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

[PATCH v6] ASoC: Intel: kbl_rt5663_rt5514_max98927: Fix kabylake_ssp_fixup function
by Harsha Priya 16 Jul '20

16 Jul '20

From: Vamshi Krishna Gopal <vamshi.krishna.gopal(a)intel.com> kabylake_ssp_fixup function uses snd_soc_dpcm to identify the codecs DAIs. The hw parameters are changed based on the codec DAI, the stream is intended for. The earlier approach to get snd_soc_dpcm was using container_of() macro on snd_pcm_hw_params. The structures have been modified over time and snd_soc_dpcm does not have snd_pcm_hw_params as a reference but as a copy. This causes the current driver to crash when used. This patch changes the way snd_soc_dpcm is extracted. The snd_soc_pcm_runtime holds 2 dpcm instances (one for playback and one for capture). The 2 codecs on this SSP are dmic and speakers. One is for capture and one is for playback respectively. Based on the direction of the stream, the snd_soc_dpcm is extracted from the snd_soc_pcm_runtime structure. Tested for all use cases of the driver. Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Signed-off-by: Vamshi Krishna Gopal <vamshi.krishna.gopal(a)intel.com> Tested-by: Lukasz Majczak <lma(a)semihalf.com> --- v1 -> v2: - Extract dmic from SSP0 as every BE should have own fixup function. v2 -> v3: - Restore naming in the dapm route table to not confuse with other drivers - Fixed indentations v3 -> v4: - Updated code and commit description according to solution proposed by Harsha v4 -> v5: - Cosmetic Changes v5 -> v6: - Dmic regression seen with v4 fixed - Using available routines for obtaining dpcm information --- --- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 26 ++++++++++++------- 1 file changed, 17 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 584e4f9cedc2..b261b1c466a8 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -379,22 +379,30 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_interval *chan = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - struct snd_soc_dpcm *dpcm = container_of( - params, struct snd_soc_dpcm, hw_params); - struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link; - struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link; + struct snd_soc_dpcm *dpcm, *rtd_dpcm; + + /* + * This macro will be called for playback stream + */ + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_PLAYBACK, dpcm) + rtd_dpcm = dpcm; + /* + * This macro will be called for capture stream + */ + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_CAPTURE, dpcm) + rtd_dpcm = dpcm; /* * The ADSP will convert the FE rate to 48k, stereo, 24 bit */ - if (!strcmp(fe_dai_link->name, "Kbl Audio Port") || - !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") || - !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) { + if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Port") || + !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Headset Playback") || + !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Capture Port")) { rate->min = rate->max = 48000; chan->min = chan->max = 2; snd_mask_none(fmt); snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); - } else if (!strcmp(fe_dai_link->name, "Kbl Audio DMIC cap")) { + } else if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio DMIC cap")) { if (params_channels(params) == 2 || DMIC_CH(dmic_constraints) == 2) chan->min = chan->max = 2; @@ -405,7 +413,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, * The speaker on the SSP0 supports S16_LE and not S24_LE. * thus changing the mask here */ - if (!strcmp(be_dai_link->name, "SSP0-Codec")) + if (!strcmp(rtd_dpcm->be->dai_link->name, "SSP0-Codec")) snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); return 0; -- 2.17.1

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[PATCH v3] ASoC: Intel: kbl_rt5663_rt5514_max98927: Split be_hw_params_fixup function
by Lukasz Majczak 16 Jul '20

16 Jul '20

Split be_hw_params_fixup function for different codecs as current common function, leads to crash while trying to get snd_soc_dpcm with container_of() macro in kabylake_ssp_fixup(). The crash call path looks as below: soc_pcm_hw_params() snd_soc_dai_hw_params(codec_dai, substream, &codec_params); rtd->dai_link->be_hw_params_fixup(rtd, params) kabylake_ssp_fixup() In this case, codec_params is just a copy of an internal structure and is not embedded into struct snd_soc_dpcm thus we cannot use container_of() on it. v1 -> v2: - Extract dmic from SSP0 as every BE should have own fixup function. v2 -> v3: - Restore naming in the dapm route table to not confuse with other drivers - Fixed indentations Signed-off-by: Lukasz Majczak <lma(a)semihalf.com> --- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 130 +++++++++++------- 1 file changed, 84 insertions(+), 46 deletions(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 1b1f8d7a4ea3..b9596cf3a6c4 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -328,42 +328,51 @@ static const struct snd_soc_ops kabylake_rt5663_fe_ops = { .startup = kbl_fe_startup, }; -static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) +static void kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params, snd_pcm_format_t pcm_fmt) { struct snd_interval *rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *chan = hw_param_interval(params, + struct snd_interval *channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - struct snd_soc_dpcm *dpcm = container_of( - params, struct snd_soc_dpcm, hw_params); - struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link; - struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link; /* * The ADSP will convert the FE rate to 48k, stereo, 24 bit */ - if (!strcmp(fe_dai_link->name, "Kbl Audio Port") || - !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") || - !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) { - rate->min = rate->max = 48000; - chan->min = chan->max = 2; - snd_mask_none(fmt); - snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); - } else if (!strcmp(fe_dai_link->name, "Kbl Audio DMIC cap")) { - if (params_channels(params) == 2 || - DMIC_CH(dmic_constraints) == 2) - chan->min = chan->max = 2; - else - chan->min = chan->max = 4; - } - /* - * The speaker on the SSP0 supports S16_LE and not S24_LE. - * thus changing the mask here - */ - if (!strcmp(be_dai_link->name, "SSP0-Codec")) - snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); + + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + snd_mask_none(fmt); + snd_mask_set_format(fmt, pcm_fmt); +} + +static int kabylake_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + kabylake_ssp_fixup(rtd, params, SNDRV_PCM_FORMAT_S16_LE); + return 0; +} + +static int kabylake_ssp1_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + + kabylake_ssp_fixup(rtd, params, SNDRV_PCM_FORMAT_S24_LE); + return 0; +} + +static int kabylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + if (params_channels(params) == 2 || DMIC_CH(dmic_constraints) == 2) + channels->min = channels->max = 2; + else + channels->min = channels->max = 4; return 0; } @@ -400,20 +409,6 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, int ret = 0, j; for_each_rtd_codec_dais(rtd, j, codec_dai) { - if (!strcmp(codec_dai->component->name, RT5514_DEV_NAME)) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0, 8, 16); - if (ret < 0) { - dev_err(rtd->dev, "set TDM slot err:%d\n", ret); - return ret; - } - - ret = snd_soc_dai_set_sysclk(codec_dai, - RT5514_SCLK_S_MCLK, 24576000, SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(rtd->dev, "set sysclk err: %d\n", ret); - return ret; - } - } if (!strcmp(codec_dai->component->name, MAXIM_DEV0_NAME)) { ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x30, 3, 8, 16); if (ret < 0) { @@ -433,10 +428,36 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, return ret; } +static int kabylake_dmic01_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int ret = 0; + + ret = snd_soc_dai_set_tdm_slot(rtd->codec_dai, 0xF, 0, 8, 16); + if (ret < 0) { + dev_err(rtd->dev, "set TDM slot err:%d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(rtd->codec_dai, + RT5514_SCLK_S_MCLK, 24576000, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "set sysclk err: %d\n", ret); + return ret; + } + + return ret; +} + static struct snd_soc_ops kabylake_ssp0_ops = { .hw_params = kabylake_ssp0_hw_params, }; +static struct snd_soc_ops kabylake_dmic01_ops = { + .hw_params = kabylake_dmic01_hw_params, +}; + static const unsigned int channels_dmic[] = { 4, }; @@ -507,14 +528,19 @@ SND_SOC_DAILINK_DEF(ssp0_pin, SND_SOC_DAILINK_DEF(ssp0_codec, DAILINK_COMP_ARRAY( /* Left */ COMP_CODEC(MAXIM_DEV0_NAME, KBL_MAXIM_CODEC_DAI), - /* Right */COMP_CODEC(MAXIM_DEV1_NAME, KBL_MAXIM_CODEC_DAI), - /* dmic */ COMP_CODEC(RT5514_DEV_NAME, KBL_REALTEK_DMIC_CODEC_DAI))); + /* Right */COMP_CODEC(MAXIM_DEV1_NAME, KBL_MAXIM_CODEC_DAI))); SND_SOC_DAILINK_DEF(ssp1_pin, DAILINK_COMP_ARRAY(COMP_CPU("SSP1 Pin"))); SND_SOC_DAILINK_DEF(ssp1_codec, DAILINK_COMP_ARRAY(COMP_CODEC(RT5663_DEV_NAME, KBL_REALTEK_CODEC_DAI))); +SND_SOC_DAILINK_DEF(dmic01_pin, + DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin"))); +SND_SOC_DAILINK_DEF(dmic01_codec, + DAILINK_COMP_ARRAY( + COMP_CODEC(RT5514_DEV_NAME, KBL_REALTEK_DMIC_CODEC_DAI))); + SND_SOC_DAILINK_DEF(idisp1_pin, DAILINK_COMP_ARRAY(COMP_CPU("iDisp1 Pin"))); SND_SOC_DAILINK_DEF(idisp1_codec, @@ -618,9 +644,8 @@ static struct snd_soc_dai_link kabylake_dais[] = { SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .ignore_pmdown_time = 1, - .be_hw_params_fixup = kabylake_ssp_fixup, + .be_hw_params_fixup = kabylake_ssp0_fixup, .dpcm_playback = 1, - .dpcm_capture = 1, .ops = &kabylake_ssp0_ops, SND_SOC_DAILINK_REG(ssp0_pin, ssp0_codec, platform), }, @@ -632,12 +657,25 @@ static struct snd_soc_dai_link kabylake_dais[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .ignore_pmdown_time = 1, - .be_hw_params_fixup = kabylake_ssp_fixup, + .be_hw_params_fixup = kabylake_ssp1_fixup, .ops = &kabylake_rt5663_ops, .dpcm_playback = 1, .dpcm_capture = 1, SND_SOC_DAILINK_REG(ssp1_pin, ssp1_codec, platform), }, + { + .name = "dmic01", + .id = 2, + .no_pcm = 1, + .dai_fmt = SND_SOC_DAIFMT_DSP_B | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = kabylake_dmic_fixup, + .dpcm_capture = 1, + .ops = &kabylake_dmic01_ops, + SND_SOC_DAILINK_REG(dmic01_pin, dmic01_codec, platform), + }, { .name = "iDisp1", .id = 3, base-commit: a4f6fc98cd2fa1774bcaeb248c67156ef9402a56 -- 2.25.1

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[PATCH] topology: hda-dsp: Change processing domain of srcintc module to LL
by Mateusz Gorski 16 Jul '20

16 Jul '20

In order to make this topology universal for all FW branches, change the processing domain of srcintc module from DP to LL. Signed-off-by: Mateusz Gorski <mateusz.gorski(a)linux.intel.com> --- topology/hda-dsp/skl_hda_dsp_generic-tplg.conf | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/topology/hda-dsp/skl_hda_dsp_generic-tplg.conf b/topology/hda-dsp/skl_hda_dsp_generic-tplg.conf index 32ba9a7..ed2ea80 100644 --- a/topology/hda-dsp/skl_hda_dsp_generic-tplg.conf +++ b/topology/hda-dsp/skl_hda_dsp_generic-tplg.conf @@ -1684,7 +1684,7 @@ SectionVendorTuples."media0_in srcintc 0" { SKL_TKN_U32_PIPE_PRIORITY "0" SKL_TKN_U32_PMODE "0" SKL_TKN_U32_D0I3_CAPS "1" - SKL_TKN_U32_PROC_DOMAIN "1" + SKL_TKN_U32_PROC_DOMAIN "0" SKL_TKN_U32_PIPE_DIRECTION "0" SKL_TKN_U32_NUM_CONFIGS "1" SKL_TKN_U32_DMA_BUF_SIZE "2" -- 2.17.1

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[PATCH v2 0/5] crypto: add sha256() function
by Eric Biggers 16 Jul '20

16 Jul '20

This series adds a function sha256() to the sha256 library so that users who want to compute a hash in one step can just call sha256() instead of sha256_init() + sha256_update() + sha256_final(). Patches 3-5 then convert some users to use it. Changed v1 => v2: - Added sparc patch to fix a build breakage caused by a static variable already named "sha256". - Added Reviewed-by, Acked-by, and Tested-by tags. Eric Biggers (5): crypto: sparc - rename sha256 to sha256_alg crypto: lib/sha256 - add sha256() function efi: use sha256() instead of open coding mptcp: use sha256() instead of open coding ASoC: cros_ec_codec: use sha256() instead of open coding arch/sparc/crypto/sha256_glue.c | 14 ++++++------ drivers/firmware/efi/embedded-firmware.c | 9 +++----- include/crypto/sha.h | 1 + lib/crypto/sha256.c | 10 +++++++++ net/mptcp/crypto.c | 15 +++---------- sound/soc/codecs/cros_ec_codec.c | 27 ++---------------------- 6 files changed, 26 insertions(+), 50 deletions(-) base-commit: 57c8aa43b9f272c382c253573c82be5cb68fe22d -- 2.27.0

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[PATCH] ALSA: hda/realtek - fixup for yet another Intel reference board
by Takashi Iwai 16 Jul '20

16 Jul '20

From: PeiSen Hou <pshou(a)realtek.com.tw> Add headset_jack for the intel reference board support with 10ec:1230. Signed-off-by: PeiSen Hou <pshou(a)realtek.com.tw> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- 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 45df33d1bd1a..1b06c4261248 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7581,6 +7581,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x10cf, 0x1629, "Lifebook U7x7", ALC255_FIXUP_LIFEBOOK_U7x7_HEADSET_MIC), SND_PCI_QUIRK(0x10cf, 0x1845, "Lifebook U904", ALC269_FIXUP_LIFEBOOK_EXTMIC), SND_PCI_QUIRK(0x10ec, 0x10f2, "Intel Reference board", ALC700_FIXUP_INTEL_REFERENCE), + SND_PCI_QUIRK(0x10ec, 0x1230, "Intel Reference board", ALC225_FIXUP_HEADSET_JACK), SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-SZ6", ALC269_FIXUP_HEADSET_MODE), SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), -- 2.16.4

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[PATCH] sound/pci/bt87x: Replaces 'blacklist' with 'no_audio_list'
by Paul Schulz 16 Jul '20

16 Jul '20

This patch removes 'blacklist' and replaces it with the more descriptive 'no_audio_list' This is a functionally trivial patch and has no other effect. Signed-off-by: Paul Schulz <paul(a)mawsonlakes.org> --- sound/pci/bt87x.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index 6567504665b9..b74b55bdb146 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -801,7 +801,7 @@ MODULE_DEVICE_TABLE(pci, snd_bt87x_ids); * (DVB cards use the audio function to transfer MPEG data) */ static struct { unsigned short subvendor, subdevice; -} blacklist[] = { +} no_audio_list[] = { {0x0071, 0x0101}, /* Nebula Electronics DigiTV */ {0x11bd, 0x001c}, /* Pinnacle PCTV Sat */ {0x11bd, 0x0026}, /* Pinnacle PCTV SAT CI */ @@ -817,7 +817,7 @@ static struct { static struct pci_driver driver; -/* return the id of the card, or a negative value if it's blacklisted */ +/* return the id of the card, or a negative value if it's in the no_audio_list */ static int snd_bt87x_detect_card(struct pci_dev *pci) { int i; @@ -827,9 +827,9 @@ static int snd_bt87x_detect_card(struct pci_dev *pci) if (supported && supported->driver_data > 0) return supported->driver_data; - for (i = 0; i < ARRAY_SIZE(blacklist); ++i) - if (blacklist[i].subvendor == pci->subsystem_vendor && - blacklist[i].subdevice == pci->subsystem_device) { + for (i = 0; i < ARRAY_SIZE(no_audio_list); ++i) + if (no_audio_list[i].subvendor == pci->subsystem_vendor && + no_audio_list[i].subdevice == pci->subsystem_device) { dev_dbg(&pci->dev, "card %#04x-%#04x:%#04x has no audio\n", pci->device, pci->subsystem_vendor, pci->subsystem_device); -- 2.25.1

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ASUS UX563 speaker no ouput
by Kailang 16 Jul '20

16 Jul '20

Hi Takashi, ASUS UX563 speaker also no sound. Add quirk to model ALC294_FIXUP_ASUS_HPE will solve it. BR, Kailang > -----Original Message----- > From: Takashi Iwai <tiwai(a)suse.de> > Sent: Thursday, July 16, 2020 3:19 PM > To: Kailang <kailang(a)realtek.com> > Cc: (alsa-devel(a)alsa-project.org) <alsa-devel(a)alsa-project.org> > Subject: Re: ASUS UX533 and UX534 speaker no ouput > > On Thu, 16 Jul 2020 09:12:26 +0200, > Kailang wrote: > > > > Hi Takashi, > > > > Attach patch will solve speaker no output issue for ASUS platform. > > Many Thanks. > > Pulled now. Thanks. > > > Takashi > > ------Please consider the environment before printing this e-mail.

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ASUS UX533 and UX534 speaker no ouput
by Kailang 16 Jul '20

16 Jul '20

Hi Takashi, Attach patch will solve speaker no output issue for ASUS platform. Many Thanks. BR, Kailang

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[PATCH] ASoC: soc-component: Add missed return for snd_soc_pcm_component_mmap
by Shengjiu Wang 16 Jul '20

16 Jul '20

Add missed return for snd_soc_pcm_component_mmap, otherwise it always return -EINVAL. Fixes: e2329eeba45f ("ASoC: soc-component: add soc_component_err()") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/soc-component.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c index af9909c5492f..cde7b9c609bb 100644 --- a/sound/soc/soc-component.c +++ b/sound/soc/soc-component.c @@ -705,7 +705,7 @@ int snd_soc_pcm_component_mmap(struct snd_pcm_substream *substream, /* FIXME. it returns 1st mmap now */ for_each_rtd_components(rtd, i, component) if (component->driver->mmap) - soc_component_ret( + return soc_component_ret( component, component->driver->mmap(component, substream, vma)); -- 2.27.0

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[PATCH] soundwire: fix the kernel-doc comment
by Vinod Koul 16 Jul '20

16 Jul '20

sdw_startup_stream() and sdw_shutdown_stream() argument has been updated but not the comments, so update these as well to fix warning with W=1 drivers/soundwire/stream.c:1859: warning: Function parameter or member 'sdw_substream' not described in 'sdw_startup_stream' drivers/soundwire/stream.c:1859: warning: Excess function parameter 'stream' description in 'sdw_startup_stream' drivers/soundwire/stream.c:1903: warning: Function parameter or member 'sdw_substream' not described in 'sdw_shutdown_stream' drivers/soundwire/stream.c:1903: warning: Excess function parameter 'stream' description in 'sdw_shutdown_stream' Signed-off-by: Vinod Koul <vkoul(a)kernel.org> --- drivers/soundwire/stream.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index 6bc2ff29c202..37290a799023 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -1851,7 +1851,7 @@ static int set_stream(struct snd_pcm_substream *substream, /** * sdw_startup_stream() - Startup SoundWire stream * - * @stream: Soundwire stream + * @sdw_substream: Soundwire stream * * Documentation/driver-api/soundwire/stream.rst explains this API in detail */ @@ -1895,7 +1895,7 @@ EXPORT_SYMBOL(sdw_startup_stream); /** * sdw_shutdown_stream() - Shutdown SoundWire stream * - * @stream: Soundwire stream + * @sdw_substream: Soundwire stream * * Documentation/driver-api/soundwire/stream.rst explains this API in detail */ -- 2.26.2

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