June 2021 - Alsa-devel - mailman.alsa-project.org

Current version not working with Focusrite Scarlett 18i20
by GitHub issues - opened 09 Jun '21

09 Jun '21

alsa-project/alsa-lib issue #150 was opened from TheBitFighter: Hello, I hope I am in the right repository for this. My arch system updated alsa-card-profiles and alsa-utils today which caused my interface, a Focusrite Scarlett 18i20 1st gen to stop working. The card was still showing up in pulse and showed output, however no sound was actually played. Input was still working fine. It sounded like the output stage of the interface had been disabled, and the problem could only be fixed by booting Windows and setting new routing settings in Scarlett MixControl. The interface had exhibited some resetting and popping behavior for a few versions now (when stopping playback, changing application and playing something else there), but continued to work (given a few instances where sample rate played mismatched sample rate sent which caused weird audio slowdowns). It only stopped working today. What I did initially was reinstall the older versions of these two packages, which did not fix the problem. I now downgraded alsa-libs (1.2.5-2 -> 1.2.4-2), alsa-card-profiles (1:0.3.29-2 -> 1:0.3.28-1), lib32-alsa-lib (1.2.5-2 -> 1.2.4-1), alsa-topology-conf (1.2.5-1 -> 1.2.4-1), and alsa-ucm-conf (1.2.5-1 -> 1.2.4-1) which fixed the problem. (alsa-utils is at 1.2.5-2) `cat /etc/asound/version` now reports `Advanced Linux Sound Architecture Driver Version k5.12.9-arch1-1.` This also fixed the popping/resetting issues I had before. Thanks for the help! If you need any further information I am happy to help. Issue URL : https://github.com/alsa-project/alsa-lib/issues/150 Repository URL: https://github.com/alsa-project/alsa-lib

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opensuse tumbleweed no sound after alsa lib update (tumbleweed 2010605)
by GitHub issues - opened 09 Jun '21

09 Jun '21

alsa-project/alsa-lib issue #149 was opened from pawanyadav: sound is not working alsa (never installed pipewire and pulseaudio) asound.conf defaults.pcm.card 1 defaults.pcm.device 1 aplay -l **** List of PLAYBACK Hardware Devices **** card 0: HDMI [HDA ATI HDMI], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Generic [HD-Audio Generic], device 0: ALC887-VD Analog [ALC887-VD Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Generic [HD-Audio Generic], device 1: ALC887-VD Digital [ALC887-VD Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 cat /proc/asound/cards 0 [HDMI ]: HDA-Intel - HDA ATI HDMI HDA ATI HDMI at 0xfcf40000 irq 67 1 [Generic ]: HDA-Intel - HD-Audio Generic HD-Audio Generic at 0xfce00000 irq 69 speaker-test 1.2.4 Playback device is default Stream parameters are 48000Hz, S16_LE, 1 channels Using 16 octaves of pink noise ALSA lib confmisc.c:855:(parse_card) cannot find card '1dmix:1' ALSA lib conf.c:5091:(_snd_config_evaluate) function snd_func_card_id returned error: No such device ALSA lib confmisc.c:422:(snd_func_concat) error evaluating strings ALSA lib conf.c:5091:(_snd_config_evaluate) function snd_func_concat returned error: No such device ALSA lib confmisc.c:1334:(snd_func_refer) error evaluating name ALSA lib conf.c:5091:(_snd_config_evaluate) function snd_func_refer returned error: No such device ALSA lib conf.c:5579:(snd_config_expand) Evaluate error: No such device ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM dmix:1dmix:1 Playback open error: -19,No such device Issue URL : https://github.com/alsa-project/alsa-lib/issues/149 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2] ASoC: qcom: Add four speaker support on MI2S secondary
by Srinivasa Rao Mandadapu 09 Jun '21

09 Jun '21

Add four speaker support on MI2S secondary block by using I2S SD1 line on gpio52 pin, and add channel map control support in the lpass-cpu audio driver. Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- Changes Since V1: -- removed set_channel_map/get_channel_map implementation as default kcontrols added in pcm_new API. sound/soc/qcom/lpass-cpu.c | 32 ++++++++++++++++++++++++++++++++ sound/soc/qcom/lpass-sc7180.c | 1 + sound/soc/qcom/lpass.h | 2 ++ 3 files changed, 35 insertions(+) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index c62d2612e8f5..15d69e311b38 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -29,6 +29,14 @@ #define LPASS_CPU_I2S_SD0_1_2_MASK GENMASK(2, 0) #define LPASS_CPU_I2S_SD0_1_2_3_MASK GENMASK(3, 0) +/* + * Channel maps for Quad channel playbacks on MI2S Secondary + */ +static struct snd_pcm_chmap_elem lpass_quad_chmaps = { + .channels = 4, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_RL, + SNDRV_CHMAP_FR, SNDRV_CHMAP_RR } +}; static int lpass_cpu_init_i2sctl_bitfields(struct device *dev, struct lpaif_i2sctl *i2sctl, struct regmap *map) { @@ -324,6 +332,25 @@ const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops = { }; EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_dai_ops); +int lpass_cpu_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai) +{ + int ret; + struct snd_soc_dai_driver *drv = dai->driver; + struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + + if (drvdata->mi2s_playback_sd_mode[dai->id] == LPAIF_I2SCTL_MODE_QUAD01) { + ret = snd_pcm_add_chmap_ctls(rtd->pcm, SNDRV_PCM_STREAM_PLAYBACK, + &lpass_quad_chmaps, drv->playback.channels_max, 0, + NULL); + if (ret < 0) + return ret; + } + + return 0; +} +EXPORT_SYMBOL_GPL(lpass_cpu_pcm_new); + int asoc_qcom_lpass_cpu_dai_probe(struct snd_soc_dai *dai) { struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); @@ -856,6 +883,11 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) PTR_ERR(drvdata->mi2s_bit_clk[dai_id])); return PTR_ERR(drvdata->mi2s_bit_clk[dai_id]); } + if (drvdata->mi2s_playback_sd_mode[dai_id] == + LPAIF_I2SCTL_MODE_QUAD01) { + variant->dai_driver[dai_id].playback.channels_min = 4; + variant->dai_driver[dai_id].playback.channels_max = 4; + } } /* Allocation for i2sctl regmap fields */ diff --git a/sound/soc/qcom/lpass-sc7180.c b/sound/soc/qcom/lpass-sc7180.c index 8c168d3c589e..77a556b27cf0 100644 --- a/sound/soc/qcom/lpass-sc7180.c +++ b/sound/soc/qcom/lpass-sc7180.c @@ -58,6 +58,7 @@ static struct snd_soc_dai_driver sc7180_lpass_cpu_dai_driver[] = { }, .probe = &asoc_qcom_lpass_cpu_dai_probe, .ops = &asoc_qcom_lpass_cpu_dai_ops, + .pcm_new = lpass_cpu_pcm_new, }, { .id = LPASS_DP_RX, .name = "Hdmi", diff --git a/sound/soc/qcom/lpass.h b/sound/soc/qcom/lpass.h index 83b2e08ade06..623ddccdafff 100644 --- a/sound/soc/qcom/lpass.h +++ b/sound/soc/qcom/lpass.h @@ -259,5 +259,7 @@ void asoc_qcom_lpass_cpu_platform_shutdown(struct platform_device *pdev); int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev); int asoc_qcom_lpass_cpu_dai_probe(struct snd_soc_dai *dai); extern const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops; +int lpass_cpu_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai); #endif /* __LPASS_H__ */ -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 2

[PATCH] usb-audio: Add support for Denon DN-X1600
by Damien Zammit 09 Jun '21

09 Jun '21

This patch applies clean to tiwai/sound.git repo on hash 82b954a96c1550bae2cb1f1f4f084425ab237cd7 and was tested on this version. This provides support for Denon DN-X1600 hardware mixer. The device itself supports 44100, 48000 and 96000 (Hz) sample rates, but switching rates via software is currently not working. Therefore, this patch hardcodes the sample rate to 48000Hz which enables all 8 channels to function correctly when the correct sample rate is selected on the hardware itself. MIDI also tested and works. Signed-off-by: Damien Zammit <damien(a)zamaudio.com> Tested-by: xalmoxis(a)gmail.com --- sound/usb/clock.c | 7 +++++ sound/usb/pcm.c | 4 +++ sound/usb/quirks-table.h | 70 ++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 81 insertions(+) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 0afae839d..52de52288 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -534,6 +534,13 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip, * rate. */ clock = snd_usb_clock_find_source(chip, fmt, false); + + /* Denon DN-X1600 hardcoded + * Sample rate seems to be set on the hardware itself + */ + if (chip->usb_id == USB_ID(0x154e, 0x500e)) + return 0; + if (clock < 0) return clock; } diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index c66831ee1..6913a5b02 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -197,6 +197,10 @@ int snd_usb_init_pitch(struct snd_usb_audio *chip, if (!(fmt->attributes & UAC_EP_CS_ATTR_PITCH_CONTROL)) return 0; + /* Denon DN-X1600 doesn't properly handle this, bail out too */ + if (chip->usb_id == USB_ID(0x154e, 0x500e)) + return 0; + usb_audio_dbg(chip, "enable PITCH for EP 0x%x\n", fmt->endpoint); switch (fmt->protocol) { diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index bdba37d0f..19bb499c1 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3044,6 +3044,76 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } }, +/* Denon DN-X1600 */ +{ + USB_AUDIO_DEVICE(0x154e, 0x500e), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Denon", + .product_name = "DN-X1600", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]){ + { + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE, + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0x0, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ADAPTIVE, + .maxpacksize = 0x138, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { + 48000 + } + } + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0x0, + .endpoint = 0x85, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ADAPTIVE, + .maxpacksize = 0x138, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { + 48000 + } + } + }, + { + .ifnum = 4, + .type = QUIRK_MIDI_STANDARD_INTERFACE, + }, + { + .ifnum = -1 + } + } + } +}, + /* Microsoft XboxLive Headset/Xbox Communicator */ { USB_DEVICE(0x045e, 0x0283), -- 2.13.1

2 1

[PATCH 0/3] ALSA: pcm:firewire: allow to operate for period elapse event in process context
by Takashi Sakamoto 09 Jun '21

09 Jun '21

Hi, This patchset comes from my former RFC: [RFC][PATCH 0/3] ALSA: pcm/firewire: allow to queue period elapse event in process context * https://lore.kernel.org/alsa-devel/20210606091838.80812-1-o-takashi@sakamoc… All of drivers in ALSA firewire stack have two chances to process isochronous packets of any isochronous context; in software IRQ context for 1394 OHCI, and in process context of ALSA PCM application. In the process context, callbacks of .pointer and .ack are utilized. The callbacks are done by ALSA PCM core under acquiring lock of PCM substream, In design of ALSA PCM core, call of snd_pcm_period_elapsed() is used for drivers to awaken user processes from waiting for available frames. The function voluntarily acquires lock of PCM substream, therefore it is not called in the process context since it causes dead lock. As a workaround to avoid the dead lock, all of drivers in ALSA firewire stack use workqueue to delegate the call. This patchset is my attempt for the issue. A variant of 'snd_pcm_period_elapsed()' without lock acquisition is going to be added, named 'snd_pcm_period_elapsed_under_stream_lock()'. The call is available in callbacks of .pointer and .ack of snd_pcm_ops structure. Changes from RFC: * dismiss inlining * rename function name * improve function comments I tested the patchset with ALSA OXFW driver and ftrace, covering axfer and jackd cases since pulseaudio and pipewire are programmed with timer-based scheduling model and ALSA runtime expects drivers not to call snd_pcm_period_elapsed(). The configuration of tracer and filters is: ``` $ sudo trace-cmd record -p function_graph \ -l :mod:snd_firewire_lib \ -l fw_iso_context_queue:mod:firewire_core \ -l snd_pcm_ioctl:mod:snd_pcm \ -l snd_pcm_period_elapsed*:mod:snd_pcm ``` The runtime of axfer without '-M' option can often call .pointer and .ack. Below is a sample about .pointer case via SNDRV_PCM_IOCT_HWSYNC. The new snd_pcm_period_elapsed_under_stream_lock() should be called. ``` 3929.769359: funcgraph_entry: | snd_pcm_ioctl() { 3929.769360: funcgraph_entry: | amdtp_domain_stream_pcm_pointer() { 3929.769361: funcgraph_entry: | irq_target_callback() { 3929.769361: funcgraph_entry: | process_rx_packets() { 3929.769362: funcgraph_entry: | process_ctx_payloads() { 3929.769363: funcgraph_entry: 0.391 us | process_it_ctx_payloads(); 3929.769363: funcgraph_entry: | snd_pcm_period_elapsed_under_stream_lock() { 3929.769364: funcgraph_entry: 0.411 us | amdtp_domain_stream_pcm_pointer(); 3929.769365: funcgraph_exit: 1.723 us | } 3929.769365: funcgraph_exit: 3.106 us | } 3929.769366: funcgraph_entry: | queue_packet() { 3929.769366: funcgraph_entry: 0.561 us | fw_iso_context_queue(); 3929.769367: funcgraph_exit: 1.122 us | } 3929.769367: funcgraph_exit: 5.731 us | } 3929.769367: funcgraph_entry: | process_ctxs_in_domain() { 3929.769368: funcgraph_entry: | process_tx_packets() { 3929.769369: funcgraph_entry: | process_ctx_payloads() { 3929.769369: funcgraph_entry: 0.321 us | process_ir_ctx_payloads(); 3929.769369: funcgraph_exit: 0.962 us | } 3929.769370: funcgraph_entry: 0.491 us | fw_iso_context_queue(); 3929.769371: funcgraph_exit: 2.364 us | } 3929.769371: funcgraph_exit: 3.427 us | } 3929.769371: funcgraph_exit: 10.038 us | } 3929.769372: funcgraph_exit: 11.271 us | } 3929.769372: funcgraph_exit: 13.606 us | } ``` The runtime of jackd heavily relies on period wakeup scheduled in invocation of interrupt handler. Below is a sample about software IRQ case of 1394 OHCI. The snd_pcm_period_elapsed() should be called. ``` 5318.980502: funcgraph_entry: | irq_target_callback() { 5318.980503: funcgraph_entry: | process_rx_packets() { 5318.980503: funcgraph_entry: | process_ctx_payloads() { 5318.980503: funcgraph_entry: 0.531 us | process_it_ctx_payloads(); 5318.980505: funcgraph_entry: | snd_pcm_period_elapsed() { 5318.980505: funcgraph_entry: | snd_pcm_period_elapsed_under_stream_lock() { 5318.980505: funcgraph_entry: 0.180 us | amdtp_domain_stream_pcm_pointer(); 5318.980508: funcgraph_exit: 3.106 us | } 5318.980508: funcgraph_exit: 3.497 us | } 5318.980508: funcgraph_exit: 4.990 us | } 5318.980508: funcgraph_entry: | queue_packet() { 5318.980509: funcgraph_entry: 0.350 us | fw_iso_context_queue(); 5318.980509: funcgraph_exit: 0.702 us | } 5318.980509: funcgraph_entry: | queue_packet() { 5318.980509: funcgraph_entry: 0.291 us | fw_iso_context_queue(); 5318.980510: funcgraph_exit: 0.651 us | } 5318.980510: funcgraph_entry: | queue_packet() { 5318.980510: funcgraph_entry: 0.280 us | fw_iso_context_queue(); 5318.980511: funcgraph_exit: 0.641 us | } 5318.980511: funcgraph_exit: 8.416 us | } 5318.980511: funcgraph_entry: | process_ctxs_in_domain() { 5318.980512: funcgraph_entry: | process_tx_packets() { 5318.980512: funcgraph_entry: | process_ctx_payloads() { 5318.980512: funcgraph_entry: 0.531 us | process_ir_ctx_payloads(); 5318.980513: funcgraph_entry: | snd_pcm_period_elapsed() { 5318.980513: funcgraph_entry: | snd_pcm_period_elapsed_under_stream_lock() { 5318.980513: funcgraph_entry: 0.170 us | amdtp_domain_stream_pcm_pointer(); 5318.980514: funcgraph_exit: 0.842 us | } 5318.980514: funcgraph_exit: 1.242 us | } 5318.980514: funcgraph_exit: 2.335 us | } 5318.980515: funcgraph_entry: 0.301 us | fw_iso_context_queue(); 5318.980515: funcgraph_entry: 0.291 us | fw_iso_context_queue(); 5318.980516: funcgraph_entry: 0.290 us | fw_iso_context_queue(); 5318.980516: funcgraph_exit: 4.198 us | } 5318.980516: funcgraph_exit: 5.119 us | } 5318.980516: funcgraph_exit: 14.077 us | } ``` ALSA OXFW driver works well to select appropriate kernel API by distinguishing running context. I also tested for error cases that the sequence of tx packets is invalid or packet queueing fails. In both cases, the error doesn't bring system corruption. Takashi Sakamoto (3): ALSA: pcm: add snd_pcm_period_elapsed() variant without acquiring lock of PCM substream ALSA: firewire-lib: operate for period elapse event in process context ALSA: firewire-lib: obsolete workqueue for period update include/sound/pcm.h | 1 + sound/core/pcm_lib.c | 68 +++++++++++++++++++++++++++-------- sound/firewire/amdtp-stream.c | 46 ++++++++---------------- sound/firewire/amdtp-stream.h | 1 - 4 files changed, 68 insertions(+), 48 deletions(-) -- 2.27.0

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[PATCH] Asoc: qcom: Add four speaker support on MI2S secondary
by Srinivasa Rao Mandadapu 09 Jun '21

09 Jun '21

Add four speaker support on MI2S secondary block by using I2S SD1 line on gpio52 pin, and add channel map control support in the lpass-cpu audio driver. Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- sound/soc/qcom/lpass-cpu.c | 106 +++++++++++++++++++++++++++++++++- sound/soc/qcom/lpass-sc7180.c | 1 + sound/soc/qcom/lpass.h | 5 ++ 3 files changed, 111 insertions(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index c62d2612e8f5..88a038a107a7 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -28,6 +28,19 @@ #define LPASS_CPU_I2S_SD2_3_MASK GENMASK(3, 2) #define LPASS_CPU_I2S_SD0_1_2_MASK GENMASK(2, 0) #define LPASS_CPU_I2S_SD0_1_2_3_MASK GENMASK(3, 0) +#define LPASS_CHMAP_IDX_UNKNOWN -1 + +/* + * Channel maps for multi-channel playbacks on MI2S Secondary, up to 4 n_ch + */ +static struct snd_pcm_chmap_elem lpass_chmaps[] = { + { .channels = 2, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR } }, + { .channels = 4, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } }, + { } +}; static int lpass_cpu_init_i2sctl_bitfields(struct device *dev, struct lpaif_i2sctl *i2sctl, struct regmap *map) @@ -86,6 +99,9 @@ static int lpass_cpu_daiops_startup(struct snd_pcm_substream *substream, clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]); return ret; } + if (drvdata->chmap_info && drvdata->chmap_info->max_channels == 4) + drvdata->chmap_info->chmap = lpass_chmaps; + return 0; } @@ -96,6 +112,7 @@ static void lpass_cpu_daiops_shutdown(struct snd_pcm_substream *substream, clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]); clk_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]); + drvdata->chmap_idx = LPASS_CHMAP_IDX_UNKNOWN; } static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, @@ -224,9 +241,14 @@ static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, ret); return ret; } - if (channels >= 2) + if (channels >= 2) { ret = regmap_fields_write(i2sctl->spkmono, id, LPAIF_I2SCTL_SPKMONO_STEREO); + if (channels == 4) + drvdata->chmap_idx = 1; + else + drvdata->chmap_idx = 0; + } else ret = regmap_fields_write(i2sctl->spkmono, id, LPAIF_I2SCTL_SPKMONO_MONO); @@ -324,6 +346,84 @@ const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops = { }; EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_dai_ops); +static int lpass_cpu_chmap_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + unsigned const char *map; + unsigned int i; + struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); + struct lpass_data *drvdata = info->private_data; + + if (info->max_channels == 2) + map = &lpass_chmaps[0].map[0]; + else if (info->max_channels == 4) + map = &lpass_chmaps[1].map[0]; + + for (i = 0; i < info->max_channels; i++) { + if (drvdata->chmap_idx == LPASS_CHMAP_IDX_UNKNOWN) + ucontrol->value.integer.value[i] = 0; + else + ucontrol->value.integer.value[i] = map[i]; + } + + return 0; +} + +static int lpass_cpu_chmap_ctl_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + unsigned int i; + struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); + struct lpass_data *drvdata = info->private_data; + unsigned char *map; + + if (info->max_channels == 2) + map = &lpass_chmaps[0].map[0]; + else if (info->max_channels == 4) + map = &lpass_chmaps[1].map[0]; + + for (i = 0; i < info->max_channels; i++) { + if (drvdata->chmap_idx == LPASS_CHMAP_IDX_UNKNOWN) + map[i] = 0; + else + map[i] = ucontrol->value.integer.value[i]; + } + + return 0; +} + +int lpass_cpu_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai) +{ + int ret; + + struct snd_soc_dai_driver *drv = dai->driver; + struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + + ret = snd_pcm_add_chmap_ctls(rtd->pcm, SNDRV_PCM_STREAM_PLAYBACK, + snd_pcm_alt_chmaps, drv->playback.channels_max, 0, + &drvdata->chmap_info); + if (ret < 0) + return ret; + + + /* + * override handlers + */ + drvdata->chmap_info->private_data = drvdata; + drvdata->chmap_info->kctl->get = lpass_cpu_chmap_ctl_get; + drvdata->chmap_info->kctl->put = lpass_cpu_chmap_ctl_set; + + /* + * default chmap supported is stereo + */ + drvdata->chmap_info->chmap = lpass_chmaps; + drvdata->chmap_idx = LPASS_CHMAP_IDX_UNKNOWN; + + return 0; +} +EXPORT_SYMBOL_GPL(lpass_cpu_pcm_new); + int asoc_qcom_lpass_cpu_dai_probe(struct snd_soc_dai *dai) { struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); @@ -856,6 +956,10 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) PTR_ERR(drvdata->mi2s_bit_clk[dai_id])); return PTR_ERR(drvdata->mi2s_bit_clk[dai_id]); } + if (drvdata->mi2s_playback_sd_mode[dai_id] == LPAIF_I2SCTL_MODE_QUAD01) { + variant->dai_driver[dai_id].playback.channels_min = 4; + variant->dai_driver[dai_id].playback.channels_max = 4; + } } /* Allocation for i2sctl regmap fields */ diff --git a/sound/soc/qcom/lpass-sc7180.c b/sound/soc/qcom/lpass-sc7180.c index 8c168d3c589e..77a556b27cf0 100644 --- a/sound/soc/qcom/lpass-sc7180.c +++ b/sound/soc/qcom/lpass-sc7180.c @@ -58,6 +58,7 @@ static struct snd_soc_dai_driver sc7180_lpass_cpu_dai_driver[] = { }, .probe = &asoc_qcom_lpass_cpu_dai_probe, .ops = &asoc_qcom_lpass_cpu_dai_ops, + .pcm_new = lpass_cpu_pcm_new, }, { .id = LPASS_DP_RX, .name = "Hdmi", diff --git a/sound/soc/qcom/lpass.h b/sound/soc/qcom/lpass.h index 83b2e08ade06..fb82591d550c 100644 --- a/sound/soc/qcom/lpass.h +++ b/sound/soc/qcom/lpass.h @@ -76,6 +76,9 @@ struct lpass_data { /* regmap backed by the low-power audio interface (LPAIF) registers */ struct regmap *lpaif_map; struct regmap *hdmiif_map; + struct snd_pcm_chmap *chmap_info; + + int chmap_idx; /* interrupts from the low-power audio interface (LPAIF) */ int lpaif_irq; @@ -259,5 +262,7 @@ void asoc_qcom_lpass_cpu_platform_shutdown(struct platform_device *pdev); int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev); int asoc_qcom_lpass_cpu_dai_probe(struct snd_soc_dai *dai); extern const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops; +int lpass_cpu_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai); #endif /* __LPASS_H__ */ -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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[PATCH 0/9] ASoC: tidyup snd_soc_of_parse_daifmt()
by Kuninori Morimoto 09 Jun '21

09 Jun '21

Hi Mark I want to add new audio-graph-card2 sound card driver, and this is last part of necessary soc-core cleanup for it. Current some drivers are using DT, and Then, snd_soc_of_parse_daifmt() parses daifmt, but bitclock/frame provider parsing part is one of headache, because we are assuming below both cases. A) node { bitclock-master; frame-master; ... }; B) link { bitclock-master = <&xxx>; frame-master = <&xxx>; ... }; The original was style A), and style B) was added later. snd_soc_of_parse_daifmt() parses A) style as original style, and user need to update to B) style for clock_provider part if needed. In such case, user need to re-parse it, like below. daifmt = snd_soc_of_parse_daifmt(..., &bitclkmaster, &framemaster); daifmt &= ~SND_SOC_DAIFMT_MASTER_MASK; if (codec == bitclkmaster) daifmt |= (codec == framemaster) ? SND_SOC_DAIFMT_CBM_CFM : SND_SOC_DAIFMT_CBM_CFS; else daifmt |= (codec == framemaster) ? SND_SOC_DAIFMT_CBS_CFM : SND_SOC_DAIFMT_CBS_CFS; This patch-set adds new functions, and handle these more simply. Unfortunately, there are too many use-case, do it by 1 function was implessible. style A) bit_frame = snd_soc_daifmt_parse_clock_provider(); daifmt = snd_soc_daifmt_parse_format(...) | /* format part */ snd_soc_daifmt_clock_provider_pickup(bit_frame); /* clock part */ style B) snd_soc_daifmt_parse_clock_provider(..., &bit, &frame); daifmt = snd_soc_daifmt_parse_format(...) | /* format part */ snd_soc_daifmt_clock_provider_pickup( /* clock part */ ((codec == bit) << 4) + (codec == frame)); Kuninori Morimoto (9): ASoC: soc-core: don't use discriminatory terms on snd_soc_runtime_get_dai_fmt() ASoC: soc-core: add snd_soc_daifmt_clock_provider_pickup() ASoC: soc-core: add snd_soc_daifmt_clock_provider_fliped() ASoC: soc-core: add snd_soc_daifmt_parse_format/clock_provider() ASoC: atmel: switch to use snd_soc_daifmt_parse_format/clock_provider() ASoC: fsl: switch to use snd_soc_daifmt_parse_format/clock_provider() ASoC: meson: switch to use snd_soc_daifmt_parse_format/clock_provider() ASoC: simple-card-utils: switch to use snd_soc_daifmt_parse_format/clock_provider() ASoC: soc-core: remove snd_soc_of_parse_daifmt() include/sound/soc.h | 13 +++- sound/soc/atmel/mikroe-proto.c | 18 ++--- sound/soc/fsl/fsl-asoc-card.c | 16 +--- sound/soc/generic/simple-card-utils.c | 19 ++--- sound/soc/meson/meson-card-utils.c | 15 ++-- sound/soc/soc-core.c | 103 ++++++++++++++++---------- 6 files changed, 99 insertions(+), 85 deletions(-) -- 2.25.1

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[bug report] media: sound/usb: Use Media Controller API to share media resources
by Dan Carpenter 08 Jun '21

08 Jun '21

Hello Shuah Khan, The patch 66354f18fe5f: "media: sound/usb: Use Media Controller API to share media resources" from Apr 1, 2019, leads to the following static checker warning: sound/usb/media.c:287 snd_media_device_create() warn: 'mdev' can also be NULL sound/usb/media.c 270 271 mdev = media_device_usb_allocate(usbdev, KBUILD_MODNAME, THIS_MODULE); ^^^^ If CONFIG_MEDIA_CONTROLLER is disabled then "mdev" is NULL. 272 if (IS_ERR(mdev)) 273 return -ENOMEM; 274 275 /* save media device - avoid lookups */ 276 chip->media_dev = mdev; 277 278 snd_mixer_init: 279 /* Create media entities for mixer and control dev */ 280 ret = snd_media_mixer_init(chip); 281 /* media_device might be registered, print error and continue */ 282 if (ret) 283 dev_err(&usbdev->dev, 284 "Couldn't create media mixer entities. Error: %d\n", 285 ret); 286 287 if (!media_devnode_is_registered(mdev->devnode)) { ^^^^^^^^^^^^^ dereferenced without checking here 288 /* dont'register if snd_media_mixer_init() failed */ 289 if (ret) 290 goto create_fail; 291 292 /* register media_device */ 293 ret = media_device_register(mdev); 294 create_fail: 295 if (ret) { 296 snd_media_mixer_delete(chip); 297 media_device_delete(mdev, KBUILD_MODNAME, THIS_MODULE); 298 /* clear saved media_dev */ 299 chip->media_dev = NULL; 300 dev_err(&usbdev->dev, 301 "Couldn't register media device. Error: %d\n", 302 ret); 303 return ret; 304 } 305 } 306 307 return ret; 308 } regards, dan carpenter

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alsamixer: use background color instead of COLOR_BLACK
by GitHub pull_request - opened 08 Jun '21

08 Jun '21

alsa-project/alsa-utils pull request #77 was opened from r-burns: Hi! I have a cosmetic "improvement" I'd like to share. Scare quotes because this is possibly opinionated, but I'd still like to know what you think. I think on typical terminals, COLOR_BLACK is the same or quite similar to the background color, so this does not make much of a difference. But on "solarized" color schemes and similar, the COLOR_BLACK may be quite different from the background color, so in my opinion it looks odd to have the alsamixer panel stand out this way. For example, I'm using the "Nord" color scheme on a pitch black #000000 background, and here's how the mixer looks with/without this patch (blank terminal, htop, and ncmpcpp also provided for comparison): ![image](https://user-images.githubusercontent.com/52847440/107131140-dbf46500-6888-11eb-938a-9a3258bf7184.png) As you can see, the current alsamixer in the center looks a bit strange and sticks out against my chosen background color and the other TUIs I'm using. Htop and ncmpcpp agree on using the builtin `-1` background color for their own backgrounds, but alsamixer uses COLOR_BLACK which stands out from the actual background. The alsamixer on the bottom right is using this patch, and blends in seamlessly. Again, this is just my opinion! Feel free to close if this is wrong or superfluous. Nonetheless, I thought I'd share this patch, in case anyone else feels the same way. Request URL : https://github.com/alsa-project/alsa-utils/pull/77 Patch URL : https://github.com/alsa-project/alsa-utils/pull/77.patch Repository URL: https://github.com/alsa-project/alsa-utils

2 1

[PATCH 00/15] ASoC: fixes for SoundWire codec drivers
by Pierre-Louis Bossart 08 Jun '21

08 Jun '21

The SOF CI exposed a set of issues with suspend/resume, error handling, register access and mixer values. These fixes were indentified with a tag so that they can be applied by linux-stable and distributions. Thanks to the Realtek and Maxim teams for their help. Bard Liao (3): ASoC: rt5682-sdw: set regcache_cache_only false before reading RT5682_DEVICE_ID ASoC: rt711-sdca-sdw: add readable for SDW_SDCA_CTL() registers ASoC: rt711-sdca: handle mbq_regmap in rt711_sdca_io_init Jack Yu (1): ASoC: rt715-sdca: fix clock stop prepare timeout issue Oder Chiou (1): ASoC: rt5682: Fix a problem with error handling in the io init function of the soundwire Pierre-Louis Bossart (10): ASoC: max98373-sdw: add missing memory allocation check ASoC: max98373-sdw: use first_hw_init flag on resume ASoC: rt1308-sdw: use first_hw_init flag on resume ASoC: rt1316-sdw: use first_hw_init flag on resume ASoC: rt5682-sdw: use first_hw_init flag on resume ASoC: rt700-sdw: use first_hw_init flag on resume ASoC: rt711-sdca-sdw: use first_hw_init flag on resume ASoC: rt711-sdw: use first_hw_init flag on resume ASoC: rt715-sdca-sdw: use first_hw_init flag on resume ASoC: rt715-sdw: use first_hw_init flag on resume sound/soc/codecs/max98373-sdw.c | 14 ++++++++------ sound/soc/codecs/max98373.h | 2 +- sound/soc/codecs/rt1308-sdw.c | 2 +- sound/soc/codecs/rt1316-sdw.c | 2 +- sound/soc/codecs/rt5682-sdw.c | 19 +++++++++++-------- sound/soc/codecs/rt700-sdw.c | 2 +- sound/soc/codecs/rt711-sdca-sdw.c | 12 +++++++++++- sound/soc/codecs/rt711-sdca.c | 4 ++++ sound/soc/codecs/rt711-sdw.c | 2 +- sound/soc/codecs/rt715-sdca-sdw.c | 3 ++- sound/soc/codecs/rt715-sdca-sdw.h | 1 + sound/soc/codecs/rt715-sdca.c | 9 ++++++--- sound/soc/codecs/rt715-sdca.h | 3 ++- sound/soc/codecs/rt715-sdw.c | 2 +- 14 files changed, 51 insertions(+), 26 deletions(-) -- 2.25.1

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