Move expand uma logger into statistics calculator.
Bug: webrtc:370424996
Change-Id: I525758eaa5430a4d1cf63cfd663de0079e7d3d68
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364100
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43106}
diff --git a/modules/audio_coding/neteq/statistics_calculator.cc b/modules/audio_coding/neteq/statistics_calculator.cc
index b4fd5e8..4f0f74a 100644
--- a/modules/audio_coding/neteq/statistics_calculator.cc
+++ b/modules/audio_coding/neteq/statistics_calculator.cc
@@ -112,7 +112,7 @@
counter_ = 0;
}
-StatisticsCalculator::StatisticsCalculator()
+StatisticsCalculator::StatisticsCalculator(TickTimer* tick_timer)
: preemptive_samples_(0),
accelerate_samples_(0),
expanded_speech_samples_(0),
@@ -129,7 +129,13 @@
1000),
buffer_full_counter_("WebRTC.Audio.JitterBufferFullPerMinute",
60000, // 60 seconds report interval.
- 100) {}
+ 100),
+ expand_uma_logger_("WebRTC.Audio.ExpandRatePercent",
+ 10, // Report once every 10 s.
+ tick_timer),
+ speech_expand_uma_logger_("WebRTC.Audio.SpeechExpandRatePercent",
+ 10, // Report once every 10 s.
+ tick_timer) {}
StatisticsCalculator::~StatisticsCalculator() = default;
@@ -260,6 +266,12 @@
timestamps_since_last_report_ = 0;
}
lifetime_stats_.total_samples_received += num_samples;
+ expand_uma_logger_.UpdateSampleCounter(lifetime_stats_.concealed_samples,
+ fs_hz);
+ speech_expand_uma_logger_.UpdateSampleCounter(
+ lifetime_stats_.concealed_samples -
+ lifetime_stats_.silent_concealed_samples,
+ fs_hz);
}
void StatisticsCalculator::JitterBufferDelay(size_t num_samples,