Renamed new EncodeInternal to EncodeImpl to ensure proper backwards compatibility.

webrtc/modules/audio_coding/codecs/audio_encoder.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 71 matching lines @@
 
   // Returns the current target bitrate in bits/s. The value -1 means that the
   // codec adapts the target automatically, and a current target cannot be
   // provided.
   virtual int GetTargetBitrate() const = 0;
 
   // Accepts one 10 ms block of input audio (i.e., SampleRateHz() / 100 *
   // NumChannels() samples). Multi-channel audio must be sample-interleaved.
   // The encoder appends zero or more bytes of output to |encoded| and returns
   // additional encoding information.  Encode() checks some preconditions, calls
-  // EncodeInternal() which does the actual work, and then checks some
+  // EncodeImpl() which does the actual work, and then checks some
   // postconditions.
   EncodedInfo Encode(uint32_t rtp_timestamp,
                      rtc::ArrayView<const int16_t> audio,
                      rtc::Buffer* encoded);
 
   // Deprecated interface to Encode (remove eventually, bug 5591). May incur a
   // copy. The encoder produces zero or more bytes of output in |encoded| and
   // returns additional encoding information.  The caller is responsible for
   // making sure that |max_encoded_bytes| is not smaller than the number of
   // bytes actually produced by the encoder.
@@ skipping 55 matching lines @@
   // Tells the encoder what average bitrate we'd like it to produce. The
   // encoder is free to adjust or disregard the given bitrate (the default
   // implementation does the latter).
   virtual void SetTargetBitrate(int target_bps);
 
  protected:
   // Subclasses implement this to perform the actual encoding. Called by
   // Encode(). For compatibility reasons, this is implemented by default as a
   // call to the older version of EncodeInternal(). At least one of the two
   // interfaces of EncodeInternal _must_ be implemented by a subclass.
   // Preferably this one.

[kwiberg-webrtc, 2016/03/03 13:39:15]

This comment needs some tweaking.

-  virtual EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
-                                     rtc::ArrayView<const int16_t> audio,
-                                     rtc::Buffer* encoded);
+  virtual EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
+                                 rtc::ArrayView<const int16_t> audio,
+                                 rtc::Buffer* encoded);
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_