| Index: webrtc/audio_send_stream.h
|
| diff --git a/webrtc/audio_send_stream.h b/webrtc/audio_send_stream.h
|
| index 806b3d4a3cc3258644119f693f144db021a4836d..ffb5f9c4d56595905c2c1c9e45f671694633dfb4 100644
|
| --- a/webrtc/audio_send_stream.h
|
| +++ b/webrtc/audio_send_stream.h
|
| @@ -56,7 +56,7 @@ class AudioSendStream {
|
|
|
| std::string ToString() const;
|
|
|
| - // Receive-stream specific RTP settings.
|
| + // Send-stream specific RTP settings.
|
| struct Rtp {
|
| std::string ToString() const;
|
|
|
| @@ -66,6 +66,9 @@ class AudioSendStream {
|
| // RTP header extensions used for the sent stream.
|
| std::vector<RtpExtension> extensions;
|
|
|
| + // See NackConfig for description.
|
| + NackConfig nack;
|
| +
|
| // RTCP CNAME, see RFC 3550.
|
| std::string c_name;
|
| } rtp;
|
|
|