Error AudioDriverPulseAudio::init() {
active = false;
thread_exited = false;
exit_thread = false;
mix_rate = GLOBAL_DEF_RST("audio/mix_rate", DEFAULT_MIX_RATE);
pa_ml = pa_mainloop_new();
ERR_FAIL_COND_V(pa_ml == NULL, ERR_CANT_OPEN);
pa_ctx = pa_context_new(pa_mainloop_get_api(pa_ml), "Godot");
ERR_FAIL_COND_V(pa_ctx == NULL, ERR_CANT_OPEN);
pa_ready = 0;
pa_context_set_state_callback(pa_ctx, pa_state_cb, (void *)this);
int ret = pa_context_connect(pa_ctx, NULL, PA_CONTEXT_NOFLAGS, NULL);
if (ret < 0) {
if (pa_ctx) {
pa_context_unref(pa_ctx);
pa_ctx = NULL;
}
if (pa_ml) {
pa_mainloop_free(pa_ml);
pa_ml = NULL;
}
return ERR_CANT_OPEN;
}
while (pa_ready == 0) {
pa_mainloop_iterate(pa_ml, 1, NULL);
}
if (pa_ready < 0) {
if (pa_ctx) {
pa_context_disconnect(pa_ctx);
pa_context_unref(pa_ctx);
pa_ctx = NULL;
}
if (pa_ml) {
pa_mainloop_free(pa_ml);
pa_ml = NULL;
}
return ERR_CANT_OPEN;
}
Error err = init_device();
if (err == OK) {
mutex = Mutex::create();
thread = Thread::create(AudioDriverPulseAudio::thread_func, this);
}
return OK;
}
Error AudioDriverWASAPI::init() {
mix_rate = GLOBAL_DEF_RST("audio/mix_rate", DEFAULT_MIX_RATE);
Error err = init_render_device();
if (err != OK) {
ERR_PRINT("WASAPI: init_render_device error");
}
exit_thread = false;
thread_exited = false;
mutex = Mutex::create(true);
thread = Thread::create(thread_func, this);
return OK;
}
Error AudioDriverCoreAudio::init() {
mutex = Mutex::create();
AudioComponentDescription desc;
zeromem(&desc, sizeof(desc));
desc.componentType = kAudioUnitType_Output;
#ifdef OSX_ENABLED
desc.componentSubType = kAudioUnitSubType_HALOutput;
#else
desc.componentSubType = kAudioUnitSubType_RemoteIO;
#endif
desc.componentManufacturer = kAudioUnitManufacturer_Apple;
AudioComponent comp = AudioComponentFindNext(NULL, &desc);
ERR_FAIL_COND_V(comp == NULL, FAILED);
OSStatus result = AudioComponentInstanceNew(comp, &audio_unit);
ERR_FAIL_COND_V(result != noErr, FAILED);
#ifdef OSX_ENABLED
AudioObjectPropertyAddress prop;
prop.mSelector = kAudioHardwarePropertyDefaultOutputDevice;
prop.mScope = kAudioObjectPropertyScopeGlobal;
prop.mElement = kAudioObjectPropertyElementMaster;
result = AudioObjectAddPropertyListener(kAudioObjectSystemObject, &prop, &output_device_address_cb, this);
ERR_FAIL_COND_V(result != noErr, FAILED);
prop.mSelector = kAudioHardwarePropertyDefaultInputDevice;
result = AudioObjectAddPropertyListener(kAudioObjectSystemObject, &prop, &input_device_address_cb, this);
ERR_FAIL_COND_V(result != noErr, FAILED);
#endif
AudioStreamBasicDescription strdesc;
zeromem(&strdesc, sizeof(strdesc));
UInt32 size = sizeof(strdesc);
result = AudioUnitGetProperty(audio_unit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, kOutputBus, &strdesc, &size);
ERR_FAIL_COND_V(result != noErr, FAILED);
switch (strdesc.mChannelsPerFrame) {
case 2: // Stereo
case 4: // Surround 3.1
case 6: // Surround 5.1
case 8: // Surround 7.1
channels = strdesc.mChannelsPerFrame;
break;
default:
// Unknown number of channels, default to stereo
channels = 2;
break;
}
zeromem(&strdesc, sizeof(strdesc));
size = sizeof(strdesc);
result = AudioUnitGetProperty(audio_unit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, kInputBus, &strdesc, &size);
ERR_FAIL_COND_V(result != noErr, FAILED);
switch (strdesc.mChannelsPerFrame) {
case 1: // Mono
capture_channels = 1;
break;
case 2: // Stereo
capture_channels = 2;
break;
default:
// Unknown number of channels, default to stereo
capture_channels = 2;
break;
}
mix_rate = GLOBAL_DEF_RST("audio/mix_rate", DEFAULT_MIX_RATE);
zeromem(&strdesc, sizeof(strdesc));
strdesc.mFormatID = kAudioFormatLinearPCM;
strdesc.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked;
strdesc.mChannelsPerFrame = channels;
strdesc.mSampleRate = mix_rate;
strdesc.mFramesPerPacket = 1;
strdesc.mBitsPerChannel = 16;
strdesc.mBytesPerFrame = strdesc.mBitsPerChannel * strdesc.mChannelsPerFrame / 8;
strdesc.mBytesPerPacket = strdesc.mBytesPerFrame * strdesc.mFramesPerPacket;
result = AudioUnitSetProperty(audio_unit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, kOutputBus, &strdesc, sizeof(strdesc));
ERR_FAIL_COND_V(result != noErr, FAILED);
strdesc.mChannelsPerFrame = capture_channels;
result = AudioUnitSetProperty(audio_unit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, kInputBus, &strdesc, sizeof(strdesc));
ERR_FAIL_COND_V(result != noErr, FAILED);
int latency = GLOBAL_DEF_RST("audio/output_latency", DEFAULT_OUTPUT_LATENCY);
// Sample rate is independent of channels (ref: https://stackoverflow.com/questions/11048825/audio-sample-frequency-rely-on-channels)
buffer_frames = closest_power_of_2(latency * mix_rate / 1000);
#ifdef OSX_ENABLED
result = AudioUnitSetProperty(audio_unit, kAudioDevicePropertyBufferFrameSize, kAudioUnitScope_Global, kOutputBus, &buffer_frames, sizeof(UInt32));
ERR_FAIL_COND_V(result != noErr, FAILED);
#endif
unsigned int buffer_size = buffer_frames * channels;
samples_in.resize(buffer_size);
input_buf.resize(buffer_size);
input_buffer.resize(buffer_size * 8);
input_position = 0;
input_size = 0;
print_verbose("CoreAudio: detected " + itos(channels) + " channels");
print_verbose("CoreAudio: audio buffer frames: " + itos(buffer_frames) + " calculated latency: " + itos(buffer_frames * 1000 / mix_rate) + "ms");
AURenderCallbackStruct callback;
zeromem(&callback, sizeof(AURenderCallbackStruct));
callback.inputProc = &AudioDriverCoreAudio::output_callback;
callback.inputProcRefCon = this;
result = AudioUnitSetProperty(audio_unit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, kOutputBus, &callback, sizeof(callback));
ERR_FAIL_COND_V(result != noErr, FAILED);
zeromem(&callback, sizeof(AURenderCallbackStruct));
callback.inputProc = &AudioDriverCoreAudio::input_callback;
callback.inputProcRefCon = this;
result = AudioUnitSetProperty(audio_unit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, 0, &callback, sizeof(callback));
ERR_FAIL_COND_V(result != noErr, FAILED);
result = AudioUnitInitialize(audio_unit);
ERR_FAIL_COND_V(result != noErr, FAILED);
return OK;
}
void register_core_settings() {
//since in register core types, globals may not e present
GLOBAL_DEF_RST("network/limits/packet_peer_stream/max_buffer_po2", (16));
}
Error AudioDriverRtAudio::init() {
active = false;
mutex = Mutex::create(true);
dac = memnew(RtAudio);
ERR_EXPLAIN("Cannot initialize RtAudio audio driver: No devices present.")
ERR_FAIL_COND_V(dac->getDeviceCount() < 1, ERR_UNAVAILABLE);
// FIXME: Adapt to the OutputFormat -> SpeakerMode change
/*
String channels = GLOBAL_DEF_RST("audio/output","stereo");
if (channels=="5.1")
output_format=OUTPUT_5_1;
else if (channels=="quad")
output_format=OUTPUT_QUAD;
else if (channels=="mono")
output_format=OUTPUT_MONO;
else
output_format=OUTPUT_STEREO;
*/
RtAudio::StreamParameters parameters;
parameters.deviceId = dac->getDefaultOutputDevice();
RtAudio::StreamOptions options;
// set the desired numberOfBuffers
options.numberOfBuffers = 4;
parameters.firstChannel = 0;
mix_rate = GLOBAL_DEF_RST("audio/mix_rate", DEFAULT_MIX_RATE);
int latency = GLOBAL_DEF("audio/output_latency", DEFAULT_OUTPUT_LATENCY);
unsigned int buffer_frames = closest_power_of_2(latency * mix_rate / 1000);
print_verbose("Audio buffer frames: " + itos(buffer_frames) + " calculated latency: " + itos(buffer_frames * 1000 / mix_rate) + "ms");
short int tries = 2;
while (tries >= 0) {
switch (speaker_mode) {
case SPEAKER_MODE_STEREO: parameters.nChannels = 2; break;
case SPEAKER_SURROUND_51: parameters.nChannels = 6; break;
case SPEAKER_SURROUND_71: parameters.nChannels = 8; break;
};
try {
dac->openStream(¶meters, NULL, RTAUDIO_SINT32, mix_rate, &buffer_frames, &callback, this, &options);
active = true;
break;
} catch (RtAudioError) {
// try with less channels
ERR_PRINT("Unable to open audio, retrying with fewer channels...");
switch (speaker_mode) {
case SPEAKER_SURROUND_51: speaker_mode = SPEAKER_MODE_STEREO; break;
case SPEAKER_SURROUND_71: speaker_mode = SPEAKER_SURROUND_51; break;
}
tries--;
}
}
return active ? OK : ERR_UNAVAILABLE;
}
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;
}
Error AudioDriverCoreAudio::capture_init() {
AudioComponentDescription desc;
zeromem(&desc, sizeof(desc));
desc.componentType = kAudioUnitType_Output;
#ifdef OSX_ENABLED
desc.componentSubType = kAudioUnitSubType_HALOutput;
#else
desc.componentSubType = kAudioUnitSubType_RemoteIO;
#endif
desc.componentManufacturer = kAudioUnitManufacturer_Apple;
AudioComponent comp = AudioComponentFindNext(NULL, &desc);
ERR_FAIL_COND_V(comp == NULL, FAILED);
OSStatus result = AudioComponentInstanceNew(comp, &input_unit);
ERR_FAIL_COND_V(result != noErr, FAILED);
#ifdef OSX_ENABLED
AudioObjectPropertyAddress prop;
prop.mSelector = kAudioHardwarePropertyDefaultInputDevice;
prop.mScope = kAudioObjectPropertyScopeGlobal;
prop.mElement = kAudioObjectPropertyElementMaster;
result = AudioObjectAddPropertyListener(kAudioObjectSystemObject, &prop, &input_device_address_cb, this);
ERR_FAIL_COND_V(result != noErr, FAILED);
#endif
UInt32 flag = 1;
result = AudioUnitSetProperty(input_unit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, kInputBus, &flag, sizeof(flag));
ERR_FAIL_COND_V(result != noErr, FAILED);
flag = 0;
result = AudioUnitSetProperty(input_unit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, kOutputBus, &flag, sizeof(flag));
ERR_FAIL_COND_V(result != noErr, FAILED);
UInt32 size;
#ifdef OSX_ENABLED
AudioDeviceID deviceId;
size = sizeof(AudioDeviceID);
AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultInputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
result = AudioObjectGetPropertyData(kAudioObjectSystemObject, &property, 0, NULL, &size, &deviceId);
ERR_FAIL_COND_V(result != noErr, FAILED);
result = AudioUnitSetProperty(input_unit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &deviceId, sizeof(AudioDeviceID));
ERR_FAIL_COND_V(result != noErr, FAILED);
#endif
AudioStreamBasicDescription strdesc;
zeromem(&strdesc, sizeof(strdesc));
size = sizeof(strdesc);
result = AudioUnitGetProperty(input_unit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, kInputBus, &strdesc, &size);
ERR_FAIL_COND_V(result != noErr, FAILED);
switch (strdesc.mChannelsPerFrame) {
case 1: // Mono
capture_channels = 1;
break;
case 2: // Stereo
capture_channels = 2;
break;
default:
// Unknown number of channels, default to stereo
capture_channels = 2;
break;
}
mix_rate = GLOBAL_DEF_RST("audio/mix_rate", DEFAULT_MIX_RATE);
zeromem(&strdesc, sizeof(strdesc));
strdesc.mFormatID = kAudioFormatLinearPCM;
strdesc.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked;
strdesc.mChannelsPerFrame = capture_channels;
strdesc.mSampleRate = mix_rate;
strdesc.mFramesPerPacket = 1;
strdesc.mBitsPerChannel = 16;
strdesc.mBytesPerFrame = strdesc.mBitsPerChannel * strdesc.mChannelsPerFrame / 8;
strdesc.mBytesPerPacket = strdesc.mBytesPerFrame * strdesc.mFramesPerPacket;
result = AudioUnitSetProperty(input_unit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, kInputBus, &strdesc, sizeof(strdesc));
ERR_FAIL_COND_V(result != noErr, FAILED);
AURenderCallbackStruct callback;
zeromem(&callback, sizeof(AURenderCallbackStruct));
callback.inputProc = &AudioDriverCoreAudio::input_callback;
callback.inputProcRefCon = this;
result = AudioUnitSetProperty(input_unit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, kInputBus, &callback, sizeof(callback));
ERR_FAIL_COND_V(result != noErr, FAILED);
result = AudioUnitInitialize(input_unit);
ERR_FAIL_COND_V(result != noErr, FAILED);
return OK;
}
