video_coding: Updating offline tests
1. Adding parameters to the input arguments. 2. Introducing delay in the RTPSend callback 3. Clean-up Review URL: http://webrtc-codereview.appspot.com/116006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@414 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
mikhal@webrtc.org
parent
6f555dcafe
commit
d70b77d19f
@ -21,7 +21,7 @@ using namespace webrtc;
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*****************************/
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// Basic callback implementation
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// passes the encoded frame directly to the encoder
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// Packetization callback implmentation
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// Packetization callback implementation
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VCMEncodeCompleteCallback::VCMEncodeCompleteCallback(FILE* encodedFile):
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_encodedFile(encodedFile),
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_encodedBytes(0),
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@ -90,10 +90,10 @@ VCMEncodeCompleteCallback::SendData(
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// TODO(hlundin): Remove assert once we've piped PictureID into VCM
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// through the WebRtcRTPHeader.
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assert(rtpInfo.type.Video.codec != kRTPVideoVP8);
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_VCMReceiver->IncomingPacket(payloadData, payloadSize, rtpInfo);
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int ret = _VCMReceiver->IncomingPacket(payloadData, payloadSize, rtpInfo);
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_encodeComplete = true;
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return 0;
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return ret;
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}
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float
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@ -133,7 +133,7 @@ VCMEncodeCompleteCallback::ResetByteCount()
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/***********************************/
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// Encode Complete callback implementation
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// passes the encoded frame via the RTP module to the decoder
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// Packetization callback implmentation
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// Packetization callback implementation
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WebRtc_Word32
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VCMRTPEncodeCompleteCallback::SendData(
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@ -148,7 +148,6 @@ VCMRTPEncodeCompleteCallback::SendData(
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_frameType = frameType;
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_encodedBytes+= payloadSize;
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_encodeComplete = true;
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//printf("encoded = %d Bytes\n", payloadSize);
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return _RTPModule->SendOutgoingData(frameType,
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payloadType,
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timeStamp,
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@ -178,7 +177,7 @@ VCMRTPEncodeCompleteCallback::EncodeComplete()
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return false;
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}
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// Decoded Frame Callback Implmentation
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// Decoded Frame Callback Implementation
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WebRtc_Word32
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VCMDecodeCompleteCallback::FrameToRender(VideoFrame& videoFrame)
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@ -196,12 +195,14 @@ VCMRTPEncodeCompleteCallback::EncodeComplete()
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}
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int
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VCMDecodeCompleteCallback::PSNRLastFrame(const VideoFrame& sourceFrame, double *YPSNRptr)
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VCMDecodeCompleteCallback::PSNRLastFrame(const VideoFrame& sourceFrame,
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double *YPSNRptr)
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{
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double mse = 0.0;
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double mseLogSum = 0.0;
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WebRtc_Word32 frameBytes = sourceFrame.Height() * sourceFrame.Width(); // only Y
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// Y only
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WebRtc_Word32 frameBytes = sourceFrame.Height() * sourceFrame.Width();
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WebRtc_UWord8 *ref = sourceFrame.Buffer();
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if (_lastDecodedFrame.Height() == 0)
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{
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@ -233,12 +234,16 @@ VCMDecodeCompleteCallback::DecodedBytes()
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return _decodedBytes;
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}
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RTPSendCompleteCallback::RTPSendCompleteCallback(RtpRtcp* rtp, const char* filename):
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RTPSendCompleteCallback::RTPSendCompleteCallback(RtpRtcp* rtp,
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const char* filename):
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_sendCount(0),
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_rtp(rtp),
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_lossPct(0),
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_burstLength(0),
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_networkDelayMs(0),
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_prevLossState(0),
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_totalSentLength(0),
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_rtpPackets(),
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_rtpDump(NULL)
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{
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if (filename != NULL)
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@ -254,11 +259,20 @@ RTPSendCompleteCallback::~RTPSendCompleteCallback()
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_rtpDump->Stop();
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RtpDump::DestroyRtpDump(_rtpDump);
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}
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// Delete remaining packets
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while (!_rtpPackets.Empty())
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{
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// Take first packet in list
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delete static_cast<rtpPacket*>((_rtpPackets.First())->GetItem());
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_rtpPackets.PopFront();
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}
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}
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int
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RTPSendCompleteCallback::SendPacket(int channel, const void *data, int len)
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{
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_sendCount++;
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_totalSentLength += len;
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bool transmitPacket = true;
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// Packet Loss
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@ -268,8 +282,7 @@ RTPSendCompleteCallback::SendPacket(int channel, const void *data, int len)
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if (PacketLoss(_lossPct))
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{
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// drop
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//printf("\tDrop packet, sendCount = %d\n", _sendCount);
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return len;
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transmitPacket = false;
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}
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}
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else
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@ -291,7 +304,7 @@ RTPSendCompleteCallback::SendPacket(int channel, const void *data, int len)
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// to the transition probabilities:
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double probTrans10 = 100 * (1.0 / _burstLength);
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double probTrans11 = (100.0 - probTrans10);
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double probTrans01 = (probTrans10 * ( _lossPct / (100.0 - _lossPct) ) );
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double probTrans01 = (probTrans10 * ( _lossPct / (100.0 - _lossPct)));
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// Note: Random loss (Bernoulli) model is a special case where:
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// burstLength = 100.0 / (100.0 - _lossPct) (i.e., p10 + p01 = 100)
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@ -299,25 +312,24 @@ RTPSendCompleteCallback::SendPacket(int channel, const void *data, int len)
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if (_prevLossState == 0 )
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{
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// previous packet was received
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if (PacketLoss(probTrans01) )
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if (PacketLoss(probTrans01))
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{
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// drop, update previous state to loss
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_prevLossState = 1;
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return len;
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transmitPacket = false;
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}
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}
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else if (_prevLossState == 1)
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{
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_prevLossState = 0;
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// previous packet was lost
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if (PacketLoss(probTrans11) )
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if (PacketLoss(probTrans11))
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{
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// drop, update previous state to loss
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_prevLossState = 1;
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return len;
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transmitPacket = false;
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}
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}
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// no drop, update previous state to received
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_prevLossState = 0;
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}
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if (_rtpDump != NULL)
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@ -327,17 +339,53 @@ RTPSendCompleteCallback::SendPacket(int channel, const void *data, int len)
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return -1;
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}
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}
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if(_rtp->IncomingPacket((const WebRtc_UWord8*)data, len) == 0)
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WebRtc_UWord64 now = VCMTickTime::MillisecondTimestamp();
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// Insert outgoing packet into list
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if (transmitPacket)
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{
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return len;
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rtpPacket* newPacket = new rtpPacket();
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memcpy(newPacket->data, data, len);
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newPacket->length = len;
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// Simulate receive time
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newPacket->receiveTime = now + _networkDelayMs;
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_rtpPackets.PushBack(newPacket);
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}
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return -1;
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// Are we ready to send packets to the receiver?
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rtpPacket* packet = NULL;
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while (!_rtpPackets.Empty())
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{
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// Take first packet in list
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packet = static_cast<rtpPacket*>((_rtpPackets.First())->GetItem());
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WebRtc_Word64 timeToReceive = packet->receiveTime - now;
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if (timeToReceive > 0)
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{
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// No available packets to send
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break;
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}
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_rtpPackets.PopFront();
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// Send to receive side
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if (_rtp->IncomingPacket((const WebRtc_UWord8*)packet->data,
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packet->length) < 0)
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{
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delete packet;
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packet = NULL;
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// Will return an error after the first packet that goes wrong
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return -1;
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}
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delete packet;
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packet = NULL;
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}
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return len; // OK
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}
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int
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RTPSendCompleteCallback::SendRTCPPacket(int channel, const void *data, int len)
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{
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if(_rtp->IncomingPacket((const WebRtc_UWord8*)data, len) == 0)
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if (_rtp->IncomingPacket((const WebRtc_UWord8*)data, len) == 0)
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{
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return len;
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}
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@ -366,13 +414,15 @@ RTPSendCompleteCallback::PacketLoss(double lossPct)
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}
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WebRtc_Word32
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PacketRequester::ResendPackets(const WebRtc_UWord16* sequenceNumbers, WebRtc_UWord16 length)
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PacketRequester::ResendPackets(const WebRtc_UWord16* sequenceNumbers,
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WebRtc_UWord16 length)
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{
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return _rtp.SendNACK(sequenceNumbers, length);
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}
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WebRtc_Word32
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PSNRfromFiles(const WebRtc_Word8 *refFileName, const WebRtc_Word8 *testFileName, WebRtc_Word32 width, WebRtc_Word32 height, double *YPSNRptr)
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PSNRfromFiles(const WebRtc_Word8 *refFileName, const WebRtc_Word8 *testFileName,
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WebRtc_Word32 width, WebRtc_Word32 height, double *YPSNRptr)
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{
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FILE *refFp = fopen(refFileName, "rb");
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if( refFp == NULL ) {
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@ -392,9 +442,10 @@ PSNRfromFiles(const WebRtc_Word8 *refFileName, const WebRtc_Word8 *testFileName,
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double mseLogSum = 0.0;
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WebRtc_Word32 frames = 0;
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WebRtc_Word32 frameBytes = 3*width*height/2; // bytes in one frame I420
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WebRtc_UWord8 *ref = new WebRtc_UWord8[frameBytes]; // space for one frame I420
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WebRtc_UWord8 *test = new WebRtc_UWord8[frameBytes]; // space for one frame I420
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// Allocating size for one I420 frame.
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WebRtc_Word32 frameBytes = 3 * width * height >> 1;
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WebRtc_UWord8 *ref = new WebRtc_UWord8[frameBytes];
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WebRtc_UWord8 *test = new WebRtc_UWord8[frameBytes];
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WebRtc_Word32 refBytes = (WebRtc_Word32) fread(ref, 1, frameBytes, refFp);
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WebRtc_Word32 testBytes = (WebRtc_Word32) fread(test, 1, frameBytes, testFp);
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@ -440,16 +491,9 @@ PSNRfromFiles(const WebRtc_Word8 *refFileName, const WebRtc_Word8 *testFileName,
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return 0;
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}
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static double
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similarity
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(
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unsigned long sum_s,
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unsigned long sum_r,
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unsigned long sum_sq_s,
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unsigned long sum_sq_r,
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unsigned long sum_sxr,
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int count
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similarity(unsigned long sum_s, unsigned long sum_r, unsigned long sum_sq_s,
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unsigned long sum_sq_r, unsigned long sum_sxr, int count
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)
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{
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int64_t ssim_n, ssim_d;
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@ -472,13 +516,7 @@ similarity
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}
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static double
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ssim_8x8_c
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(
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unsigned char *s,
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int sp,
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unsigned char *r,
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int rp
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)
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ssim_8x8_c(unsigned char *s, int sp, unsigned char *r, int rp)
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{
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unsigned long sum_s = 0;
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unsigned long sum_r = 0;
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@ -487,9 +525,9 @@ ssim_8x8_c
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unsigned long sum_sxr = 0;
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int i,j;
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for(i=0;i<8;i++,s+=sp,r+=rp)
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for (i = 0;i < 8; i++, s += sp,r += rp)
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{
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for(j=0;j<8;j++)
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for (j = 0; j < 8; j++)
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{
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sum_s += s[j];
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sum_r += r[j];
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@ -498,20 +536,13 @@ ssim_8x8_c
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sum_sxr += s[j] * r[j];
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}
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}
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return similarity(sum_s, sum_r, sum_sq_s, sum_sq_r, sum_sxr, 64);
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}
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#if defined(WEBRTC_USE_SSE2)
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#include <xmmintrin.h>
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static double
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ssim_8x8_sse2
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(
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unsigned char *s,
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int sp,
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unsigned char *r,
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int rp
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)
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ssim_8x8_sse2(unsigned char *s, int sp, unsigned char *r, int rp)
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{
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int i;
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const __m128i z = _mm_setzero_si128();
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@ -521,7 +552,7 @@ ssim_8x8_sse2
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__m128i sum_sq_r_32 = _mm_setzero_si128();
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__m128i sum_sxr_32 = _mm_setzero_si128();
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for(i=0;i<8;i++,s+=sp,r+=rp)
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for (i = 0;i < 8; i++,s += sp,r += rp)
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{
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const __m128i s_8 = _mm_loadl_epi64((__m128i*)(s));
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const __m128i r_8 = _mm_loadl_epi64((__m128i*)(r));
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@ -577,14 +608,8 @@ ssim_8x8_sse2
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#endif
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double
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SSIM_frame
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(
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unsigned char *img1,
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unsigned char *img2,
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int stride_img1,
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int stride_img2,
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int width,
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int height)
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SSIM_frame(unsigned char *img1, unsigned char *img2, int stride_img1,
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int stride_img2,int width, int height)
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{
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int i,j;
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unsigned int samples = 0;
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@ -592,7 +617,7 @@ SSIM_frame
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double (*ssim_8x8)(unsigned char*, int, unsigned char*, int rp);
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ssim_8x8 = ssim_8x8_c;
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if(WebRtc_GetCPUInfo(kSSE2))
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if (WebRtc_GetCPUInfo(kSSE2))
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{
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#if defined(WEBRTC_USE_SSE2)
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ssim_8x8 = ssim_8x8_sse2;
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@ -600,11 +625,12 @@ SSIM_frame
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}
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// sample point start with each 4x4 location
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for(i=0; i < height-8; i+=4, img1 += stride_img1*4, img2 += stride_img2*4)
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for (i = 0; i < height-8; i += 4, img1 += stride_img1 * 4,
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img2 += stride_img2 * 4)
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{
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for(j=0; j < width-8; j+=4 )
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for (j = 0; j < width - 8; j += 4 )
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{
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double v = ssim_8x8(img1+j, stride_img1, img2+j, stride_img2);
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double v = ssim_8x8(img1 + j, stride_img1, img2 + j, stride_img2);
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ssim_total += v;
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samples++;
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}
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@ -614,17 +640,20 @@ SSIM_frame
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}
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WebRtc_Word32
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SSIMfromFiles(const WebRtc_Word8 *refFileName, const WebRtc_Word8 *testFileName, WebRtc_Word32 width, WebRtc_Word32 height, double *SSIMptr)
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SSIMfromFiles(const WebRtc_Word8 *refFileName, const WebRtc_Word8 *testFileName,
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WebRtc_Word32 width, WebRtc_Word32 height, double *SSIMptr)
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{
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FILE *refFp = fopen(refFileName, "rb");
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if( refFp == NULL ) {
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if ( refFp == NULL )
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{
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// cannot open reference file
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fprintf(stderr, "Cannot open file %s\n", refFileName);
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return -1;
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}
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FILE *testFp = fopen(testFileName, "rb");
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if( testFp == NULL ) {
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if ( testFp == NULL )
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{
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// cannot open test file
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fprintf(stderr, "Cannot open file %s\n", testFileName);
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return -2;
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@ -632,7 +661,7 @@ SSIMfromFiles(const WebRtc_Word8 *refFileName, const WebRtc_Word8 *testFileName,
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int frames = 0;
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const int frameBytes = 3*width*height/2; // bytes in one frame I420
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const int frameBytes = 3 * width *height / 2; // bytes in one frame I420
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unsigned char *ref = new unsigned char[frameBytes];
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unsigned char *test = new unsigned char[frameBytes];
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@ -674,24 +703,29 @@ ConvertCodecType(const char* plname)
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if (strncmp(plname,"VP8" , 3) == 0)
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{
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return kRTPVideoVP8;
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}else if (strncmp(plname,"H263" , 5) == 0)
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}
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else if (strncmp(plname,"H263" , 5) == 0)
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{
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return kRTPVideoH263;
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}else if (strncmp(plname, "H263-1998",10) == 0)
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}
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else if (strncmp(plname, "H263-1998",10) == 0)
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{
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return kRTPVideoH263;
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}else if (strncmp(plname,"I420" , 5) == 0)
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}
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else if (strncmp(plname,"I420" , 5) == 0)
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{
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return kRTPVideoI420;
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}else
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}
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else
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{
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return kRTPVideoNoVideo; // defualt value
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return kRTPVideoNoVideo; // Default value
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}
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}
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WebRtc_Word32
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SendStatsTest::SendStatistics(const WebRtc_UWord32 bitRate, const WebRtc_UWord32 frameRate)
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SendStatsTest::SendStatistics(const WebRtc_UWord32 bitRate,
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const WebRtc_UWord32 frameRate)
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{
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TEST(frameRate <= _frameRate);
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TEST(bitRate > 0 && bitRate < 100000);
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@ -17,6 +17,7 @@
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#include "module_common_types.h"
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#include "tick_time.h"
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#include "test_util.h"
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#include "list_wrapper.h"
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#include <string.h>
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#include <fstream>
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@ -28,19 +29,24 @@ enum { kMaxWaitEncTimeMs = 100 };
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class CmdArgs
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{
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public:
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CmdArgs() : codecName(""), codecType(webrtc::kVideoCodecVP8), width(-1),
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height(-1), bitRate(-1), frameRate(-1),
|
||||
inputFile(""), outputFile(""), testNum(-1)
|
||||
{}
|
||||
std::string codecName;
|
||||
webrtc::VideoCodecType codecType;
|
||||
int width;
|
||||
int height;
|
||||
int bitRate;
|
||||
int frameRate;
|
||||
std::string inputFile;
|
||||
std::string outputFile;
|
||||
int testNum;
|
||||
CmdArgs() : codecName(""), codecType(webrtc::kVideoCodecVP8), width(-1),
|
||||
height(-1), bitRate(-1), frameRate(-1), packetLoss(0), rtt(0),
|
||||
protectionMode(0), camaEnable(0), inputFile(""), outputFile(""),
|
||||
testNum(-1)
|
||||
{}
|
||||
std::string codecName;
|
||||
webrtc::VideoCodecType codecType;
|
||||
int width;
|
||||
int height;
|
||||
int bitRate;
|
||||
int frameRate;
|
||||
int packetLoss;
|
||||
int rtt;
|
||||
int protectionMode;
|
||||
int camaEnable;
|
||||
std::string inputFile;
|
||||
std::string outputFile;
|
||||
int testNum;
|
||||
};
|
||||
|
||||
// forward declaration
|
||||
@ -50,7 +56,8 @@ namespace webrtc
|
||||
class RtpDump;
|
||||
}
|
||||
|
||||
// definition of general test function to be used by VCM tester (mainly send side)
|
||||
// Definition of general test function to be used by VCM tester
|
||||
// (mainly send side)
|
||||
/*
|
||||
Includes the following:
|
||||
1. General Callback definition for VCM test functions - no RTP.
|
||||
@ -60,31 +67,34 @@ namespace webrtc
|
||||
3. Caluclate PSNR from file function (for now: does not deal with frame drops)
|
||||
*/
|
||||
|
||||
//Send Side - Packetization callback - send an encoded frame directly to the VCMReceiver
|
||||
// Send Side - Packetization callback - send an encoded frame to the VCMReceiver
|
||||
class VCMEncodeCompleteCallback: public webrtc::VCMPacketizationCallback
|
||||
{
|
||||
public:
|
||||
// constructor input: file in which encoded data will be written, and test parameters
|
||||
// constructor input: file in which encoded data will be written
|
||||
VCMEncodeCompleteCallback(FILE* encodedFile);
|
||||
virtual ~VCMEncodeCompleteCallback();
|
||||
// Register transport callback
|
||||
void RegisterTransportCallback(webrtc::VCMPacketizationCallback* transport);
|
||||
// process encoded data received from the encoder, pass stream to the VCMReceiver module
|
||||
// Process encoded data received from the encoder, pass stream to the
|
||||
// VCMReceiver module
|
||||
WebRtc_Word32 SendData(const webrtc::FrameType frameType,
|
||||
const WebRtc_UWord8 payloadType, const WebRtc_UWord32 timeStamp,
|
||||
const WebRtc_UWord8* payloadData, const WebRtc_UWord32 payloadSize,
|
||||
const webrtc::RTPFragmentationHeader& fragmentationHeader,
|
||||
const webrtc::RTPVideoTypeHeader* videoTypeHdr);
|
||||
// Register exisitng VCM. Currently - encode and decode with the same vcm module.
|
||||
void RegisterReceiverVCM(webrtc::VideoCodingModule *vcm) { _VCMReceiver = vcm; }
|
||||
// Register exisitng VCM. Currently - encode and decode under same module.
|
||||
void RegisterReceiverVCM(webrtc::VideoCodingModule *vcm) {_VCMReceiver = vcm;}
|
||||
// Return size of last encoded frame encoded data (all frames in the sequence)
|
||||
// Good for only one call - after which will reset value (to allow detection of frame drop)
|
||||
// Good for only one call - after which will reset value
|
||||
// (to allow detection of frame drop)
|
||||
float EncodedBytes();
|
||||
// return encode complete (true/false)
|
||||
// Return encode complete (true/false)
|
||||
bool EncodeComplete();
|
||||
// Inform callback of codec used
|
||||
void SetCodecType(webrtc::RTPVideoCodecTypes codecType) { _codecType = codecType; }
|
||||
// inform callback of frame dimensions
|
||||
void SetCodecType(webrtc::RTPVideoCodecTypes codecType)
|
||||
{_codecType = codecType;}
|
||||
// Inform callback of frame dimensions
|
||||
void SetFrameDimensions(WebRtc_Word32 width, WebRtc_Word32 height)
|
||||
{
|
||||
_width = width;
|
||||
@ -94,8 +104,7 @@ public:
|
||||
void Initialize();
|
||||
void ResetByteCount();
|
||||
|
||||
// conversion function for payload type (needed for the callback function)
|
||||
// RTPVideoVideoCodecTypes ConvertPayloadType(WebRtc_UWord8 payloadType);
|
||||
// Conversion function for payload type (needed for the callback function)
|
||||
|
||||
private:
|
||||
FILE* _encodedFile;
|
||||
@ -111,7 +120,8 @@ private:
|
||||
|
||||
}; // end of VCMEncodeCompleteCallback
|
||||
|
||||
//Send Side - Packetization callback - packetize an encoded frame via the RTP module
|
||||
// Send Side - Packetization callback - packetize an encoded frame via the
|
||||
// RTP module
|
||||
class VCMRTPEncodeCompleteCallback: public webrtc::VCMPacketizationCallback
|
||||
{
|
||||
public:
|
||||
@ -122,7 +132,8 @@ public:
|
||||
_RTPModule(rtp) {}
|
||||
|
||||
virtual ~VCMRTPEncodeCompleteCallback() {}
|
||||
// process encoded data received from the encoder, pass stream to the RTP module
|
||||
// Process encoded data received from the encoder, pass stream to the
|
||||
// RTP module
|
||||
WebRtc_Word32 SendData(const webrtc::FrameType frameType,
|
||||
const WebRtc_UWord8 payloadType, const WebRtc_UWord32 timeStamp,
|
||||
const WebRtc_UWord8* payloadData, const WebRtc_UWord32 payloadSize,
|
||||
@ -134,9 +145,10 @@ public:
|
||||
// return encode complete (true/false)
|
||||
bool EncodeComplete();
|
||||
// Inform callback of codec used
|
||||
void SetCodecType(webrtc::RTPVideoCodecTypes codecType) { _codecType = codecType; }
|
||||
void SetCodecType(webrtc::RTPVideoCodecTypes codecType)
|
||||
{_codecType = codecType;}
|
||||
|
||||
// inform callback of frame dimensions
|
||||
// Inform callback of frame dimensions
|
||||
void SetFrameDimensions(WebRtc_Word16 width, WebRtc_Word16 height)
|
||||
{
|
||||
_width = width;
|
||||
@ -171,13 +183,21 @@ private:
|
||||
webrtc::VideoFrame _lastDecodedFrame;
|
||||
}; // end of VCMDecodeCompleCallback class
|
||||
|
||||
///
|
||||
|
||||
typedef struct
|
||||
{
|
||||
WebRtc_Word8 data[1650]; // max packet size
|
||||
WebRtc_Word32 length;
|
||||
WebRtc_Word64 receiveTime;
|
||||
} rtpPacket;
|
||||
|
||||
|
||||
class RTPSendCompleteCallback: public webrtc::Transport
|
||||
{
|
||||
public:
|
||||
// constructor input: (reeive side) rtp module to send encoded data to
|
||||
RTPSendCompleteCallback(webrtc::RtpRtcp* rtp,
|
||||
const char* filename = NULL);
|
||||
// constructor input: (receive side) rtp module to send encoded data to
|
||||
RTPSendCompleteCallback(webrtc::RtpRtcp* rtp,
|
||||
const char* filename = NULL);
|
||||
virtual ~RTPSendCompleteCallback();
|
||||
// Send Packet to receive side RTP module
|
||||
virtual int SendPacket(int channel, const void *data, int len);
|
||||
@ -187,21 +207,29 @@ public:
|
||||
void SetLossPct(double lossPct);
|
||||
// Set average size of burst loss
|
||||
void SetBurstLength(double burstLength);
|
||||
// return send count
|
||||
int SendCount() { return _sendCount; }
|
||||
// Set network delay in the network
|
||||
void SetNetworkDelay(WebRtc_UWord32 networkDelayMs)
|
||||
{_networkDelayMs = networkDelayMs;};
|
||||
// Return send count
|
||||
int SendCount() {return _sendCount; }
|
||||
// Return accumulated length in bytes of transmitted packets
|
||||
WebRtc_UWord32 TotalSentLength() {return _totalSentLength;}
|
||||
private:
|
||||
// randomly decide weather to drop a packet or not, based on the channel model
|
||||
// Randomly decide whether to drop packets, based on the channel model
|
||||
bool PacketLoss(double lossPct);
|
||||
|
||||
WebRtc_UWord32 _sendCount;
|
||||
webrtc::RtpRtcp* _rtp;
|
||||
double _lossPct;
|
||||
double _burstLength;
|
||||
bool _prevLossState;
|
||||
webrtc::RtpDump* _rtpDump;
|
||||
WebRtc_UWord32 _sendCount;
|
||||
webrtc::RtpRtcp* _rtp;
|
||||
double _lossPct;
|
||||
double _burstLength;
|
||||
WebRtc_UWord32 _networkDelayMs;
|
||||
bool _prevLossState;
|
||||
WebRtc_UWord32 _totalSentLength;
|
||||
webrtc::ListWrapper _rtpPackets;
|
||||
webrtc::RtpDump* _rtpDump;
|
||||
};
|
||||
|
||||
// used in multi thread test
|
||||
// Used in multi thread test
|
||||
class SendSharedState
|
||||
{
|
||||
public:
|
||||
@ -245,7 +273,7 @@ SSIMfromFiles(const WebRtc_Word8 *refFileName,
|
||||
const WebRtc_Word8 *testFileName, WebRtc_Word32 width,
|
||||
WebRtc_Word32 height, double *SSIMptr);
|
||||
|
||||
// codec type conversion
|
||||
// Codec type conversion
|
||||
webrtc::RTPVideoCodecTypes
|
||||
ConvertCodecType(const char* plname);
|
||||
|
||||
@ -255,7 +283,7 @@ public:
|
||||
SendStatsTest() : _frameRate(15) {}
|
||||
WebRtc_Word32 SendStatistics(const WebRtc_UWord32 bitRate,
|
||||
const WebRtc_UWord32 frameRate);
|
||||
void SetTargetFrameRate(WebRtc_UWord32 frameRate) { _frameRate = frameRate; }
|
||||
void SetTargetFrameRate(WebRtc_UWord32 frameRate) {_frameRate = frameRate;}
|
||||
private:
|
||||
WebRtc_UWord32 _frameRate;
|
||||
};
|
||||
|
||||
@ -121,6 +121,26 @@ int ParseArguments(int argc, char **argv, CmdArgs& args)
|
||||
args.testNum = n;
|
||||
break;
|
||||
}
|
||||
case 'p':
|
||||
{
|
||||
args.packetLoss = atoi(argv[i+1]);
|
||||
break;
|
||||
}
|
||||
case 'r':
|
||||
{
|
||||
args.rtt = atoi(argv[i+1]);
|
||||
break;
|
||||
}
|
||||
case 'm':
|
||||
{
|
||||
args.protectionMode = atoi(argv[i+1]);
|
||||
break;
|
||||
}
|
||||
case 'e':
|
||||
{
|
||||
args.camaEnable = atoi(argv[i+1]);
|
||||
break;
|
||||
}
|
||||
default:
|
||||
return -1;
|
||||
}
|
||||
@ -136,10 +156,12 @@ int main(int argc, char **argv)
|
||||
if (ParseArguments(argc, argv, args) != 0)
|
||||
{
|
||||
printf("Unable to parse input arguments\n");
|
||||
printf("args: -n <test #> -w <width> -h <height> -f <fps> -b <bps> -c <codec>"
|
||||
" -i <input file> -o <output file>\n");
|
||||
printf("args: -n <test #> -w <width> -h <height> -f <fps> -b <bps> "
|
||||
"-c <codec> -i <input file> -o <output file> -p <packet loss> "
|
||||
"-r <round-trip-time> -e <cama enable> -m <protection mode> \n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
int ret = 0;
|
||||
switch (args.testNum)
|
||||
{
|
||||
|
||||
Loading…
Reference in New Issue
Block a user