webrtc/audio_send_stream.h - Issue 1479023002: Prepare the AudioSendStream to be hooked up to send-side BWE.

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Unified Diff: webrtc/audio_send_stream.h

Issue 1479023002: Prepare the AudioSendStream to be hooked up to send-side BWE. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: Comments addressed. Created 5 years, 1 month 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 c5db82b91bbf156733967831181a8b5f41ecd28b..2af0b34f9fb98425a80da665d491535a1bc0fa8e 100644

--- a/webrtc/audio_send_stream.h

+++ b/webrtc/audio_send_stream.h

@@ -59,7 +59,7 @@ class AudioSendStream : public SendStream {

// Sender SSRC.

uint32_t ssrc = 0;

- // RTP header extensions used for the received stream.

+ // RTP header extensions used for the sent stream.

std::vector<RtpExtension> extensions;

// RTCP CNAME, see RFC 3550.

« webrtc/audio/audio_send_stream_unittest.cc ('K') | « webrtc/audio/audio_send_stream_unittest.cc ('k') | webrtc/call/call.cc » ('j') | webrtc/voice_engine/channel.cc » ('J')