From 3526860f26febbe46960f9b37f5dbd5ccc109ea8 Mon Sep 17 00:00:00 2001 From: Rui Salvaterra Date: Mon, 22 Jan 2024 11:45:12 +0000 Subject: [PATCH 01/21] ALSA: hda: Replace numeric device IDs with constant values MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit We have self-explanatory constants for Intel HDA devices, let's use them instead of magic numbers and code comments. Signed-off-by: Rui Salvaterra Reviewed-by: Amadeusz Sławiński Link: https://lore.kernel.org/r/20240122114512.55808-2-rsalvaterra@gmail.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 2276adc84478..66f013ee160d 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1729,8 +1729,8 @@ static int default_bdl_pos_adj(struct azx *chip) /* some exceptions: Atoms seem problematic with value 1 */ if (chip->pci->vendor == PCI_VENDOR_ID_INTEL) { switch (chip->pci->device) { - case 0x0f04: /* Baytrail */ - case 0x2284: /* Braswell */ + case PCI_DEVICE_ID_INTEL_HDA_BYT: + case PCI_DEVICE_ID_INTEL_HDA_BSW: return 32; } } From 56beedc88405fd8022edfd1c2e63d1bc6c95efcb Mon Sep 17 00:00:00 2001 From: Rui Salvaterra Date: Mon, 22 Jan 2024 11:45:13 +0000 Subject: [PATCH 02/21] ALSA: hda: Increase default bdl_pos_adj for Apollo Lake MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Apollo Lake seems to also suffer from IRQ timing issues. After being up for ~4 minutes, a Pentium N4200 system ends up falling back to workqueue-based IRQ handling: [ 208.019906] snd_hda_intel 0000:00:0e.0: IRQ timing workaround is activated for card #0. Suggest a bigger bdl_pos_adj. Unfortunately, the Baytrail and Braswell workaround value of 32 samples isn't enough to fix the issue here. Default to 64 samples. Signed-off-by: Rui Salvaterra Reviewed-by: Amadeusz Sławiński Link: https://lore.kernel.org/r/20240122114512.55808-3-rsalvaterra@gmail.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 66f013ee160d..1b550c42db09 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1732,6 +1732,8 @@ static int default_bdl_pos_adj(struct azx *chip) case PCI_DEVICE_ID_INTEL_HDA_BYT: case PCI_DEVICE_ID_INTEL_HDA_BSW: return 32; + case PCI_DEVICE_ID_INTEL_HDA_APL: + return 64; } } From a2ed0a44d637ef9deca595054c206da7d6cbdcbc Mon Sep 17 00:00:00 2001 From: Vitaly Rodionov Date: Mon, 22 Jan 2024 18:47:10 +0000 Subject: [PATCH 03/21] ALSA: hda/cs8409: Suppress vmaster control for Dolphin models Customer has reported an issue with specific desktop platform where two CS42L42 codecs are connected to CS8409 HDA bridge. If "Master Volume Control" is created then on Ubuntu OS UCM left/right balance slider in UI audio settings has no effect. This patch will fix this issue for a target paltform. Fixes: 20e507724113 ("ALSA: hda/cs8409: Add support for dolphin") Signed-off-by: Vitaly Rodionov Cc: Link: https://lore.kernel.org/r/20240122184710.5802-1-vitalyr@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cs8409.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_cs8409.c b/sound/pci/hda/patch_cs8409.c index 627899959ffe..e41316e2e983 100644 --- a/sound/pci/hda/patch_cs8409.c +++ b/sound/pci/hda/patch_cs8409.c @@ -1371,6 +1371,7 @@ void dolphin_fixups(struct hda_codec *codec, const struct hda_fixup *fix, int ac spec->scodecs[CS8409_CODEC1] = &dolphin_cs42l42_1; spec->scodecs[CS8409_CODEC1]->codec = codec; spec->num_scodecs = 2; + spec->gen.suppress_vmaster = 1; codec->patch_ops = cs8409_dolphin_patch_ops; From a969210066054ea109d8b7aff29a9b1c98776841 Mon Sep 17 00:00:00 2001 From: Julian Sikorski Date: Tue, 23 Jan 2024 09:49:35 +0100 Subject: [PATCH 04/21] ALSA: usb-audio: Add a quirk for Yamaha YIT-W12TX transmitter The device fails to initialize otherwise, giving the following error: [ 3676.671641] usb 2-1.1: 1:1: cannot get freq at ep 0x1 Signed-off-by: Julian Sikorski Cc: Link: https://lore.kernel.org/r/20240123084935.2745-1-belegdol+github@gmail.com Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 07cc6a201579..43c14a81a1e2 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2031,6 +2031,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_CTL_MSG_DELAY_1M | QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x0499, 0x1509, /* Steinberg UR22 */ QUIRK_FLAG_GENERIC_IMPLICIT_FB), + DEVICE_FLG(0x0499, 0x3108, /* Yamaha YIT-W12TX */ + QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x04d8, 0xfeea, /* Benchmark DAC1 Pre */ QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x04e8, 0xa051, /* Samsung USBC Headset (AKG) */ From 67794f882adca00d043899ac248bc002751da9f6 Mon Sep 17 00:00:00 2001 From: Alexander Tsoy Date: Tue, 23 Jan 2024 16:46:35 +0300 Subject: [PATCH 05/21] ALSA: usb-audio: Skip setting clock selector for single connections Since commit 086b957cc17f5 ("ALSA: usb-audio: Skip the clock selector inquiry for single connections") we are already skipping clock selector inquiry if only one clock source is connected, but we are still sending a set request. Lets skip that too. This should fix errors when setting a sample rate on devices that don't have any controls present within the clock selector. An example of such device is the new revision of MOTU M Series (07fd:000b): AudioControl Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 11 (CLOCK_SELECTOR) bClockID 1 bNrInPins 1 baCSourceID(0) 2 bmControls 0x00 iClockSelector 0 Perhaps we also should check if clock selectors are readable and writeable like we already do for clock sources, but this is out of scope of this patch. Link: https://bugzilla.kernel.org/show_bug.cgi?id=217601 Signed-off-by: Alexander Tsoy Link: https://lore.kernel.org/r/20240123134635.54026-1-alexander@tsoy.me Signed-off-by: Takashi Iwai --- sound/usb/clock.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 33db334e6556..94e4aaeafe58 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -325,7 +325,8 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, visited, validate); if (ret > 0) { /* Skip setting clock selector again for some devices */ - if (chip->quirk_flags & QUIRK_FLAG_SKIP_CLOCK_SELECTOR) + if (chip->quirk_flags & QUIRK_FLAG_SKIP_CLOCK_SELECTOR || + pins == 1) return ret; err = uac_clock_selector_set_val(chip, entity_id, cur); if (err < 0) From fcfc9f711d1e2fc7876ac12b1b16c509404b9625 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Wed, 24 Jan 2024 14:21:47 +0800 Subject: [PATCH 06/21] ALSA: hda/realtek - Add speaker pin verbtable for Dell dual speaker platform SSID 0x0c0d platform. It can't mute speaker when HP plugged. This patch add quirk to fill speaker pin verbtable. And disable speaker passthrough. Signed-off-by: Kailang Yang Cc: Link: https://lore.kernel.org/r/38b82976a875451d833d514cee34ff6a@realtek.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 19f2eb26659d..bfefe9ccbc6b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -439,6 +439,10 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) alc_update_coef_idx(codec, 0x67, 0xf000, 0x3000); fallthrough; case 0x10ec0215: + case 0x10ec0285: + case 0x10ec0289: + alc_update_coef_idx(codec, 0x36, 1<<13, 0); + fallthrough; case 0x10ec0230: case 0x10ec0233: case 0x10ec0235: @@ -452,9 +456,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0283: case 0x10ec0286: case 0x10ec0288: - case 0x10ec0285: case 0x10ec0298: - case 0x10ec0289: case 0x10ec0300: alc_update_coef_idx(codec, 0x10, 1<<9, 0); break; @@ -9732,6 +9734,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0b71, "Dell Inspiron 16 Plus 7620", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS), SND_PCI_QUIRK(0x1028, 0x0beb, "Dell XPS 15 9530 (2023)", ALC289_FIXUP_DELL_CS35L41_SPI_2), SND_PCI_QUIRK(0x1028, 0x0c03, "Dell Precision 5340", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0c0d, "Dell Oasis", ALC289_FIXUP_RTK_AMP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x0c19, "Dell Precision 3340", ALC236_FIXUP_DELL_DUAL_CODECS), SND_PCI_QUIRK(0x1028, 0x0c1a, "Dell Precision 3340", ALC236_FIXUP_DELL_DUAL_CODECS), SND_PCI_QUIRK(0x1028, 0x0c1b, "Dell Precision 3440", ALC236_FIXUP_DELL_DUAL_CODECS), From 7c4298534ce3c4b85099aba53442ec82ef17578b Mon Sep 17 00:00:00 2001 From: Jacob Siverskog Date: Wed, 24 Jan 2024 11:18:24 +0100 Subject: [PATCH 07/21] ALSA: usb-audio: fix typo fix typo in midi fallback log. Signed-off-by: Jacob Siverskog Fixes: ff49d1df79ae ("ALSA: usb-audio: USB MIDI 2.0 UMP support") Link: https://lore.kernel.org/r/20240124101827.35433-1-jacob@teenage.engineering Signed-off-by: Takashi Iwai --- sound/usb/midi2.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/midi2.c b/sound/usb/midi2.c index 1ec177fe284e..820d3e4b672a 100644 --- a/sound/usb/midi2.c +++ b/sound/usb/midi2.c @@ -1085,7 +1085,7 @@ int snd_usb_midi_v2_create(struct snd_usb_audio *chip, } if ((quirk && quirk->type != QUIRK_MIDI_STANDARD_INTERFACE) || iface->num_altsetting < 2) { - usb_audio_info(chip, "Quirk or no altest; falling back to MIDI 1.0\n"); + usb_audio_info(chip, "Quirk or no altset; falling back to MIDI 1.0\n"); goto fallback_to_midi1; } hostif = &iface->altsetting[1]; From d62ccb59afcd94e8d301433974672b156392289d Mon Sep 17 00:00:00 2001 From: Li RongQing Date: Wed, 24 Jan 2024 20:08:34 +0800 Subject: [PATCH 08/21] ALSA: virtio: remove duplicate check if queue is broken virtqueue_enable_cb() will call virtqueue_poll() which will check if queue is broken at beginning, so remove the virtqueue_is_broken() call Signed-off-by: Li RongQing Reviewed-by: Stefan Hajnoczi Link: https://lore.kernel.org/r/20240124120834.49410-1-lirongqing@baidu.com Signed-off-by: Takashi Iwai --- sound/virtio/virtio_card.c | 2 -- sound/virtio/virtio_ctl_msg.c | 2 -- sound/virtio/virtio_pcm_msg.c | 2 -- 3 files changed, 6 deletions(-) diff --git a/sound/virtio/virtio_card.c b/sound/virtio/virtio_card.c index e2847c040f75..b158c3cb8e5f 100644 --- a/sound/virtio/virtio_card.c +++ b/sound/virtio/virtio_card.c @@ -91,8 +91,6 @@ static void virtsnd_event_notify_cb(struct virtqueue *vqueue) virtsnd_event_dispatch(snd, event); virtsnd_event_send(vqueue, event, true, GFP_ATOMIC); } - if (unlikely(virtqueue_is_broken(vqueue))) - break; } while (!virtqueue_enable_cb(vqueue)); spin_unlock_irqrestore(&queue->lock, flags); } diff --git a/sound/virtio/virtio_ctl_msg.c b/sound/virtio/virtio_ctl_msg.c index 18dc5aca2e0c..9dabea01277f 100644 --- a/sound/virtio/virtio_ctl_msg.c +++ b/sound/virtio/virtio_ctl_msg.c @@ -303,8 +303,6 @@ void virtsnd_ctl_notify_cb(struct virtqueue *vqueue) virtqueue_disable_cb(vqueue); while ((msg = virtqueue_get_buf(vqueue, &length))) virtsnd_ctl_msg_complete(msg); - if (unlikely(virtqueue_is_broken(vqueue))) - break; } while (!virtqueue_enable_cb(vqueue)); spin_unlock_irqrestore(&queue->lock, flags); } diff --git a/sound/virtio/virtio_pcm_msg.c b/sound/virtio/virtio_pcm_msg.c index 542446c4c7ba..8c32efaf4c52 100644 --- a/sound/virtio/virtio_pcm_msg.c +++ b/sound/virtio/virtio_pcm_msg.c @@ -358,8 +358,6 @@ static inline void virtsnd_pcm_notify_cb(struct virtio_snd_queue *queue) virtqueue_disable_cb(queue->vqueue); while ((msg = virtqueue_get_buf(queue->vqueue, &written_bytes))) virtsnd_pcm_msg_complete(msg, written_bytes); - if (unlikely(virtqueue_is_broken(queue->vqueue))) - break; } while (!virtqueue_enable_cb(queue->vqueue)); spin_unlock_irqrestore(&queue->lock, flags); } From d915a6850e27efb383cd4400caadfe47792623df Mon Sep 17 00:00:00 2001 From: Alexander Tsoy Date: Wed, 24 Jan 2024 16:02:39 +0300 Subject: [PATCH 09/21] ALSA: usb-audio: Add delay quirk for MOTU M Series 2nd revision Audio control requests that sets sampling frequency sometimes fail on this card. Adding delay between control messages eliminates that problem. Link: https://bugzilla.kernel.org/show_bug.cgi?id=217601 Cc: Signed-off-by: Alexander Tsoy Link: https://lore.kernel.org/r/20240124130239.358298-1-alexander@tsoy.me Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 43c14a81a1e2..be0ce1fc4f26 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2075,6 +2075,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_GENERIC_IMPLICIT_FB), DEVICE_FLG(0x0763, 0x2031, /* M-Audio Fast Track C600 */ QUIRK_FLAG_GENERIC_IMPLICIT_FB), + DEVICE_FLG(0x07fd, 0x000b, /* MOTU M Series 2nd hardware revision */ + QUIRK_FLAG_CTL_MSG_DELAY_1M), DEVICE_FLG(0x08bb, 0x2702, /* LineX FM Transmitter */ QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x0951, 0x16ad, /* Kingston HyperX */ From 7822baa844a87cbb93308c1032c3d47d4079bb8a Mon Sep 17 00:00:00 2001 From: Sean Young Date: Wed, 24 Jan 2024 15:15:24 +0000 Subject: [PATCH 10/21] ALSA: usb-audio: add quirk for RODE NT-USB+ MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The RODE NT-USB+ is marketed as a professional usb microphone, however the usb audio interface is a mess: [ 1.130977] usb 1-5: new full-speed USB device number 2 using xhci_hcd [ 1.503906] usb 1-5: config 1 has an invalid interface number: 5 but max is 4 [ 1.503912] usb 1-5: config 1 has no interface number 4 [ 1.519689] usb 1-5: New USB device found, idVendor=19f7, idProduct=0035, bcdDevice= 1.09 [ 1.519695] usb 1-5: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [ 1.519697] usb 1-5: Product: RØDE NT-USB+ [ 1.519699] usb 1-5: Manufacturer: RØDE [ 1.519700] usb 1-5: SerialNumber: 1D773A1A [ 8.327495] usb 1-5: 1:1: cannot get freq at ep 0x82 [ 8.344500] usb 1-5: 1:2: cannot get freq at ep 0x82 [ 8.365499] usb 1-5: 2:1: cannot get freq at ep 0x2 Add QUIRK_FLAG_GET_SAMPLE_RATE to work around the broken sample rate get. I have asked Rode support to fix it, but they show no interest. Signed-off-by: Sean Young Cc: Link: https://lore.kernel.org/r/20240124151524.23314-1-sean@mess.org Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index be0ce1fc4f26..591c3a354696 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2183,6 +2183,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_FIXED_RATE), DEVICE_FLG(0x1bcf, 0x2283, /* NexiGo N930AF FHD Webcam */ QUIRK_FLAG_GET_SAMPLE_RATE), + DEVICE_FLG(0x19f7, 0x0035, /* RODE NT-USB+ */ + QUIRK_FLAG_GET_SAMPLE_RATE), /* Vendor matches */ VENDOR_FLG(0x045e, /* MS Lifecam */ From 668abe6dc7b61941fa5c724c06797efb0b87f070 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 24 Jan 2024 16:53:07 +0100 Subject: [PATCH 11/21] ALSA: usb-audio: Sort quirk table entries The quirk table entries should be put in the USB ID order, but some entries have been put in random places. Re-sort them. Fixes: bf990c102319 ("ALSA: usb-audio: add quirk to fix Hamedal C20 disconnect issue") Fixes: fd28941cff1c ("ALSA: usb-audio: Add new quirk FIXED_RATE for JBL Quantum810 Wireless") Fixes: dfd5fe19db7d ("ALSA: usb-audio: Add FIXED_RATE quirk for JBL Quantum610 Wireless") Fixes: 4a63e68a2951 ("ALSA: usb-audio: Fix microphone sound on Nexigo webcam.") Fixes: 7822baa844a8 ("ALSA: usb-audio: add quirk for RODE NT-USB+") Fixes: 4fb7c24f69c4 ("ALSA: usb-audio: Add quirk for Fiero SC-01") Fixes: 2307a0e1ca0b ("ALSA: usb-audio: Add quirk for Fiero SC-01 (fw v1.0.0)") Link: https://lore.kernel.org/r/20240124155307.16996-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 36 ++++++++++++++++++------------------ 1 file changed, 18 insertions(+), 18 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 591c3a354696..09712e61c606 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2037,6 +2037,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x04e8, 0xa051, /* Samsung USBC Headset (AKG) */ QUIRK_FLAG_SKIP_CLOCK_SELECTOR | QUIRK_FLAG_CTL_MSG_DELAY_5M), + DEVICE_FLG(0x0525, 0xa4ad, /* Hamedal C20 usb camero */ + QUIRK_FLAG_IFACE_SKIP_CLOSE), DEVICE_FLG(0x054c, 0x0b8c, /* Sony WALKMAN NW-A45 DAC */ QUIRK_FLAG_SET_IFACE_FIRST), DEVICE_FLG(0x0556, 0x0014, /* Phoenix Audio TMX320VC */ @@ -2083,8 +2085,14 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_CTL_MSG_DELAY_1M), DEVICE_FLG(0x0b0e, 0x0349, /* Jabra 550a */ QUIRK_FLAG_CTL_MSG_DELAY_1M), + DEVICE_FLG(0x0ecb, 0x205c, /* JBL Quantum610 Wireless */ + QUIRK_FLAG_FIXED_RATE), + DEVICE_FLG(0x0ecb, 0x2069, /* JBL Quantum810 Wireless */ + QUIRK_FLAG_FIXED_RATE), DEVICE_FLG(0x0fd9, 0x0008, /* Hauppauge HVR-950Q */ QUIRK_FLAG_SHARE_MEDIA_DEVICE | QUIRK_FLAG_ALIGN_TRANSFER), + DEVICE_FLG(0x1224, 0x2a25, /* Jieli Technology USB PHY 2.0 */ + QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x1395, 0x740a, /* Sennheiser DECT */ QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x1397, 0x0507, /* Behringer UMC202HD */ @@ -2117,6 +2125,10 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_ITF_USB_DSD_DAC | QUIRK_FLAG_CTL_MSG_DELAY), DEVICE_FLG(0x1901, 0x0191, /* GE B850V3 CP2114 audio interface */ QUIRK_FLAG_GET_SAMPLE_RATE), + DEVICE_FLG(0x19f7, 0x0035, /* RODE NT-USB+ */ + QUIRK_FLAG_GET_SAMPLE_RATE), + DEVICE_FLG(0x1bcf, 0x2283, /* NexiGo N930AF FHD Webcam */ + QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x2040, 0x7200, /* Hauppauge HVR-950Q */ QUIRK_FLAG_SHARE_MEDIA_DEVICE | QUIRK_FLAG_ALIGN_TRANSFER), DEVICE_FLG(0x2040, 0x7201, /* Hauppauge HVR-950Q-MXL */ @@ -2159,6 +2171,12 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x2912, 0x30c8, /* Audioengine D1 */ QUIRK_FLAG_GET_SAMPLE_RATE), + DEVICE_FLG(0x2b53, 0x0023, /* Fiero SC-01 (firmware v1.0.0 @ 48 kHz) */ + QUIRK_FLAG_GENERIC_IMPLICIT_FB), + DEVICE_FLG(0x2b53, 0x0024, /* Fiero SC-01 (firmware v1.0.0 @ 96 kHz) */ + QUIRK_FLAG_GENERIC_IMPLICIT_FB), + DEVICE_FLG(0x2b53, 0x0031, /* Fiero SC-01 (firmware v1.1.0) */ + QUIRK_FLAG_GENERIC_IMPLICIT_FB), DEVICE_FLG(0x30be, 0x0101, /* Schiit Hel */ QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x413c, 0xa506, /* Dell AE515 sound bar */ @@ -2167,24 +2185,6 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_ALIGN_TRANSFER), DEVICE_FLG(0x534d, 0x2109, /* MacroSilicon MS2109 */ QUIRK_FLAG_ALIGN_TRANSFER), - DEVICE_FLG(0x1224, 0x2a25, /* Jieli Technology USB PHY 2.0 */ - QUIRK_FLAG_GET_SAMPLE_RATE), - DEVICE_FLG(0x2b53, 0x0023, /* Fiero SC-01 (firmware v1.0.0 @ 48 kHz) */ - QUIRK_FLAG_GENERIC_IMPLICIT_FB), - DEVICE_FLG(0x2b53, 0x0024, /* Fiero SC-01 (firmware v1.0.0 @ 96 kHz) */ - QUIRK_FLAG_GENERIC_IMPLICIT_FB), - DEVICE_FLG(0x2b53, 0x0031, /* Fiero SC-01 (firmware v1.1.0) */ - QUIRK_FLAG_GENERIC_IMPLICIT_FB), - DEVICE_FLG(0x0525, 0xa4ad, /* Hamedal C20 usb camero */ - QUIRK_FLAG_IFACE_SKIP_CLOSE), - DEVICE_FLG(0x0ecb, 0x205c, /* JBL Quantum610 Wireless */ - QUIRK_FLAG_FIXED_RATE), - DEVICE_FLG(0x0ecb, 0x2069, /* JBL Quantum810 Wireless */ - QUIRK_FLAG_FIXED_RATE), - DEVICE_FLG(0x1bcf, 0x2283, /* NexiGo N930AF FHD Webcam */ - QUIRK_FLAG_GET_SAMPLE_RATE), - DEVICE_FLG(0x19f7, 0x0035, /* RODE NT-USB+ */ - QUIRK_FLAG_GET_SAMPLE_RATE), /* Vendor matches */ VENDOR_FLG(0x045e, /* MS Lifecam */ From eeca59a6e8e610e593d17e12f67324c615ec5ef2 Mon Sep 17 00:00:00 2001 From: Alexander Tsoy Date: Thu, 25 Jan 2024 23:54:57 +0300 Subject: [PATCH 12/21] ALSA: usb-audio: Support read-only clock selector control Clock selector control might be read-only. Add corresponding checks to prevent sending control requests that would fail. Signed-off-by: Alexander Tsoy Link: https://lore.kernel.org/r/20240125205457.28258-1-alexander@tsoy.me Signed-off-by: Takashi Iwai --- sound/usb/clock.c | 23 ++++++++++++++++++++++- 1 file changed, 22 insertions(+), 1 deletion(-) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 94e4aaeafe58..7b259641adb5 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -261,6 +261,8 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, int ret, i, cur, err, pins, clock_id; const u8 *sources; int proto = fmt->protocol; + bool readable, writeable; + u32 bmControls; entity_id &= 0xff; @@ -292,11 +294,27 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, sources = GET_VAL(selector, proto, baCSourceID); cur = 0; + if (proto == UAC_VERSION_3) + bmControls = le32_to_cpu(*(__le32 *)(&selector->v3.baCSourceID[0] + pins)); + else + bmControls = *(__u8 *)(&selector->v2.baCSourceID[0] + pins); + + readable = uac_v2v3_control_is_readable(bmControls, + UAC2_CX_CLOCK_SELECTOR); + writeable = uac_v2v3_control_is_writeable(bmControls, + UAC2_CX_CLOCK_SELECTOR); + if (pins == 1) { ret = 1; goto find_source; } + /* for now just warn about buggy device */ + if (!readable) + usb_audio_warn(chip, + "%s(): clock selector control is not readable, id %d\n", + __func__, clock_id); + /* the entity ID we are looking at is a selector. * find out what it currently selects */ ret = uac_clock_selector_get_val(chip, clock_id); @@ -326,7 +344,7 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, if (ret > 0) { /* Skip setting clock selector again for some devices */ if (chip->quirk_flags & QUIRK_FLAG_SKIP_CLOCK_SELECTOR || - pins == 1) + pins == 1 || !writeable) return ret; err = uac_clock_selector_set_val(chip, entity_id, cur); if (err < 0) @@ -337,6 +355,9 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, return ret; find_others: + if (!writeable) + return -ENXIO; + /* The current clock source is invalid, try others. */ for (i = 1; i <= pins; i++) { if (i == cur) From efb56d84dd9c3de3c99fc396abb57c6d330038b5 Mon Sep 17 00:00:00 2001 From: David Senoner Date: Fri, 26 Jan 2024 16:56:26 +0100 Subject: [PATCH 13/21] ALSA: hda/realtek: Fix the external mic not being recognised for Acer Swift 1 SF114-32 If you connect an external headset/microphone to the 3.5mm jack on the Acer Swift 1 SF114-32 it does not recognize the microphone. This fixes that and gives the user the ability to choose between internal and headset mic. Signed-off-by: David Senoner Cc: Link: https://lore.kernel.org/r/20240126155626.2304465-1-seda18@rolmail.net Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bfefe9ccbc6b..ef51e51fb6a5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9655,6 +9655,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x1247, "Acer vCopperbox", ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS), SND_PCI_QUIRK(0x1025, 0x1248, "Acer Veriton N4660G", ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1269, "Acer SWIFT SF314-54", ALC256_FIXUP_ACER_HEADSET_MIC), + SND_PCI_QUIRK(0x1025, 0x126a, "Acer Swift SF114-32", ALC256_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x128f, "Acer Veriton Z6860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1290, "Acer Veriton Z4860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), From f0d78972f27dc1d1d51fbace2713ad3cdc60a877 Mon Sep 17 00:00:00 2001 From: Luka Guzenko Date: Sun, 28 Jan 2024 16:57:04 +0100 Subject: [PATCH 14/21] ALSA: hda/realtek: Enable Mute LED on HP Laptop 14-fq0xxx This HP Laptop uses ALC236 codec with COEF 0x07 controlling the mute LED. Enable existing quirk for this device. Signed-off-by: Luka Guzenko Cc: Link: https://lore.kernel.org/r/20240128155704.2333812-1-l.guzenko@web.de Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ef51e51fb6a5..bfa244028b99 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9856,6 +9856,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8786, "HP OMEN 15", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8787, "HP OMEN 15", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8788, "HP OMEN 15", ALC285_FIXUP_HP_MUTE_LED), + SND_PCI_QUIRK(0x103c, 0x87b7, "HP Laptop 14-fq0xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), SND_PCI_QUIRK(0x103c, 0x87c8, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87e5, "HP ProBook 440 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87e7, "HP ProBook 450 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), From c0787fcff88bce36d21e5b726bdeab694baba6a5 Mon Sep 17 00:00:00 2001 From: Alexander Tsoy Date: Sun, 28 Jan 2024 16:23:38 +0300 Subject: [PATCH 15/21] Revert "ALSA: usb-audio: Skip setting clock selector for single connections" This reverts commit 67794f882adca00d043899ac248bc002751da9f6. We need to explicitly set up the clock selector to workaround a problem with the Behringer mixers. This was originally done in d2e8f641257d ("ALSA: usb-audio: Explicitly set up the clock selector") The problem with MOTU M Series mentioned in commit message was fixed in a different way by checking control capabilities of clock selectors. Signed-off-by: Alexander Tsoy Link: https://lore.kernel.org/r/20240128132338.819273-1-alexander@tsoy.me Signed-off-by: Takashi Iwai --- sound/usb/clock.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 7b259641adb5..a8204c6d6fac 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -344,7 +344,7 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, if (ret > 0) { /* Skip setting clock selector again for some devices */ if (chip->quirk_flags & QUIRK_FLAG_SKIP_CLOCK_SELECTOR || - pins == 1 || !writeable) + !writeable) return ret; err = uac_clock_selector_set_val(chip, entity_id, cur); if (err < 0) From 346f59d1e8ed0eed41c80e1acb657e484c308e6a Mon Sep 17 00:00:00 2001 From: Alexander Tsoy Date: Mon, 29 Jan 2024 15:12:54 +0300 Subject: [PATCH 16/21] ALSA: usb-audio: Check presence of valid altsetting control Many devices with a single alternate setting do not have a Valid Alternate Setting Control and validation performed by validate_sample_rate_table_v2v3() doesn't work on them and is not really needed. So check the presense of control before sending altsetting validation requests. MOTU Microbook IIc is suffering the most without this check. It takes up to 40 seconds to bootup due to how slow it switches sampling rates: [ 2659.164824] usb 3-2: New USB device found, idVendor=07fd, idProduct=0004, bcdDevice= 0.60 [ 2659.164827] usb 3-2: New USB device strings: Mfr=1, Product=2, SerialNumber=0 [ 2659.164829] usb 3-2: Product: MicroBook IIc [ 2659.164830] usb 3-2: Manufacturer: MOTU [ 2659.166204] usb 3-2: Found last interface = 3 [ 2679.322298] usb 3-2: No valid sample rate available for 1:1, assuming a firmware bug [ 2679.322306] usb 3-2: 1:1: add audio endpoint 0x3 [ 2679.322321] usb 3-2: Creating new data endpoint #3 [ 2679.322552] usb 3-2: 1:1 Set sample rate 96000, clock 1 [ 2684.362250] usb 3-2: 2:1: cannot get freq (v2/v3): err -110 [ 2694.444700] usb 3-2: No valid sample rate available for 2:1, assuming a firmware bug [ 2694.444707] usb 3-2: 2:1: add audio endpoint 0x84 [ 2694.444721] usb 3-2: Creating new data endpoint #84 [ 2699.482103] usb 3-2: 2:1 Set sample rate 96000, clock 1 Signed-off-by: Alexander Tsoy Link: https://lore.kernel.org/r/20240129121254.3454481-1-alexander@tsoy.me Signed-off-by: Takashi Iwai --- sound/usb/format.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/usb/format.c b/sound/usb/format.c index ab5fed9f55b6..3b45d0ee7693 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -470,9 +470,11 @@ static int validate_sample_rate_table_v2v3(struct snd_usb_audio *chip, int clock) { struct usb_device *dev = chip->dev; + struct usb_host_interface *alts; unsigned int *table; unsigned int nr_rates; int i, err; + u32 bmControls; /* performing the rate verification may lead to unexpected USB bus * behavior afterwards by some unknown reason. Do this only for the @@ -481,6 +483,24 @@ static int validate_sample_rate_table_v2v3(struct snd_usb_audio *chip, if (!(chip->quirk_flags & QUIRK_FLAG_VALIDATE_RATES)) return 0; /* don't perform the validation as default */ + alts = snd_usb_get_host_interface(chip, fp->iface, fp->altsetting); + if (!alts) + return 0; + + if (fp->protocol == UAC_VERSION_3) { + struct uac3_as_header_descriptor *as = snd_usb_find_csint_desc( + alts->extra, alts->extralen, NULL, UAC_AS_GENERAL); + bmControls = le32_to_cpu(as->bmControls); + } else { + struct uac2_as_header_descriptor *as = snd_usb_find_csint_desc( + alts->extra, alts->extralen, NULL, UAC_AS_GENERAL); + bmControls = as->bmControls; + } + + if (!uac_v2v3_control_is_readable(bmControls, + UAC2_AS_VAL_ALT_SETTINGS)) + return 0; + table = kcalloc(fp->nr_rates, sizeof(*table), GFP_KERNEL); if (!table) return -ENOMEM; From c16dfab33f99fc3ff43d48253bc2784ccb84c1de Mon Sep 17 00:00:00 2001 From: Kenzo Gomez Date: Sat, 27 Jan 2024 17:46:21 +0100 Subject: [PATCH 17/21] ALSA: hda: cs35l41: Support additional ASUS Zenbook UX3402VA Add new model entry into configuration table. Signed-off-by: Kenzo Gomez Link: https://lore.kernel.org/r/20240127164621.26431-1-kenzo.sgomez@gmail.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda_property.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index 35277ce890a4..59504852adc6 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -76,6 +76,7 @@ static const struct cs35l41_config cs35l41_config_table[] = { { "10431533", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, { "10431573", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, { "10431663", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 1000, 4500, 24 }, + { "104316A3", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, { "104316D3", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, { "104316F3", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, { "104317F3", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, @@ -410,6 +411,7 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CSC3551", "10431533", generic_dsd_config }, { "CSC3551", "10431573", generic_dsd_config }, { "CSC3551", "10431663", generic_dsd_config }, + { "CSC3551", "104316A3", generic_dsd_config }, { "CSC3551", "104316D3", generic_dsd_config }, { "CSC3551", "104316F3", generic_dsd_config }, { "CSC3551", "104317F3", generic_dsd_config }, From be220d2e5544ff094142d263db5cf94d034b5e39 Mon Sep 17 00:00:00 2001 From: Chhayly Leang Date: Fri, 26 Jan 2024 15:09:12 +0700 Subject: [PATCH 18/21] ALSA: hda: cs35l41: Support ASUS Zenbook UM3402YAR Adds sound support for ASUS Zenbook UM3402YAR with missing DSD Signed-off-by: Chhayly Leang Link: https://lore.kernel.org/r/20240126080912.87422-1-clw.leang@gmail.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda_property.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index 59504852adc6..d74cf11eef1e 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -76,6 +76,7 @@ static const struct cs35l41_config cs35l41_config_table[] = { { "10431533", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, { "10431573", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, { "10431663", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 1000, 4500, 24 }, + { "10431683", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, { "104316A3", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, { "104316D3", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, { "104316F3", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, @@ -411,6 +412,7 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CSC3551", "10431533", generic_dsd_config }, { "CSC3551", "10431573", generic_dsd_config }, { "CSC3551", "10431663", generic_dsd_config }, + { "CSC3551", "10431683", generic_dsd_config }, { "CSC3551", "104316A3", generic_dsd_config }, { "CSC3551", "104316D3", generic_dsd_config }, { "CSC3551", "104316F3", generic_dsd_config }, From f7c4cb4a3f77867612b45c6327f80eac58a8ce65 Mon Sep 17 00:00:00 2001 From: Ivan Orlov Date: Thu, 25 Jan 2024 22:35:19 +0000 Subject: [PATCH 19/21] ALSA: pcm: Add missing formats to formats list Add 4 missing formats to 'snd_pcm_format_names' array in order to be able to get their names with 'snd_pcm_format_name' function. Signed-off-by: Ivan Orlov Link: https://lore.kernel.org/r/20240125223522.1122765-1-ivan.orlov0322@gmail.com Signed-off-by: Takashi Iwai --- sound/core/pcm.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index a09f0154e6a7..d0788126cbab 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -211,6 +211,10 @@ static const char * const snd_pcm_format_names[] = { FORMAT(DSD_U32_LE), FORMAT(DSD_U16_BE), FORMAT(DSD_U32_BE), + FORMAT(S20_LE), + FORMAT(S20_BE), + FORMAT(U20_LE), + FORMAT(U20_BE), }; /** From d1eb913c8df4574df2861db03d2411023bd7930a Mon Sep 17 00:00:00 2001 From: Ivan Orlov Date: Thu, 25 Jan 2024 22:35:20 +0000 Subject: [PATCH 20/21] ALSA: pcm: Fix snd_pcm_format_name function Fix snd_pcm_format_name so it won't return NULL-pointer in case if it can't find the format in the 'snd_pcm_format_names' list. Return "Unknown" instead, as it is done if the number passed to the function is larger than a list size. Signed-off-by: Ivan Orlov Link: https://lore.kernel.org/r/20240125223522.1122765-2-ivan.orlov0322@gmail.com Signed-off-by: Takashi Iwai --- sound/core/pcm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index d0788126cbab..d9b338088d10 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -225,9 +225,11 @@ static const char * const snd_pcm_format_names[] = { */ const char *snd_pcm_format_name(snd_pcm_format_t format) { - if ((__force unsigned int)format >= ARRAY_SIZE(snd_pcm_format_names)) + unsigned int format_num = (__force unsigned int)format; + + if (format_num >= ARRAY_SIZE(snd_pcm_format_names) || !snd_pcm_format_names[format_num]) return "Unknown"; - return snd_pcm_format_names[(__force unsigned int)format]; + return snd_pcm_format_names[format_num]; } EXPORT_SYMBOL_GPL(snd_pcm_format_name); From 3e39acf56ededdebd1033349a16b704839b94b28 Mon Sep 17 00:00:00 2001 From: Ivan Orlov Date: Thu, 25 Jan 2024 22:35:21 +0000 Subject: [PATCH 21/21] ALSA: core: Add sound core KUnit test At the moment, we have a decent amount of integration tests (selftests) covering different aspects of the sound subsystem. However, a lot of of sound-related in-kernel functions remains uncovered. This patch introduces the KUnit test for the core part of the sound subsystem. It includes 10 test cases: - Coverage of the format-related inline functions from 'pcm.h' header file: snd_pcm_format_physical_width, snd_pcm_format_width, snd_pcm_format_signed, test_format_endianness - Coverage of the available bytes counting functions from 'pcm.h' header: snd_pcm_capture_avail, snd_pcm_playback_avail - Coverage of functions from pcm_misc: snd_pcm_format_set_silence, snd_pcm_format_name - Coverage of card-related functions from init.c: snd_card_set_id, snd_component_add This patch depends on the previous patches in this patch series as they contain fix for the bug, which was found during the test development. Without them, the test doesn't pass. Signed-off-by: Ivan Orlov Link: https://lore.kernel.org/r/20240125223522.1122765-3-ivan.orlov0322@gmail.com Signed-off-by: Takashi Iwai --- MAINTAINERS | 6 + sound/core/Kconfig | 16 ++ sound/core/Makefile | 2 + sound/core/sound_kunit.c | 310 +++++++++++++++++++++++++++++++++++++++ 4 files changed, 334 insertions(+) create mode 100644 sound/core/sound_kunit.c diff --git a/MAINTAINERS b/MAINTAINERS index 8d1052fa6a69..dbc61d565760 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -20484,6 +20484,12 @@ F: include/uapi/sound/compress_* F: sound/core/compress_offload.c F: sound/soc/soc-compress.c +SOUND - CORE KUNIT TEST +M: Ivan Orlov +L: linux-sound@vger.kernel.org +S: Supported +F: sound/core/sound_kunit.c + SOUND - DMAENGINE HELPERS M: Lars-Peter Clausen S: Supported diff --git a/sound/core/Kconfig b/sound/core/Kconfig index e41818e59a15..664c6ee2b5a1 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -39,6 +39,22 @@ config SND_UMP_LEGACY_RAWMIDI legacy MIDI 1.0 byte streams is created for each UMP Endpoint. The device contains 16 substreams corresponding to UMP groups. +config SND_CORE_TEST + tristate "Sound core KUnit test" + depends on KUNIT + default KUNIT_ALL_TESTS + help + This options enables the sound core functions KUnit test. + + KUnit tests run during boot and output the results to the debug + log in TAP format (https://testanything.org/). Only useful for + kernel devs running KUnit test harness and are not for inclusion + into a production build. + + For more information on KUnit and unit tests in general, refer + to the KUnit documentation in Documentation/dev-tools/kunit/. + + config SND_COMPRESS_OFFLOAD tristate diff --git a/sound/core/Makefile b/sound/core/Makefile index a6b444ee2832..1d34e6950317 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -49,6 +49,8 @@ obj-$(CONFIG_SND_SEQ_DEVICE) += snd-seq-device.o obj-$(CONFIG_SND_RAWMIDI) += snd-rawmidi.o obj-$(CONFIG_SND_UMP) += snd-ump.o +obj-$(CONFIG_SND_CORE_TEST) += sound_kunit.o + obj-$(CONFIG_SND_OSSEMUL) += oss/ obj-$(CONFIG_SND_SEQUENCER) += seq/ diff --git a/sound/core/sound_kunit.c b/sound/core/sound_kunit.c new file mode 100644 index 000000000000..5d5a7bf88de4 --- /dev/null +++ b/sound/core/sound_kunit.c @@ -0,0 +1,310 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +/* + * Sound core KUnit test + * Author: Ivan Orlov + */ + +#include +#include +#include + +#define SILENCE_BUFFER_SIZE 2048 +#define SILENCE(...) { __VA_ARGS__ } +#define DEFINE_FORMAT(fmt, pbits, wd, endianness, signd, silence_arr) { \ + .format = SNDRV_PCM_FORMAT_##fmt, .physical_bits = pbits, \ + .width = wd, .le = endianness, .sd = signd, .silence = silence_arr, \ + .name = #fmt, \ +} + +#define WRONG_FORMAT (SNDRV_PCM_FORMAT_LAST + 1) + +#define VALID_NAME "ValidName" +#define NAME_W_SPEC_CHARS "In%v@1id name" +#define NAME_W_SPACE "Test name" +#define NAME_W_SPACE_REMOVED "Testname" + +#define TEST_FIRST_COMPONENT "Component1" +#define TEST_SECOND_COMPONENT "Component2" + +struct snd_format_test_data { + snd_pcm_format_t format; + int physical_bits; + int width; + int le; + int sd; + unsigned char silence[8]; + unsigned char *name; +}; + +struct avail_test_data { + snd_pcm_uframes_t buffer_size; + snd_pcm_uframes_t hw_ptr; + snd_pcm_uframes_t appl_ptr; + snd_pcm_uframes_t expected_avail; +}; + +static struct snd_format_test_data valid_fmt[] = { + DEFINE_FORMAT(S8, 8, 8, -1, 1, SILENCE()), + DEFINE_FORMAT(U8, 8, 8, -1, 0, SILENCE(0x80)), + DEFINE_FORMAT(S16_LE, 16, 16, 1, 1, SILENCE()), + DEFINE_FORMAT(S16_BE, 16, 16, 0, 1, SILENCE()), + DEFINE_FORMAT(U16_LE, 16, 16, 1, 0, SILENCE(0x00, 0x80)), + DEFINE_FORMAT(U16_BE, 16, 16, 0, 0, SILENCE(0x80, 0x00)), + DEFINE_FORMAT(S24_LE, 32, 24, 1, 1, SILENCE()), + DEFINE_FORMAT(S24_BE, 32, 24, 0, 1, SILENCE()), + DEFINE_FORMAT(U24_LE, 32, 24, 1, 0, SILENCE(0x00, 0x00, 0x80)), + DEFINE_FORMAT(U24_BE, 32, 24, 0, 0, SILENCE(0x00, 0x80, 0x00, 0x00)), + DEFINE_FORMAT(S32_LE, 32, 32, 1, 1, SILENCE()), + DEFINE_FORMAT(S32_BE, 32, 32, 0, 1, SILENCE()), + DEFINE_FORMAT(U32_LE, 32, 32, 1, 0, SILENCE(0x00, 0x00, 0x00, 0x80)), + DEFINE_FORMAT(U32_BE, 32, 32, 0, 0, SILENCE(0x80, 0x00, 0x00, 0x00)), + DEFINE_FORMAT(FLOAT_LE, 32, 32, 1, -1, SILENCE()), + DEFINE_FORMAT(FLOAT_BE, 32, 32, 0, -1, SILENCE()), + DEFINE_FORMAT(FLOAT64_LE, 64, 64, 1, -1, SILENCE()), + DEFINE_FORMAT(FLOAT64_BE, 64, 64, 0, -1, SILENCE()), + DEFINE_FORMAT(IEC958_SUBFRAME_LE, 32, 32, 1, -1, SILENCE()), + DEFINE_FORMAT(IEC958_SUBFRAME_BE, 32, 32, 0, -1, SILENCE()), + DEFINE_FORMAT(MU_LAW, 8, 8, -1, -1, SILENCE(0x7f)), + DEFINE_FORMAT(A_LAW, 8, 8, -1, -1, SILENCE(0x55)), + DEFINE_FORMAT(IMA_ADPCM, 4, 4, -1, -1, SILENCE()), + DEFINE_FORMAT(G723_24, 3, 3, -1, -1, SILENCE()), + DEFINE_FORMAT(G723_40, 5, 5, -1, -1, SILENCE()), + DEFINE_FORMAT(DSD_U8, 8, 8, 1, 0, SILENCE(0x69)), + DEFINE_FORMAT(DSD_U16_LE, 16, 16, 1, 0, SILENCE(0x69, 0x69)), + DEFINE_FORMAT(DSD_U32_LE, 32, 32, 1, 0, SILENCE(0x69, 0x69, 0x69, 0x69)), + DEFINE_FORMAT(DSD_U16_BE, 16, 16, 0, 0, SILENCE(0x69, 0x69)), + DEFINE_FORMAT(DSD_U32_BE, 32, 32, 0, 0, SILENCE(0x69, 0x69, 0x69, 0x69)), + DEFINE_FORMAT(S20_LE, 32, 20, 1, 1, SILENCE()), + DEFINE_FORMAT(S20_BE, 32, 20, 0, 1, SILENCE()), + DEFINE_FORMAT(U20_LE, 32, 20, 1, 0, SILENCE(0x00, 0x00, 0x08, 0x00)), + DEFINE_FORMAT(U20_BE, 32, 20, 0, 0, SILENCE(0x00, 0x08, 0x00, 0x00)), + DEFINE_FORMAT(S24_3LE, 24, 24, 1, 1, SILENCE()), + DEFINE_FORMAT(S24_3BE, 24, 24, 0, 1, SILENCE()), + DEFINE_FORMAT(U24_3LE, 24, 24, 1, 0, SILENCE(0x00, 0x00, 0x80)), + DEFINE_FORMAT(U24_3BE, 24, 24, 0, 0, SILENCE(0x80, 0x00, 0x00)), + DEFINE_FORMAT(S20_3LE, 24, 20, 1, 1, SILENCE()), + DEFINE_FORMAT(S20_3BE, 24, 20, 0, 1, SILENCE()), + DEFINE_FORMAT(U20_3LE, 24, 20, 1, 0, SILENCE(0x00, 0x00, 0x08)), + DEFINE_FORMAT(U20_3BE, 24, 20, 0, 0, SILENCE(0x08, 0x00, 0x00)), + DEFINE_FORMAT(S18_3LE, 24, 18, 1, 1, SILENCE()), + DEFINE_FORMAT(S18_3BE, 24, 18, 0, 1, SILENCE()), + DEFINE_FORMAT(U18_3LE, 24, 18, 1, 0, SILENCE(0x00, 0x00, 0x02)), + DEFINE_FORMAT(U18_3BE, 24, 18, 0, 0, SILENCE(0x02, 0x00, 0x00)), + DEFINE_FORMAT(G723_24_1B, 8, 3, -1, -1, SILENCE()), + DEFINE_FORMAT(G723_40_1B, 8, 5, -1, -1, SILENCE()), +}; + +static void test_phys_format_size(struct kunit *test) +{ + u32 i; + + for (i = 0; i < ARRAY_SIZE(valid_fmt); i++) { + KUNIT_EXPECT_EQ(test, snd_pcm_format_physical_width(valid_fmt[i].format), + valid_fmt[i].physical_bits); + } + + KUNIT_EXPECT_EQ(test, snd_pcm_format_physical_width(WRONG_FORMAT), -EINVAL); + KUNIT_EXPECT_EQ(test, snd_pcm_format_physical_width(-1), -EINVAL); +} + +static void test_format_width(struct kunit *test) +{ + u32 i; + + for (i = 0; i < ARRAY_SIZE(valid_fmt); i++) { + KUNIT_EXPECT_EQ(test, snd_pcm_format_width(valid_fmt[i].format), + valid_fmt[i].width); + } + + KUNIT_EXPECT_EQ(test, snd_pcm_format_width(WRONG_FORMAT), -EINVAL); + KUNIT_EXPECT_EQ(test, snd_pcm_format_width(-1), -EINVAL); +} + +static void test_format_signed(struct kunit *test) +{ + u32 i; + + for (i = 0; i < ARRAY_SIZE(valid_fmt); i++) { + KUNIT_EXPECT_EQ(test, snd_pcm_format_signed(valid_fmt[i].format), + valid_fmt[i].sd < 0 ? -EINVAL : valid_fmt[i].sd); + KUNIT_EXPECT_EQ(test, snd_pcm_format_unsigned(valid_fmt[i].format), + valid_fmt[i].sd < 0 ? -EINVAL : 1 - valid_fmt[i].sd); + } + + KUNIT_EXPECT_EQ(test, snd_pcm_format_width(WRONG_FORMAT), -EINVAL); + KUNIT_EXPECT_EQ(test, snd_pcm_format_width(-1), -EINVAL); +} + +static void test_format_endianness(struct kunit *test) +{ + u32 i; + + for (i = 0; i < ARRAY_SIZE(valid_fmt); i++) { + KUNIT_EXPECT_EQ(test, snd_pcm_format_little_endian(valid_fmt[i].format), + valid_fmt[i].le < 0 ? -EINVAL : valid_fmt[i].le); + KUNIT_EXPECT_EQ(test, snd_pcm_format_big_endian(valid_fmt[i].format), + valid_fmt[i].le < 0 ? -EINVAL : 1 - valid_fmt[i].le); + } + + KUNIT_EXPECT_EQ(test, snd_pcm_format_little_endian(WRONG_FORMAT), -EINVAL); + KUNIT_EXPECT_EQ(test, snd_pcm_format_little_endian(-1), -EINVAL); + KUNIT_EXPECT_EQ(test, snd_pcm_format_big_endian(WRONG_FORMAT), -EINVAL); + KUNIT_EXPECT_EQ(test, snd_pcm_format_big_endian(-1), -EINVAL); +} + +static void _test_fill_silence(struct kunit *test, struct snd_format_test_data *data, + u8 *buffer, size_t samples_count) +{ + size_t sample_bytes = data->physical_bits >> 3; + u32 i; + + KUNIT_ASSERT_EQ(test, snd_pcm_format_set_silence(data->format, buffer, samples_count), 0); + for (i = 0; i < samples_count * sample_bytes; i++) + KUNIT_EXPECT_EQ(test, buffer[i], data->silence[i % sample_bytes]); +} + +static void test_format_fill_silence(struct kunit *test) +{ + u32 buf_samples[] = { 10, 20, 32, 64, 129, 260 }; + u8 *buffer; + u32 i, j; + + buffer = kunit_kzalloc(test, SILENCE_BUFFER_SIZE, GFP_KERNEL); + + for (i = 0; i < ARRAY_SIZE(buf_samples); i++) { + for (j = 0; j < ARRAY_SIZE(valid_fmt); j++) + _test_fill_silence(test, &valid_fmt[j], buffer, buf_samples[i]); + } + + KUNIT_EXPECT_EQ(test, snd_pcm_format_set_silence(WRONG_FORMAT, buffer, 20), -EINVAL); + KUNIT_EXPECT_EQ(test, snd_pcm_format_set_silence(SNDRV_PCM_FORMAT_LAST, buffer, 0), 0); +} + +static snd_pcm_uframes_t calculate_boundary(snd_pcm_uframes_t buffer_size) +{ + snd_pcm_uframes_t boundary = buffer_size; + + while (boundary * 2 <= 0x7fffffffUL - buffer_size) + boundary *= 2; + return boundary; +} + +static struct avail_test_data p_avail_data[] = { + /* buf_size + hw_ptr < appl_ptr => avail = buf_size + hw_ptr - appl_ptr + boundary */ + { 128, 1000, 1129, 1073741824UL - 1 }, + /* + * buf_size + hw_ptr - appl_ptr >= boundary => + * => avail = buf_size + hw_ptr - appl_ptr - boundary + */ + { 128, 1073741824UL, 10, 118 }, + /* standard case: avail = buf_size + hw_ptr - appl_ptr */ + { 128, 1000, 1001, 127 }, +}; + +static void test_playback_avail(struct kunit *test) +{ + struct snd_pcm_runtime *r = kunit_kzalloc(test, sizeof(*r), GFP_KERNEL); + u32 i; + + r->status = kunit_kzalloc(test, sizeof(*r->status), GFP_KERNEL); + r->control = kunit_kzalloc(test, sizeof(*r->control), GFP_KERNEL); + + for (i = 0; i < ARRAY_SIZE(p_avail_data); i++) { + r->buffer_size = p_avail_data[i].buffer_size; + r->boundary = calculate_boundary(r->buffer_size); + r->status->hw_ptr = p_avail_data[i].hw_ptr; + r->control->appl_ptr = p_avail_data[i].appl_ptr; + KUNIT_EXPECT_EQ(test, snd_pcm_playback_avail(r), p_avail_data[i].expected_avail); + } +} + +static struct avail_test_data c_avail_data[] = { + /* hw_ptr - appl_ptr < 0 => avail = hw_ptr - appl_ptr + boundary */ + { 128, 1000, 1001, 1073741824UL - 1 }, + /* standard case: avail = hw_ptr - appl_ptr */ + { 128, 1001, 1000, 1 }, +}; + +static void test_capture_avail(struct kunit *test) +{ + struct snd_pcm_runtime *r = kunit_kzalloc(test, sizeof(*r), GFP_KERNEL); + u32 i; + + r->status = kunit_kzalloc(test, sizeof(*r->status), GFP_KERNEL); + r->control = kunit_kzalloc(test, sizeof(*r->control), GFP_KERNEL); + + for (i = 0; i < ARRAY_SIZE(c_avail_data); i++) { + r->buffer_size = c_avail_data[i].buffer_size; + r->boundary = calculate_boundary(r->buffer_size); + r->status->hw_ptr = c_avail_data[i].hw_ptr; + r->control->appl_ptr = c_avail_data[i].appl_ptr; + KUNIT_EXPECT_EQ(test, snd_pcm_capture_avail(r), c_avail_data[i].expected_avail); + } +} + +static void test_card_set_id(struct kunit *test) +{ + struct snd_card *card = kunit_kzalloc(test, sizeof(*card), GFP_KERNEL); + + snd_card_set_id(card, VALID_NAME); + KUNIT_EXPECT_STREQ(test, card->id, VALID_NAME); + + /* clear the first id character so we can set it again */ + card->id[0] = '\0'; + snd_card_set_id(card, NAME_W_SPEC_CHARS); + KUNIT_EXPECT_STRNEQ(test, card->id, NAME_W_SPEC_CHARS); + + card->id[0] = '\0'; + snd_card_set_id(card, NAME_W_SPACE); + kunit_info(test, "%s", card->id); + KUNIT_EXPECT_STREQ(test, card->id, NAME_W_SPACE_REMOVED); +} + +static void test_pcm_format_name(struct kunit *test) +{ + u32 i; + const char *name; + + for (i = 0; i < ARRAY_SIZE(valid_fmt); i++) { + name = snd_pcm_format_name(valid_fmt[i].format); + KUNIT_ASSERT_NOT_NULL_MSG(test, name, "Don't have name for %s", valid_fmt[i].name); + KUNIT_EXPECT_STREQ(test, name, valid_fmt[i].name); + } + + KUNIT_ASSERT_STREQ(test, snd_pcm_format_name(WRONG_FORMAT), "Unknown"); + KUNIT_ASSERT_STREQ(test, snd_pcm_format_name(-1), "Unknown"); +} + +static void test_card_add_component(struct kunit *test) +{ + struct snd_card *card = kunit_kzalloc(test, sizeof(*card), GFP_KERNEL); + + snd_component_add(card, TEST_FIRST_COMPONENT); + KUNIT_ASSERT_STREQ(test, card->components, TEST_FIRST_COMPONENT); + + snd_component_add(card, TEST_SECOND_COMPONENT); + KUNIT_ASSERT_STREQ(test, card->components, TEST_FIRST_COMPONENT " " TEST_SECOND_COMPONENT); +} + +static struct kunit_case sound_utils_cases[] = { + KUNIT_CASE(test_phys_format_size), + KUNIT_CASE(test_format_width), + KUNIT_CASE(test_format_endianness), + KUNIT_CASE(test_format_signed), + KUNIT_CASE(test_format_fill_silence), + KUNIT_CASE(test_playback_avail), + KUNIT_CASE(test_capture_avail), + KUNIT_CASE(test_card_set_id), + KUNIT_CASE(test_pcm_format_name), + KUNIT_CASE(test_card_add_component), + {}, +}; + +static struct kunit_suite sound_utils_suite = { + .name = "sound-core-test", + .test_cases = sound_utils_cases, +}; + +kunit_test_suite(sound_utils_suite); +MODULE_AUTHOR("Ivan Orlov"); +MODULE_LICENSE("GPL");