Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(256)

Side by Side Diff: webrtc/video/video_send_stream.h

Issue 1413333002: system_wrappers: rename interface -> include (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebased again! Created 5 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
« no previous file with comments | « webrtc/video/video_receive_stream.cc ('k') | webrtc/video/video_send_stream_tests.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) 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_VIDEO_VIDEO_SEND_STREAM_H_ 11 #ifndef WEBRTC_VIDEO_VIDEO_SEND_STREAM_H_
12 #define WEBRTC_VIDEO_VIDEO_SEND_STREAM_H_ 12 #define WEBRTC_VIDEO_VIDEO_SEND_STREAM_H_
13 13
14 #include <map> 14 #include <map>
15 #include <vector> 15 #include <vector>
16 16
17 #include "webrtc/call.h" 17 #include "webrtc/call.h"
18 #include "webrtc/call/transport_adapter.h" 18 #include "webrtc/call/transport_adapter.h"
19 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" 19 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
20 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" 20 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
21 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" 21 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
22 #include "webrtc/video/encoded_frame_callback_adapter.h" 22 #include "webrtc/video/encoded_frame_callback_adapter.h"
23 #include "webrtc/video/send_statistics_proxy.h" 23 #include "webrtc/video/send_statistics_proxy.h"
24 #include "webrtc/video/video_capture_input.h" 24 #include "webrtc/video/video_capture_input.h"
25 #include "webrtc/video_receive_stream.h" 25 #include "webrtc/video_receive_stream.h"
26 #include "webrtc/video_send_stream.h" 26 #include "webrtc/video_send_stream.h"
27 27
28 namespace webrtc { 28 namespace webrtc {
29 29
30 class CallStats; 30 class CallStats;
31 class CongestionController; 31 class CongestionController;
(...skipping 59 matching lines...) Expand 10 before | Expand all | Expand 10 after
91 // start bitrate initially, instead of the one reported by VideoEngine (which 91 // start bitrate initially, instead of the one reported by VideoEngine (which
92 // defaults to too high). 92 // defaults to too high).
93 bool use_config_bitrate_; 93 bool use_config_bitrate_;
94 94
95 SendStatisticsProxy stats_proxy_; 95 SendStatisticsProxy stats_proxy_;
96 }; 96 };
97 } // namespace internal 97 } // namespace internal
98 } // namespace webrtc 98 } // namespace webrtc
99 99
100 #endif // WEBRTC_VIDEO_VIDEO_SEND_STREAM_H_ 100 #endif // WEBRTC_VIDEO_VIDEO_SEND_STREAM_H_
OLDNEW
« no previous file with comments | « webrtc/video/video_receive_stream.cc ('k') | webrtc/video/video_send_stream_tests.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698