| /* |
| * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| #ifndef WEBRTC_API_CALL_AUDIO_STATE_H_ |
| #define WEBRTC_API_CALL_AUDIO_STATE_H_ |
| |
| #include "webrtc/base/refcount.h" |
| #include "webrtc/base/scoped_ref_ptr.h" |
| |
| namespace webrtc { |
| |
| class AudioDeviceModule; |
| class VoiceEngine; |
| |
| // WORK IN PROGRESS |
| // This class is under development and is not yet intended for for use outside |
| // of WebRtc/Libjingle. Please use the VoiceEngine API instead. |
| // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690 |
| |
| // AudioState holds the state which must be shared between multiple instances of |
| // webrtc::Call for audio processing purposes. |
| class AudioState : public rtc::RefCountInterface { |
| public: |
| struct Config { |
| // VoiceEngine used for audio streams and audio/video synchronization. |
| // AudioState will tickle the VoE refcount to keep it alive for as long as |
| // the AudioState itself. |
| VoiceEngine* voice_engine = nullptr; |
| |
| // The AudioDeviceModule associated with the Calls. |
| AudioDeviceModule* audio_device_module = nullptr; |
| }; |
| |
| // TODO(solenberg): Replace scoped_refptr with shared_ptr once we can use it. |
| static rtc::scoped_refptr<AudioState> Create( |
| const AudioState::Config& config); |
| |
| virtual ~AudioState() {} |
| }; |
| } // namespace webrtc |
| |
| #endif // WEBRTC_API_CALL_AUDIO_STATE_H_ |