Added a kEchoCancellation constraint to turn off the audio processing.

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 147 matching lines @@
       device_info_.device.matched_output.sample_rate,
       device_info_.device.matched_output.frames_per_buffer,
       device_info_.device.input.effects));
 
   if (render_view_id_ == -1) {
     // Return true here to allow injecting a new source via
     // SetCapturerSourceForTesting() at a later state.
     return true;
   }
 
+  MediaAudioConstraints audio_constraints(constraints_,
+                                          device_info_.device.input.effects);
+  if (!audio_constraints.IsValid())
+    return false;
+
   media::ChannelLayout channel_layout = static_cast<media::ChannelLayout>(
       device_info_.device.input.channel_layout);
   DVLOG(1) << "Audio input hardware channel layout: " << channel_layout;
   UMA_HISTOGRAM_ENUMERATION("WebRTC.AudioInputChannelLayout",
                             channel_layout, media::CHANNEL_LAYOUT_MAX + 1);
 
   // Verify that the reported input channel configuration is supported.
   if (channel_layout != media::CHANNEL_LAYOUT_MONO &&
       channel_layout != media::CHANNEL_LAYOUT_STEREO) {
     DLOG(ERROR) << channel_layout
@@ skipping 38 matching lines @@
 
 WebRtcAudioCapturer::WebRtcAudioCapturer(
     int render_view_id,
     const StreamDeviceInfo& device_info,
     const blink::WebMediaConstraints& constraints,
     WebRtcAudioDeviceImpl* audio_device,
     MediaStreamAudioSource* audio_source)
     : constraints_(constraints),
       audio_processor_(
           new talk_base::RefCountedObject<MediaStreamAudioProcessor>(
-              constraints, device_info.device.input.effects,
-              device_info.device.type, audio_device)),
+              constraints, device_info.device.input.effects, audio_device)),
       running_(false),
       render_view_id_(render_view_id),
       device_info_(device_info),
       volume_(0),
       peer_connection_mode_(false),
       key_pressed_(false),
       need_audio_processing_(false),
       audio_device_(audio_device),
       audio_source_(audio_source),
       audio_power_monitor_(
@@ skipping 86 matching lines @@
   media::AudioParameters params(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                                 channel_layout, 0, sample_rate,
                                 16, buffer_size,
                                 device_info_.device.input.effects);
 
   {
     base::AutoLock auto_lock(lock_);
     // Notify the |audio_processor_| of the new format.
     audio_processor_->OnCaptureFormatChanged(params);
 
-    need_audio_processing_ = NeedsAudioProcessing(
-        constraints_, device_info_.device.input.effects);
+    MediaAudioConstraints audio_constraints(constraints_,
+                                            device_info_.device.input.effects);
+    need_audio_processing_ = audio_constraints.NeedsAudioProcessing();
     // Notify all tracks about the new format.
     tracks_.TagAll();
   }
 
   if (source.get())
     source->Initialize(params, this, session_id());
 
   Start();
 }
 
@@ skipping 268 matching lines @@
   DCHECK(aec_dump_file.IsValid());
   audio_processor_->StartAecDump(aec_dump_file.Pass());
 }
 
 void WebRtcAudioCapturer::StopAecDump() {
   DCHECK(thread_checker_.CalledOnValidThread());
   audio_processor_->StopAecDump();
 }
 
 }  // namespace content