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

Issue 2932073002: s/WebRtcVideoChannel2/WebRtcVideoChannel and s/WebRtcVideoEngine2/WebRtcVideoEngine (Closed)
Patch Set: . Created 3 years, 6 months ago
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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
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26 // webrtc/api/rtp/. 26 // webrtc/api/rtp/.
27 // 27 //
28 // For a start, this object is just a collection of the objects needed 28 // For a start, this object is just a collection of the objects needed
29 // by the VideoSendStream constructor. The plan is to move ownership 29 // by the VideoSendStream constructor. The plan is to move ownership
30 // of all RTP-related objects here, and add methods to create per-ssrc 30 // of all RTP-related objects here, and add methods to create per-ssrc
31 // objects which would then be passed to VideoSendStream. Eventually, 31 // objects which would then be passed to VideoSendStream. Eventually,
32 // direct accessors like packet_router() should be removed. 32 // direct accessors like packet_router() should be removed.
33 // 33 //
34 // This should also have a reference to the underlying 34 // This should also have a reference to the underlying
35 // webrtc::Transport(s). Currently, webrtc::Transport is implemented by 35 // webrtc::Transport(s). Currently, webrtc::Transport is implemented by
36 // WebRtcVideoChannel2 and WebRtcVoiceMediaChannel, and owned by 36 // WebRtcVideoChannel and WebRtcVoiceMediaChannel, and owned by
37 // WebrtcSession. Video and audio always uses different transport 37 // WebrtcSession. Video and audio always uses different transport
38 // objects, even in the common case where they are bundled over the 38 // objects, even in the common case where they are bundled over the
39 // same underlying transport. 39 // same underlying transport.
40 // 40 //
41 // Extracting the logic of the webrtc::Transport from BaseChannel and 41 // Extracting the logic of the webrtc::Transport from BaseChannel and
42 // subclasses into a separate class seems to be a prerequesite for 42 // subclasses into a separate class seems to be a prerequesite for
43 // moving the transport here. 43 // moving the transport here.
44 class RtpTransportControllerSendInterface { 44 class RtpTransportControllerSendInterface {
45 public: 45 public:
46 virtual ~RtpTransportControllerSendInterface() {} 46 virtual ~RtpTransportControllerSendInterface() {}
47 virtual PacketRouter* packet_router() = 0; 47 virtual PacketRouter* packet_router() = 0;
48 // Currently returning the same pointer, but with different types. 48 // Currently returning the same pointer, but with different types.
49 virtual SendSideCongestionController* send_side_cc() = 0; 49 virtual SendSideCongestionController* send_side_cc() = 0;
50 virtual TransportFeedbackObserver* transport_feedback_observer() = 0; 50 virtual TransportFeedbackObserver* transport_feedback_observer() = 0;
51 51
52 virtual RtpPacketSender* packet_sender() = 0; 52 virtual RtpPacketSender* packet_sender() = 0;
53 }; 53 };
54 54
55 } // namespace webrtc 55 } // namespace webrtc
56 56
57 #endif // WEBRTC_CALL_RTP_TRANSPORT_CONTROLLER_SEND_INTERFACE_H_ 57 #endif // WEBRTC_CALL_RTP_TRANSPORT_CONTROLLER_SEND_INTERFACE_H_
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