Index: webrtc/call/rtx_receive_stream.h |
diff --git a/webrtc/call/rtx_receive_stream.h b/webrtc/call/rtx_receive_stream.h |
index 418775c7780d3d349694e10b99063237d31e3d76..c288a27edac7266c50a49effe38e852ee4a83225 100644 |
--- a/webrtc/call/rtx_receive_stream.h |
+++ b/webrtc/call/rtx_receive_stream.h |
@@ -17,13 +17,20 @@ |
namespace webrtc { |
+class ReceiveStatistics; |
+ |
// This class is responsible for RTX decapsulation. The resulting media packets |
// are passed on to a sink representing the associated media stream. |
class RtxReceiveStream : public RtpPacketSinkInterface { |
public: |
RtxReceiveStream(RtpPacketSinkInterface* media_sink, |
std::map<int, int> associated_payload_types, |
- uint32_t media_ssrc); |
+ uint32_t media_ssrc, |
+ // TODO(nisse): Delete this argument, and |
+ // corresponding member variable, by moving the |
+ // responsibility for rtcp feedback to |
+ // RtpStreamReceiverController. |
+ ReceiveStatistics* rtp_receive_statistics = nullptr); |
~RtxReceiveStream() override; |
// RtpPacketSinkInterface. |
void OnRtpPacket(const RtpPacketReceived& packet) override; |
@@ -35,6 +42,7 @@ class RtxReceiveStream : public RtpPacketSinkInterface { |
// TODO(nisse): Ultimately, the media receive stream shouldn't care about the |
// ssrc, and we should delete this. |
const uint32_t media_ssrc_; |
+ ReceiveStatistics* const rtp_receive_statistics_; |
}; |
} // namespace webrtc |