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tighten audio latency by clearing the network stack after starves

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This commit is contained in:
Stephen Birarda 2013-06-26 18:17:50 -07:00
parent 1c0b256f92
commit 14718980e5
1 changed files with 17 additions and 4 deletions
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21
interface/src/Audio.cpp
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@ -136,7 +136,8 @@ inline void Audio::performIO(int16_t* inputLeft, int16_t* outputLeft, int16_t* o
// if there is anything in the ring buffer, decide what to do:
if (ringBuffer->getEndOfLastWrite()) {
if (!ringBuffer->isStarted() && ringBuffer->diffLastWriteNextOutput() < (PACKET_LENGTH_SAMPLES + _jitterBufferSamples * (ringBuffer->isStereo() ? 2 : 1))) {
if (!ringBuffer->isStarted() && ringBuffer->diffLastWriteNextOutput() <
(PACKET_LENGTH_SAMPLES + _jitterBufferSamples * (ringBuffer->isStereo() ? 2 : 1))) {
//
// If not enough audio has arrived to start playback, keep waiting
//
@ -147,8 +148,7 @@ inline void Audio::performIO(int16_t* inputLeft, int16_t* outputLeft, int16_t* o
PACKET_LENGTH_SAMPLES,
_jitterBufferSamples);
#endif
} else if (ringBuffer->isStarted() && (ringBuffer->diffLastWriteNextOutput()
< PACKET_LENGTH_SAMPLES * (ringBuffer->isStereo() ? 2 : 1))) {
} else if (ringBuffer->isStarted() && ringBuffer->diffLastWriteNextOutput() == 0) {
//
// If we have started and now have run out of audio to send to the audio device,
// this means we've starved and should restart.
@ -159,7 +159,7 @@ inline void Audio::performIO(int16_t* inputLeft, int16_t* outputLeft, int16_t* o
_packetsReceivedThisPlayback = 0;
_wasStarved = 10; // Frames for which to render the indication that the system was starved.
#ifdef SHOW_AUDIO_DEBUG
printLog("Starved, remaining samples = %.0f\n",
printLog("Starved, remaining samples = %d\n",
ringBuffer->diffLastWriteNextOutput());
#endif
@ -423,7 +423,20 @@ void Audio::addReceivedAudioToBuffer(unsigned char* receivedData, int receivedBy
gettimeofday(&_firstPacketReceivedTime, NULL);
}
if (_ringBuffer.diffLastWriteNextOutput() + PACKET_LENGTH_SAMPLES >
PACKET_LENGTH_SAMPLES + (ceilf((float) (_jitterBufferSamples * 2) / PACKET_LENGTH_SAMPLES) * PACKET_LENGTH_SAMPLES)) {
// this packet would give us more than the required amount for play out
// discard the first packet in the buffer
_ringBuffer.setNextOutput(_ringBuffer.getNextOutput() + PACKET_LENGTH_SAMPLES);
if (_ringBuffer.getNextOutput() == _ringBuffer.getBuffer() + RING_BUFFER_LENGTH_SAMPLES) {
_ringBuffer.setNextOutput(_ringBuffer.getBuffer());
}
}
//printf("Got audio packet %d\n", _packetsReceivedThisPlayback);
_ringBuffer.parseData((unsigned char*) receivedData, PACKET_LENGTH_BYTES + sizeof(PACKET_HEADER));
_lastReceiveTime = currentReceiveTime;