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..31829023f43485214447caf9b6d6e1e4d973a84b 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"

[tommi (sloooow) - chröme, 2013/02/08 14:56:57]

remove

[phoglund_chromium, 2013/02/08 16:10:38]

Done.

 #include "base/synchronization/lock.h"
 #include "base/threading/thread_checker.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
@@ -26,6 +27,7 @@ namespace content {
 
 class WebRtcAudioCapturerSink;
 class WebRtcLocalAudioRenderer;
+class CaptureBuffer;

[tommi (sloooow) - chröme, 2013/02/08 14:56:57]

can we make this a private class inside WebRtcAudioCapturer?

[phoglund_chromium, 2013/02/08 16:10:38]

Done.

 
 // This class manages the capture data flow by getting data from its
 // |source_|, and passing it to its |sink_|.
@@ -165,11 +167,16 @@ 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, 2013/02/08 14:56:57]

params_

[phoglund_chromium, 2013/02/08 16:10:38]

Done.

   base::Lock lock_;
 
   // A list of sinks that the audio data is fed to.
@@ -183,7 +190,7 @@ class CONTENT_EXPORT WebRtcAudioCapturer
 
   // Buffers used for temporary storage during capture callbacks.
   // Allocated during initialization.
-  scoped_array<int16> buffer_;
+  scoped_refptr<CaptureBuffer> buffer_;
   std::string device_id_;
   bool running_;