git-svn-id: http://webrtc.googlecode.com/svn/trunk@4 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
niklase@google.com
481
modules/audio_coding/main/test/Channel.cpp
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481
modules/audio_coding/main/test/Channel.cpp
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@@ -0,0 +1,481 @@
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/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include <assert.h>
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#include <iostream>
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#include "audio_coding_module.h"
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#include "Channel.h"
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#include "tick_util.h"
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#include "typedefs.h"
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#include "common_types.h"
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using namespace webrtc;
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WebRtc_Word32
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Channel::SendData(
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const FrameType frameType,
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const WebRtc_UWord8 payloadType,
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const WebRtc_UWord32 timeStamp,
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const WebRtc_UWord8* payloadData,
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const WebRtc_UWord16 payloadSize,
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const RTPFragmentationHeader* fragmentation)
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{
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WebRtcRTPHeader rtpInfo;
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WebRtc_Word32 status;
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WebRtc_UWord16 payloadDataSize = payloadSize;
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rtpInfo.header.markerBit = false;
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rtpInfo.header.ssrc = 0;
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rtpInfo.header.sequenceNumber = _seqNo++;
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rtpInfo.header.payloadType = payloadType;
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rtpInfo.header.timestamp = timeStamp;
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if(frameType == kAudioFrameCN)
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{
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rtpInfo.type.Audio.isCNG = true;
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}
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else
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{
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rtpInfo.type.Audio.isCNG = false;
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}
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if(frameType == kFrameEmpty)
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{
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// Skip this frame
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return 0;
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}
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rtpInfo.type.Audio.channel = 1;
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// Treat fragmentation separately
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if(fragmentation != NULL)
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{
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if((fragmentation->fragmentationTimeDiff[1] <= 0x3fff) && // silence for too long send only new data
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(fragmentation->fragmentationVectorSize == 2))
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{
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// only 0x80 if we have multiple blocks
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_payloadData[0] = 0x80 + fragmentation->fragmentationPlType[1];
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WebRtc_UWord32 REDheader = (((WebRtc_UWord32)fragmentation->fragmentationTimeDiff[1]) << 10) + fragmentation->fragmentationLength[1];
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_payloadData[1] = WebRtc_UWord8((REDheader >> 16) & 0x000000FF);
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_payloadData[2] = WebRtc_UWord8((REDheader >> 8) & 0x000000FF);
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_payloadData[3] = WebRtc_UWord8(REDheader & 0x000000FF);
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_payloadData[4] = fragmentation->fragmentationPlType[0];
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// copy the RED data
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memcpy(_payloadData + 5,
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payloadData + fragmentation->fragmentationOffset[1],
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fragmentation->fragmentationLength[1]);
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// copy the normal data
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memcpy(_payloadData + 5 + fragmentation->fragmentationLength[1],
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payloadData + fragmentation->fragmentationOffset[0],
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fragmentation->fragmentationLength[0]);
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payloadDataSize += 5;
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} else
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{
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// single block (newest one)
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memcpy(_payloadData,
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payloadData + fragmentation->fragmentationOffset[0],
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fragmentation->fragmentationLength[0]);
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payloadDataSize = WebRtc_UWord16(fragmentation->fragmentationLength[0]);
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rtpInfo.header.payloadType = fragmentation->fragmentationPlType[0];
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}
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}
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else
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{
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memcpy(_payloadData, payloadData, payloadDataSize);
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if(_isStereo)
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{
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if(_leftChannel)
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{
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memcpy(&_rtpInfo, &rtpInfo, sizeof(WebRtcRTPHeader));
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_leftChannel = false;
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rtpInfo.type.Audio.channel = 1;
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}
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else
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{
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memcpy(&rtpInfo, &_rtpInfo, sizeof(WebRtcRTPHeader));
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_leftChannel = true;
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rtpInfo.type.Audio.channel = 2;
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}
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}
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}
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_channelCritSect->Enter();
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if(_saveBitStream)
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{
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//fwrite(payloadData, sizeof(WebRtc_UWord8), payloadSize, _bitStreamFile);
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}
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if(!_isStereo)
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{
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CalcStatistics(rtpInfo, payloadSize);
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}
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_lastInTimestamp = timeStamp;
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_totalBytes += payloadDataSize;
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_channelCritSect->Leave();
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if(_useFECTestWithPacketLoss)
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{
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_packetLoss += 1;
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if(_packetLoss == 3)
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{
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_packetLoss = 0;
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return 0;
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}
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}
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//status = _receiverACM->IncomingPayload((WebRtc_Word8*)_payloadData, payloadSize, payloadType, timeStamp);
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status = _receiverACM->IncomingPacket((WebRtc_Word8*)_payloadData, payloadDataSize, rtpInfo);
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//delete [] payloadData;
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return status;
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}
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void
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Channel::CalcStatistics(
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WebRtcRTPHeader& rtpInfo,
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WebRtc_UWord16 payloadSize)
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{
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int n;
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if((rtpInfo.header.payloadType != _lastPayloadType) &&
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(_lastPayloadType != -1))
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{
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// payload-type is changed.
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// we have to terminate the calculations on the previous payload type
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// we ignore the last packet in that payload type just to make things
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// easier.
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for(n = 0; n < MAX_NUM_PAYLOADS; n++)
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{
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if(_lastPayloadType == _payloadStats[n].payloadType)
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{
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_payloadStats[n].newPacket = true;
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break;
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}
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}
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}
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_lastPayloadType = rtpInfo.header.payloadType;
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bool newPayload = true;
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ACMTestPayloadStats* currentPayloadStr;
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for(n = 0; n < MAX_NUM_PAYLOADS; n++)
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{
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if(rtpInfo.header.payloadType == _payloadStats[n].payloadType)
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{
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newPayload = false;
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currentPayloadStr = &_payloadStats[n];
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break;
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}
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}
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if(!newPayload)
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{
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if(!currentPayloadStr->newPacket)
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{
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WebRtc_UWord32 lastFrameSizeSample = (WebRtc_UWord32)((WebRtc_UWord32)rtpInfo.header.timestamp -
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(WebRtc_UWord32)currentPayloadStr->lastTimestamp);
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assert(lastFrameSizeSample > 0);
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int k = 0;
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while((currentPayloadStr->frameSizeStats[k].frameSizeSample !=
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lastFrameSizeSample) &&
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(currentPayloadStr->frameSizeStats[k].frameSizeSample != 0))
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{
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k++;
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}
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ACMTestFrameSizeStats* currentFrameSizeStats =
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&(currentPayloadStr->frameSizeStats[k]);
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currentFrameSizeStats->frameSizeSample = (WebRtc_Word16)lastFrameSizeSample;
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// increment the number of encoded samples.
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currentFrameSizeStats->totalEncodedSamples +=
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lastFrameSizeSample;
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// increment the number of recveived packets
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currentFrameSizeStats->numPackets++;
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// increment the total number of bytes (this is based on
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// the previous payload we don't know the frame-size of
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// the current payload.
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currentFrameSizeStats->totalPayloadLenByte +=
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currentPayloadStr->lastPayloadLenByte;
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// store the maximum payload-size (this is based on
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// the previous payload we don't know the frame-size of
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// the current payload.
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if(currentFrameSizeStats->maxPayloadLen <
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currentPayloadStr->lastPayloadLenByte)
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{
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currentFrameSizeStats->maxPayloadLen =
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currentPayloadStr->lastPayloadLenByte;
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}
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// store the current values for the next time
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currentPayloadStr->lastTimestamp = rtpInfo.header.timestamp;
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currentPayloadStr->lastPayloadLenByte = payloadSize;
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}
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else
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{
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currentPayloadStr->newPacket = false;
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currentPayloadStr->lastPayloadLenByte = payloadSize;
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currentPayloadStr->lastTimestamp = rtpInfo.header.timestamp;
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currentPayloadStr->payloadType = rtpInfo.header.payloadType;
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}
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}
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else
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{
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n = 0;
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while(_payloadStats[n].payloadType != -1)
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{
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n++;
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}
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// first packet
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_payloadStats[n].newPacket = false;
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_payloadStats[n].lastPayloadLenByte = payloadSize;
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_payloadStats[n].lastTimestamp = rtpInfo.header.timestamp;
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_payloadStats[n].payloadType = rtpInfo.header.payloadType;
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}
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}
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Channel::Channel(WebRtc_Word16 chID) :
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_receiverACM(NULL),
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_seqNo(0),
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_channelCritSect(CriticalSectionWrapper::CreateCriticalSection()),
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_bitStreamFile(NULL),
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_saveBitStream(false),
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_lastPayloadType(-1),
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_isStereo(false),
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_leftChannel(true),
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_useFECTestWithPacketLoss(false),
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_packetLoss(0),
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_lastInTimestamp(0),
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_chID(chID),
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_beginTime(TickTime::MillisecondTimestamp()),
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_totalBytes(0)
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{
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int n;
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int k;
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for(n = 0; n < MAX_NUM_PAYLOADS; n++)
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{
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_payloadStats[n].payloadType = -1;
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_payloadStats[n].newPacket = true;
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for(k = 0; k < MAX_NUM_FRAMESIZES; k++)
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{
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_payloadStats[n].frameSizeStats[k].frameSizeSample = 0;
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_payloadStats[n].frameSizeStats[k].maxPayloadLen = 0;
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_payloadStats[n].frameSizeStats[k].numPackets = 0;
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_payloadStats[n].frameSizeStats[k].totalPayloadLenByte = 0;
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_payloadStats[n].frameSizeStats[k].totalEncodedSamples = 0;
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}
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}
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if(chID >= 0)
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{
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_saveBitStream = true;
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char bitStreamFileName[500];
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sprintf(bitStreamFileName, "bitStream_%d.dat", chID);
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_bitStreamFile = fopen(bitStreamFileName, "wb");
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}
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else
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{
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_saveBitStream = false;
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}
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}
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Channel::~Channel()
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{
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delete _channelCritSect;
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}
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void
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Channel::RegisterReceiverACM(AudioCodingModule* acm)
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{
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_receiverACM = acm;
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return;
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}
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void
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Channel::ResetStats()
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{
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int n;
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int k;
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_channelCritSect->Enter();
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_lastPayloadType = -1;
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for(n = 0; n < MAX_NUM_PAYLOADS; n++)
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{
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_payloadStats[n].payloadType = -1;
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_payloadStats[n].newPacket = true;
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for(k = 0; k < MAX_NUM_FRAMESIZES; k++)
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{
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_payloadStats[n].frameSizeStats[k].frameSizeSample = 0;
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_payloadStats[n].frameSizeStats[k].maxPayloadLen = 0;
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_payloadStats[n].frameSizeStats[k].numPackets = 0;
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_payloadStats[n].frameSizeStats[k].totalPayloadLenByte = 0;
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_payloadStats[n].frameSizeStats[k].totalEncodedSamples = 0;
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}
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}
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_beginTime = TickTime::MillisecondTimestamp();
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_totalBytes = 0;
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_channelCritSect->Leave();
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}
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WebRtc_Word16
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Channel::Stats(CodecInst& codecInst, ACMTestPayloadStats& payloadStats)
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{
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_channelCritSect->Enter();
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int n;
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payloadStats.payloadType = -1;
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for(n = 0; n < MAX_NUM_PAYLOADS; n++)
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{
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if(_payloadStats[n].payloadType == codecInst.pltype)
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{
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memcpy(&payloadStats, &_payloadStats[n], sizeof(ACMTestPayloadStats));
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break;
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}
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}
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if(payloadStats.payloadType == -1)
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{
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_channelCritSect->Leave();
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return -1;
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}
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for(n = 0; n < MAX_NUM_FRAMESIZES; n++)
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{
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if(payloadStats.frameSizeStats[n].frameSizeSample == 0)
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{
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_channelCritSect->Leave();
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return 0;
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}
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payloadStats.frameSizeStats[n].usageLenSec =
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(double)payloadStats.frameSizeStats[n].totalEncodedSamples
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/ (double)codecInst.plfreq;
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payloadStats.frameSizeStats[n].rateBitPerSec =
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payloadStats.frameSizeStats[n].totalPayloadLenByte * 8 /
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payloadStats.frameSizeStats[n].usageLenSec;
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}
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_channelCritSect->Leave();
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return 0;
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}
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void
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Channel::Stats(WebRtc_UWord32* numPackets)
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{
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_channelCritSect->Enter();
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int k;
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int n;
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memset(numPackets, 0, MAX_NUM_PAYLOADS * sizeof(WebRtc_UWord32));
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for(k = 0; k < MAX_NUM_PAYLOADS; k++)
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{
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if(_payloadStats[k].payloadType == -1)
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{
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break;
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}
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numPackets[k] = 0;
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for(n = 0; n < MAX_NUM_FRAMESIZES; n++)
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{
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if(_payloadStats[k].frameSizeStats[n].frameSizeSample == 0)
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{
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break;
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}
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numPackets[k] +=
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_payloadStats[k].frameSizeStats[n].numPackets;
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}
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}
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_channelCritSect->Leave();
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}
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void
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Channel::Stats(WebRtc_UWord8* payloadType, WebRtc_UWord32* payloadLenByte)
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{
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_channelCritSect->Enter();
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int k;
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int n;
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memset(payloadLenByte, 0, MAX_NUM_PAYLOADS * sizeof(WebRtc_UWord32));
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for(k = 0; k < MAX_NUM_PAYLOADS; k++)
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{
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if(_payloadStats[k].payloadType == -1)
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{
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break;
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}
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payloadType[k] = (WebRtc_UWord8)_payloadStats[k].payloadType;
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payloadLenByte[k] = 0;
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for(n = 0; n < MAX_NUM_FRAMESIZES; n++)
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{
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if(_payloadStats[k].frameSizeStats[n].frameSizeSample == 0)
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{
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break;
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}
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payloadLenByte[k] += (WebRtc_UWord16)
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_payloadStats[k].frameSizeStats[n].totalPayloadLenByte;
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}
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}
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_channelCritSect->Leave();
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}
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void
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Channel::PrintStats(CodecInst& codecInst)
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{
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ACMTestPayloadStats payloadStats;
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Stats(codecInst, payloadStats);
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printf("%s %d kHz\n",
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codecInst.plname,
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codecInst.plfreq / 1000);
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printf("=====================================================\n");
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if(payloadStats.payloadType == -1)
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{
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printf("No Packets are sent with payload-type %d (%s)\n\n",
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codecInst.pltype,
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codecInst.plname);
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return;
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}
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for(int k = 0; k < MAX_NUM_FRAMESIZES; k++)
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{
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if(payloadStats.frameSizeStats[k].frameSizeSample == 0)
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{
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break;
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}
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printf("Frame-size.................... %d samples\n",
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payloadStats.frameSizeStats[k].frameSizeSample);
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printf("Average Rate.................. %.0f bits/sec\n",
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payloadStats.frameSizeStats[k].rateBitPerSec);
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printf("Maximum Payload-Size.......... %d Bytes\n",
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payloadStats.frameSizeStats[k].maxPayloadLen);
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printf("Maximum Instantaneous Rate.... %.0f bits/sec\n",
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((double)payloadStats.frameSizeStats[k].maxPayloadLen * 8.0 *
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(double)codecInst.plfreq) /
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(double)payloadStats.frameSizeStats[k].frameSizeSample);
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printf("Number of Packets............. %u\n",
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(unsigned int)payloadStats.frameSizeStats[k].numPackets);
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printf("Duration...................... %0.3f sec\n\n",
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payloadStats.frameSizeStats[k].usageLenSec);
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}
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}
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WebRtc_UWord32
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Channel::LastInTimestamp()
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{
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WebRtc_UWord32 timestamp;
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_channelCritSect->Enter();
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timestamp = _lastInTimestamp;
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_channelCritSect->Leave();
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return timestamp;
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}
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double
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Channel::BitRate()
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{
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double rate;
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WebRtc_UWord64 currTime = TickTime::MillisecondTimestamp();
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_channelCritSect->Enter();
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rate = ((double)_totalBytes * 8.0)/ (double)(currTime - _beginTime);
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_channelCritSect->Leave();
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return rate;
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}
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