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Side by Side Diff: webrtc/audio_send_stream.h

Issue 1942823002: Remove webrtc/base/scoped_ptr.h (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: fix additional scoped_ptr uses that had been added Created 4 years, 7 months ago
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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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