| Index: content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc
|
| diff --git a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc
|
| new file mode 100644
|
| index 0000000000000000000000000000000000000000..c86881b07a90eed64bca82048846e137536a2b21
|
| --- /dev/null
|
| +++ b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc
|
| @@ -0,0 +1,161 @@
|
| +// Copyright 2014 The Chromium Authors. All rights reserved.
|
| +// Use of this source code is governed by a BSD-style license that can be
|
| +// found in the LICENSE file.
|
| +
|
| +#include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
|
| +
|
| +#include "base/location.h"
|
| +#include "base/logging.h"
|
| +#include "content/renderer/media/media_stream_audio_processor.h"
|
| +#include "content/renderer/media/webrtc/peer_connection_dependency_factory.h"
|
| +#include "content/renderer/media/webrtc/webrtc_audio_sink_adapter.h"
|
| +#include "content/renderer/media/webrtc_local_audio_track.h"
|
| +#include "content/renderer/render_thread_impl.h"
|
| +#include "third_party/webrtc/api/mediastreaminterface.h"
|
| +
|
| +namespace content {
|
| +
|
| +static const char kAudioTrackKind[] = "audio";
|
| +
|
| +scoped_refptr<WebRtcLocalAudioTrackAdapter>
|
| +WebRtcLocalAudioTrackAdapter::Create(
|
| + const std::string& label,
|
| + webrtc::AudioSourceInterface* track_source) {
|
| + // TODO(tommi): Change this so that the signaling thread is one of the
|
| + // parameters to this method.
|
| + scoped_refptr<base::SingleThreadTaskRunner> signaling_task_runner;
|
| + RenderThreadImpl* current = RenderThreadImpl::current();
|
| + if (current) {
|
| + PeerConnectionDependencyFactory* pc_factory =
|
| + current->GetPeerConnectionDependencyFactory();
|
| + signaling_task_runner = pc_factory->GetWebRtcSignalingThread();
|
| + LOG_IF(ERROR, !signaling_task_runner) << "No signaling thread!";
|
| + } else {
|
| + LOG(WARNING) << "Assuming single-threaded operation for unit test.";
|
| + }
|
| +
|
| + rtc::RefCountedObject<WebRtcLocalAudioTrackAdapter>* adapter =
|
| + new rtc::RefCountedObject<WebRtcLocalAudioTrackAdapter>(
|
| + label, track_source, std::move(signaling_task_runner));
|
| + return adapter;
|
| +}
|
| +
|
| +WebRtcLocalAudioTrackAdapter::WebRtcLocalAudioTrackAdapter(
|
| + const std::string& label,
|
| + webrtc::AudioSourceInterface* track_source,
|
| + scoped_refptr<base::SingleThreadTaskRunner> signaling_task_runner)
|
| + : webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>(label),
|
| + owner_(NULL),
|
| + track_source_(track_source),
|
| + signaling_task_runner_(std::move(signaling_task_runner)) {}
|
| +
|
| +WebRtcLocalAudioTrackAdapter::~WebRtcLocalAudioTrackAdapter() {
|
| +}
|
| +
|
| +void WebRtcLocalAudioTrackAdapter::Initialize(WebRtcLocalAudioTrack* owner) {
|
| + DCHECK(!owner_);
|
| + DCHECK(owner);
|
| + owner_ = owner;
|
| +}
|
| +
|
| +void WebRtcLocalAudioTrackAdapter::SetAudioProcessor(
|
| + scoped_refptr<MediaStreamAudioProcessor> processor) {
|
| + DCHECK(processor.get());
|
| + DCHECK(!audio_processor_);
|
| + audio_processor_ = std::move(processor);
|
| +}
|
| +
|
| +void WebRtcLocalAudioTrackAdapter::SetLevel(
|
| + scoped_refptr<MediaStreamAudioLevelCalculator::Level> level) {
|
| + DCHECK(level.get());
|
| + DCHECK(!level_);
|
| + level_ = std::move(level);
|
| +}
|
| +
|
| +std::string WebRtcLocalAudioTrackAdapter::kind() const {
|
| + return kAudioTrackKind;
|
| +}
|
| +
|
| +bool WebRtcLocalAudioTrackAdapter::set_enabled(bool enable) {
|
| + // If we're not called on the signaling thread, we need to post a task to
|
| + // change the state on the correct thread.
|
| + if (signaling_task_runner_ &&
|
| + !signaling_task_runner_->BelongsToCurrentThread()) {
|
| + signaling_task_runner_->PostTask(FROM_HERE,
|
| + base::Bind(
|
| + base::IgnoreResult(&WebRtcLocalAudioTrackAdapter::set_enabled),
|
| + this, enable));
|
| + return true;
|
| + }
|
| +
|
| + return webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>::
|
| + set_enabled(enable);
|
| +}
|
| +
|
| +void WebRtcLocalAudioTrackAdapter::AddSink(
|
| + webrtc::AudioTrackSinkInterface* sink) {
|
| + DCHECK(!signaling_task_runner_ ||
|
| + signaling_task_runner_->RunsTasksOnCurrentThread());
|
| + DCHECK(sink);
|
| +#ifndef NDEBUG
|
| + // Verify that |sink| has not been added.
|
| + for (ScopedVector<WebRtcAudioSinkAdapter>::const_iterator it =
|
| + sink_adapters_.begin();
|
| + it != sink_adapters_.end(); ++it) {
|
| + DCHECK(!(*it)->IsEqual(sink));
|
| + }
|
| +#endif
|
| +
|
| + std::unique_ptr<WebRtcAudioSinkAdapter> adapter(
|
| + new WebRtcAudioSinkAdapter(sink));
|
| + owner_->AddSink(adapter.get());
|
| + sink_adapters_.push_back(adapter.release());
|
| +}
|
| +
|
| +void WebRtcLocalAudioTrackAdapter::RemoveSink(
|
| + webrtc::AudioTrackSinkInterface* sink) {
|
| + DCHECK(!signaling_task_runner_ ||
|
| + signaling_task_runner_->RunsTasksOnCurrentThread());
|
| + DCHECK(sink);
|
| + for (ScopedVector<WebRtcAudioSinkAdapter>::iterator it =
|
| + sink_adapters_.begin();
|
| + it != sink_adapters_.end(); ++it) {
|
| + if ((*it)->IsEqual(sink)) {
|
| + owner_->RemoveSink(*it);
|
| + sink_adapters_.erase(it);
|
| + return;
|
| + }
|
| + }
|
| +}
|
| +
|
| +bool WebRtcLocalAudioTrackAdapter::GetSignalLevel(int* level) {
|
| + DCHECK(!signaling_task_runner_ ||
|
| + signaling_task_runner_->RunsTasksOnCurrentThread());
|
| +
|
| + // |level_| is only set once, so it's safe to read without first acquiring a
|
| + // mutex.
|
| + if (!level_)
|
| + return false;
|
| + const float signal_level = level_->GetCurrent();
|
| + DCHECK_GE(signal_level, 0.0f);
|
| + DCHECK_LE(signal_level, 1.0f);
|
| + // Convert from float in range [0.0,1.0] to an int in range [0,32767].
|
| + *level = static_cast<int>(signal_level * std::numeric_limits<int16_t>::max() +
|
| + 0.5f /* rounding to nearest int */);
|
| + return true;
|
| +}
|
| +
|
| +rtc::scoped_refptr<webrtc::AudioProcessorInterface>
|
| +WebRtcLocalAudioTrackAdapter::GetAudioProcessor() {
|
| + DCHECK(!signaling_task_runner_ ||
|
| + signaling_task_runner_->RunsTasksOnCurrentThread());
|
| + return audio_processor_.get();
|
| +}
|
| +
|
| +webrtc::AudioSourceInterface* WebRtcLocalAudioTrackAdapter::GetSource() const {
|
| + DCHECK(!signaling_task_runner_ ||
|
| + signaling_task_runner_->RunsTasksOnCurrentThread());
|
| + return track_source_;
|
| +}
|
| +
|
| +} // namespace content
|
|
|