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Side by Side Diff: webrtc/audio_receive_stream.h

Issue 1588693002: Revert of Storing raw audio sink for default audio track. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 11 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_RECEIVE_STREAM_H_ 11 #ifndef WEBRTC_AUDIO_RECEIVE_STREAM_H_
12 #define WEBRTC_AUDIO_RECEIVE_STREAM_H_ 12 #define WEBRTC_AUDIO_RECEIVE_STREAM_H_
13 13
14 #include <map> 14 #include <map>
15 #include <string> 15 #include <string>
16 #include <vector> 16 #include <vector>
17 17
18 #include "webrtc/base/scoped_ref_ptr.h" 18 #include "webrtc/base/scoped_ptr.h"
19 #include "webrtc/config.h" 19 #include "webrtc/config.h"
20 #include "webrtc/stream.h" 20 #include "webrtc/stream.h"
21 #include "webrtc/transport.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 class AudioDecoder; 26 class AudioDecoder;
27 class AudioSinkInterface; 27 class AudioSinkInterface;
28 28
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99 99
100 // TODO(pbos): Remove config option once combined A/V BWE is always on. 100 // TODO(pbos): Remove config option once combined A/V BWE is always on.
101 bool combined_audio_video_bwe = false; 101 bool combined_audio_video_bwe = false;
102 }; 102 };
103 103
104 virtual Stats GetStats() const = 0; 104 virtual Stats GetStats() const = 0;
105 105
106 // Sets an audio sink that receives unmixed audio from the receive stream. 106 // Sets an audio sink that receives unmixed audio from the receive stream.
107 // Ownership of the sink is passed to the stream and can be used by the 107 // Ownership of the sink is passed to the stream and can be used by the
108 // caller to do lifetime management (i.e. when the sink's dtor is called). 108 // caller to do lifetime management (i.e. when the sink's dtor is called).
109 // Only one sink can be set and passing a null sink clears an existing one. 109 // Only one sink can be set and passing a null sink, clears an existing one.
110 // NOTE: Audio must still somehow be pulled through AudioTransport for audio 110 // NOTE: Audio must still somehow be pulled through AudioTransport for audio
111 // to stream through this sink. In practice, this happens if mixed audio 111 // to stream through this sink. In practice, this happens if mixed audio
112 // is being pulled+rendered and/or if audio is being pulled for the purposes 112 // is being pulled+rendered and/or if audio is being pulled for the purposes
113 // of feeding to the AEC. 113 // of feeding to the AEC.
114 virtual void SetSink(const rtc::scoped_refptr<AudioSinkInterface>& sink) = 0; 114 virtual void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) = 0;
115 }; 115 };
116 } // namespace webrtc 116 } // namespace webrtc
117 117
118 #endif // WEBRTC_AUDIO_RECEIVE_STREAM_H_ 118 #endif // WEBRTC_AUDIO_RECEIVE_STREAM_H_
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