moonlight-qt/app/streaming/audio/renderers/slaud.cpp
2020-04-03 19:04:11 -07:00

145 lines
4.2 KiB
C++

#include "slaud.h"
#include <SDL.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 * sizeof(short) * opusConfig->channelCount;
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;
}
int SLAudioRenderer::getCapabilities()
{
return CAPABILITY_SLOW_OPUS_DECODER | CAPABILITY_SUPPORTS_ARBITRARY_AUDIO_DURATION;
}
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);
}