AnotherFoxGuy/webrtc-audio-processing
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Bump to WebRTC M120 release

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Some API deprecation -- ExperimentalAgc and ExperimentalNs are gone.
We're continuing to carry iSAC even though it's gone upstream, but maybe
we'll want to drop that soon.
This commit is contained in:
Arun Raghavan
2023-12-12 10:42:58 -05:00
parent 9a202fb8c2
commit c6abf6cd3f
479 changed files with 20900 additions and 11996 deletions
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29
webrtc/modules/audio_processing/aec3/render_delay_controller_metrics.cc
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@ -37,16 +37,13 @@ enum class DelayChangesCategory {
kNumCategories
};
constexpr int kMaxSkewShiftCount = 20;
} // namespace
RenderDelayControllerMetrics::RenderDelayControllerMetrics() = default;
void RenderDelayControllerMetrics::Update(
absl::optional<size_t> delay_samples,
size_t buffer_delay_blocks,
absl::optional<int> skew_shift_blocks,
absl::optional<size_t> buffer_delay_blocks,
ClockdriftDetector::Level clockdrift) {
++call_counter_;
@ -54,6 +51,8 @@ void RenderDelayControllerMetrics::Update(
size_t delay_blocks;
if (delay_samples) {
++reliable_delay_estimate_counter_;
// Add an offset by 1 (metric is halved before reporting) to reserve 0 for
// absent delay.
delay_blocks = (*delay_samples) / kBlockSize + 2;
} else {
delay_blocks = 0;
@ -64,21 +63,21 @@ void RenderDelayControllerMetrics::Update(
delay_blocks_ = delay_blocks;
}
if (skew_shift_blocks) {
skew_shift_count_ = std::min(kMaxSkewShiftCount, skew_shift_count_);
}
} else if (++initial_call_counter_ == 5 * kNumBlocksPerSecond) {
initial_update = false;
}
if (call_counter_ == kMetricsReportingIntervalBlocks) {
int value_to_report = static_cast<int>(delay_blocks_);
// Divide by 2 to compress metric range.
value_to_report = std::min(124, value_to_report >> 1);
RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.EchoCanceller.EchoPathDelay",
value_to_report, 0, 124, 125);
value_to_report = static_cast<int>(buffer_delay_blocks + 2);
value_to_report = std::min(124, value_to_report >> 1);
// Divide by 2 to compress metric range.
// Offset by 1 to reserve 0 for absent delay.
value_to_report = buffer_delay_blocks ? (*buffer_delay_blocks + 2) >> 1 : 0;
value_to_report = std::min(124, value_to_report);
RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.EchoCanceller.BufferDelay",
value_to_report, 0, 124, 125);
@ -120,20 +119,8 @@ void RenderDelayControllerMetrics::Update(
"WebRTC.Audio.EchoCanceller.Clockdrift", static_cast<int>(clockdrift),
static_cast<int>(ClockdriftDetector::Level::kNumCategories));
metrics_reported_ = true;
call_counter_ = 0;
ResetMetrics();
} else {
metrics_reported_ = false;
}
if (!initial_update && ++skew_report_timer_ == 60 * kNumBlocksPerSecond) {
RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.EchoCanceller.MaxSkewShiftCount",
skew_shift_count_, 0, kMaxSkewShiftCount,
kMaxSkewShiftCount + 1);
skew_shift_count_ = 0;
skew_report_timer_ = 0;
}
}