Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(193)

Side by Side Diff: webrtc/call/video_send_stream.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) 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2013 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_CALL_VIDEO_SEND_STREAM_H_ 11 #ifndef WEBRTC_CALL_VIDEO_SEND_STREAM_H_
12 #define WEBRTC_CALL_VIDEO_SEND_STREAM_H_ 12 #define WEBRTC_CALL_VIDEO_SEND_STREAM_H_
13 13
14 #include <map> 14 #include <map>
15 #include <string> 15 #include <string>
16 #include <utility> 16 #include <utility>
17 #include <vector> 17 #include <vector>
18 18
19 #include "webrtc/api/call/transport.h" 19 #include "webrtc/api/call/transport.h"
20 #include "webrtc/api/rtpparameters.h"
21 #include "webrtc/call/rtp_config.h"
22 #include "webrtc/call/video_config.h"
20 #include "webrtc/common_types.h" 23 #include "webrtc/common_types.h"
21 #include "webrtc/common_video/include/frame_callback.h" 24 #include "webrtc/common_video/include/frame_callback.h"
22 #include "webrtc/config.h"
23 #include "webrtc/media/base/videosinkinterface.h" 25 #include "webrtc/media/base/videosinkinterface.h"
24 #include "webrtc/media/base/videosourceinterface.h" 26 #include "webrtc/media/base/videosourceinterface.h"
25 #include "webrtc/rtc_base/platform_file.h" 27 #include "webrtc/rtc_base/platform_file.h"
26 28
27 namespace webrtc { 29 namespace webrtc {
28 30
29 class VideoEncoder; 31 class VideoEncoder;
30 32
31 class VideoSendStream { 33 class VideoSendStream {
32 public: 34 public:
(...skipping 240 matching lines...) Expand 10 before | Expand all | Expand 10 after
273 EnableEncodedFrameRecording(std::vector<rtc::PlatformFile>(), 0); 275 EnableEncodedFrameRecording(std::vector<rtc::PlatformFile>(), 0);
274 } 276 }
275 277
276 protected: 278 protected:
277 virtual ~VideoSendStream() {} 279 virtual ~VideoSendStream() {}
278 }; 280 };
279 281
280 } // namespace webrtc 282 } // namespace webrtc
281 283
282 #endif // WEBRTC_CALL_VIDEO_SEND_STREAM_H_ 284 #endif // WEBRTC_CALL_VIDEO_SEND_STREAM_H_
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698