Audio codec usage statistics

webrtc/modules/audio_coding/codecs/audio_encoder.h

Index: webrtc/modules/audio_coding/codecs/audio_encoder.h
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.h b/webrtc/modules/audio_coding/codecs/audio_encoder.h
index a0f1839e73393c4ad017c6d6d5393a740187f64d..001a120649a20c6a7ef8b6de0fe15a47275ee49e 100644
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.h
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.h
@@ -54,6 +54,8 @@ class AudioEncoder {
     std::vector<EncodedInfoLeaf> redundant;
   };
 
+  AudioEncoder();
+
   virtual ~AudioEncoder() = default;
 
   // Returns the input sample rate in Hz and the number of input channels.
@@ -136,6 +138,11 @@ class AudioEncoder {
                                  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
@@ -144,6 +151,12 @@ class AudioEncoder {
   // 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_