Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(30)

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

Issue 3004723002: Move RtpExtension to api/ directory and config.h/.cc to call/. (Closed)
Patch Set: Fix nit. Created 3 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/call/BUILD.gn ('k') | webrtc/call/audio_send_stream.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) 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_CALL_AUDIO_RECEIVE_STREAM_H_ 11 #ifndef WEBRTC_CALL_AUDIO_RECEIVE_STREAM_H_
12 #define WEBRTC_CALL_AUDIO_RECEIVE_STREAM_H_ 12 #define WEBRTC_CALL_AUDIO_RECEIVE_STREAM_H_
13 13
14 #include <map> 14 #include <map>
15 #include <memory> 15 #include <memory>
16 #include <string> 16 #include <string>
17 #include <vector> 17 #include <vector>
18 18
19 #include "webrtc/api/audio_codecs/audio_decoder_factory.h" 19 #include "webrtc/api/audio_codecs/audio_decoder_factory.h"
20 #include "webrtc/api/call/transport.h" 20 #include "webrtc/api/call/transport.h"
21 #include "webrtc/api/rtpparameters.h"
21 #include "webrtc/api/rtpreceiverinterface.h" 22 #include "webrtc/api/rtpreceiverinterface.h"
23 #include "webrtc/call/rtp_config.h"
22 #include "webrtc/common_types.h" 24 #include "webrtc/common_types.h"
23 #include "webrtc/config.h"
24 #include "webrtc/rtc_base/optional.h" 25 #include "webrtc/rtc_base/optional.h"
25 #include "webrtc/rtc_base/scoped_ref_ptr.h" 26 #include "webrtc/rtc_base/scoped_ref_ptr.h"
26 #include "webrtc/typedefs.h" 27 #include "webrtc/typedefs.h"
27 28
28 namespace webrtc { 29 namespace webrtc {
29 class AudioSinkInterface; 30 class AudioSinkInterface;
30 31
31 // WORK IN PROGRESS 32 // WORK IN PROGRESS
32 // This class is under development and is not yet intended for for use outside 33 // This class is under development and is not yet intended for for use outside
33 // of WebRtc/Libjingle. Please use the VoiceEngine API instead. 34 // of WebRtc/Libjingle. Please use the VoiceEngine API instead.
(...skipping 114 matching lines...) Expand 10 before | Expand all | Expand 10 after
148 virtual void SetGain(float gain) = 0; 149 virtual void SetGain(float gain) = 0;
149 150
150 virtual std::vector<RtpSource> GetSources() const = 0; 151 virtual std::vector<RtpSource> GetSources() const = 0;
151 152
152 protected: 153 protected:
153 virtual ~AudioReceiveStream() {} 154 virtual ~AudioReceiveStream() {}
154 }; 155 };
155 } // namespace webrtc 156 } // namespace webrtc
156 157
157 #endif // WEBRTC_CALL_AUDIO_RECEIVE_STREAM_H_ 158 #endif // WEBRTC_CALL_AUDIO_RECEIVE_STREAM_H_
OLDNEW
« no previous file with comments | « webrtc/call/BUILD.gn ('k') | webrtc/call/audio_send_stream.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698