Remove the deprecated EncodeInternal interface from AudioEncoder

webrtc/modules/audio_coding/codecs/audio_encoder.cc

Index: webrtc/modules/audio_coding/codecs/audio_encoder.cc
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.cc b/webrtc/modules/audio_coding/codecs/audio_encoder.cc
index 6f793e253142e0e233c7389ef5810ed18d57aa37..fa262c46446baeb329edef103d134e0874841a5c 100644
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.cc
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.cc
@@ -37,55 +37,6 @@ AudioEncoder::EncodedInfo AudioEncoder::Encode(
   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::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 = 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;
 }
@@ -104,4 +55,9 @@ void AudioEncoder::SetProjectedPacketLossRate(double fraction) {}
 
 void AudioEncoder::SetTargetBitrate(int target_bps) {}
 
+size_t AudioEncoder::MaxEncodedBytes() const {
+  RTC_CHECK(false);
+  return 0;
+}
+
 }  // namespace webrtc