Make AudioOutputDevice restartable and reinitializable

content/renderer/media/webrtc_local_audio_track.cc

 // Copyright (c) 2012 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_local_audio_track.h"
 
 #include <stdint.h>
 
 #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"
 
+#include "base/debug/stack_trace.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) {
@@ skipping 17 matching lines @@
   if (webaudio_source_.get()) {
     return media::AudioParameters();
   } else {
     return capturer_->GetOutputFormat();
   }
 }
 
 void WebRtcLocalAudioTrack::Capture(const media::AudioBus& audio_bus,
                                     base::TimeTicks estimated_capture_time,
                                     bool force_report_nonzero_energy) {
+/*
+  static bool first_time = true;
+  if (first_time) {
+    first_time = false;
+    base::debug::StackTrace st;
+    st.Print();
+  }
+*/
   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()
@@ skipping 25 matching lines @@
   // TODO(jiayl): we should not pass the real audio data down if the track is
   // disabled. This is currently done so to feed input to WebRTC typing
   // detection and should be changed when audio processing is moved from
   // WebRTC to the track.
   for (const auto& sink : sinks)
     sink->OnData(audio_bus, estimated_capture_time);
 }
 
 void WebRtcLocalAudioTrack::OnSetFormat(
     const media::AudioParameters& params) {
+  printf("******* WebRtcLocalAudioTrack::OnSetFormat %p\n", this);
+//  static bool first_time = true;
+//  if (first_time) {
+//    first_time = false;
+//    base::debug::StackTrace st;
+//    st.Print();
+//  }
+
   DVLOG(1) << "WebRtcLocalAudioTrack::OnSetFormat()";
   // If the source is restarted, we might have changed to another capture
   // thread.
   capture_thread_checker_.DetachFromThread();
   DCHECK(capture_thread_checker_.CalledOnValidThread());
 
   audio_parameters_ = params;
   level_calculator_.reset(new MediaStreamAudioLevelCalculator());
 
   base::AutoLock auto_lock(lock_);
@@ skipping 120 matching lines @@
     (*it)->Reset();
   }
 }
 
 webrtc::AudioTrackInterface* WebRtcLocalAudioTrack::GetAudioAdapter() {
   DCHECK(main_render_thread_checker_.CalledOnValidThread());
   return adapter_.get();
 }
 
 }  // namespace content