Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(19)

Side by Side Diff: webrtc/media/base/mediachannel.h

Issue 2437503004: Set actual transport overhead in rtp_rtcp (Closed)
Patch Set: Response to comments of honghaiz3 Created 4 years, 1 month 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
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2004 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
(...skipping 377 matching lines...) Expand 10 before | Expand all | Expand 10 after
388 const rtc::PacketTime& packet_time) = 0; 388 const rtc::PacketTime& packet_time) = 0;
389 // Called when a RTCP packet is received. 389 // Called when a RTCP packet is received.
390 virtual void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet, 390 virtual void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
391 const rtc::PacketTime& packet_time) = 0; 391 const rtc::PacketTime& packet_time) = 0;
392 // Called when the socket's ability to send has changed. 392 // Called when the socket's ability to send has changed.
393 virtual void OnReadyToSend(bool ready) = 0; 393 virtual void OnReadyToSend(bool ready) = 0;
394 // Called when the network route used for sending packets changed. 394 // Called when the network route used for sending packets changed.
395 virtual void OnNetworkRouteChanged( 395 virtual void OnNetworkRouteChanged(
396 const std::string& transport_name, 396 const std::string& transport_name,
397 const rtc::NetworkRoute& network_route) = 0; 397 const rtc::NetworkRoute& network_route) = 0;
398 // Called when the rtp transport overhead changed.
399 virtual void OnTransportOverheadChange(int transport_overhead_per_packet) = 0;
398 // Creates a new outgoing media stream with SSRCs and CNAME as described 400 // Creates a new outgoing media stream with SSRCs and CNAME as described
399 // by sp. 401 // by sp.
400 virtual bool AddSendStream(const StreamParams& sp) = 0; 402 virtual bool AddSendStream(const StreamParams& sp) = 0;
401 // Removes an outgoing media stream. 403 // Removes an outgoing media stream.
402 // ssrc must be the first SSRC of the media stream if the stream uses 404 // ssrc must be the first SSRC of the media stream if the stream uses
403 // multiple SSRCs. 405 // multiple SSRCs.
404 virtual bool RemoveSendStream(uint32_t ssrc) = 0; 406 virtual bool RemoveSendStream(uint32_t ssrc) = 0;
405 // Creates a new incoming media stream with SSRCs and CNAME as described 407 // Creates a new incoming media stream with SSRCs and CNAME as described
406 // by sp. 408 // by sp.
407 virtual bool AddRecvStream(const StreamParams& sp) = 0; 409 virtual bool AddRecvStream(const StreamParams& sp) = 0;
(...skipping 734 matching lines...) Expand 10 before | Expand all | Expand 10 after
1142 // Signal when the media channel is ready to send the stream. Arguments are: 1144 // Signal when the media channel is ready to send the stream. Arguments are:
1143 // writable(bool) 1145 // writable(bool)
1144 sigslot::signal1<bool> SignalReadyToSend; 1146 sigslot::signal1<bool> SignalReadyToSend;
1145 // Signal for notifying that the remote side has closed the DataChannel. 1147 // Signal for notifying that the remote side has closed the DataChannel.
1146 sigslot::signal1<uint32_t> SignalStreamClosedRemotely; 1148 sigslot::signal1<uint32_t> SignalStreamClosedRemotely;
1147 }; 1149 };
1148 1150
1149 } // namespace cricket 1151 } // namespace cricket
1150 1152
1151 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ 1153 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698