Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(143)

Side by Side Diff: webrtc/stream.h

Issue 1333483002: Wire up PacketTime to ReceiveStreams. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 3 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 | « webrtc/call.h ('k') | webrtc/test/fake_network_pipe.cc » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 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 #ifndef WEBRTC_STREAM_H_ 10 #ifndef WEBRTC_STREAM_H_
(...skipping 24 matching lines...) Expand all
35 virtual bool DeliverRtcp(const uint8_t* packet, size_t length) = 0; 35 virtual bool DeliverRtcp(const uint8_t* packet, size_t length) = 0;
36 36
37 protected: 37 protected:
38 virtual ~Stream() {} 38 virtual ~Stream() {}
39 }; 39 };
40 40
41 // Common base class for receive streams. 41 // Common base class for receive streams.
42 class ReceiveStream : public Stream { 42 class ReceiveStream : public Stream {
43 public: 43 public:
44 // Called when a RTP packet is received. 44 // Called when a RTP packet is received.
45 virtual bool DeliverRtp(const uint8_t* packet, size_t length) = 0; 45 virtual bool DeliverRtp(const uint8_t* packet,
46 size_t length,
47 const PacketTime& packet_time) = 0;
46 }; 48 };
47 49
48 // Common base class for send streams. 50 // Common base class for send streams.
49 // A tag class that denotes send stream type. 51 // A tag class that denotes send stream type.
50 class SendStream : public Stream {}; 52 class SendStream : public Stream {};
51 53
52 } // namespace webrtc 54 } // namespace webrtc
53 55
54 #endif // WEBRTC_STREAM_H_ 56 #endif // WEBRTC_STREAM_H_
OLDNEW
« no previous file with comments | « webrtc/call.h ('k') | webrtc/test/fake_network_pipe.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698