| /* |
| * 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_AUDIO_AUDIO_STATE_H_ |
| #define WEBRTC_AUDIO_AUDIO_STATE_H_ |
| |
| #include "webrtc/api/call/audio_state.h" |
| #include "webrtc/audio/scoped_voe_interface.h" |
| #include "webrtc/base/constructormagic.h" |
| #include "webrtc/base/criticalsection.h" |
| #include "webrtc/base/thread_checker.h" |
| #include "webrtc/voice_engine/include/voe_base.h" |
| |
| namespace webrtc { |
| namespace internal { |
| |
| class AudioState final : public webrtc::AudioState, |
| public webrtc::VoiceEngineObserver { |
| public: |
| explicit AudioState(const AudioState::Config& config); |
| ~AudioState() override; |
| |
| VoiceEngine* voice_engine(); |
| bool typing_noise_detected() const; |
| |
| private: |
| // rtc::RefCountInterface implementation. |
| int AddRef() const override; |
| int Release() const override; |
| |
| // webrtc::VoiceEngineObserver implementation. |
| void CallbackOnError(int channel_id, int err_code) override; |
| |
| rtc::ThreadChecker thread_checker_; |
| rtc::ThreadChecker process_thread_checker_; |
| const webrtc::AudioState::Config config_; |
| |
| // We hold one interface pointer to the VoE to make sure it is kept alive. |
| ScopedVoEInterface<VoEBase> voe_base_; |
| |
| // The critical section isn't strictly needed in this case, but xSAN bots may |
| // trigger on unprotected cross-thread access. |
| rtc::CriticalSection crit_sect_; |
| bool typing_noise_detected_ GUARDED_BY(crit_sect_) = false; |
| |
| // Reference count; implementation copied from rtc::RefCountedObject. |
| mutable volatile int ref_count_ = 0; |
| |
| RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioState); |
| }; |
| } // namespace internal |
| } // namespace webrtc |
| |
| #endif // WEBRTC_AUDIO_AUDIO_STATE_H_ |