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; |