| 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; | 
|  |