Possible solution to synchronization problems in webrtc audio capturer.

content/renderer/media/webrtc_audio_capturer.h

Index: content/renderer/media/webrtc_audio_capturer.h
diff --git a/content/renderer/media/webrtc_audio_capturer.h b/content/renderer/media/webrtc_audio_capturer.h
index afb072f5f7ab8204e1863eff744ac7d889ce963b..453a0b0308103daccc690512adbef43e5e643f55 100644
--- a/content/renderer/media/webrtc_audio_capturer.h
+++ b/content/renderer/media/webrtc_audio_capturer.h
@@ -10,6 +10,7 @@
 
 #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"
@@ -165,13 +166,22 @@ class CONTENT_EXPORT WebRtcAudioCapturer
 
   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_;