Only turn on the audio processing by default for MEDIA_DEVICE_AUDIO_CAPTURE

content/renderer/media/webrtc_audio_capturer.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/webrtc_audio_capturer.h"
 
 #include "base/bind.h"
 #include "base/logging.h"
 #include "base/metrics/histogram.h"
 #include "base/strings/string_util.h"
@@ skipping 286 matching lines @@
   // also apply the new |constraints|.
   // The idea is to get rid of any dependency of the microphone parameters
   // which would normally be used by default.
   // bits_per_sample is always 16 for now.
   int buffer_size = GetBufferSize(sample_rate);
   media::AudioParameters params(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                                 channel_layout, 0, sample_rate,
                                 16, buffer_size, effects);
   scoped_refptr<MediaStreamAudioProcessor> new_audio_processor(
       new talk_base::RefCountedObject<MediaStreamAudioProcessor>(
-          params, constraints, effects, audio_device_));
+          params, constraints, effects, device_info_.device.type,
+          audio_device_));
   {
     base::AutoLock auto_lock(lock_);
     audio_processor_ = new_audio_processor;
     need_audio_processing_ = NeedsAudioProcessing(constraints, effects);
 
     // Notify all tracks about the new format.
     tracks_.TagAll();
   }
 
   if (source.get())
@@ skipping 250 matching lines @@
     const scoped_refptr<media::AudioCapturerSource>& source,
     media::AudioParameters params) {
   // Create a new audio stream as source which uses the new source.
   SetCapturerSource(source, params.channel_layout(),
                     static_cast<float>(params.sample_rate()),
                     params.effects(),
                     constraints_);
 }
 
 }  // namespace content