webrtc/audio_send_stream.h - Issue 1418503010: Move some send stream configuration into webrtc::AudioSendStream.

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Unified Diff: webrtc/audio_send_stream.h

Issue 1418503010: Move some send stream configuration into webrtc::AudioSendStream. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: rebase 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 89b73e6e3ed2a95f9ef79b6f31f0bd3e8b725b0e..c5db82b91bbf156733967831181a8b5f41ecd28b 100644

--- a/webrtc/audio_send_stream.h

+++ b/webrtc/audio_send_stream.h

@@ -61,6 +61,9 @@ class AudioSendStream : public SendStream {

// RTP header extensions used for the received stream.

std::vector<RtpExtension> extensions;

+ // RTCP CNAME, see RFC 3550.

+ std::string c_name;

} rtp;

// Transport for outgoing packets. The transport is expected to exist for

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