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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_ptr.h" |
18 #include "webrtc/config.h" | 19 #include "webrtc/config.h" |
19 #include "webrtc/stream.h" | 20 #include "webrtc/stream.h" |
20 #include "webrtc/transport.h" | 21 #include "webrtc/transport.h" |
21 #include "webrtc/typedefs.h" | 22 #include "webrtc/typedefs.h" |
22 | 23 |
23 namespace webrtc { | 24 namespace webrtc { |
24 | 25 |
25 class AudioDecoder; | 26 class AudioDecoder; |
| 27 class AudioSinkInterface; |
26 | 28 |
27 // WORK IN PROGRESS | 29 // WORK IN PROGRESS |
28 // This class is under development and is not yet intended for for use outside | 30 // This class is under development and is not yet intended for for use outside |
29 // of WebRtc/Libjingle. Please use the VoiceEngine API instead. | 31 // of WebRtc/Libjingle. Please use the VoiceEngine API instead. |
30 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690 | 32 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690 |
31 | 33 |
32 class AudioReceiveStream : public ReceiveStream { | 34 class AudioReceiveStream : public ReceiveStream { |
33 public: | 35 public: |
34 struct Stats { | 36 struct Stats { |
35 uint32_t remote_ssrc = 0; | 37 uint32_t remote_ssrc = 0; |
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93 // AudioDecoder instances once the Config is submitted to | 95 // AudioDecoder instances once the Config is submitted to |
94 // Call::CreateReceiveStream(). | 96 // Call::CreateReceiveStream(). |
95 // TODO(solenberg): Use unique_ptr<> once our std lib fully supports C++11. | 97 // TODO(solenberg): Use unique_ptr<> once our std lib fully supports C++11. |
96 std::map<uint8_t, AudioDecoder*> decoder_map; | 98 std::map<uint8_t, AudioDecoder*> decoder_map; |
97 | 99 |
98 // 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. |
99 bool combined_audio_video_bwe = false; | 101 bool combined_audio_video_bwe = false; |
100 }; | 102 }; |
101 | 103 |
102 virtual Stats GetStats() const = 0; | 104 virtual Stats GetStats() const = 0; |
| 105 |
| 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 |
| 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. |
| 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 |
| 112 // is being pulled+rendered and/or if audio is being pulled for the purposes |
| 113 // of feeding to the AEC. |
| 114 virtual void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) = 0; |
103 }; | 115 }; |
104 } // namespace webrtc | 116 } // namespace webrtc |
105 | 117 |
106 #endif // WEBRTC_AUDIO_RECEIVE_STREAM_H_ | 118 #endif // WEBRTC_AUDIO_RECEIVE_STREAM_H_ |
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