snd_hdac_i915_init to before probe_work. ASoC: SOF: Intel: Move binding to display driver outside of deferred probe ALSA: hda/i915: Remove extra argument from snd_hdac_i915_init sound/hda/hdac_i915.c | 25 ++++++++------- sound/pci/hda/hda_intel.c | 60 ++++++++++++++++++----------------- sound/soc/intel/avs/core.c | 13 +++++--- sound/soc/intel/skylake/skl.c | 31 ++++++------------ sound/soc/sof/core.c | 19 ++++------- 5 files changed, 70 insertions(+), 78 deletions(-) -- 2.39.2

5 23

Re: [PATCH 3/4] ASoC: amd: acp: Add machine driver that enables sound for systems with a ES8336 codec
by Zhu Ning 18 Aug '23

18 Aug '23

> > This is saying that the machine driver should disable jack detection > > over suspend and restart it during resume. The machine driver should > > suspend before the rest of the card which should mean that the CODEC > > gets powered off then. The core can't tell if jack detection is > > supposed to work over suspend, it is a standard wake event on systems > > like phones, but it sounds like on this system the power gets removed > > from the device so that can't work. > > > > Sorry, I don't understand what you are trying to say here. My intention > is to find a way to have sound working when suspending/resuming > with jack inserted by not fudging the jack in the machine driver but > fixing it in the CODEC. > > > I don't understand why that would be expected to help? The main effect > > of keeping the bias on all the time would be to consume more power. > > I don't fully understand the whole bias thing (I did try hard to > understand it from the code), but in this specific instance it helps for > suspending the CODEC. If idle_bias_on is true then idle_bias_off will be > false, that would mean that during suspend (with jack inserted) > in snd_soc_suspend() in switch (snd_soc_dapm_get_bias_level(dapm)) > the flow goes in case SND_SOC_BIAS_STANDBY and since dapm->idle_bias_off > is false it doesn't break from the case but falls through to > SND_SOC_BIAS_OFF case which in turn calls the CODEC component's suspend > callback. > > The reason the sound stops working with jack inserted is that the CODEC > suspend callback never gets called. It only gets called when the jack > is not inserted. Hi Postevca, I just reproduced the bug on my side. You are right that when jack inserted, the codec driver will never enter SND_SOC_BIAS_OFF or suspend or resume. This makes sense because the codec needs micbias power to detect button. Problem is that ALSA or dapm thought the codec power is on during suspend, but the system power is turned off and the registers are reset. The system power is controlled by EC and neither dapm or codec driver can do anything about it. The system power is just turned off. Disabling the jack detection before suspend is ok because the system stops working during suspend anyway. Maybe a detailed comment is needed to explain why?

1 0

[PATCH] ALSA: aoa: Fix typos in PCM fix patch
by Takashi Iwai 18 Aug '23

18 Aug '23

There was typos in the previous fix for PCM to detach the struct device that caused build errors. Corrected here. Fixes: bc41a7228ced ("ALSA: pcm: Don't embed device") Reported-by: kernel test robot <lkp(a)intel.com> Closes: https://lore.kernel.org/oe-kbuild-all/202308181347.q3XPr3Lm-lkp@intel.com/ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/aoa/soundbus/i2sbus/pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/aoa/soundbus/i2sbus/pcm.c b/sound/aoa/soundbus/i2sbus/pcm.c index 3680eb6eabc9..07df5cc0f2d7 100644 --- a/sound/aoa/soundbus/i2sbus/pcm.c +++ b/sound/aoa/soundbus/i2sbus/pcm.c @@ -972,7 +972,7 @@ i2sbus_attach_codec(struct soundbus_dev *dev, struct snd_card *card, goto out_put_ci_module; snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_PLAYBACK, &i2sbus_playback_ops); - dev->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK]->dev.parent = + dev->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].dev->parent = &dev->ofdev.dev; i2sdev->out.created = 1; } @@ -989,7 +989,7 @@ i2sbus_attach_codec(struct soundbus_dev *dev, struct snd_card *card, goto out_put_ci_module; snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_CAPTURE, &i2sbus_record_ops); - dev->pcm->streams[SNDRV_PCM_STREAM_CAPTURE]->dev.parent = + dev->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].dev->parent = &dev->ofdev.dev; i2sdev->in.created = 1; } -- 2.35.3

1 0

[tiwai-sound:topic/dev-split-kref 3/11] sound/aoa/soundbus/i2sbus/pcm.c:975:47: error: member reference type 'struct snd_pcm_str' is not a pointer; did you mean to use '.'?
by kernel test robot 18 Aug '23

18 Aug '23

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/dev-split-kref head: 6e1b44385310dd465cae48213eae65d0b8095d13 commit: 7a3dce3fb37c25de3f3c347f4e69ba358cb81a3d [3/11] ALSA: pcm: Don't embed device config: powerpc-g5_defconfig (https://download.01.org/0day-ci/archive/20230818/202308181347.q3XPr3Lm-lkp@…) compiler: clang version 15.0.7 (https://github.com/llvm/llvm-project.git 8dfdcc7b7bf66834a761bd8de445840ef68e4d1a) reproduce: (https://download.01.org/0day-ci/archive/20230818/202308181347.q3XPr3Lm-lkp@…) 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/202308181347.q3XPr3Lm-lkp@intel.com/ All errors (new ones prefixed by >>): >> sound/aoa/soundbus/i2sbus/pcm.c:975:47: error: member reference type 'struct snd_pcm_str' is not a pointer; did you mean to use '.'? dev->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK]->dev.parent = ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~^~ . >> sound/aoa/soundbus/i2sbus/pcm.c:975:52: error: member reference type 'struct device *' is a pointer; did you mean to use '->'? dev->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK]->dev.parent = ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~^ -> sound/aoa/soundbus/i2sbus/pcm.c:992:46: error: member reference type 'struct snd_pcm_str' is not a pointer; did you mean to use '.'? dev->pcm->streams[SNDRV_PCM_STREAM_CAPTURE]->dev.parent = ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~^~ . sound/aoa/soundbus/i2sbus/pcm.c:992:51: error: member reference type 'struct device *' is a pointer; did you mean to use '->'? dev->pcm->streams[SNDRV_PCM_STREAM_CAPTURE]->dev.parent = ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~^ -> 4 errors generated. vim +975 sound/aoa/soundbus/i2sbus/pcm.c 866 867 int 868 i2sbus_attach_codec(struct soundbus_dev *dev, struct snd_card *card, 869 struct codec_info *ci, void *data) 870 { 871 int err, in = 0, out = 0; 872 struct transfer_info *tmp; 873 struct i2sbus_dev *i2sdev = soundbus_dev_to_i2sbus_dev(dev); 874 struct codec_info_item *cii; 875 876 if (!dev->pcmname || dev->pcmid == -1) { 877 printk(KERN_ERR "i2sbus: pcm name and id must be set!\n"); 878 return -EINVAL; 879 } 880 881 list_for_each_entry(cii, &dev->codec_list, list) { 882 if (cii->codec_data == data) 883 return -EALREADY; 884 } 885 886 if (!ci->transfers || !ci->transfers->formats 887 || !ci->transfers->rates || !ci->usable) 888 return -EINVAL; 889 890 /* we currently code the i2s transfer on the clock, and support only 891 * 32 and 64 */ 892 if (ci->bus_factor != 32 && ci->bus_factor != 64) 893 return -EINVAL; 894 895 /* If you want to fix this, you need to keep track of what transport infos 896 * are to be used, which codecs they belong to, and then fix all the 897 * sysclock/busclock stuff above to depend on which is usable */ 898 list_for_each_entry(cii, &dev->codec_list, list) { 899 if (cii->codec->sysclock_factor != ci->sysclock_factor) { 900 printk(KERN_DEBUG 901 "cannot yet handle multiple different sysclocks!\n"); 902 return -EINVAL; 903 } 904 if (cii->codec->bus_factor != ci->bus_factor) { 905 printk(KERN_DEBUG 906 "cannot yet handle multiple different bus clocks!\n"); 907 return -EINVAL; 908 } 909 } 910 911 tmp = ci->transfers; 912 while (tmp->formats && tmp->rates) { 913 if (tmp->transfer_in) 914 in = 1; 915 else 916 out = 1; 917 tmp++; 918 } 919 920 cii = kzalloc(sizeof(struct codec_info_item), GFP_KERNEL); 921 if (!cii) 922 return -ENOMEM; 923 924 /* use the private data to point to the codec info */ 925 cii->sdev = soundbus_dev_get(dev); 926 cii->codec = ci; 927 cii->codec_data = data; 928 929 if (!cii->sdev) { 930 printk(KERN_DEBUG 931 "i2sbus: failed to get soundbus dev reference\n"); 932 err = -ENODEV; 933 goto out_free_cii; 934 } 935 936 if (!try_module_get(THIS_MODULE)) { 937 printk(KERN_DEBUG "i2sbus: failed to get module reference!\n"); 938 err = -EBUSY; 939 goto out_put_sdev; 940 } 941 942 if (!try_module_get(ci->owner)) { 943 printk(KERN_DEBUG 944 "i2sbus: failed to get module reference to codec owner!\n"); 945 err = -EBUSY; 946 goto out_put_this_module; 947 } 948 949 if (!dev->pcm) { 950 err = snd_pcm_new(card, dev->pcmname, dev->pcmid, 0, 0, 951 &dev->pcm); 952 if (err) { 953 printk(KERN_DEBUG "i2sbus: failed to create pcm\n"); 954 goto out_put_ci_module; 955 } 956 } 957 958 /* ALSA yet again sucks. 959 * If it is ever fixed, remove this line. See below. */ 960 out = in = 1; 961 962 if (!i2sdev->out.created && out) { 963 if (dev->pcm->card != card) { 964 /* eh? */ 965 printk(KERN_ERR 966 "Can't attach same bus to different cards!\n"); 967 err = -EINVAL; 968 goto out_put_ci_module; 969 } 970 err = snd_pcm_new_stream(dev->pcm, SNDRV_PCM_STREAM_PLAYBACK, 1); 971 if (err) 972 goto out_put_ci_module; 973 snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_PLAYBACK, 974 &i2sbus_playback_ops); > 975 dev->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK]->dev.parent = 976 &dev->ofdev.dev; 977 i2sdev->out.created = 1; 978 } 979 980 if (!i2sdev->in.created && in) { 981 if (dev->pcm->card != card) { 982 printk(KERN_ERR 983 "Can't attach same bus to different cards!\n"); 984 err = -EINVAL; 985 goto out_put_ci_module; 986 } 987 err = snd_pcm_new_stream(dev->pcm, SNDRV_PCM_STREAM_CAPTURE, 1); 988 if (err) 989 goto out_put_ci_module; 990 snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_CAPTURE, 991 &i2sbus_record_ops); 992 dev->pcm->streams[SNDRV_PCM_STREAM_CAPTURE]->dev.parent = 993 &dev->ofdev.dev; 994 i2sdev->in.created = 1; 995 } 996 997 /* so we have to register the pcm after adding any substream 998 * to it because alsa doesn't create the devices for the 999 * substreams when we add them later. 1000 * Therefore, force in and out on both busses (above) and 1001 * register the pcm now instead of just after creating it. 1002 */ 1003 err = snd_device_register(card, dev->pcm); 1004 if (err) { 1005 printk(KERN_ERR "i2sbus: error registering new pcm\n"); 1006 goto out_put_ci_module; 1007 } 1008 /* no errors any more, so let's add this to our list */ 1009 list_add(&cii->list, &dev->codec_list); 1010 1011 dev->pcm->private_data = i2sdev; 1012 dev->pcm->private_free = i2sbus_private_free; 1013 1014 /* well, we really should support scatter/gather DMA */ 1015 snd_pcm_set_managed_buffer_all( 1016 dev->pcm, SNDRV_DMA_TYPE_DEV, 1017 &macio_get_pci_dev(i2sdev->macio)->dev, 1018 64 * 1024, 64 * 1024); 1019 1020 return 0; 1021 out_put_ci_module: 1022 module_put(ci->owner); 1023 out_put_this_module: 1024 module_put(THIS_MODULE); 1025 out_put_sdev: 1026 soundbus_dev_put(dev); 1027 out_free_cii: 1028 kfree(cii); 1029 return err; 1030 } 1031 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[PATCH RFC 0/6] ALSA: Fix UAF with delayed kobj release
by Takashi Iwai 18 Aug '23

18 Aug '23

Hi, this is a test patch set for addressing the UAF problems with delayed kobj releases reported by Curtis: https://lore.kernel.org/r/20230801171928.1460120-1-cujomalainey@chromium.org The patch introduced a simple helper for allocating memory with a refcount, and converts the card object, control, PCM and compress objects with the new type. With the refcount, the actual memory release is delayed until all referrer are gone. It's just a RFC and only lightly tested. I myself am not sure whether this is the best way to go. It might be better to take Curtis' approach, just converting the device to its own allocation, too. (But I don't know whether Curtis' patch set covers all cases -- can still be a UAF of card_dev due to devres vs kobj release?) thanks, Takashi === Takashi Iwai (6): ALSA: core: Introduced referenced memory allocator ALSA: core: Fix potential UAF by delayed kobject release of card_dev ALSA: core: Associate memory reference with device initialization ALSA: pcm: Release memory with reference ALSA: control: Reference card by ctl_dev ALSA: compress: Reference card by the device include/sound/core.h | 7 ++- sound/core/compress_offload.c | 2 +- sound/core/control.c | 2 +- sound/core/hwdep.c | 2 +- sound/core/init.c | 105 +++++++++++++++++++++++++++------ sound/core/pcm.c | 6 +- sound/core/rawmidi.c | 2 +- sound/core/seq/seq_clientmgr.c | 2 +- sound/core/timer.c | 2 +- 9 files changed, 101 insertions(+), 29 deletions(-) -- 2.35.3

2 23

[PATCH v2 0/5] ASoC: Convert remaining Realtek codecs to GPIO descriptors
by Linus Walleij 18 Aug '23

18 Aug '23

After dropping unused headers a few Realtek devices actually using the GPIO descriptors remain. Converting them to use optional GPIO descriptors is pretty straight-forward. Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- Changes in v2: - Rebased on asoc-6.6 - Link to v1: https://lore.kernel.org/r/20230817-descriptors-asoc-rt-v1-0-434f5f177cee@li… --- Linus Walleij (5): ASoC: rt5640: Convert to just use GPIO descriptors ASoC: rt5665: Convert to use GPIO descriptors ASoC: rt5668: Convert to use GPIO descriptors ASoC: rt5682: Convert to use GPIO descriptors ASoC: rt5682s: Convert to use GPIO descriptors include/sound/rt5665.h | 2 -- include/sound/rt5668.h | 3 --- include/sound/rt5682.h | 3 --- include/sound/rt5682s.h | 3 --- sound/soc/codecs/rt5640.c | 55 ++++++++++++------------------------------- sound/soc/codecs/rt5640.h | 2 +- sound/soc/codecs/rt5665.c | 17 +++++++------ sound/soc/codecs/rt5668.c | 17 +++++++------ sound/soc/codecs/rt5682-i2c.c | 11 ++++----- sound/soc/codecs/rt5682-sdw.c | 5 ++++ sound/soc/codecs/rt5682.c | 20 ++++++++++++---- sound/soc/codecs/rt5682.h | 3 +++ sound/soc/codecs/rt5682s.c | 16 ++++++------- sound/soc/codecs/rt5682s.h | 2 ++ 14 files changed, 68 insertions(+), 91 deletions(-) --- base-commit: 17b9f4387ebabb19b871bbe2d06562e48e4e7130 change-id: 20230817-descriptors-asoc-rt-9c072934d376 Best regards, -- Linus Walleij <linus.walleij(a)linaro.org>

2 6

[PATCH 0/5] ASoC: Convert remaining Realtek codecs to GPIO descriptors
by Linus Walleij 18 Aug '23

18 Aug '23

After dropping unused headers a few Realtek devices actually using the GPIO descriptors remain. Converting them to use optional GPIO descriptors is pretty straight-forward. Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- Linus Walleij (5): ASoC: rt5640: Convert to just use GPIO descriptors ASoC: rt5665: Convert to use GPIO descriptors ASoC: rt5668: Convert to use GPIO descriptors ASoC: rt5682: Convert to use GPIO descriptors ASoC: rt5682s: Convert to use GPIO descriptors include/sound/rt5665.h | 2 -- include/sound/rt5668.h | 3 --- include/sound/rt5682.h | 3 --- include/sound/rt5682s.h | 3 --- sound/soc/codecs/rt5640.c | 55 ++++++++++++------------------------------- sound/soc/codecs/rt5640.h | 2 +- sound/soc/codecs/rt5665.c | 17 +++++++------ sound/soc/codecs/rt5668.c | 17 +++++++------ sound/soc/codecs/rt5682-i2c.c | 11 ++++----- sound/soc/codecs/rt5682-sdw.c | 4 ++++ sound/soc/codecs/rt5682.c | 20 ++++++++++++---- sound/soc/codecs/rt5682.h | 3 +++ sound/soc/codecs/rt5682s.c | 16 ++++++------- sound/soc/codecs/rt5682s.h | 2 ++ 14 files changed, 67 insertions(+), 91 deletions(-) --- base-commit: c5b06b35a73b80f1af931c2371de084fa77b5c23 change-id: 20230817-descriptors-asoc-rt-9c072934d376 Best regards, -- Linus Walleij <linus.walleij(a)linaro.org>

2 7

[PATCH v1] ASoC: tas2781: fixed register access error when switching to other chips
by Shenghao Ding 18 Aug '23

18 Aug '23

fixed register access error when switching to other tas2781 -- refresh the page inside regmap on the switched tas2781 Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- Changes in v1: - fixed register access error when switching to other tas2781 --- sound/soc/codecs/tas2781-comlib.c | 19 ++++++++++--------- 1 file changed, 10 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/tas2781-comlib.c b/sound/soc/codecs/tas2781-comlib.c index a88c6c28a394..ffb26e4a7e2f 100644 --- a/sound/soc/codecs/tas2781-comlib.c +++ b/sound/soc/codecs/tas2781-comlib.c @@ -57,16 +57,17 @@ static int tasdevice_change_chn_book(struct tasdevice_priv *tas_priv, if (client->addr != tasdev->dev_addr) { client->addr = tasdev->dev_addr; - if (tasdev->cur_book == book) { - ret = regmap_write(map, - TASDEVICE_PAGE_SELECT, 0); - if (ret < 0) { - dev_err(tas_priv->dev, "%s, E=%d\n", - __func__, ret); - goto out; - } + /* All tas2781s share the same regmap, clear the page + * inside regmap once switching to another tas2781. + * Register 0 at any pages and any books inside tas2781 + * is the same one for page-switching. + */ + ret = regmap_write(map, TASDEVICE_PAGE_SELECT, 0); + if (ret < 0) { + dev_err(tas_priv->dev, "%s, E=%d\n", + __func__, ret); + goto out; } - goto out; } if (tasdev->cur_book != book) { -- 2.34.1

2 1

[PATCH v2 0/2] ASoC: cs35l56: Update ACPI HID and property
by Richard Fitzgerald 18 Aug '23

18 Aug '23

These two patches add an ACPI HID and update the way the platform- specific firmware identifier is extracted from the ACPI. CHANGES SINCE V1: - Rebased to apply on v6.4 and v6.5. - Change kstrdup() to devm_kstrdup() To apply on v6.6 a 1-line change is needed: static int cs35l56_get_firmware_uid(struct cs35l56_private *cs35l56) { - struct device *dev = cs35l56->dev; + struct device *dev = cs35l56->base.dev; Maciej Strozek (1): ASoC: cs35l56: Read firmware uuid from a device property instead of _SUB Simon Trimmer (1): ASoC: cs35l56: Add an ACPI match table sound/soc/codecs/cs35l56-i2c.c | 9 +++++++++ sound/soc/codecs/cs35l56-spi.c | 9 +++++++++ sound/soc/codecs/cs35l56.c | 31 ++++++++++++------------------- 3 files changed, 30 insertions(+), 19 deletions(-) -- 2.30.2

2 3

[PATCH 0/2] ASoC: cs35l56: Update ACPI HID and property
by Richard Fitzgerald 18 Aug '23

18 Aug '23

These two patches add an ACPI HID and update the way the platform- specific firmware identifier is extracted from the ACPI. Maciej Strozek (1): ASoC: cs35l56: Read firmware uuid from a device property instead of _SUB Simon Trimmer (1): ASoC: cs35l56: Add an ACPI match table sound/soc/codecs/cs35l56-i2c.c | 9 +++++++++ sound/soc/codecs/cs35l56-spi.c | 9 +++++++++ sound/soc/codecs/cs35l56.c | 31 ++++++++++++++----------------- 3 files changed, 32 insertions(+), 17 deletions(-) -- 2.30.2

2 6

[PATCH v2 01/11] ASoC: dt-bindings: Add common sound card properties
by Krzysztof Kozlowski 18 Aug '23

18 Aug '23

Almost every board machine / sound cards has "audio-routing" and "model" properties, so move them to common schema to have only one definition of these properties. For amlogic,gx-sound-card, drop the minItems: 2 from the "audio-routing", because any limit here - lower or upper- is rather meaningless. This will also fix `dtbs_check` warnings like: meson-gxm-s912-libretech-pc.dtb: sound: audio-routing: ['AU2 INL', 'ACODEC LOLN', 'AU2 INR', 'ACODEC LORN', '7J4-14 LEFT', 'AU2 OUTL', '7J4-11 RIGHT', 'AU2 OUTR'] is too long Reviewed-by: AngeloGioacchino Del Regno <angelogioacchino.delregno(a)collabora.com> Reviewed-by: Rob Herring <robh(a)kernel.org> Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Changes in v2: 1. Drop minItems/maxItems from amlogic,gx-sound-card --- .../sound/amlogic,axg-sound-card.yaml | 15 +++-------- .../bindings/sound/amlogic,gx-sound-card.yaml | 17 +++--------- .../devicetree/bindings/sound/fsl,rpmsg.yaml | 17 +++--------- .../bindings/sound/google,sc7180-trogdor.yaml | 17 +++--------- .../sound/google,sc7280-herobrine.yaml | 17 +++--------- .../bindings/sound/imx-audio-card.yaml | 18 +++---------- .../bindings/sound/sound-card-common.yaml | 27 +++++++++++++++++++ 7 files changed, 50 insertions(+), 78 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/sound-card-common.yaml diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.yaml b/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.yaml index bf1234550343..5db718e4d0e7 100644 --- a/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.yaml +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.yaml @@ -9,6 +9,9 @@ title: Amlogic AXG sound card maintainers: - Jerome Brunet <jbrunet(a)baylibre.com> +allOf: + - $ref: sound-card-common.yaml# + properties: compatible: const: amlogic,axg-sound-card @@ -17,23 +20,12 @@ properties: $ref: /schemas/types.yaml#/definitions/phandle-array description: list of auxiliary devices - audio-routing: - $ref: /schemas/types.yaml#/definitions/non-unique-string-array - description: - A list of the connections between audio components. Each entry is a - pair of strings, the first being the connection's sink, the second - being the connection's source. - audio-widgets: $ref: /schemas/types.yaml#/definitions/non-unique-string-array description: A list off component DAPM widget. Each entry is a pair of strings, the first being the widget type, the second being the widget name - model: - $ref: /schemas/types.yaml#/definitions/string - description: User specified audio sound card name - patternProperties: "^dai-link-[0-9]+$": type: object @@ -108,7 +100,6 @@ patternProperties: - sound-dai required: - - model - dai-link-0 unevaluatedProperties: false diff --git a/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml b/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml index b358fd601ed3..d4277d342e69 100644 --- a/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml +++ b/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml @@ -9,6 +9,9 @@ title: Amlogic GX sound card maintainers: - Jerome Brunet <jbrunet(a)baylibre.com> +allOf: + - $ref: sound-card-common.yaml# + properties: compatible: items: @@ -18,14 +21,6 @@ properties: $ref: /schemas/types.yaml#/definitions/phandle-array description: list of auxiliary devices - audio-routing: - $ref: /schemas/types.yaml#/definitions/non-unique-string-array - minItems: 2 - description: |- - A list of the connections between audio components. Each entry is a - pair of strings, the first being the connection's sink, the second - being the connection's source. - audio-widgets: $ref: /schemas/types.yaml#/definitions/non-unique-string-array minItems: 2 @@ -33,10 +28,6 @@ properties: A list off component DAPM widget. Each entry is a pair of strings, the first being the widget type, the second being the widget name - model: - $ref: /schemas/types.yaml#/definitions/string - description: User specified audio sound card name - patternProperties: "^dai-link-[0-9]+$": type: object @@ -86,7 +77,7 @@ required: - model - dai-link-0 -additionalProperties: false +unevaluatedProperties: false examples: - | diff --git a/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml b/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml index 6df0e03a1d4b..188f38baddec 100644 --- a/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml +++ b/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml @@ -17,6 +17,9 @@ description: | such as SAI, MICFIL, .etc through building rpmsg channels between Cortex-A and Cortex-M. +allOf: + - $ref: sound-card-common.yaml# + properties: compatible: enum: @@ -27,10 +30,6 @@ properties: - fsl,imx8ulp-rpmsg-audio - fsl,imx93-rpmsg-audio - model: - $ref: /schemas/types.yaml#/definitions/string - description: User specified audio sound card name - clocks: items: - description: Peripheral clock for register access @@ -66,13 +65,6 @@ properties: $ref: /schemas/types.yaml#/definitions/phandle description: The phandle to a node of audio codec - audio-routing: - $ref: /schemas/types.yaml#/definitions/non-unique-string-array - description: | - A list of the connections between audio components. Each entry is a - pair of strings, the first being the connection's sink, the second - being the connection's source. - fsl,enable-lpa: $ref: /schemas/types.yaml#/definitions/flag description: enable low power audio path. @@ -101,9 +93,8 @@ properties: required: - compatible - - model -additionalProperties: false +unevaluatedProperties: false examples: - | diff --git a/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml b/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml index ba5b7728cf33..bac940553965 100644 --- a/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml +++ b/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml @@ -13,23 +13,15 @@ maintainers: description: This binding describes the SC7180 sound card which uses LPASS for audio. +allOf: + - $ref: sound-card-common.yaml# + properties: compatible: enum: - google,sc7180-trogdor - google,sc7180-coachz - audio-routing: - $ref: /schemas/types.yaml#/definitions/non-unique-string-array - description: - A list of the connections between audio components. Each entry is a - pair of strings, the first being the connection's sink, the second - being the connection's source. - - model: - $ref: /schemas/types.yaml#/definitions/string - description: User specified audio sound card name - "#address-cells": const: 1 @@ -86,11 +78,10 @@ patternProperties: required: - compatible - - model - "#address-cells" - "#size-cells" -additionalProperties: false +unevaluatedProperties: false examples: diff --git a/Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml b/Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml index 0b1a01a4c14e..ec4b6e547ca6 100644 --- a/Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml +++ b/Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml @@ -13,22 +13,14 @@ maintainers: description: This binding describes the SC7280 sound card which uses LPASS for audio. +allOf: + - $ref: sound-card-common.yaml# + properties: compatible: enum: - google,sc7280-herobrine - audio-routing: - $ref: /schemas/types.yaml#/definitions/non-unique-string-array - description: - A list of the connections between audio components. Each entry is a - pair of strings, the first being the connection's sink, the second - being the connection's source. - - model: - $ref: /schemas/types.yaml#/definitions/string - description: User specified audio sound card name - "#address-cells": const: 1 @@ -97,11 +89,10 @@ patternProperties: required: - compatible - - model - "#address-cells" - "#size-cells" -additionalProperties: false +unevaluatedProperties: false examples: diff --git a/Documentation/devicetree/bindings/sound/imx-audio-card.yaml b/Documentation/devicetree/bindings/sound/imx-audio-card.yaml index b6f5d486600e..f7ad5ea2491e 100644 --- a/Documentation/devicetree/bindings/sound/imx-audio-card.yaml +++ b/Documentation/devicetree/bindings/sound/imx-audio-card.yaml @@ -9,23 +9,14 @@ title: NXP i.MX audio sound card. maintainers: - Shengjiu Wang <shengjiu.wang(a)nxp.com> +allOf: + - $ref: sound-card-common.yaml# + properties: compatible: enum: - fsl,imx-audio-card - model: - $ref: /schemas/types.yaml#/definitions/string - description: User specified audio sound card name - - audio-routing: - $ref: /schemas/types.yaml#/definitions/non-unique-string-array - description: - A list of the connections between audio components. Each entry is a - pair of strings, the first being the connection's sink, the second - being the connection's source. Valid names could be power supplies, - MicBias of codec and the jacks on the board. - patternProperties: ".*-dai-link$": description: @@ -84,9 +75,8 @@ patternProperties: required: - compatible - - model -additionalProperties: false +unevaluatedProperties: false examples: - | diff --git a/Documentation/devicetree/bindings/sound/sound-card-common.yaml b/Documentation/devicetree/bindings/sound/sound-card-common.yaml new file mode 100644 index 000000000000..3a941177f684 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/sound-card-common.yaml @@ -0,0 +1,27 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/sound-card-common.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Board Sound Card Common Properties + +maintainers: + - Mark Brown <broonie(a)kernel.org> + +properties: + audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the connections between audio components. Each entry is a + pair of strings, the first being the connection's sink, the second + being the connection's source. + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User specified audio sound card name + +required: + - model + +additionalProperties: true -- 2.34.1

4 20

[PATCH 00/12] ASoC: dt-bindings: Add sound-card-common.yaml
by Krzysztof Kozlowski 18 Aug '23

18 Aug '23

Hi, Almost every board machine / sound cards has "audio-routing" and "model" properties, so we can make things simpler by introducing one common binding. It is also expected that given property has only one definition. If the patchset looks good, then the next steps will be: 1. Convert more vendor audio-routing properties to audio-routing, 2. Add dai-links children to common binding. Best regards, Krzysztof Krzysztof Kozlowski (12): ASoC: dt-bindings: amlogic,gx-sound-card: correct maxItems constraints ASoC: dt-bindings: Add common sound card properties ASoC: dt-bindings: mediatek,mt8188-mt6359: use common sound card ASoC: dt-bindings: samsung,aries-wm8994: use common sound card ASoC: dt-bindings: samsung,midas-audio: use common sound card ASoC: dt-bindings: samsung,odroid: use common sound card ASoC: dt-bindings: samsung,tm2: use common sound card ASoC: samsung: odroid: use of_property_present to check for property ASoC: samsung: aries_wm8994: parse audio-routing ASoC: samsung: midas_wm1811: parse audio-routing ASoC: samsung: odroid: parse audio-routing ASoC: samsung: tm2_wm5110: parse audio-routing .../sound/amlogic,axg-sound-card.yaml | 15 +++-------- .../bindings/sound/amlogic,gx-sound-card.yaml | 16 +++++------ .../devicetree/bindings/sound/fsl,rpmsg.yaml | 17 +++--------- .../bindings/sound/google,sc7180-trogdor.yaml | 17 +++--------- .../sound/google,sc7280-herobrine.yaml | 17 +++--------- .../bindings/sound/imx-audio-card.yaml | 18 +++---------- .../sound/mediatek,mt8188-mt6359.yaml | 17 +++++------- .../bindings/sound/samsung,aries-wm8994.yaml | 16 +++++------ .../bindings/sound/samsung,midas-audio.yaml | 16 +++++------ .../bindings/sound/samsung,odroid.yaml | 14 +++++----- .../bindings/sound/samsung,tm2.yaml | 16 +++++------ .../bindings/sound/sound-card-common.yaml | 27 +++++++++++++++++++ sound/soc/samsung/aries_wm8994.c | 10 ++++--- sound/soc/samsung/midas_wm1811.c | 10 ++++--- sound/soc/samsung/odroid.c | 15 ++++++----- sound/soc/samsung/tm2_wm5110.c | 10 ++++--- 16 files changed, 119 insertions(+), 132 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/sound-card-common.yaml -- 2.34.1

5 30

[PATCH v1] ALSA: hda: cs35l41: Support systems with missing _DSD properties
by Stefan Binding 18 Aug '23

18 Aug '23

Some systems using CS35L41 with HDA were released without some required _DSD properties in ACPI. To support these special cases, add an api to configure the correct properties for systems with this issue. This initial commit moves the no _DSD support for Lenovo Legion Laptops (CLSA0100, CLSA0101) into a new framework which can be extended to support additional laptops in the future. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> --- sound/pci/hda/Makefile | 2 +- sound/pci/hda/cs35l41_hda.c | 65 ++++++------------------- sound/pci/hda/cs35l41_hda.h | 1 + sound/pci/hda/cs35l41_hda_property.c | 73 ++++++++++++++++++++++++++++ sound/pci/hda/cs35l41_hda_property.h | 18 +++++++ 5 files changed, 108 insertions(+), 51 deletions(-) create mode 100644 sound/pci/hda/cs35l41_hda_property.c create mode 100644 sound/pci/hda/cs35l41_hda_property.h diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index c6e6509e7b8e..5506255be895 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -28,7 +28,7 @@ snd-hda-codec-via-objs := patch_via.o snd-hda-codec-hdmi-objs := patch_hdmi.o hda_eld.o # side codecs -snd-hda-scodec-cs35l41-objs := cs35l41_hda.o +snd-hda-scodec-cs35l41-objs := cs35l41_hda.o cs35l41_hda_property.o snd-hda-scodec-cs35l41-i2c-objs := cs35l41_hda_i2c.o snd-hda-scodec-cs35l41-spi-objs := cs35l41_hda_spi.o snd-hda-scodec-cs35l56-objs := cs35l56_hda.o diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 825e551be9bb..f9b77353c266 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -19,6 +19,7 @@ #include "hda_component.h" #include "cs35l41_hda.h" #include "hda_cs_dsp_ctl.h" +#include "cs35l41_hda_property.h" #define CS35L41_FIRMWARE_ROOT "cirrus/" #define CS35L41_PART "cs35l41" @@ -1315,8 +1316,7 @@ static int cs35l41_hda_apply_properties(struct cs35l41_hda *cs35l41) return cs35l41_hda_channel_map(cs35l41->dev, 0, NULL, 1, &hw_cfg->spk_pos); } -static int cs35l41_get_speaker_id(struct device *dev, int amp_index, - int num_amps, int fixed_gpio_id) +int cs35l41_get_speaker_id(struct device *dev, int amp_index, int num_amps, int fixed_gpio_id) { struct gpio_desc *speaker_id_desc; int speaker_id = -ENODEV; @@ -1370,49 +1370,6 @@ static int cs35l41_get_speaker_id(struct device *dev, int amp_index, return speaker_id; } -/* - * Device CLSA010(0/1) doesn't have _DSD so a gpiod_get by the label reset won't work. - * And devices created by serial-multi-instantiate don't have their device struct - * pointing to the correct fwnode, so acpi_dev must be used here. - * And devm functions expect that the device requesting the resource has the correct - * fwnode. - */ -static int cs35l41_no_acpi_dsd(struct cs35l41_hda *cs35l41, struct device *physdev, int id, - const char *hid) -{ - struct cs35l41_hw_cfg *hw_cfg = &cs35l41->hw_cfg; - - /* check I2C address to assign the index */ - cs35l41->index = id == 0x40 ? 0 : 1; - cs35l41->channel_index = 0; - cs35l41->reset_gpio = gpiod_get_index(physdev, NULL, 0, GPIOD_OUT_HIGH); - cs35l41->speaker_id = cs35l41_get_speaker_id(physdev, 0, 0, 2); - hw_cfg->spk_pos = cs35l41->index; - hw_cfg->gpio2.func = CS35L41_INTERRUPT; - hw_cfg->gpio2.valid = true; - hw_cfg->valid = true; - - if (strncmp(hid, "CLSA0100", 8) == 0) { - hw_cfg->bst_type = CS35L41_EXT_BOOST_NO_VSPK_SWITCH; - } else if (strncmp(hid, "CLSA0101", 8) == 0) { - hw_cfg->bst_type = CS35L41_EXT_BOOST; - hw_cfg->gpio1.func = CS35l41_VSPK_SWITCH; - hw_cfg->gpio1.valid = true; - } else { - /* - * Note: CLSA010(0/1) are special cases which use a slightly different design. - * All other HIDs e.g. CSC3551 require valid ACPI _DSD properties to be supported. - */ - dev_err(cs35l41->dev, "Error: ACPI _DSD Properties are missing for HID %s.\n", hid); - hw_cfg->valid = false; - hw_cfg->gpio1.valid = false; - hw_cfg->gpio2.valid = false; - return -EINVAL; - } - - return 0; -} - static int cs35l41_hda_read_acpi(struct cs35l41_hda *cs35l41, const char *hid, int id) { struct cs35l41_hw_cfg *hw_cfg = &cs35l41->hw_cfg; @@ -1438,12 +1395,17 @@ static int cs35l41_hda_read_acpi(struct cs35l41_hda *cs35l41, const char *hid, i sub = NULL; cs35l41->acpi_subsystem_id = sub; + ret = cs35l41_add_dsd_properties(cs35l41, physdev, id, hid); + if (!ret) { + dev_info(cs35l41->dev, "Using extra _DSD properties, bypassing _DSD in ACPI\n"); + goto put_physdev; + } + property = "cirrus,dev-index"; ret = device_property_count_u32(physdev, property); - if (ret <= 0) { - ret = cs35l41_no_acpi_dsd(cs35l41, physdev, id, hid); - goto err_put_physdev; - } + if (ret <= 0) + goto err; + if (ret > ARRAY_SIZE(values)) { ret = -EINVAL; goto err; @@ -1533,7 +1495,10 @@ static int cs35l41_hda_read_acpi(struct cs35l41_hda *cs35l41, const char *hid, i err: dev_err(cs35l41->dev, "Failed property %s: %d\n", property, ret); -err_put_physdev: + hw_cfg->valid = false; + hw_cfg->gpio1.valid = false; + hw_cfg->gpio2.valid = false; +put_physdev: put_device(physdev); return ret; diff --git a/sound/pci/hda/cs35l41_hda.h b/sound/pci/hda/cs35l41_hda.h index bdb35f3be68a..b93bf762976e 100644 --- a/sound/pci/hda/cs35l41_hda.h +++ b/sound/pci/hda/cs35l41_hda.h @@ -83,5 +83,6 @@ extern const struct dev_pm_ops cs35l41_hda_pm_ops; int cs35l41_hda_probe(struct device *dev, const char *device_name, int id, int irq, struct regmap *regmap); void cs35l41_hda_remove(struct device *dev); +int cs35l41_get_speaker_id(struct device *dev, int amp_index, int num_amps, int fixed_gpio_id); #endif /*__CS35L41_HDA_H__*/ diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c new file mode 100644 index 000000000000..673f23257a09 --- /dev/null +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -0,0 +1,73 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// CS35L41 ALSA HDA Property driver +// +// Copyright 2023 Cirrus Logic, Inc. +// +// Author: Stefan Binding <sbinding(a)opensource.cirrus.com> + +#include <linux/gpio/consumer.h> +#include <linux/string.h> +#include "cs35l41_hda_property.h" + +/* + * Device CLSA010(0/1) doesn't have _DSD so a gpiod_get by the label reset won't work. + * And devices created by serial-multi-instantiate don't have their device struct + * pointing to the correct fwnode, so acpi_dev must be used here. + * And devm functions expect that the device requesting the resource has the correct + * fwnode. + */ +static int lenovo_legion_no_acpi(struct cs35l41_hda *cs35l41, struct device *physdev, int id, + const char *hid) +{ + struct cs35l41_hw_cfg *hw_cfg = &cs35l41->hw_cfg; + + /* check I2C address to assign the index */ + cs35l41->index = id == 0x40 ? 0 : 1; + cs35l41->channel_index = 0; + cs35l41->reset_gpio = gpiod_get_index(physdev, NULL, 0, GPIOD_OUT_HIGH); + cs35l41->speaker_id = cs35l41_get_speaker_id(physdev, 0, 0, 2); + hw_cfg->spk_pos = cs35l41->index; + hw_cfg->gpio2.func = CS35L41_INTERRUPT; + hw_cfg->gpio2.valid = true; + hw_cfg->valid = true; + + if (strcmp(hid, "CLSA0100") == 0) { + hw_cfg->bst_type = CS35L41_EXT_BOOST_NO_VSPK_SWITCH; + } else if (strcmp(hid, "CLSA0101") == 0) { + hw_cfg->bst_type = CS35L41_EXT_BOOST; + hw_cfg->gpio1.func = CS35l41_VSPK_SWITCH; + hw_cfg->gpio1.valid = true; + } + + return 0; +} + +struct cs35l41_prop_model { + const char *hid; + const char *ssid; + int (*add_prop)(struct cs35l41_hda *cs35l41, struct device *physdev, int id, + const char *hid); +}; + +const struct cs35l41_prop_model cs35l41_prop_model_table[] = { + { "CLSA0100", NULL, lenovo_legion_no_acpi }, + { "CLSA0101", NULL, lenovo_legion_no_acpi }, + {} +}; + +int cs35l41_add_dsd_properties(struct cs35l41_hda *cs35l41, struct device *physdev, int id, + const char *hid) +{ + const struct cs35l41_prop_model *model; + + for (model = cs35l41_prop_model_table; model->hid > 0; model++) { + if (!strcmp(model->hid, hid) && + (!model->ssid || + (cs35l41->acpi_subsystem_id && + !strcmp(model->ssid, cs35l41->acpi_subsystem_id)))) + return model->add_prop(cs35l41, physdev, id, hid); + } + + return -ENOENT; +} diff --git a/sound/pci/hda/cs35l41_hda_property.h b/sound/pci/hda/cs35l41_hda_property.h new file mode 100644 index 000000000000..fd834042e2fd --- /dev/null +++ b/sound/pci/hda/cs35l41_hda_property.h @@ -0,0 +1,18 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * CS35L41 ALSA HDA Property driver + * + * Copyright 2023 Cirrus Logic, Inc. + * + * Author: Stefan Binding <sbinding(a)opensource.cirrus.com> + */ + +#ifndef CS35L41_HDA_PROP_H +#define CS35L41_HDA_PROP_H + +#include <linux/device.h> +#include "cs35l41_hda.h" + +int cs35l41_add_dsd_properties(struct cs35l41_hda *cs35l41, struct device *physdev, int id, + const char *hid); +#endif /* CS35L41_HDA_PROP_H */ -- 2.34.1

3 2

[PATCH v4 0/6] ASoC: mediatek: Add support for MT7986 SoC
by Maso Huang 17 Aug '23

17 Aug '23

Changes in v4: - fix typo of authors and copyright - refine with bitfield helper for readability [2/6] - refine with human readable error message [3/6] - use "mt7986-wm8960-sound" as compatible string [4/6] [5/6] - refine properties based on reviewer's suggestions [5/6] - remove assigned-clocks and assigned-clocks-parents [6/6] - constrain clocks per variants [6/6] Changes in v3: - merge clk api to mt7986-afe-pcm.c, remove [2/7] in v2 - refine based on reviewer's suggestions [1/6] [2/6] - fix the comment format, move in clk api, and simplify based on reviewer's suggestions [3/6] - remove redundant prefix in dt-binding [6/6] Changes in v2: - v1 title: [PATCH 0/7] ASoC: mediatek: Add support for MT79xx SoC - add missing maintainers - rename mt79xx to mt7986 in all files - use clk bulk api in mt7986-afe-clk.c [2/7] - refine mt79xx-afe-pcm.c based on reviewer's suggestions [4/7] - refine dt-binding files based on reviewer's suggestions [6/7] [7/7] - transpose [3/7] and [4/7] in v1 to fix test build errors Maso Huang (6): ASoC: mediatek: mt7986: add common header ASoC: mediatek: mt7986: support etdm in platform driver ASoC: mediatek: mt7986: add platform driver ASoC: mediatek: mt7986: add machine driver with wm8960 ASoC: dt-bindings: mediatek,mt7986-wm8960: add mt7986-wm8960 document ASoC: dt-bindings: mediatek,mt7986-afe: add audio afe document .../bindings/sound/mediatek,mt7986-afe.yaml | 160 +++++ .../sound/mediatek,mt7986-wm8960.yaml | 67 ++ sound/soc/mediatek/Kconfig | 20 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/mt7986/Makefile | 9 + sound/soc/mediatek/mt7986/mt7986-afe-common.h | 49 ++ sound/soc/mediatek/mt7986/mt7986-afe-pcm.c | 622 ++++++++++++++++++ sound/soc/mediatek/mt7986/mt7986-dai-etdm.c | 411 ++++++++++++ sound/soc/mediatek/mt7986/mt7986-reg.h | 196 ++++++ sound/soc/mediatek/mt7986/mt7986-wm8960.c | 196 ++++++ 10 files changed, 1731 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mediatek,mt7986-afe.yaml create mode 100644 Documentation/devicetree/bindings/sound/mediatek,mt7986-wm8960.yaml create mode 100644 sound/soc/mediatek/mt7986/Makefile create mode 100644 sound/soc/mediatek/mt7986/mt7986-afe-common.h create mode 100644 sound/soc/mediatek/mt7986/mt7986-afe-pcm.c create mode 100644 sound/soc/mediatek/mt7986/mt7986-dai-etdm.c create mode 100644 sound/soc/mediatek/mt7986/mt7986-reg.h create mode 100644 sound/soc/mediatek/mt7986/mt7986-wm8960.c -- 2.18.0

4 9

[PATCH v2] ASoC: tas2783: Add source files for tas2783 soundwire driver
by Baojun Xu 17 Aug '23

17 Aug '23

Add source file and header file for tas2783 driver. Update Kconfig and Makefile for tas2783 driver. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- Change in v2: - change all registers access to regmap. - change header file include order. - remove tas2783_sdw_read(). - remove regmap_bus and relative function. - change custom access to regmap. - return if get calibration data from EFI failed. - remove mutex protect for null function. - remove tasdevice_startup() as no actions needed. - remove sysclk function as no action needed. - remove tasdevice_mute(). - remove crc8 look up table generate. - remove first_hw_init flag. - change some define name to TAS2783_XXX. - remove SMS_HTONL(), use le_to_cpup(). - remove enum channel define. - change struct and element name for FW. - remove global address elements. - remove irq and gpio in structure. - remove tasdevice structure. --- sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2783-sdw.c | 872 +++++++++++++++++++++++++++++++++ sound/soc/codecs/tas2783.h | 113 +++++ 4 files changed, 999 insertions(+) create mode 100644 sound/soc/codecs/tas2783-sdw.c create mode 100644 sound/soc/codecs/tas2783.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c2de4ee72183..a4d334faca5e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -240,6 +240,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_TAS2781_COMLIB imply SND_SOC_TAS2781_FMWLIB imply SND_SOC_TAS2781_I2C + imply SND_SOC_TAS2783 imply SND_SOC_TAS5086 imply SND_SOC_TAS571X imply SND_SOC_TAS5720 @@ -1758,6 +1759,17 @@ config SND_SOC_TAS2781_I2C algo coefficient setting, for one, two or even multiple TAS2781 chips. +config SND_SOC_TAS2783 + tristate "Texas Instruments TAS2783 speaker amplifier (sdw)" + depends on SOUNDWIRE + select REGMAP + select CRC32_SARWATE + help + Enable support for Texas Instruments TAS2783 Smart Amplifier + Digital input mono Class-D and DSP-inside audio power amplifiers. + Note the TAS2783 driver implements a flexible and configurable + algo coff setting, for one, two, even multiple TAS2783 chips. + config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index b48a9a323b84..3298dfa3b9d5 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -276,6 +276,7 @@ snd-soc-tas2770-objs := tas2770.o snd-soc-tas2781-comlib-objs := tas2781-comlib.o snd-soc-tas2781-fmwlib-objs := tas2781-fmwlib.o snd-soc-tas2781-i2c-objs := tas2781-i2c.o +snd-soc-tas2783-objs := tas2783-sdw.o snd-soc-tfa9879-objs := tfa9879.o snd-soc-tfa989x-objs := tfa989x.o snd-soc-tlv320adc3xxx-objs := tlv320adc3xxx.o @@ -648,6 +649,7 @@ obj-$(CONFIG_SND_SOC_TAS2780) += snd-soc-tas2780.o obj-$(CONFIG_SND_SOC_TAS2781_COMLIB) += snd-soc-tas2781-comlib.o obj-$(CONFIG_SND_SOC_TAS2781_FMWLIB) += snd-soc-tas2781-fmwlib.o obj-$(CONFIG_SND_SOC_TAS2781_I2C) += snd-soc-tas2781-i2c.o +obj-$(CONFIG_SND_SOC_TAS2783) += snd-soc-tas2783.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o obj-$(CONFIG_SND_SOC_TAS5720) += snd-soc-tas5720.o diff --git a/sound/soc/codecs/tas2783-sdw.c b/sound/soc/codecs/tas2783-sdw.c new file mode 100644 index 000000000000..f916ac9c5520 --- /dev/null +++ b/sound/soc/codecs/tas2783-sdw.c @@ -0,0 +1,871 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// ALSA SoC Texas Instruments TAS2783 Audio Smart Amplifier. +// +// Copyright (C) 2023 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2783 driver implements a flexible and configurable +// algo coefficient setting for single TAS2783 chips. +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// Author: Kevin Lu <kevin-lu(a)ti.com> +// + +#include <linux/crc32.h> +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/efi.h> +#include <linux/err.h> +#include <linux/firmware.h> +#include <linux/init.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/pm.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_registers.h> +#include <linux/soundwire/sdw_type.h> +#include <sound/pcm_params.h> +#include <sound/sdw.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/tas2781-tlv.h> + +#include "tas2783.h" + +static const unsigned int tas2783_calibration_reg[] = { + TAS2783_CALIBRATION_RE, + TAS2783_CALIBRATION_RE_LOW, + TAS2783_CALIBRATION_INV_RE, + TAS2783_CALIBRATION_POW, + TAS2783_CALIBRATION_TLIMIT, + 0, +}; + +static const struct reg_default tas2783_reg_defaults[] = { + // Default values for ROM mode. Activated. + { 0x8002, 0x1a}, // Put AMP in power down mode, B0P0R02. + { 0x800e, 0x44}, // VSense from slot 4, B0P0R0E. + { 0x800f, 0x40}, // ISense from slot 0, B0P0R0E. + { 0x8097, 0xc8}, // SARBurstMask = 0, B0P1R17. + { 0x80b5, 0x74}, + { 0x8099, 0x20}, + { 0xfe8d, 0x0d}, + { 0xfebe, 0x4a}, + { 0x8230, 0x00}, + { 0x8231, 0x00}, + { 0x8232, 0x00}, + { 0x8233, 0x01}, // YMEM[70] = 1. B0P4R30. + { 0x8418, 0x00}, // 0 dB in B0P8R18. + { 0x8419, 0x00}, + { 0x841a, 0x00}, + { 0x841b, 0x00}, + { 0x8428, 0x40}, // Unmute channel, B0P8R28. + { 0x8429, 0x00}, + { 0x842a, 0x00}, + { 0x842b, 0x00}, + { 0x8548, 0x00}, // 0dB in B0PAR48. + { 0x8549, 0x00}, + { 0x854a, 0x00}, + { 0x854b, 0x00}, + { 0x8558, 0x40}, // unmute channel, B0PAR58. + { 0x8559, 0x00}, + { 0x855a, 0x00}, + { 0x855b, 0x00}, + { 0x800a, 0x3a}, // Enable both channel in B0P0RA. + { 0x805c, 0xd9}, // Enable clock detected in B0P0R5C. + { 0x8002, 0x00}, // Put AMP in active mode, B0P0R02. + /* Below register was used to select function, entity, CS, CN. */ + { SDW_SDCA_CTL(1, 1, 2, 0), 0}, // 0x40400088. + { SDW_SDCA_CTL(1, 1, 1, 0), 0}, // 0x40400090. + { SDW_SDCA_CTL(1, 2, 1, 0), 0}, // 0x40400108. +}; + +static bool tas2783_readable_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + case 0x8000 ... 0xc000: // Page 0 ~ 127. + case 0xfe80 ... 0xfeff: // Page 253. + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, TAS2783_SDCA_ENT_UDMPU21, + TAS2783_SDCA_CTL_UDMPU_CLUSTER, 0): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, TAS2783_SDCA_ENT_FU21, + TAS2783_SDCA_CTL_FU_MUTE, TAS2783_DEVICE_CHANNEL_LEFT): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, TAS2783_SDCA_ENT_FU21, + TAS2783_SDCA_CTL_FU_MUTE, TAS2783_DEVICE_CHANNEL_RIGHT): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, TAS2783_SDCA_ENT_PDE23, + TAS2783_SDCA_CTL_REQ_POWER_STATE, 0): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, TAS2783_SDCA_ENT_PDE22, + TAS2783_SDCA_CTL_REQ_POWER_STATE, 0): + return true; + default: + return false; + } +} + +static bool tas2783_volatile_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + case 0x8001: + // Only reset register was volatiled. + return true; + default: + return false; + } +} + +static const struct regmap_config tasdevice_regmap = { + .reg_bits = 32, + .val_bits = 8, + .readable_reg = tas2783_readable_register, + .volatile_reg = tas2783_volatile_register, + .max_register = 0x41008000 + TASDEVICE_REG(0xa1, 0x60, 0x7f), + .reg_defaults = tas2783_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas2783_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .use_single_read = true, + .use_single_write = true, +}; + +static int tasdevice_clamp(int val, int max, unsigned int invert) +{ + /* Keep in valid area, out of range value don't care. */ + if (val > max) + val = max; + if (invert) + val = max - val; + if (val < 0) + val = 0; + return val; +} + +static int tas2783_digital_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct regmap *map = tas_dev->regmap; + int val = 0, ret; + + if (!map) { + ret = -EINVAL; + dev_err(tas_dev->dev, "%s, regmap doesn't exist.\n", + __func__); + goto out; + } + /* Read the primary device as the whole */ + ret = regmap_read(map, mc->reg, &val); + dev_dbg(tas_dev->dev, "%s, get digital vol %d from %x with %d\n", + __func__, val, mc->reg, ret); + if (ret) { + dev_err(tas_dev->dev, "%s, get digital vol error %x.\n", + __func__, ret); + goto out; + } + ucontrol->value.integer.value[0] = + tasdevice_clamp(val, mc->max, mc->invert); + +out: + return ret; +} + +static int tas2783_digital_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct regmap *map = tas_dev->regmap; + int val, ret; + + if (!map) { + dev_err(tas_dev->dev, "%s, regmap doesn't exist.\n", + __func__); + return -EINVAL; + } + val = tasdevice_clamp(ucontrol->value.integer.value[0], + mc->max, mc->invert); + + ret = regmap_write(map, mc->reg, val); + if (ret != 0) { + dev_err(tas_dev->dev, "Write @%#x..%#x:%d\n", + mc->reg, val, ret); + return ret; + } + dev_dbg(tas_dev->dev, "%s, Put digital vol %d into %x.\n", + __func__, val, mc->reg); + + return 1; +} + +static int tas2783_amp_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct regmap *map = tas_dev->regmap; + unsigned char mask; + int ret, val; + + if (!map) { + dev_err(tas_dev->dev, "%s, regmap doesn't exist.\n", + __func__); + return -EINVAL; + } + /* Read the primary device */ + ret = regmap_read(map, mc->reg, &val); + dev_dbg(tas_dev->dev, "%s, get AMP vol %d from %x with %d\n", + __func__, val, mc->reg, ret); + + mask = (1 << fls(mc->max)) - 1; + mask <<= mc->shift; + val = (val & mask) >> mc->shift; + ucontrol->value.integer.value[0] = tasdevice_clamp(val, mc->max, + mc->invert); + + return ret; +} + +static int tas2783_amp_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct regmap *map = tas_dev->regmap; + unsigned char mask; + int val, ret; + + if (!map) { + dev_err(tas_dev->dev, "%s, regmap doesn't exist.\n", + __func__); + return -EINVAL; + } + mask = (1 << fls(mc->max)) - 1; + mask <<= mc->shift; + val = tasdevice_clamp(ucontrol->value.integer.value[0], mc->max, + mc->invert); + ret = regmap_update_bits(map, mc->reg, mask, val << mc->shift); + if (ret != 0) { + dev_err(tas_dev->dev, "Write @%#x..%#x:%d\n", + mc->reg, val, ret); + return ret; + } + + dev_dbg(tas_dev->dev, "wr amp %x into 0x%08x by %x shift %d.\n", + val, mc->reg, mask, mc->shift); + + return ret; +} + +static const struct snd_kcontrol_new tas2783_snd_controls[] = { + SOC_SINGLE_RANGE_EXT_TLV("Amp Gain Volume", TAS2783_AMP_LEVEL, + 1, 0, 20, 0, tas2783_amp_getvol, + tas2783_amp_putvol, amp_vol_tlv), + SOC_SINGLE_RANGE_EXT_TLV("Digital Volume", TAS2783_DVC_LVL, + 0, 0, 200, 1, tas2783_digital_getvol, + tas2783_digital_putvol, dvc_tlv), +}; + +static void tas2783_apply_calib( + struct tasdevice_priv *tas_priv, unsigned int *cali_data) +{ + struct regmap *map = tas_priv->regmap; + int i = 0, ret; + u8 *reg_start; + + if (!map) { + dev_err(tas_priv->dev, "%s, regmap doesn't exist.\n", + __func__); + return; + } + if (!tas_priv->sdw_peripheral) { + dev_err(tas_priv->dev, "%s, slaver doesn't exist.\n", + __func__); + return; + } + if ((tas_priv->sdw_peripheral->id.unique_id < + TAS2783_DEVICE_ID_START) || + (tas_priv->sdw_peripheral->id.unique_id > + TAS2783_DEVICE_ID_END)) { + dev_err(tas_priv->dev, "%s, wrong unique id: %x.\n", + __func__, tas_priv->sdw_peripheral->id.unique_id); + return; + } + reg_start = + (u8 *)(cali_data+(tas_priv->sdw_peripheral->id.unique_id - + TAS2783_DEVICE_ID_START)*TAS2783_CALIBRATION_SIZE); + while (tas2783_calibration_reg[i] != 0) { + ret = regmap_bulk_write(map, tas2783_calibration_reg[i], + reg_start + i, 4); + if (ret != 0) { + dev_err(tas_priv->dev, "Cali failed %x:%d\n", + tas2783_calibration_reg[i], ret); + break; + } + i++; + } +} + +static int tas2783_calibration(struct tasdevice_priv *tas_priv) +{ + efi_guid_t efi_guid = EFI_GUID(0x1f52d2a1, 0xbb3a, 0x457d, 0xbc, + 0x09, 0x43, 0xa3, 0xf4, 0x31, 0x0a, 0x92); + static efi_char16_t efi_name[] = TAS2783_CALIDATA_NAME; + struct tm *tm = &tas_priv->tm; + unsigned int attr, crc; + unsigned int *tmp_val; + efi_status_t status; + + tas_priv->cali_data.total_sz = TAS2783_MAX_CALIDATA_SIZE; + /* Get real size of UEFI variable */ + status = efi.get_variable(efi_name, &efi_guid, &attr, + &tas_priv->cali_data.total_sz, tas_priv->cali_data.data); + dev_dbg(tas_priv->dev, "cali get %lx bytes with result : %ld\n", + tas_priv->cali_data.total_sz, status); + if (status == EFI_BUFFER_TOO_SMALL) { + status = efi.get_variable(efi_name, &efi_guid, &attr, + &tas_priv->cali_data.total_sz, + tas_priv->cali_data.data); + dev_dbg(tas_priv->dev, "cali get %lx bytes result:%ld\n", + tas_priv->cali_data.total_sz, status); + } + /* Failed got calibration data from EFI. */ + if (status != 0) { + dev_dbg(tas_priv->dev, "cali get %lx error with:%ld\n", + tas_priv->cali_data.total_sz, status); + return 0; + } + /* Print all content of calibration data for debug. */ + for (int i = 0; i < tas_priv->cali_data.total_sz; i += 4) { + dev_dbg(tas_priv->dev, "cali get %02x %02x %02x %02x", + tas_priv->cali_data.data[i], + tas_priv->cali_data.data[i+1], + tas_priv->cali_data.data[i+2], + tas_priv->cali_data.data[i+3]); + } + + tmp_val = (unsigned int *)tas_priv->cali_data.data; + + crc = crc32(~0, tas_priv->cali_data.data, 84) ^ ~0; + dev_dbg(tas_priv->dev, "cali crc 0x%08x PK tmp_val 0x%08x\n", + crc, tmp_val[21]); + + if (crc == tmp_val[21]) { + time64_to_tm(tmp_val[20], 0, tm); + dev_dbg(tas_priv->dev, "%4ld-%2d-%2d, %2d:%2d:%2d\n", + tm->tm_year, tm->tm_mon, tm->tm_mday, + tm->tm_hour, tm->tm_min, tm->tm_sec); + tas2783_apply_calib(tas_priv, tmp_val); + } else { + dev_dbg(tas_priv->dev, "CRC error!\n"); + tas_priv->cali_data.total_sz = 0; + } + + return 0; +} + +static void tasdevice_rca_ready(const struct firmware *fmw, void *context) +{ + struct tasdevice_priv *tas_dev = + (struct tasdevice_priv *) context; + struct regmap *map = tas_dev->regmap; + struct tas2783_firmware_node *p; + int offset = 0, num_nodes = 0, img_sz, ret; + unsigned char *buf; + + mutex_lock(&tas_dev->codec_lock); + + if (!fmw || !fmw->data) { + dev_err(tas_dev->dev, + "Failed to read %s, no side - effect on driver running\n", + tas_dev->rca_binaryname); + ret = -EINVAL; + goto out; + } + if (!map) { + dev_err(tas_dev->dev, "Failed to load regmap.\n"); + ret = -EINVAL; + goto out; + } + buf = (unsigned char *)fmw->data; + + img_sz = le32_to_cpup((__le32 *)&buf[offset]); + dev_dbg(tas_dev->dev, "Got %x:%lx.\n", img_sz, fmw->size); + offset += sizeof(img_sz); + if (img_sz != fmw->size) { + dev_err(tas_dev->dev, + "Size not match, %d %u", (int)fmw->size, img_sz); + ret = -EINVAL; + goto out; + } + + while ((offset < img_sz) && (num_nodes < TAS2783_MAX_NODES)) { + /* Store firmware into context of driver. */ + p = (struct tas2783_firmware_node *)(buf + offset); + tas_dev->firmware_node[num_nodes].vendor_id = + p->vendor_id; + tas_dev->firmware_node[num_nodes].file_id = p->file_id; + tas_dev->firmware_node[num_nodes].version_id = + p->version_id; + tas_dev->firmware_node[num_nodes].length = p->length; + tas_dev->firmware_node[num_nodes].download_addr = + p->download_addr; + tas_dev->firmware_node[num_nodes].start_addr = + ((char *)p) + sizeof(unsigned int)*5; + + ret = regmap_bulk_write(map, p->download_addr, + p->start_addr, p->length); + dev_dbg(tas_dev->dev, "Wr %d :%x:%x:%x:%x:%x:%x %d.\n", + num_nodes, p->vendor_id, p->file_id, + p->version_id, p->length, p->download_addr, + p->start_addr[0], ret); + + offset += sizeof(unsigned int)*5 + p->length; + num_nodes++; + } + + tas2783_calibration(tas_dev); + +out: + mutex_unlock(&tas_dev->codec_lock); + if (fmw) + release_firmware(fmw); +} + +static const struct snd_soc_dapm_widget tasdevice_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("ASI", "ASI Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("ASI OUT", "ASI Capture", 0, SND_SOC_NOPM, + 0, 0), + SND_SOC_DAPM_OUTPUT("OUT"), + SND_SOC_DAPM_INPUT("DMIC") +}; + +static const struct snd_soc_dapm_route tasdevice_audio_map[] = { + {"OUT", NULL, "ASI"}, + {"ASI OUT", NULL, "DMIC"} +}; + +static int tasdevice_set_sdw_stream( + struct snd_soc_dai *dai, void *sdw_stream, int direction) +{ + struct sdw_stream_data *stream; + + if (!sdw_stream) + return 0; + + stream = kzalloc(sizeof(*stream), GFP_KERNEL); + if (!stream) + return -ENOMEM; + + stream->sdw_stream = sdw_stream; + + /* Use tx_mask or rx_mask to set dma_data */ + snd_soc_dai_dma_data_set(dai, direction, stream); + + return 0; +} + +static void tasdevice_sdw_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct sdw_stream_data *stream; + + stream = snd_soc_dai_get_dma_data(dai, substream); + snd_soc_dai_set_dma_data(dai, substream, NULL); + kfree(stream); +} + +static int tasdevice_sdw_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tasdevice = + snd_soc_component_get_drvdata(component); + struct sdw_stream_config stream_config = {0}; + struct sdw_port_config port_config = {0}; + struct sdw_stream_data *stream; + int retval; + + dev_dbg(dai->dev, "%s %s", __func__, dai->name); + stream = snd_soc_dai_get_dma_data(dai, substream); + + if (!stream) + return -EINVAL; + + if (!tasdevice->sdw_peripheral) + return -EINVAL; + + /* SoundWire specific configuration */ + snd_sdw_params_to_config(substream, params, + &stream_config, &port_config); + + /* port 1 for playback */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + port_config.num = 1; + else + port_config.num = 2; + + retval = sdw_stream_add_slave(tasdevice->sdw_peripheral, + &stream_config, &port_config, 1, stream->sdw_stream); + if (retval) { + dev_err(dai->dev, "Unable to configure port\n"); + return retval; + } + + return 0; +} + +static int tasdevice_sdw_pcm_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tasdevice = + snd_soc_component_get_drvdata(component); + struct sdw_stream_data *stream = + snd_soc_dai_get_dma_data(dai, substream); + + if (!tasdevice->sdw_peripheral) + return -EINVAL; + + sdw_stream_remove_slave(tasdevice->sdw_peripheral, stream->sdw_stream); + return 0; +} + +static const struct snd_soc_dai_ops tasdevice_dai_ops = { + .hw_params = tasdevice_sdw_hw_params, + .hw_free = tasdevice_sdw_pcm_hw_free, + .set_stream = tasdevice_set_sdw_stream, + .shutdown = tasdevice_sdw_shutdown, +}; + +static struct snd_soc_dai_driver tasdevice_dai_driver[] = { + { + .name = "tas2783-codec", + .id = 0, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 4, + .rates = TAS2783_DEVICE_RATES, + .formats = TAS2783_DEVICE_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 4, + .rates = TAS2783_DEVICE_RATES, + .formats = TAS2783_DEVICE_FORMATS, + }, + .ops = &tasdevice_dai_ops, + .symmetric_rate = 1, + }, +}; + +static void tas2783_reset(struct tasdevice_priv *tas_dev) +{ + struct regmap *map = tas_dev->regmap; + unsigned char value_sdw; + int ret; + + if (!map) { + dev_err(tas_dev->dev, "Failed to load regmap.\n"); + return; + } + value_sdw = TAS2873_REG_SWRESET_RESET; + ret = regmap_write(map, TAS2873_REG_SWRESET, value_sdw); + dev_dbg(tas_dev->dev, "%s TAS2783 was reseted %d.\n", + __func__, ret); + usleep_range(1000, 1050); +} + +static int tasdevice_codec_probe(struct snd_soc_component *codec) +{ + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + int ret; + + dev_dbg(tas_dev->dev, "%s called for TAS2783 start.\n", + __func__); + /* Codec Lock Hold */ + mutex_lock(&tas_dev->codec_lock); + + tas2783_reset(tas_dev); + + tas_dev->codec = codec; + + scnprintf(tas_dev->rca_binaryname, 64, "MY_SWFT_x%01x.bin", + tas_dev->sdw_peripheral->id.unique_id); + + ret = request_firmware_nowait(THIS_MODULE, FW_ACTION_UEVENT, + tas_dev->rca_binaryname, tas_dev->dev, GFP_KERNEL, + tas_dev, tasdevice_rca_ready); + dev_dbg(tas_dev->dev, + "%s: request_firmware %x open status: 0x%08x\n", + __func__, tas_dev->sdw_peripheral->id.unique_id, ret); + + /* Codec Lock Release*/ + mutex_unlock(&tas_dev->codec_lock); + + dev_dbg(tas_dev->dev, "%s was called end.\n", __func__); + return ret; +} + +static const struct snd_soc_component_driver + soc_codec_driver_tasdevice = { + .probe = tasdevice_codec_probe, + .controls = tas2783_snd_controls, + .num_controls = ARRAY_SIZE(tas2783_snd_controls), + .dapm_widgets = tasdevice_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tasdevice_dapm_widgets), + .dapm_routes = tasdevice_audio_map, + .num_dapm_routes = ARRAY_SIZE(tasdevice_audio_map), + .idle_bias_on = 1, + .endianness = 1, +}; + +static int tasdevice_init(struct tasdevice_priv *tas_dev) +{ + int ret; + + mutex_init(&tas_dev->dev_lock); + + dev_set_drvdata(tas_dev->dev, tas_dev); + + mutex_init(&tas_dev->codec_lock); + ret = devm_snd_soc_register_component(tas_dev->dev, + &soc_codec_driver_tasdevice, + tasdevice_dai_driver, ARRAY_SIZE(tasdevice_dai_driver)); + if (ret) { + dev_err(tas_dev->dev, "%s: codec register error:0x%x\n", + __func__, ret); + } + dev_dbg(tas_dev->dev, "%s was called for TAS2783.\n", __func__); + + return ret; +} + +static int tasdevice_read_prop(struct sdw_slave *slave) +{ + struct sdw_slave_prop *prop = &slave->prop; + struct sdw_dpn_prop *dpn; + unsigned long addr; + int nval, i, j; + u32 bit; + + prop->scp_int1_mask = + SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; + prop->quirks = SDW_SLAVE_QUIRKS_INVALID_INITIAL_PARITY; + + prop->paging_support = true; + + /* first we need to allocate memory for set bits in port lists */ + prop->source_ports = 0x04; /* BITMAP: 00000100 */ + prop->sink_ports = 0x2; /* BITMAP: 00000010 */ + + nval = hweight32(prop->source_ports); + prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->src_dpn_prop), GFP_KERNEL); + if (!prop->src_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->src_dpn_prop; + addr = prop->source_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* do this again for sink now */ + nval = hweight32(prop->sink_ports); + prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->sink_dpn_prop), GFP_KERNEL); + if (!prop->sink_dpn_prop) + return -ENOMEM; + + j = 0; + dpn = prop->sink_dpn_prop; + addr = prop->sink_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[j].num = bit; + dpn[j].type = SDW_DPN_FULL; + dpn[j].simple_ch_prep_sm = true; + dpn[j].ch_prep_timeout = 10; + j++; + } + + /* set the timeout values */ + prop->clk_stop_timeout = 20; + + dev_dbg(&slave->dev, "%s was performed.\n", __func__); + + return 0; +} + +static int tasdevice_io_init(struct device *dev, struct sdw_slave *slave) +{ + struct tasdevice_priv *tasdevice = dev_get_drvdata(dev); + + if (tasdevice->hw_init) + return 0; + + /* PM runtime is only enabled when + * a Slave reports as Attached + * set autosuspend parameters + */ + pm_runtime_set_autosuspend_delay(&slave->dev, 3000); + pm_runtime_use_autosuspend(&slave->dev); + + /* update count of parent 'active' children */ + pm_runtime_set_active(&slave->dev); + + /* make sure the device does not suspend immediately */ + pm_runtime_mark_last_busy(&slave->dev); + + pm_runtime_enable(&slave->dev); + + pm_runtime_get_noresume(&slave->dev); + + /* Mark Slave initialization complete */ + tasdevice->hw_init = true; + + pm_runtime_mark_last_busy(&slave->dev); + pm_runtime_put_autosuspend(&slave->dev); + + dev_dbg(&slave->dev, "%s hw_init complete\n", __func__); + return 0; +} + +static int tasdevice_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct tasdevice_priv *tasdevice = dev_get_drvdata(&slave->dev); + + /* Update the status */ + tasdevice->status = status; + + if (status == SDW_SLAVE_UNATTACHED) + tasdevice->hw_init = false; + + /* Perform initialization only if slave status + * is present and hw_init flag is false + */ + if (tasdevice->hw_init || tasdevice->status != SDW_SLAVE_ATTACHED) + return 0; + + /* perform I/O transfers required for Slave initialization */ + return tasdevice_io_init(&slave->dev, slave); +} + +/* + * slave_ops: callbacks for get_clock_stop_mode, clock_stop and + * port_prep are not defined for now + */ +static const struct sdw_slave_ops tasdevice_sdw_ops = { + .read_prop = tasdevice_read_prop, + .update_status = tasdevice_update_status, +}; + +static void tasdevice_remove(struct tasdevice_priv *tas_dev) +{ + snd_soc_unregister_component(tas_dev->dev); + + mutex_destroy(&tas_dev->dev_lock); + mutex_destroy(&tas_dev->codec_lock); +} + +static int tasdevice_sdw_probe(struct sdw_slave *peripheral, + const struct sdw_device_id *id) +{ + struct device *dev = &peripheral->dev; + struct tasdevice_priv *tas_dev; + int ret; + + dev_dbg(dev, "%s was called.\n", __func__); + + tas_dev = devm_kzalloc(dev, sizeof(*tas_dev), GFP_KERNEL); + if (!tas_dev) { + ret = -ENOMEM; + goto out; + } + tas_dev->dev = &peripheral->dev; + tas_dev->chip_id = id->driver_data; + tas_dev->sdw_peripheral = peripheral; + tas_dev->hw_init = false; + + dev_dbg(dev, "%d chip id %x for TAS2783.\n", + peripheral->id.unique_id, tas_dev->chip_id); + + dev_set_drvdata(dev, tas_dev); + + tas_dev->regmap = devm_regmap_init_sdw(peripheral, + &tasdevice_regmap); + if (IS_ERR(tas_dev->regmap)) { + ret = PTR_ERR(tas_dev->regmap); + dev_err(dev, "Failed devm_regmap_init: %d\n", ret); + goto out; + } + ret = tasdevice_init(tas_dev); + +out: + if (ret < 0 && tas_dev != NULL) + tasdevice_remove(tas_dev); + + return ret; + +} + +static int tasdevice_sdw_remove(struct sdw_slave *peripheral) +{ + struct tasdevice_priv *tas_dev = + dev_get_drvdata(&peripheral->dev); + + if (tas_dev) + tasdevice_remove(tas_dev); + + return 0; +} + +static const struct sdw_device_id tasdevice_sdw_id[] = { + SDW_SLAVE_ENTRY(0x0102, 0x0, 0), + {}, +}; +MODULE_DEVICE_TABLE(sdw, tasdevice_sdw_id); + +static struct sdw_driver tasdevice_sdw_driver = { + .driver = { + .name = "slave-tas2783", + }, + .probe = tasdevice_sdw_probe, + .remove = tasdevice_sdw_remove, + .ops = &tasdevice_sdw_ops, + .id_table = tasdevice_sdw_id, +}; + +module_sdw_driver(tasdevice_sdw_driver); + +MODULE_AUTHOR("Baojun Xu <baojun.xu(a)ti.com>"); +MODULE_DESCRIPTION("ASoC TAS2783 SoundWire Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2783.h b/sound/soc/codecs/tas2783.h new file mode 100644 index 000000000000..09575e741400 --- /dev/null +++ b/sound/soc/codecs/tas2783.h @@ -0,0 +1,113 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * ALSA SoC Texas Instruments TAS2783 Audio Smart Amplifier. + * + * Copyright (C) 2023 Texas Instruments Incorporated + * https://www.ti.com + * + * The TAS2783 driver implements a flexible and configurable + * algo coff setting for single TAS2783 chips. + * + * Author: Baojun Xu <baojun.xu(a)ti.com> + */ + +#ifndef __TAS2783_H__ +#define __TAS2783_H__ + +#define TAS2783_DEVICE_RATES (SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_88200) + +#define TAS2783_DEVICE_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/* BOOK, PAGE Control Register */ +#define TASDEVICE_REG(book, page, reg) ((book * 256 * 256) + 0x8000 +\ + (page * 128) + reg) + +/*Software Reset */ +#define TAS2873_REG_SWRESET TASDEVICE_REG(0x0, 0X0, 0x01) +#define TAS2873_REG_SWRESET_RESET BIT(0) + +/* Volume control */ +#define TAS2783_DVC_LVL TASDEVICE_REG(0x0, 0x00, 0x1A) +#define TAS2783_AMP_LEVEL TASDEVICE_REG(0x0, 0x00, 0x03) +#define TAS2783_AMP_LEVEL_MASK GENMASK(5, 1) + +/* Calibration data */ +#define TAS2783_CALIBRATION_RE TASDEVICE_REG(0x0, 0x17, 0x74) +#define TAS2783_CALIBRATION_RE_LOW TASDEVICE_REG(0x0, 0x18, 0x14) +#define TAS2783_CALIBRATION_INV_RE TASDEVICE_REG(0x0, 0x18, 0x0c) +#define TAS2783_CALIBRATION_POW TASDEVICE_REG(0x0, 0x0d, 0x3c) +#define TAS2783_CALIBRATION_TLIMIT TASDEVICE_REG(0x0, 0x18, 0x7c) + +#define TAS2783_CALIBRATION_SIZE 5 // int data number. + +#define TAS2783_DEVICE_ID_START 0x08 // Unique id start +#define TAS2783_DEVICE_ID_END 0x0F // Unique id end + +/* TAS2783 SDCA Control - function number */ +#define FUNC_NUM_SMART_AMP 0x01 + +/* TAS2783 SDCA entity */ +#define TAS2783_SDCA_ENT_PDE23 0x0C +#define TAS2783_SDCA_ENT_PDE22 0x0B +#define TAS2783_SDCA_ENT_FU21 0x01 +#define TAS2783_SDCA_ENT_UDMPU21 0x10 + +/* TAS2783 SDCA control */ +#define TAS2783_SDCA_CTL_REQ_POWER_STATE 0x01 +#define TAS2783_SDCA_CTL_FU_MUTE 0x01 +#define TAS2783_SDCA_CTL_UDMPU_CLUSTER 0x10 + +#define TAS2783_DEVICE_CHANNEL_LEFT 1 +#define TAS2783_DEVICE_CHANNEL_RIGHT 2 + +#define TAS2783_MAX_NODES 100 +#define TAS2783_MAX_CALIDATA_SIZE 252 +#define TAS2783_CALIDATA_NAME L"CALI_DATA" + +struct tas2783_firmware_node { + unsigned int vendor_id; + unsigned int file_id; + unsigned int version_id; + unsigned int length; + unsigned int download_addr; + unsigned char *start_addr; +}; + +struct calibration_data { + unsigned long total_sz; + unsigned char data[TAS2783_MAX_CALIDATA_SIZE]; +}; + +struct tasdevice_priv { + struct tas2783_firmware_node firmware_node[TAS2783_MAX_NODES]; + struct snd_soc_component *component; + struct calibration_data cali_data; + struct sdw_slave *sdw_peripheral; + struct sdw_bus_params params; + struct mutex codec_lock; + struct regmap *regmap; + struct mutex dev_lock; + struct device *dev; + struct tm tm; + + enum sdw_slave_status status; + + bool hw_init; + + unsigned int chip_id; + unsigned char rca_binaryname[64]; + unsigned char dev_name[32]; + + void *codec; +}; + +struct sdw_stream_data { + struct sdw_stream_runtime *sdw_stream; +}; + +#endif /*__TAS2783_H__ */ -- 2.34.1

3 3

[PATCH v3 1/2] ALSA: hda/tas2781: Add tas2781 HDA driver
by Shenghao Ding 17 Aug '23

17 Aug '23

Create tas2781 side codec HDA driver for Lenovo Laptops. The quantity of the speakers has been define in ACPI. All of the tas2781s in the laptop will be aggregated as one audio speaker. The code supports realtek codec as the primary codec. Code offers several controls for digtial/analog gain setting during playback, and other for eq params setting in case of different audio profiles, such as music, voice, movie, etc. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- Changes in v3: - oprimize the indentation - fix the ambiguous error message - The check of the given value to kcontrol's put func is in the sound/tas2781-comlib.c - In tasdevice_info_programs, corrected max value - Seperate snd_kcontrol_new tas2781_dsp_controls into two standalone kcontrol. - All the controls set as const static - Add static for page_array & rgno_array - Remove uncessary blank lines - Add descriptions for tas2781_save_calibration - remove global addr handling in the code - checking subid in switch statement in function tas2781_hda_bind - add force firmware load Kcontrol - rename the kcontrol name to be more undertandable - remove Superfluous cast in tasdevice_fw_ready - correct weird line break in function tas2781_acpi_get_i2c_resource - correct Referencing adev after acpi_dev_put() in tas2781_hda_read_acpi - As to checking the given value in tasdevice_set_profile_id, it seems done by the tasdevice_info_profile - replace strcpy with strscpy in tas2781_hda_read_acpi - rewrite the subid judgement - Add tiwai(a)suse.de into Cc list - remove the cast in tas2781_acpi_get_i2c_resource - remove else in tas2781_acpi_get_i2c_resource - fix the return value in tasdevice_set_profile_id - remove unneeded NL in tasdevice_config_get - Unifiy the comment style - remove ret = 0 in tasdevice_fw_ready - remove ret in tas2781_save_calibration - remove unused ret in tas2781_hda_playback - add force firmware load Kcontrol --- sound/pci/hda/Kconfig | 15 + sound/pci/hda/Makefile | 2 + sound/pci/hda/tas2781_hda_i2c.c | 858 ++++++++++++++++++++++++++++++++ 3 files changed, 875 insertions(+) create mode 100644 sound/pci/hda/tas2781_hda_i2c.c diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 886255a03e8b..e66257277492 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -130,6 +130,21 @@ config SND_HDA_SCODEC_CS35L41_SPI comment "Set to Y if you want auto-loading the side codec driver" depends on SND_HDA=y && SND_HDA_SCODEC_CS35L41_SPI=m +config SND_HDA_SCODEC_TAS2781_I2C + tristate "Build TAS2781 HD-audio side codec support for I2C Bus" + depends on I2C + depends on ACPI + depends on SND_SOC + select SND_SOC_TAS2781_COMLIB + select SND_SOC_TAS2781_FMWLIB + select CRC32_SARWATE + help + Say Y or M here to include TAS2781 I2C HD-audio side codec support + in snd-hda-intel driver, such as ALC287. + +comment "Set to Y if you want auto-loading the side codec driver" + depends on SND_HDA=y && SND_HDA_SCODEC_TAS2781_I2C=m + config SND_HDA_CODEC_REALTEK tristate "Build Realtek HD-audio codec support" select SND_HDA_GENERIC diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index 00d306104484..1c76609690d6 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -32,6 +32,7 @@ snd-hda-scodec-cs35l41-objs := cs35l41_hda.o snd-hda-scodec-cs35l41-i2c-objs := cs35l41_hda_i2c.o snd-hda-scodec-cs35l41-spi-objs := cs35l41_hda_spi.o snd-hda-cs-dsp-ctls-objs := hda_cs_dsp_ctl.o +snd-hda-scodec-tas2781-i2c-objs := tas2781_hda_i2c.o # common driver obj-$(CONFIG_SND_HDA) := snd-hda-codec.o @@ -56,6 +57,7 @@ obj-$(CONFIG_SND_HDA_SCODEC_CS35L41) += snd-hda-scodec-cs35l41.o obj-$(CONFIG_SND_HDA_SCODEC_CS35L41_I2C) += snd-hda-scodec-cs35l41-i2c.o obj-$(CONFIG_SND_HDA_SCODEC_CS35L41_SPI) += snd-hda-scodec-cs35l41-spi.o obj-$(CONFIG_SND_HDA_CS_DSP_CONTROLS) += snd-hda-cs-dsp-ctls.o +obj-$(CONFIG_SND_HDA_SCODEC_TAS2781_I2C) += snd-hda-scodec-tas2781-i2c.o # this must be the last entry after codec drivers; # otherwise the codec patches won't be hooked before the PCI probe diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c new file mode 100644 index 000000000000..4ccbe12ad451 --- /dev/null +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -0,0 +1,858 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// TAS2781 HDA I2C driver +// +// Copyright 2023 Texas Instruments, Inc. +// +// Author: Shenghao Ding <shenghao-ding(a)ti.com> + +#include <linux/acpi.h> +#include <linux/crc8.h> +#include <linux/crc32.h> +#include <linux/efi.h> +#include <linux/firmware.h> +#include <linux/i2c.h> +#include <linux/mod_devicetable.h> +#include <linux/module.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <sound/hda_codec.h> +#include <sound/soc.h> +#include <sound/tas2781.h> +#include <sound/tlv.h> +#include <sound/tas2781-tlv.h> + +#include "hda_local.h" +#include "hda_auto_parser.h" +#include "hda_component.h" +#include "hda_jack.h" +#include "hda_generic.h" + +#define TASDEVICE_SPEAKER_CALIBRATION_SIZE 20 + +/* No standard control callbacks for SNDRV_CTL_ELEM_IFACE_CARD + * Define two controls, one is Volume control callbacks, the other is + * flag setting control callbacks. + */ + +/* Volume control callbacks for tas2781 */ +#define ACARD_SINGLE_RANGE_EXT_TLV(xname, xreg, xshift, xmin, xmax, xinvert, \ + xhandler_get, xhandler_put, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_CARD, .name = (xname),\ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw_range, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .rreg = xreg, .shift = xshift, \ + .rshift = xshift, .min = xmin, .max = xmax, \ + .invert = xinvert} } + +/* Flag control callbacks for tas2781 */ +#define ACARD_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_CARD, .name = xname, \ + .info = snd_ctl_boolean_mono_info, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = xdata } + +enum calib_data { + R0_VAL = 0, + INV_R0, + R0LOW, + POWER, + TLIM, + CALIB_MAX +}; + +static int tas2781_get_i2c_res(struct acpi_resource *ares, void *data) +{ + struct tasdevice_priv *tas_priv = data; + struct acpi_resource_i2c_serialbus *sb; + + if (i2c_acpi_get_i2c_resource(ares, &sb)) { + if (tas_priv->ndev < TASDEVICE_MAX_CHANNELS && + sb->slave_address != TAS2781_GLOBAL_ADDR) { + tas_priv->tasdevice[tas_priv->ndev].dev_addr = + (unsigned int)sb->slave_address; + tas_priv->ndev++; + } + } + return 1; +} + +static int tas2781_read_acpi(struct tasdevice_priv *p, const char *hid) +{ + struct acpi_device *adev; + struct device *physdev; + LIST_HEAD(resources); + const char *sub; + int ret; + + adev = acpi_dev_get_first_match_dev(hid, NULL, -1); + if (!adev) { + dev_err(p->dev, + "Failed to find an ACPI device for %s\n", hid); + return -ENODEV; + } + + ret = acpi_dev_get_resources(adev, &resources, tas2781_get_i2c_res, p); + if (ret < 0) + goto err; + + acpi_dev_free_resource_list(&resources); + strscpy(p->dev_name, hid, sizeof(p->dev_name)); + physdev = get_device(acpi_get_first_physical_node(adev)); + acpi_dev_put(adev); + + /* No side-effect to the playback even if subsystem_id is NULL*/ + sub = acpi_get_subsystem_id(ACPI_HANDLE(physdev)); + if (IS_ERR(sub)) + sub = NULL; + + p->acpi_subsystem_id = sub; + + put_device(physdev); + + return 0; + +err: + dev_err(p->dev, "read acpi error, ret: %d\n", ret); + put_device(physdev); + + return ret; +} + +static void tas2781_hda_playback_hook(struct device *dev, int action) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + + dev_dbg(tas_priv->dev, "%s: action = %d\n", __func__, action); + switch (action) { + case HDA_GEN_PCM_ACT_OPEN: + pm_runtime_get_sync(dev); + mutex_lock(&tas_priv->codec_lock); + tasdevice_tuning_switch(tas_priv, 0); + mutex_unlock(&tas_priv->codec_lock); + break; + case HDA_GEN_PCM_ACT_CLOSE: + mutex_lock(&tas_priv->codec_lock); + tasdevice_tuning_switch(tas_priv, 1); + mutex_unlock(&tas_priv->codec_lock); + + pm_runtime_mark_last_busy(dev); + pm_runtime_put_autosuspend(dev); + break; + default: + dev_dbg(tas_priv->dev, "Playback action not supported: %d\n", + action); + break; + } +} + +static int tasdevice_info_profile(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = tas_priv->rcabin.ncfgs - 1; + + return 0; +} + +static int tasdevice_get_profile_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = tas_priv->rcabin.profile_cfg_id; + + return 0; +} + +static int tasdevice_hda_clamp(int val, int max) +{ + if (val > max) + val = max; + + if (val < 0) + val = 0; + return val; +} + +static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + int nr_profile = ucontrol->value.integer.value[0]; + int max = tas_priv->rcabin.ncfgs - 1; + int val, ret = 0; + + val = tasdevice_hda_clamp(nr_profile, max); + + if (tas_priv->rcabin.profile_cfg_id != nr_profile) { + tas_priv->rcabin.profile_cfg_id = nr_profile; + ret = 1; + } + + return ret; +} + +static int tasdevice_info_programs(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct tasdevice_fw *tas_fw = tas_priv->fmw; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = tas_fw->nr_programs - 1; + + return 0; +} + +static int tasdevice_info_config(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct tasdevice_fw *tas_fw = tas_priv->fmw; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = tas_fw->nr_configurations - 1; + + return 0; +} + +static int tasdevice_program_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = tas_priv->cur_prog; + + return 0; +} + +static int tasdevice_program_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct tasdevice_fw *tas_fw = tas_priv->fmw; + int nr_program = ucontrol->value.integer.value[0]; + int max = tas_fw->nr_programs - 1; + int val, ret = 0; + + val = tasdevice_hda_clamp(nr_program, max); + + if (tas_priv->cur_prog != nr_program) { + tas_priv->cur_prog = nr_program; + ret = 1; + } + + return ret; +} + +static int tasdevice_config_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = tas_priv->cur_conf; + + return 0; +} + +static int tasdevice_config_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct tasdevice_fw *tas_fw = tas_priv->fmw; + int nr_config = ucontrol->value.integer.value[0]; + int max = tas_fw->nr_configurations - 1; + int val, ret = 0; + + val = tasdevice_hda_clamp(nr_config, max); + + if (tas_priv->cur_conf != nr_config) { + tas_priv->cur_conf = nr_config; + ret = 1; + } + + return ret; +} + +/* + * tas2781_digital_getvol - get the volum control + * @kcontrol: control pointer + * @ucontrol: User data + * Customer Kcontrol for tas2781 is primarily for regmap booking, paging + * depends on internal regmap mechanism. + * tas2781 contains book and page two-level register map, especially + * book switching will set the register BXXP00R7F, after switching to the + * correct book, then leverage the mechanism for paging to access the + * register. + */ +static int tas2781_digital_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + return tasdevice_digital_getvol(tas_priv, ucontrol, mc); +} + +static int tas2781_amp_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + return tasdevice_amp_getvol(tas_priv, ucontrol, mc); +} + +static int tas2781_digital_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + /* The check of the given value is in tasdevice_digital_putvol. */ + return tasdevice_digital_putvol(tas_priv, ucontrol, mc); +} + +static int tas2781_amp_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + /* The check of the given value is in tasdevice_amp_putvol. */ + return tasdevice_amp_putvol(tas_priv, ucontrol, mc); +} + +static int tas2781_force_fwload_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = (int)tas_priv->force_fwload_status; + dev_dbg(tas_priv->dev, "%s : Force FWload %s\n", __func__, + tas_priv->force_fwload_status ? "ON" : "OFF"); + + return 0; +} + +static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + bool change, val = (bool)ucontrol->value.integer.value[0]; + + if (tas_priv->force_fwload_status == val) + change = false; + else { + change = true; + tas_priv->force_fwload_status = val; + } + dev_dbg(tas_priv->dev, "%s : Force FWload %s\n", __func__, + tas_priv->force_fwload_status ? "ON" : "OFF"); + + return change; +} + +static const struct snd_kcontrol_new tas2781_snd_controls[] = { + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain", TAS2781_AMP_LEVEL, + 1, 0, 20, 0, tas2781_amp_getvol, + tas2781_amp_putvol, amp_vol_tlv), + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Digital Gain", TAS2781_DVC_LVL, + 0, 0, 200, 1, tas2781_digital_getvol, + tas2781_digital_putvol, dvc_tlv), + ACARD_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0, + tas2781_force_fwload_get, tas2781_force_fwload_put), +}; + +static const struct snd_kcontrol_new tas2781_prof_ctrl = { + .name = "Speaker Profile Id", + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_profile, + .get = tasdevice_get_profile_id, + .put = tasdevice_set_profile_id, +}; + +static const struct snd_kcontrol_new tas2781_dsp_prog_ctrl = { + .name = "Speaker Program Id", + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_programs, + .get = tasdevice_program_get, + .put = tasdevice_program_put, +}; + +static const struct snd_kcontrol_new tas2781_dsp_conf_ctrl = { + .name = "Speaker Config Id", + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_config, + .get = tasdevice_config_get, + .put = tasdevice_config_put, +}; + +static void tas2781_apply_calib(struct tasdevice_priv *tas_priv) +{ + static const unsigned char page_array[CALIB_MAX] = { + 0x17, 0x18, 0x18, 0x0d, 0x18 + }; + static const unsigned char rgno_array[CALIB_MAX] = { + 0x74, 0x0c, 0x14, 0x3c, 0x7c + }; + unsigned char *data; + int i, j, rc; + + for (i = 0; i < tas_priv->ndev; i++) { + data = tas_priv->cali_data.data + + i * TASDEVICE_SPEAKER_CALIBRATION_SIZE; + for (j = 0; j < CALIB_MAX; j++) { + rc = tasdevice_dev_bulk_write(tas_priv, i, + TASDEVICE_REG(0, page_array[j], rgno_array[j]), + &(data[4 * j]), 4); + if (rc < 0) + dev_err(tas_priv->dev, + "chn %d calib %d bulk_wr err = %d\n", + i, j, rc); + } + } +} + +/* Update the calibrate data, including speaker impedance, f0, etc, into algo. + * Calibrate data is done by manufacturer in the factory. These data are used + * by Algo for calucating the speaker temperature, speaker membrance excursion + * and f0 in real time during playback. + */ +static int tas2781_save_calibration(struct tasdevice_priv *tas_priv) +{ + efi_guid_t efi_guid = EFI_GUID(0x02f9af02, 0x7734, 0x4233, 0xb4, 0x3d, + 0x93, 0xfe, 0x5a, 0xa3, 0x5d, 0xb3); + static efi_char16_t efi_name[] = L"CALI_DATA"; + struct tm *tm = &tas_priv->tm; + unsigned int attr, crc; + unsigned int *tmp_val; + efi_status_t status; + + /* Lenovo devices */ + if (tas_priv->catlog_id == LENOVO) + efi_guid = EFI_GUID(0x1f52d2a1, 0xbb3a, 0x457d, 0xbc, 0x09, + 0x43, 0xa3, 0xf4, 0x31, 0x0a, 0x92); + + tas_priv->cali_data.total_sz = 0; + /* Get real size of UEFI variable */ + status = efi.get_variable(efi_name, &efi_guid, &attr, + &tas_priv->cali_data.total_sz, tas_priv->cali_data.data); + if (status == EFI_BUFFER_TOO_SMALL) { + /* Allocate data buffer of data_size bytes */ + tas_priv->cali_data.data = devm_kzalloc(tas_priv->dev, + tas_priv->cali_data.total_sz, GFP_KERNEL); + if (!tas_priv->cali_data.data) + return -ENOMEM; + /* Get variable contents into buffer */ + status = efi.get_variable(efi_name, &efi_guid, &attr, + &tas_priv->cali_data.total_sz, + tas_priv->cali_data.data); + if (status != EFI_SUCCESS) + return -EINVAL; + } + + tmp_val = (unsigned int *)tas_priv->cali_data.data; + + crc = crc32(~0, tas_priv->cali_data.data, 84) ^ ~0; + dev_dbg(tas_priv->dev, "cali crc 0x%08x PK tmp_val 0x%08x\n", + crc, tmp_val[21]); + + if (crc == tmp_val[21]) { + time64_to_tm(tmp_val[20], 0, tm); + dev_dbg(tas_priv->dev, "%4ld-%2d-%2d, %2d:%2d:%2d\n", + tm->tm_year, tm->tm_mon, tm->tm_mday, + tm->tm_hour, tm->tm_min, tm->tm_sec); + tas2781_apply_calib(tas_priv); + } else + tas_priv->cali_data.total_sz = 0; + + return 0; +} + +static void tasdev_fw_ready(const struct firmware *fmw, void *context) +{ + struct tasdevice_priv *tas_priv = context; + struct hda_codec *codec = tas_priv->codec; + int i, ret; + + pm_runtime_get_sync(tas_priv->dev); + mutex_lock(&tas_priv->codec_lock); + + ret = tasdevice_rca_parser(tas_priv, fmw); + if (ret) + goto out; + + ret = snd_ctl_add(codec->card, + snd_ctl_new1(&tas2781_prof_ctrl, tas_priv)); + if (ret) { + dev_err(tas_priv->dev, + "Failed to add KControl %s = %d\n", + tas2781_prof_ctrl.name, ret); + goto out; + } + + for (i = 0; i < ARRAY_SIZE(tas2781_snd_controls); i++) { + ret = snd_ctl_add(codec->card, + snd_ctl_new1(&tas2781_snd_controls[i], tas_priv)); + if (ret) { + dev_err(tas_priv->dev, + "Failed to add KControl %s = %d\n", + tas2781_snd_controls[i].name, ret); + goto out; + } + } + + tasdevice_dsp_remove(tas_priv); + + tas_priv->fw_state = TASDEVICE_DSP_FW_PENDING; + scnprintf(tas_priv->coef_binaryname, 64, "TAS2XXX%04X.bin", + codec->core.subsystem_id & 0xffff); + ret = tasdevice_dsp_parser(tas_priv); + if (ret) { + dev_err(tas_priv->dev, "dspfw load %s error\n", + tas_priv->coef_binaryname); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + goto out; + } + + ret = snd_ctl_add(codec->card, + snd_ctl_new1(&tas2781_dsp_prog_ctrl, tas_priv)); + if (ret) { + dev_err(tas_priv->dev, + "Failed to add KControl %s = %d\n", + tas2781_dsp_prog_ctrl.name, ret); + goto out; + } + + ret = snd_ctl_add(codec->card, + snd_ctl_new1(&tas2781_dsp_conf_ctrl, tas_priv)); + if (ret) { + dev_err(tas_priv->dev, + "Failed to add KControl %s = %d\n", + tas2781_dsp_conf_ctrl.name, ret); + goto out; + } + + tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; + tasdevice_prmg_load(tas_priv, 0); + + /* If calibrated data occurs error, dsp will still works with default + * calibrated data inside algo. + */ + tas2781_save_calibration(tas_priv); + +out: + if (tas_priv->fw_state == TASDEVICE_DSP_FW_FAIL) { + /*If DSP FW fail, kcontrol won't be created */ + tasdevice_config_info_remove(tas_priv); + tasdevice_dsp_remove(tas_priv); + } + mutex_unlock(&tas_priv->codec_lock); + if (fmw) + release_firmware(fmw); + pm_runtime_mark_last_busy(tas_priv->dev); + pm_runtime_put_autosuspend(tas_priv->dev); +} + +static int tas2781_hda_bind(struct device *dev, struct device *master, + void *master_data) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + struct hda_component *comps = master_data; + struct hda_codec *codec; + unsigned int subid; + int ret; + + if (!comps || tas_priv->index < 0 || + tas_priv->index >= HDA_MAX_COMPONENTS) + return -EINVAL; + + comps = &comps[tas_priv->index]; + if (comps->dev) + return -EBUSY; + + codec = comps->codec; + subid = codec->core.subsystem_id >> 16; + + switch (subid) { + case 0x17aa: + tas_priv->catlog_id = LENOVO; + break; + default: + tas_priv->catlog_id = OTHERS; + break; + } + + pm_runtime_get_sync(dev); + + comps->dev = dev; + + strscpy(comps->name, dev_name(dev), sizeof(comps->name)); + + ret = tascodec_init(tas_priv, codec, tasdev_fw_ready); + if (ret) + return ret; + + comps->playback_hook = tas2781_hda_playback_hook; + + pm_runtime_mark_last_busy(dev); + pm_runtime_put_autosuspend(dev); + + return 0; +} + +static void tas2781_hda_unbind(struct device *dev, + struct device *master, void *master_data) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + struct hda_component *comps = master_data; + + if (comps[tas_priv->index].dev == dev) + memset(&comps[tas_priv->index], 0, sizeof(*comps)); + + tasdevice_config_info_remove(tas_priv); + tasdevice_dsp_remove(tas_priv); + + tas_priv->fw_state = TASDEVICE_DSP_FW_PENDING; +} + +static const struct component_ops tas2781_hda_comp_ops = { + .bind = tas2781_hda_bind, + .unbind = tas2781_hda_unbind, +}; + +static void tas2781_hda_remove(struct device *dev) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + + pm_runtime_get_sync(tas_priv->dev); + pm_runtime_disable(tas_priv->dev); + + component_del(tas_priv->dev, &tas2781_hda_comp_ops); + + pm_runtime_put_noidle(tas_priv->dev); + + tasdevice_remove(tas_priv); +} + +static int tas2781_hda_i2c_probe(struct i2c_client *clt) +{ + struct tasdevice_priv *tas_priv; + const char *device_name; + int ret; + + if (strstr(dev_name(&clt->dev), "TIAS2781")) + device_name = "TIAS2781"; + else + return -ENODEV; + + tas_priv = tasdevice_kzalloc(clt); + if (!tas_priv) + return -ENOMEM; + + tas_priv->irq_info.irq = clt->irq; + ret = tas2781_read_acpi(tas_priv, device_name); + if (ret) + return dev_err_probe(tas_priv->dev, ret, + "Platform not supported\n"); + + ret = tasdevice_init(tas_priv); + if (ret) + goto err; + + pm_runtime_set_autosuspend_delay(tas_priv->dev, 3000); + pm_runtime_use_autosuspend(tas_priv->dev); + pm_runtime_mark_last_busy(tas_priv->dev); + pm_runtime_set_active(tas_priv->dev); + pm_runtime_get_noresume(tas_priv->dev); + pm_runtime_enable(tas_priv->dev); + + pm_runtime_put_autosuspend(tas_priv->dev); + + ret = component_add(tas_priv->dev, &tas2781_hda_comp_ops); + if (ret) { + dev_err(tas_priv->dev, "Register component failed: %d\n", ret); + pm_runtime_disable(tas_priv->dev); + goto err; + } + + tas2781_reset(tas_priv); +err: + if (ret) + tas2781_hda_remove(&clt->dev); + return ret; +} + +static void tas2781_hda_i2c_remove(struct i2c_client *clt) +{ + tas2781_hda_remove(&clt->dev); +} + +static int tas2781_runtime_suspend(struct device *dev) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + int i; + + dev_dbg(tas_priv->dev, "Runtime Suspend\n"); + + mutex_lock(&tas_priv->codec_lock); + + if (tas_priv->playback_started) { + tasdevice_tuning_switch(tas_priv, 1); + tas_priv->playback_started = false; + } + + for (i = 0; i < tas_priv->ndev; i++) { + tas_priv->tasdevice[i].cur_book = -1; + tas_priv->tasdevice[i].cur_prog = -1; + tas_priv->tasdevice[i].cur_conf = -1; + } + + regcache_cache_only(tas_priv->regmap, true); + regcache_mark_dirty(tas_priv->regmap); + + mutex_unlock(&tas_priv->codec_lock); + + return 0; +} + +static int tas2781_runtime_resume(struct device *dev) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + unsigned long calib_data_sz = + tas_priv->ndev * TASDEVICE_SPEAKER_CALIBRATION_SIZE; + int ret; + + dev_dbg(tas_priv->dev, "Runtime Resume\n"); + + mutex_lock(&tas_priv->codec_lock); + + regcache_cache_only(tas_priv->regmap, false); + ret = regcache_sync(tas_priv->regmap); + if (ret) { + dev_err(tas_priv->dev, + "Failed to restore register cache: %d\n", ret); + goto out; + } + + tasdevice_prmg_load(tas_priv, tas_priv->cur_prog); + + /* If calibrated data occurs error, dsp will still works with default + * calibrated data inside algo. + */ + if (tas_priv->cali_data.total_sz > calib_data_sz) + tas2781_apply_calib(tas_priv); + +out: + mutex_unlock(&tas_priv->codec_lock); + + return ret; +} + +static int tas2781_system_suspend(struct device *dev) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + int ret; + + dev_dbg(tas_priv->dev, "System Suspend\n"); + + ret = pm_runtime_force_suspend(dev); + if (ret) + return ret; + + /* Shutdown chip before system suspend */ + regcache_cache_only(tas_priv->regmap, false); + tasdevice_tuning_switch(tas_priv, 1); + regcache_cache_only(tas_priv->regmap, true); + regcache_mark_dirty(tas_priv->regmap); + + /* + * Reset GPIO may be shared, so cannot reset here. + * However beyond this point, amps may be powered down. + */ + return 0; +} + +static int tas2781_system_resume(struct device *dev) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + unsigned long calib_data_sz = + tas_priv->ndev * TASDEVICE_SPEAKER_CALIBRATION_SIZE; + int i, ret; + + dev_dbg(tas_priv->dev, "System Resume\n"); + + ret = pm_runtime_force_resume(dev); + if (ret) + return ret; + + mutex_lock(&tas_priv->codec_lock); + + for (i = 0; i < tas_priv->ndev; i++) { + tas_priv->tasdevice[i].cur_book = -1; + tas_priv->tasdevice[i].cur_prog = -1; + tas_priv->tasdevice[i].cur_conf = -1; + } + tas2781_reset(tas_priv); + tasdevice_prmg_load(tas_priv, tas_priv->cur_prog); + + /* If calibrated data occurs error, dsp will still work with default + * calibrated data inside algo. + */ + if (tas_priv->cali_data.total_sz > calib_data_sz) + tas2781_apply_calib(tas_priv); + mutex_unlock(&tas_priv->codec_lock); + + return 0; +} + +static const struct dev_pm_ops tas2781_hda_pm_ops = { + RUNTIME_PM_OPS(tas2781_runtime_suspend, tas2781_runtime_resume, NULL) + SYSTEM_SLEEP_PM_OPS(tas2781_system_suspend, tas2781_system_resume) +}; + +static const struct i2c_device_id tas2781_hda_i2c_id[] = { + { "tas2781-hda", 0 }, + {} +}; + +static const struct acpi_device_id tas2781_acpi_hda_match[] = { + {"TIAS2781", 0 }, + {} +}; +MODULE_DEVICE_TABLE(acpi, tas2781_acpi_hda_match); + +static struct i2c_driver tas2781_hda_i2c_driver = { + .driver = { + .name = "tas2781-hda", + .acpi_match_table = tas2781_acpi_hda_match, + .pm = &tas2781_hda_pm_ops, + }, + .id_table = tas2781_hda_i2c_id, + .probe_new = tas2781_hda_i2c_probe, + .remove = tas2781_hda_i2c_remove, +}; +module_i2c_driver(tas2781_hda_i2c_driver); + +MODULE_DESCRIPTION("TAS2781 HDA Driver"); +MODULE_AUTHOR("Shenghao Ding, TI, <shenghao-ding(a)ti.com>"); +MODULE_LICENSE("GPL"); +MODULE_IMPORT_NS(SND_SOC_TAS2781_FMWLIB); -- 2.34.1

2 2

Fix 'errno undeclared' compile error with older gcc versions
by GitHub pull_request - opened 17 Aug '23

17 Aug '23

alsa-project/alsa-gobject pull request #83 was opened from michele-perrone: It looks like errno.h has to be included explicitly when compiling with older gcc versions. This issue was encountered on the following system: - Ubuntu 20.04 LTS - Linux 5.4.0-156 - gcc 10.5.0 - Meson 0.53.2 Request URL : https://github.com/alsa-project/alsa-gobject/pull/83 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/83.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

Re: [PATCH] Input: pwm-beeper - Support volume setting via sysfs
by Dmitry Torokhov 17 Aug '23

17 Aug '23

On Sat, May 13, 2023 at 11:02:30PM +0200, Marek Vasut wrote: > On 5/13/23 03:51, Marek Vasut wrote: > > On 5/13/23 03:12, Jeff LaBundy wrote: > > > Hi Marek, > > > > Hi, > > > > > On Fri, May 12, 2023 at 08:55:51PM +0200, Marek Vasut wrote: > > > > The PWM beeper volume can be controlled by adjusting the PWM duty cycle, > > > > expose volume setting via sysfs, so users can make the beeper quieter. > > > > This patch adds sysfs attribute 'volume' in range 0..50000, i.e. from 0 > > > > to 50% in 1/1000th of percent steps, this resolution should be > > > > sufficient. > > > > > > > > The reason for 50000 cap on volume or PWM duty cycle is because > > > > duty cycle > > > > above 50% again reduces the loudness, the PWM wave form is inverted wave > > > > form of the one for duty cycle below 50% and the beeper gets quieter the > > > > closer the setting is to 100% . Hence, 50% cap where the wave > > > > form yields > > > > the loudest result. > > > > > > > > Signed-off-by: Marek Vasut <marex(a)denx.de> > > > > --- > > > > An alternative option would be to extend the userspace input > > > > ABI, e.g. by > > > > using SND_TONE top 16bits to encode the duty cycle in 0..50000 > > > > range, and > > > > bottom 16bit to encode the existing frequency in Hz . Since frequency in > > > > Hz is likely to be below some 25 kHz for audible bell, this fits > > > > in 16bits > > > > just fine. Thoughts ? > > > > --- > > > > > > Thanks for the patch; this seems like a useful feature. > > > > > > My first thought is that 50000 seems like an oddly specific limit to > > > impose > > > upon user space. Ideally, user space need not even care that the > > > beeper is > > > implemented via PWM and why 50000 is significant. > > > > > > Instead, what about accepting 0..255 as the LED subsystem does for > > > brightness, > > > then map these values to 0..50000 internally? In fact, the leds-pwm > > > driver > > > does something similar. > > > > The pwm_set_relative_duty_cycle() function can map whatever range to > > whatever other range of the PWM already, so that's not an issues here. > > It seems to me the 0..127 or 0..255 range is a bit too limiting . I > > think even for the LEDs the reason for that limit is legacy design, but > > here I might be wrong. > > > > > I'm also curious as to whether this function should be a rogue sysfs > > > control > > > limited to this driver, or a generic operation in input. For > > > example, input > > > already allows user space to specify the magnitude of an FF effect; > > > perhaps > > > something similar is warranted here? > > > > See the "An alternative ..." part above, I was wondering about this too, > > whether this can be added into the input ABI, but I am somewhat > > reluctant to fiddle with the ABI. > > Thinking about this further, we could try and add some > > EV_SND SND_TONE_WITH_VOLUME > > to avoid overloading EV_SND SND_TONE , and at the same time allow the user > to set both frequency and volume for the tone without any race condition > between the two. > > The EV_SND SND_TONE_WITH_VOLUME would still take one 32bit parameter, except > this time the parameter 16 LSbits would be the frequency and 16 MSbits would > be the volume. > > But again, here I would like input from the maintainers. Beeper was supposed to be an extremely simple device with minimal controls. I wonder if there is need for volume controls, etc, etc are we not better moving it over to the sound subsystem. We already have: sound/drivers/pcsp/pcsp.c and sound/pci/hda/hda_beep.c there, can we have other "advanced" beepers there as well? Adding sound maintainers to CC... Thanks. -- Dmitry

6 21

[PATCH -next v2] ALSA: Make SND_PCMTEST depend on DEBUG_FS
by Ruan Jinjie 17 Aug '23

17 Aug '23

Since pcmtest is a test module that manipulates or gets notification via debugfs, without DEBUG_FS it can not work fine. So make SND_PCMTEST depend on DEBUG_FS. Signed-off-by: Ruan Jinjie <ruanjinjie(a)huawei.com> --- v2: - make it depend on DEBUG_FS instead of dropping error check. --- sound/drivers/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig index 41c171468c1e..6debd8e95cb7 100644 --- a/sound/drivers/Kconfig +++ b/sound/drivers/Kconfig @@ -111,6 +111,7 @@ config SND_ALOOP config SND_PCMTEST tristate "Virtual PCM test driver" + depends on DEBUG_FS select SND_PCM help Say 'Y' or 'M' to include support for the Virtual PCM test driver. -- 2.34.1

3 2

Re: [PATCH -next] ALSA: pcmtest: Drop unnecessary error check for debugfs_create_dir
by Takashi Iwai 17 Aug '23

17 Aug '23

On Thu, 17 Aug 2023 08:39:22 +0200, Ruan Jinjie wrote: > > This patch removes the error checking for debugfs_create_dir in > pcmtest.c. This is because the DebugFS kernel API is developed > in a way that the caller can safely ignore the errors that > occur during the creation of DebugFS nodes. The debugfs APIs have > a IS_ERR() judge in start_creating() which can handle it > gracefully. So these checks are unnecessary. > > Signed-off-by: Ruan Jinjie <ruanjinjie(a)huawei.com> I believe debugfs is mandatory in this case (it's a test module that manipulates / gets notification via debugfs), hence it makes sense to check the error. Maybe we should add a dependency on CONFIG_DEBUG_FS in Kconfig? thanks, Takashi > --- > sound/drivers/pcmtest.c | 2 -- > 1 file changed, 2 deletions(-) > > diff --git a/sound/drivers/pcmtest.c b/sound/drivers/pcmtest.c > index 7f170429eb8f..9360b3bb771e 100644 > --- a/sound/drivers/pcmtest.c > +++ b/sound/drivers/pcmtest.c > @@ -669,8 +669,6 @@ static int init_debug_files(int buf_count) > char len_file_name[32]; > > driver_debug_dir = debugfs_create_dir("pcmtest", NULL); > - if (IS_ERR(driver_debug_dir)) > - return PTR_ERR(driver_debug_dir); > debugfs_create_u8("pc_test", 0444, driver_debug_dir, &playback_capture_test); > debugfs_create_u8("ioctl_test", 0444, driver_debug_dir, &ioctl_reset_test); > > -- > 2.34.1 >

2 3

[PATCH RFC 0/9] ALSA: Don't embed struct devices
by Takashi Iwai 17 Aug '23

17 Aug '23

Hi, this is another set of patches to attempt papering over the UAF problems that are seen when the delayed kobject release is enabled, as initially reported by Curtis: https://lore.kernel.org/r/20230801171928.1460120-1-cujomalainey@chromium.org There was a previous patch set with a different approach (using the device refcount dependencies), but this is a sort of step-back to the old way. https://lore.kernel.org/r/20230807135207.17708-1-tiwai@suse.de After discussions and evaluations, we agreed that decoupling the struct device from each sound component object is the safest (and easiest) way as of now. For applying the changes more consistently, I introduced a new helper for the struct device allocation and initialization, and applied all components. A couple of more changes for card_dev refcount managed aren't included in this patch set, though. They might be good to have, but this patch set should suffice for the currently seen UAF problems. For a long-term solution, we may restructure the device management, then the struct devices may be embedded again in each object. But, it'll need lots of other changes and cleanups, a big TODO. The latest patches are found in topic/dev-split branch of sound.git tree. Takashi === Takashi Iwai (9): ALSA: core: Introduce snd_device_alloc() ALSA: control: Don't embed ctl_dev ALSA: pcm: Don't embed device ALSA: hwdep: Don't embed device ALSA: rawmidi: Don't embed device ALSA: compress: Don't embed device ALSA: timer: Create device with snd_device_alloc() ALSA: seq: Create device with snd_device_alloc() ALSA: core: Drop snd_device_initialize() include/sound/compress_driver.h | 2 +- include/sound/core.h | 4 ++-- include/sound/hwdep.h | 2 +- include/sound/pcm.h | 2 +- include/sound/rawmidi.h | 2 +- sound/aoa/soundbus/i2sbus/pcm.c | 4 ++-- sound/core/compress_offload.c | 16 ++++++++------ sound/core/control.c | 14 ++++++------ sound/core/control_led.c | 4 ++-- sound/core/hwdep.c | 38 ++++++++++++++++++--------------- sound/core/init.c | 28 +++++++++++++++--------- sound/core/pcm.c | 22 +++++++++++-------- sound/core/rawmidi.c | 29 +++++++++++-------------- sound/core/seq/seq_clientmgr.c | 16 ++++++++------ sound/core/timer.c | 16 ++++++++------ sound/core/ump.c | 8 +++---- sound/pci/hda/hda_hwdep.c | 4 ++-- sound/usb/media.c | 4 ++-- 18 files changed, 119 insertions(+), 96 deletions(-) -- 2.35.3

3 12

[PATCH -next] ALSA: hda: cs35l41: change cs35l41_prop_model to static
by Yang Yingliang 17 Aug '23

17 Aug '23

cs35l41_prop_model is only used in cs35l41_hda_property.c now, change it to static. Fixes: ef4ba63f12b0 ("ALSA: hda: cs35l41: Support systems with missing _DSD properties") Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/pci/hda/cs35l41_hda_property.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index 673f23257a09..2d8a3edb868b 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -50,7 +50,7 @@ struct cs35l41_prop_model { const char *hid); }; -const struct cs35l41_prop_model cs35l41_prop_model_table[] = { +static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CLSA0100", NULL, lenovo_legion_no_acpi }, { "CLSA0101", NULL, lenovo_legion_no_acpi }, {} -- 2.25.1

2 1

[PATCH -next] ALSA: pcmtest: Drop unnecessary error check for debugfs_create_dir
by Ruan Jinjie 17 Aug '23

17 Aug '23

This patch removes the error checking for debugfs_create_dir in pcmtest.c. This is because the DebugFS kernel API is developed in a way that the caller can safely ignore the errors that occur during the creation of DebugFS nodes. The debugfs APIs have a IS_ERR() judge in start_creating() which can handle it gracefully. So these checks are unnecessary. Signed-off-by: Ruan Jinjie <ruanjinjie(a)huawei.com> --- sound/drivers/pcmtest.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/drivers/pcmtest.c b/sound/drivers/pcmtest.c index 7f170429eb8f..9360b3bb771e 100644 --- a/sound/drivers/pcmtest.c +++ b/sound/drivers/pcmtest.c @@ -669,8 +669,6 @@ static int init_debug_files(int buf_count) char len_file_name[32]; driver_debug_dir = debugfs_create_dir("pcmtest", NULL); - if (IS_ERR(driver_debug_dir)) - return PTR_ERR(driver_debug_dir); debugfs_create_u8("pc_test", 0444, driver_debug_dir, &playback_capture_test); debugfs_create_u8("ioctl_test", 0444, driver_debug_dir, &ioctl_reset_test); -- 2.34.1

2 1

SplitPCM: Fix Device variable in SplitPCMDevice macro
by GitHub pull_request - opened 17 Aug '23

17 Aug '23

alsa-project/alsa-ucm-conf pull request #339 was opened from WeirdTreeThing: Currently, the variable name is dev instead of Device as the comment would suggest, update the name to fit the comment. Also, the variable gets ignored due to it getting set to nothing right away, remove that line to allow the variable to work as intended. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/339 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/339.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: SOF: ipc4-pcm: fix possible null pointer deference
by Peter Ujfalusi 16 Aug '23

16 Aug '23

From: Chao Song <chao.song(a)linux.intel.com> The call to snd_sof_find_spcm_dai() could return NULL, add nullable check for the return value to avoid null pointer defenrece. Fixes: 7cb19007baba ("ASoC: SOF: ipc4-pcm: add hw_params") Signed-off-by: Chao Song <chao.song(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/ipc4-pcm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/sof/ipc4-pcm.c b/sound/soc/sof/ipc4-pcm.c index 802cbf73594e..db19cd03ecad 100644 --- a/sound/soc/sof/ipc4-pcm.c +++ b/sound/soc/sof/ipc4-pcm.c @@ -709,6 +709,9 @@ static int sof_ipc4_pcm_hw_params(struct snd_soc_component *component, struct snd_sof_pcm *spcm; spcm = snd_sof_find_spcm_dai(component, rtd); + if (!spcm) + return -EINVAL; + time_info = spcm->stream[substream->stream].private; /* delay calculation is not supported by current fw_reg ABI */ if (!time_info) -- 2.41.0

2 1

[PATCH v3 00/28] Add support for QMC HDLC, framer infrastruture and PEF2256 framer
by Herve Codina 16 Aug '23

16 Aug '23

Hi, I have a system where I need to handle an HDLC interface and some audio data. The HDLC data are transferred using a TDM bus on which a PEF2256 (E1/T1 framer) is present. The PEF2256 transfers data from/to the TDM bus to/from the E1 line. This PEF2256 is connected to a PowerQUICC SoC for the control path and the TDM is connected to the SoC (QMC component) for the data path. From the QMC HDLC driver, I need to handle HDLC data using the QMC, carrier detection using the PEF2256 (E1 line carrier) and set/get some PEF2256 configuration. The QMC HDLC driver considers the PEF2256 as a generic framer. It performs operations that involve the PEF2256 through the generic framer API. The audio data are exchanged with the PEF2256 using a CPU DAI connected to the TDM bus through the QMC and the PEF2256 needs to be seen as a codec in order to be linked to the CPU DAI. The codec handles the carrier detection using the PEF2256 and reports the carrier state to the ALSA subsystem using the ASoC jack detection. The codec, even if instantiated by the PEF2256 driver, considers the PEF2256 as a generic framer. The generic framer has: - 2 consumers (QMC HDLC drv and codec) - 1 provider (PEF2256) So, the design is the following: +------------------+ +---------+ | QMC | <- TDM -> | PEF2256 | <-> E1 +---------+ | +-------------+ | | | | CPU DAI | <-data--> | QMC channel | | | | +---------+ | +-------------+ | | | +--------------+ | +-------------+ | | | | QMC HDLC drv | <-data--> | QMC channel | | | | +--------------+ | +-------------+ | | | ^ +------------------+ | | | +--------+ +-------------+ | | +-> | framer | <-> | PEF2256 drv | <- local bus ->| | | | | | +---------+ +-> | | | | | +--------+ | +-------+ | +-------------------> | codec | | | +-------+ | +-------------+ Further more, the TDM timeslots used by the QMC HDLC driver need to be configured at runtime (QMC dynamic timeslots). Several weeks ago, I sent two series related to this topic: - Add the Lantiq PEF2256 audio support [1] - RFC Add support for QMC HDLC and PHY [2] This current series is a rework of these two series taking into account feedbacks previously received. In order to implement all of this, I do the following: 1) Perform some fixes (patches 1, 2, 3, 4) 2) Introduce the QMC HDLC driver (patches 5, 6, 7) 3) Add QMC dynamic timeslot support (patches 8 - 18) 4) Add timeslots change support in QMC HDLC (patch 19) 5) Introduce framer infrastructure (patch 20) 6) Add PEF2256 framer provider (patches 11, 22, 23, 24, 25) 7) Add framer codec as a framer consumer (patch 26) 8) Add framer support as a framer consumer in QMC HDLC (patch 27, 28) The series contains the full story and detailed modifications. If needed, the series can be split and/or commmits can be squashed. Let me know. Compare to the previous iteration https://lore.kernel.org/linux-kernel/20230726150225.483464-1-herve.codina@b… This v3 series mainly: - Fixes some implementation details. - Adds a new patch (patch 5) to remove existing inline keyword in the QMC driver. - Squashes patches related to the QMC HDLC binding together. Best regards, Hervé [1]: https://lore.kernel.org/all/20230417171601.74656-1-herve.codina@bootlin.com/ [2]: https://lore.kernel.org/all/20230323103154.264546-1-herve.codina@bootlin.co… Changes v2 -> v3 - Patches 1, 2, 3, 4 Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - New patch Remove inline keyword from the existing registers accessors helpers - Patch 6 (patches 5, 27 in v2) Update the binding title Squash patch 27 - Patch 7 (patch 6 in v2) Remove the cast in netdev_to_qmc_hdlc() Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patch 8 (patch 7 in v2): No change - Patches 9, 10 (patches 8, 9 in v2) Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patch 11 (patch 10 in v2) Remove inline keyword from the introduced qmc_clrsetbits16() helper Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patches 12, 13, 14, 15, 16, 17, 18, 19, 20 Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patch 21 (patch 20 in v2) Remove unneeded framer NULL pointer check Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patch 22 (patch 21 in v2) Change sclkr and sclkx clocks description Remove the framer phandle property from the framer subnodes (ie. from framer-codec nodes) - Patch 23 (patch 22 in v2) Initialize 'disabled' variable at declaration Fix commit log Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patch 24 (patch 23 in v2) Remove inline keyword from the existing registers accessors helpers Use dev_warn_ratelimited() in default interrupt handler Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patch 25 (patch 24 in v2) Replace #include "linux/bitfield.h" by #include <linux/bitfield.h> Fold the pinctrl anonymous struct into the struct pef2256_pinctrl Update commit log Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patch 26 (patch 25 in v2) Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patch 27 (patch 26 in v2) Fix error message Changed the ch.max computation in framer_dai_hw_rule_channels_by_format() Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patch 28 Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' Changes v1 -> v2 - Patches 1, 2 (New in v2) Fix __iomem addresses declaration - Patch 19 (17 in v1) Fix a compilation warning - Patch 26 (24 in v1) Fix a typo in Kconfig file Fix issues raised by sparse (make C=1) Herve Codina (28): soc: fsl: cpm1: tsa: Fix __iomem addresses declaration soc: fsl: cpm1: qmc: Fix __iomem addresses declaration soc: fsl: cpm1: qmc: Fix rx channel reset soc: fsl: cpm1: qmc: Extend the API to provide Rx status soc: fsl: cpm1: qmc: Remove inline function specifiers dt-bindings: net: Add support for QMC HDLC net: wan: Add support for QMC HDLC MAINTAINERS: Add the Freescale QMC HDLC driver entry soc: fsl: cpm1: qmc: Introduce available timeslots masks soc: fsl: cpm1: qmc: Rename qmc_setup_tsa* to qmc_init_tsa* soc: fsl: cpm1: qmc: Introduce qmc_chan_setup_tsa* soc: fsl: cpm1: qmc: Remove no more needed checks from qmc_check_chans() soc: fsl: cpm1: qmc: Check available timeslots in qmc_check_chans() soc: fsl: cpm1: qmc: Add support for disabling channel TSA entries soc: fsl: cpm1: qmc: Split Tx and Rx TSA entries setup soc: fsl: cpm1: qmc: Introduce is_tsa_64rxtx flag soc: fsl: cpm1: qmc: Handle timeslot entries at channel start() and stop() soc: fsl: cpm1: qmc: Remove timeslots handling from setup_chan() soc: fsl: cpm1: qmc: Introduce functions to change timeslots at runtime wan: qmc_hdlc: Add runtime timeslots changes support net: wan: Add framer framework support dt-bindings: net: Add the Lantiq PEF2256 E1/T1/J1 framer mfd: core: Ensure disabled devices are skiped without aborting net: wan: framer: Add support for the Lantiq PEF2256 framer pinctrl: Add support for the Lantic PEF2256 pinmux MAINTAINERS: Add the Lantiq PEF2256 driver entry ASoC: codecs: Add support for the framer codec net: wan: fsl_qmc_hdlc: Add framer support .../devicetree/bindings/net/fsl,qmc-hdlc.yaml | 46 + .../bindings/net/lantiq,pef2256.yaml | 219 +++++ MAINTAINERS | 17 + drivers/mfd/mfd-core.c | 17 +- drivers/net/wan/Kconfig | 14 + drivers/net/wan/Makefile | 3 + drivers/net/wan/framer/Kconfig | 35 + drivers/net/wan/framer/Makefile | 7 + drivers/net/wan/framer/framer-core.c | 886 ++++++++++++++++++ drivers/net/wan/framer/pef2256/Makefile | 8 + drivers/net/wan/framer/pef2256/pef2256-regs.h | 250 +++++ drivers/net/wan/framer/pef2256/pef2256.c | 880 +++++++++++++++++ drivers/net/wan/fsl_qmc_hdlc.c | 820 ++++++++++++++++ drivers/pinctrl/Kconfig | 14 + drivers/pinctrl/Makefile | 1 + drivers/pinctrl/pinctrl-pef2256-regs.h | 65 ++ drivers/pinctrl/pinctrl-pef2256.c | 308 ++++++ drivers/soc/fsl/qe/qmc.c | 501 ++++++++-- drivers/soc/fsl/qe/tsa.c | 22 +- include/linux/framer/framer-provider.h | 194 ++++ include/linux/framer/framer.h | 199 ++++ include/linux/framer/pef2256.h | 31 + include/soc/fsl/qe/qmc.h | 25 +- sound/soc/codecs/Kconfig | 15 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/framer-codec.c | 413 ++++++++ sound/soc/fsl/fsl_qmc_audio.c | 2 +- 27 files changed, 4872 insertions(+), 122 deletions(-) create mode 100644 Documentation/devicetree/bindings/net/fsl,qmc-hdlc.yaml create mode 100644 Documentation/devicetree/bindings/net/lantiq,pef2256.yaml create mode 100644 drivers/net/wan/framer/Kconfig create mode 100644 drivers/net/wan/framer/Makefile create mode 100644 drivers/net/wan/framer/framer-core.c create mode 100644 drivers/net/wan/framer/pef2256/Makefile create mode 100644 drivers/net/wan/framer/pef2256/pef2256-regs.h create mode 100644 drivers/net/wan/framer/pef2256/pef2256.c create mode 100644 drivers/net/wan/fsl_qmc_hdlc.c create mode 100644 drivers/pinctrl/pinctrl-pef2256-regs.h create mode 100644 drivers/pinctrl/pinctrl-pef2256.c create mode 100644 include/linux/framer/framer-provider.h create mode 100644 include/linux/framer/framer.h create mode 100644 include/linux/framer/pef2256.h create mode 100644 sound/soc/codecs/framer-codec.c -- 2.41.0

5 36

Fwd: [Bug 217440] New: ALC236 audio disappears from HP 15z-fc000 on warm boot
by Joseph C. Sible 16 Aug '23

16 Aug '23

Hello, Forwarding my regression bug report submission per the instructions in "Reporting regressions". #regzbot introduced: 5aec989130 https://bugzilla.kernel.org/show_bug.cgi?id=217440 Regards, Joseph C. Sible ---------- Forwarded message --------- From: <bugzilla-daemon(a)kernel.org> Date: Sat, May 13, 2023 at 3:46 PM Subject: [Bug 217440] New: ALC236 audio disappears from HP 15z-fc000 on warm boot To: <josephcsible(a)gmail.com> https://bugzilla.kernel.org/show_bug.cgi?id=217440 Bug ID: 217440 Summary: ALC236 audio disappears from HP 15z-fc000 on warm boot Product: Drivers Version: 2.5 Kernel Version: 6.3.2 Hardware: All OS: Linux Status: NEW Severity: normal Priority: P3 Component: Sound(ALSA) Assignee: perex(a)perex.cz Reporter: josephcsible(a)gmail.com Regression: Yes Bisected 5aec98913095ed3b4424ed6c5fdeb6964e9734da commit-id: On an HP 15z-fc000 laptop that uses Realtek ALC236 for sound, once the system is warm booted ("reboot" from within Linux), sound stops working completely and the device doesn't even show up anymore. Once this happens, the only way to get it to come back is to cold boot ("poweroff" from within Linux, wait for it to turn completely off, and turn it back on with the physical power button). In particular, once you boot from a "bad" kernel, no number of warm boots with "good" kernels will ever bring sound back. I bisected this to commit 5aec989, which first appeared in 5.15-rc6. In addition, I confirmed by doing a custom build of 6.3.2 with reverts of both f30741c (just to avoid a conflict) and 5aec989, which works correctly for me. -- You may reply to this email to add a comment. You are receiving this mail because: You reported the bug.

5 48

[PATCH] -- fix Boss MS-3 midi-usb communication
by john_candlish＠mail.candlish.net 16 Aug '23

16 Aug '23

This one line change gets MIDI communication flowing between ALSA and the Boss MS-3 multi-effect switcher. ///// *** sound/usb/midi.c 2023-08-10 22:46:31.016346930 +0200 --- sound/usb/midi.c-orig 2023-07-19 16:37:03.000000000 +0200 *************** *** 2106,2113 **** struct usb_endpoint_descriptor *epd; int i, out_eps = 0, in_eps = 0; ! if (USB_ID_VENDOR(umidi->usb_id) == 0x0582 && ! USB_ID_PRODUCT(umidi->usb_id) != 0x01f5) snd_usbmidi_switch_roland_altsetting(umidi); if (endpoint[0].out_ep || endpoint[0].in_ep) --- 2106,2112 ---- struct usb_endpoint_descriptor *epd; int i, out_eps = 0, in_eps = 0; ! if (USB_ID_VENDOR(umidi->usb_id) == 0x0582) snd_usbmidi_switch_roland_altsetting(umidi); if (endpoint[0].out_ep || endpoint[0].in_ep) ///// There are doubtless other Roland/Boss USB_ID_PRODUCT id's that are effected and it would be nice to make the USB_ID_PRODUCT value a tunable, but that is beyond my competence.

2 3

Build fails
by GitHub issues - edited 16 Aug '23

16 Aug '23

alsa-project/alsa-utils issue #230 was edited from symdeb: Follow INSTALL $ls -l total 148 -rw-rw-r-- 1 user user 440 Aug 16 17:40 acinclude.m4 drwxrwxr-x 3 user user 4096 Aug 16 17:40 alsaconf drwxrwxr-x 3 user user 4096 Aug 16 17:40 alsactl drwxrwxr-x 2 user user 4096 Aug 16 17:40 alsa-info drwxrwxr-x 2 user user 4096 Aug 16 17:40 alsaloop drwxrwxr-x 2 user user 4096 Aug 16 17:40 alsamixer drwxrwxr-x 2 user user 4096 Aug 16 17:40 alsaucm drwxrwxr-x 2 user user 4096 Aug 16 17:40 amidi drwxrwxr-x 2 user user 4096 Aug 16 17:40 amixer drwxrwxr-x 2 user user 4096 Aug 16 17:40 aplay drwxrwxr-x 3 user user 4096 Aug 16 17:40 axfer drwxrwxr-x 3 user user 4096 Aug 16 17:40 bat -rw-rw-r-- 1 user user 1055 Aug 16 17:40 ChangeLog -rw-rw-r-- 1 user user 17587 Aug 16 17:40 configure.ac -rw-rw-r-- 1 user user 18002 Aug 16 17:40 COPYING -rwxrwxr-x 1 user user 780 Aug 16 17:40 gitcompile drwxrwxr-x 2 user user 4096 Aug 16 17:40 iecset drwxrwxr-x 2 user user 4096 Aug 16 17:40 include -rw-rw-r-- 1 user user 1595 Aug 16 17:40 INSTALL drwxrwxr-x 2 user user 4096 Aug 16 17:40 m4 -rw-rw-r-- 1 user user 1063 Aug 16 17:40 Makefile.am drwxrwxr-x 2 user user 4096 Aug 16 17:40 nhlt drwxrwxr-x 2 user user 4096 Aug 16 17:40 po -rw-rw-r-- 1 user user 1392 Aug 16 17:40 README.md drwxrwxr-x 6 user user 4096 Aug 16 17:40 seq drwxrwxr-x 3 user user 4096 Aug 16 17:40 speaker-test -rw-rw-r-- 1 user user 187 Aug 16 17:40 TODO drwxrwxr-x 3 user user 4096 Aug 16 17:40 topology drwxrwxr-x 2 user user 4096 Aug 16 17:40 utils $aclocal $gettextize Copying file ABOUT-NLS Copying file config.rpath Copying file po/Makefile.in.in Copying file po/Makevars.template Copying file po/Rules-quot Copying file po/boldquot.sed Copying file po/en(a)boldquot.header Copying file po/en(a)quot.header Copying file po/insert-header.sin Copying file po/quot.sed Copying file po/remove-potcdate.sin Creating po/ChangeLog Copying file m4/gettext.m4 Copying file m4/host-cpu-c-abi.m4 Copying file m4/iconv.m4 Copying file m4/intlmacosx.m4 Copying file m4/lib-ld.m4 Copying file m4/lib-link.m4 Copying file m4/lib-prefix.m4 Copying file m4/nls.m4 Copying file m4/po.m4 Copying file m4/progtest.m4 Updating EXTRA_DIST in m4/Makefile.am (backup is in m4/Makefile.am~) Updating configure.ac (backup is in configure.ac~) Adding an entry to ChangeLog (backup is in ChangeLog~) Please run 'aclocal -I m4' to regenerate the aclocal.m4 file. You need aclocal from GNU automake 1.9 (or newer) to do this. Then run 'autoconf' to regenerate the configure file. You will also need config.guess and config.sub, which you can get from the CVS of the 'config' project at https://savannah.gnu.org/. The commands to fetch them are $ wget 'https://savannah.gnu.org/cgi-bin/viewcvs/*checkout*/config/config/config.gu…' $ wget 'https://savannah.gnu.org/cgi-bin/viewcvs/*checkout*/config/config/config.sub' You might also want to copy the convenience header file gettext.h from the /usr/share/gettext directory into your package. It is a wrapper around <libintl.h> that implements the configure --disable-nls option. Press Return to acknowledge the previous three paragraphs. $autoheader configure.ac:480: error: `po/Makefile.in' is already registered with AC_CONFIG_FILES. ./lib/autoconf/status.m4:289: AC_CONFIG_FILES is expanded from... configure.ac:480: the top level autom4te: error: /usr/bin/m4 failed with exit status: 1 autoheader: error: '/usr/bin/autom4te' failed with exit status: 1 $automake --foreign --copy --add-missing configure.ac:480: error: `po/Makefile.in' is already registered with AC_CONFIG_FILES. ./lib/autoconf/status.m4:289: AC_CONFIG_FILES is expanded from... configure.ac:480: the top level autom4te: error: /usr/bin/m4 failed with exit status: 1 $aclocal -I m4 configure.ac:480: error: `po/Makefile.in' is already registered with AC_CONFIG_FILES. ./lib/autoconf/status.m4:289: AC_CONFIG_FILES is expanded from... configure.ac:480: the top level autom4te: error: /usr/bin/m4 failed with exit status: 1 aclocal: error: autom4te failed with exit status: 1 ./configure bash: ./configure: No such file or directory aclocal --version aclocal (GNU automake) 1.16.5 Issue URL : https://github.com/alsa-project/alsa-utils/issues/230 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Build fails
by GitHub issues - edited 16 Aug '23

16 Aug '23

alsa-project/alsa-utils issue #230 was edited from symdeb: Follow INSTALL $ls -l total 148 -rw-rw-r-- 1 user user 440 Aug 16 17:40 acinclude.m4 drwxrwxr-x 3 user user 4096 Aug 16 17:40 alsaconf drwxrwxr-x 3 user user 4096 Aug 16 17:40 alsactl drwxrwxr-x 2 user user 4096 Aug 16 17:40 alsa-info drwxrwxr-x 2 user user 4096 Aug 16 17:40 alsaloop drwxrwxr-x 2 user user 4096 Aug 16 17:40 alsamixer drwxrwxr-x 2 user user 4096 Aug 16 17:40 alsaucm drwxrwxr-x 2 user user 4096 Aug 16 17:40 amidi drwxrwxr-x 2 user user 4096 Aug 16 17:40 amixer drwxrwxr-x 2 user user 4096 Aug 16 17:40 aplay drwxrwxr-x 3 user user 4096 Aug 16 17:40 axfer drwxrwxr-x 3 user user 4096 Aug 16 17:40 bat -rw-rw-r-- 1 user user 1055 Aug 16 17:40 ChangeLog -rw-rw-r-- 1 user user 17587 Aug 16 17:40 configure.ac -rw-rw-r-- 1 user user 18002 Aug 16 17:40 COPYING -rwxrwxr-x 1 user user 780 Aug 16 17:40 gitcompile drwxrwxr-x 2 user user 4096 Aug 16 17:40 iecset drwxrwxr-x 2 user user 4096 Aug 16 17:40 include -rw-rw-r-- 1 user user 1595 Aug 16 17:40 INSTALL drwxrwxr-x 2 user user 4096 Aug 16 17:40 m4 -rw-rw-r-- 1 user user 1063 Aug 16 17:40 Makefile.am drwxrwxr-x 2 user user 4096 Aug 16 17:40 nhlt drwxrwxr-x 2 user user 4096 Aug 16 17:40 po -rw-rw-r-- 1 user user 1392 Aug 16 17:40 README.md drwxrwxr-x 6 user user 4096 Aug 16 17:40 seq drwxrwxr-x 3 user user 4096 Aug 16 17:40 speaker-test -rw-rw-r-- 1 user user 187 Aug 16 17:40 TODO drwxrwxr-x 3 user user 4096 Aug 16 17:40 topology drwxrwxr-x 2 user user 4096 Aug 16 17:40 utils $aclocal $gettextize Copying file ABOUT-NLS Copying file config.rpath Copying file po/Makefile.in.in Copying file po/Makevars.template Copying file po/Rules-quot Copying file po/boldquot.sed Copying file po/en(a)boldquot.header Copying file po/en(a)quot.header Copying file po/insert-header.sin Copying file po/quot.sed Copying file po/remove-potcdate.sin Creating po/ChangeLog Copying file m4/gettext.m4 Copying file m4/host-cpu-c-abi.m4 Copying file m4/iconv.m4 Copying file m4/intlmacosx.m4 Copying file m4/lib-ld.m4 Copying file m4/lib-link.m4 Copying file m4/lib-prefix.m4 Copying file m4/nls.m4 Copying file m4/po.m4 Copying file m4/progtest.m4 Updating EXTRA_DIST in m4/Makefile.am (backup is in m4/Makefile.am~) Updating configure.ac (backup is in configure.ac~) Adding an entry to ChangeLog (backup is in ChangeLog~) Please run 'aclocal -I m4' to regenerate the aclocal.m4 file. You need aclocal from GNU automake 1.9 (or newer) to do this. Then run 'autoconf' to regenerate the configure file. You will also need config.guess and config.sub, which you can get from the CVS of the 'config' project at https://savannah.gnu.org/. The commands to fetch them are $ wget 'https://savannah.gnu.org/cgi-bin/viewcvs/*checkout*/config/config/config.gu…' $ wget 'https://savannah.gnu.org/cgi-bin/viewcvs/*checkout*/config/config/config.sub' You might also want to copy the convenience header file gettext.h from the /usr/share/gettext directory into your package. It is a wrapper around <libintl.h> that implements the configure --disable-nls option. Press Return to acknowledge the previous three paragraphs. $autoheader configure.ac:480: error: `po/Makefile.in' is already registered with AC_CONFIG_FILES. ./lib/autoconf/status.m4:289: AC_CONFIG_FILES is expanded from... configure.ac:480: the top level autom4te: error: /usr/bin/m4 failed with exit status: 1 autoheader: error: '/usr/bin/autom4te' failed with exit status: 1 $automake --foreign --copy --add-missing configure.ac:480: error: `po/Makefile.in' is already registered with AC_CONFIG_FILES. ./lib/autoconf/status.m4:289: AC_CONFIG_FILES is expanded from... configure.ac:480: the top level autom4te: error: /usr/bin/m4 failed with exit status: 1 $aclocal -I m4 configure.ac:480: error: `po/Makefile.in' is already registered with AC_CONFIG_FILES. ./lib/autoconf/status.m4:289: AC_CONFIG_FILES is expanded from... configure.ac:480: the top level autom4te: error: /usr/bin/m4 failed with exit status: 1 aclocal: error: autom4te failed with exit status: 1 ./configure bash: ./configure: No such file or directory Issue URL : https://github.com/alsa-project/alsa-utils/issues/230 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Build fails
by GitHub issues - edited 16 Aug '23

16 Aug '23

alsa-project/alsa-utils issue #230 was edited from symdeb: Follow INSTALL $ls -l total 148 -rw-rw-r-- 1 user user 440 Aug 16 17:40 acinclude.m4 drwxrwxr-x 3 user user 4096 Aug 16 17:40 alsaconf drwxrwxr-x 3 user user 4096 Aug 16 17:40 alsactl drwxrwxr-x 2 user user 4096 Aug 16 17:40 alsa-info drwxrwxr-x 2 user user 4096 Aug 16 17:40 alsaloop drwxrwxr-x 2 user user 4096 Aug 16 17:40 alsamixer drwxrwxr-x 2 user user 4096 Aug 16 17:40 alsaucm drwxrwxr-x 2 user user 4096 Aug 16 17:40 amidi drwxrwxr-x 2 user user 4096 Aug 16 17:40 amixer drwxrwxr-x 2 user user 4096 Aug 16 17:40 aplay drwxrwxr-x 3 user user 4096 Aug 16 17:40 axfer drwxrwxr-x 3 user user 4096 Aug 16 17:40 bat -rw-rw-r-- 1 user user 1055 Aug 16 17:40 ChangeLog -rw-rw-r-- 1 user user 17587 Aug 16 17:40 configure.ac -rw-rw-r-- 1 user user 18002 Aug 16 17:40 COPYING -rwxrwxr-x 1 user user 780 Aug 16 17:40 gitcompile drwxrwxr-x 2 user user 4096 Aug 16 17:40 iecset drwxrwxr-x 2 user user 4096 Aug 16 17:40 include -rw-rw-r-- 1 user user 1595 Aug 16 17:40 INSTALL drwxrwxr-x 2 user user 4096 Aug 16 17:40 m4 -rw-rw-r-- 1 user user 1063 Aug 16 17:40 Makefile.am drwxrwxr-x 2 user user 4096 Aug 16 17:40 nhlt drwxrwxr-x 2 user user 4096 Aug 16 17:40 po -rw-rw-r-- 1 user user 1392 Aug 16 17:40 README.md drwxrwxr-x 6 user user 4096 Aug 16 17:40 seq drwxrwxr-x 3 user user 4096 Aug 16 17:40 speaker-test -rw-rw-r-- 1 user user 187 Aug 16 17:40 TODO drwxrwxr-x 3 user user 4096 Aug 16 17:40 topology drwxrwxr-x 2 user user 4096 Aug 16 17:40 utils $aclocal $gettextize Copying file ABOUT-NLS Copying file config.rpath Copying file po/Makefile.in.in Copying file po/Makevars.template Copying file po/Rules-quot Copying file po/boldquot.sed Copying file po/en(a)boldquot.header Copying file po/en(a)quot.header Copying file po/insert-header.sin Copying file po/quot.sed Copying file po/remove-potcdate.sin Creating po/ChangeLog Copying file m4/gettext.m4 Copying file m4/host-cpu-c-abi.m4 Copying file m4/iconv.m4 Copying file m4/intlmacosx.m4 Copying file m4/lib-ld.m4 Copying file m4/lib-link.m4 Copying file m4/lib-prefix.m4 Copying file m4/nls.m4 Copying file m4/po.m4 Copying file m4/progtest.m4 Updating EXTRA_DIST in m4/Makefile.am (backup is in m4/Makefile.am~) Updating configure.ac (backup is in configure.ac~) Adding an entry to ChangeLog (backup is in ChangeLog~) Please run 'aclocal -I m4' to regenerate the aclocal.m4 file. You need aclocal from GNU automake 1.9 (or newer) to do this. Then run 'autoconf' to regenerate the configure file. You will also need config.guess and config.sub, which you can get from the CVS of the 'config' project at https://savannah.gnu.org/. The commands to fetch them are $ wget 'https://savannah.gnu.org/cgi-bin/viewcvs/*checkout*/config/config/config.gu…' $ wget 'https://savannah.gnu.org/cgi-bin/viewcvs/*checkout*/config/config/config.sub' You might also want to copy the convenience header file gettext.h from the /usr/share/gettext directory into your package. It is a wrapper around <libintl.h> that implements the configure --disable-nls option. Press Return to acknowledge the previous three paragraphs. $autoheader configure.ac:480: error: `po/Makefile.in' is already registered with AC_CONFIG_FILES. ./lib/autoconf/status.m4:289: AC_CONFIG_FILES is expanded from... configure.ac:480: the top level autom4te: error: /usr/bin/m4 failed with exit status: 1 autoheader: error: '/usr/bin/autom4te' failed with exit status: 1 $automake --foreign --copy --add-missing configure.ac:480: error: `po/Makefile.in' is already registered with AC_CONFIG_FILES. ./lib/autoconf/status.m4:289: AC_CONFIG_FILES is expanded from... configure.ac:480: the top level autom4te: error: /usr/bin/m4 failed with exit status: 1 aclocal -I m4 configure.ac:480: error: `po/Makefile.in' is already registered with AC_CONFIG_FILES. ./lib/autoconf/status.m4:289: AC_CONFIG_FILES is expanded from... configure.ac:480: the top level autom4te: error: /usr/bin/m4 failed with exit status: 1 aclocal: error: autom4te failed with exit status: 1 ./configure bash: ./configure: No such file or directory Issue URL : https://github.com/alsa-project/alsa-utils/issues/230 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Build fails
by GitHub issues - edited 16 Aug '23

16 Aug '23

alsa-project/alsa-utils issue #230 was edited from symdeb: Follow INSTALL $ls -l total 148 -rw-rw-r-- 1 user user 440 Aug 16 17:40 acinclude.m4 drwxrwxr-x 3 user user 4096 Aug 16 17:40 alsaconf drwxrwxr-x 3 user user 4096 Aug 16 17:40 alsactl drwxrwxr-x 2 user user 4096 Aug 16 17:40 alsa-info drwxrwxr-x 2 user user 4096 Aug 16 17:40 alsaloop drwxrwxr-x 2 user user 4096 Aug 16 17:40 alsamixer drwxrwxr-x 2 user user 4096 Aug 16 17:40 alsaucm drwxrwxr-x 2 user user 4096 Aug 16 17:40 amidi drwxrwxr-x 2 user user 4096 Aug 16 17:40 amixer drwxrwxr-x 2 user user 4096 Aug 16 17:40 aplay drwxrwxr-x 3 user user 4096 Aug 16 17:40 axfer drwxrwxr-x 3 user user 4096 Aug 16 17:40 bat -rw-rw-r-- 1 user user 1055 Aug 16 17:40 ChangeLog -rw-rw-r-- 1 user user 17587 Aug 16 17:40 configure.ac -rw-rw-r-- 1 user user 18002 Aug 16 17:40 COPYING -rwxrwxr-x 1 user user 780 Aug 16 17:40 gitcompile drwxrwxr-x 2 user user 4096 Aug 16 17:40 iecset drwxrwxr-x 2 user user 4096 Aug 16 17:40 include -rw-rw-r-- 1 user user 1595 Aug 16 17:40 INSTALL drwxrwxr-x 2 user user 4096 Aug 16 17:40 m4 -rw-rw-r-- 1 user user 1063 Aug 16 17:40 Makefile.am drwxrwxr-x 2 user user 4096 Aug 16 17:40 nhlt drwxrwxr-x 2 user user 4096 Aug 16 17:40 po -rw-rw-r-- 1 user user 1392 Aug 16 17:40 README.md drwxrwxr-x 6 user user 4096 Aug 16 17:40 seq drwxrwxr-x 3 user user 4096 Aug 16 17:40 speaker-test -rw-rw-r-- 1 user user 187 Aug 16 17:40 TODO drwxrwxr-x 3 user user 4096 Aug 16 17:40 topology drwxrwxr-x 2 user user 4096 Aug 16 17:40 utils $aclocal $gettextize Copying file ABOUT-NLS Copying file config.rpath Copying file po/Makefile.in.in Copying file po/Makevars.template Copying file po/Rules-quot Copying file po/boldquot.sed Copying file po/en(a)boldquot.header Copying file po/en(a)quot.header Copying file po/insert-header.sin Copying file po/quot.sed Copying file po/remove-potcdate.sin Creating po/ChangeLog Copying file m4/gettext.m4 Copying file m4/host-cpu-c-abi.m4 Copying file m4/iconv.m4 Copying file m4/intlmacosx.m4 Copying file m4/lib-ld.m4 Copying file m4/lib-link.m4 Copying file m4/lib-prefix.m4 Copying file m4/nls.m4 Copying file m4/po.m4 Copying file m4/progtest.m4 Updating EXTRA_DIST in m4/Makefile.am (backup is in m4/Makefile.am~) Updating configure.ac (backup is in configure.ac~) Adding an entry to ChangeLog (backup is in ChangeLog~) Please run 'aclocal -I m4' to regenerate the aclocal.m4 file. You need aclocal from GNU automake 1.9 (or newer) to do this. Then run 'autoconf' to regenerate the configure file. You will also need config.guess and config.sub, which you can get from the CVS of the 'config' project at https://savannah.gnu.org/. The commands to fetch them are $ wget 'https://savannah.gnu.org/cgi-bin/viewcvs/*checkout*/config/config/config.gu…' $ wget 'https://savannah.gnu.org/cgi-bin/viewcvs/*checkout*/config/config/config.sub' You might also want to copy the convenience header file gettext.h from the /usr/share/gettext directory into your package. It is a wrapper around <libintl.h> that implements the configure --disable-nls option. Press Return to acknowledge the previous three paragraphs. $autoheader configure.ac:480: error: `po/Makefile.in' is already registered with AC_CONFIG_FILES. ./lib/autoconf/status.m4:289: AC_CONFIG_FILES is expanded from... configure.ac:480: the top level autom4te: error: /usr/bin/m4 failed with exit status: 1 autoheader: error: '/usr/bin/autom4te' failed with exit status: 1 $automake --foreign --copy --add-missing configure.ac:480: error: `po/Makefile.in' is already registered with AC_CONFIG_FILES. ./lib/autoconf/status.m4:289: AC_CONFIG_FILES is expanded from... configure.ac:480: the top level autom4te: error: /usr/bin/m4 failed with exit status: 1 aclocal -I m4 configure.ac:480: error: `po/Makefile.in' is already registered with AC_CONFIG_FILES. ./lib/autoconf/status.m4:289: AC_CONFIG_FILES is expanded from... configure.ac:480: the top level autom4te: error: /usr/bin/m4 failed with exit status: 1 aclocal: error: autom4te failed with exit status: 1 Issue URL : https://github.com/alsa-project/alsa-utils/issues/230 Repository URL: https://github.com/alsa-project/alsa-utils

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Build fails
by GitHub issues - opened 16 Aug '23

16 Aug '23

alsa-project/alsa-utils issue #230 was opened from symdeb: Follow INSTALL $ls -l total 148 -rw-rw-r-- 1 user user 440 Aug 16 17:40 acinclude.m4 drwxrwxr-x 3 user user 4096 Aug 16 17:40 alsaconf drwxrwxr-x 3 user user 4096 Aug 16 17:40 alsactl drwxrwxr-x 2 user user 4096 Aug 16 17:40 alsa-info drwxrwxr-x 2 user user 4096 Aug 16 17:40 alsaloop drwxrwxr-x 2 user user 4096 Aug 16 17:40 alsamixer drwxrwxr-x 2 user user 4096 Aug 16 17:40 alsaucm drwxrwxr-x 2 user user 4096 Aug 16 17:40 amidi drwxrwxr-x 2 user user 4096 Aug 16 17:40 amixer drwxrwxr-x 2 user user 4096 Aug 16 17:40 aplay drwxrwxr-x 3 user user 4096 Aug 16 17:40 axfer drwxrwxr-x 3 user user 4096 Aug 16 17:40 bat -rw-rw-r-- 1 user user 1055 Aug 16 17:40 ChangeLog -rw-rw-r-- 1 user user 17587 Aug 16 17:40 configure.ac -rw-rw-r-- 1 user user 18002 Aug 16 17:40 COPYING -rwxrwxr-x 1 user user 780 Aug 16 17:40 gitcompile drwxrwxr-x 2 user user 4096 Aug 16 17:40 iecset drwxrwxr-x 2 user user 4096 Aug 16 17:40 include -rw-rw-r-- 1 user user 1595 Aug 16 17:40 INSTALL drwxrwxr-x 2 user user 4096 Aug 16 17:40 m4 -rw-rw-r-- 1 user user 1063 Aug 16 17:40 Makefile.am drwxrwxr-x 2 user user 4096 Aug 16 17:40 nhlt drwxrwxr-x 2 user user 4096 Aug 16 17:40 po -rw-rw-r-- 1 user user 1392 Aug 16 17:40 README.md drwxrwxr-x 6 user user 4096 Aug 16 17:40 seq drwxrwxr-x 3 user user 4096 Aug 16 17:40 speaker-test -rw-rw-r-- 1 user user 187 Aug 16 17:40 TODO drwxrwxr-x 3 user user 4096 Aug 16 17:40 topology drwxrwxr-x 2 user user 4096 Aug 16 17:40 utils $aclocal $gettextize Copying file ABOUT-NLS Copying file config.rpath Copying file po/Makefile.in.in Copying file po/Makevars.template Copying file po/Rules-quot Copying file po/boldquot.sed Copying file po/en(a)boldquot.header Copying file po/en(a)quot.header Copying file po/insert-header.sin Copying file po/quot.sed Copying file po/remove-potcdate.sin Creating po/ChangeLog Copying file m4/gettext.m4 Copying file m4/host-cpu-c-abi.m4 Copying file m4/iconv.m4 Copying file m4/intlmacosx.m4 Copying file m4/lib-ld.m4 Copying file m4/lib-link.m4 Copying file m4/lib-prefix.m4 Copying file m4/nls.m4 Copying file m4/po.m4 Copying file m4/progtest.m4 Updating EXTRA_DIST in m4/Makefile.am (backup is in m4/Makefile.am~) Updating configure.ac (backup is in configure.ac~) Adding an entry to ChangeLog (backup is in ChangeLog~) Please run 'aclocal -I m4' to regenerate the aclocal.m4 file. You need aclocal from GNU automake 1.9 (or newer) to do this. Then run 'autoconf' to regenerate the configure file. You will also need config.guess and config.sub, which you can get from the CVS of the 'config' project at https://savannah.gnu.org/. The commands to fetch them are $ wget 'https://savannah.gnu.org/cgi-bin/viewcvs/*checkout*/config/config/config.gu…' $ wget 'https://savannah.gnu.org/cgi-bin/viewcvs/*checkout*/config/config/config.sub' You might also want to copy the convenience header file gettext.h from the /usr/share/gettext directory into your package. It is a wrapper around <libintl.h> that implements the configure --disable-nls option. Press Return to acknowledge the previous three paragraphs. $autoheader configure.ac:480: error: `po/Makefile.in' is already registered with AC_CONFIG_FILES. ./lib/autoconf/status.m4:289: AC_CONFIG_FILES is expanded from... configure.ac:480: the top level autom4te: error: /usr/bin/m4 failed with exit status: 1 autoheader: error: '/usr/bin/autom4te' failed with exit status: 1 $automake --foreign --copy --add-missing configure.ac:480: error: `po/Makefile.in' is already registered with AC_CONFIG_FILES. ./lib/autoconf/status.m4:289: AC_CONFIG_FILES is expanded from... configure.ac:480: the top level autom4te: error: /usr/bin/m4 failed with exit status: 1 Issue URL : https://github.com/alsa-project/alsa-utils/issues/230 Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH RESEND 1/3] ASoC: codecs: tlv320aic32x4: Fix Wvoid-pointer-to-enum-cast warning
by Krzysztof Kozlowski 16 Aug '23

16 Aug '23

'type' is an enum, thus cast of pointer on 64-bit compile test with W=1 causes: tlv320aic32x4.c:1352:18: error: cast to smaller integer type 'enum aic32x4_type' from 'void *' [-Werror,-Wvoid-pointer-to-enum-cast] Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Resending because thread got duplicated/messed on the lists. I also dropped patch for aw88261 from the series, because it was fixed other way by Arnd. --- sound/soc/codecs/tlv320aic32x4.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index ffe1828a4b7e..6829834a412f 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -1349,7 +1349,7 @@ int aic32x4_probe(struct device *dev, struct regmap *regmap) return -ENOMEM; aic32x4->dev = dev; - aic32x4->type = (enum aic32x4_type)dev_get_drvdata(dev); + aic32x4->type = (uintptr_t)dev_get_drvdata(dev); dev_set_drvdata(dev, aic32x4); -- 2.34.1

3 4

[PATCH v2] ALSA: hda: intel-dsp-cfg: Add Chromebook quirk to ADL/RPL
by Brady Norander 16 Aug '23

16 Aug '23

AlderLake and RaptorLake Chromebooks currently use the HDA driver by default. Add a quirk to use the SOF driver on these platforms, which is needed for functional internal audio. Signed-off-by: Brady Norander <bradynorander(a)gmail.com> --- v2: Removed quirk from SKUs not used in Chrome platforms sound/hda/intel-dsp-config.c | 60 +++++++++++++++++++++++++++++++++--- 1 file changed, 56 insertions(+), 4 deletions(-) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index dcf2453138a5..24a948baf1bc 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -167,10 +167,10 @@ static const struct config_entry config_table[] = { #endif /* - * CoffeeLake, CannonLake, CometLake, IceLake, TigerLake use legacy - * HDAudio driver except for Google Chromebooks and when DMICs are - * present. Two cases are required since Coreboot does not expose NHLT - * tables. + * CoffeeLake, CannonLake, CometLake, IceLake, TigerLake, AlderLake, + * RaptorLake use legacy HDAudio driver except for Google Chromebooks + * and when DMICs are present. Two cases are required since Coreboot + * does not expose NHLT tables. * * When the Chromebook quirk is not present, it's based on information * that no such device exists. When the quirk is present, it could be @@ -408,6 +408,19 @@ static const struct config_entry config_table[] = { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = PCI_DEVICE_ID_INTEL_HDA_RPL_S, }, + { + .flags = FLAG_SOF, + .device = PCI_DEVICE_ID_INTEL_HDA_ADL_P, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Google Chromebooks", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + } + }, + {} + } + }, { .flags = FLAG_SOF, .device = PCI_DEVICE_ID_INTEL_HDA_ADL_P, @@ -434,14 +447,53 @@ static const struct config_entry config_table[] = { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = PCI_DEVICE_ID_INTEL_HDA_ADL_M, }, + { + .flags = FLAG_SOF, + .device = PCI_DEVICE_ID_INTEL_HDA_ADL_N, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Google Chromebooks", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + } + }, + {} + } + }, { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = PCI_DEVICE_ID_INTEL_HDA_ADL_N, }, + { + .flags = FLAG_SOF, + .device = PCI_DEVICE_ID_INTEL_HDA_RPL_P_0, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Google Chromebooks", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + } + }, + {} + } + }, { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = PCI_DEVICE_ID_INTEL_HDA_RPL_P_0, }, + { + .flags = FLAG_SOF, + .device = PCI_DEVICE_ID_INTEL_HDA_RPL_P_1, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Google Chromebooks", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + } + }, + {} + } + }, { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = PCI_DEVICE_ID_INTEL_HDA_RPL_P_1, -- 2.41.0

4 7

[PATCH] ALSA: hda: cs35l41: Fix the loop check in cs35l41_add_dsd_properties
by Takashi Iwai 16 Aug '23

16 Aug '23

model->hid is a pointer, and should be rather NULL-checked in the loop of cs35l41_prop_model_table. Fixes: ef4ba63f12b0 ("ALSA: hda: cs35l41: Support systems with missing _DSD properties") Reported-by: kernel test robot <lkp(a)intel.com> Closes: https://lore.kernel.org/oe-kbuild-all/202308160506.8lCEeFDG-lkp@intel.com/ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/cs35l41_hda_property.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index 673f23257a09..48dcc3f1ef88 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -61,7 +61,7 @@ int cs35l41_add_dsd_properties(struct cs35l41_hda *cs35l41, struct device *physd { const struct cs35l41_prop_model *model; - for (model = cs35l41_prop_model_table; model->hid > 0; model++) { + for (model = cs35l41_prop_model_table; model->hid; model++) { if (!strcmp(model->hid, hid) && (!model->ssid || (cs35l41->acpi_subsystem_id && -- 2.35.3

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[tiwai-sound:for-next 28/28] sound/pci/hda/cs35l41_hda_property.c:64:59: warning: ordered comparison of pointer with integer zero
by kernel test robot 16 Aug '23

16 Aug '23

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: ef4ba63f12b03532378395a8611f2f6e22ece67b commit: ef4ba63f12b03532378395a8611f2f6e22ece67b [28/28] ALSA: hda: cs35l41: Support systems with missing _DSD properties config: loongarch-allyesconfig (https://download.01.org/0day-ci/archive/20230816/202308160506.8lCEeFDG-lkp@…) compiler: loongarch64-linux-gcc (GCC) 12.3.0 reproduce: (https://download.01.org/0day-ci/archive/20230816/202308160506.8lCEeFDG-lkp@…) 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/202308160506.8lCEeFDG-lkp@intel.com/ All warnings (new ones prefixed by >>): sound/pci/hda/cs35l41_hda_property.c: In function 'cs35l41_add_dsd_properties': >> sound/pci/hda/cs35l41_hda_property.c:64:59: warning: ordered comparison of pointer with integer zero [-Wextra] 64 | for (model = cs35l41_prop_model_table; model->hid > 0; model++) { | ^ vim +64 sound/pci/hda/cs35l41_hda_property.c 58 59 int cs35l41_add_dsd_properties(struct cs35l41_hda *cs35l41, struct device *physdev, int id, 60 const char *hid) 61 { 62 const struct cs35l41_prop_model *model; 63 > 64 for (model = cs35l41_prop_model_table; model->hid > 0; model++) { -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

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[PATCH 0/2] ASoC: cs35l56: Don't patch firmware that is already patched
by Richard Fitzgerald 15 Aug '23

15 Aug '23

Use the FIRMWARE_MISSING flag in the CS35L56 to determine whether it is safe to download a firmware patch. Richard Fitzgerald (2): ASoC: wm_adsp: Support powering-up DSP without trying to load firmware ASoC: cs35l56: Don't overwrite a patched firmware sound/soc/codecs/cs35l56.c | 19 ++++++++++++++++--- sound/soc/codecs/wm_adsp.c | 16 +++++++++------- sound/soc/codecs/wm_adsp.h | 2 +- 3 files changed, 26 insertions(+), 11 deletions(-) -- 2.30.2

2 3

[PATCH 0/2] ASoC: SOF: topology: simplify kcontrol names with token
by Pierre-Louis Bossart 15 Aug '23

15 Aug '23

The use of the widget name as a prefix for the kcontrol name is quite useful in the case of multiple pipelines going to the same endpoint, but it's overkill in simpler cases. This patchset extends the existing DAPM code to drop the widget name prefix and make the kcontrol names simpler when there's no possible ambiguity, e.g. "gain.2.1 Main Playback Volume" becomes just "Main Playback Volume". Jyri Sarha (2): ASoC: dapm: Add a flag for not having widget name in kcontrol name ASoC: SOF: topology: Add a token for dropping widget name in kcontrol name include/sound/soc-dapm.h | 1 + include/uapi/sound/sof/tokens.h | 6 +++++- sound/soc/soc-dapm.c | 2 ++ sound/soc/sof/topology.c | 22 ++++++++++++++++++++++ 4 files changed, 30 insertions(+), 1 deletion(-) -- 2.39.2

2 3

[PATCH] ASoC: SOF: Intel: Refactor code for HDA stream creation
by Pierre-Louis Bossart 15 Aug '23

15 Aug '23

From: Chao Song <chao.song(a)linux.intel.com> Existing HDA stream creation is split into two for loops for capture and playback, but most of the code in the two for loops are duplicated. This patch refactors HDA stream creation with a single for loop, thus remove code duplication. No functional change in this patch. Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Chao Song <chao.song(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/intel/hda-stream.c | 68 ++++---------------------------- 1 file changed, 8 insertions(+), 60 deletions(-) diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index b13acb959653..0b0087abcc50 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -869,8 +869,8 @@ int hda_dsp_stream_init(struct snd_sof_dev *sdev) return -ENOMEM; } - /* create capture streams */ - for (i = 0; i < num_capture; i++) { + /* create capture and playback streams */ + for (i = 0; i < num_total; i++) { struct sof_intel_hda_stream *hda_stream; hda_stream = devm_kzalloc(sdev->dev, sizeof(*hda_stream), @@ -909,69 +909,17 @@ int hda_dsp_stream_init(struct snd_sof_dev *sdev) hstream->index = i; sd_offset = SOF_STREAM_SD_OFFSET(hstream); hstream->sd_addr = sdev->bar[HDA_DSP_HDA_BAR] + sd_offset; - hstream->stream_tag = i + 1; hstream->opened = false; hstream->running = false; - hstream->direction = SNDRV_PCM_STREAM_CAPTURE; - /* memory alloc for stream BDL */ - ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, &pci->dev, - HDA_DSP_BDL_SIZE, &hstream->bdl); - if (ret < 0) { - dev_err(sdev->dev, "error: stream bdl dma alloc failed\n"); - return -ENOMEM; - } - hstream->posbuf = (__le32 *)(bus->posbuf.area + - (hstream->index) * 8); - - list_add_tail(&hstream->list, &bus->stream_list); - } - - /* create playback streams */ - for (i = num_capture; i < num_total; i++) { - struct sof_intel_hda_stream *hda_stream; - - hda_stream = devm_kzalloc(sdev->dev, sizeof(*hda_stream), - GFP_KERNEL); - if (!hda_stream) - return -ENOMEM; - - hda_stream->sdev = sdev; - - hext_stream = &hda_stream->hext_stream; - - if (sdev->bar[HDA_DSP_PP_BAR]) { - hext_stream->pphc_addr = sdev->bar[HDA_DSP_PP_BAR] + - SOF_HDA_PPHC_BASE + SOF_HDA_PPHC_INTERVAL * i; - - hext_stream->pplc_addr = sdev->bar[HDA_DSP_PP_BAR] + - SOF_HDA_PPLC_BASE + SOF_HDA_PPLC_MULTI * num_total + - SOF_HDA_PPLC_INTERVAL * i; + if (i < num_capture) { + hstream->stream_tag = i + 1; + hstream->direction = SNDRV_PCM_STREAM_CAPTURE; + } else { + hstream->stream_tag = i - num_capture + 1; + hstream->direction = SNDRV_PCM_STREAM_PLAYBACK; } - hstream = &hext_stream->hstream; - - /* do we support SPIB */ - if (sdev->bar[HDA_DSP_SPIB_BAR]) { - hstream->spib_addr = sdev->bar[HDA_DSP_SPIB_BAR] + - SOF_HDA_SPIB_BASE + SOF_HDA_SPIB_INTERVAL * i + - SOF_HDA_SPIB_SPIB; - - hstream->fifo_addr = sdev->bar[HDA_DSP_SPIB_BAR] + - SOF_HDA_SPIB_BASE + SOF_HDA_SPIB_INTERVAL * i + - SOF_HDA_SPIB_MAXFIFO; - } - - hstream->bus = bus; - hstream->sd_int_sta_mask = 1 << i; - hstream->index = i; - sd_offset = SOF_STREAM_SD_OFFSET(hstream); - hstream->sd_addr = sdev->bar[HDA_DSP_HDA_BAR] + sd_offset; - hstream->stream_tag = i - num_capture + 1; - hstream->opened = false; - hstream->running = false; - hstream->direction = SNDRV_PCM_STREAM_PLAYBACK; - /* mem alloc for stream BDL */ ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, &pci->dev, HDA_DSP_BDL_SIZE, &hstream->bdl); -- 2.39.2

2 1

[PATCH 0/4] ASoC: Intel: RPL/MTL machine updates for 6.6
by Pierre-Louis Bossart 15 Aug '23

15 Aug '23

4 commits with the match tables for RaptorLake and MeteorLake devices. Balamurugan C (3): ASoC: Intel: soc-acpi: Add entry for sof_es8336 in RPL match table. ASoC: Intel: soc-acpi: Add entry for HDMI_In capture support in RPL match table ASoC: Intel: RPL: Add entry for HDMI-In capture support on non-I2S codec boards. Chao Song (1): ASoC: Intel: soc-acpi: add support for Dell SKU0C87 devices sound/soc/intel/boards/sof_es8336.c | 10 ++++ sound/soc/intel/boards/sof_ssp_amp.c | 9 ++++ .../intel/common/soc-acpi-intel-mtl-match.c | 52 +++++++++++++++++++ .../intel/common/soc-acpi-intel-rpl-match.c | 30 +++++++++++ 4 files changed, 101 insertions(+) -- 2.39.2

2 5

[PATCH] ASoC: SOF: remove duplicate code due to merge
by Pierre-Louis Bossart 15 Aug '23

15 Aug '23

Commit 81113108491e ("ASoC: Merge up fixes") added the same code twice, remove the extra call. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/ipc4-topology.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index b43663822613..633f6040d712 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -1734,9 +1734,6 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, /* update pipeline memory usage */ sof_ipc4_update_resource_usage(sdev, swidget, &copier_data->base_config); - /* update pipeline memory usage */ - sof_ipc4_update_resource_usage(sdev, swidget, &copier_data->base_config); - /* copy IPC data */ memcpy(*ipc_config_data, (void *)copier_data, sizeof(*copier_data)); if (gtw_cfg_config_length) -- 2.39.2

3 2

[PATCH -next 0/2] Use devm_kmemdup to replace devm_kmalloc + memcpy
by Li Zetao 15 Aug '23

15 Aug '23

This patch set use the helper function devm_kmemdup() to replace devm_kmalloc + memcpy, which is the same as implementing the function separately. Li Zetao (2): ASoC: tas5805m: Use devm_kmemdup to replace devm_kmalloc + memcpy ASoC: SOF: ipc3: Use devm_kmemdup to replace devm_kmalloc + memcpy sound/soc/codecs/tas5805m.c | 3 +-- sound/soc/sof/ipc3.c | 5 +---- 2 files changed, 2 insertions(+), 6 deletions(-) -- 2.34.1

4 5

[GIT PULL] ASoC fixes for v6.5-rc6
by Mark Brown 15 Aug '23

15 Aug '23

The following changes since commit 5d0c230f1de8c7515b6567d9afba1f196fb4e2f4: Linux 6.5-rc4 (2023-07-30 13:23:47 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.5-rc6 for you to fetch changes up to 37aba3190891d4de189bd5192ee95220e295f34d: ASoC: rt1308-sdw: fix random louder sound (2023-08-13 18:16:32 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.5 A fairly large collection of fixes here, mostly SOF and Intel related. The one core fix is Hans' change which reduces the log spam when working out new use cases for DPCM. ---------------------------------------------------------------- Bard Liao (1): ASoC: max98363: don't return on success reading revision ID Daniel Baluta (1): ASoC: fsl: micfil: Use dual license micfil code Hans de Goede (1): ASoC: lower "no backend DAIs enabled for ... Port" log severity Jerome Brunet (1): ASoC: meson: axg-tdm-formatter: fix channel slot allocation Peter Ujfalusi (1): ASoC: SOF: ipc4-topology: Update the basecfg for copier earlier Pierre-Louis Bossart (1): ASoC: Intel: sof-sdw: update jack detection quirk for LunarLake RVP Ranjani Sridharan (1): ASoC: SOF: intel: hda: Clean up link DMA for IPC3 during stop Shuming Fan (1): ASoC: rt1308-sdw: fix random louder sound Xia Fukun (1): ASoC: SOF: Fix incorrect use of sizeof in sof_ipc3_do_rx_work() Zhang Shurong (1): ASoC: rt5665: add missed regulator_bulk_disable jairaj-arava (1): ASoC: Intel: sof-sdw-cs42142: fix for codec button mapping sound/soc/codecs/max98363.c | 9 ++++--- sound/soc/codecs/rt1308-sdw.c | 13 +++++++++- sound/soc/codecs/rt5665.c | 2 ++ sound/soc/fsl/fsl_micfil.c | 4 +-- sound/soc/fsl/fsl_micfil.h | 2 +- sound/soc/intel/boards/sof_sdw.c | 2 +- sound/soc/intel/boards/sof_sdw_cs42l42.c | 6 ++--- sound/soc/meson/axg-tdm-formatter.c | 42 ++++++++++++++++++++------------ sound/soc/soc-pcm.c | 8 ++++-- sound/soc/sof/intel/hda-dai-ops.c | 11 ++++++++- sound/soc/sof/intel/hda-dai.c | 5 ++-- sound/soc/sof/intel/hda.h | 2 ++ sound/soc/sof/ipc3.c | 2 +- sound/soc/sof/ipc4-topology.c | 6 ++--- 14 files changed, 76 insertions(+), 38 deletions(-)

2 1

[PATCH] ASoC: soc-jack: calling snd_soc_jack_report causes a null pointer access
by mnlife 15 Aug '23

15 Aug '23

When snd_soc_card_jack_new is not called or the call fails, calling this function causes a null pointer access Signed-off-by: mnlife <mnlife(a)126.com> --- sound/soc/soc-jack.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index f951acb2ce362..b2cc13b9c77b7 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -37,7 +37,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) struct snd_soc_jack_pin *pin; unsigned int sync = 0; - if (!jack) + if (!jack || !jack->jack) return; trace_snd_soc_jack_report(jack, mask, status); -- 2.25.1

2 1

[PATCH v1] ASoC: Intel: Add rpl_nau8318_8825 driver
by Ajye Huang 15 Aug '23

15 Aug '23

Boards were using this in older kernels before adl and rpl ids were split. Add this back to maintain support. Signed-off-by: Ajye Huang <ajye_huang(a)compal.corp-partner.google.com> --- sound/soc/intel/boards/sof_nau8825.c | 10 ++++++++++ sound/soc/intel/common/soc-acpi-intel-rpl-match.c | 12 ++++++++++++ 2 files changed, 22 insertions(+) diff --git a/sound/soc/intel/boards/sof_nau8825.c b/sound/soc/intel/boards/sof_nau8825.c index 4fc6e1c6aef3..46b7ecf6f9f1 100644 --- a/sound/soc/intel/boards/sof_nau8825.c +++ b/sound/soc/intel/boards/sof_nau8825.c @@ -684,6 +684,16 @@ static const struct platform_device_id board_ids[] = { SOF_BT_OFFLOAD_SSP(2) | SOF_SSP_BT_OFFLOAD_PRESENT), }, + { + .name = "rpl_nau8318_8825", + .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_NAU8318_SPEAKER_AMP_PRESENT | + SOF_NAU8825_SSP_AMP(1) | + SOF_NAU8825_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c index 4eefdb2dd45c..1dd699181765 100644 --- a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c @@ -351,6 +351,11 @@ static const struct snd_soc_acpi_codecs rpl_rt1019p_amp = { .codecs = {"RTL1019"} }; +static const struct snd_soc_acpi_codecs rpl_nau8318_amp = { + .num_codecs = 1, + .codecs = {"NVTN2012"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_machines[] = { { .comp_ids = &rpl_rt5682_hp, @@ -373,6 +378,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_machines[] = { .quirk_data = &rpl_max98373_amp, .sof_tplg_filename = "sof-rpl-max98373-nau8825.tplg", }, + { + .id = "10508825", + .drv_name = "rpl_nau8318_8825", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &rpl_nau8318_amp, + .sof_tplg_filename = "sof-rpl-nau8318-nau8825.tplg", + }, { .comp_ids = &rpl_rt5682_hp, .drv_name = "rpl_rt1019_rt5682", -- 2.25.1

3 3

[PATCH 00/25] ALSA: Generic PCM copy ops using iov_iter
by Takashi Iwai 15 Aug '23

15 Aug '23

Hi, this is a revised patch set for cleaning up the PCM copy ops using iov_iter to deal with kernel / user-space pointers consistently. The previous patch set was based on sockptr_t: https://lore.kernel.org/r/20230731154718.31048-1-tiwai@suse.de But this approach was NAK'ed as iov_iter is a preferred way for achieving the purpose: https://lore.kernel.org/r/87edkce118.wl-tiwai@suse.de The patch set starts with the missing export of import_ubuf() function, followed by the new copy_ops using iov_iter, extensions of some helpers and replacements of the existing code. It resulted in a good amount of code reduction. Takashi === Cc: Alexander Viro <viro(a)zeniv.linux.org.uk> Cc: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Cc: Andrey Utkin <andrey_utkin(a)fastmail.com> Cc: Anton Sviridenko <anton(a)corp.bluecherry.net> Cc: Arnaud Pouliquen <arnaud.pouliquen(a)foss.st.com> Cc: Banajit Goswami <bgoswami(a)quicinc.com> Cc: Bluecherry Maintainers <maintainers(a)bluecherrydvr.com> Cc: Claudiu Beznea <claudiu.beznea(a)microchip.com> Cc: Ismael Luceno <ismael(a)iodev.co.uk> Cc: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Mark Brown <broonie(a)kernel.org> Cc: Mauro Carvalho Chehab <mchehab(a)kernel.org> Cc: Oleksandr Andrushchenko <oleksandr_andrushchenko(a)epam.com> Cc: Olivier Moysan <olivier.moysan(a)foss.st.com> Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Cc: linux-media(a)vger.kernel.org Cc: xen-devel(a)lists.xenproject.org === Takashi Iwai (25): iov_iter: Export import_ubuf() ALSA: pcm: Add copy ops with iov_iter ALSA: core: Add memory copy helpers between iov_iter and iomem ALSA: dummy: Convert to generic PCM copy ops ALSA: gus: Convert to generic PCM copy ops ALSA: emu8000: Convert to generic PCM copy ops ALSA: es1938: Convert to generic PCM copy ops ALSA: korg1212: Convert to generic PCM copy ops ALSA: nm256: Convert to generic PCM copy ops ALSA: rme32: Convert to generic PCM copy ops ALSA: rme96: Convert to generic PCM copy ops ALSA: hdsp: Convert to generic PCM copy ops ALSA: rme9652: Convert to generic PCM copy ops ALSA: sh: Convert to generic PCM copy ops ALSA: xen: Convert to generic PCM copy ops ALSA: pcmtest: Update comment about PCM copy ops media: solo6x10: Convert to generic PCM copy ops ASoC: component: Add generic PCM copy ops ASoC: mediatek: Convert to generic PCM copy ops ASoC: qcom: Convert to generic PCM copy ops ASoC: dmaengine: Convert to generic PCM copy ops ASoC: dmaengine: Use iov_iter for process callback, too ALSA: doc: Update description for the new PCM copy ops ASoC: pcm: Drop obsoleted PCM copy_user ops ALSA: pcm: Drop obsoleted PCM copy_user and copy_kernel ops .../kernel-api/writing-an-alsa-driver.rst | 58 ++++------- drivers/media/pci/solo6x10/solo6x10-g723.c | 38 +------- include/sound/dmaengine_pcm.h | 2 +- include/sound/pcm.h | 13 ++- include/sound/soc-component.h | 14 +-- lib/iov_iter.c | 1 + sound/core/memory.c | 56 +++++++++-- sound/core/pcm_lib.c | 95 ++++++++++--------- sound/core/pcm_native.c | 2 +- sound/drivers/dummy.c | 12 +-- sound/drivers/pcmtest.c | 2 +- sound/isa/gus/gus_pcm.c | 23 +---- sound/isa/sb/emu8000_pcm.c | 74 ++++----------- sound/pci/es1938.c | 30 +----- sound/pci/korg1212/korg1212.c | 50 +++------- sound/pci/nm256/nm256.c | 42 ++------ sound/pci/rme32.c | 50 +++------- sound/pci/rme96.c | 42 ++------ sound/pci/rme9652/hdsp.c | 42 ++------ sound/pci/rme9652/rme9652.c | 46 ++------- sound/sh/sh_dac_audio.c | 25 +---- sound/soc/atmel/mchp-pdmc.c | 2 +- sound/soc/mediatek/common/mtk-btcvsd.c | 23 ++--- sound/soc/qcom/lpass-platform.c | 13 +-- sound/soc/soc-component.c | 10 +- sound/soc/soc-generic-dmaengine-pcm.c | 18 ++-- sound/soc/soc-pcm.c | 4 +- sound/soc/stm/stm32_sai_sub.c | 2 +- sound/xen/xen_snd_front_alsa.c | 55 ++--------- 29 files changed, 263 insertions(+), 581 deletions(-) -- 2.35.3

5 43

[PATCH v1 1/1] minmax: Deduplicate __unconst_integer_typeof()
by Andy Shevchenko 15 Aug '23

15 Aug '23

It appears that compiler_types.h already have an implementation of the __unconst_integer_typeof() called __unqual_scalar_typeof(). Use it instead of the copy. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- As the initial code is in Linux Next via ASoC tree, it's assumed to go via that tree as well. include/linux/minmax.h | 26 +++----------------------- 1 file changed, 3 insertions(+), 23 deletions(-) diff --git a/include/linux/minmax.h b/include/linux/minmax.h index 83aebc244cba..69bbe987fa87 100644 --- a/include/linux/minmax.h +++ b/include/linux/minmax.h @@ -2,6 +2,7 @@ #ifndef _LINUX_MINMAX_H #define _LINUX_MINMAX_H +#include <linux/compiler_types.h> #include <linux/const.h> #include <linux/types.h> @@ -134,27 +135,6 @@ */ #define max_t(type, x, y) __careful_cmp((type)(x), (type)(y), >) -/* - * Remove a const qualifier from integer types - * _Generic(foo, type-name: association, ..., default: association) performs a - * comparison against the foo type (not the qualified type). - * Do not use the const keyword in the type-name as it will not match the - * unqualified type of foo. - */ -#define __unconst_integer_type_cases(type) \ - unsigned type: (unsigned type)0, \ - signed type: (signed type)0 - -#define __unconst_integer_typeof(x) typeof( \ - _Generic((x), \ - char: (char)0, \ - __unconst_integer_type_cases(char), \ - __unconst_integer_type_cases(short), \ - __unconst_integer_type_cases(int), \ - __unconst_integer_type_cases(long), \ - __unconst_integer_type_cases(long long), \ - default: (x))) - /* * Do not check the array parameter using __must_be_array(). * In the following legit use-case where the "array" passed is a simple pointer, @@ -169,13 +149,13 @@ * 'int *buff' and 'int buff[N]' types. * * The array can be an array of const items. - * typeof() keeps the const qualifier. Use __unconst_integer_typeof() in order + * typeof() keeps the const qualifier. Use __unqual_scalar_typeof() in order * to discard the const qualifier for the __element variable. */ #define __minmax_array(op, array, len) ({ \ typeof(&(array)[0]) __array = (array); \ typeof(len) __len = (len); \ - __unconst_integer_typeof(__array[0]) __element = __array[--__len]; \ + __unqual_scalar_typeof(__array[0]) __element = __array[--__len];\ while (__len--) \ __element = op(__element, __array[__len]); \ __element; }) -- 2.40.0.1.gaa8946217a0b

2 1

[PATCH 1/4] ASoC: aw88261: Fix unitialized ret of aw88261_reg_update()
by Krzysztof Kozlowski 15 Aug '23

15 Aug '23

There is a branch in if() clause where aw88261_reg_update() could return uninitialized value. Pounted out by W=1 clang build: sound/soc/codecs/aw88261.c:651:7: error: variable 'ret' is used uninitialized whenever 'if' condition is false [-Werror,-Wsometimes-uninitialized] if (aw_dev->prof_cur != aw_dev->prof_index) { ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/aw88261.c:660:9: note: uninitialized use occurs here return ret; ^~~ sound/soc/codecs/aw88261.c:651:3: note: remove the 'if' if its condition is always true if (aw_dev->prof_cur != aw_dev->prof_index) { ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/codecs/aw88261.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/aw88261.c b/sound/soc/codecs/aw88261.c index 82923b454dd4..fddba2f2de41 100644 --- a/sound/soc/codecs/aw88261.c +++ b/sound/soc/codecs/aw88261.c @@ -636,7 +636,7 @@ static int aw88261_dev_stop(struct aw_device *aw_dev) static int aw88261_reg_update(struct aw88261 *aw88261, bool force) { struct aw_device *aw_dev = aw88261->aw_pa; - int ret; + int ret = 0; if (force) { ret = regmap_write(aw_dev->regmap, -- 2.34.1

3 9

[PATCH] ASoC: rt5677: fix -Wvoid-pointer-to-enum-cast warning
by Justin Stitt 15 Aug '23

15 Aug '23

`match_id->data` is a void* and as such is being truncated when cast to `enum rt5677_type` which is only int-width. There is likely no data loss occurring, though, as `enum rt5677_type` consists of only two fields from 0 to 1 wherein obviously no data loss happens from pointer-width -> int-width. Link: https://github.com/ClangBuiltLinux/linux/issues/1910 Reported-by: Nathan Chancellor <nathan(a)kernel.org> Signed-off-by: Justin Stitt <justinstitt(a)google.com> --- Note: I'm not sure if `uintptr_t` is the correct solution here as I've also seen a cast to `unsigned long` suffice. Any thoughts on the semantically correct option? --- sound/soc/codecs/rt5677.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index ad14d18860fc..278cb0b265e5 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -5578,7 +5578,7 @@ static int rt5677_i2c_probe(struct i2c_client *i2c) match_id = of_match_device(rt5677_of_match, &i2c->dev); if (match_id) - rt5677->type = (enum rt5677_type)match_id->data; + rt5677->type = (uintptr_t)match_id->data; } else if (ACPI_HANDLE(&i2c->dev)) { const struct acpi_device_id *acpi_id; --- base-commit: 2ccdd1b13c591d306f0401d98dedc4bdcd02b421 change-id: 20230814-void-sound-soc-codecs-rt5677-dfc041c1a734 Best regards, -- Justin Stitt <justinstitt(a)google.com>

3 2

[PATCH] ALSA: hda: intel-dsp-cfg: Add Chromebook quirk to ADL/RPL
by Brady Norander 15 Aug '23

15 Aug '23

AlderLake and RaptorLake Chromebooks currently use the HDA driver by default. Add a quirk to use the SOF driver on these platforms, which is needed for functional internal audio. Signed-off-by: Brady Norander <bradynorander(a)gmail.com> --- sound/hda/intel-dsp-config.c | 151 ++++++++++++++++++++++++++++++++++- 1 file changed, 147 insertions(+), 4 deletions(-) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index dcf2453138a5..5fb5abb76679 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -167,10 +167,10 @@ static const struct config_entry config_table[] = { #endif /* - * CoffeeLake, CannonLake, CometLake, IceLake, TigerLake use legacy - * HDAudio driver except for Google Chromebooks and when DMICs are - * present. Two cases are required since Coreboot does not expose NHLT - * tables. + * CoffeeLake, CannonLake, CometLake, IceLake, TigerLake, AlderLake, + * RaptorLake use legacy HDAudio driver except for Google Chromebooks + * and when DMICs are present. Two cases are required since Coreboot + * does not expose NHLT tables. * * When the Chromebook quirk is not present, it's based on information * that no such device exists. When the quirk is present, it could be @@ -400,14 +400,53 @@ static const struct config_entry config_table[] = { /* Alder Lake / Raptor Lake */ #if IS_ENABLED(CONFIG_SND_SOC_SOF_ALDERLAKE) + { + .flags = FLAG_SOF, + .device = PCI_DEVICE_ID_INTEL_HDA_ADL_S, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Google Chromebooks", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + } + }, + {} + } + }, { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = PCI_DEVICE_ID_INTEL_HDA_ADL_S, }, + { + .flags = FLAG_SOF, + .device = PCI_DEVICE_ID_INTEL_HDA_RPL_S, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Google Chromebooks", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + } + }, + {} + } + }, { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = PCI_DEVICE_ID_INTEL_HDA_RPL_S, }, + { + .flags = FLAG_SOF, + .device = PCI_DEVICE_ID_INTEL_HDA_ADL_P, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Google Chromebooks", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + } + }, + {} + } + }, { .flags = FLAG_SOF, .device = PCI_DEVICE_ID_INTEL_HDA_ADL_P, @@ -417,10 +456,36 @@ static const struct config_entry config_table[] = { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = PCI_DEVICE_ID_INTEL_HDA_ADL_P, }, + { + .flags = FLAG_SOF, + .device = PCI_DEVICE_ID_INTEL_HDA_ADL_PX, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Google Chromebooks", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + } + }, + {} + } + }, { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = PCI_DEVICE_ID_INTEL_HDA_ADL_PX, }, + { + .flags = FLAG_SOF, + .device = PCI_DEVICE_ID_INTEL_HDA_ADL_PS, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Google Chromebooks", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + } + }, + {} + } + }, { .flags = FLAG_SOF, .device = PCI_DEVICE_ID_INTEL_HDA_ADL_PS, @@ -430,26 +495,104 @@ static const struct config_entry config_table[] = { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = PCI_DEVICE_ID_INTEL_HDA_ADL_PS, }, + { + .flags = FLAG_SOF, + .device = PCI_DEVICE_ID_INTEL_HDA_ADL_M, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Google Chromebooks", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + } + }, + {} + } + }, { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = PCI_DEVICE_ID_INTEL_HDA_ADL_M, }, + { + .flags = FLAG_SOF, + .device = PCI_DEVICE_ID_INTEL_HDA_ADL_N, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Google Chromebooks", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + } + }, + {} + } + }, { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = PCI_DEVICE_ID_INTEL_HDA_ADL_N, }, + { + .flags = FLAG_SOF, + .device = PCI_DEVICE_ID_INTEL_HDA_RPL_P_0, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Google Chromebooks", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + } + }, + {} + } + }, { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = PCI_DEVICE_ID_INTEL_HDA_RPL_P_0, }, + { + .flags = FLAG_SOF, + .device = PCI_DEVICE_ID_INTEL_HDA_RPL_P_1, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Google Chromebooks", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + } + }, + {} + } + }, { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = PCI_DEVICE_ID_INTEL_HDA_RPL_P_1, }, + { + .flags = FLAG_SOF, + .device = PCI_DEVICE_ID_INTEL_HDA_RPL_M, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Google Chromebooks", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + } + }, + {} + } + }, { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = PCI_DEVICE_ID_INTEL_HDA_RPL_M, }, + { + .flags = FLAG_SOF, + .device = PCI_DEVICE_ID_INTEL_HDA_RPL_PX, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Google Chromebooks", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + } + }, + {} + } + }, { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = PCI_DEVICE_ID_INTEL_HDA_RPL_PX, -- 2.41.0

4 7

[PATCH v2 00/39] ASoC: merge DAI call back functions into ops
by Kuninori Morimoto 15 Aug '23

15 Aug '23

Hi Mark snd_soc_dai_driver has .ops for call back functions (A), but it also has other call back functions (B). It is duplicated and confusable. struct snd_soc_dai_driver { ... ^ int (*probe)(...); | int (*remove)(...); (B) int (*compress_new)(...); | int (*pcm_new)(...); v ... (A) const struct snd_soc_dai_ops *ops; ... } This patch merges (B) into (A). -------- 8< -------- 8< -------- 8< -------- 8< -------- 8< -------- v1 -> v2 - add Reviewed-by on each patches. - fixup git-log typo (thesse -> there) - add missing patch for au1x Link: https://lore.kernel.org/r/87a5vauuzg.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (39): ASoC: soc-dai.h: merge DAI call back functions into ops ASoC: ti: merge DAI call back functions into ops ASoC: adi: merge DAI call back functions into ops ASoC: adi: merge DAI call back functions into ops ASoC: amd: merge DAI call back functions into ops ASoC: dwc: merge DAI call back functions into ops ASoC: pxa: merge DAI call back functions into ops ASoC: bcm: merge DAI call back functions into ops ASoC: fsl: merge DAI call back functions into ops ASoC: img: merge DAI call back functions into ops ASoC: sof: merge DAI call back functions into ops ASoC: sti: merge DAI call back functions into ops ASoC: stm: merge DAI call back functions into ops ASoC: pxa: merge DAI call back functions into ops ASoC: rsnd: merge DAI call back functions into ops ASoC: qcom: merge DAI call back functions into ops ASoC: au1x: merge DAI call back functions into ops ASoC: ux500: merge DAI call back functions into ops ASoC: sunxi: merge DAI call back functions into ops ASoC: tegra: merge DAI call back functions into ops ASoC: atmel: merge DAI call back functions into ops ASoC: intel: merge DAI call back functions into ops ASoC: meson: merge DAI call back functions into ops ASoC: jz4740: merge DAI call back functions into ops ASoC: cirrus: merge DAI call back functions into ops ASoC: drm/vc4: merge DAI call back functions into ops ASoC: samsung: merge DAI call back functions into ops ASoC: mediatek: merge DAI call back functions into ops ASoC: rockchip: merge DAI call back functions into ops ASoC: uniphier: merge DAI call back functions into ops ASoC: loongson: merge DAI call back functions into ops ASoC: starfive: merge DAI call back functions into ops ASoC: hisilicon: merge DAI call back functions into ops ASoC: codecs/wm*: merge DAI call back functions into ops ASoC: soc-topology: merge DAI call back functions into ops ASoC: codecs/cs47lxx: merge DAI call back functions into ops ASoC: codecs/cx2072x: merge DAI call back functions into ops ASoC: codecs/hdmi-codec: merge DAI call back functions into ops ASoC: soc-dai.h: remove unused call back functions drivers/gpu/drm/vc4/vc4_hdmi.c | 6 +- include/sound/soc-dai.h | 26 +-- sound/soc/adi/axi-i2s.c | 2 +- sound/soc/adi/axi-spdif.c | 2 +- sound/soc/amd/acp/acp-i2s.c | 24 +-- sound/soc/amd/acp/acp-rembrandt.c | 3 - sound/soc/amd/acp/acp-renoir.c | 2 - sound/soc/amd/acp/amd.h | 1 - sound/soc/atmel/atmel-i2s.c | 16 +- sound/soc/atmel/mchp-i2s-mcc.c | 24 +-- sound/soc/atmel/mchp-pdmc.c | 18 +-- sound/soc/atmel/mchp-spdifrx.c | 14 +- sound/soc/atmel/mchp-spdiftx.c | 18 +-- sound/soc/au1x/ac97c.c | 10 +- sound/soc/au1x/psc-ac97.c | 2 +- sound/soc/bcm/bcm2835-i2s.c | 24 +-- sound/soc/cirrus/ep93xx-i2s.c | 2 +- sound/soc/codecs/cs47l15.c | 6 +- sound/soc/codecs/cs47l24.c | 8 +- sound/soc/codecs/cs47l35.c | 8 +- sound/soc/codecs/cs47l85.c | 8 +- sound/soc/codecs/cs47l90.c | 8 +- sound/soc/codecs/cs47l92.c | 6 +- sound/soc/codecs/cx2072x.c | 11 +- sound/soc/codecs/hdmi-codec.c | 44 ++--- sound/soc/codecs/wm2200.c | 12 +- sound/soc/codecs/wm5102.c | 6 +- sound/soc/codecs/wm5110.c | 8 +- sound/soc/codecs/wm8994.c | 2 +- sound/soc/dwc/dwc-i2s.c | 18 +-- sound/soc/fsl/fsl_asrc.c | 16 +- sound/soc/fsl/fsl_aud2htx.c | 10 +- sound/soc/fsl/fsl_easrc.c | 16 +- sound/soc/fsl/fsl_esai.c | 20 +-- sound/soc/fsl/fsl_micfil.c | 14 +- sound/soc/fsl/fsl_sai.c | 24 +-- sound/soc/fsl/fsl_spdif.c | 17 +- sound/soc/fsl/fsl_ssi.c | 3 +- sound/soc/fsl/fsl_xcvr.c | 16 +- sound/soc/generic/audio-graph-card.c | 2 +- sound/soc/hisilicon/hi6210-i2s.c | 2 +- sound/soc/img/img-i2s-in.c | 14 +- sound/soc/img/img-i2s-out.c | 14 +- sound/soc/img/img-parallel-out.c | 14 +- sound/soc/img/img-spdif-in.c | 12 +- sound/soc/img/img-spdif-out.c | 12 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- sound/soc/intel/avs/probes.c | 10 +- sound/soc/intel/catpt/pcm.c | 8 +- sound/soc/intel/keembay/kmb_platform.c | 4 +- sound/soc/jz4740/jz4740-i2s.c | 3 +- sound/soc/loongson/loongson_i2s.c | 16 +- sound/soc/mediatek/mt8195/mt8195-dai-etdm.c | 56 ++++--- sound/soc/meson/aiu-acodec-ctrl.c | 4 +- sound/soc/meson/aiu-codec-ctrl.c | 4 +- sound/soc/meson/aiu-fifo-i2s.c | 3 + sound/soc/meson/aiu-fifo-spdif.c | 3 + sound/soc/meson/aiu.c | 6 - sound/soc/meson/axg-frddr.c | 4 +- sound/soc/meson/axg-pdm.c | 18 +-- sound/soc/meson/axg-spdifin.c | 4 +- sound/soc/meson/axg-tdm-interface.c | 6 +- sound/soc/meson/axg-toddr.c | 4 +- sound/soc/meson/g12a-toacodec.c | 4 +- sound/soc/meson/g12a-tohdmitx.c | 4 +- sound/soc/pxa/mmp-sspa.c | 2 +- sound/soc/pxa/pxa-ssp.c | 4 +- sound/soc/qcom/lpass-apq8016.c | 4 - sound/soc/qcom/lpass-cpu.c | 40 +++-- sound/soc/qcom/lpass-ipq806x.c | 1 - sound/soc/qcom/lpass-sc7180.c | 5 +- sound/soc/qcom/lpass-sc7280.c | 2 - sound/soc/qcom/lpass.h | 4 +- sound/soc/qcom/qdsp6/q6afe-dai.c | 88 +++++----- sound/soc/qcom/qdsp6/q6asm-dai.c | 6 +- sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 3 - sound/soc/rockchip/rockchip_i2s.c | 2 +- sound/soc/rockchip/rockchip_i2s_tdm.c | 2 +- sound/soc/rockchip/rockchip_pdm.c | 2 +- sound/soc/rockchip/rockchip_spdif.c | 2 +- sound/soc/samsung/i2s.c | 5 +- sound/soc/samsung/pcm.c | 18 +-- sound/soc/sh/rcar/core.c | 27 ++-- sound/soc/soc-dai.c | 44 ++--- sound/soc/soc-topology.c | 6 +- sound/soc/sof/sof-client-probes.c | 6 +- sound/soc/starfive/jh7110_tdm.c | 16 +- sound/soc/sti/sti_uniperif.c | 6 +- sound/soc/stm/stm32_i2s.c | 2 +- sound/soc/stm/stm32_sai_sub.c | 18 ++- sound/soc/stm/stm32_spdifrx.c | 2 +- sound/soc/sunxi/sun4i-codec.c | 6 +- sound/soc/sunxi/sun4i-i2s.c | 18 +-- sound/soc/sunxi/sun4i-spdif.c | 2 +- sound/soc/sunxi/sun50i-dmic.c | 2 +- sound/soc/tegra/tegra20_ac97.c | 10 +- sound/soc/tegra/tegra20_i2s.c | 2 +- sound/soc/tegra/tegra20_spdif.c | 2 +- sound/soc/tegra/tegra210_admaif.c | 12 +- sound/soc/tegra/tegra30_i2s.c | 2 +- sound/soc/ti/davinci-i2s.c | 22 +-- sound/soc/ti/davinci-mcasp.c | 27 ++-- sound/soc/ti/omap-dmic.c | 22 +-- sound/soc/ti/omap-mcbsp.c | 28 ++-- sound/soc/ti/omap-mcpdm.c | 22 +-- sound/soc/uniphier/aio-cpu.c | 161 +++++++++++++++---- sound/soc/uniphier/aio-ld11.c | 62 ++----- sound/soc/uniphier/aio-pxs2.c | 55 +------ sound/soc/uniphier/aio.h | 10 +- sound/soc/ux500/ux500_msp_dai.c | 2 +- 110 files changed, 798 insertions(+), 694 deletions(-) -- 2.25.1

5 44

[PATCH] ASoC: ak4613: Simplify probe()
by Biju Das 15 Aug '23

15 Aug '23

Simpilfy probe() by replacing of_device_get_match_data() and id lookup for retrieving match data by i2c_get_match_data(). While at it, drop unused local variable np from probe(). Signed-off-by: Biju Das <biju.das.jz(a)bp.renesas.com> --- sound/soc/codecs/ak4613.c | 11 +---------- 1 file changed, 1 insertion(+), 10 deletions(-) diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index ad56caec9dac..619a817ee91c 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -880,20 +880,11 @@ static void ak4613_parse_of(struct ak4613_priv *priv, static int ak4613_i2c_probe(struct i2c_client *i2c) { struct device *dev = &i2c->dev; - struct device_node *np = dev->of_node; const struct regmap_config *regmap_cfg; struct regmap *regmap; struct ak4613_priv *priv; - regmap_cfg = NULL; - if (np) - regmap_cfg = of_device_get_match_data(dev); - else { - const struct i2c_device_id *id = - i2c_match_id(ak4613_i2c_id, i2c); - regmap_cfg = (const struct regmap_config *)id->driver_data; - } - + regmap_cfg = i2c_get_match_data(i2c); if (!regmap_cfg) return -EINVAL; -- 2.25.1

2 1

[PATCH] ASoC: SOF: Intel: fix hda_sdw_check_wakeen_irq()
by Arnd Bergmann 14 Aug '23

14 Aug '23

From: Arnd Bergmann <arnd(a)arndb.de> This function cannot work, as the 'chip' variable is never initialized: sound/soc/sof/intel/hda.c:423:6: error: variable 'chip' is uninitialized when used here [-Werror,-Wuninitialized] if (chip && chip->check_sdw_wakeen_irq) ^~~~ Set it the same way that other functions in this file do. Fixes: 9362ab78f175d ("ASoC: SOF: Intel: add abstraction for SoundWire wake-ups") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- There hasn't been a new linux-next in a few days, so there is a good chance someone else already fixed this in the meantime. Sending out my fix in case that hasn't happened yet. --- sound/soc/sof/intel/hda.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 04c748a72b137..6c9c7b390cf53 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -415,7 +415,7 @@ bool hda_sdw_check_wakeen_irq_common(struct snd_sof_dev *sdev) static bool hda_sdw_check_wakeen_irq(struct snd_sof_dev *sdev) { u32 interface_mask = hda_get_interface_mask(sdev); - const struct sof_intel_dsp_desc *chip; + const struct sof_intel_dsp_desc *chip = get_chip_info(sdev->pdata); if (!(interface_mask & BIT(SOF_DAI_INTEL_ALH))) return false; -- 2.39.2

2 1

[PATCH] ALSA: hda: Zhaoxin: Add HDAC PCI IDs and HDMI Codec Vendor IDs
by Tony W Wang-oc 14 Aug '23

14 Aug '23

Add newer HD Audio PCI IDs, and HDMI codec vendor IDs for Zhaoxin. The quirk of AZX_DCAPS_RIRB_PRE_DELAY implies that write response from codec to memory needs delay when synchronized to memory after interrupt happened. For KX-5000 needs raising the priority of write cycles additionally, so add specific register setting codes for it. Because Zhaoxin hardware limitation, set BDL position to 128 to increase interrupt interval. Because Zhaoxin HDMI codec need driver to do actual clean-ups for the linked codec when switch from one codec to another, set no_sticky_stream for Zhaoxin HDMI codec too. Signed-off-by: Tony W Wang-oc <TonyWWang-oc(a)zhaoxin.com> --- sound/pci/hda/hda_controller.c | 17 +++++++++++- sound/pci/hda/hda_controller.h | 2 ++ sound/pci/hda/hda_intel.c | 51 +++++++++++++++++++++++++++++++++- sound/pci/hda/patch_hdmi.c | 13 +++++++++ 4 files changed, 81 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 406779625fb5..7b7a7735c482 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1044,6 +1044,16 @@ void azx_stop_chip(struct azx *chip) } EXPORT_SYMBOL_GPL(azx_stop_chip); +static void azx_rirb_zxdelay(struct azx *chip, int enable) +{ + if (chip->remap_diu_addr) { + if (!enable) + writel(0x0, (char *)chip->remap_diu_addr + 0x490a8); + else + writel(0x1000000, (char *)chip->remap_diu_addr + 0x490a8); + } +} + /* * interrupt handler */ @@ -1103,9 +1113,14 @@ irqreturn_t azx_interrupt(int irq, void *dev_id) azx_writeb(chip, RIRBSTS, RIRB_INT_MASK); active = true; if (status & RIRB_INT_RESPONSE) { - if (chip->driver_caps & AZX_DCAPS_CTX_WORKAROUND) + if ((chip->driver_caps & AZX_DCAPS_CTX_WORKAROUND) || + (chip->driver_caps & AZX_DCAPS_RIRB_PRE_DELAY)) { + azx_rirb_zxdelay(chip, 1); udelay(80); + } snd_hdac_bus_update_rirb(bus); + if (chip->driver_caps & AZX_DCAPS_RIRB_PRE_DELAY) + azx_rirb_zxdelay(chip, 0); } } } while (active && ++repeat < 10); diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index 8556031bcd68..ba51e33329c1 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -45,6 +45,7 @@ #define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28) /* CORBRP clears itself after reset */ #define AZX_DCAPS_NO_MSI64 (1 << 29) /* Stick to 32-bit MSIs */ #define AZX_DCAPS_SEPARATE_STREAM_TAG (1 << 30) /* capture and playback use separate stream tag */ +#define AZX_DCAPS_RIRB_PRE_DELAY (1 << 31) /* RIRB need delay */ enum { AZX_SNOOP_TYPE_NONE, @@ -145,6 +146,7 @@ struct azx { /* GTS present */ unsigned int gts_present:1; + void __iomem *remap_diu_addr; #ifdef CONFIG_SND_HDA_DSP_LOADER struct azx_dev saved_azx_dev; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 176567f0d0e0..d5dce7e7ef4a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -238,6 +238,7 @@ enum { AZX_DRIVER_CMEDIA, AZX_DRIVER_ZHAOXIN, AZX_DRIVER_LOONGSON, + AZX_DRIVER_ZHAOXINHDMI, AZX_DRIVER_GENERIC, AZX_NUM_DRIVERS, /* keep this as last entry */ }; @@ -350,6 +351,7 @@ static const char * const driver_short_names[] = { [AZX_DRIVER_CMEDIA] = "HDA C-Media", [AZX_DRIVER_ZHAOXIN] = "HDA Zhaoxin", [AZX_DRIVER_LOONGSON] = "HDA Loongson", + [AZX_DRIVER_ZHAOXINHDMI] = "HDA Zhaoxin HDMI", [AZX_DRIVER_GENERIC] = "HD-Audio Generic", }; @@ -371,6 +373,29 @@ static void update_pci_byte(struct pci_dev *pci, unsigned int reg, pci_write_config_byte(pci, reg, data); } +static int azx_init_pci_zx(struct azx *chip) +{ + struct snd_card *card = chip->card; + unsigned int diu_reg; + struct pci_dev *diu_pci = NULL; + + diu_pci = pci_get_device(PCI_VENDOR_ID_ZHAOXIN, 0x3a03, NULL); + if (!diu_pci) { + dev_err(card->dev, "hda no KX-5000 device.\n"); + return -ENXIO; + } + pci_read_config_dword(diu_pci, PCI_BASE_ADDRESS_0, &diu_reg); + chip->remap_diu_addr = ioremap(diu_reg, 0x50000); + dev_info(card->dev, "hda %x %p\n", diu_reg, chip->remap_diu_addr); + return 0; +} + +static void azx_free_pci_zx(struct azx *chip) +{ + if (chip->remap_diu_addr) + iounmap(chip->remap_diu_addr); +} + static void azx_init_pci(struct azx *chip) { int snoop_type = azx_get_snoop_type(chip); @@ -1360,6 +1385,10 @@ static void azx_free(struct azx *chip) hda->init_failed = 1; /* to be sure */ complete_all(&hda->probe_wait); + if (chip->driver_type == AZX_DRIVER_ZHAOXINHDMI) { + azx_free_pci_zx(chip); + } + if (use_vga_switcheroo(hda)) { if (chip->disabled && hda->probe_continued) snd_hda_unlock_devices(&chip->bus); @@ -1743,10 +1772,11 @@ static int default_bdl_pos_adj(struct azx *chip) switch (chip->driver_type) { /* - * increase the bdl size for Glenfly Gpus for hardware + * increase the bdl size for Glenfly/Zhaoxin Gpus for hardware * limitation on hdac interrupt interval */ case AZX_DRIVER_GFHDMI: + case AZX_DRIVER_ZHAOXINHDMI: return 128; case AZX_DRIVER_ICH: case AZX_DRIVER_PCH: @@ -1876,6 +1906,10 @@ static int azx_first_init(struct azx *chip) bus->access_sdnctl_in_dword = 1; } + chip->remap_diu_addr = NULL; + if (chip->driver_type == AZX_DRIVER_ZHAOXINHDMI) + azx_init_pci_zx(chip); + err = pcim_iomap_regions(pci, 1 << 0, "ICH HD audio"); if (err < 0) return err; @@ -1977,6 +2011,7 @@ static int azx_first_init(struct azx *chip) chip->capture_streams = ATIHDMI_NUM_CAPTURE; break; case AZX_DRIVER_GFHDMI: + case AZX_DRIVER_ZHAOXINHDMI: case AZX_DRIVER_GENERIC: default: chip->playback_streams = ICH6_NUM_PLAYBACK; @@ -2739,6 +2774,20 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_HDMI }, /* Zhaoxin */ { PCI_VDEVICE(ZHAOXIN, 0x3288), .driver_data = AZX_DRIVER_ZHAOXIN }, + { PCI_VDEVICE(ZHAOXIN, 0x9141), + .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_NO_64BIT }, + { PCI_VDEVICE(ZHAOXIN, 0x9142), + .driver_data = AZX_DRIVER_ZHAOXINHDMI | AZX_DCAPS_POSFIX_LPIB | + AZX_DCAPS_NO_MSI | AZX_DCAPS_RIRB_PRE_DELAY | AZX_DCAPS_NO_64BIT }, + { PCI_VDEVICE(ZHAOXIN, 0x9144), + .driver_data = AZX_DRIVER_ZHAOXINHDMI | AZX_DCAPS_POSFIX_LPIB | + AZX_DCAPS_NO_MSI | AZX_DCAPS_RIRB_PRE_DELAY | AZX_DCAPS_NO_64BIT }, + { PCI_VDEVICE(ZHAOXIN, 0x9145), + .driver_data = AZX_DRIVER_ZHAOXINHDMI | AZX_DCAPS_POSFIX_LPIB | + AZX_DCAPS_NO_MSI | AZX_DCAPS_RIRB_PRE_DELAY | AZX_DCAPS_NO_64BIT }, + { PCI_VDEVICE(ZHAOXIN, 0x9146), + .driver_data = AZX_DRIVER_ZHAOXINHDMI | AZX_DCAPS_POSFIX_LPIB | + AZX_DCAPS_NO_MSI | AZX_DCAPS_RIRB_PRE_DELAY | AZX_DCAPS_NO_64BIT }, /* Loongson HDAudio*/ { PCI_VDEVICE(LOONGSON, PCI_DEVICE_ID_LOONGSON_HDA), .driver_data = AZX_DRIVER_LOONGSON }, diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 1cde2a69bdb4..58ae77aa1ad9 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4501,6 +4501,8 @@ static int patch_gf_hdmi(struct hda_codec *codec) return 0; } +static int patch_zx_hdmi(struct hda_codec *codec) { return patch_gf_hdmi(codec); } + /* * patch entries */ @@ -4607,6 +4609,17 @@ HDA_CODEC_ENTRY(0x11069f80, "VX900 HDMI/DP", patch_via_hdmi), HDA_CODEC_ENTRY(0x11069f81, "VX900 HDMI/DP", patch_via_hdmi), HDA_CODEC_ENTRY(0x11069f84, "VX11 HDMI/DP", patch_generic_hdmi), HDA_CODEC_ENTRY(0x11069f85, "VX11 HDMI/DP", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x1d179f86, "ZX-100S HDMI/DP", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x1d179f87, "ZX-100S HDMI/DP", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x1d179f88, "KX-5000 HDMI/DP", patch_zx_hdmi), +HDA_CODEC_ENTRY(0x1d179f89, "KX-5000 HDMI/DP", patch_zx_hdmi), +HDA_CODEC_ENTRY(0x1d179f8a, "KX-6000 HDMI/DP", patch_zx_hdmi), +HDA_CODEC_ENTRY(0x1d179f8b, "KX-6000 HDMI/DP", patch_zx_hdmi), +HDA_CODEC_ENTRY(0x1d179f8c, "KX-6000G HDMI/DP", patch_zx_hdmi), +HDA_CODEC_ENTRY(0x1d179f8d, "KX-6000G HDMI/DP", patch_zx_hdmi), +HDA_CODEC_ENTRY(0x1d179f8e, "KX-8000 HDMI/DP", patch_zx_hdmi), +HDA_CODEC_ENTRY(0x1d179f8f, "KX-8000 HDMI/DP", patch_zx_hdmi), +HDA_CODEC_ENTRY(0x1d179f90, "KX-8000 HDMI/DP", patch_zx_hdmi), HDA_CODEC_ENTRY(0x80860054, "IbexPeak HDMI", patch_i915_cpt_hdmi), HDA_CODEC_ENTRY(0x80862800, "Geminilake HDMI", patch_i915_glk_hdmi), HDA_CODEC_ENTRY(0x80862801, "Bearlake HDMI", patch_generic_hdmi), -- 2.17.1

3 4

[PATCH -next 0/5] sound: Remove redundant of_match_ptr() macro
by Ruan Jinjie 14 Aug '23

14 Aug '23

Since these sound drivers depend on CONFIG_OF, there is no need to wrap the macro of_match_ptr() here. Ruan Jinjie (5): ASoC: atmel: Remove redundant of_match_ptr() ASoC: atmel: tse850-pcm5142: Remove redundant of_match_ptr() ASoC: sunxi: sun4i-spdif: Remove redundant of_match_ptr() ASoC: ssm3515: Remove redundant of_match_ptr() ALSA: Remove redundant of_match_ptr() sound/drivers/serial-generic.c | 2 +- sound/soc/atmel/atmel-i2s.c | 2 +- sound/soc/atmel/atmel-pdmic.c | 2 +- sound/soc/atmel/mchp-i2s-mcc.c | 2 +- sound/soc/atmel/mchp-spdifrx.c | 2 +- sound/soc/atmel/mchp-spdiftx.c | 2 +- sound/soc/atmel/tse850-pcm5142.c | 2 +- sound/soc/codecs/ssm3515.c | 2 +- sound/soc/sunxi/sun4i-spdif.c | 2 +- 9 files changed, 9 insertions(+), 9 deletions(-) -- 2.34.1

4 10

[PATCH 00/14] ASoC: rt: Drop unused GPIO includes
by Linus Walleij 14 Aug '23

14 Aug '23

These drivers include legacy GPIO headers for no reason at all, so get rid of the includes. Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- Linus Walleij (14): ASoC: rt1011: Drop GPIO includes ASoC: rt1015: Drop GPIO include ASoC: rt1015p: Drop legacy GPIO include ASoC: rt1016: Drop GPIO include ASoC: rt1019: Drop GPIO include ASoC: rt1305: Drop GPIO includes ASoC: rt1308: Drop GPIO includes ASoC: rt5514-spi: Drop GPIO include ASoC: rt5514: Drop GPIO include ASoC: rt5645: Drop legacy GPIO include ASoC: rt5659: Drop legacy GPIO include ASoC: rt5660: Drop GPIO includes ASoC: rt5682-sdw: Drop GPIO includes ASoC: rt715: Drop GPIO includes sound/soc/codecs/rt1011.c | 2 -- sound/soc/codecs/rt1015.c | 1 - sound/soc/codecs/rt1015p.c | 1 - sound/soc/codecs/rt1016.c | 1 - sound/soc/codecs/rt1019.c | 1 - sound/soc/codecs/rt1305.c | 2 -- sound/soc/codecs/rt1308.c | 2 -- sound/soc/codecs/rt5514-spi.c | 1 - sound/soc/codecs/rt5514.c | 1 - sound/soc/codecs/rt5645.c | 1 - sound/soc/codecs/rt5659.c | 1 - sound/soc/codecs/rt5660.c | 2 -- sound/soc/codecs/rt5682-sdw.c | 2 -- sound/soc/codecs/rt715.c | 3 --- 14 files changed, 21 deletions(-) --- base-commit: 06c2afb862f9da8dc5efa4b6076a0e48c3fbaaa5 change-id: 20230811-descriptors-asoc-221835f35f32 Best regards, -- Linus Walleij <linus.walleij(a)linaro.org>

2 15

[PATCH] ASoC: rt1308-sdw: fix random louder sound
by shumingf＠realtek.com 14 Aug '23

14 Aug '23

From: Shuming Fan <shumingf(a)realtek.com> This patch uses a vendor register to check whether the system hibernated ever. The driver will only set the preset when the driver brings up or the system hibernated. It will avoid the unknown issue that makes the speaker output louder and can't control the volume. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1308-sdw.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt1308-sdw.c b/sound/soc/codecs/rt1308-sdw.c index c98cd7abef6a..63d4abf964d4 100644 --- a/sound/soc/codecs/rt1308-sdw.c +++ b/sound/soc/codecs/rt1308-sdw.c @@ -52,6 +52,7 @@ static bool rt1308_volatile_register(struct device *dev, unsigned int reg) case 0x300a: case 0xc000: case 0xc710: + case 0xcf01: case 0xc860 ... 0xc863: case 0xc870 ... 0xc873: return true; @@ -213,7 +214,7 @@ static int rt1308_io_init(struct device *dev, struct sdw_slave *slave) { struct rt1308_sdw_priv *rt1308 = dev_get_drvdata(dev); int ret = 0; - unsigned int tmp; + unsigned int tmp, hibernation_flag; if (rt1308->hw_init) return 0; @@ -231,6 +232,10 @@ static int rt1308_io_init(struct device *dev, struct sdw_slave *slave) pm_runtime_get_noresume(&slave->dev); + regmap_read(rt1308->regmap, 0xcf01, &hibernation_flag); + if ((hibernation_flag != 0x00) && rt1308->first_hw_init) + goto _preset_ready_; + /* sw reset */ regmap_write(rt1308->regmap, RT1308_SDW_RESET, 0); @@ -271,6 +276,12 @@ static int rt1308_io_init(struct device *dev, struct sdw_slave *slave) regmap_write(rt1308->regmap, 0xc100, 0xd7); regmap_write(rt1308->regmap, 0xc101, 0xd7); + /* apply BQ params */ + rt1308_apply_bq_params(rt1308); + + regmap_write(rt1308->regmap, 0xcf01, 0x01); + +_preset_ready_: if (rt1308->first_hw_init) { regcache_cache_bypass(rt1308->regmap, false); regcache_mark_dirty(rt1308->regmap); -- 2.34.1

2 1

[PATCH -next] ASoC: q6dsp: Remove unused declaration
by Yue Haibing 14 Aug '23

14 Aug '23

Commit 5477518b8a0e ("ASoC: qdsp6: audioreach: add q6apm support") declared but never implemented these. Signed-off-by: Yue Haibing <yuehaibing(a)huawei.com> --- sound/soc/qcom/qdsp6/q6apm.h | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6apm.h b/sound/soc/qcom/qdsp6/q6apm.h index 8ee40732ce9e..f486bd639b9f 100644 --- a/sound/soc/qcom/qdsp6/q6apm.h +++ b/sound/soc/qcom/qdsp6/q6apm.h @@ -126,7 +126,6 @@ int q6apm_graph_media_format_shmem(struct q6apm_graph *graph, struct audioreach_module_config *cfg); /* read/write related */ -int q6apm_send_eos_nowait(struct q6apm_graph *graph); int q6apm_read(struct q6apm_graph *graph); int q6apm_write_async(struct q6apm_graph *graph, uint32_t len, uint32_t msw_ts, uint32_t lsw_ts, uint32_t wflags); @@ -144,7 +143,6 @@ int q6apm_send_cmd_sync(struct q6apm *apm, struct gpr_pkt *pkt, /* Callback for graph specific */ struct audioreach_module *q6apm_find_module_by_mid(struct q6apm_graph *graph, uint32_t mid); -void q6apm_set_fe_dai_ops(struct snd_soc_dai_driver *dai_drv); int q6apm_graph_get_rx_shmem_module_iid(struct q6apm_graph *graph); bool q6apm_is_adsp_ready(void); -- 2.34.1

2 1

[PATCH -next] ASoC: cygnus: Remove unused declarations
by Yue Haibing 14 Aug '23

14 Aug '23

Commit a6ee05d94e8f ("ASoC: cygnus: Add Cygnus audio DAI driver") declared but never implemented these. Signed-off-by: Yue Haibing <yuehaibing(a)huawei.com> --- sound/soc/bcm/cygnus-ssp.h | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/bcm/cygnus-ssp.h b/sound/soc/bcm/cygnus-ssp.h index 74152b2d770d..4925e03c3c30 100644 --- a/sound/soc/bcm/cygnus-ssp.h +++ b/sound/soc/bcm/cygnus-ssp.h @@ -117,8 +117,6 @@ struct cygnus_audio { unsigned long vco_rate; }; -extern int cygnus_ssp_get_mode(struct snd_soc_dai *cpu_dai); -extern int cygnus_ssp_add_pll_tweak_controls(struct snd_soc_pcm_runtime *rtd); extern int cygnus_ssp_set_custom_fsync_width(struct snd_soc_dai *cpu_dai, int len); extern int cygnus_soc_platform_register(struct device *dev, -- 2.34.1

2 1

[PATCH 0/2] ASoC: simple-card: Introduce playback-only/capture-only DAI link flags
by Daniel Baluta 13 Aug '23

13 Aug '23

From: Daniel Baluta <daniel.baluta(a)nxp.com> This patch series introduces capture-only/playback-only DAI link properties for simple-card. Daniel Baluta (2): ASoC: simple-card: Introduce playback-only/capture only DAI link flags ASoC: dt-bindings: simple-card: Document new DAI flags playback-only/capture-only .../bindings/sound/simple-card.yaml | 8 ++++++ include/sound/simple_card_utils.h | 5 ++++ sound/soc/generic/simple-card-utils.c | 27 +++++++++++++++++++ sound/soc/generic/simple-card.c | 10 +++++++ 4 files changed, 50 insertions(+) -- 2.25.1

5 7

[PATCH AUTOSEL 4.14 8/9] ASoc: codecs: ES8316: Fix DMIC config
by Sasha Levin 13 Aug '23

13 Aug '23

From: Edgar <ljijcj(a)163.com> [ Upstream commit d20d35d1ad62c6cca36368c1e8f29335a068659e ] According to the datasheet, the DMIC config should be changed to { 0, 2 ,3 } Signed-off-by: Edgar <ljijcj(a)163.com> Link: https://lore.kernel.org/r/20230719054722.401954-1-ljijcj@163.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/es8316.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index 0410f2e5183c3..d35a157bad57f 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -140,7 +140,7 @@ static const char * const es8316_dmic_txt[] = { "dmic data at high level", "dmic data at low level", }; -static const unsigned int es8316_dmic_values[] = { 0, 1, 2 }; +static const unsigned int es8316_dmic_values[] = { 0, 2, 3 }; static const struct soc_enum es8316_dmic_src_enum = SOC_VALUE_ENUM_SINGLE(ES8316_ADC_DMIC, 0, 3, ARRAY_SIZE(es8316_dmic_txt), -- 2.40.1

1 0

[PATCH AUTOSEL 4.14 2/9] ASoC: da7219: Flush pending AAD IRQ when suspending
by Sasha Levin 13 Aug '23

13 Aug '23

From: Dmytro Maluka <dmy(a)semihalf.com> [ Upstream commit 91e292917dad64ab8d1d5ca2ab3069ad9dac6f72 ] da7219_aad_suspend() disables jack detection, which should prevent generating new interrupts by DA7219 while suspended. However, there is a theoretical possibility that there is a pending interrupt generated just before suspending DA7219 and not handled yet, so the IRQ handler may still run after DA7219 is suspended. To prevent that, wait until the pending IRQ handling is done. This patch arose as an attempt to fix the following I2C failure occurring sometimes during system suspend or resume: [ 355.876211] i2c_designware i2c_designware.3: Transfer while suspended [ 355.876245] WARNING: CPU: 2 PID: 3576 at drivers/i2c/busses/i2c-designware-master.c:570 i2c_dw_xfer+0x411/0x440 ... [ 355.876462] Call Trace: [ 355.876468] <TASK> [ 355.876475] ? update_load_avg+0x1b3/0x615 [ 355.876484] __i2c_transfer+0x101/0x1d8 [ 355.876494] i2c_transfer+0x74/0x10d [ 355.876504] regmap_i2c_read+0x6a/0x9c [ 355.876513] _regmap_raw_read+0x179/0x223 [ 355.876521] regmap_raw_read+0x1e1/0x28e [ 355.876527] regmap_bulk_read+0x17d/0x1ba [ 355.876532] ? __wake_up+0xed/0x1bb [ 355.876542] da7219_aad_irq_thread+0x54/0x2c9 [snd_soc_da7219 5fb8ebb2179cf2fea29af090f3145d68ed8e2184] [ 355.876556] irq_thread+0x13c/0x231 [ 355.876563] ? irq_forced_thread_fn+0x5f/0x5f [ 355.876570] ? irq_thread_fn+0x4d/0x4d [ 355.876576] kthread+0x13a/0x152 [ 355.876581] ? synchronize_irq+0xc3/0xc3 [ 355.876587] ? kthread_blkcg+0x31/0x31 [ 355.876592] ret_from_fork+0x1f/0x30 [ 355.876601] </TASK> which indicates that the AAD IRQ handler is unexpectedly running when DA7219 is suspended, and as a result, is trying to read data from DA7219 over I2C and is hitting the I2C driver "Transfer while suspended" failure. However, with this patch the above failure is still reproducible. So this patch does not fix any real observed issue so far, but at least is useful for confirming that the above issue is not caused by a pending IRQ but rather looks like a DA7219 hardware issue with an IRQ unexpectedly generated after jack detection is already disabled. Signed-off-by: Dmytro Maluka <dmy(a)semihalf.com> Link: https://lore.kernel.org/r/20230717193737.161784-2-dmy@semihalf.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/da7219-aad.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index 1d1d10dd92ae2..793c8768f7c44 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -854,6 +854,8 @@ void da7219_aad_suspend(struct snd_soc_codec *codec) } } } + + synchronize_irq(da7219_aad->irq); } void da7219_aad_resume(struct snd_soc_codec *codec) -- 2.40.1

1 0

[PATCH AUTOSEL 4.19 11/13] ASoC: atmel: Fix the 8K sample parameter in I2SC master
by Sasha Levin 13 Aug '23

13 Aug '23

From: Guiting Shen <aarongt.shen(a)gmail.com> [ Upstream commit f85739c0b2b0d98a32f5ca4fcc5501d2b76df4f6 ] The 8K sample parameter of 12.288Mhz main system bus clock doesn't work because the I2SC_MR.IMCKDIV must not be 0 according to the sama5d2 series datasheet(I2SC Mode Register of Register Summary). So use the 6.144Mhz instead of 12.288Mhz to support 8K sample. Signed-off-by: Guiting Shen <aarongt.shen(a)gmail.com> Link: https://lore.kernel.org/r/20230715030620.62328-1-aarongt.shen@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/atmel/atmel-i2s.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/atmel/atmel-i2s.c b/sound/soc/atmel/atmel-i2s.c index 99cc731505769..ab7f761174742 100644 --- a/sound/soc/atmel/atmel-i2s.c +++ b/sound/soc/atmel/atmel-i2s.c @@ -174,11 +174,14 @@ struct atmel_i2s_gck_param { #define I2S_MCK_12M288 12288000UL #define I2S_MCK_11M2896 11289600UL +#define I2S_MCK_6M144 6144000UL /* mck = (32 * (imckfs+1) / (imckdiv+1)) * fs */ static const struct atmel_i2s_gck_param gck_params[] = { + /* mck = 6.144Mhz */ + { 8000, I2S_MCK_6M144, 1, 47}, /* mck = 768 fs */ + /* mck = 12.288MHz */ - { 8000, I2S_MCK_12M288, 0, 47}, /* mck = 1536 fs */ { 16000, I2S_MCK_12M288, 1, 47}, /* mck = 768 fs */ { 24000, I2S_MCK_12M288, 3, 63}, /* mck = 512 fs */ { 32000, I2S_MCK_12M288, 3, 47}, /* mck = 384 fs */ -- 2.40.1

1 0

[PATCH AUTOSEL 4.19 10/13] ASoc: codecs: ES8316: Fix DMIC config
by Sasha Levin 13 Aug '23

13 Aug '23

From: Edgar <ljijcj(a)163.com> [ Upstream commit d20d35d1ad62c6cca36368c1e8f29335a068659e ] According to the datasheet, the DMIC config should be changed to { 0, 2 ,3 } Signed-off-by: Edgar <ljijcj(a)163.com> Link: https://lore.kernel.org/r/20230719054722.401954-1-ljijcj@163.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/es8316.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index 57130edaf3aba..d3a44b21143d4 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -140,7 +140,7 @@ static const char * const es8316_dmic_txt[] = { "dmic data at high level", "dmic data at low level", }; -static const unsigned int es8316_dmic_values[] = { 0, 1, 2 }; +static const unsigned int es8316_dmic_values[] = { 0, 2, 3 }; static const struct soc_enum es8316_dmic_src_enum = SOC_VALUE_ENUM_SINGLE(ES8316_ADC_DMIC, 0, 3, ARRAY_SIZE(es8316_dmic_txt), -- 2.40.1

1 0

[PATCH AUTOSEL 4.19 03/13] ASoC: da7219: Check for failure reading AAD IRQ events
by Sasha Levin 13 Aug '23

13 Aug '23

From: Dmytro Maluka <dmy(a)semihalf.com> [ Upstream commit f0691dc16206f21b13c464434366e2cd632b8ed7 ] When handling an AAD interrupt, if IRQ events read failed (for example, due to i2c "Transfer while suspended" failure, i.e. when attempting to read it while DA7219 is suspended, which may happen due to a spurious AAD interrupt), the events array contains garbage uninitialized values. So instead of trying to interprete those values and doing any actions based on them (potentially resulting in misbehavior, e.g. reporting bogus events), refuse to handle the interrupt. Signed-off-by: Dmytro Maluka <dmy(a)semihalf.com> Link: https://lore.kernel.org/r/20230717193737.161784-3-dmy@semihalf.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/da7219-aad.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index 29e24a0f923d5..7e18e007a639f 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -351,11 +351,15 @@ static irqreturn_t da7219_aad_irq_thread(int irq, void *data) struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); u8 events[DA7219_AAD_IRQ_REG_MAX]; u8 statusa; - int i, report = 0, mask = 0; + int i, ret, report = 0, mask = 0; /* Read current IRQ events */ - regmap_bulk_read(da7219->regmap, DA7219_ACCDET_IRQ_EVENT_A, - events, DA7219_AAD_IRQ_REG_MAX); + ret = regmap_bulk_read(da7219->regmap, DA7219_ACCDET_IRQ_EVENT_A, + events, DA7219_AAD_IRQ_REG_MAX); + if (ret) { + dev_warn_ratelimited(component->dev, "Failed to read IRQ events: %d\n", ret); + return IRQ_NONE; + } if (!events[DA7219_AAD_IRQ_REG_A] && !events[DA7219_AAD_IRQ_REG_B]) return IRQ_NONE; -- 2.40.1

1 0

[PATCH AUTOSEL 4.19 02/13] ASoC: da7219: Flush pending AAD IRQ when suspending
by Sasha Levin 13 Aug '23

13 Aug '23

From: Dmytro Maluka <dmy(a)semihalf.com> [ Upstream commit 91e292917dad64ab8d1d5ca2ab3069ad9dac6f72 ] da7219_aad_suspend() disables jack detection, which should prevent generating new interrupts by DA7219 while suspended. However, there is a theoretical possibility that there is a pending interrupt generated just before suspending DA7219 and not handled yet, so the IRQ handler may still run after DA7219 is suspended. To prevent that, wait until the pending IRQ handling is done. This patch arose as an attempt to fix the following I2C failure occurring sometimes during system suspend or resume: [ 355.876211] i2c_designware i2c_designware.3: Transfer while suspended [ 355.876245] WARNING: CPU: 2 PID: 3576 at drivers/i2c/busses/i2c-designware-master.c:570 i2c_dw_xfer+0x411/0x440 ... [ 355.876462] Call Trace: [ 355.876468] <TASK> [ 355.876475] ? update_load_avg+0x1b3/0x615 [ 355.876484] __i2c_transfer+0x101/0x1d8 [ 355.876494] i2c_transfer+0x74/0x10d [ 355.876504] regmap_i2c_read+0x6a/0x9c [ 355.876513] _regmap_raw_read+0x179/0x223 [ 355.876521] regmap_raw_read+0x1e1/0x28e [ 355.876527] regmap_bulk_read+0x17d/0x1ba [ 355.876532] ? __wake_up+0xed/0x1bb [ 355.876542] da7219_aad_irq_thread+0x54/0x2c9 [snd_soc_da7219 5fb8ebb2179cf2fea29af090f3145d68ed8e2184] [ 355.876556] irq_thread+0x13c/0x231 [ 355.876563] ? irq_forced_thread_fn+0x5f/0x5f [ 355.876570] ? irq_thread_fn+0x4d/0x4d [ 355.876576] kthread+0x13a/0x152 [ 355.876581] ? synchronize_irq+0xc3/0xc3 [ 355.876587] ? kthread_blkcg+0x31/0x31 [ 355.876592] ret_from_fork+0x1f/0x30 [ 355.876601] </TASK> which indicates that the AAD IRQ handler is unexpectedly running when DA7219 is suspended, and as a result, is trying to read data from DA7219 over I2C and is hitting the I2C driver "Transfer while suspended" failure. However, with this patch the above failure is still reproducible. So this patch does not fix any real observed issue so far, but at least is useful for confirming that the above issue is not caused by a pending IRQ but rather looks like a DA7219 hardware issue with an IRQ unexpectedly generated after jack detection is already disabled. Signed-off-by: Dmytro Maluka <dmy(a)semihalf.com> Link: https://lore.kernel.org/r/20230717193737.161784-2-dmy@semihalf.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/da7219-aad.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index 2c7d5088e6f27..29e24a0f923d5 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -859,6 +859,8 @@ void da7219_aad_suspend(struct snd_soc_component *component) } } } + + synchronize_irq(da7219_aad->irq); } void da7219_aad_resume(struct snd_soc_component *component) -- 2.40.1

1 0

[PATCH AUTOSEL 5.4 11/14] ASoC: atmel: Fix the 8K sample parameter in I2SC master
by Sasha Levin 13 Aug '23

13 Aug '23

From: Guiting Shen <aarongt.shen(a)gmail.com> [ Upstream commit f85739c0b2b0d98a32f5ca4fcc5501d2b76df4f6 ] The 8K sample parameter of 12.288Mhz main system bus clock doesn't work because the I2SC_MR.IMCKDIV must not be 0 according to the sama5d2 series datasheet(I2SC Mode Register of Register Summary). So use the 6.144Mhz instead of 12.288Mhz to support 8K sample. Signed-off-by: Guiting Shen <aarongt.shen(a)gmail.com> Link: https://lore.kernel.org/r/20230715030620.62328-1-aarongt.shen@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/atmel/atmel-i2s.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/atmel/atmel-i2s.c b/sound/soc/atmel/atmel-i2s.c index d870f56c44cfc..0341b31197670 100644 --- a/sound/soc/atmel/atmel-i2s.c +++ b/sound/soc/atmel/atmel-i2s.c @@ -163,11 +163,14 @@ struct atmel_i2s_gck_param { #define I2S_MCK_12M288 12288000UL #define I2S_MCK_11M2896 11289600UL +#define I2S_MCK_6M144 6144000UL /* mck = (32 * (imckfs+1) / (imckdiv+1)) * fs */ static const struct atmel_i2s_gck_param gck_params[] = { + /* mck = 6.144Mhz */ + { 8000, I2S_MCK_6M144, 1, 47}, /* mck = 768 fs */ + /* mck = 12.288MHz */ - { 8000, I2S_MCK_12M288, 0, 47}, /* mck = 1536 fs */ { 16000, I2S_MCK_12M288, 1, 47}, /* mck = 768 fs */ { 24000, I2S_MCK_12M288, 3, 63}, /* mck = 512 fs */ { 32000, I2S_MCK_12M288, 3, 47}, /* mck = 384 fs */ -- 2.40.1

1 0

[PATCH AUTOSEL 5.4 10/14] ASoc: codecs: ES8316: Fix DMIC config
by Sasha Levin 13 Aug '23

13 Aug '23

From: Edgar <ljijcj(a)163.com> [ Upstream commit d20d35d1ad62c6cca36368c1e8f29335a068659e ] According to the datasheet, the DMIC config should be changed to { 0, 2 ,3 } Signed-off-by: Edgar <ljijcj(a)163.com> Link: https://lore.kernel.org/r/20230719054722.401954-1-ljijcj@163.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/es8316.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index 131f41cccbe65..dd2df9a903e05 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -153,7 +153,7 @@ static const char * const es8316_dmic_txt[] = { "dmic data at high level", "dmic data at low level", }; -static const unsigned int es8316_dmic_values[] = { 0, 1, 2 }; +static const unsigned int es8316_dmic_values[] = { 0, 2, 3 }; static const struct soc_enum es8316_dmic_src_enum = SOC_VALUE_ENUM_SINGLE(ES8316_ADC_DMIC, 0, 3, ARRAY_SIZE(es8316_dmic_txt), -- 2.40.1

1 0

[PATCH AUTOSEL 5.4 03/14] ASoC: da7219: Check for failure reading AAD IRQ events
by Sasha Levin 13 Aug '23

13 Aug '23

From: Dmytro Maluka <dmy(a)semihalf.com> [ Upstream commit f0691dc16206f21b13c464434366e2cd632b8ed7 ] When handling an AAD interrupt, if IRQ events read failed (for example, due to i2c "Transfer while suspended" failure, i.e. when attempting to read it while DA7219 is suspended, which may happen due to a spurious AAD interrupt), the events array contains garbage uninitialized values. So instead of trying to interprete those values and doing any actions based on them (potentially resulting in misbehavior, e.g. reporting bogus events), refuse to handle the interrupt. Signed-off-by: Dmytro Maluka <dmy(a)semihalf.com> Link: https://lore.kernel.org/r/20230717193737.161784-3-dmy@semihalf.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/da7219-aad.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index a652f154d8e52..befe26749bc2b 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -347,11 +347,15 @@ static irqreturn_t da7219_aad_irq_thread(int irq, void *data) struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); u8 events[DA7219_AAD_IRQ_REG_MAX]; u8 statusa; - int i, report = 0, mask = 0; + int i, ret, report = 0, mask = 0; /* Read current IRQ events */ - regmap_bulk_read(da7219->regmap, DA7219_ACCDET_IRQ_EVENT_A, - events, DA7219_AAD_IRQ_REG_MAX); + ret = regmap_bulk_read(da7219->regmap, DA7219_ACCDET_IRQ_EVENT_A, + events, DA7219_AAD_IRQ_REG_MAX); + if (ret) { + dev_warn_ratelimited(component->dev, "Failed to read IRQ events: %d\n", ret); + return IRQ_NONE; + } if (!events[DA7219_AAD_IRQ_REG_A] && !events[DA7219_AAD_IRQ_REG_B]) return IRQ_NONE; -- 2.40.1

1 0

[PATCH AUTOSEL 5.4 02/14] ASoC: da7219: Flush pending AAD IRQ when suspending
by Sasha Levin 13 Aug '23

13 Aug '23

From: Dmytro Maluka <dmy(a)semihalf.com> [ Upstream commit 91e292917dad64ab8d1d5ca2ab3069ad9dac6f72 ] da7219_aad_suspend() disables jack detection, which should prevent generating new interrupts by DA7219 while suspended. However, there is a theoretical possibility that there is a pending interrupt generated just before suspending DA7219 and not handled yet, so the IRQ handler may still run after DA7219 is suspended. To prevent that, wait until the pending IRQ handling is done. This patch arose as an attempt to fix the following I2C failure occurring sometimes during system suspend or resume: [ 355.876211] i2c_designware i2c_designware.3: Transfer while suspended [ 355.876245] WARNING: CPU: 2 PID: 3576 at drivers/i2c/busses/i2c-designware-master.c:570 i2c_dw_xfer+0x411/0x440 ... [ 355.876462] Call Trace: [ 355.876468] <TASK> [ 355.876475] ? update_load_avg+0x1b3/0x615 [ 355.876484] __i2c_transfer+0x101/0x1d8 [ 355.876494] i2c_transfer+0x74/0x10d [ 355.876504] regmap_i2c_read+0x6a/0x9c [ 355.876513] _regmap_raw_read+0x179/0x223 [ 355.876521] regmap_raw_read+0x1e1/0x28e [ 355.876527] regmap_bulk_read+0x17d/0x1ba [ 355.876532] ? __wake_up+0xed/0x1bb [ 355.876542] da7219_aad_irq_thread+0x54/0x2c9 [snd_soc_da7219 5fb8ebb2179cf2fea29af090f3145d68ed8e2184] [ 355.876556] irq_thread+0x13c/0x231 [ 355.876563] ? irq_forced_thread_fn+0x5f/0x5f [ 355.876570] ? irq_thread_fn+0x4d/0x4d [ 355.876576] kthread+0x13a/0x152 [ 355.876581] ? synchronize_irq+0xc3/0xc3 [ 355.876587] ? kthread_blkcg+0x31/0x31 [ 355.876592] ret_from_fork+0x1f/0x30 [ 355.876601] </TASK> which indicates that the AAD IRQ handler is unexpectedly running when DA7219 is suspended, and as a result, is trying to read data from DA7219 over I2C and is hitting the I2C driver "Transfer while suspended" failure. However, with this patch the above failure is still reproducible. So this patch does not fix any real observed issue so far, but at least is useful for confirming that the above issue is not caused by a pending IRQ but rather looks like a DA7219 hardware issue with an IRQ unexpectedly generated after jack detection is already disabled. Signed-off-by: Dmytro Maluka <dmy(a)semihalf.com> Link: https://lore.kernel.org/r/20230717193737.161784-2-dmy@semihalf.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/da7219-aad.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index 4f2a96e9fd45b..a652f154d8e52 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -855,6 +855,8 @@ void da7219_aad_suspend(struct snd_soc_component *component) } } } + + synchronize_irq(da7219_aad->irq); } void da7219_aad_resume(struct snd_soc_component *component) -- 2.40.1

1 0

[PATCH AUTOSEL 5.10 15/25] ASoC: atmel: Fix the 8K sample parameter in I2SC master
by Sasha Levin 13 Aug '23

13 Aug '23

From: Guiting Shen <aarongt.shen(a)gmail.com> [ Upstream commit f85739c0b2b0d98a32f5ca4fcc5501d2b76df4f6 ] The 8K sample parameter of 12.288Mhz main system bus clock doesn't work because the I2SC_MR.IMCKDIV must not be 0 according to the sama5d2 series datasheet(I2SC Mode Register of Register Summary). So use the 6.144Mhz instead of 12.288Mhz to support 8K sample. Signed-off-by: Guiting Shen <aarongt.shen(a)gmail.com> Link: https://lore.kernel.org/r/20230715030620.62328-1-aarongt.shen@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/atmel/atmel-i2s.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/atmel/atmel-i2s.c b/sound/soc/atmel/atmel-i2s.c index d870f56c44cfc..0341b31197670 100644 --- a/sound/soc/atmel/atmel-i2s.c +++ b/sound/soc/atmel/atmel-i2s.c @@ -163,11 +163,14 @@ struct atmel_i2s_gck_param { #define I2S_MCK_12M288 12288000UL #define I2S_MCK_11M2896 11289600UL +#define I2S_MCK_6M144 6144000UL /* mck = (32 * (imckfs+1) / (imckdiv+1)) * fs */ static const struct atmel_i2s_gck_param gck_params[] = { + /* mck = 6.144Mhz */ + { 8000, I2S_MCK_6M144, 1, 47}, /* mck = 768 fs */ + /* mck = 12.288MHz */ - { 8000, I2S_MCK_12M288, 0, 47}, /* mck = 1536 fs */ { 16000, I2S_MCK_12M288, 1, 47}, /* mck = 768 fs */ { 24000, I2S_MCK_12M288, 3, 63}, /* mck = 512 fs */ { 32000, I2S_MCK_12M288, 3, 47}, /* mck = 384 fs */ -- 2.40.1

1 0

[PATCH AUTOSEL 5.10 14/25] ASoc: codecs: ES8316: Fix DMIC config
by Sasha Levin 13 Aug '23

13 Aug '23

From: Edgar <ljijcj(a)163.com> [ Upstream commit d20d35d1ad62c6cca36368c1e8f29335a068659e ] According to the datasheet, the DMIC config should be changed to { 0, 2 ,3 } Signed-off-by: Edgar <ljijcj(a)163.com> Link: https://lore.kernel.org/r/20230719054722.401954-1-ljijcj@163.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/es8316.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index 03ad34a275da2..dafbf73d4eede 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -153,7 +153,7 @@ static const char * const es8316_dmic_txt[] = { "dmic data at high level", "dmic data at low level", }; -static const unsigned int es8316_dmic_values[] = { 0, 1, 2 }; +static const unsigned int es8316_dmic_values[] = { 0, 2, 3 }; static const struct soc_enum es8316_dmic_src_enum = SOC_VALUE_ENUM_SINGLE(ES8316_ADC_DMIC, 0, 3, ARRAY_SIZE(es8316_dmic_txt), -- 2.40.1

1 0

[PATCH AUTOSEL 5.10 07/25] ASoC: da7219: Check for failure reading AAD IRQ events
by Sasha Levin 13 Aug '23

13 Aug '23

From: Dmytro Maluka <dmy(a)semihalf.com> [ Upstream commit f0691dc16206f21b13c464434366e2cd632b8ed7 ] When handling an AAD interrupt, if IRQ events read failed (for example, due to i2c "Transfer while suspended" failure, i.e. when attempting to read it while DA7219 is suspended, which may happen due to a spurious AAD interrupt), the events array contains garbage uninitialized values. So instead of trying to interprete those values and doing any actions based on them (potentially resulting in misbehavior, e.g. reporting bogus events), refuse to handle the interrupt. Signed-off-by: Dmytro Maluka <dmy(a)semihalf.com> Link: https://lore.kernel.org/r/20230717193737.161784-3-dmy@semihalf.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/da7219-aad.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index 7fbda445be8e8..b316d613a7090 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -347,11 +347,15 @@ static irqreturn_t da7219_aad_irq_thread(int irq, void *data) struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); u8 events[DA7219_AAD_IRQ_REG_MAX]; u8 statusa; - int i, report = 0, mask = 0; + int i, ret, report = 0, mask = 0; /* Read current IRQ events */ - regmap_bulk_read(da7219->regmap, DA7219_ACCDET_IRQ_EVENT_A, - events, DA7219_AAD_IRQ_REG_MAX); + ret = regmap_bulk_read(da7219->regmap, DA7219_ACCDET_IRQ_EVENT_A, + events, DA7219_AAD_IRQ_REG_MAX); + if (ret) { + dev_warn_ratelimited(component->dev, "Failed to read IRQ events: %d\n", ret); + return IRQ_NONE; + } if (!events[DA7219_AAD_IRQ_REG_A] && !events[DA7219_AAD_IRQ_REG_B]) return IRQ_NONE; -- 2.40.1

1 0

[PATCH AUTOSEL 5.10 06/25] ASoC: da7219: Flush pending AAD IRQ when suspending
by Sasha Levin 13 Aug '23

13 Aug '23

From: Dmytro Maluka <dmy(a)semihalf.com> [ Upstream commit 91e292917dad64ab8d1d5ca2ab3069ad9dac6f72 ] da7219_aad_suspend() disables jack detection, which should prevent generating new interrupts by DA7219 while suspended. However, there is a theoretical possibility that there is a pending interrupt generated just before suspending DA7219 and not handled yet, so the IRQ handler may still run after DA7219 is suspended. To prevent that, wait until the pending IRQ handling is done. This patch arose as an attempt to fix the following I2C failure occurring sometimes during system suspend or resume: [ 355.876211] i2c_designware i2c_designware.3: Transfer while suspended [ 355.876245] WARNING: CPU: 2 PID: 3576 at drivers/i2c/busses/i2c-designware-master.c:570 i2c_dw_xfer+0x411/0x440 ... [ 355.876462] Call Trace: [ 355.876468] <TASK> [ 355.876475] ? update_load_avg+0x1b3/0x615 [ 355.876484] __i2c_transfer+0x101/0x1d8 [ 355.876494] i2c_transfer+0x74/0x10d [ 355.876504] regmap_i2c_read+0x6a/0x9c [ 355.876513] _regmap_raw_read+0x179/0x223 [ 355.876521] regmap_raw_read+0x1e1/0x28e [ 355.876527] regmap_bulk_read+0x17d/0x1ba [ 355.876532] ? __wake_up+0xed/0x1bb [ 355.876542] da7219_aad_irq_thread+0x54/0x2c9 [snd_soc_da7219 5fb8ebb2179cf2fea29af090f3145d68ed8e2184] [ 355.876556] irq_thread+0x13c/0x231 [ 355.876563] ? irq_forced_thread_fn+0x5f/0x5f [ 355.876570] ? irq_thread_fn+0x4d/0x4d [ 355.876576] kthread+0x13a/0x152 [ 355.876581] ? synchronize_irq+0xc3/0xc3 [ 355.876587] ? kthread_blkcg+0x31/0x31 [ 355.876592] ret_from_fork+0x1f/0x30 [ 355.876601] </TASK> which indicates that the AAD IRQ handler is unexpectedly running when DA7219 is suspended, and as a result, is trying to read data from DA7219 over I2C and is hitting the I2C driver "Transfer while suspended" failure. However, with this patch the above failure is still reproducible. So this patch does not fix any real observed issue so far, but at least is useful for confirming that the above issue is not caused by a pending IRQ but rather looks like a DA7219 hardware issue with an IRQ unexpectedly generated after jack detection is already disabled. Signed-off-by: Dmytro Maluka <dmy(a)semihalf.com> Link: https://lore.kernel.org/r/20230717193737.161784-2-dmy@semihalf.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/da7219-aad.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index 48081d71c22c9..7fbda445be8e8 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -854,6 +854,8 @@ void da7219_aad_suspend(struct snd_soc_component *component) } } } + + synchronize_irq(da7219_aad->irq); } void da7219_aad_resume(struct snd_soc_component *component) -- 2.40.1

1 0

[PATCH AUTOSEL 5.15 21/31] ALSA: usb-audio: Add quirk for Microsoft Modern Wireless Headset
by Sasha Levin 13 Aug '23

13 Aug '23

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 3da435063777f8d861ba5a165344e3f75f839357 ] Microsoft Modern Wireless Headset (appearing on the host as "Microsoft USB Link") has a playback and a capture mixer volume/switch, but they are fairly broken. The descriptor reports wrong dB ranges for playback, and the capture volume/switch don't influence on the actual recording at all. Moreover, there seem instabilities in the connection, and at best, we should disable the runtime PM. So this ended up with a quirk entry for: - Correct the playback dB range; I picked up some reasonable values but it's a guess work - Disable the capture mixer; it's completely useless and confuses PA/PW - Suppress get-sample-rate, apply the delay for message handling, and suppress the auto-suspend The behavior of the wheel control on the headset is somehow flaky, too, but it's an issue of HID. Link: https://bugzilla.suse.com/show_bug.cgi?id=1207129 Link: https://lore.kernel.org/r/20230725092057.15115-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/mixer_maps.c | 14 ++++++++++++++ sound/usb/quirks.c | 3 +++ 2 files changed, 17 insertions(+) diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 3f8f6056ff6a5..8e0915f02b26f 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -366,6 +366,15 @@ static const struct usbmix_name_map corsair_virtuoso_map[] = { { 0 } }; +/* Microsoft USB Link headset */ +/* a guess work: raw playback volume values are from 2 to 129 */ +static const struct usbmix_dB_map ms_usb_link_dB = { -3225, 0, true }; +static const struct usbmix_name_map ms_usb_link_map[] = { + { 9, NULL, .dB = &ms_usb_link_dB }, + { 10, NULL }, /* Headset Capture volume; seems non-working, disabled */ + { 0 } /* terminator */ +}; + /* ASUS ROG Zenith II with Realtek ALC1220-VB */ static const struct usbmix_name_map asus_zenith_ii_map[] = { { 19, NULL, 12 }, /* FU, Input Gain Pad - broken response, disabled */ @@ -625,6 +634,11 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = { .id = USB_ID(0x1395, 0x0025), .map = sennheiser_pc8_map, }, + { + /* Microsoft USB Link headset */ + .id = USB_ID(0x045e, 0x083c), + .map = ms_usb_link_map, + }, { 0 } /* terminator */ }; diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 8de572e774ddc..e5eab3b84b3f7 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1747,6 +1747,9 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x041e, 0x4080, /* Creative Live Cam VF0610 */ QUIRK_FLAG_GET_SAMPLE_RATE), + DEVICE_FLG(0x045e, 0x083c, /* MS USB Link headset */ + QUIRK_FLAG_GET_SAMPLE_RATE | QUIRK_FLAG_CTL_MSG_DELAY | + QUIRK_FLAG_DISABLE_AUTOSUSPEND), DEVICE_FLG(0x046d, 0x084c, /* Logitech ConferenceCam Connect */ QUIRK_FLAG_GET_SAMPLE_RATE | QUIRK_FLAG_CTL_MSG_DELAY_1M), DEVICE_FLG(0x046d, 0x0991, /* Logitech QuickCam Pro */ -- 2.40.1

1 0

[PATCH AUTOSEL 5.15 20/31] ASoC: atmel: Fix the 8K sample parameter in I2SC master
by Sasha Levin 13 Aug '23

13 Aug '23

From: Guiting Shen <aarongt.shen(a)gmail.com> [ Upstream commit f85739c0b2b0d98a32f5ca4fcc5501d2b76df4f6 ] The 8K sample parameter of 12.288Mhz main system bus clock doesn't work because the I2SC_MR.IMCKDIV must not be 0 according to the sama5d2 series datasheet(I2SC Mode Register of Register Summary). So use the 6.144Mhz instead of 12.288Mhz to support 8K sample. Signed-off-by: Guiting Shen <aarongt.shen(a)gmail.com> Link: https://lore.kernel.org/r/20230715030620.62328-1-aarongt.shen@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/atmel/atmel-i2s.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/atmel/atmel-i2s.c b/sound/soc/atmel/atmel-i2s.c index 6b3d9c05eaf27..4cb0605f6daa2 100644 --- a/sound/soc/atmel/atmel-i2s.c +++ b/sound/soc/atmel/atmel-i2s.c @@ -163,11 +163,14 @@ struct atmel_i2s_gck_param { #define I2S_MCK_12M288 12288000UL #define I2S_MCK_11M2896 11289600UL +#define I2S_MCK_6M144 6144000UL /* mck = (32 * (imckfs+1) / (imckdiv+1)) * fs */ static const struct atmel_i2s_gck_param gck_params[] = { + /* mck = 6.144Mhz */ + { 8000, I2S_MCK_6M144, 1, 47}, /* mck = 768 fs */ + /* mck = 12.288MHz */ - { 8000, I2S_MCK_12M288, 0, 47}, /* mck = 1536 fs */ { 16000, I2S_MCK_12M288, 1, 47}, /* mck = 768 fs */ { 24000, I2S_MCK_12M288, 3, 63}, /* mck = 512 fs */ { 32000, I2S_MCK_12M288, 3, 47}, /* mck = 384 fs */ -- 2.40.1

1 0

[PATCH AUTOSEL 5.15 19/31] ASoC: rt711-sdca: fix for JD event handling in ClockStop Mode0
by Sasha Levin 13 Aug '23

13 Aug '23

From: Shuming Fan <shumingf(a)realtek.com> [ Upstream commit 23adeb7056acd4fd866969f4afb91441776cc4f5 ] When the system suspends, peripheral SDCA interrupts are disabled. When system level resume is invoked, the peripheral SDCA interrupts should be enabled to handle JD events. Enable SDCA interrupts in resume sequence when ClockStop Mode0 is applied. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> Reported-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Link: https://lore.kernel.org/r/20230721090711.128247-1-shumingf@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt711-sdca-sdw.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt711-sdca-sdw.c b/sound/soc/codecs/rt711-sdca-sdw.c index 31e77d462ef34..4faf6b8544ddd 100644 --- a/sound/soc/codecs/rt711-sdca-sdw.c +++ b/sound/soc/codecs/rt711-sdca-sdw.c @@ -442,8 +442,16 @@ static int __maybe_unused rt711_sdca_dev_resume(struct device *dev) if (!rt711->first_hw_init) return 0; - if (!slave->unattach_request) + if (!slave->unattach_request) { + if (rt711->disable_irq == true) { + mutex_lock(&rt711->disable_irq_lock); + sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_0); + sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); + rt711->disable_irq = false; + mutex_unlock(&rt711->disable_irq_lock); + } goto regmap_sync; + } time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT711_PROBE_TIMEOUT)); -- 2.40.1

1 0

[PATCH AUTOSEL 5.15 18/31] ASoC: rt711: fix for JD event handling in ClockStop Mode0
by Sasha Levin 13 Aug '23

13 Aug '23

From: Shuming Fan <shumingf(a)realtek.com> [ Upstream commit b69de265bd0e877015a00fbba453ef72af162e0f ] When the system suspends, peripheral Imp-defined interrupt is disabled. When system level resume is invoked, the peripheral Imp-defined interrupts should be enabled to handle JD events. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> Reported-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Link: https://lore.kernel.org/r/20230721090654.128230-1-shumingf@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt711-sdw.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt711-sdw.c b/sound/soc/codecs/rt711-sdw.c index 4fe68bcf2a7c2..9545b8a7eb192 100644 --- a/sound/soc/codecs/rt711-sdw.c +++ b/sound/soc/codecs/rt711-sdw.c @@ -541,8 +541,15 @@ static int __maybe_unused rt711_dev_resume(struct device *dev) if (!rt711->first_hw_init) return 0; - if (!slave->unattach_request) + if (!slave->unattach_request) { + if (rt711->disable_irq == true) { + mutex_lock(&rt711->disable_irq_lock); + sdw_write_no_pm(slave, SDW_SCP_INTMASK1, SDW_SCP_INT1_IMPL_DEF); + rt711->disable_irq = false; + mutex_unlock(&rt711->disable_irq_lock); + } goto regmap_sync; + } time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT711_PROBE_TIMEOUT)); -- 2.40.1

1 0

[PATCH AUTOSEL 5.15 17/31] ASoc: codecs: ES8316: Fix DMIC config
by Sasha Levin 13 Aug '23

13 Aug '23

From: Edgar <ljijcj(a)163.com> [ Upstream commit d20d35d1ad62c6cca36368c1e8f29335a068659e ] According to the datasheet, the DMIC config should be changed to { 0, 2 ,3 } Signed-off-by: Edgar <ljijcj(a)163.com> Link: https://lore.kernel.org/r/20230719054722.401954-1-ljijcj@163.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/es8316.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index b36ccfc54cd69..93549f8ee130c 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -153,7 +153,7 @@ static const char * const es8316_dmic_txt[] = { "dmic data at high level", "dmic data at low level", }; -static const unsigned int es8316_dmic_values[] = { 0, 1, 2 }; +static const unsigned int es8316_dmic_values[] = { 0, 2, 3 }; static const struct soc_enum es8316_dmic_src_enum = SOC_VALUE_ENUM_SINGLE(ES8316_ADC_DMIC, 0, 3, ARRAY_SIZE(es8316_dmic_txt), -- 2.40.1

1 0

[PATCH AUTOSEL 5.15 16/31] ASoC: rt5682-sdw: fix for JD event handling in ClockStop Mode0
by Sasha Levin 13 Aug '23

13 Aug '23

From: Shuming Fan <shumingf(a)realtek.com> [ Upstream commit 02fb23d72720df2b6be3f29fc5787ca018eb92c3 ] When the system suspends, peripheral Imp-defined interrupt is disabled. When system level resume is invoked, the peripheral Imp-defined interrupts should be enabled to handle JD events. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> Reported-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Link: https://lore.kernel.org/r/20230721090643.128213-1-shumingf@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5682-sdw.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c index f04e18c32489d..9fdd9afe00da4 100644 --- a/sound/soc/codecs/rt5682-sdw.c +++ b/sound/soc/codecs/rt5682-sdw.c @@ -786,8 +786,15 @@ static int __maybe_unused rt5682_dev_resume(struct device *dev) if (!rt5682->first_hw_init) return 0; - if (!slave->unattach_request) + if (!slave->unattach_request) { + if (rt5682->disable_irq == true) { + mutex_lock(&rt5682->disable_irq_lock); + sdw_write_no_pm(slave, SDW_SCP_INTMASK1, SDW_SCP_INT1_IMPL_DEF); + rt5682->disable_irq = false; + mutex_unlock(&rt5682->disable_irq_lock); + } goto regmap_sync; + } time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT5682_PROBE_TIMEOUT)); -- 2.40.1

1 0

[PATCH AUTOSEL 5.15 09/31] ASoC: da7219: Check for failure reading AAD IRQ events
by Sasha Levin 13 Aug '23

13 Aug '23

From: Dmytro Maluka <dmy(a)semihalf.com> [ Upstream commit f0691dc16206f21b13c464434366e2cd632b8ed7 ] When handling an AAD interrupt, if IRQ events read failed (for example, due to i2c "Transfer while suspended" failure, i.e. when attempting to read it while DA7219 is suspended, which may happen due to a spurious AAD interrupt), the events array contains garbage uninitialized values. So instead of trying to interprete those values and doing any actions based on them (potentially resulting in misbehavior, e.g. reporting bogus events), refuse to handle the interrupt. Signed-off-by: Dmytro Maluka <dmy(a)semihalf.com> Link: https://lore.kernel.org/r/20230717193737.161784-3-dmy@semihalf.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/da7219-aad.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index 6b368ab945be5..7d18de959439f 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -347,11 +347,15 @@ static irqreturn_t da7219_aad_irq_thread(int irq, void *data) struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); u8 events[DA7219_AAD_IRQ_REG_MAX]; u8 statusa; - int i, report = 0, mask = 0; + int i, ret, report = 0, mask = 0; /* Read current IRQ events */ - regmap_bulk_read(da7219->regmap, DA7219_ACCDET_IRQ_EVENT_A, - events, DA7219_AAD_IRQ_REG_MAX); + ret = regmap_bulk_read(da7219->regmap, DA7219_ACCDET_IRQ_EVENT_A, + events, DA7219_AAD_IRQ_REG_MAX); + if (ret) { + dev_warn_ratelimited(component->dev, "Failed to read IRQ events: %d\n", ret); + return IRQ_NONE; + } if (!events[DA7219_AAD_IRQ_REG_A] && !events[DA7219_AAD_IRQ_REG_B]) return IRQ_NONE; -- 2.40.1

1 0

[PATCH AUTOSEL 5.15 08/31] ASoC: da7219: Flush pending AAD IRQ when suspending
by Sasha Levin 13 Aug '23

13 Aug '23

From: Dmytro Maluka <dmy(a)semihalf.com> [ Upstream commit 91e292917dad64ab8d1d5ca2ab3069ad9dac6f72 ] da7219_aad_suspend() disables jack detection, which should prevent generating new interrupts by DA7219 while suspended. However, there is a theoretical possibility that there is a pending interrupt generated just before suspending DA7219 and not handled yet, so the IRQ handler may still run after DA7219 is suspended. To prevent that, wait until the pending IRQ handling is done. This patch arose as an attempt to fix the following I2C failure occurring sometimes during system suspend or resume: [ 355.876211] i2c_designware i2c_designware.3: Transfer while suspended [ 355.876245] WARNING: CPU: 2 PID: 3576 at drivers/i2c/busses/i2c-designware-master.c:570 i2c_dw_xfer+0x411/0x440 ... [ 355.876462] Call Trace: [ 355.876468] <TASK> [ 355.876475] ? update_load_avg+0x1b3/0x615 [ 355.876484] __i2c_transfer+0x101/0x1d8 [ 355.876494] i2c_transfer+0x74/0x10d [ 355.876504] regmap_i2c_read+0x6a/0x9c [ 355.876513] _regmap_raw_read+0x179/0x223 [ 355.876521] regmap_raw_read+0x1e1/0x28e [ 355.876527] regmap_bulk_read+0x17d/0x1ba [ 355.876532] ? __wake_up+0xed/0x1bb [ 355.876542] da7219_aad_irq_thread+0x54/0x2c9 [snd_soc_da7219 5fb8ebb2179cf2fea29af090f3145d68ed8e2184] [ 355.876556] irq_thread+0x13c/0x231 [ 355.876563] ? irq_forced_thread_fn+0x5f/0x5f [ 355.876570] ? irq_thread_fn+0x4d/0x4d [ 355.876576] kthread+0x13a/0x152 [ 355.876581] ? synchronize_irq+0xc3/0xc3 [ 355.876587] ? kthread_blkcg+0x31/0x31 [ 355.876592] ret_from_fork+0x1f/0x30 [ 355.876601] </TASK> which indicates that the AAD IRQ handler is unexpectedly running when DA7219 is suspended, and as a result, is trying to read data from DA7219 over I2C and is hitting the I2C driver "Transfer while suspended" failure. However, with this patch the above failure is still reproducible. So this patch does not fix any real observed issue so far, but at least is useful for confirming that the above issue is not caused by a pending IRQ but rather looks like a DA7219 hardware issue with an IRQ unexpectedly generated after jack detection is already disabled. Signed-off-by: Dmytro Maluka <dmy(a)semihalf.com> Link: https://lore.kernel.org/r/20230717193737.161784-2-dmy@semihalf.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/da7219-aad.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index 7998fdd3b378e..6b368ab945be5 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -854,6 +854,8 @@ void da7219_aad_suspend(struct snd_soc_component *component) } } } + + synchronize_irq(da7219_aad->irq); } void da7219_aad_resume(struct snd_soc_component *component) -- 2.40.1

1 0

[PATCH AUTOSEL 6.1 44/47] ALSA: usb-audio: Update for native DSD support quirks
by Sasha Levin 13 Aug '23

13 Aug '23

From: Jussi Laako <jussi(a)sonarnerd.net> [ Upstream commit f7fea075edfa085c25eb34c44ceacf3602537f98 ] Maintenance patch for native DSD support. Remove incorrect T+A device quirks. Move set of device quirks to vendor quirks. Add set of missing device and vendor quirks. Signed-off-by: Jussi Laako <jussi(a)sonarnerd.net> Link: https://lore.kernel.org/r/20230726165645.404311-1-jussi@sonarnerd.net Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/quirks.c | 34 ++++++++++++++++++++++++++++------ 1 file changed, 28 insertions(+), 6 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index d4a7ffef82194..4667d543f7481 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1874,8 +1874,10 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, /* XMOS based USB DACs */ switch (chip->usb_id) { - case USB_ID(0x1511, 0x0037): /* AURALiC VEGA */ - case USB_ID(0x21ed, 0xd75a): /* Accuphase DAC-60 option card */ + case USB_ID(0x139f, 0x5504): /* Nagra DAC */ + case USB_ID(0x20b1, 0x3089): /* Mola-Mola DAC */ + case USB_ID(0x2522, 0x0007): /* LH Labs Geek Out 1V5 */ + case USB_ID(0x2522, 0x0009): /* LH Labs Geek Pulse X Inifinity 2V0 */ case USB_ID(0x2522, 0x0012): /* LH Labs VI DAC Infinity */ case USB_ID(0x2772, 0x0230): /* Pro-Ject Pre Box S2 Digital */ if (fp->altsetting == 2) @@ -1885,14 +1887,18 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, case USB_ID(0x0d8c, 0x0316): /* Hegel HD12 DSD */ case USB_ID(0x10cb, 0x0103): /* The Bit Opus #3; with fp->dsd_raw */ case USB_ID(0x16d0, 0x06b2): /* NuPrime DAC-10 */ - case USB_ID(0x16d0, 0x09dd): /* Encore mDSD */ + case USB_ID(0x16d0, 0x06b4): /* NuPrime Audio HD-AVP/AVA */ case USB_ID(0x16d0, 0x0733): /* Furutech ADL Stratos */ + case USB_ID(0x16d0, 0x09d8): /* NuPrime IDA-8 */ case USB_ID(0x16d0, 0x09db): /* NuPrime Audio DAC-9 */ + case USB_ID(0x16d0, 0x09dd): /* Encore mDSD */ case USB_ID(0x1db5, 0x0003): /* Bryston BDA3 */ + case USB_ID(0x20a0, 0x4143): /* WaveIO USB Audio 2.0 */ case USB_ID(0x22e1, 0xca01): /* HDTA Serenade DSD */ case USB_ID(0x249c, 0x9326): /* M2Tech Young MkIII */ case USB_ID(0x2616, 0x0106): /* PS Audio NuWave DAC */ case USB_ID(0x2622, 0x0041): /* Audiolab M-DAC+ */ + case USB_ID(0x278b, 0x5100): /* Rotel RC-1590 */ case USB_ID(0x27f7, 0x3002): /* W4S DAC-2v2SE */ case USB_ID(0x29a2, 0x0086): /* Mutec MC3+ USB */ case USB_ID(0x6b42, 0x0042): /* MSB Technology */ @@ -1902,9 +1908,6 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, /* Amanero Combo384 USB based DACs with native DSD support */ case USB_ID(0x16d0, 0x071a): /* Amanero - Combo384 */ - case USB_ID(0x2ab6, 0x0004): /* T+A DAC8DSD-V2.0, MP1000E-V2.0, MP2000R-V2.0, MP2500R-V2.0, MP3100HV-V2.0 */ - case USB_ID(0x2ab6, 0x0005): /* T+A USB HD Audio 1 */ - case USB_ID(0x2ab6, 0x0006): /* T+A USB HD Audio 2 */ if (fp->altsetting == 2) { switch (le16_to_cpu(chip->dev->descriptor.bcdDevice)) { case 0x199: @@ -2049,6 +2052,9 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_IFACE_DELAY), DEVICE_FLG(0x0644, 0x805f, /* TEAC Model 12 */ QUIRK_FLAG_FORCE_IFACE_RESET), + DEVICE_FLG(0x0644, 0x806b, /* TEAC UD-701 */ + QUIRK_FLAG_ITF_USB_DSD_DAC | QUIRK_FLAG_CTL_MSG_DELAY | + QUIRK_FLAG_IFACE_DELAY), DEVICE_FLG(0x06f8, 0xb000, /* Hercules DJ Console (Windows Edition) */ QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x06f8, 0xd002, /* Hercules DJ Console (Macintosh Edition) */ @@ -2087,6 +2093,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_ITF_USB_DSD_DAC | QUIRK_FLAG_CTL_MSG_DELAY), DEVICE_FLG(0x154e, 0x3006, /* Marantz SA-14S1 */ QUIRK_FLAG_ITF_USB_DSD_DAC | QUIRK_FLAG_CTL_MSG_DELAY), + DEVICE_FLG(0x154e, 0x300b, /* Marantz SA-KI RUBY / SA-12 */ + QUIRK_FLAG_DSD_RAW), DEVICE_FLG(0x154e, 0x500e, /* Denon DN-X1600 */ QUIRK_FLAG_IGNORE_CLOCK_SOURCE), DEVICE_FLG(0x1686, 0x00dd, /* Zoom R16/24 */ @@ -2131,6 +2139,10 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_SHARE_MEDIA_DEVICE | QUIRK_FLAG_ALIGN_TRANSFER), DEVICE_FLG(0x21b4, 0x0081, /* AudioQuest DragonFly */ QUIRK_FLAG_GET_SAMPLE_RATE), + DEVICE_FLG(0x21b4, 0x0230, /* Ayre QB-9 Twenty */ + QUIRK_FLAG_DSD_RAW), + DEVICE_FLG(0x21b4, 0x0232, /* Ayre QX-5 Twenty */ + QUIRK_FLAG_DSD_RAW), DEVICE_FLG(0x2522, 0x0007, /* LH Labs Geek Out HD Audio 1V5 */ QUIRK_FLAG_SET_IFACE_FIRST), DEVICE_FLG(0x2708, 0x0002, /* Audient iD14 */ @@ -2173,12 +2185,18 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_VALIDATE_RATES), VENDOR_FLG(0x1235, /* Focusrite Novation */ QUIRK_FLAG_VALIDATE_RATES), + VENDOR_FLG(0x1511, /* AURALiC */ + QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x152a, /* Thesycon devices */ QUIRK_FLAG_DSD_RAW), + VENDOR_FLG(0x18d1, /* iBasso devices */ + QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x1de7, /* Phoenix Audio */ QUIRK_FLAG_GET_SAMPLE_RATE), VENDOR_FLG(0x20b1, /* XMOS based devices */ QUIRK_FLAG_DSD_RAW), + VENDOR_FLG(0x21ed, /* Accuphase Laboratory */ + QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x22d9, /* Oppo */ QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x23ba, /* Playback Design */ @@ -2194,10 +2212,14 @@ 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(0x2d87, /* Cayin device */ + QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x3336, /* HEM devices */ QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x3353, /* Khadas devices */ QUIRK_FLAG_DSD_RAW), + VENDOR_FLG(0x35f4, /* MSB Technology */ + QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x3842, /* EVGA */ QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0xc502, /* HiBy devices */ -- 2.40.1

1 0

[PATCH AUTOSEL 6.1 29/47] ALSA: usb-audio: Add quirk for Microsoft Modern Wireless Headset
by Sasha Levin 13 Aug '23

13 Aug '23

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 3da435063777f8d861ba5a165344e3f75f839357 ] Microsoft Modern Wireless Headset (appearing on the host as "Microsoft USB Link") has a playback and a capture mixer volume/switch, but they are fairly broken. The descriptor reports wrong dB ranges for playback, and the capture volume/switch don't influence on the actual recording at all. Moreover, there seem instabilities in the connection, and at best, we should disable the runtime PM. So this ended up with a quirk entry for: - Correct the playback dB range; I picked up some reasonable values but it's a guess work - Disable the capture mixer; it's completely useless and confuses PA/PW - Suppress get-sample-rate, apply the delay for message handling, and suppress the auto-suspend The behavior of the wheel control on the headset is somehow flaky, too, but it's an issue of HID. Link: https://bugzilla.suse.com/show_bug.cgi?id=1207129 Link: https://lore.kernel.org/r/20230725092057.15115-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/mixer_maps.c | 14 ++++++++++++++ sound/usb/quirks.c | 3 +++ 2 files changed, 17 insertions(+) diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index f4bd1e8ae4b6c..23260aa1919d3 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -374,6 +374,15 @@ static const struct usbmix_name_map corsair_virtuoso_map[] = { { 0 } }; +/* Microsoft USB Link headset */ +/* a guess work: raw playback volume values are from 2 to 129 */ +static const struct usbmix_dB_map ms_usb_link_dB = { -3225, 0, true }; +static const struct usbmix_name_map ms_usb_link_map[] = { + { 9, NULL, .dB = &ms_usb_link_dB }, + { 10, NULL }, /* Headset Capture volume; seems non-working, disabled */ + { 0 } /* terminator */ +}; + /* ASUS ROG Zenith II with Realtek ALC1220-VB */ static const struct usbmix_name_map asus_zenith_ii_map[] = { { 19, NULL, 12 }, /* FU, Input Gain Pad - broken response, disabled */ @@ -668,6 +677,11 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = { .id = USB_ID(0x1395, 0x0025), .map = sennheiser_pc8_map, }, + { + /* Microsoft USB Link headset */ + .id = USB_ID(0x045e, 0x083c), + .map = ms_usb_link_map, + }, { 0 } /* terminator */ }; diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 6cf55b7f7a041..d4a7ffef82194 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2011,6 +2011,9 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x041e, 0x4080, /* Creative Live Cam VF0610 */ QUIRK_FLAG_GET_SAMPLE_RATE), + DEVICE_FLG(0x045e, 0x083c, /* MS USB Link headset */ + QUIRK_FLAG_GET_SAMPLE_RATE | QUIRK_FLAG_CTL_MSG_DELAY | + QUIRK_FLAG_DISABLE_AUTOSUSPEND), DEVICE_FLG(0x046d, 0x084c, /* Logitech ConferenceCam Connect */ QUIRK_FLAG_GET_SAMPLE_RATE | QUIRK_FLAG_CTL_MSG_DELAY_1M), DEVICE_FLG(0x046d, 0x0991, /* Logitech QuickCam Pro */ -- 2.40.1

1 0

[PATCH AUTOSEL 6.1 28/47] ASoC: atmel: Fix the 8K sample parameter in I2SC master
by Sasha Levin 13 Aug '23

13 Aug '23

From: Guiting Shen <aarongt.shen(a)gmail.com> [ Upstream commit f85739c0b2b0d98a32f5ca4fcc5501d2b76df4f6 ] The 8K sample parameter of 12.288Mhz main system bus clock doesn't work because the I2SC_MR.IMCKDIV must not be 0 according to the sama5d2 series datasheet(I2SC Mode Register of Register Summary). So use the 6.144Mhz instead of 12.288Mhz to support 8K sample. Signed-off-by: Guiting Shen <aarongt.shen(a)gmail.com> Link: https://lore.kernel.org/r/20230715030620.62328-1-aarongt.shen@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/atmel/atmel-i2s.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/atmel/atmel-i2s.c b/sound/soc/atmel/atmel-i2s.c index 425d66edbf867..5e43ff0b537a3 100644 --- a/sound/soc/atmel/atmel-i2s.c +++ b/sound/soc/atmel/atmel-i2s.c @@ -163,11 +163,14 @@ struct atmel_i2s_gck_param { #define I2S_MCK_12M288 12288000UL #define I2S_MCK_11M2896 11289600UL +#define I2S_MCK_6M144 6144000UL /* mck = (32 * (imckfs+1) / (imckdiv+1)) * fs */ static const struct atmel_i2s_gck_param gck_params[] = { + /* mck = 6.144Mhz */ + { 8000, I2S_MCK_6M144, 1, 47}, /* mck = 768 fs */ + /* mck = 12.288MHz */ - { 8000, I2S_MCK_12M288, 0, 47}, /* mck = 1536 fs */ { 16000, I2S_MCK_12M288, 1, 47}, /* mck = 768 fs */ { 24000, I2S_MCK_12M288, 3, 63}, /* mck = 512 fs */ { 32000, I2S_MCK_12M288, 3, 47}, /* mck = 384 fs */ -- 2.40.1

1 0

[PATCH AUTOSEL 6.1 27/47] ASoC: rt711-sdca: fix for JD event handling in ClockStop Mode0
by Sasha Levin 13 Aug '23

13 Aug '23

From: Shuming Fan <shumingf(a)realtek.com> [ Upstream commit 23adeb7056acd4fd866969f4afb91441776cc4f5 ] When the system suspends, peripheral SDCA interrupts are disabled. When system level resume is invoked, the peripheral SDCA interrupts should be enabled to handle JD events. Enable SDCA interrupts in resume sequence when ClockStop Mode0 is applied. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> Reported-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Link: https://lore.kernel.org/r/20230721090711.128247-1-shumingf@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt711-sdca-sdw.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt711-sdca-sdw.c b/sound/soc/codecs/rt711-sdca-sdw.c index e23cec4c457de..487d3010ddc19 100644 --- a/sound/soc/codecs/rt711-sdca-sdw.c +++ b/sound/soc/codecs/rt711-sdca-sdw.c @@ -442,8 +442,16 @@ static int __maybe_unused rt711_sdca_dev_resume(struct device *dev) if (!rt711->first_hw_init) return 0; - if (!slave->unattach_request) + if (!slave->unattach_request) { + if (rt711->disable_irq == true) { + mutex_lock(&rt711->disable_irq_lock); + sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_0); + sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); + rt711->disable_irq = false; + mutex_unlock(&rt711->disable_irq_lock); + } goto regmap_sync; + } time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT711_PROBE_TIMEOUT)); -- 2.40.1

1 0

[PATCH AUTOSEL 6.1 26/47] ASoC: rt711: fix for JD event handling in ClockStop Mode0
by Sasha Levin 13 Aug '23

13 Aug '23

From: Shuming Fan <shumingf(a)realtek.com> [ Upstream commit b69de265bd0e877015a00fbba453ef72af162e0f ] When the system suspends, peripheral Imp-defined interrupt is disabled. When system level resume is invoked, the peripheral Imp-defined interrupts should be enabled to handle JD events. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> Reported-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Link: https://lore.kernel.org/r/20230721090654.128230-1-shumingf@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt711-sdw.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt711-sdw.c b/sound/soc/codecs/rt711-sdw.c index 4fe68bcf2a7c2..9545b8a7eb192 100644 --- a/sound/soc/codecs/rt711-sdw.c +++ b/sound/soc/codecs/rt711-sdw.c @@ -541,8 +541,15 @@ static int __maybe_unused rt711_dev_resume(struct device *dev) if (!rt711->first_hw_init) return 0; - if (!slave->unattach_request) + if (!slave->unattach_request) { + if (rt711->disable_irq == true) { + mutex_lock(&rt711->disable_irq_lock); + sdw_write_no_pm(slave, SDW_SCP_INTMASK1, SDW_SCP_INT1_IMPL_DEF); + rt711->disable_irq = false; + mutex_unlock(&rt711->disable_irq_lock); + } goto regmap_sync; + } time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT711_PROBE_TIMEOUT)); -- 2.40.1

1 0

[PATCH AUTOSEL 6.1 25/47] ASoc: codecs: ES8316: Fix DMIC config
by Sasha Levin 13 Aug '23

13 Aug '23

From: Edgar <ljijcj(a)163.com> [ Upstream commit d20d35d1ad62c6cca36368c1e8f29335a068659e ] According to the datasheet, the DMIC config should be changed to { 0, 2 ,3 } Signed-off-by: Edgar <ljijcj(a)163.com> Link: https://lore.kernel.org/r/20230719054722.401954-1-ljijcj@163.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/es8316.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index 87775378362e7..c4e4ab93fdb6d 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -153,7 +153,7 @@ static const char * const es8316_dmic_txt[] = { "dmic data at high level", "dmic data at low level", }; -static const unsigned int es8316_dmic_values[] = { 0, 1, 2 }; +static const unsigned int es8316_dmic_values[] = { 0, 2, 3 }; static const struct soc_enum es8316_dmic_src_enum = SOC_VALUE_ENUM_SINGLE(ES8316_ADC_DMIC, 0, 3, ARRAY_SIZE(es8316_dmic_txt), -- 2.40.1

1 0

[PATCH AUTOSEL 6.1 24/47] ASoC: rt5682-sdw: fix for JD event handling in ClockStop Mode0
by Sasha Levin 13 Aug '23

13 Aug '23

From: Shuming Fan <shumingf(a)realtek.com> [ Upstream commit 02fb23d72720df2b6be3f29fc5787ca018eb92c3 ] When the system suspends, peripheral Imp-defined interrupt is disabled. When system level resume is invoked, the peripheral Imp-defined interrupts should be enabled to handle JD events. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> Reported-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Link: https://lore.kernel.org/r/20230721090643.128213-1-shumingf@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5682-sdw.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c index c1a94229dc7e3..868a61c8b0608 100644 --- a/sound/soc/codecs/rt5682-sdw.c +++ b/sound/soc/codecs/rt5682-sdw.c @@ -786,8 +786,15 @@ static int __maybe_unused rt5682_dev_resume(struct device *dev) if (!rt5682->first_hw_init) return 0; - if (!slave->unattach_request) + if (!slave->unattach_request) { + if (rt5682->disable_irq == true) { + mutex_lock(&rt5682->disable_irq_lock); + sdw_write_no_pm(slave, SDW_SCP_INTMASK1, SDW_SCP_INT1_IMPL_DEF); + rt5682->disable_irq = false; + mutex_unlock(&rt5682->disable_irq_lock); + } goto regmap_sync; + } time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT5682_PROBE_TIMEOUT)); -- 2.40.1

1 0

[PATCH AUTOSEL 6.1 15/47] ASoC: nau8821: Add DMI quirk mechanism for active-high jack-detect
by Sasha Levin 13 Aug '23

13 Aug '23

From: Edson Juliano Drosdeck <edson.drosdeck(a)gmail.com> [ Upstream commit 1bc40efdaf4a0ccfdb10a1c8e4b458f4764e8e5f ] Add a quirk mechanism to allow specifying that active-high jack-detection should be used on platforms where this info is not available in devicetree. And add an entry for the Positivo CW14Q01P-V2 to the DMI table, so that jack-detection will work properly on this laptop. Signed-off-by: Edson Juliano Drosdeck <edson.drosdeck(a)gmail.com> Link: https://lore.kernel.org/r/20230719200241.4865-1-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/nau8821.c | 41 ++++++++++++++++++++++++++++++++++++++ 1 file changed, 41 insertions(+) diff --git a/sound/soc/codecs/nau8821.c b/sound/soc/codecs/nau8821.c index 4a72b94e84104..efd92656a060d 100644 --- a/sound/soc/codecs/nau8821.c +++ b/sound/soc/codecs/nau8821.c @@ -10,6 +10,7 @@ #include <linux/acpi.h> #include <linux/clk.h> #include <linux/delay.h> +#include <linux/dmi.h> #include <linux/init.h> #include <linux/i2c.h> #include <linux/module.h> @@ -25,6 +26,13 @@ #include <sound/tlv.h> #include "nau8821.h" +#define NAU8821_JD_ACTIVE_HIGH BIT(0) + +static int nau8821_quirk; +static int quirk_override = -1; +module_param_named(quirk, quirk_override, uint, 0444); +MODULE_PARM_DESC(quirk, "Board-specific quirk override"); + #define NAU_FREF_MAX 13500000 #define NAU_FVCO_MAX 100000000 #define NAU_FVCO_MIN 90000000 @@ -1696,6 +1704,33 @@ static int nau8821_setup_irq(struct nau8821 *nau8821) return 0; } +/* Please keep this list alphabetically sorted */ +static const struct dmi_system_id nau8821_quirk_table[] = { + { + /* Positivo CW14Q01P-V2 */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Positivo Tecnologia SA"), + DMI_MATCH(DMI_BOARD_NAME, "CW14Q01P-V2"), + }, + .driver_data = (void *)(NAU8821_JD_ACTIVE_HIGH), + }, + {} +}; + +static void nau8821_check_quirks(void) +{ + const struct dmi_system_id *dmi_id; + + if (quirk_override != -1) { + nau8821_quirk = quirk_override; + return; + } + + dmi_id = dmi_first_match(nau8821_quirk_table); + if (dmi_id) + nau8821_quirk = (unsigned long)dmi_id->driver_data; +} + static int nau8821_i2c_probe(struct i2c_client *i2c) { struct device *dev = &i2c->dev; @@ -1716,6 +1751,12 @@ static int nau8821_i2c_probe(struct i2c_client *i2c) nau8821->dev = dev; nau8821->irq = i2c->irq; + + nau8821_check_quirks(); + + if (nau8821_quirk & NAU8821_JD_ACTIVE_HIGH) + nau8821->jkdet_polarity = 0; + nau8821_print_device_properties(nau8821); nau8821_reset_chip(nau8821->regmap); -- 2.40.1

1 0

[PATCH AUTOSEL 6.1 14/47] ASoC: da7219: Check for failure reading AAD IRQ events
by Sasha Levin 13 Aug '23

13 Aug '23

From: Dmytro Maluka <dmy(a)semihalf.com> [ Upstream commit f0691dc16206f21b13c464434366e2cd632b8ed7 ] When handling an AAD interrupt, if IRQ events read failed (for example, due to i2c "Transfer while suspended" failure, i.e. when attempting to read it while DA7219 is suspended, which may happen due to a spurious AAD interrupt), the events array contains garbage uninitialized values. So instead of trying to interprete those values and doing any actions based on them (potentially resulting in misbehavior, e.g. reporting bogus events), refuse to handle the interrupt. Signed-off-by: Dmytro Maluka <dmy(a)semihalf.com> Link: https://lore.kernel.org/r/20230717193737.161784-3-dmy@semihalf.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/da7219-aad.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index 49b1622e7bada..9251490548e8c 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -353,11 +353,15 @@ static irqreturn_t da7219_aad_irq_thread(int irq, void *data) struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); u8 events[DA7219_AAD_IRQ_REG_MAX]; u8 statusa; - int i, report = 0, mask = 0; + int i, ret, report = 0, mask = 0; /* Read current IRQ events */ - regmap_bulk_read(da7219->regmap, DA7219_ACCDET_IRQ_EVENT_A, - events, DA7219_AAD_IRQ_REG_MAX); + ret = regmap_bulk_read(da7219->regmap, DA7219_ACCDET_IRQ_EVENT_A, + events, DA7219_AAD_IRQ_REG_MAX); + if (ret) { + dev_warn_ratelimited(component->dev, "Failed to read IRQ events: %d\n", ret); + return IRQ_NONE; + } if (!events[DA7219_AAD_IRQ_REG_A] && !events[DA7219_AAD_IRQ_REG_B]) return IRQ_NONE; -- 2.40.1

1 0

[PATCH AUTOSEL 6.1 13/47] ASoC: da7219: Flush pending AAD IRQ when suspending
by Sasha Levin 13 Aug '23

13 Aug '23

From: Dmytro Maluka <dmy(a)semihalf.com> [ Upstream commit 91e292917dad64ab8d1d5ca2ab3069ad9dac6f72 ] da7219_aad_suspend() disables jack detection, which should prevent generating new interrupts by DA7219 while suspended. However, there is a theoretical possibility that there is a pending interrupt generated just before suspending DA7219 and not handled yet, so the IRQ handler may still run after DA7219 is suspended. To prevent that, wait until the pending IRQ handling is done. This patch arose as an attempt to fix the following I2C failure occurring sometimes during system suspend or resume: [ 355.876211] i2c_designware i2c_designware.3: Transfer while suspended [ 355.876245] WARNING: CPU: 2 PID: 3576 at drivers/i2c/busses/i2c-designware-master.c:570 i2c_dw_xfer+0x411/0x440 ... [ 355.876462] Call Trace: [ 355.876468] <TASK> [ 355.876475] ? update_load_avg+0x1b3/0x615 [ 355.876484] __i2c_transfer+0x101/0x1d8 [ 355.876494] i2c_transfer+0x74/0x10d [ 355.876504] regmap_i2c_read+0x6a/0x9c [ 355.876513] _regmap_raw_read+0x179/0x223 [ 355.876521] regmap_raw_read+0x1e1/0x28e [ 355.876527] regmap_bulk_read+0x17d/0x1ba [ 355.876532] ? __wake_up+0xed/0x1bb [ 355.876542] da7219_aad_irq_thread+0x54/0x2c9 [snd_soc_da7219 5fb8ebb2179cf2fea29af090f3145d68ed8e2184] [ 355.876556] irq_thread+0x13c/0x231 [ 355.876563] ? irq_forced_thread_fn+0x5f/0x5f [ 355.876570] ? irq_thread_fn+0x4d/0x4d [ 355.876576] kthread+0x13a/0x152 [ 355.876581] ? synchronize_irq+0xc3/0xc3 [ 355.876587] ? kthread_blkcg+0x31/0x31 [ 355.876592] ret_from_fork+0x1f/0x30 [ 355.876601] </TASK> which indicates that the AAD IRQ handler is unexpectedly running when DA7219 is suspended, and as a result, is trying to read data from DA7219 over I2C and is hitting the I2C driver "Transfer while suspended" failure. However, with this patch the above failure is still reproducible. So this patch does not fix any real observed issue so far, but at least is useful for confirming that the above issue is not caused by a pending IRQ but rather looks like a DA7219 hardware issue with an IRQ unexpectedly generated after jack detection is already disabled. Signed-off-by: Dmytro Maluka <dmy(a)semihalf.com> Link: https://lore.kernel.org/r/20230717193737.161784-2-dmy@semihalf.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/da7219-aad.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index bba73c44c219f..49b1622e7bada 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -863,6 +863,8 @@ void da7219_aad_suspend(struct snd_soc_component *component) } } } + + synchronize_irq(da7219_aad->irq); } void da7219_aad_resume(struct snd_soc_component *component) -- 2.40.1

1 0

[PATCH AUTOSEL 6.4 49/54] ALSA: usb-audio: Update for native DSD support quirks
by Sasha Levin 13 Aug '23

13 Aug '23

From: Jussi Laako <jussi(a)sonarnerd.net> [ Upstream commit f7fea075edfa085c25eb34c44ceacf3602537f98 ] Maintenance patch for native DSD support. Remove incorrect T+A device quirks. Move set of device quirks to vendor quirks. Add set of missing device and vendor quirks. Signed-off-by: Jussi Laako <jussi(a)sonarnerd.net> Link: https://lore.kernel.org/r/20230726165645.404311-1-jussi@sonarnerd.net Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/quirks.c | 34 ++++++++++++++++++++++++++++------ 1 file changed, 28 insertions(+), 6 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index d4a7ffef82194..4667d543f7481 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1874,8 +1874,10 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, /* XMOS based USB DACs */ switch (chip->usb_id) { - case USB_ID(0x1511, 0x0037): /* AURALiC VEGA */ - case USB_ID(0x21ed, 0xd75a): /* Accuphase DAC-60 option card */ + case USB_ID(0x139f, 0x5504): /* Nagra DAC */ + case USB_ID(0x20b1, 0x3089): /* Mola-Mola DAC */ + case USB_ID(0x2522, 0x0007): /* LH Labs Geek Out 1V5 */ + case USB_ID(0x2522, 0x0009): /* LH Labs Geek Pulse X Inifinity 2V0 */ case USB_ID(0x2522, 0x0012): /* LH Labs VI DAC Infinity */ case USB_ID(0x2772, 0x0230): /* Pro-Ject Pre Box S2 Digital */ if (fp->altsetting == 2) @@ -1885,14 +1887,18 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, case USB_ID(0x0d8c, 0x0316): /* Hegel HD12 DSD */ case USB_ID(0x10cb, 0x0103): /* The Bit Opus #3; with fp->dsd_raw */ case USB_ID(0x16d0, 0x06b2): /* NuPrime DAC-10 */ - case USB_ID(0x16d0, 0x09dd): /* Encore mDSD */ + case USB_ID(0x16d0, 0x06b4): /* NuPrime Audio HD-AVP/AVA */ case USB_ID(0x16d0, 0x0733): /* Furutech ADL Stratos */ + case USB_ID(0x16d0, 0x09d8): /* NuPrime IDA-8 */ case USB_ID(0x16d0, 0x09db): /* NuPrime Audio DAC-9 */ + case USB_ID(0x16d0, 0x09dd): /* Encore mDSD */ case USB_ID(0x1db5, 0x0003): /* Bryston BDA3 */ + case USB_ID(0x20a0, 0x4143): /* WaveIO USB Audio 2.0 */ case USB_ID(0x22e1, 0xca01): /* HDTA Serenade DSD */ case USB_ID(0x249c, 0x9326): /* M2Tech Young MkIII */ case USB_ID(0x2616, 0x0106): /* PS Audio NuWave DAC */ case USB_ID(0x2622, 0x0041): /* Audiolab M-DAC+ */ + case USB_ID(0x278b, 0x5100): /* Rotel RC-1590 */ case USB_ID(0x27f7, 0x3002): /* W4S DAC-2v2SE */ case USB_ID(0x29a2, 0x0086): /* Mutec MC3+ USB */ case USB_ID(0x6b42, 0x0042): /* MSB Technology */ @@ -1902,9 +1908,6 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, /* Amanero Combo384 USB based DACs with native DSD support */ case USB_ID(0x16d0, 0x071a): /* Amanero - Combo384 */ - case USB_ID(0x2ab6, 0x0004): /* T+A DAC8DSD-V2.0, MP1000E-V2.0, MP2000R-V2.0, MP2500R-V2.0, MP3100HV-V2.0 */ - case USB_ID(0x2ab6, 0x0005): /* T+A USB HD Audio 1 */ - case USB_ID(0x2ab6, 0x0006): /* T+A USB HD Audio 2 */ if (fp->altsetting == 2) { switch (le16_to_cpu(chip->dev->descriptor.bcdDevice)) { case 0x199: @@ -2049,6 +2052,9 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_IFACE_DELAY), DEVICE_FLG(0x0644, 0x805f, /* TEAC Model 12 */ QUIRK_FLAG_FORCE_IFACE_RESET), + DEVICE_FLG(0x0644, 0x806b, /* TEAC UD-701 */ + QUIRK_FLAG_ITF_USB_DSD_DAC | QUIRK_FLAG_CTL_MSG_DELAY | + QUIRK_FLAG_IFACE_DELAY), DEVICE_FLG(0x06f8, 0xb000, /* Hercules DJ Console (Windows Edition) */ QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x06f8, 0xd002, /* Hercules DJ Console (Macintosh Edition) */ @@ -2087,6 +2093,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_ITF_USB_DSD_DAC | QUIRK_FLAG_CTL_MSG_DELAY), DEVICE_FLG(0x154e, 0x3006, /* Marantz SA-14S1 */ QUIRK_FLAG_ITF_USB_DSD_DAC | QUIRK_FLAG_CTL_MSG_DELAY), + DEVICE_FLG(0x154e, 0x300b, /* Marantz SA-KI RUBY / SA-12 */ + QUIRK_FLAG_DSD_RAW), DEVICE_FLG(0x154e, 0x500e, /* Denon DN-X1600 */ QUIRK_FLAG_IGNORE_CLOCK_SOURCE), DEVICE_FLG(0x1686, 0x00dd, /* Zoom R16/24 */ @@ -2131,6 +2139,10 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_SHARE_MEDIA_DEVICE | QUIRK_FLAG_ALIGN_TRANSFER), DEVICE_FLG(0x21b4, 0x0081, /* AudioQuest DragonFly */ QUIRK_FLAG_GET_SAMPLE_RATE), + DEVICE_FLG(0x21b4, 0x0230, /* Ayre QB-9 Twenty */ + QUIRK_FLAG_DSD_RAW), + DEVICE_FLG(0x21b4, 0x0232, /* Ayre QX-5 Twenty */ + QUIRK_FLAG_DSD_RAW), DEVICE_FLG(0x2522, 0x0007, /* LH Labs Geek Out HD Audio 1V5 */ QUIRK_FLAG_SET_IFACE_FIRST), DEVICE_FLG(0x2708, 0x0002, /* Audient iD14 */ @@ -2173,12 +2185,18 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_VALIDATE_RATES), VENDOR_FLG(0x1235, /* Focusrite Novation */ QUIRK_FLAG_VALIDATE_RATES), + VENDOR_FLG(0x1511, /* AURALiC */ + QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x152a, /* Thesycon devices */ QUIRK_FLAG_DSD_RAW), + VENDOR_FLG(0x18d1, /* iBasso devices */ + QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x1de7, /* Phoenix Audio */ QUIRK_FLAG_GET_SAMPLE_RATE), VENDOR_FLG(0x20b1, /* XMOS based devices */ QUIRK_FLAG_DSD_RAW), + VENDOR_FLG(0x21ed, /* Accuphase Laboratory */ + QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x22d9, /* Oppo */ QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x23ba, /* Playback Design */ @@ -2194,10 +2212,14 @@ 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(0x2d87, /* Cayin device */ + QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x3336, /* HEM devices */ QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x3353, /* Khadas devices */ QUIRK_FLAG_DSD_RAW), + VENDOR_FLG(0x35f4, /* MSB Technology */ + QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x3842, /* EVGA */ QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0xc502, /* HiBy devices */ -- 2.40.1

1 0

[PATCH AUTOSEL 6.4 34/54] ALSA: usb-audio: Add quirk for Microsoft Modern Wireless Headset
by Sasha Levin 13 Aug '23

13 Aug '23

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 3da435063777f8d861ba5a165344e3f75f839357 ] Microsoft Modern Wireless Headset (appearing on the host as "Microsoft USB Link") has a playback and a capture mixer volume/switch, but they are fairly broken. The descriptor reports wrong dB ranges for playback, and the capture volume/switch don't influence on the actual recording at all. Moreover, there seem instabilities in the connection, and at best, we should disable the runtime PM. So this ended up with a quirk entry for: - Correct the playback dB range; I picked up some reasonable values but it's a guess work - Disable the capture mixer; it's completely useless and confuses PA/PW - Suppress get-sample-rate, apply the delay for message handling, and suppress the auto-suspend The behavior of the wheel control on the headset is somehow flaky, too, but it's an issue of HID. Link: https://bugzilla.suse.com/show_bug.cgi?id=1207129 Link: https://lore.kernel.org/r/20230725092057.15115-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/mixer_maps.c | 14 ++++++++++++++ sound/usb/quirks.c | 3 +++ 2 files changed, 17 insertions(+) diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index f4bd1e8ae4b6c..23260aa1919d3 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -374,6 +374,15 @@ static const struct usbmix_name_map corsair_virtuoso_map[] = { { 0 } }; +/* Microsoft USB Link headset */ +/* a guess work: raw playback volume values are from 2 to 129 */ +static const struct usbmix_dB_map ms_usb_link_dB = { -3225, 0, true }; +static const struct usbmix_name_map ms_usb_link_map[] = { + { 9, NULL, .dB = &ms_usb_link_dB }, + { 10, NULL }, /* Headset Capture volume; seems non-working, disabled */ + { 0 } /* terminator */ +}; + /* ASUS ROG Zenith II with Realtek ALC1220-VB */ static const struct usbmix_name_map asus_zenith_ii_map[] = { { 19, NULL, 12 }, /* FU, Input Gain Pad - broken response, disabled */ @@ -668,6 +677,11 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = { .id = USB_ID(0x1395, 0x0025), .map = sennheiser_pc8_map, }, + { + /* Microsoft USB Link headset */ + .id = USB_ID(0x045e, 0x083c), + .map = ms_usb_link_map, + }, { 0 } /* terminator */ }; diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 6cf55b7f7a041..d4a7ffef82194 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2011,6 +2011,9 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x041e, 0x4080, /* Creative Live Cam VF0610 */ QUIRK_FLAG_GET_SAMPLE_RATE), + DEVICE_FLG(0x045e, 0x083c, /* MS USB Link headset */ + QUIRK_FLAG_GET_SAMPLE_RATE | QUIRK_FLAG_CTL_MSG_DELAY | + QUIRK_FLAG_DISABLE_AUTOSUSPEND), DEVICE_FLG(0x046d, 0x084c, /* Logitech ConferenceCam Connect */ QUIRK_FLAG_GET_SAMPLE_RATE | QUIRK_FLAG_CTL_MSG_DELAY_1M), DEVICE_FLG(0x046d, 0x0991, /* Logitech QuickCam Pro */ -- 2.40.1

1 0

[PATCH AUTOSEL 6.4 33/54] ASoC: atmel: Fix the 8K sample parameter in I2SC master
by Sasha Levin 13 Aug '23

13 Aug '23

From: Guiting Shen <aarongt.shen(a)gmail.com> [ Upstream commit f85739c0b2b0d98a32f5ca4fcc5501d2b76df4f6 ] The 8K sample parameter of 12.288Mhz main system bus clock doesn't work because the I2SC_MR.IMCKDIV must not be 0 according to the sama5d2 series datasheet(I2SC Mode Register of Register Summary). So use the 6.144Mhz instead of 12.288Mhz to support 8K sample. Signed-off-by: Guiting Shen <aarongt.shen(a)gmail.com> Link: https://lore.kernel.org/r/20230715030620.62328-1-aarongt.shen@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/atmel/atmel-i2s.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/atmel/atmel-i2s.c b/sound/soc/atmel/atmel-i2s.c index 49930baf5e4d6..69a88dc651652 100644 --- a/sound/soc/atmel/atmel-i2s.c +++ b/sound/soc/atmel/atmel-i2s.c @@ -163,11 +163,14 @@ struct atmel_i2s_gck_param { #define I2S_MCK_12M288 12288000UL #define I2S_MCK_11M2896 11289600UL +#define I2S_MCK_6M144 6144000UL /* mck = (32 * (imckfs+1) / (imckdiv+1)) * fs */ static const struct atmel_i2s_gck_param gck_params[] = { + /* mck = 6.144Mhz */ + { 8000, I2S_MCK_6M144, 1, 47}, /* mck = 768 fs */ + /* mck = 12.288MHz */ - { 8000, I2S_MCK_12M288, 0, 47}, /* mck = 1536 fs */ { 16000, I2S_MCK_12M288, 1, 47}, /* mck = 768 fs */ { 24000, I2S_MCK_12M288, 3, 63}, /* mck = 512 fs */ { 32000, I2S_MCK_12M288, 3, 47}, /* mck = 384 fs */ -- 2.40.1

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[PATCH AUTOSEL 6.4 32/54] ASoC: rt711-sdca: fix for JD event handling in ClockStop Mode0
by Sasha Levin 13 Aug '23

13 Aug '23

From: Shuming Fan <shumingf(a)realtek.com> [ Upstream commit 23adeb7056acd4fd866969f4afb91441776cc4f5 ] When the system suspends, peripheral SDCA interrupts are disabled. When system level resume is invoked, the peripheral SDCA interrupts should be enabled to handle JD events. Enable SDCA interrupts in resume sequence when ClockStop Mode0 is applied. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> Reported-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Link: https://lore.kernel.org/r/20230721090711.128247-1-shumingf@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt711-sdca-sdw.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt711-sdca-sdw.c b/sound/soc/codecs/rt711-sdca-sdw.c index 51f3335343e08..76ed61e47316d 100644 --- a/sound/soc/codecs/rt711-sdca-sdw.c +++ b/sound/soc/codecs/rt711-sdca-sdw.c @@ -441,8 +441,16 @@ static int __maybe_unused rt711_sdca_dev_resume(struct device *dev) if (!rt711->first_hw_init) return 0; - if (!slave->unattach_request) + if (!slave->unattach_request) { + if (rt711->disable_irq == true) { + mutex_lock(&rt711->disable_irq_lock); + sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_0); + sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); + rt711->disable_irq = false; + mutex_unlock(&rt711->disable_irq_lock); + } goto regmap_sync; + } time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT711_PROBE_TIMEOUT)); -- 2.40.1

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[PATCH AUTOSEL 6.4 31/54] ASoC: rt711: fix for JD event handling in ClockStop Mode0
by Sasha Levin 13 Aug '23

13 Aug '23

From: Shuming Fan <shumingf(a)realtek.com> [ Upstream commit b69de265bd0e877015a00fbba453ef72af162e0f ] When the system suspends, peripheral Imp-defined interrupt is disabled. When system level resume is invoked, the peripheral Imp-defined interrupts should be enabled to handle JD events. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> Reported-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Link: https://lore.kernel.org/r/20230721090654.128230-1-shumingf@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt711-sdw.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt711-sdw.c b/sound/soc/codecs/rt711-sdw.c index 4fe68bcf2a7c2..9545b8a7eb192 100644 --- a/sound/soc/codecs/rt711-sdw.c +++ b/sound/soc/codecs/rt711-sdw.c @@ -541,8 +541,15 @@ static int __maybe_unused rt711_dev_resume(struct device *dev) if (!rt711->first_hw_init) return 0; - if (!slave->unattach_request) + if (!slave->unattach_request) { + if (rt711->disable_irq == true) { + mutex_lock(&rt711->disable_irq_lock); + sdw_write_no_pm(slave, SDW_SCP_INTMASK1, SDW_SCP_INT1_IMPL_DEF); + rt711->disable_irq = false; + mutex_unlock(&rt711->disable_irq_lock); + } goto regmap_sync; + } time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT711_PROBE_TIMEOUT)); -- 2.40.1

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[PATCH AUTOSEL 6.4 30/54] ASoC: rt712-sdca: fix for JD event handling in ClockStop Mode0
by Sasha Levin 13 Aug '23

13 Aug '23

From: Shuming Fan <shumingf(a)realtek.com> [ Upstream commit 7a8735c1551e489351172d0da96128f6f8b52b2d ] When the system suspends, peripheral SDCA interrupts are disabled. When system level resume is invoked, the peripheral SDCA interrupts should be enabled to handle JD events. Enable SDCA interrupts in resume sequence when ClockStop Mode0 is applied. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> Reported-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Link: https://lore.kernel.org/r/20230721090721.128264-1-shumingf@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt712-sdca-sdw.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt712-sdca-sdw.c b/sound/soc/codecs/rt712-sdca-sdw.c index 3f319459dfec3..1c9e10fea3ddd 100644 --- a/sound/soc/codecs/rt712-sdca-sdw.c +++ b/sound/soc/codecs/rt712-sdca-sdw.c @@ -441,8 +441,16 @@ static int __maybe_unused rt712_sdca_dev_resume(struct device *dev) if (!rt712->first_hw_init) return 0; - if (!slave->unattach_request) + if (!slave->unattach_request) { + if (rt712->disable_irq == true) { + mutex_lock(&rt712->disable_irq_lock); + sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_0); + sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); + rt712->disable_irq = false; + mutex_unlock(&rt712->disable_irq_lock); + } goto regmap_sync; + } time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT712_PROBE_TIMEOUT)); -- 2.40.1

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[PATCH AUTOSEL 6.4 29/54] ASoc: codecs: ES8316: Fix DMIC config
by Sasha Levin 13 Aug '23

13 Aug '23

From: Edgar <ljijcj(a)163.com> [ Upstream commit d20d35d1ad62c6cca36368c1e8f29335a068659e ] According to the datasheet, the DMIC config should be changed to { 0, 2 ,3 } Signed-off-by: Edgar <ljijcj(a)163.com> Link: https://lore.kernel.org/r/20230719054722.401954-1-ljijcj@163.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/es8316.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index ccecfdf700649..9f5522dee501d 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -153,7 +153,7 @@ static const char * const es8316_dmic_txt[] = { "dmic data at high level", "dmic data at low level", }; -static const unsigned int es8316_dmic_values[] = { 0, 1, 2 }; +static const unsigned int es8316_dmic_values[] = { 0, 2, 3 }; static const struct soc_enum es8316_dmic_src_enum = SOC_VALUE_ENUM_SINGLE(ES8316_ADC_DMIC, 0, 3, ARRAY_SIZE(es8316_dmic_txt), -- 2.40.1

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[PATCH AUTOSEL 6.4 28/54] ASoC: rt5682-sdw: fix for JD event handling in ClockStop Mode0
by Sasha Levin 13 Aug '23

13 Aug '23

From: Shuming Fan <shumingf(a)realtek.com> [ Upstream commit 02fb23d72720df2b6be3f29fc5787ca018eb92c3 ] When the system suspends, peripheral Imp-defined interrupt is disabled. When system level resume is invoked, the peripheral Imp-defined interrupts should be enabled to handle JD events. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> Reported-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Link: https://lore.kernel.org/r/20230721090643.128213-1-shumingf@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5682-sdw.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c index 23f17f70d7e9b..9622aaf1b3e63 100644 --- a/sound/soc/codecs/rt5682-sdw.c +++ b/sound/soc/codecs/rt5682-sdw.c @@ -753,8 +753,15 @@ static int __maybe_unused rt5682_dev_resume(struct device *dev) if (!rt5682->first_hw_init) return 0; - if (!slave->unattach_request) + if (!slave->unattach_request) { + if (rt5682->disable_irq == true) { + mutex_lock(&rt5682->disable_irq_lock); + sdw_write_no_pm(slave, SDW_SCP_INTMASK1, SDW_SCP_INT1_IMPL_DEF); + rt5682->disable_irq = false; + mutex_unlock(&rt5682->disable_irq_lock); + } goto regmap_sync; + } time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT5682_PROBE_TIMEOUT)); -- 2.40.1

1 0

[PATCH AUTOSEL 6.4 17/54] ASoC: nau8821: Add DMI quirk mechanism for active-high jack-detect
by Sasha Levin 13 Aug '23

13 Aug '23

From: Edson Juliano Drosdeck <edson.drosdeck(a)gmail.com> [ Upstream commit 1bc40efdaf4a0ccfdb10a1c8e4b458f4764e8e5f ] Add a quirk mechanism to allow specifying that active-high jack-detection should be used on platforms where this info is not available in devicetree. And add an entry for the Positivo CW14Q01P-V2 to the DMI table, so that jack-detection will work properly on this laptop. Signed-off-by: Edson Juliano Drosdeck <edson.drosdeck(a)gmail.com> Link: https://lore.kernel.org/r/20230719200241.4865-1-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/nau8821.c | 41 ++++++++++++++++++++++++++++++++++++++ 1 file changed, 41 insertions(+) diff --git a/sound/soc/codecs/nau8821.c b/sound/soc/codecs/nau8821.c index fee970427a243..42de7588fdb68 100644 --- a/sound/soc/codecs/nau8821.c +++ b/sound/soc/codecs/nau8821.c @@ -10,6 +10,7 @@ #include <linux/acpi.h> #include <linux/clk.h> #include <linux/delay.h> +#include <linux/dmi.h> #include <linux/init.h> #include <linux/i2c.h> #include <linux/module.h> @@ -25,6 +26,13 @@ #include <sound/tlv.h> #include "nau8821.h" +#define NAU8821_JD_ACTIVE_HIGH BIT(0) + +static int nau8821_quirk; +static int quirk_override = -1; +module_param_named(quirk, quirk_override, uint, 0444); +MODULE_PARM_DESC(quirk, "Board-specific quirk override"); + #define NAU_FREF_MAX 13500000 #define NAU_FVCO_MAX 100000000 #define NAU_FVCO_MIN 90000000 @@ -1792,6 +1800,33 @@ static int nau8821_setup_irq(struct nau8821 *nau8821) return 0; } +/* Please keep this list alphabetically sorted */ +static const struct dmi_system_id nau8821_quirk_table[] = { + { + /* Positivo CW14Q01P-V2 */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Positivo Tecnologia SA"), + DMI_MATCH(DMI_BOARD_NAME, "CW14Q01P-V2"), + }, + .driver_data = (void *)(NAU8821_JD_ACTIVE_HIGH), + }, + {} +}; + +static void nau8821_check_quirks(void) +{ + const struct dmi_system_id *dmi_id; + + if (quirk_override != -1) { + nau8821_quirk = quirk_override; + return; + } + + dmi_id = dmi_first_match(nau8821_quirk_table); + if (dmi_id) + nau8821_quirk = (unsigned long)dmi_id->driver_data; +} + static int nau8821_i2c_probe(struct i2c_client *i2c) { struct device *dev = &i2c->dev; @@ -1812,6 +1847,12 @@ static int nau8821_i2c_probe(struct i2c_client *i2c) nau8821->dev = dev; nau8821->irq = i2c->irq; + + nau8821_check_quirks(); + + if (nau8821_quirk & NAU8821_JD_ACTIVE_HIGH) + nau8821->jkdet_polarity = 0; + nau8821_print_device_properties(nau8821); nau8821_reset_chip(nau8821->regmap); -- 2.40.1

1 0

[PATCH AUTOSEL 6.4 16/54] ASoC: da7219: Check for failure reading AAD IRQ events
by Sasha Levin 13 Aug '23

13 Aug '23

From: Dmytro Maluka <dmy(a)semihalf.com> [ Upstream commit f0691dc16206f21b13c464434366e2cd632b8ed7 ] When handling an AAD interrupt, if IRQ events read failed (for example, due to i2c "Transfer while suspended" failure, i.e. when attempting to read it while DA7219 is suspended, which may happen due to a spurious AAD interrupt), the events array contains garbage uninitialized values. So instead of trying to interprete those values and doing any actions based on them (potentially resulting in misbehavior, e.g. reporting bogus events), refuse to handle the interrupt. Signed-off-by: Dmytro Maluka <dmy(a)semihalf.com> Link: https://lore.kernel.org/r/20230717193737.161784-3-dmy@semihalf.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/da7219-aad.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index 07c5d9d354b6d..70c175744772c 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -361,11 +361,15 @@ static irqreturn_t da7219_aad_irq_thread(int irq, void *data) struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); u8 events[DA7219_AAD_IRQ_REG_MAX]; u8 statusa; - int i, report = 0, mask = 0; + int i, ret, report = 0, mask = 0; /* Read current IRQ events */ - regmap_bulk_read(da7219->regmap, DA7219_ACCDET_IRQ_EVENT_A, - events, DA7219_AAD_IRQ_REG_MAX); + ret = regmap_bulk_read(da7219->regmap, DA7219_ACCDET_IRQ_EVENT_A, + events, DA7219_AAD_IRQ_REG_MAX); + if (ret) { + dev_warn_ratelimited(component->dev, "Failed to read IRQ events: %d\n", ret); + return IRQ_NONE; + } if (!events[DA7219_AAD_IRQ_REG_A] && !events[DA7219_AAD_IRQ_REG_B]) return IRQ_NONE; -- 2.40.1

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[PATCH AUTOSEL 6.4 15/54] ASoC: da7219: Flush pending AAD IRQ when suspending
by Sasha Levin 13 Aug '23

13 Aug '23

From: Dmytro Maluka <dmy(a)semihalf.com> [ Upstream commit 91e292917dad64ab8d1d5ca2ab3069ad9dac6f72 ] da7219_aad_suspend() disables jack detection, which should prevent generating new interrupts by DA7219 while suspended. However, there is a theoretical possibility that there is a pending interrupt generated just before suspending DA7219 and not handled yet, so the IRQ handler may still run after DA7219 is suspended. To prevent that, wait until the pending IRQ handling is done. This patch arose as an attempt to fix the following I2C failure occurring sometimes during system suspend or resume: [ 355.876211] i2c_designware i2c_designware.3: Transfer while suspended [ 355.876245] WARNING: CPU: 2 PID: 3576 at drivers/i2c/busses/i2c-designware-master.c:570 i2c_dw_xfer+0x411/0x440 ... [ 355.876462] Call Trace: [ 355.876468] <TASK> [ 355.876475] ? update_load_avg+0x1b3/0x615 [ 355.876484] __i2c_transfer+0x101/0x1d8 [ 355.876494] i2c_transfer+0x74/0x10d [ 355.876504] regmap_i2c_read+0x6a/0x9c [ 355.876513] _regmap_raw_read+0x179/0x223 [ 355.876521] regmap_raw_read+0x1e1/0x28e [ 355.876527] regmap_bulk_read+0x17d/0x1ba [ 355.876532] ? __wake_up+0xed/0x1bb [ 355.876542] da7219_aad_irq_thread+0x54/0x2c9 [snd_soc_da7219 5fb8ebb2179cf2fea29af090f3145d68ed8e2184] [ 355.876556] irq_thread+0x13c/0x231 [ 355.876563] ? irq_forced_thread_fn+0x5f/0x5f [ 355.876570] ? irq_thread_fn+0x4d/0x4d [ 355.876576] kthread+0x13a/0x152 [ 355.876581] ? synchronize_irq+0xc3/0xc3 [ 355.876587] ? kthread_blkcg+0x31/0x31 [ 355.876592] ret_from_fork+0x1f/0x30 [ 355.876601] </TASK> which indicates that the AAD IRQ handler is unexpectedly running when DA7219 is suspended, and as a result, is trying to read data from DA7219 over I2C and is hitting the I2C driver "Transfer while suspended" failure. However, with this patch the above failure is still reproducible. So this patch does not fix any real observed issue so far, but at least is useful for confirming that the above issue is not caused by a pending IRQ but rather looks like a DA7219 hardware issue with an IRQ unexpectedly generated after jack detection is already disabled. Signed-off-by: Dmytro Maluka <dmy(a)semihalf.com> Link: https://lore.kernel.org/r/20230717193737.161784-2-dmy@semihalf.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/da7219-aad.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index 993a0d00bc48d..07c5d9d354b6d 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -910,6 +910,8 @@ void da7219_aad_suspend(struct snd_soc_component *component) } } } + + synchronize_irq(da7219_aad->irq); } void da7219_aad_resume(struct snd_soc_component *component) -- 2.40.1

1 0

"alsactl restore" does not load the driver state
by GitHub issues - edited 13 Aug '23

13 Aug '23

alsa-project/alsa-utils issue #229 was edited from YuGiOhJCJ: Hello, When I type "alsactl restore", it does not load the driver state. Steps to reproduce: 1. Open a first terminal dedicated to alsa-mixer 2. Open a second terminal where you will type the following commands 3. In the first terminal, press F4, set the audio volume for capture to 40 4. In the second terminal, type "sudo alsactl store" in order to save the driver state to /var/lib/alsa/asound.state 5. In the first terminal, set the audio volume for capture to 35 6. In the second terminal, type "sudo alsactl restore" in order to load the driver state from /var/lib/alsa/asound.state Expected results: I should see in the first terminal that the audio volume for capture is restored to 40. Current results: I see in the first terminal that the audio volume for capture is not restored to 40. It is still set to 35! How to fix that please? Thank you. Best regards. About my system: Packages: - alsa-lib-1.2.9 - alsa-plugins-1.2.7.1 - alsa-utils-1.2.9 Operating system: Slackware 14.2 (64-bit) Architecture: x86_64 Kernel: Linux 5.18.10 ![image](https://github.com/alsa-project/alsa-utils/assets/2247896/d4cc947f-b8db-493f-bf03-f18af9c821b3) Issue URL : https://github.com/alsa-project/alsa-utils/issues/229 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

"alsactl restore" does not load the driver state
by GitHub issues - opened 13 Aug '23

13 Aug '23

alsa-project/alsa-utils issue #229 was opened from YuGiOhJCJ: Hello, When I type "alsactl restore", it does not load the driver state. Steps to reproduce: 1. Open a first terminal dedicated to alsa-mixer 2. Open a second terminal where you will type the following commands 3. In the first terminal, press F4, set the audio volume for capture to 40 4. In the second terminal, type "sudo alsactl store" in order to save the driver state to /var/lib/alsa/asound.state 5. In the first terminal, set the audio volume for capture to 35 6. In the second terminal, type "sudo alsactl restore" in order to load the driver state to /var/lib/alsa/asound.state Expected results: I should see in the first terminal that the audio volume for capture is restored to 40. Current results: I see in the first terminal that the audio volume for capture is not restored to 40. It is still set to 35! How to fix that please? Thank you. Best regards. About my system: Packages: - alsa-lib-1.2.9 - alsa-plugins-1.2.7.1 - alsa-utils-1.2.9 Operating system: Slackware 14.2 (64-bit) Architecture: x86_64 Kernel: Linux 5.18.10 ![image](https://github.com/alsa-project/alsa-utils/assets/2247896/d4cc947f-b8db-493f-bf03-f18af9c821b3) Issue URL : https://github.com/alsa-project/alsa-utils/issues/229 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH v2 0/4] Device tree and config updates for the Librem 5 devkit
by Guido Günther 12 Aug '23

12 Aug '23

The device tree updates ensure the A53 cores don't get powered off and fix a DT warning. This isn't testable with the arm64 default config unless we enable the rsi wifi modules too so do this as well. While at that include a binding file update. This is against next-20230809 that include David's option,gtm601 conversion. v2: - drop "dt-bindings: mmc: Fix reference to pwr-seq-simple" already picked up by Ulf Hansson, thanks! - add acked-by from Conor Dooley, thanks! - rework sdio changes as proposed by Sherry Sun Guido Günther (4): dt-bindings: sound: gtm601: Add description arm64: dts: imx8mq-librem5-devkit: Mark buck2 as always on arm64: dts: imx8mq-librem5-devkit: Drop power-supply arm64: defconfig: Enable Redpine 91X wlan driver .../bindings/sound/option,gtm601.yaml | 6 +++-- .../dts/freescale/imx8mq-librem5-devkit.dts | 26 ++++++++----------- arch/arm64/configs/defconfig | 1 + 3 files changed, 16 insertions(+), 17 deletions(-) -- 2.40.1

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[tiwai-sound:test/pcm-iov_iter 3/25] sound/core/memory.c:108:53: warning: passing argument 2 of 'copy_from_iter' makes integer from pointer without a cast
by kernel test robot 12 Aug '23

12 Aug '23

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git test/pcm-iov_iter head: da1be13e050f24d088d24fd65517d56f27a2eeb1 commit: 516e284c8b17a5129fcfbfb1f2f4e11093e13d3f [3/25] ALSA: core: Add memory copy helpers between iov_iter and iomem config: i386-randconfig-r033-20230812 (https://download.01.org/0day-ci/archive/20230812/202308120931.Aj1gaY7K-lkp@…) compiler: gcc-12 (Debian 12.2.0-14) 12.2.0 reproduce: (https://download.01.org/0day-ci/archive/20230812/202308120931.Aj1gaY7K-lkp@…) 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/202308120931.Aj1gaY7K-lkp@intel.com/ All warnings (new ones prefixed by >>): sound/core/memory.c: In function 'copy_from_iter_toio': >> sound/core/memory.c:108:53: warning: passing argument 2 of 'copy_from_iter' makes integer from pointer without a cast [-Wint-conversion] 108 | return !copy_from_iter((void __force *)dst, src, count) ? -EFAULT : 0; | ^~~ | | | struct iov_iter * In file included from include/sound/pcm.h:19, from sound/core/memory.c:12: include/linux/uio.h:199:42: note: expected 'size_t' {aka 'unsigned int'} but argument is of type 'struct iov_iter *' 199 | size_t copy_from_iter(void *addr, size_t bytes, struct iov_iter *i) | ~~~~~~~^~~~~ >> sound/core/memory.c:108:58: warning: passing argument 3 of 'copy_from_iter' makes pointer from integer without a cast [-Wint-conversion] 108 | return !copy_from_iter((void __force *)dst, src, count) ? -EFAULT : 0; | ^~~~~ | | | size_t {aka unsigned int} include/linux/uio.h:199:66: note: expected 'struct iov_iter *' but argument is of type 'size_t' {aka 'unsigned int'} 199 | size_t copy_from_iter(void *addr, size_t bytes, struct iov_iter *i) | ~~~~~~~~~~~~~~~~~^ vim +/copy_from_iter +108 sound/core/memory.c 93 94 /** 95 * copy_from_iter_toio - copy data from iov_iter to mmio-space 96 * @dst: the destination pointer on mmio-space 97 * @src: the source iov_iter 98 * @count: the data size to copy in bytes 99 * 100 * Copies the data from iov_iter to mmio-space. 101 * Only the single segment iov_iter is assumed for now. 102 * 103 * Return: Zero if successful, or non-zero on failure. 104 */ 105 int copy_from_iter_toio(void __iomem *dst, struct iov_iter *src, size_t count) 106 { 107 #if defined(__i386__) || defined(CONFIG_SPARC32) > 108 return !copy_from_iter((void __force *)dst, src, count) ? -EFAULT : 0; -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

MSI x570s Carbon Max WiFi ALC 4080
by GitHub issues - edited 11 Aug '23

11 Aug '23

alsa-project/alsa-ucm-conf issue #338 was edited from zpackrat: Fedora 38 patched to current, issue has existed for months. Rear output jacks for Center/Sub and Rear Speakers are reversed. I use a 5.1 speaker setup. When I run sound tests in the gnome audio settings the issue is verifiable. Also audio that requires center channel for dialogue has issues. This is specifically the rear audio I/O ports. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/338 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

MSI x570s Carbon Max WiFi ALC 4080
by GitHub issues - opened 11 Aug '23

11 Aug '23

alsa-project/alsa-ucm-conf issue #338 was opened from zpackrat: Fedora 38 patched to current, issue has existed for months. Rear output jacks for Center/Sub and Rear Speakers are reversed. I use a 5.1 speaker setup. When I run sound tests in the gnome audio settings the issue is verifiable. Also audio that requires center channel for dialogue has issues. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/338 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[RFC] pcm: Add MSBITS subformat options
by GitHub pull_request - edited 11 Aug '23

11 Aug '23

alsa-project/alsa-lib pull request #342 was edited from crojewsk-intel: Improve granularity of format selection for S32/U32 formats by adding masks representing 20, 24 and 32 most significant bits. As a preparation step introduce `snd_pcm_subformat_value()` for easy name-to-subformat conversions. While at it, fix few warnings that pop up in subformat-related code during compilation. Request URL : https://github.com/alsa-project/alsa-lib/pull/342 Patch URL : https://github.com/alsa-project/alsa-lib/pull/342.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[RFC] pcm: Add MSBITS subformat options
by GitHub pull_request - edited 11 Aug '23

11 Aug '23

alsa-project/alsa-lib pull request #342 was edited from crojewsk-intel: Improve granularity of format selection for S32/U32 formats by adding masks representing 20 and 24 most significant bits. As a preparation step introduce `snd_pcm_subformat_value()` for easy name-to-subformat conversions. While at it, fix few warnings that pop up in subformat-related code during compilation. Request URL : https://github.com/alsa-project/alsa-lib/pull/342 Patch URL : https://github.com/alsa-project/alsa-lib/pull/342.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v1] ALSA: hda/cs8409: Support new Dell Dolphin Variants
by Stefan Binding 11 Aug '23

11 Aug '23

Add 4 new Dell Dolphin Systems, same configuration as older systems. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> --- sound/pci/hda/patch_cs8409-tables.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_cs8409-tables.c b/sound/pci/hda/patch_cs8409-tables.c index b288874e401e..36b411d1a960 100644 --- a/sound/pci/hda/patch_cs8409-tables.c +++ b/sound/pci/hda/patch_cs8409-tables.c @@ -550,6 +550,10 @@ const struct snd_pci_quirk cs8409_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0C50, "Dolphin", CS8409_DOLPHIN), SND_PCI_QUIRK(0x1028, 0x0C51, "Dolphin", CS8409_DOLPHIN), SND_PCI_QUIRK(0x1028, 0x0C52, "Dolphin", CS8409_DOLPHIN), + SND_PCI_QUIRK(0x1028, 0x0C73, "Dolphin", CS8409_DOLPHIN), + SND_PCI_QUIRK(0x1028, 0x0C75, "Dolphin", CS8409_DOLPHIN), + SND_PCI_QUIRK(0x1028, 0x0C7D, "Dolphin", CS8409_DOLPHIN), + SND_PCI_QUIRK(0x1028, 0x0C7F, "Dolphin", CS8409_DOLPHIN), {} /* terminator */ }; -- 2.34.1

2 1

[PATCH v2 0/3] soundwire: allow for more than 8 devices, keep IDA for wake-capable devices
by Bard Liao 11 Aug '23

11 Aug '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 (3): soundwire: extend parameters of new_peripheral_assigned() callback soundwire: bus: add callbacks for device_number allocation soundwire: intel_auxdevice: add hybrid IDA-based device_number allocation v2: - Move the device numver allocation from bus.c to intel_auxdevice.c - List the codecs with wake capability in a quirk table, and assign device number according to that quirk table. drivers/soundwire/bus.c | 18 +++--- drivers/soundwire/intel_auxdevice.c | 85 ++++++++++++++++++++++++++--- include/linux/soundwire/sdw.h | 12 ++-- include/linux/soundwire/sdw_intel.h | 7 +++ 4 files changed, 99 insertions(+), 23 deletions(-) -- 2.25.1

2 4

[PATCH 0/2] soundwire: improve pm_runtime handling
by Bard Liao 11 Aug '23

11 Aug '23

This patchset improves the pm_runtime behavior in rare corner cases identified by the Intel CI in the last 6 months. a) in stress-tests, it's not uncommon to see the following type of warnings when the codec reports as ATTACHED "rt711 sdw:0:025d:0711:00: runtime PM trying to activate child device sdw:0:025d:0711:00 but parent (sdw-master-0) is not active" This warning was not correlated with any functional issue, but it exposed a design issue on when to enable pm_runtime. The recommended practice in the pm_runtime documentation is to keep the devices in 'suspended' mode and mark them as 'active' when they are really functional. Pierre-Louis Bossart (2): soundwire: intel_auxdevice: enable pm_runtime earlier on startup soundWire: intel_auxdevice: resume 'sdw-master' on startup and system resume drivers/soundwire/intel_auxdevice.c | 27 +++++++++++++++++++-------- 1 file changed, 19 insertions(+), 8 deletions(-) -- 2.25.1

3 5

[PATCH -next] ASoC: SOF: Intel: Remove duplicated include in lnl.c
by Yang Li 10 Aug '23

10 Aug '23

./sound/soc/sof/intel/lnl.c: hda.h is included more than once. Signed-off-by: Yang Li <yang.lee(a)linux.alibaba.com> --- sound/soc/sof/intel/lnl.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/sof/intel/lnl.c b/sound/soc/sof/intel/lnl.c index 3d919b0b6891..db94b45e53af 100644 --- a/sound/soc/sof/intel/lnl.c +++ b/sound/soc/sof/intel/lnl.c @@ -16,7 +16,6 @@ #include "hda-ipc.h" #include "../sof-audio.h" #include "mtl.h" -#include "hda.h" #include <sound/hda-mlink.h> /* LunarLake ops */ -- 2.20.1.7.g153144c

3 2

[PATCH v1 0/5] Device tree and config updates for the Librem 5 devkit
by Guido Günther 10 Aug '23

10 Aug '23

The device tree updates ensure the A53 don't get powered off and fix a DT warning. This isn't testable with the arm64 default config unless we enable the rsi wifi modules too so do this as well. While at that include two binding file updates. This is against next-20230809 that include David's option,gtm601 conversion. Guido Günther (5): dt-bindings: sound: gtm601: Add description dt-bindings: mmc: Fix reference to pwr-seq-simple arm64: dts: imx8mq-librem5-devkit: Mark buck2 as always on arm64: dts: imx8mq-librem5-devkit: Drop power-supply arm64: defconfig: Enable Redpine 91X wlan driver .../devicetree/bindings/mmc/mmc-controller.yaml | 2 +- .../devicetree/bindings/sound/option,gtm601.yaml | 6 ++++-- .../boot/dts/freescale/imx8mq-librem5-devkit.dts | 14 +++++--------- arch/arm64/configs/defconfig | 1 + 4 files changed, 11 insertions(+), 12 deletions(-) -- 2.40.1

6 10

[PATCH] ASoC: rt715: Add software reset in io init
by Jack Yu 10 Aug '23

10 Aug '23

Add software reset before setting preset registers to make sure all the registers are the default value before preset. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt715.c | 56 ++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt715.h | 7 +++++ 2 files changed, 63 insertions(+) diff --git a/sound/soc/codecs/rt715.c b/sound/soc/codecs/rt715.c index 79416bb48814..c95f481cd235 100644 --- a/sound/soc/codecs/rt715.c +++ b/sound/soc/codecs/rt715.c @@ -52,6 +52,60 @@ static int rt715_index_write(struct regmap *regmap, unsigned int reg, return ret; } +static int rt715_index_write_nid(struct regmap *regmap, + unsigned int nid, unsigned int reg, unsigned int value) +{ + int ret; + unsigned int addr = ((RT715_PRIV_INDEX_W_H_2 | nid) << 8) | reg; + + ret = regmap_write(regmap, addr, value); + if (ret < 0) + pr_err("Failed to set private value: %06x <= %04x ret=%d\n", + addr, value, ret); + + return ret; +} + +static int rt715_index_read_nid(struct regmap *regmap, + unsigned int nid, unsigned int reg, unsigned int *value) +{ + int ret; + unsigned int addr = ((RT715_PRIV_INDEX_W_H_2 | nid) << 8) | reg; + + *value = 0; + ret = regmap_read(regmap, addr, value); + if (ret < 0) + pr_err("Failed to get private value: %06x => %04x ret=%d\n", + addr, *value, ret); + + return ret; +} + +static int rt715_index_update_bits(struct regmap *regmap, unsigned int nid, + unsigned int reg, unsigned int mask, unsigned int val) +{ + unsigned int tmp, orig; + int ret; + + ret = rt715_index_read_nid(regmap, nid, reg, &orig); + if (ret < 0) + return ret; + + tmp = orig & ~mask; + tmp |= val & mask; + + return rt715_index_write_nid(regmap, nid, reg, tmp); +} + +static void rt715_reset(struct regmap *regmap) +{ + regmap_write(regmap, RT715_FUNC_RESET, 0); + rt715_index_update_bits(regmap, RT715_VENDOR_REGISTERS, + RT715_VD_CLEAR_CTRL, RT715_CLEAR_HIDDEN_REG, + RT715_CLEAR_HIDDEN_REG); +} + + static void rt715_get_gain(struct rt715_priv *rt715, unsigned int addr_h, unsigned int addr_l, unsigned int val_h, unsigned int *r_val, unsigned int *l_val) @@ -1040,6 +1094,8 @@ int rt715_io_init(struct device *dev, struct sdw_slave *slave) pm_runtime_get_noresume(&slave->dev); + rt715_reset(rt715->regmap); + /* Mute nid=08h/09h */ regmap_write(rt715->regmap, RT715_SET_GAIN_LINE_ADC_H, 0xb080); regmap_write(rt715->regmap, RT715_SET_GAIN_MIX_ADC_H, 0xb080); diff --git a/sound/soc/codecs/rt715.h b/sound/soc/codecs/rt715.h index 12a0ae656d09..6e37bf64e12f 100644 --- a/sound/soc/codecs/rt715.h +++ b/sound/soc/codecs/rt715.h @@ -48,6 +48,7 @@ struct rt715_priv { #define RT715_INLINE_CMD 0x55 /* Index (NID:20h) */ +#define RT715_VD_CLEAR_CTRL 0x01 #define RT715_SDW_INPUT_SEL 0x39 #define RT715_EXT_DMIC_CLK_CTRL2 0x54 @@ -71,6 +72,8 @@ struct rt715_priv { #define RT715_READ_HDA_0 0x2015 #define RT715_PRIV_INDEX_W_H 0x7520 #define RT715_PRIV_INDEX_W_L 0x85a0 +#define RT715_PRIV_INDEX_W_H_2 0x7500 +#define RT715_PRIV_INDEX_W_L_2 0x8580 #define RT715_PRIV_DATA_W_H 0x7420 #define RT715_PRIV_DATA_W_L 0x84a0 #define RT715_PRIV_INDEX_R_H 0x9d20 @@ -198,6 +201,10 @@ struct rt715_priv { #define RT715_SET_DMIC4_CONFIG_DEFAULT4\ (RT715_VERB_SET_CONFIG_DEFAULT4 | RT715_DMIC4) +/* vendor register clear ctrl-1 (0x01)(NID:20h) */ +#define RT715_CLEAR_HIDDEN_REG (0x1 << 15) + + #define RT715_MUTE_SFT 7 #define RT715_DIR_IN_SFT 6 #define RT715_DIR_OUT_SFT 7 -- 2.34.1

2 1

[PATCH] ASoC: meson: axg-tdm-formatter: fix channel slot allocation
by Jerome Brunet 10 Aug '23

10 Aug '23

When the tdm lane mask is computed, the driver currently fills the 1st lane before moving on to the next. If the stream has less channels than the lanes can accommodate, slots will be disabled on the last lanes. Unfortunately, the HW distribute channels in a different way. It distribute channels in pair on each lanes before moving on the next slots. This difference leads to problems if a device has an interface with more than 1 lane and with more than 2 slots per lane. For example: a playback interface with 2 lanes and 4 slots each (total 8 slots - zero based numbering) - Playing a 8ch stream: - All slots activated by the driver - channel #2 will be played on lane #1 - slot #0 following HW placement - Playing a 4ch stream: - Lane #1 disabled by the driver - channel #2 will be played on lane #0 - slot #2 This behaviour is obviously not desirable. Change the way slots are activated on the TDM lanes to follow what the HW does and make sure each channel always get mapped to the same slot/lane. Fixes: 1a11d88f499c ("ASoC: meson: add tdm formatter base driver") Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/axg-tdm-formatter.c | 42 ++++++++++++++++++----------- 1 file changed, 26 insertions(+), 16 deletions(-) diff --git a/sound/soc/meson/axg-tdm-formatter.c b/sound/soc/meson/axg-tdm-formatter.c index 9883dc777f63..63333a2b0a9c 100644 --- a/sound/soc/meson/axg-tdm-formatter.c +++ b/sound/soc/meson/axg-tdm-formatter.c @@ -30,27 +30,32 @@ int axg_tdm_formatter_set_channel_masks(struct regmap *map, struct axg_tdm_stream *ts, unsigned int offset) { - unsigned int val, ch = ts->channels; - unsigned long mask; - int i, j; + unsigned int ch = ts->channels; + u32 val[AXG_TDM_NUM_LANES]; + int i, j, k; + + /* + * We need to mimick the slot distribution used by the HW to keep the + * channel placement consistent regardless of the number of channel + * in the stream. This is why the odd algorithm below is used. + */ + memset(val, 0, sizeof(*val) * AXG_TDM_NUM_LANES); /* * Distribute the channels of the stream over the available slots - * of each TDM lane + * of each TDM lane. We need to go over the 32 slots ... */ - for (i = 0; i < AXG_TDM_NUM_LANES; i++) { - val = 0; - mask = ts->mask[i]; - - for (j = find_first_bit(&mask, 32); - (j < 32) && ch; - j = find_next_bit(&mask, 32, j + 1)) { - val |= 1 << j; - ch -= 1; + for (i = 0; (i < 32) && ch; i += 2) { + /* ... of all the lanes ... */ + for (j = 0; j < AXG_TDM_NUM_LANES; j++) { + /* ... then distribute the channels in pairs */ + for (k = 0; k < 2; k++) { + if ((BIT(i + k) & ts->mask[j]) && ch) { + val[j] |= BIT(i + k); + ch -= 1; + } + } } - - regmap_write(map, offset, val); - offset += regmap_get_reg_stride(map); } /* @@ -63,6 +68,11 @@ int axg_tdm_formatter_set_channel_masks(struct regmap *map, return -EINVAL; } + for (i = 0; i < AXG_TDM_NUM_LANES; i++) { + regmap_write(map, offset, val[i]); + offset += regmap_get_reg_stride(map); + } + return 0; } EXPORT_SYMBOL_GPL(axg_tdm_formatter_set_channel_masks); -- 2.40.1

2 1

[PATCH] ASoC: SOF: Intel: Initialize chip in hda_sdw_check_wakeen_irq()
by Nathan Chancellor 10 Aug '23

10 Aug '23

Clang warns (or errors with CONFIG_WERROR): sound/soc/sof/intel/hda.c:423:6: error: variable 'chip' is uninitialized when used here [-Werror,-Wuninitialized] 423 | if (chip && chip->check_sdw_wakeen_irq) | ^~~~ sound/soc/sof/intel/hda.c:418:39: note: initialize the variable 'chip' to silence this warning 418 | const struct sof_intel_dsp_desc *chip; | ^ | = NULL 1 error generated. Add the missing initialization, following the pattern of the other irq functions. Fixes: 9362ab78f175 ("ASoC: SOF: Intel: add abstraction for SoundWire wake-ups") Signed-off-by: Nathan Chancellor <nathan(a)kernel.org> --- sound/soc/sof/intel/hda.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 04c748a72b13..15e6779efaa3 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -420,6 +420,7 @@ static bool hda_sdw_check_wakeen_irq(struct snd_sof_dev *sdev) if (!(interface_mask & BIT(SOF_DAI_INTEL_ALH))) return false; + chip = get_chip_info(sdev->pdata); if (chip && chip->check_sdw_wakeen_irq) return chip->check_sdw_wakeen_irq(sdev); --- base-commit: 59146c3cd326a622e9041614842346aada11ca99 change-id: 20230809-intel-hda-missing-chip-init-dcccbe6365a4 Best regards, -- Nathan Chancellor <nathan(a)kernel.org>

3 7

[PATCH -next] ASoC: SOF: amd: Remove duplicated include
by GUO Zihua 10 Aug '23

10 Aug '23

Remove duplicated include of sof-audio.h. Resolves checkincludes message. Signed-off-by: GUO Zihua <guozihua(a)huawei.com> --- sound/soc/sof/amd/acp-common.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/sof/amd/acp-common.c b/sound/soc/sof/amd/acp-common.c index 3a0c7688dcfe..2d72c6d55dc8 100644 --- a/sound/soc/sof/amd/acp-common.c +++ b/sound/soc/sof/amd/acp-common.c @@ -13,7 +13,6 @@ #include "../sof-priv.h" #include "../sof-audio.h" #include "../ops.h" -#include "../sof-audio.h" #include "acp.h" #include "acp-dsp-offset.h" #include <sound/sof/xtensa.h> -- 2.17.1

1 0

[PATCH] ALSA: dice: add stream format parameters for Weiss devices
by Takashi Sakamoto 10 Aug '23

10 Aug '23

This patch is for kernel prepatch v6.5. Hard-coded stream format parameters are added for Weiss Engineering FireWire devices. When the device vendor and model match, the parameters are copied into the stream format cache. This allows for setting all supported sampling rates up to 192kHz, and consequently adjusting the number of available I/O channels. Signed-off-by: Rolf Anderegg <rolf.anderegg(a)weiss.ch> Signed-off-by: Michele Perrone <michele.perrone(a)weiss.ch> Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/dice/Makefile | 2 +- sound/firewire/dice/dice-weiss.c | 104 +++++++++++++++++++++++++++++++ sound/firewire/dice/dice.c | 63 +++++++++++++++++++ sound/firewire/dice/dice.h | 1 + 4 files changed, 169 insertions(+), 1 deletion(-) create mode 100644 sound/firewire/dice/dice-weiss.c diff --git a/sound/firewire/dice/Makefile b/sound/firewire/dice/Makefile index a5f3fbf28b8c..bac8712f9014 100644 --- a/sound/firewire/dice/Makefile +++ b/sound/firewire/dice/Makefile @@ -2,5 +2,5 @@ snd-dice-objs := dice-transaction.o dice-stream.o dice-proc.o dice-midi.o \ dice-pcm.o dice-hwdep.o dice.o dice-tcelectronic.o \ dice-alesis.o dice-extension.o dice-mytek.o dice-presonus.o \ - dice-harman.o dice-focusrite.o + dice-harman.o dice-focusrite.o dice-weiss.o obj-$(CONFIG_SND_DICE) += snd-dice.o diff --git a/sound/firewire/dice/dice-weiss.c b/sound/firewire/dice/dice-weiss.c new file mode 100644 index 000000000000..129d43408956 --- /dev/null +++ b/sound/firewire/dice/dice-weiss.c @@ -0,0 +1,104 @@ +// SPDX-License-Identifier: GPL-2.0 +// dice-weiss.c - a part of driver for DICE based devices +// +// Copyright (c) 2023 Rolf Anderegg and Michele Perrone + +#include "dice.h" + +struct dice_weiss_spec { + unsigned int tx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT]; + unsigned int rx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT]; +}; + +// Weiss DAC202: 192kHz 2-channel DAC +static const struct dice_weiss_spec dac202 = { + .tx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, +}; + +// Weiss MAN301: 192kHz 2-channel music archive network player +static const struct dice_weiss_spec man301 = { + .tx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, +}; + +// Weiss INT202: 192kHz unidirectional 2-channel digital Firewire nterface +static const struct dice_weiss_spec int202 = { + .tx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, +}; + +// Weiss INT203: 192kHz bidirectional 2-channel digital Firewire nterface +static const struct dice_weiss_spec int203 = { + .tx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, +}; + +// Weiss ADC2: 192kHz A/D converter with microphone preamps and line nputs +static const struct dice_weiss_spec adc2 = { + .tx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, +}; + +// Weiss DAC2/Minerva: 192kHz 2-channel DAC +static const struct dice_weiss_spec dac2_minerva = { + .tx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, +}; + +// Weiss Vesta: 192kHz 2-channel Firewire to AES/EBU interface +static const struct dice_weiss_spec vesta = { + .tx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, +}; + +// Weiss AFI1: 192kHz 24-channel Firewire to ADAT or AES/EBU interface +static const struct dice_weiss_spec afi1 = { + .tx_pcm_chs = {{24, 16, 8}, {0, 0, 0} }, + .rx_pcm_chs = {{24, 16, 8}, {0, 0, 0} }, +}; + +int snd_dice_detect_weiss_formats(struct snd_dice *dice) +{ + static const struct { + u32 model_id; + const struct dice_weiss_spec *spec; + } *entry, entries[] = { + {0x000007, &dac202}, + {0x000008, &dac202}, // Maya edition: same audio I/O as DAC202. + {0x000006, &int202}, + {0x00000a, &int203}, + {0x00000b, &man301}, + {0x000001, &adc2}, + {0x000003, &dac2_minerva}, + {0x000002, &vesta}, + {0x000004, &afi1}, + }; + struct fw_csr_iterator it; + int key, val, model_id; + int i; + + model_id = 0; + fw_csr_iterator_init(&it, dice->unit->directory); + while (fw_csr_iterator_next(&it, &key, &val)) { + if (key == CSR_MODEL) { + model_id = val; + break; + } + } + + for (i = 0; i < ARRAY_SIZE(entries); ++i) { + entry = entries + i; + if (entry->model_id == model_id) + break; + } + if (i == ARRAY_SIZE(entries)) + return -ENODEV; + + memcpy(dice->tx_pcm_chs, entry->spec->tx_pcm_chs, + MAX_STREAMS * SND_DICE_RATE_MODE_COUNT * sizeof(unsigned int)); + memcpy(dice->rx_pcm_chs, entry->spec->rx_pcm_chs, + MAX_STREAMS * SND_DICE_RATE_MODE_COUNT * sizeof(unsigned int)); + + return 0; +} diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index 6c93e6e4982c..d362e4251c68 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -392,6 +392,69 @@ static const struct ieee1394_device_id dice_id_table[] = { .model_id = 0x0000de, .driver_data = (kernel_ulong_t)snd_dice_detect_focusrite_pro40_tcd3070_formats, }, + // Weiss DAC202: 192kHz 2-channel DAC + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x000007, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + // Weiss DAC202: 192kHz 2-channel DAC (Maya edition) + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x000008, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + // Weiss MAN301: 192kHz 2-channel music archive network player + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x00000b, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + // Weiss INT202: 192kHz unidirectional 2-channel digital Firewire face + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x000006, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + // Weiss INT203: 192kHz bidirectional 2-channel digital Firewire face + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x00000a, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + // Weiss ADC2: 192kHz A/D converter with microphone preamps and inputs + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x000001, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + // Weiss DAC2/Minerva: 192kHz 2-channel DAC + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x000003, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + // Weiss Vesta: 192kHz 2-channel Firewire to AES/EBU interface + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x000002, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + // Weiss AFI1: 192kHz 24-channel Firewire to ADAT or AES/EBU face + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x000004, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, { .match_flags = IEEE1394_MATCH_VERSION, .version = DICE_INTERFACE, diff --git a/sound/firewire/dice/dice.h b/sound/firewire/dice/dice.h index 674f7d552c2e..4c0ad7335998 100644 --- a/sound/firewire/dice/dice.h +++ b/sound/firewire/dice/dice.h @@ -232,5 +232,6 @@ int snd_dice_detect_mytek_formats(struct snd_dice *dice); int snd_dice_detect_presonus_formats(struct snd_dice *dice); int snd_dice_detect_harman_formats(struct snd_dice *dice); int snd_dice_detect_focusrite_pro40_tcd3070_formats(struct snd_dice *dice); +int snd_dice_detect_weiss_formats(struct snd_dice *dice); #endif -- 2.39.2

2 6

[PATCH v2] ASoC: SOF: ipc4-topology: Update the basecfg for copier earlier
by Peter Ujfalusi 10 Aug '23

10 Aug '23

The sof_ipc4_update_resource_usage() call updates the CPC value in basecfg and it must be done prior to making a copy of the copier configuration for the init message. Other module types do the resource update as last step or in case of a process module at the correct time, before the memcpy. Fixes: d8a2c9879349 ("ASoC: SOF: ipc4-loader/topology: Query the CPC value from manifest") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Adrian Bonislawski <adrian.bonislawski(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Chao Song <chao.song(a)linux.intel.com> --- Hi, changes since v1: - created on top of for-6.5 branch The conflict caused by this for 6.6 is trivial as it is just a move two lines patch. Regards, Peter sound/soc/sof/ipc4-topology.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index a4e1a70b607d..11361e1cd688 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -1731,6 +1731,9 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, *ipc_config_size = ipc_size; + /* update pipeline memory usage */ + sof_ipc4_update_resource_usage(sdev, swidget, &copier_data->base_config); + /* copy IPC data */ memcpy(*ipc_config_data, (void *)copier_data, sizeof(*copier_data)); if (gtw_cfg_config_length) @@ -1743,9 +1746,6 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, gtw_cfg_config_length, &ipc4_copier->dma_config_tlv, dma_config_tlv_size); - /* update pipeline memory usage */ - sof_ipc4_update_resource_usage(sdev, swidget, &copier_data->base_config); - return 0; } -- 2.41.0

2 1

[PATCH] ASoC: SOF: ipc4-topology: Update the basecfg for copier earlier
by Peter Ujfalusi 10 Aug '23

10 Aug '23

The sof_ipc4_update_resource_usage() call updates the CPC value in basecfg and it must be done prior to making a copy of the copier configuration for the init message. Other module types do the resource update as last step or in case of a process module at the correct time, before the memcpy. Fixes: d8a2c9879349 ("ASoC: SOF: ipc4-loader/topology: Query the CPC value from manifest") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Adrian Bonislawski <adrian.bonislawski(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Chao Song <chao.song(a)linux.intel.com> --- sound/soc/sof/ipc4-topology.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 0196cbfc0998..633f6040d712 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -1731,6 +1731,9 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, *ipc_config_size = ipc_size; + /* update pipeline memory usage */ + sof_ipc4_update_resource_usage(sdev, swidget, &copier_data->base_config); + /* copy IPC data */ memcpy(*ipc_config_data, (void *)copier_data, sizeof(*copier_data)); if (gtw_cfg_config_length) @@ -1743,9 +1746,6 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, gtw_cfg_config_length, &ipc4_copier->dma_config_tlv, dma_config_tlv_size); - /* update pipeline memory usage */ - sof_ipc4_update_resource_usage(sdev, swidget, &copier_data->base_config); - /* * Restore gateway config length now that IPC payload is prepared. This avoids * counting the DMA CONFIG TLV multiple times -- 2.41.0

2 4

[PATCH] MAINTAINERS: update Claudiu Beznea's email address
by Claudiu Beznea 10 Aug '23

10 Aug '23

Update MAINTAINERS entries with a valid email address as the Microchip one is no longer valid. Acked-by: Conor Dooley <conor.dooley(a)microchip.com> Acked-by: Nicolas Ferre <nicolas.ferre(a)microchip.com> Signed-off-by: Claudiu Beznea <claudiu.beznea(a)tuxon.dev> --- Changes in v2: - collected tags - extended the recipients list to include individual subsystem maintainers and lists instead using only linux-kernel(a)vger.kernel.org as suggested initially by get_maintainers.pl MAINTAINERS | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 128fd295f86f..d48d8e857f57 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -2351,7 +2351,7 @@ F: drivers/phy/mediatek/ ARM/MICROCHIP (ARM64) SoC support M: Conor Dooley <conor(a)kernel.org> M: Nicolas Ferre <nicolas.ferre(a)microchip.com> -M: Claudiu Beznea <claudiu.beznea(a)microchip.com> +M: Claudiu Beznea <claudiu.beznea(a)tuxon.dev> L: linux-arm-kernel(a)lists.infradead.org (moderated for non-subscribers) S: Supported T: git https://git.kernel.org/pub/scm/linux/kernel/git/at91/linux.git @@ -2360,7 +2360,7 @@ F: arch/arm64/boot/dts/microchip/ ARM/Microchip (AT91) SoC support M: Nicolas Ferre <nicolas.ferre(a)microchip.com> M: Alexandre Belloni <alexandre.belloni(a)bootlin.com> -M: Claudiu Beznea <claudiu.beznea(a)microchip.com> +M: Claudiu Beznea <claudiu.beznea(a)tuxon.dev> L: linux-arm-kernel(a)lists.infradead.org (moderated for non-subscribers) S: Supported W: http://www.linux4sam.org @@ -3265,7 +3265,7 @@ F: include/uapi/linux/atm* ATMEL MACB ETHERNET DRIVER M: Nicolas Ferre <nicolas.ferre(a)microchip.com> -M: Claudiu Beznea <claudiu.beznea(a)microchip.com> +M: Claudiu Beznea <claudiu.beznea(a)tuxon.dev> S: Supported F: drivers/net/ethernet/cadence/ @@ -13880,7 +13880,7 @@ F: Documentation/devicetree/bindings/serial/atmel,at91-usart.yaml F: drivers/spi/spi-at91-usart.c MICROCHIP AUDIO ASOC DRIVERS -M: Claudiu Beznea <claudiu.beznea(a)microchip.com> +M: Claudiu Beznea <claudiu.beznea(a)tuxon.dev> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Supported F: Documentation/devicetree/bindings/sound/atmel* @@ -13903,7 +13903,7 @@ S: Maintained F: drivers/crypto/atmel-ecc.* MICROCHIP EIC DRIVER -M: Claudiu Beznea <claudiu.beznea(a)microchip.com> +M: Claudiu Beznea <claudiu.beznea(a)tuxon.dev> L: linux-arm-kernel(a)lists.infradead.org (moderated for non-subscribers) S: Supported F: Documentation/devicetree/bindings/interrupt-controller/microchip,sama7g5-eic.yaml @@ -13976,7 +13976,7 @@ F: drivers/video/fbdev/atmel_lcdfb.c F: include/video/atmel_lcdc.h MICROCHIP MCP16502 PMIC DRIVER -M: Claudiu Beznea <claudiu.beznea(a)microchip.com> +M: Claudiu Beznea <claudiu.beznea(a)tuxon.dev> L: linux-arm-kernel(a)lists.infradead.org (moderated for non-subscribers) S: Supported F: Documentation/devicetree/bindings/regulator/mcp16502-regulator.txt @@ -14003,7 +14003,7 @@ F: Documentation/devicetree/bindings/mtd/atmel-nand.txt F: drivers/mtd/nand/raw/atmel/* MICROCHIP OTPC DRIVER -M: Claudiu Beznea <claudiu.beznea(a)microchip.com> +M: Claudiu Beznea <claudiu.beznea(a)tuxon.dev> L: linux-arm-kernel(a)lists.infradead.org (moderated for non-subscribers) S: Supported F: Documentation/devicetree/bindings/nvmem/microchip,sama7g5-otpc.yaml @@ -14042,7 +14042,7 @@ F: Documentation/devicetree/bindings/fpga/microchip,mpf-spi-fpga-mgr.yaml F: drivers/fpga/microchip-spi.c MICROCHIP PWM DRIVER -M: Claudiu Beznea <claudiu.beznea(a)microchip.com> +M: Claudiu Beznea <claudiu.beznea(a)tuxon.dev> L: linux-arm-kernel(a)lists.infradead.org (moderated for non-subscribers) L: linux-pwm(a)vger.kernel.org S: Supported @@ -14058,7 +14058,7 @@ F: drivers/iio/adc/at91-sama5d2_adc.c F: include/dt-bindings/iio/adc/at91-sama5d2_adc.h MICROCHIP SAMA5D2-COMPATIBLE SHUTDOWN CONTROLLER -M: Claudiu Beznea <claudiu.beznea(a)microchip.com> +M: Claudiu Beznea <claudiu.beznea(a)tuxon.dev> S: Supported F: Documentation/devicetree/bindings/power/reset/atmel,sama5d2-shdwc.yaml F: drivers/power/reset/at91-sama5d2_shdwc.c @@ -14075,7 +14075,7 @@ S: Supported F: drivers/spi/spi-atmel.* MICROCHIP SSC DRIVER -M: Claudiu Beznea <claudiu.beznea(a)microchip.com> +M: Claudiu Beznea <claudiu.beznea(a)tuxon.dev> L: linux-arm-kernel(a)lists.infradead.org (moderated for non-subscribers) S: Supported F: Documentation/devicetree/bindings/misc/atmel-ssc.txt @@ -14104,7 +14104,7 @@ F: drivers/usb/gadget/udc/atmel_usba_udc.* MICROCHIP WILC1000 WIFI DRIVER M: Ajay Singh <ajay.kathat(a)microchip.com> -M: Claudiu Beznea <claudiu.beznea(a)microchip.com> +M: Claudiu Beznea <claudiu.beznea(a)tuxon.dev> L: linux-wireless(a)vger.kernel.org S: Supported F: drivers/net/wireless/microchip/wilc1000/ -- 2.39.2

6 7

[PATCH v2 0/2] Add HP G11 Quirks and update Dell Oasis Quirks
by Stefan Binding 10 Aug '23

10 Aug '23

Add quirks to support HP G11 using CS35l41 HDA, using both SPI and I2C. One older HP laptop has an SSID removed. In addition, all Dell Oasis laptops using CS35L41 have been switch to use SPI, with one SSID removed. Changes from v1: - Add fixes tags Stefan Binding (2): ALSA: hda/realtek: Add quirks for HP G11 Laptops ALSA: hda/realtek: Switch Dell Oasis models to use SPI sound/pci/hda/patch_realtek.c | 17 +++++++++++------ 1 file changed, 11 insertions(+), 6 deletions(-) -- 2.34.1

2 3

[PATCH v3 0/3] ASoC: mt8188-mt6359: add SOF support
by Trevor Wu 10 Aug '23

10 Aug '23

This series introduces dynamic pinctrl and adds support for the SOF on the mt8188-mt6359 machine driver. Changes since v2: - revise mtk_sof_check_tplg_be_dai_link_fixup(). Changes since v1: - minor changes to incorporate the suggestions from the reviewer. Trevor Wu (3): ASoC: mediatek: mt8188-mt6359: support dynamic pinctrl ASoC: mediatek: common: revise SOF common code ASoC: mediatek: mt8188-mt6359: add SOF support .../soc/mediatek/common/mtk-dsp-sof-common.c | 113 +++++++-- .../soc/mediatek/common/mtk-dsp-sof-common.h | 8 + sound/soc/mediatek/mt8188/mt8188-mt6359.c | 239 +++++++++++++++++- 3 files changed, 332 insertions(+), 28 deletions(-) -- 2.18.0

1 3

Question: license compliance for libatopology 2.0.0
by GitHub issues - opened 09 Aug '23

09 Aug '23

alsa-project/alsa-lib issue #343 was opened from lucasgonze: Given a binary linked with libatopology 2.0.0, and given that the LGPL requires source code to be bundled, it is unclear how to find the corresponding repo and tag (or release). Thanks in advance for tips. Issue URL : https://github.com/alsa-project/alsa-lib/issues/343 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: samsung: midas_wm1811: Fix 'Headphone Switch' control creation
by Marek Szyprowski 09 Aug '23

09 Aug '23

'Headphone Switch' control is already registered from sound/soc/codecs/wm_hubs.c:479, so duplicating it in midas_wm1811 causes following probe failure: midas-audio sound: control 2:0:0:Headphone Switch:0 is already present midas-audio sound: ASoC: Failed to add Headphone Switch: -16 midas-audio sound: Failed to register card: -16 midas-audio: probe of sound failed with error -16 Fix this by dropping duplicated control. Fixes: d27224a45e54 ("ASoC: samsung: midas_wm1811: Map missing jack kcontrols") Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> --- sound/soc/samsung/midas_wm1811.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/samsung/midas_wm1811.c b/sound/soc/samsung/midas_wm1811.c index 44b32f5cddca..126098fdcf1b 100644 --- a/sound/soc/samsung/midas_wm1811.c +++ b/sound/soc/samsung/midas_wm1811.c @@ -257,7 +257,6 @@ static const struct snd_kcontrol_new midas_controls[] = { SOC_DAPM_PIN_SWITCH("Main Mic"), SOC_DAPM_PIN_SWITCH("Sub Mic"), SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("Headphone"), SOC_DAPM_PIN_SWITCH("FM In"), }; -- 2.34.1

2 1

[PATCH] ASoC: rsnd: call of_node_put() when break
by Kuninori Morimoto 09 Aug '23

09 Aug '23

We need to call of_node_put() when break from for_each_child_of_node(). This patch add missing of_node_put(). Reported-by: kernel test robot <lkp(a)intel.com> Reported-by: Julia Lawall <julia.lawall(a)inria.fr> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 069add7b1b71..e29c2fee9521 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1282,6 +1282,7 @@ static int rsnd_dai_of_node(struct rsnd_priv *priv, int *is_graph) i++; if (i >= RSND_MAX_COMPONENT) { dev_info(dev, "reach to max component\n"); + of_node_put(node); break; } } @@ -1301,6 +1302,7 @@ static int rsnd_dai_of_node(struct rsnd_priv *priv, int *is_graph) i++; if (i >= RSND_MAX_COMPONENT) { dev_info(dev, "reach to max component\n"); + of_node_put(node); break; } } -- 2.25.1

2 1

[PATCH 0/5] ASoC: cs35l56: Bugfixes
by Richard Fitzgerald 09 Aug '23

09 Aug '23

Miscellaneous bugfixes for the cs35l56 codec driver. Richard Fitzgerald (3): ASoC: cs35l56: Avoid uninitialized variable in cs35l56_set_asp_slot_positions() ASoC: cs35l56: Don't rely on GPIOD_OUT_LOW to set RESET initially low ASoC: cs35l56: Wait for control port ready during system-resume Simon Trimmer (2): ASoC: wm_adsp: Expose the DSP power down actions as wm_adsp_power_down() ASoC: cs35l56: Call wm_adsp_power_down() before reloading firmware sound/soc/codecs/cs35l56.c | 33 +++++++++++++++++---------------- sound/soc/codecs/wm_adsp.c | 8 +++++++- sound/soc/codecs/wm_adsp.h | 1 + 3 files changed, 25 insertions(+), 17 deletions(-) -- 2.30.2

2 6

[PATCH v1 0/2] Add HP G11 Quirks and update Dell Oasis Quirks
by Stefan Binding 09 Aug '23

09 Aug '23

Add quirks to support HP G11 using CS35l41 HDA, using both SPI and I2C. One older HP laptop has an SSID removed. In addition, all Dell Oasis laptops using CS35L41 have been switch to use SPI, with one SSID removed. Stefan Binding (2): ALSA: hda/realtek: Add quirks for HP G11 Laptops ALSA: hda/realtek: Switch Dell Oasis models to use SPI sound/pci/hda/patch_realtek.c | 17 +++++++++++------ 1 file changed, 11 insertions(+), 6 deletions(-) -- 2.34.1

2 3

[PATCH] ASoC: codecs: aw88261: avoid uninitialized variable warning
by Arnd Bergmann 08 Aug '23

08 Aug '23

From: Arnd Bergmann <arnd(a)arndb.de> aw88261_reg_update() returns an unintialized error code in the success path: sound/soc/codecs/aw88261.c:651:7: error: variable 'ret' is used uninitialized whenever 'if' condition is false [-Werror,-Wsometimes-uninitialized] if (aw_dev->prof_cur != aw_dev->prof_index) { ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/aw88261.c:660:9: note: uninitialized use occurs here return ret; ^~~ sound/soc/codecs/aw88261.c:651:3: note: remove the 'if' if its condition is always true if (aw_dev->prof_cur != aw_dev->prof_index) { ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ Return zero instead here. Fixes: 028a2ae256916 ("ASoC: codecs: Add aw88261 amplifier driver") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/aw88261.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/aw88261.c b/sound/soc/codecs/aw88261.c index 82923b454dd47..6e2266b713862 100644 --- a/sound/soc/codecs/aw88261.c +++ b/sound/soc/codecs/aw88261.c @@ -652,6 +652,8 @@ static int aw88261_reg_update(struct aw88261 *aw88261, bool force) ret = aw88261_dev_fw_update(aw88261); if (ret) return ret; + } else { + ret = 0; } } -- 2.39.2

3 2

[PATCH 01/12] ASoC: intel: sof_sdw: Printk's should end with a newline
by Charles Keepax 08 Aug '23

08 Aug '23

Add the missing new lines. Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/intel/boards/sof_sdw.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index fd27e211211bd..8f3329dcf7062 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -524,7 +524,7 @@ int sdw_prepare(struct snd_pcm_substream *substream) sdw_stream = snd_soc_dai_get_stream(dai, substream->stream); if (IS_ERR(sdw_stream)) { - dev_err(rtd->dev, "no stream found for DAI %s", dai->name); + dev_err(rtd->dev, "no stream found for DAI %s\n", dai->name); return PTR_ERR(sdw_stream); } @@ -543,7 +543,7 @@ int sdw_trigger(struct snd_pcm_substream *substream, int cmd) sdw_stream = snd_soc_dai_get_stream(dai, substream->stream); if (IS_ERR(sdw_stream)) { - dev_err(rtd->dev, "no stream found for DAI %s", dai->name); + dev_err(rtd->dev, "no stream found for DAI %s\n", dai->name); return PTR_ERR(sdw_stream); } @@ -565,7 +565,7 @@ int sdw_trigger(struct snd_pcm_substream *substream, int cmd) } if (ret) - dev_err(rtd->dev, "%s trigger %d failed: %d", __func__, cmd, ret); + dev_err(rtd->dev, "%s trigger %d failed: %d\n", __func__, cmd, ret); return ret; } @@ -630,7 +630,7 @@ int sdw_hw_free(struct snd_pcm_substream *substream) sdw_stream = snd_soc_dai_get_stream(dai, substream->stream); if (IS_ERR(sdw_stream)) { - dev_err(rtd->dev, "no stream found for DAI %s", dai->name); + dev_err(rtd->dev, "no stream found for DAI %s\n", dai->name); return PTR_ERR(sdw_stream); } @@ -1339,7 +1339,7 @@ static int get_slave_info(const struct snd_soc_acpi_link_adr *adr_link, return -EINVAL; if (index >= SDW_MAX_CPU_DAIS) { - dev_err(dev, " cpu_dai_id array overflows"); + dev_err(dev, "cpu_dai_id array overflows\n"); return -EINVAL; } @@ -1490,7 +1490,7 @@ static int create_sdw_dailink(struct snd_soc_card *card, int *link_index, return -ENOMEM; if (cpu_dai_index >= sdw_cpu_dai_num) { - dev_err(dev, "invalid cpu dai index %d", + dev_err(dev, "invalid cpu dai index %d\n", cpu_dai_index); return -EINVAL; } @@ -1503,12 +1503,12 @@ static int create_sdw_dailink(struct snd_soc_card *card, int *link_index, * not be larger than sdw_be_num */ if (*link_index >= sdw_be_num) { - dev_err(dev, "invalid dai link index %d", *link_index); + dev_err(dev, "invalid dai link index %d\n", *link_index); return -EINVAL; } if (*cpu_id >= sdw_cpu_dai_num) { - dev_err(dev, " invalid cpu dai index %d", *cpu_id); + dev_err(dev, "invalid cpu dai index %d\n", *cpu_id); return -EINVAL; } @@ -1531,7 +1531,7 @@ static int create_sdw_dailink(struct snd_soc_card *card, int *link_index, ret = set_codec_init_func(card, adr_link, dai_links + (*link_index)++, playback, group_id, adr_index, dai_index); if (ret < 0) { - dev_err(dev, "failed to init codec %d", codec_index); + dev_err(dev, "failed to init codec %d\n", codec_index); return ret; } @@ -1675,7 +1675,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) endpoint = adr_link->adr_d[i].endpoints; if (endpoint->aggregated && !endpoint->group_id) { - dev_err(dev, "invalid group id on link %x", + dev_err(dev, "invalid group id on link %x\n", adr_link->mask); continue; } @@ -1698,7 +1698,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) &be_id, &codec_conf_index, &ignore_pch_dmic, append_dai_type, i, j); if (ret < 0) { - dev_err(dev, "failed to create dai link %d", link_index); + dev_err(dev, "failed to create dai link %d\n", link_index); return ret; } } -- 2.30.2

2 12

[PATCH -next] ASoC: imx-audio-rpmsg: Remove redundant initialization owner in imx_audio_rpmsg_driver
by Li Zetao 08 Aug '23

08 Aug '23

The module_rpmsg_driver() will set "THIS_MODULE" to driver.owner when register a rpmsg_driver driver, so it is redundant initialization to set driver.owner in the statement. Remove it for clean code. Signed-off-by: Li Zetao <lizetao1(a)huawei.com> --- sound/soc/fsl/imx-audio-rpmsg.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/fsl/imx-audio-rpmsg.c b/sound/soc/fsl/imx-audio-rpmsg.c index d5234ac4b09b..289e47c03d40 100644 --- a/sound/soc/fsl/imx-audio-rpmsg.c +++ b/sound/soc/fsl/imx-audio-rpmsg.c @@ -116,7 +116,6 @@ static struct rpmsg_device_id imx_audio_rpmsg_id_table[] = { static struct rpmsg_driver imx_audio_rpmsg_driver = { .drv.name = "imx_audio_rpmsg", - .drv.owner = THIS_MODULE, .id_table = imx_audio_rpmsg_id_table, .probe = imx_audio_rpmsg_probe, .callback = imx_audio_rpmsg_cb, -- 2.34.1

2 1

[linux-next:master] BUILD REGRESSION 71cd4fc492ec41e4acd85e98bbf7a13753fc1e03
by kernel test robot 08 Aug '23

08 Aug '23

tree/branch: https://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git master branch HEAD: 71cd4fc492ec41e4acd85e98bbf7a13753fc1e03 Add linux-next specific files for 20230808 Error/Warning reports: https://lore.kernel.org/oe-kbuild-all/202307251531.p8ZLFTMZ-lkp@intel.com https://lore.kernel.org/oe-kbuild-all/202308081459.US5rLYAY-lkp@intel.com Error/Warning: (recently discovered and may have been fixed) ../lib/gcc/loongarch64-linux/12.3.0/plugin/include/config/loongarch/loongarch-opts.h:31:10: fatal error: loongarch-def.h: No such file or directory drivers/gpu/drm/amd/amdgpu/../display/dc/dce/dmub_replay.c:37: warning: This comment starts with '/**', but isn't a kernel-doc comment. Refer Documentation/doc-guide/kernel-doc.rst sound/soc/codecs/aw88261.c:651:7: warning: variable 'ret' is used uninitialized whenever 'if' condition is false [-Wsometimes-uninitialized] Unverified Error/Warning (likely false positive, please contact us if interested): drivers/gpu/drm/tests/drm_exec_test.c:166 test_prepare_array() error: uninitialized symbol 'ret'. drivers/mtd/nand/raw/qcom_nandc.c:2590 qcom_op_cmd_mapping() error: uninitialized symbol 'ret'. drivers/mtd/nand/raw/qcom_nandc.c:3017 qcom_check_op() warn: was && intended here instead of ||? kernel/futex/waitwake.c:422 futex_wait_multiple_setup() warn: bitwise AND condition is false here sh4-linux-gcc: internal compiler error: Segmentation fault signal terminated program cc1 {standard input}: Warning: end of file not at end of a line; newline inserted {standard input}:927: Error: pcrel too far Error/Warning ids grouped by kconfigs: gcc_recent_errors |-- alpha-allyesconfig | `-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dce-dmub_replay.c:warning:This-comment-starts-with-but-isn-t-a-kernel-doc-comment.-Refer-Documentation-doc-guide-kernel-doc.rst |-- arc-allyesconfig | `-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dce-dmub_replay.c:warning:This-comment-starts-with-but-isn-t-a-kernel-doc-comment.-Refer-Documentation-doc-guide-kernel-doc.rst |-- arm-allmodconfig | `-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dce-dmub_replay.c:warning:This-comment-starts-with-but-isn-t-a-kernel-doc-comment.-Refer-Documentation-doc-guide-kernel-doc.rst |-- arm-allyesconfig | `-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dce-dmub_replay.c:warning:This-comment-starts-with-but-isn-t-a-kernel-doc-comment.-Refer-Documentation-doc-guide-kernel-doc.rst |-- arm64-allyesconfig | `-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dce-dmub_replay.c:warning:This-comment-starts-with-but-isn-t-a-kernel-doc-comment.-Refer-Documentation-doc-guide-kernel-doc.rst |-- csky-randconfig-m041-20230808 | `-- drivers-gpu-drm-tests-drm_exec_test.c-test_prepare_array()-error:uninitialized-symbol-ret-. |-- i386-allyesconfig | `-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dce-dmub_replay.c:warning:This-comment-starts-with-but-isn-t-a-kernel-doc-comment.-Refer-Documentation-doc-guide-kernel-doc.rst |-- loongarch-allmodconfig | `-- lib-gcc-loongarch64-linux-..-plugin-include-config-loongarch-loongarch-opts.h:fatal-error:loongarch-def.h:No-such-file-or-directory |-- mips-allmodconfig | `-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dce-dmub_replay.c:warning:This-comment-starts-with-but-isn-t-a-kernel-doc-comment.-Refer-Documentation-doc-guide-kernel-doc.rst |-- mips-allyesconfig | `-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dce-dmub_replay.c:warning:This-comment-starts-with-but-isn-t-a-kernel-doc-comment.-Refer-Documentation-doc-guide-kernel-doc.rst |-- parisc-allyesconfig | `-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dce-dmub_replay.c:warning:This-comment-starts-with-but-isn-t-a-kernel-doc-comment.-Refer-Documentation-doc-guide-kernel-doc.rst |-- powerpc-allmodconfig | `-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dce-dmub_replay.c:warning:This-comment-starts-with-but-isn-t-a-kernel-doc-comment.-Refer-Documentation-doc-guide-kernel-doc.rst |-- riscv-allmodconfig | `-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dce-dmub_replay.c:warning:This-comment-starts-with-but-isn-t-a-kernel-doc-comment.-Refer-Documentation-doc-guide-kernel-doc.rst |-- riscv-allyesconfig | `-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dce-dmub_replay.c:warning:This-comment-starts-with-but-isn-t-a-kernel-doc-comment.-Refer-Documentation-doc-guide-kernel-doc.rst |-- riscv-randconfig-m031-20230808 | |-- drivers-gpu-drm-tests-drm_exec_test.c-test_prepare_array()-error:uninitialized-symbol-ret-. | |-- drivers-mtd-nand-raw-qcom_nandc.c-qcom_check_op()-warn:was-intended-here-instead-of | |-- drivers-mtd-nand-raw-qcom_nandc.c-qcom_op_cmd_mapping()-error:uninitialized-symbol-ret-. | `-- kernel-futex-waitwake.c-futex_wait_multiple_setup()-warn:bitwise-AND-condition-is-false-here |-- s390-allyesconfig | `-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dce-dmub_replay.c:warning:This-comment-starts-with-but-isn-t-a-kernel-doc-comment.-Refer-Documentation-doc-guide-kernel-doc.rst |-- s390-randconfig-r044-20230808 | `-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dce-dmub_replay.c:warning:This-comment-starts-with-but-isn-t-a-kernel-doc-comment.-Refer-Documentation-doc-guide-kernel-doc.rst |-- sh-allmodconfig | |-- sh4-linux-gcc:internal-compiler-error:Segmentation-fault-signal-terminated-program-cc1 | |-- standard-input:Error:pcrel-too-far | `-- standard-input:Warning:end-of-file-not-at-end-of-a-line-newline-inserted |-- sparc-allyesconfig | `-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dce-dmub_replay.c:warning:This-comment-starts-with-but-isn-t-a-kernel-doc-comment.-Refer-Documentation-doc-guide-kernel-doc.rst |-- x86_64-allyesconfig | `-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dce-dmub_replay.c:warning:This-comment-starts-with-but-isn-t-a-kernel-doc-comment.-Refer-Documentation-doc-guide-kernel-doc.rst `-- xtensa-randconfig-r022-20230808 `-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dce-dmub_replay.c:warning:This-comment-starts-with-but-isn-t-a-kernel-doc-comment.-Refer-Documentation-doc-guide-kernel-doc.rst clang_recent_errors `-- arm64-randconfig-r006-20230808 `-- sound-soc-codecs-aw88261.c:warning:variable-ret-is-used-uninitialized-whenever-if-condition-is-false elapsed time: 721m configs tested: 106 configs skipped: 5 tested configs: alpha allyesconfig gcc alpha defconfig gcc arc allyesconfig gcc arc defconfig gcc arc randconfig-r034-20230808 gcc arc randconfig-r043-20230808 gcc arm allmodconfig gcc arm allyesconfig gcc arm defconfig gcc arm randconfig-r012-20230808 clang arm randconfig-r046-20230808 clang arm64 allyesconfig gcc arm64 defconfig gcc arm64 randconfig-r001-20230808 clang csky defconfig gcc csky randconfig-r024-20230808 gcc hexagon randconfig-r015-20230808 clang hexagon randconfig-r031-20230808 clang hexagon randconfig-r041-20230808 clang hexagon randconfig-r045-20230808 clang i386 allyesconfig gcc i386 buildonly-randconfig-r004-20230808 clang i386 buildonly-randconfig-r005-20230808 clang i386 buildonly-randconfig-r006-20230808 clang i386 debian-10.3 gcc i386 defconfig gcc i386 randconfig-i001-20230808 clang i386 randconfig-i002-20230808 clang i386 randconfig-i003-20230808 clang i386 randconfig-i004-20230808 clang i386 randconfig-i005-20230808 clang i386 randconfig-i006-20230808 clang i386 randconfig-i011-20230808 gcc i386 randconfig-i012-20230808 gcc i386 randconfig-i013-20230808 gcc i386 randconfig-i014-20230808 gcc i386 randconfig-i015-20230808 gcc i386 randconfig-i016-20230808 gcc i386 randconfig-r014-20230808 gcc loongarch allmodconfig gcc loongarch allnoconfig gcc loongarch defconfig gcc m68k allmodconfig gcc m68k allyesconfig gcc m68k defconfig gcc m68k randconfig-r004-20230808 gcc mips allmodconfig gcc mips allyesconfig gcc nios2 defconfig gcc parisc allyesconfig gcc parisc defconfig gcc parisc randconfig-r002-20230808 gcc parisc randconfig-r016-20230808 gcc parisc64 defconfig gcc powerpc allmodconfig gcc powerpc allnoconfig gcc powerpc randconfig-r005-20230808 clang riscv allmodconfig gcc riscv allnoconfig gcc riscv allyesconfig gcc riscv defconfig gcc riscv randconfig-r042-20230808 gcc riscv rv32_defconfig gcc s390 allmodconfig gcc s390 allyesconfig gcc s390 defconfig gcc s390 randconfig-r011-20230808 gcc s390 randconfig-r023-20230808 gcc s390 randconfig-r044-20230808 gcc sh allmodconfig gcc sh randconfig-r006-20230808 gcc sh randconfig-r025-20230808 gcc sparc allyesconfig gcc sparc defconfig gcc um allmodconfig clang um allnoconfig clang um allyesconfig clang um defconfig gcc um i386_defconfig gcc um randconfig-r033-20230808 gcc um x86_64_defconfig gcc x86_64 allyesconfig gcc x86_64 buildonly-randconfig-r001-20230808 clang x86_64 buildonly-randconfig-r002-20230808 clang x86_64 buildonly-randconfig-r003-20230808 clang x86_64 defconfig gcc x86_64 kexec gcc x86_64 randconfig-r003-20230808 clang x86_64 randconfig-r021-20230808 gcc x86_64 randconfig-x001-20230808 gcc x86_64 randconfig-x002-20230808 gcc x86_64 randconfig-x003-20230808 gcc x86_64 randconfig-x004-20230808 gcc x86_64 randconfig-x005-20230808 gcc x86_64 randconfig-x006-20230808 gcc x86_64 randconfig-x011-20230808 clang x86_64 randconfig-x012-20230808 clang x86_64 randconfig-x013-20230808 clang x86_64 randconfig-x014-20230808 clang x86_64 randconfig-x015-20230808 clang x86_64 randconfig-x016-20230808 clang x86_64 rhel-8.3-rust clang x86_64 rhel-8.3 gcc xtensa randconfig-r013-20230808 gcc xtensa randconfig-r022-20230808 gcc xtensa randconfig-r035-20230808 gcc -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[PATCH] ASoC: amd: acp3x-rt5682-max9836: Configure jack as not detecting Line Out
by Alper Nebi Yasak 08 Aug '23

08 Aug '23

The RT5682, RT1015 and RT1015p codecs used in this driver do not seem capable of distinguishing Line Out connections from Headphone, but the driver configures its jack object as if it can. Remove the wrong value from the jack creation call to avoid any confusion. Signed-off-by: Alper Nebi Yasak <alpernebiyasak(a)gmail.com> --- Depends on [1] which applies onto of linux-next. Untested, see [2]: [1] https://lore.kernel.org/alsa-devel/20230802175737.263412-5-alpernebiyasak@g… [2] https://lore.kernel.org/alsa-devel/3d6794c9-60d3-b8f3-99d2-afdc69593d9f@gma… sound/soc/amd/acp3x-rt5682-max9836.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index 3bf7e1e83c36..28ad5f5b9a76 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -98,7 +98,7 @@ static int acp3x_5682_init(struct snd_soc_pcm_runtime *rtd) rt5682_dai_bclk = clk_get(component->dev, "rt5682-dai-bclk"); ret = snd_soc_card_jack_new_pins(card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_LINEOUT | + SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3, &pco_jack, base-commit: 626c67169f9972fffcdf3bc3864de421f162ebf5 prerequisite-patch-id: 7158530856f8f97324906b342528a5ffe7b4032b -- 2.40.1

4 3

[PATCH 00/20] ASoC: SOF: Intel: add LunarLake support
by Pierre-Louis Bossart 08 Aug '23

08 Aug '23

IMPORTANT NOTE: this patchset is dependent on Takashi's hda-intel-6.6 tag. The PCI parts will not compile without merging this tag into the ASoC tree. This patchset first fixes a number of errors made in the hda-mlink support, then adds Lunar Lake definitions. The main contribution is the hda-dai changes where the HDaudio DMA is now used for SSP, DMIC and SoundWire. In previous hardware the GPDMA (aka DesignWare) was used and controlled by the audio firmware. The volume of code is minimized with the abstraction added in previous kernel cycles. Due to cross-dependencies between ASoC and SoundWire trees, the full support for jack detection will be deferred to the next kernel cycle. There's not much point to ask for a sync of the two trees to support one patch for each tree - we are at -rc5 already. Pierre-Louis Bossart (18): ASoC: SOF: Intel: hda-mlink: fix off-by-one error ASoC: SOF: Intel: fix u16/32 confusion in LSDIID ASoC: SOF: ipc4-topology: restore gateway config length ASoC: SOF: Intel: hda: add interface definitions for ACE2.x ASoC: SOF: Intel: split MTL and LNL operations ASoC: SOF: Intel: LNL: enable DMIC/SSP offload in probe/resume ASoC: SOF: Intel: hda-dai-ops: add/select DMA ops for SSP ASoC: SOF: Intel: hda-dai: add ops for SSP ASoC: SOF: Intel: hda-dai: add DMIC support ASoC: SOF: Intel: hda-dai-ops: only allocate/release streams for first CPU DAI ASoC: SOF: Intel: hda-dai-ops: add ops for SoundWire ASoC: SOF: Intel: hda-dai: add helpers for SoundWire callbacks ASoC: SOF: Intel: hda: add hw_params/free/trigger callbacks ASoC: SOF: Intel: add abstraction for SoundWire wake-ups ASoC: SOF: Intel: hda-mlink: add helper to get sublink LSDIID register ASoC: SOF: Intel: hda-dai-ops: reset device count for SoundWire DAIs ASoC: SOF: IPC4: clarify 'pipeline_ids' usage and logs ASoC: SOF: Intel: hda-mlink: add sublink to dev_dbg() log Ranjani Sridharan (2): ASoC: SOF: ipc4: avoid uninitialized default instance 0 ASoC: SOF: Intel: LNL: Add support for Lunarlake platform include/sound/hda-mlink.h | 4 + sound/soc/sof/intel/Kconfig | 16 +++ sound/soc/sof/intel/Makefile | 4 +- sound/soc/sof/intel/cnl.c | 2 + sound/soc/sof/intel/hda-dai-ops.c | 179 ++++++++++++++++++++++++- sound/soc/sof/intel/hda-dai.c | 211 +++++++++++++++++++++++++++++- sound/soc/sof/intel/hda-mlink.c | 45 ++++++- sound/soc/sof/intel/hda.c | 58 +++++++- sound/soc/sof/intel/hda.h | 21 +++ sound/soc/sof/intel/icl.c | 1 + sound/soc/sof/intel/lnl.c | 189 ++++++++++++++++++++++++++ sound/soc/sof/intel/mtl.c | 23 ++-- sound/soc/sof/intel/mtl.h | 22 +++- sound/soc/sof/intel/pci-lnl.c | 71 ++++++++++ sound/soc/sof/intel/shim.h | 1 + sound/soc/sof/intel/tgl.c | 4 + sound/soc/sof/ipc4-pcm.c | 21 +-- sound/soc/sof/ipc4-topology.c | 7 + sound/soc/sof/ipc4-topology.h | 4 +- sound/soc/sof/topology.c | 2 + 20 files changed, 842 insertions(+), 43 deletions(-) create mode 100644 sound/soc/sof/intel/lnl.c create mode 100644 sound/soc/sof/intel/pci-lnl.c -- 2.39.2

2 23

[PATCH 0/2] ASoC: Intel: sof_sdw: fix two settings
by Pierre-Louis Bossart 08 Aug '23

08 Aug '23

Fix button mapping for CS42L42 and jack detection for LunarLake RVP. Pierre-Louis Bossart (1): ASoC: Intel: sof-sdw: update jack detection quirk for LunarLake RVP jairaj-arava (1): ASoC: Intel: sof-sdw-cs42142: fix for codec button mapping sound/soc/intel/boards/sof_sdw.c | 2 +- sound/soc/intel/boards/sof_sdw_cs42l42.c | 6 +++--- 2 files changed, 4 insertions(+), 4 deletions(-) -- 2.39.2

2 5

[PATCH -next] ASoC: SOF: ipc3-dtrace: Switch to memdup_user_nul() helper
by Yang Yingliang 08 Aug '23

08 Aug '23

Use memdup_user_nul() helper instead of open-coding to simplify the code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/sof/ipc3-dtrace.c | 12 +++--------- 1 file changed, 3 insertions(+), 9 deletions(-) diff --git a/sound/soc/sof/ipc3-dtrace.c b/sound/soc/sof/ipc3-dtrace.c index 35da85a45a9a..bd07f0472efd 100644 --- a/sound/soc/sof/ipc3-dtrace.c +++ b/sound/soc/sof/ipc3-dtrace.c @@ -196,15 +196,9 @@ static ssize_t dfsentry_trace_filter_write(struct file *file, const char __user return -EINVAL; } - string = kmalloc(count + 1, GFP_KERNEL); - if (!string) - return -ENOMEM; - - if (copy_from_user(string, from, count)) { - ret = -EFAULT; - goto error; - } - string[count] = '\0'; + string = memdup_user_nul(from, count); + if (IS_ERR(string)) + return PTR_ERR(string); ret = trace_filter_parse(sdev, string, &num_elems, &elems); if (ret < 0) -- 2.25.1

2 1

Regression fix not progressing towards mainline/stable
by Thorsten Leemhuis 08 Aug '23

08 Aug '23

Hi everyone, I noticed a regression fix that afaics lingers in the thesofproject's git repo instead of progressing towards the stable trees and wondered what's up here. I'm talking about about the fix "ASoC: SOF: intel: hda: Clean up link DMA for IPC3 during stop" for this ticket: https://github.com/thesofproject/linux/issues/4455 Two more tickets about it: https://github.com/thesofproject/linux/issues/4482 https://bugzilla.kernel.org/show_bug.cgi?id=217673 It seems the fix is ready and reviewed for two weeks now, but not even in -next by now. That's not how it should be for regression fixes, as per https://docs.kernel.org/process/handling-regressions.html it ideally should be in -stable soon or already. Is this a mistake, am I missing something, or is there a good reason for this? Ciao, Thorsten

2 2

[PATCH] ALSA: hda/tegra: refactor deprecated strncpy
by Justin Stitt 08 Aug '23

08 Aug '23

`strncpy` is deprecated for use on NUL-terminated destination strings [1]. A suitable replacement is `strscpy` [2] due to the fact that it guarantees NUL-termination on its destination buffer argument which is _not_ the case for `strncpy`! It should be noted that the current implementation is unlikely to have a bug because `drv_name` is a string literal with a size of 9 while `card->driver` has a size of 16. However, it is probably worthwhile to switch to a more robust and less ambiguous interface. [1]: www.kernel.org/doc/html/latest/process/deprecated.html#strncpy-on-nul-termi… [2]: https://manpages.debian.org/testing/linux-manual-4.8/strscpy.9.en.html Link: https://github.com/KSPP/linux/issues/90 Cc: linux-hardening(a)vger.kernel.org Signed-off-by: Justin Stitt <justinstitt(a)google.com> --- sound/pci/hda/hda_tegra.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 9d0ab043880b..f16740500d46 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -378,14 +378,14 @@ static int hda_tegra_first_init(struct azx *chip, struct platform_device *pdev) } /* driver name */ - strncpy(card->driver, drv_name, sizeof(card->driver)); + strscpy(card->driver, drv_name, sizeof(card->driver)); /* shortname for card */ sname = of_get_property(np, "nvidia,model", NULL); if (!sname) sname = drv_name; if (strlen(sname) > sizeof(card->shortname)) dev_info(card->dev, "truncating shortname for card\n"); - strncpy(card->shortname, sname, sizeof(card->shortname)); + strscpy(card->shortname, sname, sizeof(card->shortname)); /* longname for card */ snprintf(card->longname, sizeof(card->longname), --- base-commit: c1a515d3c0270628df8ae5f5118ba859b85464a2 change-id: 20230803-sound-pci-hda-c3a712b72a35 Best regards, -- Justin Stitt <justinstitt(a)google.com>

2 1

[PATCH -next] ALSA: info: Remove unused function declarations
by Yue Haibing 08 Aug '23

08 Aug '23

These declarations is never used since beginning of git history. Signed-off-by: Yue Haibing <yuehaibing(a)huawei.com> --- include/sound/info.h | 2 -- 1 file changed, 2 deletions(-) diff --git a/include/sound/info.h b/include/sound/info.h index 7c13bf52cc81..adbc506860d6 100644 --- a/include/sound/info.h +++ b/include/sound/info.h @@ -118,8 +118,6 @@ struct snd_info_entry *snd_info_create_card_entry(struct snd_card *card, const char *name, struct snd_info_entry *parent); void snd_info_free_entry(struct snd_info_entry *entry); -int snd_info_store_text(struct snd_info_entry *entry); -int snd_info_restore_text(struct snd_info_entry *entry); int snd_info_card_create(struct snd_card *card); int snd_info_card_register(struct snd_card *card); -- 2.34.1

2 1

[PATCH v2 1/2] ALSA: pcmtest: Move buffer iterator initialization to prepare callback
by Ivan Orlov 08 Aug '23

08 Aug '23

Trigger callback is not the best place for buffer iterator initialization, so move it out to the prepare callback, where it have to be. Minor enhancement: remove blank line. Signed-off-by: Ivan Orlov <ivan.orlov0322(a)gmail.com> --- V1 -> V2: - Split changes in the patch into two different patches sound/drivers/pcmtest.c | 33 ++++++++++++++++----------------- 1 file changed, 16 insertions(+), 17 deletions(-) diff --git a/sound/drivers/pcmtest.c b/sound/drivers/pcmtest.c index 08e14b5eb772..dc7c9c758537 100644 --- a/sound/drivers/pcmtest.c +++ b/sound/drivers/pcmtest.c @@ -92,7 +92,6 @@ MODULE_PARM_DESC(inject_trigger_err, "Inject EINVAL error in the 'trigger' callb module_param(inject_open_err, bool, 0600); MODULE_PARM_DESC(inject_open_err, "Inject EBUSY error in the 'open' callback"); - struct pcmtst { struct snd_pcm *pcm; struct snd_card *card; @@ -405,25 +404,9 @@ static int snd_pcmtst_pcm_close(struct snd_pcm_substream *substream) static int snd_pcmtst_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct pcmtst_buf_iter *v_iter = runtime->private_data; - if (inject_trigger_err) return -EINVAL; - v_iter->sample_bytes = runtime->sample_bits / 8; - v_iter->period_bytes = frames_to_bytes(runtime, runtime->period_size); - if (runtime->access == SNDRV_PCM_ACCESS_RW_NONINTERLEAVED || - runtime->access == SNDRV_PCM_ACCESS_MMAP_NONINTERLEAVED) { - v_iter->chan_block = runtime->dma_bytes / runtime->channels; - v_iter->interleaved = false; - } else { - v_iter->interleaved = true; - } - // We want to record RATE * ch_cnt samples per sec, it is rate * sample_bytes * ch_cnt bytes - v_iter->s_rw_ch = runtime->rate / TIMER_PER_SEC; - v_iter->b_rw = v_iter->s_rw_ch * v_iter->sample_bytes * runtime->channels; - return 0; } @@ -454,8 +437,24 @@ static void pcmtst_pdev_release(struct device *dev) static int snd_pcmtst_pcm_prepare(struct snd_pcm_substream *substream) { + struct snd_pcm_runtime *runtime = substream->runtime; + struct pcmtst_buf_iter *v_iter = runtime->private_data; + if (inject_prepare_err) return -EINVAL; + + v_iter->sample_bytes = samples_to_bytes(runtime, 1); + v_iter->period_bytes = snd_pcm_lib_period_bytes(substream); + v_iter->interleaved = true; + if (runtime->access == SNDRV_PCM_ACCESS_RW_NONINTERLEAVED || + runtime->access == SNDRV_PCM_ACCESS_MMAP_NONINTERLEAVED) { + v_iter->chan_block = snd_pcm_lib_buffer_bytes(substream) / runtime->channels; + v_iter->interleaved = false; + } + // We want to record RATE * ch_cnt samples per sec, it is rate * sample_bytes * ch_cnt bytes + v_iter->s_rw_ch = runtime->rate / TIMER_PER_SEC; + v_iter->b_rw = v_iter->s_rw_ch * v_iter->sample_bytes * runtime->channels; + return 0; } -- 2.34.1

2 2

[PATCH v2 00/28] Add support for QMC HDLC, framer infrastruture and PEF2256 framer
by Herve Codina 08 Aug '23

08 Aug '23

Hi, I have a system where I need to handle an HDLC interface and some audio data. The HDLC data are transferred using a TDM bus on which a PEF2256 (E1/T1 framer) is present. The PEF2256 transfers data from/to the TDM bus to/from the E1 line. This PEF2256 is connected to a PowerQUICC SoC for the control path and the TDM is connected to the SoC (QMC component) for the data path. From the QMC HDLC driver, I need to handle HDLC data using the QMC, carrier detection using the PEF2256 (E1 line carrier) and set/get some PEF2256 configuration. The QMC HDLC driver considers the PEF2256 as a generic framer. It performs operations that involve the PEF2256 through the generic framer API. The audio data are exchanged with the PEF2256 using a CPU DAI connected to the TDM bus through the QMC and the PEF2256 needs to be seen as a codec in order to be linked to the CPU DAI. The codec handles the carrier detection using the PEF2256 and reports the carrier state to the ALSA subsystem using the ASoC jack detection. The codec, even if instantiated by the PEF2256 driver, considers the PEF2256 as a generic framer. The generic framer has: - 2 consumers (QMC HDLC drv and codec) - 1 provider (PEF2256) So, the design is the following: +------------------+ +---------+ | QMC | <- TDM -> | PEF2256 | <-> E1 +---------+ | +-------------+ | | | | CPU DAI | <-data--> | QMC channel | | | | +---------+ | +-------------+ | | | +--------------+ | +-------------+ | | | | QMC HDLC drv | <-data--> | QMC channel | | | | +--------------+ | +-------------+ | | | ^ +------------------+ | | | +--------+ +-------------+ | | +-> | framer | <-> | PEF2256 drv | <- local bus ->| | | | | | +---------+ +-> | | | | | +--------+ | +-------+ | +-------------------> | codec | | | +-------+ | +-------------+ Further more, the TDM timeslots used by the QMC HDLC driver need to be configured at runtime (QMC dynamic timeslots). Several weeks ago, I sent two series related to this topic: - Add the Lantiq PEF2256 audio support [1] - RFC Add support for QMC HDLC and PHY [2] This current series is a rework of these two series taking into account feedbacks previously received. In order to implement all of this, I do the following: 1) Perform some fixes (patches 1, 2, 3, 4) 2) Introduce the QMC HDLC driver (patches 5, 6, 7) 3) Add QMC dynamic timeslot support (patches 8 - 18) 4) Add timeslots change support in QMC HDLC (patch 19) 5) Introduce framer infrastructure (patch 20) 6) Add PEF2256 framer provider (patches 11, 22, 23, 24, 25) 7) Add framer codec as a framer consumer (patch 26) 8) Add framer support as a framer consumer in QMC HDLC (patch 27, 28) The series contains the full story and detailed modifications. If needed, the series can be split and/or commmits can be squashed. Let me know. Compare to the previous iteration https://lore.kernel.org/linux-kernel/20230725092417.43706-1-herve.codina@bo… This v2 series mainly: - Adds 2 patches to fix some compilation warning present on existing files (__iomem addresses declaration issues). - Fixes compilation warning on some patches. - Reword a commit log. Best regards, Hervé [1]: https://lore.kernel.org/all/20230417171601.74656-1-herve.codina@bootlin.com/ [2]: https://lore.kernel.org/all/20230323103154.264546-1-herve.codina@bootlin.co… Changes v1 -> v2 - Patches 1, 2 (New in v2) Fix __iomem addresses declaration - Patch 19 (17 in v1) Fix a compilation warning - Patch 26 (24 in v1) Fix a typo in Kconfig file Fix issues raised by sparse (make C=1) Herve Codina (28): soc: fsl: cpm1: tsa: Fix __iomem addresses declaration soc: fsl: cpm1: qmc: Fix __iomem addresses declaration soc: fsl: cpm1: qmc: Fix rx channel reset soc: fsl: cpm1: qmc: Extend the API to provide Rx status dt-bindings: net: Add support for QMC HDLC net: wan: Add support for QMC HDLC MAINTAINERS: Add the Freescale QMC HDLC driver entry soc: fsl: cpm1: qmc: Introduce available timeslots masks soc: fsl: cpm1: qmc: Rename qmc_setup_tsa* to qmc_init_tsa* soc: fsl: cpm1: qmc: Introduce qmc_chan_setup_tsa* soc: fsl: cpm1: qmc: Remove no more needed checks from qmc_check_chans() soc: fsl: cpm1: qmc: Check available timeslots in qmc_check_chans() soc: fsl: cpm1: qmc: Add support for disabling channel TSA entries soc: fsl: cpm1: qmc: Split Tx and Rx TSA entries setup soc: fsl: cpm1: qmc: Introduce is_tsa_64rxtx flag soc: fsl: cpm1: qmc: Handle timeslot entries at channel start() and stop() soc: fsl: cpm1: qmc: Remove timeslots handling from setup_chan() soc: fsl: cpm1: qmc: Introduce functions to change timeslots at runtime wan: qmc_hdlc: Add runtime timeslots changes support net: wan: Add framer framework support dt-bindings: net: Add the Lantiq PEF2256 E1/T1/J1 framer mfd: core: Ensure disabled devices are skiped without aborting net: wan: framer: Add support for the Lantiq PEF2256 framer pinctrl: Add support for the Lantic PEF2256 pinmux MAINTAINERS: Add the Lantiq PEF2256 driver entry ASoC: codecs: Add support for the framer codec dt-bindings: net: fsl,qmc-hdlc: Add framer support net: wan: fsl_qmc_hdlc: Add framer support .../devicetree/bindings/net/fsl,qmc-hdlc.yaml | 46 + .../bindings/net/lantiq,pef2256.yaml | 226 +++++ MAINTAINERS | 17 + drivers/mfd/mfd-core.c | 18 +- drivers/net/wan/Kconfig | 14 + drivers/net/wan/Makefile | 3 + drivers/net/wan/framer/Kconfig | 35 + drivers/net/wan/framer/Makefile | 7 + drivers/net/wan/framer/framer-core.c | 935 ++++++++++++++++++ drivers/net/wan/framer/pef2256/Makefile | 8 + drivers/net/wan/framer/pef2256/pef2256-regs.h | 250 +++++ drivers/net/wan/framer/pef2256/pef2256.c | 880 +++++++++++++++++ drivers/net/wan/fsl_qmc_hdlc.c | 820 +++++++++++++++ drivers/pinctrl/Kconfig | 14 + drivers/pinctrl/Makefile | 1 + drivers/pinctrl/pinctrl-pef2256-regs.h | 65 ++ drivers/pinctrl/pinctrl-pef2256.c | 310 ++++++ drivers/soc/fsl/qe/qmc.c | 501 ++++++++-- drivers/soc/fsl/qe/tsa.c | 22 +- include/linux/framer/framer-provider.h | 194 ++++ include/linux/framer/framer.h | 215 ++++ include/linux/framer/pef2256.h | 31 + include/soc/fsl/qe/qmc.h | 25 +- sound/soc/codecs/Kconfig | 15 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/framer-codec.c | 423 ++++++++ sound/soc/fsl/fsl_qmc_audio.c | 2 +- 27 files changed, 4957 insertions(+), 122 deletions(-) create mode 100644 Documentation/devicetree/bindings/net/fsl,qmc-hdlc.yaml create mode 100644 Documentation/devicetree/bindings/net/lantiq,pef2256.yaml create mode 100644 drivers/net/wan/framer/Kconfig create mode 100644 drivers/net/wan/framer/Makefile create mode 100644 drivers/net/wan/framer/framer-core.c create mode 100644 drivers/net/wan/framer/pef2256/Makefile create mode 100644 drivers/net/wan/framer/pef2256/pef2256-regs.h create mode 100644 drivers/net/wan/framer/pef2256/pef2256.c create mode 100644 drivers/net/wan/fsl_qmc_hdlc.c create mode 100644 drivers/pinctrl/pinctrl-pef2256-regs.h create mode 100644 drivers/pinctrl/pinctrl-pef2256.c create mode 100644 include/linux/framer/framer-provider.h create mode 100644 include/linux/framer/framer.h create mode 100644 include/linux/framer/pef2256.h create mode 100644 sound/soc/codecs/framer-codec.c -- 2.41.0

8 90

[PATCH 00/27] ASoC: Map missing jack kcontrols
by Alper Nebi Yasak 08 Aug '23

08 Aug '23

This patchset adds missing jack kcontrols for each independently detectable audio peripheral and maps jack pins to those kcontrols accordingly, with the primary intent to enable/improve jack detection handling in PulseAudio and PipeWire through JackControl UCM values. Usually it's just splitting a joint anything-is-connected "Headset Jack" kcontrol (from [1]) into those like "Headphone Jack" and "Headset Mic" (similar to a previous series for Intel Chromebooks [2]). This split is important to avoid automatically switching to a nonexistent external microphone when a headphone-only device is connected. When the underlying hardware seems to support it, this also adds a "Line Out" kcontrol. This is important in case the hardware can actually support a line-level connection via a different configuration (bypassing output amplifiers?), or simply for userspace to display "Line Out" instead of "Headphones" to the user for connected line-out devices. Beyond the mappings, I had to add PIN_SWITCH card kcontrols and DAPM widgets to avoid "unknown pin" errors on my devices, so tried to do them for all. For Intel devices I saw a pattern of routing things to "Platform Clock" and added to that as well. Looking at patch 5/7 of a Mediatek-related series [3], I can only guess that routes could be further improved, but don't know exactly how for each device. And one more concern is I don't know if the names conflict with any controls from codecs, although I tried to keep to names of existing widgets. As far as I can tell, the root cause for most of why these are missing originates to things being developed for ChromeOS, whose userspace reads the jack input device and doesn't care for these kcontrols. There's non-ChromeOS cases as well, maybe things got copy-pasted around and people didn't need or couldn't figure out how to get more specific than a single jack kcontrol. The secondary intent in this patchset is to fix this *everywhere*, so future copy-pastes result in the right behaviour. For more context also see: [1] ASoC: soc-card: Create jack kcontrol without pins https://lore.kernel.org/alsa-devel/20220408041114.6024-1-akihiko.odaki@gmai… [2] ASoC: Intel: Chromebooks: remap jack pins https://lore.kernel.org/alsa-devel/20220616214055.134943-1-pierre-louis.bos… [3] ASoC: mediatek: Allow separate handling of headphone and headset mic jack https://lore.kernel.org/alsa-devel/20220922235951.252532-1-nfraprado@collab… [4] ASoC: rk3399_gru_sound: Add DAPM pins, kcontrols for jack detection https://lore.kernel.org/alsa-devel/20200721182709.6895-1-alpernebiyasak@gma… This applies onto next-20230802. Unfortunately most of it is untested except for a few Chromebooks I have (Kevin, Lick, Hana, Cozmo), because I'm intentionally generalizing to everything. Alper Nebi Yasak (27): ASoC: amd: acp-da7219-max98357a: Map missing jack kcontrols ASoC: amd: acp-rt5645: Map missing jack kcontrols ASoC: amd: acp: Map missing jack kcontrols ASoC: amd: acp3x-rt5682-max9836: Map missing jack kcontrols ASoC: Intel: avs: da7219: Map missing jack kcontrols ASoC: Intel: bxt_da7219_max98357a: Map missing Line Out jack kcontrol ASoC: Intel: bytcr_wm5102: Map missing Line Out jack kcontrol ASoC: Intel: kbl_da7219_max98357a: Map missing Line Out jack kcontrol ASoC: Intel: kbl_da7219_max98927: Map missing Line Out jack kcontrol ASoC: Intel: sof_da7219_max98373: Map missing Line Out jack kcontrol ASoC: imx-es8328: Map missing jack kcontrols ASoC: mediatek: mt8173-max98090: Configure jack as a Headset jack ASoC: mediatek: mt8173-rt5650-rt5514: Map missing jack kcontrols ASoC: mediatek: mt8173-rt5650-rt5676: Map missing jack kcontrols ASoC: mediatek: mt8173-rt5650: Map missing jack kcontrols ASoC: mediatek: mt8183-da7219-max98357: Map missing jack kcontrols ASoC: mediatek: mt8183-mt6358-ts3a227-max98357: Map missing jack kcontrols ASoC: mediatek: mt8186-mt6366-da7219-max98357: Map missing jack kcontrols ASoC: qcom: apq8016_sbc: Map missing jack kcontrols ASoC: qcom: sc7180: Map missing jack kcontrols ASoC: qcom: sc7280: Map missing jack kcontrols ASoC: qcom: sdm845: Map missing jack kcontrols ASoC: rk3399-gru-sound: Map missing Line Out jack kcontrol ASoC: rockchip: rockchip_rt5645: Map missing jack kcontrols ASoC: samsung: littlemill: Map missing jack kcontrols ASoC: samsung: lowland: Split Line Out jack kcontrol from Headphone ASoC: samsung: midas_wm1811: Map missing jack kcontrols sound/soc/amd/acp-da7219-max98357a.c | 41 +++- sound/soc/amd/acp-rt5645.c | 22 +- sound/soc/amd/acp/acp-mach-common.c | 226 ++++++++++++++---- sound/soc/amd/acp3x-rt5682-max9836.c | 23 +- sound/soc/fsl/imx-es8328.c | 25 +- sound/soc/intel/avs/boards/da7219.c | 34 ++- sound/soc/intel/boards/bxt_da7219_max98357a.c | 7 + sound/soc/intel/boards/bytcr_wm5102.c | 7 + sound/soc/intel/boards/kbl_da7219_max98357a.c | 7 + sound/soc/intel/boards/kbl_da7219_max98927.c | 7 + sound/soc/intel/boards/sof_da7219_max98373.c | 10 + sound/soc/mediatek/mt8173/mt8173-max98090.c | 2 +- .../mediatek/mt8173/mt8173-rt5650-rt5514.c | 23 +- .../mediatek/mt8173/mt8173-rt5650-rt5676.c | 23 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 23 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 43 +++- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 53 +++- .../mt8186/mt8186-mt6366-da7219-max98357.c | 6 + sound/soc/qcom/apq8016_sbc.c | 37 ++- sound/soc/qcom/sc7180.c | 40 +++- sound/soc/qcom/sc7280.c | 34 ++- sound/soc/qcom/sdm845.c | 32 ++- sound/soc/rockchip/rk3399_gru_sound.c | 7 +- sound/soc/rockchip/rockchip_rt5645.c | 22 +- sound/soc/samsung/littlemill.c | 27 ++- sound/soc/samsung/lowland.c | 8 +- sound/soc/samsung/midas_wm1811.c | 25 +- 27 files changed, 668 insertions(+), 146 deletions(-) base-commit: 626c67169f9972fffcdf3bc3864de421f162ebf5 -- 2.40.1

4 34

Re: linux-next: Tree for Aug 7 (sound/soc/fsl/)
by Randy Dunlap 08 Aug '23

08 Aug '23

On 8/6/23 22:47, Stephen Rothwell wrote: > Hi all, > > Changes since 20230804: > on PPC32: WARNING: unmet direct dependencies detected for SND_SOC_MPC5200_AC97 Depends on [n]: SOUND [=y] && SND [=y] && SND_SOC [=y] && SND_POWERPC_SOC [=y] && PPC_MPC52xx [=y] && PPC_BESTCOMM [=n] Selected by [y]: - SND_MPC52xx_SOC_PCM030 [=y] && SOUND [=y] && SND [=y] && SND_SOC [=y] && SND_POWERPC_SOC [=y] && PPC_MPC5200_SIMPLE [=y] - SND_MPC52xx_SOC_EFIKA [=y] && SOUND [=y] && SND [=y] && SND_SOC [=y] && SND_POWERPC_SOC [=y] && PPC_EFIKA [=y] powerpc-linux-ld: sound/soc/fsl/mpc5200_psc_ac97.o: in function `psc_ac97_of_remove': mpc5200_psc_ac97.c:(.text.psc_ac97_of_remove+0x1c): undefined reference to `mpc5200_audio_dma_destroy' powerpc-linux-ld: sound/soc/fsl/mpc5200_psc_ac97.o: in function `psc_ac97_of_probe': mpc5200_psc_ac97.c:(.text.psc_ac97_of_probe+0x20): undefined reference to `mpc5200_audio_dma_create' Full randconfig file is attached. -- ~Randy

1 0

[PATCH] ASoC: lower "no backend DAIs enabled for ... Port" log severity
by Hans de Goede 08 Aug '23

08 Aug '23

If SNDRV_PCM_IOCTL_PREPARE is called when the mixer settings linking frontend and backend have not been setup yet this results in e.g. the following errors getting logged: [ 43.244549] Baytrail Audio Port: ASoC: no backend DAIs enabled for Baytrail Audio Port [ 43.244744] Baytrail Audio Port: ASoC: error at dpcm_fe_dai_prepare on Baytrail Audio Port: -22 pipewire triggers this leading to 96 repeats of this in the log after the user has logged into a GNOME session. IMHO userspace should not be able to get the kernel to spam dmesg like this. Lower the severity of the "no backend DAIs enabled" log message for dai-s with the dynamic flag set to avoid this. And also changes _soc_pcm_ret() to not log for -EINVAL errors, to fix the other error. Userspace passing wrong parameters should not lead to dmesg messages. Link: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/3407 Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/soc-pcm.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 8896227e4fb7..fdc1a1c654d6 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -38,6 +38,7 @@ static inline int _soc_pcm_ret(struct snd_soc_pcm_runtime *rtd, switch (ret) { case -EPROBE_DEFER: case -ENOTSUPP: + case -EINVAL: break; default: dev_err(rtd->dev, @@ -2466,7 +2467,11 @@ static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream) /* there is no point preparing this FE if there are no BEs */ if (list_empty(&fe->dpcm[stream].be_clients)) { - dev_err(fe->dev, "ASoC: no backend DAIs enabled for %s\n", + if (fe->dai_link->dynamic) + dev_dbg(fe->dev, "ASoC: no backend DAIs enabled for %s\n", + fe->dai_link->name); + else + dev_err(fe->dev, "ASoC: no backend DAIs enabled for %s\n", fe->dai_link->name); ret = -EINVAL; goto out; -- 2.41.0

4 9

[PATCH V4 0/3] ASoC: codecs: Add awinic AW88261 audio amplifier driver
by wangweidong.a＠awinic.com 07 Aug '23

07 Aug '23

From: Weidong Wang <wangweidong.a(a)awinic.com> The awinic AW88261 is an I2S/TDM input, high efficiency digital Smart K audio amplifier v3 -> v4: Modify the dev_err_probe usage Changed the use of sizeof in kzalloc Changed the function name Merge aw88261_device.c into aw88261.c Weidong Wang (3): ASoC: dt-bindings: Add schema for "awinic,aw88261" ASoC: codecs: Add code for bin parsing compatible with aw88261 ASoC: codecs: Add aw88261 amplifier driver .../bindings/sound/awinic,aw88395.yaml | 4 +- sound/soc/codecs/Kconfig | 15 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/aw88261.c | 1298 +++++++++++++++++ sound/soc/codecs/aw88261.h | 459 ++++++ sound/soc/codecs/aw88395/aw88395_lib.c | 193 ++- sound/soc/codecs/aw88395/aw88395_reg.h | 1 + 7 files changed, 1954 insertions(+), 18 deletions(-) create mode 100644 sound/soc/codecs/aw88261.c create mode 100644 sound/soc/codecs/aw88261.h base-commit: c1a515d3c0270628df8ae5f5118ba859b85464a2 -- 2.41.0

2 4

[PATCH] ASoC: jz4740: Update to use maple tree register cache
by Mark Brown 07 Aug '23

07 Aug '23

The maple tree register cache is based on a much more modern data structure than the rbtree cache and makes optimisation choices which are probably more appropriate for modern systems than those made by the rbtree cache. In v6.5 it has also acquired the ability to generate multi-register writes in sync operations, bringing performance up to parity with the rbtree cache there. Update the jz4740 driver to use the more modern data structure. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/jz4740.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c index 7c25acf6ff0d..d1cea93bdb59 100644 --- a/sound/soc/codecs/jz4740.c +++ b/sound/soc/codecs/jz4740.c @@ -301,7 +301,7 @@ static const struct regmap_config jz4740_codec_regmap_config = { .reg_defaults = jz4740_codec_reg_defaults, .num_reg_defaults = ARRAY_SIZE(jz4740_codec_reg_defaults), - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_MAPLE, }; static int jz4740_codec_probe(struct platform_device *pdev) --- base-commit: 06c2afb862f9da8dc5efa4b6076a0e48c3fbaaa5 change-id: 20230630-asoc-jz-maple-aacbd3a12bb6 Best regards, -- Mark Brown <broonie(a)kernel.org>

1 1

[PATCH 00/16] ASoC: SoundWire codecs: improve pm_runtime handling
by Pierre-Louis Bossart 07 Aug '23

07 Aug '23

This patchset improves the pm_runtime behavior in rare corner cases identified by the Intel CI in the last 6 months. a) in stress-tests, it's not uncommon to see the following type of warnings when the codec reports as ATTACHED "rt711 sdw:0:025d:0711:00: runtime PM trying to activate child device sdw:0:025d:0711:00 but parent (sdw-master-0) is not active" This warning was not correlated with any functional issue, but it exposed a design issue on when to enable pm_runtime. The recommended practice in the pm_runtime documentation is to keep the devices in 'suspended' mode and mark them as 'active' when they are really functional. b) enabling pm_runtime when the codec reports as ATTACHED also creates a problematic case when the ASoC pm_runtime_get_sync() will silently fail due to the -EACCESS error handling. This can happen when playback starts before the codec is enumerated. This patchset modifies the initial stages so that codecs are pm_runtime enabled in the .probe() callback, but become pm_runtime 'active' only when they report present. This is better aligned with the design of the pm_runtime helpers and improved CI results significantly. This patchset modifies all existing SoundWire codecs (except Qualcomm ones), but the pattern of changes is exactly the same in all patches. Pierre-Louis Bossart (16): ASoC: SoundWire codecs: return error status in probe ASoC: SoundWire codecs: make regmap cache-only in probe ASoC: rt5682-sdw: make regmap cache-only in probe ASoC: rt711: enable pm_runtime in probe, keep status as 'suspended' ASoC: rt711-sdca: enable pm_runtime in probe, keep status as 'suspended' ASoC: rt700: enable pm_runtime in probe, keep status as 'suspended' ASoC: rt1712-sdca: enable pm_runtime in probe, keep status as 'suspended' ASoC: rt712-sdca-dmic: enable pm_runtime in probe, keep status as 'suspended' ASoC: rt715: enable pm_runtime in probe, keep status as 'suspended' ASoC: rt715-sdca: enable pm_runtime in probe, keep status as 'suspended' ASoC: rt1308-sdw: enable pm_runtime in probe, keep status as 'suspended' ASoC: rt1316-sdw: enable pm_runtime in probe, keep status as 'suspended' ASoC: rt1318-sdw: enable pm_runtime in probe, keep status as 'suspended' ASoC: rt5682-sdw: enable pm_runtime in probe, keep status as 'suspended' ASoC: max98363: enable pm_runtime in probe, keep status as 'suspended' ASoC: max98373-sdw: enable pm_runtime in probe, keep status as 'suspended' sound/soc/codecs/max98363.c | 41 +++++++++++++--------- sound/soc/codecs/max98373-sdw.c | 47 ++++++++++++++----------- sound/soc/codecs/rt1308-sdw.c | 54 +++++++++++++++------------- sound/soc/codecs/rt1316-sdw.c | 44 ++++++++++++++--------- sound/soc/codecs/rt1318-sdw.c | 43 ++++++++++++++--------- sound/soc/codecs/rt5682-sdw.c | 56 ++++++++++++++++++------------ sound/soc/codecs/rt5682.c | 3 ++ sound/soc/codecs/rt700-sdw.c | 7 ++-- sound/soc/codecs/rt700.c | 46 +++++++++++++++--------- sound/soc/codecs/rt711-sdca-sdw.c | 3 +- sound/soc/codecs/rt711-sdca.c | 48 +++++++++++++++++-------- sound/soc/codecs/rt711-sdw.c | 7 ++-- sound/soc/codecs/rt711.c | 47 ++++++++++++++++--------- sound/soc/codecs/rt712-sdca-dmic.c | 47 +++++++++++++++---------- sound/soc/codecs/rt712-sdca-sdw.c | 3 +- sound/soc/codecs/rt712-sdca.c | 45 ++++++++++++++++-------- sound/soc/codecs/rt715-sdca-sdw.c | 5 +-- sound/soc/codecs/rt715-sdca.c | 41 ++++++++++++++++------ sound/soc/codecs/rt715-sdw.c | 9 ++--- sound/soc/codecs/rt715.c | 41 +++++++++++++++------- 20 files changed, 392 insertions(+), 245 deletions(-) -- 2.39.2

2 17

[PATCH v2 1/3] ASoC: dt-bindings: Convert maxim,max98925 to DT schema
by Krzysztof Kozlowski 07 Aug '23

07 Aug '23

Convert the Maxim Integrated MAX98925/MAX98926/MAX98927 speaker amplifier bindings to DT schema format. Changes during conversion: 1. Add "sound-dai-cells", already used by DTS. 2. Use "maxim,interleave-mode" instead previous "interleave-mode" and undocumented but used interleave_mode. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Changes in v2: 1. Use maxim,interleave-mode as bool. Ryan, As original author, I added you as bindings maintainer. Is that okay or maybe someone else from Maxim should take care about the bindings? --- .../devicetree/bindings/sound/max9892x.txt | 44 --------- .../bindings/sound/maxim,max98925.yaml | 98 +++++++++++++++++++ 2 files changed, 98 insertions(+), 44 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/max9892x.txt create mode 100644 Documentation/devicetree/bindings/sound/maxim,max98925.yaml diff --git a/Documentation/devicetree/bindings/sound/max9892x.txt b/Documentation/devicetree/bindings/sound/max9892x.txt deleted file mode 100644 index 98cb9ba5b328..000000000000 --- a/Documentation/devicetree/bindings/sound/max9892x.txt +++ /dev/null @@ -1,44 +0,0 @@ -Maxim Integrated MAX98925/MAX98926/MAX98927 Speaker Amplifier - -This device supports I2C. - -Required properties: - - - compatible : should be one of the following - - "maxim,max98925" - - "maxim,max98926" - - "maxim,max98927" - - - vmon-slot-no : slot number used to send voltage information - or in inteleave mode this will be used as - interleave slot. - MAX98925/MAX98926 slot range : 0 ~ 30, Default : 0 - MAX98927 slot range : 0 ~ 15, Default : 0 - - - imon-slot-no : slot number used to send current information - MAX98925/MAX98926 slot range : 0 ~ 30, Default : 0 - MAX98927 slot range : 0 ~ 15, Default : 0 - - - interleave-mode : When using two MAX9892X in a system it is - possible to create ADC data that that will - overflow the frame size. Digital Audio Interleave - mode provides a means to output VMON and IMON data - from two devices on a single DOUT line when running - smaller frames sizes such as 32 BCLKS per LRCLK or - 48 BCLKS per LRCLK. - Range : 0 (off), 1 (on), Default : 0 - - - reg : the I2C address of the device for I2C - -Optional properties: - - reset-gpios : GPIO to reset the device - -Example: - -codec: max98927@3a { - compatible = "maxim,max98927"; - vmon-slot-no = <0>; - imon-slot-no = <1>; - interleave-mode = <0>; - reg = <0x3a>; -}; diff --git a/Documentation/devicetree/bindings/sound/maxim,max98925.yaml b/Documentation/devicetree/bindings/sound/maxim,max98925.yaml new file mode 100644 index 000000000000..32fd86204a7a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98925.yaml @@ -0,0 +1,98 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max98925.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim Integrated MAX98925/MAX98926/MAX98927 speaker amplifier + +maintainers: + - Ryan Lee <ryans.lee(a)maximintegrated.com> + +properties: + compatible: + enum: + - maxim,max98925 + - maxim,max98926 + - maxim,max98927 + + reg: + maxItems: 1 + + reset-gpios: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + vmon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 30 + default: 0 + description: + Slot number used to send voltage information or in inteleave mode this + will be used as interleave slot. + + imon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 30 + default: 0 + description: + Slot number used to send current information. + + maxim,interleave-mode: + type: boolean + description: + When using two MAX9892X in a system it is possible to create ADC data + that will overflow the frame size. When enabled, the Digital Audio + Interleave mode provides a means to output VMON and IMON data from two + devices on a single DOUT line when running smaller frames sizes such as + 32 BCLKS per LRCLK or 48 BCLKS per LRCLK. + +required: + - compatible + - reg + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + contains: + enum: + - maxim,max98927 + then: + properties: + vmon-slot-no: + minimum: 0 + maximum: 15 + + imon-slot-no: + minimum: 0 + maximum: 15 + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + #include <dt-bindings/gpio/gpio.h> + audio-codec@3a { + compatible = "maxim,max98927"; + reg = <0x3a>; + #sound-dai-cells = <0>; + + pinctrl-0 = <&speaker_default>; + pinctrl-names = "default"; + + reset-gpios = <&tlmm 69 GPIO_ACTIVE_LOW>; + + vmon-slot-no = <1>; + imon-slot-no = <0>; + }; + }; -- 2.34.1

2 3

[PATCH v2] ASoC: rt5665: add missed regulator_bulk_disable
by Zhang Shurong 07 Aug '23

07 Aug '23

The driver forgets to call regulator_bulk_disable() Add the missed call to fix it. Fixes: 33ada14a26c8 ("ASoC: add rt5665 codec driver") Signed-off-by: Zhang Shurong <zhang_shurong(a)foxmail.com> --- Changes in v2: - move regulator_bulk_disable() in rt5665_remove() sound/soc/codecs/rt5665.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 83c367af91da..525713c33d71 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -4472,6 +4472,8 @@ static void rt5665_remove(struct snd_soc_component *component) struct rt5665_priv *rt5665 = snd_soc_component_get_drvdata(component); regmap_write(rt5665->regmap, RT5665_RESET, 0); + + regulator_bulk_disable(ARRAY_SIZE(rt5665->supplies), rt5665->supplies); } #ifdef CONFIG_PM -- 2.30.2

2 1

[PATCH 00/24] ALSA: Generic PCM copy ops using sockptr_t
by Takashi Iwai 07 Aug '23

07 Aug '23

Hi, this is a patch set to clean up the PCM copy ops using sockptr_t as a "universal" pointer, inspired by the recent patch from Andy Shevchenko: https://lore.kernel.org/r/20230721100146.67293-1-andriy.shevchenko@linux.in… Even though it sounds a bit weird, sockptr_t is a generic type that is used already in wide ranges, and it can fit our purpose, too. With sockptr_t, the former split of copy_user and copy_kernel PCM ops can be unified again gracefully. The patch set introduces the new PCM ops, converting users, and drops the old PCM ops. Most of conversions are straightforward, simply replacing copy_*_user() with copy_*_sockptr() variants. Note that the conversion in ASoC will fix a potential problem of ASoC PCM that has been for long time. Since ASoC component takes care of only copy_user, the conversion form/to kernel space might have been missing. With this patch set, both cases are handled with sockptr_t by a single callback. The patches are lightly tested (with a faked PCM copy implementation on HD-audio), while most of patches are only compile-tested. Takashi === Cc: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Cc: Andrey Utkin <andrey_utkin(a)fastmail.com> Cc: Anton Sviridenko <anton(a)corp.bluecherry.net> Cc: Arnaud Pouliquen <arnaud.pouliquen(a)foss.st.com> Cc: Banajit Goswami <bgoswami(a)quicinc.com> Cc: Bluecherry Maintainers <maintainers(a)bluecherrydvr.com> Cc: Claudiu Beznea <claudiu.beznea(a)microchip.com> Cc: Ismael Luceno <ismael(a)iodev.co.uk> Cc: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Mark Brown <broonie(a)kernel.org> Cc: Mauro Carvalho Chehab <mchehab(a)kernel.org> Cc: Oleksandr Andrushchenko <oleksandr_andrushchenko(a)epam.com> Cc: Olivier Moysan <olivier.moysan(a)foss.st.com> Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Cc: linux-media(a)vger.kernel.org Cc: xen-devel(a)lists.xenproject.org === Takashi Iwai (24): ALSA: pcm: Add copy ops with universal sockptr_t ALSA: core: Add memory copy helpers between sockptr and iomem ALSA: dummy: Convert to generic PCM copy ops ALSA: gus: Convert to generic PCM copy ops ALSA: emu8000: Convert to generic PCM copy ops ALSA: es1938: Convert to generic PCM copy ops ALSA: korg1212: Convert to generic PCM copy ops ALSA: nm256: Convert to generic PCM copy ops ALSA: rme32: Convert to generic PCM copy ops ALSA: rme96: Convert to generic PCM copy ops ALSA: hdsp: Convert to generic PCM copy ops ALSA: rme9652: Convert to generic PCM copy ops ALSA: sh: Convert to generic PCM copy ops ALSA: xen: Convert to generic PCM copy ops ALSA: pcmtest: Update comment about PCM copy ops media: solo6x10: Convert to generic PCM copy ops ASoC: component: Add generic PCM copy ops ASoC: mediatek: Convert to generic PCM copy ops ASoC: qcom: Convert to generic PCM copy ops ASoC: dmaengine: Convert to generic PCM copy ops ASoC: dmaengine: Use sockptr_t for process callback, too ALSA: doc: Update description for the new PCM copy ops ASoC: pcm: Drop obsoleted PCM copy_user ops ALSA: pcm: Drop obsoleted PCM copy_user and copy_kernel ops .../kernel-api/writing-an-alsa-driver.rst | 59 +++++--------- drivers/media/pci/solo6x10/solo6x10-g723.c | 41 ++-------- include/sound/dmaengine_pcm.h | 2 +- include/sound/pcm.h | 12 +-- include/sound/soc-component.h | 14 ++-- sound/core/memory.c | 39 +++++++++ sound/core/pcm_lib.c | 81 +++++++++---------- sound/core/pcm_native.c | 2 +- sound/drivers/dummy.c | 12 +-- sound/drivers/pcmtest.c | 2 +- sound/isa/gus/gus_pcm.c | 23 +----- sound/isa/sb/emu8000_pcm.c | 79 +++++------------- sound/pci/es1938.c | 31 ++----- sound/pci/korg1212/korg1212.c | 46 +++-------- sound/pci/nm256/nm256.c | 42 ++-------- sound/pci/rme32.c | 50 +++--------- sound/pci/rme96.c | 48 +++-------- sound/pci/rme9652/hdsp.c | 42 ++-------- sound/pci/rme9652/rme9652.c | 46 ++--------- sound/sh/sh_dac_audio.c | 25 +----- sound/soc/atmel/mchp-pdmc.c | 2 +- sound/soc/mediatek/common/mtk-btcvsd.c | 22 ++--- sound/soc/qcom/lpass-platform.c | 12 +-- sound/soc/soc-component.c | 10 +-- sound/soc/soc-generic-dmaengine-pcm.c | 18 ++--- sound/soc/soc-pcm.c | 4 +- sound/soc/stm/stm32_sai_sub.c | 2 +- sound/xen/xen_snd_front_alsa.c | 55 +++---------- 28 files changed, 251 insertions(+), 570 deletions(-) -- 2.35.3

3 33

[PATCH v2] ASoC: lower "no backend DAIs enabled for ... Port" log severity
by Hans de Goede 07 Aug '23

07 Aug '23

If SNDRV_PCM_IOCTL_PREPARE is called when the mixer settings linking frontend and backend have not been setup yet this results in e.g. the following errors getting logged: [ 43.244549] Baytrail Audio Port: ASoC: no backend DAIs enabled for Baytrail Audio Port [ 43.244744] Baytrail Audio Port: ASoC: error at dpcm_fe_dai_prepare on Baytrail Audio Port: -22 pipewire triggers this leading to 96 lines getting logged after the user has logged into a GNOME session. Change the actual "no backend DAIs enabled for ... Port" error to dev_err_once() to avoid it getting repeated 48 times. While at it also improve the error by hinting the user how to fix this. To not make developing new UCM profiles harder, also log the error at dev_dbg() level all the time (vs once). So that e.g. dyndbg can be used to (re)enable the messages. Also changes _soc_pcm_ret() to not log for -EINVAL errors, to fix the other error getting logged 48 times. Userspace passing wrong parameters should not lead to dmesg messages. Link: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/3407 Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- Changes in v2: - Switch the logging to a combination of dev_err_once() with improved log-message + dev_dbg() with shorter log message --- sound/soc/soc-pcm.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 8896227e4fb7..3aa6b988cb4b 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -38,6 +38,7 @@ static inline int _soc_pcm_ret(struct snd_soc_pcm_runtime *rtd, switch (ret) { case -EPROBE_DEFER: case -ENOTSUPP: + case -EINVAL: break; default: dev_err(rtd->dev, @@ -2466,8 +2467,11 @@ static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream) /* there is no point preparing this FE if there are no BEs */ if (list_empty(&fe->dpcm[stream].be_clients)) { - dev_err(fe->dev, "ASoC: no backend DAIs enabled for %s\n", - fe->dai_link->name); + /* dev_err_once() for visibility, dev_dbg() for debugging UCM profiles */ + dev_err_once(fe->dev, "ASoC: no backend DAIs enabled for %s, possibly missing ALSA mixer-based routing or UCM profile\n", + fe->dai_link->name); + dev_dbg(fe->dev, "ASoC: no backend DAIs enabled for %s\n", + fe->dai_link->name); ret = -EINVAL; goto out; } -- 2.41.0

3 2

[PATCH] sound: core: fix device ownership model in card and pcm
by cujomalainey＠chromium.org 07 Aug '23

07 Aug '23

From: Curtis Malainey <cujomalainey(a)chromium.org> The current implementation of how devices are released is valid for production use cases (root control of memory is handled by card_dev, all other devices are no-ops). This model does not work though in a kernel hacking environment where KASAN and delayed release on kobj is enabled. If the card_dev device is released before any of the child objects a use-after-free bug is caught by KASAN as the delayed release still has a reference to the devices that were freed by the card_dev release. Also both snd_card and snd_pcm both own two devices internally, so even if they released independently, the shared struct would result in another use after free. Solution is to move the child devices into their own memory so they can be handled independently and released on their own schedule. Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> Cc: Doug Anderson <dianders(a)chromium.org> --- include/sound/core.h | 2 +- include/sound/pcm.h | 2 +- sound/core/control.c | 21 +++++++++++++++------ sound/core/control_led.c | 4 ++-- sound/core/pcm.c | 30 ++++++++++++++++++++---------- sound/usb/media.c | 4 ++-- 6 files changed, 41 insertions(+), 22 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index f6e0dd648b80c..a08ab8c8cfb6d 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -96,7 +96,7 @@ struct snd_card { private data */ struct list_head devices; /* devices */ - struct device ctl_dev; /* control device */ + struct device *ctl_dev; /* control device */ unsigned int last_numid; /* last used numeric ID */ struct rw_semaphore controls_rwsem; /* controls lock (list and values) */ rwlock_t ctl_files_rwlock; /* ctl_files list lock */ diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 19f564606ac42..0243a13e9ac47 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -510,7 +510,7 @@ struct snd_pcm_str { #endif #endif struct snd_kcontrol *chmap_kctl; /* channel-mapping controls */ - struct device dev; + struct device *dev; }; struct snd_pcm { diff --git a/sound/core/control.c b/sound/core/control.c index 8386b53acdcd4..8bbaf3dffce62 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -2315,7 +2315,7 @@ static int snd_ctl_dev_register(struct snd_device *device) int err; err = snd_register_device(SNDRV_DEVICE_TYPE_CONTROL, card, -1, - &snd_ctl_f_ops, card, &card->ctl_dev); + &snd_ctl_f_ops, card, card->ctl_dev); if (err < 0) return err; down_read(&card->controls_rwsem); @@ -2351,7 +2351,7 @@ static int snd_ctl_dev_disconnect(struct snd_device *device) up_read(&snd_ctl_layer_rwsem); up_read(&card->controls_rwsem); - return snd_unregister_device(&card->ctl_dev); + return snd_unregister_device(card->ctl_dev); } /* @@ -2373,10 +2373,15 @@ static int snd_ctl_dev_free(struct snd_device *device) xa_destroy(&card->ctl_hash); #endif up_write(&card->controls_rwsem); - put_device(&card->ctl_dev); + put_device(card->ctl_dev); return 0; } +static void snd_ctl_dev_release(struct device *dev) +{ + kfree(dev); +} + /* * create control core: * called from init.c @@ -2394,13 +2399,17 @@ int snd_ctl_create(struct snd_card *card) return -ENXIO; if (snd_BUG_ON(card->number < 0 || card->number >= SNDRV_CARDS)) return -ENXIO; + card->ctl_dev = kzalloc(sizeof(*card->ctl_dev), GFP_KERNEL); + if (!card->ctl_dev) + return -ENOMEM; - snd_device_initialize(&card->ctl_dev, card); - dev_set_name(&card->ctl_dev, "controlC%d", card->number); + snd_device_initialize(card->ctl_dev, card); + card->ctl_dev->release = snd_ctl_dev_release; + dev_set_name(card->ctl_dev, "controlC%d", card->number); err = snd_device_new(card, SNDRV_DEV_CONTROL, card, &ops); if (err < 0) - put_device(&card->ctl_dev); + put_device(card->ctl_dev); return err; } diff --git a/sound/core/control_led.c b/sound/core/control_led.c index ee77547bf8dcb..760e46cf25cc8 100644 --- a/sound/core/control_led.c +++ b/sound/core/control_led.c @@ -688,7 +688,7 @@ static void snd_ctl_led_sysfs_add(struct snd_card *card) goto cerr; led->cards[card->number] = led_card; snprintf(link_name, sizeof(link_name), "led-%s", led->name); - WARN(sysfs_create_link(&card->ctl_dev.kobj, &led_card->dev.kobj, link_name), + WARN(sysfs_create_link(&card->ctl_dev->kobj, &led_card->dev.kobj, link_name), "can't create symlink to controlC%i device\n", card->number); WARN(sysfs_create_link(&led_card->dev.kobj, &card->card_dev.kobj, "card"), "can't create symlink to card%i\n", card->number); @@ -714,7 +714,7 @@ static void snd_ctl_led_sysfs_remove(struct snd_card *card) if (!led_card) continue; snprintf(link_name, sizeof(link_name), "led-%s", led->name); - sysfs_remove_link(&card->ctl_dev.kobj, link_name); + sysfs_remove_link(&card->ctl_dev->kobj, link_name); sysfs_remove_link(&led_card->dev.kobj, "card"); device_unregister(&led_card->dev); led->cards[card->number] = NULL; diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 9d95e37311230..9026ccc56dbe7 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -604,7 +604,7 @@ static const struct attribute_group *pcm_dev_attr_groups[]; #ifdef CONFIG_PM_SLEEP static int do_pcm_suspend(struct device *dev) { - struct snd_pcm_str *pstr = container_of(dev, struct snd_pcm_str, dev); + struct snd_pcm_str *pstr = dev_get_drvdata(dev); if (!pstr->pcm->no_device_suspend) snd_pcm_suspend_all(pstr->pcm); @@ -622,6 +622,11 @@ static const struct device_type pcm_dev_type = { .pm = &pcm_dev_pm_ops, }; +static void snd_pcm_dev_release(struct device *dev) +{ + kfree(dev); +} + /** * snd_pcm_new_stream - create a new PCM stream * @pcm: the pcm instance @@ -641,6 +646,10 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) struct snd_pcm_str *pstr = &pcm->streams[stream]; struct snd_pcm_substream *substream, *prev; + pstr->dev = kzalloc(sizeof(*pstr->dev), GFP_KERNEL); + if (!pstr->dev) + return -ENOMEM; + dev_set_drvdata(pstr->dev, pstr); #if IS_ENABLED(CONFIG_SND_PCM_OSS) mutex_init(&pstr->oss.setup_mutex); #endif @@ -650,10 +659,11 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) if (!substream_count) return 0; - snd_device_initialize(&pstr->dev, pcm->card); - pstr->dev.groups = pcm_dev_attr_groups; - pstr->dev.type = &pcm_dev_type; - dev_set_name(&pstr->dev, "pcmC%iD%i%c", pcm->card->number, pcm->device, + snd_device_initialize(pstr->dev, pcm->card); + pstr->dev->release = snd_pcm_dev_release; + pstr->dev->groups = pcm_dev_attr_groups; + pstr->dev->type = &pcm_dev_type; + dev_set_name(pstr->dev, "pcmC%iD%i%c", pcm->card->number, pcm->device, stream == SNDRV_PCM_STREAM_PLAYBACK ? 'p' : 'c'); if (!pcm->internal) { @@ -699,7 +709,7 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) prev = substream; } return 0; -} +} EXPORT_SYMBOL(snd_pcm_new_stream); static int _snd_pcm_new(struct snd_card *card, const char *id, int device, @@ -847,7 +857,7 @@ static void snd_pcm_free_stream(struct snd_pcm_str * pstr) #endif free_chmap(pstr); if (pstr->substream_count) - put_device(&pstr->dev); + put_device(pstr->dev); } #if IS_ENABLED(CONFIG_SND_PCM_OSS) @@ -1017,7 +1027,7 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream) static ssize_t pcm_class_show(struct device *dev, struct device_attribute *attr, char *buf) { - struct snd_pcm_str *pstr = container_of(dev, struct snd_pcm_str, dev); + struct snd_pcm_str *pstr = dev_get_drvdata(dev); struct snd_pcm *pcm = pstr->pcm; const char *str; static const char *strs[SNDRV_PCM_CLASS_LAST + 1] = { @@ -1078,7 +1088,7 @@ static int snd_pcm_dev_register(struct snd_device *device) /* register pcm */ err = snd_register_device(devtype, pcm->card, pcm->device, &snd_pcm_f_ops[cidx], pcm, - &pcm->streams[cidx].dev); + pcm->streams[cidx].dev); if (err < 0) { list_del_init(&pcm->list); goto unlock; @@ -1125,7 +1135,7 @@ static int snd_pcm_dev_disconnect(struct snd_device *device) pcm_call_notify(pcm, n_disconnect); for (cidx = 0; cidx < 2; cidx++) { - snd_unregister_device(&pcm->streams[cidx].dev); + snd_unregister_device(pcm->streams[cidx].dev); free_chmap(&pcm->streams[cidx]); } mutex_unlock(&pcm->open_mutex); diff --git a/sound/usb/media.c b/sound/usb/media.c index 840f42cb9272c..d48db6f3ae659 100644 --- a/sound/usb/media.c +++ b/sound/usb/media.c @@ -35,7 +35,7 @@ int snd_media_stream_init(struct snd_usb_substream *subs, struct snd_pcm *pcm, { struct media_device *mdev; struct media_ctl *mctl; - struct device *pcm_dev = &pcm->streams[stream].dev; + struct device *pcm_dev = pcm->streams[stream].dev; u32 intf_type; int ret = 0; u16 mixer_pad; @@ -163,7 +163,7 @@ void snd_media_stop_pipeline(struct snd_usb_substream *subs) static int snd_media_mixer_init(struct snd_usb_audio *chip) { - struct device *ctl_dev = &chip->card->ctl_dev; + struct device *ctl_dev = chip->card->ctl_dev; struct media_intf_devnode *ctl_intf; struct usb_mixer_interface *mixer; struct media_device *mdev = chip->media_dev; -- 2.41.0.585.gd2178a4bd4-goog

6 16

[PATCH v1 0/5] Add I2S support for the StarFive JH7110 SoC
by Xingyu Wu 07 Aug '23

07 Aug '23

This patch series adds I2S support for the StarFive JH7110 RISC-V SoC based on Designware I2S controller. There has three I2S channels (RX/TX0/TX1) on the JH7110 SoC, one of which is for record(RX) and two for playback(TX). The first patch adds the ops to get data from platform bus in the I2S driver. The second patch adds support for the StarFive JH7110 SoC in the Designware I2S bindings. The third patch adds support for the StarFive JH7110 SoC in the Designware I2S driver. The fourth patch fixes the name of I2STX1 pinmux. The last patch adds device node of I2S RX/TX0/TX1 in JH7110 dts. This patch series is based on Linux-next which is merge clock, syscon and dma nodes for the StarFive JH7110 SoC. And these should be applied after the following patchset: https://lore.kernel.org/all/20230724055440.100947-1-xingyu.wu@starfivetech.… The series has been tested and works normally on the VisionFive 2 board by plugging an audio expansion board. Xingyu Wu (5): ASoC: dwc: Use ops to get platform data ASoC: dt-bindings: snps,designware-i2s: Add StarFive JH7110 SoC support ASoC: dwc: i2s: Add StarFive JH7110 SoC support riscv: dts: starfive: pinfunc: Fix the pins name of I2STX1 riscv: dts: starfive: Add the nodes and pins of I2Srx/I2Stx0/I2Stx1 .../bindings/sound/snps,designware-i2s.yaml | 101 +++++- arch/riscv/boot/dts/starfive/jh7110-pinfunc.h | 4 +- .../jh7110-starfive-visionfive-2.dtsi | 58 ++++ arch/riscv/boot/dts/starfive/jh7110.dtsi | 65 ++++ include/sound/designware_i2s.h | 3 + sound/soc/dwc/dwc-i2s.c | 318 ++++++++++++++++-- sound/soc/dwc/local.h | 1 + 7 files changed, 523 insertions(+), 27 deletions(-) base-commit: 0fee53365d051781156b75c6f6e6015757e71fd8 prerequisite-patch-id: 0159f09bb0a1ff711a00ae17ef5b12662c9c7d3d -- 2.25.1

3 13

[PATCH v2 0/3] ASoC: mt8188-mt6359: add SOF support
by Trevor Wu 07 Aug '23

07 Aug '23

This series introduces dynamic pinctrl and adds support for the SOF on the mt8188-mt6359 machine driver. Changes since v1: - minor changes to incorporate the suggestions from the reviewer. Trevor Wu (3): ASoC: mediatek: mt8188-mt6359: support dynamic pinctrl ASoC: mediatek: common: revise SOF common code ASoC: mediatek: mt8188-mt6359: add SOF support .../soc/mediatek/common/mtk-dsp-sof-common.c | 106 ++++++-- .../soc/mediatek/common/mtk-dsp-sof-common.h | 8 + sound/soc/mediatek/mt8188/mt8188-mt6359.c | 239 +++++++++++++++++- 3 files changed, 325 insertions(+), 28 deletions(-) -- 2.18.0

3 6

[PATCH v5] um/drivers: fix hostaudio build errors
by Randy Dunlap 06 Aug '23

06 Aug '23

Use "select" to ensure that the required kconfig symbols are set as expected. Drop HOSTAUDIO since it is now equivalent to UML_SOUND. Set CONFIG_SOUND=m in ARCH=um defconfig files to maintain the status quo of the default configs. Allow SOUND with UML regardless of HAS_IOMEM. Otherwise there is a kconfig warning for unmet dependencies. (This was not an issue when SOUND was defined in arch/um/drivers/Kconfig. I have done 50 randconfig builds and didn't find any issues.) This fixes build errors when CONFIG_SOUND is not set: ld: arch/um/drivers/hostaudio_kern.o: in function `hostaudio_cleanup_module': hostaudio_kern.c:(.exit.text+0xa): undefined reference to `unregister_sound_mixer' ld: hostaudio_kern.c:(.exit.text+0x15): undefined reference to `unregister_sound_dsp' ld: arch/um/drivers/hostaudio_kern.o: in function `hostaudio_init_module': hostaudio_kern.c:(.init.text+0x19): undefined reference to `register_sound_dsp' ld: hostaudio_kern.c:(.init.text+0x31): undefined reference to `register_sound_mixer' ld: hostaudio_kern.c:(.init.text+0x49): undefined reference to `unregister_sound_dsp' and this kconfig warning: WARNING: unmet direct dependencies detected for SOUND Fixes: 1da177e4c3f4 ("Linux-2.6.12-rc2") Fixes: d886e87cb82b ("sound: make OSS sound core optional") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Reported-by: kernel test robot <lkp(a)intel.com> Closes: lore.kernel.org/r/202307141416.vxuRVpFv-lkp@intel.com Cc: Richard Weinberger <richard(a)nod.at> Cc: Anton Ivanov <anton.ivanov(a)cambridgegreys.com> Cc: Johannes Berg <johannes(a)sipsolutions.net> Cc: linux-um(a)lists.infradead.org Cc: Tejun Heo <tj(a)kernel.org> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Masahiro Yamada <masahiroy(a)kernel.org> Cc: Nathan Chancellor <nathan(a)kernel.org> Cc: Nick Desaulniers <ndesaulniers(a)google.com> Cc: Nicolas Schier <nicolas(a)fjasle.eu> Cc: linux-kbuild(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org --- v2: don't delete the HOSTAUDIO Kconfig entry (Masahiro) v3: drop HOSTAUDIO and use CONFIG_UML_SOUND for it in Makefile (Takashi); add SOUND depends on "|| UML" to HAS_IOMEM v4: use depends on instead of select for SOUND (Masahiro); use Closes: instead of Link: v5: update ARCH=um defconfig files (Masahiro) arch/um/configs/i386_defconfig | 1 + arch/um/configs/x86_64_defconfig | 1 + arch/um/drivers/Kconfig | 16 +++------------- arch/um/drivers/Makefile | 2 +- sound/Kconfig | 2 +- 5 files changed, 7 insertions(+), 15 deletions(-) diff -- a/arch/um/drivers/Kconfig b/arch/um/drivers/Kconfig --- a/arch/um/drivers/Kconfig +++ b/arch/um/drivers/Kconfig @@ -111,24 +111,14 @@ config SSL_CHAN config UML_SOUND tristate "Sound support" + depends on SOUND + select SOUND_OSS_CORE help This option enables UML sound support. If enabled, it will pull in - soundcore and the UML hostaudio relay, which acts as a intermediary + the UML hostaudio relay, which acts as a intermediary between the host's dsp and mixer devices and the UML sound system. It is safe to say 'Y' here. -config SOUND - tristate - default UML_SOUND - -config SOUND_OSS_CORE - bool - default UML_SOUND - -config HOSTAUDIO - tristate - default UML_SOUND - endmenu menu "UML Network Devices" diff -- a/sound/Kconfig b/sound/Kconfig --- a/sound/Kconfig +++ b/sound/Kconfig @@ -1,7 +1,7 @@ # SPDX-License-Identifier: GPL-2.0-only menuconfig SOUND tristate "Sound card support" - depends on HAS_IOMEM + depends on HAS_IOMEM || UML help If you have a sound card in your computer, i.e. if it can say more than an occasional beep, say Y. diff -- a/arch/um/drivers/Makefile b/arch/um/drivers/Makefile --- a/arch/um/drivers/Makefile +++ b/arch/um/drivers/Makefile @@ -54,7 +54,7 @@ obj-$(CONFIG_UML_NET) += net.o obj-$(CONFIG_MCONSOLE) += mconsole.o obj-$(CONFIG_MMAPPER) += mmapper_kern.o obj-$(CONFIG_BLK_DEV_UBD) += ubd.o -obj-$(CONFIG_HOSTAUDIO) += hostaudio.o +obj-$(CONFIG_UML_SOUND) += hostaudio.o obj-$(CONFIG_NULL_CHAN) += null.o obj-$(CONFIG_PORT_CHAN) += port.o obj-$(CONFIG_PTY_CHAN) += pty.o diff -- a/arch/um/configs/i386_defconfig b/arch/um/configs/i386_defconfig --- a/arch/um/configs/i386_defconfig +++ b/arch/um/configs/i386_defconfig @@ -34,6 +34,7 @@ CONFIG_TTY_CHAN=y CONFIG_XTERM_CHAN=y CONFIG_CON_CHAN="pts" CONFIG_SSL_CHAN="pts" +CONFIG_SOUND=m CONFIG_UML_SOUND=m CONFIG_DEVTMPFS=y CONFIG_DEVTMPFS_MOUNT=y diff -- a/arch/um/configs/x86_64_defconfig b/arch/um/configs/x86_64_defconfig --- a/arch/um/configs/x86_64_defconfig +++ b/arch/um/configs/x86_64_defconfig @@ -32,6 +32,7 @@ CONFIG_TTY_CHAN=y CONFIG_XTERM_CHAN=y CONFIG_CON_CHAN="pts" CONFIG_SSL_CHAN="pts" +CONFIG_SOUND=m CONFIG_UML_SOUND=m CONFIG_DEVTMPFS=y CONFIG_DEVTMPFS_MOUNT=y

2 1

change order in which kernel initializes devices
by Fourhundred Thecat 06 Aug '23

06 Aug '23

Hello, I am using monolithic kernel without loadable module support. Everything is compiled in statically. Now I have snd-aloop (CONFIG_SND_ALOOP=y) and when i boot my kernel, it is recognized as primary device. My real hw card is as second device. Is there any way I can change the order, in which kernel initializes the sound devices, so that my real (hw) sound card is recognized first ? If I was using modules, I could simply load snd-aloop later. I am compiling my own kernel, so I would be happy to patch the code if necessary, if somebody could point me into the right section. Here is output of aplay -l, in case it helps: **** List of PLAYBACK Hardware Devices **** card 0: Loopback [Loopback], device 0: Loopback PCM [Loopback PCM] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 card 0: Loopback [Loopback], device 1: Loopback PCM [Loopback PCM] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 card 1: PCH [HDA Intel PCH], device 0: Generic Analog [Generic Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 PS: why are there 2 loopback devices, with 8 subdevices? When I am alredy at it, can I change tat in the kernel, so that I only have 1 loopback device for simplicity ? thank you,

1 0

[PATCH 1/2] ASoC: dt-bindings: wlf,wm8960: Describe the power supplies
by Fabio Estevam 06 Aug '23

06 Aug '23

From: Fabio Estevam <festevam(a)denx.de> WM8960 has the following power supplies: - AVDD - DBVDD - DCVDD - SPKVDD1 - SPKVDD1 Add them to bindings to improve the hardware description. Signed-off-by: Fabio Estevam <festevam(a)denx.de> --- .../devicetree/bindings/sound/wlf,wm8960.yaml | 20 +++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8960.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8960.yaml index ee8eba7f0104..82519f6720ef 100644 --- a/Documentation/devicetree/bindings/sound/wlf,wm8960.yaml +++ b/Documentation/devicetree/bindings/sound/wlf,wm8960.yaml @@ -26,6 +26,21 @@ properties: '#sound-dai-cells': const: 0 + AVDD-supply: + description: Analogue supply. + + DBVDD-supply: + description: Digital Buffer Supply. + + DCVDD-supply: + description: Digital Core Supply. + + SPKVDD1-supply: + description: Supply for speaker drivers 1. + + SPKVDD2-supply: + description: Supply for speaker drivers 2. + wlf,capless: type: boolean description: @@ -84,5 +99,10 @@ examples: wlf,hp-cfg = <3 2 3>; wlf,gpio-cfg = <1 3>; wlf,shared-lrclk; + DCVDD-supply = <&reg_audio>; + DBVDD-supply = <&reg_audio>; + AVDD-supply = <&reg_audio>; + SPKVDD1-supply = <&reg_audio>; + SPKVDD2-supply = <&reg_audio>; }; }; -- 2.34.1

5 6

[PATCH] ASoC: max98363: don't return on success reading revision ID
by Bard Liao 04 Aug '23

04 Aug '23

max98363_io_init needs to keep going when we read revision ID successfully. Fixes: 18c0af945fa3 ("ASoC: max98363: add soundwire amplifier driver") Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/codecs/max98363.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/max98363.c b/sound/soc/codecs/max98363.c index b5c69bba0e48..2dfaf4fcfbd3 100644 --- a/sound/soc/codecs/max98363.c +++ b/sound/soc/codecs/max98363.c @@ -185,10 +185,10 @@ static int max98363_io_init(struct sdw_slave *slave) pm_runtime_get_noresume(dev); ret = regmap_read(max98363->regmap, MAX98363_R21FF_REV_ID, &reg); - if (!ret) { + if (!ret) dev_info(dev, "Revision ID: %X\n", reg); - return ret; - } + else + goto out; if (max98363->first_hw_init) { regcache_cache_bypass(max98363->regmap, false); @@ -198,10 +198,11 @@ static int max98363_io_init(struct sdw_slave *slave) max98363->first_hw_init = true; max98363->hw_init = true; +out: pm_runtime_mark_last_busy(dev); pm_runtime_put_autosuspend(dev); - return 0; + return ret; } #define MAX98363_RATES SNDRV_PCM_RATE_8000_192000 -- 2.25.1

2 1

[PATCH v2 00/15] ASoC: add multi Component support
by Kuninori Morimoto 04 Aug '23

04 Aug '23

Hi Mark These are v2 of multi Component support. On below HW case, we would like to use it as "2 Cards", but unfortunately it is impossible in intuitive way, or possible but not intuitive way. In reality, it is handled as "1 big Card" today. +-- basic board --------+ |+--------+ | || CPU ch0| <--> CodecA | || ch1| <-+ | |+--------+ | | +-------------|---------+ +-- expansion board ----+ | | | | +-> CodecB| +-----------------------+ To handling it as intuitive "2 Cards", this patch-set adds multi Component support. To enable this patch-set, I included [01/15] patch into this patch-set which is posted but not yet accepted. v1 -> v2 - include [01/15] patch into this patch-set - add related member to To - add DT member to [14/15][15/15] Link: https://lore.kernel.org/all/87a6b6cofh.wl-kuninori.morimoto.gx@renesas.com/ Link: https://lore.kernel.org/r/87r0q5blta.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/20230623-asoc-fix-meson-probe-v1-1-82b2c2ec5ca4@k… Kuninori Morimoto (15): ASoC: soc-core: protect dlc->of_node under mutex ASoC: soc-core.c: initialize dlc on snd_soc_get_dai_id() ASoC: soc-core.c: cleanup soc_dai_link_sanity_check() ASoC: soc-dai.c: add DAI get/match functions ASoC: soc-core.c: enable multi Component ASoC: soc-core.c: add snd_soc_get_dai_via_args() ASoC: soc-core.c: add snd_soc_dlc_use_cpu_as_platform() ASoC: soc-core.c: add snd_soc_copy_dai_args() ASoC: simple-card-utils.c: enable multi Component support ASoC: simple-card.c: enable multi Component support ASoC: rsnd: use DAI driver ID instead of DAI ID ASoC: rsnd: cleanup rsnd_dai_of_node() ASoC: rsnd: enable multi Component support for Audio Graph Card/Card2 ASoC: dt-bindings: renesas,rsnd.yaml: add common port-def ASoC: dt-bindings: renesas,rsnd.yaml: enable multi ports for multi Component support .../bindings/sound/renesas,rsnd.yaml | 60 ++-- include/sound/simple_card_utils.h | 5 +- include/sound/soc-dai.h | 3 + include/sound/soc.h | 6 + sound/soc/generic/audio-graph-card.c | 2 +- sound/soc/generic/audio-graph-card2.c | 2 +- sound/soc/generic/simple-card-utils.c | 23 +- sound/soc/generic/simple-card.c | 20 +- sound/soc/sh/rcar/core.c | 146 +++++++--- sound/soc/sh/rcar/rsnd.h | 4 + sound/soc/soc-core.c | 273 +++++++++++++----- 11 files changed, 378 insertions(+), 166 deletions(-) -- 2.25.1

3 24

[PATCH] ASoC: SOF: mediatek: mt8186 modify dram type as non-cache
by Trevor Wu 04 Aug '23

04 Aug '23

To prevent incorrect access between the host and DSP sides, we need to modify DRAM as a non-cache memory type. Additionally, we can retrieve the size of shared DMA from the device tree. Signed-off-by: Trevor Wu <trevor.wu(a)mediatek.com> Reviewed-by: Yaochun Hung <yc.hung(a)mediatek.com> Reviewed-by: Kuan-Hsun Cheng <Allen-KH.Cheng(a)mediatek.com> --- sound/soc/sof/mediatek/mt8186/mt8186.c | 40 +++++++++++++++----------- 1 file changed, 23 insertions(+), 17 deletions(-) diff --git a/sound/soc/sof/mediatek/mt8186/mt8186.c b/sound/soc/sof/mediatek/mt8186/mt8186.c index 3e0ea0e109e2..f587edf9e0a7 100644 --- a/sound/soc/sof/mediatek/mt8186/mt8186.c +++ b/sound/soc/sof/mediatek/mt8186/mt8186.c @@ -111,6 +111,14 @@ static int platform_parse_resource(struct platform_device *pdev, void *data) dev_dbg(dev, "DMA %pR\n", &res); + adsp->pa_shared_dram = (phys_addr_t)res.start; + adsp->shared_size = resource_size(&res); + if (adsp->pa_shared_dram & DRAM_REMAP_MASK) { + dev_err(dev, "adsp shared dma memory(%#x) is not 4K-aligned\n", + (u32)adsp->pa_shared_dram); + return -EINVAL; + } + ret = of_reserved_mem_device_init(dev); if (ret) { dev_err(dev, "of_reserved_mem_device_init failed\n"); @@ -244,23 +252,18 @@ static int adsp_shared_base_ioremap(struct platform_device *pdev, void *data) { struct device *dev = &pdev->dev; struct mtk_adsp_chip_info *adsp = data; - u32 shared_size; /* remap shared-dram base to be non-cachable */ - shared_size = TOTAL_SIZE_SHARED_DRAM_FROM_TAIL; - adsp->pa_shared_dram = adsp->pa_dram + adsp->dramsize - shared_size; - if (adsp->va_dram) { - adsp->shared_dram = adsp->va_dram + DSP_DRAM_SIZE - shared_size; - } else { - adsp->shared_dram = devm_ioremap(dev, adsp->pa_shared_dram, - shared_size); - if (!adsp->shared_dram) { - dev_err(dev, "ioremap failed for shared DRAM\n"); - return -ENOMEM; - } + adsp->shared_dram = devm_ioremap(dev, adsp->pa_shared_dram, + adsp->shared_size); + if (!adsp->shared_dram) { + dev_err(dev, "failed to ioremap base %pa size %#x\n", + adsp->shared_dram, adsp->shared_size); + return -ENOMEM; } - dev_dbg(dev, "shared-dram vbase=%p, phy addr :%pa, size=%#x\n", - adsp->shared_dram, &adsp->pa_shared_dram, shared_size); + + dev_dbg(dev, "shared-dram vbase=%p, phy addr :%pa, size=%#x\n", + adsp->shared_dram, &adsp->pa_shared_dram, adsp->shared_size); return 0; } @@ -307,9 +310,12 @@ static int mt8186_dsp_probe(struct snd_sof_dev *sdev) return -ENOMEM; } - sdev->bar[SOF_FW_BLK_TYPE_SRAM] = devm_ioremap_wc(sdev->dev, - priv->adsp->pa_dram, - priv->adsp->dramsize); + priv->adsp->va_sram = sdev->bar[SOF_FW_BLK_TYPE_IRAM]; + + sdev->bar[SOF_FW_BLK_TYPE_SRAM] = devm_ioremap(sdev->dev, + priv->adsp->pa_dram, + priv->adsp->dramsize); + if (!sdev->bar[SOF_FW_BLK_TYPE_SRAM]) { dev_err(sdev->dev, "failed to ioremap base %pa size %#x\n", &priv->adsp->pa_dram, priv->adsp->dramsize); -- 2.18.0

4 3

[PATCH -next] ASoC: ti: davinci-evm: simplify the code with module_platform_driver
by Yang Yingliang 04 Aug '23

04 Aug '23

The init/exit() of driver only calls platform_driver_register/unregister, it can be simpilfied with module_platform_driver. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/ti/davinci-evm.c | 13 +------------ 1 file changed, 1 insertion(+), 12 deletions(-) diff --git a/sound/soc/ti/davinci-evm.c b/sound/soc/ti/davinci-evm.c index 983d69b951b0..544cb3da50eb 100644 --- a/sound/soc/ti/davinci-evm.c +++ b/sound/soc/ti/davinci-evm.c @@ -257,18 +257,7 @@ static struct platform_driver davinci_evm_driver = { }, }; -static int __init evm_init(void) -{ - return platform_driver_register(&davinci_evm_driver); -} - -static void __exit evm_exit(void) -{ - platform_driver_unregister(&davinci_evm_driver); -} - -module_init(evm_init); -module_exit(evm_exit); +module_platform_driver(davinci_evm_driver); MODULE_AUTHOR("Vladimir Barinov"); MODULE_DESCRIPTION("TI DAVINCI EVM ASoC driver"); -- 2.25.1

2 1

[PATCH v2 0/9] sound: Use -EPROBE_DEFER instead of i915 module loading.
by Maarten Lankhorst 04 Aug '23

04 Aug '23

Explicitly loading i915 becomes a problem when upstreaming the new intel driver for Tiger Lake and higher graphics (xe). By loading i915, it doesn't wait for driver load of xe, and will fail completely before it loads. -EPROBE_DEFER has to be returned before any device is created in probe(), otherwise the removal of the device will cause EPROBE_DEFER to try again in an infinite loop. The conversion is done in gradual steps. First I add an argument to snd_hdac_i915_init to allow for -EPROBE_DEFER so I can convert each driver separately. Then I convert each driver to move snd_hdac_i915_init out of the workqueue. Finally I drop the ability to choose modprobe behavior after the last user is converted. I suspect the avs and skylake drivers used snd_hdac_i915_init purely for the modprobe, but I don't have the hardware to test if it can be safely removed. It can still be done easily in a followup patch to simplify probing. --- New since first version: - snd_hda_core.gpu_bind is added as a mechanism to force gpu binding, for testing. snd_hda_core.gpu_bind=0 forces waiting for GPU bind to off, snd_hda_core.gpu_bind=1 forces waiting for gpu bind. Default setting depends on whether kernel booted with nomodeset. - Incorporated all feedback review. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Cezary Rojewski <cezary.rojewski(a)intel.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Cc: Bard Liao <yung-chuan.liao(a)linux.intel.com> Cc: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Cc: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Daniel Baluta <daniel.baluta(a)nxp.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Cc: sound-open-firmware(a)alsa-project.org Maarten Lankhorst (9): ALSA: hda/intel: Fix error handling in azx_probe() ALSA: hda/i915: Allow override of gpu binding. ALSA: hda/i915: Add an allow_modprobe argument to snd_hdac_i915_init ALSA: hda/i915: Allow xe as match for i915_component_master_match ASoC: Intel: avs: Move snd_hdac_i915_init to before probe_work. ASoC: Intel: Skylake: Move snd_hdac_i915_init to before probe_work. ALSA: hda/intel: Move snd_hdac_i915_init to before probe_work. ASoC: SOF: Intel: Remove deferred probe for SOF ALSA: hda/i915: Remove extra argument from snd_hdac_i915_init sound/hda/hdac_i915.c | 25 ++++++++------- sound/pci/hda/hda_intel.c | 60 ++++++++++++++++++----------------- sound/soc/intel/avs/core.c | 13 +++++--- sound/soc/intel/skylake/skl.c | 31 ++++++------------ sound/soc/sof/Kconfig | 19 ----------- sound/soc/sof/core.c | 38 ++-------------------- sound/soc/sof/intel/Kconfig | 1 - sound/soc/sof/intel/hda.c | 32 +++++++++++-------- sound/soc/sof/sof-pci-dev.c | 3 +- sound/soc/sof/sof-priv.h | 5 --- 10 files changed, 85 insertions(+), 142 deletions(-) -- 2.39.2

6 37

[PATCH] ALSA: usb-audio: Add support for Mythware XA001AU capture and playback interfaces.
by dengxiang 04 Aug '23

04 Aug '23

This patch adds a USB quirk for Mythware XA001AU USB interface. Signed-off-by: dengxiang <dengxiang(a)nfschina.com> --- sound/usb/quirks-table.h | 29 +++++++++++++++++++++++++++++ 1 file changed, 29 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index efb4a3311cc5..5d72dc8441cb 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -4507,6 +4507,35 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + /* Advanced modes of the Mythware XA001AU. + * For the standard mode, Mythware XA001AU has ID ffad:a001 + */ + USB_DEVICE_VENDOR_SPEC(0xffad, 0xa001), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Mythware", + .product_name = "XA001AU", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE, + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE, + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE, + }, + { + .ifnum = -1 + } + } + } +}, #undef USB_DEVICE_VENDOR_SPEC #undef USB_AUDIO_DEVICE -- 2.30.2

2 2

[GIT PULL] ASoC fixes for v6.5-rc4
by Mark Brown 04 Aug '23

04 Aug '23

The following changes since commit 5d0c230f1de8c7515b6567d9afba1f196fb4e2f4: Linux 6.5-rc4 (2023-07-30 13:23:47 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.5-rc4 for you to fetch changes up to f803ec63686dec863a33cad87218d7d99c4b5e92: ASoC: fsl: micfil: Use dual license micfil code (2023-08-03 14:54:06 +0100) ---------------------------------------------------------------- ASoC: Fix for v6.5 Not really a fix, but rather a licensing update for the fsl_micfil driver. ---------------------------------------------------------------- Daniel Baluta (1): ASoC: fsl: micfil: Use dual license micfil code sound/soc/fsl/fsl_micfil.c | 4 ++-- sound/soc/fsl/fsl_micfil.h | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-)

2 1

[PATCH 0/5] ALSA/PCI: hda: add ARL-S support, config for MTL/LNL
by Pierre-Louis Bossart 04 Aug '23

04 Aug '23

Add a new PCI ID and i915 connectivity check for ArrowLake-S, and configuration updates for MTL/LNL. Kai Vehmanen (2): ALSA: hda: add HD Audio PCI ID for Intel Arrow Lake-S ALSA: hda/i915: extend connectivity check to cover Intel ARL Pierre-Louis Bossart (3): PCI: add ArrowLake-S PCI ID for Intel HDAudio subsystem. ALSA: hda: intel-dsp-cfg: use common include for MeteorLake ALSA: hda: intel-dsp-cfg: add LunarLake support include/linux/pci_ids.h | 1 + sound/hda/hdac_i915.c | 14 ++++++++++---- sound/hda/intel-dsp-config.c | 10 +++++++++- sound/pci/hda/hda_intel.c | 2 ++ 4 files changed, 22 insertions(+), 5 deletions(-) -- 2.39.2

4 12

[PATCH] ALSA: pcmtest: Move buffer iterator initialization to prepare callback
by Ivan Orlov 04 Aug '23

04 Aug '23

Trigger callback is not the best place for buffer iterator initialization, so move it out to the prepare callback, where it have to be. Minor enhancement: remove redundant definitions and blank line. Signed-off-by: Ivan Orlov <ivan.orlov0322(a)gmail.com> --- sound/drivers/pcmtest.c | 41 +++++++++++++++++++---------------------- 1 file changed, 19 insertions(+), 22 deletions(-) diff --git a/sound/drivers/pcmtest.c b/sound/drivers/pcmtest.c index 08e14b5eb772..7f170429eb8f 100644 --- a/sound/drivers/pcmtest.c +++ b/sound/drivers/pcmtest.c @@ -41,8 +41,6 @@ #include <linux/debugfs.h> #include <linux/delay.h> -#define DEVNAME "pcmtestd" -#define CARD_NAME "pcm-test-card" #define TIMER_PER_SEC 5 #define TIMER_INTERVAL (HZ / TIMER_PER_SEC) #define DELAY_JIFFIES HZ @@ -74,11 +72,11 @@ static u8 ioctl_reset_test; static struct dentry *driver_debug_dir; module_param(index, int, 0444); -MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard"); +MODULE_PARM_DESC(index, "Index value for pcmtest soundcard"); module_param(id, charp, 0444); -MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard"); +MODULE_PARM_DESC(id, "ID string for pcmtest soundcard"); module_param(enable, bool, 0444); -MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard."); +MODULE_PARM_DESC(enable, "Enable pcmtest soundcard."); module_param(fill_mode, short, 0600); MODULE_PARM_DESC(fill_mode, "Buffer fill mode: rand(0) or pattern(1)"); module_param(inject_delay, int, 0600); @@ -92,7 +90,6 @@ MODULE_PARM_DESC(inject_trigger_err, "Inject EINVAL error in the 'trigger' callb module_param(inject_open_err, bool, 0600); MODULE_PARM_DESC(inject_open_err, "Inject EBUSY error in the 'open' callback"); - struct pcmtst { struct snd_pcm *pcm; struct snd_card *card; @@ -405,25 +402,9 @@ static int snd_pcmtst_pcm_close(struct snd_pcm_substream *substream) static int snd_pcmtst_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct pcmtst_buf_iter *v_iter = runtime->private_data; - if (inject_trigger_err) return -EINVAL; - v_iter->sample_bytes = runtime->sample_bits / 8; - v_iter->period_bytes = frames_to_bytes(runtime, runtime->period_size); - if (runtime->access == SNDRV_PCM_ACCESS_RW_NONINTERLEAVED || - runtime->access == SNDRV_PCM_ACCESS_MMAP_NONINTERLEAVED) { - v_iter->chan_block = runtime->dma_bytes / runtime->channels; - v_iter->interleaved = false; - } else { - v_iter->interleaved = true; - } - // We want to record RATE * ch_cnt samples per sec, it is rate * sample_bytes * ch_cnt bytes - v_iter->s_rw_ch = runtime->rate / TIMER_PER_SEC; - v_iter->b_rw = v_iter->s_rw_ch * v_iter->sample_bytes * runtime->channels; - return 0; } @@ -454,8 +435,24 @@ static void pcmtst_pdev_release(struct device *dev) static int snd_pcmtst_pcm_prepare(struct snd_pcm_substream *substream) { + struct snd_pcm_runtime *runtime = substream->runtime; + struct pcmtst_buf_iter *v_iter = runtime->private_data; + if (inject_prepare_err) return -EINVAL; + + v_iter->sample_bytes = samples_to_bytes(runtime, 1); + v_iter->period_bytes = snd_pcm_lib_period_bytes(substream); + v_iter->interleaved = true; + if (runtime->access == SNDRV_PCM_ACCESS_RW_NONINTERLEAVED || + runtime->access == SNDRV_PCM_ACCESS_MMAP_NONINTERLEAVED) { + v_iter->chan_block = snd_pcm_lib_buffer_bytes(substream) / runtime->channels; + v_iter->interleaved = false; + } + // We want to record RATE * ch_cnt samples per sec, it is rate * sample_bytes * ch_cnt bytes + v_iter->s_rw_ch = runtime->rate / TIMER_PER_SEC; + v_iter->b_rw = v_iter->s_rw_ch * v_iter->sample_bytes * runtime->channels; + return 0; } -- 2.34.1

2 2

[PATCH v1 1/2] ASoC: tas2783: Add source files for tas2783 soundwire driver
by Baojun.Xu 04 Aug '23

04 Aug '23

From: Baojun Xu <baojun.xu(a)ti.com> Add source file and header file for tas2783 driver. Update Kconfig and Makefile for tas2783 driver. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- Change in v1: - Add tas2783-sdw.c - Add tas2783.h - Update Makefile for tas2783 driver. - Update Kconfig for tas2783 driver. --- sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2783-sdw.c | 1166 ++++++++++++++++++++++++++++++++ sound/soc/codecs/tas2783.h | 178 +++++ 4 files changed, 1358 insertions(+) create mode 100644 sound/soc/codecs/tas2783-sdw.c create mode 100644 sound/soc/codecs/tas2783.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c2de4ee72183..318ad867030c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -240,6 +240,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_TAS2781_COMLIB imply SND_SOC_TAS2781_FMWLIB imply SND_SOC_TAS2781_I2C + imply SND_SOC_TAS2783 imply SND_SOC_TAS5086 imply SND_SOC_TAS571X imply SND_SOC_TAS5720 @@ -1762,6 +1763,17 @@ config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" depends on I2C +config SND_SOC_TAS2783 + tristate "Texas Instruments TAS2781 speaker amplifier (sdw)" + depends on SOUNDWIRE + select REGMAP + select CRC8 + help + Enable support for Texas Instruments TAS2783 Smart Amplifier + Digital input mono Class-D and DSP-inside audio power amplifiers. + Note the TAS2783 driver implements a flexible and configurable + algo coff setting, for one, two, even multiple TAS2783 chips. + config SND_SOC_TAS571X tristate "Texas Instruments TAS571x power amplifiers" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index b48a9a323b84..3298dfa3b9d5 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -276,6 +276,7 @@ snd-soc-tas2770-objs := tas2770.o snd-soc-tas2781-comlib-objs := tas2781-comlib.o snd-soc-tas2781-fmwlib-objs := tas2781-fmwlib.o snd-soc-tas2781-i2c-objs := tas2781-i2c.o +snd-soc-tas2783-objs := tas2783-sdw.o snd-soc-tfa9879-objs := tfa9879.o snd-soc-tfa989x-objs := tfa989x.o snd-soc-tlv320adc3xxx-objs := tlv320adc3xxx.o @@ -648,6 +649,7 @@ obj-$(CONFIG_SND_SOC_TAS2780) += snd-soc-tas2780.o obj-$(CONFIG_SND_SOC_TAS2781_COMLIB) += snd-soc-tas2781-comlib.o obj-$(CONFIG_SND_SOC_TAS2781_FMWLIB) += snd-soc-tas2781-fmwlib.o obj-$(CONFIG_SND_SOC_TAS2781_I2C) += snd-soc-tas2781-i2c.o +obj-$(CONFIG_SND_SOC_TAS2783) += snd-soc-tas2783.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o obj-$(CONFIG_SND_SOC_TAS5720) += snd-soc-tas5720.o diff --git a/sound/soc/codecs/tas2783-sdw.c b/sound/soc/codecs/tas2783-sdw.c new file mode 100644 index 000000000000..93b1e7a1f311 --- /dev/null +++ b/sound/soc/codecs/tas2783-sdw.c @@ -0,0 +1,1165 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// ALSA SoC Texas Instruments TAS2783 Audio Smart Amplifier +// +// Copyright (C) 2023 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2783 driver implements a flexible and configurable +// algo coefficient setting for single TAS2783 chips. +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// Author: Kevin Lu <kevin-lu(a)ti.com> +// + +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/pm_runtime.h> +#include <linux/err.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_registers.h> +#include <linux/soundwire/sdw_type.h> +#include <linux/init.h> +#include <linux/pm.h> +#include <linux/of.h> +#include <linux/slab.h> +#include <linux/crc8.h> +#include <linux/crc32.h> +#include <linux/efi.h> +#include <linux/firmware.h> +#include <linux/of_gpio.h> +#include <linux/interrupt.h> +#include <sound/soc.h> +#include <sound/sdw.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> +#include <sound/tas2781-tlv.h> + +#include "tas2783.h" + +static const unsigned int tas2783_calibration_reg[] = { + TAS2783_CALIBRATION_RE, + TAS2783_CALIBRATION_RE_LOW, + TAS2783_CALIBRATION_INV_RE, + TAS2783_CALIBRATION_POW, + TAS2783_CALIBRATION_TLIMIT, + 0, +}; + +static const struct reg_default tas2783_reg_defaults[] = { + // Default values for ROM mode. Activated. + { 0x8002, 0x1a}, + { 0x800e, 0x44}, + { 0x800f, 0x40}, + { 0x8097, 0xc8}, + { 0x80b5, 0x74}, + { 0x8099, 0x20}, + { 0xfe8d, 0x0d}, + { 0xfebe, 0x4a}, + { 0x8230, 0x00}, + { 0x8231, 0x00}, + { 0x8232, 0x00}, + { 0x8233, 0x01}, + { 0x8418, 0x00}, + { 0x8419, 0x00}, + { 0x841a, 0x00}, + { 0x841b, 0x00}, + { 0x8428, 0x40}, + { 0x8429, 0x00}, + { 0x842a, 0x00}, + { 0x842b, 0x00}, + { 0x8548, 0x00}, + { 0x8549, 0x00}, + { 0x854a, 0x00}, + { 0x854b, 0x00}, + { 0x8558, 0x40}, + { 0x8559, 0x00}, + { 0x855a, 0x00}, + { 0x855b, 0x00}, + { 0x800a, 0x3a}, + { 0x800e, 0x44}, + { 0x800f, 0x40}, + { 0x805c, 0xd9}, + { 0x8002, 0x00}, +}; + +static bool tas2783_readable_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + case 0x8000 ... 0xc000: // Page 0 ~ 127. + case 0xfe80 ... 0xfeff: // Page 253. + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, TAS2783_SDCA_ENT_UDMPU21, + TAS2783_SDCA_CTL_UDMPU_CLUSTER, 0): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, TAS2783_SDCA_ENT_FU21, + TAS2783_SDCA_CTL_FU_MUTE, CH_L): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, TAS2783_SDCA_ENT_FU21, + TAS2783_SDCA_CTL_FU_MUTE, CH_R): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, TAS2783_SDCA_ENT_PDE23, + TAS2783_SDCA_CTL_REQ_POWER_STATE, 0): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, TAS2783_SDCA_ENT_PDE22, + TAS2783_SDCA_CTL_REQ_POWER_STATE, 0): + return true; + default: + return false; + } +} + +static bool tas2783_volatile_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + case 0x8001: + // Only reset register was volatiled. + return true; + default: + return false; + } +} + +static const struct regmap_config tasdevice_regmap = { + .reg_bits = 32, + .val_bits = 8, + .readable_reg = tas2783_readable_register, + .volatile_reg = tas2783_volatile_register, + .max_register = 0x41008000 + TASDEVICE_REG(0xa1, 0x60, 0x7f), + .reg_defaults = tas2783_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas2783_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .use_single_read = true, + .use_single_write = true, +}; + +static int tas2783_sdw_read(void *context, const void *reg_buf, + const size_t reg_size, void *val_buf, size_t val_size) +{ + struct sdw_slave *peripheral = context; + u8 *buf8 = val_buf; + unsigned int reg, bytes; + int ret; + + reg = le32_to_cpu(*(const __le32 *)reg_buf); + reg += TAS2783_SDW_ADDR_OFFSET; + + while (val_size) { + /* to end of page */ + bytes = SDW_REG_NO_PAGE - (reg & SDW_REGADDR); + if (bytes > val_size) + bytes = val_size; + + ret = sdw_nread_no_pm(peripheral, reg, bytes, buf8); + if (ret != 0) { + dev_err(&peripheral->dev, "Read @%#x..%#x:%d\n", + reg, reg + bytes - 1, ret); + return ret; + } + + swab32_array((u32 *)buf8, bytes / 4); + val_size -= bytes; + reg += bytes; + buf8 += bytes; + } + + return 0; +} + +/* Convert for LSB or MSB. */ +static inline void tas2783_swab_copy( + void *dest, const void *src, size_t nbytes) +{ + u32 *dest32 = dest; + const u32 *src32 = src; + + for (; nbytes > 0; nbytes -= 4) + *dest32++ = swab32(*src32++); +} + +static int tas2783_sdw_gather_write(void *context, const void *reg_buf, + size_t reg_size, const void *val_buf, size_t val_size) +{ + struct sdw_slave *peripheral = context; + const u8 *src_be = val_buf; + u32 val_le_buf[64]; /* Define u32 so it is 32-bit aligned */ + unsigned int reg, bytes; + int ret; + + reg = le32_to_cpu(*(const __le32 *)reg_buf); + + while (val_size) { + /* to end of page */ + bytes = SDW_REG_NO_PAGE - (reg & SDW_REGADDR); + if (bytes > val_size) + bytes = val_size; + if (bytes > sizeof(val_le_buf)) + bytes = sizeof(val_le_buf); + + tas2783_swab_copy(val_le_buf, src_be, bytes); + + ret = sdw_nwrite_no_pm(peripheral, reg, bytes, + (u8 *)val_le_buf); + if (ret != 0) { + dev_err(&peripheral->dev, "Write @%#x..%#x:%d\n", + reg, reg + bytes - 1, ret); + return ret; + } + + val_size -= bytes; + reg += bytes; + src_be += bytes; + } + + return 0; +} + +static int tas2783_sdw_write( + void *context, const void *val_buf, size_t val_size) +{ + const u8 *src_buf = val_buf; + + /* First word of val_buf contains the destination address */ + return tas2783_sdw_gather_write(context, &src_buf[0], 4, + &src_buf[4], val_size - 4); +} + +static int tasdevice_clamp(int val, int max, unsigned int invert) +{ + if (val > max) + val = max; + if (invert) + val = max - val; + if (val < 0) + val = 0; + return val; +} + +/* + * Registers are big-endian on I2C and SPI but little-endian on SoundWire. + * Exported firmware controls are big-endian on I2C/SPI but little-endian + * on SoundWire. + * Firmware files are always big-endian and are opaque blobs. + * Present a big-endian regmap and hide the endianness swap, + * so that the ALSA byte controls always have the same byte order, + * and firmware file blobs can be written verbatim. + */ +static const struct regmap_bus tas2783_regmap_bus_sdw = { + .read = tas2783_sdw_read, + .write = tas2783_sdw_write, + .gather_write = tas2783_sdw_gather_write, + .reg_format_endian_default = REGMAP_ENDIAN_LITTLE, + .val_format_endian_default = REGMAP_ENDIAN_BIG, +}; + +static int tas2783_digital_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int val; + unsigned int invert = mc->invert; + int max = mc->max; + int ret = 0; + + /* Read the primary device as the whole */ + ret = tasdevice_dev_read(tas_dev, &mc->reg, (unsigned int *)&val); + dev_dbg(tas_dev->dev, "%s, get digital vol %d from %x with %d\n", + __func__, val, mc->reg, ret); + if (ret) { + dev_err(tas_dev->dev, "%s, get digital vol error %x.\n", + __func__, ret); + goto out; + } + + if (val > max) + val = max; + if (invert) + val = max - val; + if (val < 0) + val = 0; + ucontrol->value.integer.value[0] = val; + +out: + return ret; +} + +static int tas2783_digital_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int val, ret = 0; + unsigned int invert = mc->invert; + int max = mc->max; + unsigned char value = 0; + + val = ucontrol->value.integer.value[0]; + if (val > max) + val = max; + if (invert) + val = max - val; + if (val < 0) + val = 0; + + value = (u8)val; + ret = sdw_nwrite_no_pm(tas_dev->sdw_slave, mc->reg, 1, &value); + if (ret != 0) { + dev_err(tas_dev->dev, "Write @%#x..%#x:%d\n", + mc->reg, val, ret); + return ret; + } + dev_dbg(tas_dev->dev, "%s, Put digital vol %d into %x.\n", + __func__, val, mc->reg); + + return 1; +} + +static int tas2783_amp_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int val; + unsigned char mask = 0; + int max = mc->max; + int ret = 0; + unsigned int invert = mc->invert; + unsigned char value = 0; + + /* Read the primary device */ + ret = sdw_nread_no_pm(tas_dev->sdw_slave, mc->reg, 1, &value); + dev_dbg(tas_dev->dev, "%s, get AMP vol %d from %x with %d\n", + __func__, value, mc->reg, ret); + + mask = (1 << fls(mc->max)) - 1; + mask <<= mc->shift; + val = value; + val = (val & mask) >> mc->shift; + if (val > max) + val = max; + if (invert) + val = max - val; + if (val < 0) + val = 0; + ucontrol->value.integer.value[0] = val; + + return ret; +} + +static int tas2783_amp_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int val = 0, ret = 0, max = mc->max; + unsigned char mask = 0, value = 0; + unsigned int invert = mc->invert; + + mask = (1 << fls(mc->max)) - 1; + mask <<= mc->shift; + val = tasdevice_clamp(ucontrol->value.integer.value[0], max, + invert); + + ret = sdw_nread_no_pm(tas_dev->sdw_slave, mc->reg, 1, &value); + if (ret != 0) { + dev_err(tas_dev->dev, "Read @%x.%x:%d\n", + mc->reg, value, ret); + return ret; + } + value &= (~TAS2783_AMP_LEVEL_MASK); + value |= (val << mc->shift); + ret = sdw_nwrite_no_pm(tas_dev->sdw_slave, mc->reg, 1, &value); + if (ret != 0) { + dev_err(tas_dev->dev, "Write @%#x..%#x:%d\n", + mc->reg, val, ret); + return ret; + } + dev_dbg(tas_dev->dev, "amp %x into 0x%08x by %lx shift %d.\n", + val, mc->reg, TAS2783_AMP_LEVEL_MASK, mc->shift); + + return ret; +} + +static const struct snd_kcontrol_new tas2783_snd_controls[] = { + SOC_SINGLE_RANGE_EXT_TLV("Amp Gain Volume", TAS2783_AMP_LEVEL, + 1, 0, 20, 0, tas2783_amp_getvol, + tas2783_amp_putvol, amp_vol_tlv), + SOC_SINGLE_RANGE_EXT_TLV("Digital Volume", TAS2783_DVC_LVL, + 0, 0, 200, 1, tas2783_digital_getvol, + tas2783_digital_putvol, dvc_tlv), +}; + +static void tas2783_apply_calib( + struct tasdevice_priv *tas_priv, unsigned int *cali_data) +{ + unsigned int reg_address = 0, i = 0; + unsigned int *reg_start; + + if (!tas_priv->sdw_slave) + return; + if ((tas_priv->sdw_slave->id.unique_id < TAS2783_DEVICE_ID_START) + || (tas_priv->sdw_slave->id.unique_id > TAS2783_DEVICE_ID_END)) + return; + reg_start = cali_data+(tas_priv->sdw_slave->id.unique_id - + TAS2783_DEVICE_ID_START)*TAS2783_CALIBRATION_SIZE; + while (tas2783_calibration_reg[i] != 0) { + reg_address = tas2783_calibration_reg[i]; + tas2783_sdw_gather_write(tas_priv->sdw_slave, + &reg_address, 4, reg_start+i, 4); + } +} + +static int tas2783_calibration(struct tasdevice_priv *tas_priv) +{ + efi_guid_t efi_guid = EFI_GUID(0x1f52d2a1, 0xbb3a, 0x457d, 0xbc, + 0x09, 0x43, 0xa3, 0xf4, 0x31, 0x0a, 0x92); + static efi_char16_t efi_name[] = L"CALI_DATA"; + struct tm *tm = &tas_priv->tm; + unsigned int attr, crc; + unsigned int *tmp_val; + efi_status_t status; + + tas_priv->cali_data.total_sz = 128; + /* Get real size of UEFI variable */ + status = efi.get_variable(efi_name, &efi_guid, &attr, + &tas_priv->cali_data.total_sz, tas_priv->cali_data.data); + dev_dbg(tas_priv->dev, "cali get %lx bytes with result : %ld\n", + tas_priv->cali_data.total_sz, status); + if (status == EFI_BUFFER_TOO_SMALL) { + status = efi.get_variable(efi_name, &efi_guid, &attr, + &tas_priv->cali_data.total_sz, + tas_priv->cali_data.data); + dev_dbg(tas_priv->dev, "cali get %lx bytes result:%ld\n", + tas_priv->cali_data.total_sz, status); + } + for (int i = 0; i < tas_priv->cali_data.total_sz; i += 4) { + dev_dbg(tas_priv->dev, "cali get %02x %02x %02x %02x", + tas_priv->cali_data.data[i], + tas_priv->cali_data.data[i+1], + tas_priv->cali_data.data[i+2], + tas_priv->cali_data.data[i+3]); + } + + tmp_val = (unsigned int *)tas_priv->cali_data.data; + + crc = crc32(~0, tas_priv->cali_data.data, 84) ^ ~0; + dev_dbg(tas_priv->dev, "cali crc 0x%08x PK tmp_val 0x%08x\n", + crc, tmp_val[21]); + + if (crc == tmp_val[21]) { + time64_to_tm(tmp_val[20], 0, tm); + dev_dbg(tas_priv->dev, "%4ld-%2d-%2d, %2d:%2d:%2d\n", + tm->tm_year, tm->tm_mon, tm->tm_mday, + tm->tm_hour, tm->tm_min, tm->tm_sec); + tas2783_apply_calib(tas_priv, tmp_val); + } else { + dev_dbg(tas_priv->dev, "CRC error!\n"); + tas_priv->cali_data.total_sz = 0; + } + + return 0; +} + +static void tasdevice_rca_ready(const struct firmware *fmw, void *context) +{ + struct tasdevice_priv *tas_dev = + (struct tasdevice_priv *) context; + struct MYSWFTFile *p; + unsigned char *buf = NULL, value_sdw; + int offset = 0, i = 0, num_files = 0, img_sz = 0; + int ret = 0; + + mutex_lock(&tas_dev->codec_lock); + + if (!fmw || !fmw->data) { + dev_err(tas_dev->dev, + "Failed to read %s, no side - effect on driver running\n", + tas_dev->rca_binaryname); + ret = -1; + goto out; + } + buf = (unsigned char *)fmw->data; + + img_sz = SMS_HTONL(buf[offset+3], buf[offset + 2], + buf[offset + 1], buf[offset]); + dev_dbg(tas_dev->dev, "Got %x:%lx.\n", img_sz, fmw->size); + offset += sizeof(img_sz); + if (img_sz != fmw->size) { + dev_err(tas_dev->dev, + "Size not match, %d %u", (int)fmw->size, img_sz); + ret = -1; + goto out; + } + + while ((offset < img_sz) && (num_files < MAX_NO_FILES)) { + p = (struct MYSWFTFile *)(buf + offset); + tas_dev->m_swftFile[num_files].m_vendor_id = + p->m_vendor_id; + tas_dev->m_swftFile[num_files].m_file_id = p->m_file_id; + tas_dev->m_swftFile[num_files].m_version = p->m_version; + tas_dev->m_swftFile[num_files].m_Length = p->m_Length; + tas_dev->m_swftFile[num_files].m_downloadAddr = + p->m_downloadAddr; + tas_dev->m_swftFile[num_files].m_startAdress = + ((char *)p) + sizeof(unsigned int)*5; + + ret = sdw_nwrite_no_pm(tas_dev->sdw_peripheral, + tas_dev->m_swftFile[num_files].m_downloadAddr, + tas_dev->m_swftFile[num_files].m_Length, + tas_dev->m_swftFile[num_files].m_startAdress); + dev_dbg(tas_dev->dev, "Write %x into %x result : %d\n", + tas_dev->m_swftFile[num_files].m_Length, + tas_dev->m_swftFile[num_files].m_downloadAddr, + ret); + + dev_dbg(tas_dev->dev, "Write %d :%x:%x:%x:%x:%x:%x.\n", + num_files, + tas_dev->m_swftFile[num_files].m_vendor_id, + tas_dev->m_swftFile[num_files].m_file_id, + tas_dev->m_swftFile[num_files].m_version, + tas_dev->m_swftFile[num_files].m_Length, + tas_dev->m_swftFile[num_files].m_downloadAddr, + tas_dev->m_swftFile[num_files].m_startAdress[0]); + + ret = sdw_nread_no_pm(tas_dev->sdw_peripheral, + tas_dev->m_swftFile[num_files].m_downloadAddr, + 1, &value_sdw); + dev_dbg(tas_dev->dev, "Got block %d :%x.\n", + num_files, value_sdw); + + offset += sizeof(unsigned int)*5 + p->m_Length; + num_files++; + } + + i = 0x50; + while (i < 0x56) { + ret = sdw_nread_no_pm(tas_dev->sdw_peripheral, i, 1, + &value_sdw); + dev_dbg(tas_dev->dev, "Got device id:%x\n", value_sdw); + i++; + } + tas2783_calibration(tas_dev); + +out: + mutex_unlock(&tas_dev->codec_lock); + if (fmw) + release_firmware(fmw); +} + +static void tasdevice_config_info_remove(void *context) +{ + struct tasdevice_priv *tas_dev = + (struct tasdevice_priv *) context; + dev_dbg(tas_dev->dev, "%s was performed\n", __func__); +} + +static int tasdevice_dapm_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *codec = + snd_soc_dapm_to_component(w->dapm); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + int state = 1; + + /* Codec Lock Hold */ + mutex_lock(&tas_dev->codec_lock); + dev_dbg(tas_dev->dev, "dapm event : %x\n", event); + if (event == SND_SOC_DAPM_PRE_PMD) + state = 0; + /* Codec Lock Release*/ + mutex_unlock(&tas_dev->codec_lock); + + return 0; +} + +static const struct snd_soc_dapm_widget tasdevice_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("ASI", "ASI Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT_E("ASI OUT", "ASI Capture", 0, SND_SOC_NOPM, + 0, 0, tasdevice_dapm_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_SPK("SPK", tasdevice_dapm_event), + SND_SOC_DAPM_OUTPUT("OUT"), + SND_SOC_DAPM_INPUT("DMIC") +}; + +static const struct snd_soc_dapm_route tasdevice_audio_map[] = { + {"SPK", NULL, "ASI"}, + {"OUT", NULL, "SPK"}, + {"ASI OUT", NULL, "DMIC"} +}; + +static int tasdevice_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int ret = 0; + + return ret; +} + +static int tasdevice_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct tasdevice_priv *tas_dev = + snd_soc_dai_get_drvdata(codec_dai); + + tas_dev->sysclk = freq; + + return 0; +} + +static int tasdevice_set_sdw_stream( + struct snd_soc_dai *dai, void *sdw_stream, int direction) +{ + struct sdw_stream_data *stream; + + if (!sdw_stream) + return 0; + + stream = kzalloc(sizeof(*stream), GFP_KERNEL); + if (!stream) + return -ENOMEM; + + stream->sdw_stream = sdw_stream; + + /* Use tx_mask or rx_mask to set dma_data */ + snd_soc_dai_dma_data_set(dai, direction, stream); + + return 0; +} + +static void tasdevice_sdw_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct sdw_stream_data *stream; + + stream = snd_soc_dai_get_dma_data(dai, substream); + snd_soc_dai_set_dma_data(dai, substream, NULL); + kfree(stream); +} + +static int tasdevice_sdw_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tasdevice = + snd_soc_component_get_drvdata(component); + struct sdw_stream_config stream_config = {0}; + struct sdw_port_config port_config = {0}; + struct sdw_stream_data *stream; + int retval; + + dev_dbg(dai->dev, "%s %s", __func__, dai->name); + stream = snd_soc_dai_get_dma_data(dai, substream); + + if (!stream) + return -EINVAL; + + if (!tasdevice->sdw_slave) + return -EINVAL; + + /* SoundWire specific configuration */ + snd_sdw_params_to_config(substream, params, + &stream_config, &port_config); + + /* port 1 for playback */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + port_config.num = 1; + else + port_config.num = 2; + + retval = sdw_stream_add_slave(tasdevice->sdw_slave, + &stream_config, &port_config, 1, stream->sdw_stream); + if (retval) { + dev_err(dai->dev, "Unable to configure port\n"); + return retval; + } + + return 0; +} + +static int tasdevice_sdw_pcm_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tasdevice = + snd_soc_component_get_drvdata(component); + struct sdw_stream_data *stream = + snd_soc_dai_get_dma_data(dai, substream); + + if (!tasdevice->sdw_slave) + return -EINVAL; + + sdw_stream_remove_slave(tasdevice->sdw_slave, stream->sdw_stream); + return 0; +} + +static int tasdevice_mute(struct snd_soc_dai *dai, int mute, + int direction) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tasdevice = + snd_soc_component_get_drvdata(component); + int ret = 0; + unsigned char value_sdw = 0; + + dev_dbg(tasdevice->dev, "%s: mute %d, %d\n", + __func__, mute, direction); + if (mute == 0) {// Unmute. + // FU21 Unmute + value_sdw = 0; + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + SDW_SDCA_CTL(1, 1, 1, 0), 1, &value_sdw); + dev_dbg(tasdevice->dev, "Unmuted %lx %x\n", + SDW_SDCA_CTL(1, 1, 1, 0), ret); + value_sdw = 0; + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + 0x40400090, 1, &value_sdw); + dev_dbg(tasdevice->dev, "Unmuted %x\n", ret); + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + 0x40402090, 1, &value_sdw); + dev_dbg(tasdevice->dev, "Unmuted %x\n", ret); + + // FU23 Unmute + value_sdw = 0; + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + SDW_SDCA_CTL(1, 2, 1, 0), 1, &value_sdw); + dev_dbg(tasdevice->dev, "Unmuted %lx %x\n", + SDW_SDCA_CTL(1, 2, 1, 0), ret); + value_sdw = 0; + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + 0x40400108, 1, &value_sdw); + dev_dbg(tasdevice->dev, "Unmuted %x\n", ret); + value_sdw = 0; + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + 0x8002, 1, &value_sdw); + dev_dbg(tasdevice->dev, "Unmuted %x\n", ret); + } else { + // FU21 mute + value_sdw = 1; + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + SDW_SDCA_CTL(1, 1, 1, 0), 1, &value_sdw); + dev_dbg(tasdevice->dev, "muted %lx %x\n", + SDW_SDCA_CTL(1, 1, 1, 0), ret); + value_sdw = 1; + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + 0x40400090, 1, &value_sdw); + dev_dbg(tasdevice->dev, "muted %x\n", ret); + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + 0x40402090, 1, &value_sdw); + dev_dbg(tasdevice->dev, "muted %x\n", ret); + // FU23 mute + value_sdw = 1; + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + SDW_SDCA_CTL(1, 2, 1, 0), 1, &value_sdw); + dev_dbg(tasdevice->dev, "muted %lx %x\n", + SDW_SDCA_CTL(1, 2, 1, 0), ret); + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + 0x40400108, 1, &value_sdw); + dev_dbg(tasdevice->dev, "muted %x\n", ret); + value_sdw = 0x1a; + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + 0x8002, 1, &value_sdw); + dev_dbg(tasdevice->dev, "muted %x\n", ret); + } + + if (!tasdevice->sdw_slave) + return -EINVAL; + + return 0; +} + +static const struct snd_soc_dai_ops tasdevice_dai_ops = { + .startup = tasdevice_startup, + .mute_stream = tasdevice_mute, + .hw_params = tasdevice_sdw_hw_params, + .set_sysclk = tasdevice_set_dai_sysclk, + .hw_free = tasdevice_sdw_pcm_hw_free, + .set_stream = tasdevice_set_sdw_stream, + .shutdown = tasdevice_sdw_shutdown, +}; + +static struct snd_soc_dai_driver tasdevice_dai_driver[] = { + { + .name = "tas2783_codec", + .id = 0, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 4, + .rates = TASDEVICE_RATES, + .formats = TASDEVICE_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 1, + .rates = TASDEVICE_RATES, + .formats = TASDEVICE_FORMATS, + }, + .ops = &tasdevice_dai_ops, + .symmetric_rate = 1, + }, +}; + +static void tas2783_reset(struct tasdevice_priv *tas_dev) +{ + int ret = 0; + unsigned char value_sdw; + + value_sdw = TAS2873_REG_SWRESET_RESET; + ret = sdw_nwrite_no_pm(tas_dev->sdw_peripheral, + TAS2873_REG_SWRESET, 1, &value_sdw); + dev_dbg(tas_dev->dev, "%s TAS2783 was reseted.\n", __func__); + usleep_range(1000, 1050); +} + +static int tasdevice_codec_probe(struct snd_soc_component *codec) +{ + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + int ret = 0; + + dev_dbg(tas_dev->dev, "%s called for TAS2783 start.\n", + __func__); + /* Codec Lock Hold */ + mutex_lock(&tas_dev->codec_lock); + + crc8_populate_msb(tas_dev->crc8_lkp_tbl, + TASDEVICE_CRC8_POLYNOMIAL); + tas_dev->codec = codec; + + scnprintf(tas_dev->rca_binaryname, 64, "MY_SWFT_x%01x.bin", + tas_dev->sdw_peripheral->id.unique_id); + + ret = request_firmware_nowait(THIS_MODULE, FW_ACTION_UEVENT, + tas_dev->rca_binaryname, tas_dev->dev, GFP_KERNEL, + tas_dev, tasdevice_rca_ready); + dev_dbg(tas_dev->dev, + "%s: request_firmware %x open status: 0x%08x\n", + __func__, tas_dev->sdw_peripheral->id.unique_id, ret); + + /* Codec Lock Release*/ + mutex_unlock(&tas_dev->codec_lock); + + tas2783_reset(tas_dev); + + dev_dbg(tas_dev->dev, "%s was called end.\n", __func__); + return ret; +} + +static void tasdevice_deinit(void *context) +{ + struct tasdevice_priv *tas_dev = + (struct tasdevice_priv *) context; + + tasdevice_config_info_remove(tas_dev); +} + +static void tasdevice_codec_remove( + struct snd_soc_component *codec) +{ + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + /* Codec Lock Hold */ + mutex_lock(&tas_dev->codec_lock); + tasdevice_deinit(tas_dev); + /* Codec Lock Release*/ + mutex_unlock(&tas_dev->codec_lock); +} + +static const struct snd_soc_component_driver + soc_codec_driver_tasdevice = { + .probe = tasdevice_codec_probe, + .remove = tasdevice_codec_remove, + .controls = tas2783_snd_controls, + .num_controls = ARRAY_SIZE(tas2783_snd_controls), + .dapm_widgets = tasdevice_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tasdevice_dapm_widgets), + .dapm_routes = tasdevice_audio_map, + .num_dapm_routes = ARRAY_SIZE(tasdevice_audio_map), + .idle_bias_on = 1, + .endianness = 1, +}; + +int tasdevice_dev_read(struct tasdevice_priv *tas_dev, unsigned int *reg, + unsigned int *val) +{ + int ret = 0; + + mutex_lock(&tas_dev->dev_lock); + ret = tas2783_sdw_read(tas_dev->sdw_peripheral, + reg, 4, (void *)val, 4); + dev_dbg(tas_dev->dev, "%s, read %x from reg %x. Result is (%d)\n", + __func__, *val, *reg, ret); + mutex_unlock(&tas_dev->dev_lock); + return ret; +} + +static int tasdevice_init(struct tasdevice_priv *tas_dev) +{ + int ret = 0; + + tas_dev->cur_prog = -1; + tas_dev->cur_conf = -1; + + dev_dbg(tas_dev->dev, "%s was called for TAS2783.\n", __func__); + mutex_init(&tas_dev->dev_lock); + + dev_set_drvdata(tas_dev->dev, tas_dev); + + mutex_init(&tas_dev->codec_lock); + ret = devm_snd_soc_register_component(tas_dev->dev, + &soc_codec_driver_tasdevice, + tasdevice_dai_driver, ARRAY_SIZE(tasdevice_dai_driver)); + if (ret) { + dev_err(tas_dev->dev, "%s: codec register error:0x%08x\n", + __func__, ret); + } + dev_dbg(tas_dev->dev, "%s was called for TAS2783.\n", __func__); + + return ret; +} + +static int tasdevice_read_prop(struct sdw_slave *slave) +{ + struct sdw_slave_prop *prop = &slave->prop; + int nval; + int i, j; + u32 bit; + unsigned long addr; + struct sdw_dpn_prop *dpn; + + prop->scp_int1_mask = + SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; + prop->quirks = SDW_SLAVE_QUIRKS_INVALID_INITIAL_PARITY; + + prop->paging_support = true; + + /* first we need to allocate memory for set bits in port lists */ + prop->source_ports = 0x04; /* BITMAP: 00000100 */ + prop->sink_ports = 0x2; /* BITMAP: 00000010 */ + + nval = hweight32(prop->source_ports); + prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->src_dpn_prop), GFP_KERNEL); + if (!prop->src_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->src_dpn_prop; + addr = prop->source_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* do this again for sink now */ + nval = hweight32(prop->sink_ports); + prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->sink_dpn_prop), GFP_KERNEL); + if (!prop->sink_dpn_prop) + return -ENOMEM; + + j = 0; + dpn = prop->sink_dpn_prop; + addr = prop->sink_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[j].num = bit; + dpn[j].type = SDW_DPN_FULL; + dpn[j].simple_ch_prep_sm = true; + dpn[j].ch_prep_timeout = 10; + j++; + } + + /* set the timeout values */ + prop->clk_stop_timeout = 20; + + dev_dbg(&slave->dev, "%s was performed.\n", __func__); + + return 0; +} + +static int tasdevice_io_init(struct device *dev, struct sdw_slave *slave) +{ + struct tasdevice_priv *tasdevice = dev_get_drvdata(dev); + + if (tasdevice->hw_init) + return 0; + + if (tasdevice->first_hw_init) { + regcache_cache_only(tasdevice->regmap, false); + regcache_cache_bypass(tasdevice->regmap, true); + } else { + /* PM runtime is only enabled when + * a Slave reports as Attached + * set autosuspend parameters + */ + pm_runtime_set_autosuspend_delay(&slave->dev, 3000); + pm_runtime_use_autosuspend(&slave->dev); + + /* update count of parent 'active' children */ + pm_runtime_set_active(&slave->dev); + + /* make sure the device does not suspend immediately */ + pm_runtime_mark_last_busy(&slave->dev); + + pm_runtime_enable(&slave->dev); + } + + pm_runtime_get_noresume(&slave->dev); + + if (tasdevice->first_hw_init) { + regcache_cache_bypass(tasdevice->regmap, false); + regcache_mark_dirty(tasdevice->regmap); + } else + tasdevice->first_hw_init = true; + + /* Mark Slave initialization complete */ + tasdevice->hw_init = true; + + pm_runtime_mark_last_busy(&slave->dev); + pm_runtime_put_autosuspend(&slave->dev); + + dev_dbg(&slave->dev, "%s hw_init complete\n", __func__); + return 0; +} + +static int tasdevice_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct tasdevice_priv *tasdevice = dev_get_drvdata(&slave->dev); + + /* Update the status */ + tasdevice->status = status; + + if (status == SDW_SLAVE_UNATTACHED) + tasdevice->hw_init = false; + + /* Perform initialization only if slave status + * is present and hw_init flag is false + */ + if (tasdevice->hw_init || tasdevice->status != SDW_SLAVE_ATTACHED) + return 0; + + /* perform I/O transfers required for Slave initialization */ + return tasdevice_io_init(&slave->dev, slave); +} + +/* + * slave_ops: callbacks for get_clock_stop_mode, clock_stop and + * port_prep are not defined for now + */ +static const struct sdw_slave_ops tasdevice_sdw_ops = { + .read_prop = tasdevice_read_prop, + .update_status = tasdevice_update_status, +}; + +static void tasdevice_remove(struct tasdevice_priv *tas_dev) +{ + snd_soc_unregister_component(tas_dev->dev); + if (gpio_is_valid(tas_dev->rst_gpio)) + gpio_free(tas_dev->rst_gpio); + + if (gpio_is_valid(tas_dev->irq_info.irq_gpio)) + gpio_free(tas_dev->irq_info.irq_gpio); + + mutex_destroy(&tas_dev->dev_lock); + mutex_destroy(&tas_dev->codec_lock); +} + +static int tasdevice_sdw_probe(struct sdw_slave *peripheral, + const struct sdw_device_id *id) +{ + struct device *dev = &peripheral->dev; + struct tasdevice_priv *tas_dev; + int ret = 0; + + dev_dbg(dev, "%s was called for TAS2783.\n", __func__); + + tas_dev = devm_kzalloc(dev, sizeof(*tas_dev), GFP_KERNEL); + if (!tas_dev) { + ret = -ENOMEM; + goto out; + } + tas_dev->dev = &peripheral->dev; + tas_dev->sdw_peripheral = peripheral; + tas_dev->chip_id = id->driver_data; + tas_dev->sdw_slave = peripheral; + tas_dev->hw_init = false; + tas_dev->first_hw_init = false; + + dev_dbg(dev, "%d chip id %x for TAS2783.\n", + peripheral->id.unique_id, tas_dev->chip_id); + + dev_set_drvdata(dev, tas_dev); + + tas_dev->regmap = devm_regmap_init(dev, &tas2783_regmap_bus_sdw, + peripheral, &tasdevice_regmap); + if (IS_ERR(tas_dev->regmap)) { + ret = PTR_ERR(tas_dev->regmap); + dev_err(dev, "Failed devm_regmap_init: %d\n", ret); + goto out; + } + ret = tasdevice_init(tas_dev); + +out: + if (ret < 0 && tas_dev != NULL) + tasdevice_remove(tas_dev); + + dev_dbg(dev, "%s was called for TAS2783.\n", __func__); + return ret; + +} + +static int tasdevice_sdw_remove(struct sdw_slave *peripheral) +{ + struct tasdevice_priv *tas_dev = + dev_get_drvdata(&peripheral->dev); + + if (tas_dev) + tasdevice_remove(tas_dev); + + return 0; +} + +static const struct sdw_device_id tasdevice_sdw_id[] = { + SDW_SLAVE_ENTRY(0x0102, 0x0, 0), + {}, +}; +MODULE_DEVICE_TABLE(sdw, tasdevice_sdw_id); + +static struct sdw_driver tasdevice_sdw_driver = { + .driver = { + .name = "slave-tas2783", + }, + .probe = tasdevice_sdw_probe, + .remove = tasdevice_sdw_remove, + .ops = &tasdevice_sdw_ops, + .id_table = tasdevice_sdw_id, +}; + +module_sdw_driver(tasdevice_sdw_driver); + +MODULE_AUTHOR("Baojun Xu <baojun.xu(a)ti.com>"); +MODULE_DESCRIPTION("ASoC TAS2783 SoundWire Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2783.h b/sound/soc/codecs/tas2783.h new file mode 100644 index 000000000000..0fa5b3855b51 --- /dev/null +++ b/sound/soc/codecs/tas2783.h @@ -0,0 +1,178 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier + * + * Copyright (C) 2023 Texas Instruments Incorporated + * https://www.ti.com + * + * The TAS2783 driver implements a flexible and configurable + * algo coff setting for single TAS2783 chips. + * + * Author: Baojun Xu <baojun.xu(a)ti.com> + */ + +#ifndef __TAS2783_H__ +#define __TAS2783_H__ + +#define TASDEVICE_RATES (SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_88200) +#define TAS2783_SDW_ADDR_OFFSET 0x8000 + +#define TASDEVICE_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +/* BOOK, PAGE Control Register */ +#define TASDEVICE_REG(book, page, reg) ((book * 256 * 256) + 0x8000 +\ + (page * 128) + reg) + +/*Software Reset */ +#define TAS2873_REG_SWRESET TASDEVICE_REG(0x0, 0X0, 0x01) +#define TAS2873_REG_SWRESET_RESET BIT(0) + +/* Volume control */ +#define TAS2783_DVC_LVL TASDEVICE_REG(0x0, 0x00, 0x1A) +#define TAS2783_AMP_LEVEL TASDEVICE_REG(0x0, 0x00, 0x03) +#define TAS2783_AMP_LEVEL_MASK GENMASK(5, 1) + +/* Calibration data */ +#define TAS2783_CALIBRATION_RE TASDEVICE_REG(0x0, 0x17, 0x74) +#define TAS2783_CALIBRATION_RE_LOW TASDEVICE_REG(0x0, 0x18, 0x14) +#define TAS2783_CALIBRATION_INV_RE TASDEVICE_REG(0x0, 0x18, 0x0c) +#define TAS2783_CALIBRATION_POW TASDEVICE_REG(0x0, 0x0d, 0x3c) +#define TAS2783_CALIBRATION_TLIMIT TASDEVICE_REG(0x0, 0x18, 0x7c) + +#define TAS2783_CALIBRATION_SIZE 5 // int data number. + +#define TAS2783_DEVICE_ID_START 0x08 // Unique id start +#define TAS2783_DEVICE_ID_END 0x0F // Unique id end + +/* TAS2783 SDCA Control - function number */ +#define FUNC_NUM_SMART_AMP 0x01 + +/* TAS2783 SDCA entity */ +#define TAS2783_SDCA_ENT_PDE23 0x0C +#define TAS2783_SDCA_ENT_PDE22 0x0B +#define TAS2783_SDCA_ENT_FU21 0x01 +#define TAS2783_SDCA_ENT_UDMPU21 0x10 + +/* TAS2783 SDCA control */ +#define TAS2783_SDCA_CTL_REQ_POWER_STATE 0x01 +#define TAS2783_SDCA_CTL_FU_MUTE 0x01 +#define TAS2783_SDCA_CTL_UDMPU_CLUSTER 0x10 + +#define CH_L 1 +#define CH_R 2 + +#define TASDEVICE_CRC8_POLYNOMIAL 0x4d + +#define MAX_NO_FILES 100 +#define MAX_CALIBRATION_DATA_SIZE 252 + +#define SMS_HTONL(a, b, c, d) ((((a)&0x000000FF)<<24) | \ + (((b)&0x000000FF)<<16) | \ + (((c)&0x000000FF)<<8) | \ + ((d)&0x000000FF)) + +enum channel { + top_left_Chn, + top_right_chn, + bottom_left_Chn, + bottom_right_chn, + max_chn, +}; + +struct tasdevice { + unsigned int dev_addr; + unsigned int err_code; + short cur_prog; + short cur_conf; + bool is_loading; + bool is_loaderr; + bool is_calibrated_data_loaded; + unsigned char cur_book; + struct tasdevice_fw *cali_data_fmw; +}; + +struct MYSWFTFile { + unsigned int m_vendor_id; + unsigned int m_file_id; + unsigned int m_version; + unsigned int m_Length; + unsigned int m_downloadAddr; + unsigned char *m_startAdress; +}; + +/* + * This item is used to store the generic i2c address of + * all the tas2781 devices for I2C broadcast during the multi-device + * writes, useless in mono case. + */ +struct global_addr { + unsigned char cur_book; + unsigned int dev_addr; + int ref_cnt; +}; + +struct calibration_data { + unsigned long total_sz; + unsigned char data[MAX_CALIBRATION_DATA_SIZE]; +}; + +struct tas_control { + struct snd_kcontrol_new *tasdevice_profile_controls; + int nr_controls; +}; + +struct tasdevice_irqinfo { + int irq_gpio; + int irq; +}; + +struct tasdevice_priv { + struct device *dev; + struct snd_soc_component *component; + struct sdw_slave *sdw_slave; + enum sdw_slave_status status; + struct sdw_bus_params params; + struct regmap *regmap; + struct mutex codec_lock; + struct mutex dev_lock; + int rst_gpio; + struct tasdevice tasdevice[max_chn]; + struct tasdevice_fw *fmw; + struct tasdevice_irqinfo irq_info; + struct tas_control tas_ctrl; + struct sdw_slave *sdw_peripheral; + struct global_addr glb_addr; + struct calibration_data cali_data; + struct tm tm; + int cur_prog; + int cur_conf; + unsigned int chip_id; + bool hw_init; + bool first_hw_init; + unsigned char *bq_params; + unsigned int magic_num; + unsigned char ndev; + unsigned char dev_name[32]; + unsigned char rca_binaryname[64]; + unsigned char coef_binaryname[64]; + unsigned char crc8_lkp_tbl[CRC8_TABLE_SIZE]; + + struct MYSWFTFile m_swftFile[MAX_NO_FILES]; + + void *codec; + unsigned int sysclk; + bool is_glb_calibrated_data_loaded; +}; + +struct sdw_stream_data { + struct sdw_stream_runtime *sdw_stream; +}; + +int tasdevice_dev_read(struct tasdevice_priv *tasdevice, + unsigned int *reg, unsigned int *value); + +#endif /*__TAS2783_H__ */ -- 2.34.1

4 4

[PATCH v3 0/6] ASoC: mediatek: Add support for MT7986 SoC
by Maso Huang 04 Aug '23

04 Aug '23

Changes in v3: - merge clk api to mt7986-afe-pcm.c, remove [2/7] in v2 - refine based on reviewer's suggestions [1/6] [2/6] - fix the comment format, move in clk api, and simplify based on reviewer's suggestions [3/6] - remove redundant prefix in dt-binding [6/6] Changes in v2: - v1 title: [PATCH 0/7] ASoC: mediatek: Add support for MT79xx SoC - add missing maintainers - rename mt79xx to mt7986 in all files - use clk bulk api in mt7986-afe-clk.c [2/7] - refine mt79xx-afe-pcm.c based on reviewer's suggestions [4/7] - refine dt-binding files based on reviewer's suggestions [6/7] [7/7] - transpose [3/7] and [4/7] in v1 to fix test build errors Maso Huang (6): ASoC: mediatek: mt7986: add common header ASoC: mediatek: mt7986: support etdm in platform driver ASoC: mediatek: mt7986: add platform driver ASoC: mediatek: mt7986: add machine driver with wm8960 ASoC: dt-bindings: mediatek,mt7986-wm8960: add mt7986-wm8960 document ASoC: dt-bindings: mediatek,mt7986-afe: add audio afe document .../bindings/sound/mediatek,mt7986-afe.yaml | 89 +++ .../sound/mediatek,mt7986-wm8960.yaml | 53 ++ sound/soc/mediatek/Kconfig | 20 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/mt7986/Makefile | 9 + sound/soc/mediatek/mt7986/mt7986-afe-common.h | 49 ++ sound/soc/mediatek/mt7986/mt7986-afe-pcm.c | 622 ++++++++++++++++++ sound/soc/mediatek/mt7986/mt7986-dai-etdm.c | 420 ++++++++++++ sound/soc/mediatek/mt7986/mt7986-reg.h | 206 ++++++ sound/soc/mediatek/mt7986/mt7986-wm8960.c | 184 ++++++ 10 files changed, 1653 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mediatek,mt7986-afe.yaml create mode 100644 Documentation/devicetree/bindings/sound/mediatek,mt7986-wm8960.yaml create mode 100644 sound/soc/mediatek/mt7986/Makefile create mode 100644 sound/soc/mediatek/mt7986/mt7986-afe-common.h create mode 100644 sound/soc/mediatek/mt7986/mt7986-afe-pcm.c create mode 100644 sound/soc/mediatek/mt7986/mt7986-dai-etdm.c create mode 100644 sound/soc/mediatek/mt7986/mt7986-reg.h create mode 100644 sound/soc/mediatek/mt7986/mt7986-wm8960.c -- 2.18.0

5 16

Re: [PATCH v3 6/6] ASoC: dt-bindings: mediatek,mt7986-afe: add audio afe document
by Krzysztof Kozlowski 04 Aug '23

04 Aug '23

On 01/08/2023 10:25, Maso Huang (黃加竹) wrote: > On Fri, 2023-07-28 at 14:51 +0200, Krzysztof Kozlowski wrote: >> >> External email : Please do not click links or open attachments until >> you have verified the sender or the content. >> On 28/07/2023 11:08, Maso Huang wrote: >>> Add mt7986 audio afe document. >>> >>> Signed-off-by: Maso Huang <maso.huang(a)mediatek.com> >> >> Thank you for your patch. There is something to discuss/improve. >> >>> + >>> +required: >>> + - compatible >>> + - reg >>> + - interrupts >>> + - clocks >>> + - clock-names >>> + - assigned-clocks >>> + - assigned-clock-parents >> >> You should constrain your clocks per variants. I doubt that they are >> really so flexible/optional on each SoC... or maybe missing clocks >> are >> result of unimplemented parts in the driver? But then this should not >> really affect bindings. Bindings still should require such clocks. >> Your >> DTS can always provide a <0>, if needed. >> >> > > Hi Krzysztof, > > After internal check, assigned-clocks and assigned-clock-parents are > not required on this SoC. > Maybe we can just drop these two options? It's separate issue, but yes - why requiring them? I wrote about missing constraints for your clocks in the bindings. Best regards, Krzysztof

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[PATCH -next] ASoC: q6dsp: Make g_apm staic
by GUO Zihua 04 Aug '23

04 Aug '23

g_apm is not referred elsewhere, so make it static. This resolves sparse warning: warning: symbol 'g_apm' was not declared. Should it be static? Signed-off-by: GUO Zihua <guozihua(a)huawei.com> --- sound/soc/qcom/qdsp6/q6apm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6apm.c b/sound/soc/qcom/qdsp6/q6apm.c index 5d44d07acd69..2a2a5bd98110 100644 --- a/sound/soc/qcom/qdsp6/q6apm.c +++ b/sound/soc/qcom/qdsp6/q6apm.c @@ -27,7 +27,7 @@ struct apm_graph_mgmt_cmd { #define APM_GRAPH_MGMT_PSIZE(p, n) ALIGN(struct_size(p, sub_graph_id_list, n), 8) -struct q6apm *g_apm; +static struct q6apm *g_apm; int q6apm_send_cmd_sync(struct q6apm *apm, struct gpr_pkt *pkt, uint32_t rsp_opcode) { -- 2.17.1

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[RFC] aplay: Add option for specifying subformat
by GitHub pull_request - opened 04 Aug '23

04 Aug '23

alsa-project/alsa-utils pull request #228 was opened from crojewsk-intel: Make subformat first-class citizen by allowing users to specify it just like it is the case for the format. Default to SND_PCM_SUBFORMAT_STD so they are no surprises. Request URL : https://github.com/alsa-project/alsa-utils/pull/228 Patch URL : https://github.com/alsa-project/alsa-utils/pull/228.patch Repository URL: https://github.com/alsa-project/alsa-utils

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[RFC] pcm: Add MSBITS_20 and MSBITS_24 subformat options
by GitHub pull_request - opened 04 Aug '23

04 Aug '23

alsa-project/alsa-lib pull request #342 was opened from crojewsk-intel: Improve granularity of format selection for S32/U32 formats by adding masks representing 20 and 24 most significant bits. As a preparation step introduce `snd_pcm_subformat_value()` for easy name-to-subformat conversions. While at it, fix few warnings that pop up in subformat-related code during compilation. Request URL : https://github.com/alsa-project/alsa-lib/pull/342 Patch URL : https://github.com/alsa-project/alsa-lib/pull/342.patch Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH] ASoC: soc-acpi: Add missing kernel doc
by Charles Keepax 03 Aug '23

03 Aug '23

The UID field in snd_soc_acpi_link_adr is not documented, add kernel doc for it. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- include/sound/soc-acpi.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/sound/soc-acpi.h b/include/sound/soc-acpi.h index 630bf7367fe69..6d31d535e8f6d 100644 --- a/include/sound/soc-acpi.h +++ b/include/sound/soc-acpi.h @@ -151,6 +151,7 @@ struct snd_soc_acpi_link_adr { * all firmware/topology related fields. * * @id: ACPI ID (usually the codec's) used to find a matching machine driver. + * @uid: ACPI Unique ID, can be used to disambiguate matches. * @comp_ids: list of compatible audio codecs using the same machine driver, * firmware and topology * @link_mask: describes required board layout, e.g. for SoundWire. -- 2.30.2

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[PATCH v2 RESEND*3] ASoC: fsl MPC52xx drivers require PPC_BESTCOMM
by Randy Dunlap 03 Aug '23

03 Aug '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> Acked-by: Shengjiu Wang <shengjiu.wang(a)gmail.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

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[PATCH] ASoC: fsl: micfil: Use dual license micfil code
by Daniel Baluta 03 Aug '23

03 Aug '23

From: Daniel Baluta <daniel.baluta(a)nxp.com> We need this in order to easily reuse register definitions and some functions with Sound Open Firmware driver. According to Documentation/process/license-rules.rst: "Dual BSD/GPL" The module is dual licensed under a GPL v2 variant or BSD license choice. The exact variant of the BSD license can only be determined via the license information in the corresponding source files. so use "Dual BSD/GPL" for license string. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/fsl/fsl_micfil.c | 4 ++-- sound/soc/fsl/fsl_micfil.h | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index fe28b27e50d0..97cf315781ab 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -1,4 +1,4 @@ -// SPDX-License-Identifier: GPL-2.0 +// SPDX-License-Identifier: GPL-2.0 OR BSD-3-Clause // Copyright 2018 NXP #include <linux/bitfield.h> @@ -1254,4 +1254,4 @@ module_platform_driver(fsl_micfil_driver); MODULE_AUTHOR("Cosmin-Gabriel Samoila <cosmin.samoila(a)nxp.com>"); MODULE_DESCRIPTION("NXP PDM Microphone Interface (MICFIL) driver"); -MODULE_LICENSE("GPL v2"); +MODULE_LICENSE("Dual BSD/GPL"); diff --git a/sound/soc/fsl/fsl_micfil.h b/sound/soc/fsl/fsl_micfil.h index 9237a1c4cb8f..fee9fe3d9119 100644 --- a/sound/soc/fsl/fsl_micfil.h +++ b/sound/soc/fsl/fsl_micfil.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 OR BSD-3-Clause */ /* * PDM Microphone Interface for the NXP i.MX SoC * Copyright 2018 NXP -- 2.25.1

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[PATCH] ASoC: mediatek: mt8186: Remove unused mutex.
by Sebastian Andrzej Siewior 03 Aug '23

03 Aug '23

The mutex mutex_request_dram has no user. Remove mutex_request_dram. Signed-off-by: Sebastian Andrzej Siewior <bigeasy(a)linutronix.de> --- sound/soc/mediatek/mt8186/mt8186-afe-clk.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/mediatek/mt8186/mt8186-afe-clk.c b/sound/soc/mediatek/mt8186/mt8186-afe-clk.c index 539e3a023bc4e..70ec101890d36 100644 --- a/sound/soc/mediatek/mt8186/mt8186-afe-clk.c +++ b/sound/soc/mediatek/mt8186/mt8186-afe-clk.c @@ -13,8 +13,6 @@ #include "mt8186-afe-clk.h" #include "mt8186-audsys-clk.h" -static DEFINE_MUTEX(mutex_request_dram); - static const char *aud_clks[CLK_NUM] = { [CLK_AFE] = "aud_afe_clk", [CLK_DAC] = "aud_dac_clk", -- 2.40.1

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[PATCH v2 0/3] Add PWM-DAC audio support for StarFive JH7110 RISC-V SoC
by Hal Feng 03 Aug '23

03 Aug '23

This patchset adds PWM-DAC audio support for the StarFive JH7110 SoC. The PWM-DAC module does not require a hardware codec, but a dummy codec is needed for the driver. The dummy spdif codec driver, which is already upstream, is compatible with the one which JH7110 PWM-DAC needed. So we use it as the dummy codec driver for the JH7110 PWM-DAC module. The third patch depends on tag next-20230728 in linux-next branch. Changes since v1: - Rebase on tag v6.5-rc3. - Drop patch 1 and 2. - Drop the unneeded space and line in patch 3. - Use the dummy spdif codec driver instead of adding a new one. - Change "dai_link->stop_dma_first = 1" to "dai_link->trigger_stop = SND_SOC_TRIGGER_ORDER_LDC" in patch 4. - Drop the unneeded "status = "okay;" in patch 5. - Change some node names in patch 5. --- v1: https://lore.kernel.org/all/20230626110909.38718-1-hal.feng@starfivetech.co… Hal Feng (3): ASoC: dt-bindings: Add StarFive JH7110 PWM-DAC controller ASoC: starfive: Add JH7110 PWM-DAC driver riscv: dts: starfive: Add JH7110 PWM-DAC support .../sound/starfive,jh7110-pwmdac.yaml | 76 ++ MAINTAINERS | 7 + .../jh7110-starfive-visionfive-2.dtsi | 99 +++ arch/riscv/boot/dts/starfive/jh7110.dtsi | 156 ++++ sound/soc/starfive/Kconfig | 9 + sound/soc/starfive/Makefile | 1 + sound/soc/starfive/jh7110_pwmdac.c | 787 ++++++++++++++++++ 7 files changed, 1135 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/starfive,jh7110-pwmdac.yaml create mode 100644 sound/soc/starfive/jh7110_pwmdac.c base-commit: 6eaae198076080886b9e7d57f4ae06fa782f90ef -- 2.38.1

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