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video coding tests: Adding a Normal distribution to simulate packet arrival times

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Review URL: http://webrtc-codereview.appspot.com/138003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@483 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
mikhal@webrtc.org 2011-08-29 23:38:04 +00:00
parent 8571af7be6
commit 46171cf546
2 changed files with 25 additions and 2 deletions
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23
src/modules/video_coding/main/test/test_util.cc
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@ -15,6 +15,19 @@
using namespace webrtc; using namespace webrtc;
// Normal Distribution
#define PI 3.14159265
double
NormalDist(double mean, double stdDev)
{
// Creating a Normal distribution variable from two independent uniform
// variables based on the Box-Muller transform
double uniform1 = (std::rand() + 1.0) / (RAND_MAX + 1.0);
double uniform2 = (std::rand() + 1.0) / (RAND_MAX + 1.0);
return (mean + stdDev * sqrt(-2 * log(uniform1)) * cos(2 * PI * uniform2));
}
/****************************** /******************************
* VCMEncodeCompleteCallback * VCMEncodeCompleteCallback
*****************************/ *****************************/
@ -199,6 +212,7 @@ RTPSendCompleteCallback::RTPSendCompleteCallback(RtpRtcp* rtp,
_lossPct(0), _lossPct(0),
_burstLength(0), _burstLength(0),
_networkDelayMs(0), _networkDelayMs(0),
_jitterVar(0),
_prevLossState(0), _prevLossState(0),
_totalSentLength(0), _totalSentLength(0),
_rtpPackets(), _rtpPackets(),
@ -307,8 +321,13 @@ RTPSendCompleteCallback::SendPacket(int channel, const void *data, int len)
rtpPacket* newPacket = new rtpPacket(); rtpPacket* newPacket = new rtpPacket();
memcpy(newPacket->data, data, len); memcpy(newPacket->data, data, len);
newPacket->length = len; newPacket->length = len;
// Simulate receive time // Simulate receive time = network delay + packet jitter
newPacket->receiveTime = now + _networkDelayMs; // simulated as a Normal distribution random variable with
// mean = networkDelay and variance = jitterVar
WebRtc_Word32
simulatedDelay = (WebRtc_Word32)NormalDist(_networkDelayMs,
sqrt(_jitterVar));
newPacket->receiveTime = now + simulatedDelay;
_rtpPackets.PushBack(newPacket); _rtpPackets.PushBack(newPacket);
} }

4
src/modules/video_coding/main/test/test_util.h
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@ -208,6 +208,9 @@ public:
// Set network delay in the network // Set network delay in the network
void SetNetworkDelay(WebRtc_UWord32 networkDelayMs) void SetNetworkDelay(WebRtc_UWord32 networkDelayMs)
{_networkDelayMs = networkDelayMs;}; {_networkDelayMs = networkDelayMs;};
// Set Packet jitter delay
void SetJitterVar(WebRtc_UWord32 jitterVar)
{_jitterVar = jitterVar;};
// Return send count // Return send count
int SendCount() {return _sendCount; } int SendCount() {return _sendCount; }
// Return accumulated length in bytes of transmitted packets // Return accumulated length in bytes of transmitted packets
@ -221,6 +224,7 @@ private:
double _lossPct; double _lossPct;
double _burstLength; double _burstLength;
WebRtc_UWord32 _networkDelayMs; WebRtc_UWord32 _networkDelayMs;
double _jitterVar;
bool _prevLossState; bool _prevLossState;
WebRtc_UWord32 _totalSentLength; WebRtc_UWord32 _totalSentLength;
webrtc::ListWrapper _rtpPackets; webrtc::ListWrapper _rtpPackets;