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"

[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"
 #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;
 }
 
 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_|.
 // It allows clients to inject their own capture data source by calling
 // SetCapturerSource(). It is also possible to enable a local sink and
 // register a callback which the sink can call when it wants to read captured
 // data cached in a FIFO for local loopback rendering.
 // The threading model for this class is rather complex since it will be
 // created on the main render thread, captured data is provided on a dedicated
 // AudioInputDevice thread, and methods can be called either on the Libjingle
@@ skipping 119 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, 2013/02/08 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.
   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.
   // Allocated during initialization.
-  scoped_array<int16> buffer_;
+  scoped_refptr<CaptureBuffer> buffer_;
   std::string device_id_;
   bool running_;
 
   // Callback object which is called during OnDeviceStopped().
   // Informs a local sink that it should stop asking for data.
   base::Closure on_device_stopped_cb_;
 
   // Contains copies of captured audio frames. Only utilized in loopback
   // mode when a local sink has been set.
   scoped_ptr<media::AudioFifo> loopback_fifo_;
 
   // True when FIFO is utilized, false otherwise.
   bool buffering_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioCapturer);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_