blob: 0e3f5f6f20f6ff6da6c9b60ca7f53028c48d6666 [file] [log] [blame]
/*
* 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 "webrtc/audio/audio_state.h"
#include "webrtc/modules/audio_device/include/audio_device.h"
#include "webrtc/rtc_base/atomicops.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/logging.h"
#include "webrtc/voice_engine/include/voe_errors.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);
// Only one AudioState should be created per VoiceEngine.
RTC_CHECK(voe_base_->RegisterVoiceEngineObserver(*this) != -1);
auto* const device = voe_base_->audio_device_module();
RTC_DCHECK(device);
// This is needed for the Chrome implementation of RegisterAudioCallback.
device->RegisterAudioCallback(nullptr);
device->RegisterAudioCallback(&audio_transport_proxy_);
}
AudioState::~AudioState() {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
voe_base_->DeRegisterVoiceEngineObserver();
}
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());
rtc::CritScope lock(&crit_sect_);
return typing_noise_detected_;
}
// Reference count; implementation copied from rtc::RefCountedObject.
int AudioState::AddRef() const {
return rtc::AtomicOps::Increment(&ref_count_);
}
// Reference count; implementation copied from rtc::RefCountedObject.
int AudioState::Release() const {
int count = rtc::AtomicOps::Decrement(&ref_count_);
if (!count) {
delete this;
}
return count;
}
void AudioState::CallbackOnError(int channel_id, int err_code) {
RTC_DCHECK(process_thread_checker_.CalledOnValidThread());
// All call sites in VoE, as of this writing, specify -1 as channel_id.
RTC_DCHECK(channel_id == -1);
LOG(LS_INFO) << "VoiceEngine error " << err_code << " reported on channel "
<< channel_id << ".";
if (err_code == VE_TYPING_NOISE_WARNING) {
rtc::CritScope lock(&crit_sect_);
typing_noise_detected_ = true;
} else if (err_code == VE_TYPING_NOISE_OFF_WARNING) {
rtc::CritScope lock(&crit_sect_);
typing_noise_detected_ = false;
}
}
} // namespace internal
rtc::scoped_refptr<AudioState> AudioState::Create(
const AudioState::Config& config) {
return rtc::scoped_refptr<AudioState>(new internal::AudioState(config));
}
} // namespace webrtc