Give Audio{De,En}coderIsac* an "Impl" suffix, to free up the original names

webrtc/modules/audio_coding/acm2/rent_a_codec.cc

 /*
  *  Copyright (c) 2015 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 136 matching lines @@
 namespace {
 
 // Returns a new speech encoder, or null on error.
 // TODO(kwiberg): Don't handle errors here (bug 5033)
 std::unique_ptr<AudioEncoder> CreateEncoder(
     const CodecInst& speech_inst,
     const rtc::scoped_refptr<LockedIsacBandwidthInfo>& bwinfo) {
 #if defined(WEBRTC_CODEC_ISACFX)
   if (STR_CASE_CMP(speech_inst.plname, "isac") == 0)
     return std::unique_ptr<AudioEncoder>(
-        new AudioEncoderIsacFix(speech_inst, bwinfo));
+        new AudioEncoderIsacFixImpl(speech_inst, bwinfo));
 #endif
 #if defined(WEBRTC_CODEC_ISAC)
   if (STR_CASE_CMP(speech_inst.plname, "isac") == 0)
     return std::unique_ptr<AudioEncoder>(
-        new AudioEncoderIsac(speech_inst, bwinfo));
+        new AudioEncoderIsacFloatImpl(speech_inst, bwinfo));
 #endif
 #ifdef WEBRTC_CODEC_OPUS
   if (STR_CASE_CMP(speech_inst.plname, "opus") == 0)
     return std::unique_ptr<AudioEncoder>(new AudioEncoderOpus(speech_inst));
 #endif
   if (STR_CASE_CMP(speech_inst.plname, "pcmu") == 0)
     return std::unique_ptr<AudioEncoder>(new AudioEncoderPcmU(speech_inst));
   if (STR_CASE_CMP(speech_inst.plname, "pcma") == 0)
     return std::unique_ptr<AudioEncoder>(new AudioEncoderPcmA(speech_inst));
   if (STR_CASE_CMP(speech_inst.plname, "l16") == 0)
@@ skipping 49 matching lines @@
       FATAL();
   }
   return std::unique_ptr<AudioEncoder>(new AudioEncoderCng(std::move(config)));
 }
 
 std::unique_ptr<AudioDecoder> CreateIsacDecoder(
     int sample_rate_hz,
     const rtc::scoped_refptr<LockedIsacBandwidthInfo>& bwinfo) {
 #if defined(WEBRTC_CODEC_ISACFX)
   return std::unique_ptr<AudioDecoder>(
-      new AudioDecoderIsacFix(sample_rate_hz, bwinfo));
+      new AudioDecoderIsacFixImpl(sample_rate_hz, bwinfo));
 #elif defined(WEBRTC_CODEC_ISAC)
   return std::unique_ptr<AudioDecoder>(
-      new AudioDecoderIsac(sample_rate_hz, bwinfo));
+      new AudioDecoderIsacFloatImpl(sample_rate_hz, bwinfo));
 #else
   FATAL() << "iSAC is not supported.";
   return std::unique_ptr<AudioDecoder>();
 #endif
 }
 
 }  // namespace
 
 RentACodec::RentACodec() {
 #if defined(WEBRTC_CODEC_ISACFX) || defined(WEBRTC_CODEC_ISAC)
@@ skipping 59 matching lines @@
   }
   return encoder_stack;
 }
 
 std::unique_ptr<AudioDecoder> RentACodec::RentIsacDecoder(int sample_rate_hz) {
   return CreateIsacDecoder(sample_rate_hz, isac_bandwidth_info_);
 }
 
 }  // namespace acm2
 }  // namespace webrtc