Audio codec usage statistics

webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.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 30 matching lines @@
   int RtpTimestampRateHz() const override;
   size_t Num10MsFramesInNextPacket() const override;
   size_t Max10MsFramesInAPacket() const override;
   int GetTargetBitrate() const override;
   void Reset() override;
 
  protected:
   EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
                          rtc::ArrayView<const int16_t> audio,
                          rtc::Buffer* encoded) override;
+  const char* GetCodecName() const override;
 
  private:
   // The encoder state for one channel.
   struct EncoderState {
     G722EncInst* encoder;
     std::unique_ptr<int16_t[]> speech_buffer;   // Queued up for encoding.
     rtc::Buffer encoded_buffer;                 // Already encoded.
     EncoderState();
     ~EncoderState();
   };
 
   size_t SamplesPerChannel() const;
 
   const size_t num_channels_;
   const int payload_type_;
   const size_t num_10ms_frames_per_packet_;
   size_t num_10ms_frames_buffered_;
   uint32_t first_timestamp_in_buffer_;
   const std::unique_ptr<EncoderState[]> encoders_;
   rtc::Buffer interleave_buffer_;
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderG722);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_