Files
moonlight-qt/app/streaming/audio/renderers/slaud.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

148 lines
4.2 KiB
C++

#include "slaud.h"
#include "SDL_compat.h"
SLAudioRenderer::SLAudioRenderer()
: m_AudioContext(nullptr),
m_AudioStream(nullptr),
m_AudioBuffer(nullptr)
{
SLAudio_SetLogFunction(SLAudioRenderer::slLogCallback, nullptr);
}
bool SLAudioRenderer::prepareForPlayback(const OPUS_MULTISTREAM_CONFIGURATION* opusConfig)
{
m_AudioContext = SLAudio_CreateContext();
if (m_AudioContext == nullptr) {
SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
"SLAudio_CreateContext() failed");
return false;
}
// This number is pretty conservative (especially for surround), but
// it's hard to avoid since we get crushed by CPU limitations.
m_MaxQueuedAudioMs = 40 * opusConfig->channelCount / 2;
m_AudioBufferSize = opusConfig->samplesPerFrame *
opusConfig->channelCount *
getAudioBufferSampleSize();
m_AudioStream = SLAudio_CreateStream(m_AudioContext,
opusConfig->sampleRate,
opusConfig->channelCount,
m_AudioBufferSize,
1);
if (m_AudioStream == nullptr) {
SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
"SLAudio_CreateStream() failed");
return false;
}
SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
"Using SLAudio renderer with %d samples per frame",
opusConfig->samplesPerFrame);
return true;
}
void SLAudioRenderer::remapChannels(POPUS_MULTISTREAM_CONFIGURATION opusConfig) {
OPUS_MULTISTREAM_CONFIGURATION originalConfig = *opusConfig;
// The Moonlight's default channel order is FL,FR,C,LFE,RL,RR,SL,SR
// SLAudio expects FL,C,FR,RL,RR,(SL,SR),LFE for 5.1/7.1 so we swap the channels around to match
if (opusConfig->channelCount == 3 || opusConfig->channelCount >= 6) {
// Swap FR and C
opusConfig->mapping[1] = originalConfig.mapping[2];
opusConfig->mapping[2] = originalConfig.mapping[1];
}
if (opusConfig->channelCount >= 6) {
// SLAudio expects the LFE channel at the end
opusConfig->mapping[opusConfig->channelCount - 1] = originalConfig.mapping[3];
// Slide the other channels down
memcpy(&opusConfig->mapping[3],
&originalConfig.mapping[4],
opusConfig->channelCount - 4);
}
}
void* SLAudioRenderer::getAudioBuffer(int* size)
{
SDL_assert(*size == m_AudioBufferSize);
if (m_AudioBuffer == nullptr) {
m_AudioBuffer = SLAudio_BeginFrame(m_AudioStream);
}
return m_AudioBuffer;
}
SLAudioRenderer::~SLAudioRenderer()
{
if (m_AudioBuffer != nullptr) {
memset(m_AudioBuffer, 0, m_AudioBufferSize);
SLAudio_SubmitFrame(m_AudioStream);
}
if (m_AudioStream != nullptr) {
SLAudio_FreeStream(m_AudioStream);
}
if (m_AudioContext != nullptr) {
SLAudio_FreeContext(m_AudioContext);
}
}
bool SLAudioRenderer::submitAudio(int bytesWritten)
{
if (bytesWritten == 0) {
// This buffer will be reused next time
return true;
}
if (LiGetPendingAudioDuration() < m_MaxQueuedAudioMs) {
SLAudio_SubmitFrame(m_AudioStream);
m_AudioBuffer = nullptr;
}
else {
SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
"Too many queued audio frames: %d",
LiGetPendingAudioFrames());
}
return true;
}
IAudioRenderer::AudioFormat SLAudioRenderer::getAudioBufferFormat()
{
return AudioFormat::Sint16NE;
}
void SLAudioRenderer::slLogCallback(void*, ESLAudioLog logLevel, const char *message)
{
SDL_LogPriority priority;
switch (logLevel)
{
case k_ESLAudioLogError:
priority = SDL_LOG_PRIORITY_ERROR;
break;
case k_ESLAudioLogWarning:
priority = SDL_LOG_PRIORITY_WARN;
break;
case k_ESLAudioLogInfo:
priority = SDL_LOG_PRIORITY_INFO;
break;
default:
case k_ESLAudioLogDebug:
priority = SDL_LOG_PRIORITY_DEBUG;
break;
}
SDL_LogMessage(SDL_LOG_CATEGORY_APPLICATION,
priority,
"SLAudio: %s",
message);
}