AudioDecoderIsacT: Require caller to always specify sample rate

webrtc/modules/audio_coding/acm2/audio_coding_module_impl.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 692 matching lines @@
 int AudioCodingModuleImpl::PlayoutFrequency() const {
   WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
                "PlayoutFrequency()");
   return receiver_.last_output_sample_rate_hz();
 }
 
 int AudioCodingModuleImpl::RegisterReceiveCodec(const CodecInst& codec) {
   rtc::CritScope lock(&acm_crit_sect_);
   auto* ef = encoder_factory_.get();
   return RegisterReceiveCodecUnlocked(
-      codec, [ef] { return ef->rent_a_codec.RentIsacDecoder(); });
+      codec, [&] { return ef->rent_a_codec.RentIsacDecoder(codec.plfreq); });
 }
 
 int AudioCodingModuleImpl::RegisterReceiveCodec(
     const CodecInst& codec,
     FunctionView<std::unique_ptr<AudioDecoder>()> isac_factory) {
   rtc::CritScope lock(&acm_crit_sect_);
   return RegisterReceiveCodecUnlocked(codec, isac_factory);
 }
 
 int AudioCodingModuleImpl::RegisterReceiveCodecUnlocked(
@@ skipping 251 matching lines @@
   return receiver_.LeastRequiredDelayMs();
 }
 
 void AudioCodingModuleImpl::GetDecodingCallStatistics(
       AudioDecodingCallStats* call_stats) const {
   receiver_.GetDecodingCallStatistics(call_stats);
 }
 
 }  // namespace acm2
 }  // namespace webrtc