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d319534143
Bug: webrtc:4690 Change-Id: I3b8950fdb13835964c5bf41162731eff5048bf1a Reviewed-on: https://webrtc-review.googlesource.com/23820 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20823}
112 lines
3.7 KiB
C++
112 lines
3.7 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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#include "audio/audio_state.h"
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#include "modules/audio_device/include/audio_device.h"
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#include "rtc_base/atomicops.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/logging.h"
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#include "rtc_base/ptr_util.h"
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#include "rtc_base/thread.h"
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#include "voice_engine/transmit_mixer.h"
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namespace webrtc {
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namespace internal {
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AudioState::AudioState(const AudioState::Config& config)
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: config_(config),
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voe_base_(config.voice_engine),
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audio_transport_proxy_(voe_base_->audio_transport(),
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config_.audio_processing.get(),
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config_.audio_mixer) {
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process_thread_checker_.DetachFromThread();
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RTC_DCHECK(config_.audio_mixer);
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}
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AudioState::~AudioState() {
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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}
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VoiceEngine* AudioState::voice_engine() {
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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return config_.voice_engine;
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}
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rtc::scoped_refptr<AudioMixer> AudioState::mixer() {
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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return config_.audio_mixer;
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}
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bool AudioState::typing_noise_detected() const {
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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// TODO(solenberg): Remove const_cast once AudioState owns transmit mixer
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// functionality.
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voe::TransmitMixer* transmit_mixer =
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const_cast<AudioState*>(this)->voe_base_->transmit_mixer();
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return transmit_mixer->typing_noise_detected();
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}
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void AudioState::SetPlayout(bool enabled) {
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RTC_LOG(INFO) << "SetPlayout(" << enabled << ")";
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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const bool currently_enabled = (null_audio_poller_ == nullptr);
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if (enabled == currently_enabled) {
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return;
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}
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VoEBase* const voe = VoEBase::GetInterface(voice_engine());
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RTC_DCHECK(voe);
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if (enabled) {
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null_audio_poller_.reset();
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}
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// Will stop/start playout of the underlying device, if necessary, and
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// remember the setting for when it receives subsequent calls of
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// StartPlayout.
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voe->SetPlayout(enabled);
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if (!enabled) {
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null_audio_poller_ =
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rtc::MakeUnique<NullAudioPoller>(&audio_transport_proxy_);
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}
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voe->Release();
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}
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void AudioState::SetRecording(bool enabled) {
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RTC_LOG(INFO) << "SetRecording(" << enabled << ")";
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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// TODO(henrika): keep track of state as in SetPlayout().
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VoEBase* const voe = VoEBase::GetInterface(voice_engine());
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RTC_DCHECK(voe);
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// Will stop/start recording of the underlying device, if necessary, and
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// remember the setting for when it receives subsequent calls of
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// StartPlayout.
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voe->SetRecording(enabled);
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voe->Release();
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}
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// Reference count; implementation copied from rtc::RefCountedObject.
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void AudioState::AddRef() const {
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rtc::AtomicOps::Increment(&ref_count_);
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}
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// Reference count; implementation copied from rtc::RefCountedObject.
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rtc::RefCountReleaseStatus AudioState::Release() const {
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if (rtc::AtomicOps::Decrement(&ref_count_) == 0) {
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delete this;
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return rtc::RefCountReleaseStatus::kDroppedLastRef;
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}
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return rtc::RefCountReleaseStatus::kOtherRefsRemained;
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}
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} // namespace internal
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rtc::scoped_refptr<AudioState> AudioState::Create(
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const AudioState::Config& config) {
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return rtc::scoped_refptr<AudioState>(new internal::AudioState(config));
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}
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} // namespace webrtc
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