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

webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc

 /*
  *  Copyright (c) 2014 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 735 matching lines @@
       : AudioCodingModuleTestOldApi(),
         receive_thread_(CbReceiveThread, this, "receive"),
         codec_registration_thread_(CbCodecRegistrationThread,
                                    this,
                                    "codec_registration"),
         test_complete_(EventWrapper::Create()),
         codec_registered_(false),
         receive_packet_count_(0),
         next_insert_packet_time_ms_(0),
         fake_clock_(new SimulatedClock(0)) {
-    AudioEncoderIsac::Config config;
+    AudioEncoderIsacFloatImpl::Config config;
     config.payload_type = kPayloadType;
-    isac_encoder_.reset(new AudioEncoderIsac(config));
+    isac_encoder_.reset(new AudioEncoderIsacFloatImpl(config));
     clock_ = fake_clock_.get();
   }
 
   void SetUp() override {
     AudioCodingModuleTestOldApi::SetUp();
     // Set up input audio source to read from specified file, loop after 5
     // seconds, and deliver blocks of 10 ms.
     const std::string input_file_name =
         webrtc::test::ResourcePath("audio_coding/speech_mono_16kHz", "pcm");
     audio_loop_.Init(input_file_name, 5 * kSampleRateHz, kNumSamples10ms);
@@ skipping 106 matching lines @@
     return true;
   }
 
   rtc::PlatformThread receive_thread_;
   rtc::PlatformThread codec_registration_thread_;
   const std::unique_ptr<EventWrapper> test_complete_;
   rtc::CriticalSection crit_sect_;
   bool codec_registered_ GUARDED_BY(crit_sect_);
   int receive_packet_count_ GUARDED_BY(crit_sect_);
   int64_t next_insert_packet_time_ms_ GUARDED_BY(crit_sect_);
-  std::unique_ptr<AudioEncoderIsac> isac_encoder_;
+  std::unique_ptr<AudioEncoderIsacFloatImpl> isac_encoder_;
   std::unique_ptr<SimulatedClock> fake_clock_;
   test::AudioLoop audio_loop_;
 };
 
 #if defined(WEBRTC_IOS)
 #define MAYBE_DoTest DISABLED_DoTest
 #else
 #define MAYBE_DoTest DoTest
 #endif
 #if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
@@ skipping 1060 matching lines @@
   Run(16000, 8000, 1000);
 }
 
 TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle8KhzTo16Khz) {
   Run(8000, 16000, 1000);
 }
 
 #endif
 
 }  // namespace webrtc