mirror of
https://github.com/moonlight-stream/moonlight-qt
synced 2024-11-16 16:27:59 +00:00
284 lines
8.8 KiB
C++
284 lines
8.8 KiB
C++
#include "session.hpp"
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#include <Limelight.h>
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#include <SDL.h>
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#define MAX_CHANNELS 6
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#define SAMPLES_PER_FRAME 240
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#define MIN_QUEUED_FRAMES 2
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#define MAX_QUEUED_FRAMES 4
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#define STOP_THE_WORLD_LIMIT 20
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#define DROP_RATIO_DENOM 32
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SDL_AudioDeviceID Session::s_AudioDevice;
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OpusMSDecoder* Session::s_OpusDecoder;
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short Session::s_OpusDecodeBuffer[MAX_CHANNELS * SAMPLES_PER_FRAME];
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int Session::s_ChannelCount;
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int Session::s_PendingDrops;
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int Session::s_PendingHardDrops;
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unsigned int Session::s_SampleIndex;
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Uint32 Session::s_BaselinePendingData;
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int Session::sdlDetermineAudioConfiguration()
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{
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SDL_AudioSpec want, have;
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SDL_AudioDeviceID dev;
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int ret;
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if (SDL_InitSubSystem(SDL_INIT_AUDIO) != 0) {
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SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
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"SDL_InitSubSystem(SDL_INIT_AUDIO) failed: %s",
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SDL_GetError());
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return AUDIO_CONFIGURATION_STEREO;
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}
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SDL_zero(want);
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want.freq = 48000;
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want.format = AUDIO_S16;
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want.channels = 6;
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want.samples = 1024;
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// Try to open for 5.1 surround sound, but allow SDL to tell us that's
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// not available.
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dev = SDL_OpenAudioDevice(NULL, 0, &want, &have, SDL_AUDIO_ALLOW_CHANNELS_CHANGE);
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if (dev == 0) {
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SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
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"Failed to open audio device");
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// We'll probably have issues during audio stream init, but we'll
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// try anyway
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ret = AUDIO_CONFIGURATION_STEREO;
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goto Exit;
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}
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SDL_CloseAudioDevice(dev);
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SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
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"Audio device has %d channels", have.channels);
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if (have.channels > 2) {
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// We don't support quadraphonic or 7.1 surround, but SDL
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// should be able to downmix or upmix better from 5.1 than
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// from stereo, so use 5.1 in non-stereo cases.
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ret = AUDIO_CONFIGURATION_51_SURROUND;
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}
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else {
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ret = AUDIO_CONFIGURATION_STEREO;
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}
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Exit:
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SDL_QuitSubSystem(SDL_INIT_AUDIO);
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return ret;
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}
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bool Session::testAudio(int audioConfiguration)
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{
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SDL_AudioSpec want, have;
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SDL_AudioDeviceID dev;
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bool ret;
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if (SDL_InitSubSystem(SDL_INIT_AUDIO) != 0) {
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SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
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"Audio test - SDL_InitSubSystem(SDL_INIT_AUDIO) failed: %s",
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SDL_GetError());
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return false;
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}
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SDL_zero(want);
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want.freq = 48000;
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want.format = AUDIO_S16;
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want.samples = SAMPLES_PER_FRAME;
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switch (audioConfiguration) {
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case AUDIO_CONFIGURATION_STEREO:
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want.channels = 2;
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break;
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case AUDIO_CONFIGURATION_51_SURROUND:
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want.channels = 6;
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break;
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default:
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SDL_assert(false);
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ret = false;
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goto Exit;
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}
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// Test audio device for functionality
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dev = SDL_OpenAudioDevice(NULL, 0, &want, &have, 0);
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if (dev == 0) {
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SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
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"Audio test - Failed to open audio device: %s",
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SDL_GetError());
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ret = false;
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goto Exit;
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}
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SDL_CloseAudioDevice(dev);
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SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
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"Audio test - Successful with %d channels",
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want.channels);
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ret = true;
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Exit:
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SDL_QuitSubSystem(SDL_INIT_AUDIO);
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return ret;
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}
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int Session::sdlAudioInit(int /* audioConfiguration */,
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POPUS_MULTISTREAM_CONFIGURATION opusConfig,
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void* /* arContext */, int /* arFlags */)
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{
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SDL_AudioSpec want, have;
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int error;
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SDL_assert(!SDL_WasInit(SDL_INIT_AUDIO));
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if (SDL_InitSubSystem(SDL_INIT_AUDIO) != 0) {
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SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
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"SDL_InitSubSystem(SDL_INIT_AUDIO) failed: %s",
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SDL_GetError());
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return -1;
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}
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SDL_zero(want);
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want.freq = opusConfig->sampleRate;
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want.format = AUDIO_S16;
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want.channels = opusConfig->channelCount;
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// This is supposed to be a power of 2, but our
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// frames contain a non-power of 2 number of samples,
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// so the slop would require buffering another full frame.
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// Specifying non-Po2 seems to work for our supported platforms.
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want.samples = SAMPLES_PER_FRAME;
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s_AudioDevice = SDL_OpenAudioDevice(NULL, 0, &want, &have, 0);
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if (s_AudioDevice == 0) {
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SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
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"Failed to open audio device: %s",
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SDL_GetError());
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SDL_QuitSubSystem(SDL_INIT_AUDIO);
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return -1;
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}
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s_OpusDecoder = 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_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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SDL_CloseAudioDevice(s_AudioDevice);
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s_AudioDevice = 0;
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SDL_QuitSubSystem(SDL_INIT_AUDIO);
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return -2;
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}
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// SDL counts pending samples in the queued
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// audio size using the WASAPI backend. This
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// includes silence, which can throw off our
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// pending data count. Get a baseline so we
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// can exclude that data.
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s_BaselinePendingData = 0;
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#ifdef Q_OS_WIN32
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for (int i = 0; i < 100; i++) {
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s_BaselinePendingData = qMax(s_BaselinePendingData, SDL_GetQueuedAudioSize(s_AudioDevice));
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SDL_Delay(10);
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}
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#endif
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s_BaselinePendingData *= 2;
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SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
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"Baseline pending audio data: %d bytes",
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s_BaselinePendingData);
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s_ChannelCount = opusConfig->channelCount;
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s_SampleIndex = 0;
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s_PendingDrops = s_PendingHardDrops = 0;
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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::sdlAudioStart()
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{
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// Unpause the audio device
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SDL_PauseAudioDevice(s_AudioDevice, 0);
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}
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void Session::sdlAudioStop()
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{
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// Pause the audio device
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SDL_PauseAudioDevice(s_AudioDevice, 1);
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}
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void Session::sdlAudioCleanup()
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{
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SDL_CloseAudioDevice(s_AudioDevice);
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s_AudioDevice = 0;
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opus_multistream_decoder_destroy(s_OpusDecoder);
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s_OpusDecoder = nullptr;
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SDL_QuitSubSystem(SDL_INIT_AUDIO);
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SDL_assert(!SDL_WasInit(SDL_INIT_AUDIO));
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}
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void Session::sdlAudioDecodeAndPlaySample(char* sampleData, int sampleLength)
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{
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int samplesDecoded;
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s_SampleIndex++;
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Uint32 queuedAudio = qMax((int)SDL_GetQueuedAudioSize(s_AudioDevice) - (int)s_BaselinePendingData, 0);
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Uint32 framesQueued = queuedAudio / (SAMPLES_PER_FRAME * s_ChannelCount * sizeof(short));
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// We must check this prior to the below checks to ensure we don't
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// underflow if framesQueued - s_PendingHardDrops < 0.
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if (framesQueued <= MIN_QUEUED_FRAMES) {
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s_PendingDrops = s_PendingHardDrops = 0;
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}
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// Pend enough drops to get us back to MIN_QUEUED_FRAMES
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else if (framesQueued - s_PendingHardDrops > STOP_THE_WORLD_LIMIT) {
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s_PendingHardDrops = framesQueued - MIN_QUEUED_FRAMES;
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SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
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"Pending hard drop of %u audio frames",
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s_PendingHardDrops);
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}
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else if (framesQueued - s_PendingHardDrops - s_PendingDrops > MAX_QUEUED_FRAMES) {
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s_PendingDrops = framesQueued - MIN_QUEUED_FRAMES;
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}
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// Determine if this frame should be dropped
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if (s_PendingHardDrops != 0) {
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// Hard drops happen all at once to forcefully
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// resync with the source.
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s_PendingHardDrops--;
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return;
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}
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else if (s_PendingDrops != 0 && s_SampleIndex % DROP_RATIO_DENOM == 0) {
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// Normal drops are interspersed with the audio data
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// to hide the glitches.
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s_PendingDrops--;
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return;
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}
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samplesDecoded = opus_multistream_decode(s_OpusDecoder,
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(unsigned char*)sampleData,
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sampleLength,
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s_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 (SDL_QueueAudio(s_AudioDevice,
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s_OpusDecodeBuffer,
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sizeof(short) * samplesDecoded * s_ChannelCount) < 0) {
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SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
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"Failed to queue audio sample: %s",
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SDL_GetError());
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}
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}
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}
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