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;
 
+  // Returns a pointer to a statically allocated string with the codec
+  // name, e.g. "g711A" or "g711U". The default implementation returns
+  // null, which means no name. Used to set
+  // EncodedInfoLeaf::encoder_name in AudioEncoderPcm::EncodeImpl
+  virtual const char* GetCodecName() const;

[ossu, 2016/05/12 12:50:14]

I think this should have a name that refers to logging as well, since that's
what it's used for (and the name must match something in a chromium XML file,
IIRC).

[aleloi, 2016/05/12 13:25:47]

The names in WebRTC and histograms.xml do not have to match. Only numbers are
sent in the UMA communication between WebRTC and chromium. I define names in the
XML to know what the numbers correspond to.

I am not sure about this and about what to change to. It's not specifically
tailored to logging, the name could in principle be used for something else.

+
  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_;
 };
 
 struct CodecInst;
 
 class AudioEncoderPcmA final : public AudioEncoderPcm {
  public:
   struct Config : public AudioEncoderPcm::Config {
     Config() : AudioEncoderPcm::Config(8) {}
   };
 
   explicit AudioEncoderPcmA(const Config& config)
       : AudioEncoderPcm(config, kSampleRateHz) {}
   explicit AudioEncoderPcmA(const CodecInst& codec_inst);
 
  protected:
   size_t EncodeCall(const int16_t* audio,
                     size_t input_len,
                     uint8_t* encoded) override;
 
   size_t BytesPerSample() const override;
 
+  const char* GetCodecName() const override;
+
  private:
   static const int kSampleRateHz = 8000;
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderPcmA);
 };
 
 class AudioEncoderPcmU final : public AudioEncoderPcm {
  public:
   struct Config : public AudioEncoderPcm::Config {
     Config() : AudioEncoderPcm::Config(0) {}
   };
 
   explicit AudioEncoderPcmU(const Config& config)
       : AudioEncoderPcm(config, kSampleRateHz) {}
   explicit AudioEncoderPcmU(const CodecInst& codec_inst);
 
  protected:
   size_t EncodeCall(const int16_t* audio,
                     size_t input_len,
                     uint8_t* encoded) override;
 
   size_t BytesPerSample() const override;
 
+  const char* GetCodecName() 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_