#include "slaud.h" #include 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); }