Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(63)

Side by Side Diff: webrtc/voice_engine/transmit_mixer.h

Issue 2245413002: Revert of Move FilePlayer and FileRecorder to Voice Engine (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 4 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/voice_engine/output_mixer.h ('k') | webrtc/voice_engine/voice_engine.gyp » ('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) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 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_VOICE_ENGINE_TRANSMIT_MIXER_H 11 #ifndef WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H
12 #define WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H 12 #define WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H
13 13
14 #include "webrtc/base/criticalsection.h" 14 #include "webrtc/base/criticalsection.h"
15 #include "webrtc/common_audio/resampler/include/push_resampler.h" 15 #include "webrtc/common_audio/resampler/include/push_resampler.h"
16 #include "webrtc/common_types.h" 16 #include "webrtc/common_types.h"
17 #include "webrtc/modules/audio_processing/typing_detection.h" 17 #include "webrtc/modules/audio_processing/typing_detection.h"
18 #include "webrtc/modules/include/module_common_types.h" 18 #include "webrtc/modules/include/module_common_types.h"
19 #include "webrtc/voice_engine/file_player.h" 19 #include "webrtc/modules/utility/include/file_player.h"
20 #include "webrtc/voice_engine/file_recorder.h" 20 #include "webrtc/modules/utility/include/file_recorder.h"
21 #include "webrtc/voice_engine/include/voe_base.h" 21 #include "webrtc/voice_engine/include/voe_base.h"
22 #include "webrtc/voice_engine/level_indicator.h" 22 #include "webrtc/voice_engine/level_indicator.h"
23 #include "webrtc/voice_engine/monitor_module.h" 23 #include "webrtc/voice_engine/monitor_module.h"
24 #include "webrtc/voice_engine/voice_engine_defines.h" 24 #include "webrtc/voice_engine/voice_engine_defines.h"
25 25
26 namespace webrtc { 26 namespace webrtc {
27 27
28 class AudioProcessing; 28 class AudioProcessing;
29 class ProcessThread; 29 class ProcessThread;
30 class VoEExternalMedia; 30 class VoEExternalMedia;
(...skipping 194 matching lines...) Expand 10 before | Expand all | Expand 10 after
225 bool _mute; 225 bool _mute;
226 bool stereo_codec_; 226 bool stereo_codec_;
227 bool swap_stereo_channels_; 227 bool swap_stereo_channels_;
228 }; 228 };
229 229
230 } // namespace voe 230 } // namespace voe
231 231
232 } // namespace webrtc 232 } // namespace webrtc
233 233
234 #endif // WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H 234 #endif // WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H
OLDNEW
« no previous file with comments | « webrtc/voice_engine/output_mixer.h ('k') | webrtc/voice_engine/voice_engine.gyp » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698