Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(28)

Side by Side Diff: webrtc/call/rtp_demuxer.h

Issue 2920993002: Add RSID-based demuxing to RtpDemuxer (Closed)
Patch Set: Created 3 years, 6 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
« no previous file with comments | « no previous file | webrtc/call/rtp_demuxer.cc » ('j') | webrtc/call/rtp_demuxer_unittest.cc » ('J')
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 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 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 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10
10 #ifndef WEBRTC_CALL_RTP_DEMUXER_H_ 11 #ifndef WEBRTC_CALL_RTP_DEMUXER_H_
11 #define WEBRTC_CALL_RTP_DEMUXER_H_ 12 #define WEBRTC_CALL_RTP_DEMUXER_H_
12 13
13 #include <map> 14 #include <map>
15 #include <set>
16 #include <string>
14 17
15 namespace webrtc { 18 namespace webrtc {
16 19
17 class RtpPacketReceived; 20 class RtpPacketReceived;
18 21
19 // This class represents a receiver of an already parsed RTP packets. 22 // This class represents a receiver of an already parsed RTP packets.
20 class RtpPacketSinkInterface { 23 class RtpPacketSinkInterface {
21 public: 24 public:
22 virtual ~RtpPacketSinkInterface() {} 25 virtual ~RtpPacketSinkInterface() {}
23 virtual void OnRtpPacket(const RtpPacketReceived& packet) = 0; 26 virtual void OnRtpPacket(const RtpPacketReceived& packet) = 0;
24 }; 27 };
25 28
26 // This class represents the RTP demuxing, for a single RTP session (i.e., one 29 // This class represents the RTP demuxing, for a single RTP session (i.e., one
27 // ssrc space, see RFC 7656). It isn't thread aware, leaving responsibility of 30 // ssrc space, see RFC 7656). It isn't thread aware, leaving responsibility of
28 // multithreading issues to the user of this class. 31 // multithreading issues to the user of this class.
29 // TODO(nisse): Should be extended to also do MID-based demux and payload-type 32 // TODO(nisse): Should be extended to also do MID-based demux and payload-type
30 // demux. 33 // demux.
31 class RtpDemuxer { 34 class RtpDemuxer {
32 public: 35 public:
33 RtpDemuxer(); 36 RtpDemuxer();
34 ~RtpDemuxer(); 37 ~RtpDemuxer();
35 38
36 // Registers a sink. The same sink can be registered for multiple ssrcs, and 39 // Registers a sink. The same sink can be registered for multiple ssrcs, and
37 // the same ssrc can have multiple sinks. Null pointer is not allowed. 40 // the same ssrc can have multiple sinks. Null pointer is not allowed.
38 void AddSink(uint32_t ssrc, RtpPacketSinkInterface* sink); 41 void AddSink(uint32_t ssrc, RtpPacketSinkInterface* sink);
42
43 // Registers a sink's association to an RSID. Null pointer is not allowed.
44 void AddSink(const std::string& rsid, RtpPacketSinkInterface* sink);
45
39 // Removes a sink. Returns deletion count (a sink may be registered 46 // Removes a sink. Returns deletion count (a sink may be registered
40 // for multiple ssrcs). Null pointer is not allowed. 47 // for multiple SSRCs and/or RSIDs). Null pointer is not allowed.
41 size_t RemoveSink(const RtpPacketSinkInterface* sink); 48 size_t RemoveSink(const RtpPacketSinkInterface* sink);
42 49
43 // Returns true if at least one matching sink was found, otherwise false. 50 // Returns true if at least one matching sink was found, otherwise false.
44 bool OnRtpPacket(const RtpPacketReceived& packet); 51 bool OnRtpPacket(const RtpPacketReceived& packet);
45 52
46 private: 53 private:
54 // Records a sink<->SSRC association. This can happen by explicit
55 // configuration by AddSink(ssrc...), or by inferred configuration from an
56 // RSID-based configuration which is resolved to an SSRC upon
57 // packet reception.
58 void RecordSsrcToSinkAssociation(uint32_t ssrc, RtpPacketSinkInterface* sink);
59
60 // When a new packet arrives, we attempt to resolve extra associations,
61 // such as which RSIDs are associated with which SSRCs.
62 void FindSsrcAssociations(const RtpPacketReceived& packet);
63
64 // This records the association SSRCs to sinks. Other associations, such
65 // as by RSID, also end up here once the RSID, etc., is resolved to an SSRC.
47 std::multimap<uint32_t, RtpPacketSinkInterface*> sinks_; 66 std::multimap<uint32_t, RtpPacketSinkInterface*> sinks_;
67
68 // A sink may be associated with an RSID - RTP Stream ID. This tag has a
69 // one-to-one association with an SSRC, but that SSRC is not yet known.
70 // When it becomes known, the association of the sink to the RSID is deleted
71 // from this container, and moved into |sinks_|.
72 std::multimap<std::string, RtpPacketSinkInterface*> rsid_sinks_;
73
74 // Iterating over |rsid_sinks_| for each incoming and performing multiple
75 // string comparisons is of non-trivial cost. To avoid this cost, we only
76 // check RSIDs for the first packet on each incoming SSRC stream.
77 // (If RSID associations are added later, we check again.)
78 std::set<uint32_t> processed_ssrcs_;
48 }; 79 };
49 80
50 } // namespace webrtc 81 } // namespace webrtc
51 82
52 #endif // WEBRTC_CALL_RTP_DEMUXER_H_ 83 #endif // WEBRTC_CALL_RTP_DEMUXER_H_
OLDNEW
« no previous file with comments | « no previous file | webrtc/call/rtp_demuxer.cc » ('j') | webrtc/call/rtp_demuxer_unittest.cc » ('J')

Powered by Google App Engine
This is Rietveld 408576698