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

[alsa-devel] [PATCH 00/12] ASoC: SOF: Data probing
by Cezary Rojewski 27 Jan '20

27 Jan '20

This set of patches achieves few goals in order to enable data probing feature for audio DSP. First, provide new and alter existing interfaces (page allocation, runtime flow adjustments) to make them compress friendly. For HDA part, work has been done to account for compress streams when servicing IRQs, setting up BDLs and assigning DMAs. Finally, the end goal which are the probe APIs and usage itself. Probes can be treated as endpoints which allow for data extraction from or injection to target module - a great ally when debugging problematic audio issues such as distortions, glitches or gaps. Compress streams are a weapon of choice here to provide a lightweight implementation. While all available IPCs have been defined, current implementation covers extraction only, with injection scheduled for a later date. Initial review and development of probes can be found under: https://github.com/thesofproject/linux/pull/1276 with the hda-compress-enable set of patches being separated and reviewed on: https://github.com/thesofproject/linux/pull/1571 Tested on CML-U with rt5682 i2s board. Cezary Rojewski (12): ALSA: hda: Allow for compress stream to hdac_ext_stream assignment ALSA: hda: Prepare for compress stream support ALSA: hda: Interrupt servicing and BDL setup for compress streams ALSA: core: Expand DMA buffer information ALSA: core: Implement compress page allocation and free routines ASoC: SOF: Intel: Account for compress streams when servicing IRQs ASoC: SOF: Implement Probe IPC API ASoC: SOF: Generic probe compress operations ASoC: SOF: Intel: Probe compress operations ASoC: SOF: Provide probe debugfs support ASoC: SOF: Intel: Add Probe compress CPU DAIs ASoC: Intel: sof_rt5682: Add compress probe DAI links include/sound/compress_driver.h | 40 +++- include/sound/hdaudio.h | 2 + include/sound/hdaudio_ext.h | 2 + include/sound/sof/header.h | 11 ++ sound/core/compress_offload.c | 42 ++++ sound/hda/ext/hdac_ext_stream.c | 49 ++++- sound/hda/hdac_controller.c | 4 +- sound/hda/hdac_stream.c | 52 +++-- sound/soc/intel/boards/sof_rt5682.c | 20 +- sound/soc/sof/Kconfig | 9 + sound/soc/sof/Makefile | 3 +- sound/soc/sof/compress.c | 139 ++++++++++++++ sound/soc/sof/compress.h | 29 +++ sound/soc/sof/debug.c | 208 ++++++++++++++++++++ sound/soc/sof/intel/Kconfig | 10 + sound/soc/sof/intel/Makefile | 1 + sound/soc/sof/intel/apl.c | 9 + sound/soc/sof/intel/cnl.c | 9 + sound/soc/sof/intel/hda-compress.c | 132 +++++++++++++ sound/soc/sof/intel/hda-dai.c | 28 +++ sound/soc/sof/intel/hda-ipc.c | 4 +- sound/soc/sof/intel/hda-stream.c | 26 ++- sound/soc/sof/intel/hda.h | 30 +++ sound/soc/sof/ops.h | 43 +++++ sound/soc/sof/pcm.c | 11 +- sound/soc/sof/probe.c | 287 ++++++++++++++++++++++++++++ sound/soc/sof/probe.h | 85 ++++++++ sound/soc/sof/sof-priv.h | 24 +++ 28 files changed, 1266 insertions(+), 43 deletions(-) create mode 100644 sound/soc/sof/compress.c create mode 100644 sound/soc/sof/compress.h create mode 100644 sound/soc/sof/intel/hda-compress.c create mode 100644 sound/soc/sof/probe.c create mode 100644 sound/soc/sof/probe.h -- 2.17.1

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[alsa-devel] [PATCH v2 0/3] SOF: Fix HDMI probe errors on GLK devices
by Kai Vehmanen 27 Jan '20

27 Jan '20

Hi Mark and Pierre, here's a patch series to address an issue that popped up after SOF changed from hdac-hdmi to snd-hda-codec-hdmi: "failed to get afg sub nodes on ChromeOS devices" https://github.com/thesofproject/linux/issues/1642 v2 changes: - rebased on top of broonie/for-next - added Takashi's reviewed-by to the one HDA patch, agreed the series should go via asoc tree This is fairly hard to hit and only occurs on some devices (most current reports are Gemini Lake based Chromebooks), but when it does, it is rather severe as the whole SOF probe fails because of this, and user is left without sound. First fix is to optimize out one power down/up cycle from the probe process. On platforms such as GLK, each time audio driver does acomp's get_power(), graphics typically needs to do a modeset due to clocking requirements for the HDA bus. This can cause display to flash when audio is probed, plus in the reported cases on GLK, can lead to probe failures. The above change doesn't cover all reported cases, so additionally retry logic is added to snd_hda_get_sub_nodes() calls on Intel platforms. Multiple other approaches were investigated, but a simple retry -- although less than ideal -- in the end proved to be most reliable solution. HDA dump on one affected Acer Chromebook looks like this: # codec initialization goes as normal with multiple successful cmds udevd-9486 [001] .... 3910.087791: hda_send_cmd: [0000:00:0e.0:2] val=0x207f1c00 udevd-9486 [001] .... 3910.087857: hda_get_response: [0000:00:0e.0:2] val=0x18560010 udevd-9486 [001] .... 3910.087858: hda_send_cmd: [0000:00:0e.0:2] val=0x207f0700 udevd-9486 [001] .... 3910.087931: hda_get_response: [0000:00:0e.0:2] val=0x00000000 udevd-9486 [001] .... 3910.087932: hda_send_cmd: [0000:00:0e.0:2] val=0x20bf8100 udevd-9486 [001] .... 3910.088040: hda_get_response: [0000:00:0e.0:2] val=0x00000001 udevd-9486 [001] .... 3910.088044: hda_send_cmd: [0000:00:0e.0:2] val=0x20b78103 # here get params for AC_PAR_NODE_COUNT fails, 0xffffffff is returned udevd-9486 [001] .... 3910.088048: hda_send_cmd: [0000:00:0e.0:2] val=0x201f0004 udevd-9486 [001] .... 3911.090131: hda_get_response: [0000:00:0e.0:2] val=0xffffffff # retry is successful udevd-9486 [001] .... 3911.090152: hda_send_cmd: [0000:00:0e.0:2] val=0x201f0004 udevd-9486 [001] .... 3911.090288: hda_get_response: [0000:00:0e.0:2] val=0x00020006 Kai Vehmanen (3): ASoC: SOF: Intel: refactor i915_get/put functions ASoC: SOF: Intel: do not disable i915 power during probe ALSA: hda/hdmi - add retry logic to parse_intel_hdmi() sound/pci/hda/patch_hdmi.c | 7 +++++-- sound/soc/sof/intel/hda-codec.c | 21 ++++++--------------- sound/soc/sof/intel/hda.c | 3 ++- sound/soc/sof/intel/hda.h | 7 +++---- 4 files changed, 16 insertions(+), 22 deletions(-) -- 2.17.1

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[alsa-devel] [PATCH] ASoC: amd: Fix for Subsequent Playback issue.
by Ravulapati Vishnu vardhan rao 27 Jan '20

27 Jan '20

If we play audio back to back, which kills one playback and immediately start another, we can hear clicks. This patch fixes the issue. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 5c3ec3c..aecc3c0 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -349,13 +349,6 @@ static int acp3x_dma_close(struct snd_soc_component *component, component = snd_soc_rtdcom_lookup(prtd, DRV_NAME); adata = dev_get_drvdata(component->dev); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - adata->play_stream = NULL; - adata->i2ssp_play_stream = NULL; - } else { - adata->capture_stream = NULL; - adata->i2ssp_capture_stream = NULL; - } /* Disable ACP irq, when the current stream is being closed and * another stream is also not active. @@ -363,6 +356,13 @@ static int acp3x_dma_close(struct snd_soc_component *component, if (!adata->play_stream && !adata->capture_stream && !adata->i2ssp_play_stream && !adata->i2ssp_capture_stream) rv_writel(0, adata->acp3x_base + mmACP_EXTERNAL_INTR_ENB); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + adata->play_stream = NULL; + adata->i2ssp_play_stream = NULL; + } else { + adata->capture_stream = NULL; + adata->i2ssp_capture_stream = NULL; + } return 0; } -- 2.7.4

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[alsa-devel] ASUS T100TAM UCM
by Michal Feix 27 Jan '20

27 Jan '20

Hello, there is no sound working via speakers or headphones jack on Asus T100TAM. Sound works correctly via HDMI or Bluetooth headphones. The chip on this Baytrail machine is correctly recognized as bytcr_rt5640. From dmesg: bytcr_rt5640 bytcr_rt5640: quirk DMIC1_MAP enabled bytcr_rt5640 bytcr_rt5640: quirk realtek,jack-detect-source 3 bytcr_rt5640 bytcr_rt5640: quirk realtek,over-current-threshold-microamp 2000 bytcr_rt5640 bytcr_rt5640: quirk realtek,over-current-scale-factor 1 bytcr_rt5640 bytcr_rt5640: quirk MCLK_EN enabled bytcr_rt5640 bytcr_rt5640: snd-soc-dummy-dai <-> media-cpu-dai mapping ok bytcr_rt5640 bytcr_rt5640: snd-soc-dummy-dai <-> deepbuffer-cpu-dai mapping ok bytcr_rt5640 bytcr_rt5640: rt5640-aif1 <-> ssp2-port mapping ok input: bytcr-rt5640 Headset as /devices/platform/80860F28:00/bytcr_rt5640/sound/card0/input17 After bootup, kernel log is filled with: "Baytrail Audio Port: ASoC: no backend DAIs enabled for Baytrail Audio Port" messages. I do understand this is due to the missing UCM profile for the machine. I've successfully uploaded the output of alsa-info.sh. I'm on Arch linux, with latest alsa-ucm-conf 1.2.1.2 which seems not to be sufficient to make sound working on this thing. Is there anything I can provide to help smart Alsa people to help with this? Thank you, -- Michal

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[alsa-devel] [PATCH] ALSA: hda: Add Clevo W65_67SB the power_save blacklist
by Hans de Goede 26 Jan '20

26 Jan '20

Using HDA power-saving on the Clevo W65_67SB causes the first 0.5 seconds of audio to be missing every time audio starts playing. This commit adds the Clevo W65_67SB the power_save blacklist to avoid this issue. Cc: stable(a)vger.kernel.org BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1525104 Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 5b92f290cbb0..54d9ea1750f9 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2185,6 +2185,8 @@ static struct snd_pci_quirk power_save_blacklist[] = { /* https://bugzilla.redhat.com/show_bug.cgi?id=1581607 */ SND_PCI_QUIRK(0x1558, 0x3501, "Clevo W35xSS_370SS", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ + SND_PCI_QUIRK(0x1558, 0x6504, "Clevo W65_67SB", 0), + /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ SND_PCI_QUIRK(0x1028, 0x0497, "Dell Precision T3600", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ /* Note the P55A-UD3 and Z87-D3HP share the subsys id for the HDA dev */ -- 2.23.0

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[alsa-devel] [PATCH v2] ASoC: cros_ec_codec: Support setting bclk ratio
by Yu-Hsuan Hsu 26 Jan '20

26 Jan '20

Support setting bclk ratio from machine drivers. Signed-off-by: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> --- sound/soc/codecs/cros_ec_codec.c | 25 ++++++++++++++++++++++--- 1 file changed, 22 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/cros_ec_codec.c b/sound/soc/codecs/cros_ec_codec.c index 6a24f570c5e86f..d3dc42aa682565 100644 --- a/sound/soc/codecs/cros_ec_codec.c +++ b/sound/soc/codecs/cros_ec_codec.c @@ -45,6 +45,9 @@ struct cros_ec_codec_priv { /* DMIC */ atomic_t dmic_probed; + /* I2S_RX */ + uint32_t i2s_rx_bclk_ratio; + /* WoV */ bool wov_enabled; uint8_t *wov_audio_shm_p; @@ -259,6 +262,7 @@ static int i2s_rx_hw_params(struct snd_pcm_substream *substream, snd_soc_component_get_drvdata(component); struct ec_param_ec_codec_i2s_rx p; enum ec_codec_i2s_rx_sample_depth depth; + uint32_t bclk; int ret; if (params_rate(params) != 48000) @@ -284,15 +288,29 @@ static int i2s_rx_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - dev_dbg(component->dev, "set bclk to %u\n", - snd_soc_params_to_bclk(params)); + if (priv->i2s_rx_bclk_ratio) + bclk = params_rate(params) * priv->i2s_rx_bclk_ratio; + else + bclk = snd_soc_params_to_bclk(params); + + dev_dbg(component->dev, "set bclk to %u\n", bclk); p.cmd = EC_CODEC_I2S_RX_SET_BCLK; - p.set_bclk_param.bclk = snd_soc_params_to_bclk(params); + p.set_bclk_param.bclk = bclk; return send_ec_host_command(priv->ec_device, EC_CMD_EC_CODEC_I2S_RX, (uint8_t *)&p, sizeof(p), NULL, 0); } +static int i2s_rx_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_component *component = dai->component; + struct cros_ec_codec_priv *priv = + snd_soc_component_get_drvdata(component); + + priv->i2s_rx_bclk_ratio = ratio; + return 0; +} + static int i2s_rx_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_component *component = dai->component; @@ -340,6 +358,7 @@ static int i2s_rx_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) static const struct snd_soc_dai_ops i2s_rx_dai_ops = { .hw_params = i2s_rx_hw_params, .set_fmt = i2s_rx_set_fmt, + .set_bclk_ratio = i2s_rx_set_bclk_ratio, }; static int i2s_rx_event(struct snd_soc_dapm_widget *w, -- 2.25.0.341.g760bfbb309-goog

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[alsa-devel] [PATCH v2] ALSA: usb-audio: add quirks for Line6 Helix devices fw>=2.82
by Nicola Lunghi 26 Jan '20

26 Jan '20

With firmware 2.82 Line6 changed the usb id of some of the Helix devices but the quirks is still needed. Add it to the quirk list for line6 helix family of devices. Thanks to Jens for pointing out the missing ids. Signed-off-by: Nicola Lunghi <nick83ola(a)gmail.com> --- sound/usb/format.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/usb/format.c b/sound/usb/format.c index d79db71305f6..53922f73467f 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -296,6 +296,9 @@ static int line6_parse_audio_format_rates_quirk(struct snd_usb_audio *chip, case USB_ID(0x0E41, 0x4242): /* Line6 Helix Rack */ case USB_ID(0x0E41, 0x4244): /* Line6 Helix LT */ case USB_ID(0x0E41, 0x4246): /* Line6 HX-Stomp */ + case USB_ID(0x0E41, 0x4248): /* Line6 Helix >= fw 2.82 */ + case USB_ID(0x0E41, 0x4249): /* Line6 Helix Rack >= fw 2.82 */ + case USB_ID(0x0E41, 0x424a): /* Line6 Helix LT >= fw 2.82 */ /* supported rates: 48Khz */ kfree(fp->rate_table); fp->rate_table = kmalloc(sizeof(int), GFP_KERNEL); -- 2.20.1

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[alsa-devel] [PATCH] ALSA: usb-audio: add quirks for Line6 Helix fw 2.82
by Nicola Lunghi 25 Jan '20

25 Jan '20

Line6 Helix fw 2.82 still needs a quirk to set the audio rate to 48khz but in respect to older firmware has changed the USB ID of the device. Add it to the quirk list for line6 helix family of devices. Signed-off-by: Nicola Lunghi <nick83ola(a)gmail.com> --- sound/usb/format.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/format.c b/sound/usb/format.c index d79db71305f6..e2b98f181d05 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -296,6 +296,7 @@ static int line6_parse_audio_format_rates_quirk(struct snd_usb_audio *chip, case USB_ID(0x0E41, 0x4242): /* Line6 Helix Rack */ case USB_ID(0x0E41, 0x4244): /* Line6 Helix LT */ case USB_ID(0x0E41, 0x4246): /* Line6 HX-Stomp */ + case USB_ID(0x0E41, 0x4248): /* Line6 Helix >= fw 2.82 */ /* supported rates: 48Khz */ kfree(fp->rate_table); fp->rate_table = kmalloc(sizeof(int), GFP_KERNEL); -- 2.20.1

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Re: [alsa-devel] Line 6 Helix quirks
by nick83ola 25 Jan '20

25 Jan '20

Hi I verified on my hx_stomp with the latest driver and I can confirm tha the quirks works here Sadly I don't have an helix to verify it What model it is? you can try to recompile the kernel adding the line with you usb device to the see my patch https://lkml.org/lkml/2019/7/1/850 + case USB_ID(0x0E41, 0x4244): /* HELIX */ + case USB_ID(0x0E41, 0x4246): /* HX STOMP */ + case USB_ID(0x0E41, 0x4248): /* what kind of HELIX ???????? */ Cheers Nicola Lunghi On Sat, 19 Oct 2019 at 06:34, Carlo Calica <carlo(a)calica.com> wrote: > > Hi, > > I'm attempting to use my Helix with the newish 5.3.x kernel. It isn't working. Looks like the quirk isn't enabled for the idProduct (4248) of my Helix. My Helix's firmware is 2.81.0. I have yet to compile the kernel to see if adding the id is enough. Thanks. If you need any more info, please let me know. > > dmesg gives: > > [89014.296954] usb 3-2: new high-speed USB device number 2 using xhci_hcd > [89014.521720] usb 3-2: New USB device found, idVendor=0e41, idProduct=4248, bcdDevice= 2.00 > [89014.521723] usb 3-2: New USB device strings: Mfr=1, Product=2, SerialNumber=3 > [89014.521724] usb 3-2: Product: HELIX > [89014.521726] usb 3-2: Manufacturer: LINE 6 > [89014.521727] usb 3-2: SerialNumber: 2796112 > [89014.555715] usb 3-2: parse_audio_format_rates_v2v3(): unable to retrieve number of sample rates (clock 16) > [89014.559724] usb 3-2: parse_audio_format_rates_v2v3(): unable to retrieve number of sample rates (clock 16) > [89014.575990] input: LINE 6 HELIX as /devices/pci0000:00/0000:00:01.3/0000:03:00.2/0000:16:08.0/0000:1c:00.0/usb3/3-2/3-2:1.5/0003:0E41:4248.000C/input/input50 > [89014.628069] hid-generic 0003:0E41:4248.000C: input,hidraw8: USB HID v1.11 Keyboard [LINE 6 HELIX ] on usb-0000:1c:00.0-2/input5 > > > -- > Carlo J. Calica

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[alsa-devel] [PATCH] ASoC: rt1015: Remove unnecessary const
by Nathan Chancellor 24 Jan '20

24 Jan '20

Clang warns: ../sound/soc/codecs/rt1015.c:392:14: warning: duplicate 'const' declaration specifier [-Wduplicate-decl-specifier] static const SOC_ENUM_SINGLE_DECL(rt1015_boost_mode_enum, 0, 0, ^ ../include/sound/soc.h:355:2: note: expanded from macro 'SOC_ENUM_SINGLE_DECL' SOC_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xtexts) ^ ../include/sound/soc.h:352:2: note: expanded from macro 'SOC_ENUM_DOUBLE_DECL' const struct soc_enum name = SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, \ ^ 1 warning generated. Remove the const after static to fix it. Fixes: df31007400c3 ("ASoC: rt1015: add rt1015 amplifier driver") Link: https://github.com/ClangBuiltLinux/linux/issues/845 Signed-off-by: Nathan Chancellor <natechancellor(a)gmail.com> --- sound/soc/codecs/rt1015.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 4a9c5b54008f..6d490e2dbc25 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -389,7 +389,7 @@ static const char * const rt1015_boost_mode[] = { "Bypass", "Adaptive", "Fixed Adaptive" }; -static const SOC_ENUM_SINGLE_DECL(rt1015_boost_mode_enum, 0, 0, +static SOC_ENUM_SINGLE_DECL(rt1015_boost_mode_enum, 0, 0, rt1015_boost_mode); static int rt1015_boost_mode_get(struct snd_kcontrol *kcontrol, -- 2.25.0

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