April 2020 - Alsa-devel - mailman.alsa-project.org

[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

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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

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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

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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

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[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

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[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

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[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

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[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

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[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

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[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

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