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;
}
