Index: webrtc/audio_receive_stream.h |
diff --git a/webrtc/audio_receive_stream.h b/webrtc/audio_receive_stream.h |
index 9a8601de9bd023749bba80d238dc5731c9613592..29735683b3b769e0c18a6db775743fb3a007a350 100644 |
--- a/webrtc/audio_receive_stream.h |
+++ b/webrtc/audio_receive_stream.h |
@@ -17,6 +17,7 @@ |
#include "webrtc/config.h" |
#include "webrtc/stream.h" |
+#include "webrtc/transport.h" |
#include "webrtc/typedefs.h" |
namespace webrtc { |
@@ -44,6 +45,9 @@ class AudioReceiveStream : public ReceiveStream { |
std::vector<RtpExtension> extensions; |
} rtp; |
+ Transport* receive_transport = nullptr; |
+ Transport* rtcp_send_transport = nullptr; |
+ |
// Underlying VoiceEngine handle, used to map AudioReceiveStream to |
// lower-level components. Temporarily used while VoiceEngine channels are |
// created outside of Call. |