| Index: webrtc/audio_send_stream.h
|
| diff --git a/webrtc/audio_send_stream.h b/webrtc/audio_send_stream.h
|
| index d8e98bb0ec9cff818c36d965e090076364eef333..045b8fd717488ddad8c017524180cb84b10eaf41 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;
|
|
|