Add send transports to individual webrtc::Call streams.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1273363005
Cr-Original-Commit-Position: refs/heads/master@{#9807}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: 4fbae2b79134572135d9d5fe35a7d1ccdeea3a4d
diff --git a/video_receive_stream.h b/video_receive_stream.h
index 9525336..e613006 100644
--- a/video_receive_stream.h
+++ b/video_receive_stream.h
@@ -86,6 +86,10 @@
};
struct Config {
+ Config() = delete;
+ explicit Config(newapi::Transport* rtcp_send_transport)
+ : rtcp_send_transport(rtcp_send_transport) {}
+
std::string ToString() const;
// Decoders for every payload that we can receive.
@@ -137,6 +141,9 @@
std::vector<RtpExtension> extensions;
} rtp;
+ // Transport for outgoing packets (RTCP).
+ newapi::Transport* rtcp_send_transport = nullptr;
+
// VideoRenderer will be called for each decoded frame. 'nullptr' disables
// rendering of this stream.
VideoRenderer* renderer = nullptr;