Support RtpEncodingParameters::active in voice engine.

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 63241332a3e9eb6540e923460671c4795fa4683d..856984f368cf3666cd9e9ac4be39cd4b7d04f416 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -1215,16 +1215,19 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
     return rtp_parameters_;
   }
 
-  void set_rtp_parameters(const webrtc::RtpParameters& parameters) {
+  void SetRtpParameters(const webrtc::RtpParameters& parameters) {
     RTC_CHECK_EQ(1UL, parameters.encodings.size());
     rtp_parameters_ = parameters;
+    // parameters.encodings[0].active could have changed.
+    UpdateSendState();
   }
 
  private:
   void UpdateSendState() {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
     RTC_DCHECK(stream_);
-    if (send_ && source_ != nullptr) {
+    RTC_DCHECK(rtp_parameters_.encodings.size() == 1u);

[the sun, 2016/05/03 08:14:04]

nit: use .._EQ like above

+    if (send_ && source_ != nullptr && rtp_parameters_.encodings[0].active) {
       stream_->Start();
     } else {  // !send || source_ = nullptr
       stream_->Stop();
@@ -1456,7 +1459,7 @@ bool WebRtcVoiceMediaChannel::SetRtpParameters(
   // Codecs are handled at the WebRtcVoiceMediaChannel level.
   webrtc::RtpParameters reduced_params = parameters;
   reduced_params.codecs.clear();
-  it->second->set_rtp_parameters(reduced_params);
+  it->second->SetRtpParameters(reduced_params);
   return true;
 }