| Index: webrtc/config.cc
|
| diff --git a/webrtc/config.cc b/webrtc/config.cc
|
| index c5d29d480397eda61fe24e6968e7b1ed9a8ac1b7..ddff931e241fc1ea22f01b09dd149c754eb9686c 100644
|
| --- a/webrtc/config.cc
|
| +++ b/webrtc/config.cc
|
| @@ -35,16 +35,20 @@ const char* RtpExtension::kAbsSendTime =
|
| const char* RtpExtension::kVideoRotation = "urn:3gpp:video-orientation";
|
| const char* RtpExtension::kAudioLevel =
|
| "urn:ietf:params:rtp-hdrext:ssrc-audio-level";
|
| +const char* RtpExtension::kTransportSequenceNumber =
|
| + "http://www.webrtc.org/experiments/rtp-hdrext/transport-sequence-number";
|
|
|
| bool RtpExtension::IsSupportedForAudio(const std::string& name) {
|
| return name == webrtc::RtpExtension::kAbsSendTime ||
|
| - name == webrtc::RtpExtension::kAudioLevel;
|
| + name == webrtc::RtpExtension::kAudioLevel ||
|
| + name == webrtc::RtpExtension::kTransportSequenceNumber;
|
| }
|
|
|
| bool RtpExtension::IsSupportedForVideo(const std::string& name) {
|
| return name == webrtc::RtpExtension::kTOffset ||
|
| name == webrtc::RtpExtension::kAbsSendTime ||
|
| - name == webrtc::RtpExtension::kVideoRotation;
|
| + name == webrtc::RtpExtension::kVideoRotation ||
|
| + name == webrtc::RtpExtension::kTransportSequenceNumber;
|
| }
|
|
|
| VideoStream::VideoStream()
|
|
|