blob: 7d5e6e291f2739a7f43579b2873b225952ff33af [file] [log] [blame]
/*
* 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/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