Audio codec usage statistics

webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.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 37 matching lines @@
   EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
                          rtc::ArrayView<const int16_t> audio,
                          rtc::Buffer* encoded) override;
 
   virtual size_t EncodeCall(const int16_t* audio,
                             size_t input_len,
                             uint8_t* encoded) = 0;
 
   virtual size_t BytesPerSample() const = 0;
 
+  const char* GetCodecName() const override;

[kwiberg-webrtc, 2016/05/11 00:47:00]

AudioEncoderPcm is also base class for the 16-bit pcm encoder, which IIRC isn't
called g711. You probably need to move this override to the leaf classes.

[aleloi, 2016/05/11 11:16:13]

Acknowledged.

+
  private:
   const int sample_rate_hz_;
   const size_t num_channels_;
   const int payload_type_;
   const size_t num_10ms_frames_per_packet_;
   const size_t full_frame_samples_;
   std::vector<int16_t> speech_buffer_;
   uint32_t first_timestamp_in_buffer_;
 };
 
@@ skipping 39 matching lines @@
   size_t BytesPerSample() const override;
 
  private:
   static const int kSampleRateHz = 8000;
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderPcmU);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_