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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_SEND_STREAM_H_ | 11 #ifndef WEBRTC_AUDIO_SEND_STREAM_H_ |
12 #define WEBRTC_AUDIO_SEND_STREAM_H_ | 12 #define WEBRTC_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/base/scoped_ptr.h" | |
19 #include "webrtc/config.h" | 18 #include "webrtc/config.h" |
20 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h" | 19 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h" |
21 #include "webrtc/transport.h" | 20 #include "webrtc/transport.h" |
22 #include "webrtc/typedefs.h" | 21 #include "webrtc/typedefs.h" |
23 | 22 |
24 namespace webrtc { | 23 namespace webrtc { |
25 | 24 |
26 // WORK IN PROGRESS | 25 // WORK IN PROGRESS |
27 // This class is under development and is not yet intended for for use outside | 26 // This class is under development and is not yet intended for for use outside |
28 // of WebRtc/Libjingle. Please use the VoiceEngine API instead. | 27 // of WebRtc/Libjingle. Please use the VoiceEngine API instead. |
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100 virtual bool SendTelephoneEvent(int payload_type, int event, | 99 virtual bool SendTelephoneEvent(int payload_type, int event, |
101 int duration_ms) = 0; | 100 int duration_ms) = 0; |
102 virtual Stats GetStats() const = 0; | 101 virtual Stats GetStats() const = 0; |
103 | 102 |
104 protected: | 103 protected: |
105 virtual ~AudioSendStream() {} | 104 virtual ~AudioSendStream() {} |
106 }; | 105 }; |
107 } // namespace webrtc | 106 } // namespace webrtc |
108 | 107 |
109 #endif // WEBRTC_AUDIO_SEND_STREAM_H_ | 108 #endif // WEBRTC_AUDIO_SEND_STREAM_H_ |
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