Improves resolution when logging rate in the ADB class

webrtc/modules/audio_device/audio_device_buffer.cc

 /*
  *  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 380 matching lines @@
   RTC_DCHECK(task_queue_.IsCurrent());
 
   int64_t now_time = rtc::TimeMillis();
   int64_t next_callback_time = now_time + kTimerIntervalInMilliseconds;
   int64_t time_since_last = rtc::TimeDiff(now_time, last_log_stat_time_);
   last_log_stat_time_ = now_time;
 
   // Log the latest statistics but skip the first 10 seconds since we are not
   // sure of the exact starting point. I.e., the first log printout will be
   // after ~20 seconds.
   if (++num_stat_reports_ > 1) {

[hlundin-webrtc, 2016/09/20 07:30:25]

You should add && time_since_last > 0 to the if statement, to avoid div-by-zero.
(Probably not very likely in reality, but at the devil lives on the outskirts of
probability.)

[henrika_webrtc, 2016/09/20 09:05:32]

Thanks!

     uint32_t diff_samples = rec_samples_ - last_rec_samples_;
-    uint32_t rate = diff_samples / kTimerIntervalInSeconds;
+    float rate = diff_samples / (static_cast<float>(time_since_last) / 1000.0);
     LOG(INFO) << "[REC : " << time_since_last << "msec, "
               << rec_sample_rate_ / 1000
               << "kHz] callbacks: " << rec_callbacks_ - last_rec_callbacks_
               << ", "
               << "samples: " << diff_samples << ", "
-              << "rate: " << rate << ", "
+              << "rate: " << static_cast<int>(rate + 0.5) << ", "
               << "level: " << max_rec_level_;
 
     diff_samples = play_samples_ - last_play_samples_;
-    rate = diff_samples / kTimerIntervalInSeconds;
+    rate = diff_samples / (static_cast<float>(time_since_last) / 1000.0);
     LOG(INFO) << "[PLAY: " << time_since_last << "msec, "
               << play_sample_rate_ / 1000
               << "kHz] callbacks: " << play_callbacks_ - last_play_callbacks_
               << ", "
               << "samples: " << diff_samples << ", "
-              << "rate: " << rate << ", "
+              << "rate: " << static_cast<int>(rate + 0.5) << ", "
               << "level: " << max_play_level_;
   }
 
   // Count number of times we detect "no audio" corresponding to a case where
   // all level measurements have been zero.
   if (max_rec_level_ == 0) {
     ++num_rec_level_is_zero_;
   }
 
   last_rec_callbacks_ = rec_callbacks_;
@@ skipping 60 matching lines @@
     int16_t max_abs = WebRtcSpl_MaxAbsValueW16(
         static_cast<int16_t*>(const_cast<void*>(audio_buffer)),
         num_samples * play_channels_);
     if (max_abs > max_play_level_) {
       max_play_level_ = max_abs;
     }
   }
 }
 
 }  // namespace webrtc