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[PATCH -next 1/2] ALSA: mts64: Switch to use list_for_each_entry() helper
by Yang Yingliang 17 Sep '22

17 Sep '22

Use list_for_each_entry() helper instead of list_for_each() and list_entry() to simplify code a bit. No functional change. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/drivers/mts64.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) diff --git a/sound/drivers/mts64.c b/sound/drivers/mts64.c index d3bc9e8c407d..746ac25a319e 100644 --- a/sound/drivers/mts64.c +++ b/sound/drivers/mts64.c @@ -752,7 +752,6 @@ static int snd_mts64_rawmidi_create(struct snd_card *card) struct mts64 *mts = card->private_data; struct snd_rawmidi *rmidi; struct snd_rawmidi_substream *substream; - struct list_head *list; int err; err = snd_rawmidi_new(card, CARD_NAME, 0, @@ -778,16 +777,16 @@ static int snd_mts64_rawmidi_create(struct snd_card *card) /* name substreams */ /* output */ - list_for_each(list, - &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substreams) { - substream = list_entry(list, struct snd_rawmidi_substream, list); + list_for_each_entry(substream, + &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substreams, + list) { sprintf(substream->name, "Miditerminal %d", substream->number+1); } /* input */ - list_for_each(list, - &rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substreams) { - substream = list_entry(list, struct snd_rawmidi_substream, list); + list_for_each_entry(substream, + &rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substreams, + list) { mts->midi_input_substream[substream->number] = substream; switch(substream->number) { case MTS64_SMPTE_SUBSTREAM: -- 2.25.1

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[PATCH -next] ALSA: ppc: Switch to use for_each_child_of_node() macro
by Yang Yingliang 17 Sep '22

17 Sep '22

Use for_each_child_of_node() macro instead of open coding it. No functional change. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/ppc/tumbler.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/ppc/tumbler.c b/sound/ppc/tumbler.c index c65e74d7cd0a..f3f8ad7c3df8 100644 --- a/sound/ppc/tumbler.c +++ b/sound/ppc/tumbler.c @@ -1060,8 +1060,7 @@ static struct device_node *find_audio_device(const char *name) if (! gpiop) return NULL; - for (np = of_get_next_child(gpiop, NULL); np; - np = of_get_next_child(gpiop, np)) { + for_each_child_of_node(gpiop, np) { const char *property = of_get_property(np, "audio-gpio", NULL); if (property && strcmp(property, name) == 0) break; @@ -1080,8 +1079,7 @@ static struct device_node *find_compatible_audio_device(const char *name) if (!gpiop) return NULL; - for (np = of_get_next_child(gpiop, NULL); np; - np = of_get_next_child(gpiop, np)) { + for_each_child_of_node(gpiop, np) { if (of_device_is_compatible(np, name)) break; } -- 2.25.1

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[bug report] ASoC: rockchip: i2s: use regmap_read_poll_timeout to poll I2S_CLR
by Dan Carpenter 16 Sep '22

16 Sep '22

Hello Judy Hsiao, The patch fbb0ec656ee5: "ASoC: rockchip: i2s: use regmap_read_poll_timeout to poll I2S_CLR" from Sep 14, 2022, leads to the following Smatch static checker warning: sound/soc/rockchip/rockchip_i2s.c:165 rockchip_snd_txctrl() warn: sleeping in atomic context sound/soc/rockchip/rockchip_i2s.c 126 static int rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) 127 { 128 unsigned int val = 0; 129 int ret = 0; 130 131 spin_lock(&i2s->lock); ^^^^^^^^^^^^^^^^^^^^^ Holding a spin lock. 132 if (on) { 133 ret = regmap_update_bits(i2s->regmap, I2S_DMACR, 134 I2S_DMACR_TDE_ENABLE, 135 I2S_DMACR_TDE_ENABLE); 136 if (ret < 0) 137 goto end; 138 ret = regmap_update_bits(i2s->regmap, I2S_XFER, 139 I2S_XFER_TXS_START | I2S_XFER_RXS_START, 140 I2S_XFER_TXS_START | I2S_XFER_RXS_START); 141 if (ret < 0) 142 goto end; 143 i2s->tx_start = true; 144 } else { 145 i2s->tx_start = false; 146 147 ret = regmap_update_bits(i2s->regmap, I2S_DMACR, 148 I2S_DMACR_TDE_ENABLE, 149 I2S_DMACR_TDE_DISABLE); 150 if (ret < 0) 151 goto end; 152 153 if (!i2s->rx_start) { 154 ret = regmap_update_bits(i2s->regmap, I2S_XFER, 155 I2S_XFER_TXS_START | I2S_XFER_RXS_START, 156 I2S_XFER_TXS_STOP | I2S_XFER_RXS_STOP); 157 if (ret < 0) 158 goto end; 159 udelay(150); 160 ret = regmap_update_bits(i2s->regmap, I2S_CLR, 161 I2S_CLR_TXC | I2S_CLR_RXC, 162 I2S_CLR_TXC | I2S_CLR_RXC); 163 if (ret < 0) 164 goto end; --> 165 ret = regmap_read_poll_timeout(i2s->regmap, 166 I2S_CLR, 167 val, 168 val != 0, 169 20, ^^ 170 200); ^^^ If the sleep_us or timeout_us values are non-zero then it can sleep so it's a sleeping in atomic bug. 171 if (ret < 0) 172 dev_warn(i2s->dev, "fail to clear: %d\n", ret); 173 } 174 } 175 end: 176 spin_unlock(&i2s->lock); 177 if (ret < 0) 178 dev_err(i2s->dev, "lrclk update failed\n"); 179 180 return ret; 181 } regards, dan carpenter

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[PATCH] soundwire: qcom: update status from device id 1
by Srinivas Kandagatla 16 Sep '22

16 Sep '22

By default autoenumeration is enabled on QCom SoundWire controller which means the core should not be dealing with device 0 w.r.t enumeration. Currently device 0 status is also shared with SoundWire core which confuses the core sometimes and we endup adding 0:0:0:0 slave device. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- drivers/soundwire/qcom.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index e21a3306bf01..871e4d8b32c7 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -428,7 +428,7 @@ 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 = 1; 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) { @@ -448,7 +448,7 @@ 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 = 1; i <= SDW_MAX_DEVICES; i++) { u32 s; s = (val >> (i * 2)); -- 2.21.0

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[PATCH] ASoC: fsl_spdif: add ALSA event on dpll locked
by robert.rosengren＠axis.com 16 Sep '22

16 Sep '22

From: Robert Rosengren <robert.rosengren(a)axis.com> Add an ALSA event on the RX Sample Rate controller upon the dpll locked interrupt, making it possible for audio applications to monitor changes in the hardware. Signed-off-by: Robert Rosengren <robert.rosengren(a)axis.com> --- sound/soc/fsl/fsl_spdif.c | 18 ++++++++++++++++-- 1 file changed, 16 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 7fc1c96929bb..1679a21ffdca 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -44,6 +44,8 @@ static u8 srpc_dpll_locked[] = { 0x0, 0x1, 0x2, 0x3, 0x4, 0xa, 0xb }; #define DEFAULT_RXCLK_SRC 1 +#define RX_SAMPLE_RATE_KCONTROL "RX Sample Rate" + /** * struct fsl_spdif_soc_data: soc specific data * @@ -122,6 +124,7 @@ struct fsl_spdif_priv { const struct fsl_spdif_soc_data *soc; struct spdif_mixer_control fsl_spdif_control; struct snd_soc_dai_driver cpu_dai_drv; + struct snd_soc_dai *dai; struct platform_device *pdev; struct regmap *regmap; bool dpll_locked; @@ -223,9 +226,19 @@ static void spdif_irq_dpll_lock(struct fsl_spdif_priv *spdif_priv) locked &= SRPC_DPLL_LOCKED; dev_dbg(&pdev->dev, "isr: Rx dpll %s \n", - locked ? "locked" : "loss lock"); + locked ? "locked" : "loss lock"); spdif_priv->dpll_locked = locked ? true : false; + + if (spdif_priv->dai) { + struct snd_soc_component *component = spdif_priv->dai->component; + struct snd_kcontrol *kctl = snd_soc_card_get_kcontrol(component->card, + RX_SAMPLE_RATE_KCONTROL); + + if (kctl) + snd_ctl_notify(component->card->snd_card, + SNDRV_CTL_EVENT_MASK_VALUE, &kctl->id); + } } /* Receiver found illegal symbol interrupt handler */ @@ -1197,7 +1210,7 @@ static struct snd_kcontrol_new fsl_spdif_ctrls[] = { /* DPLL lock info get controller */ { .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = "RX Sample Rate", + .name = RX_SAMPLE_RATE_KCONTROL, .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, .info = fsl_spdif_rxrate_info, @@ -1241,6 +1254,7 @@ static struct snd_kcontrol_new fsl_spdif_ctrls_rcm[] = { static int fsl_spdif_dai_probe(struct snd_soc_dai *dai) { struct fsl_spdif_priv *spdif_private = snd_soc_dai_get_drvdata(dai); + spdif_private->dai = dai; snd_soc_dai_init_dma_data(dai, &spdif_private->dma_params_tx, &spdif_private->dma_params_rx); -- 2.30.2

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[PATCH v4 0/4] ASoC: qcom: add support for sc8280xp machine
by Srinivas Kandagatla 15 Sep '22

15 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 v3: - fixed few spellings in commit log - removed build dependency with APR 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 | 12 ++ 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, 387 insertions(+), 146 deletions(-) create mode 100644 sound/soc/qcom/sc8280xp.c -- 2.21.0

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[PATCH] ALSA: hda/realtek: Re-arrange quirk table entries
by Takashi Iwai 15 Sep '22

15 Sep '22

A few entries have been mistakenly inserted in wrong positions without considering the SSID ordering. Place them at right positions. Fixes: b7557267c233 ("ALSA: hda/realtek: Add quirk for ASUS GA402") Fixes: 94db9cc8f8fa ("ALSA: hda/realtek: Add quirk for ASUS GU603") Fixes: 739d0959fbed ("ALSA: hda: Add quirk for ASUS Flow x13") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d309304cdc46..90fde53e6956 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9341,10 +9341,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x13b0, "ASUS Z550SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), + SND_PCI_QUIRK(0x1043, 0x1662, "ASUS GV301QH", ALC294_FIXUP_ASUS_DUAL_SPK), + SND_PCI_QUIRK(0x1043, 0x16b2, "ASUS GU603", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x1740, "ASUS UX430UA", ALC295_FIXUP_ASUS_DACS), 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, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x18f1, "Asus FX505DT", ALC256_FIXUP_ASUS_HEADSET_MIC), @@ -9361,13 +9362,12 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC), + SND_PCI_QUIRK(0x1043, 0x1d42, "ASUS Zephyrus G14 2022", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1d4e, "ASUS TM420", ALC256_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502), SND_PCI_QUIRK(0x1043, 0x1e51, "ASUS Zephyrus M15", ALC294_FIXUP_ASUS_GU502_PINS), SND_PCI_QUIRK(0x1043, 0x1e8e, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401), - SND_PCI_QUIRK(0x1043, 0x1d42, "ASUS Zephyrus G14 2022", ALC289_FIXUP_ASUS_GA401), - SND_PCI_QUIRK(0x1043, 0x16b2, "ASUS GU603", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), -- 2.35.3

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[PATCH 1/1] ALSA: hda/realtek: Enable 4-speaker output Dell Precision 5530 laptop
by Callum Osmotherly 15 Sep '22

15 Sep '22

Callum Osmotherly <callum.osmotherly(a)gmail.com> Just as with the 5570 (and the other Dell laptops), this enables the two subwoofer speakers on the Dell Precision 5530 together with the main ones, significantly increasing the audio quality. I've tested this myself on a 5530 and can confirm it's working as expected. Signed-off-by: Callum Osmotherly <callum.osmotherly(a)gmail.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 799f6bf266dd..bdef329adb10 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9114,6 +9114,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0871, "Dell Precision 3630", ALC255_FIXUP_DELL_HEADSET_MIC), SND_PCI_QUIRK(0x1028, 0x0872, "Dell Precision 3630", ALC255_FIXUP_DELL_HEADSET_MIC), SND_PCI_QUIRK(0x1028, 0x0873, "Dell Precision 3930", ALC255_FIXUP_DUMMY_LINEOUT_VERB), + SND_PCI_QUIRK(0x1028, 0x087d, "Dell Precision 5530", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x08ad, "Dell WYSE AIO", ALC225_FIXUP_DELL_WYSE_AIO_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x08ae, "Dell WYSE NB", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0935, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB), -- 2.37.3

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[bug report] ASoC: SOF: compress: Add copy function for capture case
by Dan Carpenter 15 Sep '22

15 Sep '22

Hello Laurentiu Mihalcea, The patch 1a01e1927802: "ASoC: SOF: compress: Add copy function for capture case" from Aug 22, 2022, leads to the following Smatch static checker warning: sound/soc/sof/compress.c:315 sof_compr_copy_playback() warn: special assignment for copy_ function '+=' sound/soc/sof/compress.c 300 static int sof_compr_copy_playback(struct snd_compr_runtime *rtd, 301 char __user *buf, size_t count) 302 { 303 void *ptr; 304 unsigned int offset, n; 305 int ret; 306 307 div_u64_rem(rtd->total_bytes_available, rtd->buffer_size, &offset); 308 ptr = rtd->dma_area + offset; 309 n = rtd->buffer_size - offset; 310 311 if (count < n) { 312 ret = copy_from_user(ptr, buf, count); 313 } else { 314 ret = copy_from_user(ptr, buf, n); --> 315 ret += copy_from_user(rtd->dma_area, buf + n, count - n); This API is not ideal. The static checker warning is based on the theory that if you combine the number of bytes copied from the first copy with the number of bytes from the second copy, is that useful? "We copied 100 bytes. But not necessarily consecutive bytes, so you'll have to guess which they were!" Generally if we're copying from the user and the destination buffer is too small we return the number of bytes which were copied. But if the user memory is invalid we return -EFAULT. Here the user memory is invalid but we are returning a truncated number of bytes, so it suggests that the kernel buffer is too small which is kind of misleading. I guess user space kind of gets what they deserve for passing invalid memory... Also it looks like we can't really return -EFAULT because _soc_component_ret() is not able to accept that... 316 } 317 318 return count - ret; 319 } regards, dan carpenter

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[PATCH 00/10] Support for CS42L83 on Apple machines
by Martin Povišer 15 Sep '22

15 Sep '22

Hi all, there's a CS42L83 headphone jack codec found in Apple computers (in the recent 'Apple Silicon' ones as well as in earlier models, one example [1]). The part isn't publicly documented, but it appears almost identical to CS42L42, for which we have a driver in kernel. This series adapts the CS42L42 driver to the new part, and makes one change in anticipation of a machine driver for the Apple computers. Patch 1 adds new compatible to the cs42l42 schema. Patches 2 to 7 are taken from Richard's recent series [2] adding soundwire support to cs42l42. They are useful refactorings to build on in later patches, and also this way our work doesn't diverge. I made one fix: I added a call of common_remove at the end of i2c_probe should the cs42l42_init call fail (both before and after the split to cs42l42-i2c.c). Also s/Soundwire/SoundWire/ in the changelogs. Patch 8 exports some regmap-related symbols from cs42l42.c so they can be used to create cs42l83 regmap in cs42l83-i2c.c later. Patch 9 is the cs42l83 support proper. Patch 10 implements 'set_bclk_ratio' on the cs42l42 core. This will be called by the upcoming ASoC machine driver for 'Apple Silicon' Macs. (We have touched on this change to be made in earlier discussion, see [3] and replies.) Best, Martin [1] https://www.ifixit.com/Teardown/MacBook+Pro+13-Inch+Touch+Bar+2018+Teardown… [2] https://lore.kernel.org/alsa-devel/20220819125230.42731-1-rf@opensource.cir… [3] https://lore.kernel.org/asahi/8961DDD2-93FF-4A18-BCA2-90FCE298F517@cutebit.… Martin Povišer (5): ASoC: dt-bindings: cs42l42: Add 'cs42l83' compatible ASoC: cs42l42: Split probe() and remove() into stages ASoC: cs42l42: Export regmap elements to the core namespace ASoC: cs42l83: Extend CS42L42 support to new part ASoC: cs42l42: Implement 'set_bclk_ratio' Richard Fitzgerald (5): ASoC: cs42l42: Add bitclock frequency argument to cs42l42_pll_config() ASoC: cs42l42: Use cs42l42->dev instead of &i2c_client->dev ASoC: cs42l42: Split cs42l42_resume into two functions ASoC: cs42l42: Pass component and dai defs into common probe ASoC: cs42l42: Split I2C identity into separate module .../bindings/sound/cirrus,cs42l42.yaml | 1 + MAINTAINERS | 1 + include/sound/cs42l42.h | 1 + sound/soc/codecs/Kconfig | 15 +- sound/soc/codecs/Makefile | 6 +- sound/soc/codecs/cs42l42-i2c.c | 112 ++++++++ sound/soc/codecs/cs42l42.c | 256 +++++++++--------- sound/soc/codecs/cs42l42.h | 24 +- sound/soc/codecs/cs42l83-i2c.c | 248 +++++++++++++++++ 9 files changed, 538 insertions(+), 126 deletions(-) create mode 100644 sound/soc/codecs/cs42l42-i2c.c create mode 100644 sound/soc/codecs/cs42l83-i2c.c -- 2.33.0

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