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

webrtc/modules/audio_coding/acm2/rent_a_codec.cc

Index: webrtc/modules/audio_coding/acm2/rent_a_codec.cc
diff --git a/webrtc/modules/audio_coding/acm2/rent_a_codec.cc b/webrtc/modules/audio_coding/acm2/rent_a_codec.cc
index 491df316ba53097861b0fb735580165e589a626d..3bc1464908fc006d480deccddf1fb6c1f8f011cd 100644
--- a/webrtc/modules/audio_coding/acm2/rent_a_codec.cc
+++ b/webrtc/modules/audio_coding/acm2/rent_a_codec.cc
@@ -154,12 +154,12 @@ std::unique_ptr<AudioEncoder> CreateEncoder(
 #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)
@@ -229,10 +229,10 @@ std::unique_ptr<AudioDecoder> CreateIsacDecoder(
     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>();