Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(227)

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

Issue 2059703002: Move webrtc/audio_*.h to webrtc/api/call (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebased on top of d565b73121b1b7672fb7d1f115bbbbb137a838eb Created 4 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
« no previous file with comments | « webrtc/media/BUILD.gn ('k') | webrtc/media/base/mediaengine.h » ('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) 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_FAKEMEDIAENGINE_H_ 11 #ifndef WEBRTC_MEDIA_BASE_FAKEMEDIAENGINE_H_
12 #define WEBRTC_MEDIA_BASE_FAKEMEDIAENGINE_H_ 12 #define WEBRTC_MEDIA_BASE_FAKEMEDIAENGINE_H_
13 13
14 #include <list> 14 #include <list>
15 #include <map> 15 #include <map>
16 #include <memory> 16 #include <memory>
17 #include <set> 17 #include <set>
18 #include <string> 18 #include <string>
19 #include <vector> 19 #include <vector>
20 20
21 #include "webrtc/audio_sink.h" 21 #include "webrtc/api/call/audio_sink.h"
22 #include "webrtc/base/copyonwritebuffer.h" 22 #include "webrtc/base/copyonwritebuffer.h"
23 #include "webrtc/base/networkroute.h" 23 #include "webrtc/base/networkroute.h"
24 #include "webrtc/base/stringutils.h" 24 #include "webrtc/base/stringutils.h"
25 #include "webrtc/media/base/audiosource.h" 25 #include "webrtc/media/base/audiosource.h"
26 #include "webrtc/media/base/mediaengine.h" 26 #include "webrtc/media/base/mediaengine.h"
27 #include "webrtc/media/base/rtputils.h" 27 #include "webrtc/media/base/rtputils.h"
28 #include "webrtc/media/base/streamparams.h" 28 #include "webrtc/media/base/streamparams.h"
29 #include "webrtc/p2p/base/sessiondescription.h" 29 #include "webrtc/p2p/base/sessiondescription.h"
30 30
31 using webrtc::RtpExtension; 31 using webrtc::RtpExtension;
(...skipping 927 matching lines...) Expand 10 before | Expand all | Expand 10 after
959 959
960 private: 960 private:
961 std::vector<FakeDataMediaChannel*> channels_; 961 std::vector<FakeDataMediaChannel*> channels_;
962 std::vector<DataCodec> data_codecs_; 962 std::vector<DataCodec> data_codecs_;
963 DataChannelType last_channel_type_; 963 DataChannelType last_channel_type_;
964 }; 964 };
965 965
966 } // namespace cricket 966 } // namespace cricket
967 967
968 #endif // WEBRTC_MEDIA_BASE_FAKEMEDIAENGINE_H_ 968 #endif // WEBRTC_MEDIA_BASE_FAKEMEDIAENGINE_H_
OLDNEW
« no previous file with comments | « webrtc/media/BUILD.gn ('k') | webrtc/media/base/mediaengine.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698