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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: rebase again Created 4 years, 6 months ago
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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;
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