OLD | NEW |
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_API_CALL_AUDIO_SEND_STREAM_H_ | 11 #ifndef WEBRTC_API_CALL_AUDIO_SEND_STREAM_H_ |
12 #define WEBRTC_API_CALL_AUDIO_SEND_STREAM_H_ | 12 #define WEBRTC_API_CALL_AUDIO_SEND_STREAM_H_ |
13 | 13 |
14 #include <memory> | 14 #include <memory> |
15 #include <string> | 15 #include <string> |
16 #include <vector> | 16 #include <vector> |
17 | 17 |
| 18 #include "webrtc/api/call/transport.h" |
18 #include "webrtc/base/optional.h" | 19 #include "webrtc/base/optional.h" |
19 #include "webrtc/config.h" | 20 #include "webrtc/config.h" |
20 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h" | 21 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h" |
21 #include "webrtc/transport.h" | |
22 #include "webrtc/typedefs.h" | 22 #include "webrtc/typedefs.h" |
23 | 23 |
24 namespace webrtc { | 24 namespace webrtc { |
25 | 25 |
26 // WORK IN PROGRESS | 26 // WORK IN PROGRESS |
27 // This class is under development and is not yet intended for for use outside | 27 // This class is under development and is not yet intended for for use outside |
28 // of WebRtc/Libjingle. Please use the VoiceEngine API instead. | 28 // of WebRtc/Libjingle. Please use the VoiceEngine API instead. |
29 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690 | 29 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690 |
30 | 30 |
31 class AudioSendStream { | 31 class AudioSendStream { |
(...skipping 104 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
136 virtual void SetMuted(bool muted) = 0; | 136 virtual void SetMuted(bool muted) = 0; |
137 | 137 |
138 virtual Stats GetStats() const = 0; | 138 virtual Stats GetStats() const = 0; |
139 | 139 |
140 protected: | 140 protected: |
141 virtual ~AudioSendStream() {} | 141 virtual ~AudioSendStream() {} |
142 }; | 142 }; |
143 } // namespace webrtc | 143 } // namespace webrtc |
144 | 144 |
145 #endif // WEBRTC_API_CALL_AUDIO_SEND_STREAM_H_ | 145 #endif // WEBRTC_API_CALL_AUDIO_SEND_STREAM_H_ |
OLD | NEW |