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

[PATCH] ASoC: Intel: sof_rt5682: add 512FS MCLK clock configuration
by Mac Chiang 19 Jan '22

19 Jan '22

codec system clock source support 512FS MCLK synchronous directly, so no need to set PLL configuration when MCLK 24.576MHz. Suggested-by: Shuming Fan <shumingf(a)realtek.com> Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- sound/soc/intel/boards/sof_rt5682.c | 17 +++++++++++------ 1 file changed, 11 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index bd6d2e7dea53..a6efffc14cad 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -61,7 +61,6 @@ #define SOF_MAX98390_SPEAKER_AMP_PRESENT BIT(24) #define SOF_MAX98390_TWEETER_SPEAKER_PRESENT BIT(25) - /* Default: MCLK on, MCLK 19.2M, SSP0 */ static unsigned long sof_rt5682_quirk = SOF_RT5682_MCLK_EN | SOF_RT5682_SSP_CODEC(0); @@ -362,6 +361,9 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream, if (sof_rt5682_quirk & SOF_RT5682S_HEADPHONE_CODEC_PRESENT) { pll_id = RT5682S_PLL2; clk_id = RT5682S_SCLK_S_PLL2; + + if (pll_in == 24576000) + clk_id = RT5682S_SCLK_S_MCLK; } else { pll_id = RT5682_PLL1; clk_id = RT5682_SCLK_S_PLL1; @@ -369,11 +371,14 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream, pll_out = params_rate(params) * 512; - /* Configure pll for codec */ - ret = snd_soc_dai_set_pll(codec_dai, pll_id, pll_source, pll_in, - pll_out); - if (ret < 0) - dev_err(rtd->dev, "snd_soc_dai_set_pll err = %d\n", ret); + /* when MCLK is 512FS, no need to set PLL configuration additionally. */ + if (pll_in != 24576000) { + /* Configure pll for codec */ + ret = snd_soc_dai_set_pll(codec_dai, pll_id, pll_source, pll_in, + pll_out); + if (ret < 0) + dev_err(rtd->dev, "snd_soc_dai_set_pll err = %d\n", ret); + } /* Configure sysclk for codec */ ret = snd_soc_dai_set_sysclk(codec_dai, clk_id, -- 2.20.1

4 3

[PATCH] ASoC: soc-pcm: use GFP_ATOMIC in dpcm_create_debugfs_state()
by Christophe JAILLET 19 Jan '22

19 Jan '22

The commit below states that dpcm_be_connect() may be called from atomic context. It changes a GFP_KERNEL into a GFP_ATOMIC to deal with it. Another memory allocation is done in dpcm_create_debugfs_state() which is called by dpcm_be_connect(). Also use GFP_ATOMIC there to be consistent and be compliant with atomic context. Fixes: d8a9c6e1f676 ("ASoC: soc-pcm: use GFP_ATOMIC for dpcm structure") Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- Not clear to me how dpcm_be_connect() can be called from an atomic context, though. But better safe than sorry. --- 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 7abfc48b26ca..1a536a2b9dc3 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -212,7 +212,7 @@ static void dpcm_create_debugfs_state(struct snd_soc_dpcm *dpcm, int stream) { char *name; - name = kasprintf(GFP_KERNEL, "%s:%s", dpcm->be->dai_link->name, + name = kasprintf(GFP_ATOMIC, "%s:%s", dpcm->be->dai_link->name, stream ? "capture" : "playback"); if (name) { dpcm->debugfs_state = debugfs_create_dir( -- 2.32.0

4 4

[PATCH v2 0/1] ALSA: hda/realtek: fix speakers and micmute on HP 855
by Alexander Sergeyev 19 Jan '22

19 Jan '22

Rebased and re-tested on the latest sound.git tree. Alexander Sergeyev (1): ALSA: hda/realtek: fix speakers and micmute on HP 855 G8 sound/pci/hda/patch_realtek.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) -- 2.34.1

3 7

[PATCH, RFC] Enable S/PDIF output on Intel Hades Canyon NUC
by Dag B 18 Jan '22

18 Jan '22

Hi. The current snd-hda-intel/snd_hda_codec_realtek combo in the Linux kernel does not appear to enable the S/PDIF (toslink, iec958) port on Intel's Hades Canyon NUC. None of the current models/quirks changes this. To my knowledge. (The physical port in question is a minijack/mini toslink combo port.) I have found a number of people having the same issue with S/PDIF under Linux, on this particular hardware. None of these people reporting being able to find a solution, apart from a warm reboot into Linux from Windows. Until ... https://www.insanelymac.com/?app=core&module=system&controller=content&do=f… So. cyber4o claims the BIOS is wrong, and came up with the patch= solution first, then a modification to sound/pci/hda/patch_realtek.c to avoid having to boot to windows first. (IIUIC). I have been able to employ this solution and get the optical output enabled. I have (until now) used model=dell-headset-multi to enable my headphones with (on-wire) microphone in the front jack. (Kudos to arch-linux docs) The suggested fix for the missing optical out didn't cover headset config, so some modifications were needed. I'll freely admit to have very limited knowledge of what I am doing. Monkey see, monkey do. Anyways: I took cyber4o's solution and chained in the dell-headset-multi fixup. I now have optical output, my headset and on-wire microphone works. My current patch is below, and is purely intended as an illustration of how to make this work. I would be very happy if someone with an actual clue took the patch below and beat it into shape acceptable for inclusion in the kernel. Reading the forum post above may be wise, in case I have missed something. Some things which still make me scratch my head: - There is a slider named 'Digital' in the 'Capture' tab of alsamixer. Not sure what it does. - 'Headset Mic Boost', 'Headphone Mic Boost', 'Internal Mic Boost', 'Internal Mic Boost 1' all appear in *both* the 'Playback' *and* 'Capture' tabs of alsamixer. Is this on purpose? Bug in alsamixer? - There is no 'Headset' output slider in the 'Playback' tab. I do have 'Headphone' and 'Line out', though. And Mic boost for both headset and headphone. Not a problem, just ... 'asymmetrical'? - Marketing materials name the jack in the rear as 'Stereo headphones/TOSLINK combo rear jack'. Can we pretend it doesn't support a microphone (no idea...), and just name it (the copper bits) Line-Out? Less confusion? Again, really no clue about where these strings come from or what the 'correct' action is. - Inserting the jack in the front does not automatically mute the (optical) output in the back. Independent of the Auto-Mute state. Unsure if this is a corner case not covered by current code or not. Auto-mute only offers Enabled/Disabled here. (And not 'Disabled', 'Speaker Only' and 'Line-Out+Speaker'. I propose to mute S/PDIF together with Line Out.) /lib/firmware/nuc.patch: (loaded by patch= parameter to snd-hda-intel) [codec] 0x10ec0700 0x80862073 0 [pincfg] 0x1e 0x03441060 --- patch_realtek.c.orig 2021-12-10 21:07:29.324951606 -0000 +++ patch_realtek.c 2021-12-14 10:23:18.452173524 -0000 @@ -4669,6 +4669,12 @@ } } +static void alc700_fixup_nuc_spdif(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + snd_hda_override_wcaps(codec, 0x6, 0x611); +} + /* Line2 = mic mute hotkey * GPIO2 = mic mute LED */ @@ -6667,6 +6673,7 @@ ALC225_FIXUP_DELL_WYSE_MIC_NO_PRESENCE, ALC225_FIXUP_S3_POP_NOISE, ALC700_FIXUP_INTEL_REFERENCE, + ALC700_FIXUP_NUC_SPDIF, ALC274_FIXUP_DELL_BIND_DACS, ALC274_FIXUP_DELL_AIO_LINEOUT_VERB, ALC298_FIXUP_TPT470_DOCK_FIX, @@ -6975,6 +6982,12 @@ .chained = true, .chain_id = ALC269_FIXUP_THINKPAD_ACPI, }, + [ALC700_FIXUP_NUC_SPDIF] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc700_fixup_nuc_spdif, + .chained = true, + .chain_id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, + }, [ALC269_FIXUP_DELL1_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -8761,6 +8774,7 @@ SND_PCI_QUIRK(0x10cf, 0x1757, "Lifebook E752", ALC269_FIXUP_LIFEBOOK_HP_PIN), SND_PCI_QUIRK(0x10cf, 0x1845, "Lifebook U904", ALC269_FIXUP_LIFEBOOK_EXTMIC), SND_PCI_QUIRK(0x10ec, 0x10f2, "Intel Reference board", ALC700_FIXUP_INTEL_REFERENCE), + SND_PCI_QUIRK(0x10ec, 0x2073, "Intel NUC8 Hades Canyon", ALC700_FIXUP_NUC_SPDIF), SND_PCI_QUIRK(0x10ec, 0x118c, "Medion EE4254 MD62100", ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE), SND_PCI_QUIRK(0x10ec, 0x1230, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x10ec, 0x1252, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), @@ -9024,6 +9038,7 @@ {.id = ALC298_FIXUP_TPT470_DOCK, .name = "tpt470-dock"}, {.id = ALC233_FIXUP_LENOVO_MULTI_CODECS, .name = "dual-codecs"}, {.id = ALC700_FIXUP_INTEL_REFERENCE, .name = "alc700-ref"}, + {.id = ALC700_FIXUP_NUC_SPDIF, .name = "nuc-hc"}, {.id = ALC269_FIXUP_SONY_VAIO, .name = "vaio"}, {.id = ALC269_FIXUP_DELL_M101Z, .name = "dell-m101z"}, {.id = ALC269_FIXUP_ASUS_G73JW, .name = "asus-g73jw"},

1 0

Re: Behringer UMC202HD issues and a partial solution
by Geraldo Nascimento 18 Jan '22

18 Jan '22

On Sat, Jan 15, 2022 at 12:29:18PM +0530, Nandakumar Edamana wrote: > Could compile and reload the module successfully, but the stuttering is > still there. Hi Nandakumar, OK, my mistake, scratch that patch. The codepath that activates Implicit Feedback when implicit_fb=1 is passed to the module as an option is the same that should activate when the quirk is in place. Like I said, it's very strange that your device skips implicit feedback when the quirk is in place, but can pick implicit feedback when implicit_fb=1 is passed to the module as an option. I've prepared some printk's hooked up to every codepath I can think of that will skip implicit feedback. Please test and share your results, always with dyndbg on. --- implicit.c 2022-01-14 16:41:53.946606013 -0300 +++ implicit.c 2022-01-15 21:45:15.108655890 -0300 @@ -90,8 +90,10 @@ static int add_implicit_fb_sync_ep(struc if (!alts) { iface = usb_ifnum_to_if(chip->dev, ifnum); - if (!iface || iface->num_altsetting < 2) + if (!iface || iface->num_altsetting < 2) { + printk(KERN_WARNING "implicit: add_implicit_fb_sync_ep(): not a usb_host_interface or less than 2 altsettings"); return 0; + } alts = &iface->altsetting[1]; } @@ -242,17 +244,23 @@ static int __add_generic_implicit_fb(str struct usb_endpoint_descriptor *epd; alts = snd_usb_get_host_interface(chip, iface, altset); - if (!alts) + if (!alts) { + printk(KERN_WARNING "implicit: __add_generic_implicit_fb(): not a usb_host_interface"); return 0; + } if ((alts->desc.bInterfaceClass != USB_CLASS_VENDOR_SPEC && alts->desc.bInterfaceClass != USB_CLASS_AUDIO) || - alts->desc.bNumEndpoints < 1) + alts->desc.bNumEndpoints < 1) { + printk(KERN_WARNING "implicit: __add_generic_implicit_fb(): no endpoints or wrong Interface Class"); return 0; + } epd = get_endpoint(alts, 0); if (!usb_endpoint_is_isoc_in(epd) || - (epd->bmAttributes & USB_ENDPOINT_SYNCTYPE) != USB_ENDPOINT_SYNC_ASYNC) + (epd->bmAttributes & USB_ENDPOINT_SYNCTYPE) != USB_ENDPOINT_SYNC_ASYNC) { + printk(KERN_WARNING "implicit: __add_generic_implicit_fb(): not Isochronous IN or not Asynchronous"); return 0; + } return add_implicit_fb_sync_ep(chip, fmt, epd->bEndpointAddress, 0, iface, alts); } @@ -262,8 +270,10 @@ static int add_generic_implicit_fb(struc struct audioformat *fmt, struct usb_host_interface *alts) { - if ((fmt->ep_attr & USB_ENDPOINT_SYNCTYPE) != USB_ENDPOINT_SYNC_ASYNC) + if ((fmt->ep_attr & USB_ENDPOINT_SYNCTYPE) != USB_ENDPOINT_SYNC_ASYNC) { + printk(KERN_WARNING "implicit: add_generic_implicit_fb(): not Asynchronous"); return 0; + } if (__add_generic_implicit_fb(chip, fmt, alts->desc.bInterfaceNumber + 1, @@ -302,8 +312,10 @@ static int audioformat_implicit_fb_quirk if (p) { switch (p->type) { case IMPLICIT_FB_GENERIC: + printk(KERN_WARNING "implicit: audioformat_implicit_fb_quirk(): matched GENERIC inside playback_implicit_fb_quirks[]"); return add_generic_implicit_fb(chip, fmt, alts); case IMPLICIT_FB_NONE: + printk(KERN_WARNING "implicit: audioformat_implicit_fb_quirk(): matched SKIP_DEV inside playback_implicit_fb_quirks[]"); return 0; /* No quirk */ case IMPLICIT_FB_FIXED: return add_implicit_fb_sync_ep(chip, fmt, p->ep_num, 0, @@ -311,6 +323,10 @@ static int audioformat_implicit_fb_quirk } } + else { + printk(KERN_WARNING "implicit: audioformat_implicit_fb_quirk(): no matches found inside playback_implicit_fb_quirks[]"); + } + /* Special handling for devices with capture quirks */ p = find_implicit_fb_entry(chip, capture_implicit_fb_quirks, alts); if (p) { @@ -354,6 +370,7 @@ static int audioformat_implicit_fb_quirk return add_generic_implicit_fb(chip, fmt, alts); /* No quirk */ + printk(KERN_WARNING "implicit: audioformat_implicit_fb_quirk(): returning 0 because no quirk was found"); return 0; }

2 2

[PATCH] ASoC: SOF: compr: Add compress ops implementation
by Daniel Baluta 18 Jan '22

18 Jan '22

From: Daniel Baluta <daniel.baluta(a)nxp.com> Implement snd_compress_ops. There are a lot of similarities with PCM implementation. For now we use sof_ipc_pcm_params to transfer compress parameters to SOF firmware. This will be changed in the future once we either add new compress parameters to SOF or enhance existing sof_ipc_pcm_params structure to support all native compress params. Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/sof/compress.c | 270 ++++++++++++++++++++++++++++++++++++++- sound/soc/sof/sof-priv.h | 6 + 2 files changed, 271 insertions(+), 5 deletions(-) diff --git a/sound/soc/sof/compress.c b/sound/soc/sof/compress.c index 01ca85f0b87f..2db1517d906d 100644 --- a/sound/soc/sof/compress.c +++ b/sound/soc/sof/compress.c @@ -10,6 +10,22 @@ #include "sof-audio.h" #include "sof-priv.h" +static void sof_set_transferred_bytes(struct sof_compr_stream *sstream, + u64 host_pos, u64 buffer_size) +{ + u64 prev_pos; + unsigned int copied; + + div64_u64_rem(sstream->copied_total, buffer_size, &prev_pos); + + if (host_pos < prev_pos) + copied = (buffer_size - prev_pos) + host_pos; + else + copied = host_pos - prev_pos; + + sstream->copied_total += copied; +} + static void snd_sof_compr_fragment_elapsed_work(struct work_struct *work) { struct snd_sof_pcm_stream *sps = @@ -29,16 +45,16 @@ void snd_sof_compr_init_elapsed_work(struct work_struct *work) */ void snd_sof_compr_fragment_elapsed(struct snd_compr_stream *cstream) { - struct snd_soc_component *component; - struct snd_soc_pcm_runtime *rtd; + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_compr_runtime *runtime = cstream->runtime; + struct sof_compr_stream *sstream = runtime->private_data; + struct snd_soc_component *component = + snd_soc_rtdcom_lookup(rtd, SOF_AUDIO_PCM_DRV_NAME); struct snd_sof_pcm *spcm; if (!cstream) return; - rtd = cstream->private_data; - component = snd_soc_rtdcom_lookup(rtd, SOF_AUDIO_PCM_DRV_NAME); - spcm = snd_sof_find_spcm_dai(component, rtd); if (!spcm) { dev_err(component->dev, @@ -46,6 +62,250 @@ void snd_sof_compr_fragment_elapsed(struct snd_compr_stream *cstream) return; } + sof_set_transferred_bytes(sstream, spcm->stream[cstream->direction].posn.host_posn, + runtime->buffer_size); + /* use the same workqueue-based solution as for PCM, cf. snd_sof_pcm_elapsed */ schedule_work(&spcm->stream[cstream->direction].period_elapsed_work); } + +static int create_page_table(struct snd_soc_component *component, + struct snd_compr_stream *cstream, + unsigned char *dma_area, size_t size) +{ + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_dma_buffer *dmab = cstream->runtime->dma_buffer_p; + int dir = cstream->direction; + struct snd_sof_pcm *spcm; + + spcm = snd_sof_find_spcm_dai(component, rtd); + if (!spcm) + return -EINVAL; + + return snd_sof_create_page_table(component->dev, dmab, + spcm->stream[dir].page_table.area, size); +} + +int sof_compr_open(struct snd_soc_component *component, + struct snd_compr_stream *cstream) +{ + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_compr_runtime *runtime = cstream->runtime; + struct sof_compr_stream *sstream; + struct snd_sof_pcm *spcm; + int dir; + + sstream = kzalloc(sizeof(*sstream), GFP_KERNEL); + if (!sstream) + return -ENOMEM; + + spcm = snd_sof_find_spcm_dai(component, rtd); + if (!spcm) { + kfree(sstream); + return -EINVAL; + } + + dir = cstream->direction; + + if (spcm->stream[dir].cstream) { + kfree(sstream); + return -EBUSY; + } + + spcm->stream[dir].cstream = cstream; + spcm->stream[dir].posn.host_posn = 0; + spcm->stream[dir].posn.dai_posn = 0; + spcm->prepared[dir] = false; + + runtime->private_data = sstream; + + return 0; +} + +int sof_compr_free(struct snd_soc_component *component, + struct snd_compr_stream *cstream) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_compr_runtime *runtime = cstream->runtime; + struct sof_compr_stream *sstream = runtime->private_data; + struct sof_ipc_stream stream; + struct sof_ipc_reply reply; + struct snd_sof_pcm *spcm; + int ret = 0; + + spcm = snd_sof_find_spcm_dai(component, rtd); + if (!spcm) + return -EINVAL; + + stream.hdr.size = sizeof(stream); + stream.hdr.cmd = SOF_IPC_GLB_STREAM_MSG | SOF_IPC_STREAM_PCM_FREE; + stream.comp_id = spcm->stream[cstream->direction].comp_id; + + if (spcm->prepared[cstream->direction]) { + ret = sof_ipc_tx_message(sdev->ipc, stream.hdr.cmd, + &stream, sizeof(stream), + &reply, sizeof(reply)); + if (!ret) + spcm->prepared[cstream->direction] = false; + } + + cancel_work_sync(&spcm->stream[cstream->direction].period_elapsed_work); + spcm->stream[cstream->direction].cstream = NULL; + kfree(sstream); + + return ret; +} + +int sof_compr_set_params(struct snd_soc_component *component, + struct snd_compr_stream *cstream, struct snd_compr_params *params) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); + struct snd_soc_pcm_runtime *rtd_pcm = cstream->private_data; + struct snd_compr_runtime *rtd = cstream->runtime; + struct sof_compr_stream *sstream = rtd->private_data; + struct sof_ipc_pcm_params_reply ipc_params_reply; + struct sof_ipc_pcm_params pcm; + struct snd_sof_pcm *spcm; + int ret; + + spcm = snd_sof_find_spcm_dai(component, rtd_pcm); + if (!spcm) + return -EINVAL; + + cstream->dma_buffer.dev.type = SNDRV_DMA_TYPE_DEV_SG; + cstream->dma_buffer.dev.dev = sdev->dev; + ret = snd_compr_malloc_pages(cstream, rtd->buffer_size); + if (ret < 0) + return ret; + + create_page_table(component, cstream, rtd->dma_area, rtd->dma_bytes); + + memset(&pcm, 0, sizeof(pcm)); + + pcm.params.buffer.pages = PFN_UP(rtd->dma_bytes); + pcm.hdr.size = sizeof(pcm); + pcm.hdr.cmd = SOF_IPC_GLB_STREAM_MSG | SOF_IPC_STREAM_PCM_PARAMS; + + pcm.comp_id = spcm->stream[cstream->direction].comp_id; + pcm.params.hdr.size = sizeof(pcm.params); + pcm.params.buffer.phy_addr = spcm->stream[cstream->direction].page_table.addr; + pcm.params.buffer.size = rtd->dma_bytes; + pcm.params.direction = cstream->direction; + pcm.params.channels = params->codec.ch_out; + pcm.params.rate = params->codec.sample_rate; + pcm.params.buffer_fmt = SOF_IPC_BUFFER_INTERLEAVED; + pcm.params.frame_fmt = SOF_IPC_FRAME_S32_LE; + pcm.params.sample_container_bytes = + snd_pcm_format_physical_width(SNDRV_PCM_FORMAT_S32) >> 3; + pcm.params.host_period_bytes = params->buffer.fragment_size; + + ret = sof_ipc_tx_message(sdev->ipc, pcm.hdr.cmd, &pcm, sizeof(pcm), + &ipc_params_reply, sizeof(ipc_params_reply)); + if (ret < 0) { + dev_err(component->dev, "error ipc failed\n"); + return ret; + } + + sstream->posn_offset = sdev->stream_box.offset + ipc_params_reply.posn_offset; + sstream->sample_rate = params->codec.sample_rate; + spcm->prepared[cstream->direction] = true; + + return 0; +} + +int sof_compr_get_params(struct snd_soc_component *component, + struct snd_compr_stream *cstream, struct snd_codec *params) +{ + return 0; +} + +int sof_compr_trigger(struct snd_soc_component *component, + struct snd_compr_stream *cstream, int cmd) +{ + struct sof_ipc_stream stream; + struct sof_ipc_reply reply; + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); + struct snd_sof_pcm *spcm; + + spcm = snd_sof_find_spcm_dai(component, rtd); + if (!spcm) + return -EINVAL; + + stream.hdr.size = sizeof(stream); + stream.hdr.cmd = SOF_IPC_GLB_STREAM_MSG; + stream.comp_id = spcm->stream[cstream->direction].comp_id; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_START; + break; + case SNDRV_PCM_TRIGGER_STOP: + stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_STOP; + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_PAUSE; + break; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_RELEASE; + break; + default: + dev_err(component->dev, "error: unhandled trigger cmd %d\n", cmd); + break; + } + + return sof_ipc_tx_message(sdev->ipc, stream.hdr.cmd, + &stream, sizeof(stream), + &reply, sizeof(reply)); +} + +int sof_compr_copy(struct snd_soc_component *component, + struct snd_compr_stream *cstream, + char __user *buf, size_t count) +{ + struct snd_compr_runtime *rtd = cstream->runtime; + unsigned int offset, n; + void *ptr; + int ret; + + if (count > rtd->buffer_size) + count = rtd->buffer_size; + + div_u64_rem(rtd->total_bytes_available, rtd->buffer_size, &offset); + ptr = rtd->dma_area + offset; + n = rtd->buffer_size - offset; + + if (count < n) { + ret = copy_from_user(ptr, buf, count); + } else { + ret = copy_from_user(ptr, buf, n); + ret += copy_from_user(rtd->dma_area, buf + n, count - n); + } + + return count - ret; +} + +static int sof_compr_pointer(struct snd_soc_component *component, + struct snd_compr_stream *cstream, + struct snd_compr_tstamp *tstamp) +{ + struct snd_compr_runtime *runtime = cstream->runtime; + struct sof_compr_stream *sstream = runtime->private_data; + + tstamp->sampling_rate = sstream->sample_rate; + tstamp->copied_total = sstream->copied_total; + + return 0; +} + +struct snd_compress_ops sof_compressed_ops = { + .open = sof_compr_open, + .free = sof_compr_free, + .set_params = sof_compr_set_params, + .get_params = sof_compr_get_params, + .trigger = sof_compr_trigger, + .pointer = sof_compr_pointer, + .copy = sof_compr_copy, +}; +EXPORT_SYMBOL(sof_compressed_ops); diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 087935192ce8..d001a762a866 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -108,6 +108,12 @@ enum sof_debugfs_access_type { SOF_DEBUGFS_ACCESS_D0_ONLY, }; +struct sof_compr_stream { + unsigned int copied_total; + unsigned int sample_rate; + size_t posn_offset; +}; + struct snd_sof_dev; struct snd_sof_ipc_msg; struct snd_sof_ipc; -- 2.27.0

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[PATCH] ASoC: amd: fix unmet dependency on GPIOLIB for SND_SOC_DMIC
by Julian Braha 18 Jan '22

18 Jan '22

When SND_SOC_AMD_YC_MACH is selected, and GPIOLIB is not selected, Kbuild gives the following warning: WARNING: unmet direct dependencies detected for SND_SOC_DMIC Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n] Selected by [y]: - SND_SOC_AMD_YC_MACH [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_AMD_ACP6x [=y] This is because SND_SOC_AMD_YC_MACH selects SND_SOC_DMIC, without selecting or depending on GPIOLIB, despite SND_SOC_DMIC depending on GPIOLIB. This unmet dependency bug was detected by Kismet, a static analysis tool for Kconfig. Please advise if this is not the appropriate solution. Signed-off-by: Julian Braha <julianbraha(a)gmail.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 7a9e45094f37..4aa067c5f054 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -88,7 +88,7 @@ config SND_SOC_AMD_ACP6x config SND_SOC_AMD_YC_MACH tristate "AMD YC support for DMIC" select SND_SOC_DMIC - depends on SND_SOC_AMD_ACP6x + depends on SND_SOC_AMD_ACP6x && GPIOLIB help This option enables machine driver for Yellow Carp platform using dmic. ACP IP has PDM Decoder block with DMA controller. -- 2.32.0

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[PATCH] ASoC: au1x: fix unmet dependency on SND_SOC_I2C_AND_SPI for SND_SOC_WM8731
by Julian Braha 18 Jan '22

18 Jan '22

When SND_SOC_DB1200 is selected, and SND_SOC_I2C_AND_SPI is not selected, Kbuild gives the following warning: WARNING: unmet direct dependencies detected for SND_SOC_WM8731 Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_I2C_AND_SPI [=n] Selected by [y]: - SND_SOC_DB1200 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_AU1XPSC [=y] This is because SND_SOC_DB1200 selects SND_SOC_WM8731 without selecting or depending on SND_SOC_I2C_AND_SPI, despite SND_SOC_WM8731 depending on SND_SOC_I2C_AND_SPI. This unmet dependency bug was detected by Kismet, a static analysis tool for Kconfig. Please advise if this is not the appropriate solution. Signed-off-by: Julian Braha <julianbraha(a)gmail.com> --- sound/soc/au1x/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/au1x/Kconfig b/sound/soc/au1x/Kconfig index 38de7c0efbc7..3dccdfddbf9d 100644 --- a/sound/soc/au1x/Kconfig +++ b/sound/soc/au1x/Kconfig @@ -54,6 +54,7 @@ config SND_SOC_DB1000 config SND_SOC_DB1200 tristate "DB1200/DB1300/DB1550 Audio support" depends on SND_SOC_AU1XPSC + depends on SND_SOC_I2C_AND_SPI select SND_SOC_AU1XPSC_AC97 select SND_SOC_AC97_CODEC select SND_SOC_WM9712 -- 2.32.0

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[PATCH] ALSA: hda: cs35l41: Make cs35l41_hda_remove() return void
by Uwe Kleine-König 18 Jan '22

18 Jan '22

Up to now cs35l41_hda_remove() returns zero unconditionally. Make it return void instead which makes it easier to see in the callers that there is no error to handle. Also the return value of i2c and spi remove callbacks is ignored anyway. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/pci/hda/cs35l41_hda.c | 4 +--- sound/pci/hda/cs35l41_hda.h | 2 +- sound/pci/hda/cs35l41_hda_i2c.c | 4 +++- sound/pci/hda/cs35l41_hda_spi.c | 4 +++- 4 files changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 30b40d865863..ce3782826830 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -508,7 +508,7 @@ int cs35l41_hda_probe(struct device *dev, const char *device_name, int id, int i } EXPORT_SYMBOL_GPL(cs35l41_hda_probe); -int cs35l41_hda_remove(struct device *dev) +void cs35l41_hda_remove(struct device *dev) { struct cs35l41_hda *cs35l41 = dev_get_drvdata(dev); @@ -517,8 +517,6 @@ int cs35l41_hda_remove(struct device *dev) if (!cs35l41->vspk_always_on) gpiod_set_value_cansleep(cs35l41->reset_gpio, 0); gpiod_put(cs35l41->reset_gpio); - - return 0; } EXPORT_SYMBOL_GPL(cs35l41_hda_remove); diff --git a/sound/pci/hda/cs35l41_hda.h b/sound/pci/hda/cs35l41_hda.h index 76c69a8a22f6..8ecaddf5f132 100644 --- a/sound/pci/hda/cs35l41_hda.h +++ b/sound/pci/hda/cs35l41_hda.h @@ -64,6 +64,6 @@ struct cs35l41_hda { int cs35l41_hda_probe(struct device *dev, const char *device_name, int id, int irq, struct regmap *regmap); -int cs35l41_hda_remove(struct device *dev); +void cs35l41_hda_remove(struct device *dev); #endif /*__CS35L41_HDA_H__*/ diff --git a/sound/pci/hda/cs35l41_hda_i2c.c b/sound/pci/hda/cs35l41_hda_i2c.c index 4a9462fb5c14..d4240b8ded10 100644 --- a/sound/pci/hda/cs35l41_hda_i2c.c +++ b/sound/pci/hda/cs35l41_hda_i2c.c @@ -32,7 +32,9 @@ static int cs35l41_hda_i2c_probe(struct i2c_client *clt, const struct i2c_device static int cs35l41_hda_i2c_remove(struct i2c_client *clt) { - return cs35l41_hda_remove(&clt->dev); + cs35l41_hda_remove(&clt->dev); + + return 0; } static const struct i2c_device_id cs35l41_hda_i2c_id[] = { diff --git a/sound/pci/hda/cs35l41_hda_spi.c b/sound/pci/hda/cs35l41_hda_spi.c index 77426e96c58f..d63c487bc3a9 100644 --- a/sound/pci/hda/cs35l41_hda_spi.c +++ b/sound/pci/hda/cs35l41_hda_spi.c @@ -30,7 +30,9 @@ static int cs35l41_hda_spi_probe(struct spi_device *spi) static int cs35l41_hda_spi_remove(struct spi_device *spi) { - return cs35l41_hda_remove(&spi->dev); + cs35l41_hda_remove(&spi->dev); + + return 0; } static const struct spi_device_id cs35l41_hda_spi_id[] = { base-commit: 0c947b893d69231a9add855939da7c66237ab44f -- 2.34.1

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[PATCH v2 1/6] ALSA: hda: cs35l41: Avoid overwriting register patch
by Lucas Tanure 18 Jan '22

18 Jan '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> regmap_register_patch can't be used to apply the probe sequence as a patch is already registers with the regmap by cs35l41_register_errata_patch and only a single patch can be attached to a single regmap. The driver doesn't currently rely on a cache sync to re-apply this probe sequence so simply switch it to a multi write. Fixes: 7b2f3eb492da ("ALSA: hda: cs35l41: Add support for CS35L41 in HDA systems") Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> --- V2: Add Fixes tag --- sound/pci/hda/cs35l41_hda.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 30b40d865863..c47c5f0b4e59 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -480,7 +480,7 @@ int cs35l41_hda_probe(struct device *dev, const char *device_name, int id, int i acpi_hw_cfg = NULL; if (cs35l41->reg_seq->probe) { - ret = regmap_register_patch(cs35l41->regmap, cs35l41->reg_seq->probe, + ret = regmap_multi_reg_write(cs35l41->regmap, cs35l41->reg_seq->probe, cs35l41->reg_seq->num_probe); if (ret) { dev_err(cs35l41->dev, "Fail to apply probe reg patch: %d\n", ret); -- 2.34.1

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