WebRtcVoiceCodecs: Eliminate some useless copying

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 377 matching lines @@
     for (const AudioCodec& codec : codecs) {
       if (IsCodec(codec, kDtmfCodecName) || IsCodec(codec, kCnCodecName)) {
         // Skip telephone-event/CN codec, which will be handled later.
         continue;
       }
 
       // We'll use the first codec in the list to actually send audio data.
       // Be sure to use the payload type requested by the remote side.
       // Ignore codecs we don't know about. The negotiation step should prevent
       // this, but double-check to be sure.
-      webrtc::CodecInst voe_codec = {0};
-      if (!ToCodecInst(codec, &voe_codec)) {
+      if (!ToCodecInst(codec, out)) {
         LOG(LS_WARNING) << "Unknown codec " << ToString(codec);
         continue;
       }
-      *out = voe_codec;
       return &codec;
     }
     return nullptr;
   }
 
  private:
   static const int kMaxNumPacketSize = 6;
   struct CodecPref {
     const char* name;
     int clockrate;
@@ skipping 2177 matching lines @@
     }
   } else {
     LOG(LS_INFO) << "Stopping playout for channel #" << channel;
     engine()->voe()->base()->StopPlayout(channel);
   }
   return true;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE