1 /* 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 3 * 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 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef WEBRTC_AUDIO_AUDIO_STATE_H_ 12 #define WEBRTC_AUDIO_AUDIO_STATE_H_ 13 14 #include "webrtc/audio_state.h" 15 #include "webrtc/audio/scoped_voe_interface.h" 16 #include "webrtc/base/constructormagic.h" 17 #include "webrtc/base/criticalsection.h" 18 #include "webrtc/base/thread_checker.h" 19 #include "webrtc/voice_engine/include/voe_base.h" 20 21 namespace webrtc { 22 namespace internal { 23 24 class AudioState final : public webrtc::AudioState, 25 public webrtc::VoiceEngineObserver { 26 public: 27 explicit AudioState(const AudioState::Config& config); 28 ~AudioState() override; 29 30 VoiceEngine* voice_engine(); 31 bool typing_noise_detected() const; 32 33 private: 34 // rtc::RefCountInterface implementation. 35 int AddRef() const override; 36 int Release() const override; 37 38 // webrtc::VoiceEngineObserver implementation. 39 void CallbackOnError(int channel_id, int err_code) override; 40 41 rtc::ThreadChecker thread_checker_; 42 rtc::ThreadChecker process_thread_checker_; 43 const webrtc::AudioState::Config config_; 44 45 // We hold one interface pointer to the VoE to make sure it is kept alive. 46 ScopedVoEInterface<VoEBase> voe_base_; 47 48 // The critical section isn't strictly needed in this case, but xSAN bots may 49 // trigger on unprotected cross-thread access. 50 mutable rtc::CriticalSection crit_sect_; 51 bool typing_noise_detected_ GUARDED_BY(crit_sect_) = false; 52 53 // Reference count; implementation copied from rtc::RefCountedObject. 54 mutable volatile int ref_count_ = 0; 55 56 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioState); 57 }; 58 } // namespace internal 59 } // namespace webrtc 60 61 #endif // WEBRTC_AUDIO_AUDIO_STATE_H_ 62