Introduced new scheme for controlling the functionality inside the audio processing module

webrtc/media/engine/webrtcvoiceengine.cc

Index: webrtc/media/engine/webrtcvoiceengine.cc
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index d0ca9aea617322719caf5cf0184c92128e25cb07..0e835238533580da156950a9eb0005da3725fe73 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -872,8 +872,9 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
 
   LOG(LS_INFO) << "Level control: "
                << (!!level_control_ ? *level_control_ : -1);
+  webrtc::AudioProcessing::Config apm_config;
   if (level_control_) {
-    config.Set<webrtc::LevelControl>(new webrtc::LevelControl(*level_control_));
+    apm_config.level_controller.enabled = *level_control_;
   }
 
   // We check audioproc for the benefit of tests, since FakeWebRtcVoiceEngine
@@ -883,6 +884,9 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
     audioproc->SetExtraOptions(config);
   }
 
+  RTC_DCHECK(apm_config.Validate());
+  voe_wrapper_->base()->audio_processing()->ApplyConfig(apm_config);

[the sun, 2016/08/30 18:05:35]

It looks to me, reading the comment above, that this call should be within the
above conditional. As a bonus, you can use the "audioproc" variable, rather than
fetching the APM* again.

[peah-webrtc, 2016/08/31 08:08:31]

Great point!
Done.

+
   if (options.recording_sample_rate) {
     LOG(LS_INFO) << "Recording sample rate is "
                  << *options.recording_sample_rate;