Repo created

TMessagesProj/jni/voip/webrtc/call/rtx_receive_stream.h

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+/*
+ *  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 CALL_RTX_RECEIVE_STREAM_H_
+#define CALL_RTX_RECEIVE_STREAM_H_
+
+#include <cstdint>
+#include <map>
+
+#include "api/sequence_checker.h"
+#include "call/rtp_packet_sink_interface.h"
+#include "rtc_base/system/no_unique_address.h"
+
+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,
+                   // 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;
+
+  // Update payload types post construction. Must be called from the same
+  // calling context as `OnRtpPacket` is called on.
+  void SetAssociatedPayloadTypes(std::map<int, int> associated_payload_types);
+
+  // RtpPacketSinkInterface.
+  void OnRtpPacket(const RtpPacketReceived& packet) override;
+
+ private:
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker packet_checker_;
+  RtpPacketSinkInterface* const media_sink_;
+  // Map from rtx payload type -> media payload type.
+  std::map<int, int> associated_payload_types_ RTC_GUARDED_BY(&packet_checker_);
+  // 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
+
+#endif  // CALL_RTX_RECEIVE_STREAM_H_