Renamed new EncodeInternal to EncodeImpl to ensure proper backwards compatibility.

webrtc/modules/audio_coding/codecs/audio_encoder.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 14 matching lines @@
 
 AudioEncoder::EncodedInfo AudioEncoder::Encode(
     uint32_t rtp_timestamp,
     rtc::ArrayView<const int16_t> audio,
     rtc::Buffer* encoded) {
   TRACE_EVENT0("webrtc", "AudioEncoder::Encode");
   RTC_CHECK_EQ(audio.size(),
                static_cast<size_t>(NumChannels() * SampleRateHz() / 100));
 
   const size_t old_size = encoded->size();
-  EncodedInfo info = EncodeInternal(rtp_timestamp, audio, encoded);
+  EncodedInfo info = EncodeImpl(rtp_timestamp, audio, encoded);
   RTC_CHECK_EQ(encoded->size() - old_size, info.encoded_bytes);
   return info;
 }
 
 AudioEncoder::EncodedInfo AudioEncoder::Encode(
     uint32_t rtp_timestamp,
     rtc::ArrayView<const int16_t> audio,
     size_t max_encoded_bytes,
     uint8_t* encoded) {
   return DEPRECATED_Encode(rtp_timestamp, audio, max_encoded_bytes, encoded);
 }
 
 AudioEncoder::EncodedInfo AudioEncoder::DEPRECATED_Encode(
     uint32_t rtp_timestamp,
     rtc::ArrayView<const int16_t> audio,
     size_t max_encoded_bytes,
     uint8_t* encoded) {
   TRACE_EVENT0("webrtc", "AudioEncoder::Encode");
   RTC_CHECK_EQ(audio.size(),
                static_cast<size_t>(NumChannels() * SampleRateHz() / 100));
   EncodedInfo info =
       EncodeInternal(rtp_timestamp, audio, max_encoded_bytes, encoded);
   RTC_CHECK_LE(info.encoded_bytes, max_encoded_bytes);
   return info;
 }
 
-AudioEncoder::EncodedInfo AudioEncoder::EncodeInternal(
+AudioEncoder::EncodedInfo AudioEncoder::EncodeImpl(
     uint32_t rtp_timestamp,
     rtc::ArrayView<const int16_t> audio,
     rtc::Buffer* encoded)
 {
   EncodedInfo info;
   encoded->AppendData(MaxEncodedBytes(), [&] (rtc::ArrayView<uint8_t> encoded) {
       info = EncodeInternal(rtp_timestamp, audio,
                             encoded.size(), encoded.data());
       return info.encoded_bytes;
     });
   return info;
 }
 
 AudioEncoder::EncodedInfo AudioEncoder::EncodeInternal(
     uint32_t rtp_timestamp,
     rtc::ArrayView<const int16_t> audio,
     size_t max_encoded_bytes,
     uint8_t* encoded)
 {
   rtc::Buffer temp_buffer;
-  EncodedInfo info = EncodeInternal(rtp_timestamp, audio, &temp_buffer);
+  EncodedInfo info = EncodeImpl(rtp_timestamp, audio, &temp_buffer);
   RTC_DCHECK_LE(temp_buffer.size(), max_encoded_bytes);
   std::memcpy(encoded, temp_buffer.data(), info.encoded_bytes);
   return info;
 }
 
 bool AudioEncoder::SetFec(bool enable) {
   return !enable;
 }
 
 bool AudioEncoder::SetDtx(bool enable) {
   return !enable;
 }
 
 bool AudioEncoder::SetApplication(Application application) {
   return false;
 }
 
 void AudioEncoder::SetMaxPlaybackRate(int frequency_hz) {}
 
 void AudioEncoder::SetProjectedPacketLossRate(double fraction) {}
 
 void AudioEncoder::SetTargetBitrate(int target_bps) {}
 
 }  // namespace webrtc