| 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.
|
|
|