bool PulseAudio::PulseInit() { m_pa_error = 0; m_pa_connected = 0; // create pulseaudio main loop and context // also register the async state callback which is called when the connection to the pa server has changed m_pa_ml = pa_mainloop_new(); m_pa_mlapi = pa_mainloop_get_api(m_pa_ml); m_pa_ctx = pa_context_new(m_pa_mlapi, "dolphin-emu"); m_pa_error = pa_context_connect(m_pa_ctx, nullptr, PA_CONTEXT_NOFLAGS, nullptr); pa_context_set_state_callback(m_pa_ctx, StateCallback, this); // wait until we're connected to the pulseaudio server while (m_pa_connected == 0 && m_pa_error >= 0) m_pa_error = pa_mainloop_iterate(m_pa_ml, 1, nullptr); if (m_pa_connected == 2 || m_pa_error < 0) { ERROR_LOG(AUDIO, "PulseAudio failed to initialize: %s", pa_strerror(m_pa_error)); return false; } // create a new audio stream with our sample format // also connect the callbacks for this stream pa_sample_spec ss; ss.format = PA_SAMPLE_S16LE; ss.channels = 2; ss.rate = m_mixer->GetSampleRate(); m_pa_s = pa_stream_new(m_pa_ctx, "Playback", &ss, nullptr); pa_stream_set_write_callback(m_pa_s, WriteCallback, this); pa_stream_set_underflow_callback(m_pa_s, UnderflowCallback, this); // connect this audio stream to the default audio playback // limit buffersize to reduce latency m_pa_ba.fragsize = -1; m_pa_ba.maxlength = -1; // max buffer, so also max latency m_pa_ba.minreq = -1; // don't read every byte, try to group them _a bit_ m_pa_ba.prebuf = -1; // start as early as possible m_pa_ba.tlength = BUFFER_SIZE; // designed latency, only change this flag for low latency output pa_stream_flags flags = pa_stream_flags(PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_ADJUST_LATENCY | PA_STREAM_AUTO_TIMING_UPDATE); m_pa_error = pa_stream_connect_playback(m_pa_s, nullptr, &m_pa_ba, flags, nullptr, nullptr); if (m_pa_error < 0) { ERROR_LOG(AUDIO, "PulseAudio failed to initialize: %s", pa_strerror(m_pa_error)); return false; } INFO_LOG(AUDIO, "Pulse successfully initialized"); return true; }
Error AudioDriverPulseAudio::init_device() { // If there is a specified device check that it is really present if (device_name != "Default") { Array list = get_device_list(); if (list.find(device_name) == -1) { device_name = "Default"; new_device = "Default"; } } // Detect the amount of channels PulseAudio is using // Note: If using an even amount of channels (2, 4, etc) channels and pa_map.channels will be equal, // if not then pa_map.channels will have the real amount of channels PulseAudio is using and channels // will have the amount of channels Godot is using (in this case it's pa_map.channels + 1) detect_channels(); switch (pa_map.channels) { case 1: // Mono case 3: // Surround 2.1 case 5: // Surround 5.0 case 7: // Surround 7.0 channels = pa_map.channels + 1; break; case 2: // Stereo case 4: // Surround 4.0 case 6: // Surround 5.1 case 8: // Surround 7.1 channels = pa_map.channels; break; default: WARN_PRINTS("PulseAudio: Unsupported number of channels: " + itos(pa_map.channels)); pa_channel_map_init_stereo(&pa_map); channels = 2; break; } int latency = GLOBAL_DEF_RST("audio/output_latency", DEFAULT_OUTPUT_LATENCY); buffer_frames = closest_power_of_2(latency * mix_rate / 1000); pa_buffer_size = buffer_frames * pa_map.channels; print_verbose("PulseAudio: detected " + itos(pa_map.channels) + " channels"); print_verbose("PulseAudio: audio buffer frames: " + itos(buffer_frames) + " calculated latency: " + itos(buffer_frames * 1000 / mix_rate) + "ms"); pa_sample_spec spec; spec.format = PA_SAMPLE_S16LE; spec.channels = pa_map.channels; spec.rate = mix_rate; pa_str = pa_stream_new(pa_ctx, "Sound", &spec, &pa_map); if (pa_str == NULL) { ERR_PRINTS("PulseAudio: pa_stream_new error: " + String(pa_strerror(pa_context_errno(pa_ctx)))); ERR_FAIL_V(ERR_CANT_OPEN); } pa_buffer_attr attr; // set to appropriate buffer length (in bytes) from global settings // Note: PulseAudio defaults to 4 fragments, which means that the actual // latency is tlength / fragments. It seems that the PulseAudio has no way // to get the fragments number so we're hardcoding this to the default of 4 const int fragments = 4; attr.tlength = pa_buffer_size * sizeof(int16_t) * fragments; // set them to be automatically chosen attr.prebuf = (uint32_t)-1; attr.maxlength = (uint32_t)-1; attr.minreq = (uint32_t)-1; const char *dev = device_name == "Default" ? NULL : device_name.utf8().get_data(); pa_stream_flags flags = pa_stream_flags(PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_ADJUST_LATENCY | PA_STREAM_AUTO_TIMING_UPDATE); int error_code = pa_stream_connect_playback(pa_str, dev, &attr, flags, NULL, NULL); ERR_FAIL_COND_V(error_code < 0, ERR_CANT_OPEN); samples_in.resize(buffer_frames * channels); samples_out.resize(pa_buffer_size); // Reset audio input to keep synchronisation. input_position = 0; input_size = 0; return OK; }