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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 #ifndef WEBRTC_CALL_AUDIO_STATE_H_ | 10 #ifndef WEBRTC_CALL_AUDIO_STATE_H_ |
11 #define WEBRTC_CALL_AUDIO_STATE_H_ | 11 #define WEBRTC_CALL_AUDIO_STATE_H_ |
12 | 12 |
13 #include "webrtc/api/audio/audio_mixer.h" | 13 #include "webrtc/api/audio/audio_mixer.h" |
14 #include "webrtc/base/refcount.h" | 14 #include "webrtc/base/refcount.h" |
15 #include "webrtc/base/scoped_ref_ptr.h" | 15 #include "webrtc/base/scoped_ref_ptr.h" |
16 | 16 |
17 namespace webrtc { | 17 namespace webrtc { |
18 | 18 |
| 19 class AudioProcessing; |
19 class VoiceEngine; | 20 class VoiceEngine; |
20 | 21 |
21 // WORK IN PROGRESS | 22 // WORK IN PROGRESS |
22 // This class is under development and is not yet intended for for use outside | 23 // This class is under development and is not yet intended for for use outside |
23 // of WebRtc/Libjingle. Please use the VoiceEngine API instead. | 24 // of WebRtc/Libjingle. Please use the VoiceEngine API instead. |
24 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690 | 25 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690 |
25 | 26 |
26 // AudioState holds the state which must be shared between multiple instances of | 27 // AudioState holds the state which must be shared between multiple instances of |
27 // webrtc::Call for audio processing purposes. | 28 // webrtc::Call for audio processing purposes. |
28 class AudioState : public rtc::RefCountInterface { | 29 class AudioState : public rtc::RefCountInterface { |
29 public: | 30 public: |
30 struct Config { | 31 struct Config { |
31 // VoiceEngine used for audio streams and audio/video synchronization. | 32 // VoiceEngine used for audio streams and audio/video synchronization. |
32 // AudioState will tickle the VoE refcount to keep it alive for as long as | 33 // AudioState will tickle the VoE refcount to keep it alive for as long as |
33 // the AudioState itself. | 34 // the AudioState itself. |
34 VoiceEngine* voice_engine = nullptr; | 35 VoiceEngine* voice_engine = nullptr; |
35 | 36 |
36 // The audio mixer connected to active receive streams. One per | 37 // The audio mixer connected to active receive streams. One per |
37 // AudioState. | 38 // AudioState. |
38 rtc::scoped_refptr<AudioMixer> audio_mixer; | 39 rtc::scoped_refptr<AudioMixer> audio_mixer; |
| 40 |
| 41 // The audio processing module. |
| 42 rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing; |
39 }; | 43 }; |
40 | 44 |
| 45 virtual AudioProcessing* audio_processing() = 0; |
| 46 |
41 // TODO(solenberg): Replace scoped_refptr with shared_ptr once we can use it. | 47 // TODO(solenberg): Replace scoped_refptr with shared_ptr once we can use it. |
42 static rtc::scoped_refptr<AudioState> Create( | 48 static rtc::scoped_refptr<AudioState> Create( |
43 const AudioState::Config& config); | 49 const AudioState::Config& config); |
44 | 50 |
45 virtual ~AudioState() {} | 51 virtual ~AudioState() {} |
46 }; | 52 }; |
47 } // namespace webrtc | 53 } // namespace webrtc |
48 | 54 |
49 #endif // WEBRTC_CALL_AUDIO_STATE_H_ | 55 #endif // WEBRTC_CALL_AUDIO_STATE_H_ |
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