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Improves resolution when logging rate in the ADB class.

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Trivial patch which fixes an issue where logged rate estimates could be
invalid. E.g. on iOS, two successive timer interrupts can be ~10.5 seconds
and not exactly 10.0 (which is usually the case on Android). In those
cases we could log a rate estimate of e.g. ~51000Hz instead of ~48000Hz.

This CL fixes that.

BUG=NONE

Review-Url: https://codereview.webrtc.org/2350103002
Cr-Commit-Position: refs/heads/master@{#14305}
This commit is contained in:
henrika 2016-09-20 04:44:04 -07:00 committed by Commit bot
parent 6b19b560ac
commit a6d26ec6a2
1 changed files with 5 additions and 5 deletions
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10
webrtc/modules/audio_device/audio_device_buffer.cc
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@ -398,25 +398,25 @@ void AudioDeviceBuffer::LogStats() {
// 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) {
if (++num_stat_reports_ > 1 && time_since_last > 0) {
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_;
}