MediaStream audio object graph untangling and clean-ups.

content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc

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
index 20a3969d11312d81c0b99354bc08c3f476666884..064cfa2eceff90ef9848cacc4f54d78371f40559 100644
--- a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc
+++ b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc
@@ -46,8 +46,7 @@ WebRtcLocalAudioTrackAdapter::WebRtcLocalAudioTrackAdapter(
     : webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>(label),
       owner_(NULL),
       track_source_(track_source),
-      signaling_thread_(signaling_thread),
-      signal_level_(0) {
+      signaling_thread_(signaling_thread) {
   signaling_thread_checker_.DetachFromThread();

[mcasas, 2016/02/26 01:28:19]

DCHECK(signaling_thread_);
(this should not happen but probably some
or another unit tests will roll with it by 
skipping some functionality, in that case
probs the Create() method in l.34 should
add an
#if !defined(NDEBUG)
  if(!signalling_thread.get())
    signaling_thread = base::ThreadTaskRunnerHandle::Get();
#endif
WDYT?

[miu, 2016/02/27 03:46:36]

I actually have some threading fixes to this class that I'll be introducing in
my next CL.  ;-)  Basically, the AddSink() and RemoveSink() methods are
currently called on the WebRTC signaling thread, but all other MediaStream code
adds and removes sinks/tracks/whatever on the main thread.  So, my next CL will
make things more consistent.

   capture_thread_.DetachFromThread();
 }
@@ -72,6 +71,12 @@ void WebRtcLocalAudioTrackAdapter::SetAudioProcessor(
   audio_processor_ = processor;
 }
 
+void WebRtcLocalAudioTrackAdapter::SetLevel(
+    scoped_refptr<MediaStreamAudioLevelCalculator::Level> level) {

[mcasas, 2016/02/26 01:28:19]

Paranoia nit:

DCHECK(capture_thread_.CalledOnValidThread() ||
       signaling_thread_checker_.CalledOnValidThread());

?
(Here and in l.134) ?

[miu, 2016/02/27 03:46:36]

Actually, it's not called from either of those threads; it's called from the
main thread.  It really does not matter what thread calls this since it is just
set once before the audio flow begins and then left alone forever after.

+  base::AutoLock auto_lock(lock_);
+  level_ = level;
+}
+
 std::string WebRtcLocalAudioTrackAdapter::kind() const {
   return kAudioTrackKind;
 }
@@ -126,10 +131,19 @@ void WebRtcLocalAudioTrackAdapter::RemoveSink(
 }
 
 bool WebRtcLocalAudioTrackAdapter::GetSignalLevel(int* level) {
-  DCHECK(signaling_thread_checker_.CalledOnValidThread());
-
-  base::AutoLock auto_lock(lock_);
-  *level = signal_level_;
+  // Note: Called on the signaling thread by WebRtc.
+  float signal_level = 0.0f;
+  {
+    base::AutoLock auto_lock(lock_);
+    if (!level_)
+      return false;
+    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;
 }
 
@@ -140,12 +154,6 @@ WebRtcLocalAudioTrackAdapter::GetAudioProcessor() {
   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());
   return track_source_;