December 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: dma-sh7760: cleanup a debug printk
by Dan Carpenter 18 Dec '18

18 Dec '18

The intent was to print the address as a hexadecimal but there is an extra "u" in the "0x%08ulx" format specification so it is displayed as decimal. Fixes: aef3b06ac697 ("[ALSA] SH7760 ASoC support") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/sh/dma-sh7760.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index 922fb6aa3ed1..5aee11c94f2a 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -202,7 +202,7 @@ static int camelot_prepare(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct camelot_pcm *cam = &cam_pcm_data[rtd->cpu_dai->id]; - pr_debug("PCM data: addr 0x%08ulx len %d\n", + pr_debug("PCM data: addr 0x%08lx len %d\n", (u32)runtime->dma_addr, runtime->dma_bytes); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { -- 2.17.1

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[alsa-devel] [PATCH v2 1/5] ASoC: qcom: sdm845: Add board specific dapm widgets
by Cheng-Yi Chiang 18 Dec '18

18 Dec '18

Add board specific dapm widgets so these widgets can be used in the route. Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- sound/soc/qcom/sdm845.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 58593db2ab151..95d8d4422dae0 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -212,6 +212,14 @@ static int sdm845_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } +static const struct snd_soc_dapm_widget sdm845_snd_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), +}; + static void sdm845_add_be_ops(struct snd_soc_card *card) { struct snd_soc_dai_link *link; @@ -243,6 +251,8 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev) goto data_alloc_fail; } + card->dapm_widgets = sdm845_snd_widgets; + card->num_dapm_widgets = ARRAY_SIZE(sdm845_snd_widgets); card->dev = dev; dev_set_drvdata(dev, card); ret = qcom_snd_parse_of(card); -- 2.20.0.rc0.387.gc7a69e6b6c-goog

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[alsa-devel] [PATCH 0/9] ASoC: audio/simple card tidyup/cleanup
by Kuninori Morimoto 18 Dec '18

18 Dec '18

Hi Mark We could merged normal sound and DPCM sound on audio/simple card. This is nice, but I noticed few issues. 1) : for "reg" property issue, again. 2) - 5) : missing 1CPU-1Codec DPCM case, and cleanup for audio-graph 6) - 9) : missing 1CPU-1Codec DPCM case, and cleanup for simple-card Kuninori Morimoto (9): 1) ASoC: simple-card-utils: check "reg" property on asoc_simple_card_get_dai_id() 2) ASoC: audio-graph-card: add asoc_graph_card_get_conversion() 3) ASoC: audio-graph-card: add 1 CPU : 1 Codec support again 4) ASoC: audio-graph-card: cleanup DAI link loop method 5) ASoC: audio-graph-card: reduce naming prefix 6) ASoC: simple-card: add asoc_simple_card_get_conversion() 7) ASoC: simple-card: add 1 CPU : 1 Codec support again 8) ASoC: simple-card: cleanup DAI link loop method 9) ASoC: simple-card: reduce naming prefix sound/soc/generic/audio-graph-card.c | 461 +++++++++++++++++++--------------- sound/soc/generic/simple-card-utils.c | 14 +- sound/soc/generic/simple-card.c | 453 +++++++++++++++++++-------------- 3 files changed, 544 insertions(+), 384 deletions(-) -- 2.7.4

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[alsa-devel] PATCH] ALSA: hda/realtek: Enable audio jacks of ASUS UX433FN variant with ALC294
by Wandrille RONCE 17 Dec '18

17 Dec '18

From 5a244e278c7bb626c901d668d359f47681e40647 Mon Sep 17 00:00:00 2001 From: Wandrille RONCE <w(a)ndrille.fr> Date: Mon, 17 Dec 2018 19:30:00 +0100 Subject: [PATCH] ALSA: hda/realtek: Enable audio jacks of ASUS UX433FN variant with ALC294 --- 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 90f30e769a42..6f34f81125e0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6584,6 +6584,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x13b0, "ASUS Z550SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), SND_PCI_QUIRK(0x1043, 0x14a1, "ASUS UX533FD", ALC294_FIXUP_ASUS_SPK), + SND_PCI_QUIRK(0x1043, 0x16ce, "ASUS UX433FN", ALC294_FIXUP_ASUS_SPK), SND_PCI_QUIRK(0x1043, 0x10a1, "ASUS UX391UA", ALC294_FIXUP_ASUS_SPK), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), -- 2.19.1

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Re: [alsa-devel] [Sound-open-firmware] [PATCH v3 13/14] ASoC: SOF: Add xtensa support
by rander.wang 17 Dec '18

17 Dec '18

在 12/13/2018 2:15 PM, rander.wang 写道: > > 在 12/12/2018 8:00 AM, Pierre-Louis Bossart 写道: >> >> On 12/11/18 5:08 PM, Andy Shevchenko wrote: >>> On Tue, Dec 11, 2018 at 03:23:17PM -0600, Pierre-Louis Bossart wrote: >>>> Add common directory for xtensa architecture >>>> + for (i = 0; i <= stack_words - 4; i += 4) { >>>> + dev_err(sdev->dev, "0x%8.8x: 0x%8.8x 0x%8.8x 0x%8.8x >>>> 0x%8.8x\n", >>>> + stack_ptr + i, stack[i], stack[i + 1], stack[i + 2], >>>> + stack[i + 3]); >>>> + } >>>> + >>>> + /* deal with any remaining words */ >>>> + switch (stack_words - i) { >>>> + case 0: >>>> + break; >>>> + case 1: >>>> + dev_err(sdev->dev, "0x%8.8x: 0x%8.8x\n", >>>> + stack_ptr + stack_words - 1, stack[stack_words - 1]); >>>> + break; >>>> + case 2: >>>> + dev_err(sdev->dev, "0x%8.8x: 0x%8.8x 0x%8.8x\n", >>>> + stack_ptr + stack_words - 2, stack[stack_words - 2], >>>> + stack[stack_words - 1]); >>>> + break; >>>> + case 3: >>>> + dev_err(sdev->dev, "0x%8.8x: 0x%8.8x 0x%8.8x 0x%8.8x\n", >>>> + stack_ptr + stack_words - 3, stack[stack_words - 3], >>>> + stack[stack_words - 2], stack[stack_words - 1]); >>>> + break; >>>> + default: >>>> + break; >>>> + } >>> hex_dump_to_buffer(). >> I vaguely recall there was a reason not to do it, we'll relook at this. >> > we don't want to convert a blob of data to "hex ASCII" in memory > (hex_dump_to_buffer). > > Here, we just want to dump the content of call stack, like: > > [ 3.250628] sof-audio sof-audio: 0x0049fbb0: 0x8000f2d0 0x0049fc00 > 0x6f6c6c61 0x00632e63 > [ 3.250670] sof-audio sof-audio: 0x0049fbb4: 0x00ffffff 0x0010b5d4 > 0x00000000 0x2000004c > [ 3.250710] sof-audio sof-audio: 0x0049fbb8: 0x0040d000 0x00000000 > 0x0000024c 0x2000930c > [ 3.250747] sof-audio sof-audio: 0x0049fbbc: 0x8000454c 0x0049fc30 > 0x00401e20 0x00000001 > [ 3.250787] sof-audio sof-audio: 0x0049fbc0: 0x8000454c 0x0049fc30 > 0x00401e20 0x00000001 > [ 3.250827] sof-audio sof-audio: 0x0049fbc4: 0x00000000 0x00000001 > 0x00000010 0x0049fcb8 > [ 3.250865] sof-audio sof-audio: 0x0049fbc8: 0x0040a000 0x00401e98 > 0x00000001 0x00000002 > [ 3.250889] sof-audio sof-audio: 0x0049fbcc: 0x80005ac5 0x0049fc60 > 0x00000000 0x00000000 Add more details: we use another tool to analysis the hex numbers with a ELF file to get the correct information, so we do it by ourself. Thanks! Rander >> _______________________________________________ >> Sound-open-firmware mailing list >> Sound-open-firmware(a)alsa-project.org >> http://mailman.alsa-project.org/mailman/listinfo/sound-open-firmware

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Re: [alsa-devel] [PATCH 01/21] ASoC: SOF: Intel: Add BYT, CHT and BSW DSP HW support.
by Yang, Libin 17 Dec '18

17 Dec '18

Hi Sakamoto, > >-------- Forwarded Message -------- >Subject: Re: [alsa-devel] [PATCH 01/21] ASoC: SOF: Intel: Add BYT, CHT and >BSW DSP HW support. >Date: Wed, 12 Dec 2018 13:08:48 +0900 >From: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> >To: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>, >alsa-devel(a)alsa-project.org >CC: Daniel Baluta <daniel.baluta(a)gmail.com>, andriy.shevchenko(a)intel.com, >tiwai(a)suse.de, Pan Xiuli <xiuli.pan(a)linux.intel.com>, >liam.r.girdwood(a)linux.intel.com, vkoul(a)kernel.org, broonie(a)kernel.org, >sound-open-firmware(a)alsa-project.org, Rander Wang ><rander.wang(a)linux.intel.com>, Alan Cox <alan(a)linux.intel.com> > >Hi, > >I have small nitpicking. > >On 2018/12/12 6:30, Pierre-Louis Bossart wrote: >> From: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> >> >> Add support for the audio DSP hardware found on Intel Baytrail, >> Cherrytrail and Braswell based devices. >> >> Signed-off-by: Rander Wang <rander.wang(a)linux.intel.com> >> Signed-off-by: Pan Xiuli <xiuli.pan(a)linux.intel.com> >> Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> >> Signed-off-by: Pierre-Louis Bossart >> <pierre-louis.bossart(a)linux.intel.com> >> --- >> sound/soc/sof/intel/byt.c | 805 >+++++++++++++++++++++++++++++++++++++ >> sound/soc/sof/intel/shim.h | 159 ++++++++ >> 2 files changed, 964 insertions(+) >> create mode 100644 sound/soc/sof/intel/byt.c >> create mode 100644 sound/soc/sof/intel/shim.h .. >> diff --git a/sound/soc/sof/intel/shim.h b/sound/soc/sof/intel/shim.h >> new file mode 100644 index 000000000000..2fdcb2d262f4 >> --- /dev/null >> +++ b/sound/soc/sof/intel/shim.h >> @@ -0,0 +1,159 @@ >> ... >> +extern struct snd_sof_dsp_ops sof_byt_ops; extern struct >> +snd_sof_dsp_ops sof_cht_ops; > >These two symbols are found in this patch. > >> +extern struct snd_sof_dsp_ops sof_hsw_ops; > >This symbol is added in second patch and this first patch has no reference to it. > >> +extern struct snd_sof_dsp_ops sof_bdw_ops; > >This symbol is added in third patch and this first patch has no reference to it. > >> + >> +#endif > >I think it better to add these extern declarations when adding corresponding >symbols (Of course no issues as is). > >Furthermore, if you have no plan to change contents of these symbols in >kernel run time, it's better to have 'const' qualifier to locate the symbol to >readonly section. However, user of these symbols is machine driver and you >have a plan to implement it later. They can get 'const' >when you work for arrangement of existent/new codes. > >In this meaning, it might be good to add a new member to 'struct >snd_soc_acpi_mach' for new drivers to refer to the const ops, then 'pdata' >member is used from the existent drivers. I made a patch based on your suggestion. Could you please help review to see if my understanding is right? https://github.com/thesofproject/linux/pull/457 Regards, Libin > > >Regards > >Takashi Sakamoto >_______________________________________________ >Alsa-devel mailing list >Alsa-devel(a)alsa-project.org >http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

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[alsa-devel] [PATCH] ASoC: pcm512x: Implement the digital_mute interface
by Dimitris Papavasiliou 17 Dec '18

17 Dec '18

Clicks and pops of various volumes can be produced while the device is opened, closed, put into and taken out of standby, or reconfigured. Fix this, by implementing the digital_mute interface, so that the output is muted during such operations. Signed-off-by: Dimitris Papavasiliou <dpapavas(a)gmail.com> --- Notes: Although the datasheet isn't very specific about it, PCM512x_ANALOG_MUTE_DET is only updated once ramp-down to digital mute is complete (followed by analog mute, assuming PCM512x_ANALOG_MUTE_CTRL is left with the default value, as is now the case). I've measured the time it takes for the polling to finish and it seems consistent with that assumption, including properly tracking changes in the ramp-down parameters. Thus, polling the register ensures that the device has been completely muted before returning. When unmuting, the time required for PCM512x_ANALOG_MUTE_DET to reflect the change, doesn't depend on the ramp-up parameters (as might be expected, since analog mute is likely lifted first). Nevertheless the value is consistently substantially larger than zero, so that it seemingly reflects a particular stage in the unmuting process. It also does no harm and seems to help minimize clicks, so I've kept the delay. I've tested this patch both in my day-to-day use of my DAC for more than a month, as well as by running a script that opens the device, performs several sample rate and format changes and closes it, every one second. Without the patch, several different kinds of clicks and pops are produced. After the patch, I've run the script for several hours with the DAC being consistently virtually silent. sound/soc/codecs/pcm512x.c | 121 ++++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/pcm512x.h | 2 + 2 files changed, 121 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index f0f2d4fd3769..6cb1653be804 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -53,6 +53,8 @@ struct pcm512x_priv { unsigned long overclock_pll; unsigned long overclock_dac; unsigned long overclock_dsp; + int mute; + struct mutex mutex; }; /* @@ -384,6 +386,61 @@ static const struct soc_enum pcm512x_veds = SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_2, PCM512x_VEDS_SHIFT, 4, pcm512x_ramp_step_text); +static int pcm512x_update_mute(struct pcm512x_priv *pcm512x) +{ + return regmap_update_bits( + pcm512x->regmap, PCM512x_MUTE, PCM512x_RQML | PCM512x_RQMR, + (!!(pcm512x->mute & 0x5) << PCM512x_RQML_SHIFT) + | (!!(pcm512x->mute & 0x3) << PCM512x_RQMR_SHIFT)); +} + +static int pcm512x_digital_playback_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct pcm512x_priv *pcm512x = snd_soc_component_get_drvdata(component); + + mutex_lock(&pcm512x->mutex); + ucontrol->value.integer.value[0] = !(pcm512x->mute & 0x4); + ucontrol->value.integer.value[1] = !(pcm512x->mute & 0x2); + mutex_unlock(&pcm512x->mutex); + + return 0; +} + +static int pcm512x_digital_playback_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct pcm512x_priv *pcm512x = snd_soc_component_get_drvdata(component); + int ret, changed = 0; + + mutex_lock(&pcm512x->mutex); + + if ((pcm512x->mute & 0x4) == (ucontrol->value.integer.value[0] << 2)) { + pcm512x->mute ^= 0x4; + changed = 1; + } + if ((pcm512x->mute & 0x2) == (ucontrol->value.integer.value[1] << 1)) { + pcm512x->mute ^= 0x2; + changed = 1; + } + + if (changed) { + ret = pcm512x_update_mute(pcm512x); + if (ret != 0) { + dev_err(component->dev, + "Failed to update digital mute: %d\n", ret); + mutex_unlock(&pcm512x->mutex); + return ret; + } + } + + mutex_unlock(&pcm512x->mutex); + + return changed; +} + static const struct snd_kcontrol_new pcm512x_controls[] = { SOC_DOUBLE_R_TLV("Digital Playback Volume", PCM512x_DIGITAL_VOLUME_2, PCM512x_DIGITAL_VOLUME_3, 0, 255, 1, digital_tlv), @@ -391,8 +448,15 @@ SOC_DOUBLE_TLV("Analogue Playback Volume", PCM512x_ANALOG_GAIN_CTRL, PCM512x_LAGN_SHIFT, PCM512x_RAGN_SHIFT, 1, 1, analog_tlv), SOC_DOUBLE_TLV("Analogue Playback Boost Volume", PCM512x_ANALOG_GAIN_BOOST, PCM512x_AGBL_SHIFT, PCM512x_AGBR_SHIFT, 1, 0, boost_tlv), -SOC_DOUBLE("Digital Playback Switch", PCM512x_MUTE, PCM512x_RQML_SHIFT, - PCM512x_RQMR_SHIFT, 1, 1), +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Digital Playback Switch", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ctl_boolean_stereo_info, + .get = pcm512x_digital_playback_switch_get, + .put = pcm512x_digital_playback_switch_put +}, SOC_SINGLE("Deemphasis Switch", PCM512x_DSP, PCM512x_DEMP_SHIFT, 1, 1), SOC_ENUM("DSP Program", pcm512x_dsp_program), @@ -1319,10 +1383,61 @@ static int pcm512x_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } +static int pcm512x_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_component *component = dai->component; + struct pcm512x_priv *pcm512x = snd_soc_component_get_drvdata(component); + int ret; + unsigned int mute_det; + + mutex_lock(&pcm512x->mutex); + + if (mute) { + pcm512x->mute |= 0x1; + ret = regmap_update_bits(pcm512x->regmap, PCM512x_MUTE, + PCM512x_RQML | PCM512x_RQMR, + PCM512x_RQML | PCM512x_RQMR); + if (ret != 0) { + dev_err(component->dev, + "Failed to set digital mute: %d\n", ret); + mutex_unlock(&pcm512x->mutex); + return ret; + } + + regmap_read_poll_timeout(pcm512x->regmap, + PCM512x_ANALOG_MUTE_DET, + mute_det, (mute_det & 0x3) == 0, + 200, 10000); + + mutex_unlock(&pcm512x->mutex); + } else { + pcm512x->mute &= ~0x1; + ret = pcm512x_update_mute(pcm512x); + if (ret != 0) { + dev_err(component->dev, + "Failed to update digital mute: %d\n", ret); + mutex_unlock(&pcm512x->mutex); + return ret; + } + + regmap_read_poll_timeout(pcm512x->regmap, + PCM512x_ANALOG_MUTE_DET, + mute_det, + (mute_det & 0x3) + == ((~pcm512x->mute >> 1) & 0x3), + 200, 10000); + } + + mutex_unlock(&pcm512x->mutex); + + return 0; +} + static const struct snd_soc_dai_ops pcm512x_dai_ops = { .startup = pcm512x_dai_startup, .hw_params = pcm512x_hw_params, .set_fmt = pcm512x_set_fmt, + .digital_mute = pcm512x_digital_mute, }; static struct snd_soc_dai_driver pcm512x_dai = { @@ -1388,6 +1503,8 @@ int pcm512x_probe(struct device *dev, struct regmap *regmap) if (!pcm512x) return -ENOMEM; + mutex_init(&pcm512x->mutex); + dev_set_drvdata(dev, pcm512x); pcm512x->regmap = regmap; diff --git a/sound/soc/codecs/pcm512x.h b/sound/soc/codecs/pcm512x.h index d70d9c0c2088..9dda8693498e 100644 --- a/sound/soc/codecs/pcm512x.h +++ b/sound/soc/codecs/pcm512x.h @@ -112,7 +112,9 @@ #define PCM512x_RQST_SHIFT 4 /* Page 0, Register 3 - mute */ +#define PCM512x_RQMR (1 << 0) #define PCM512x_RQMR_SHIFT 0 +#define PCM512x_RQML (1 << 4) #define PCM512x_RQML_SHIFT 4 /* Page 0, Register 4 - PLL */ -- 2.11.0

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[alsa-devel] [PATCH 0/3] Add Xilinx I2S driver
by Maruthi Srinivas Bayyavarapu 17 Dec '18

17 Dec '18

Xilinx I2S IP enables I2S audio. Each IP can either work in playback or capture mode exclusively. The driver supports both by reading the configuration in device tree. Patchset includes devicetree bindings documentation, driver and build enablement. Maruthi Srinivas Bayyavarapu (3): dt-bindings: ASoC: xlnx,i2s: Document i2s bindings ASoC: xlnx: Add i2s driver ASoC: xlnx: enable i2s driver build .../devicetree/bindings/sound/xlnx,i2s.txt | 28 ++++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/xilinx/Kconfig | 8 + sound/soc/xilinx/Makefile | 2 + sound/soc/xilinx/xlnx_i2s.c | 185 +++++++++++++++++++++ 6 files changed, 225 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/xlnx,i2s.txt create mode 100644 sound/soc/xilinx/Kconfig create mode 100644 sound/soc/xilinx/Makefile create mode 100644 sound/soc/xilinx/xlnx_i2s.c -- 2.7.4

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[alsa-devel] ASoC: About graph node issue without 'reg' property
by Kunihiko Hayashi 17 Dec '18

17 Dec '18

Hello Morimoto-san, Mark, I found that your commit b6f3fc005a2c ("ASoC: simple-card-utils: fixup asoc_simple_card_get_dai_id() counting") replaced with of_graph_parse_endpoint(), that needs 'reg' property in each port that has an endpoint. Currently devicetrees for UniPhier SoCs have also audio graph nodes [1], and these graph nodes include an endpoint for each port. Although the port doesn't have 'reg' property, it has worked fine before. And after applying the commit, it doesn't work. It is written in Documentation/devicetree/bindings/graph.txt that common scheme is to include the 'reg' property. Maybe I think that we'd better add 'reg' properties to each port, however, there might be same matter in other SoCs, so I'm not sure whether your patch breaks the backword compatibility or our devicetrees are wrong. I'd be glad if I could hear something about this issue from you. Thank you, [1] https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/arc… --- Best Regards, Kunihiko Hayashi

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[alsa-devel] [PATCH v2] ALSA: hda: add mute LED support for HP EliteBook 840 G4
by Mantas Mikulėnas 16 Dec '18

16 Dec '18

Tested with 4.19.9. v2: Changed from CXT_FIXUP_MUTE_LED_GPIO to CXT_FIXUP_HP_DOCK because that's what the existing fixups for EliteBooks use. Signed-off-by: Mantas Mikulėnas <grawity(a)gmail.com> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 950e02e71766..51cc6589443f 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -923,6 +923,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x103c, 0x8079, "HP EliteBook 840 G3", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x807C, "HP EliteBook 820 G3", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x80FD, "HP ProBook 640 G2", CXT_FIXUP_HP_DOCK), + SND_PCI_QUIRK(0x103c, 0x828c, "HP EliteBook 840 G4", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x83b3, "HP EliteBook 830 G5", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x83d3, "HP ProBook 640 G4", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE), -- 2.20.1

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