// Submits any buffers that have currently been queued,
// assuming they are needed based on current queue depth
void sound_xaudio2::submit_needed()
{
XAUDIO2_VOICE_STATE state;
m_sourceVoice->GetState(&state, XAUDIO2_VOICE_NOSAMPLESPLAYED);
std::lock_guard<std::mutex> lock(m_buffer_lock);
// If we have buffers queued into XAudio and our current in-memory buffer
// isn't yet full, there's no need to submit it
if (state.BuffersQueued >= 1 && m_queue.empty())
return;
// We do however want to achieve some kind of minimal latency, so if the queued buffers
// are greater than 2, flush them to re-sync the audio
if (state.BuffersQueued > 2)
{
m_sourceVoice->FlushSourceBuffers();
m_overflows++;
}
// Roll the buffer
roll_buffer();
// Submit the next buffer
submit_next_queued();
}
int sound_xaudio2::init(osd_options const &options)
{
HRESULT result;
WAVEFORMATEX format = {0};
auto init_start = std::chrono::system_clock::now();
std::chrono::milliseconds init_time;
CoInitializeEx(nullptr, COINIT_MULTITHREADED);
// Make sure our XAudio2Create entrypoint is bound
if (!XAudio2Create)
{
osd_printf_error("Could not find XAudio2. Please try to reinstall DirectX runtime package.\n");
return 1;
}
// Create the IXAudio2 object
HR_GOERR(this->XAudio2Create(m_xAudio2.GetAddressOf(), 0, XAUDIO2_DEFAULT_PROCESSOR));
// make a format description for what we want
format.wBitsPerSample = 16;
format.wFormatTag = WAVE_FORMAT_PCM;
format.nChannels = 2;
format.nSamplesPerSec = sample_rate();
format.nBlockAlign = format.wBitsPerSample * format.nChannels / 8;
format.nAvgBytesPerSec = format.nSamplesPerSec * format.nBlockAlign;
m_sample_bytes = format.nBlockAlign;
// Create the buffers
create_buffers(format);
// Initialize our events
m_hEventBufferCompleted = CreateEvent(nullptr, FALSE, FALSE, nullptr);
m_hEventDataAvailable = CreateEvent(nullptr, FALSE, FALSE, nullptr);
m_hEventExiting = CreateEvent(nullptr, FALSE, FALSE, nullptr);
// create the voices and start them
HR_GOERR(create_voices(format));
HR_GOERR(m_sourceVoice->Start());
// Start the thread listening
m_audioThread = std::thread([](sound_xaudio2* self) { self->process_audio(); }, this);
init_time = std::chrono::duration_cast<std::chrono::milliseconds>(std::chrono::system_clock::now() - init_start);
osd_printf_verbose("Sound: XAudio2 initialized. %d ms.\n", static_cast<int>(init_time.count()));
m_initialized = TRUE;
return 0;
Error:
this->exit();
return 1;
}
int sound_xaudio2::init(osd_options const &options)
{
HRESULT result;
// Make sure our XAudio2Create entrypoint is bound
int status = XAudio2Create.initialize();
if (status != 0)
{
osd_printf_error("Could not find XAudio2 library\n");
return 1;
}
// Create the IXAudio2 object
IXAudio2 *temp_xaudio2 = nullptr;
HR_RET1(this->XAudio2Create(&temp_xaudio2, 0, XAUDIO2_DEFAULT_PROCESSOR));
m_xAudio2 = xaudio2_ptr(temp_xaudio2);
// make a format description for what we want
WAVEFORMATEX format = { 0 };
format.wBitsPerSample = 16;
format.wFormatTag = WAVE_FORMAT_PCM;
format.nChannels = 2;
format.nSamplesPerSec = sample_rate();
format.nBlockAlign = format.wBitsPerSample * format.nChannels / 8;
format.nAvgBytesPerSec = format.nSamplesPerSec * format.nBlockAlign;
m_sample_bytes = format.nBlockAlign;
// Create the buffers
create_buffers(format);
// Initialize our events
m_hEventBufferCompleted = CreateEvent(nullptr, FALSE, FALSE, nullptr);
m_hEventDataAvailable = CreateEvent(nullptr, FALSE, FALSE, nullptr);
m_hEventExiting = CreateEvent(nullptr, FALSE, FALSE, nullptr);
// create the voices and start them
HR_RET1(create_voices(format));
HR_RET1(m_sourceVoice->Start());
// Start the thread listening
m_audioThread = std::thread([](sound_xaudio2* self) { self->process_audio(); }, this);
osd_printf_verbose("Sound: XAudio2 initialized\n");
return 0;
}
// Submits any buffers that have currently been queued,
// assuming they are needed based on current queue depth
void sound_xaudio2::submit_needed()
{
XAUDIO2_VOICE_STATE state;
m_sourceVoice->GetState(&state, XAUDIO2_VOICE_NOSAMPLESPLAYED);
// If we have a buffer on the queue, no reason to submit
if (state.BuffersQueued >= 1)
return;
std::lock_guard<std::mutex> lock(m_buffer_lock);
// Roll the buffer
roll_buffer();
// Submit the next buffer
submit_next_queued();
}
void sound_xaudio2::set_mastervolume(int attenuation)
{
assert(m_sourceVoice);
HRESULT result;
// clamp the attenuation to 0-32 range
attenuation = MAX(MIN(attenuation, 0), -32);
// Ranges from 1.0 to XAUDIO2_MAX_VOLUME_LEVEL indicate additional gain
// Ranges from 0 to 1.0 indicate a reduced volume level
// 0 indicates silence
// We only support a reduction from 1.0, so we generate values in the range 0.0 to 1.0
float scaledVolume = (32.0f + attenuation) / 32.0f;
// set the master volume
HR_RETV(m_sourceVoice->SetVolume(scaledVolume));
}
void sound_xaudio2::exit()
{
// Wait on processing thread to end
SetEvent(m_hEventExiting);
m_audioThread.join();
CloseHandle(m_hEventBufferCompleted);
CloseHandle(m_hEventDataAvailable);
CloseHandle(m_hEventExiting);
m_sourceVoice.reset();
m_masterVoice.reset();
m_xAudio2.reset();
m_buffer.reset();
m_buffer_pool.reset();
if (m_overflows != 0 || m_underflows != 0)
osd_printf_verbose("Sound: overflows=%u, underflows=%u\n", m_overflows, m_underflows);
osd_printf_verbose("Sound: XAudio2 deinitialized\n");
}
void sound_xaudio2::submit_buffer(std::unique_ptr<BYTE[]> audioData, DWORD audioLength) const
{
assert(audioLength != 0);
XAUDIO2_BUFFER buf = { 0 };
buf.AudioBytes = audioLength;
buf.pAudioData = audioData.get();
buf.PlayBegin = 0;
buf.PlayLength = audioLength / m_sample_bytes;
buf.Flags = XAUDIO2_END_OF_STREAM;
buf.pContext = audioData.get();
HRESULT result;
if (FAILED(result = m_sourceVoice->SubmitSourceBuffer(&buf)))
{
osd_printf_verbose("Sound: XAudio2 failed to submit source buffer (non-fatal). Error: 0x%X\n", static_cast<unsigned int>(result));
m_buffer_pool->return_to_pool(audioData.release());
return;
}
// If we succeeded, relinquish the buffer allocation to the XAudio2 runtime
// The buffer will be freed on the OnBufferCompleted callback
audioData.release();
}
