#if USE_ALSA #include "audiostream.h" #include #include "cfg/cfg.h" #include "cfg/option.h" class AlsaAudioBackend : public AudioBackend { snd_pcm_t *handle = nullptr; bool pcm_blocking = true; snd_pcm_uframes_t buffer_size = 0; snd_pcm_uframes_t period_size = 0; snd_pcm_t *handle_record = nullptr; public: AlsaAudioBackend() : AudioBackend("alsa", "Advanced Linux Sound Architecture") {} bool init() override { snd_pcm_hw_params_t *params; std::string device = cfgLoadStr("alsa", "device", ""); int rc = -1; if (!device.empty() && device != "auto") { rc = snd_pcm_open(&handle, device.c_str(), SND_PCM_STREAM_PLAYBACK, 0); if (rc < 0) WARN_LOG(AUDIO, "ALSA: Cannot open device %s. Trying auto", device.c_str()); } if (rc < 0) { INFO_LOG(AUDIO, "ALSA: trying to determine audio device"); // trying default device device = "default"; rc = snd_pcm_open(&handle, device.c_str(), SND_PCM_STREAM_PLAYBACK, 0); // "default" didn't work, try first device if (rc < 0) { device = "plughw:0,0,0"; rc = snd_pcm_open(&handle, device.c_str(), SND_PCM_STREAM_PLAYBACK, 0); if (rc < 0) { device = "plughw:0,0"; rc = snd_pcm_open(&handle, device.c_str(), SND_PCM_STREAM_PLAYBACK, 0); } } // first didn't work, try second if (rc < 0) { device = "plughw:1,0"; rc = snd_pcm_open(&handle, device.c_str(), SND_PCM_STREAM_PLAYBACK, 0); } // try pulse audio backend if (rc < 0) { device = "pulse"; rc = snd_pcm_open(&handle, device.c_str(), SND_PCM_STREAM_PLAYBACK, 0); } if (rc < 0) INFO_LOG(AUDIO, "ALSA: unable to automatically determine audio device."); } if (rc < 0) { ERROR_LOG(AUDIO, "ALSA: unable to open PCM device %s: %s", device.c_str(), snd_strerror(rc)); return false; } INFO_LOG(AUDIO, "ALSA: Successfully initialized \"%s\"", device.c_str()); /* Allocate a hardware parameters object. */ snd_pcm_hw_params_alloca(¶ms); /* Fill it in with default values. */ rc=snd_pcm_hw_params_any(handle, params); if (rc < 0) { ERROR_LOG(AUDIO, "ALSA: Error:snd_pcm_hw_params_any %s", snd_strerror(rc)); term(); return false; } /* Set the desired hardware parameters. */ /* Interleaved mode */ rc=snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED); if (rc < 0) { ERROR_LOG(AUDIO, "ALSA: Error:snd_pcm_hw_params_set_access %s", snd_strerror(rc)); term(); return false; } /* Signed 16-bit little-endian format */ rc=snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE); if (rc < 0) { ERROR_LOG(AUDIO, "ALSA: Error:snd_pcm_hw_params_set_format %s", snd_strerror(rc)); term(); return false; } /* Two channels (stereo) */ rc=snd_pcm_hw_params_set_channels(handle, params, 2); if (rc < 0) { ERROR_LOG(AUDIO, "ALSA: Error:snd_pcm_hw_params_set_channels %s", snd_strerror(rc)); term(); return false; } // 44100 samples/second rc = snd_pcm_hw_params_set_rate(handle, params, 44100, 0); if (rc < 0) { ERROR_LOG(AUDIO, "ALSA: Error:snd_pcm_hw_params_set_rate %s", snd_strerror(rc)); term(); return false; } // Period size (512) period_size = std::min(SAMPLE_COUNT, (u32)config::AudioBufferSize / 4); rc = snd_pcm_hw_params_set_period_size_near(handle, params, &period_size, nullptr); if (rc < 0) { ERROR_LOG(AUDIO, "ALSA: Error:snd_pcm_hw_params_set_periods_near %s", snd_strerror(rc)); term(); return false; } INFO_LOG(AUDIO, "ALSA: period size set to %zd", (size_t)period_size); // Sample buffer size buffer_size = config::AudioBufferSize; rc = snd_pcm_hw_params_set_buffer_size_near(handle, params, &buffer_size); if (rc < 0) { ERROR_LOG(AUDIO, "ALSA: Error:snd_pcm_hw_params_set_buffer_size_near %s", snd_strerror(rc)); term(); return false; } INFO_LOG(AUDIO, "ALSA: buffer size set to %ld", buffer_size); /* Write the parameters to the driver */ rc = snd_pcm_hw_params(handle, params); if (rc < 0) { ERROR_LOG(AUDIO, "ALSA: Unable to set hw parameters: %s", snd_strerror(rc)); term(); return false; } return true; } bool initRecord(u32 sampling_freq) override { int err; if ((err = snd_pcm_open(&handle_record, "default", SND_PCM_STREAM_CAPTURE, 0)) < 0) { ERROR_LOG(AUDIO, "ALSA: Cannot open default audio capture device: %s", snd_strerror(err)); return false; } snd_pcm_hw_params_t *hw_params; if ((err = snd_pcm_hw_params_malloc(&hw_params)) < 0) { ERROR_LOG(AUDIO, "ALSA: Cannot allocate hardware parameter structure: %s", snd_strerror(err)); snd_pcm_close(handle_record); return false; } if ((err = snd_pcm_hw_params_any(handle_record, hw_params)) < 0) { ERROR_LOG(AUDIO, "ALSA: Cannot initialize hardware parameter structure: %s", snd_strerror(err)); snd_pcm_hw_params_free(hw_params); snd_pcm_close(handle_record); return false; } if ((err = snd_pcm_hw_params_set_access(handle_record, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) { ERROR_LOG(AUDIO, "ALSA: Cannot set access type: %s\n", snd_strerror(err)); snd_pcm_hw_params_free(hw_params); snd_pcm_close(handle_record); return false; } if ((err = snd_pcm_hw_params_set_format(handle_record, hw_params, SND_PCM_FORMAT_S16_LE)) < 0) { ERROR_LOG(AUDIO, "ALSA: Cannot set sample format: %s", snd_strerror(err)); snd_pcm_hw_params_free(hw_params); snd_pcm_close(handle_record); return false; } if ((err = snd_pcm_hw_params_set_rate(handle_record, hw_params, sampling_freq, 0)) < 0) { ERROR_LOG(AUDIO, "ALSA: Cannot set sample rate to %d Hz: %s", sampling_freq, snd_strerror(err)); snd_pcm_hw_params_free(hw_params); snd_pcm_close(handle_record); return false; } if ((err = snd_pcm_hw_params_set_channels(handle_record, hw_params, 1)) < 0) { ERROR_LOG(AUDIO, "ALSA: Cannot set channel count: %s", snd_strerror(err)); snd_pcm_hw_params_free(hw_params); snd_pcm_close(handle_record); return false; } if ((err = snd_pcm_hw_params(handle_record, hw_params)) < 0) { ERROR_LOG(AUDIO, "ALSA: Cannot set parameters: %s", snd_strerror(err)); snd_pcm_hw_params_free(hw_params); snd_pcm_close(handle_record); return false; } snd_pcm_hw_params_free(hw_params); snd_pcm_nonblock(handle_record, 1); if ((err = snd_pcm_prepare(handle_record)) < 0) { ERROR_LOG(AUDIO, "ALSA: Cannot prepare device: %s", snd_strerror(err)); snd_pcm_close(handle_record); return false; } INFO_LOG(AUDIO, "ALSA: Successfully initialized capture device"); return true; } void termRecord() override { snd_pcm_close(handle_record); } u32 record(void* frame, u32 samples) override { int err = snd_pcm_readi(handle_record, frame, samples); if (err < (int)samples) { if (err < 0) { DEBUG_LOG(AUDIO, "ALSA: Recording error: %s", snd_strerror(err)); err = 0; err = snd_pcm_prepare(handle_record); } u8 *buffer = (u8 *)frame + err; memset(buffer, 0, (samples - err) * 2); } return err; } u32 push(const void* frame, u32 samples, bool wait) override { if (wait != pcm_blocking) { snd_pcm_nonblock(handle, wait ? 0 : 1); pcm_blocking = wait; } int rc = snd_pcm_writei(handle, frame, samples); if (rc < 0) { snd_pcm_recover(handle, rc, 1); if (rc == -EPIPE) { // EPIPE means underrun // Write some silence then our samples const size_t silence_size = buffer_size - samples; void *silence = alloca(silence_size * 4); memset(silence, 0, silence_size * 4); snd_pcm_writei(handle, silence, silence_size); snd_pcm_writei(handle, frame, samples); } } return 1; } void term() override { snd_pcm_drop(handle); snd_pcm_close(handle); } std::vector getDeviceList() { std::vector result; char **hints; int err = snd_device_name_hint(-1, "pcm", (void***)&hints); // Error initializing ALSA if (err != 0) return result; // special value to automatically detect on initialization result.emplace_back("auto"); char** n = hints; while (*n != NULL) { // Get the type (NULL/Input/Output) char *type = snd_device_name_get_hint(*n, "IOID"); char *name = snd_device_name_get_hint(*n, "NAME"); if (name != NULL) { // We only want output or special devices (like "default" or "pulse") // TODO Only those with type == NULL? if (type == NULL || strcmp(type, "Output") == 0) { // TODO Check if device works (however we need to hash the resulting list then) /*snd_pcm_t *handle; int rc = snd_pcm_open(&handle, name, SND_PCM_STREAM_PLAYBACK, 0); if (rc == 0) { result.push_back(name); snd_pcm_close(handle); } */ result.emplace_back(name); } } if (type != NULL) free(type); if (name != NULL) free(name); n++; } snd_device_name_free_hint((void**)hints); return result; } Option* getOptions(int *count) override { *count = 1; static Option result; result.name = "device"; result.caption = "Device"; result.type = Option::list; result.values = getDeviceList(); return &result; } }; static AlsaAudioBackend alsaBackend; #endif