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(this CL is based on the work by Taylor and Steve in https://webrtc-review.googlesource.com/c/src/+/10201) This SetAudioPlayout method lets applications disable audio playout while still processing incoming audio data and generating statistics on the received audio. This may be useful if the application wants to set up media flows as soon as possible, but isn't ready to play audio yet. Currently, native applications don't have any API point to control this, unless they completely implement their own AudioDeviceModule. The SetAudioRecording works in a similar fashion but for the recorded audio. One difference is that calling SetAudioRecording(false) does not keep any audio processing alive. TBR=solenberg Bug: webrtc:7313 Change-Id: I0aa075f6bfef9818f1080f85a8ff7842fb0750aa Reviewed-on: https://webrtc-review.googlesource.com/16180 Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20499}
76 lines
2.4 KiB
C++
76 lines
2.4 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 AUDIO_AUDIO_STATE_H_
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#define AUDIO_AUDIO_STATE_H_
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#include <memory>
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#include "audio/audio_transport_proxy.h"
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#include "audio/null_audio_poller.h"
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#include "audio/scoped_voe_interface.h"
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#include "call/audio_state.h"
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#include "rtc_base/constructormagic.h"
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#include "rtc_base/criticalsection.h"
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#include "rtc_base/refcount.h"
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#include "rtc_base/thread_checker.h"
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#include "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:
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explicit AudioState(const AudioState::Config& config);
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~AudioState() override;
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AudioProcessing* audio_processing() override {
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RTC_DCHECK(config_.audio_processing);
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return config_.audio_processing.get();
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}
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void SetPlayout(bool enabled) override;
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void SetRecording(bool enabled) 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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void AddRef() const override;
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rtc::RefCountReleaseStatus Release() const 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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// Reference count; implementation copied from rtc::RefCountedObject.
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// TODO(nisse): Use RefCountedObject or RefCountedBase instead.
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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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// Null audio poller is used to continue polling the audio streams if audio
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// playout is disabled so that audio processing still happens and the audio
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// stats are still updated.
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std::unique_ptr<NullAudioPoller> null_audio_poller_;
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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 // AUDIO_AUDIO_STATE_H_
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