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[PATCH AUTOSEL 5.19 03/22] ASoC: nau8824: Fix semaphore unbalance at error paths
by Sasha Levin 14 Sep '22

14 Sep '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 5628560e90395d3812800a8e44a01c32ffa429ec ] The semaphore of nau8824 wasn't properly unlocked at some error handling code paths, hence this may result in the unbalance (and potential lock-up). Fix them to handle the semaphore up properly. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Link: https://lore.kernel.org/r/20220823081000.2965-3-tiwai@suse.de Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/nau8824.c | 17 ++++++++++------- 1 file changed, 10 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 2a7c935085353..c263858745ee2 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -1043,6 +1043,7 @@ static int nau8824_hw_params(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct nau8824 *nau8824 = snd_soc_component_get_drvdata(component); unsigned int val_len = 0, osr, ctrl_val, bclk_fs, bclk_div; + int err = -EINVAL; nau8824_sema_acquire(nau8824, HZ); @@ -1059,7 +1060,7 @@ static int nau8824_hw_params(struct snd_pcm_substream *substream, osr &= NAU8824_DAC_OVERSAMPLE_MASK; if (nau8824_clock_check(nau8824, substream->stream, nau8824->fs, osr)) - return -EINVAL; + goto error; regmap_update_bits(nau8824->regmap, NAU8824_REG_CLK_DIVIDER, NAU8824_CLK_DAC_SRC_MASK, osr_dac_sel[osr].clk_src << NAU8824_CLK_DAC_SRC_SFT); @@ -1069,7 +1070,7 @@ static int nau8824_hw_params(struct snd_pcm_substream *substream, osr &= NAU8824_ADC_SYNC_DOWN_MASK; if (nau8824_clock_check(nau8824, substream->stream, nau8824->fs, osr)) - return -EINVAL; + goto error; regmap_update_bits(nau8824->regmap, NAU8824_REG_CLK_DIVIDER, NAU8824_CLK_ADC_SRC_MASK, osr_adc_sel[osr].clk_src << NAU8824_CLK_ADC_SRC_SFT); @@ -1090,7 +1091,7 @@ static int nau8824_hw_params(struct snd_pcm_substream *substream, else if (bclk_fs <= 256) bclk_div = 0; else - return -EINVAL; + goto error; regmap_update_bits(nau8824->regmap, NAU8824_REG_PORT0_I2S_PCM_CTRL_2, NAU8824_I2S_LRC_DIV_MASK | NAU8824_I2S_BLK_DIV_MASK, @@ -1111,15 +1112,17 @@ static int nau8824_hw_params(struct snd_pcm_substream *substream, val_len |= NAU8824_I2S_DL_32; break; default: - return -EINVAL; + goto error; } regmap_update_bits(nau8824->regmap, NAU8824_REG_PORT0_I2S_PCM_CTRL_1, NAU8824_I2S_DL_MASK, val_len); + err = 0; + error: nau8824_sema_release(nau8824); - return 0; + return err; } static int nau8824_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) @@ -1128,8 +1131,6 @@ static int nau8824_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) struct nau8824 *nau8824 = snd_soc_component_get_drvdata(component); unsigned int ctrl1_val = 0, ctrl2_val = 0; - nau8824_sema_acquire(nau8824, HZ); - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: ctrl2_val |= NAU8824_I2S_MS_MASTER; @@ -1171,6 +1172,8 @@ static int nau8824_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } + nau8824_sema_acquire(nau8824, HZ); + regmap_update_bits(nau8824->regmap, NAU8824_REG_PORT0_I2S_PCM_CTRL_1, NAU8824_I2S_DF_MASK | NAU8824_I2S_BP_MASK | NAU8824_I2S_PCMB_EN, ctrl1_val); -- 2.35.1

1 0

[PATCH linux-next] soundwire: intel: Remove unused parameter ret
by cgel.zte＠gmail.com 14 Sep '22

14 Sep '22

From: Jinpeng Cui <cui.jinpeng2(a)zte.com.cn> The parameter ret = 0 has not been used. Return 0 directly instead of returning ret. Reported-by: Zeal Robot <zealci(a)zte.com.cn> Signed-off-by: Jinpeng Cui <cui.jinpeng2(a)zte.com.cn> --- drivers/soundwire/intel.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index b006ae6efcb0..26fdf3577454 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -395,7 +395,6 @@ static int intel_shim_init(struct sdw_intel *sdw, bool clock_stop) { void __iomem *shim = sdw->link_res->shim; unsigned int link_id = sdw->instance; - int ret = 0; u16 ioctl = 0, act = 0; mutex_lock(sdw->link_res->shim_lock); @@ -427,7 +426,7 @@ static int intel_shim_init(struct sdw_intel *sdw, bool clock_stop) mutex_unlock(sdw->link_res->shim_lock); - return ret; + return 0; } static void intel_shim_wake(struct sdw_intel *sdw, bool wake_enable) -- 2.25.1

2 1

[PATCH] ALSA: patch_realtek: Fix Dell Inspiron Plus 16
by Philipp Jungkamp 14 Sep '22

14 Sep '22

The Dell Inspiron Plus 16 has top speakers connected on NID 0x17, which the codec reports as unconnected. These speakers should be connected to the DAC on NID 0x03. --- I don't possess this Laptop myself. The owner of this Laptop reports that the speakers are now active, though there are still problem's. > There are still 2 small issues present: > 1. 0,5s delay before the top speakers turn on > 2. Popping sound from headphones 5s after the music stops playing > and afterwards when it starts playing again. This can be fixed by > disabling power save on the audio driver [...]. I've only written this patch because the issue seemed similar to that on my Lenovo Yoga 9 14IAP7. sound/pci/hda/patch_realtek.c | 34 ++++++++++++++++++++++++++++++++++ 1 file changed, 34 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 38930cf5aace..e6f229e418c3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6896,6 +6896,32 @@ static void alc287_fixup_yoga9_14iap7_bass_spk_pin(struct hda_codec *codec, } } +static void alc295_fixup_dell_inspiron_top_speakers(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + static const struct hda_pintbl pincfgs[] = { + { 0x14, 0x90170151 }, + { 0x17, 0x90170150 }, + { } + }; + static const hda_nid_t conn[] = { 0x02, 0x03 }; + static const hda_nid_t preferred_pairs[] = { + 0x14, 0x02, + 0x17, 0x03, + 0x21, 0x02, + 0 + }; + struct alc_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_apply_pincfgs(codec, pincfgs); + snd_hda_override_conn_list(codec, 0x17, ARRAY_SIZE(conn), conn); + spec->gen.preferred_dacs = preferred_pairs; + break; + } +} + enum { ALC269_FIXUP_GPIO2, ALC269_FIXUP_SONY_VAIO, @@ -7137,6 +7163,7 @@ enum { ALC287_FIXUP_LEGION_16ITHG6, ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK, ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN, + ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -9066,6 +9093,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK, }, + [ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc295_fixup_dell_inspiron_top_speakers, + .chained = true, + .chain_id = ALC269_FIXUP_DELL4_MIC_NO_PRESENCE, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -9165,6 +9198,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0a9d, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0a9e, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0b19, "Dell XPS 15 9520", ALC289_FIXUP_DUAL_SPK), + SND_PCI_QUIRK(0x1028, 0x0b71, "Dell Inspiron 16 Plus 7620", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- 2.37.3

2 1

[PATCH] ALSA: hda/realtek: Enable 4-speaker output Dell Precision 5560 laptop
by acheronfail 14 Sep '22

14 Sep '22

This is my first patch, so let me know if I need to do anything differently. :) The Dell Precision 5570 uses the same 4-speakers-on-ALC289 just like the previous Precision 5560. I replicated that patch onto this one, and can confirm that the audio is much better (the woofers are now working); I've tested it on my Dell Precision 5570. --- 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 799f6bf266dd..b0ad4856b877 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9130,6 +9130,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0a9d, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0a9e, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0b19, "Dell XPS 15 9520", ALC289_FIXUP_DUAL_SPK), + SND_PCI_QUIRK(0x1028, 0x0b1a, "Dell Precision 5570", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- 2.37.3

2 1

RE: [PATCH 1/2] ASoC: sma1303: Add driver for Iron Device SMA1303 Amp
by Ki-Seok Jo 14 Sep '22

14 Sep '22

Dear Mark Brown, Thanks for your detailed feedback. I'm so late in replying. COVID-19 is going around again in Korea. I also suffered from COVID-19 this time. Please take care of yourself from COVID-19. Anyway, I proceeded with the update again by referring to your review. And I added my opinion to the following. Please check the below. If there is anything wrong, please give me a feedback again. Thank you for your patience and cooperation. Best regards, Ki Seok 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. -----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. # To solve the any issue by changing the value whenever there is a problem during use, we registered to control all functions. # Since the verification was completed by mass production with the corresponding setting, we removed the controls in the userspace except for a few things as you mentioned. > +// SPDX-License-Identifier: GPL-2.0-or-later > +/* sma1303.c -- sma1303 ALSA SoC Audio driver > + * > + * Copyright 2019 Iron Device Corporation 2019? # I missed this point. Thank you! # Changed 2019 to 2022. Please make the entire block a C++ one so things look more intentional. # I have modified the code a bit. I'm sorry, but I couldn't figure out exactly what you were talking about. # If there is anything that needs to be corrected, please reply again. > +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. # Added the return value for checking error. # It's ok if the value is '0', and if a problem occurs, it returns to a value of less than 0. # I have one question. All return values have to match the Linux standard error code? However I wouldn't expect the power controls to exist at all, generally power is managed via DAPM rather than userspace. # Removed the mixer control to control power from userspace. # Automatically power control as DAC Event(DAPM), bias, set hw parameter and so on.. > +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? # In chip, this is a function to set not to output on I2S. # The output will always be on. # So, We removed this. > +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(). # It's already included in that. # This mixer control is an additional control for debugging. # So, We removed this. > +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() # It's already included in that. # This mixer control is also an additional control for debugging. # So, We removed this. > +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. # I'm sorry. I think I named it wrong. # This control is a feature that sets whether output is enabled or not. Output means SDO, that is, data exported from the chip. # So I modified this. > +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. # That's right. I added the control in route. > +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? # Yes. I missed this value. # This function has been removed as users do not need to control it > +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. # This function also includes PWM on/off. # We have changed the enum. > +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. # We have other control for normal volume. # So, this control has been removed. I've stopped reviewing at this point. # Thanks for the detailed feedback.

1 0

[PATCH v1] ASoC: rockchip: i2s: use regmap_read_poll_timeout to poll I2S_CLR
by Judy Hsiao 14 Sep '22

14 Sep '22

Use regmap_read_poll_timeout to poll I2S_CLR. It also fixes the 'rockchip-i2s ff070000.i2s; fail to clear' and return -EBUSY when the read of I2S_CLR exceeds the retry limit. Fixes: 0ff9f8b9f592 ("ASoC: rockchip: i2s: Fix error code when fail to read I2S_CLR") Signed-off-by: Judy Hsiao <judyhsiao(a)chromium.org> --- sound/soc/rockchip/rockchip_i2s.c | 39 +++++++++++++------------------ 1 file changed, 16 insertions(+), 23 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index f5f3540a9e18..02905eec615e 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -163,18 +163,14 @@ static int rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) I2S_CLR_TXC | I2S_CLR_RXC); if (ret < 0) goto end; - regmap_read(i2s->regmap, I2S_CLR, &val); - - /* Should wait for clear operation to finish */ - while (val) { - regmap_read(i2s->regmap, I2S_CLR, &val); - retry--; - if (!retry) { - dev_warn(i2s->dev, "fail to clear\n"); - ret = -EBUSY; - break; - } - } + ret = regmap_read_poll_timeout(i2s->regmap, + I2S_CLR, + val, + val != 0, + 20, + 200); + if (ret < 0) + dev_warn(i2s->dev, "fail to clear\n"); } } end: @@ -226,17 +222,14 @@ static int rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) I2S_CLR_TXC | I2S_CLR_RXC); if (ret < 0) goto end; - regmap_read(i2s->regmap, I2S_CLR, &val); - /* Should wait for clear operation to finish */ - while (val) { - regmap_read(i2s->regmap, I2S_CLR, &val); - retry--; - if (!retry) { - dev_warn(i2s->dev, "fail to clear\n"); - ret = -EBUSY; - break; - } - } + ret = regmap_read_poll_timeout(i2s->regmap, + I2S_CLR, + val, + val != 0, + 20, + 200); + if (ret < 0) + dev_warn(i2s->dev, "fail to clear\n"); } } end: -- 2.37.2.789.g6183377224-goog

2 1

[PATCH] ASoC: meson: aiu-fifo.c: use devm_kzalloc(), and remove .remove function
by Kuninori Morimoto 14 Sep '22

14 Sep '22

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current aiu-fifo.c is using kzalloc()/kfree(), but we can replace it by devm_kzalloc(), and remove kfree(). This patch do it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/meson/aiu-fifo.c | 10 +--------- sound/soc/meson/aiu-fifo.h | 1 - sound/soc/meson/aiu.c | 2 -- 3 files changed, 1 insertion(+), 12 deletions(-) diff --git a/sound/soc/meson/aiu-fifo.c b/sound/soc/meson/aiu-fifo.c index d67ff4cdabd5..843e5067e07d 100644 --- a/sound/soc/meson/aiu-fifo.c +++ b/sound/soc/meson/aiu-fifo.c @@ -196,7 +196,7 @@ int aiu_fifo_dai_probe(struct snd_soc_dai *dai) { struct aiu_fifo *fifo; - fifo = kzalloc(sizeof(*fifo), GFP_KERNEL); + fifo = devm_kzalloc(dai->dev, sizeof(*fifo), GFP_KERNEL); if (!fifo) return -ENOMEM; @@ -204,11 +204,3 @@ int aiu_fifo_dai_probe(struct snd_soc_dai *dai) return 0; } - -int aiu_fifo_dai_remove(struct snd_soc_dai *dai) -{ - kfree(dai->playback_dma_data); - - return 0; -} - diff --git a/sound/soc/meson/aiu-fifo.h b/sound/soc/meson/aiu-fifo.h index 42ce266677cc..fb323a4385f7 100644 --- a/sound/soc/meson/aiu-fifo.h +++ b/sound/soc/meson/aiu-fifo.h @@ -26,7 +26,6 @@ struct aiu_fifo { }; int aiu_fifo_dai_probe(struct snd_soc_dai *dai); -int aiu_fifo_dai_remove(struct snd_soc_dai *dai); snd_pcm_uframes_t aiu_fifo_pointer(struct snd_soc_component *component, struct snd_pcm_substream *substream); diff --git a/sound/soc/meson/aiu.c b/sound/soc/meson/aiu.c index 88e611e64d14..7e632aa31368 100644 --- a/sound/soc/meson/aiu.c +++ b/sound/soc/meson/aiu.c @@ -123,7 +123,6 @@ static struct snd_soc_dai_driver aiu_cpu_dai_drv[] = { .ops = &aiu_fifo_i2s_dai_ops, .pcm_new = aiu_fifo_pcm_new, .probe = aiu_fifo_i2s_dai_probe, - .remove = aiu_fifo_dai_remove, }, [CPU_SPDIF_FIFO] = { .name = "SPDIF FIFO", @@ -139,7 +138,6 @@ static struct snd_soc_dai_driver aiu_cpu_dai_drv[] = { .ops = &aiu_fifo_spdif_dai_ops, .pcm_new = aiu_fifo_pcm_new, .probe = aiu_fifo_spdif_dai_probe, - .remove = aiu_fifo_dai_remove, }, [CPU_I2S_ENCODER] = { .name = "I2S Encoder", -- 2.25.1

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[linux-next:master] BUILD REGRESSION 0caac1da994900d12a9be6106edb8e98696712a3
by kernel test robot 14 Sep '22

14 Sep '22

tree/branch: https://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git master branch HEAD: 0caac1da994900d12a9be6106edb8e98696712a3 Add linux-next specific files for 20220913 Error/Warning reports: https://lore.kernel.org/linux-mm/202209042337.FQi69rLV-lkp@intel.com https://lore.kernel.org/linux-mm/202209060229.dVuyxjBv-lkp@intel.com https://lore.kernel.org/linux-mm/202209080718.y5QmlNKH-lkp@intel.com Error/Warning: (recently discovered and may have been fixed) drivers/gpu/drm/amd/amdgpu/imu_v11_0_3.c:139:6: warning: no previous prototype for 'imu_v11_0_3_program_rlc_ram' [-Wmissing-prototypes] drivers/gpu/drm/drm_atomic_helper.c:802: warning: expecting prototype for drm_atomic_helper_check_wb_connector_state(). Prototype was for drm_atomic_helper_check_wb_encoder_state() instead drivers/hwmon/emc2305.c:194 emc2305_set_cur_state() warn: impossible condition '(val > 255) => (0-255 > 255)' drivers/scsi/qla2xxx/qla_os.c:2854:23: warning: assignment to 'struct trace_array *' from 'int' makes pointer from integer without a cast [-Wint-conversion] drivers/scsi/qla2xxx/qla_os.c:2854:25: error: implicit declaration of function 'trace_array_get_by_name'; did you mean 'trace_array_set_clr_event'? [-Werror=implicit-function-declaration] drivers/scsi/qla2xxx/qla_os.c:2869:9: error: implicit declaration of function 'trace_array_put' [-Werror=implicit-function-declaration] kernel/bpf/memalloc.c:499 bpf_mem_alloc_destroy() error: potentially dereferencing uninitialized 'c'. mm/damon/vaddr.c:158 __damon_va_three_regions() error: uninitialized symbol 'start'. net/mac80211/iface.c:251 ieee80211_can_powered_addr_change() warn: inconsistent returns '&local->mtx'. sound/soc/codecs/tas2562.c:442:13: warning: variable 'ret' set but not used [-Wunused-but-set-variable] Error/Warning ids grouped by kconfigs: gcc_recent_errors |-- alpha-allyesconfig | |-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-imu_v11_0_3_program_rlc_ram | |-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead | |-- drivers-scsi-qla2xxx-qla_os.c:error:implicit-declaration-of-function-trace_array_get_by_name | |-- drivers-scsi-qla2xxx-qla_os.c:error:implicit-declaration-of-function-trace_array_put | |-- drivers-scsi-qla2xxx-qla_os.c:warning:assignment-to-struct-trace_array-from-int-makes-pointer-from-integer-without-a-cast | `-- sound-soc-codecs-tas2562.c:warning:variable-ret-set-but-not-used |-- alpha-buildonly-randconfig-r003-20220911 | |-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead | `-- sound-soc-codecs-tas2562.c:warning:variable-ret-set-but-not-used |-- arc-allyesconfig | |-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-imu_v11_0_3_program_rlc_ram | |-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead | `-- sound-soc-codecs-tas2562.c:warning:variable-ret-set-but-not-used |-- arc-buildonly-randconfig-r002-20220912 | `-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead |-- arc-randconfig-r012-20220913 | `-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead |-- arc-randconfig-r021-20220911 | `-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead |-- arc-randconfig-r043-20220911 | `-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead |-- arc-randconfig-r043-20220912 | `-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead |-- arc-randconfig-s052-20220911 | |-- kernel-bpf-hashtab.c:sparse:sparse:cast-removes-address-space-__percpu-of-expression | |-- kernel-bpf-hashtab.c:sparse:sparse:incorrect-type-in-assignment-(different-address-spaces)-expected-void-noderef-__percpu-assigned-pptr-got-void | |-- kernel-bpf-hashtab.c:sparse:sparse:incorrect-type-in-assignment-(different-address-spaces)-expected-void-ptr_to_pptr-got-void-noderef-__percpu-assigned-pptr | |-- kernel-bpf-memalloc.c:sparse:sparse:incorrect-type-in-argument-(different-address-spaces)-expected-void-noderef-__percpu-__pdata-got-void | |-- kernel-bpf-memalloc.c:sparse:sparse:incorrect-type-in-argument-(different-address-spaces)-expected-void-noderef-__percpu-__pdata-got-void-pptr | |-- kernel-bpf-memalloc.c:sparse:sparse:incorrect-type-in-initializer-(different-address-spaces)-expected-void-pptr-got-void-noderef-__percpu | `-- kernel-exit.c:sparse:sparse:incorrect-type-in-initializer-(different-address-spaces)-expected-struct-sighand_struct-sighand-got-struct-sighand_struct-noderef-__rcu-sighand |-- arm-allyesconfig | |-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-imu_v11_0_3_program_rlc_ram | |-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead | `-- sound-soc-codecs-tas2562.c:warning:variable-ret-set-but-not-used |-- arm-defconfig | `-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead |-- arm-randconfig-r001-20220911 | |-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-imu_v11_0_3_program_rlc_ram | |-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead | `-- sound-soc-codecs-tas2562.c:warning:variable-ret-set-but-not-used |-- arm-randconfig-r006-20220911 | |-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead | `-- sound-soc-codecs-tas2562.c:warning:variable-ret-set-but-not-used |-- arm64-allyesconfig | |-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-imu_v11_0_3_program_rlc_ram | |-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead | `-- sound-soc-codecs-tas2562.c:warning:variable-ret-set-but-not-used |-- csky-buildonly-randconfig-r001-20220911 clang_recent_errors |-- i386-randconfig-a012-20220912 | `-- sound-soc-intel-skylake-skl.c:warning:unused-variable-skl |-- i386-randconfig-a015-20220912 | `-- drivers-extcon-extcon-usbc-tusb320.c:warning:expecting-prototype-for-drivers-extcon-extcon-tusb320c().-Prototype-was-for-TUSB320_REG8()-instead |-- powerpc-buildonly-randconfig-r006-20220912 | `-- drivers-extcon-extcon-usbc-tusb320.c:warning:expecting-prototype-for-drivers-extcon-extcon-tusb320c().-Prototype-was-for-TUSB320_REG8()-instead |-- powerpc-randconfig-r004-20220911 | `-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-function-imu_v11_0_3_program_rlc_ram `-- riscv-randconfig-r042-20220912 `-- drivers-extcon-extcon-usbc-tusb320.c:warning:expecting-prototype-for-drivers-extcon-extcon-tusb320c().-Prototype-was-for-TUSB320_REG8()-instead elapsed time: 725m configs tested: 63 configs skipped: 2 gcc tested configs: um i386_defconfig um x86_64_defconfig i386 randconfig-a001-20220912 i386 randconfig-a002-20220912 x86_64 defconfig i386 randconfig-a004-20220912 x86_64 rhel-8.3-kvm i386 randconfig-a003-20220912 x86_64 rhel-8.3-func i386 randconfig-a006-20220912 x86_64 rhel-8.3 x86_64 rhel-8.3-kunit i386 randconfig-a005-20220912 x86_64 rhel-8.3-kselftests m68k allmodconfig x86_64 rhel-8.3-syz arc allyesconfig x86_64 allyesconfig alpha allyesconfig i386 defconfig m68k allyesconfig powerpc allnoconfig powerpc allmodconfig arm defconfig mips allyesconfig x86_64 randconfig-a001-20220912 x86_64 randconfig-a006-20220912 sh allmodconfig x86_64 randconfig-a004-20220912 riscv randconfig-r042-20220911 x86_64 randconfig-a002-20220912 i386 allyesconfig arc randconfig-r043-20220912 x86_64 randconfig-a005-20220912 arm64 allyesconfig x86_64 randconfig-a003-20220912 arm allyesconfig arc randconfig-r043-20220911 s390 randconfig-r044-20220911 ia64 allmodconfig csky allnoconfig arc allnoconfig alpha allnoconfig riscv allnoconfig clang tested configs: i386 randconfig-a014-20220912 i386 randconfig-a013-20220912 x86_64 randconfig-a014-20220912 i386 randconfig-a011-20220912 x86_64 randconfig-a011-20220912 i386 randconfig-a015-20220912 x86_64 randconfig-a012-20220912 i386 randconfig-a012-20220912 x86_64 randconfig-a016-20220912 i386 randconfig-a016-20220912 x86_64 randconfig-a013-20220912 riscv randconfig-r042-20220912 x86_64 randconfig-a015-20220912 hexagon randconfig-r041-20220912 hexagon randconfig-r045-20220911 hexagon randconfig-r041-20220911 hexagon randconfig-r045-20220912 s390 randconfig-r044-20220912 x86_64 rhel-8.3-rust -- 0-DAY CI Kernel Test Service https://01.org/lkp

1 0

[PATCH v2 0/5] soundwire: Fixes for spurious and missing UNATTACH
by Richard Fitzgerald 13 Sep '22

13 Sep '22

The bus and cadence code has several bugs that cause UNATTACH notifications to either be sent spuriously or to be missed. These can be seen occasionally with a single peripheral on the bus, but are much more frequent with multiple peripherals, where several peripherals could change state and report in consecutive PINGs. The root of all of these bugs seems to be a code design flaw that assumed every PING status change would be handled separately. However, PINGs are handled by a workqueue function and there is no guarantee when that function will be scheduled to run or how much CPU time it will receive. PINGs will continue while the work function is handling a snapshot of a previous PING so the code must take account that (a) status could change during the work function and (b) there can be a backlog of changes before the IRQ work function runs again. Tested with 4 peripherals on 1 bus, and 8 peripherals on 2 buses. CHANGES SINCE V1: Patch #3 replaced with a better solution to the same bug. Patches #4 and #5 added to fix some more bugs that were found. Richard Fitzgerald (4): soundwire: bus: Don't lose unattach notifications soundwire: bus: Don't re-enumerate before status is UNATTACHED soundwire: cadence: Fix lost ATTACHED interrupts when enumerating soundwire: bus: Don't exit early if no device IDs were programmed Simon Trimmer (1): soundwire: cadence: fix updating slave status when a bus has multiple peripherals drivers/soundwire/bus.c | 40 +++++++++++----- drivers/soundwire/cadence_master.c | 75 ++++++++++++++++-------------- 2 files changed, 68 insertions(+), 47 deletions(-) -- 2.30.2

2 13

[PATCH 00/10] ASoC: mediatek: Set i2s clock sharing from machine drivers
by Nícolas F. R. A. Prado 13 Sep '22

13 Sep '22

The i2s ports on MediaTek SoCs only support a single data lane. In order to achieve full-duplex operation thus two i2s ports, one for input and one for output, need to be used together and sharing a single clock from one of the ports. This clock sharing setting was previously read by the sound platform driver from the devicetree, but given that the input/output pairing is closely related to which codecs are connected to which ports, the machine sound driver can infer and set it, so that no DT property is required. At this point only mt8183-kukui was using the DT property, but given that this property was never documented, and that the API introduced in this series makes it obsolete, the undocumented DT property can safely be removed. This series adds a function to allow setting the i2s shared clocks, makes use of it in the machine drivers as required, and removes the no longer required DT properties and support for them in the drivers, for all of mt8192, mt8183 and mt8186. Nícolas F. R. A. Prado (10): ASoC: mediatek: mt8192: Allow setting shared clocks from machine driver ASoC: mediatek: mt8192-mt6359: Make i2s9 share the clock from i2s8 ASoC: mediatek: mt8192: Remove clock share parsing from DT ASoC: mediatek: mt8183: Allow setting shared clocks from machine driver ASoC: mediatek: mt8183: Configure shared clocks ASoC: mediatek: mt8183: Remove clock share parsing from DT arm64: dts: mediatek: kukui: Remove i2s-share properties ASoC: mediatek: mt8186: Allow setting shared clocks from machine driver ASoC: mediatek: mt8186: Configure shared clocks ASoC: mediatek: mt8186: Remove clock share parsing from DT .../arm64/boot/dts/mediatek/mt8183-kukui.dtsi | 5 -- sound/soc/mediatek/mt8183/mt8183-afe-common.h | 3 ++ .../mediatek/mt8183/mt8183-da7219-max98357.c | 33 +++++++++++++ sound/soc/mediatek/mt8183/mt8183-dai-i2s.c | 45 ++++++++--------- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 33 +++++++++++++ sound/soc/mediatek/mt8186/mt8186-afe-common.h | 3 ++ sound/soc/mediatek/mt8186/mt8186-dai-i2s.c | 44 ++++++++--------- .../mt8186/mt8186-mt6366-da7219-max98357.c | 18 +++++++ .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 18 +++++++ sound/soc/mediatek/mt8192/mt8192-afe-common.h | 3 ++ sound/soc/mediatek/mt8192/mt8192-dai-i2s.c | 49 ++++++++----------- .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 9 ++++ 12 files changed, 180 insertions(+), 83 deletions(-) -- 2.37.3

4 22

[PATCH] ASoC: ak4458: Add ak4458_reset in device probe and remove
by Shengjiu Wang 13 Sep '22

13 Sep '22

This patch fixup this warning when CONFIG_PM not defined linux/sound/soc/codecs/ak4458.c:631:13: error: 'ak4458_reset' defined but\ not used [-Werror=unused-function] 631 | static void ak4458_reset(struct ak4458_priv *ak4458, bool active) | ^~~~~~~~~~~~ cc1: all warnings being treated as errors Fixes: e9e7df88996d64 ("ASoC: ak4458: Remove component probe() and remove()") Reported-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/ak4458.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c index b534212096ee..626310859814 100644 --- a/sound/soc/codecs/ak4458.c +++ b/sound/soc/codecs/ak4458.c @@ -781,12 +781,16 @@ static int ak4458_i2c_probe(struct i2c_client *i2c) pm_runtime_enable(&i2c->dev); regcache_cache_only(ak4458->regmap, true); + ak4458_reset(ak4458, false); return 0; } static int ak4458_i2c_remove(struct i2c_client *i2c) { + struct ak4458_priv *ak4458 = i2c_get_clientdata(i2c); + + ak4458_reset(ak4458, true); pm_runtime_disable(&i2c->dev); return 0; -- 2.34.1

2 1

[PATCH v9 2/2] ASoC: sun50i-dmic: dt-bindings: add DT bindings for DMIC controller
by Ban Tao 13 Sep '22

13 Sep '22

DT binding documentation for this new ASoC driver. Signed-off-by: Ban Tao <fengzheng923(a)gmail.com> Reviewed-by: Rob Herring <robh(a)kernel.org> Reviewed-by: Maxime Ripard <maxime(a)cerno.tech> --- v1->v2: 1.Fix some build errors. v2->v3: 1.Fix some build errors. v3->v4: 1.None. v4->v5: 1.Add interrupt. 2.Keep clock and reset index. v5->v6: 1.None. v6->v7: 1.None. v7->v8: 1.Fix some build errors. v8->v9: 1.None. --- .../bindings/sound/allwinner,sun50i-h6-dmic.yaml | 79 ++++++++++++++++++++++ 1 file changed, 79 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml new file mode 100644 index 0000000..0cfc07f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml @@ -0,0 +1,79 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/allwinner,sun50i-h6-dmic.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Allwinner H6 DMIC Device Tree Bindings + +maintainers: + - Ban Tao <fengzheng923(a)gmail.com> + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + const: allwinner,sun50i-h6-dmic + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Bus Clock + - description: Module Clock + + clock-names: + items: + - const: bus + - const: mod + + dmas: + items: + - description: RX DMA Channel + + dma-names: + items: + - const: rx + + resets: + maxItems: 1 + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + - resets + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + + #include <dt-bindings/clock/sun50i-h6-ccu.h> + #include <dt-bindings/reset/sun50i-h6-ccu.h> + + dmic: dmic@5095000 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun50i-h6-dmic"; + reg = <0x05095000 0x400>; + interrupts = <GIC_SPI 22 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&ccu CLK_BUS_DMIC>, <&ccu CLK_DMIC>; + clock-names = "bus", "mod"; + dmas = <&dma 7>; + dma-names = "rx"; + resets = <&ccu RST_BUS_DMIC>; + }; + +... -- 2.7.4

4 8

[-next] ASoC: Intel: fix compiler warning of function ‘probe_codec’
by Wang Yufen 13 Sep '22

13 Sep '22

Move the variable ‘skl’ define to CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC enabled, to fix the compiler warning: sound/soc/intel/skylake/skl.c: In function ‘probe_codec’: sound/soc/intel/skylake/skl.c:729:18: warning: unused variable ‘skl’ [-Wunused-variable] 729 | struct skl_dev *skl = bus_to_skl(bus); | ^~~ Signed-off-by: Wang Yufen <wangyufen(a)huawei.com> --- sound/soc/intel/skylake/skl.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index c7c1cad..52a041d 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -726,8 +726,8 @@ static int probe_codec(struct hdac_bus *bus, int addr) unsigned int cmd = (addr << 28) | (AC_NODE_ROOT << 20) | (AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID; unsigned int res = -1; - struct skl_dev *skl = bus_to_skl(bus); #if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC) + struct skl_dev *skl = bus_to_skl(bus); struct hdac_hda_priv *hda_codec; #endif struct hda_codec *codec; -- 1.8.3.1

2 1

[PATCH] ASoC: imx-card: Fix refcount issue with of_node_put
by Shengjiu Wang 13 Sep '22

13 Sep '22

imx_card_parse_of will search all the node with loop, if there is defer probe happen in the middle of loop, the previous released codec node will be released twice, then cause refcount issue. Here assign NULL to pointer of released nodes to fix the issue. Fixes: aa736700f42f ("ASoC: imx-card: Add imx-card machine driver") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/imx-card.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/fsl/imx-card.c b/sound/soc/fsl/imx-card.c index 14be29530fb5..3f128ced4180 100644 --- a/sound/soc/fsl/imx-card.c +++ b/sound/soc/fsl/imx-card.c @@ -698,6 +698,10 @@ static int imx_card_parse_of(struct imx_card_data *data) of_node_put(cpu); of_node_put(codec); of_node_put(platform); + + cpu = NULL; + codec = NULL; + platform = NULL; } return 0; -- 2.34.1

2 1

[PATCH 1/2] ASoC: nau8825: Add ADCOUT IO drive strength control
by David Lin 13 Sep '22

13 Sep '22

Add a property to control the driving of ADCOUT. Signed-off-by: David Lin <CTLIN0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 6 ++++++ sound/soc/codecs/nau8825.h | 3 +++ 2 files changed, 9 insertions(+) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 8213273f501e..c41ac7b9217b 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1983,6 +1983,10 @@ static void nau8825_init_regs(struct nau8825 *nau8825) /* Disable short Frame Sync detection logic */ regmap_update_bits(regmap, NAU8825_REG_LEFT_TIME_SLOT, NAU8825_DIS_FS_SHORT_DET, NAU8825_DIS_FS_SHORT_DET); + /* ADCDAT IO drive strength control */ + regmap_update_bits(regmap, NAU8825_REG_CHARGE_PUMP, + NAU8825_ADCOUT_DS_MASK, + nau8825->adcout_ds << NAU8825_ADCOUT_DS_SFT); } static const struct regmap_config nau8825_regmap_config = { @@ -2521,6 +2525,7 @@ static void nau8825_print_device_properties(struct nau8825 *nau8825) nau8825->jack_eject_debounce); dev_dbg(dev, "crosstalk-enable: %d\n", nau8825->xtalk_enable); + dev_dbg(dev, "adcout-drive-str: %d\n", nau8825->adcout_ds); } static int nau8825_read_device_properties(struct device *dev, @@ -2587,6 +2592,7 @@ static int nau8825_read_device_properties(struct device *dev, nau8825->jack_eject_debounce = 0; nau8825->xtalk_enable = device_property_read_bool(dev, "nuvoton,crosstalk-enable"); + nau8825->adcout_ds = device_property_read_bool(dev, "nuvoton,adcout-drive-str"); nau8825->mclk = devm_clk_get(dev, "mclk"); if (PTR_ERR(nau8825->mclk) == -EPROBE_DEFER) { diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 887bbff03ec6..6d112b6145df 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -418,6 +418,8 @@ #define NAU8825_POWERUP_HP_DRV_L (1 << 0) /* CHARGE_PUMP (0x80) */ +#define NAU8825_ADCOUT_DS_SFT 12 +#define NAU8825_ADCOUT_DS_MASK (1 << NAU8825_ADCOUT_DS_SFT) #define NAU8825_JAMNODCLOW (1 << 10) #define NAU8825_POWER_DOWN_DACR (1 << 9) #define NAU8825_POWER_DOWN_DACL (1 << 8) @@ -477,6 +479,7 @@ struct nau8825 { int imp_rms[NAU8825_XTALK_IMM]; int xtalk_enable; bool xtalk_baktab_initialized; /* True if initialized. */ + bool adcout_ds; }; int nau8825_enable_jack_detect(struct snd_soc_component *component, -- 2.25.1

3 5

[PATCH 0/7] soundwire: Fix driver removal
by Richard Fitzgerald 13 Sep '22

13 Sep '22

Removal of child drivers and the bus driver was broken and would result in a slew of various errors. Most of these were caused by the code shutting down in the wrong order, shutting down the bus driver first. The bus driver should be shut down after the child drivers have been removed (compare with the I2C and SPI subsystem for example). These patches fix that. A secondary problem was over the cleanup of child drivers. The removal functions were not the opposite of the probe function, and the ownership of struct sdw_slave is tricky because it mixes two separate usages and currently has to be "owned" by the bus driver. Tested with 4 peripherals on 1 bus and 8 peripherals on 2 buses. Richard Fitzgerald (7): soundwire: bus: Do not forcibly disable child pm_runtime soundwire: intel_init: Separate shutdown and cleanup ASoC: SOF: Intel: Don't disable Soundwire interrupt before the bus has shut down soundwire: bus: Add remove callback to struct sdw_master_ops soundwire: intel: Don't disable interrupt until children are removed soundwire: intel: Don't disable pm_runtime until children are removed soundwire: bus: Fix premature removal of sdw_slave objects drivers/soundwire/bus.c | 37 ++++++++++++++++++++++++----- drivers/soundwire/intel.c | 13 ++++++++-- drivers/soundwire/intel_init.c | 25 +++++++++++++++---- drivers/soundwire/slave.c | 21 ++++++++++++---- include/linux/soundwire/sdw.h | 3 ++- include/linux/soundwire/sdw_intel.h | 2 ++ sound/soc/sof/intel/hda.c | 16 ++++++++++--- 7 files changed, 96 insertions(+), 21 deletions(-) -- 2.30.2

3 14

[PATCH] dt-bindings: sound: ts3a227e: add control of debounce times
by Astrid Rost 13 Sep '22

13 Sep '22

Add devicetree parameters to control the insertion, release and press debounce times. Signed-off-by: Astrid Rost <astrid.rost(a)axis.com> --- Documentation/devicetree/bindings/sound/ts3a227e.txt | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/ts3a227e.txt b/Documentation/devicetree/bindings/sound/ts3a227e.txt index 21ab45bc7e8f..a4aa4154c54c 100644 --- a/Documentation/devicetree/bindings/sound/ts3a227e.txt +++ b/Documentation/devicetree/bindings/sound/ts3a227e.txt @@ -17,6 +17,15 @@ Optional properies: Select 0/1/2/3/4/5/6/7 to specify MICBIAS voltage 2.1V/2.2V/2.3V/2.4V/2.5V/2.6V/2.7V/2.8V Default value is "1" (2.2V). + - ti,debounce-release: key release debounce (datasheet section 9.6.7). + Select 0/1 to specify the release debounce time 0ms/20ms + Default value is "1" (20 ms). + - ti,debounce-press: key press debounce (datasheet section 9.6.7). + Select 0/1/2/3 to specify the press debounce time 2ms/40ms/80ms/120ms + Default value is "2" (80 ms). + - ti,debounce-insertion: headset insertion debounce (datasheet section 9.6.5). + Select 0/1/2/3/4/5/6/7 to specify the insertion debounce time 2ms/30ms/60ms/90ms/120ms/150ms/1s/2s + Default value is "3" (90 ms). Examples: -- 2.20.1

4 10

[PATCH v3 00/14] ASoC/qcom/arm64: Qualcomm ADSP DTS and binding fixes
by Krzysztof Kozlowski 13 Sep '22

13 Sep '22

Hi, Dependencies/merging ==================== 1. The DTS patches are independent. 2. The binding patches should come together, because of context changes. Could be one of: Qualcomm SoC, ASoC or DT tree. Changes since v2 ================ 1. Patch 9: rename and extend commit msg. 2. Add Rb tags. Changes since v1 ================ 1. Patch 9: New patch. 2. Patch 10: Correct also sound/qcom,q6apm-dai.yaml (Rob). 2. Patch 13: New patch. 3. Add Rb/Tb tags. Best regards, Krzysztof Krzysztof Kozlowski (14): arm64: dts: qcom: sdm630: align APR services node names with dtschema arm64: dts: qcom: sdm845: align APR services node names with dtschema arm64: dts: qcom: sm8250: align APR services node names with dtschema arm64: dts: qcom: msm8996: fix APR services nodes arm64: dts: qcom: sdm845: align dai node names with dtschema arm64: dts: qcom: msm8996: align dai node names with dtschema arm64: dts: qcom: qrb5165-rb5: align dai node names with dtschema arm64: dts: qcom: sm8250: use generic name for LPASS clock controller ASoC: dt-bindings:: qcom,q6asm: convert to dtschema dt-bindings: soc: qcom: apr: correct service children ASoC: dt-bindings: qcom,q6dsp-lpass-ports: cleanup example ASoC: dt-bindings: qcom,q6dsp-lpass-clocks: cleanup example ASoC: dt-bindings: qcom,q6apm-dai: adjust indentation in example dt-bindings: soc: qcom: apr: add missing properties .../bindings/soc/qcom/qcom,apr.yaml | 111 +++++++++++++++-- .../bindings/sound/qcom,q6apm-dai.yaml | 21 ++-- .../bindings/sound/qcom,q6asm-dais.yaml | 112 ++++++++++++++++++ .../devicetree/bindings/sound/qcom,q6asm.txt | 70 ----------- .../sound/qcom,q6dsp-lpass-clocks.yaml | 36 +++--- .../sound/qcom,q6dsp-lpass-ports.yaml | 64 +++++----- arch/arm64/boot/dts/qcom/msm8996.dtsi | 10 +- arch/arm64/boot/dts/qcom/qrb5165-rb5.dts | 4 +- arch/arm64/boot/dts/qcom/sdm630.dtsi | 8 +- arch/arm64/boot/dts/qcom/sdm845-db845c.dts | 2 +- .../boot/dts/qcom/sdm845-xiaomi-beryllium.dts | 2 +- .../boot/dts/qcom/sdm845-xiaomi-polaris.dts | 4 +- arch/arm64/boot/dts/qcom/sdm845.dtsi | 8 +- arch/arm64/boot/dts/qcom/sm8250.dtsi | 10 +- 14 files changed, 293 insertions(+), 169 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6asm-dais.yaml delete mode 100644 Documentation/devicetree/bindings/sound/qcom,q6asm.txt -- 2.34.1

3 17

RE: Re: [PATCH 1/5] ASoC: dt-bindings: fsl_rpmsg: Add a property to assign platform driver name
by Chancel Liu 13 Sep '22

13 Sep '22

> > Add a string property to assign ASoC platform driver name. It also > > represents the rpmsg channel this sound card sits on. This property > > can be omitted if there is only one sound card and it sits on > > "rpmsg-audio-channel". > > > > Signed-off-by: Chancel Liu <chancel.liu(a)nxp.com> > > --- > > .../devicetree/bindings/sound/fsl,rpmsg.yaml | 34 +++++++++++++++++-- > > 1 file changed, 32 insertions(+), 2 deletions(-) > > > > diff --git a/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml > b/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml > > index d370c98a62c7..35e3cb9f768b 100644 > > --- a/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml > > +++ b/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml > > @@ -11,8 +11,11 @@ maintainers: > > > > description: | > > fsl_rpmsg is a virtual audio device. Mapping to real hardware devices > > - are SAI, DMA controlled by Cortex M core. What we see from Linux > > - side is a device which provides audio service by rpmsg channel. > > + are SAI, MICFIL, DMA controlled by Cortex M core. What we see from > > + Linux side is a device which provides audio service by rpmsg channel. > > + We can create different sound cards which access different hardwares > > + such as SAI, MICFIL, .etc through building rpmsg channels between > > + Cortex-A and Cortex-M. > > > > properties: > > compatible: > > @@ -85,6 +88,14 @@ properties: > > This is a boolean property. If present, the receiving function > > will be enabled. > > > > + fsl,platform: > > + $ref: /schemas/types.yaml#/definitions/string > > + description: | > > + A string property to assign ASoC platform driver name. > > No, this is not a property of hardware. Naming of some drivers in some > systems does not fit DTS and bindings. > This property aims to tell the ASoC driver which rpmsg channel the sound card depends on. If there are several sound cards sit on rpmsg, we should pass correct information in dts node to specify the name of rpmsg channel. That is why I meant to add this property. I just want to use a string property to distinguish different names of rpmsg channel. Actually this property is hardware-related. As we discussed before, this kind of sound card based on rpmsg works under this mechanism Cortex-A core tells the Cortex-M core configuration of the PCM parameters then Cortex-M controls real hardware devices. This property specifying rpmsg channel represents the real hardware audio controller. That's my idea adding this property. Do you have any better suggestion? > > It also > > + represents the rpmsg channel this sound card sits on. This property > > + can be omitted if there is only one sound card and it sits on > > + "rpmsg-audio-channel". > > + > > required: > > - compatible > > - model > > @@ -107,3 +118,22 @@ examples: > > <&clk IMX8MN_AUDIO_PLL2_OUT>; > > clock-names = "ipg", "mclk", "dma", "pll8k", "pll11k"; > > }; > > + > > + - | > > + #include <dt-bindings/clock/imx8mm-clock.h> > > + > > + rpmsg_micfil: rpmsg_micfil { > > Node names should be generic. > https://eur01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fdevicetre > e-specification.readthedocs.io%2Fen%2Flatest%2Fchapter2-devicetree-basics.h > tml%23generic-names-recommendation&data=05%7C01%7Cchancel.liu% > 40nxp.com%7C8d5f4ca9669349c597b908da8a6a0311%7C686ea1d3bc2b4c6fa > 92cd99c5c301635%7C0%7C0%7C637974485190445475%7CUnknown%7CTWFp > bGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6 > Mn0%3D%7C3000%7C%7C%7C&sdata=SgDveDWfYyVYhBrG9gpi2aPgGV4j > PwtqbQOtbaE%2FI2s%3D&reserved=0 > > Also: no underscores in node names. > > Best regards, > Krzysztof Thanks for your reminder. We will rename the node in patches for next version like: rpmsg_micfil: audio-controller { // property; }; Regards, Chancel Liu

2 1

[PATCH] ALSA: hda/hdmi: change type for the 'assigned' variable
by Jaroslav Kysela 13 Sep '22

13 Sep '22

This change converts the assigned value from int type to the bool type to retain consistency with other structure members like 'setup', 'non_pcm' etc. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/pci/hda/patch_hdmi.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 22766810635b..4576af4cff90 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -53,7 +53,7 @@ MODULE_PARM_DESC(enable_all_pins, "Forcibly enable all pins"); struct hdmi_spec_per_cvt { hda_nid_t cvt_nid; - int assigned; + bool assigned; /* the stream has been assigned */ unsigned int channels_min; unsigned int channels_max; u32 rates; @@ -1193,7 +1193,7 @@ static int hdmi_pcm_open_no_pin(struct hda_pcm_stream *hinfo, return err; per_cvt = get_cvt(spec, cvt_idx); - per_cvt->assigned = 1; + per_cvt->assigned = true; hinfo->nid = per_cvt->cvt_nid; pin_cvt_fixup(codec, NULL, per_cvt->cvt_nid); @@ -1262,7 +1262,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, per_cvt = get_cvt(spec, cvt_idx); /* Claim converter */ - per_cvt->assigned = 1; + per_cvt->assigned = true; set_bit(pcm_idx, &spec->pcm_in_use); per_pin = get_pin(spec, pin_idx); @@ -1297,7 +1297,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, snd_hdmi_eld_update_pcm_info(&eld->info, hinfo); if (hinfo->channels_min > hinfo->channels_max || !hinfo->rates || !hinfo->formats) { - per_cvt->assigned = 0; + per_cvt->assigned = false; hinfo->nid = 0; snd_hda_spdif_ctls_unassign(codec, pcm_idx); err = -ENODEV; @@ -1756,7 +1756,7 @@ static void silent_stream_enable(struct hda_codec *codec, } per_cvt = get_cvt(spec, cvt_idx); - per_cvt->assigned = 1; + per_cvt->assigned = true; per_pin->cvt_nid = per_cvt->cvt_nid; per_pin->silent_stream = true; @@ -1816,7 +1816,7 @@ static void silent_stream_disable(struct hda_codec *codec, cvt_idx = cvt_nid_to_cvt_index(codec, per_pin->cvt_nid); if (cvt_idx >= 0 && cvt_idx < spec->num_cvts) { per_cvt = get_cvt(spec, cvt_idx); - per_cvt->assigned = 0; + per_cvt->assigned = false; } if (spec->silent_stream_type == SILENT_STREAM_I915) { @@ -2212,7 +2212,7 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, goto unlock; } per_cvt = get_cvt(spec, cvt_idx); - per_cvt->assigned = 0; + per_cvt->assigned = false; hinfo->nid = 0; azx_stream(get_azx_dev(substream))->stripe = 0; -- 2.35.3

2 1

[v2] ALSA: hda: Fix Nvidia dp infoframe
by Mohan Kumar 13 Sep '22

13 Sep '22

Nvidia HDA HW expects infoframe data bytes order same for both HDMI and DP i.e infoframe data starts from 5th bytes offset. As dp infoframe structure has 4th byte as valid infoframe data, use hdmi infoframe structure for nvidia dp infoframe to match HW behvaior. Signed-off-by: Mohan Kumar <mkumard(a)nvidia.com> --- sound/pci/hda/patch_hdmi.c | 23 +++++++++++++++++++---- 1 file changed, 19 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 6c209cd26c0c..9127dd1b1a9c 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -170,6 +170,8 @@ struct hdmi_spec { bool dyn_pcm_no_legacy; /* hdmi interrupt trigger control flag for Nvidia codec */ bool hdmi_intr_trig_ctrl; + bool nv_dp_workaround; /* workaround DP audio infoframe for Nvidia */ + bool intel_hsw_fixup; /* apply Intel platform-specific fixups */ /* * Non-generic VIA/NVIDIA specific @@ -679,15 +681,24 @@ static void hdmi_pin_setup_infoframe(struct hda_codec *codec, int ca, int active_channels, int conn_type) { + struct hdmi_spec *spec = codec->spec; union audio_infoframe ai; memset(&ai, 0, sizeof(ai)); - if (conn_type == 0) { /* HDMI */ + if ((conn_type == 0) || /* HDMI */ + /* Nvidia DisplayPort: Nvidia HW expects same layout as HDMI */ + (conn_type == 1 && spec->nv_dp_workaround)) { struct hdmi_audio_infoframe *hdmi_ai = &ai.hdmi; - hdmi_ai->type = 0x84; - hdmi_ai->ver = 0x01; - hdmi_ai->len = 0x0a; + if (conn_type == 0) { /* HDMI */ + hdmi_ai->type = 0x84; + hdmi_ai->ver = 0x01; + hdmi_ai->len = 0x0a; + } else {/* Nvidia DP */ + hdmi_ai->type = 0x84; + hdmi_ai->ver = 0x1b; + hdmi_ai->len = 0x11 << 2; + } hdmi_ai->CC02_CT47 = active_channels - 1; hdmi_ai->CA = ca; hdmi_checksum_audio_infoframe(hdmi_ai); @@ -3617,6 +3628,7 @@ static int patch_nvhdmi_2ch(struct hda_codec *codec) spec->pcm_playback.rates = SUPPORTED_RATES; spec->pcm_playback.maxbps = SUPPORTED_MAXBPS; spec->pcm_playback.formats = SUPPORTED_FORMATS; + spec->nv_dp_workaround = true; return 0; } @@ -3756,6 +3768,7 @@ static int patch_nvhdmi(struct hda_codec *codec) spec->chmap.ops.chmap_cea_alloc_validate_get_type = nvhdmi_chmap_cea_alloc_validate_get_type; spec->chmap.ops.chmap_validate = nvhdmi_chmap_validate; + spec->nv_dp_workaround = true; codec->link_down_at_suspend = 1; @@ -3779,6 +3792,7 @@ static int patch_nvhdmi_legacy(struct hda_codec *codec) spec->chmap.ops.chmap_cea_alloc_validate_get_type = nvhdmi_chmap_cea_alloc_validate_get_type; spec->chmap.ops.chmap_validate = nvhdmi_chmap_validate; + spec->nv_dp_workaround = true; codec->link_down_at_suspend = 1; @@ -3992,6 +4006,7 @@ static int tegra_hdmi_init(struct hda_codec *codec) spec->chmap.ops.chmap_cea_alloc_validate_get_type = nvhdmi_chmap_cea_alloc_validate_get_type; spec->chmap.ops.chmap_validate = nvhdmi_chmap_validate; + spec->nv_dp_workaround = true; return 0; } -- 2.17.1

3 3

[PATCH] ALSA: hda: Fix tegra hda dp infoframe struct
by Mohan Kumar 13 Sep '22

13 Sep '22

Tegra HDA HW expects infoframe data bytes order same for both HDMI and DP i.e infoframe data starts from 5th bytes offset. This hw behavior mandates to have dummy bytes for dp infoframe structure for Tegra. Signed-off-by: Mohan Kumar <mkumard(a)nvidia.com> --- sound/pci/hda/patch_hdmi.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 6c209cd26c0c..a52e764db2e0 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -218,6 +218,9 @@ struct dp_audio_infoframe { u8 type; /* 0x84 */ u8 len; /* 0x1b */ u8 ver; /* 0x11 << 2 */ +#if IS_ENABLED(CONFIG_SND_HDA_TEGRA) + u8 checksum; /* Tegra HW expects infoframe bytes from 5th offset */ +#endif u8 CC02_CT47; /* match with HDMI infoframe from this on */ u8 SS01_SF24; -- 2.17.1

3 2

[PATCH] ALSA: hda/realtek: Add quirk for Huawei WRT-WX9
by huangwenhui 13 Sep '22

13 Sep '22

Fixes headphone and headset microphone detection on Huawei WRT-WX9. Signed-off-by: huangwenhui <huangwenhuia(a)uniontech.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 38930cf5aace..0996a8fd008c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9569,6 +9569,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), SND_PCI_QUIRK(0x1849, 0x1233, "ASRock NUC Box 1100", ALC233_FIXUP_NO_AUDIO_JACK), SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MACH-WX9", ALC256_FIXUP_HUAWEI_MACH_WX9_PINS), + SND_PCI_QUIRK(0x19e5, 0x320f, "Huawei WRT-WX9 ", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1b35, 0x1235, "CZC B20", ALC269_FIXUP_CZC_B20), SND_PCI_QUIRK(0x1b35, 0x1236, "CZC TMI", ALC269_FIXUP_CZC_TMI), SND_PCI_QUIRK(0x1b35, 0x1237, "CZC L101", ALC269_FIXUP_CZC_L101), -- 2.20.1

2 1

[PATCH] ALSA: hda/tegra: set depop delay for tegra
by Mohan Kumar 13 Sep '22

13 Sep '22

Reduce the suspend time by setting depop delay to 10ms for tegra. Signed-off-by: Mohan Kumar <mkumard(a)nvidia.com> --- sound/pci/hda/patch_hdmi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 6c209cd26c0c..acaf6b790ee1 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -3984,6 +3984,7 @@ static int tegra_hdmi_init(struct hda_codec *codec) generic_hdmi_init_per_pins(codec); + codec->depop_delay = 10; codec->patch_ops.build_pcms = tegra_hdmi_build_pcms; spec->chmap.ops.chmap_cea_alloc_validate_get_type = nvhdmi_chmap_cea_alloc_validate_get_type; -- 2.17.1

2 1

[PATCH] ALSA: hda: add Intel 5 Series / 3400 PCI DID
by Kai Vehmanen 13 Sep '22

13 Sep '22

Handle 0x3b57 variant with same AZX_DCAPS_INTEL_PCH_NOPM capabilities as 0x3b56. In practise this allow use of HDMI/DP display audio via i915. BugLink: https://gitlab.freedesktop.org/drm/intel/-/issues/2751 Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.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 7e605ce43f41..2a93bc64c2d8 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2543,6 +2543,8 @@ static const struct pci_device_id azx_ids[] = { /* 5 Series/3400 */ { PCI_DEVICE(0x8086, 0x3b56), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH_NOPM }, + { PCI_DEVICE(0x8086, 0x3b57), + .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH_NOPM }, /* Poulsbo */ { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH_BASE }, base-commit: 0186f7325909dec0cb52814384062f6de3b87941 -- 2.37.3

2 1

Re: [PATCH] ASoC: ak4458.c: move ak4458_reset() under CONFIG_PM
by S.J. Wang 13 Sep '22

13 Sep '22

> Hi Mark > > > > This patch fixup this warning > > > > > > linux/sound/soc/codecs/ak4458.c:631:13: error: 'ak4458_reset' defined > but\ > > > not used [-Werror=unused-function] > > > 631 | static void ak4458_reset(struct ak4458_priv *ak4458, bool active) > > > | ^~~~~~~~~~~~ > > > cc1: all warnings being treated as errors > > > > > > Fixes: e9e7df88996d64 ("ASoC: ak4458: Remove component probe() and > > > remove()") > > > > Is this the best fix, or should we be adding _reset() calls into the > > device level probe() and remove() functions? > > Hmm... I'm not familiar with ak4458.c. > Is your opinion for original patch ? or this patch ? > This patch is just trying to fix the compile issue. > > Thank you for your help !! > Thanks for catching this issue. Without or with _reset() in probe() and remove(), the driver can work in both case. Best regards Wang shengjiu

2 1

[PATCH] ASoC: ak4458.c: move ak4458_reset() under CONFIG_PM
by Kuninori Morimoto 13 Sep '22

13 Sep '22

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This patch fixup this warning linux/sound/soc/codecs/ak4458.c:631:13: error: 'ak4458_reset' defined but\ not used [-Werror=unused-function] 631 | static void ak4458_reset(struct ak4458_priv *ak4458, bool active) | ^~~~~~~~~~~~ cc1: all warnings being treated as errors Fixes: e9e7df88996d64 ("ASoC: ak4458: Remove component probe() and remove()") Cc: Shengjiu Wang <shengjiu.wang(a)nxp.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/ak4458.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c index b534212096ee..4735aae3bfbd 100644 --- a/sound/soc/codecs/ak4458.c +++ b/sound/soc/codecs/ak4458.c @@ -628,6 +628,7 @@ static struct snd_soc_dai_driver ak4497_dai = { .ops = &ak4458_dai_ops, }; +#ifdef CONFIG_PM static void ak4458_reset(struct ak4458_priv *ak4458, bool active) { if (ak4458->reset_gpiod) { @@ -636,7 +637,6 @@ static void ak4458_reset(struct ak4458_priv *ak4458, bool active) } } -#ifdef CONFIG_PM static int __maybe_unused ak4458_runtime_suspend(struct device *dev) { struct ak4458_priv *ak4458 = dev_get_drvdata(dev); -- 2.25.1

2 2

[linux-next:master] BUILD REGRESSION 044b771be9c5de9d817dfafb829d2f049c71c3b4
by kernel test robot 13 Sep '22

13 Sep '22

tree/branch: https://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git master branch HEAD: 044b771be9c5de9d817dfafb829d2f049c71c3b4 Add linux-next specific files for 20220912 Error/Warning reports: https://lore.kernel.org/linux-media/202209020437.eXEOdmfe-lkp@intel.com https://lore.kernel.org/linux-mm/202209042337.FQi69rLV-lkp@intel.com https://lore.kernel.org/linux-mm/202209060229.dVuyxjBv-lkp@intel.com https://lore.kernel.org/linux-mm/202209080718.y5QmlNKH-lkp@intel.com Error/Warning: (recently discovered and may have been fixed) ERROR: modpost: "ioread64" [drivers/pci/switch/switchtec.ko] undefined! drivers/gpu/drm/amd/amdgpu/imu_v11_0_3.c:139:6: warning: no previous prototype for 'imu_v11_0_3_program_rlc_ram' [-Wmissing-prototypes] drivers/gpu/drm/drm_atomic_helper.c:802: warning: expecting prototype for drm_atomic_helper_check_wb_connector_state(). Prototype was for drm_atomic_helper_check_wb_encoder_state() instead drivers/pinctrl/pinctrl-amd.c:288 amd_gpio_dbg_show() warn: format string contains non-ascii character '\x9a' drivers/pinctrl/pinctrl-amd.c:288 amd_gpio_dbg_show() warn: format string contains non-ascii character '\xa1' drivers/pinctrl/pinctrl-amd.c:370 amd_gpio_dbg_show() warn: format string contains non-ascii character '\x95' drivers/scsi/qla2xxx/qla_os.c:2854:23: warning: assignment to 'struct trace_array *' from 'int' makes pointer from integer without a cast [-Wint-conversion] drivers/scsi/qla2xxx/qla_os.c:2854:25: error: implicit declaration of function 'trace_array_get_by_name'; did you mean 'trace_array_set_clr_event'? [-Werror=implicit-function-declaration] drivers/scsi/qla2xxx/qla_os.c:2869:9: error: implicit declaration of function 'trace_array_put' [-Werror=implicit-function-declaration] kernel/bpf/memalloc.c:499 bpf_mem_alloc_destroy() error: potentially dereferencing uninitialized 'c'. sound/soc/codecs/tas2562.c:442:13: warning: variable 'ret' set but not used [-Wunused-but-set-variable] Error/Warning ids grouped by kconfigs: gcc_recent_errors |-- alpha-allyesconfig | |-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-imu_v11_0_3_program_rlc_ram | |-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead | |-- drivers-scsi-qla2xxx-qla_os.c:error:implicit-declaration-of-function-trace_array_get_by_name | |-- drivers-scsi-qla2xxx-qla_os.c:error:implicit-declaration-of-function-trace_array_put | |-- drivers-scsi-qla2xxx-qla_os.c:warning:assignment-to-struct-trace_array-from-int-makes-pointer-from-integer-without-a-cast | `-- sound-soc-codecs-tas2562.c:warning:variable-ret-set-but-not-used |-- alpha-randconfig-r005-20220911 | |-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-imu_v11_0_3_program_rlc_ram | `-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead |-- alpha-randconfig-r033-20220911 | |-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-imu_v11_0_3_program_rlc_ram | `-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead |-- alpha-randconfig-r035-20220911 | |-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead | `-- sound-soc-codecs-tas2562.c:warning:variable-ret-set-but-not-used |-- arc-allyesconfig | |-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-imu_v11_0_3_program_rlc_ram | |-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead | `-- sound-soc-codecs-tas2562.c:warning:variable-ret-set-but-not-used |-- arc-axs101_defconfig | `-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead |-- arc-randconfig-r016-20220912 | |-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-imu_v11_0_3_program_rlc_ram | `-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead |-- arc-randconfig-r032-20220912 | `-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead |-- arc-randconfig-r043-20220911 | `-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead |-- arc-randconfig-r043-20220912 | `-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead |-- arm-allyesconfig | |-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-imu_v11_0_3_program_rlc_ram | |-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead | `-- sound-soc-codecs-tas2562.c:warning:variable-ret-set-but-not-used |-- arm-defconfig | `-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead |-- arm-randconfig-c034-20220911 | `-- net-bluetooth-mgmt.c:WARNING:kzalloc-should-be-used-for-rp-instead-of-kmalloc-memset |-- arm-randconfig-r032-20220911 | |-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead | `-- sound-soc-codecs-tas2562.c:warning:variable-ret-set-but-not-used |-- arm64-allyesconfig | |-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-imu_v11_0_3_program_rlc_ram | |-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead | `-- sound-soc-codecs-tas2562.c:warning:variable-ret-set-but-not-used |-- arm64-buildonly-randconfig-r002-20220912 | |-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-imu_v11_0_3_program_rlc_ram | |-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead | `-- sound-soc-codecs-tas2562.c:warning:variable-ret-set-but-not-used clang_recent_errors |-- arm64-randconfig-r011-20220912 | `-- drivers-extcon-extcon-usbc-tusb320.c:warning:expecting-prototype-for-drivers-extcon-extcon-tusb320c().-Prototype-was-for-TUSB320_REG8()-instead |-- hexagon-allyesconfig | |-- drivers-extcon-extcon-usbc-tusb320.c:warning:expecting-prototype-for-drivers-extcon-extcon-tusb320c().-Prototype-was-for-TUSB320_REG8()-instead | `-- drivers-staging-media-deprecated-cpia2-cpia2_usb.c:warning:variable-frame_count-set-but-not-used |-- i386-randconfig-a012-20220912 | `-- sound-soc-intel-skylake-skl.c:warning:unused-variable-skl |-- i386-randconfig-a015-20220912 | `-- drivers-extcon-extcon-usbc-tusb320.c:warning:expecting-prototype-for-drivers-extcon-extcon-tusb320c().-Prototype-was-for-TUSB320_REG8()-instead |-- mips-randconfig-r021-20220911 | `-- drivers-extcon-extcon-usbc-tusb320.c:warning:expecting-prototype-for-drivers-extcon-extcon-tusb320c().-Prototype-was-for-TUSB320_REG8()-instead |-- powerpc-randconfig-r002-20220911 | `-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-function-imu_v11_0_3_program_rlc_ram |-- riscv-randconfig-r015-20220912 | `-- drivers-extcon-extcon-usbc-tusb320.c:warning:expecting-prototype-for-drivers-extcon-extcon-tusb320c().-Prototype-was-for-TUSB320_REG8()-instead `-- riscv-randconfig-r042-20220912 `-- drivers-extcon-extcon-usbc-tusb320.c:warning:expecting-prototype-for-drivers-extcon-extcon-tusb320c().-Prototype-was-for-TUSB320_REG8()-instead elapsed time: 724m configs tested: 82 configs skipped: 2 gcc tested configs: um i386_defconfig um x86_64_defconfig x86_64 defconfig x86_64 randconfig-a001-20220912 arc axs101_defconfig x86_64 randconfig-a002-20220912 i386 randconfig-a001-20220912 x86_64 rhel-8.3 x86_64 randconfig-a003-20220912 x86_64 rhel-8.3-func arm stm32_defconfig alpha allyesconfig sh se7724_defconfig x86_64 allyesconfig arm defconfig alpha allnoconfig i386 randconfig-a002-20220912 x86_64 rhel-8.3-kunit riscv allnoconfig m68k allmodconfig x86_64 rhel-8.3-kvm x86_64 randconfig-a006-20220912 i386 randconfig-a004-20220912 x86_64 randconfig-a004-20220912 csky allnoconfig arc allnoconfig sh se7705_defconfig i386 defconfig i386 allyesconfig m68k q40_defconfig i386 randconfig-a003-20220912 arc allyesconfig x86_64 rhel-8.3-kselftests x86_64 rhel-8.3-syz arm iop32x_defconfig x86_64 randconfig-a005-20220912 powerpc allnoconfig m68k allyesconfig riscv randconfig-r042-20220911 arm trizeps4_defconfig i386 randconfig-a006-20220912 arc randconfig-r043-20220912 mips allyesconfig sparc64 defconfig i386 randconfig-a005-20220912 powerpc allmodconfig arm64 allyesconfig sh allmodconfig s390 debug_defconfig arm allyesconfig arc randconfig-r043-20220911 s390 randconfig-r044-20220911 ia64 allmodconfig sh sh2007_defconfig sh ecovec24_defconfig sh alldefconfig sh rsk7269_defconfig clang tested configs: x86_64 randconfig-a011-20220912 x86_64 randconfig-a012-20220912 i386 randconfig-a013-20220912 x86_64 randconfig-a013-20220912 i386 randconfig-a011-20220912 x86_64 randconfig-a014-20220912 mips rs90_defconfig arm pcm027_defconfig x86_64 randconfig-a016-20220912 i386 randconfig-a012-20220912 arm ixp4xx_defconfig x86_64 randconfig-a015-20220912 i386 randconfig-a014-20220912 arm vexpress_defconfig i386 randconfig-a015-20220912 riscv randconfig-r042-20220912 hexagon randconfig-r041-20220912 i386 randconfig-a016-20220912 hexagon randconfig-r045-20220911 hexagon randconfig-r041-20220911 hexagon randconfig-r045-20220912 s390 randconfig-r044-20220912 mips malta_qemu_32r6_defconfig powerpc gamecube_defconfig x86_64 rhel-8.3-rust -- 0-DAY CI Kernel Test Service https://01.org/lkp

1 0

[PATCH] soundwire: bus: conditionally recheck UNATTACHED status
by Bard Liao 12 Sep '22

12 Sep '22

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> In configurations with two amplifiers on the same link, the Intel CI reports occasional enumeration/initialization timeouts during suspend-resume stress tests, where one of the two amplifiers becomes UNATTACHED immediately after being enumerated. This problem was reported both with Maxim and Realtek codecs, which pointed initially to a problem with status handling on the Intel side. The Cadence IP integrated on Intel platforms throws an interrupt when the status changes, and the information is kept with sticky bits until cleared. We initially added more checks to make sure the edge detection did not miss any transition, but that did not improve the results significantly. With the recent addition of the read_ping_status() callback, we were able to show that the status in sticky bits is shown as UNATTACHED even though the PING frames show the problematic device as ATTACHED. That completely breaks the entire logic where we assumed that a peripheral would always re-attach as device0. The resume timeouts make sense in that in those cases, the enumeration/initialization never happens a second time. One possible explanation is that this problem typically happens when a bus clash is reported, so it could very well be that the detection is fooled by a transient electrical issue or conflict between two peripherals. This patch conditionally double-checks the status reported in the sticky bits with the actual PING frame status. If the peripheral reports as attached in PING frames, the early detection based on sticky bits is discarded. Note that this patch only corrects issues of false positives on the manager side. If the peripheral lost and regain sync, then it would report as attached on Device0. A peripheral that would not reset its dev_num would not be compliant with the MIPI specification. BugLink: https://github.com/thesofproject/linux/issues/3638 BugLink: https://github.com/thesofproject/linux/issues/3325 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- Hi Vinod, You will need the "ASoC/soundwire: log actual PING status on resume issues" series which is applied at ASoC tree before appling this patch. --- drivers/soundwire/bus.c | 19 +++++++++++++++++++ drivers/soundwire/intel.c | 1 + include/linux/soundwire/sdw.h | 3 +++ 3 files changed, 23 insertions(+) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index 2772973eebb1..d0d486f07673 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -1767,6 +1767,25 @@ int sdw_handle_slave_status(struct sdw_bus *bus, slave->status != SDW_SLAVE_UNATTACHED) { dev_warn(&slave->dev, "Slave %d state check1: UNATTACHED, status was %d\n", i, slave->status); + + if (bus->recheck_unattached && bus->ops->read_ping_status) { + u32 ping_status; + + mutex_lock(&bus->msg_lock); + + ping_status = bus->ops->read_ping_status(bus); + + mutex_unlock(&bus->msg_lock); + + ping_status >>= (i * 2); + ping_status &= 0x3; + + if (ping_status != 0) { + dev_warn(&slave->dev, "Slave %d state in PING frame is %d, ignoring earlier detection\n", + i, ping_status); + continue; + } + } sdw_modify_slave_status(slave, SDW_SLAVE_UNATTACHED); } } diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index 25ec9c272239..0c6e674dbf85 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -1311,6 +1311,7 @@ static int intel_link_probe(struct auxiliary_device *auxdev, bus->link_id = auxdev->id; bus->dev_num_ida_min = INTEL_DEV_NUM_IDA_MIN; + bus->recheck_unattached = true; sdw_cdns_probe(cdns); diff --git a/include/linux/soundwire/sdw.h b/include/linux/soundwire/sdw.h index a2b31d25ea27..51ac71984260 100644 --- a/include/linux/soundwire/sdw.h +++ b/include/linux/soundwire/sdw.h @@ -892,6 +892,8 @@ struct sdw_master_ops { * @dev_num_ida_min: if set, defines the minimum values for the IDA * used to allocate system-unique device numbers. This value needs to be * identical across all SoundWire bus in the system. + * @recheck_unattached: if set, double-check UNATTACHED status changes + * by reading PING frame status. */ struct sdw_bus { struct device *dev; @@ -917,6 +919,7 @@ struct sdw_bus { bool multi_link; int hw_sync_min_links; int dev_num_ida_min; + bool recheck_unattached; }; int sdw_bus_master_add(struct sdw_bus *bus, struct device *parent, -- 2.25.1

3 2

[PATCH] ASoC: rt5640: Fix the issue of the abnormal JD2 status
by Oder Chiou 12 Sep '22

12 Sep '22

The patch fixes the issue of the abnormal JD2 status. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Reported-by: Sameer Pujar <spujar(a)nvidia.com> --- sound/soc/codecs/rt5640.c | 64 +++++++++++++++++++++++++++------------ sound/soc/codecs/rt5640.h | 14 +++++++++ 2 files changed, 59 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 5a844329800..0f8e6dd214b 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2494,7 +2494,7 @@ static void rt5640_enable_jack_detect(struct snd_soc_component *component, /* Select JD-source */ snd_soc_component_update_bits(component, RT5640_JD_CTRL, - RT5640_JD_MASK, rt5640->jd_src); + RT5640_JD_MASK, rt5640->jd_src << RT5640_JD_SFT); /* Selecting GPIO01 as an interrupt */ snd_soc_component_update_bits(component, RT5640_GPIO_CTRL1, @@ -2504,12 +2504,8 @@ static void rt5640_enable_jack_detect(struct snd_soc_component *component, snd_soc_component_update_bits(component, RT5640_GPIO_CTRL3, RT5640_GP1_PF_MASK, RT5640_GP1_PF_OUT); - /* Enabling jd2 in general control 1 */ snd_soc_component_write(component, RT5640_DUMMY1, 0x3f41); - /* Enabling jd2 in general control 2 */ - snd_soc_component_write(component, RT5640_DUMMY2, 0x4001); - rt5640_set_ovcd_params(component); /* @@ -2518,12 +2514,25 @@ static void rt5640_enable_jack_detect(struct snd_soc_component *component, * pin 0/1 instead of it being stuck to 1. So we invert the JD polarity * on systems where the hardware does not already do this. */ - if (rt5640->jd_inverted) - snd_soc_component_write(component, RT5640_IRQ_CTRL1, - RT5640_IRQ_JD_NOR); - else - snd_soc_component_write(component, RT5640_IRQ_CTRL1, - RT5640_IRQ_JD_NOR | RT5640_JD_P_INV); + if (rt5640->jd_inverted) { + if (rt5640->jd_src == RT5640_JD_SRC_JD1_IN4P) + snd_soc_component_write(component, RT5640_IRQ_CTRL1, + RT5640_IRQ_JD_NOR); + else if (rt5640->jd_src == RT5640_JD_SRC_JD2_IN4N) + snd_soc_component_update_bits(component, RT5640_DUMMY2, + RT5640_IRQ_JD2_MASK | RT5640_JD2_MASK, + RT5640_IRQ_JD2_NOR | RT5640_JD2_EN); + } else { + if (rt5640->jd_src == RT5640_JD_SRC_JD1_IN4P) + snd_soc_component_write(component, RT5640_IRQ_CTRL1, + RT5640_IRQ_JD_NOR | RT5640_JD_P_INV); + else if (rt5640->jd_src == RT5640_JD_SRC_JD2_IN4N) + snd_soc_component_update_bits(component, RT5640_DUMMY2, + RT5640_IRQ_JD2_MASK | RT5640_JD2_P_MASK | + RT5640_JD2_MASK, + RT5640_IRQ_JD2_NOR | RT5640_JD2_P_INV | + RT5640_JD2_EN); + } rt5640->jack = jack; if (rt5640->jack->status & SND_JACK_MICROPHONE) { @@ -2725,10 +2734,8 @@ static int rt5640_probe(struct snd_soc_component *component) if (device_property_read_u32(component->dev, "realtek,jack-detect-source", &val) == 0) { - if (val <= RT5640_JD_SRC_GPIO4) - rt5640->jd_src = val << RT5640_JD_SFT; - else if (val == RT5640_JD_SRC_HDA_HEADER) - rt5640->jd_src = RT5640_JD_SRC_HDA_HEADER; + if (val <= RT5640_JD_SRC_HDA_HEADER) + rt5640->jd_src = val; else dev_warn(component->dev, "Warning: Invalid jack-detect-source value: %d, leaving jack-detect disabled\n", val); @@ -2809,12 +2816,31 @@ static int rt5640_resume(struct snd_soc_component *component) regcache_sync(rt5640->regmap); if (rt5640->jack) { - if (rt5640->jd_src == RT5640_JD_SRC_HDA_HEADER) + if (rt5640->jd_src == RT5640_JD_SRC_HDA_HEADER) { snd_soc_component_update_bits(component, RT5640_DUMMY2, 0x1100, 0x1100); - else - snd_soc_component_write(component, RT5640_DUMMY2, - 0x4001); + } else { + if (rt5640->jd_inverted) { + if (rt5640->jd_src == RT5640_JD_SRC_JD2_IN4N) + snd_soc_component_update_bits( + component, RT5640_DUMMY2, + RT5640_IRQ_JD2_MASK | + RT5640_JD2_MASK, + RT5640_IRQ_JD2_NOR | + RT5640_JD2_EN); + + } else { + if (rt5640->jd_src == RT5640_JD_SRC_JD2_IN4N) + snd_soc_component_update_bits( + component, RT5640_DUMMY2, + RT5640_IRQ_JD2_MASK | + RT5640_JD2_P_MASK | + RT5640_JD2_MASK, + RT5640_IRQ_JD2_NOR | + RT5640_JD2_P_INV | + RT5640_JD2_EN); + } + } queue_delayed_work(system_long_wq, &rt5640->jack_work, 0); } diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index 505c9351405..f58b88e3325 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -1984,6 +1984,20 @@ #define RT5640_M_MONO_ADC_R_SFT 12 #define RT5640_MCLK_DET (0x1 << 11) +/* General Control 1 (0xfb) */ +#define RT5640_IRQ_JD2_MASK (0x1 << 12) +#define RT5640_IRQ_JD2_SFT 12 +#define RT5640_IRQ_JD2_BP (0x0 << 12) +#define RT5640_IRQ_JD2_NOR (0x1 << 12) +#define RT5640_JD2_P_MASK (0x1 << 10) +#define RT5640_JD2_P_SFT 10 +#define RT5640_JD2_P_NOR (0x0 << 10) +#define RT5640_JD2_P_INV (0x1 << 10) +#define RT5640_JD2_MASK (0x1 << 8) +#define RT5640_JD2_SFT 8 +#define RT5640_JD2_DIS (0x0 << 8) +#define RT5640_JD2_EN (0x1 << 8) + /* Codec Private Register definition */ /* MIC Over current threshold scale factor (0x15) */ -- 2.34.1

2 2

[PATCH v2] ASoC: Intel: sof_rt5682: Add support for jsl_rt5682_rt1019
by Sean Hong 12 Sep '22

12 Sep '22

This patch adds the driver data for rt5682 support jsl_rt5682_rt1019. Signed-off-by: Sean Hong <sean.hong(a)quanta.corp-partner.google.com> --- sound/soc/intel/boards/sof_rt5682.c | 9 +++++++++ sound/soc/intel/common/soc-acpi-intel-jsl-match.c | 12 ++++++++++++ 2 files changed, 21 insertions(+) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 045965312..3a840f3a9 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -1100,6 +1100,15 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(1) | SOF_RT5682_NUM_HDMIDEV(4)), }, + { + .name = "jsl_rt5682_rt1019", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_MCLK_24MHZ | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT1019_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1)), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c index b95c4b2cd..e374bf163 100644 --- a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c @@ -29,6 +29,11 @@ static const struct snd_soc_acpi_codecs rt1015p_spk = { .codecs = {"RTL1015"} }; +static struct snd_soc_acpi_codecs rt1019p_spk = { + .num_codecs = 1, + .codecs = {"RTL1019"} +}; + static const struct snd_soc_acpi_codecs mx98360a_spk = { .num_codecs = 1, .codecs = {"MX98360A"} @@ -78,6 +83,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[] = { .quirk_data = &mx98360a_spk, .sof_tplg_filename = "sof-jsl-rt5682-mx98360a.tplg", }, + { + .comp_ids = &rt5682_rt5682s_hp, + .drv_name = "jsl_rt5682_rt1019", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &rt1019p_spk, + .sof_tplg_filename = "sof-jsl-rt5682-rt1019.tplg", + }, { .id = "10134242", .drv_name = "jsl_cs4242_mx98360a", -- 2.25.1

3 3

[PATCH] ASoC: wm8978: Support the recording function of codec
by Mengchen Li 12 Sep '22

12 Sep '22

If we want to use the microphone function of the wm8978 codec, we must enable the microphone bias circuit and power up the bias circuit,otherwise the microphone can not vibrate for sound source sampling.The bit that controls the output voltage of the bias circuit is the fourth bit of the WM8978_POWER_MANAGEMENT_1 register,so this bit must be set to 1.This patch can support the recording function of the wm8978 codec. Signed-off-by: Mengchen Li <mengchenli64(a)gmail.com> --- sound/soc/codecs/wm8978.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index a682f80..fdd5429 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -822,7 +822,9 @@ static int wm8978_hw_params(struct snd_pcm_substream *substream, snd_soc_component_write(component, WM8978_AUDIO_INTERFACE, iface_ctl); snd_soc_component_write(component, WM8978_ADDITIONAL_CONTROL, add_ctl); - + /* Enable MICBEN */ + snd_soc_component_write(component, WM8978_POWER_MANAGEMENT_1, + snd_soc_component_read(component, WM8978_POWER_MANAGEMENT_1) | 0x10); if (wm8978->sysclk != current_clk_id) { if (wm8978->sysclk == WM8978_PLL) /* Run CODEC from PLL instead of MCLK */ -- 2.7.4

2 1

[PATCH v1] ALSA: dummy: Add customizable volume min/max.
by YJ Lee 12 Sep '22

12 Sep '22

Add module parameters to support customized min/max volume leveling, which will be useful to test devices with different volume granularity. Signed-off-by: YJ Lee <yunjunlee(a)chromium.org> --- sound/drivers/dummy.c | 34 ++++++++++++++++++++++++---------- 1 file changed, 24 insertions(+), 10 deletions(-) diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index 2a7fc49c1a7c5..64fb2778f1e9a 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -42,6 +42,8 @@ MODULE_LICENSE("GPL"); #define USE_CHANNELS_MAX 2 #define USE_PERIODS_MIN 1 #define USE_PERIODS_MAX 1024 +#define USE_MIXER_VOLUME_LEVEL_MIN -50 +#define USE_MIXER_VOLUME_LEVEL_MAX 100 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ @@ -50,6 +52,8 @@ static char *model[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = NULL}; static int pcm_devs[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; static int pcm_substreams[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 8}; //static int midi_devs[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 2}; +static int mixer_volume_level_min = USE_MIXER_VOLUME_LEVEL_MIN; +static int mixer_volume_level_max = USE_MIXER_VOLUME_LEVEL_MAX; #ifdef CONFIG_HIGH_RES_TIMERS static bool hrtimer = 1; #endif @@ -69,6 +73,10 @@ module_param_array(pcm_substreams, int, NULL, 0444); MODULE_PARM_DESC(pcm_substreams, "PCM substreams # (1-128) for dummy driver."); //module_param_array(midi_devs, int, NULL, 0444); //MODULE_PARM_DESC(midi_devs, "MIDI devices # (0-2) for dummy driver."); +module_param(mixer_volume_level_min, int, 0444); +MODULE_PARM_DESC(mixer_volume_level_min, "Minimum mixer volume level for dummy driver. Default: -50"); +module_param(mixer_volume_level_max, int, 0444); +MODULE_PARM_DESC(mixer_volume_level_max, "Maximum mixer volume level for dummy driver. Default: 100"); module_param(fake_buffer, bool, 0444); MODULE_PARM_DESC(fake_buffer, "Fake buffer allocations."); #ifdef CONFIG_HIGH_RES_TIMERS @@ -713,8 +721,8 @@ static int snd_dummy_volume_info(struct snd_kcontrol *kcontrol, { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 2; - uinfo->value.integer.min = -50; - uinfo->value.integer.max = 100; + uinfo->value.integer.min = mixer_volume_level_min; + uinfo->value.integer.max = mixer_volume_level_max; return 0; } @@ -739,15 +747,15 @@ static int snd_dummy_volume_put(struct snd_kcontrol *kcontrol, int left, right; left = ucontrol->value.integer.value[0]; - if (left < -50) - left = -50; - if (left > 100) - left = 100; + if (left < mixer_volume_level_min) + left = mixer_volume_level_min; + if (left > mixer_volume_level_max) + left = mixer_volume_level_max; right = ucontrol->value.integer.value[1]; - if (right < -50) - right = -50; - if (right > 100) - right = 100; + if (right < mixer_volume_level_min) + right = mixer_volume_level_min; + if (right > mixer_volume_level_max) + right = mixer_volume_level_max; spin_lock_irq(&dummy->mixer_lock); change = dummy->mixer_volume[addr][0] != left || dummy->mixer_volume[addr][1] != right; @@ -1076,6 +1084,12 @@ static int snd_dummy_probe(struct platform_device *devptr) dummy->pcm_hw.channels_max = m->channels_max; } + if (mixer_volume_level_min > mixer_volume_level_max) { + pr_warn("snd-dummy: Invalid mixer volume level: min=%d, max=%d. Fall back to default value.\n", + mixer_volume_level_min, mixer_volume_level_max); + mixer_volume_level_min = USE_MIXER_VOLUME_LEVEL_MIN; + mixer_volume_level_max = USE_MIXER_VOLUME_LEVEL_MAX; + } err = snd_card_dummy_new_mixer(dummy); if (err < 0) return err; -- 2.31.0

2 2

[PATCH v1] ALSA: dummy: Fix trailing whitespaces.
by YJ Lee 12 Sep '22

12 Sep '22

Fix checkpatch.pl ERROR: trailing whitespaces. Signed-off-by: YJ Lee <yunjunlee(a)chromium.org> --- sound/drivers/dummy.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index 2a7fc49c1a7c5..fcf1ee00bd214 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -296,7 +296,7 @@ static void dummy_systimer_callback(struct timer_list *t) struct dummy_systimer_pcm *dpcm = from_timer(dpcm, t, timer); unsigned long flags; int elapsed = 0; - + spin_lock_irqsave(&dpcm->lock, flags); dummy_systimer_update(dpcm); dummy_systimer_rearm(dpcm); @@ -717,7 +717,7 @@ static int snd_dummy_volume_info(struct snd_kcontrol *kcontrol, uinfo->value.integer.max = 100; return 0; } - + static int snd_dummy_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -766,7 +766,7 @@ static const DECLARE_TLV_DB_SCALE(db_scale_dummy, -4500, 30, 0); .private_value = addr } #define snd_dummy_capsrc_info snd_ctl_boolean_stereo_info - + static int snd_dummy_capsrc_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1100,7 +1100,7 @@ static int snd_dummy_suspend(struct device *pdev) snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); return 0; } - + static int snd_dummy_resume(struct device *pdev) { struct snd_card *card = dev_get_drvdata(pdev); -- 2.37.2.789.g6183377224-goog

2 2

[PATCH] ASoC: ts3a227e: allow enabling the jack detect in driver
by Astrid Rost 12 Sep '22

12 Sep '22

ti,jack-detect enables the jack detection input device Signed-off-by: Astrid Rost <astrid.rost(a)axis.com> --- sound/soc/codecs/Kconfig | 2 ++ sound/soc/codecs/ts3a227e.c | 62 ++++++++++++++++++++++++++++++++++++- 2 files changed, 63 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d16b4efb88a7..cb86e52cd02f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1671,6 +1671,8 @@ config SND_SOC_TLV320ADCX140 config SND_SOC_TS3A227E tristate "TI Headset/Mic detect and keypress chip" depends on I2C + select SND_JACK + select SND_JACK_INPUT_DEV config SND_SOC_TSCS42XX tristate "Tempo Semiconductor TSCS42xx CODEC" diff --git a/sound/soc/codecs/ts3a227e.c b/sound/soc/codecs/ts3a227e.c index d8ab0810fceb..b5d0d32cafdb 100644 --- a/sound/soc/codecs/ts3a227e.c +++ b/sound/soc/codecs/ts3a227e.c @@ -38,6 +38,32 @@ static const int ts3a227e_buttons[] = { SND_JACK_BTN_3, }; +/* Headphones jack detection DAPM pin */ +static struct snd_soc_jack_pin ts3a227e_jack_pins[] = { + + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Internal Speaker", + /* disable speaker when hp jack is inserted */ + .mask = SND_JACK_HEADPHONE, + .invert = 1, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Internal Mic", + /* disable microphone when microphone jack is inserted */ + .mask = SND_JACK_MICROPHONE, + .invert = 1, + }, + +}; + #define TS3A227E_NUM_BUTTONS 4 #define TS3A227E_JACK_MASK (SND_JACK_HEADPHONE | \ SND_JACK_MICROPHONE | \ @@ -250,7 +276,12 @@ int ts3a227e_enable_jack_detect(struct snd_soc_component *component, } EXPORT_SYMBOL_GPL(ts3a227e_enable_jack_detect); -static struct snd_soc_component_driver ts3a227e_soc_driver; +static int ts3a227e_probe(struct snd_soc_component *component); + +static const struct snd_soc_component_driver ts3a227e_soc_driver = { + .name = "Audio Accessory Detection ts3a227e", + .probe = ts3a227e_probe, +}; static const struct regmap_config ts3a227e_regmap_config = { .val_bits = 8, @@ -355,6 +386,35 @@ static int ts3a227e_resume(struct device *dev) } #endif +static int ts3a227e_probe(struct snd_soc_component *component) +{ + int ret = 0; + bool enable = 0; + struct snd_soc_card *card = component->card; + struct ts3a227e *ts3a227e = snd_soc_component_get_drvdata(component); + + enable = device_property_read_bool(component->dev, "ti,jack-detect"); + if (enable) { + /* Enable Headset and 4 Buttons Jack detection */ + ts3a227e->jack = devm_kzalloc(component->dev, + sizeof(struct snd_soc_jack), GFP_KERNEL); + + ret = snd_soc_card_jack_new_pins(card, "Headset Jack", + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, + ts3a227e->jack, + ts3a227e_jack_pins, + ARRAY_SIZE(ts3a227e_jack_pins)); + if (ret) + return ret; + + ret = ts3a227e_enable_jack_detect(component, ts3a227e->jack); + } + + return ret; +} + static const struct dev_pm_ops ts3a227e_pm = { SET_SYSTEM_SLEEP_PM_OPS(ts3a227e_suspend, ts3a227e_resume) }; -- 2.20.1

4 5

Buffers used in ASoC Intel Atom SST driver
by Takashi Iwai 11 Sep '22

11 Sep '22

Hi, while cleaning up / fixing the memory allocation code, I noticed that Intel SST driver is the only driver that currently uses the SNDRV_DMA_TYPE_CONTINUOUS pages for its PCM buffer. Is there any reason not to use the standard device buffer (SNDRV_DMA_TYPE_DEV)? i.e. the change like below. thanks, Takashi -- 8< -- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ASoC: Intel: sst: Switch to standard device pages Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index a56dd48c045f..c75616a5fd0a 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -676,10 +676,9 @@ static int sst_soc_pcm_new(struct snd_soc_component *component, if (dai->driver->playback.channels_min || dai->driver->capture.channels_min) { - snd_pcm_set_managed_buffer_all(pcm, - SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_DMA), - SST_MIN_BUFFER, SST_MAX_BUFFER); + snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV, + pcm->card->dev, + SST_MIN_BUFFER, SST_MAX_BUFFER); } return 0; } -- 2.35.3

4 5

[PATCH] ASoC: nau8825: Add ADCOUT IO drive strength control
by David Lin 11 Sep '22

11 Sep '22

Add a property to control the driving of ADCOUT. Signed-off-by: David Lin <CTLIN0(a)nuvoton.com> --- Documentation/devicetree/bindings/sound/nau8825.txt | 2 ++ sound/soc/codecs/nau8825.c | 6 ++++++ sound/soc/codecs/nau8825.h | 3 +++ 3 files changed, 11 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/nau8825.txt b/Documentation/devicetree/bindings/sound/nau8825.txt index 388a7bc60b1f..54972cc2ca1a 100644 --- a/Documentation/devicetree/bindings/sound/nau8825.txt +++ b/Documentation/devicetree/bindings/sound/nau8825.txt @@ -71,6 +71,8 @@ Optional properties: - nuvoton,crosstalk-enable: make crosstalk function enable if set. + - nuvoton,adcout-drive-str: ADCOUT IO drive strength. 0 - normal, 1 - stronger. + - clocks: list of phandle and clock specifier pairs according to common clock bindings for the clocks described in clock-names - clock-names: should include "mclk" for the MCLK master clock diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 8213273f501e..fc00e600cf2c 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1983,6 +1983,10 @@ static void nau8825_init_regs(struct nau8825 *nau8825) /* Disable short Frame Sync detection logic */ regmap_update_bits(regmap, NAU8825_REG_LEFT_TIME_SLOT, NAU8825_DIS_FS_SHORT_DET, NAU8825_DIS_FS_SHORT_DET); + /* ADCDAT IO drive strength control */ + regmap_update_bits(remap, NAU8825_REG_CHARGE_PUMP, + NAU8825_ADCOUT_DS_MASK, + nau8825->adcout_ds << NAU8825_ADCOUT_DS_SFT); } static const struct regmap_config nau8825_regmap_config = { @@ -2521,6 +2525,7 @@ static void nau8825_print_device_properties(struct nau8825 *nau8825) nau8825->jack_eject_debounce); dev_dbg(dev, "crosstalk-enable: %d\n", nau8825->xtalk_enable); + dev_dbg(dev, "adcout-drive-str: %d\n", nau8825->adcout_ds); } static int nau8825_read_device_properties(struct device *dev, @@ -2587,6 +2592,7 @@ static int nau8825_read_device_properties(struct device *dev, nau8825->jack_eject_debounce = 0; nau8825->xtalk_enable = device_property_read_bool(dev, "nuvoton,crosstalk-enable"); + nau8825->adcout_ds = device_property_read_bool(dev, "nuvoton,adcout-drive-str"); nau8825->mclk = devm_clk_get(dev, "mclk"); if (PTR_ERR(nau8825->mclk) == -EPROBE_DEFER) { diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 887bbff03ec6..6d112b6145df 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -418,6 +418,8 @@ #define NAU8825_POWERUP_HP_DRV_L (1 << 0) /* CHARGE_PUMP (0x80) */ +#define NAU8825_ADCOUT_DS_SFT 12 +#define NAU8825_ADCOUT_DS_MASK (1 << NAU8825_ADCOUT_DS_SFT) #define NAU8825_JAMNODCLOW (1 << 10) #define NAU8825_POWER_DOWN_DACR (1 << 9) #define NAU8825_POWER_DOWN_DACL (1 << 8) @@ -477,6 +479,7 @@ struct nau8825 { int imp_rms[NAU8825_XTALK_IMM]; int xtalk_enable; bool xtalk_baktab_initialized; /* True if initialized. */ + bool adcout_ds; }; int nau8825_enable_jack_detect(struct snd_soc_component *component, -- 2.25.1

4 3

[PATCH] ASoC: wm8731: fix build when I2C is not enabled
by Randy Dunlap 11 Sep '22

11 Sep '22

Building wm8731-i2c.c has a kconfig warning and build errors when I2C is not enabled, so prevent that kconfig-uration. WARNING: unmet direct dependencies detected for SND_SOC_WM8731_I2C Depends on [n]: SOUND [=y] && !UML && SND [=m] && SND_SOC [=m] && I2C [=n] Selected by [m]: - SND_SOC_DB1200 [=m] && SOUND [=y] && !UML && SND [=m] && SND_SOC [=m] && SND_SOC_AU1XPSC [=m] and subsequent build errors: ../sound/soc/codecs/wm8731-i2c.c:64:1: warning: data definition has no type or storage class 64 | module_i2c_driver(wm8731_i2c_driver); ../sound/soc/codecs/wm8731-i2c.c:64:1: error: type defaults to 'int' in declaration of 'module_i2c_driver' [-Werror=implicit-int] ../sound/soc/codecs/wm8731-i2c.c:64:1: warning: parameter names (without types) in function declaration ../sound/soc/codecs/wm8731-i2c.c:55:26: warning: 'wm8731_i2c_driver' defined but not used [-Wunused-variable] 55 | static struct i2c_driver wm8731_i2c_driver = { Fixes: 9dc15f81baf2 ("ASoC: wm8731: Factor out the I2C and SPI bus code into separate modules") Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: alsa-devel(a)alsa-project.org Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai --- sound/soc/au1x/Kconfig | 1 + 1 file changed, 1 insertion(+) --- a/sound/soc/au1x/Kconfig +++ b/sound/soc/au1x/Kconfig @@ -54,6 +54,7 @@ config SND_SOC_DB1000 config SND_SOC_DB1200 tristate "DB1200/DB1300/DB1550 Audio support" depends on SND_SOC_AU1XPSC + depends on I2C select SND_SOC_AU1XPSC_AC97 select SND_SOC_AC97_CODEC select SND_SOC_WM9712

1 0

[PATCH] ALSA: hda: Fix hang at HD-audio codec unbinding due to refcount saturation
by Takashi Iwai 10 Sep '22

10 Sep '22

We fixed the potential deadlock at dynamic unbinding the HD-audio codec at the commit 7206998f578d ("ALSA: hda: Fix potential deadlock at codec unbinding"), but ironically, this caused another potential deadlock. The current code uses refcount_dec() and waits for the pending task with wait_event for dropping the refcount to 0. This works fine when PCMs are assigned and actually waiting for the refcount drop. Meanwhile, when there was no PCM assigned, the refcount_dec() call itself was supposed to drop to zero -- alas, it doesn't in reality; refcount_dec() complains, spews kernel warning and it saturates instead of dropping to 0, due to the nature of refcount_dec() implementation. This eventually blocks the wait_event() wakeup and the code get stuck there. For avoiding the problem, we call refcount_dec_and_test() and skips the sync-wait if it already reaches to zero. The patch does a slight code reshuffling to make sure to invoke other disconnect calls before the sync-wait, too. Fixes: 7206998f578d ("ALSA: hda: Fix potential deadlock at codec unbinding") Reported-by: Ville Syrjälä <ville.syrjala(a)linux.intel.com> Tested-by: Ville Syrjälä <ville.syrjala(a)linux.intel.com> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/YxtflWQnslMHVlU7@intel.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_bind.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index cae9a975cbcc..1a868dd9dc4b 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -157,10 +157,10 @@ static int hda_codec_driver_remove(struct device *dev) return codec->bus->core.ext_ops->hdev_detach(&codec->core); } - refcount_dec(&codec->pcm_ref); snd_hda_codec_disconnect_pcms(codec); snd_hda_jack_tbl_disconnect(codec); - wait_event(codec->remove_sleep, !refcount_read(&codec->pcm_ref)); + if (!refcount_dec_and_test(&codec->pcm_ref)) + wait_event(codec->remove_sleep, !refcount_read(&codec->pcm_ref)); snd_power_sync_ref(codec->bus->card); if (codec->patch_ops.free) -- 2.35.3

1 0

hda codec unbind refcount hang
by Ville Syrjälä 10 Sep '22

10 Sep '22

Hi Takashi, commit 7206998f578d ("ALSA: hda: Fix potential deadlock at codec unbinding") introduced a problem on at least one of my older machines. The problem happens when hda_codec_driver_remove() encounters a codec without any pcms (and thus the refcount is 1) and tries to call refcount_dec(). Turns out refcount_dec() doesn't like to be used for dropping the refcount to 0, and instead if spews a warning and does its saturate thing. The subsequent wait_event() is then permanently stuck waiting on the saturated refcount. I've definitely seen the same kind of pattern used elsewhere in the kernel as well, so the fact that refcount_t can't be used to implement it is a bit of surprise to me. I guess most other places still use atomic_t instead. -- Ville Syrjälä Intel

2 4

aseqdump crashing after running for a bit
by GitHub issues - edited 10 Sep '22

10 Sep '22

alsa-project/alsa-utils issue #170 was edited from BlastBolt5: I have been working on a script to control pipewire sources and sinks using faders on my midi keyboard but aseqdump keeps erroring out with > Port unsubscribed 128:0 I don't have any other midi device to test it out and see if it's specific to this conrtoller but I'm using the Korg nanoKontrol2. If i need to provide anything else just let me know. Issue URL : https://github.com/alsa-project/alsa-utils/issues/170 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

aseqdump crashing after running for a bit
by GitHub issues - opened 10 Sep '22

10 Sep '22

alsa-project/alsa-utils issue #170 was opened from BlastBolt5: I have been working on a script to control pipewire sources and sinks using faders on my midi keyboard but it keeps erroring out with > Port unsubscribed 128:0 I don't have any other midi device to test it out and see if it's specific to this conrtoller but I'm using the Korg nanoKontrol2. If i need to provide anything else just let me know. Issue URL : https://github.com/alsa-project/alsa-utils/issues/170 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH 0/8] Add support for the internal RK3308 audio codec
by luca.ceresoli＠bootlin.com 10 Sep '22

10 Sep '22

From: Luca Ceresoli <luca.ceresoli(a)bootlin.com> This series of patches adds support to use the internal audio codec of the Rockchip RK3308 SoC. This codec is internally connected to the I2S peripherals on the same chip, and it has some peculiarities arising from that interconnection. For proper bidirectional operation with the internal codec, the I2S peripheral needs two clock sources (tx and rx), while connection with an external codec commonly needs only one. Since v5.16 there is a driver for the I2S in sound/soc/rockchip/rockchip_i2s_tdm.c, but it does not correctly handle receiving those two clocks via the .set_sysclk op. Patch 5 fixes that. However the simple-audio-card and audio-graph-card sound card drivers do not support calling .set_sysclk twice, thus patch 6 makes the .init op of struct asoc_simple_priv overridable by a driver in order to be able to call .set_sysclk twice and thus configure both clocks. Patch 7 adds the codec driver and patch 8 builds on top of all the above by implementing a simple RK3308-specific audio card, based on audio-graph-card. This card sets all the I2S input clocks. Patches 1-2 add DT bindings for the codec and the card. Patches 3-4 add to the SoC DT file two I2S controllers (those which are internally connected to the internal codec) and the codec itself. Luca Luca Ceresoli (8): ASoC: rockchip: rk3308: add internal audio codec bindings ASoC: rockchip: rk3308: add audio card bindings arm64: dts: rockchip: add i2s_8ch_2 and i2s_8ch_3 arm64: dts: rockchip: add the internal audio codec ASoC: rockchip: i2s-tdm: Fix clk_id usage in .set_sysclk() ASoC: audio-graph: let dai_link->init be overridable ASoC: codecs: Add RK3308 internal audio codec driver ASoC: rockchip: add new RK3308 sound card .../rockchip,rk3308-audio-graph-card.yaml | 50 + .../bindings/sound/rockchip,rk3308-codec.yaml | 102 + MAINTAINERS | 14 + arch/arm64/boot/dts/rockchip/rk3308.dtsi | 68 + .../dt-bindings/sound/rockchip,rk3308-codec.h | 15 + include/sound/simple_card_utils.h | 1 + sound/soc/codecs/Kconfig | 11 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rk3308_codec.c | 2122 +++++++++++++++++ sound/soc/codecs/rk3308_codec.h | 648 +++++ sound/soc/generic/audio-graph-card.c | 2 + sound/soc/rockchip/Kconfig | 14 + sound/soc/rockchip/Makefile | 1 + sound/soc/rockchip/rockchip_i2s_tdm.c | 18 +- sound/soc/rockchip/rockchip_rk3308_card.c | 97 + 15 files changed, 3159 insertions(+), 6 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk3308-audio-graph-card.yaml create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk3308-codec.yaml create mode 100644 include/dt-bindings/sound/rockchip,rk3308-codec.h create mode 100644 sound/soc/codecs/rk3308_codec.c create mode 100644 sound/soc/codecs/rk3308_codec.h create mode 100644 sound/soc/rockchip/rockchip_rk3308_card.c -- 2.34.1

6 23

[PATCH] ASoC: hdmi-codec.c: use devm_kzalloc() for DMA data
by Kuninori Morimoto 10 Sep '22

10 Sep '22

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> hdmi-codec.c is using kzalloc(), but we can replace it to devm_kzalloc() and then, we can remove .remove callback. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/hdmi-codec.c | 10 +--------- 1 file changed, 1 insertion(+), 9 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 863e679d2ac1..0b1cdb2d6049 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -823,7 +823,7 @@ static int hdmi_dai_probe(struct snd_soc_dai *dai) if (ret) return ret; - daifmt = kzalloc(sizeof(*daifmt), GFP_KERNEL); + daifmt = devm_kzalloc(dai->dev, sizeof(*daifmt), GFP_KERNEL); if (!daifmt) return -ENOMEM; @@ -890,17 +890,10 @@ static int hdmi_dai_spdif_probe(struct snd_soc_dai *dai) return 0; } -static int hdmi_codec_dai_remove(struct snd_soc_dai *dai) -{ - kfree(dai->playback_dma_data); - return 0; -} - static const struct snd_soc_dai_driver hdmi_i2s_dai = { .name = "i2s-hifi", .id = DAI_ID_I2S, .probe = hdmi_dai_probe, - .remove = hdmi_codec_dai_remove, .playback = { .stream_name = "I2S Playback", .channels_min = 2, @@ -925,7 +918,6 @@ static const struct snd_soc_dai_driver hdmi_spdif_dai = { .name = "spdif-hifi", .id = DAI_ID_SPDIF, .probe = hdmi_dai_spdif_probe, - .remove = hdmi_codec_dai_remove, .playback = { .stream_name = "SPDIF Playback", .channels_min = 2, -- 2.25.1

3 2

[PATCH v3] ASoC: fsl_asrc: Add initialization finishing check in runtime resume
by Shengjiu Wang 10 Sep '22

10 Sep '22

If the initialization is not finished, then filling input data to the FIFO may fail. So it is better to add initialization finishing check in the runtime resume for suspend & resume case. And consider the case of three instances working in parallel, increase the retry times to 50 for more initialization time. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Reviewed-by: Nicolin Chen <nicolinc(a)gmail.com> --- changes in v3: - update warning message. changes in v2: - update comments. sound/soc/fsl/fsl_asrc.c | 29 ++++++++++++++++++++++++++++- 1 file changed, 28 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index aa5edf32d988..936aef5d2767 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -20,6 +20,7 @@ #define IDEAL_RATIO_DECIMAL_DEPTH 26 #define DIVIDER_NUM 64 +#define INIT_RETRY_NUM 50 #define pair_err(fmt, ...) \ dev_err(&asrc->pdev->dev, "Pair %c: " fmt, 'A' + index, ##__VA_ARGS__) @@ -27,6 +28,9 @@ #define pair_dbg(fmt, ...) \ dev_dbg(&asrc->pdev->dev, "Pair %c: " fmt, 'A' + index, ##__VA_ARGS__) +#define pair_warn(fmt, ...) \ + dev_warn(&asrc->pdev->dev, "Pair %c: " fmt, 'A' + index, ##__VA_ARGS__) + /* Corresponding to process_option */ static unsigned int supported_asrc_rate[] = { 5512, 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, @@ -579,7 +583,7 @@ static void fsl_asrc_start_pair(struct fsl_asrc_pair *pair) { struct fsl_asrc *asrc = pair->asrc; enum asrc_pair_index index = pair->index; - int reg, retry = 10, i; + int reg, retry = INIT_RETRY_NUM, i; /* Enable the current pair */ regmap_update_bits(asrc->regmap, REG_ASRCTR, @@ -592,6 +596,10 @@ static void fsl_asrc_start_pair(struct fsl_asrc_pair *pair) reg &= ASRCFG_INIRQi_MASK(index); } while (!reg && --retry); + /* NOTE: Doesn't treat initialization timeout as an error */ + if (!retry) + pair_warn("initialization isn't finished\n"); + /* Make the input fifo to ASRC STALL level */ regmap_read(asrc->regmap, REG_ASRCNCR, &reg); for (i = 0; i < pair->channels * 4; i++) @@ -1257,6 +1265,7 @@ static int fsl_asrc_runtime_resume(struct device *dev) { struct fsl_asrc *asrc = dev_get_drvdata(dev); struct fsl_asrc_priv *asrc_priv = asrc->private; + int reg, retry = INIT_RETRY_NUM; int i, ret; u32 asrctr; @@ -1295,6 +1304,24 @@ static int fsl_asrc_runtime_resume(struct device *dev) regmap_update_bits(asrc->regmap, REG_ASRCTR, ASRCTR_ASRCEi_ALL_MASK, asrctr); + /* Wait for status of initialization for all enabled pairs */ + do { + udelay(5); + regmap_read(asrc->regmap, REG_ASRCFG, &reg); + reg = (reg >> ASRCFG_INIRQi_SHIFT(0)) & 0x7; + } while ((reg != ((asrctr >> ASRCTR_ASRCEi_SHIFT(0)) & 0x7)) && --retry); + + /* + * NOTE: Doesn't treat initialization timeout as an error + * Some of the pairs may success, then still can continue. + */ + if (!retry) { + for (i = ASRC_PAIR_A; i < ASRC_PAIR_MAX_NUM; i++) { + if ((asrctr & ASRCTR_ASRCEi_MASK(i)) && !(reg & (1 << i))) + dev_warn(dev, "Pair %c initialization isn't finished\n", 'A' + i); + } + } + return 0; disable_asrck_clk: -- 2.34.1

2 1

Re: [PATCH] ASoC: mediatek: mt6359: fix platform_get_irq error checking
by Mark Brown 10 Sep '22

10 Sep '22

On Fri, Sep 09, 2022 at 02:45:11PM +0800, Yu Zhe wrote: > The platform_get_irq() function returns negative error codes on error, > fix the checking. This doesn't apply against current code, please check and resend.

1 0

[PATCH 0/7] ASoC: SOF: ipc4: Add support for 'mtrace' log extraction
by Peter Ujfalusi 10 Sep '22

10 Sep '22

Hi, The traditional dtrace used by SOF IPC3 is not available with firmware built as IPC4. With the new IPC implementation we have a new log extraction infrastructure and 'mtrace' is one way to get the logs out from the firmware for debugging. The protocol is relatively simple: The shared sram's debug window is split up to 'slots' Each DSP core will get a dedicated slot assigned for log output. The function of a slots can be checked in a descriptor table. The slot used for logging has the following layout: u32 host_read_ptr; u32 dsp_write_ptr; u8 buffer[]; The two pointers are offsets within the buffer. The firmware manages the write pointer, the host updates the read pointer value and based on the two pointers the log entries can be extracted. The firmware also sends notification when new entries are available in the log slot. The user space tool to read the log entries can be found in the SOF repository: https://github.com/thesofproject/sof/blob/main/tools/mtrace/mtrace-reader.py Regards, Peter --- Peter Ujfalusi (7): ASoC: SOF: ipc4: Only print LOG BUFFER update message info if requested ASoC: SOF: ipc4: Add macro to get core ID from log buffer status message ASoC: SOF: ipc4: Add define for the outbox window index ASoC: SOF: ipc4: Configure the debug box offset ASoC: SOF: ipc4: Add support for mtrace log extraction ASoC: SOF: Intel: icl: Set IPC4-specific DSP ops ASoC: SOF: Intel: Add mtrace type information for IPC4 include/sound/sof/ipc4/header.h | 5 + sound/soc/sof/Makefile | 3 +- sound/soc/sof/intel/apl.c | 2 + sound/soc/sof/intel/cnl.c | 2 + sound/soc/sof/intel/icl.c | 30 +- sound/soc/sof/intel/mtl.c | 2 + sound/soc/sof/intel/tgl.c | 2 + sound/soc/sof/ipc4-loader.c | 2 + sound/soc/sof/ipc4-mtrace.c | 643 ++++++++++++++++++++++++++++++++ sound/soc/sof/ipc4-priv.h | 18 +- sound/soc/sof/ipc4.c | 22 +- 11 files changed, 724 insertions(+), 7 deletions(-) create mode 100644 sound/soc/sof/ipc4-mtrace.c -- 2.37.3

2 8

[linux-next:master] BUILD REGRESSION 9a82ccda91ed2b40619cb3c10d446ae1f97bab6e
by kernel test robot 10 Sep '22

10 Sep '22

tree/branch: https://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git master branch HEAD: 9a82ccda91ed2b40619cb3c10d446ae1f97bab6e Add linux-next specific files for 20220909 Error/Warning reports: https://lore.kernel.org/linux-mm/202209042337.FQi69rLV-lkp@intel.com https://lore.kernel.org/linux-mm/202209080718.y5QmlNKH-lkp@intel.com Error/Warning: (recently discovered and may have been fixed) ./drivers/gpu/drm/drm_atomic_helper.c:802: warning: expecting prototype for drm_atomic_helper_check_wb_connector_state(). Prototype was for drm_atomic_helper_check_wb_encoder_state() instead ERROR: modpost: "__divdi3" [drivers/gpu/drm/vkms/vkms.ko] undefined! ERROR: modpost: "__udivdi3" [drivers/gpu/drm/vkms/vkms.ko] undefined! arm-linux-gnueabi-ld: vkms_formats.c:(.text+0x824): undefined reference to `__aeabi_ldivmod' drivers/base/regmap/regmap-mmio.c:222:17: error: implicit declaration of function 'writesb'; did you mean 'writeb'? [-Werror=implicit-function-declaration] drivers/base/regmap/regmap-mmio.c:225:17: error: implicit declaration of function 'writesw'; did you mean 'writew'? [-Werror=implicit-function-declaration] drivers/base/regmap/regmap-mmio.c:228:17: error: implicit declaration of function 'writesl'; did you mean 'writel'? [-Werror=implicit-function-declaration] drivers/base/regmap/regmap-mmio.c:358:17: error: implicit declaration of function 'readsb'; did you mean 'readb'? [-Werror=implicit-function-declaration] drivers/base/regmap/regmap-mmio.c:361:17: error: implicit declaration of function 'readsw'; did you mean 'readw'? [-Werror=implicit-function-declaration] drivers/base/regmap/regmap-mmio.c:364:17: error: implicit declaration of function 'readsl'; did you mean 'readl'? [-Werror=implicit-function-declaration] drivers/gpu/drm/amd/amdgpu/imu_v11_0_3.c:139:6: warning: no previous prototype for 'imu_v11_0_3_program_rlc_ram' [-Wmissing-prototypes] drivers/gpu/drm/drm_atomic_helper.c:802: warning: expecting prototype for drm_atomic_helper_check_wb_connector_state(). Prototype was for drm_atomic_helper_check_wb_encoder_state() instead drivers/gpu/drm/vkms/vkms_formats.c:259: undefined reference to `__divdi3' drivers/gpu/drm/vkms/vkms_plane.c:110 vkms_plane_atomic_update() warn: variable dereferenced before check 'fb' (see line 108) drivers/scsi/qla2xxx/qla_os.c:2854:23: warning: assignment to 'struct trace_array *' from 'int' makes pointer from integer without a cast [-Wint-conversion] drivers/scsi/qla2xxx/qla_os.c:2854:25: error: implicit declaration of function 'trace_array_get_by_name'; did you mean 'trace_array_set_clr_event'? [-Werror=implicit-function-declaration] drivers/scsi/qla2xxx/qla_os.c:2869:9: error: implicit declaration of function 'trace_array_put' [-Werror=implicit-function-declaration] kernel/bpf/memalloc.c:499 bpf_mem_alloc_destroy() error: potentially dereferencing uninitialized 'c'. ld: drivers/gpu/drm/vkms/vkms_formats.c:260: undefined reference to `__divdi3' ld: vkms_formats.c:(.text+0x362): undefined reference to `__divdi3' ld: vkms_formats.c:(.text+0x3b2): undefined reference to `__divdi3' ld: vkms_formats.c:(.text+0x3ba): undefined reference to `__divdi3' ld: vkms_formats.c:(.text+0x47f): undefined reference to `__divdi3' microblaze-linux-ld: drivers/gpu/drm/vkms/vkms_formats.c:260: undefined reference to `__divdi3' mips-linux-ld: vkms_formats.c:(.text+0x8b8): undefined reference to `__divdi3' mips-linux-ld: vkms_formats.c:(.text.argb_u16_to_RGB565+0xd0): undefined reference to `__divdi3' sound/soc/codecs/tas2562.c:442:13: warning: variable 'ret' set but not used [-Wunused-but-set-variable] vkms_formats.c:(.text+0x266): undefined reference to `__divdi3' vkms_formats.c:(.text+0x338): undefined reference to `__divdi3' vkms_formats.c:(.text+0x388): undefined reference to `__divdi3' vkms_formats.c:(.text+0x390): undefined reference to `__divdi3' vkms_formats.c:(.text+0x455): undefined reference to `__divdi3' vkms_formats.c:(.text+0x804): undefined reference to `__aeabi_ldivmod' vkms_formats.c:(.text+0x89c): undefined reference to `__divdi3' vkms_formats.c:(.text.argb_u16_to_RGB565+0xb0): undefined reference to `__divdi3' Error/Warning ids grouped by kconfigs: gcc_recent_errors |-- alpha-allyesconfig | |-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-imu_v11_0_3_program_rlc_ram | |-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead | |-- drivers-scsi-qla2xxx-qla_os.c:error:implicit-declaration-of-function-trace_array_get_by_name | |-- drivers-scsi-qla2xxx-qla_os.c:error:implicit-declaration-of-function-trace_array_put | |-- drivers-scsi-qla2xxx-qla_os.c:warning:assignment-to-struct-trace_array-from-int-makes-pointer-from-integer-without-a-cast | `-- sound-soc-codecs-tas2562.c:warning:variable-ret-set-but-not-used |-- arc-allyesconfig | |-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-imu_v11_0_3_program_rlc_ram | |-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead | `-- sound-soc-codecs-tas2562.c:warning:variable-ret-set-but-not-used |-- arc-randconfig-r025-20220908 | `-- sound-soc-codecs-tas2562.c:warning:variable-ret-set-but-not-used |-- arc-randconfig-s033-20220907 | |-- kernel-bpf-hashtab.c:sparse:sparse:cast-removes-address-space-__percpu-of-expression | |-- kernel-bpf-hashtab.c:sparse:sparse:incorrect-type-in-assignment-(different-address-spaces)-expected-void-noderef-__percpu-assigned-pptr-got-void | |-- kernel-bpf-hashtab.c:sparse:sparse:incorrect-type-in-assignment-(different-address-spaces)-expected-void-ptr_to_pptr-got-void-noderef-__percpu-assigned-pptr | |-- kernel-bpf-memalloc.c:sparse:sparse:incorrect-type-in-argument-(different-address-spaces)-expected-void-noderef-__percpu-__pdata-got-void | |-- kernel-bpf-memalloc.c:sparse:sparse:incorrect-type-in-argument-(different-address-spaces)-expected-void-noderef-__percpu-__pdata-got-void-pptr | |-- kernel-bpf-memalloc.c:sparse:sparse:incorrect-type-in-initializer-(different-address-spaces)-expected-void-pptr-got-void-noderef-__percpu | `-- kernel-exit.c:sparse:sparse:incorrect-type-in-initializer-(different-address-spaces)-expected-struct-sighand_struct-sighand-got-struct-sighand_struct-noderef-__rcu-sighand |-- arm-allyesconfig | |-- arm-linux-gnueabi-ld:vkms_formats.c:(.text):undefined-reference-to-__aeabi_ldivmod | |-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-imu_v11_0_3_program_rlc_ram | |-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead | |-- sound-soc-codecs-tas2562.c:warning:variable-ret-set-but-not-used | `-- vkms_formats.c:(.text):undefined-reference-to-__aeabi_ldivmod |-- arm-defconfig | `-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead |-- arm64-allyesconfig | |-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-imu_v11_0_3_program_rlc_ram | |-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead | `-- sound-soc-codecs-tas2562.c:warning:variable-ret-set-but-not-used |-- arm64-randconfig-r021-20220908 | `-- sound-soc-codecs-tas2562.c:warning:variable-ret-set-but-not-used |-- arm64-randconfig-r034-20220909 | `-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead |-- i386-allyesconfig | |-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-imu_v11_0_3_program_rlc_ram | |-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead | |-- ld:vkms_formats.c:(.text):undefined-reference-to-__divdi3 | |-- sound-soc-codecs-tas2562.c:warning:variable-ret-set-but-not-used | `-- vkms_formats.c:(.text):undefined-reference-to-__divdi3 |-- i386-defconfig | `-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead |-- i386-randconfig-a003 | |-- ERROR:__divdi3-drivers-gpu-drm-vkms-vkms.ko-undefined | |-- ERROR:__udivdi3-drivers-gpu-drm-vkms-vkms.ko-undefined | `-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead |-- i386-randconfig-a012 clang_recent_errors |-- arm64-randconfig-r006-20220908 | `-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-function-imu_v11_0_3_program_rlc_ram |-- i386-randconfig-a002 | `-- drivers-extcon-extcon-usbc-tusb320.c:warning:expecting-prototype-for-drivers-extcon-extcon-tusb320c().-Prototype-was-for-TUSB320_REG8()-instead |-- i386-randconfig-a006 | `-- ld.lld:error:undefined-symbol:__udivdi3 |-- i386-randconfig-a013 | `-- ld.lld:error:undefined-symbol:__udivdi3 |-- i386-randconfig-a015 | `-- drivers-extcon-extcon-usbc-tusb320.c:warning:expecting-prototype-for-drivers-extcon-extcon-tusb320c().-Prototype-was-for-TUSB320_REG8()-instead |-- riscv-randconfig-r001-20220908 | `-- ld.lld:error:undefined-symbol:__udivdi3 |-- riscv-randconfig-r013-20220907 | `-- drivers-extcon-extcon-usbc-tusb320.c:warning:expecting-prototype-for-drivers-extcon-extcon-tusb320c().-Prototype-was-for-TUSB320_REG8()-instead |-- x86_64-randconfig-a003 | `-- drivers-extcon-extcon-usbc-tusb320.c:warning:expecting-prototype-for-drivers-extcon-extcon-tusb320c().-Prototype-was-for-TUSB320_REG8()-instead |-- x86_64-randconfig-a012 | `-- drivers-extcon-extcon-usbc-tusb320.c:warning:expecting-prototype-for-drivers-extcon-extcon-tusb320c().-Prototype-was-for-TUSB320_REG8()-instead `-- x86_64-randconfig-a016 `-- drivers-extcon-extcon-usbc-tusb320.c:warning:expecting-prototype-for-drivers-extcon-extcon-tusb320c().-Prototype-was-for-TUSB320_REG8()-instead elapsed time: 729m configs tested: 75 configs skipped: 2 gcc tested configs: um i386_defconfig um x86_64_defconfig i386 randconfig-a001 i386 randconfig-a003 x86_64 defconfig i386 defconfig i386 randconfig-a005 m68k allmodconfig arc allyesconfig i386 allyesconfig arm defconfig x86_64 randconfig-a004 alpha allyesconfig x86_64 randconfig-a002 x86_64 rhel-8.3 m68k allyesconfig powerpc allnoconfig x86_64 allyesconfig x86_64 randconfig-a015 arm allyesconfig x86_64 randconfig-a006 arm64 allyesconfig x86_64 randconfig-a013 powerpc allmodconfig x86_64 randconfig-a011 sh allmodconfig ia64 allmodconfig mips allyesconfig mips loongson3_defconfig arc randconfig-r043-20220907 x86_64 rhel-8.3-func parisc alldefconfig x86_64 rhel-8.3-kunit sh migor_defconfig powerpc ppc40x_defconfig x86_64 rhel-8.3-kvm openrisc defconfig sh apsh4a3a_defconfig x86_64 rhel-8.3-kselftests x86_64 rhel-8.3-syz i386 randconfig-a014 alpha allnoconfig i386 randconfig-a012 riscv allnoconfig sh edosk7705_defconfig csky allnoconfig i386 randconfig-a016 sh shx3_defconfig arc allnoconfig ia64 bigsur_defconfig s390 zfcpdump_defconfig sh rsk7201_defconfig powerpc bamboo_defconfig i386 randconfig-c001 clang tested configs: i386 randconfig-a002 i386 randconfig-a004 i386 randconfig-a006 x86_64 randconfig-a005 x86_64 randconfig-a001 x86_64 randconfig-a014 x86_64 randconfig-a003 hexagon randconfig-r041-20220907 riscv randconfig-r042-20220907 x86_64 randconfig-a016 x86_64 randconfig-a012 hexagon randconfig-r045-20220907 s390 randconfig-r044-20220907 i386 randconfig-a013 i386 randconfig-a015 i386 randconfig-a011 mips ip22_defconfig mips ath79_defconfig powerpc ppa8548_defconfig x86_64 rhel-8.3-rust x86_64 randconfig-k001 -- 0-DAY CI Kernel Test Service https://01.org/lkp

1 0

build failure of next-20220909 due to e9e7df88996d ("ASoC: ak4458: Remove component probe() and remove()")
by Sudip Mukherjee (Codethink) 09 Sep '22

09 Sep '22

Hi All, The builds of loongarch, alpha, s390, csky and s390allmodconfig have failed to build next-20220909 with the error: sound/soc/codecs/ak4458.c:631:13: error: 'ak4458_reset' defined but not used [-Werror=unused-function] 631 | static void ak4458_reset(struct ak4458_priv *ak4458, bool active) | ^~~~~~~~~~~~ git bisect pointed to e9e7df88996d ("ASoC: ak4458: Remove component probe() and remove()") I will be happy to test any patch or provide any extra log if needed. -- Regards Sudip

1 0

[PATCH v2 1/3] slimbus: qcom-ngd: use correct error in message of pdr_add_lookup() failure
by Krzysztof Kozlowski 09 Sep '22

09 Sep '22

Use correct error code, instead of previous 'ret' value, when printing error from pdr_add_lookup() failure. Cc: <stable(a)vger.kernel.org> Fixes: e1ae85e1830e ("slimbus: qcom-ngd-ctrl: add Protection Domain Restart Support") Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Changes since v1: 1. Correct typo 2. Return 'ret' instead of again PTR_ERR --- drivers/slimbus/qcom-ngd-ctrl.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index 0aa8408464ad..f4f330b9fa72 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -1581,8 +1581,9 @@ static int qcom_slim_ngd_ctrl_probe(struct platform_device *pdev) pds = pdr_add_lookup(ctrl->pdr, "avs/audio", "msm/adsp/audio_pd"); if (IS_ERR(pds) && PTR_ERR(pds) != -EALREADY) { + ret = PTR_ERR(pds); dev_err(dev, "pdr add lookup failed: %d\n", ret); - return PTR_ERR(pds); + return ret; } platform_driver_register(&qcom_slim_ngd_driver); -- 2.34.1

2 3

po: Add Georgian translation
by GitHub pull_request - opened 09 Sep '22

09 Sep '22

alsa-project/alsa-utils pull request #169 was opened from NorwayFun: po: Add Georgian translation Request URL : https://github.com/alsa-project/alsa-utils/pull/169 Patch URL : https://github.com/alsa-project/alsa-utils/pull/169.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] sound: oss: dmasound: remove software_input_volume declaration
by Gaosheng Cui 09 Sep '22

09 Sep '22

expand_read_bal has been removed since commit fc37449f7959 ("The next round of scheduled OSS code removal"). software_input_volume has been removed since commit 0a1b42db4bf9 ("sound: sound/oss/dmasound/: cleanups"). so remove the declare for them from header file. Signed-off-by: Gaosheng Cui <cuigaosheng1(a)huawei.com> --- sound/oss/dmasound/dmasound.h | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/oss/dmasound/dmasound.h b/sound/oss/dmasound/dmasound.h index ad8ce6a1c25c..f065840c0efb 100644 --- a/sound/oss/dmasound/dmasound.h +++ b/sound/oss/dmasound/dmasound.h @@ -250,7 +250,4 @@ extern int dmasound_catchRadius; #define SW_INPUT_VOLUME_SCALE 4 #define SW_INPUT_VOLUME_DEFAULT (128 / SW_INPUT_VOLUME_SCALE) -extern int expand_read_bal; /* Balance factor for reading */ -extern uint software_input_volume; /* software implemented recording volume! */ - #endif /* _dmasound_h_ */ -- 2.25.1

2 1

[PATCH 0/2] Remove unused declarations
by Gaosheng Cui 09 Sep '22

09 Sep '22

This series contains a few cleanup patches, to remove unused declarations which have been removed. Thanks! Gaosheng Cui (2): ALSA: line6: remove line6_set_raw declaration ALSA: memalloc: remove snd_dma_sg_ops declaration sound/core/memalloc_local.h | 4 ---- sound/usb/line6/driver.h | 2 -- 2 files changed, 6 deletions(-) -- 2.25.1

2 3

[PATCH v9 1/2] ASoC: sunxi: Add Allwinner H6 Digital MIC driver
by Ban Tao 09 Sep '22

09 Sep '22

The Allwinner H6 and later SoCs have an DMIC block which is capable of capture. Signed-off-by: Ban Tao <fengzheng923(a)gmail.com> Tested-by: Samuel Holland <samuel(a)sholland.org> --- v1->v2: 1.Fix some compilation errors. 2.Modify some code styles. v2->v3: None. v3->v4: 1.add sig_bits. v4->v5: None. v5->v6: 1.Modify RXFIFO_CTL_MODE to mode 1. v6->v7: 1.Modify dmic_rate_s to be a global variable. 2.Changed some macro names to make more sense. 3.Align code format. 4.Add a depends on PM to Kconfig entry. v7->v8: None. v8->v9: 1.Modified Kconfig. 2.Added a comment. --- MAINTAINERS | 7 + sound/soc/sunxi/Kconfig | 7 + sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun50i-dmic.c | 406 ++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 421 insertions(+) create mode 100644 sound/soc/sunxi/sun50i-dmic.c diff --git a/MAINTAINERS b/MAINTAINERS index e9d5b05..839f625 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -796,6 +796,13 @@ L: linux-media(a)vger.kernel.org S: Maintained F: drivers/staging/media/sunxi/cedrus/ +ALLWINNER DMIC DRIVERS +M: Ban Tao <fengzheng923(a)gmail.com> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained +F: Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml +F: sound/soc/sunxi/sun50i-dmic.c + ALPHA PORT M: Richard Henderson <rth(a)twiddle.net> M: Ivan Kokshaysky <ink(a)jurassic.park.msu.ru> diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index ddcaaa9..1f18f01 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -56,6 +56,13 @@ config SND_SUN4I_SPDIF Say Y or M to add support for the S/PDIF audio block in the Allwinner A10 and affiliated SoCs. +config SND_SUN50I_DMIC + tristate "Allwinner H6 DMIC Support" + select SND_SOC_GENERIC_DMAENGINE_PCM + help + Say Y or M to add support for the DMIC audio block in the Allwinner + H6 and affiliated SoCs. + config SND_SUN8I_ADDA_PR_REGMAP tristate select REGMAP diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile index a86be34..4483fe9 100644 --- a/sound/soc/sunxi/Makefile +++ b/sound/soc/sunxi/Makefile @@ -6,3 +6,4 @@ obj-$(CONFIG_SND_SUN8I_CODEC_ANALOG) += sun8i-codec-analog.o obj-$(CONFIG_SND_SUN50I_CODEC_ANALOG) += sun50i-codec-analog.o obj-$(CONFIG_SND_SUN8I_CODEC) += sun8i-codec.o obj-$(CONFIG_SND_SUN8I_ADDA_PR_REGMAP) += sun8i-adda-pr-regmap.o +obj-$(CONFIG_SND_SUN50I_DMIC) += sun50i-dmic.o diff --git a/sound/soc/sunxi/sun50i-dmic.c b/sound/soc/sunxi/sun50i-dmic.c new file mode 100644 index 0000000..cd3c07f --- /dev/null +++ b/sound/soc/sunxi/sun50i-dmic.c @@ -0,0 +1,406 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +// +// This driver supports the DMIC in Allwinner's H6 SoCs. +// +// Copyright 2021 Ban Tao <fengzheng923(a)gmail.com> + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/of_device.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <linux/reset.h> +#include <sound/dmaengine_pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#define SUN50I_DMIC_EN_CTL (0x00) + #define SUN50I_DMIC_EN_CTL_GLOBE BIT(8) + #define SUN50I_DMIC_EN_CTL_CHAN(v) ((v) << 0) + #define SUN50I_DMIC_EN_CTL_CHAN_MASK GENMASK(7, 0) +#define SUN50I_DMIC_SR (0x04) + #define SUN50I_DMIC_SR_SAMPLE_RATE(v) ((v) << 0) + #define SUN50I_DMIC_SR_SAMPLE_RATE_MASK GENMASK(2, 0) +#define SUN50I_DMIC_CTL (0x08) + #define SUN50I_DMIC_CTL_OVERSAMPLE_RATE BIT(0) +#define SUN50I_DMIC_DATA (0x10) +#define SUN50I_DMIC_INTC (0x14) + #define SUN50I_DMIC_FIFO_DRQ_EN BIT(2) +#define SUN50I_DMIC_INT_STA (0x18) + #define SUN50I_DMIC_INT_STA_OVERRUN_IRQ_PENDING BIT(1) + #define SUN50I_DMIC_INT_STA_DATA_IRQ_PENDING BIT(0) +#define SUN50I_DMIC_RXFIFO_CTL (0x1c) + #define SUN50I_DMIC_RXFIFO_CTL_FLUSH BIT(31) + #define SUN50I_DMIC_RXFIFO_CTL_MODE_MASK BIT(9) + #define SUN50I_DMIC_RXFIFO_CTL_MODE_LSB (0 << 9) + #define SUN50I_DMIC_RXFIFO_CTL_MODE_MSB (1 << 9) + #define SUN50I_DMIC_RXFIFO_CTL_SAMPLE_MASK BIT(8) + #define SUN50I_DMIC_RXFIFO_CTL_SAMPLE_16 (0 << 8) + #define SUN50I_DMIC_RXFIFO_CTL_SAMPLE_24 (1 << 8) +#define SUN50I_DMIC_CH_NUM (0x24) + #define SUN50I_DMIC_CH_NUM_N(v) ((v) << 0) + #define SUN50I_DMIC_CH_NUM_N_MASK GENMASK(2, 0) +#define SUN50I_DMIC_CNT (0x2c) + #define SUN50I_DMIC_CNT_N (1 << 0) +#define SUN50I_DMIC_HPF_CTRL (0x38) +#define SUN50I_DMIC_VERSION (0x50) + +struct sun50i_dmic_dev { + struct clk *dmic_clk; + struct clk *bus_clk; + struct reset_control *rst; + struct regmap *regmap; + struct snd_dmaengine_dai_dma_data dma_params_rx; +}; + +struct dmic_rate { + unsigned int samplerate; + unsigned int rate_bit; +}; + +const static struct dmic_rate dmic_rate_s[] = { + {48000, 0x0}, + {44100, 0x0}, + {32000, 0x1}, + {24000, 0x2}, + {22050, 0x2}, + {16000, 0x3}, + {12000, 0x4}, + {11025, 0x4}, + {8000, 0x5}, +}; + +static int sun50i_dmic_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); + + /* only support capture */ + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_FLUSH, + SUN50I_DMIC_RXFIFO_CTL_FLUSH); + regmap_write(host->regmap, SUN50I_DMIC_CNT, SUN50I_DMIC_CNT_N); + + return 0; +} + +static int sun50i_dmic_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + int i = 0; + unsigned long rate = params_rate(params); + unsigned int mclk = 0; + unsigned int channels = params_channels(params); + unsigned int chan_en = (1 << channels) - 1; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + + /* DMIC num is N+1 */ + regmap_update_bits(host->regmap, SUN50I_DMIC_CH_NUM, + SUN50I_DMIC_CH_NUM_N_MASK, + SUN50I_DMIC_CH_NUM_N(channels - 1)); + regmap_write(host->regmap, SUN50I_DMIC_HPF_CTRL, chan_en); + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_CHAN_MASK, + SUN50I_DMIC_EN_CTL_CHAN(chan_en)); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_SAMPLE_MASK, + SUN50I_DMIC_RXFIFO_CTL_SAMPLE_16); + break; + case SNDRV_PCM_FORMAT_S24_LE: + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_SAMPLE_MASK, + SUN50I_DMIC_RXFIFO_CTL_SAMPLE_24); + break; + default: + dev_err(cpu_dai->dev, "Invalid format!\n"); + return -EINVAL; + } + /* The hardware supports FIFO mode 1 for 24-bit samples */ + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_MODE_MASK, + SUN50I_DMIC_RXFIFO_CTL_MODE_MSB); + + switch (rate) { + case 11025: + case 22050: + case 44100: + mclk = 22579200; + break; + case 8000: + case 12000: + case 16000: + case 24000: + case 32000: + case 48000: + mclk = 24576000; + break; + default: + dev_err(cpu_dai->dev, "Invalid rate!\n"); + return -EINVAL; + } + + if (clk_set_rate(host->dmic_clk, mclk)) { + dev_err(cpu_dai->dev, "mclk : %u not support\n", mclk); + return -EINVAL; + } + + for (i = 0; i < ARRAY_SIZE(dmic_rate_s); i++) { + if (dmic_rate_s[i].samplerate == rate) { + regmap_update_bits(host->regmap, SUN50I_DMIC_SR, + SUN50I_DMIC_SR_SAMPLE_RATE_MASK, + SUN50I_DMIC_SR_SAMPLE_RATE(dmic_rate_s[i].rate_bit)); + break; + } + } + + switch (params_physical_width(params)) { + case 16: + host->dma_params_rx.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + break; + case 32: + host->dma_params_rx.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + break; + default: + dev_err(cpu_dai->dev, "Unsupported physical sample width: %d\n", + params_physical_width(params)); + return -EINVAL; + } + + /* oversamplerate adjust */ + if (params_rate(params) >= 24000) + regmap_update_bits(host->regmap, SUN50I_DMIC_CTL, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE); + else + regmap_update_bits(host->regmap, SUN50I_DMIC_CTL, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE, 0); + + return 0; +} + +static int sun50i_dmic_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + int ret = 0; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* DRQ ENABLE */ + regmap_update_bits(host->regmap, SUN50I_DMIC_INTC, + SUN50I_DMIC_FIFO_DRQ_EN, + SUN50I_DMIC_FIFO_DRQ_EN); + /* Global enable */ + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_GLOBE, + SUN50I_DMIC_EN_CTL_GLOBE); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + /* DRQ DISABLE */ + regmap_update_bits(host->regmap, SUN50I_DMIC_INTC, + SUN50I_DMIC_FIFO_DRQ_EN, 0); + /* Global disable */ + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_GLOBE, 0); + break; + default: + ret = -EINVAL; + break; + } + return ret; +} + +static int sun50i_dmic_soc_dai_probe(struct snd_soc_dai *dai) +{ + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, NULL, &host->dma_params_rx); + + return 0; +} + +static const struct snd_soc_dai_ops sun50i_dmic_dai_ops = { + .startup = sun50i_dmic_startup, + .trigger = sun50i_dmic_trigger, + .hw_params = sun50i_dmic_hw_params, +}; + +static const struct regmap_config sun50i_dmic_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = SUN50I_DMIC_VERSION, + .cache_type = REGCACHE_NONE, +}; + +#define SUN50I_DMIC_RATES (SNDRV_PCM_RATE_8000_48000) +#define SUN50I_DMIC_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_driver sun50i_dmic_dai = { + .capture = { + .channels_min = 1, + .channels_max = 8, + .rates = SUN50I_DMIC_RATES, + .formats = SUN50I_DMIC_FORMATS, + .sig_bits = 21, + }, + .probe = sun50i_dmic_soc_dai_probe, + .ops = &sun50i_dmic_dai_ops, + .name = "dmic", +}; + +static const struct of_device_id sun50i_dmic_of_match[] = { + { + .compatible = "allwinner,sun50i-h6-dmic", + }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, sun50i_dmic_of_match); + +static const struct snd_soc_component_driver sun50i_dmic_component = { + .name = "sun50i-dmic", +}; + +static int sun50i_dmic_runtime_suspend(struct device *dev) +{ + struct sun50i_dmic_dev *host = dev_get_drvdata(dev); + + clk_disable_unprepare(host->dmic_clk); + clk_disable_unprepare(host->bus_clk); + + return 0; +} + +static int sun50i_dmic_runtime_resume(struct device *dev) +{ + struct sun50i_dmic_dev *host = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(host->dmic_clk); + if (ret) + return ret; + + ret = clk_prepare_enable(host->bus_clk); + if (ret) { + clk_disable_unprepare(host->dmic_clk); + return ret; + } + + return 0; +} + +static int sun50i_dmic_probe(struct platform_device *pdev) +{ + struct sun50i_dmic_dev *host; + struct resource *res; + int ret; + void __iomem *base; + + host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); + if (!host) + return -ENOMEM; + + /* Get the addresses */ + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) + return dev_err_probe(&pdev->dev, PTR_ERR(base), + "get resource failed.\n"); + + host->regmap = devm_regmap_init_mmio(&pdev->dev, base, + &sun50i_dmic_regmap_config); + + /* Clocks */ + host->bus_clk = devm_clk_get(&pdev->dev, "bus"); + if (IS_ERR(host->bus_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->bus_clk), + "failed to get bus clock.\n"); + + host->dmic_clk = devm_clk_get(&pdev->dev, "mod"); + if (IS_ERR(host->dmic_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->dmic_clk), + "failed to get dmic clock.\n"); + + host->dma_params_rx.addr = res->start + SUN50I_DMIC_DATA; + host->dma_params_rx.maxburst = 8; + + platform_set_drvdata(pdev, host); + + host->rst = devm_reset_control_get_optional_exclusive(&pdev->dev, NULL); + if (IS_ERR(host->rst)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->rst), + "Failed to get reset.\n"); + reset_control_deassert(host->rst); + + ret = devm_snd_soc_register_component(&pdev->dev, &sun50i_dmic_component, + &sun50i_dmic_dai, 1); + if (ret) + return dev_err_probe(&pdev->dev, ret, + "failed to register component.\n"); + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = sun50i_dmic_runtime_resume(&pdev->dev); + if (ret) + goto err_disable_runtime_pm; + } + + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) + goto err_suspend; + + return 0; +err_suspend: + if (!pm_runtime_status_suspended(&pdev->dev)) + sun50i_dmic_runtime_suspend(&pdev->dev); +err_disable_runtime_pm: + pm_runtime_disable(&pdev->dev); + return ret; +} + +static int sun50i_dmic_remove(struct platform_device *pdev) +{ + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + sun50i_dmic_runtime_suspend(&pdev->dev); + + return 0; +} + +static const struct dev_pm_ops sun50i_dmic_pm = { + SET_RUNTIME_PM_OPS(sun50i_dmic_runtime_suspend, + sun50i_dmic_runtime_resume, NULL) +}; + +static struct platform_driver sun50i_dmic_driver = { + .driver = { + .name = "sun50i-dmic", + .of_match_table = of_match_ptr(sun50i_dmic_of_match), + .pm = &sun50i_dmic_pm, + }, + .probe = sun50i_dmic_probe, + .remove = sun50i_dmic_remove, +}; + +module_platform_driver(sun50i_dmic_driver); + +MODULE_DESCRIPTION("Allwinner sun50i DMIC SoC Interface"); +MODULE_AUTHOR("Ban Tao <fengzheng923(a)gmail.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:sun50i-dmic"); -- 2.7.4

2 1

[linux-next:master] BUILD REGRESSION 47c191411b68a771261be3dc0bd6f68394cef358
by kernel test robot 09 Sep '22

09 Sep '22

tree/branch: https://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git master branch HEAD: 47c191411b68a771261be3dc0bd6f68394cef358 Add linux-next specific files for 20220908 Error/Warning reports: https://lore.kernel.org/linux-mm/202209042337.FQi69rLV-lkp@intel.com https://lore.kernel.org/linux-mm/202209080718.y5QmlNKH-lkp@intel.com https://lore.kernel.org/llvm/202209090343.JPAFJt74-lkp@intel.com Error/Warning: (recently discovered and may have been fixed) ERROR: modpost: "__divdi3" [drivers/gpu/drm/vkms/vkms.ko] undefined! ERROR: modpost: "__udivdi3" [drivers/gpu/drm/vkms/vkms.ko] undefined! arm-linux-gnueabi-ld: vkms_formats.c:(.text+0x824): undefined reference to `__aeabi_ldivmod' drivers/base/regmap/regmap-mmio.c:222:17: error: implicit declaration of function 'writesb'; did you mean 'writeb'? [-Werror=implicit-function-declaration] drivers/base/regmap/regmap-mmio.c:225:17: error: implicit declaration of function 'writesw'; did you mean 'writew'? [-Werror=implicit-function-declaration] drivers/base/regmap/regmap-mmio.c:228:17: error: implicit declaration of function 'writesl'; did you mean 'writel'? [-Werror=implicit-function-declaration] drivers/base/regmap/regmap-mmio.c:232:17: error: implicit declaration of function 'writesq'; did you mean 'writeq'? [-Werror=implicit-function-declaration] drivers/base/regmap/regmap-mmio.c:232:17: error: implicit declaration of function 'writesq'; did you mean 'writesl'? [-Werror=implicit-function-declaration] drivers/base/regmap/regmap-mmio.c:358:17: error: implicit declaration of function 'readsb'; did you mean 'readb'? [-Werror=implicit-function-declaration] drivers/base/regmap/regmap-mmio.c:361:17: error: implicit declaration of function 'readsw'; did you mean 'readw'? [-Werror=implicit-function-declaration] drivers/base/regmap/regmap-mmio.c:364:17: error: implicit declaration of function 'readsl'; did you mean 'readl'? [-Werror=implicit-function-declaration] drivers/base/regmap/regmap-mmio.c:368:17: error: implicit declaration of function 'readsq'; did you mean 'readq'? [-Werror=implicit-function-declaration] drivers/base/regmap/regmap-mmio.c:368:17: error: implicit declaration of function 'readsq'; did you mean 'readsl'? [-Werror=implicit-function-declaration] drivers/crypto/aspeed/aspeed-hace.c:133 aspeed_hace_probe() warn: platform_get_irq() does not return zero drivers/gpu/drm/amd/amdgpu/imu_v11_0_3.c:139:6: warning: no previous prototype for 'imu_v11_0_3_program_rlc_ram' [-Wmissing-prototypes] drivers/gpu/drm/drm_atomic_helper.c:802: warning: expecting prototype for drm_atomic_helper_check_wb_connector_state(). Prototype was for drm_atomic_helper_check_wb_encoder_state() instead drivers/gpu/drm/vkms/vkms_formats.c:259: undefined reference to `__divdi3' drivers/pinctrl/pinctrl-amd.c:288 amd_gpio_dbg_show() warn: format string contains non-ascii character '\x9a' drivers/pinctrl/pinctrl-amd.c:288 amd_gpio_dbg_show() warn: format string contains non-ascii character '\xa1' drivers/pinctrl/pinctrl-amd.c:370 amd_gpio_dbg_show() warn: format string contains non-ascii character '\x95' drivers/scsi/qla2xxx/qla_os.c:2854:23: warning: assignment to 'struct trace_array *' from 'int' makes pointer from integer without a cast [-Wint-conversion] drivers/scsi/qla2xxx/qla_os.c:2854:25: error: implicit declaration of function 'trace_array_get_by_name'; did you mean 'trace_array_set_clr_event'? [-Werror=implicit-function-declaration] drivers/scsi/qla2xxx/qla_os.c:2869:9: error: implicit declaration of function 'trace_array_put' [-Werror=implicit-function-declaration] fs/btrfs/volumes.c:6549 __btrfs_map_block() error: we previously assumed 'mirror_num_ret' could be null (see line 6376) ld: drivers/gpu/drm/vkms/vkms_formats.c:260: undefined reference to `__divdi3' ld: vkms_formats.c:(.text+0x362): undefined reference to `__divdi3' ld: vkms_formats.c:(.text+0x3b2): undefined reference to `__divdi3' ld: vkms_formats.c:(.text+0x3ba): undefined reference to `__divdi3' ld: vkms_formats.c:(.text+0x47f): undefined reference to `__divdi3' mips-linux-ld: vkms_formats.c:(.text.argb_u16_to_RGB565+0xd0): undefined reference to `__divdi3' mm/kasan/kasan_test_module.c:90:26: sparse: struct kasan_rcu_info * mm/kasan/kasan_test_module.c:90:26: sparse: struct kasan_rcu_info [noderef] __rcu * sound/soc/codecs/tas2562.c:442:13: warning: variable 'ret' set but not used [-Wunused-but-set-variable] vkms_formats.c:(.text+0x266): undefined reference to `__divdi3' vkms_formats.c:(.text+0x338): undefined reference to `__divdi3' vkms_formats.c:(.text+0x388): undefined reference to `__divdi3' vkms_formats.c:(.text+0x390): undefined reference to `__divdi3' vkms_formats.c:(.text+0x455): undefined reference to `__divdi3' vkms_formats.c:(.text+0x804): undefined reference to `__aeabi_ldivmod' vkms_formats.c:(.text.argb_u16_to_RGB565+0xb0): undefined reference to `__divdi3' Error/Warning ids grouped by kconfigs: gcc_recent_errors |-- alpha-allyesconfig | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-readsb | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-readsl | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-readsq | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-readsw | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-writesb | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-writesl | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-writesq | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-writesw | |-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-imu_v11_0_3_program_rlc_ram | |-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead | |-- drivers-scsi-qla2xxx-qla_os.c:error:implicit-declaration-of-function-trace_array_get_by_name | |-- drivers-scsi-qla2xxx-qla_os.c:error:implicit-declaration-of-function-trace_array_put | |-- drivers-scsi-qla2xxx-qla_os.c:warning:assignment-to-struct-trace_array-from-int-makes-pointer-from-integer-without-a-cast | `-- sound-soc-codecs-tas2562.c:warning:variable-ret-set-but-not-used |-- alpha-randconfig-r013-20220907 | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-readsb | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-readsl | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-readsq | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-readsw | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-writesb | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-writesl | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-writesq | `-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-writesw |-- alpha-randconfig-r034-20220907 | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-readsb | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-readsl | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-readsq | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-readsw | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-writesb | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-writesl | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-writesq | `-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-writesw |-- arc-allyesconfig | |-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-imu_v11_0_3_program_rlc_ram | |-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead | `-- sound-soc-codecs-tas2562.c:warning:variable-ret-set-but-not-used |-- arc-randconfig-r003-20220907 | `-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead |-- arc-randconfig-r026-20220907 | |-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead | `-- sound-soc-codecs-tas2562.c:warning:variable-ret-set-but-not-used |-- arc-randconfig-s033-20220907 | |-- kernel-bpf-hashtab.c:sparse:sparse:cast-removes-address-space-__percpu-of-expression | |-- kernel-bpf-hashtab.c:sparse:sparse:incorrect-type-in-assignment-(different-address-spaces)-expected-void-noderef-__percpu-assigned-pptr-got-void | |-- kernel-bpf-hashtab.c:sparse:sparse:incorrect-type-in-assignment-(different-address-spaces)-expected-void-ptr_to_pptr-got-void-noderef-__percpu-assigned-pptr | |-- kernel-bpf-memalloc.c:sparse:sparse:incorrect-type-in-argument-(different-address-spaces)-expected-void-noderef-__percpu-__pdata-got-void | |-- kernel-bpf-memalloc.c:sparse:sparse:incorrect-type-in-argument-(different-address-spaces)-expected-void-noderef-__percpu-__pdata-got-void-pptr | |-- kernel-bpf-memalloc.c:sparse:sparse:incorrect-type-in-initializer-(different-address-spaces)-expected-void-pptr-got-void-noderef-__percpu | `-- kernel-exit.c:sparse:sparse:incorrect-type-in-initializer-(different-address-spaces)-expected-struct-sighand_struct-sighand-got-struct-sighand_struct-noderef-__rcu-sighand clang_recent_errors |-- i386-randconfig-a002 | `-- drivers-extcon-extcon-usbc-tusb320.c:warning:expecting-prototype-for-drivers-extcon-extcon-tusb320c().-Prototype-was-for-TUSB320_REG8()-instead |-- i386-randconfig-a006 | `-- ld.lld:error:undefined-symbol:__udivdi3 |-- i386-randconfig-a013 | `-- ld.lld:error:undefined-symbol:__udivdi3 |-- i386-randconfig-a015 | `-- drivers-extcon-extcon-usbc-tusb320.c:warning:expecting-prototype-for-drivers-extcon-extcon-tusb320c().-Prototype-was-for-TUSB320_REG8()-instead |-- powerpc-randconfig-r021-20220907 | |-- arch-powerpc-math-emu-fre.c:warning:no-previous-prototype-for-function-fre | |-- arch-powerpc-math-emu-frsqrtes.c:warning:no-previous-prototype-for-function-frsqrtes | |-- arch-powerpc-math-emu-fsqrt.c:warning:no-previous-prototype-for-function-fsqrt | |-- arch-powerpc-math-emu-fsqrts.c:warning:no-previous-prototype-for-function-fsqrts | |-- arch-powerpc-math-emu-mtfsf.c:warning:no-previous-prototype-for-function-mtfsf | `-- arch-powerpc-math-emu-mtfsfi.c:warning:no-previous-prototype-for-function-mtfsfi |-- x86_64-randconfig-a003 | `-- drivers-extcon-extcon-usbc-tusb320.c:warning:expecting-prototype-for-drivers-extcon-extcon-tusb320c().-Prototype-was-for-TUSB320_REG8()-instead |-- x86_64-randconfig-a012 | `-- drivers-extcon-extcon-usbc-tusb320.c:warning:expecting-prototype-for-drivers-extcon-extcon-tusb320c().-Prototype-was-for-TUSB320_REG8()-instead `-- x86_64-randconfig-a016 `-- drivers-extcon-extcon-usbc-tusb320.c:warning:expecting-prototype-for-drivers-extcon-extcon-tusb320c().-Prototype-was-for-TUSB320_REG8()-instead elapsed time: 734m configs tested: 75 configs skipped: 4 gcc tested configs: um i386_defconfig um x86_64_defconfig m68k allmodconfig x86_64 randconfig-a011 powerpc allnoconfig arc allyesconfig powerpc allmodconfig alpha allyesconfig mips allyesconfig m68k allyesconfig x86_64 randconfig-a015 i386 randconfig-a014 sh allmodconfig x86_64 randconfig-a002 x86_64 defconfig x86_64 randconfig-a013 i386 defconfig i386 randconfig-a001 x86_64 rhel-8.3-func i386 randconfig-a003 x86_64 randconfig-a006 arc randconfig-r043-20220908 x86_64 randconfig-a004 i386 randconfig-a005 arm defconfig i386 randconfig-a012 i386 randconfig-a016 x86_64 rhel-8.3-kunit x86_64 rhel-8.3-kvm arm allyesconfig x86_64 rhel-8.3-kselftests x86_64 rhel-8.3-syz x86_64 rhel-8.3 i386 allyesconfig arm64 allyesconfig arc randconfig-r043-20220907 i386 randconfig-c001 s390 randconfig-r044-20220908 riscv randconfig-r042-20220908 x86_64 allyesconfig ia64 allmodconfig csky allnoconfig arc allnoconfig alpha allnoconfig riscv allnoconfig m68k m5275evb_defconfig sh ap325rxa_defconfig arm cerfcube_defconfig powerpc wii_defconfig xtensa cadence_csp_defconfig arm mvebu_v7_defconfig clang tested configs: x86_64 randconfig-a012 i386 randconfig-a013 x86_64 randconfig-a014 x86_64 randconfig-a001 i386 randconfig-a002 x86_64 randconfig-a003 x86_64 randconfig-a016 i386 randconfig-a011 x86_64 randconfig-a005 hexagon randconfig-r041-20220907 i386 randconfig-a015 hexagon randconfig-r041-20220908 i386 randconfig-a006 i386 randconfig-a004 riscv randconfig-r042-20220907 hexagon randconfig-r045-20220908 hexagon randconfig-r045-20220907 s390 randconfig-r044-20220907 x86_64 rhel-8.3-rust powerpc icon_defconfig arm spear13xx_defconfig arm palmz72_defconfig powerpc mpc832x_rdb_defconfig x86_64 randconfig-k001 -- 0-DAY CI Kernel Test Service https://01.org/lkp

1 0

[PATCH v2] ASoC: fsl_asrc: Add initialization finishing check in runtime resume
by Shengjiu Wang 08 Sep '22

08 Sep '22

If the initialization is not finished, then filling input data to the FIFO may fail. So it is better to add initialization finishing check in the runtime resume for suspend & resume case. And consider the case of three instances working in parallel, increase the retry times to 50 for more initialization time. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2: - update comments. sound/soc/fsl/fsl_asrc.c | 22 +++++++++++++++++++++- 1 file changed, 21 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index aa5edf32d988..b394b762025d 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -20,6 +20,7 @@ #define IDEAL_RATIO_DECIMAL_DEPTH 26 #define DIVIDER_NUM 64 +#define INIT_TRY_NUM 50 #define pair_err(fmt, ...) \ dev_err(&asrc->pdev->dev, "Pair %c: " fmt, 'A' + index, ##__VA_ARGS__) @@ -579,7 +580,7 @@ static void fsl_asrc_start_pair(struct fsl_asrc_pair *pair) { struct fsl_asrc *asrc = pair->asrc; enum asrc_pair_index index = pair->index; - int reg, retry = 10, i; + int reg, retry = INIT_TRY_NUM, i; /* Enable the current pair */ regmap_update_bits(asrc->regmap, REG_ASRCTR, @@ -592,6 +593,10 @@ static void fsl_asrc_start_pair(struct fsl_asrc_pair *pair) reg &= ASRCFG_INIRQi_MASK(index); } while (!reg && --retry); + /* NOTE: Doesn't treat initialization timeout as error */ + if (!retry) + dev_warn(&asrc->pdev->dev, "initialization isn't finished\n"); + /* Make the input fifo to ASRC STALL level */ regmap_read(asrc->regmap, REG_ASRCNCR, &reg); for (i = 0; i < pair->channels * 4; i++) @@ -1257,6 +1262,7 @@ static int fsl_asrc_runtime_resume(struct device *dev) { struct fsl_asrc *asrc = dev_get_drvdata(dev); struct fsl_asrc_priv *asrc_priv = asrc->private; + int reg, retry = INIT_TRY_NUM; int i, ret; u32 asrctr; @@ -1295,6 +1301,20 @@ static int fsl_asrc_runtime_resume(struct device *dev) regmap_update_bits(asrc->regmap, REG_ASRCTR, ASRCTR_ASRCEi_ALL_MASK, asrctr); + /* Wait for status of initialization for every enabled pairs */ + do { + udelay(5); + regmap_read(asrc->regmap, REG_ASRCFG, &reg); + reg = (reg >> ASRCFG_INIRQi_SHIFT(0)) & 0x7; + } while ((reg != ((asrctr >> ASRCTR_ASRCEi_SHIFT(0)) & 0x7)) && --retry); + + /* + * NOTE: Doesn't treat initialization timeout as error + * Some of pair maybe success, then still can continue. + */ + if (!retry) + dev_warn(dev, "initialization isn't finished\n"); + return 0; disable_asrck_clk: -- 2.34.1

2 1

Re: [PATCH 09/12] ASoC: cs42l42: Split I2C identity into separate module
by Martin Povišer 08 Sep '22

08 Sep '22

> On 19. 8. 2022, at 14:52, Richard Fitzgerald <rf(a)opensource.cirrus.com> wrote: > > Split the I2C bus driver definition and probe()/remove() into a > separate module so that a Soundwire build of CS42L42 support does > not have a spurious dependency on I2C. > > Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> > --- > +static int cs42l42_i2c_probe(struct i2c_client *i2c_client) > +{ > + struct device *dev = &i2c_client->dev; > + struct cs42l42_private *cs42l42; > + struct regmap *regmap; > + int ret; > + > + cs42l42 = devm_kzalloc(dev, sizeof(*cs42l42), GFP_KERNEL); > + if (!cs42l42) > + return -ENOMEM; > + > + regmap = devm_regmap_init_i2c(i2c_client, &cs42l42_regmap); > + if (IS_ERR(regmap)) { > + ret = PTR_ERR(regmap); > + dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); > + return ret; > + } > + > + cs42l42->dev = dev; > + cs42l42->regmap = regmap; > + cs42l42->irq = i2c_client->irq; > + > + ret = cs42l42_common_probe(cs42l42, &cs42l42_soc_component, &cs42l42_dai); > + if (ret) > + return ret; > + > + return cs42l42_init(cs42l42); The common_remove should probably be called here if the init fails. Best, Martin > +} > + > +static int cs42l42_i2c_remove(struct i2c_client *i2c_client) > +{ > + struct cs42l42_private *cs42l42 = dev_get_drvdata(&i2c_client->dev); > + > + cs42l42_common_remove(cs42l42); > + > + return 0; > +}

1 0

[PATCH v3 0/4] ASoC: qcom: add support for sc8280xp machine
by Srinivas Kandagatla 08 Sep '22

08 Sep '22

This patchset adds support for SC8280XP SoC machine driver. First patch moves some of the commonly used code to common from sm8250 machine driver and the follow on code adds minimal support for sc8280xp. Currently this driver is only tested with SmartSpeakers and Headset on Lenovo Thinkpad X13s. Support for sm8450 is tested and I will post the patches soon. Thanks, Srini Changes since v2: - put SoundWire specific symbols under IS_ENABLED check as not all Qcom plaforms need SoundWire that use common.c Srinivas Kandagatla (4): ASoC: qcom: common: use EXPORT_SYMBOL_GPL instead of EXPORT_SYMBOL ASoC: dt-bindings: qcom,sm8250: add compatibles for sm8450 and sm8250 ASoC: qcom: sm8250: move some code to common ASoC: qcom: add machine driver for sc8280xp .../bindings/sound/qcom,sm8250.yaml | 2 + sound/soc/qcom/Kconfig | 11 ++ sound/soc/qcom/Makefile | 2 + sound/soc/qcom/common.c | 173 +++++++++++++++++- sound/soc/qcom/common.h | 35 ++++ sound/soc/qcom/sc8280xp.c | 157 ++++++++++++++++ sound/soc/qcom/sm8250.c | 152 +-------------- 7 files changed, 386 insertions(+), 146 deletions(-) create mode 100644 sound/soc/qcom/sc8280xp.c -- 2.21.0

2 9

[PATCH v2 0/4] ASoC: qcom: add support for sc8280xp machine
by Srinivas Kandagatla 08 Sep '22

08 Sep '22

This patchset adds support for SC8280XP SoC machine driver. First patch moves some of the commonly used code to common from sm8250 machine driver and the follow on code adds minimal support for sc8280xp. Currently this driver is only tested with SmartSpeakers and Headset on Lenovo Thinkpad X13s. Support for sm8450 is tested and I will post the patches soon. Thanks, Srini Changes since v1: - make all the exported symbol as EXPORT_SYMBOL_GPL Srinivas Kandagatla (4): ASoC: qcom: common: use EXPORT_SYMBOL_GPL instead of EXPORT_SYMBOL ASoC: dt-bindings: qcom,sm8250: add compatibles for sm8450 and sm8250 ASoC: qcom: sm8250: move some code to common ASoC: qcom: add machine driver for sc8280xp .../bindings/sound/qcom,sm8250.yaml | 2 + sound/soc/qcom/Kconfig | 11 ++ sound/soc/qcom/Makefile | 2 + sound/soc/qcom/common.c | 171 +++++++++++++++++- sound/soc/qcom/common.h | 12 ++ sound/soc/qcom/sc8280xp.c | 157 ++++++++++++++++ sound/soc/qcom/sm8250.c | 152 +--------------- 7 files changed, 361 insertions(+), 146 deletions(-) create mode 100644 sound/soc/qcom/sc8280xp.c -- 2.21.0

2 7

About ALSA dummy module: support customized mixer volume leveling
by YJ Lee 08 Sep '22

08 Sep '22

Hello Takashi (and the alsa community), This is YJ Lee from chromium.org. I'm thinking about extending the ALSA dummy module to support customized volume leveling. It will be very useful to test devices with different volume granularity. Currently this module is using hard-coded mixer volume leveling, from min=-50 to max=100. See L716: https://git.kernel.org/pub/scm/linux/kernel/git/stable/linux-stable-rc.git/… My plan (A) is to expose 2 more additional module parameters and replace a few occurrences, from L742-750. Another plan (B) is to leverage the current existing module parameter:model, and make the dummy-module to be able to accept models with customized volume leveling. New dummy models can be provided as configuration files, and be used in the same way as how currently existing dummy models (L146-213) are used. However, it's a bit over-engineering from the original purpose. What do you think about these plans? Any insight you (and the alsa community) can provide will be greatly appreciated. Best, YJ

2 4

[PATCH 0/2] ASoC: soc-dapm.c: random cleanup retry
by Kuninori Morimoto 08 Sep '22

08 Sep '22

Hi Mark These are remains of my previous cleanup patch-set. [1/2] adds comment why kzalloc()/kfree() are needed on snd_soc_dai_link_event_pre_pmu(). [2/2] is adjusted to [1/1] patch. Link: https://lore.kernel.org/all/8735d59zt9.wl-kuninori.morimoto.gx@renesas.com/ Kuninori Morimoto (2): ASoC: soc-dapm.c: add comment for kzalloc()/kfree() on snd_soc_dai_link_event_pre_pmu() ASoC: soc-dapm.c: tidyup snd_soc_dai_link_event_pre_pmu() sound/soc/soc-dapm.c | 20 +++++++++++++++----- 1 file changed, 15 insertions(+), 5 deletions(-) -- 2.25.1

2 3

[PATCH] ASoC: ak4458: Remove component probe() and remove()
by Shengjiu Wang 08 Sep '22

08 Sep '22

Most function in ak4458_probe() and ak4458_remove() are duplicate with dai ops, so remove them and move dsd_path setting to dai ops. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/ak4458.c | 53 ++++++--------------------------------- 1 file changed, 7 insertions(+), 46 deletions(-) diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c index ea33cc83c86c..b534212096ee 100644 --- a/sound/soc/codecs/ak4458.c +++ b/sound/soc/codecs/ak4458.c @@ -447,6 +447,13 @@ static int ak4458_hw_params(struct snd_pcm_substream *substream, snd_soc_component_update_bits(component, AK4458_0B_CONTROL7, AK4458_DCHAIN_MASK, dchn); + if (ak4458->drvdata->type == AK4497) { + ret = snd_soc_component_update_bits(component, AK4458_09_DSD2, + 0x4, (ak4458->dsd_path << 2)); + if (ret < 0) + return ret; + } + ret = ak4458_rstn_control(component, 0); if (ret) return ret; @@ -629,48 +636,6 @@ static void ak4458_reset(struct ak4458_priv *ak4458, bool active) } } -static int ak4458_init(struct snd_soc_component *component) -{ - struct ak4458_priv *ak4458 = snd_soc_component_get_drvdata(component); - int ret; - - /* External Mute ON */ - if (ak4458->mute_gpiod) - gpiod_set_value_cansleep(ak4458->mute_gpiod, 1); - - ak4458_reset(ak4458, false); - - ret = snd_soc_component_update_bits(component, AK4458_00_CONTROL1, - 0x80, 0x80); /* ACKS bit = 1; 10000000 */ - if (ret < 0) - return ret; - - if (ak4458->drvdata->type == AK4497) { - ret = snd_soc_component_update_bits(component, AK4458_09_DSD2, - 0x4, (ak4458->dsd_path << 2)); - if (ret < 0) - return ret; - } - - return ak4458_rstn_control(component, 1); -} - -static int ak4458_probe(struct snd_soc_component *component) -{ - struct ak4458_priv *ak4458 = snd_soc_component_get_drvdata(component); - - ak4458->fs = 48000; - - return ak4458_init(component); -} - -static void ak4458_remove(struct snd_soc_component *component) -{ - struct ak4458_priv *ak4458 = snd_soc_component_get_drvdata(component); - - ak4458_reset(ak4458, true); -} - #ifdef CONFIG_PM static int __maybe_unused ak4458_runtime_suspend(struct device *dev) { @@ -714,8 +679,6 @@ static int __maybe_unused ak4458_runtime_resume(struct device *dev) #endif /* CONFIG_PM */ static const struct snd_soc_component_driver soc_codec_dev_ak4458 = { - .probe = ak4458_probe, - .remove = ak4458_remove, .controls = ak4458_snd_controls, .num_controls = ARRAY_SIZE(ak4458_snd_controls), .dapm_widgets = ak4458_dapm_widgets, @@ -728,8 +691,6 @@ static const struct snd_soc_component_driver soc_codec_dev_ak4458 = { }; static const struct snd_soc_component_driver soc_codec_dev_ak4497 = { - .probe = ak4458_probe, - .remove = ak4458_remove, .controls = ak4497_snd_controls, .num_controls = ARRAY_SIZE(ak4497_snd_controls), .dapm_widgets = ak4497_dapm_widgets, -- 2.34.1

3 2

[v2] ASoC: max98390: Remove unnecessary amp on/off conrtol
by Steve Lee 08 Sep '22

08 Sep '22

The Amp is already control in userspace before trigger calibrate function. Remove unnecessary control in calibrate function and add condition to check calibration is ready. Signed-off-by: Steve Lee <steve.lee.analog(a)gmail.com> --- sound/soc/codecs/max98390.c | 17 ++++++++++------- 1 file changed, 10 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index 91c0bf3d76fc..4b2ee70c29d7 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -635,10 +635,19 @@ static int max98390_dsm_calib_get(struct snd_kcontrol *kcontrol, static int max98390_dsm_calib_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { + unsigned int val; struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct max98390_priv *max98390 = + snd_soc_component_get_drvdata(component); - max98390_dsm_calibrate(component); + regmap_read(max98390->regmap, MAX98390_R23FF_GLOBAL_EN, &val); + if (val == 0x1) + max98390_dsm_calibrate(component); + else { + dev_err(component->dev, "AMP is not ready to run calibration\n"); + return -ECANCELED; + } return 0; } @@ -826,9 +835,6 @@ static int max98390_dsm_calibrate(struct snd_soc_component *component) struct max98390_priv *max98390 = snd_soc_component_get_drvdata(component); - regmap_write(max98390->regmap, MAX98390_R203A_AMP_EN, 0x81); - regmap_write(max98390->regmap, MAX98390_R23FF_GLOBAL_EN, 0x01); - regmap_read(max98390->regmap, THERMAL_RDC_RD_BACK_BYTE1, &rdc); regmap_read(max98390->regmap, @@ -847,9 +853,6 @@ static int max98390_dsm_calibrate(struct snd_soc_component *component) dev_info(component->dev, "rdc resistance about %d.%02d ohm, reg=0x%X temp reg=0x%X\n", rdc_integer, rdc_factor, rdc_cal_result, temp); - regmap_write(max98390->regmap, MAX98390_R23FF_GLOBAL_EN, 0x00); - regmap_write(max98390->regmap, MAX98390_R203A_AMP_EN, 0x80); - return 0; } -- 2.17.1

2 1

[PATCH linux-next] ASoC: sti-sas: Remove the unneeded result variable
by cgel.zte＠gmail.com 08 Sep '22

08 Sep '22

From: ye xingchen <ye.xingchen(a)zte.com.cn> Return the value regmap_write() and sti_sas_init_sas_registers() directly instead of storing it in another redundant variable. Reported-by: Zeal Robot <zealci(a)zte.com.cn> Signed-off-by: ye xingchen <ye.xingchen(a)zte.com.cn> --- sound/soc/codecs/sti-sas.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/sti-sas.c b/sound/soc/codecs/sti-sas.c index f076878908ee..99545bcb2ba9 100644 --- a/sound/soc/codecs/sti-sas.c +++ b/sound/soc/codecs/sti-sas.c @@ -96,11 +96,8 @@ static int sti_sas_write_reg(void *context, unsigned int reg, unsigned int value) { struct sti_sas_data *drvdata = context; - int status; - - status = regmap_write(drvdata->dac.regmap, reg, value); - return status; + return regmap_write(drvdata->dac.regmap, reg, value); } static int sti_sas_init_sas_registers(struct snd_soc_component *component, @@ -385,11 +382,8 @@ static int sti_sas_resume(struct snd_soc_component *component) static int sti_sas_component_probe(struct snd_soc_component *component) { struct sti_sas_data *drvdata = dev_get_drvdata(component->dev); - int ret; - ret = sti_sas_init_sas_registers(component, drvdata); - - return ret; + return sti_sas_init_sas_registers(component, drvdata); } static struct snd_soc_component_driver sti_sas_driver = { -- 2.25.1

2 1

[GIT PULL] ASoC fixes for v6.0-rc4
by Mark Brown 08 Sep '22

08 Sep '22

The following changes since commit 568035b01cfb107af8d2e4bd2fb9aea22cf5b868: Linux 6.0-rc1 (2022-08-14 15:50:18 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.0-rc4 for you to fetch changes up to b1cd3fd42db7593a2d24c06f1c53b8c886592080: ASoC: fsl_aud2htx: Add error handler for pm_runtime_enable (2022-08-25 14:17:31 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.0 Quite a few fixes here, all driver specific and fairly small. ---------------------------------------------------------------- Bard Liao (1): ASoC: SOF: ipc4-topology: fix alh_group_ida max value Jiaxin Yu (1): ASoC: mediatek: mt8186: fix DMIC record noise Mark Brown (3): Merge remote-tracking branch 'asoc/for-5.20' into asoc-6.0 ASoC: SOF: Kconfig: Fix the dependency for client modules ASoC: nau8xxx: Implement hw constraint for rates Nathan Chancellor (1): ASoC: mchp-spdiftx: Fix clang -Wbitfield-constant-conversion Peter Ujfalusi (2): ASoC: SOF: Kconfig: Make IPC_FLOOD_TEST depend on SND_SOC_SOF ASoC: SOF: Kconfig: Make IPC_MESSAGE_INJECTOR depend on SND_SOC_SOF Pieterjan Camerlynck (1): ASoC: fsl_sai: fix incorrect mclk number in error message Richard Fitzgerald (1): ASoC: cs42l42: Only report button state if there was a button interrupt Shengjiu Wang (3): ASoC: fsl_mqs: Fix supported clock DAI format ASoC: fsl_aud2htx: register platform component before registering cpu dai ASoC: fsl_aud2htx: Add error handler for pm_runtime_enable Srinivas Kandagatla (1): ASoC: qcom: sm8250: add missing module owner Takashi Iwai (5): ASoC: nau8821: Implement hw constraint for rates ASoC: nau8824: Fix semaphore unbalance at error paths ASoC: nau8824: Implement hw constraint for rates ASoC: nau8825: Implement hw constraint for rates ASoC: nau8540: Implement hw constraint for rates Yang Yingliang (1): ASoC: SOF: ipc4-topology: fix wrong use of sizeof in sof_ipc4_widget_setup_comp_src() sound/soc/amd/yc/acp6x-mach.c | 28 +++++++++ sound/soc/atmel/mchp-spdiftx.c | 2 +- sound/soc/codecs/cs42l42.c | 13 ++-- sound/soc/codecs/nau8540.c | 40 ++++++++---- sound/soc/codecs/nau8821.c | 66 ++++++++++--------- sound/soc/codecs/nau8824.c | 80 +++++++++++++---------- sound/soc/codecs/nau8825.c | 83 +++++++++++++----------- sound/soc/codecs/rt5640.c | 5 +- sound/soc/codecs/tas2770.c | 98 +++++++++++++---------------- sound/soc/codecs/tas2770.h | 5 ++ sound/soc/codecs/tlv320aic32x4.c | 9 +++ sound/soc/fsl/fsl_aud2htx.c | 16 +++-- sound/soc/fsl/fsl_mqs.c | 2 +- sound/soc/fsl/fsl_sai.c | 2 +- sound/soc/intel/avs/pcm.c | 4 +- sound/soc/intel/boards/sof_es8336.c | 4 +- sound/soc/mediatek/mt8186/mt8186-dai-adda.c | 3 - sound/soc/qcom/sm8250.c | 1 + sound/soc/sh/rz-ssi.c | 26 ++++---- sound/soc/soc-pcm.c | 3 + sound/soc/sof/Kconfig | 2 + sound/soc/sof/debug.c | 6 +- sound/soc/sof/intel/hda.c | 2 +- sound/soc/sof/ipc3-topology.c | 2 +- sound/soc/sof/ipc4-topology.c | 4 +- 25 files changed, 299 insertions(+), 207 deletions(-)

2 1

Alsaloop long-running session leads to CPU spikes and stalls
by GitHub issues - opened 08 Sep '22

08 Sep '22

alsa-project/alsa-utils issue #168 was opened from ebrodlic: We have a recurring issue on an embedded system where we use alsaloop to continually forward audio between: EC25E modem -> system speaker (output) system microphone (input) -> EC25E modem The two alsaloop processes run constantly in the background, and are in charge of enabling the GSM call audio functionality on our devices. What ends up happening is: - System boots up and starts the 2 processes - System usage is normal levels for 2-3 hours (very low CPU utilization) - After cca 3 hours of runtime, one alsaloop process will spike a single core of the CPU up to 100% continually - After a seemingly random amount of time of the single CPU core being fully utilized, the system may show reduced responsiveness, eventually showing recurring kernel warnings regarding CPU stalls ``` [Tue Aug 30 16:50:42 2022] rcu: INFO: rcu_preempt self-detected stall on CPU [Tue Aug 30 16:50:42 2022] rcu: 3-....: (5048 ticks this GP) idle=752/1/0x4000000000000002 softirq=1819133/1819133 fqs=2493 [Tue Aug 30 16:50:42 2022] (t=5250 jiffies g=3235733 q=26917) [Tue Aug 30 16:50:42 2022] Task dump for CPU 3: [Tue Aug 30 16:50:42 2022] alsaloop R running task 0 761 1 0x00000202 [Tue Aug 30 16:50:42 2022] Call trace: [Tue Aug 30 16:50:42 2022] dump_backtrace+0x0/0x140 [Tue Aug 30 16:50:42 2022] show_stack+0x14/0x20 [Tue Aug 30 16:50:42 2022] sched_show_task+0x110/0x138 [Tue Aug 30 16:50:42 2022] dump_cpu_task+0x40/0x4c [Tue Aug 30 16:50:42 2022] rcu_dump_cpu_stacks+0x94/0xd0 [Tue Aug 30 16:50:42 2022] rcu_sched_clock_irq+0x75c/0x9c8 [Tue Aug 30 16:50:42 2022] update_process_times+0x2c/0x68 [Tue Aug 30 16:50:42 2022] tick_sched_handle.isra.0+0x30/0x50 [Tue Aug 30 16:50:42 2022] tick_sched_timer+0x48/0x98 [Tue Aug 30 16:50:42 2022] __hrtimer_run_queues+0x110/0x1b0 [Tue Aug 30 16:50:42 2022] hrtimer_interrupt+0xe4/0x240 [Tue Aug 30 16:50:42 2022] arch_timer_handler_phys+0x30/0x40 [Tue Aug 30 16:50:42 2022] handle_percpu_devid_irq+0x80/0x140 [Tue Aug 30 16:50:42 2022] generic_handle_irq+0x24/0x38 [Tue Aug 30 16:50:42 2022] __handle_domain_irq+0x60/0xb8 [Tue Aug 30 16:50:42 2022] gic_handle_irq+0x5c/0x148 [Tue Aug 30 16:50:42 2022] el1_irq+0xb8/0x180 [Tue Aug 30 16:50:42 2022] snd_power_wait+0x0/0xe8 [Tue Aug 30 16:50:42 2022] snd_pcm_ioctl+0x2c/0x40 [Tue Aug 30 16:50:42 2022] do_vfs_ioctl+0x964/0xb48 [Tue Aug 30 16:50:42 2022] ksys_ioctl+0x78/0xa8 [Tue Aug 30 16:50:42 2022] __arm64_sys_ioctl+0x1c/0xb8 [Tue Aug 30 16:50:42 2022] el0_svc_common.constprop.0+0x68/0x160 [Tue Aug 30 16:50:42 2022] el0_svc_handler+0x20/0x80 [Tue Aug 30 16:50:42 2022] el0_svc+0x8/0x208 [Tue Aug 30 16:50:42 2022] retire_capture_urb: 307 callbacks suppressed ``` Commands used to start the two processes: `/usr/bin/alsaloop -P ec25 -C default -t 500000 -n -r 8000 -f S16_LE -c 1` `/usr/bin/alsaloop -C ec25 -P default -t 500000 -n -r 8000 -f S16_LE -c 1` Our current alsa .asoundrc conf file: ``` pcm.dmixer { type dmix ipc_key 1024 ipc_key_add_uid false # let multiple users share ipc_perm 0666 # IPC permissions for multi-user sharing (octal, default 0600) slave { pcm "hw:tas5722,0" } bindings { 0 0 1 1 } } pcm.dsnooped { type dsnoop ipc_key 1027 ipc_key_add_uid false # let multiple users share ipc_perm 0666 # IPC permissions for multi-user sharing (octal, default 0600) slave { pcm "hw:ics43432,0" channels 1 format S32_LE } } pcm.asymed { type asym playback.pcm "dmixer" capture.pcm "softvol" } pcm.!default { type plug slave.pcm "asymed" } pcm_slave.ec25 { pcm "hw:EC25E,0" format S16_LE channels 1 rate 8000 } pcm.ec25 { type plug slave ec25 } pcm.softvol { type softvol slave.pcm dsnooped control { name "Boost Capture Volume" card ics43432 } min_dB -3.0 max_dB 50.0 } ``` This one is not very easy to debug so it would be great if I could get some pointers in are we dealing with: - potential bugs - alsaloop misuse in approach - problems in existing configuration Any info would be appreciated! Issue URL : https://github.com/alsa-project/alsa-utils/issues/168 Repository URL: https://github.com/alsa-project/alsa-utils

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ucm2: Add support for MT8195 Cherry Tomato Chromebook with SOF
by GitHub pull_request - edited 08 Sep '22

08 Sep '22

alsa-project/alsa-ucm-conf pull request #213 was edited from kholk: Add support for the Acer Chromebook Spin 513 CP513-2H, powered by MediaTek Kompanio 1380 (MT8195) with a HiFi 4 Audio DSP running SoundOpenFirmware. This machine uses MT6359 PMIC, with RT1019 and RT5682 as headphones and speaker codecs/amps. This configuration has been successfully tested on Debian 11 with PipeWire 0.3.56 + wireplumber. `Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno(a)collabora.com>` Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/213 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/213.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: Add support for MT8195 Cherry Tomato Chromebook with SOF
by GitHub pull_request - opened 08 Sep '22

08 Sep '22

alsa-project/alsa-ucm-conf pull request #213 was opened from kholk: Add support for the Acer Chromebook Spin 513 CP513-2H, powered by MediaTek Kompanio 1380 (MT8195) with a HiFi 4 Audio DSP running SoundOpenFirmware. This machine uses MT6359 PMIC, with RT1019 and RT5682 as headphones and speaker codecs/amps. This configuration has been successfully tested on Debian 11 with PipeWire 0.3.56 + wireplumber. Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno(a)collabora.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/213 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/213.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2 00/14] ASoC/qcom/arm64: Qualcomm ADSP DTS and binding fixes
by Krzysztof Kozlowski 08 Sep '22

08 Sep '22

Hi, Dependencies/merging ==================== 1. The DTS patches are independent. 2. The binding patches should come together, because of context changes. Could be one of: Qualcomm SoC, ASoC or DT tree. Changes since v1 ================ 1. Patch 9: New patch. 2. Patch 10: Correct also sound/qcom,q6apm-dai.yaml (Rob). 2. Patch 13: New patch. 3. Add Rb/Tb tags. Best regards, Krzysztof Krzysztof Kozlowski (14): arm64: dts: qcom: sdm630: align APR services node names with dtschema arm64: dts: qcom: sdm845: align APR services node names with dtschema arm64: dts: qcom: sm8250: align APR services node names with dtschema arm64: dts: qcom: msm8996: fix APR services nodes arm64: dts: qcom: sdm845: align dai node names with dtschema arm64: dts: qcom: msm8996: align dai node names with dtschema arm64: dts: qcom: qrb5165-rb5: align dai node names with dtschema arm64: dts: qcom: sm8250: use generic name for LPASS clock controller ASoC: dt-bindings:: qcom,q6asm: convert to dtschema dt-bindings: soc: qcom: apr: correct service children ASoC: dt-bindings: qcom,q6dsp-lpass-ports: cleanup example ASoC: dt-bindings: qcom,q6dsp-lpass-clocks: cleanup example ASoC: dt-bindings: qcom,q6apm-dai: adjust indentation in example dt-bindings: soc: qcom: apr: add missing properties .../bindings/soc/qcom/qcom,apr.yaml | 111 +++++++++++++++-- .../bindings/sound/qcom,q6apm-dai.yaml | 21 ++-- .../devicetree/bindings/sound/qcom,q6asm.txt | 70 ----------- .../devicetree/bindings/sound/qcom,q6asm.yaml | 112 ++++++++++++++++++ .../sound/qcom,q6dsp-lpass-clocks.yaml | 36 +++--- .../sound/qcom,q6dsp-lpass-ports.yaml | 64 +++++----- arch/arm64/boot/dts/qcom/msm8996.dtsi | 10 +- arch/arm64/boot/dts/qcom/qrb5165-rb5.dts | 4 +- arch/arm64/boot/dts/qcom/sdm630.dtsi | 8 +- arch/arm64/boot/dts/qcom/sdm845-db845c.dts | 2 +- .../boot/dts/qcom/sdm845-xiaomi-beryllium.dts | 2 +- .../boot/dts/qcom/sdm845-xiaomi-polaris.dts | 4 +- arch/arm64/boot/dts/qcom/sdm845.dtsi | 8 +- arch/arm64/boot/dts/qcom/sm8250.dtsi | 10 +- 14 files changed, 293 insertions(+), 169 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/qcom,q6asm.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6asm.yaml -- 2.34.1

2 17

[PATCH] slimbus: qcom-ngd-ctrl: allow compile testing without QCOM_RPROC_COMMON
by Krzysztof Kozlowski 08 Sep '22

08 Sep '22

The Qualcomm common remote-proc code (CONFIG_QCOM_RPROC_COMMON) has necessary stubs, so it is not needed for compile testing. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- drivers/slimbus/Kconfig | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/drivers/slimbus/Kconfig b/drivers/slimbus/Kconfig index 1235b7dc8496..2ed821f75816 100644 --- a/drivers/slimbus/Kconfig +++ b/drivers/slimbus/Kconfig @@ -22,7 +22,8 @@ config SLIM_QCOM_CTRL config SLIM_QCOM_NGD_CTRL tristate "Qualcomm SLIMbus Satellite Non-Generic Device Component" - depends on HAS_IOMEM && DMA_ENGINE && NET && QCOM_RPROC_COMMON + depends on HAS_IOMEM && DMA_ENGINE && NET + depends on QCOM_RPROC_COMMON || COMPILE_TEST depends on ARCH_QCOM || COMPILE_TEST select QCOM_QMI_HELPERS select QCOM_PDR_HELPERS -- 2.34.1

2 1

[linux-next:master] BUILD REGRESSION 5957ac6635a1a12d4aa2661bbf04d3085a73372a
by kernel test robot 08 Sep '22

08 Sep '22

tree/branch: https://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git master branch HEAD: 5957ac6635a1a12d4aa2661bbf04d3085a73372a Add linux-next specific files for 20220907 Error/Warning reports: https://lore.kernel.org/linux-mm/202209070728.o3stvgVt-lkp@intel.com https://lore.kernel.org/linux-mm/202209080545.qMIVj7YM-lkp@intel.com https://lore.kernel.org/linux-mm/202209080718.y5QmlNKH-lkp@intel.com Error/Warning: (recently discovered and may have been fixed) ERROR: modpost: "__divdi3" [drivers/gpu/drm/vkms/vkms.ko] undefined! ERROR: modpost: "__udivdi3" [drivers/gpu/drm/vkms/vkms.ko] undefined! arm-linux-gnueabi-ld: vkms_formats.c:(.text+0x824): undefined reference to `__aeabi_ldivmod' arm-linux-gnueabi-ld: warning: orphan section `.data.rel.ro.local' from `arch/arm/boot/compressed/fdt.o' being placed in section `.data.rel.ro.local' arm-linux-gnueabi-ld: warning: orphan section `.data.rel.ro.local' from `arch/arm/boot/compressed/fdt_ro.o' being placed in section `.data.rel.ro.local' arm-linux-gnueabi-ld: warning: orphan section `.data.rel.ro.local' from `arch/arm/boot/compressed/fdt_rw.o' being placed in section `.data.rel.ro.local' arm-linux-gnueabi-ld: warning: orphan section `.data.rel.ro.local' from `arch/arm/boot/compressed/fdt_wip.o' being placed in section `.data.rel.ro.local' arm-linux-gnueabi-ld: warning: orphan section `.printk_index' from `arch/arm/boot/compressed/fdt.o' being placed in section `.printk_index' arm-linux-gnueabi-ld: warning: orphan section `.printk_index' from `arch/arm/boot/compressed/fdt_ro.o' being placed in section `.printk_index' arm-linux-gnueabi-ld: warning: orphan section `.printk_index' from `arch/arm/boot/compressed/fdt_rw.o' being placed in section `.printk_index' arm-linux-gnueabi-ld: warning: orphan section `.printk_index' from `arch/arm/boot/compressed/fdt_wip.o' being placed in section `.printk_index' drivers/base/regmap/regmap-mmio.c:221:17: error: implicit declaration of function 'writesb'; did you mean 'writeb'? [-Werror=implicit-function-declaration] drivers/base/regmap/regmap-mmio.c:224:17: error: implicit declaration of function 'writesw'; did you mean 'writew'? [-Werror=implicit-function-declaration] drivers/base/regmap/regmap-mmio.c:227:17: error: implicit declaration of function 'writesl'; did you mean 'writel'? [-Werror=implicit-function-declaration] drivers/base/regmap/regmap-mmio.c:231:17: error: implicit declaration of function 'writesq'; did you mean 'writeq'? [-Werror=implicit-function-declaration] drivers/base/regmap/regmap-mmio.c:231:17: error: implicit declaration of function 'writesq'; did you mean 'writesl'? [-Werror=implicit-function-declaration] drivers/base/regmap/regmap-mmio.c:358:17: error: implicit declaration of function 'readsb'; did you mean 'readb'? [-Werror=implicit-function-declaration] drivers/base/regmap/regmap-mmio.c:361:17: error: implicit declaration of function 'readsw'; did you mean 'readw'? [-Werror=implicit-function-declaration] drivers/base/regmap/regmap-mmio.c:364:17: error: implicit declaration of function 'readsl'; did you mean 'readl'? [-Werror=implicit-function-declaration] drivers/base/regmap/regmap-mmio.c:368:17: error: implicit declaration of function 'readsq'; did you mean 'readq'? [-Werror=implicit-function-declaration] drivers/base/regmap/regmap-mmio.c:368:17: error: implicit declaration of function 'readsq'; did you mean 'readsl'? [-Werror=implicit-function-declaration] drivers/clk/xilinx/clk-xlnx-clock-wizard.c:431: undefined reference to `devm_platform_ioremap_resource' drivers/gpu/drm/amd/amdgpu/imu_v11_0_3.c:139:6: warning: no previous prototype for 'imu_v11_0_3_program_rlc_ram' [-Wmissing-prototypes] drivers/gpu/drm/drm_atomic_helper.c:802: warning: expecting prototype for drm_atomic_helper_check_wb_connector_state(). Prototype was for drm_atomic_helper_check_wb_encoder_state() instead drivers/gpu/drm/vkms/vkms_plane.c:110 vkms_plane_atomic_update() warn: variable dereferenced before check 'fb' (see line 108) drivers/scsi/qla2xxx/qla_os.c:2854:23: warning: assignment to 'struct trace_array *' from 'int' makes pointer from integer without a cast [-Wint-conversion] fs/btrfs/volumes.c:6549 __btrfs_map_block() error: we previously assumed 'mirror_num_ret' could be null (see line 6376) include/linux/string.h:303:42: warning: 'strnlen' specified bound 4 exceeds source size 3 [-Wstringop-overread] kernel/bpf/memalloc.c:499 bpf_mem_alloc_destroy() error: potentially dereferencing uninitialized 'c'. ld: vkms_formats.c:(.text+0x3ba): undefined reference to `__divdi3' microblaze-linux-ld: vkms_formats.o:(.text+0xc74): undefined reference to `__divdi3' mips-linux-ld: vkms_formats.c:(.text+0x384): undefined reference to `__divdi3' mips-linux-ld: vkms_formats.c:(.text.argb_u16_to_RGB565+0xd0): undefined reference to `__divdi3' sound/soc/codecs/tas2562.c:442:13: warning: variable 'ret' set but not used [-Wunused-but-set-variable] vkms_formats.c:(.text+0x266): undefined reference to `__divdi3' vkms_formats.c:(.text+0x364): undefined reference to `__divdi3' vkms_formats.c:(.text+0x390): undefined reference to `__divdi3' vkms_formats.c:(.text+0x804): undefined reference to `__aeabi_ldivmod' vkms_formats.c:(.text.argb_u16_to_RGB565+0xb0): undefined reference to `__divdi3' vkms_formats.o:(.text+0xb28): undefined reference to `__divdi3' xtensa-linux-ld: vkms_formats.c:(.text+0x560): undefined reference to `__divdi3' Error/Warning ids grouped by kconfigs: gcc_recent_errors |-- alpha-allyesconfig | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-readsb | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-readsl | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-readsq | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-readsw | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-writesb | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-writesl | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-writesq | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-writesw | |-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-imu_v11_0_3_program_rlc_ram | |-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead | |-- drivers-scsi-qla2xxx-qla_os.c:warning:assignment-to-struct-trace_array-from-int-makes-pointer-from-integer-without-a-cast | `-- sound-soc-codecs-tas2562.c:warning:variable-ret-set-but-not-used |-- alpha-randconfig-r015-20220907 | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-readsb | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-readsl | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-readsq | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-readsw | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-writesb | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-writesl | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-writesq | `-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-writesw |-- arc-allyesconfig | |-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-imu_v11_0_3_program_rlc_ram | |-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead | `-- sound-soc-codecs-tas2562.c:warning:variable-ret-set-but-not-used |-- arc-randconfig-r001-20220907 | `-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead |-- arm-allyesconfig | |-- arm-linux-gnueabi-ld:vkms_formats.c:(.text):undefined-reference-to-__aeabi_ldivmod | |-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-imu_v11_0_3_program_rlc_ram | |-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead | |-- sound-soc-codecs-tas2562.c:warning:variable-ret-set-but-not-used | `-- vkms_formats.c:(.text):undefined-reference-to-__aeabi_ldivmod |-- arm-buildonly-randconfig-r006-20220907 | |-- arm-linux-gnueabi-ld:warning:orphan-section-data.rel.ro.local-from-arch-arm-boot-compressed-fdt.o-being-placed-in-section-.data.rel.ro.local | |-- arm-linux-gnueabi-ld:warning:orphan-section-data.rel.ro.local-from-arch-arm-boot-compressed-fdt_ro.o-being-placed-in-section-.data.rel.ro.local | |-- arm-linux-gnueabi-ld:warning:orphan-section-data.rel.ro.local-from-arch-arm-boot-compressed-fdt_rw.o-being-placed-in-section-.data.rel.ro.local | |-- arm-linux-gnueabi-ld:warning:orphan-section-data.rel.ro.local-from-arch-arm-boot-compressed-fdt_wip.o-being-placed-in-section-.data.rel.ro.local | |-- arm-linux-gnueabi-ld:warning:orphan-section-printk_index-from-arch-arm-boot-compressed-fdt.o-being-placed-in-section-.printk_index | |-- arm-linux-gnueabi-ld:warning:orphan-section-printk_index-from-arch-arm-boot-compressed-fdt_ro.o-being-placed-in-section-.printk_index | |-- arm-linux-gnueabi-ld:warning:orphan-section-printk_index-from-arch-arm-boot-compressed-fdt_rw.o-being-placed-in-section-.printk_index | `-- arm-linux-gnueabi-ld:warning:orphan-section-printk_index-from-arch-arm-boot-compressed-fdt_wip.o-being-placed-in-section-.printk_index |-- arm-defconfig | `-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead |-- arm64-allyesconfig | |-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-imu_v11_0_3_program_rlc_ram | |-- drivers-gpu-drm-drm_atomic_helper.c:warning:expecting-prototype-for-drm_atomic_helper_check_wb_connector_state().-Prototype-was-for-drm_atomic_helper_check_wb_encoder_state()-instead | `-- sound-soc-codecs-tas2562.c:warning:variable-ret-set-but-not-used |-- arm64-randconfig-r033-20220907 clang_recent_errors |-- i386-randconfig-a006 | `-- ld.lld:error:undefined-symbol:__udivdi3 |-- i386-randconfig-a013 | `-- ld.lld:error:undefined-symbol:__udivdi3 |-- powerpc-randconfig-r013-20220907 | `-- drivers-extcon-extcon-usbc-tusb320.c:warning:expecting-prototype-for-drivers-extcon-extcon-tusb320c().-Prototype-was-for-TUSB320_REG8()-instead `-- riscv-randconfig-r021-20220907 `-- drivers-extcon-extcon-usbc-tusb320.c:warning:expecting-prototype-for-drivers-extcon-extcon-tusb320c().-Prototype-was-for-TUSB320_REG8()-instead elapsed time: 723m configs tested: 35 configs skipped: 2 gcc tested configs: um i386_defconfig um x86_64_defconfig i386 defconfig x86_64 rhel-8.3-kselftests x86_64 rhel-8.3-func x86_64 defconfig arc randconfig-r043-20220907 x86_64 rhel-8.3-kunit x86_64 rhel-8.3-kvm x86_64 rhel-8.3 x86_64 rhel-8.3-syz x86_64 randconfig-a013 x86_64 allyesconfig x86_64 randconfig-a011 x86_64 randconfig-a004 m68k allyesconfig arm defconfig m68k allmodconfig alpha allyesconfig powerpc allnoconfig arc allyesconfig ia64 allmodconfig arm allyesconfig arm64 allyesconfig sh allmodconfig mips allyesconfig powerpc allmodconfig i386 allyesconfig clang tested configs: hexagon randconfig-r041-20220907 riscv randconfig-r042-20220907 hexagon randconfig-r045-20220907 s390 randconfig-r044-20220907 i386 randconfig-a013 i386 randconfig-a006 x86_64 rhel-8.3-rust -- 0-DAY CI Kernel Test Service https://01.org/lkp

1 0

Some Motu M4 have a different usb device ID
by GitHub issues - opened 07 Sep '22

07 Sep '22

alsa-project/alsa-ucm-conf issue #212 was opened from Big-FG: I just wanted to draw some attention regarding the Motu M4 UCM2 profie currently USB-Audio.conf has the M4 with usb id 07fd:000b ``` if.motu-m24 { Condition { Type String Haystack "${CardComponents}" Needle "USB07fd:000b" } True.If.M4 { Condition { Type String Haystack "${CardLongName}" Needle "MOTU M4" } True.Define.ProfileName "MOTU/M4" False.Define.ProfileName "MOTU/M2" } } ``` but mine is apparently different with 07fd:0008 `Bus 003 Device 002: ID 07fd:0008 Mark of the Unicorn M Series` after adjusting the USB-Audio.conf with my usb ID the profile loads and works correctly. Im not sure if mine is just a weird one or how many people this may be the case for. Im fine with modifying the conf myself for my card if this is just specific to me as my unit wasnt purchased new but was a used one so Im not sure what might be different. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/212 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 0/5] ASoC: Fixes for MT8195 SOF support
by AngeloGioacchino Del Regno 07 Sep '22

07 Sep '22

This series fixes Sound Open Firmware support for MT8195 by making sure that the sound card driver is actually able to probe and IPC can finally happen. It is now possible to get DSP support for audio. Tested on MT8195 Tomato - Acer Chromebook Spin 513 CP513-2H (Pipewire). AngeloGioacchino Del Regno (5): ASoC: mediatek: mt8195-mt6359: Properly register sound card for SOF ASoC: SOF: mediatek: mt8195: Import namespace SND_SOC_SOF_MTK_COMMON ASoC: SOF: mediatek: mt8195: Add mailbox generic callbacks for IPC ASoC: SOF: mediatek: mt8195: Add generic pcm_{open,close} callbacks ASoC: SOF: mediatek: mt8195: Add devicetree support to select topologies sound/soc/mediatek/mt8195/mt8195-mt6359.c | 6 +++++ sound/soc/sof/mediatek/mt8195/mt8195.c | 32 ++++++++++++++++------- 2 files changed, 29 insertions(+), 9 deletions(-) -- 2.37.2

2 6

[PATCH] ASoC: ts3a227e: add parameters to control debounce times
by Astrid Rost 07 Sep '22

07 Sep '22

Add devicetree parameters to control the insertion, release and press debounce times. Signed-off-by: Astrid Rost <astrid.rost(a)axis.com> --- sound/soc/codecs/ts3a227e.c | 48 +++++++++++++++++++++++++++++++------ 1 file changed, 41 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/ts3a227e.c b/sound/soc/codecs/ts3a227e.c index d8ab0810fceb..fea274a0a91b 100644 --- a/sound/soc/codecs/ts3a227e.c +++ b/sound/soc/codecs/ts3a227e.c @@ -78,12 +78,20 @@ static const int ts3a227e_buttons[] = { #define ADC_COMPLETE_INT_DISABLE 0x04 #define INTB_DISABLE 0x08 +/* TS3A227E_REG_SETTING_1 0x4 */ +#define DEBOUNCE_INSERTION_SETTING_SFT (0) +#define DEBOUNCE_INSERTION_SETTING_MASK (0x7 << DEBOUNCE_PRESS_SETTING_SFT) + /* TS3A227E_REG_SETTING_2 0x05 */ #define KP_ENABLE 0x04 /* TS3A227E_REG_SETTING_3 0x06 */ -#define MICBIAS_SETTING_SFT (3) +#define MICBIAS_SETTING_SFT 3 #define MICBIAS_SETTING_MASK (0x7 << MICBIAS_SETTING_SFT) +#define DEBOUNCE_RELEASE_SETTING_SFT 2 +#define DEBOUNCE_RELEASE_SETTING_MASK (0x1 << DEBOUNCE_RELEASE_SETTING_SFT) +#define DEBOUNCE_PRESS_SETTING_SFT 0 +#define DEBOUNCE_PRESS_SETTING_MASK (0x3 << DEBOUNCE_PRESS_SETTING_SFT) /* TS3A227E_REG_ACCESSORY_STATUS 0x0b */ #define TYPE_3_POLE 0x01 @@ -136,7 +144,7 @@ static bool ts3a227e_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { case TS3A227E_REG_INTERRUPT ... TS3A227E_REG_INTERRUPT_DISABLE: - case TS3A227E_REG_SETTING_2: + case TS3A227E_REG_SETTING_1 ... TS3A227E_REG_SETTING_2: case TS3A227E_REG_SWITCH_STATUS_1 ... TS3A227E_REG_ADC_OUTPUT: return true; default: @@ -269,16 +277,42 @@ static const struct regmap_config ts3a227e_regmap_config = { static int ts3a227e_parse_device_property(struct ts3a227e *ts3a227e, struct device *dev) { - u32 micbias; + u32 value; + u32 setting3_value = 0; + u32 setting3_mask = 0; int err; - err = device_property_read_u32(dev, "ti,micbias", &micbias); + err = device_property_read_u32(dev, "ti,micbias", &value); if (!err) { - regmap_update_bits(ts3a227e->regmap, TS3A227E_REG_SETTING_3, - MICBIAS_SETTING_MASK, - (micbias & 0x07) << MICBIAS_SETTING_SFT); + setting3_mask = MICBIAS_SETTING_MASK; + setting3_value = + (value << MICBIAS_SETTING_SFT) & MICBIAS_SETTING_MASK; + } + + err = device_property_read_u32(dev, "ti,debounce-release", &value); + if (!err) { + setting3_mask |= DEBOUNCE_RELEASE_SETTING_MASK; + setting3_value |= (value << DEBOUNCE_RELEASE_SETTING_SFT) & + DEBOUNCE_RELEASE_SETTING_MASK; } + err = device_property_read_u32(dev, "ti,debounce-press", &value); + if (!err) { + setting3_mask |= DEBOUNCE_PRESS_SETTING_MASK; + setting3_value |= (value << DEBOUNCE_PRESS_SETTING_SFT) & + DEBOUNCE_PRESS_SETTING_MASK; + } + + if (setting3_mask) + regmap_update_bits(ts3a227e->regmap, TS3A227E_REG_SETTING_3, + setting3_mask, setting3_value); + + err = device_property_read_u32(dev, "ti,debounce-insertion", &value); + if (!err) + regmap_update_bits(ts3a227e->regmap, TS3A227E_REG_SETTING_1, + DEBOUNCE_INSERTION_SETTING_MASK, (value << DEBOUNCE_INSERTION_SETTING_SFT) + & DEBOUNCE_INSERTION_SETTING_MASK); + return 0; } -- 2.20.1

1 0

[PATCH] ASoC: simple-card-utils: switch to using gpiod API
by Dmitry Torokhov 07 Sep '22

07 Sep '22

This patch switches the driver away from legacy gpio/of_gpio API to gpiod API, and removes use of of_get_named_gpio_flags() which I want to make private to gpiolib. Signed-off-by: Dmitry Torokhov <dmitry.torokhov(a)gmail.com> --- sound/soc/generic/simple-card-utils.c | 28 ++++++++++++++------------- 1 file changed, 15 insertions(+), 13 deletions(-) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 1b201dd09259..bef16833c487 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -9,7 +9,6 @@ #include <linux/gpio/consumer.h> #include <linux/module.h> #include <linux/of.h> -#include <linux/of_gpio.h> #include <linux/of_graph.h> #include <sound/jack.h> #include <sound/pcm_params.h> @@ -729,12 +728,12 @@ int asoc_simple_init_jack(struct snd_soc_card *card, char *pin) { struct device *dev = card->dev; - enum of_gpio_flags flags; + struct gpio_desc *desc; char prop[128]; char *pin_name; char *gpio_name; int mask; - int det; + int error; if (!prefix) prefix = ""; @@ -742,36 +741,39 @@ int asoc_simple_init_jack(struct snd_soc_card *card, sjack->gpio.gpio = -ENOENT; if (is_hp) { - snprintf(prop, sizeof(prop), "%shp-det-gpio", prefix); + snprintf(prop, sizeof(prop), "%shp-det", prefix); pin_name = pin ? pin : "Headphones"; gpio_name = "Headphone detection"; mask = SND_JACK_HEADPHONE; } else { - snprintf(prop, sizeof(prop), "%smic-det-gpio", prefix); + snprintf(prop, sizeof(prop), "%smic-det", prefix); pin_name = pin ? pin : "Mic Jack"; gpio_name = "Mic detection"; mask = SND_JACK_MICROPHONE; } - det = of_get_named_gpio_flags(dev->of_node, prop, 0, &flags); - if (det == -EPROBE_DEFER) - return -EPROBE_DEFER; + desc = gpiod_get_optional(dev, prop, GPIOD_IN); + error = PTR_ERR_OR_ZERO(desc); + if (error) + return error; + + if (desc) { + error = gpiod_set_consumer_name(desc, gpio_name); + if (error) + return error; - if (gpio_is_valid(det)) { sjack->pin.pin = pin_name; sjack->pin.mask = mask; sjack->gpio.name = gpio_name; sjack->gpio.report = mask; - sjack->gpio.gpio = det; - sjack->gpio.invert = !!(flags & OF_GPIO_ACTIVE_LOW); + sjack->gpio.desc = desc; sjack->gpio.debounce_time = 150; snd_soc_card_jack_new_pins(card, pin_name, mask, &sjack->jack, &sjack->pin, 1); - snd_soc_jack_add_gpios(&sjack->jack, 1, - &sjack->gpio); + snd_soc_jack_add_gpios(&sjack->jack, 1, &sjack->gpio); } return 0; -- 2.37.2.789.g6183377224-goog -- Dmitry

3 2

[PATCH] ASoC: dt-bindings: qcom,q6core: remove binding
by Krzysztof Kozlowski 07 Sep '22

07 Sep '22

qcom,q6core is already described in soc/qcom/qcom,apr.yaml. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- .../devicetree/bindings/sound/qcom,q6core.txt | 21 ------------------- 1 file changed, 21 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/qcom,q6core.txt diff --git a/Documentation/devicetree/bindings/sound/qcom,q6core.txt b/Documentation/devicetree/bindings/sound/qcom,q6core.txt deleted file mode 100644 index 5cd4cc9b1fde..000000000000 --- a/Documentation/devicetree/bindings/sound/qcom,q6core.txt +++ /dev/null @@ -1,21 +0,0 @@ -Qualcomm ADSP Core service binding - -Q6CORE is one of the APR audio service on Q6DSP. -Please refer to qcom,apr.txt for details of the common apr service bindings -used by the apr service device. - -- but must contain the following property: - -- compatible: - Usage: required - Value type: <stringlist> - Definition: must be "qcom,q6core-v<MAJOR-NUMBER>.<MINOR-NUMBER>". - Or "qcom,q6core" where the version number can be queried - from DSP. - example "qcom,q6core-v2.0" - -= EXAMPLE -apr-service@3 { - compatible = "qcom,q6core"; - reg = <APR_SVC_ADSP_CORE>; -}; -- 2.34.1

2 1

[PATCH 0/4] ASoC: soc-dapm.c: random cleanup
by Kuninori Morimoto 07 Sep '22

07 Sep '22

Hi Mark These are cleanup patches for soc-dapm.c. Each patches are not related, very random cleanup. Kuninori Morimoto (4): ASoC: soc-dapm.c: don't use kzalloc() for param on snd_soc_dai_link_event_pre_pmu() ASoC: soc-dapm.c: don't use WARN_ON() at snd_soc_dai_link_event_pre_pmu() ASoC: soc-dapm.c: tidyup snd_soc_dai_link_event_pre_pmu() ASoC: soc-dapm.c: fixup snd_soc_dapm_new_control_unlocked() error handling sound/soc/soc-dapm.c | 77 +++++++++++++++++++------------------------- 1 file changed, 33 insertions(+), 44 deletions(-) -- 2.25.1

2 8

utils: cplay: Add support for ID3v2 tag skip
by GitHub pull_request - edited 07 Sep '22

07 Sep '22

alsa-project/tinycompress pull request #15 was edited from LaurentiuM1234: This patch will allow playing MP3 files containing ID3v2 tag at the beginning of the file. This is done simply by parsing the ID3v2 header, extracting the header size and seeking to that position (+10 bytes from the header size which is not included in the 'header size' field of the ID3v2 header) Signed-off-by: Laurentiu Mihalcea <laurentiu.mihalcea(a)nxp.com> Request URL : https://github.com/alsa-project/tinycompress/pull/15 Patch URL : https://github.com/alsa-project/tinycompress/pull/15.patch Repository URL: https://github.com/alsa-project/tinycompress

1 0

utils: cplay: Add support for ID3v2 header skip
by GitHub pull_request - opened 07 Sep '22

07 Sep '22

alsa-project/tinycompress pull request #15 was opened from LaurentiuM1234: This patch will allow playing MP3 files containing ID3v2 tag at the beginning of the file. This is done simply by parsing the ID3v2 header, extracting the header size and seeking to that position (+10 bytes from the header size which is not included in the 'header size' field of the ID3v2 header) Signed-off-by: Laurentiu Mihalcea <laurentiu.mihalcea(a)nxp.com> Request URL : https://github.com/alsa-project/tinycompress/pull/15 Patch URL : https://github.com/alsa-project/tinycompress/pull/15.patch Repository URL: https://github.com/alsa-project/tinycompress

1 0

[PATCH] ALSA: usb-audio: Clear fixed clock rate at closing EP
by Takashi Iwai 07 Sep '22

07 Sep '22

The recent commit c11117b634f4 ("ALSA: usb-audio: Refcount multiple accesses on the single clock") tries to manage the clock rate shared by several endpoints. This was intended for avoiding the unmatched rate by a different endpoint, but unfortunately, it introduced a regression for PulseAudio and pipewire, too; those applications try to probe the multiple possible rates (44.1k and 48kHz) and setting up the normal rate fails but only the last rate is applied. The cause is that the last sample rate is still left to the clock reference even after closing the endpoint, and this value is still used at the next open. It happens only when applications set up via PCM prepare but don't start/stop the stream; the rate is reset when the stream is stopped, but it's not cleared at close. This patch addresses the issue above, simply by clearing the rate set in the clock reference at the last close of each endpoint. Fixes: c11117b634f4 ("ALSA: usb-audio: Refcount multiple accesses on the single clock") Reported-by: Jason A. Donenfeld <Jason(a)zx2c4.com> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/all/YxXIWv8dYmg1tnXP@zx2c4.com/ Link: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/2620 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/endpoint.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index a42f2ce19455..8c8f9a851f89 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -925,6 +925,8 @@ void snd_usb_endpoint_close(struct snd_usb_audio *chip, endpoint_set_interface(chip, ep, false); if (!--ep->opened) { + if (ep->clock_ref && !atomic_read(&ep->clock_ref->locked)) + ep->clock_ref->rate = 0; ep->iface = 0; ep->altsetting = 0; ep->cur_audiofmt = NULL; -- 2.35.3

2 1

[PATCH] ALSA: usb-audio: Don't refcount multiple accesses on the single clock
by Jason A. Donenfeld 07 Sep '22

07 Sep '22

This reverts commit 03a8b0df757f1beb21ba1626e23ca7412e48b525. This reverts commit c11117b634f4f832c4420d3cf41c44227f140ce1. Pipewire and PulseAudio start devices with 44.1khz before changing them to 48khz (or something different). By locking the rate, daemons are unable to enumerate possible rates, and so they never change them to a more optimal rate. This revert patch should allow 48khz audio again. Fixes: c11117b634f4 ("ALSA: usb-audio: Refcount multiple accesses on the single clock") Cc: stable(a)vger.kernel.org Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Niklāvs Koļesņikovs <89q1r14hd(a)relay.firefox.com> Cc: Wim Taymans <wtaymans(a)redhat.com> Link: https://lore.kernel.org/all/YxXIWv8dYmg1tnXP@zx2c4.com/ Link: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/2620 Signed-off-by: Jason A. Donenfeld <Jason(a)zx2c4.com> --- sound/usb/card.c | 1 - sound/usb/card.h | 3 +- sound/usb/endpoint.c | 90 +++++--------------------------------------- sound/usb/usbaudio.h | 1 - 4 files changed, 10 insertions(+), 85 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index d356743de2ff..12bca2e85505 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -643,7 +643,6 @@ static int snd_usb_audio_create(struct usb_interface *intf, INIT_LIST_HEAD(&chip->pcm_list); INIT_LIST_HEAD(&chip->ep_list); INIT_LIST_HEAD(&chip->iface_ref_list); - INIT_LIST_HEAD(&chip->clock_ref_list); INIT_LIST_HEAD(&chip->midi_list); INIT_LIST_HEAD(&chip->mixer_list); diff --git a/sound/usb/card.h b/sound/usb/card.h index ca75f2206170..87f042d06ce0 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -44,7 +44,6 @@ struct audioformat { struct snd_usb_substream; struct snd_usb_iface_ref; -struct snd_usb_clock_ref; struct snd_usb_endpoint; struct snd_usb_power_domain; @@ -63,7 +62,6 @@ struct snd_urb_ctx { struct snd_usb_endpoint { struct snd_usb_audio *chip; struct snd_usb_iface_ref *iface_ref; - struct snd_usb_clock_ref *clock_ref; int opened; /* open refcount; protect with chip->mutex */ atomic_t running; /* running status */ @@ -140,6 +138,7 @@ struct snd_usb_endpoint { unsigned int cur_period_frames; unsigned int cur_period_bytes; unsigned int cur_buffer_periods; + unsigned char cur_clock; spinlock_t lock; struct list_head list; diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 0d7b73bf7945..0f5e9224a799 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -35,14 +35,6 @@ struct snd_usb_iface_ref { struct list_head list; }; -/* clock refcounting */ -struct snd_usb_clock_ref { - unsigned char clock; - atomic_t locked; - int rate; - struct list_head list; -}; - /* * snd_usb_endpoint is a model that abstracts everything related to an * USB endpoint and its streaming. @@ -599,25 +591,6 @@ iface_ref_find(struct snd_usb_audio *chip, int iface) return ip; } -/* Similarly, a refcount object for clock */ -static struct snd_usb_clock_ref * -clock_ref_find(struct snd_usb_audio *chip, int clock) -{ - struct snd_usb_clock_ref *ref; - - list_for_each_entry(ref, &chip->clock_ref_list, list) - if (ref->clock == clock) - return ref; - - ref = kzalloc(sizeof(*ref), GFP_KERNEL); - if (!ref) - return NULL; - ref->clock = clock; - atomic_set(&ref->locked, 0); - list_add_tail(&ref->list, &chip->clock_ref_list); - return ref; -} - /* * Get the existing endpoint object corresponding EP * Returns NULL if not present. @@ -795,14 +768,6 @@ snd_usb_endpoint_open(struct snd_usb_audio *chip, goto unlock; } - if (fp->protocol != UAC_VERSION_1) { - ep->clock_ref = clock_ref_find(chip, fp->clock); - if (!ep->clock_ref) { - ep = NULL; - goto unlock; - } - } - ep->cur_audiofmt = fp; ep->cur_channels = fp->channels; ep->cur_rate = params_rate(params); @@ -812,6 +777,7 @@ snd_usb_endpoint_open(struct snd_usb_audio *chip, ep->cur_period_frames = params_period_size(params); ep->cur_period_bytes = ep->cur_period_frames * ep->cur_frame_bytes; ep->cur_buffer_periods = params_periods(params); + ep->cur_clock = fp->clock; if (ep->type == SND_USB_ENDPOINT_TYPE_SYNC) endpoint_set_syncinterval(chip, ep); @@ -928,8 +894,8 @@ void snd_usb_endpoint_close(struct snd_usb_audio *chip, ep->altsetting = 0; ep->cur_audiofmt = NULL; ep->cur_rate = 0; + ep->cur_clock = 0; ep->iface_ref = NULL; - ep->clock_ref = NULL; usb_audio_dbg(chip, "EP 0x%x closed\n", ep->ep_num); } mutex_unlock(&chip->mutex); @@ -941,8 +907,6 @@ void snd_usb_endpoint_suspend(struct snd_usb_endpoint *ep) ep->need_setup = true; if (ep->iface_ref) ep->iface_ref->need_setup = true; - if (ep->clock_ref) - ep->clock_ref->rate = 0; } /* @@ -1350,33 +1314,6 @@ static int snd_usb_endpoint_set_params(struct snd_usb_audio *chip, return 0; } -static int init_sample_rate(struct snd_usb_audio *chip, - struct snd_usb_endpoint *ep) -{ - struct snd_usb_clock_ref *clock = ep->clock_ref; - int err; - - if (clock) { - if (atomic_read(&clock->locked)) - return 0; - if (clock->rate == ep->cur_rate) - return 0; - if (clock->rate && clock->rate != ep->cur_rate) { - usb_audio_dbg(chip, "Mismatched sample rate %d vs %d for EP 0x%x\n", - clock->rate, ep->cur_rate, ep->ep_num); - return -EINVAL; - } - } - - err = snd_usb_init_sample_rate(chip, ep->cur_audiofmt, ep->cur_rate); - if (err < 0) - return err; - - if (clock) - clock->rate = ep->cur_rate; - return 0; -} - /* * snd_usb_endpoint_configure: Configure the endpoint * @@ -1406,7 +1343,8 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip, * to update at each EP configuration */ if (ep->cur_audiofmt->protocol == UAC_VERSION_1) { - err = init_sample_rate(chip, ep); + err = snd_usb_init_sample_rate(chip, ep->cur_audiofmt, + ep->cur_rate); if (err < 0) goto unlock; } @@ -1436,7 +1374,7 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip, if (err < 0) goto unlock; - err = init_sample_rate(chip, ep); + err = snd_usb_init_sample_rate(chip, ep->cur_audiofmt, ep->cur_rate); if (err < 0) goto unlock; @@ -1469,15 +1407,15 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip, /* get the current rate set to the given clock by any endpoint */ int snd_usb_endpoint_get_clock_rate(struct snd_usb_audio *chip, int clock) { - struct snd_usb_clock_ref *ref; + struct snd_usb_endpoint *ep; int rate = 0; if (!clock) return 0; mutex_lock(&chip->mutex); - list_for_each_entry(ref, &chip->clock_ref_list, list) { - if (ref->clock == clock) { - rate = ref->rate; + list_for_each_entry(ep, &chip->ep_list, list) { + if (ep->cur_clock == clock && ep->cur_rate) { + rate = ep->cur_rate; break; } } @@ -1518,9 +1456,6 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) if (atomic_inc_return(&ep->running) != 1) return 0; - if (ep->clock_ref) - atomic_inc(&ep->clock_ref->locked); - ep->active_mask = 0; ep->unlink_mask = 0; ep->phase = 0; @@ -1630,9 +1565,6 @@ void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep, bool keep_pending) if (ep->sync_source) WRITE_ONCE(ep->sync_source->sync_sink, NULL); stop_urbs(ep, false, keep_pending); - if (ep->clock_ref) - if (!atomic_dec_return(&ep->clock_ref->locked)) - ep->clock_ref->rate = 0; } } @@ -1659,16 +1591,12 @@ void snd_usb_endpoint_free_all(struct snd_usb_audio *chip) { struct snd_usb_endpoint *ep, *en; struct snd_usb_iface_ref *ip, *in; - struct snd_usb_clock_ref *cp, *cn; list_for_each_entry_safe(ep, en, &chip->ep_list, list) kfree(ep); list_for_each_entry_safe(ip, in, &chip->iface_ref_list, list) kfree(ip); - - list_for_each_entry_safe(cp, cn, &chip->clock_ref_list, list) - kfree(cp); } /* diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index ffbb4b0d09a0..044cd7ab27cb 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -45,7 +45,6 @@ struct snd_usb_audio { struct list_head pcm_list; /* list of pcm streams */ struct list_head ep_list; /* list of audio-related endpoints */ struct list_head iface_ref_list; /* list of interface refcounts */ - struct list_head clock_ref_list; /* list of clock refcounts */ int pcm_devs; struct list_head midi_list; /* list of midi interfaces */ -- 2.37.3

2 7

[PATCH] ALSA: hda/ca0132: fix potential memory leak in dspxfr_image()
by Jianglei Nie 07 Sep '22

07 Sep '22

dspxfr_image() allocates DSP ports for the download stream with dsp_allocate_ports_format(). When gets some error, the allocated DSP ports are not released, which will lead to a memory leak. We can fix it by releasing DSP ports with dsp_free_ports() when getting some error. Signed-off-by: Jianglei Nie <niejianglei2021(a)163.com> --- sound/pci/hda/patch_ca0132.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 208933792787..6b8f45e14075 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -3455,6 +3455,7 @@ static int dspxfr_image(struct hda_codec *codec, &port_map_mask); if (status < 0) { codec_dbg(codec, "alloc ports fail\n"); + dsp_free_ports(codec); goto exit; } @@ -3463,6 +3464,7 @@ static int dspxfr_image(struct hda_codec *codec, WIDGET_CHIP_CTRL, stream_id, 0, &response); if (status < 0) { codec_dbg(codec, "set stream chan fail\n"); + dsp_free_ports(codec); goto exit; } @@ -3470,6 +3472,7 @@ static int dspxfr_image(struct hda_codec *codec, if (!is_valid(fls_data)) { codec_dbg(codec, "FLS check fail\n"); status = -EINVAL; + dsp_free_ports(codec); goto exit; } status = dspxfr_one_seg(codec, fls_data, reloc, -- 2.25.1

2 1

Query for snd_pcm_readi function.
by Web X 07 Sep '22

07 Sep '22

Hi, I am using alsa open source library for my project to read and write audio data. For that I am using snd_pcm_readi function to read audio data. I am able to read audio data. But issue is that sometimes it will take so much time to read data.(200-230 msec). Normally this function executed in 1-3 ms that is normal case, but after every 2-4 frames it is taking 200-300 msec. So this is affecting my whole application process. I have also try to set this call as nonblock with the help of snd_pcm_nonblock function. With that I have observed some improvements but still after 2-3 frames it is taking 100msec time to get a data. Please do needful as soon as possible. Let me know if you want more details.

2 1

[PATCH] ALSA: emu10k1: Fix out of bounds access in snd_emu10k1_pcm_channel_alloc()
by Tasos Sahanidis 07 Sep '22

07 Sep '22

The voice allocator sometimes begins allocating from near the end of the array and then wraps around, however snd_emu10k1_pcm_channel_alloc() accesses the newly allocated voices as if it never wrapped around. This results in out of bounds access if the first voice has a high enough index so that first_voice + requested_voice_count > NUM_G (64). The more voices are requested, the more likely it is for this to occur. This was initially discovered using PipeWire, however it can be reproduced by calling aplay multiple times with 16 channels: aplay -r 48000 -D plughw:CARD=Live,DEV=3 -c 16 /dev/zero UBSAN: array-index-out-of-bounds in sound/pci/emu10k1/emupcm.c:127:40 index 65 is out of range for type 'snd_emu10k1_voice [64]' CPU: 1 PID: 31977 Comm: aplay Tainted: G W IOE 6.0.0-rc2-emu10k1+ #7 Hardware name: ASUSTEK COMPUTER INC P5W DH Deluxe/P5W DH Deluxe, BIOS 3002 07/22/2010 Call Trace: <TASK> dump_stack_lvl+0x49/0x63 dump_stack+0x10/0x16 ubsan_epilogue+0x9/0x3f __ubsan_handle_out_of_bounds.cold+0x44/0x49 snd_emu10k1_playback_hw_params+0x3bc/0x420 [snd_emu10k1] snd_pcm_hw_params+0x29f/0x600 [snd_pcm] snd_pcm_common_ioctl+0x188/0x1410 [snd_pcm] ? exit_to_user_mode_prepare+0x35/0x170 ? do_syscall_64+0x69/0x90 ? syscall_exit_to_user_mode+0x26/0x50 ? do_syscall_64+0x69/0x90 ? exit_to_user_mode_prepare+0x35/0x170 snd_pcm_ioctl+0x27/0x40 [snd_pcm] __x64_sys_ioctl+0x95/0xd0 do_syscall_64+0x5c/0x90 ? do_syscall_64+0x69/0x90 ? do_syscall_64+0x69/0x90 entry_SYSCALL_64_after_hwframe+0x63/0xcd Signed-off-by: Tasos Sahanidis <tasos(a)tasossah.com> --- sound/pci/emu10k1/emupcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index b2701a4452d8..48af77ae8020 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -124,7 +124,7 @@ static int snd_emu10k1_pcm_channel_alloc(struct snd_emu10k1_pcm * epcm, int voic epcm->voices[0]->epcm = epcm; if (voices > 1) { for (i = 1; i < voices; i++) { - epcm->voices[i] = &epcm->emu->voices[epcm->voices[0]->number + i]; + epcm->voices[i] = &epcm->emu->voices[(epcm->voices[0]->number + i) % NUM_G]; epcm->voices[i]->epcm = epcm; } } -- 2.25.1

2 1

Patch for fixing top firing speakers for Dell Inspiron 16 Plus (7620) laptop
by Miroslav Lakota 07 Sep '22

07 Sep '22

Dear Sir or Madam, I would like to submit a kernel patch fixing speakers on the laptop Dell Inspiron 16 Plus 7620. The patch has been tested for a month with 5.19 and 6.0RC kernels. The current issue: - The top firing speakers on the laptop do not work. They are not detected by the audio driver. The solution: - assign correct pins and DACs to the ALC chip Status after application of the patch: - Both bottom and top firing speakers work as expected. One small issue remains - the top firing speakers are enabled with a 0,5s delay. This is however not very noticeable in real life and the fact that they work already means a significant improvement. I am completely new to submiting kernel patches so I apologise beforehand if I made any mistake in the formal process. Best regards, Miroslav Lakota >From 6f91aba8189a0b04fd977d218321e89df7e0673e Mon Sep 17 00:00:00 2001 From: Philipp Jungkamp <p.jungkamp(a)gmx.net> Date: Mon, 15 Aug 2022 11:17:21 +0200 Subject: [PATCH] ALSA: patch_realtek: Fix Dell Inspiron Plus 16 --- sound/pci/hda/patch_realtek.c | 48 +++++++++++++++++++++++++++++++++++ 1 file changed, 48 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fd630d62b5a0..9b0b4ddcc36d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6854,6 +6854,46 @@ static void alc287_fixup_yoga9_14iap7_bass_spk_pin(struct hda_codec *codec, } } +static void alc295_fixup_dell_inspiron_top_speakers(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + /* + * The Pin Complex 0x17 for the top speakers is wrongly reported as + * unconnected. + * The top speakers should also have a sequence number lower than + * the bottom speakers on NID 0x14. + */ + static const struct hda_pintbl pincfgs[] = { + { 0x14, 0x90170151 }, + { 0x17, 0x90170150 }, + { } + }; + /* + * Avoid DAC 0x06 and 0x08, as they have no volume controls. + * DAC 0x02 and 0x03 would be fine. + */ + static const hda_nid_t conn[] = { 0x02, 0x03 }; + /* + * Prefer both bottom speakers (0x14) and headphones (0x21) connected to DAC 0x02. + * Top speakers (0x21) are connected to DAC 0x03. + */ + static const hda_nid_t preferred_pairs[] = { + 0x14, 0x02, + 0x17, 0x03, + 0x21, 0x02, + 0 + }; + struct alc_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_apply_pincfgs(codec, pincfgs); + snd_hda_override_conn_list(codec, 0x17, ARRAY_SIZE(conn), conn); + spec->gen.preferred_dacs = preferred_pairs; + break; + } +} + enum { ALC269_FIXUP_GPIO2, ALC269_FIXUP_SONY_VAIO, @@ -7094,6 +7134,7 @@ enum { ALC287_FIXUP_LEGION_16ITHG6, ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK, ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN, + ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -9017,6 +9058,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK, }, + [ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc295_fixup_dell_inspiron_top_speakers, + .chained = true, + .chain_id = ALC269_FIXUP_DELL4_MIC_NO_PRESENCE, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -9116,6 +9163,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0a9d, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0a9e, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0b19, "Dell XPS 15 9520", ALC289_FIXUP_DUAL_SPK), + SND_PCI_QUIRK(0x1028, 0x0b71, "Dell Inspiron 16 Plus 7620", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- 2.37.2

2 1

[linux-next:master] BUILD REGRESSION 840126e36e8ff272cb63158646433fa1324533d9
by kernel test robot 07 Sep '22

07 Sep '22

tree/branch: https://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git master branch HEAD: 840126e36e8ff272cb63158646433fa1324533d9 Add linux-next specific files for 20220906 Error/Warning reports: https://lore.kernel.org/linux-mm/202209021204.DcLzOllr-lkp@intel.com https://lore.kernel.org/linux-mm/202209042337.FQi69rLV-lkp@intel.com https://lore.kernel.org/linux-mm/202209060229.dVuyxjBv-lkp@intel.com https://lore.kernel.org/linux-mm/202209070728.o3stvgVt-lkp@intel.com https://lore.kernel.org/llvm/202208312208.HjwleIeN-lkp@intel.com Error/Warning: (recently discovered and may have been fixed) ERROR: modpost: "__aeabi_ldivmod" [drivers/gpu/drm/vkms/vkms.ko] undefined! ERROR: modpost: "__aeabi_uldivmod" [drivers/gpu/drm/vkms/vkms.ko] undefined! ERROR: modpost: "__divdi3" [drivers/gpu/drm/vkms/vkms.ko] undefined! ERROR: modpost: "__udivdi3" [drivers/gpu/drm/vkms/vkms.ko] undefined! arm-linux-gnueabi-ld: vkms_formats.c:(.text+0x1e98): undefined reference to `__divdi3' drivers/base/regmap/regmap-mmio.c:221:17: error: implicit declaration of function 'writesb'; did you mean 'writeb'? [-Werror=implicit-function-declaration] drivers/base/regmap/regmap-mmio.c:224:17: error: implicit declaration of function 'writesw'; did you mean 'writew'? [-Werror=implicit-function-declaration] drivers/base/regmap/regmap-mmio.c:227:17: error: implicit declaration of function 'writesl'; did you mean 'writel'? [-Werror=implicit-function-declaration] drivers/base/regmap/regmap-mmio.c:231:17: error: implicit declaration of function 'writesq'; did you mean 'writeq'? [-Werror=implicit-function-declaration] drivers/base/regmap/regmap-mmio.c:231:17: error: implicit declaration of function 'writesq'; did you mean 'writesl'? [-Werror=implicit-function-declaration] drivers/base/regmap/regmap-mmio.c:358:17: error: implicit declaration of function 'readsb'; did you mean 'readb'? [-Werror=implicit-function-declaration] drivers/base/regmap/regmap-mmio.c:361:17: error: implicit declaration of function 'readsw'; did you mean 'readw'? [-Werror=implicit-function-declaration] drivers/base/regmap/regmap-mmio.c:364:17: error: implicit declaration of function 'readsl'; did you mean 'readl'? [-Werror=implicit-function-declaration] drivers/base/regmap/regmap-mmio.c:368:17: error: implicit declaration of function 'readsq'; did you mean 'readq'? [-Werror=implicit-function-declaration] drivers/base/regmap/regmap-mmio.c:368:17: error: implicit declaration of function 'readsq'; did you mean 'readsl'? [-Werror=implicit-function-declaration] drivers/gpu/drm/amd/amdgpu/imu_v11_0_3.c:139:6: warning: no previous prototype for 'imu_v11_0_3_program_rlc_ram' [-Wmissing-prototypes] drivers/gpu/drm/drm_atomic_helper.c:802: warning: expecting prototype for drm_atomic_helper_check_wb_connector_state(). Prototype was for drm_atomic_helper_check_wb_encoder_state() instead drivers/gpu/drm/vkms/vkms_formats.c:(.text+0x4b0): undefined reference to `__divdi3' drivers/gpu/drm/vkms/vkms_formats.c:259: undefined reference to `__divdi3' drivers/gpu/drm/vkms/vkms_plane.c:105 vkms_plane_atomic_update() warn: variable dereferenced before check 'fb' (see line 103) drivers/scsi/qla2xxx/qla_os.c:2854:23: warning: assignment to 'struct trace_array *' from 'int' makes pointer from integer without a cast [-Wint-conversion] drivers/usb/host/ehci-platform.c:56:19: warning: 'hcd_name' defined but not used [-Wunused-const-variable=] drivers/usb/host/ohci-platform.c:44:19: warning: 'hcd_name' defined but not used [-Wunused-const-variable=] include/linux/string.h:303:42: warning: 'strnlen' specified bound 4 exceeds source size 3 [-Wstringop-overread] kernel/bpf/memalloc.c:344 bpf_mem_alloc_destroy() error: potentially dereferencing uninitialized 'c'. kismet: WARNING: unmet direct dependencies detected for PINCTRL_IMX when selected by PINCTRL_IMX8MM ld: drivers/gpu/drm/vkms/vkms_formats.c:260: undefined reference to `__divdi3' ld: vkms_formats.c:(.text+0x47f): undefined reference to `__divdi3' mips-linux-ld: vkms_formats.c:(.text+0x384): undefined reference to `__divdi3' mips-linux-ld: vkms_formats.c:(.text.argb_u16_to_RGB565+0xd0): undefined reference to `__divdi3' mipsel-linux-ld: drivers/gpu/drm/vkms/vkms_formats.c:(.text+0x4d8): undefined reference to `__divdi3' sound/soc/codecs/tas2562.c:442:13: warning: variable 'ret' set but not used [-Wunused-but-set-variable] vkms_formats.c:(.text+0x455): undefined reference to `__divdi3' vkms_formats.c:(.text.argb_u16_to_RGB565+0xb0): undefined reference to `__divdi3' Unverified Error/Warning (likely false positive, please contact us if interested): drivers/usb/host/ehci-atmel.c:28:19: warning: unused variable 'hcd_name' [-Wunused-const-variable] drivers/usb/host/ehci-exynos.c:35:19: warning: unused variable 'hcd_name' [-Wunused-const-variable] drivers/usb/host/ehci-npcm7xx.c:27:19: warning: unused variable 'hcd_name' [-Wunused-const-variable] drivers/usb/host/ehci-orion.c:68:19: warning: unused variable 'hcd_name' [-Wunused-const-variable] drivers/usb/host/ehci-platform.c:56:19: warning: unused variable 'hcd_name' [-Wunused-const-variable] drivers/usb/host/ehci-spear.c:27:19: warning: unused variable 'hcd_name' [-Wunused-const-variable] drivers/usb/host/ohci-platform.c:44:19: warning: unused variable 'hcd_name' [-Wunused-const-variable] Error/Warning ids grouped by kconfigs: gcc_recent_errors |-- alpha-allyesconfig | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-readsb | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-readsl | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-readsq | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-readsw | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-writesb | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-writesl | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-writesq | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-writesw | |-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-imu_v11_0_3_program_rlc_ram | `-- drivers-scsi-qla2xxx-qla_os.c:warning:assignment-to-struct-trace_array-from-int-makes-pointer-from-integer-without-a-cast |-- alpha-buildonly-randconfig-r001-20220906 | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-readsb | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-readsl | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-readsq | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-readsw | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-writesb | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-writesl | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-writesq | |-- drivers-base-regmap-regmap-mmio.c:error:implicit-declaration-of-function-writesw | `-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-imu_v11_0_3_program_rlc_ram |-- arc-allyesconfig | `-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-imu_v11_0_3_program_rlc_ram |-- arm-allyesconfig | `-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-imu_v11_0_3_program_rlc_ram |-- arm-randconfig-r006-20220906 | |-- ERROR:__aeabi_ldivmod-drivers-gpu-drm-vkms-vkms.ko-undefined | `-- ERROR:__aeabi_uldivmod-drivers-gpu-drm-vkms-vkms.ko-undefined |-- arm-randconfig-r013-20220905 | `-- arm-linux-gnueabi-ld:vkms_formats.c:(.text):undefined-reference-to-__divdi3 |-- arm64-allyesconfig | `-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-imu_v11_0_3_program_rlc_ram |-- csky-randconfig-r006-20220906 | `-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-imu_v11_0_3_program_rlc_ram |-- csky-randconfig-s031-20220906 | |-- ERROR:__divdi3-drivers-gpu-drm-vkms-vkms.ko-undefined | `-- kernel-exit.c:sparse:sparse:incorrect-type-in-initializer-(different-address-spaces)-expected-struct-sighand_struct-sighand-got-struct-sighand_struct-noderef-__rcu-sighand |-- i386-allyesconfig | `-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-imu_v11_0_3_program_rlc_ram |-- i386-randconfig-a001 | |-- drivers-usb-host-ehci-platform.c:warning:hcd_name-defined-but-not-used | `-- drivers-usb-host-ohci-platform.c:warning:hcd_name-defined-but-not-used |-- i386-randconfig-a003 | |-- ERROR:__divdi3-drivers-gpu-drm-vkms-vkms.ko-undefined | `-- ERROR:__udivdi3-drivers-gpu-drm-vkms-vkms.ko-undefined |-- i386-randconfig-a012 | |-- drivers-gpu-drm-vkms-vkms_formats.c:undefined-reference-to-__divdi3 | `-- ld:drivers-gpu-drm-vkms-vkms_formats.c:undefined-reference-to-__divdi3 |-- i386-randconfig-a014 | |-- ld:vkms_formats.c:(.text):undefined-reference-to-__divdi3 clang_recent_errors |-- arm64-randconfig-r005-20220906 | `-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-function-imu_v11_0_3_program_rlc_ram |-- hexagon-randconfig-r001-20220906 | `-- drivers-extcon-extcon-usbc-tusb320.c:warning:expecting-prototype-for-drivers-extcon-extcon-tusb320c().-Prototype-was-for-TUSB320_REG8()-instead |-- hexagon-randconfig-r045-20220907 | |-- drivers-usb-host-ehci-exynos.c:warning:unused-variable-hcd_name | |-- drivers-usb-host-ehci-npcm7xx.c:warning:unused-variable-hcd_name | `-- drivers-usb-host-ehci-orion.c:warning:unused-variable-hcd_name |-- i386-randconfig-a002 | `-- drivers-extcon-extcon-usbc-tusb320.c:warning:expecting-prototype-for-drivers-extcon-extcon-tusb320c().-Prototype-was-for-TUSB320_REG8()-instead |-- i386-randconfig-a011 | |-- drivers-usb-host-ehci-platform.c:warning:unused-variable-hcd_name | `-- drivers-usb-host-ohci-platform.c:warning:unused-variable-hcd_name |-- i386-randconfig-a015 | `-- drivers-extcon-extcon-usbc-tusb320.c:warning:expecting-prototype-for-drivers-extcon-extcon-tusb320c().-Prototype-was-for-TUSB320_REG8()-instead |-- riscv-randconfig-r004-20220906 | `-- drivers-gpu-drm-amd-amdgpu-imu_v11_0_3.c:warning:no-previous-prototype-for-function-imu_v11_0_3_program_rlc_ram |-- riscv-randconfig-r042-20220907 | |-- drivers-usb-host-ehci-atmel.c:warning:unused-variable-hcd_name | `-- drivers-usb-host-ehci-spear.c:warning:unused-variable-hcd_name |-- s390-randconfig-r036-20220906 | |-- drivers-scsi-qla2xxx-qla_os.c:error:call-to-undeclared-function-trace_array_get_by_name-ISO-C99-and-later-do-not-support-implicit-function-declarations | |-- drivers-scsi-qla2xxx-qla_os.c:error:call-to-undeclared-function-trace_array_put-ISO-C99-and-later-do-not-support-implicit-function-declarations | `-- drivers-scsi-qla2xxx-qla_os.c:error:incompatible-integer-to-pointer-conversion-assigning-to-struct-trace_array-from-int |-- x86_64-randconfig-a003 | `-- drivers-extcon-extcon-usbc-tusb320.c:warning:expecting-prototype-for-drivers-extcon-extcon-tusb320c().-Prototype-was-for-TUSB320_REG8()-instead |-- x86_64-randconfig-a012 | `-- drivers-extcon-extcon-usbc-tusb320.c:warning:expecting-prototype-for-drivers-extcon-extcon-tusb320c().-Prototype-was-for-TUSB320_REG8()-instead `-- x86_64-randconfig-a016 `-- drivers-extcon-extcon-usbc-tusb320.c:warning:expecting-prototype-for-drivers-extcon-extcon-tusb320c().-Prototype-was-for-TUSB320_REG8()-instead elapsed time: 721m configs tested: 64 configs skipped: 2 gcc tested configs: arc randconfig-r043-20220906 riscv randconfig-r042-20220906 um x86_64_defconfig s390 randconfig-r044-20220906 um i386_defconfig i386 randconfig-a014 i386 randconfig-a001 i386 randconfig-a012 i386 defconfig i386 randconfig-a016 i386 randconfig-a003 m68k allmodconfig i386 randconfig-a005 arc allyesconfig alpha allyesconfig m68k allyesconfig x86_64 defconfig x86_64 rhel-8.3 arm defconfig x86_64 rhel-8.3-func x86_64 randconfig-a002 x86_64 randconfig-a013 x86_64 rhel-8.3-kunit x86_64 randconfig-a011 i386 allyesconfig x86_64 randconfig-a006 x86_64 rhel-8.3-kselftests x86_64 allyesconfig x86_64 rhel-8.3-kvm x86_64 rhel-8.3-syz x86_64 randconfig-a015 x86_64 randconfig-a004 ia64 allmodconfig arm allyesconfig microblaze mmu_defconfig openrisc alldefconfig arm pxa3xx_defconfig sh r7785rp_defconfig sh sh7710voipgw_defconfig arm64 allyesconfig powerpc allnoconfig powerpc allmodconfig mips allyesconfig csky allnoconfig arc allnoconfig alpha allnoconfig riscv allnoconfig i386 randconfig-c001 sh allmodconfig clang tested configs: hexagon randconfig-r041-20220906 hexagon randconfig-r045-20220906 i386 randconfig-a013 i386 randconfig-a011 i386 randconfig-a002 i386 randconfig-a015 i386 randconfig-a006 i386 randconfig-a004 x86_64 randconfig-a001 x86_64 randconfig-a016 x86_64 randconfig-a003 x86_64 randconfig-a012 x86_64 randconfig-a014 x86_64 randconfig-a005 x86_64 rhel-8.3-rust -- 0-DAY CI Kernel Test Service https://01.org/lkp

1 0

[PATCH] ASoC: fsl_asrc: Add initialization finishing check in runtime resume
by Shengjiu Wang 06 Sep '22

06 Sep '22

If the initialization is not finished, then filling input data to the FIFO may fail. So it is better to add initialization finishing check in the runtime resume for suspend & resume case. And consider the case of three instances working in parallel, increase the retry times to 50 for more initialization time. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_asrc.c | 19 ++++++++++++++++++- 1 file changed, 18 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index aa5edf32d988..bae263b90ac1 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -20,6 +20,7 @@ #define IDEAL_RATIO_DECIMAL_DEPTH 26 #define DIVIDER_NUM 64 +#define INIT_TRY_NUM 50 #define pair_err(fmt, ...) \ dev_err(&asrc->pdev->dev, "Pair %c: " fmt, 'A' + index, ##__VA_ARGS__) @@ -579,7 +580,7 @@ static void fsl_asrc_start_pair(struct fsl_asrc_pair *pair) { struct fsl_asrc *asrc = pair->asrc; enum asrc_pair_index index = pair->index; - int reg, retry = 10, i; + int reg, retry = INIT_TRY_NUM, i; /* Enable the current pair */ regmap_update_bits(asrc->regmap, REG_ASRCTR, @@ -592,6 +593,10 @@ static void fsl_asrc_start_pair(struct fsl_asrc_pair *pair) reg &= ASRCFG_INIRQi_MASK(index); } while (!reg && --retry); + /* FIXME: Doesn't treat initialization timeout as error */ + if (!retry) + dev_warn(&asrc->pdev->dev, "initialization isn't finished\n"); + /* Make the input fifo to ASRC STALL level */ regmap_read(asrc->regmap, REG_ASRCNCR, &reg); for (i = 0; i < pair->channels * 4; i++) @@ -1257,6 +1262,7 @@ static int fsl_asrc_runtime_resume(struct device *dev) { struct fsl_asrc *asrc = dev_get_drvdata(dev); struct fsl_asrc_priv *asrc_priv = asrc->private; + int reg, retry = INIT_TRY_NUM; int i, ret; u32 asrctr; @@ -1295,6 +1301,17 @@ static int fsl_asrc_runtime_resume(struct device *dev) regmap_update_bits(asrc->regmap, REG_ASRCTR, ASRCTR_ASRCEi_ALL_MASK, asrctr); + /* Wait for status of initialization for every enabled pairs */ + do { + udelay(5); + regmap_read(asrc->regmap, REG_ASRCFG, &reg); + reg = (reg >> ASRCFG_INIRQi_SHIFT(0)) & 0x7; + } while ((reg != ((asrctr >> ASRCTR_ASRCEi_SHIFT(0)) & 0x7)) && --retry); + + /* FIXME: Doesn't treat initialization timeout as error */ + if (!retry) + dev_warn(dev, "initialization isn't finished\n"); + return 0; disable_asrck_clk: -- 2.34.1

3 6

snd_pcm_readi() takes to much time.
by GitHub issues - opened 06 Sep '22

06 Sep '22

alsa-project/alsa-lib issue #270 was opened from coder0201200: I am using alsa open source library for my project to read and write audio data. For that I am using snd_pcm_readi function to read audio data. I am able to read audio data. But issue is that sometimes it will take so much time to read data.(200-230 msec). Normally this function executed in 1-3 ms that is normal case, but after every 2-4 frames it is taking 200-300 msec. So this is affecting my whole application process. I have also try to set this call as nonblock with the help of snd_pcm_nonblock function. With that I have observed some improvements but still after 2-3 frames it is taking 100msec time to get a data. Issue URL : https://github.com/alsa-project/alsa-lib/issues/270 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v1 1/2] ALSA: hda: cs35l41: Call put_device() in the scope of get_device()
by Andy Shevchenko 06 Sep '22

06 Sep '22

When put_device() is called in another function it's hard to realize that and easy to "fix" the code in a wrong way. Instead, move put_device() to be in the same scope as get_device(), so we prevent appearance of any attempts to "fix" the code. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/pci/hda/cs35l41_hda.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 15e2a0009080..12e955931044 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -1154,7 +1154,6 @@ static int cs35l41_no_acpi_dsd(struct cs35l41_hda *cs35l41, struct device *physd hw_cfg->gpio2.func = CS35L41_INTERRUPT; hw_cfg->gpio2.valid = true; hw_cfg->valid = true; - put_device(physdev); if (strncmp(hid, "CLSA0100", 8) == 0) { hw_cfg->bst_type = CS35L41_EXT_BOOST_NO_VSPK_SWITCH; @@ -1204,9 +1203,10 @@ static int cs35l41_hda_read_acpi(struct cs35l41_hda *cs35l41, const char *hid, i property = "cirrus,dev-index"; ret = device_property_count_u32(physdev, property); - if (ret <= 0) - return cs35l41_no_acpi_dsd(cs35l41, physdev, id, hid); - + if (ret <= 0) { + ret = cs35l41_no_acpi_dsd(cs35l41, physdev, id, hid); + goto err_put_physdev; + } if (ret > ARRAY_SIZE(values)) { ret = -EINVAL; goto err; @@ -1295,8 +1295,9 @@ static int cs35l41_hda_read_acpi(struct cs35l41_hda *cs35l41, const char *hid, i return 0; err: - put_device(physdev); dev_err(cs35l41->dev, "Failed property %s: %d\n", property, ret); +err_put_physdev: + put_device(physdev); return ret; } -- 2.35.1

2 2

[PATCH] ALSA: hda: beep: Simplify keep-power-at-enable behavior
by Takashi Iwai 06 Sep '22

06 Sep '22

The recent fix for IDT codecs to keep the power up while the beep is enabled can be better integrated into the beep helper code. This patch cleans up the code with refactoring. Fixes: 414d38ba8710 ("ALSA: hda/sigmatel: Keep power up while beep is enabled") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_beep.c | 15 +++++++++++++-- sound/pci/hda/hda_beep.h | 1 + sound/pci/hda/patch_sigmatel.c | 25 ++----------------------- 3 files changed, 16 insertions(+), 25 deletions(-) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 53a2b89f8983..e63621bcb214 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -118,6 +118,12 @@ static int snd_hda_beep_event(struct input_dev *dev, unsigned int type, return 0; } +static void turn_on_beep(struct hda_beep *beep) +{ + if (beep->keep_power_at_enable) + snd_hda_power_up_pm(beep->codec); +} + static void turn_off_beep(struct hda_beep *beep) { cancel_work_sync(&beep->beep_work); @@ -125,6 +131,8 @@ static void turn_off_beep(struct hda_beep *beep) /* turn off beep */ generate_tone(beep, 0); } + if (beep->keep_power_at_enable) + snd_hda_power_down_pm(beep->codec); } /** @@ -140,7 +148,9 @@ int snd_hda_enable_beep_device(struct hda_codec *codec, int enable) enable = !!enable; if (beep->enabled != enable) { beep->enabled = enable; - if (!enable) + if (enable) + turn_on_beep(beep); + else turn_off_beep(beep); return 1; } @@ -167,7 +177,8 @@ static int beep_dev_disconnect(struct snd_device *device) input_unregister_device(beep->dev); else input_free_device(beep->dev); - turn_off_beep(beep); + if (beep->enabled) + turn_off_beep(beep); return 0; } diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h index a25358a4807a..db76e3ddba65 100644 --- a/sound/pci/hda/hda_beep.h +++ b/sound/pci/hda/hda_beep.h @@ -25,6 +25,7 @@ struct hda_beep { unsigned int enabled:1; unsigned int linear_tone:1; /* linear tone for IDT/STAC codec */ unsigned int playing:1; + unsigned int keep_power_at_enable:1; /* set by driver */ struct work_struct beep_work; /* scheduled task for beep event */ struct mutex mutex; void (*power_hook)(struct hda_beep *beep, bool on); diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 7f340f18599c..a794a01a68ca 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4311,6 +4311,8 @@ static int stac_parse_auto_config(struct hda_codec *codec) if (codec->beep) { /* IDT/STAC codecs have linear beep tone parameter */ codec->beep->linear_tone = spec->linear_tone_beep; + /* keep power up while beep is enabled */ + codec->beep->keep_power_at_enable = 1; /* if no beep switch is available, make its own one */ caps = query_amp_caps(codec, nid, HDA_OUTPUT); if (!(caps & AC_AMPCAP_MUTE)) { @@ -4444,28 +4446,6 @@ static int stac_suspend(struct hda_codec *codec) return 0; } - -static int stac_check_power_status(struct hda_codec *codec, hda_nid_t nid) -{ -#ifdef CONFIG_SND_HDA_INPUT_BEEP - struct sigmatel_spec *spec = codec->spec; -#endif - int ret = snd_hda_gen_check_power_status(codec, nid); - -#ifdef CONFIG_SND_HDA_INPUT_BEEP - if (nid == spec->gen.beep_nid && codec->beep) { - if (codec->beep->enabled != spec->beep_power_on) { - spec->beep_power_on = codec->beep->enabled; - if (spec->beep_power_on) - snd_hda_power_up_pm(codec); - else - snd_hda_power_down_pm(codec); - } - ret |= spec->beep_power_on; - } -#endif - return ret; -} #else #define stac_suspend NULL #endif /* CONFIG_PM */ @@ -4478,7 +4458,6 @@ static const struct hda_codec_ops stac_patch_ops = { .unsol_event = snd_hda_jack_unsol_event, #ifdef CONFIG_PM .suspend = stac_suspend, - .check_power_status = stac_check_power_status, #endif }; -- 2.35.3

1 0

[PATCH] ALSA: hda: Once again fix regression of page allocations with IOMMU
by Takashi Iwai 06 Sep '22

06 Sep '22

The last fix for trying to recover the regression on AMD platforms, unfortunately, leaded to yet another regression: it turned out that IOMMUs don't like the usage of raw page allocations. This is yet another attempt for addressing the log saga; at this time, we re-use the existing buffer allocation mechanism with SG-pages although we require only single pages. The SG buffer allocation itself was confirmed to work for stream buffers, so it's relatively easy to adapt for other places. The only problem is: although the HD-audio code is accessing the address directly via dmab->address field, SG-pages don't set up it. For the ease of adaption, we now set up the dmab->addr field from the address of the first page as default, so that it can run with the HD-audio driver code as-is without the excessive call of snd_sgbuf_get_addr() multiple times; that's the only change in the memalloc helper side. The rest is nothing but a flip of the dma_type field in the HD-audio side. Fixes: a8d302a0b770 ("ALSA: memalloc: Revive x86-specific WC page allocations again") Reported-by: Mikhail Gavrilov <mikhail.v.gavrilov(a)gmail.com> Tested-by: Mikhail Gavrilov <mikhail.v.gavrilov(a)gmail.com> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/CABXGCsO+kB2t5QyHY-rUe76npr1m0-5JOtt8g8SiHUo34ur7… Link: https://bugzilla.kernel.org/show_bug.cgi?id=216112 Link: https://bugzilla.kernel.org/show_bug.cgi?id=216363 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/memalloc.c | 9 +++++++-- sound/pci/hda/hda_intel.c | 2 +- 2 files changed, 8 insertions(+), 3 deletions(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index b665ac66ccbe..cfcd8eff4139 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -543,10 +543,13 @@ static void *snd_dma_noncontig_alloc(struct snd_dma_buffer *dmab, size_t size) dmab->dev.need_sync = dma_need_sync(dmab->dev.dev, sg_dma_address(sgt->sgl)); p = dma_vmap_noncontiguous(dmab->dev.dev, size, sgt); - if (p) + if (p) { dmab->private_data = sgt; - else + /* store the first page address for convenience */ + dmab->addr = snd_sgbuf_get_addr(dmab, 0); + } else { dma_free_noncontiguous(dmab->dev.dev, size, sgt, dmab->dev.dir); + } return p; } @@ -780,6 +783,8 @@ static void *snd_dma_sg_fallback_alloc(struct snd_dma_buffer *dmab, size_t size) if (!p) goto error; dmab->private_data = sgbuf; + /* store the first page address for convenience */ + dmab->addr = snd_sgbuf_get_addr(dmab, 0); return p; error: diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index a77165bd92a9..b20694fd69de 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1817,7 +1817,7 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, /* use the non-cached pages in non-snoop mode */ if (!azx_snoop(chip)) - azx_bus(chip)->dma_type = SNDRV_DMA_TYPE_DEV_WC; + azx_bus(chip)->dma_type = SNDRV_DMA_TYPE_DEV_WC_SG; if (chip->driver_type == AZX_DRIVER_NVIDIA) { dev_dbg(chip->card->dev, "Enable delay in RIRB handling\n"); -- 2.35.3

1 0

[PATCH] ASoC: max98390: Remove unnecessary amp on/off conrtol
by Steve Lee 06 Sep '22

06 Sep '22

The Amp is already control in userspace before trigger calibrate function. Remove unnecessary control in calibrate function. Signed-off-by: Steve Lee <steve.lee.analog(a)gmail.com> --- sound/soc/codecs/max98390.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index 91c0bf3d76fc..99e769190568 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -826,9 +826,6 @@ static int max98390_dsm_calibrate(struct snd_soc_component *component) struct max98390_priv *max98390 = snd_soc_component_get_drvdata(component); - regmap_write(max98390->regmap, MAX98390_R203A_AMP_EN, 0x81); - regmap_write(max98390->regmap, MAX98390_R23FF_GLOBAL_EN, 0x01); - regmap_read(max98390->regmap, THERMAL_RDC_RD_BACK_BYTE1, &rdc); regmap_read(max98390->regmap, @@ -847,9 +844,6 @@ static int max98390_dsm_calibrate(struct snd_soc_component *component) dev_info(component->dev, "rdc resistance about %d.%02d ohm, reg=0x%X temp reg=0x%X\n", rdc_integer, rdc_factor, rdc_cal_result, temp); - regmap_write(max98390->regmap, MAX98390_R23FF_GLOBAL_EN, 0x00); - regmap_write(max98390->regmap, MAX98390_R203A_AMP_EN, 0x80); - return 0; } -- 2.17.1

3 4

[PATCH] ALSA: usb-audio: Fix an out-of-bounds bug in __snd_usb_parse_audio_interface()
by Dongxiang Ke 06 Sep '22

06 Sep '22

There may be a bad USB audio device with a USB ID of (0x04fa, 0x4201) and the number of it's interfaces less than 4, an out-of-bounds read bug occurs when parsing the interface descriptor for this device. Fix this by checking the number of interfaces. Signed-off-by: Dongxiang Ke <kdx.glider(a)gmail.com> --- sound/usb/stream.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/stream.c b/sound/usb/stream.c index ceb93d798182..40ce8a1cb318 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -1105,7 +1105,7 @@ static int __snd_usb_parse_audio_interface(struct snd_usb_audio *chip, * Dallas DS4201 workaround: It presents 5 altsettings, but the last * one misses syncpipe, and does not produce any sound. */ - if (chip->usb_id == USB_ID(0x04fa, 0x4201)) + if (chip->usb_id == USB_ID(0x04fa, 0x4201) && num >= 4) num = 4; for (i = 0; i < num; i++) { -- 2.30.2

2 1

[PATCH] ALSA: hda/tegra: Align BDL entry to 4KB boundary
by Mohan Kumar 06 Sep '22

06 Sep '22

AZA HW may send a burst read/write request crossing 4K memory boundary. The 4KB boundary is not guaranteed by Tegra HDA HW. Make SW change to include the flag AZX_DCAPS_4K_BDLE_BOUNDARY to align BDLE to 4K boundary. Signed-off-by: Mohan Kumar <mkumard(a)nvidia.com> --- sound/pci/hda/hda_tegra.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 7debb2c76aa6..976a112c7d00 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -474,7 +474,8 @@ MODULE_DEVICE_TABLE(of, hda_tegra_match); static int hda_tegra_probe(struct platform_device *pdev) { const unsigned int driver_flags = AZX_DCAPS_CORBRP_SELF_CLEAR | - AZX_DCAPS_PM_RUNTIME; + AZX_DCAPS_PM_RUNTIME | + AZX_DCAPS_4K_BDLE_BOUNDARY; struct snd_card *card; struct azx *chip; struct hda_tegra *hda; -- 2.17.1

2 1

[PATCH v3 0/4] ASoC platform driver for Apple MCA
by Martin Povišer 06 Sep '22

06 Sep '22

Hi all, sending what should be the final touches on Apple MCA driver. It most likely goes without saying but please do not merge the DT additions into the ASoC tree. Martin Changes since v2: - fix Mark's style nit - order compatibles in schema - redo the DT additions for t8103 - remove stray unlock in mca_be_prepare (and rename the locking patch) Link: https://lore.kernel.org/asahi/20220819125430.4920-1-povik+lin@cutebit.org/T… Changes since v1: - reflect the set_fmt provider/consumer change in 6.0-rc1 - fix the probe function requesting optional reset - unroll 'dma-names' in schema - fix binding schema/code discrepancy in register ranges - minor things Link: https://lore.kernel.org/asahi/20220808224153.3634-1-povik+lin@cutebit.org/T… Changes since 'macaudio RFC v2' [0]: - addition of locking (extra commit) - transition to set_bclk_ratio (instead of getting the bclk ratio from set_sysclk) - using shared reset control and documenting the reset in binding - formatting, comments, and a minor fix to hw driving Link: https://lore.kernel.org/asahi/20220606191910.16580-1-povik+lin@cutebit.org/… Martin Povišer (4): dt-bindings: sound: Add Apple MCA I2S transceiver arm64: dts: apple: t8103: Add MCA and its support ASoC: apple: mca: Start new platform driver ASoC: apple: mca: Add locking .../devicetree/bindings/sound/apple,mca.yaml | 131 ++ MAINTAINERS | 8 + arch/arm64/boot/dts/apple/t8103-jxxx.dtsi | 4 + arch/arm64/boot/dts/apple/t8103.dtsi | 73 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/apple/Kconfig | 9 + sound/soc/apple/Makefile | 3 + sound/soc/apple/mca.c | 1167 +++++++++++++++++ 9 files changed, 1397 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/apple,mca.yaml create mode 100644 sound/soc/apple/Kconfig create mode 100644 sound/soc/apple/Makefile create mode 100644 sound/soc/apple/mca.c -- 2.33.0

4 9

amidi: --send shouldn't require a seekable file
by GitHub issues - opened 06 Sep '22

06 Sep '22

alsa-project/alsa-utils issue #167 was opened from BasedUser: Just like `--receive`, `--send` should instead send events to a given port until the sent file returns EOF, or `amidi` is terminated. Pipes and FIFOs cannot currently be used to create a persistent MIDI source. Issue URL : https://github.com/alsa-project/alsa-utils/issues/167 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH 0/2] Untested TAS2562 power setting fixes
by Martin Povišer 05 Sep '22

05 Sep '22

The tas2562 driver does the same thing with the setting of PWR_CTRL field as the tas2764/tas2770 drivers were doing. Link: https://lore.kernel.org/alsa-devel/20220808141246.5749-1-povik+lin@cutebit.… Link: https://lore.kernel.org/alsa-devel/20220825140241.53963-1-povik+lin@cutebit… These are blindly written patches without testing since I don't have the hardware. (I even tried TI's formal sample request program but was refused there. CCing @ti.com addresses I found on other series recently submitted.) Martin Povišer (2): ASoC: tas2562: Drop conflicting set_bias_level power setting ASoC: tas2562: Fix mute/unmute sound/soc/codecs/tas2562.c | 90 ++++++++++++-------------------------- 1 file changed, 29 insertions(+), 61 deletions(-) -- 2.33.0

2 5

[PATCH v6 0/2] lib/string_helpers: Introduce parse_int_array_user()
by Cezary Rojewski 05 Sep '22

05 Sep '22

Continuation of recent upstream discussion [1] regarding user string tokenization. First, parse_int_array_user() is introduced to allow for splitting specified user string into a sequence of integers. Makes use of get_options() internally so the parsing logic is not duplicated. With that done, redundant parts of the sound driver are removed. Originally similar functionality was added for the SOF sound driver. As more users are on the horizon, it is desirable to update existing string_helpers code and provide a unified solution. Changes in v6: - minor improvements in SOF code: dropped superfluous array-empty checks as suggested by Andy Changes in v5: - fixed kernel doc for parse_int_array_user() Changes in v4: - renamed the function to parse_int_array_user() - at the name several local variable names have been reworded to match the above Changes in v3: - relocated tokenize_user_input() implementation to string_helpers as requested by Matthew Changes in v2: - reused get_options() so no parsing logic is duplicated - simplified __user variant with help of memdup_user_nul() Both suggested by Andy, thanks for thorough review [1]: https://lore.kernel.org/alsa-devel/20220707091301.1282291-1-cezary.rojewski… Cezary Rojewski (2): lib/string_helpers: Introduce parse_int_array_user() ASoC: SOF: Remove strsplit_u32() and tokenize_input() include/linux/string_helpers.h | 2 + lib/string_helpers.c | 44 +++++++++++++ sound/soc/sof/sof-client-probes.c | 104 +++++------------------------- 3 files changed, 61 insertions(+), 89 deletions(-) -- 2.25.1

2 3

[PATCH] ASoC: SOF: Introduce function sof_of_machine_select
by Chunxu Li 05 Sep '22

05 Sep '22

>From current design in sof_machine_check the SOF can only support ACPI type machine. In sof_machine_check if there is no ACPI machine exist, the function will return -ENODEV directly, that's we don't expected if we do not base on ACPI machine. So we add a new function named sof_of_machine_select that we can pass sof_machine_check and obtain info required by snd_sof_new_platform_drv. Signed-off-by: Chunxu Li <chunxu.li(a)mediatek.com> --- sound/soc/sof/sof-audio.c | 29 +++++++++++++++++++++++++++++ 1 file changed, 29 insertions(+) diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index c18e723435bd..ea9663d448eb 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -785,6 +785,28 @@ int sof_dai_get_bclk(struct snd_soc_pcm_runtime *rtd) } EXPORT_SYMBOL(sof_dai_get_bclk); +static struct snd_sof_of_mach *sof_of_machine_select(struct snd_sof_dev *sdev) +{ + struct snd_sof_pdata *sof_pdata = sdev->pdata; + const struct sof_dev_desc *desc = sof_pdata->desc; + struct snd_sof_of_mach *mach = desc->of_machines; + + if (!mach) + return NULL; + + for (; mach->compatible; mach++) { + if (of_machine_is_compatible(mach->compatible)) { + sof_pdata->tplg_filename = mach->sof_tplg_filename; + if (mach->fw_filename) + sof_pdata->fw_filename = mach->fw_filename; + + return mach; + } + } + + return NULL; +} + /* * SOF Driver enumeration. */ @@ -795,6 +817,7 @@ int sof_machine_check(struct snd_sof_dev *sdev) struct snd_soc_acpi_mach *mach; if (!IS_ENABLED(CONFIG_SND_SOC_SOF_FORCE_NOCODEC_MODE)) { + const struct snd_sof_of_mach *of_mach; /* find machine */ mach = snd_sof_machine_select(sdev); @@ -804,6 +827,12 @@ int sof_machine_check(struct snd_sof_dev *sdev) return 0; } + of_mach = sof_of_machine_select(sdev); + if (of_mach) { + sof_pdata->of_machine = of_mach; + return 0; + } + if (!IS_ENABLED(CONFIG_SND_SOC_SOF_NOCODEC)) { dev_err(sdev->dev, "error: no matching ASoC machine driver found - aborting probe\n"); return -ENODEV; -- 2.25.1

3 2

[PATCH v2] ASoC: wm_adsp: Handle optional legacy support
by Cristian Ciocaltea 05 Sep '22

05 Sep '22

The tracing capabilities for the speaker protection fw enabled via commit c55b3e46cb99 ("ASoC: wm_adsp: Add trace caps to speaker protection FW") are not be available on all platforms, such as the Valve's Steam Deck which is based on the Halo Core DSP. As a consequence, whenever the firmware is loaded, a rather misleading 'Failed to parse legacy: -19' error message is written to the kernel ring buffer: [ 288.977412] steamdeck kernel: cs35l41 spi-VLV1776:01: DSP1: Firmware version: 3 [ 288.978002] steamdeck kernel: cs35l41 spi-VLV1776:01: DSP1: cs35l41-dsp1-spk-prot.wmfw: Fri 02 Apr 2021 21:03:50 W. Europe Daylight Time [ 289.094065] steamdeck kernel: cs35l41 spi-VLV1776:01: DSP1: Firmware: 400a4 vendor: 0x2 v0.33.0, 2 algorithms [ 289.095073] steamdeck kernel: cs35l41 spi-VLV1776:01: DSP1: 0: ID cd v29.53.0 XM@94 YM@e [ 289.095665] steamdeck kernel: cs35l41 spi-VLV1776:01: DSP1: 1: ID f20b v0.0.1 XM@170 YM@0 [ 289.096275] steamdeck kernel: cs35l41 spi-VLV1776:01: DSP1: Protection: C:\Users\ocanavan\Desktop\cirrusTune_july2021.bin [ 291.172383] steamdeck kernel: cs35l41 spi-VLV1776:01: DSP1: Failed to parse legacy: -19 Update wm_adsp_buffer_init() to print a more descriptive info message when wm_adsp_buffer_parse_legacy() returns -ENODEV. Fixes: c55b3e46cb99 ("ASoC: wm_adsp: Add trace caps to speaker protection FW") Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea(a)collabora.com> --- Changes in v2: - Add a generic handling of the issue, not bound to a particular firmware, as suggested by Charles - Update commit message, including the subject, to reflect the changes sound/soc/codecs/wm_adsp.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index cfaa45ede916..8a2e9771bb50 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1602,7 +1602,9 @@ static int wm_adsp_buffer_init(struct wm_adsp *dsp) if (list_empty(&dsp->buffer_list)) { /* Fall back to legacy support */ ret = wm_adsp_buffer_parse_legacy(dsp); - if (ret) + if (ret == -ENODEV) + adsp_info(dsp, "Legacy support not available\n"); + else if (ret) adsp_warn(dsp, "Failed to parse legacy: %d\n", ret); } base-commit: 568035b01cfb107af8d2e4bd2fb9aea22cf5b868 -- 2.37.2

3 2

[PATCH] ASoC: apple: mca: Postpone requesting of DMA channels
by Martin Povišer 05 Sep '22

05 Sep '22

Move the requesting of DMA channels further down from 'probe' to 'pcm_new'. This is to spare the allocated DMA channel resources as we typically only ever use one or two of the clusters for PCM streaming. Before we would request DMA channels for all clusters. (This is prompted by a change in the Audio DMA Controller driver, which will now be allocating cache SRAM to channels.) Signed-off-by: Martin Povišer <povik+lin(a)cutebit.org> --- sound/soc/apple/mca.c | 79 +++++++++++++++++++++++++------------------ 1 file changed, 46 insertions(+), 33 deletions(-) diff --git a/sound/soc/apple/mca.c b/sound/soc/apple/mca.c index aa67d57c9a9b..69643524796e 100644 --- a/sound/soc/apple/mca.c +++ b/sound/soc/apple/mca.c @@ -885,6 +885,43 @@ static snd_pcm_uframes_t mca_pointer(struct snd_soc_component *component, return snd_dmaengine_pcm_pointer(substream); } +static struct dma_chan *mca_request_dma_channel(struct mca_cluster *cl, unsigned int stream) +{ + bool is_tx = (stream == SNDRV_PCM_STREAM_PLAYBACK); +#ifndef USE_RXB_FOR_CAPTURE + char *name = devm_kasprintf(cl->host->dev, GFP_KERNEL, + is_tx ? "tx%da" : "rx%da", cl->no); +#else + char *name = devm_kasprintf(cl->host->dev, GFP_KERNEL, + is_tx ? "tx%da" : "rx%db", cl->no); +#endif + return of_dma_request_slave_channel(cl->host->dev->of_node, name); + +} + +static void mca_pcm_free(struct snd_soc_component *component, + struct snd_pcm *pcm) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_chip(pcm); + struct mca_cluster *cl = mca_dai_to_cluster(asoc_rtd_to_cpu(rtd, 0)); + unsigned int i; + + if (rtd->dai_link->no_pcm) + return; + + for_each_pcm_streams(i) { + struct snd_pcm_substream *substream = + rtd->pcm->streams[i].substream; + + if (!substream || !cl->dma_chans[i]) + continue; + + dma_release_channel(cl->dma_chans[i]); + cl->dma_chans[i] = NULL; + } +} + + static int mca_pcm_new(struct snd_soc_component *component, struct snd_soc_pcm_runtime *rtd) { @@ -897,17 +934,21 @@ static int mca_pcm_new(struct snd_soc_component *component, for_each_pcm_streams(i) { struct snd_pcm_substream *substream = rtd->pcm->streams[i].substream; - struct dma_chan *chan = cl->dma_chans[i]; + struct dma_chan *chan; if (!substream) continue; - if (!chan) { - dev_err(component->dev, "missing DMA channel for stream %d on SERDES %d\n", - i, cl->no); + chan = mca_request_dma_channel(cl, i); + + if (IS_ERR_OR_NULL(chan)) { + dev_err(component->dev, "unable to obtain DMA channel (stream %d cluster %d): %pe\n", + i, cl->no, chan); + mca_pcm_free(component, rtd->pcm); return -EINVAL; } + cl->dma_chans[i] = chan; snd_pcm_set_managed_buffer(substream, SNDRV_DMA_TYPE_DEV_IRAM, chan->device->dev, 512 * 1024 * 6, SIZE_MAX); @@ -924,6 +965,7 @@ static const struct snd_soc_component_driver mca_component = { .trigger = mca_trigger, .pointer = mca_pointer, .pcm_construct = mca_pcm_new, + .pcm_destruct = mca_pcm_free, }; static void apple_mca_release(struct mca_data *mca) @@ -1019,7 +1061,6 @@ static int apple_mca_probe(struct platform_device *pdev) struct snd_soc_dai_driver *fe = &dai_drivers[mca->nclusters + i]; struct snd_soc_dai_driver *be = &dai_drivers[i]; - int stream; cl->host = mca; cl->no = i; @@ -1041,34 +1082,6 @@ static int apple_mca_probe(struct platform_device *pdev) goto err_release; } - for_each_pcm_streams(stream) { - struct dma_chan *chan; - bool is_tx = (stream == SNDRV_PCM_STREAM_PLAYBACK); -#ifndef USE_RXB_FOR_CAPTURE - char *name = devm_kasprintf(&pdev->dev, GFP_KERNEL, - is_tx ? "tx%da" : "rx%da", - i); -#else - char *name = devm_kasprintf(&pdev->dev, GFP_KERNEL, - is_tx ? "tx%da" : "rx%db", - i); -#endif - - chan = of_dma_request_slave_channel(pdev->dev.of_node, - name); - if (IS_ERR(chan)) { - if (PTR_ERR(chan) != -EPROBE_DEFER) - dev_err(&pdev->dev, - "no %s DMA channel: %ld\n", - name, PTR_ERR(chan)); - - ret = PTR_ERR(chan); - goto err_release; - } - - cl->dma_chans[stream] = chan; - } - fe->id = i; fe->name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "mca-pcm-%d", i); -- 2.33.0

2 1

[PATCH 0/3] ASoC: soc-pcm.c: random cleanup
by Kuninori Morimoto 05 Sep '22

05 Sep '22

Hi Mark These are not related, but random cleanup patches for soc-pcm.c Kuninori Morimoto (3): ASoC: soc-pcm.c: remove unnecessary codec2codec_close_delayed_work() ASoC: soc-pcm.c: add soc_pcm_ret() ASoC: soc-pcm.c: check fe condition at out of loop sound/soc/soc-pcm.c | 112 +++++++++++++++++++------------------------- 1 file changed, 47 insertions(+), 65 deletions(-) -- 2.25.1

2 4

[PATCH] ALSA: hda/sigmatel: Fix unused variable warning for beep power change
by Takashi Iwai 05 Sep '22

05 Sep '22

The newly added stac_check_power_status() caused a compile warning when CONFIG_SND_HDA_INPUT_BEEP is disabled. Fix it. Fixes: 414d38ba8710 ("ALSA: hda/sigmatel: Keep power up while beep is enabled") Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_sigmatel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 066bfccd2587..7f340f18599c 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4447,7 +4447,9 @@ static int stac_suspend(struct hda_codec *codec) static int stac_check_power_status(struct hda_codec *codec, hda_nid_t nid) { +#ifdef CONFIG_SND_HDA_INPUT_BEEP struct sigmatel_spec *spec = codec->spec; +#endif int ret = snd_hda_gen_check_power_status(codec, nid); #ifdef CONFIG_SND_HDA_INPUT_BEEP -- 2.35.3

1 0

[PATCH] ALSA: usb-audio: Refcount multiple accesses on the single clock
by Takashi Iwai 05 Sep '22

05 Sep '22

When a clock source is connected to multiple nodes / endpoints, the current USB-audio driver tries to set up at each time one of them is configured. Although it reads the current rate and updates only if it differs, some devices seem unhappy with this behavior and spew the errors when reading/updating the rate unnecessarily. This patch tries to reduce the redundant clock setup by introducing a refcount for each clock source. When the stream is actually running, a clock rate is "locked", and it bypasses the clock and/or refuse to change any longer. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=215934 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/card.c | 1 + sound/usb/card.h | 3 +- sound/usb/endpoint.c | 90 +++++++++++++++++++++++++++++++++++++++----- sound/usb/usbaudio.h | 1 + 4 files changed, 85 insertions(+), 10 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 376962291c4d..0fff96a5d3ab 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -635,6 +635,7 @@ static int snd_usb_audio_create(struct usb_interface *intf, INIT_LIST_HEAD(&chip->pcm_list); INIT_LIST_HEAD(&chip->ep_list); INIT_LIST_HEAD(&chip->iface_ref_list); + INIT_LIST_HEAD(&chip->clock_ref_list); INIT_LIST_HEAD(&chip->midi_list); INIT_LIST_HEAD(&chip->mixer_list); diff --git a/sound/usb/card.h b/sound/usb/card.h index 87f042d06ce0..ca75f2206170 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -44,6 +44,7 @@ struct audioformat { struct snd_usb_substream; struct snd_usb_iface_ref; +struct snd_usb_clock_ref; struct snd_usb_endpoint; struct snd_usb_power_domain; @@ -62,6 +63,7 @@ struct snd_urb_ctx { struct snd_usb_endpoint { struct snd_usb_audio *chip; struct snd_usb_iface_ref *iface_ref; + struct snd_usb_clock_ref *clock_ref; int opened; /* open refcount; protect with chip->mutex */ atomic_t running; /* running status */ @@ -138,7 +140,6 @@ struct snd_usb_endpoint { unsigned int cur_period_frames; unsigned int cur_period_bytes; unsigned int cur_buffer_periods; - unsigned char cur_clock; spinlock_t lock; struct list_head list; diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 743b8287cfcd..df5a70013a85 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -35,6 +35,14 @@ struct snd_usb_iface_ref { struct list_head list; }; +/* clock refcounting */ +struct snd_usb_clock_ref { + unsigned char clock; + atomic_t locked; + int rate; + struct list_head list; +}; + /* * snd_usb_endpoint is a model that abstracts everything related to an * USB endpoint and its streaming. @@ -591,6 +599,25 @@ iface_ref_find(struct snd_usb_audio *chip, int iface) return ip; } +/* Similarly, a refcount object for clock */ +static struct snd_usb_clock_ref * +clock_ref_find(struct snd_usb_audio *chip, int clock) +{ + struct snd_usb_clock_ref *ref; + + list_for_each_entry(ref, &chip->clock_ref_list, list) + if (ref->clock == clock) + return ref; + + ref = kzalloc(sizeof(*ref), GFP_KERNEL); + if (!ref) + return NULL; + ref->clock = clock; + atomic_set(&ref->locked, 0); + list_add_tail(&ref->list, &chip->clock_ref_list); + return ref; +} + /* * Get the existing endpoint object corresponding EP * Returns NULL if not present. @@ -768,6 +795,14 @@ snd_usb_endpoint_open(struct snd_usb_audio *chip, goto unlock; } + if (fp->protocol != UAC_VERSION_1) { + ep->clock_ref = clock_ref_find(chip, fp->clock); + if (!ep->clock_ref) { + ep = NULL; + goto unlock; + } + } + ep->cur_audiofmt = fp; ep->cur_channels = fp->channels; ep->cur_rate = params_rate(params); @@ -777,7 +812,6 @@ snd_usb_endpoint_open(struct snd_usb_audio *chip, ep->cur_period_frames = params_period_size(params); ep->cur_period_bytes = ep->cur_period_frames * ep->cur_frame_bytes; ep->cur_buffer_periods = params_periods(params); - ep->cur_clock = fp->clock; if (ep->type == SND_USB_ENDPOINT_TYPE_SYNC) endpoint_set_syncinterval(chip, ep); @@ -894,8 +928,8 @@ void snd_usb_endpoint_close(struct snd_usb_audio *chip, ep->altsetting = 0; ep->cur_audiofmt = NULL; ep->cur_rate = 0; - ep->cur_clock = 0; ep->iface_ref = NULL; + ep->clock_ref = NULL; usb_audio_dbg(chip, "EP 0x%x closed\n", ep->ep_num); } mutex_unlock(&chip->mutex); @@ -907,6 +941,8 @@ void snd_usb_endpoint_suspend(struct snd_usb_endpoint *ep) ep->need_setup = true; if (ep->iface_ref) ep->iface_ref->need_setup = true; + if (ep->clock_ref) + ep->clock_ref->rate = 0; } /* @@ -1314,6 +1350,33 @@ static int snd_usb_endpoint_set_params(struct snd_usb_audio *chip, return 0; } +static int init_sample_rate(struct snd_usb_audio *chip, + struct snd_usb_endpoint *ep) +{ + struct snd_usb_clock_ref *clock = ep->clock_ref; + int err; + + if (clock) { + if (atomic_read(&clock->locked)) + return 0; + if (clock->rate == ep->cur_rate) + return 0; + if (clock->rate && clock->rate != ep->cur_rate) { + usb_audio_dbg(chip, "Mismatched sample rate %d vs %d for EP 0x%x\n", + clock->rate, ep->cur_rate, ep->ep_num); + return -EINVAL; + } + } + + err = snd_usb_init_sample_rate(chip, ep->cur_audiofmt, ep->cur_rate); + if (err < 0) + return err; + + if (clock) + clock->rate = ep->cur_rate; + return 0; +} + /* * snd_usb_endpoint_configure: Configure the endpoint * @@ -1343,8 +1406,7 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip, * to update at each EP configuration */ if (ep->cur_audiofmt->protocol == UAC_VERSION_1) { - err = snd_usb_init_sample_rate(chip, ep->cur_audiofmt, - ep->cur_rate); + err = init_sample_rate(chip, ep); if (err < 0) goto unlock; } @@ -1374,7 +1436,7 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip, if (err < 0) goto unlock; - err = snd_usb_init_sample_rate(chip, ep->cur_audiofmt, ep->cur_rate); + err = init_sample_rate(chip, ep); if (err < 0) goto unlock; @@ -1407,15 +1469,15 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip, /* get the current rate set to the given clock by any endpoint */ int snd_usb_endpoint_get_clock_rate(struct snd_usb_audio *chip, int clock) { - struct snd_usb_endpoint *ep; + struct snd_usb_clock_ref *ref; int rate = 0; if (!clock) return 0; mutex_lock(&chip->mutex); - list_for_each_entry(ep, &chip->ep_list, list) { - if (ep->cur_clock == clock && ep->cur_rate) { - rate = ep->cur_rate; + list_for_each_entry(ref, &chip->clock_ref_list, list) { + if (ref->clock == clock) { + rate = ref->rate; break; } } @@ -1456,6 +1518,9 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) if (atomic_inc_return(&ep->running) != 1) return 0; + if (ep->clock_ref) + atomic_inc(&ep->clock_ref->locked); + ep->active_mask = 0; ep->unlink_mask = 0; ep->phase = 0; @@ -1565,6 +1630,9 @@ void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep, bool keep_pending) if (ep->sync_source) WRITE_ONCE(ep->sync_source->sync_sink, NULL); stop_urbs(ep, false, keep_pending); + if (ep->clock_ref) + if (!atomic_dec_return(&ep->clock_ref->locked)) + ep->clock_ref->rate = 0; } } @@ -1591,12 +1659,16 @@ void snd_usb_endpoint_free_all(struct snd_usb_audio *chip) { struct snd_usb_endpoint *ep, *en; struct snd_usb_iface_ref *ip, *in; + struct snd_usb_clock_ref *cp, *cn; list_for_each_entry_safe(ep, en, &chip->ep_list, list) kfree(ep); list_for_each_entry_safe(ip, in, &chip->iface_ref_list, list) kfree(ip); + + list_for_each_entry_safe(cp, cn, &chip->clock_ref_list, list) + kfree(ip); } /* diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 044cd7ab27cb..ffbb4b0d09a0 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -45,6 +45,7 @@ struct snd_usb_audio { struct list_head pcm_list; /* list of pcm streams */ struct list_head ep_list; /* list of audio-related endpoints */ struct list_head iface_ref_list; /* list of interface refcounts */ + struct list_head clock_ref_list; /* list of clock refcounts */ int pcm_devs; struct list_head midi_list; /* list of midi interfaces */ -- 2.35.3

2 2

[PATCH] MAINTAINERS: rectify entry in ARM/APPLE MACHINE SOUND DRIVERS
by Lukas Bulwahn 05 Sep '22

05 Sep '22

Commit 3df5d0d97289 ("ASoC: apple: mca: Start new platform driver") adds a new sound driver at the location "sound/soc/apple/", but it adds a file entry referring to the non-existing location "drivers/sound/apple/*". Hence, ./scripts/get_maintainer.pl --self-test=patterns complains about a broken reference. Repair this file reference in ARM/APPLE MACHINE SOUND DRIVERS. Signed-off-by: Lukas Bulwahn <lukas.bulwahn(a)gmail.com> --- Martin, please ack. Mark, please pick this patch on top of the commit above. MAINTAINERS | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index 29443aad5031..c239d7e69158 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -1904,7 +1904,7 @@ L: asahi(a)lists.linux.dev L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Maintained F: Documentation/devicetree/bindings/sound/apple,* -F: drivers/sound/apple/* +F: sound/soc/apple/ ARM/ARTPEC MACHINE SUPPORT M: Jesper Nilsson <jesper.nilsson(a)axis.com> -- 2.17.1

2 2

[PATCH] ALSA: pcm: oss: Fix race at SNDCTL_DSP_SYNC
by Takashi Iwai 05 Sep '22

05 Sep '22

There is a small race window at snd_pcm_oss_sync() that is called from OSS PCM SNDCTL_DSP_SYNC ioctl; namely the function calls snd_pcm_oss_make_ready() at first, then takes the params_lock mutex for the rest. When the stream is set up again by another thread between them, it leads to inconsistency, and may result in unexpected results such as NULL dereference of OSS buffer as a fuzzer spotted recently. The fix is simply to cover snd_pcm_oss_make_ready() call into the same params_lock mutex with snd_pcm_oss_make_ready_locked() variant. Reported-and-tested-by: butt3rflyh4ck <butterflyhuangxx(a)gmail.com> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/CAFcO6XN7JDM4xSXGhtusQfS2mSBcx50VJKwQpCq=WeLt57aa… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/oss/pcm_oss.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 90c3a367d7de..02df915eb3c6 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -1672,14 +1672,14 @@ static int snd_pcm_oss_sync(struct snd_pcm_oss_file *pcm_oss_file) runtime = substream->runtime; if (atomic_read(&substream->mmap_count)) goto __direct; - err = snd_pcm_oss_make_ready(substream); - if (err < 0) - return err; atomic_inc(&runtime->oss.rw_ref); if (mutex_lock_interruptible(&runtime->oss.params_lock)) { atomic_dec(&runtime->oss.rw_ref); return -ERESTARTSYS; } + err = snd_pcm_oss_make_ready_locked(substream); + if (err < 0) + goto unlock; format = snd_pcm_oss_format_from(runtime->oss.format); width = snd_pcm_format_physical_width(format); if (runtime->oss.buffer_used > 0) { -- 2.35.3

2 1

A new null-ptr-deref Write bug in snd_pcm_format_set_silence
by butt3rflyh4ck 05 Sep '22

05 Sep '22

Hi, there is a new null-ptr-deref Write bug in snd_pcm_format_set_slience in sound/core/pcm_misc.c in the latest upstream kernel and can reproduce it. We call SNDCTL_DSP_SYNC and SNDCTL_DSP_SPEED in multiple threads to trigger the vulnerability. See the Call Trace: ================================================================== Call Trace: <TASK> __dump_stack lib/dump_stack.c:88 [inline] dump_stack_lvl+0xcd/0x134 lib/dump_stack.c:106 kasan_report+0xb1/0x1e0 mm/kasan/report.c:495 check_region_inline mm/kasan/generic.c:183 [inline] kasan_check_range+0x13d/0x180 mm/kasan/generic.c:189 memset+0x20/0x40 mm/kasan/shadow.c:44 snd_pcm_format_set_silence sound/core/pcm_misc.c:441 [inline] snd_pcm_format_set_silence+0x215/0x350 sound/core/pcm_misc.c:424 snd_pcm_oss_sync+0x60e/0x800 sound/core/oss/pcm_oss.c:1690 snd_pcm_oss_ioctl+0x2087/0x3420 sound/core/oss/pcm_oss.c:2634 vfs_ioctl fs/ioctl.c:51 [inline] __do_sys_ioctl fs/ioctl.c:870 [inline] __se_sys_ioctl fs/ioctl.c:856 [inline] __x64_sys_ioctl+0x193/0x200 fs/ioctl.c:856 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x35/0xb0 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x63/0xcd ================================================================== We can see the function snd_pcm_format_set_silence code below: ``` int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int samples) { int width; unsigned char *dst; const unsigned char *pat; if (!valid_format(format)) return -EINVAL; if (samples == 0) return 0; width = pcm_formats[(INT)format].phys; /* physical width */ pat = pcm_formats[(INT)format].silence; if (!width || !pat) return -EINVAL; /* signed or 1 byte data */ if (pcm_formats[(INT)format].signd == 1 || width <= 8) { unsigned int bytes = samples * width / 8; memset(data, *pat, bytes); ///// [1] ---------> data is NULL return 0; } ...... } ``` [1], the data pointer is NULL, we can know snd_pcm_format_set_silence called in line 1690 in sound/core/oss/pcm_oss.c from call stack trace. let we see code below: ``` static int snd_pcm_oss_sync(struct snd_pcm_oss_file *pcm_oss_file) { int err = 0; unsigned int saved_f_flags; struct snd_pcm_substream *substream; struct snd_pcm_runtime *runtime; snd_pcm_format_t format; unsigned long width; size_t size; substream = pcm_oss_file->streams[SNDRV_PCM_STREAM_PLAYBACK]; if (substream != NULL) { runtime = substream->runtime; if (atomic_read(&substream->mmap_count)) goto __direct; err = snd_pcm_oss_make_ready(substream); if (err < 0) return err; atomic_inc(&runtime->oss.rw_ref); if (mutex_lock_interruptible(&runtime->oss.params_lock)) { atomic_dec(&runtime->oss.rw_ref); return -ERESTARTSYS; } format = snd_pcm_oss_format_from(runtime->oss.format); width = snd_pcm_format_physical_width(format); if (runtime->oss.buffer_used > 0) { #ifdef OSS_DEBUG pcm_dbg(substream->pcm, "sync: buffer_used\n"); #endif size = (8 * (runtime->oss.period_bytes - runtime->oss.buffer_used) + 7) / width; snd_pcm_format_set_silence(format, runtime->oss.buffer + runtime->oss.buffer_used, ///// [2] size); err = snd_pcm_oss_sync1(substream, runtime->oss.period_bytes); if (err < 0) goto unlock; } else if (runtime->oss.period_ptr > 0) { ``` [2] runtime->oss.buffer + runtime->oss.buffer_used is the data pointer, but runtime->oss.buffer is NULL here but it doesn't make sense. runtime->oss.buffter is allocated by kvzalloc, if runtime->oss_buffer is NULL, it would return an ENOMEM error. Maybe I think there is a race condition, the runtime->oss.buffer is freed and set to NULL but we can use runtime->oss.buffter via ioctl. ###reproduce it [ 167.258988][T25615] ================================================================== [ 167.265917][T25615] BUG: KASAN: null-ptr-deref in snd_pcm_format_set_silence+0x215/0x350 [ 167.266704][T25615] Write of size 16383 at addr 0000000000000001 by task snd_pcm_format_/25615 [ 167.267506][T25615] [ 167.267732][T25615] CPU: 0 PID: 25615 Comm: snd_pcm_format_ Not tainted 6.0.0-rc3-00299-gd895ec7938c4 #11 [ 167.268617][T25615] Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS 1.14.0-2 04/01/2014 [ 167.269410][T25615] Call Trace: [ 167.269697][T25615] <TASK> [ 167.269977][T25615] dump_stack_lvl+0xcd/0x134 [ 167.270428][T25615] ? snd_pcm_format_set_silence+0x215/0x350 [ 167.270985][T25615] kasan_report+0xb1/0x1e0 [ 167.271405][T25615] ? snd_pcm_format_set_silence+0x215/0x350 [ 167.271964][T25615] kasan_check_range+0x13d/0x180 [ 167.272440][T25615] memset+0x20/0x40 [ 167.272809][T25615] snd_pcm_format_set_silence+0x215/0x350 [ 167.273366][T25615] snd_pcm_oss_sync+0x60e/0x800 [ 167.273831][T25615] snd_pcm_oss_ioctl+0x2087/0x3420 [ 167.274320][T25615] ? snd_pcm_oss_release+0x300/0x300 [ 167.274817][T25615] ? __fget_files+0x26a/0x440 [ 167.275262][T25615] ? bpf_lsm_file_ioctl+0x5/0x10 [ 167.275731][T25615] ? snd_pcm_oss_release+0x300/0x300 [ 167.276222][T25615] __x64_sys_ioctl+0x193/0x200 [ 167.276677][T25615] do_syscall_64+0x35/0xb0 [ 167.277108][T25615] entry_SYSCALL_64_after_hwframe+0x63/0xcd [ 167.277679][T25615] RIP: 0033:0x44af9d [ 167.278058][T25615] Code: 66 2e 0f 1f 84 00 00 00 00 00 0f 1f 00 f3 0f 1e fa 48 89 f8 48 89 f7 48 89 d6 48 89 ca 4d 89 c2 4d 89 c8 4c 8b 4c 24 08 0f 05 <48> 3d 01 f0 ff ff 73 01 c3 48 c7 c1 b8 ff ff 8 [ 167.279792][T25615] RSP: 002b:00007f0cb636a178 EFLAGS: 00000246 ORIG_RAX: 0000000000000010 [ 167.280546][T25615] RAX: ffffffffffffffda RBX: 0000000000000000 RCX: 000000000044af9d [ 167.281285][T25615] RDX: 0000000000000000 RSI: 0000000000005001 RDI: 0000000000000003 [ 167.282001][T25615] RBP: 00007f0cb636a1a0 R08: 0000000000000000 R09: 0000000000000000 [ 167.282715][T25615] R10: 0000000000000000 R11: 0000000000000246 R12: 00007ffcfea8e08e [ 167.283432][T25615] R13: 00007ffcfea8e08f R14: 0000000000000000 R15: 00007f0cb636a640 [ 167.284164][T25615] </TASK> [ 167.284453][T25615] ================================================================== If needed I would provide a reproduce. Regards, butt3rflyh4ck. -- Active Defense Lab of Venustech

2 3

[PATCH] ALSA: usb-audio: Register card at the last interface
by Takashi Iwai 04 Sep '22

04 Sep '22

The USB-audio driver matches per interface, and as default, it registers the card instance at the very first instance. This can be a problem for the devices that have multiple interfaces to be probed, as the udev rule isn't applied properly for the later appearing interfaces. Although we introduced the delayed_register option and the quirks for covering those shortcomings, it's nothing but a workaround for specific devices. This patch is an another attempt to fix the problem in a more generic way. Now the driver checks the whole USB device descriptor at the very first time when an interface is attached to a sound card. It looks at each matching interface in the descriptor and remembers the last matching one. The snd_card_register() is invoked only when this last interface is probed. After this change, the quirks for the delayed registration become superfluous, hence they are removed along with the patch. OTOH, the delayed_register option is still kept, as it might be useful for some corner cases (e.g. a special driver overtakes the interface probe from the standard driver, and the last interface probe may miss). Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/card.c | 32 +++++++++++++++++++++++++------- sound/usb/quirks.c | 42 ------------------------------------------ sound/usb/quirks.h | 2 -- sound/usb/usbaudio.h | 1 + 4 files changed, 26 insertions(+), 51 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 706d249a9ad6..3aea241435fb 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -690,7 +690,7 @@ static bool get_alias_id(struct usb_device *dev, unsigned int *id) return false; } -static bool check_delayed_register_option(struct snd_usb_audio *chip, int iface) +static int check_delayed_register_option(struct snd_usb_audio *chip) { int i; unsigned int id, inum; @@ -699,14 +699,31 @@ static bool check_delayed_register_option(struct snd_usb_audio *chip, int iface) if (delayed_register[i] && sscanf(delayed_register[i], "%x:%x", &id, &inum) == 2 && id == chip->usb_id) - return iface < inum; + return inum; } - return false; + return -1; } static const struct usb_device_id usb_audio_ids[]; /* defined below */ +/* look for the last interface that matches with our ids and remember it */ +static void find_last_interface(struct snd_usb_audio *chip) +{ + struct usb_host_config *config = chip->dev->actconfig; + struct usb_interface *intf; + int i; + + if (!config) + return; + for (i = 0; i < config->desc.bNumInterfaces; i++) { + intf = config->interface[i]; + if (usb_match_id(intf, usb_audio_ids)) + chip->last_iface = intf->altsetting[0].desc.bInterfaceNumber; + } + usb_audio_dbg(chip, "Found last interface = %d\n", chip->last_iface); +} + /* look for the corresponding quirk */ static const struct snd_usb_audio_quirk * get_alias_quirk(struct usb_device *dev, unsigned int id) @@ -813,6 +830,7 @@ static int usb_audio_probe(struct usb_interface *intf, err = -ENODEV; goto __error; } + find_last_interface(chip); } if (chip->num_interfaces >= MAX_CARD_INTERFACES) { @@ -862,11 +880,11 @@ static int usb_audio_probe(struct usb_interface *intf, chip->need_delayed_register = false; /* clear again */ } - /* we are allowed to call snd_card_register() many times, but first - * check to see if a device needs to skip it or do anything special + /* register card if we reach to the last interface or to the specified + * one given via option */ - if (!snd_usb_registration_quirk(chip, ifnum) && - !check_delayed_register_option(chip, ifnum)) { + if (check_delayed_register_option(chip) == ifnum || + chip->last_iface == ifnum) { err = snd_card_register(chip->card); if (err < 0) goto __error; diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 5b4d8f5eade2..8c3b0be909eb 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1728,48 +1728,6 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip, } } -/* - * registration quirk: - * the registration is skipped if a device matches with the given ID, - * unless the interface reaches to the defined one. This is for delaying - * the registration until the last known interface, so that the card and - * devices appear at the same time. - */ - -struct registration_quirk { - unsigned int usb_id; /* composed via USB_ID() */ - unsigned int interface; /* the interface to trigger register */ -}; - -#define REG_QUIRK_ENTRY(vendor, product, iface) \ - { .usb_id = USB_ID(vendor, product), .interface = (iface) } - -static const struct registration_quirk registration_quirks[] = { - REG_QUIRK_ENTRY(0x0951, 0x16d8, 2), /* Kingston HyperX AMP */ - REG_QUIRK_ENTRY(0x0951, 0x16ed, 2), /* Kingston HyperX Cloud Alpha S */ - REG_QUIRK_ENTRY(0x0951, 0x16ea, 2), /* Kingston HyperX Cloud Flight S */ - REG_QUIRK_ENTRY(0x0ecb, 0x1f46, 2), /* JBL Quantum 600 */ - REG_QUIRK_ENTRY(0x0ecb, 0x1f47, 2), /* JBL Quantum 800 */ - REG_QUIRK_ENTRY(0x0ecb, 0x1f4c, 2), /* JBL Quantum 400 */ - REG_QUIRK_ENTRY(0x0ecb, 0x2039, 2), /* JBL Quantum 400 */ - REG_QUIRK_ENTRY(0x0ecb, 0x203c, 2), /* JBL Quantum 600 */ - REG_QUIRK_ENTRY(0x0ecb, 0x203e, 2), /* JBL Quantum 800 */ - { 0 } /* terminator */ -}; - -/* return true if skipping registration */ -bool snd_usb_registration_quirk(struct snd_usb_audio *chip, int iface) -{ - const struct registration_quirk *q; - - for (q = registration_quirks; q->usb_id; q++) - if (chip->usb_id == q->usb_id) - return iface < q->interface; - - /* Register as normal */ - return false; -} - /* * driver behavior quirk flags */ diff --git a/sound/usb/quirks.h b/sound/usb/quirks.h index 31abb7cb01a5..f9bfd5ac7bab 100644 --- a/sound/usb/quirks.h +++ b/sound/usb/quirks.h @@ -48,8 +48,6 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip, struct audioformat *fp, int stream); -bool snd_usb_registration_quirk(struct snd_usb_audio *chip, int iface); - void snd_usb_init_quirk_flags(struct snd_usb_audio *chip); #endif /* __USBAUDIO_QUIRKS_H */ diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index ffbb4b0d09a0..2c6575029b1c 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -37,6 +37,7 @@ struct snd_usb_audio { unsigned int quirk_flags; unsigned int need_delayed_register:1; /* warn for delayed registration */ int num_interfaces; + int last_iface; int num_suspended_intf; int sample_rate_read_error; -- 2.35.3

1 0

[PATCH v5 0/2] lib/string_helpers: Introduce parse_int_array_user()
by Cezary Rojewski 04 Sep '22

04 Sep '22

Continuation of recent upstream discussion [1] regarding user string tokenization. First, parse_int_array_user() is introduced to allow for splitting specified user string into a sequence of integers. Makes use of get_options() internally so the parsing logic is not duplicated. With that done, redundant parts of the sound driver are removed. Originally similar functionality was added for the SOF sound driver. As more users are on the horizon, it is desirable to update existing string_helpers code and provide a unified solution. Changes in v5: - fixed kernel doc for parse_int_array_user() Changes in v4: - renamed the function to parse_int_array_user() - at the name several local variable names have been reworded to match the above Changes in v3: - relocated tokenize_user_input() implementation to string_helpers as requested by Matthew Changes in v2: - reused get_options() so no parsing logic is duplicated - simplified __user variant with help of memdup_user_nul() Both suggested by Andy, thanks for thorough review [1]: https://lore.kernel.org/alsa-devel/20220707091301.1282291-1-cezary.rojewski… Cezary Rojewski (2): lib/string_helpers: Introduce parse_int_array_user() ASoC: SOF: Remove strsplit_u32() and tokenize_input() include/linux/string_helpers.h | 2 + lib/string_helpers.c | 44 +++++++++++++ sound/soc/sof/sof-client-probes.c | 104 ++++++------------------------ 3 files changed, 64 insertions(+), 86 deletions(-) -- 2.25.1

2 6

[PATCH] ALSA: hda/sigmatel: Keep power up while beep is enabled
by Takashi Iwai 04 Sep '22

04 Sep '22

It seems that the beep playback doesn't work well on IDT codec devices when the codec auto-pm is enabled. Keep the power on while the beep switch is enabled. Link: https://bugzilla.suse.com/show_bug.cgi?id=1200544 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_sigmatel.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 61df4d33c48f..066bfccd2587 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -209,6 +209,7 @@ struct sigmatel_spec { /* beep widgets */ hda_nid_t anabeep_nid; + bool beep_power_on; /* SPDIF-out mux */ const char * const *spdif_labels; @@ -4443,6 +4444,26 @@ static int stac_suspend(struct hda_codec *codec) return 0; } + +static int stac_check_power_status(struct hda_codec *codec, hda_nid_t nid) +{ + struct sigmatel_spec *spec = codec->spec; + int ret = snd_hda_gen_check_power_status(codec, nid); + +#ifdef CONFIG_SND_HDA_INPUT_BEEP + if (nid == spec->gen.beep_nid && codec->beep) { + if (codec->beep->enabled != spec->beep_power_on) { + spec->beep_power_on = codec->beep->enabled; + if (spec->beep_power_on) + snd_hda_power_up_pm(codec); + else + snd_hda_power_down_pm(codec); + } + ret |= spec->beep_power_on; + } +#endif + return ret; +} #else #define stac_suspend NULL #endif /* CONFIG_PM */ @@ -4455,6 +4476,7 @@ static const struct hda_codec_ops stac_patch_ops = { .unsol_event = snd_hda_jack_unsol_event, #ifdef CONFIG_PM .suspend = stac_suspend, + .check_power_status = stac_check_power_status, #endif }; -- 2.35.3

1 0

[PATCH v2] ASoC: rsnd: Add check for rsnd_mod_power_on
by Jiasheng Jiang 02 Sep '22

02 Sep '22

As rsnd_mod_power_on() can return negative numbers, it should be better to check the return value and deal with the exception. Fixes: e7d850dd10f4 ("ASoC: rsnd: use mod base common method on SSI-parent") Signed-off-by: Jiasheng Jiang <jiasheng(a)iscas.ac.cn> --- Changelog: v1 -> v2: 1. Add check for rsnd_mod_power_on in other files. --- sound/soc/sh/rcar/ctu.c | 6 +++++- sound/soc/sh/rcar/dvc.c | 6 +++++- sound/soc/sh/rcar/mix.c | 6 +++++- sound/soc/sh/rcar/src.c | 5 ++++- sound/soc/sh/rcar/ssi.c | 4 +++- 5 files changed, 22 insertions(+), 5 deletions(-) diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c index 6156445bcb69..e39eb2ac7e95 100644 --- a/sound/soc/sh/rcar/ctu.c +++ b/sound/soc/sh/rcar/ctu.c @@ -171,7 +171,11 @@ static int rsnd_ctu_init(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) { - rsnd_mod_power_on(mod); + int ret; + + ret = rsnd_mod_power_on(mod); + if (ret < 0) + return ret; rsnd_ctu_activation(mod); diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index 5137e03a9d7c..16befcbc312c 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -186,7 +186,11 @@ static int rsnd_dvc_init(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) { - rsnd_mod_power_on(mod); + int ret; + + ret = rsnd_mod_power_on(mod); + if (ret < 0) + return ret; rsnd_dvc_activation(mod); diff --git a/sound/soc/sh/rcar/mix.c b/sound/soc/sh/rcar/mix.c index 3572c2c5686c..1de0e085804c 100644 --- a/sound/soc/sh/rcar/mix.c +++ b/sound/soc/sh/rcar/mix.c @@ -146,7 +146,11 @@ static int rsnd_mix_init(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) { - rsnd_mod_power_on(mod); + int ret; + + ret = rsnd_mod_power_on(mod); + if (ret < 0) + return ret; rsnd_mix_activation(mod); diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 0ea84ae57c6a..f832165e46bc 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -463,11 +463,14 @@ static int rsnd_src_init(struct rsnd_mod *mod, struct rsnd_priv *priv) { struct rsnd_src *src = rsnd_mod_to_src(mod); + int ret; /* reset sync convert_rate */ src->sync.val = 0; - rsnd_mod_power_on(mod); + ret = rsnd_mod_power_on(mod); + if (ret < 0) + return ret; rsnd_src_activation(mod); diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 43c5e27dc5c8..7ade6c5ed96f 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -480,7 +480,9 @@ static int rsnd_ssi_init(struct rsnd_mod *mod, ssi->usrcnt++; - rsnd_mod_power_on(mod); + ret = rsnd_mod_power_on(mod); + if (ret < 0) + return ret; rsnd_ssi_config_init(mod, io); -- 2.25.1

3 2

[PATCH v4 0/2] lib/string_helpers: Introduce parse_int_array_user()
by Cezary Rojewski 02 Sep '22

02 Sep '22

Continuation of recent upstream discussion [1] regarding user string tokenization. First, parse_int_array_user() is introduced to allow for splitting specified user string into a sequence of integers. Makes use of get_options() internally so the parsing logic is not duplicated. With that done, redundant parts of the sound driver are removed. Originally similar functionality was added for the SOF sound driver. As more users are on the horizon, it is desirable to update existing string_helpers code and provide a unified solution. Changes in v4: - renamed the function to parse_int_array_user() - at the name several local variable names have been reworded to match the above Changes in v3: - relocated tokenize_user_input() implementation to string_helpers as requested by Matthew Changes in v2: - reused get_options() so no parsing logic is duplicated - simplified __user variant with help of memdup_user_nul() Both suggested by Andy, thanks for thourough review [1]: https://lore.kernel.org/alsa-devel/20220707091301.1282291-1-cezary.rojewski… Cezary Rojewski (2): lib/string_helpers: Introduce parse_int_array_user() ASoC: SOF: Remove strsplit_u32() and tokenize_input() include/linux/string_helpers.h | 2 + lib/string_helpers.c | 45 +++++++++++++ sound/soc/sof/sof-client-probes.c | 104 ++++++------------------------ 3 files changed, 65 insertions(+), 86 deletions(-) -- 2.25.1

2 4

[PATCH v3 0/2] lib/string_helpers: Introduce tokenize_user_input()
by Cezary Rojewski 02 Sep '22

02 Sep '22

Continuation of recent upstream discussion [1] regarding user string tokenization. First, tokenize_user_input() is introduced to allow for splitting specified user string into a sequence of integers. Makes use of get_options() internally so the parsing logic is not duplicated. With that done, redundant parts of the sound driver are removed. Originally similar functionality was added for the SOF sound driver. As more users are on the horizon, it is desirable to update existing string_helpers code and provide a unified solution. Changes in v3: - relocated tokenize_user_input() implementation to string_helpers as requested by Matthew Changes in v2: - reused get_options() so no parsing logic is duplicated - simplified __user variant with help of memdup_user_nul() Both suggested by Andy, thanks for thourough review [1]: https://lore.kernel.org/alsa-devel/20220707091301.1282291-1-cezary.rojewski… Cezary Rojewski (2): lib/string_helpers: Introduce tokenize_user_input() ASoC: SOF: Remove strsplit_u32() and tokenize_input() include/linux/string_helpers.h | 2 + lib/string_helpers.c | 45 +++++++++++++++ sound/soc/sof/sof-client-probes.c | 93 ++++--------------------------- 3 files changed, 59 insertions(+), 81 deletions(-) -- 2.25.1

2 7

[PATCH v2 0/2] Introduce sof_of_machine_select
by Chunxu Li 02 Sep '22

02 Sep '22

From: "chunxu.li" <chunxu.li(a)mediatek.com> Change since v1: - remove the callback of_machine_select defined in sof-priv.h - move sof_of_machine_select to common code, and called in sof_machine_check - rename .board field to .compatible in structure snd_sof_of_mach In these patches, we introduce function sof_of_machine_select for SOF Chunxu Li (2): ASoC: SOF: Introduce function sof_of_machine_select ASoC: SOF: mediatek: Add sof_mt8186_machs for mt8186 include/sound/sof.h | 2 ++ sound/soc/sof/mediatek/mt8186/mt8186.c | 9 ++++++++ sound/soc/sof/pcm.c | 8 ++++++- sound/soc/sof/sof-audio.c | 30 ++++++++++++++++++++++++++ sound/soc/sof/sof-of-dev.h | 7 ++++++ 5 files changed, 55 insertions(+), 1 deletion(-) -- 2.25.1

3 5

[PATCH] ALSA: aloop: Fix random zeros in capture data when using jiffies timer
by Pattara Teerapong 02 Sep '22

02 Sep '22

In loopback_jiffies_timer_pos_update(), we are getting jiffies twice. First time for playback, second time for capture. Jiffies can be updated between these two calls and if the capture jiffies is larger, extra zeros will be filled in the capture buffer. Change to get jiffies once and use it for both playback and capture. Signed-off-by: Pattara Teerapong <pteerapong(a)chromium.org> --- sound/drivers/aloop.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 9b4a7cdb103a..12f12a294df5 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -605,17 +605,18 @@ static unsigned int loopback_jiffies_timer_pos_update cable->streams[SNDRV_PCM_STREAM_PLAYBACK]; struct loopback_pcm *dpcm_capt = cable->streams[SNDRV_PCM_STREAM_CAPTURE]; - unsigned long delta_play = 0, delta_capt = 0; + unsigned long delta_play = 0, delta_capt = 0, cur_jiffies; unsigned int running, count1, count2; + cur_jiffies = jiffies; running = cable->running ^ cable->pause; if (running & (1 << SNDRV_PCM_STREAM_PLAYBACK)) { - delta_play = jiffies - dpcm_play->last_jiffies; + delta_play = cur_jiffies - dpcm_play->last_jiffies; dpcm_play->last_jiffies += delta_play; } if (running & (1 << SNDRV_PCM_STREAM_CAPTURE)) { - delta_capt = jiffies - dpcm_capt->last_jiffies; + delta_capt = cur_jiffies - dpcm_capt->last_jiffies; dpcm_capt->last_jiffies += delta_capt; } -- 2.37.2.789.g6183377224-goog

2 1

retire_capture_urb: Corrected urb data len
by Sean Anderson 02 Sep '22

02 Sep '22

Hi all, I have a "FiiO DigiHug USB Audio" sound card (1852:7022) [3]. I have had no problems with the audio, but I did notice a large number of message like retire_capture_urb: 4992 callbacks suppressed in my dmesg [1]. This is caused by the "Corrected urb data len." warning. The patch adding this warning [2] makes it seem like this warning should be an uncommon occurance. However, based on the number of suppressed callbacks, this seems to be happening at a rate of around 500 Hz. Is this buggy hardware? Or is this a bug in the driver? Does there need to be a quirk? Or perhaps the warning above should be a debug instead? I'm using 5.15.0-46-generic. I know, I know; this is a vendor kernel. However, I went through the commits for sound/usb/pcm.c and didn't see any major differences. I also previously saw this issue on 5.4, so this probably isn't a regression. --Sean [1] I would have liked to post the actual message, but I was unable to figure out how to disable ratelimiting. I tried setting /proc/sys/kernel/printk_ratelimit to 0, /proc/sys/kernel/printk_ratelimit_burst to 5000, and /proc/sys/kernel/printk to 8. However, nothing seemed to have any effect (except that the callbacks suppressed messages were disabled). I find this very strange, given that the callbacks suppressed message uses KERN_WARN, and so does dev_warn_ratelimited. [2] https://lore.kernel.org/all/4B28A659.6070303@ladisch.de/T/ [3] Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.10 bDeviceClass 0 bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 8 idVendor 0x1852 GYROCOM C&C Co., LTD idProduct 0x7022 bcdDevice 0.01 iManufacturer 1 FiiO iProduct 2 DigiHug USB Audio iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 0x0182 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 500mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 3 Human Interface Device bInterfaceSubClass 0 bInterfaceProtocol 0 iInterface 0 HID Device Descriptor: bLength 9 bDescriptorType 33 bcdHID 1.00 bCountryCode 0 Not supported bNumDescriptors 1 bDescriptorType 34 Report wDescriptorLength 58 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0012 1x 18 bytes bInterval 32 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 3 FiiO USB DAC-E10 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 0x003e bInCollection 2 baInterfaceNr(0) 2 baInterfaceNr(1) 3 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 5 wTerminalType 0x0605 SPDIF interface bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 9 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0605 SPDIF interface bAssocTerminal 0 bSourceID 16 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 7 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 5 iTerminal 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 16 bSourceID 9 bControlSize 1 bmaControls(0) 0x01 Mute Control bmaControls(1) 0x02 Volume Control bmaControls(2) 0x02 Volume Control iFeature 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 7 bDelay 0 frames wFormatTag 0x0001 PCM AudioStreaming Interface Descriptor: bLength 26 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 6 Discrete tSamFreq[ 0] 8000 tSamFreq[ 1] 16000 tSamFreq[ 2] 32000 tSamFreq[ 3] 44100 tSamFreq[ 4] 48000 tSamFreq[ 5] 96000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0184 1x 388 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0002 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 2 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 7 bDelay 0 frames wFormatTag 0x0001 PCM AudioStreaming Interface Descriptor: bLength 26 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 6 Discrete tSamFreq[ 0] 8000 tSamFreq[ 1] 16000 tSamFreq[ 2] 32000 tSamFreq[ 3] 44100 tSamFreq[ 4] 48000 tSamFreq[ 5] 96000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0246 1x 582 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0002 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 9 bDelay 0 frames wFormatTag 0x0001 PCM AudioStreaming Interface Descriptor: bLength 20 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 4 Discrete tSamFreq[ 0] 32000 tSamFreq[ 1] 44100 tSamFreq[ 2] 48000 tSamFreq[ 3] 96000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0184 1x 388 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0002 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 2 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 9 bDelay 0 frames wFormatTag 0x0001 PCM AudioStreaming Interface Descriptor: bLength 20 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 4 Discrete tSamFreq[ 0] 32000 tSamFreq[ 1] 44100 tSamFreq[ 2] 48000 tSamFreq[ 3] 96000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0246 1x 582 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0002 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 3 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 9 bDelay 0 frames wFormatTag 0x2001 IEC1937_AC-3 AudioStreaming Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 3 (FORMAT_TYPE_III) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 1 Discrete tSamFreq[ 0] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x00c0 1x 192 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0002 can't get debug descriptor: Resource temporarily unavailable Device Status: 0x0000 (Bus Powered)

2 5

[PATCH] ASoC: rsnd: Add check for rsnd_mod_power_on
by Jiasheng Jiang 02 Sep '22

02 Sep '22

As rsnd_mod_power_on() can return negative numbers, it should be better to check the return value and deal with the exception. Fixes: e7d850dd10f4 ("ASoC: rsnd: use mod base common method on SSI-parent") Signed-off-by: Jiasheng Jiang <jiasheng(a)iscas.ac.cn> --- sound/soc/sh/rcar/ssi.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 43c5e27dc5c8..7ade6c5ed96f 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -480,7 +480,9 @@ static int rsnd_ssi_init(struct rsnd_mod *mod, ssi->usrcnt++; - rsnd_mod_power_on(mod); + ret = rsnd_mod_power_on(mod); + if (ret < 0) + return ret; rsnd_ssi_config_init(mod, io); -- 2.25.1

2 1

[PATCH] ALSA: usb-audio: Clean up endpoint setups at PCM prepare
by Takashi Iwai 01 Sep '22

01 Sep '22

This patch cleans up the superfluous checks and calls for setting up the endpoints at PCM prepare callback: - Drop stop_endpoints() and sync_pending_stops() calls; the stream is guaranteed to have been already stopped and synced at each PCM prepare call by ALSA PCM core - Call snd_usb_endpoint_prepare() unconditionally; the check for endpoint->need_setup is done in snd_pcm_hw_endpoint_prepare() itself - Apply snd_usb_set_format_quirk() only when the endpoint is actually set up (i.e. the return code from snd_usb_endpoint_prepare() > 0) - Move a few lines back into snd_usb_pcm_prepare(); it's even easier to follow than a small useless function Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/pcm.c | 40 ++++++++++------------------------------ 1 file changed, 10 insertions(+), 30 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index b604f7e95e82..4ed53a3dc922 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -433,35 +433,6 @@ static void close_endpoints(struct snd_usb_audio *chip, } } -static int configure_endpoints(struct snd_usb_audio *chip, - struct snd_usb_substream *subs) -{ - int err; - - if (subs->data_endpoint->need_setup) { - /* stop any running stream beforehand */ - if (stop_endpoints(subs, false)) - sync_pending_stops(subs); - if (subs->sync_endpoint) { - err = snd_usb_endpoint_prepare(chip, subs->sync_endpoint); - if (err < 0) - return err; - } - err = snd_usb_endpoint_prepare(chip, subs->data_endpoint); - if (err < 0) - return err; - snd_usb_set_format_quirk(subs, subs->cur_audiofmt); - } else { - if (subs->sync_endpoint) { - err = snd_usb_endpoint_prepare(chip, subs->sync_endpoint); - if (err < 0) - return err; - } - } - - return 0; -} - /* * hw_params callback * @@ -640,9 +611,18 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) goto unlock; } - ret = configure_endpoints(chip, subs); + if (subs->sync_endpoint) { + ret = snd_usb_endpoint_prepare(chip, subs->sync_endpoint); + if (ret < 0) + goto unlock; + } + + ret = snd_usb_endpoint_prepare(chip, subs->data_endpoint); if (ret < 0) goto unlock; + else if (ret > 0) + snd_usb_set_format_quirk(subs, subs->cur_audiofmt); + ret = 0; /* reset the pointer */ subs->buffer_bytes = frames_to_bytes(runtime, runtime->buffer_size); -- 2.35.3

1 0

[PATCH 1/2] ASoC: apple: mca: Unselect COMMON_CLK in Kconfig
by Martin Povišer 01 Sep '22

01 Sep '22

The MCA driver selects COMMON_CLK, which leads to misconfiguration on platforms with HAVE_LEGACY_CLK (under compile test). Kconfig catches the conflict with the following warning: WARNING: unmet direct dependencies detected for COMMON_CLK Depends on [n]: !HAVE_LEGACY_CLK [=y] Selected by [m]: - SND_SOC_APPLE_MCA [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && (ARCH_APPLE || COMPILE_TEST [=y]) Eventually the build fails with errors like: >> drivers/clk/clk.c:867:6: error: redefinition of 'clk_unprepare' 867 | void clk_unprepare(struct clk *clk) | ^~~~~~~~~~~~~ In file included from drivers/clk/clk.c:9: include/linux/clk.h:303:20: note: previous definition of 'clk_unprepare' with type 'void(struct clk *)' 303 | static inline void clk_unprepare(struct clk *clk) | ^~~~~~~~~~~~~ which appears to be because COMMON_CLK is selected but HAVE_CLK_PREPARE is not. In the end it seems we had no business selecting COMMON_CLK from an unrelated driver like that, so remove the selection. The linux/clk.h API is there anyway. Fixes: 3df5d0d97289 ("ASoC: apple: mca: Start new platform driver") Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Martin Povišer <povik+lin(a)cutebit.org> --- sound/soc/apple/Kconfig | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/apple/Kconfig b/sound/soc/apple/Kconfig index 0ba955657e98..793f7782e0d7 100644 --- a/sound/soc/apple/Kconfig +++ b/sound/soc/apple/Kconfig @@ -2,7 +2,6 @@ config SND_SOC_APPLE_MCA tristate "Apple Silicon MCA driver" depends on ARCH_APPLE || COMPILE_TEST select SND_DMAENGINE_PCM - select COMMON_CLK default ARCH_APPLE help This option enables an ASoC platform driver for MCA peripherals found -- 2.33.0

2 2

[PATCH] ALSA: usb-audio: Split endpoint setups for hw_params and prepare
by Takashi Iwai 01 Sep '22

01 Sep '22

One of the former changes for the endpoint management was the more consistent setup of endpoints at hw_params. snd_usb_endpoint_configure() is a single function that does the full setup, and it's called from both PCM hw_params and prepare callbacks. Although the EP setup at the prepare phase is usually skipped (by checking need_setup flag), it may be still effective in some cases like suspend/resume that requires the interface setup again. As it's a full and single setup, the invocation of snd_usb_endpoint_configure() includes not only the USB interface setup but also the buffer release and allocation. OTOH, doing the buffer release and re-allocation at PCM prepare phase is rather superfluous, and better to be done only in the hw_params phase. For those optimizations, this patch splits the endpoint setup to two phases: snd_usb_endpoint_set_params() and snd_usb_endpoint_prepare(), to be called from hw_params and from prepare, respectively. Note that this patch changes the driver operation slightly, effectively moving the USB interface setup again to PCM prepare stage instead of hw_params stage, while the buffer allocation and such initializations are still done at hw_params stage. And, the change of the USB interface setup timing (moving to prepare) gave an interesting "fix", too: it was reported that the recent kernels caused silent output at the beginning on playbacks on some devices on Android, and this change casually fixed the regression. It seems that those devices are picky about the sample rate change (or the interface change?), and don't follow the too immediate rate changes. Meanwhile, Android operates the PCM in the following order: - open, then hw_params with the possibly highest sample rate - close without prepare - re-open, hw_params with the normal sample rate - prepare, and start streaming This procedure ended up the hw_params twice with different rates, and because the recent kernel did set up the sample rate twice one and after, it screwed up the device. OTOH, the earlier kernels didn't set up the USB interface at hw_params, hence this problem didn't appear. Now, with this patch, the USB interface setup is again back to the prepare phase, and it works around the problem automagically. Although we should address the sample rate problem in a more solid way in future, let's keep things working as before for now. Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management") Cc: <stable(a)vger.kernel.org> Reported-by: chihhao chen <chihhao.chen(a)mediatek.com> Link: https://lore.kernel.org/r/87e6d6ae69d68dc588ac9acc8c0f24d6188375c3.camel@me… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/endpoint.c | 23 +++++++++-------------- sound/usb/endpoint.h | 6 ++++-- sound/usb/pcm.c | 14 ++++++++++---- 3 files changed, 23 insertions(+), 20 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 0d7b73bf7945..a42f2ce19455 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -758,7 +758,8 @@ bool snd_usb_endpoint_compatible(struct snd_usb_audio *chip, * The endpoint needs to be closed via snd_usb_endpoint_close() later. * * Note that this function doesn't configure the endpoint. The substream - * needs to set it up later via snd_usb_endpoint_configure(). + * needs to set it up later via snd_usb_endpoint_set_params() and + * snd_usb_endpoint_prepare(). */ struct snd_usb_endpoint * snd_usb_endpoint_open(struct snd_usb_audio *chip, @@ -1290,12 +1291,13 @@ static int sync_ep_set_params(struct snd_usb_endpoint *ep) /* * snd_usb_endpoint_set_params: configure an snd_usb_endpoint * + * It's called either from hw_params callback. * Determine the number of URBs to be used on this endpoint. * An endpoint must be configured before it can be started. * An endpoint that is already running can not be reconfigured. */ -static int snd_usb_endpoint_set_params(struct snd_usb_audio *chip, - struct snd_usb_endpoint *ep) +int snd_usb_endpoint_set_params(struct snd_usb_audio *chip, + struct snd_usb_endpoint *ep) { const struct audioformat *fmt = ep->cur_audiofmt; int err; @@ -1378,18 +1380,18 @@ static int init_sample_rate(struct snd_usb_audio *chip, } /* - * snd_usb_endpoint_configure: Configure the endpoint + * snd_usb_endpoint_prepare: Prepare the endpoint * * This function sets up the EP to be fully usable state. - * It's called either from hw_params or prepare callback. + * It's called either from prepare callback. * The function checks need_setup flag, and performs nothing unless needed, * so it's safe to call this multiple times. * * This returns zero if unchanged, 1 if the configuration has changed, * or a negative error code. */ -int snd_usb_endpoint_configure(struct snd_usb_audio *chip, - struct snd_usb_endpoint *ep) +int snd_usb_endpoint_prepare(struct snd_usb_audio *chip, + struct snd_usb_endpoint *ep) { bool iface_first; int err = 0; @@ -1410,9 +1412,6 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip, if (err < 0) goto unlock; } - err = snd_usb_endpoint_set_params(chip, ep); - if (err < 0) - goto unlock; goto done; } @@ -1440,10 +1439,6 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip, if (err < 0) goto unlock; - err = snd_usb_endpoint_set_params(chip, ep); - if (err < 0) - goto unlock; - err = snd_usb_select_mode_quirk(chip, ep->cur_audiofmt); if (err < 0) goto unlock; diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index 6a9af04cf175..e67ea28faa54 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -17,8 +17,10 @@ snd_usb_endpoint_open(struct snd_usb_audio *chip, bool is_sync_ep); void snd_usb_endpoint_close(struct snd_usb_audio *chip, struct snd_usb_endpoint *ep); -int snd_usb_endpoint_configure(struct snd_usb_audio *chip, - struct snd_usb_endpoint *ep); +int snd_usb_endpoint_set_params(struct snd_usb_audio *chip, + struct snd_usb_endpoint *ep); +int snd_usb_endpoint_prepare(struct snd_usb_audio *chip, + struct snd_usb_endpoint *ep); int snd_usb_endpoint_get_clock_rate(struct snd_usb_audio *chip, int clock); bool snd_usb_endpoint_compatible(struct snd_usb_audio *chip, diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index d45d1d7e6664..b604f7e95e82 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -443,17 +443,17 @@ static int configure_endpoints(struct snd_usb_audio *chip, if (stop_endpoints(subs, false)) sync_pending_stops(subs); if (subs->sync_endpoint) { - err = snd_usb_endpoint_configure(chip, subs->sync_endpoint); + err = snd_usb_endpoint_prepare(chip, subs->sync_endpoint); if (err < 0) return err; } - err = snd_usb_endpoint_configure(chip, subs->data_endpoint); + err = snd_usb_endpoint_prepare(chip, subs->data_endpoint); if (err < 0) return err; snd_usb_set_format_quirk(subs, subs->cur_audiofmt); } else { if (subs->sync_endpoint) { - err = snd_usb_endpoint_configure(chip, subs->sync_endpoint); + err = snd_usb_endpoint_prepare(chip, subs->sync_endpoint); if (err < 0) return err; } @@ -551,7 +551,13 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, subs->cur_audiofmt = fmt; mutex_unlock(&chip->mutex); - ret = configure_endpoints(chip, subs); + if (subs->sync_endpoint) { + ret = snd_usb_endpoint_set_params(chip, subs->sync_endpoint); + if (ret < 0) + goto unlock; + } + + ret = snd_usb_endpoint_set_params(chip, subs->data_endpoint); unlock: if (ret < 0) -- 2.35.3

1 0

[PATCH] ALSA: asihpi - Remove useless code in hpi_meter_get_peak()
by Valentina Goncharenko 01 Sep '22

01 Sep '22

The hpi_meter_get_peak() function contains the expression "hm.obj_index = hm.obj_index", which does not carry any semantic load. Found by Linux Verification Center (linuxtesting.org) with SVACE. Fixes: 719f82d3987a ("ALSA: Add support of AudioScience ASI boards") Signed-off-by: Valentina Goncharenko <goncharenko.vp(a)ispras.ru> --- sound/pci/asihpi/hpifunc.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/asihpi/hpifunc.c b/sound/pci/asihpi/hpifunc.c index 1de05383126a..24047fafef51 100644 --- a/sound/pci/asihpi/hpifunc.c +++ b/sound/pci/asihpi/hpifunc.c @@ -2020,7 +2020,6 @@ u16 hpi_meter_get_peak(u32 h_control, short an_peakdB[HPI_MAX_CHANNELS] HPI_CONTROL_GET_STATE); if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index)) return HPI_ERROR_INVALID_HANDLE; - hm.obj_index = hm.obj_index; hm.u.c.attribute = HPI_METER_PEAK; hpi_send_recv(&hm, &hr); -- 2.25.1

2 1

[PATCH 1/2] ASoC: codecs: wcd934x: add Slimbus dependency
by Krzysztof Kozlowski 01 Sep '22

01 Sep '22

The WCD934X codec is a Slimbus driver, so it must depend on SLIMBUS, also for compile tests: ERROR: modpost: "slim_stream_prepare" [sound/soc/codecs/snd-soc-wcd934x.ko] undefined! Reported-by: kernel test robot <lkp(a)intel.com> Fixes: 5b7f4e5de61b ("ASoC: codecs: allow compile testing without MFD drivers") Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 968d0701f2e8..2fc37f4e061c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1744,6 +1744,7 @@ config SND_SOC_WCD_MBHC config SND_SOC_WCD934X tristate "WCD9340/WCD9341 Codec" depends on COMMON_CLK + depends on SLIMBUS select SND_SOC_WCD_MBHC depends on MFD_WCD934X || COMPILE_TEST help -- 2.34.1

2 2

mt8195-demo: redefine soundcard initialization
by GitHub pull_request - opened 01 Sep '22

01 Sep '22

alsa-project/alsa-ucm-conf pull request #211 was opened from fchiby: The previous initialization in Bootsequence set all Switch to off after boot. Whithout alsactl, alsa-ucm-conf does't work. So remove the setting off of the Switch in the BootSequence and define Enable/DisableSequence in each device section for a proper setting. Signed-off-by: Fadwa Chiby <fchiby(a)baylibre.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/211 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/211.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH v2] soundwire: qcom: fix device status array range
by Srinivas Kandagatla 01 Sep '22

01 Sep '22

This patch updates device status array range from 11 to 12 as we will be reading status from device number 0 to device number 11 inclusive. Without this patch we can potentially access status array out of range during auto-enumeration. Fixes: aa1262ca6695 ("soundwire: qcom: Check device status before reading devid") Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- drivers/soundwire/qcom.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 9df970eeca45..16a4a51978ca 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -169,7 +169,7 @@ struct qcom_swrm_ctrl { u8 wcmd_id; struct qcom_swrm_port_config pconfig[QCOM_SDW_MAX_PORTS]; struct sdw_stream_runtime *sruntime[SWRM_MAX_DAIS]; - enum sdw_slave_status status[SDW_MAX_DEVICES]; + enum sdw_slave_status status[SDW_MAX_DEVICES + 1]; int (*reg_read)(struct qcom_swrm_ctrl *ctrl, int reg, u32 *val); int (*reg_write)(struct qcom_swrm_ctrl *ctrl, int reg, int val); u32 slave_status; @@ -420,7 +420,7 @@ static int qcom_swrm_get_alert_slave_dev_num(struct qcom_swrm_ctrl *ctrl) ctrl->reg_read(ctrl, SWRM_MCP_SLV_STATUS, &val); - for (dev_num = 0; dev_num < SDW_MAX_DEVICES; dev_num++) { + for (dev_num = 0; dev_num <= SDW_MAX_DEVICES; dev_num++) { status = (val >> (dev_num * SWRM_MCP_SLV_STATUS_SZ)); if ((status & SWRM_MCP_SLV_STATUS_MASK) == SDW_SLAVE_ALERT) { @@ -440,7 +440,7 @@ static void qcom_swrm_get_device_status(struct qcom_swrm_ctrl *ctrl) ctrl->reg_read(ctrl, SWRM_MCP_SLV_STATUS, &val); ctrl->slave_status = val; - for (i = 0; i < SDW_MAX_DEVICES; i++) { + for (i = 0; i <= SDW_MAX_DEVICES; i++) { u32 s; s = (val >> (i * 2)); -- 2.25.1

2 1

[PATCH 0/6] ASoC: codecs: lpass: add support fro sm8450 and sc8280xp
by Srinivas Kandagatla 01 Sep '22

01 Sep '22

THis patchset adds support for SM8450 and SC8280XP SoC. Tested SmartSpeakers and Headset on SM8450 MTP and Lenovo Thinkpad X13s. THanks, Srini Srinivas Kandagatla (6): ASoC: qcom: dt-bindings: add sm8450 and sc8280xp compatibles ASoC: codecs: wsa-macro: add support for sm8450 and sc8280xp ASoC: codecs: tx-macro: add support for sm8450 and sc8280xp ASoC: codecs: rx-macro: add support for sm8450 and sc8280xp ASoC: codecs: va-macro: clear the frame sync counter before enabling ASoC: codecs: tx-macro: add support for sm8450 and sc8280xp .../bindings/sound/qcom,lpass-rx-macro.yaml | 2 + .../bindings/sound/qcom,lpass-tx-macro.yaml | 2 + .../bindings/sound/qcom,lpass-va-macro.yaml | 2 + .../bindings/sound/qcom,lpass-wsa-macro.yaml | 2 + sound/soc/codecs/lpass-rx-macro.c | 2 + sound/soc/codecs/lpass-tx-macro.c | 2 + sound/soc/codecs/lpass-va-macro.c | 72 +++++++++++++++++-- sound/soc/codecs/lpass-wsa-macro.c | 2 + 8 files changed, 82 insertions(+), 4 deletions(-) -- 2.21.0

3 15

[PATCH] sound: soc/codecs: tlv320adc3xxx: needs GPIOLIB
by Randy Dunlap 31 Aug '22

31 Aug '22

Fix build errors when CONFIG_GPIOLIB is not enabled: ../sound/soc/codecs/tlv320adc3xxx.c: In function 'adc3xxx_i2c_probe': ../sound/soc/codecs/tlv320adc3xxx.c:1352:28: error: implicit declaration of function 'devm_gpiod_get'; did you mean 'devm_gpio_free'? [-Werror=implicit-function-declaration] 1352 | adc3xxx->rst_pin = devm_gpiod_get(dev, "reset", GPIOD_OUT_LOW); ../sound/soc/codecs/tlv320adc3xxx.c:1352:57: error: 'GPIOD_OUT_LOW' undeclared (first use in this function); did you mean 'GPIOF_INIT_LOW'? 1352 | adc3xxx->rst_pin = devm_gpiod_get(dev, "reset", GPIOD_OUT_LOW); CC lib/dynamic_debug.o ../sound/soc/codecs/tlv320adc3xxx.c:1400:9: error: implicit declaration of function 'gpiod_set_value_cansleep'; did you mean 'gpio_set_value_cansleep'? [-Werror=implicit-function-declaration] 1400 | gpiod_set_value_cansleep(adc3xxx->rst_pin, 1); Fixes: e9a3b57efd28 ("ASoC: codec: tlv320adc3xxx: New codec driver") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Reported-by: kernel test robot <lkp(a)intel.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Ricard Wanderlof <ricardw(a)axis.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1567,6 +1567,7 @@ config SND_SOC_TFA989X config SND_SOC_TLV320ADC3XXX tristate "Texas Instruments TLV320ADC3001/3101 audio ADC" depends on I2C + depends on GPIOLIB help Enable support for Texas Instruments TLV320ADC3001 and TLV320ADC3101 ADCs.

1 1

Support MSI MPG Z690 EDGE WIFI DDR4 (ALC4080)
by GitHub issues - edited 31 Aug '22

31 Aug '22

alsa-project/alsa-ucm-conf issue #210 was edited from tekstryder: Per @perexg request, filing a new issue after testing git master branch changes for the ALC4080 and finding any profile(s) to be missing/nonfunctional. USB ID: `0db0:1feb` In version 1.2.7.2 of .../ucm2/USB-Audio/Realtek/ALC4080-HiFi.conf I need to change: - Define.SPDIFIndex "3" to Define.SPDIFIndex "2" For proper playback via S/PDIF. After that change the release version is working great as far as I could test. Please let me know if I can provide any other useful information. Attaching the output of `alsaucm -c hw:2 dump text` Attaching the output of `pactl list sinks` [ucm-dump.txt](https://github.com/alsa-project/alsa-ucm-conf/files/9463734/u… [sinks.txt](https://github.com/alsa-project/alsa-ucm-conf/files/9463784/sink… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/210 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Support MSI MPG Z690 EDGE WIFI DDR4 (ALC4080)
by GitHub issues - opened 31 Aug '22

31 Aug '22

alsa-project/alsa-ucm-conf issue #210 was opened from tekstryder: Per @perexg request, filing a new issue after testing git master branch changes for the ALC4080 and finding any profile(s) to be missing/nonfunctional. USB ID: `0db0:1feb` In version 1.2.7.2 of .../ucm2/USB-Audio/Realtek/ALC4080-HiFi.conf I need to change: - Define.SPDIFIndex "3" to Define.SPDIFIndex "Define.SPDIFIndex "2"" For proper playback via S/PDIF. After that change the release version is working great as far as I could test. Please let me know if I can provide any other useful information. Attaching the output of `alsaucm -c hw:2 dump text` Attaching the output of `pactl list sinks` [ucm-dump.txt](https://github.com/alsa-project/alsa-ucm-conf/files/9463734/u… [sinks.txt](https://github.com/alsa-project/alsa-ucm-conf/files/9463784/sink… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/210 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2 0/2] libfs: Introduce tokenize_user_input()
by Cezary Rojewski 31 Aug '22

31 Aug '22

Continuation of recent upstream discussion [1] regarding user string tokenization. First, tokenize_user_input() is introduced to allow for splitting specified user string into a sequence of integers. Makes use of get_options() internally so the parsing logic is not duplicated. With that done, redundant parts of the sound driver are removed. Originally similar functionality was added for the SOF sound driver. As more users are on the horizon, it is desirable to update existing fs code and provide a unified solution. Changes in v2: - reused get_options() so no parsing logic is duplicated - simplified __user variant with help of memdup_user_nul() Both suggested by Andy, thanks for thourough review [1]: https://lore.kernel.org/alsa-devel/20220707091301.1282291-1-cezary.rojewski… Cezary Rojewski (2): libfs: Introduce tokenize_user_input() ASoC: SOF: Remove strsplit_u32() and tokenize_input() fs/libfs.c | 45 +++++++++++++++ include/linux/fs.h | 1 + sound/soc/sof/sof-client-probes.c | 92 ++++--------------------------- 3 files changed, 57 insertions(+), 81 deletions(-) -- 2.25.1

2 4

[PATCH] ALSA: usb-audio: Drop superfluous interface setup at parsing
by Takashi Iwai 31 Aug '22

31 Aug '22

We reset each interface that is being parsed for each stream, but this is superfluous and even can lead to spurious errors. Since the interface is set up properly at opening the endpoint for each actual stream operation, let's drop the superfluous one. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/stream.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/usb/stream.c b/sound/usb/stream.c index ceb93d798182..99578e9a8af0 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -1221,12 +1221,6 @@ static int __snd_usb_parse_audio_interface(struct snd_usb_audio *chip, if (err < 0) return err; } - - /* try to set the interface... */ - usb_set_interface(chip->dev, iface_no, 0); - snd_usb_init_pitch(chip, fp); - snd_usb_init_sample_rate(chip, fp, fp->rate_max); - usb_set_interface(chip->dev, iface_no, altno); } return 0; } -- 2.35.3

1 0

[PATCH 1/2] ALSA: usb-audio: Inform the delayed registration more properly
by Takashi Iwai 31 Aug '22

31 Aug '22

The info message that was added in the commit a4aad5636c72 ("ALSA: usb-audio: Inform devices that need delayed registration") is actually useful to know the need for the delayed registration. However, it turned out that this doesn't catch the all cases; namely, this warned only when a PCM stream is attached onto the existing PCM instance, but it doesn't count for a newly created PCM instance. This made confusion as if there were no further delayed registration. This patch moves the check to the code path for either adding a stream or creating a PCM instance. Also, make it simpler by checking the card->registered flag instead of querying each snd_device state. Fixes: a4aad5636c72 ("ALSA: usb-audio: Inform devices that need delayed registration") Link: https://bugzilla.kernel.org/show_bug.cgi?id=216082 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/stream.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/usb/stream.c b/sound/usb/stream.c index ceb93d798182..40b7821c6c99 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -495,6 +495,10 @@ static int __snd_usb_add_audio_stream(struct snd_usb_audio *chip, return 0; } } + + if (chip->card->registered) + chip->need_delayed_register = true; + /* look for an empty stream */ list_for_each_entry(as, &chip->pcm_list, list) { if (as->fmt_type != fp->fmt_type) @@ -502,9 +506,6 @@ static int __snd_usb_add_audio_stream(struct snd_usb_audio *chip, subs = &as->substream[stream]; if (subs->ep_num) continue; - if (snd_device_get_state(chip->card, as->pcm) != - SNDRV_DEV_BUILD) - chip->need_delayed_register = true; err = snd_pcm_new_stream(as->pcm, stream, 1); if (err < 0) return err; -- 2.35.3

1 1

[PATCH v3 1/2] ASoC: dt-bindings: max98396: Document data monitor properties
by Daniel Mack 31 Aug '22

31 Aug '22

This device features a data monitor that puts the device in software reset upon a configurable set of events. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- v1 -> v2: fix a typo and remove a stray blank line v2 -> v3: Rename device tree properties, drop $ref for standard unit suffix properties .../bindings/sound/adi,max98396.yaml | 34 +++++++++++++++++++ 1 file changed, 34 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/adi,max98396.yaml b/Documentation/devicetree/bindings/sound/adi,max98396.yaml index 8d2ef991db40..7985b1410017 100644 --- a/Documentation/devicetree/bindings/sound/adi,max98396.yaml +++ b/Documentation/devicetree/bindings/sound/adi,max98396.yaml @@ -78,6 +78,40 @@ properties: interleaved on a single output channel. type: boolean + adi,dmon-stuck-enable: + description: + Enables the "data monitor stuck" feature. Once the data monitor is + enabled, it actively monitors the selected input data (from DIN) to the + speaker amplifier. Once a data error is detected, the data monitor + automatically places the device into software shutdown. + type: boolean + + adi,dmon-stuck-threshold-bits: + description: + Sets the threshold for the "data monitor stuck" feature, in bits. + enum: [9, 11, 13, 15] + default: 15 + + adi,dmon-magnitude-enable: + description: + Enables the "data monitor magnitude" feature. Once the data monitor is + enabled, it actively monitors the selected input data (from DIN) to the + speaker amplifier. Once a data error is detected, the data monitor + automatically places the device into software shutdown. + type: boolean + + adi,dmon-magnitude-threshold-bits: + description: + Sets the threshold for the "data monitor magnitude" feature, in bits. + enum: [2, 3, 4, 5] + default: 5 + + adi,dmon-duration-ms: + description: + Sets the duration for the "data monitor" feature, in milliseconds. + enum: [64, 256, 1024, 4096] + default: 64 + reset-gpios: maxItems: 1 -- 2.37.2

3 3

[PATCH 0/5] ASoC: codecs: minor cppcheck cleanups
by Pierre-Louis Bossart 31 Aug '22

31 Aug '22

Small number of cleanups that were either missed in previous versions or detected by new cppcheck version. Pierre-Louis Bossart (5): ASoC: hdmi-codec: remove unused definitions ASoC: wcd-mbhc-v2: remove always-true condition ASoC: wcd9335: remove always-true condition ASoC: fsl: fsl-utils: remove useless assignment ASoC: ti: omap-mcbsp: remove useless assignment sound/soc/codecs/hdmi-codec.c | 4 ---- sound/soc/codecs/wcd-mbhc-v2.c | 2 +- sound/soc/codecs/wcd9335.c | 4 +--- sound/soc/fsl/fsl_utils.c | 2 +- sound/soc/ti/omap-mcbsp.c | 2 +- 5 files changed, 4 insertions(+), 10 deletions(-) -- 2.34.1

3 7

[PATCH 0/2] ASoC: SOF: compress: Add support for timestamp on capture
by Laurentiu Mihalcea 31 Aug '22

31 Aug '22

The purpose of this patch series is to add support for timestamping on capture direction using the compress API. This is simply done by splitting sof_compr_copy into 2 functions: sof_compr_copy_playback and sof_compr_copy_capture. Each of these functions handles one of the possible directions: capture or playback and is called in sof_compr_copy based on the stream's direction. The only difference between sof_compr_copy_playback and sof_compr_copy_capture is the fact that on playback case we need to copy data from user space and on capture we need to copy data to user space. Laurentiu Mihalcea (2): ASoC: SOF: compress: Move sof_compr_copy functionality ASoC: SOF: compress: Add copy function for capture case sound/soc/sof/compress.c | 47 +++++++++++++++++++++++++++++++++------- 1 file changed, 39 insertions(+), 8 deletions(-) -- 2.34.1

2 3

[PATCH v2] ASoC: amd: acp: Modify dai_id macros to be more generic
by Venkata Prasad Potturu 31 Aug '22

31 Aug '22

Change dai_id macros to make I2S instances in order. Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- Changes since v1: -- Change commit description. sound/soc/amd/acp/amd.h | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/amd/acp/amd.h b/sound/soc/amd/acp/amd.h index af9603724a68..60a7c95f134f 100644 --- a/sound/soc/amd/acp/amd.h +++ b/sound/soc/amd/acp/amd.h @@ -21,9 +21,9 @@ #define ACP3X_DEV 3 #define ACP6X_DEV 6 -#define I2S_SP_INSTANCE 0x00 -#define I2S_BT_INSTANCE 0x01 -#define DMIC_INSTANCE 0x02 +#define DMIC_INSTANCE 0x00 +#define I2S_SP_INSTANCE 0x01 +#define I2S_BT_INSTANCE 0x02 #define I2S_HS_INSTANCE 0x03 #define MEM_WINDOW_START 0x4080000 -- 2.25.1

2 1

[PATCH 0/6] ASoC: qcom: add support for sc8280xp machine
by Srinivas Kandagatla 31 Aug '22

31 Aug '22

THis patchset adds support for SC8280XP SoC machine driver. First patch moves some of the commonly used code to common from sm8250 machine driver and the follow on code adds minimal support for sc8280xp. Currently this driver is only tested with SmartSpeakers and Headset on Lenovo Thinkpad X13s. Support for sm8450 is tested and I will post the patches soon. Thanks, Srini Srinivas Kandagatla (3): ASoC: dt-bindings: qcom,sm8250: add compatibles for sm8450 and sm8250 ASoC: qcom: sm8250: move some code to common ASoC: qcom: add machine driver for sc8280xp .../bindings/sound/qcom,sm8250.yaml | 2 + sound/soc/qcom/Kconfig | 11 ++ sound/soc/qcom/Makefile | 2 + sound/soc/qcom/common.c | 169 ++++++++++++++++++ sound/soc/qcom/common.h | 12 ++ sound/soc/qcom/sc8280xp.c | 157 ++++++++++++++++ sound/soc/qcom/sm8250.c | 152 +--------------- 7 files changed, 360 insertions(+), 145 deletions(-) create mode 100644 sound/soc/qcom/sc8280xp.c -- 2.21.0

3 6

How does PlaybackChannelPos# Work?
by GitHub issues - opened 31 Aug '22

31 Aug '22

alsa-project/alsa-ucm-conf issue #209 was opened from 13r0ck: I found https://github.com/alsa-project/alsa-lib/commit/0ba999591b78e07a4b8e610ebce… but doesn't seem to work when setting audio jack output. I have a mostly working configuration for a motherboard's 3.5mm audio jack output. The only problem is that the C/Sub and Rear jacks are flipped. I am assuming setting the `PlaybackChannelPos` in the speaker should work, but it doesn't for some reason. ``` SectionDevice."Speaker" { Commant "Speakers" Value { PlaybackChannel2 "BL" PlaybackChannel3 "BR" PlaybackChannel4 "FC" PlaybackChannel5 "LFE" } } ``` Any help would be greatly appreciated! Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/209 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: amd: fix spelling mistake: "i.e" -> "i.e."
by Syed Saba Kareem 30 Aug '22

30 Aug '22

From: Syed Saba kareem <Syed.SabaKareem(a)amd.com> trivial fix to spelling mistake in Kconfig File. Reported by : Randy Dunlap <rdunlap(a)infradead.org> Signed-off-by: Syed Saba Kareem <Syed.SabaKareem(a)amd.com> --- sound/soc/amd/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 68837d42736d..150786279257 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -122,7 +122,7 @@ config SND_SOC_AMD_RPL_ACP6x tristate "AMD Audio Coprocessor-v6.2 RPL support" depends on X86 && PCI help - This option enables Audio Coprocessor i.e ACP v6.2 support on + This option enables Audio Coprocessor i.e. ACP v6.2 support on AMD RPL platform. By enabling this flag build will be triggered for ACP PCI driver. Say m if you have such a device. -- 2.25.1

3 2

[PATCH 1/3] slimbus: qcom-ngd: use correct error in message of pdr_add_lookup() failure
by Krzysztof Kozlowski 30 Aug '22

30 Aug '22

Use cprrect error code, instead of previous 'ret' value, when printing error from pdr_add_lookup() failure. Cc: <stable(a)vger.kernel.org> Fixes: e1ae85e1830e ("slimbus: qcom-ngd-ctrl: add Protection Domain Restart Support") Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- drivers/slimbus/qcom-ngd-ctrl.c | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index 0aa8408464ad..6fe6abb86061 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -1581,6 +1581,7 @@ static int qcom_slim_ngd_ctrl_probe(struct platform_device *pdev) pds = pdr_add_lookup(ctrl->pdr, "avs/audio", "msm/adsp/audio_pd"); if (IS_ERR(pds) && PTR_ERR(pds) != -EALREADY) { + ret = PTR_ERR(pds); dev_err(dev, "pdr add lookup failed: %d\n", ret); return PTR_ERR(pds); } -- 2.34.1

2 4

[PATCH] ASoC: codecs: allow compile testing without MFD drivers
by Krzysztof Kozlowski 30 Aug '22

30 Aug '22

Motorola CPCAP, Lochnagar Sound, Rockchip RK817 and Qualcomm WCD9340/WCD9341 do not depend on parent MFD driver in build time and can be compile tested without respective MFD part for increased build coverage. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/codecs/Kconfig | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 5926b33ba09e..94b7bb85d236 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -610,7 +610,7 @@ config SND_SOC_BT_SCO config SND_SOC_CPCAP tristate "Motorola CPCAP codec" - depends on MFD_CPCAP + depends on MFD_CPCAP || COMPILE_TEST config SND_SOC_CQ0093VC tristate @@ -972,7 +972,7 @@ config SND_SOC_LM49453 config SND_SOC_LOCHNAGAR_SC tristate "Lochnagar Sound Card" - depends on MFD_LOCHNAGAR + depends on MFD_LOCHNAGAR || COMPILE_TEST help This driver support the sound card functionality of the Cirrus Logic Lochnagar audio development board. @@ -1197,7 +1197,7 @@ config SND_SOC_RK3328 config SND_SOC_RK817 tristate "Rockchip RK817 audio CODEC" - depends on MFD_RK808 + depends on MFD_RK808 || COMPILE_TEST select REGMAP_I2C config SND_SOC_RL6231 @@ -1745,7 +1745,7 @@ config SND_SOC_WCD934X tristate "WCD9340/WCD9341 Codec" depends on COMMON_CLK select SND_SOC_WCD_MBHC - depends on MFD_WCD934X + depends on MFD_WCD934X || COMPILE_TEST help The WCD9340/9341 is a audio codec IC Integrated in Qualcomm SoCs like SDM845. -- 2.34.1

2 5

[PATCH] ASoC: codecs: rk817: fix missing I2C dependency in compile test
by Krzysztof Kozlowski 30 Aug '22

30 Aug '22

SND_SOC_RK817 uses I2C regmap so compile testing without parent MFD_RK808, requires I2C: WARNING: unmet direct dependencies detected for REGMAP_I2C Depends on [n]: I2C [=n] Selected by [y]: - SND_SOC_RK817 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && (MFD_RK808 [=n] || COMPILE_TEST [=y]) Reported-by: kernel test robot <lkp(a)intel.com> Fixes: 5b7f4e5de61b ("ASoC: codecs: allow compile testing without MFD drivers") Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Commit SHA from linux-next (recent). --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 94b7bb85d236..968d0701f2e8 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1197,7 +1197,7 @@ config SND_SOC_RK3328 config SND_SOC_RK817 tristate "Rockchip RK817 audio CODEC" - depends on MFD_RK808 || COMPILE_TEST + depends on MFD_RK808 || COMPILE_TEST && I2C select REGMAP_I2C config SND_SOC_RL6231 -- 2.34.1

2 1

[PATCH v2 00/13] Add Pink Sardine platform ASoC driver
by Syed Saba Kareem 30 Aug '22

30 Aug '22

Pink Sardine platform is new APU series based on acp6.2 design. This patch set adds an ASoC driver for the ACP (Audio CoProcessor) block on AMD Pink Sardine APU with DMIC endpoint support. changes since v1: -Removed subraction logic in acp62.h header file and changed the address in acp62_chip_offset_byte.h file. -Fixed indentation in pci driver file. -Removed comments which states function names in pci driver file. -Replaced some hex values with decimals in pci driver file. -Corrected MODULE_DESCRIPTION in pdm driver file. -Removed structure variable which is used for byte count in header file and added new local variable for denoting the byte count. -Added Kfree for freeing the runtime->private data. -Replaced X86 && PCI with X86 && PCI && ACPI in Kconfig file. Syed Saba Kareem (13): ASoC: amd: add Pink Sardine platform ACP IP register header ASoC: amd: add Pink Sardine ACP PCI driver ASoC: amd: add acp6.2 init/de-init functions ASoC: amd: add platform devices for acp6.2 pdm driver and dmic driver ASoC: amd: add acp6.2 pdm platform driver ASoC: amd: add acp6.2 irq handler ASoC: amd: add acp6.2 pdm driver dma ops ASoC: amd: add acp6.2 pci driver pm ops ASoC: amd: add acp6.2 pdm driver pm ops ASoC: amd: enable Pink Sardine acp6.2 drivers build ASoC: amd: create platform device for acp6.2 machine driver ASoC: amd: add Pink Sardine machine driver using dmic ASoC: amd: enable Pink sardine platform machine driver build. include/sound/acp62_chip_offset_byte.h | 444 ++++++++++++++++++++++++ sound/soc/amd/Kconfig | 21 ++ sound/soc/amd/Makefile | 1 + sound/soc/amd/ps/Makefile | 9 + sound/soc/amd/ps/acp62.h | 98 ++++++ sound/soc/amd/ps/pci-ps.c | 351 +++++++++++++++++++ sound/soc/amd/ps/ps-mach.c | 79 +++++ sound/soc/amd/ps/ps-pdm-dma.c | 452 +++++++++++++++++++++++++ 8 files changed, 1455 insertions(+) create mode 100644 include/sound/acp62_chip_offset_byte.h create mode 100644 sound/soc/amd/ps/Makefile create mode 100644 sound/soc/amd/ps/acp62.h create mode 100644 sound/soc/amd/ps/pci-ps.c create mode 100644 sound/soc/amd/ps/ps-mach.c create mode 100644 sound/soc/amd/ps/ps-pdm-dma.c -- 2.25.1

2 14

[PATCH 0/5] Create a new sound card to access MICFIL based on rpmsg channel
by Chancel Liu 30 Aug '22

30 Aug '22

At a previous time, we have successfully created a virtual sound card based on rpmsg. The sound card works under this mechanism Cortex-A core tells the Cortex-M core the format, rate, channel, .etc configuration of the PCM parameters and Cortex-M controls real hardware devices such as SAI and DMA. From the view of Linux side, the sound card is bound to a rpmsg channel through which it can access SAI. Here these patches are introduced to create a new virtual sound card to access MICFIL based on a new created rpmsg channel. It's easy to create a new rpmsg channel for MICFIL through rpmsg name service announcment. Also the other ASoC components bound to this rpmsg MICFIL sound card will be registered with these patches. If other sound cards using different hardware devices needs to be created over rpmsg in the future, these patches can be referred. Chancel Liu (5): ASoC: dt-bindings: fsl_rpmsg: Add a property to assign platform driver name ASoC: imx-audio-rpmsg: Create rpmsg channel for MICFIL ASoC: imx-pcm-rpmsg: Register different platform drivers ASoC: fsl_rpmsg: Register different CPU DAI drivers ASoC: imx-rpmsg: Assign platform driver used by machine driver to link with .../devicetree/bindings/sound/fsl,rpmsg.yaml | 34 +++++++++++++++++-- sound/soc/fsl/fsl_rpmsg.c | 2 +- sound/soc/fsl/imx-audio-rpmsg.c | 3 +- sound/soc/fsl/imx-pcm-rpmsg.c | 10 ++++-- sound/soc/fsl/imx-rpmsg.c | 6 +++- 5 files changed, 47 insertions(+), 8 deletions(-) -- 2.25.1

3 8

[PATCH 0/1] soundwire: bus: Fix for missing UNATTACH when re-enumerating
by Richard Fitzgerald 30 Aug '22

30 Aug '22

This is an alternative fix for the race where a device that is reset can be found on #0 and its ID reprogrammed before the bus code has handled a PING showing it unattached. The other possible fix is here: https://lore.kernel.org/all/20220825122241.273090-4-rf@opensource.cirrus.co… Richard Fitzgerald (1): soundwire: bus: Don't re-enumerate before status is UNATTACHED drivers/soundwire/bus.c | 7 +++++++ 1 file changed, 7 insertions(+) -- 2.30.2

1 2

[PATCH 0/3] soundwire: Fixes for spurious and missing UNATTACH
by Richard Fitzgerald 30 Aug '22

30 Aug '22

The bus and cadence code has several bugs that cause UNATTACH notifications to either be sent spuriously or to be missed. These can be seen occasionally with a single peripheral on the bus, but are much more frequent with multiple peripherals, where several peripherals could change state and report in consecutive PINGs. The root of all of these bugs seems to be a code design flaw that assumed every PING status change would be handled separately. However, PINGs are handled by a workqueue function and there is no guarantee when that function will be scheduled to run or how much CPU time it will receive. PINGs will continue while the work function is handling a snapshot of a previous PING so the code must take account that (a) status could change during the work function and (b) there can be a backlog of changes before the IRQ work function runs again. Richard Fitzgerald (2): soundwire: bus: Don't lose unattach notifications soundwire: bus: Fix lost UNATTACH when re-enumerating Simon Trimmer (1): soundwire: cadence: fix updating slave status when a bus has multiple peripherals drivers/soundwire/bus.c | 44 +++++++++++++++------ drivers/soundwire/cadence_master.c | 63 +++++++++++++----------------- 2 files changed, 59 insertions(+), 48 deletions(-) -- 2.30.2

2 11

[PATCH][next] ALSA: hda/hdmi: Replace zero-length array with DECLARE_FLEX_ARRAY() helper
by Gustavo A. R. Silva 30 Aug '22

30 Aug '22

Zero-length arrays are deprecated and we are moving towards adopting C99 flexible-array members, instead. So, replace zero-length array declaration in union audio_infoframe with the new DECLARE_FLEX_ARRAY() helper macro. This helper allows for a flexible-array member in a union. Link: https://github.com/KSPP/linux/issues/193 Link: https://gcc.gnu.org/onlinedocs/gcc/Zero-Length.html Signed-off-by: Gustavo A. R. Silva <gustavoars(a)kernel.org> --- sound/pci/hda/patch_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 6c209cd26c0c..2191d445d74e 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -229,7 +229,7 @@ struct dp_audio_infoframe { union audio_infoframe { struct hdmi_audio_infoframe hdmi; struct dp_audio_infoframe dp; - u8 bytes[0]; + DECLARE_FLEX_ARRAY(u8, bytes); }; /* -- 2.34.1

3 2

[PATCH 00/11] soundwire: register cleanups
by Bard Liao 30 Aug '22

30 Aug '22

Clean-up and reorder/regroup SoundWire SHIM registers. Pierre-Louis Bossart (11): soundwire: intel: cleanup definition of LCOUNT soundwire: intel: regroup definitions for LCTL soundwire: intel: remove IPPTR unused definition soundwire: intel: cleanup SHIM SYNC soundwire: intel: remove unused PDM capabilities soundwire: intel: add comment for control stream cap/chmap soundwire: intel: cleanup PCM stream capabilities soundwire: intel: cleanup PCM Stream channel map and channel count soundwire: intel: cleanup IO control soundwire: intel: cleanup AC Timing Control soundwire: intel: cleanup WakeEnable and WakeStatus drivers/soundwire/intel.c | 5 +-- drivers/soundwire/intel_init.c | 2 +- include/linux/soundwire/sdw_intel.h | 63 +++++++++++++++++------------ 3 files changed, 41 insertions(+), 29 deletions(-) -- 2.25.1

2 12

[PATCH 0/3] soundwire: allocate device_number with IDA
by Bard Liao 30 Aug '22

30 Aug '22

The device_number is currently allocated in the scope of each bus and does not need to be unique at the system level. This leads e.g. on Dell devices with three or four Device1 on different bus segments. To make the device_number unique at the system level, and unified with the HDaudio/iDISP SDI values, this series allocates the dev_number with an IDA restricted between 4 and 11 (inclusive). Pierre-Louis Bossart (3): soundwire: bus: rename sdw_ida as sdw_bus_ida soundwire: bus: allow device number to be unique at system level soundwire: intel: set dev_num_ida_min drivers/soundwire/bus.c | 29 ++++++++++++++++++++--------- drivers/soundwire/intel.c | 4 ++++ include/linux/soundwire/sdw.h | 4 ++++ 3 files changed, 28 insertions(+), 9 deletions(-) -- 2.25.1

3 6

[PATCH] dt-bindings: iio: Add missing (unevaluated|additional)Properties on child nodes
by Rob Herring 29 Aug '22

29 Aug '22

In order to ensure only documented properties are present, node schemas must have unevaluatedProperties or additionalProperties set to false (typically). Signed-off-by: Rob Herring <robh(a)kernel.org> --- Documentation/devicetree/bindings/iio/adc/qcom,spmi-vadc.yaml | 1 + .../devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml | 2 ++ Documentation/devicetree/bindings/iio/adc/ti,ads1015.yaml | 1 + .../devicetree/bindings/iio/adc/xlnx,zynqmp-ams.yaml | 1 + Documentation/devicetree/bindings/iio/addac/adi,ad74413r.yaml | 1 + Documentation/devicetree/bindings/iio/dac/adi,ad3552r.yaml | 1 + Documentation/devicetree/bindings/iio/dac/adi,ad5770r.yaml | 4 ++++ Documentation/devicetree/bindings/iio/dac/adi,ltc2688.yaml | 1 + 8 files changed, 12 insertions(+) diff --git a/Documentation/devicetree/bindings/iio/adc/qcom,spmi-vadc.yaml b/Documentation/devicetree/bindings/iio/adc/qcom,spmi-vadc.yaml index 74a4a9d95798..8bac0c4120dd 100644 --- a/Documentation/devicetree/bindings/iio/adc/qcom,spmi-vadc.yaml +++ b/Documentation/devicetree/bindings/iio/adc/qcom,spmi-vadc.yaml @@ -58,6 +58,7 @@ required: patternProperties: "^.*@[0-9a-f]+$": type: object + additionalProperties: false description: | Represents the external channels which are connected to the ADC. For compatible property "qcom,spmi-vadc" following channels, also known as diff --git a/Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml b/Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml index 92f9472a77ae..1970503389aa 100644 --- a/Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml +++ b/Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml @@ -72,6 +72,7 @@ additionalProperties: false patternProperties: "^filter@[0-9]+$": type: object + unevaluatedProperties: false description: child node properties: @@ -225,6 +226,7 @@ patternProperties: patternProperties: "^dfsdm-dai+$": type: object + additionalProperties: false description: child node properties: diff --git a/Documentation/devicetree/bindings/iio/adc/ti,ads1015.yaml b/Documentation/devicetree/bindings/iio/adc/ti,ads1015.yaml index a3b79438a13a..2c3c2cf2145c 100644 --- a/Documentation/devicetree/bindings/iio/adc/ti,ads1015.yaml +++ b/Documentation/devicetree/bindings/iio/adc/ti,ads1015.yaml @@ -43,6 +43,7 @@ additionalProperties: false patternProperties: "^channel@[0-7]+$": type: object + additionalProperties: false description: Child nodes needed for each channel that the platform uses. diff --git a/Documentation/devicetree/bindings/iio/adc/xlnx,zynqmp-ams.yaml b/Documentation/devicetree/bindings/iio/adc/xlnx,zynqmp-ams.yaml index 3698b4b0900f..be93c109d6ac 100644 --- a/Documentation/devicetree/bindings/iio/adc/xlnx,zynqmp-ams.yaml +++ b/Documentation/devicetree/bindings/iio/adc/xlnx,zynqmp-ams.yaml @@ -134,6 +134,7 @@ properties: ams-pl@400: type: object + additionalProperties: false description: PL-SYSMON is capable of monitoring off chip voltage and temperature. PL-SYSMON block has DRP, JTAG and I2C interface to enable monitoring diff --git a/Documentation/devicetree/bindings/iio/addac/adi,ad74413r.yaml b/Documentation/devicetree/bindings/iio/addac/adi,ad74413r.yaml index baa65a521bad..03bb90a7f4f8 100644 --- a/Documentation/devicetree/bindings/iio/addac/adi,ad74413r.yaml +++ b/Documentation/devicetree/bindings/iio/addac/adi,ad74413r.yaml @@ -63,6 +63,7 @@ additionalProperties: false patternProperties: "^channel@[0-3]$": type: object + additionalProperties: false description: Represents the external channels which are connected to the device. properties: diff --git a/Documentation/devicetree/bindings/iio/dac/adi,ad3552r.yaml b/Documentation/devicetree/bindings/iio/dac/adi,ad3552r.yaml index 9c48c76993fe..fee0f023a8c8 100644 --- a/Documentation/devicetree/bindings/iio/dac/adi,ad3552r.yaml +++ b/Documentation/devicetree/bindings/iio/dac/adi,ad3552r.yaml @@ -78,6 +78,7 @@ patternProperties: custom-output-range-config: type: object + additionalProperties: false description: Configuration of custom range when adi,output-range-microvolt is not present. The formulas for calculation the output voltages are diff --git a/Documentation/devicetree/bindings/iio/dac/adi,ad5770r.yaml b/Documentation/devicetree/bindings/iio/dac/adi,ad5770r.yaml index 24ac40180ac1..c2d9ae3039b2 100644 --- a/Documentation/devicetree/bindings/iio/dac/adi,ad5770r.yaml +++ b/Documentation/devicetree/bindings/iio/dac/adi,ad5770r.yaml @@ -62,6 +62,7 @@ properties: connected to the DAC. Channel 0 can act both as a current source and sink. type: object + additionalProperties: false properties: reg: @@ -85,6 +86,7 @@ properties: description: Represents an external channel which are connected to the DAC. type: object + additionalProperties: false properties: reg: @@ -101,6 +103,7 @@ properties: description: Represents an external channel which are connected to the DAC. type: object + additionalProperties: false properties: reg: @@ -116,6 +119,7 @@ properties: patternProperties: "^channel@([3-5])$": type: object + additionalProperties: false description: Represents the external channels which are connected to the DAC. properties: reg: diff --git a/Documentation/devicetree/bindings/iio/dac/adi,ltc2688.yaml b/Documentation/devicetree/bindings/iio/dac/adi,ltc2688.yaml index 48f9e7d29423..15cc6bf59b13 100644 --- a/Documentation/devicetree/bindings/iio/dac/adi,ltc2688.yaml +++ b/Documentation/devicetree/bindings/iio/dac/adi,ltc2688.yaml @@ -48,6 +48,7 @@ properties: patternProperties: "^channel@([0-9]|1[0-5])$": type: object + additionalProperties: false properties: reg: -- 2.34.1

3 4

[PATCH linux-next] ASoC: codecs: max98088: remove redundant ret variable
by cgel.zte＠gmail.com 29 Aug '22

29 Aug '22

From: Jinpeng Cui <cui.jinpeng2(a)zte.com.cn> Return value from devm_snd_soc_register_component() directly instead of taking this in another redundant variable. Reported-by: Zeal Robot <zealci(a)zte.com.cn> Signed-off-by: Jinpeng Cui <cui.jinpeng2(a)zte.com.cn> --- sound/soc/codecs/max98088.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index b208fb530a8b..405ec16be2b6 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1749,7 +1749,6 @@ MODULE_DEVICE_TABLE(i2c, max98088_i2c_id); static int max98088_i2c_probe(struct i2c_client *i2c) { struct max98088_priv *max98088; - int ret; const struct i2c_device_id *id; max98088 = devm_kzalloc(&i2c->dev, sizeof(struct max98088_priv), @@ -1772,9 +1771,8 @@ static int max98088_i2c_probe(struct i2c_client *i2c) i2c_set_clientdata(i2c, max98088); max98088->pdata = i2c->dev.platform_data; - ret = devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_max98088, + return devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_max98088, &max98088_dai[0], 2); - return ret; } #if defined(CONFIG_OF) -- 2.25.1

2 1

[PATCH 0/5] TAS2764 fixes/extensions
by Martin Povišer 29 Aug '22

29 Aug '22

First three patches are fixes analogical to those recently done to the TAS2770 driver. Link: https://lore.kernel.org/asahi/20220808141246.5749-1-povik+lin@cutebit.org/T… The latter two add IRQ handler to log faults and expose a new control. Martin Povišer (5): ASoC: tas2764: Allow mono streams ASoC: tas2764: Drop conflicting set_bias_level power setting ASoC: tas2764: Fix mute/unmute ASoC: tas2764: Add IRQ handling ASoC: tas2764: Export highpass filter setting sound/soc/codecs/tas2764.c | 181 ++++++++++++++++++++++++++----------- sound/soc/codecs/tas2764.h | 23 +++++ 2 files changed, 152 insertions(+), 52 deletions(-) -- 2.33.0

2 6

[RESEND v2] ASoC: amd: acp: Modify dai_id macros to be more generic
by Venkata Prasad Potturu 29 Aug '22

29 Aug '22

Change dai_id macros to make I2S instances in order. Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- sound/soc/amd/acp/amd.h | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/amd/acp/amd.h b/sound/soc/amd/acp/amd.h index af9603724a68..60a7c95f134f 100644 --- a/sound/soc/amd/acp/amd.h +++ b/sound/soc/amd/acp/amd.h @@ -21,9 +21,9 @@ #define ACP3X_DEV 3 #define ACP6X_DEV 6 -#define I2S_SP_INSTANCE 0x00 -#define I2S_BT_INSTANCE 0x01 -#define DMIC_INSTANCE 0x02 +#define DMIC_INSTANCE 0x00 +#define I2S_SP_INSTANCE 0x01 +#define I2S_BT_INSTANCE 0x02 #define I2S_HS_INSTANCE 0x03 #define MEM_WINDOW_START 0x4080000 -- 2.25.1

2 1

[PATCH -next 0/2] ASoC: Cleanup deprecated regmap-irq functionality
by Aidan MacDonald 29 Aug '22

29 Aug '22

Update two ASoC codec drivers to remove uses of regmap-irq type registers, which have recently been deprecated by the "regmap-irq cleanups and refactoring" series in linux-next. Link: https://lore.kernel.org/lkml/20220623211420.918875-1-aidanmacdonald.0x0@gma… Aidan MacDonald (2): sound: soc: codecs: wcd9335: Convert irq chip to config regs sound: soc: codecs: wcd938x: Remove spurious type_base from irq chip sound/soc/codecs/wcd9335.c | 10 ++++++++-- sound/soc/codecs/wcd938x.c | 1 - 2 files changed, 8 insertions(+), 3 deletions(-) base-commit: 353f7988dd8413c47718f7ca79c030b6fb62cfe5 prerequisite-patch-id: a45db1cac7537769dc10087fc61f977dc150744c prerequisite-patch-id: cb8aa009c7bb7a6575eb05e3af65342dc8d0efa3 prerequisite-patch-id: 8d32557e53b894d1fb17250d2d0eb3673f068d37 prerequisite-patch-id: 5b293867ef81e3697892ac51b941bb53680a70dc prerequisite-patch-id: 8138d39a4817e804141bfe8c2ad37d9c55456a40 prerequisite-patch-id: b01216129e887519d441cf556bbc75c397871773 prerequisite-patch-id: b5dcf0c0609113c2d81bc557c1fc95ef23f40811 prerequisite-patch-id: 622ca1c10e851b1889aaa567c1d2a0adf43cca44 prerequisite-patch-id: d28c5187f9bf0e43f27b2f4aa8dcf7fd91842a03 prerequisite-patch-id: d29d54b756be73304f844abeeaf9b46a5c0119d5 prerequisite-patch-id: 5d405790ae89c0831b46a359f86e94bac5a67470 prerequisite-patch-id: c90120e79acbb52ffa148bfedee1df9d35b5eced prerequisite-patch-id: 2e35247a5cfe5a28565c9272b85fc6835011b032 -- 2.35.1

2 4

[PATCH] ASoC: soc-pcm.c: call __soc_pcm_close() in soc_pcm_close()
by Kuninori Morimoto 29 Aug '22

29 Aug '22

commit b7898396f4bbe16 ("ASoC: soc-pcm: Fix and cleanup DPCM locking") added __soc_pcm_close() for non-lock version of soc_pcm_close(). But soc_pcm_close() is not using it. It is no problem, but confusable. static int __soc_pcm_close(...) { => return soc_pcm_clean(rtd, substream, 0); } static int soc_pcm_close(...) { ... snd_soc_dpcm_mutex_lock(rtd); => soc_pcm_clean(rtd, substream, 0); snd_soc_dpcm_mutex_unlock(rtd); return 0; } This patch use it. Fixes: b7898396f4bbe16 ("ASoC: soc-pcm: Fix and cleanup DPCM locking") Cc: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 5b99bf2dbd08..3ca073edf7e7 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -723,7 +723,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); snd_soc_dpcm_mutex_lock(rtd); - soc_pcm_clean(rtd, substream, 0); + __soc_pcm_close(rtd, substream); snd_soc_dpcm_mutex_unlock(rtd); return 0; } -- 2.25.1

2 1

ucm2: HDA - add support for Internal Mic
by GitHub pull_request - edited 28 Aug '22

28 Aug '22

alsa-project/alsa-ucm-conf pull request #207 was edited from kasper93: Simple patch to support also "Internal Mic". Everything seems to be working fine after this change, except that I don't have "Internal Mic Jack", I'm lying there a little to set it as such. Everything switches nicely when connecting/disconnecting headphones mic. But there are two input devices to select and selection doesn't have effect. When headphone mic is connected the internal one should be hidden (it is handled somehow when using HDA driver), but there is no Jack and I don't know how to handle it properly. (It is marked as conflicting device with the other one though) Except this minor annoyance it is fine. For reference: [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/9439697/… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/207 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/207.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: HDA - add support for Internal Mic
by GitHub pull_request - edited 28 Aug '22

28 Aug '22

alsa-project/alsa-ucm-conf pull request #207 was edited from kasper93: Simple patch to support also "Internal Mic". Everything seems to be working fine after this change, except that I don't have "Internal Mic Jack", I'm lying there a little to set it as such. Everything switches nicely when connecting/disconnecting headphones mic. But there are two input devices to select and selection doesn't have effect. When headphone mic is connected the internal one should be hidden, but there is no Jack and I don't know how to handle it properly. (It is marked as conflicting device with the other one though) Except this minor annoyance it is fine. For reference: [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/9439697/… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/207 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/207.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: HDA - add support for Internal Mic
by GitHub pull_request - opened 28 Aug '22

28 Aug '22

alsa-project/alsa-ucm-conf pull request #207 was opened from kasper93: Simple patch to support also "Internal Mic". Everything seems to be working fine after this change, except that I don't have "Internal Mic Jack", I'm lying there a little to set it as such. Everything switches nicely when connecting/disconnecting headphones mic. But there are two input devices to select and selection doesn't have effect. I believe when headphone mic is connected the internal one should be hidden, but there is no Jack and I don't know how to handle it properly. (It is marked as conflicting device with the other one though) Except this minor annoyance it is fine. For reference: [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/9439697/… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/207 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/207.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ALSA: usb-audio: Add quirk for LH Labs Geek Out HD Audio 1V5
by Takashi Iwai 28 Aug '22

28 Aug '22

The USB DAC from LH Labs (2522:0007) seems requiring the same quirk as Sony Walkman to set up the interface like UAC1; otherwise it gets the constant errors "usb_set_interface failed (-71)". This patch adds a quirk entry for addressing the buggy behavior. Reported-by: Lennert Van Alboom <lennert(a)vanalboom.org> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/T3VPXtCc4uFws9Gfh2RjX6OdwM1RqfC6VqQr--_LMDyB2x5N3… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 168fd802d70b..9bfead5efc4c 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1903,6 +1903,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_SHARE_MEDIA_DEVICE | QUIRK_FLAG_ALIGN_TRANSFER), DEVICE_FLG(0x21b4, 0x0081, /* AudioQuest DragonFly */ QUIRK_FLAG_GET_SAMPLE_RATE), + DEVICE_FLG(0x2522, 0x0007, /* LH Labs Geek Out HD Audio 1V5 */ + QUIRK_FLAG_SET_IFACE_FIRST), DEVICE_FLG(0x2708, 0x0002, /* Audient iD14 */ QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x2912, 0x30c8, /* Audioengine D1 */ -- 2.35.3

1 0

[PATCH] ALSA: hda/realtek: Add speaker AMP init for Samsung laptops with ALC298
by Kacper Michajłow 28 Aug '22

28 Aug '22

Magic initialization sequence was extracted from Windows driver and cleaned up manually. Fixes internal speakers output. Link: https://bugzilla.kernel.org/show_bug.cgi?id=207423 Link: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1851518 Cc: Takashi Iwai <tiwai(a)suse.com> Signed-off-by: Kacper Michajłow <kasper93(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 63 +++++++++++++++++++++++++++++++---- 1 file changed, 56 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 47e72cf76608..38930cf5aace 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4700,6 +4700,48 @@ static void alc236_fixup_hp_mute_led_micmute_vref(struct hda_codec *codec, alc236_fixup_hp_micmute_led_vref(codec, fix, action); } +static inline void alc298_samsung_write_coef_pack(struct hda_codec *codec, + const unsigned short coefs[2]) +{ + alc_write_coef_idx(codec, 0x23, coefs[0]); + alc_write_coef_idx(codec, 0x25, coefs[1]); + alc_write_coef_idx(codec, 0x26, 0xb011); +} + +struct alc298_samsung_amp_desc { + unsigned char nid; + unsigned short init_seq[2][2]; +}; + +static void alc298_fixup_samsung_amp(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + int i, j; + static const unsigned short init_seq[][2] = { + { 0x19, 0x00 }, { 0x20, 0xc0 }, { 0x22, 0x44 }, { 0x23, 0x08 }, + { 0x24, 0x85 }, { 0x25, 0x41 }, { 0x35, 0x40 }, { 0x36, 0x01 }, + { 0x38, 0x81 }, { 0x3a, 0x03 }, { 0x3b, 0x81 }, { 0x40, 0x3e }, + { 0x41, 0x07 }, { 0x400, 0x1 } + }; + static const struct alc298_samsung_amp_desc amps[] = { + { 0x3a, { { 0x18, 0x1 }, { 0x26, 0x0 } } }, + { 0x39, { { 0x18, 0x2 }, { 0x26, 0x1 } } } + }; + + if (action != HDA_FIXUP_ACT_INIT) + return; + + for (i = 0; i < ARRAY_SIZE(amps); i++) { + alc_write_coef_idx(codec, 0x22, amps[i].nid); + + for (j = 0; j < ARRAY_SIZE(amps[i].init_seq); j++) + alc298_samsung_write_coef_pack(codec, amps[i].init_seq[j]); + + for (j = 0; j < ARRAY_SIZE(init_seq); j++) + alc298_samsung_write_coef_pack(codec, init_seq[j]); + } +} + #if IS_REACHABLE(CONFIG_INPUT) static void gpio2_mic_hotkey_event(struct hda_codec *codec, struct hda_jack_callback *event) @@ -7030,6 +7072,7 @@ enum { ALC236_FIXUP_HP_GPIO_LED, ALC236_FIXUP_HP_MUTE_LED, ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF, + ALC298_FIXUP_SAMSUNG_AMP, ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, ALC256_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, ALC295_FIXUP_ASUS_MIC_NO_PRESENCE, @@ -8396,6 +8439,12 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc236_fixup_hp_mute_led_micmute_vref, }, + [ALC298_FIXUP_SAMSUNG_AMP] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc298_fixup_samsung_amp, + .chained = true, + .chain_id = ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET + }, [ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -9342,13 +9391,13 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x10ec, 0x1254, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), 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), - SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), - SND_PCI_QUIRK(0x144d, 0xc189, "Samsung Galaxy Flex Book (NT950QCG-X716)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), - SND_PCI_QUIRK(0x144d, 0xc18a, "Samsung Galaxy Book Ion (NP930XCJ-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), + SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_AMP), + SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_AMP), + SND_PCI_QUIRK(0x144d, 0xc189, "Samsung Galaxy Flex Book (NT950QCG-X716)", ALC298_FIXUP_SAMSUNG_AMP), + SND_PCI_QUIRK(0x144d, 0xc18a, "Samsung Galaxy Book Ion (NP930XCJ-K01US)", ALC298_FIXUP_SAMSUNG_AMP), SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8), - SND_PCI_QUIRK(0x144d, 0xc812, "Samsung Notebook Pen S (NT950SBE-X58)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), - SND_PCI_QUIRK(0x144d, 0xc830, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), + SND_PCI_QUIRK(0x144d, 0xc812, "Samsung Notebook Pen S (NT950SBE-X58)", ALC298_FIXUP_SAMSUNG_AMP), + SND_PCI_QUIRK(0x144d, 0xc830, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_AMP), SND_PCI_QUIRK(0x144d, 0xc832, "Samsung Galaxy Book Flex Alpha (NP730QCJ)", ALC256_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), @@ -9716,7 +9765,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC299_FIXUP_PREDATOR_SPK, .name = "predator-spk"}, {.id = ALC298_FIXUP_HUAWEI_MBX_STEREO, .name = "huawei-mbx-stereo"}, {.id = ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, .name = "alc256-medion-headset"}, - {.id = ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, .name = "alc298-samsung-headphone"}, + {.id = ALC298_FIXUP_SAMSUNG_AMP, .name = "alc298-samsung-amp"}, {.id = ALC256_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, .name = "alc256-samsung-headphone"}, {.id = ALC255_FIXUP_XIAOMI_HEADSET_MIC, .name = "alc255-xiaomi-headset"}, {.id = ALC274_FIXUP_HP_MIC, .name = "alc274-hp-mic-detect"}, -- 2.37.2

2 1

USB DAC broken since commit bf6313a0ff766925462e97b4e733d5952de02367 (5.10.0-rc5)
by Lennert Van Alboom 27 Aug '22

27 Aug '22

Hi, My USB DAC fails to work on recent kernels (anything past 5.10). The device is the following: Bus 003 Device 005: ID 2522:0007 LH Labs Geek Out HD Audio 1V5 Syslog shows error messages which have changed over the different kernel versions - this is the output from debian's 5.17.0-2-amd64: [66136.185642] usb 3-6: uac_clock_source_is_valid(): cannot get clock validity for id 41 [66136.185643] usb 3-6: clock source 41 is not valid, cannot use [66136.185956] usb 3-6: 1:0: usb_set_interface failed (-71) [66136.186183] usb 3-6: 1:0: usb_set_interface failed (-71) >From the first point in git where it fails (5.10.0-rc5-00025-gbf6313a0ff76): Aug 25 12:39:37 Nesbitt kernel: [ 5295.633079] usb 3-3: new high-speed USB device number 8 using xhci_hcd Aug 25 12:39:37 Nesbitt kernel: [ 5295.782144] usb 3-3: New USB device found, idVendor=2522, idProduct=0007, bcdDevice=15.02 Aug 25 12:39:37 Nesbitt kernel: [ 5295.782148] usb 3-3: New USB device strings: Mfr=1, Product=3, SerialNumber=0 Aug 25 12:39:37 Nesbitt kernel: [ 5295.782150] usb 3-3: Product: Geek Out HD Audio 1V5 Aug 25 12:39:37 Nesbitt kernel: [ 5295.782151] usb 3-3: Manufacturer: LH Labs Aug 25 12:39:37 Nesbitt kernel: [ 5296.162338] usb 3-3: 1:2 : unsupported format bits 0x100000000 Aug 25 12:39:38 Nesbitt mtp-probe: checking bus 3, device 8: "/sys/devices/pci0000:00/0000:00:14.0/usb3/3-3" Aug 25 12:39:38 Nesbitt mtp-probe: bus: 3, device: 8 was not an MTP device Aug 25 12:39:38 Nesbitt systemd-udevd[10311]: controlC1: Process '/usr/sbin/alsactl -E HOME=/run/alsa -E XDG_RUNTIME_DIR=/run/alsa/runtime restore 1' failed with exit code 99. Aug 25 12:39:38 Nesbitt mtp-probe: checking bus 3, device 8: "/sys/devices/pci0000:00/0000:00:14.0/usb3/3-3" Aug 25 12:39:38 Nesbitt mtp-probe: bus: 3, device: 8 was not an MTP device Aug 25 12:39:43 Nesbitt kernel: [ 5301.921810] usb 3-3: 1:1: usb_set_interface failed (-110) Aug 25 12:39:43 Nesbitt kernel: [ 5301.922047] usb 3-3: 1:0: usb_set_interface failed (-71) Aug 25 12:39:43 Nesbitt kernel: [ 5301.922354] usb 3-3: 1:0: usb_set_interface failed (-71) Aug 25 12:39:43 Nesbitt kernel: [ 5301.922915] usb 3-3: 1:0: usb_set_interface failed (-71) I did a git bisect (from kernel source, not on debian-specific) and the offending commit has been found: # git bisect bad bf6313a0ff766925462e97b4e733d5952de02367 is the first bad commit commit bf6313a0ff766925462e97b4e733d5952de02367 Author: Takashi Iwai <tiwai(a)suse.de> Date: Mon Nov 23 09:53:31 2020 +0100 ALSA: usb-audio: Refactor endpoint management This is an intensive surgery for the endpoint and stream management for achieving more robust and clean code. The goals of this patch are: - More clear endpoint resource changes - The interface altsetting control in a single place Below are brief description of the whole changes. First off, most of the endpoint operations are moved into endpoint.c, so that the snd_usb_endpoint object is only referred in other places. The endpoint object is acquired and released via the new functions snd_usb_endpoint_open() and snd_usb_endpoint_close() that are called at PCM hw_params and hw_free callbacks, respectively. Those are ref-counted and EPs can manage the multiple opens. The open callback receives the audioformat and hw_params arguments, and those are used for initializing the EP parameters; especially the endpoint, interface and altset numbers are read from there, as well as the PCM parameters like the format, rate and channels. Those are stored in snd_usb_endpoint object. If it's the secondary open, the function checks whether the given parameters are compatible with the already opened EP setup, too. The coupling with a sync EP (including an implicit feedback sync) is done by the sole snd_usb_endpoint_set_sync() call. The configuration of each endpoint is done in a single shot via snd_usb_endpoint_configure() call. This is the place where most of PCM configurations are done. A few flags and special handling in the snd_usb_substream are dropped along with this change. A significant difference wrt the configuration from the previous code is the order of USB host interface setups. Now the interface is always disabled at beginning and (re-)enabled at the last step of snd_usb_endpoint_configure(), in order to be compliant with the standard UAC2/3. For UAC1, the interface is set before the parameter setups since there seem devices that require it (e.g. Yamaha THR10), just like how it was done in the previous driver code. The start/stop are almost same as before, also single-shots. The URB callbacks need to be set via snd_usb_endpoint_set_callback() like the previous code at the trigger phase, too. Finally, the flag for the re-setup is set at the device suspend through the full EP list, instead of PCM trigger. This catches the overlooked cases where the PCM hasn't been running yet but the device needs the full setup after resume. Tested-by: Keith Milner <kamilner(a)superlative.org> Tested-by: Dylan Robinson <dylan_robinson(a)motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-26-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> sound/usb/card.c | 8 +- sound/usb/card.h | 11 +- sound/usb/clock.c | 13 +- sound/usb/endpoint.c | 662 ++++++++++++++++++++++++++------------------------- sound/usb/endpoint.h | 40 ++-- sound/usb/pcm.c | 616 +++++++++++++++++++---------------------------- 6 files changed, 616 insertions(+), 734 deletions(-) That's a lot of code though. I'd like to be able to use my DAC on kernels newer than 5.10 so any hints on how to debug and fix this would be greatly appreciated. After doing this 17-point bisect I absolutely don't mind building a few more to fix this. For info, the debian bug report: https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=1011196 For info, a related report from Arch linux for the same device vendor: https://bugs.archlinux.org/task/70636 Thanks, Lennert

2 6

Re: [PATCH 10/13] ASoC: amd: enable Pink Sardine acp6.2 drivers build
by Syed Saba Kareem 27 Aug '22

27 Aug '22

On 8/12/22 19:35, Mark Brown wrote: > [CAUTION: External Email] ACP Driver has strict dependency on ACPI COMPILE_TEST will relax the flag options. As ACP PCI driver has strict dependency,compile test flag cannot be used in this case.

1 0

Tascam US122L, JACK, PulseAudio and OBS
by GitHub issues - opened 27 Aug '22

27 Aug '22

alsa-project/alsa-lib issue #269 was opened from stephematician: Apologies if this is the wrong place - however I'm getting an error most of the time in a very specific situation. My setup is a TASCAM US122L USB interface, which ALSA is capable of recognising and setting up as a usb_stream device. I then run jackd for this device, load the pulseaudio module for a JACK source (at this point I can record via parecord), and then attempt to add a PulseAudio input to OBS. At this point, I'll get an input for less than a second, then it'll cut out and I'll get an error in JACK. I've attached the output of alsa-info; showing that the device is recognised. The contents of my .asoundrc are somewhat widely used: ``` # The usb_stream plugin configuration pcm.!usb_stream { @args [ CARD ] @args.CARD { type string default "1" } type usb_stream card $CARD } ctl.!usb_stream { @args [ CARD ] @args.CARD { type string default "1" } type hw card $CARD } ``` I run jackd with a command like: ``` /usr/bin/jackd -v -p128 -dalsa -dusb_stream:CARD=US122L -r44100 -p256 -n2 -C ``` I then set up a source in PulseAudio using `pacmd load-module module-jack-source`. Typically at this point, `parecord --device=jack_in test.wav` records no problems. Then I'll run OBS and try to add a PulseAudio input, at which point I usually get a very short period of input (less than 1 second?) and then the input will cut out and I'll see the following error in jackd output: ``` Couldn't configure usb_stream : Input/output error ALSA: prepare error for capture on "usb_stream:CARD=US122L" (Input/output error) ``` Upon tracking this down, it seems that some call to snd_pcm_prepare by JACK is receiving an error code; but I'm not an expert in this library (or alsa) so it's hard for me to even guess what the problem could be. [alsa-info.txt](https://github.com/alsa-project/alsa-lib/files/9438112/alsa-… I have also raised this issue on JACK in case it's a problem with that library (link here - but details are the same https://github.com/jackaudio/jack1/issues/118#issue-1353053534) Issue URL : https://github.com/alsa-project/alsa-lib/issues/269 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Request for setup of new repositories
by Takashi Sakamoto 27 Aug '22

27 Aug '22

Hi Jaroslav, Iwai-san, Thanks for your maintenance for alsa-project organization in github.com. Currently I'd like to add new three repositories under the organization as a part of my work for ALSA firewire stack. I've been maintaining libhinawa since 2014 and recently realized that current design is not necessarily convenient since it includes two functions; operation to Linux FireWire cdev, and operation of ALSA HwDep cdev. Currently I'm working for new library to split the latter operation. Then I'd like you to setup below repositories: * 'libhitaki' * 'libhitaki-doc' * 'hitaki-rs' The library itself and its Rust binding are utilized by 'snd-firewire-ctl-services'[2], thus it's preferable to register them under 'GObject Introspection' Team. Thanks for your assist for my work. [1] https://github.com/alsa-project/libhinawa [2] https://github.com/alsa-project/snd-firewire-ctl-services Regards Takashi Sakamoto

2 20

USB-Audio: Fix for ALC4080 (ASUS ROG Strix B660-F Gaming WiFi)
by GitHub pull_request - edited 27 Aug '22

27 Aug '22

alsa-project/alsa-ucm-conf pull request #206 was edited from RomanAverin: Fix for ALC4080 (ASUS ROG Strix B660-F Gaming WiFi) Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/206 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/206.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Fix for ALC4080 (ASUS ROG Strix B660-F Gaming WiFi)
by GitHub pull_request - edited 27 Aug '22

27 Aug '22

alsa-project/alsa-ucm-conf pull request #206 was edited from RomanAverin: Fix for ALC4080 (ASUS ROG Strix B660-F Gaming WiFi) Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/206 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/206.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Update USB-Audio.conf
by GitHub pull_request - opened 27 Aug '22

27 Aug '22

alsa-project/alsa-ucm-conf pull request #206 was opened from RomanAverin: Fix for ALC4080 (ASUS ROG Strix B660-F Gaming WiFi) Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/206 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/206.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Asus ROG Strix B660-F (ALC4080) output issue: No audio front headphone, No audio S/PDIF
by GitHub issues - opened 27 Aug '22

27 Aug '22

alsa-project/alsa-ucm-conf issue #205 was opened from RomanAverin: I have problems playing audio from the front headphone jack, as well as S/PDIF. Kernel-5.19.4, Fedora 36, X11, Gnome 42 1. usb-device detect as `Bus 001 Device 004: ID 0b05:1a16 ASUSTek Computer, Inc. USB Audio` 2. Sound cards device by ` arecord -l ` ``` **** List of CAPTURE Hardware Devices **** card 0: Audio [USB Audio], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Audio [USB Audio], device 1: USB Audio [USB Audio #1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Audio [USB Audio], device 2: USB Audio [USB Audio #2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 3: J250a [Jabra UC VOICE 250a], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` 3. Info from `alsa-info.sh` uploaded [http://alsa-project.org/db/?f=4376afd5f9282787f5ce07b8dd473e48639b6201](htt… 4. Trying to change the device id in the `/usr/share/alsa/ucm2/USB-Audio/Realtek/ALC4080.conf` on my id `Regex "USB(0b05:1a16)"` (default 0b05:1996) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/205 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

MPG X570S EDGE MAX WIFI compatibility
by GitHub issues - edited 26 Aug '22

26 Aug '22

alsa-project/alsa-ucm-conf issue #172 was edited from yhojann-cl: Please, add `0db0:151f` to `realtek-alc4080` chip compatibility in MPG X570S EDGE MAX WIFI motherboard in https://github.com/alsa-project/alsa-ucm-conf/blob/master/ucm2/USB-Audio/US… : ``` $ lsusb | grep 0db0 Bus 001 Device 003: ID 0db0:151f Micro Star International USB Audio ``` In `ucm2/USB-Audio/USB-Audio.conf` add `151f` to regex of `0db0`: ``` Regex "USB((0b05:(1996|1a2[07]))|(0db0:(1feb|419c|a073|151f)))" ``` My Motherboard is MSI X570S and output audio works fine but: - Motherboard microphone plug is not work. - Frontal microphone plug is not work. - When change output from gnome settings to hdmi audio does not work, the audio continues to play in the plug and not in the monitor. - When I change the input or output from digital to analog or sinput it does not change at all. - The strangest thing is that the microphone input captures the audio that is heard in the system as if it were the output, audacious records the audio from the desktop when it tries to monitor the microphone. - When plugging or unplugging the microphone, the system detects this change and tries to change the audio input automatically, but it does not capture the microphone and the desktop continues to be heard. - When plug headphone in frontal plug the gnome setting change automaticaly to the headphone item, but the sound play in speakers, not in headphone. - On Audacity when capture Alsa + `USB Audio: #2 (hw: 1,2)` works fine!, but can not set on gnome settings. My alsa data: http://alsa-project.org/db/?f=e29a6593e48350efeba75f69d5ece62379973c35 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/172 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

MPG X570S EDGE MAX WIFI compatibility
by GitHub issues - edited 26 Aug '22

26 Aug '22

alsa-project/alsa-ucm-conf issue #172 was edited from yhojann-cl: Please, add `0db0:151f` to `realtek-alc4080` chip compatibility in MPG X570S EDGE MAX WIFI motherboard in https://github.com/alsa-project/alsa-ucm-conf/blob/master/ucm2/USB-Audio/US… : ``` $ lsusb | grep 0db0 Bus 001 Device 003: ID 0db0:151f Micro Star International USB Audio ``` In `ucm2/USB-Audio/USB-Audio.conf` add `151f` to regex of `0db0`: ``` Regex "USB((0b05:(1996|1a2[07]))|(0db0:(1feb|419c|a073|151f)))" ``` My Motherboard is MSI X570S and output audio works fine but: - Motherboard microphone plug is not work. - Frontal microphone plug is not work. - When change output from gnome settings to hdmi audio does not work, the audio continues to play in the plug and not in the monitor. - When I change the input or output from digital to analog or sinput it does not change at all. - The strangest thing is that the microphone input captures the audio that is heard in the system as if it were the output, audacious records the audio from the desktop when it tries to monitor the microphone. - When plugging or unplugging the microphone, the system detects this change and tries to change the audio input automatically, but it does not capture the microphone and the desktop continues to be heard. - When plug headphone in frontal plug the gnome setting change automaticaly to the headphone item, but the sound play in speakers, not in headphone. - On Audacity when capture Alsa + `USB Audio: #2 (hw: 1,2)` works fine!, but can not set on gnome settings. My alsa data: http://alsa-project.org/db/?f=df4d1f6c126260eabc5f159e64cff37efa064a38 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/172 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

MPG X570S EDGE MAX WIFI compatibility
by GitHub issues - edited 26 Aug '22

26 Aug '22

alsa-project/alsa-ucm-conf issue #172 was edited from yhojann-cl: Please, add `0db0:151f` to `realtek-alc4080` chip compatibility in MPG X570S EDGE MAX WIFI motherboard: ``` $ lsusb | grep 0db0 Bus 001 Device 003: ID 0db0:151f Micro Star International USB Audio ``` In `ucm2/USB-Audio/USB-Audio.conf` add `151f` to regex of `0db0`: ``` Regex "USB((0b05:(1996|1a2[07]))|(0db0:(1feb|419c|a073|151f)))" ``` My Motherboard is MSI X570S and output audio works fine but: - Motherboard microphone plug is not work. - Frontal microphone plug is not work. - When change output from gnome settings to hdmi audio does not work, the audio continues to play in the plug and not in the monitor. - When I change the input or output from digital to analog or sinput it does not change at all. - The strangest thing is that the microphone input captures the audio that is heard in the system as if it were the output, audacious records the audio from the desktop when it tries to monitor the microphone. - When plugging or unplugging the microphone, the system detects this change and tries to change the audio input automatically, but it does not capture the microphone and the desktop continues to be heard. - When plug headphone in frontal plug the gnome setting change automaticaly to the headphone item, but the sound play in speakers, not in headphone. - On Audacity when capture Alsa + `USB Audio: #2 (hw: 1,2)` works fine!, but can not set on gnome settings. My alsa data: http://alsa-project.org/db/?f=df4d1f6c126260eabc5f159e64cff37efa064a38 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/172 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

MPG X570S EDGE MAX WIFI compatibility
by GitHub issues - edited 26 Aug '22

26 Aug '22

alsa-project/alsa-ucm-conf issue #172 was edited from yhojann-cl: Please, add `0db0:151f` to `realtek-alc4080` chip compatibility in MPG X570S EDGE MAX WIFI motherboard: ``` $ lsusb | grep 0db0 Bus 001 Device 003: ID 0db0:151f Micro Star International USB Audio ``` In `ucm2/USB-Audio/USB-Audio.conf` add `151f` to regex of `0db0`: ``` Regex "USB((0b05:(1996|1a2[07]))|(0db0:(1feb|419c|a073|151f)))" ``` My Motherboard is MSI X570S and output audio works fine but: - Motherboard microphone plug is not work. - Frontal microphone plug is not work. - When change output from gnome settings to hdmi audio does not work, the audio continues to play in the plug and not in the monitor. - When I change the input or output from digital to analog or sinput it does not change at all. - The strangest thing is that the microphone input captures the audio that is heard in the system as if it were the output, audacious records the audio from the desktop when it tries to monitor the microphone. - When plugging or unplugging the microphone, the system detects this change and tries to change the audio input automatically, but it does not capture the microphone and the desktop continues to be heard. - When plug headphone in frontal plug the gnome setting change automaticaly to the headphone item, but the sound play in speakers, not in headphone. - On Audacity when capture Alsa + "USB Audio: #2 (hw: 1,2)" works fine!, but can not set on gnome settings. My alsa data: http://alsa-project.org/db/?f=df4d1f6c126260eabc5f159e64cff37efa064a38 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/172 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: Fix misspelling of 'system' as 'ststem'.
by Maurycy Z 26 Aug '22

26 Aug '22

Signed-off-by: Maurycy Zalewski <10maurycy10(a)gmail.com> --- sound/pci/ice1712/ice1712.c | 2 +- sound/pci/ice1712/ice1724.c | 2 +- sound/soc/fsl/Kconfig | 2 +- sound/soc/generic/Kconfig | 4 ++-- 4 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index a5241a287..ff6ce91bf 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2247,7 +2247,7 @@ static int snd_ice1712_read_eeprom(struct snd_ice1712 *ice, (snd_ice1712_read_i2c(ice, dev, 0x03) << 24); if (ice->eeprom.subvendor == 0 || ice->eeprom.subvendor == (unsigned int)-1) { - /* invalid subvendor from EEPROM, try the PCI subststem ID instead */ + /* invalid subvendor from EEPROM, try the PCI subsystem ID instead */ u16 vendor, device; pci_read_config_word(ice->pci, PCI_SUBSYSTEM_VENDOR_ID, &vendor); pci_read_config_word(ice->pci, PCI_SUBSYSTEM_ID, &device); diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 6fab2ad85..9e81e6335 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -2266,7 +2266,7 @@ static int snd_vt1724_read_eeprom(struct snd_ice1712 *ice, if (ice->eeprom.subvendor == 0 || ice->eeprom.subvendor == (unsigned int)-1) { /* invalid subvendor from EEPROM, try the PCI - * subststem ID instead + * subsystem ID instead */ u16 vendor, device; pci_read_config_word(ice->pci, PCI_SUBSYSTEM_VENDOR_ID, diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 614eceda6..c7fe12fbd 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -366,7 +366,7 @@ config SND_SOC_IMX_CARD help This option enables audio sound card support for i.MX boards with OF-graph DT bindings. - It also support DPCM of single CPU multi Codec ststem. + It also support DPCM of single CPU multi Codec system. endif # SND_IMX_SOC diff --git a/sound/soc/generic/Kconfig b/sound/soc/generic/Kconfig index b6df4e26b..955ac8d3f 100644 --- a/sound/soc/generic/Kconfig +++ b/sound/soc/generic/Kconfig @@ -7,7 +7,7 @@ config SND_SIMPLE_CARD select SND_SIMPLE_CARD_UTILS help This option enables generic simple sound card support - It also support DPCM of multi CPU single Codec ststem. + It also support DPCM of multi CPU single Codec system. config SND_AUDIO_GRAPH_CARD tristate "ASoC Audio Graph sound card support" @@ -16,7 +16,7 @@ config SND_AUDIO_GRAPH_CARD help This option enables generic simple sound card support with OF-graph DT bindings. - It also support DPCM of multi CPU single Codec ststem. + It also support DPCM of multi CPU single Codec system. config SND_AUDIO_GRAPH_CARD2 tristate "ASoC Audio Graph sound card2 support" -- 2.37.2

2 1

[PATCH] ALSA: control: Re-order bounds checking in get_ctl_id_hash()
by Dan Carpenter 26 Aug '22

26 Aug '22

These two checks are in the reverse order so it might read one element beyond the end of the array. First check if the "i" is within bounds before using it. Fixes: 6ab55ec0a938 ("ALSA: control: Fix an out-of-bounds bug in get_ctl_id_hash()") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/core/control.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/control.c b/sound/core/control.c index e8fc4c511e5f..a7271927d875 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -391,7 +391,7 @@ static unsigned long get_ctl_id_hash(const struct snd_ctl_elem_id *id) h = id->iface; h = MULTIPLIER * h + id->device; h = MULTIPLIER * h + id->subdevice; - for (i = 0; id->name[i] && i < SNDRV_CTL_ELEM_ID_NAME_MAXLEN; i++) + for (i = 0; i < SNDRV_CTL_ELEM_ID_NAME_MAXLEN && id->name[i]; i++) h = MULTIPLIER * h + id->name[i]; h = MULTIPLIER * h + id->index; h &= LONG_MAX; -- 2.35.1

2 1

Add interactive pause/resume functionality
by GitHub pull_request - edited 26 Aug '22

26 Aug '22

alsa-project/tinycompress pull request #14 was edited from LaurentiuM1234: We want to be able to interactively pause and resume the stream. Initially, when trying to play an MP3 song, cplay would hang during the polling on compress_write. The cause for this was the fact that, after calling `get_codec_mp3`, the file pointer would point to the first MP3 data, leaving out the first MP3 header. I'm assuming that this would cause the codec to be unable to process the received frames since it wouldn't be able to find the first MP3 header. In order to be able to use the pause/resume functionality on MP3 streams, this problem had to be addressed before adding the actual functionality. Request URL : https://github.com/alsa-project/tinycompress/pull/14 Patch URL : https://github.com/alsa-project/tinycompress/pull/14.patch Repository URL: https://github.com/alsa-project/tinycompress

1 0

Add interactive pause/resume functionality
by GitHub pull_request - edited 26 Aug '22

26 Aug '22

alsa-project/tinycompress pull request #14 was edited from LaurentiuM1234: We want to be able to interactively pause and resume the stream. Initially, when trying to play an MP3 song, cplay would hang during the polling on compress_write. The cause for this was probably the fact that, after calling `get_codec_mp3`, the file pointer would point to the first MP3 data, leaving out the first MP3 header. I'm assuming that this would cause the codec to hang since it wouldn't be able to find the first MP3 header. In order to be able to use the pause/resume functionality on MP3 streams, this problem had to be addressed before adding the actual functionality. Request URL : https://github.com/alsa-project/tinycompress/pull/14 Patch URL : https://github.com/alsa-project/tinycompress/pull/14.patch Repository URL: https://github.com/alsa-project/tinycompress

1 0

Add interactive pause/resume functionality
by GitHub pull_request - edited 26 Aug '22

26 Aug '22

alsa-project/tinycompress pull request #14 was edited from LaurentiuM1234: We want to be able to interactively pause and resume the stream Request URL : https://github.com/alsa-project/tinycompress/pull/14 Patch URL : https://github.com/alsa-project/tinycompress/pull/14.patch Repository URL: https://github.com/alsa-project/tinycompress

1 0

[PATCH 1/2] sound: realtek: Add pincfg for ASUS G533Z
by Luke D. Jones 26 Aug '22

26 Aug '22

Fixes up the pincfg for ASUS ROG Strix G15 (G533Z) laptop. Signed-off-by: Luke D. Jones <luke(a)ljones.dev> --- sound/pci/hda/patch_realtek.c | 15 ++++++++++++++- 1 file changed, 14 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fd630d62b5a0..e9cad49ee2a5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5828,7 +5828,7 @@ static void alc_fixup_headset_mode_alc255_no_hp_mic(struct hda_codec *codec, struct alc_spec *spec = codec->spec; spec->parse_flags |= HDA_PINCFG_HEADSET_MIC; alc255_set_default_jack_type(codec); - } + } else alc_fixup_headset_mode(codec, fix, action); } @@ -7022,6 +7022,7 @@ enum { ALC294_FIXUP_ASUS_GX502_HP, ALC294_FIXUP_ASUS_GX502_PINS, ALC294_FIXUP_ASUS_GX502_VERBS, + ALC285_FIXUP_ASUS_G533Z_PINS, ALC294_FIXUP_ASUS_GU502_HP, ALC294_FIXUP_ASUS_GU502_PINS, ALC294_FIXUP_ASUS_GU502_VERBS, @@ -8363,6 +8364,17 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc294_fixup_gu502_hp, }, + [ALC285_FIXUP_ASUS_G533Z_PINS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x14, 0x90170120 }, + { 0x19, 0x03A11050 }, /* front HP mic */ + { 0x1B, 0x03A11C30 }, /* rear external mic */ + { 0x21, 0x03211420 }, /* front HP out */ + { } + }, + .chained = false, + }, [ALC294_FIXUP_ASUS_COEF_1B] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -9294,6 +9306,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_DUAL_SPK), SND_PCI_QUIRK(0x1043, 0x1662, "ASUS GV301QH", ALC294_FIXUP_ASUS_DUAL_SPK), SND_PCI_QUIRK(0x1043, 0x1881, "ASUS Zephyrus S/M", ALC294_FIXUP_ASUS_GX502_PINS), + SND_PCI_QUIRK(0x1043, 0x1c92, "ASUS ROG Strix G15", ALC285_FIXUP_ASUS_G533Z_PINS), SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x18f1, "Asus FX505DT", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x194e, "ASUS UX563FD", ALC294_FIXUP_ASUS_HPE), -- 2.37.2

2 2

[PATCH v8 1/2] ASoC: sunxi: Add Allwinner H6 Digital MIC driver
by Ban Tao 26 Aug '22

26 Aug '22

The Allwinner H6 and later SoCs have an DMIC block which is capable of capture. Signed-off-by: Ban Tao <fengzheng923(a)gmail.com> --- v1->v2: 1.Fix some compilation errors. 2.Modify some code styles. v2->v3: None. v3->v4: 1.add sig_bits. v4->v5: None. v5->v6: 1.Modify RXFIFO_CTL_MODE to mode 1. v6->v7: 1.Modify dmic_rate_s to be a global variable. 2.Changed some macro names to make more sense. 3.Align code format. 4.Add a depends on PM to Kconfig entry. v7->v8: None. --- MAINTAINERS | 7 + sound/soc/sunxi/Kconfig | 8 + sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun50i-dmic.c | 405 ++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 421 insertions(+) create mode 100644 sound/soc/sunxi/sun50i-dmic.c diff --git a/MAINTAINERS b/MAINTAINERS index e9d5b05..839f625 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -796,6 +796,13 @@ L: linux-media(a)vger.kernel.org S: Maintained F: drivers/staging/media/sunxi/cedrus/ +ALLWINNER DMIC DRIVERS +M: Ban Tao <fengzheng923(a)gmail.com> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained +F: Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml +F: sound/soc/sunxi/sun50i-dmic.c + ALPHA PORT M: Richard Henderson <rth(a)twiddle.net> M: Ivan Kokshaysky <ink(a)jurassic.park.msu.ru> diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index ddcaaa9..8543234 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -56,6 +56,14 @@ config SND_SUN4I_SPDIF Say Y or M to add support for the S/PDIF audio block in the Allwinner A10 and affiliated SoCs. +config SND_SUN50I_DMIC + tristate "Allwinner H6 DMIC Support" + depends on (OF && PM && ARCH_SUNXI) || COMPILE_TEST + select SND_SOC_GENERIC_DMAENGINE_PCM + help + Say Y or M to add support for the DMIC audio block in the Allwinner + H6 and affiliated SoCs. + config SND_SUN8I_ADDA_PR_REGMAP tristate select REGMAP diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile index a86be34..4483fe9 100644 --- a/sound/soc/sunxi/Makefile +++ b/sound/soc/sunxi/Makefile @@ -6,3 +6,4 @@ obj-$(CONFIG_SND_SUN8I_CODEC_ANALOG) += sun8i-codec-analog.o obj-$(CONFIG_SND_SUN50I_CODEC_ANALOG) += sun50i-codec-analog.o obj-$(CONFIG_SND_SUN8I_CODEC) += sun8i-codec.o obj-$(CONFIG_SND_SUN8I_ADDA_PR_REGMAP) += sun8i-adda-pr-regmap.o +obj-$(CONFIG_SND_SUN50I_DMIC) += sun50i-dmic.o diff --git a/sound/soc/sunxi/sun50i-dmic.c b/sound/soc/sunxi/sun50i-dmic.c new file mode 100644 index 0000000..76b3378 --- /dev/null +++ b/sound/soc/sunxi/sun50i-dmic.c @@ -0,0 +1,405 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +// +// This driver supports the DMIC in Allwinner's H6 SoCs. +// +// Copyright 2021 Ban Tao <fengzheng923(a)gmail.com> + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/of_device.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <linux/reset.h> +#include <sound/dmaengine_pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#define SUN50I_DMIC_EN_CTL (0x00) + #define SUN50I_DMIC_EN_CTL_GLOBE BIT(8) + #define SUN50I_DMIC_EN_CTL_CHAN(v) ((v) << 0) + #define SUN50I_DMIC_EN_CTL_CHAN_MASK GENMASK(7, 0) +#define SUN50I_DMIC_SR (0x04) + #define SUN50I_DMIC_SR_SAMPLE_RATE(v) ((v) << 0) + #define SUN50I_DMIC_SR_SAMPLE_RATE_MASK GENMASK(2, 0) +#define SUN50I_DMIC_CTL (0x08) + #define SUN50I_DMIC_CTL_OVERSAMPLE_RATE BIT(0) +#define SUN50I_DMIC_DATA (0x10) +#define SUN50I_DMIC_INTC (0x14) + #define SUN50I_DMIC_FIFO_DRQ_EN BIT(2) +#define SUN50I_DMIC_INT_STA (0x18) + #define SUN50I_DMIC_INT_STA_OVERRUN_IRQ_PENDING BIT(1) + #define SUN50I_DMIC_INT_STA_DATA_IRQ_PENDING BIT(0) +#define SUN50I_DMIC_RXFIFO_CTL (0x1c) + #define SUN50I_DMIC_RXFIFO_CTL_FLUSH BIT(31) + #define SUN50I_DMIC_RXFIFO_CTL_MODE_MASK BIT(9) + #define SUN50I_DMIC_RXFIFO_CTL_MODE_LSB (0 << 9) + #define SUN50I_DMIC_RXFIFO_CTL_MODE_MSB (1 << 9) + #define SUN50I_DMIC_RXFIFO_CTL_SAMPLE_MASK BIT(8) + #define SUN50I_DMIC_RXFIFO_CTL_SAMPLE_16 (0 << 8) + #define SUN50I_DMIC_RXFIFO_CTL_SAMPLE_24 (1 << 8) +#define SUN50I_DMIC_CH_NUM (0x24) + #define SUN50I_DMIC_CH_NUM_N(v) ((v) << 0) + #define SUN50I_DMIC_CH_NUM_N_MASK GENMASK(2, 0) +#define SUN50I_DMIC_CNT (0x2c) + #define SUN50I_DMIC_CNT_N BIT(0) +#define SUN50I_DMIC_HPF_CTRL (0x38) +#define SUN50I_DMIC_VERSION (0x50) + +struct sun50i_dmic_dev { + struct clk *dmic_clk; + struct clk *bus_clk; + struct reset_control *rst; + struct regmap *regmap; + struct snd_dmaengine_dai_dma_data dma_params_rx; +}; + +struct dmic_rate { + unsigned int samplerate; + unsigned int rate_bit; +}; + +static struct dmic_rate dmic_rate_s[] = { + {48000, 0x0}, + {44100, 0x0}, + {32000, 0x1}, + {24000, 0x2}, + {22050, 0x2}, + {16000, 0x3}, + {12000, 0x4}, + {11025, 0x4}, + {8000, 0x5}, +}; + +static int sun50i_dmic_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); + + /* only support capture */ + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_FLUSH, + SUN50I_DMIC_RXFIFO_CTL_FLUSH); + regmap_write(host->regmap, SUN50I_DMIC_CNT, SUN50I_DMIC_CNT_N); + + return 0; +} + +static int sun50i_dmic_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + int i = 0; + unsigned long rate = params_rate(params); + unsigned int mclk = 0; + unsigned int channels = params_channels(params); + unsigned int chan_en = (1 << channels) - 1; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + + /* DMIC num is N+1 */ + regmap_update_bits(host->regmap, SUN50I_DMIC_CH_NUM, + SUN50I_DMIC_CH_NUM_N_MASK, + SUN50I_DMIC_CH_NUM_N(channels - 1)); + regmap_write(host->regmap, SUN50I_DMIC_HPF_CTRL, chan_en); + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_CHAN_MASK, + SUN50I_DMIC_EN_CTL_CHAN(chan_en)); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_SAMPLE_MASK, + SUN50I_DMIC_RXFIFO_CTL_SAMPLE_16); + break; + case SNDRV_PCM_FORMAT_S24_LE: + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_SAMPLE_MASK, + SUN50I_DMIC_RXFIFO_CTL_SAMPLE_24); + break; + default: + dev_err(cpu_dai->dev, "Invalid format!\n"); + return -EINVAL; + } + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_MODE_MASK, + SUN50I_DMIC_RXFIFO_CTL_MODE_MSB); + + switch (rate) { + case 11025: + case 22050: + case 44100: + mclk = 22579200; + break; + case 8000: + case 12000: + case 16000: + case 24000: + case 32000: + case 48000: + mclk = 24576000; + break; + default: + dev_err(cpu_dai->dev, "Invalid rate!\n"); + return -EINVAL; + } + + if (clk_set_rate(host->dmic_clk, mclk)) { + dev_err(cpu_dai->dev, "mclk : %u not support\n", mclk); + return -EINVAL; + } + + for (i = 0; i < ARRAY_SIZE(dmic_rate_s); i++) { + if (dmic_rate_s[i].samplerate == rate) { + regmap_update_bits(host->regmap, SUN50I_DMIC_SR, + SUN50I_DMIC_SR_SAMPLE_RATE_MASK, + SUN50I_DMIC_SR_SAMPLE_RATE(dmic_rate_s[i].rate_bit)); + break; + } + } + + switch (params_physical_width(params)) { + case 16: + host->dma_params_rx.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + break; + case 32: + host->dma_params_rx.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + break; + default: + dev_err(cpu_dai->dev, "Unsupported physical sample width: %d\n", + params_physical_width(params)); + return -EINVAL; + } + + /* oversamplerate adjust */ + if (params_rate(params) >= 24000) + regmap_update_bits(host->regmap, SUN50I_DMIC_CTL, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE); + else + regmap_update_bits(host->regmap, SUN50I_DMIC_CTL, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE, 0); + + return 0; +} + +static int sun50i_dmic_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + int ret = 0; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* DRQ ENABLE */ + regmap_update_bits(host->regmap, SUN50I_DMIC_INTC, + SUN50I_DMIC_FIFO_DRQ_EN, + SUN50I_DMIC_FIFO_DRQ_EN); + /* Global enable */ + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_GLOBE, + SUN50I_DMIC_EN_CTL_GLOBE); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + /* DRQ DISABLE */ + regmap_update_bits(host->regmap, SUN50I_DMIC_INTC, + SUN50I_DMIC_FIFO_DRQ_EN, 0); + /* Global disable */ + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_GLOBE, 0); + break; + default: + ret = -EINVAL; + break; + } + return ret; +} + +static int sun50i_dmic_soc_dai_probe(struct snd_soc_dai *dai) +{ + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, NULL, &host->dma_params_rx); + + return 0; +} + +static const struct snd_soc_dai_ops sun50i_dmic_dai_ops = { + .startup = sun50i_dmic_startup, + .trigger = sun50i_dmic_trigger, + .hw_params = sun50i_dmic_hw_params, +}; + +static const struct regmap_config sun50i_dmic_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = SUN50I_DMIC_VERSION, + .cache_type = REGCACHE_NONE, +}; + +#define SUN50I_DMIC_RATES (SNDRV_PCM_RATE_8000_48000) +#define SUN50I_DMIC_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_driver sun50i_dmic_dai = { + .capture = { + .channels_min = 1, + .channels_max = 8, + .rates = SUN50I_DMIC_RATES, + .formats = SUN50I_DMIC_FORMATS, + .sig_bits = 21, + }, + .probe = sun50i_dmic_soc_dai_probe, + .ops = &sun50i_dmic_dai_ops, + .name = "dmic", +}; + +static const struct of_device_id sun50i_dmic_of_match[] = { + { + .compatible = "allwinner,sun50i-h6-dmic", + }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, sun50i_dmic_of_match); + +static const struct snd_soc_component_driver sun50i_dmic_component = { + .name = "sun50i-dmic", +}; + +static int sun50i_dmic_runtime_suspend(struct device *dev) +{ + struct sun50i_dmic_dev *host = dev_get_drvdata(dev); + + clk_disable_unprepare(host->dmic_clk); + clk_disable_unprepare(host->bus_clk); + + return 0; +} + +static int sun50i_dmic_runtime_resume(struct device *dev) +{ + struct sun50i_dmic_dev *host = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(host->dmic_clk); + if (ret) + return ret; + + ret = clk_prepare_enable(host->bus_clk); + if (ret) { + clk_disable_unprepare(host->dmic_clk); + return ret; + } + + return 0; +} + +static int sun50i_dmic_probe(struct platform_device *pdev) +{ + struct sun50i_dmic_dev *host; + struct resource *res; + int ret; + void __iomem *base; + + host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); + if (!host) + return -ENOMEM; + + /* Get the addresses */ + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) + return dev_err_probe(&pdev->dev, PTR_ERR(base), + "get resource failed.\n"); + + host->regmap = devm_regmap_init_mmio(&pdev->dev, base, + &sun50i_dmic_regmap_config); + + /* Clocks */ + host->bus_clk = devm_clk_get(&pdev->dev, "bus"); + if (IS_ERR(host->bus_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->bus_clk), + "failed to get bus clock.\n"); + + host->dmic_clk = devm_clk_get(&pdev->dev, "mod"); + if (IS_ERR(host->dmic_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->dmic_clk), + "failed to get dmic clock.\n"); + + host->dma_params_rx.addr = res->start + SUN50I_DMIC_DATA; + host->dma_params_rx.maxburst = 8; + + platform_set_drvdata(pdev, host); + + host->rst = devm_reset_control_get_optional_exclusive(&pdev->dev, NULL); + if (IS_ERR(host->rst)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->rst), + "Failed to get reset.\n"); + reset_control_deassert(host->rst); + + ret = devm_snd_soc_register_component(&pdev->dev, &sun50i_dmic_component, + &sun50i_dmic_dai, 1); + if (ret) + return dev_err_probe(&pdev->dev, ret, + "failed to register component.\n"); + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = sun50i_dmic_runtime_resume(&pdev->dev); + if (ret) + goto err_disable_runtime_pm; + } + + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) + goto err_suspend; + + return 0; +err_suspend: + if (!pm_runtime_status_suspended(&pdev->dev)) + sun50i_dmic_runtime_suspend(&pdev->dev); +err_disable_runtime_pm: + pm_runtime_disable(&pdev->dev); + return ret; +} + +static int sun50i_dmic_remove(struct platform_device *pdev) +{ + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + sun50i_dmic_runtime_suspend(&pdev->dev); + + return 0; +} + +static const struct dev_pm_ops sun50i_dmic_pm = { + SET_RUNTIME_PM_OPS(sun50i_dmic_runtime_suspend, + sun50i_dmic_runtime_resume, NULL) +}; + +static struct platform_driver sun50i_dmic_driver = { + .driver = { + .name = "sun50i-dmic", + .of_match_table = of_match_ptr(sun50i_dmic_of_match), + .pm = &sun50i_dmic_pm, + }, + .probe = sun50i_dmic_probe, + .remove = sun50i_dmic_remove, +}; + +module_platform_driver(sun50i_dmic_driver); + +MODULE_DESCRIPTION("Allwinner sun50i DMIC SoC Interface"); +MODULE_AUTHOR("Ban Tao <fengzheng923(a)gmail.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:sun50i-dmic"); -- 2.7.4

3 3

[PATCH] ASoC: wm_adsp: Silent parsing error on loading speaker protection fw
by Cristian Ciocaltea 26 Aug '22

26 Aug '22

The tracing capabilities for the speaker protection fw enabled via commit c55b3e46cb99 ("ASoC: wm_adsp: Add trace caps to speaker protection FW") are not be available on all platforms, such as the Valve's Steam Deck which is based on the Halo Core DSP. As a consequence, whenever the firmware is loaded, a rather misleading 'Failed to parse legacy: -19' error message is written to the kernel ring buffer: [ 288.977412] steamdeck kernel: cs35l41 spi-VLV1776:01: DSP1: Firmware version: 3 [ 288.978002] steamdeck kernel: cs35l41 spi-VLV1776:01: DSP1: cs35l41-dsp1-spk-prot.wmfw: Fri 02 Apr 2021 21:03:50 W. Europe Daylight Time [ 289.094065] steamdeck kernel: cs35l41 spi-VLV1776:01: DSP1: Firmware: 400a4 vendor: 0x2 v0.33.0, 2 algorithms [ 289.095073] steamdeck kernel: cs35l41 spi-VLV1776:01: DSP1: 0: ID cd v29.53.0 XM@94 YM@e [ 289.095665] steamdeck kernel: cs35l41 spi-VLV1776:01: DSP1: 1: ID f20b v0.0.1 XM@170 YM@0 [ 289.096275] steamdeck kernel: cs35l41 spi-VLV1776:01: DSP1: Protection: C:\Users\ocanavan\Desktop\cirrusTune_july2021.bin [ 291.172383] steamdeck kernel: cs35l41 spi-VLV1776:01: DSP1: Failed to parse legacy: -19 Update wm_adsp_buffer_init() to *not* report the ENODEV error when the firmware type is WM_ADSP_FW_SPK_PROT. Fixes: c55b3e46cb99 ("ASoC: wm_adsp: Add trace caps to speaker protection FW") Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea(a)collabora.com> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index cfaa45ede916..7514fc03b468 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1602,7 +1602,7 @@ static int wm_adsp_buffer_init(struct wm_adsp *dsp) if (list_empty(&dsp->buffer_list)) { /* Fall back to legacy support */ ret = wm_adsp_buffer_parse_legacy(dsp); - if (ret) + if (ret && (dsp->fw != WM_ADSP_FW_SPK_PROT || ret != -ENODEV)) adsp_warn(dsp, "Failed to parse legacy: %d\n", ret); } base-commit: 568035b01cfb107af8d2e4bd2fb9aea22cf5b868 -- 2.37.2

2 2

[PATCH v2 1/2] ASoC: dt-bindings: max98396: Document data monitor properties
by Daniel Mack 25 Aug '22

25 Aug '22

This device features a data monitor that puts the device in software reset upon a configurable set of events. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- v1 -> v2: fix a typo and remove a stray blank line .../bindings/sound/adi,max98396.yaml | 37 +++++++++++++++++++ 1 file changed, 37 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/adi,max98396.yaml b/Documentation/devicetree/bindings/sound/adi,max98396.yaml index 8d2ef991db40..f4eefe10c494 100644 --- a/Documentation/devicetree/bindings/sound/adi,max98396.yaml +++ b/Documentation/devicetree/bindings/sound/adi,max98396.yaml @@ -78,6 +78,43 @@ properties: interleaved on a single output channel. type: boolean + adi,dmon-stuck-enable: + description: + Enables the "data monitor stuck" feature. Once the data monitor is + enabled, it actively monitors the selected input data (from DIN) to the + speaker amplifier. Once a data error is detected, the data monitor + automatically places the device into software shutdown. + type: boolean + + adi,dmon-stuck-threshold-bits: + description: + Sets the threshold for the "data monitor stuck" feature, in bits. + $ref: "/schemas/types.yaml#/definitions/uint32" + enum: [9, 11, 13, 15] + default: 15 + + adi,dmon-magnitude-enable: + description: + Enables the "data monitor magnitude" feature. Once the data monitor is + enabled, it actively monitors the selected input data (from DIN) to the + speaker amplifier. Once a data error is detected, the data monitor + automatically places the device into software shutdown. + type: boolean + + adi,dmon-magnitude-threshold-bits: + description: + Sets the threshold for the "data monitor magnitude" feature, in bits. + $ref: "/schemas/types.yaml#/definitions/uint32" + enum: [2, 3, 4, 5] + default: 5 + + adi,dmon-duration-msecs: + description: + Sets the duration for the "data monitor" feature, in milliseconds. + $ref: "/schemas/types.yaml#/definitions/uint32" + enum: [64, 256, 1024, 4096] + default: 64 + reset-gpios: maxItems: 1 -- 2.37.2

2 2

[PATCH 1/2] lib/string_helpers: Introduce strsplit_u32()
by Cezary Rojewski 25 Aug '22

25 Aug '22

Add strsplit_u32() and its __user variant to allow for splitting specified string into array of u32 tokens. Originally this functionality was added for the SOF sound driver. As more users are on the horizon, relocate it so it becomes a common good. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- include/linux/string_helpers.h | 3 + lib/string_helpers.c | 96 +++++++++++++++++++++++++++++++ sound/soc/sof/sof-client-probes.c | 51 +--------------- 3 files changed, 100 insertions(+), 50 deletions(-) diff --git a/include/linux/string_helpers.h b/include/linux/string_helpers.h index 4d72258d42fd..a4630ddfca27 100644 --- a/include/linux/string_helpers.h +++ b/include/linux/string_helpers.h @@ -126,4 +126,7 @@ static inline const char *str_enabled_disabled(bool v) return v ? "enabled" : "disabled"; } +int strsplit_u32(const char *str, const char *delim, u32 **tkns, size_t *num_tkns); +int strsplit_u32_user(const char __user *from, size_t count, loff_t *ppos, const char *delim, + u32 **tkns, size_t *num_tkns); #endif diff --git a/lib/string_helpers.c b/lib/string_helpers.c index 5ed3beb066e6..bb24f0c62539 100644 --- a/lib/string_helpers.c +++ b/lib/string_helpers.c @@ -984,3 +984,99 @@ void fortify_panic(const char *name) } EXPORT_SYMBOL(fortify_panic); #endif /* CONFIG_FORTIFY_SOURCE */ + +/** + * strsplit_u32 - Split string into sequence of u32 tokens + * @str: The string to split into tokens. + * @delim: The string containing delimiter characters. + * @tkns: Returned u32 sequence pointer. + * @num_tkns: Returned number of tokens obtained. + * + * On success @num_tkns and @tkns are assigned the number of tokens extracted + * and the array itself respectively. + * Caller takes responsibility for freeing @tkns when no longer needed. + */ +int strsplit_u32(const char *str, const char *delim, u32 **tkns, size_t *num_tkns) +{ + size_t max_count = 32; + size_t count = 0; + char *s, **p; + u32 *buf, *tmp; + int ret = 0; + + p = (char **)&str; + *tkns = NULL; + *num_tkns = 0; + + buf = kcalloc(max_count, sizeof(*buf), GFP_KERNEL); + if (!buf) + return -ENOMEM; + + while ((s = strsep(p, delim)) != NULL) { + ret = kstrtouint(s, 0, buf + count); + if (ret) + goto free_buf; + + if (++count > max_count) { + max_count *= 2; + tmp = krealloc(buf, max_count * sizeof(*buf), GFP_KERNEL); + if (!tmp) { + ret = -ENOMEM; + goto free_buf; + } + buf = tmp; + } + } + + if (!count) + goto free_buf; + *tkns = kmemdup(buf, count * sizeof(*buf), GFP_KERNEL); + if (*tkns == NULL) { + ret = -ENOMEM; + goto free_buf; + } + *num_tkns = count; + +free_buf: + kfree(buf); + return ret; +} +EXPORT_SYMBOL(strsplit_u32); + +/** + * strsplit_u32_user - Split string into sequence of u32 tokens + * @from: The user space buffer to read from + * @ppos: The current position in the buffer + * @count: The maximum number of bytes to read + * @delim: The string containing delimiter characters. + * @tkns: Returned u32 sequence pointer. + * @num_tkns: Returned number of tokens obtained. + * + * On success @num_tkns and @tkns are assigned the number of tokens extracted + * and the array itself respectively. + * Caller takes responsibility for freeing @tkns when no longer needed. + */ +int strsplit_u32_user(const char __user *from, size_t count, loff_t *ppos, const char *delim, + u32 **tkns, size_t *num_tkns) +{ + char *buf; + int ret; + + buf = kmalloc(count + 1, GFP_KERNEL); + if (!buf) + return -ENOMEM; + + ret = simple_write_to_buffer(buf, count, ppos, from, count); + if (ret != count) { + ret = (ret < 0) ? ret : -EIO; + goto free_buf; + } + + buf[count] = '\0'; + ret = strsplit_u32(buf, delim, tkns, num_tkns); + +free_buf: + kfree(buf); + return ret; +} +EXPORT_SYMBOL(strsplit_u32_user); diff --git a/sound/soc/sof/sof-client-probes.c b/sound/soc/sof/sof-client-probes.c index 1f1ea93a7fbf..48ebbe58e2b9 100644 --- a/sound/soc/sof/sof-client-probes.c +++ b/sound/soc/sof/sof-client-probes.c @@ -12,6 +12,7 @@ #include <linux/debugfs.h> #include <linux/module.h> #include <linux/pm_runtime.h> +#include <linux/string_helpers.h> #include <sound/soc.h> #include <sound/sof/header.h> #include "sof-client.h" @@ -410,56 +411,6 @@ static const struct snd_compress_ops sof_probes_compressed_ops = { .copy = sof_probes_compr_copy, }; -/** - * strsplit_u32 - Split string into sequence of u32 tokens - * @buf: String to split into tokens. - * @delim: String containing delimiter characters. - * @tkns: Returned u32 sequence pointer. - * @num_tkns: Returned number of tokens obtained. - */ -static int strsplit_u32(char *buf, const char *delim, u32 **tkns, size_t *num_tkns) -{ - char *s; - u32 *data, *tmp; - size_t count = 0; - size_t cap = 32; - int ret = 0; - - *tkns = NULL; - *num_tkns = 0; - data = kcalloc(cap, sizeof(*data), GFP_KERNEL); - if (!data) - return -ENOMEM; - - while ((s = strsep(&buf, delim)) != NULL) { - ret = kstrtouint(s, 0, data + count); - if (ret) - goto exit; - if (++count >= cap) { - cap *= 2; - tmp = krealloc(data, cap * sizeof(*data), GFP_KERNEL); - if (!tmp) { - ret = -ENOMEM; - goto exit; - } - data = tmp; - } - } - - if (!count) - goto exit; - *tkns = kmemdup(data, count * sizeof(*data), GFP_KERNEL); - if (!(*tkns)) { - ret = -ENOMEM; - goto exit; - } - *num_tkns = count; - -exit: - kfree(data); - return ret; -} - static int tokenize_input(const char __user *from, size_t count, loff_t *ppos, u32 **tkns, size_t *num_tkns) { -- 2.25.1

5 29

[PATCH -next] ASoC: sigmadsp: switch to use kmemdup_nul() helper
by Yang Yingliang 25 Aug '22

25 Aug '22

Use kmemdup_nul() helper instead of open-coding to simplify the code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/codecs/sigmadsp.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c index b992216aee55..3047a6fbb380 100644 --- a/sound/soc/codecs/sigmadsp.c +++ b/sound/soc/codecs/sigmadsp.c @@ -227,13 +227,11 @@ static int sigma_fw_load_control(struct sigmadsp *sigmadsp, if (!ctrl) return -ENOMEM; - name = kzalloc(name_len + 1, GFP_KERNEL); + name = kmemdup_nul(ctrl_chunk->name, name_len, GFP_KERNEL); if (!name) { ret = -ENOMEM; goto err_free_ctrl; } - memcpy(name, ctrl_chunk->name, name_len); - name[name_len] = '\0'; ctrl->name = name; /* -- 2.25.1

2 1

[PATCH v2 0/2] ASoC: mediatek: dt-bindings: modify machine
by Chunxu Li 25 Aug '22

25 Aug '22

From: "chunxu.li" <chunxu.li(a)mediatek.com> Changes since V1: - remove unnecessary quotes Add SOF related field. 1. Add a property "mediatek,adsp", Only when adsp phandle could be retrieved, from DTS, the SOF related part of machine driver is executed. 2. Add a property "mediatek,dai-link" to support dai-links could be specified from DTS chunxu.li (2): ASoC: mediatek: dt-bindings: modify machine bindings for SOF ASoC: mediatek: dt-bindings: modify machine bindings for SOF .../bindings/sound/mt8186-mt6366-da7219-max98357.yaml | 10 ++++++++++ .../bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml | 10 ++++++++++ 2 files changed, 20 insertions(+) -- 2.25.1

3 7

[PATCH 1/2] ASoC: fsl_aud2htx: register platform component before registering cpu dai
by Shengjiu Wang 25 Aug '22

25 Aug '22

There is no defer probe when adding platform component to snd_soc_pcm_runtime(rtd), the code is in snd_soc_add_pcm_runtime() snd_soc_register_card() -> snd_soc_bind_card() -> snd_soc_add_pcm_runtime() -> adding cpu dai -> adding codec dai -> adding platform component. So if the platform component is not ready at that time, then the sound card still registered successfully, but platform component is empty, the sound card can't be used. As there is defer probe checking for cpu dai component, then register platform component before cpu dai to avoid such issue. And the behavior of imx_pcm_dma_init() is same as common devm_snd_dmaengine_pcm_register(), so use devm_snd_dmaengine_pcm_register() instead Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_aud2htx.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_aud2htx.c b/sound/soc/fsl/fsl_aud2htx.c index 873295f59ad7..bc1b9c6df95c 100644 --- a/sound/soc/fsl/fsl_aud2htx.c +++ b/sound/soc/fsl/fsl_aud2htx.c @@ -234,6 +234,16 @@ static int fsl_aud2htx_probe(struct platform_device *pdev) regcache_cache_only(aud2htx->regmap, true); + /* + * Register platform component before registering cpu dai for there + * is not defer probe for platform component in snd_soc_add_pcm_runtime(). + */ + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) { + dev_err(&pdev->dev, "failed to pcm register\n"); + return ret; + } + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_aud2htx_component, &fsl_aud2htx_dai, 1); @@ -242,10 +252,6 @@ static int fsl_aud2htx_probe(struct platform_device *pdev) return ret; } - ret = imx_pcm_dma_init(pdev); - if (ret) - dev_err(&pdev->dev, "failed to init imx pcm dma: %d\n", ret); - return ret; } -- 2.34.1

2 2

[PATCH 1/2] ASoC: dt-bindings: max98396: Document data monitor properties
by Daniel Mack 25 Aug '22

25 Aug '22

This device features a data monitor that puts the device in software reset upon a configurable set of events. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- .../bindings/sound/adi,max98396.yaml | 37 +++++++++++++++++++ 1 file changed, 37 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/adi,max98396.yaml b/Documentation/devicetree/bindings/sound/adi,max98396.yaml index 8d2ef991db40..f4eefe10c494 100644 --- a/Documentation/devicetree/bindings/sound/adi,max98396.yaml +++ b/Documentation/devicetree/bindings/sound/adi,max98396.yaml @@ -78,6 +78,43 @@ properties: interleaved on a single output channel. type: boolean + adi,dmon-stuck-enable: + description: + Enables the "data monitor stuck" feature. Once the data monitor is + enabled, it actively monitors the selected input data (from DIN) to the + speaker amplifier. Once a data error is detected, the data monitor + automatically places the device into software shutdown. + type: boolean + + adi,dmon-stuck-threshold-bits: + description: + Sets the threshold for the "data monitor stuck" feature, in bits. + $ref: "/schemas/types.yaml#/definitions/uint32" + enum: [9, 11, 13, 15] + default: 15 + + adi,dmon-magnitude-enable: + description: + Enables the "data monitor magnitude" feature. Once the data monitor is + enabled, it actively monitors the selected input data (from DIN) to the + speaker amplifier. Once a data error is detected, the data monitor + automatically places the device into software shutdown. + type: boolean + + adi,dmon-magnitude-threshold-bits: + description: + Sets the threshold for the "data monitor magnitude" feature, in bits. + $ref: "/schemas/types.yaml#/definitions/uint32" + enum: [2, 3, 4, 5] + default: 5 + + adi,dmon-duration-msecs: + description: + Sets the duration for the "data monitor" feature, in milliseconds. + $ref: "/schemas/types.yaml#/definitions/uint32" + enum: [64, 256, 1024, 4096] + default: 64 + reset-gpios: maxItems: 1 -- 2.37.2

1 1

[PATCH] ASoC: dapm: Export new 'graph.dot' file in debugfs
by Martin Povišer 25 Aug '22

25 Aug '22

Provide a DOT summary of the DAPM graph in a newly added 'graph.dot' file in debugfs, placed in the card's DAPM directory. Signed-off-by: Martin Povišer <povik+lin(a)cutebit.org> --- Sample output: https://cutebit.org/macaudio-j274.svg (With unupstreamed sound drivers on Mac mini (2020)) The helper bufprintf macro triggers checkpath.pl: ERROR: Macros with complex values should be enclosed in parentheses #47: FILE: sound/soc/soc-dapm.c:2235: +#define bufprintf(...) \ + ret += scnprintf(buf + ret, bufsize - ret, __VA_ARGS__) but adding in {} to the macro body interferes with the if/else constructions later, so I left it as-is. sound/soc/soc-dapm.c | 141 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 141 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 73b8bd452ca7..86524908c3fd 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2210,6 +2210,143 @@ static const struct file_operations dapm_bias_fops = { .llseek = default_llseek, }; +static ssize_t dapm_graph_read_file(struct file *file, char __user *user_buf, + size_t count, loff_t *ppos) +{ + struct snd_soc_card *card = file->private_data; + struct snd_soc_dapm_context *dapm; + struct snd_soc_dapm_path *p; + struct snd_soc_dapm_widget *w; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dapm_widget *wdone[16]; + struct snd_soc_dai *dai; + int i, num_wdone = 0, cluster = 0; + char *buf; + ssize_t bufsize; + ssize_t ret = 0; + + bufsize = 1024 * card->num_dapm_widgets; + buf = kmalloc(bufsize, GFP_KERNEL); + if (!buf) + return -ENOMEM; + + mutex_lock(&card->dapm_mutex); + +#define bufprintf(...) \ + ret += scnprintf(buf + ret, bufsize - ret, __VA_ARGS__) + + bufprintf("digraph dapm {\n"); + bufprintf("label=\"%s\";\n", card->name); + + /* + * Print the user-visible PCM devices of the card. + */ + bufprintf("subgraph cluster_%d {\n", cluster++); + bufprintf("label=\"PCM devices\";style=filled;fillcolor=lightgray;\n"); + for_each_card_rtds(card, rtd) { + if (rtd->dai_link->no_pcm) + continue; + + bufprintf("w%pK [label=\"%d: %s\"];\n", rtd, + rtd->pcm->device, rtd->dai_link->name); + } + bufprintf("};\n"); + + /* + * Print the playback/capture widgets of CPU-side DAIs, and link + * them to the PCM devices. Keep a list of already printed + * widgets in 'wdone', so they will be skipped later. Do not put + * these widgets in a component cluster like we will do with + * the other widgets later, since that just clutters the graph. + */ + for_each_card_rtds(card, rtd) { + for_each_rtd_cpu_dais(rtd, i, dai) { + if (dai->playback_widget) { + w = dai->playback_widget; + bufprintf("w%pK [label=\"%s\"];\n", w, w->name); + if (!rtd->dai_link->no_pcm) + bufprintf("w%pK -> w%pK;\n", rtd, w); + wdone[num_wdone] = w; + if (num_wdone < ARRAY_SIZE(wdone)) + num_wdone++; + } + + if (dai->capture_widget) { + w = dai->capture_widget; + bufprintf("w%pK [label=\"%s\"];\n", w, w->name); + if (!rtd->dai_link->no_pcm) + bufprintf("w%pK -> w%pK;\n", w, rtd); + wdone[num_wdone] = w; + if (num_wdone < ARRAY_SIZE(wdone)) + num_wdone++; + } + } + } + + for_each_card_dapms(card, dapm) { + const char *prefix = soc_dapm_prefix(dapm); + + if (dapm != &card->dapm) { + bufprintf("subgraph cluster_%d {\n", cluster++); + if (prefix && dapm->component) + bufprintf("label=\"%s (%s)\";\n", prefix, + dapm->component->name); + else if (dapm->component) + bufprintf("label=\"%s\";\n", + dapm->component->name); + } + + for_each_card_widgets(dapm->card, w) { + const char *name = w->name; + bool skip = false; + + if (w->dapm != dapm) + continue; + + if (list_empty(&w->edges[0]) && list_empty(&w->edges[1])) + continue; + + for (i = 0; i < num_wdone; i++) + if (wdone[i] == w) + skip = true; + if (skip) + continue; + + if (prefix && strlen(name) > strlen(prefix) + 1) + name += strlen(prefix) + 1; + + bufprintf("w%pK [label=\"%s\"];\n", w, name); + } + + if (dapm != &card->dapm) + bufprintf("}\n"); + } + + list_for_each_entry(p, &card->paths, list) { + if (p->name) + bufprintf("w%pK -> w%pK [label=\"%s\"];\n", + p->source, p->sink, p->name); + else + bufprintf("w%pK -> w%pK;\n", p->source, p->sink); + } + + bufprintf("}\n"); +#undef bufprintf + + mutex_unlock(&card->dapm_mutex); + + ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret); + + kfree(buf); + return ret; +} + +static const struct file_operations dapm_graph_fops = { + .open = simple_open, + .read = dapm_graph_read_file, + .llseek = default_llseek, +}; + void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm, struct dentry *parent) { @@ -2220,6 +2357,10 @@ void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm, debugfs_create_file("bias_level", 0444, dapm->debugfs_dapm, dapm, &dapm_bias_fops); + + if (dapm == &dapm->card->dapm) + debugfs_create_file("graph.dot", 0444, dapm->debugfs_dapm, + dapm->card, &dapm_graph_fops); } static void dapm_debugfs_add_widget(struct snd_soc_dapm_widget *w) -- 2.33.0

5 10

[PATCH] ASoC: max98088: add support for noise gate reg
by Tommaso Merciai 25 Aug '22

25 Aug '22

Add support for Noise Gate Threshold reg (ANTH reg 0x40 bit 4 to 7) References: - https://datasheets.maximintegrated.com/en/ds/MAX98089.pdf, p75 Signed-off-by: Tommaso Merciai <tommaso.merciai(a)amarulasolutions.com> --- sound/soc/codecs/max98088.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 5ef2e1279ee7..76ef71ab9c83 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -474,6 +474,9 @@ static const struct snd_kcontrol_new max98088_snd_controls[] = { max98088_mic2pre_get, max98088_mic2pre_set, max98088_micboost_tlv), + SOC_SINGLE("Noise Gate Threshold", M98088_REG_40_MICAGC_THRESH, + 4, 15, 0), + SOC_SINGLE("INA Volume", M98088_REG_37_LVL_INA, 0, 7, 1), SOC_SINGLE("INB Volume", M98088_REG_38_LVL_INB, 0, 7, 1), -- 2.25.1

2 1

[PATCH 0/3] ASoC: SOF: mediatek: update SOF driver for mt8186
by Chunxu Li 25 Aug '22

25 Aug '22

In these patches, we update SOF driver for mt8186 Chunxu Li (3): ASoC: SOF: mediatek: Add dai driver for mt8186 ASoC: SOF: mediatek: add snd_sof_dsp_ops callbacks for pcm and mail box ASoC: SOF: mediatek: Use generic implementation for .ipc_msg_data field sound/soc/sof/mediatek/mt8186/mt8186.c | 49 ++++++++++++++++++++++---- 1 file changed, 42 insertions(+), 7 deletions(-) -- 2.25.1

2 4

[PATCH] ASoC: codecs: add suspend and resume for ES8316
by Zhu Ning 25 Aug '22

25 Aug '22

The registers may be lost after suspend due to powerdown. regcache_sync solves this issue. Signed-off-by: Zhu Ning <zhuning(a)everest-semi.com> --- sound/soc/codecs/es8316.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index de7185f73e1e..8643014472ae 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -767,9 +767,31 @@ static void es8316_remove(struct snd_soc_component *component) clk_disable_unprepare(es8316->mclk); } +static int es8316_resume(struct snd_soc_component *component) +{ + struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(es8316->regmap, false); + regcache_sync(es8316->regmap); + + return 0; +} + +static int es8316_suspend(struct snd_soc_component *component) +{ + struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(es8316->regmap, true); + regcache_mark_dirty(es8316->regmap); + + return 0; +} + static const struct snd_soc_component_driver soc_component_dev_es8316 = { .probe = es8316_probe, .remove = es8316_remove, + .resume = es8316_resume, + .suspend = es8316_suspend, .set_jack = es8316_set_jack, .controls = es8316_snd_controls, .num_controls = ARRAY_SIZE(es8316_snd_controls), -- 2.36.1

2 1

[PATCH v2 1/6] sysfs: do not create empty directories if no attributes are present
by Greg Kroah-Hartman 25 Aug '22

25 Aug '22

When creating an attribute group, if it is named a subdirectory is created and the sysfs files are placed into that subdirectory. If no files are created, normally the directory would still be present, but it would be empty. Clean this up by removing the directory if no files were successfully created in the group at all. Cc: Vinod Koul <vkoul(a)kernel.org> Cc: Bard Liao <yung-chuan.liao(a)linux.intel.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Sanyog Kale <sanyog.r.kale(a)intel.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- v2: new patch Note, totally untested! The following soundwire patches will need this, if a soundwire developer could test this out, it would be most apreciated. fs/sysfs/group.c | 19 ++++++++++++++++--- 1 file changed, 16 insertions(+), 3 deletions(-) diff --git a/fs/sysfs/group.c b/fs/sysfs/group.c index eeb0e3099421..9fe0b47db47f 100644 --- a/fs/sysfs/group.c +++ b/fs/sysfs/group.c @@ -31,12 +31,14 @@ static void remove_files(struct kernfs_node *parent, kernfs_remove_by_name(parent, (*bin_attr)->attr.name); } +/* returns -ERROR if error, or >= 0 for number of files actually created */ static int create_files(struct kernfs_node *parent, struct kobject *kobj, kuid_t uid, kgid_t gid, const struct attribute_group *grp, int update) { struct attribute *const *attr; struct bin_attribute *const *bin_attr; + int files_created = 0; int error = 0, i; if (grp->attrs) { @@ -65,6 +67,8 @@ static int create_files(struct kernfs_node *parent, struct kobject *kobj, gid, NULL); if (unlikely(error)) break; + + files_created++; } if (error) { remove_files(parent, grp); @@ -95,12 +99,15 @@ static int create_files(struct kernfs_node *parent, struct kobject *kobj, NULL); if (error) break; + files_created++; } if (error) remove_files(parent, grp); } exit: - return error; + if (error) + return error; + return files_created; } @@ -146,10 +153,16 @@ static int internal_create_group(struct kobject *kobj, int update, kn = kobj->sd; kernfs_get(kn); error = create_files(kn, kobj, uid, gid, grp, update); - if (error) { + if (error <= 0) { + /* + * If an error happened _OR_ if no files were created in the + * attribute group, and we have a name for this group, delete + * the name so there's not an empty directory. + */ if (grp->name) kernfs_remove(kn); - } + } else + error = 0; kernfs_put(kn); if (grp->name && update) -- 2.37.2

2 8

[PATCH AUTOSEL 5.15 02/20] ASoC: sh: rz-ssi: Improve error handling in rz_ssi_probe() error path
by Sasha Levin 25 Aug '22

25 Aug '22

From: Biju Das <biju.das.jz(a)bp.renesas.com> [ Upstream commit c75ed9f54ce8d349fee557f2b471a4d637ed2a6b ] We usually do cleanup in reverse order of init. Currently in case of error rz_ssi_release_dma_channels() done in the reverse order. This patch improves error handling in rz_ssi_probe() error path. While at it, use "goto cleanup" style to reduce code duplication. Reported-by: Pavel Machek <pavel(a)denx.de> Signed-off-by: Biju Das <biju.das.jz(a)bp.renesas.com> Link: https://lore.kernel.org/r/20220728092612.38858-1-biju.das.jz@bp.renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sh/rz-ssi.c | 26 +++++++++++++++----------- 1 file changed, 15 insertions(+), 11 deletions(-) diff --git a/sound/soc/sh/rz-ssi.c b/sound/soc/sh/rz-ssi.c index 6d794eaaf4c3..2e33a1fa0a6f 100644 --- a/sound/soc/sh/rz-ssi.c +++ b/sound/soc/sh/rz-ssi.c @@ -1022,32 +1022,36 @@ static int rz_ssi_probe(struct platform_device *pdev) ssi->rstc = devm_reset_control_get_exclusive(&pdev->dev, NULL); if (IS_ERR(ssi->rstc)) { - rz_ssi_release_dma_channels(ssi); - return PTR_ERR(ssi->rstc); + ret = PTR_ERR(ssi->rstc); + goto err_reset; } reset_control_deassert(ssi->rstc); pm_runtime_enable(&pdev->dev); ret = pm_runtime_resume_and_get(&pdev->dev); if (ret < 0) { - rz_ssi_release_dma_channels(ssi); - pm_runtime_disable(ssi->dev); - reset_control_assert(ssi->rstc); - return dev_err_probe(ssi->dev, ret, "pm_runtime_resume_and_get failed\n"); + dev_err(&pdev->dev, "pm_runtime_resume_and_get failed\n"); + goto err_pm; } ret = devm_snd_soc_register_component(&pdev->dev, &rz_ssi_soc_component, rz_ssi_soc_dai, ARRAY_SIZE(rz_ssi_soc_dai)); if (ret < 0) { - rz_ssi_release_dma_channels(ssi); - - pm_runtime_put(ssi->dev); - pm_runtime_disable(ssi->dev); - reset_control_assert(ssi->rstc); dev_err(&pdev->dev, "failed to register snd component\n"); + goto err_snd_soc; } + return 0; + +err_snd_soc: + pm_runtime_put(ssi->dev); +err_pm: + pm_runtime_disable(ssi->dev); + reset_control_assert(ssi->rstc); +err_reset: + rz_ssi_release_dma_channels(ssi); + return ret; } -- 2.35.1

1 0

[PATCH AUTOSEL 5.19 34/38] ALSA: hda/cs8409: Support new Dolphin Variants
by Sasha Levin 25 Aug '22

25 Aug '22

From: Stefan Binding <sbinding(a)opensource.cirrus.com> [ Upstream commit 1ff954f9ab054675b9eb02dd14add8f7aa376d71 ] Add 4 new Dolphin Systems, same configuration as older systems. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220816151901.1398007-1-sbinding@opensource.cirr… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_cs8409-tables.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_cs8409-tables.c b/sound/pci/hda/patch_cs8409-tables.c index 4f4cc8215917..5b140301ca66 100644 --- a/sound/pci/hda/patch_cs8409-tables.c +++ b/sound/pci/hda/patch_cs8409-tables.c @@ -546,6 +546,10 @@ const struct snd_pci_quirk cs8409_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0BD6, "Dolphin", CS8409_DOLPHIN), SND_PCI_QUIRK(0x1028, 0x0BD7, "Dolphin", CS8409_DOLPHIN), SND_PCI_QUIRK(0x1028, 0x0BD8, "Dolphin", CS8409_DOLPHIN), + SND_PCI_QUIRK(0x1028, 0x0C43, "Dolphin", CS8409_DOLPHIN), + SND_PCI_QUIRK(0x1028, 0x0C50, "Dolphin", CS8409_DOLPHIN), + SND_PCI_QUIRK(0x1028, 0x0C51, "Dolphin", CS8409_DOLPHIN), + SND_PCI_QUIRK(0x1028, 0x0C52, "Dolphin", CS8409_DOLPHIN), {} /* terminator */ }; -- 2.35.1

1 0

[PATCH AUTOSEL 5.19 18/38] ALSA: hda/realtek: Add quirks for ASUS Zenbooks using CS35L41
by Sasha Levin 25 Aug '22

25 Aug '22

From: Stefan Binding <sbinding(a)opensource.cirrus.com> [ Upstream commit 461122b999bda2ebef2086a35d8990f9ccac5ab8 ] These Asus Zenbook laptop use Realtek HDA codec combined with 2xCS35L41 Amplifiers using SPI. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220815141953.25197-1-sbinding@opensource.cirrus… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 619e6025ba97..7aaa021f550d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9246,6 +9246,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1271, "ASUS X430UN", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1290, "ASUS X441SA", ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x12a0, "ASUS X441UV", ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1043, 0x12af, "ASUS UX582ZS", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x12e0, "ASUS X541SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x12f0, "ASUS X541UV", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1313, "Asus K42JZ", ALC269VB_FIXUP_ASUS_MIC_NO_PRESENCE), @@ -9266,6 +9267,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x19e1, "ASUS UX581LV", ALC295_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x1043, 0x1a30, "ASUS X705UD", ALC256_FIXUP_ASUS_MIC), + SND_PCI_QUIRK(0x1043, 0x1a8f, "ASUS UX582ZS", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1b11, "ASUS UX431DA", ALC294_FIXUP_ASUS_COEF_1B), SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), -- 2.35.1

1 0

[PATCH AUTOSEL 5.19 03/38] ASoC: rt5640: Fix the JD voltage dropping issue
by Sasha Levin 25 Aug '22

25 Aug '22

From: Oder Chiou <oder_chiou(a)realtek.com> [ Upstream commit afb176d45870048eea540991b082208270824037 ] The patch fixes the JD voltage dropping issue in the HDA JD using. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Reported-by: Mohan Kumar D <mkumard(a)nvidia.com> Link: https://lore.kernel.org/r/20220808052836.25791-1-oder_chiou@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5640.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 18b3da9211e3..5ada0d318d0f 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1986,7 +1986,7 @@ static int rt5640_set_bias_level(struct snd_soc_component *component, snd_soc_component_write(component, RT5640_PWR_MIXER, 0x0000); if (rt5640->jd_src == RT5640_JD_SRC_HDA_HEADER) snd_soc_component_write(component, RT5640_PWR_ANLG1, - 0x0018); + 0x2818); else snd_soc_component_write(component, RT5640_PWR_ANLG1, 0x0000); @@ -2592,7 +2592,8 @@ static void rt5640_enable_hda_jack_detect( snd_soc_component_update_bits(component, RT5640_DUMMY1, 0x400, 0x0); snd_soc_component_update_bits(component, RT5640_PWR_ANLG1, - RT5640_PWR_VREF2, RT5640_PWR_VREF2); + RT5640_PWR_VREF2 | RT5640_PWR_MB | RT5640_PWR_BG, + RT5640_PWR_VREF2 | RT5640_PWR_MB | RT5640_PWR_BG); usleep_range(10000, 15000); snd_soc_component_update_bits(component, RT5640_PWR_ANLG1, RT5640_PWR_FV2, RT5640_PWR_FV2); -- 2.35.1

1 0

[PATCH AUTOSEL 5.19 02/38] ASoC: sh: rz-ssi: Improve error handling in rz_ssi_probe() error path
by Sasha Levin 25 Aug '22

25 Aug '22

From: Biju Das <biju.das.jz(a)bp.renesas.com> [ Upstream commit c75ed9f54ce8d349fee557f2b471a4d637ed2a6b ] We usually do cleanup in reverse order of init. Currently in case of error rz_ssi_release_dma_channels() done in the reverse order. This patch improves error handling in rz_ssi_probe() error path. While at it, use "goto cleanup" style to reduce code duplication. Reported-by: Pavel Machek <pavel(a)denx.de> Signed-off-by: Biju Das <biju.das.jz(a)bp.renesas.com> Link: https://lore.kernel.org/r/20220728092612.38858-1-biju.das.jz@bp.renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sh/rz-ssi.c | 26 +++++++++++++++----------- 1 file changed, 15 insertions(+), 11 deletions(-) diff --git a/sound/soc/sh/rz-ssi.c b/sound/soc/sh/rz-ssi.c index e392de7a262e..3d74acffec11 100644 --- a/sound/soc/sh/rz-ssi.c +++ b/sound/soc/sh/rz-ssi.c @@ -1016,32 +1016,36 @@ static int rz_ssi_probe(struct platform_device *pdev) ssi->rstc = devm_reset_control_get_exclusive(&pdev->dev, NULL); if (IS_ERR(ssi->rstc)) { - rz_ssi_release_dma_channels(ssi); - return PTR_ERR(ssi->rstc); + ret = PTR_ERR(ssi->rstc); + goto err_reset; } reset_control_deassert(ssi->rstc); pm_runtime_enable(&pdev->dev); ret = pm_runtime_resume_and_get(&pdev->dev); if (ret < 0) { - rz_ssi_release_dma_channels(ssi); - pm_runtime_disable(ssi->dev); - reset_control_assert(ssi->rstc); - return dev_err_probe(ssi->dev, ret, "pm_runtime_resume_and_get failed\n"); + dev_err(&pdev->dev, "pm_runtime_resume_and_get failed\n"); + goto err_pm; } ret = devm_snd_soc_register_component(&pdev->dev, &rz_ssi_soc_component, rz_ssi_soc_dai, ARRAY_SIZE(rz_ssi_soc_dai)); if (ret < 0) { - rz_ssi_release_dma_channels(ssi); - - pm_runtime_put(ssi->dev); - pm_runtime_disable(ssi->dev); - reset_control_assert(ssi->rstc); dev_err(&pdev->dev, "failed to register snd component\n"); + goto err_snd_soc; } + return 0; + +err_snd_soc: + pm_runtime_put(ssi->dev); +err_pm: + pm_runtime_disable(ssi->dev); + reset_control_assert(ssi->rstc); +err_reset: + rz_ssi_release_dma_channels(ssi); + return ret; } -- 2.35.1

1 0

[PATCH] ARM: pxa: pxa2xx-ac97-lib: fix tests for platform_get_irq() failure
by Yu Zhe 25 Aug '22

25 Aug '22

The platform_get_irq() returns negative error codes. It can't actually return zero. Signed-off-by: Yu Zhe <yuzhe(a)nfschina.com> --- sound/arm/pxa2xx-ac97-lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index e55c0421718b..1f4c4749a6fb 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -402,7 +402,7 @@ int pxa2xx_ac97_hw_probe(struct platform_device *dev) goto err_clk2; irq = platform_get_irq(dev, 0); - if (!irq) + if (irq < 0) goto err_irq; ret = request_irq(irq, pxa2xx_ac97_irq, 0, "AC97", NULL); -- 2.11.0

1 0

[PATCH 0/2] dt-bindings: sound: Add SOF field for mt8186
by Chunxu Li 24 Aug '22

24 Aug '22

From: "chunxu.li" <chunxu.li(a)mediatek.com> These patches add SOF field for mt8186 machine driver chunxu.li (2): ASoC: mediatek: dt-bindings: modify machine bindings for SOF ASoC: mediatek: dt-bindings: modify machine bindings for SOF .../bindings/sound/mt8186-mt6366-da7219-max98357.yaml | 10 ++++++++++ .../bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml | 10 ++++++++++ 2 files changed, 20 insertions(+) -- 2.25.1

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Have I got a missing ucm files problem ?
by GitHub issues - opened 24 Aug '22

24 Aug '22

alsa-project/alsa-ucm-conf issue #204 was opened from xevilstar: Hello, this is my alsa-info http://alsa-project.org/db/?f=cc1f9ff2eb1e223ebf6024aa1605db1d737ff5fa I am having problems with callaudiod not finding the headset and mic on my audio cards and one of the members on the callaudiod git suggested me I am missing some ucm files .... This is a raspberrypi4 can you please help me ? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/204 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 1/5] soundwire: sysfs: move sdw_slave_dev_attr_group into the existing list of groups
by Greg Kroah-Hartman 24 Aug '22

24 Aug '22

The sysfs logic already creates a list of groups for the device, so add the sdw_slave_dev_attr_group group to that list instead of having to do a two-step process of adding a group list and then an individual group. This is a step on the way to moving all of the sysfs attribute handling into the default driver core attribute group logic so that the soundwire core does not have to do any of it manually. Cc: Vinod Koul <vkoul(a)kernel.org> Cc: Bard Liao <yung-chuan.liao(a)linux.intel.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Sanyog Kale <sanyog.r.kale(a)intel.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- drivers/soundwire/sysfs_slave.c | 15 ++++++++++----- 1 file changed, 10 insertions(+), 5 deletions(-) diff --git a/drivers/soundwire/sysfs_slave.c b/drivers/soundwire/sysfs_slave.c index 3210359cd944..83e3f6cc3250 100644 --- a/drivers/soundwire/sysfs_slave.c +++ b/drivers/soundwire/sysfs_slave.c @@ -105,7 +105,10 @@ static struct attribute *slave_attrs[] = { &dev_attr_modalias.attr, NULL, }; -ATTRIBUTE_GROUPS(slave); + +static const struct attribute_group slave_attr_group = { + .attrs = slave_attrs, +}; static struct attribute *slave_dev_attrs[] = { &dev_attr_mipi_revision.attr, @@ -190,6 +193,12 @@ static const struct attribute_group dp0_group = { .name = "dp0", }; +static const struct attribute_group *slave_groups[] = { + &slave_attr_group, + &sdw_slave_dev_attr_group, + NULL, +}; + int sdw_slave_sysfs_init(struct sdw_slave *slave) { int ret; @@ -198,10 +207,6 @@ int sdw_slave_sysfs_init(struct sdw_slave *slave) if (ret < 0) return ret; - ret = devm_device_add_group(&slave->dev, &sdw_slave_dev_attr_group); - if (ret < 0) - return ret; - if (slave->prop.dp0_prop) { ret = devm_device_add_group(&slave->dev, &dp0_group); if (ret < 0) -- 2.37.1

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[PATCH -next 1/2] ASoC: SOF: imx8ulp: change es8326_regmap_config to static
by Yang Yingliang 24 Aug '22

24 Aug '22

sof_imx8ulp_ops is only used in imx8ulp.c now, change it to static. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/sof/imx/imx8ulp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/imx/imx8ulp.c b/sound/soc/sof/imx/imx8ulp.c index 02b496165acc..afab7feab5fb 100644 --- a/sound/soc/sof/imx/imx8ulp.c +++ b/sound/soc/sof/imx/imx8ulp.c @@ -413,7 +413,7 @@ static int imx8ulp_dsp_set_power_state(struct snd_sof_dev *sdev, } /* i.MX8 ops */ -struct snd_sof_dsp_ops sof_imx8ulp_ops = { +static struct snd_sof_dsp_ops sof_imx8ulp_ops = { /* probe and remove */ .probe = imx8ulp_probe, .remove = imx8ulp_remove, -- 2.25.1

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