blob: c8ce44b8b2ed18c680547effce90ae8f4e1ea44e [file] [log] [blame]
* 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.
#include "webrtc/audio/audio_transport_proxy.h"
#include "webrtc/audio/scoped_voe_interface.h"
#include "webrtc/call/audio_state.h"
#include "webrtc/rtc_base/constructormagic.h"
#include "webrtc/rtc_base/criticalsection.h"
#include "webrtc/rtc_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 {
explicit AudioState(const AudioState::Config& config);
~AudioState() override;
AudioProcessing* audio_processing() override {
return config_.audio_processing.get();
VoiceEngine* voice_engine();
rtc::scoped_refptr<AudioMixer> mixer();
bool typing_noise_detected() const;
// 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;
// Transports mixed audio from the mixer to the audio device and
// recorded audio to the VoE AudioTransport.
AudioTransportProxy audio_transport_proxy_;
} // namespace internal
} // namespace webrtc