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Adapt the audio latency mitigation to WASAPI quirks and add a "stop the world" mode for quick resync

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This commit is contained in:
Cameron Gutman 2018-07-15 21:43:43 -07:00
parent 7f8d4c88c1
commit 68bbeafab3
2 changed files with 42 additions and 6 deletions
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46
app/streaming/audio.cpp
View File

@ -7,6 +7,7 @@
#define SAMPLES_PER_FRAME 240
#define MIN_QUEUED_FRAMES 2
#define MAX_QUEUED_FRAMES 4
#define STOP_THE_WORLD_LIMIT 20
#define DROP_RATIO_DENOM 32
SDL_AudioDeviceID Session::s_AudioDevice;
@ -14,7 +15,9 @@ OpusMSDecoder* Session::s_OpusDecoder;
short Session::s_OpusDecodeBuffer[MAX_CHANNELS * SAMPLES_PER_FRAME];
int Session::s_ChannelCount;
int Session::s_PendingDrops;
int Session::s_PendingHardDrops;
unsigned int Session::s_SampleIndex;
Uint32 Session::s_BaselinePendingData;
int Session::sdlDetermineAudioConfiguration()
{
@ -95,9 +98,26 @@ int Session::sdlAudioInit(int /* audioConfiguration */,
return -2;
}
// SDL counts pending samples in the queued
// audio size using the WASAPI backend. This
// includes silence, which can throw off our
// pending data count. Get a baseline so we
// can exclude that data.
s_BaselinePendingData = 0;
#ifdef _WIN32
for (int i = 0; i < 100; i++) {
s_BaselinePendingData = qMax(s_BaselinePendingData, SDL_GetQueuedAudioSize(s_AudioDevice));
SDL_Delay(10);
}
#endif
s_BaselinePendingData *= 2;
SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
"Baseline pending audio data: %d bytes",
s_BaselinePendingData);
s_ChannelCount = opusConfig->channelCount;
s_SampleIndex = 0;
s_PendingDrops = 0;
s_PendingDrops = s_PendingHardDrops = 0;
return 0;
}
@ -129,19 +149,33 @@ void Session::sdlAudioDecodeAndPlaySample(char* sampleData, int sampleLength)
s_SampleIndex++;
Uint32 queuedAudio = SDL_GetQueuedAudioSize(s_AudioDevice);
Uint32 queuedAudio = qMax((int)SDL_GetQueuedAudioSize(s_AudioDevice) - (int)s_BaselinePendingData, 0);
Uint32 framesQueued = queuedAudio / (SAMPLES_PER_FRAME * s_ChannelCount * sizeof(short));
if (framesQueued - s_PendingDrops > MAX_QUEUED_FRAMES) {
// Pend enough drops to get us back to MIN_QUEUED_FRAMES
s_PendingDrops += (framesQueued - MIN_QUEUED_FRAMES);
// Pend enough drops to get us back to MIN_QUEUED_FRAMES
if (framesQueued - s_PendingHardDrops > STOP_THE_WORLD_LIMIT) {
s_PendingHardDrops = framesQueued - MIN_QUEUED_FRAMES;
SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
"Pending hard drop of %u audio frames",
s_PendingHardDrops);
}
else if (framesQueued - s_PendingHardDrops - s_PendingDrops > MAX_QUEUED_FRAMES) {
s_PendingDrops = framesQueued - MIN_QUEUED_FRAMES;
}
// Determine if this frame should be dropped
if (framesQueued <= MIN_QUEUED_FRAMES) {
s_PendingDrops = 0;
s_PendingDrops = s_PendingHardDrops = 0;
}
else if (s_PendingHardDrops != 0) {
// Hard drops happen all at once to forcefully
// resync with the source.
s_PendingHardDrops--;
return;
}
else if (s_PendingDrops != 0 && s_SampleIndex % DROP_RATIO_DENOM == 0) {
// Normal drops are interspersed with the audio data
// to hide the glitches.
s_PendingDrops--;
return;
}

2
app/streaming/session.hpp
View File

@ -123,7 +123,9 @@ private:
static short s_OpusDecodeBuffer[];
static int s_ChannelCount;
static int s_PendingDrops;
static int s_PendingHardDrops;
static unsigned int s_SampleIndex;
static Uint32 s_BaselinePendingData;
static AUDIO_RENDERER_CALLBACKS k_AudioCallbacks;
static CONNECTION_LISTENER_CALLBACKS k_ConnCallbacks;