Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(1)

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

Issue 1481493004: audio_coding: remove "main" directory (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebased Created 5 years 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/modules/utility/source/coder.h ('k') | webrtc/voice_engine/voe_base_impl.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) 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_CHANNEL_H_ 11 #ifndef WEBRTC_VOICE_ENGINE_CHANNEL_H_
12 #define WEBRTC_VOICE_ENGINE_CHANNEL_H_ 12 #define WEBRTC_VOICE_ENGINE_CHANNEL_H_
13 13
14 #include "webrtc/base/scoped_ptr.h" 14 #include "webrtc/base/scoped_ptr.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_coding/main/include/audio_coding_module.h" 17 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
18 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_d efines.h" 18 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_d efines.h"
19 #include "webrtc/modules/audio_processing/rms_level.h" 19 #include "webrtc/modules/audio_processing/rms_level.h"
20 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h" 20 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
21 #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h" 21 #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
22 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" 22 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
23 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" 23 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
24 #include "webrtc/modules/utility/include/file_player.h" 24 #include "webrtc/modules/utility/include/file_player.h"
25 #include "webrtc/modules/utility/include/file_recorder.h" 25 #include "webrtc/modules/utility/include/file_recorder.h"
26 #include "webrtc/voice_engine/dtmf_inband.h" 26 #include "webrtc/voice_engine/dtmf_inband.h"
27 #include "webrtc/voice_engine/dtmf_inband_queue.h" 27 #include "webrtc/voice_engine/dtmf_inband_queue.h"
(...skipping 554 matching lines...) Expand 10 before | Expand all | Expand 10 after
582 rtc::scoped_ptr<NetworkPredictor> network_predictor_; 582 rtc::scoped_ptr<NetworkPredictor> network_predictor_;
583 // An associated send channel. 583 // An associated send channel.
584 rtc::scoped_ptr<CriticalSectionWrapper> assoc_send_channel_lock_; 584 rtc::scoped_ptr<CriticalSectionWrapper> assoc_send_channel_lock_;
585 ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_); 585 ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_);
586 }; 586 };
587 587
588 } // namespace voe 588 } // namespace voe
589 } // namespace webrtc 589 } // namespace webrtc
590 590
591 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ 591 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_
OLDNEW
« no previous file with comments | « webrtc/modules/utility/source/coder.h ('k') | webrtc/voice_engine/voe_base_impl.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698