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

 /*
  *  Copyright (c) 2012 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 "webrtc/modules/audio_coding/test/Channel.h"

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

 #include "webrtc/rtc_base/format_macros.h"
 #include "webrtc/rtc_base/timeutils.h"

 namespace webrtc {

 int32_t Channel::SendData(FrameType frameType,
                           uint8_t payloadType,
                           uint32_t timeStamp,
                           const uint8_t* payloadData,
                           size_t payloadSize,
                           const RTPFragmentationHeader* fragmentation) {
   WebRtcRTPHeader rtpInfo;
   int32_t status;
   size_t payloadDataSize = payloadSize;

   rtpInfo.header.markerBit = false;
   rtpInfo.header.ssrc = 0;
   rtpInfo.header.sequenceNumber = (external_sequence_number_ < 0) ?
       _seqNo++ : static_cast<uint16_t>(external_sequence_number_);
   rtpInfo.header.payloadType = payloadType;
   rtpInfo.header.timestamp = (external_send_timestamp_ < 0) ? timeStamp :
       static_cast<uint32_t>(external_send_timestamp_);

   if (frameType == kAudioFrameCN) {
     rtpInfo.type.Audio.isCNG = true;
   } else {
     rtpInfo.type.Audio.isCNG = false;
   }
   if (frameType == kEmptyFrame) {
     // When frame is empty, we should not transmit it. The frame size of the
     // next non-empty frame will be based on the previous frame size.
     _useLastFrameSize = _lastFrameSizeSample > 0;
     return 0;
   }

   rtpInfo.type.Audio.channel = 1;
   // Treat fragmentation separately
   if (fragmentation != NULL) {
     // If silence for too long, send only new data.
     if ((fragmentation->fragmentationVectorSize == 2) &&
         (fragmentation->fragmentationTimeDiff[1] <= 0x3fff)) {
       // only 0x80 if we have multiple blocks
       _payloadData[0] = 0x80 + fragmentation->fragmentationPlType[1];
       size_t REDheader = (fragmentation->fragmentationTimeDiff[1] << 10) +
           fragmentation->fragmentationLength[1];
       _payloadData[1] = uint8_t((REDheader >> 16) & 0x000000FF);
       _payloadData[2] = uint8_t((REDheader >> 8) & 0x000000FF);
       _payloadData[3] = uint8_t(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 = 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(uint8_t), payloadSize, _bitStreamFile);
   }

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

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

   if (num_packets_to_drop_ > 0) {
     num_packets_to_drop_--;
     return 0;
   }

   status = _receiverACM->IncomingPacket(_payloadData, payloadDataSize, rtpInfo);

   return status;
 }

 // TODO(turajs): rewite this method.
 void Channel::CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t 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 = NULL;
   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) {
       if (!_useLastFrameSize) {
         _lastFrameSizeSample = (uint32_t) ((uint32_t) rtpInfo.header.timestamp -
             (uint32_t) currentPayloadStr->lastTimestamp);
       }
       assert(_lastFrameSizeSample > 0);
       int k = 0;
       for (; k < MAX_NUM_FRAMESIZES; ++k) {
         if ((currentPayloadStr->frameSizeStats[k].frameSizeSample ==
             _lastFrameSizeSample) ||
             (currentPayloadStr->frameSizeStats[k].frameSizeSample == 0)) {
           break;
         }
       }
       if (k == MAX_NUM_FRAMESIZES) {
         // New frame size found but no space to count statistics on it. Skip it.
         printf("No memory to store statistics for payload %d : frame size %d\n",
                _lastPayloadType, _lastFrameSizeSample);
         return;
       }
       ACMTestFrameSizeStats* currentFrameSizeStats = &(currentPayloadStr
           ->frameSizeStats[k]);
       currentFrameSizeStats->frameSizeSample = (int16_t) _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;
       memset(currentPayloadStr->frameSizeStats, 0, MAX_NUM_FRAMESIZES *
              sizeof(ACMTestFrameSizeStats));
     }
   } 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;
     memset(_payloadStats[n].frameSizeStats, 0, MAX_NUM_FRAMESIZES *
            sizeof(ACMTestFrameSizeStats));
   }
 }

 Channel::Channel(int16_t chID)
     : _receiverACM(NULL),
       _seqNo(0),
       _bitStreamFile(NULL),
       _saveBitStream(false),
       _lastPayloadType(-1),
       _isStereo(false),
       _leftChannel(true),
       _lastInTimestamp(0),
       _useLastFrameSize(false),
       _lastFrameSizeSample(0),
       _packetLoss(0),
       _useFECTestWithPacketLoss(false),
       _beginTime(rtc::TimeMillis()),
       _totalBytes(0),
       external_send_timestamp_(-1),
       external_sequence_number_(-1),
       num_packets_to_drop_(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() {
 }

 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 = rtc::TimeMillis();
   _totalBytes = 0;
   _channelCritSect.Leave();
 }

 int16_t 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(uint32_t* numPackets) {
   _channelCritSect.Enter();
   int k;
   int n;
   memset(numPackets, 0, MAX_NUM_PAYLOADS * sizeof(uint32_t));
   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(uint8_t* payloadType, uint32_t* payloadLenByte) {
   _channelCritSect.Enter();

   int k;
   int n;
   memset(payloadLenByte, 0, MAX_NUM_PAYLOADS * sizeof(uint32_t));
   for (k = 0; k < MAX_NUM_PAYLOADS; k++) {
     if (_payloadStats[k].payloadType == -1) {
       break;
     }
     payloadType[k] = (uint8_t) _payloadStats[k].payloadType;
     payloadLenByte[k] = 0;
     for (n = 0; n < MAX_NUM_FRAMESIZES; n++) {
       if (_payloadStats[k].frameSizeStats[n].frameSizeSample == 0) {
         break;
       }
       payloadLenByte[k] += (uint16_t) _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.......... %" PRIuS " 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);

   }

 }

 uint32_t Channel::LastInTimestamp() {
   uint32_t timestamp;
   _channelCritSect.Enter();
   timestamp = _lastInTimestamp;
   _channelCritSect.Leave();
   return timestamp;
 }

 double Channel::BitRate() {
   double rate;
   uint64_t currTime = rtc::TimeMillis();
   _channelCritSect.Enter();
   rate = ((double) _totalBytes * 8.0) / (double) (currTime - _beginTime);
   _channelCritSect.Leave();
   return rate;
 }

 }  // namespace webrtc
	/*
	* Copyright (c) 2012 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 "webrtc/modules/audio_coding/test/Channel.h"

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

	#include "webrtc/rtc_base/format_macros.h"
	#include "webrtc/rtc_base/timeutils.h"

	namespace webrtc {

	int32_t Channel::SendData(FrameType frameType,
	uint8_t payloadType,
	uint32_t timeStamp,
	const uint8_t* payloadData,
	size_t payloadSize,
	const RTPFragmentationHeader* fragmentation) {
	WebRtcRTPHeader rtpInfo;
	int32_t status;
	size_t payloadDataSize = payloadSize;

	rtpInfo.header.markerBit = false;
	rtpInfo.header.ssrc = 0;
	rtpInfo.header.sequenceNumber = (external_sequence_number_ < 0) ?
	_seqNo++ : static_cast<uint16_t>(external_sequence_number_);
	rtpInfo.header.payloadType = payloadType;
	rtpInfo.header.timestamp = (external_send_timestamp_ < 0) ? timeStamp :
	static_cast<uint32_t>(external_send_timestamp_);

	if (frameType == kAudioFrameCN) {
	rtpInfo.type.Audio.isCNG = true;
	} else {
	rtpInfo.type.Audio.isCNG = false;
	}
	if (frameType == kEmptyFrame) {
	// When frame is empty, we should not transmit it. The frame size of the
	// next non-empty frame will be based on the previous frame size.
	_useLastFrameSize = _lastFrameSizeSample > 0;
	return 0;
	}

	rtpInfo.type.Audio.channel = 1;
	// Treat fragmentation separately
	if (fragmentation != NULL) {
	// If silence for too long, send only new data.
	if ((fragmentation->fragmentationVectorSize == 2) &&
	(fragmentation->fragmentationTimeDiff[1] <= 0x3fff)) {
	// only 0x80 if we have multiple blocks
	_payloadData[0] = 0x80 + fragmentation->fragmentationPlType[1];
	size_t REDheader = (fragmentation->fragmentationTimeDiff[1] << 10) +
	fragmentation->fragmentationLength[1];
	_payloadData[1] = uint8_t((REDheader >> 16) & 0x000000FF);
	_payloadData[2] = uint8_t((REDheader >> 8) & 0x000000FF);
	_payloadData[3] = uint8_t(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 = 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(uint8_t), payloadSize, _bitStreamFile);
	}

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

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

	if (num_packets_to_drop_ > 0) {
	num_packets_to_drop_--;
	return 0;
	}

	status = _receiverACM->IncomingPacket(_payloadData, payloadDataSize, rtpInfo);

	return status;
	}

	// TODO(turajs): rewite this method.
	void Channel::CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t 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 = NULL;
	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) {
	if (!_useLastFrameSize) {
	_lastFrameSizeSample = (uint32_t) ((uint32_t) rtpInfo.header.timestamp -
	(uint32_t) currentPayloadStr->lastTimestamp);
	}
	assert(_lastFrameSizeSample > 0);
	int k = 0;
	for (; k < MAX_NUM_FRAMESIZES; ++k) {
	if ((currentPayloadStr->frameSizeStats[k].frameSizeSample ==
	_lastFrameSizeSample) \|\|
	(currentPayloadStr->frameSizeStats[k].frameSizeSample == 0)) {
	break;
	}
	}
	if (k == MAX_NUM_FRAMESIZES) {
	// New frame size found but no space to count statistics on it. Skip it.
	printf("No memory to store statistics for payload %d : frame size %d\n",
	_lastPayloadType, _lastFrameSizeSample);
	return;
	}
	ACMTestFrameSizeStats* currentFrameSizeStats = &(currentPayloadStr
	->frameSizeStats[k]);
	currentFrameSizeStats->frameSizeSample = (int16_t) _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;
	memset(currentPayloadStr->frameSizeStats, 0, MAX_NUM_FRAMESIZES *
	sizeof(ACMTestFrameSizeStats));
	}
	} 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;
	memset(_payloadStats[n].frameSizeStats, 0, MAX_NUM_FRAMESIZES *
	sizeof(ACMTestFrameSizeStats));
	}
	}

	Channel::Channel(int16_t chID)
	: _receiverACM(NULL),
	_seqNo(0),
	_bitStreamFile(NULL),
	_saveBitStream(false),
	_lastPayloadType(-1),
	_isStereo(false),
	_leftChannel(true),
	_lastInTimestamp(0),
	_useLastFrameSize(false),
	_lastFrameSizeSample(0),
	_packetLoss(0),
	_useFECTestWithPacketLoss(false),
	_beginTime(rtc::TimeMillis()),
	_totalBytes(0),
	external_send_timestamp_(-1),
	external_sequence_number_(-1),
	num_packets_to_drop_(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() {
	}

	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 = rtc::TimeMillis();
	_totalBytes = 0;
	_channelCritSect.Leave();
	}

	int16_t 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(uint32_t* numPackets) {
	_channelCritSect.Enter();
	int k;
	int n;
	memset(numPackets, 0, MAX_NUM_PAYLOADS * sizeof(uint32_t));
	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(uint8_t* payloadType, uint32_t* payloadLenByte) {
	_channelCritSect.Enter();

	int k;
	int n;
	memset(payloadLenByte, 0, MAX_NUM_PAYLOADS * sizeof(uint32_t));
	for (k = 0; k < MAX_NUM_PAYLOADS; k++) {
	if (_payloadStats[k].payloadType == -1) {
	break;
	}
	payloadType[k] = (uint8_t) _payloadStats[k].payloadType;
	payloadLenByte[k] = 0;
	for (n = 0; n < MAX_NUM_FRAMESIZES; n++) {
	if (_payloadStats[k].frameSizeStats[n].frameSizeSample == 0) {
	break;
	}
	payloadLenByte[k] += (uint16_t) _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.......... %" PRIuS " 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);

	}

	}

	uint32_t Channel::LastInTimestamp() {
	uint32_t timestamp;
	_channelCritSect.Enter();
	timestamp = _lastInTimestamp;
	_channelCritSect.Leave();
	return timestamp;
	}

	double Channel::BitRate() {
	double rate;
	uint64_t currTime = rtc::TimeMillis();
	_channelCritSect.Enter();
	rate = ((double) _totalBytes * 8.0) / (double) (currTime - _beginTime);
	_channelCritSect.Leave();
	return rate;
	}

	} // namespace webrtc