Add chromium support for MediaStreamAudioDestinationNode - part II

content/renderer/media/webrtc_audio_capturer.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_audio_capturer.h"
 
 #include "base/bind.h"
 #include "base/logging.h"
 #include "base/metrics/histogram.h"
 #include "base/string_util.h"
@@ skipping 95 matching lines @@
   {
     base::AutoLock auto_lock(lock_);
     if (source_ == source)
       return;
 
     source_.swap(old_source);
     source_ = source;
   }
 
   // Detach the old source from normal recording.
-  if (old_source)
+  if (old_source) {
     old_source->Stop();
 
+    // TODO(henrika, crogers): I have hard coded parameters given knowledge
+    // of a file source (44.1kHz, mono) here and use these parameters to reset
+    // our member parameters. I then dispatch the new parameters both to the
+    // sink(s) and to the new source. The idea is to get rid of any dependency
+    // of the microphone parameters which are uses as base otherwise.
+
+    // I guess we could add this info to AudioCapturerSource to enable a
+    // query here. E.g.:
+
+    // source->ModifyAudioParameters(&params_);
+
+    params_.Reset(params_.format(),
+                  media::CHANNEL_LAYOUT_MONO,  // <-- get this from WebKit?
+                  44100,                       // <-- get this from WebKit?
+                  16,                          // <-- get this from WebKit?
+                  440); // requires knowledge about WebRTC @ 10ms
+
+    buffer_.reset(new int16[params_.frames_per_buffer() * params_.channels()]);
+
+    for (SinkList::const_iterator it = sinks_.begin();
+         it != sinks_.end(); ++it) {
+      (*it)->SetCaptureFormat(params_);
+    }
+  }
+
   if (source)
     source->Initialize(params_, this, this);
 }
 
 void WebRtcAudioCapturer::SetStopCallback(
     const base::Closure& on_device_stopped_cb) {
   DVLOG(1) <<  "WebRtcAudioCapturer::SetStopCallback()";
   base::AutoLock auto_lock(lock_);
   on_device_stopped_cb_ = on_device_stopped_cb;
 }
@@ skipping 241 matching lines @@
   // Inform the local renderer about the stopped device.
   // The renderer can then save resources by not asking for more data from
   // the stopped source. We are on the IO thread but the callback task will
   // be posted on the message loop of the main render thread thanks to
   // usage of BindToLoop() when the callback was initialized.
   if (!on_device_stopped_cb_.is_null())
     on_device_stopped_cb_.Run();
 }
 
 }  // namespace content