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[PATCH] ALSA: usb-audio: Apply async workaround for Scarlett 2i4 2nd gen
by Alexander Tsoy 27 Apr '20

27 Apr '20

Due to rounding error driver sometimes incorrectly calculate next packet size, which results in audible clicks on devices with synchronous playback endpoints. For example on a high speed bus and a sample rate 44.1 kHz it loses one sample every ~40.9 seconds. Fortunately playback interface on Scarlett 2i4 2nd gen has a working explicit feedback endpoint, so we can switch playback data endpoint to asynchronous mode as a workaround. Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 6c2dfd3bfcbf..351ba214a9d3 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1806,6 +1806,7 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip, */ fp->attributes &= ~UAC_EP_CS_ATTR_FILL_MAX; break; + case USB_ID(0x1235, 0x8200): /* Focusrite Scarlett 2i4 2nd gen */ case USB_ID(0x1235, 0x8202): /* Focusrite Scarlett 2i2 2nd gen */ case USB_ID(0x1235, 0x8205): /* Focusrite Scarlett Solo 2nd gen */ /* -- 2.25.3

3 19

[PATCH] ALSA: hda/realtek - Two front mics on a Lenovo ThinkCenter
by Hui Wang 27 Apr '20

27 Apr '20

This new Lenovo ThinkCenter has two front mics which can't be handled by PA so far, so apply the fixup ALC283_FIXUP_HEADSET_MIC to change the location for one of the mics. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c1a85c8f7b69..c16f63957c5a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7420,6 +7420,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x8560, "System76 Gazelle (gaze14)", ALC269_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1558, 0x8561, "System76 Gazelle (gaze14)", ALC269_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x1036, "Lenovo P520", ALC233_FIXUP_LENOVO_MULTI_CODECS), + SND_PCI_QUIRK(0x17aa, 0x1048, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), -- 2.17.1

2 1

[PATCH] ASoC: fsl_esai: Remove the tasklet
by Shengjiu Wang 27 Apr '20

27 Apr '20

Remove tasklet for it may cause the reset operation can't be handled immediately, then there will be endless xrun interrupt. Fixes: 7ccafa2b3879 ("ASoC: fsl_esai: recover the channel swap after xrun") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_esai.c | 15 ++++----------- 1 file changed, 4 insertions(+), 11 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index c7a49d03463a..1ad0859da5e2 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -32,7 +32,6 @@ * @extalclk: esai clock source to derive HCK, SCK and FS * @fsysclk: system clock source to derive HCK, SCK and FS * @spbaclk: SPBA clock (optional, depending on SoC design) - * @task: tasklet to handle the reset operation * @lock: spin lock between hw_reset() and trigger() * @fifo_depth: depth of tx/rx FIFO * @slot_width: width of each DAI slot @@ -56,7 +55,6 @@ struct fsl_esai { struct clk *extalclk; struct clk *fsysclk; struct clk *spbaclk; - struct tasklet_struct task; spinlock_t lock; /* Protect hw_reset and trigger */ u32 fifo_depth; u32 slot_width; @@ -74,6 +72,8 @@ struct fsl_esai { char name[32]; }; +static void fsl_esai_hw_reset(struct fsl_esai *esai_priv); + static irqreturn_t esai_isr(int irq, void *devid) { struct fsl_esai *esai_priv = (struct fsl_esai *)devid; @@ -87,7 +87,7 @@ static irqreturn_t esai_isr(int irq, void *devid) if ((saisr & (ESAI_SAISR_TUE | ESAI_SAISR_ROE)) && esai_priv->reset_at_xrun) { dev_dbg(&pdev->dev, "reset module for xrun\n"); - tasklet_schedule(&esai_priv->task); + fsl_esai_hw_reset(esai_priv); } if (esr & ESAI_ESR_TINIT_MASK) @@ -674,9 +674,8 @@ static void fsl_esai_trigger_stop(struct fsl_esai *esai_priv, bool tx) ESAI_xFCR_xFR, 0); } -static void fsl_esai_hw_reset(unsigned long arg) +static void fsl_esai_hw_reset(struct fsl_esai *esai_priv) { - struct fsl_esai *esai_priv = (struct fsl_esai *)arg; bool tx = true, rx = false, enabled[2]; unsigned long lock_flags; u32 tfcr, rfcr; @@ -1034,9 +1033,6 @@ static int fsl_esai_probe(struct platform_device *pdev) return ret; } - tasklet_init(&esai_priv->task, fsl_esai_hw_reset, - (unsigned long)esai_priv); - pm_runtime_enable(&pdev->dev); regcache_cache_only(esai_priv->regmap, true); @@ -1050,10 +1046,7 @@ static int fsl_esai_probe(struct platform_device *pdev) static int fsl_esai_remove(struct platform_device *pdev) { - struct fsl_esai *esai_priv = platform_get_drvdata(pdev); - pm_runtime_disable(&pdev->dev); - tasklet_kill(&esai_priv->task); return 0; } -- 2.21.0

3 2

FW: Does the SpeexDSP plugin work?
by Stuart Naylor 26 Apr '20

26 Apr '20

I have been playing with this repo and the results are actually quite good. https://github.com/voice-engine/ec I have been using exactly the same parameters with the following. I have sent this to you guys because I am sure something is very much wrong with the plugin itself. SpeexDSP works OK, the above repo works ok, alsa-plugin speex works exceptionally badly that in terms of audio quality it doesn’t work at all? # The IPC key of dmix or dsnoop plugin must be unique # If 555555 or 666666 is used by other processes, use another one # use samplerate to resample as speexdsp resample is bad #defaults.pcm.rate_converter "samplerate" pcm.!default { type asym playback.pcm "playback" capture.pcm "echo" } pcm.playback { type plug slave.pcm "dmixed" } pcm.echo { type speex slave.pcm "agc" echo yes frames 128 filter_length 4096 } pcm.agc { type speex slave.pcm "capture" agc 1 } pcm.capture { type plug slave.pcm "array" route_policy sum } pcm.dmixed { type dmix slave.pcm "hw:seeed2micvoicec" ipc_key 555555 } pcm.array { type dsnoop slave { pcm "hw:seeed2micvoicec" channels 2 } ipc_key 666666 } I really don’t understand how the above repo can do such and excellent job and then with alsa-plugins it can provide zero EC and also vocode the recording so badly? Surely there is something wrong with the implementation? Its doesn’t make and sense unless I have the above asound.conf wrong? But https://github.com/voice-engine/ec works quite effectively? Stuart Sent from Mail<https://go.microsoft.com/fwlink/?LinkId=550986> for Windows 10

1 0

Re: [PATCH] sound:hdmi:fix without unlocked before return
by Markus Elfring 26 Apr '20

26 Apr '20

> After add sanity check to pass klockwork check, > The spdif_mutex should be unlock before return true > in check_non_pcm_per_cvt(). How do you think about a wording variant like the following? Subject: [PATCH v2] ALSA: hda/hdmi: Unlock a mutex always before returning from check_non_pcm_per_cvt() Change description (according to a solution alternative): The exception handling was incomplete in an if branch for a null pointer check of the variable “spdif”. Thus assign an appropriate value to the local variable “non_pcm” by using a conditional operator instead. A mutex is also appropriately unlocked then. … > +++ b/sound/pci/hda/patch_hdmi.c > @@ -1848,8 +1848,10 @@ static bool check_non_pcm_per_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) … > - if (WARN_ON(spdif == NULL)) > + if (WARN_ON(spdif == NULL)) { > + mutex_unlock(&codec->spdif_mutex); > return true; > + } > non_pcm = !!(spdif->status & IEC958_AES0_NONAUDIO); - if (WARN_ON(spdif == NULL)) - return true; - non_pcm = !!(spdif->status & IEC958_AES0_NONAUDIO); + non_pcm = WARN_ON(!spdif) + ? true + : !!(spdif->status & IEC958_AES0_NONAUDIO); mutex_unlock(&codec->spdif_mutex); return non_pcm; } Would you like to add the tag “Fixes” to the change description? Regards, Markus

1 0

MOTU M2 stopped working after upgrade to 5.5
by Michael Gebetsroither 26 Apr '20

26 Apr '20

Hi, I'm using a MOTU M2 as an audio interface for my headset and after upgrading to kernel 5.5 (from 5.4) the interface stopped working in full duplex mode, means it does not work anymore for me with a headset at all. (The headset is a sennheiser hmdc27) If i go back to kernel 5.4 everything works (also without any cracks in the audio output). It's a bit unfortunate that a working audio interface is suddenly disabled by software. Is there any way this can be made working again? I suspect c249177944b650816069f6c49b769baaa94339dc to be the reason for this "ALSA: usb-audio: add implicit fb quirk for MOTU M Series" With kernel 5.5 i also get numerous warnings in kernel log like: | Apr 25 20:23:16 localhost kernel: [ 2631.701391] usb 5-2.3.2: Unable to change format on ep #81: already in use | Apr 25 20:23:16 localhost kernel: [ 2631.701694] usb 5-2.3.2: Unable to change format on ep #81: already in use | Apr 25 20:23:16 localhost kernel: [ 2631.701951] usb 5-2.3.2: Unable to change format on ep #81: already in use | Apr 25 20:23:16 localhost kernel: [ 2631.705521] usb 5-2.3.2: Unable to change format on ep #81: already in use | ... For audio output i've used pulseaudio with the following settings in daemon.conf: | resample-method = src-sinc-best-quality | default-sample-format = s24le | default-sample-rate = 96000 | alternate-sample-rate = 48000 And for jack to not produce cracks i've used: | sample rate: 96khz | Frames/periode: 256 | Periodes/buffer: 25 greets, gebi

2 2

[PATCH] ALSA: line6: Fix POD HD500 audio playback
by Vasily Khoruzhick 26 Apr '20

26 Apr '20

Apparently interface 1 is control interface akin to HD500X, setting LINE6_CAP_CONTROL and choosing it as ctrl_if fixes audio playback on POD HD500. Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> --- sound/usb/line6/podhd.c | 22 +++++----------------- 1 file changed, 5 insertions(+), 17 deletions(-) diff --git a/sound/usb/line6/podhd.c b/sound/usb/line6/podhd.c index d37db32ecd3b..e39dc85c355a 100644 --- a/sound/usb/line6/podhd.c +++ b/sound/usb/line6/podhd.c @@ -21,8 +21,7 @@ enum { LINE6_PODHD300, LINE6_PODHD400, - LINE6_PODHD500_0, - LINE6_PODHD500_1, + LINE6_PODHD500, LINE6_PODX3, LINE6_PODX3LIVE, LINE6_PODHD500X, @@ -318,8 +317,7 @@ static const struct usb_device_id podhd_id_table[] = { /* TODO: no need to alloc data interfaces when only audio is used */ { LINE6_DEVICE(0x5057), .driver_info = LINE6_PODHD300 }, { LINE6_DEVICE(0x5058), .driver_info = LINE6_PODHD400 }, - { LINE6_IF_NUM(0x414D, 0), .driver_info = LINE6_PODHD500_0 }, - { LINE6_IF_NUM(0x414D, 1), .driver_info = LINE6_PODHD500_1 }, + { LINE6_IF_NUM(0x414D, 0), .driver_info = LINE6_PODHD500 }, { LINE6_IF_NUM(0x414A, 0), .driver_info = LINE6_PODX3 }, { LINE6_IF_NUM(0x414B, 0), .driver_info = LINE6_PODX3LIVE }, { LINE6_IF_NUM(0x4159, 0), .driver_info = LINE6_PODHD500X }, @@ -352,23 +350,13 @@ static const struct line6_properties podhd_properties_table[] = { .ep_audio_r = 0x82, .ep_audio_w = 0x01, }, - [LINE6_PODHD500_0] = { + [LINE6_PODHD500] = { .id = "PODHD500", .name = "POD HD500", - .capabilities = LINE6_CAP_PCM + .capabilities = LINE6_CAP_PCM | LINE6_CAP_CONTROL | LINE6_CAP_HWMON, .altsetting = 1, - .ep_ctrl_r = 0x81, - .ep_ctrl_w = 0x01, - .ep_audio_r = 0x86, - .ep_audio_w = 0x02, - }, - [LINE6_PODHD500_1] = { - .id = "PODHD500", - .name = "POD HD500", - .capabilities = LINE6_CAP_PCM - | LINE6_CAP_HWMON, - .altsetting = 0, + .ctrl_if = 1, .ep_ctrl_r = 0x81, .ep_ctrl_w = 0x01, .ep_audio_r = 0x86, -- 2.25.0

2 1

[PATCH v2] soundwire: intel: Remove unused function
by Sudip Mukherjee 26 Apr '20

26 Apr '20

The function sdw_intel_init() is not used anywhere, remove it for now. And that makes sdw_intel_add_controller(), sdw_intel_acpi_cb() and link_mask unused, so remove them as well. Signed-off-by: Sudip Mukherjee <sudipm.mukherjee(a)gmail.com> --- drivers/soundwire/intel_init.c | 162 ----------------------------------------- 1 file changed, 162 deletions(-) diff --git a/drivers/soundwire/intel_init.c b/drivers/soundwire/intel_init.c index ad7053463889..d90929a5043b 100644 --- a/drivers/soundwire/intel_init.c +++ b/drivers/soundwire/intel_init.c @@ -23,10 +23,6 @@ #define SDW_LINK_BASE 0x30000 #define SDW_LINK_SIZE 0x10000 -static int link_mask; -module_param_named(sdw_link_mask, link_mask, int, 0444); -MODULE_PARM_DESC(sdw_link_mask, "Intel link mask (one bit per link)"); - static int sdw_intel_cleanup_pdev(struct sdw_intel_ctx *ctx) { struct sdw_intel_link_res *link = ctx->links; @@ -47,164 +43,6 @@ static int sdw_intel_cleanup_pdev(struct sdw_intel_ctx *ctx) return 0; } -static struct sdw_intel_ctx -*sdw_intel_add_controller(struct sdw_intel_res *res) -{ - struct platform_device_info pdevinfo; - struct platform_device *pdev; - struct sdw_intel_link_res *link; - struct sdw_intel_ctx *ctx; - struct acpi_device *adev; - int ret, i; - u8 count; - u32 caps; - - if (acpi_bus_get_device(res->handle, &adev)) - return NULL; - - /* Found controller, find links supported */ - count = 0; - ret = fwnode_property_read_u8_array(acpi_fwnode_handle(adev), - "mipi-sdw-master-count", &count, 1); - - /* Don't fail on error, continue and use hw value */ - if (ret) { - dev_err(&adev->dev, - "Failed to read mipi-sdw-master-count: %d\n", ret); - count = SDW_MAX_LINKS; - } - - /* Check SNDWLCAP.LCOUNT */ - caps = ioread32(res->mmio_base + SDW_SHIM_BASE + SDW_SHIM_LCAP); - caps &= GENMASK(2, 0); - - /* Check HW supported vs property value and use min of two */ - count = min_t(u8, caps, count); - - /* Check count is within bounds */ - if (count > SDW_MAX_LINKS) { - dev_err(&adev->dev, "Link count %d exceeds max %d\n", - count, SDW_MAX_LINKS); - return NULL; - } else if (!count) { - dev_warn(&adev->dev, "No SoundWire links detected\n"); - return NULL; - } - - dev_dbg(&adev->dev, "Creating %d SDW Link devices\n", count); - - ctx = kzalloc(sizeof(*ctx), GFP_KERNEL); - if (!ctx) - return NULL; - - ctx->count = count; - ctx->links = kcalloc(ctx->count, sizeof(*ctx->links), GFP_KERNEL); - if (!ctx->links) - goto link_err; - - link = ctx->links; - - /* Create SDW Master devices */ - for (i = 0; i < count; i++) { - if (link_mask && !(link_mask & BIT(i))) { - dev_dbg(&adev->dev, - "Link %d masked, will not be enabled\n", i); - link++; - continue; - } - - link->registers = res->mmio_base + SDW_LINK_BASE - + (SDW_LINK_SIZE * i); - link->shim = res->mmio_base + SDW_SHIM_BASE; - link->alh = res->mmio_base + SDW_ALH_BASE; - - link->ops = res->ops; - link->dev = res->dev; - - memset(&pdevinfo, 0, sizeof(pdevinfo)); - - pdevinfo.parent = res->parent; - pdevinfo.name = "int-sdw"; - pdevinfo.id = i; - pdevinfo.fwnode = acpi_fwnode_handle(adev); - - pdev = platform_device_register_full(&pdevinfo); - if (IS_ERR(pdev)) { - dev_err(&adev->dev, - "platform device creation failed: %ld\n", - PTR_ERR(pdev)); - goto pdev_err; - } - - link->pdev = pdev; - link++; - } - - return ctx; - -pdev_err: - sdw_intel_cleanup_pdev(ctx); -link_err: - kfree(ctx); - return NULL; -} - -static acpi_status sdw_intel_acpi_cb(acpi_handle handle, u32 level, - void *cdata, void **return_value) -{ - struct sdw_intel_res *res = cdata; - struct acpi_device *adev; - acpi_status status; - u64 adr; - - status = acpi_evaluate_integer(handle, METHOD_NAME__ADR, NULL, &adr); - if (ACPI_FAILURE(status)) - return AE_OK; /* keep going */ - - if (acpi_bus_get_device(handle, &adev)) { - pr_err("%s: Couldn't find ACPI handle\n", __func__); - return AE_NOT_FOUND; - } - - res->handle = handle; - - /* - * On some Intel platforms, multiple children of the HDAS - * device can be found, but only one of them is the SoundWire - * controller. The SNDW device is always exposed with - * Name(_ADR, 0x40000000), with bits 31..28 representing the - * SoundWire link so filter accordingly - */ - if ((adr & GENMASK(31, 28)) >> 28 != SDW_LINK_TYPE) - return AE_OK; /* keep going */ - - /* device found, stop namespace walk */ - return AE_CTRL_TERMINATE; -} - -/** - * sdw_intel_init() - SoundWire Intel init routine - * @parent_handle: ACPI parent handle - * @res: resource data - * - * This scans the namespace and creates SoundWire link controller devices - * based on the info queried. - */ -static void *sdw_intel_init(acpi_handle *parent_handle, - struct sdw_intel_res *res) -{ - acpi_status status; - - status = acpi_walk_namespace(ACPI_TYPE_DEVICE, - parent_handle, 1, - sdw_intel_acpi_cb, - NULL, res, NULL); - if (ACPI_FAILURE(status)) - return NULL; - - return sdw_intel_add_controller(res); -} - /** * sdw_intel_exit() - SoundWire Intel exit * @arg: callback context -- 2.11.0

2 1

[PATCH 1/2] ALSA: usb-audio: Improve frames size computation
by Alexander Tsoy 25 Apr '20

25 Apr '20

For computation of the the next frame size current value of fs/fps and accumulated fractional parts of fs/fps are used, where values are stored in Q16.16 format. This is quite natural for computing frame size for asynchronous endpoints driven by explicit feedback, since in this case fs/fps is a value provided by the feedback endpoint and it's already in the Q format. If an error is accumulated over time, the device can adjust fs/fps value to prevent buffer overruns/underruns. But for synchronous endpoints the accuracy provided by these computations is not enough. Due to accumulated error the driver periodically produces frames with incorrect size (+/- 1 audio sample). This patch fixes this issue by implementing a different algorithm for frame size computation. It is based on accumulating of the remainders from division fs/fps and it doesn't accumulate errors over time. This new method is enabled for synchronous and adaptive playback endpoints. Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> --- sound/usb/card.h | 4 ++++ sound/usb/endpoint.c | 43 ++++++++++++++++++++++++++++++++++++++----- sound/usb/endpoint.h | 1 + sound/usb/pcm.c | 2 ++ 4 files changed, 45 insertions(+), 5 deletions(-) diff --git a/sound/usb/card.h b/sound/usb/card.h index 395403a2d33f..820e564656ed 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -84,6 +84,10 @@ struct snd_usb_endpoint { dma_addr_t sync_dma; /* DMA address of syncbuf */ unsigned int pipe; /* the data i/o pipe */ + unsigned int framesize[2]; /* small/large frame sizes in samples */ + unsigned int sample_rem; /* remainder from division fs/fps */ + unsigned int sample_accum; /* sample accumulator */ + unsigned int fps; /* frames per second */ unsigned int freqn; /* nominal sampling rate in fs/fps in Q16.16 format */ unsigned int freqm; /* momentary sampling rate in fs/fps in Q16.16 format */ int freqshift; /* how much to shift the feedback value to get Q16.16 */ diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 4a9a2f6ef5a4..d8dc7cb56d43 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -124,12 +124,12 @@ int snd_usb_endpoint_implicit_feedback_sink(struct snd_usb_endpoint *ep) /* * For streaming based on information derived from sync endpoints, - * prepare_outbound_urb_sizes() will call next_packet_size() to + * prepare_outbound_urb_sizes() will call slave_next_packet_size() to * determine the number of samples to be sent in the next packet. * - * For implicit feedback, next_packet_size() is unused. + * For implicit feedback, slave_next_packet_size() is unused. */ -int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep) +int snd_usb_endpoint_slave_next_packet_size(struct snd_usb_endpoint *ep) { unsigned long flags; int ret; @@ -146,6 +146,29 @@ int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep) return ret; } +/* + * For adaptive and synchronous endpoints, prepare_outbound_urb_sizes() + * will call next_packet_size() to determine the number of samples to be + * sent in the next packet. + */ +int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep) +{ + int ret; + + if (ep->fill_max) + return ep->maxframesize; + + ep->sample_accum += ep->sample_rem; + if (ep->sample_accum >= ep->fps) { + ep->sample_accum -= ep->fps; + ret = ep->framesize[1]; + } else { + ret = ep->framesize[0]; + } + + return ret; +} + static void retire_outbound_urb(struct snd_usb_endpoint *ep, struct snd_urb_ctx *urb_ctx) { @@ -190,6 +213,8 @@ static void prepare_silent_urb(struct snd_usb_endpoint *ep, if (ctx->packet_size[i]) counts = ctx->packet_size[i]; + else if (ep->sync_master) + counts = snd_usb_endpoint_slave_next_packet_size(ep); else counts = snd_usb_endpoint_next_packet_size(ep); @@ -874,10 +899,17 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, ep->maxpacksize = fmt->maxpacksize; ep->fill_max = !!(fmt->attributes & UAC_EP_CS_ATTR_FILL_MAX); - if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_FULL) + if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_FULL) { ep->freqn = get_usb_full_speed_rate(rate); - else + ep->fps = 1000; + } else { ep->freqn = get_usb_high_speed_rate(rate); + ep->fps = 8000; + } + + ep->sample_rem = rate % ep->fps; + ep->framesize[0] = rate / ep->fps; + ep->framesize[1] = (rate + (ep->fps - 1)) / ep->fps; /* calculate the frequency in 16.16 format */ ep->freqm = ep->freqn; @@ -936,6 +968,7 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) ep->active_mask = 0; ep->unlink_mask = 0; ep->phase = 0; + ep->sample_accum = 0; snd_usb_endpoint_start_quirk(ep); diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index 63a39d4fa8d8..d23fa0a8c11b 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -28,6 +28,7 @@ void snd_usb_endpoint_release(struct snd_usb_endpoint *ep); void snd_usb_endpoint_free(struct snd_usb_endpoint *ep); int snd_usb_endpoint_implicit_feedback_sink(struct snd_usb_endpoint *ep); +int snd_usb_endpoint_slave_next_packet_size(struct snd_usb_endpoint *ep); int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep); void snd_usb_handle_sync_urb(struct snd_usb_endpoint *ep, diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index a4e4064f9aee..b50965ab3b3a 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1579,6 +1579,8 @@ static void prepare_playback_urb(struct snd_usb_substream *subs, for (i = 0; i < ctx->packets; i++) { if (ctx->packet_size[i]) counts = ctx->packet_size[i]; + else if (ep->sync_master) + counts = snd_usb_endpoint_slave_next_packet_size(ep); else counts = snd_usb_endpoint_next_packet_size(ep); -- 2.26.2

4 11

[PATCH 00/11] ASoC: QCOM: Add support for SC7180 lpass variant
by Ajit Pandey 25 Apr '20

25 Apr '20

We've common ASoC cpu driver for QCOM LPASS soc varaints. This patch chain add support for new variant SC7180 soc by doing the required modification in existing common lpass-cpu driver. Below is a brief summary of patch series: PATCH 01 ... 04: Update lpass-cpu driver files to make it more generic PATCH 05 ... 08: Add changes to support SC7180 specific configuration PATCH 09 ... 11: Add new drivers and documentation for SC7180 soc lpass Ajit Pandey (11): Documentation: device-tree: sound: Update lpass-cpu driver binding ASoC: qcom: lpass: Add struct lpass_dai to store dai clocks pointer ASoC: qcom: Add common array to initialize soc based core clocks ASoC: qcom: lpass-cpu: Make "ahbix-clk" an optional clock. ASoC: qcom: lpass: Add support for newer lpass version dt-bindings: sound: Add bindings related to lpass-cpu configuration Documentation: dt-bindings: sound: Add details for new dai properties ASoC: qcom : lpass: Add support to configure dai's connection mode device-tree: bindings: sound: lpass-cpu: Add new compatible soc ASoC: qcom: lpass-sc7180: Add platform driver for lpass audio ASoC: qcom: lpass-platform: Replace card->dev with component->dev .../devicetree/bindings/sound/qcom,lpass-cpu.txt | 158 ------------- .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 159 +++++++++++++ include/dt-bindings/sound/qcom,lpass.h | 31 +++ sound/soc/qcom/Kconfig | 5 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/lpass-apq8016.c | 39 ++- sound/soc/qcom/lpass-cpu.c | 263 ++++++++++++++------- sound/soc/qcom/lpass-lpaif-reg.h | 182 +++++++++++++- sound/soc/qcom/lpass-platform.c | 36 +-- sound/soc/qcom/lpass-sc7180.c | 192 +++++++++++++++ sound/soc/qcom/lpass.h | 40 +++- 11 files changed, 808 insertions(+), 299 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml create mode 100644 include/dt-bindings/sound/qcom,lpass.h create mode 100644 sound/soc/qcom/lpass-sc7180.c -- 1.9.1

2 20

[Help] Are there tools to change Audio Mixer other than HDA Analyzer?
by andythe_great 24 Apr '20

24 Apr '20

Hello, I could not find appropriate mailing list for this question. I think HDA Analyzer is retired. https://www.alsa-project.org/main/index.php/HDA_Analyzer So I am finding other tool to change Audio Mixer like how HDA Analyzer do it. I attached a screenshot to show what Audio Mixer is in the HDA Analyzer which I try to figure out where it is in hdajackretask. Thanks Andy

1 0

[PATCH] soundwire: intel: Remove unused function
by Sudip Mukherjee 24 Apr '20

24 Apr '20

The function sdw_intel_init() is not used anywhere, remove it for now. Signed-off-by: Sudip Mukherjee <sudipm.mukherjee(a)gmail.com> --- drivers/soundwire/intel_init.c | 23 ----------------------- 1 file changed, 23 deletions(-) diff --git a/drivers/soundwire/intel_init.c b/drivers/soundwire/intel_init.c index ad7053463889..2f37dad06321 100644 --- a/drivers/soundwire/intel_init.c +++ b/drivers/soundwire/intel_init.c @@ -183,29 +183,6 @@ static acpi_status sdw_intel_acpi_cb(acpi_handle handle, u32 level, } /** - * sdw_intel_init() - SoundWire Intel init routine - * @parent_handle: ACPI parent handle - * @res: resource data - * - * This scans the namespace and creates SoundWire link controller devices - * based on the info queried. - */ -static void *sdw_intel_init(acpi_handle *parent_handle, - struct sdw_intel_res *res) -{ - acpi_status status; - - status = acpi_walk_namespace(ACPI_TYPE_DEVICE, - parent_handle, 1, - sdw_intel_acpi_cb, - NULL, res, NULL); - if (ACPI_FAILURE(status)) - return NULL; - - return sdw_intel_add_controller(res); -} - -/** * sdw_intel_exit() - SoundWire Intel exit * @arg: callback context * -- 2.11.0

2 1

[PATCH] ALSA: pcm: oss: Place the plugin buffer overflow checks correctly (for 5.7)
by Takashi Iwai 24 Apr '20

24 Apr '20

[ This is again a forward-port of the fix applied for 5.6-base code (commit 4285de0725b1) to 5.7-base, hence neither Fixes nor Cc-to-stable tags are included here -- tiwai ] The checks of the plugin buffer overflow in the previous fix by commit f2ecf903ef06 ("ALSA: pcm: oss: Avoid plugin buffer overflow") are put in the wrong places mistakenly, which leads to the expected (repeated) sound when the rate plugin is involved. Fix in the right places. Also, at those right places, the zero check is needed for the termination node, so added there as well, and let's get it done, finally. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/oss/pcm_plugin.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 59d62f05658f..1545f8fdb4db 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -205,13 +205,14 @@ static snd_pcm_sframes_t calc_dst_frames(struct snd_pcm_substream *plug, plugin = snd_pcm_plug_first(plug); while (plugin && frames > 0) { plugin_next = plugin->next; + if (check_size && plugin->buf_frames && + frames > plugin->buf_frames) + frames = plugin->buf_frames; if (plugin->dst_frames) { frames = plugin->dst_frames(plugin, frames); if (frames < 0) return frames; } - if (check_size && frames > plugin->buf_frames) - frames = plugin->buf_frames; plugin = plugin_next; } return frames; @@ -225,14 +226,15 @@ static snd_pcm_sframes_t calc_src_frames(struct snd_pcm_substream *plug, plugin = snd_pcm_plug_last(plug); while (plugin && frames > 0) { - if (check_size && frames > plugin->buf_frames) - frames = plugin->buf_frames; plugin_prev = plugin->prev; if (plugin->src_frames) { frames = plugin->src_frames(plugin, frames); if (frames < 0) return frames; } + if (check_size && plugin->buf_frames && + frames > plugin->buf_frames) + frames = plugin->buf_frames; plugin = plugin_prev; } return frames; -- 2.16.4

1 0

[PATCH] ALSA: pcm: oss: Place the plugin buffer overflow checks correctly
by Takashi Iwai 24 Apr '20

24 Apr '20

The checks of the plugin buffer overflow in the previous fix by commit f2ecf903ef06 ("ALSA: pcm: oss: Avoid plugin buffer overflow") are put in the wrong places mistakenly, which leads to the expected (repeated) sound when the rate plugin is involved. Fix in the right places. Also, at those right places, the zero check is needed for the termination node, so added there as well, and let's get it done, finally. Fixes: f2ecf903ef06 ("ALSA: pcm: oss: Avoid plugin buffer overflow") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/oss/pcm_plugin.c | 20 ++++++++++++-------- 1 file changed, 12 insertions(+), 8 deletions(-) diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 3ecc070738e8..c1315ce98b54 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -211,21 +211,23 @@ static snd_pcm_sframes_t plug_client_size(struct snd_pcm_substream *plug, if (stream == SNDRV_PCM_STREAM_PLAYBACK) { plugin = snd_pcm_plug_last(plug); while (plugin && drv_frames > 0) { - if (check_size && drv_frames > plugin->buf_frames) - drv_frames = plugin->buf_frames; plugin_prev = plugin->prev; if (plugin->src_frames) drv_frames = plugin->src_frames(plugin, drv_frames); + if (check_size && plugin->buf_frames && + drv_frames > plugin->buf_frames) + drv_frames = plugin->buf_frames; plugin = plugin_prev; } } else if (stream == SNDRV_PCM_STREAM_CAPTURE) { plugin = snd_pcm_plug_first(plug); while (plugin && drv_frames > 0) { plugin_next = plugin->next; + if (check_size && plugin->buf_frames && + drv_frames > plugin->buf_frames) + drv_frames = plugin->buf_frames; if (plugin->dst_frames) drv_frames = plugin->dst_frames(plugin, drv_frames); - if (check_size && drv_frames > plugin->buf_frames) - drv_frames = plugin->buf_frames; plugin = plugin_next; } } else @@ -251,26 +253,28 @@ static snd_pcm_sframes_t plug_slave_size(struct snd_pcm_substream *plug, plugin = snd_pcm_plug_first(plug); while (plugin && frames > 0) { plugin_next = plugin->next; + if (check_size && plugin->buf_frames && + frames > plugin->buf_frames) + frames = plugin->buf_frames; if (plugin->dst_frames) { frames = plugin->dst_frames(plugin, frames); if (frames < 0) return frames; } - if (check_size && frames > plugin->buf_frames) - frames = plugin->buf_frames; plugin = plugin_next; } } else if (stream == SNDRV_PCM_STREAM_CAPTURE) { plugin = snd_pcm_plug_last(plug); while (plugin) { - if (check_size && frames > plugin->buf_frames) - frames = plugin->buf_frames; plugin_prev = plugin->prev; if (plugin->src_frames) { frames = plugin->src_frames(plugin, frames); if (frames < 0) return frames; } + if (check_size && plugin->buf_frames && + frames > plugin->buf_frames) + frames = plugin->buf_frames; plugin = plugin_prev; } } else -- 2.16.4

1 0

ASoC driver names
by Jaroslav Kysela 24 Apr '20

24 Apr '20

Hi all, I am fighting actually with the ASoC driver names. The current situation, where each hardware variant (card name) is mapped to the driver name is not very practical (and not correct). The driver name should describe the common part (usually the controller - like 'HDA-Intel' or bus 'USB-Audio'). I am talking about the situation, where the ASoC card name is set via the device tree (snd_soc_of_parse_card_name) like in [1], but the change may be considered for other drivers like Intel SST: Card driver name: Lenovo-YOGA-C63 Card name: Lenovo-YOGA-C630-13Q50 Card long name: LENOVO-81JL-LenovoYOGAC630_13Q50-LNVNB161216 [1] https://github.com/alsa-project/alsa-tests/pull/3/commits/1b5a552cff06c6039… My question is, can we change / add the more apropriate driver names conditionally with a new kernel CONFIG option? Will you accept this change? The goal is to group the related UCM2 configurations and do the required split inside the UCM2 top-level configuration file based on card components string or other card identificators (related to the driver). Thanks, Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

4 11

[PATCH] ASoC: snd-sof-intel-hda-common - add hda_model parameter and pass it to HDA codec driver
by Jaroslav Kysela 24 Apr '20

24 Apr '20

It may be useful to pass the specific model to the generic HDA codec routines like the legacy HDA driver (snd-hda-intel) allows. The model name "sofbus" is tricky anyway. Original proposal: https://lore.kernel.org/alsa-devel/20191203161908.7496-1-perex@perex.cz/ Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/intel/hda.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 211e91e79eae..ea0189ee8939 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -282,6 +282,10 @@ module_param_named(use_msi, hda_use_msi, bool, 0444); MODULE_PARM_DESC(use_msi, "SOF HDA use PCI MSI mode"); #endif +static char *hda_model; +module_param(hda_model, charp, 0444); +MODULE_PARM_DESC(hda_model, "Use the given HDA board model."); + #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) static int hda_dmic_num = -1; module_param_named(dmic_num, hda_dmic_num, int, 0444); @@ -503,7 +507,7 @@ static int hda_init(struct snd_sof_dev *sdev) mutex_init(&hbus->prepare_mutex); hbus->pci = pci; hbus->mixer_assigned = -1; - hbus->modelname = "sofbus"; + hbus->modelname = hda_model; /* initialise hdac bus */ bus->addr = pci_resource_start(pci, 0); -- 2.25.3

5 6

Re: [PATCH] ASoC: fsl_easrc: Check for null pointer before dereferencing “ctx” in fsl_easrc_hw_free()
by Markus Elfring 24 Apr '20

24 Apr '20

> The patch 955ac624058f: "ASoC: fsl_easrc: Add EASRC ASoC CPU DAI > drivers" from Apr 16, 2020, leads to the following Smatch complaint: > > sound/soc/fsl/fsl_easrc.c:1529 fsl_easrc_hw_free() > warn: variable dereferenced before check 'ctx' (see line 1527) Please avoid a typo in the final commit subject. Regards, Markus

1 0

[PATCH v3 0/3] Add support for different DMIC configurations
by Mateusz Gorski 24 Apr '20

24 Apr '20

Set of patches to enable DMIC capture on different hardware configurations. Information about supported DMIC configuration is read from NHLT and correct pipeline configuration is selected automatically. Also, adding additional option for topology binary name which is based on used machine driver. Changes in v2: - removed patch 1/4 swapping machine device and platform device registration order - alt_tplg_name creation now uses different field to read machine driver name - including of <sound/soc-acpi.h> moved to different patch Changes in v3: - cosmetic changes in skl-topology.c file Mateusz Gorski (3): ASoC: Intel: Skylake: Add alternative topology binary name ASoC: Intel: Multiple I/O PCM format support for pipe ASoC: Intel: Skylake: Automatic DMIC format configuration according to information from NHLT include/uapi/sound/skl-tplg-interface.h | 2 + sound/soc/intel/skylake/skl-topology.c | 188 +++++++++++++++++++++++- sound/soc/intel/skylake/skl-topology.h | 1 + 3 files changed, 186 insertions(+), 5 deletions(-) -- 2.17.1

3 7

[PATCH AUTOSEL 4.4 1/8] ALSA: hda: Don't release card at firmware loading error
by Sasha Levin 24 Apr '20

24 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 25faa4bd37c10f19e4b848b9032a17a3d44c6f09 ] At the error path of the firmware loading error, the driver tries to release the card object and set NULL to drvdata. This may be referred badly at the possible PM action, as the driver itself is still bound and the PM callbacks read the card object. Instead, we continue the probing as if it were no option set. This is often a better choice than the forced abort, too. Fixes: 5cb543dba986 ("ALSA: hda - Deferred probing with request_firmware_nowait()") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043 Link: https://lore.kernel.org/r/20200413082034.25166-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_intel.c | 19 +++++-------------- 1 file changed, 5 insertions(+), 14 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 3e3277100f08a..0fa0c33660087 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1839,24 +1839,15 @@ static void azx_firmware_cb(const struct firmware *fw, void *context) { struct snd_card *card = context; struct azx *chip = card->private_data; - struct pci_dev *pci = chip->pci; - - if (!fw) { - dev_err(card->dev, "Cannot load firmware, aborting\n"); - goto error; - } - chip->fw = fw; + if (fw) + chip->fw = fw; + else + dev_err(card->dev, "Cannot load firmware, continue without patching\n"); if (!chip->disabled) { /* continue probing */ - if (azx_probe_continue(chip)) - goto error; + azx_probe_continue(chip); } - return; /* OK */ - - error: - snd_card_free(card); - pci_set_drvdata(pci, NULL); } #endif -- 2.20.1

1 1

[PATCH AUTOSEL 4.9 02/13] ALSA: hda: Don't release card at firmware loading error
by Sasha Levin 24 Apr '20

24 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 25faa4bd37c10f19e4b848b9032a17a3d44c6f09 ] At the error path of the firmware loading error, the driver tries to release the card object and set NULL to drvdata. This may be referred badly at the possible PM action, as the driver itself is still bound and the PM callbacks read the card object. Instead, we continue the probing as if it were no option set. This is often a better choice than the forced abort, too. Fixes: 5cb543dba986 ("ALSA: hda - Deferred probing with request_firmware_nowait()") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043 Link: https://lore.kernel.org/r/20200413082034.25166-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_intel.c | 19 +++++-------------- 1 file changed, 5 insertions(+), 14 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 3234e9ca02cec..5a578ebca1055 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1828,24 +1828,15 @@ static void azx_firmware_cb(const struct firmware *fw, void *context) { struct snd_card *card = context; struct azx *chip = card->private_data; - struct pci_dev *pci = chip->pci; - - if (!fw) { - dev_err(card->dev, "Cannot load firmware, aborting\n"); - goto error; - } - chip->fw = fw; + if (fw) + chip->fw = fw; + else + dev_err(card->dev, "Cannot load firmware, continue without patching\n"); if (!chip->disabled) { /* continue probing */ - if (azx_probe_continue(chip)) - goto error; + azx_probe_continue(chip); } - return; /* OK */ - - error: - snd_card_free(card); - pci_set_drvdata(pci, NULL); } #endif -- 2.20.1

1 0

[PATCH AUTOSEL 4.14 04/21] ALSA: hda: Keep the controller initialization even if no codecs found
by Sasha Levin 24 Apr '20

24 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 9479e75fca370a5220784f7596bf598c4dad0b9b ] Currently, when the HD-audio controller driver doesn't detect any codecs, it tries to abort the probe. But this abort happens at the delayed probe, i.e. the primary probe call already returned success, hence the driver is never unbound until user does so explicitly. As a result, it may leave the HD-audio device in the running state without the runtime PM. More badly, if the device is a HD-audio bus that is tied with a GPU, GPU cannot reach to the full power down and consumes unnecessarily much power. This patch changes the logic after no-codec situation; it continues probing without the further codec initialization but keep the controller driver running normally. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043 Tested-by: Roy Spliet <nouveau(a)spliet.org> Link: https://lore.kernel.org/r/20200413082034.25166-5-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_intel.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index a868a88bc2897..9a8f5164d88e5 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2012,7 +2012,7 @@ static int azx_first_init(struct azx *chip) /* codec detection */ if (!azx_bus(chip)->codec_mask) { dev_err(card->dev, "no codecs found!\n"); - return -ENODEV; + /* keep running the rest for the runtime PM */ } if (azx_acquire_irq(chip, 0) < 0) @@ -2340,9 +2340,11 @@ static int azx_probe_continue(struct azx *chip) #endif /* create codec instances */ - err = azx_probe_codecs(chip, azx_max_codecs[chip->driver_type]); - if (err < 0) - goto out_free; + if (bus->codec_mask) { + err = azx_probe_codecs(chip, azx_max_codecs[chip->driver_type]); + if (err < 0) + goto out_free; + } #ifdef CONFIG_SND_HDA_PATCH_LOADER if (chip->fw) { @@ -2356,7 +2358,7 @@ static int azx_probe_continue(struct azx *chip) #endif } #endif - if ((probe_only[dev] & 1) == 0) { + if (bus->codec_mask && !(probe_only[dev] & 1)) { err = azx_codec_configure(chip); if (err < 0) goto out_free; -- 2.20.1

1 0

[PATCH AUTOSEL 4.14 03/21] ALSA: hda: Don't release card at firmware loading error
by Sasha Levin 24 Apr '20

24 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 25faa4bd37c10f19e4b848b9032a17a3d44c6f09 ] At the error path of the firmware loading error, the driver tries to release the card object and set NULL to drvdata. This may be referred badly at the possible PM action, as the driver itself is still bound and the PM callbacks read the card object. Instead, we continue the probing as if it were no option set. This is often a better choice than the forced abort, too. Fixes: 5cb543dba986 ("ALSA: hda - Deferred probing with request_firmware_nowait()") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043 Link: https://lore.kernel.org/r/20200413082034.25166-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_intel.c | 19 +++++-------------- 1 file changed, 5 insertions(+), 14 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 890793ad85ca1..a868a88bc2897 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2034,24 +2034,15 @@ static void azx_firmware_cb(const struct firmware *fw, void *context) { struct snd_card *card = context; struct azx *chip = card->private_data; - struct pci_dev *pci = chip->pci; - - if (!fw) { - dev_err(card->dev, "Cannot load firmware, aborting\n"); - goto error; - } - chip->fw = fw; + if (fw) + chip->fw = fw; + else + dev_err(card->dev, "Cannot load firmware, continue without patching\n"); if (!chip->disabled) { /* continue probing */ - if (azx_probe_continue(chip)) - goto error; + azx_probe_continue(chip); } - return; /* OK */ - - error: - snd_card_free(card); - pci_set_drvdata(pci, NULL); } #endif -- 2.20.1

1 0

[PATCH AUTOSEL 4.19 12/18] ALSA: hda: call runtime_allow() for all hda controllers
by Sasha Levin 24 Apr '20

24 Apr '20

From: Hui Wang <hui.wang(a)canonical.com> [ Upstream commit 9a6418487b566503c772cb6e7d3d44e652b019b0 ] Before the pci_driver->probe() is called, the pci subsystem calls runtime_forbid() and runtime_get_sync() on this pci dev, so only call runtime_put_autosuspend() is not enough to enable the runtime_pm on this device. For controllers with vgaswitcheroo feature, the pci/quirks.c will call runtime_allow() for this dev, then the controllers could enter rt_idle/suspend/resume, but for non-vgaswitcheroo controllers like Intel hda controllers, the runtime_pm is not enabled because the runtime_allow() is not called. Since it is no harm calling runtime_allow() twice, here let hda driver call runtime_allow() for all controllers. Then the runtime_pm is enabled on all controllers after the put_autosuspend() is called. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Link: https://lore.kernel.org/r/20200414142725.6020-1-hui.wang@canonical.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b2b2dba6d5f01..84894141eac53 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2478,6 +2478,7 @@ static int azx_probe_continue(struct azx *chip) if (azx_has_pm_runtime(chip)) { pm_runtime_use_autosuspend(&pci->dev); + pm_runtime_allow(&pci->dev); pm_runtime_put_autosuspend(&pci->dev); } -- 2.20.1

1 0

[PATCH AUTOSEL 4.19 06/18] ALSA: hda: Explicitly permit using autosuspend if runtime PM is supported
by Sasha Levin 24 Apr '20

24 Apr '20

From: Roy Spliet <nouveau(a)spliet.org> [ Upstream commit 3ba21113bd33d49f3c300a23fc08cf114c434995 ] This fixes runtime PM not working after a suspend-to-RAM cycle at least for the codec-less HDA device found on NVIDIA GPUs. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043 Signed-off-by: Roy Spliet <nouveau(a)spliet.org> Link: https://lore.kernel.org/r/20200413082034.25166-7-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_intel.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b993b652ead81..b2b2dba6d5f01 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2476,8 +2476,10 @@ static int azx_probe_continue(struct azx *chip) set_default_power_save(chip); - if (azx_has_pm_runtime(chip)) + if (azx_has_pm_runtime(chip)) { + pm_runtime_use_autosuspend(&pci->dev); pm_runtime_put_autosuspend(&pci->dev); + } out_free: if ((chip->driver_caps & AZX_DCAPS_I915_POWERWELL) -- 2.20.1

1 0

[PATCH AUTOSEL 4.19 05/18] ALSA: hda: Keep the controller initialization even if no codecs found
by Sasha Levin 24 Apr '20

24 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 9479e75fca370a5220784f7596bf598c4dad0b9b ] Currently, when the HD-audio controller driver doesn't detect any codecs, it tries to abort the probe. But this abort happens at the delayed probe, i.e. the primary probe call already returned success, hence the driver is never unbound until user does so explicitly. As a result, it may leave the HD-audio device in the running state without the runtime PM. More badly, if the device is a HD-audio bus that is tied with a GPU, GPU cannot reach to the full power down and consumes unnecessarily much power. This patch changes the logic after no-codec situation; it continues probing without the further codec initialization but keep the controller driver running normally. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043 Tested-by: Roy Spliet <nouveau(a)spliet.org> Link: https://lore.kernel.org/r/20200413082034.25166-5-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_intel.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 54a9b391ecce0..b993b652ead81 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2054,7 +2054,7 @@ static int azx_first_init(struct azx *chip) /* codec detection */ if (!azx_bus(chip)->codec_mask) { dev_err(card->dev, "no codecs found!\n"); - return -ENODEV; + /* keep running the rest for the runtime PM */ } if (azx_acquire_irq(chip, 0) < 0) @@ -2441,9 +2441,11 @@ static int azx_probe_continue(struct azx *chip) #endif /* create codec instances */ - err = azx_probe_codecs(chip, azx_max_codecs[chip->driver_type]); - if (err < 0) - goto out_free; + if (bus->codec_mask) { + err = azx_probe_codecs(chip, azx_max_codecs[chip->driver_type]); + if (err < 0) + goto out_free; + } #ifdef CONFIG_SND_HDA_PATCH_LOADER if (chip->fw) { @@ -2457,7 +2459,7 @@ static int azx_probe_continue(struct azx *chip) #endif } #endif - if ((probe_only[dev] & 1) == 0) { + if (bus->codec_mask && !(probe_only[dev] & 1)) { err = azx_codec_configure(chip); if (err < 0) goto out_free; -- 2.20.1

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[PATCH AUTOSEL 5.4 17/26] ALSA: hda: call runtime_allow() for all hda controllers
by Sasha Levin 24 Apr '20

24 Apr '20

From: Hui Wang <hui.wang(a)canonical.com> [ Upstream commit 9a6418487b566503c772cb6e7d3d44e652b019b0 ] Before the pci_driver->probe() is called, the pci subsystem calls runtime_forbid() and runtime_get_sync() on this pci dev, so only call runtime_put_autosuspend() is not enough to enable the runtime_pm on this device. For controllers with vgaswitcheroo feature, the pci/quirks.c will call runtime_allow() for this dev, then the controllers could enter rt_idle/suspend/resume, but for non-vgaswitcheroo controllers like Intel hda controllers, the runtime_pm is not enabled because the runtime_allow() is not called. Since it is no harm calling runtime_allow() twice, here let hda driver call runtime_allow() for all controllers. Then the runtime_pm is enabled on all controllers after the put_autosuspend() is called. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Link: https://lore.kernel.org/r/20200414142725.6020-1-hui.wang@canonical.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 9f77713c36de6..1129ec1f038a7 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2322,6 +2322,7 @@ static int azx_probe_continue(struct azx *chip) if (azx_has_pm_runtime(chip)) { pm_runtime_use_autosuspend(&pci->dev); + pm_runtime_allow(&pci->dev); pm_runtime_put_autosuspend(&pci->dev); } -- 2.20.1

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[PATCH AUTOSEL 5.4 11/26] ALSA: hda: Explicitly permit using autosuspend if runtime PM is supported
by Sasha Levin 24 Apr '20

24 Apr '20

From: Roy Spliet <nouveau(a)spliet.org> [ Upstream commit 3ba21113bd33d49f3c300a23fc08cf114c434995 ] This fixes runtime PM not working after a suspend-to-RAM cycle at least for the codec-less HDA device found on NVIDIA GPUs. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043 Signed-off-by: Roy Spliet <nouveau(a)spliet.org> Link: https://lore.kernel.org/r/20200413082034.25166-7-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_intel.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 4604d1e16150d..9f77713c36de6 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2320,8 +2320,10 @@ static int azx_probe_continue(struct azx *chip) set_default_power_save(chip); - if (azx_has_pm_runtime(chip)) + if (azx_has_pm_runtime(chip)) { + pm_runtime_use_autosuspend(&pci->dev); pm_runtime_put_autosuspend(&pci->dev); + } out_free: if (err < 0) { -- 2.20.1

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[PATCH AUTOSEL 5.4 10/26] ALSA: hda: Skip controller resume if not needed
by Sasha Levin 24 Apr '20

24 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit c4c8dd6ef807663e42a5f04ea77cd62029eb99fa ] The HD-audio controller does system-suspend and resume operations by directly calling its helpers __azx_runtime_suspend() and __azx_runtime_resume(). However, in general, we don't have to resume always the device fully at the system resume; typically, if a device has been runtime-suspended, we can leave it to runtime resume. Usually for achieving this, the driver would call pm_runtime_force_suspend() and pm_runtime_force_resume() pairs in the system suspend and resume ops. Unfortunately, this doesn't work for the resume path in our case. For handling the jack detection at the system resume, a child codec device may need the (literally) forcibly resume even if it's been runtime-suspended, and for that, the controller device must be also resumed even if it's been suspended. This patch is an attempt to improve the situation. It replaces the direct __azx_runtime_suspend()/_resume() calls with with pm_runtime_force_suspend() and pm_runtime_force_resume() with a slight trick as we've done for the codec side. More exactly: - azx_has_pm_runtime() check is dropped from azx_runtime_suspend() and azx_runtime_resume(), so that it can be properly executed from the system-suspend/resume path - The WAKEEN handling depends on the card's power state now; it's set and cleared only for the runtime-suspend - azx_resume() checks whether any codec may need the forcible resume beforehand. If the forcible resume is required, it does temporary PM refcount up/down for actually triggering the runtime resume. - A new helper function, hda_codec_need_resume(), is introduced for checking whether the codec needs a forcible runtime-resume, and the existing code is rewritten with that. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043 Link: https://lore.kernel.org/r/20200413082034.25166-6-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/hda_codec.h | 5 +++++ sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/hda_intel.c | 38 +++++++++++++++++++++++++++----------- 3 files changed, 33 insertions(+), 12 deletions(-) diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h index 9a0393cf024c2..65c056ce91128 100644 --- a/include/sound/hda_codec.h +++ b/include/sound/hda_codec.h @@ -494,6 +494,11 @@ void snd_hda_update_power_acct(struct hda_codec *codec); static inline void snd_hda_set_power_save(struct hda_bus *bus, int delay) {} #endif +static inline bool hda_codec_need_resume(struct hda_codec *codec) +{ + return !codec->relaxed_resume && codec->jacktbl.used; +} + #ifdef CONFIG_SND_HDA_PATCH_LOADER /* * patch firmware diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 6cb72336433ad..1bc45c8507858 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2948,7 +2948,7 @@ static int hda_codec_runtime_resume(struct device *dev) static int hda_codec_force_resume(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); - bool forced_resume = !codec->relaxed_resume && codec->jacktbl.used; + bool forced_resume = hda_codec_need_resume(codec); int ret; /* The get/put pair below enforces the runtime resume even if the diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 6ccf34840e8a1..4604d1e16150d 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1025,7 +1025,7 @@ static int azx_suspend(struct device *dev) chip = card->private_data; bus = azx_bus(chip); snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - __azx_runtime_suspend(chip); + pm_runtime_force_suspend(dev); if (bus->irq >= 0) { free_irq(bus->irq, chip); bus->irq = -1; @@ -1041,7 +1041,9 @@ static int azx_suspend(struct device *dev) static int azx_resume(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); + struct hda_codec *codec; struct azx *chip; + bool forced_resume = false; if (!azx_is_pm_ready(card)) return 0; @@ -1052,7 +1054,20 @@ static int azx_resume(struct device *dev) chip->msi = 0; if (azx_acquire_irq(chip, 1) < 0) return -EIO; - __azx_runtime_resume(chip, false); + + /* check for the forced resume */ + list_for_each_codec(codec, &chip->bus) { + if (hda_codec_need_resume(codec)) { + forced_resume = true; + break; + } + } + + if (forced_resume) + pm_runtime_get_noresume(dev); + pm_runtime_force_resume(dev); + if (forced_resume) + pm_runtime_put(dev); snd_power_change_state(card, SNDRV_CTL_POWER_D0); trace_azx_resume(chip); @@ -1099,12 +1114,12 @@ static int azx_runtime_suspend(struct device *dev) if (!azx_is_pm_ready(card)) return 0; chip = card->private_data; - if (!azx_has_pm_runtime(chip)) - return 0; /* enable controller wake up event */ - azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) | - STATESTS_INT_MASK); + if (snd_power_get_state(card) == SNDRV_CTL_POWER_D0) { + azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) | + STATESTS_INT_MASK); + } __azx_runtime_suspend(chip); trace_azx_runtime_suspend(chip); @@ -1115,17 +1130,18 @@ static int azx_runtime_resume(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); struct azx *chip; + bool from_rt = snd_power_get_state(card) == SNDRV_CTL_POWER_D0; if (!azx_is_pm_ready(card)) return 0; chip = card->private_data; - if (!azx_has_pm_runtime(chip)) - return 0; - __azx_runtime_resume(chip, true); + __azx_runtime_resume(chip, from_rt); /* disable controller Wake Up event*/ - azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) & - ~STATESTS_INT_MASK); + if (from_rt) { + azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) & + ~STATESTS_INT_MASK); + } trace_azx_runtime_resume(chip); return 0; -- 2.20.1

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[PATCH AUTOSEL 5.4 09/26] ALSA: hda: Keep the controller initialization even if no codecs found
by Sasha Levin 24 Apr '20

24 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 9479e75fca370a5220784f7596bf598c4dad0b9b ] Currently, when the HD-audio controller driver doesn't detect any codecs, it tries to abort the probe. But this abort happens at the delayed probe, i.e. the primary probe call already returned success, hence the driver is never unbound until user does so explicitly. As a result, it may leave the HD-audio device in the running state without the runtime PM. More badly, if the device is a HD-audio bus that is tied with a GPU, GPU cannot reach to the full power down and consumes unnecessarily much power. This patch changes the logic after no-codec situation; it continues probing without the further codec initialization but keep the controller driver running normally. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043 Tested-by: Roy Spliet <nouveau(a)spliet.org> Link: https://lore.kernel.org/r/20200413082034.25166-5-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_intel.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 2b288c523dce0..6ccf34840e8a1 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1958,7 +1958,7 @@ static int azx_first_init(struct azx *chip) /* codec detection */ if (!azx_bus(chip)->codec_mask) { dev_err(card->dev, "no codecs found!\n"); - return -ENODEV; + /* keep running the rest for the runtime PM */ } if (azx_acquire_irq(chip, 0) < 0) @@ -2269,9 +2269,11 @@ static int azx_probe_continue(struct azx *chip) #endif /* create codec instances */ - err = azx_probe_codecs(chip, azx_max_codecs[chip->driver_type]); - if (err < 0) - goto out_free; + if (bus->codec_mask) { + err = azx_probe_codecs(chip, azx_max_codecs[chip->driver_type]); + if (err < 0) + goto out_free; + } #ifdef CONFIG_SND_HDA_PATCH_LOADER if (chip->fw) { @@ -2285,7 +2287,7 @@ static int azx_probe_continue(struct azx *chip) #endif } #endif - if ((probe_only[dev] & 1) == 0) { + if (bus->codec_mask && !(probe_only[dev] & 1)) { err = azx_codec_configure(chip); if (err < 0) goto out_free; -- 2.20.1

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[PATCH AUTOSEL 5.4 08/26] ALSA: hda: Release resources at error in delayed probe
by Sasha Levin 24 Apr '20

24 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 2393e7555b531a534152ffe7bfd1862cacedaacb ] snd-hda-intel driver handles the most of its probe task in the delayed work (either via workqueue or via firmware loader). When an error happens in the later delayed probe, we can't deregister the device itself because the probe callback already returned success and the device was bound. So, for now, we set hda->init_failed flag and make the rest untouched until the device gets really unbound. However, this leaves the device up running, keeping the resources without any use that prevents other operations. In this patch, we release the resources at first when a probe error happens in the delayed probe stage, but keeps the top-level object, so that the PM and other ops can still refer to the object itself. Also for simplicity, snd_hda_intel object is allocated via devm, so that we can get rid of the explicit kfree calls. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043 Link: https://lore.kernel.org/r/20200413082034.25166-4-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_intel.c | 29 ++++++++++++++++------------- sound/pci/hda/hda_intel.h | 1 + 2 files changed, 17 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 72bbfeddea24a..2b288c523dce0 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1200,10 +1200,8 @@ static void azx_vs_set_state(struct pci_dev *pci, if (!disabled) { dev_info(chip->card->dev, "Start delayed initialization\n"); - if (azx_probe_continue(chip) < 0) { + if (azx_probe_continue(chip) < 0) dev_err(chip->card->dev, "initialization error\n"); - hda->init_failed = true; - } } } else { dev_info(chip->card->dev, "%s via vga_switcheroo\n", @@ -1336,12 +1334,15 @@ static int register_vga_switcheroo(struct azx *chip) /* * destructor */ -static int azx_free(struct azx *chip) +static void azx_free(struct azx *chip) { struct pci_dev *pci = chip->pci; struct hda_intel *hda = container_of(chip, struct hda_intel, chip); struct hdac_bus *bus = azx_bus(chip); + if (hda->freed) + return; + if (azx_has_pm_runtime(chip) && chip->running) pm_runtime_get_noresume(&pci->dev); chip->running = 0; @@ -1385,9 +1386,8 @@ static int azx_free(struct azx *chip) if (chip->driver_caps & AZX_DCAPS_I915_COMPONENT) snd_hdac_i915_exit(bus); - kfree(hda); - return 0; + hda->freed = 1; } static int azx_dev_disconnect(struct snd_device *device) @@ -1403,7 +1403,8 @@ static int azx_dev_disconnect(struct snd_device *device) static int azx_dev_free(struct snd_device *device) { - return azx_free(device->device_data); + azx_free(device->device_data); + return 0; } #ifdef SUPPORT_VGA_SWITCHEROO @@ -1717,7 +1718,7 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, if (err < 0) return err; - hda = kzalloc(sizeof(*hda), GFP_KERNEL); + hda = devm_kzalloc(&pci->dev, sizeof(*hda), GFP_KERNEL); if (!hda) { pci_disable_device(pci); return -ENOMEM; @@ -1758,7 +1759,6 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, err = azx_bus_init(chip, model[dev]); if (err < 0) { - kfree(hda); pci_disable_device(pci); return err; } @@ -2306,13 +2306,16 @@ static int azx_probe_continue(struct azx *chip) pm_runtime_put_autosuspend(&pci->dev); out_free: - if (err < 0 || !hda->need_i915_power) + if (err < 0) { + azx_free(chip); + return err; + } + + if (!hda->need_i915_power) display_power(chip, false); - if (err < 0) - hda->init_failed = 1; complete_all(&hda->probe_wait); to_hda_bus(bus)->bus_probing = 0; - return err; + return 0; } static void azx_remove(struct pci_dev *pci) diff --git a/sound/pci/hda/hda_intel.h b/sound/pci/hda/hda_intel.h index 2acfff3da1a04..3fb119f090408 100644 --- a/sound/pci/hda/hda_intel.h +++ b/sound/pci/hda/hda_intel.h @@ -27,6 +27,7 @@ struct hda_intel { unsigned int use_vga_switcheroo:1; unsigned int vga_switcheroo_registered:1; unsigned int init_failed:1; /* delayed init failed */ + unsigned int freed:1; /* resources already released */ bool need_i915_power:1; /* the hda controller needs i915 power */ }; -- 2.20.1

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[PATCH AUTOSEL 5.6 22/38] ALSA: hda: call runtime_allow() for all hda controllers
by Sasha Levin 24 Apr '20

24 Apr '20

From: Hui Wang <hui.wang(a)canonical.com> [ Upstream commit 9a6418487b566503c772cb6e7d3d44e652b019b0 ] Before the pci_driver->probe() is called, the pci subsystem calls runtime_forbid() and runtime_get_sync() on this pci dev, so only call runtime_put_autosuspend() is not enough to enable the runtime_pm on this device. For controllers with vgaswitcheroo feature, the pci/quirks.c will call runtime_allow() for this dev, then the controllers could enter rt_idle/suspend/resume, but for non-vgaswitcheroo controllers like Intel hda controllers, the runtime_pm is not enabled because the runtime_allow() is not called. Since it is no harm calling runtime_allow() twice, here let hda driver call runtime_allow() for all controllers. Then the runtime_pm is enabled on all controllers after the put_autosuspend() is called. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Link: https://lore.kernel.org/r/20200414142725.6020-1-hui.wang@canonical.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 8519051a426e7..a5fab12defde2 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2356,6 +2356,7 @@ static int azx_probe_continue(struct azx *chip) if (azx_has_pm_runtime(chip)) { pm_runtime_use_autosuspend(&pci->dev); + pm_runtime_allow(&pci->dev); pm_runtime_put_autosuspend(&pci->dev); } -- 2.20.1

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[PATCH AUTOSEL 5.6 13/38] ALSA: hda: Explicitly permit using autosuspend if runtime PM is supported
by Sasha Levin 24 Apr '20

24 Apr '20

From: Roy Spliet <nouveau(a)spliet.org> [ Upstream commit 3ba21113bd33d49f3c300a23fc08cf114c434995 ] This fixes runtime PM not working after a suspend-to-RAM cycle at least for the codec-less HDA device found on NVIDIA GPUs. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043 Signed-off-by: Roy Spliet <nouveau(a)spliet.org> Link: https://lore.kernel.org/r/20200413082034.25166-7-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_intel.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 02c6308502b1e..8519051a426e7 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2354,8 +2354,10 @@ static int azx_probe_continue(struct azx *chip) set_default_power_save(chip); - if (azx_has_pm_runtime(chip)) + if (azx_has_pm_runtime(chip)) { + pm_runtime_use_autosuspend(&pci->dev); pm_runtime_put_autosuspend(&pci->dev); + } out_free: if (err < 0) { -- 2.20.1

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[PATCH AUTOSEL 5.6 11/38] ALSA: hda: Keep the controller initialization even if no codecs found
by Sasha Levin 24 Apr '20

24 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 9479e75fca370a5220784f7596bf598c4dad0b9b ] Currently, when the HD-audio controller driver doesn't detect any codecs, it tries to abort the probe. But this abort happens at the delayed probe, i.e. the primary probe call already returned success, hence the driver is never unbound until user does so explicitly. As a result, it may leave the HD-audio device in the running state without the runtime PM. More badly, if the device is a HD-audio bus that is tied with a GPU, GPU cannot reach to the full power down and consumes unnecessarily much power. This patch changes the logic after no-codec situation; it continues probing without the further codec initialization but keep the controller driver running normally. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043 Tested-by: Roy Spliet <nouveau(a)spliet.org> Link: https://lore.kernel.org/r/20200413082034.25166-5-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_intel.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 692857904d49e..aa0be85614b6c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2009,7 +2009,7 @@ static int azx_first_init(struct azx *chip) /* codec detection */ if (!azx_bus(chip)->codec_mask) { dev_err(card->dev, "no codecs found!\n"); - return -ENODEV; + /* keep running the rest for the runtime PM */ } if (azx_acquire_irq(chip, 0) < 0) @@ -2303,9 +2303,11 @@ static int azx_probe_continue(struct azx *chip) #endif /* create codec instances */ - err = azx_probe_codecs(chip, azx_max_codecs[chip->driver_type]); - if (err < 0) - goto out_free; + if (bus->codec_mask) { + err = azx_probe_codecs(chip, azx_max_codecs[chip->driver_type]); + if (err < 0) + goto out_free; + } #ifdef CONFIG_SND_HDA_PATCH_LOADER if (chip->fw) { @@ -2319,7 +2321,7 @@ static int azx_probe_continue(struct azx *chip) #endif } #endif - if ((probe_only[dev] & 1) == 0) { + if (bus->codec_mask && !(probe_only[dev] & 1)) { err = azx_codec_configure(chip); if (err < 0) goto out_free; -- 2.20.1

1 0

[PATCH AUTOSEL 5.6 10/38] ALSA: hda: Release resources at error in delayed probe
by Sasha Levin 24 Apr '20

24 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 2393e7555b531a534152ffe7bfd1862cacedaacb ] snd-hda-intel driver handles the most of its probe task in the delayed work (either via workqueue or via firmware loader). When an error happens in the later delayed probe, we can't deregister the device itself because the probe callback already returned success and the device was bound. So, for now, we set hda->init_failed flag and make the rest untouched until the device gets really unbound. However, this leaves the device up running, keeping the resources without any use that prevents other operations. In this patch, we release the resources at first when a probe error happens in the delayed probe stage, but keeps the top-level object, so that the PM and other ops can still refer to the object itself. Also for simplicity, snd_hda_intel object is allocated via devm, so that we can get rid of the explicit kfree calls. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043 Link: https://lore.kernel.org/r/20200413082034.25166-4-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_intel.c | 29 ++++++++++++++++------------- sound/pci/hda/hda_intel.h | 1 + 2 files changed, 17 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index f41d8b7864c1e..692857904d49e 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1203,10 +1203,8 @@ static void azx_vs_set_state(struct pci_dev *pci, if (!disabled) { dev_info(chip->card->dev, "Start delayed initialization\n"); - if (azx_probe_continue(chip) < 0) { + if (azx_probe_continue(chip) < 0) dev_err(chip->card->dev, "initialization error\n"); - hda->init_failed = true; - } } } else { dev_info(chip->card->dev, "%s via vga_switcheroo\n", @@ -1339,12 +1337,15 @@ static int register_vga_switcheroo(struct azx *chip) /* * destructor */ -static int azx_free(struct azx *chip) +static void azx_free(struct azx *chip) { struct pci_dev *pci = chip->pci; struct hda_intel *hda = container_of(chip, struct hda_intel, chip); struct hdac_bus *bus = azx_bus(chip); + if (hda->freed) + return; + if (azx_has_pm_runtime(chip) && chip->running) pm_runtime_get_noresume(&pci->dev); chip->running = 0; @@ -1388,9 +1389,8 @@ static int azx_free(struct azx *chip) if (chip->driver_caps & AZX_DCAPS_I915_COMPONENT) snd_hdac_i915_exit(bus); - kfree(hda); - return 0; + hda->freed = 1; } static int azx_dev_disconnect(struct snd_device *device) @@ -1406,7 +1406,8 @@ static int azx_dev_disconnect(struct snd_device *device) static int azx_dev_free(struct snd_device *device) { - return azx_free(device->device_data); + azx_free(device->device_data); + return 0; } #ifdef SUPPORT_VGA_SWITCHEROO @@ -1773,7 +1774,7 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, if (err < 0) return err; - hda = kzalloc(sizeof(*hda), GFP_KERNEL); + hda = devm_kzalloc(&pci->dev, sizeof(*hda), GFP_KERNEL); if (!hda) { pci_disable_device(pci); return -ENOMEM; @@ -1814,7 +1815,6 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, err = azx_bus_init(chip, model[dev]); if (err < 0) { - kfree(hda); pci_disable_device(pci); return err; } @@ -2340,13 +2340,16 @@ static int azx_probe_continue(struct azx *chip) pm_runtime_put_autosuspend(&pci->dev); out_free: - if (err < 0 || !hda->need_i915_power) + if (err < 0) { + azx_free(chip); + return err; + } + + if (!hda->need_i915_power) display_power(chip, false); - if (err < 0) - hda->init_failed = 1; complete_all(&hda->probe_wait); to_hda_bus(bus)->bus_probing = 0; - return err; + return 0; } static void azx_remove(struct pci_dev *pci) diff --git a/sound/pci/hda/hda_intel.h b/sound/pci/hda/hda_intel.h index 2acfff3da1a04..3fb119f090408 100644 --- a/sound/pci/hda/hda_intel.h +++ b/sound/pci/hda/hda_intel.h @@ -27,6 +27,7 @@ struct hda_intel { unsigned int use_vga_switcheroo:1; unsigned int vga_switcheroo_registered:1; unsigned int init_failed:1; /* delayed init failed */ + unsigned int freed:1; /* resources already released */ bool need_i915_power:1; /* the hda controller needs i915 power */ }; -- 2.20.1

1 0

[PATCH v2] ASoC: soc-compress: avoid false-positive Wuninitialized warning
by Rong Chen 24 Apr '20

24 Apr '20

gcc-6.5 and earlier show a new warning: sound/soc/soc-compress.c: In function ‘soc_compr_open’: sound/soc/soc-compress.c:75:28: warning: ‘component’ is used uninitialized in this function [-Wuninitialized] struct snd_soc_component *component, *save = NULL; ^~~~~~~~~ Simplest fix is to initialize it to avoid the warning. Reported-by: kbuild test robot <lkp(a)intel.com> Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Link: https://lore.kernel.org/lkml/202004201540.vYPhhYMs%25lkp@intel.com Signed-off-by: Rong Chen <rong.a.chen(a)intel.com> --- sound/soc/soc-compress.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index ceaf976db0bb..8431ff72be63 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -72,7 +72,7 @@ static int soc_compr_components_free(struct snd_compr_stream *cstream, static int soc_compr_open(struct snd_compr_stream *cstream) { struct snd_soc_pcm_runtime *rtd = cstream->private_data; - struct snd_soc_component *component, *save = NULL; + struct snd_soc_component *component = NULL, *save = NULL; struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int ret, i; -- 2.20.1

3 2

[PATCH] ASoC: soc-compress: avoid false-positive Wuninitialized warning
by Rong Chen 24 Apr '20

24 Apr '20

gcc-6.5 and earlier show a new warning: sound/soc/soc-compress.c: In function ‘soc_compr_open’: sound/soc/soc-compress.c:75:28: warning: ‘component’ is used uninitialized in this function [-Wuninitialized] struct snd_soc_component *component, *save = NULL; ^~~~~~~~~ Simplest fix is to initialize it to avoid the warning. Reported-by: kbuild test robot <lkp(a)intel.com> Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Link: https://lore.kernel.org/lkml/202004201540.vYPhhYMs%25lkp@intel.com Signed-off-by: Rong Chen <rong.a.chen(a)intel.com> --- sound/soc/soc-compress.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 50062eb79adb..e7bd80ed7f2e 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -72,7 +72,7 @@ static int soc_compr_components_free(struct snd_compr_stream *cstream, static int soc_compr_open(struct snd_compr_stream *cstream) { struct snd_soc_pcm_runtime *rtd = cstream->private_data; - struct snd_soc_component *component, *save = NULL; + struct snd_soc_component *component = NULL, *save = NULL; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret, i; -- 2.20.1

3 3

[PATCH] ASoc: nau8810: add AUX related dapm widgets and routes
by Seven Lee 24 Apr '20

24 Apr '20

This patch implements the following features: - AUX input for recording. - An input AUX output to SPK/MOUT. Signed-off-by: Seven Lee <wtli(a)nuvoton.com> --- sound/soc/codecs/nau8810.c | 19 ++++++++++++++++++- 1 file changed, 18 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8810.c b/sound/soc/codecs/nau8810.c index de26758c30a8..2a04243ffeaa 100644 --- a/sound/soc/codecs/nau8810.c +++ b/sound/soc/codecs/nau8810.c @@ -355,6 +355,8 @@ static const struct snd_kcontrol_new nau8810_snd_controls[] = { /* Speaker Output Mixer */ static const struct snd_kcontrol_new nau8810_speaker_mixer_controls[] = { + SOC_DAPM_SINGLE("AUX Bypass Switch", NAU8810_REG_SPKMIX, + NAU8810_AUXSPK_SFT, 1, 0), SOC_DAPM_SINGLE("Line Bypass Switch", NAU8810_REG_SPKMIX, NAU8810_BYPSPK_SFT, 1, 0), SOC_DAPM_SINGLE("PCM Playback Switch", NAU8810_REG_SPKMIX, @@ -363,6 +365,8 @@ static const struct snd_kcontrol_new nau8810_speaker_mixer_controls[] = { /* Mono Output Mixer */ static const struct snd_kcontrol_new nau8810_mono_mixer_controls[] = { + SOC_DAPM_SINGLE("AUX Bypass Switch", NAU8810_REG_MONOMIX, + NAU8810_AUXMOUT_SFT, 1, 0), SOC_DAPM_SINGLE("Line Bypass Switch", NAU8810_REG_MONOMIX, NAU8810_BYPMOUT_SFT, 1, 0), SOC_DAPM_SINGLE("PCM Playback Switch", NAU8810_REG_MONOMIX, @@ -371,6 +375,8 @@ static const struct snd_kcontrol_new nau8810_mono_mixer_controls[] = { /* PGA Mute */ static const struct snd_kcontrol_new nau8810_pgaboost_mixer_controls[] = { + SOC_DAPM_SINGLE("AUX PGA Switch", NAU8810_REG_ADCBOOST, + NAU8810_AUXBSTGAIN_SFT, 0x7, 0), SOC_DAPM_SINGLE("PGA Mute Switch", NAU8810_REG_PGAGAIN, NAU8810_PGAMT_SFT, 1, 1), SOC_DAPM_SINGLE("PMIC PGA Switch", NAU8810_REG_ADCBOOST, @@ -379,6 +385,8 @@ static const struct snd_kcontrol_new nau8810_pgaboost_mixer_controls[] = { /* Input PGA */ static const struct snd_kcontrol_new nau8810_inpga[] = { + SOC_DAPM_SINGLE("AUX Switch", NAU8810_REG_INPUT_SIGNAL, + NAU8810_AUXPGA_SFT, 1, 0), SOC_DAPM_SINGLE("MicN Switch", NAU8810_REG_INPUT_SIGNAL, NAU8810_NMICPGA_SFT, 1, 0), SOC_DAPM_SINGLE("MicP Switch", NAU8810_REG_INPUT_SIGNAL, @@ -425,6 +433,8 @@ static const struct snd_soc_dapm_widget nau8810_dapm_widgets[] = { SND_SOC_DAPM_MIXER("Input Boost Stage", NAU8810_REG_POWER2, NAU8810_BST_EN_SFT, 0, nau8810_pgaboost_mixer_controls, ARRAY_SIZE(nau8810_pgaboost_mixer_controls)), + SND_SOC_DAPM_PGA("AUX Input", NAU8810_REG_POWER1, + NAU8810_AUX_EN_SFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Mic Bias", NAU8810_REG_POWER1, NAU8810_MICBIAS_EN_SFT, 0, NULL, 0), @@ -434,6 +444,7 @@ static const struct snd_soc_dapm_widget nau8810_dapm_widgets[] = { SND_SOC_DAPM_SWITCH("Digital Loopback", SND_SOC_NOPM, 0, 0, &nau8810_loopback), + SND_SOC_DAPM_INPUT("AUX"), SND_SOC_DAPM_INPUT("MICN"), SND_SOC_DAPM_INPUT("MICP"), SND_SOC_DAPM_OUTPUT("MONOOUT"), @@ -445,10 +456,12 @@ static const struct snd_soc_dapm_route nau8810_dapm_routes[] = { {"DAC", NULL, "PLL", check_mclk_select_pll}, /* Mono output mixer */ + {"Mono Mixer", "AUX Bypass Switch", "AUX Input"}, {"Mono Mixer", "PCM Playback Switch", "DAC"}, {"Mono Mixer", "Line Bypass Switch", "Input Boost Stage"}, /* Speaker output mixer */ + {"Speaker Mixer", "AUX Bypass Switch", "AUX Input"}, {"Speaker Mixer", "PCM Playback Switch", "DAC"}, {"Speaker Mixer", "Line Bypass Switch", "Input Boost Stage"}, @@ -463,14 +476,18 @@ static const struct snd_soc_dapm_route nau8810_dapm_routes[] = { /* Input Boost Stage */ {"ADC", NULL, "Input Boost Stage"}, {"ADC", NULL, "PLL", check_mclk_select_pll}, + {"Input Boost Stage", "AUX PGA Switch", "AUX Input"}, {"Input Boost Stage", "PGA Mute Switch", "Input PGA"}, {"Input Boost Stage", "PMIC PGA Switch", "MICP"}, /* Input PGA */ - {"Input PGA", NULL, "Mic Bias"}, + {"Input PGA", NULL, "Mic Bias", check_mic_enabled}, + {"Input PGA", "AUX Switch", "AUX Input"}, {"Input PGA", "MicN Switch", "MICN"}, {"Input PGA", "MicP Switch", "MICP"}, + {"AUX Input", NULL, "AUX"}, + /* Digital Looptack */ {"Digital Loopback", "Switch", "ADC"}, {"DAC", NULL, "Digital Loopback"}, -- 2.25.2

2 1

[PATCH] ASoC: wm8962: Force suspend/resume during system suspend/resume
by Shengjiu Wang 24 Apr '20

24 Apr '20

Use force_suspend/resume to make sure clocks are disabled/enabled accordingly during system suspend/resume. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/wm8962.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 0a2cfff44441..80332c147da9 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3854,6 +3854,8 @@ static int wm8962_runtime_suspend(struct device *dev) static const struct dev_pm_ops wm8962_pm = { SET_RUNTIME_PM_OPS(wm8962_runtime_suspend, wm8962_runtime_resume, NULL) + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, + pm_runtime_force_resume) }; static const struct i2c_device_id wm8962_i2c_id[] = { -- 2.21.0

3 2

[PATCH v2] ASoC: wm8962: set CLOCKING2 as non-volatile register
by Shengjiu Wang 24 Apr '20

24 Apr '20

Previously CLOCKING2 is set as a volatile register, but cause issue at suspend & resume, that some bits of CLOCKING2 is not restored at resume, for example SYSCLK_SRC bits, then the output clock is wrong. The volatile property is caused by CLASSD_CLK_DIV bits, which are controlled by the chip itself. But the datasheet claims these are read only and protected by the security key, and they are not read by the driver at all. So it should be safe to change CLOCKING2 to be non-volatile. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Changes in v2: - Change according to Charles's suggestion to use non-volatile sound/soc/codecs/wm8962.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index d9d59f45833f..0a2cfff44441 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -118,7 +118,7 @@ static const struct reg_default wm8962_reg[] = { { 5, 0x0018 }, /* R5 - ADC & DAC Control 1 */ { 6, 0x2008 }, /* R6 - ADC & DAC Control 2 */ { 7, 0x000A }, /* R7 - Audio Interface 0 */ - + { 8, 0x01E4 }, /* R8 - Clocking2 */ { 9, 0x0300 }, /* R9 - Audio Interface 1 */ { 10, 0x00C0 }, /* R10 - Left DAC volume */ { 11, 0x00C0 }, /* R11 - Right DAC volume */ @@ -788,7 +788,6 @@ static bool wm8962_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { case WM8962_CLOCKING1: - case WM8962_CLOCKING2: case WM8962_SOFTWARE_RESET: case WM8962_THERMAL_SHUTDOWN_STATUS: case WM8962_ADDITIONAL_CONTROL_4: -- 2.21.0

3 2

[bug report] ASoC: fsl_easrc: Add EASRC ASoC CPU DAI drivers
by Dan Carpenter 24 Apr '20

24 Apr '20

Hello Shengjiu Wang, This is a semi-automatic email about new static checker warnings. The patch 955ac624058f: "ASoC: fsl_easrc: Add EASRC ASoC CPU DAI drivers" from Apr 16, 2020, leads to the following Smatch complaint: sound/soc/fsl/fsl_easrc.c:1529 fsl_easrc_hw_free() warn: variable dereferenced before check 'ctx' (see line 1527) sound/soc/fsl/fsl_easrc.c 1526 struct fsl_asrc_pair *ctx = runtime->private_data; 1527 struct fsl_easrc_ctx_priv *ctx_priv = ctx->private; ^^^^^ Dereference 1528 1529 if (ctx && (ctx_priv->ctx_streams & BIT(substream->stream))) { ^^^ This check is too late, to prevent a NULL dereference. 1530 ctx_priv->ctx_streams &= ~BIT(substream->stream); 1531 fsl_easrc_release_context(ctx); regards, dan carpenter

1 0

[PATCH 00/17] ASoC: soc-dai: add snd_soc_dai_xxx()
by Kuninori Morimoto 24 Apr '20

24 Apr '20

Hi Mark We have soc-dai.c today. DAI related functions should be implemented in it. These patches are do it. Kuninori Morimoto (17): ASoC: soc-dai: add soc_dai_err() ASoC: soc-dai: don't overwide dai->driver->ops ASoC: soc-dai: add snd_soc_pcm_dai_new() ASoC: soc-dai: add snd_soc_pcm_dai_prepare() ASoC: soc-dai: add snd_soc_pcm_dai_trigger() ASoC: soc-dai: add snd_soc_pcm_dai_bespoke_trigger() ASoC: soc-dai: add snd_soc_pcm_dai_probe() ASoC: soc-dai: add snd_soc_pcm_dai_remove() ASoC: soc-dai: add snd_soc_dai_compr_start() ASoC: soc-dai: add snd_soc_dai_compr_shutdown() ASoC: soc-dai: add snd_soc_dai_compr_trigger() ASoC: soc-dai: add snd_soc_dai_compr_set_params() ASoC: soc-dai: add snd_soc_dai_compr_get_params() ASoC: soc-dai: add snd_soc_dai_compr_ack() ASoC: soc-dai: add snd_soc_dai_compr_pointer() ASoC: soc-dai: add snd_soc_dai_compr_set_metadata() ASoC: soc-dai: add snd_soc_dai_compr_get_metadata() include/sound/soc-dai.h | 41 +++- sound/soc/soc-compress.c | 104 ++++----- sound/soc/soc-core.c | 85 +------- sound/soc/soc-dai.c | 453 +++++++++++++++++++++++++++++---------- sound/soc/soc-pcm.c | 50 ++--- 5 files changed, 436 insertions(+), 297 deletions(-) -- 2.17.1

5 27

[inline assembly compliance] Issues and patches
by FRÉDÉRIC RECOULES 24 Apr '20

24 Apr '20

Dear developpers, we are academic researchers working in automated program analysis. We are currently interested in checking compliance of inline asm chunks as found in C programs. While benchmarking our tool and technique, we found 5 significant issues in ALSA. We report them to you, as well as adequate patches with explanations. * All these bugs are related to compliance between the block of asm and its surrounding "contract" (in gcc-style notation). They are akin to undefined or implementation-defined behaviours in C: they currently do not manifest themselves in your program, but at some point in time with compiler optimizations becoming more and more aggressive or changes in undocumented compiler choices regarding asm chunks, they can suddenly trigger a (hard-to-find) bug. * The typical problems come from the compiler missing dataflow information and performing undue optimizations on this wrong basis, or the compiler allocating an already used register. Actually, we demonstrate "in lab" problems with all these categories of bugs in case of inlining (especially with LTO enabler) or code refactoring. * Some of those issues may seems benign or irrealistic but it cost nothing to patch so, why not do it? We would be very interested to hear your opinion on these matters. Are you interested in such errors and patches? Also, besides the patches, we are currently working on a code analyzer prototype designed to check asm compliance and to propose patches when the chunk is not compliant. This is still work in progress and we are finalizing it. The errors and patches I reported to you came from my prototype. In case such a prototype would be made available, would you consider using it? Best regards Frédéric Recoules -----------------------------------Details------------------------------------------ The issues are the sames for the five macro-functions defined in pcm/pcm_dmix_i386.h. 1. "memory" clobber is missing ------------------------------ The memory is read from src and written from dst and sum. For instance, the compiler may believe that the memory pointed by src has never been read and decide to get rid of its initialization (dead store elimination). Patched by adding "memory" to the clobbers 2. the read-only input size is clobbered ---------------------------------------- The compiler may assume the old value of size is still present at the location and may misuse the new value (i.e. 0). Patched by moving size from read-only input to read-write output. 3. ebx is not declared in clobbers ---------------------------------- We know that prior to the version 5.0, GCC refused to give ebx as clobber. It is no longer true and may become risky as it can be used to compute the address of one of the others memory inputs. Patched by creating a switch to add ebx in clobber (and removing the manual save/restore) if GCC is newer than 5.0 or other compiler is used. The following issue concern the MIX_AREAS_16_MMX macro-function. 4. mm0 is clobbered, mm1 is used uninitialized ---------------------------------------------- Patched by adding mm0 and mm1 to clobbers and adding a pxor instruction for mm1. Then here are some minor improvements: - add "cc" to the clobbers even if it is, for now, redundant; - allow to use a register (i.e. ebp) for size - allow to pass dst_step, src_step and sum_step through immediates -----------------------------------Patches------------------------------------------ diff --git a/src/pcm/pcm_dmix_i386.h b/src/pcm/pcm_dmix_i386.h index 2778cb1d..af2f4630 100644 --- a/src/pcm/pcm_dmix_i386.h +++ b/src/pcm/pcm_dmix_i386.h @@ -26,6 +26,13 @@ * */ +#define COMMA , +#if __GNUC__ < 5 && defined(__PIC__) +# define GCC_PIC_SWITCH(before,after) before +#else +# define GCC_PIC_SWITCH(before,after) after +#endif + /* * for plain i386 */ @@ -47,13 +54,14 @@ static void MIX_AREAS_16(unsigned int size, __asm__ __volatile__ ( "\n" - "\tmovl %%ebx, %7\n" /* ebx is GOT pointer (-fPIC) */ + /* ebx is GOT pointer (-fPIC) */ + GCC_PIC_SWITCH("\tmovl %%ebx, %1\n",) /* * initialization, load ESI, EDI, EBX registers */ - "\tmovl %1, %%edi\n" - "\tmovl %2, %%esi\n" - "\tmovl %3, %%ebx\n" + "\tmovl %2, %%edi\n" + "\tmovl %3, %%esi\n" + "\tmovl %4, %%ebx\n" "\tcmpl $0, %0\n" "\tjnz 2f\n" "\tjmp 7f\n" @@ -64,9 +72,9 @@ static void MIX_AREAS_16(unsigned int size, */ "\t.p2align 4,,15\n" "1:" - "\tadd %4, %%edi\n" - "\tadd %5, %%esi\n" - "\tadd %6, %%ebx\n" + "\tadd %5, %%edi\n" + "\tadd %6, %%esi\n" + "\tadd %7, %%ebx\n" /* * sample = *src; @@ -138,15 +146,16 @@ static void MIX_AREAS_16(unsigned int size, "\tjnz 4b\n" "\tdecl %0\n" "\tjnz 1b\n" - - "7:" - "\tmovl %7, %%ebx\n" /* ebx is GOT pointer (-fPIC) */ - : /* no output regs */ - : "m" (size), "m" (dst), "m" (src), - "m" (sum), "m" (dst_step), "m" (src_step), - "m" (sum_step), "m" (old_ebx) - : "esi", "edi", "edx", "ecx", "eax" + "7:" + /* ebx is GOT pointer (-fPIC) */ + GCC_PIC_SWITCH("\tmovl %1, %%ebx\n",) + + : "+&rm" (size), GCC_PIC_SWITCH("=m","=X") (old_ebx) + : "m" (dst), "m" (src), "m" (sum), + "im" (dst_step), "im" (src_step), "im" (sum_step) + : "esi", "edi", "edx", "ecx", GCC_PIC_SWITCH(,"ebx"COMMA) "eax", + "memory", "cc" ); } @@ -171,22 +180,24 @@ static void MIX_AREAS_16_MMX(unsigned int size, __asm__ __volatile__ ( "\n" - "\tmovl %%ebx, %7\n" /* ebx is GOT pointer (-fPIC) */ + /* ebx is GOT pointer (-fPIC) */ + GCC_PIC_SWITCH("\tmovl %%ebx, %1\n",) /* - * initialization, load ESI, EDI, EBX registers + * initialization, load ESI, EDI, EBX registers, clear MM1 */ - "\tmovl %1, %%edi\n" - "\tmovl %2, %%esi\n" - "\tmovl %3, %%ebx\n" + "\tpxor %%mm1, %%mm1\n" + "\tmovl %2, %%edi\n" + "\tmovl %3, %%esi\n" + "\tmovl %4, %%ebx\n" "\tcmpl $0, %0\n" "\tjnz 2f\n" "\tjmp 5f\n" "\t.p2align 4,,15\n" "1:" - "\tadd %4, %%edi\n" - "\tadd %5, %%esi\n" - "\tadd %6, %%ebx\n" + "\tadd %5, %%edi\n" + "\tadd %6, %%esi\n" + "\tadd %7, %%ebx\n" "2:" /* @@ -230,13 +241,14 @@ static void MIX_AREAS_16_MMX(unsigned int size, "\tjnz 1b\n" "\temms\n" "5:" - "\tmovl %7, %%ebx\n" /* ebx is GOT pointer (-fPIC) */ - - : /* no output regs */ - : "m" (size), "m" (dst), "m" (src), - "m" (sum), "m" (dst_step), "m" (src_step), - "m" (sum_step), "m" (old_ebx) - : "esi", "edi", "edx", "ecx", "eax" + /* ebx is GOT pointer (-fPIC) */ + GCC_PIC_SWITCH("\tmovl %1, %%ebx\n",) + + : "+&rm" (size), GCC_PIC_SWITCH("=m","=X") (old_ebx) + : "m" (dst), "m" (src), "m" (sum), + "im" (dst_step), "im" (src_step), "im" (sum_step) + : "esi", "edi", "edx", "ecx", GCC_PIC_SWITCH(,"ebx"COMMA) "eax", + "mm0", "mm1", "memory", "cc" ); } @@ -261,13 +273,14 @@ static void MIX_AREAS_32(unsigned int size, __asm__ __volatile__ ( "\n" - "\tmovl %%ebx, %7\n" /* ebx is GOT pointer (-fPIC) */ + /* ebx is GOT pointer (-fPIC) */ + GCC_PIC_SWITCH("\tmovl %%ebx, %1\n",) /* * initialization, load ESI, EDI, EBX registers */ - "\tmovl %1, %%edi\n" - "\tmovl %2, %%esi\n" - "\tmovl %3, %%ebx\n" + "\tmovl %2, %%edi\n" + "\tmovl %3, %%esi\n" + "\tmovl %4, %%ebx\n" "\tcmpl $0, %0\n" "\tjnz 1f\n" "\tjmp 6f\n" @@ -337,19 +350,20 @@ static void MIX_AREAS_32(unsigned int size, */ "\tdecl %0\n" "\tjz 6f\n" - "\tadd %4, %%edi\n" - "\tadd %5, %%esi\n" - "\tadd %6, %%ebx\n" + "\tadd %5, %%edi\n" + "\tadd %6, %%esi\n" + "\tadd %7, %%ebx\n" "\tjmp 1b\n" - - "6:" - "\tmovl %7, %%ebx\n" /* ebx is GOT pointer (-fPIC) */ - : /* no output regs */ - : "m" (size), "m" (dst), "m" (src), - "m" (sum), "m" (dst_step), "m" (src_step), - "m" (sum_step), "m" (old_ebx) - : "esi", "edi", "edx", "ecx", "eax" + "6:" + /* ebx is GOT pointer (-fPIC) */ + GCC_PIC_SWITCH("\tmovl %1, %%ebx\n",) + + : "+&rm" (size), GCC_PIC_SWITCH("=m","=X") (old_ebx) + : "m" (dst), "m" (src), "m" (sum), + "im" (dst_step), "im" (src_step), "im" (sum_step) + : "esi", "edi", "edx", "ecx", GCC_PIC_SWITCH(,"ebx"COMMA) "eax", + "memory", "cc" ); } @@ -374,13 +388,14 @@ static void MIX_AREAS_24(unsigned int size, __asm__ __volatile__ ( "\n" - "\tmovl %%ebx, %7\n" /* ebx is GOT pointer (-fPIC) */ + /* ebx is GOT pointer (-fPIC) */ + GCC_PIC_SWITCH("\tmovl %%ebx, %1\n",) /* * initialization, load ESI, EDI, EBX registers */ - "\tmovl %1, %%edi\n" - "\tmovl %2, %%esi\n" - "\tmovl %3, %%ebx\n" + "\tmovl %2, %%edi\n" + "\tmovl %3, %%esi\n" + "\tmovl %4, %%ebx\n" "\tcmpl $0, %0\n" "\tjnz 1f\n" "\tjmp 6f\n" @@ -443,19 +458,20 @@ static void MIX_AREAS_24(unsigned int size, */ "\tdecl %0\n" "\tjz 6f\n" - "\tadd %4, %%edi\n" - "\tadd %5, %%esi\n" - "\tadd %6, %%ebx\n" + "\tadd %5, %%edi\n" + "\tadd %6, %%esi\n" + "\tadd %7, %%ebx\n" "\tjmp 1b\n" - - "6:" - "\tmovl %7, %%ebx\n" /* ebx is GOT pointer (-fPIC) */ - : /* no output regs */ - : "m" (size), "m" (dst), "m" (src), - "m" (sum), "m" (dst_step), "m" (src_step), - "m" (sum_step), "m" (old_ebx) - : "esi", "edi", "edx", "ecx", "eax" + "6:" + /* ebx is GOT pointer (-fPIC) */ + GCC_PIC_SWITCH("\tmovl %1, %%ebx\n",) + + : "+&rm" (size), GCC_PIC_SWITCH("=m","=X") (old_ebx) + : "m" (dst), "m" (src), "m" (sum), + "im" (dst_step), "im" (src_step), "im" (sum_step) + : "esi", "edi", "edx", "ecx", GCC_PIC_SWITCH(,"ebx"COMMA) "eax", + "memory", "cc" ); } @@ -480,13 +496,14 @@ static void MIX_AREAS_24_CMOV(unsigned int size, __asm__ __volatile__ ( "\n" - "\tmovl %%ebx, %7\n" /* ebx is GOT pointer (-fPIC) */ + /* ebx is GOT pointer (-fPIC) */ + GCC_PIC_SWITCH("\tmovl %%ebx, %1\n",) /* * initialization, load ESI, EDI, EBX registers */ - "\tmovl %1, %%edi\n" - "\tmovl %2, %%esi\n" - "\tmovl %3, %%ebx\n" + "\tmovl %2, %%edi\n" + "\tmovl %3, %%esi\n" + "\tmovl %4, %%ebx\n" "\tcmpl $0, %0\n" "\tjz 6f\n" @@ -541,19 +558,20 @@ static void MIX_AREAS_24_CMOV(unsigned int size, /* * while (size-- > 0) */ - "\tadd %4, %%edi\n" - "\tadd %5, %%esi\n" - "\tadd %6, %%ebx\n" + "\tadd %5, %%edi\n" + "\tadd %6, %%esi\n" + "\tadd %7, %%ebx\n" "\tdecl %0\n" "\tjnz 1b\n" - - "6:" - "\tmovl %7, %%ebx\n" /* ebx is GOT pointer (-fPIC) */ - : /* no output regs */ - : "m" (size), "m" (dst), "m" (src), - "m" (sum), "m" (dst_step), "m" (src_step), - "m" (sum_step), "m" (old_ebx) - : "esi", "edi", "edx", "ecx", "eax" + "6:" + /* ebx is GOT pointer (-fPIC) */ + GCC_PIC_SWITCH("\tmovl %1, %%ebx\n",) + + : "+&rm" (size), GCC_PIC_SWITCH("=m","=X") (old_ebx) + : "m" (dst), "m" (src), "m" (sum), + "im" (dst_step), "im" (src_step), "im" (sum_step) + : "esi", "edi", "edx", "ecx", GCC_PIC_SWITCH(,"ebx"COMMA) "eax", + "memory", "cc" ); }

2 2

[PATCH] ALSA: usb-audio: Fix racy list management in output queue
by Takashi Iwai 24 Apr '20

24 Apr '20

The linked list entry from FIFO is peeked at queue_pending_output_urbs() but the actual element pop-out is performed outside the spinlock, and it's potentially racy. Do delete the link at the right place inside the spinlock. Fixes: 8fdff6a319e7 ("ALSA: snd-usb: implement new endpoint streaming model") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/endpoint.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index d8dc7cb56d43..50104f658ed4 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -346,17 +346,17 @@ static void queue_pending_output_urbs(struct snd_usb_endpoint *ep) ep->next_packet_read_pos %= MAX_URBS; /* take URB out of FIFO */ - if (!list_empty(&ep->ready_playback_urbs)) + if (!list_empty(&ep->ready_playback_urbs)) { ctx = list_first_entry(&ep->ready_playback_urbs, struct snd_urb_ctx, ready_list); + list_del_init(&ctx->ready_list); + } } spin_unlock_irqrestore(&ep->lock, flags); if (ctx == NULL) return; - list_del_init(&ctx->ready_list); - /* copy over the length information */ for (i = 0; i < packet->packets; i++) ctx->packet_size[i] = packet->packet_size[i]; -- 2.16.4

1 0

[PATCH v2] ALSA: hda: Refactor Intel NHLT init
by Cezary Rojewski 24 Apr '20

24 Apr '20

NHLT fetch based on _DSM prevents ACPI table override mechanism from being utilized. Make use of acpi_get_table to enable it and get rid of redundant code. In consequence, NHLT can be overridden just like any other ACPI table, e.g.: DSDT or SSDT. Change has been verified on all Intel AVS architecture platforms, RVP and production laptops both. Change possible due to addition of NHLT signature to the list of standard ACPI tables: https://patchwork.kernel.org/patch/11463235/ Override helps not only with debug purposes but also allows user for table adjustment when one found on their production hardware is invalid. Shared official NHLT spec is now available to community at: https://01.org/blogs/intel-smart-sound-technology-audio-dsp NHLT support for iASL is still ongoing subject but should be available in nearest future. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- Changes in v2: - commit message update, no functional changes sound/hda/intel-nhlt.c | 49 +++++++----------------------------------- 1 file changed, 8 insertions(+), 41 deletions(-) diff --git a/sound/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c index 99a23fe7fab9..2f741d2792d8 100644 --- a/sound/hda/intel-nhlt.c +++ b/sound/hda/intel-nhlt.c @@ -4,58 +4,25 @@ #include <linux/acpi.h> #include <sound/intel-nhlt.h> -#define NHLT_ACPI_HEADER_SIG "NHLT" - -/* Unique identification for getting NHLT blobs */ -static const guid_t osc_guid = - GUID_INIT(0xA69F886E, 0x6CEB, 0x4594, - 0xA4, 0x1F, 0x7B, 0x5D, 0xCE, 0x24, 0xC5, 0x53); - struct nhlt_acpi_table *intel_nhlt_init(struct device *dev) { - acpi_handle handle; - union acpi_object *obj; - struct nhlt_resource_desc *nhlt_ptr; - struct nhlt_acpi_table *nhlt_table = NULL; - - handle = ACPI_HANDLE(dev); - if (!handle) { - dev_err(dev, "Didn't find ACPI_HANDLE\n"); - return NULL; - } + struct nhlt_acpi_table *nhlt; + acpi_status status; - obj = acpi_evaluate_dsm(handle, &osc_guid, 1, 1, NULL); - - if (!obj) - return NULL; - - if (obj->type != ACPI_TYPE_BUFFER) { - dev_dbg(dev, "No NHLT table found\n"); - ACPI_FREE(obj); + status = acpi_get_table(ACPI_SIG_NHLT, 0, + (struct acpi_table_header **)&nhlt); + if (ACPI_FAILURE(status)) { + dev_warn(dev, "NHLT table not found\n"); return NULL; } - nhlt_ptr = (struct nhlt_resource_desc *)obj->buffer.pointer; - if (nhlt_ptr->length) - nhlt_table = (struct nhlt_acpi_table *) - memremap(nhlt_ptr->min_addr, nhlt_ptr->length, - MEMREMAP_WB); - ACPI_FREE(obj); - if (nhlt_table && - (strncmp(nhlt_table->header.signature, - NHLT_ACPI_HEADER_SIG, - strlen(NHLT_ACPI_HEADER_SIG)) != 0)) { - memunmap(nhlt_table); - dev_err(dev, "NHLT ACPI header signature incorrect\n"); - return NULL; - } - return nhlt_table; + return nhlt; } EXPORT_SYMBOL_GPL(intel_nhlt_init); void intel_nhlt_free(struct nhlt_acpi_table *nhlt) { - memunmap((void *)nhlt); + acpi_put_table((struct acpi_table_header *)nhlt); } EXPORT_SYMBOL_GPL(intel_nhlt_free); -- 2.17.1

2 1

[PATCH] ALSA: hda: Match both PCI ID and SSID for driver blacklist
by Takashi Iwai 24 Apr '20

24 Apr '20

The commit 3c6fd1f07ed0 ("ALSA: hda: Add driver blacklist") added a new blacklist for the devices that are known to have empty codecs, and one of the entries was ASUS ROG Zenith II (PCI SSID 1043:874f). However, it turned out that the very same PCI SSID is used for the previous model that does have the valid HD-audio codecs and the change broke the sound on it. Since the empty codec problem appear on the certain AMD platform (PCI ID 1022:1487), this patch changes the blacklist matching to both PCI ID and SSID using pci_match_id(). Also, the entry that was removed by the previous fix for ASUS ROG Zenigh II is re-added. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index d7adae316c0d..9f995576cff1 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2091,9 +2091,10 @@ static void pcm_mmap_prepare(struct snd_pcm_substream *substream, * some HD-audio PCI entries are exposed without any codecs, and such devices * should be ignored from the beginning. */ -static const struct snd_pci_quirk driver_blacklist[] = { - SND_PCI_QUIRK(0x1462, 0xcb59, "MSI TRX40 Creator", 0), - SND_PCI_QUIRK(0x1462, 0xcb60, "MSI TRX40", 0), +static const struct pci_device_id driver_blacklist[] = { + { PCI_DEVICE_SUB(0x1022, 0x1487, 0x1043, 0x874f) }, /* ASUS ROG Zenith II / Strix */ + { PCI_DEVICE_SUB(0x1022, 0x1487, 0x1462, 0xcb59) }, /* MSI TRX40 Creator */ + { PCI_DEVICE_SUB(0x1022, 0x1487, 0x1462, 0xcb60) }, /* MSI TRX40 */ {} }; @@ -2113,7 +2114,7 @@ static int azx_probe(struct pci_dev *pci, bool schedule_probe; int err; - if (snd_pci_quirk_lookup(pci, driver_blacklist)) { + if (pci_match_id(driver_blacklist, pci)) { dev_info(&pci->dev, "Skipping the blacklisted device\n"); return -ENODEV; } -- 2.16.4

1 0

[PATCH -next] soundwire: intel: Make sdw_intel_init static
by YueHaibing 23 Apr '20

23 Apr '20

Fix sparse warning: drivers/soundwire/intel_init.c:193:6: warning: symbol 'sdw_intel_init' was not declared. Should it be static? Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- drivers/soundwire/intel_init.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/drivers/soundwire/intel_init.c b/drivers/soundwire/intel_init.c index 4b769409f6f8..ad7053463889 100644 --- a/drivers/soundwire/intel_init.c +++ b/drivers/soundwire/intel_init.c @@ -190,7 +190,8 @@ static acpi_status sdw_intel_acpi_cb(acpi_handle handle, u32 level, * This scans the namespace and creates SoundWire link controller devices * based on the info queried. */ -void *sdw_intel_init(acpi_handle *parent_handle, struct sdw_intel_res *res) +static void *sdw_intel_init(acpi_handle *parent_handle, + struct sdw_intel_res *res) { acpi_status status; -- 2.17.1

4 3

sof-hda-dsp: fix speaker on ThinkPad X1C7
by GitHub pull_request - opened 23 Apr '20

23 Apr '20

alsa-project/alsa-ucm-conf pull request #24 was opened from chrisx8: With the current ucm2 profile at `sof-hda-dsp/HiFi.conf`, built-in speakers on the ThinkPad X1 Carbon Gen 7 sounds muffled/tinny. Muting 'Speaker' and setting 'Speaker' volume to 100 in `alsamixer` fixes this issue. I applied these fixes in the ucm2 profile, so they don't have to be applied manually. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/24 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/24.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ALSA: hda: Refactor Intel NHLT init
by Cezary Rojewski 23 Apr '20

23 Apr '20

NHLT fetch based on _DSM prevents ACPI table override mechanism from being utilized. Make use of acpi_get_table to enable it and get rid of redundant code. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- sound/hda/intel-nhlt.c | 49 +++++++----------------------------------- 1 file changed, 8 insertions(+), 41 deletions(-) diff --git a/sound/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c index 99a23fe7fab9..2f741d2792d8 100644 --- a/sound/hda/intel-nhlt.c +++ b/sound/hda/intel-nhlt.c @@ -4,58 +4,25 @@ #include <linux/acpi.h> #include <sound/intel-nhlt.h> -#define NHLT_ACPI_HEADER_SIG "NHLT" - -/* Unique identification for getting NHLT blobs */ -static const guid_t osc_guid = - GUID_INIT(0xA69F886E, 0x6CEB, 0x4594, - 0xA4, 0x1F, 0x7B, 0x5D, 0xCE, 0x24, 0xC5, 0x53); - struct nhlt_acpi_table *intel_nhlt_init(struct device *dev) { - acpi_handle handle; - union acpi_object *obj; - struct nhlt_resource_desc *nhlt_ptr; - struct nhlt_acpi_table *nhlt_table = NULL; - - handle = ACPI_HANDLE(dev); - if (!handle) { - dev_err(dev, "Didn't find ACPI_HANDLE\n"); - return NULL; - } + struct nhlt_acpi_table *nhlt; + acpi_status status; - obj = acpi_evaluate_dsm(handle, &osc_guid, 1, 1, NULL); - - if (!obj) - return NULL; - - if (obj->type != ACPI_TYPE_BUFFER) { - dev_dbg(dev, "No NHLT table found\n"); - ACPI_FREE(obj); + status = acpi_get_table(ACPI_SIG_NHLT, 0, + (struct acpi_table_header **)&nhlt); + if (ACPI_FAILURE(status)) { + dev_warn(dev, "NHLT table not found\n"); return NULL; } - nhlt_ptr = (struct nhlt_resource_desc *)obj->buffer.pointer; - if (nhlt_ptr->length) - nhlt_table = (struct nhlt_acpi_table *) - memremap(nhlt_ptr->min_addr, nhlt_ptr->length, - MEMREMAP_WB); - ACPI_FREE(obj); - if (nhlt_table && - (strncmp(nhlt_table->header.signature, - NHLT_ACPI_HEADER_SIG, - strlen(NHLT_ACPI_HEADER_SIG)) != 0)) { - memunmap(nhlt_table); - dev_err(dev, "NHLT ACPI header signature incorrect\n"); - return NULL; - } - return nhlt_table; + return nhlt; } EXPORT_SYMBOL_GPL(intel_nhlt_init); void intel_nhlt_free(struct nhlt_acpi_table *nhlt) { - memunmap((void *)nhlt); + acpi_put_table((struct acpi_table_header *)nhlt); } EXPORT_SYMBOL_GPL(intel_nhlt_free); -- 2.17.1

3 7

[PATCH] ASoC: SOF: Add missing dependency on IMX_SCU
by Mark Brown 23 Apr '20

23 Apr '20

This broke PowerPC allyesconfig. Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/imx/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/imx/Kconfig b/sound/soc/sof/imx/Kconfig index 5e0c68b12292..9a5ed1860e4e 100644 --- a/sound/soc/sof/imx/Kconfig +++ b/sound/soc/sof/imx/Kconfig @@ -21,6 +21,7 @@ config SND_SOC_SOF_IMX_OF config SND_SOC_SOF_IMX8_SUPPORT bool "SOF support for i.MX8" + depends on IMX_SCU help This adds support for Sound Open Firmware for NXP i.MX8 platforms Say Y if you have such a device. -- 2.20.1

3 5

[PATCH 0/4] ASoC:: don't use snd_soc_rtdcom_lookup()
by Kuninori Morimoto 23 Apr '20

23 Apr '20

Hi Mark Cc related engineer These patches are tring to not to use snd_soc_rtdcom_lookup() function on each drivers as much as possible, because we might have same name component under multi component situation. It can't find correct component in such case. I tried to add new feature on each drivers to not to use it, but I can't test. Thus, these patches should get Acked-by or Tested-by from each drivers user/maintenor. Please test these. After these patches, Intel / SOF drivers are still using snd_soc_rtdcom_lookup(). Because it is very complex, I couldn't try not to use it. If possible, each drivers should try to not use it, and it should be removed from ASoC. Kuninori Morimoto (4): ASoC: mediatek: don't use snd_soc_rtdcom_lookup() ASoC: intel: baytrail: don't use snd_soc_rtdcom_lookup() ASoC: intel: haswell: don't use snd_soc_rtdcom_lookup() ASoC: amd: don't use snd_soc_rtdcom_lookup() sound/soc/amd/raven/acp3x-pcm-dma.c | 6 ------ sound/soc/intel/baytrail/sst-baytrail-pcm.c | 13 +++++++------ sound/soc/intel/haswell/sst-haswell-pcm.c | 4 +++- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 12 +++++------- sound/soc/mediatek/common/mtk-afe-fe-dai.h | 3 ++- sound/soc/mediatek/common/mtk-base-afe.h | 6 ++++-- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 6 ++++-- sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 11 ++++------- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 7 ++++--- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 11 ++++------- 10 files changed, 37 insertions(+), 42 deletions(-) -- 2.17.1

3 10

[PATCH][next] ASoC: fsl_easrc: fix spelling mistake "prefitler" -> "prefilter"
by Colin King 23 Apr '20

23 Apr '20

From: Colin Ian King <colin.king(a)canonical.com> There is a spelling mistake in a deb_dbg message, fix it. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/fsl/fsl_easrc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_easrc.c b/sound/soc/fsl/fsl_easrc.c index 233f26ff885c..97658e1f4989 100644 --- a/sound/soc/fsl/fsl_easrc.c +++ b/sound/soc/fsl/fsl_easrc.c @@ -1769,7 +1769,7 @@ static void fsl_easrc_dump_firmware(struct fsl_asrc *easrc) } dev_dbg(dev, "Firmware v%u dump:\n", firm->firmware_version); - dev_dbg(dev, "Num prefitler scenarios: %u\n", firm->prefil_scen); + dev_dbg(dev, "Num prefilter scenarios: %u\n", firm->prefil_scen); dev_dbg(dev, "Num interpolation scenarios: %u\n", firm->interp_scen); dev_dbg(dev, "\nInterpolation scenarios:\n"); -- 2.25.1

2 1

[PATCH v1] ASoC: Intel: Skylake: Replace guid_copy() with import_guid()
by Andy Shevchenko 23 Apr '20

23 Apr '20

There is a specific API to treat raw data as GUID, i.e. import_guid(). Use it instead of guid_copy() with explicit casting. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/intel/skylake/skl-sst-utils.c | 2 +- sound/soc/intel/skylake/skl-topology.c | 6 +++--- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/skylake/skl-sst-utils.c b/sound/soc/intel/skylake/skl-sst-utils.c index d43cbf4a71ef2..b233f89517c11 100644 --- a/sound/soc/intel/skylake/skl-sst-utils.c +++ b/sound/soc/intel/skylake/skl-sst-utils.c @@ -290,7 +290,7 @@ int snd_skl_parse_uuids(struct sst_dsp *ctx, const struct firmware *fw, goto free_uuid_list; } - guid_copy(&module->uuid, (guid_t *)&mod_entry->uuid); + import_guid(&module->uuid, mod_entry->uuid); module->id = (i | (index << 12)); module->is_loadable = mod_entry->type.load_type; diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 69cd7a81bf2ac..43a159f23a10d 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -1989,7 +1989,7 @@ static int skl_tplg_get_uuid(struct device *dev, guid_t *guid, struct snd_soc_tplg_vendor_uuid_elem *uuid_tkn) { if (uuid_tkn->token == SKL_TKN_UUID) { - guid_copy(guid, (guid_t *)&uuid_tkn->uuid); + import_guid(guid, uuid_tkn->uuid); return 0; } @@ -3376,8 +3376,8 @@ static int skl_tplg_get_manifest_tkn(struct device *dev, dev_err(dev, "Too many UUID tokens\n"); return -EINVAL; } - guid_copy(&skl->modules[uuid_index++]->uuid, - (guid_t *)&array->uuid->uuid); + import_guid(&skl->modules[uuid_index++]->uuid, + array->uuid->uuid); tuple_size += sizeof(*array->uuid); continue; -- 2.26.1

2 1

[PATCH v2 0/3] Add support for different DMIC configurations
by Mateusz Gorski 23 Apr '20

23 Apr '20

Set of patches to enable DMIC capture on different hardware configurations. Information about supported DMIC configuration is read from NHLT and correct pipeline configuration is selected automatically. Also, adding additional option for topology binary name which is based on used machine driver. Mateusz Gorski (3): ASoC: Intel: Skylake: Add alternative topology binary name ASoC: Intel: Multiple I/O PCM format support for pipe ASoC: Intel: Skylake: Automatic DMIC format configuration according to information from NHLT Changes in v2: - removed patch 1/4 swapping machine device and platform device registration order - alt_tplg_name creation now uses different field to read machine driver name - including of <sound/soc-acpi.h> moved to different patch include/uapi/sound/skl-tplg-interface.h | 2 + sound/soc/intel/skylake/skl-topology.c | 189 +++++++++++++++++++++++- sound/soc/intel/skylake/skl-topology.h | 1 + 3 files changed, 187 insertions(+), 5 deletions(-) -- 2.17.1

3 8

[PATCH] ALSA: hda: Always use jackpoll helper for jack update after resume
by Takashi Iwai 23 Apr '20

23 Apr '20

HD-audio codec driver applies a tricky procedure to forcibly perform the runtime resume by mimicking the usage count even if the device has been runtime-suspended beforehand. This was needed to assure to trigger the jack detection update after the system resume. And recently we also applied the similar logic to the HD-audio controller side. However this seems leading to some inconsistency, and eventually PCI controller gets screwed up. This patch is an attempt to fix and clean up those behavior: instead of the tricky runtime resume procedure, the existing jackpoll work is scheduled when such a forced codec resume is required. The jackpoll work will power up the codec, and this alone should suffice for the jack status update in usual cases. If the extra polling is requested (by checking codec->jackpoll_interval), the manual update is invoked after that, and the codec is powered down again. Also, we filter the spurious wake up of the codec from the controller runtime resume by checking codec->relaxed_resume flag. If this flag is set, basically we don't need to wake up explicitly, but it's supposed to be done via the audio component notifier. Fixes: c4c8dd6ef807 ("ALSA: hda: Skip controller resume if not needed") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 28 +++++++++++++++++----------- sound/pci/hda/hda_intel.c | 17 ++--------------- 2 files changed, 19 insertions(+), 26 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 86a632bf4d50..7e3ae4534df9 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -641,8 +641,18 @@ static void hda_jackpoll_work(struct work_struct *work) struct hda_codec *codec = container_of(work, struct hda_codec, jackpoll_work.work); - snd_hda_jack_set_dirty_all(codec); - snd_hda_jack_poll_all(codec); + /* for non-polling trigger: we need nothing if already powered on */ + if (!codec->jackpoll_interval && snd_hdac_is_power_on(&codec->core)) + return; + + /* the power-up/down sequence triggers the runtime resume */ + snd_hda_power_up_pm(codec); + /* update jacks manually if polling is required, too */ + if (codec->jackpoll_interval) { + snd_hda_jack_set_dirty_all(codec); + snd_hda_jack_poll_all(codec); + } + snd_hda_power_down_pm(codec); if (!codec->jackpoll_interval) return; @@ -2951,18 +2961,14 @@ static int hda_codec_runtime_resume(struct device *dev) static int hda_codec_force_resume(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); - bool forced_resume = hda_codec_need_resume(codec); int ret; - /* The get/put pair below enforces the runtime resume even if the - * device hasn't been used at suspend time. This trick is needed to - * update the jack state change during the sleep. - */ - if (forced_resume) - pm_runtime_get_noresume(dev); ret = pm_runtime_force_resume(dev); - if (forced_resume) - pm_runtime_put(dev); + /* schedule jackpoll work for jack detection update */ + if (codec->jackpoll_interval || + (pm_runtime_suspended(dev) && hda_codec_need_resume(codec))) + schedule_delayed_work(&codec->jackpoll_work, + codec->jackpoll_interval); return ret; } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 9f995576cff1..0310193ea1bd 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1004,7 +1004,8 @@ static void __azx_runtime_resume(struct azx *chip, bool from_rt) if (status && from_rt) { list_for_each_codec(codec, &chip->bus) - if (status & (1 << codec->addr)) + if (!codec->relaxed_resume && + (status & (1 << codec->addr))) schedule_delayed_work(&codec->jackpoll_work, codec->jackpoll_interval); } @@ -1044,9 +1045,7 @@ static int azx_suspend(struct device *dev) static int azx_resume(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); - struct hda_codec *codec; struct azx *chip; - bool forced_resume = false; if (!azx_is_pm_ready(card)) return 0; @@ -1058,19 +1057,7 @@ static int azx_resume(struct device *dev) if (azx_acquire_irq(chip, 1) < 0) return -EIO; - /* check for the forced resume */ - list_for_each_codec(codec, &chip->bus) { - if (hda_codec_need_resume(codec)) { - forced_resume = true; - break; - } - } - - if (forced_resume) - pm_runtime_get_noresume(dev); pm_runtime_force_resume(dev); - if (forced_resume) - pm_runtime_put(dev); snd_power_change_state(card, SNDRV_CTL_POWER_D0); trace_azx_resume(chip); -- 2.16.4

3 5

Re: [PATCH] ASoC: wm8962: restore the CLOCKING2 register in resume
by S.j. Wang 23 Apr '20

23 Apr '20

> > On Tue, Apr 21, 2020 at 08:02:15PM +0800, Shengjiu Wang wrote: > > The CLOCKING2 is a volatile register, but some bits should be restored > > when resume, for example SYSCLK_SRC. otherwise the output clock is > > wrong > > > > Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> > > --- > > sound/soc/codecs/wm8962.c | 8 ++++++++ > > 1 file changed, 8 insertions(+) > > > > diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c > > index d9d59f45833f..6e96c0c5ad2a 100644 > > --- a/sound/soc/codecs/wm8962.c > > +++ b/sound/soc/codecs/wm8962.c > > @@ -82,6 +82,7 @@ struct wm8962_priv { #endif > > > > int irq; > > + u32 regcache_clocking2; > > }; > > > > /* We can't use the same notifier block for more than one supply and > > @@ -3813,6 +3814,10 @@ static int wm8962_runtime_resume(struct > device > > *dev) > > > > regcache_sync(wm8962->regmap); > > > > + regmap_update_bits(wm8962->regmap, WM8962_CLOCKING2, > > + WM8962_SYSCLK_SRC_MASK, > > + wm8962->regcache_clocking2); > > + > > I wonder if it might just be better to make the register non-volatile? From > looking through the datasheet I am guessing this is volatile for the > CLASSD_CLK_DIV bits, which are controlled by the chip itself. But the > datasheet claims these are read only and protected by the security key, and > they are not read by the driver at all. > > Thanks, > Charles > Use non-volatile also can fix the issue, I will send v2 for it Best regards Wang shengjiu

1 0

New codec ALC245
by Kailang 23 Apr '20

23 Apr '20

Hi Takashi, Attach was ALC245 supported patch. Many Thanks. BR, Kailang

2 1

[PATCH] ALSA: usb-audio: Fix usb audio refcnt leak when getting spdif
by Xiyu Yang 23 Apr '20

23 Apr '20

snd_microii_spdif_default_get() invokes snd_usb_lock_shutdown(), which increases the refcount of the snd_usb_audio object "chip". When snd_microii_spdif_default_get() returns, local variable "chip" becomes invalid, so the refcount should be decreased to keep refcount balanced. The reference counting issue happens in several exception handling paths of snd_microii_spdif_default_get(). When those error scenarios occur such as usb_ifnum_to_if() returns NULL, the function forgets to decrease the refcnt increased by snd_usb_lock_shutdown(), causing a refcnt leak. Fix this issue by jumping to "end" label when those error scenarios occur. Signed-off-by: Xiyu Yang <xiyuyang19(a)fudan.edu.cn> Signed-off-by: Xin Tan <tanxin.ctf(a)gmail.com> --- sound/usb/mixer_quirks.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index c237e24f08d9..0f072426b84c 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -1508,11 +1508,15 @@ static int snd_microii_spdif_default_get(struct snd_kcontrol *kcontrol, /* use known values for that card: interface#1 altsetting#1 */ iface = usb_ifnum_to_if(chip->dev, 1); - if (!iface || iface->num_altsetting < 2) - return -EINVAL; + if (!iface || iface->num_altsetting < 2) { + err = -EINVAL; + goto end; + } alts = &iface->altsetting[1]; - if (get_iface_desc(alts)->bNumEndpoints < 1) - return -EINVAL; + if (get_iface_desc(alts)->bNumEndpoints < 1) { + err = -EINVAL; + goto end; + } ep = get_endpoint(alts, 0)->bEndpointAddress; err = snd_usb_ctl_msg(chip->dev, -- 2.7.4

2 1

Unrecoverable AER error when resuming from RAM (hda regression in 5.7-rc2)
by Alex Xu (Hello71) 23 Apr '20

23 Apr '20

With 5.7-rc2, after resuming from suspend to RAM, I get: [ 55.679382] pcieport 0000:00:03.1: AER: Multiple Uncorrected (Non-Fatal) error received: 0000:00:00.0 [ 55.679405] pcieport 0000:00:03.1: AER: PCIe Bus Error: severity=Uncorrected (Non-Fatal), type=Transaction Layer, (Requester ID) [ 55.679410] pcieport 0000:00:03.1: AER: device [1022:1453] error status/mask=00100000/04400000 [ 55.679414] pcieport 0000:00:03.1: AER: [20] UnsupReq (First) [ 55.679417] pcieport 0000:00:03.1: AER: TLP Header: 40000004 0a0000ff fffc0e80 00000000 [ 55.679423] amdgpu 0000:0a:00.0: AER: can't recover (no error_detected callback) [ 55.679425] snd_hda_intel 0000:0a:00.1: AER: can't recover (no error_detected callback) [ 55.679455] pcieport 0000:00:03.1: AER: device recovery failed Then the display freezes and the system basically falls apart (can't even sudo reboot -f, need to use magic sysrq). I bisected this to "ALSA: hda: Skip controller resume if not needed". Setting snd_hda_intel.power_save=0 resolves the issue. I am using an ASRock B450 Pro4 with Realtek HDA codec: [ 1.009400] snd_hda_intel 0000:0a:00.1: enabling device (0000 -> 0002) [ 1.009425] snd_hda_intel 0000:0a:00.1: Force to non-snoop mode [ 1.009653] snd_hda_intel 0000:0c:00.3: enabling device (0000 -> 0002) [ 1.021452] snd_hda_codec_generic hdaudioC0D0: ignore pin 0x7, too many assigned pins [ 1.021461] snd_hda_codec_generic hdaudioC0D0: ignore pin 0x9, too many assigned pins [ 1.021471] snd_hda_codec_generic hdaudioC0D0: ignore pin 0xb, too many assigned pins [ 1.021480] snd_hda_codec_generic hdaudioC0D0: ignore pin 0xd, too many assigned pins [ 1.021482] snd_hda_codec_generic hdaudioC0D0: autoconfig for Generic: line_outs=0 (0x0/0x0/0x0/0x0/0x0) type:line [ 1.021482] snd_hda_codec_generic hdaudioC0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 1.021483] snd_hda_codec_generic hdaudioC0D0: hp_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 1.021484] snd_hda_codec_generic hdaudioC0D0: mono: mono_out=0x0 [ 1.021484] snd_hda_codec_generic hdaudioC0D0: dig-out=0x3/0x5 [ 1.021485] snd_hda_codec_generic hdaudioC0D0: inputs: [ 1.046053] snd_hda_codec_realtek hdaudioC1D0: autoconfig for ALC892: line_outs=1 (0x14/0x0/0x0/0x0/0x0) type:line [ 1.046054] snd_hda_codec_realtek hdaudioC1D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 1.046055] snd_hda_codec_realtek hdaudioC1D0: hp_outs=1 (0x1b/0x0/0x0/0x0/0x0) [ 1.046055] snd_hda_codec_realtek hdaudioC1D0: mono: mono_out=0x0 [ 1.046056] snd_hda_codec_realtek hdaudioC1D0: inputs: [ 1.046057] snd_hda_codec_realtek hdaudioC1D0: Front Mic=0x19 [ 1.046058] snd_hda_codec_realtek hdaudioC1D0: Rear Mic=0x18 [ 1.046058] snd_hda_codec_realtek hdaudioC1D0: Line=0x1a I also have an ASUS RX 480 graphics card with HDMI audio output.

3 5

[RFC PATCH 00/16] ASoC/SOF/clk/gpio/dt: add Hifiberry DAC+ PRO support
by Pierre-Louis Bossart 22 Apr '20

22 Apr '20

The Hifiberry DAC+ / DAC+ PRO is supported in the Raspberry PI tree but until now not in the mainline and not for ACPI platforms. This patchset implements the recommendations suggested by Mark Brown back in 2018: first add a gpiochip in the PCM512x codec driver, then use these gpios from a clock driver and the machine driver. Since this patchset relies on different subsystems, sending as RFC for now. I chose to import the original code from the Raspberry PI tree as is, and add my changes on top. If there is a preference to squash the changes that'd be fine. I also don't know if I should split this series in two, one for ASoC and one for clk changes? This patchset does not add changes to the sound/soc/bcm machine drivers, but that should be trivial once all the gpio/clock is available. Thanks to Andy Shevchenko for his help navigating the gpio subsystem and flagging mistakes in the use of lookup tables, and to Rob Herring for pointers on the DT bindings verification tools. Daniel Matuschek (1): clk: hifiberry-dacpro: initial import Pierre-Louis Bossart (15): ASoC: pcm512x: expose 6 GPIOs ASoC: pcm512x: use "sclk" string to retrieve clock ASoC: Intel: sof-pcm512x: use gpiod for LED ASoC: Intel: sof-pcm512x: detect Hifiberry DAC+ PRO ASoC: Intel: sof-pcm512x: reconfigure sclk in hw_params if needed ASoC: Intel: sof-pcm512x: select HIFIBERRY_DACPRO clk clk: hifiberry-dacpro: update SDPX/copyright clk: hifiberry-dacpro: style cleanups, use devm_ clk: hifiberry-dacpro: add OF dependency clk: hifiberry-dacpro: transition to _hw functions clk: hifiberry-dacpro: add ACPI support clk: hifiberry-dacpro: add "sclk" lookup clk: hifiberry-dacpro: toggle GPIOs on prepare/unprepare clk: hifiberry-dacpro: add delay on clock prepare/deprepare ASoC: dt-bindings: add document for Hifiberry DAC+ PRO clock .../bindings/sound/hifiberry-dacpro.yaml | 38 +++ drivers/clk/Kconfig | 4 + drivers/clk/Makefile | 1 + drivers/clk/clk-hifiberry-dacpro.c | 284 ++++++++++++++++++ sound/soc/codecs/pcm512x.c | 134 ++++++++- sound/soc/intel/boards/Kconfig | 2 + sound/soc/intel/boards/sof_pcm512x.c | 190 +++++++++++- 7 files changed, 635 insertions(+), 18 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/hifiberry-dacpro.yaml create mode 100644 drivers/clk/clk-hifiberry-dacpro.c base-commit: dd8e871d4e560eeb8d22af82dde91457ad835a63 -- 2.20.1

5 67

sound:core:oss: what determines the value of plugin->extra_data in rate_src_frames()?
by Xingyu Li 22 Apr '20

22 Apr '20

Hi all, I encountered a problem about ALSA subsystem when I try to reproduce a bug https://syzkaller.appspot.com/bug?id=54576f48248190caa29334a754275ce5e1d8be… . I run the PoC https://syzkaller.appspot.com/text?tag=ReproC&x=1275090ee00000, it is supposed to crash the kernel. But it sometimes does not. Then I found the point is rate_src_frames() in sound/core/oss/rate.c. When I can not reproduce the bug,data->old_src_frames variable in rate_src_frames() and the passed arguments frames are both 11. When it can reproduce, the value of data->old_src_frames is 85. The variable data is one member extra_data in plugin variable. But I can not find where it write to plugin->extra_data. So I want to what determines the value of plugin->extra_data? Why is the value of it different at different time? Thanks

1 0

alsa-gobject: seq: Revert "seq: tstamp: use wrapper structure instead of union"
by GitHub pull_request - opened 22 Apr '20

22 Apr '20

alsa-project/alsa-gobject pull request #19 was opened from takaswie: This reverts commit 20fdc2d893ccba609cc786ebac733bf6914f9874. In the above commit, union type structure is wrapped with care of language bindings in which union-compatible feature is not expected. However, it's responsible for language bindings to parse gir and the care is not necessarily required in shared library side. In first place, ALSASeq.Tstamp is boxed type object. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Request URL : https://github.com/alsa-project/alsa-gobject/pull/19 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/19.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[GIT PULL] ASoC: tegra: Fixes for v5.7-rc3
by Thierry Reding 22 Apr '20

22 Apr '20

Hi Liam, Mark, The following changes since commit 8f3d9f354286745c751374f5f1fcafee6b3f3136: Linux 5.7-rc1 (2020-04-12 12:35:55 -0700) are available in the Git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/tegra/linux.git tags/tegra-for-5.7-asoc for you to fetch changes up to ff5d18cb04f4ecccbcf05b7f83ab6df2a0d95c16: ASoC: tegra: Enable audio mclk during tegra_asoc_utils_init() (2020-04-21 16:53:15 +0200) This is a set of patches that I was supposed to send out during the merge window[0], but I somehow messed up my scripts and then didn't notice that this was missing. I've rebased these patches on top of v5.7-rc1. The lack of these are now causing some issues with unbalanced clock reference counts. Is there any chance you can pick these up for v5.7-rc3? Thanks, Thierry [0]: https://lore.kernel.org/lkml/20200327154506.GF4437@sirena.org.uk/ ---------------------------------------------------------------- ASoC: tegra: Fixes for v5.7-rc3 This contains a couple of fixes that are needed to properly reconfigure the audio clocks on older Tegra devices. ---------------------------------------------------------------- Sowjanya Komatineni (4): ASoC: tegra: Use device managed resource APIs to get the clock ASoC: nau8825: Change Tegra clk_out_2 provider to PMC ASoC: tegra: Add audio mclk parent configuration ASoC: tegra: Enable audio mclk during tegra_asoc_utils_init() .../devicetree/bindings/sound/nau8825.txt | 2 +- sound/soc/tegra/tegra_alc5632.c | 7 +- sound/soc/tegra/tegra_asoc_utils.c | 113 ++++++++++----------- sound/soc/tegra/tegra_asoc_utils.h | 1 - sound/soc/tegra/tegra_max98090.c | 22 ++-- sound/soc/tegra/tegra_rt5640.c | 22 ++-- sound/soc/tegra/tegra_rt5677.c | 7 +- sound/soc/tegra/tegra_sgtl5000.c | 7 +- sound/soc/tegra/tegra_wm8753.c | 22 ++-- sound/soc/tegra/tegra_wm8903.c | 22 ++-- sound/soc/tegra/tegra_wm9712.c | 8 +- sound/soc/tegra/trimslice.c | 18 +--- 12 files changed, 90 insertions(+), 161 deletions(-)

2 1

[PATCH] ASoC: SOF: Fix build
by Mark Brown 22 Apr '20

22 Apr '20

The recent batch of SOF changes failed to build on some x86 configurations including an allmodconfig, revert the commits: e150ef4169a76 ASoC: SOF: Introduce extended manifest 371091417864b ASoC: SOF: ext_manifest: parse firmware version 7c024b948c206 ASoC: SOF: ext_manifest: parse compiler version 9e72f13ee541c ASoC: SOF: ext_manifest: parse windows to fix this. Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/uapi/sound/sof/ext_manifest.h | 91 ------------- sound/soc/sof/loader.c | 189 +------------------------- 2 files changed, 2 insertions(+), 278 deletions(-) delete mode 100644 include/uapi/sound/sof/ext_manifest.h diff --git a/include/uapi/sound/sof/ext_manifest.h b/include/uapi/sound/sof/ext_manifest.h deleted file mode 100644 index d49c47d08c7f..000000000000 --- a/include/uapi/sound/sof/ext_manifest.h +++ /dev/null @@ -1,91 +0,0 @@ -/* SPDX-License-Identifier: ((GPL-2.0 WITH Linux-syscall-note) OR BSD-3-Clause) */ -/* - * This file is provided under a dual BSD/GPLv2 license. When using or - * redistributing this file, you may do so under either license. - * - * Copyright(c) 2020 Intel Corporation. All rights reserved. - */ - -/* - * Extended manifest is a place to store metadata about firmware, known during - * compilation time - for example firmware version or used compiler. - * Given information are read on host side before firmware startup. - * This part of output binary is not signed. - */ - -#ifndef __SOF_FIRMWARE_EXT_MANIFEST_H__ -#define __SOF_FIRMWARE_EXT_MANIFEST_H__ - -#include <linux/const.h> -#include <sound/sof/info.h> - -/* In ASCII `XMan` */ -#define SOF_EXT_MAN_MAGIC_NUMBER 0x6e614d58 - -/* Build u32 number in format MMmmmppp */ -#define SOF_EXT_MAN_BUILD_VERSION(MAJOR, MINOR, PATH) ((uint32_t)( \ - ((MAJOR) << 24) | \ - ((MINOR) << 12) | \ - (PATH))) - -/* check extended manifest version consistency */ -#define SOF_EXT_MAN_VERSION_INCOMPATIBLE(host_ver, cli_ver) ( \ - ((host_ver) & GENMASK(31, 24)) != \ - ((cli_ver) & GENMASK(31, 24))) - -/* used extended manifest header version */ -#define SOF_EXT_MAN_VERSION SOF_EXT_MAN_BUILD_VERSION(1, 0, 0) - -/* extended manifest header, deleting any field breaks backward compatibility */ -struct sof_ext_man_header { - uint32_t magic; /*< identification number, */ - /*< EXT_MAN_MAGIC_NUMBER */ - uint32_t full_size; /*< [bytes] full size of ext_man, */ - /*< (header + content + padding) */ - uint32_t header_size; /*< [bytes] makes header extensionable, */ - /*< after append new field to ext_man header */ - /*< then backward compatible won't be lost */ - uint32_t header_version; /*< value of EXT_MAN_VERSION */ - /*< not related with following content */ - uint8_t elements[]; /*< list of ext_man_elem_* elements */ -} __packed; - -/* Now define extended manifest elements */ - -/* Extended manifest elements types */ -enum sof_ext_man_elem_type { - SOF_EXT_MAN_ELEM_FW_VERSION = 0, - SOF_EXT_MAN_ELEM_WINDOW = SOF_IPC_EXT_WINDOW, - SOF_EXT_MAN_ELEM_CC_VERSION = SOF_IPC_EXT_CC_INFO, -}; - -/* extended manifest element header */ -struct sof_ext_man_elem_header { - uint32_t type; /*< SOF_EXT_MAN_ELEM_ */ - uint32_t size; /*< in bytes, including header size */ - uint8_t blob[]; /*< type dependent content */ -} __packed; - -/* FW version */ -struct sof_ext_man_fw_version { - struct sof_ext_man_elem_header hdr; - /* use sof_ipc struct because of code re-use */ - struct sof_ipc_fw_version version; - uint32_t flags; -} __packed; - -/* extended data memory windows for IPC, trace and debug */ -struct sof_ext_man_window { - struct sof_ext_man_elem_header hdr; - /* use sof_ipc struct because of code re-use */ - struct sof_ipc_window ipc_window; -} __packed; - -/* Used C compiler description */ -struct sof_ext_man_cc_version { - struct sof_ext_man_elem_header hdr; - /* use sof_ipc struct because of code re-use */ - struct sof_ipc_cc_version cc_version; -} __packed; - -#endif /* __SOF_FIRMWARE_EXT_MANIFEST_H__ */ diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c index 8be30cd5e038..89f35db2577d 100644 --- a/sound/soc/sof/loader.c +++ b/sound/soc/sof/loader.c @@ -12,7 +12,6 @@ #include <linux/firmware.h> #include <sound/sof.h> -#include <uapi/sound/sof/ext_manifest.h> #include "ops.h" static int get_ext_windows(struct snd_sof_dev *sdev, @@ -20,21 +19,13 @@ static int get_ext_windows(struct snd_sof_dev *sdev, { const struct sof_ipc_window *w = container_of(ext_hdr, struct sof_ipc_window, ext_hdr); - size_t w_size = struct_size(w, window, w->num_windows); - - if (sdev->info_window) { - if (memcmp(sdev->info_window, w, w_size)) { - dev_err(sdev->dev, "error: mistmatch between window descriptor from extended manifest and mailbox"); - return -EINVAL; - } - return 0; - } if (w->num_windows == 0 || w->num_windows > SOF_IPC_MAX_ELEMS) return -EINVAL; /* keep a local copy of the data */ - sdev->info_window = kmemdup(w, w_size, GFP_KERNEL); + sdev->info_window = kmemdup(w, struct_size(w, window, w->num_windows), + GFP_KERNEL); if (!sdev->info_window) return -ENOMEM; @@ -49,14 +40,6 @@ static int get_cc_info(struct snd_sof_dev *sdev, const struct sof_ipc_cc_version *cc = container_of(ext_hdr, struct sof_ipc_cc_version, ext_hdr); - if (sdev->cc_version) { - if (memcmp(sdev->cc_version, cc, cc->ext_hdr.hdr.size)) { - dev_err(sdev->dev, "error: receive diverged cc_version descriptions"); - return -EINVAL; - } - return 0; - } - dev_dbg(sdev->dev, "Firmware info: used compiler %s %d:%d:%d%s used optimization flags %s\n", cc->name, cc->major, cc->minor, cc->micro, cc->desc, cc->optim); @@ -143,151 +126,6 @@ int snd_sof_fw_parse_ext_data(struct snd_sof_dev *sdev, u32 bar, u32 offset) } EXPORT_SYMBOL(snd_sof_fw_parse_ext_data); -static int ext_man_get_fw_version(struct snd_sof_dev *sdev, - const struct sof_ext_man_elem_header *hdr) -{ - const struct sof_ext_man_fw_version *v; - - v = container_of(hdr, struct sof_ext_man_fw_version, hdr); - - memcpy(&sdev->fw_ready.version, &v->version, sizeof(v->version)); - sdev->fw_ready.flags = v->flags; - - /* log ABI versions and check FW compatibility */ - return snd_sof_ipc_valid(sdev); -} - -static int ext_man_get_windows(struct snd_sof_dev *sdev, - const struct sof_ext_man_elem_header *hdr) -{ - const struct sof_ipc_ext_data_hdr *w_ipc; - const struct sof_ext_man_window *w; - - w = container_of(hdr, struct sof_ext_man_window, hdr); - w_ipc = (const struct sof_ipc_ext_data_hdr *)&w->ipc_window; - - return get_ext_windows(sdev, w_ipc); -} - -static int ext_man_get_cc_info(struct snd_sof_dev *sdev, - const struct sof_ext_man_elem_header *hdr) -{ - const struct sof_ext_man_cc_version *cc; - const struct sof_ipc_ext_data_hdr *cc_version; - - cc = container_of(hdr, struct sof_ext_man_cc_version, hdr); - cc_version = (const struct sof_ipc_ext_data_hdr *)&cc->cc_version; - - return get_cc_info(sdev, cc_version); -} - -static ssize_t snd_sof_ext_man_size(const struct firmware *fw) -{ - const struct sof_ext_man_header *head = (void *)fw->data; - - /* - * assert fw size is big enough to contain extended manifest header, - * it prevents from reading unallocated memory from `head` in following - * step. - */ - if (fw->size < sizeof(*head)) - return -EINVAL; - - /* - * When fw points to extended manifest, - * then first u32 must be equal SOF_EXT_MAN_MAGIC_NUMBER. - */ - if (head->magic == SOF_EXT_MAN_MAGIC_NUMBER) - return head->full_size; - - /* otherwise given fw don't have an extended manifest */ - return 0; -} - -/* parse extended FW manifest data structures */ -static int snd_sof_fw_ext_man_parse(struct snd_sof_dev *sdev, - const struct firmware *fw) -{ - const struct sof_ext_man_elem_header *elem_hdr; - const struct sof_ext_man_header *head; - ssize_t ext_man_size; - ssize_t remaining; - uintptr_t iptr; - int ret = 0; - - head = (struct sof_ext_man_header *)fw->data; - remaining = head->full_size - head->header_size; - ext_man_size = snd_sof_ext_man_size(fw); - - /* Assert firmware starts with extended manifest */ - if (ext_man_size < 0) { - dev_err(sdev->dev, "error: exception while reading firmware extended manifest, code %d\n", - (int)ext_man_size); - return ext_man_size; - } else if (!ext_man_size) { - dev_err(sdev->dev, "error: can't parse extended manifest when it's not present\n"); - return -EINVAL; - } - - /* incompatible version */ - if (SOF_EXT_MAN_VERSION_INCOMPATIBLE(SOF_EXT_MAN_VERSION, - head->header_version)) { - dev_err(sdev->dev, "error: extended manifest version 0x%X differ from used 0x%X\n", - head->header_version, SOF_EXT_MAN_VERSION); - return -EINVAL; - } - - /* get first extended manifest element header */ - iptr = (uintptr_t)fw->data + head->header_size; - - while (remaining > sizeof(*elem_hdr)) { - elem_hdr = (struct sof_ext_man_elem_header *)iptr; - - dev_dbg(sdev->dev, "found sof_ext_man header type %d size 0x%X\n", - elem_hdr->type, elem_hdr->size); - - if (elem_hdr->size < sizeof(*elem_hdr) || - elem_hdr->size > remaining) { - dev_err(sdev->dev, "error: invalid sof_ext_man header size, type %d size 0x%X\n", - elem_hdr->type, elem_hdr->size); - break; - } - - /* process structure data */ - switch (elem_hdr->type) { - case SOF_EXT_MAN_ELEM_FW_VERSION: - ret = ext_man_get_fw_version(sdev, elem_hdr); - break; - case SOF_EXT_MAN_ELEM_WINDOW: - ret = ext_man_get_windows(sdev, elem_hdr); - break; - case SOF_EXT_MAN_ELEM_CC_VERSION: - ret = ext_man_get_cc_info(sdev, elem_hdr); - break; - default: - dev_warn(sdev->dev, "warning: unknown sof_ext_man header type %d size 0x%X\n", - elem_hdr->type, elem_hdr->size); - break; - } - - if (ret < 0) { - dev_err(sdev->dev, "error: failed to parse sof_ext_man header type %d size 0x%X\n", - elem_hdr->type, elem_hdr->size); - break; - } - - remaining -= elem_hdr->size; - iptr += elem_hdr->size; - } - - if (remaining) { - dev_err(sdev->dev, "error: sof_ext_man header is inconsistent\n"); - ret = -EINVAL; - } - - return ret; -} - /* * IPC Firmware ready. */ @@ -635,7 +473,6 @@ int snd_sof_load_firmware_raw(struct snd_sof_dev *sdev) { struct snd_sof_pdata *plat_data = sdev->pdata; const char *fw_filename; - ssize_t ext_man_size; int ret; /* Don't request firmware again if firmware is already requested */ @@ -653,33 +490,11 @@ int snd_sof_load_firmware_raw(struct snd_sof_dev *sdev) if (ret < 0) { dev_err(sdev->dev, "error: request firmware %s failed err: %d\n", fw_filename, ret); - goto err; } else { dev_dbg(sdev->dev, "request_firmware %s successful\n", fw_filename); } - /* check for extended manifest */ - ext_man_size = snd_sof_ext_man_size(plat_data->fw); - if (ext_man_size > 0) { - ret = snd_sof_fw_ext_man_parse(sdev, plat_data->fw); - - /* when no error occurred, drop extended manifest */ - if (!ret) - plat_data->fw_offset = ext_man_size; - else - dev_err(sdev->dev, "error: firmware %s contains unsupported or invalid extended manifest: %d\n", - fw_filename, ret); - } else if (!ext_man_size) { - /* No extended manifest, so nothing to skip during FW load */ - dev_dbg(sdev->dev, "firmware doesn't contain extended manifest\n"); - } else { - ret = ext_man_size; - dev_err(sdev->dev, "error: firmware %s contains unsupported or invalid extended manifest: %d\n", - fw_filename, ret); - } - -err: kfree(fw_filename); return ret; -- 2.20.1

3 8

[PATCH] ASoC: soc-core: return true, false in snd_soc_volsw_is_stereo()
by Jason Yan 22 Apr '20

22 Apr '20

Fix the following coccicheck warning: include/sound/soc.h:1271:9-10: WARNING: return of 0/1 in function 'snd_soc_volsw_is_stereo' with return type bool Signed-off-by: Jason Yan <yanaijie(a)huawei.com> --- include/sound/soc.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 946f88a6c63d..d1e9a526363e 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1268,13 +1268,13 @@ static inline void *snd_soc_card_get_drvdata(struct snd_soc_card *card) static inline bool snd_soc_volsw_is_stereo(struct soc_mixer_control *mc) { if (mc->reg == mc->rreg && mc->shift == mc->rshift) - return 0; + return false; /* * mc->reg == mc->rreg && mc->shift != mc->rshift, or * mc->reg != mc->rreg means that the control is * stereo (bits in one register or in two registers) */ - return 1; + return true; } static inline unsigned int snd_soc_enum_val_to_item(struct soc_enum *e, -- 2.21.1

2 1

[PATCH] ASoC: dmic: Allow GPIO operations to sleep
by Mark Brown 22 Apr '20

22 Apr '20

If there is a power GPIO provided we control it from DAPM context so there is no problem with a sleeping GPIO, use the _cansleep() version of the API to allow this. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/dmic.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c index f5560a49b9e5..5d079d90fd3b 100644 --- a/sound/soc/codecs/dmic.c +++ b/sound/soc/codecs/dmic.c @@ -59,14 +59,14 @@ static int dmic_aif_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_POST_PMU: if (dmic->gpio_en) - gpiod_set_value(dmic->gpio_en, 1); + gpiod_set_value_cansleep(dmic->gpio_en, 1); if (dmic->wakeup_delay) msleep(dmic->wakeup_delay); break; case SND_SOC_DAPM_POST_PMD: if (dmic->gpio_en) - gpiod_set_value(dmic->gpio_en, 0); + gpiod_set_value_cansleep(dmic->gpio_en, 0); break; } -- 2.20.1

2 4

[PATCH 0/4] Add support for different DMIC configurations
by Mateusz Gorski 22 Apr '20

22 Apr '20

Set of patches to enable DMIC capture on different hardware configurations. Information about supported DMIC configuration is read from NHLT and correct pipeline configuration is selected automatically. Also, adding additional option for topology binary name which is based on used machine driver. Mateusz Gorski (4): ASoC: Intel: Skylake: Change the order of machine device and platform registration ASoC: Intel: Skylake: Add alternative topology binary name ASoC: Intel: Multiple I/O PCM format support for pipe ASoC: Intel: Skylake: Automatic DMIC format configuration according to information from NHLT include/uapi/sound/skl-tplg-interface.h | 2 + sound/soc/intel/skylake/skl-topology.c | 189 +++++++++++++++++++++++- sound/soc/intel/skylake/skl-topology.h | 1 + sound/soc/intel/skylake/skl.c | 10 +- 4 files changed, 192 insertions(+), 10 deletions(-) -- 2.17.1

3 8

[PATCH] ASoC: SOF: ensure all fields in header are zero'd before copying back to userspace
by Colin King 22 Apr '20

22 Apr '20

From: Colin Ian King <colin.king(a)canonical.com> Field header.tlv is uninitialized and being copied back to userspace and hence leaking data from the stack to userspace. Fix this by ensuring the header structure is zero'd. Fixes: c3078f539704 ("ASoC: SOF: Add Sound Open Firmware KControl support") Addresses-Coverity: ("Uninitialized scalar variable") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/sof/control.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/control.c b/sound/soc/sof/control.c index dfc412e2d956..97d5c1a4c1ff 100644 --- a/sound/soc/sof/control.c +++ b/sound/soc/sof/control.c @@ -362,7 +362,7 @@ int snd_sof_bytes_ext_get(struct snd_kcontrol *kcontrol, struct snd_sof_control *scontrol = be->dobj.private; struct snd_soc_component *scomp = scontrol->scomp; struct sof_ipc_ctrl_data *cdata = scontrol->control_data; - struct snd_ctl_tlv header; + struct snd_ctl_tlv header = { }; struct snd_ctl_tlv __user *tlvd = (struct snd_ctl_tlv __user *)binary_data; int data_size; -- 2.25.1

3 2

[PATCH] ALSA: usb-audio: Add connector notifier delegation
by Takashi Iwai 22 Apr '20

22 Apr '20

It turned out that ALC1220-VB USB-audio device gives the interrupt event to some PCM terminals while those don't allow the connector state request but only the actual I/O terminals return the request. The recent commit 7dc3c5a0172e ("ALSA: usb-audio: Don't create jack controls for PCM terminals") excluded those phantom terminals, so those events are ignored, too. My first thought was that this could be easily deduced from the associated terminals, but some of them have even no associate terminal ID, hence it's not too trivial to figure out. Since the number of such terminals are small and limited, this patch implements another quirk table for the simple mapping of the connectors. It's not really scalable, but let's hope that there will be not many such funky devices in future. Fixes: 7dc3c5a0172e ("ALSA: usb-audio: Don't create jack controls for PCM terminals") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer.c | 25 +++++++++++++++++++++++++ sound/usb/mixer.h | 10 ++++++++++ sound/usb/mixer_maps.c | 13 +++++++++++++ 3 files changed, 48 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index ecd5036a0b44..a88d7854513b 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -3117,6 +3117,7 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) if (map->id == state.chip->usb_id) { state.map = map->map; state.selector_map = map->selector_map; + mixer->connector_map = map->connector_map; mixer->ignore_ctl_error |= map->ignore_ctl_error; break; } @@ -3198,10 +3199,32 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) return 0; } +static int delegate_notify(struct usb_mixer_interface *mixer, int unitid, + u8 *control, u8 *channel) +{ + const struct usbmix_connector_map *map = mixer->connector_map; + + if (!map) + return unitid; + + for (; map->id; map++) { + if (map->id == unitid) { + if (control && map->control) + *control = map->control; + if (channel && map->channel) + *channel = map->channel; + return map->delegated_id; + } + } + return unitid; +} + void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid) { struct usb_mixer_elem_list *list; + unitid = delegate_notify(mixer, unitid, NULL, NULL); + for_each_mixer_elem(list, mixer, unitid) { struct usb_mixer_elem_info *info = mixer_elem_list_to_info(list); @@ -3271,6 +3294,8 @@ static void snd_usb_mixer_interrupt_v2(struct usb_mixer_interface *mixer, return; } + unitid = delegate_notify(mixer, unitid, &control, &channel); + for_each_mixer_elem(list, mixer, unitid) count++; diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index 65d6d08c96f5..41ec9dc4139b 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -6,6 +6,13 @@ struct media_mixer_ctl; +struct usbmix_connector_map { + u8 id; + u8 delegated_id; + u8 control; + u8 channel; +}; + struct usb_mixer_interface { struct snd_usb_audio *chip; struct usb_host_interface *hostif; @@ -18,6 +25,9 @@ struct usb_mixer_interface { /* the usb audio specification version this interface complies to */ int protocol; + /* optional connector delegation map */ + const struct usbmix_connector_map *connector_map; + /* Sound Blaster remote control stuff */ const struct rc_config *rc_cfg; u32 rc_code; diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index fb2c4a992951..0260c750e156 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -27,6 +27,7 @@ struct usbmix_ctl_map { u32 id; const struct usbmix_name_map *map; const struct usbmix_selector_map *selector_map; + const struct usbmix_connector_map *connector_map; int ignore_ctl_error; }; @@ -387,6 +388,15 @@ static const struct usbmix_name_map trx40_mobo_map[] = { {} }; +static const struct usbmix_connector_map trx40_mobo_connector_map[] = { + { 10, 16 }, /* (Back) Speaker */ + { 11, 17 }, /* Front Headphone */ + { 13, 7 }, /* Line */ + { 14, 8 }, /* Mic */ + { 15, 9 }, /* Front Mic */ + {} +}; + /* * Control map entries */ @@ -519,6 +529,7 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = { { /* Gigabyte TRX40 Aorus Pro WiFi */ .id = USB_ID(0x0414, 0xa002), .map = trx40_mobo_map, + .connector_map = trx40_mobo_connector_map, }, { /* ASUS ROG Zenith II */ .id = USB_ID(0x0b05, 0x1916), @@ -531,10 +542,12 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = { { /* MSI TRX40 Creator */ .id = USB_ID(0x0db0, 0x0d64), .map = trx40_mobo_map, + .connector_map = trx40_mobo_connector_map, }, { /* MSI TRX40 */ .id = USB_ID(0x0db0, 0x543d), .map = trx40_mobo_map, + .connector_map = trx40_mobo_connector_map, }, { 0 } /* terminator */ }; -- 2.16.4

1 0

[PATCH] ALSA: oxygen: use true,false for bool variables
by Jason Yan 22 Apr '20

22 Apr '20

Fix the following coccicheck warning: sound/pci/oxygen/xonar_pcm179x.c:463:1-17: WARNING: Assignment of 0/1 to bool variable sound/pci/oxygen/xonar_pcm179x.c:505:1-17: WARNING: Assignment of 0/1 to bool variable Signed-off-by: Jason Yan <yanaijie(a)huawei.com> --- sound/pci/oxygen/xonar_pcm179x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index 6a0520c4fb5a..cf801a235df9 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -460,7 +460,7 @@ static void xonar_st_init(struct oxygen *chip) data->generic.anti_pop_delay = 100; data->h6 = chip->model.dac_channels_mixer > 2; - data->has_cs2000 = 1; + data->has_cs2000 = true; data->cs2000_regs[CS2000_FUN_CFG_1] = CS2000_REF_CLK_DIV_1; data->broken_i2c = true; @@ -502,7 +502,7 @@ static void xonar_xense_init(struct oxygen *chip) xonar_init_ext_power(chip); data->generic.anti_pop_delay = 100; - data->has_cs2000 = 1; + data->has_cs2000 = true; data->cs2000_regs[CS2000_FUN_CFG_1] = CS2000_REF_CLK_DIV_1; oxygen_write16(chip, OXYGEN_I2S_A_FORMAT, -- 2.21.1

2 1

[PATCH] ALSA: usb-audio: Fix a limit check in proc_dump_substream_formats()
by Dan Carpenter 22 Apr '20

22 Apr '20

This should be ARRAY_SIZE() instead of sizeof(). The sizeof() limit is too high so it doesn't work. Fixes: 093b8494f299 ("ALSA: usb-audio: Print more information in stream proc files") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/usb/proc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/proc.c b/sound/usb/proc.c index 5a36e192ebb0..889c550c9f29 100644 --- a/sound/usb/proc.c +++ b/sound/usb/proc.c @@ -140,7 +140,7 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s snd_iprintf(buffer, " Channel map:"); for (c = 0; c < map->channels; c++) { - if (map->map[c] >= sizeof(channel_labels) || + if (map->map[c] >= ARRAY_SIZE(channel_labels) || !channel_labels[map->map[c]]) snd_iprintf(buffer, " --"); else -- 2.26.1

2 1

v2: Add support to SDM845 based platforms
by GitHub pull_request - opened 22 Apr '20

22 Apr '20

alsa-project/alsa-ucm-conf pull request #23 was opened from Srinivas-Kandagatla: Add Support to two codecs WCD934x and WSA881x and two platforms DragonBoard DB845c and Lenovo-YOGA-C630-13Q50 that use this codec. For ucm-validator to pass this PR depends on config files PR "v2: ucm-validator: add config files for DB845c and Lenovo-YOGA-C630-13Q50" (https://github.com/alsa-project/alsa-tests/pull/3) to alsa-tests Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/23 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/23.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

alsa-gobject: seq: some improvements and fixes
by GitHub pull_request - opened 22 Apr '20

22 Apr '20

alsa-project/alsa-gobject pull request #18 was opened from takaswie: In UAPI of ALSA Sequencer, some members in structures are defined as fixed-sized array. For accessor methods to the members, this patchset adds changes so that function annotations have 'not a variable length array type' in their prototype. In some cases, it's convenient to compare two instance of ALSASeq.Addr. This patchset adds a new API, 'ALSASeq.Addr.equal()' for this purpose. Finally, this patchset also adds some improvements and fixes. Request URL : https://github.com/alsa-project/alsa-gobject/pull/18 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/18.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

How to support multi FIFO address in ALSA & DMA Engine?
by Shengjiu Wang 22 Apr '20

22 Apr '20

Hi Experts Some cpu-dai devices support multi FIFO, so we need to tell DMA which fifos and how many fifos it should fill the data to. For example, when channels = 2, only one fifo is used. when channels = 4, then two fifos are used. or when channels = 2, two fifos are used, when channels= 4, four fifos are used. But in snd_dmaengine_dai_dma_data, and dma_slave_config, there is only one dma address. Do you have any idea to support multi FIFO? is there any implemented reference? Best regards Wang shengjiu

1 0

[GIT PULL] ASoC fix v5.7-rc2
by Mark Brown 21 Apr '20

21 Apr '20

The following changes since commit ccfc531695f3a4aada042f6bdb33ac6be24e1aec: ASoC: cs4270: pull reset GPIO low then high (2020-04-07 15:29:54 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.7-rc2 for you to fetch changes up to 1e060a453c8604311fb45ae2f84f67ed673329b4: ASoC: wm8960: Fix wrong clock after suspend & resume (2020-04-21 15:43:22 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.7 Quite a lot of fixes here, a lot of driver specific ones but the biggest one is the revert of changes to the startup and shutdown sequence for DAIs that went in during the merge window - they broke some older x86 platforms and attempts to fix them didn't succeed so it's safer to just roll them back and try to make sure those platforms are handled properly in any future attempt. The rockchip S/PDIF DT stuff was IIRC for validation issues. ---------------------------------------------------------------- Akshu Agrawal (1): ASoC: amd: Fix button configuration Amadeusz Sławiński (8): ASoC: topology: Add missing memory checks ASoC: topology: Check return value of soc_tplg_create_tlv ASoC: topology: Check return value of soc_tplg_*_create ASoC: topology: Check soc_tplg_add_route return value ASoC: topology: Check return value of pcm_new_ver ASoC: topology: Check return value of soc_tplg_dai_config ASoC: topology: Fix endianness issue ASoC: codecs: hdac_hdmi: Fix incorrect use of list_for_each_entry Bjorn Andersson (1): ASoC: qcom: common: Silence duplicate parse error messages Charles Keepax (4): ASoC: dapm: Fix regression introducing multiple copies of DAI widgets ASoC: dapm: Move error message to avoid some duplication ASoC: dapm: Remove dapm_connect_dai_link_widgets helper ASoC: madera: Remove a couple of stray blank lines Gyeongtaek Lee (1): ASoC: dapm: fixup dapm kcontrol widget Jason Yan (2): ASoC: intel: soc-acpi-intel-icl-match: remove useless 'rt1308_2_adr' ASoC: Intel: soc-acpi-intel-cml-match: remove useless 'rt1308_2_adr' Jerome Brunet (2): ASoC: meson: axg-card: fix codec-to-codec link setup ASoC: meson: gx-card: fix codec-to-codec link setup Johan Jonker (4): ASoC: convert rockchip spdif bindings to yaml ASoC: rockchip-spdif: add #sound-dai-cells property ASoC: rockchip-spdif: add power-domains property ASoC: rockchip-i2s: add power-domains property Krzysztof Kozlowski (1): ASoC: samsung: s3c24xx-i2s: Fix build after removal of DAI suspend/resume Mark Brown (3): Merge series "ASoC: rsnd: Fixes for multichannel HDMI audio output" from Matthias Blankertz <matthias.blankertz(a)cetitec.com>: Merge series "ASoC: meson: fix codec-to-codec link setup" from Jerome Brunet <jbrunet(a)baylibre.com>: Merge series "ASoC: rsnd: multi-SSI setup fixes" from Matthias Blankertz <matthias.blankertz(a)cetitec.com>: Matthias Blankertz (4): ASoC: rsnd: Fix parent SSI start/stop in multi-SSI mode ASoC: rsnd: Fix HDMI channel mapping for multi-SSI mode ASoC: rsnd: Don't treat master SSI in multi SSI setup as parent ASoC: rsnd: Fix "status check failed" spam for multi-SSI Olivier Moysan (1): ASoC: stm32: sai: fix sai probe Philipp Puschmann (1): ASoC: tas571x: disable regulators on failed probe Pierre-Louis Bossart (2): ASoC: soc-dai: revert all changes to DAI startup/shutdown sequence ASoC: SOF: Intel: add min/max channels for SSP on Baytrail/Broadwell Sebastian Fricke (1): soc/stm/stm32_sub_sai: Add missing '\n' in log messages Sebastian Reichel (1): ASoC: sgtl5000: Fix VAG power-on handling Shengjiu Wang (1): ASoC: wm8960: Fix wrong clock after suspend & resume Srinivas Kandagatla (1): ASoC: wsa881x: mark read_only_wordlength flag Stephan Gerhold (2): ASoC: soc-pcm: dpcm: Only allow playback/capture if supported ASoC: q6dsp6: q6afe-dai: add missing channels to MI2S DAIs YueHaibing (1): ASoC: wm89xx: Add missing dependency .../devicetree/bindings/sound/rockchip-i2s.yaml | 3 + .../devicetree/bindings/sound/rockchip-spdif.txt | 45 ------- .../devicetree/bindings/sound/rockchip-spdif.yaml | 101 ++++++++++++++ include/sound/soc-dai.h | 1 - include/sound/soc.h | 3 + sound/soc/amd/acp3x-rt5682-max9836.c | 6 +- sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/hdac_hdmi.c | 6 +- sound/soc/codecs/madera.c | 4 - sound/soc/codecs/sgtl5000.c | 34 +++++ sound/soc/codecs/sgtl5000.h | 1 + sound/soc/codecs/tas571x.c | 20 ++- sound/soc/codecs/wm8960.c | 3 +- sound/soc/codecs/wsa881x.c | 4 + sound/soc/intel/common/soc-acpi-intel-cml-match.c | 8 -- sound/soc/intel/common/soc-acpi-intel-icl-match.c | 8 -- sound/soc/meson/axg-card.c | 4 +- sound/soc/meson/gx-card.c | 4 +- sound/soc/qcom/apq8096.c | 4 +- sound/soc/qcom/qdsp6/q6afe-dai.c | 16 +++ sound/soc/qcom/sdm845.c | 4 +- sound/soc/samsung/s3c-i2s-v2.c | 57 -------- sound/soc/samsung/s3c2412-i2s.c | 56 ++++++++ sound/soc/sh/rcar/ssi.c | 11 +- sound/soc/sh/rcar/ssiu.c | 2 +- sound/soc/soc-dai.c | 11 +- sound/soc/soc-dapm.c | 147 +++++++++++---------- sound/soc/soc-pcm.c | 13 +- sound/soc/soc-topology.c | 115 ++++++++++++---- sound/soc/sof/intel/bdw.c | 16 +++ sound/soc/sof/intel/byt.c | 48 +++++++ sound/soc/stm/stm32_sai_sub.c | 14 +- 32 files changed, 509 insertions(+), 263 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/rockchip-spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip-spdif.yaml

2 1

[PATCH v2] ASoC: intel/skl/hda - fix oops on systems without i915 audio codec
by Kai Vehmanen 21 Apr '20

21 Apr '20

Recent fix for jack detection caused a regression on systems with HDA audio codec but no HDMI/DP audio via i915 graphics, leading to a kernel oops at device probe. On these systems, HDA bus instance lookup fails, as the first ASoC runtime of the card is connected to a dummy codec (as no HDMI codec is present). Fixes: 3a24f135e6cc ("ASoC: intel/skl/hda - set autosuspend timeout for hda codecs") Signed-off-by: Kai Vehmanen <kai.vehmanen(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/intel/boards/skl_hda_dsp_generic.c | 34 +++++++++++--------- 1 file changed, 18 insertions(+), 16 deletions(-) diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index 35f222ed9912..fc9290fb4e99 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -172,23 +172,25 @@ static int skl_hda_fill_card_info(struct snd_soc_acpi_mach_params *mach_params) static void skl_set_hda_codec_autosuspend_delay(struct snd_soc_card *card) { - struct snd_soc_pcm_runtime *rtd = - list_first_entry(&card->rtd_list, - struct snd_soc_pcm_runtime, list); - struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_pcm_runtime *rtd; struct hdac_hda_priv *hda_pvt; - - if (!codec_dai) - return; - - /* - * all codecs are on the same bus, so it's sufficient - * to lookup the first runtime and its codec, and set - * power save defaults for all codecs on the bus - */ - hda_pvt = snd_soc_component_get_drvdata(codec_dai->component); - snd_hda_set_power_save(hda_pvt->codec.bus, - HDA_CODEC_AUTOSUSPEND_DELAY_MS); + struct snd_soc_dai *dai; + + for_each_card_rtds(card, rtd) { + if (!strstr(rtd->dai_link->codecs->name, "ehdaudio")) + continue; + dai = asoc_rtd_to_codec(rtd, 0); + hda_pvt = snd_soc_component_get_drvdata(dai->component); + if (hda_pvt) { + /* + * all codecs are on the same bus, so it's sufficient + * to look up only the first one + */ + snd_hda_set_power_save(hda_pvt->codec.bus, + HDA_CODEC_AUTOSUSPEND_DELAY_MS); + break; + } + } } static int skl_hda_audio_probe(struct platform_device *pdev) -- 2.26.0

3 4

[PATCH] ASoC: intel/skl/hda - fix oops on systems without i915 audio codec
by Kai Vehmanen 21 Apr '20

21 Apr '20

Recent fix for jack detection caused a regression on systems with HDA audio codec but no HDMI/DP audio via i915 graphics, leading to a kernel oops at device probe. On these systems, HDA bus instance lookup fails, as the first ASoC runtime of the card is connected to a dummy codec (as no HDMI codec is present). Fixes: 3a24f135e6cc ("ASoC: intel/skl/hda - set autosuspend timeout for hda codecs") Signed-off-by: Kai Vehmanen <kai.vehmanen(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/intel/boards/skl_hda_dsp_generic.c | 34 +++++++++++--------- 1 file changed, 18 insertions(+), 16 deletions(-) diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index 35f222ed9912..fc9290fb4e99 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -172,23 +172,25 @@ static int skl_hda_fill_card_info(struct snd_soc_acpi_mach_params *mach_params) static void skl_set_hda_codec_autosuspend_delay(struct snd_soc_card *card) { - struct snd_soc_pcm_runtime *rtd = - list_first_entry(&card->rtd_list, - struct snd_soc_pcm_runtime, list); - struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_pcm_runtime *rtd; struct hdac_hda_priv *hda_pvt; - - if (!codec_dai) - return; - - /* - * all codecs are on the same bus, so it's sufficient - * to lookup the first runtime and its codec, and set - * power save defaults for all codecs on the bus - */ - hda_pvt = snd_soc_component_get_drvdata(codec_dai->component); - snd_hda_set_power_save(hda_pvt->codec.bus, - HDA_CODEC_AUTOSUSPEND_DELAY_MS); + struct snd_soc_dai *dai; + + for_each_card_rtds(card, rtd) { + if (!strstr(rtd->dai_link->codecs->name, "ehdaudio")) + continue; + dai = asoc_rtd_to_codec(rtd, 0); + hda_pvt = snd_soc_component_get_drvdata(dai->component); + if (hda_pvt) { + /* + * all codecs are on the same bus, so it's sufficient + * to look up only the first one + */ + snd_hda_set_power_save(hda_pvt->codec.bus, + HDA_CODEC_AUTOSUSPEND_DELAY_MS); + break; + } + } } static int skl_hda_audio_probe(struct platform_device *pdev) -- 2.26.0

2 2

[PATCH] ASoC: txx9: add back the hack for a too small resource_size_t
by Christoph Hellwig 21 Apr '20

21 Apr '20

Looks like I misread the Kconfig magic and this driver can be compiled into 32-bit kernels. Add back the hack to extent the range of the resource_size_t, and include the header with the txx9-specific ioremap magic for that. Fixes: acfaaf52ebfd ("ASoC: txx9: don't work around too small resource_size_t") Signed-off-by: Christoph Hellwig <hch(a)lst.de> --- sound/soc/txx9/txx9aclc-ac97.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c index 7402448bdb09..d9e348444bd0 100644 --- a/sound/soc/txx9/txx9aclc-ac97.c +++ b/sound/soc/txx9/txx9aclc-ac97.c @@ -14,6 +14,7 @@ #include <linux/interrupt.h> #include <linux/io.h> #include <linux/gfp.h> +#include <asm/mach-tx39xx/ioremap.h> /* for TXX9_DIRECTMAP_BASE */ #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> @@ -175,8 +176,6 @@ static int txx9aclc_ac97_dev_probe(struct platform_device *pdev) int err; int irq; - BUILD_BUG_ON(sizeof(drvdata->physbase) > sizeof(r->start)); - irq = platform_get_irq(pdev, 0); if (irq < 0) return irq; @@ -192,6 +191,10 @@ static int txx9aclc_ac97_dev_probe(struct platform_device *pdev) platform_set_drvdata(pdev, drvdata); drvdata->physbase = r->start; + if (sizeof(drvdata->physbase) > sizeof(r->start) && + r->start >= TXX9_DIRECTMAP_BASE && + r->start < TXX9_DIRECTMAP_BASE + 0x400000) + drvdata->physbase |= 0xf00000000ull; err = devm_request_irq(&pdev->dev, irq, txx9aclc_ac97_irq, 0, dev_name(&pdev->dev), drvdata); if (err < 0) -- 2.26.1

3 2

[PATCH v2 0/8] ASoC: add snd_compress_ops and replace
by Kuninori Morimoto 21 Apr '20

21 Apr '20

Hi Mark Current snd_soc_component_driver has compr_ops, but it is created for ALSA, not for ALSA SoC. This means it doesn't know the callback is for which component, thus, each callback needs to get component via lookup function. It is not good design, and will not work in the future, because ASoC can have multipul same name component which current lookup function can't handle. This v2 patch-set adds new snd_compress_ops and call callbacks with component. v1 is here Link: https://lore.kernel.org/alsa-devel/87blnqpuqp.wl-kuninori.morimoto.gx@renes… Kuninori Morimoto (8): ASoC: soc-compress: add snd_compress_ops ASoC: codec: wm_adsp: use snd_compress_ops ASoC: uniphier: use snd_compress_ops ASoC: qcom: q6sp6: use snd_compress_ops ASoC: intel: atom: use snd_compress_ops ASoC: sof: use snd_compress_ops ASoC: sprd: use snd_compress_ops ASoC: soc-compress: remove snd_compr_ops include/sound/soc-component.h | 40 ++++++++- sound/soc/codecs/cs47l15.c | 9 +- sound/soc/codecs/cs47l24.c | 8 +- sound/soc/codecs/cs47l35.c | 9 +- sound/soc/codecs/cs47l85.c | 9 +- sound/soc/codecs/cs47l90.c | 9 +- sound/soc/codecs/cs47l92.c | 9 +- sound/soc/codecs/wm5102.c | 9 +- sound/soc/codecs/wm5110.c | 8 +- sound/soc/codecs/wm_adsp.c | 18 ++-- sound/soc/codecs/wm_adsp.h | 18 ++-- .../intel/atom/sst-mfld-platform-compress.c | 43 +++++---- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- sound/soc/intel/atom/sst-mfld-platform.h | 2 +- sound/soc/qcom/qdsp6/q6asm-dai.c | 51 ++++++----- sound/soc/soc-compress.c | 89 ++++++++++--------- sound/soc/sof/compress.c | 7 +- sound/soc/sof/compress.h | 7 +- sound/soc/sof/pcm.c | 4 +- sound/soc/sof/sof-priv.h | 2 +- sound/soc/sprd/sprd-pcm-compress.c | 49 +++++----- sound/soc/sprd/sprd-pcm-dma.c | 2 +- sound/soc/sprd/sprd-pcm-dma.h | 2 +- sound/soc/uniphier/aio-compress.c | 45 ++++++---- sound/soc/uniphier/aio-dma.c | 2 +- sound/soc/uniphier/aio.h | 2 +- 26 files changed, 265 insertions(+), 190 deletions(-) -- 2.17.1

4 13

[PATCH v9 0/7] ASoC: Add new module driver for new ASRC
by Shengjiu Wang 21 Apr '20

21 Apr '20

Add new module driver for new ASRC in i.MX8MN, several commits are added for new property fsl,asrc-format Shengjiu Wang (7): ASoC: fsl_asrc: rename asrc_priv to asrc ASoC: dt-bindings: fsl_asrc: Add new property fsl,asrc-format ASoC: fsl-asoc-card: Support new property fsl,asrc-format ASoC: fsl_asrc: Support new property fsl,asrc-format ASoC: fsl_asrc: Move common definition to fsl_asrc_common ASoC: dt-bindings: fsl_easrc: Add document for EASRC ASoC: fsl_easrc: Add EASRC ASoC CPU DAI drivers changes in v9 - use lowercase for dt example. changes in v8 - change get_pair_priv_size to pair_priv_size variable changes in v7 - updated according to Nicoin's comments. - add get_pair_priv_size to replace PAIR_PRIVATE_SIZE changes in v6 - updated according to Nicoin's and Rob's comments. changes in v5 - Add new property fsl,asrc-format, rather than change fsl,asrc-width to fsl,asrc-formt. - code change for above change. changes in v4 - Add several commit for changing DT binding asrc-width to asrc-format - rename asrc_priv to asrc changes in v3 - add new commit "ASoC: fsl_asrc: Change asrc_width to asrc_format" - modify binding doc to yaml format - remove fsl_easrc_dma.c, make fsl_asrc_dma.c useable for easrc. changes in v2 - change i.MX815 to i.MX8MN - Add changes in Kconfig and Makefile .../devicetree/bindings/sound/fsl,asrc.txt | 4 + .../devicetree/bindings/sound/fsl,easrc.yaml | 101 + sound/soc/fsl/Kconfig | 11 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl-asoc-card.c | 24 +- sound/soc/fsl/fsl_asrc.c | 305 +-- sound/soc/fsl/fsl_asrc.h | 74 +- sound/soc/fsl/fsl_asrc_common.h | 106 + sound/soc/fsl/fsl_asrc_dma.c | 54 +- sound/soc/fsl/fsl_easrc.c | 2114 +++++++++++++++++ sound/soc/fsl/fsl_easrc.h | 651 +++++ 11 files changed, 3212 insertions(+), 234 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,easrc.yaml create mode 100644 sound/soc/fsl/fsl_asrc_common.h create mode 100644 sound/soc/fsl/fsl_easrc.c create mode 100644 sound/soc/fsl/fsl_easrc.h -- 2.21.0

2 8

[PATCH v2 1/2] dt-bindings: sound: add Microsemi ZL38060 binding
by Sven Van Asbroeck 21 Apr '20

21 Apr '20

Add devicetree binding for the Microsemi ZL38060 Connected Home Audio Processor. Signed-off-by: Sven Van Asbroeck <TheSven73(a)gmail.com> --- Tree: v5.7-rc1 v1->v2: - removed all properties related to amp-en-gpios - removed 'mscc,load-firmware' property For further details, see v1->v2 notes in 2nd patch. .../devicetree/bindings/sound/zl38060.yaml | 69 +++++++++++++++++++ 1 file changed, 69 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/zl38060.yaml diff --git a/Documentation/devicetree/bindings/sound/zl38060.yaml b/Documentation/devicetree/bindings/sound/zl38060.yaml new file mode 100644 index 000000000000..338e2a13c775 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/zl38060.yaml @@ -0,0 +1,69 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/zl38060.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: ZL38060 Connected Home Audio Processor from Microsemi. + +description: | + The ZL38060 is a "Connected Home Audio Processor" from Microsemi, + which consists of a Digital Signal Processor (DSP), several Digital + Audio Interfaces (DAIs), analog outputs, and a block of 14 GPIOs. + +maintainers: + - Jaroslav Kysela <perex(a)perex.cz> + - Takashi Iwai <tiwai(a)suse.com> + +properties: + compatible: + const: mscc,zl38060 + + reg: + description: + SPI device address. + maxItems: 1 + + spi-max-frequency: + maximum: 24000000 + + reset-gpios: + description: + A GPIO line handling reset of the chip. As the line is active low, + it should be marked GPIO_ACTIVE_LOW (see ../gpio/gpio.txt) + maxItems: 1 + + '#gpio-cells': + const: 2 + + gpio-controller: true + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + - '#gpio-cells' + - gpio-controller + - '#sound-dai-cells' + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + spi0 { + #address-cells = <1>; + #size-cells = <0>; + + codec: zl38060@0 { + gpio-controller; + #gpio-cells = <2>; + #sound-dai-cells = <0>; + compatible = "mscc,zl38060"; + reg = <0>; + spi-max-frequency = <12000000>; + reset-gpios = <&gpio1 0 GPIO_ACTIVE_LOW>; + }; + }; -- 2.17.1

2 7

[PATCH v8 0/7] ASoC: Add new module driver for new ASRC
by Shengjiu Wang 21 Apr '20

21 Apr '20

Add new module driver for new ASRC in i.MX8MN, several commits are added for new property fsl,asrc-format Shengjiu Wang (7): ASoC: fsl_asrc: rename asrc_priv to asrc ASoC: dt-bindings: fsl_asrc: Add new property fsl,asrc-format ASoC: fsl-asoc-card: Support new property fsl,asrc-format ASoC: fsl_asrc: Support new property fsl,asrc-format ASoC: fsl_asrc: Move common definition to fsl_asrc_common ASoC: dt-bindings: fsl_easrc: Add document for EASRC ASoC: fsl_easrc: Add EASRC ASoC CPU DAI drivers changes in v8 - change get_pair_priv_size to pair_priv_size variable changes in v7 - updated according to Nicoin's comments. - add get_pair_priv_size to replace PAIR_PRIVATE_SIZE changes in v6 - updated according to Nicoin's and Rob's comments. changes in v5 - Add new property fsl,asrc-format, rather than change fsl,asrc-width to fsl,asrc-formt. - code change for above change. changes in v4 - Add several commit for changing DT binding asrc-width to asrc-format - rename asrc_priv to asrc changes in v3 - add new commit "ASoC: fsl_asrc: Change asrc_width to asrc_format" - modify binding doc to yaml format - remove fsl_easrc_dma.c, make fsl_asrc_dma.c useable for easrc. changes in v2 - change i.MX815 to i.MX8MN - Add changes in Kconfig and Makefile .../devicetree/bindings/sound/fsl,asrc.txt | 4 + .../devicetree/bindings/sound/fsl,easrc.yaml | 101 + sound/soc/fsl/Kconfig | 11 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl-asoc-card.c | 24 +- sound/soc/fsl/fsl_asrc.c | 305 +-- sound/soc/fsl/fsl_asrc.h | 74 +- sound/soc/fsl/fsl_asrc_common.h | 106 + sound/soc/fsl/fsl_asrc_dma.c | 54 +- sound/soc/fsl/fsl_easrc.c | 2114 +++++++++++++++++ sound/soc/fsl/fsl_easrc.h | 651 +++++ 11 files changed, 3212 insertions(+), 234 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,easrc.yaml create mode 100644 sound/soc/fsl/fsl_asrc_common.h create mode 100644 sound/soc/fsl/fsl_easrc.c create mode 100644 sound/soc/fsl/fsl_easrc.h -- 2.21.0

3 9

[PATCH] ASoC: wm8960: Fix wrong clock after suspend & resume
by Shengjiu Wang 21 Apr '20

21 Apr '20

After suspend & resume, wm8960_hw_params may be called when bias_level is not SND_SOC_BIAS_ON, then wm8960_configure_clocking is not called. But if sample rate is changed at that time, then the output clock rate will be not correct. So judgement of bias_level is SND_SOC_BIAS_ON in wm8960_hw_params is not necessary and it causes above issue. Fixes: 3176bf2d7ccd ("ASoC: wm8960: update pll and clock setting function") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/wm8960.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 55112c1bba5e..6cf0f6612bda 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -860,8 +860,7 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, wm8960->is_stream_in_use[tx] = true; - if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_ON && - !wm8960->is_stream_in_use[!tx]) + if (!wm8960->is_stream_in_use[!tx]) return wm8960_configure_clocking(component); return 0; -- 2.21.0

3 2

[RFC PATCH v2 0/2] Introduce multi PM domains helpers
by Daniel Baluta 21 Apr '20

21 Apr '20

From: Daniel Baluta <daniel.baluta(a)nxp.com> i.MX8QXP/i.MX8QM has IPs that need multiple power domains to be up in order to work. In order to help drivers, we introduce multi PM domains helpers that are able to activate/deactivate multi PM domains. First patch introduces the helpers and second patch demonstrates how a driver can use them instead of hardcoding the PM domains handling. Changes since v1: (addressed Ranjani's comments) - enhanced description for dev_multi_pm_attach return value - renamed exit_unroll_pm label to exit_detach_pm Daniel Baluta (2): PM / domains: Introduce multi PM domains helpers ASoC: SOF: Use multi PM domains helpers drivers/base/power/common.c | 93 +++++++++++++++++++++++++++++++++++++ include/linux/pm_domain.h | 19 ++++++++ sound/soc/sof/imx/imx8.c | 60 ++++-------------------- 3 files changed, 121 insertions(+), 51 deletions(-) -- 2.17.1

4 6

[RFC PATCH 0/9] fix -Wempty-body build warnings
by Randy Dunlap 21 Apr '20

21 Apr '20

Hi, When -Wextra is used, gcc emits many warnings about an empty 'if' or 'else' body, like this: ../fs/posix_acl.c: In function ‘get_acl’: ../fs/posix_acl.c:127:22: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body] /* fall through */ ; ^ To quieten these warnings, add a new macro "do_empty()". I originally wanted to use do_nothing(), but that's already in use. It would sorta be nice if "fallthrough" could be coerced for this instead of using something like do_empty(). Or should we just use "{}" in place of ";"? This causes some odd coding style issue IMO. E.g., see this change: original: if (cmpxchg(p, ACL_NOT_CACHED, sentinel) != ACL_NOT_CACHED) /* fall through */ ; with new macro: if (cmpxchg(p, ACL_NOT_CACHED, sentinel) != ACL_NOT_CACHED) do_empty(); /* fall through */ using {}: if (cmpxchg(p, ACL_NOT_CACHED, sentinel) != ACL_NOT_CACHED) {} /* fall through */ or { /* fall through */ } or even if (cmpxchg(p, ACL_NOT_CACHED, sentinel) != ACL_NOT_CACHED) { /* fall through */ } or if (cmpxchg(p, ACL_NOT_CACHED, sentinel) != ACL_NOT_CACHED) { } /* fall through */ drivers/base/devcoredump.c | 5 +++-- drivers/dax/bus.c | 5 +++-- drivers/input/mouse/synaptics.c | 3 ++- drivers/target/target_core_pscsi.c | 3 ++- drivers/usb/core/sysfs.c | 2 +- fs/nfsd/nfs4state.c | 3 ++- fs/posix_acl.c | 2 +- include/linux/kernel.h | 8 ++++++++ sound/drivers/vx/vx_core.c | 3 ++- 9 files changed, 24 insertions(+), 10 deletions(-)

14 38

[PATCH] ASoC: wm8962: restore the CLOCKING2 register in resume
by Shengjiu Wang 21 Apr '20

21 Apr '20

The CLOCKING2 is a volatile register, but some bits should be restored when resume, for example SYSCLK_SRC. otherwise the output clock is wrong Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/wm8962.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index d9d59f45833f..6e96c0c5ad2a 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -82,6 +82,7 @@ struct wm8962_priv { #endif int irq; + u32 regcache_clocking2; }; /* We can't use the same notifier block for more than one supply and @@ -3813,6 +3814,10 @@ static int wm8962_runtime_resume(struct device *dev) regcache_sync(wm8962->regmap); + regmap_update_bits(wm8962->regmap, WM8962_CLOCKING2, + WM8962_SYSCLK_SRC_MASK, + wm8962->regcache_clocking2); + regmap_update_bits(wm8962->regmap, WM8962_ANTI_POP, WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA, WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA); @@ -3842,6 +3847,9 @@ static int wm8962_runtime_suspend(struct device *dev) WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA, 0); + regmap_read(wm8962->regmap, WM8962_CLOCKING2, + &wm8962->regcache_clocking2); + regcache_cache_only(wm8962->regmap, true); regulator_bulk_disable(ARRAY_SIZE(wm8962->supplies), -- 2.21.0

2 1

[PATCH 1/2] ASoC: SOF: Fix "no reply expected" error during firmware-boot
by Hans de Goede 21 Apr '20

21 Apr '20

At least on Canon Lake each time the SOF firmware is booted, the following error is logged in dmesg: [ 36.711803] sof-audio-pci 0000:00:1f.3: error: no reply expected, received 0x0 Since the DSP is powered down when not in used this happens everytime e.g. a notification plays, polluting dmesg with these false-positive errors. Add a check to snd_sof_ipc_reply() which makes it return success when receiving an unexpected msg_id 0 during fw-boot, fixing these false-positive errors being logged. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/sof/ipc.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/sof/ipc.c b/sound/soc/sof/ipc.c index 78aa1da7c7a9..7303b3d42f12 100644 --- a/sound/soc/sof/ipc.c +++ b/sound/soc/sof/ipc.c @@ -312,6 +312,11 @@ int snd_sof_ipc_reply(struct snd_sof_dev *sdev, u32 msg_id) { struct snd_sof_ipc_msg *msg = &sdev->ipc->msg; + /* Ignore msg_id 0 being send during fw-boot */ + if (msg->ipc_complete && sdev->fw_state == SOF_FW_BOOT_IN_PROGRESS && + msg_id == 0) + return 0; + if (msg->ipc_complete) { dev_err(sdev->dev, "error: no reply expected, received 0x%x", msg_id); -- 2.26.0

4 11

Add support to SDM845 based platforms
by GitHub pull_request - opened 21 Apr '20

21 Apr '20

alsa-project/alsa-ucm-conf pull request #22 was opened from Srinivas-Kandagatla: Add Support to two codecs WCD934x and WSA881x and two platforms DragonBoard DB845c and Lenovo-YOGA-C630-13Q50 that use this codec. For ucm-validator to pass this PR depends on config files PR "ucm-validator: add config files for DB845c and Lenovo-YOGA-C630-13Q50" to alsa-tests Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/22 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/22.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: fsl_esai: Remove expensive print in irq handler
by Shengjiu Wang 21 Apr '20

21 Apr '20

Use dev_dbg instead of dev_err in irq handler, the dev_err is expensive, we don't need the message to be printed everytime, which is almost a debug option. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_esai.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index c7a49d03463a..453392a9cd68 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -94,10 +94,10 @@ static irqreturn_t esai_isr(int irq, void *devid) dev_dbg(&pdev->dev, "isr: Transmission Initialized\n"); if (esr & ESAI_ESR_RFF_MASK) - dev_warn(&pdev->dev, "isr: Receiving overrun\n"); + dev_dbg(&pdev->dev, "isr: Receiving overrun\n"); if (esr & ESAI_ESR_TFE_MASK) - dev_warn(&pdev->dev, "isr: Transmission underrun\n"); + dev_dbg(&pdev->dev, "isr: Transmission underrun\n"); if (esr & ESAI_ESR_TLS_MASK) dev_dbg(&pdev->dev, "isr: Just transmitted the last slot\n"); -- 2.21.0

3 2

[PATCH v2 0/7] Add H6 I2S support
by Clément Péron 21 Apr '20

21 Apr '20

Hi, This is a sequel of Marcus Cooper serie[0], where remarks made by Maxime have been fixed. I have tested it on my Beelink GS1 board. Thanks, Clement 0: https://lore.kernel.org/patchwork/cover/1139949/ Changes since v1: - Fix missing header in set sign extend sample Jernej Skrabec (3): dt-bindings: ASoC: sun4i-i2s: Add H6 compatible ASoC: sun4i-i2s: Add support for H6 I2S arm64: dts: sun50i-h6: Add HDMI audio to H6 DTSI Marcus Cooper (4): ASoC: sun4i-i2s: Adjust LRCLK width ASoC: sun4i-i2s: Set sign extend sample ASoc: sun4i-i2s: Add 20 and 24 bit support ASoC: sun4i-i2s: Adjust regmap settings .../sound/allwinner,sun4i-a10-i2s.yaml | 2 + arch/arm64/boot/dts/allwinner/sun50i-h6.dtsi | 31 ++ sound/soc/sunxi/sun4i-i2s.c | 292 ++++++++++++++++-- 3 files changed, 301 insertions(+), 24 deletions(-) -- 2.20.1

2 11

Think I found an ALSA bug
by Ólafur Víðir Guðbjargarson 21 Apr '20

21 Apr '20

Dear receiver I think I found an ALSA bug. Yesterday my computer, which is running Manjaro 19.0.2 GNOME edition froze. After rebooting I checked the system log and found a message there to report to ALSA developers. This is just few lines from the log: apr 20 22:45:20 HP pulseaudio[1458]: E: [alsa-sink-ALC295 Analog] alsa- util.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. apr 20 22:45:20 HP pulseaudio[1458]: E: [alsa-sink-ALC295 Analog] alsa- util.c: snd_pcm_avail() returned a value that is exceptionally large: 2271932 bytes (12879 ms). I can't wrap this into code block here. I have attached the log as txt file. If you need further information, I will do my best to provide them. -- Kær kveðja/best regards Ólafur Víðir Guðbjargarson Tæknifræðingur/engineer technical GSM: 692 6292 olividir(a)gmail.com

1 0

[PATCH] ALSA: usb-audio: Add quirk for Focusrite Scarlett 2i2
by Gregor Pintar 21 Apr '20

21 Apr '20

Force it to use asynchronous playback. Same quirk has already been added for Focusrite Scarlett Solo (2nd gen) with a commit 46f5710f0b88 ("ALSA: usb-audio: Add quirk for Focusrite Scarlett Solo"). This also seems to prevent regular clicks when playing at 44100Hz on Scarlett 2i2 (2nd gen). I did not notice any side effects. Signed-off-by: Gregor Pintar <grpintar(a)gmail.com> --- sound/usb/quirks-table.h | 84 ++++++++++++++++++++++++++++++++++++++++ 1 file changed, 84 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index e009d584e..bc936bf79 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2840,6 +2840,90 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + /* + * Focusrite Scarlett 2i2 2nd generation + * Reports that playback should use Synch: Synchronous + * while still providing a feedback endpoint. Synchronous causes + * snapping on some sample rates. + * Force it to use Synch: Asynchronous. + */ + USB_DEVICE(0x1235, 0x8202), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 1, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels = 2, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .protocol = UAC_VERSION_2, + .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, + .rate_min = 44100, + .rate_max = 192000, + .nr_rates = 6, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, + 96000, 176400, 192000 + }, + .clock = 41 + } + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels = 2, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0, + .endpoint = 0x82, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC | + USB_ENDPOINT_USAGE_IMPLICIT_FB, + .protocol = UAC_VERSION_2, + .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, + .rate_min = 44100, + .rate_max = 192000, + .nr_rates = 6, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, + 96000, 176400, 192000 + }, + .clock = 41 + } + }, + { + .ifnum = 3, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, /* Access Music devices */ { -- 2.20.1

3 4

[PATCH 1/2] dt-bindings: ASoC: qdsp6: Suggest more generic node names
by Stephan Gerhold 21 Apr '20

21 Apr '20

Change the listed examples to use more generic node names, representing the class of the device nodes: - apr-service@<id> - dai@<id> Both names are already in use in arch/arm64/boot/dts/qcom/sdm845.dtsi. Also add #address-cells + #size-cells to the q6asm example, without them the example produces dtc warnings. Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- .../devicetree/bindings/sound/qcom,q6adm.txt | 2 +- .../devicetree/bindings/sound/qcom,q6afe.txt | 46 +++++++++---------- .../devicetree/bindings/sound/qcom,q6asm.txt | 7 ++- .../devicetree/bindings/sound/qcom,q6core.txt | 2 +- 4 files changed, 30 insertions(+), 27 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,q6adm.txt b/Documentation/devicetree/bindings/sound/qcom,q6adm.txt index bbae426cdfb1..15c353a20de8 100644 --- a/Documentation/devicetree/bindings/sound/qcom,q6adm.txt +++ b/Documentation/devicetree/bindings/sound/qcom,q6adm.txt @@ -29,7 +29,7 @@ used by the apr service device. Definition: Must be 0 = EXAMPLE -q6adm@8 { +apr-service@8 { compatible = "qcom,q6adm"; reg = <APR_SVC_ADM>; q6routing: routing { diff --git a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt index d74888b9f1bb..4916dd6a0896 100644 --- a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt +++ b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt @@ -100,7 +100,7 @@ configuration of each dai. Must contain the following properties. = EXAMPLE -q6afe@4 { +apr-service@4 { compatible = "qcom,q6afe"; reg = <APR_SVC_AFE>; @@ -110,12 +110,12 @@ q6afe@4 { #address-cells = <1>; #size-cells = <0>; - hdmi@1 { - reg = <1>; + dai@1 { + reg = <HDMI_RX>; }; - tdm@24 { - reg = <24>; + dai@24 { + reg = <PRIMARY_TDM_RX_0>; qcom,tdm-sync-mode = <1>: qcom,tdm-sync-src = <1>; qcom,tdm-data-out = <0>; @@ -125,8 +125,8 @@ q6afe@4 { }; - tdm@25 { - reg = <25>; + dai@25 { + reg = <PRIMARY_TDM_TX_0>; qcom,tdm-sync-mode = <1>: qcom,tdm-sync-src = <1>; qcom,tdm-data-out = <0>; @@ -135,43 +135,43 @@ q6afe@4 { qcom,tdm-data-align = <0>; }; - prim-mi2s-rx@16 { - reg = <16>; + dai@16 { + reg = <PRIMARY_MI2S_RX>; qcom,sd-lines = <0 2>; }; - prim-mi2s-tx@17 { - reg = <17>; + dai@17 { + reg = <PRIMARY_MI2S_TX>; qcom,sd-lines = <1>; }; - sec-mi2s-rx@18 { - reg = <18>; + dai@18 { + reg = <SECONDARY_MI2S_RX>; qcom,sd-lines = <0 3>; }; - sec-mi2s-tx@19 { - reg = <19>; + dai@19 { + reg = <SECONDARY_MI2S_TX>; qcom,sd-lines = <1>; }; - tert-mi2s-rx@20 { - reg = <20>; + dai@20 { + reg = <TERTIARY_MI2S_RX>; qcom,sd-lines = <1 3>; }; - tert-mi2s-tx@21 { - reg = <21>; + dai@21 { + reg = <TERTIARY_MI2S_TX>; qcom,sd-lines = <0>; }; - quat-mi2s-rx@22 { - reg = <22>; + dai@22 { + reg = <QUATERNARY_MI2S_RX>; qcom,sd-lines = <0>; }; - quat-mi2s-tx@23 { - reg = <23>; + dai@23 { + reg = <QUATERNARY_MI2S_TX>; qcom,sd-lines = <1>; }; }; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6asm.txt b/Documentation/devicetree/bindings/sound/qcom,q6asm.txt index 9f5378c51686..6b9a88d0ea3f 100644 --- a/Documentation/devicetree/bindings/sound/qcom,q6asm.txt +++ b/Documentation/devicetree/bindings/sound/qcom,q6asm.txt @@ -51,13 +51,16 @@ configuration of each dai. Must contain the following properties. = EXAMPLE -q6asm@7 { +apr-service@7 { compatible = "qcom,q6asm"; reg = <APR_SVC_ASM>; q6asmdai: dais { compatible = "qcom,q6asm-dais"; + #address-cells = <1>; + #size-cells = <0>; #sound-dai-cells = <1>; - mm@0 { + + dai@0 { reg = <0>; direction = <2>; is-compress-dai; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6core.txt b/Documentation/devicetree/bindings/sound/qcom,q6core.txt index 7f36ff8bec18..5cd4cc9b1fde 100644 --- a/Documentation/devicetree/bindings/sound/qcom,q6core.txt +++ b/Documentation/devicetree/bindings/sound/qcom,q6core.txt @@ -15,7 +15,7 @@ used by the apr service device. example "qcom,q6core-v2.0" = EXAMPLE -q6core@3 { +apr-service@3 { compatible = "qcom,q6core"; reg = <APR_SVC_ADSP_CORE>; }; -- 2.26.1

5 7

[PATCH 1/2] dt-bindings: Clean-up schema indentation formatting
by Rob Herring 20 Apr '20

20 Apr '20

Fix various inconsistencies in schema indentation. Most of these are list indentation which should be 2 spaces more than the start of the enclosing keyword. This doesn't matter functionally, but affects running scripts which do transforms on the schema files. Signed-off-by: Rob Herring <robh(a)kernel.org> --- .../devicetree/bindings/arm/altera.yaml | 6 +- .../amlogic/amlogic,meson-gx-ao-secure.yaml | 2 +- .../devicetree/bindings/arm/bitmain.yaml | 2 +- .../devicetree/bindings/arm/nxp/lpc32xx.yaml | 9 ++- .../bindings/arm/socionext/uniphier.yaml | 26 ++++---- .../bindings/arm/stm32/st,mlahb.yaml | 2 +- .../bindings/arm/stm32/st,stm32-syscon.yaml | 6 +- .../bindings/ata/faraday,ftide010.yaml | 4 +- .../bindings/bus/allwinner,sun8i-a23-rsb.yaml | 4 +- .../clock/allwinner,sun4i-a10-gates-clk.yaml | 8 +-- .../devicetree/bindings/clock/fsl,plldig.yaml | 17 +++-- .../devicetree/bindings/clock/qcom,mmcc.yaml | 16 ++--- .../bindings/connector/usb-connector.yaml | 6 +- .../crypto/allwinner,sun4i-a10-crypto.yaml | 14 ++-- .../bindings/crypto/allwinner,sun8i-ce.yaml | 16 ++--- .../bindings/crypto/amlogic,gxl-crypto.yaml | 2 +- .../display/allwinner,sun4i-a10-hdmi.yaml | 40 ++++++------ .../display/allwinner,sun4i-a10-tcon.yaml | 58 ++++++++--------- .../display/allwinner,sun6i-a31-mipi-dsi.yaml | 28 ++++---- .../display/allwinner,sun8i-a83t-dw-hdmi.yaml | 10 +-- .../bindings/display/bridge/lvds-codec.yaml | 18 +++--- .../display/panel/sony,acx424akp.yaml | 2 +- .../display/panel/xinpeng,xpp055c272.yaml | 4 +- .../bindings/display/renesas,cmm.yaml | 16 ++--- .../devicetree/bindings/dma/ti/k3-udma.yaml | 8 +-- .../bindings/gpio/brcm,xgs-iproc-gpio.yaml | 2 +- .../bindings/gpu/arm,mali-midgard.yaml | 18 +++--- .../devicetree/bindings/gpu/vivante,gc.yaml | 2 +- .../devicetree/bindings/i2c/i2c-rk3x.yaml | 10 +-- .../bindings/iio/adc/adi,ad7124.yaml | 4 +- .../bindings/iio/adc/lltc,ltc2496.yaml | 6 +- .../input/allwinner,sun4i-a10-lradc-keys.yaml | 4 +- .../bindings/input/touchscreen/goodix.yaml | 2 +- .../bindings/interconnect/qcom,msm8916.yaml | 4 +- .../bindings/interconnect/qcom,msm8974.yaml | 4 +- .../bindings/interconnect/qcom,qcs404.yaml | 4 +- .../allwinner,sun7i-a20-sc-nmi.yaml | 12 ++-- .../intel,ixp4xx-interrupt.yaml | 8 +-- .../interrupt-controller/st,stm32-exti.yaml | 12 ++-- .../bindings/iommu/samsung,sysmmu.yaml | 10 +-- .../bindings/mailbox/st,stm32-ipcc.yaml | 2 +- .../media/allwinner,sun4i-a10-csi.yaml | 28 ++++---- .../bindings/media/amlogic,gx-vdec.yaml | 14 ++-- .../bindings/media/renesas,ceu.yaml | 28 ++++---- .../bindings/media/renesas,vin.yaml | 8 +-- .../devicetree/bindings/media/ti,vpe.yaml | 2 +- .../memory-controllers/fsl/imx8m-ddrc.yaml | 6 +- .../bindings/mfd/st,stm32-lptimer.yaml | 4 +- .../bindings/mfd/st,stm32-timers.yaml | 4 +- .../devicetree/bindings/mfd/syscon.yaml | 12 ++-- .../devicetree/bindings/mmc/cdns,sdhci.yaml | 2 +- .../bindings/mmc/rockchip-dw-mshc.yaml | 16 ++--- .../bindings/mmc/socionext,uniphier-sd.yaml | 14 ++-- .../devicetree/bindings/mtd/denali,nand.yaml | 4 +- .../net/allwinner,sun8i-a83t-emac.yaml | 4 +- .../bindings/net/can/bosch,m_can.yaml | 52 +++++++-------- .../bindings/net/renesas,ether.yaml | 4 +- .../bindings/net/ti,cpsw-switch.yaml | 12 ++-- .../bindings/net/ti,davinci-mdio.yaml | 27 ++++---- .../bindings/phy/intel,lgm-emmc-phy.yaml | 2 +- .../devicetree/bindings/pwm/pwm-samsung.yaml | 16 ++--- .../bindings/remoteproc/st,stm32-rproc.yaml | 2 +- .../reset/brcm,bcm7216-pcie-sata-rescal.yaml | 4 +- .../devicetree/bindings/rtc/st,stm32-rtc.yaml | 38 +++++------ .../bindings/serial/amlogic,meson-uart.yaml | 16 ++--- .../devicetree/bindings/serial/rs485.yaml | 17 ++--- .../bindings/soc/amlogic/amlogic,canvas.yaml | 10 +-- .../bindings/sound/renesas,fsi.yaml | 16 ++--- .../bindings/spi/qcom,spi-qcom-qspi.yaml | 10 +-- .../devicetree/bindings/spi/renesas,hspi.yaml | 4 +- .../devicetree/bindings/spi/spi-pl022.yaml | 2 +- .../bindings/spi/st,stm32-qspi.yaml | 4 +- .../allwinner,sun4i-a10-system-control.yaml | 64 +++++++++---------- .../bindings/thermal/amlogic,thermal.yaml | 10 +-- .../bindings/timer/arm,arch_timer.yaml | 4 +- .../bindings/timer/arm,arch_timer_mmio.yaml | 4 +- .../devicetree/bindings/usb/dwc2.yaml | 8 +-- 77 files changed, 450 insertions(+), 450 deletions(-) diff --git a/Documentation/devicetree/bindings/arm/altera.yaml b/Documentation/devicetree/bindings/arm/altera.yaml index 49e0362ddc11..b388c5aa7984 100644 --- a/Documentation/devicetree/bindings/arm/altera.yaml +++ b/Documentation/devicetree/bindings/arm/altera.yaml @@ -13,8 +13,8 @@ properties: compatible: items: - enum: - - altr,socfpga-cyclone5 - - altr,socfpga-arria5 - - altr,socfpga-arria10 + - altr,socfpga-cyclone5 + - altr,socfpga-arria5 + - altr,socfpga-arria10 - const: altr,socfpga ... diff --git a/Documentation/devicetree/bindings/arm/amlogic/amlogic,meson-gx-ao-secure.yaml b/Documentation/devicetree/bindings/arm/amlogic/amlogic,meson-gx-ao-secure.yaml index 66213bd95e6e..6cc74523ebfd 100644 --- a/Documentation/devicetree/bindings/arm/amlogic/amlogic,meson-gx-ao-secure.yaml +++ b/Documentation/devicetree/bindings/arm/amlogic/amlogic,meson-gx-ao-secure.yaml @@ -25,7 +25,7 @@ select: properties: compatible: - items: + items: - const: amlogic,meson-gx-ao-secure - const: syscon diff --git a/Documentation/devicetree/bindings/arm/bitmain.yaml b/Documentation/devicetree/bindings/arm/bitmain.yaml index 0efdb4ac028e..5cd5b36cff2d 100644 --- a/Documentation/devicetree/bindings/arm/bitmain.yaml +++ b/Documentation/devicetree/bindings/arm/bitmain.yaml @@ -13,6 +13,6 @@ properties: compatible: items: - enum: - - bitmain,sophon-edge + - bitmain,sophon-edge - const: bitmain,bm1880 ... diff --git a/Documentation/devicetree/bindings/arm/nxp/lpc32xx.yaml b/Documentation/devicetree/bindings/arm/nxp/lpc32xx.yaml index 07f39d3eee7e..f7f024910e71 100644 --- a/Documentation/devicetree/bindings/arm/nxp/lpc32xx.yaml +++ b/Documentation/devicetree/bindings/arm/nxp/lpc32xx.yaml @@ -17,9 +17,8 @@ properties: - nxp,lpc3230 - nxp,lpc3240 - items: - - enum: - - ea,ea3250 - - phytec,phy3250 - - const: nxp,lpc3250 - + - enum: + - ea,ea3250 + - phytec,phy3250 + - const: nxp,lpc3250 ... diff --git a/Documentation/devicetree/bindings/arm/socionext/uniphier.yaml b/Documentation/devicetree/bindings/arm/socionext/uniphier.yaml index 65ad6d8a3c99..113f93b9ae55 100644 --- a/Documentation/devicetree/bindings/arm/socionext/uniphier.yaml +++ b/Documentation/devicetree/bindings/arm/socionext/uniphier.yaml @@ -17,45 +17,45 @@ properties: - description: LD4 SoC boards items: - enum: - - socionext,uniphier-ld4-ref + - socionext,uniphier-ld4-ref - const: socionext,uniphier-ld4 - description: Pro4 SoC boards items: - enum: - - socionext,uniphier-pro4-ace - - socionext,uniphier-pro4-ref - - socionext,uniphier-pro4-sanji + - socionext,uniphier-pro4-ace + - socionext,uniphier-pro4-ref + - socionext,uniphier-pro4-sanji - const: socionext,uniphier-pro4 - description: sLD8 SoC boards items: - enum: - - socionext,uniphier-sld8-ref + - socionext,uniphier-sld8-ref - const: socionext,uniphier-sld8 - description: PXs2 SoC boards items: - enum: - - socionext,uniphier-pxs2-gentil - - socionext,uniphier-pxs2-vodka + - socionext,uniphier-pxs2-gentil + - socionext,uniphier-pxs2-vodka - const: socionext,uniphier-pxs2 - description: LD6b SoC boards items: - enum: - - socionext,uniphier-ld6b-ref + - socionext,uniphier-ld6b-ref - const: socionext,uniphier-ld6b - description: LD11 SoC boards items: - enum: - - socionext,uniphier-ld11-global - - socionext,uniphier-ld11-ref + - socionext,uniphier-ld11-global + - socionext,uniphier-ld11-ref - const: socionext,uniphier-ld11 - description: LD20 SoC boards items: - enum: - - socionext,uniphier-ld20-global - - socionext,uniphier-ld20-ref + - socionext,uniphier-ld20-global + - socionext,uniphier-ld20-ref - const: socionext,uniphier-ld20 - description: PXs3 SoC boards items: - enum: - - socionext,uniphier-pxs3-ref + - socionext,uniphier-pxs3-ref - const: socionext,uniphier-pxs3 diff --git a/Documentation/devicetree/bindings/arm/stm32/st,mlahb.yaml b/Documentation/devicetree/bindings/arm/stm32/st,mlahb.yaml index 55f7938c4826..9f276bc9efa0 100644 --- a/Documentation/devicetree/bindings/arm/stm32/st,mlahb.yaml +++ b/Documentation/devicetree/bindings/arm/stm32/st,mlahb.yaml @@ -20,7 +20,7 @@ description: | [2]: https://wiki.st.com/stm32mpu/wiki/STM32MP15_RAM_mapping allOf: - - $ref: /schemas/simple-bus.yaml# + - $ref: /schemas/simple-bus.yaml# properties: compatible: diff --git a/Documentation/devicetree/bindings/arm/stm32/st,stm32-syscon.yaml b/Documentation/devicetree/bindings/arm/stm32/st,stm32-syscon.yaml index baff80197d5a..cf5db5e273f3 100644 --- a/Documentation/devicetree/bindings/arm/stm32/st,stm32-syscon.yaml +++ b/Documentation/devicetree/bindings/arm/stm32/st,stm32-syscon.yaml @@ -14,9 +14,9 @@ properties: compatible: oneOf: - items: - - enum: - - st,stm32mp157-syscfg - - const: syscon + - enum: + - st,stm32mp157-syscfg + - const: syscon reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/ata/faraday,ftide010.yaml b/Documentation/devicetree/bindings/ata/faraday,ftide010.yaml index bfc6357476fd..6451928dd2ce 100644 --- a/Documentation/devicetree/bindings/ata/faraday,ftide010.yaml +++ b/Documentation/devicetree/bindings/ata/faraday,ftide010.yaml @@ -26,8 +26,8 @@ properties: oneOf: - const: faraday,ftide010 - items: - - const: cortina,gemini-pata - - const: faraday,ftide010 + - const: cortina,gemini-pata + - const: faraday,ftide010 reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/bus/allwinner,sun8i-a23-rsb.yaml b/Documentation/devicetree/bindings/bus/allwinner,sun8i-a23-rsb.yaml index 80973619342d..32d33b983d66 100644 --- a/Documentation/devicetree/bindings/bus/allwinner,sun8i-a23-rsb.yaml +++ b/Documentation/devicetree/bindings/bus/allwinner,sun8i-a23-rsb.yaml @@ -21,8 +21,8 @@ properties: oneOf: - const: allwinner,sun8i-a23-rsb - items: - - const: allwinner,sun8i-a83t-rsb - - const: allwinner,sun8i-a23-rsb + - const: allwinner,sun8i-a83t-rsb + - const: allwinner,sun8i-a23-rsb reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/clock/allwinner,sun4i-a10-gates-clk.yaml b/Documentation/devicetree/bindings/clock/allwinner,sun4i-a10-gates-clk.yaml index ed1b2126a81b..9a37a357cb4e 100644 --- a/Documentation/devicetree/bindings/clock/allwinner,sun4i-a10-gates-clk.yaml +++ b/Documentation/devicetree/bindings/clock/allwinner,sun4i-a10-gates-clk.yaml @@ -52,12 +52,12 @@ properties: - const: allwinner,sun4i-a10-dram-gates-clk - items: - - const: allwinner,sun5i-a13-dram-gates-clk - - const: allwinner,sun4i-a10-gates-clk + - const: allwinner,sun5i-a13-dram-gates-clk + - const: allwinner,sun4i-a10-gates-clk - items: - - const: allwinner,sun8i-h3-apb0-gates-clk - - const: allwinner,sun4i-a10-gates-clk + - const: allwinner,sun8i-h3-apb0-gates-clk + - const: allwinner,sun4i-a10-gates-clk reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/clock/fsl,plldig.yaml b/Documentation/devicetree/bindings/clock/fsl,plldig.yaml index a203d5d498db..8141f22410dd 100644 --- a/Documentation/devicetree/bindings/clock/fsl,plldig.yaml +++ b/Documentation/devicetree/bindings/clock/fsl,plldig.yaml @@ -28,15 +28,14 @@ properties: const: 0 fsl,vco-hz: - description: Optional for VCO frequency of the PLL in Hertz. - The VCO frequency of this PLL cannot be changed during runtime - only at startup. Therefore, the output frequencies are very - limited and might not even closely match the requested frequency. - To work around this restriction the user may specify its own - desired VCO frequency for the PLL. - minimum: 650000000 - maximum: 1300000000 - default: 1188000000 + description: Optional for VCO frequency of the PLL in Hertz. The VCO frequency + of this PLL cannot be changed during runtime only at startup. Therefore, + the output frequencies are very limited and might not even closely match + the requested frequency. To work around this restriction the user may specify + its own desired VCO frequency for the PLL. + minimum: 650000000 + maximum: 1300000000 + default: 1188000000 required: - compatible diff --git a/Documentation/devicetree/bindings/clock/qcom,mmcc.yaml b/Documentation/devicetree/bindings/clock/qcom,mmcc.yaml index f684fe67db84..acc31b3991bd 100644 --- a/Documentation/devicetree/bindings/clock/qcom,mmcc.yaml +++ b/Documentation/devicetree/bindings/clock/qcom,mmcc.yaml @@ -15,15 +15,15 @@ description: | power domains. properties: - compatible : + compatible: enum: - - qcom,mmcc-apq8064 - - qcom,mmcc-apq8084 - - qcom,mmcc-msm8660 - - qcom,mmcc-msm8960 - - qcom,mmcc-msm8974 - - qcom,mmcc-msm8996 - - qcom,mmcc-msm8998 + - qcom,mmcc-apq8064 + - qcom,mmcc-apq8084 + - qcom,mmcc-msm8660 + - qcom,mmcc-msm8960 + - qcom,mmcc-msm8974 + - qcom,mmcc-msm8996 + - qcom,mmcc-msm8998 clocks: items: diff --git a/Documentation/devicetree/bindings/connector/usb-connector.yaml b/Documentation/devicetree/bindings/connector/usb-connector.yaml index 4638d7adb806..369c58e22a06 100644 --- a/Documentation/devicetree/bindings/connector/usb-connector.yaml +++ b/Documentation/devicetree/bindings/connector/usb-connector.yaml @@ -144,7 +144,7 @@ required: examples: # Micro-USB connector with HS lines routed via controller (MUIC). - - |+ + - | muic-max77843 { usb_con1: connector { compatible = "usb-b-connector"; @@ -156,7 +156,7 @@ examples: # USB-C connector attached to CC controller (s2mm005), HS lines routed # to companion PMIC (max77865), SS lines to USB3 PHY and SBU to DisplayPort. # DisplayPort video lines are routed to the connector via SS mux in USB3 PHY. - - |+ + - | ccic: s2mm005 { usb_con2: connector { compatible = "usb-c-connector"; @@ -190,7 +190,7 @@ examples: # USB-C connector attached to a typec port controller(ptn5110), which has # power delivery support and enables drp. - - |+ + - | #include <dt-bindings/usb/pd.h> typec: ptn5110 { usb_con3: connector { diff --git a/Documentation/devicetree/bindings/crypto/allwinner,sun4i-a10-crypto.yaml b/Documentation/devicetree/bindings/crypto/allwinner,sun4i-a10-crypto.yaml index 8b9a8f337f16..fc823572bcff 100644 --- a/Documentation/devicetree/bindings/crypto/allwinner,sun4i-a10-crypto.yaml +++ b/Documentation/devicetree/bindings/crypto/allwinner,sun4i-a10-crypto.yaml @@ -15,16 +15,16 @@ properties: oneOf: - const: allwinner,sun4i-a10-crypto - items: - - const: allwinner,sun5i-a13-crypto - - const: allwinner,sun4i-a10-crypto + - const: allwinner,sun5i-a13-crypto + - const: allwinner,sun4i-a10-crypto - items: - - const: allwinner,sun6i-a31-crypto - - const: allwinner,sun4i-a10-crypto + - const: allwinner,sun6i-a31-crypto + - const: allwinner,sun4i-a10-crypto - items: - - const: allwinner,sun7i-a20-crypto - - const: allwinner,sun4i-a10-crypto + - const: allwinner,sun7i-a20-crypto + - const: allwinner,sun4i-a10-crypto - items: - - const: allwinner,sun8i-a33-crypto + - const: allwinner,sun8i-a33-crypto reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/crypto/allwinner,sun8i-ce.yaml b/Documentation/devicetree/bindings/crypto/allwinner,sun8i-ce.yaml index 2c459b8c76ff..7a60d84289cc 100644 --- a/Documentation/devicetree/bindings/crypto/allwinner,sun8i-ce.yaml +++ b/Documentation/devicetree/bindings/crypto/allwinner,sun8i-ce.yaml @@ -50,16 +50,16 @@ if: const: allwinner,sun50i-h6-crypto then: properties: - clocks: - minItems: 3 - clock-names: - minItems: 3 + clocks: + minItems: 3 + clock-names: + minItems: 3 else: properties: - clocks: - maxItems: 2 - clock-names: - maxItems: 2 + clocks: + maxItems: 2 + clock-names: + maxItems: 2 required: - compatible diff --git a/Documentation/devicetree/bindings/crypto/amlogic,gxl-crypto.yaml b/Documentation/devicetree/bindings/crypto/amlogic,gxl-crypto.yaml index 5becc60a0e28..385b23d255c3 100644 --- a/Documentation/devicetree/bindings/crypto/amlogic,gxl-crypto.yaml +++ b/Documentation/devicetree/bindings/crypto/amlogic,gxl-crypto.yaml @@ -12,7 +12,7 @@ maintainers: properties: compatible: items: - - const: amlogic,gxl-crypto + - const: amlogic,gxl-crypto reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/display/allwinner,sun4i-a10-hdmi.yaml b/Documentation/devicetree/bindings/display/allwinner,sun4i-a10-hdmi.yaml index 5d4915aed1e2..75e6479397a5 100644 --- a/Documentation/devicetree/bindings/display/allwinner,sun4i-a10-hdmi.yaml +++ b/Documentation/devicetree/bindings/display/allwinner,sun4i-a10-hdmi.yaml @@ -21,8 +21,8 @@ properties: - const: allwinner,sun5i-a10s-hdmi - const: allwinner,sun6i-a31-hdmi - items: - - const: allwinner,sun7i-a20-hdmi - - const: allwinner,sun5i-a10s-hdmi + - const: allwinner,sun7i-a20-hdmi + - const: allwinner,sun5i-a10s-hdmi reg: maxItems: 1 @@ -33,32 +33,32 @@ properties: clocks: oneOf: - items: - - description: The HDMI interface clock - - description: The HDMI module clock - - description: The first video PLL - - description: The second video PLL + - description: The HDMI interface clock + - description: The HDMI module clock + - description: The first video PLL + - description: The second video PLL - items: - - description: The HDMI interface clock - - description: The HDMI module clock - - description: The HDMI DDC clock - - description: The first video PLL - - description: The second video PLL + - description: The HDMI interface clock + - description: The HDMI module clock + - description: The HDMI DDC clock + - description: The first video PLL + - description: The second video PLL clock-names: oneOf: - items: - - const: ahb - - const: mod - - const: pll-0 - - const: pll-1 + - const: ahb + - const: mod + - const: pll-0 + - const: pll-1 - items: - - const: ahb - - const: mod - - const: ddc - - const: pll-0 - - const: pll-1 + - const: ahb + - const: mod + - const: ddc + - const: pll-0 + - const: pll-1 resets: maxItems: 1 diff --git a/Documentation/devicetree/bindings/display/allwinner,sun4i-a10-tcon.yaml b/Documentation/devicetree/bindings/display/allwinner,sun4i-a10-tcon.yaml index e5344c4ae226..87cb77b32ee3 100644 --- a/Documentation/devicetree/bindings/display/allwinner,sun4i-a10-tcon.yaml +++ b/Documentation/devicetree/bindings/display/allwinner,sun4i-a10-tcon.yaml @@ -35,26 +35,26 @@ properties: - const: allwinner,sun9i-a80-tcon-tv - items: - - enum: - - allwinner,sun7i-a20-tcon0 - - allwinner,sun7i-a20-tcon1 - - const: allwinner,sun7i-a20-tcon + - enum: + - allwinner,sun7i-a20-tcon0 + - allwinner,sun7i-a20-tcon1 + - const: allwinner,sun7i-a20-tcon - items: - - enum: - - allwinner,sun50i-a64-tcon-lcd - - const: allwinner,sun8i-a83t-tcon-lcd + - enum: + - allwinner,sun50i-a64-tcon-lcd + - const: allwinner,sun8i-a83t-tcon-lcd - items: - - enum: - - allwinner,sun8i-h3-tcon-tv - - allwinner,sun50i-a64-tcon-tv - - const: allwinner,sun8i-a83t-tcon-tv + - enum: + - allwinner,sun8i-h3-tcon-tv + - allwinner,sun50i-a64-tcon-tv + - const: allwinner,sun8i-a83t-tcon-tv - items: - - enum: - - allwinner,sun50i-h6-tcon-tv - - const: allwinner,sun8i-r40-tcon-tv + - enum: + - allwinner,sun50i-h6-tcon-tv + - const: allwinner,sun8i-r40-tcon-tv reg: maxItems: 1 @@ -83,37 +83,37 @@ properties: resets: anyOf: - items: - - description: TCON Reset Line + - description: TCON Reset Line - items: - - description: TCON Reset Line - - description: TCON LVDS Reset Line + - description: TCON Reset Line + - description: TCON LVDS Reset Line - items: - - description: TCON Reset Line - - description: TCON eDP Reset Line + - description: TCON Reset Line + - description: TCON eDP Reset Line - items: - - description: TCON Reset Line - - description: TCON eDP Reset Line - - description: TCON LVDS Reset Line + - description: TCON Reset Line + - description: TCON eDP Reset Line + - description: TCON LVDS Reset Line reset-names: oneOf: - const: lcd - items: - - const: lcd - - const: lvds + - const: lcd + - const: lvds - items: - - const: lcd - - const: edp + - const: lcd + - const: edp - items: - - const: lcd - - const: edp - - const: lvds + - const: lcd + - const: edp + - const: lvds ports: type: object diff --git a/Documentation/devicetree/bindings/display/allwinner,sun6i-a31-mipi-dsi.yaml b/Documentation/devicetree/bindings/display/allwinner,sun6i-a31-mipi-dsi.yaml index 9e90c2b00960..eed05b26cdf3 100644 --- a/Documentation/devicetree/bindings/display/allwinner,sun6i-a31-mipi-dsi.yaml +++ b/Documentation/devicetree/bindings/display/allwinner,sun6i-a31-mipi-dsi.yaml @@ -76,28 +76,28 @@ required: allOf: - if: properties: - compatible: - contains: - const: allwinner,sun6i-a31-mipi-dsi + compatible: + contains: + const: allwinner,sun6i-a31-mipi-dsi then: - properties: - clocks: - minItems: 2 + properties: + clocks: + minItems: 2 - required: - - clock-names + required: + - clock-names - if: properties: - compatible: - contains: - const: allwinner,sun50i-a64-mipi-dsi + compatible: + contains: + const: allwinner,sun50i-a64-mipi-dsi then: - properties: - clocks: - minItems: 1 + properties: + clocks: + minItems: 1 additionalProperties: false diff --git a/Documentation/devicetree/bindings/display/allwinner,sun8i-a83t-dw-hdmi.yaml b/Documentation/devicetree/bindings/display/allwinner,sun8i-a83t-dw-hdmi.yaml index 4d6795690ac3..fa4769a0b26e 100644 --- a/Documentation/devicetree/bindings/display/allwinner,sun8i-a83t-dw-hdmi.yaml +++ b/Documentation/devicetree/bindings/display/allwinner,sun8i-a83t-dw-hdmi.yaml @@ -29,11 +29,11 @@ properties: - const: allwinner,sun50i-h6-dw-hdmi - items: - - enum: - - allwinner,sun8i-h3-dw-hdmi - - allwinner,sun8i-r40-dw-hdmi - - allwinner,sun50i-a64-dw-hdmi - - const: allwinner,sun8i-a83t-dw-hdmi + - enum: + - allwinner,sun8i-h3-dw-hdmi + - allwinner,sun8i-r40-dw-hdmi + - allwinner,sun50i-a64-dw-hdmi + - const: allwinner,sun8i-a83t-dw-hdmi reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/display/bridge/lvds-codec.yaml b/Documentation/devicetree/bindings/display/bridge/lvds-codec.yaml index 8f373029f5d2..e737951f5873 100644 --- a/Documentation/devicetree/bindings/display/bridge/lvds-codec.yaml +++ b/Documentation/devicetree/bindings/display/bridge/lvds-codec.yaml @@ -32,17 +32,17 @@ properties: compatible: oneOf: - items: - - enum: - - ti,ds90c185 # For the TI DS90C185 FPD-Link Serializer - - ti,ds90c187 # For the TI DS90C187 FPD-Link Serializer - - ti,sn75lvds83 # For the TI SN75LVDS83 FlatLink transmitter - - const: lvds-encoder # Generic LVDS encoder compatible fallback + - enum: + - ti,ds90c185 # For the TI DS90C185 FPD-Link Serializer + - ti,ds90c187 # For the TI DS90C187 FPD-Link Serializer + - ti,sn75lvds83 # For the TI SN75LVDS83 FlatLink transmitter + - const: lvds-encoder # Generic LVDS encoder compatible fallback - items: - - enum: - - ti,ds90cf384a # For the DS90CF384A FPD-Link LVDS Receiver - - const: lvds-decoder # Generic LVDS decoders compatible fallback + - enum: + - ti,ds90cf384a # For the DS90CF384A FPD-Link LVDS Receiver + - const: lvds-decoder # Generic LVDS decoders compatible fallback - enum: - - thine,thc63lvdm83d # For the THC63LVDM83D LVDS serializer + - thine,thc63lvdm83d # For the THC63LVDM83D LVDS serializer ports: type: object diff --git a/Documentation/devicetree/bindings/display/panel/sony,acx424akp.yaml b/Documentation/devicetree/bindings/display/panel/sony,acx424akp.yaml index 185dcc8fd1f9..78d060097052 100644 --- a/Documentation/devicetree/bindings/display/panel/sony,acx424akp.yaml +++ b/Documentation/devicetree/bindings/display/panel/sony,acx424akp.yaml @@ -18,7 +18,7 @@ properties: reg: true reset-gpios: true vddi-supply: - description: regulator that supplies the vddi voltage + description: regulator that supplies the vddi voltage enforce-video-mode: true required: diff --git a/Documentation/devicetree/bindings/display/panel/xinpeng,xpp055c272.yaml b/Documentation/devicetree/bindings/display/panel/xinpeng,xpp055c272.yaml index d9fdb58e06b4..891de2256d22 100644 --- a/Documentation/devicetree/bindings/display/panel/xinpeng,xpp055c272.yaml +++ b/Documentation/devicetree/bindings/display/panel/xinpeng,xpp055c272.yaml @@ -19,9 +19,9 @@ properties: backlight: true reset-gpios: true iovcc-supply: - description: regulator that supplies the iovcc voltage + description: regulator that supplies the iovcc voltage vci-supply: - description: regulator that supplies the vci voltage + description: regulator that supplies the vci voltage required: - compatible diff --git a/Documentation/devicetree/bindings/display/renesas,cmm.yaml b/Documentation/devicetree/bindings/display/renesas,cmm.yaml index a57037b9e9ba..005406c89507 100644 --- a/Documentation/devicetree/bindings/display/renesas,cmm.yaml +++ b/Documentation/devicetree/bindings/display/renesas,cmm.yaml @@ -21,15 +21,15 @@ properties: compatible: oneOf: - items: - - enum: - - renesas,r8a7795-cmm - - renesas,r8a7796-cmm - - renesas,r8a77965-cmm - - renesas,r8a77990-cmm - - renesas,r8a77995-cmm - - const: renesas,rcar-gen3-cmm + - enum: + - renesas,r8a7795-cmm + - renesas,r8a7796-cmm + - renesas,r8a77965-cmm + - renesas,r8a77990-cmm + - renesas,r8a77995-cmm + - const: renesas,rcar-gen3-cmm - items: - - const: renesas,rcar-gen2-cmm + - const: renesas,rcar-gen2-cmm reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/dma/ti/k3-udma.yaml b/Documentation/devicetree/bindings/dma/ti/k3-udma.yaml index 39ea05e6e5ff..85056982a242 100644 --- a/Documentation/devicetree/bindings/dma/ti/k3-udma.yaml +++ b/Documentation/devicetree/bindings/dma/ti/k3-udma.yaml @@ -69,10 +69,10 @@ properties: maxItems: 3 reg-names: - items: - - const: gcfg - - const: rchanrt - - const: tchanrt + items: + - const: gcfg + - const: rchanrt + - const: tchanrt msi-parent: true diff --git a/Documentation/devicetree/bindings/gpio/brcm,xgs-iproc-gpio.yaml b/Documentation/devicetree/bindings/gpio/brcm,xgs-iproc-gpio.yaml index 5f1ed20e43ee..4f2cbd8307a7 100644 --- a/Documentation/devicetree/bindings/gpio/brcm,xgs-iproc-gpio.yaml +++ b/Documentation/devicetree/bindings/gpio/brcm,xgs-iproc-gpio.yaml @@ -27,7 +27,7 @@ properties: gpio-controller: true '#gpio-cells': - const: 2 + const: 2 ngpios: minimum: 0 diff --git a/Documentation/devicetree/bindings/gpu/arm,mali-midgard.yaml b/Documentation/devicetree/bindings/gpu/arm,mali-midgard.yaml index 0407e45eb8c4..a7a67e0a42e5 100644 --- a/Documentation/devicetree/bindings/gpu/arm,mali-midgard.yaml +++ b/Documentation/devicetree/bindings/gpu/arm,mali-midgard.yaml @@ -16,33 +16,33 @@ properties: oneOf: - items: - enum: - - samsung,exynos5250-mali + - samsung,exynos5250-mali - const: arm,mali-t604 - items: - enum: - - samsung,exynos5420-mali + - samsung,exynos5420-mali - const: arm,mali-t628 - items: - enum: - - allwinner,sun50i-h6-mali + - allwinner,sun50i-h6-mali - const: arm,mali-t720 - items: - enum: - - amlogic,meson-gxm-mali - - realtek,rtd1295-mali + - amlogic,meson-gxm-mali + - realtek,rtd1295-mali - const: arm,mali-t820 - items: - enum: - - arm,juno-mali + - arm,juno-mali - const: arm,mali-t624 - items: - enum: - - rockchip,rk3288-mali - - samsung,exynos5433-mali + - rockchip,rk3288-mali + - samsung,exynos5433-mali - const: arm,mali-t760 - items: - enum: - - rockchip,rk3399-mali + - rockchip,rk3399-mali - const: arm,mali-t860 # "arm,mali-t830" diff --git a/Documentation/devicetree/bindings/gpu/vivante,gc.yaml b/Documentation/devicetree/bindings/gpu/vivante,gc.yaml index 0bc4b38d5cbb..e1ac6ff5a230 100644 --- a/Documentation/devicetree/bindings/gpu/vivante,gc.yaml +++ b/Documentation/devicetree/bindings/gpu/vivante,gc.yaml @@ -9,7 +9,7 @@ title: Vivante GPU Bindings description: Vivante GPU core devices maintainers: - - Lucas Stach <l.stach(a)pengutronix.de> + - Lucas Stach <l.stach(a)pengutronix.de> properties: compatible: diff --git a/Documentation/devicetree/bindings/i2c/i2c-rk3x.yaml b/Documentation/devicetree/bindings/i2c/i2c-rk3x.yaml index 61eac76c84c4..790aa7218ee0 100644 --- a/Documentation/devicetree/bindings/i2c/i2c-rk3x.yaml +++ b/Documentation/devicetree/bindings/i2c/i2c-rk3x.yaml @@ -28,14 +28,14 @@ properties: - const: rockchip,rk3399-i2c - items: - enum: - - rockchip,rk3036-i2c - - rockchip,rk3368-i2c + - rockchip,rk3036-i2c + - rockchip,rk3368-i2c - const: rockchip,rk3288-i2c - items: - enum: - - rockchip,px30-i2c - - rockchip,rk3308-i2c - - rockchip,rk3328-i2c + - rockchip,px30-i2c + - rockchip,rk3308-i2c + - rockchip,rk3328-i2c - const: rockchip,rk3399-i2c reg: diff --git a/Documentation/devicetree/bindings/iio/adc/adi,ad7124.yaml b/Documentation/devicetree/bindings/iio/adc/adi,ad7124.yaml index f0934b295edc..97087a45ce54 100644 --- a/Documentation/devicetree/bindings/iio/adc/adi,ad7124.yaml +++ b/Documentation/devicetree/bindings/iio/adc/adi,ad7124.yaml @@ -72,8 +72,8 @@ patternProperties: The channel number. It can have up to 8 channels on ad7124-4 and 16 channels on ad7124-8, numbered from 0 to 15. items: - minimum: 0 - maximum: 15 + minimum: 0 + maximum: 15 adi,reference-select: description: | diff --git a/Documentation/devicetree/bindings/iio/adc/lltc,ltc2496.yaml b/Documentation/devicetree/bindings/iio/adc/lltc,ltc2496.yaml index 118809a03279..97f521d654ea 100644 --- a/Documentation/devicetree/bindings/iio/adc/lltc,ltc2496.yaml +++ b/Documentation/devicetree/bindings/iio/adc/lltc,ltc2496.yaml @@ -7,9 +7,9 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: Linear Technology / Analog Devices LTC2496 ADC maintainers: - - Lars-Peter Clausen <lars(a)metafoo.de> - - Michael Hennerich <Michael.Hennerich(a)analog.com> - - Stefan Popa <stefan.popa(a)analog.com> + - Lars-Peter Clausen <lars(a)metafoo.de> + - Michael Hennerich <Michael.Hennerich(a)analog.com> + - Stefan Popa <stefan.popa(a)analog.com> properties: compatible: diff --git a/Documentation/devicetree/bindings/input/allwinner,sun4i-a10-lradc-keys.yaml b/Documentation/devicetree/bindings/input/allwinner,sun4i-a10-lradc-keys.yaml index 5b3b71c9c018..512a6af5aa42 100644 --- a/Documentation/devicetree/bindings/input/allwinner,sun4i-a10-lradc-keys.yaml +++ b/Documentation/devicetree/bindings/input/allwinner,sun4i-a10-lradc-keys.yaml @@ -16,8 +16,8 @@ properties: - const: allwinner,sun4i-a10-lradc-keys - const: allwinner,sun8i-a83t-r-lradc - items: - - const: allwinner,sun50i-a64-lradc - - const: allwinner,sun8i-a83t-r-lradc + - const: allwinner,sun50i-a64-lradc + - const: allwinner,sun8i-a83t-r-lradc reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/input/touchscreen/goodix.yaml b/Documentation/devicetree/bindings/input/touchscreen/goodix.yaml index c8ea9434c9cc..e81cfa56f25a 100644 --- a/Documentation/devicetree/bindings/input/touchscreen/goodix.yaml +++ b/Documentation/devicetree/bindings/input/touchscreen/goodix.yaml @@ -63,7 +63,7 @@ required: - interrupts examples: -- | + - | i2c { #address-cells = <1>; #size-cells = <0>; diff --git a/Documentation/devicetree/bindings/interconnect/qcom,msm8916.yaml b/Documentation/devicetree/bindings/interconnect/qcom,msm8916.yaml index 4107e60cab12..e1009ae4e8f7 100644 --- a/Documentation/devicetree/bindings/interconnect/qcom,msm8916.yaml +++ b/Documentation/devicetree/bindings/interconnect/qcom,msm8916.yaml @@ -10,8 +10,8 @@ maintainers: - Georgi Djakov <georgi.djakov(a)linaro.org> description: | - The Qualcomm MSM8916 interconnect providers support adjusting the - bandwidth requirements between the various NoC fabrics. + The Qualcomm MSM8916 interconnect providers support adjusting the + bandwidth requirements between the various NoC fabrics. properties: compatible: diff --git a/Documentation/devicetree/bindings/interconnect/qcom,msm8974.yaml b/Documentation/devicetree/bindings/interconnect/qcom,msm8974.yaml index 9af3c6e59cff..8004c4baf397 100644 --- a/Documentation/devicetree/bindings/interconnect/qcom,msm8974.yaml +++ b/Documentation/devicetree/bindings/interconnect/qcom,msm8974.yaml @@ -10,8 +10,8 @@ maintainers: - Brian Masney <masneyb(a)onstation.org> description: | - The Qualcomm MSM8974 interconnect providers support setting system - bandwidth requirements between various network-on-chip fabrics. + The Qualcomm MSM8974 interconnect providers support setting system + bandwidth requirements between various network-on-chip fabrics. properties: reg: diff --git a/Documentation/devicetree/bindings/interconnect/qcom,qcs404.yaml b/Documentation/devicetree/bindings/interconnect/qcom,qcs404.yaml index 8d65c5f80679..3fbb8785fbc9 100644 --- a/Documentation/devicetree/bindings/interconnect/qcom,qcs404.yaml +++ b/Documentation/devicetree/bindings/interconnect/qcom,qcs404.yaml @@ -10,8 +10,8 @@ maintainers: - Georgi Djakov <georgi.djakov(a)linaro.org> description: | - The Qualcomm QCS404 interconnect providers support adjusting the - bandwidth requirements between the various NoC fabrics. + The Qualcomm QCS404 interconnect providers support adjusting the + bandwidth requirements between the various NoC fabrics. properties: reg: diff --git a/Documentation/devicetree/bindings/interrupt-controller/allwinner,sun7i-a20-sc-nmi.yaml b/Documentation/devicetree/bindings/interrupt-controller/allwinner,sun7i-a20-sc-nmi.yaml index cf09055da78b..7cd6b8bacfa0 100644 --- a/Documentation/devicetree/bindings/interrupt-controller/allwinner,sun7i-a20-sc-nmi.yaml +++ b/Documentation/devicetree/bindings/interrupt-controller/allwinner,sun7i-a20-sc-nmi.yaml @@ -27,15 +27,15 @@ properties: deprecated: true - const: allwinner,sun7i-a20-sc-nmi - items: - - const: allwinner,sun8i-a83t-r-intc - - const: allwinner,sun6i-a31-r-intc + - const: allwinner,sun8i-a83t-r-intc + - const: allwinner,sun6i-a31-r-intc - const: allwinner,sun9i-a80-sc-nmi - items: - - const: allwinner,sun50i-a64-r-intc - - const: allwinner,sun6i-a31-r-intc + - const: allwinner,sun50i-a64-r-intc + - const: allwinner,sun6i-a31-r-intc - items: - - const: allwinner,sun50i-h6-r-intc - - const: allwinner,sun6i-a31-r-intc + - const: allwinner,sun50i-h6-r-intc + - const: allwinner,sun6i-a31-r-intc reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/interrupt-controller/intel,ixp4xx-interrupt.yaml b/Documentation/devicetree/bindings/interrupt-controller/intel,ixp4xx-interrupt.yaml index ccc507f384d2..14dced11877b 100644 --- a/Documentation/devicetree/bindings/interrupt-controller/intel,ixp4xx-interrupt.yaml +++ b/Documentation/devicetree/bindings/interrupt-controller/intel,ixp4xx-interrupt.yaml @@ -25,10 +25,10 @@ properties: compatible: items: - enum: - - intel,ixp42x-interrupt - - intel,ixp43x-interrupt - - intel,ixp45x-interrupt - - intel,ixp46x-interrupt + - intel,ixp42x-interrupt + - intel,ixp43x-interrupt + - intel,ixp45x-interrupt + - intel,ixp46x-interrupt reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/interrupt-controller/st,stm32-exti.yaml b/Documentation/devicetree/bindings/interrupt-controller/st,stm32-exti.yaml index 9e5c6608b4e3..2a5b29567926 100644 --- a/Documentation/devicetree/bindings/interrupt-controller/st,stm32-exti.yaml +++ b/Documentation/devicetree/bindings/interrupt-controller/st,stm32-exti.yaml @@ -14,13 +14,13 @@ properties: compatible: oneOf: - items: - - enum: - - st,stm32-exti - - st,stm32h7-exti + - enum: + - st,stm32-exti + - st,stm32h7-exti - items: - - enum: - - st,stm32mp1-exti - - const: syscon + - enum: + - st,stm32mp1-exti + - const: syscon "#interrupt-cells": const: 2 diff --git a/Documentation/devicetree/bindings/iommu/samsung,sysmmu.yaml b/Documentation/devicetree/bindings/iommu/samsung,sysmmu.yaml index 0e33cd9e010e..af51b91c893e 100644 --- a/Documentation/devicetree/bindings/iommu/samsung,sysmmu.yaml +++ b/Documentation/devicetree/bindings/iommu/samsung,sysmmu.yaml @@ -54,13 +54,13 @@ properties: clock-names: oneOf: - items: - - const: sysmmu + - const: sysmmu - items: - - const: sysmmu - - const: master + - const: sysmmu + - const: master - items: - - const: aclk - - const: pclk + - const: aclk + - const: pclk "#iommu-cells": const: 0 diff --git a/Documentation/devicetree/bindings/mailbox/st,stm32-ipcc.yaml b/Documentation/devicetree/bindings/mailbox/st,stm32-ipcc.yaml index 5b13d6672996..db851541d619 100644 --- a/Documentation/devicetree/bindings/mailbox/st,stm32-ipcc.yaml +++ b/Documentation/devicetree/bindings/mailbox/st,stm32-ipcc.yaml @@ -24,7 +24,7 @@ properties: maxItems: 1 clocks: - maxItems: 1 + maxItems: 1 interrupts: items: diff --git a/Documentation/devicetree/bindings/media/allwinner,sun4i-a10-csi.yaml b/Documentation/devicetree/bindings/media/allwinner,sun4i-a10-csi.yaml index 8453ee340b9f..09318830db47 100644 --- a/Documentation/devicetree/bindings/media/allwinner,sun4i-a10-csi.yaml +++ b/Documentation/devicetree/bindings/media/allwinner,sun4i-a10-csi.yaml @@ -20,11 +20,11 @@ properties: - const: allwinner,sun4i-a10-csi1 - const: allwinner,sun7i-a20-csi0 - items: - - const: allwinner,sun7i-a20-csi1 - - const: allwinner,sun4i-a10-csi1 + - const: allwinner,sun7i-a20-csi1 + - const: allwinner,sun4i-a10-csi1 - items: - - const: allwinner,sun8i-r40-csi0 - - const: allwinner,sun7i-a20-csi0 + - const: allwinner,sun8i-r40-csi0 + - const: allwinner,sun7i-a20-csi0 reg: maxItems: 1 @@ -35,24 +35,24 @@ properties: clocks: oneOf: - items: - - description: The CSI interface clock - - description: The CSI DRAM clock + - description: The CSI interface clock + - description: The CSI DRAM clock - items: - - description: The CSI interface clock - - description: The CSI ISP clock - - description: The CSI DRAM clock + - description: The CSI interface clock + - description: The CSI ISP clock + - description: The CSI DRAM clock clock-names: oneOf: - items: - - const: bus - - const: ram + - const: bus + - const: ram - items: - - const: bus - - const: isp - - const: ram + - const: bus + - const: isp + - const: ram resets: maxItems: 1 diff --git a/Documentation/devicetree/bindings/media/amlogic,gx-vdec.yaml b/Documentation/devicetree/bindings/media/amlogic,gx-vdec.yaml index 37d77e065491..5a1da4029c37 100644 --- a/Documentation/devicetree/bindings/media/amlogic,gx-vdec.yaml +++ b/Documentation/devicetree/bindings/media/amlogic,gx-vdec.yaml @@ -29,14 +29,14 @@ properties: compatible: oneOf: - items: - - enum: - - amlogic,gxbb-vdec # GXBB (S905) - - amlogic,gxl-vdec # GXL (S905X, S905D) - - amlogic,gxm-vdec # GXM (S912) - - const: amlogic,gx-vdec + - enum: + - amlogic,gxbb-vdec # GXBB (S905) + - amlogic,gxl-vdec # GXL (S905X, S905D) + - amlogic,gxm-vdec # GXM (S912) + - const: amlogic,gx-vdec - enum: - - amlogic,g12a-vdec # G12A (S905X2, S905D2) - - amlogic,sm1-vdec # SM1 (S905X3, S905D3) + - amlogic,g12a-vdec # G12A (S905X2, S905D2) + - amlogic,sm1-vdec # SM1 (S905X3, S905D3) interrupts: minItems: 2 diff --git a/Documentation/devicetree/bindings/media/renesas,ceu.yaml b/Documentation/devicetree/bindings/media/renesas,ceu.yaml index fcb5f13704a5..f2393458814e 100644 --- a/Documentation/devicetree/bindings/media/renesas,ceu.yaml +++ b/Documentation/devicetree/bindings/media/renesas,ceu.yaml @@ -32,23 +32,23 @@ properties: additionalProperties: false properties: - endpoint: - type: object - additionalProperties: false + endpoint: + type: object + additionalProperties: false # Properties described in # Documentation/devicetree/bindings/media/video-interfaces.txt - properties: - remote-endpoint: true - hsync-active: true - vsync-active: true - field-even-active: false - bus-width: - enum: [8, 16] - default: 8 - - required: - - remote-endpoint + properties: + remote-endpoint: true + hsync-active: true + vsync-active: true + field-even-active: false + bus-width: + enum: [8, 16] + default: 8 + + required: + - remote-endpoint required: - endpoint diff --git a/Documentation/devicetree/bindings/media/renesas,vin.yaml b/Documentation/devicetree/bindings/media/renesas,vin.yaml index 1ec947b4781f..ecc09f1124d4 100644 --- a/Documentation/devicetree/bindings/media/renesas,vin.yaml +++ b/Documentation/devicetree/bindings/media/renesas,vin.yaml @@ -261,13 +261,13 @@ properties: anyOf: - required: - - endpoint@0 + - endpoint@0 - required: - - endpoint@1 + - endpoint@1 - required: - - endpoint@2 + - endpoint@2 - required: - - endpoint@3 + - endpoint@3 additionalProperties: false diff --git a/Documentation/devicetree/bindings/media/ti,vpe.yaml b/Documentation/devicetree/bindings/media/ti,vpe.yaml index f3a8a350e85f..ef473f287399 100644 --- a/Documentation/devicetree/bindings/media/ti,vpe.yaml +++ b/Documentation/devicetree/bindings/media/ti,vpe.yaml @@ -17,7 +17,7 @@ description: |- properties: compatible: - const: ti,dra7-vpe + const: ti,dra7-vpe reg: items: diff --git a/Documentation/devicetree/bindings/memory-controllers/fsl/imx8m-ddrc.yaml b/Documentation/devicetree/bindings/memory-controllers/fsl/imx8m-ddrc.yaml index c9e6c22cb5be..445e46feda69 100644 --- a/Documentation/devicetree/bindings/memory-controllers/fsl/imx8m-ddrc.yaml +++ b/Documentation/devicetree/bindings/memory-controllers/fsl/imx8m-ddrc.yaml @@ -25,9 +25,9 @@ properties: compatible: items: - enum: - - fsl,imx8mn-ddrc - - fsl,imx8mm-ddrc - - fsl,imx8mq-ddrc + - fsl,imx8mn-ddrc + - fsl,imx8mm-ddrc + - fsl,imx8mq-ddrc - const: fsl,imx8m-ddrc reg: diff --git a/Documentation/devicetree/bindings/mfd/st,stm32-lptimer.yaml b/Documentation/devicetree/bindings/mfd/st,stm32-lptimer.yaml index ddf190cb800b..e675611f80d0 100644 --- a/Documentation/devicetree/bindings/mfd/st,stm32-lptimer.yaml +++ b/Documentation/devicetree/bindings/mfd/st,stm32-lptimer.yaml @@ -66,8 +66,8 @@ patternProperties: reg: description: Identify trigger hardware block. items: - minimum: 0 - maximum: 2 + minimum: 0 + maximum: 2 required: - compatible diff --git a/Documentation/devicetree/bindings/mfd/st,stm32-timers.yaml b/Documentation/devicetree/bindings/mfd/st,stm32-timers.yaml index 590849ee9f32..4acda7ce3b44 100644 --- a/Documentation/devicetree/bindings/mfd/st,stm32-timers.yaml +++ b/Documentation/devicetree/bindings/mfd/st,stm32-timers.yaml @@ -102,8 +102,8 @@ patternProperties: reg: description: Identify trigger hardware block. items: - minimum: 0 - maximum: 16 + minimum: 0 + maximum: 16 required: - compatible diff --git a/Documentation/devicetree/bindings/mfd/syscon.yaml b/Documentation/devicetree/bindings/mfd/syscon.yaml index 39375e4313d2..7a39486b215a 100644 --- a/Documentation/devicetree/bindings/mfd/syscon.yaml +++ b/Documentation/devicetree/bindings/mfd/syscon.yaml @@ -33,13 +33,13 @@ properties: compatible: anyOf: - items: - - enum: - - allwinner,sun8i-a83t-system-controller - - allwinner,sun8i-h3-system-controller - - allwinner,sun8i-v3s-system-controller - - allwinner,sun50i-a64-system-controller + - enum: + - allwinner,sun8i-a83t-system-controller + - allwinner,sun8i-h3-system-controller + - allwinner,sun8i-v3s-system-controller + - allwinner,sun50i-a64-system-controller - - const: syscon + - const: syscon - contains: const: syscon diff --git a/Documentation/devicetree/bindings/mmc/cdns,sdhci.yaml b/Documentation/devicetree/bindings/mmc/cdns,sdhci.yaml index 2f45dd0d04db..d43a0c557a44 100644 --- a/Documentation/devicetree/bindings/mmc/cdns,sdhci.yaml +++ b/Documentation/devicetree/bindings/mmc/cdns,sdhci.yaml @@ -17,7 +17,7 @@ properties: compatible: items: - enum: - - socionext,uniphier-sd4hc + - socionext,uniphier-sd4hc - const: cdns,sd4hc reg: diff --git a/Documentation/devicetree/bindings/mmc/rockchip-dw-mshc.yaml b/Documentation/devicetree/bindings/mmc/rockchip-dw-mshc.yaml index 89c3edd6a728..4ee3ed6efab4 100644 --- a/Documentation/devicetree/bindings/mmc/rockchip-dw-mshc.yaml +++ b/Documentation/devicetree/bindings/mmc/rockchip-dw-mshc.yaml @@ -30,21 +30,21 @@ properties: - items: - enum: # for Rockchip PX30 - - rockchip,px30-dw-mshc + - rockchip,px30-dw-mshc # for Rockchip RK3036 - - rockchip,rk3036-dw-mshc + - rockchip,rk3036-dw-mshc # for Rockchip RK322x - - rockchip,rk3228-dw-mshc + - rockchip,rk3228-dw-mshc # for Rockchip RK3308 - - rockchip,rk3308-dw-mshc + - rockchip,rk3308-dw-mshc # for Rockchip RK3328 - - rockchip,rk3328-dw-mshc + - rockchip,rk3328-dw-mshc # for Rockchip RK3368 - - rockchip,rk3368-dw-mshc + - rockchip,rk3368-dw-mshc # for Rockchip RK3399 - - rockchip,rk3399-dw-mshc + - rockchip,rk3399-dw-mshc # for Rockchip RV1108 - - rockchip,rv1108-dw-mshc + - rockchip,rv1108-dw-mshc - const: rockchip,rk3288-dw-mshc reg: diff --git a/Documentation/devicetree/bindings/mmc/socionext,uniphier-sd.yaml b/Documentation/devicetree/bindings/mmc/socionext,uniphier-sd.yaml index cdfac9b4411b..8d6413f48823 100644 --- a/Documentation/devicetree/bindings/mmc/socionext,uniphier-sd.yaml +++ b/Documentation/devicetree/bindings/mmc/socionext,uniphier-sd.yaml @@ -35,15 +35,15 @@ properties: oneOf: - const: host - items: - - const: host - - const: bridge + - const: host + - const: bridge - items: - - const: host - - const: hw + - const: host + - const: hw - items: - - const: host - - const: bridge - - const: hw + - const: host + - const: bridge + - const: hw resets: minItems: 1 diff --git a/Documentation/devicetree/bindings/mtd/denali,nand.yaml b/Documentation/devicetree/bindings/mtd/denali,nand.yaml index 46e6b6726bc0..c07b91592cbd 100644 --- a/Documentation/devicetree/bindings/mtd/denali,nand.yaml +++ b/Documentation/devicetree/bindings/mtd/denali,nand.yaml @@ -54,8 +54,8 @@ properties: reg: register reset oneOf: - items: - - const: nand - - const: reg + - const: nand + - const: reg - const: nand - const: reg diff --git a/Documentation/devicetree/bindings/net/allwinner,sun8i-a83t-emac.yaml b/Documentation/devicetree/bindings/net/allwinner,sun8i-a83t-emac.yaml index db36b4d86484..c7c9ad4e3f9f 100644 --- a/Documentation/devicetree/bindings/net/allwinner,sun8i-a83t-emac.yaml +++ b/Documentation/devicetree/bindings/net/allwinner,sun8i-a83t-emac.yaml @@ -19,8 +19,8 @@ properties: - const: allwinner,sun8i-v3s-emac - const: allwinner,sun50i-a64-emac - items: - - const: allwinner,sun50i-h6-emac - - const: allwinner,sun50i-a64-emac + - const: allwinner,sun50i-h6-emac + - const: allwinner,sun50i-a64-emac reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/net/can/bosch,m_can.yaml b/Documentation/devicetree/bindings/net/can/bosch,m_can.yaml index cccf8202c8f7..7a784dc4e513 100644 --- a/Documentation/devicetree/bindings/net/can/bosch,m_can.yaml +++ b/Documentation/devicetree/bindings/net/can/bosch,m_can.yaml @@ -9,7 +9,7 @@ title: Bosch MCAN controller Bindings description: Bosch MCAN controller for CAN bus maintainers: - - Sriram Dash <sriram.dash(a)samsung.com> + - Sriram Dash <sriram.dash(a)samsung.com> properties: compatible: @@ -51,31 +51,31 @@ properties: bosch,mram-cfg: description: | - Message RAM configuration data. - Multiple M_CAN instances can share the same Message RAM - and each element(e.g Rx FIFO or Tx Buffer and etc) number - in Message RAM is also configurable, so this property is - telling driver how the shared or private Message RAM are - used by this M_CAN controller. - - The format should be as follows: - <offset sidf_elems xidf_elems rxf0_elems rxf1_elems rxb_elems txe_elems txb_elems> - The 'offset' is an address offset of the Message RAM where - the following elements start from. This is usually set to - 0x0 if you're using a private Message RAM. The remain cells - are used to specify how many elements are used for each FIFO/Buffer. - - M_CAN includes the following elements according to user manual: - 11-bit Filter 0-128 elements / 0-128 words - 29-bit Filter 0-64 elements / 0-128 words - Rx FIFO 0 0-64 elements / 0-1152 words - Rx FIFO 1 0-64 elements / 0-1152 words - Rx Buffers 0-64 elements / 0-1152 words - Tx Event FIFO 0-32 elements / 0-64 words - Tx Buffers 0-32 elements / 0-576 words - - Please refer to 2.4.1 Message RAM Configuration in Bosch - M_CAN user manual for details. + Message RAM configuration data. + Multiple M_CAN instances can share the same Message RAM + and each element(e.g Rx FIFO or Tx Buffer and etc) number + in Message RAM is also configurable, so this property is + telling driver how the shared or private Message RAM are + used by this M_CAN controller. + + The format should be as follows: + <offset sidf_elems xidf_elems rxf0_elems rxf1_elems rxb_elems txe_elems txb_elems> + The 'offset' is an address offset of the Message RAM where + the following elements start from. This is usually set to + 0x0 if you're using a private Message RAM. The remain cells + are used to specify how many elements are used for each FIFO/Buffer. + + M_CAN includes the following elements according to user manual: + 11-bit Filter 0-128 elements / 0-128 words + 29-bit Filter 0-64 elements / 0-128 words + Rx FIFO 0 0-64 elements / 0-1152 words + Rx FIFO 1 0-64 elements / 0-1152 words + Rx Buffers 0-64 elements / 0-1152 words + Tx Event FIFO 0-32 elements / 0-64 words + Tx Buffers 0-32 elements / 0-576 words + + Please refer to 2.4.1 Message RAM Configuration in Bosch + M_CAN user manual for details. allOf: - $ref: /schemas/types.yaml#/definitions/int32-array - items: diff --git a/Documentation/devicetree/bindings/net/renesas,ether.yaml b/Documentation/devicetree/bindings/net/renesas,ether.yaml index 7f84df9790e2..2eaa8799e002 100644 --- a/Documentation/devicetree/bindings/net/renesas,ether.yaml +++ b/Documentation/devicetree/bindings/net/renesas,ether.yaml @@ -40,8 +40,8 @@ properties: reg: items: - - description: E-DMAC/feLic registers - - description: TSU registers + - description: E-DMAC/feLic registers + - description: TSU registers minItems: 1 interrupts: diff --git a/Documentation/devicetree/bindings/net/ti,cpsw-switch.yaml b/Documentation/devicetree/bindings/net/ti,cpsw-switch.yaml index 976f139bb66e..8fc8d3be303b 100644 --- a/Documentation/devicetree/bindings/net/ti,cpsw-switch.yaml +++ b/Documentation/devicetree/bindings/net/ti,cpsw-switch.yaml @@ -23,14 +23,14 @@ properties: oneOf: - const: ti,cpsw-switch - items: - - const: ti,am335x-cpsw-switch - - const: ti,cpsw-switch + - const: ti,am335x-cpsw-switch + - const: ti,cpsw-switch - items: - - const: ti,am4372-cpsw-switch - - const: ti,cpsw-switch + - const: ti,am4372-cpsw-switch + - const: ti,cpsw-switch - items: - - const: ti,dra7-cpsw-switch - - const: ti,cpsw-switch + - const: ti,dra7-cpsw-switch + - const: ti,cpsw-switch reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/net/ti,davinci-mdio.yaml b/Documentation/devicetree/bindings/net/ti,davinci-mdio.yaml index 242ac4935a4b..2ea14ab29254 100644 --- a/Documentation/devicetree/bindings/net/ti,davinci-mdio.yaml +++ b/Documentation/devicetree/bindings/net/ti,davinci-mdio.yaml @@ -18,25 +18,24 @@ allOf: properties: compatible: oneOf: - - const: ti,davinci_mdio - - items: - - const: ti,keystone_mdio - - const: ti,davinci_mdio - - items: - - const: ti,cpsw-mdio - - const: ti,davinci_mdio - - items: - - const: ti,am4372-mdio - - const: ti,cpsw-mdio - - const: ti,davinci_mdio + - const: ti,davinci_mdio + - items: + - const: ti,keystone_mdio + - const: ti,davinci_mdio + - items: + - const: ti,cpsw-mdio + - const: ti,davinci_mdio + - items: + - const: ti,am4372-mdio + - const: ti,cpsw-mdio + - const: ti,davinci_mdio reg: maxItems: 1 bus_freq: - maximum: 2500000 - description: - MDIO Bus frequency + maximum: 2500000 + description: MDIO Bus frequency ti,hwmods: description: TI hwmod name diff --git a/Documentation/devicetree/bindings/phy/intel,lgm-emmc-phy.yaml b/Documentation/devicetree/bindings/phy/intel,lgm-emmc-phy.yaml index 9a346d6290d9..77bb5309918e 100644 --- a/Documentation/devicetree/bindings/phy/intel,lgm-emmc-phy.yaml +++ b/Documentation/devicetree/bindings/phy/intel,lgm-emmc-phy.yaml @@ -23,7 +23,7 @@ description: |+ properties: compatible: - const: intel,lgm-emmc-phy + const: intel,lgm-emmc-phy "#phy-cells": const: 0 diff --git a/Documentation/devicetree/bindings/pwm/pwm-samsung.yaml b/Documentation/devicetree/bindings/pwm/pwm-samsung.yaml index ea7f32905172..4fe64f4dd594 100644 --- a/Documentation/devicetree/bindings/pwm/pwm-samsung.yaml +++ b/Documentation/devicetree/bindings/pwm/pwm-samsung.yaml @@ -49,17 +49,17 @@ properties: are available. oneOf: - items: - - const: timers + - const: timers - items: - - const: timers - - const: pwm-tclk0 + - const: timers + - const: pwm-tclk0 - items: - - const: timers - - const: pwm-tclk1 + - const: timers + - const: pwm-tclk1 - items: - - const: timers - - const: pwm-tclk0 - - const: pwm-tclk1 + - const: timers + - const: pwm-tclk0 + - const: pwm-tclk1 interrupts: description: diff --git a/Documentation/devicetree/bindings/remoteproc/st,stm32-rproc.yaml b/Documentation/devicetree/bindings/remoteproc/st,stm32-rproc.yaml index c0d83865e933..4ff4d3df0a06 100644 --- a/Documentation/devicetree/bindings/remoteproc/st,stm32-rproc.yaml +++ b/Documentation/devicetree/bindings/remoteproc/st,stm32-rproc.yaml @@ -25,7 +25,7 @@ properties: maxItems: 3 resets: - maxItems: 1 + maxItems: 1 st,syscfg-holdboot: allOf: diff --git a/Documentation/devicetree/bindings/reset/brcm,bcm7216-pcie-sata-rescal.yaml b/Documentation/devicetree/bindings/reset/brcm,bcm7216-pcie-sata-rescal.yaml index 512a33bdb208..dfce6738b033 100644 --- a/Documentation/devicetree/bindings/reset/brcm,bcm7216-pcie-sata-rescal.yaml +++ b/Documentation/devicetree/bindings/reset/brcm,bcm7216-pcie-sata-rescal.yaml @@ -7,7 +7,9 @@ $schema: "http://devicetree.org/meta-schemas/core.yaml#" title: BCM7216 RESCAL reset controller -description: This document describes the BCM7216 RESCAL reset controller which is responsible for controlling the reset of the SATA and PCIe0/1 instances on BCM7216. +description: This document describes the BCM7216 RESCAL reset controller + which is responsible for controlling the reset of the SATA and PCIe0/1 + instances on BCM7216. maintainers: - Florian Fainelli <f.fainelli(a)gmail.com> diff --git a/Documentation/devicetree/bindings/rtc/st,stm32-rtc.yaml b/Documentation/devicetree/bindings/rtc/st,stm32-rtc.yaml index 48c6cafca90c..57b087574aa1 100644 --- a/Documentation/devicetree/bindings/rtc/st,stm32-rtc.yaml +++ b/Documentation/devicetree/bindings/rtc/st,stm32-rtc.yaml @@ -38,10 +38,10 @@ properties: minItems: 3 maxItems: 3 description: | - Phandle/offset/mask triplet. The phandle to pwrcfg used to - access control register at offset, and change the dbp (Disable Backup - Protection) bit represented by the mask, mandatory to disable/enable backup - domain (RTC registers) write protection. + Phandle/offset/mask triplet. The phandle to pwrcfg used to + access control register at offset, and change the dbp (Disable Backup + Protection) bit represented by the mask, mandatory to disable/enable backup + domain (RTC registers) write protection. assigned-clocks: description: | @@ -78,14 +78,14 @@ allOf: const: st,stm32h7-rtc then: - properties: - clocks: - minItems: 2 - maxItems: 2 + properties: + clocks: + minItems: 2 + maxItems: 2 - required: - - clock-names - - st,syscfg + required: + - clock-names + - st,syscfg - if: properties: @@ -94,16 +94,16 @@ allOf: const: st,stm32mp1-rtc then: - properties: - clocks: - minItems: 2 - maxItems: 2 + properties: + clocks: + minItems: 2 + maxItems: 2 - assigned-clocks: false - assigned-clock-parents: false + assigned-clocks: false + assigned-clock-parents: false - required: - - clock-names + required: + - clock-names required: - compatible diff --git a/Documentation/devicetree/bindings/serial/amlogic,meson-uart.yaml b/Documentation/devicetree/bindings/serial/amlogic,meson-uart.yaml index d4178ab0d675..75ebc9952a99 100644 --- a/Documentation/devicetree/bindings/serial/amlogic,meson-uart.yaml +++ b/Documentation/devicetree/bindings/serial/amlogic,meson-uart.yaml @@ -24,18 +24,18 @@ properties: oneOf: - description: Always-on power domain UART controller items: - - enum: + - enum: + - amlogic,meson6-uart + - amlogic,meson8-uart + - amlogic,meson8b-uart + - amlogic,meson-gx-uart + - const: amlogic,meson-ao-uart + - description: Everything-Else power domain UART controller + enum: - amlogic,meson6-uart - amlogic,meson8-uart - amlogic,meson8b-uart - amlogic,meson-gx-uart - - const: amlogic,meson-ao-uart - - description: Everything-Else power domain UART controller - enum: - - amlogic,meson6-uart - - amlogic,meson8-uart - - amlogic,meson8b-uart - - amlogic,meson-gx-uart reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/serial/rs485.yaml b/Documentation/devicetree/bindings/serial/rs485.yaml index d4beaf11222d..2b8261ea6d9c 100644 --- a/Documentation/devicetree/bindings/serial/rs485.yaml +++ b/Documentation/devicetree/bindings/serial/rs485.yaml @@ -6,13 +6,12 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: RS485 serial communications Bindings -description: The RTS signal is capable of automatically controlling - line direction for the built-in half-duplex mode. - The properties described hereafter shall be given to a - half-duplex capable UART node. +description: The RTS signal is capable of automatically controlling line + direction for the built-in half-duplex mode. The properties described + hereafter shall be given to a half-duplex capable UART node. maintainers: - - Rob Herring <robh(a)kernel.org> + - Rob Herring <robh(a)kernel.org> properties: rs485-rts-delay: @@ -37,9 +36,11 @@ properties: $ref: /schemas/types.yaml#/definitions/flag linux,rs485-enabled-at-boot-time: - description: enables the rs485 feature at boot time. It can be disabled later with proper ioctl. + description: enables the rs485 feature at boot time. It can be disabled + later with proper ioctl. $ref: /schemas/types.yaml#/definitions/flag rs485-rx-during-tx: - description: enables the receiving of data even while sending data. - $ref: /schemas/types.yaml#/definitions/flag + description: enables the receiving of data even while sending data. + $ref: /schemas/types.yaml#/definitions/flag +... diff --git a/Documentation/devicetree/bindings/soc/amlogic/amlogic,canvas.yaml b/Documentation/devicetree/bindings/soc/amlogic/amlogic,canvas.yaml index cb008fd188d8..02b2d5ba01d6 100644 --- a/Documentation/devicetree/bindings/soc/amlogic/amlogic,canvas.yaml +++ b/Documentation/devicetree/bindings/soc/amlogic/amlogic,canvas.yaml @@ -26,11 +26,11 @@ properties: compatible: oneOf: - items: - - enum: - - amlogic,meson8-canvas - - amlogic,meson8b-canvas - - amlogic,meson8m2-canvas - - const: amlogic,canvas + - enum: + - amlogic,meson8-canvas + - amlogic,meson8b-canvas + - amlogic,meson8m2-canvas + - const: amlogic,canvas - const: amlogic,canvas # GXBB and newer SoCs reg: diff --git a/Documentation/devicetree/bindings/sound/renesas,fsi.yaml b/Documentation/devicetree/bindings/sound/renesas,fsi.yaml index d1b65554e681..91cf4176abd5 100644 --- a/Documentation/devicetree/bindings/sound/renesas,fsi.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,fsi.yaml @@ -17,16 +17,16 @@ properties: oneOf: # for FSI2 SoC - items: - - enum: - - renesas,fsi2-sh73a0 - - renesas,fsi2-r8a7740 - - enum: - - renesas,sh_fsi2 + - enum: + - renesas,fsi2-sh73a0 + - renesas,fsi2-r8a7740 + - enum: + - renesas,sh_fsi2 # for Generic - items: - - enum: - - renesas,sh_fsi - - renesas,sh_fsi2 + - enum: + - renesas,sh_fsi + - renesas,sh_fsi2 reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/spi/qcom,spi-qcom-qspi.yaml b/Documentation/devicetree/bindings/spi/qcom,spi-qcom-qspi.yaml index 0cf470eaf2a0..406286149a6b 100644 --- a/Documentation/devicetree/bindings/spi/qcom,spi-qcom-qspi.yaml +++ b/Documentation/devicetree/bindings/spi/qcom,spi-qcom-qspi.yaml @@ -8,12 +8,12 @@ $schema: "http://devicetree.org/meta-schemas/core.yaml#" title: Qualcomm Quad Serial Peripheral Interface (QSPI) maintainers: - - Mukesh Savaliya <msavaliy(a)codeaurora.org> - - Akash Asthana <akashast(a)codeaurora.org> + - Mukesh Savaliya <msavaliy(a)codeaurora.org> + - Akash Asthana <akashast(a)codeaurora.org> -description: - The QSPI controller allows SPI protocol communication in single, dual, or quad - wire transmission modes for read/write access to slaves such as NOR flash. +description: The QSPI controller allows SPI protocol communication in single, + dual, or quad wire transmission modes for read/write access to slaves such + as NOR flash. allOf: - $ref: /spi/spi-controller.yaml# diff --git a/Documentation/devicetree/bindings/spi/renesas,hspi.yaml b/Documentation/devicetree/bindings/spi/renesas,hspi.yaml index c429cf4bea5b..f492cb9fea12 100644 --- a/Documentation/devicetree/bindings/spi/renesas,hspi.yaml +++ b/Documentation/devicetree/bindings/spi/renesas,hspi.yaml @@ -16,8 +16,8 @@ properties: compatible: items: - enum: - - renesas,hspi-r8a7778 # R-Car M1A - - renesas,hspi-r8a7779 # R-Car H1 + - renesas,hspi-r8a7778 # R-Car M1A + - renesas,hspi-r8a7779 # R-Car H1 - const: renesas,hspi reg: diff --git a/Documentation/devicetree/bindings/spi/spi-pl022.yaml b/Documentation/devicetree/bindings/spi/spi-pl022.yaml index dfb697c69341..22ba4e90655b 100644 --- a/Documentation/devicetree/bindings/spi/spi-pl022.yaml +++ b/Documentation/devicetree/bindings/spi/spi-pl022.yaml @@ -51,7 +51,7 @@ properties: pl022,rt: description: indicates the controller should run the message pump with realtime - priority to minimise the transfer latency on the bus (boolean) + priority to minimise the transfer latency on the bus (boolean) type: boolean dmas: diff --git a/Documentation/devicetree/bindings/spi/st,stm32-qspi.yaml b/Documentation/devicetree/bindings/spi/st,stm32-qspi.yaml index 3665a5fe6b7f..1a342ce1f798 100644 --- a/Documentation/devicetree/bindings/spi/st,stm32-qspi.yaml +++ b/Documentation/devicetree/bindings/spi/st,stm32-qspi.yaml @@ -24,8 +24,8 @@ properties: reg-names: items: - - const: qspi - - const: qspi_mm + - const: qspi + - const: qspi_mm clocks: maxItems: 1 diff --git a/Documentation/devicetree/bindings/sram/allwinner,sun4i-a10-system-control.yaml b/Documentation/devicetree/bindings/sram/allwinner,sun4i-a10-system-control.yaml index 4b5509436588..f5825935fd22 100644 --- a/Documentation/devicetree/bindings/sram/allwinner,sun4i-a10-system-control.yaml +++ b/Documentation/devicetree/bindings/sram/allwinner,sun4i-a10-system-control.yaml @@ -29,8 +29,8 @@ properties: - const: allwinner,sun4i-a10-system-control - const: allwinner,sun5i-a13-system-control - items: - - const: allwinner,sun7i-a20-system-control - - const: allwinner,sun4i-a10-system-control + - const: allwinner,sun7i-a20-system-control + - const: allwinner,sun4i-a10-system-control - const: allwinner,sun8i-a23-system-control - const: allwinner,sun8i-h3-system-control - const: allwinner,sun50i-a64-sram-controller @@ -38,11 +38,11 @@ properties: - const: allwinner,sun50i-a64-system-control - const: allwinner,sun50i-h5-system-control - items: - - const: allwinner,sun50i-h6-system-control - - const: allwinner,sun50i-a64-system-control + - const: allwinner,sun50i-h6-system-control + - const: allwinner,sun50i-a64-system-control - items: - - const: allwinner,suniv-f1c100s-system-control - - const: allwinner,sun4i-a10-system-control + - const: allwinner,suniv-f1c100s-system-control + - const: allwinner,sun4i-a10-system-control reg: maxItems: 1 @@ -69,44 +69,44 @@ patternProperties: - const: allwinner,sun4i-a10-sram-d - const: allwinner,sun50i-a64-sram-c - items: - - const: allwinner,sun5i-a13-sram-a3-a4 - - const: allwinner,sun4i-a10-sram-a3-a4 + - const: allwinner,sun5i-a13-sram-a3-a4 + - const: allwinner,sun4i-a10-sram-a3-a4 - items: - - const: allwinner,sun7i-a20-sram-a3-a4 - - const: allwinner,sun4i-a10-sram-a3-a4 + - const: allwinner,sun7i-a20-sram-a3-a4 + - const: allwinner,sun4i-a10-sram-a3-a4 - items: - - const: allwinner,sun5i-a13-sram-c1 - - const: allwinner,sun4i-a10-sram-c1 + - const: allwinner,sun5i-a13-sram-c1 + - const: allwinner,sun4i-a10-sram-c1 - items: - - const: allwinner,sun7i-a20-sram-c1 - - const: allwinner,sun4i-a10-sram-c1 + - const: allwinner,sun7i-a20-sram-c1 + - const: allwinner,sun4i-a10-sram-c1 - items: - - const: allwinner,sun8i-a23-sram-c1 - - const: allwinner,sun4i-a10-sram-c1 + - const: allwinner,sun8i-a23-sram-c1 + - const: allwinner,sun4i-a10-sram-c1 - items: - - const: allwinner,sun8i-h3-sram-c1 - - const: allwinner,sun4i-a10-sram-c1 + - const: allwinner,sun8i-h3-sram-c1 + - const: allwinner,sun4i-a10-sram-c1 - items: - - const: allwinner,sun50i-a64-sram-c1 - - const: allwinner,sun4i-a10-sram-c1 + - const: allwinner,sun50i-a64-sram-c1 + - const: allwinner,sun4i-a10-sram-c1 - items: - - const: allwinner,sun50i-h5-sram-c1 - - const: allwinner,sun4i-a10-sram-c1 + - const: allwinner,sun50i-h5-sram-c1 + - const: allwinner,sun4i-a10-sram-c1 - items: - - const: allwinner,sun50i-h6-sram-c1 - - const: allwinner,sun4i-a10-sram-c1 + - const: allwinner,sun50i-h6-sram-c1 + - const: allwinner,sun4i-a10-sram-c1 - items: - - const: allwinner,sun5i-a13-sram-d - - const: allwinner,sun4i-a10-sram-d + - const: allwinner,sun5i-a13-sram-d + - const: allwinner,sun4i-a10-sram-d - items: - - const: allwinner,sun7i-a20-sram-d - - const: allwinner,sun4i-a10-sram-d + - const: allwinner,sun7i-a20-sram-d + - const: allwinner,sun4i-a10-sram-d - items: - - const: allwinner,suniv-f1c100s-sram-d - - const: allwinner,sun4i-a10-sram-d + - const: allwinner,suniv-f1c100s-sram-d + - const: allwinner,sun4i-a10-sram-d - items: - - const: allwinner,sun50i-h6-sram-c - - const: allwinner,sun50i-a64-sram-c + - const: allwinner,sun50i-h6-sram-c + - const: allwinner,sun50i-a64-sram-c required: - "#address-cells" diff --git a/Documentation/devicetree/bindings/thermal/amlogic,thermal.yaml b/Documentation/devicetree/bindings/thermal/amlogic,thermal.yaml index e43ec50bda37..999c6b365f1d 100644 --- a/Documentation/devicetree/bindings/thermal/amlogic,thermal.yaml +++ b/Documentation/devicetree/bindings/thermal/amlogic,thermal.yaml @@ -13,11 +13,11 @@ description: Binding for Amlogic Thermal properties: compatible: - items: - - enum: - - amlogic,g12a-cpu-thermal - - amlogic,g12a-ddr-thermal - - const: amlogic,g12a-thermal + items: + - enum: + - amlogic,g12a-cpu-thermal + - amlogic,g12a-ddr-thermal + - const: amlogic,g12a-thermal reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/timer/arm,arch_timer.yaml b/Documentation/devicetree/bindings/timer/arm,arch_timer.yaml index fa255672e8e5..135186f83925 100644 --- a/Documentation/devicetree/bindings/timer/arm,arch_timer.yaml +++ b/Documentation/devicetree/bindings/timer/arm,arch_timer.yaml @@ -28,10 +28,10 @@ properties: - arm,armv7-timer - items: - enum: - - arm,armv7-timer + - arm,armv7-timer - items: - enum: - - arm,armv8-timer + - arm,armv8-timer interrupts: items: diff --git a/Documentation/devicetree/bindings/timer/arm,arch_timer_mmio.yaml b/Documentation/devicetree/bindings/timer/arm,arch_timer_mmio.yaml index 582bbef62b95..6ff718ede184 100644 --- a/Documentation/devicetree/bindings/timer/arm,arch_timer_mmio.yaml +++ b/Documentation/devicetree/bindings/timer/arm,arch_timer_mmio.yaml @@ -20,7 +20,7 @@ properties: compatible: items: - enum: - - arm,armv7-timer-mem + - arm,armv7-timer-mem reg: maxItems: 1 @@ -77,7 +77,7 @@ patternProperties: - description: physical timer irq - description: virtual timer irq - reg : + reg: minItems: 1 maxItems: 2 items: diff --git a/Documentation/devicetree/bindings/usb/dwc2.yaml b/Documentation/devicetree/bindings/usb/dwc2.yaml index 0d6d850a7f17..fb2f62aef5fa 100644 --- a/Documentation/devicetree/bindings/usb/dwc2.yaml +++ b/Documentation/devicetree/bindings/usb/dwc2.yaml @@ -62,14 +62,14 @@ properties: resets: items: - - description: common reset - - description: ecc reset + - description: common reset + - description: ecc reset minItems: 1 reset-names: items: - - const: dwc2 - - const: dwc2-ecc + - const: dwc2 + - const: dwc2-ecc minItems: 1 phys: -- 2.20.1

12 23

[PATCH] soundwire: intel: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
by Kuninori Morimoto 20 Apr '20

20 Apr '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Now ALSA SoC needs to use asoc_rtd_to_codec(), otherwise, it will be compile error. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- drivers/soundwire/intel.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index 3c83e76c6bf9..ed8d576bf5dc 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -669,11 +669,11 @@ static int sdw_stream_setup(struct snd_pcm_substream *substream, /* Set stream pointer on all CODEC DAIs */ for (i = 0; i < rtd->num_codecs; i++) { - ret = snd_soc_dai_set_sdw_stream(rtd->codec_dais[i], sdw_stream, + ret = snd_soc_dai_set_sdw_stream(asoc_rtd_to_codec(rtd, i), sdw_stream, substream->stream); if (ret < 0) { dev_err(dai->dev, "failed to set stream pointer on codec dai %s", - rtd->codec_dais[i]->name); + asoc_rtd_to_codec(rtd, i)->name); goto release_stream; } } -- 2.17.1

3 6

[PATCH] ASoC: wm89xx: Fix build error without CONFIG_I2C
by YueHaibing 20 Apr '20

20 Apr '20

sound/soc/codecs/wm8900.o: In function `wm8900_i2c_probe': wm8900.c:(.text+0xa36): undefined reference to `__devm_regmap_init_i2c' sound/soc/codecs/wm8900.o: In function `wm8900_modinit': wm8900.c:(.init.text+0xb): undefined reference to `i2c_register_driver' sound/soc/codecs/wm8900.o: In function `wm8900_exit': wm8900.c:(.exit.text+0x8): undefined reference to `i2c_del_driver' sound/soc/codecs/wm8988.o: In function `wm8988_i2c_probe': wm8988.c:(.text+0x857): undefined reference to `__devm_regmap_init_i2c' sound/soc/codecs/wm8988.o: In function `wm8988_modinit': wm8988.c:(.init.text+0xb): undefined reference to `i2c_register_driver' sound/soc/codecs/wm8988.o: In function `wm8988_exit': wm8988.c:(.exit.text+0x8): undefined reference to `i2c_del_driver' sound/soc/codecs/wm8995.o: In function `wm8995_i2c_probe': wm8995.c:(.text+0x1c4f): undefined reference to `__devm_regmap_init_i2c' sound/soc/codecs/wm8995.o: In function `wm8995_modinit': wm8995.c:(.init.text+0xb): undefined reference to `i2c_register_driver' sound/soc/codecs/wm8995.o: In function `wm8995_exit': wm8995.c:(.exit.text+0x8): undefined reference to `i2c_del_driver' Add missing I2C dependency to fix this. Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: ea00d95200d02ece ("ASoC: Use imply for SND_SOC_ALL_CODECS") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/Kconfig | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e6a0c5d05fa5..2d8d0bba0c0a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1525,6 +1525,7 @@ config SND_SOC_WM8804_SPI config SND_SOC_WM8900 tristate + depends on I2C config SND_SOC_WM8903 tristate "Wolfson Microelectronics WM8903 CODEC" @@ -1576,6 +1577,7 @@ config SND_SOC_WM8985 config SND_SOC_WM8988 tristate + depends on I2C config SND_SOC_WM8990 tristate @@ -1594,6 +1596,7 @@ config SND_SOC_WM8994 config SND_SOC_WM8995 tristate + depends on I2C config SND_SOC_WM8996 tristate -- 2.17.1

3 5

[PATCH] ASoC: wcd934x: remove unneeded semicolon
by Jason Yan 20 Apr '20

20 Apr '20

Fix the following coccicheck warning: sound/soc/codecs/wcd934x.c:3621:2-3: Unneeded semicolon sound/soc/codecs/wcd934x.c:4270:2-3: Unneeded semicolon sound/soc/codecs/wcd934x.c:4018:2-3: Unneeded semicolon sound/soc/codecs/wcd934x.c:4043:2-3: Unneeded semicolon sound/soc/codecs/wcd934x.c:3804:2-3: Unneeded semicolon sound/soc/codecs/wcd934x.c:3866:2-3: Unneeded semicolon sound/soc/codecs/wcd934x.c:3596:2-3: Unneeded semicolon sound/soc/codecs/wcd934x.c:3542:2-3: Unneeded semicolon sound/soc/codecs/wcd934x.c:3673:2-3: Unneeded semicolon sound/soc/codecs/wcd934x.c:3723:2-3: Unneeded semicolon sound/soc/codecs/wcd934x.c:2492:2-3: Unneeded semicolon sound/soc/codecs/wcd934x.c:1790:3-4: Unneeded semicolon sound/soc/codecs/wcd934x.c:1806:3-4: Unneeded semicolon sound/soc/codecs/wcd934x.c:1812:2-3: Unneeded semicolon Signed-off-by: Jason Yan <yanaijie(a)huawei.com> --- sound/soc/codecs/wcd934x.c | 28 ++++++++++++++-------------- 1 file changed, 14 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index 5269857e2746..a514eb32e3e6 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -1787,7 +1787,7 @@ static int wcd934x_hw_params(struct snd_pcm_substream *substream, params_rate(params)); return -EINVAL; - }; + } ret = wcd934x_set_decimator_rate(dai, tx_fs_rate, params_rate(params)); @@ -1803,13 +1803,13 @@ static int wcd934x_hw_params(struct snd_pcm_substream *substream, dev_err(wcd->dev, "Invalid format 0x%x\n", params_width(params)); return -EINVAL; - }; + } break; default: dev_err(wcd->dev, "Invalid stream type %d\n", substream->stream); return -EINVAL; - }; + } wcd->dai[dai->id].sconfig.rate = params_rate(params); wcd934x_slim_set_hw_params(wcd, &wcd->dai[dai->id], substream->stream); @@ -2489,7 +2489,7 @@ static int wcd934x_compander_set(struct snd_kcontrol *kc, break; default: break; - }; + } return 0; } @@ -3539,7 +3539,7 @@ static int wcd934x_codec_enable_mix_path(struct snd_soc_dapm_widget *w, val += offset_val; snd_soc_component_write(comp, gain_reg, val); break; - }; + } return 0; } @@ -3593,7 +3593,7 @@ static int wcd934x_codec_enable_main_path(struct snd_soc_dapm_widget *w, snd_soc_component_write(comp, gain_reg, snd_soc_component_read32(comp, gain_reg)); break; - }; + } return 0; } @@ -3618,7 +3618,7 @@ static int wcd934x_codec_ear_dac_event(struct snd_soc_dapm_widget *w, wcd_clsh_ctrl_set_state(wcd->clsh_ctrl, WCD_CLSH_EVENT_POST_PA, WCD_CLSH_STATE_EAR, CLS_H_NORMAL); break; - }; + } return 0; } @@ -3670,7 +3670,7 @@ static int wcd934x_codec_hphl_dac_event(struct snd_soc_dapm_widget *w, break; default: break; - }; + } return 0; } @@ -3720,7 +3720,7 @@ static int wcd934x_codec_hphr_dac_event(struct snd_soc_dapm_widget *w, break; default: break; - }; + } return 0; } @@ -3801,7 +3801,7 @@ static int wcd934x_codec_enable_hphl_pa(struct snd_soc_dapm_widget *w, */ usleep_range(20000, 20100); break; - }; + } return 0; } @@ -3863,7 +3863,7 @@ static int wcd934x_codec_enable_hphr_pa(struct snd_soc_dapm_widget *w, */ usleep_range(20000, 20100); break; - }; + } return 0; } @@ -4015,7 +4015,7 @@ static int wcd934x_codec_enable_dmic(struct snd_soc_dapm_widget *w, dev_err(comp->dev, "%s: Invalid DMIC Selection\n", __func__); return -EINVAL; - }; + } switch (event) { case SND_SOC_DAPM_PRE_PMU: @@ -4040,7 +4040,7 @@ static int wcd934x_codec_enable_dmic(struct snd_soc_dapm_widget *w, snd_soc_component_update_bits(comp, dmic_clk_reg, dmic_clk_en, 0); break; - }; + } return 0; } @@ -4267,7 +4267,7 @@ static int wcd934x_codec_enable_dec(struct snd_soc_dapm_widget *w, WCD934X_DEC_PWR_LVL_MASK, WCD934X_DEC_PWR_LVL_DF); break; - }; + } out: kfree(wname); return ret; -- 2.21.1

2 1

[PATCH 0/2] ASoC: Fix dependency issues of SND_SOC
by Wei Li 20 Apr '20

20 Apr '20

Fix dependency issues of SND_SOC introduced by commit ea00d95200d02 ("ASoC: Use imply for SND_SOC_ALL_CODECS"). Wei Li (2): ASoC: wm89xx: Fix build errors caused by I2C dependency ASoC: Fix wrong dependency of da7210 and wm8983 sound/soc/codecs/Kconfig | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) -- 2.17.1

2 3

[PATCH] ASoC: wcd934x: remove unnecessary comparisons to bool
by Jason Yan 20 Apr '20

20 Apr '20

Fix the following coccicheck warning: sound/soc/codecs/wcd934x.c:3881:8-17: WARNING: Comparison of 0/1 to bool variable Signed-off-by: Jason Yan <yanaijie(a)huawei.com> --- sound/soc/codecs/wcd934x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index a514eb32e3e6..531b8b79e55f 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -3878,7 +3878,7 @@ static u32 wcd934x_get_dmic_sample_rate(struct snd_soc_component *comp, u16 adc_mux_ctl_reg, tx_fs_reg; u32 dmic_fs; - while (dec_found == 0 && adc_mux_index < WCD934X_MAX_VALID_ADC_MUX) { + while (!dec_found && adc_mux_index < WCD934X_MAX_VALID_ADC_MUX) { if (adc_mux_index < 4) { adc_mux_ctl_reg = WCD934X_CDC_TX_INP_MUX_ADC_MUX0_CFG0 + (adc_mux_index * 2); -- 2.21.1

2 1

[PATCH v3] ASoC: Intel: boards: split woofer and tweeter support
by mac.chiang＠intel.com 20 Apr '20

20 Apr '20

From: Mac Chiang <mac.chiang(a)intel.com> Support Woofer stereo speakers by default and optionally Tweeter stereo speakers with a DMI quirk Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- Changelog: v2: - considering woofer speakers settings as general. - separate out the tweeter speakers and consistency of woofer+tweeter when hardware-specific. - check NULL condition if devm_kzalloc are failed v3: - fixed multiple alignments - add invalid value return --- sound/soc/intel/boards/cml_rt1011_rt5682.c | 222 +++++++++++++++++++++++------ 1 file changed, 180 insertions(+), 42 deletions(-) diff --git a/sound/soc/intel/boards/cml_rt1011_rt5682.c b/sound/soc/intel/boards/cml_rt1011_rt5682.c index ed6c26a..2872345 100644 --- a/sound/soc/intel/boards/cml_rt1011_rt5682.c +++ b/sound/soc/intel/boards/cml_rt1011_rt5682.c @@ -30,6 +30,36 @@ #define CML_RT5682_CODEC_DAI "rt5682-aif1" #define NAME_SIZE 32 +#define SOF_RT1011_SPEAKER_WL BIT(0) +#define SOF_RT1011_SPEAKER_WR BIT(1) +#define SOF_RT1011_SPEAKER_TL BIT(2) +#define SOF_RT1011_SPEAKER_TR BIT(3) +#define SPK_CH 4 + +/* Default: Woofer speakers */ +static unsigned long sof_rt1011_quirk = SOF_RT1011_SPEAKER_WL | + SOF_RT1011_SPEAKER_WR; + +static int sof_rt1011_quirk_cb(const struct dmi_system_id *id) +{ + sof_rt1011_quirk = (unsigned long)id->driver_data; + return 1; +} + +static const struct dmi_system_id sof_rt1011_quirk_table[] = { + { + .callback = sof_rt1011_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + DMI_MATCH(DMI_PRODUCT_NAME, "Helios"), + }, + .driver_data = (void *)(SOF_RT1011_SPEAKER_WL | SOF_RT1011_SPEAKER_WR | + SOF_RT1011_SPEAKER_TL | SOF_RT1011_SPEAKER_TR), + }, + { + } +}; + static struct snd_soc_jack hdmi_jack[3]; struct hdmi_pcm { @@ -48,15 +78,16 @@ struct card_private { static const struct snd_kcontrol_new cml_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone Jack"), SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("TL Ext Spk"), - SOC_DAPM_PIN_SWITCH("TR Ext Spk"), SOC_DAPM_PIN_SWITCH("WL Ext Spk"), SOC_DAPM_PIN_SWITCH("WR Ext Spk"), }; +static const struct snd_kcontrol_new cml_rt1011_tt_controls[] = { + SOC_DAPM_PIN_SWITCH("TL Ext Spk"), + SOC_DAPM_PIN_SWITCH("TR Ext Spk"), +}; + static const struct snd_soc_dapm_widget cml_rt1011_rt5682_widgets[] = { - SND_SOC_DAPM_SPK("TL Ext Spk", NULL), - SND_SOC_DAPM_SPK("TR Ext Spk", NULL), SND_SOC_DAPM_SPK("WL Ext Spk", NULL), SND_SOC_DAPM_SPK("WR Ext Spk", NULL), SND_SOC_DAPM_HP("Headphone Jack", NULL), @@ -64,10 +95,13 @@ static const struct snd_soc_dapm_widget cml_rt1011_rt5682_widgets[] = { SND_SOC_DAPM_MIC("SoC DMIC", NULL), }; +static const struct snd_soc_dapm_widget cml_rt1011_tt_widgets[] = { + SND_SOC_DAPM_SPK("TL Ext Spk", NULL), + SND_SOC_DAPM_SPK("TR Ext Spk", NULL), +}; + static const struct snd_soc_dapm_route cml_rt1011_rt5682_map[] = { - /*speaker*/ - {"TL Ext Spk", NULL, "TL SPO"}, - {"TR Ext Spk", NULL, "TR SPO"}, + /*WL/WR speaker*/ {"WL Ext Spk", NULL, "WL SPO"}, {"WR Ext Spk", NULL, "WR SPO"}, @@ -82,6 +116,12 @@ static const struct snd_soc_dapm_route cml_rt1011_rt5682_map[] = { {"DMic", NULL, "SoC DMIC"}, }; +static const struct snd_soc_dapm_route cml_rt1011_tt_map[] = { + /*TL/TR speaker*/ + {"TL Ext Spk", NULL, "TL SPO" }, + {"TR Ext Spk", NULL, "TR SPO" }, +}; + static int cml_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) { struct card_private *ctx = snd_soc_card_get_drvdata(rtd->card); @@ -121,6 +161,35 @@ static int cml_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) return ret; }; +static int cml_rt1011_spk_init(struct snd_soc_pcm_runtime *rtd) +{ + int ret = 0; + struct snd_soc_card *card = rtd->card; + + if (sof_rt1011_quirk & (SOF_RT1011_SPEAKER_TL | + SOF_RT1011_SPEAKER_TR)) { + + ret = snd_soc_add_card_controls(card, cml_rt1011_tt_controls, + ARRAY_SIZE(cml_rt1011_tt_controls)); + if (ret) + return ret; + + ret = snd_soc_dapm_new_controls(&card->dapm, + cml_rt1011_tt_widgets, + ARRAY_SIZE(cml_rt1011_tt_widgets)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(&card->dapm, cml_rt1011_tt_map, + ARRAY_SIZE(cml_rt1011_tt_map)); + + if (ret) + return ret; + } + + return ret; +} + static int cml_rt5682_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -191,30 +260,38 @@ static int cml_rt1011_hw_params(struct snd_pcm_substream *substream, * The feedback is captured for each codec individually. * Hence all 4 codecs use 1 Tx slot each for feedback. */ - if (!strcmp(codec_dai->component->name, "i2c-10EC1011:00")) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, - 0x4, 0x1, 4, 24); - if (ret < 0) - break; - } - if (!strcmp(codec_dai->component->name, "i2c-10EC1011:02")) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, - 0x1, 0x1, 4, 24); - if (ret < 0) - break; + if (sof_rt1011_quirk & (SOF_RT1011_SPEAKER_WL | + SOF_RT1011_SPEAKER_WR)) { + if (!strcmp(codec_dai->component->name, "i2c-10EC1011:00")) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, + 0x4, 0x1, 4, 24); + if (ret < 0) + break; + } + + if (!strcmp(codec_dai->component->name, "i2c-10EC1011:01")) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, + 0x8, 0x2, 4, 24); + if (ret < 0) + break; + } } - /* TDM Rx slot 2 is used for Right Woofer & Tweeters pair */ - if (!strcmp(codec_dai->component->name, "i2c-10EC1011:01")) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, - 0x8, 0x2, 4, 24); - if (ret < 0) - break; - } - if (!strcmp(codec_dai->component->name, "i2c-10EC1011:03")) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, - 0x2, 0x2, 4, 24); - if (ret < 0) - break; + + if (sof_rt1011_quirk & (SOF_RT1011_SPEAKER_TL | + SOF_RT1011_SPEAKER_TR)) { + if (!strcmp(codec_dai->component->name, "i2c-10EC1011:02")) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, + 0x1, 0x1, 4, 24); + if (ret < 0) + break; + } + + if (!strcmp(codec_dai->component->name, "i2c-10EC1011:03")) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, + 0x2, 0x2, 4, 24); + if (ret < 0) + break; + } } } if (ret < 0) @@ -302,9 +379,7 @@ SND_SOC_DAILINK_DEF(ssp1_pin, SND_SOC_DAILINK_DEF(ssp1_codec, DAILINK_COMP_ARRAY( /* WL */ COMP_CODEC("i2c-10EC1011:00", CML_RT1011_CODEC_DAI), - /* WR */ COMP_CODEC("i2c-10EC1011:01", CML_RT1011_CODEC_DAI), - /* TL */ COMP_CODEC("i2c-10EC1011:02", CML_RT1011_CODEC_DAI), - /* TR */ COMP_CODEC("i2c-10EC1011:03", CML_RT1011_CODEC_DAI))); + /* WR */ COMP_CODEC("i2c-10EC1011:01", CML_RT1011_CODEC_DAI))); SND_SOC_DAILINK_DEF(dmic_pin, DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin"))); @@ -398,6 +473,7 @@ static struct snd_soc_dai_link cml_rt1011_rt5682_dailink[] = { .dpcm_playback = 1, .dpcm_capture = 1, /* Capture stream provides Feedback */ .no_pcm = 1, + .init = cml_rt1011_spk_init, .ops = &cml_rt1011_ops, SND_SOC_DAILINK_REG(ssp1_pin, ssp1_codec, platform), }, @@ -412,14 +488,6 @@ static struct snd_soc_codec_conf rt1011_conf[] = { .dlc = COMP_CODEC_CONF("i2c-10EC1011:01"), .name_prefix = "WR", }, - { - .dlc = COMP_CODEC_CONF("i2c-10EC1011:02"), - .name_prefix = "TL", - }, - { - .dlc = COMP_CODEC_CONF("i2c-10EC1011:03"), - .name_prefix = "TR", - }, }; /* Cometlake audio machine driver for RT1011 and RT5682 */ @@ -441,10 +509,12 @@ static struct snd_soc_card snd_soc_card_cml = { static int snd_cml_rt1011_probe(struct platform_device *pdev) { + struct snd_soc_dai_link_component *rt1011_dais_components; + struct snd_soc_codec_conf *rt1011_dais_confs; struct card_private *ctx; struct snd_soc_acpi_mach *mach; const char *platform_name; - int ret; + int ret, i; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -455,6 +525,73 @@ static int snd_cml_rt1011_probe(struct platform_device *pdev) snd_soc_card_cml.dev = &pdev->dev; platform_name = mach->mach_params.platform; + dmi_check_system(sof_rt1011_quirk_table); + + dev_info(&pdev->dev, "sof_rt1011_quirk = %lx\n", sof_rt1011_quirk); + + if (sof_rt1011_quirk & (SOF_RT1011_SPEAKER_TL | + SOF_RT1011_SPEAKER_TR)) { + rt1011_dais_confs = devm_kzalloc(&pdev->dev, + sizeof(struct snd_soc_codec_conf) * + SPK_CH, GFP_KERNEL); + + if (!rt1011_dais_confs) + return -ENOMEM; + + rt1011_dais_components = devm_kzalloc(&pdev->dev, + sizeof(struct snd_soc_dai_link_component) * + SPK_CH, GFP_KERNEL); + + if (!rt1011_dais_components) + return -ENOMEM; + + for (i = 0; i < SPK_CH; i++) { + rt1011_dais_confs[i].dlc.name = devm_kasprintf(&pdev->dev, + GFP_KERNEL, + "i2c-10EC1011:0%d", + i); + + if (!rt1011_dais_confs[i].dlc.name) + return -ENOMEM; + + switch (i) { + case 0: + rt1011_dais_confs[i].name_prefix = "WL"; + break; + case 1: + rt1011_dais_confs[i].name_prefix = "WR"; + break; + case 2: + rt1011_dais_confs[i].name_prefix = "TL"; + break; + case 3: + rt1011_dais_confs[i].name_prefix = "TR"; + break; + default: + return -EINVAL; + } + rt1011_dais_components[i].name = devm_kasprintf(&pdev->dev, + GFP_KERNEL, + "i2c-10EC1011:0%d", + i); + if (!rt1011_dais_components[i].name) + return -ENOMEM; + + rt1011_dais_components[i].dai_name = CML_RT1011_CODEC_DAI; + } + + snd_soc_card_cml.codec_conf = rt1011_dais_confs; + snd_soc_card_cml.num_configs = SPK_CH; + + for (i = 0; i < ARRAY_SIZE(cml_rt1011_rt5682_dailink); i++) { + if (!strcmp(cml_rt1011_rt5682_dailink[i].codecs->dai_name, + CML_RT1011_CODEC_DAI)) { + cml_rt1011_rt5682_dailink[i].codecs = rt1011_dais_components; + cml_rt1011_rt5682_dailink[i].num_codecs = SPK_CH; + } + } + } + /* set platform name for each dailink */ ret = snd_soc_fixup_dai_links_platform_name(&snd_soc_card_cml, platform_name); @@ -482,5 +619,6 @@ MODULE_DESCRIPTION("Cometlake Audio Machine driver - RT1011 and RT5682 in I2S mo MODULE_AUTHOR("Naveen Manohar <naveen.m(a)intel.com>"); MODULE_AUTHOR("Sathya Prakash M R <sathya.prakash.m.r(a)intel.com>"); MODULE_AUTHOR("Shuming Fan <shumingf(a)realtek.com>"); +MODULE_AUTHOR("Mac Chiang <mac.chiang(a)intel.com>"); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("platform:cml_rt1011_rt5682"); -- 2.7.4

2 1

[PATCH] ASoC: wcd9335: remove unneeded semicolon
by Jason Yan 20 Apr '20

20 Apr '20

Fix the following coccicheck warning: sound/soc/codecs/wcd9335.c:2606:2-3: Unneeded semicolon sound/soc/codecs/wcd9335.c:3619:2-3: Unneeded semicolon sound/soc/codecs/wcd9335.c:2849:2-3: Unneeded semicolon sound/soc/codecs/wcd9335.c:2955:2-3: Unneeded semicolon sound/soc/codecs/wcd9335.c:2988:2-3: Unneeded semicolon sound/soc/codecs/wcd9335.c:3960:2-3: Unneeded semicolon sound/soc/codecs/wcd9335.c:3776:2-3: Unneeded semicolon sound/soc/codecs/wcd9335.c:3924:2-3: Unneeded semicolon sound/soc/codecs/wcd9335.c:3355:2-3: Unneeded semicolon sound/soc/codecs/wcd9335.c:3832:2-3: Unneeded semicolon sound/soc/codecs/wcd9335.c:3079:2-3: Unneeded semicolon sound/soc/codecs/wcd9335.c:3089:2-3: Unneeded semicolon sound/soc/codecs/wcd9335.c:3232:2-3: Unneeded semicolon sound/soc/codecs/wcd9335.c:3878:2-3: Unneeded semicolon sound/soc/codecs/wcd9335.c:3578:2-3: Unneeded semicolon sound/soc/codecs/wcd9335.c:3728:2-3: Unneeded semicolon sound/soc/codecs/wcd9335.c:1922:3-4: Unneeded semicolon sound/soc/codecs/wcd9335.c:1938:3-4: Unneeded semicolon sound/soc/codecs/wcd9335.c:1944:2-3: Unneeded semicolon sound/soc/codecs/wcd9335.c:3144:2-3: Unneeded semicolon sound/soc/codecs/wcd9335.c:2568:2-3: Unneeded semicolon sound/soc/codecs/wcd9335.c:2219:2-3: Unneeded semicolon Signed-off-by: Jason Yan <yanaijie(a)huawei.com> --- sound/soc/codecs/wcd9335.c | 44 +++++++++++++++++++------------------- 1 file changed, 22 insertions(+), 22 deletions(-) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 700cc1212770..fb073f4dc7ed 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -1919,7 +1919,7 @@ static int wcd9335_hw_params(struct snd_pcm_substream *substream, __func__, params_rate(params)); return -EINVAL; - }; + } ret = wcd9335_set_decimator_rate(dai, tx_fs_rate, params_rate(params)); @@ -1935,13 +1935,13 @@ static int wcd9335_hw_params(struct snd_pcm_substream *substream, dev_err(wcd->dev, "%s: Invalid format 0x%x\n", __func__, params_width(params)); return -EINVAL; - }; + } break; default: dev_err(wcd->dev, "Invalid stream type %d\n", substream->stream); return -EINVAL; - }; + } wcd->dai[dai->id].sconfig.rate = params_rate(params); wcd9335_slim_set_hw_params(wcd, &wcd->dai[dai->id], substream->stream); @@ -2216,7 +2216,7 @@ static int wcd9335_set_compander(struct snd_kcontrol *kc, break; default: break; - }; + } return 0; } @@ -2565,7 +2565,7 @@ static int wcd9335_micbias_control(struct snd_soc_component *component, 0xC0, 0x00); } break; - }; + } return 0; } @@ -2603,7 +2603,7 @@ static int __wcd9335_codec_enable_micbias(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_POST_PMD: wcd9335_micbias_control(comp, micb_num, MICB_DISABLE, true); break; - }; + } return 0; } @@ -2846,7 +2846,7 @@ static int wcd9335_codec_enable_dec(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_POST_PMD: snd_soc_component_update_bits(comp, tx_vol_ctl_reg, 0x10, 0x00); break; - }; + } out: kfree(wname); return ret; @@ -2952,7 +2952,7 @@ static int wcd9335_codec_enable_dmic(struct snd_soc_dapm_widget *w, dev_err(comp->dev, "%s: Invalid DMIC Selection\n", __func__); return -EINVAL; - }; + } switch (event) { case SND_SOC_DAPM_PRE_PMU: @@ -2985,7 +2985,7 @@ static int wcd9335_codec_enable_dmic(struct snd_soc_dapm_widget *w, dmic_rate_val << dmic_rate_shift); } break; - }; + } return 0; } @@ -3076,7 +3076,7 @@ static int wcd9335_codec_enable_mix_path(struct snd_soc_dapm_widget *w, dev_err(comp->dev, "%s: No gain register avail for %s\n", __func__, w->name); return 0; - }; + } switch (event) { case SND_SOC_DAPM_POST_PMU: @@ -3086,7 +3086,7 @@ static int wcd9335_codec_enable_mix_path(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_POST_PMD: break; - }; + } return 0; } @@ -3141,7 +3141,7 @@ static u16 wcd9335_interp_get_primary_reg(u16 reg, u16 *ind) prim_int_reg = WCD9335_CDC_RX8_RX_PATH_CTL; *ind = 8; break; - }; + } return prim_int_reg; } @@ -3229,7 +3229,7 @@ static int wcd9335_codec_enable_prim_interpolator( wcd9335_codec_hd2_control(comp, prim_int_reg, event); } break; - }; + } return 0; } @@ -3352,7 +3352,7 @@ static int wcd9335_codec_enable_interpolator(struct snd_soc_dapm_widget *w, wcd9335_config_compander(comp, w->shift, event); wcd9335_codec_enable_prim_interpolator(comp, reg, event); break; - }; + } return 0; } @@ -3575,7 +3575,7 @@ static int wcd9335_codec_hphl_dac_event(struct snd_soc_dapm_widget *w, ((hph_mode == CLS_H_LOHIFI) ? CLS_H_HIFI : hph_mode)); break; - }; + } return 0; } @@ -3616,7 +3616,7 @@ static int wcd9335_codec_ear_dac_event(struct snd_soc_dapm_widget *w, wcd_clsh_ctrl_set_state(wcd->clsh_ctrl, WCD_CLSH_EVENT_POST_PA, WCD_CLSH_STATE_EAR, CLS_H_NORMAL); break; - }; + } return 0; } @@ -3725,7 +3725,7 @@ static int wcd9335_codec_hphr_dac_event(struct snd_soc_dapm_widget *w, WCD_CLSH_STATE_HPHR, ((hph_mode == CLS_H_LOHIFI) ? CLS_H_HIFI : hph_mode)); break; - }; + } return 0; } @@ -3773,7 +3773,7 @@ static int wcd9335_codec_enable_hphl_pa(struct snd_soc_dapm_widget *w, */ usleep_range(5000, 5500); break; - }; + } return 0; } @@ -3829,7 +3829,7 @@ static int wcd9335_codec_enable_lineout_pa(struct snd_soc_dapm_widget *w, */ usleep_range(5000, 5500); break; - }; + } return 0; } @@ -3875,7 +3875,7 @@ static int wcd9335_codec_enable_rx_bias(struct snd_soc_dapm_widget *w, WCD9335_ANA_RX_BIAS_ENABLE_MASK, WCD9335_ANA_RX_BIAS_DISABLE); break; - }; + } return 0; } @@ -3921,7 +3921,7 @@ static int wcd9335_codec_enable_hphr_pa(struct snd_soc_dapm_widget *w, */ usleep_range(5000, 5500); break; - }; + } return 0; } @@ -3957,7 +3957,7 @@ static int wcd9335_codec_enable_ear_pa(struct snd_soc_dapm_widget *w, usleep_range(5000, 5500); break; - }; + } return 0; } -- 2.21.1

2 1

[PATCH] ASoC: topology: remove unneeded semicolon
by Jason Yan 20 Apr '20

20 Apr '20

Fix the following coccicheck warning: sound/soc/soc-topology.c:2344:2-3: Unneeded semicolon Signed-off-by: Jason Yan <yanaijie(a)huawei.com> --- sound/soc/soc-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 33e8d189ba2f..a5cd73742dc5 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -2390,7 +2390,7 @@ static int soc_tplg_link_elems_load(struct soc_tplg *tplg, tplg->pos += le32_to_cpu(hdr->size) + le32_to_cpu(hdr->payload_size); return 0; - }; + } /* check the element size and count */ link = (struct snd_soc_tplg_link_config *)tplg->pos; -- 2.21.1

2 1

[PATCH 0/2] ASoC: rsnd: multi-SSI setup fixes
by Matthias Blankertz 20 Apr '20

20 Apr '20

Fix rsnd_dai_call() operations being performed twice for the master SSI in multi-SSI setups, and fix the rsnd_ssi_stop operation for multi-SSI setups. The only visible effect of these issues was some "status check failed" spam when the rsnd_ssi_stop was called, but overall the code is cleaner now, and some questionable writes to the SSICR register which did not lead to any observable misbehaviour but were contrary to the datasheet are fixed. Mark: The first patch kind of reverts my "ASoC: rsnd: Fix parent SSI start/stop in multi-SSI mode" from a few days ago and achieves the same effect in a simpler fashion, if you would prefer a clean patch series based on v5.6 drop me a note. Greetings, Matthias Matthias Blankertz (2): ASoC: rsnd: Don't treat master SSI in multi SSI setup as parent ASoC: rsnd: Fix "status check failed" spam for multi-SSI sound/soc/sh/rcar/ssi.c | 18 +++++++++++++----- 1 file changed, 13 insertions(+), 5 deletions(-) base-commit: 15a5760cb8b6d5c1ebbf1d2e1f0b77380ab68a82 -- 2.26.1

3 4

[PATCH 0/2] ASoC: meson: fix codec-to-codec link setup
by Jerome Brunet 20 Apr '20

20 Apr '20

This patchset fixes the problem reported by Marc in this thread [0] The problem was due to an error in the meson card drivers which had the "no_pcm" dai_link property set on codec-to-codec links [0]: https://lore.kernel.org/r/20200417122732.GC5315@sirena.org.uk Jerome Brunet (2): ASoC: meson: axg-card: fix codec-to-codec link setup ASoC: meson: gx-card: fix codec-to-codec link setup sound/soc/meson/axg-card.c | 4 +++- sound/soc/meson/gx-card.c | 4 +++- 2 files changed, 6 insertions(+), 2 deletions(-) -- 2.25.2

2 3

[PATCH] ASoC: dapm: fixup dapm kcontrol widget
by Gyeongtaek Lee 20 Apr '20

20 Apr '20

snd_soc_dapm_kcontrol widget which is created by autodisable control should contain correct on_val, mask and shift because it is set when the widget is powered and changed value is applied on registers by following code in dapm_seq_run_coalesced(). mask |= w->mask << w->shift; if (w->power) value |= w->on_val << w->shift; else value |= w->off_val << w->shift; Shift on the mask in dapm_kcontrol_data_alloc() is removed to prevent double shift. And, on_val in dapm_kcontrol_set_value() is modified to get correct value in the dapm_seq_run_coalesced(). Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Cc: stable(a)vger.kernel.org --- sound/soc/soc-dapm.c | 20 +++++++++++++++++--- 1 file changed, 17 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a4143ca190d0..e2632841b321 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -423,7 +423,7 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, memset(&template, 0, sizeof(template)); template.reg = e->reg; - template.mask = e->mask << e->shift_l; + template.mask = e->mask; template.shift = e->shift_l; template.off_val = snd_soc_enum_item_to_val(e, 0); template.on_val = template.off_val; @@ -546,8 +546,22 @@ static bool dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol, if (data->value == value) return false; - if (data->widget) - data->widget->on_val = value; + if (data->widget) { + switch (dapm_kcontrol_get_wlist(kcontrol)->widgets[0]->id) { + case snd_soc_dapm_switch: + case snd_soc_dapm_mixer: + case snd_soc_dapm_mixer_named_ctl: + data->widget->on_val = value & data->widget->mask; + break; + case snd_soc_dapm_demux: + case snd_soc_dapm_mux: + data->widget->on_val = value >> data->widget->shift; + break; + default: + data->widget->on_val = value; + break; + } + } data->value = value; base-commit: ecdcf8a833d770347115f739a37583cdc6ddbf7a -- 2.21.0

2 1

[PATCH 0/2] ASoC: qdsp6: fix default FE dais and routings.
by Srinivas Kandagatla 20 Apr '20

20 Apr '20

QDSP6 Frontend dais can be configured to work in rx or tx or both rx/tx mode, however the default routing do not honour this DT configuration making sound card fail to probe. FE dais are also not fully honouring device tree configuration. Fix both of them. Originally issue was reported by Vinod Koul Srinivas Kandagatla (2): ASoC: qdsp6: q6asm-dai: only enable dais from device tree ASoC: qdsp6: q6routing: remove default routing sound/soc/qcom/qdsp6/q6asm-dai.c | 30 +++++++++++++++++++++++------- sound/soc/qcom/qdsp6/q6routing.c | 19 ------------------- 2 files changed, 23 insertions(+), 26 deletions(-) -- 2.21.0

4 11

wm8960 codec driver question
by Alexey Shinkin 20 Apr '20

20 Apr '20

Hi, I hope this is the right place to ask . In the driver ( soc/codecs/wm8960.c ) there is a function that sets pll - the wm8960_set_pll() , line 1185 . On exit , the function turns on the pll clocking like this : /* Turn it on */ snd_soc_component_update_bits(component, WM8960_POWER2, 0x1, 0x1); msleep(250); snd_soc_component_update_bits(component, WM8960_CLOCK1, 0x1, 0x1); What is the reason for the 250 ms delay ? I did not find anything related in the chip datasheet . I found the delay very annoying in my app that generates an audible feedback to touchscreen events. Looks like it is safe to comment out the msleep() call , could there be any implications ? Best regards, Alex Shinkin

2 1

[PATCH] ALSA: usx2y: Fix potential NULL dereference
by Takashi Iwai 20 Apr '20

20 Apr '20

The error handling code in usX2Y_rate_set() may hit a potential NULL dereference when an error occurs before allocating all us->urb[]. Add a proper NULL check for fixing the corner case. Reported-by: Lin Yi <teroincn(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/usx2y/usbusx2yaudio.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 37d290fe9d43..ecaf41265dcd 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -681,6 +681,8 @@ static int usX2Y_rate_set(struct usX2Ydev *usX2Y, int rate) us->submitted = 2*NOOF_SETRATE_URBS; for (i = 0; i < NOOF_SETRATE_URBS; ++i) { struct urb *urb = us->urb[i]; + if (!urb) + continue; if (urb->status) { if (!err) err = -ENODEV; -- 2.16.4

1 0

sound: usb: usx2y: is it a null pointer deference in function usX2Y_rate_set?
by 亿一 20 Apr '20

20 Apr '20

Hi, all: when reviewing function usX2Y_rate_set, here may exist a NULL pointer deference if kmalloc_array failed or usb_alloc_urb failed, in cleanup, we should judge whether us->urb[i] is NULL first. static int usX2Y_rate_set(struct usX2Ydev *usX2Y, int rate) { us = kzalloc(sizeof(*us) + sizeof(struct urb*) * NOOF_SETRATE_URBS, GFP_KERNEL); if (NULL == us) { err = -ENOMEM; goto cleanup; } usbdata = kmalloc_array(NOOF_SETRATE_URBS, sizeof(int), GFP_KERNEL); if (NULL == usbdata) { err = -ENOMEM; goto cleanup; } for (i = 0; i < NOOF_SETRATE_URBS; ++i) { if (NULL == (us->urb[i] = usb_alloc_urb(0, GFP_KERNEL))) { err = -ENOMEM; goto cleanup; } ... } ... cleanup: if (us) { us->submitted = 2*NOOF_SETRATE_URBS; for (i = 0; i < NOOF_SETRATE_URBS; ++i) { struct urb *urb = us->urb[i]; if (urb->status) { if (!err) err = -ENODEV; usb_kill_urb(urb); } usb_free_urb(urb);

2 2

PROBLEM: No sound on ASUS ROG Zenith Extreme in 5.6.5
by Johnathan Smithinovic 20 Apr '20

20 Apr '20

PROBLEM: No sound on ASUS ROG Zenith Extreme since 5.6.5 When I upgraded from 5.6.4 to 5.6.5 I noticed that my sound stopped working. My motherboard is a first generation "ASUS ROG Zenith Extreme". (I'm running Ubuntu 18.04 in case that matters.) I'm pretty sure it comes from stable kernel release commit 7f722d9aa00fde795834840cc84e13a94efb29ed (git://git.kernel.org/pub/scm/linux/kernel/git/stable/linux.git) which notes "commit 3c6fd1f07ed03a04debbb9a9d782205f1ef5e2ab upstream." ("ALSA: hda: Add driver blacklist"). When I comment out 'SND_PCI_QUIRK(0x1043, 0x874f, "ASUS ROG Zenith II / Strix", 0),' in sound/pci/hda/hda_intel.c my sound works again. (In case I chose incorrect places to write this mail to I'm sorry. I tried to follow https://www.kernel.org/doc/html/latest/admin-guide/reporting-bugs.html while keeping this report minimal.) Kind regards John

2 3

MIPS ioremap cleanups v2
by Christoph Hellwig 20 Apr '20

20 Apr '20

Hi Thomas, below is a bunch of cleanups for the MIPS ioremap code. Compile tested only. Changes since v1: - don't remove cpu_has_64bit_gp_regs - add a new prep patch to stop txx9aclc-ac97 from poking into ioremap internals

3 10

[PATCH 0/8] ASoC: add snd_compress_ops and replace
by Kuninori Morimoto 20 Apr '20

20 Apr '20

Hi Mark Current snd_soc_component_driver has compr_ops, but it is created for ALSA, not for ALSA SoC. This means it doesn't know the callback is for which component, thus, each callback needs to get component via lookup function. It is not good design, and will not work in the future, because ASoC can have multipul same name component which current lookup function can't handle. This patch-set adds new snd_compress_ops and call callbacks with component. Kuninori Morimoto (8): ASoC: soc-compress: add snd_compress_ops ASoC: codec: wm_adsp: use snd_compress_ops ASoC: uniphier: use snd_compress_ops ASoC: qcom: q6sp6: use snd_compress_ops ASoC: intel: atom: use snd_compress_ops ASoC: sof: use snd_compress_ops ASoC: sprd: use snd_compress_ops ASoC: soc-compress: remove snd_compr_ops include/sound/soc-component.h | 40 ++++++++- sound/soc/codecs/cs47l15.c | 9 +- sound/soc/codecs/cs47l24.c | 8 +- sound/soc/codecs/cs47l35.c | 9 +- sound/soc/codecs/cs47l85.c | 9 +- sound/soc/codecs/cs47l90.c | 9 +- sound/soc/codecs/cs47l92.c | 9 +- sound/soc/codecs/wm5102.c | 9 +- sound/soc/codecs/wm5110.c | 8 +- sound/soc/codecs/wm_adsp.c | 18 ++-- sound/soc/codecs/wm_adsp.h | 18 ++-- .../intel/atom/sst-mfld-platform-compress.c | 43 +++++---- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- sound/soc/intel/atom/sst-mfld-platform.h | 2 +- sound/soc/qcom/qdsp6/q6asm-dai.c | 51 ++++++----- sound/soc/soc-compress.c | 89 ++++++++++--------- sound/soc/sof/compress.c | 34 ++++--- sound/soc/sof/compress.h | 34 ++++--- sound/soc/sof/pcm.c | 4 +- sound/soc/sof/sof-priv.h | 2 +- sound/soc/sprd/sprd-pcm-compress.c | 49 +++++----- sound/soc/sprd/sprd-pcm-dma.c | 2 +- sound/soc/sprd/sprd-pcm-dma.h | 2 +- sound/soc/uniphier/aio-compress.c | 45 ++++++---- sound/soc/uniphier/aio-dma.c | 2 +- sound/soc/uniphier/aio.h | 2 +- 26 files changed, 299 insertions(+), 210 deletions(-) -- 2.17.1

2 10

[PATCH] ALSA: usb-audio: Add static mapping table for ALC1220-VB-based mobos
by Takashi Iwai 20 Apr '20

20 Apr '20

TRX40 mobos from MSI and others with ALC1220-VB USB-audio device need yet more quirks for the proper control names. This patch provides the mapping table for those boards, correcting the FU names for volume and mute controls as well as the terminal names for jack controls. It also improves build_connector_control() not to add the directional suffix blindly if the string is given from the mapping table. With this patch applied, the new UCM profiles will be effective. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206873 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer.c | 12 +++++++++--- sound/usb/mixer_maps.c | 24 +++++++++++++++++++++--- sound/usb/quirks-table.h | 14 ++++++++++++++ 3 files changed, 44 insertions(+), 6 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index e7b9040a54e6..ecd5036a0b44 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1776,8 +1776,10 @@ static void build_connector_control(struct usb_mixer_interface *mixer, { struct snd_kcontrol *kctl; struct usb_mixer_elem_info *cval; + const struct usbmix_name_map *map; - if (check_ignored_ctl(find_map(imap, term->id, 0))) + map = find_map(imap, term->id, 0); + if (check_ignored_ctl(map)) return; cval = kzalloc(sizeof(*cval), GFP_KERNEL); @@ -1809,8 +1811,12 @@ static void build_connector_control(struct usb_mixer_interface *mixer, usb_mixer_elem_info_free(cval); return; } - get_connector_control_name(mixer, term, is_input, kctl->id.name, - sizeof(kctl->id.name)); + + if (check_mapped_name(map, kctl->id.name, sizeof(kctl->id.name))) + strlcat(kctl->id.name, " Jack", sizeof(kctl->id.name)); + else + get_connector_control_name(mixer, term, is_input, kctl->id.name, + sizeof(kctl->id.name)); kctl->private_free = snd_usb_mixer_elem_free; snd_usb_mixer_add_control(&cval->head, kctl); } diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index b4e77000f441..fb2c4a992951 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -369,6 +369,24 @@ static const struct usbmix_name_map asus_rog_map[] = { {} }; +/* TRX40 mobos with Realtek ALC1220-VB */ +static const struct usbmix_name_map trx40_mobo_map[] = { + { 18, NULL }, /* OT, IEC958 - broken response, disabled */ + { 19, NULL, 12 }, /* FU, Input Gain Pad - broken response, disabled */ + { 16, "Speaker" }, /* OT */ + { 22, "Speaker Playback" }, /* FU */ + { 7, "Line" }, /* IT */ + { 19, "Line Capture" }, /* FU */ + { 17, "Front Headphone" }, /* OT */ + { 23, "Front Headphone Playback" }, /* FU */ + { 8, "Mic" }, /* IT */ + { 20, "Mic Capture" }, /* FU */ + { 9, "Front Mic" }, /* IT */ + { 21, "Front Mic Capture" }, /* FU */ + { 24, "IEC958 Playback" }, /* FU */ + {} +}; + /* * Control map entries */ @@ -500,7 +518,7 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = { }, { /* Gigabyte TRX40 Aorus Pro WiFi */ .id = USB_ID(0x0414, 0xa002), - .map = asus_rog_map, + .map = trx40_mobo_map, }, { /* ASUS ROG Zenith II */ .id = USB_ID(0x0b05, 0x1916), @@ -512,11 +530,11 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = { }, { /* MSI TRX40 Creator */ .id = USB_ID(0x0db0, 0x0d64), - .map = asus_rog_map, + .map = trx40_mobo_map, }, { /* MSI TRX40 */ .id = USB_ID(0x0db0, 0x543d), - .map = asus_rog_map, + .map = trx40_mobo_map, }, { 0 } /* terminator */ }; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index e009d584e7d0..7598d6d6740c 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3635,4 +3635,18 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } }, +#define ALC1220_VB_DESKTOP(vend, prod) { \ + USB_DEVICE(vend, prod), \ + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { \ + .vendor_name = "Realtek", \ + .product_name = "ALC1220-VB-DT", \ + .profile_name = "Realtek-ALC1220-VB-Desktop", \ + .ifnum = QUIRK_NO_INTERFACE \ + } \ +} +ALC1220_VB_DESKTOP(0x0414, 0xa002), /* Gigabyte TRX40 Aorus Pro WiFi */ +ALC1220_VB_DESKTOP(0x0db0, 0x0d64), /* MSI TRX40 Creator */ +ALC1220_VB_DESKTOP(0x0db0, 0x543d), /* MSI TRX40 */ +#undef ALC1220_VB_DESKTOP + #undef USB_DEVICE_VENDOR_SPEC -- 2.16.4

1 0

[PATCH] ALSA: usb-audio: Print more information in stream proc files
by Takashi Iwai 19 Apr '20

19 Apr '20

For more debug and usability information, add the entry showing the DSD raw states and the channel mapping in each stream proc file. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/proc.c | 53 +++++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 53 insertions(+) diff --git a/sound/usb/proc.c b/sound/usb/proc.c index 4174ad11fca6..5a36e192ebb0 100644 --- a/sound/usb/proc.c +++ b/sound/usb/proc.c @@ -54,6 +54,38 @@ void snd_usb_audio_create_proc(struct snd_usb_audio *chip) proc_audio_usbid_read); } +static const char * const channel_labels[] = { + [SNDRV_CHMAP_NA] = "N/A", + [SNDRV_CHMAP_MONO] = "MONO", + [SNDRV_CHMAP_FL] = "FL", + [SNDRV_CHMAP_FR] = "FR", + [SNDRV_CHMAP_FC] = "FC", + [SNDRV_CHMAP_LFE] = "LFE", + [SNDRV_CHMAP_RL] = "RL", + [SNDRV_CHMAP_RR] = "RR", + [SNDRV_CHMAP_FLC] = "FLC", + [SNDRV_CHMAP_FRC] = "FRC", + [SNDRV_CHMAP_RC] = "RC", + [SNDRV_CHMAP_SL] = "SL", + [SNDRV_CHMAP_SR] = "SR", + [SNDRV_CHMAP_TC] = "TC", + [SNDRV_CHMAP_TFL] = "TFL", + [SNDRV_CHMAP_TFC] = "TFC", + [SNDRV_CHMAP_TFR] = "TFR", + [SNDRV_CHMAP_TRL] = "TRL", + [SNDRV_CHMAP_TRC] = "TRC", + [SNDRV_CHMAP_TRR] = "TRR", + [SNDRV_CHMAP_TFLC] = "TFLC", + [SNDRV_CHMAP_TFRC] = "TFRC", + [SNDRV_CHMAP_LLFE] = "LLFE", + [SNDRV_CHMAP_RLFE] = "RLFE", + [SNDRV_CHMAP_TSL] = "TSL", + [SNDRV_CHMAP_TSR] = "TSR", + [SNDRV_CHMAP_BC] = "BC", + [SNDRV_CHMAP_RLC] = "RLC", + [SNDRV_CHMAP_RRC] = "RRC", +}; + /* * proc interface for list the supported pcm formats */ @@ -97,6 +129,27 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s snd_iprintf(buffer, " Data packet interval: %d us\n", 125 * (1 << fp->datainterval)); snd_iprintf(buffer, " Bits: %d\n", fp->fmt_bits); + + if (fp->dsd_raw) + snd_iprintf(buffer, " DSD raw: DOP=%d, bitrev=%d\n", + fp->dsd_dop, fp->dsd_bitrev); + + if (fp->chmap) { + const struct snd_pcm_chmap_elem *map = fp->chmap; + int c; + + snd_iprintf(buffer, " Channel map:"); + for (c = 0; c < map->channels; c++) { + if (map->map[c] >= sizeof(channel_labels) || + !channel_labels[map->map[c]]) + snd_iprintf(buffer, " --"); + else + snd_iprintf(buffer, " %s", + channel_labels[map->map[c]]); + } + snd_iprintf(buffer, "\n"); + } + // snd_iprintf(buffer, " Max Packet Size = %d\n", fp->maxpacksize); // snd_iprintf(buffer, " EP Attribute = %#x\n", fp->attributes); } -- 2.16.4

1 0

No Sound Realtek ALC294ASUS UX534FT
by LACOUR François 19 Apr '20

19 Apr '20

Hi, I can't find a issue to this problem. Have you ? Thanks François Sent with [ProtonMail](https://protonmail.com) Secure Email.

1 0

[PATCH] ALSA: hda: Remove ASUS ROG Zenith from the blacklist
by Takashi Iwai 19 Apr '20

19 Apr '20

The commit 3c6fd1f07ed0 ("ALSA: hda: Add driver blacklist") added a new blacklist for the devices that are known to have empty codecs, and one of the entries was ASUS ROG Zenith II (PCI SSID 1043:874f). However, it turned out that the very same PCI SSID is used for the previous model that does have the valid HD-audio codecs and the change broke the sound on it. This patch reverts the corresponding entry as a temporary solution. Although Zenith II and co will see get the empty HD-audio bus again, it'd be merely resource wastes and won't affect the functionality, so it's no end of the world. We'll need to address this later, e.g. by either switching to DMI string matching or using PCI ID & SSID pairs. Fixes: 3c6fd1f07ed0 ("ALSA: hda: Add driver blacklist") Reported-by: Johnathan Smithinovic <johnathan.smithinovic(a)gmx.at> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index a5fab12defde..d7adae316c0d 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2092,7 +2092,6 @@ static void pcm_mmap_prepare(struct snd_pcm_substream *substream, * should be ignored from the beginning. */ static const struct snd_pci_quirk driver_blacklist[] = { - SND_PCI_QUIRK(0x1043, 0x874f, "ASUS ROG Zenith II / Strix", 0), SND_PCI_QUIRK(0x1462, 0xcb59, "MSI TRX40 Creator", 0), SND_PCI_QUIRK(0x1462, 0xcb60, "MSI TRX40", 0), {} -- 2.16.4

1 0

[PATCH 0/7] Add H6 I2S support
by Clément Péron 18 Apr '20

18 Apr '20

Hi, This is a sequel of Marcus Cooper serie[0], where remarks made by Maxime have been fixed. I have tested it on my Beelink GS1 board. Thanks, Clement 0: https://lore.kernel.org/patchwork/cover/1139949/ Jernej Skrabec (3): dt-bindings: ASoC: sun4i-i2s: Add H6 compatible ASoC: sun4i-i2s: Add support for H6 I2S arm64: dts: sun50i-h6: Add HDMI audio to H6 DTSI Marcus Cooper (4): ASoC: sun4i-i2s: Adjust LRCLK width ASoC: sun4i-i2s: Set sign extend sample ASoc: sun4i-i2s: Add 20 and 24 bit support ASoC: sun4i-i2s: Adjust regmap settings .../sound/allwinner,sun4i-a10-i2s.yaml | 2 + arch/arm64/boot/dts/allwinner/sun50i-h6.dtsi | 31 ++ sound/soc/sunxi/sun4i-i2s.c | 286 ++++++++++++++++-- 3 files changed, 295 insertions(+), 24 deletions(-) -- 2.20.1

1 8

[PATCH v2] ALSA: usb-audio: Filter out unsupported sample rates on Focusrite devices
by Alexander Tsoy 18 Apr '20

18 Apr '20

Many Focusrite devices supports a limited set of sample rates per altsetting. These includes audio interfaces with ADAT ports: - Scarlett 18i6, 18i8 1st gen, 18i20 1st gen; - Scarlett 18i8 2nd gen, 18i20 2nd gen; - Scarlett 18i8 3rd gen, 18i20 3rd gen; - Clarett 2Pre USB, 4Pre USB, 8Pre USB. Maximum rate is exposed in the last 4 bytes of Format Type descriptor which has a non-standard bLength = 10. Tested-by: Alexey Skobkin <skobkin-ru(a)ya.ru> Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> --- Notes: v2: updated description sound/usb/format.c | 51 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 51 insertions(+) diff --git a/sound/usb/format.c b/sound/usb/format.c index 50e1874c847c..5ffb457cc88c 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -277,6 +277,52 @@ static bool s1810c_valid_sample_rate(struct audioformat *fp, return false; } +/* + * Many Focusrite devices supports a limited set of sampling rates per + * altsetting. Maximum rate is exposed in the last 4 bytes of Format Type + * descriptor which has a non-standard bLength = 10. + */ +static bool focusrite_valid_sample_rate(struct snd_usb_audio *chip, + struct audioformat *fp, + unsigned int rate) +{ + struct usb_interface *iface; + struct usb_host_interface *alts; + unsigned char *fmt; + unsigned int max_rate; + + iface = usb_ifnum_to_if(chip->dev, fp->iface); + if (!iface) + return true; + + alts = &iface->altsetting[fp->altset_idx]; + fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, + NULL, UAC_FORMAT_TYPE); + if (!fmt) + return true; + + if (fmt[0] == 10) { /* bLength */ + max_rate = combine_quad(&fmt[6]); + + /* Validate max rate */ + if (max_rate != 48000 && + max_rate != 96000 && + max_rate != 192000 && + max_rate != 384000) { + + usb_audio_info(chip, + "%u:%d : unexpected max rate: %u\n", + fp->iface, fp->altsetting, max_rate); + + return true; + } + + return rate <= max_rate; + } + + return true; +} + /* * Helper function to walk the array of sample rate triplets reported by * the device. The problem is that we need to parse whole array first to @@ -319,6 +365,11 @@ static int parse_uac2_sample_rate_range(struct snd_usb_audio *chip, !s1810c_valid_sample_rate(fp, rate)) goto skip_rate; + /* Filter out invalid rates on Focusrite devices */ + if (USB_ID_VENDOR(chip->usb_id) == 0x1235 && + !focusrite_valid_sample_rate(chip, fp, rate)) + goto skip_rate; + if (fp->rate_table) fp->rate_table[nr_rates] = rate; if (!fp->rate_min || rate < fp->rate_min) -- 2.25.3

2 1

[PATCH] ALSA: hda/realtek - Fix unexpected init_amp override
by Takashi Iwai 18 Apr '20

18 Apr '20

The commit 1c76aa5fb48d ("ALSA: hda/realtek - Allow skipping spec->init_amp detection") changed the way to assign spec->init_amp field that specifies the way to initialize the amp. Along with the change, the commit also replaced a few fixups that set spec->init_amp in HDA_FIXUP_ACT_PROBE with HDA_FIXUP_ACT_PRE_PROBE. This was rather aligning to the other fixups, and not supposed to change the actual behavior. However, this change turned out to cause a regression on FSC S7020, which hit exactly the above. The reason was that there is still one place that overrides spec->init_amp after HDA_FIXUP_ACT_PRE_PROBE call, namely in alc_ssid_check(). This patch fixes the regression by adding the proper spec->init_amp override check, i.e. verifying whether it's still ALC_INIT_UNDEFINED. Fixes: 1c76aa5fb48d ("ALSA: hda/realtek - Allow skipping spec->init_amp detection") Cc: <stable(a)vger.kernel.org> BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207329 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index dc5557d79c43..54e1c9c0a33f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -797,9 +797,11 @@ static void alc_ssid_check(struct hda_codec *codec, const hda_nid_t *ports) { if (!alc_subsystem_id(codec, ports)) { struct alc_spec *spec = codec->spec; - codec_dbg(codec, - "realtek: Enable default setup for auto mode as fallback\n"); - spec->init_amp = ALC_INIT_DEFAULT; + if (spec->init_amp == ALC_INIT_UNDEFINED) { + codec_dbg(codec, + "realtek: Enable default setup for auto mode as fallback\n"); + spec->init_amp = ALC_INIT_DEFAULT; + } } } -- 2.25.0

1 0

[PATCH] ALSA: usb-audio: Filter out unsupported sample rates on Focusrite devices
by Alexander Tsoy 18 Apr '20

18 Apr '20

Many Focusrite devices supports a limited set of sample rates per altsetting. This includes interfaces with ADAT ports and interfaces with dedicated feedback channels: - Scarlett 18i6, 18i8 1st gen, 18i20 1st gen; - Scarlett 18i8 2nd gen, 18i20 2nd gen; - all Scarlett 3rd gen devices (feedback channels works only up to 96 kHz); - Clarett 2Pre USB, 4Pre USB, 8Pre USB. Tested-by: Alexey Skobkin <skobkin-ru(a)ya.ru> Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> --- sound/usb/format.c | 51 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 51 insertions(+) diff --git a/sound/usb/format.c b/sound/usb/format.c index 50e1874c847c..5ffb457cc88c 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -277,6 +277,52 @@ static bool s1810c_valid_sample_rate(struct audioformat *fp, return false; } +/* + * Many Focusrite devices supports a limited set of sampling rates per + * altsetting. Maximum rate is exposed in the last 4 bytes of Format Type + * descriptor which has a non-standard bLength = 10. + */ +static bool focusrite_valid_sample_rate(struct snd_usb_audio *chip, + struct audioformat *fp, + unsigned int rate) +{ + struct usb_interface *iface; + struct usb_host_interface *alts; + unsigned char *fmt; + unsigned int max_rate; + + iface = usb_ifnum_to_if(chip->dev, fp->iface); + if (!iface) + return true; + + alts = &iface->altsetting[fp->altset_idx]; + fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, + NULL, UAC_FORMAT_TYPE); + if (!fmt) + return true; + + if (fmt[0] == 10) { /* bLength */ + max_rate = combine_quad(&fmt[6]); + + /* Validate max rate */ + if (max_rate != 48000 && + max_rate != 96000 && + max_rate != 192000 && + max_rate != 384000) { + + usb_audio_info(chip, + "%u:%d : unexpected max rate: %u\n", + fp->iface, fp->altsetting, max_rate); + + return true; + } + + return rate <= max_rate; + } + + return true; +} + /* * Helper function to walk the array of sample rate triplets reported by * the device. The problem is that we need to parse whole array first to @@ -319,6 +365,11 @@ static int parse_uac2_sample_rate_range(struct snd_usb_audio *chip, !s1810c_valid_sample_rate(fp, rate)) goto skip_rate; + /* Filter out invalid rates on Focusrite devices */ + if (USB_ID_VENDOR(chip->usb_id) == 0x1235 && + !focusrite_valid_sample_rate(chip, fp, rate)) + goto skip_rate; + if (fp->rate_table) fp->rate_table[nr_rates] = rate; if (!fp->rate_min || rate < fp->rate_min) -- 2.25.3

1 1

[PATCH trivial 0/6] Fix misspellings of "Analog Devices"
by Geert Uytterhoeven 18 Apr '20

18 Apr '20

Hi all, In several files the company also known as ADI is spelled as "Analog Device". However, according to https://www.analog.com/, the company name is spelled "Analog Devices". Hence this patch series, one per subsystem, fixes these misspellings. Thanks for your comments! Geert Uytterhoeven (6): dt-bindings: Fix misspellings of "Analog Devices" dma: Fix misspelling of "Analog Devices" drm: Fix misspellings of "Analog Devices" iio: Fix misspellings of "Analog Devices" ALSA: Fix misspellings of "Analog Devices" ASoC: Fix misspellings of "Analog Devices" .../devicetree/bindings/display/bridge/adi,adv7123.txt | 4 ++-- .../devicetree/bindings/display/bridge/adi,adv7511.txt | 4 ++-- Documentation/devicetree/bindings/dma/adi,axi-dmac.txt | 2 +- Documentation/devicetree/bindings/iio/dac/ad5755.txt | 2 +- drivers/dma/Kconfig | 2 +- drivers/gpu/drm/bridge/adv7511/Kconfig | 2 +- drivers/gpu/drm/drm_fb_cma_helper.c | 2 +- drivers/gpu/drm/tegra/fb.c | 2 +- drivers/iio/adc/ad7791.c | 2 +- drivers/iio/trigger/iio-trig-hrtimer.c | 2 +- drivers/staging/iio/Documentation/overview.txt | 2 +- sound/isa/ad1816a/ad1816a.c | 2 +- sound/pci/ac97/ac97_patch.c | 2 +- sound/pci/hda/Kconfig | 4 ++-- sound/soc/codecs/ad1980.c | 2 +- sound/soc/codecs/ad73311.c | 2 +- sound/soc/codecs/wm8782.c | 2 +- 17 files changed, 20 insertions(+), 20 deletions(-) -- 2.17.1 Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

8 14

[PATCH AUTOSEL 4.4 19/19] ALSA: hda: Fix potential access overflow in beep helper
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 0ad3f0b384d58f3bd1f4fb87d0af5b8f6866f41a ] The beep control helper function blindly stores the values in two stereo channels no matter whether the actual control is mono or stereo. This is practically harmless, but it annoys the recently introduced sanity check, resulting in an error when the checker is enabled. This patch corrects the behavior to store only on the defined array member. Fixes: 0401e8548eac ("ALSA: hda - Move beep helper functions to hda_beep.c") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207139 Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200407084402.25589-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_beep.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index c397e7da0eacf..7ccfb09535e14 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -310,8 +310,12 @@ int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_beep *beep = codec->beep; + int chs = get_amp_channels(kcontrol); + if (beep && (!beep->enabled || !ctl_has_mute(kcontrol))) { - ucontrol->value.integer.value[0] = + if (chs & 1) + ucontrol->value.integer.value[0] = beep->enabled; + if (chs & 2) ucontrol->value.integer.value[1] = beep->enabled; return 0; } -- 2.20.1

1 0

[PATCH AUTOSEL 4.4 18/19] ALSA: ice1724: Fix invalid access for enumerated ctl items
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit c47914c00be346bc5b48c48de7b0da5c2d1a296c ] The access to Analog Capture Source control value implemented in prodigy_hifi.c is wrong, as caught by the recently introduced sanity check; it should be accessing value.enumerated.item[] instead of value.integer.value[]. This patch corrects the wrong access pattern. Fixes: 6b8d6e5518e2 ("[ALSA] ICE1724: Added support for Audiotrak Prodigy 7.1 HiFi & HD2, Hercules Fortissimo IV") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207139 Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200407084402.25589-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/ice1712/prodigy_hifi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c index 2697402b51957..41f6450a2539f 100644 --- a/sound/pci/ice1712/prodigy_hifi.c +++ b/sound/pci/ice1712/prodigy_hifi.c @@ -569,7 +569,7 @@ static int wm_adc_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); mutex_lock(&ice->gpio_mutex); - ucontrol->value.integer.value[0] = wm_get(ice, WM_ADC_MUX) & 0x1f; + ucontrol->value.enumerated.item[0] = wm_get(ice, WM_ADC_MUX) & 0x1f; mutex_unlock(&ice->gpio_mutex); return 0; } @@ -583,7 +583,7 @@ static int wm_adc_mux_enum_put(struct snd_kcontrol *kcontrol, mutex_lock(&ice->gpio_mutex); oval = wm_get(ice, WM_ADC_MUX); - nval = (oval & 0xe0) | ucontrol->value.integer.value[0]; + nval = (oval & 0xe0) | ucontrol->value.enumerated.item[0]; if (nval != oval) { wm_put(ice, WM_ADC_MUX, nval); change = 1; -- 2.20.1

1 0

[PATCH AUTOSEL 4.4 08/19] ASoC: Intel: atom: Take the drv->lock mutex before calling sst_send_slot_map()
by Sasha Levin 18 Apr '20

18 Apr '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 81630dc042af998b9f58cd8e2c29dab9777ea176 ] sst_send_slot_map() uses sst_fill_and_send_cmd_unlocked() because in some places it is called with the drv->lock mutex already held. So it must always be called with the mutex locked. This commit adds missing locking in the sst_set_be_modules() code-path. Fixes: 24c8d14192cc ("ASoC: Intel: mrfld: add DSP core controls") Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200402185359.3424-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/atom/sst-atom-controls.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index d55388e082e18..43c960bb4b211 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -963,7 +963,9 @@ static int sst_set_be_modules(struct snd_soc_dapm_widget *w, dev_dbg(c->dev, "Enter: widget=%s\n", w->name); if (SND_SOC_DAPM_EVENT_ON(event)) { + mutex_lock(&drv->lock); ret = sst_send_slot_map(drv); + mutex_unlock(&drv->lock); if (ret) return ret; ret = sst_send_pipe_module_params(w, k); -- 2.20.1

1 0

[PATCH AUTOSEL 4.4 06/19] ASoC: dpcm: allow start or stop during pause for backend
by Sasha Levin 18 Apr '20

18 Apr '20

From: 이경택 <gt82.lee(a)samsung.com> [ Upstream commit 21fca8bdbb64df1297e8c65a746c4c9f4a689751 ] soc_compr_trigger_fe() allows start or stop after pause_push. In dpcm_be_dai_trigger(), however, only pause_release is allowed command after pause_push. So, start or stop after pause in compress offload is always returned as error if the compress offload is used with dpcm. To fix the problem, SND_SOC_DPCM_STATE_PAUSED should be allowed for start or stop command. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Reviewed-by: Vinod Koul <vkoul(a)kernel.org> Link: https://lore.kernel.org/r/004d01d607c1$7a3d5250$6eb7f6f0$@samsung.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-pcm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 7cffa98ec313c..d4bf3dc6b0158 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1951,7 +1951,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && - (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); @@ -1981,7 +1982,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; case SNDRV_PCM_TRIGGER_STOP: - if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) -- 2.20.1

1 0

[PATCH AUTOSEL 4.9 22/23] ALSA: hda: Fix potential access overflow in beep helper
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 0ad3f0b384d58f3bd1f4fb87d0af5b8f6866f41a ] The beep control helper function blindly stores the values in two stereo channels no matter whether the actual control is mono or stereo. This is practically harmless, but it annoys the recently introduced sanity check, resulting in an error when the checker is enabled. This patch corrects the behavior to store only on the defined array member. Fixes: 0401e8548eac ("ALSA: hda - Move beep helper functions to hda_beep.c") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207139 Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200407084402.25589-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_beep.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index c397e7da0eacf..7ccfb09535e14 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -310,8 +310,12 @@ int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_beep *beep = codec->beep; + int chs = get_amp_channels(kcontrol); + if (beep && (!beep->enabled || !ctl_has_mute(kcontrol))) { - ucontrol->value.integer.value[0] = + if (chs & 1) + ucontrol->value.integer.value[0] = beep->enabled; + if (chs & 2) ucontrol->value.integer.value[1] = beep->enabled; return 0; } -- 2.20.1

1 0

[PATCH AUTOSEL 4.9 21/23] ALSA: ice1724: Fix invalid access for enumerated ctl items
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit c47914c00be346bc5b48c48de7b0da5c2d1a296c ] The access to Analog Capture Source control value implemented in prodigy_hifi.c is wrong, as caught by the recently introduced sanity check; it should be accessing value.enumerated.item[] instead of value.integer.value[]. This patch corrects the wrong access pattern. Fixes: 6b8d6e5518e2 ("[ALSA] ICE1724: Added support for Audiotrak Prodigy 7.1 HiFi & HD2, Hercules Fortissimo IV") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207139 Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200407084402.25589-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/ice1712/prodigy_hifi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c index 2697402b51957..41f6450a2539f 100644 --- a/sound/pci/ice1712/prodigy_hifi.c +++ b/sound/pci/ice1712/prodigy_hifi.c @@ -569,7 +569,7 @@ static int wm_adc_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); mutex_lock(&ice->gpio_mutex); - ucontrol->value.integer.value[0] = wm_get(ice, WM_ADC_MUX) & 0x1f; + ucontrol->value.enumerated.item[0] = wm_get(ice, WM_ADC_MUX) & 0x1f; mutex_unlock(&ice->gpio_mutex); return 0; } @@ -583,7 +583,7 @@ static int wm_adc_mux_enum_put(struct snd_kcontrol *kcontrol, mutex_lock(&ice->gpio_mutex); oval = wm_get(ice, WM_ADC_MUX); - nval = (oval & 0xe0) | ucontrol->value.integer.value[0]; + nval = (oval & 0xe0) | ucontrol->value.enumerated.item[0]; if (nval != oval) { wm_put(ice, WM_ADC_MUX, nval); change = 1; -- 2.20.1

1 0

[PATCH AUTOSEL 4.9 09/23] ASoC: Intel: atom: Take the drv->lock mutex before calling sst_send_slot_map()
by Sasha Levin 18 Apr '20

18 Apr '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 81630dc042af998b9f58cd8e2c29dab9777ea176 ] sst_send_slot_map() uses sst_fill_and_send_cmd_unlocked() because in some places it is called with the drv->lock mutex already held. So it must always be called with the mutex locked. This commit adds missing locking in the sst_set_be_modules() code-path. Fixes: 24c8d14192cc ("ASoC: Intel: mrfld: add DSP core controls") Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200402185359.3424-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/atom/sst-atom-controls.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index 0838478c4c3f9..92aa1518b7723 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -976,7 +976,9 @@ static int sst_set_be_modules(struct snd_soc_dapm_widget *w, dev_dbg(c->dev, "Enter: widget=%s\n", w->name); if (SND_SOC_DAPM_EVENT_ON(event)) { + mutex_lock(&drv->lock); ret = sst_send_slot_map(drv); + mutex_unlock(&drv->lock); if (ret) return ret; ret = sst_send_pipe_module_params(w, k); -- 2.20.1

1 0

[PATCH AUTOSEL 4.9 07/23] ASoC: dpcm: allow start or stop during pause for backend
by Sasha Levin 18 Apr '20

18 Apr '20

From: 이경택 <gt82.lee(a)samsung.com> [ Upstream commit 21fca8bdbb64df1297e8c65a746c4c9f4a689751 ] soc_compr_trigger_fe() allows start or stop after pause_push. In dpcm_be_dai_trigger(), however, only pause_release is allowed command after pause_push. So, start or stop after pause in compress offload is always returned as error if the compress offload is used with dpcm. To fix the problem, SND_SOC_DPCM_STATE_PAUSED should be allowed for start or stop command. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Reviewed-by: Vinod Koul <vkoul(a)kernel.org> Link: https://lore.kernel.org/r/004d01d607c1$7a3d5250$6eb7f6f0$@samsung.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-pcm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 9df0c8102dc03..b67d105b76e46 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2062,7 +2062,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && - (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); @@ -2092,7 +2093,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; case SNDRV_PCM_TRIGGER_STOP: - if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) -- 2.20.1

1 0

[PATCH AUTOSEL 4.14 26/28] ALSA: hda: Fix potential access overflow in beep helper
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 0ad3f0b384d58f3bd1f4fb87d0af5b8f6866f41a ] The beep control helper function blindly stores the values in two stereo channels no matter whether the actual control is mono or stereo. This is practically harmless, but it annoys the recently introduced sanity check, resulting in an error when the checker is enabled. This patch corrects the behavior to store only on the defined array member. Fixes: 0401e8548eac ("ALSA: hda - Move beep helper functions to hda_beep.c") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207139 Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200407084402.25589-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_beep.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index c397e7da0eacf..7ccfb09535e14 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -310,8 +310,12 @@ int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_beep *beep = codec->beep; + int chs = get_amp_channels(kcontrol); + if (beep && (!beep->enabled || !ctl_has_mute(kcontrol))) { - ucontrol->value.integer.value[0] = + if (chs & 1) + ucontrol->value.integer.value[0] = beep->enabled; + if (chs & 2) ucontrol->value.integer.value[1] = beep->enabled; return 0; } -- 2.20.1

1 0

[PATCH AUTOSEL 4.14 25/28] ALSA: ice1724: Fix invalid access for enumerated ctl items
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit c47914c00be346bc5b48c48de7b0da5c2d1a296c ] The access to Analog Capture Source control value implemented in prodigy_hifi.c is wrong, as caught by the recently introduced sanity check; it should be accessing value.enumerated.item[] instead of value.integer.value[]. This patch corrects the wrong access pattern. Fixes: 6b8d6e5518e2 ("[ALSA] ICE1724: Added support for Audiotrak Prodigy 7.1 HiFi & HD2, Hercules Fortissimo IV") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207139 Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200407084402.25589-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/ice1712/prodigy_hifi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c index 2697402b51957..41f6450a2539f 100644 --- a/sound/pci/ice1712/prodigy_hifi.c +++ b/sound/pci/ice1712/prodigy_hifi.c @@ -569,7 +569,7 @@ static int wm_adc_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); mutex_lock(&ice->gpio_mutex); - ucontrol->value.integer.value[0] = wm_get(ice, WM_ADC_MUX) & 0x1f; + ucontrol->value.enumerated.item[0] = wm_get(ice, WM_ADC_MUX) & 0x1f; mutex_unlock(&ice->gpio_mutex); return 0; } @@ -583,7 +583,7 @@ static int wm_adc_mux_enum_put(struct snd_kcontrol *kcontrol, mutex_lock(&ice->gpio_mutex); oval = wm_get(ice, WM_ADC_MUX); - nval = (oval & 0xe0) | ucontrol->value.integer.value[0]; + nval = (oval & 0xe0) | ucontrol->value.enumerated.item[0]; if (nval != oval) { wm_put(ice, WM_ADC_MUX, nval); change = 1; -- 2.20.1

1 0

[PATCH AUTOSEL 4.14 10/28] ASoC: Intel: atom: Take the drv->lock mutex before calling sst_send_slot_map()
by Sasha Levin 18 Apr '20

18 Apr '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 81630dc042af998b9f58cd8e2c29dab9777ea176 ] sst_send_slot_map() uses sst_fill_and_send_cmd_unlocked() because in some places it is called with the drv->lock mutex already held. So it must always be called with the mutex locked. This commit adds missing locking in the sst_set_be_modules() code-path. Fixes: 24c8d14192cc ("ASoC: Intel: mrfld: add DSP core controls") Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200402185359.3424-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/atom/sst-atom-controls.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index 0f3604b559424..b04853bdaf377 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -974,7 +974,9 @@ static int sst_set_be_modules(struct snd_soc_dapm_widget *w, dev_dbg(c->dev, "Enter: widget=%s\n", w->name); if (SND_SOC_DAPM_EVENT_ON(event)) { + mutex_lock(&drv->lock); ret = sst_send_slot_map(drv); + mutex_unlock(&drv->lock); if (ret) return ret; ret = sst_send_pipe_module_params(w, k); -- 2.20.1

1 0

[PATCH AUTOSEL 4.14 08/28] ASoC: dpcm: allow start or stop during pause for backend
by Sasha Levin 18 Apr '20

18 Apr '20

From: 이경택 <gt82.lee(a)samsung.com> [ Upstream commit 21fca8bdbb64df1297e8c65a746c4c9f4a689751 ] soc_compr_trigger_fe() allows start or stop after pause_push. In dpcm_be_dai_trigger(), however, only pause_release is allowed command after pause_push. So, start or stop after pause in compress offload is always returned as error if the compress offload is used with dpcm. To fix the problem, SND_SOC_DPCM_STATE_PAUSED should be allowed for start or stop command. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Reviewed-by: Vinod Koul <vkoul(a)kernel.org> Link: https://lore.kernel.org/r/004d01d607c1$7a3d5250$6eb7f6f0$@samsung.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-pcm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index e75822dd9930f..fd4b71729eedd 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2048,7 +2048,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && - (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); @@ -2078,7 +2079,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; case SNDRV_PCM_TRIGGER_STOP: - if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) -- 2.20.1

1 0

[PATCH AUTOSEL 4.19 36/47] ASoC: Intel: bytcr_rt5640: Add quirk for MPMAN MPWIN895CL tablet
by Sasha Levin 18 Apr '20

18 Apr '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit c8b78f24c1247b7bd0882885c672d9dec5800bc6 ] The MPMAN MPWIN895CL tablet almost fully works with out default settings. The only problem is that it has only 1 speaker so any sounds only playing on the right channel get lost. Add a quirk for this model using the default settings + MONO_SPEAKER. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200405133726.24154-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index e58240e18b301..f29014a7d6723 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -588,6 +588,17 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + /* MPMAN MPWIN895CL */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "MPMAN"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "MPWIN8900CL"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* MSI S100 tablet */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Micro-Star International Co., Ltd."), -- 2.20.1

1 0

[PATCH AUTOSEL 4.19 34/47] ALSA: hda: Fix potential access overflow in beep helper
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 0ad3f0b384d58f3bd1f4fb87d0af5b8f6866f41a ] The beep control helper function blindly stores the values in two stereo channels no matter whether the actual control is mono or stereo. This is practically harmless, but it annoys the recently introduced sanity check, resulting in an error when the checker is enabled. This patch corrects the behavior to store only on the defined array member. Fixes: 0401e8548eac ("ALSA: hda - Move beep helper functions to hda_beep.c") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207139 Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200407084402.25589-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_beep.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 066b5b59c4d7c..0224011a240fb 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -297,8 +297,12 @@ int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_beep *beep = codec->beep; + int chs = get_amp_channels(kcontrol); + if (beep && (!beep->enabled || !ctl_has_mute(kcontrol))) { - ucontrol->value.integer.value[0] = + if (chs & 1) + ucontrol->value.integer.value[0] = beep->enabled; + if (chs & 2) ucontrol->value.integer.value[1] = beep->enabled; return 0; } -- 2.20.1

1 0

[PATCH AUTOSEL 4.19 33/47] ALSA: ice1724: Fix invalid access for enumerated ctl items
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit c47914c00be346bc5b48c48de7b0da5c2d1a296c ] The access to Analog Capture Source control value implemented in prodigy_hifi.c is wrong, as caught by the recently introduced sanity check; it should be accessing value.enumerated.item[] instead of value.integer.value[]. This patch corrects the wrong access pattern. Fixes: 6b8d6e5518e2 ("[ALSA] ICE1724: Added support for Audiotrak Prodigy 7.1 HiFi & HD2, Hercules Fortissimo IV") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207139 Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200407084402.25589-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/ice1712/prodigy_hifi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c index c97b5528e4b80..317bbb725b299 100644 --- a/sound/pci/ice1712/prodigy_hifi.c +++ b/sound/pci/ice1712/prodigy_hifi.c @@ -550,7 +550,7 @@ static int wm_adc_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); mutex_lock(&ice->gpio_mutex); - ucontrol->value.integer.value[0] = wm_get(ice, WM_ADC_MUX) & 0x1f; + ucontrol->value.enumerated.item[0] = wm_get(ice, WM_ADC_MUX) & 0x1f; mutex_unlock(&ice->gpio_mutex); return 0; } @@ -564,7 +564,7 @@ static int wm_adc_mux_enum_put(struct snd_kcontrol *kcontrol, mutex_lock(&ice->gpio_mutex); oval = wm_get(ice, WM_ADC_MUX); - nval = (oval & 0xe0) | ucontrol->value.integer.value[0]; + nval = (oval & 0xe0) | ucontrol->value.enumerated.item[0]; if (nval != oval) { wm_put(ice, WM_ADC_MUX, nval); change = 1; -- 2.20.1

1 0

[PATCH AUTOSEL 4.19 30/47] ALSA: hda/realtek - Add quirk for MSI GL63
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 1d3aa4a5516d2e4933fe3cca11d3349ef63bc547 ] MSI GL63 laptop requires the similar quirk like other MSI models, ALC1220_FIXUP_CLEVO_P950. The board BIOS doesn't provide a PCI SSID for the device, hence we need to take the codec SSID (1462:1275) instead. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207157 Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200408135645.21896-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 23aab2fdac468..72d3b74a42cbb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2444,6 +2444,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1458, 0xa0b8, "Gigabyte AZ370-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), SND_PCI_QUIRK(0x1458, 0xa0cd, "Gigabyte X570 Aorus Master", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1228, "MSI-GP63", ALC1220_FIXUP_CLEVO_P950), + SND_PCI_QUIRK(0x1462, 0x1275, "MSI-GL63", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1293, "MSI-GP65", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD), -- 2.20.1

1 0

[PATCH AUTOSEL 4.19 12/47] ASoC: Intel: atom: Take the drv->lock mutex before calling sst_send_slot_map()
by Sasha Levin 18 Apr '20

18 Apr '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 81630dc042af998b9f58cd8e2c29dab9777ea176 ] sst_send_slot_map() uses sst_fill_and_send_cmd_unlocked() because in some places it is called with the drv->lock mutex already held. So it must always be called with the mutex locked. This commit adds missing locking in the sst_set_be_modules() code-path. Fixes: 24c8d14192cc ("ASoC: Intel: mrfld: add DSP core controls") Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200402185359.3424-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/atom/sst-atom-controls.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index 3672d36b4b66f..d32d48e82a2af 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -974,7 +974,9 @@ static int sst_set_be_modules(struct snd_soc_dapm_widget *w, dev_dbg(c->dev, "Enter: widget=%s\n", w->name); if (SND_SOC_DAPM_EVENT_ON(event)) { + mutex_lock(&drv->lock); ret = sst_send_slot_map(drv); + mutex_unlock(&drv->lock); if (ret) return ret; ret = sst_send_pipe_module_params(w, k); -- 2.20.1

1 0

[PATCH AUTOSEL 4.19 10/47] ASoC: dpcm: allow start or stop during pause for backend
by Sasha Levin 18 Apr '20

18 Apr '20

From: 이경택 <gt82.lee(a)samsung.com> [ Upstream commit 21fca8bdbb64df1297e8c65a746c4c9f4a689751 ] soc_compr_trigger_fe() allows start or stop after pause_push. In dpcm_be_dai_trigger(), however, only pause_release is allowed command after pause_push. So, start or stop after pause in compress offload is always returned as error if the compress offload is used with dpcm. To fix the problem, SND_SOC_DPCM_STATE_PAUSED should be allowed for start or stop command. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Reviewed-by: Vinod Koul <vkoul(a)kernel.org> Link: https://lore.kernel.org/r/004d01d607c1$7a3d5250$6eb7f6f0$@samsung.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-pcm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 356d4e754561b..a0d1ce0edaf9a 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2266,7 +2266,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && - (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); @@ -2296,7 +2297,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; case SNDRV_PCM_TRIGGER_STOP: - if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) -- 2.20.1

1 0

[PATCH AUTOSEL 5.4 68/78] ASoC: SOF: trace: fix unconditional free in trace release
by Sasha Levin 18 Apr '20

18 Apr '20

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit e6110114d18d330c05fd6de9f31283fd086a5a3a ] Check if DMA pages were successfully allocated in initialization before calling free. For many types of memory (like sgbufs) the extra free is harmless, but not all backends track allocation state, so add an explicit check. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200124213625.30186-5-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/trace.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/trace.c b/sound/soc/sof/trace.c index 4c3cff031fd66..fd6f5913782bf 100644 --- a/sound/soc/sof/trace.c +++ b/sound/soc/sof/trace.c @@ -328,7 +328,10 @@ void snd_sof_free_trace(struct snd_sof_dev *sdev) { snd_sof_release_trace(sdev); - snd_dma_free_pages(&sdev->dmatb); - snd_dma_free_pages(&sdev->dmatp); + if (sdev->dma_trace_pages) { + snd_dma_free_pages(&sdev->dmatb); + snd_dma_free_pages(&sdev->dmatp); + sdev->dma_trace_pages = 0; + } } EXPORT_SYMBOL(snd_sof_free_trace); -- 2.20.1

1 0

[PATCH AUTOSEL 5.4 59/78] ALSA: hda/realtek - Add quirk for Lenovo Carbon X1 8th gen
by Sasha Levin 18 Apr '20

18 Apr '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit ca707b3f00b4f31a6e1eb37e8ae99f15f2bb1fe5 ] The audio setup on the Lenovo Carbon X1 8th gen is the same as that on the Lenovo Carbon X1 7th gen, as such it needs the same ALC285_FIXUP_THINKPAD_HEADSET_JACK quirk. This fixes volume control of the speaker not working among other things. BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1820196 Cc: stable(a)vger.kernel.org Suggested-by: Jaroslav Kysela <perex(a)perex.cz> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Link: https://lore.kernel.org/r/20200402174311.238614-1-hdegoede@redhat.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 190c1aa6a4e62..7c327a37eba10 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7300,6 +7300,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x225d, "Thinkpad T480", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2292, "Thinkpad X1 Yoga 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), SND_PCI_QUIRK(0x17aa, 0x2293, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), + SND_PCI_QUIRK(0x17aa, 0x22be, "Thinkpad X1 Carbon 8th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x310c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), -- 2.20.1

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[PATCH AUTOSEL 5.4 58/78] ALSA: usb-audio: Add Pioneer DJ DJM-250MK2 quirk
by Sasha Levin 18 Apr '20

18 Apr '20

From: František Kučera <franta-linux(a)frantovo.cz> [ Upstream commit 73d8c94084341e2895169a0462dbc18167f01683 ] Pioneer DJ DJM-250MK2 is a mixer that acts like a USB sound card. The MIDI controller part is standard but the PCM part is "vendor specific". Output is enabled by this quirk: 8 channels, 48 000 Hz, S24_3LE. Input is not working. Signed-off-by: František Kučera <franta-linux(a)frantovo.cz> Link: https://lore.kernel.org/r/20200401095907.3387-1-konference@frantovo.cz Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/quirks-table.h | 42 ++++++++++++++++++++++++++++++++++++++++ 1 file changed, 42 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index d187aa6d50db0..dcaf9eed9a415 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3592,5 +3592,47 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } } }, +{ + /* + * Pioneer DJ DJM-250MK2 + * PCM is 8 channels out @ 48 fixed (endpoints 0x01). + * The output from computer to the mixer is usable. + * + * The input (phono or line to computer) is not working. + * It should be at endpoint 0x82 and probably also 8 channels, + * but it seems that it works only with Pioneer proprietary software. + * Even on officially supported OS, the Audacity was unable to record + * and Mixxx to recognize the control vinyls. + */ + USB_DEVICE_VENDOR_SPEC(0x2b73, 0x0017), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, // outputs + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .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 = -1 + } + } + } +}, #undef USB_DEVICE_VENDOR_SPEC -- 2.20.1

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[PATCH AUTOSEL 5.4 56/78] ASoC: Intel: bytcr_rt5640: Add quirk for MPMAN MPWIN895CL tablet
by Sasha Levin 18 Apr '20

18 Apr '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit c8b78f24c1247b7bd0882885c672d9dec5800bc6 ] The MPMAN MPWIN895CL tablet almost fully works with out default settings. The only problem is that it has only 1 speaker so any sounds only playing on the right channel get lost. Add a quirk for this model using the default settings + MONO_SPEAKER. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200405133726.24154-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 243f683bc02a7..e62e1d7815aa9 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -591,6 +591,17 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + /* MPMAN MPWIN895CL */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "MPMAN"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "MPWIN8900CL"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* MSI S100 tablet */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Micro-Star International Co., Ltd."), -- 2.20.1

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[PATCH AUTOSEL 5.4 54/78] ALSA: hda: Fix potential access overflow in beep helper
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 0ad3f0b384d58f3bd1f4fb87d0af5b8f6866f41a ] The beep control helper function blindly stores the values in two stereo channels no matter whether the actual control is mono or stereo. This is practically harmless, but it annoys the recently introduced sanity check, resulting in an error when the checker is enabled. This patch corrects the behavior to store only on the defined array member. Fixes: 0401e8548eac ("ALSA: hda - Move beep helper functions to hda_beep.c") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207139 Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200407084402.25589-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_beep.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index b7d9160ed8688..c6e1e03a5e4da 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -290,8 +290,12 @@ int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_beep *beep = codec->beep; + int chs = get_amp_channels(kcontrol); + if (beep && (!beep->enabled || !ctl_has_mute(kcontrol))) { - ucontrol->value.integer.value[0] = + if (chs & 1) + ucontrol->value.integer.value[0] = beep->enabled; + if (chs & 2) ucontrol->value.integer.value[1] = beep->enabled; return 0; } -- 2.20.1

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[PATCH AUTOSEL 5.4 53/78] ALSA: ice1724: Fix invalid access for enumerated ctl items
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit c47914c00be346bc5b48c48de7b0da5c2d1a296c ] The access to Analog Capture Source control value implemented in prodigy_hifi.c is wrong, as caught by the recently introduced sanity check; it should be accessing value.enumerated.item[] instead of value.integer.value[]. This patch corrects the wrong access pattern. Fixes: 6b8d6e5518e2 ("[ALSA] ICE1724: Added support for Audiotrak Prodigy 7.1 HiFi & HD2, Hercules Fortissimo IV") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207139 Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200407084402.25589-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/ice1712/prodigy_hifi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c index 9d71e9d5c9a06..3cf41c11a4050 100644 --- a/sound/pci/ice1712/prodigy_hifi.c +++ b/sound/pci/ice1712/prodigy_hifi.c @@ -536,7 +536,7 @@ static int wm_adc_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); mutex_lock(&ice->gpio_mutex); - ucontrol->value.integer.value[0] = wm_get(ice, WM_ADC_MUX) & 0x1f; + ucontrol->value.enumerated.item[0] = wm_get(ice, WM_ADC_MUX) & 0x1f; mutex_unlock(&ice->gpio_mutex); return 0; } @@ -550,7 +550,7 @@ static int wm_adc_mux_enum_put(struct snd_kcontrol *kcontrol, mutex_lock(&ice->gpio_mutex); oval = wm_get(ice, WM_ADC_MUX); - nval = (oval & 0xe0) | ucontrol->value.integer.value[0]; + nval = (oval & 0xe0) | ucontrol->value.enumerated.item[0]; if (nval != oval) { wm_put(ice, WM_ADC_MUX, nval); change = 1; -- 2.20.1

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[PATCH AUTOSEL 5.4 49/78] ALSA: hda/realtek - Add quirk for MSI GL63
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 1d3aa4a5516d2e4933fe3cca11d3349ef63bc547 ] MSI GL63 laptop requires the similar quirk like other MSI models, ALC1220_FIXUP_CLEVO_P950. The board BIOS doesn't provide a PCI SSID for the device, hence we need to take the codec SSID (1462:1275) instead. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207157 Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200408135645.21896-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 128db2e6bc64e..190c1aa6a4e62 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2449,6 +2449,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1458, 0xa0b8, "Gigabyte AZ370-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), SND_PCI_QUIRK(0x1458, 0xa0cd, "Gigabyte X570 Aorus Master", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1228, "MSI-GP63", ALC1220_FIXUP_CLEVO_P950), + SND_PCI_QUIRK(0x1462, 0x1275, "MSI-GL63", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1293, "MSI-GP65", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD), -- 2.20.1

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[PATCH AUTOSEL 5.4 48/78] ALSA: hda: Add driver blacklist
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 3c6fd1f07ed03a04debbb9a9d782205f1ef5e2ab ] The recent AMD platform exposes an HD-audio bus but without any actual codecs, which is internally tied with a USB-audio device, supposedly. It results in "no codecs" error of HD-audio bus driver, and it's nothing but a waste of resources. This patch introduces a static blacklist table for skipping such a known bogus PCI SSID entry. As of writing this patch, the known SSIDs are: * 1043:874f - ASUS ROG Zenith II / Strix * 1462:cb59 - MSI TRX40 Creator * 1462:cb60 - MSI TRX40 BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206543 Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200408140449.22319-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_intel.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 85beb172d8108..1db9d0579c72a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2024,6 +2024,17 @@ static void pcm_mmap_prepare(struct snd_pcm_substream *substream, #endif } +/* Blacklist for skipping the whole probe: + * some HD-audio PCI entries are exposed without any codecs, and such devices + * should be ignored from the beginning. + */ +static const struct snd_pci_quirk driver_blacklist[] = { + SND_PCI_QUIRK(0x1043, 0x874f, "ASUS ROG Zenith II / Strix", 0), + SND_PCI_QUIRK(0x1462, 0xcb59, "MSI TRX40 Creator", 0), + SND_PCI_QUIRK(0x1462, 0xcb60, "MSI TRX40", 0), + {} +}; + static const struct hda_controller_ops pci_hda_ops = { .disable_msi_reset_irq = disable_msi_reset_irq, .pcm_mmap_prepare = pcm_mmap_prepare, @@ -2059,6 +2070,11 @@ static int azx_probe(struct pci_dev *pci, bool schedule_probe; int err; + if (snd_pci_quirk_lookup(pci, driver_blacklist)) { + dev_info(&pci->dev, "Skipping the blacklisted device\n"); + return -ENODEV; + } + if (dev >= SNDRV_CARDS) return -ENODEV; if (!enable[dev]) { -- 2.20.1

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[PATCH AUTOSEL 5.4 19/78] ASoC: Intel: atom: Take the drv->lock mutex before calling sst_send_slot_map()
by Sasha Levin 18 Apr '20

18 Apr '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 81630dc042af998b9f58cd8e2c29dab9777ea176 ] sst_send_slot_map() uses sst_fill_and_send_cmd_unlocked() because in some places it is called with the drv->lock mutex already held. So it must always be called with the mutex locked. This commit adds missing locking in the sst_set_be_modules() code-path. Fixes: 24c8d14192cc ("ASoC: Intel: mrfld: add DSP core controls") Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200402185359.3424-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/atom/sst-atom-controls.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index baef461a99f19..2c3798034b1de 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -966,7 +966,9 @@ static int sst_set_be_modules(struct snd_soc_dapm_widget *w, dev_dbg(c->dev, "Enter: widget=%s\n", w->name); if (SND_SOC_DAPM_EVENT_ON(event)) { + mutex_lock(&drv->lock); ret = sst_send_slot_map(drv); + mutex_unlock(&drv->lock); if (ret) return ret; ret = sst_send_pipe_module_params(w, k); -- 2.20.1

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[PATCH AUTOSEL 5.4 16/78] ASoC: dpcm: allow start or stop during pause for backend
by Sasha Levin 18 Apr '20

18 Apr '20

From: 이경택 <gt82.lee(a)samsung.com> [ Upstream commit 21fca8bdbb64df1297e8c65a746c4c9f4a689751 ] soc_compr_trigger_fe() allows start or stop after pause_push. In dpcm_be_dai_trigger(), however, only pause_release is allowed command after pause_push. So, start or stop after pause in compress offload is always returned as error if the compress offload is used with dpcm. To fix the problem, SND_SOC_DPCM_STATE_PAUSED should be allowed for start or stop command. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Reviewed-by: Vinod Koul <vkoul(a)kernel.org> Link: https://lore.kernel.org/r/004d01d607c1$7a3d5250$6eb7f6f0$@samsung.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-pcm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index d978df95c5c65..cc4e9aa80fb0d 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2222,7 +2222,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && - (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); @@ -2252,7 +2253,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; case SNDRV_PCM_TRIGGER_STOP: - if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) -- 2.20.1

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[PATCH AUTOSEL 5.5 70/75] ASoC: SOF: trace: fix unconditional free in trace release
by Sasha Levin 18 Apr '20

18 Apr '20

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit e6110114d18d330c05fd6de9f31283fd086a5a3a ] Check if DMA pages were successfully allocated in initialization before calling free. For many types of memory (like sgbufs) the extra free is harmless, but not all backends track allocation state, so add an explicit check. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200124213625.30186-5-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/trace.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/trace.c b/sound/soc/sof/trace.c index b0e4556c8536a..5bf63fa67fb09 100644 --- a/sound/soc/sof/trace.c +++ b/sound/soc/sof/trace.c @@ -343,7 +343,10 @@ void snd_sof_free_trace(struct snd_sof_dev *sdev) snd_sof_release_trace(sdev); - snd_dma_free_pages(&sdev->dmatb); - snd_dma_free_pages(&sdev->dmatp); + if (sdev->dma_trace_pages) { + snd_dma_free_pages(&sdev->dmatb); + snd_dma_free_pages(&sdev->dmatp); + sdev->dma_trace_pages = 0; + } } EXPORT_SYMBOL(snd_sof_free_trace); -- 2.20.1

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[PATCH AUTOSEL 5.5 64/75] ALSA: hda/realtek - Add quirk for Lenovo Carbon X1 8th gen
by Sasha Levin 18 Apr '20

18 Apr '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit ca707b3f00b4f31a6e1eb37e8ae99f15f2bb1fe5 ] The audio setup on the Lenovo Carbon X1 8th gen is the same as that on the Lenovo Carbon X1 7th gen, as such it needs the same ALC285_FIXUP_THINKPAD_HEADSET_JACK quirk. This fixes volume control of the speaker not working among other things. BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1820196 Cc: stable(a)vger.kernel.org Suggested-by: Jaroslav Kysela <perex(a)perex.cz> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Link: https://lore.kernel.org/r/20200402174311.238614-1-hdegoede@redhat.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 62eac992ebd55..e2c77d2ab26e9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7300,6 +7300,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x225d, "Thinkpad T480", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2292, "Thinkpad X1 Yoga 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), SND_PCI_QUIRK(0x17aa, 0x2293, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), + SND_PCI_QUIRK(0x17aa, 0x22be, "Thinkpad X1 Carbon 8th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x310c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), -- 2.20.1

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[PATCH AUTOSEL 5.5 63/75] ALSA: usb-audio: Add Pioneer DJ DJM-250MK2 quirk
by Sasha Levin 18 Apr '20

18 Apr '20

From: František Kučera <franta-linux(a)frantovo.cz> [ Upstream commit 73d8c94084341e2895169a0462dbc18167f01683 ] Pioneer DJ DJM-250MK2 is a mixer that acts like a USB sound card. The MIDI controller part is standard but the PCM part is "vendor specific". Output is enabled by this quirk: 8 channels, 48 000 Hz, S24_3LE. Input is not working. Signed-off-by: František Kučera <franta-linux(a)frantovo.cz> Link: https://lore.kernel.org/r/20200401095907.3387-1-konference@frantovo.cz Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/quirks-table.h | 42 ++++++++++++++++++++++++++++++++++++++++ 1 file changed, 42 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index d187aa6d50db0..dcaf9eed9a415 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3592,5 +3592,47 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } } }, +{ + /* + * Pioneer DJ DJM-250MK2 + * PCM is 8 channels out @ 48 fixed (endpoints 0x01). + * The output from computer to the mixer is usable. + * + * The input (phono or line to computer) is not working. + * It should be at endpoint 0x82 and probably also 8 channels, + * but it seems that it works only with Pioneer proprietary software. + * Even on officially supported OS, the Audacity was unable to record + * and Mixxx to recognize the control vinyls. + */ + USB_DEVICE_VENDOR_SPEC(0x2b73, 0x0017), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, // outputs + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .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 = -1 + } + } + } +}, #undef USB_DEVICE_VENDOR_SPEC -- 2.20.1

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[PATCH AUTOSEL 5.5 61/75] ASoC: Intel: bytcr_rt5640: Add quirk for MPMAN MPWIN895CL tablet
by Sasha Levin 18 Apr '20

18 Apr '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit c8b78f24c1247b7bd0882885c672d9dec5800bc6 ] The MPMAN MPWIN895CL tablet almost fully works with out default settings. The only problem is that it has only 1 speaker so any sounds only playing on the right channel get lost. Add a quirk for this model using the default settings + MONO_SPEAKER. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200405133726.24154-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 243f683bc02a7..e62e1d7815aa9 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -591,6 +591,17 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + /* MPMAN MPWIN895CL */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "MPMAN"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "MPWIN8900CL"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* MSI S100 tablet */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Micro-Star International Co., Ltd."), -- 2.20.1

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[PATCH AUTOSEL 5.5 59/75] ALSA: hda: Fix potential access overflow in beep helper
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 0ad3f0b384d58f3bd1f4fb87d0af5b8f6866f41a ] The beep control helper function blindly stores the values in two stereo channels no matter whether the actual control is mono or stereo. This is practically harmless, but it annoys the recently introduced sanity check, resulting in an error when the checker is enabled. This patch corrects the behavior to store only on the defined array member. Fixes: 0401e8548eac ("ALSA: hda - Move beep helper functions to hda_beep.c") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207139 Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200407084402.25589-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_beep.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index b7d9160ed8688..c6e1e03a5e4da 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -290,8 +290,12 @@ int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_beep *beep = codec->beep; + int chs = get_amp_channels(kcontrol); + if (beep && (!beep->enabled || !ctl_has_mute(kcontrol))) { - ucontrol->value.integer.value[0] = + if (chs & 1) + ucontrol->value.integer.value[0] = beep->enabled; + if (chs & 2) ucontrol->value.integer.value[1] = beep->enabled; return 0; } -- 2.20.1

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[PATCH AUTOSEL 5.5 58/75] ALSA: ice1724: Fix invalid access for enumerated ctl items
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit c47914c00be346bc5b48c48de7b0da5c2d1a296c ] The access to Analog Capture Source control value implemented in prodigy_hifi.c is wrong, as caught by the recently introduced sanity check; it should be accessing value.enumerated.item[] instead of value.integer.value[]. This patch corrects the wrong access pattern. Fixes: 6b8d6e5518e2 ("[ALSA] ICE1724: Added support for Audiotrak Prodigy 7.1 HiFi & HD2, Hercules Fortissimo IV") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207139 Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200407084402.25589-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/ice1712/prodigy_hifi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c index 9d71e9d5c9a06..3cf41c11a4050 100644 --- a/sound/pci/ice1712/prodigy_hifi.c +++ b/sound/pci/ice1712/prodigy_hifi.c @@ -536,7 +536,7 @@ static int wm_adc_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); mutex_lock(&ice->gpio_mutex); - ucontrol->value.integer.value[0] = wm_get(ice, WM_ADC_MUX) & 0x1f; + ucontrol->value.enumerated.item[0] = wm_get(ice, WM_ADC_MUX) & 0x1f; mutex_unlock(&ice->gpio_mutex); return 0; } @@ -550,7 +550,7 @@ static int wm_adc_mux_enum_put(struct snd_kcontrol *kcontrol, mutex_lock(&ice->gpio_mutex); oval = wm_get(ice, WM_ADC_MUX); - nval = (oval & 0xe0) | ucontrol->value.integer.value[0]; + nval = (oval & 0xe0) | ucontrol->value.enumerated.item[0]; if (nval != oval) { wm_put(ice, WM_ADC_MUX, nval); change = 1; -- 2.20.1

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[PATCH AUTOSEL 5.5 54/75] ALSA: hda/realtek - Add quirk for MSI GL63
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 1d3aa4a5516d2e4933fe3cca11d3349ef63bc547 ] MSI GL63 laptop requires the similar quirk like other MSI models, ALC1220_FIXUP_CLEVO_P950. The board BIOS doesn't provide a PCI SSID for the device, hence we need to take the codec SSID (1462:1275) instead. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207157 Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200408135645.21896-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f44d8e2589337..62eac992ebd55 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2449,6 +2449,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1458, 0xa0b8, "Gigabyte AZ370-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), SND_PCI_QUIRK(0x1458, 0xa0cd, "Gigabyte X570 Aorus Master", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1228, "MSI-GP63", ALC1220_FIXUP_CLEVO_P950), + SND_PCI_QUIRK(0x1462, 0x1275, "MSI-GL63", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1293, "MSI-GP65", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD), -- 2.20.1

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[PATCH AUTOSEL 5.5 53/75] ALSA: hda: Add driver blacklist
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 3c6fd1f07ed03a04debbb9a9d782205f1ef5e2ab ] The recent AMD platform exposes an HD-audio bus but without any actual codecs, which is internally tied with a USB-audio device, supposedly. It results in "no codecs" error of HD-audio bus driver, and it's nothing but a waste of resources. This patch introduces a static blacklist table for skipping such a known bogus PCI SSID entry. As of writing this patch, the known SSIDs are: * 1043:874f - ASUS ROG Zenith II / Strix * 1462:cb59 - MSI TRX40 Creator * 1462:cb60 - MSI TRX40 BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206543 Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200408140449.22319-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_intel.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 9d3b28bcba9b4..d2d5a19bdecce 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2074,6 +2074,17 @@ static void pcm_mmap_prepare(struct snd_pcm_substream *substream, #endif } +/* Blacklist for skipping the whole probe: + * some HD-audio PCI entries are exposed without any codecs, and such devices + * should be ignored from the beginning. + */ +static const struct snd_pci_quirk driver_blacklist[] = { + SND_PCI_QUIRK(0x1043, 0x874f, "ASUS ROG Zenith II / Strix", 0), + SND_PCI_QUIRK(0x1462, 0xcb59, "MSI TRX40 Creator", 0), + SND_PCI_QUIRK(0x1462, 0xcb60, "MSI TRX40", 0), + {} +}; + static const struct hda_controller_ops pci_hda_ops = { .disable_msi_reset_irq = disable_msi_reset_irq, .pcm_mmap_prepare = pcm_mmap_prepare, @@ -2090,6 +2101,11 @@ static int azx_probe(struct pci_dev *pci, bool schedule_probe; int err; + if (snd_pci_quirk_lookup(pci, driver_blacklist)) { + dev_info(&pci->dev, "Skipping the blacklisted device\n"); + return -ENODEV; + } + if (dev >= SNDRV_CARDS) return -ENODEV; if (!enable[dev]) { -- 2.20.1

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[PATCH AUTOSEL 5.5 20/75] ASoC: Intel: atom: Take the drv->lock mutex before calling sst_send_slot_map()
by Sasha Levin 18 Apr '20

18 Apr '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 81630dc042af998b9f58cd8e2c29dab9777ea176 ] sst_send_slot_map() uses sst_fill_and_send_cmd_unlocked() because in some places it is called with the drv->lock mutex already held. So it must always be called with the mutex locked. This commit adds missing locking in the sst_set_be_modules() code-path. Fixes: 24c8d14192cc ("ASoC: Intel: mrfld: add DSP core controls") Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200402185359.3424-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/atom/sst-atom-controls.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index baef461a99f19..2c3798034b1de 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -966,7 +966,9 @@ static int sst_set_be_modules(struct snd_soc_dapm_widget *w, dev_dbg(c->dev, "Enter: widget=%s\n", w->name); if (SND_SOC_DAPM_EVENT_ON(event)) { + mutex_lock(&drv->lock); ret = sst_send_slot_map(drv); + mutex_unlock(&drv->lock); if (ret) return ret; ret = sst_send_pipe_module_params(w, k); -- 2.20.1

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[PATCH AUTOSEL 5.5 17/75] ASoC: dpcm: allow start or stop during pause for backend
by Sasha Levin 18 Apr '20

18 Apr '20

From: 이경택 <gt82.lee(a)samsung.com> [ Upstream commit 21fca8bdbb64df1297e8c65a746c4c9f4a689751 ] soc_compr_trigger_fe() allows start or stop after pause_push. In dpcm_be_dai_trigger(), however, only pause_release is allowed command after pause_push. So, start or stop after pause in compress offload is always returned as error if the compress offload is used with dpcm. To fix the problem, SND_SOC_DPCM_STATE_PAUSED should be allowed for start or stop command. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Reviewed-by: Vinod Koul <vkoul(a)kernel.org> Link: https://lore.kernel.org/r/004d01d607c1$7a3d5250$6eb7f6f0$@samsung.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-pcm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 8de29f48442f6..e4ff0796526c7 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2256,7 +2256,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && - (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); @@ -2286,7 +2287,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; case SNDRV_PCM_TRIGGER_STOP: - if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) -- 2.20.1

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[PATCH AUTOSEL 5.6 68/73] ALSA: hda/realtek - Add quirk for Lenovo Carbon X1 8th gen
by Sasha Levin 18 Apr '20

18 Apr '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit ca707b3f00b4f31a6e1eb37e8ae99f15f2bb1fe5 ] The audio setup on the Lenovo Carbon X1 8th gen is the same as that on the Lenovo Carbon X1 7th gen, as such it needs the same ALC285_FIXUP_THINKPAD_HEADSET_JACK quirk. This fixes volume control of the speaker not working among other things. BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1820196 Cc: stable(a)vger.kernel.org Suggested-by: Jaroslav Kysela <perex(a)perex.cz> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Link: https://lore.kernel.org/r/20200402174311.238614-1-hdegoede@redhat.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5cced684cf41f..3f3484455080d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7300,6 +7300,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x225d, "Thinkpad T480", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2292, "Thinkpad X1 Yoga 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), SND_PCI_QUIRK(0x17aa, 0x2293, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), + SND_PCI_QUIRK(0x17aa, 0x22be, "Thinkpad X1 Carbon 8th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x310c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), -- 2.20.1

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[PATCH AUTOSEL 5.6 67/73] ALSA: usb-audio: Add Pioneer DJ DJM-250MK2 quirk
by Sasha Levin 18 Apr '20

18 Apr '20

From: František Kučera <franta-linux(a)frantovo.cz> [ Upstream commit 73d8c94084341e2895169a0462dbc18167f01683 ] Pioneer DJ DJM-250MK2 is a mixer that acts like a USB sound card. The MIDI controller part is standard but the PCM part is "vendor specific". Output is enabled by this quirk: 8 channels, 48 000 Hz, S24_3LE. Input is not working. Signed-off-by: František Kučera <franta-linux(a)frantovo.cz> Link: https://lore.kernel.org/r/20200401095907.3387-1-konference@frantovo.cz Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/quirks-table.h | 42 ++++++++++++++++++++++++++++++++++++++++ 1 file changed, 42 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index d187aa6d50db0..dcaf9eed9a415 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3592,5 +3592,47 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } } }, +{ + /* + * Pioneer DJ DJM-250MK2 + * PCM is 8 channels out @ 48 fixed (endpoints 0x01). + * The output from computer to the mixer is usable. + * + * The input (phono or line to computer) is not working. + * It should be at endpoint 0x82 and probably also 8 channels, + * but it seems that it works only with Pioneer proprietary software. + * Even on officially supported OS, the Audacity was unable to record + * and Mixxx to recognize the control vinyls. + */ + USB_DEVICE_VENDOR_SPEC(0x2b73, 0x0017), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, // outputs + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .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 = -1 + } + } + } +}, #undef USB_DEVICE_VENDOR_SPEC -- 2.20.1

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[PATCH AUTOSEL 5.6 65/73] ASoC: Intel: bytcr_rt5640: Add quirk for MPMAN MPWIN895CL tablet
by Sasha Levin 18 Apr '20

18 Apr '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit c8b78f24c1247b7bd0882885c672d9dec5800bc6 ] The MPMAN MPWIN895CL tablet almost fully works with out default settings. The only problem is that it has only 1 speaker so any sounds only playing on the right channel get lost. Add a quirk for this model using the default settings + MONO_SPEAKER. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200405133726.24154-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 6bd9ae813be28..d14d5f7db1689 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -591,6 +591,17 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + /* MPMAN MPWIN895CL */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "MPMAN"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "MPWIN8900CL"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* MSI S100 tablet */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Micro-Star International Co., Ltd."), -- 2.20.1

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[PATCH AUTOSEL 5.6 63/73] ALSA: hda: Fix potential access overflow in beep helper
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 0ad3f0b384d58f3bd1f4fb87d0af5b8f6866f41a ] The beep control helper function blindly stores the values in two stereo channels no matter whether the actual control is mono or stereo. This is practically harmless, but it annoys the recently introduced sanity check, resulting in an error when the checker is enabled. This patch corrects the behavior to store only on the defined array member. Fixes: 0401e8548eac ("ALSA: hda - Move beep helper functions to hda_beep.c") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207139 Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200407084402.25589-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_beep.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index f5fd62ed4df5b..841523f6b88de 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -290,8 +290,12 @@ int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_beep *beep = codec->beep; + int chs = get_amp_channels(kcontrol); + if (beep && (!beep->enabled || !ctl_has_mute(kcontrol))) { - ucontrol->value.integer.value[0] = + if (chs & 1) + ucontrol->value.integer.value[0] = beep->enabled; + if (chs & 2) ucontrol->value.integer.value[1] = beep->enabled; return 0; } -- 2.20.1

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[PATCH AUTOSEL 5.6 62/73] ALSA: ice1724: Fix invalid access for enumerated ctl items
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit c47914c00be346bc5b48c48de7b0da5c2d1a296c ] The access to Analog Capture Source control value implemented in prodigy_hifi.c is wrong, as caught by the recently introduced sanity check; it should be accessing value.enumerated.item[] instead of value.integer.value[]. This patch corrects the wrong access pattern. Fixes: 6b8d6e5518e2 ("[ALSA] ICE1724: Added support for Audiotrak Prodigy 7.1 HiFi & HD2, Hercules Fortissimo IV") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207139 Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200407084402.25589-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/ice1712/prodigy_hifi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c index 91f83cef0e56c..9aa12a67d3705 100644 --- a/sound/pci/ice1712/prodigy_hifi.c +++ b/sound/pci/ice1712/prodigy_hifi.c @@ -536,7 +536,7 @@ static int wm_adc_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); mutex_lock(&ice->gpio_mutex); - ucontrol->value.integer.value[0] = wm_get(ice, WM_ADC_MUX) & 0x1f; + ucontrol->value.enumerated.item[0] = wm_get(ice, WM_ADC_MUX) & 0x1f; mutex_unlock(&ice->gpio_mutex); return 0; } @@ -550,7 +550,7 @@ static int wm_adc_mux_enum_put(struct snd_kcontrol *kcontrol, mutex_lock(&ice->gpio_mutex); oval = wm_get(ice, WM_ADC_MUX); - nval = (oval & 0xe0) | ucontrol->value.integer.value[0]; + nval = (oval & 0xe0) | ucontrol->value.enumerated.item[0]; if (nval != oval) { wm_put(ice, WM_ADC_MUX, nval); change = 1; -- 2.20.1

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[PATCH AUTOSEL 5.6 58/73] ALSA: hda/realtek - Add quirk for MSI GL63
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 1d3aa4a5516d2e4933fe3cca11d3349ef63bc547 ] MSI GL63 laptop requires the similar quirk like other MSI models, ALC1220_FIXUP_CLEVO_P950. The board BIOS doesn't provide a PCI SSID for the device, hence we need to take the codec SSID (1462:1275) instead. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207157 Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200408135645.21896-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 63e1a56f705b7..5cced684cf41f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2449,6 +2449,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1458, 0xa0b8, "Gigabyte AZ370-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), SND_PCI_QUIRK(0x1458, 0xa0cd, "Gigabyte X570 Aorus Master", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1228, "MSI-GP63", ALC1220_FIXUP_CLEVO_P950), + SND_PCI_QUIRK(0x1462, 0x1275, "MSI-GL63", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1293, "MSI-GP65", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD), -- 2.20.1

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[PATCH AUTOSEL 5.6 57/73] ALSA: hda: Add driver blacklist
by Sasha Levin 18 Apr '20

18 Apr '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 3c6fd1f07ed03a04debbb9a9d782205f1ef5e2ab ] The recent AMD platform exposes an HD-audio bus but without any actual codecs, which is internally tied with a USB-audio device, supposedly. It results in "no codecs" error of HD-audio bus driver, and it's nothing but a waste of resources. This patch introduces a static blacklist table for skipping such a known bogus PCI SSID entry. As of writing this patch, the known SSIDs are: * 1043:874f - ASUS ROG Zenith II / Strix * 1462:cb59 - MSI TRX40 Creator * 1462:cb60 - MSI TRX40 BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206543 Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200408140449.22319-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_intel.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 92a042e34d3e5..bd093593f8fbd 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2076,6 +2076,17 @@ static void pcm_mmap_prepare(struct snd_pcm_substream *substream, #endif } +/* Blacklist for skipping the whole probe: + * some HD-audio PCI entries are exposed without any codecs, and such devices + * should be ignored from the beginning. + */ +static const struct snd_pci_quirk driver_blacklist[] = { + SND_PCI_QUIRK(0x1043, 0x874f, "ASUS ROG Zenith II / Strix", 0), + SND_PCI_QUIRK(0x1462, 0xcb59, "MSI TRX40 Creator", 0), + SND_PCI_QUIRK(0x1462, 0xcb60, "MSI TRX40", 0), + {} +}; + static const struct hda_controller_ops pci_hda_ops = { .disable_msi_reset_irq = disable_msi_reset_irq, .pcm_mmap_prepare = pcm_mmap_prepare, @@ -2092,6 +2103,11 @@ static int azx_probe(struct pci_dev *pci, bool schedule_probe; int err; + if (snd_pci_quirk_lookup(pci, driver_blacklist)) { + dev_info(&pci->dev, "Skipping the blacklisted device\n"); + return -ENODEV; + } + if (dev >= SNDRV_CARDS) return -ENODEV; if (!enable[dev]) { -- 2.20.1

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[PATCH AUTOSEL 5.6 22/73] ASoC: Intel: atom: Take the drv->lock mutex before calling sst_send_slot_map()
by Sasha Levin 18 Apr '20

18 Apr '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 81630dc042af998b9f58cd8e2c29dab9777ea176 ] sst_send_slot_map() uses sst_fill_and_send_cmd_unlocked() because in some places it is called with the drv->lock mutex already held. So it must always be called with the mutex locked. This commit adds missing locking in the sst_set_be_modules() code-path. Fixes: 24c8d14192cc ("ASoC: Intel: mrfld: add DSP core controls") Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200402185359.3424-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/atom/sst-atom-controls.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index baef461a99f19..2c3798034b1de 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -966,7 +966,9 @@ static int sst_set_be_modules(struct snd_soc_dapm_widget *w, dev_dbg(c->dev, "Enter: widget=%s\n", w->name); if (SND_SOC_DAPM_EVENT_ON(event)) { + mutex_lock(&drv->lock); ret = sst_send_slot_map(drv); + mutex_unlock(&drv->lock); if (ret) return ret; ret = sst_send_pipe_module_params(w, k); -- 2.20.1

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[PATCH AUTOSEL 5.6 19/73] ASoC: dpcm: allow start or stop during pause for backend
by Sasha Levin 18 Apr '20

18 Apr '20

From: 이경택 <gt82.lee(a)samsung.com> [ Upstream commit 21fca8bdbb64df1297e8c65a746c4c9f4a689751 ] soc_compr_trigger_fe() allows start or stop after pause_push. In dpcm_be_dai_trigger(), however, only pause_release is allowed command after pause_push. So, start or stop after pause in compress offload is always returned as error if the compress offload is used with dpcm. To fix the problem, SND_SOC_DPCM_STATE_PAUSED should be allowed for start or stop command. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Reviewed-by: Vinod Koul <vkoul(a)kernel.org> Link: https://lore.kernel.org/r/004d01d607c1$7a3d5250$6eb7f6f0$@samsung.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-pcm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 2c59b3688ca01..8f6f0ad502880 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2236,7 +2236,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && - (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); @@ -2266,7 +2267,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; case SNDRV_PCM_TRIGGER_STOP: - if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) -- 2.20.1

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[PATCH AUTOSEL 5.6 18/73] ASoC: qcom: q6asm-dai: Add SNDRV_PCM_INFO_BATCH flag
by Sasha Levin 18 Apr '20

18 Apr '20

From: Stephan Gerhold <stephan(a)gerhold.net> [ Upstream commit 7f2430cda819a9ecb1df5a0f3ef4f1c20db3f811 ] At the moment, playing audio with PulseAudio with the qdsp6 driver results in distorted sound. It seems like its timer-based scheduling does not work properly with qdsp6 since setting tsched=0 in the PulseAudio configuration avoids the issue. Apparently this happens when the pointer() callback is not accurate enough. There is a SNDRV_PCM_INFO_BATCH flag that can be used to stop PulseAudio from using timer-based scheduling by default. According to https://www.alsa-project.org/pipermail/alsa-devel/2014-March/073816.html: The flag is being used in the sense explained in the previous audio meeting -- the data transfer granularity isn't fine enough but aligned to the period size (or less). q6asm-dai reports the position as multiple of prtd->pcm_count = snd_pcm_lib_period_bytes(substream) so it indeed just a multiple of the period size. Therefore adding the flag here seems appropriate and makes audio work out of the box. Fixes: 2a9e92d371db ("ASoC: qdsp6: q6asm: Add q6asm dai driver") Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Link: https://lore.kernel.org/r/20200330175210.47518-1-stephan@gerhold.net Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/qcom/qdsp6/q6asm-dai.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index c0d422d0ab94f..d7dc80ede8927 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -73,7 +73,7 @@ struct q6asm_dai_data { }; static const struct snd_pcm_hardware q6asm_dai_hardware_capture = { - .info = (SNDRV_PCM_INFO_MMAP | + .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_BATCH | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | @@ -95,7 +95,7 @@ static const struct snd_pcm_hardware q6asm_dai_hardware_capture = { }; static struct snd_pcm_hardware q6asm_dai_hardware_playback = { - .info = (SNDRV_PCM_INFO_MMAP | + .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_BATCH | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | -- 2.20.1

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State of hda-analyser.py
by Moosa Mahsoom 18 Apr '20

18 Apr '20

Hi, I am a final year student from BITS Pilani, India. I have been facing some issues with my laptop audio and hence wanted to use the hda_analyser script. However, I noticed that the python script doesn't seem to compatible with the current pygtk libraries. If possible,I would like to contribute towards bringing the scripts to parity. Yours Sincerely, Moosa Mahsoom

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Bug WD19 dockstation
by Gregorio Rodrigo 17 Apr '20

17 Apr '20

Hi everyone, I am not sure if this is the proper way to submit a bug/ticket/question. I posted a message here https://github.com/edrose/dell-dock-audio-fix/issues/2#issuecomment-6152907… and got the suggestion to write to the alsa mail list. I have a WD19 docking station and the lineout is not recognized. It seems it is recognized as an input. However it works if I run the speaker-test command. Anyone knows if I can apply those UCM settings or how could I do it? I copy here the message. I have a WD19 docking station and i am facing the same problem. Debian GNU/Linux bullseye/sid \n \l Linux xps 5.5.0-1-amd64 #1 <https://github.com/edrose/dell-dock-audio-fix/issues/1> SMP Debian 5.5.13-2 (2020-03-30) x86_64 GNU/Linux pactl list cards Card #1 <https://github.com/edrose/dell-dock-audio-fix/issues/1> Name: alsa_card.usb-Generic_USB_Audio_200901010001-00 Driver: module-alsa-card.c Owner Module: 25 Properties: alsa.card = "2" alsa.card_name = "WD19 Dock" alsa.long_card_name = "Dell-WD15-Dock" alsa.driver_name = "snd_usb_audio" device.bus_path = "pci-0000:00:14.0-usb-0:9.3.4:1.0" sysfs.path = "/devices/pci0000:00/0000:00:14.0/usb3/3-9/3-9.3/3-9.3.4/3-9.3.4:1.0/sound/card2" udev.id = "usb-Generic_USB_Audio_200901010001-00" device.bus = "usb" device.vendor.id = "0bda" device.vendor.name = "Realtek Semiconductor Corp." device.product.id = "402e" device.product.name = "USB Audio" device.serial = "Generic_USB_Audio_200901010001" device.string = "2" device.description = "USB Audio" module-udev-detect.discovered = "1" device.icon_name = "audio-card-usb" Profiles: input:analog-stereo: Analog Stereo Input (sinks: 0, sources: 1, priority: 65, available: yes) output:analog-stereo: Analog Stereo Output (sinks: 1, sources: 0, priority: 6500, available: yes) output:analog-stereo+input:analog-stereo: Analog Stereo Duplex (sinks: 1, sources: 1, priority: 6565, available: yes) off: Off (sinks: 0, sources: 0, priority: 0, available: yes) Active Profile: output:analog-stereo+input:analog-stereo Ports: analog-input-mic: Microphone (priority: 8700, latency offset: 0 usec) Properties: device.icon_name = "audio-input-microphone" Part of profile(s): input:analog-stereo, output:analog-stereo+input:analog-stereo analog-output-headphones: Headphones (priority: 9900, latency offset: 0 usec) Properties: device.icon_name = "audio-headphones" Part of profile(s): output:analog-stereo, output:analog-stereo+input:analog-stereo http://alsa-project.org/db/?f=616054544b8748431c354f1affbfd705c88fad37 If i run manualy speaker-test and select the proper subdevice, i get sound over the lineoutput. I cannot set it as default (or i dont know how to do it). It seems that if i am able to set the UCM2 configuration from WD15 should work. Any advise? Thank you in advance.

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[PATCH] ASoC: stm32: sai: fix sai probe
by Olivier Moysan 17 Apr '20

17 Apr '20

pcm config must be set before snd_dmaengine_pcm_register() call. Fixes: 0d6defc7e0e4 ("ASoC: stm32: sai: manage rebind issue") Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- sound/soc/stm/stm32_sai_sub.c | 12 +++++------- 1 file changed, 5 insertions(+), 7 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 34a7c3d6fb91..41f01c3e639e 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -1547,6 +1547,9 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) return ret; } + if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) + conf = &stm32_sai_pcm_config_spdif; + ret = snd_dmaengine_pcm_register(&pdev->dev, conf, 0); if (ret) { if (ret != -EPROBE_DEFER) @@ -1556,15 +1559,10 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) ret = snd_soc_register_component(&pdev->dev, &stm32_component, &sai->cpu_dai_drv, 1); - if (ret) { + if (ret) snd_dmaengine_pcm_unregister(&pdev->dev); - return ret; - } - - if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) - conf = &stm32_sai_pcm_config_spdif; - return 0; + return ret; } static int stm32_sai_sub_remove(struct platform_device *pdev) -- 2.17.1

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[PATCH] ASoC: SOF: Intel: add min/max channels for SSP on Baytrail/Broadwell
by Pierre-Louis Bossart 17 Apr '20

17 Apr '20

Major regressions were detected by SOF CI on CherryTrail and Broadwell: [ 25.705750] SSP2-Codec: ASoC: no backend playback stream [ 27.923378] SSP2-Codec: ASoC: no users playback at close - state This is root-caused to the introduction of the DAI capability checks with snd_soc_dai_stream_valid(). Its use in soc-pcm.c makes it a requirement for all DAIs to report at least a non-zero min_channels field. For some reason the SSP structures used for SKL+ did provide this information but legacy platforms didn't. Fixes: 9b5db059366ae2 ("ASoC: soc-pcm: dpcm: Only allow playback/capture if supported") Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/intel/bdw.c | 16 +++++++++++++ sound/soc/sof/intel/byt.c | 48 +++++++++++++++++++++++++++++++++++++++ 2 files changed, 64 insertions(+) diff --git a/sound/soc/sof/intel/bdw.c b/sound/soc/sof/intel/bdw.c index 6c23c5769330..a32a3ef78ec5 100644 --- a/sound/soc/sof/intel/bdw.c +++ b/sound/soc/sof/intel/bdw.c @@ -567,9 +567,25 @@ static void bdw_set_mach_params(const struct snd_soc_acpi_mach *mach, static struct snd_soc_dai_driver bdw_dai[] = { { .name = "ssp0-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "ssp1-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, }; diff --git a/sound/soc/sof/intel/byt.c b/sound/soc/sof/intel/byt.c index f84391294f12..29fd1d86156c 100644 --- a/sound/soc/sof/intel/byt.c +++ b/sound/soc/sof/intel/byt.c @@ -459,21 +459,69 @@ static void byt_set_mach_params(const struct snd_soc_acpi_mach *mach, static struct snd_soc_dai_driver byt_dai[] = { { .name = "ssp0-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "ssp1-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "ssp2-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + } }, { .name = "ssp3-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "ssp4-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "ssp5-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, }; -- 2.20.1

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[PATCH 0/2] ASoC: rsnd: Fixes for multichannel HDMI audio output
by Matthias Blankertz 17 Apr '20

17 Apr '20

This fixes two issues in the snd-soc-rcar driver blocking multichannel HDMI audio out: The parent SSI in a multi-SSI configuration is not correctly set up and started, and the SSI->HDMI channel mapping is wrong. With these patches, the following device tree snippet can be used on an r8a7795-based platform (Salvator-X) to enable multichannel HDMI audio on HDMI0: rsnd_port1: port@1 { rsnd_endpoint1: endpoint { remote-endpoint = <&dw_hdmi0_snd_in>; dai-format = "i2s"; bitclock-master = <&rsnd_endpoint1>; frame-master = <&rsnd_endpoint1>; playback = <&ssi0 &ssi1 &ssi2 &ssi9>; }; }; With a capable receiver attached, all of 2ch (stereo), 6ch (e.g. 5.1) and 8ch audio output should work. Matthias Blankertz (2): ASoC: rsnd: Fix parent SSI start/stop in multi-SSI mode ASoC: rsnd: Fix HDMI channel mapping for multi-SSI mode sound/soc/sh/rcar/ssi.c | 8 ++++---- sound/soc/sh/rcar/ssiu.c | 2 +- 2 files changed, 5 insertions(+), 5 deletions(-) base-commit: 7111951b8d4973bda27ff663f2cf18b663d15b48 -- 2.26.0

3 6

Question about snd_soc_card_register()
by Sridharan, Ranjani 17 Apr '20

17 Apr '20

Hi Takashi, While working on implementing the probes features in SOF using a separate card for the probe DAI links, I noticed that calling snd_soc_register_card() results in incrementing the usage_count for the device that registers the card by 2 and it is not decremented until the card is freed. Is this the expected behaviour? Typically, we register a separate platform device for the Intel machines which in turn register the card and none of them ever enable runtime PM. So this has no impact on the parent device's runtime PM status. I'd like to avoid creating a separate platform device just to register the card if possible while also enabling runtime PM . But when I do this today, the device cannot enter runtime suspend at all. Could you please shed some light on this? Thanks, Ranjani

3 6

Re: next/master bisection: baseline.dmesg.alert on meson-sm1-sei610
by Mark Brown 17 Apr '20

17 Apr '20

On Fri, Apr 17, 2020 at 02:53:36AM -0700, kernelci.org bot wrote: > next/master bisection: baseline.dmesg.alert on meson-sm1-sei610 > Summary: > Start: a3ca59b9af21e Add linux-next specific files for 20200416 > Plain log: https://storage.kernelci.org//next/master/next-20200416/arm64/defconfig+CON… > HTML log: https://storage.kernelci.org//next/master/next-20200416/arm64/defconfig+CON… > Result: 9b5db059366ae ASoC: soc-pcm: dpcm: Only allow playback/capture if supported This change to check the playback and capture constraints is also causing issues on at least Meson - I'm a bit worried that this is also causing oopses here, not just audio problems so copying in Morimoto-san too. We should fix the things that are broken by this change since it's a thing we should be able to rely on and do enforce for non-DPCm links but I'm wondering if we should revert for 5.7 and reapply for 5.8. Including complete report so people have it. > Checks: > revert: PASS > verify: PASS > Parameters: > Tree: next > URL: https://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git > Branch: master > Target: meson-sm1-sei610 > CPU arch: arm64 > Lab: lab-baylibre > Compiler: gcc-8 > Config: defconfig+CONFIG_RANDOMIZE_BASE=y > Test case: baseline.dmesg.alert > > Breaking commit found: > > ------------------------------------------------------------------------------- > commit 9b5db059366ae2087e07892b5fc108f81f4ec189 > Author: Stephan Gerhold <stephan(a)gerhold.net> > Date: Wed Apr 15 12:49:28 2020 +0200 > > ASoC: soc-pcm: dpcm: Only allow playback/capture if supported > > At the moment, PCM devices for DPCM are only created based on the > dpcm_playback/capture parameters of the DAI link, without considering > if the CPU/FE DAI is actually capable of playback/capture. > > Normally the dpcm_playback/capture parameter should match the > capabilities of the CPU DAI. However, there is no way to set that > parameter from the device tree (e.g. with simple-audio-card or > qcom sound cards). dpcm_playback/capture are always both set to 1. > > This causes problems when the CPU DAI does only support playback > or capture. Attemting to open that PCM device with an unsupported > stream type then results in a null pointer dereference: > > Unable to handle kernel NULL pointer dereference at virtual address 0000000000000128 > Internal error: Oops: 96000044 [#1] PREEMPT SMP > CPU: 3 PID: 1582 Comm: arecord Not tainted 5.7.0-rc1 > pc : invalidate_paths_ep+0x30/0xe0 > lr : snd_soc_dapm_dai_get_connected_widgets+0x170/0x1a8 > Call trace: > invalidate_paths_ep+0x30/0xe0 > snd_soc_dapm_dai_get_connected_widgets+0x170/0x1a8 > dpcm_path_get+0x38/0xd0 > dpcm_fe_dai_open+0x70/0x920 > snd_pcm_open_substream+0x564/0x840 > snd_pcm_open+0xfc/0x228 > snd_pcm_capture_open+0x4c/0x78 > snd_open+0xac/0x1a8 > ... > > ... because the DAI playback/capture_widget is not set in that case. > > We could add checks there to fix the problem (maybe we should > anyway), but much easier is to not expose the device as > playback/capture in the first place. Attemting to use that > device would always fail later anyway. > > Add checks for snd_soc_dai_stream_valid() to the DPCM case > to avoid exposing playback/capture if it is not supported. > > Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> > Link: https://lore.kernel.org/r/20200415104928.86091-1-stephan@gerhold.net > Signed-off-by: Mark Brown <broonie(a)kernel.org> > > diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c > index 289aebc155293..1f302de440525 100644 > --- a/sound/soc/soc-pcm.c > +++ b/sound/soc/soc-pcm.c > @@ -2911,8 +2911,17 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) > int i; > > if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm) { > - playback = rtd->dai_link->dpcm_playback; > - capture = rtd->dai_link->dpcm_capture; > + cpu_dai = asoc_rtd_to_cpu(rtd, 0); > + if (rtd->num_cpus > 1) { > + dev_err(rtd->dev, > + "DPCM doesn't support Multi CPU yet\n"); > + return -EINVAL; > + } > + > + playback = rtd->dai_link->dpcm_playback && > + snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK); > + capture = rtd->dai_link->dpcm_capture && > + snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE); > } else { > /* Adapt stream for codec2codec links */ > int cpu_capture = rtd->dai_link->params ? > ------------------------------------------------------------------------------- > > > Git bisection log: > > ------------------------------------------------------------------------------- > git bisect start > # good: [87b0f983f66f23762921129fd35966eddc3f2dae] net: mscc: ocelot: fix untagged packet drops when enslaving to vlan aware bridge > git bisect good 87b0f983f66f23762921129fd35966eddc3f2dae > # bad: [a3ca59b9af21e68069555ffff1ad89bd2a7c40fc] Add linux-next specific files for 20200416 > git bisect bad a3ca59b9af21e68069555ffff1ad89bd2a7c40fc > # bad: [feb09551bfe34ccf0ba462188a1aee651be0f2c3] Merge remote-tracking branch 'i2c/i2c/for-next' > git bisect bad feb09551bfe34ccf0ba462188a1aee651be0f2c3 > # good: [a4721ced760684d1776bf31f7925aa41bb3f4846] Merge v5.7-rc1 into drm-misc-fixes > git bisect good a4721ced760684d1776bf31f7925aa41bb3f4846 > # bad: [e75043c693af6a10c9e2087adeef243cf05ce3bd] Merge remote-tracking branch 'rockchip/for-next' > git bisect bad e75043c693af6a10c9e2087adeef243cf05ce3bd > # bad: [08a315860a1b462fef5969d2e6ab4cdab26df8e3] Merge remote-tracking branch 'arm-soc/for-next' > git bisect bad 08a315860a1b462fef5969d2e6ab4cdab26df8e3 > # bad: [80dbfa57508c36c2b3798b387ef3eb92954b9c09] Merge remote-tracking branch 'slave-dma-fixes/fixes' > git bisect bad 80dbfa57508c36c2b3798b387ef3eb92954b9c09 > # good: [a723380c6f0a3723db72da6c6097b32b23f5564c] Merge remote-tracking branch 'sound-current/for-linus' > git bisect good a723380c6f0a3723db72da6c6097b32b23f5564c > # bad: [5bd70440cb0a6f5c6a84019bb2aa93ab8310a5cd] ASoC: soc-dai: revert all changes to DAI startup/shutdown sequence > git bisect bad 5bd70440cb0a6f5c6a84019bb2aa93ab8310a5cd > # good: [4d1a015a203c0249e3332ea217a38ec978118daa] ASoC: convert rockchip spdif bindings to yaml > git bisect good 4d1a015a203c0249e3332ea217a38ec978118daa > # good: [0f2a3b02274c02eb97697c4d89c019d1d21ac225] ASoC: wsa881x: mark read_only_wordlength flag > git bisect good 0f2a3b02274c02eb97697c4d89c019d1d21ac225 > # good: [aa7812737f2877e192d57626cbe8825cc7cf6de9] ASoC: sgtl5000: Fix VAG power-on handling > git bisect good aa7812737f2877e192d57626cbe8825cc7cf6de9 > # bad: [0c824ec094b5cda766c80d88c2036e28c24a4cb1] ASoC: q6dsp6: q6afe-dai: add missing channels to MI2S DAIs > git bisect bad 0c824ec094b5cda766c80d88c2036e28c24a4cb1 > # bad: [9b5db059366ae2087e07892b5fc108f81f4ec189] ASoC: soc-pcm: dpcm: Only allow playback/capture if supported > git bisect bad 9b5db059366ae2087e07892b5fc108f81f4ec189 > # first bad commit: [9b5db059366ae2087e07892b5fc108f81f4ec189] ASoC: soc-pcm: dpcm: Only allow playback/capture if supported > -------------------------------------------------------------------------------

3 4

[PATCH] ASoC: Intel: atom: remove unused 'sst_mix_voip_controls'
by Jason Yan 17 Apr '20

17 Apr '20

Fix the following gcc warning: sound/soc/intel/atom/sst-atom-controls.c:710:30: warning: ‘sst_mix_voip_controls’ defined but not used [-Wunused-const-variable=] SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_voip_controls); ^ Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Jason Yan <yanaijie(a)huawei.com> --- sound/soc/intel/atom/sst-atom-controls.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index 69f3af4524ab..541f0059ba7a 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -707,7 +707,6 @@ SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_pcm2_controls); SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_sprot_l0_controls); SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_media_l1_controls); SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_media_l2_controls); -SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_voip_controls); SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_codec0_controls); SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_codec1_controls); SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_modem_controls); -- 2.21.1

2 1

[PATCH v2] ASoC: bdw-rt5650: remove 3-channel capture support
by Brent Lu 17 Apr '20

17 Apr '20

Implement a constrain to exclude 3-channel capture since only 2 and 4 channel capture are supported on the platform. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/bdw-rt5650.c | 31 +++++++++++++++++++++++++++++++ 1 file changed, 31 insertions(+) diff --git a/sound/soc/intel/boards/bdw-rt5650.c b/sound/soc/intel/boards/bdw-rt5650.c index af2f502..eedbdad 100644 --- a/sound/soc/intel/boards/bdw-rt5650.c +++ b/sound/soc/intel/boards/bdw-rt5650.c @@ -83,6 +83,36 @@ static struct snd_soc_jack_pin mic_jack_pin = { .mask = SND_JACK_MICROPHONE, }; +static const unsigned int channels[] = { + 2, 4, +}; + +static const struct snd_pcm_hw_constraint_list constraints_channels = { + .count = ARRAY_SIZE(channels), + .list = channels, + .mask = 0, +}; + +static int bdw_fe_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + /* + * On this platform for PCM device we support, + * 2 or 4 channel capture + */ + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + + return 0; +} + +static const struct snd_soc_ops bdw_rt5650_fe_ops = { + .startup = bdw_fe_startup, +}; + static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { @@ -234,6 +264,7 @@ static struct snd_soc_dai_link bdw_rt5650_dais[] = { .name = "System PCM", .stream_name = "System Playback", .dynamic = 1, + .ops = &bdw_rt5650_fe_ops, #if !IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL) .init = bdw_rt5650_rtd_init, #endif -- 2.7.4

4 6

[PATCH alsa-lib] conf: USB-Audio: Fix S/PDIF output of ASUS Xonar AE
by Ivan Mironov 17 Apr '20

17 Apr '20

ASUS Xonar AE is a PCI-Express card containing USB controller: USB controller [0c03]: ASMedia Technology Inc. ASM1042A USB 3.0 Host Controller [1b21:1142] and the actual USB sound card: ID 0b05:180f ASUSTek Computer, Inc. XONAR SOUND CARD As other Xonar USB sound cards, it uses second device for digital output. Signed-off-by: Ivan Mironov <mironov.ivan(a)gmail.com> diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf index 89811086db1d..3f4c421b7e2c 100644 --- a/src/conf/cards/USB-Audio.conf +++ b/src/conf/cards/USB-Audio.conf @@ -40,6 +40,7 @@ USB-Audio.pcm.iec958_device { "USB Sound Blaster HD" 1 "Xonar U7" 1 "ASUS XONAR U5" 1 + "XONAR SOUND CARD" 1 # The below don't have digital in/out, so prevent them from being opened. "Andrea PureAudio USB-SA Headset" 999 -- 2.25.1

2 1

alsa / dma: Sound stops playing after xrun
by Martin Fuzzey 17 Apr '20

17 Apr '20

Hi all, I am seeing a sound problem leading to no more sound after an XRUN . I see more or less what is happening and have a work around but am not entirely sure of the correct solution. The problem occurs on an i.MX6DL using mainline kernel 5.4 with the fsl_ssi driver and a SGTL5000 audio codec connected over I2S. The userspace is Android 8 using tinnyhal + tinyalsa. This uses ioctl to push samples rather than mmap. The scenario is: 1) A buffer underrun occurs, causing -EPIPE to be returned to userspace. 2) Userpsapce (the tinyalsa component) does a pcm_prepare() to recover (which completes OK). 3) Userspace starts sending data again and then, when the start threshold is exceeded this kernel log is generated fsl-ssi-dai 2028000.ssi: Timeout waiting TX FIFO filling 4) From this point on no more sound is played, further writes timeout after 10s and return -EIO. My analsysis is as follows: When the underrun occurs snd_dmaengine_pcm_trigger(SNDRV_PCM_TRIGGER_STOP) is performed which does dmaengine_terminate_async() When the stream is restarted snd_dmaengine_pcm_trigger(SNDRV_PCM_TRIGGER_START does dmaengine_submit() dma_async_issue_pending() Because dmaengine_terminate_async() is asynchronus it sometimes completes after the dmaengine_submit(), causing the new DMA request to be cancelled before it has started. This results in the "Timeout waiting TX FIFO filling" message (in fsl_ssi_config_enable()) because it enables the SSI and expects data to be transfered to the FIFO by DMA. The message is only a warning, (void function with no error return) So without DMA running the buffer quicky fills up, resulting in all future writes timeouting waiting for buffer space. Where I'm not sure is if this is an ALSA bug or a bug in the i.MX6 SDMA controller driver. IE is it OK to do dmaengine_terminate_async() and later dmaengine_submit() on the same DMA channel without waiting for the terminate to complete? My workaround is to wait for the terminate to complete before returning -EPIPE (just after releasing the lock which prevents sleeping earlier). Ie this: diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 2236b5e..b03dac3 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -18,6 +18,9 @@ #include <sound/pcm_params.h> #include <sound/timer.h> +#include <linux/dmaengine.h> +#include <sound/dmaengine_pcm.h> + #include "pcm_local.h" #ifdef CONFIG_SND_PCM_XRUN_DEBUG @@ -2239,6 +2242,10 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream, } } _end_unlock: + /* MF: Workaround for broken sound after XRUN. */ + if (err == -EPIPE) + dmaengine_synchronize(snd_dmaengine_pcm_get_chan(substream)); + runtime->twake = 0; if (xfer > 0 && err >= 0) snd_pcm_update_state(substream, runtime); Regards, Martin

1 0

[PATCH] slimbus: ngd: remove redundant assignment
by Srinivas Kandagatla 17 Apr '20

17 Apr '20

remove redundant assignment, as this is already done fewlines lines before. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- drivers/slimbus/qcom-ngd-ctrl.c | 1 - 1 file changed, 1 deletion(-) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index 7426b5884218..743ee7b4e63f 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -1365,7 +1365,6 @@ static int of_qcom_slim_ngd_register(struct device *parent, platform_device_add(ngd->pdev); ngd->base = ctrl->base + ngd->id * data->offset + (ngd->id - 1) * data->size; - ctrl->ngd = ngd; return 0; } -- 2.21.0

1 0

[PATCH] slimbus: ngd: get drvdata from correct device
by Srinivas Kandagatla 17 Apr '20

17 Apr '20

Get drvdata directly from parent instead of ngd dev, as ngd dev can probe defer and previously set drvdata will become null. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- drivers/slimbus/qcom-ngd-ctrl.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index fc2575fef51b..7426b5884218 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -1361,7 +1361,6 @@ static int of_qcom_slim_ngd_register(struct device *parent, ngd->pdev->driver_override = QCOM_SLIM_NGD_DRV_NAME; ngd->pdev->dev.of_node = node; ctrl->ngd = ngd; - platform_set_drvdata(ngd->pdev, ctrl); platform_device_add(ngd->pdev); ngd->base = ctrl->base + ngd->id * data->offset + @@ -1376,12 +1375,13 @@ static int of_qcom_slim_ngd_register(struct device *parent, static int qcom_slim_ngd_probe(struct platform_device *pdev) { - struct qcom_slim_ngd_ctrl *ctrl = platform_get_drvdata(pdev); struct device *dev = &pdev->dev; + struct qcom_slim_ngd_ctrl *ctrl = dev_get_drvdata(dev->parent); int ret; ctrl->ctrl.dev = dev; + platform_set_drvdata(pdev, ctrl); pm_runtime_use_autosuspend(dev); pm_runtime_set_autosuspend_delay(dev, QCOM_SLIM_NGD_AUTOSUSPEND); pm_runtime_set_suspended(dev); -- 2.21.0

1 0

[PATCH v2] slimbus: core: Set fwnode for a device when setting of_node
by Saravana Kannan 17 Apr '20

17 Apr '20

When setting the of_node for a newly created device, also set the fwnode. This allows fw_devlink feature to work for slimbus devices. Signed-off-by: Saravana Kannan <saravanak(a)google.com> --- drivers/slimbus/core.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/drivers/slimbus/core.c b/drivers/slimbus/core.c index 526e3215d8fe..44228a5b246d 100644 --- a/drivers/slimbus/core.c +++ b/drivers/slimbus/core.c @@ -163,8 +163,10 @@ static int slim_add_device(struct slim_controller *ctrl, INIT_LIST_HEAD(&sbdev->stream_list); spin_lock_init(&sbdev->stream_list_lock); - if (node) + if (node) { sbdev->dev.of_node = of_node_get(node); + sbdev->dev.fwnode = of_fwnode_handle(node); + } dev_set_name(&sbdev->dev, "%x:%x:%x:%x", sbdev->e_addr.manf_id, -- 2.25.1.696.g5e7596f4ac-goog

3 3

[PATCH v1] slimbus: core: Fix mismatch in of_node_get/put
by Saravana Kannan 17 Apr '20

17 Apr '20

Also, remove some unnecessary NULL checks. The functions in question already do NULL checks. Signed-off-by: Saravana Kannan <saravanak(a)google.com> --- drivers/slimbus/core.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/drivers/slimbus/core.c b/drivers/slimbus/core.c index 44228a5b246d..ae1e248a8fb8 100644 --- a/drivers/slimbus/core.c +++ b/drivers/slimbus/core.c @@ -162,11 +162,8 @@ static int slim_add_device(struct slim_controller *ctrl, sbdev->ctrl = ctrl; INIT_LIST_HEAD(&sbdev->stream_list); spin_lock_init(&sbdev->stream_list_lock); - - if (node) { - sbdev->dev.of_node = of_node_get(node); - sbdev->dev.fwnode = of_fwnode_handle(node); - } + sbdev->dev.of_node = of_node_get(node); + sbdev->dev.fwnode = of_fwnode_handle(node); dev_set_name(&sbdev->dev, "%x:%x:%x:%x", sbdev->e_addr.manf_id, @@ -285,6 +282,7 @@ EXPORT_SYMBOL_GPL(slim_register_controller); /* slim_remove_device: Remove the effect of slim_add_device() */ static void slim_remove_device(struct slim_device *sbdev) { + of_node_put(sbdev->dev.of_node); device_unregister(&sbdev->dev); } -- 2.25.1.696.g5e7596f4ac-goog

2 1

[PATCH] ALSA: au88x0: remove some dead code
by Jason Yan 17 Apr '20

17 Apr '20

Fix the following gcc warning: sound/pci/au88x0/au88x0_a3ddata.c:62:25: warning: ‘A3dHrirDImpulse’ defined but not used [-Wunused-const-variable=] static const a3d_Hrtf_t A3dHrirDImpulse = { ^~~~~~~~~~~~~~~ sound/pci/au88x0/au88x0_a3ddata.c:50:25: warning: ‘A3dHrirSatTest’ defined but not used [-Wunused-const-variable=] static const a3d_Hrtf_t A3dHrirSatTest = { ^~~~~~~~~~~~~~ sound/pci/au88x0/au88x0_a3ddata.c:33:25: warning: ‘A3dHrirOnes’ defined but not used [-Wunused-const-variable=] static const a3d_Hrtf_t A3dHrirOnes = { ^~~~~~~~~~~ Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Jason Yan <yanaijie(a)huawei.com> --- sound/pci/au88x0/au88x0_a3d.c | 23 ------------------- sound/pci/au88x0/au88x0_a3ddata.c | 38 ------------------------------- 2 files changed, 61 deletions(-) diff --git a/sound/pci/au88x0/au88x0_a3d.c b/sound/pci/au88x0/au88x0_a3d.c index 2db183f8826a..4d3b79a34c69 100644 --- a/sound/pci/au88x0/au88x0_a3d.c +++ b/sound/pci/au88x0/au88x0_a3d.c @@ -512,31 +512,8 @@ static void a3dsrc_ProgramPipe(a3dsrc_t * a) /* Single spike leads to identity transfer function. */ a3dsrc_SetHrtfCurrent(a, A3dHrirImpulse, A3dHrirImpulse); a3dsrc_SetHrtfTarget(a, A3dHrirImpulse, A3dHrirImpulse); - - /* Test: Sounds saturated. */ - //a3dsrc_SetHrtfCurrent(a, A3dHrirSatTest, A3dHrirSatTest); - //a3dsrc_SetHrtfTarget(a, A3dHrirSatTest, A3dHrirSatTest); } -/* VDB = Vortex audio Dataflow Bus */ -#if 0 -static void a3dsrc_ClearVDBData(a3dsrc_t * a, unsigned long aa) -{ - vortex_t *vortex = (vortex_t *) (a->vortex); - - // ((aa >> 2) << 8) - (aa >> 2) - hwwrite(vortex->mmio, - a3d_addrS(a->slice, A3D_SLICE_VDBDest) + (a->source << 2), 0); - hwwrite(vortex->mmio, - a3d_addrS(a->slice, - A3D_SLICE_VDBDest + 4) + (a->source << 2), 0); - /* - hwwrite(vortex->mmio, 0x19c00 + (((aa>>2)*255*4)+aa)*8, 0); - hwwrite(vortex->mmio, 0x19c04 + (((aa>>2)*255*4)+aa)*8, 0); - */ -} -#endif - /* A3D HwSource stuff. */ static void vortex_A3dSourceHw_Initialize(vortex_t * v, int source, int slice) diff --git a/sound/pci/au88x0/au88x0_a3ddata.c b/sound/pci/au88x0/au88x0_a3ddata.c index 18623cb6bc52..cc41ea67a6d3 100644 --- a/sound/pci/au88x0/au88x0_a3ddata.c +++ b/sound/pci/au88x0/au88x0_a3ddata.c @@ -30,44 +30,6 @@ static const a3d_Hrtf_t A3dHrirImpulse = { 0, 0, 0 }; -static const a3d_Hrtf_t A3dHrirOnes = { - 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, - 0x7fff, - 0x7fff, - 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, - 0x7fff, - 0x7fff, - 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, - 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, - 0x7fff, - 0x7fff, - 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, - 0x7fff, - 0x7fff, - 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff -}; - -static const a3d_Hrtf_t A3dHrirSatTest = { - 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, - 0x7fff, - 0x7fff, - 0x8001, 0x8001, 0x8001, 0x8001, 0x8001, 0x8001, 0x8001, 0x8001, - 0x8001, - 0x8001, - 0x7fff, 0x0000, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, - 0, 0, 0, - 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 -}; - -static const a3d_Hrtf_t A3dHrirDImpulse = { - 0, 0x7fff, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, - 0, 0, 0, - 0, 0, 0, 0, - 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, - 0, 0, 0, - 0, 0, 0 -}; - static const a3d_ItdDline_t A3dItdDlineZeros = { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 -- 2.21.1

2 2

[PATCH] fix sw_params handling of timestamp types in direct plugins
by sylvain.bertrand＠gmail.com 17 Apr '20

17 Apr '20

In pcms using direct plugins (dmix/dsnoop/dshare), the timestamp type could be different from the terminating hw plugin, then the kernel driver. Be sure such pcms have plugins using consistently the same timestamp type. signed-off-by: Sylvain Bertrand <sylvain.bertrand(a)legeek.net> --- src/conf/alsa.conf | 1 + src/conf/pcm/dmix.conf | 4 ++++ src/conf/pcm/dsnoop.conf | 4 ++++ src/pcm/pcm_direct.c | 36 +++++++++++++++++++++++++++++++++++- src/pcm/pcm_direct.h | 2 ++ src/pcm/pcm_dmix.c | 4 ++++ src/pcm/pcm_dshare.c | 4 ++++ src/pcm/pcm_dsnoop.c | 4 ++++ src/pcm/pcm_local.h | 2 ++ 9 files changed, 60 insertions(+), 1 deletion(-) diff --git a/src/conf/alsa.conf b/src/conf/alsa.conf index a091b810..0e01c887 100644 --- a/src/conf/alsa.conf +++ b/src/conf/alsa.conf @@ -69,6 +69,7 @@ defaults.pcm.minperiodtime 5000 # in us defaults.pcm.ipc_key 5678293 defaults.pcm.ipc_gid audio defaults.pcm.ipc_perm 0660 +defaults.pcm.tstamp_type "default" defaults.pcm.dmix.max_periods 0 defaults.pcm.dmix.channels 2 defaults.pcm.dmix.rate 48000 diff --git a/src/conf/pcm/dmix.conf b/src/conf/pcm/dmix.conf index 7fa5c8b2..50e573da 100644 --- a/src/conf/pcm/dmix.conf +++ b/src/conf/pcm/dmix.conf @@ -56,6 +56,10 @@ pcm.!dmix { @func refer name defaults.pcm.ipc_perm } + tstamp_type { + @func refer + name defaults.pcm.tstamp_type + } slave { pcm { type hw diff --git a/src/conf/pcm/dsnoop.conf b/src/conf/pcm/dsnoop.conf index abbd44f7..f4336e5f 100644 --- a/src/conf/pcm/dsnoop.conf +++ b/src/conf/pcm/dsnoop.conf @@ -49,6 +49,10 @@ pcm.!dsnoop { @func refer name defaults.pcm.ipc_perm } + tstamp_type { + @func refer + name defaults.pcm.tstamp_type + } slave { pcm { type hw diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index 54d99005..aa60a477 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -991,8 +991,11 @@ int snd_pcm_direct_hw_free(snd_pcm_t *pcm ATTRIBUTE_UNUSED) return 0; } -int snd_pcm_direct_sw_params(snd_pcm_t *pcm ATTRIBUTE_UNUSED, snd_pcm_sw_params_t * params ATTRIBUTE_UNUSED) +int snd_pcm_direct_sw_params(snd_pcm_t *pcm, snd_pcm_sw_params_t *params) { + if (params->tstamp_type != pcm->tstamp_type) + return -EINVAL; + /* values are cached in the pcm structure */ return 0; } @@ -1318,6 +1321,15 @@ int snd_pcm_direct_initialize_slave(snd_pcm_direct_t *dmix, snd_pcm_t *spcm, str return ret; } + if (dmix->tstamp_type != -1) { + ret = snd_pcm_sw_params_set_tstamp_type(spcm, &sw_params, + dmix->tstamp_type); + if (ret < 0) { + SNDERR("unable to set tstamp type"); + return ret; + } + } + if (dmix->type != SND_PCM_TYPE_DMIX && dmix->type != SND_PCM_TYPE_DSHARE) goto __skip_silencing; @@ -1878,6 +1890,7 @@ int snd_pcm_direct_parse_open_conf(snd_config_t *root, snd_config_t *conf, rec->var_periodsize = 0; rec->direct_memory_access = 1; rec->hw_ptr_alignment = SND_PCM_HW_PTR_ALIGNMENT_AUTO; + rec->tstamp_type = -1; /* read defaults */ if (snd_config_search(root, "defaults.pcm.dmix_max_periods", &n) >= 0) { @@ -1941,6 +1954,27 @@ int snd_pcm_direct_parse_open_conf(snd_config_t *root, snd_config_t *conf, continue; } + if (strcmp(id, "tstamp_type") == 0) { + const char *str; + err = snd_config_get_string(n, &str); + if (err < 0) { + SNDERR("Invalid type for %s", id); + return -EINVAL; + } + if (strcmp(str, "default") == 0) + rec->tstamp_type = -1; + else if (strcmp(str, "gettimeofday") == 0) + rec->tstamp_type = SND_PCM_TSTAMP_TYPE_GETTIMEOFDAY; + else if (strcmp(str, "monotonic") == 0) + rec->tstamp_type = SND_PCM_TSTAMP_TYPE_MONOTONIC; + else if (strcmp(str, "monotonic_raw") == 0) + rec->tstamp_type = SND_PCM_TSTAMP_TYPE_MONOTONIC_RAW; + else { + SNDERR("The field tstamp_type is invalid : %s", str); + return -EINVAL; + } + continue; + } if (strcmp(id, "ipc_gid") == 0) { char *group; char *endp; diff --git a/src/pcm/pcm_direct.h b/src/pcm/pcm_direct.h index 221edbe1..8a236970 100644 --- a/src/pcm/pcm_direct.h +++ b/src/pcm/pcm_direct.h @@ -173,6 +173,7 @@ struct snd_pcm_direct { unsigned int recoveries; /* mirror of executed recoveries on slave */ int direct_memory_access; /* use arch-optimized buffer RW */ snd_pcm_direct_hw_ptr_alignment_t hw_ptr_alignment; + int tstamp_type; /* cached from conf, can be -1(default) on top of real types */ union { struct { int shmid_sum; /* IPC global sum ring buffer memory identification */ @@ -357,6 +358,7 @@ struct snd_pcm_direct_open_conf { int var_periodsize; int direct_memory_access; snd_pcm_direct_hw_ptr_alignment_t hw_ptr_alignment; + int tstamp_type; snd_config_t *slave; snd_config_t *bindings; }; diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index d533f40c..843fa316 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -1038,6 +1038,7 @@ int snd_pcm_dmix_open(snd_pcm_t **pcmp, const char *name, dmix->ipc_key = opts->ipc_key; dmix->ipc_perm = opts->ipc_perm; dmix->ipc_gid = opts->ipc_gid; + dmix->tstamp_type = opts->tstamp_type; dmix->semid = -1; dmix->shmid = -1; @@ -1237,6 +1238,9 @@ pcm.name { # roundup # rounddown # auto (default) + tstamp_type STR # timestamp type + # STR can be one of the below strings : + # default, gettimeofday, monotonic, monotonic_raw slave STR # or slave { # Slave definition diff --git a/src/pcm/pcm_dshare.c b/src/pcm/pcm_dshare.c index 59448cfb..6a99452b 100644 --- a/src/pcm/pcm_dshare.c +++ b/src/pcm/pcm_dshare.c @@ -723,6 +723,7 @@ int snd_pcm_dshare_open(snd_pcm_t **pcmp, const char *name, dshare->ipc_key = opts->ipc_key; dshare->ipc_perm = opts->ipc_perm; dshare->ipc_gid = opts->ipc_gid; + dshare->tstamp_type = opts->tstamp_type; dshare->semid = -1; dshare->shmid = -1; @@ -929,6 +930,9 @@ pcm.name { # roundup # rounddown # auto (default) + tstamp_type STR # timestamp type + # STR can be one of the below strings : + # default, gettimeofday, monotonic, monotonic_raw slave STR # or slave { # Slave definition diff --git a/src/pcm/pcm_dsnoop.c b/src/pcm/pcm_dsnoop.c index 24f472c7..c64df381 100644 --- a/src/pcm/pcm_dsnoop.c +++ b/src/pcm/pcm_dsnoop.c @@ -591,6 +591,7 @@ int snd_pcm_dsnoop_open(snd_pcm_t **pcmp, const char *name, dsnoop->ipc_key = opts->ipc_key; dsnoop->ipc_perm = opts->ipc_perm; dsnoop->ipc_gid = opts->ipc_gid; + dsnoop->tstamp_type = opts->tstamp_type; dsnoop->semid = -1; dsnoop->shmid = -1; @@ -780,6 +781,9 @@ pcm.name { # roundup # rounddown # auto (default) + tstamp_type STR # timestamp type + # STR can be one of the below strings : + # default, gettimeofday, monotonic, monotonic_raw slave STR # or slave { # Slave definition diff --git a/src/pcm/pcm_local.h b/src/pcm/pcm_local.h index 05ed935f..89d4125b 100644 --- a/src/pcm/pcm_local.h +++ b/src/pcm/pcm_local.h @@ -928,6 +928,8 @@ int INTERNAL(snd_pcm_hw_params_set_buffer_size_last)(snd_pcm_t *pcm, snd_pcm_hw_ int snd_pcm_sw_params_set_tstamp_mode(snd_pcm_t *pcm, snd_pcm_sw_params_t *params, snd_pcm_tstamp_t val); int INTERNAL(snd_pcm_sw_params_get_tstamp_mode)(const snd_pcm_sw_params_t *params, snd_pcm_tstamp_t *val); +int snd_pcm_sw_params_set_tstamp_type(snd_pcm_t *pcm, snd_pcm_sw_params_t *params, snd_pcm_tstamp_type_t val); +int snd_pcm_sw_params_get_tstamp_type(const snd_pcm_sw_params_t *params, snd_pcm_tstamp_type_t *val); int snd_pcm_sw_params_set_avail_min(snd_pcm_t *pcm, snd_pcm_sw_params_t *params, snd_pcm_uframes_t val); int INTERNAL(snd_pcm_sw_params_get_avail_min)(const snd_pcm_sw_params_t *params, snd_pcm_uframes_t *val); int snd_pcm_sw_params_set_start_threshold(snd_pcm_t *pcm, snd_pcm_sw_params_t *params, snd_pcm_uframes_t val);

2 1

[PATCH] ASoC: soc-pcm: dpcm: Only allow playback/capture if supported
by Stephan Gerhold 17 Apr '20

17 Apr '20

At the moment, PCM devices for DPCM are only created based on the dpcm_playback/capture parameters of the DAI link, without considering if the CPU/FE DAI is actually capable of playback/capture. Normally the dpcm_playback/capture parameter should match the capabilities of the CPU DAI. However, there is no way to set that parameter from the device tree (e.g. with simple-audio-card or qcom sound cards). dpcm_playback/capture are always both set to 1. This causes problems when the CPU DAI does only support playback or capture. Attemting to open that PCM device with an unsupported stream type then results in a null pointer dereference: Unable to handle kernel NULL pointer dereference at virtual address 0000000000000128 Internal error: Oops: 96000044 [#1] PREEMPT SMP CPU: 3 PID: 1582 Comm: arecord Not tainted 5.7.0-rc1 pc : invalidate_paths_ep+0x30/0xe0 lr : snd_soc_dapm_dai_get_connected_widgets+0x170/0x1a8 Call trace: invalidate_paths_ep+0x30/0xe0 snd_soc_dapm_dai_get_connected_widgets+0x170/0x1a8 dpcm_path_get+0x38/0xd0 dpcm_fe_dai_open+0x70/0x920 snd_pcm_open_substream+0x564/0x840 snd_pcm_open+0xfc/0x228 snd_pcm_capture_open+0x4c/0x78 snd_open+0xac/0x1a8 ... ... because the DAI playback/capture_widget is not set in that case. We could add checks there to fix the problem (maybe we should anyway), but much easier is to not expose the device as playback/capture in the first place. Attemting to use that device would always fail later anyway. Add checks for snd_soc_dai_stream_valid() to the DPCM case to avoid exposing playback/capture if it is not supported. Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- This can be reproduced with q6asm-dai with a device tree configuration like: &q6asmdai { dai@0 { reg = <0>; direction = <Q6ASM_DAI_RX>; }; }; Then a simple "arecord output.wav" will result in the crash above, since it attempts to use that FE DAI for capture (even though it is limited to playback). --- sound/soc/soc-pcm.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 454735f8fa92..77a680da366f 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2911,8 +2911,17 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) int i; if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm) { - playback = rtd->dai_link->dpcm_playback; - capture = rtd->dai_link->dpcm_capture; + cpu_dai = asoc_rtd_to_cpu(rtd, 0); + if (rtd->num_cpus > 1) { + dev_err(rtd->dev, + "DPCM doesn't support Multi CPU yet\n"); + return -EINVAL; + } + + playback = rtd->dai_link->dpcm_playback && + snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK); + capture = rtd->dai_link->dpcm_capture && + snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE); } else { /* Adapt stream for codec2codec links */ int cpu_capture = rtd->dai_link->params ? -- 2.26.1

3 4

sof-hda-dsp: Reverse order and priority of headset and headphone mic
by GitHub pull_request - opened 17 Apr '20

17 Apr '20

alsa-project/alsa-ucm-conf pull request #21 was opened from juimonen: Headset mic is the analog input source in 99% of the cases compared to headphone mic, so change it's priority higher. Jaska Uimonen <jaska.uimonen(a)linux.intel.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/21 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/21.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: qcom: lpass-cpu: support full duplex operation
by Srinivas Kandagatla 16 Apr '20

16 Apr '20

From: Takahide Higuchi <takahidehiguchi(a)gmail.com> This patch fixes a bug where playback on bidirectional I2S interface stops when we start recording on the same interface. We use regmap_update_bits instead of regmap_write so that we will not clear SPKEN and SPKMODE bits when we start/stop recording. Signed-off-by: Takahide Higuchi <takahidehiguchi(a)gmail.com> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/lpass-apq8016.c | 2 ++ sound/soc/qcom/lpass-cpu.c | 24 ++++++++++++++++++------ sound/soc/qcom/lpass-lpaif-reg.h | 2 +- 3 files changed, 21 insertions(+), 7 deletions(-) diff --git a/sound/soc/qcom/lpass-apq8016.c b/sound/soc/qcom/lpass-apq8016.c index 6575da549237..85079c697faa 100644 --- a/sound/soc/qcom/lpass-apq8016.c +++ b/sound/soc/qcom/lpass-apq8016.c @@ -121,6 +121,8 @@ static struct snd_soc_dai_driver apq8016_lpass_cpu_dai_driver[] = { }, .probe = &asoc_qcom_lpass_cpu_dai_probe, .ops = &asoc_qcom_lpass_cpu_dai_ops, + .symmetric_samplebits = 1, + .symmetric_rates = 1, }, }; diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index dbce7e92baf3..dc8acb380b6f 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -72,6 +72,7 @@ static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, snd_pcm_format_t format = params_format(params); unsigned int channels = params_channels(params); unsigned int rate = params_rate(params); + unsigned int mask; unsigned int regval; int bitwidth, ret; @@ -81,6 +82,9 @@ static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, return bitwidth; } + mask = LPAIF_I2SCTL_LOOPBACK_MASK | + LPAIF_I2SCTL_WSSRC_MASK | + LPAIF_I2SCTL_BITWIDTH_MASK; regval = LPAIF_I2SCTL_LOOPBACK_DISABLE | LPAIF_I2SCTL_WSSRC_INTERNAL; @@ -100,6 +104,7 @@ static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + mask |= LPAIF_I2SCTL_SPKMODE_MASK | LPAIF_I2SCTL_SPKMONO_MASK; switch (channels) { case 1: regval |= LPAIF_I2SCTL_SPKMODE_SD0; @@ -127,6 +132,7 @@ static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } } else { + mask |= LPAIF_I2SCTL_MICMODE_MASK | LPAIF_I2SCTL_MICMONO_MASK; switch (channels) { case 1: regval |= LPAIF_I2SCTL_MICMODE_SD0; @@ -155,9 +161,9 @@ static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, } } - ret = regmap_write(drvdata->lpaif_map, - LPAIF_I2SCTL_REG(drvdata->variant, dai->driver->id), - regval); + ret = regmap_update_bits(drvdata->lpaif_map, + LPAIF_I2SCTL_REG(drvdata->variant, dai->driver->id), + mask, regval); if (ret) { dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret); return ret; @@ -178,11 +184,17 @@ static int lpass_cpu_daiops_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + unsigned int mask; int ret; - ret = regmap_write(drvdata->lpaif_map, - LPAIF_I2SCTL_REG(drvdata->variant, dai->driver->id), - 0); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + mask = LPAIF_I2SCTL_SPKMODE_MASK; + else + mask = LPAIF_I2SCTL_MICMODE_MASK; + + ret = regmap_update_bits(drvdata->lpaif_map, + LPAIF_I2SCTL_REG(drvdata->variant, dai->driver->id), + mask, 0); if (ret) dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret); diff --git a/sound/soc/qcom/lpass-lpaif-reg.h b/sound/soc/qcom/lpass-lpaif-reg.h index 3d74ae123e9d..7a2b9cf99976 100644 --- a/sound/soc/qcom/lpass-lpaif-reg.h +++ b/sound/soc/qcom/lpass-lpaif-reg.h @@ -56,7 +56,7 @@ #define LPAIF_I2SCTL_MICMODE_6CH (7 << LPAIF_I2SCTL_MICMODE_SHIFT) #define LPAIF_I2SCTL_MICMODE_8CH (8 << LPAIF_I2SCTL_MICMODE_SHIFT) -#define LPAIF_I2SCTL_MIMONO_MASK GENMASK(3, 3) +#define LPAIF_I2SCTL_MICMONO_MASK GENMASK(3, 3) #define LPAIF_I2SCTL_MICMONO_SHIFT 3 #define LPAIF_I2SCTL_MICMONO_STEREO (0 << LPAIF_I2SCTL_MICMONO_SHIFT) #define LPAIF_I2SCTL_MICMONO_MONO (1 << LPAIF_I2SCTL_MICMONO_SHIFT) -- 2.21.0

3 3

[PATCH v1 1/2] dt-bindings: sound: add Microsemi ZL38060 binding
by Sven Van Asbroeck 16 Apr '20

16 Apr '20

Add devicetree binding for the Microsemi ZL38060 Connected Home Audio Processor. Signed-off-by: Sven Van Asbroeck <TheSven73(a)gmail.com> --- Tree: v5.7-rc1 .../devicetree/bindings/sound/zl38060.yaml | 90 +++++++++++++++++++ 1 file changed, 90 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/zl38060.yaml diff --git a/Documentation/devicetree/bindings/sound/zl38060.yaml b/Documentation/devicetree/bindings/sound/zl38060.yaml new file mode 100644 index 000000000000..8e1976bee76b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/zl38060.yaml @@ -0,0 +1,90 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/zl38060.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: ZL38060 Connected Home Audio Processor from Microsemi. + +description: | + The ZL38060 is a "Connected Home Audio Processor" from Microsemi, + which consists of a Digital Signal Processor (DSP), several Digital + Audio Interfaces (DAIs), analog outputs, and a block of 14 GPIOs. + +maintainers: + - Jaroslav Kysela <perex(a)perex.cz> + - Takashi Iwai <tiwai(a)suse.com> + +properties: + compatible: + const: mscc,zl38060 + + reg: + description: + SPI device address. + maxItems: 1 + + spi-max-frequency: + maximum: 24000000 + + reset-gpios: + description: + A GPIO line handling reset of the chip. As the line is active low, + it should be marked GPIO_ACTIVE_LOW (see ../gpio/gpio.txt) + maxItems: 1 + + mscc,load-firmware: + description: + If present, the driver will attempt to load firmware into the chip via + the SPI interface. + type: boolean + + '#gpio-cells': + const: 2 + + gpio-controller: true + + '#sound-dai-cells': + const: 0 + + amp-en-gpios: + description: + An optional GPIO line which is active when the codec is playing back + audio. Can be used to control an external amplifier. + maxItems: 1 + + mscc,amp-startup-delay-ms: + description: + An optional delay between amp-en-gpio becoming active, and the start + of audio playback. In milliseconds. + $ref: /schemas/types.yaml#/definitions/uint32 + +required: + - compatible + - reg + - '#gpio-cells' + - gpio-controller + - '#sound-dai-cells' + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + spi0 { + #address-cells = <1>; + #size-cells = <0>; + + codec: zl38060@0 { + gpio-controller; + #gpio-cells = <2>; + #sound-dai-cells = <0>; + compatible = "mscc,zl38060"; + reg = <0>; + spi-max-frequency = <12000000>; + reset-gpios = <&gpio1 0 GPIO_ACTIVE_LOW>; + amp-en-gpios = <&codec 3 GPIO_ACTIVE_HIGH>; + mscc,amp-startup-delay-ms = <12>; + mscc,load-firmware; + }; + }; -- 2.17.1

2 4

[alsa-devel] [PATCH] ASoC: stm32: sai: fix master clock management
by Olivier Moysan 16 Apr '20

16 Apr '20

When master clock is used, master clock rate is set exclusively. Parent clocks of master clock cannot be changed after a call to clk_set_rate_exclusive(). So the parent clock of SAI kernel clock must be set before. Ensure also that exclusive rate operations are balanced in STM32 SAI driver. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- sound/soc/stm/stm32_sai_sub.c | 64 +++++++++++++++++++++++++++++++------------ 1 file changed, 47 insertions(+), 17 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 3dd54bc54fa1..77aa38183955 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -70,6 +70,7 @@ #define SAI_IEC60958_STATUS_BYTES 24 #define SAI_MCLK_NAME_LEN 32 +#define SAI_RATE_11K 11025 /** * struct stm32_sai_sub_data - private data of SAI sub block (block A or B) @@ -311,6 +312,25 @@ static int stm32_sai_set_clk_div(struct stm32_sai_sub_data *sai, return ret; } +static int stm32_sai_set_parent_clock(struct stm32_sai_sub_data *sai, + unsigned int rate) +{ + struct platform_device *pdev = sai->pdev; + struct clk *parent_clk = sai->pdata->clk_x8k; + int ret; + + if (!(rate % SAI_RATE_11K)) + parent_clk = sai->pdata->clk_x11k; + + ret = clk_set_parent(sai->sai_ck, parent_clk); + if (ret) + dev_err(&pdev->dev, " Error %d setting sai_ck parent clock. %s", + ret, ret == -EBUSY ? + "Active stream rates conflict\n" : "\n"); + + return ret; +} + static long stm32_sai_mclk_round_rate(struct clk_hw *hw, unsigned long rate, unsigned long *prate) { @@ -492,25 +512,29 @@ static int stm32_sai_set_sysclk(struct snd_soc_dai *cpu_dai, struct stm32_sai_sub_data *sai = snd_soc_dai_get_drvdata(cpu_dai); int ret; - if (dir == SND_SOC_CLOCK_OUT) { + if (dir == SND_SOC_CLOCK_OUT && sai->sai_mclk) { ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, SAI_XCR1_NODIV, (unsigned int)~SAI_XCR1_NODIV); if (ret < 0) return ret; - dev_dbg(cpu_dai->dev, "SAI MCLK frequency is %uHz\n", freq); - sai->mclk_rate = freq; + /* If master clock is used, set parent clock now */ + ret = stm32_sai_set_parent_clock(sai, freq); + if (ret) + return ret; - if (sai->sai_mclk) { - ret = clk_set_rate_exclusive(sai->sai_mclk, - sai->mclk_rate); - if (ret) { - dev_err(cpu_dai->dev, - "Could not set mclk rate\n"); - return ret; - } + ret = clk_set_rate_exclusive(sai->sai_mclk, freq); + if (ret) { + dev_err(cpu_dai->dev, + ret == -EBUSY ? + "Active streams have incompatible rates" : + "Could not set mclk rate\n"); + return ret; } + + dev_dbg(cpu_dai->dev, "SAI MCLK frequency is %uHz\n", freq); + sai->mclk_rate = freq; } return 0; @@ -917,11 +941,13 @@ static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai, int div = 0, cr1 = 0; int sai_clk_rate, mclk_ratio, den; unsigned int rate = params_rate(params); + int ret; - if (!(rate % 11025)) - clk_set_parent(sai->sai_ck, sai->pdata->clk_x11k); - else - clk_set_parent(sai->sai_ck, sai->pdata->clk_x8k); + if (!sai->sai_mclk) { + ret = stm32_sai_set_parent_clock(sai, rate); + if (ret) + return ret; + } sai_clk_rate = clk_get_rate(sai->sai_ck); if (STM_SAI_IS_F4(sai->pdata)) { @@ -1080,9 +1106,13 @@ static void stm32_sai_shutdown(struct snd_pcm_substream *substream, regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, SAI_XCR1_NODIV, SAI_XCR1_NODIV); - clk_disable_unprepare(sai->sai_ck); + /* Release mclk rate only if rate was actually set */ + if (sai->mclk_rate) { + clk_rate_exclusive_put(sai->sai_mclk); + sai->mclk_rate = 0; + } - clk_rate_exclusive_put(sai->sai_mclk); + clk_disable_unprepare(sai->sai_ck); spin_lock_irqsave(&sai->irq_lock, flags); sai->substream = NULL; -- 2.7.4

2 3

[PATCH 1/3] ASoC: codecs: hdac_hdmi: Fix incorrect use of list_for_each_entry
by Amadeusz Sławiński 16 Apr '20

16 Apr '20

If we don't find any pcm, pcm will point at address at an offset from the the list head and not a meaningful structure. Fix this by returning correct pcm if found and NULL if not. Found with coccinelle. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- sound/soc/codecs/hdac_hdmi.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index fba9b749839d..f26b77faed59 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -142,14 +142,14 @@ static struct hdac_hdmi_pcm * hdac_hdmi_get_pcm_from_cvt(struct hdac_hdmi_priv *hdmi, struct hdac_hdmi_cvt *cvt) { - struct hdac_hdmi_pcm *pcm = NULL; + struct hdac_hdmi_pcm *pcm; list_for_each_entry(pcm, &hdmi->pcm_list, head) { if (pcm->cvt == cvt) - break; + return pcm; } - return pcm; + return NULL; } static void hdac_hdmi_jack_report(struct hdac_hdmi_pcm *pcm, -- 2.17.1

2 4

Volume is not restored after boot
by GitHub issues - opened 16 Apr '20

16 Apr '20

alsa-project/alsa-utils issue #39 was opened from Oscar65: I have archlinux updated system last sunday. ``` $ uname -a Linux mypc 5.6.3-arch1-1 #1 SMP PREEMPT Wed, 08 Apr 2020 07:47:16 +0000 x86_64 GNU/Linux ``` I checked with alsa-restore.service and alsa-state.service. None of them work as expected. My sound card is: ``` # lshw -C sound *-multimedia description: Audio device product: 82801H (ICH8 Family) HD Audio Controller vendor: Intel Corporation physical id: 1b bus info: pci@0000:00:1b.0 version: 02 width: 64 bits clock: 33MHz capabilities: pm msi pciexpress bus_master cap_list configuration: driver=snd_hda_intel latency=0 resources: irq:29 memory:ffaf8000-ffafbfff ``` Every time that my system boots the volume levels are different than last time a powered off. Issue URL : https://github.com/alsa-project/alsa-utils/issues/39 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

alsactl: Wrong line if (!sched_setscheduler(0, SCHED_IDLE, &sched_param))
by GitHub pull_request - opened 16 Apr '20

16 Apr '20

alsa-project/alsa-utils pull request #38 was opened from Oscar65: As man page says: "If successful, the sched_setparam() function shall return zero." So the if must be without ! Without update I got this output in the syslog (journalctl): abr 16 09:25:30 mypc alsactl[1652]: alsactl 1.2.2 daemon started abr 16 09:25:30 mypc alsactl[1652]: /usr/bin/alsactl: do_nice:165sched_setparam failed: No such file or directory That I think that is a wrong message. The call to sched_setparam is correct. This is the output with strace: getpriority(PRIO_PROCESS, 0) = 20 setpriority(PRIO_PROCESS, 0, 19) = 0 getpriority(PRIO_PROCESS, 0) = 1 sched_getparam(0, [0]) = 0 sched_setscheduler(0, SCHED_IDLE, [0]) = 0 write(2, "/usr/bin/alsactl: do_nice:165: ", 31/usr/bin/alsactl: do_nice:165: ) = 31 write(2, "sched_setparam failed: No such f"..., 48sched_setparam failed: No such file or directory) = 48 write(2, "\n", 1) = 1 Call to sched_setscheduler returns 0, so it means that the call was successful. Request URL : https://github.com/alsa-project/alsa-utils/pull/38 Patch URL : https://github.com/alsa-project/alsa-utils/pull/38.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[RFC TEST] ASoC: soc-dai: revert all changes to DAI startup/shutdown sequence
by Pierre-Louis Bossart 16 Apr '20

16 Apr '20

On Baytrail/Cherrytrail, the Atom/SST driver fails miserably: [ 9.741953] intel_sst_acpi 80860F28:00: FW Version 01.0c.00.01 [ 9.832992] intel_sst_acpi 80860F28:00: FW sent error response 0x40034 [ 9.833019] intel_sst_acpi 80860F28:00: FW alloc failed ret -4 [ 9.833028] intel_sst_acpi 80860F28:00: sst_get_stream returned err -5 [ 9.833033] sst-mfld-platform sst-mfld-platform: ASoC: DAI prepare error: -5 [ 9.833037] Baytrail Audio Port: ASoC: prepare FE Baytrail Audio Port failed [ 9.853942] intel_sst_acpi 80860F28:00: FW sent error response 0x40034 [ 9.853974] intel_sst_acpi 80860F28:00: FW alloc failed ret -4 [ 9.853984] intel_sst_acpi 80860F28:00: sst_get_stream returned err -5 [ 9.853990] sst-mfld-platform sst-mfld-platform: ASoC: DAI prepare error: -5 [ 9.853994] Baytrail Audio Port: ASoC: prepare FE Baytrail Audio Port failed Commit b56be800f1292 ("ASoC: soc-pcm: call snd_soc_dai_startup()/shutdown() once") was the initial problematic commit. Commit 1ba616bd1a6d5e ("ASoC: soc-dai: fix DAI startup/shutdown sequence") was an attempt to fix things but it does not work on Baytrail, reverting all changes seems necessary for now. Fixes: 1ba616bd1a6d5e ("ASoC: soc-dai: fix DAI startup/shutdown sequence") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- Sending as RFC since I don't have a good understanding of the root-cause and for others to confirm my findings. Tested on top of v5.7-rc1. include/sound/soc-dai.h | 1 - sound/soc/soc-dai.c | 11 ++--------- 2 files changed, 2 insertions(+), 10 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index d4825b82c7a3..b33abe93b905 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -351,7 +351,6 @@ struct snd_soc_dai { /* bit field */ unsigned int probed:1; - unsigned int started[SNDRV_PCM_STREAM_LAST + 1]; }; static inline struct snd_soc_pcm_stream * diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 8f3cad8db89a..31c41559034b 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -295,24 +295,17 @@ int snd_soc_dai_startup(struct snd_soc_dai *dai, { int ret = 0; - if (!dai->started[substream->stream] && - dai->driver->ops->startup) + if (dai->driver->ops->startup) ret = dai->driver->ops->startup(substream, dai); - if (ret == 0) - dai->started[substream->stream] = 1; - return ret; } void snd_soc_dai_shutdown(struct snd_soc_dai *dai, struct snd_pcm_substream *substream) { - if (dai->started[substream->stream] && - dai->driver->ops->shutdown) + if (dai->driver->ops->shutdown) dai->driver->ops->shutdown(substream, dai); - - dai->started[substream->stream] = 0; } int snd_soc_dai_prepare(struct snd_soc_dai *dai, -- 2.20.1

3 6

[PATCH v2 0/5] Add support for SOF on i.MX8M
by Daniel Baluta 16 Apr '20

16 Apr '20

From: Daniel Baluta <daniel.baluta(a)nxp.com> This patch series adds support for SOF on i.MX8M family. First board from this family that has a DSP is i.MX8MP. First 2 patches are trying to fix some compilation issues, the next two are adding the imx8m support and the last one adds the devicetree binding. Changes since v2: - add reviewed by from Rob to DT patch - fix ownership for patch 2 Daniel Baluta (3): ASoC: SOF: imx: Add i.MX8M HW support ASoC: SOF: Add i.MX8MP device descriptor dt-bindings: dsp: fsl: Add fsl,imx8mp-dsp entry Pierre-Louis Bossart (1): ASoC: SOF: imx: fix undefined reference issue YueHaibing (1): ASoC: SOF: imx8: Fix randbuild error .../devicetree/bindings/dsp/fsl,dsp.yaml | 2 + sound/soc/sof/imx/Kconfig | 32 +- sound/soc/sof/imx/Makefile | 2 + sound/soc/sof/imx/imx8m.c | 279 ++++++++++++++++++ sound/soc/sof/sof-of-dev.c | 14 + 5 files changed, 325 insertions(+), 4 deletions(-) create mode 100644 sound/soc/sof/imx/imx8m.c -- 2.17.1

2 6

[PATCH 00/24] ASoC: SOF: topology and firmware IPC updates for 5.8
by Pierre-Louis Bossart 16 Apr '20

16 Apr '20

Long series made of a relatively small changes from multiple SOF contributors. I didn't find a good way to split this series since it tracks SOF minor ABI changes (backwards-compatible with older firmware files) and needs to be kept in-order. Future series should be much shorter. The main addition is support for an extended firmware manifest, which helps retrieve capabilities directly from the firmware file instead of the current IPC mechanism (still supported but will be deprecated). The IPC is realigned with the firmware, along with type cleanups, and the DMIC interface is simplified. The topology changes are mainly about a multi-cpu DAI fix, a new DC blocking component, better parsing of tuples and new parameters for ALH (SoundWire) and HDaudio DAIs. New tokens are also added to clarify the firmware behavior in the case of dependent pipelines, e.g. for echo reference generation. Artur Kloniecki (1): ASoC: SOF: Add XRUN flags field to struct sof_ipc_buffer. Bard Liao (5): ASoC: SOF: topology: fix: handle DAI widget connections properly with multiple CPU DAI's ASoC: SOF: align sof_ipc_dai_alh_params with FW ASoC: SOF: topology: Get ALH rate amd channels from topology ASoC: SOF: topology: fix: parse hda_tokens to &config->hda ASoC: SOF: topology: Get HDA rate and channels from topology Jaska Uimonen (2): ASoC: SOF: topology: stop parsing when all tokens have been found ASoC: SOF: topology: handle multiple sets of tuple arrays Karol Trzcinski (6): ASoC: SOF: Mark get_ext* function ext_hdr arguments as const ASoC: SOF: Introduce offset in firmware data ASoC: SOF: Introduce extended manifest ASoC: SOF: ext_manifest: parse firmware version ASoC: SOF: ext_manifest: parse windows ASoC: SOF: ext_manifest: parse compiler version Pan Xiuli (6): ASoC: SOF: add probe support extend data ASoC: SOF: add debug ABI version ASoC: SOF: change type char to uint8_t in info.h ASoC: SOF: change type char to uint8_t in trace.h ASoC: SOF: change type char to uint8_t in topology.h ASoC: SOF: make sof_ipc_cc_version to fixed length Sebastiano Carlucci (1): ASoC: SOF: topology: Add support for DC Blocker Seppo Ingalsuo (3): ASoC: SOF: Intel: Fix typo in header file comment text ASoC: SOF: Intel: Change DMIC load IPC to fixed length ASoC: SOF: Intel: Rename deprecated DMIC IPC struct field include/sound/sof.h | 3 + include/sound/sof/dai-intel.h | 20 +- include/sound/sof/info.h | 26 ++- include/sound/sof/topology.h | 16 +- include/sound/sof/trace.h | 2 +- include/uapi/sound/sof/abi.h | 2 +- include/uapi/sound/sof/ext_manifest.h | 91 ++++++++ include/uapi/sound/sof/tokens.h | 8 + sound/soc/sof/intel/hda-loader.c | 9 +- sound/soc/sof/loader.c | 226 ++++++++++++++++-- sound/soc/sof/topology.c | 323 ++++++++++++++++---------- 11 files changed, 568 insertions(+), 158 deletions(-) create mode 100644 include/uapi/sound/sof/ext_manifest.h base-commit: 83b35f4586e235bfb785a7947b555ad8f3d96887 -- 2.20.1

2 25

[alsa-devel] [PATCH] ASoC: pcm: fix error handling when try_module_get() fails.
by Ranjani Sridharan 16 Apr '20

16 Apr '20

Handle error before returning when try_module_get() fails to prevent inconsistent mutex lock/unlock. Fixes: 52034add7 (ASoC: pcm: update module refcount if module_get_upon_open is set) Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/soc-pcm.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index d21247546f7f..be80a12fba27 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -518,8 +518,10 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) continue; if (component->driver->module_get_upon_open && - !try_module_get(component->dev->driver->owner)) - return -ENODEV; + !try_module_get(component->dev->driver->owner)) { + ret = -ENODEV; + goto module_err; + } ret = component->driver->ops->open(substream); if (ret < 0) { @@ -636,7 +638,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) component_err: soc_pcm_components_close(substream, component); - +module_err: if (cpu_dai->driver->ops->shutdown) cpu_dai->driver->ops->shutdown(substream, cpu_dai); out: -- 2.17.1

2 5

[alsa-devel] [PATCH] ASoC: wcd9335: Fix missing regmap requirement
by Marc Gonzalez 16 Apr '20

16 Apr '20

wcd9335.c: undefined reference to 'devm_regmap_add_irq_chip' Signed-off-by: Marc Gonzalez <marc.w.gonzalez(a)free.fr> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 419142111b6d..33778dc99108 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1162,6 +1162,7 @@ config SND_SOC_WCD9335 tristate "WCD9335 Codec" depends on SLIMBUS select REGMAP_SLIMBUS + select REGMAP_IRQ help The WCD9335 is a standalone Hi-Fi audio CODEC IC, supports Qualcomm Technologies, Inc. (QTI) multimedia solutions, -- 2.17.1

3 4

Baytrail audio not working on next branch
by Jaime Perez 15 Apr '20

15 Apr '20

tested 20200409 20200413 20200414 tags maybe related issue https://bugzilla.kernel.org/show_bug.cgi?id=201787

2 2

[PATCH 1/2] ASoC: topology: Remove unneeded semicolon
by Amadeusz Sławiński 15 Apr '20

15 Apr '20

There is unnecessary semicolon after last bracket of if statement, remove it. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- sound/soc/soc-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 33e8d189ba2f..a5cd73742dc5 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -2390,7 +2390,7 @@ static int soc_tplg_link_elems_load(struct soc_tplg *tplg, tplg->pos += le32_to_cpu(hdr->size) + le32_to_cpu(hdr->payload_size); return 0; - }; + } /* check the element size and count */ link = (struct snd_soc_tplg_link_config *)tplg->pos; -- 2.17.1

2 3