1 /* 2 * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef CALL_RTX_RECEIVE_STREAM_H_ 12 #define CALL_RTX_RECEIVE_STREAM_H_ 13 14 #include <cstdint> 15 #include <map> 16 17 #include "api/sequence_checker.h" 18 #include "call/rtp_packet_sink_interface.h" 19 #include "rtc_base/system/no_unique_address.h" 20 21 namespace webrtc { 22 23 class ReceiveStatistics; 24 25 // This class is responsible for RTX decapsulation. The resulting media packets 26 // are passed on to a sink representing the associated media stream. 27 class RtxReceiveStream : public RtpPacketSinkInterface { 28 public: 29 RtxReceiveStream(RtpPacketSinkInterface* media_sink, 30 std::map<int, int> associated_payload_types, 31 uint32_t media_ssrc, 32 // TODO(nisse): Delete this argument, and 33 // corresponding member variable, by moving the 34 // responsibility for rtcp feedback to 35 // RtpStreamReceiverController. 36 ReceiveStatistics* rtp_receive_statistics = nullptr); 37 ~RtxReceiveStream() override; 38 39 // Update payload types post construction. Must be called from the same 40 // calling context as `OnRtpPacket` is called on. 41 void SetAssociatedPayloadTypes(std::map<int, int> associated_payload_types); 42 43 // RtpPacketSinkInterface. 44 void OnRtpPacket(const RtpPacketReceived& packet) override; 45 46 private: 47 RTC_NO_UNIQUE_ADDRESS SequenceChecker packet_checker_; 48 RtpPacketSinkInterface* const media_sink_; 49 // Map from rtx payload type -> media payload type. 50 std::map<int, int> associated_payload_types_ RTC_GUARDED_BY(&packet_checker_); 51 // TODO(nisse): Ultimately, the media receive stream shouldn't care about the 52 // ssrc, and we should delete this. 53 const uint32_t media_ssrc_; 54 ReceiveStatistics* const rtp_receive_statistics_; 55 }; 56 57 } // namespace webrtc 58 59 #endif // CALL_RTX_RECEIVE_STREAM_H_ 60