Reland "APM: log both applied and recommended input volume stats"
This is a reland of commit 8d7273357d92fab881561d886ce8dfe94e6e2238 Root cause: audioproc_f doesn't call `metrics::Enable()` and therefore the stats reporter crashed when `metrics::HistogramFactoryGetCountsLinear()` returned a nullptr. Bug fix: Added `InputVolumeStatsReporter::cannot_log_stats_`, a const flag that is set to true if any histogram factory returns a nullptr. When true, the class does nothing. This CL also includes other code readability improvements that were not part of the original CL. Original change's description: > APM: log both applied and recommended input volume stats > > This CL replaces the existing `WebRTC.Audio.ApmAnalogGain.*` stats > with `WebRTC.Audio.Apm.AppliedInputVolume.*` and adds the > `WebRTC.Audio.Apm.RecommendedInputVolume.*` stats. > > Bug: webrtc:7494 > Change-Id: I70be710d20b1589fc814cbce3d3329ac1500686f > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280220 > Reviewed-by: Hanna Silen <silen@webrtc.org> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#38468} Bug: webrtc:7494 Change-Id: I8373d16beb06b84f439d2c2274ededea7c5e95b0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280661 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Hanna Silen <silen@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38484}
This commit is contained in:
Alessio Bazzica
WebRTC LUCI CQ
parent
aebba7b468
commit
fbe5d7c3d4
@ -426,9 +426,13 @@ rtc_library("input_volume_stats_reporter") {
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"../../../rtc_base:gtest_prod",
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"../../../rtc_base:logging",
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"../../../rtc_base:safe_minmax",
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"../../../rtc_base:stringutils",
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"../../../system_wrappers:metrics",
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]
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absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
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absl_deps = [
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"//third_party/abseil-cpp/absl/strings",
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"//third_party/abseil-cpp/absl/types:optional",
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]
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}
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rtc_library("input_volume_stats_reporter_unittests") {
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@ -436,7 +440,9 @@ rtc_library("input_volume_stats_reporter_unittests") {
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sources = [ "input_volume_stats_reporter_unittest.cc" ]
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deps = [
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":input_volume_stats_reporter",
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"../../../rtc_base:stringutils",
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"../../../system_wrappers:metrics",
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"../../../test:test_support",
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]
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absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
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}
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@ -12,36 +12,91 @@
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#include <cmath>
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#include "absl/strings/string_view.h"
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#include "rtc_base/logging.h"
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#include "rtc_base/numerics/safe_minmax.h"
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#include "rtc_base/strings/string_builder.h"
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#include "system_wrappers/include/metrics.h"
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namespace webrtc {
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namespace {
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using InputVolumeType = InputVolumeStatsReporter::InputVolumeType;
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constexpr int kFramesIn60Seconds = 6000;
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constexpr int kMinInputVolume = 0;
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constexpr int kMaxInputVolume = 255;
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constexpr int kMaxUpdate = kMaxInputVolume - kMinInputVolume;
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float ComputeAverageUpdate(int sum_updates, int num_updates) {
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int ComputeAverageUpdate(int sum_updates, int num_updates) {
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RTC_DCHECK_GE(sum_updates, 0);
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RTC_DCHECK_LE(sum_updates, kMaxUpdate * kFramesIn60Seconds);
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RTC_DCHECK_GE(num_updates, 0);
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RTC_DCHECK_LE(num_updates, kFramesIn60Seconds);
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if (num_updates == 0) {
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return 0.0f;
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return 0;
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}
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return std::round(static_cast<float>(sum_updates) /
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static_cast<float>(num_updates));
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}
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constexpr absl::string_view MetricNamePrefix(
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InputVolumeType input_volume_type) {
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switch (input_volume_type) {
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case InputVolumeType::kApplied:
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return "WebRTC.Audio.Apm.AppliedInputVolume.";
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case InputVolumeType::kRecommended:
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return "WebRTC.Audio.Apm.RecommendedInputVolume.";
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}
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}
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metrics::Histogram* CreateRateHistogram(InputVolumeType input_volume_type,
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absl::string_view name) {
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char buffer[64];
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rtc::SimpleStringBuilder builder(buffer);
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builder << MetricNamePrefix(input_volume_type) << name;
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return metrics::HistogramFactoryGetCountsLinear(/*name=*/builder.str(),
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/*min=*/1,
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/*max=*/kFramesIn60Seconds,
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/*bucket_count=*/50);
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}
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metrics::Histogram* CreateAverageHistogram(InputVolumeType input_volume_type,
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absl::string_view name) {
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char buffer[64];
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rtc::SimpleStringBuilder builder(buffer);
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builder << MetricNamePrefix(input_volume_type) << name;
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return metrics::HistogramFactoryGetCountsLinear(/*name=*/builder.str(),
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/*min=*/1,
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/*max=*/kMaxUpdate,
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/*bucket_count=*/50);
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}
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} // namespace
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InputVolumeStatsReporter::InputVolumeStatsReporter() = default;
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InputVolumeStatsReporter::InputVolumeStatsReporter(InputVolumeType type)
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: histograms_(
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{.decrease_rate = CreateRateHistogram(type, "DecreaseRate"),
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.decrease_average = CreateAverageHistogram(type, "DecreaseAverage"),
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.increase_rate = CreateRateHistogram(type, "IncreaseRate"),
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.increase_average = CreateAverageHistogram(type, "IncreaseAverage"),
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.update_rate = CreateRateHistogram(type, "UpdateRate"),
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.update_average = CreateAverageHistogram(type, "UpdateAverage")}),
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cannot_log_stats_(!histograms_.AllPointersSet()) {
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if (cannot_log_stats_) {
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RTC_LOG(LS_WARNING) << "Will not log any `" << MetricNamePrefix(type)
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<< "*` histogram stats.";
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}
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}
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InputVolumeStatsReporter::~InputVolumeStatsReporter() = default;
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void InputVolumeStatsReporter::UpdateStatistics(int input_volume) {
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if (cannot_log_stats_) {
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// Since the stats cannot be logged, do not bother updating them.
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return;
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}
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RTC_DCHECK_GE(input_volume, kMinInputVolume);
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RTC_DCHECK_LE(input_volume, kMaxInputVolume);
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if (previous_input_volume_.has_value() &&
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@ -65,56 +120,31 @@ void InputVolumeStatsReporter::UpdateStatistics(int input_volume) {
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}
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void InputVolumeStatsReporter::LogVolumeUpdateStats() const {
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const float average_decrease = ComputeAverageUpdate(
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volume_update_stats_.sum_decreases, volume_update_stats_.num_decreases);
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const float average_increase = ComputeAverageUpdate(
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volume_update_stats_.sum_increases, volume_update_stats_.num_increases);
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const int num_updates =
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volume_update_stats_.num_decreases + volume_update_stats_.num_increases;
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const float average_update = ComputeAverageUpdate(
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volume_update_stats_.sum_decreases + volume_update_stats_.sum_increases,
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num_updates);
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RTC_HISTOGRAM_COUNTS_LINEAR(
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/*name=*/"WebRTC.Audio.ApmAnalogGainDecreaseRate",
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/*sample=*/volume_update_stats_.num_decreases,
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/*min=*/1,
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/*max=*/kFramesIn60Seconds,
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/*bucket_count=*/50);
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// Decrease rate and average.
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metrics::HistogramAdd(histograms_.decrease_rate,
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volume_update_stats_.num_decreases);
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if (volume_update_stats_.num_decreases > 0) {
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RTC_HISTOGRAM_COUNTS_LINEAR(
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/*name=*/"WebRTC.Audio.ApmAnalogGainDecreaseAverage",
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/*sample=*/average_decrease,
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/*min=*/1,
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/*max=*/kMaxUpdate,
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/*bucket_count=*/50);
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int average_decrease = ComputeAverageUpdate(
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volume_update_stats_.sum_decreases, volume_update_stats_.num_decreases);
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metrics::HistogramAdd(histograms_.decrease_average, average_decrease);
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}
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RTC_HISTOGRAM_COUNTS_LINEAR(
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/*name=*/"WebRTC.Audio.ApmAnalogGainIncreaseRate",
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/*sample=*/volume_update_stats_.num_increases,
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/*min=*/1,
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/*max=*/kFramesIn60Seconds,
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/*bucket_count=*/50);
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// Increase rate and average.
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metrics::HistogramAdd(histograms_.increase_rate,
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volume_update_stats_.num_increases);
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if (volume_update_stats_.num_increases > 0) {
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RTC_HISTOGRAM_COUNTS_LINEAR(
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/*name=*/"WebRTC.Audio.ApmAnalogGainIncreaseAverage",
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/*sample=*/average_increase,
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/*min=*/1,
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/*max=*/kMaxUpdate,
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/*bucket_count=*/50);
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int average_increase = ComputeAverageUpdate(
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volume_update_stats_.sum_increases, volume_update_stats_.num_increases);
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metrics::HistogramAdd(histograms_.increase_average, average_increase);
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}
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RTC_HISTOGRAM_COUNTS_LINEAR(
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/*name=*/"WebRTC.Audio.ApmAnalogGainUpdateRate",
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/*sample=*/num_updates,
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/*min=*/1,
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/*max=*/kFramesIn60Seconds,
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/*bucket_count=*/50);
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// Update rate and average.
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int num_updates =
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volume_update_stats_.num_decreases + volume_update_stats_.num_increases;
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metrics::HistogramAdd(histograms_.update_rate, num_updates);
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if (num_updates > 0) {
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RTC_HISTOGRAM_COUNTS_LINEAR(
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/*name=*/"WebRTC.Audio.ApmAnalogGainUpdateAverage",
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/*sample=*/average_update,
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/*min=*/1,
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/*max=*/kMaxUpdate,
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/*bucket_count=*/50);
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int average_update = ComputeAverageUpdate(
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volume_update_stats_.sum_decreases + volume_update_stats_.sum_increases,
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num_updates);
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metrics::HistogramAdd(histograms_.update_average, average_update);
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}
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}
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@ -13,6 +13,7 @@
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#include "absl/types/optional.h"
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#include "rtc_base/gtest_prod_util.h"
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#include "system_wrappers/include/metrics.h"
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namespace webrtc {
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@ -21,7 +22,12 @@ namespace webrtc {
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// the statistics into a histogram.
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class InputVolumeStatsReporter {
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public:
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InputVolumeStatsReporter();
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enum class InputVolumeType {
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kApplied = 0,
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kRecommended = 1,
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};
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explicit InputVolumeStatsReporter(InputVolumeType input_volume_type);
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InputVolumeStatsReporter(const InputVolumeStatsReporter&) = delete;
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InputVolumeStatsReporter operator=(const InputVolumeStatsReporter&) = delete;
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~InputVolumeStatsReporter();
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@ -57,6 +63,23 @@ class InputVolumeStatsReporter {
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// Computes aggregate stat and logs them into a histogram.
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void LogVolumeUpdateStats() const;
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// Histograms.
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struct Histograms {
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metrics::Histogram* const decrease_rate;
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metrics::Histogram* const decrease_average;
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metrics::Histogram* const increase_rate;
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metrics::Histogram* const increase_average;
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metrics::Histogram* const update_rate;
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metrics::Histogram* const update_average;
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bool AllPointersSet() const {
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return !!decrease_rate && !!decrease_average && !!increase_rate &&
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!!increase_average && !!update_rate && !!update_average;
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}
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} histograms_;
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// True if the stats cannot be logged.
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const bool cannot_log_stats_;
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int log_volume_update_stats_counter_ = 0;
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absl::optional<int> previous_input_volume_ = absl::nullopt;
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};
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@ -10,24 +10,71 @@
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#include "modules/audio_processing/agc2/input_volume_stats_reporter.h"
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#include "absl/strings/string_view.h"
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#include "rtc_base/strings/string_builder.h"
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#include "system_wrappers/include/metrics.h"
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#include "test/gmock.h"
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namespace webrtc {
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namespace {
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using InputVolumeType = InputVolumeStatsReporter::InputVolumeType;
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constexpr int kFramesIn60Seconds = 6000;
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class InputVolumeStatsReporterTest : public ::testing::Test {
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constexpr absl::string_view kLabelPrefix = "WebRTC.Audio.Apm.";
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class InputVolumeStatsReporterTest
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: public ::testing::TestWithParam<InputVolumeType> {
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public:
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InputVolumeStatsReporterTest() {}
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InputVolumeStatsReporterTest() { metrics::Reset(); }
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protected:
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void SetUp() override { metrics::Reset(); }
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InputVolumeType InputVolumeType() const { return GetParam(); }
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std::string DecreaseRateLabel() const {
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return (rtc::StringBuilder(kLabelPrefix)
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<< VolumeTypeLabel() << "DecreaseRate")
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.str();
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}
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std::string DecreaseAverageLabel() const {
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return (rtc::StringBuilder(kLabelPrefix)
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<< VolumeTypeLabel() << "DecreaseAverage")
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.str();
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}
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std::string IncreaseRateLabel() const {
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return (rtc::StringBuilder(kLabelPrefix)
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<< VolumeTypeLabel() << "IncreaseRate")
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.str();
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}
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std::string IncreaseAverageLabel() const {
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return (rtc::StringBuilder(kLabelPrefix)
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<< VolumeTypeLabel() << "IncreaseAverage")
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.str();
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}
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std::string UpdateRateLabel() const {
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return (rtc::StringBuilder(kLabelPrefix)
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<< VolumeTypeLabel() << "UpdateRate")
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.str();
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}
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std::string UpdateAverageLabel() const {
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return (rtc::StringBuilder(kLabelPrefix)
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<< VolumeTypeLabel() << "UpdateAverage")
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.str();
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}
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private:
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absl::string_view VolumeTypeLabel() const {
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switch (InputVolumeType()) {
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case InputVolumeType::kApplied:
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return "AppliedInputVolume.";
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case InputVolumeType::kRecommended:
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return "RecommendedInputVolume.";
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}
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}
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};
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TEST_F(InputVolumeStatsReporterTest, CheckLogVolumeUpdateStatsEmpty) {
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InputVolumeStatsReporter stats_reporter;
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TEST_P(InputVolumeStatsReporterTest, CheckLogVolumeUpdateStatsEmpty) {
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InputVolumeStatsReporter stats_reporter(InputVolumeType());
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constexpr int kInputVolume = 10;
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stats_reporter.UpdateStatistics(kInputVolume);
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// Update almost until the periodic logging and reset.
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@ -35,25 +82,22 @@ TEST_F(InputVolumeStatsReporterTest, CheckLogVolumeUpdateStatsEmpty) {
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stats_reporter.UpdateStatistics(kInputVolume + 2);
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stats_reporter.UpdateStatistics(kInputVolume);
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}
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EXPECT_METRIC_THAT(metrics::Samples("WebRTC.Audio.ApmAnalogGainUpdateRate"),
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EXPECT_METRIC_THAT(metrics::Samples(UpdateRateLabel()),
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::testing::ElementsAre());
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EXPECT_METRIC_THAT(metrics::Samples("WebRTC.Audio.ApmAnalogGainDecreaseRate"),
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EXPECT_METRIC_THAT(metrics::Samples(DecreaseRateLabel()),
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::testing::ElementsAre());
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EXPECT_METRIC_THAT(metrics::Samples("WebRTC.Audio.ApmAnalogGainIncreaseRate"),
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EXPECT_METRIC_THAT(metrics::Samples(IncreaseRateLabel()),
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::testing::ElementsAre());
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EXPECT_METRIC_THAT(metrics::Samples(UpdateAverageLabel()),
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::testing::ElementsAre());
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EXPECT_METRIC_THAT(metrics::Samples(DecreaseAverageLabel()),
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::testing::ElementsAre());
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EXPECT_METRIC_THAT(metrics::Samples(IncreaseAverageLabel()),
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::testing::ElementsAre());
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EXPECT_METRIC_THAT(
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metrics::Samples("WebRTC.Audio.ApmAnalogGainUpdateAverage"),
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::testing::ElementsAre());
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EXPECT_METRIC_THAT(
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metrics::Samples("WebRTC.Audio.ApmAnalogGainDecreaseAverage"),
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::testing::ElementsAre());
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EXPECT_METRIC_THAT(
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metrics::Samples("WebRTC.Audio.ApmAnalogGainIncreaseAverage"),
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::testing::ElementsAre());
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}
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TEST_F(InputVolumeStatsReporterTest, CheckLogVolumeUpdateStatsNotEmpty) {
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InputVolumeStatsReporter stats_reporter;
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TEST_P(InputVolumeStatsReporterTest, CheckLogVolumeUpdateStatsNotEmpty) {
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InputVolumeStatsReporter stats_reporter(InputVolumeType());
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constexpr int kInputVolume = 10;
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stats_reporter.UpdateStatistics(kInputVolume);
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// Update until periodic logging.
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@ -67,30 +111,30 @@ TEST_F(InputVolumeStatsReporterTest, CheckLogVolumeUpdateStatsNotEmpty) {
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stats_reporter.UpdateStatistics(kInputVolume);
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}
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EXPECT_METRIC_THAT(
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metrics::Samples("WebRTC.Audio.ApmAnalogGainUpdateRate"),
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metrics::Samples(UpdateRateLabel()),
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::testing::ElementsAre(::testing::Pair(kFramesIn60Seconds - 1, 1),
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::testing::Pair(kFramesIn60Seconds, 1)));
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EXPECT_METRIC_THAT(
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metrics::Samples("WebRTC.Audio.ApmAnalogGainDecreaseRate"),
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metrics::Samples(DecreaseRateLabel()),
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::testing::ElementsAre(::testing::Pair(kFramesIn60Seconds / 2 - 1, 1),
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::testing::Pair(kFramesIn60Seconds / 2, 1)));
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EXPECT_METRIC_THAT(
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metrics::Samples("WebRTC.Audio.ApmAnalogGainIncreaseRate"),
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metrics::Samples(IncreaseRateLabel()),
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::testing::ElementsAre(::testing::Pair(kFramesIn60Seconds / 2, 2)));
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EXPECT_METRIC_THAT(
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metrics::Samples("WebRTC.Audio.ApmAnalogGainUpdateAverage"),
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metrics::Samples(UpdateAverageLabel()),
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::testing::ElementsAre(::testing::Pair(2, 1), ::testing::Pair(3, 1)));
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EXPECT_METRIC_THAT(
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metrics::Samples("WebRTC.Audio.ApmAnalogGainDecreaseAverage"),
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metrics::Samples(DecreaseAverageLabel()),
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::testing::ElementsAre(::testing::Pair(2, 1), ::testing::Pair(3, 1)));
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EXPECT_METRIC_THAT(
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metrics::Samples("WebRTC.Audio.ApmAnalogGainIncreaseAverage"),
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metrics::Samples(IncreaseAverageLabel()),
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::testing::ElementsAre(::testing::Pair(2, 1), ::testing::Pair(3, 1)));
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}
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} // namespace
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TEST_F(InputVolumeStatsReporterTest, CheckVolumeUpdateStatsForEmptyStats) {
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InputVolumeStatsReporter stats_reporter;
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TEST_P(InputVolumeStatsReporterTest, CheckVolumeUpdateStatsForEmptyStats) {
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InputVolumeStatsReporter stats_reporter(InputVolumeType());
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const auto& update_stats = stats_reporter.volume_update_stats();
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EXPECT_EQ(update_stats.num_decreases, 0);
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EXPECT_EQ(update_stats.sum_decreases, 0);
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@ -98,10 +142,10 @@ TEST_F(InputVolumeStatsReporterTest, CheckVolumeUpdateStatsForEmptyStats) {
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EXPECT_EQ(update_stats.sum_increases, 0);
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}
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|
||||
TEST_F(InputVolumeStatsReporterTest,
|
||||
TEST_P(InputVolumeStatsReporterTest,
|
||||
CheckVolumeUpdateStatsAfterNoVolumeChange) {
|
||||
constexpr int kInputVolume = 10;
|
||||
InputVolumeStatsReporter stats_reporter;
|
||||
InputVolumeStatsReporter stats_reporter(InputVolumeType());
|
||||
stats_reporter.UpdateStatistics(kInputVolume);
|
||||
stats_reporter.UpdateStatistics(kInputVolume);
|
||||
stats_reporter.UpdateStatistics(kInputVolume);
|
||||
@ -112,10 +156,10 @@ TEST_F(InputVolumeStatsReporterTest,
|
||||
EXPECT_EQ(update_stats.sum_increases, 0);
|
||||
}
|
||||
|
||||
TEST_F(InputVolumeStatsReporterTest,
|
||||
TEST_P(InputVolumeStatsReporterTest,
|
||||
CheckVolumeUpdateStatsAfterVolumeIncrease) {
|
||||
constexpr int kInputVolume = 10;
|
||||
InputVolumeStatsReporter stats_reporter;
|
||||
InputVolumeStatsReporter stats_reporter(InputVolumeType());
|
||||
stats_reporter.UpdateStatistics(kInputVolume);
|
||||
stats_reporter.UpdateStatistics(kInputVolume + 4);
|
||||
stats_reporter.UpdateStatistics(kInputVolume + 5);
|
||||
@ -126,10 +170,10 @@ TEST_F(InputVolumeStatsReporterTest,
|
||||
EXPECT_EQ(update_stats.sum_increases, 5);
|
||||
}
|
||||
|
||||
TEST_F(InputVolumeStatsReporterTest,
|
||||
TEST_P(InputVolumeStatsReporterTest,
|
||||
CheckVolumeUpdateStatsAfterVolumeDecrease) {
|
||||
constexpr int kInputVolume = 10;
|
||||
InputVolumeStatsReporter stats_reporter;
|
||||
InputVolumeStatsReporter stats_reporter(InputVolumeType());
|
||||
stats_reporter.UpdateStatistics(kInputVolume);
|
||||
stats_reporter.UpdateStatistics(kInputVolume - 4);
|
||||
stats_reporter.UpdateStatistics(kInputVolume - 5);
|
||||
@ -140,8 +184,8 @@ TEST_F(InputVolumeStatsReporterTest,
|
||||
EXPECT_EQ(stats_update.sum_increases, 0);
|
||||
}
|
||||
|
||||
TEST_F(InputVolumeStatsReporterTest, CheckVolumeUpdateStatsAfterReset) {
|
||||
InputVolumeStatsReporter stats_reporter;
|
||||
TEST_P(InputVolumeStatsReporterTest, CheckVolumeUpdateStatsAfterReset) {
|
||||
InputVolumeStatsReporter stats_reporter(InputVolumeType());
|
||||
constexpr int kInputVolume = 10;
|
||||
stats_reporter.UpdateStatistics(kInputVolume);
|
||||
// Update until the periodic reset.
|
||||
@ -169,4 +213,9 @@ TEST_F(InputVolumeStatsReporterTest, CheckVolumeUpdateStatsAfterReset) {
|
||||
EXPECT_EQ(stats_after_reset.sum_increases, 3);
|
||||
}
|
||||
|
||||
INSTANTIATE_TEST_SUITE_P(,
|
||||
InputVolumeStatsReporterTest,
|
||||
::testing::Values(InputVolumeType::kApplied,
|
||||
InputVolumeType::kRecommended));
|
||||
|
||||
} // namespace webrtc
|
||||
|
||||
@ -291,7 +291,11 @@ AudioProcessingImpl::AudioProcessingImpl(
|
||||
MinimizeProcessingForUnusedOutput(),
|
||||
field_trial::IsEnabled("WebRTC-TransientSuppressorForcedOff")),
|
||||
capture_(),
|
||||
capture_nonlocked_() {
|
||||
capture_nonlocked_(),
|
||||
applied_input_volume_stats_reporter_(
|
||||
InputVolumeStatsReporter::InputVolumeType::kApplied),
|
||||
recommended_input_volume_stats_reporter_(
|
||||
InputVolumeStatsReporter::InputVolumeType::kRecommended) {
|
||||
RTC_LOG(LS_INFO) << "Injected APM submodules:"
|
||||
"\nEcho control factory: "
|
||||
<< !!echo_control_factory_
|
||||
@ -1361,6 +1365,10 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() {
|
||||
stats_reporter_.UpdateStatistics(capture_.stats);
|
||||
|
||||
UpdateRecommendedInputVolumeLocked();
|
||||
if (capture_.recommended_input_volume.has_value()) {
|
||||
recommended_input_volume_stats_reporter_.UpdateStatistics(
|
||||
*capture_.recommended_input_volume);
|
||||
}
|
||||
|
||||
if (submodules_.capture_levels_adjuster) {
|
||||
submodules_.capture_levels_adjuster->ApplyPostLevelAdjustment(
|
||||
|
||||
@ -541,6 +541,8 @@ class AudioProcessingImpl : public AudioProcessing {
|
||||
|
||||
InputVolumeStatsReporter applied_input_volume_stats_reporter_
|
||||
RTC_GUARDED_BY(mutex_capture_);
|
||||
InputVolumeStatsReporter recommended_input_volume_stats_reporter_
|
||||
RTC_GUARDED_BY(mutex_capture_);
|
||||
|
||||
// Lock protection not needed.
|
||||
std::unique_ptr<
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user