MediaStream audio: Refactor 3 separate "glue" implementations into one.

content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc

-// 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