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Fixed/improved Standard Deviation code used to get jitter

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This commit is contained in:
Philip Rosedale 2013-03-11 17:13:18 -07:00
parent 74452b7cc5
commit 3e62aa5169
1 changed files with 21 additions and 23 deletions
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44
interface/src/Audio.cpp
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@ -57,6 +57,7 @@ bool stopAudioReceiveThread = false;
timeval firstPlaybackTimer;
int packetsReceivedThisPlayback = 0;
float usecsAtStartup = 0;
#define LOG_SAMPLE_DELAY 0
@ -224,10 +225,16 @@ void *receiveAudioViaUDP(void *args) {
int16_t *receivedData = new int16_t[PACKET_LENGTH_SAMPLES];
ssize_t receivedBytes;
// Init Jitter timer values
timeval previousReceiveTime, currentReceiveTime = {};
gettimeofday(&previousReceiveTime, NULL);
gettimeofday(&currentReceiveTime, NULL);
int totalPacketsReceived = 0;
stdev.reset();
if (LOG_SAMPLE_DELAY) {
gettimeofday(&previousReceiveTime, NULL);
char *directory = new char[50];
char *filename = new char[50];
@ -245,30 +252,24 @@ void *receiveAudioViaUDP(void *args) {
while (!stopAudioReceiveThread) {
if (sharedAudioData->audioSocket->receive((void *)receivedData, &receivedBytes)) {
bool firstSample = (currentReceiveTime.tv_sec == 0);
gettimeofday(&currentReceiveTime, NULL);
totalPacketsReceived++;
if (LOG_SAMPLE_DELAY) {
if (!firstSample) {
// write time difference (in microseconds) between packet receipts to file
double timeDiff = diffclock(&previousReceiveTime, &currentReceiveTime);
logFile << timeDiff << std::endl;
}
// write time difference (in microseconds) between packet receipts to file
double timeDiff = diffclock(&previousReceiveTime, &currentReceiveTime);
logFile << timeDiff << std::endl;
}
// Compute and report standard deviation for jitter calculation
if (firstSample) {
double tDiff = diffclock(&previousReceiveTime, &currentReceiveTime);
//printf("tDiff %4.1f\n", tDiff);
// Discard first few received packets for computing jitter (often they pile up on start)
if (totalPacketsReceived > 3) stdev.addValue(tDiff);
if (stdev.getSamples() > 500) {
sharedAudioData->measuredJitter = stdev.getStDev();
printf("Avg: %4.2f, Stdev: %4.2f\n", stdev.getAverage(), sharedAudioData->measuredJitter);
stdev.reset();
} else {
double tDiff = diffclock(&previousReceiveTime, &currentReceiveTime);
//printf(\n";
stdev.addValue(tDiff);
if (stdev.getSamples() > 500) {
sharedAudioData->measuredJitter = stdev.getStDev();
printf("Avg: %4.2f, Stdev: %4.2f\n", stdev.getAverage(), sharedAudioData->measuredJitter);
stdev.reset();
}
}
AudioRingBuffer *ringBuffer = sharedAudioData->ringBuffer;
@ -280,14 +281,11 @@ void *receiveAudioViaUDP(void *args) {
else {
//printf("Audio packet received at %6.0f\n", usecTimestampNow()/1000);
}
if (packetsReceivedThisPlayback == 1) gettimeofday(&firstPlaybackTimer, NULL);
ringBuffer->parseData(receivedData, PACKET_LENGTH_BYTES);
if (LOG_SAMPLE_DELAY) {
gettimeofday(&previousReceiveTime, NULL);
}
previousReceiveTime = currentReceiveTime;
}
}