Support RtpEncodingParameters::active in voice engine.

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  Copyright (c) 2004 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 1197 matching lines @@
   // Accessor to the VoE channel ID.
   int channel() const {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
     return config_.voe_channel_id;
   }
 
   const webrtc::RtpParameters& rtp_parameters() const {
     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();
     }
   }
 
   rtc::ThreadChecker worker_thread_checker_;
   rtc::ThreadChecker audio_capture_thread_checker_;
   webrtc::AudioTransport* const voe_audio_transport_ = nullptr;
   webrtc::Call* call_ = nullptr;
@@ skipping 211 matching lines @@
     return false;
   }
 
   if (!SetChannelParameters(it->second->channel(), parameters)) {
     LOG(LS_WARNING) << "Failed to set RtpParameters.";
     return false;
   }
   // 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;
 }
 
 bool WebRtcVoiceMediaChannel::ValidateRtpParameters(
     const webrtc::RtpParameters& rtp_parameters) {
   if (rtp_parameters.encodings.size() != 1) {
     LOG(LS_ERROR)
         << "Attempted to set RtpParameters without exactly one encoding";
     return false;
   }
@@ skipping 1124 matching lines @@
     }
   } else {
     LOG(LS_INFO) << "Stopping playout for channel #" << channel;
     engine()->voe()->base()->StopPlayout(channel);
   }
   return true;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE