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Update remaining audio test code to not use WebRtcRTPHeader.

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Bug: webrtc:5876
Change-Id: I5b1abcec4a0ef52b6dd36d1fe94dbfd3f88f28a7
Reviewed-on: https://webrtc-review.googlesource.com/c/123235
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26736}
This commit is contained in:
Niels Möller 2019-02-18 12:00:06 +01:00 committed by Commit Bot
parent a0b1fb9ac7
commit bf47495979
7 changed files with 56 additions and 58 deletions
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49
modules/audio_coding/test/Channel.cc
View File

@ -24,21 +24,20 @@ int32_t Channel::SendData(FrameType frameType,
const uint8_t* payloadData,
size_t payloadSize,
const RTPFragmentationHeader* fragmentation) {
WebRtcRTPHeader rtpInfo;
RTPHeader rtp_header;
int32_t status;
size_t payloadDataSize = payloadSize;
rtpInfo.header.markerBit = false;
rtpInfo.header.ssrc = 0;
rtpInfo.header.sequenceNumber =
rtp_header.markerBit = false;
rtp_header.ssrc = 0;
rtp_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_);
rtp_header.payloadType = payloadType;
rtp_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
@ -75,16 +74,16 @@ int32_t Channel::SendData(FrameType frameType,
memcpy(_payloadData, payloadData + fragmentation->fragmentationOffset[0],
fragmentation->fragmentationLength[0]);
payloadDataSize = fragmentation->fragmentationLength[0];
rtpInfo.header.payloadType = fragmentation->fragmentationPlType[0];
rtp_header.payloadType = fragmentation->fragmentationPlType[0];
}
} else {
memcpy(_payloadData, payloadData, payloadDataSize);
if (_isStereo) {
if (_leftChannel) {
memcpy(&_rtpInfo, &rtpInfo, sizeof(WebRtcRTPHeader));
_rtp_header = rtp_header;
_leftChannel = false;
} else {
memcpy(&rtpInfo, &_rtpInfo, sizeof(WebRtcRTPHeader));
rtp_header = _rtp_header;
_leftChannel = true;
}
}
@ -96,7 +95,7 @@ int32_t Channel::SendData(FrameType frameType,
}
if (!_isStereo) {
CalcStatistics(rtpInfo, payloadSize);
CalcStatistics(rtp_header, payloadSize);
}
_useLastFrameSize = false;
_lastInTimestamp = timeStamp;
@ -116,16 +115,16 @@ int32_t Channel::SendData(FrameType frameType,
return 0;
}
status = _receiverACM->IncomingPacket(_payloadData, payloadDataSize,
rtpInfo.header);
status =
_receiverACM->IncomingPacket(_payloadData, payloadDataSize, rtp_header);
return status;
}
// TODO(turajs): rewite this method.
void Channel::CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t payloadSize) {
void Channel::CalcStatistics(const RTPHeader& rtp_header, size_t payloadSize) {
int n;
if ((rtpInfo.header.payloadType != _lastPayloadType) &&
if ((rtp_header.payloadType != _lastPayloadType) &&
(_lastPayloadType != -1)) {
// payload-type is changed.
// we have to terminate the calculations on the previous payload type
@ -138,12 +137,12 @@ void Channel::CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t payloadSize) {
}
}
}
_lastPayloadType = rtpInfo.header.payloadType;
_lastPayloadType = rtp_header.payloadType;
bool newPayload = true;
ACMTestPayloadStats* currentPayloadStr = NULL;
for (n = 0; n < MAX_NUM_PAYLOADS; n++) {
if (rtpInfo.header.payloadType == _payloadStats[n].payloadType) {
if (rtp_header.payloadType == _payloadStats[n].payloadType) {
newPayload = false;
currentPayloadStr = &_payloadStats[n];
break;
@ -154,7 +153,7 @@ void Channel::CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t payloadSize) {
if (!currentPayloadStr->newPacket) {
if (!_useLastFrameSize) {
_lastFrameSizeSample =
(uint32_t)((uint32_t)rtpInfo.header.timestamp -
(uint32_t)((uint32_t)rtp_header.timestamp -
(uint32_t)currentPayloadStr->lastTimestamp);
}
assert(_lastFrameSizeSample > 0);
@ -194,13 +193,13 @@ void Channel::CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t payloadSize) {
currentPayloadStr->lastPayloadLenByte;
}
// store the current values for the next time
currentPayloadStr->lastTimestamp = rtpInfo.header.timestamp;
currentPayloadStr->lastTimestamp = rtp_header.timestamp;
currentPayloadStr->lastPayloadLenByte = payloadSize;
} else {
currentPayloadStr->newPacket = false;
currentPayloadStr->lastPayloadLenByte = payloadSize;
currentPayloadStr->lastTimestamp = rtpInfo.header.timestamp;
currentPayloadStr->payloadType = rtpInfo.header.payloadType;
currentPayloadStr->lastTimestamp = rtp_header.timestamp;
currentPayloadStr->payloadType = rtp_header.payloadType;
memset(currentPayloadStr->frameSizeStats, 0,
MAX_NUM_FRAMESIZES * sizeof(ACMTestFrameSizeStats));
}
@ -212,8 +211,8 @@ void Channel::CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t payloadSize) {
// first packet
_payloadStats[n].newPacket = false;
_payloadStats[n].lastPayloadLenByte = payloadSize;
_payloadStats[n].lastTimestamp = rtpInfo.header.timestamp;
_payloadStats[n].payloadType = rtpInfo.header.payloadType;
_payloadStats[n].lastTimestamp = rtp_header.timestamp;
_payloadStats[n].payloadType = rtp_header.payloadType;
memset(_payloadStats[n].frameSizeStats, 0,
MAX_NUM_FRAMESIZES * sizeof(ACMTestFrameSizeStats));
}

4
modules/audio_coding/test/Channel.h
View File

@ -81,7 +81,7 @@ class Channel : public AudioPacketizationCallback {
}
private:
void CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t payloadSize);
void CalcStatistics(const RTPHeader& rtp_header, size_t payloadSize);
AudioCodingModule* _receiverACM;
uint16_t _seqNo;
@ -94,7 +94,7 @@ class Channel : public AudioPacketizationCallback {
int16_t _lastPayloadType;
ACMTestPayloadStats _payloadStats[MAX_NUM_PAYLOADS];
bool _isStereo;
WebRtcRTPHeader _rtpInfo;
RTPHeader _rtp_header;
bool _leftChannel;
uint32_t _lastInTimestamp;
bool _useLastFrameSize;

6
modules/audio_coding/test/EncodeDecodeTest.cc
View File

@ -155,7 +155,7 @@ bool Receiver::IncomingPacket() {
if (!_rtpStream->EndOfFile()) {
if (_firstTime) {
_firstTime = false;
_realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload,
_realPayloadSizeBytes = _rtpStream->Read(&_rtpHeader, _incomingPayload,
_payloadSizeBytes, &_nextTime);
if (_realPayloadSizeBytes == 0) {
if (_rtpStream->EndOfFile()) {
@ -168,8 +168,8 @@ bool Receiver::IncomingPacket() {
}
EXPECT_EQ(0, _acm->IncomingPacket(_incomingPayload, _realPayloadSizeBytes,
_rtpInfo.header));
_realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload,
_rtpHeader));
_realPayloadSizeBytes = _rtpStream->Read(&_rtpHeader, _incomingPayload,
_payloadSizeBytes, &_nextTime);
if (_realPayloadSizeBytes == 0 && _rtpStream->EndOfFile()) {
_firstTime = true;

2
modules/audio_coding/test/EncodeDecodeTest.h
View File

@ -84,7 +84,7 @@ class Receiver {
AudioCodingModule* _acm;
uint8_t _incomingPayload[MAX_INCOMING_PAYLOAD];
RTPStream* _rtpStream;
WebRtcRTPHeader _rtpInfo;
RTPHeader _rtpHeader;
size_t _realPayloadSizeBytes;
size_t _payloadSizeBytes;
uint32_t _nextTime;

7
modules/audio_coding/test/PacketLossTest.cc
View File

@ -44,7 +44,7 @@ void ReceiverWithPacketLoss::Setup(AudioCodingModule* acm,
bool ReceiverWithPacketLoss::IncomingPacket() {
if (!_rtpStream->EndOfFile()) {
if (packet_counter_ == 0) {
_realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload,
_realPayloadSizeBytes = _rtpStream->Read(&_rtpHeader, _incomingPayload,
_payloadSizeBytes, &_nextTime);
if (_realPayloadSizeBytes == 0) {
if (_rtpStream->EndOfFile()) {
@ -57,11 +57,10 @@ bool ReceiverWithPacketLoss::IncomingPacket() {
}
if (!PacketLost()) {
_acm->IncomingPacket(_incomingPayload, _realPayloadSizeBytes,
_rtpInfo.header);
_acm->IncomingPacket(_incomingPayload, _realPayloadSizeBytes, _rtpHeader);
}
packet_counter_++;
_realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload,
_realPayloadSizeBytes = _rtpStream->Read(&_rtpHeader, _incomingPayload,
_payloadSizeBytes, &_nextTime);
if (_realPayloadSizeBytes == 0 && _rtpStream->EndOfFile()) {
packet_counter_ = 0;

38
modules/audio_coding/test/RTPFile.cc
View File

@ -25,19 +25,19 @@
namespace webrtc {
void RTPStream::ParseRTPHeader(WebRtcRTPHeader* rtpInfo,
void RTPStream::ParseRTPHeader(RTPHeader* rtp_header,
const uint8_t* rtpHeader) {
rtpInfo->header.payloadType = rtpHeader[1];
rtpInfo->header.sequenceNumber =
rtp_header->payloadType = rtpHeader[1];
rtp_header->sequenceNumber =
(static_cast<uint16_t>(rtpHeader[2]) << 8) | rtpHeader[3];
rtpInfo->header.timestamp = (static_cast<uint32_t>(rtpHeader[4]) << 24) |
(static_cast<uint32_t>(rtpHeader[5]) << 16) |
(static_cast<uint32_t>(rtpHeader[6]) << 8) |
rtpHeader[7];
rtpInfo->header.ssrc = (static_cast<uint32_t>(rtpHeader[8]) << 24) |
(static_cast<uint32_t>(rtpHeader[9]) << 16) |
(static_cast<uint32_t>(rtpHeader[10]) << 8) |
rtpHeader[11];
rtp_header->timestamp = (static_cast<uint32_t>(rtpHeader[4]) << 24) |
(static_cast<uint32_t>(rtpHeader[5]) << 16) |
(static_cast<uint32_t>(rtpHeader[6]) << 8) |
rtpHeader[7];
rtp_header->ssrc = (static_cast<uint32_t>(rtpHeader[8]) << 24) |
(static_cast<uint32_t>(rtpHeader[9]) << 16) |
(static_cast<uint32_t>(rtpHeader[10]) << 8) |
rtpHeader[11];
}
void RTPStream::MakeRTPheader(uint8_t* rtpHeader,
@ -101,7 +101,7 @@ void RTPBuffer::Write(const uint8_t payloadType,
_queueRWLock->ReleaseLockExclusive();
}
size_t RTPBuffer::Read(WebRtcRTPHeader* rtpInfo,
size_t RTPBuffer::Read(RTPHeader* rtp_header,
uint8_t* payloadData,
size_t payloadSize,
uint32_t* offset) {
@ -109,11 +109,11 @@ size_t RTPBuffer::Read(WebRtcRTPHeader* rtpInfo,
RTPPacket* packet = _rtpQueue.front();
_rtpQueue.pop();
_queueRWLock->ReleaseLockShared();
rtpInfo->header.markerBit = 1;
rtpInfo->header.payloadType = packet->payloadType;
rtpInfo->header.sequenceNumber = packet->seqNo;
rtpInfo->header.ssrc = 0;
rtpInfo->header.timestamp = packet->timeStamp;
rtp_header->markerBit = 1;
rtp_header->payloadType = packet->payloadType;
rtp_header->sequenceNumber = packet->seqNo;
rtp_header->ssrc = 0;
rtp_header->timestamp = packet->timeStamp;
if (packet->payloadSize > 0 && payloadSize >= packet->payloadSize) {
memcpy(payloadData, packet->payloadData, packet->payloadSize);
} else {
@ -199,7 +199,7 @@ void RTPFile::Write(const uint8_t payloadType,
EXPECT_EQ(payloadSize, fwrite(payloadData, 1, payloadSize, _rtpFile));
}
size_t RTPFile::Read(WebRtcRTPHeader* rtpInfo,
size_t RTPFile::Read(RTPHeader* rtp_header,
uint8_t* payloadData,
size_t payloadSize,
uint32_t* offset) {
@ -220,7 +220,7 @@ size_t RTPFile::Read(WebRtcRTPHeader* rtpInfo,
EXPECT_GT(plen, 11);
EXPECT_EQ(1u, fread(rtpHeader, 12, 1, _rtpFile));
ParseRTPHeader(rtpInfo, rtpHeader);
ParseRTPHeader(rtp_header, rtpHeader);
EXPECT_EQ(lengthBytes, plen + 8);
if (plen == 0) {

8
modules/audio_coding/test/RTPFile.h
View File

@ -33,7 +33,7 @@ class RTPStream {
// Returns the packet's payload size. Zero should be treated as an
// end-of-stream (in the case that EndOfFile() is true) or an error.
virtual size_t Read(WebRtcRTPHeader* rtpInfo,
virtual size_t Read(RTPHeader* rtp_Header,
uint8_t* payloadData,
size_t payloadSize,
uint32_t* offset) = 0;
@ -46,7 +46,7 @@ class RTPStream {
uint32_t timeStamp,
uint32_t ssrc);
void ParseRTPHeader(WebRtcRTPHeader* rtpInfo, const uint8_t* rtpHeader);
void ParseRTPHeader(RTPHeader* rtp_header, const uint8_t* rtpHeader);
};
class RTPPacket {
@ -81,7 +81,7 @@ class RTPBuffer : public RTPStream {
const size_t payloadSize,
uint32_t frequency) override;
size_t Read(WebRtcRTPHeader* rtpInfo,
size_t Read(RTPHeader* rtp_header,
uint8_t* payloadData,
size_t payloadSize,
uint32_t* offset) override;
@ -114,7 +114,7 @@ class RTPFile : public RTPStream {
const size_t payloadSize,
uint32_t frequency) override;
size_t Read(WebRtcRTPHeader* rtpInfo,
size_t Read(RTPHeader* rtp_header,
uint8_t* payloadData,
size_t payloadSize,
uint32_t* offset) override;