rtx_receive_stream.h (2261B)
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 #include "rtc_base/thread_annotations.h" 21 22 namespace webrtc { 23 24 class ReceiveStatistics; 25 26 // This class is responsible for RTX decapsulation. The resulting media packets 27 // are passed on to a sink representing the associated media stream. 28 class RtxReceiveStream : public RtpPacketSinkInterface { 29 public: 30 RtxReceiveStream(RtpPacketSinkInterface* media_sink, 31 std::map<int, int> associated_payload_types, 32 uint32_t media_ssrc, 33 // TODO(nisse): Delete this argument, and 34 // corresponding member variable, by moving the 35 // responsibility for rtcp feedback to 36 // RtpStreamReceiverController. 37 ReceiveStatistics* rtp_receive_statistics = nullptr); 38 ~RtxReceiveStream() override; 39 40 // Update payload types post construction. Must be called from the same 41 // calling context as `OnRtpPacket` is called on. 42 void SetAssociatedPayloadTypes(std::map<int, int> associated_payload_types); 43 44 // RtpPacketSinkInterface. 45 void OnRtpPacket(const RtpPacketReceived& packet) override; 46 47 private: 48 RTC_NO_UNIQUE_ADDRESS SequenceChecker packet_checker_; 49 RtpPacketSinkInterface* const media_sink_; 50 // Map from rtx payload type -> media payload type. 51 std::map<int, int> associated_payload_types_ RTC_GUARDED_BY(&packet_checker_); 52 // TODO(nisse): Ultimately, the media receive stream shouldn't care about the 53 // ssrc, and we should delete this. 54 const uint32_t media_ssrc_; 55 ReceiveStatistics* const rtp_receive_statistics_; 56 }; 57 58 } // namespace webrtc 59 60 #endif // CALL_RTX_RECEIVE_STREAM_H_