Fix variable shadowing in the Audio Processing Module
Bug: webrtc:42223409
Change-Id: I7ba3678ae4ff30e5637fca8894da2928b7e25f12
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/382741
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#44231}
diff --git a/modules/audio_processing/aec3/matched_filter.cc b/modules/audio_processing/aec3/matched_filter.cc
index 7f9e113..b5ed883 100644
--- a/modules/audio_processing/aec3/matched_filter.cc
+++ b/modules/audio_processing/aec3/matched_filter.cc
@@ -9,6 +9,9 @@
*/
#include "modules/audio_processing/aec3/matched_filter.h"
+#include <string>
+#include <vector>
+
// Defines WEBRTC_ARCH_X86_FAMILY, used below.
#include "rtc_base/system/arch.h"
@@ -21,11 +24,10 @@
#include <algorithm>
#include <cstddef>
#include <initializer_list>
-#include <iterator>
-#include <numeric>
#include <optional>
#include "api/array_view.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
#include "modules/audio_processing/aec3/downsampled_render_buffer.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "rtc_base/checks.h"
@@ -361,20 +363,20 @@
const float alpha = smoothing * e / x2_sum;
const __m128 alpha_128 = _mm_set1_ps(alpha);
// filter = filter + smoothing * (y - filter * x) * x / x * x.
- float* h_p = &h[0];
- const float* x_p =
+ float* h_p2 = &h[0];
+ const float* x_p2 =
chunk1 != h_size ? scratch_memory.data() : &x[x_start_index];
// Perform 128 bit vector operations.
const int limit_by_4 = h_size >> 2;
- for (int k = limit_by_4; k > 0; --k, h_p += 4, x_p += 4) {
+ for (int k = limit_by_4; k > 0; --k, h_p2 += 4, x_p2 += 4) {
// Load the data into 128 bit vectors.
- __m128 h_k = _mm_loadu_ps(h_p);
- const __m128 x_k = _mm_loadu_ps(x_p);
+ __m128 h_k = _mm_loadu_ps(h_p2);
+ const __m128 x_k = _mm_loadu_ps(x_p2);
// Compute h = h + alpha * x.
const __m128 alpha_x = _mm_mul_ps(alpha_128, x_k);
h_k = _mm_add_ps(h_k, alpha_x);
// Store the result.
- _mm_storeu_ps(h_p, h_k);
+ _mm_storeu_ps(h_p2, h_k);
}
*filters_updated = true;
}
@@ -464,26 +466,26 @@
const float alpha = smoothing * e / x2_sum;
const __m128 alpha_128 = _mm_set1_ps(alpha);
// filter = filter + smoothing * (y - filter * x) * x / x * x.
- float* h_p = &h[0];
+ float* h_p2 = &h[0];
x_p = &x[x_start_index];
// Perform the loop in two chunks.
for (int limit : {chunk1, chunk2}) {
// Perform 128 bit vector operations.
const int limit_by_4 = limit >> 2;
- for (int k = limit_by_4; k > 0; --k, h_p += 4, x_p += 4) {
+ for (int k = limit_by_4; k > 0; --k, h_p2 += 4, x_p += 4) {
// Load the data into 128 bit vectors.
- __m128 h_k = _mm_loadu_ps(h_p);
+ __m128 h_k = _mm_loadu_ps(h_p2);
const __m128 x_k = _mm_loadu_ps(x_p);
// Compute h = h + alpha * x.
const __m128 alpha_x = _mm_mul_ps(alpha_128, x_k);
h_k = _mm_add_ps(h_k, alpha_x);
// Store the result.
- _mm_storeu_ps(h_p, h_k);
+ _mm_storeu_ps(h_p2, h_k);
}
// Perform non-vector operations for any remaining items.
- for (int k = limit - limit_by_4 * 4; k > 0; --k, ++h_p, ++x_p) {
- *h_p += alpha * *x_p;
+ for (int k = limit - limit_by_4 * 4; k > 0; --k, ++h_p2, ++x_p) {
+ *h_p2 += alpha * *x_p;
}
x_p = &x[0];
}
@@ -543,10 +545,10 @@
const float alpha = smoothing * e / x2_sum;
// filter = filter + smoothing * (y - filter * x) * x / x * x.
- size_t x_index = x_start_index;
+ size_t x_index2 = x_start_index;
for (size_t k = 0; k < h.size(); ++k) {
- h[k] += alpha * x[x_index];
- x_index = x_index < (x.size() - 1) ? x_index + 1 : 0;
+ h[k] += alpha * x[x_index2];
+ x_index2 = x_index2 < (x.size() - 1) ? x_index2 + 1 : 0;
}
*filters_updated = true;
}
diff --git a/modules/audio_processing/aec3/matched_filter_avx2.cc b/modules/audio_processing/aec3/matched_filter_avx2.cc
index 8c2ffcb..6babff8 100644
--- a/modules/audio_processing/aec3/matched_filter_avx2.cc
+++ b/modules/audio_processing/aec3/matched_filter_avx2.cc
@@ -10,6 +10,10 @@
#include <immintrin.h>
+#include <algorithm>
+#include <cstddef>
+
+#include "api/array_view.h"
#include "modules/audio_processing/aec3/matched_filter.h"
#include "rtc_base/checks.h"
@@ -90,9 +94,9 @@
s_acum += s_inst_hadd_256[5];
e_128[3] = s_acum - y[i];
- __m128 accumulated_error = _mm_load_ps(a_p);
- accumulated_error = _mm_fmadd_ps(e_128, e_128, accumulated_error);
- _mm_storeu_ps(a_p, accumulated_error);
+ __m128 acum_error = _mm_load_ps(a_p);
+ acum_error = _mm_fmadd_ps(e_128, e_128, acum_error);
+ _mm_storeu_ps(a_p, acum_error);
}
// Sum components together.
x2_sum_256 = _mm256_add_ps(x2_sum_256, x2_sum_256_8);
@@ -114,20 +118,20 @@
const __m256 alpha_256 = _mm256_set1_ps(alpha);
// filter = filter + smoothing * (y - filter * x) * x / x * x.
- float* h_p = &h[0];
- const float* x_p =
+ float* h_p2 = &h[0];
+ const float* x_p2 =
chunk1 != h_size ? scratch_memory.data() : &x[x_start_index];
// Perform 256 bit vector operations.
const int limit_by_8 = h_size >> 3;
- for (int k = limit_by_8; k > 0; --k, h_p += 8, x_p += 8) {
+ for (int k = limit_by_8; k > 0; --k, h_p2 += 8, x_p2 += 8) {
// Load the data into 256 bit vectors.
- __m256 h_k = _mm256_loadu_ps(h_p);
- __m256 x_k = _mm256_loadu_ps(x_p);
+ __m256 h_k = _mm256_loadu_ps(h_p2);
+ __m256 x_k = _mm256_loadu_ps(x_p2);
// Compute h = h + alpha * x.
h_k = _mm256_fmadd_ps(x_k, alpha_256, h_k);
// Store the result.
- _mm256_storeu_ps(h_p, h_k);
+ _mm256_storeu_ps(h_p2, h_k);
}
*filters_updated = true;
}
@@ -224,27 +228,27 @@
const __m256 alpha_256 = _mm256_set1_ps(alpha);
// filter = filter + smoothing * (y - filter * x) * x / x * x.
- float* h_p = &h[0];
+ float* h_p2 = &h[0];
x_p = &x[x_start_index];
// Perform the loop in two chunks.
for (int limit : {chunk1, chunk2}) {
// Perform 256 bit vector operations.
const int limit_by_8 = limit >> 3;
- for (int k = limit_by_8; k > 0; --k, h_p += 8, x_p += 8) {
+ for (int k = limit_by_8; k > 0; --k, h_p2 += 8, x_p += 8) {
// Load the data into 256 bit vectors.
- __m256 h_k = _mm256_loadu_ps(h_p);
+ __m256 h_k = _mm256_loadu_ps(h_p2);
__m256 x_k = _mm256_loadu_ps(x_p);
// Compute h = h + alpha * x.
h_k = _mm256_fmadd_ps(x_k, alpha_256, h_k);
// Store the result.
- _mm256_storeu_ps(h_p, h_k);
+ _mm256_storeu_ps(h_p2, h_k);
}
// Perform non-vector operations for any remaining items.
- for (int k = limit - limit_by_8 * 8; k > 0; --k, ++h_p, ++x_p) {
- *h_p += alpha * *x_p;
+ for (int k = limit - limit_by_8 * 8; k > 0; --k, ++h_p2, ++x_p) {
+ *h_p2 += alpha * *x_p;
}
x_p = &x[0];
diff --git a/modules/audio_processing/ns/noise_estimator.cc b/modules/audio_processing/ns/noise_estimator.cc
index 5367545..214db52 100644
--- a/modules/audio_processing/ns/noise_estimator.cc
+++ b/modules/audio_processing/ns/noise_estimator.cc
@@ -11,8 +11,14 @@
#include "modules/audio_processing/ns/noise_estimator.h"
#include <algorithm>
+#include <array>
+#include <cstddef>
+#include <cstdint>
+#include "api/array_view.h"
#include "modules/audio_processing/ns/fast_math.h"
+#include "modules/audio_processing/ns/ns_common.h"
+#include "modules/audio_processing/ns/suppression_params.h"
#include "rtc_base/checks.h"
namespace webrtc {
@@ -129,9 +135,9 @@
} else {
// Use pink noise estimate.
float use_band = i < kStartBand ? kStartBand : i;
- float denom = PowApproximation(use_band, parametric_exp);
- RTC_DCHECK_NE(denom, 0.f);
- parametric_noise_spectrum_[i] = parametric_num / denom;
+ float parametric_denom = PowApproximation(use_band, parametric_exp);
+ RTC_DCHECK_NE(parametric_denom, 0.f);
+ parametric_noise_spectrum_[i] = parametric_num / parametric_denom;
}
}
diff --git a/modules/audio_processing/residual_echo_detector.cc b/modules/audio_processing/residual_echo_detector.cc
index 06ac149..0dbbc72 100644
--- a/modules/audio_processing/residual_echo_detector.cc
+++ b/modules/audio_processing/residual_echo_detector.cc
@@ -11,9 +11,13 @@
#include "modules/audio_processing/residual_echo_detector.h"
#include <algorithm>
+#include <atomic>
+#include <cstddef>
#include <numeric>
#include <optional>
+#include "api/array_view.h"
+#include "api/audio/audio_processing.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
@@ -135,25 +139,25 @@
if (echo_likelihood_ > 1.1f) {
// Make sure we don't spam the log.
if (log_counter_ < 5 && best_delay != -1) {
- size_t read_index = kLookbackFrames + next_insertion_index_ - best_delay;
- if (read_index >= kLookbackFrames) {
- read_index -= kLookbackFrames;
+ size_t read_index_high_echo =
+ kLookbackFrames + next_insertion_index_ - best_delay;
+ if (read_index_high_echo >= kLookbackFrames) {
+ read_index_high_echo -= kLookbackFrames;
}
- RTC_DCHECK_LT(read_index, render_power_.size());
- RTC_LOG_F(LS_ERROR) << "Echo detector internal state: {"
- "Echo likelihood: "
- << echo_likelihood_ << ", Best Delay: " << best_delay
- << ", Covariance: "
- << covariances_[best_delay].covariance()
- << ", Last capture power: " << capture_power
- << ", Capture mean: " << capture_mean
- << ", Capture_standard deviation: "
- << capture_std_deviation << ", Last render power: "
- << render_power_[read_index]
- << ", Render mean: " << render_power_mean_[read_index]
- << ", Render standard deviation: "
- << render_power_std_dev_[read_index]
- << ", Reliability: " << reliability_ << "}";
+ RTC_DCHECK_LT(read_index_high_echo, render_power_.size());
+ RTC_LOG_F(LS_ERROR)
+ << "Echo detector internal state: {"
+ "Echo likelihood: "
+ << echo_likelihood_ << ", Best Delay: " << best_delay
+ << ", Covariance: " << covariances_[best_delay].covariance()
+ << ", Last capture power: " << capture_power
+ << ", Capture mean: " << capture_mean
+ << ", Capture_standard deviation: " << capture_std_deviation
+ << ", Last render power: " << render_power_[read_index_high_echo]
+ << ", Render mean: " << render_power_mean_[read_index_high_echo]
+ << ", Render standard deviation: "
+ << render_power_std_dev_[read_index_high_echo]
+ << ", Reliability: " << reliability_ << "}";
log_counter_++;
}
}
diff --git a/modules/audio_processing/test/aec_dump_based_simulator.cc b/modules/audio_processing/test/aec_dump_based_simulator.cc
index 5b4bc49..d972af6 100644
--- a/modules/audio_processing/test/aec_dump_based_simulator.cc
+++ b/modules/audio_processing/test/aec_dump_based_simulator.cc
@@ -19,6 +19,7 @@
#include <iostream>
#include <memory>
#include <optional>
+#include <sstream> // no-presubmit-check TODO(webrtc:8982)
#include <string>
#include <utility>
@@ -50,7 +51,7 @@
return false;
} else {
const int16_t* frame_data = frame.data.data();
- for (int k = 0; k < frame.num_channels * frame.samples_per_channel; ++k) {
+ for (int k = 0; k < frame.num_channels_ * frame.samples_per_channel_; ++k) {
if (msg.output_data().data()[k] != frame_data[k]) {
return false;
}
diff --git a/modules/audio_processing/test/audio_processing_simulator.h b/modules/audio_processing/test/audio_processing_simulator.h
index 42279dd..8b6f95c 100644
--- a/modules/audio_processing/test/audio_processing_simulator.h
+++ b/modules/audio_processing/test/audio_processing_simulator.h
@@ -40,41 +40,41 @@
struct Int16Frame {
void SetFormat(int sample_rate_hz, int num_channels) {
- this->sample_rate_hz = sample_rate_hz;
- samples_per_channel =
+ sample_rate_hz_ = sample_rate_hz;
+ samples_per_channel_ =
rtc::CheckedDivExact(sample_rate_hz, kChunksPerSecond);
- this->num_channels = num_channels;
+ num_channels_ = num_channels;
config = StreamConfig(sample_rate_hz, num_channels);
- data.resize(num_channels * samples_per_channel);
+ data.resize(num_channels * samples_per_channel_);
}
void CopyTo(ChannelBuffer<float>* dest) {
RTC_DCHECK(dest);
- RTC_CHECK_EQ(num_channels, dest->num_channels());
- RTC_CHECK_EQ(samples_per_channel, dest->num_frames());
+ RTC_CHECK_EQ(num_channels_, dest->num_channels());
+ RTC_CHECK_EQ(samples_per_channel_, dest->num_frames());
// Copy the data from the input buffer.
- std::vector<float> tmp(samples_per_channel * num_channels);
+ std::vector<float> tmp(samples_per_channel_ * num_channels_);
S16ToFloat(data.data(), tmp.size(), tmp.data());
- Deinterleave(tmp.data(), samples_per_channel, num_channels,
+ Deinterleave(tmp.data(), samples_per_channel_, num_channels_,
dest->channels());
}
void CopyFrom(const ChannelBuffer<float>& src) {
- RTC_CHECK_EQ(src.num_channels(), num_channels);
- RTC_CHECK_EQ(src.num_frames(), samples_per_channel);
- data.resize(num_channels * samples_per_channel);
+ RTC_CHECK_EQ(src.num_channels(), num_channels_);
+ RTC_CHECK_EQ(src.num_frames(), samples_per_channel_);
+ data.resize(num_channels_ * samples_per_channel_);
int16_t* dest_data = data.data();
- for (int ch = 0; ch < num_channels; ++ch) {
- for (int sample = 0; sample < samples_per_channel; ++sample) {
- dest_data[sample * num_channels + ch] =
+ for (int ch = 0; ch < num_channels_; ++ch) {
+ for (int sample = 0; sample < samples_per_channel_; ++sample) {
+ dest_data[sample * num_channels_ + ch] =
src.channels()[ch][sample] * 32767;
}
}
}
- int sample_rate_hz;
- int samples_per_channel;
- int num_channels;
+ int sample_rate_hz_;
+ int samples_per_channel_;
+ int num_channels_;
StreamConfig config;
