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Revert "APM: log both applied and recommended input volume stats"

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This reverts commit 8d7273357d92fab881561d886ce8dfe94e6e2238.

Reason for revert: revert needed to land https://webrtc-review.googlesource.com/c/src/+/280600

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: I4a2acfd5a983d9397932b2879cfa057deaf0eb2b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280581
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Auto-Submit: Alessio Bazzica <alessiob@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#38476}
This commit is contained in:
Alessio Bazzica
2022-10-26 13:25:44 +00:00
committed by WebRTC LUCI CQ
parent 137162e16e
commit 35b3c63ba4
6 changed files with 74 additions and 216 deletions
Download Patch File Download Diff File Expand all files Collapse all files

2
modules/audio_processing/agc2/BUILD.gn
View File

@ -436,9 +436,7 @@ rtc_library("input_volume_stats_reporter_unittests") {
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" ]
}

141
modules/audio_processing/agc2/input_volume_stats_reporter.cc
View File

@ -19,8 +19,6 @@
namespace webrtc {
namespace {
using InputVolumeType = InputVolumeStatsReporter::InputVolumeType;
constexpr int kFramesIn60Seconds = 6000;
constexpr int kMinInputVolume = 0;
constexpr int kMaxInputVolume = 255;
@ -37,102 +35,9 @@ float ComputeAverageUpdate(int sum_updates, int num_updates) {
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(
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;
InputVolumeStatsReporter::~InputVolumeStatsReporter() = default;
@ -169,19 +74,47 @@ void InputVolumeStatsReporter::LogVolumeUpdateStats() const {
const float average_update = ComputeAverageUpdate(
volume_update_stats_.sum_decreases + volume_update_stats_.sum_increases,
num_updates);
metrics::HistogramAdd(decrease_rate_histogram_,
volume_update_stats_.num_decreases);
RTC_HISTOGRAM_COUNTS_LINEAR(
/*name=*/"WebRTC.Audio.ApmAnalogGainDecreaseRate",
/*sample=*/volume_update_stats_.num_decreases,
/*min=*/1,
/*max=*/kFramesIn60Seconds,
/*bucket_count=*/50);
if (volume_update_stats_.num_decreases > 0) {
metrics::HistogramAdd(decrease_average_histogram_, average_decrease);
RTC_HISTOGRAM_COUNTS_LINEAR(
/*name=*/"WebRTC.Audio.ApmAnalogGainDecreaseAverage",
/*sample=*/average_decrease,
/*min=*/1,
/*max=*/kMaxUpdate,
/*bucket_count=*/50);
}
metrics::HistogramAdd(increase_rate_histogram_,
volume_update_stats_.num_increases);
RTC_HISTOGRAM_COUNTS_LINEAR(
/*name=*/"WebRTC.Audio.ApmAnalogGainIncreaseRate",
/*sample=*/volume_update_stats_.num_increases,
/*min=*/1,
/*max=*/kFramesIn60Seconds,
/*bucket_count=*/50);
if (volume_update_stats_.num_increases > 0) {
metrics::HistogramAdd(increase_average_histogram_, average_increase);
RTC_HISTOGRAM_COUNTS_LINEAR(
/*name=*/"WebRTC.Audio.ApmAnalogGainIncreaseAverage",
/*sample=*/average_increase,
/*min=*/1,
/*max=*/kMaxUpdate,
/*bucket_count=*/50);
}
metrics::HistogramAdd(update_rate_histogram_, num_updates);
RTC_HISTOGRAM_COUNTS_LINEAR(
/*name=*/"WebRTC.Audio.ApmAnalogGainUpdateRate",
/*sample=*/num_updates,
/*min=*/1,
/*max=*/kFramesIn60Seconds,
/*bucket_count=*/50);
if (num_updates > 0) {
metrics::HistogramAdd(update_average_histogram_, average_update);
RTC_HISTOGRAM_COUNTS_LINEAR(
/*name=*/"WebRTC.Audio.ApmAnalogGainUpdateAverage",
/*sample=*/average_update,
/*min=*/1,
/*max=*/kMaxUpdate,
/*bucket_count=*/50);
}
}

16
modules/audio_processing/agc2/input_volume_stats_reporter.h
View File

@ -13,7 +13,6 @@
#include "absl/types/optional.h"
#include "rtc_base/gtest_prod_util.h"
#include "system_wrappers/include/metrics.h"
namespace webrtc {
@ -22,12 +21,7 @@ namespace webrtc {
// the statistics into a histogram.
class InputVolumeStatsReporter {
public:
enum class InputVolumeType {
kApplied = 0,
kRecommended = 1,
};
explicit InputVolumeStatsReporter(InputVolumeType input_volume_type);
InputVolumeStatsReporter();
InputVolumeStatsReporter(const InputVolumeStatsReporter&) = delete;
InputVolumeStatsReporter operator=(const InputVolumeStatsReporter&) = delete;
~InputVolumeStatsReporter();
@ -63,14 +57,6 @@ class InputVolumeStatsReporter {
// 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;
};

113
modules/audio_processing/agc2/input_volume_stats_reporter_unittest.cc
View File

@ -10,71 +10,24 @@
#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;
constexpr absl::string_view kLabelPrefix = "WebRTC.Audio.Apm.";
class InputVolumeStatsReporterTest
: public ::testing::TestWithParam<InputVolumeType> {
class InputVolumeStatsReporterTest : public ::testing::Test {
public:
InputVolumeStatsReporterTest() { metrics::Reset(); }
InputVolumeStatsReporterTest() {}
protected:
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.";
}
}
void SetUp() override { metrics::Reset(); }
};
TEST_P(InputVolumeStatsReporterTest, CheckLogVolumeUpdateStatsEmpty) {
InputVolumeStatsReporter stats_reporter(InputVolumeType());
TEST_F(InputVolumeStatsReporterTest, CheckLogVolumeUpdateStatsEmpty) {
InputVolumeStatsReporter stats_reporter;
constexpr int kInputVolume = 10;
stats_reporter.UpdateStatistics(kInputVolume);
// Update almost until the periodic logging and reset.
@ -82,22 +35,25 @@ TEST_P(InputVolumeStatsReporterTest, CheckLogVolumeUpdateStatsEmpty) {
stats_reporter.UpdateStatistics(kInputVolume + 2);
stats_reporter.UpdateStatistics(kInputVolume);
}
EXPECT_METRIC_THAT(metrics::Samples(UpdateRateLabel()),
EXPECT_METRIC_THAT(metrics::Samples("WebRTC.Audio.ApmAnalogGainUpdateRate"),
::testing::ElementsAre());
EXPECT_METRIC_THAT(metrics::Samples(DecreaseRateLabel()),
EXPECT_METRIC_THAT(metrics::Samples("WebRTC.Audio.ApmAnalogGainDecreaseRate"),
::testing::ElementsAre());
EXPECT_METRIC_THAT(metrics::Samples(IncreaseRateLabel()),
EXPECT_METRIC_THAT(metrics::Samples("WebRTC.Audio.ApmAnalogGainIncreaseRate"),
::testing::ElementsAre());
EXPECT_METRIC_THAT(metrics::Samples(UpdateAverageLabel()),
EXPECT_METRIC_THAT(
metrics::Samples("WebRTC.Audio.ApmAnalogGainUpdateAverage"),
::testing::ElementsAre());
EXPECT_METRIC_THAT(metrics::Samples(DecreaseAverageLabel()),
EXPECT_METRIC_THAT(
metrics::Samples("WebRTC.Audio.ApmAnalogGainDecreaseAverage"),
::testing::ElementsAre());
EXPECT_METRIC_THAT(metrics::Samples(IncreaseAverageLabel()),
EXPECT_METRIC_THAT(
metrics::Samples("WebRTC.Audio.ApmAnalogGainIncreaseAverage"),
::testing::ElementsAre());
}
TEST_P(InputVolumeStatsReporterTest, CheckLogVolumeUpdateStatsNotEmpty) {
InputVolumeStatsReporter stats_reporter(InputVolumeType());
TEST_F(InputVolumeStatsReporterTest, CheckLogVolumeUpdateStatsNotEmpty) {
InputVolumeStatsReporter stats_reporter;
constexpr int kInputVolume = 10;
stats_reporter.UpdateStatistics(kInputVolume);
// Update until periodic logging.
@ -111,30 +67,30 @@ TEST_P(InputVolumeStatsReporterTest, CheckLogVolumeUpdateStatsNotEmpty) {
stats_reporter.UpdateStatistics(kInputVolume);
}
EXPECT_METRIC_THAT(
metrics::Samples(UpdateRateLabel()),
metrics::Samples("WebRTC.Audio.ApmAnalogGainUpdateRate"),
::testing::ElementsAre(::testing::Pair(kFramesIn60Seconds - 1, 1),
::testing::Pair(kFramesIn60Seconds, 1)));
EXPECT_METRIC_THAT(
metrics::Samples(DecreaseRateLabel()),
metrics::Samples("WebRTC.Audio.ApmAnalogGainDecreaseRate"),
::testing::ElementsAre(::testing::Pair(kFramesIn60Seconds / 2 - 1, 1),
::testing::Pair(kFramesIn60Seconds / 2, 1)));
EXPECT_METRIC_THAT(
metrics::Samples(IncreaseRateLabel()),
metrics::Samples("WebRTC.Audio.ApmAnalogGainIncreaseRate"),
::testing::ElementsAre(::testing::Pair(kFramesIn60Seconds / 2, 2)));
EXPECT_METRIC_THAT(
metrics::Samples(UpdateAverageLabel()),
metrics::Samples("WebRTC.Audio.ApmAnalogGainUpdateAverage"),
::testing::ElementsAre(::testing::Pair(2, 1), ::testing::Pair(3, 1)));
EXPECT_METRIC_THAT(
metrics::Samples(DecreaseAverageLabel()),
metrics::Samples("WebRTC.Audio.ApmAnalogGainDecreaseAverage"),
::testing::ElementsAre(::testing::Pair(2, 1), ::testing::Pair(3, 1)));
EXPECT_METRIC_THAT(
metrics::Samples(IncreaseAverageLabel()),
metrics::Samples("WebRTC.Audio.ApmAnalogGainIncreaseAverage"),
::testing::ElementsAre(::testing::Pair(2, 1), ::testing::Pair(3, 1)));
}
} // namespace
TEST_P(InputVolumeStatsReporterTest, CheckVolumeUpdateStatsForEmptyStats) {
InputVolumeStatsReporter stats_reporter(InputVolumeType());
TEST_F(InputVolumeStatsReporterTest, CheckVolumeUpdateStatsForEmptyStats) {
InputVolumeStatsReporter stats_reporter;
const auto& update_stats = stats_reporter.volume_update_stats();
EXPECT_EQ(update_stats.num_decreases, 0);
EXPECT_EQ(update_stats.sum_decreases, 0);
@ -142,10 +98,10 @@ TEST_P(InputVolumeStatsReporterTest, CheckVolumeUpdateStatsForEmptyStats) {
EXPECT_EQ(update_stats.sum_increases, 0);
}
TEST_P(InputVolumeStatsReporterTest,
TEST_F(InputVolumeStatsReporterTest,
CheckVolumeUpdateStatsAfterNoVolumeChange) {
constexpr int kInputVolume = 10;
InputVolumeStatsReporter stats_reporter(InputVolumeType());
InputVolumeStatsReporter stats_reporter;
stats_reporter.UpdateStatistics(kInputVolume);
stats_reporter.UpdateStatistics(kInputVolume);
stats_reporter.UpdateStatistics(kInputVolume);
@ -156,10 +112,10 @@ TEST_P(InputVolumeStatsReporterTest,
EXPECT_EQ(update_stats.sum_increases, 0);
}
TEST_P(InputVolumeStatsReporterTest,
TEST_F(InputVolumeStatsReporterTest,
CheckVolumeUpdateStatsAfterVolumeIncrease) {
constexpr int kInputVolume = 10;
InputVolumeStatsReporter stats_reporter(InputVolumeType());
InputVolumeStatsReporter stats_reporter;
stats_reporter.UpdateStatistics(kInputVolume);
stats_reporter.UpdateStatistics(kInputVolume + 4);
stats_reporter.UpdateStatistics(kInputVolume + 5);
@ -170,10 +126,10 @@ TEST_P(InputVolumeStatsReporterTest,
EXPECT_EQ(update_stats.sum_increases, 5);
}
TEST_P(InputVolumeStatsReporterTest,
TEST_F(InputVolumeStatsReporterTest,
CheckVolumeUpdateStatsAfterVolumeDecrease) {
constexpr int kInputVolume = 10;
InputVolumeStatsReporter stats_reporter(InputVolumeType());
InputVolumeStatsReporter stats_reporter;
stats_reporter.UpdateStatistics(kInputVolume);
stats_reporter.UpdateStatistics(kInputVolume - 4);
stats_reporter.UpdateStatistics(kInputVolume - 5);
@ -184,8 +140,8 @@ TEST_P(InputVolumeStatsReporterTest,
EXPECT_EQ(stats_update.sum_increases, 0);
}
TEST_P(InputVolumeStatsReporterTest, CheckVolumeUpdateStatsAfterReset) {
InputVolumeStatsReporter stats_reporter(InputVolumeType());
TEST_F(InputVolumeStatsReporterTest, CheckVolumeUpdateStatsAfterReset) {
InputVolumeStatsReporter stats_reporter;
constexpr int kInputVolume = 10;
stats_reporter.UpdateStatistics(kInputVolume);
// Update until the periodic reset.
@ -213,9 +169,4 @@ TEST_P(InputVolumeStatsReporterTest, CheckVolumeUpdateStatsAfterReset) {
EXPECT_EQ(stats_after_reset.sum_increases, 3);
}
INSTANTIATE_TEST_SUITE_P(,
InputVolumeStatsReporterTest,
::testing::Values(InputVolumeType::kApplied,
InputVolumeType::kRecommended));
} // namespace webrtc

10
modules/audio_processing/audio_processing_impl.cc
View File

@ -291,11 +291,7 @@ AudioProcessingImpl::AudioProcessingImpl(
MinimizeProcessingForUnusedOutput(),
field_trial::IsEnabled("WebRTC-TransientSuppressorForcedOff")),
capture_(),
capture_nonlocked_(),
applied_input_volume_stats_reporter_(
InputVolumeStatsReporter::InputVolumeType::kApplied),
recommended_input_volume_stats_reporter_(
InputVolumeStatsReporter::InputVolumeType::kRecommended) {
capture_nonlocked_() {
RTC_LOG(LS_INFO) << "Injected APM submodules:"
"\nEcho control factory: "
<< !!echo_control_factory_
@ -1365,10 +1361,6 @@ 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(

2
modules/audio_processing/audio_processing_impl.h
View File

@ -541,8 +541,6 @@ 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<