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[sound:topic/asoc-pcm-addr-fixes 20/24] sound/soc/amd/renoir/acp3x-pdm-dma.c:245:18: error: 'runtime' undeclared
by kernel test robot 01 Aug '21

01 Aug '21

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/asoc-pcm-addr-fixes head: 3ed85d6b39dbdab30d06d6c686739e12c0793cde commit: c4bb3c6064991e5668115533a840609b5dedfb8e [20/24] ASoC: amd: Fix reference to PCM buffer address config: x86_64-randconfig-a006-20210728 (attached as .config) compiler: gcc-10 (Ubuntu 10.3.0-1ubuntu1~20.04) 10.3.0 reproduce (this is a W=1 build): # https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/commit/?id=… git remote add sound https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git git fetch --no-tags sound topic/asoc-pcm-addr-fixes git checkout c4bb3c6064991e5668115533a840609b5dedfb8e # save the attached .config to linux build tree mkdir build_dir make W=1 O=build_dir ARCH=x86_64 SHELL=/bin/bash sound/soc/amd/renoir/ If you fix the issue, kindly add following tag as appropriate Reported-by: kernel test robot <lkp(a)intel.com> All errors (new ones prefixed by >>): sound/soc/amd/renoir/acp3x-pdm-dma.c: In function 'acp_pdm_dma_hw_params': >> sound/soc/amd/renoir/acp3x-pdm-dma.c:245:18: error: 'runtime' undeclared (first use in this function) 245 | rtd->dma_addr = runtime->dma_addr; | ^~~~~~~ sound/soc/amd/renoir/acp3x-pdm-dma.c:245:18: note: each undeclared identifier is reported only once for each function it appears in vim +/runtime +245 sound/soc/amd/renoir/acp3x-pdm-dma.c 232 233 static int acp_pdm_dma_hw_params(struct snd_soc_component *component, 234 struct snd_pcm_substream *substream, 235 struct snd_pcm_hw_params *params) 236 { 237 struct pdm_stream_instance *rtd; 238 size_t size, period_bytes; 239 240 rtd = substream->runtime->private_data; 241 if (!rtd) 242 return -EINVAL; 243 size = params_buffer_bytes(params); 244 period_bytes = params_period_bytes(params); > 245 rtd->dma_addr = runtime->dma_addr; 246 rtd->num_pages = (PAGE_ALIGN(size) >> PAGE_SHIFT); 247 config_acp_dma(rtd, substream->stream); 248 init_pdm_ring_buffer(MEM_WINDOW_START, size, period_bytes, 249 rtd->acp_base); 250 return 0; 251 } 252 --- 0-DAY CI Kernel Test Service, Intel Corporation https://lists.01.org/hyperkitty/list/kbuild-all@lists.01.org

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[PATCH] ALSA: memalloc: Fix regression with SNDRV_DMA_TYPE_CONTINUOUS
by Takashi Iwai 01 Aug '21

01 Aug '21

The recent code refactoring made the mmap of continuous pages to be done via the own helper snd_dma_continuous_mmap() with remap_pfn_range(). There I overlooked that dmab->addr isn't set for the allocation with SNDRV_DMA_TYPE_CONTINUOUS. This resulted always in an error at mmap with this buffer type on the system such as Intel SST Baytrail driver. This patch fixes the regression by passing the correct address. Fixes: 30b7ba6972d5 ("ALSA: core: Add continuous and vmalloc mmap ops") Reported-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/8d6674da-7d7b-803e-acc9-7de6cb1223fa@redhat.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/memalloc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 83b79edfa52d..439a358ecfe9 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -215,7 +215,7 @@ static int snd_dma_continuous_mmap(struct snd_dma_buffer *dmab, struct vm_area_struct *area) { return remap_pfn_range(area, area->vm_start, - dmab->addr >> PAGE_SHIFT, + page_to_pfn(virt_to_page(dmab->area)), area->vm_end - area->vm_start, area->vm_page_prot); } -- 2.26.2

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[5.14 regression] "ALSA: core: Add continuous and vmalloc mmap ops" breaks Intel SST audio
by Hans de Goede 01 Aug '21

01 Aug '21

Hi All;, As discussed in the "5.14 regression, Intel SST (BYT) audio no longer works" thread, audio on X86 devices using the Intel SST driver no longer works with 5.14. After poking at this for a while I've found the culprit: 30b7ba6972d5 ("ALSA: core: Add continuous and vmalloc mmap ops") If I revert that single commit then everything works fine again with 5.14. This is with a Fedora 34 userspace using pipewire as audiodaemon Regards, Hans

2 3

[PATCH] ALSA: usb-audio: make array static const, makes object smaller
by Colin King 01 Aug '21

01 Aug '21

From: Colin Ian King <colin.king(a)canonical.com> Don't populate array names_to_check on the stack but instead it static. Makes the object code smaller by 56 bytes. Before: text data bss dec hex filename 103512 34380 0 137892 21aa4 ./sound/usb/mixer.o After: text data bss dec hex filename 103264 34572 0 137836 21a6c ./sound/usb/mixer.o gcc version 10.2.0) Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/usb/mixer.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index f4cdaf1ba44a..aec2499284a5 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1572,8 +1572,9 @@ static size_t append_ctl_name(struct snd_kcontrol *kctl, const char *str) static void check_no_speaker_on_headset(struct snd_kcontrol *kctl, struct snd_card *card) { - const char *names_to_check[] = { - "Headset", "headset", "Headphone", "headphone", NULL}; + static const char *names_to_check[] = { + "Headset", "headset", "Headphone", "headphone", NULL + }; const char **s; bool found = false; -- 2.31.1

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Incorrect device setting for M-Audio Audiophile USB sound card
by GitHub issues - opened 31 Jul '21

31 Jul '21

alsa-project/alsa-lib issue #166 was opened from michele-perrone: ### M-Audio Audiophile USB - fix needed There is a problem with the ALSA configuration of the "M-Audio Audiophile USB" sound card. From the linux kernel documentation (https://www.kernel.org/doc/html/latest/sound/cards/audiophile-usb.html) you can see that this card has four audio interfaces: * Analog 2-channel Input (Ai) * Analog 2-channel Output (Ao) * Digital 2-channel Input (Di) * Digital 2-channel Output (Do) These interfaces are distributed over three devices of the same card: * hw:USBtm,0 is Ao in playback and Di in capture * hw:USBtm,1 is Do in playback and Ai in capture * hw:USBtm,2 is Do in AC3/DTS passthrough mode Now, this is where ALSA makes a bit of a mess. If you look at the output of `aplay - L` (see attachment aplay_out.txt), you can see that the analog audio output is associated to CARD=USBtm,DEV=0 (which is correct), **but** you can also see that two digital outputs are assigned, one to CARD=USBtm,DEV=0 (**wrong!**) and another one to CARD=USBtm,DEV=1 (correct). Speaking now of `arecord - L` (see attachment arecord_out.txt), there is only one input assigned to CARD=USBtm,DEV=0, and it is marked as "Front output / input" instead of SPDIF digital input. And most importantly, the analog input configuration, which should correspond to CARD=USBtm,DEV=1, is completely absent. The result of this is that audio applications relying on ALSA don't know where to pick the correct inputs and outputs. If want both Ao and Ai to work in Jack, I have to select hw:USBtm,1 for the input and hw:USBtm,0 for the output. Alternatively, I have to create an `.asoundrc` file with the `asym` module (see attachment) and then run Jack with the name of my virtual device. However, this is just a workaround. I'm not familiar with the alsa source but I know my way around C code. If you tell me where to look, I may be able to help fixing this. Cheers! Michele Attachments: [aplay_out.txt](https://github.com/alsa-project/alsa-lib/files/6911115/aplay… [arecord_out.txt](https://github.com/alsa-project/alsa-lib/files/6911116/are… [asoundrc.txt](https://github.com/alsa-project/alsa-lib/files/6911119/asound… Issue URL : https://github.com/alsa-project/alsa-lib/issues/166 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH] ALSA: pcxhr: use __func__ to get funcion's name in an output message
by Jason Wang 31 Jul '21

31 Jul '21

Prefer using '"%s...", __func__' to get current function's name in an output message. Signed-off-by: Jason Wang <wangborong(a)cdjrlc.com> --- sound/pci/pcxhr/pcxhr.c | 47 ++++++++++++++++++++--------------------- 1 file changed, 23 insertions(+), 24 deletions(-) diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index 96b9371c201a..242bd7e04b3e 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -366,7 +366,7 @@ static int pcxhr_sub_set_clock(struct pcxhr_mgr *mgr, mgr->codec_speed = speed; /* save new codec speed */ } - dev_dbg(&mgr->pci->dev, "pcxhr_sub_set_clock to %dHz (realfreq=%d)\n", + dev_dbg(&mgr->pci->dev, "%s to %dHz (realfreq=%d)\n", __func__, rate, realfreq); return 0; } @@ -499,7 +499,7 @@ static int pcxhr_set_stream_state(struct snd_pcxhr *chip, else { if (stream->status != PCXHR_STREAM_STATUS_SCHEDULE_STOP) { dev_err(chip->card->dev, - "pcxhr_set_stream_state CANNOT be stopped\n"); + "%s CANNOT be stopped\n", __func__); return -EINVAL; } start = 0; @@ -524,7 +524,7 @@ static int pcxhr_set_stream_state(struct snd_pcxhr *chip, err = pcxhr_send_msg(chip->mgr, &rmh); if (err) dev_err(chip->card->dev, - "ERROR pcxhr_set_stream_state err=%x;\n", err); + "ERROR %s err=%x;\n", __func__, err); stream->status = start ? PCXHR_STREAM_STATUS_STARTED : PCXHR_STREAM_STATUS_STOPPED; return err; @@ -570,7 +570,7 @@ static int pcxhr_set_format(struct pcxhr_stream *stream) break; default: dev_err(chip->card->dev, - "error pcxhr_set_format() : unknown format\n"); + "error %s() : unknown format\n", __func__); return -EINVAL; } @@ -615,7 +615,7 @@ static int pcxhr_set_format(struct pcxhr_stream *stream) err = pcxhr_send_msg(chip->mgr, &rmh); if (err) dev_err(chip->card->dev, - "ERROR pcxhr_set_format err=%x;\n", err); + "ERROR %s err=%x;\n", __func__, err); return err; } @@ -630,7 +630,7 @@ static int pcxhr_update_r_buffer(struct pcxhr_stream *stream) stream_num = is_capture ? 0 : subs->number; dev_dbg(chip->card->dev, - "pcxhr_update_r_buffer(pcm%c%d) : addr(%p) bytes(%zx) subs(%d)\n", + "%s(pcm%c%d) : addr(%p) bytes(%zx) subs(%d)\n", __func__, is_capture ? 'c' : 'p', chip->chip_idx, (void *)(long)subs->runtime->dma_addr, subs->runtime->dma_bytes, subs->number); @@ -721,21 +721,20 @@ static void pcxhr_start_linked_stream(struct pcxhr_mgr *mgr) } if (capture_mask == 0 && playback_mask == 0) { mutex_unlock(&mgr->setup_mutex); - dev_err(&mgr->pci->dev, "pcxhr_start_linked_stream : no pipes\n"); + dev_err(&mgr->pci->dev, "%s : no pipes\n", __func__); return; } - dev_dbg(&mgr->pci->dev, "pcxhr_start_linked_stream : " - "playback_mask=%x capture_mask=%x\n", - playback_mask, capture_mask); + dev_dbg(&mgr->pci->dev, "%s : playback_mask=%x capture_mask=%x\n", + __func__, playback_mask, capture_mask); /* synchronous stop of all the pipes concerned */ err = pcxhr_set_pipe_state(mgr, playback_mask, capture_mask, 0); if (err) { mutex_unlock(&mgr->setup_mutex); - dev_err(&mgr->pci->dev, "pcxhr_start_linked_stream : " + dev_err(&mgr->pci->dev, "%s : " "error stop pipes (P%x C%x)\n", - playback_mask, capture_mask); + __func__, playback_mask, capture_mask); return; } @@ -778,9 +777,9 @@ static void pcxhr_start_linked_stream(struct pcxhr_mgr *mgr) err = pcxhr_set_pipe_state(mgr, playback_mask, capture_mask, 1); if (err) { mutex_unlock(&mgr->setup_mutex); - dev_err(&mgr->pci->dev, "pcxhr_start_linked_stream : " + dev_err(&mgr->pci->dev, "%s : " "error start pipes (P%x C%x)\n", - playback_mask, capture_mask); + __func__, playback_mask, capture_mask); return; } @@ -889,7 +888,7 @@ static int pcxhr_hardware_timer(struct pcxhr_mgr *mgr, int start) } err = pcxhr_send_msg(mgr, &rmh); if (err < 0) - dev_err(&mgr->pci->dev, "error pcxhr_hardware_timer err(%x)\n", + dev_err(&mgr->pci->dev, "error %s err(%x)\n", __func__, err); return err; } @@ -904,7 +903,7 @@ static int pcxhr_prepare(struct snd_pcm_substream *subs) int err = 0; dev_dbg(chip->card->dev, - "pcxhr_prepare : period_size(%lx) periods(%x) buffer_size(%lx)\n", + "%s : period_size(%lx) periods(%x) buffer_size(%lx)\n", __func__, subs->runtime->period_size, subs->runtime->periods, subs->runtime->buffer_size); @@ -997,12 +996,12 @@ static int pcxhr_open(struct snd_pcm_substream *subs) runtime->hw = pcxhr_caps; if( subs->stream == SNDRV_PCM_STREAM_PLAYBACK ) { - dev_dbg(chip->card->dev, "pcxhr_open playback chip%d subs%d\n", - chip->chip_idx, subs->number); + dev_dbg(chip->card->dev, "%s playback chip%d subs%d\n", + __func__, chip->chip_idx, subs->number); stream = &chip->playback_stream[subs->number]; } else { - dev_dbg(chip->card->dev, "pcxhr_open capture chip%d subs%d\n", - chip->chip_idx, subs->number); + dev_dbg(chip->card->dev, "%s capture chip%d subs%d\n", + __func__, chip->chip_idx, subs->number); if (mgr->mono_capture) runtime->hw.channels_max = 1; else @@ -1011,8 +1010,8 @@ static int pcxhr_open(struct snd_pcm_substream *subs) } if (stream->status != PCXHR_STREAM_STATUS_FREE){ /* streams in use */ - dev_err(chip->card->dev, "pcxhr_open chip%d subs%d in use\n", - chip->chip_idx, subs->number); + dev_err(chip->card->dev, "%s chip%d subs%d in use\n", + __func__, chip->chip_idx, subs->number); mutex_unlock(&mgr->setup_mutex); return -EBUSY; } @@ -1077,7 +1076,7 @@ static int pcxhr_close(struct snd_pcm_substream *subs) mutex_lock(&mgr->setup_mutex); - dev_dbg(chip->card->dev, "pcxhr_close chip%d subs%d\n", + dev_dbg(chip->card->dev, "%s chip%d subs%d\n", __func__, chip->chip_idx, subs->number); /* sample rate released */ @@ -1572,7 +1571,7 @@ static int pcxhr_probe(struct pci_dev *pci, /* init setup mutex*/ mutex_init(&mgr->setup_mutex); - mgr->prmh = kmalloc(sizeof(*mgr->prmh) + + mgr->prmh = kmalloc(sizeof(*mgr->prmh) + sizeof(u32) * (PCXHR_SIZE_MAX_LONG_STATUS - PCXHR_SIZE_MAX_STATUS), GFP_KERNEL); -- 2.32.0

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[PATCH] ALSA: core: Fix double calls of snd_card_free() via devres
by Takashi Iwai 31 Jul '21

31 Jul '21

At the transition to the devres-managed card release, we've put the check of double-free at trigger_card_release(). But this wasn't enough, as the code path calls snd_card_free() again, and it would lead to the doubly snd_card_free() calls. Actually the v1 patch was correct to handle this, but I forgot that corner case and moved the check to the more obvious place as I thought it's clearer. But, as usual, devils live in details. This patch corrects the check of the double-free to the right place, with a bit more comments. Fixes: e8ad415b7a55 ("ALSA: core: Add managed card creation") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/init.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/core/init.c b/sound/core/init.c index e985185ebc91..ac335f5906c6 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -605,6 +605,15 @@ int snd_card_free(struct snd_card *card) DECLARE_COMPLETION_ONSTACK(released); int ret; + /* The call of snd_card_free() is allowed from various code paths; + * a manual call from the driver and the call via devres_free, and + * we need to avoid double-free. Moreover, the release via devres + * may call snd_card_free() twice due to its nature, we need to have + * the check here at the beginning. + */ + if (card->releasing) + return 0; + card->release_completion = &released; ret = snd_card_free_when_closed(card); if (ret) @@ -813,10 +822,7 @@ EXPORT_SYMBOL_GPL(snd_card_add_dev_attr); static void trigger_card_free(void *data) { - struct snd_card *card = data; - - if (!card->releasing) - snd_card_free(data); + snd_card_free(data); } /** -- 2.26.2

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[PATCH] ASoC: amd: Drop superfluous mmap callbacks
by Takashi Iwai 30 Jul '21

30 Jul '21

All ASoC AMD drivers just call the standard mmap handler, hence those are superfluous. Let's drop them. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/amd/acp-pcm-dma.c | 8 -------- sound/soc/amd/raven/acp3x-pcm-dma.c | 8 -------- sound/soc/amd/renoir/acp3x-pdm-dma.c | 8 -------- 3 files changed, 24 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 143155a840ac..8f202423e6ce 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -1035,13 +1035,6 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_soc_component *component, return bytes_to_frames(runtime, pos); } -static int acp_dma_mmap(struct snd_soc_component *component, - struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - return snd_pcm_lib_default_mmap(substream, vma); -} - static int acp_dma_prepare(struct snd_soc_component *component, struct snd_pcm_substream *substream) { @@ -1205,7 +1198,6 @@ static const struct snd_soc_component_driver acp_asoc_platform = { .hw_params = acp_dma_hw_params, .trigger = acp_dma_trigger, .pointer = acp_dma_pointer, - .mmap = acp_dma_mmap, .prepare = acp_dma_prepare, .pcm_construct = acp_dma_new, }; diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 8148b0d22e88..5f24546eeed0 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -320,13 +320,6 @@ static int acp3x_dma_new(struct snd_soc_component *component, return 0; } -static int acp3x_dma_mmap(struct snd_soc_component *component, - struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - return snd_pcm_lib_default_mmap(substream, vma); -} - static int acp3x_dma_close(struct snd_soc_component *component, struct snd_pcm_substream *substream) { @@ -370,7 +363,6 @@ static const struct snd_soc_component_driver acp3x_i2s_component = { .close = acp3x_dma_close, .hw_params = acp3x_dma_hw_params, .pointer = acp3x_dma_pointer, - .mmap = acp3x_dma_mmap, .pcm_construct = acp3x_dma_new, }; diff --git a/sound/soc/amd/renoir/acp3x-pdm-dma.c b/sound/soc/amd/renoir/acp3x-pdm-dma.c index bd20622b0933..9988a50a81b0 100644 --- a/sound/soc/amd/renoir/acp3x-pdm-dma.c +++ b/sound/soc/amd/renoir/acp3x-pdm-dma.c @@ -291,13 +291,6 @@ static int acp_pdm_dma_new(struct snd_soc_component *component, return 0; } -static int acp_pdm_dma_mmap(struct snd_soc_component *component, - struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - return snd_pcm_lib_default_mmap(substream, vma); -} - static int acp_pdm_dma_close(struct snd_soc_component *component, struct snd_pcm_substream *substream) { @@ -375,7 +368,6 @@ static const struct snd_soc_component_driver acp_pdm_component = { .close = acp_pdm_dma_close, .hw_params = acp_pdm_dma_hw_params, .pointer = acp_pdm_dma_pointer, - .mmap = acp_pdm_dma_mmap, .pcm_construct = acp_pdm_dma_new, }; -- 2.26.2

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[PATCH] ASoC: nau8824: Fix open coded prefix handling
by Mark Brown 30 Jul '21

30 Jul '21

As with the component layer code the nau8824 driver had been doing some open coded pin manipulation which will have been broken now the core is fixed to handle this properly, remove the open coding to avoid the issue. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/nau8824.c | 42 ++++++-------------------------------- 1 file changed, 6 insertions(+), 36 deletions(-) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 15bd8335f667..906cdd2fb54a 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -828,36 +828,6 @@ static void nau8824_int_status_clear_all(struct regmap *regmap) } } -static void nau8824_dapm_disable_pin(struct nau8824 *nau8824, const char *pin) -{ - struct snd_soc_dapm_context *dapm = nau8824->dapm; - const char *prefix = dapm->component->name_prefix; - char prefixed_pin[80]; - - if (prefix) { - snprintf(prefixed_pin, sizeof(prefixed_pin), "%s %s", - prefix, pin); - snd_soc_dapm_disable_pin(dapm, prefixed_pin); - } else { - snd_soc_dapm_disable_pin(dapm, pin); - } -} - -static void nau8824_dapm_enable_pin(struct nau8824 *nau8824, const char *pin) -{ - struct snd_soc_dapm_context *dapm = nau8824->dapm; - const char *prefix = dapm->component->name_prefix; - char prefixed_pin[80]; - - if (prefix) { - snprintf(prefixed_pin, sizeof(prefixed_pin), "%s %s", - prefix, pin); - snd_soc_dapm_force_enable_pin(dapm, prefixed_pin); - } else { - snd_soc_dapm_force_enable_pin(dapm, pin); - } -} - static void nau8824_eject_jack(struct nau8824 *nau8824) { struct snd_soc_dapm_context *dapm = nau8824->dapm; @@ -866,8 +836,8 @@ static void nau8824_eject_jack(struct nau8824 *nau8824) /* Clear all interruption status */ nau8824_int_status_clear_all(regmap); - nau8824_dapm_disable_pin(nau8824, "SAR"); - nau8824_dapm_disable_pin(nau8824, "MICBIAS"); + snd_soc_dapm_disable_pin(dapm, "SAR"); + snd_soc_dapm_disable_pin(dapm, "MICBIAS"); snd_soc_dapm_sync(dapm); /* Enable the insertion interruption, disable the ejection @@ -897,8 +867,8 @@ static void nau8824_jdet_work(struct work_struct *work) struct regmap *regmap = nau8824->regmap; int adc_value, event = 0, event_mask = 0; - nau8824_dapm_enable_pin(nau8824, "MICBIAS"); - nau8824_dapm_enable_pin(nau8824, "SAR"); + snd_soc_dapm_enable_pin(dapm. "MICBIAS"); + snd_soc_dapm_enable_pin(dapm, "SAR"); snd_soc_dapm_sync(dapm); msleep(100); @@ -909,8 +879,8 @@ static void nau8824_jdet_work(struct work_struct *work) if (adc_value < HEADSET_SARADC_THD) { event |= SND_JACK_HEADPHONE; - nau8824_dapm_disable_pin(nau8824, "SAR"); - nau8824_dapm_disable_pin(nau8824, "MICBIAS"); + snd_soc_dapm_disable_pin(dapm, "SAR"); + snd_soc_dapm_disable_pin(dapm, "MICBIAS"); snd_soc_dapm_sync(dapm); } else { event |= SND_JACK_HEADSET; -- 2.20.1

1 1

[PATCH] ALSA: pcm - fix mmap capability check for the snd-dummy driver
by Jaroslav Kysela 30 Jul '21

30 Jul '21

The snd-dummy driver (fake_buffer configuration) uses the ops->page callback for the mmap operations. Allow mmap for this case, too. Cc: <stable(a)vger.kernel.org> Fixes: c4824ae7db41 ("ALSA: pcm: Fix mmap capability check") Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/core/pcm_native.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 6a2971a7e6a1..09c0e2a6489c 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -246,7 +246,7 @@ static bool hw_support_mmap(struct snd_pcm_substream *substream) if (!(substream->runtime->hw.info & SNDRV_PCM_INFO_MMAP)) return false; - if (substream->ops->mmap) + if (substream->ops->mmap || substream->ops->page) return true; switch (substream->dma_buffer.dev.type) { -- 2.31.1

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