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[PATCH -next] ASoC: fsl: lpc3xxx: Make some symbols static
by Yue Haibing 02 Aug '24

02 Aug '24

These symbols are not used outside of the files, make them static to fix sparse warnings: sound/soc/fsl/lpc3xxx-i2s.c:261:30: warning: symbol 'lpc3xxx_i2s_dai_ops' was not declared. Should it be static? sound/soc/fsl/lpc3xxx-i2s.c:271:27: warning: symbol 'lpc3xxx_i2s_dai_driver' was not declared. Should it be static? sound/soc/fsl/lpc3xxx-pcm.c:55:39: warning: symbol 'lpc3xxx_soc_platform_driver' was not declared. Should it be static? Signed-off-by: Yue Haibing <yuehaibing(a)huawei.com> --- sound/soc/fsl/lpc3xxx-i2s.c | 4 ++-- sound/soc/fsl/lpc3xxx-pcm.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/lpc3xxx-i2s.c b/sound/soc/fsl/lpc3xxx-i2s.c index 62ef624d6dd4..bdb320bb3719 100644 --- a/sound/soc/fsl/lpc3xxx-i2s.c +++ b/sound/soc/fsl/lpc3xxx-i2s.c @@ -258,7 +258,7 @@ static int lpc3xxx_i2s_dai_probe(struct snd_soc_dai *dai) return 0; } -const struct snd_soc_dai_ops lpc3xxx_i2s_dai_ops = { +static const struct snd_soc_dai_ops lpc3xxx_i2s_dai_ops = { .probe = lpc3xxx_i2s_dai_probe, .startup = lpc3xxx_i2s_startup, .shutdown = lpc3xxx_i2s_shutdown, @@ -268,7 +268,7 @@ const struct snd_soc_dai_ops lpc3xxx_i2s_dai_ops = { .set_fmt = lpc3xxx_i2s_set_dai_fmt, }; -struct snd_soc_dai_driver lpc3xxx_i2s_dai_driver = { +static struct snd_soc_dai_driver lpc3xxx_i2s_dai_driver = { .playback = { .channels_min = 1, .channels_max = 2, diff --git a/sound/soc/fsl/lpc3xxx-pcm.c b/sound/soc/fsl/lpc3xxx-pcm.c index c0d499b9b8ba..e6abaf63895a 100644 --- a/sound/soc/fsl/lpc3xxx-pcm.c +++ b/sound/soc/fsl/lpc3xxx-pcm.c @@ -52,7 +52,7 @@ static const struct snd_dmaengine_pcm_config lpc3xxx_dmaengine_pcm_config = { .prealloc_buffer_size = 128 * 1024, }; -const struct snd_soc_component_driver lpc3xxx_soc_platform_driver = { +static const struct snd_soc_component_driver lpc3xxx_soc_platform_driver = { .name = "lpc32xx-pcm", }; -- 2.34.1

2 1

[PATCH 00/12] ALSA/ASoC: use snd_pcm_direction_name()
by Kuninori Morimoto 02 Aug '24

02 Aug '24

Hi Tatasi, Mark Many drivers are selecting strings "playback" / "capture" by own handling, but we have snd_pcm_direction_name() function for it. This patch use it. One note is that snd_pcm_direction_name() will select "Playback" and "Capture", instead of "playback" / "capture". Almost all drivers are using it as dev_dbg() or dev_err() so no problem. But some other drivers are using it as other purpose. It might be issue (?). For example ASoC debugfs dir name will be changed by this patch. Kuninori Morimoto (12): ALSA: pci: pcxhr: use snd_pcm_direction_name() ALSA: pci: rme9652: use snd_pcm_direction_name() ALSA: trace: use snd_pcm_direction_name() ALSA: aloop: use snd_pcm_direction_name() ALSA: pcm_timer: use snd_pcm_direction_name() ASoC: stm: use snd_pcm_direction_name() ASoC: sof: pcm: use snd_pcm_direction_name() ASoC: sof: intel: use snd_pcm_direction_name() ASoC: fsl: lpc3xxx-i2s: use snd_pcm_direction_name() ASoC: tegra: use snd_pcm_direction_name() ASoC: soc-pcm: use snd_pcm_direction_name() ASoC: soc-dapm: use snd_pcm_direction_name() include/trace/events/asoc.h | 3 ++- sound/core/pcm_timer.c | 3 +-- sound/drivers/aloop.c | 3 +-- sound/pci/pcxhr/pcxhr_mix22.c | 2 +- sound/pci/rme9652/hdspm.c | 6 ++---- sound/soc/fsl/lpc3xxx-i2s.c | 3 +-- sound/soc/soc-dapm.c | 3 +-- sound/soc/soc-pcm.c | 30 +++++++++++++++--------------- sound/soc/sof/intel/hda-stream.c | 4 +--- sound/soc/sof/pcm.c | 2 +- sound/soc/stm/stm32_i2s.c | 4 ++-- sound/soc/tegra/tegra210_i2s.c | 16 ++++++++-------- 12 files changed, 36 insertions(+), 43 deletions(-) -- 2.43.0

3 14

[PATCH RESEND 00/31] AMD SOF based generic SoundWire machine driver
by Vijendar Mukunda 02 Aug '24

02 Aug '24

This patch series majorly consists of below changes. - Rename structures, macros and codec helper names used in Intel SoundWire generic driver to make it generic. - Move Intel SoundWire driver common structures, macros and codec helper functions to common placeholder so that it can be used by other platform machine driver. - Refactor few SoundWire common codec helper functions. - AMD SOF based generic SoundWire machine driver for ACP 6.3 variant. This work started a couple of months ago to avoid duplication of code that wasn't really Intel-specific in the "sof_sdw" machine driver. The code went through multiple iterations, was tested for multiple weeks and a couple of build issues reported by the Intel kbuild bots were corrected. This is the initial version of SoundWire machine driver for AMD platforms. Additional code refactoring will be done in the next step on the AMD side. Link: https://github.com/thesofproject/linux/pull/5068 Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Vijendar Mukunda (31): ASoC: intel: rename codec_info and dai_info structures names ASoC: intel: rename soundwire common header macros ASoC: intel: rename soundwire machine driver soc ops ASoC: intel: rename soundwire codec helper functions ASoC: intel: rename maxim codec macros ASoC: intel: rename ignore_pch_dmic variable name ASoC: intel/sdw-utils: move soundwire machine driver soc ops ASoC: intel: move soundwire machine driver common structures ASoC: intel/sdw_utils: move soundwire machine driver helper functions ASoC: intel/sdw_utils: move dmic codec helper function ASoC: intel/sdw_utils: move rtk dmic helper functions ASoC: intel/sdw_utils: move rt712 sdca helper functions ASoC: intel/sdw_utils: move rt722 sdca helper functions ASoC: intel: split soundwire machine driver private data ASoC: intel/sdw_utils: move rt5682 codec helper function ASoC: intel/sdw_utils: move rtk jack common helper functions ASoC: intel/sdw_utils: move rt700 and rt711 codec helper functions ASoC: intel/sdw_utils: move rtk amp codec helper functions ASoC: intel/sdw_utils: move cirrus soundwire codec helper functions ASoC: intel/sdw_utils: move maxim codec helper functions ASoC: intel/sdw_utils: move dai id common macros ASoC: intel/sdw_utils: move soundwire dai type macros ASoC: intel/sdw_utils: move soundwire codec_info_list structure ASoC: intel/sdw_utils: move machine driver dai link helper functions ASoC: sdw_utils: refactor sof_sdw_card_late_probe function ASoC: intel/sdw_utils: refactor init_dai_link() and init_simple_dai_link() ASoC: soc-acpi: add pci revision id field in mach params structure ASoC: amd: acp: add soundwire machines for acp6.3 based platform ASoC: SOF: amd: add alternate machines for acp6.3 based platform ASoC: SOF: amd: update mach params subsystem_rev variable ASoC: amd/sdw_utils: add sof based soundwire generic machine driver include/sound/soc-acpi.h | 2 + include/sound/soc_sdw_utils.h | 216 ++++ sound/soc/Kconfig | 2 + sound/soc/Makefile | 1 + sound/soc/amd/acp/Kconfig | 22 + sound/soc/amd/acp/Makefile | 4 + sound/soc/amd/acp/acp-sdw-sof-mach.c | 742 +++++++++++ sound/soc/amd/acp/amd-acp63-acpi-match.c | 90 ++ sound/soc/amd/acp/soc_amd_sdw_common.h | 44 + sound/soc/amd/mach-config.h | 1 + sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/Makefile | 9 - sound/soc/intel/boards/sof_sdw.c | 1142 ++--------------- sound/soc/intel/boards/sof_sdw_common.h | 166 +-- sound/soc/intel/boards/sof_sdw_hdmi.c | 14 +- sound/soc/sdw_utils/Kconfig | 6 + sound/soc/sdw_utils/Makefile | 11 + .../soc_sdw_bridge_cs35l56.c} | 56 +- .../soc_sdw_cs42l42.c} | 13 +- .../soc_sdw_cs42l43.c} | 38 +- .../soc_sdw_cs_amp.c} | 18 +- .../soc_sdw_dmic.c} | 10 +- .../soc_sdw_maxim.c} | 56 +- .../soc_sdw_rt5682.c} | 12 +- .../soc_sdw_rt700.c} | 12 +- .../soc_sdw_rt711.c} | 38 +- .../soc_sdw_rt712_sdca.c} | 10 +- .../soc_sdw_rt722_sdca.c} | 10 +- .../soc_sdw_rt_amp.c} | 32 +- .../soc_sdw_rt_amp_coeff_tables.h} | 6 +- .../soc_sdw_rt_dmic.c} | 11 +- .../soc_sdw_rt_sdca_jack_common.c} | 40 +- sound/soc/sdw_utils/soc_sdw_utils.c | 990 ++++++++++++++ sound/soc/sof/amd/Kconfig | 1 + sound/soc/sof/amd/acp-common.c | 3 + sound/soc/sof/amd/acp.c | 1 + sound/soc/sof/amd/acp.h | 1 + sound/soc/sof/amd/pci-acp63.c | 1 + 38 files changed, 2475 insertions(+), 1357 deletions(-) create mode 100644 include/sound/soc_sdw_utils.h create mode 100644 sound/soc/amd/acp/acp-sdw-sof-mach.c create mode 100644 sound/soc/amd/acp/amd-acp63-acpi-match.c create mode 100644 sound/soc/amd/acp/soc_amd_sdw_common.h create mode 100644 sound/soc/sdw_utils/Kconfig create mode 100644 sound/soc/sdw_utils/Makefile rename sound/soc/{intel/boards/bridge_cs35l56.c => sdw_utils/soc_sdw_bridge_cs35l56.c} (63%) rename sound/soc/{intel/boards/sof_sdw_cs42l42.c => sdw_utils/soc_sdw_cs42l42.c} (82%) rename sound/soc/{intel/boards/sof_sdw_cs42l43.c => sdw_utils/soc_sdw_cs42l43.c} (73%) rename sound/soc/{intel/boards/sof_sdw_cs_amp.c => sdw_utils/soc_sdw_cs_amp.c} (66%) rename sound/soc/{intel/boards/sof_sdw_dmic.c => sdw_utils/soc_sdw_dmic.c} (71%) rename sound/soc/{intel/boards/sof_sdw_maxim.c => sdw_utils/soc_sdw_maxim.c} (63%) rename sound/soc/{intel/boards/sof_sdw_rt5682.c => sdw_utils/soc_sdw_rt5682.c} (82%) rename sound/soc/{intel/boards/sof_sdw_rt700.c => sdw_utils/soc_sdw_rt700.c} (81%) rename sound/soc/{intel/boards/sof_sdw_rt711.c => sdw_utils/soc_sdw_rt711.c} (70%) rename sound/soc/{intel/boards/sof_sdw_rt712_sdca.c => sdw_utils/soc_sdw_rt712_sdca.c} (74%) rename sound/soc/{intel/boards/sof_sdw_rt722_sdca.c => sdw_utils/soc_sdw_rt722_sdca.c} (68%) rename sound/soc/{intel/boards/sof_sdw_rt_amp.c => sdw_utils/soc_sdw_rt_amp.c} (87%) rename sound/soc/{intel/boards/sof_sdw_amp_coeff_tables.h => sdw_utils/soc_sdw_rt_amp_coeff_tables.h} (97%) rename sound/soc/{intel/boards/sof_sdw_rt_dmic.c => sdw_utils/soc_sdw_rt_dmic.c} (70%) rename sound/soc/{intel/boards/sof_sdw_rt_sdca_jack_common.c => sdw_utils/soc_sdw_rt_sdca_jack_common.c} (78%) create mode 100644 sound/soc/sdw_utils/soc_sdw_utils.c -- 2.34.1

4 34

common: pcm: hdmi/sof-hda-dsp/sof-soundwire: Syntax 7 dependent macro to generate the hdmi device sections
by GitHub pull_request - edited 02 Aug '24

02 Aug '24

alsa-project/alsa-ucm-conf pull request #413 was edited from ujfalusi: Hi, The PR will introduce a generic `HdmiDevice` macro, which can replace duplicated conditions in sof UCM files and potentially used by other devices to generate the needed section for the hdmi PCM devices to allow passthrough. The PR is marked as DNM and is in draft since the change depends on Syntax 7 support, which is only available in git version of alsa-lib, it is not a simple ask for a user (or developer) to use it and the change would break UCM for released alsa-lib. @perexg, is this something that you were suggesting? I think the macro and it's definition is generic enough, I hope that it will be usable for non SOF setups. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/413 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/413.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

common: pcm/split: add support up to 32 / 8 channels
by GitHub pull_request - edited 02 Aug '24

02 Aug '24

alsa-project/alsa-ucm-conf pull request #417 was edited from perexg: **DO NOT MERGE - after 1.2.12!** MOTU 828 has up to 32 hardware channels. Add support for this configuration to common/pcm/split.conf. Also, add support for up to 8 channels for the abstact devices (like 7.1 surround). Note that this change requires recent alsa-lib (Syntax 7). Link: https://github.com/alsa-project/alsa-ucm-conf/pull/416 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/417 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/417.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Lag between microphone and speaker
by Guillermo Bernaldo de Quiros Maraver 02 Aug '24

02 Aug '24

Hi, good afternoon! First of all, I'm new to the mailing list so I'm not sure if this is the correct place to ask this question. I ask it here because I don't know a better place to ask it so, if this is not the correct place, please let me know. The thing is, I have a USB sound card plugged to a Beaglebone AI64. This works correctly (I mean, I can record my voice and I can play a wav file without problems. using the arecord and aplay tools) The problem comes with a software I'm writing which needs to read audio from the microphone, then do a process (which takes less than 20ms) and then, after that process, I have to write the audio to the speaker. The software, right now, consists of several files, but, the most important components of the software are Recorder and Player CPP classes which I show you at the end. The important part is the next: 1º I init the alsa context using the next calls: audioPlayer.OpenDevice("plughw:CARD=Device,DEV=0"); micRecorder.OpenDevice("plughw:CARD=Device,DEV=0"); 2º I'm using the sample rate of 16KHz, a sample size of 16bit and 1 channel. So, assuming I've done correctly, this should give me a total frame rate of 342 samples, then, if I multiply it by 2 bytes, I get 684 bytes of total size. 3º Finally, in the main, I do the next thing: while(1) { //bool keyPressed = id.IsKeyPressed(0); //printf("Key is pressed: %s\n", keyPressed ? "true" : "false"); const int8_t *pAudioBuffer = micRecorder.ReadAudio(); if(pAudioBuffer) { // Process a single frame of audio // Note: output_frame can be the same buffer as input_frame for in-place processing if(keyPressed) { // DO A PROCESS WHICH TAKES 19ms more or less } else { memcpy(buff_in,pAudioBuffer, buff_size); } // Opcionalmente, leemos/escribinos de/a fichero. //write(noisy_raw_fd, pAudioBuffer, micRecorder.GetBufferSize()); //write(clean_raw_fd, g_out_buffer, g_out_buffer_size); if(audioPlayer.WriteAudio((int8_t*)buff_in)<0) { return 1; } } //printf("Numero de frames escritos: %d del total de: %d\n", pcm, frames); } This works correctly, the problem is after some time I start to note that there is a difference of time between my speaking and my listening and I don't know why. I've enabled all traces in both the recorder and the player, but I don't see any error. Is just that there is a difference between my speaking and my listening. The difference is up to 10 seconds or more. Please, can anyone help me with this problem? PD: Please, be aware I'm just a software engineer, I'm not an expert in all technical stuff related to audio. I did this code based on some examples I found on internet which are: I used this code to implement the player class: https://gist.github.com/ghedo/963382/815c98d1ba0eda1b486eb9d80d9a91a81d9952… I used this code to implement the recorder class: https://gist.github.com/albanpeignier/104902 Here is the code // Player.h #pragma once #include <iostream> #include <string> #include <alsa/asoundlib.h> #define SAMPLE_RATE 16000 #define FRAME_SIZE 342 #define CHANNELS 1 class Player { public: Player(const std::string &devname = "", int sampleRate = SAMPLE_RATE, int channels = CHANNELS); ~Player(); int OpenDevice(const std::string &devname = "", int sampleRate = SAMPLE_RATE, int channels = CHANNELS); void CloseDevice(); size_t GetFrameSize(); size_t GetBufferSize(); int WriteAudio(const int8_t *pAudioBuffer); private: int InitAudioContext(); private: int m_SampleRate; int m_Channels; std::string m_DevName; snd_pcm_t *m_AudioHandle; size_t m_FrameBufferSize; size_t m_BufferSize; int8_t *m_pAudioBuffer; }; // Player.cpp #include "Player.h" Player::Player(const std::string &devname, int sampleRate, int channels): m_DevName{devname}, m_SampleRate{sampleRate}, m_Channels{channels}, m_AudioHandle{nullptr}, m_FrameBufferSize{FRAME_SIZE} { } Player::~Player() { CloseDevice(); } int Player::OpenDevice(const std::string &devname, int sampleRate, int channels) { if(!devname.empty()) m_DevName = devname; if(sampleRate != SAMPLE_RATE) m_SampleRate = sampleRate; if(channels != CHANNELS) m_Channels = channels; return InitAudioContext(); } void Player::CloseDevice() { std::cout << "buffer freed" << std::endl; snd_pcm_close (m_AudioHandle); //free(m_pAudioBuffer); std::cout << "audio interface closed" << std::endl; } size_t Player::GetFrameSize() { return m_FrameBufferSize; } size_t Player::GetBufferSize() { return m_BufferSize; } int Player::WriteAudio(const int8_t *pAudioBuffer) { int pcm; if (pcm = snd_pcm_writei(m_AudioHandle, pAudioBuffer, FRAME_SIZE) == -EPIPE) { printf("XRUN.\n"); snd_pcm_prepare(m_AudioHandle); } else if (pcm < 0) { printf("ERROR. Can't write to PCM device. %s\n", snd_strerror(pcm)); return 0; } return 1; } int Player::InitAudioContext() { std::cout << "Iniciando Player" << std::endl; int i; int err; snd_pcm_hw_params_t *hw_params; snd_pcm_format_t format = SND_PCM_FORMAT_S16_LE; if ((err = snd_pcm_open (&m_AudioHandle, m_DevName.c_str(), SND_PCM_STREAM_PLAYBACK, 0)) < 0) { std::cerr << "cannot open audio device " << m_DevName << " Error: " << snd_strerror (err) << std::endl; return (1); } std::cout << "audio interface opened" << std::endl; if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0) { std::cerr << "cannot allocate hardware parameters structure: " << snd_strerror(err) << std::endl; return (1); } std::cout << "hw_params allocated" << std::endl; if ((err = snd_pcm_hw_params_any (m_AudioHandle, hw_params)) < 0) { std::cerr << "cannot initialize hardware parameter structure: " << snd_strerror(err) << std::endl; return (1); } std::cout << "hw_params initialized" << std::endl; if ((err = snd_pcm_hw_params_set_access (m_AudioHandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) { std::cerr << "cannot set access type: " << snd_strerror (err) << std::endl; return (1); } std::cout << "hw_params access setted" << std::endl; if ((err = snd_pcm_hw_params_set_format (m_AudioHandle, hw_params, format)) < 0) { std::cerr << "cannot set sample format: " << snd_strerror (err) << std::endl; return (1); } std::cout << "hw_params format setted" << std::endl; if ((err = snd_pcm_hw_params_set_rate_near (m_AudioHandle, hw_params, (unsigned int *)&m_SampleRate, 0)) < 0) { std::cerr << "cannot set sample rate: " << snd_strerror (err) << std::endl; return (1); } std::cout << "hw_params rate setted" << std::endl; if ((err = snd_pcm_hw_params_set_channels (m_AudioHandle, hw_params, CHANNELS)) < 0) { std::cerr << "cannot set channel count: " << snd_strerror (err) << std::endl; return (1); } std::cout << "hw_params channels setted" << std::endl; if ((err = snd_pcm_hw_params (m_AudioHandle, hw_params)) < 0) { std::cerr << "cannot set parameters: " << snd_strerror (err) << std::endl; return (1); } std::cout << "hw_params setted" << std::endl; snd_pcm_hw_params_free (hw_params); std::cout << "hw_params freed" << std::endl; if ((err = snd_pcm_prepare (m_AudioHandle)) < 0) { std::cerr << "cannot prepare audio interface for use: " << snd_strerror (err) << std::endl; return (1); } std::cout << "audio interface prepared" << std::endl; std::cout << "format width: " << snd_pcm_format_width(format) << std::endl; // Tamaño total en bytes del buffer de memoria reservado m_BufferSize = m_FrameBufferSize * snd_pcm_format_width(SND_PCM_FORMAT_S16_LE) / 8 * CHANNELS; return 0; } // Recorder.h #pragma once #include <iostream> #include <string> #include <alsa/asoundlib.h> #define SAMPLE_RATE 16000 #define FRAME_SIZE 342 #define CHANNELS 1 class Recorder { public: Recorder(const std::string &devname = "", int sampleRate = SAMPLE_RATE, int channels = CHANNELS); ~Recorder(); int OpenDevice(const std::string &devname = "", int sampleRate = SAMPLE_RATE, int channels = CHANNELS); void CloseDevice(); size_t GetFrameSize(); size_t GetBufferSize(); const int8_t *ReadAudio(); private: int InitAudioContext(); private: int m_SampleRate; int m_Channels; std::string m_DevName; snd_pcm_t *m_AudioHandle; size_t m_FrameBufferSize; size_t m_BufferSize; int8_t *m_pAudioBuffer; }; // Recorder.cpp #include "Recorder.h" /* * Recuperacion ante errores de alsa. */ static int xrun_recovery(snd_pcm_t *handle, int err) { if (err == -EPIPE) { /* under-run */ printf("underrun!\n"); err = snd_pcm_prepare(handle); if (err < 0) printf("Can't recovery from underrun, prepare failed: %s\n", snd_strerror(err)); return 0; } else if (err == -ESTRPIPE) { while ((err = snd_pcm_resume(handle)) == -EAGAIN) sleep(1); /* wait until the suspend flag is released */ if (err < 0) { err = snd_pcm_prepare(handle); if (err < 0) printf("Can't recovery from suspend, prepare failed: %s\n", snd_strerror(err)); } return 0; } return err; } Recorder::Recorder(const std::string &devname, int sampleRate, int channels): m_DevName{devname}, m_SampleRate{sampleRate}, m_Channels{channels}, m_AudioHandle{nullptr}, m_FrameBufferSize{FRAME_SIZE} { } Recorder::~Recorder() { CloseDevice(); } int Recorder::OpenDevice(const std::string &devname, int sampleRate, int channels) { if(!devname.empty()) m_DevName = devname; if(sampleRate != SAMPLE_RATE) m_SampleRate = sampleRate; if(channels != CHANNELS) m_Channels = channels; return InitAudioContext(); } size_t Recorder::GetFrameSize() { return m_FrameBufferSize; } /* * Tamaño total en bytes del buffer de memoria reservado */ size_t Recorder::GetBufferSize() { return m_BufferSize; } /* * Lee en el puntero de memoria resevado, las muestras de audio leidas de la capturadora de audio (microfono). */ const int8_t *Recorder::ReadAudio() { int err; if ((err = snd_pcm_readi(m_AudioHandle, m_pAudioBuffer, m_FrameBufferSize)) != m_FrameBufferSize) { // Si el error es de tipo PIPE o STRPIPE, tratamos de recuperar el dispositivo. if(xrun_recovery(m_AudioHandle, err) < 0) { std::cerr << "read from audio interface failed " << snd_strerror(err) << std::endl; return nullptr; } } return m_pAudioBuffer; } void Recorder::CloseDevice() { std::cout << "buffer freed" << std::endl; snd_pcm_close (m_AudioHandle); free(m_pAudioBuffer); std::cout << "audio interface closed" << std::endl; } int Recorder::InitAudioContext() { std::cout << "Iniciando Grabador" << std::endl; int i; int err; snd_pcm_hw_params_t *hw_params; snd_pcm_format_t format = SND_PCM_FORMAT_S16_LE; if ((err = snd_pcm_open (&m_AudioHandle, m_DevName.c_str(), SND_PCM_STREAM_CAPTURE, 0)) < 0) { std::cerr << "cannot open audio device " << m_DevName << " Error: " << snd_strerror (err) << std::endl; return (1); } std::cout << "audio interface opened" << std::endl; if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0) { std::cerr << "cannot allocate hardware parameters structure: " << snd_strerror(err) << std::endl; return (1); } std::cout << "hw_params allocated" << std::endl; if ((err = snd_pcm_hw_params_any (m_AudioHandle, hw_params)) < 0) { std::cerr << "cannot initialize hardware parameter structure: " << snd_strerror(err) << std::endl; return (1); } std::cout << "hw_params initialized" << std::endl; if ((err = snd_pcm_hw_params_set_access (m_AudioHandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) { std::cerr << "cannot set access type: " << snd_strerror (err) << std::endl; return (1); } std::cout << "hw_params access setted" << std::endl; if ((err = snd_pcm_hw_params_set_format (m_AudioHandle, hw_params, format)) < 0) { std::cerr << "cannot set sample format: " << snd_strerror (err) << std::endl; return (1); } std::cout << "hw_params format setted" << std::endl; if ((err = snd_pcm_hw_params_set_rate_near (m_AudioHandle, hw_params, (unsigned int *)&m_SampleRate, 0)) < 0) { std::cerr << "cannot set sample rate: " << snd_strerror (err) << std::endl; return (1); } std::cout << "hw_params rate setted" << std::endl; if ((err = snd_pcm_hw_params_set_channels (m_AudioHandle, hw_params, CHANNELS)) < 0) { std::cerr << "cannot set channel count: " << snd_strerror (err) << std::endl; return (1); } std::cout << "hw_params channels setted" << std::endl; if ((err = snd_pcm_hw_params (m_AudioHandle, hw_params)) < 0) { std::cerr << "cannot set parameters: " << snd_strerror (err) << std::endl; return (1); } std::cout << "hw_params setted" << std::endl; snd_pcm_hw_params_free (hw_params); std::cout << "hw_params freed" << std::endl; if ((err = snd_pcm_prepare (m_AudioHandle)) < 0) { std::cerr << "cannot prepare audio interface for use: " << snd_strerror (err) << std::endl; return (1); } std::cout << "audio interface prepared" << std::endl; std::cout << "formt width: " << snd_pcm_format_width(format) << std::endl; // Tamaño total en bytes del buffer de memoria reservado m_BufferSize = m_FrameBufferSize * snd_pcm_format_width(format) / 8 * CHANNELS; // Puntero al buffer de memoria reservado. m_pAudioBuffer = (int8_t *)malloc(m_BufferSize); return 0; }

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[PATCH] ASoC: SOF: Remove libraries from topology lookups
by cujomalainey＠chromium.org 01 Aug '24

01 Aug '24

From: Curtis Malainey <cujomalainey(a)chromium.org> Default firmware shipped in open source are not licensed for 3P libraries, therefore topologies should not reference them. If a OS wants to use 3P (that they have licensed) then they should use the appropriate topology override mechanisms. Fixes: 8a7d5d85ed2161 ("ASoC: SOF: mediatek: mt8195: Add devicetree support to select topologies") Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> Cc: Wojciech Macek <wmacek(a)google.com> --- sound/soc/sof/mediatek/mt8195/mt8195.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/mediatek/mt8195/mt8195.c b/sound/soc/sof/mediatek/mt8195/mt8195.c index 24ae1d4959be..1c6e035fd313 100644 --- a/sound/soc/sof/mediatek/mt8195/mt8195.c +++ b/sound/soc/sof/mediatek/mt8195/mt8195.c @@ -573,7 +573,7 @@ static const struct snd_sof_dsp_ops sof_mt8195_ops = { static struct snd_sof_of_mach sof_mt8195_machs[] = { { .compatible = "google,tomato", - .sof_tplg_filename = "sof-mt8195-mt6359-rt1019-rt5682-dts.tplg" + .sof_tplg_filename = "sof-mt8195-mt6359-rt1019-rt5682.tplg" }, { .compatible = "mediatek,mt8195", .sof_tplg_filename = "sof-mt8195.tplg" -- 2.46.0.rc2.264.g509ed76dc8-goog

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[PATCH] slimbus: generate MODULE_ALIAS() from MODULE_DEVICE_TABLE()
by Masahiro Yamada 01 Aug '24

01 Aug '24

Commit 9e663f4811c6 ("slimbus: core: add support to uevent") added the MODALIAS=slim:* uevent variable, but modpost does not generate the corresponding MODULE_ALIAS(). To support automatic module loading, slimbus drivers still need to manually add MODULE_ALIAS("slim:<manf_id>:<prod_code>:*"), as seen in sound/soc/codecs/wcd9335.c. To automate this, make modpost generate the proper MODULE_ALIAS() from MODULE_DEVICE_TABLE(slim, ). Signed-off-by: Masahiro Yamada <masahiroy(a)kernel.org> --- scripts/mod/devicetable-offsets.c | 4 ++++ scripts/mod/file2alias.c | 11 +++++++++++ 2 files changed, 15 insertions(+) diff --git a/scripts/mod/devicetable-offsets.c b/scripts/mod/devicetable-offsets.c index 518200813d4e..9c7b404defbd 100644 --- a/scripts/mod/devicetable-offsets.c +++ b/scripts/mod/devicetable-offsets.c @@ -153,6 +153,10 @@ int main(void) DEVID_FIELD(i3c_device_id, part_id); DEVID_FIELD(i3c_device_id, extra_info); + DEVID(slim_device_id); + DEVID_FIELD(slim_device_id, manf_id); + DEVID_FIELD(slim_device_id, prod_code); + DEVID(spi_device_id); DEVID_FIELD(spi_device_id, name); diff --git a/scripts/mod/file2alias.c b/scripts/mod/file2alias.c index 5d1c61fa5a55..99dce93a4188 100644 --- a/scripts/mod/file2alias.c +++ b/scripts/mod/file2alias.c @@ -960,6 +960,16 @@ static int do_i3c_entry(const char *filename, void *symval, return 1; } +static int do_slim_entry(const char *filename, void *symval, char *alias) +{ + DEF_FIELD(symval, slim_device_id, manf_id); + DEF_FIELD(symval, slim_device_id, prod_code); + + sprintf(alias, "slim:%x:%x:*", manf_id, prod_code); + + return 1; +} + /* Looks like: spi:S */ static int do_spi_entry(const char *filename, void *symval, char *alias) @@ -1555,6 +1565,7 @@ static const struct devtable devtable[] = { {"rpmsg", SIZE_rpmsg_device_id, do_rpmsg_entry}, {"i2c", SIZE_i2c_device_id, do_i2c_entry}, {"i3c", SIZE_i3c_device_id, do_i3c_entry}, + {"slim", SIZE_slim_device_id, do_slim_entry}, {"spi", SIZE_spi_device_id, do_spi_entry}, {"dmi", SIZE_dmi_system_id, do_dmi_entry}, {"platform", SIZE_platform_device_id, do_platform_entry}, -- 2.43.0

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[PATCH 0/3] slimbus: use 'time_left' instead of 'timeout' with wait_for_*() functions
by Wolfram Sang 01 Aug '24

01 Aug '24

There is a confusing pattern in the kernel to use a variable named 'timeout' to store the result of wait_for_*() functions causing patterns like: timeout = wait_for_completion_timeout(...) if (!timeout) return -ETIMEDOUT; with all kinds of permutations. Use 'time_left' as a variable to make the code obvious and self explaining. This is part of a tree-wide series. The rest of the patches can be found here (some parts may still be WIP): git://git.kernel.org/pub/scm/linux/kernel/git/wsa/linux.git i2c/time_left Because these patches are generated, I audit them before sending. This is why I will send series step by step. Build bot is happy with these patches, though. No functional changes intended. Wolfram Sang (3): slimbus: messaging: use 'time_left' variable with wait_for_completion_timeout() slimbus: qcom-ctrl: use 'time_left' variable with wait_for_completion_timeout() slimbus: qcom-ngd-ctrl: use 'time_left' variable with wait_for_completion_timeout() drivers/slimbus/messaging.c | 9 +++++---- drivers/slimbus/qcom-ctrl.c | 7 ++++--- drivers/slimbus/qcom-ngd-ctrl.c | 29 ++++++++++++++++------------- 3 files changed, 25 insertions(+), 20 deletions(-) -- 2.43.0

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Re: [PATCH RESEND 00/31] AMD SOF based generic SoundWire machine driver
by Mukunda,Vijendar 01 Aug '24

01 Aug '24

On 01/08/24 16:27, Charles Keepax wrote: > On Thu, Aug 01, 2024 at 02:44:15PM +0530, Vijendar Mukunda wrote: >> This patch series majorly consists of below changes. >> - Rename structures, macros and codec helper names used in Intel >> SoundWire generic driver to make it generic. >> - Move Intel SoundWire driver common structures, macros and codec >> helper functions to common placeholder so that it can be used by >> other platform machine driver. >> - Refactor few SoundWire common codec helper functions. >> - AMD SOF based generic SoundWire machine driver for ACP 6.3 variant. >> >> This work started a couple of months ago to avoid duplication of code >> that wasn't really Intel-specific in the "sof_sdw" machine driver. >> The code went through multiple iterations, was tested for multiple weeks >> and a couple of build issues reported by the Intel kbuild bots were >> corrected. >> >> This is the initial version of SoundWire machine driver for AMD >> platforms. Additional code refactoring will be done in the next step on >> the AMD side. >> >> Link: https://github.com/thesofproject/linux/pull/5068 >> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> >> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> >> > Would be good to also include the linux-sound mailing list on the > cover letter, if not the individuals given they are basically the > same on each patch. > > Also assuming your mail timed out again, you should be able to do > a separate git send-email for the later patches using the > --in-reply-to option to give the Message-ID of the cover letter > and they should append to the same chain. > > Thanks, > Charles Thanks, Charles, for your suggestion. I have shared the rest of the Patch series. - Vijendar

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