Possible solution to synchronization problems in webrtc audio capturer.

content/renderer/media/webrtc_audio_capturer.h

 // 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.
 
 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_
 
 #include <list>
 #include <string>
 
 #include "base/callback.h"
 #include "base/memory/ref_counted.h"
+#include "base/synchronization/condition_variable.h"
 #include "base/synchronization/lock.h"
 #include "base/threading/thread_checker.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/renderer/media/webrtc_local_audio_renderer.h"
 #include "media/audio/audio_input_device.h"
 #include "media/base/audio_capturer_source.h"
 #include "media/base/audio_fifo.h"
 
 namespace media {
 class AudioBus;
@@ skipping 135 matching lines @@
 
  protected:
   friend class base::RefCountedThreadSafe<WebRtcAudioCapturer>;
   virtual ~WebRtcAudioCapturer();
 
  private:
   typedef std::list<WebRtcAudioCapturerSink*> SinkList;
 
   WebRtcAudioCapturer();
 
+  // Reconfigures the capturer with a new buffer size and capture parameters.
+  // Must be called without holding the lock. Returns true on success.
+  bool Reconfigure(int sample_rate, media::AudioParameters::Format format,
+                   media::ChannelLayout channel_layout);
+
   // Used to DCHECK that we are called on the correct thread.
   base::ThreadChecker thread_checker_;
 
   // Protects |source_|, |sinks_|, |running_|, |on_device_stopped_cb_|,
-  // |loopback_fifo_| and |buffering_|.
+  // |loopback_fifo_|, |params| and |buffering_|.

[tommi (sloooow) - chröme, 2013/02/07 14:57:47]

params_

   base::Lock lock_;
 
+  // Keeps us from reconfiguring the buffer while the buffer is being used.
+  base::ConditionVariable buffer_in_use_cv_;
+  bool buffer_in_use_;
+
   // A list of sinks that the audio data is fed to.
   SinkList sinks_;
 
   // The audio data source from the browser process.
   scoped_refptr<media::AudioCapturerSource> source_;
 
   // Cached values of utilized audio parameters. Platform dependent.
   media::AudioParameters params_;
 
   // Buffers used for temporary storage during capture callbacks.
@@ skipping 12 matching lines @@
 
   // True when FIFO is utilized, false otherwise.
   bool buffering_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioCapturer);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_