Make an enum class out of NetEqDecoder, and hide the neteq_decoders_ table

webrtc/modules/audio_coding/neteq/include/neteq.h

 /*
  *  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 157 matching lines @@
   // |samples_per_channel| elements. If more than one channel is written,
   // the samples are interleaved.
   // The speech type is written to |type|, if |type| is not NULL.
   // Returns kOK on success, or kFail in case of an error.
   virtual int GetAudio(size_t max_length, int16_t* output_audio,
                        size_t* samples_per_channel, int* num_channels,
                        NetEqOutputType* type) = 0;
 
   // Associates |rtp_payload_type| with |codec| and stores the information in
   // the codec database. Returns 0 on success, -1 on failure.
-  virtual int RegisterPayloadType(enum NetEqDecoder codec,
+  virtual int RegisterPayloadType(NetEqDecoder codec,
                                   uint8_t rtp_payload_type) = 0;
 
   // Provides an externally created decoder object |decoder| to insert in the
   // decoder database. The decoder implements a decoder of type |codec| and
   // associates it with |rtp_payload_type|. The decoder will produce samples
   // at the rate |sample_rate_hz|. Returns kOK on success, kFail on failure.
   virtual int RegisterExternalDecoder(AudioDecoder* decoder,
-                                      enum NetEqDecoder codec,
+                                      NetEqDecoder codec,
                                       uint8_t rtp_payload_type,
                                       int sample_rate_hz) = 0;
 
   // Removes |rtp_payload_type| from the codec database. Returns 0 on success,
   // -1 on failure.
   virtual int RemovePayloadType(uint8_t rtp_payload_type) = 0;
 
   // Sets a minimum delay in millisecond for packet buffer. The minimum is
   // maintained unless a higher latency is dictated by channel condition.
   // Returns true if the minimum is successfully applied, otherwise false is
@@ skipping 82 matching lines @@
 
  protected:
   NetEq() {}
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(NetEq);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_NETEQ_H_