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 71 matching lines @@
     chunk_size_ = input_params.frames_per_buffer();
     audio_fifo_.reset(new AudioPullFifo(
         downmix_early_ ? output_params.channels() : input_params.channels(),
         chunk_size_,
         base::Bind(&AudioConverter::SourceCallback, base::Unretained(this))));
   }
 }
 
 AudioConverter::~AudioConverter() {}
 
+void AudioConverter::PrimeWithSilence() {
+  if (!resampler_)
+    return;
+  DVLOG(1) << "AudioConverter::PrimeWithSilence";

[DaleCurtis, 2015/07/07 16:38:17]

Remove before submit and invert conditional; i.e. if (resampler_)
PrimeWithSilence().

[henrika (OOO until Aug 14), 2015/07/08 12:22:59]

Done.

+  resampler_->PrimeWithSilence();
+}
+
 void AudioConverter::AddInput(InputCallback* input) {
   DCHECK(std::find(transform_inputs_.begin(), transform_inputs_.end(), input) ==
          transform_inputs_.end());
   transform_inputs_.push_back(input);
 }
 
 void AudioConverter::RemoveInput(InputCallback* input) {
   DCHECK(std::find(transform_inputs_.begin(), transform_inputs_.end(), input) !=
          transform_inputs_.end());
   transform_inputs_.remove(input);
@@ skipping 141 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