Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(1869)

Unified Diff: webrtc/audio_send_stream.h

Issue 1955363003: Configure VoE NACK through AudioSendStream::Config. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: misc Created 4 years, 6 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
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;

Powered by Google App Engine
This is Rietveld 408576698