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/*
* 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 "audio/audio_state.h"
#include "modules/audio_device/include/audio_device.h"
#include "rtc_base/atomicops.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/ptr_util.h"
#include "rtc_base/thread.h"
#include "voice_engine/transmit_mixer.h"
namespace webrtc {
namespace internal {
AudioState::AudioState(const AudioState::Config& config)
: config_(config),
voe_base_(config.voice_engine),
audio_transport_proxy_(voe_base_->audio_transport(),
config_.audio_processing.get(),
config_.audio_mixer) {
process_thread_checker_.DetachFromThread();
RTC_DCHECK(config_.audio_mixer);
}
AudioState::~AudioState() {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
}
VoiceEngine* AudioState::voice_engine() {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
return config_.voice_engine;
}
rtc::scoped_refptr<AudioMixer> AudioState::mixer() {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
return config_.audio_mixer;
}
bool AudioState::typing_noise_detected() const {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
// TODO(solenberg): Remove const_cast once AudioState owns transmit mixer
// functionality.
voe::TransmitMixer* transmit_mixer =
const_cast<AudioState*>(this)->voe_base_->transmit_mixer();
return transmit_mixer->typing_noise_detected();
}
void AudioState::SetPlayout(bool enabled) {
RTC_LOG(INFO) << "SetPlayout(" << enabled << ")";
RTC_DCHECK(thread_checker_.CalledOnValidThread());
const bool currently_enabled = (null_audio_poller_ == nullptr);
if (enabled == currently_enabled) {
return;
}
VoEBase* const voe = VoEBase::GetInterface(voice_engine());
RTC_DCHECK(voe);
if (enabled) {
null_audio_poller_.reset();
}
// Will stop/start playout of the underlying device, if necessary, and
// remember the setting for when it receives subsequent calls of
// StartPlayout.
voe->SetPlayout(enabled);
if (!enabled) {
null_audio_poller_ =
rtc::MakeUnique<NullAudioPoller>(&audio_transport_proxy_);
}
voe->Release();
}
void AudioState::SetRecording(bool enabled) {
RTC_LOG(INFO) << "SetRecording(" << enabled << ")";
RTC_DCHECK(thread_checker_.CalledOnValidThread());
// TODO(henrika): keep track of state as in SetPlayout().
VoEBase* const voe = VoEBase::GetInterface(voice_engine());
RTC_DCHECK(voe);
// Will stop/start recording of the underlying device, if necessary, and
// remember the setting for when it receives subsequent calls of
// StartPlayout.
voe->SetRecording(enabled);
voe->Release();
}
// Reference count; implementation copied from rtc::RefCountedObject.
void AudioState::AddRef() const {
rtc::AtomicOps::Increment(&ref_count_);
}
// Reference count; implementation copied from rtc::RefCountedObject.
rtc::RefCountReleaseStatus AudioState::Release() const {
if (rtc::AtomicOps::Decrement(&ref_count_) == 0) {
delete this;
return rtc::RefCountReleaseStatus::kDroppedLastRef;
}
return rtc::RefCountReleaseStatus::kOtherRefsRemained;
}
} // namespace internal
rtc::scoped_refptr<AudioState> AudioState::Create(
const AudioState::Config& config) {
return rtc::scoped_refptr<AudioState>(new internal::AudioState(config));
}
} // namespace webrtc
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