Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(159)

Side by Side Diff: webrtc/audio/audio_receive_stream.h

Issue 1417683006: modules: more interface -> include renames (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fix last incorrectly wrapped pragma message 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 | « no previous file | webrtc/call/call.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) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 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_AUDIO_AUDIO_RECEIVE_STREAM_H_ 11 #ifndef WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
12 #define WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ 12 #define WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
13 13
14 #include "webrtc/audio_receive_stream.h" 14 #include "webrtc/audio_receive_stream.h"
15 #include "webrtc/audio/scoped_voe_interface.h" 15 #include "webrtc/audio/scoped_voe_interface.h"
16 #include "webrtc/base/thread_checker.h" 16 #include "webrtc/base/thread_checker.h"
17 #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" 17 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
18 #include "webrtc/voice_engine/include/voe_base.h" 18 #include "webrtc/voice_engine/include/voe_base.h"
19 19
20 namespace webrtc { 20 namespace webrtc {
21 21
22 class RemoteBitrateEstimator; 22 class RemoteBitrateEstimator;
23 class VoiceEngine; 23 class VoiceEngine;
24 24
25 namespace internal { 25 namespace internal {
26 26
27 class AudioReceiveStream final : public webrtc::AudioReceiveStream { 27 class AudioReceiveStream final : public webrtc::AudioReceiveStream {
(...skipping 25 matching lines...) Expand all
53 // We hold one interface pointer to the VoE to make sure it is kept alive. 53 // We hold one interface pointer to the VoE to make sure it is kept alive.
54 ScopedVoEInterface<VoEBase> voe_base_; 54 ScopedVoEInterface<VoEBase> voe_base_;
55 rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_; 55 rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_;
56 56
57 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream); 57 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream);
58 }; 58 };
59 } // namespace internal 59 } // namespace internal
60 } // namespace webrtc 60 } // namespace webrtc
61 61
62 #endif // WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ 62 #endif // WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
OLDNEW
« no previous file with comments | « no previous file | webrtc/call/call.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698