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snd_pcm_hw_open open '/dev/snd/pcmC0D0p' failed (-16)
by GitHub issues - opened 21 Oct '24

21 Oct '24

alsa-project/alsa-lib issue #412 was opened from SVENsuzhou: Hi, I am using pjsip to make a sip call function. But I got the following error in alsa-lib. ``` Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.315 pjsua_call.c !Answering call 0: code=200 Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.315 inv0x7f989c003fc8 ..SDP negotiation done: Success Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.316 pjsua_media.c ...Call 0: updating media.. Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.316 pjsua_media.c .....Media stream call00:0 is destroyed Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.316 udp0x7f989c010710 ....UDP media transport started Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.316 pjsua_aud.c ....Audio channel update.. Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.316 udp0x7f989c010710 .....UDP media transport attached Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.317 strm0x5559ce2eb538 .....Encoder stream started Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.317 strm0x5559ce2eb538 .....Decoder stream started Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.317 pjsua_media.c ....Audio updated, stream #0: PCMA (sendrecv) Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.317 pjsua_aud.c ...Conf connect: 1 --> 0 Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.317 pjsua_aud.c ....Set sound device: capture=-1, playback=-2, mode=0, use_default_settings=0 Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.317 pjsua_aud.c .....Opening sound device (speaker + mic) PCM@16000/1/20ms Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.318 alsa_dev.c ......ALSA lib /usr/src/debug/alsa-lib/1.2.11/src/pcm/pcm_hw.c:1785:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-16): Device o Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.318 pjsua_aud.c .....Opening sound device (speaker + mic) PCM@44100/1/20ms Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: start media transmit failed: pjsua_conf_connect(id, sink.id) error: Unknown error from audio driver (PJMEDIA_EAUD_SYSERR) (status=420002) [/usr/src/debug/pjproject/2.14.1/pjsip/src/pjsua2/media.cpp:218] Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.318 alsa_dev.c ......ALSA lib /usr/src/debug/alsa-lib/1.2.11/src/pcm/pcm_hw.c:1785:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-16): Device o Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.318 pjsua_aud.c .....Opening sound device (speaker + mic) PCM@48000/1/20ms Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.319 alsa_dev.c ......ALSA lib /usr/src/debug/alsa-lib/1.2.11/src/pcm/pcm_hw.c:1785:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-16): Device o Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.319 pjsua_aud.c .....Opening sound device (speaker + mic) PCM@32000/1/20ms Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.319 alsa_dev.c ......ALSA lib /usr/src/debug/alsa-lib/1.2.11/src/pcm/pcm_hw.c:1785:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-16): Device o Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.319 pjsua_aud.c .....Opening sound device (speaker + mic) PCM@16000/1/20ms Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.319 alsa_dev.c ......ALSA lib /usr/src/debug/alsa-lib/1.2.11/src/pcm/pcm_hw.c:1785:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-16): Device o Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.319 pjsua_aud.c .....Opening sound device (speaker + mic) PCM@8000/1/20ms Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.320 alsa_dev.c ......ALSA lib /usr/src/debug/alsa-lib/1.2.11/src/pcm/pcm_hw.c:1785:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-16): Device o Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.320 pjsua_aud.c .....Unable to open sound device: Unknown error from audio driver (PJMEDIA_EAUD_SYSERR) [status=420002] Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.320 pjsua_aud.c ....Error opening sound device: Unknown error from audio driver (PJMEDIA_EAUD_SYSERR) [status=420002] Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.320 media.cpp ...pjsua_conf_connect(id, sink.id) error: Unknown error from audio driver (PJMEDIA_EAUD_SYSERR) (status=420002) [/usr/src/debug/pjproject/2.14.1/pjsip/src/pjsua2/media.cpp:218] Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.321 pjsua_core.c ....TX 966 bytes Response msg 200/INVITE/cseq=30946 (tdta0x5559ce2e7378) to UDP 10.57.0.200:5060: Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: SIP/2.0 200 OK Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: Via: SIP/2.0/UDP 10.57.0.200:5060;rport=5060;received=10.57.0.200;branch=z9hG4bKPje5a12777-db72-4db0-a5a5-a72743589a27 Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: Call-ID: 0bc91889-099c-42de-beb2-4fedefac6e7f Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: answer incoming call, remote uri: <sip:passenger_pei@10.57.0.200>, state: CONNECTING Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: From: <sip:passenger_pei@10.57.0.200>;tag=9efc50b5-c57a-4aa7-823e-2c3df064a27c ``` I try to use `lsof /dev/snd/*` to check which process hold the file handle. And I got nothing. Can you give me some leads on troubleshooting? Issue URL : https://github.com/alsa-project/alsa-lib/issues/412 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v6] MAINTAINERS: update entries in TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) DRIVERS and add entries for haptic driver
by Shenghao Ding 20 Oct '24

20 Oct '24

Due to internal re-org, Kevin has no longer maintained audio driver. Due to job responsibility change, drop entries for the audio converter and add entries for both haptics drivers and middle-power audio amplifier drivers. Add audio converter, set the Status as Supported. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v6: - fixed spelling mistakes v5: - | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202410021557.FByBO9Dp-lkp@intel.com/ v4: - Add Touch Screen support v3: - Add Audio converter section - Set the section of LOW/MIDDLE-POWER AUDIO AMPLIFIER as Supported v2: - Add the detailed information of the maintained drivers. --- MAINTAINERS | 32 ++++++++++++++++++++++++-------- 1 file changed, 24 insertions(+), 8 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 87198c5e589e..d1484b5658e3 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -22910,25 +22910,42 @@ S: Maintained F: Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml F: sound/soc/ti/ -TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS +TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) & HAPTICS DRIVERS M: Shenghao Ding <shenghao-ding(a)ti.com> -M: Kevin Lu <kevin-lu(a)ti.com> M: Baojun Xu <baojun.xu(a)ti.com> L: linux-sound(a)vger.kernel.org -S: Maintained +S: Supported +F: Documentation/devicetree/bindings/input/ti,drv260x.yaml +F: Documentation/devicetree/bindings/input/ti,drv266x.yaml F: Documentation/devicetree/bindings/sound/tas2552.txt +F: Documentation/devicetree/bindings/sound/tas5720.txt F: Documentation/devicetree/bindings/sound/ti,tas2562.yaml F: Documentation/devicetree/bindings/sound/ti,tas2770.yaml +F: Documentation/devicetree/bindings/sound/ti,tas2781.yaml F: Documentation/devicetree/bindings/sound/ti,tas27xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5086.txt +F: Documentation/devicetree/bindings/sound/ti,tas57xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5805m.yaml F: Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml +F: drivers/input/misc/drv2*.c +F: include/sound/tas2*.h +F: include/sound/tas5086.h +F: include/sound/tpa6130a2-plat.h +F: sound/pci/hda/tas2781_hda_i2c.c +F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/tas5*.* +F: sound/soc/codecs/tpa6130a2.* + +TEXAS INSTRUMENTS AUDIO Converter (ASoC) and Touch Screen DRIVERS +L: linux-sound(a)vger.kernel.org +S: Maintained F: Documentation/devicetree/bindings/sound/ti,pcm1681.yaml F: Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml +F: Documentation/devicetree/bindings/sound/ti,pcm6240.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320*.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320adcx140.yaml -F: include/sound/tas2*.h +F: drivers/input/touchscreen/tsc2*.* F: include/sound/tlv320*.h -F: include/sound/tpa6130a2-plat.h -F: sound/pci/hda/tas2781_hda_i2c.c F: sound/soc/codecs/pcm1681.c F: sound/soc/codecs/pcm1789*.* F: sound/soc/codecs/pcm179x*.* @@ -22938,9 +22955,8 @@ F: sound/soc/codecs/pcm3060*.* F: sound/soc/codecs/pcm3168a*.* F: sound/soc/codecs/pcm5102a.c F: sound/soc/codecs/pcm512x*.* -F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/pcm6240.* F: sound/soc/codecs/tlv320*.* -F: sound/soc/codecs/tpa6130a2.* TEXAS INSTRUMENTS DMA DRIVERS M: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> -- 2.34.1

1 0

[PATCH v2] ASoC: codecs: lpass-rx-macro: fix RXn(rx,n) macro for DSM_CTL and SEC7 regs
by Alexey Klimov 18 Oct '24

18 Oct '24

Turns out some registers of pre-2.5 version of rxmacro codecs are not located at the expected offsets but 0xc further away in memory. So far the detected registers are CDC_RX_RX2_RX_PATH_SEC7 and CDC_RX_RX2_RX_PATH_DSM_CTL. CDC_RX_RXn_RX_PATH_DSM_CTL(rx, n) macro incorrectly generates the address 0x540 for RX2 but it should be 0x54C and it also overwrites CDC_RX_RX2_RX_PATH_SEC7 which is located at 0x540. The same goes for CDC_RX_RXn_RX_PATH_SEC7(rx, n). Fix this by introducing additional rxn_reg_stride2 offset. For 2.5 version and above this offset will be equal to 0. With such change the corresponding RXn() macros will generate the same values for 2.5 codec version for all RX paths and the same old values for pre-2.5 version for RX0 and RX1. However for the latter case with RX2 path it will also add rxn_reg_stride2 on top. While at this, also remove specific if-check for INTERP_AUX from rx_macro_digital_mute() and rx_macro_enable_interp_clk(). These if-check was used to handle such special offset for AUX interpolator but since CDC_RX_RXn_RX_PATH_SEC7(rx, n) and CDC_RX_RXn_RX_PATH_DSM_CTL(rx, n) macros will generate the correst addresses of dsm register, they are no longer needed. Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Cc: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> Signed-off-by: Alexey Klimov <alexey.klimov(a)linaro.org> --- Changes in v2: - updated macros as suggested by Mark and Dmitry; - removed INTERP_AUX if-check in two functions; Link to previous patch: https://lore.kernel.org/lkml/20240925043823.520218-1-alexey.klimov@linaro.o… sound/soc/codecs/lpass-rx-macro.c | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c index ef7a70fa6966..febbbe073962 100644 --- a/sound/soc/codecs/lpass-rx-macro.c +++ b/sound/soc/codecs/lpass-rx-macro.c @@ -202,12 +202,14 @@ #define CDC_RX_RXn_RX_PATH_SEC3(rx, n) (0x042c + rx->rxn_reg_stride * n) #define CDC_RX_RX0_RX_PATH_SEC4 (0x0430) #define CDC_RX_RX0_RX_PATH_SEC7 (0x0434) -#define CDC_RX_RXn_RX_PATH_SEC7(rx, n) (0x0434 + rx->rxn_reg_stride * n) +#define CDC_RX_RXn_RX_PATH_SEC7(rx, n) \ + (0x0434 + (rx->rxn_reg_stride * n) + ((n > 1) ? rx->rxn_reg_stride2 : 0)) #define CDC_RX_DSM_OUT_DELAY_SEL_MASK GENMASK(2, 0) #define CDC_RX_DSM_OUT_DELAY_TWO_SAMPLE 0x2 #define CDC_RX_RX0_RX_PATH_MIX_SEC0 (0x0438) #define CDC_RX_RX0_RX_PATH_MIX_SEC1 (0x043C) -#define CDC_RX_RXn_RX_PATH_DSM_CTL(rx, n) (0x0440 + rx->rxn_reg_stride * n) +#define CDC_RX_RXn_RX_PATH_DSM_CTL(rx, n) \ + (0x0440 + (rx->rxn_reg_stride * n) + ((n > 1) ? rx->rxn_reg_stride2 : 0)) #define CDC_RX_RXn_DSM_CLK_EN_MASK BIT(0) #define CDC_RX_RX0_RX_PATH_DSM_CTL (0x0440) #define CDC_RX_RX0_RX_PATH_DSM_DATA1 (0x0444) @@ -645,6 +647,7 @@ struct rx_macro { int rx_mclk_cnt; enum lpass_codec_version codec_version; int rxn_reg_stride; + int rxn_reg_stride2; bool is_ear_mode_on; bool hph_pwr_mode; bool hph_hd2_mode; @@ -1929,9 +1932,6 @@ static int rx_macro_digital_mute(struct snd_soc_dai *dai, int mute, int stream) CDC_RX_PATH_PGA_MUTE_MASK, 0x0); } - if (j == INTERP_AUX) - dsm_reg = CDC_RX_RXn_RX_PATH_DSM_CTL(rx, 2); - int_mux_cfg0 = CDC_RX_INP_MUX_RX_INT0_CFG0 + j * 8; int_mux_cfg1 = int_mux_cfg0 + 4; int_mux_cfg0_val = snd_soc_component_read(component, int_mux_cfg0); @@ -2702,9 +2702,6 @@ static int rx_macro_enable_interp_clk(struct snd_soc_component *component, main_reg = CDC_RX_RXn_RX_PATH_CTL(rx, interp_idx); dsm_reg = CDC_RX_RXn_RX_PATH_DSM_CTL(rx, interp_idx); - if (interp_idx == INTERP_AUX) - dsm_reg = CDC_RX_RXn_RX_PATH_DSM_CTL(rx, 2); - rx_cfg2_reg = CDC_RX_RXn_RX_PATH_CFG2(rx, interp_idx); if (SND_SOC_DAPM_EVENT_ON(event)) { @@ -3821,6 +3818,7 @@ static int rx_macro_probe(struct platform_device *pdev) case LPASS_CODEC_VERSION_2_0: case LPASS_CODEC_VERSION_2_1: rx->rxn_reg_stride = 0x80; + rx->rxn_reg_stride2 = 0xc; def_count = ARRAY_SIZE(rx_defaults) + ARRAY_SIZE(rx_pre_2_5_defaults); reg_defaults = kmalloc_array(def_count, sizeof(struct reg_default), GFP_KERNEL); if (!reg_defaults) @@ -3834,6 +3832,7 @@ static int rx_macro_probe(struct platform_device *pdev) case LPASS_CODEC_VERSION_2_7: case LPASS_CODEC_VERSION_2_8: rx->rxn_reg_stride = 0xc0; + rx->rxn_reg_stride2 = 0x0; def_count = ARRAY_SIZE(rx_defaults) + ARRAY_SIZE(rx_2_5_defaults); reg_defaults = kmalloc_array(def_count, sizeof(struct reg_default), GFP_KERNEL); if (!reg_defaults) -- 2.45.2

1 0

[PATCH] ASoC: fsl_micfil: Add a flag to distinguish with different volume control types
by Chancel Liu 18 Oct '24

18 Oct '24

On i.MX8MM the register of volume control has positive and negative values. It is different from other platforms like i.MX8MP and i.MX93 which only have positive values. Add a volume_sx flag to use SX_TLV volume control for this kind of platform. Use common TLV volume control for other platforms. Fixes: cdfa92eb90f5 ("ASoC: fsl_micfil: Correct the number of steps on SX controls") Signed-off-by: Chancel Liu <chancel.liu(a)nxp.com> --- sound/soc/fsl/fsl_micfil.c | 43 +++++++++++++++++++++++++++++++++++++- 1 file changed, 42 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index 4161b674745b..0679d762f08a 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -78,6 +78,7 @@ struct fsl_micfil_soc_data { bool imx; bool use_edma; bool use_verid; + bool volume_sx; u64 formats; }; @@ -87,6 +88,7 @@ static struct fsl_micfil_soc_data fsl_micfil_imx8mm = { .fifo_depth = 8, .dataline = 0xf, .formats = SNDRV_PCM_FMTBIT_S16_LE, + .volume_sx = true, }; static struct fsl_micfil_soc_data fsl_micfil_imx8mp = { @@ -95,6 +97,7 @@ static struct fsl_micfil_soc_data fsl_micfil_imx8mp = { .fifo_depth = 32, .dataline = 0xf, .formats = SNDRV_PCM_FMTBIT_S32_LE, + .volume_sx = false, }; static struct fsl_micfil_soc_data fsl_micfil_imx93 = { @@ -105,6 +108,7 @@ static struct fsl_micfil_soc_data fsl_micfil_imx93 = { .formats = SNDRV_PCM_FMTBIT_S32_LE, .use_edma = true, .use_verid = true, + .volume_sx = false, }; static const struct of_device_id fsl_micfil_dt_ids[] = { @@ -328,7 +332,26 @@ static int hwvad_detected(struct snd_kcontrol *kcontrol, return 0; } -static const struct snd_kcontrol_new fsl_micfil_snd_controls[] = { +static const struct snd_kcontrol_new fsl_micfil_volume_controls[] = { + SOC_SINGLE_TLV("CH0 Volume", REG_MICFIL_OUT_CTRL, + MICFIL_OUTGAIN_CHX_SHIFT(0), 0xF, 0, gain_tlv), + SOC_SINGLE_TLV("CH1 Volume", REG_MICFIL_OUT_CTRL, + MICFIL_OUTGAIN_CHX_SHIFT(1), 0xF, 0, gain_tlv), + SOC_SINGLE_TLV("CH2 Volume", REG_MICFIL_OUT_CTRL, + MICFIL_OUTGAIN_CHX_SHIFT(2), 0xF, 0, gain_tlv), + SOC_SINGLE_TLV("CH3 Volume", REG_MICFIL_OUT_CTRL, + MICFIL_OUTGAIN_CHX_SHIFT(3), 0xF, 0, gain_tlv), + SOC_SINGLE_TLV("CH4 Volume", REG_MICFIL_OUT_CTRL, + MICFIL_OUTGAIN_CHX_SHIFT(4), 0xF, 0, gain_tlv), + SOC_SINGLE_TLV("CH5 Volume", REG_MICFIL_OUT_CTRL, + MICFIL_OUTGAIN_CHX_SHIFT(5), 0xF, 0, gain_tlv), + SOC_SINGLE_TLV("CH6 Volume", REG_MICFIL_OUT_CTRL, + MICFIL_OUTGAIN_CHX_SHIFT(6), 0xF, 0, gain_tlv), + SOC_SINGLE_TLV("CH7 Volume", REG_MICFIL_OUT_CTRL, + MICFIL_OUTGAIN_CHX_SHIFT(7), 0xF, 0, gain_tlv), +}; + +static const struct snd_kcontrol_new fsl_micfil_volume_sx_controls[] = { SOC_SINGLE_SX_TLV("CH0 Volume", REG_MICFIL_OUT_CTRL, MICFIL_OUTGAIN_CHX_SHIFT(0), 0x8, 0xF, gain_tlv), SOC_SINGLE_SX_TLV("CH1 Volume", REG_MICFIL_OUT_CTRL, @@ -345,6 +368,9 @@ static const struct snd_kcontrol_new fsl_micfil_snd_controls[] = { MICFIL_OUTGAIN_CHX_SHIFT(6), 0x8, 0xF, gain_tlv), SOC_SINGLE_SX_TLV("CH7 Volume", REG_MICFIL_OUT_CTRL, MICFIL_OUTGAIN_CHX_SHIFT(7), 0x8, 0xF, gain_tlv), +}; + +static const struct snd_kcontrol_new fsl_micfil_snd_controls[] = { SOC_ENUM_EXT("MICFIL Quality Select", fsl_micfil_quality_enum, micfil_quality_get, micfil_quality_set), @@ -844,6 +870,20 @@ static int fsl_micfil_dai_probe(struct snd_soc_dai *cpu_dai) return 0; } +static int fsl_micfil_component_probe(struct snd_soc_component *component) +{ + struct fsl_micfil *micfil = snd_soc_component_get_drvdata(component); + + if (micfil->soc->volume_sx) + snd_soc_add_component_controls(component, fsl_micfil_volume_sx_controls, + ARRAY_SIZE(fsl_micfil_volume_sx_controls)); + else + snd_soc_add_component_controls(component, fsl_micfil_volume_controls, + ARRAY_SIZE(fsl_micfil_volume_controls)); + + return 0; +} + static const struct snd_soc_dai_ops fsl_micfil_dai_ops = { .probe = fsl_micfil_dai_probe, .startup = fsl_micfil_startup, @@ -865,6 +905,7 @@ static struct snd_soc_dai_driver fsl_micfil_dai = { static const struct snd_soc_component_driver fsl_micfil_component = { .name = "fsl-micfil-dai", + .probe = fsl_micfil_component_probe, .controls = fsl_micfil_snd_controls, .num_controls = ARRAY_SIZE(fsl_micfil_snd_controls), .legacy_dai_naming = 1, -- 2.46.2

3 2

[PATCH] Use card->dev in replace of the &pdev->dev argument in the dev_err function
by Liu Jing 18 Oct '24

18 Oct '24

Because card->dev = &pdev->dev is already defined in the rx51_soc_probe function, and then &pdev->dev is still used. Signed-off-by: Liu Jing <liujing(a)cmss.chinamobile.com> --- sound/soc/ti/rx51.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/ti/rx51.c b/sound/soc/ti/rx51.c index 77296237575a..d9900c69e536 100644 --- a/sound/soc/ti/rx51.c +++ b/sound/soc/ti/rx51.c @@ -371,7 +371,7 @@ static int rx51_soc_probe(struct platform_device *pdev) dai_node = of_parse_phandle(np, "nokia,cpu-dai", 0); if (!dai_node) { - dev_err(&pdev->dev, "McBSP node is not provided\n"); + dev_err(card->dev, "McBSP node is not provided\n"); return -EINVAL; } rx51_dai[0].cpus->dai_name = NULL; @@ -381,7 +381,7 @@ static int rx51_soc_probe(struct platform_device *pdev) dai_node = of_parse_phandle(np, "nokia,audio-codec", 0); if (!dai_node) { - dev_err(&pdev->dev, "Codec node is not provided\n"); + dev_err(card->dev, "Codec node is not provided\n"); return -EINVAL; } rx51_dai[0].codecs->name = NULL; @@ -389,7 +389,7 @@ static int rx51_soc_probe(struct platform_device *pdev) dai_node = of_parse_phandle(np, "nokia,audio-codec", 1); if (!dai_node) { - dev_err(&pdev->dev, "Auxiliary Codec node is not provided\n"); + dev_err(card->dev, "Auxiliary Codec node is not provided\n"); return -EINVAL; } rx51_aux_dev[0].dlc.name = NULL; @@ -399,7 +399,7 @@ static int rx51_soc_probe(struct platform_device *pdev) dai_node = of_parse_phandle(np, "nokia,headphone-amplifier", 0); if (!dai_node) { - dev_err(&pdev->dev, "Headphone amplifier node is not provided\n"); + dev_err(card->dev, "Headphone amplifier node is not provided\n"); return -EINVAL; } rx51_aux_dev[1].dlc.name = NULL; @@ -408,7 +408,7 @@ static int rx51_soc_probe(struct platform_device *pdev) rx51_codec_conf[1].dlc.of_node = dai_node; } - pdata = devm_kzalloc(&pdev->dev, sizeof(*pdata), GFP_KERNEL); + pdata = devm_kzalloc(card->dev, sizeof(*pdata), GFP_KERNEL); if (pdata == NULL) return -ENOMEM; @@ -439,7 +439,7 @@ static int rx51_soc_probe(struct platform_device *pdev) err = devm_snd_soc_register_card(card->dev, card); if (err) { - dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", err); + dev_err(card->dev, "snd_soc_register_card failed (%d)\n", err); return err; } -- 2.27.0

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[PATCH 5.10 0/1] soundwire: cadence: Fix error check in cdns_xfer_msg()
by Daniil Dulov 18 Oct '24

18 Oct '24

Svacer reports redundant comparison in cdns_xfer_msg(). The problem is present in 5.10 stable release and can be fixed by the following upstream patch that can be cleanly applied to 5.10 stable branch.

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[PATCH alsa-ucm-conf v5 1/2] sof-soundwire: Change map control names to make them unique and user friendly
by Maciej Strozek 18 Oct '24

18 Oct '24

Both 'Speaker Digital' and 'cs42l43 PDM2' names were found in other alsa controls, confusing the MixerElems paths. Make them unique to avoid errors for hardware mutes. Also add 'Capture' string into microphone controls to make them work properly. Signed-off-by: Maciej Strozek <mstrozek(a)opensource.cirrus.com> --- v5: Change PlaybackMixerElem to just "Speaker" v4: Add "CaptureMixer" to headset config v3: Make Headset Microphone more user-friendly and contain the word "Capture" v2: Rename to "Speaker Playback" instead of "Speaker Combined" --- ucm2/codecs/cs42l43/init.conf | 10 ++++++++-- ucm2/sof-soundwire/cs35l56-bridge.conf | 6 +++--- ucm2/sof-soundwire/cs42l43-dmic.conf | 2 +- ucm2/sof-soundwire/cs42l43.conf | 3 ++- 4 files changed, 14 insertions(+), 7 deletions(-) diff --git a/ucm2/codecs/cs42l43/init.conf b/ucm2/codecs/cs42l43/init.conf index 10a6e1f..b2796e2 100644 --- a/ucm2/codecs/cs42l43/init.conf +++ b/ucm2/codecs/cs42l43/init.conf @@ -2,13 +2,19 @@ LibraryConfig.remap.Config { ctl.default.map { - "name='cs42l43 PDM2 Switch'" { + "name='cs42l43 Microphone Capture Switch'" { "name='cs42l43 Decimator 3 Switch'".vindex.0 0 "name='cs42l43 Decimator 4 Switch'".vindex.1 0 } - "name='cs42l43 PDM2 Volume'" { + "name='cs42l43 Microphone Capture Volume'" { "name='cs42l43 Decimator 3 Volume'".vindex.0 0 "name='cs42l43 Decimator 4 Volume'".vindex.1 0 } + "name='cs42l43 Headset Microphone Capture Switch'" { + "name='cs42l43 Decimator 1 Switch'".vindex.0 0 + } + "name='cs42l43 Headset Microphone Capture Volume'" { + "name='cs42l43 Decimator 1 Volume'".vindex.0 0 + } } } diff --git a/ucm2/sof-soundwire/cs35l56-bridge.conf b/ucm2/sof-soundwire/cs35l56-bridge.conf index 5435b11..176f880 100644 --- a/ucm2/sof-soundwire/cs35l56-bridge.conf +++ b/ucm2/sof-soundwire/cs35l56-bridge.conf @@ -2,11 +2,11 @@ LibraryConfig.remap.Config { ctl.default.map { - "name='Speaker Digital Switch'" { + "name='Speaker Playback Switch'" { "name='AMPL Speaker Switch'".vindex.0 0 "name='AMPR Speaker Switch'".vindex.1 0 } - "name='Speaker Digital Volume'" { + "name='Speaker Playback Volume'" { "name='AMPL Speaker Volume'".vindex.0 0 "name='AMPR Speaker Volume'".vindex.1 0 } @@ -56,6 +56,6 @@ SectionDevice."Speaker" { PlaybackPriority 100 PlaybackPCM "hw:${CardId},0" PlaybackMixer "default:${CardId}" - PlaybackMixerElem "Speaker Digital" + PlaybackMixerElem "Speaker" } } diff --git a/ucm2/sof-soundwire/cs42l43-dmic.conf b/ucm2/sof-soundwire/cs42l43-dmic.conf index d5f895d..b68a70d 100644 --- a/ucm2/sof-soundwire/cs42l43-dmic.conf +++ b/ucm2/sof-soundwire/cs42l43-dmic.conf @@ -21,6 +21,6 @@ SectionDevice."Mic" { CapturePriority 100 CapturePCM "hw:${CardId},4" CaptureMixer "default:${CardId}" - CaptureMixerElem "cs42l43 PDM2" + CaptureMixerElem "cs42l43 Microphone" } } diff --git a/ucm2/sof-soundwire/cs42l43.conf b/ucm2/sof-soundwire/cs42l43.conf index 1702cf0..ddca67a 100644 --- a/ucm2/sof-soundwire/cs42l43.conf +++ b/ucm2/sof-soundwire/cs42l43.conf @@ -40,7 +40,8 @@ SectionDevice."Headset" { Value { CapturePriority 200 CapturePCM "hw:${CardId},4" - CaptureMixerElem "cs42l43 Decimator 1" + CaptureMixer "default:${CardId}" + CaptureMixerElem "cs42l43 Headset Microphone" JackControl "Headset Mic Jack" } } -- 2.39.5

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Qualcomm: fix codec initialisation
by GitHub pull_request - opened 18 Oct '24

18 Oct '24

alsa-project/alsa-ucm-conf pull request #454 was opened from jhovold: Fix the Qualcomm configurations that are using the same identifier for lazy includes of codec initialisation files which means that only the last one is actually included. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/454 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/454.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Revert "ucm2: Qualcomm: x1e80100: add USB DisplayPort playback"
by GitHub pull_request - opened 18 Oct '24

18 Oct '24

alsa-project/alsa-ucm-conf pull request #453 was opened from jhovold: This reverts commit 1dcb58e4fac989bafe0705e49a7f029727207fb3. The recently added DisplayPort support depends on kernel changes that never made it upstream and has some usability issues that remain to be resolved. Revert for now to avoid breaking existing setups. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/453 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/453.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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ALSA error: snd_pcm_hw_params_set_format failed: Invalid argument.
by GitHub issues - opened 17 Oct '24

17 Oct '24

alsa-project/alsa-plugins issue #62 was opened from linuxandos2: Playing a flac with a samplerate higer than 96000 with audacious gives: > _ALSA error: snd_pcm_hw_params_set_format failed: Invalid argument._ Output device for audacious is hw:1.0 - USB Audio the device is a soudbaster blasterx G6. output of _cat /proc/asound/card1/stream0_ > Creative Technology Ltd Sound BlasterX G6 at usb-0000:00:14.0-4.1, high speed : > USB Audio > > Playback: > Status: Stop > Interface 1 > Altset 1 > Format: S24_3LE > Channels: 2 > Endpoint: 0x01 (1 OUT) (ASYNC) > Rates: 44100, 48000, 88200, 96000, 176400, 192000, 352800, 384000 > Data packet interval: 125 us > Bits: 24 > Channel map: FL FR > Sync Endpoint: 0x81 (1 IN) > Sync EP Interface: 1 > Sync EP Altset: 1 > Implicit Feedback Mode: No > Interface 1 > Altset 2 > Format: S32_LE > Channels: 2 > Endpoint: 0x01 (1 OUT) (ASYNC) > Rates: 44100, 48000, 88200, 96000, 176400, 192000, 352800, 384000 > Data packet interval: 125 us > Bits: 32 > Channel map: FL FR > Sync Endpoint: 0x81 (1 IN) > Sync EP Interface: 1 > Sync EP Altset: 2 > Implicit Feedback Mode: No > > Capture: > Status: Stop > Interface 2 > Altset 1 > Format: S24_3LE > Channels: 2 > Endpoint: 0x82 (2 IN) (ASYNC) > Rates: 44100, 48000, 88200, 96000, 176400, 192000, 352800, 384000 > Data packet interval: 125 us > Bits: 24 > Channel map: FL FR > Interface 2 > Altset 2 > Format: S32_LE > Channels: 2 > Endpoint: 0x82 (2 IN) (ASYNC) > Rates: 44100, 48000, 88200, 96000, 176400, 192000, 352800, 384000 > Data packet interval: 125 us > Bits: 32 > Channel map: FL FR > linux: gentoo kernel: 6.9.9-gentoo-x86_64 alsa-lib-1.2.12 alsa-topology-conf-1.2.5.1 alsa-ucm-conf-1.2.12 alsa-plugins-1.2.12 alsa-utils-1.2.12 selecting > pulse - PulseAudio Sound Server in audacious results in no sound and audacious hangs. Pipewire and wireplumber are used. On an old gentoo- system on another partition all samplerates play successfully. Using the same kernel: linux-6.8.1-gentoo as in the old system on the new one does not solve the problem. the versions of alsa are: alsa-lib-1.2.10-r2 alsa-topology-conf-1.2.5.1 alsa-ucm-conf-1.2.10-r1 alsa-plugins-1.2.7.1-r1 alsa-utils-1.2.10-r1 Issue URL : https://github.com/alsa-project/alsa-plugins/issues/62 Repository URL: https://github.com/alsa-project/alsa-plugins

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