92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
99 lines
3.1 KiB
C++
99 lines
3.1 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 "voice_engine/include/voe_errors.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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// Only one AudioState should be created per VoiceEngine.
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RTC_CHECK(voe_base_->RegisterVoiceEngineObserver(*this) != -1);
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auto* const device = voe_base_->audio_device_module();
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RTC_DCHECK(device);
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// This is needed for the Chrome implementation of RegisterAudioCallback.
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device->RegisterAudioCallback(nullptr);
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device->RegisterAudioCallback(&audio_transport_proxy_);
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}
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AudioState::~AudioState() {
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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voe_base_->DeRegisterVoiceEngineObserver();
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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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rtc::CritScope lock(&crit_sect_);
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return typing_noise_detected_;
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}
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// Reference count; implementation copied from rtc::RefCountedObject.
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int AudioState::AddRef() const {
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return rtc::AtomicOps::Increment(&ref_count_);
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}
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// Reference count; implementation copied from rtc::RefCountedObject.
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int AudioState::Release() const {
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int count = rtc::AtomicOps::Decrement(&ref_count_);
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if (!count) {
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delete this;
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}
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return count;
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}
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void AudioState::CallbackOnError(int channel_id, int err_code) {
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RTC_DCHECK(process_thread_checker_.CalledOnValidThread());
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// All call sites in VoE, as of this writing, specify -1 as channel_id.
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RTC_DCHECK(channel_id == -1);
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LOG(LS_INFO) << "VoiceEngine error " << err_code << " reported on channel "
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<< channel_id << ".";
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if (err_code == VE_TYPING_NOISE_WARNING) {
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rtc::CritScope lock(&crit_sect_);
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typing_noise_detected_ = true;
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} else if (err_code == VE_TYPING_NOISE_OFF_WARNING) {
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rtc::CritScope lock(&crit_sect_);
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typing_noise_detected_ = false;
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}
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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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