Audio codec usage statistics

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 13 matching lines @@
 // This is the interface class for encoders in AudioCoding module. Each codec
 // type must have an implementation of this class.
 class AudioEncoder {
  public:
   struct EncodedInfoLeaf {
     size_t encoded_bytes = 0;
     uint32_t encoded_timestamp = 0;
     int payload_type = 0;
     bool send_even_if_empty = false;
     bool speech = true;
+    const char* encoder_name = nullptr;
   };
 
   // This is the main struct for auxiliary encoding information. Each encoded
   // packet should be accompanied by one EncodedInfo struct, containing the
   // total number of |encoded_bytes|, the |encoded_timestamp| and the
   // |payload_type|. If the packet contains redundant encodings, the |redundant|
   // vector will be populated with EncodedInfoLeaf structs. Each struct in the
   // vector represents one encoding; the order of structs in the vector is the
   // same as the order in which the actual payloads are written to the byte
   // stream. When EncoderInfoLeaf structs are present in the vector, the main
@@ skipping 96 matching lines @@
   // This function is deprecated. It was used to return the maximum number of
   // bytes that can be produced by the encoder at each Encode() call.  Since the
   // Encode interface was changed to use rtc::Buffer, this is no longer
   // applicable. It is only kept in to avoid breaking subclasses that still have
   // it implemented (with the override attribute). It will be removed as soon
   // as these subclasses have been given a chance to change.
   virtual size_t MaxEncodedBytes() const;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_