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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
Download Patch File Download Diff File Expand all files Collapse all files

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) {