dd31071e19
All audio in calls is now routed through AudioTransportProxy. The AudioTransport implemented by VoEBaseImpl is disconnected from AudioDevice and replaced by an empty proxy layer that forwards calls to the old Transport. This is a refactoring CL in preparation for landing https://codereview.webrtc.org/2436033002/, which will connect the new AudioMixer. In the planned configuration, the currently empty AudioTransportProxy will query the new mixer for audio instead of polling data from the old Transport. Mixed audio will be passed to an AudioProcessing interface. AudioTransportProxy is initialized with an AudioProcessing*, which is currently unused. No presubmit since we implement an interface with non-const references. NOPRESUBMIT=True BUG=webrtc:6346 Review-Url: https://codereview.webrtc.org/2454373002 Cr-Commit-Position: refs/heads/master@{#15133}
69 lines
2.2 KiB
C++
69 lines
2.2 KiB
C++
/*
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* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_AUDIO_AUDIO_STATE_H_
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#define WEBRTC_AUDIO_AUDIO_STATE_H_
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#include "webrtc/api/call/audio_state.h"
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#include "webrtc/audio/audio_transport_proxy.h"
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#include "webrtc/audio/scoped_voe_interface.h"
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/base/criticalsection.h"
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#include "webrtc/base/thread_checker.h"
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#include "webrtc/voice_engine/include/voe_base.h"
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namespace webrtc {
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namespace internal {
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class AudioState final : public webrtc::AudioState,
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public webrtc::VoiceEngineObserver {
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public:
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explicit AudioState(const AudioState::Config& config);
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~AudioState() override;
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VoiceEngine* voice_engine();
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rtc::scoped_refptr<AudioMixer> mixer();
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bool typing_noise_detected() const;
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private:
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// rtc::RefCountInterface implementation.
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int AddRef() const override;
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int Release() const override;
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// webrtc::VoiceEngineObserver implementation.
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void CallbackOnError(int channel_id, int err_code) override;
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rtc::ThreadChecker thread_checker_;
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rtc::ThreadChecker process_thread_checker_;
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const webrtc::AudioState::Config config_;
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// We hold one interface pointer to the VoE to make sure it is kept alive.
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ScopedVoEInterface<VoEBase> voe_base_;
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// The critical section isn't strictly needed in this case, but xSAN bots may
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// trigger on unprotected cross-thread access.
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rtc::CriticalSection crit_sect_;
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bool typing_noise_detected_ GUARDED_BY(crit_sect_) = false;
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// Reference count; implementation copied from rtc::RefCountedObject.
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mutable volatile int ref_count_ = 0;
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// Transports mixed audio from the mixer to the audio device and
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// recorded audio to the VoE AudioTransport.
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AudioTransportProxy audio_transport_proxy_;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioState);
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};
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} // namespace internal
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} // namespace webrtc
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#endif // WEBRTC_AUDIO_AUDIO_STATE_H_
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