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 36 matching lines @@
     EncodedInfo();
     EncodedInfo(const EncodedInfo&);
     EncodedInfo(EncodedInfo&&);
     ~EncodedInfo();
     EncodedInfo& operator=(const EncodedInfo&);
     EncodedInfo& operator=(EncodedInfo&&);
 
     std::vector<EncodedInfoLeaf> redundant;
   };
 
+  AudioEncoder();
+
   virtual ~AudioEncoder() = default;
 
   // Returns the input sample rate in Hz and the number of input channels.
   // These are constants set at instantiation time.
   virtual int SampleRateHz() const = 0;
   virtual size_t NumChannels() const = 0;
 
   // Returns the rate at which the RTP timestamps are updated. The default
   // implementation returns SampleRateHz().
   virtual int RtpTimestampRateHz() const;
@@ skipping 62 matching lines @@
   // implementation does the latter).
   virtual void SetTargetBitrate(int target_bps);
 
  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;
 
+  // Returns a pointer to a statically allocated string with the codec
+  // name, e.g. "Opus" or "iSAC". The default implementation returns
+  // null, which means no name
+  virtual const char* GetCodecName() const;
+
  private:
   // 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;
+
+  // Every time AudioEncoder::Encode() is called, this number is
+  // incremented. Every 500 increments, this number is set to zero and
+  // a value is logged to the UMA histogram
+  // "WebRTC.Audio.Encoder.CodecType"
+  size_t encoded_packets;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_