| /* |
| * 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 "modules/audio_coding/test/Channel.h" |
| |
| #include <assert.h> |
| #include <iostream> |
| |
| #include "rtc_base/format_macros.h" |
| #include "rtc_base/time_utils.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 == 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; |
| } |
| |
| // 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; |
| } else { |
| memcpy(&rtpInfo, &_rtpInfo, sizeof(WebRtcRTPHeader)); |
| _leftChannel = true; |
| } |
| } |
| } |
| |
| _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(); |
| } |
| |
| 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 |
