Adding a new ChangeLogger class to handle UMA logging of bitrates

webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.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 232 matching lines @@
   struct InputData {
     uint32_t input_timestamp;
     const int16_t* audio;
     uint16_t length_per_channel;
     uint8_t audio_channel;
     // If a re-mix is required (up or down), this buffer will store a re-mixed
     // version of the input.
     int16_t buffer[WEBRTC_10MS_PCM_AUDIO];
   };
 
+  // This member class writes values to the named UMA histogram, but only if
+  // the value has changed since the last time (and always for the first call).
+  class ChangeLogger {
+   public:
+    explicit ChangeLogger(const std::string& histogram_name)
+        : histogram_name_(histogram_name) {}
+    // Logs the new value if it is different from the last logged value, or if
+    // this is the first call.
+    void MaybeLog(int value);
+
+   private:
+    int last_value_ = 0;
+    int first_time_ = true;
+    const std::string histogram_name_;
+  };
+
   int Add10MsDataInternal(const AudioFrame& audio_frame, InputData* input_data)
       EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
   int Encode(const InputData& input_data)
       EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
 
   int InitializeReceiverSafe() EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
 
   bool HaveValidEncoder(const char* caller_name) const
       EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
 
@@ skipping 15 matching lines @@
   // Change required states after starting to receive the codec corresponding
   // to |index|.
   int UpdateUponReceivingCodec(int index);
 
   CriticalSectionWrapper* acm_crit_sect_;
   int id_;  // TODO(henrik.lundin) Make const.
   uint32_t expected_codec_ts_ GUARDED_BY(acm_crit_sect_);
   uint32_t expected_in_ts_ GUARDED_BY(acm_crit_sect_);
   ACMResampler resampler_ GUARDED_BY(acm_crit_sect_);
   AcmReceiver receiver_;  // AcmReceiver has it's own internal lock.
+  ChangeLogger bitrate_logger_ GUARDED_BY(acm_crit_sect_);
   CodecManager codec_manager_ GUARDED_BY(acm_crit_sect_);
 
   // This is to keep track of CN instances where we can send DTMFs.
   uint8_t previous_pltype_ GUARDED_BY(acm_crit_sect_);
 
   // Used when payloads are pushed into ACM without any RTP info
   // One example is when pre-encoded bit-stream is pushed from
   // a file.
   // IMPORTANT: this variable is only used in IncomingPayload(), therefore,
   // no lock acquired when interacting with this variable. If it is going to
@@ skipping 85 matching lines @@
   int playout_frequency_hz_;
   // TODO(henrik.lundin): All members below this line are temporary and should
   // be removed after refactoring is completed.
   rtc::scoped_ptr<acm2::AudioCodingModuleImpl> acm_old_;
   CodecInst current_send_codec_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_AUDIO_CODING_MODULE_IMPL_H_