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AGC2 size_t -> int

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Bug: webrtc:7494
Change-Id: I5ecf242e83b509931c1764a37339d11506c5afc6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213341
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33600}
This commit is contained in:
Alessio Bazzica
2021-03-31 09:48:49 +02:00
committed by Commit Bot
parent 2178d1ae69
commit b995bb86df
3 changed files with 21 additions and 21 deletions
Download Patch File Download Diff File Expand all files Collapse all files

18
modules/audio_processing/agc2/agc2_common.h
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@ -19,9 +19,9 @@ constexpr float kMinFloatS16Value = -32768.f;
constexpr float kMaxFloatS16Value = 32767.f;
constexpr float kMaxAbsFloatS16Value = 32768.0f;
constexpr size_t kFrameDurationMs = 10;
constexpr size_t kSubFramesInFrame = 20;
constexpr size_t kMaximalNumberOfSamplesPerChannel = 480;
constexpr int kFrameDurationMs = 10;
constexpr int kSubFramesInFrame = 20;
constexpr int kMaximalNumberOfSamplesPerChannel = 480;
constexpr float kAttackFilterConstant = 0.f;
@ -38,7 +38,7 @@ constexpr float kLimiterThresholdForAgcGainDbfs = -kHeadroomDbfs;
constexpr float kVadConfidenceThreshold = 0.9f;
// The amount of 'memory' of the Level Estimator. Decides leak factors.
constexpr size_t kFullBufferSizeMs = 1200;
constexpr int kFullBufferSizeMs = 1200;
constexpr float kFullBufferLeakFactor = 1.f - 1.f / kFullBufferSizeMs;
constexpr float kInitialSpeechLevelEstimateDbfs = -30.f;
@ -51,12 +51,12 @@ constexpr int kDefaultLevelEstimatorAdjacentSpeechFramesThreshold = 1;
constexpr float kDefaultInitialSaturationMarginDb = 20.f;
constexpr float kDefaultExtraSaturationMarginDb = 2.f;
constexpr size_t kPeakEnveloperSuperFrameLengthMs = 400;
constexpr int kPeakEnveloperSuperFrameLengthMs = 400;
static_assert(kFullBufferSizeMs % kPeakEnveloperSuperFrameLengthMs == 0,
"Full buffer size should be a multiple of super frame length for "
"optimal Saturation Protector performance.");
constexpr size_t kPeakEnveloperBufferSize =
constexpr int kPeakEnveloperBufferSize =
kFullBufferSizeMs / kPeakEnveloperSuperFrameLengthMs + 1;
// This value is 10 ** (-1/20 * frame_size_ms / satproc_attack_ms),
@ -76,9 +76,9 @@ constexpr float kDecayFilterConstant = 0.9998848773724686f;
// Number of interpolation points for each region of the limiter.
// These values have been tuned to limit the interpolated gain curve error given
// the limiter parameters and allowing a maximum error of +/- 32768^-1.
constexpr size_t kInterpolatedGainCurveKneePoints = 22;
constexpr size_t kInterpolatedGainCurveBeyondKneePoints = 10;
constexpr size_t kInterpolatedGainCurveTotalPoints =
constexpr int kInterpolatedGainCurveKneePoints = 22;
constexpr int kInterpolatedGainCurveBeyondKneePoints = 10;
constexpr int kInterpolatedGainCurveTotalPoints =
kInterpolatedGainCurveKneePoints + kInterpolatedGainCurveBeyondKneePoints;
} // namespace webrtc

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

@ -25,7 +25,7 @@ constexpr float kInitialFilterStateLevel = 0.f;
} // namespace
FixedDigitalLevelEstimator::FixedDigitalLevelEstimator(
size_t sample_rate_hz,
int sample_rate_hz,
ApmDataDumper* apm_data_dumper)
: apm_data_dumper_(apm_data_dumper),
filter_state_level_(kInitialFilterStateLevel) {
@ -52,8 +52,8 @@ std::array<float, kSubFramesInFrame> FixedDigitalLevelEstimator::ComputeLevel(
for (size_t channel_idx = 0; channel_idx < float_frame.num_channels();
++channel_idx) {
const auto channel = float_frame.channel(channel_idx);
for (size_t sub_frame = 0; sub_frame < kSubFramesInFrame; ++sub_frame) {
for (size_t sample_in_sub_frame = 0;
for (int sub_frame = 0; sub_frame < kSubFramesInFrame; ++sub_frame) {
for (int sample_in_sub_frame = 0;
sample_in_sub_frame < samples_in_sub_frame_; ++sample_in_sub_frame) {
envelope[sub_frame] =
std::max(envelope[sub_frame],
@ -66,14 +66,14 @@ std::array<float, kSubFramesInFrame> FixedDigitalLevelEstimator::ComputeLevel(
// Make sure envelope increases happen one step earlier so that the
// corresponding *gain decrease* doesn't miss a sudden signal
// increase due to interpolation.
for (size_t sub_frame = 0; sub_frame < kSubFramesInFrame - 1; ++sub_frame) {
for (int sub_frame = 0; sub_frame < kSubFramesInFrame - 1; ++sub_frame) {
if (envelope[sub_frame] < envelope[sub_frame + 1]) {
envelope[sub_frame] = envelope[sub_frame + 1];
}
}
// Add attack / decay smoothing.
for (size_t sub_frame = 0; sub_frame < kSubFramesInFrame; ++sub_frame) {
for (int sub_frame = 0; sub_frame < kSubFramesInFrame; ++sub_frame) {
const float envelope_value = envelope[sub_frame];
if (envelope_value > filter_state_level_) {
envelope[sub_frame] = envelope_value * (1 - kAttackFilterConstant) +
@ -97,9 +97,9 @@ std::array<float, kSubFramesInFrame> FixedDigitalLevelEstimator::ComputeLevel(
return envelope;
}
void FixedDigitalLevelEstimator::SetSampleRate(size_t sample_rate_hz) {
samples_in_frame_ = rtc::CheckedDivExact(sample_rate_hz * kFrameDurationMs,
static_cast<size_t>(1000));
void FixedDigitalLevelEstimator::SetSampleRate(int sample_rate_hz) {
samples_in_frame_ =
rtc::CheckedDivExact(sample_rate_hz * kFrameDurationMs, 1000);
samples_in_sub_frame_ =
rtc::CheckedDivExact(samples_in_frame_, kSubFramesInFrame);
CheckParameterCombination();

8
modules/audio_processing/agc2/fixed_digital_level_estimator.h
View File

@ -31,7 +31,7 @@ class FixedDigitalLevelEstimator {
// kSubFramesInSample. For kFrameDurationMs=10 and
// kSubFramesInSample=20, this means that sample_rate_hz has to be
// divisible by 2000.
FixedDigitalLevelEstimator(size_t sample_rate_hz,
FixedDigitalLevelEstimator(int sample_rate_hz,
ApmDataDumper* apm_data_dumper);
// The input is assumed to be in FloatS16 format. Scaled input will
@ -43,7 +43,7 @@ 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(size_t sample_rate_hz);
void SetSampleRate(int sample_rate_hz);
// Resets the level estimator internal state.
void Reset();
@ -55,8 +55,8 @@ class FixedDigitalLevelEstimator {
ApmDataDumper* const apm_data_dumper_ = nullptr;
float filter_state_level_;
size_t samples_in_frame_;
size_t samples_in_sub_frame_;
int samples_in_frame_;
int samples_in_sub_frame_;
RTC_DISALLOW_COPY_AND_ASSIGN(FixedDigitalLevelEstimator);
};