Add encode/decode time tracing to audio_coding.

webrtc/modules/audio_coding/codecs/audio_decoder.h

 /*
  *  Copyright (c) 2012 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 23 matching lines @@
   AudioDecoder() = default;
   virtual ~AudioDecoder() = default;
 
   // Decodes |encode_len| bytes from |encoded| and writes the result in
   // |decoded|. The maximum bytes allowed to be written into |decoded| is
   // |max_decoded_bytes|. Returns the total number of samples across all
   // channels. If the decoder produced comfort noise, |speech_type|
   // is set to kComfortNoise, otherwise it is kSpeech. The desired output
   // sample rate is provided in |sample_rate_hz|, which must be valid for the
   // codec at hand.
-  virtual int Decode(const uint8_t* encoded,
-                     size_t encoded_len,
-                     int sample_rate_hz,
-                     size_t max_decoded_bytes,
-                     int16_t* decoded,
-                     SpeechType* speech_type);
+  int Decode(const uint8_t* encoded,
+             size_t encoded_len,
+             int sample_rate_hz,
+             size_t max_decoded_bytes,
+             int16_t* decoded,
+             SpeechType* speech_type);
 
   // Same as Decode(), but interfaces to the decoders redundant decode function.
   // The default implementation simply calls the regular Decode() method.
-  virtual int DecodeRedundant(const uint8_t* encoded,
-                              size_t encoded_len,
-                              int sample_rate_hz,
-                              size_t max_decoded_bytes,
-                              int16_t* decoded,
-                              SpeechType* speech_type);
+  int DecodeRedundant(const uint8_t* encoded,
+                      size_t encoded_len,
+                      int sample_rate_hz,
+                      size_t max_decoded_bytes,
+                      int16_t* decoded,
+                      SpeechType* speech_type);
 
   // Indicates if the decoder implements the DecodePlc method.
   virtual bool HasDecodePlc() const;
 
   // Calls the packet-loss concealment of the decoder to update the state after
   // one or several lost packets. The caller has to make sure that the
   // memory allocated in |decoded| should accommodate |num_frames| frames.
   virtual size_t DecodePlc(size_t num_frames, int16_t* decoded);
 
   // Resets the decoder state (empty buffers etc.).
@@ skipping 31 matching lines @@
 
   virtual size_t Channels() const = 0;
 
  protected:
   static SpeechType ConvertSpeechType(int16_t type);
 
   virtual int DecodeInternal(const uint8_t* encoded,
                              size_t encoded_len,
                              int sample_rate_hz,
                              int16_t* decoded,
-                             SpeechType* speech_type);
+                             SpeechType* speech_type) = 0;
 
   virtual int DecodeRedundantInternal(const uint8_t* encoded,
                                       size_t encoded_len,
                                       int sample_rate_hz,
                                       int16_t* decoded,
                                       SpeechType* speech_type);
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoder);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_AUDIO_DECODER_H_