AudioDecoderIsacT: Require caller to always specify sample rate

webrtc/voice_engine/channel.cc

 /*
  *  Copyright (c) 2012 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 31 matching lines @@
 #include "webrtc/voice_engine/utility.h"
 
 namespace webrtc {
 namespace voe {
 
 namespace {
 
 bool RegisterReceiveCodec(std::unique_ptr<AudioCodingModule>* acm,
                           acm2::RentACodec* rac,
                           const CodecInst& ci) {
-  const int result =
-      (*acm)->RegisterReceiveCodec(ci, [&] { return rac->RentIsacDecoder(); });
+  const int result = (*acm)->RegisterReceiveCodec(
+      ci, [&] { return rac->RentIsacDecoder(ci.plfreq); });
   return result == 0;
 }
 
 }  // namespace
 
 const int kTelephoneEventAttenuationdB = 10;
 
 class TransportFeedbackProxy : public TransportFeedbackObserver {
  public:
   TransportFeedbackProxy() : feedback_observer_(nullptr) {
@@ skipping 3513 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc