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

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 131 matching lines @@
   // the range [0.0, 1.0]. The encoder would typically use this information to
   // adjust channel coding efforts, such as FEC. The default implementation
   // does nothing.
   virtual void SetProjectedPacketLossRate(double fraction);
 
   // 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);
 
+  // Causes this encoder to let go of any other encoders it contains, and
+  // returns a pointer to an array where they are stored (which is required to
+  // live as long as this encoder). Unless the returned array is empty, you may
+  // not call any methods on this encoder afterwards, except for the
+  // destructor. The default implementation just returns an empty array.
+  // NOTE: This method is subject to change. Do not call or override it.
+  virtual rtc::ArrayView<std::unique_ptr<AudioEncoder>>
+  ReclaimContainedEncoders();

[kwiberg-webrtc, 2016/06/22 13:46:12]

This method is something of a hack. Hopefully, we'll be able to get rid of it
when we implement the prophesied encoder factory (since we should then set all
encoder properties at creation, and thus will lose no config state by recreating
encoders, which we should not end up doing anyway because we don't set a bunch
of properties after the fact).

+
  protected:
   // Subclasses implement this to perform the actual encoding. Called by
   // Encode().
   virtual EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
                                  rtc::ArrayView<const int16_t> audio,
                                  rtc::Buffer* encoded) = 0;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_