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Switch away from hz to samples per channel for FrameCombiner et al

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This simplifies the following steps:
* FrameCombiner infers the sample rate from channel size
* Sends the inferred sample rate to FixedDigitalLevelEstimator
  and Limiter.
* Those classes then convert the sample rate to channel size.
  Along the way perform checks that the derived channel size value
  is a legal value (which has already been done by FrameCombiner).

To:
* FrameCombiner sends channel size to FixedDigitalLevelEstimator and
  Limiter.

Bug: chromium:335805780
Change-Id: I6d2953ba5ee99771f3ff5bf4f4a049a8a29b5577
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352581
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42480}
This commit is contained in:
Tommi 2024-06-13 15:28:34 +02:00 committed by WebRTC LUCI CQ
parent da485a1b46
commit 093824c4d2
10 changed files with 95 additions and 65 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

7
modules/audio_mixer/frame_combiner.cc
View File

@ -106,10 +106,7 @@ void MixToFloatFrame(rtc::ArrayView<const AudioFrame* const> mix_list,
}
void RunLimiter(AudioFrameView<float> mixing_buffer_view, Limiter* limiter) {
const size_t sample_rate = mixing_buffer_view.samples_per_channel() * 1000 /
AudioMixerImpl::kFrameDurationInMs;
// TODO(alessiob): Avoid calling SetSampleRate every time.
limiter->SetSampleRate(sample_rate);
limiter->SetSamplesPerChannel(mixing_buffer_view.samples_per_channel());
limiter->Process(mixing_buffer_view);
}
@ -134,7 +131,7 @@ constexpr size_t FrameCombiner::kMaximumChannelSize;
FrameCombiner::FrameCombiner(bool use_limiter)
: data_dumper_(new ApmDataDumper(0)),
limiter_(static_cast<size_t>(48000), data_dumper_.get(), "AudioMixer"),
limiter_(data_dumper_.get(), kMaximumChannelSize, "AudioMixer"),
use_limiter_(use_limiter) {
static_assert(kMaximumChannelSize * kMaximumNumberOfChannels <=
AudioFrame::kMaxDataSizeSamples,

1
modules/audio_processing/BUILD.gn
View File

@ -108,6 +108,7 @@ rtc_library("gain_controller2") {
":apm_logging",
":audio_buffer",
":audio_frame_view",
"../../api/audio:audio_frame_api",
"../../api/audio:audio_processing",
"../../common_audio",
"../../rtc_base:checks",

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

@ -148,6 +148,7 @@ rtc_library("fixed_digital") {
"..:apm_logging",
"..:audio_frame_view",
"../../../api:array_view",
"../../../api/audio:audio_frame_api",
"../../../common_audio",
"../../../rtc_base:checks",
"../../../rtc_base:gtest_prod",
@ -388,6 +389,7 @@ rtc_library("fixed_digital_unittests") {
"..:apm_logging",
"..:audio_frame_view",
"../../../api:array_view",
"../../../api/audio:audio_frame_api",
"../../../common_audio",
"../../../rtc_base:checks",
"../../../rtc_base:gunit_helpers",

16
modules/audio_processing/agc2/fixed_digital_level_estimator.cc
View File

@ -14,6 +14,7 @@
#include <cmath>
#include "api/array_view.h"
#include "api/audio/audio_frame.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "rtc_base/checks.h"
@ -34,14 +35,17 @@ constexpr float kDecayFilterConstant = 0.9971259f;
} // namespace
FixedDigitalLevelEstimator::FixedDigitalLevelEstimator(
int sample_rate_hz,
size_t samples_per_channel,
ApmDataDumper* apm_data_dumper)
: apm_data_dumper_(apm_data_dumper),
filter_state_level_(kInitialFilterStateLevel) {
SetSampleRate(sample_rate_hz);
SetSamplesPerChannel(samples_per_channel);
CheckParameterCombination();
RTC_DCHECK(apm_data_dumper_);
apm_data_dumper_->DumpRaw("agc2_level_estimator_samplerate", sample_rate_hz);
// Convert `samples_per_channel` to sample rate for
// `agc2_level_estimator_samplerate`.
apm_data_dumper_->DumpRaw("agc2_level_estimator_samplerate",
samples_per_channel * kDefaultAudioBuffersPerSec);
}
void FixedDigitalLevelEstimator::CheckParameterCombination() {
@ -106,9 +110,9 @@ std::array<float, kSubFramesInFrame> FixedDigitalLevelEstimator::ComputeLevel(
return envelope;
}
void FixedDigitalLevelEstimator::SetSampleRate(int sample_rate_hz) {
samples_in_frame_ =
rtc::CheckedDivExact(sample_rate_hz * kFrameDurationMs, 1000);
void FixedDigitalLevelEstimator::SetSamplesPerChannel(
size_t samples_per_channel) {
samples_in_frame_ = static_cast<int>(samples_per_channel);
samples_in_sub_frame_ =
rtc::CheckedDivExact(samples_in_frame_, kSubFramesInFrame);
CheckParameterCombination();

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

@ -25,12 +25,16 @@ class ApmDataDumper;
// filtering.
class FixedDigitalLevelEstimator {
public:
// Sample rates are allowed if the number of samples in a frame
// (sample_rate_hz * kFrameDurationMs / 1000) is divisible by
// `samples_per_channel` is expected to be derived from this formula:
// sample_rate_hz * kFrameDurationMs / 1000
// or
// sample_rate_hz / 100
// I.e. the number of samples for 10ms of the given sample rate. The
// expectation is that samples per channel is divisible by
// kSubFramesInSample. For kFrameDurationMs=10 and
// kSubFramesInSample=20, this means that sample_rate_hz has to be
// divisible by 2000.
FixedDigitalLevelEstimator(int sample_rate_hz,
// kSubFramesInSample=20, this means that the original sample rate has to be
// divisible by 2000 and therefore `samples_per_channel` by 20.
FixedDigitalLevelEstimator(size_t samples_per_channel,
ApmDataDumper* apm_data_dumper);
FixedDigitalLevelEstimator(const FixedDigitalLevelEstimator&) = delete;
@ -46,7 +50,7 @@ class FixedDigitalLevelEstimator {
// Rate may be changed at any time (but not concurrently) from the
// value passed to the constructor. The class is not thread safe.
void SetSampleRate(int sample_rate_hz);
void SetSamplesPerChannel(size_t samples_per_channel);
// Resets the level estimator internal state.
void Reset();

41
modules/audio_processing/agc2/fixed_digital_level_estimator_unittest.cc
View File

@ -12,6 +12,7 @@
#include <limits>
#include "api/audio/audio_frame.h"
#include "common_audio/include/audio_util.h"
#include "modules/audio_processing/agc2/agc2_common.h"
#include "modules/audio_processing/agc2/agc2_testing_common.h"
@ -26,17 +27,17 @@ constexpr float kInputLevel = 10000.f;
// Run audio at specified settings through the level estimator, and
// verify that the output level falls within the bounds.
void TestLevelEstimator(int sample_rate_hz,
void TestLevelEstimator(size_t samples_per_channel,
int num_channels,
float input_level_linear_scale,
float expected_min,
float expected_max) {
ApmDataDumper apm_data_dumper(0);
FixedDigitalLevelEstimator level_estimator(sample_rate_hz, &apm_data_dumper);
FixedDigitalLevelEstimator level_estimator(samples_per_channel,
&apm_data_dumper);
const VectorFloatFrame vectors_with_float_frame(
num_channels, rtc::CheckedDivExact(sample_rate_hz, 100),
input_level_linear_scale);
num_channels, samples_per_channel, input_level_linear_scale);
for (int i = 0; i < 500; ++i) {
const auto level = level_estimator.ComputeLevel(
@ -56,7 +57,7 @@ void TestLevelEstimator(int sample_rate_hz,
// Returns time it takes for the level estimator to decrease its level
// estimate by 'level_reduction_db'.
float TimeMsToDecreaseLevel(int sample_rate_hz,
float TimeMsToDecreaseLevel(size_t samples_per_channel,
int num_channels,
float input_level_db,
float level_reduction_db) {
@ -64,10 +65,11 @@ float TimeMsToDecreaseLevel(int sample_rate_hz,
RTC_DCHECK_GT(level_reduction_db, 0);
const VectorFloatFrame vectors_with_float_frame(
num_channels, rtc::CheckedDivExact(sample_rate_hz, 100), input_level);
num_channels, samples_per_channel, input_level);
ApmDataDumper apm_data_dumper(0);
FixedDigitalLevelEstimator level_estimator(sample_rate_hz, &apm_data_dumper);
FixedDigitalLevelEstimator level_estimator(samples_per_channel,
&apm_data_dumper);
// Give the LevelEstimator plenty of time to ramp up and stabilize
float last_level = 0.f;
@ -78,8 +80,8 @@ float TimeMsToDecreaseLevel(int sample_rate_hz,
}
// Set input to 0.
VectorFloatFrame vectors_with_zero_float_frame(
num_channels, rtc::CheckedDivExact(sample_rate_hz, 100), 0);
VectorFloatFrame vectors_with_zero_float_frame(num_channels,
samples_per_channel, 0);
const float reduced_level_linear =
DbfsToFloatS16(input_level_db - level_reduction_db);
@ -102,21 +104,22 @@ float TimeMsToDecreaseLevel(int sample_rate_hz,
} // namespace
TEST(GainController2FixedDigitalLevelEstimator, EstimatorShouldNotCrash) {
TestLevelEstimator(8000, 1, 0, std::numeric_limits<float>::lowest(),
TestLevelEstimator(SampleRateToDefaultChannelSize(8000u), 1, 0,
std::numeric_limits<float>::lowest(),
std::numeric_limits<float>::max());
}
TEST(GainController2FixedDigitalLevelEstimator,
EstimatorShouldEstimateConstantLevel) {
TestLevelEstimator(10000, 1, kInputLevel, kInputLevel * 0.99,
kInputLevel * 1.01);
TestLevelEstimator(SampleRateToDefaultChannelSize(10000u), 1, kInputLevel,
kInputLevel * 0.99, kInputLevel * 1.01);
}
TEST(GainController2FixedDigitalLevelEstimator,
EstimatorShouldEstimateConstantLevelForManyChannels) {
constexpr size_t num_channels = 10;
TestLevelEstimator(20000, num_channels, kInputLevel, kInputLevel * 0.99,
kInputLevel * 1.01);
TestLevelEstimator(SampleRateToDefaultChannelSize(20000u), num_channels,
kInputLevel, kInputLevel * 0.99, kInputLevel * 1.01);
}
TEST(GainController2FixedDigitalLevelEstimator, TimeToDecreaseForLowLevel) {
@ -125,7 +128,8 @@ TEST(GainController2FixedDigitalLevelEstimator, TimeToDecreaseForLowLevel) {
constexpr float kExpectedTime = kLevelReductionDb * test::kDecayMs;
const float time_to_decrease =
TimeMsToDecreaseLevel(22000, 1, kInitialLowLevel, kLevelReductionDb);
TimeMsToDecreaseLevel(SampleRateToDefaultChannelSize(22000u), 1,
kInitialLowLevel, kLevelReductionDb);
EXPECT_LE(kExpectedTime * 0.9, time_to_decrease);
EXPECT_LE(time_to_decrease, kExpectedTime * 1.1);
@ -136,8 +140,8 @@ TEST(GainController2FixedDigitalLevelEstimator,
constexpr float kLevelReductionDb = 25;
constexpr float kExpectedTime = kLevelReductionDb * test::kDecayMs;
const float time_to_decrease =
TimeMsToDecreaseLevel(26000, 1, 0, kLevelReductionDb);
const float time_to_decrease = TimeMsToDecreaseLevel(
SampleRateToDefaultChannelSize(26000u), 1, 0, kLevelReductionDb);
EXPECT_LE(kExpectedTime * 0.9, time_to_decrease);
EXPECT_LE(time_to_decrease, kExpectedTime * 1.1);
@ -150,7 +154,8 @@ TEST(GainController2FixedDigitalLevelEstimator,
constexpr size_t kNumChannels = 10;
const float time_to_decrease =
TimeMsToDecreaseLevel(28000, kNumChannels, 0, kLevelReductionDb);
TimeMsToDecreaseLevel(SampleRateToDefaultChannelSize(28000u),
kNumChannels, 0, kLevelReductionDb);
EXPECT_LE(kExpectedTime * 0.9, time_to_decrease);
EXPECT_LE(time_to_decrease, kExpectedTime * 1.1);

29
modules/audio_processing/agc2/limiter.cc
View File

@ -86,24 +86,25 @@ void ScaleSamples(MonoView<const float> per_sample_scaling_factors,
}
}
}
void CheckLimiterSampleRate(int sample_rate_hz) {
// Check that per_sample_scaling_factors_ is large enough.
RTC_DCHECK_LE(sample_rate_hz,
kMaximalNumberOfSamplesPerChannel * 1000 / kFrameDurationMs);
}
} // namespace
Limiter::Limiter(int sample_rate_hz,
ApmDataDumper* apm_data_dumper,
Limiter::Limiter(ApmDataDumper* apm_data_dumper,
size_t samples_per_channel,
absl::string_view histogram_name)
: interp_gain_curve_(apm_data_dumper, histogram_name),
level_estimator_(sample_rate_hz, apm_data_dumper),
level_estimator_(samples_per_channel, apm_data_dumper),
apm_data_dumper_(apm_data_dumper) {
CheckLimiterSampleRate(sample_rate_hz);
RTC_DCHECK_LE(samples_per_channel, kMaximalNumberOfSamplesPerChannel);
}
// Deprecated ctor.
Limiter::Limiter(int sample_rate_hz,
ApmDataDumper* apm_data_dumper,
absl::string_view histogram_name_prefix)
: Limiter(apm_data_dumper,
SampleRateToDefaultChannelSize(sample_rate_hz),
histogram_name_prefix) {}
Limiter::~Limiter() = default;
void Limiter::Process(AudioFrameView<float> signal) {
@ -140,9 +141,9 @@ InterpolatedGainCurve::Stats Limiter::GetGainCurveStats() const {
return interp_gain_curve_.get_stats();
}
void Limiter::SetSampleRate(int sample_rate_hz) {
CheckLimiterSampleRate(sample_rate_hz);
level_estimator_.SetSampleRate(sample_rate_hz);
void Limiter::SetSamplesPerChannel(size_t samples_per_channel) {
RTC_DCHECK_LE(samples_per_channel, kMaximalNumberOfSamplesPerChannel);
level_estimator_.SetSamplesPerChannel(samples_per_channel);
}
void Limiter::Reset() {

28
modules/audio_processing/agc2/limiter.h
View File

@ -14,6 +14,7 @@
#include <vector>
#include "absl/strings/string_view.h"
#include "api/audio/audio_frame.h"
#include "modules/audio_processing/agc2/fixed_digital_level_estimator.h"
#include "modules/audio_processing/agc2/interpolated_gain_curve.h"
#include "modules/audio_processing/include/audio_frame_view.h"
@ -23,9 +24,16 @@ class ApmDataDumper;
class Limiter {
public:
Limiter(int sample_rate_hz,
ApmDataDumper* apm_data_dumper,
// See `SetSamplesPerChannel()` for valid values for `samples_per_channel`.
Limiter(ApmDataDumper* apm_data_dumper,
size_t samples_per_channel,
absl::string_view histogram_name_prefix);
[[deprecated("Use constructor that accepts samples_per_channel")]] Limiter(
int sample_rate_hz,
ApmDataDumper* apm_data_dumper,
absl::string_view histogram_name_prefix);
Limiter(const Limiter& limiter) = delete;
Limiter& operator=(const Limiter& limiter) = delete;
~Limiter();
@ -34,12 +42,16 @@ class Limiter {
void Process(AudioFrameView<float> signal);
InterpolatedGainCurve::Stats GetGainCurveStats() const;
// Supported rates must be
// * supported by FixedDigitalLevelEstimator
// * below kMaximalNumberOfSamplesPerChannel*1000/kFrameDurationMs
// so that samples_per_channel fit in the
// per_sample_scaling_factors_ array.
void SetSampleRate(int sample_rate_hz);
// Supported values must be
// * Supported by FixedDigitalLevelEstimator
// * Below or equal to kMaximalNumberOfSamplesPerChannel so that samples
// fit in the per_sample_scaling_factors_ array.
void SetSamplesPerChannel(size_t samples_per_channel);
[[deprecated("Use SetSamplesPerChannel")]] void SetSampleRate(
int sample_rate_hz) {
SetSamplesPerChannel(SampleRateToDefaultChannelSize(sample_rate_hz));
}
// Resets the internal state.
void Reset();

15
modules/audio_processing/agc2/limiter_unittest.cc
View File

@ -10,6 +10,7 @@
#include "modules/audio_processing/agc2/limiter.h"
#include "api/audio/audio_frame.h"
#include "common_audio/include/audio_util.h"
#include "modules/audio_processing/agc2/agc2_common.h"
#include "modules/audio_processing/agc2/agc2_testing_common.h"
@ -20,33 +21,33 @@
namespace webrtc {
TEST(Limiter, LimiterShouldConstructAndRun) {
const int sample_rate_hz = 48000;
const size_t samples_per_channel = SampleRateToDefaultChannelSize(48000);
ApmDataDumper apm_data_dumper(0);
Limiter limiter(sample_rate_hz, &apm_data_dumper, "");
Limiter limiter(&apm_data_dumper, samples_per_channel, "");
VectorFloatFrame vectors_with_float_frame(1, sample_rate_hz / 100,
VectorFloatFrame vectors_with_float_frame(1, samples_per_channel,
kMaxAbsFloatS16Value);
limiter.Process(vectors_with_float_frame.float_frame_view());
}
TEST(Limiter, OutputVolumeAboveThreshold) {
const int sample_rate_hz = 48000;
const size_t samples_per_channel = SampleRateToDefaultChannelSize(48000);
const float input_level =
(kMaxAbsFloatS16Value + DbfsToFloatS16(test::kLimiterMaxInputLevelDbFs)) /
2.f;
ApmDataDumper apm_data_dumper(0);
Limiter limiter(sample_rate_hz, &apm_data_dumper, "");
Limiter limiter(&apm_data_dumper, samples_per_channel, "");
// Give the level estimator time to adapt.
for (int i = 0; i < 5; ++i) {
VectorFloatFrame vectors_with_float_frame(1, sample_rate_hz / 100,
VectorFloatFrame vectors_with_float_frame(1, samples_per_channel,
input_level);
limiter.Process(vectors_with_float_frame.float_frame_view());
}
VectorFloatFrame vectors_with_float_frame(1, sample_rate_hz / 100,
VectorFloatFrame vectors_with_float_frame(1, samples_per_channel,
input_level);
limiter.Process(vectors_with_float_frame.float_frame_view());
rtc::ArrayView<const float> channel =

5
modules/audio_processing/gain_controller2.cc
View File

@ -13,6 +13,7 @@
#include <memory>
#include <utility>
#include "api/audio/audio_frame.h"
#include "common_audio/include/audio_util.h"
#include "modules/audio_processing/agc2/agc2_common.h"
#include "modules/audio_processing/agc2/cpu_features.h"
@ -94,7 +95,9 @@ GainController2::GainController2(
fixed_gain_applier_(
/*hard_clip_samples=*/false,
/*initial_gain_factor=*/DbToRatio(config.fixed_digital.gain_db)),
limiter_(sample_rate_hz, &data_dumper_, /*histogram_name_prefix=*/"Agc2"),
limiter_(&data_dumper_,
SampleRateToDefaultChannelSize(sample_rate_hz),
/*histogram_name_prefix=*/"Agc2"),
calls_since_last_limiter_log_(0) {
RTC_DCHECK(Validate(config));
data_dumper_.InitiateNewSetOfRecordings();