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

[PATCH] ASoC: dt-bindings: qcom,q6core: remove binding
by Krzysztof Kozlowski 07 Sep '22

07 Sep '22

qcom,q6core is already described in soc/qcom/qcom,apr.yaml. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- .../devicetree/bindings/sound/qcom,q6core.txt | 21 ------------------- 1 file changed, 21 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/qcom,q6core.txt diff --git a/Documentation/devicetree/bindings/sound/qcom,q6core.txt b/Documentation/devicetree/bindings/sound/qcom,q6core.txt deleted file mode 100644 index 5cd4cc9b1fde..000000000000 --- a/Documentation/devicetree/bindings/sound/qcom,q6core.txt +++ /dev/null @@ -1,21 +0,0 @@ -Qualcomm ADSP Core service binding - -Q6CORE is one of the APR audio service on Q6DSP. -Please refer to qcom,apr.txt for details of the common apr service bindings -used by the apr service device. - -- but must contain the following property: - -- compatible: - Usage: required - Value type: <stringlist> - Definition: must be "qcom,q6core-v<MAJOR-NUMBER>.<MINOR-NUMBER>". - Or "qcom,q6core" where the version number can be queried - from DSP. - example "qcom,q6core-v2.0" - -= EXAMPLE -apr-service@3 { - compatible = "qcom,q6core"; - reg = <APR_SVC_ADSP_CORE>; -}; -- 2.34.1

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[PATCH 0/4] ASoC: soc-dapm.c: random cleanup
by Kuninori Morimoto 07 Sep '22

07 Sep '22

Hi Mark These are cleanup patches for soc-dapm.c. Each patches are not related, very random cleanup. Kuninori Morimoto (4): ASoC: soc-dapm.c: don't use kzalloc() for param on snd_soc_dai_link_event_pre_pmu() ASoC: soc-dapm.c: don't use WARN_ON() at snd_soc_dai_link_event_pre_pmu() ASoC: soc-dapm.c: tidyup snd_soc_dai_link_event_pre_pmu() ASoC: soc-dapm.c: fixup snd_soc_dapm_new_control_unlocked() error handling sound/soc/soc-dapm.c | 77 +++++++++++++++++++------------------------- 1 file changed, 33 insertions(+), 44 deletions(-) -- 2.25.1

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utils: cplay: Add support for ID3v2 tag skip
by GitHub pull_request - edited 07 Sep '22

07 Sep '22

alsa-project/tinycompress pull request #15 was edited from LaurentiuM1234: This patch will allow playing MP3 files containing ID3v2 tag at the beginning of the file. This is done simply by parsing the ID3v2 header, extracting the header size and seeking to that position (+10 bytes from the header size which is not included in the 'header size' field of the ID3v2 header) Signed-off-by: Laurentiu Mihalcea <laurentiu.mihalcea(a)nxp.com> Request URL : https://github.com/alsa-project/tinycompress/pull/15 Patch URL : https://github.com/alsa-project/tinycompress/pull/15.patch Repository URL: https://github.com/alsa-project/tinycompress

1 0

utils: cplay: Add support for ID3v2 header skip
by GitHub pull_request - opened 07 Sep '22

07 Sep '22

alsa-project/tinycompress pull request #15 was opened from LaurentiuM1234: This patch will allow playing MP3 files containing ID3v2 tag at the beginning of the file. This is done simply by parsing the ID3v2 header, extracting the header size and seeking to that position (+10 bytes from the header size which is not included in the 'header size' field of the ID3v2 header) Signed-off-by: Laurentiu Mihalcea <laurentiu.mihalcea(a)nxp.com> Request URL : https://github.com/alsa-project/tinycompress/pull/15 Patch URL : https://github.com/alsa-project/tinycompress/pull/15.patch Repository URL: https://github.com/alsa-project/tinycompress

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[PATCH] ALSA: usb-audio: Clear fixed clock rate at closing EP
by Takashi Iwai 07 Sep '22

07 Sep '22

The recent commit c11117b634f4 ("ALSA: usb-audio: Refcount multiple accesses on the single clock") tries to manage the clock rate shared by several endpoints. This was intended for avoiding the unmatched rate by a different endpoint, but unfortunately, it introduced a regression for PulseAudio and pipewire, too; those applications try to probe the multiple possible rates (44.1k and 48kHz) and setting up the normal rate fails but only the last rate is applied. The cause is that the last sample rate is still left to the clock reference even after closing the endpoint, and this value is still used at the next open. It happens only when applications set up via PCM prepare but don't start/stop the stream; the rate is reset when the stream is stopped, but it's not cleared at close. This patch addresses the issue above, simply by clearing the rate set in the clock reference at the last close of each endpoint. Fixes: c11117b634f4 ("ALSA: usb-audio: Refcount multiple accesses on the single clock") Reported-by: Jason A. Donenfeld <Jason(a)zx2c4.com> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/all/YxXIWv8dYmg1tnXP@zx2c4.com/ Link: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/2620 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/endpoint.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index a42f2ce19455..8c8f9a851f89 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -925,6 +925,8 @@ void snd_usb_endpoint_close(struct snd_usb_audio *chip, endpoint_set_interface(chip, ep, false); if (!--ep->opened) { + if (ep->clock_ref && !atomic_read(&ep->clock_ref->locked)) + ep->clock_ref->rate = 0; ep->iface = 0; ep->altsetting = 0; ep->cur_audiofmt = NULL; -- 2.35.3

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[PATCH] ALSA: usb-audio: Don't refcount multiple accesses on the single clock
by Jason A. Donenfeld 07 Sep '22

07 Sep '22

This reverts commit 03a8b0df757f1beb21ba1626e23ca7412e48b525. This reverts commit c11117b634f4f832c4420d3cf41c44227f140ce1. Pipewire and PulseAudio start devices with 44.1khz before changing them to 48khz (or something different). By locking the rate, daemons are unable to enumerate possible rates, and so they never change them to a more optimal rate. This revert patch should allow 48khz audio again. Fixes: c11117b634f4 ("ALSA: usb-audio: Refcount multiple accesses on the single clock") Cc: stable(a)vger.kernel.org Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Niklāvs Koļesņikovs <89q1r14hd(a)relay.firefox.com> Cc: Wim Taymans <wtaymans(a)redhat.com> Link: https://lore.kernel.org/all/YxXIWv8dYmg1tnXP@zx2c4.com/ Link: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/2620 Signed-off-by: Jason A. Donenfeld <Jason(a)zx2c4.com> --- sound/usb/card.c | 1 - sound/usb/card.h | 3 +- sound/usb/endpoint.c | 90 +++++--------------------------------------- sound/usb/usbaudio.h | 1 - 4 files changed, 10 insertions(+), 85 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index d356743de2ff..12bca2e85505 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -643,7 +643,6 @@ static int snd_usb_audio_create(struct usb_interface *intf, INIT_LIST_HEAD(&chip->pcm_list); INIT_LIST_HEAD(&chip->ep_list); INIT_LIST_HEAD(&chip->iface_ref_list); - INIT_LIST_HEAD(&chip->clock_ref_list); INIT_LIST_HEAD(&chip->midi_list); INIT_LIST_HEAD(&chip->mixer_list); diff --git a/sound/usb/card.h b/sound/usb/card.h index ca75f2206170..87f042d06ce0 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -44,7 +44,6 @@ struct audioformat { struct snd_usb_substream; struct snd_usb_iface_ref; -struct snd_usb_clock_ref; struct snd_usb_endpoint; struct snd_usb_power_domain; @@ -63,7 +62,6 @@ struct snd_urb_ctx { struct snd_usb_endpoint { struct snd_usb_audio *chip; struct snd_usb_iface_ref *iface_ref; - struct snd_usb_clock_ref *clock_ref; int opened; /* open refcount; protect with chip->mutex */ atomic_t running; /* running status */ @@ -140,6 +138,7 @@ struct snd_usb_endpoint { unsigned int cur_period_frames; unsigned int cur_period_bytes; unsigned int cur_buffer_periods; + unsigned char cur_clock; spinlock_t lock; struct list_head list; diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 0d7b73bf7945..0f5e9224a799 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -35,14 +35,6 @@ struct snd_usb_iface_ref { struct list_head list; }; -/* clock refcounting */ -struct snd_usb_clock_ref { - unsigned char clock; - atomic_t locked; - int rate; - struct list_head list; -}; - /* * snd_usb_endpoint is a model that abstracts everything related to an * USB endpoint and its streaming. @@ -599,25 +591,6 @@ iface_ref_find(struct snd_usb_audio *chip, int iface) return ip; } -/* Similarly, a refcount object for clock */ -static struct snd_usb_clock_ref * -clock_ref_find(struct snd_usb_audio *chip, int clock) -{ - struct snd_usb_clock_ref *ref; - - list_for_each_entry(ref, &chip->clock_ref_list, list) - if (ref->clock == clock) - return ref; - - ref = kzalloc(sizeof(*ref), GFP_KERNEL); - if (!ref) - return NULL; - ref->clock = clock; - atomic_set(&ref->locked, 0); - list_add_tail(&ref->list, &chip->clock_ref_list); - return ref; -} - /* * Get the existing endpoint object corresponding EP * Returns NULL if not present. @@ -795,14 +768,6 @@ snd_usb_endpoint_open(struct snd_usb_audio *chip, goto unlock; } - if (fp->protocol != UAC_VERSION_1) { - ep->clock_ref = clock_ref_find(chip, fp->clock); - if (!ep->clock_ref) { - ep = NULL; - goto unlock; - } - } - ep->cur_audiofmt = fp; ep->cur_channels = fp->channels; ep->cur_rate = params_rate(params); @@ -812,6 +777,7 @@ snd_usb_endpoint_open(struct snd_usb_audio *chip, ep->cur_period_frames = params_period_size(params); ep->cur_period_bytes = ep->cur_period_frames * ep->cur_frame_bytes; ep->cur_buffer_periods = params_periods(params); + ep->cur_clock = fp->clock; if (ep->type == SND_USB_ENDPOINT_TYPE_SYNC) endpoint_set_syncinterval(chip, ep); @@ -928,8 +894,8 @@ void snd_usb_endpoint_close(struct snd_usb_audio *chip, ep->altsetting = 0; ep->cur_audiofmt = NULL; ep->cur_rate = 0; + ep->cur_clock = 0; ep->iface_ref = NULL; - ep->clock_ref = NULL; usb_audio_dbg(chip, "EP 0x%x closed\n", ep->ep_num); } mutex_unlock(&chip->mutex); @@ -941,8 +907,6 @@ void snd_usb_endpoint_suspend(struct snd_usb_endpoint *ep) ep->need_setup = true; if (ep->iface_ref) ep->iface_ref->need_setup = true; - if (ep->clock_ref) - ep->clock_ref->rate = 0; } /* @@ -1350,33 +1314,6 @@ static int snd_usb_endpoint_set_params(struct snd_usb_audio *chip, return 0; } -static int init_sample_rate(struct snd_usb_audio *chip, - struct snd_usb_endpoint *ep) -{ - struct snd_usb_clock_ref *clock = ep->clock_ref; - int err; - - if (clock) { - if (atomic_read(&clock->locked)) - return 0; - if (clock->rate == ep->cur_rate) - return 0; - if (clock->rate && clock->rate != ep->cur_rate) { - usb_audio_dbg(chip, "Mismatched sample rate %d vs %d for EP 0x%x\n", - clock->rate, ep->cur_rate, ep->ep_num); - return -EINVAL; - } - } - - err = snd_usb_init_sample_rate(chip, ep->cur_audiofmt, ep->cur_rate); - if (err < 0) - return err; - - if (clock) - clock->rate = ep->cur_rate; - return 0; -} - /* * snd_usb_endpoint_configure: Configure the endpoint * @@ -1406,7 +1343,8 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip, * to update at each EP configuration */ if (ep->cur_audiofmt->protocol == UAC_VERSION_1) { - err = init_sample_rate(chip, ep); + err = snd_usb_init_sample_rate(chip, ep->cur_audiofmt, + ep->cur_rate); if (err < 0) goto unlock; } @@ -1436,7 +1374,7 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip, if (err < 0) goto unlock; - err = init_sample_rate(chip, ep); + err = snd_usb_init_sample_rate(chip, ep->cur_audiofmt, ep->cur_rate); if (err < 0) goto unlock; @@ -1469,15 +1407,15 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip, /* get the current rate set to the given clock by any endpoint */ int snd_usb_endpoint_get_clock_rate(struct snd_usb_audio *chip, int clock) { - struct snd_usb_clock_ref *ref; + struct snd_usb_endpoint *ep; int rate = 0; if (!clock) return 0; mutex_lock(&chip->mutex); - list_for_each_entry(ref, &chip->clock_ref_list, list) { - if (ref->clock == clock) { - rate = ref->rate; + list_for_each_entry(ep, &chip->ep_list, list) { + if (ep->cur_clock == clock && ep->cur_rate) { + rate = ep->cur_rate; break; } } @@ -1518,9 +1456,6 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) if (atomic_inc_return(&ep->running) != 1) return 0; - if (ep->clock_ref) - atomic_inc(&ep->clock_ref->locked); - ep->active_mask = 0; ep->unlink_mask = 0; ep->phase = 0; @@ -1630,9 +1565,6 @@ void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep, bool keep_pending) if (ep->sync_source) WRITE_ONCE(ep->sync_source->sync_sink, NULL); stop_urbs(ep, false, keep_pending); - if (ep->clock_ref) - if (!atomic_dec_return(&ep->clock_ref->locked)) - ep->clock_ref->rate = 0; } } @@ -1659,16 +1591,12 @@ void snd_usb_endpoint_free_all(struct snd_usb_audio *chip) { struct snd_usb_endpoint *ep, *en; struct snd_usb_iface_ref *ip, *in; - struct snd_usb_clock_ref *cp, *cn; list_for_each_entry_safe(ep, en, &chip->ep_list, list) kfree(ep); list_for_each_entry_safe(ip, in, &chip->iface_ref_list, list) kfree(ip); - - list_for_each_entry_safe(cp, cn, &chip->clock_ref_list, list) - kfree(cp); } /* diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index ffbb4b0d09a0..044cd7ab27cb 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -45,7 +45,6 @@ struct snd_usb_audio { struct list_head pcm_list; /* list of pcm streams */ struct list_head ep_list; /* list of audio-related endpoints */ struct list_head iface_ref_list; /* list of interface refcounts */ - struct list_head clock_ref_list; /* list of clock refcounts */ int pcm_devs; struct list_head midi_list; /* list of midi interfaces */ -- 2.37.3

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[PATCH] ALSA: hda/ca0132: fix potential memory leak in dspxfr_image()
by Jianglei Nie 07 Sep '22

07 Sep '22

dspxfr_image() allocates DSP ports for the download stream with dsp_allocate_ports_format(). When gets some error, the allocated DSP ports are not released, which will lead to a memory leak. We can fix it by releasing DSP ports with dsp_free_ports() when getting some error. Signed-off-by: Jianglei Nie <niejianglei2021(a)163.com> --- sound/pci/hda/patch_ca0132.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 208933792787..6b8f45e14075 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -3455,6 +3455,7 @@ static int dspxfr_image(struct hda_codec *codec, &port_map_mask); if (status < 0) { codec_dbg(codec, "alloc ports fail\n"); + dsp_free_ports(codec); goto exit; } @@ -3463,6 +3464,7 @@ static int dspxfr_image(struct hda_codec *codec, WIDGET_CHIP_CTRL, stream_id, 0, &response); if (status < 0) { codec_dbg(codec, "set stream chan fail\n"); + dsp_free_ports(codec); goto exit; } @@ -3470,6 +3472,7 @@ static int dspxfr_image(struct hda_codec *codec, if (!is_valid(fls_data)) { codec_dbg(codec, "FLS check fail\n"); status = -EINVAL; + dsp_free_ports(codec); goto exit; } status = dspxfr_one_seg(codec, fls_data, reloc, -- 2.25.1

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Query for snd_pcm_readi function.
by Web X 07 Sep '22

07 Sep '22

Hi, I am using alsa open source library for my project to read and write audio data. For that I am using snd_pcm_readi function to read audio data. I am able to read audio data. But issue is that sometimes it will take so much time to read data.(200-230 msec). Normally this function executed in 1-3 ms that is normal case, but after every 2-4 frames it is taking 200-300 msec. So this is affecting my whole application process. I have also try to set this call as nonblock with the help of snd_pcm_nonblock function. With that I have observed some improvements but still after 2-3 frames it is taking 100msec time to get a data. Please do needful as soon as possible. Let me know if you want more details.

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[PATCH] ALSA: emu10k1: Fix out of bounds access in snd_emu10k1_pcm_channel_alloc()
by Tasos Sahanidis 07 Sep '22

07 Sep '22

The voice allocator sometimes begins allocating from near the end of the array and then wraps around, however snd_emu10k1_pcm_channel_alloc() accesses the newly allocated voices as if it never wrapped around. This results in out of bounds access if the first voice has a high enough index so that first_voice + requested_voice_count > NUM_G (64). The more voices are requested, the more likely it is for this to occur. This was initially discovered using PipeWire, however it can be reproduced by calling aplay multiple times with 16 channels: aplay -r 48000 -D plughw:CARD=Live,DEV=3 -c 16 /dev/zero UBSAN: array-index-out-of-bounds in sound/pci/emu10k1/emupcm.c:127:40 index 65 is out of range for type 'snd_emu10k1_voice [64]' CPU: 1 PID: 31977 Comm: aplay Tainted: G W IOE 6.0.0-rc2-emu10k1+ #7 Hardware name: ASUSTEK COMPUTER INC P5W DH Deluxe/P5W DH Deluxe, BIOS 3002 07/22/2010 Call Trace: <TASK> dump_stack_lvl+0x49/0x63 dump_stack+0x10/0x16 ubsan_epilogue+0x9/0x3f __ubsan_handle_out_of_bounds.cold+0x44/0x49 snd_emu10k1_playback_hw_params+0x3bc/0x420 [snd_emu10k1] snd_pcm_hw_params+0x29f/0x600 [snd_pcm] snd_pcm_common_ioctl+0x188/0x1410 [snd_pcm] ? exit_to_user_mode_prepare+0x35/0x170 ? do_syscall_64+0x69/0x90 ? syscall_exit_to_user_mode+0x26/0x50 ? do_syscall_64+0x69/0x90 ? exit_to_user_mode_prepare+0x35/0x170 snd_pcm_ioctl+0x27/0x40 [snd_pcm] __x64_sys_ioctl+0x95/0xd0 do_syscall_64+0x5c/0x90 ? do_syscall_64+0x69/0x90 ? do_syscall_64+0x69/0x90 entry_SYSCALL_64_after_hwframe+0x63/0xcd Signed-off-by: Tasos Sahanidis <tasos(a)tasossah.com> --- sound/pci/emu10k1/emupcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index b2701a4452d8..48af77ae8020 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -124,7 +124,7 @@ static int snd_emu10k1_pcm_channel_alloc(struct snd_emu10k1_pcm * epcm, int voic epcm->voices[0]->epcm = epcm; if (voices > 1) { for (i = 1; i < voices; i++) { - epcm->voices[i] = &epcm->emu->voices[epcm->voices[0]->number + i]; + epcm->voices[i] = &epcm->emu->voices[(epcm->voices[0]->number + i) % NUM_G]; epcm->voices[i]->epcm = epcm; } } -- 2.25.1

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Patch for fixing top firing speakers for Dell Inspiron 16 Plus (7620) laptop
by Miroslav Lakota 07 Sep '22

07 Sep '22

Dear Sir or Madam, I would like to submit a kernel patch fixing speakers on the laptop Dell Inspiron 16 Plus 7620. The patch has been tested for a month with 5.19 and 6.0RC kernels. The current issue: - The top firing speakers on the laptop do not work. They are not detected by the audio driver. The solution: - assign correct pins and DACs to the ALC chip Status after application of the patch: - Both bottom and top firing speakers work as expected. One small issue remains - the top firing speakers are enabled with a 0,5s delay. This is however not very noticeable in real life and the fact that they work already means a significant improvement. I am completely new to submiting kernel patches so I apologise beforehand if I made any mistake in the formal process. Best regards, Miroslav Lakota >From 6f91aba8189a0b04fd977d218321e89df7e0673e Mon Sep 17 00:00:00 2001 From: Philipp Jungkamp <p.jungkamp(a)gmx.net> Date: Mon, 15 Aug 2022 11:17:21 +0200 Subject: [PATCH] ALSA: patch_realtek: Fix Dell Inspiron Plus 16 --- sound/pci/hda/patch_realtek.c | 48 +++++++++++++++++++++++++++++++++++ 1 file changed, 48 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fd630d62b5a0..9b0b4ddcc36d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6854,6 +6854,46 @@ static void alc287_fixup_yoga9_14iap7_bass_spk_pin(struct hda_codec *codec, } } +static void alc295_fixup_dell_inspiron_top_speakers(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + /* + * The Pin Complex 0x17 for the top speakers is wrongly reported as + * unconnected. + * The top speakers should also have a sequence number lower than + * the bottom speakers on NID 0x14. + */ + static const struct hda_pintbl pincfgs[] = { + { 0x14, 0x90170151 }, + { 0x17, 0x90170150 }, + { } + }; + /* + * Avoid DAC 0x06 and 0x08, as they have no volume controls. + * DAC 0x02 and 0x03 would be fine. + */ + static const hda_nid_t conn[] = { 0x02, 0x03 }; + /* + * Prefer both bottom speakers (0x14) and headphones (0x21) connected to DAC 0x02. + * Top speakers (0x21) are connected to DAC 0x03. + */ + static const hda_nid_t preferred_pairs[] = { + 0x14, 0x02, + 0x17, 0x03, + 0x21, 0x02, + 0 + }; + struct alc_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_apply_pincfgs(codec, pincfgs); + snd_hda_override_conn_list(codec, 0x17, ARRAY_SIZE(conn), conn); + spec->gen.preferred_dacs = preferred_pairs; + break; + } +} + enum { ALC269_FIXUP_GPIO2, ALC269_FIXUP_SONY_VAIO, @@ -7094,6 +7134,7 @@ enum { ALC287_FIXUP_LEGION_16ITHG6, ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK, ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN, + ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -9017,6 +9058,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK, }, + [ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc295_fixup_dell_inspiron_top_speakers, + .chained = true, + .chain_id = ALC269_FIXUP_DELL4_MIC_NO_PRESENCE, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -9116,6 +9163,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0a9d, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0a9e, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0b19, "Dell XPS 15 9520", ALC289_FIXUP_DUAL_SPK), + SND_PCI_QUIRK(0x1028, 0x0b71, "Dell Inspiron 16 Plus 7620", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- 2.37.2

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