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justify jitter simulation

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This commit is contained in:
Zach Pomerantz 2016-09-16 11:40:37 -07:00
parent 53f5899a69
commit 80bce953f7
1 changed files with 12 additions and 2 deletions
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14
libraries/audio-client/src/AudioClient.cpp
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@ -572,12 +572,21 @@ void AudioClient::Gate::insert(QSharedPointer<ReceivedMessage> message) {
return; return;
} }
// Throttle the current packet until the next flush
_queue.push(message); _queue.push(message);
_index++; _index++;
// When appropriate, flush all held packets to the received audio stream
if (_isSimulatingJitter) { if (_isSimulatingJitter) {
if (randFloat() < 0.6f) { // The JITTER_FLUSH_CHANCE defines the discrete probability density function of jitter (ms),
flush(); // 60% of the time, it works every time // where f(t) = pow(1 - JITTER_FLUSH_CHANCE, (t / 10) * JITTER_FLUSH_CHANCE
// for t (ms) = 10, 20, ... (because typical packet timegap is 10ms),
// because there is a JITTER_FLUSH_CHANCE of any packet instigating a flush of all held packets.
static const float JITTER_FLUSH_CHANCE = 0.6f;
// It is set at 0.6 to give a low chance of spikes (>30ms, 2.56%) so that they are obvious,
// but settled within the measured 5s window in audio network stats.
if (randFloat() < JITTER_FLUSH_CHANCE) {
flush();
} }
} else if (!(_index % _threshold)) { } else if (!(_index % _threshold)) {
flush(); flush();
@ -585,6 +594,7 @@ void AudioClient::Gate::insert(QSharedPointer<ReceivedMessage> message) {
} }
void AudioClient::Gate::flush() { void AudioClient::Gate::flush() {
// Send all held packets to the received audio stream to be (eventually) played
while (!_queue.empty()) { while (!_queue.empty()) {
_audioClient->_receivedAudioStream.parseData(*_queue.front()); _audioClient->_receivedAudioStream.parseData(*_queue.front());
_queue.pop(); _queue.pop();