zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Fix for RTP extension audio level.

Browse Source 
Review URL: http://webrtc-codereview.appspot.com/339002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1334 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pwestin@webrtc.org
2012-01-04 17:04:51 +00:00
parent d77a6614fa
commit 6c1d41583a
8 changed files with 53 additions and 44 deletions
Download Patch File Download Diff File Expand all files Collapse all files

7
src/modules/rtp_rtcp/source/rtp_sender.cc
View File

@ -1314,10 +1314,10 @@ RTPSender::BuildRTPHeaderExtension(WebRtc_UWord8* dataBuffer) const
WebRtc_UWord16 total_block_length = 0;
RTPExtensionType type = _rtpHeaderExtensionMap.First();
while (type != NONE)
while (type != kRtpExtensionNone)
{
WebRtc_UWord8 block_length = 0;
if (type == TRANSMISSION_TIME_OFFSET)
if (type == kRtpExtensionTransmissionTimeOffset)
{
block_length = BuildTransmissionTimeOffsetExtension(
dataBuffer + kHeaderLength + total_block_length);
@ -1363,7 +1363,8 @@ RTPSender::BuildTransmissionTimeOffsetExtension(WebRtc_UWord8* dataBuffer) const
// Get id defined by user.
WebRtc_UWord8 id;
if (_rtpHeaderExtensionMap.GetId(TRANSMISSION_TIME_OFFSET, &id) != 0) {
if (_rtpHeaderExtensionMap.GetId(kRtpExtensionTransmissionTimeOffset, &id)
!= 0) {
// Not registered.
return 0;
}