August 2022 - Alsa-devel - mailman.alsa-project.org

Add ASUS ROG Zenith II Extreme configuration
by GitHub pull_request - opened 18 Aug '22

18 Aug '22

alsa-project/alsa-lib pull request #263 was opened from nabajour: This points iec958_device to the correct PCM device. complements this [ALSA kernel patch](https://bugzilla.kernel.org/show_bug.cgi?id=211005). Request URL : https://github.com/alsa-project/alsa-lib/pull/263 Patch URL : https://github.com/alsa-project/alsa-lib/pull/263.patch Repository URL: https://github.com/alsa-project/alsa-lib

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[bug report] ASoC: amd: Add common framework to support I2S on ACP SOC
by Dan Carpenter 18 Aug '22

18 Aug '22

Hello Ajit Kumar Pandey, The patch 623621a9f9e1: "ASoC: amd: Add common framework to support I2S on ACP SOC" from Oct 19, 2021, leads to the following Smatch static checker warning: sound/soc/amd/acp/acp-platform.c:199 acp_dma_open() warn: '&stream->list' not removed from list sound/soc/amd/acp/acp-platform.c 173 static int acp_dma_open(struct snd_soc_component *component, struct snd_pcm_substream *substream) 174 { 175 struct snd_pcm_runtime *runtime = substream->runtime; 176 struct device *dev = component->dev; 177 struct acp_dev_data *adata = dev_get_drvdata(dev); 178 struct acp_stream *stream; 179 int ret; 180 181 stream = kzalloc(sizeof(*stream), GFP_KERNEL); 182 if (!stream) 183 return -ENOMEM; 184 185 stream->substream = substream; 186 187 spin_lock_irq(&adata->acp_lock); 188 list_add_tail(&stream->list, &adata->stream_list); Stream added to the list 189 spin_unlock_irq(&adata->acp_lock); 190 191 if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) 192 runtime->hw = acp_pcm_hardware_playback; 193 else 194 runtime->hw = acp_pcm_hardware_capture; 195 196 ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); 197 if (ret < 0) { 198 dev_err(component->dev, "set integer constraint failed\n"); --> 199 kfree(stream); Use after free. 200 return ret; 201 } 202 runtime->private_data = stream; 203 204 writel(1, ACP_EXTERNAL_INTR_ENB(adata)); 205 206 return ret; Could we delay adding the stream to the list until right before this "return 0;"? 207 } regards, dan carpenter

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[PATCH 1/2] ASoC: Intel: Skylake: fix error message of NHLT blob selection
by Icenowy Zheng 18 Aug '22

18 Aug '22

The query to select NHLT blob had been changed to use parameters of the pipeline instead of PCM, however the error message returned when it fails still uses PCM properties. Fix this inconsistency. Fixes: 87b265260046 ("ASoC: Intel: Skylake: Select proper format for NHLT blob") Signed-off-by: Icenowy Zheng <uwu(a)icenowy.me> --- sound/soc/intel/skylake/skl-topology.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 9bdf020a2b64..19994ec8bba1 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -1864,7 +1864,8 @@ static int skl_tplg_be_fill_pipe_params(struct snd_soc_dai *dai, } else { dev_err(dai->dev, "Blob NULL for id:%d type:%d dirn:%d ch:%d, freq:%d, fmt:%d\n", mconfig->vbus_id, link_type, params->stream, - params->ch, params->s_freq, params->s_fmt); + pipe_fmt->channels, pipe_fmt->freq, + pipe_fmt->bps); return -EINVAL; } -- 2.37.1

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[RESEND PATCH v2 0/6] ALSA: hda: Unify codec construction
by Cezary Rojewski 18 Aug '22

18 Aug '22

A follow up to the recent HDAudio fixes series [1]. Given the recently reported regression [2], before the page fault occurring on codec shutdown can be fixed, codec construction procedure needs to be updated for skylake and sof-intel drivers. Drivers: pci-hda and avs need no changes - already making use of snd_hda_codec_device_init(). As subject touches code used by the sof-driver, additional review has been conducted on thesofproject/linux [3]. Changes in v2: - dropped snd_hda_ext_core <-> snd_hda_codec dependency by calling snd_hda_codec_device_init() directly in skylake and sof drivers probe enumeration routines, as suggested by Takashi - skylake/sof portion of the change has been split into two separate patches - new functions that aim to replace hdac_ext codec init & exit functionality are added first - for skylake and sof drivers both - third patch in the series now combines the "field -> pointer" change for hdac_hda_priv->codec plus the codec-enumeration adjustments for skylake and sof drivers Both above are here to keep git bisect happy, as suggested by Pierre [1]: https://lore.kernel.org/alsa-devel/20220706120230.427296-7-cezary.rojewski@… [2]: https://lore.kernel.org/alsa-devel/3c40df55-3aee-1e08-493b-7b30cd84dc00@lin… [3]: https://github.com/thesofproject/linux/pull/3775 Cezary Rojewski (6): ASoC: Intel: Skylake: Introduce HDA codec init and exit routines ASoC: SOF: Intel: Introduce HDA codec init and exit routines ASoC: Intel: Drop hdac_ext usage for codec device creation ALSA: hda: Always free codec on the device release ALSA: hda: Remove codec init and exit routines ALSA: hda: Fix page fault in snd_hda_codec_shutdown() include/sound/hda_codec.h | 2 - include/sound/hdaudio_ext.h | 3 -- sound/hda/ext/hdac_ext_bus.c | 53 ------------------- sound/pci/hda/hda_codec.c | 49 ++++++++--------- sound/soc/codecs/hdac_hda.c | 26 ++++----- sound/soc/codecs/hdac_hda.h | 2 +- sound/soc/intel/boards/hda_dsp_common.c | 2 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 +- sound/soc/intel/skylake/skl.c | 53 ++++++++++++++----- sound/soc/sof/intel/hda-codec.c | 55 ++++++++++++++------ 10 files changed, 113 insertions(+), 134 deletions(-) -- 2.25.1

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[PATCH] ALSA: hda/realtek: Add quirk for Clevo NS50PU, NS70PU
by Werner Sembach 18 Aug '22

18 Aug '22

From: Christoffer Sandberg <cs(a)tuxedo.de> Fixes headset microphone detection on Clevo NS50PU and NS70PU. Signed-off-by: Christoffer Sandberg <cs(a)tuxedo.de> Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> Cc: <stable(a)vger.kernel.org> --- 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 fd630d62b5a0..fceac608214e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9389,6 +9389,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x70f4, "Clevo NH77EPY", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x70f6, "Clevo NH77DPQ-Y", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x7716, "Clevo NS50PU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x7717, "Clevo NS70PU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x7718, "Clevo L140PU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8228, "Clevo NR40BU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8520, "Clevo NH50D[CD]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), -- 2.34.1

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HDA ATI HDMI not supported
by GitHub issues - edited 18 Aug '22

18 Aug '22

alsa-project/alsa-ucm-conf issue #194 was edited from yangxiaohua2009: I am using the latest ucm2 file and alsalib alsainfo [http://alsa-project.org/db/?f=92c58f3a6b72fee05b678386858dbc816d25dbf3](htt… The `alsaucm reload` command returns error saying: `ALSA lib parser.c:244:(error_node) UCM is not supported for this HDA model (HDA ATI HDMI at 0xfeb60000 irq 28)` the HDMI works but the I2S and HDMI all use fallback ucm config. despite the fact that I already added the acpes8336.conf in conf.d and acpes8336 folder. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/194 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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RE: [PATCH 1/2] ASoC: sma1303: Add driver for Iron Device SMA1303 Amp
by Ki-Seok Jo 18 Aug '22

18 Aug '22

Dear Mark Brown, Thank you for your detailed feedback. I will apply what you told me as soon as possible and request it again. I'll ask you again if there's anything wrong with the correction. I appreciate your quick response. Thank you very much for your time Best Regards, Kiseok Jo Iron Device Corporation Tel.: +82-2-541-2896 Mobile: +82-10-3320-0376 Email: kiseok.jo(a)irondevice.com The information in this email and any attachment is confidential and may be legally protected. It is intended solely for the addressee. Access to this email by anyone else is unauthorised. If you are not the intended recipient, any disclosure or actions taken as a result of the information in this email is prohibited and may be unlawful. If you are not the intended recipient, please notify us immediately and then delete this e-mail and any attachment. Thank you. -----Original Message----- From: Mark Brown <broonie(a)kernel.org> Sent: Wednesday, August 17, 2022 9:09 PM To: Ki-Seok Jo <kiseok.jo(a)irondevice.com> Cc: Gyu-Hwa Park <gyuhwa.park(a)irondevice.com>; alsa-devel(a)alsa-project.org; Suk-Min Kang <sukmin.kang(a)irondevice.com> Subject: Re: [PATCH 1/2] ASoC: sma1303: Add driver for Iron Device SMA1303 Amp On Wed, Aug 17, 2022 at 12:29:37PM +0900, Kiseok Jo wrote: I got part way through reviewing this driver. The top level feedback here is that the driver appears to be exposing everything in the device to userspace as enumeration controls rather than using standard features like DAPM, or appropriate other control types where there's something that's a better fit than an enum. If there's a strong reason for the driver to do something so unusual then theere should be something in here which explains what's going on. > +// SPDX-License-Identifier: GPL-2.0-or-later > +/* sma1303.c -- sma1303 ALSA SoC Audio driver > + * > + * Copyright 2019 Iron Device Corporation 2019? Please make the entire block a C++ one so things look more intentional. > +static int power_up_down_control_put(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) { > + struct snd_soc_component *component = > + snd_soc_kcontrol_component(kcontrol); > + struct sma1303_priv *sma1303 = snd_soc_component_get_drvdata(component); > + int sel = (int)ucontrol->value.integer.value[0]; > + > + if (sel && !(sma1303->force_amp_power_down)) > + sma1303_startup(component); > + else > + sma1303_shutdown(component); > + > + return 0; > +} This and all the other controls need to return 1 on change, please make sure a card with your driver your driver passes mixer-test cleanly - it will spot this and other problems. However I wouldn't expect the power controls to exist at all, generally power is managed via DAPM rather than userspace. > +static int force_sdo_bypass_put(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) { > + struct snd_soc_component *component = > + snd_soc_kcontrol_component(kcontrol); > + struct sma1303_priv *sma1303 = snd_soc_component_get_drvdata(component); > + int sel = (int)ucontrol->value.integer.value[0]; > + > + sma1303->sdo_bypass_flag = (bool)sel; > + > + return 0; > +} What is this bypassing? > +static const char * const sma1303_input_format_text[] = { > + "I2S", "LJ", "Reserved", "Reserved", > + "RJ_16", "RJ_18", "RJ_20", "RJ_24"}; > + > +static const struct soc_enum sma1303_input_format_enum = > +SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(sma1303_input_format_text), > + sma1303_input_format_text); This should be controlled by the machine driver through set_dai_fmt() and hw_params(). > +static const char * const sma1303_pcm_n_slot_text[] = { > + "Slot_1", "Slot_2", "Slot_3", "Slot_4", "Slot_5", "Slot_6", > + "Slot_7", "Slot_8", "Slot_9", "Slot_10", "Slot_11", "Slot_12", > + "Slot_13", "Slot_14", "Slot_15", "Slot_16"}; > + > +static const struct soc_enum sma1303_pcm_n_slot_enum = > +SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(sma1303_pcm_n_slot_text), > + sma1303_pcm_n_slot_text); set_tdm_slot() > +static const char * const sma1303_port_config_text[] = { > + "IN_Port", "Reserved", "OUT_Port", "Reserved"}; This should be DT properties, possibly using the pinctrl schema depending on what's actually going on. > +static const char * const sma1303_port_out_sel_text[] = { > + "Disable", "Format_C", "Mixer_out", "After_DSP", > + "Postscaler", "Reserved", "Reserved", "Reserved"}; > + > +static const struct soc_enum sma1303_port_out_sel_enum = > + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(sma1303_port_out_sel_text), > + sma1303_port_out_sel_text); This looks like it should be a DAPM routing control. > +static const char * const sma1303_spk_off_slope_text[] = { > + "00", "01", "10", "11"}; > + > +static const struct soc_enum sma1303_spk_off_slope_enum = > + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(sma1303_spk_off_slope_text), > + sma1303_spk_off_slope_text); Why make this an enum and not just a numerical control with values 0-3? > +static const char * const sma1303_spkmode_text[] = { > + "Off", "Mono", "Reserved", "Reserved", > + "Stereo", "Reserved", "Reserved", "Reserved"}; > + > +static const struct soc_enum sma1303_spkmode_enum = > + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(sma1303_spkmode_text), > + sma1303_spkmode_text); Use a value enum to eliminate the reserved values from user selection, however if this is controlling if the device has mono or stereo speakers it should be a DT property since that's unlikely to change at runtime. > +static const char * const sma1303_input_gain_text[] = { > + "Gain_0dB", "Gain_M6dB", "Gain_M12dB", "Gain_MInf"}; > + > +static const struct soc_enum sma1303_input_gain_enum = > +SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(sma1303_input_gain_text), > + sma1303_input_gain_text); This should just be a normal volume control. I've stopped reviewing at this point.

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snd_hda_intel: Woke up but no data
by GitHub issues - edited 18 Aug '22

18 Aug '22

alsa-project/alsa-lib issue #262 was edited from shmohawk: ``` Aug 17 01:11:23 hellokitty pulseaudio[12938]: [alsa-sink-CX20590 Analog] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write. Aug 17 01:11:23 hellokitty pulseaudio[12938]: [alsa-sink-CX20590 Analog] alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. Aug 17 01:11:23 hellokitty pulseaudio[12938]: [alsa-sink-CX20590 Analog] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. ``` It is symptomous with a crackling sound for ~5 seconds after any playback starts. alsa-lib 1.2.7.2 alsa-utils 1.2.7 alsa-plugins 1.2.2 Issue URL : https://github.com/alsa-project/alsa-lib/issues/262 Repository URL: https://github.com/alsa-project/alsa-lib

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snd_hda_intel: Woke up but no data
by GitHub issues - opened 17 Aug '22

17 Aug '22

alsa-project/alsa-lib issue #262 was opened from shmohawk: ``` Aug 17 01:11:23 hellokitty pulseaudio[12938]: [alsa-sink-CX20590 Analog] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write. Aug 17 01:11:23 hellokitty pulseaudio[12938]: [alsa-sink-CX20590 Analog] alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. Aug 17 01:11:23 hellokitty pulseaudio[12938]: [alsa-sink-CX20590 Analog] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. ``` It is symptomous with a crackling sound for ~5 seconds after any playback starts. Issue URL : https://github.com/alsa-project/alsa-lib/issues/262 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH] ASoC: fsl_sai: Remove unnecessary FIFO reset in ISR
by Shengjiu Wang 17 Aug '22

17 Aug '22

The FIFO reset drops the words in the FIFO, which may cause channel swap when SAI module is running, especially when the DMA speed is low. So it is not good to do FIFO reset in ISR, then remove the operation. Fixes: e2681a1bf5ae ("ASoC: fsl_sai: Add isr to deal with error flag") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index d430eece1d6b..a7fa6f0bf83d 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -114,11 +114,8 @@ static irqreturn_t fsl_sai_isr(int irq, void *devid) if (flags & FSL_SAI_CSR_SEF) dev_dbg(dev, "isr: Tx Frame sync error detected\n"); - if (flags & FSL_SAI_CSR_FEF) { + if (flags & FSL_SAI_CSR_FEF) dev_dbg(dev, "isr: Transmit underrun detected\n"); - /* FIFO reset for safety */ - xcsr |= FSL_SAI_CSR_FR; - } if (flags & FSL_SAI_CSR_FWF) dev_dbg(dev, "isr: Enabled transmit FIFO is empty\n"); @@ -148,11 +145,8 @@ static irqreturn_t fsl_sai_isr(int irq, void *devid) if (flags & FSL_SAI_CSR_SEF) dev_dbg(dev, "isr: Rx Frame sync error detected\n"); - if (flags & FSL_SAI_CSR_FEF) { + if (flags & FSL_SAI_CSR_FEF) dev_dbg(dev, "isr: Receive overflow detected\n"); - /* FIFO reset for safety */ - xcsr |= FSL_SAI_CSR_FR; - } if (flags & FSL_SAI_CSR_FWF) dev_dbg(dev, "isr: Enabled receive FIFO is full\n"); -- 2.34.1

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