Remove the last piece of deprecated synchronous IO code.

content/renderer/media/webaudio_capturer_source.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.
 
 #include "content/renderer/media/webaudio_capturer_source.h"
 
 #include "base/logging.h"
 #include "base/time/time.h"
 #include "content/renderer/media/webrtc_audio_capturer.h"
 #include "content/renderer/media/webrtc_local_audio_track.h"
@@ skipping 30 matching lines @@
   }
 
   ChannelLayout channel_layout =
       number_of_channels == 1 ? CHANNEL_LAYOUT_MONO : CHANNEL_LAYOUT_STEREO;
 
   base::AutoLock auto_lock(lock_);
   // Set the format used by this WebAudioCapturerSource. We are using 10ms data
   // as buffer size since that is the native buffer size of WebRtc packet
   // running on.
   params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
-                channel_layout, number_of_channels, 0, sample_rate, 16,
+                channel_layout, number_of_channels, sample_rate, 16,
                 sample_rate / 100);
   audio_format_changed_ = true;
 
   wrapper_bus_ = AudioBus::CreateWrapper(params_.channels());
   capture_bus_ = AudioBus::Create(params_);
   audio_data_.reset(
       new int16[params_.frames_per_buffer() * params_.channels()]);
   fifo_.reset(new AudioFifo(
       params_.channels(),
       kMaxNumberOfBuffersInFifo * params_.frames_per_buffer()));
@@ skipping 63 matching lines @@
     // TODO(xians): Avoid this interleave/deinterleave operation.
     capture_bus_->ToInterleaved(capture_bus_->frames(),
                                 params_.bits_per_sample() / 8,
                                 audio_data_.get());
     track_->Capture(audio_data_.get(), delay, volume, key_pressed,
                     need_audio_processing);
   }
 }
 
 }  // namespace content