10111bc495
This is a refactoring change in preparation for enabling AudioMixer with the goal to have a small CL as possible for passing audio through the new audio mixer in WebRTC. The dependent CL https://codereview.webrtc.org/2436033002/ enables the mixer. An object of class AudioState is shared across different webrtc audio connections. It is created in tests and in WebRTCVoiceEngine. AudioState is constructed by passing a Config struct, where one argument is scoped_refptr<AudioMixer>. Populating this field has now been mandatory. Tests and WebRTCVoiceEngine create and pass either a AudioMixerImpl. WebRTCVoiceEngine passes a real AudioMixer, which is currently unused. An alternative would have tests pass a mocked audio mixer. We chose not to do that, because we believe that tests should use the real thing unless there are reasons against it. Construction time is not an issue, because the real mixer is relatively lightweight. We couldn't find a way to test any mixer-related changes in AudioState before the mixes is connected. The next dependent CL https://codereview.webrtc.org/2436033002/ contains unit tests for mixer usage. BUG=webrtc:6346 Review-Url: https://codereview.webrtc.org/2469743002 Cr-Commit-Position: refs/heads/master@{#15134}
99 lines
3.2 KiB
C++
99 lines
3.2 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 "webrtc/audio/audio_state.h"
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#include "webrtc/base/atomicops.h"
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#include "webrtc/base/checks.h"
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#include "webrtc/base/logging.h"
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#include "webrtc/modules/audio_device/include/audio_device.h"
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#include "webrtc/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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voe_base_->audio_processing(),
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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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