Index: webrtc/modules/audio_device/audio_device_buffer.cc |
diff --git a/webrtc/modules/audio_device/audio_device_buffer.cc b/webrtc/modules/audio_device/audio_device_buffer.cc |
index 0c85fdaf833944b46aca88635eb5ad4f95620456..28d6a97e8b0bde5f81c195c51fcf88d0e36ce913 100644 |
--- a/webrtc/modules/audio_device/audio_device_buffer.cc |
+++ b/webrtc/modules/audio_device/audio_device_buffer.cc |
@@ -400,23 +400,23 @@ void AudioDeviceBuffer::LogStats() { |
// 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 st
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_; |
} |