Normalize codec names to those used by AcmCodecDatabase.

webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h

 /*
  *  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 41 matching lines @@
     bool enforce_frame_size = false;
   };
 
   explicit AudioEncoderIsacT(const Config& config);
   explicit AudioEncoderIsacT(
       const CodecInst& codec_inst,
       const rtc::scoped_refptr<LockedIsacBandwidthInfo>& bwinfo);
   AudioEncoderIsacT(int payload_type, const SdpAudioFormat& format);
   ~AudioEncoderIsacT() override;
 
-  static constexpr const char* GetPayloadName() { return "isac"; }
+  static constexpr const char* GetPayloadName() { return "ISAC"; }
   static rtc::Optional<AudioCodecInfo> QueryAudioEncoder(
       const SdpAudioFormat& format);
 
   int SampleRateHz() const override;
   size_t NumChannels() const override;
   size_t Num10MsFramesInNextPacket() const override;
   size_t Max10MsFramesInAPacket() const override;
   int GetTargetBitrate() const override;
   EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
                          rtc::ArrayView<const int16_t> audio,
@@ skipping 22 matching lines @@
 
   // Timestamp of the previously encoded packet.
   uint32_t last_encoded_timestamp_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderIsacT);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_