/* * 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 #include #include "audio_coding_module.h" #include "Channel.h" #include "tick_util.h" #include "typedefs.h" #include "common_types.h" using 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), _useFECTestWithPacketLoss(false), _packetLoss(0), _lastInTimestamp(0), _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; }