moonlight-qt/app/streaming/audio/renderers/sdlaud.cpp
Cameron Gutman 7d544c1ce4 Move audio capabilities out of the renderer classes
Since the removal of libsoundio, all renderers support arbitrary audio duration, so we can avoid having to start an audio session just to query capabilities.
2025-10-25 19:17:38 -05:00

138 lines
4.2 KiB
C++

#include "sdl.h"
#include <Limelight.h>
SdlAudioRenderer::SdlAudioRenderer()
: m_AudioDevice(0),
m_AudioBuffer(nullptr)
{
SDL_assert(!SDL_WasInit(SDL_INIT_AUDIO));
if (SDL_InitSubSystem(SDL_INIT_AUDIO) != 0) {
SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
"SDL_InitSubSystem(SDL_INIT_AUDIO) failed: %s",
SDL_GetError());
SDL_assert(SDL_WasInit(SDL_INIT_AUDIO));
}
}
bool SdlAudioRenderer::prepareForPlayback(const OPUS_MULTISTREAM_CONFIGURATION* opusConfig)
{
SDL_AudioSpec want, have;
SDL_zero(want);
want.freq = opusConfig->sampleRate;
want.format = AUDIO_F32SYS;
want.channels = opusConfig->channelCount;
// On PulseAudio systems, setting a value too small can cause underruns for other
// applications sharing this output device. We impose a floor of 480 samples (10 ms)
// to mitigate this issue. Otherwise, we will buffer up to 3 frames of audio which
// is 15 ms at regular 5 ms frames and 30 ms at 10 ms frames for slow connections.
// The buffering helps avoid audio underruns due to network jitter.
want.samples = SDL_max(480, opusConfig->samplesPerFrame * 3);
m_FrameSize = opusConfig->samplesPerFrame *
opusConfig->channelCount *
getAudioBufferSampleSize();
m_AudioDevice = SDL_OpenAudioDevice(NULL, 0, &want, &have, 0);
if (m_AudioDevice == 0) {
SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
"Failed to open audio device: %s",
SDL_GetError());
return false;
}
m_AudioBuffer = SDL_malloc(m_FrameSize);
if (m_AudioBuffer == nullptr) {
SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
"Failed to allocate audio buffer");
return false;
}
SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
"Desired audio buffer: %u samples (%u bytes)",
want.samples,
want.samples * want.channels * getAudioBufferSampleSize());
SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
"Obtained audio buffer: %u samples (%u bytes)",
have.samples,
have.size);
SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
"SDL audio driver: %s",
SDL_GetCurrentAudioDriver());
// Start playback
SDL_PauseAudioDevice(m_AudioDevice, 0);
return true;
}
SdlAudioRenderer::~SdlAudioRenderer()
{
if (m_AudioDevice != 0) {
// Stop playback
SDL_PauseAudioDevice(m_AudioDevice, 1);
SDL_CloseAudioDevice(m_AudioDevice);
}
if (m_AudioBuffer != nullptr) {
SDL_free(m_AudioBuffer);
}
SDL_QuitSubSystem(SDL_INIT_AUDIO);
SDL_assert(!SDL_WasInit(SDL_INIT_AUDIO));
}
void* SdlAudioRenderer::getAudioBuffer(int*)
{
return m_AudioBuffer;
}
bool SdlAudioRenderer::submitAudio(int bytesWritten)
{
if (bytesWritten == 0) {
// Nothing to do
return true;
}
// Don't queue if there's already more than 30 ms of audio data waiting
// in Moonlight's audio queue.
if (LiGetPendingAudioDuration() > 30) {
return true;
}
// Provide backpressure on the queue to ensure too many frames don't build up
// in SDL's audio queue, but don't wait forever to avoid a deadlock if the
// audio device fails.
for (int i = 0; i < 100; i++) {
// Our device may enter a permanent error status upon removal, so we need
// to recreate the audio device to pick up the new default audio device.
if (SDL_GetAudioDeviceStatus(m_AudioDevice) == SDL_AUDIO_STOPPED) {
return false;
}
// Only queue more samples where there are 10 frames or less in SDL's queue
if (SDL_GetQueuedAudioSize(m_AudioDevice) / m_FrameSize <= 10) {
break;
}
SDL_Delay(1);
}
if (SDL_QueueAudio(m_AudioDevice, m_AudioBuffer, bytesWritten) < 0) {
SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
"Failed to queue audio sample: %s",
SDL_GetError());
}
return true;
}
IAudioRenderer::AudioFormat SdlAudioRenderer::getAudioBufferFormat()
{
return AudioFormat::Float32NE;
}