nisse | e4bcd6d | 2017-05-16 11:47:04 | [diff] [blame] | 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 | #ifndef WEBRTC_CALL_RTP_DEMUXER_H_ |
| 11 | #define WEBRTC_CALL_RTP_DEMUXER_H_ |
| 12 | |
| 13 | #include <map> |
| 14 | |
| 15 | namespace webrtc { |
| 16 | |
| 17 | class RtpPacketReceived; |
nisse | d76b7b2 | 2017-06-01 11:02:35 | [diff] [blame] | 18 | class RtpPacketSinkInterface; |
nisse | e4bcd6d | 2017-05-16 11:47:04 | [diff] [blame] | 19 | |
| 20 | // This class represents the RTP demuxing, for a single RTP session (i.e., one |
| 21 | // ssrc space, see RFC 7656). It isn't thread aware, leaving responsibility of |
| 22 | // multithreading issues to the user of this class. |
| 23 | // TODO(nisse): Should be extended to also do MID-based demux and payload-type |
| 24 | // demux. |
| 25 | class RtpDemuxer { |
| 26 | public: |
| 27 | RtpDemuxer(); |
| 28 | ~RtpDemuxer(); |
| 29 | |
| 30 | // Registers a sink. The same sink can be registered for multiple ssrcs, and |
| 31 | // the same ssrc can have multiple sinks. Null pointer is not allowed. |
| 32 | void AddSink(uint32_t ssrc, RtpPacketSinkInterface* sink); |
| 33 | // Removes a sink. Returns deletion count (a sink may be registered |
| 34 | // for multiple ssrcs). Null pointer is not allowed. |
| 35 | size_t RemoveSink(const RtpPacketSinkInterface* sink); |
| 36 | |
| 37 | // Returns true if at least one matching sink was found, otherwise false. |
| 38 | bool OnRtpPacket(const RtpPacketReceived& packet); |
| 39 | |
| 40 | private: |
| 41 | std::multimap<uint32_t, RtpPacketSinkInterface*> sinks_; |
| 42 | }; |
| 43 | |
| 44 | } // namespace webrtc |
| 45 | |
| 46 | #endif // WEBRTC_CALL_RTP_DEMUXER_H_ |