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Implement audio latency mitigations

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This commit is contained in:
Cameron Gutman 2018-07-15 20:17:08 -07:00
parent 8c65ee8bc5
commit 7f8d4c88c1
2 changed files with 34 additions and 1 deletions
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33
app/streaming/audio.cpp
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@ -5,11 +5,16 @@
#define MAX_CHANNELS 6 #define MAX_CHANNELS 6
#define SAMPLES_PER_FRAME 240 #define SAMPLES_PER_FRAME 240
#define MIN_QUEUED_FRAMES 2
#define MAX_QUEUED_FRAMES 4
#define DROP_RATIO_DENOM 32
SDL_AudioDeviceID Session::s_AudioDevice; SDL_AudioDeviceID Session::s_AudioDevice;
OpusMSDecoder* Session::s_OpusDecoder; OpusMSDecoder* Session::s_OpusDecoder;
short Session::s_OpusDecodeBuffer[MAX_CHANNELS * SAMPLES_PER_FRAME]; short Session::s_OpusDecodeBuffer[MAX_CHANNELS * SAMPLES_PER_FRAME];
int Session::s_ChannelCount; int Session::s_ChannelCount;
int Session::s_PendingDrops;
unsigned int Session::s_SampleIndex;
int Session::sdlDetermineAudioConfiguration() int Session::sdlDetermineAudioConfiguration()
{ {
@ -60,7 +65,12 @@ int Session::sdlAudioInit(int /* audioConfiguration */,
want.freq = opusConfig->sampleRate; want.freq = opusConfig->sampleRate;
want.format = AUDIO_S16; want.format = AUDIO_S16;
want.channels = opusConfig->channelCount; want.channels = opusConfig->channelCount;
want.samples = 1024;
// This is supposed to be a power of 2, but our
// frames contain a non-power of 2 number of samples,
// so the slop would require buffering another full frame.
// Specifying non-Po2 seems to work for our supported platforms.
want.samples = SAMPLES_PER_FRAME;
s_AudioDevice = SDL_OpenAudioDevice(NULL, 0, &want, &have, 0); s_AudioDevice = SDL_OpenAudioDevice(NULL, 0, &want, &have, 0);
if (s_AudioDevice == 0) { if (s_AudioDevice == 0) {
@ -86,6 +96,8 @@ int Session::sdlAudioInit(int /* audioConfiguration */,
} }
s_ChannelCount = opusConfig->channelCount; s_ChannelCount = opusConfig->channelCount;
s_SampleIndex = 0;
s_PendingDrops = 0;
return 0; return 0;
} }
@ -115,6 +127,25 @@ void Session::sdlAudioDecodeAndPlaySample(char* sampleData, int sampleLength)
{ {
int samplesDecoded; int samplesDecoded;
s_SampleIndex++;
Uint32 queuedAudio = SDL_GetQueuedAudioSize(s_AudioDevice);
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);
}
// Determine if this frame should be dropped
if (framesQueued <= MIN_QUEUED_FRAMES) {
s_PendingDrops = 0;
}
else if (s_PendingDrops != 0 && s_SampleIndex % DROP_RATIO_DENOM == 0) {
s_PendingDrops--;
return;
}
samplesDecoded = opus_multistream_decode(s_OpusDecoder, samplesDecoded = opus_multistream_decode(s_OpusDecoder,
(unsigned char*)sampleData, (unsigned char*)sampleData,
sampleLength, sampleLength,

2
app/streaming/session.hpp
View File

@ -122,6 +122,8 @@ private:
static OpusMSDecoder* s_OpusDecoder; static OpusMSDecoder* s_OpusDecoder;
static short s_OpusDecodeBuffer[]; static short s_OpusDecodeBuffer[];
static int s_ChannelCount; static int s_ChannelCount;
static int s_PendingDrops;
static unsigned int s_SampleIndex;
static AUDIO_RENDERER_CALLBACKS k_AudioCallbacks; static AUDIO_RENDERER_CALLBACKS k_AudioCallbacks;
static CONNECTION_LISTENER_CALLBACKS k_ConnCallbacks; static CONNECTION_LISTENER_CALLBACKS k_ConnCallbacks;