Don't recreate the speech encoder if we don't have to

webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.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 169 matching lines @@
 }
 
 void AudioEncoderCng::SetProjectedPacketLossRate(double fraction) {
   speech_encoder_->SetProjectedPacketLossRate(fraction);
 }
 
 void AudioEncoderCng::SetTargetBitrate(int bits_per_second) {
   speech_encoder_->SetTargetBitrate(bits_per_second);
 }
 
+rtc::ArrayView<std::unique_ptr<AudioEncoder>>
+AudioEncoderCng::ReclaimContainedEncoders() {
+  return rtc::ArrayView<std::unique_ptr<AudioEncoder>>(&speech_encoder_, 1);
+}
+
 AudioEncoder::EncodedInfo AudioEncoderCng::EncodePassive(
     size_t frames_to_encode,
     rtc::Buffer* encoded) {
   bool force_sid = last_frame_active_;
   bool output_produced = false;
   const size_t samples_per_10ms_frame = SamplesPer10msFrame();
   AudioEncoder::EncodedInfo info;
 
   for (size_t i = 0; i < frames_to_encode; ++i) {
     // It's important not to pass &info.encoded_bytes directly to
@@ skipping 42 matching lines @@
     }
   }
   return info;
 }
 
 size_t AudioEncoderCng::SamplesPer10msFrame() const {
   return rtc::CheckedDivExact(10 * SampleRateHz(), 1000);
 }
 
 }  // namespace webrtc