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

[PATCH] ASoC: simple-card-utils: switch to using gpiod API
by Dmitry Torokhov 07 Sep '22

07 Sep '22

This patch switches the driver away from legacy gpio/of_gpio API to gpiod API, and removes use of of_get_named_gpio_flags() which I want to make private to gpiolib. Signed-off-by: Dmitry Torokhov <dmitry.torokhov(a)gmail.com> --- sound/soc/generic/simple-card-utils.c | 28 ++++++++++++++------------- 1 file changed, 15 insertions(+), 13 deletions(-) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 1b201dd09259..bef16833c487 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -9,7 +9,6 @@ #include <linux/gpio/consumer.h> #include <linux/module.h> #include <linux/of.h> -#include <linux/of_gpio.h> #include <linux/of_graph.h> #include <sound/jack.h> #include <sound/pcm_params.h> @@ -729,12 +728,12 @@ int asoc_simple_init_jack(struct snd_soc_card *card, char *pin) { struct device *dev = card->dev; - enum of_gpio_flags flags; + struct gpio_desc *desc; char prop[128]; char *pin_name; char *gpio_name; int mask; - int det; + int error; if (!prefix) prefix = ""; @@ -742,36 +741,39 @@ int asoc_simple_init_jack(struct snd_soc_card *card, sjack->gpio.gpio = -ENOENT; if (is_hp) { - snprintf(prop, sizeof(prop), "%shp-det-gpio", prefix); + snprintf(prop, sizeof(prop), "%shp-det", prefix); pin_name = pin ? pin : "Headphones"; gpio_name = "Headphone detection"; mask = SND_JACK_HEADPHONE; } else { - snprintf(prop, sizeof(prop), "%smic-det-gpio", prefix); + snprintf(prop, sizeof(prop), "%smic-det", prefix); pin_name = pin ? pin : "Mic Jack"; gpio_name = "Mic detection"; mask = SND_JACK_MICROPHONE; } - det = of_get_named_gpio_flags(dev->of_node, prop, 0, &flags); - if (det == -EPROBE_DEFER) - return -EPROBE_DEFER; + desc = gpiod_get_optional(dev, prop, GPIOD_IN); + error = PTR_ERR_OR_ZERO(desc); + if (error) + return error; + + if (desc) { + error = gpiod_set_consumer_name(desc, gpio_name); + if (error) + return error; - if (gpio_is_valid(det)) { sjack->pin.pin = pin_name; sjack->pin.mask = mask; sjack->gpio.name = gpio_name; sjack->gpio.report = mask; - sjack->gpio.gpio = det; - sjack->gpio.invert = !!(flags & OF_GPIO_ACTIVE_LOW); + sjack->gpio.desc = desc; sjack->gpio.debounce_time = 150; snd_soc_card_jack_new_pins(card, pin_name, mask, &sjack->jack, &sjack->pin, 1); - snd_soc_jack_add_gpios(&sjack->jack, 1, - &sjack->gpio); + snd_soc_jack_add_gpios(&sjack->jack, 1, &sjack->gpio); } return 0; -- 2.37.2.789.g6183377224-goog -- Dmitry

3 2

[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

2 1

[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

2 8

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

1 0

[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

2 1

[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

2 7

[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

2 1

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.

2 1

[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

2 1

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

2 1

[linux-next:master] BUILD REGRESSION 840126e36e8ff272cb63158646433fa1324533d9
by kernel test robot 07 Sep '22

07 Sep '22

tree/branch: https://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git master branch HEAD: 840126e36e8ff272cb63158646433fa1324533d9 Add linux-next specific files for 20220906 Error/Warning reports: https://lore.kernel.org/linux-mm/202209021204.DcLzOllr-lkp@intel.com https://lore.kernel.org/linux-mm/202209042337.FQi69rLV-lkp@intel.com https://lore.kernel.org/linux-mm/202209060229.dVuyxjBv-lkp@intel.com https://lore.kernel.org/linux-mm/202209070728.o3stvgVt-lkp@intel.com https://lore.kernel.org/llvm/202208312208.HjwleIeN-lkp@intel.com Error/Warning: (recently discovered and may have been fixed) ERROR: modpost: "__aeabi_ldivmod" [drivers/gpu/drm/vkms/vkms.ko] undefined! ERROR: modpost: "__aeabi_uldivmod" [drivers/gpu/drm/vkms/vkms.ko] undefined! ERROR: modpost: "__divdi3" [drivers/gpu/drm/vkms/vkms.ko] undefined! ERROR: modpost: "__udivdi3" [drivers/gpu/drm/vkms/vkms.ko] undefined! arm-linux-gnueabi-ld: vkms_formats.c:(.text+0x1e98): undefined reference to `__divdi3' drivers/base/regmap/regmap-mmio.c:221:17: error: implicit declaration of function 'writesb'; did you mean 'writeb'? [-Werror=implicit-function-declaration] drivers/base/regmap/regmap-mmio.c:224:17: error: implicit declaration of function 'writesw'; did you mean 'writew'? [-Werror=implicit-function-declaration] drivers/base/regmap/regmap-mmio.c:227:17: error: implicit declaration of function 'writesl'; did you mean 'writel'? [-Werror=implicit-function-declaration] drivers/base/regmap/regmap-mmio.c:231:17: error: implicit declaration of function 'writesq'; did you mean 'writeq'? [-Werror=implicit-function-declaration] drivers/base/regmap/regmap-mmio.c:231:17: error: implicit declaration of function 'writesq'; did you mean 'writesl'? [-Werror=implicit-function-declaration] drivers/base/regmap/regmap-mmio.c:358:17: error: implicit declaration of function 'readsb'; did you mean 'readb'? [-Werror=implicit-function-declaration] drivers/base/regmap/regmap-mmio.c:361:17: error: implicit declaration of function 'readsw'; did you mean 'readw'? [-Werror=implicit-function-declaration] drivers/base/regmap/regmap-mmio.c:364:17: error: implicit declaration of function 'readsl'; did you mean 'readl'? [-Werror=implicit-function-declaration] drivers/base/regmap/regmap-mmio.c:368:17: error: implicit declaration of function 'readsq'; did you mean 'readq'? [-Werror=implicit-function-declaration] drivers/base/regmap/regmap-mmio.c:368:17: error: implicit declaration of function 'readsq'; did you mean 'readsl'? [-Werror=implicit-function-declaration] drivers/gpu/drm/amd/amdgpu/imu_v11_0_3.c:139:6: warning: no previous prototype for 'imu_v11_0_3_program_rlc_ram' [-Wmissing-prototypes] drivers/gpu/drm/drm_atomic_helper.c:802: warning: expecting prototype for drm_atomic_helper_check_wb_connector_state(). Prototype was for drm_atomic_helper_check_wb_encoder_state() instead drivers/gpu/drm/vkms/vkms_formats.c:(.text+0x4b0): undefined reference to `__divdi3' drivers/gpu/drm/vkms/vkms_formats.c:259: undefined reference to `__divdi3' drivers/gpu/drm/vkms/vkms_plane.c:105 vkms_plane_atomic_update() warn: variable dereferenced before check 'fb' (see line 103) drivers/scsi/qla2xxx/qla_os.c:2854:23: warning: assignment to 'struct trace_array *' from 'int' makes pointer from integer without a cast [-Wint-conversion] drivers/usb/host/ehci-platform.c:56:19: warning: 'hcd_name' defined but not used [-Wunused-const-variable=] drivers/usb/host/ohci-platform.c:44:19: warning: 'hcd_name' defined but not used [-Wunused-const-variable=] include/linux/string.h:303:42: warning: 'strnlen' specified bound 4 exceeds source size 3 [-Wstringop-overread] kernel/bpf/memalloc.c:344 bpf_mem_alloc_destroy() error: potentially dereferencing uninitialized 'c'. kismet: WARNING: unmet direct dependencies detected for PINCTRL_IMX when selected by PINCTRL_IMX8MM ld: drivers/gpu/drm/vkms/vkms_formats.c:260: undefined reference to `__divdi3' ld: vkms_formats.c:(.text+0x47f): undefined reference to `__divdi3' mips-linux-ld: vkms_formats.c:(.text+0x384): undefined reference to `__divdi3' mips-linux-ld: vkms_formats.c:(.text.argb_u16_to_RGB565+0xd0): undefined reference to `__divdi3' mipsel-linux-ld: drivers/gpu/drm/vkms/vkms_formats.c:(.text+0x4d8): undefined reference to `__divdi3' sound/soc/codecs/tas2562.c:442:13: warning: variable 'ret' set but not used [-Wunused-but-set-variable] vkms_formats.c:(.text+0x455): undefined reference to `__divdi3' vkms_formats.c:(.text.argb_u16_to_RGB565+0xb0): undefined reference to `__divdi3' Unverified Error/Warning (likely false positive, please contact us if interested): drivers/usb/host/ehci-atmel.c:28:19: warning: unused variable 'hcd_name' [-Wunused-const-variable] drivers/usb/host/ehci-exynos.c:35:19: warning: unused variable 'hcd_name' [-Wunused-const-variable] drivers/usb/host/ehci-npcm7xx.c:27:19: warning: unused variable 'hcd_name' [-Wunused-const-variable] drivers/usb/host/ehci-orion.c:68:19: warning: unused variable 'hcd_name' [-Wunused-const-variable] drivers/usb/host/ehci-platform.c:56:19: warning: unused variable 'hcd_name' [-Wunused-const-variable] drivers/usb/host/ehci-spear.c:27:19: warning: unused variable 'hcd_name' [-Wunused-const-variable] drivers/usb/host/ohci-platform.c:44:19: warning: unused variable 'hcd_name' [-Wunused-const-variable] Error/Warning ids grouped by kconfigs: gcc_recent_errors |-- alpha-allyesconfig | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-readsb | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-readsl | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-readsq | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-readsw | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-writesb | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-writesl | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-writesq | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-writesw | |-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-imu_v11_0_3_program_rlc_ram | `-- drivers-scsi-qla2xxx-qla_os.c:warning:assignment-to-struct-trace_array-from-int-makes-pointer-from-integer-without-a-cast |-- alpha-buildonly-randconfig-r001-20220906 | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-readsb | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-readsl | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-readsq | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-readsw | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-writesb | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-writesl | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-writesq | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-writesw | `-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-imu_v11_0_3_program_rlc_ram |-- arc-allyesconfig | `-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-imu_v11_0_3_program_rlc_ram |-- arm-allyesconfig | `-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-imu_v11_0_3_program_rlc_ram |-- arm-randconfig-r006-20220906 | |-- ERROR:__aeabi_ldivmod-drivers-gpu-drm-vkms-vkms.ko-undefined | `-- ERROR:__aeabi_uldivmod-drivers-gpu-drm-vkms-vkms.ko-undefined |-- arm-randconfig-r013-20220905 | `-- arm-linux-gnueabi-ld:vkms_formats.c:(.text):undefined-reference-to-__divdi3 |-- arm64-allyesconfig | `-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-imu_v11_0_3_program_rlc_ram |-- csky-randconfig-r006-20220906 | `-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-imu_v11_0_3_program_rlc_ram |-- csky-randconfig-s031-20220906 | |-- ERROR:__divdi3-drivers-gpu-drm-vkms-vkms.ko-undefined | `-- kernel-exit.c:sparse:sparse:incorrect-type-in-initializer-(different-address-spaces)-expected-struct-sighand_struct-sighand-got-struct-sighand_struct-noderef-__rcu-sighand |-- i386-allyesconfig | `-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-imu_v11_0_3_program_rlc_ram |-- i386-randconfig-a001 | |-- drivers-usb-host-ehci-platform.c:warning:hcd_name-defined-but-not-used | `-- drivers-usb-host-ohci-platform.c:warning:hcd_name-defined-but-not-used |-- i386-randconfig-a003 | |-- ERROR:__divdi3-drivers-gpu-drm-vkms-vkms.ko-undefined | `-- ERROR:__udivdi3-drivers-gpu-drm-vkms-vkms.ko-undefined |-- i386-randconfig-a012 | |-- drivers-gpu-drm-vkms-vkms_formats.c:undefined-reference-to-__divdi3 | `-- ld:drivers-gpu-drm-vkms-vkms_formats.c:undefined-reference-to-__divdi3 |-- i386-randconfig-a014 | |-- ld:vkms_formats.c:(.text):undefined-reference-to-__divdi3 clang_recent_errors |-- arm64-randconfig-r005-20220906 | `-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-function-imu_v11_0_3_program_rlc_ram |-- hexagon-randconfig-r001-20220906 | `-- drivers-extcon-extcon-usbc-tusb320.c:warning:expecting-prototype-for-drivers-extcon-extcon-tusb320c().-Prototype-was-for-TUSB320_REG8()-instead |-- hexagon-randconfig-r045-20220907 | |-- drivers-usb-host-ehci-exynos.c:warning:unused-variable-hcd_name | |-- drivers-usb-host-ehci-npcm7xx.c:warning:unused-variable-hcd_name | `-- drivers-usb-host-ehci-orion.c:warning:unused-variable-hcd_name |-- i386-randconfig-a002 | `-- drivers-extcon-extcon-usbc-tusb320.c:warning:expecting-prototype-for-drivers-extcon-extcon-tusb320c().-Prototype-was-for-TUSB320_REG8()-instead |-- i386-randconfig-a011 | |-- drivers-usb-host-ehci-platform.c:warning:unused-variable-hcd_name | `-- drivers-usb-host-ohci-platform.c:warning:unused-variable-hcd_name |-- i386-randconfig-a015 | `-- drivers-extcon-extcon-usbc-tusb320.c:warning:expecting-prototype-for-drivers-extcon-extcon-tusb320c().-Prototype-was-for-TUSB320_REG8()-instead |-- riscv-randconfig-r004-20220906 | `-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-function-imu_v11_0_3_program_rlc_ram |-- riscv-randconfig-r042-20220907 | |-- drivers-usb-host-ehci-atmel.c:warning:unused-variable-hcd_name | `-- drivers-usb-host-ehci-spear.c:warning:unused-variable-hcd_name |-- s390-randconfig-r036-20220906 | |-- drivers-scsi-qla2xxx-qla_os.c:error:call-to-undeclared-function-trace_array_get_by_name-ISO-C99-and-later-do-not-support-implicit-function-declarations | |-- drivers-scsi-qla2xxx-qla_os.c:error:call-to-undeclared-function-trace_array_put-ISO-C99-and-later-do-not-support-implicit-function-declarations | `-- drivers-scsi-qla2xxx-qla_os.c:error:incompatible-integer-to-pointer-conversion-assigning-to-struct-trace_array-from-int |-- x86_64-randconfig-a003 | `-- drivers-extcon-extcon-usbc-tusb320.c:warning:expecting-prototype-for-drivers-extcon-extcon-tusb320c().-Prototype-was-for-TUSB320_REG8()-instead |-- x86_64-randconfig-a012 | `-- drivers-extcon-extcon-usbc-tusb320.c:warning:expecting-prototype-for-drivers-extcon-extcon-tusb320c().-Prototype-was-for-TUSB320_REG8()-instead `-- x86_64-randconfig-a016 `-- drivers-extcon-extcon-usbc-tusb320.c:warning:expecting-prototype-for-drivers-extcon-extcon-tusb320c().-Prototype-was-for-TUSB320_REG8()-instead elapsed time: 721m configs tested: 64 configs skipped: 2 gcc tested configs: arc randconfig-r043-20220906 riscv randconfig-r042-20220906 um x86_64_defconfig s390 randconfig-r044-20220906 um i386_defconfig i386 randconfig-a014 i386 randconfig-a001 i386 randconfig-a012 i386 defconfig i386 randconfig-a016 i386 randconfig-a003 m68k allmodconfig i386 randconfig-a005 arc allyesconfig alpha allyesconfig m68k allyesconfig x86_64 defconfig x86_64 rhel-8.3 arm defconfig x86_64 rhel-8.3-func x86_64 randconfig-a002 x86_64 randconfig-a013 x86_64 rhel-8.3-kunit x86_64 randconfig-a011 i386 allyesconfig x86_64 randconfig-a006 x86_64 rhel-8.3-kselftests x86_64 allyesconfig x86_64 rhel-8.3-kvm x86_64 rhel-8.3-syz x86_64 randconfig-a015 x86_64 randconfig-a004 ia64 allmodconfig arm allyesconfig microblaze mmu_defconfig openrisc alldefconfig arm pxa3xx_defconfig sh r7785rp_defconfig sh sh7710voipgw_defconfig arm64 allyesconfig powerpc allnoconfig powerpc allmodconfig mips allyesconfig csky allnoconfig arc allnoconfig alpha allnoconfig riscv allnoconfig i386 randconfig-c001 sh allmodconfig clang tested configs: hexagon randconfig-r041-20220906 hexagon randconfig-r045-20220906 i386 randconfig-a013 i386 randconfig-a011 i386 randconfig-a002 i386 randconfig-a015 i386 randconfig-a006 i386 randconfig-a004 x86_64 randconfig-a001 x86_64 randconfig-a016 x86_64 randconfig-a003 x86_64 randconfig-a012 x86_64 randconfig-a014 x86_64 randconfig-a005 x86_64 rhel-8.3-rust -- 0-DAY CI Kernel Test Service https://01.org/lkp

1 0

[PATCH] ASoC: fsl_asrc: Add initialization finishing check in runtime resume
by Shengjiu Wang 06 Sep '22

06 Sep '22

If the initialization is not finished, then filling input data to the FIFO may fail. So it is better to add initialization finishing check in the runtime resume for suspend & resume case. And consider the case of three instances working in parallel, increase the retry times to 50 for more initialization time. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_asrc.c | 19 ++++++++++++++++++- 1 file changed, 18 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index aa5edf32d988..bae263b90ac1 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -20,6 +20,7 @@ #define IDEAL_RATIO_DECIMAL_DEPTH 26 #define DIVIDER_NUM 64 +#define INIT_TRY_NUM 50 #define pair_err(fmt, ...) \ dev_err(&asrc->pdev->dev, "Pair %c: " fmt, 'A' + index, ##__VA_ARGS__) @@ -579,7 +580,7 @@ static void fsl_asrc_start_pair(struct fsl_asrc_pair *pair) { struct fsl_asrc *asrc = pair->asrc; enum asrc_pair_index index = pair->index; - int reg, retry = 10, i; + int reg, retry = INIT_TRY_NUM, i; /* Enable the current pair */ regmap_update_bits(asrc->regmap, REG_ASRCTR, @@ -592,6 +593,10 @@ static void fsl_asrc_start_pair(struct fsl_asrc_pair *pair) reg &= ASRCFG_INIRQi_MASK(index); } while (!reg && --retry); + /* FIXME: Doesn't treat initialization timeout as error */ + if (!retry) + dev_warn(&asrc->pdev->dev, "initialization isn't finished\n"); + /* Make the input fifo to ASRC STALL level */ regmap_read(asrc->regmap, REG_ASRCNCR, &reg); for (i = 0; i < pair->channels * 4; i++) @@ -1257,6 +1262,7 @@ static int fsl_asrc_runtime_resume(struct device *dev) { struct fsl_asrc *asrc = dev_get_drvdata(dev); struct fsl_asrc_priv *asrc_priv = asrc->private; + int reg, retry = INIT_TRY_NUM; int i, ret; u32 asrctr; @@ -1295,6 +1301,17 @@ static int fsl_asrc_runtime_resume(struct device *dev) regmap_update_bits(asrc->regmap, REG_ASRCTR, ASRCTR_ASRCEi_ALL_MASK, asrctr); + /* Wait for status of initialization for every enabled pairs */ + do { + udelay(5); + regmap_read(asrc->regmap, REG_ASRCFG, &reg); + reg = (reg >> ASRCFG_INIRQi_SHIFT(0)) & 0x7; + } while ((reg != ((asrctr >> ASRCTR_ASRCEi_SHIFT(0)) & 0x7)) && --retry); + + /* FIXME: Doesn't treat initialization timeout as error */ + if (!retry) + dev_warn(dev, "initialization isn't finished\n"); + return 0; disable_asrck_clk: -- 2.34.1

3 6

snd_pcm_readi() takes to much time.
by GitHub issues - opened 06 Sep '22

06 Sep '22

alsa-project/alsa-lib issue #270 was opened from coder0201200: 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. Issue URL : https://github.com/alsa-project/alsa-lib/issues/270 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v1 1/2] ALSA: hda: cs35l41: Call put_device() in the scope of get_device()
by Andy Shevchenko 06 Sep '22

06 Sep '22

When put_device() is called in another function it's hard to realize that and easy to "fix" the code in a wrong way. Instead, move put_device() to be in the same scope as get_device(), so we prevent appearance of any attempts to "fix" the code. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/pci/hda/cs35l41_hda.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 15e2a0009080..12e955931044 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -1154,7 +1154,6 @@ static int cs35l41_no_acpi_dsd(struct cs35l41_hda *cs35l41, struct device *physd hw_cfg->gpio2.func = CS35L41_INTERRUPT; hw_cfg->gpio2.valid = true; hw_cfg->valid = true; - put_device(physdev); if (strncmp(hid, "CLSA0100", 8) == 0) { hw_cfg->bst_type = CS35L41_EXT_BOOST_NO_VSPK_SWITCH; @@ -1204,9 +1203,10 @@ static int cs35l41_hda_read_acpi(struct cs35l41_hda *cs35l41, const char *hid, i property = "cirrus,dev-index"; ret = device_property_count_u32(physdev, property); - if (ret <= 0) - return cs35l41_no_acpi_dsd(cs35l41, physdev, id, hid); - + if (ret <= 0) { + ret = cs35l41_no_acpi_dsd(cs35l41, physdev, id, hid); + goto err_put_physdev; + } if (ret > ARRAY_SIZE(values)) { ret = -EINVAL; goto err; @@ -1295,8 +1295,9 @@ static int cs35l41_hda_read_acpi(struct cs35l41_hda *cs35l41, const char *hid, i return 0; err: - put_device(physdev); dev_err(cs35l41->dev, "Failed property %s: %d\n", property, ret); +err_put_physdev: + put_device(physdev); return ret; } -- 2.35.1

2 2

[PATCH] ALSA: hda: beep: Simplify keep-power-at-enable behavior
by Takashi Iwai 06 Sep '22

06 Sep '22

The recent fix for IDT codecs to keep the power up while the beep is enabled can be better integrated into the beep helper code. This patch cleans up the code with refactoring. Fixes: 414d38ba8710 ("ALSA: hda/sigmatel: Keep power up while beep is enabled") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_beep.c | 15 +++++++++++++-- sound/pci/hda/hda_beep.h | 1 + sound/pci/hda/patch_sigmatel.c | 25 ++----------------------- 3 files changed, 16 insertions(+), 25 deletions(-) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 53a2b89f8983..e63621bcb214 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -118,6 +118,12 @@ static int snd_hda_beep_event(struct input_dev *dev, unsigned int type, return 0; } +static void turn_on_beep(struct hda_beep *beep) +{ + if (beep->keep_power_at_enable) + snd_hda_power_up_pm(beep->codec); +} + static void turn_off_beep(struct hda_beep *beep) { cancel_work_sync(&beep->beep_work); @@ -125,6 +131,8 @@ static void turn_off_beep(struct hda_beep *beep) /* turn off beep */ generate_tone(beep, 0); } + if (beep->keep_power_at_enable) + snd_hda_power_down_pm(beep->codec); } /** @@ -140,7 +148,9 @@ int snd_hda_enable_beep_device(struct hda_codec *codec, int enable) enable = !!enable; if (beep->enabled != enable) { beep->enabled = enable; - if (!enable) + if (enable) + turn_on_beep(beep); + else turn_off_beep(beep); return 1; } @@ -167,7 +177,8 @@ static int beep_dev_disconnect(struct snd_device *device) input_unregister_device(beep->dev); else input_free_device(beep->dev); - turn_off_beep(beep); + if (beep->enabled) + turn_off_beep(beep); return 0; } diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h index a25358a4807a..db76e3ddba65 100644 --- a/sound/pci/hda/hda_beep.h +++ b/sound/pci/hda/hda_beep.h @@ -25,6 +25,7 @@ struct hda_beep { unsigned int enabled:1; unsigned int linear_tone:1; /* linear tone for IDT/STAC codec */ unsigned int playing:1; + unsigned int keep_power_at_enable:1; /* set by driver */ struct work_struct beep_work; /* scheduled task for beep event */ struct mutex mutex; void (*power_hook)(struct hda_beep *beep, bool on); diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 7f340f18599c..a794a01a68ca 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4311,6 +4311,8 @@ static int stac_parse_auto_config(struct hda_codec *codec) if (codec->beep) { /* IDT/STAC codecs have linear beep tone parameter */ codec->beep->linear_tone = spec->linear_tone_beep; + /* keep power up while beep is enabled */ + codec->beep->keep_power_at_enable = 1; /* if no beep switch is available, make its own one */ caps = query_amp_caps(codec, nid, HDA_OUTPUT); if (!(caps & AC_AMPCAP_MUTE)) { @@ -4444,28 +4446,6 @@ static int stac_suspend(struct hda_codec *codec) return 0; } - -static int stac_check_power_status(struct hda_codec *codec, hda_nid_t nid) -{ -#ifdef CONFIG_SND_HDA_INPUT_BEEP - struct sigmatel_spec *spec = codec->spec; -#endif - int ret = snd_hda_gen_check_power_status(codec, nid); - -#ifdef CONFIG_SND_HDA_INPUT_BEEP - if (nid == spec->gen.beep_nid && codec->beep) { - if (codec->beep->enabled != spec->beep_power_on) { - spec->beep_power_on = codec->beep->enabled; - if (spec->beep_power_on) - snd_hda_power_up_pm(codec); - else - snd_hda_power_down_pm(codec); - } - ret |= spec->beep_power_on; - } -#endif - return ret; -} #else #define stac_suspend NULL #endif /* CONFIG_PM */ @@ -4478,7 +4458,6 @@ static const struct hda_codec_ops stac_patch_ops = { .unsol_event = snd_hda_jack_unsol_event, #ifdef CONFIG_PM .suspend = stac_suspend, - .check_power_status = stac_check_power_status, #endif }; -- 2.35.3

1 0

[PATCH] ALSA: hda: Once again fix regression of page allocations with IOMMU
by Takashi Iwai 06 Sep '22

06 Sep '22

The last fix for trying to recover the regression on AMD platforms, unfortunately, leaded to yet another regression: it turned out that IOMMUs don't like the usage of raw page allocations. This is yet another attempt for addressing the log saga; at this time, we re-use the existing buffer allocation mechanism with SG-pages although we require only single pages. The SG buffer allocation itself was confirmed to work for stream buffers, so it's relatively easy to adapt for other places. The only problem is: although the HD-audio code is accessing the address directly via dmab->address field, SG-pages don't set up it. For the ease of adaption, we now set up the dmab->addr field from the address of the first page as default, so that it can run with the HD-audio driver code as-is without the excessive call of snd_sgbuf_get_addr() multiple times; that's the only change in the memalloc helper side. The rest is nothing but a flip of the dma_type field in the HD-audio side. Fixes: a8d302a0b770 ("ALSA: memalloc: Revive x86-specific WC page allocations again") Reported-by: Mikhail Gavrilov <mikhail.v.gavrilov(a)gmail.com> Tested-by: Mikhail Gavrilov <mikhail.v.gavrilov(a)gmail.com> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/CABXGCsO+kB2t5QyHY-rUe76npr1m0-5JOtt8g8SiHUo34ur7… Link: https://bugzilla.kernel.org/show_bug.cgi?id=216112 Link: https://bugzilla.kernel.org/show_bug.cgi?id=216363 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/memalloc.c | 9 +++++++-- sound/pci/hda/hda_intel.c | 2 +- 2 files changed, 8 insertions(+), 3 deletions(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index b665ac66ccbe..cfcd8eff4139 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -543,10 +543,13 @@ static void *snd_dma_noncontig_alloc(struct snd_dma_buffer *dmab, size_t size) dmab->dev.need_sync = dma_need_sync(dmab->dev.dev, sg_dma_address(sgt->sgl)); p = dma_vmap_noncontiguous(dmab->dev.dev, size, sgt); - if (p) + if (p) { dmab->private_data = sgt; - else + /* store the first page address for convenience */ + dmab->addr = snd_sgbuf_get_addr(dmab, 0); + } else { dma_free_noncontiguous(dmab->dev.dev, size, sgt, dmab->dev.dir); + } return p; } @@ -780,6 +783,8 @@ static void *snd_dma_sg_fallback_alloc(struct snd_dma_buffer *dmab, size_t size) if (!p) goto error; dmab->private_data = sgbuf; + /* store the first page address for convenience */ + dmab->addr = snd_sgbuf_get_addr(dmab, 0); return p; error: diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index a77165bd92a9..b20694fd69de 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1817,7 +1817,7 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, /* use the non-cached pages in non-snoop mode */ if (!azx_snoop(chip)) - azx_bus(chip)->dma_type = SNDRV_DMA_TYPE_DEV_WC; + azx_bus(chip)->dma_type = SNDRV_DMA_TYPE_DEV_WC_SG; if (chip->driver_type == AZX_DRIVER_NVIDIA) { dev_dbg(chip->card->dev, "Enable delay in RIRB handling\n"); -- 2.35.3

1 0

[PATCH] ASoC: max98390: Remove unnecessary amp on/off conrtol
by Steve Lee 06 Sep '22

06 Sep '22

The Amp is already control in userspace before trigger calibrate function. Remove unnecessary control in calibrate function. Signed-off-by: Steve Lee <steve.lee.analog(a)gmail.com> --- sound/soc/codecs/max98390.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index 91c0bf3d76fc..99e769190568 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -826,9 +826,6 @@ static int max98390_dsm_calibrate(struct snd_soc_component *component) struct max98390_priv *max98390 = snd_soc_component_get_drvdata(component); - regmap_write(max98390->regmap, MAX98390_R203A_AMP_EN, 0x81); - regmap_write(max98390->regmap, MAX98390_R23FF_GLOBAL_EN, 0x01); - regmap_read(max98390->regmap, THERMAL_RDC_RD_BACK_BYTE1, &rdc); regmap_read(max98390->regmap, @@ -847,9 +844,6 @@ static int max98390_dsm_calibrate(struct snd_soc_component *component) dev_info(component->dev, "rdc resistance about %d.%02d ohm, reg=0x%X temp reg=0x%X\n", rdc_integer, rdc_factor, rdc_cal_result, temp); - regmap_write(max98390->regmap, MAX98390_R23FF_GLOBAL_EN, 0x00); - regmap_write(max98390->regmap, MAX98390_R203A_AMP_EN, 0x80); - return 0; } -- 2.17.1

3 4

[PATCH] ALSA: usb-audio: Fix an out-of-bounds bug in __snd_usb_parse_audio_interface()
by Dongxiang Ke 06 Sep '22

06 Sep '22

There may be a bad USB audio device with a USB ID of (0x04fa, 0x4201) and the number of it's interfaces less than 4, an out-of-bounds read bug occurs when parsing the interface descriptor for this device. Fix this by checking the number of interfaces. Signed-off-by: Dongxiang Ke <kdx.glider(a)gmail.com> --- sound/usb/stream.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/stream.c b/sound/usb/stream.c index ceb93d798182..40ce8a1cb318 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -1105,7 +1105,7 @@ static int __snd_usb_parse_audio_interface(struct snd_usb_audio *chip, * Dallas DS4201 workaround: It presents 5 altsettings, but the last * one misses syncpipe, and does not produce any sound. */ - if (chip->usb_id == USB_ID(0x04fa, 0x4201)) + if (chip->usb_id == USB_ID(0x04fa, 0x4201) && num >= 4) num = 4; for (i = 0; i < num; i++) { -- 2.30.2

2 1

[PATCH] ALSA: hda/tegra: Align BDL entry to 4KB boundary
by Mohan Kumar 06 Sep '22

06 Sep '22

AZA HW may send a burst read/write request crossing 4K memory boundary. The 4KB boundary is not guaranteed by Tegra HDA HW. Make SW change to include the flag AZX_DCAPS_4K_BDLE_BOUNDARY to align BDLE to 4K boundary. Signed-off-by: Mohan Kumar <mkumard(a)nvidia.com> --- sound/pci/hda/hda_tegra.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 7debb2c76aa6..976a112c7d00 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -474,7 +474,8 @@ MODULE_DEVICE_TABLE(of, hda_tegra_match); static int hda_tegra_probe(struct platform_device *pdev) { const unsigned int driver_flags = AZX_DCAPS_CORBRP_SELF_CLEAR | - AZX_DCAPS_PM_RUNTIME; + AZX_DCAPS_PM_RUNTIME | + AZX_DCAPS_4K_BDLE_BOUNDARY; struct snd_card *card; struct azx *chip; struct hda_tegra *hda; -- 2.17.1

2 1

[PATCH v3 0/4] ASoC platform driver for Apple MCA
by Martin Povišer 06 Sep '22

06 Sep '22

Hi all, sending what should be the final touches on Apple MCA driver. It most likely goes without saying but please do not merge the DT additions into the ASoC tree. Martin Changes since v2: - fix Mark's style nit - order compatibles in schema - redo the DT additions for t8103 - remove stray unlock in mca_be_prepare (and rename the locking patch) Link: https://lore.kernel.org/asahi/20220819125430.4920-1-povik+lin@cutebit.org/T… Changes since v1: - reflect the set_fmt provider/consumer change in 6.0-rc1 - fix the probe function requesting optional reset - unroll 'dma-names' in schema - fix binding schema/code discrepancy in register ranges - minor things Link: https://lore.kernel.org/asahi/20220808224153.3634-1-povik+lin@cutebit.org/T… Changes since 'macaudio RFC v2' [0]: - addition of locking (extra commit) - transition to set_bclk_ratio (instead of getting the bclk ratio from set_sysclk) - using shared reset control and documenting the reset in binding - formatting, comments, and a minor fix to hw driving Link: https://lore.kernel.org/asahi/20220606191910.16580-1-povik+lin@cutebit.org/… Martin Povišer (4): dt-bindings: sound: Add Apple MCA I2S transceiver arm64: dts: apple: t8103: Add MCA and its support ASoC: apple: mca: Start new platform driver ASoC: apple: mca: Add locking .../devicetree/bindings/sound/apple,mca.yaml | 131 ++ MAINTAINERS | 8 + arch/arm64/boot/dts/apple/t8103-jxxx.dtsi | 4 + arch/arm64/boot/dts/apple/t8103.dtsi | 73 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/apple/Kconfig | 9 + sound/soc/apple/Makefile | 3 + sound/soc/apple/mca.c | 1167 +++++++++++++++++ 9 files changed, 1397 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/apple,mca.yaml create mode 100644 sound/soc/apple/Kconfig create mode 100644 sound/soc/apple/Makefile create mode 100644 sound/soc/apple/mca.c -- 2.33.0

4 9

amidi: --send shouldn't require a seekable file
by GitHub issues - opened 06 Sep '22

06 Sep '22

alsa-project/alsa-utils issue #167 was opened from BasedUser: Just like `--receive`, `--send` should instead send events to a given port until the sent file returns EOF, or `amidi` is terminated. Pipes and FIFOs cannot currently be used to create a persistent MIDI source. Issue URL : https://github.com/alsa-project/alsa-utils/issues/167 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH 0/2] Untested TAS2562 power setting fixes
by Martin Povišer 05 Sep '22

05 Sep '22

The tas2562 driver does the same thing with the setting of PWR_CTRL field as the tas2764/tas2770 drivers were doing. Link: https://lore.kernel.org/alsa-devel/20220808141246.5749-1-povik+lin@cutebit.… Link: https://lore.kernel.org/alsa-devel/20220825140241.53963-1-povik+lin@cutebit… These are blindly written patches without testing since I don't have the hardware. (I even tried TI's formal sample request program but was refused there. CCing @ti.com addresses I found on other series recently submitted.) Martin Povišer (2): ASoC: tas2562: Drop conflicting set_bias_level power setting ASoC: tas2562: Fix mute/unmute sound/soc/codecs/tas2562.c | 90 ++++++++++++-------------------------- 1 file changed, 29 insertions(+), 61 deletions(-) -- 2.33.0

2 5

[PATCH v6 0/2] lib/string_helpers: Introduce parse_int_array_user()
by Cezary Rojewski 05 Sep '22

05 Sep '22

Continuation of recent upstream discussion [1] regarding user string tokenization. First, parse_int_array_user() is introduced to allow for splitting specified user string into a sequence of integers. Makes use of get_options() internally so the parsing logic is not duplicated. With that done, redundant parts of the sound driver are removed. Originally similar functionality was added for the SOF sound driver. As more users are on the horizon, it is desirable to update existing string_helpers code and provide a unified solution. Changes in v6: - minor improvements in SOF code: dropped superfluous array-empty checks as suggested by Andy Changes in v5: - fixed kernel doc for parse_int_array_user() Changes in v4: - renamed the function to parse_int_array_user() - at the name several local variable names have been reworded to match the above Changes in v3: - relocated tokenize_user_input() implementation to string_helpers as requested by Matthew Changes in v2: - reused get_options() so no parsing logic is duplicated - simplified __user variant with help of memdup_user_nul() Both suggested by Andy, thanks for thorough review [1]: https://lore.kernel.org/alsa-devel/20220707091301.1282291-1-cezary.rojewski… Cezary Rojewski (2): lib/string_helpers: Introduce parse_int_array_user() ASoC: SOF: Remove strsplit_u32() and tokenize_input() include/linux/string_helpers.h | 2 + lib/string_helpers.c | 44 +++++++++++++ sound/soc/sof/sof-client-probes.c | 104 +++++------------------------- 3 files changed, 61 insertions(+), 89 deletions(-) -- 2.25.1

2 3

[PATCH] ASoC: SOF: Introduce function sof_of_machine_select
by Chunxu Li 05 Sep '22

05 Sep '22

>From current design in sof_machine_check the SOF can only support ACPI type machine. In sof_machine_check if there is no ACPI machine exist, the function will return -ENODEV directly, that's we don't expected if we do not base on ACPI machine. So we add a new function named sof_of_machine_select that we can pass sof_machine_check and obtain info required by snd_sof_new_platform_drv. Signed-off-by: Chunxu Li <chunxu.li(a)mediatek.com> --- sound/soc/sof/sof-audio.c | 29 +++++++++++++++++++++++++++++ 1 file changed, 29 insertions(+) diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index c18e723435bd..ea9663d448eb 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -785,6 +785,28 @@ int sof_dai_get_bclk(struct snd_soc_pcm_runtime *rtd) } EXPORT_SYMBOL(sof_dai_get_bclk); +static struct snd_sof_of_mach *sof_of_machine_select(struct snd_sof_dev *sdev) +{ + struct snd_sof_pdata *sof_pdata = sdev->pdata; + const struct sof_dev_desc *desc = sof_pdata->desc; + struct snd_sof_of_mach *mach = desc->of_machines; + + if (!mach) + return NULL; + + for (; mach->compatible; mach++) { + if (of_machine_is_compatible(mach->compatible)) { + sof_pdata->tplg_filename = mach->sof_tplg_filename; + if (mach->fw_filename) + sof_pdata->fw_filename = mach->fw_filename; + + return mach; + } + } + + return NULL; +} + /* * SOF Driver enumeration. */ @@ -795,6 +817,7 @@ int sof_machine_check(struct snd_sof_dev *sdev) struct snd_soc_acpi_mach *mach; if (!IS_ENABLED(CONFIG_SND_SOC_SOF_FORCE_NOCODEC_MODE)) { + const struct snd_sof_of_mach *of_mach; /* find machine */ mach = snd_sof_machine_select(sdev); @@ -804,6 +827,12 @@ int sof_machine_check(struct snd_sof_dev *sdev) return 0; } + of_mach = sof_of_machine_select(sdev); + if (of_mach) { + sof_pdata->of_machine = of_mach; + return 0; + } + if (!IS_ENABLED(CONFIG_SND_SOC_SOF_NOCODEC)) { dev_err(sdev->dev, "error: no matching ASoC machine driver found - aborting probe\n"); return -ENODEV; -- 2.25.1

3 2

[PATCH v2] ASoC: wm_adsp: Handle optional legacy support
by Cristian Ciocaltea 05 Sep '22

05 Sep '22

The tracing capabilities for the speaker protection fw enabled via commit c55b3e46cb99 ("ASoC: wm_adsp: Add trace caps to speaker protection FW") are not be available on all platforms, such as the Valve's Steam Deck which is based on the Halo Core DSP. As a consequence, whenever the firmware is loaded, a rather misleading 'Failed to parse legacy: -19' error message is written to the kernel ring buffer: [ 288.977412] steamdeck kernel: cs35l41 spi-VLV1776:01: DSP1: Firmware version: 3 [ 288.978002] steamdeck kernel: cs35l41 spi-VLV1776:01: DSP1: cs35l41-dsp1-spk-prot.wmfw: Fri 02 Apr 2021 21:03:50 W. Europe Daylight Time [ 289.094065] steamdeck kernel: cs35l41 spi-VLV1776:01: DSP1: Firmware: 400a4 vendor: 0x2 v0.33.0, 2 algorithms [ 289.095073] steamdeck kernel: cs35l41 spi-VLV1776:01: DSP1: 0: ID cd v29.53.0 XM@94 YM@e [ 289.095665] steamdeck kernel: cs35l41 spi-VLV1776:01: DSP1: 1: ID f20b v0.0.1 XM@170 YM@0 [ 289.096275] steamdeck kernel: cs35l41 spi-VLV1776:01: DSP1: Protection: C:\Users\ocanavan\Desktop\cirrusTune_july2021.bin [ 291.172383] steamdeck kernel: cs35l41 spi-VLV1776:01: DSP1: Failed to parse legacy: -19 Update wm_adsp_buffer_init() to print a more descriptive info message when wm_adsp_buffer_parse_legacy() returns -ENODEV. Fixes: c55b3e46cb99 ("ASoC: wm_adsp: Add trace caps to speaker protection FW") Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea(a)collabora.com> --- Changes in v2: - Add a generic handling of the issue, not bound to a particular firmware, as suggested by Charles - Update commit message, including the subject, to reflect the changes sound/soc/codecs/wm_adsp.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index cfaa45ede916..8a2e9771bb50 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1602,7 +1602,9 @@ static int wm_adsp_buffer_init(struct wm_adsp *dsp) if (list_empty(&dsp->buffer_list)) { /* Fall back to legacy support */ ret = wm_adsp_buffer_parse_legacy(dsp); - if (ret) + if (ret == -ENODEV) + adsp_info(dsp, "Legacy support not available\n"); + else if (ret) adsp_warn(dsp, "Failed to parse legacy: %d\n", ret); } base-commit: 568035b01cfb107af8d2e4bd2fb9aea22cf5b868 -- 2.37.2

3 2

[PATCH] ASoC: apple: mca: Postpone requesting of DMA channels
by Martin Povišer 05 Sep '22

05 Sep '22

Move the requesting of DMA channels further down from 'probe' to 'pcm_new'. This is to spare the allocated DMA channel resources as we typically only ever use one or two of the clusters for PCM streaming. Before we would request DMA channels for all clusters. (This is prompted by a change in the Audio DMA Controller driver, which will now be allocating cache SRAM to channels.) Signed-off-by: Martin Povišer <povik+lin(a)cutebit.org> --- sound/soc/apple/mca.c | 79 +++++++++++++++++++++++++------------------ 1 file changed, 46 insertions(+), 33 deletions(-) diff --git a/sound/soc/apple/mca.c b/sound/soc/apple/mca.c index aa67d57c9a9b..69643524796e 100644 --- a/sound/soc/apple/mca.c +++ b/sound/soc/apple/mca.c @@ -885,6 +885,43 @@ static snd_pcm_uframes_t mca_pointer(struct snd_soc_component *component, return snd_dmaengine_pcm_pointer(substream); } +static struct dma_chan *mca_request_dma_channel(struct mca_cluster *cl, unsigned int stream) +{ + bool is_tx = (stream == SNDRV_PCM_STREAM_PLAYBACK); +#ifndef USE_RXB_FOR_CAPTURE + char *name = devm_kasprintf(cl->host->dev, GFP_KERNEL, + is_tx ? "tx%da" : "rx%da", cl->no); +#else + char *name = devm_kasprintf(cl->host->dev, GFP_KERNEL, + is_tx ? "tx%da" : "rx%db", cl->no); +#endif + return of_dma_request_slave_channel(cl->host->dev->of_node, name); + +} + +static void mca_pcm_free(struct snd_soc_component *component, + struct snd_pcm *pcm) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_chip(pcm); + struct mca_cluster *cl = mca_dai_to_cluster(asoc_rtd_to_cpu(rtd, 0)); + unsigned int i; + + if (rtd->dai_link->no_pcm) + return; + + for_each_pcm_streams(i) { + struct snd_pcm_substream *substream = + rtd->pcm->streams[i].substream; + + if (!substream || !cl->dma_chans[i]) + continue; + + dma_release_channel(cl->dma_chans[i]); + cl->dma_chans[i] = NULL; + } +} + + static int mca_pcm_new(struct snd_soc_component *component, struct snd_soc_pcm_runtime *rtd) { @@ -897,17 +934,21 @@ static int mca_pcm_new(struct snd_soc_component *component, for_each_pcm_streams(i) { struct snd_pcm_substream *substream = rtd->pcm->streams[i].substream; - struct dma_chan *chan = cl->dma_chans[i]; + struct dma_chan *chan; if (!substream) continue; - if (!chan) { - dev_err(component->dev, "missing DMA channel for stream %d on SERDES %d\n", - i, cl->no); + chan = mca_request_dma_channel(cl, i); + + if (IS_ERR_OR_NULL(chan)) { + dev_err(component->dev, "unable to obtain DMA channel (stream %d cluster %d): %pe\n", + i, cl->no, chan); + mca_pcm_free(component, rtd->pcm); return -EINVAL; } + cl->dma_chans[i] = chan; snd_pcm_set_managed_buffer(substream, SNDRV_DMA_TYPE_DEV_IRAM, chan->device->dev, 512 * 1024 * 6, SIZE_MAX); @@ -924,6 +965,7 @@ static const struct snd_soc_component_driver mca_component = { .trigger = mca_trigger, .pointer = mca_pointer, .pcm_construct = mca_pcm_new, + .pcm_destruct = mca_pcm_free, }; static void apple_mca_release(struct mca_data *mca) @@ -1019,7 +1061,6 @@ static int apple_mca_probe(struct platform_device *pdev) struct snd_soc_dai_driver *fe = &dai_drivers[mca->nclusters + i]; struct snd_soc_dai_driver *be = &dai_drivers[i]; - int stream; cl->host = mca; cl->no = i; @@ -1041,34 +1082,6 @@ static int apple_mca_probe(struct platform_device *pdev) goto err_release; } - for_each_pcm_streams(stream) { - struct dma_chan *chan; - bool is_tx = (stream == SNDRV_PCM_STREAM_PLAYBACK); -#ifndef USE_RXB_FOR_CAPTURE - char *name = devm_kasprintf(&pdev->dev, GFP_KERNEL, - is_tx ? "tx%da" : "rx%da", - i); -#else - char *name = devm_kasprintf(&pdev->dev, GFP_KERNEL, - is_tx ? "tx%da" : "rx%db", - i); -#endif - - chan = of_dma_request_slave_channel(pdev->dev.of_node, - name); - if (IS_ERR(chan)) { - if (PTR_ERR(chan) != -EPROBE_DEFER) - dev_err(&pdev->dev, - "no %s DMA channel: %ld\n", - name, PTR_ERR(chan)); - - ret = PTR_ERR(chan); - goto err_release; - } - - cl->dma_chans[stream] = chan; - } - fe->id = i; fe->name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "mca-pcm-%d", i); -- 2.33.0

2 1

[PATCH 0/3] ASoC: soc-pcm.c: random cleanup
by Kuninori Morimoto 05 Sep '22

05 Sep '22

Hi Mark These are not related, but random cleanup patches for soc-pcm.c Kuninori Morimoto (3): ASoC: soc-pcm.c: remove unnecessary codec2codec_close_delayed_work() ASoC: soc-pcm.c: add soc_pcm_ret() ASoC: soc-pcm.c: check fe condition at out of loop sound/soc/soc-pcm.c | 112 +++++++++++++++++++------------------------- 1 file changed, 47 insertions(+), 65 deletions(-) -- 2.25.1

2 4

[PATCH] ALSA: hda/sigmatel: Fix unused variable warning for beep power change
by Takashi Iwai 05 Sep '22

05 Sep '22

The newly added stac_check_power_status() caused a compile warning when CONFIG_SND_HDA_INPUT_BEEP is disabled. Fix it. Fixes: 414d38ba8710 ("ALSA: hda/sigmatel: Keep power up while beep is enabled") Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_sigmatel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 066bfccd2587..7f340f18599c 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4447,7 +4447,9 @@ static int stac_suspend(struct hda_codec *codec) static int stac_check_power_status(struct hda_codec *codec, hda_nid_t nid) { +#ifdef CONFIG_SND_HDA_INPUT_BEEP struct sigmatel_spec *spec = codec->spec; +#endif int ret = snd_hda_gen_check_power_status(codec, nid); #ifdef CONFIG_SND_HDA_INPUT_BEEP -- 2.35.3

1 0

[PATCH] ALSA: usb-audio: Refcount multiple accesses on the single clock
by Takashi Iwai 05 Sep '22

05 Sep '22

When a clock source is connected to multiple nodes / endpoints, the current USB-audio driver tries to set up at each time one of them is configured. Although it reads the current rate and updates only if it differs, some devices seem unhappy with this behavior and spew the errors when reading/updating the rate unnecessarily. This patch tries to reduce the redundant clock setup by introducing a refcount for each clock source. When the stream is actually running, a clock rate is "locked", and it bypasses the clock and/or refuse to change any longer. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=215934 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/card.c | 1 + sound/usb/card.h | 3 +- sound/usb/endpoint.c | 90 +++++++++++++++++++++++++++++++++++++++----- sound/usb/usbaudio.h | 1 + 4 files changed, 85 insertions(+), 10 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 376962291c4d..0fff96a5d3ab 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -635,6 +635,7 @@ 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 87f042d06ce0..ca75f2206170 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -44,6 +44,7 @@ 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; @@ -62,6 +63,7 @@ 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 */ @@ -138,7 +140,6 @@ 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 743b8287cfcd..df5a70013a85 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -35,6 +35,14 @@ 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. @@ -591,6 +599,25 @@ 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. @@ -768,6 +795,14 @@ 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); @@ -777,7 +812,6 @@ 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); @@ -894,8 +928,8 @@ 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); @@ -907,6 +941,8 @@ 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; } /* @@ -1314,6 +1350,33 @@ 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 * @@ -1343,8 +1406,7 @@ 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 = snd_usb_init_sample_rate(chip, ep->cur_audiofmt, - ep->cur_rate); + err = init_sample_rate(chip, ep); if (err < 0) goto unlock; } @@ -1374,7 +1436,7 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip, if (err < 0) goto unlock; - err = snd_usb_init_sample_rate(chip, ep->cur_audiofmt, ep->cur_rate); + err = init_sample_rate(chip, ep); if (err < 0) goto unlock; @@ -1407,15 +1469,15 @@ 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_endpoint *ep; + struct snd_usb_clock_ref *ref; int rate = 0; if (!clock) return 0; mutex_lock(&chip->mutex); - list_for_each_entry(ep, &chip->ep_list, list) { - if (ep->cur_clock == clock && ep->cur_rate) { - rate = ep->cur_rate; + list_for_each_entry(ref, &chip->clock_ref_list, list) { + if (ref->clock == clock) { + rate = ref->rate; break; } } @@ -1456,6 +1518,9 @@ 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; @@ -1565,6 +1630,9 @@ 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; } } @@ -1591,12 +1659,16 @@ 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(ip); } /* diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 044cd7ab27cb..ffbb4b0d09a0 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -45,6 +45,7 @@ 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 */ -- 2.35.3

2 2

[PATCH] MAINTAINERS: rectify entry in ARM/APPLE MACHINE SOUND DRIVERS
by Lukas Bulwahn 05 Sep '22

05 Sep '22

Commit 3df5d0d97289 ("ASoC: apple: mca: Start new platform driver") adds a new sound driver at the location "sound/soc/apple/", but it adds a file entry referring to the non-existing location "drivers/sound/apple/*". Hence, ./scripts/get_maintainer.pl --self-test=patterns complains about a broken reference. Repair this file reference in ARM/APPLE MACHINE SOUND DRIVERS. Signed-off-by: Lukas Bulwahn <lukas.bulwahn(a)gmail.com> --- Martin, please ack. Mark, please pick this patch on top of the commit above. MAINTAINERS | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index 29443aad5031..c239d7e69158 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -1904,7 +1904,7 @@ L: asahi(a)lists.linux.dev L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Maintained F: Documentation/devicetree/bindings/sound/apple,* -F: drivers/sound/apple/* +F: sound/soc/apple/ ARM/ARTPEC MACHINE SUPPORT M: Jesper Nilsson <jesper.nilsson(a)axis.com> -- 2.17.1

2 2

[PATCH] ALSA: pcm: oss: Fix race at SNDCTL_DSP_SYNC
by Takashi Iwai 05 Sep '22

05 Sep '22

There is a small race window at snd_pcm_oss_sync() that is called from OSS PCM SNDCTL_DSP_SYNC ioctl; namely the function calls snd_pcm_oss_make_ready() at first, then takes the params_lock mutex for the rest. When the stream is set up again by another thread between them, it leads to inconsistency, and may result in unexpected results such as NULL dereference of OSS buffer as a fuzzer spotted recently. The fix is simply to cover snd_pcm_oss_make_ready() call into the same params_lock mutex with snd_pcm_oss_make_ready_locked() variant. Reported-and-tested-by: butt3rflyh4ck <butterflyhuangxx(a)gmail.com> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/CAFcO6XN7JDM4xSXGhtusQfS2mSBcx50VJKwQpCq=WeLt57aa… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/oss/pcm_oss.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 90c3a367d7de..02df915eb3c6 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -1672,14 +1672,14 @@ static int snd_pcm_oss_sync(struct snd_pcm_oss_file *pcm_oss_file) runtime = substream->runtime; if (atomic_read(&substream->mmap_count)) goto __direct; - err = snd_pcm_oss_make_ready(substream); - if (err < 0) - return err; atomic_inc(&runtime->oss.rw_ref); if (mutex_lock_interruptible(&runtime->oss.params_lock)) { atomic_dec(&runtime->oss.rw_ref); return -ERESTARTSYS; } + err = snd_pcm_oss_make_ready_locked(substream); + if (err < 0) + goto unlock; format = snd_pcm_oss_format_from(runtime->oss.format); width = snd_pcm_format_physical_width(format); if (runtime->oss.buffer_used > 0) { -- 2.35.3

2 1

A new null-ptr-deref Write bug in snd_pcm_format_set_silence
by butt3rflyh4ck 05 Sep '22

05 Sep '22

Hi, there is a new null-ptr-deref Write bug in snd_pcm_format_set_slience in sound/core/pcm_misc.c in the latest upstream kernel and can reproduce it. We call SNDCTL_DSP_SYNC and SNDCTL_DSP_SPEED in multiple threads to trigger the vulnerability. See the Call Trace: ================================================================== Call Trace: <TASK> __dump_stack lib/dump_stack.c:88 [inline] dump_stack_lvl+0xcd/0x134 lib/dump_stack.c:106 kasan_report+0xb1/0x1e0 mm/kasan/report.c:495 check_region_inline mm/kasan/generic.c:183 [inline] kasan_check_range+0x13d/0x180 mm/kasan/generic.c:189 memset+0x20/0x40 mm/kasan/shadow.c:44 snd_pcm_format_set_silence sound/core/pcm_misc.c:441 [inline] snd_pcm_format_set_silence+0x215/0x350 sound/core/pcm_misc.c:424 snd_pcm_oss_sync+0x60e/0x800 sound/core/oss/pcm_oss.c:1690 snd_pcm_oss_ioctl+0x2087/0x3420 sound/core/oss/pcm_oss.c:2634 vfs_ioctl fs/ioctl.c:51 [inline] __do_sys_ioctl fs/ioctl.c:870 [inline] __se_sys_ioctl fs/ioctl.c:856 [inline] __x64_sys_ioctl+0x193/0x200 fs/ioctl.c:856 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x35/0xb0 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x63/0xcd ================================================================== We can see the function snd_pcm_format_set_silence code below: ``` int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int samples) { int width; unsigned char *dst; const unsigned char *pat; if (!valid_format(format)) return -EINVAL; if (samples == 0) return 0; width = pcm_formats[(INT)format].phys; /* physical width */ pat = pcm_formats[(INT)format].silence; if (!width || !pat) return -EINVAL; /* signed or 1 byte data */ if (pcm_formats[(INT)format].signd == 1 || width <= 8) { unsigned int bytes = samples * width / 8; memset(data, *pat, bytes); ///// [1] ---------> data is NULL return 0; } ...... } ``` [1], the data pointer is NULL, we can know snd_pcm_format_set_silence called in line 1690 in sound/core/oss/pcm_oss.c from call stack trace. let we see code below: ``` static int snd_pcm_oss_sync(struct snd_pcm_oss_file *pcm_oss_file) { int err = 0; unsigned int saved_f_flags; struct snd_pcm_substream *substream; struct snd_pcm_runtime *runtime; snd_pcm_format_t format; unsigned long width; size_t size; substream = pcm_oss_file->streams[SNDRV_PCM_STREAM_PLAYBACK]; if (substream != NULL) { runtime = substream->runtime; if (atomic_read(&substream->mmap_count)) goto __direct; err = snd_pcm_oss_make_ready(substream); if (err < 0) return err; atomic_inc(&runtime->oss.rw_ref); if (mutex_lock_interruptible(&runtime->oss.params_lock)) { atomic_dec(&runtime->oss.rw_ref); return -ERESTARTSYS; } format = snd_pcm_oss_format_from(runtime->oss.format); width = snd_pcm_format_physical_width(format); if (runtime->oss.buffer_used > 0) { #ifdef OSS_DEBUG pcm_dbg(substream->pcm, "sync: buffer_used\n"); #endif size = (8 * (runtime->oss.period_bytes - runtime->oss.buffer_used) + 7) / width; snd_pcm_format_set_silence(format, runtime->oss.buffer + runtime->oss.buffer_used, ///// [2] size); err = snd_pcm_oss_sync1(substream, runtime->oss.period_bytes); if (err < 0) goto unlock; } else if (runtime->oss.period_ptr > 0) { ``` [2] runtime->oss.buffer + runtime->oss.buffer_used is the data pointer, but runtime->oss.buffer is NULL here but it doesn't make sense. runtime->oss.buffter is allocated by kvzalloc, if runtime->oss_buffer is NULL, it would return an ENOMEM error. Maybe I think there is a race condition, the runtime->oss.buffer is freed and set to NULL but we can use runtime->oss.buffter via ioctl. ###reproduce it [ 167.258988][T25615] ================================================================== [ 167.265917][T25615] BUG: KASAN: null-ptr-deref in snd_pcm_format_set_silence+0x215/0x350 [ 167.266704][T25615] Write of size 16383 at addr 0000000000000001 by task snd_pcm_format_/25615 [ 167.267506][T25615] [ 167.267732][T25615] CPU: 0 PID: 25615 Comm: snd_pcm_format_ Not tainted 6.0.0-rc3-00299-gd895ec7938c4 #11 [ 167.268617][T25615] Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS 1.14.0-2 04/01/2014 [ 167.269410][T25615] Call Trace: [ 167.269697][T25615] <TASK> [ 167.269977][T25615] dump_stack_lvl+0xcd/0x134 [ 167.270428][T25615] ? snd_pcm_format_set_silence+0x215/0x350 [ 167.270985][T25615] kasan_report+0xb1/0x1e0 [ 167.271405][T25615] ? snd_pcm_format_set_silence+0x215/0x350 [ 167.271964][T25615] kasan_check_range+0x13d/0x180 [ 167.272440][T25615] memset+0x20/0x40 [ 167.272809][T25615] snd_pcm_format_set_silence+0x215/0x350 [ 167.273366][T25615] snd_pcm_oss_sync+0x60e/0x800 [ 167.273831][T25615] snd_pcm_oss_ioctl+0x2087/0x3420 [ 167.274320][T25615] ? snd_pcm_oss_release+0x300/0x300 [ 167.274817][T25615] ? __fget_files+0x26a/0x440 [ 167.275262][T25615] ? bpf_lsm_file_ioctl+0x5/0x10 [ 167.275731][T25615] ? snd_pcm_oss_release+0x300/0x300 [ 167.276222][T25615] __x64_sys_ioctl+0x193/0x200 [ 167.276677][T25615] do_syscall_64+0x35/0xb0 [ 167.277108][T25615] entry_SYSCALL_64_after_hwframe+0x63/0xcd [ 167.277679][T25615] RIP: 0033:0x44af9d [ 167.278058][T25615] Code: 66 2e 0f 1f 84 00 00 00 00 00 0f 1f 00 f3 0f 1e fa 48 89 f8 48 89 f7 48 89 d6 48 89 ca 4d 89 c2 4d 89 c8 4c 8b 4c 24 08 0f 05 <48> 3d 01 f0 ff ff 73 01 c3 48 c7 c1 b8 ff ff 8 [ 167.279792][T25615] RSP: 002b:00007f0cb636a178 EFLAGS: 00000246 ORIG_RAX: 0000000000000010 [ 167.280546][T25615] RAX: ffffffffffffffda RBX: 0000000000000000 RCX: 000000000044af9d [ 167.281285][T25615] RDX: 0000000000000000 RSI: 0000000000005001 RDI: 0000000000000003 [ 167.282001][T25615] RBP: 00007f0cb636a1a0 R08: 0000000000000000 R09: 0000000000000000 [ 167.282715][T25615] R10: 0000000000000000 R11: 0000000000000246 R12: 00007ffcfea8e08e [ 167.283432][T25615] R13: 00007ffcfea8e08f R14: 0000000000000000 R15: 00007f0cb636a640 [ 167.284164][T25615] </TASK> [ 167.284453][T25615] ================================================================== If needed I would provide a reproduce. Regards, butt3rflyh4ck. -- Active Defense Lab of Venustech

2 3

[PATCH] ALSA: usb-audio: Register card at the last interface
by Takashi Iwai 04 Sep '22

04 Sep '22

The USB-audio driver matches per interface, and as default, it registers the card instance at the very first instance. This can be a problem for the devices that have multiple interfaces to be probed, as the udev rule isn't applied properly for the later appearing interfaces. Although we introduced the delayed_register option and the quirks for covering those shortcomings, it's nothing but a workaround for specific devices. This patch is an another attempt to fix the problem in a more generic way. Now the driver checks the whole USB device descriptor at the very first time when an interface is attached to a sound card. It looks at each matching interface in the descriptor and remembers the last matching one. The snd_card_register() is invoked only when this last interface is probed. After this change, the quirks for the delayed registration become superfluous, hence they are removed along with the patch. OTOH, the delayed_register option is still kept, as it might be useful for some corner cases (e.g. a special driver overtakes the interface probe from the standard driver, and the last interface probe may miss). Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/card.c | 32 +++++++++++++++++++++++++------- sound/usb/quirks.c | 42 ------------------------------------------ sound/usb/quirks.h | 2 -- sound/usb/usbaudio.h | 1 + 4 files changed, 26 insertions(+), 51 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 706d249a9ad6..3aea241435fb 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -690,7 +690,7 @@ static bool get_alias_id(struct usb_device *dev, unsigned int *id) return false; } -static bool check_delayed_register_option(struct snd_usb_audio *chip, int iface) +static int check_delayed_register_option(struct snd_usb_audio *chip) { int i; unsigned int id, inum; @@ -699,14 +699,31 @@ static bool check_delayed_register_option(struct snd_usb_audio *chip, int iface) if (delayed_register[i] && sscanf(delayed_register[i], "%x:%x", &id, &inum) == 2 && id == chip->usb_id) - return iface < inum; + return inum; } - return false; + return -1; } static const struct usb_device_id usb_audio_ids[]; /* defined below */ +/* look for the last interface that matches with our ids and remember it */ +static void find_last_interface(struct snd_usb_audio *chip) +{ + struct usb_host_config *config = chip->dev->actconfig; + struct usb_interface *intf; + int i; + + if (!config) + return; + for (i = 0; i < config->desc.bNumInterfaces; i++) { + intf = config->interface[i]; + if (usb_match_id(intf, usb_audio_ids)) + chip->last_iface = intf->altsetting[0].desc.bInterfaceNumber; + } + usb_audio_dbg(chip, "Found last interface = %d\n", chip->last_iface); +} + /* look for the corresponding quirk */ static const struct snd_usb_audio_quirk * get_alias_quirk(struct usb_device *dev, unsigned int id) @@ -813,6 +830,7 @@ static int usb_audio_probe(struct usb_interface *intf, err = -ENODEV; goto __error; } + find_last_interface(chip); } if (chip->num_interfaces >= MAX_CARD_INTERFACES) { @@ -862,11 +880,11 @@ static int usb_audio_probe(struct usb_interface *intf, chip->need_delayed_register = false; /* clear again */ } - /* we are allowed to call snd_card_register() many times, but first - * check to see if a device needs to skip it or do anything special + /* register card if we reach to the last interface or to the specified + * one given via option */ - if (!snd_usb_registration_quirk(chip, ifnum) && - !check_delayed_register_option(chip, ifnum)) { + if (check_delayed_register_option(chip) == ifnum || + chip->last_iface == ifnum) { err = snd_card_register(chip->card); if (err < 0) goto __error; diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 5b4d8f5eade2..8c3b0be909eb 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1728,48 +1728,6 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip, } } -/* - * registration quirk: - * the registration is skipped if a device matches with the given ID, - * unless the interface reaches to the defined one. This is for delaying - * the registration until the last known interface, so that the card and - * devices appear at the same time. - */ - -struct registration_quirk { - unsigned int usb_id; /* composed via USB_ID() */ - unsigned int interface; /* the interface to trigger register */ -}; - -#define REG_QUIRK_ENTRY(vendor, product, iface) \ - { .usb_id = USB_ID(vendor, product), .interface = (iface) } - -static const struct registration_quirk registration_quirks[] = { - REG_QUIRK_ENTRY(0x0951, 0x16d8, 2), /* Kingston HyperX AMP */ - REG_QUIRK_ENTRY(0x0951, 0x16ed, 2), /* Kingston HyperX Cloud Alpha S */ - REG_QUIRK_ENTRY(0x0951, 0x16ea, 2), /* Kingston HyperX Cloud Flight S */ - REG_QUIRK_ENTRY(0x0ecb, 0x1f46, 2), /* JBL Quantum 600 */ - REG_QUIRK_ENTRY(0x0ecb, 0x1f47, 2), /* JBL Quantum 800 */ - REG_QUIRK_ENTRY(0x0ecb, 0x1f4c, 2), /* JBL Quantum 400 */ - REG_QUIRK_ENTRY(0x0ecb, 0x2039, 2), /* JBL Quantum 400 */ - REG_QUIRK_ENTRY(0x0ecb, 0x203c, 2), /* JBL Quantum 600 */ - REG_QUIRK_ENTRY(0x0ecb, 0x203e, 2), /* JBL Quantum 800 */ - { 0 } /* terminator */ -}; - -/* return true if skipping registration */ -bool snd_usb_registration_quirk(struct snd_usb_audio *chip, int iface) -{ - const struct registration_quirk *q; - - for (q = registration_quirks; q->usb_id; q++) - if (chip->usb_id == q->usb_id) - return iface < q->interface; - - /* Register as normal */ - return false; -} - /* * driver behavior quirk flags */ diff --git a/sound/usb/quirks.h b/sound/usb/quirks.h index 31abb7cb01a5..f9bfd5ac7bab 100644 --- a/sound/usb/quirks.h +++ b/sound/usb/quirks.h @@ -48,8 +48,6 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip, struct audioformat *fp, int stream); -bool snd_usb_registration_quirk(struct snd_usb_audio *chip, int iface); - void snd_usb_init_quirk_flags(struct snd_usb_audio *chip); #endif /* __USBAUDIO_QUIRKS_H */ diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index ffbb4b0d09a0..2c6575029b1c 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -37,6 +37,7 @@ struct snd_usb_audio { unsigned int quirk_flags; unsigned int need_delayed_register:1; /* warn for delayed registration */ int num_interfaces; + int last_iface; int num_suspended_intf; int sample_rate_read_error; -- 2.35.3

1 0

[PATCH v5 0/2] lib/string_helpers: Introduce parse_int_array_user()
by Cezary Rojewski 04 Sep '22

04 Sep '22

Continuation of recent upstream discussion [1] regarding user string tokenization. First, parse_int_array_user() is introduced to allow for splitting specified user string into a sequence of integers. Makes use of get_options() internally so the parsing logic is not duplicated. With that done, redundant parts of the sound driver are removed. Originally similar functionality was added for the SOF sound driver. As more users are on the horizon, it is desirable to update existing string_helpers code and provide a unified solution. Changes in v5: - fixed kernel doc for parse_int_array_user() Changes in v4: - renamed the function to parse_int_array_user() - at the name several local variable names have been reworded to match the above Changes in v3: - relocated tokenize_user_input() implementation to string_helpers as requested by Matthew Changes in v2: - reused get_options() so no parsing logic is duplicated - simplified __user variant with help of memdup_user_nul() Both suggested by Andy, thanks for thorough review [1]: https://lore.kernel.org/alsa-devel/20220707091301.1282291-1-cezary.rojewski… Cezary Rojewski (2): lib/string_helpers: Introduce parse_int_array_user() ASoC: SOF: Remove strsplit_u32() and tokenize_input() include/linux/string_helpers.h | 2 + lib/string_helpers.c | 44 +++++++++++++ sound/soc/sof/sof-client-probes.c | 104 ++++++------------------------ 3 files changed, 64 insertions(+), 86 deletions(-) -- 2.25.1

2 6

[PATCH] ALSA: hda/sigmatel: Keep power up while beep is enabled
by Takashi Iwai 04 Sep '22

04 Sep '22

It seems that the beep playback doesn't work well on IDT codec devices when the codec auto-pm is enabled. Keep the power on while the beep switch is enabled. Link: https://bugzilla.suse.com/show_bug.cgi?id=1200544 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_sigmatel.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 61df4d33c48f..066bfccd2587 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -209,6 +209,7 @@ struct sigmatel_spec { /* beep widgets */ hda_nid_t anabeep_nid; + bool beep_power_on; /* SPDIF-out mux */ const char * const *spdif_labels; @@ -4443,6 +4444,26 @@ static int stac_suspend(struct hda_codec *codec) return 0; } + +static int stac_check_power_status(struct hda_codec *codec, hda_nid_t nid) +{ + struct sigmatel_spec *spec = codec->spec; + int ret = snd_hda_gen_check_power_status(codec, nid); + +#ifdef CONFIG_SND_HDA_INPUT_BEEP + if (nid == spec->gen.beep_nid && codec->beep) { + if (codec->beep->enabled != spec->beep_power_on) { + spec->beep_power_on = codec->beep->enabled; + if (spec->beep_power_on) + snd_hda_power_up_pm(codec); + else + snd_hda_power_down_pm(codec); + } + ret |= spec->beep_power_on; + } +#endif + return ret; +} #else #define stac_suspend NULL #endif /* CONFIG_PM */ @@ -4455,6 +4476,7 @@ static const struct hda_codec_ops stac_patch_ops = { .unsol_event = snd_hda_jack_unsol_event, #ifdef CONFIG_PM .suspend = stac_suspend, + .check_power_status = stac_check_power_status, #endif }; -- 2.35.3

1 0

[PATCH v2] ASoC: rsnd: Add check for rsnd_mod_power_on
by Jiasheng Jiang 02 Sep '22

02 Sep '22

As rsnd_mod_power_on() can return negative numbers, it should be better to check the return value and deal with the exception. Fixes: e7d850dd10f4 ("ASoC: rsnd: use mod base common method on SSI-parent") Signed-off-by: Jiasheng Jiang <jiasheng(a)iscas.ac.cn> --- Changelog: v1 -> v2: 1. Add check for rsnd_mod_power_on in other files. --- sound/soc/sh/rcar/ctu.c | 6 +++++- sound/soc/sh/rcar/dvc.c | 6 +++++- sound/soc/sh/rcar/mix.c | 6 +++++- sound/soc/sh/rcar/src.c | 5 ++++- sound/soc/sh/rcar/ssi.c | 4 +++- 5 files changed, 22 insertions(+), 5 deletions(-) diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c index 6156445bcb69..e39eb2ac7e95 100644 --- a/sound/soc/sh/rcar/ctu.c +++ b/sound/soc/sh/rcar/ctu.c @@ -171,7 +171,11 @@ static int rsnd_ctu_init(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) { - rsnd_mod_power_on(mod); + int ret; + + ret = rsnd_mod_power_on(mod); + if (ret < 0) + return ret; rsnd_ctu_activation(mod); diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index 5137e03a9d7c..16befcbc312c 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -186,7 +186,11 @@ static int rsnd_dvc_init(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) { - rsnd_mod_power_on(mod); + int ret; + + ret = rsnd_mod_power_on(mod); + if (ret < 0) + return ret; rsnd_dvc_activation(mod); diff --git a/sound/soc/sh/rcar/mix.c b/sound/soc/sh/rcar/mix.c index 3572c2c5686c..1de0e085804c 100644 --- a/sound/soc/sh/rcar/mix.c +++ b/sound/soc/sh/rcar/mix.c @@ -146,7 +146,11 @@ static int rsnd_mix_init(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) { - rsnd_mod_power_on(mod); + int ret; + + ret = rsnd_mod_power_on(mod); + if (ret < 0) + return ret; rsnd_mix_activation(mod); diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 0ea84ae57c6a..f832165e46bc 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -463,11 +463,14 @@ static int rsnd_src_init(struct rsnd_mod *mod, struct rsnd_priv *priv) { struct rsnd_src *src = rsnd_mod_to_src(mod); + int ret; /* reset sync convert_rate */ src->sync.val = 0; - rsnd_mod_power_on(mod); + ret = rsnd_mod_power_on(mod); + if (ret < 0) + return ret; rsnd_src_activation(mod); diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 43c5e27dc5c8..7ade6c5ed96f 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -480,7 +480,9 @@ static int rsnd_ssi_init(struct rsnd_mod *mod, ssi->usrcnt++; - rsnd_mod_power_on(mod); + ret = rsnd_mod_power_on(mod); + if (ret < 0) + return ret; rsnd_ssi_config_init(mod, io); -- 2.25.1

3 2

[PATCH v4 0/2] lib/string_helpers: Introduce parse_int_array_user()
by Cezary Rojewski 02 Sep '22

02 Sep '22

Continuation of recent upstream discussion [1] regarding user string tokenization. First, parse_int_array_user() is introduced to allow for splitting specified user string into a sequence of integers. Makes use of get_options() internally so the parsing logic is not duplicated. With that done, redundant parts of the sound driver are removed. Originally similar functionality was added for the SOF sound driver. As more users are on the horizon, it is desirable to update existing string_helpers code and provide a unified solution. Changes in v4: - renamed the function to parse_int_array_user() - at the name several local variable names have been reworded to match the above Changes in v3: - relocated tokenize_user_input() implementation to string_helpers as requested by Matthew Changes in v2: - reused get_options() so no parsing logic is duplicated - simplified __user variant with help of memdup_user_nul() Both suggested by Andy, thanks for thourough review [1]: https://lore.kernel.org/alsa-devel/20220707091301.1282291-1-cezary.rojewski… Cezary Rojewski (2): lib/string_helpers: Introduce parse_int_array_user() ASoC: SOF: Remove strsplit_u32() and tokenize_input() include/linux/string_helpers.h | 2 + lib/string_helpers.c | 45 +++++++++++++ sound/soc/sof/sof-client-probes.c | 104 ++++++------------------------ 3 files changed, 65 insertions(+), 86 deletions(-) -- 2.25.1

2 4

[PATCH v3 0/2] lib/string_helpers: Introduce tokenize_user_input()
by Cezary Rojewski 02 Sep '22

02 Sep '22

Continuation of recent upstream discussion [1] regarding user string tokenization. First, tokenize_user_input() is introduced to allow for splitting specified user string into a sequence of integers. Makes use of get_options() internally so the parsing logic is not duplicated. With that done, redundant parts of the sound driver are removed. Originally similar functionality was added for the SOF sound driver. As more users are on the horizon, it is desirable to update existing string_helpers code and provide a unified solution. Changes in v3: - relocated tokenize_user_input() implementation to string_helpers as requested by Matthew Changes in v2: - reused get_options() so no parsing logic is duplicated - simplified __user variant with help of memdup_user_nul() Both suggested by Andy, thanks for thourough review [1]: https://lore.kernel.org/alsa-devel/20220707091301.1282291-1-cezary.rojewski… Cezary Rojewski (2): lib/string_helpers: Introduce tokenize_user_input() ASoC: SOF: Remove strsplit_u32() and tokenize_input() include/linux/string_helpers.h | 2 + lib/string_helpers.c | 45 +++++++++++++++ sound/soc/sof/sof-client-probes.c | 93 ++++--------------------------- 3 files changed, 59 insertions(+), 81 deletions(-) -- 2.25.1

2 7

[PATCH v2 0/2] Introduce sof_of_machine_select
by Chunxu Li 02 Sep '22

02 Sep '22

From: "chunxu.li" <chunxu.li(a)mediatek.com> Change since v1: - remove the callback of_machine_select defined in sof-priv.h - move sof_of_machine_select to common code, and called in sof_machine_check - rename .board field to .compatible in structure snd_sof_of_mach In these patches, we introduce function sof_of_machine_select for SOF Chunxu Li (2): ASoC: SOF: Introduce function sof_of_machine_select ASoC: SOF: mediatek: Add sof_mt8186_machs for mt8186 include/sound/sof.h | 2 ++ sound/soc/sof/mediatek/mt8186/mt8186.c | 9 ++++++++ sound/soc/sof/pcm.c | 8 ++++++- sound/soc/sof/sof-audio.c | 30 ++++++++++++++++++++++++++ sound/soc/sof/sof-of-dev.h | 7 ++++++ 5 files changed, 55 insertions(+), 1 deletion(-) -- 2.25.1

3 5

[PATCH] ALSA: aloop: Fix random zeros in capture data when using jiffies timer
by Pattara Teerapong 02 Sep '22

02 Sep '22

In loopback_jiffies_timer_pos_update(), we are getting jiffies twice. First time for playback, second time for capture. Jiffies can be updated between these two calls and if the capture jiffies is larger, extra zeros will be filled in the capture buffer. Change to get jiffies once and use it for both playback and capture. Signed-off-by: Pattara Teerapong <pteerapong(a)chromium.org> --- sound/drivers/aloop.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 9b4a7cdb103a..12f12a294df5 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -605,17 +605,18 @@ static unsigned int loopback_jiffies_timer_pos_update cable->streams[SNDRV_PCM_STREAM_PLAYBACK]; struct loopback_pcm *dpcm_capt = cable->streams[SNDRV_PCM_STREAM_CAPTURE]; - unsigned long delta_play = 0, delta_capt = 0; + unsigned long delta_play = 0, delta_capt = 0, cur_jiffies; unsigned int running, count1, count2; + cur_jiffies = jiffies; running = cable->running ^ cable->pause; if (running & (1 << SNDRV_PCM_STREAM_PLAYBACK)) { - delta_play = jiffies - dpcm_play->last_jiffies; + delta_play = cur_jiffies - dpcm_play->last_jiffies; dpcm_play->last_jiffies += delta_play; } if (running & (1 << SNDRV_PCM_STREAM_CAPTURE)) { - delta_capt = jiffies - dpcm_capt->last_jiffies; + delta_capt = cur_jiffies - dpcm_capt->last_jiffies; dpcm_capt->last_jiffies += delta_capt; } -- 2.37.2.789.g6183377224-goog

2 1

[PATCH] ASoC: rsnd: Add check for rsnd_mod_power_on
by Jiasheng Jiang 02 Sep '22

02 Sep '22

As rsnd_mod_power_on() can return negative numbers, it should be better to check the return value and deal with the exception. Fixes: e7d850dd10f4 ("ASoC: rsnd: use mod base common method on SSI-parent") Signed-off-by: Jiasheng Jiang <jiasheng(a)iscas.ac.cn> --- sound/soc/sh/rcar/ssi.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 43c5e27dc5c8..7ade6c5ed96f 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -480,7 +480,9 @@ static int rsnd_ssi_init(struct rsnd_mod *mod, ssi->usrcnt++; - rsnd_mod_power_on(mod); + ret = rsnd_mod_power_on(mod); + if (ret < 0) + return ret; rsnd_ssi_config_init(mod, io); -- 2.25.1

2 1

[PATCH] ALSA: usb-audio: Clean up endpoint setups at PCM prepare
by Takashi Iwai 01 Sep '22

01 Sep '22

This patch cleans up the superfluous checks and calls for setting up the endpoints at PCM prepare callback: - Drop stop_endpoints() and sync_pending_stops() calls; the stream is guaranteed to have been already stopped and synced at each PCM prepare call by ALSA PCM core - Call snd_usb_endpoint_prepare() unconditionally; the check for endpoint->need_setup is done in snd_pcm_hw_endpoint_prepare() itself - Apply snd_usb_set_format_quirk() only when the endpoint is actually set up (i.e. the return code from snd_usb_endpoint_prepare() > 0) - Move a few lines back into snd_usb_pcm_prepare(); it's even easier to follow than a small useless function Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/pcm.c | 40 ++++++++++------------------------------ 1 file changed, 10 insertions(+), 30 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index b604f7e95e82..4ed53a3dc922 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -433,35 +433,6 @@ static void close_endpoints(struct snd_usb_audio *chip, } } -static int configure_endpoints(struct snd_usb_audio *chip, - struct snd_usb_substream *subs) -{ - int err; - - if (subs->data_endpoint->need_setup) { - /* stop any running stream beforehand */ - if (stop_endpoints(subs, false)) - sync_pending_stops(subs); - if (subs->sync_endpoint) { - err = snd_usb_endpoint_prepare(chip, subs->sync_endpoint); - if (err < 0) - return err; - } - err = snd_usb_endpoint_prepare(chip, subs->data_endpoint); - if (err < 0) - return err; - snd_usb_set_format_quirk(subs, subs->cur_audiofmt); - } else { - if (subs->sync_endpoint) { - err = snd_usb_endpoint_prepare(chip, subs->sync_endpoint); - if (err < 0) - return err; - } - } - - return 0; -} - /* * hw_params callback * @@ -640,9 +611,18 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) goto unlock; } - ret = configure_endpoints(chip, subs); + if (subs->sync_endpoint) { + ret = snd_usb_endpoint_prepare(chip, subs->sync_endpoint); + if (ret < 0) + goto unlock; + } + + ret = snd_usb_endpoint_prepare(chip, subs->data_endpoint); if (ret < 0) goto unlock; + else if (ret > 0) + snd_usb_set_format_quirk(subs, subs->cur_audiofmt); + ret = 0; /* reset the pointer */ subs->buffer_bytes = frames_to_bytes(runtime, runtime->buffer_size); -- 2.35.3

1 0

[PATCH 1/2] ASoC: apple: mca: Unselect COMMON_CLK in Kconfig
by Martin Povišer 01 Sep '22

01 Sep '22

The MCA driver selects COMMON_CLK, which leads to misconfiguration on platforms with HAVE_LEGACY_CLK (under compile test). Kconfig catches the conflict with the following warning: WARNING: unmet direct dependencies detected for COMMON_CLK Depends on [n]: !HAVE_LEGACY_CLK [=y] Selected by [m]: - SND_SOC_APPLE_MCA [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && (ARCH_APPLE || COMPILE_TEST [=y]) Eventually the build fails with errors like: >> drivers/clk/clk.c:867:6: error: redefinition of 'clk_unprepare' 867 | void clk_unprepare(struct clk *clk) | ^~~~~~~~~~~~~ In file included from drivers/clk/clk.c:9: include/linux/clk.h:303:20: note: previous definition of 'clk_unprepare' with type 'void(struct clk *)' 303 | static inline void clk_unprepare(struct clk *clk) | ^~~~~~~~~~~~~ which appears to be because COMMON_CLK is selected but HAVE_CLK_PREPARE is not. In the end it seems we had no business selecting COMMON_CLK from an unrelated driver like that, so remove the selection. The linux/clk.h API is there anyway. Fixes: 3df5d0d97289 ("ASoC: apple: mca: Start new platform driver") Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Martin Povišer <povik+lin(a)cutebit.org> --- sound/soc/apple/Kconfig | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/apple/Kconfig b/sound/soc/apple/Kconfig index 0ba955657e98..793f7782e0d7 100644 --- a/sound/soc/apple/Kconfig +++ b/sound/soc/apple/Kconfig @@ -2,7 +2,6 @@ config SND_SOC_APPLE_MCA tristate "Apple Silicon MCA driver" depends on ARCH_APPLE || COMPILE_TEST select SND_DMAENGINE_PCM - select COMMON_CLK default ARCH_APPLE help This option enables an ASoC platform driver for MCA peripherals found -- 2.33.0

2 2

[PATCH] ALSA: usb-audio: Split endpoint setups for hw_params and prepare
by Takashi Iwai 01 Sep '22

01 Sep '22

One of the former changes for the endpoint management was the more consistent setup of endpoints at hw_params. snd_usb_endpoint_configure() is a single function that does the full setup, and it's called from both PCM hw_params and prepare callbacks. Although the EP setup at the prepare phase is usually skipped (by checking need_setup flag), it may be still effective in some cases like suspend/resume that requires the interface setup again. As it's a full and single setup, the invocation of snd_usb_endpoint_configure() includes not only the USB interface setup but also the buffer release and allocation. OTOH, doing the buffer release and re-allocation at PCM prepare phase is rather superfluous, and better to be done only in the hw_params phase. For those optimizations, this patch splits the endpoint setup to two phases: snd_usb_endpoint_set_params() and snd_usb_endpoint_prepare(), to be called from hw_params and from prepare, respectively. Note that this patch changes the driver operation slightly, effectively moving the USB interface setup again to PCM prepare stage instead of hw_params stage, while the buffer allocation and such initializations are still done at hw_params stage. And, the change of the USB interface setup timing (moving to prepare) gave an interesting "fix", too: it was reported that the recent kernels caused silent output at the beginning on playbacks on some devices on Android, and this change casually fixed the regression. It seems that those devices are picky about the sample rate change (or the interface change?), and don't follow the too immediate rate changes. Meanwhile, Android operates the PCM in the following order: - open, then hw_params with the possibly highest sample rate - close without prepare - re-open, hw_params with the normal sample rate - prepare, and start streaming This procedure ended up the hw_params twice with different rates, and because the recent kernel did set up the sample rate twice one and after, it screwed up the device. OTOH, the earlier kernels didn't set up the USB interface at hw_params, hence this problem didn't appear. Now, with this patch, the USB interface setup is again back to the prepare phase, and it works around the problem automagically. Although we should address the sample rate problem in a more solid way in future, let's keep things working as before for now. Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management") Cc: <stable(a)vger.kernel.org> Reported-by: chihhao chen <chihhao.chen(a)mediatek.com> Link: https://lore.kernel.org/r/87e6d6ae69d68dc588ac9acc8c0f24d6188375c3.camel@me… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/endpoint.c | 23 +++++++++-------------- sound/usb/endpoint.h | 6 ++++-- sound/usb/pcm.c | 14 ++++++++++---- 3 files changed, 23 insertions(+), 20 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 0d7b73bf7945..a42f2ce19455 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -758,7 +758,8 @@ bool snd_usb_endpoint_compatible(struct snd_usb_audio *chip, * The endpoint needs to be closed via snd_usb_endpoint_close() later. * * Note that this function doesn't configure the endpoint. The substream - * needs to set it up later via snd_usb_endpoint_configure(). + * needs to set it up later via snd_usb_endpoint_set_params() and + * snd_usb_endpoint_prepare(). */ struct snd_usb_endpoint * snd_usb_endpoint_open(struct snd_usb_audio *chip, @@ -1290,12 +1291,13 @@ static int sync_ep_set_params(struct snd_usb_endpoint *ep) /* * snd_usb_endpoint_set_params: configure an snd_usb_endpoint * + * It's called either from hw_params callback. * Determine the number of URBs to be used on this endpoint. * An endpoint must be configured before it can be started. * An endpoint that is already running can not be reconfigured. */ -static int snd_usb_endpoint_set_params(struct snd_usb_audio *chip, - struct snd_usb_endpoint *ep) +int snd_usb_endpoint_set_params(struct snd_usb_audio *chip, + struct snd_usb_endpoint *ep) { const struct audioformat *fmt = ep->cur_audiofmt; int err; @@ -1378,18 +1380,18 @@ static int init_sample_rate(struct snd_usb_audio *chip, } /* - * snd_usb_endpoint_configure: Configure the endpoint + * snd_usb_endpoint_prepare: Prepare the endpoint * * This function sets up the EP to be fully usable state. - * It's called either from hw_params or prepare callback. + * It's called either from prepare callback. * The function checks need_setup flag, and performs nothing unless needed, * so it's safe to call this multiple times. * * This returns zero if unchanged, 1 if the configuration has changed, * or a negative error code. */ -int snd_usb_endpoint_configure(struct snd_usb_audio *chip, - struct snd_usb_endpoint *ep) +int snd_usb_endpoint_prepare(struct snd_usb_audio *chip, + struct snd_usb_endpoint *ep) { bool iface_first; int err = 0; @@ -1410,9 +1412,6 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip, if (err < 0) goto unlock; } - err = snd_usb_endpoint_set_params(chip, ep); - if (err < 0) - goto unlock; goto done; } @@ -1440,10 +1439,6 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip, if (err < 0) goto unlock; - err = snd_usb_endpoint_set_params(chip, ep); - if (err < 0) - goto unlock; - err = snd_usb_select_mode_quirk(chip, ep->cur_audiofmt); if (err < 0) goto unlock; diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index 6a9af04cf175..e67ea28faa54 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -17,8 +17,10 @@ snd_usb_endpoint_open(struct snd_usb_audio *chip, bool is_sync_ep); void snd_usb_endpoint_close(struct snd_usb_audio *chip, struct snd_usb_endpoint *ep); -int snd_usb_endpoint_configure(struct snd_usb_audio *chip, - struct snd_usb_endpoint *ep); +int snd_usb_endpoint_set_params(struct snd_usb_audio *chip, + struct snd_usb_endpoint *ep); +int snd_usb_endpoint_prepare(struct snd_usb_audio *chip, + struct snd_usb_endpoint *ep); int snd_usb_endpoint_get_clock_rate(struct snd_usb_audio *chip, int clock); bool snd_usb_endpoint_compatible(struct snd_usb_audio *chip, diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index d45d1d7e6664..b604f7e95e82 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -443,17 +443,17 @@ static int configure_endpoints(struct snd_usb_audio *chip, if (stop_endpoints(subs, false)) sync_pending_stops(subs); if (subs->sync_endpoint) { - err = snd_usb_endpoint_configure(chip, subs->sync_endpoint); + err = snd_usb_endpoint_prepare(chip, subs->sync_endpoint); if (err < 0) return err; } - err = snd_usb_endpoint_configure(chip, subs->data_endpoint); + err = snd_usb_endpoint_prepare(chip, subs->data_endpoint); if (err < 0) return err; snd_usb_set_format_quirk(subs, subs->cur_audiofmt); } else { if (subs->sync_endpoint) { - err = snd_usb_endpoint_configure(chip, subs->sync_endpoint); + err = snd_usb_endpoint_prepare(chip, subs->sync_endpoint); if (err < 0) return err; } @@ -551,7 +551,13 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, subs->cur_audiofmt = fmt; mutex_unlock(&chip->mutex); - ret = configure_endpoints(chip, subs); + if (subs->sync_endpoint) { + ret = snd_usb_endpoint_set_params(chip, subs->sync_endpoint); + if (ret < 0) + goto unlock; + } + + ret = snd_usb_endpoint_set_params(chip, subs->data_endpoint); unlock: if (ret < 0) -- 2.35.3

1 0

[PATCH] ALSA: asihpi - Remove useless code in hpi_meter_get_peak()
by Valentina Goncharenko 01 Sep '22

01 Sep '22

The hpi_meter_get_peak() function contains the expression "hm.obj_index = hm.obj_index", which does not carry any semantic load. Found by Linux Verification Center (linuxtesting.org) with SVACE. Fixes: 719f82d3987a ("ALSA: Add support of AudioScience ASI boards") Signed-off-by: Valentina Goncharenko <goncharenko.vp(a)ispras.ru> --- sound/pci/asihpi/hpifunc.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/asihpi/hpifunc.c b/sound/pci/asihpi/hpifunc.c index 1de05383126a..24047fafef51 100644 --- a/sound/pci/asihpi/hpifunc.c +++ b/sound/pci/asihpi/hpifunc.c @@ -2020,7 +2020,6 @@ u16 hpi_meter_get_peak(u32 h_control, short an_peakdB[HPI_MAX_CHANNELS] HPI_CONTROL_GET_STATE); if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index)) return HPI_ERROR_INVALID_HANDLE; - hm.obj_index = hm.obj_index; hm.u.c.attribute = HPI_METER_PEAK; hpi_send_recv(&hm, &hr); -- 2.25.1

2 1

[PATCH 1/2] ASoC: codecs: wcd934x: add Slimbus dependency
by Krzysztof Kozlowski 01 Sep '22

01 Sep '22

The WCD934X codec is a Slimbus driver, so it must depend on SLIMBUS, also for compile tests: ERROR: modpost: "slim_stream_prepare" [sound/soc/codecs/snd-soc-wcd934x.ko] undefined! Reported-by: kernel test robot <lkp(a)intel.com> Fixes: 5b7f4e5de61b ("ASoC: codecs: allow compile testing without MFD drivers") Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 968d0701f2e8..2fc37f4e061c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1744,6 +1744,7 @@ config SND_SOC_WCD_MBHC config SND_SOC_WCD934X tristate "WCD9340/WCD9341 Codec" depends on COMMON_CLK + depends on SLIMBUS select SND_SOC_WCD_MBHC depends on MFD_WCD934X || COMPILE_TEST help -- 2.34.1

2 2

mt8195-demo: redefine soundcard initialization
by GitHub pull_request - opened 01 Sep '22

01 Sep '22

alsa-project/alsa-ucm-conf pull request #211 was opened from fchiby: The previous initialization in Bootsequence set all Switch to off after boot. Whithout alsactl, alsa-ucm-conf does't work. So remove the setting off of the Switch in the BootSequence and define Enable/DisableSequence in each device section for a proper setting. Signed-off-by: Fadwa Chiby <fchiby(a)baylibre.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/211 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/211.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2] soundwire: qcom: fix device status array range
by Srinivas Kandagatla 01 Sep '22

01 Sep '22

This patch updates device status array range from 11 to 12 as we will be reading status from device number 0 to device number 11 inclusive. Without this patch we can potentially access status array out of range during auto-enumeration. Fixes: aa1262ca6695 ("soundwire: qcom: Check device status before reading devid") Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- drivers/soundwire/qcom.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 9df970eeca45..16a4a51978ca 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -169,7 +169,7 @@ struct qcom_swrm_ctrl { u8 wcmd_id; struct qcom_swrm_port_config pconfig[QCOM_SDW_MAX_PORTS]; struct sdw_stream_runtime *sruntime[SWRM_MAX_DAIS]; - enum sdw_slave_status status[SDW_MAX_DEVICES]; + enum sdw_slave_status status[SDW_MAX_DEVICES + 1]; int (*reg_read)(struct qcom_swrm_ctrl *ctrl, int reg, u32 *val); int (*reg_write)(struct qcom_swrm_ctrl *ctrl, int reg, int val); u32 slave_status; @@ -420,7 +420,7 @@ static int qcom_swrm_get_alert_slave_dev_num(struct qcom_swrm_ctrl *ctrl) ctrl->reg_read(ctrl, SWRM_MCP_SLV_STATUS, &val); - for (dev_num = 0; dev_num < SDW_MAX_DEVICES; dev_num++) { + for (dev_num = 0; dev_num <= SDW_MAX_DEVICES; dev_num++) { status = (val >> (dev_num * SWRM_MCP_SLV_STATUS_SZ)); if ((status & SWRM_MCP_SLV_STATUS_MASK) == SDW_SLAVE_ALERT) { @@ -440,7 +440,7 @@ static void qcom_swrm_get_device_status(struct qcom_swrm_ctrl *ctrl) ctrl->reg_read(ctrl, SWRM_MCP_SLV_STATUS, &val); ctrl->slave_status = val; - for (i = 0; i < SDW_MAX_DEVICES; i++) { + for (i = 0; i <= SDW_MAX_DEVICES; i++) { u32 s; s = (val >> (i * 2)); -- 2.25.1

2 1

[PATCH 0/6] ASoC: codecs: lpass: add support fro sm8450 and sc8280xp
by Srinivas Kandagatla 01 Sep '22

01 Sep '22

THis patchset adds support for SM8450 and SC8280XP SoC. Tested SmartSpeakers and Headset on SM8450 MTP and Lenovo Thinkpad X13s. THanks, Srini Srinivas Kandagatla (6): ASoC: qcom: dt-bindings: add sm8450 and sc8280xp compatibles ASoC: codecs: wsa-macro: add support for sm8450 and sc8280xp ASoC: codecs: tx-macro: add support for sm8450 and sc8280xp ASoC: codecs: rx-macro: add support for sm8450 and sc8280xp ASoC: codecs: va-macro: clear the frame sync counter before enabling ASoC: codecs: tx-macro: add support for sm8450 and sc8280xp .../bindings/sound/qcom,lpass-rx-macro.yaml | 2 + .../bindings/sound/qcom,lpass-tx-macro.yaml | 2 + .../bindings/sound/qcom,lpass-va-macro.yaml | 2 + .../bindings/sound/qcom,lpass-wsa-macro.yaml | 2 + sound/soc/codecs/lpass-rx-macro.c | 2 + sound/soc/codecs/lpass-tx-macro.c | 2 + sound/soc/codecs/lpass-va-macro.c | 72 +++++++++++++++++-- sound/soc/codecs/lpass-wsa-macro.c | 2 + 8 files changed, 82 insertions(+), 4 deletions(-) -- 2.21.0

3 15

[PATCH] sound: soc/codecs: tlv320adc3xxx: needs GPIOLIB
by Randy Dunlap 31 Aug '22

31 Aug '22

Fix build errors when CONFIG_GPIOLIB is not enabled: ../sound/soc/codecs/tlv320adc3xxx.c: In function 'adc3xxx_i2c_probe': ../sound/soc/codecs/tlv320adc3xxx.c:1352:28: error: implicit declaration of function 'devm_gpiod_get'; did you mean 'devm_gpio_free'? [-Werror=implicit-function-declaration] 1352 | adc3xxx->rst_pin = devm_gpiod_get(dev, "reset", GPIOD_OUT_LOW); ../sound/soc/codecs/tlv320adc3xxx.c:1352:57: error: 'GPIOD_OUT_LOW' undeclared (first use in this function); did you mean 'GPIOF_INIT_LOW'? 1352 | adc3xxx->rst_pin = devm_gpiod_get(dev, "reset", GPIOD_OUT_LOW); CC lib/dynamic_debug.o ../sound/soc/codecs/tlv320adc3xxx.c:1400:9: error: implicit declaration of function 'gpiod_set_value_cansleep'; did you mean 'gpio_set_value_cansleep'? [-Werror=implicit-function-declaration] 1400 | gpiod_set_value_cansleep(adc3xxx->rst_pin, 1); Fixes: e9a3b57efd28 ("ASoC: codec: tlv320adc3xxx: New codec driver") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Reported-by: kernel test robot <lkp(a)intel.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Ricard Wanderlof <ricardw(a)axis.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1567,6 +1567,7 @@ config SND_SOC_TFA989X config SND_SOC_TLV320ADC3XXX tristate "Texas Instruments TLV320ADC3001/3101 audio ADC" depends on I2C + depends on GPIOLIB help Enable support for Texas Instruments TLV320ADC3001 and TLV320ADC3101 ADCs.

1 1

Support MSI MPG Z690 EDGE WIFI DDR4 (ALC4080)
by GitHub issues - edited 31 Aug '22

31 Aug '22

alsa-project/alsa-ucm-conf issue #210 was edited from tekstryder: Per @perexg request, filing a new issue after testing git master branch changes for the ALC4080 and finding any profile(s) to be missing/nonfunctional. USB ID: `0db0:1feb` In version 1.2.7.2 of .../ucm2/USB-Audio/Realtek/ALC4080-HiFi.conf I need to change: - Define.SPDIFIndex "3" to Define.SPDIFIndex "2" For proper playback via S/PDIF. After that change the release version is working great as far as I could test. Please let me know if I can provide any other useful information. Attaching the output of `alsaucm -c hw:2 dump text` Attaching the output of `pactl list sinks` [ucm-dump.txt](https://github.com/alsa-project/alsa-ucm-conf/files/9463734/u… [sinks.txt](https://github.com/alsa-project/alsa-ucm-conf/files/9463784/sink… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/210 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Support MSI MPG Z690 EDGE WIFI DDR4 (ALC4080)
by GitHub issues - opened 31 Aug '22

31 Aug '22

alsa-project/alsa-ucm-conf issue #210 was opened from tekstryder: Per @perexg request, filing a new issue after testing git master branch changes for the ALC4080 and finding any profile(s) to be missing/nonfunctional. USB ID: `0db0:1feb` In version 1.2.7.2 of .../ucm2/USB-Audio/Realtek/ALC4080-HiFi.conf I need to change: - Define.SPDIFIndex "3" to Define.SPDIFIndex "Define.SPDIFIndex "2"" For proper playback via S/PDIF. After that change the release version is working great as far as I could test. Please let me know if I can provide any other useful information. Attaching the output of `alsaucm -c hw:2 dump text` Attaching the output of `pactl list sinks` [ucm-dump.txt](https://github.com/alsa-project/alsa-ucm-conf/files/9463734/u… [sinks.txt](https://github.com/alsa-project/alsa-ucm-conf/files/9463784/sink… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/210 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2 0/2] libfs: Introduce tokenize_user_input()
by Cezary Rojewski 31 Aug '22

31 Aug '22

Continuation of recent upstream discussion [1] regarding user string tokenization. First, tokenize_user_input() is introduced to allow for splitting specified user string into a sequence of integers. Makes use of get_options() internally so the parsing logic is not duplicated. With that done, redundant parts of the sound driver are removed. Originally similar functionality was added for the SOF sound driver. As more users are on the horizon, it is desirable to update existing fs code and provide a unified solution. Changes in v2: - reused get_options() so no parsing logic is duplicated - simplified __user variant with help of memdup_user_nul() Both suggested by Andy, thanks for thourough review [1]: https://lore.kernel.org/alsa-devel/20220707091301.1282291-1-cezary.rojewski… Cezary Rojewski (2): libfs: Introduce tokenize_user_input() ASoC: SOF: Remove strsplit_u32() and tokenize_input() fs/libfs.c | 45 +++++++++++++++ include/linux/fs.h | 1 + sound/soc/sof/sof-client-probes.c | 92 ++++--------------------------- 3 files changed, 57 insertions(+), 81 deletions(-) -- 2.25.1

2 4

[PATCH] ALSA: usb-audio: Drop superfluous interface setup at parsing
by Takashi Iwai 31 Aug '22

31 Aug '22

We reset each interface that is being parsed for each stream, but this is superfluous and even can lead to spurious errors. Since the interface is set up properly at opening the endpoint for each actual stream operation, let's drop the superfluous one. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/stream.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/usb/stream.c b/sound/usb/stream.c index ceb93d798182..99578e9a8af0 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -1221,12 +1221,6 @@ static int __snd_usb_parse_audio_interface(struct snd_usb_audio *chip, if (err < 0) return err; } - - /* try to set the interface... */ - usb_set_interface(chip->dev, iface_no, 0); - snd_usb_init_pitch(chip, fp); - snd_usb_init_sample_rate(chip, fp, fp->rate_max); - usb_set_interface(chip->dev, iface_no, altno); } return 0; } -- 2.35.3

1 0

[PATCH 1/2] ALSA: usb-audio: Inform the delayed registration more properly
by Takashi Iwai 31 Aug '22

31 Aug '22

The info message that was added in the commit a4aad5636c72 ("ALSA: usb-audio: Inform devices that need delayed registration") is actually useful to know the need for the delayed registration. However, it turned out that this doesn't catch the all cases; namely, this warned only when a PCM stream is attached onto the existing PCM instance, but it doesn't count for a newly created PCM instance. This made confusion as if there were no further delayed registration. This patch moves the check to the code path for either adding a stream or creating a PCM instance. Also, make it simpler by checking the card->registered flag instead of querying each snd_device state. Fixes: a4aad5636c72 ("ALSA: usb-audio: Inform devices that need delayed registration") Link: https://bugzilla.kernel.org/show_bug.cgi?id=216082 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/stream.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/usb/stream.c b/sound/usb/stream.c index ceb93d798182..40b7821c6c99 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -495,6 +495,10 @@ static int __snd_usb_add_audio_stream(struct snd_usb_audio *chip, return 0; } } + + if (chip->card->registered) + chip->need_delayed_register = true; + /* look for an empty stream */ list_for_each_entry(as, &chip->pcm_list, list) { if (as->fmt_type != fp->fmt_type) @@ -502,9 +506,6 @@ static int __snd_usb_add_audio_stream(struct snd_usb_audio *chip, subs = &as->substream[stream]; if (subs->ep_num) continue; - if (snd_device_get_state(chip->card, as->pcm) != - SNDRV_DEV_BUILD) - chip->need_delayed_register = true; err = snd_pcm_new_stream(as->pcm, stream, 1); if (err < 0) return err; -- 2.35.3

1 1

[PATCH v3 1/2] ASoC: dt-bindings: max98396: Document data monitor properties
by Daniel Mack 31 Aug '22

31 Aug '22

This device features a data monitor that puts the device in software reset upon a configurable set of events. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- v1 -> v2: fix a typo and remove a stray blank line v2 -> v3: Rename device tree properties, drop $ref for standard unit suffix properties .../bindings/sound/adi,max98396.yaml | 34 +++++++++++++++++++ 1 file changed, 34 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/adi,max98396.yaml b/Documentation/devicetree/bindings/sound/adi,max98396.yaml index 8d2ef991db40..7985b1410017 100644 --- a/Documentation/devicetree/bindings/sound/adi,max98396.yaml +++ b/Documentation/devicetree/bindings/sound/adi,max98396.yaml @@ -78,6 +78,40 @@ properties: interleaved on a single output channel. type: boolean + adi,dmon-stuck-enable: + description: + Enables the "data monitor stuck" feature. Once the data monitor is + enabled, it actively monitors the selected input data (from DIN) to the + speaker amplifier. Once a data error is detected, the data monitor + automatically places the device into software shutdown. + type: boolean + + adi,dmon-stuck-threshold-bits: + description: + Sets the threshold for the "data monitor stuck" feature, in bits. + enum: [9, 11, 13, 15] + default: 15 + + adi,dmon-magnitude-enable: + description: + Enables the "data monitor magnitude" feature. Once the data monitor is + enabled, it actively monitors the selected input data (from DIN) to the + speaker amplifier. Once a data error is detected, the data monitor + automatically places the device into software shutdown. + type: boolean + + adi,dmon-magnitude-threshold-bits: + description: + Sets the threshold for the "data monitor magnitude" feature, in bits. + enum: [2, 3, 4, 5] + default: 5 + + adi,dmon-duration-ms: + description: + Sets the duration for the "data monitor" feature, in milliseconds. + enum: [64, 256, 1024, 4096] + default: 64 + reset-gpios: maxItems: 1 -- 2.37.2

3 3

[PATCH 0/5] ASoC: codecs: minor cppcheck cleanups
by Pierre-Louis Bossart 31 Aug '22

31 Aug '22

Small number of cleanups that were either missed in previous versions or detected by new cppcheck version. Pierre-Louis Bossart (5): ASoC: hdmi-codec: remove unused definitions ASoC: wcd-mbhc-v2: remove always-true condition ASoC: wcd9335: remove always-true condition ASoC: fsl: fsl-utils: remove useless assignment ASoC: ti: omap-mcbsp: remove useless assignment sound/soc/codecs/hdmi-codec.c | 4 ---- sound/soc/codecs/wcd-mbhc-v2.c | 2 +- sound/soc/codecs/wcd9335.c | 4 +--- sound/soc/fsl/fsl_utils.c | 2 +- sound/soc/ti/omap-mcbsp.c | 2 +- 5 files changed, 4 insertions(+), 10 deletions(-) -- 2.34.1

3 7

[PATCH 0/2] ASoC: SOF: compress: Add support for timestamp on capture
by Laurentiu Mihalcea 31 Aug '22

31 Aug '22

The purpose of this patch series is to add support for timestamping on capture direction using the compress API. This is simply done by splitting sof_compr_copy into 2 functions: sof_compr_copy_playback and sof_compr_copy_capture. Each of these functions handles one of the possible directions: capture or playback and is called in sof_compr_copy based on the stream's direction. The only difference between sof_compr_copy_playback and sof_compr_copy_capture is the fact that on playback case we need to copy data from user space and on capture we need to copy data to user space. Laurentiu Mihalcea (2): ASoC: SOF: compress: Move sof_compr_copy functionality ASoC: SOF: compress: Add copy function for capture case sound/soc/sof/compress.c | 47 +++++++++++++++++++++++++++++++++------- 1 file changed, 39 insertions(+), 8 deletions(-) -- 2.34.1

2 3

[PATCH v2] ASoC: amd: acp: Modify dai_id macros to be more generic
by Venkata Prasad Potturu 31 Aug '22

31 Aug '22

Change dai_id macros to make I2S instances in order. Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- Changes since v1: -- Change commit description. sound/soc/amd/acp/amd.h | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/amd/acp/amd.h b/sound/soc/amd/acp/amd.h index af9603724a68..60a7c95f134f 100644 --- a/sound/soc/amd/acp/amd.h +++ b/sound/soc/amd/acp/amd.h @@ -21,9 +21,9 @@ #define ACP3X_DEV 3 #define ACP6X_DEV 6 -#define I2S_SP_INSTANCE 0x00 -#define I2S_BT_INSTANCE 0x01 -#define DMIC_INSTANCE 0x02 +#define DMIC_INSTANCE 0x00 +#define I2S_SP_INSTANCE 0x01 +#define I2S_BT_INSTANCE 0x02 #define I2S_HS_INSTANCE 0x03 #define MEM_WINDOW_START 0x4080000 -- 2.25.1

2 1

[PATCH 0/6] ASoC: qcom: add support for sc8280xp machine
by Srinivas Kandagatla 31 Aug '22

31 Aug '22

THis patchset adds support for SC8280XP SoC machine driver. First patch moves some of the commonly used code to common from sm8250 machine driver and the follow on code adds minimal support for sc8280xp. Currently this driver is only tested with SmartSpeakers and Headset on Lenovo Thinkpad X13s. Support for sm8450 is tested and I will post the patches soon. Thanks, Srini Srinivas Kandagatla (3): ASoC: dt-bindings: qcom,sm8250: add compatibles for sm8450 and sm8250 ASoC: qcom: sm8250: move some code to common ASoC: qcom: add machine driver for sc8280xp .../bindings/sound/qcom,sm8250.yaml | 2 + sound/soc/qcom/Kconfig | 11 ++ sound/soc/qcom/Makefile | 2 + sound/soc/qcom/common.c | 169 ++++++++++++++++++ sound/soc/qcom/common.h | 12 ++ sound/soc/qcom/sc8280xp.c | 157 ++++++++++++++++ sound/soc/qcom/sm8250.c | 152 +--------------- 7 files changed, 360 insertions(+), 145 deletions(-) create mode 100644 sound/soc/qcom/sc8280xp.c -- 2.21.0

3 6

How does PlaybackChannelPos# Work?
by GitHub issues - opened 31 Aug '22

31 Aug '22

alsa-project/alsa-ucm-conf issue #209 was opened from 13r0ck: I found https://github.com/alsa-project/alsa-lib/commit/0ba999591b78e07a4b8e610ebce… but doesn't seem to work when setting audio jack output. I have a mostly working configuration for a motherboard's 3.5mm audio jack output. The only problem is that the C/Sub and Rear jacks are flipped. I am assuming setting the `PlaybackChannelPos` in the speaker should work, but it doesn't for some reason. ``` SectionDevice."Speaker" { Commant "Speakers" Value { PlaybackChannel2 "BL" PlaybackChannel3 "BR" PlaybackChannel4 "FC" PlaybackChannel5 "LFE" } } ``` Any help would be greatly appreciated! Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/209 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: amd: fix spelling mistake: "i.e" -> "i.e."
by Syed Saba Kareem 30 Aug '22

30 Aug '22

From: Syed Saba kareem <Syed.SabaKareem(a)amd.com> trivial fix to spelling mistake in Kconfig File. Reported by : Randy Dunlap <rdunlap(a)infradead.org> Signed-off-by: Syed Saba Kareem <Syed.SabaKareem(a)amd.com> --- sound/soc/amd/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 68837d42736d..150786279257 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -122,7 +122,7 @@ config SND_SOC_AMD_RPL_ACP6x tristate "AMD Audio Coprocessor-v6.2 RPL support" depends on X86 && PCI help - This option enables Audio Coprocessor i.e ACP v6.2 support on + This option enables Audio Coprocessor i.e. ACP v6.2 support on AMD RPL platform. By enabling this flag build will be triggered for ACP PCI driver. Say m if you have such a device. -- 2.25.1

3 2

[PATCH 1/3] slimbus: qcom-ngd: use correct error in message of pdr_add_lookup() failure
by Krzysztof Kozlowski 30 Aug '22

30 Aug '22

Use cprrect error code, instead of previous 'ret' value, when printing error from pdr_add_lookup() failure. Cc: <stable(a)vger.kernel.org> Fixes: e1ae85e1830e ("slimbus: qcom-ngd-ctrl: add Protection Domain Restart Support") Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- drivers/slimbus/qcom-ngd-ctrl.c | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index 0aa8408464ad..6fe6abb86061 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -1581,6 +1581,7 @@ static int qcom_slim_ngd_ctrl_probe(struct platform_device *pdev) pds = pdr_add_lookup(ctrl->pdr, "avs/audio", "msm/adsp/audio_pd"); if (IS_ERR(pds) && PTR_ERR(pds) != -EALREADY) { + ret = PTR_ERR(pds); dev_err(dev, "pdr add lookup failed: %d\n", ret); return PTR_ERR(pds); } -- 2.34.1

2 4

[PATCH] ASoC: codecs: allow compile testing without MFD drivers
by Krzysztof Kozlowski 30 Aug '22

30 Aug '22

Motorola CPCAP, Lochnagar Sound, Rockchip RK817 and Qualcomm WCD9340/WCD9341 do not depend on parent MFD driver in build time and can be compile tested without respective MFD part for increased build coverage. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/codecs/Kconfig | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 5926b33ba09e..94b7bb85d236 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -610,7 +610,7 @@ config SND_SOC_BT_SCO config SND_SOC_CPCAP tristate "Motorola CPCAP codec" - depends on MFD_CPCAP + depends on MFD_CPCAP || COMPILE_TEST config SND_SOC_CQ0093VC tristate @@ -972,7 +972,7 @@ config SND_SOC_LM49453 config SND_SOC_LOCHNAGAR_SC tristate "Lochnagar Sound Card" - depends on MFD_LOCHNAGAR + depends on MFD_LOCHNAGAR || COMPILE_TEST help This driver support the sound card functionality of the Cirrus Logic Lochnagar audio development board. @@ -1197,7 +1197,7 @@ config SND_SOC_RK3328 config SND_SOC_RK817 tristate "Rockchip RK817 audio CODEC" - depends on MFD_RK808 + depends on MFD_RK808 || COMPILE_TEST select REGMAP_I2C config SND_SOC_RL6231 @@ -1745,7 +1745,7 @@ config SND_SOC_WCD934X tristate "WCD9340/WCD9341 Codec" depends on COMMON_CLK select SND_SOC_WCD_MBHC - depends on MFD_WCD934X + depends on MFD_WCD934X || COMPILE_TEST help The WCD9340/9341 is a audio codec IC Integrated in Qualcomm SoCs like SDM845. -- 2.34.1

2 5

[PATCH] ASoC: codecs: rk817: fix missing I2C dependency in compile test
by Krzysztof Kozlowski 30 Aug '22

30 Aug '22

SND_SOC_RK817 uses I2C regmap so compile testing without parent MFD_RK808, requires I2C: WARNING: unmet direct dependencies detected for REGMAP_I2C Depends on [n]: I2C [=n] Selected by [y]: - SND_SOC_RK817 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && (MFD_RK808 [=n] || COMPILE_TEST [=y]) Reported-by: kernel test robot <lkp(a)intel.com> Fixes: 5b7f4e5de61b ("ASoC: codecs: allow compile testing without MFD drivers") Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Commit SHA from linux-next (recent). --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 94b7bb85d236..968d0701f2e8 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1197,7 +1197,7 @@ config SND_SOC_RK3328 config SND_SOC_RK817 tristate "Rockchip RK817 audio CODEC" - depends on MFD_RK808 || COMPILE_TEST + depends on MFD_RK808 || COMPILE_TEST && I2C select REGMAP_I2C config SND_SOC_RL6231 -- 2.34.1

2 1

[PATCH v2 00/13] Add Pink Sardine platform ASoC driver
by Syed Saba Kareem 30 Aug '22

30 Aug '22

Pink Sardine platform is new APU series based on acp6.2 design. This patch set adds an ASoC driver for the ACP (Audio CoProcessor) block on AMD Pink Sardine APU with DMIC endpoint support. changes since v1: -Removed subraction logic in acp62.h header file and changed the address in acp62_chip_offset_byte.h file. -Fixed indentation in pci driver file. -Removed comments which states function names in pci driver file. -Replaced some hex values with decimals in pci driver file. -Corrected MODULE_DESCRIPTION in pdm driver file. -Removed structure variable which is used for byte count in header file and added new local variable for denoting the byte count. -Added Kfree for freeing the runtime->private data. -Replaced X86 && PCI with X86 && PCI && ACPI in Kconfig file. Syed Saba Kareem (13): ASoC: amd: add Pink Sardine platform ACP IP register header ASoC: amd: add Pink Sardine ACP PCI driver ASoC: amd: add acp6.2 init/de-init functions ASoC: amd: add platform devices for acp6.2 pdm driver and dmic driver ASoC: amd: add acp6.2 pdm platform driver ASoC: amd: add acp6.2 irq handler ASoC: amd: add acp6.2 pdm driver dma ops ASoC: amd: add acp6.2 pci driver pm ops ASoC: amd: add acp6.2 pdm driver pm ops ASoC: amd: enable Pink Sardine acp6.2 drivers build ASoC: amd: create platform device for acp6.2 machine driver ASoC: amd: add Pink Sardine machine driver using dmic ASoC: amd: enable Pink sardine platform machine driver build. include/sound/acp62_chip_offset_byte.h | 444 ++++++++++++++++++++++++ sound/soc/amd/Kconfig | 21 ++ sound/soc/amd/Makefile | 1 + sound/soc/amd/ps/Makefile | 9 + sound/soc/amd/ps/acp62.h | 98 ++++++ sound/soc/amd/ps/pci-ps.c | 351 +++++++++++++++++++ sound/soc/amd/ps/ps-mach.c | 79 +++++ sound/soc/amd/ps/ps-pdm-dma.c | 452 +++++++++++++++++++++++++ 8 files changed, 1455 insertions(+) create mode 100644 include/sound/acp62_chip_offset_byte.h create mode 100644 sound/soc/amd/ps/Makefile create mode 100644 sound/soc/amd/ps/acp62.h create mode 100644 sound/soc/amd/ps/pci-ps.c create mode 100644 sound/soc/amd/ps/ps-mach.c create mode 100644 sound/soc/amd/ps/ps-pdm-dma.c -- 2.25.1

2 14

[PATCH 0/5] Create a new sound card to access MICFIL based on rpmsg channel
by Chancel Liu 30 Aug '22

30 Aug '22

At a previous time, we have successfully created a virtual sound card based on rpmsg. The sound card works under this mechanism Cortex-A core tells the Cortex-M core the format, rate, channel, .etc configuration of the PCM parameters and Cortex-M controls real hardware devices such as SAI and DMA. From the view of Linux side, the sound card is bound to a rpmsg channel through which it can access SAI. Here these patches are introduced to create a new virtual sound card to access MICFIL based on a new created rpmsg channel. It's easy to create a new rpmsg channel for MICFIL through rpmsg name service announcment. Also the other ASoC components bound to this rpmsg MICFIL sound card will be registered with these patches. If other sound cards using different hardware devices needs to be created over rpmsg in the future, these patches can be referred. Chancel Liu (5): ASoC: dt-bindings: fsl_rpmsg: Add a property to assign platform driver name ASoC: imx-audio-rpmsg: Create rpmsg channel for MICFIL ASoC: imx-pcm-rpmsg: Register different platform drivers ASoC: fsl_rpmsg: Register different CPU DAI drivers ASoC: imx-rpmsg: Assign platform driver used by machine driver to link with .../devicetree/bindings/sound/fsl,rpmsg.yaml | 34 +++++++++++++++++-- sound/soc/fsl/fsl_rpmsg.c | 2 +- sound/soc/fsl/imx-audio-rpmsg.c | 3 +- sound/soc/fsl/imx-pcm-rpmsg.c | 10 ++++-- sound/soc/fsl/imx-rpmsg.c | 6 +++- 5 files changed, 47 insertions(+), 8 deletions(-) -- 2.25.1

3 8

[PATCH 0/1] soundwire: bus: Fix for missing UNATTACH when re-enumerating
by Richard Fitzgerald 30 Aug '22

30 Aug '22

This is an alternative fix for the race where a device that is reset can be found on #0 and its ID reprogrammed before the bus code has handled a PING showing it unattached. The other possible fix is here: https://lore.kernel.org/all/20220825122241.273090-4-rf@opensource.cirrus.co… Richard Fitzgerald (1): soundwire: bus: Don't re-enumerate before status is UNATTACHED drivers/soundwire/bus.c | 7 +++++++ 1 file changed, 7 insertions(+) -- 2.30.2

1 2

[PATCH 0/3] soundwire: Fixes for spurious and missing UNATTACH
by Richard Fitzgerald 30 Aug '22

30 Aug '22

The bus and cadence code has several bugs that cause UNATTACH notifications to either be sent spuriously or to be missed. These can be seen occasionally with a single peripheral on the bus, but are much more frequent with multiple peripherals, where several peripherals could change state and report in consecutive PINGs. The root of all of these bugs seems to be a code design flaw that assumed every PING status change would be handled separately. However, PINGs are handled by a workqueue function and there is no guarantee when that function will be scheduled to run or how much CPU time it will receive. PINGs will continue while the work function is handling a snapshot of a previous PING so the code must take account that (a) status could change during the work function and (b) there can be a backlog of changes before the IRQ work function runs again. Richard Fitzgerald (2): soundwire: bus: Don't lose unattach notifications soundwire: bus: Fix lost UNATTACH when re-enumerating Simon Trimmer (1): soundwire: cadence: fix updating slave status when a bus has multiple peripherals drivers/soundwire/bus.c | 44 +++++++++++++++------ drivers/soundwire/cadence_master.c | 63 +++++++++++++----------------- 2 files changed, 59 insertions(+), 48 deletions(-) -- 2.30.2

2 11

[PATCH][next] ALSA: hda/hdmi: Replace zero-length array with DECLARE_FLEX_ARRAY() helper
by Gustavo A. R. Silva 30 Aug '22

30 Aug '22

Zero-length arrays are deprecated and we are moving towards adopting C99 flexible-array members, instead. So, replace zero-length array declaration in union audio_infoframe with the new DECLARE_FLEX_ARRAY() helper macro. This helper allows for a flexible-array member in a union. Link: https://github.com/KSPP/linux/issues/193 Link: https://gcc.gnu.org/onlinedocs/gcc/Zero-Length.html Signed-off-by: Gustavo A. R. Silva <gustavoars(a)kernel.org> --- sound/pci/hda/patch_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 6c209cd26c0c..2191d445d74e 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -229,7 +229,7 @@ struct dp_audio_infoframe { union audio_infoframe { struct hdmi_audio_infoframe hdmi; struct dp_audio_infoframe dp; - u8 bytes[0]; + DECLARE_FLEX_ARRAY(u8, bytes); }; /* -- 2.34.1

3 2

[PATCH 00/11] soundwire: register cleanups
by Bard Liao 30 Aug '22

30 Aug '22

Clean-up and reorder/regroup SoundWire SHIM registers. Pierre-Louis Bossart (11): soundwire: intel: cleanup definition of LCOUNT soundwire: intel: regroup definitions for LCTL soundwire: intel: remove IPPTR unused definition soundwire: intel: cleanup SHIM SYNC soundwire: intel: remove unused PDM capabilities soundwire: intel: add comment for control stream cap/chmap soundwire: intel: cleanup PCM stream capabilities soundwire: intel: cleanup PCM Stream channel map and channel count soundwire: intel: cleanup IO control soundwire: intel: cleanup AC Timing Control soundwire: intel: cleanup WakeEnable and WakeStatus drivers/soundwire/intel.c | 5 +-- drivers/soundwire/intel_init.c | 2 +- include/linux/soundwire/sdw_intel.h | 63 +++++++++++++++++------------ 3 files changed, 41 insertions(+), 29 deletions(-) -- 2.25.1

2 12

[PATCH 0/3] soundwire: allocate device_number with IDA
by Bard Liao 30 Aug '22

30 Aug '22

The device_number is currently allocated in the scope of each bus and does not need to be unique at the system level. This leads e.g. on Dell devices with three or four Device1 on different bus segments. To make the device_number unique at the system level, and unified with the HDaudio/iDISP SDI values, this series allocates the dev_number with an IDA restricted between 4 and 11 (inclusive). Pierre-Louis Bossart (3): soundwire: bus: rename sdw_ida as sdw_bus_ida soundwire: bus: allow device number to be unique at system level soundwire: intel: set dev_num_ida_min drivers/soundwire/bus.c | 29 ++++++++++++++++++++--------- drivers/soundwire/intel.c | 4 ++++ include/linux/soundwire/sdw.h | 4 ++++ 3 files changed, 28 insertions(+), 9 deletions(-) -- 2.25.1

3 6

[PATCH] dt-bindings: iio: Add missing (unevaluated|additional)Properties on child nodes
by Rob Herring 29 Aug '22

29 Aug '22

In order to ensure only documented properties are present, node schemas must have unevaluatedProperties or additionalProperties set to false (typically). Signed-off-by: Rob Herring <robh(a)kernel.org> --- Documentation/devicetree/bindings/iio/adc/qcom,spmi-vadc.yaml | 1 + .../devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml | 2 ++ Documentation/devicetree/bindings/iio/adc/ti,ads1015.yaml | 1 + .../devicetree/bindings/iio/adc/xlnx,zynqmp-ams.yaml | 1 + Documentation/devicetree/bindings/iio/addac/adi,ad74413r.yaml | 1 + Documentation/devicetree/bindings/iio/dac/adi,ad3552r.yaml | 1 + Documentation/devicetree/bindings/iio/dac/adi,ad5770r.yaml | 4 ++++ Documentation/devicetree/bindings/iio/dac/adi,ltc2688.yaml | 1 + 8 files changed, 12 insertions(+) diff --git a/Documentation/devicetree/bindings/iio/adc/qcom,spmi-vadc.yaml b/Documentation/devicetree/bindings/iio/adc/qcom,spmi-vadc.yaml index 74a4a9d95798..8bac0c4120dd 100644 --- a/Documentation/devicetree/bindings/iio/adc/qcom,spmi-vadc.yaml +++ b/Documentation/devicetree/bindings/iio/adc/qcom,spmi-vadc.yaml @@ -58,6 +58,7 @@ required: patternProperties: "^.*@[0-9a-f]+$": type: object + additionalProperties: false description: | Represents the external channels which are connected to the ADC. For compatible property "qcom,spmi-vadc" following channels, also known as diff --git a/Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml b/Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml index 92f9472a77ae..1970503389aa 100644 --- a/Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml +++ b/Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml @@ -72,6 +72,7 @@ additionalProperties: false patternProperties: "^filter@[0-9]+$": type: object + unevaluatedProperties: false description: child node properties: @@ -225,6 +226,7 @@ patternProperties: patternProperties: "^dfsdm-dai+$": type: object + additionalProperties: false description: child node properties: diff --git a/Documentation/devicetree/bindings/iio/adc/ti,ads1015.yaml b/Documentation/devicetree/bindings/iio/adc/ti,ads1015.yaml index a3b79438a13a..2c3c2cf2145c 100644 --- a/Documentation/devicetree/bindings/iio/adc/ti,ads1015.yaml +++ b/Documentation/devicetree/bindings/iio/adc/ti,ads1015.yaml @@ -43,6 +43,7 @@ additionalProperties: false patternProperties: "^channel@[0-7]+$": type: object + additionalProperties: false description: Child nodes needed for each channel that the platform uses. diff --git a/Documentation/devicetree/bindings/iio/adc/xlnx,zynqmp-ams.yaml b/Documentation/devicetree/bindings/iio/adc/xlnx,zynqmp-ams.yaml index 3698b4b0900f..be93c109d6ac 100644 --- a/Documentation/devicetree/bindings/iio/adc/xlnx,zynqmp-ams.yaml +++ b/Documentation/devicetree/bindings/iio/adc/xlnx,zynqmp-ams.yaml @@ -134,6 +134,7 @@ properties: ams-pl@400: type: object + additionalProperties: false description: PL-SYSMON is capable of monitoring off chip voltage and temperature. PL-SYSMON block has DRP, JTAG and I2C interface to enable monitoring diff --git a/Documentation/devicetree/bindings/iio/addac/adi,ad74413r.yaml b/Documentation/devicetree/bindings/iio/addac/adi,ad74413r.yaml index baa65a521bad..03bb90a7f4f8 100644 --- a/Documentation/devicetree/bindings/iio/addac/adi,ad74413r.yaml +++ b/Documentation/devicetree/bindings/iio/addac/adi,ad74413r.yaml @@ -63,6 +63,7 @@ additionalProperties: false patternProperties: "^channel@[0-3]$": type: object + additionalProperties: false description: Represents the external channels which are connected to the device. properties: diff --git a/Documentation/devicetree/bindings/iio/dac/adi,ad3552r.yaml b/Documentation/devicetree/bindings/iio/dac/adi,ad3552r.yaml index 9c48c76993fe..fee0f023a8c8 100644 --- a/Documentation/devicetree/bindings/iio/dac/adi,ad3552r.yaml +++ b/Documentation/devicetree/bindings/iio/dac/adi,ad3552r.yaml @@ -78,6 +78,7 @@ patternProperties: custom-output-range-config: type: object + additionalProperties: false description: Configuration of custom range when adi,output-range-microvolt is not present. The formulas for calculation the output voltages are diff --git a/Documentation/devicetree/bindings/iio/dac/adi,ad5770r.yaml b/Documentation/devicetree/bindings/iio/dac/adi,ad5770r.yaml index 24ac40180ac1..c2d9ae3039b2 100644 --- a/Documentation/devicetree/bindings/iio/dac/adi,ad5770r.yaml +++ b/Documentation/devicetree/bindings/iio/dac/adi,ad5770r.yaml @@ -62,6 +62,7 @@ properties: connected to the DAC. Channel 0 can act both as a current source and sink. type: object + additionalProperties: false properties: reg: @@ -85,6 +86,7 @@ properties: description: Represents an external channel which are connected to the DAC. type: object + additionalProperties: false properties: reg: @@ -101,6 +103,7 @@ properties: description: Represents an external channel which are connected to the DAC. type: object + additionalProperties: false properties: reg: @@ -116,6 +119,7 @@ properties: patternProperties: "^channel@([3-5])$": type: object + additionalProperties: false description: Represents the external channels which are connected to the DAC. properties: reg: diff --git a/Documentation/devicetree/bindings/iio/dac/adi,ltc2688.yaml b/Documentation/devicetree/bindings/iio/dac/adi,ltc2688.yaml index 48f9e7d29423..15cc6bf59b13 100644 --- a/Documentation/devicetree/bindings/iio/dac/adi,ltc2688.yaml +++ b/Documentation/devicetree/bindings/iio/dac/adi,ltc2688.yaml @@ -48,6 +48,7 @@ properties: patternProperties: "^channel@([0-9]|1[0-5])$": type: object + additionalProperties: false properties: reg: -- 2.34.1

3 4

[PATCH linux-next] ASoC: codecs: max98088: remove redundant ret variable
by cgel.zte＠gmail.com 29 Aug '22

29 Aug '22

From: Jinpeng Cui <cui.jinpeng2(a)zte.com.cn> Return value from devm_snd_soc_register_component() directly instead of taking this in another redundant variable. Reported-by: Zeal Robot <zealci(a)zte.com.cn> Signed-off-by: Jinpeng Cui <cui.jinpeng2(a)zte.com.cn> --- sound/soc/codecs/max98088.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index b208fb530a8b..405ec16be2b6 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1749,7 +1749,6 @@ MODULE_DEVICE_TABLE(i2c, max98088_i2c_id); static int max98088_i2c_probe(struct i2c_client *i2c) { struct max98088_priv *max98088; - int ret; const struct i2c_device_id *id; max98088 = devm_kzalloc(&i2c->dev, sizeof(struct max98088_priv), @@ -1772,9 +1771,8 @@ static int max98088_i2c_probe(struct i2c_client *i2c) i2c_set_clientdata(i2c, max98088); max98088->pdata = i2c->dev.platform_data; - ret = devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_max98088, + return devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_max98088, &max98088_dai[0], 2); - return ret; } #if defined(CONFIG_OF) -- 2.25.1

2 1

[PATCH 0/5] TAS2764 fixes/extensions
by Martin Povišer 29 Aug '22

29 Aug '22

First three patches are fixes analogical to those recently done to the TAS2770 driver. Link: https://lore.kernel.org/asahi/20220808141246.5749-1-povik+lin@cutebit.org/T… The latter two add IRQ handler to log faults and expose a new control. Martin Povišer (5): ASoC: tas2764: Allow mono streams ASoC: tas2764: Drop conflicting set_bias_level power setting ASoC: tas2764: Fix mute/unmute ASoC: tas2764: Add IRQ handling ASoC: tas2764: Export highpass filter setting sound/soc/codecs/tas2764.c | 181 ++++++++++++++++++++++++++----------- sound/soc/codecs/tas2764.h | 23 +++++ 2 files changed, 152 insertions(+), 52 deletions(-) -- 2.33.0

2 6

[RESEND v2] ASoC: amd: acp: Modify dai_id macros to be more generic
by Venkata Prasad Potturu 29 Aug '22

29 Aug '22

Change dai_id macros to make I2S instances in order. Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- sound/soc/amd/acp/amd.h | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/amd/acp/amd.h b/sound/soc/amd/acp/amd.h index af9603724a68..60a7c95f134f 100644 --- a/sound/soc/amd/acp/amd.h +++ b/sound/soc/amd/acp/amd.h @@ -21,9 +21,9 @@ #define ACP3X_DEV 3 #define ACP6X_DEV 6 -#define I2S_SP_INSTANCE 0x00 -#define I2S_BT_INSTANCE 0x01 -#define DMIC_INSTANCE 0x02 +#define DMIC_INSTANCE 0x00 +#define I2S_SP_INSTANCE 0x01 +#define I2S_BT_INSTANCE 0x02 #define I2S_HS_INSTANCE 0x03 #define MEM_WINDOW_START 0x4080000 -- 2.25.1

2 1

[PATCH -next 0/2] ASoC: Cleanup deprecated regmap-irq functionality
by Aidan MacDonald 28 Aug '22

28 Aug '22

Update two ASoC codec drivers to remove uses of regmap-irq type registers, which have recently been deprecated by the "regmap-irq cleanups and refactoring" series in linux-next. Link: https://lore.kernel.org/lkml/20220623211420.918875-1-aidanmacdonald.0x0@gma… Aidan MacDonald (2): sound: soc: codecs: wcd9335: Convert irq chip to config regs sound: soc: codecs: wcd938x: Remove spurious type_base from irq chip sound/soc/codecs/wcd9335.c | 10 ++++++++-- sound/soc/codecs/wcd938x.c | 1 - 2 files changed, 8 insertions(+), 3 deletions(-) base-commit: 353f7988dd8413c47718f7ca79c030b6fb62cfe5 prerequisite-patch-id: a45db1cac7537769dc10087fc61f977dc150744c prerequisite-patch-id: cb8aa009c7bb7a6575eb05e3af65342dc8d0efa3 prerequisite-patch-id: 8d32557e53b894d1fb17250d2d0eb3673f068d37 prerequisite-patch-id: 5b293867ef81e3697892ac51b941bb53680a70dc prerequisite-patch-id: 8138d39a4817e804141bfe8c2ad37d9c55456a40 prerequisite-patch-id: b01216129e887519d441cf556bbc75c397871773 prerequisite-patch-id: b5dcf0c0609113c2d81bc557c1fc95ef23f40811 prerequisite-patch-id: 622ca1c10e851b1889aaa567c1d2a0adf43cca44 prerequisite-patch-id: d28c5187f9bf0e43f27b2f4aa8dcf7fd91842a03 prerequisite-patch-id: d29d54b756be73304f844abeeaf9b46a5c0119d5 prerequisite-patch-id: 5d405790ae89c0831b46a359f86e94bac5a67470 prerequisite-patch-id: c90120e79acbb52ffa148bfedee1df9d35b5eced prerequisite-patch-id: 2e35247a5cfe5a28565c9272b85fc6835011b032 -- 2.35.1

2 4

[PATCH] ASoC: soc-pcm.c: call __soc_pcm_close() in soc_pcm_close()
by Kuninori Morimoto 28 Aug '22

28 Aug '22

commit b7898396f4bbe16 ("ASoC: soc-pcm: Fix and cleanup DPCM locking") added __soc_pcm_close() for non-lock version of soc_pcm_close(). But soc_pcm_close() is not using it. It is no problem, but confusable. static int __soc_pcm_close(...) { => return soc_pcm_clean(rtd, substream, 0); } static int soc_pcm_close(...) { ... snd_soc_dpcm_mutex_lock(rtd); => soc_pcm_clean(rtd, substream, 0); snd_soc_dpcm_mutex_unlock(rtd); return 0; } This patch use it. Fixes: b7898396f4bbe16 ("ASoC: soc-pcm: Fix and cleanup DPCM locking") Cc: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 5b99bf2dbd08..3ca073edf7e7 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -723,7 +723,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); snd_soc_dpcm_mutex_lock(rtd); - soc_pcm_clean(rtd, substream, 0); + __soc_pcm_close(rtd, substream); snd_soc_dpcm_mutex_unlock(rtd); return 0; } -- 2.25.1

2 1

ucm2: HDA - add support for Internal Mic
by GitHub pull_request - edited 28 Aug '22

28 Aug '22

alsa-project/alsa-ucm-conf pull request #207 was edited from kasper93: Simple patch to support also "Internal Mic". Everything seems to be working fine after this change, except that I don't have "Internal Mic Jack", I'm lying there a little to set it as such. Everything switches nicely when connecting/disconnecting headphones mic. But there are two input devices to select and selection doesn't have effect. When headphone mic is connected the internal one should be hidden (it is handled somehow when using HDA driver), but there is no Jack and I don't know how to handle it properly. (It is marked as conflicting device with the other one though) Except this minor annoyance it is fine. For reference: [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/9439697/… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/207 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/207.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: HDA - add support for Internal Mic
by GitHub pull_request - edited 28 Aug '22

28 Aug '22

alsa-project/alsa-ucm-conf pull request #207 was edited from kasper93: Simple patch to support also "Internal Mic". Everything seems to be working fine after this change, except that I don't have "Internal Mic Jack", I'm lying there a little to set it as such. Everything switches nicely when connecting/disconnecting headphones mic. But there are two input devices to select and selection doesn't have effect. When headphone mic is connected the internal one should be hidden, but there is no Jack and I don't know how to handle it properly. (It is marked as conflicting device with the other one though) Except this minor annoyance it is fine. For reference: [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/9439697/… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/207 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/207.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: HDA - add support for Internal Mic
by GitHub pull_request - opened 28 Aug '22

28 Aug '22

alsa-project/alsa-ucm-conf pull request #207 was opened from kasper93: Simple patch to support also "Internal Mic". Everything seems to be working fine after this change, except that I don't have "Internal Mic Jack", I'm lying there a little to set it as such. Everything switches nicely when connecting/disconnecting headphones mic. But there are two input devices to select and selection doesn't have effect. I believe when headphone mic is connected the internal one should be hidden, but there is no Jack and I don't know how to handle it properly. (It is marked as conflicting device with the other one though) Except this minor annoyance it is fine. For reference: [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/9439697/… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/207 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/207.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ALSA: usb-audio: Add quirk for LH Labs Geek Out HD Audio 1V5
by Takashi Iwai 28 Aug '22

28 Aug '22

The USB DAC from LH Labs (2522:0007) seems requiring the same quirk as Sony Walkman to set up the interface like UAC1; otherwise it gets the constant errors "usb_set_interface failed (-71)". This patch adds a quirk entry for addressing the buggy behavior. Reported-by: Lennert Van Alboom <lennert(a)vanalboom.org> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/T3VPXtCc4uFws9Gfh2RjX6OdwM1RqfC6VqQr--_LMDyB2x5N3… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 168fd802d70b..9bfead5efc4c 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1903,6 +1903,8 @@ 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(0x2522, 0x0007, /* LH Labs Geek Out HD Audio 1V5 */ + QUIRK_FLAG_SET_IFACE_FIRST), DEVICE_FLG(0x2708, 0x0002, /* Audient iD14 */ QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x2912, 0x30c8, /* Audioengine D1 */ -- 2.35.3

1 0

[PATCH] ALSA: hda/realtek: Add speaker AMP init for Samsung laptops with ALC298
by Kacper Michajłow 28 Aug '22

28 Aug '22

Magic initialization sequence was extracted from Windows driver and cleaned up manually. Fixes internal speakers output. Link: https://bugzilla.kernel.org/show_bug.cgi?id=207423 Link: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1851518 Cc: Takashi Iwai <tiwai(a)suse.com> Signed-off-by: Kacper Michajłow <kasper93(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 63 +++++++++++++++++++++++++++++++---- 1 file changed, 56 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 47e72cf76608..38930cf5aace 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4700,6 +4700,48 @@ static void alc236_fixup_hp_mute_led_micmute_vref(struct hda_codec *codec, alc236_fixup_hp_micmute_led_vref(codec, fix, action); } +static inline void alc298_samsung_write_coef_pack(struct hda_codec *codec, + const unsigned short coefs[2]) +{ + alc_write_coef_idx(codec, 0x23, coefs[0]); + alc_write_coef_idx(codec, 0x25, coefs[1]); + alc_write_coef_idx(codec, 0x26, 0xb011); +} + +struct alc298_samsung_amp_desc { + unsigned char nid; + unsigned short init_seq[2][2]; +}; + +static void alc298_fixup_samsung_amp(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + int i, j; + static const unsigned short init_seq[][2] = { + { 0x19, 0x00 }, { 0x20, 0xc0 }, { 0x22, 0x44 }, { 0x23, 0x08 }, + { 0x24, 0x85 }, { 0x25, 0x41 }, { 0x35, 0x40 }, { 0x36, 0x01 }, + { 0x38, 0x81 }, { 0x3a, 0x03 }, { 0x3b, 0x81 }, { 0x40, 0x3e }, + { 0x41, 0x07 }, { 0x400, 0x1 } + }; + static const struct alc298_samsung_amp_desc amps[] = { + { 0x3a, { { 0x18, 0x1 }, { 0x26, 0x0 } } }, + { 0x39, { { 0x18, 0x2 }, { 0x26, 0x1 } } } + }; + + if (action != HDA_FIXUP_ACT_INIT) + return; + + for (i = 0; i < ARRAY_SIZE(amps); i++) { + alc_write_coef_idx(codec, 0x22, amps[i].nid); + + for (j = 0; j < ARRAY_SIZE(amps[i].init_seq); j++) + alc298_samsung_write_coef_pack(codec, amps[i].init_seq[j]); + + for (j = 0; j < ARRAY_SIZE(init_seq); j++) + alc298_samsung_write_coef_pack(codec, init_seq[j]); + } +} + #if IS_REACHABLE(CONFIG_INPUT) static void gpio2_mic_hotkey_event(struct hda_codec *codec, struct hda_jack_callback *event) @@ -7030,6 +7072,7 @@ enum { ALC236_FIXUP_HP_GPIO_LED, ALC236_FIXUP_HP_MUTE_LED, ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF, + ALC298_FIXUP_SAMSUNG_AMP, ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, ALC256_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, ALC295_FIXUP_ASUS_MIC_NO_PRESENCE, @@ -8396,6 +8439,12 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc236_fixup_hp_mute_led_micmute_vref, }, + [ALC298_FIXUP_SAMSUNG_AMP] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc298_fixup_samsung_amp, + .chained = true, + .chain_id = ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET + }, [ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -9342,13 +9391,13 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x10ec, 0x1254, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-SZ6", ALC269_FIXUP_HEADSET_MODE), SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC), - SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), - SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), - SND_PCI_QUIRK(0x144d, 0xc189, "Samsung Galaxy Flex Book (NT950QCG-X716)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), - SND_PCI_QUIRK(0x144d, 0xc18a, "Samsung Galaxy Book Ion (NP930XCJ-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), + SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_AMP), + SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_AMP), + SND_PCI_QUIRK(0x144d, 0xc189, "Samsung Galaxy Flex Book (NT950QCG-X716)", ALC298_FIXUP_SAMSUNG_AMP), + SND_PCI_QUIRK(0x144d, 0xc18a, "Samsung Galaxy Book Ion (NP930XCJ-K01US)", ALC298_FIXUP_SAMSUNG_AMP), SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8), - SND_PCI_QUIRK(0x144d, 0xc812, "Samsung Notebook Pen S (NT950SBE-X58)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), - SND_PCI_QUIRK(0x144d, 0xc830, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), + SND_PCI_QUIRK(0x144d, 0xc812, "Samsung Notebook Pen S (NT950SBE-X58)", ALC298_FIXUP_SAMSUNG_AMP), + SND_PCI_QUIRK(0x144d, 0xc830, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_AMP), SND_PCI_QUIRK(0x144d, 0xc832, "Samsung Galaxy Book Flex Alpha (NP730QCJ)", ALC256_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), @@ -9716,7 +9765,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC299_FIXUP_PREDATOR_SPK, .name = "predator-spk"}, {.id = ALC298_FIXUP_HUAWEI_MBX_STEREO, .name = "huawei-mbx-stereo"}, {.id = ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, .name = "alc256-medion-headset"}, - {.id = ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, .name = "alc298-samsung-headphone"}, + {.id = ALC298_FIXUP_SAMSUNG_AMP, .name = "alc298-samsung-amp"}, {.id = ALC256_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, .name = "alc256-samsung-headphone"}, {.id = ALC255_FIXUP_XIAOMI_HEADSET_MIC, .name = "alc255-xiaomi-headset"}, {.id = ALC274_FIXUP_HP_MIC, .name = "alc274-hp-mic-detect"}, -- 2.37.2

2 1

USB DAC broken since commit bf6313a0ff766925462e97b4e733d5952de02367 (5.10.0-rc5)
by Lennert Van Alboom 27 Aug '22

27 Aug '22

Hi, My USB DAC fails to work on recent kernels (anything past 5.10). The device is the following: Bus 003 Device 005: ID 2522:0007 LH Labs Geek Out HD Audio 1V5 Syslog shows error messages which have changed over the different kernel versions - this is the output from debian's 5.17.0-2-amd64: [66136.185642] usb 3-6: uac_clock_source_is_valid(): cannot get clock validity for id 41 [66136.185643] usb 3-6: clock source 41 is not valid, cannot use [66136.185956] usb 3-6: 1:0: usb_set_interface failed (-71) [66136.186183] usb 3-6: 1:0: usb_set_interface failed (-71) >From the first point in git where it fails (5.10.0-rc5-00025-gbf6313a0ff76): Aug 25 12:39:37 Nesbitt kernel: [ 5295.633079] usb 3-3: new high-speed USB device number 8 using xhci_hcd Aug 25 12:39:37 Nesbitt kernel: [ 5295.782144] usb 3-3: New USB device found, idVendor=2522, idProduct=0007, bcdDevice=15.02 Aug 25 12:39:37 Nesbitt kernel: [ 5295.782148] usb 3-3: New USB device strings: Mfr=1, Product=3, SerialNumber=0 Aug 25 12:39:37 Nesbitt kernel: [ 5295.782150] usb 3-3: Product: Geek Out HD Audio 1V5 Aug 25 12:39:37 Nesbitt kernel: [ 5295.782151] usb 3-3: Manufacturer: LH Labs Aug 25 12:39:37 Nesbitt kernel: [ 5296.162338] usb 3-3: 1:2 : unsupported format bits 0x100000000 Aug 25 12:39:38 Nesbitt mtp-probe: checking bus 3, device 8: "/sys/devices/pci0000:00/0000:00:14.0/usb3/3-3" Aug 25 12:39:38 Nesbitt mtp-probe: bus: 3, device: 8 was not an MTP device Aug 25 12:39:38 Nesbitt systemd-udevd[10311]: controlC1: Process '/usr/sbin/alsactl -E HOME=/run/alsa -E XDG_RUNTIME_DIR=/run/alsa/runtime restore 1' failed with exit code 99. Aug 25 12:39:38 Nesbitt mtp-probe: checking bus 3, device 8: "/sys/devices/pci0000:00/0000:00:14.0/usb3/3-3" Aug 25 12:39:38 Nesbitt mtp-probe: bus: 3, device: 8 was not an MTP device Aug 25 12:39:43 Nesbitt kernel: [ 5301.921810] usb 3-3: 1:1: usb_set_interface failed (-110) Aug 25 12:39:43 Nesbitt kernel: [ 5301.922047] usb 3-3: 1:0: usb_set_interface failed (-71) Aug 25 12:39:43 Nesbitt kernel: [ 5301.922354] usb 3-3: 1:0: usb_set_interface failed (-71) Aug 25 12:39:43 Nesbitt kernel: [ 5301.922915] usb 3-3: 1:0: usb_set_interface failed (-71) I did a git bisect (from kernel source, not on debian-specific) and the offending commit has been found: # git bisect bad bf6313a0ff766925462e97b4e733d5952de02367 is the first bad commit commit bf6313a0ff766925462e97b4e733d5952de02367 Author: Takashi Iwai <tiwai(a)suse.de> Date: Mon Nov 23 09:53:31 2020 +0100 ALSA: usb-audio: Refactor endpoint management This is an intensive surgery for the endpoint and stream management for achieving more robust and clean code. The goals of this patch are: - More clear endpoint resource changes - The interface altsetting control in a single place Below are brief description of the whole changes. First off, most of the endpoint operations are moved into endpoint.c, so that the snd_usb_endpoint object is only referred in other places. The endpoint object is acquired and released via the new functions snd_usb_endpoint_open() and snd_usb_endpoint_close() that are called at PCM hw_params and hw_free callbacks, respectively. Those are ref-counted and EPs can manage the multiple opens. The open callback receives the audioformat and hw_params arguments, and those are used for initializing the EP parameters; especially the endpoint, interface and altset numbers are read from there, as well as the PCM parameters like the format, rate and channels. Those are stored in snd_usb_endpoint object. If it's the secondary open, the function checks whether the given parameters are compatible with the already opened EP setup, too. The coupling with a sync EP (including an implicit feedback sync) is done by the sole snd_usb_endpoint_set_sync() call. The configuration of each endpoint is done in a single shot via snd_usb_endpoint_configure() call. This is the place where most of PCM configurations are done. A few flags and special handling in the snd_usb_substream are dropped along with this change. A significant difference wrt the configuration from the previous code is the order of USB host interface setups. Now the interface is always disabled at beginning and (re-)enabled at the last step of snd_usb_endpoint_configure(), in order to be compliant with the standard UAC2/3. For UAC1, the interface is set before the parameter setups since there seem devices that require it (e.g. Yamaha THR10), just like how it was done in the previous driver code. The start/stop are almost same as before, also single-shots. The URB callbacks need to be set via snd_usb_endpoint_set_callback() like the previous code at the trigger phase, too. Finally, the flag for the re-setup is set at the device suspend through the full EP list, instead of PCM trigger. This catches the overlooked cases where the PCM hasn't been running yet but the device needs the full setup after resume. Tested-by: Keith Milner <kamilner(a)superlative.org> Tested-by: Dylan Robinson <dylan_robinson(a)motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-26-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> sound/usb/card.c | 8 +- sound/usb/card.h | 11 +- sound/usb/clock.c | 13 +- sound/usb/endpoint.c | 662 ++++++++++++++++++++++++++------------------------- sound/usb/endpoint.h | 40 ++-- sound/usb/pcm.c | 616 +++++++++++++++++++---------------------------- 6 files changed, 616 insertions(+), 734 deletions(-) That's a lot of code though. I'd like to be able to use my DAC on kernels newer than 5.10 so any hints on how to debug and fix this would be greatly appreciated. After doing this 17-point bisect I absolutely don't mind building a few more to fix this. For info, the debian bug report: https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=1011196 For info, a related report from Arch linux for the same device vendor: https://bugs.archlinux.org/task/70636 Thanks, Lennert

2 6

Re: [PATCH 10/13] ASoC: amd: enable Pink Sardine acp6.2 drivers build
by Syed Saba Kareem 27 Aug '22

27 Aug '22

On 8/12/22 19:35, Mark Brown wrote: > [CAUTION: External Email] ACP Driver has strict dependency on ACPI COMPILE_TEST will relax the flag options. As ACP PCI driver has strict dependency,compile test flag cannot be used in this case.

1 0

Tascam US122L, JACK, PulseAudio and OBS
by GitHub issues - opened 27 Aug '22

27 Aug '22

alsa-project/alsa-lib issue #269 was opened from stephematician: Apologies if this is the wrong place - however I'm getting an error most of the time in a very specific situation. My setup is a TASCAM US122L USB interface, which ALSA is capable of recognising and setting up as a usb_stream device. I then run jackd for this device, load the pulseaudio module for a JACK source (at this point I can record via parecord), and then attempt to add a PulseAudio input to OBS. At this point, I'll get an input for less than a second, then it'll cut out and I'll get an error in JACK. I've attached the output of alsa-info; showing that the device is recognised. The contents of my .asoundrc are somewhat widely used: ``` # The usb_stream plugin configuration pcm.!usb_stream { @args [ CARD ] @args.CARD { type string default "1" } type usb_stream card $CARD } ctl.!usb_stream { @args [ CARD ] @args.CARD { type string default "1" } type hw card $CARD } ``` I run jackd with a command like: ``` /usr/bin/jackd -v -p128 -dalsa -dusb_stream:CARD=US122L -r44100 -p256 -n2 -C ``` I then set up a source in PulseAudio using `pacmd load-module module-jack-source`. Typically at this point, `parecord --device=jack_in test.wav` records no problems. Then I'll run OBS and try to add a PulseAudio input, at which point I usually get a very short period of input (less than 1 second?) and then the input will cut out and I'll see the following error in jackd output: ``` Couldn't configure usb_stream : Input/output error ALSA: prepare error for capture on "usb_stream:CARD=US122L" (Input/output error) ``` Upon tracking this down, it seems that some call to snd_pcm_prepare by JACK is receiving an error code; but I'm not an expert in this library (or alsa) so it's hard for me to even guess what the problem could be. [alsa-info.txt](https://github.com/alsa-project/alsa-lib/files/9438112/alsa-… I have also raised this issue on JACK in case it's a problem with that library (link here - but details are the same https://github.com/jackaudio/jack1/issues/118#issue-1353053534) Issue URL : https://github.com/alsa-project/alsa-lib/issues/269 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Request for setup of new repositories
by Takashi Sakamoto 27 Aug '22

27 Aug '22

Hi Jaroslav, Iwai-san, Thanks for your maintenance for alsa-project organization in github.com. Currently I'd like to add new three repositories under the organization as a part of my work for ALSA firewire stack. I've been maintaining libhinawa since 2014 and recently realized that current design is not necessarily convenient since it includes two functions; operation to Linux FireWire cdev, and operation of ALSA HwDep cdev. Currently I'm working for new library to split the latter operation. Then I'd like you to setup below repositories: * 'libhitaki' * 'libhitaki-doc' * 'hitaki-rs' The library itself and its Rust binding are utilized by 'snd-firewire-ctl-services'[2], thus it's preferable to register them under 'GObject Introspection' Team. Thanks for your assist for my work. [1] https://github.com/alsa-project/libhinawa [2] https://github.com/alsa-project/snd-firewire-ctl-services Regards Takashi Sakamoto

2 20

USB-Audio: Fix for ALC4080 (ASUS ROG Strix B660-F Gaming WiFi)
by GitHub pull_request - edited 27 Aug '22

27 Aug '22

alsa-project/alsa-ucm-conf pull request #206 was edited from RomanAverin: Fix for ALC4080 (ASUS ROG Strix B660-F Gaming WiFi) Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/206 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/206.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Fix for ALC4080 (ASUS ROG Strix B660-F Gaming WiFi)
by GitHub pull_request - edited 27 Aug '22

27 Aug '22

alsa-project/alsa-ucm-conf pull request #206 was edited from RomanAverin: Fix for ALC4080 (ASUS ROG Strix B660-F Gaming WiFi) Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/206 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/206.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Update USB-Audio.conf
by GitHub pull_request - opened 27 Aug '22

27 Aug '22

alsa-project/alsa-ucm-conf pull request #206 was opened from RomanAverin: Fix for ALC4080 (ASUS ROG Strix B660-F Gaming WiFi) Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/206 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/206.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Asus ROG Strix B660-F (ALC4080) output issue: No audio front headphone, No audio S/PDIF
by GitHub issues - opened 27 Aug '22

27 Aug '22

alsa-project/alsa-ucm-conf issue #205 was opened from RomanAverin: I have problems playing audio from the front headphone jack, as well as S/PDIF. Kernel-5.19.4, Fedora 36, X11, Gnome 42 1. usb-device detect as `Bus 001 Device 004: ID 0b05:1a16 ASUSTek Computer, Inc. USB Audio` 2. Sound cards device by ` arecord -l ` ``` **** List of CAPTURE Hardware Devices **** card 0: Audio [USB Audio], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Audio [USB Audio], device 1: USB Audio [USB Audio #1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Audio [USB Audio], device 2: USB Audio [USB Audio #2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 3: J250a [Jabra UC VOICE 250a], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` 3. Info from `alsa-info.sh` uploaded [http://alsa-project.org/db/?f=4376afd5f9282787f5ce07b8dd473e48639b6201](htt… 4. Trying to change the device id in the `/usr/share/alsa/ucm2/USB-Audio/Realtek/ALC4080.conf` on my id `Regex "USB(0b05:1a16)"` (default 0b05:1996) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/205 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

MPG X570S EDGE MAX WIFI compatibility
by GitHub issues - edited 26 Aug '22

26 Aug '22

alsa-project/alsa-ucm-conf issue #172 was edited from yhojann-cl: Please, add `0db0:151f` to `realtek-alc4080` chip compatibility in MPG X570S EDGE MAX WIFI motherboard in https://github.com/alsa-project/alsa-ucm-conf/blob/master/ucm2/USB-Audio/US… : ``` $ lsusb | grep 0db0 Bus 001 Device 003: ID 0db0:151f Micro Star International USB Audio ``` In `ucm2/USB-Audio/USB-Audio.conf` add `151f` to regex of `0db0`: ``` Regex "USB((0b05:(1996|1a2[07]))|(0db0:(1feb|419c|a073|151f)))" ``` My Motherboard is MSI X570S and output audio works fine but: - Motherboard microphone plug is not work. - Frontal microphone plug is not work. - When change output from gnome settings to hdmi audio does not work, the audio continues to play in the plug and not in the monitor. - When I change the input or output from digital to analog or sinput it does not change at all. - The strangest thing is that the microphone input captures the audio that is heard in the system as if it were the output, audacious records the audio from the desktop when it tries to monitor the microphone. - When plugging or unplugging the microphone, the system detects this change and tries to change the audio input automatically, but it does not capture the microphone and the desktop continues to be heard. - When plug headphone in frontal plug the gnome setting change automaticaly to the headphone item, but the sound play in speakers, not in headphone. - On Audacity when capture Alsa + `USB Audio: #2 (hw: 1,2)` works fine!, but can not set on gnome settings. My alsa data: http://alsa-project.org/db/?f=e29a6593e48350efeba75f69d5ece62379973c35 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/172 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

MPG X570S EDGE MAX WIFI compatibility
by GitHub issues - edited 26 Aug '22

26 Aug '22

alsa-project/alsa-ucm-conf issue #172 was edited from yhojann-cl: Please, add `0db0:151f` to `realtek-alc4080` chip compatibility in MPG X570S EDGE MAX WIFI motherboard in https://github.com/alsa-project/alsa-ucm-conf/blob/master/ucm2/USB-Audio/US… : ``` $ lsusb | grep 0db0 Bus 001 Device 003: ID 0db0:151f Micro Star International USB Audio ``` In `ucm2/USB-Audio/USB-Audio.conf` add `151f` to regex of `0db0`: ``` Regex "USB((0b05:(1996|1a2[07]))|(0db0:(1feb|419c|a073|151f)))" ``` My Motherboard is MSI X570S and output audio works fine but: - Motherboard microphone plug is not work. - Frontal microphone plug is not work. - When change output from gnome settings to hdmi audio does not work, the audio continues to play in the plug and not in the monitor. - When I change the input or output from digital to analog or sinput it does not change at all. - The strangest thing is that the microphone input captures the audio that is heard in the system as if it were the output, audacious records the audio from the desktop when it tries to monitor the microphone. - When plugging or unplugging the microphone, the system detects this change and tries to change the audio input automatically, but it does not capture the microphone and the desktop continues to be heard. - When plug headphone in frontal plug the gnome setting change automaticaly to the headphone item, but the sound play in speakers, not in headphone. - On Audacity when capture Alsa + `USB Audio: #2 (hw: 1,2)` works fine!, but can not set on gnome settings. My alsa data: http://alsa-project.org/db/?f=df4d1f6c126260eabc5f159e64cff37efa064a38 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/172 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

MPG X570S EDGE MAX WIFI compatibility
by GitHub issues - edited 26 Aug '22

26 Aug '22

alsa-project/alsa-ucm-conf issue #172 was edited from yhojann-cl: Please, add `0db0:151f` to `realtek-alc4080` chip compatibility in MPG X570S EDGE MAX WIFI motherboard: ``` $ lsusb | grep 0db0 Bus 001 Device 003: ID 0db0:151f Micro Star International USB Audio ``` In `ucm2/USB-Audio/USB-Audio.conf` add `151f` to regex of `0db0`: ``` Regex "USB((0b05:(1996|1a2[07]))|(0db0:(1feb|419c|a073|151f)))" ``` My Motherboard is MSI X570S and output audio works fine but: - Motherboard microphone plug is not work. - Frontal microphone plug is not work. - When change output from gnome settings to hdmi audio does not work, the audio continues to play in the plug and not in the monitor. - When I change the input or output from digital to analog or sinput it does not change at all. - The strangest thing is that the microphone input captures the audio that is heard in the system as if it were the output, audacious records the audio from the desktop when it tries to monitor the microphone. - When plugging or unplugging the microphone, the system detects this change and tries to change the audio input automatically, but it does not capture the microphone and the desktop continues to be heard. - When plug headphone in frontal plug the gnome setting change automaticaly to the headphone item, but the sound play in speakers, not in headphone. - On Audacity when capture Alsa + `USB Audio: #2 (hw: 1,2)` works fine!, but can not set on gnome settings. My alsa data: http://alsa-project.org/db/?f=df4d1f6c126260eabc5f159e64cff37efa064a38 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/172 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

MPG X570S EDGE MAX WIFI compatibility
by GitHub issues - edited 26 Aug '22

26 Aug '22

alsa-project/alsa-ucm-conf issue #172 was edited from yhojann-cl: Please, add `0db0:151f` to `realtek-alc4080` chip compatibility in MPG X570S EDGE MAX WIFI motherboard: ``` $ lsusb | grep 0db0 Bus 001 Device 003: ID 0db0:151f Micro Star International USB Audio ``` In `ucm2/USB-Audio/USB-Audio.conf` add `151f` to regex of `0db0`: ``` Regex "USB((0b05:(1996|1a2[07]))|(0db0:(1feb|419c|a073|151f)))" ``` My Motherboard is MSI X570S and output audio works fine but: - Motherboard microphone plug is not work. - Frontal microphone plug is not work. - When change output from gnome settings to hdmi audio does not work, the audio continues to play in the plug and not in the monitor. - When I change the input or output from digital to analog or sinput it does not change at all. - The strangest thing is that the microphone input captures the audio that is heard in the system as if it were the output, audacious records the audio from the desktop when it tries to monitor the microphone. - When plugging or unplugging the microphone, the system detects this change and tries to change the audio input automatically, but it does not capture the microphone and the desktop continues to be heard. - When plug headphone in frontal plug the gnome setting change automaticaly to the headphone item, but the sound play in speakers, not in headphone. - On Audacity when capture Alsa + "USB Audio: #2 (hw: 1,2)" works fine!, but can not set on gnome settings. My alsa data: http://alsa-project.org/db/?f=df4d1f6c126260eabc5f159e64cff37efa064a38 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/172 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: Fix misspelling of 'system' as 'ststem'.
by Maurycy Z 26 Aug '22

26 Aug '22

Signed-off-by: Maurycy Zalewski <10maurycy10(a)gmail.com> --- sound/pci/ice1712/ice1712.c | 2 +- sound/pci/ice1712/ice1724.c | 2 +- sound/soc/fsl/Kconfig | 2 +- sound/soc/generic/Kconfig | 4 ++-- 4 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index a5241a287..ff6ce91bf 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2247,7 +2247,7 @@ static int snd_ice1712_read_eeprom(struct snd_ice1712 *ice, (snd_ice1712_read_i2c(ice, dev, 0x03) << 24); if (ice->eeprom.subvendor == 0 || ice->eeprom.subvendor == (unsigned int)-1) { - /* invalid subvendor from EEPROM, try the PCI subststem ID instead */ + /* invalid subvendor from EEPROM, try the PCI subsystem ID instead */ u16 vendor, device; pci_read_config_word(ice->pci, PCI_SUBSYSTEM_VENDOR_ID, &vendor); pci_read_config_word(ice->pci, PCI_SUBSYSTEM_ID, &device); diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 6fab2ad85..9e81e6335 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -2266,7 +2266,7 @@ static int snd_vt1724_read_eeprom(struct snd_ice1712 *ice, if (ice->eeprom.subvendor == 0 || ice->eeprom.subvendor == (unsigned int)-1) { /* invalid subvendor from EEPROM, try the PCI - * subststem ID instead + * subsystem ID instead */ u16 vendor, device; pci_read_config_word(ice->pci, PCI_SUBSYSTEM_VENDOR_ID, diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 614eceda6..c7fe12fbd 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -366,7 +366,7 @@ config SND_SOC_IMX_CARD help This option enables audio sound card support for i.MX boards with OF-graph DT bindings. - It also support DPCM of single CPU multi Codec ststem. + It also support DPCM of single CPU multi Codec system. endif # SND_IMX_SOC diff --git a/sound/soc/generic/Kconfig b/sound/soc/generic/Kconfig index b6df4e26b..955ac8d3f 100644 --- a/sound/soc/generic/Kconfig +++ b/sound/soc/generic/Kconfig @@ -7,7 +7,7 @@ config SND_SIMPLE_CARD select SND_SIMPLE_CARD_UTILS help This option enables generic simple sound card support - It also support DPCM of multi CPU single Codec ststem. + It also support DPCM of multi CPU single Codec system. config SND_AUDIO_GRAPH_CARD tristate "ASoC Audio Graph sound card support" @@ -16,7 +16,7 @@ config SND_AUDIO_GRAPH_CARD help This option enables generic simple sound card support with OF-graph DT bindings. - It also support DPCM of multi CPU single Codec ststem. + It also support DPCM of multi CPU single Codec system. config SND_AUDIO_GRAPH_CARD2 tristate "ASoC Audio Graph sound card2 support" -- 2.37.2

2 1

[PATCH] ALSA: control: Re-order bounds checking in get_ctl_id_hash()
by Dan Carpenter 26 Aug '22

26 Aug '22

These two checks are in the reverse order so it might read one element beyond the end of the array. First check if the "i" is within bounds before using it. Fixes: 6ab55ec0a938 ("ALSA: control: Fix an out-of-bounds bug in get_ctl_id_hash()") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/core/control.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/control.c b/sound/core/control.c index e8fc4c511e5f..a7271927d875 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -391,7 +391,7 @@ static unsigned long get_ctl_id_hash(const struct snd_ctl_elem_id *id) h = id->iface; h = MULTIPLIER * h + id->device; h = MULTIPLIER * h + id->subdevice; - for (i = 0; id->name[i] && i < SNDRV_CTL_ELEM_ID_NAME_MAXLEN; i++) + for (i = 0; i < SNDRV_CTL_ELEM_ID_NAME_MAXLEN && id->name[i]; i++) h = MULTIPLIER * h + id->name[i]; h = MULTIPLIER * h + id->index; h &= LONG_MAX; -- 2.35.1

2 1

Add interactive pause/resume functionality
by GitHub pull_request - edited 26 Aug '22

26 Aug '22

alsa-project/tinycompress pull request #14 was edited from LaurentiuM1234: We want to be able to interactively pause and resume the stream. Initially, when trying to play an MP3 song, cplay would hang during the polling on compress_write. The cause for this was the fact that, after calling `get_codec_mp3`, the file pointer would point to the first MP3 data, leaving out the first MP3 header. I'm assuming that this would cause the codec to be unable to process the received frames since it wouldn't be able to find the first MP3 header. In order to be able to use the pause/resume functionality on MP3 streams, this problem had to be addressed before adding the actual functionality. Request URL : https://github.com/alsa-project/tinycompress/pull/14 Patch URL : https://github.com/alsa-project/tinycompress/pull/14.patch Repository URL: https://github.com/alsa-project/tinycompress

1 0

Add interactive pause/resume functionality
by GitHub pull_request - edited 26 Aug '22

26 Aug '22

alsa-project/tinycompress pull request #14 was edited from LaurentiuM1234: We want to be able to interactively pause and resume the stream. Initially, when trying to play an MP3 song, cplay would hang during the polling on compress_write. The cause for this was probably the fact that, after calling `get_codec_mp3`, the file pointer would point to the first MP3 data, leaving out the first MP3 header. I'm assuming that this would cause the codec to hang since it wouldn't be able to find the first MP3 header. In order to be able to use the pause/resume functionality on MP3 streams, this problem had to be addressed before adding the actual functionality. Request URL : https://github.com/alsa-project/tinycompress/pull/14 Patch URL : https://github.com/alsa-project/tinycompress/pull/14.patch Repository URL: https://github.com/alsa-project/tinycompress

1 0

Add interactive pause/resume functionality
by GitHub pull_request - edited 26 Aug '22

26 Aug '22

alsa-project/tinycompress pull request #14 was edited from LaurentiuM1234: We want to be able to interactively pause and resume the stream Request URL : https://github.com/alsa-project/tinycompress/pull/14 Patch URL : https://github.com/alsa-project/tinycompress/pull/14.patch Repository URL: https://github.com/alsa-project/tinycompress

1 0

[PATCH 1/2] sound: realtek: Add pincfg for ASUS G533Z
by Luke D. Jones 26 Aug '22

26 Aug '22

Fixes up the pincfg for ASUS ROG Strix G15 (G533Z) laptop. Signed-off-by: Luke D. Jones <luke(a)ljones.dev> --- sound/pci/hda/patch_realtek.c | 15 ++++++++++++++- 1 file changed, 14 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fd630d62b5a0..e9cad49ee2a5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5828,7 +5828,7 @@ static void alc_fixup_headset_mode_alc255_no_hp_mic(struct hda_codec *codec, struct alc_spec *spec = codec->spec; spec->parse_flags |= HDA_PINCFG_HEADSET_MIC; alc255_set_default_jack_type(codec); - } + } else alc_fixup_headset_mode(codec, fix, action); } @@ -7022,6 +7022,7 @@ enum { ALC294_FIXUP_ASUS_GX502_HP, ALC294_FIXUP_ASUS_GX502_PINS, ALC294_FIXUP_ASUS_GX502_VERBS, + ALC285_FIXUP_ASUS_G533Z_PINS, ALC294_FIXUP_ASUS_GU502_HP, ALC294_FIXUP_ASUS_GU502_PINS, ALC294_FIXUP_ASUS_GU502_VERBS, @@ -8363,6 +8364,17 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc294_fixup_gu502_hp, }, + [ALC285_FIXUP_ASUS_G533Z_PINS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x14, 0x90170120 }, + { 0x19, 0x03A11050 }, /* front HP mic */ + { 0x1B, 0x03A11C30 }, /* rear external mic */ + { 0x21, 0x03211420 }, /* front HP out */ + { } + }, + .chained = false, + }, [ALC294_FIXUP_ASUS_COEF_1B] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -9294,6 +9306,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_DUAL_SPK), SND_PCI_QUIRK(0x1043, 0x1662, "ASUS GV301QH", ALC294_FIXUP_ASUS_DUAL_SPK), SND_PCI_QUIRK(0x1043, 0x1881, "ASUS Zephyrus S/M", ALC294_FIXUP_ASUS_GX502_PINS), + SND_PCI_QUIRK(0x1043, 0x1c92, "ASUS ROG Strix G15", ALC285_FIXUP_ASUS_G533Z_PINS), SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x18f1, "Asus FX505DT", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x194e, "ASUS UX563FD", ALC294_FIXUP_ASUS_HPE), -- 2.37.2

2 2

[PATCH v8 1/2] ASoC: sunxi: Add Allwinner H6 Digital MIC driver
by Ban Tao 26 Aug '22

26 Aug '22

The Allwinner H6 and later SoCs have an DMIC block which is capable of capture. Signed-off-by: Ban Tao <fengzheng923(a)gmail.com> --- v1->v2: 1.Fix some compilation errors. 2.Modify some code styles. v2->v3: None. v3->v4: 1.add sig_bits. v4->v5: None. v5->v6: 1.Modify RXFIFO_CTL_MODE to mode 1. v6->v7: 1.Modify dmic_rate_s to be a global variable. 2.Changed some macro names to make more sense. 3.Align code format. 4.Add a depends on PM to Kconfig entry. v7->v8: None. --- MAINTAINERS | 7 + sound/soc/sunxi/Kconfig | 8 + sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun50i-dmic.c | 405 ++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 421 insertions(+) create mode 100644 sound/soc/sunxi/sun50i-dmic.c diff --git a/MAINTAINERS b/MAINTAINERS index e9d5b05..839f625 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -796,6 +796,13 @@ L: linux-media(a)vger.kernel.org S: Maintained F: drivers/staging/media/sunxi/cedrus/ +ALLWINNER DMIC DRIVERS +M: Ban Tao <fengzheng923(a)gmail.com> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained +F: Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml +F: sound/soc/sunxi/sun50i-dmic.c + ALPHA PORT M: Richard Henderson <rth(a)twiddle.net> M: Ivan Kokshaysky <ink(a)jurassic.park.msu.ru> diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index ddcaaa9..8543234 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -56,6 +56,14 @@ config SND_SUN4I_SPDIF Say Y or M to add support for the S/PDIF audio block in the Allwinner A10 and affiliated SoCs. +config SND_SUN50I_DMIC + tristate "Allwinner H6 DMIC Support" + depends on (OF && PM && ARCH_SUNXI) || COMPILE_TEST + select SND_SOC_GENERIC_DMAENGINE_PCM + help + Say Y or M to add support for the DMIC audio block in the Allwinner + H6 and affiliated SoCs. + config SND_SUN8I_ADDA_PR_REGMAP tristate select REGMAP diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile index a86be34..4483fe9 100644 --- a/sound/soc/sunxi/Makefile +++ b/sound/soc/sunxi/Makefile @@ -6,3 +6,4 @@ obj-$(CONFIG_SND_SUN8I_CODEC_ANALOG) += sun8i-codec-analog.o obj-$(CONFIG_SND_SUN50I_CODEC_ANALOG) += sun50i-codec-analog.o obj-$(CONFIG_SND_SUN8I_CODEC) += sun8i-codec.o obj-$(CONFIG_SND_SUN8I_ADDA_PR_REGMAP) += sun8i-adda-pr-regmap.o +obj-$(CONFIG_SND_SUN50I_DMIC) += sun50i-dmic.o diff --git a/sound/soc/sunxi/sun50i-dmic.c b/sound/soc/sunxi/sun50i-dmic.c new file mode 100644 index 0000000..76b3378 --- /dev/null +++ b/sound/soc/sunxi/sun50i-dmic.c @@ -0,0 +1,405 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +// +// This driver supports the DMIC in Allwinner's H6 SoCs. +// +// Copyright 2021 Ban Tao <fengzheng923(a)gmail.com> + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/of_device.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <linux/reset.h> +#include <sound/dmaengine_pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#define SUN50I_DMIC_EN_CTL (0x00) + #define SUN50I_DMIC_EN_CTL_GLOBE BIT(8) + #define SUN50I_DMIC_EN_CTL_CHAN(v) ((v) << 0) + #define SUN50I_DMIC_EN_CTL_CHAN_MASK GENMASK(7, 0) +#define SUN50I_DMIC_SR (0x04) + #define SUN50I_DMIC_SR_SAMPLE_RATE(v) ((v) << 0) + #define SUN50I_DMIC_SR_SAMPLE_RATE_MASK GENMASK(2, 0) +#define SUN50I_DMIC_CTL (0x08) + #define SUN50I_DMIC_CTL_OVERSAMPLE_RATE BIT(0) +#define SUN50I_DMIC_DATA (0x10) +#define SUN50I_DMIC_INTC (0x14) + #define SUN50I_DMIC_FIFO_DRQ_EN BIT(2) +#define SUN50I_DMIC_INT_STA (0x18) + #define SUN50I_DMIC_INT_STA_OVERRUN_IRQ_PENDING BIT(1) + #define SUN50I_DMIC_INT_STA_DATA_IRQ_PENDING BIT(0) +#define SUN50I_DMIC_RXFIFO_CTL (0x1c) + #define SUN50I_DMIC_RXFIFO_CTL_FLUSH BIT(31) + #define SUN50I_DMIC_RXFIFO_CTL_MODE_MASK BIT(9) + #define SUN50I_DMIC_RXFIFO_CTL_MODE_LSB (0 << 9) + #define SUN50I_DMIC_RXFIFO_CTL_MODE_MSB (1 << 9) + #define SUN50I_DMIC_RXFIFO_CTL_SAMPLE_MASK BIT(8) + #define SUN50I_DMIC_RXFIFO_CTL_SAMPLE_16 (0 << 8) + #define SUN50I_DMIC_RXFIFO_CTL_SAMPLE_24 (1 << 8) +#define SUN50I_DMIC_CH_NUM (0x24) + #define SUN50I_DMIC_CH_NUM_N(v) ((v) << 0) + #define SUN50I_DMIC_CH_NUM_N_MASK GENMASK(2, 0) +#define SUN50I_DMIC_CNT (0x2c) + #define SUN50I_DMIC_CNT_N BIT(0) +#define SUN50I_DMIC_HPF_CTRL (0x38) +#define SUN50I_DMIC_VERSION (0x50) + +struct sun50i_dmic_dev { + struct clk *dmic_clk; + struct clk *bus_clk; + struct reset_control *rst; + struct regmap *regmap; + struct snd_dmaengine_dai_dma_data dma_params_rx; +}; + +struct dmic_rate { + unsigned int samplerate; + unsigned int rate_bit; +}; + +static struct dmic_rate dmic_rate_s[] = { + {48000, 0x0}, + {44100, 0x0}, + {32000, 0x1}, + {24000, 0x2}, + {22050, 0x2}, + {16000, 0x3}, + {12000, 0x4}, + {11025, 0x4}, + {8000, 0x5}, +}; + +static int sun50i_dmic_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); + + /* only support capture */ + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_FLUSH, + SUN50I_DMIC_RXFIFO_CTL_FLUSH); + regmap_write(host->regmap, SUN50I_DMIC_CNT, SUN50I_DMIC_CNT_N); + + return 0; +} + +static int sun50i_dmic_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + int i = 0; + unsigned long rate = params_rate(params); + unsigned int mclk = 0; + unsigned int channels = params_channels(params); + unsigned int chan_en = (1 << channels) - 1; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + + /* DMIC num is N+1 */ + regmap_update_bits(host->regmap, SUN50I_DMIC_CH_NUM, + SUN50I_DMIC_CH_NUM_N_MASK, + SUN50I_DMIC_CH_NUM_N(channels - 1)); + regmap_write(host->regmap, SUN50I_DMIC_HPF_CTRL, chan_en); + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_CHAN_MASK, + SUN50I_DMIC_EN_CTL_CHAN(chan_en)); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_SAMPLE_MASK, + SUN50I_DMIC_RXFIFO_CTL_SAMPLE_16); + break; + case SNDRV_PCM_FORMAT_S24_LE: + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_SAMPLE_MASK, + SUN50I_DMIC_RXFIFO_CTL_SAMPLE_24); + break; + default: + dev_err(cpu_dai->dev, "Invalid format!\n"); + return -EINVAL; + } + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_MODE_MASK, + SUN50I_DMIC_RXFIFO_CTL_MODE_MSB); + + switch (rate) { + case 11025: + case 22050: + case 44100: + mclk = 22579200; + break; + case 8000: + case 12000: + case 16000: + case 24000: + case 32000: + case 48000: + mclk = 24576000; + break; + default: + dev_err(cpu_dai->dev, "Invalid rate!\n"); + return -EINVAL; + } + + if (clk_set_rate(host->dmic_clk, mclk)) { + dev_err(cpu_dai->dev, "mclk : %u not support\n", mclk); + return -EINVAL; + } + + for (i = 0; i < ARRAY_SIZE(dmic_rate_s); i++) { + if (dmic_rate_s[i].samplerate == rate) { + regmap_update_bits(host->regmap, SUN50I_DMIC_SR, + SUN50I_DMIC_SR_SAMPLE_RATE_MASK, + SUN50I_DMIC_SR_SAMPLE_RATE(dmic_rate_s[i].rate_bit)); + break; + } + } + + switch (params_physical_width(params)) { + case 16: + host->dma_params_rx.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + break; + case 32: + host->dma_params_rx.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + break; + default: + dev_err(cpu_dai->dev, "Unsupported physical sample width: %d\n", + params_physical_width(params)); + return -EINVAL; + } + + /* oversamplerate adjust */ + if (params_rate(params) >= 24000) + regmap_update_bits(host->regmap, SUN50I_DMIC_CTL, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE); + else + regmap_update_bits(host->regmap, SUN50I_DMIC_CTL, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE, 0); + + return 0; +} + +static int sun50i_dmic_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + int ret = 0; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* DRQ ENABLE */ + regmap_update_bits(host->regmap, SUN50I_DMIC_INTC, + SUN50I_DMIC_FIFO_DRQ_EN, + SUN50I_DMIC_FIFO_DRQ_EN); + /* Global enable */ + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_GLOBE, + SUN50I_DMIC_EN_CTL_GLOBE); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + /* DRQ DISABLE */ + regmap_update_bits(host->regmap, SUN50I_DMIC_INTC, + SUN50I_DMIC_FIFO_DRQ_EN, 0); + /* Global disable */ + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_GLOBE, 0); + break; + default: + ret = -EINVAL; + break; + } + return ret; +} + +static int sun50i_dmic_soc_dai_probe(struct snd_soc_dai *dai) +{ + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, NULL, &host->dma_params_rx); + + return 0; +} + +static const struct snd_soc_dai_ops sun50i_dmic_dai_ops = { + .startup = sun50i_dmic_startup, + .trigger = sun50i_dmic_trigger, + .hw_params = sun50i_dmic_hw_params, +}; + +static const struct regmap_config sun50i_dmic_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = SUN50I_DMIC_VERSION, + .cache_type = REGCACHE_NONE, +}; + +#define SUN50I_DMIC_RATES (SNDRV_PCM_RATE_8000_48000) +#define SUN50I_DMIC_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_driver sun50i_dmic_dai = { + .capture = { + .channels_min = 1, + .channels_max = 8, + .rates = SUN50I_DMIC_RATES, + .formats = SUN50I_DMIC_FORMATS, + .sig_bits = 21, + }, + .probe = sun50i_dmic_soc_dai_probe, + .ops = &sun50i_dmic_dai_ops, + .name = "dmic", +}; + +static const struct of_device_id sun50i_dmic_of_match[] = { + { + .compatible = "allwinner,sun50i-h6-dmic", + }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, sun50i_dmic_of_match); + +static const struct snd_soc_component_driver sun50i_dmic_component = { + .name = "sun50i-dmic", +}; + +static int sun50i_dmic_runtime_suspend(struct device *dev) +{ + struct sun50i_dmic_dev *host = dev_get_drvdata(dev); + + clk_disable_unprepare(host->dmic_clk); + clk_disable_unprepare(host->bus_clk); + + return 0; +} + +static int sun50i_dmic_runtime_resume(struct device *dev) +{ + struct sun50i_dmic_dev *host = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(host->dmic_clk); + if (ret) + return ret; + + ret = clk_prepare_enable(host->bus_clk); + if (ret) { + clk_disable_unprepare(host->dmic_clk); + return ret; + } + + return 0; +} + +static int sun50i_dmic_probe(struct platform_device *pdev) +{ + struct sun50i_dmic_dev *host; + struct resource *res; + int ret; + void __iomem *base; + + host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); + if (!host) + return -ENOMEM; + + /* Get the addresses */ + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) + return dev_err_probe(&pdev->dev, PTR_ERR(base), + "get resource failed.\n"); + + host->regmap = devm_regmap_init_mmio(&pdev->dev, base, + &sun50i_dmic_regmap_config); + + /* Clocks */ + host->bus_clk = devm_clk_get(&pdev->dev, "bus"); + if (IS_ERR(host->bus_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->bus_clk), + "failed to get bus clock.\n"); + + host->dmic_clk = devm_clk_get(&pdev->dev, "mod"); + if (IS_ERR(host->dmic_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->dmic_clk), + "failed to get dmic clock.\n"); + + host->dma_params_rx.addr = res->start + SUN50I_DMIC_DATA; + host->dma_params_rx.maxburst = 8; + + platform_set_drvdata(pdev, host); + + host->rst = devm_reset_control_get_optional_exclusive(&pdev->dev, NULL); + if (IS_ERR(host->rst)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->rst), + "Failed to get reset.\n"); + reset_control_deassert(host->rst); + + ret = devm_snd_soc_register_component(&pdev->dev, &sun50i_dmic_component, + &sun50i_dmic_dai, 1); + if (ret) + return dev_err_probe(&pdev->dev, ret, + "failed to register component.\n"); + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = sun50i_dmic_runtime_resume(&pdev->dev); + if (ret) + goto err_disable_runtime_pm; + } + + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) + goto err_suspend; + + return 0; +err_suspend: + if (!pm_runtime_status_suspended(&pdev->dev)) + sun50i_dmic_runtime_suspend(&pdev->dev); +err_disable_runtime_pm: + pm_runtime_disable(&pdev->dev); + return ret; +} + +static int sun50i_dmic_remove(struct platform_device *pdev) +{ + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + sun50i_dmic_runtime_suspend(&pdev->dev); + + return 0; +} + +static const struct dev_pm_ops sun50i_dmic_pm = { + SET_RUNTIME_PM_OPS(sun50i_dmic_runtime_suspend, + sun50i_dmic_runtime_resume, NULL) +}; + +static struct platform_driver sun50i_dmic_driver = { + .driver = { + .name = "sun50i-dmic", + .of_match_table = of_match_ptr(sun50i_dmic_of_match), + .pm = &sun50i_dmic_pm, + }, + .probe = sun50i_dmic_probe, + .remove = sun50i_dmic_remove, +}; + +module_platform_driver(sun50i_dmic_driver); + +MODULE_DESCRIPTION("Allwinner sun50i DMIC SoC Interface"); +MODULE_AUTHOR("Ban Tao <fengzheng923(a)gmail.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:sun50i-dmic"); -- 2.7.4

3 3

[PATCH] ASoC: wm_adsp: Silent parsing error on loading speaker protection fw
by Cristian Ciocaltea 26 Aug '22

26 Aug '22

The tracing capabilities for the speaker protection fw enabled via commit c55b3e46cb99 ("ASoC: wm_adsp: Add trace caps to speaker protection FW") are not be available on all platforms, such as the Valve's Steam Deck which is based on the Halo Core DSP. As a consequence, whenever the firmware is loaded, a rather misleading 'Failed to parse legacy: -19' error message is written to the kernel ring buffer: [ 288.977412] steamdeck kernel: cs35l41 spi-VLV1776:01: DSP1: Firmware version: 3 [ 288.978002] steamdeck kernel: cs35l41 spi-VLV1776:01: DSP1: cs35l41-dsp1-spk-prot.wmfw: Fri 02 Apr 2021 21:03:50 W. Europe Daylight Time [ 289.094065] steamdeck kernel: cs35l41 spi-VLV1776:01: DSP1: Firmware: 400a4 vendor: 0x2 v0.33.0, 2 algorithms [ 289.095073] steamdeck kernel: cs35l41 spi-VLV1776:01: DSP1: 0: ID cd v29.53.0 XM@94 YM@e [ 289.095665] steamdeck kernel: cs35l41 spi-VLV1776:01: DSP1: 1: ID f20b v0.0.1 XM@170 YM@0 [ 289.096275] steamdeck kernel: cs35l41 spi-VLV1776:01: DSP1: Protection: C:\Users\ocanavan\Desktop\cirrusTune_july2021.bin [ 291.172383] steamdeck kernel: cs35l41 spi-VLV1776:01: DSP1: Failed to parse legacy: -19 Update wm_adsp_buffer_init() to *not* report the ENODEV error when the firmware type is WM_ADSP_FW_SPK_PROT. Fixes: c55b3e46cb99 ("ASoC: wm_adsp: Add trace caps to speaker protection FW") Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea(a)collabora.com> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index cfaa45ede916..7514fc03b468 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1602,7 +1602,7 @@ static int wm_adsp_buffer_init(struct wm_adsp *dsp) if (list_empty(&dsp->buffer_list)) { /* Fall back to legacy support */ ret = wm_adsp_buffer_parse_legacy(dsp); - if (ret) + if (ret && (dsp->fw != WM_ADSP_FW_SPK_PROT || ret != -ENODEV)) adsp_warn(dsp, "Failed to parse legacy: %d\n", ret); } base-commit: 568035b01cfb107af8d2e4bd2fb9aea22cf5b868 -- 2.37.2

2 2

[PATCH v2 1/2] ASoC: dt-bindings: max98396: Document data monitor properties
by Daniel Mack 25 Aug '22

25 Aug '22

This device features a data monitor that puts the device in software reset upon a configurable set of events. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- v1 -> v2: fix a typo and remove a stray blank line .../bindings/sound/adi,max98396.yaml | 37 +++++++++++++++++++ 1 file changed, 37 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/adi,max98396.yaml b/Documentation/devicetree/bindings/sound/adi,max98396.yaml index 8d2ef991db40..f4eefe10c494 100644 --- a/Documentation/devicetree/bindings/sound/adi,max98396.yaml +++ b/Documentation/devicetree/bindings/sound/adi,max98396.yaml @@ -78,6 +78,43 @@ properties: interleaved on a single output channel. type: boolean + adi,dmon-stuck-enable: + description: + Enables the "data monitor stuck" feature. Once the data monitor is + enabled, it actively monitors the selected input data (from DIN) to the + speaker amplifier. Once a data error is detected, the data monitor + automatically places the device into software shutdown. + type: boolean + + adi,dmon-stuck-threshold-bits: + description: + Sets the threshold for the "data monitor stuck" feature, in bits. + $ref: "/schemas/types.yaml#/definitions/uint32" + enum: [9, 11, 13, 15] + default: 15 + + adi,dmon-magnitude-enable: + description: + Enables the "data monitor magnitude" feature. Once the data monitor is + enabled, it actively monitors the selected input data (from DIN) to the + speaker amplifier. Once a data error is detected, the data monitor + automatically places the device into software shutdown. + type: boolean + + adi,dmon-magnitude-threshold-bits: + description: + Sets the threshold for the "data monitor magnitude" feature, in bits. + $ref: "/schemas/types.yaml#/definitions/uint32" + enum: [2, 3, 4, 5] + default: 5 + + adi,dmon-duration-msecs: + description: + Sets the duration for the "data monitor" feature, in milliseconds. + $ref: "/schemas/types.yaml#/definitions/uint32" + enum: [64, 256, 1024, 4096] + default: 64 + reset-gpios: maxItems: 1 -- 2.37.2

2 2

[PATCH 1/2] lib/string_helpers: Introduce strsplit_u32()
by Cezary Rojewski 25 Aug '22

25 Aug '22

Add strsplit_u32() and its __user variant to allow for splitting specified string into array of u32 tokens. Originally this functionality was added for the SOF sound driver. As more users are on the horizon, relocate it so it becomes a common good. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- include/linux/string_helpers.h | 3 + lib/string_helpers.c | 96 +++++++++++++++++++++++++++++++ sound/soc/sof/sof-client-probes.c | 51 +--------------- 3 files changed, 100 insertions(+), 50 deletions(-) diff --git a/include/linux/string_helpers.h b/include/linux/string_helpers.h index 4d72258d42fd..a4630ddfca27 100644 --- a/include/linux/string_helpers.h +++ b/include/linux/string_helpers.h @@ -126,4 +126,7 @@ static inline const char *str_enabled_disabled(bool v) return v ? "enabled" : "disabled"; } +int strsplit_u32(const char *str, const char *delim, u32 **tkns, size_t *num_tkns); +int strsplit_u32_user(const char __user *from, size_t count, loff_t *ppos, const char *delim, + u32 **tkns, size_t *num_tkns); #endif diff --git a/lib/string_helpers.c b/lib/string_helpers.c index 5ed3beb066e6..bb24f0c62539 100644 --- a/lib/string_helpers.c +++ b/lib/string_helpers.c @@ -984,3 +984,99 @@ void fortify_panic(const char *name) } EXPORT_SYMBOL(fortify_panic); #endif /* CONFIG_FORTIFY_SOURCE */ + +/** + * strsplit_u32 - Split string into sequence of u32 tokens + * @str: The string to split into tokens. + * @delim: The string containing delimiter characters. + * @tkns: Returned u32 sequence pointer. + * @num_tkns: Returned number of tokens obtained. + * + * On success @num_tkns and @tkns are assigned the number of tokens extracted + * and the array itself respectively. + * Caller takes responsibility for freeing @tkns when no longer needed. + */ +int strsplit_u32(const char *str, const char *delim, u32 **tkns, size_t *num_tkns) +{ + size_t max_count = 32; + size_t count = 0; + char *s, **p; + u32 *buf, *tmp; + int ret = 0; + + p = (char **)&str; + *tkns = NULL; + *num_tkns = 0; + + buf = kcalloc(max_count, sizeof(*buf), GFP_KERNEL); + if (!buf) + return -ENOMEM; + + while ((s = strsep(p, delim)) != NULL) { + ret = kstrtouint(s, 0, buf + count); + if (ret) + goto free_buf; + + if (++count > max_count) { + max_count *= 2; + tmp = krealloc(buf, max_count * sizeof(*buf), GFP_KERNEL); + if (!tmp) { + ret = -ENOMEM; + goto free_buf; + } + buf = tmp; + } + } + + if (!count) + goto free_buf; + *tkns = kmemdup(buf, count * sizeof(*buf), GFP_KERNEL); + if (*tkns == NULL) { + ret = -ENOMEM; + goto free_buf; + } + *num_tkns = count; + +free_buf: + kfree(buf); + return ret; +} +EXPORT_SYMBOL(strsplit_u32); + +/** + * strsplit_u32_user - Split string into sequence of u32 tokens + * @from: The user space buffer to read from + * @ppos: The current position in the buffer + * @count: The maximum number of bytes to read + * @delim: The string containing delimiter characters. + * @tkns: Returned u32 sequence pointer. + * @num_tkns: Returned number of tokens obtained. + * + * On success @num_tkns and @tkns are assigned the number of tokens extracted + * and the array itself respectively. + * Caller takes responsibility for freeing @tkns when no longer needed. + */ +int strsplit_u32_user(const char __user *from, size_t count, loff_t *ppos, const char *delim, + u32 **tkns, size_t *num_tkns) +{ + char *buf; + int ret; + + buf = kmalloc(count + 1, GFP_KERNEL); + if (!buf) + return -ENOMEM; + + ret = simple_write_to_buffer(buf, count, ppos, from, count); + if (ret != count) { + ret = (ret < 0) ? ret : -EIO; + goto free_buf; + } + + buf[count] = '\0'; + ret = strsplit_u32(buf, delim, tkns, num_tkns); + +free_buf: + kfree(buf); + return ret; +} +EXPORT_SYMBOL(strsplit_u32_user); diff --git a/sound/soc/sof/sof-client-probes.c b/sound/soc/sof/sof-client-probes.c index 1f1ea93a7fbf..48ebbe58e2b9 100644 --- a/sound/soc/sof/sof-client-probes.c +++ b/sound/soc/sof/sof-client-probes.c @@ -12,6 +12,7 @@ #include <linux/debugfs.h> #include <linux/module.h> #include <linux/pm_runtime.h> +#include <linux/string_helpers.h> #include <sound/soc.h> #include <sound/sof/header.h> #include "sof-client.h" @@ -410,56 +411,6 @@ static const struct snd_compress_ops sof_probes_compressed_ops = { .copy = sof_probes_compr_copy, }; -/** - * strsplit_u32 - Split string into sequence of u32 tokens - * @buf: String to split into tokens. - * @delim: String containing delimiter characters. - * @tkns: Returned u32 sequence pointer. - * @num_tkns: Returned number of tokens obtained. - */ -static int strsplit_u32(char *buf, const char *delim, u32 **tkns, size_t *num_tkns) -{ - char *s; - u32 *data, *tmp; - size_t count = 0; - size_t cap = 32; - int ret = 0; - - *tkns = NULL; - *num_tkns = 0; - data = kcalloc(cap, sizeof(*data), GFP_KERNEL); - if (!data) - return -ENOMEM; - - while ((s = strsep(&buf, delim)) != NULL) { - ret = kstrtouint(s, 0, data + count); - if (ret) - goto exit; - if (++count >= cap) { - cap *= 2; - tmp = krealloc(data, cap * sizeof(*data), GFP_KERNEL); - if (!tmp) { - ret = -ENOMEM; - goto exit; - } - data = tmp; - } - } - - if (!count) - goto exit; - *tkns = kmemdup(data, count * sizeof(*data), GFP_KERNEL); - if (!(*tkns)) { - ret = -ENOMEM; - goto exit; - } - *num_tkns = count; - -exit: - kfree(data); - return ret; -} - static int tokenize_input(const char __user *from, size_t count, loff_t *ppos, u32 **tkns, size_t *num_tkns) { -- 2.25.1

5 29

[PATCH -next] ASoC: sigmadsp: switch to use kmemdup_nul() helper
by Yang Yingliang 25 Aug '22

25 Aug '22

Use kmemdup_nul() helper instead of open-coding to simplify the code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/codecs/sigmadsp.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c index b992216aee55..3047a6fbb380 100644 --- a/sound/soc/codecs/sigmadsp.c +++ b/sound/soc/codecs/sigmadsp.c @@ -227,13 +227,11 @@ static int sigma_fw_load_control(struct sigmadsp *sigmadsp, if (!ctrl) return -ENOMEM; - name = kzalloc(name_len + 1, GFP_KERNEL); + name = kmemdup_nul(ctrl_chunk->name, name_len, GFP_KERNEL); if (!name) { ret = -ENOMEM; goto err_free_ctrl; } - memcpy(name, ctrl_chunk->name, name_len); - name[name_len] = '\0'; ctrl->name = name; /* -- 2.25.1

2 1

[PATCH v2 0/2] ASoC: mediatek: dt-bindings: modify machine
by Chunxu Li 25 Aug '22

25 Aug '22

From: "chunxu.li" <chunxu.li(a)mediatek.com> Changes since V1: - remove unnecessary quotes Add SOF related field. 1. Add a property "mediatek,adsp", Only when adsp phandle could be retrieved, from DTS, the SOF related part of machine driver is executed. 2. Add a property "mediatek,dai-link" to support dai-links could be specified from DTS chunxu.li (2): ASoC: mediatek: dt-bindings: modify machine bindings for SOF ASoC: mediatek: dt-bindings: modify machine bindings for SOF .../bindings/sound/mt8186-mt6366-da7219-max98357.yaml | 10 ++++++++++ .../bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml | 10 ++++++++++ 2 files changed, 20 insertions(+) -- 2.25.1

3 7

[PATCH 1/2] ASoC: fsl_aud2htx: register platform component before registering cpu dai
by Shengjiu Wang 25 Aug '22

25 Aug '22

There is no defer probe when adding platform component to snd_soc_pcm_runtime(rtd), the code is in snd_soc_add_pcm_runtime() snd_soc_register_card() -> snd_soc_bind_card() -> snd_soc_add_pcm_runtime() -> adding cpu dai -> adding codec dai -> adding platform component. So if the platform component is not ready at that time, then the sound card still registered successfully, but platform component is empty, the sound card can't be used. As there is defer probe checking for cpu dai component, then register platform component before cpu dai to avoid such issue. And the behavior of imx_pcm_dma_init() is same as common devm_snd_dmaengine_pcm_register(), so use devm_snd_dmaengine_pcm_register() instead Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_aud2htx.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_aud2htx.c b/sound/soc/fsl/fsl_aud2htx.c index 873295f59ad7..bc1b9c6df95c 100644 --- a/sound/soc/fsl/fsl_aud2htx.c +++ b/sound/soc/fsl/fsl_aud2htx.c @@ -234,6 +234,16 @@ static int fsl_aud2htx_probe(struct platform_device *pdev) regcache_cache_only(aud2htx->regmap, true); + /* + * Register platform component before registering cpu dai for there + * is not defer probe for platform component in snd_soc_add_pcm_runtime(). + */ + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) { + dev_err(&pdev->dev, "failed to pcm register\n"); + return ret; + } + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_aud2htx_component, &fsl_aud2htx_dai, 1); @@ -242,10 +252,6 @@ static int fsl_aud2htx_probe(struct platform_device *pdev) return ret; } - ret = imx_pcm_dma_init(pdev); - if (ret) - dev_err(&pdev->dev, "failed to init imx pcm dma: %d\n", ret); - return ret; } -- 2.34.1

2 2

[PATCH 1/2] ASoC: dt-bindings: max98396: Document data monitor properties
by Daniel Mack 25 Aug '22

25 Aug '22

This device features a data monitor that puts the device in software reset upon a configurable set of events. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- .../bindings/sound/adi,max98396.yaml | 37 +++++++++++++++++++ 1 file changed, 37 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/adi,max98396.yaml b/Documentation/devicetree/bindings/sound/adi,max98396.yaml index 8d2ef991db40..f4eefe10c494 100644 --- a/Documentation/devicetree/bindings/sound/adi,max98396.yaml +++ b/Documentation/devicetree/bindings/sound/adi,max98396.yaml @@ -78,6 +78,43 @@ properties: interleaved on a single output channel. type: boolean + adi,dmon-stuck-enable: + description: + Enables the "data monitor stuck" feature. Once the data monitor is + enabled, it actively monitors the selected input data (from DIN) to the + speaker amplifier. Once a data error is detected, the data monitor + automatically places the device into software shutdown. + type: boolean + + adi,dmon-stuck-threshold-bits: + description: + Sets the threshold for the "data monitor stuck" feature, in bits. + $ref: "/schemas/types.yaml#/definitions/uint32" + enum: [9, 11, 13, 15] + default: 15 + + adi,dmon-magnitude-enable: + description: + Enables the "data monitor magnitude" feature. Once the data monitor is + enabled, it actively monitors the selected input data (from DIN) to the + speaker amplifier. Once a data error is detected, the data monitor + automatically places the device into software shutdown. + type: boolean + + adi,dmon-magnitude-threshold-bits: + description: + Sets the threshold for the "data monitor magnitude" feature, in bits. + $ref: "/schemas/types.yaml#/definitions/uint32" + enum: [2, 3, 4, 5] + default: 5 + + adi,dmon-duration-msecs: + description: + Sets the duration for the "data monitor" feature, in milliseconds. + $ref: "/schemas/types.yaml#/definitions/uint32" + enum: [64, 256, 1024, 4096] + default: 64 + reset-gpios: maxItems: 1 -- 2.37.2

1 1

[PATCH] ASoC: dapm: Export new 'graph.dot' file in debugfs
by Martin Povišer 25 Aug '22

25 Aug '22

Provide a DOT summary of the DAPM graph in a newly added 'graph.dot' file in debugfs, placed in the card's DAPM directory. Signed-off-by: Martin Povišer <povik+lin(a)cutebit.org> --- Sample output: https://cutebit.org/macaudio-j274.svg (With unupstreamed sound drivers on Mac mini (2020)) The helper bufprintf macro triggers checkpath.pl: ERROR: Macros with complex values should be enclosed in parentheses #47: FILE: sound/soc/soc-dapm.c:2235: +#define bufprintf(...) \ + ret += scnprintf(buf + ret, bufsize - ret, __VA_ARGS__) but adding in {} to the macro body interferes with the if/else constructions later, so I left it as-is. sound/soc/soc-dapm.c | 141 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 141 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 73b8bd452ca7..86524908c3fd 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2210,6 +2210,143 @@ static const struct file_operations dapm_bias_fops = { .llseek = default_llseek, }; +static ssize_t dapm_graph_read_file(struct file *file, char __user *user_buf, + size_t count, loff_t *ppos) +{ + struct snd_soc_card *card = file->private_data; + struct snd_soc_dapm_context *dapm; + struct snd_soc_dapm_path *p; + struct snd_soc_dapm_widget *w; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dapm_widget *wdone[16]; + struct snd_soc_dai *dai; + int i, num_wdone = 0, cluster = 0; + char *buf; + ssize_t bufsize; + ssize_t ret = 0; + + bufsize = 1024 * card->num_dapm_widgets; + buf = kmalloc(bufsize, GFP_KERNEL); + if (!buf) + return -ENOMEM; + + mutex_lock(&card->dapm_mutex); + +#define bufprintf(...) \ + ret += scnprintf(buf + ret, bufsize - ret, __VA_ARGS__) + + bufprintf("digraph dapm {\n"); + bufprintf("label=\"%s\";\n", card->name); + + /* + * Print the user-visible PCM devices of the card. + */ + bufprintf("subgraph cluster_%d {\n", cluster++); + bufprintf("label=\"PCM devices\";style=filled;fillcolor=lightgray;\n"); + for_each_card_rtds(card, rtd) { + if (rtd->dai_link->no_pcm) + continue; + + bufprintf("w%pK [label=\"%d: %s\"];\n", rtd, + rtd->pcm->device, rtd->dai_link->name); + } + bufprintf("};\n"); + + /* + * Print the playback/capture widgets of CPU-side DAIs, and link + * them to the PCM devices. Keep a list of already printed + * widgets in 'wdone', so they will be skipped later. Do not put + * these widgets in a component cluster like we will do with + * the other widgets later, since that just clutters the graph. + */ + for_each_card_rtds(card, rtd) { + for_each_rtd_cpu_dais(rtd, i, dai) { + if (dai->playback_widget) { + w = dai->playback_widget; + bufprintf("w%pK [label=\"%s\"];\n", w, w->name); + if (!rtd->dai_link->no_pcm) + bufprintf("w%pK -> w%pK;\n", rtd, w); + wdone[num_wdone] = w; + if (num_wdone < ARRAY_SIZE(wdone)) + num_wdone++; + } + + if (dai->capture_widget) { + w = dai->capture_widget; + bufprintf("w%pK [label=\"%s\"];\n", w, w->name); + if (!rtd->dai_link->no_pcm) + bufprintf("w%pK -> w%pK;\n", w, rtd); + wdone[num_wdone] = w; + if (num_wdone < ARRAY_SIZE(wdone)) + num_wdone++; + } + } + } + + for_each_card_dapms(card, dapm) { + const char *prefix = soc_dapm_prefix(dapm); + + if (dapm != &card->dapm) { + bufprintf("subgraph cluster_%d {\n", cluster++); + if (prefix && dapm->component) + bufprintf("label=\"%s (%s)\";\n", prefix, + dapm->component->name); + else if (dapm->component) + bufprintf("label=\"%s\";\n", + dapm->component->name); + } + + for_each_card_widgets(dapm->card, w) { + const char *name = w->name; + bool skip = false; + + if (w->dapm != dapm) + continue; + + if (list_empty(&w->edges[0]) && list_empty(&w->edges[1])) + continue; + + for (i = 0; i < num_wdone; i++) + if (wdone[i] == w) + skip = true; + if (skip) + continue; + + if (prefix && strlen(name) > strlen(prefix) + 1) + name += strlen(prefix) + 1; + + bufprintf("w%pK [label=\"%s\"];\n", w, name); + } + + if (dapm != &card->dapm) + bufprintf("}\n"); + } + + list_for_each_entry(p, &card->paths, list) { + if (p->name) + bufprintf("w%pK -> w%pK [label=\"%s\"];\n", + p->source, p->sink, p->name); + else + bufprintf("w%pK -> w%pK;\n", p->source, p->sink); + } + + bufprintf("}\n"); +#undef bufprintf + + mutex_unlock(&card->dapm_mutex); + + ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret); + + kfree(buf); + return ret; +} + +static const struct file_operations dapm_graph_fops = { + .open = simple_open, + .read = dapm_graph_read_file, + .llseek = default_llseek, +}; + void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm, struct dentry *parent) { @@ -2220,6 +2357,10 @@ void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm, debugfs_create_file("bias_level", 0444, dapm->debugfs_dapm, dapm, &dapm_bias_fops); + + if (dapm == &dapm->card->dapm) + debugfs_create_file("graph.dot", 0444, dapm->debugfs_dapm, + dapm->card, &dapm_graph_fops); } static void dapm_debugfs_add_widget(struct snd_soc_dapm_widget *w) -- 2.33.0

5 10

[PATCH] ASoC: max98088: add support for noise gate reg
by Tommaso Merciai 25 Aug '22

25 Aug '22

Add support for Noise Gate Threshold reg (ANTH reg 0x40 bit 4 to 7) References: - https://datasheets.maximintegrated.com/en/ds/MAX98089.pdf, p75 Signed-off-by: Tommaso Merciai <tommaso.merciai(a)amarulasolutions.com> --- sound/soc/codecs/max98088.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 5ef2e1279ee7..76ef71ab9c83 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -474,6 +474,9 @@ static const struct snd_kcontrol_new max98088_snd_controls[] = { max98088_mic2pre_get, max98088_mic2pre_set, max98088_micboost_tlv), + SOC_SINGLE("Noise Gate Threshold", M98088_REG_40_MICAGC_THRESH, + 4, 15, 0), + SOC_SINGLE("INA Volume", M98088_REG_37_LVL_INA, 0, 7, 1), SOC_SINGLE("INB Volume", M98088_REG_38_LVL_INB, 0, 7, 1), -- 2.25.1

2 1

[PATCH 0/3] ASoC: SOF: mediatek: update SOF driver for mt8186
by Chunxu Li 25 Aug '22

25 Aug '22

In these patches, we update SOF driver for mt8186 Chunxu Li (3): ASoC: SOF: mediatek: Add dai driver for mt8186 ASoC: SOF: mediatek: add snd_sof_dsp_ops callbacks for pcm and mail box ASoC: SOF: mediatek: Use generic implementation for .ipc_msg_data field sound/soc/sof/mediatek/mt8186/mt8186.c | 49 ++++++++++++++++++++++---- 1 file changed, 42 insertions(+), 7 deletions(-) -- 2.25.1

2 4

[PATCH] ASoC: codecs: add suspend and resume for ES8316
by Zhu Ning 25 Aug '22

25 Aug '22

The registers may be lost after suspend due to powerdown. regcache_sync solves this issue. Signed-off-by: Zhu Ning <zhuning(a)everest-semi.com> --- sound/soc/codecs/es8316.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index de7185f73e1e..8643014472ae 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -767,9 +767,31 @@ static void es8316_remove(struct snd_soc_component *component) clk_disable_unprepare(es8316->mclk); } +static int es8316_resume(struct snd_soc_component *component) +{ + struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(es8316->regmap, false); + regcache_sync(es8316->regmap); + + return 0; +} + +static int es8316_suspend(struct snd_soc_component *component) +{ + struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(es8316->regmap, true); + regcache_mark_dirty(es8316->regmap); + + return 0; +} + static const struct snd_soc_component_driver soc_component_dev_es8316 = { .probe = es8316_probe, .remove = es8316_remove, + .resume = es8316_resume, + .suspend = es8316_suspend, .set_jack = es8316_set_jack, .controls = es8316_snd_controls, .num_controls = ARRAY_SIZE(es8316_snd_controls), -- 2.36.1

2 1

[PATCH v2 1/6] sysfs: do not create empty directories if no attributes are present
by Greg Kroah-Hartman 25 Aug '22

25 Aug '22

When creating an attribute group, if it is named a subdirectory is created and the sysfs files are placed into that subdirectory. If no files are created, normally the directory would still be present, but it would be empty. Clean this up by removing the directory if no files were successfully created in the group at all. Cc: Vinod Koul <vkoul(a)kernel.org> Cc: Bard Liao <yung-chuan.liao(a)linux.intel.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Sanyog Kale <sanyog.r.kale(a)intel.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- v2: new patch Note, totally untested! The following soundwire patches will need this, if a soundwire developer could test this out, it would be most apreciated. fs/sysfs/group.c | 19 ++++++++++++++++--- 1 file changed, 16 insertions(+), 3 deletions(-) diff --git a/fs/sysfs/group.c b/fs/sysfs/group.c index eeb0e3099421..9fe0b47db47f 100644 --- a/fs/sysfs/group.c +++ b/fs/sysfs/group.c @@ -31,12 +31,14 @@ static void remove_files(struct kernfs_node *parent, kernfs_remove_by_name(parent, (*bin_attr)->attr.name); } +/* returns -ERROR if error, or >= 0 for number of files actually created */ static int create_files(struct kernfs_node *parent, struct kobject *kobj, kuid_t uid, kgid_t gid, const struct attribute_group *grp, int update) { struct attribute *const *attr; struct bin_attribute *const *bin_attr; + int files_created = 0; int error = 0, i; if (grp->attrs) { @@ -65,6 +67,8 @@ static int create_files(struct kernfs_node *parent, struct kobject *kobj, gid, NULL); if (unlikely(error)) break; + + files_created++; } if (error) { remove_files(parent, grp); @@ -95,12 +99,15 @@ static int create_files(struct kernfs_node *parent, struct kobject *kobj, NULL); if (error) break; + files_created++; } if (error) remove_files(parent, grp); } exit: - return error; + if (error) + return error; + return files_created; } @@ -146,10 +153,16 @@ static int internal_create_group(struct kobject *kobj, int update, kn = kobj->sd; kernfs_get(kn); error = create_files(kn, kobj, uid, gid, grp, update); - if (error) { + if (error <= 0) { + /* + * If an error happened _OR_ if no files were created in the + * attribute group, and we have a name for this group, delete + * the name so there's not an empty directory. + */ if (grp->name) kernfs_remove(kn); - } + } else + error = 0; kernfs_put(kn); if (grp->name && update) -- 2.37.2

2 8

[PATCH AUTOSEL 5.15 02/20] ASoC: sh: rz-ssi: Improve error handling in rz_ssi_probe() error path
by Sasha Levin 25 Aug '22

25 Aug '22

From: Biju Das <biju.das.jz(a)bp.renesas.com> [ Upstream commit c75ed9f54ce8d349fee557f2b471a4d637ed2a6b ] We usually do cleanup in reverse order of init. Currently in case of error rz_ssi_release_dma_channels() done in the reverse order. This patch improves error handling in rz_ssi_probe() error path. While at it, use "goto cleanup" style to reduce code duplication. Reported-by: Pavel Machek <pavel(a)denx.de> Signed-off-by: Biju Das <biju.das.jz(a)bp.renesas.com> Link: https://lore.kernel.org/r/20220728092612.38858-1-biju.das.jz@bp.renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sh/rz-ssi.c | 26 +++++++++++++++----------- 1 file changed, 15 insertions(+), 11 deletions(-) diff --git a/sound/soc/sh/rz-ssi.c b/sound/soc/sh/rz-ssi.c index 6d794eaaf4c3..2e33a1fa0a6f 100644 --- a/sound/soc/sh/rz-ssi.c +++ b/sound/soc/sh/rz-ssi.c @@ -1022,32 +1022,36 @@ static int rz_ssi_probe(struct platform_device *pdev) ssi->rstc = devm_reset_control_get_exclusive(&pdev->dev, NULL); if (IS_ERR(ssi->rstc)) { - rz_ssi_release_dma_channels(ssi); - return PTR_ERR(ssi->rstc); + ret = PTR_ERR(ssi->rstc); + goto err_reset; } reset_control_deassert(ssi->rstc); pm_runtime_enable(&pdev->dev); ret = pm_runtime_resume_and_get(&pdev->dev); if (ret < 0) { - rz_ssi_release_dma_channels(ssi); - pm_runtime_disable(ssi->dev); - reset_control_assert(ssi->rstc); - return dev_err_probe(ssi->dev, ret, "pm_runtime_resume_and_get failed\n"); + dev_err(&pdev->dev, "pm_runtime_resume_and_get failed\n"); + goto err_pm; } ret = devm_snd_soc_register_component(&pdev->dev, &rz_ssi_soc_component, rz_ssi_soc_dai, ARRAY_SIZE(rz_ssi_soc_dai)); if (ret < 0) { - rz_ssi_release_dma_channels(ssi); - - pm_runtime_put(ssi->dev); - pm_runtime_disable(ssi->dev); - reset_control_assert(ssi->rstc); dev_err(&pdev->dev, "failed to register snd component\n"); + goto err_snd_soc; } + return 0; + +err_snd_soc: + pm_runtime_put(ssi->dev); +err_pm: + pm_runtime_disable(ssi->dev); + reset_control_assert(ssi->rstc); +err_reset: + rz_ssi_release_dma_channels(ssi); + return ret; } -- 2.35.1

1 0

[PATCH AUTOSEL 5.19 34/38] ALSA: hda/cs8409: Support new Dolphin Variants
by Sasha Levin 25 Aug '22

25 Aug '22

From: Stefan Binding <sbinding(a)opensource.cirrus.com> [ Upstream commit 1ff954f9ab054675b9eb02dd14add8f7aa376d71 ] Add 4 new Dolphin Systems, same configuration as older systems. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220816151901.1398007-1-sbinding@opensource.cirr… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- 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 4f4cc8215917..5b140301ca66 100644 --- a/sound/pci/hda/patch_cs8409-tables.c +++ b/sound/pci/hda/patch_cs8409-tables.c @@ -546,6 +546,10 @@ const struct snd_pci_quirk cs8409_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0BD6, "Dolphin", CS8409_DOLPHIN), SND_PCI_QUIRK(0x1028, 0x0BD7, "Dolphin", CS8409_DOLPHIN), SND_PCI_QUIRK(0x1028, 0x0BD8, "Dolphin", CS8409_DOLPHIN), + SND_PCI_QUIRK(0x1028, 0x0C43, "Dolphin", CS8409_DOLPHIN), + SND_PCI_QUIRK(0x1028, 0x0C50, "Dolphin", CS8409_DOLPHIN), + SND_PCI_QUIRK(0x1028, 0x0C51, "Dolphin", CS8409_DOLPHIN), + SND_PCI_QUIRK(0x1028, 0x0C52, "Dolphin", CS8409_DOLPHIN), {} /* terminator */ }; -- 2.35.1

1 0

[PATCH AUTOSEL 5.19 18/38] ALSA: hda/realtek: Add quirks for ASUS Zenbooks using CS35L41
by Sasha Levin 25 Aug '22

25 Aug '22

From: Stefan Binding <sbinding(a)opensource.cirrus.com> [ Upstream commit 461122b999bda2ebef2086a35d8990f9ccac5ab8 ] These Asus Zenbook laptop use Realtek HDA codec combined with 2xCS35L41 Amplifiers using SPI. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220815141953.25197-1-sbinding@opensource.cirrus… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 619e6025ba97..7aaa021f550d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9246,6 +9246,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1271, "ASUS X430UN", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1290, "ASUS X441SA", ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x12a0, "ASUS X441UV", ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1043, 0x12af, "ASUS UX582ZS", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x12e0, "ASUS X541SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x12f0, "ASUS X541UV", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1313, "Asus K42JZ", ALC269VB_FIXUP_ASUS_MIC_NO_PRESENCE), @@ -9266,6 +9267,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x19e1, "ASUS UX581LV", ALC295_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x1043, 0x1a30, "ASUS X705UD", ALC256_FIXUP_ASUS_MIC), + SND_PCI_QUIRK(0x1043, 0x1a8f, "ASUS UX582ZS", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1b11, "ASUS UX431DA", ALC294_FIXUP_ASUS_COEF_1B), SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), -- 2.35.1

1 0

[PATCH AUTOSEL 5.19 03/38] ASoC: rt5640: Fix the JD voltage dropping issue
by Sasha Levin 25 Aug '22

25 Aug '22

From: Oder Chiou <oder_chiou(a)realtek.com> [ Upstream commit afb176d45870048eea540991b082208270824037 ] The patch fixes the JD voltage dropping issue in the HDA JD using. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Reported-by: Mohan Kumar D <mkumard(a)nvidia.com> Link: https://lore.kernel.org/r/20220808052836.25791-1-oder_chiou@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5640.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 18b3da9211e3..5ada0d318d0f 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1986,7 +1986,7 @@ static int rt5640_set_bias_level(struct snd_soc_component *component, snd_soc_component_write(component, RT5640_PWR_MIXER, 0x0000); if (rt5640->jd_src == RT5640_JD_SRC_HDA_HEADER) snd_soc_component_write(component, RT5640_PWR_ANLG1, - 0x0018); + 0x2818); else snd_soc_component_write(component, RT5640_PWR_ANLG1, 0x0000); @@ -2592,7 +2592,8 @@ static void rt5640_enable_hda_jack_detect( snd_soc_component_update_bits(component, RT5640_DUMMY1, 0x400, 0x0); snd_soc_component_update_bits(component, RT5640_PWR_ANLG1, - RT5640_PWR_VREF2, RT5640_PWR_VREF2); + RT5640_PWR_VREF2 | RT5640_PWR_MB | RT5640_PWR_BG, + RT5640_PWR_VREF2 | RT5640_PWR_MB | RT5640_PWR_BG); usleep_range(10000, 15000); snd_soc_component_update_bits(component, RT5640_PWR_ANLG1, RT5640_PWR_FV2, RT5640_PWR_FV2); -- 2.35.1

1 0

[PATCH AUTOSEL 5.19 02/38] ASoC: sh: rz-ssi: Improve error handling in rz_ssi_probe() error path
by Sasha Levin 25 Aug '22

25 Aug '22

From: Biju Das <biju.das.jz(a)bp.renesas.com> [ Upstream commit c75ed9f54ce8d349fee557f2b471a4d637ed2a6b ] We usually do cleanup in reverse order of init. Currently in case of error rz_ssi_release_dma_channels() done in the reverse order. This patch improves error handling in rz_ssi_probe() error path. While at it, use "goto cleanup" style to reduce code duplication. Reported-by: Pavel Machek <pavel(a)denx.de> Signed-off-by: Biju Das <biju.das.jz(a)bp.renesas.com> Link: https://lore.kernel.org/r/20220728092612.38858-1-biju.das.jz@bp.renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sh/rz-ssi.c | 26 +++++++++++++++----------- 1 file changed, 15 insertions(+), 11 deletions(-) diff --git a/sound/soc/sh/rz-ssi.c b/sound/soc/sh/rz-ssi.c index e392de7a262e..3d74acffec11 100644 --- a/sound/soc/sh/rz-ssi.c +++ b/sound/soc/sh/rz-ssi.c @@ -1016,32 +1016,36 @@ static int rz_ssi_probe(struct platform_device *pdev) ssi->rstc = devm_reset_control_get_exclusive(&pdev->dev, NULL); if (IS_ERR(ssi->rstc)) { - rz_ssi_release_dma_channels(ssi); - return PTR_ERR(ssi->rstc); + ret = PTR_ERR(ssi->rstc); + goto err_reset; } reset_control_deassert(ssi->rstc); pm_runtime_enable(&pdev->dev); ret = pm_runtime_resume_and_get(&pdev->dev); if (ret < 0) { - rz_ssi_release_dma_channels(ssi); - pm_runtime_disable(ssi->dev); - reset_control_assert(ssi->rstc); - return dev_err_probe(ssi->dev, ret, "pm_runtime_resume_and_get failed\n"); + dev_err(&pdev->dev, "pm_runtime_resume_and_get failed\n"); + goto err_pm; } ret = devm_snd_soc_register_component(&pdev->dev, &rz_ssi_soc_component, rz_ssi_soc_dai, ARRAY_SIZE(rz_ssi_soc_dai)); if (ret < 0) { - rz_ssi_release_dma_channels(ssi); - - pm_runtime_put(ssi->dev); - pm_runtime_disable(ssi->dev); - reset_control_assert(ssi->rstc); dev_err(&pdev->dev, "failed to register snd component\n"); + goto err_snd_soc; } + return 0; + +err_snd_soc: + pm_runtime_put(ssi->dev); +err_pm: + pm_runtime_disable(ssi->dev); + reset_control_assert(ssi->rstc); +err_reset: + rz_ssi_release_dma_channels(ssi); + return ret; } -- 2.35.1

1 0

[PATCH] ARM: pxa: pxa2xx-ac97-lib: fix tests for platform_get_irq() failure
by Yu Zhe 25 Aug '22

25 Aug '22

The platform_get_irq() returns negative error codes. It can't actually return zero. Signed-off-by: Yu Zhe <yuzhe(a)nfschina.com> --- sound/arm/pxa2xx-ac97-lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index e55c0421718b..1f4c4749a6fb 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -402,7 +402,7 @@ int pxa2xx_ac97_hw_probe(struct platform_device *dev) goto err_clk2; irq = platform_get_irq(dev, 0); - if (!irq) + if (irq < 0) goto err_irq; ret = request_irq(irq, pxa2xx_ac97_irq, 0, "AC97", NULL); -- 2.11.0

1 0

[PATCH 0/2] dt-bindings: sound: Add SOF field for mt8186
by Chunxu Li 24 Aug '22

24 Aug '22

From: "chunxu.li" <chunxu.li(a)mediatek.com> These patches add SOF field for mt8186 machine driver chunxu.li (2): ASoC: mediatek: dt-bindings: modify machine bindings for SOF ASoC: mediatek: dt-bindings: modify machine bindings for SOF .../bindings/sound/mt8186-mt6366-da7219-max98357.yaml | 10 ++++++++++ .../bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml | 10 ++++++++++ 2 files changed, 20 insertions(+) -- 2.25.1

3 5

Have I got a missing ucm files problem ?
by GitHub issues - opened 24 Aug '22

24 Aug '22

alsa-project/alsa-ucm-conf issue #204 was opened from xevilstar: Hello, this is my alsa-info http://alsa-project.org/db/?f=cc1f9ff2eb1e223ebf6024aa1605db1d737ff5fa I am having problems with callaudiod not finding the headset and mic on my audio cards and one of the members on the callaudiod git suggested me I am missing some ucm files .... This is a raspberrypi4 can you please help me ? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/204 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 1/5] soundwire: sysfs: move sdw_slave_dev_attr_group into the existing list of groups
by Greg Kroah-Hartman 24 Aug '22

24 Aug '22

The sysfs logic already creates a list of groups for the device, so add the sdw_slave_dev_attr_group group to that list instead of having to do a two-step process of adding a group list and then an individual group. This is a step on the way to moving all of the sysfs attribute handling into the default driver core attribute group logic so that the soundwire core does not have to do any of it manually. Cc: Vinod Koul <vkoul(a)kernel.org> Cc: Bard Liao <yung-chuan.liao(a)linux.intel.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Sanyog Kale <sanyog.r.kale(a)intel.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- drivers/soundwire/sysfs_slave.c | 15 ++++++++++----- 1 file changed, 10 insertions(+), 5 deletions(-) diff --git a/drivers/soundwire/sysfs_slave.c b/drivers/soundwire/sysfs_slave.c index 3210359cd944..83e3f6cc3250 100644 --- a/drivers/soundwire/sysfs_slave.c +++ b/drivers/soundwire/sysfs_slave.c @@ -105,7 +105,10 @@ static struct attribute *slave_attrs[] = { &dev_attr_modalias.attr, NULL, }; -ATTRIBUTE_GROUPS(slave); + +static const struct attribute_group slave_attr_group = { + .attrs = slave_attrs, +}; static struct attribute *slave_dev_attrs[] = { &dev_attr_mipi_revision.attr, @@ -190,6 +193,12 @@ static const struct attribute_group dp0_group = { .name = "dp0", }; +static const struct attribute_group *slave_groups[] = { + &slave_attr_group, + &sdw_slave_dev_attr_group, + NULL, +}; + int sdw_slave_sysfs_init(struct sdw_slave *slave) { int ret; @@ -198,10 +207,6 @@ int sdw_slave_sysfs_init(struct sdw_slave *slave) if (ret < 0) return ret; - ret = devm_device_add_group(&slave->dev, &sdw_slave_dev_attr_group); - if (ret < 0) - return ret; - if (slave->prop.dp0_prop) { ret = devm_device_add_group(&slave->dev, &dp0_group); if (ret < 0) -- 2.37.1

3 23

[PATCH -next 1/2] ASoC: SOF: imx8ulp: change es8326_regmap_config to static
by Yang Yingliang 24 Aug '22

24 Aug '22

sof_imx8ulp_ops is only used in imx8ulp.c now, change it to static. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/sof/imx/imx8ulp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/imx/imx8ulp.c b/sound/soc/sof/imx/imx8ulp.c index 02b496165acc..afab7feab5fb 100644 --- a/sound/soc/sof/imx/imx8ulp.c +++ b/sound/soc/sof/imx/imx8ulp.c @@ -413,7 +413,7 @@ static int imx8ulp_dsp_set_power_state(struct snd_sof_dev *sdev, } /* i.MX8 ops */ -struct snd_sof_dsp_ops sof_imx8ulp_ops = { +static struct snd_sof_dsp_ops sof_imx8ulp_ops = { /* probe and remove */ .probe = imx8ulp_probe, .remove = imx8ulp_remove, -- 2.25.1

2 2

[PATCH 1/2] ALSA: seq: oss: Fix data-race for max_midi_devs access
by Takashi Iwai 24 Aug '22

24 Aug '22

ALSA OSS sequencer refers to a global variable max_midi_devs at creating a new port, storing it to its own field. Meanwhile this variable may be changed by other sequencer events at snd_seq_oss_midi_check_exit_port() in parallel, which may cause a data race. OTOH, this data race itself is almost harmless, as the access to the MIDI device is done via get_mdev() and it's protected with a refcount, hence its presence is guaranteed. Though, it's sill better to address the data-race from the code sanity POV, and this patch adds the proper spinlock for the protection. Reported-by: Abhishek Shah <abhishek.shah(a)columbia.edu> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/CAEHB2493pZRXs863w58QWnUTtv3HHfg85aYhLn5HJHCwxqtH… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/oss/seq_oss_midi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c index 1e3bf086f867..07efb38f58ac 100644 --- a/sound/core/seq/oss/seq_oss_midi.c +++ b/sound/core/seq/oss/seq_oss_midi.c @@ -270,7 +270,9 @@ snd_seq_oss_midi_clear_all(void) void snd_seq_oss_midi_setup(struct seq_oss_devinfo *dp) { + spin_lock_irq(&register_lock); dp->max_mididev = max_midi_devs; + spin_unlock_irq(&register_lock); } /* -- 2.35.3

2 5

[PATCH 0/4] ASoC: SOF: Intel: override mclk_id for ES8336 support
by Pierre-Louis Bossart 24 Aug '22

24 Aug '22

This patchset solves a known issue with ES8336 platforms wrt MCLK selection. Most of the devices use the MCLK0 signal, but some devices do use the MCLK1 signal. The MCLK is defined in the topology, it would be a nightmare to generate more topology files just for one MCLK difference. With a minor extension to the intel-nhlt library, the MCLK information can be found by parsing the NHLT table, and we can override the mclk_id at boot time. The only known issues for this platform remain the detection of GPIO and microphone connections, currently only possible with manual quirks. Thanks to Eugene J. Markow for testing this patchset. Pierre-Louis Bossart (4): ASoC: SOF: add quirk to override topology mclk_id ALSA: hda: intel-nhlt: add intel_nhlt_ssp_mclk_mask() ASoC: SOF: Intel: hda: override mclk_id after parsing NHLT SSP blob ASoC: SOF: Intel: hda: refine SSP count support include/sound/intel-nhlt.h | 7 ++++ sound/hda/intel-nhlt.c | 61 +++++++++++++++++++++++++++++++++++ sound/soc/sof/intel/hda.c | 34 +++++++++++++++++++ sound/soc/sof/intel/hda.h | 2 ++ sound/soc/sof/intel/mtl.c | 2 +- sound/soc/sof/intel/tgl.c | 8 ++--- sound/soc/sof/ipc3-topology.c | 7 ++++ sound/soc/sof/sof-priv.h | 4 +++ 8 files changed, 120 insertions(+), 5 deletions(-) -- 2.34.1

3 12

[PATCH v2] ASoC: codes: src4xxx: Avoid clang -Wsometimes-uninitialized in src4xxx_hw_params()
by Nathan Chancellor 24 Aug '22

24 Aug '22

Clang warns: sound/soc/codecs/src4xxx.c:280:3: error: variable 'd' is used uninitialized whenever switch default is taken [-Werror,-Wsometimes-uninitialized] default: ^~~~~~~ sound/soc/codecs/src4xxx.c:298:59: note: uninitialized use occurs here ret = regmap_write(src4xxx->regmap, SRC4XXX_RCV_PLL_11, d); ^ sound/soc/codecs/src4xxx.c:223:20: note: initialize the variable 'd' to silence this warning int val, pj, jd, d; ^ = 0 sound/soc/codecs/src4xxx.c:280:3: error: variable 'jd' is used uninitialized whenever switch default is taken [-Werror,-Wsometimes-uninitialized] default: ^~~~~~~ sound/soc/codecs/src4xxx.c:293:59: note: uninitialized use occurs here ret = regmap_write(src4xxx->regmap, SRC4XXX_RCV_PLL_10, jd); ^~ sound/soc/codecs/src4xxx.c:223:17: note: initialize the variable 'jd' to silence this warning int val, pj, jd, d; ^ = 0 sound/soc/codecs/src4xxx.c:280:3: error: variable 'pj' is used uninitialized whenever switch default is taken [-Werror,-Wsometimes-uninitialized] default: ^~~~~~~ sound/soc/codecs/src4xxx.c:288:59: note: uninitialized use occurs here ret = regmap_write(src4xxx->regmap, SRC4XXX_RCV_PLL_0F, pj); ^~ sound/soc/codecs/src4xxx.c:223:13: note: initialize the variable 'pj' to silence this warning int val, pj, jd, d; ^ = 0 3 errors generated. The datasheet does not have any default values for these regmap values so pick some arbitrary values and print to the user that this is the case to silence the warnings. Link: https://github.com/ClangBuiltLinux/linux/issues/1691 Reported-by: kernel test robot <lkp(a)intel.com> Reported-by: "Sudip Mukherjee (Codethink)" <sudipm.mukherjee(a)gmail.com> Suggested-by: Matt Flax <flatmax(a)flatmax.com> Signed-off-by: Nathan Chancellor <nathan(a)kernel.org> --- v1 -> v2: https://lore.kernel.org/20220822183101.1115095-1-nathan@kernel.org/ * Don't return early, just initialize the values to some arbitrary numbers and try to hobble along, as other parts of the chip may be functional. * Add message and comment to describe this situation. sound/soc/codecs/src4xxx.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/src4xxx.c b/sound/soc/codecs/src4xxx.c index a8f143057b41..db4e280dd055 100644 --- a/sound/soc/codecs/src4xxx.c +++ b/sound/soc/codecs/src4xxx.c @@ -280,9 +280,14 @@ static int src4xxx_hw_params(struct snd_pcm_substream *substream, default: /* don't error out here, * other parts of the chip are still functional + * Dummy initialize variables to avoid + * -Wsometimes-uninitialized from clang. */ dev_info(component->dev, - "Couldn't set the RCV PLL as this master clock rate is unknown\n"); + "Couldn't set the RCV PLL as this master clock rate is unknown. Chosen regmap values may not match real world values.\n"); + pj = 0x0; + jd = 0xff; + d = 0xff; break; } ret = regmap_write(src4xxx->regmap, SRC4XXX_RCV_PLL_0F, pj); base-commit: 94f072748337424c9cf92cd018532a34db3a5516 -- 2.37.2

3 3

[PATCH] ASoC: fsl_sai: Update slots number according to bclk_ratio
by Shengjiu Wang 24 Aug '22

24 Aug '22

The bclk_ratio is set by .set_bclk_ratio API. bclk_ratio = slots * slot_width So if slots is not set by .set_tdm_slot, then it can be calculated by bclk_ratio. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 7523bb944b21..acf76ea21138 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -535,12 +535,14 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, u32 pins, bclk; int ret, i; - if (sai->slots) - slots = sai->slots; - if (sai->slot_width) slot_width = sai->slot_width; + if (sai->slots) + slots = sai->slots; + else if (sai->bclk_ratio) + slots = sai->bclk_ratio / slot_width; + pins = DIV_ROUND_UP(channels, slots); /* -- 2.17.1

2 1

[PATCH] ASoC: SOF: imx: imx8ulp: declare ops structure as static
by Pierre-Louis Bossart 24 Aug '22

24 Aug '22

Sparse warning: sound/soc/sof/imx/imx8ulp.c:416:24: error: symbol 'sof_imx8ulp_ops' was not declared. Should it be static? Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/imx/imx8ulp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/imx/imx8ulp.c b/sound/soc/sof/imx/imx8ulp.c index 02b496165acc3..afab7feab5fb3 100644 --- a/sound/soc/sof/imx/imx8ulp.c +++ b/sound/soc/sof/imx/imx8ulp.c @@ -413,7 +413,7 @@ static int imx8ulp_dsp_set_power_state(struct snd_sof_dev *sdev, } /* i.MX8 ops */ -struct snd_sof_dsp_ops sof_imx8ulp_ops = { +static struct snd_sof_dsp_ops sof_imx8ulp_ops = { /* probe and remove */ .probe = imx8ulp_probe, .remove = imx8ulp_remove, -- 2.34.1

2 1

[PATCH] ASoC: dt-bindings: Add missing (unevaluated|additional)Properties on child nodes
by Rob Herring 24 Aug '22

24 Aug '22

In order to ensure only documented properties are present, node schemas must have unevaluatedProperties or additionalProperties set to false (typically). Signed-off-by: Rob Herring <robh(a)kernel.org> --- .../bindings/sound/amlogic,gx-sound-card.yaml | 2 ++ .../bindings/sound/audio-graph-port.yaml | 2 ++ .../bindings/sound/google,sc7180-trogdor.yaml | 4 ++++ .../devicetree/bindings/sound/imx-audio-card.yaml | 2 ++ .../sound/mt8192-mt6359-rt1015-rt5682.yaml | 4 ++++ .../devicetree/bindings/sound/qcom,sm8250.yaml | 6 ++++++ .../devicetree/bindings/sound/renesas,rsnd.yaml | 14 ++++++++++++-- .../bindings/sound/samsung,aries-wm8994.yaml | 2 ++ .../bindings/sound/samsung,midas-audio.yaml | 2 ++ .../devicetree/bindings/sound/samsung,snow.yaml | 2 ++ .../devicetree/bindings/sound/st,stm32-sai.yaml | 1 + 11 files changed, 39 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml b/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml index b4b35edcb493..5b8d59245f82 100644 --- a/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml +++ b/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml @@ -40,6 +40,7 @@ properties: patternProperties: "^dai-link-[0-9]+$": type: object + additionalProperties: false description: |- dai-link child nodes: Container for dai-link level properties and the CODEC sub-nodes. @@ -63,6 +64,7 @@ patternProperties: patternProperties: "^codec-[0-9]+$": type: object + additionalProperties: false description: |- Codecs: dai-link representing backend links should have at least one subnode. diff --git a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml index 5c368674d11a..c54ca5eb0baa 100644 --- a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml +++ b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml @@ -27,6 +27,8 @@ properties: patternProperties: "^endpoint(@[0-9a-f]+)?": $ref: /schemas/graph.yaml#/$defs/endpoint-base + unevaluatedProperties: false + properties: mclk-fs: description: | diff --git a/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml b/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml index 233caa0ade87..67ccddd44489 100644 --- a/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml +++ b/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml @@ -61,6 +61,8 @@ patternProperties: cpu: description: Holds subnode which indicates cpu dai. type: object + additionalProperties: false + properties: sound-dai: maxItems: 1 @@ -68,6 +70,8 @@ patternProperties: codec: description: Holds subnode which indicates codec dai. type: object + additionalProperties: false + properties: sound-dai: maxItems: 1 diff --git a/Documentation/devicetree/bindings/sound/imx-audio-card.yaml b/Documentation/devicetree/bindings/sound/imx-audio-card.yaml index bb3a435722c7..b6f5d486600e 100644 --- a/Documentation/devicetree/bindings/sound/imx-audio-card.yaml +++ b/Documentation/devicetree/bindings/sound/imx-audio-card.yaml @@ -58,6 +58,7 @@ patternProperties: cpu: description: Holds subnode which indicates cpu dai. type: object + additionalProperties: false properties: sound-dai: maxItems: 1 @@ -65,6 +66,7 @@ patternProperties: codec: description: Holds subnode which indicates codec dai. type: object + additionalProperties: false properties: sound-dai: minItems: 1 diff --git a/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml b/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml index 4fa179909c62..478be7e3fa29 100644 --- a/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml +++ b/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml @@ -30,6 +30,8 @@ properties: headset-codec: type: object + additionalProperties: false + properties: sound-dai: $ref: /schemas/types.yaml#/definitions/phandle @@ -38,6 +40,8 @@ properties: speaker-codecs: type: object + additionalProperties: false + properties: sound-dai: minItems: 1 diff --git a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml index e6e27d09783e..a3a4289f713e 100644 --- a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml @@ -71,6 +71,8 @@ patternProperties: cpu: description: Holds subnode which indicates cpu dai. type: object + additionalProperties: false + properties: sound-dai: maxItems: 1 @@ -78,6 +80,8 @@ patternProperties: platform: description: Holds subnode which indicates platform dai. type: object + additionalProperties: false + properties: sound-dai: maxItems: 1 @@ -85,6 +89,8 @@ patternProperties: codec: description: Holds subnode which indicates codec dai. type: object + additionalProperties: false + properties: sound-dai: minItems: 1 diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml index e17c0245f77a..268895c90bd5 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml @@ -129,6 +129,8 @@ properties: patternProperties: "^dvc-[0-1]$": type: object + additionalProperties: false + properties: dmas: maxItems: 1 @@ -145,7 +147,7 @@ properties: patternProperties: "^mix-[0-1]$": type: object - # no properties + additionalProperties: false additionalProperties: false rcar_sound,ctu: @@ -154,7 +156,7 @@ properties: patternProperties: "^ctu-[0-7]$": type: object - # no properties + additionalProperties: false additionalProperties: false rcar_sound,src: @@ -163,6 +165,8 @@ properties: patternProperties: "^src-[0-9]$": type: object + additionalProperties: false + properties: interrupts: maxItems: 1 @@ -186,6 +190,8 @@ properties: patternProperties: "^ssiu-[0-9]+$": type: object + additionalProperties: false + properties: dmas: maxItems: 2 @@ -206,6 +212,8 @@ properties: patternProperties: "^ssi-[0-9]$": type: object + additionalProperties: false + properties: interrupts: maxItems: 1 @@ -243,6 +251,8 @@ properties: patternProperties: "^dai([0-9]+)?$": type: object + additionalProperties: false + properties: playback: $ref: /schemas/types.yaml#/definitions/phandle-array diff --git a/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml b/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml index a01c4ad929b8..447e013f6e17 100644 --- a/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml +++ b/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml @@ -23,6 +23,7 @@ properties: cpu: type: object + additionalProperties: false properties: sound-dai: minItems: 2 @@ -34,6 +35,7 @@ properties: - sound-dai codec: + additionalProperties: false type: object properties: sound-dai: diff --git a/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml b/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml index ec50bcb4af5f..31095913e330 100644 --- a/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml +++ b/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml @@ -19,6 +19,7 @@ properties: cpu: type: object + additionalProperties: false properties: sound-dai: maxItems: 1 @@ -28,6 +29,7 @@ properties: codec: type: object + additionalProperties: false properties: sound-dai: maxItems: 1 diff --git a/Documentation/devicetree/bindings/sound/samsung,snow.yaml b/Documentation/devicetree/bindings/sound/samsung,snow.yaml index 51a83d3c7274..3d49aa4c9be2 100644 --- a/Documentation/devicetree/bindings/sound/samsung,snow.yaml +++ b/Documentation/devicetree/bindings/sound/samsung,snow.yaml @@ -19,6 +19,7 @@ properties: codec: type: object + additionalProperties: false properties: sound-dai: description: List of phandles to the CODEC and HDMI IP nodes. @@ -30,6 +31,7 @@ properties: cpu: type: object + additionalProperties: false properties: sound-dai: description: Phandle to the Samsung I2S controller. diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml index fe2e15504ebc..1a3abc949505 100644 --- a/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml +++ b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml @@ -60,6 +60,7 @@ required: patternProperties: "^audio-controller@[0-9a-f]+$": type: object + additionalProperties: false description: Two subnodes corresponding to SAI sub-block instances A et B can be defined. Subnode can be omitted for unsused sub-block. -- 2.34.1

2 1

[PATCH] ASoC: SOF: Intel: hda: Skip IMR boot after a firmware crash or boot failure
by Peter Ujfalusi 24 Aug '22

24 Aug '22

To make sure that we start from a clean state next time when the DSP is powered up after a firmware crash or boot failure we must skip the IMR booting attempt. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(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/sof/intel/hda-dsp.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index eddfd77ad90f..671c3e02d7df 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -620,8 +620,13 @@ static int hda_suspend(struct snd_sof_dev *sdev, bool runtime_suspend) /* * The memory used for IMR boot loses its content in deeper than S3 state * We must not try IMR boot on next power up (as it will fail). + * + * In case of firmware crash or boot failure set the skip_imr_boot to true + * as well in order to try to re-load the firmware to do a 'cold' boot. */ - if (sdev->system_suspend_target > SOF_SUSPEND_S3) + if (sdev->system_suspend_target > SOF_SUSPEND_S3 || + sdev->fw_state == SOF_FW_CRASHED || + sdev->fw_state == SOF_FW_BOOT_FAILED) hda->skip_imr_boot = true; hda_sdw_int_enable(sdev, false); -- 2.37.2

2 1

[PATCH] ASoC: SOF: ipc4-loader: Verify ext manifest magic number
by Peter Ujfalusi 24 Aug '22

24 Aug '22

Firmware image must start with an extended manifest. Add a check to make sure that the image does contain it. The magic number (the first u32 of a firmware image if manifest is present) for an IPC4 image must be 0x31454124 (ASCI "$AE1"). Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(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> --- sound/soc/sof/ipc4-loader.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/sof/ipc4-loader.c b/sound/soc/sof/ipc4-loader.c index 9fadae8fd011..8bd2132b4f41 100644 --- a/sound/soc/sof/ipc4-loader.c +++ b/sound/soc/sof/ipc4-loader.c @@ -40,6 +40,17 @@ static size_t sof_ipc4_fw_parse_ext_man(struct snd_sof_dev *sdev) ext_man_hdr = (struct sof_ext_manifest4_hdr *)fw->data; + /* + * At the start of the firmware image we must have an extended manifest. + * Verify that the magic number is correct. + */ + if (ext_man_hdr->id != SOF_EXT_MAN4_MAGIC_NUMBER) { + dev_err(sdev->dev, + "Unexpected extended manifest magic number: %#x\n", + ext_man_hdr->id); + return -EINVAL; + } + fw_hdr_offset = ipc4_data->manifest_fw_hdr_offset; if (!fw_hdr_offset) return -EINVAL; -- 2.37.2

2 1

[PATCH] ALSA: control: Fix an out-of-bounds bug in get_ctl_id_hash()
by Zheyu Ma 24 Aug '22

24 Aug '22

Since the user can control the arguments provided to the kernel by the ioctl() system call, an out-of-bounds bug occurs when the 'id->name' provided by the user does not end with '\0'. The following log can reveal it: [ 10.002313] BUG: KASAN: stack-out-of-bounds in snd_ctl_find_id+0x36c/0x3a0 [ 10.002895] Read of size 1 at addr ffff888109f5fe28 by task snd/439 [ 10.004934] Call Trace: [ 10.007140] snd_ctl_find_id+0x36c/0x3a0 [ 10.007489] snd_ctl_ioctl+0x6cf/0x10e0 Fix this by checking the bound of 'id->name' in the loop. Fixes: c27e1efb61c5 ("ALSA: control: Use xarray for faster lookups") Signed-off-by: Zheyu Ma <zheyuma97(a)gmail.com> --- sound/core/control.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index f3e893715369..e8fc4c511e5f 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -385,14 +385,14 @@ static bool elem_id_matches(const struct snd_kcontrol *kctl, #define MULTIPLIER 37 static unsigned long get_ctl_id_hash(const struct snd_ctl_elem_id *id) { + int i; unsigned long h; - const unsigned char *p; h = id->iface; h = MULTIPLIER * h + id->device; h = MULTIPLIER * h + id->subdevice; - for (p = id->name; *p; p++) - h = MULTIPLIER * h + *p; + for (i = 0; id->name[i] && i < SNDRV_CTL_ELEM_ID_NAME_MAXLEN; i++) + h = MULTIPLIER * h + id->name[i]; h = MULTIPLIER * h + id->index; h &= LONG_MAX; return h; -- 2.25.1

2 1

arcam_av.c: Include missing string.h
by GitHub pull_request - opened 24 Aug '22

24 Aug '22

alsa-project/alsa-plugins pull request #47 was opened from kraj: bzero() function needs this header to be included Signed-off-by: Khem Raj <raj.khem(a)gmail.com> Request URL : https://github.com/alsa-project/alsa-plugins/pull/47 Patch URL : https://github.com/alsa-project/alsa-plugins/pull/47.patch Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH v2] drm/bridge: dw-hdmi-i2s: set insert_pcuv bit if hardware supports it
by Geraldo Nascimento 23 Aug '22

23 Aug '22

Starting with version 2.10a of Synopsys DesignWare HDMI controller the insert_pcuv bit was introduced. On RK3399pro SoM (Radxa Rock Pi N10) for example if we neglect to set this bit and proceed to enable hdmi_sound and i2s2 on the device tree there will be extreme clipping of sound output, to the point that music sounds like white noise. Problem could also manifest as just mild cracking depending of HDMI audio implementation of sink. Setting insert_pcuv bit (bit 2 of aud_conf2 Audio Sample register) fixes this. Signed-off-by: Geraldo Nascimento <geraldogabriel(a)gmail.com> --- v1->v2: SoC->SoM on description, better commenting, minor style changes, conditional application of fix for L-PCM only --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c @@ -42,6 +42,7 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, struct dw_hdmi *hdmi = audio->hdmi; u8 conf0 = 0; u8 conf1 = 0; + u8 conf2 = 0; u8 inputclkfs = 0; /* it cares I2S only */ @@ -101,6 +102,16 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, return -EINVAL; } + /* + * dw-hdmi introduced insert_pcuv bit in + * version 2.10a. + * + * This single bit (bit 2 of HDMI_AUD_CONF2) + * when set to 1 will enable the insertion of the PCUV + * (Parity, Channel Status, User bit and Validity) + * bits on the incoming audio stream. + * + * Support is limited to Linear PCM audio. If + * neglected, the lack of valid PCUV bits + * on L-PCM streams will cause anything from + * mild cracking to full blown extreme + * clipping depending of HDMI audio + * implementation of the sink. + * + */ + + if (hdmi_read(audio, HDMI_DESIGN_ID) >= 0x21 && + !(hparms->iec.status[0] & IEC958_AES0_NONAUDIO)) + conf2 = HDMI_AUD_CONF2_INSERT_PCUV; + dw_hdmi_set_sample_rate(hdmi, hparms->sample_rate); dw_hdmi_set_channel_status(hdmi, hparms->iec.status); dw_hdmi_set_channel_count(hdmi, hparms->channels); @@ -109,6 +120,7 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, hdmi_write(audio, inputclkfs, HDMI_AUD_INPUTCLKFS); hdmi_write(audio, conf0, HDMI_AUD_CONF0); hdmi_write(audio, conf1, HDMI_AUD_CONF1); + hdmi_write(audio, conf2, HDMI_AUD_CONF2); return 0; } --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h @@ -931,6 +931,11 @@ enum { HDMI_AUD_CONF1_WIDTH_16 = 0x10, HDMI_AUD_CONF1_WIDTH_24 = 0x18, +/* AUD_CONF2 field values */ + HDMI_AUD_CONF2_HBR = 0x01, + HDMI_AUD_CONF2_NLPCM = 0x02, + HDMI_AUD_CONF2_INSERT_PCUV = 0x04, + /* AUD_CTS3 field values */ HDMI_AUD_CTS3_N_SHIFT_OFFSET = 5, HDMI_AUD_CTS3_N_SHIFT_MASK = 0xe0,

1 0

[PATCH 0/3] ASoC: SOF: Intel: hda: Correct Firmware State Register use
by Peter Ujfalusi 23 Aug '22

23 Aug '22

Hi, The FSR (Firmware State Register) holds the ROM state information, it does not contain error information. The FSR itself is a bit more complicated as well as the state depends on the module currently in use. The error code from ROM or the status code from the firmware is located at the next register. Fix the handling of the FSR in order to provide usable and human readable (in most cases) report on the status and error. Regards, Peter --- Peter Ujfalusi (3): ASoC: SOF: Intel: hda: Correct the ROM/FW state reporting code ASoC: SOF: Intel: hda-loader: Use the FSR state definitions during bootup ASoC: SOF: Intel: hda: Drop no longer used ROM state definitions sound/soc/sof/intel/hda-loader.c | 10 +-- sound/soc/sof/intel/hda.c | 147 ++++++++++++++++++++++++++----- sound/soc/sof/intel/hda.h | 69 +++++++++++++-- 3 files changed, 194 insertions(+), 32 deletions(-) -- 2.37.0

2 5

[PATCH] ASoC: mediatek: mt8186: rename sound card name
by Chunxu Li 23 Aug '22

23 Aug '22

From: "chunxu.li" <chunxu.li(a)mediatek.com> The field 'topology_shortname' in 'snd_soc_card' is defined as char[32], Current card name will be truncated when SOF is enabled, so rename the sound card name. Signed-off-by: chunxu.li <chunxu.li(a)mediatek.com> --- sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c | 2 +- sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c b/sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c index 84ee5d95a9f0..17a15bec41da 100644 --- a/sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c +++ b/sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c @@ -962,7 +962,7 @@ mt8186_mt6366_da7219_max98357_controls[] = { }; static struct snd_soc_card mt8186_mt6366_da7219_max98357_soc_card = { - .name = "mt8186_mt6366_da7219_max98357", + .name = "mt8186_da7219_max98357", .owner = THIS_MODULE, .dai_link = mt8186_mt6366_da7219_max98357_dai_links, .num_links = ARRAY_SIZE(mt8186_mt6366_da7219_max98357_dai_links), diff --git a/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c b/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c index 6c41706a5621..393d179d61de 100644 --- a/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c +++ b/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c @@ -938,7 +938,7 @@ mt8186_mt6366_rt1019_rt5682s_controls[] = { }; static struct snd_soc_card mt8186_mt6366_rt1019_rt5682s_soc_card = { - .name = "mt8186_mt6366_rt1019_rt5682s", + .name = "mt8186_rt1019_rt5682s", .owner = THIS_MODULE, .dai_link = mt8186_mt6366_rt1019_rt5682s_dai_links, .num_links = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_dai_links), -- 2.25.1

2 1

[PATCH linux-next] ASoC: atmel_ssc_dai: Remove the unneeded result variable
by cgel.zte＠gmail.com 23 Aug '22

23 Aug '22

From: ye xingchen <ye.xingchen(a)zte.com.cn> Return the value from asoc_ssc_init() directly instead of storing it in another redundant variable. Reported-by: Zeal Robot <zealci(a)zte.com.cn> Signed-off-by: ye xingchen <ye.xingchen(a)zte.com.cn> --- sound/soc/atmel/atmel_ssc_dai.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index e868b7e028d6..3763454436c1 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -891,7 +891,6 @@ static int asoc_ssc_init(struct device *dev) int atmel_ssc_set_audio(int ssc_id) { struct ssc_device *ssc; - int ret; /* If we can grab the SSC briefly to parent the DAI device off it */ ssc = ssc_request(ssc_id); @@ -903,9 +902,7 @@ int atmel_ssc_set_audio(int ssc_id) ssc_info[ssc_id].ssc = ssc; } - ret = asoc_ssc_init(&ssc->pdev->dev); - - return ret; + return asoc_ssc_init(&ssc->pdev->dev); } EXPORT_SYMBOL_GPL(atmel_ssc_set_audio); -- 2.25.1

2 1

[RESEND PATCH] ASoC: fsl_sai: Add support multi fifo sdma script
by Shengjiu Wang 23 Aug '22

23 Aug '22

With disabling combine mode, the multiple successive FIFO registers or non successive FIFO registers of SAI module can work with the sdma multi fifo script. This patch is to configure the necessary information to the SDMA engine driver for support multi fifo script. 'words_per_fifo' is the channels for each dataline 'n_fifos_src' and 'n_fifos_dst' are the fifo number 'stride_fifos_src' and 'stride_fifos_dst' are the stride between enable FIFOs 'maxburst' is the multiply of datalines Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in resend: - rebase to latest for-6.1 sound/soc/fsl/fsl_sai.c | 37 ++++++++++++++++++++++++++++++++++++- sound/soc/fsl/fsl_sai.h | 3 +++ 2 files changed, 39 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index a7fa6f0bf83d..0f92906d7a29 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -527,6 +527,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, u32 slot_width = word_width; int adir = tx ? RX : TX; u32 pins, bclk; + u32 watermark; int ret, i; if (sai->slots) @@ -619,7 +620,15 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, FSL_SAI_CR5_FBT_MASK, val_cr5); } - if (hweight8(dl_cfg[dl_cfg_idx].mask[tx]) <= 1) + /* + * Combine mode has limation: + * - Can't used for singel dataline/FIFO case except the FIFO0 + * - Can't used for multi dataline/FIFO case except the enabled FIFOs + * are successive and start from FIFO0 + * + * So for common usage, all multi fifo case disable the combine mode. + */ + if (hweight8(dl_cfg[dl_cfg_idx].mask[tx]) <= 1 || sai->is_multi_fifo_dma) regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx, ofs), FSL_SAI_CR4_FCOMB_MASK, 0); else @@ -630,6 +639,26 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, dma_params->addr = sai->res->start + FSL_SAI_xDR0(tx) + dl_cfg[dl_cfg_idx].start_off[tx] * 0x4; + if (sai->is_multi_fifo_dma) { + sai->audio_config[tx].words_per_fifo = min(slots, channels); + if (tx) { + sai->audio_config[tx].n_fifos_dst = pins; + sai->audio_config[tx].stride_fifos_dst = dl_cfg[dl_cfg_idx].next_off[tx]; + } else { + sai->audio_config[tx].n_fifos_src = pins; + sai->audio_config[tx].stride_fifos_src = dl_cfg[dl_cfg_idx].next_off[tx]; + } + dma_params->maxburst = sai->audio_config[tx].words_per_fifo * pins; + dma_params->peripheral_config = &sai->audio_config[tx]; + dma_params->peripheral_size = sizeof(sai->audio_config[tx]); + + watermark = tx ? (sai->soc_data->fifo_depth - dma_params->maxburst) : + (dma_params->maxburst - 1); + regmap_update_bits(sai->regmap, FSL_SAI_xCR1(tx, ofs), + FSL_SAI_CR1_RFW_MASK(sai->soc_data->fifo_depth), + watermark); + } + /* Find a proper tcre setting */ for (i = 0; i < sai->soc_data->pins; i++) { trce_mask = (1 << (i + 1)) - 1; @@ -1257,6 +1286,7 @@ static int fsl_sai_probe(struct platform_device *pdev) char tmp[8]; int irq, ret, i; int index; + u32 dmas[4]; sai = devm_kzalloc(dev, sizeof(*sai), GFP_KERNEL); if (!sai) @@ -1313,6 +1343,11 @@ static int fsl_sai_probe(struct platform_device *pdev) fsl_asoc_get_pll_clocks(&pdev->dev, &sai->pll8k_clk, &sai->pll11k_clk); + /* Use Multi FIFO mode depending on the support from SDMA script */ + ret = of_property_read_u32_array(np, "dmas", dmas, 4); + if (!sai->soc_data->use_edma && !ret && dmas[2] == IMX_DMATYPE_MULTI_SAI) + sai->is_multi_fifo_dma = true; + /* read dataline mask for rx and tx*/ ret = fsl_sai_read_dlcfg(sai); if (ret < 0) { diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 17956b5731dc..697f6690068c 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -6,6 +6,7 @@ #ifndef __FSL_SAI_H #define __FSL_SAI_H +#include <linux/dma/imx-dma.h> #include <sound/dmaengine_pcm.h> #define FSL_SAI_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ @@ -281,6 +282,7 @@ struct fsl_sai { bool is_lsb_first; bool is_dsp_mode; bool is_pdm_mode; + bool is_multi_fifo_dma; bool synchronous[2]; struct fsl_sai_dl_cfg *dl_cfg; unsigned int dl_cfg_cnt; @@ -300,6 +302,7 @@ struct fsl_sai { struct pm_qos_request pm_qos_req; struct pinctrl *pinctrl; struct pinctrl_state *pins_state; + struct sdma_peripheral_config audio_config[2]; }; #define TX 1 -- 2.34.1

2 1

[PATCH] ASoC: SOF: compress: Remove dai_posn variable
by Daniel Baluta 23 Aug '22

23 Aug '22

From: Daniel Baluta <daniel.baluta(a)nxp.com> dai_posn is set but never used. Initial intention was to use dai_posn to shorthen one code line but it looks fine without it too. Fixes: c1a731c71359 ("ASoC: SOF: compress: Add support for computing timestamps") Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/sof/compress.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/sof/compress.c b/sound/soc/sof/compress.c index 174b3d8e67dd..c8ae778a50d1 100644 --- a/sound/soc/sof/compress.c +++ b/sound/soc/sof/compress.c @@ -327,7 +327,6 @@ static int sof_compr_pointer(struct snd_soc_component *component, struct snd_compr_stream *cstream, struct snd_compr_tstamp *tstamp) { - u64 dai_posn; struct snd_sof_pcm *spcm; struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct sof_compr_stream *sstream = cstream->runtime->private_data; @@ -336,8 +335,6 @@ static int sof_compr_pointer(struct snd_soc_component *component, if (!spcm) return -EINVAL; - dai_posn = spcm->stream[cstream->direction].posn.dai_posn; - tstamp->sampling_rate = sstream->sampling_rate; tstamp->copied_total = sstream->copied_total; tstamp->pcm_io_frames = div_u64(spcm->stream[cstream->direction].posn.dai_posn, -- 2.27.0

2 1

[PATCH] ASoC: fsl_sai: Add support multi fifo sdma script
by Shengjiu Wang 23 Aug '22

23 Aug '22

With disabling combine mode, the multiple successive FIFO registers or non successive FIFO registers of SAI module can work with the sdma multi fifo script. This patch is to configure the necessary information to the SDMA engine driver for support multi fifo script. 'words_per_fifo' is the channels for each dataline 'n_fifos_src' and 'n_fifos_dst' are the fifo number 'stride_fifos_src' and 'stride_fifos_dst' are the stride between enable FIFOs 'maxburst' is the multiply of datalines Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 37 ++++++++++++++++++++++++++++++++++++- sound/soc/fsl/fsl_sai.h | 3 +++ 2 files changed, 39 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index acf76ea21138..a2b375e2b56c 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -533,6 +533,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, u32 slot_width = word_width; int adir = tx ? RX : TX; u32 pins, bclk; + u32 watermark; int ret, i; if (sai->slot_width) @@ -627,7 +628,15 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, FSL_SAI_CR5_FBT_MASK, val_cr5); } - if (hweight8(dl_cfg[dl_cfg_idx].mask[tx]) <= 1) + /* + * Combine mode has limation: + * - Can't used for singel dataline/FIFO case except the FIFO0 + * - Can't used for multi dataline/FIFO case except the enabled FIFOs + * are successive and start from FIFO0 + * + * So for common usage, all multi fifo case disable the combine mode. + */ + if (hweight8(dl_cfg[dl_cfg_idx].mask[tx]) <= 1 || sai->is_multi_fifo_dma) regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx, ofs), FSL_SAI_CR4_FCOMB_MASK, 0); else @@ -638,6 +647,26 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, dma_params->addr = sai->res->start + FSL_SAI_xDR0(tx) + dl_cfg[dl_cfg_idx].start_off[tx] * 0x4; + if (sai->is_multi_fifo_dma) { + sai->audio_config[tx].words_per_fifo = min(slots, channels); + if (tx) { + sai->audio_config[tx].n_fifos_dst = pins; + sai->audio_config[tx].stride_fifos_dst = dl_cfg[dl_cfg_idx].next_off[tx]; + } else { + sai->audio_config[tx].n_fifos_src = pins; + sai->audio_config[tx].stride_fifos_src = dl_cfg[dl_cfg_idx].next_off[tx]; + } + dma_params->maxburst = sai->audio_config[tx].words_per_fifo * pins; + dma_params->peripheral_config = &sai->audio_config[tx]; + dma_params->peripheral_size = sizeof(sai->audio_config[tx]); + + watermark = tx ? (sai->soc_data->fifo_depth - dma_params->maxburst) : + (dma_params->maxburst - 1); + regmap_update_bits(sai->regmap, FSL_SAI_xCR1(tx, ofs), + FSL_SAI_CR1_RFW_MASK(sai->soc_data->fifo_depth), + watermark); + } + /* Find a proper tcre setting */ for (i = 0; i < sai->soc_data->pins; i++) { trce_mask = (1 << (i + 1)) - 1; @@ -1265,6 +1294,7 @@ static int fsl_sai_probe(struct platform_device *pdev) char tmp[8]; int irq, ret, i; int index; + u32 dmas[4]; sai = devm_kzalloc(dev, sizeof(*sai), GFP_KERNEL); if (!sai) @@ -1321,6 +1351,11 @@ static int fsl_sai_probe(struct platform_device *pdev) fsl_asoc_get_pll_clocks(&pdev->dev, &sai->pll8k_clk, &sai->pll11k_clk); + /* Use Multi FIFO mode depending on the support from SDMA script */ + ret = of_property_read_u32_array(np, "dmas", dmas, 4); + if (!sai->soc_data->use_edma && !ret && dmas[2] == IMX_DMATYPE_MULTI_SAI) + sai->is_multi_fifo_dma = true; + /* read dataline mask for rx and tx*/ ret = fsl_sai_read_dlcfg(sai); if (ret < 0) { diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 17956b5731dc..697f6690068c 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -6,6 +6,7 @@ #ifndef __FSL_SAI_H #define __FSL_SAI_H +#include <linux/dma/imx-dma.h> #include <sound/dmaengine_pcm.h> #define FSL_SAI_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ @@ -281,6 +282,7 @@ struct fsl_sai { bool is_lsb_first; bool is_dsp_mode; bool is_pdm_mode; + bool is_multi_fifo_dma; bool synchronous[2]; struct fsl_sai_dl_cfg *dl_cfg; unsigned int dl_cfg_cnt; @@ -300,6 +302,7 @@ struct fsl_sai { struct pm_qos_request pm_qos_req; struct pinctrl *pinctrl; struct pinctrl_state *pins_state; + struct sdma_peripheral_config audio_config[2]; }; #define TX 1 -- 2.17.1

2 2

[PATCH] ASoC: mchp-spdiftx: Fix clang -Wbitfield-constant-conversion
by Nathan Chancellor 23 Aug '22

23 Aug '22

A recent change in clang strengthened its -Wbitfield-constant-conversion to warn when 1 is assigned to a 1-bit signed integer bitfield, as it can only be 0 or -1, not 1: sound/soc/atmel/mchp-spdiftx.c:505:20: error: implicit truncation from 'int' to bit-field changes value from 1 to -1 [-Werror,-Wbitfield-constant-conversion] dev->gclk_enabled = 1; ^ ~ 1 error generated. The actual value of the field is never checked, just that it is not zero, so there is not a real bug here. However, it is simple enough to silence the warning by making the bitfield unsigned, which matches the mchp-spdifrx driver. Fixes: 06ca24e98e6b ("ASoC: mchp-spdiftx: add driver for S/PDIF TX Controller") Link: https://github.com/ClangBuiltLinux/linux/issues/1686 Link: https://github.com/llvm/llvm-project/commit/82afc9b169a67e8b8a1862fb9c41a2c… Signed-off-by: Nathan Chancellor <nathan(a)kernel.org> --- sound/soc/atmel/mchp-spdiftx.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/atmel/mchp-spdiftx.c b/sound/soc/atmel/mchp-spdiftx.c index 4850a177803d..ab2d7a791f39 100644 --- a/sound/soc/atmel/mchp-spdiftx.c +++ b/sound/soc/atmel/mchp-spdiftx.c @@ -196,7 +196,7 @@ struct mchp_spdiftx_dev { struct clk *pclk; struct clk *gclk; unsigned int fmt; - int gclk_enabled:1; + unsigned int gclk_enabled:1; }; static inline int mchp_spdiftx_is_running(struct mchp_spdiftx_dev *dev) base-commit: 15205c2829ca2cbb5ece5ceaafe1171a8470e62b -- 2.37.1

4 7

[PATCH 0/2] ASoC: SOF: Kconfig: Fix the dependency for client modules
by Peter Ujfalusi 23 Aug '22

23 Aug '22

Hi, There is still a chance to end up with a client driver selected as built in while the core SOF is as module. Fix this by making the client drivers depend on SND_SOC_SOF. Regards, Peter --- Peter Ujfalusi (2): ASoC: SOF: Kconfig: Make IPC_FLOOD_TEST depend on SND_SOC_SOF ASoC: SOF: Kconfig: Make IPC_MESSAGE_INJECTOR depend on SND_SOC_SOF sound/soc/sof/Kconfig | 2 ++ 1 file changed, 2 insertions(+) -- 2.37.2

2 3

[PATCH] ASoC: SOF: ipc4-topology: fix alh_group_ida max value
by Pierre-Louis Bossart 23 Aug '22

23 Aug '22

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> group_id is from 0 ~ ALH_MULTI_GTW_COUNT - 1, not 0 ~ ALH_MULTI_GTW_COUNT. Fixes: 0e357b5290532 ("ASoC: SOF: ipc4-topology: add SoundWire/ALH aggregation support" Reported-by: kernel test robot <lkp(a)intel.com> Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/ipc4-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index c6abfaf5d5326..64929dc9af397 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -1251,7 +1251,7 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, if (blob->alh_cfg.count > 1) { int group_id; - group_id = ida_alloc_max(&alh_group_ida, ALH_MULTI_GTW_COUNT, + group_id = ida_alloc_max(&alh_group_ida, ALH_MULTI_GTW_COUNT - 1, GFP_KERNEL); if (group_id < 0) -- 2.34.1

2 2

[PATCH 0/5] ASoC: nau8xxx: Implement hw constraint for rates
by Takashi Iwai 23 Aug '22

23 Aug '22

Hi, this is a series of patches to address the issues on nau8xxx codecs I've stumbled upon while dealing with a bug report for Steam Deck. Most of them are to implement the missing hw constraint for rate restrictions while one patch is to fix the semaphore unbalance in nau8824 driver. Takashi === Takashi Iwai (5): ASoC: nau8821: Implement hw constraint for rates ASoC: nau8824: Fix semaphore unbalance at error paths ASoC: nau8824: Implement hw constraint for rates ASoC: nau8825: Implement hw constraint for rates ASoC: nau8540: Implement hw constraint for rates sound/soc/codecs/nau8540.c | 40 +++++++++++++----- sound/soc/codecs/nau8821.c | 66 ++++++++++++++++-------------- sound/soc/codecs/nau8824.c | 80 ++++++++++++++++++++---------------- sound/soc/codecs/nau8825.c | 83 +++++++++++++++++++++----------------- 4 files changed, 156 insertions(+), 113 deletions(-) -- 2.35.3

2 6

[PATCH 0/2] soundwire: remove use of __func__ in dev_dbg
by Bard Liao 23 Aug '22

23 Aug '22

The module and function information can be added with 'modprobe foo dyndbg=+pmf' Pierre-Louis Bossart (2): soundwire: bus: remove use of __func__ in dev_dbg soundwire: intel: remove use of __func__ in dev_dbg drivers/soundwire/bus.c | 16 ++++++++-------- drivers/soundwire/intel.c | 14 +++++++------- 2 files changed, 15 insertions(+), 15 deletions(-) -- 2.25.1

2 3

[PATCH] soundwire: dmi-quirks: add remapping for HP Omen 16-k0005TX
by Bard Liao 23 Aug '22

23 Aug '22

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> The DSDT for this device has a number of problems: a) it lists rt711 on link0 and link1, but link1 is disabled b) the rt711 entry on link0 uses the wrong v2 instead of v3 (SDCA) c) the rt1316 amplifier on link3 is not listed. Add a remapping table to work-around these BIOS shenanigans. BugLink: https://github.com/thesofproject/sof/issues/5955 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/dmi-quirks.c | 27 +++++++++++++++++++++++++++ 1 file changed, 27 insertions(+) diff --git a/drivers/soundwire/dmi-quirks.c b/drivers/soundwire/dmi-quirks.c index 747983743a14..f81cdd83ec26 100644 --- a/drivers/soundwire/dmi-quirks.c +++ b/drivers/soundwire/dmi-quirks.c @@ -55,7 +55,26 @@ static const struct adr_remap dell_sku_0A3E[] = { {} }; +/* + * The HP Omen 16-k0005TX does not expose the correct version of RT711 on link0 + * and does not expose a RT1316 on link3 + */ +static const struct adr_remap hp_omen_16[] = { + /* rt711-sdca on link0 */ + { + 0x000020025d071100ull, + 0x000030025d071101ull + }, + /* rt1316-sdca on link3 */ + { + 0x000120025d071100ull, + 0x000330025d131601ull + }, + {} +}; + static const struct dmi_system_id adr_remap_quirk_table[] = { + /* TGL devices */ { .matches = { DMI_MATCH(DMI_SYS_VENDOR, "HP"), @@ -78,6 +97,14 @@ static const struct dmi_system_id adr_remap_quirk_table[] = { }, .driver_data = (void *)dell_sku_0A3E, }, + /* ADL devices */ + { + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "HP"), + DMI_MATCH(DMI_PRODUCT_NAME, "OMEN by HP Gaming Laptop 16-k0xxx"), + }, + .driver_data = (void *)hp_omen_16, + }, {} }; -- 2.25.1

2 1

[PATCH v3 linux-next] soundwire: intel: Remove unnecessary TODO
by Khalid Masum 23 Aug '22

23 Aug '22

The capabilities enabled for multi-link are required as part of the programming sequences, even when a stream uses a single link we still use the syncArm/syncGo sequences. Therefore the TODO is no longer necessary. Suggested-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Khalid Masum <khalid.masum.92(a)gmail.com> --- Previous discussions: - v2 Link: https://lore.kernel.org/lkml/20220816175954.5259-1-khalid.masum.92@gmail.co… - v1 Link: https://lore.kernel.org/lkml/20220814080416.7531-1-khalid.masum.92@gmail.co… Changes since v2: - Update commit title. Changes since v1: - Remove the todo, leave rest of the code as is. - Update commit message. drivers/soundwire/bus.c | 1 - 1 file changed, 1 deletion(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index 8d4000664fa3..18158825a6d8 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -75,7 +75,6 @@ int sdw_bus_master_add(struct sdw_bus *bus, struct device *parent, /* * Initialize multi_link flag - * TODO: populate this flag by reading property from FW node */ bus->multi_link = false; if (bus->ops->read_prop) { -- 2.37.1

2 1

[PATCH v2] soundwire: qcom: Update error prints to debug prints
by Srinivasa Rao Mandadapu 23 Aug '22

23 Aug '22

Update error prints to debug prints to avoid redundant logging in kernel boot time, as these prints are informative prints in irq handler. Signed-off-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> Reviewed-by: Andrew Halaney <ahalaney(a)redhat.com> --- Changes since v1: -- Remove redundant __func__ argument. -- Replace dev_dbg with dev_dbg_ratelimited. drivers/soundwire/qcom.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 9df970e..976ae75 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -573,11 +573,10 @@ static irqreturn_t qcom_swrm_irq_handler(int irq, void *dev_id) break; case SWRM_INTERRUPT_STATUS_NEW_SLAVE_ATTACHED: case SWRM_INTERRUPT_STATUS_CHANGE_ENUM_SLAVE_STATUS: - dev_err_ratelimited(swrm->dev, "%s: SWR new slave attached\n", - __func__); + dev_dbg_ratelimited(swrm->dev, "SWR new slave attached\n"); swrm->reg_read(swrm, SWRM_MCP_SLV_STATUS, &slave_status); if (swrm->slave_status == slave_status) { - dev_err(swrm->dev, "Slave status not changed %x\n", + dev_dbg(swrm->dev, "Slave status not changed %x\n", slave_status); } else { qcom_swrm_get_device_status(swrm); -- 2.7.4

3 2

[PATCH] ASoC: fsl_mqs: Fix supported clock DAI format
by Shengjiu Wang 23 Aug '22

23 Aug '22

The MQS works as codec DAI, not cpu DAI. It is clock consumer, not clock privider. Fixes: 3b14c15a333b ("ASoC: fsl: Update to use set_fmt_new callback") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_mqs.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_mqs.c b/sound/soc/fsl/fsl_mqs.c index c1e2f671191b..4922e6795b73 100644 --- a/sound/soc/fsl/fsl_mqs.c +++ b/sound/soc/fsl/fsl_mqs.c @@ -122,7 +122,7 @@ static int fsl_mqs_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) } switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { - case SND_SOC_DAIFMT_BP_FP: + case SND_SOC_DAIFMT_CBC_CFC: break; default: return -EINVAL; -- 2.34.1

2 1

Re: [PATCH] drm/bridge: dw-hdmi-i2s: set insert_pcuv bit if hardware supports it
by Geraldo Nascimento 23 Aug '22

23 Aug '22

On Mon, Aug 22, 2022 at 02:27:26PM +0100, Mark Brown wrote: > On Sat, Aug 20, 2022 at 09:36:53PM -0300, Geraldo Nascimento wrote: > > > + /* > > + * dw-hdmi introduced insert_pcuv bit in version 2.10a. > > + * When set (1'b1), this bit enables the insertion of the PCUV > > + * (Parity, Channel Status, User bit and Validity) bits on the > > + * incoming audio stream (support limited to Linear PCM audio) > > + */ > > + > > + if (hdmi_read(audio, HDMI_DESIGN_ID) >= 0x21) > > + conf2 |= HDMI_AUD_CONF2_INSERT_PCUV; > > So what if we're not handlign linear PCM? Hi Mark, I think we may be able to restrict the fix for L-PCM only by checking byte 0 of iec.status, specifically bit 1. Our define is called IEC958_AES0_NONAUDIO but https://tech.ebu.ch/docs/tech/tech3250.pdf calls it the "Linear PCM identification" bit. There's also a supplement to AES/EBU 3250 in https://tech.ebu.ch/docs/n/n009_1.pdf Let me know if the following is OK. I'll be happy to submit V2 if it is. Thanks, Geraldo Nascimento --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c @@ -42,6 +42,7 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, struct dw_hdmi *hdmi = audio->hdmi; u8 conf0 = 0; u8 conf1 = 0; + u8 conf2 = 0; u8 inputclkfs = 0; /* it cares I2S only */ @@ -101,6 +102,18 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, return -EINVAL; } + /* + * dw-hdmi introduced insert_pcuv bit in version 2.10a. + * When set (1'b1), this bit enables the insertion of the PCUV + * (Parity, Channel Status, User bit and Validity) bits on the + * incoming audio stream (support limited to Linear PCM audio) + */ + + if (hdmi_read(audio, HDMI_DESIGN_ID) >= 0x21 && + !(hparms->iec.status[0] & IEC958_AES0_NONAUDIO)) + conf2 = HDMI_AUD_CONF2_INSERT_PCUV; + dw_hdmi_set_sample_rate(hdmi, hparms->sample_rate); dw_hdmi_set_channel_status(hdmi, hparms->iec.status); dw_hdmi_set_channel_count(hdmi, hparms->channels); @@ -109,6 +122,7 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, hdmi_write(audio, inputclkfs, HDMI_AUD_INPUTCLKFS); hdmi_write(audio, conf0, HDMI_AUD_CONF0); hdmi_write(audio, conf1, HDMI_AUD_CONF1); + hdmi_write(audio, conf2, HDMI_AUD_CONF2); return 0; }

2 1

[PATCH] soundwire: qcom: remove duplicate reset control get
by Srinivas Kandagatla 23 Aug '22

23 Aug '22

Looks like adding clock gate flag patch forgot to remove the old code that gets reset control. This causes below crash on platforms that do not need reset. [ 15.653501] reset_control_reset+0x124/0x170 [ 15.653508] qcom_swrm_init+0x50/0x1a0 [ 15.653514] qcom_swrm_probe+0x320/0x668 [ 15.653519] platform_probe+0x68/0xe0 [ 15.653529] really_probe+0xbc/0x2a8 [ 15.653535] __driver_probe_device+0x7c/0xe8 [ 15.653541] driver_probe_device+0x40/0x110 [ 15.653547] __device_attach_driver+0x98/0xd0 [ 15.653553] bus_for_each_drv+0x68/0xd0 [ 15.653559] __device_attach+0xf4/0x188 [ 15.653565] device_initial_probe+0x14/0x20 Fix this by removing old code. Reported-by: Amit Pundir <amit.pundir(a)linaro.org> Fixes: 1fd0d85affe4 ("soundwire: qcom: Add flag for software clock gating check") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- drivers/soundwire/qcom.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 9df970eeca45..a43961ad4614 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -1356,10 +1356,6 @@ static int qcom_swrm_probe(struct platform_device *pdev) ctrl->bus.compute_params = &qcom_swrm_compute_params; ctrl->bus.clk_stop_timeout = 300; - ctrl->audio_cgcr = devm_reset_control_get_exclusive(dev, "swr_audio_cgcr"); - if (IS_ERR(ctrl->audio_cgcr)) - dev_err(dev, "Failed to get audio_cgcr reset required for soundwire-v1.6.0\n"); - ret = qcom_swrm_get_port_config(ctrl); if (ret) goto err_clk; -- 2.21.0

3 2

Re: [PATCH v8 1/2] ASoC: sunxi: Add Allwinner H6 Digital MIC driver
by 班涛 23 Aug '22

23 Aug '22

Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> 于2022年8月12日周五 23:20写道： > On 8/11/2022 4:49 PM, Ban Tao wrote: > > The Allwinner H6 and later SoCs have an DMIC block > > which is capable of capture. > > > > Signed-off-by: Ban Tao <fengzheng923(a)gmail.com> > > --- > > ... > > > + > > +#define SUN50I_DMIC_RATES (SNDRV_PCM_RATE_8000_48000) > > Seeing supported rates in hw_params callback, you probably want to take > a look at SNDRV_PCM_RATE_KNOT? > No, I don't need SNDRV_PCM_RATE_KNOT. Dmic does not support unknown rates. > > > +#define SUN50I_DMIC_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | > SNDRV_PCM_FMTBIT_S24_LE) > > + > > +static struct snd_soc_dai_driver sun50i_dmic_dai = { > > + .capture = { > > + .channels_min = 1, > > + .channels_max = 8, > > + .rates = SUN50I_DMIC_RATES, > > + .formats = SUN50I_DMIC_FORMATS, > > + .sig_bits = 21, > > + }, > > + .probe = sun50i_dmic_soc_dai_probe, > > + .ops = &sun50i_dmic_dai_ops, > > + .name = "dmic", > > +}; > > + > > ... > >

1 0

[PATCH] ALSA: hda: intel-nhlt: Correct the handling of fmt_config flexible array
by Peter Ujfalusi 23 Aug '22

23 Aug '22

The struct nhlt_format's fmt_config is a flexible array, it must not be used as normal array. When moving to the next nhlt_fmt_cfg we need to take into account the data behind the ->config.caps (indicated by ->config.size). Fixes: a864e8f159b13 ("ALSA: hda: intel-nhlt: verify config type") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Jaska Uimonen <jaska.uimonen(a)linux.intel.com> --- sound/hda/intel-nhlt.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c index 9db5ccd9aa2d..13bb0ccfb36c 100644 --- a/sound/hda/intel-nhlt.c +++ b/sound/hda/intel-nhlt.c @@ -55,16 +55,22 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt) /* find max number of channels based on format_configuration */ if (fmt_configs->fmt_count) { + struct nhlt_fmt_cfg *fmt_cfg = fmt_configs->fmt_config; + dev_dbg(dev, "found %d format definitions\n", fmt_configs->fmt_count); for (i = 0; i < fmt_configs->fmt_count; i++) { struct wav_fmt_ext *fmt_ext; - fmt_ext = &fmt_configs->fmt_config[i].fmt_ext; + fmt_ext = &fmt_cfg->fmt_ext; if (fmt_ext->fmt.channels > max_ch) max_ch = fmt_ext->fmt.channels; + + /* Move to the next nhlt_fmt_cfg */ + fmt_cfg = (struct nhlt_fmt_cfg *)(fmt_cfg->config.caps + + fmt_cfg->config.size); } dev_dbg(dev, "max channels found %d\n", max_ch); } else { -- 2.37.2

2 1

data-race in snd_seq_oss_midi_check_exit_port / snd_seq_oss_midi_setup
by Abhishek Shah 23 Aug '22

23 Aug '22

Hi all, We found a race involving the *max_midi_devs* variable. We see an interleaving where the following check here <https://elixir.bootlin.com/linux/v5.18-rc5/source/sound/core/seq/oss/seq_os…> passes before the *snd_seq_oss_midi_check_exit_port*() finishes, but this check should not pass if *max_midi_devs* will become zero, but we are not sure of its implications in terms of security impact. Please let us know what you think. Thanks! *-------------------Report---------------------* *write* to 0xffffffff88382f80 of 4 bytes by task 6541 on cpu 0: snd_seq_oss_midi_check_exit_port+0x1a6/0x270 sound/core/seq/oss/seq_oss_midi.c:237 receive_announce+0x193/0x1b0 sound/core/seq/oss/seq_oss_init.c:143 snd_seq_deliver_single_event+0x30d/0x4e0 sound/core/seq/seq_clientmgr.c:640 deliver_to_subscribers sound/core/seq/seq_clientmgr.c:695 [inline] snd_seq_deliver_event+0x38c/0x490 sound/core/seq/seq_clientmgr.c:830 snd_seq_kernel_client_dispatch+0x189/0x1a0 sound/core/seq/seq_clientmgr.c:2339 snd_seq_system_broadcast+0x98/0xd0 sound/core/seq/seq_system.c:86 snd_seq_ioctl_delete_port+0x9a/0xc0 sound/core/seq/seq_clientmgr.c:1356 snd_seq_ioctl+0x198/0x2d0 sound/core/seq/seq_clientmgr.c:2173 vfs_ioctl fs/ioctl.c:51 [inline] __do_sys_ioctl fs/ioctl.c:870 [inline] __se_sys_ioctl+0xe1/0x150 fs/ioctl.c:856 __x64_sys_ioctl+0x43/0x50 fs/ioctl.c:856 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x3d/0x90 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x44/0xae *read* to 0xffffffff88382f80 of 4 bytes by task 6542 on cpu 1: snd_seq_oss_midi_setup+0x1b/0x40 sound/core/seq/oss/seq_oss_midi.c:273 snd_seq_oss_open+0x364/0x900 sound/core/seq/oss/seq_oss_init.c:198 odev_open+0x55/0x70 sound/core/seq/oss/seq_oss.c:128 soundcore_open+0x315/0x3a0 sound/sound_core.c:593 chrdev_open+0x373/0x3f0 fs/char_dev.c:414 do_dentry_open+0x543/0x8f0 fs/open.c:824 vfs_open+0x47/0x50 fs/open.c:958 do_open fs/namei.c:3476 [inline] path_openat+0x1906/0x1dc0 fs/namei.c:3609 do_filp_open+0xef/0x200 fs/namei.c:3636 do_sys_openat2+0xa5/0x2a0 fs/open.c:1213 do_sys_open fs/open.c:1229 [inline] __do_sys_openat fs/open.c:1245 [inline] __se_sys_openat fs/open.c:1240 [inline] __x64_sys_openat+0xf0/0x120 fs/open.c:1240 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x3d/0x90 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x44/0xae Reported by Kernel Concurrency Sanitizer on: CPU: 1 PID: 6542 Comm: syz-executor2-n Not tainted 5.18.0-rc5+ #107 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS 1.15.0-1 04/01/2014 *Reproducing Inputs* Input CPU 0: r0 = openat$sndseq(0xffffffffffffff9c, &(0x7f0000000040)='/dev/snd/seq\x00', 0x0) ioctl$SNDRV_SEQ_IOCTL_CREATE_PORT(r0, 0xc0a85320, &(0x7f0000000240)={{0x80}, 'port1\x00', 0x10}) ioctl$SNDRV_SEQ_IOCTL_SET_CLIENT_POOL(r0, 0x40a85321, &(0x7f0000000100)={0x80}) Input CPU 1: r0 = openat$sequencer2(0xffffff9c, &(0x7f0000000000)='/dev/sequencer2\x00', 0x0, 0x0) ioctl$SNDCTL_SYNTH_INFO(r0, 0xc08c5102, &(0x7f0000000200)={"02961a3ce6d4828f8b5559726313251b55fa11d8d65406f1f33c9af8e3f8", 0xffffffff})

3 4

[PATCH 0/5] ALSA: Drop hackish GFP giveaway for CONTINUOUS pages
by Takashi Iwai 23 Aug '22

23 Aug '22

Hi, this is a series of cleanup patches for dropping the current hackish way of passing the GFP_* flags for CONTINOUS and VMALLOC memory allocations. There are only three users for this legacy feature, and all of them seem superfluous. And, if any driver requires the memory restriction in future, it can now pass the proper device pointer for specifying the DMA mask. Takashi === Takashi Iwai (5): ALSA: vx: Drop superfluous GFP setup ALSA: pdaudiocf: Drop superfluous GFP setup ASoC: Intel: sst: Switch to standard device pages ALSA: memalloc: Drop special handling of GFP for CONTINUOUS allocation ALSA: doc: Drop snd_dma_continuous_data() usages .../kernel-api/writing-an-alsa-driver.rst | 21 ++-- include/sound/memalloc.h | 3 - sound/core/memalloc.c | 113 ++++++++---------- sound/drivers/vx/vx_pcm.c | 3 +- sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c | 3 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 7 +- 6 files changed, 61 insertions(+), 89 deletions(-) -- 2.35.3

2 6

[PATCH v2 0/4] ASoC platform driver for Apple MCA
by Martin Povišer 23 Aug '22

23 Aug '22

Hi all, this is v2 of the ASoC platform driver for MCA peripheral on Apple M1, to be followed by a machine driver later for sound support on the new Apple devices. This time I attached a DTS patch for t8103 (M1), which I guess should go through Marcan's (in CC) tree. Martin Changes since v1: - reflect the set_fmt provider/consumer change in 6.0-rc1 - fix the probe function requesting optional reset - unroll 'dma-names' in schema - fix binding schema/code discrepancy in register ranges - minor things Changes since 'macaudio RFC v2' [0]: - addition of locking (extra commit) - transition to set_bclk_ratio (instead of getting the bclk ratio from set_sysclk) - using shared reset control and documenting the reset in binding - formatting, comments, and a minor fix to hw driving [0] https://lore.kernel.org/asahi/20220606191910.16580-1-povik+lin@cutebit.org/ Martin Povišer (4): dt-bindings: sound: Add Apple MCA I2S transceiver arm64: dts: apple: t8103: Add MCA and its support ASoC: apple: mca: Start new platform driver ASoC: apple: mca: Add locks on foreign cluster access .../devicetree/bindings/sound/apple,mca.yaml | 131 ++ MAINTAINERS | 8 + arch/arm64/boot/dts/apple/t8103.dtsi | 70 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/apple/Kconfig | 9 + sound/soc/apple/Makefile | 3 + sound/soc/apple/mca.c | 1170 +++++++++++++++++ 8 files changed, 1393 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/apple,mca.yaml create mode 100644 sound/soc/apple/Kconfig create mode 100644 sound/soc/apple/Makefile create mode 100644 sound/soc/apple/mca.c -- 2.33.0

4 19

DSD hardware quirks and usb vendor id hardcoding
by GitHub issues - opened 23 Aug '22

23 Aug '22

alsa-project/alsa-lib issue #266 was opened from soholt: I am using tinyusb for UAC2 audio class and line 492 https://github.com/hathach/tinyusb/blob/master/src/class/audio/audio.h allows: "AUDIO_DATA_FORMAT_TYPE_I_RAW_DATA = 0x80000000" cat /proc/asound/ reports it as "Format: SPECIAL", but I wanted to be "SPECIAL DSD_U32_xx", so for fun I tried: 0x80000001 which reported as = S32_LE SPECIAL, 0x80000002 = U8 SPECIAL, 0x80000003 = U8 S32_LE SPECIAL 0x800000ff = U8 S32_LE FLOAT_LE MU_LAW A_LAW SPECIAL - seem to recursively add more Then I found quirks line 1582 https://github.com/torvalds/linux/blob/master/sound/usb/quirks.c "The UAC standards don't have a designated bit field to denote DSD-capable interfaces, hence all hardware that is known to support this format has to be listed here." So if I change my usb vendor id to XMOS 0x20b1, only then "Format: SPECIAL" becomes "Format: SPECIAL DSD_U32_BE" I have no idea if range 0x80000001++ is used for anything or if it could be used for DSD formats like: 0x80000001 = SPECIAL DSD_U8 0x80000002 = SPECIAL DSD_U16_LE 0x80000003 = SPECIAL DSD_U16_BE 0x80000004 = SPECIAL DSD_U32_LE 0x80000005 = SPECIAL DSD_U32_BE so no future hardcoding is required. Thanks, Gin Issue URL : https://github.com/alsa-project/alsa-lib/issues/266 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ALSA: hda: cleanup definitions for multi-link registers
by Pierre-Louis Bossart 23 Aug '22

23 Aug '22

For some reason two masks are used without the AZX prefix, and the pattern MLCLT should be ML_LCTL for consistency. Pure rename, no functionality change. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- include/sound/hda_register.h | 20 +++++++++++--------- sound/hda/ext/hdac_ext_controller.c | 16 ++++++++-------- sound/pci/hda/hda_intel.c | 14 +++++++------- 3 files changed, 26 insertions(+), 24 deletions(-) diff --git a/include/sound/hda_register.h b/include/sound/hda_register.h index ad8b71b1dbb67..d37cf43546eba 100644 --- a/include/sound/hda_register.h +++ b/include/sound/hda_register.h @@ -260,7 +260,18 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define AZX_REG_ML_LCAP 0x00 #define AZX_REG_ML_LCTL 0x04 + +#define AZX_ML_LCTL_CPA BIT(23) +#define AZX_ML_LCTL_CPA_SHIFT 23 +#define AZX_ML_LCTL_SPA BIT(16) +#define AZX_ML_LCTL_SPA_SHIFT 16 +#define AZX_ML_LCTL_SCF GENMASK(3, 0) + #define AZX_REG_ML_LOSIDV 0x08 + +/* bit0 is reserved, with BIT(1) mapping to stream1 */ +#define AZX_ML_LOSIDV_STREAM_MASK 0xFFFE + #define AZX_REG_ML_LSDIID 0x0C #define AZX_REG_ML_LPSOO 0x10 #define AZX_REG_ML_LPSIO 0x12 @@ -268,15 +279,6 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define AZX_REG_ML_LOUTPAY 0x20 #define AZX_REG_ML_LINPAY 0x30 -/* bit0 is reserved, with BIT(1) mapping to stream1 */ -#define ML_LOSIDV_STREAM_MASK 0xFFFE - -#define ML_LCTL_SCF_MASK 0xF -#define AZX_MLCTL_SPA (0x1 << 16) -#define AZX_MLCTL_CPA (0x1 << 23) -#define AZX_MLCTL_SPA_SHIFT 16 -#define AZX_MLCTL_CPA_SHIFT 23 - /* registers for DMA Resume Capability Structure */ #define AZX_DRSM_CAP_ID 0x5 #define AZX_REG_DRSM_CTL 0x4 diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index a42f66f561f57..80876b9a87f46 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -170,7 +170,7 @@ static int check_hdac_link_power_active(struct hdac_ext_link *link, bool enable) { int timeout; u32 val; - int mask = (1 << AZX_MLCTL_CPA_SHIFT); + int mask = (1 << AZX_ML_LCTL_CPA_SHIFT); udelay(3); timeout = 150; @@ -178,10 +178,10 @@ static int check_hdac_link_power_active(struct hdac_ext_link *link, bool enable) do { val = readl(link->ml_addr + AZX_REG_ML_LCTL); if (enable) { - if (((val & mask) >> AZX_MLCTL_CPA_SHIFT)) + if (((val & mask) >> AZX_ML_LCTL_CPA_SHIFT)) return 0; } else { - if (!((val & mask) >> AZX_MLCTL_CPA_SHIFT)) + if (!((val & mask) >> AZX_ML_LCTL_CPA_SHIFT)) return 0; } udelay(3); @@ -197,7 +197,7 @@ static int check_hdac_link_power_active(struct hdac_ext_link *link, bool enable) int snd_hdac_ext_bus_link_power_up(struct hdac_ext_link *link) { snd_hdac_updatel(link->ml_addr, AZX_REG_ML_LCTL, - AZX_MLCTL_SPA, AZX_MLCTL_SPA); + AZX_ML_LCTL_SPA, AZX_ML_LCTL_SPA); return check_hdac_link_power_active(link, true); } @@ -209,7 +209,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_power_up); */ int snd_hdac_ext_bus_link_power_down(struct hdac_ext_link *link) { - snd_hdac_updatel(link->ml_addr, AZX_REG_ML_LCTL, AZX_MLCTL_SPA, 0); + snd_hdac_updatel(link->ml_addr, AZX_REG_ML_LCTL, AZX_ML_LCTL_SPA, 0); return check_hdac_link_power_active(link, false); } @@ -226,7 +226,7 @@ int snd_hdac_ext_bus_link_power_up_all(struct hdac_bus *bus) list_for_each_entry(hlink, &bus->hlink_list, list) { snd_hdac_updatel(hlink->ml_addr, AZX_REG_ML_LCTL, - AZX_MLCTL_SPA, AZX_MLCTL_SPA); + AZX_ML_LCTL_SPA, AZX_ML_LCTL_SPA); ret = check_hdac_link_power_active(hlink, true); if (ret < 0) return ret; @@ -247,7 +247,7 @@ int snd_hdac_ext_bus_link_power_down_all(struct hdac_bus *bus) list_for_each_entry(hlink, &bus->hlink_list, list) { snd_hdac_updatel(hlink->ml_addr, AZX_REG_ML_LCTL, - AZX_MLCTL_SPA, 0); + AZX_ML_LCTL_SPA, 0); ret = check_hdac_link_power_active(hlink, false); if (ret < 0) return ret; @@ -281,7 +281,7 @@ int snd_hdac_ext_bus_link_get(struct hdac_bus *bus, * clear the register to invalidate all the output streams */ snd_hdac_updatew(link->ml_addr, AZX_REG_ML_LOSIDV, - ML_LOSIDV_STREAM_MASK, 0); + AZX_ML_LOSIDV_STREAM_MASK, 0); /* * wait for 521usec for codec to report status * HDA spec section 4.3 - Codec Discovery diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 7720978dc1326..bf9df9bc8f1bf 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -489,14 +489,14 @@ static int intel_ml_lctl_set_power(struct azx *chip, int state) * If other links are enabled for stream, they need similar fix */ val = readl(bus->mlcap + AZX_ML_BASE + AZX_REG_ML_LCTL); - val &= ~AZX_MLCTL_SPA; - val |= state << AZX_MLCTL_SPA_SHIFT; + val &= ~AZX_ML_LCTL_SPA; + val |= state << AZX_ML_LCTL_SPA_SHIFT; writel(val, bus->mlcap + AZX_ML_BASE + AZX_REG_ML_LCTL); /* wait for CPA */ timeout = 50; while (timeout) { if (((readl(bus->mlcap + AZX_ML_BASE + AZX_REG_ML_LCTL)) & - AZX_MLCTL_CPA) == (state << AZX_MLCTL_CPA_SHIFT)) + AZX_ML_LCTL_CPA) == (state << AZX_ML_LCTL_CPA_SHIFT)) return 0; timeout--; udelay(10); @@ -514,15 +514,15 @@ static void intel_init_lctl(struct azx *chip) /* 0. check lctl register value is correct or not */ val = readl(bus->mlcap + AZX_ML_BASE + AZX_REG_ML_LCTL); /* if SCF is already set, let's use it */ - if ((val & ML_LCTL_SCF_MASK) != 0) + if ((val & AZX_ML_LCTL_SCF) != 0) return; /* * Before operating on SPA, CPA must match SPA. * Any deviation may result in undefined behavior. */ - if (((val & AZX_MLCTL_SPA) >> AZX_MLCTL_SPA_SHIFT) != - ((val & AZX_MLCTL_CPA) >> AZX_MLCTL_CPA_SHIFT)) + if (((val & AZX_ML_LCTL_SPA) >> AZX_ML_LCTL_SPA_SHIFT) != + ((val & AZX_ML_LCTL_CPA) >> AZX_ML_LCTL_CPA_SHIFT)) return; /* 1. turn link down: set SPA to 0 and wait CPA to 0 */ @@ -532,7 +532,7 @@ static void intel_init_lctl(struct azx *chip) goto set_spa; /* 2. update SCF to select a properly audio clock*/ - val &= ~ML_LCTL_SCF_MASK; + val &= ~AZX_ML_LCTL_SCF; val |= intel_get_lctl_scf(chip); writel(val, bus->mlcap + AZX_ML_BASE + AZX_REG_ML_LCTL); -- 2.34.1

2 1

Adding Focusrite Scarlett 2i4 gen2
by GitHub pull_request - opened 23 Aug '22

23 Aug '22

alsa-project/alsa-ucm-conf pull request #203 was opened from PatrickLang: This cleans up the Focusrite Scarlett 2i4 Gen2 device a bit. This device is split into two stereo pairs for output, and has 2 mic/line/instrument inputs. There are no toggles or mixer controls exposed to Alsa. It has hardware switches for: - Headphones - can monitor line outputs 1+2 or 3+4 - Headphone input/playback mix - Monitor out (Line 1+2) - Each input has a level pot, and switches for pad and Z (line/inst) - There is 1 physical +48v switch Please let me know if any more info would be helpful. I have captured some more notes showing before/after state at https://gist.github.com/PatrickLang/79a3090eea6f2d640179457b5d428134#scarle… I'm planning to write a UCM2 profile for the more complicated Focusrite Scarlett 18i8 Gen2 next which has more inputs, but also multiple mix busses controllable through Alsa Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/203 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/203.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: codes: src4xxx: Avoid clang -Wsometimes-uninitialized in src4xxx_hw_params()
by Nathan Chancellor 23 Aug '22

23 Aug '22

Clang warns: sound/soc/codecs/src4xxx.c:280:3: error: variable 'd' is used uninitialized whenever switch default is taken [-Werror,-Wsometimes-uninitialized] default: ^~~~~~~ sound/soc/codecs/src4xxx.c:298:59: note: uninitialized use occurs here ret = regmap_write(src4xxx->regmap, SRC4XXX_RCV_PLL_11, d); ^ sound/soc/codecs/src4xxx.c:223:20: note: initialize the variable 'd' to silence this warning int val, pj, jd, d; ^ = 0 sound/soc/codecs/src4xxx.c:280:3: error: variable 'jd' is used uninitialized whenever switch default is taken [-Werror,-Wsometimes-uninitialized] default: ^~~~~~~ sound/soc/codecs/src4xxx.c:293:59: note: uninitialized use occurs here ret = regmap_write(src4xxx->regmap, SRC4XXX_RCV_PLL_10, jd); ^~ sound/soc/codecs/src4xxx.c:223:17: note: initialize the variable 'jd' to silence this warning int val, pj, jd, d; ^ = 0 sound/soc/codecs/src4xxx.c:280:3: error: variable 'pj' is used uninitialized whenever switch default is taken [-Werror,-Wsometimes-uninitialized] default: ^~~~~~~ sound/soc/codecs/src4xxx.c:288:59: note: uninitialized use occurs here ret = regmap_write(src4xxx->regmap, SRC4XXX_RCV_PLL_0F, pj); ^~ sound/soc/codecs/src4xxx.c:223:13: note: initialize the variable 'pj' to silence this warning int val, pj, jd, d; ^ = 0 3 errors generated. According to the comment in the default case, other parts of the chip are still functional without these values so just return 0 in the default case to avoid using these variables uninitialized. Link: https://github.com/ClangBuiltLinux/linux/issues/1691 Reported-by: kernel test robot <lkp(a)intel.com> Reported-by: "Sudip Mukherjee (Codethink)" <sudipm.mukherjee(a)gmail.com> Signed-off-by: Nathan Chancellor <nathan(a)kernel.org> --- sound/soc/codecs/src4xxx.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/src4xxx.c b/sound/soc/codecs/src4xxx.c index a8f143057b41..cf45caa4bf7f 100644 --- a/sound/soc/codecs/src4xxx.c +++ b/sound/soc/codecs/src4xxx.c @@ -283,7 +283,7 @@ static int src4xxx_hw_params(struct snd_pcm_substream *substream, */ dev_info(component->dev, "Couldn't set the RCV PLL as this master clock rate is unknown\n"); - break; + return 0; } ret = regmap_write(src4xxx->regmap, SRC4XXX_RCV_PLL_0F, pj); if (ret < 0) base-commit: 94f072748337424c9cf92cd018532a34db3a5516 -- 2.37.2

2 3

Re: [PATCH/RESEND 2 v1] ASoC: trival spelling fix.
by Maurycy Z 22 Aug '22

22 Aug '22

"system" is mis-spelled as "ststem" in the help for SND_AUDIO_GRAPH_CARD. Acked-by: Randy Dunlap <rdunlap(a)infradead.org> Signed-off-by: Maurycy Zalewski <10maurycy10(a)gmail.com> --- diff --git a/sound/soc/generic/Kconfig b/sound/soc/generic/Kconfig index b6df4e26b..72f3446e1 100644 --- a/sound/soc/generic/Kconfig +++ b/sound/soc/generic/Kconfig @@ -16,7 +16,7 @@ config SND_AUDIO_GRAPH_CARD help This option enables generic simple sound card support with OF-graph DT bindings. - It also support DPCM of multi CPU single Codec ststem. + It also support DPCM of multi CPU single Codec system. config SND_AUDIO_GRAPH_CARD2 tristate "ASoC Audio Graph sound card2 support"

1 0

data-race in snd_seq_client_use_ptr / snd_seq_client_use_ptr
by Abhishek Shah 22 Aug '22

22 Aug '22

Hi all, We found a data race involving the *card_requested[card]* variable here <https://elixir.bootlin.com/linux/v5.18-rc5/source/sound/core/seq/seq_client…>, which can cause *snd_request_card* to be called multiple times. We are not sure if this has security implications however, but we would still like to report it. Please let us know what you think. Thanks! *----------Report-----------* *write* to 0xffffffff88382191 of 1 bytes by task 6541 on cpu 0: snd_seq_client_use_ptr+0x254/0x610 sound/core/seq/seq_clientmgr.c:146 snd_seq_info_clients_read+0xd8/0x480 sound/core/seq/seq_clientmgr.c:2475 snd_info_seq_show+0x81/0xa0 sound/core/info.c:361 seq_read_iter+0x2d2/0x8e0 fs/seq_file.c:230 seq_read+0x1c9/0x210 fs/seq_file.c:162 pde_read fs/proc/inode.c:311 [inline] proc_reg_read+0x123/0x1b0 fs/proc/inode.c:323 vfs_read+0x1b5/0x6e0 fs/read_write.c:480 ksys_read+0xde/0x190 fs/read_write.c:620 __do_sys_read fs/read_write.c:630 [inline] __se_sys_read fs/read_write.c:628 [inline] __x64_sys_read+0x43/0x50 fs/read_write.c:628 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x3d/0x90 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x44/0xae *read* to 0xffffffff88382191 of 1 bytes by task 6542 on cpu 1: snd_seq_client_use_ptr+0x189/0x610 sound/core/seq/seq_clientmgr.c:145 snd_seq_info_clients_read+0xd8/0x480 sound/core/seq/seq_clientmgr.c:2475 snd_info_seq_show+0x81/0xa0 sound/core/info.c:361 seq_read_iter+0x2d2/0x8e0 fs/seq_file.c:230 seq_read+0x1c9/0x210 fs/seq_file.c:162 pde_read fs/proc/inode.c:311 [inline] proc_reg_read+0x123/0x1b0 fs/proc/inode.c:323 vfs_read+0x1b5/0x6e0 fs/read_write.c:480 ksys_read+0xde/0x190 fs/read_write.c:620 __do_sys_read fs/read_write.c:630 [inline] __se_sys_read fs/read_write.c:628 [inline] __x64_sys_read+0x43/0x50 fs/read_write.c:628 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x3d/0x90 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x44/0xae Reported by Kernel Concurrency Sanitizer on: CPU: 1 PID: 6542 Comm: syz-executor2-n Not tainted 5.18.0-rc5+ #107 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS 1.15.0-1 04/01/2014 *Reproducing Inputs* Input CPU 0: r0 = openat$procfs(0xffffffffffffff9c, &(0x7f00000004c0)='/proc/asound/seq/clients\x00', 0x0, 0x0) read$ptp(r0, &(0x7f0000001280)=""/4096, 0x1000) Input CPU 1: r0 = openat$procfs(0xffffff9c, &(0x7f0000000000)='/proc/asound/seq/clients\x00', 0x0, 0x0) syz_fuse_handle_req(r0, &(0x7f0000000040)="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", 0x2000, &(0x7f00000048c0)={0x0, &(0x7f00000020c0)={0x18}, 0x0, 0x0, 0x0, 0x0, 0x0, &(0x7f0000002280)={0x18, 0x0, 0x0, {0x9}}, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0})

3 2

[PATCH v2] ASoC: codecs: add uspport for the TI SRC4392 codec
by Matt Flax 22 Aug '22

22 Aug '22

The src4xxx keyword is used for future capability to integrate other codecs similar to the src4392 to the same code base. Signed-off-by: Matt Flax <flatmax(a)flatmax.com> --- Notes: V2 changes : * Fixed indents in Kconfig * Comment header of source code is now one block * Documented the devicetree binding * sample rate converter MCLK src is non-user settable * hushed spurious dev_info output * src4xxx_remove was empty is removed. * Using switch in hw_params to select port reg. for clarity * Removed interrupt reg. settings - will re-intro. properly later * probe function is now EXPORT_SYMBOL_GPL .../devicetree/bindings/sound/src4xxx.txt | 20 + sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/src4xxx-i2c.c | 55 ++ sound/soc/codecs/src4xxx.c | 508 ++++++++++++++++++ sound/soc/codecs/src4xxx.h | 115 ++++ 6 files changed, 715 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/src4xxx.txt create mode 100644 sound/soc/codecs/src4xxx-i2c.c create mode 100644 sound/soc/codecs/src4xxx.c create mode 100644 sound/soc/codecs/src4xxx.h diff --git a/Documentation/devicetree/bindings/sound/src4xxx.txt b/Documentation/devicetree/bindings/sound/src4xxx.txt new file mode 100644 index 000000000000..f8d845d42f42 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/src4xxx.txt @@ -0,0 +1,20 @@ +Texas Instruments src4392 DT bindings + +SRC4392 is a digital audio codec that can be connected via +I2C or SPI. Currently, only I2C bus is supported. + +Required properties: + + - compatible: "ti,src4392" + +Required properties on I2C: + + - reg: the I2C address + +Examples: + + src4392@70 { + compatible = "ti,src4392"; + reg = <0x70>; + #sound-dai-cells = <0>; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d16b4efb88a7..9c9d74074888 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -205,6 +205,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_SIMPLE_AMPLIFIER imply SND_SOC_SIMPLE_MUX imply SND_SOC_SPDIF + imply SND_SOC_SRC4XXX_I2C imply SND_SOC_SSM2305 imply SND_SOC_SSM2518 imply SND_SOC_SSM2602_SPI @@ -1471,6 +1472,18 @@ config SND_SOC_SIMPLE_MUX config SND_SOC_SPDIF tristate "S/PDIF CODEC" +config SND_SOC_SRC4XXX_I2C + tristate + depends on I2C + select SND_SOC_SRC4XXX + +config SND_SOC_SRC4XXX + tristate "Texas Instruments SRC4XXX DIR/DIT and SRC codecs" + help + Enable support for the TI SRC4XXX family of codecs. These include the + scr4392 which has digital receivers, transmitters, and + a sample rate converter, including numerous ports. + config SND_SOC_SSM2305 tristate "Analog Devices SSM2305 Class-D Amplifier" help diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 92fd441d426a..a599b727c65b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -231,6 +231,8 @@ snd-soc-sigmadsp-regmap-objs := sigmadsp-regmap.o snd-soc-si476x-objs := si476x.o snd-soc-spdif-tx-objs := spdif_transmitter.o snd-soc-spdif-rx-objs := spdif_receiver.o +snd-soc-src4xxx-objs := src4xxx.o +snd-soc-src4xxx-i2c-objs := src4xxx-i2c.o snd-soc-ssm2305-objs := ssm2305.o snd-soc-ssm2518-objs := ssm2518.o snd-soc-ssm2602-objs := ssm2602.o @@ -579,6 +581,8 @@ obj-$(CONFIG_SND_SOC_SIGMADSP_I2C) += snd-soc-sigmadsp-i2c.o obj-$(CONFIG_SND_SOC_SIGMADSP_REGMAP) += snd-soc-sigmadsp-regmap.o obj-$(CONFIG_SND_SOC_SI476X) += snd-soc-si476x.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif-rx.o snd-soc-spdif-tx.o +obj-$(CONFIG_SND_SOC_SRC4XXX) += snd-soc-src4xxx.o +obj-$(CONFIG_SND_SOC_SRC4XXX_I2C) += snd-soc-src4xxx-i2c.o obj-$(CONFIG_SND_SOC_SSM2305) += snd-soc-ssm2305.o obj-$(CONFIG_SND_SOC_SSM2518) += snd-soc-ssm2518.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o diff --git a/sound/soc/codecs/src4xxx-i2c.c b/sound/soc/codecs/src4xxx-i2c.c new file mode 100644 index 000000000000..a42d1b463dbf --- /dev/null +++ b/sound/soc/codecs/src4xxx-i2c.c @@ -0,0 +1,55 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * Driver for SRC4XXX codecs + * + * Copyright 2021-2022 Deqx Pty Ltd + * Author: Matt Flax <flatmax(a)flatmax.com> + */ + +#include <linux/i2c.h> +#include <linux/mod_devicetable.h> +#include <linux/module.h> +#include <linux/regmap.h> + +#include "src4xxx.h" + +static int src4xxx_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + return src4xxx_probe(&i2c->dev, + devm_regmap_init_i2c(i2c, &src4xxx_regmap_config), NULL); +} + +static int src4xxx_i2c_remove(struct i2c_client *i2c) +{ + src4xxx_remove(&i2c->dev); + return 0; +} + +static const struct i2c_device_id src4xxx_i2c_ids[] = { + { "src4392", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, src4xxx_i2c_ids); + +static const struct of_device_id src4xxx_of_match[] = { + { .compatible = "ti,src4392", }, + { } +}; +MODULE_DEVICE_TABLE(of, src4xxx_of_match); + + +static struct i2c_driver src4xxx_i2c_driver = { + .driver = { + .name = "src4xxx", + .of_match_table = of_match_ptr(src4xxx_of_match), + }, + .probe = src4xxx_i2c_probe, + .remove = src4xxx_i2c_remove, + .id_table = src4xxx_i2c_ids, +}; +module_i2c_driver(src4xxx_i2c_driver); + +MODULE_DESCRIPTION("ASoC SRC4392 CODEC I2C driver"); +MODULE_AUTHOR("Matt Flax <flatmax(a)flatmax.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/src4xxx.c b/sound/soc/codecs/src4xxx.c new file mode 100644 index 000000000000..46f693a59186 --- /dev/null +++ b/sound/soc/codecs/src4xxx.c @@ -0,0 +1,508 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * TI SRC4xxx Audio Codec driver + * + * Copyright 2021-2022 Deqx Pty Ltd + * Author: Matt Flax <flatmax(a)flatmax.com> + */ + +#include <linux/module.h> + +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "src4xxx.h" + +struct src4xxx { + struct regmap *regmap; + bool master[2]; + int mclk_hz; + struct device *dev; +}; + +enum {SRC4XXX_PORTA, SRC4XXX_PORTB}; + +/* SRC attenuation */ +static const DECLARE_TLV_DB_SCALE(src_tlv, -12750, 50, 0); + +static const struct snd_kcontrol_new src4xxx_controls[] = { + SOC_DOUBLE_R_TLV("SRC vol", + SRC4XXX_SCR_CTL_30, SRC4XXX_SCR_CTL_31, 0, 255, 1, src_tlv), +}; + +/* I2S port control */ +static const char * const port_out_src_text[] = { + "loopback", "other_port", "DIR", "SRC" +}; +static SOC_ENUM_SINGLE_DECL(porta_out_src_enum, SRC4XXX_PORTA_CTL_03, 4, + port_out_src_text); +static SOC_ENUM_SINGLE_DECL(portb_out_src_enum, SRC4XXX_PORTB_CTL_05, 4, + port_out_src_text); +static const struct snd_kcontrol_new porta_out_control = + SOC_DAPM_ENUM("Port A source select", porta_out_src_enum); +static const struct snd_kcontrol_new portb_out_control = + SOC_DAPM_ENUM("Port B source select", portb_out_src_enum); + +/* Digital audio transmitter control */ +static const char * const dit_mux_text[] = {"Port A", "Port B", "DIR", "SRC"}; +static SOC_ENUM_SINGLE_DECL(dit_mux_enum, SRC4XXX_TX_CTL_07, 3, dit_mux_text); +static const struct snd_kcontrol_new dit_mux_control = + SOC_DAPM_ENUM("DIT source", dit_mux_enum); + +/* SRC control */ +static const char * const src_in_text[] = {"Port A", "Port B", "DIR"}; +static SOC_ENUM_SINGLE_DECL(src_in_enum, SRC4XXX_SCR_CTL_2D, 0, src_in_text); +static const struct snd_kcontrol_new src_in_control = + SOC_DAPM_ENUM("SRC source select", src_in_enum); + +/* DIR control */ +static const char * const dir_in_text[] = {"Ch 1", "Ch 2", "Ch 3", "Ch 4"}; +static SOC_ENUM_SINGLE_DECL(dir_in_enum, SRC4XXX_RCV_CTL_0D, 0, dir_in_text); +static const struct snd_kcontrol_new dir_in_control = + SOC_DAPM_ENUM("Digital Input", dir_in_enum); + +static const struct snd_soc_dapm_widget src4xxx_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("loopback_A"), + SND_SOC_DAPM_INPUT("other_port_A"), + SND_SOC_DAPM_INPUT("DIR_A"), + SND_SOC_DAPM_INPUT("SRC_A"), + SND_SOC_DAPM_MUX("Port A source", + SND_SOC_NOPM, 0, 0, &porta_out_control), + + SND_SOC_DAPM_INPUT("loopback_B"), + SND_SOC_DAPM_INPUT("other_port_B"), + SND_SOC_DAPM_INPUT("DIR_B"), + SND_SOC_DAPM_INPUT("SRC_B"), + SND_SOC_DAPM_MUX("Port B source", + SND_SOC_NOPM, 0, 0, &portb_out_control), + + SND_SOC_DAPM_INPUT("Port_A"), + SND_SOC_DAPM_INPUT("Port_B"), + SND_SOC_DAPM_INPUT("DIR_"), + + /* Digital audio receivers and transmitters */ + SND_SOC_DAPM_OUTPUT("DIR_OUT"), + SND_SOC_DAPM_OUTPUT("SRC_OUT"), + SND_SOC_DAPM_MUX("DIT Out Src", SRC4XXX_PWR_RST_01, + SRC4XXX_ENABLE_DIT_SHIFT, 1, &dit_mux_control), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF_A_RX", "Playback A", 0, + SRC4XXX_PWR_RST_01, SRC4XXX_ENABLE_PORT_A_SHIFT, 1), + SND_SOC_DAPM_AIF_OUT("AIF_A_TX", "Capture A", 0, + SRC4XXX_PWR_RST_01, SRC4XXX_ENABLE_PORT_A_SHIFT, 1), + SND_SOC_DAPM_AIF_IN("AIF_B_RX", "Playback B", 0, + SRC4XXX_PWR_RST_01, SRC4XXX_ENABLE_PORT_B_SHIFT, 1), + SND_SOC_DAPM_AIF_OUT("AIF_B_TX", "Capture B", 0, + SRC4XXX_PWR_RST_01, SRC4XXX_ENABLE_PORT_B_SHIFT, 1), + + SND_SOC_DAPM_MUX("SRC source", SND_SOC_NOPM, 0, 0, &src_in_control), + + SND_SOC_DAPM_INPUT("MCLK"), + SND_SOC_DAPM_INPUT("RXMCLKI"), + SND_SOC_DAPM_INPUT("RXMCLKO"), + + SND_SOC_DAPM_INPUT("RX1"), + SND_SOC_DAPM_INPUT("RX2"), + SND_SOC_DAPM_INPUT("RX3"), + SND_SOC_DAPM_INPUT("RX4"), + SND_SOC_DAPM_MUX("Digital Input", SRC4XXX_PWR_RST_01, + SRC4XXX_ENABLE_DIR_SHIFT, 1, &dir_in_control), +}; + +static const struct snd_soc_dapm_route src4xxx_audio_routes[] = { + /* I2S Input to Output Routing */ + {"Port A source", "loopback", "loopback_A"}, + {"Port A source", "other_port", "other_port_A"}, + {"Port A source", "DIR", "DIR_A"}, + {"Port A source", "SRC", "SRC_A"}, + {"Port B source", "loopback", "loopback_B"}, + {"Port B source", "other_port", "other_port_B"}, + {"Port B source", "DIR", "DIR_B"}, + {"Port B source", "SRC", "SRC_B"}, + /* DIT muxing */ + {"DIT Out Src", "Port A", "Capture A"}, + {"DIT Out Src", "Port B", "Capture B"}, + {"DIT Out Src", "DIR", "DIR_OUT"}, + {"DIT Out Src", "SRC", "SRC_OUT"}, + + /* SRC input selection */ + {"SRC source", "Port A", "Port_A"}, + {"SRC source", "Port B", "Port_B"}, + {"SRC source", "DIR", "DIR_"}, + /* SRC mclk selection */ + {"SRC mclk source", "Master (MCLK)", "MCLK"}, + {"SRC mclk source", "Master (RXCLKI)", "RXMCLKI"}, + {"SRC mclk source", "Recovered receiver clk", "RXMCLKO"}, + /* DIR input selection */ + {"Digital Input", "Ch 1", "RX1"}, + {"Digital Input", "Ch 2", "RX2"}, + {"Digital Input", "Ch 3", "RX3"}, + {"Digital Input", "Ch 4", "RX4"}, +}; + + +static const struct snd_soc_component_driver src4xxx_driver = { + .controls = src4xxx_controls, + .num_controls = ARRAY_SIZE(src4xxx_controls), + + .dapm_widgets = src4xxx_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(src4xxx_dapm_widgets), + .dapm_routes = src4xxx_audio_routes, + .num_dapm_routes = ARRAY_SIZE(src4xxx_audio_routes), +}; + +static int src4xxx_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + struct src4xxx *src4xxx = snd_soc_component_get_drvdata(component); + unsigned int ctrl; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ctrl = SRC4XXX_BUS_MASTER; + src4xxx->master[dai->id] = true; + break; + case SND_SOC_DAIFMT_CBS_CFS: + ctrl = 0; + src4xxx->master[dai->id] = false; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl |= SRC4XXX_BUS_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl |= SRC4XXX_BUS_LEFT_J; + break; + case SND_SOC_DAIFMT_RIGHT_J: + ctrl |= SRC4XXX_BUS_RIGHT_J_24; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + regmap_update_bits(src4xxx->regmap, SRC4XXX_BUS_FMT(dai->id), + SRC4XXX_BUS_FMT_MS_MASK, ctrl); + + return 0; +} + +static int src4xxx_set_mclk_hz(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_component *component = codec_dai->component; + struct src4xxx *src4xxx = snd_soc_component_get_drvdata(component); + + dev_info(component->dev, "changing mclk rate from %d to %d Hz\n", + src4xxx->mclk_hz, freq); + src4xxx->mclk_hz = freq; + + return 0; +} + +static int src4xxx_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 src4xxx *src4xxx = snd_soc_component_get_drvdata(component); + unsigned int mclk_div; + int val, pj, jd, d; + int reg; + int ret; + + switch (dai->id) { + case SRC4XXX_PORTB: + reg = SRC4XXX_PORTB_CTL_06; + break; + default: + reg = SRC4XXX_PORTA_CTL_04; + } + + if (src4xxx->master[dai->id]) { + mclk_div = src4xxx->mclk_hz/params_rate(params); + if (src4xxx->mclk_hz != mclk_div*params_rate(params)) { + dev_err(component->dev, + "mclk %d / rate %d has a remainder.\n", + src4xxx->mclk_hz, params_rate(params)); + return -EINVAL; + } + + val = ((int)mclk_div - 128) / 128; + if ((val < 0) | (val > 3)) { + dev_err(component->dev, + "div register setting %d is out of range\n", + val); + dev_err(component->dev, + "unsupported sample rate %d Hz for the master clock of %d Hz\n", + params_rate(params), src4xxx->mclk_hz); + return -EINVAL; + } + + /* set the TX DIV */ + ret = regmap_update_bits(src4xxx->regmap, + SRC4XXX_TX_CTL_07, SRC4XXX_TX_MCLK_DIV_MASK, + val<<SRC4XXX_TX_MCLK_DIV_SHIFT); + if (ret) { + dev_err(component->dev, + "Couldn't set the TX's div register to %d << %d = 0x%x\n", + val, SRC4XXX_TX_MCLK_DIV_SHIFT, + val<<SRC4XXX_TX_MCLK_DIV_SHIFT); + return ret; + } + + /* set the PLL for the digital receiver */ + switch (src4xxx->mclk_hz) { + case 24576000: + pj = 0x22; + jd = 0x00; + d = 0x00; + break; + case 22579200: + pj = 0x22; + jd = 0x1b; + d = 0xa3; + break; + default: + /* don't error out here, + * other parts of the chip are still functional + */ + dev_info(component->dev, + "Couldn't set the RCV PLL as this master clock rate is unknown\n"); + } + ret = regmap_write(src4xxx->regmap, SRC4XXX_RCV_PLL_0F, pj); + if (ret < 0) + dev_err(component->dev, + "Failed to update PLL register 0x%x\n", + SRC4XXX_RCV_PLL_0F); + ret = regmap_write(src4xxx->regmap, SRC4XXX_RCV_PLL_10, jd); + if (ret < 0) + dev_err(component->dev, + "Failed to update PLL register 0x%x\n", + SRC4XXX_RCV_PLL_10); + ret = regmap_write(src4xxx->regmap, SRC4XXX_RCV_PLL_11, d); + if (ret < 0) + dev_err(component->dev, + "Failed to update PLL register 0x%x\n", + SRC4XXX_RCV_PLL_11); + + ret = regmap_update_bits(src4xxx->regmap, + SRC4XXX_TX_CTL_07, SRC4XXX_TX_MCLK_DIV_MASK, + val<<SRC4XXX_TX_MCLK_DIV_SHIFT); + if (ret < 0) { + dev_err(component->dev, + "Couldn't set the TX's div register to %d << %d = 0x%x\n", + val, SRC4XXX_TX_MCLK_DIV_SHIFT, + val<<SRC4XXX_TX_MCLK_DIV_SHIFT); + return ret; + } + + return regmap_update_bits(src4xxx->regmap, reg, + SRC4XXX_MCLK_DIV_MASK, val); + } else + dev_info(dai->dev, "not setting up MCLK as not master\n"); + + return 0; +}; + +static const struct snd_soc_dai_ops src4xxx_dai_ops = { + .hw_params = src4xxx_hw_params, + .set_sysclk = src4xxx_set_mclk_hz, + .set_fmt = src4xxx_set_dai_fmt, +}; + +#define SRC4XXX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) +#define SRC4XXX_RATES (SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000|\ + SNDRV_PCM_RATE_88200|\ + SNDRV_PCM_RATE_96000|\ + SNDRV_PCM_RATE_176400|\ + SNDRV_PCM_RATE_192000) + +static struct snd_soc_dai_driver src4xxx_dai_driver[] = { + { + .id = SRC4XXX_PORTA, + .name = "src4xxx-portA", + .playback = { + .stream_name = "Playback A", + .channels_min = 2, + .channels_max = 2, + .rates = SRC4XXX_RATES, + .formats = SRC4XXX_FORMATS, + }, + .capture = { + .stream_name = "Capture A", + .channels_min = 2, + .channels_max = 2, + .rates = SRC4XXX_RATES, + .formats = SRC4XXX_FORMATS, + }, + .ops = &src4xxx_dai_ops, + }, + { + .id = SRC4XXX_PORTB, + .name = "src4xxx-portB", + .playback = { + .stream_name = "Playback B", + .channels_min = 2, + .channels_max = 2, + .rates = SRC4XXX_RATES, + .formats = SRC4XXX_FORMATS, + }, + .capture = { + .stream_name = "Capture B", + .channels_min = 2, + .channels_max = 2, + .rates = SRC4XXX_RATES, + .formats = SRC4XXX_FORMATS, + }, + .ops = &src4xxx_dai_ops, + }, +}; + +static const struct reg_default src4xxx_reg_defaults[] = { + { SRC4XXX_PWR_RST_01, 0x00 }, /* all powered down intially */ + { SRC4XXX_PORTA_CTL_03, 0x00 }, + { SRC4XXX_PORTA_CTL_04, 0x00 }, + { SRC4XXX_PORTB_CTL_05, 0x00 }, + { SRC4XXX_PORTB_CTL_06, 0x00 }, + { SRC4XXX_TX_CTL_07, 0x00 }, + { SRC4XXX_TX_CTL_08, 0x00 }, + { SRC4XXX_TX_CTL_09, 0x00 }, + { SRC4XXX_SRC_DIT_IRQ_MSK_0B, 0x00 }, + { SRC4XXX_SRC_DIT_IRQ_MODE_0C, 0x00 }, + { SRC4XXX_RCV_CTL_0D, 0x00 }, + { SRC4XXX_RCV_CTL_0E, 0x00 }, + { SRC4XXX_RCV_PLL_0F, 0x00 }, /* not spec. in the datasheet */ + { SRC4XXX_RCV_PLL_10, 0xff }, /* not spec. in the datasheet */ + { SRC4XXX_RCV_PLL_11, 0xff }, /* not spec. in the datasheet */ + { SRC4XXX_RVC_IRQ_MSK_16, 0x00 }, + { SRC4XXX_RVC_IRQ_MSK_17, 0x00 }, + { SRC4XXX_RVC_IRQ_MODE_18, 0x00 }, + { SRC4XXX_RVC_IRQ_MODE_19, 0x00 }, + { SRC4XXX_RVC_IRQ_MODE_1A, 0x00 }, + { SRC4XXX_GPIO_1_1B, 0x00 }, + { SRC4XXX_GPIO_2_1C, 0x00 }, + { SRC4XXX_GPIO_3_1D, 0x00 }, + { SRC4XXX_GPIO_4_1E, 0x00 }, + { SRC4XXX_SCR_CTL_2D, 0x00 }, + { SRC4XXX_SCR_CTL_2E, 0x00 }, + { SRC4XXX_SCR_CTL_2F, 0x00 }, + { SRC4XXX_SCR_CTL_30, 0x00 }, + { SRC4XXX_SCR_CTL_31, 0x00 }, +}; + +int src4xxx_probe(struct device *dev, struct regmap *regmap, + void (*switch_mode)(struct device *dev)) +{ + struct src4xxx *src4xxx; + int ret; + + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + src4xxx = devm_kzalloc(dev, sizeof(*src4xxx), GFP_KERNEL); + if (!src4xxx) + return -ENOMEM; + + src4xxx->regmap = regmap; + src4xxx->dev = dev; + src4xxx->mclk_hz = 0; /* mclk has not been configured yet */ + dev_set_drvdata(dev, src4xxx); + + ret = regmap_write(regmap, SRC4XXX_PWR_RST_01, SRC4XXX_RESET); + if (ret < 0) + dev_err(dev, "Failed to issue reset: %d\n", ret); + usleep_range(1, 500); /* sleep for more then 500 ns */ + ret = regmap_write(regmap, SRC4XXX_PWR_RST_01, SRC4XXX_POWER_DOWN); + if (ret < 0) + dev_err(dev, "Failed to decommission reset: %d\n", ret); + usleep_range(500, 1000); /* sleep for 500 us or more */ + + ret = regmap_update_bits(src4xxx->regmap, SRC4XXX_PWR_RST_01, + SRC4XXX_POWER_ENABLE, SRC4XXX_POWER_ENABLE); + if (ret < 0) + dev_err(dev, "Failed to port A and B : %d\n", ret); + + /* set receiver to use master clock (rcv mclk is most likely jittery) */ + ret = regmap_update_bits(src4xxx->regmap, SRC4XXX_RCV_CTL_0D, + SRC4XXX_RXCLK_MCLK, SRC4XXX_RXCLK_MCLK); + if (ret < 0) + dev_err(dev, + "Failed to enable mclk as the PLL1 DIR reference : %d\n", ret); + + /* default to leaving the PLL2 running on loss of lock, divide by 8 */ + ret = regmap_update_bits(src4xxx->regmap, SRC4XXX_RCV_CTL_0E, + SRC4XXX_PLL2_DIV_8 | SRC4XXX_REC_MCLK_EN | SRC4XXX_PLL2_LOL, + SRC4XXX_PLL2_DIV_8 | SRC4XXX_REC_MCLK_EN | SRC4XXX_PLL2_LOL); + if (ret < 0) + dev_err(dev, "Failed to enable mclk rec and div : %d\n", ret); + + ret = devm_snd_soc_register_component(dev, &src4xxx_driver, + src4xxx_dai_driver, ARRAY_SIZE(src4xxx_dai_driver)); + if (ret == 0) + dev_info(dev, "src4392 probe ok %d\n", ret); + return ret; +} +EXPORT_SYMBOL_GPL(src4xxx_probe); + +static bool src4xxx_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case SRC4XXX_RES_00: + case SRC4XXX_GLOBAL_ITR_STS_02: + case SRC4XXX_SRC_DIT_STS_0A: + case SRC4XXX_NON_AUDIO_D_12: + case SRC4XXX_RVC_STS_13: + case SRC4XXX_RVC_STS_14: + case SRC4XXX_RVC_STS_15: + case SRC4XXX_SUB_CODE_1F: + case SRC4XXX_SUB_CODE_20: + case SRC4XXX_SUB_CODE_21: + case SRC4XXX_SUB_CODE_22: + case SRC4XXX_SUB_CODE_23: + case SRC4XXX_SUB_CODE_24: + case SRC4XXX_SUB_CODE_25: + case SRC4XXX_SUB_CODE_26: + case SRC4XXX_SUB_CODE_27: + case SRC4XXX_SUB_CODE_28: + case SRC4XXX_PC_PREAMBLE_HI_29: + case SRC4XXX_PC_PREAMBLE_LO_2A: + case SRC4XXX_PD_PREAMBLE_HI_2B: + case SRC4XXX_PC_PREAMBLE_LO_2C: + case SRC4XXX_IO_RATIO_32: + case SRC4XXX_IO_RATIO_33: + return true; + } + + if (reg > SRC4XXX_IO_RATIO_33 && reg < SRC4XXX_PAGE_SEL_7F) + return true; + + return false; +} + +const struct regmap_config src4xxx_regmap_config = { + .val_bits = 8, + .reg_bits = 8, + .max_register = SRC4XXX_IO_RATIO_33, + + .reg_defaults = src4xxx_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(src4xxx_reg_defaults), + .volatile_reg = src4xxx_volatile_register, + .cache_type = REGCACHE_RBTREE, +}; +EXPORT_SYMBOL_GPL(src4xxx_regmap_config); + +MODULE_DESCRIPTION("ASoC SRC4XXX CODEC driver"); +MODULE_AUTHOR("Matt Flax <flatmax(a)flatmax.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/src4xxx.h b/sound/soc/codecs/src4xxx.h new file mode 100644 index 000000000000..0407db8a8445 --- /dev/null +++ b/sound/soc/codecs/src4xxx.h @@ -0,0 +1,115 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * src4xxx.h -- SRC4XXX ALSA SoC audio driver + * + * Copyright 2021-2022 Deqx Pty Ltd + * Author: Matt R Flax <flatmax(a)flatmax.com> + */ + +#ifndef __SRC4XXX_H__ +#define __SRC4XXX_H__ + +#define SRC4XXX_RES_00 0x00 +#define SRC4XXX_PWR_RST_01 0x01 +#define SRC4XXX_RESET 0x80 +#define SRC4XXX_POWER_DOWN 0x00 +#define SRC4XXX_POWER_ENABLE 0x20 +#define SRC4XXX_ENABLE_SRC 0x1 +#define SRC4XXX_ENABLE_SRC_SHIFT 0 +#define SRC4XXX_ENABLE_DIR 0x2 +#define SRC4XXX_ENABLE_DIR_SHIFT 1 +#define SRC4XXX_ENABLE_DIT 0x4 +#define SRC4XXX_ENABLE_DIT_SHIFT 2 +#define SRC4XXX_ENABLE_PORT_B 0x8 +#define SRC4XXX_ENABLE_PORT_B_SHIFT 3 +#define SRC4XXX_ENABLE_PORT_A 0x10 +#define SRC4XXX_ENABLE_PORT_A_SHIFT 4 + +#define SRC4XXX_PORTA_CTL_03 0x03 +#define SRC4XXX_BUS_MASTER 0x8 +#define SRC4XXX_BUS_LEFT_J 0x0 +#define SRC4XXX_BUS_I2S 0x1 +#define SRC4XXX_BUS_RIGHT_J_16 0x4 +#define SRC4XXX_BUS_RIGHT_J_18 0x5 +#define SRC4XXX_BUS_RIGHT_J_20 0x6 +#define SRC4XXX_BUS_RIGHT_J_24 0x7 +#define SRC4XXX_BUS_FMT_MS_MASK 0xf + +#define SRC4XXX_PORTA_CTL_04 0x04 +#define SRC4XXX_MCLK_DIV_MASK 0x3 + +#define SRC4XXX_BUS_FMT(id) (SRC4XXX_PORTA_CTL_03+2*id) +#define SRC4XXX_BUS_CLK(id) (SRC4XXX_PORTA_CTL_04+2*id) + +#define SRC4XXX_PORTB_CTL_05 0x05 +#define SRC4XXX_PORTB_CTL_06 0x06 + +#define SRC4XXX_TX_CTL_07 0x07 +#define SRC4XXX_TX_MCLK_DIV_MASK 0x60 +#define SRC4XXX_TX_MCLK_DIV_SHIFT 5 + +#define SRC4XXX_TX_CTL_08 0x08 +#define SRC4XXX_TX_CTL_09 0x09 +#define SRC4XXX_SRC_DIT_IRQ_MSK_0B 0x0B +#define SRC4XXX_SRC_BTI_EN 0x01 +#define SRC4XXX_SRC_TSLIP_EN 0x02 +#define SRC4XXX_SRC_DIT_IRQ_MODE_0C 0x0C +#define SRC4XXX_RCV_CTL_0D 0x0D +#define SRC4XXX_RXCLK_RXCKI 0x0 +#define SRC4XXX_RXCLK_MCLK 0x8 +#define SRC4XXX_RCV_CTL_0E 0x0E +#define SRC4XXX_REC_MCLK_EN 0x1 +#define SRC4XXX_PLL2_DIV_0 (0x0<<1) +#define SRC4XXX_PLL2_DIV_2 (0x1<<1) +#define SRC4XXX_PLL2_DIV_4 (0x2<<1) +#define SRC4XXX_PLL2_DIV_8 (0x3<<1) +#define SRC4XXX_PLL2_LOL 0x8 +#define SRC4XXX_RCV_PLL_0F 0x0F +#define SRC4XXX_RCV_PLL_10 0x10 +#define SRC4XXX_RCV_PLL_11 0x11 +#define SRC4XXX_RVC_IRQ_MSK_16 0x16 +#define SRC4XXX_RVC_IRQ_MSK_17 0x17 +#define SRC4XXX_RVC_IRQ_MODE_18 0x18 +#define SRC4XXX_RVC_IRQ_MODE_19 0x19 +#define SRC4XXX_RVC_IRQ_MODE_1A 0x1A +#define SRC4XXX_GPIO_1_1B 0x1B +#define SRC4XXX_GPIO_2_1C 0x1C +#define SRC4XXX_GPIO_3_1D 0x1D +#define SRC4XXX_GPIO_4_1E 0x1E +#define SRC4XXX_SCR_CTL_2D 0x2D +#define SRC4XXX_SCR_CTL_2E 0x2E +#define SRC4XXX_SCR_CTL_2F 0x2F +#define SRC4XXX_SCR_CTL_30 0x30 +#define SRC4XXX_SCR_CTL_31 0x31 +#define SRC4XXX_PAGE_SEL_7F 0x7F + +// read only registers +#define SRC4XXX_GLOBAL_ITR_STS_02 0x02 +#define SRC4XXX_SRC_DIT_STS_0A 0x0A +#define SRC4XXX_NON_AUDIO_D_12 0x12 +#define SRC4XXX_RVC_STS_13 0x13 +#define SRC4XXX_RVC_STS_14 0x14 +#define SRC4XXX_RVC_STS_15 0x15 +#define SRC4XXX_SUB_CODE_1F 0x1F +#define SRC4XXX_SUB_CODE_20 0x20 +#define SRC4XXX_SUB_CODE_21 0x21 +#define SRC4XXX_SUB_CODE_22 0x22 +#define SRC4XXX_SUB_CODE_23 0x23 +#define SRC4XXX_SUB_CODE_24 0x24 +#define SRC4XXX_SUB_CODE_25 0x25 +#define SRC4XXX_SUB_CODE_26 0x26 +#define SRC4XXX_SUB_CODE_27 0x27 +#define SRC4XXX_SUB_CODE_28 0x28 +#define SRC4XXX_PC_PREAMBLE_HI_29 0x29 +#define SRC4XXX_PC_PREAMBLE_LO_2A 0x2A +#define SRC4XXX_PD_PREAMBLE_HI_2B 0x2B +#define SRC4XXX_PC_PREAMBLE_LO_2C 0x2C +#define SRC4XXX_IO_RATIO_32 0x32 +#define SRC4XXX_IO_RATIO_33 0x33 + +int src4xxx_probe(struct device *dev, struct regmap *regmap, + void (*switch_mode)(struct device *dev)); +void src4xxx_remove(struct device *dev); +extern const struct regmap_config src4xxx_regmap_config; + +#endif /* __SRC4XXX_H__ */ -- 2.34.1

4 12

[PATCH] ASoC: ops: Don't modify the driver's plaform_max when reading state
by Mark Brown 22 Aug '22

22 Aug '22

Currently snd_soc_info_volsw() will set a platform_max based on the limit the control has if one is not already set. This isn't really great, we shouldn't be modifying the passed in driver data especially in a path like this which may not ever be executed or where we may execute other callbacks before this one. Instead make this function leave the data unchanged, and clarify things a bit by referring to max rather than platform_max within the function. platform_max is now applied as a limit after working out the natural maximum value for the control. This means that platform_max is no longer treated as a direct register value for controls were min is non-zero. The put() callbacks already validate on this basis, and there do not appear to be any in tree users that would be affected. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-ops.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 2d39370ddeca..93e72a016b4d 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -176,20 +176,21 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, { struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - int platform_max; + int max; - if (!mc->platform_max) - mc->platform_max = mc->max; - platform_max = mc->platform_max; + max = uinfo->value.integer.max = mc->max - mc->min; + if (mc->platform_max && mc->platform_max < max) + max = mc->platform_max; - if (platform_max == 1 && !strstr(kcontrol->id.name, " Volume")) + if (max == 1 && !strstr(kcontrol->id.name, " Volume")) uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; else uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1; uinfo->value.integer.min = 0; - uinfo->value.integer.max = platform_max - mc->min; + uinfo->value.integer.max = max; + return 0; } EXPORT_SYMBOL_GPL(snd_soc_info_volsw); -- 2.30.2

2 6

build failure of next-20220822 due to 4e6bedd3c396 ("ASoC: codecs: add support for the TI SRC4392 codec")
by Sudip Mukherjee (Codethink) 22 Aug '22

22 Aug '22

Hi All, Not sure if it has been reported, builds of arm64 with clang failed to build next-20220822 with the error: sound/soc/codecs/src4xxx.c:280:3: error: variable 'd' is used uninitialized whenever switch default is taken [-Werror,-Wsometimes-uninitialized] default: ^~~~~~~ sound/soc/codecs/src4xxx.c:298:59: note: uninitialized use occurs here ret = regmap_write(src4xxx->regmap, SRC4XXX_RCV_PLL_11, d); ^ sound/soc/codecs/src4xxx.c:223:20: note: initialize the variable 'd' to silence this warning int val, pj, jd, d; ^ = 0 sound/soc/codecs/src4xxx.c:280:3: error: variable 'jd' is used uninitialized whenever switch default is taken [-Werror,-Wsometimes-uninitialized] default: ^~~~~~~ sound/soc/codecs/src4xxx.c:293:59: note: uninitialized use occurs here ret = regmap_write(src4xxx->regmap, SRC4XXX_RCV_PLL_10, jd); ^~ sound/soc/codecs/src4xxx.c:223:17: note: initialize the variable 'jd' to silence this warning int val, pj, jd, d; ^ = 0 sound/soc/codecs/src4xxx.c:280:3: error: variable 'pj' is used uninitialized whenever switch default is taken [-Werror,-Wsometimes-uninitialized] default: ^~~~~~~ sound/soc/codecs/src4xxx.c:288:59: note: uninitialized use occurs here ret = regmap_write(src4xxx->regmap, SRC4XXX_RCV_PLL_0F, pj); ^~ sound/soc/codecs/src4xxx.c:223:13: note: initialize the variable 'pj' to silence this warning int val, pj, jd, d; ^ = 0 3 errors generated. git bisect pointed to 4e6bedd3c396 ("ASoC: codecs: add support for the TI SRC4392 codec"). I will be happy to test any patch or provide any extra log if needed. -- Regards Sudip

2 1

[PATCH] ASoC: soc-pcm.c: summarize related settings at soc_new_pcm()
by Kuninori Morimoto 22 Aug '22

22 Aug '22

soc_new_pcm() sets pcm->no_device_suspend, but it sets other pcm->xxx at the same function with different timing. pcm->no_device_suspend setup timing has no effect. This patch tidyup it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-pcm.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index ce4a4151e784..4baa24fbf924 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2898,6 +2898,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) rtd->pcm = pcm; pcm->nonatomic = rtd->dai_link->nonatomic; pcm->private_data = rtd; + pcm->no_device_suspend = true; if (rtd->dai_link->no_pcm || rtd->dai_link->params) { if (playback) @@ -2952,8 +2953,6 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) ret = snd_soc_pcm_component_new(rtd); if (ret < 0) return ret; - - pcm->no_device_suspend = true; out: dev_dbg(rtd->card->dev, "%s <-> %s mapping ok\n", soc_codec_dai_name(rtd), soc_cpu_dai_name(rtd)); -- 2.25.1

2 1

[PATCH] ASoC: samsung: Use iio_get_channel_type() accessor.
by Jonathan Cameron 22 Aug '22

22 Aug '22

From: Jonathan Cameron <Jonathan.Cameron(a)huawei.com> struct iio_chan_spec is meant to be opaque to IIO consumer drivers which should only use the interfaces in linux/iio/consumer.h. Use the provided accessor function to find get the type of the channel instead of directly reading it form the structure. Signed-off-by: Jonathan Cameron <Jonathan.Cameron(a)huawei.com> Cc: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> Cc: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/samsung/aries_wm8994.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/aries_wm8994.c b/sound/soc/samsung/aries_wm8994.c index e7d52d27132e..0fbbf3b02c09 100644 --- a/sound/soc/samsung/aries_wm8994.c +++ b/sound/soc/samsung/aries_wm8994.c @@ -1,7 +1,6 @@ // SPDX-License-Identifier: GPL-2.0+ #include <linux/extcon.h> #include <linux/iio/consumer.h> -#include <linux/iio/iio.h> #include <linux/input-event-codes.h> #include <linux/mfd/wm8994/registers.h> #include <linux/module.h> @@ -543,6 +542,7 @@ static int aries_audio_probe(struct platform_device *pdev) struct aries_wm8994_data *priv; struct snd_soc_dai_link *dai_link; const struct of_device_id *match; + enum iio_chan_type channel_type; int ret, i; if (!np) @@ -594,7 +594,11 @@ static int aries_audio_probe(struct platform_device *pdev) return dev_err_probe(dev, PTR_ERR(priv->adc), "Failed to get ADC channel"); - if (priv->adc->channel->type != IIO_VOLTAGE) + ret = iio_get_channel_type(priv->adc, &channel_type); + if (ret) + return dev_err_probe(dev, ret, + "Failed to get ADC channel type"); + if (channel_type != IIO_VOLTAGE) return -EINVAL; priv->gpio_headset_key = devm_gpiod_get(dev, "headset-key", -- 2.37.2

2 1

Re: [PATCH] drm/bridge: dw-hdmi-i2s: set insert_pcuv bit if hardware supports it
by Geraldo Nascimento 22 Aug '22

22 Aug '22

On Mon, Aug 22, 2022 at 02:27:26PM +0100, Mark Brown wrote: > On Sat, Aug 20, 2022 at 09:36:53PM -0300, Geraldo Nascimento wrote: > > > + /* > > + * dw-hdmi introduced insert_pcuv bit in version 2.10a. > > + * When set (1'b1), this bit enables the insertion of the PCUV > > + * (Parity, Channel Status, User bit and Validity) bits on the > > + * incoming audio stream (support limited to Linear PCM audio) > > + */ > > + > > + if (hdmi_read(audio, HDMI_DESIGN_ID) >= 0x21) > > + conf2 |= HDMI_AUD_CONF2_INSERT_PCUV; > > So what if we're not handlign linear PCM? Good question, Mark. For HBR and NL-PCM there are different bits that need to be set in aud_conf2 Sample Register, and then insert_pcuv bit needs to be dropped. I don't have a sink handy however to test the other modes. Do you have a suggestion on how to make this the default for L-PCM specifically? Thanks, Geraldo Nascimento

1 0

[PATCH] ASoC: mediatek: mt8186: fix DMIC record noise
by Jiaxin Yu 22 Aug '22

22 Aug '22

When the first DMIC recording is power down, mtkaif_dmic will be reset. This will cause configuration error in the second DMIC recording. So do not reset mtkaif_dmic except in "MTKAIF_DMIC Switch" kcontrol. Signed-off-by: Jiaxin Yu <jiaxin.yu(a)mediatek.com> --- sound/soc/mediatek/mt8186/mt8186-dai-adda.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/mediatek/mt8186/mt8186-dai-adda.c b/sound/soc/mediatek/mt8186/mt8186-dai-adda.c index 266704556f37..094402470dc2 100644 --- a/sound/soc/mediatek/mt8186/mt8186-dai-adda.c +++ b/sound/soc/mediatek/mt8186/mt8186-dai-adda.c @@ -271,9 +271,6 @@ static int mtk_adda_ul_event(struct snd_soc_dapm_widget *w, /* should delayed 1/fs(smallest is 8k) = 125us before afe off */ usleep_range(125, 135); mt8186_afe_gpio_request(afe->dev, false, MT8186_DAI_ADDA, 1); - - /* reset dmic */ - afe_priv->mtkaif_dmic = 0; break; default: break; -- 2.25.1

2 1

[PATCH] Add quirk to enable Avid Mbox 3 support
by connerknoxpublic＠gmail.com 22 Aug '22

22 Aug '22

From: Conner Knox <connerknoxpublic(a)gmail.com> Hi all, I believe these are the right emails to send this patch to, but if I've missed something please let me know. This patch enables the Avid Mbox 3 audio interface to be used through ALSA. It locks the device at a 48khz sample rate. Then it mutes and raises the appropriate channels/inputs to allow for expected functionality. Thanks for your time. --- sound/usb/quirks-table.h | 76 ++++++++++ sound/usb/quirks.c | 311 +++++++++++++++++++++++++++++++++++++++ 2 files changed, 387 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 0ea39565e623..5f43863e5a5f 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2978,6 +2978,82 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +/* DIGIDESIGN MBOX 3 */ +{ + USB_DEVICE(0x0dba, 0x5000), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Digidesign", + .product_name = "Mbox 3", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 4, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0x00, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { + 48000 + } + } + }, + { + .ifnum = 3, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 4, + .iface = 3, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x81, + .attributes = 0x00, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .maxpacksize = 0x009c, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { + 48000 + } + } + }, + { + .ifnum = 4, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } + }, + { + .ifnum = -1 + } + } + } +}, { /* Tascam US122 MKII - playback-only support */ USB_DEVICE_VENDOR_SPEC(0x0644, 0x8021), diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index ab9f3da49941..e7caa7eeb955 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1020,6 +1020,313 @@ static int snd_usb_axefx3_boot_quirk(struct usb_device *dev) return 0; } +static void mbox3_setup_48_24_magic(struct usb_device *dev) +{ + u8 com_buff[4]; + + /* Load 48000Hz rate into buffer */ + com_buff[0] = 0x80; + com_buff[1] = 0xbb; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + + /* Set 48000Hz sample rate */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 0x01, 0x21, 0x0100, 0x0001, &com_buff, 0x0004); + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 0x01, 0x21, 0x0100, 0x8101, &com_buff, 0x0004); + + /* Set clock source to Internal (as opposed to S/PDIF) */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x01; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0100, 0x8001, &com_buff, 1); + + /* Mute the hardware loopbacks to start the device in a known state. + * The Mbox 3 is little endian and + * the data written here seems a bit odd. + * 0x8000 (shown in big endian form) is muted + * and volume increases to 0xffff it seems. + * I've yet to observe 0xffff being sent. + * Instead, full volume seems to be 0x0000. + * Per my understanding of 16 bit integers, this is seems strange, but maybe I'm missing something + */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x80; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + /* Analogue input 1 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0110, 0x4001, &com_buff, 2); + /* Analogue input 1 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0111, 0x4001, &com_buff, 2); + /* Analogue input 2 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0114, 0x4001, &com_buff, 2); + /* Analogue input 2 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0115, 0x4001, &com_buff, 2); + /* Analogue input 3 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0118, 0x4001, &com_buff, 2); + /* Analogue input 3 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0119, 0x4001, &com_buff, 2); + /* Analogue input 4 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x011c, 0x4001, &com_buff, 2); + /* Analogue input 4 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x011d, 0x4001, &com_buff, 2); + + /* Set software sends to output */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + /* Analogue software return 1 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0100, 0x4001, &com_buff, 2); + /* Analogue software return 1 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0101, 0x4001, &com_buff, 2); + /* Analogue software return 2 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0104, 0x4001, &com_buff, 2); + /* Analogue software return 2 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0105, 0x4001, &com_buff, 2); + /* Analogue software return 3 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0108, 0x4001, &com_buff, 2); + /* Analogue software return 3 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0109, 0x4001, &com_buff, 2); + /* Analogue software return 4 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x010c, 0x4001, &com_buff, 2); + /* Analogue software return 4 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x010d, 0x4001, &com_buff, 2); + + /* Return to muting sends */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x80; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + /* Analogue fx return left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0120, 0x4001, &com_buff, 2); + /* Analogue fx return right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0121, 0x4001, &com_buff, 2); + + /* Analogue software input 1 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0100, 0x4201, &com_buff, 2); + /* Analogue software input 2 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0101, 0x4201, &com_buff, 2); + /* Analogue software input 3 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0102, 0x4201, &com_buff, 2); + /* Analogue software input 4 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0103, 0x4201, &com_buff, 2); + /* Analogue input 1 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0104, 0x4201, &com_buff, 2); + /* Analogue input 2 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0105, 0x4201, &com_buff, 2); + /* Analogue input 3 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0106, 0x4201, &com_buff, 2); + /* Analogue input 4 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0107, 0x4201, &com_buff, 2); + + /* Toggle allowing host control */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x02; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 3, 0x21, 0x0000, 0x2001, &com_buff, 1); + + /* Do not dim fx returns */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 3, 0x21, 0x0002, 0x2001, &com_buff, 1); + + /* Do not set fx returns to mono */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 3, 0x21, 0x0001, 0x2001, &com_buff, 1); + + /* Mute the S/PDIF hardware loopback + * same odd volume logic here as above + */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x80; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + /* S/PDIF hardware input 1 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0112, 0x4001, &com_buff, 2); + /* S/PDIF hardware input 1 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0113, 0x4001, &com_buff, 2); + /* S/PDIF hardware input 2 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0116, 0x4001, &com_buff, 2); + /* S/PDIF hardware input 2 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0117, 0x4001, &com_buff, 2); + /* S/PDIF hardware input 3 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x011a, 0x4001, &com_buff, 2); + /* S/PDIF hardware input 3 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x011b, 0x4001, &com_buff, 2); + /* S/PDIF hardware input 4 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x011e, 0x4001, &com_buff, 2); + /* S/PDIF hardware input 4 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x011f, 0x4001, &com_buff, 2); + /* S/PDIF software return 1 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0102, 0x4001, &com_buff, 2); + /* S/PDIF software return 1 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0103, 0x4001, &com_buff, 2); + /* S/PDIF software return 2 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0106, 0x4001, &com_buff, 2); + /* S/PDIF software return 2 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0107, 0x4001, &com_buff, 2); + /* S/PDIF software return 3 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x010a, 0x4001, &com_buff, 2); + /* S/PDIF software return 3 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x010b, 0x4001, &com_buff, 2); + /* S/PDIF software return 4 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x010e, 0x4001, &com_buff, 2); + /* S/PDIF software return 4 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x010f, 0x4001, &com_buff, 2); + /* S/PDIF fx returns left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0122, 0x4001, &com_buff, 2); + /* S/PDIF fx returns right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0123, 0x4001, &com_buff, 2); + + /* Set the dropdown "Effect" to the first option */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0200, 0x4301, &com_buff, 1); + + /* Set the effect duration to 0 */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0400, 0x4301, &com_buff, 2); + + /* Set the effect volume and feedback to 0 */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + /* feedback: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0500, 0x4301, &com_buff, 1); + /* volume: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0300, 0x4301, &com_buff, 1); + + /* Set soft button hold duration */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x05; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 3, 0x21, 0x0005, 0x2001, &com_buff, 1); + + /* Use dim LEDs for button of state */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 3, 0x21, 0x0004, 0x2001, &com_buff, 1); + + return; +} + +#define MBOX3_DESCRIPTOR_SIZE 464 + +static int snd_usb_mbox3_boot_quirk(struct usb_device *dev) +{ + struct usb_host_config *config = dev->actconfig; + int err; + int descriptor_size; + + descriptor_size = le16_to_cpu(get_cfg_desc(config)->wTotalLength); + + if (descriptor_size != MBOX3_DESCRIPTOR_SIZE) { + dev_err(&dev->dev, "Invalid descriptor size=%d.\n", descriptor_size); + return -ENODEV; + } + + dev_dbg(&dev->dev, "device initialised!\n"); + + err = usb_get_descriptor(dev, USB_DT_DEVICE, 0, + &dev->descriptor, sizeof(dev->descriptor)); + config = dev->actconfig; + if (err < 0) + dev_dbg(&dev->dev, "error usb_get_descriptor: %d\n", err); + + err = usb_reset_configuration(dev); + if (err < 0) + dev_dbg(&dev->dev, "error usb_reset_configuration: %d\n", err); + dev_dbg(&dev->dev, "mbox3_boot: new boot length = %d\n", + le16_to_cpu(get_cfg_desc(config)->wTotalLength)); + + mbox3_setup_48_24_magic(dev); + dev_info(&dev->dev, "Digidesign Mbox 3: 24bit 48kHz"); + + return 0; /* Successful boot */ +} #define MICROBOOK_BUF_SIZE 128 @@ -1324,6 +1631,10 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, case USB_ID(0x0dba, 0x3000): /* Digidesign Mbox 2 */ return snd_usb_mbox2_boot_quirk(dev); + case USB_ID(0x0dba, 0x5000): + /* Digidesign Mbox 3 */ + return snd_usb_mbox3_boot_quirk(dev); + case USB_ID(0x1235, 0x0010): /* Focusrite Novation Saffire 6 USB */ case USB_ID(0x1235, 0x0018): /* Focusrite Novation Twitch */ -- 2.35.1

4 8

The cs35l41 driver does not work on several laptop models that lack the _DSD block.
by XiaoYan Li 22 Aug '22

22 Aug '22

Hi, Several laptop models employ CSC3551 but lack a corresponding _DSD block in the ACPI table, including but not limited to: - Zenbook S 13 OLED (UM5302TA) - HP Envy x360 ey0xxx - Lenovo Legion 7i Gen 7 Here are some relevant links: - https://bugzilla.kernel.org/show_bug.cgi?id=216194 - https://lore.kernel.org/all/20220703053225.2203-1-xw897002528@gmail.com/T/#u - https://lore.kernel.org/all/20220811152923.10143-1-faenkhauser@gmail.com/T/… The following is a snippet of the ACPI table from my laptop (UM5302TA): --- Device (SPKR) { Name (_HID, "CSC3551") // _HID: Hardware ID Name (_SUB, "10431F12") // _SUB: Subsystem ID Name (_UID, One) // _UID: Unique ID Method (_CRS, 0, NotSerialized) // _CRS: Current Resource Settings { Name (RBUF, ResourceTemplate () { I2cSerialBusV2 (0x0040, ControllerInitiated, 0x000F4240, AddressingMode7Bit, "\\_SB.I2CA", 0x00, ResourceConsumer, , Exclusive, ) I2cSerialBusV2 (0x0041, ControllerInitiated, 0x000F4240, AddressingMode7Bit, "\\_SB.I2CA", 0x00, ResourceConsumer, , Exclusive, ) GpioIo (Exclusive, PullDown, 0x0000, 0x0000, IoRestrictionOutputOnly, "\\_SB.GPIO", 0x00, ResourceConsumer, , ) { // Pin list 0x0004 } GpioIo (Exclusive, PullUp, 0x0000, 0x0000, IoRestrictionInputOnly, "\\_SB.GPIO", 0x00, ResourceConsumer, , ) { // Pin list 0x009B } GpioIo (Shared, PullUp, 0x0064, 0x0000, IoRestrictionInputOnly, "\\_SB.GPIO", 0x00, ResourceConsumer, , ) { // Pin list 0x0009 } GpioInt (Edge, ActiveBoth, Shared, PullUp, 0x0064, "\\_SB.GPIO", 0x00, ResourceConsumer, , ) { // Pin list 0x0009 } }) Return (RBUF) /* \_SB_.I2CA.SPKR._CRS.RBUF */ } Method (_STA, 0, NotSerialized) // _STA: Status { If ((AMPD == Zero)) { Return (Zero) } Else { Return (0x0F) } } Method (_DIS, 0, NotSerialized) // _DIS: Disable Device { } } --- The Cirrus driver for Windows works perfectly. I don't see anything related to the _DSD stuff in the INF files that came with the driver (if I read them correctly). For reference, here are the two INF files from UM5302TA: https://gist.github.com/lixiaoyan/7e1f11b153b38339007f6ec9aa149f81 I think we should support those devices out of the box, as the Windows driver does. This problem does not appear to be limited to a single model; rather, it appears to be a widespread issue. At least, we should add some quirks for the models listed above. Thanks.

1 1

missing sound on kernel-5.15
by GitHub issues - opened 22 Aug '22

22 Aug '22

alsa-project/alsa-lib issue #264 was opened from chihhao-chen: I am testing many types of headsets on Android platform with different kernel versions. On kernel-5.10, calling sequence before playback is snd_usb_hw_params -> set highest sampling rate, no SAMPLING_FREQ_CONTROL USB request in this stage snd_pcm_release snd_usb_hw_params -> set proper sampling rate, still no SAMPLING_FREQ_CONTROL USB request snd_usb_pcm_parepare -> in configure_endpoint SAMPLING_FREQ_CONTROL USB request was sent With the same calling sequence, a slightly behavior change on kernel-5.15 is snd_usb_hw_params -> set highest sampling rate, SAMPLING_FREQ_CONTROL USB request was sent snd_pcm_release snd_usb_hw_params -> set proper sampling rate, SAMPLING_FREQ_CONTROL USB request was sent too (because configure_endpoint was moved to snd_usb_hw_params function) snd_usb_pcm_parepare -> no SAMPLING_FREQ_CONTROL USB request because of USB EP flag check I checked all USB packets and confirmed audio data output was correct. But there may be missing sound problem for the first sound with many headsets (for example Samsung headset). I found this issue is related to two-times sampling rate set request. (I tried forcibly skip first USB request, everything became okay.) So do you guys know why configure_endpoint was moved to snd_usb_hw_params? Or could you provide patch for this problem? Many thanks Chihhao Issue URL : https://github.com/alsa-project/alsa-lib/issues/264 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Do we need to check state for hw_free ?
by Kuninori Morimoto 22 Aug '22

22 Aug '22

Hi ALSA ML I wonder do we need to check state (A) during dpcm_be_dai_hw_free() ? I think users count check (B) is very enough... void dpcm_be_dai_hw_free() { ... (B) if (be->dpcm[stream].users > 1) continue; ^ if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && | (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && | (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) && (A) (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED) && | (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && | (be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND)) v continue; ... __soc_pcm_hw_free(be, be_substream); } Thank you for your help !! Best regards --- Kuninori Morimoto

1 0

bug: Fix verbose logging in GitHub Workflow
by GitHub pull_request - opened 22 Aug '22

22 Aug '22

alsa-project/alsa-ucm-conf pull request #202 was opened from PatrickLang: I noticed the GitHub Workflow `ucm_validate` was setting `LOGLEVEL` instead of `V` which is used in the `Makefile` to set the log level. The tests in this repo are still failing, but fixing the logging first should help. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/202 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/202.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2] ALSA: memalloc: Revive x86-specific WC page allocations again
by Takashi Iwai 21 Aug '22

21 Aug '22

We dropped the x86-specific hack for WC-page allocations with a hope that the standard dma_alloc_wc() works nowadays. Alas, it doesn't, and we need to take back some workaround again, but in a different form, as the previous one was broken for some platforms. This patch re-introduces the x86-specific WC-page allocations, but it uses rather the manual page allocations instead of dma_alloc_coherent(). The use of dma_alloc_coherent() was also a potential problem in the recent addition of the fallback allocation for noncontig pages, and this patch eliminates both at once. Fixes: 9882d63bea14 ("ALSA: memalloc: Drop x86-specific hack for WC allocations") Cc: <stable(a)vger.kernel.org> BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=216363 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: Use gfp_t sound/core/memalloc.c | 87 +++++++++++++++++++++++++++++++++++-------- 1 file changed, 71 insertions(+), 16 deletions(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index d3885cb02270..b665ac66ccbe 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -20,6 +20,13 @@ static const struct snd_malloc_ops *snd_dma_get_ops(struct snd_dma_buffer *dmab); +#ifdef CONFIG_SND_DMA_SGBUF +static void *do_alloc_fallback_pages(struct device *dev, size_t size, + dma_addr_t *addr, bool wc); +static void do_free_fallback_pages(void *p, size_t size, bool wc); +static void *snd_dma_sg_fallback_alloc(struct snd_dma_buffer *dmab, size_t size); +#endif + /* a cast to gfp flag from the dev pointer; for CONTINUOUS and VMALLOC types */ static inline gfp_t snd_mem_get_gfp_flags(const struct snd_dma_buffer *dmab, gfp_t default_gfp) @@ -277,16 +284,21 @@ EXPORT_SYMBOL(snd_sgbuf_get_chunk_size); /* * Continuous pages allocator */ -static void *snd_dma_continuous_alloc(struct snd_dma_buffer *dmab, size_t size) +static void *do_alloc_pages(size_t size, dma_addr_t *addr, gfp_t gfp) { - gfp_t gfp = snd_mem_get_gfp_flags(dmab, GFP_KERNEL); void *p = alloc_pages_exact(size, gfp); if (p) - dmab->addr = page_to_phys(virt_to_page(p)); + *addr = page_to_phys(virt_to_page(p)); return p; } +static void *snd_dma_continuous_alloc(struct snd_dma_buffer *dmab, size_t size) +{ + return do_alloc_pages(size, &dmab->addr, + snd_mem_get_gfp_flags(dmab, GFP_KERNEL)); +} + static void snd_dma_continuous_free(struct snd_dma_buffer *dmab) { free_pages_exact(dmab->area, dmab->bytes); @@ -463,6 +475,25 @@ static const struct snd_malloc_ops snd_dma_dev_ops = { /* * Write-combined pages */ +/* x86-specific allocations */ +#ifdef CONFIG_SND_DMA_SGBUF +static void *snd_dma_wc_alloc(struct snd_dma_buffer *dmab, size_t size) +{ + return do_alloc_fallback_pages(dmab->dev.dev, size, &dmab->addr, true); +} + +static void snd_dma_wc_free(struct snd_dma_buffer *dmab) +{ + do_free_fallback_pages(dmab->area, dmab->bytes, true); +} + +static int snd_dma_wc_mmap(struct snd_dma_buffer *dmab, + struct vm_area_struct *area) +{ + area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); + return snd_dma_continuous_mmap(dmab, area); +} +#else static void *snd_dma_wc_alloc(struct snd_dma_buffer *dmab, size_t size) { return dma_alloc_wc(dmab->dev.dev, size, &dmab->addr, DEFAULT_GFP); @@ -479,6 +510,7 @@ static int snd_dma_wc_mmap(struct snd_dma_buffer *dmab, return dma_mmap_wc(dmab->dev.dev, area, dmab->area, dmab->addr, dmab->bytes); } +#endif /* CONFIG_SND_DMA_SGBUF */ static const struct snd_malloc_ops snd_dma_wc_ops = { .alloc = snd_dma_wc_alloc, @@ -486,10 +518,6 @@ static const struct snd_malloc_ops snd_dma_wc_ops = { .mmap = snd_dma_wc_mmap, }; -#ifdef CONFIG_SND_DMA_SGBUF -static void *snd_dma_sg_fallback_alloc(struct snd_dma_buffer *dmab, size_t size); -#endif - /* * Non-contiguous pages allocator */ @@ -669,6 +697,37 @@ static const struct snd_malloc_ops snd_dma_sg_wc_ops = { .get_chunk_size = snd_dma_noncontig_get_chunk_size, }; +/* manual page allocations with wc setup */ +static void *do_alloc_fallback_pages(struct device *dev, size_t size, + dma_addr_t *addr, bool wc) +{ + gfp_t gfp = GFP_KERNEL | __GFP_NORETRY | __GFP_NOWARN; + void *p; + + again: + p = do_alloc_pages(size, addr, gfp); + if (!p || (*addr + size - 1) & ~dev->coherent_dma_mask) { + if (IS_ENABLED(CONFIG_ZONE_DMA32) && !(gfp & GFP_DMA32)) { + gfp |= GFP_DMA32; + goto again; + } + if (IS_ENABLED(CONFIG_ZONE_DMA) && !(gfp & GFP_DMA)) { + gfp = (gfp & ~GFP_DMA32) | GFP_DMA; + goto again; + } + } + if (p && wc) + set_memory_wc((unsigned long)(p), size >> PAGE_SHIFT); + return p; +} + +static void do_free_fallback_pages(void *p, size_t size, bool wc) +{ + if (wc) + set_memory_wb((unsigned long)(p), size >> PAGE_SHIFT); + free_pages_exact(p, size); +} + /* Fallback SG-buffer allocations for x86 */ struct snd_dma_sg_fallback { size_t count; @@ -679,14 +738,11 @@ struct snd_dma_sg_fallback { static void __snd_dma_sg_fallback_free(struct snd_dma_buffer *dmab, struct snd_dma_sg_fallback *sgbuf) { + bool wc = dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG_FALLBACK; size_t i; - if (sgbuf->count && dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG_FALLBACK) - set_pages_array_wb(sgbuf->pages, sgbuf->count); for (i = 0; i < sgbuf->count && sgbuf->pages[i]; i++) - dma_free_coherent(dmab->dev.dev, PAGE_SIZE, - page_address(sgbuf->pages[i]), - sgbuf->addrs[i]); + do_free_fallback_pages(page_address(sgbuf->pages[i]), PAGE_SIZE, wc); kvfree(sgbuf->pages); kvfree(sgbuf->addrs); kfree(sgbuf); @@ -698,6 +754,7 @@ static void *snd_dma_sg_fallback_alloc(struct snd_dma_buffer *dmab, size_t size) struct page **pages; size_t i, count; void *p; + bool wc = dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG_FALLBACK; sgbuf = kzalloc(sizeof(*sgbuf), GFP_KERNEL); if (!sgbuf) @@ -712,15 +769,13 @@ static void *snd_dma_sg_fallback_alloc(struct snd_dma_buffer *dmab, size_t size) goto error; for (i = 0; i < count; sgbuf->count++, i++) { - p = dma_alloc_coherent(dmab->dev.dev, PAGE_SIZE, - &sgbuf->addrs[i], DEFAULT_GFP); + p = do_alloc_fallback_pages(dmab->dev.dev, PAGE_SIZE, + &sgbuf->addrs[i], wc); if (!p) goto error; sgbuf->pages[i] = virt_to_page(p); } - if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG_FALLBACK) - set_pages_array_wc(pages, count); p = vmap(pages, count, VM_MAP, PAGE_KERNEL); if (!p) goto error; -- 2.35.3

1 0

[PATCH] ALSA: memalloc: Revive x86-specific WC page allocations again
by Takashi Iwai 21 Aug '22

21 Aug '22

We dropped the x86-specific hack for WC-page allocations with a hope that the standard dma_alloc_wc() works nowadays. Alas, it doesn't, and we need to take back some workaround again, but in a different form, as the previous one was broken for some platforms. This patch re-introduces the x86-specific WC-page allocations, but it uses rather the manual page allocations instead of dma_alloc_coherent(). The use of dma_alloc_coherent() was also a potential problem in the recent addition of the fallback allocation for noncontig pages, and this patch eliminates both at once. Fixes: 9882d63bea14 ("ALSA: memalloc: Drop x86-specific hack for WC allocations") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=216363 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/memalloc.c | 87 +++++++++++++++++++++++++++++++++++-------- 1 file changed, 71 insertions(+), 16 deletions(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index d3885cb02270..90d74855d988 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -20,6 +20,13 @@ static const struct snd_malloc_ops *snd_dma_get_ops(struct snd_dma_buffer *dmab); +#ifdef CONFIG_SND_DMA_SGBUF +static void *do_alloc_fallback_pages(struct device *dev, size_t size, + dma_addr_t *addr, bool wc); +static void do_free_fallback_pages(void *p, size_t size, bool wc); +static void *snd_dma_sg_fallback_alloc(struct snd_dma_buffer *dmab, size_t size); +#endif + /* a cast to gfp flag from the dev pointer; for CONTINUOUS and VMALLOC types */ static inline gfp_t snd_mem_get_gfp_flags(const struct snd_dma_buffer *dmab, gfp_t default_gfp) @@ -277,16 +284,21 @@ EXPORT_SYMBOL(snd_sgbuf_get_chunk_size); /* * Continuous pages allocator */ -static void *snd_dma_continuous_alloc(struct snd_dma_buffer *dmab, size_t size) +static void *do_alloc_pages(size_t size, dma_addr_t *addr, unsigned long gfp) { - gfp_t gfp = snd_mem_get_gfp_flags(dmab, GFP_KERNEL); void *p = alloc_pages_exact(size, gfp); if (p) - dmab->addr = page_to_phys(virt_to_page(p)); + *addr = page_to_phys(virt_to_page(p)); return p; } +static void *snd_dma_continuous_alloc(struct snd_dma_buffer *dmab, size_t size) +{ + return do_alloc_pages(size, &dmab->addr, + snd_mem_get_gfp_flags(dmab, GFP_KERNEL)); +} + static void snd_dma_continuous_free(struct snd_dma_buffer *dmab) { free_pages_exact(dmab->area, dmab->bytes); @@ -463,6 +475,25 @@ static const struct snd_malloc_ops snd_dma_dev_ops = { /* * Write-combined pages */ +/* x86-specific allocations */ +#ifdef CONFIG_SND_DMA_SGBUF +static void *snd_dma_wc_alloc(struct snd_dma_buffer *dmab, size_t size) +{ + return do_alloc_fallback_pages(dmab->dev.dev, size, &dmab->addr, true); +} + +static void snd_dma_wc_free(struct snd_dma_buffer *dmab) +{ + do_free_fallback_pages(dmab->area, dmab->bytes, true); +} + +static int snd_dma_wc_mmap(struct snd_dma_buffer *dmab, + struct vm_area_struct *area) +{ + area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); + return snd_dma_continuous_mmap(dmab, area); +} +#else static void *snd_dma_wc_alloc(struct snd_dma_buffer *dmab, size_t size) { return dma_alloc_wc(dmab->dev.dev, size, &dmab->addr, DEFAULT_GFP); @@ -479,6 +510,7 @@ static int snd_dma_wc_mmap(struct snd_dma_buffer *dmab, return dma_mmap_wc(dmab->dev.dev, area, dmab->area, dmab->addr, dmab->bytes); } +#endif /* CONFIG_SND_DMA_SGBUF */ static const struct snd_malloc_ops snd_dma_wc_ops = { .alloc = snd_dma_wc_alloc, @@ -486,10 +518,6 @@ static const struct snd_malloc_ops snd_dma_wc_ops = { .mmap = snd_dma_wc_mmap, }; -#ifdef CONFIG_SND_DMA_SGBUF -static void *snd_dma_sg_fallback_alloc(struct snd_dma_buffer *dmab, size_t size); -#endif - /* * Non-contiguous pages allocator */ @@ -669,6 +697,37 @@ static const struct snd_malloc_ops snd_dma_sg_wc_ops = { .get_chunk_size = snd_dma_noncontig_get_chunk_size, }; +/* manual page allocations with wc setup */ +static void *do_alloc_fallback_pages(struct device *dev, size_t size, + dma_addr_t *addr, bool wc) +{ + void *p; + unsigned long gfp = DEFAULT_GFP & ~__GFP_COMP; + + again: + p = do_alloc_pages(size, addr, gfp); + if (!p || (*addr + size - 1) & ~dev->coherent_dma_mask) { + if (IS_ENABLED(CONFIG_ZONE_DMA32) && !(gfp & GFP_DMA32)) { + gfp |= GFP_DMA32; + goto again; + } + if (IS_ENABLED(CONFIG_ZONE_DMA) && !(gfp & GFP_DMA)) { + gfp = (gfp & ~GFP_DMA32) | GFP_DMA; + goto again; + } + } + if (p && wc) + set_memory_wc((unsigned long)(p), size >> PAGE_SHIFT); + return p; +} + +static void do_free_fallback_pages(void *p, size_t size, bool wc) +{ + if (wc) + set_memory_wb((unsigned long)(p), size >> PAGE_SHIFT); + free_pages_exact(p, size); +} + /* Fallback SG-buffer allocations for x86 */ struct snd_dma_sg_fallback { size_t count; @@ -679,14 +738,11 @@ struct snd_dma_sg_fallback { static void __snd_dma_sg_fallback_free(struct snd_dma_buffer *dmab, struct snd_dma_sg_fallback *sgbuf) { + bool wc = dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG_FALLBACK; size_t i; - if (sgbuf->count && dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG_FALLBACK) - set_pages_array_wb(sgbuf->pages, sgbuf->count); for (i = 0; i < sgbuf->count && sgbuf->pages[i]; i++) - dma_free_coherent(dmab->dev.dev, PAGE_SIZE, - page_address(sgbuf->pages[i]), - sgbuf->addrs[i]); + do_free_fallback_pages(page_address(sgbuf->pages[i]), PAGE_SIZE, wc); kvfree(sgbuf->pages); kvfree(sgbuf->addrs); kfree(sgbuf); @@ -698,6 +754,7 @@ static void *snd_dma_sg_fallback_alloc(struct snd_dma_buffer *dmab, size_t size) struct page **pages; size_t i, count; void *p; + bool wc = dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG_FALLBACK; sgbuf = kzalloc(sizeof(*sgbuf), GFP_KERNEL); if (!sgbuf) @@ -712,15 +769,13 @@ static void *snd_dma_sg_fallback_alloc(struct snd_dma_buffer *dmab, size_t size) goto error; for (i = 0; i < count; sgbuf->count++, i++) { - p = dma_alloc_coherent(dmab->dev.dev, PAGE_SIZE, - &sgbuf->addrs[i], DEFAULT_GFP); + p = do_alloc_fallback_pages(dmab->dev.dev, PAGE_SIZE, + &sgbuf->addrs[i], wc); if (!p) goto error; sgbuf->pages[i] = virt_to_page(p); } - if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG_FALLBACK) - set_pages_array_wc(pages, count); p = vmap(pages, count, VM_MAP, PAGE_KERNEL); if (!p) goto error; -- 2.35.3

2 2

Re: Rockchip I2S commit possibly ignored
by sugar zhang 21 Aug '22

21 Aug '22

Hi Geraldo, Mark, 在 2022/8/17 1:54, Geraldo Nascimento 写道: > On Tue, Aug 16, 2022 at 04:22:37PM +0100, Mark Brown wrote: >> On Tue, Aug 16, 2022 at 10:37:18AM -0300, Geraldo Nascimento wrote: >> >>> This isn't my patch, it's a patch by Sugar Zhang from Rockchip that was >>> supposed to have been applied one year ago, give or take 10 days. I >>> can't resend a patch that wasn't authored by me. >> There's absolutely no problem with reposting someone else's patch - just >> add your Signed-off-by at the end of the signoff chain. > Mark, I'm in no position to lecture anyone, least of who, you, hard-working > ASoC maintainer that you are. But there are *tons* of problems with > reposting someone else's patch, even if they had been previously given > the green-light but misteriously vanished. > > You are assuming responsibility for someone else's work! Let's see in > this case what would happen: > > In this case, the Rockchip I2S missing reset affair, I have tested > the proposed changes, and they are OK, *if* you add the reset properties > in the device tree. > > For example, I was playing with HDMI sound on the RK3399pro Radxa Rock Pi > N10 and I added the following properties to my private rk3399.dtsi: > > --- a/arch/arm64/boot/dts/rockchip/rk3399.dtsi > +++ b/arch/arm64/boot/dts/rockchip/rk3399.dtsi > @@ -1695,6 +1695,8 @@ i2s2: i2s@ff8a0000 { > dma-names = "tx", "rx"; > clock-names = "i2s_clk", "i2s_hclk"; > clocks = <&cru SCLK_I2S2_8CH>, <&cru HCLK_I2S2_8CH>; > + reset-names = "reset-m", "reset-h"; > + resets = <&cru SRST_I2S2_8CH>, <&cru SRST_H_I2S2_8CH>; > power-domains = <&power RK3399_PD_SDIOAUDIO>; > #sound-dai-cells = <0>; > status = "disabled"; > > --- > > > And once I okay &i2s2 and &hdmi_sound inside rk3399pro-vmarc-som.dtsi I > get kind of working HDMI sound, very clipped, I have to software adjust > the volume to 30% to get clean, not speaker-blowing sound. But I > digress, the point that it kinda works is not the point at all. > > My main point is that without adding "resets" and "reset-names" to *at > least* every Rockchip device tree that enables sound over HDMI, just an > example, you get systems with non-working HDMI. I just tested it, I > omitted "resets" and "reset-names" from my device tree and then had > to SSH into the black screen to revert the changes to my boot partition. Actually, the reset is used for i2s SLAVE mode which may fail to clear if the MASTER side stop the clk before i2s done. in this situation, we do reset the controller and bring it back to normal state. But, for HDMI sound, the i2s works as MASTER mode and should not be fail to clear. could you share your I2S and HDMI register dump to me to dig out the root cause. OTOH，I have noticed that some patchs ignored on upstream, I will check and repost those patchs. > So it's not trivial to RESEND this. It amounts to device tree overhaul > of arch/arm64/boot/dts/rockchip > > This would require community effort of say, equivalent to LibreELEC > resources, which are not many, but they have enough patience to test > every proposed change to Rockchip device trees, and could help upstream > this. > > Thank you, > Geraldo Nascimento -- Best Regards! 张学广/Sugar 瑞芯微电子股份有限公司 Rockchip Electronics Co., Ltd.

2 1

Rockchip/rk3399-gru-sound: Use 48KHz sample rate for all devices
by GitHub pull_request - opened 20 Aug '22

20 Aug '22

alsa-project/alsa-ucm-conf pull request #201 was opened from alpernebbi: The rk3399-gru-kevin board has problems when trying to simultaneously playback and record audio with the default PulseAudio sample rate of 44.1KHz. When the following command is run, the playback starts a few seconds after the recording finishes. arecord -vvv -d 4 /dev/null & sleep 0.2; speaker-test -l 1 -p 100000 -t sine When the sample rates are set to 48KHz either with PA configuration or in the UCM, playback and recording both start immediately. Another example is when a music player is running and we start `arecord`, the music starts stuttering but `arecord` can't record anything either. Apparently, this is a hardware limitation due to the I2S lines being shared between the devices. Set all the device rates to 48KHz like Chrome OS does to make things work smoothly. [1] https://chromium-review.googlesource.com/389695 [2] https://chromium-review.googlesource.com/898682 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/201 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/201.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH RESEND 0/3] ASoC: mediatek: support SOF for mt8186
by Chunxu Li 20 Aug '22

20 Aug '22

From: "chunxu.li" <chunxu.li(a)mediatek.com> In these patches, we add SOF support for mt8186 which have HiFi5 DSP inside. Chunxu Li (3): ASoC: mediatek: mt8186: support DSP downlink ASoC: mediatek: mt8186: add SOF support on mt8186-mt6366-rt1019-rt5682s ASoC: mediatek: mt8186: add SOF support on mt8186-mt6366-da7219-max98357 sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 3 + sound/soc/mediatek/mt8186/mt8186-dai-i2s.c | 12 ++ .../mt8186/mt8186-mt6366-da7219-max98357.c | 168 ++++++++++++++++-- .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 168 ++++++++++++++++-- 4 files changed, 325 insertions(+), 26 deletions(-) -- 2.25.1

2 4

[RESEND PATCH v2 1/2] dt-bindings: dsp: fsl: Add SOF compatile string for i.MX8ULP
by Shengjiu Wang 20 Aug '22

20 Aug '22

Add SOF compatile string "fsl,imx8ulp-dsp" for supporting DSP device on i.MX8ULP platform. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Acked-by: Rob Herring <robh(a)kernel.org> --- changes in v2 - add acked-by Rob Documentation/devicetree/bindings/dsp/fsl,dsp.yaml | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/dsp/fsl,dsp.yaml b/Documentation/devicetree/bindings/dsp/fsl,dsp.yaml index e66ef2da7879..9af40da5688e 100644 --- a/Documentation/devicetree/bindings/dsp/fsl,dsp.yaml +++ b/Documentation/devicetree/bindings/dsp/fsl,dsp.yaml @@ -20,6 +20,7 @@ properties: - fsl,imx8qxp-dsp - fsl,imx8qm-dsp - fsl,imx8mp-dsp + - fsl,imx8ulp-dsp - fsl,imx8qxp-hifi4 - fsl,imx8qm-hifi4 - fsl,imx8mp-hifi4 -- 2.34.1

3 4

[PATCH -next 1/3] ASoC: amd: acp: add missing platform_device_unregister() in acp_pci_probe()
by Yang Yingliang 20 Aug '22

20 Aug '22

Add missing platform_device_unregister() in error path in acp_pci_probe(). Fixes: c49f5e74a11e ("ASoC: amd: acp: Add error handling cases") Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/amd/acp/acp-pci.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/amd/acp/acp-pci.c b/sound/soc/amd/acp/acp-pci.c index 2c8e960cc9a6..5bb23ebe1216 100644 --- a/sound/soc/amd/acp/acp-pci.c +++ b/sound/soc/amd/acp/acp-pci.c @@ -104,6 +104,7 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id addr = pci_resource_start(pci, 0); chip->base = devm_ioremap(&pci->dev, addr, pci_resource_len(pci, 0)); if (!chip->base) { + platform_device_unregister(dmic_dev); ret = -ENOMEM; goto release_regions; } -- 2.25.1

2 3

[PATCH] ASoC: amd: acp: remove unnecessary NULL checks
by Dan Carpenter 20 Aug '22

20 Aug '22

The list iterator can never be NULL. Delete the bogus NULL checks. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/amd/acp/acp-platform.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/acp/acp-platform.c b/sound/soc/amd/acp/acp-platform.c index beee53aedeaf..85a81add4ef9 100644 --- a/sound/soc/amd/acp/acp-platform.c +++ b/sound/soc/amd/acp/acp-platform.c @@ -106,14 +106,14 @@ static irqreturn_t i2s_irq_handler(int irq, void *data) spin_lock(&adata->acp_lock); list_for_each_entry(stream, &adata->stream_list, list) { - if (stream && (ext_intr_stat & stream->irq_bit)) { + if (ext_intr_stat & stream->irq_bit) { writel(stream->irq_bit, ACP_EXTERNAL_INTR_STAT(adata, rsrc->irqp_used)); snd_pcm_period_elapsed(stream->substream); i2s_flag = 1; } if (adata->rsrc->no_of_ctrls == 2) { - if (stream && (ext_intr_stat1 & stream->irq_bit)) { + if (ext_intr_stat1 & stream->irq_bit) { writel(stream->irq_bit, ACP_EXTERNAL_INTR_STAT(adata, (rsrc->irqp_used - 1))); snd_pcm_period_elapsed(stream->substream); -- 2.35.1

2 1

[PATCH] ASoC: Change handling of unimplemented set_bclk_ratio
by Martin Povišer 20 Aug '22

20 Aug '22

If a 'set_bclk_ratio' call is attempted on a DAI not implementing the method, make it an -ENOSUPP error instead of -EINVAL. Assume the DAI can still be okay with the ratio, just does not care to register a handler. No current in-tree users of snd_soc_dai_set_bclk_ratio seem to inspect the return value, but -ENOSUPP disables an error print from within the common soc_dai_ret return filter. With the new behavior a machine driver can do a blanket 'set_bclk_ratio' on all DAIs on a bus, some of which may care about the ratio, some of which may not. Signed-off-by: Martin Povišer <povik+lin(a)cutebit.org> --- sound/soc/soc-dai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index d530e8c2b77b..49752af0e205 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -124,7 +124,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll); */ int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) { - int ret = -EINVAL; + int ret = -ENOTSUPP; if (dai->driver->ops && dai->driver->ops->set_bclk_ratio) -- 2.33.0

2 1

[PATCH] soundwire: qcom: remove unneeded check
by Steev Klimaszewski 19 Aug '22

19 Aug '22

commit 1fd0d85affe4d6 ("soundwire: qcom: Add flag for software clock gating check") added a flag for software clock gating check, however in commit 33ba01788889666 ("soundwire: qcom: Add support for controlling audio CGCR from HLOS") the same check was added without the flag, so we would still end up failing the software clock gating check. Originally reported by Amit Pundir on DB845c, I also saw it on the Lenovo Yoga C630, right before the splat, we would see qcom-soundwire wcd934x-soundwire.6.auto: Failed to get audio_cgcr reset required for soundwire-v1.6.0 however, SDM845 has a soundwire-v1.3.0 Since the flag was added in 1fd0d85affe, lets just remove this one. Fixes: 33ba01788889 ("soundwire: qcom: Add support for controlling audio CGCR from HLOS") Signed-off-by: Steev Klimaszewski <steev(a)kali.org> Reported-by: Amit Pundir <amit.pundir(a)linaro.org> --- drivers/soundwire/qcom.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 709a7c1e0704..b621f7fb866f 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -1355,10 +1355,6 @@ static int qcom_swrm_probe(struct platform_device *pdev) ctrl->bus.compute_params = &qcom_swrm_compute_params; ctrl->bus.clk_stop_timeout = 300; - ctrl->audio_cgcr = devm_reset_control_get_exclusive(dev, "swr_audio_cgcr"); - if (IS_ERR(ctrl->audio_cgcr)) - dev_err(dev, "Failed to get audio_cgcr reset required for soundwire-v1.6.0\n"); - ret = qcom_swrm_get_port_config(ctrl); if (ret) goto err_clk; -- 2.35.1

2 2

[PATCH v2] ASoC: SOF: Fix compilation when HDA_AUDIO_CODEC config is disabled
by Cezary Rojewski 19 Aug '22

19 Aug '22

hda_codec_device_init() expects three parameters, not two. Fixes: 3fd63658caed ("ASoC: Intel: Drop hdac_ext usage for codec device creation") Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- Changes in v2: - adjusted SHA-1 provided in Fixes tag sound/soc/sof/intel/hda-codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index 73336648cd25..1e9afc48394c 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -213,7 +213,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, put_device(&codec->core.dev); } #else - codec = hda_codec_device_init(&hbus->core, address); + codec = hda_codec_device_init(&hbus->core, address, HDA_DEV_ASOC); ret = PTR_ERR_OR_ZERO(codec); #endif -- 2.25.1

3 2

[PATCH] ASoC: SOF: Fix compilation when HDA_AUDIO_CODEC config is disabled
by Cezary Rojewski 19 Aug '22

19 Aug '22

hda_codec_device_init() expects three parameters, not two. Fixes: f2183b08c81a ("ASoC: Intel: Drop hdac_ext usage for codec device creation") Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- sound/soc/sof/intel/hda-codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index 73336648cd25..1e9afc48394c 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -213,7 +213,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, put_device(&codec->core.dev); } #else - codec = hda_codec_device_init(&hbus->core, address); + codec = hda_codec_device_init(&hbus->core, address, HDA_DEV_ASOC); ret = PTR_ERR_OR_ZERO(codec); #endif -- 2.25.1

2 2

chromium - cracking pulseaudio issue (small buffer size settings - 512)
by GitHub issues - edited 19 Aug '22

19 Aug '22

alsa-project/alsa-lib issue #262 was edited from shmohawk: ``` Aug 17 01:11:23 hellokitty pulseaudio[12938]: [alsa-sink-CX20590 Analog] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write. Aug 17 01:11:23 hellokitty pulseaudio[12938]: [alsa-sink-CX20590 Analog] alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. Aug 17 01:11:23 hellokitty pulseaudio[12938]: [alsa-sink-CX20590 Analog] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. ``` It is symptomous with a crackling sound for ~5 seconds after any playback starts. alsa-lib 1.2.7.2 alsa-utils 1.2.7 alsa-plugins 1.2.2 Issue URL : https://github.com/alsa-project/alsa-lib/issues/262 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 0/3] ASoC platform driver for Apple MCA
by Martin Povišer 19 Aug '22

19 Aug '22

Hi, for review I am posting another version of the ASoC platform driver to be used on Apple M1/M2 platforms (some previous version was attached to the macaudio RFC v2 [0]). Martin Changes since [0]: - addition of locking (extra commit) - transition to set_bclk_ratio (instead of getting the bclk ratio from set_sysclk) - using shared reset control and documenting the reset in binding - formatting, comments, and a minor fix to hw driving [0] https://lore.kernel.org/asahi/20220606191910.16580-1-povik+lin@cutebit.org/ Martin Povišer (3): dt-bindings: sound: Add Apple MCA I2S transceiver ASoC: apple: mca: Start new platform driver ASoC: apple: mca: Add locks on foreign cluster access .../devicetree/bindings/sound/apple,mca.yaml | 109 ++ MAINTAINERS | 8 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/apple/Kconfig | 9 + sound/soc/apple/Makefile | 3 + sound/soc/apple/mca.c | 1173 +++++++++++++++++ 7 files changed, 1304 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/apple,mca.yaml create mode 100644 sound/soc/apple/Kconfig create mode 100644 sound/soc/apple/Makefile create mode 100644 sound/soc/apple/mca.c -- 2.33.0

3 11

ucm2: Add intial acpes8336 support
by GitHub pull_request - opened 19 Aug '22

19 Aug '22

alsa-project/alsa-ucm-conf pull request #200 was opened from yangxiaohua2009: This add support for es8336 on AMD Jadeite platform. ASLAINFO http://alsa-project.org/db/?f=b5aae41350b63c40c7743ba43ff8a7178ee93229 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/200 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/200.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

How trivial would it be to port the DICE driver from FFADO?
by forevernoob＠tutanota.com 19 Aug '22

19 Aug '22

Hello, I'm new to these kinds of mailing lists so please forgive me if I'm doing something dumb. I would love to see Allen & Heath Zed R16 as well as Midas Venice F32 support in ALSA. Currently (at least according to libffado-2.4.4/configuration), they are supported via the DICE driver: { # Allen and Heath Zed R16. Information from Brendan Pike. vendorid = 0x000004C4; modelid = 0x00000000; vendorname = "Allen and Heath"; modelname = "Zed R16"; driver = "DICE"; mixer = "Generic_Dice_EAP"; }, { # Midas Venice F32. Information from Jano Svitok. vendorid = 0x0010C73F; modelid = 0x00000001; vendorname = "Midas"; modelname = "Venice F32"; driver = "DICE"; mixer = "Generic_Dice_EAP"; }, In their source code I think it's this file: http://subversion.ffado.org/browser/branches/2.4.x/libffado/src/dice/dice_e… So my question would be: How difficult would it be to port this driver (more specifically: Port it sufficiently enough to support the aforementioned devices) ? Are we talking trivial enough for someone with minimal C experience, a dedicated team of ALSA devs or something else? Thanks in advance, FN

2 2

cracking pulseaudio issue
by GitHub issues - edited 19 Aug '22

19 Aug '22

alsa-project/alsa-lib issue #262 was edited from shmohawk: ``` Aug 17 01:11:23 hellokitty pulseaudio[12938]: [alsa-sink-CX20590 Analog] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write. Aug 17 01:11:23 hellokitty pulseaudio[12938]: [alsa-sink-CX20590 Analog] alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. Aug 17 01:11:23 hellokitty pulseaudio[12938]: [alsa-sink-CX20590 Analog] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. ``` It is symptomous with a crackling sound for ~5 seconds after any playback starts. alsa-lib 1.2.7.2 alsa-utils 1.2.7 alsa-plugins 1.2.2 Issue URL : https://github.com/alsa-project/alsa-lib/issues/262 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

snd_hda_intel: Woke up but no data
by GitHub issues - reopened 19 Aug '22

19 Aug '22

alsa-project/alsa-lib issue #262 was reopened from shmohawk: ``` Aug 17 01:11:23 hellokitty pulseaudio[12938]: [alsa-sink-CX20590 Analog] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write. Aug 17 01:11:23 hellokitty pulseaudio[12938]: [alsa-sink-CX20590 Analog] alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. Aug 17 01:11:23 hellokitty pulseaudio[12938]: [alsa-sink-CX20590 Analog] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. ``` It is symptomous with a crackling sound for ~5 seconds after any playback starts. alsa-lib 1.2.7.2 alsa-utils 1.2.7 alsa-plugins 1.2.2 Issue URL : https://github.com/alsa-project/alsa-lib/issues/262 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Raising volume past 100%
by David Griffith 18 Aug '22

18 Aug '22

After reinstalling Debian on a laptop, I once again had to face the problem of how to get volume out of the built-in speakers boosted. The last time I reinstalled, I somehow got the MATE volume control applet to let me turn the volume past 100% using the mouse wheel. Over the past two days I've been trying to rediscover what I neglected to write down. I'm already aware that by bringing up the Sound Preferences dialog, I can raise volume to 153% and no further. I also know that I can use pulseaudio-ctl(1) or pactl(1) to set the volume at any arbitrary level past 100%. I remember that I went looking for solutions other people found for this that involved editing /etc/asound.conf or ~/.asoundrc, but the best I can get with that is add an extra slider to alsamixer's display that acts as apreamplifier. I was going off of http://alien.slackbook.org/blog/adding-an-alsa-software-pre-amp-to-fix-low-…. What could I have possibly done to make this work the last time? -- David Griffith dave(a)661.org A: Because it fouls the order in which people normally read text. Q: Why is top-posting such a bad thing? A: Top-posting. Q: What is the most annoying thing in e-mail?

1 0

[PATCH 0/2] ASoC: Add a driver for the Iron Device SMA1303 Amp
by Kiseok Jo 18 Aug '22

18 Aug '22

This adds basic audio support for the Iron Device SMA1303 amplifier. Kiseok Jo (2): ASoC: sma1303: Add driver for Iron Device SMA1303 Amp ASoC: dt-bindings: sma1303: Iron Device SMA1303 Amp .../bindings/sound/irondevice,sma1303.yaml | 56 + sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sma1303.c | 3388 +++++++++++++++++ sound/soc/codecs/sma1303.h | 604 +++ 5 files changed, 4057 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/irondevice,sma1303.yaml create mode 100644 sound/soc/codecs/sma1303.c create mode 100644 sound/soc/codecs/sma1303.h -- 2.30.2

2 3

[PATCH -next] ASoC: SOF: ipc4-topology: fix wrong use of sizeof in sof_ipc4_widget_setup_comp_src()
by Yang Yingliang 18 Aug '22

18 Aug '22

It should be size of the struct sof_ipc4_src, not data pointer pass to sof_update_ipc_object(). Fixes: b85f4fc40d56 ("ASoC: SOF: add ipc4 SRC module support") Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/sof/ipc4-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index af072b484a60..c6abfaf5d532 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -771,7 +771,7 @@ static int sof_ipc4_widget_setup_comp_src(struct snd_sof_widget *swidget) goto err; ret = sof_update_ipc_object(scomp, src, SOF_SRC_TOKENS, swidget->tuples, - swidget->num_tuples, sizeof(src), 1); + swidget->num_tuples, sizeof(*src), 1); if (ret) { dev_err(scomp->dev, "Parsing SRC tokens failed\n"); goto err; -- 2.25.1

3 2

Add ASUS ROG Zenith II Extreme configuration
by GitHub pull_request - opened 18 Aug '22

18 Aug '22

alsa-project/alsa-lib pull request #263 was opened from nabajour: This points iec958_device to the correct PCM device. complements this [ALSA kernel patch](https://bugzilla.kernel.org/show_bug.cgi?id=211005). Request URL : https://github.com/alsa-project/alsa-lib/pull/263 Patch URL : https://github.com/alsa-project/alsa-lib/pull/263.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[bug report] ASoC: amd: Add common framework to support I2S on ACP SOC
by Dan Carpenter 18 Aug '22

18 Aug '22

Hello Ajit Kumar Pandey, The patch 623621a9f9e1: "ASoC: amd: Add common framework to support I2S on ACP SOC" from Oct 19, 2021, leads to the following Smatch static checker warning: sound/soc/amd/acp/acp-platform.c:199 acp_dma_open() warn: '&stream->list' not removed from list sound/soc/amd/acp/acp-platform.c 173 static int acp_dma_open(struct snd_soc_component *component, struct snd_pcm_substream *substream) 174 { 175 struct snd_pcm_runtime *runtime = substream->runtime; 176 struct device *dev = component->dev; 177 struct acp_dev_data *adata = dev_get_drvdata(dev); 178 struct acp_stream *stream; 179 int ret; 180 181 stream = kzalloc(sizeof(*stream), GFP_KERNEL); 182 if (!stream) 183 return -ENOMEM; 184 185 stream->substream = substream; 186 187 spin_lock_irq(&adata->acp_lock); 188 list_add_tail(&stream->list, &adata->stream_list); Stream added to the list 189 spin_unlock_irq(&adata->acp_lock); 190 191 if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) 192 runtime->hw = acp_pcm_hardware_playback; 193 else 194 runtime->hw = acp_pcm_hardware_capture; 195 196 ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); 197 if (ret < 0) { 198 dev_err(component->dev, "set integer constraint failed\n"); --> 199 kfree(stream); Use after free. 200 return ret; 201 } 202 runtime->private_data = stream; 203 204 writel(1, ACP_EXTERNAL_INTR_ENB(adata)); 205 206 return ret; Could we delay adding the stream to the list until right before this "return 0;"? 207 } regards, dan carpenter

1 0

[PATCH 1/2] ASoC: Intel: Skylake: fix error message of NHLT blob selection
by Icenowy Zheng 18 Aug '22

18 Aug '22

The query to select NHLT blob had been changed to use parameters of the pipeline instead of PCM, however the error message returned when it fails still uses PCM properties. Fix this inconsistency. Fixes: 87b265260046 ("ASoC: Intel: Skylake: Select proper format for NHLT blob") Signed-off-by: Icenowy Zheng <uwu(a)icenowy.me> --- sound/soc/intel/skylake/skl-topology.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 9bdf020a2b64..19994ec8bba1 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -1864,7 +1864,8 @@ static int skl_tplg_be_fill_pipe_params(struct snd_soc_dai *dai, } else { dev_err(dai->dev, "Blob NULL for id:%d type:%d dirn:%d ch:%d, freq:%d, fmt:%d\n", mconfig->vbus_id, link_type, params->stream, - params->ch, params->s_freq, params->s_fmt); + pipe_fmt->channels, pipe_fmt->freq, + pipe_fmt->bps); return -EINVAL; } -- 2.37.1

3 9