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

error: possibly undefined macro: AM_PATH_ALSA
by GitHub issues - edited 27 Apr '20

27 Apr '20

alsa-project/alsa-utils issue #40 was edited from zeerd: Sorry, I pressed the wrong button. I deleted this. Issue URL : https://github.com/alsa-project/alsa-utils/issues/40 Repository URL: https://github.com/alsa-project/alsa-utils

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error: possibly undefined macro: AM_PATH_ALSA
by GitHub issues - opened 27 Apr '20

27 Apr '20

alsa-project/alsa-utils issue #40 was opened from zeerd: Hello! Seems alsa-utils using gettext, but have no check in the configure.ac But my question is , could we just ignore the using of gettext by some parameter of configure? My target-board-sdk has no gettext. Issue URL : https://github.com/alsa-project/alsa-utils/issues/40 Repository URL: https://github.com/alsa-project/alsa-utils

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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

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[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

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[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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