New class RtxReceiveStream.

webrtc/call/rtx_receive_stream.h

Index: webrtc/call/rtx_receive_stream.h
diff --git a/webrtc/call/rtx_receive_stream.h b/webrtc/call/rtx_receive_stream.h
new file mode 100644
index 0000000000000000000000000000000000000000..06b64d363d3e78d7a8c8569184c07c45689e333c
--- /dev/null
+++ b/webrtc/call/rtx_receive_stream.h
@@ -0,0 +1,40 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_CALL_RTX_RECEIVE_STREAM_H_
+#define WEBRTC_CALL_RTX_RECEIVE_STREAM_H_
+
+#include <map>
+
+#include "webrtc/call/rtp_demuxer.h"
+
+namespace webrtc {
+
+class RtxReceiveStream : public RtpPacketSinkInterface {
+ public:
+  RtxReceiveStream(RtpPacketSinkInterface* media_sink,
+                   const std::map<int, int> rtx_payload_type_map,

[danilchap, 2017/05/17 13:43:56]

remove const
(it is helpful for const&, but when passing by value it is recommended not to
write it at least in the declaration)

[nisse-webrtc, 2017/05/18 09:13:20]

Ooop, I left this in as an experiment, I would have expected std::move on a
const value to give a compilation error, but I didn't get any (instead, things
are always copied). See
http://stackoverflow.com/questions/28595117/why-can-we-use-stdmove-on-a-const....

I'm removing const.

+                   uint32_t media_ssrc);
+
+  // RtpPacketSinkInterface.
+  void OnRtpPacket(const RtpPacketReceived& packet) override;
+
+ private:
+  RtpPacketSinkInterface* const media_sink_;
+  // Mapping rtx_payload_type_map_[rtx] = associated.
+  const std::map<int, int> rtx_payload_type_map_;
+  // TODO(nisse): Ultimately, the media receive stream shouldn't care about the
+  // ssrc, and we should delete this.
+  const uint32_t media_ssrc_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_CALL_RTX_RECEIVE_STREAM_H_