mirror of
https://github.com/moonlight-stream/moonlight-qt
synced 2024-12-16 06:12:28 +00:00
187 lines
6.5 KiB
C++
187 lines
6.5 KiB
C++
#include "../session.h"
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#include "renderers/renderer.h"
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#ifdef HAVE_SOUNDIO
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#include "renderers/soundioaudiorenderer.h"
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#endif
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#ifdef HAVE_SLAUDIO
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#include "renderers/slaud.h"
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#endif
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#include "renderers/sdl.h"
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#include <Limelight.h>
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IAudioRenderer* Session::createAudioRenderer()
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{
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#if defined(HAVE_SOUNDIO)
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if (qgetenv("ML_AUDIO") == "SDL") {
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return new SdlAudioRenderer();
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}
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return new SoundIoAudioRenderer();
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#elif defined(HAVE_SLAUDIO)
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return new SLAudioRenderer();
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#else
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return new SdlAudioRenderer();
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#endif
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}
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bool Session::testAudio(int audioConfiguration)
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{
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IAudioRenderer* audioRenderer;
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audioRenderer = createAudioRenderer();
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if (audioRenderer == nullptr) {
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return false;
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}
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// Build a fake OPUS_MULTISTREAM_CONFIGURATION to give
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// the renderer the channel count and sample rate.
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OPUS_MULTISTREAM_CONFIGURATION opusConfig = {};
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opusConfig.sampleRate = 48000;
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switch (audioConfiguration)
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{
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case AUDIO_CONFIGURATION_STEREO:
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opusConfig.channelCount = 2;
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break;
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case AUDIO_CONFIGURATION_51_SURROUND:
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opusConfig.channelCount = 6;
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break;
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default:
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SDL_assert(false);
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return false;
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}
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bool ret = audioRenderer->prepareForPlayback(&opusConfig);
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delete audioRenderer;
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return ret;
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}
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int Session::arInit(int /* audioConfiguration */,
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const POPUS_MULTISTREAM_CONFIGURATION opusConfig,
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void* /* arContext */, int /* arFlags */)
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{
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int error;
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SDL_memcpy(&s_ActiveSession->m_AudioConfig, opusConfig, sizeof(*opusConfig));
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s_ActiveSession->m_OpusDecoder =
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opus_multistream_decoder_create(opusConfig->sampleRate,
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opusConfig->channelCount,
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opusConfig->streams,
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opusConfig->coupledStreams,
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opusConfig->mapping,
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&error);
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if (s_ActiveSession->m_OpusDecoder == NULL) {
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SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
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"Failed to create decoder: %d",
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error);
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return -1;
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}
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s_ActiveSession->m_AudioRenderer = s_ActiveSession->createAudioRenderer();
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if (!s_ActiveSession->m_AudioRenderer->prepareForPlayback(opusConfig)) {
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delete s_ActiveSession->m_AudioRenderer;
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opus_multistream_decoder_destroy(s_ActiveSession->m_OpusDecoder);
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return -2;
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}
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SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
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"Audio stream has %d channels",
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opusConfig->channelCount);
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return 0;
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}
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void Session::arCleanup()
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{
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delete s_ActiveSession->m_AudioRenderer;
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s_ActiveSession->m_AudioRenderer = nullptr;
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opus_multistream_decoder_destroy(s_ActiveSession->m_OpusDecoder);
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s_ActiveSession->m_OpusDecoder = nullptr;
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}
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void Session::arDecodeAndPlaySample(char* sampleData, int sampleLength)
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{
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int samplesDecoded;
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// Set this thread to high priority to reduce
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// the chance of missing our sample delivery time
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if (s_ActiveSession->m_AudioSampleCount == 0) {
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if (SDL_SetThreadPriority(SDL_THREAD_PRIORITY_HIGH) < 0) {
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SDL_LogWarn(SDL_LOG_CATEGORY_APPLICATION,
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"Unable to set audio thread to high priority: %s",
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SDL_GetError());
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}
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}
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// See if we need to drop this sample
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if (s_ActiveSession->m_DropAudioEndTime != 0) {
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if (SDL_TICKS_PASSED(SDL_GetTicks(), s_ActiveSession->m_DropAudioEndTime)) {
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// Avoid calling SDL_GetTicks() now
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s_ActiveSession->m_DropAudioEndTime = 0;
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SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
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"Audio drop window has ended");
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}
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else {
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// We're still in the drop window
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return;
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}
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}
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s_ActiveSession->m_AudioSampleCount++;
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if (s_ActiveSession->m_AudioRenderer != nullptr) {
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samplesDecoded = opus_multistream_decode(s_ActiveSession->m_OpusDecoder,
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(unsigned char*)sampleData,
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sampleLength,
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s_ActiveSession->m_OpusDecodeBuffer,
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SAMPLES_PER_FRAME,
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0);
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if (samplesDecoded > 0) {
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if (!s_ActiveSession->m_AudioRenderer->submitAudio(s_ActiveSession->m_OpusDecodeBuffer,
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static_cast<int>(
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sizeof(short) *
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samplesDecoded *
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s_ActiveSession->m_AudioConfig.channelCount))) {
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SDL_LogWarn(SDL_LOG_CATEGORY_APPLICATION,
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"Reinitializing audio renderer after failure");
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delete s_ActiveSession->m_AudioRenderer;
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s_ActiveSession->m_AudioRenderer = nullptr;
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}
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}
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}
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// Only try to recreate the audio renderer every 200 samples (1 second)
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// to avoid thrashing if the audio device is unavailable. It is
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// safe to reinitialize here because we can't be torn down while
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// the audio decoder/playback thread is still alive.
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if (s_ActiveSession->m_AudioRenderer == nullptr && (s_ActiveSession->m_AudioSampleCount % 200) == 0) {
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// Since we're doing this inline and audio initialization takes time, we need
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// to drop samples to account for the time we've spent blocking audio rendering
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// so we return to real-time playback and don't accumulate latency.
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Uint32 audioReinitStartTime = SDL_GetTicks();
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s_ActiveSession->m_AudioRenderer = s_ActiveSession->createAudioRenderer();
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if (!s_ActiveSession->m_AudioRenderer->prepareForPlayback(&s_ActiveSession->m_AudioConfig)) {
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delete s_ActiveSession->m_AudioRenderer;
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s_ActiveSession->m_AudioRenderer = nullptr;
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}
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Uint32 audioReinitStopTime = SDL_GetTicks();
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s_ActiveSession->m_DropAudioEndTime = audioReinitStopTime + (audioReinitStopTime - audioReinitStartTime);
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SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
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"Audio reinitialization took %d ms - starting drop window",
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audioReinitStopTime - audioReinitStartTime);
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}
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}
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