05db352f56
The test code could read outside the allocated memory. The bug could up until now not be triggered by the production code, but coming changes would uncover it. COAUTHOR=kwiberg@webrtc.org R=minyue@webrtc.org Review URL: https://webrtc-codereview.appspot.com/34929004 Cr-Commit-Position: refs/heads/master@{#8216} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8216 4adac7df-926f-26a2-2b94-8c16560cd09d
411 lines
14 KiB
C++
411 lines
14 KiB
C++
/*
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* Copyright (c) 2012 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 "webrtc/modules/audio_coding/main/test/Channel.h"
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#include <assert.h>
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#include <iostream>
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#include "webrtc/base/format_macros.h"
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#include "webrtc/system_wrappers/interface/tick_util.h"
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#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
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namespace webrtc {
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int32_t Channel::SendData(FrameType frameType,
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uint8_t payloadType,
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uint32_t timeStamp,
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const uint8_t* payloadData,
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size_t payloadSize,
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const RTPFragmentationHeader* fragmentation) {
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WebRtcRTPHeader rtpInfo;
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int32_t status;
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size_t 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 = (external_sequence_number_ < 0) ?
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_seqNo++ : static_cast<uint16_t>(external_sequence_number_);
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rtpInfo.header.payloadType = payloadType;
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rtpInfo.header.timestamp = (external_send_timestamp_ < 0) ? timeStamp :
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static_cast<uint32_t>(external_send_timestamp_);
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if (frameType == kAudioFrameCN) {
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rtpInfo.type.Audio.isCNG = true;
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} else {
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rtpInfo.type.Audio.isCNG = false;
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}
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if (frameType == kFrameEmpty) {
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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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// If silence for too long, send only new data.
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if ((fragmentation->fragmentationVectorSize == 2) &&
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(fragmentation->fragmentationTimeDiff[1] <= 0x3fff)) {
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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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size_t REDheader = (fragmentation->fragmentationTimeDiff[1] << 10) +
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fragmentation->fragmentationLength[1];
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_payloadData[1] = uint8_t((REDheader >> 16) & 0x000000FF);
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_payloadData[2] = uint8_t((REDheader >> 8) & 0x000000FF);
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_payloadData[3] = uint8_t(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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// single block (newest one)
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memcpy(_payloadData, payloadData + fragmentation->fragmentationOffset[0],
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fragmentation->fragmentationLength[0]);
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payloadDataSize = fragmentation->fragmentationLength[0];
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rtpInfo.header.payloadType = fragmentation->fragmentationPlType[0];
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}
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} else {
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memcpy(_payloadData, payloadData, payloadDataSize);
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if (_isStereo) {
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if (_leftChannel) {
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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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} else {
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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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//fwrite(payloadData, sizeof(uint8_t), payloadSize, _bitStreamFile);
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}
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if (!_isStereo) {
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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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_packetLoss += 1;
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if (_packetLoss == 3) {
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_packetLoss = 0;
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return 0;
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}
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}
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if (num_packets_to_drop_ > 0) {
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num_packets_to_drop_--;
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return 0;
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}
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status = _receiverACM->IncomingPacket(_payloadData, payloadDataSize, rtpInfo);
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return status;
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}
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// TODO(turajs): rewite this method.
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void Channel::CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t payloadSize) {
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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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// 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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if (_lastPayloadType == _payloadStats[n].payloadType) {
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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 = NULL;
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for (n = 0; n < MAX_NUM_PAYLOADS; n++) {
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if (rtpInfo.header.payloadType == _payloadStats[n].payloadType) {
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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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if (!currentPayloadStr->newPacket) {
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uint32_t lastFrameSizeSample = (uint32_t)(
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(uint32_t) rtpInfo.header.timestamp
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- (uint32_t) 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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k++;
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}
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ACMTestFrameSizeStats* currentFrameSizeStats = &(currentPayloadStr
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->frameSizeStats[k]);
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currentFrameSizeStats->frameSizeSample = (int16_t) lastFrameSizeSample;
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// increment the number of encoded samples.
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currentFrameSizeStats->totalEncodedSamples += 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 += currentPayloadStr
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->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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currentFrameSizeStats->maxPayloadLen = currentPayloadStr
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->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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} else {
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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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memset(currentPayloadStr->frameSizeStats, 0, MAX_NUM_FRAMESIZES *
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sizeof(ACMTestFrameSizeStats));
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}
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} else {
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n = 0;
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while (_payloadStats[n].payloadType != -1) {
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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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memset(_payloadStats[n].frameSizeStats, 0, MAX_NUM_FRAMESIZES *
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sizeof(ACMTestFrameSizeStats));
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}
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}
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Channel::Channel(int16_t 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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_lastInTimestamp(0),
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_packetLoss(0),
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_useFECTestWithPacketLoss(false),
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_beginTime(TickTime::MillisecondTimestamp()),
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_totalBytes(0),
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external_send_timestamp_(-1),
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external_sequence_number_(-1),
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num_packets_to_drop_(0) {
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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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_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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_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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_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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} else {
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_saveBitStream = false;
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}
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}
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Channel::~Channel() {
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delete _channelCritSect;
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}
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void Channel::RegisterReceiverACM(AudioCodingModule* acm) {
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_receiverACM = acm;
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return;
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}
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void Channel::ResetStats() {
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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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_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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_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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int16_t Channel::Stats(CodecInst& codecInst,
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ACMTestPayloadStats& payloadStats) {
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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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if (_payloadStats[n].payloadType == codecInst.pltype) {
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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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_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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if (payloadStats.frameSizeStats[n].frameSizeSample == 0) {
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_channelCritSect->Leave();
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return 0;
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}
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payloadStats.frameSizeStats[n].usageLenSec = (double) payloadStats
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.frameSizeStats[n].totalEncodedSamples / (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 Channel::Stats(uint32_t* numPackets) {
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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(uint32_t));
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for (k = 0; k < MAX_NUM_PAYLOADS; k++) {
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if (_payloadStats[k].payloadType == -1) {
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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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if (_payloadStats[k].frameSizeStats[n].frameSizeSample == 0) {
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break;
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}
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numPackets[k] += _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 Channel::Stats(uint8_t* payloadType, uint32_t* payloadLenByte) {
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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(uint32_t));
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for (k = 0; k < MAX_NUM_PAYLOADS; k++) {
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if (_payloadStats[k].payloadType == -1) {
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break;
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}
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payloadType[k] = (uint8_t) _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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if (_payloadStats[k].frameSizeStats[n].frameSizeSample == 0) {
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break;
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}
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payloadLenByte[k] += (uint16_t) _payloadStats[k].frameSizeStats[n]
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.totalPayloadLenByte;
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}
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}
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_channelCritSect->Leave();
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}
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void Channel::PrintStats(CodecInst& codecInst) {
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ACMTestPayloadStats payloadStats;
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Stats(codecInst, payloadStats);
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printf("%s %d kHz\n", codecInst.plname, codecInst.plfreq / 1000);
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printf("=====================================================\n");
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if (payloadStats.payloadType == -1) {
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printf("No Packets are sent with payload-type %d (%s)\n\n",
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codecInst.pltype, 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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if (payloadStats.frameSizeStats[k].frameSizeSample == 0) {
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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.......... %" PRIuS " Bytes\n",
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payloadStats.frameSizeStats[k].maxPayloadLen);
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printf(
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"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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uint32_t Channel::LastInTimestamp() {
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uint32_t 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 Channel::BitRate() {
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double rate;
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uint64_t 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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} // namespace webrtc
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