MediaStream audio object graph untangling and clean-ups.

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_thread;
+  scoped_refptr<base::SingleThreadTaskRunner> signaling_task_runner;
   RenderThreadImpl* current = RenderThreadImpl::current();
   if (current) {
     PeerConnectionDependencyFactory* pc_factory =
         current->GetPeerConnectionDependencyFactory();
-    signaling_thread = pc_factory->GetWebRtcSignalingThread();
+    signaling_task_runner = pc_factory->GetWebRtcSignalingThread();
+    CHECK(signaling_task_runner);
+  } else {
+    LOG(WARNING) << "Assuming single-threaded operation for unit test.";
   }
 
-  LOG_IF(ERROR, !signaling_thread.get()) << "No signaling thread!";
-
   rtc::RefCountedObject<WebRtcLocalAudioTrackAdapter>* adapter =
       new rtc::RefCountedObject<WebRtcLocalAudioTrackAdapter>(
-          label, track_source, signaling_thread);
+          label, track_source, signaling_task_runner);
   return adapter;
 }
 
 WebRtcLocalAudioTrackAdapter::WebRtcLocalAudioTrackAdapter(
     const std::string& label,
     webrtc::AudioSourceInterface* track_source,
-    const scoped_refptr<base::SingleThreadTaskRunner>& signaling_thread)
+    const scoped_refptr<base::SingleThreadTaskRunner>& signaling_task_runner)
     : webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>(label),
       owner_(NULL),
       track_source_(track_source),
-      signaling_thread_(signaling_thread),
-      signal_level_(0) {
-  signaling_thread_checker_.DetachFromThread();
-  capture_thread_.DetachFromThread();
-}
+      signaling_task_runner_(signaling_task_runner) {}
 
 WebRtcLocalAudioTrackAdapter::~WebRtcLocalAudioTrackAdapter() {
 }
 
 void WebRtcLocalAudioTrackAdapter::Initialize(WebRtcLocalAudioTrack* owner) {
   DCHECK(!owner_);
   DCHECK(owner);
   owner_ = owner;
 }
 
 void WebRtcLocalAudioTrackAdapter::SetAudioProcessor(
     const scoped_refptr<MediaStreamAudioProcessor>& processor) {
-  // SetAudioProcessor will be called when a new capture thread has been
-  // initialized, so we need to detach from any current capture thread we're
-  // checking and attach to the current one.
-  capture_thread_.DetachFromThread();
-  DCHECK(capture_thread_.CalledOnValidThread());
-  base::AutoLock auto_lock(lock_);
+  // |audio_processor_| must only be set once.  This eliminates the need to
+  // protect |audio_processor_| with a mutex.
+  DCHECK(processor.get());
+  DCHECK(!audio_processor_);
   audio_processor_ = processor;
 }
 
+void WebRtcLocalAudioTrackAdapter::SetLevel(
+    scoped_refptr<MediaStreamAudioLevelCalculator::Level> level) {
+  // |level_| must only be set once.  This eliminates the need to protect
+  // |level_| with a mutex.
+  DCHECK(level.get());
+  DCHECK(!level_);
+  level_ = 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_thread_.get() && !signaling_thread_->BelongsToCurrentThread()) {
-    signaling_thread_->PostTask(FROM_HERE,
+  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_thread_checker_.CalledOnValidThread());
+  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
 
   scoped_ptr<WebRtcAudioSinkAdapter> adapter(
       new WebRtcAudioSinkAdapter(sink));
   owner_->AddSink(adapter.get());
   sink_adapters_.push_back(adapter.release());
 }
 
 void WebRtcLocalAudioTrackAdapter::RemoveSink(
     webrtc::AudioTrackSinkInterface* sink) {
-  DCHECK(signaling_thread_checker_.CalledOnValidThread());
+  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_thread_checker_.CalledOnValidThread());
+  DCHECK(!signaling_task_runner_ ||
+         signaling_task_runner_->RunsTasksOnCurrentThread());
 
-  base::AutoLock auto_lock(lock_);
-  *level = signal_level_;
+  // |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_thread_checker_.CalledOnValidThread());
-  base::AutoLock auto_lock(lock_);
+  DCHECK(!signaling_task_runner_ ||
+         signaling_task_runner_->RunsTasksOnCurrentThread());
   return audio_processor_.get();
 }
 
-void WebRtcLocalAudioTrackAdapter::SetSignalLevel(int signal_level) {
-  DCHECK(capture_thread_.CalledOnValidThread());
-  base::AutoLock auto_lock(lock_);
-  signal_level_ = signal_level;
-}
-
 webrtc::AudioSourceInterface* WebRtcLocalAudioTrackAdapter::GetSource() const {
-  DCHECK(signaling_thread_checker_.CalledOnValidThread());
+  DCHECK(!signaling_task_runner_ ||
+         signaling_task_runner_->RunsTasksOnCurrentThread());
   return track_source_;
 }
 
 }  // namespace content