Index: webrtc/audio_receive_stream.h |
diff --git a/webrtc/audio_receive_stream.h b/webrtc/audio_receive_stream.h |
index 9a8601de9bd023749bba80d238dc5731c9613592..70d6480b10c1cc3e9d140b93a90b614d1008413e 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 { |
@@ -44,9 +45,13 @@ class AudioReceiveStream : public ReceiveStream { |
std::vector<RtpExtension> extensions; |
} rtp; |
- // Underlying VoiceEngine handle, used to map AudioReceiveStream to |
- // lower-level components. Temporarily used while VoiceEngine channels are |
- // created outside of Call. |
+ Transport* receive_transport = nullptr; |
+ Transport* rtcp_send_transport = nullptr; |
+ |
+ // Underlying VoiceEngine handle, used to map AudioReceiveStream to lower- |
+ // level components. |
+ // TODO(solenberg): Remove when VoiceEngine channels are created outside |
+ // of Call. |
int voe_channel_id = -1; |
// Identifier for an A/V synchronization group. Empty string to disable. |