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}
diff --git a/modules/audio_processing/agc2/BUILD.gn b/modules/audio_processing/agc2/BUILD.gn
index 976f96e..b7a4030 100644
--- a/modules/audio_processing/agc2/BUILD.gn
+++ b/modules/audio_processing/agc2/BUILD.gn
@@ -401,7 +401,9 @@
sources = [ "input_volume_stats_reporter_unittest.cc" ]
deps = [
":input_volume_stats_reporter",
+ "../../../rtc_base:stringutils",
"../../../system_wrappers:metrics",
"../../../test:test_support",
]
+ absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
}
diff --git a/modules/audio_processing/agc2/input_volume_stats_reporter.cc b/modules/audio_processing/agc2/input_volume_stats_reporter.cc
index 4a8b016..6d70186 100644
--- a/modules/audio_processing/agc2/input_volume_stats_reporter.cc
+++ b/modules/audio_processing/agc2/input_volume_stats_reporter.cc
@@ -19,6 +19,8 @@
namespace webrtc {
namespace {
+using InputVolumeType = InputVolumeStatsReporter::InputVolumeType;
+
constexpr int kFramesIn60Seconds = 6000;
constexpr int kMinInputVolume = 0;
constexpr int kMaxInputVolume = 255;
@@ -35,9 +37,102 @@
return std::round(static_cast<float>(sum_updates) /
static_cast<float>(num_updates));
}
+
+metrics::Histogram* CreateRateHistogram(absl::string_view name) {
+ return metrics::HistogramFactoryGetCounts(
+ name, /*min=*/1, /*max=*/kFramesIn60Seconds, /*bucket_count=*/50);
+}
+
+metrics::Histogram* CreateAverageHistogram(absl::string_view name) {
+ return metrics::HistogramFactoryGetCounts(name, /*min=*/1, /*max=*/kMaxUpdate,
+ /*bucket_count=*/50);
+}
+
+metrics::Histogram* CreateDecreaseRateHistogram(
+ InputVolumeType input_volume_type) {
+ switch (input_volume_type) {
+ case InputVolumeType::kApplied:
+ return CreateRateHistogram(
+ "WebRTC.Audio.Apm.AppliedInputVolume.DecreaseRate");
+ case InputVolumeType::kRecommended:
+ return CreateRateHistogram(
+ "WebRTC.Audio.Apm.RecommendedInputVolume.DecreaseRate");
+ }
+}
+
+metrics::Histogram* CreateDecreaseAverageHistogram(
+ InputVolumeType input_volume_type) {
+ switch (input_volume_type) {
+ case InputVolumeType::kApplied:
+ return CreateAverageHistogram(
+ "WebRTC.Audio.Apm.AppliedInputVolume.DecreaseAverage");
+ case InputVolumeType::kRecommended:
+ return CreateAverageHistogram(
+ "WebRTC.Audio.Apm.RecommendedInputVolume.DecreaseAverage");
+ }
+}
+
+metrics::Histogram* CreateIncreaseRateHistogram(
+ InputVolumeType input_volume_type) {
+ switch (input_volume_type) {
+ case InputVolumeType::kApplied:
+ return CreateRateHistogram(
+ "WebRTC.Audio.Apm.AppliedInputVolume.IncreaseRate");
+ case InputVolumeType::kRecommended:
+ return CreateRateHistogram(
+ "WebRTC.Audio.Apm.RecommendedInputVolume.IncreaseRate");
+ }
+}
+
+metrics::Histogram* CreateIncreaseAverageHistogram(
+ InputVolumeType input_volume_type) {
+ switch (input_volume_type) {
+ case InputVolumeType::kApplied:
+ return CreateAverageHistogram(
+ "WebRTC.Audio.Apm.AppliedInputVolume.IncreaseAverage");
+ case InputVolumeType::kRecommended:
+ return CreateAverageHistogram(
+ "WebRTC.Audio.Apm.RecommendedInputVolume.IncreaseAverage");
+ }
+}
+
+metrics::Histogram* CreateUpdateRateHistogram(
+ InputVolumeType input_volume_type) {
+ switch (input_volume_type) {
+ case InputVolumeType::kApplied:
+ return CreateRateHistogram(
+ "WebRTC.Audio.Apm.AppliedInputVolume.UpdateRate");
+ case InputVolumeType::kRecommended:
+ return CreateRateHistogram(
+ "WebRTC.Audio.Apm.RecommendedInputVolume.UpdateRate");
+ }
+}
+
+metrics::Histogram* CreateUpdateAverageHistogram(
+ InputVolumeType input_volume_type) {
+ switch (input_volume_type) {
+ case InputVolumeType::kApplied:
+ return CreateAverageHistogram(
+ "WebRTC.Audio.Apm.AppliedInputVolume.UpdateAverage");
+ case InputVolumeType::kRecommended:
+ return CreateAverageHistogram(
+ "WebRTC.Audio.Apm.RecommendedInputVolume.UpdateAverage");
+ }
+}
+
} // namespace
-InputVolumeStatsReporter::InputVolumeStatsReporter() = default;
+InputVolumeStatsReporter::InputVolumeStatsReporter(
+ InputVolumeType input_volume_type)
+ : decrease_rate_histogram_(CreateDecreaseRateHistogram(input_volume_type)),
+ decrease_average_histogram_(
+ CreateDecreaseAverageHistogram(input_volume_type)),
+ increase_rate_histogram_(CreateIncreaseRateHistogram(input_volume_type)),
+ increase_average_histogram_(
+ CreateIncreaseAverageHistogram(input_volume_type)),
+ update_rate_histogram_(CreateUpdateRateHistogram(input_volume_type)),
+ update_average_histogram_(
+ CreateUpdateAverageHistogram(input_volume_type)) {}
InputVolumeStatsReporter::~InputVolumeStatsReporter() = default;
@@ -74,47 +169,19 @@
const float average_update = ComputeAverageUpdate(
volume_update_stats_.sum_decreases + volume_update_stats_.sum_increases,
num_updates);
- RTC_HISTOGRAM_COUNTS_LINEAR(
- /*name=*/"WebRTC.Audio.ApmAnalogGainDecreaseRate",
- /*sample=*/volume_update_stats_.num_decreases,
- /*min=*/1,
- /*max=*/kFramesIn60Seconds,
- /*bucket_count=*/50);
+ metrics::HistogramAdd(decrease_rate_histogram_,
+ volume_update_stats_.num_decreases);
if (volume_update_stats_.num_decreases > 0) {
- RTC_HISTOGRAM_COUNTS_LINEAR(
- /*name=*/"WebRTC.Audio.ApmAnalogGainDecreaseAverage",
- /*sample=*/average_decrease,
- /*min=*/1,
- /*max=*/kMaxUpdate,
- /*bucket_count=*/50);
+ metrics::HistogramAdd(decrease_average_histogram_, average_decrease);
}
- RTC_HISTOGRAM_COUNTS_LINEAR(
- /*name=*/"WebRTC.Audio.ApmAnalogGainIncreaseRate",
- /*sample=*/volume_update_stats_.num_increases,
- /*min=*/1,
- /*max=*/kFramesIn60Seconds,
- /*bucket_count=*/50);
+ metrics::HistogramAdd(increase_rate_histogram_,
+ volume_update_stats_.num_increases);
if (volume_update_stats_.num_increases > 0) {
- RTC_HISTOGRAM_COUNTS_LINEAR(
- /*name=*/"WebRTC.Audio.ApmAnalogGainIncreaseAverage",
- /*sample=*/average_increase,
- /*min=*/1,
- /*max=*/kMaxUpdate,
- /*bucket_count=*/50);
+ metrics::HistogramAdd(increase_average_histogram_, average_increase);
}
- RTC_HISTOGRAM_COUNTS_LINEAR(
- /*name=*/"WebRTC.Audio.ApmAnalogGainUpdateRate",
- /*sample=*/num_updates,
- /*min=*/1,
- /*max=*/kFramesIn60Seconds,
- /*bucket_count=*/50);
+ metrics::HistogramAdd(update_rate_histogram_, num_updates);
if (num_updates > 0) {
- RTC_HISTOGRAM_COUNTS_LINEAR(
- /*name=*/"WebRTC.Audio.ApmAnalogGainUpdateAverage",
- /*sample=*/average_update,
- /*min=*/1,
- /*max=*/kMaxUpdate,
- /*bucket_count=*/50);
+ metrics::HistogramAdd(update_average_histogram_, average_update);
}
}
diff --git a/modules/audio_processing/agc2/input_volume_stats_reporter.h b/modules/audio_processing/agc2/input_volume_stats_reporter.h
index 0040754..757e6ff 100644
--- a/modules/audio_processing/agc2/input_volume_stats_reporter.h
+++ b/modules/audio_processing/agc2/input_volume_stats_reporter.h
@@ -13,6 +13,7 @@
#include "absl/types/optional.h"
#include "rtc_base/gtest_prod_util.h"
+#include "system_wrappers/include/metrics.h"
namespace webrtc {
@@ -21,7 +22,12 @@
// the statistics into a histogram.
class InputVolumeStatsReporter {
public:
- InputVolumeStatsReporter();
+ enum class InputVolumeType {
+ kApplied = 0,
+ kRecommended = 1,
+ };
+
+ explicit InputVolumeStatsReporter(InputVolumeType input_volume_type);
InputVolumeStatsReporter(const InputVolumeStatsReporter&) = delete;
InputVolumeStatsReporter operator=(const InputVolumeStatsReporter&) = delete;
~InputVolumeStatsReporter();
@@ -57,6 +63,14 @@
// Computes aggregate stat and logs them into a histogram.
void LogVolumeUpdateStats() const;
+ // Histograms.
+ metrics::Histogram* decrease_rate_histogram_;
+ metrics::Histogram* decrease_average_histogram_;
+ metrics::Histogram* increase_rate_histogram_;
+ metrics::Histogram* increase_average_histogram_;
+ metrics::Histogram* update_rate_histogram_;
+ metrics::Histogram* update_average_histogram_;
+
int log_volume_update_stats_counter_ = 0;
absl::optional<int> previous_input_volume_ = absl::nullopt;
};
diff --git a/modules/audio_processing/agc2/input_volume_stats_reporter_unittest.cc b/modules/audio_processing/agc2/input_volume_stats_reporter_unittest.cc
index f8cd010..a3e2cca 100644
--- a/modules/audio_processing/agc2/input_volume_stats_reporter_unittest.cc
+++ b/modules/audio_processing/agc2/input_volume_stats_reporter_unittest.cc
@@ -10,24 +10,71 @@
#include "modules/audio_processing/agc2/input_volume_stats_reporter.h"
+#include "absl/strings/string_view.h"
+#include "rtc_base/strings/string_builder.h"
#include "system_wrappers/include/metrics.h"
#include "test/gmock.h"
namespace webrtc {
namespace {
+using InputVolumeType = InputVolumeStatsReporter::InputVolumeType;
+
constexpr int kFramesIn60Seconds = 6000;
-class InputVolumeStatsReporterTest : public ::testing::Test {
+constexpr absl::string_view kLabelPrefix = "WebRTC.Audio.Apm.";
+
+class InputVolumeStatsReporterTest
+ : public ::testing::TestWithParam<InputVolumeType> {
public:
- InputVolumeStatsReporterTest() {}
+ InputVolumeStatsReporterTest() { metrics::Reset(); }
protected:
- void SetUp() override { metrics::Reset(); }
+ InputVolumeType InputVolumeType() const { return GetParam(); }
+ std::string DecreaseRateLabel() const {
+ return (rtc::StringBuilder(kLabelPrefix)
+ << VolumeTypeLabel() << "DecreaseRate")
+ .str();
+ }
+ std::string DecreaseAverageLabel() const {
+ return (rtc::StringBuilder(kLabelPrefix)
+ << VolumeTypeLabel() << "DecreaseAverage")
+ .str();
+ }
+ std::string IncreaseRateLabel() const {
+ return (rtc::StringBuilder(kLabelPrefix)
+ << VolumeTypeLabel() << "IncreaseRate")
+ .str();
+ }
+ std::string IncreaseAverageLabel() const {
+ return (rtc::StringBuilder(kLabelPrefix)
+ << VolumeTypeLabel() << "IncreaseAverage")
+ .str();
+ }
+ std::string UpdateRateLabel() const {
+ return (rtc::StringBuilder(kLabelPrefix)
+ << VolumeTypeLabel() << "UpdateRate")
+ .str();
+ }
+ std::string UpdateAverageLabel() const {
+ return (rtc::StringBuilder(kLabelPrefix)
+ << VolumeTypeLabel() << "UpdateAverage")
+ .str();
+ }
+
+ private:
+ absl::string_view VolumeTypeLabel() const {
+ switch (InputVolumeType()) {
+ case InputVolumeType::kApplied:
+ return "AppliedInputVolume.";
+ case InputVolumeType::kRecommended:
+ return "RecommendedInputVolume.";
+ }
+ }
};
-TEST_F(InputVolumeStatsReporterTest, CheckLogVolumeUpdateStatsEmpty) {
- InputVolumeStatsReporter stats_reporter;
+TEST_P(InputVolumeStatsReporterTest, CheckLogVolumeUpdateStatsEmpty) {
+ InputVolumeStatsReporter stats_reporter(InputVolumeType());
constexpr int kInputVolume = 10;
stats_reporter.UpdateStatistics(kInputVolume);
// Update almost until the periodic logging and reset.
@@ -35,25 +82,22 @@
stats_reporter.UpdateStatistics(kInputVolume + 2);
stats_reporter.UpdateStatistics(kInputVolume);
}
- EXPECT_METRIC_THAT(metrics::Samples("WebRTC.Audio.ApmAnalogGainUpdateRate"),
+ EXPECT_METRIC_THAT(metrics::Samples(UpdateRateLabel()),
::testing::ElementsAre());
- EXPECT_METRIC_THAT(metrics::Samples("WebRTC.Audio.ApmAnalogGainDecreaseRate"),
+ EXPECT_METRIC_THAT(metrics::Samples(DecreaseRateLabel()),
::testing::ElementsAre());
- EXPECT_METRIC_THAT(metrics::Samples("WebRTC.Audio.ApmAnalogGainIncreaseRate"),
+ EXPECT_METRIC_THAT(metrics::Samples(IncreaseRateLabel()),
::testing::ElementsAre());
- EXPECT_METRIC_THAT(
- metrics::Samples("WebRTC.Audio.ApmAnalogGainUpdateAverage"),
- ::testing::ElementsAre());
- EXPECT_METRIC_THAT(
- metrics::Samples("WebRTC.Audio.ApmAnalogGainDecreaseAverage"),
- ::testing::ElementsAre());
- EXPECT_METRIC_THAT(
- metrics::Samples("WebRTC.Audio.ApmAnalogGainIncreaseAverage"),
- ::testing::ElementsAre());
+ EXPECT_METRIC_THAT(metrics::Samples(UpdateAverageLabel()),
+ ::testing::ElementsAre());
+ EXPECT_METRIC_THAT(metrics::Samples(DecreaseAverageLabel()),
+ ::testing::ElementsAre());
+ EXPECT_METRIC_THAT(metrics::Samples(IncreaseAverageLabel()),
+ ::testing::ElementsAre());
}
-TEST_F(InputVolumeStatsReporterTest, CheckLogVolumeUpdateStatsNotEmpty) {
- InputVolumeStatsReporter stats_reporter;
+TEST_P(InputVolumeStatsReporterTest, CheckLogVolumeUpdateStatsNotEmpty) {
+ InputVolumeStatsReporter stats_reporter(InputVolumeType());
constexpr int kInputVolume = 10;
stats_reporter.UpdateStatistics(kInputVolume);
// Update until periodic logging.
@@ -67,30 +111,30 @@
stats_reporter.UpdateStatistics(kInputVolume);
}
EXPECT_METRIC_THAT(
- metrics::Samples("WebRTC.Audio.ApmAnalogGainUpdateRate"),
+ metrics::Samples(UpdateRateLabel()),
::testing::ElementsAre(::testing::Pair(kFramesIn60Seconds - 1, 1),
::testing::Pair(kFramesIn60Seconds, 1)));
EXPECT_METRIC_THAT(
- metrics::Samples("WebRTC.Audio.ApmAnalogGainDecreaseRate"),
+ metrics::Samples(DecreaseRateLabel()),
::testing::ElementsAre(::testing::Pair(kFramesIn60Seconds / 2 - 1, 1),
::testing::Pair(kFramesIn60Seconds / 2, 1)));
EXPECT_METRIC_THAT(
- metrics::Samples("WebRTC.Audio.ApmAnalogGainIncreaseRate"),
+ metrics::Samples(IncreaseRateLabel()),
::testing::ElementsAre(::testing::Pair(kFramesIn60Seconds / 2, 2)));
EXPECT_METRIC_THAT(
- metrics::Samples("WebRTC.Audio.ApmAnalogGainUpdateAverage"),
+ metrics::Samples(UpdateAverageLabel()),
::testing::ElementsAre(::testing::Pair(2, 1), ::testing::Pair(3, 1)));
EXPECT_METRIC_THAT(
- metrics::Samples("WebRTC.Audio.ApmAnalogGainDecreaseAverage"),
+ metrics::Samples(DecreaseAverageLabel()),
::testing::ElementsAre(::testing::Pair(2, 1), ::testing::Pair(3, 1)));
EXPECT_METRIC_THAT(
- metrics::Samples("WebRTC.Audio.ApmAnalogGainIncreaseAverage"),
+ metrics::Samples(IncreaseAverageLabel()),
::testing::ElementsAre(::testing::Pair(2, 1), ::testing::Pair(3, 1)));
}
} // namespace
-TEST_F(InputVolumeStatsReporterTest, CheckVolumeUpdateStatsForEmptyStats) {
- InputVolumeStatsReporter stats_reporter;
+TEST_P(InputVolumeStatsReporterTest, CheckVolumeUpdateStatsForEmptyStats) {
+ InputVolumeStatsReporter stats_reporter(InputVolumeType());
const auto& update_stats = stats_reporter.volume_update_stats();
EXPECT_EQ(update_stats.num_decreases, 0);
EXPECT_EQ(update_stats.sum_decreases, 0);
@@ -98,10 +142,10 @@
EXPECT_EQ(update_stats.sum_increases, 0);
}
-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 @@
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 @@
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 @@
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 @@
EXPECT_EQ(stats_after_reset.sum_increases, 3);
}
+INSTANTIATE_TEST_SUITE_P(,
+ InputVolumeStatsReporterTest,
+ ::testing::Values(InputVolumeType::kApplied,
+ InputVolumeType::kRecommended));
+
} // namespace webrtc
diff --git a/modules/audio_processing/audio_processing_impl.cc b/modules/audio_processing/audio_processing_impl.cc
index 76e3804..a0415e2 100644
--- a/modules/audio_processing/audio_processing_impl.cc
+++ b/modules/audio_processing/audio_processing_impl.cc
@@ -291,7 +291,11 @@
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 @@
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(
diff --git a/modules/audio_processing/audio_processing_impl.h b/modules/audio_processing/audio_processing_impl.h
index bc582b5..5daea90 100644
--- a/modules/audio_processing/audio_processing_impl.h
+++ b/modules/audio_processing/audio_processing_impl.h
@@ -541,6 +541,8 @@
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<