commit bf47495979b8e938fd4f05ca13e6a3a6ab639387
author Niels Möller <nisse@webrtc.org> Mon Feb 18 12:00:06 2019 +0100
committer Commit Bot <commit-bot@chromium.org> Mon Feb 18 13:29:35 2019 +0000
tree 9dc8ad343e4253a688988f94ff4bb50301c62b8c
parent a0b1fb9ac7698948736aa53c6d48972cb40779b5

Update remaining audio test code to not use WebRtcRTPHeader.

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}

modules/audio_coding/test/RTPFile.cc

diff --git a/modules/audio_coding/test/RTPFile.cc b/modules/audio_coding/test/RTPFile.cc
index cfe93fa..1273fa8 100644
--- a/modules/audio_coding/test/RTPFile.cc
+++ b/modules/audio_coding/test/RTPFile.cc
@@ -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 @@
   _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 @@
   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 @@
   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 @@
   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) {