Fixes issue where Web Speech API drops a frame every 5.1 seconds

media/base/audio_converter.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.
 //
 // AudioConverter is a complete mixing, resampling, buffering, and channel
 // mixing solution for converting data from one set of AudioParameters to
 // another.
 //
 // For efficiency, pieces are only invoked when necessary; i.e.,
 //    - The resampler is only used if sample rates differ.
@@ skipping 72 matching lines @@
   void AddInput(InputCallback* input);
   void RemoveInput(InputCallback* input);
 
   // Flushes all buffered data.
   void Reset();
 
   // The maximum size in frames that guarantees we will only make a single call
   // to each input's ProvideInput for more data.
   int ChunkSize() const;
 
+  // See SincResampler::PrimeWithSilence.
+  void PrimeWithSilence();
+
   bool empty() const { return transform_inputs_.empty(); }
 
  private:
   // Provides input to the MultiChannelResampler.  Called by the resampler when
   // more data is necessary.
   void ProvideInput(int resampler_frame_delay, AudioBus* audio_bus);
 
   // Provides input to the AudioPullFifo.  Called by the fifo when more data is
   // necessary.
   void SourceCallback(int fifo_frame_delay, AudioBus* audio_bus);
@@ skipping 35 matching lines @@
   // value from the input AudioParameters class.  Preserved to recreate internal
   // AudioBus structures on demand in response to varying frame size requests.
   const int input_channel_count_;
 
   DISALLOW_COPY_AND_ASSIGN(AudioConverter);
 };
 
 }  // namespace media
 
 #endif  // MEDIA_BASE_AUDIO_CONVERTER_H_