Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(585)

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

Issue 3004723002: Move RtpExtension to api/ directory and config.h/.cc to call/. (Closed)
Patch Set: Rebase Created 3 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
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
11 #ifndef WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ 11 #ifndef WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_
12 #define WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ 12 #define WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_
13 13
14 #include <memory> 14 #include <memory>
15 #include <string> 15 #include <string>
16 #include <vector> 16 #include <vector>
17 17
18 #include "webrtc/api/rtpparameters.h" 18 #include "webrtc/api/rtpparameters.h"
19 #include "webrtc/api/rtpreceiverinterface.h" 19 #include "webrtc/api/rtpreceiverinterface.h"
20 #include "webrtc/api/video/video_timing.h" 20 #include "webrtc/api/video/video_timing.h"
21 #include "webrtc/config.h" 21 #include "webrtc/call/video_config.h"
22 #include "webrtc/media/base/codec.h" 22 #include "webrtc/media/base/codec.h"
23 #include "webrtc/media/base/mediaconstants.h" 23 #include "webrtc/media/base/mediaconstants.h"
24 #include "webrtc/media/base/streamparams.h" 24 #include "webrtc/media/base/streamparams.h"
25 #include "webrtc/media/base/videosinkinterface.h" 25 #include "webrtc/media/base/videosinkinterface.h"
26 #include "webrtc/media/base/videosourceinterface.h" 26 #include "webrtc/media/base/videosourceinterface.h"
27 #include "webrtc/rtc_base/basictypes.h" 27 #include "webrtc/rtc_base/basictypes.h"
28 #include "webrtc/rtc_base/buffer.h" 28 #include "webrtc/rtc_base/buffer.h"
29 #include "webrtc/rtc_base/copyonwritebuffer.h" 29 #include "webrtc/rtc_base/copyonwritebuffer.h"
30 #include "webrtc/rtc_base/dscp.h" 30 #include "webrtc/rtc_base/dscp.h"
31 #include "webrtc/rtc_base/logging.h" 31 #include "webrtc/rtc_base/logging.h"
(...skipping 1209 matching lines...) Expand 10 before | Expand all | Expand 10 after
1241 const char*, 1241 const char*,
1242 size_t> SignalDataReceived; 1242 size_t> SignalDataReceived;
1243 // Signal when the media channel is ready to send the stream. Arguments are: 1243 // Signal when the media channel is ready to send the stream. Arguments are:
1244 // writable(bool) 1244 // writable(bool)
1245 sigslot::signal1<bool> SignalReadyToSend; 1245 sigslot::signal1<bool> SignalReadyToSend;
1246 }; 1246 };
1247 1247
1248 } // namespace cricket 1248 } // namespace cricket
1249 1249
1250 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ 1250 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698