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

media/base/audio_converter.cc

 // 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 implementation.  Uses MultiChannelSincResampler for resampling
 // audio, ChannelMixer for channel mixing, and AudioPullFifo for buffering.
 //
 // Delay estimates are provided to InputCallbacks based on the frame delay
 // information reported via the resampler and FIFO units.
 
@@ skipping 99 matching lines @@
   if (resampler_)
     resampler_->Flush();
 }
 
 int AudioConverter::ChunkSize() const {
   if (!resampler_)
     return chunk_size_;
   return resampler_->ChunkSize();
 }
 
+void AudioConverter::PrimeWithSilence() {
+  if (resampler_) {
+    resampler_->PrimeWithSilence();
+  }
+}
+
 void AudioConverter::ConvertWithDelay(const base::TimeDelta& initial_delay,
                                       AudioBus* dest) {
   initial_delay_ = initial_delay;
 
   if (transform_inputs_.empty()) {
     dest->Zero();
     return;
   }
 
   // Determine if channel mixing should be done and if it should be done before
@@ skipping 113 matching lines @@
   else
     SourceCallback(0, dest);
 }
 
 void AudioConverter::CreateUnmixedAudioIfNecessary(int frames) {
   if (!unmixed_audio_ || unmixed_audio_->frames() != frames)
     unmixed_audio_ = AudioBus::Create(input_channel_count_, frames);
 }
 
 }  // namespace media