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

[PATCH v2 0/3] sound: realtek: add quirks for ASUS G513, G533Z, GA503R
by Luke D. Jones 18 Sep '22

18 Sep '22

This series adds quirks to fix pincfg for the ASUS ROG G513 and G533Z laptops, and adds a mapping for the ASUS ROG GA503R to use the previous G15/G14 quirks which fix the DAC mapping. Changelog: V2: - Use smallcaps in hex string - Chain the G533Z quirk with the G513 quirk as they use the same base pincfg - Add the quirk for GA503R series to use GA401 quirk Luke D. Jones (3): sound: realtek: Add pincfg for ASUS G513 HP jack sound: realtek: Add pincfg for ASUS G533Z HP jack sound: realtek: Add quirk for ASUS GA503R laptop sound/pci/hda/patch_realtek.c | 25 ++++++++++++++++++++++++- 1 file changed, 24 insertions(+), 1 deletion(-) -- 2.37.3

3 6

USB-Audio: Motu M4: no duplex profile
by GitHub issues - opened 18 Sep '22

18 Sep '22

alsa-project/alsa-ucm-conf issue #214 was opened from pisquaredover6: I have compiled alsa-ucm-conf v1.2.7.2 with the patches [3a6acaa5090a0c5a0d29c8a4906b6a60378774e2](https://github.com/alsa-project/a… and [ad1618cdd33f5c78ee1a0a9d3db362ffcb9f2596](https://github.com/alsa-project/a…. With this, I have simplex audio, but no duplex profiles any more. Is there any chance we can add a duplex profile, as it used to be the case before alsa-ucm-conf introduced the M4? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/214 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: amd: yc: Add ASUS UM5302TA into DMI table
by Xiaoyan Li 17 Sep '22

17 Sep '22

ASUS Zenbook S 13 OLED (UM5302TA) needs this quirk to get the built-in microphone working properly. Link: https://bugzilla.kernel.org/show_bug.cgi?id=216270 Signed-off-by: Xiaoyan Li <lxy.lixiaoyan(a)gmail.com> --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index e0b24e1daef3..5eab3baf3573 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -171,6 +171,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "21J6"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "ASUSTeK COMPUTER INC."), + DMI_MATCH(DMI_PRODUCT_NAME, "UM5302TA"), + } + }, {} }; -- 2.37.3

2 1

[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

1 1

[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

2 1

[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

1 1

[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

2 5

[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

4 5

[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

2 6

[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

1 0