MediaStream audio object graph untangling and clean-ups.

content/renderer/media/webrtc_local_audio_track.cc

Index: content/renderer/media/webrtc_local_audio_track.cc
diff --git a/content/renderer/media/webrtc_local_audio_track.cc b/content/renderer/media/webrtc_local_audio_track.cc
index cb48668eaeccc09a670c0a150b5d439e8d515778..cb62cf05efc02d6746a120c6b33597e86435f8ab 100644
--- a/content/renderer/media/webrtc_local_audio_track.cc
+++ b/content/renderer/media/webrtc_local_audio_track.cc
@@ -9,82 +9,52 @@
 #include <limits>
 
 #include "content/public/renderer/media_stream_audio_sink.h"
-#include "content/renderer/media/media_stream_audio_level_calculator.h"
 #include "content/renderer/media/media_stream_audio_processor.h"
 #include "content/renderer/media/media_stream_audio_sink_owner.h"
 #include "content/renderer/media/media_stream_audio_track_sink.h"
-#include "content/renderer/media/webaudio_capturer_source.h"
 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
-#include "content/renderer/media/webrtc_audio_capturer.h"
 
 namespace content {
 
 WebRtcLocalAudioTrack::WebRtcLocalAudioTrack(
-    WebRtcLocalAudioTrackAdapter* adapter,
-    const scoped_refptr<WebRtcAudioCapturer>& capturer,
-    WebAudioCapturerSource* webaudio_source)
-    : MediaStreamAudioTrack(true),
-      adapter_(adapter),
-      capturer_(capturer),
-      webaudio_source_(webaudio_source) {
-  DCHECK(capturer.get() || webaudio_source);
+    scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter)
+    : MediaStreamAudioTrack(true), adapter_(adapter) {

[tommi (sloooow) - chröme, 2016/03/03 11:07:31]

use std::move()? (assuming |adapter| is passed by value on purpose.

[miu, 2016/03/05 02:55:31]

Done.

   signal_thread_checker_.DetachFromThread();
+  DVLOG(1) << "WebRtcLocalAudioTrack::WebRtcLocalAudioTrack()";
 
   adapter_->Initialize(this);
-
-  DVLOG(1) << "WebRtcLocalAudioTrack::WebRtcLocalAudioTrack()";
 }
 
 WebRtcLocalAudioTrack::~WebRtcLocalAudioTrack() {
   DCHECK(main_render_thread_checker_.CalledOnValidThread());
   DVLOG(1) << "WebRtcLocalAudioTrack::~WebRtcLocalAudioTrack()";
-  // Users might not call Stop() on the track.
-  Stop();
+  // Ensure the track is stopped.
+  MediaStreamAudioTrack::Stop();
 }
 
 media::AudioParameters WebRtcLocalAudioTrack::GetOutputFormat() const {
   DCHECK(main_render_thread_checker_.CalledOnValidThread());
-  if (webaudio_source_.get()) {
-    return media::AudioParameters();
-  } else {
-    return capturer_->GetOutputFormat();
-  }
+  base::AutoLock auto_lock(lock_);
+  return audio_parameters_;
 }
 
 void WebRtcLocalAudioTrack::Capture(const media::AudioBus& audio_bus,
-                                    base::TimeTicks estimated_capture_time,
-                                    bool force_report_nonzero_energy) {
+                                    base::TimeTicks estimated_capture_time) {
   DCHECK(capture_thread_checker_.CalledOnValidThread());
   DCHECK(!estimated_capture_time.is_null());
 
-  // Calculate the signal level regardless of whether the track is disabled or
-  // enabled.  If |force_report_nonzero_energy| is true, |audio_bus| contains
-  // post-processed data that may be all zeros even though the signal contained
-  // energy before the processing.  In this case, report nonzero energy even if
-  // the energy of the data in |audio_bus| is zero.
-  const float minimum_signal_level =
-      force_report_nonzero_energy ? 1.0f / std::numeric_limits<int16_t>::max()
-                                  : 0.0f;
-  const float signal_level = std::max(
-      minimum_signal_level,
-      std::min(1.0f, level_calculator_->Calculate(audio_bus)));
-  const int signal_level_as_pcm16 =
-      static_cast<int>(signal_level * std::numeric_limits<int16_t>::max() +
-                       0.5f /* rounding to nearest int */);
-  adapter_->SetSignalLevel(signal_level_as_pcm16);
-
-  scoped_refptr<WebRtcAudioCapturer> capturer;
   SinkList::ItemList sinks;
   SinkList::ItemList sinks_to_notify_format;
   {
     base::AutoLock auto_lock(lock_);
-    capturer = capturer_;
     sinks = sinks_.Items();
     sinks_.RetrieveAndClearTags(&sinks_to_notify_format);
   }
 
   // Notify the tracks on when the format changes. This will do nothing if
-  // |sinks_to_notify_format| is empty.
+  // |sinks_to_notify_format| is empty. Note that accessing |audio_parameters_|
+  // without holding the |lock_| is valid since |audio_parameters_| is only
+  // changed on the current thread.
   for (const auto& sink : sinks_to_notify_format)
     sink->OnSetFormat(audio_parameters_);
 
@@ -105,24 +75,12 @@ void WebRtcLocalAudioTrack::OnSetFormat(
   capture_thread_checker_.DetachFromThread();
   DCHECK(capture_thread_checker_.CalledOnValidThread());
 
-  audio_parameters_ = params;
-  level_calculator_.reset(new MediaStreamAudioLevelCalculator());
-
   base::AutoLock auto_lock(lock_);
+  audio_parameters_ = params;
   // Remember to notify all sinks of the new format.
   sinks_.TagAll();
 }
 
-void WebRtcLocalAudioTrack::SetAudioProcessor(
-    const scoped_refptr<MediaStreamAudioProcessor>& processor) {
-  // if the |processor| does not have audio processing, which can happen if
-  // kDisableAudioTrackProcessing is set set or all the constraints in
-  // the |processor| are turned off. In such case, we pass NULL to the
-  // adapter to indicate that no stats can be gotten from the processor.
-  adapter_->SetAudioProcessor(processor->has_audio_processing() ?
-      processor : NULL);
-}
-
 void WebRtcLocalAudioTrack::AddSink(MediaStreamAudioSink* sink) {
   // This method is called from webrtc, on the signaling thread, when the local
   // description is set and from the main thread from WebMediaPlayerMS::load
@@ -166,63 +124,22 @@ void WebRtcLocalAudioTrack::RemoveSink(MediaStreamAudioSink* sink) {
     removed_item->Reset();
 }
 
-void WebRtcLocalAudioTrack::Start() {
-  DCHECK(main_render_thread_checker_.CalledOnValidThread());
-  DVLOG(1) << "WebRtcLocalAudioTrack::Start()";
-  if (webaudio_source_.get()) {
-    // If the track is hooking up with WebAudio, do NOT add the track to the
-    // capturer as its sink otherwise two streams in different clock will be
-    // pushed through the same track.
-    webaudio_source_->Start(this);
-  } else if (capturer_.get()) {
-    capturer_->AddTrack(this);
-  }
-
-  SinkList::ItemList sinks;
-  {
-    base::AutoLock auto_lock(lock_);
-    sinks = sinks_.Items();
-  }
-  for (SinkList::ItemList::const_iterator it = sinks.begin();
-       it != sinks.end();
-       ++it) {
-    (*it)->OnReadyStateChanged(blink::WebMediaStreamSource::ReadyStateLive);
-  }
-}
-
 void WebRtcLocalAudioTrack::SetEnabled(bool enabled) {
   DCHECK(main_render_thread_checker_.CalledOnValidThread());
   if (adapter_.get())
     adapter_->set_enabled(enabled);
 }
 
-void WebRtcLocalAudioTrack::Stop() {
+void WebRtcLocalAudioTrack::OnStop() {
   DCHECK(main_render_thread_checker_.CalledOnValidThread());
-  DVLOG(1) << "WebRtcLocalAudioTrack::Stop()";
-  if (!capturer_.get() && !webaudio_source_.get())
-    return;
-
-  if (webaudio_source_.get()) {
-    // Called Stop() on the |webaudio_source_| explicitly so that
-    // |webaudio_source_| won't push more data to the track anymore.
-    // Also note that the track is not registered as a sink to the |capturer_|
-    // in such case and no need to call RemoveTrack().
-    webaudio_source_->Stop();
-  } else {
-    // It is necessary to call RemoveTrack on the |capturer_| to avoid getting
-    // audio callback after Stop().
-    capturer_->RemoveTrack(this);
-  }
+  DVLOG(1) << "WebRtcLocalAudioTrack::OnStop()";
 
-  // Protect the pointers using the lock when accessing |sinks_| and
-  // setting the |capturer_| to NULL.
+  // Protect the pointers using the lock when accessing |sinks_|.
   SinkList::ItemList sinks;
   {
     base::AutoLock auto_lock(lock_);
     sinks = sinks_.Items();
     sinks_.Clear();
-    webaudio_source_ = NULL;
-    capturer_ = NULL;
   }
 
   for (SinkList::ItemList::const_iterator it = sinks.begin();