Fold AudioEncoderMutable into AudioEncoder

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.
  */
 
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
 #include "webrtc/base/checks.h"
 
 namespace webrtc {
 
-AudioEncoder::EncodedInfo::EncodedInfo() : EncodedInfoLeaf() {
-}
+AudioEncoder::EncodedInfo::EncodedInfo() = default;
 
-AudioEncoder::EncodedInfo::~EncodedInfo() {
+AudioEncoder::EncodedInfo::~EncodedInfo() = default;
+
+int AudioEncoder::RtpTimestampRateHz() const {
+  return SampleRateHz();
 }
 
 AudioEncoder::EncodedInfo AudioEncoder::Encode(uint32_t rtp_timestamp,
                                                const int16_t* audio,
                                                size_t num_samples_per_channel,
                                                size_t max_encoded_bytes,
                                                uint8_t* encoded) {
   CHECK_EQ(num_samples_per_channel,
            static_cast<size_t>(SampleRateHz() / 100));
   EncodedInfo info =
       EncodeInternal(rtp_timestamp, audio, max_encoded_bytes, encoded);
   CHECK_LE(info.encoded_bytes, max_encoded_bytes);
   return info;
 }
 
-int AudioEncoder::RtpTimestampRateHz() const {
-  return SampleRateHz();
+bool AudioEncoder::SetFec(bool enable) {
+  return !enable;
 }
 
+bool AudioEncoder::SetDtx(bool enable) {
+  return !enable;
+}
+
+bool AudioEncoder::SetApplication(Application application) {
+  return false;
+}
+
+bool AudioEncoder::SetMaxPlaybackRate(int frequency_hz) {
+  return true;
+}
+
+void AudioEncoder::SetProjectedPacketLossRate(double fraction) {}
+
+void AudioEncoder::SetTargetBitrate(int target_bps) {}
+
+void AudioEncoder::SetMaxBitrate(int max_bps) {}
+
+void AudioEncoder::SetMaxPayloadSize(int max_payload_size_bytes) {}
+
 }  // namespace webrtc