mirror of
https://github.com/moonlight-stream/moonlight-qt
synced 2024-12-16 14:22:28 +00:00
145 lines
4.2 KiB
C++
145 lines
4.2 KiB
C++
#include "slaud.h"
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#include <SDL.h>
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SLAudioRenderer::SLAudioRenderer()
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: m_AudioContext(nullptr),
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m_AudioStream(nullptr),
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m_AudioBuffer(nullptr)
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{
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SLAudio_SetLogFunction(SLAudioRenderer::slLogCallback, nullptr);
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}
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bool SLAudioRenderer::prepareForPlayback(const OPUS_MULTISTREAM_CONFIGURATION* opusConfig)
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{
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m_AudioContext = SLAudio_CreateContext();
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if (m_AudioContext == nullptr) {
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SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
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"SLAudio_CreateContext() failed");
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return false;
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}
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// This number is pretty conservative (especially for surround), but
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// it's hard to avoid since we get crushed by CPU limitations.
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m_MaxQueuedAudioMs = 40 * opusConfig->channelCount / 2;
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m_AudioBufferSize = opusConfig->samplesPerFrame * sizeof(short) * opusConfig->channelCount;
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m_AudioStream = SLAudio_CreateStream(m_AudioContext,
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opusConfig->sampleRate,
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opusConfig->channelCount,
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m_AudioBufferSize,
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1);
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if (m_AudioStream == nullptr) {
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SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
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"SLAudio_CreateStream() failed");
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return false;
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}
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SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
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"Using SLAudio renderer with %d samples per frame",
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opusConfig->samplesPerFrame);
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return true;
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}
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void SLAudioRenderer::remapChannels(POPUS_MULTISTREAM_CONFIGURATION opusConfig) {
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OPUS_MULTISTREAM_CONFIGURATION originalConfig = *opusConfig;
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// The Moonlight's default channel order is FL,FR,C,LFE,RL,RR,SL,SR
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// SLAudio expects FL,C,FR,RL,RR,(SL,SR),LFE for 5.1/7.1 so we swap the channels around to match
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if (opusConfig->channelCount == 3 || opusConfig->channelCount >= 6) {
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// Swap FR and C
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opusConfig->mapping[1] = originalConfig.mapping[2];
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opusConfig->mapping[2] = originalConfig.mapping[1];
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}
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if (opusConfig->channelCount >= 6) {
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// SLAudio expects the LFE channel at the end
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opusConfig->mapping[opusConfig->channelCount - 1] = originalConfig.mapping[3];
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// Slide the other channels down
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memcpy(&opusConfig->mapping[3],
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&originalConfig.mapping[4],
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opusConfig->channelCount - 4);
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}
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}
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void* SLAudioRenderer::getAudioBuffer(int* size)
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{
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SDL_assert(*size == m_AudioBufferSize);
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if (m_AudioBuffer == nullptr) {
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m_AudioBuffer = SLAudio_BeginFrame(m_AudioStream);
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}
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return m_AudioBuffer;
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}
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SLAudioRenderer::~SLAudioRenderer()
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{
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if (m_AudioBuffer != nullptr) {
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memset(m_AudioBuffer, 0, m_AudioBufferSize);
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SLAudio_SubmitFrame(m_AudioStream);
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}
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if (m_AudioStream != nullptr) {
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SLAudio_FreeStream(m_AudioStream);
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}
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if (m_AudioContext != nullptr) {
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SLAudio_FreeContext(m_AudioContext);
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}
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}
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bool SLAudioRenderer::submitAudio(int bytesWritten)
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{
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if (bytesWritten == 0) {
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// This buffer will be reused next time
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return true;
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}
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if (LiGetPendingAudioDuration() < m_MaxQueuedAudioMs) {
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SLAudio_SubmitFrame(m_AudioStream);
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m_AudioBuffer = nullptr;
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}
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else {
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SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
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"Too many queued audio frames: %d",
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LiGetPendingAudioFrames());
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}
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return true;
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}
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int SLAudioRenderer::getCapabilities()
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{
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return CAPABILITY_SLOW_OPUS_DECODER | CAPABILITY_SUPPORTS_ARBITRARY_AUDIO_DURATION;
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}
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void SLAudioRenderer::slLogCallback(void*, ESLAudioLog logLevel, const char *message)
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{
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SDL_LogPriority priority;
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switch (logLevel)
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{
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case k_ESLAudioLogError:
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priority = SDL_LOG_PRIORITY_ERROR;
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break;
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case k_ESLAudioLogWarning:
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priority = SDL_LOG_PRIORITY_WARN;
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break;
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case k_ESLAudioLogInfo:
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priority = SDL_LOG_PRIORITY_INFO;
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break;
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default:
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case k_ESLAudioLogDebug:
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priority = SDL_LOG_PRIORITY_DEBUG;
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break;
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}
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SDL_LogMessage(SDL_LOG_CATEGORY_APPLICATION,
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priority,
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"SLAudio: %s",
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message);
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}
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