webrtc/audio/audio_state.h
Fredrik Solenberg 63e6072a43 Add AudioState::audio_transport() to prepare clients for moving ADM initialization out of VoiceEngine.
(See: https://webrtc-review.googlesource.com/c/src/+/23820)

Bug: webrtc:4690
Change-Id: I474a327303aa0c9b5b34c2055ae3a35e466a7d9f
Reviewed-on: https://webrtc-review.googlesource.com/24720
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20810}
2017-11-21 10:51:02 +00:00

79 lines
2.5 KiB
C++

/*
* 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 AUDIO_AUDIO_STATE_H_
#define AUDIO_AUDIO_STATE_H_
#include <memory>
#include "audio/audio_transport_proxy.h"
#include "audio/null_audio_poller.h"
#include "audio/scoped_voe_interface.h"
#include "call/audio_state.h"
#include "rtc_base/constructormagic.h"
#include "rtc_base/criticalsection.h"
#include "rtc_base/refcount.h"
#include "rtc_base/thread_checker.h"
#include "voice_engine/include/voe_base.h"
namespace webrtc {
namespace internal {
class AudioState final : public webrtc::AudioState {
public:
explicit AudioState(const AudioState::Config& config);
~AudioState() override;
AudioProcessing* audio_processing() override {
RTC_DCHECK(config_.audio_processing);
return config_.audio_processing.get();
}
AudioTransport* audio_transport() override {
return &audio_transport_proxy_;
}
void SetPlayout(bool enabled) override;
void SetRecording(bool enabled) override;
VoiceEngine* voice_engine();
rtc::scoped_refptr<AudioMixer> mixer();
bool typing_noise_detected() const;
private:
// rtc::RefCountInterface implementation.
void AddRef() const override;
rtc::RefCountReleaseStatus Release() const 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_;
// Reference count; implementation copied from rtc::RefCountedObject.
// TODO(nisse): Use RefCountedObject or RefCountedBase instead.
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_;
// Null audio poller is used to continue polling the audio streams if audio
// playout is disabled so that audio processing still happens and the audio
// stats are still updated.
std::unique_ptr<NullAudioPoller> null_audio_poller_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioState);
};
} // namespace internal
} // namespace webrtc
#endif // AUDIO_AUDIO_STATE_H_