November 2024 - Alsa-devel - mailman.alsa-project.org

topology: Fix strcat() to uninitialized memory in add_define()
by GitHub pull_request - opened 07 Nov '24

07 Nov '24

alsa-project/alsa-utils pull request #277 was opened from rfvirgil: If *defs == NULL the realloc() will be a simple malloc() and return a pointer to uninitialized memory. Doing a strcat() to this is bad, so in that case strcpy() the string into the newly allocated buffer. This fixes the buffer overflow bug, such as: *** buffer overflow detected ***: terminated Program received signal SIGABRT, Aborted. __pthread_kill_implementation (no_tid=0, signo=6, threadid=<optimized out>) at ./nptl/pthread_kill.c:44 warning: 44 ./nptl/pthread_kill.c: No such file or directory (gdb) bt at ../sysdeps/posix/libc_fatal.c:132 at ./debug/fortify_fail.c:24 dest=dest@entry=0x5555555682c0 "hUUU\005PLATFORM=mtl,PREPROCESS_PLUGINS=nhlt,NHLT_BIN=nhlt-sof-mtl-sdw-cs42l42-l0-max98363-l2-4ch-48k-96k.bin,NUM_DMICS=4,PDM1_MIC_A_ENABLE=1,PDM1_MIC_B_ENABLE=1,DMIC0_ID=3,DMIC1_ID=4,DMIC1_ENABLE=passth"..., src=src@entry=0x7fffffffe0fa "PLATFORM=mtl,PREPROCESS_PLUGINS=nhlt,NHLT_BIN=nhlt-sof-mtl-sdw-cs42l42-l0-max98363-l2-4ch-48k-96k.bin,NUM_DMICS=4,PDM1_MIC_A_ENABLE=1,PDM1_MIC_B_ENABLE=1,DMIC0_ID=3,DMIC1_ID=4,DMIC1_ENABLE=passthrough"..., destlen=<optimized out>) at ./debug/strcat_chk.c:34 __src=0x7fffffffe0fa "PLATFORM=mtl,PREPROCESS_PLUGINS=nhlt,NHLT_BIN=nhlt-sof-mtl-sdw-cs42l42-l0-max98363-l2-4ch-48k-96k.bin,NUM_DMICS=4,PDM1_MIC_A_ENABLE=1,PDM1_MIC_B_ENABLE=1,DMIC0_ID=3,DMIC1_ID=4,DMIC1_ENABLE=passthrough"..., __dest=<optimized out>) at /usr/include/x86_64-linux-gnu/bits/string_fortified.h:130 d=0x7fffffffe0fa "PLATFORM=mtl,PREPROCESS_PLUGINS=nhlt,NHLT_BIN=nhlt-sof-mtl-sdw-cs42l42-l0-max98363-l2-4ch-48k-96k.bin,NUM_DMICS=4,PDM1_MIC_A_ENABLE=1,PDM1_MIC_B_ENABLE=1,DMIC0_ID=3,DMIC1_ID=4,DMIC1_ENABLE=passthrough"..., defs=<synthetic pointer>) at topology.c:400 Request URL : https://github.com/alsa-project/alsa-utils/pull/277 Patch URL : https://github.com/alsa-project/alsa-utils/pull/277.patch Repository URL: https://github.com/alsa-project/alsa-utils

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[BUG] HDA Intel PCH, HDMI , HDA Intel PCH, ALC886 Digital, no IEC958 / HDMI output/passthrough to AVR
by schorpp 07 Nov '24

07 Nov '24

[BUG] HDA Intel PCH, HDMI , HDA Intel PCH, ALC886 Digital, no IEC958 / HDMI output/passthrough to AVR ALSA versions 1.0.25...current Kernel Driver versions 3.12.40...6.x Hardware: DFI Mainboard MB330-CRM https://www.dfi.com/de/product/index/273#specification Reproducible Always Steps to reproduce and full report: https://www-vdr--portal-de.translate.goog/forum/index.php?thread/136519-gel… Software crosscheck with CM106 USB Soundbox iec958: works OK. Where to start in the drivers source code to fix it? y tom

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BridgeCast v2 not working properly despite V2 configuration files
by GitHub issues - edited 06 Nov '24

06 Nov '24

alsa-project/alsa-ucm-conf issue #458 was edited from acerammos: Hi everybody, I (sadly) made the firmware update for the Roland BridgeCast and it came out broken. It was working flawlessly before the update, thanks to you i had all the Output/Input channels available and i was able to assign sound sources on them individually through the system settings gui (Kmix - Plasma). Today, as a now "V2", it is over. The BridgeCast seems to be assigned to the new conf files with its new usb tag, to `BridgeCastV2-Hifi.conf` and not former the `BridgeCast-Hifi.conf`. It seems to be a good start, thanks to Fulmer for the merge and first testing Roland's update. But now, i have a weird mix of "auxiliary channels" instead of proper audio outputs like before : ![Copie d'écran_20241106_154428](https://github.com/user-attachments/assets/4240924b-… > _(Tell me if you need Fr>En translation on that screenshot)_ Note that "chromium" (wich is in fact a client like spotify, playing music) cannot be assigned to a specific output, because BridgeCast is now detected to be "only one channel" so the option is missing. An exemple after also enabling my speakers : ![Copie d'écran_20241106_155745](https://github.com/user-attachments/assets/d01d1a67-… > _Old behaviour allowed me to choose between all the output channels of the BridgeCast, like "Games", "Aux", "Music", Chat" or "System"_ Something else i've (maybe) noticed is the differences for the channel maps, between the `cat /proc/asound/cardX/stream0` channels output and what is written in the `Roland/BridgeCastV2-Hifi.conf`. `cat /proc/asound/cardX/stream0 :` ``` Roland BRIDGE CAST V2 at usb-0000:00:14.0-1, high speed : USB Audio Playback: Status: Running Interface = 1 Altset = 1 Packet Size = 392 Momentary freq = 48000 Hz (0x6.0000) Interface 1 Altset 1 Format: S32_LE Channels: 14 Endpoint: 0x0d (13 OUT) (ASYNC) Rates: 48000 Data packet interval: 125 us Bits: 24 Channel map: FL FR FC LFE RL RR FLC FRC RC SL SR TC -- -- Sync Endpoint: 0x8e (14 IN) Sync EP Interface: 2 Sync EP Altset: 1 Implicit Feedback Mode: Yes Capture: Status: Stop Interface 2 Altset 1 Format: S32_LE Channels: 6 Endpoint: 0x8e (14 IN) (ASYNC) Rates: 48000 Data packet interval: 125 us Bits: 24 Channel map: FL FR FC LFE RL RR Sync Endpoint: 0x0d (13 OUT) Sync EP Interface: 1 Sync EP Altset: 1 Implicit Feedback Mode: Yes ``` `BridgeCastV2-Hifi.conf :` ``` Macro [ { SplitPCM { Name "bc_stereo_out" Direction Playback Channels 2 HWChannels 14 HWChannelPos0 FL HWChannelPos1 FR HWChannelPos2 FL HWChannelPos3 FR HWChannelPos4 FL HWChannelPos5 FR HWChannelPos6 FL HWChannelPos7 FR HWChannelPos8 FL HWChannelPos9 FR HWChannelPos10 FL HWChannelPos11 FR HWChannelPos12 FL HWChannelPos13 FR } } { SplitPCM { Name "bc_stereo_in" Direction Capture Channels 2 HWChannels 6 HWChannelPos0 FL HWChannelPos1 FR HWChannelPos2 FL HWChannelPos3 FR HWChannelPos4 FL HWChannelPos5 FR } } ] ``` I'm completely new to this kind of code/linux driver wizardry so i haven't been able to gather more data, but i'm not afraid to be messing around, so if i can be of any help, screenshot, command outputs, random file testing, i'll gladly help. Regards, and thank you all ! > _Edits : Spelling/Grammar mistakes_ Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/458 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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BridgeCast v2 not working properly despite V2 configuration files
by GitHub issues - edited 06 Nov '24

06 Nov '24

alsa-project/alsa-ucm-conf issue #458 was edited from acerammos: Hi everybody, I (sadly) made the firmware update for the Roland BridgeCast and it came out broken. It was working flawlessly before the update, thanks to you i had all the Output/Input channels available and i was able to assign sound sources on them individually through the system settings gui (Kmix - Plasma). Today, as a now "V2", it is over. The BridgeCast seems to be assigned to the new conf files with its new usb tag, to `BridgeCastV2-Hifi.conf` and not former the `BridgeCast-Hifi.conf`. It seems to be a good start, thanks to Fulmer for the merge and first testing Roland's update. But now, i have a weird mix of "auxiliary channels" instead of proper audio outputs like before : ![Copie d'écran_20241106_154428](https://github.com/user-attachments/assets/4240924b-… > _(Tell me if you need Fr>En translation on that screenshot)_ Note that "chromium" (wich is in fact a client like spotify, playing music) cannot be assigned to a specific output, because BridgeCast is now detected to be "only one channel" so the option is missing. An exemple after also enabling my speakers : ![Copie d'écran_20241106_155745](https://github.com/user-attachments/assets/d01d1a67-… > _Old behaviour allowed me to choose between all the output channels of the BridgeCast, like "Games", "Aux", "Music", Chat" or "System"_ Something else i've (maybe) noticed is the differences for the channel maps, between the `cat /proc/asound/cardX/stream0` channels output and what is written in the `Roland/BridgeCastV2-Hifi.conf`. `cat /proc/asound/cardX/stream0 :` ``` Roland BRIDGE CAST V2 at usb-0000:00:14.0-1, high speed : USB Audio Playback: Status: Running Interface = 1 Altset = 1 Packet Size = 392 Momentary freq = 48000 Hz (0x6.0000) Interface 1 Altset 1 Format: S32_LE Channels: 14 Endpoint: 0x0d (13 OUT) (ASYNC) Rates: 48000 Data packet interval: 125 us Bits: 24 Channel map: FL FR FC LFE RL RR FLC FRC RC SL SR TC -- -- Sync Endpoint: 0x8e (14 IN) Sync EP Interface: 2 Sync EP Altset: 1 Implicit Feedback Mode: Yes Capture: Status: Stop Interface 2 Altset 1 Format: S32_LE Channels: 6 Endpoint: 0x8e (14 IN) (ASYNC) Rates: 48000 Data packet interval: 125 us Bits: 24 Channel map: FL FR FC LFE RL RR Sync Endpoint: 0x0d (13 OUT) Sync EP Interface: 1 Sync EP Altset: 1 Implicit Feedback Mode: Yes ``` `BridgeCastV2-Hifi.conf :` ``` Macro [ { SplitPCM { Name "bc_stereo_out" Direction Playback Channels 2 HWChannels 14 HWChannelPos0 FL HWChannelPos1 FR HWChannelPos2 FL HWChannelPos3 FR HWChannelPos4 FL HWChannelPos5 FR HWChannelPos6 FL HWChannelPos7 FR HWChannelPos8 FL HWChannelPos9 FR HWChannelPos10 FL HWChannelPos11 FR HWChannelPos12 FL HWChannelPos13 FR } } { SplitPCM { Name "bc_stereo_in" Direction Capture Channels 2 HWChannels 6 HWChannelPos0 FL HWChannelPos1 FR HWChannelPos2 FL HWChannelPos3 FR HWChannelPos4 FL HWChannelPos5 FR } } ] ``` I'm completely new to this kind of code/linux driver wizardry so i haven't been able to gather more data, but i'm not afraid to be messing around, so if i can be of any help, screenshot, command outputs, random file testing, i'll gladly help. Regards, and thank you all ! > _Edits : Spellings/Grammar mistakes_ Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/458 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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BridgeCast v2 not working properly despite V2 configuration files
by GitHub issues - edited 06 Nov '24

06 Nov '24

alsa-project/alsa-ucm-conf issue #458 was edited from acerammos: Hi everybody, I (sadly) made the firmware update for the Roland BridgeCast and it came out broken. It was working flawlessly before the update, thanks to you i had all the Output/Input channels available and i was able to assign sound sources on them individually through the system settings gui (Kmix - Plasma). Today, as a now "V2", it is done. The BridgeCast seems to be assigned to the correct conf file with its new usb tag, to `BridgeCastV2-Hifi.conf` and not former the `BridgeCast-Hifi.conf`. It seems to be a good start, thanks to Fulmer for the merge and first testing Roland's update. But now, i have a weird mix of "auxiliary channels" instead of proper audio outputs like before : ![Copie d'écran_20241106_154428](https://github.com/user-attachments/assets/4240924b-… > _(Tell me if you need Fr>En translation on that screenshot)_ Note that "chromium" (wich is in fact a client like spotify, playing music) cannot be assigned to a specific output, because BridgeCast is now detected to be "only one channel" so the option is missing. An exemple after also enabling my speakers : ![Copie d'écran_20241106_155745](https://github.com/user-attachments/assets/d01d1a67-… > _Old behaviour allowed me to choose between all the output channels of the BridgeCast, like "Games", "Aux", "Music", Chat" or "System"_ Something else i've (maybe) noticed is the differences for the channel maps, between the `cat /proc/asound/cardX/stream0` channels output and what is written in the `Roland/BridgeCastV2-Hifi.conf`. `cat /proc/asound/cardX/stream0 :` ``` Roland BRIDGE CAST V2 at usb-0000:00:14.0-1, high speed : USB Audio Playback: Status: Running Interface = 1 Altset = 1 Packet Size = 392 Momentary freq = 48000 Hz (0x6.0000) Interface 1 Altset 1 Format: S32_LE Channels: 14 Endpoint: 0x0d (13 OUT) (ASYNC) Rates: 48000 Data packet interval: 125 us Bits: 24 Channel map: FL FR FC LFE RL RR FLC FRC RC SL SR TC -- -- Sync Endpoint: 0x8e (14 IN) Sync EP Interface: 2 Sync EP Altset: 1 Implicit Feedback Mode: Yes Capture: Status: Stop Interface 2 Altset 1 Format: S32_LE Channels: 6 Endpoint: 0x8e (14 IN) (ASYNC) Rates: 48000 Data packet interval: 125 us Bits: 24 Channel map: FL FR FC LFE RL RR Sync Endpoint: 0x0d (13 OUT) Sync EP Interface: 1 Sync EP Altset: 1 Implicit Feedback Mode: Yes ``` `BridgeCastV2-Hifi.conf :` ``` Macro [ { SplitPCM { Name "bc_stereo_out" Direction Playback Channels 2 HWChannels 14 HWChannelPos0 FL HWChannelPos1 FR HWChannelPos2 FL HWChannelPos3 FR HWChannelPos4 FL HWChannelPos5 FR HWChannelPos6 FL HWChannelPos7 FR HWChannelPos8 FL HWChannelPos9 FR HWChannelPos10 FL HWChannelPos11 FR HWChannelPos12 FL HWChannelPos13 FR } } { SplitPCM { Name "bc_stereo_in" Direction Capture Channels 2 HWChannels 6 HWChannelPos0 FL HWChannelPos1 FR HWChannelPos2 FL HWChannelPos3 FR HWChannelPos4 FL HWChannelPos5 FR } } ] ``` I'm completely new to this kind of code/linux driver wizardry so i haven't been able to gather more data, but i'm not afraid to be messing around, so if i can be of any help, screenshot, command outputs, random file testing, i'll gladly help. Regards, and thank you all ! Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/458 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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BridgeCast v2 not working despite V2 module
by GitHub issues - opened 06 Nov '24

06 Nov '24

alsa-project/alsa-ucm-conf issue #458 was opened from acerammos: Hi everybody, I (sadly) made the firmware update for the Roland BridgeCast and it came out broken. It was working flawlessly before the update, thanks to you i had all the Output/Input channels available and i was able to assign sound sources on them individually through the system settings gui (Kmix - Plasma). Today, as a now "V2", it is done. The BridgeCast seems to be assigned to the correct conf file with its new usb tag, to `BridgeCastV2-Hifi.conf` and not former the `BridgeCast-Hifi.conf`. It seems to be a good start, thanks to Fulmer for the merge and first testing Roland's update. But now, i have a weird mix of "auxiliary channels" instead of proper audio outputs like before : ![Copie d'écran_20241106_154428](https://github.com/user-attachments/assets/4240924b-… > _(Tell me if you need Fr>En translation on that screenshot)_ Note that "chromium" (wich is in fact a client like spotify, playing music) cannot be assigned to a specific output, because BridgeCast is now detected to be "only one channel" so the option is missing. An exemple after also enabling my speakers : ![Copie d'écran_20241106_155745](https://github.com/user-attachments/assets/d01d1a67-… > _Old behaviour allowed me to choose between all the output channels of the BridgeCast, like "Games", "Aux", "Music", Chat" or "System"_ Something else i've (maybe) noticed is the differences for the channel maps, between the `cat /proc/asound/cardX/stream0` channels output and what is written in the `Roland/BridgeCastV2-Hifi.conf`. `cat /proc/asound/cardX/stream0 :` ``` Roland BRIDGE CAST V2 at usb-0000:00:14.0-1, high speed : USB Audio Playback: Status: Running Interface = 1 Altset = 1 Packet Size = 392 Momentary freq = 48000 Hz (0x6.0000) Interface 1 Altset 1 Format: S32_LE Channels: 14 Endpoint: 0x0d (13 OUT) (ASYNC) Rates: 48000 Data packet interval: 125 us Bits: 24 Channel map: FL FR FC LFE RL RR FLC FRC RC SL SR TC -- -- Sync Endpoint: 0x8e (14 IN) Sync EP Interface: 2 Sync EP Altset: 1 Implicit Feedback Mode: Yes Capture: Status: Stop Interface 2 Altset 1 Format: S32_LE Channels: 6 Endpoint: 0x8e (14 IN) (ASYNC) Rates: 48000 Data packet interval: 125 us Bits: 24 Channel map: FL FR FC LFE RL RR Sync Endpoint: 0x0d (13 OUT) Sync EP Interface: 1 Sync EP Altset: 1 Implicit Feedback Mode: Yes ``` `BridgeCastV2-Hifi.conf :` ``` Macro [ { SplitPCM { Name "bc_stereo_out" Direction Playback Channels 2 HWChannels 14 HWChannelPos0 FL HWChannelPos1 FR HWChannelPos2 FL HWChannelPos3 FR HWChannelPos4 FL HWChannelPos5 FR HWChannelPos6 FL HWChannelPos7 FR HWChannelPos8 FL HWChannelPos9 FR HWChannelPos10 FL HWChannelPos11 FR HWChannelPos12 FL HWChannelPos13 FR } } { SplitPCM { Name "bc_stereo_in" Direction Capture Channels 2 HWChannels 6 HWChannelPos0 FL HWChannelPos1 FR HWChannelPos2 FL HWChannelPos3 FR HWChannelPos4 FL HWChannelPos5 FR } } ] ``` I'm completely new to this kind of code/linux driver wizardry so i haven't been able to gather more data, but i'm not afraid to be messing around, so if i can be of any help, screenshot, command outputs, random file testing, i'll gladly help. Regards, and thank you all ! Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/458 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH v4 0/2] Add a driver for the Iron Device SMA1307 Amp
by Kiseok Jo 06 Nov '24

06 Nov '24

This adds basic audio support for the Iron Device SMA1307 amplifier Kiseok Jo (2): This adds the schema binding for the Iron Device SMA1307 Amp The Iron Device SMA1307 is a boosted digital speaker amplifier .../bindings/sound/irondevice,sma1307.yaml | 53 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sma1307.c | 2052 +++++++++++++++++ sound/soc/codecs/sma1307.h | 444 ++++ 5 files changed, 2561 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/irondevice,sma1307.yaml create mode 100644 sound/soc/codecs/sma1307.c create mode 100644 sound/soc/codecs/sma1307.h -- 2.39.2

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Help to solve pop when sound loops
by GitHub issues - edited 06 Nov '24

06 Nov '24

alsa-project/alsa-lib issue #417 was edited from charlesmulder: Hi. Apologies for asking for help here. I don't know where else to go. I'm working on a wavetable project for college. I've managed to create a wavetable containing a sine wave. The pitch can be altered via a frequency variable (f). There is an audible popping sound when the sound loops. I have noticed that increasing the buffer size, postpones the pop. I've been reading up on `buffer > period > frames`, but I don't know how to go about removing the pop. Would really appreciate some guidance. Thanks C ```c #include <stdio.h> #include <alsa/asoundlib.h> #include <math.h> unsigned char sinuc( float ); #define TWOPI 2*M_PI #define ALSA_INFO 1 static char *device = "default"; /* playback device */ unsigned char buffer[24*1024]; int main(void) { // ALSA playback related int err; snd_pcm_t *playback_handle; // pcm snd_pcm_hw_params_t *hw_params; unsigned int val, val2; int dir; snd_pcm_uframes_t period_size; snd_pcm_format_t format; // Wavetable related unsigned int N = 1024; float f = 440; // A4 note //float f = 261.626; // C4 note unsigned int fs = 44100; // playback if ((err = snd_pcm_open(&playback_handle, device, SND_PCM_STREAM_PLAYBACK, 0)) < 0) { printf("Playback open error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } // ALSA configuration /* Allocate a hardware parameters object. */ snd_pcm_hw_params_alloca(&hw_params); /* Fill it in with default values. */ snd_pcm_hw_params_any(playback_handle, hw_params); /* Set the desired hardware parameters. */ /* Interleaved mode */ snd_pcm_hw_params_set_access(playback_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED); /* Unsigned 8-bit little-endian format */ snd_pcm_hw_params_set_format(playback_handle, hw_params, SND_PCM_FORMAT_U8); /* Two channels (stereo) */ snd_pcm_hw_params_set_channels(playback_handle, hw_params, 1); /* 44100 bits/second sampling rate (CD quality) */ snd_pcm_hw_params_set_rate_near(playback_handle, hw_params, &fs, &dir); /* Set period size to 32 frames. */ period_size = 32; snd_pcm_hw_params_set_period_size_near(playback_handle, hw_params, &period_size, &dir); /* Write the parameters to the driver */ err = snd_pcm_hw_params(playback_handle, hw_params); if (err < 0) { fprintf(stderr, "unable to set hw parameters: %s\n", snd_strerror(err)); exit(1); } // Wavetable init unsigned char wavetable[N]; // wavetable buffer float angle_inc = TWOPI/(float)N; // sine wave angle increment float index_inc = N*f/(float)fs; // wavetable index increment // Populate wavetable with a sine wave for( int n = 0; n < N; n++ ) { wavetable[n] = sinuc( angle_inc * n ); // 0 - 255 range } // ALSA Sample Buffer // period = 940 frames // buffer = 15052 frames float n = 0; for (int i = 0; i < sizeof(buffer)/sizeof(char); i++) { buffer[i] = wavetable[(int)n]; //printf("%d\n",buffer[i]); n = n+index_inc; if( (int)n >= N ) { n = 0; } } if ((err = snd_pcm_prepare (playback_handle)) < 0) { fprintf (stderr, "cannot prepare audio interface for use (%s)\n", snd_strerror (err)); exit (1); } //for (int i = 0; i < 16; i++) { while(1) { period_size = snd_pcm_writei(playback_handle, buffer, sizeof(buffer)); if (period_size < 0) period_size = snd_pcm_recover(playback_handle, period_size, 0); if (period_size < 0) { printf("snd_pcm_writei failed: %s\n", snd_strerror(period_size)); break; } if (period_size > 0 && period_size < (long)sizeof(buffer)) printf("Short write (expected %li, wrote %li)\n", (long)sizeof(buffer), period_size); } // pass the remaining samples, otherwise they're dropped in close err = snd_pcm_drain(playback_handle); if (err < 0) printf("snd_pcm_drain failed: %s\n", snd_strerror(err)); //snd_pcm_hw_params_free(hw_params); snd_pcm_close(playback_handle); return 0; } /** * Sine unsigned char. * Scales sine output to a char * Original range -1 to 1. * New range 0 - 255. */ unsigned char sinuc( float angle ) { return (sinf( angle ) * 255 + 255) / 2; } ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/417 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH 0/8] ASoC: cleasnup rtd and its ID
by Kuninori Morimoto 05 Nov '24

05 Nov '24

Hi Mark rtd has own ID, but it is naming "num" for it. The naming is confusable. This patch rename it to "id". And many functions request both "rtd" and its "id" as function parameter, but rtd itself has rtd->id. This patch cleanup it. And, Qcom driver want to use irregular rtd ID because of its topology, and thus, soc-core need irregular calculation. I'm not sure why only Qcom needs such calculation, but this patch also cleanup it. But I guess we want to cleanup is not soc-core but Qcom side (?) Kuninori Morimoto (8): ASoC: rename rtd->num to rtd->id ASoC: fsl: switch to use rtd->id from rtd->num ASoC: meson: switch to use rtd->id from rtd->num ASoC: sh: switch to use rtd->id from rtd->num ASoC: generic: switch to use rtd->id from rtd->num ASoC: remove rtd->num ASoC: soc-core: do rtd->id trick at snd_soc_add_pcm_runtime() ASoC: cleanup function parameter for rtd and its id include/sound/soc-dai.h | 5 ++- include/sound/soc.h | 8 ++--- sound/soc/fsl/imx-card.c | 6 ++-- sound/soc/generic/simple-card-utils.c | 10 +++--- sound/soc/meson/axg-card.c | 6 ++-- sound/soc/meson/gx-card.c | 2 +- sound/soc/sh/rcar/core.c | 2 +- sound/soc/soc-compress.c | 9 +++-- sound/soc/soc-core.c | 50 +++++++++++++++------------ sound/soc/soc-dai.c | 4 +-- sound/soc/soc-pcm.c | 16 ++++----- 11 files changed, 60 insertions(+), 58 deletions(-) -- 2.43.0

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[PATCH 0/2] Soundwire dai link init logic refactor
by Vijendar Mukunda 05 Nov '24

05 Nov '24

This patch series refactors the SoundWire dai link init logic for Intel and AMD generic SoundWire machine driver and also updates the stream_name in dai_links structure. Link: https://github.com/thesofproject/linux/pull/5218 Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Liam Girdwood <liam.r.girdwood(a)intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Vijendar Mukunda (2): ASoC: sdw_utils/intel/amd: refactor dai link init logic ASoC: sdw_utils: Update stream_name in dai_links structure include/sound/soc_sdw_utils.h | 5 +++-- sound/soc/amd/acp/acp-sdw-sof-mach.c | 8 ++++---- sound/soc/intel/boards/sof_sdw.c | 12 ++++++------ sound/soc/sdw_utils/soc_sdw_utils.c | 10 ++++++---- 4 files changed, 19 insertions(+), 16 deletions(-) -- 2.34.1

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