src/modules/audio_coding/main/test/Channel.cc - vendor/opensource/webrtc - Git at Google

 /*
  *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */

 #include <assert.h>
 #include <iostream>

 #include "audio_coding_module.h"
 #include "Channel.h"
 #include "tick_util.h"
 #include "typedefs.h"
 #include "common_types.h"

 namespace webrtc {

 WebRtc_Word32
 Channel::SendData(
         const FrameType       frameType,
         const WebRtc_UWord8   payloadType,
         const WebRtc_UWord32  timeStamp,
         const WebRtc_UWord8*  payloadData,
         const WebRtc_UWord16  payloadSize,
         const RTPFragmentationHeader* fragmentation)
 {
     WebRtcRTPHeader rtpInfo;
     WebRtc_Word32   status;
     WebRtc_UWord16  payloadDataSize = payloadSize;

     rtpInfo.header.markerBit = false;
     rtpInfo.header.ssrc = 0;
     rtpInfo.header.sequenceNumber = _seqNo++;
     rtpInfo.header.payloadType = payloadType;
     rtpInfo.header.timestamp = timeStamp;
     if(frameType == kAudioFrameCN)
     {
         rtpInfo.type.Audio.isCNG = true;
     }
     else
     {
         rtpInfo.type.Audio.isCNG = false;
     }
     if(frameType == kFrameEmpty)
     {
         // Skip this frame
         return 0;
     }

     rtpInfo.type.Audio.channel = 1;
     // Treat fragmentation separately
     if(fragmentation != NULL)
     {
         if((fragmentation->fragmentationTimeDiff[1] <= 0x3fff) && // silence for too long send only new data
             (fragmentation->fragmentationVectorSize == 2))
         {
             // only 0x80 if we have multiple blocks
             _payloadData[0] = 0x80 + fragmentation->fragmentationPlType[1];
             WebRtc_UWord32 REDheader =  (((WebRtc_UWord32)fragmentation->fragmentationTimeDiff[1]) << 10) + fragmentation->fragmentationLength[1];
             _payloadData[1] = WebRtc_UWord8((REDheader >> 16) & 0x000000FF);
             _payloadData[2] = WebRtc_UWord8((REDheader >> 8) & 0x000000FF);
             _payloadData[3] = WebRtc_UWord8(REDheader & 0x000000FF);

             _payloadData[4] = fragmentation->fragmentationPlType[0];
             // copy the RED data
             memcpy(_payloadData + 5,
                 payloadData + fragmentation->fragmentationOffset[1],
                 fragmentation->fragmentationLength[1]);
             // copy the normal data
             memcpy(_payloadData + 5 + fragmentation->fragmentationLength[1],
                 payloadData + fragmentation->fragmentationOffset[0],
                 fragmentation->fragmentationLength[0]);
             payloadDataSize += 5;
         } else
         {
             // single block (newest one)
             memcpy(_payloadData,
                 payloadData + fragmentation->fragmentationOffset[0],
                 fragmentation->fragmentationLength[0]);
             payloadDataSize = WebRtc_UWord16(fragmentation->fragmentationLength[0]);
             rtpInfo.header.payloadType = fragmentation->fragmentationPlType[0];
         }
     }
     else
     {
         memcpy(_payloadData, payloadData, payloadDataSize);
         if(_isStereo)
         {
             if(_leftChannel)
             {
                 memcpy(&_rtpInfo, &rtpInfo, sizeof(WebRtcRTPHeader));
                 _leftChannel = false;
                 rtpInfo.type.Audio.channel = 1;
             }
             else
             {
                 memcpy(&rtpInfo, &_rtpInfo, sizeof(WebRtcRTPHeader));
                 _leftChannel = true;
                 rtpInfo.type.Audio.channel = 2;
             }
         }
     }

     _channelCritSect->Enter();
     if(_saveBitStream)
     {
         //fwrite(payloadData, sizeof(WebRtc_UWord8), payloadSize, _bitStreamFile);
     }

     if(!_isStereo)
     {
         CalcStatistics(rtpInfo, payloadSize);
     }
     _lastInTimestamp = timeStamp;
     _totalBytes += payloadDataSize;
     _channelCritSect->Leave();

     if(_useFECTestWithPacketLoss)
     {
         _packetLoss += 1;
         if(_packetLoss == 3)
         {
             _packetLoss = 0;
             return 0;
         }
     }


     //status = _receiverACM->IncomingPayload((WebRtc_Word8*)_payloadData, payloadSize, payloadType, timeStamp);
     status = _receiverACM->IncomingPacket((WebRtc_Word8*)_payloadData, payloadDataSize, rtpInfo);

     //delete [] payloadData;


     return status;
 }

 void
 Channel::CalcStatistics(
     WebRtcRTPHeader& rtpInfo,
     WebRtc_UWord16   payloadSize)
 {
     int n;
     if((rtpInfo.header.payloadType != _lastPayloadType) &&
         (_lastPayloadType != -1))
     {
         // payload-type is changed.
         // we have to terminate the calculations on the previous payload type
         // we ignore the last packet in that payload type just to make things
         // easier.
         for(n = 0; n < MAX_NUM_PAYLOADS; n++)
         {
             if(_lastPayloadType == _payloadStats[n].payloadType)
             {
                 _payloadStats[n].newPacket = true;
                 break;
             }
         }
     }
     _lastPayloadType = rtpInfo.header.payloadType;

     bool newPayload = true;
     ACMTestPayloadStats* currentPayloadStr;
     for(n = 0; n < MAX_NUM_PAYLOADS; n++)
     {
         if(rtpInfo.header.payloadType == _payloadStats[n].payloadType)
         {
             newPayload = false;
             currentPayloadStr = &_payloadStats[n];
             break;
         }
     }

     if(!newPayload)
     {
         if(!currentPayloadStr->newPacket)
         {
             WebRtc_UWord32 lastFrameSizeSample = (WebRtc_UWord32)((WebRtc_UWord32)rtpInfo.header.timestamp -
                 (WebRtc_UWord32)currentPayloadStr->lastTimestamp);
             assert(lastFrameSizeSample > 0);
             int k = 0;
             while((currentPayloadStr->frameSizeStats[k].frameSizeSample !=
                 lastFrameSizeSample) &&
                 (currentPayloadStr->frameSizeStats[k].frameSizeSample != 0))
             {
                 k++;
             }
             ACMTestFrameSizeStats* currentFrameSizeStats =
                 &(currentPayloadStr->frameSizeStats[k]);
             currentFrameSizeStats->frameSizeSample = (WebRtc_Word16)lastFrameSizeSample;

             // increment the number of encoded samples.
             currentFrameSizeStats->totalEncodedSamples +=
                 lastFrameSizeSample;
             // increment the number of recveived packets
             currentFrameSizeStats->numPackets++;
             // increment the total number of bytes (this is based on
             // the previous payload we don't know the frame-size of
             // the current payload.
             currentFrameSizeStats->totalPayloadLenByte +=
                 currentPayloadStr->lastPayloadLenByte;
             // store the maximum payload-size (this is based on
             // the previous payload we don't know the frame-size of
             // the current payload.
             if(currentFrameSizeStats->maxPayloadLen <
                 currentPayloadStr->lastPayloadLenByte)
             {
                 currentFrameSizeStats->maxPayloadLen =
                     currentPayloadStr->lastPayloadLenByte;
             }
             // store the current values for the next time
             currentPayloadStr->lastTimestamp = rtpInfo.header.timestamp;
             currentPayloadStr->lastPayloadLenByte = payloadSize;
         }
         else
         {
             currentPayloadStr->newPacket          = false;
             currentPayloadStr->lastPayloadLenByte = payloadSize;
             currentPayloadStr->lastTimestamp      = rtpInfo.header.timestamp;
             currentPayloadStr->payloadType        = rtpInfo.header.payloadType;
         }
     }
     else
     {
         n = 0;
         while(_payloadStats[n].payloadType != -1)
         {
             n++;
         }
         // first packet
         _payloadStats[n].newPacket          = false;
         _payloadStats[n].lastPayloadLenByte = payloadSize;
         _payloadStats[n].lastTimestamp      = rtpInfo.header.timestamp;
         _payloadStats[n].payloadType        = rtpInfo.header.payloadType;
     }
 }

 Channel::Channel(WebRtc_Word16 chID) :
 _receiverACM(NULL),
 _seqNo(0),
 _channelCritSect(CriticalSectionWrapper::CreateCriticalSection()),
 _bitStreamFile(NULL),
 _saveBitStream(false),
 _lastPayloadType(-1),
 _isStereo(false),
 _leftChannel(true),
 _lastInTimestamp(0),
 _packetLoss(0),
 _useFECTestWithPacketLoss(false),
 _chID(chID),
 _beginTime(TickTime::MillisecondTimestamp()),
 _totalBytes(0)
 {
     int n;
     int k;
     for(n = 0; n < MAX_NUM_PAYLOADS; n++)
     {
         _payloadStats[n].payloadType = -1;
         _payloadStats[n].newPacket   = true;
         for(k = 0; k < MAX_NUM_FRAMESIZES; k++)
         {
             _payloadStats[n].frameSizeStats[k].frameSizeSample = 0;
             _payloadStats[n].frameSizeStats[k].maxPayloadLen   = 0;
             _payloadStats[n].frameSizeStats[k].numPackets      = 0;
             _payloadStats[n].frameSizeStats[k].totalPayloadLenByte = 0;
             _payloadStats[n].frameSizeStats[k].totalEncodedSamples = 0;
         }
     }
     if(chID >= 0)
     {
         _saveBitStream = true;
         char bitStreamFileName[500];
         sprintf(bitStreamFileName, "bitStream_%d.dat", chID);
         _bitStreamFile = fopen(bitStreamFileName, "wb");
     }
     else
     {
         _saveBitStream = false;
     }
 }

 Channel::~Channel()
 {
     delete _channelCritSect;
 }

 void
 Channel::RegisterReceiverACM(AudioCodingModule* acm)
 {
     _receiverACM = acm;
     return;
 }

 void
 Channel::ResetStats()
 {
     int n;
     int k;
     _channelCritSect->Enter();
     _lastPayloadType = -1;
     for(n = 0; n < MAX_NUM_PAYLOADS; n++)
     {
         _payloadStats[n].payloadType = -1;
         _payloadStats[n].newPacket   = true;
         for(k = 0; k < MAX_NUM_FRAMESIZES; k++)
         {
             _payloadStats[n].frameSizeStats[k].frameSizeSample = 0;
             _payloadStats[n].frameSizeStats[k].maxPayloadLen   = 0;
             _payloadStats[n].frameSizeStats[k].numPackets      = 0;
             _payloadStats[n].frameSizeStats[k].totalPayloadLenByte = 0;
             _payloadStats[n].frameSizeStats[k].totalEncodedSamples = 0;
         }
     }
     _beginTime = TickTime::MillisecondTimestamp();
     _totalBytes = 0;
     _channelCritSect->Leave();
 }

 WebRtc_Word16
 Channel::Stats(CodecInst& codecInst, ACMTestPayloadStats& payloadStats)
 {
     _channelCritSect->Enter();
     int n;
     payloadStats.payloadType = -1;
     for(n = 0; n < MAX_NUM_PAYLOADS; n++)
     {
         if(_payloadStats[n].payloadType == codecInst.pltype)
         {
             memcpy(&payloadStats, &_payloadStats[n], sizeof(ACMTestPayloadStats));
             break;
         }
     }
     if(payloadStats.payloadType == -1)
     {
         _channelCritSect->Leave();
         return -1;
     }
     for(n = 0; n < MAX_NUM_FRAMESIZES; n++)
     {
         if(payloadStats.frameSizeStats[n].frameSizeSample == 0)
         {
             _channelCritSect->Leave();
             return 0;
         }
         payloadStats.frameSizeStats[n].usageLenSec =
             (double)payloadStats.frameSizeStats[n].totalEncodedSamples
             / (double)codecInst.plfreq;

         payloadStats.frameSizeStats[n].rateBitPerSec =
             payloadStats.frameSizeStats[n].totalPayloadLenByte * 8 /
             payloadStats.frameSizeStats[n].usageLenSec;

     }
     _channelCritSect->Leave();
     return 0;
 }

 void
 Channel::Stats(WebRtc_UWord32* numPackets)
 {
     _channelCritSect->Enter();
     int k;
     int n;
     memset(numPackets, 0, MAX_NUM_PAYLOADS * sizeof(WebRtc_UWord32));
     for(k = 0; k < MAX_NUM_PAYLOADS; k++)
     {
         if(_payloadStats[k].payloadType == -1)
         {
             break;
         }
         numPackets[k] = 0;
         for(n = 0; n < MAX_NUM_FRAMESIZES; n++)
         {
             if(_payloadStats[k].frameSizeStats[n].frameSizeSample == 0)
             {
                 break;
             }
             numPackets[k] +=
                 _payloadStats[k].frameSizeStats[n].numPackets;
         }
     }
     _channelCritSect->Leave();
 }

 void
 Channel::Stats(WebRtc_UWord8* payloadType, WebRtc_UWord32* payloadLenByte)
 {
     _channelCritSect->Enter();

     int k;
     int n;
     memset(payloadLenByte, 0, MAX_NUM_PAYLOADS * sizeof(WebRtc_UWord32));
     for(k = 0; k < MAX_NUM_PAYLOADS; k++)
     {
         if(_payloadStats[k].payloadType == -1)
         {
             break;
         }
         payloadType[k] = (WebRtc_UWord8)_payloadStats[k].payloadType;
         payloadLenByte[k] = 0;
         for(n = 0; n < MAX_NUM_FRAMESIZES; n++)
         {
             if(_payloadStats[k].frameSizeStats[n].frameSizeSample == 0)
             {
                 break;
             }
             payloadLenByte[k] += (WebRtc_UWord16)
                 _payloadStats[k].frameSizeStats[n].totalPayloadLenByte;
         }
     }

     _channelCritSect->Leave();
 }


 void
 Channel::PrintStats(CodecInst& codecInst)
 {
     ACMTestPayloadStats payloadStats;
     Stats(codecInst, payloadStats);
     printf("%s %d kHz\n",
         codecInst.plname,
         codecInst.plfreq / 1000);
     printf("=====================================================\n");
     if(payloadStats.payloadType == -1)
     {
         printf("No Packets are sent with payload-type %d (%s)\n\n",
             codecInst.pltype,
             codecInst.plname);
         return;
     }
     for(int k = 0; k < MAX_NUM_FRAMESIZES; k++)
     {
         if(payloadStats.frameSizeStats[k].frameSizeSample == 0)
         {
             break;
         }
         printf("Frame-size.................... %d samples\n",
             payloadStats.frameSizeStats[k].frameSizeSample);
         printf("Average Rate.................. %.0f bits/sec\n",
             payloadStats.frameSizeStats[k].rateBitPerSec);
         printf("Maximum Payload-Size.......... %d Bytes\n",
             payloadStats.frameSizeStats[k].maxPayloadLen);
         printf("Maximum Instantaneous Rate.... %.0f bits/sec\n",
             ((double)payloadStats.frameSizeStats[k].maxPayloadLen * 8.0 *
             (double)codecInst.plfreq) /
             (double)payloadStats.frameSizeStats[k].frameSizeSample);
         printf("Number of Packets............. %u\n",
                (unsigned int)payloadStats.frameSizeStats[k].numPackets);
         printf("Duration...................... %0.3f sec\n\n",
             payloadStats.frameSizeStats[k].usageLenSec);

     }

 }

 WebRtc_UWord32
 Channel::LastInTimestamp()
 {
     WebRtc_UWord32 timestamp;
     _channelCritSect->Enter();
     timestamp = _lastInTimestamp;
     _channelCritSect->Leave();
     return timestamp;
 }

 double
 Channel::BitRate()
 {
     double rate;
     WebRtc_UWord64 currTime = TickTime::MillisecondTimestamp();
     _channelCritSect->Enter();
     rate =   ((double)_totalBytes * 8.0)/ (double)(currTime - _beginTime);
     _channelCritSect->Leave();
     return rate;
 }

 } // namespace webrtc
	/*
	* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
	*
	* Use of this source code is governed by a BSD-style license
	* that can be found in the LICENSE file in the root of the source
	* tree. An additional intellectual property rights grant can be found
	* in the file PATENTS. All contributing project authors may
	* be found in the AUTHORS file in the root of the source tree.
	*/

	#include <assert.h>
	#include <iostream>

	#include "audio_coding_module.h"
	#include "Channel.h"
	#include "tick_util.h"
	#include "typedefs.h"
	#include "common_types.h"

	namespace webrtc {

	WebRtc_Word32
	Channel::SendData(
	const FrameType frameType,
	const WebRtc_UWord8 payloadType,
	const WebRtc_UWord32 timeStamp,
	const WebRtc_UWord8* payloadData,
	const WebRtc_UWord16 payloadSize,
	const RTPFragmentationHeader* fragmentation)
	{
	WebRtcRTPHeader rtpInfo;
	WebRtc_Word32 status;
	WebRtc_UWord16 payloadDataSize = payloadSize;

	rtpInfo.header.markerBit = false;
	rtpInfo.header.ssrc = 0;
	rtpInfo.header.sequenceNumber = _seqNo++;
	rtpInfo.header.payloadType = payloadType;
	rtpInfo.header.timestamp = timeStamp;
	if(frameType == kAudioFrameCN)
	{
	rtpInfo.type.Audio.isCNG = true;
	}
	else
	{
	rtpInfo.type.Audio.isCNG = false;
	}
	if(frameType == kFrameEmpty)
	{
	// Skip this frame
	return 0;
	}

	rtpInfo.type.Audio.channel = 1;
	// Treat fragmentation separately
	if(fragmentation != NULL)
	{
	if((fragmentation->fragmentationTimeDiff[1] <= 0x3fff) && // silence for too long send only new data
	(fragmentation->fragmentationVectorSize == 2))
	{
	// only 0x80 if we have multiple blocks
	_payloadData[0] = 0x80 + fragmentation->fragmentationPlType[1];
	WebRtc_UWord32 REDheader = (((WebRtc_UWord32)fragmentation->fragmentationTimeDiff[1]) << 10) + fragmentation->fragmentationLength[1];
	_payloadData[1] = WebRtc_UWord8((REDheader >> 16) & 0x000000FF);
	_payloadData[2] = WebRtc_UWord8((REDheader >> 8) & 0x000000FF);
	_payloadData[3] = WebRtc_UWord8(REDheader & 0x000000FF);

	_payloadData[4] = fragmentation->fragmentationPlType[0];
	// copy the RED data
	memcpy(_payloadData + 5,
	payloadData + fragmentation->fragmentationOffset[1],
	fragmentation->fragmentationLength[1]);
	// copy the normal data
	memcpy(_payloadData + 5 + fragmentation->fragmentationLength[1],
	payloadData + fragmentation->fragmentationOffset[0],
	fragmentation->fragmentationLength[0]);
	payloadDataSize += 5;
	} else
	{
	// single block (newest one)
	memcpy(_payloadData,
	payloadData + fragmentation->fragmentationOffset[0],
	fragmentation->fragmentationLength[0]);
	payloadDataSize = WebRtc_UWord16(fragmentation->fragmentationLength[0]);
	rtpInfo.header.payloadType = fragmentation->fragmentationPlType[0];
	}
	}
	else
	{
	memcpy(_payloadData, payloadData, payloadDataSize);
	if(_isStereo)
	{
	if(_leftChannel)
	{
	memcpy(&_rtpInfo, &rtpInfo, sizeof(WebRtcRTPHeader));
	_leftChannel = false;
	rtpInfo.type.Audio.channel = 1;
	}
	else
	{
	memcpy(&rtpInfo, &_rtpInfo, sizeof(WebRtcRTPHeader));
	_leftChannel = true;
	rtpInfo.type.Audio.channel = 2;
	}
	}
	}

	_channelCritSect->Enter();
	if(_saveBitStream)
	{
	//fwrite(payloadData, sizeof(WebRtc_UWord8), payloadSize, _bitStreamFile);
	}

	if(!_isStereo)
	{
	CalcStatistics(rtpInfo, payloadSize);
	}
	_lastInTimestamp = timeStamp;
	_totalBytes += payloadDataSize;
	_channelCritSect->Leave();

	if(_useFECTestWithPacketLoss)
	{
	_packetLoss += 1;
	if(_packetLoss == 3)
	{
	_packetLoss = 0;
	return 0;
	}
	}


	//status = _receiverACM->IncomingPayload((WebRtc_Word8*)_payloadData, payloadSize, payloadType, timeStamp);
	status = _receiverACM->IncomingPacket((WebRtc_Word8*)_payloadData, payloadDataSize, rtpInfo);

	//delete [] payloadData;



	return status;
	}

	void
	Channel::CalcStatistics(
	WebRtcRTPHeader& rtpInfo,
	WebRtc_UWord16 payloadSize)
	{
	int n;
	if((rtpInfo.header.payloadType != _lastPayloadType) &&
	(_lastPayloadType != -1))
	{
	// payload-type is changed.
	// we have to terminate the calculations on the previous payload type
	// we ignore the last packet in that payload type just to make things
	// easier.
	for(n = 0; n < MAX_NUM_PAYLOADS; n++)
	{
	if(_lastPayloadType == _payloadStats[n].payloadType)
	{
	_payloadStats[n].newPacket = true;
	break;
	}
	}
	}
	_lastPayloadType = rtpInfo.header.payloadType;

	bool newPayload = true;
	ACMTestPayloadStats* currentPayloadStr;
	for(n = 0; n < MAX_NUM_PAYLOADS; n++)
	{
	if(rtpInfo.header.payloadType == _payloadStats[n].payloadType)
	{
	newPayload = false;
	currentPayloadStr = &_payloadStats[n];
	break;
	}
	}

	if(!newPayload)
	{
	if(!currentPayloadStr->newPacket)
	{
	WebRtc_UWord32 lastFrameSizeSample = (WebRtc_UWord32)((WebRtc_UWord32)rtpInfo.header.timestamp -
	(WebRtc_UWord32)currentPayloadStr->lastTimestamp);
	assert(lastFrameSizeSample > 0);
	int k = 0;
	while((currentPayloadStr->frameSizeStats[k].frameSizeSample !=
	lastFrameSizeSample) &&
	(currentPayloadStr->frameSizeStats[k].frameSizeSample != 0))
	{
	k++;
	}
	ACMTestFrameSizeStats* currentFrameSizeStats =
	&(currentPayloadStr->frameSizeStats[k]);
	currentFrameSizeStats->frameSizeSample = (WebRtc_Word16)lastFrameSizeSample;

	// increment the number of encoded samples.
	currentFrameSizeStats->totalEncodedSamples +=
	lastFrameSizeSample;
	// increment the number of recveived packets
	currentFrameSizeStats->numPackets++;
	// increment the total number of bytes (this is based on
	// the previous payload we don't know the frame-size of
	// the current payload.
	currentFrameSizeStats->totalPayloadLenByte +=
	currentPayloadStr->lastPayloadLenByte;
	// store the maximum payload-size (this is based on
	// the previous payload we don't know the frame-size of
	// the current payload.
	if(currentFrameSizeStats->maxPayloadLen <
	currentPayloadStr->lastPayloadLenByte)
	{
	currentFrameSizeStats->maxPayloadLen =
	currentPayloadStr->lastPayloadLenByte;
	}
	// store the current values for the next time
	currentPayloadStr->lastTimestamp = rtpInfo.header.timestamp;
	currentPayloadStr->lastPayloadLenByte = payloadSize;
	}
	else
	{
	currentPayloadStr->newPacket = false;
	currentPayloadStr->lastPayloadLenByte = payloadSize;
	currentPayloadStr->lastTimestamp = rtpInfo.header.timestamp;
	currentPayloadStr->payloadType = rtpInfo.header.payloadType;
	}
	}
	else
	{
	n = 0;
	while(_payloadStats[n].payloadType != -1)
	{
	n++;
	}
	// first packet
	_payloadStats[n].newPacket = false;
	_payloadStats[n].lastPayloadLenByte = payloadSize;
	_payloadStats[n].lastTimestamp = rtpInfo.header.timestamp;
	_payloadStats[n].payloadType = rtpInfo.header.payloadType;
	}
	}

	Channel::Channel(WebRtc_Word16 chID) :
	_receiverACM(NULL),
	_seqNo(0),
	_channelCritSect(CriticalSectionWrapper::CreateCriticalSection()),
	_bitStreamFile(NULL),
	_saveBitStream(false),
	_lastPayloadType(-1),
	_isStereo(false),
	_leftChannel(true),
	_lastInTimestamp(0),
	_packetLoss(0),
	_useFECTestWithPacketLoss(false),
	_chID(chID),
	_beginTime(TickTime::MillisecondTimestamp()),
	_totalBytes(0)
	{
	int n;
	int k;
	for(n = 0; n < MAX_NUM_PAYLOADS; n++)
	{
	_payloadStats[n].payloadType = -1;
	_payloadStats[n].newPacket = true;
	for(k = 0; k < MAX_NUM_FRAMESIZES; k++)
	{
	_payloadStats[n].frameSizeStats[k].frameSizeSample = 0;
	_payloadStats[n].frameSizeStats[k].maxPayloadLen = 0;
	_payloadStats[n].frameSizeStats[k].numPackets = 0;
	_payloadStats[n].frameSizeStats[k].totalPayloadLenByte = 0;
	_payloadStats[n].frameSizeStats[k].totalEncodedSamples = 0;
	}
	}
	if(chID >= 0)
	{
	_saveBitStream = true;
	char bitStreamFileName[500];
	sprintf(bitStreamFileName, "bitStream_%d.dat", chID);
	_bitStreamFile = fopen(bitStreamFileName, "wb");
	}
	else
	{
	_saveBitStream = false;
	}
	}

	Channel::~Channel()
	{
	delete _channelCritSect;
	}

	void
	Channel::RegisterReceiverACM(AudioCodingModule* acm)
	{
	_receiverACM = acm;
	return;
	}

	void
	Channel::ResetStats()
	{
	int n;
	int k;
	_channelCritSect->Enter();
	_lastPayloadType = -1;
	for(n = 0; n < MAX_NUM_PAYLOADS; n++)
	{
	_payloadStats[n].payloadType = -1;
	_payloadStats[n].newPacket = true;
	for(k = 0; k < MAX_NUM_FRAMESIZES; k++)
	{
	_payloadStats[n].frameSizeStats[k].frameSizeSample = 0;
	_payloadStats[n].frameSizeStats[k].maxPayloadLen = 0;
	_payloadStats[n].frameSizeStats[k].numPackets = 0;
	_payloadStats[n].frameSizeStats[k].totalPayloadLenByte = 0;
	_payloadStats[n].frameSizeStats[k].totalEncodedSamples = 0;
	}
	}
	_beginTime = TickTime::MillisecondTimestamp();
	_totalBytes = 0;
	_channelCritSect->Leave();
	}

	WebRtc_Word16
	Channel::Stats(CodecInst& codecInst, ACMTestPayloadStats& payloadStats)
	{
	_channelCritSect->Enter();
	int n;
	payloadStats.payloadType = -1;
	for(n = 0; n < MAX_NUM_PAYLOADS; n++)
	{
	if(_payloadStats[n].payloadType == codecInst.pltype)
	{
	memcpy(&payloadStats, &_payloadStats[n], sizeof(ACMTestPayloadStats));
	break;
	}
	}
	if(payloadStats.payloadType == -1)
	{
	_channelCritSect->Leave();
	return -1;
	}
	for(n = 0; n < MAX_NUM_FRAMESIZES; n++)
	{
	if(payloadStats.frameSizeStats[n].frameSizeSample == 0)
	{
	_channelCritSect->Leave();
	return 0;
	}
	payloadStats.frameSizeStats[n].usageLenSec =
	(double)payloadStats.frameSizeStats[n].totalEncodedSamples
	/ (double)codecInst.plfreq;

	payloadStats.frameSizeStats[n].rateBitPerSec =
	payloadStats.frameSizeStats[n].totalPayloadLenByte * 8 /
	payloadStats.frameSizeStats[n].usageLenSec;

	}
	_channelCritSect->Leave();
	return 0;
	}

	void
	Channel::Stats(WebRtc_UWord32* numPackets)
	{
	_channelCritSect->Enter();
	int k;
	int n;
	memset(numPackets, 0, MAX_NUM_PAYLOADS * sizeof(WebRtc_UWord32));
	for(k = 0; k < MAX_NUM_PAYLOADS; k++)
	{
	if(_payloadStats[k].payloadType == -1)
	{
	break;
	}
	numPackets[k] = 0;
	for(n = 0; n < MAX_NUM_FRAMESIZES; n++)
	{
	if(_payloadStats[k].frameSizeStats[n].frameSizeSample == 0)
	{
	break;
	}
	numPackets[k] +=
	_payloadStats[k].frameSizeStats[n].numPackets;
	}
	}
	_channelCritSect->Leave();
	}

	void
	Channel::Stats(WebRtc_UWord8* payloadType, WebRtc_UWord32* payloadLenByte)
	{
	_channelCritSect->Enter();

	int k;
	int n;
	memset(payloadLenByte, 0, MAX_NUM_PAYLOADS * sizeof(WebRtc_UWord32));
	for(k = 0; k < MAX_NUM_PAYLOADS; k++)
	{
	if(_payloadStats[k].payloadType == -1)
	{
	break;
	}
	payloadType[k] = (WebRtc_UWord8)_payloadStats[k].payloadType;
	payloadLenByte[k] = 0;
	for(n = 0; n < MAX_NUM_FRAMESIZES; n++)
	{
	if(_payloadStats[k].frameSizeStats[n].frameSizeSample == 0)
	{
	break;
	}
	payloadLenByte[k] += (WebRtc_UWord16)
	_payloadStats[k].frameSizeStats[n].totalPayloadLenByte;
	}
	}

	_channelCritSect->Leave();
	}


	void
	Channel::PrintStats(CodecInst& codecInst)
	{
	ACMTestPayloadStats payloadStats;
	Stats(codecInst, payloadStats);
	printf("%s %d kHz\n",
	codecInst.plname,
	codecInst.plfreq / 1000);
	printf("=====================================================\n");
	if(payloadStats.payloadType == -1)
	{
	printf("No Packets are sent with payload-type %d (%s)\n\n",
	codecInst.pltype,
	codecInst.plname);
	return;
	}
	for(int k = 0; k < MAX_NUM_FRAMESIZES; k++)
	{
	if(payloadStats.frameSizeStats[k].frameSizeSample == 0)
	{
	break;
	}
	printf("Frame-size.................... %d samples\n",
	payloadStats.frameSizeStats[k].frameSizeSample);
	printf("Average Rate.................. %.0f bits/sec\n",
	payloadStats.frameSizeStats[k].rateBitPerSec);
	printf("Maximum Payload-Size.......... %d Bytes\n",
	payloadStats.frameSizeStats[k].maxPayloadLen);
	printf("Maximum Instantaneous Rate.... %.0f bits/sec\n",
	((double)payloadStats.frameSizeStats[k].maxPayloadLen * 8.0 *
	(double)codecInst.plfreq) /
	(double)payloadStats.frameSizeStats[k].frameSizeSample);
	printf("Number of Packets............. %u\n",
	(unsigned int)payloadStats.frameSizeStats[k].numPackets);
	printf("Duration...................... %0.3f sec\n\n",
	payloadStats.frameSizeStats[k].usageLenSec);

	}

	}

	WebRtc_UWord32
	Channel::LastInTimestamp()
	{
	WebRtc_UWord32 timestamp;
	_channelCritSect->Enter();
	timestamp = _lastInTimestamp;
	_channelCritSect->Leave();
	return timestamp;
	}

	double
	Channel::BitRate()
	{
	double rate;
	WebRtc_UWord64 currTime = TickTime::MillisecondTimestamp();
	_channelCritSect->Enter();
	rate = ((double)_totalBytes * 8.0)/ (double)(currTime - _beginTime);
	_channelCritSect->Leave();
	return rate;
	}

	} // namespace webrtc