Attempt to reland: Allow intelligibility to compile in apm (https://codereview.webrtc.org/1182323005/)
Revert of original: https://codereview.webrtc.org/1187033005/
Changes in original:
- Added files to gyp and BUILD
- Made minor fixes to get everything to compile
and intelligibility_proc to run
- Added comments
- Auto-reformatting
New Changes:
- Added <numeric> header to intelligibility_enhancer.cc to address buildbot errors
- Switched to use WAV for i/o in intelligibility_proc.cc to address windows errors
- clean up
Note: Patch 1 duplicates Patch 7 of https://codereview.webrtc.org/1182323005/
R=andrew@webrtc.org
Review URL: https://codereview.webrtc.org/1190733004.
Cr-Commit-Position: refs/heads/master@{#9486}
diff --git a/webrtc/modules/audio_processing/BUILD.gn b/webrtc/modules/audio_processing/BUILD.gn
index 092be1e..ce750b6 100644
--- a/webrtc/modules/audio_processing/BUILD.gn
+++ b/webrtc/modules/audio_processing/BUILD.gn
@@ -89,6 +89,10 @@
"high_pass_filter_impl.cc",
"high_pass_filter_impl.h",
"include/audio_processing.h",
+ "intelligibility/intelligibility_enhancer.cc",
+ "intelligibility/intelligibility_enhancer.h",
+ "intelligibility/intelligibility_utils.cc",
+ "intelligibility/intelligibility_utils.h",
"level_estimator_impl.cc",
"level_estimator_impl.h",
"noise_suppression_impl.cc",
diff --git a/webrtc/modules/audio_processing/audio_processing.gypi b/webrtc/modules/audio_processing/audio_processing.gypi
index 8eb2775..a4f9b39 100644
--- a/webrtc/modules/audio_processing/audio_processing.gypi
+++ b/webrtc/modules/audio_processing/audio_processing.gypi
@@ -99,6 +99,10 @@
'high_pass_filter_impl.cc',
'high_pass_filter_impl.h',
'include/audio_processing.h',
+ 'intelligibility/intelligibility_enhancer.cc',
+ 'intelligibility/intelligibility_enhancer.h',
+ 'intelligibility/intelligibility_utils.cc',
+ 'intelligibility/intelligibility_utils.h',
'level_estimator_impl.cc',
'level_estimator_impl.h',
'noise_suppression_impl.cc',
diff --git a/webrtc/modules/audio_processing/audio_processing_tests.gypi b/webrtc/modules/audio_processing/audio_processing_tests.gypi
index 7658e10..a05c67b 100644
--- a/webrtc/modules/audio_processing/audio_processing_tests.gypi
+++ b/webrtc/modules/audio_processing/audio_processing_tests.gypi
@@ -59,6 +59,20 @@
'beamformer/nonlinear_beamformer_test.cc',
],
}, # nonlinear_beamformer_test
+ {
+ 'target_name': 'intelligibility_proc',
+ 'type': 'executable',
+ 'dependencies': [
+ 'audioproc_test_utils',
+ '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
+ '<(DEPTH)/testing/gtest.gyp:gtest',
+ '<(webrtc_root)/modules/modules.gyp:audio_processing',
+ '<(webrtc_root)/test/test.gyp:test_support',
+ ],
+ 'sources': [
+ 'intelligibility/intelligibility_proc.cc',
+ ],
+ }, # intelligibility_proc
],
'conditions': [
['enable_protobuf==1', {
diff --git a/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.cc b/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.cc
index 932eff1..3029e21 100644
--- a/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.cc
+++ b/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.cc
@@ -8,12 +8,20 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+//
+// Implements core class for intelligibility enhancer.
+//
+// Details of the model and algorithm can be found in the original paper:
+// http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=&arnumber=6882788
+//
+
#include "webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h"
#include <cmath>
#include <cstdlib>
#include <algorithm>
+#include <numeric>
#include "webrtc/base/checks.h"
#include "webrtc/common_audio/vad/include/webrtc_vad.h"
@@ -27,13 +35,16 @@
const int IntelligibilityEnhancer::kErbResolution = 2;
const int IntelligibilityEnhancer::kWindowSizeMs = 2;
-// The size of the chunk provided by APM, in milliseconds.
-const int IntelligibilityEnhancer::kChunkSizeMs = 10;
+const int IntelligibilityEnhancer::kChunkSizeMs = 10; // Size provided by APM.
const int IntelligibilityEnhancer::kAnalyzeRate = 800;
const int IntelligibilityEnhancer::kVarianceRate = 2;
const float IntelligibilityEnhancer::kClipFreq = 200.0f;
const float IntelligibilityEnhancer::kConfigRho = 0.02f;
const float IntelligibilityEnhancer::kKbdAlpha = 1.5f;
+
+// To disable gain update smoothing, set gain limit to be VERY high.
+// TODO(ekmeyerson): Add option to disable gain smoothing altogether
+// to avoid the extra computation.
const float IntelligibilityEnhancer::kGainChangeLimit = 0.0125f;
using VarianceType = intelligibility::VarianceArray::StepType;
@@ -41,12 +52,14 @@
IntelligibilityEnhancer::TransformCallback::TransformCallback(
IntelligibilityEnhancer* parent,
IntelligibilityEnhancer::AudioSource source)
- : parent_(parent),
- source_(source) {}
+ : parent_(parent), source_(source) {
+}
void IntelligibilityEnhancer::TransformCallback::ProcessAudioBlock(
const complex<float>* const* in_block,
- int in_channels, int frames, int /* out_channels */,
+ int in_channels,
+ int frames,
+ int /* out_channels */,
complex<float>* const* out_block) {
DCHECK_EQ(parent_->freqs_, frames);
for (int i = 0; i < in_channels; ++i) {
@@ -57,13 +70,14 @@
IntelligibilityEnhancer::IntelligibilityEnhancer(int erb_resolution,
int sample_rate_hz,
int channels,
- int cv_type, float cv_alpha,
+ int cv_type,
+ float cv_alpha,
int cv_win,
int analysis_rate,
int variance_rate,
float gain_limit)
- : freqs_(RealFourier::ComplexLength(RealFourier::FftOrder(
- sample_rate_hz * kWindowSizeMs / 1000))),
+ : freqs_(RealFourier::ComplexLength(
+ RealFourier::FftOrder(sample_rate_hz * kWindowSizeMs / 1000))),
window_size_(1 << RealFourier::FftOrder(freqs_)),
chunk_length_(sample_rate_hz * kChunkSizeMs / 1000),
bank_size_(GetBankSize(sample_rate_hz, erb_resolution)),
@@ -72,7 +86,9 @@
channels_(channels),
analysis_rate_(analysis_rate),
variance_rate_(variance_rate),
- clear_variance_(freqs_, static_cast<VarianceType>(cv_type), cv_win,
+ clear_variance_(freqs_,
+ static_cast<VarianceType>(cv_type),
+ cv_win,
cv_alpha),
noise_variance_(freqs_, VarianceType::kStepInfinite, 475, 0.01f),
filtered_clear_var_(new float[bank_size_]),
@@ -83,58 +99,51 @@
gains_eq_(new float[bank_size_]),
gain_applier_(freqs_, gain_limit),
temp_out_buffer_(nullptr),
- input_audio_(new float*[channels]),
+ input_audio_(new float* [channels]),
kbd_window_(new float[window_size_]),
render_callback_(this, AudioSource::kRenderStream),
capture_callback_(this, AudioSource::kCaptureStream),
block_count_(0),
analysis_step_(0),
- vad_high_(nullptr),
- vad_low_(nullptr),
+ vad_high_(WebRtcVad_Create()),
+ vad_low_(WebRtcVad_Create()),
vad_tmp_buffer_(new int16_t[chunk_length_]) {
DCHECK_LE(kConfigRho, 1.0f);
CreateErbBank();
- WebRtcVad_Create(&vad_high_);
WebRtcVad_Init(vad_high_);
- WebRtcVad_set_mode(vad_high_, 0); // high likelihood of speech
- WebRtcVad_Create(&vad_low_);
+ WebRtcVad_set_mode(vad_high_, 0); // High likelihood of speech.
WebRtcVad_Init(vad_low_);
- WebRtcVad_set_mode(vad_low_, 3); // low likelihood of speech
+ WebRtcVad_set_mode(vad_low_, 3); // Low likelihood of speech.
- temp_out_buffer_ = static_cast<float**>(malloc(
- sizeof(*temp_out_buffer_) * channels_ +
- sizeof(**temp_out_buffer_) * chunk_length_ * channels_));
+ temp_out_buffer_ = static_cast<float**>(
+ malloc(sizeof(*temp_out_buffer_) * channels_ +
+ sizeof(**temp_out_buffer_) * chunk_length_ * channels_));
for (int i = 0; i < channels_; ++i) {
- temp_out_buffer_[i] = reinterpret_cast<float*>(temp_out_buffer_ + channels_)
- + chunk_length_ * i;
+ temp_out_buffer_[i] =
+ reinterpret_cast<float*>(temp_out_buffer_ + channels_) +
+ chunk_length_ * i;
}
+ // Assumes all rho equal.
for (int i = 0; i < bank_size_; ++i) {
rho_[i] = kConfigRho * kConfigRho;
}
float freqs_khz = kClipFreq / 1000.0f;
- int erb_index = static_cast<int>(ceilf(11.17f * logf((freqs_khz + 0.312f) /
- (freqs_khz + 14.6575f))
- + 43.0f));
+ int erb_index = static_cast<int>(ceilf(
+ 11.17f * logf((freqs_khz + 0.312f) / (freqs_khz + 14.6575f)) + 43.0f));
start_freq_ = max(1, erb_index * kErbResolution);
WindowGenerator::KaiserBesselDerived(kKbdAlpha, window_size_,
kbd_window_.get());
- render_mangler_.reset(new LappedTransform(channels_, channels_,
- chunk_length_,
- kbd_window_.get(),
- window_size_,
- window_size_ / 2,
- &render_callback_));
- capture_mangler_.reset(new LappedTransform(channels_, channels_,
- chunk_length_,
- kbd_window_.get(),
- window_size_,
- window_size_ / 2,
- &capture_callback_));
+ render_mangler_.reset(new LappedTransform(
+ channels_, channels_, chunk_length_, kbd_window_.get(), window_size_,
+ window_size_ / 2, &render_callback_));
+ capture_mangler_.reset(new LappedTransform(
+ channels_, channels_, chunk_length_, kbd_window_.get(), window_size_,
+ window_size_ / 2, &capture_callback_));
}
IntelligibilityEnhancer::~IntelligibilityEnhancer() {
@@ -150,7 +159,9 @@
has_voice_low_ = WebRtcVad_Process(vad_low_, sample_rate_hz_,
vad_tmp_buffer_.get(), chunk_length_) == 1;
+ // Process and enhance chunk of |audio|
render_mangler_->ProcessChunk(audio, temp_out_buffer_);
+
for (int i = 0; i < channels_; ++i) {
memcpy(audio[i], temp_out_buffer_[i],
chunk_length_ * sizeof(**temp_out_buffer_));
@@ -161,21 +172,25 @@
for (int i = 0; i < chunk_length_; ++i) {
vad_tmp_buffer_[i] = (int16_t)audio[0][i];
}
- // TODO(bercic): the VAD was always detecting voice in the noise stream,
- // no matter what the aggressiveness, so it was temporarily disabled here
+ // TODO(bercic): The VAD was always detecting voice in the noise stream,
+ // no matter what the aggressiveness, so it was temporarily disabled here.
- //if (WebRtcVad_Process(vad_high_, sample_rate_hz_, vad_tmp_buffer_.get(),
- // chunk_length_) == 1) {
- // printf("capture HAS speech\n");
- // return;
- //}
- //printf("capture NO speech\n");
+ #if 0
+ if (WebRtcVad_Process(vad_high_, sample_rate_hz_, vad_tmp_buffer_.get(),
+ chunk_length_) == 1) {
+ printf("capture HAS speech\n");
+ return;
+ }
+ printf("capture NO speech\n");
+ #endif
+
capture_mangler_->ProcessChunk(audio, temp_out_buffer_);
}
void IntelligibilityEnhancer::DispatchAudio(
IntelligibilityEnhancer::AudioSource source,
- const complex<float>* in_block, complex<float>* out_block) {
+ const complex<float>* in_block,
+ complex<float>* out_block) {
switch (source) {
case kRenderStream:
ProcessClearBlock(in_block, out_block);
@@ -196,6 +211,9 @@
return;
}
+ // For now, always assumes enhancement is necessary.
+ // TODO(ekmeyerson): Change to only enhance if necessary,
+ // based on experiments with different cutoffs.
if (has_voice_low_ || true) {
clear_variance_.Step(in_block, false);
power_target = std::accumulate(clear_variance_.variance(),
@@ -221,23 +239,25 @@
FilterVariance(clear_variance_.variance(), filtered_clear_var_.get());
FilterVariance(noise_variance_.variance(), filtered_noise_var_.get());
- /* lambda binary search */
+ // Bisection search for optimal |lambda|
float lambda_bot = -1.0f, lambda_top = -10e-18f, lambda;
float power_bot, power_top, power;
- SolveEquation14(lambda_top, start_freq_, gains_eq_.get());
- power_top = DotProduct(gains_eq_.get(), filtered_clear_var_.get(),
- bank_size_);
- SolveEquation14(lambda_bot, start_freq_, gains_eq_.get());
- power_bot = DotProduct(gains_eq_.get(), filtered_clear_var_.get(),
- bank_size_);
+ SolveForGainsGivenLambda(lambda_top, start_freq_, gains_eq_.get());
+ power_top =
+ DotProduct(gains_eq_.get(), filtered_clear_var_.get(), bank_size_);
+ SolveForGainsGivenLambda(lambda_bot, start_freq_, gains_eq_.get());
+ power_bot =
+ DotProduct(gains_eq_.get(), filtered_clear_var_.get(), bank_size_);
DCHECK(power_target >= power_bot && power_target <= power_top);
- float power_ratio = 2.0f;
+ float power_ratio = 2.0f; // Ratio of achieved power to target power.
+ const float kConvergeThresh = 0.001f; // TODO(ekmeyerson): Find best values
+ const int kMaxIters = 100; // for these, based on experiments.
int iters = 0;
- while (fabs(power_ratio - 1.0f) > 0.001f && iters <= 100) {
+ while (fabs(power_ratio - 1.0f) > kConvergeThresh && iters <= kMaxIters) {
lambda = lambda_bot + (lambda_top - lambda_bot) / 2.0f;
- SolveEquation14(lambda, start_freq_, gains_eq_.get());
+ SolveForGainsGivenLambda(lambda, start_freq_, gains_eq_.get());
power = DotProduct(gains_eq_.get(), filtered_clear_var_.get(), bank_size_);
if (power < power_target) {
lambda_bot = lambda;
@@ -248,7 +268,7 @@
++iters;
}
- /* b = filterbank' * b */
+ // (ERB gain) = filterbank' * (freq gain)
float* gains = gain_applier_.target();
for (int i = 0; i < freqs_; ++i) {
gains[i] = 0.0f;
@@ -265,8 +285,8 @@
int IntelligibilityEnhancer::GetBankSize(int sample_rate, int erb_resolution) {
float freq_limit = sample_rate / 2000.0f;
- int erb_scale = ceilf(11.17f * logf((freq_limit + 0.312f) /
- (freq_limit + 14.6575f)) + 43.0f);
+ int erb_scale = ceilf(
+ 11.17f * logf((freq_limit + 0.312f) / (freq_limit + 14.6575f)) + 43.0f);
return erb_scale * erb_resolution;
}
@@ -283,29 +303,29 @@
center_freqs_[i] *= 0.5f * sample_rate_hz_ / last_center_freq;
}
- filter_bank_ = static_cast<float**>(malloc(
- sizeof(*filter_bank_) * bank_size_ +
- sizeof(**filter_bank_) * freqs_ * bank_size_));
+ filter_bank_ = static_cast<float**>(
+ malloc(sizeof(*filter_bank_) * bank_size_ +
+ sizeof(**filter_bank_) * freqs_ * bank_size_));
for (int i = 0; i < bank_size_; ++i) {
- filter_bank_[i] = reinterpret_cast<float*>(filter_bank_ + bank_size_) +
- freqs_ * i;
+ filter_bank_[i] =
+ reinterpret_cast<float*>(filter_bank_ + bank_size_) + freqs_ * i;
}
for (int i = 1; i <= bank_size_; ++i) {
int lll, ll, rr, rrr;
lll = round(center_freqs_[max(1, i - lf) - 1] * freqs_ /
- (0.5f * sample_rate_hz_));
- ll = round(center_freqs_[max(1, i ) - 1] * freqs_ /
- (0.5f * sample_rate_hz_));
+ (0.5f * sample_rate_hz_));
+ ll =
+ round(center_freqs_[max(1, i) - 1] * freqs_ / (0.5f * sample_rate_hz_));
lll = min(freqs_, max(lll, 1)) - 1;
- ll = min(freqs_, max(ll, 1)) - 1;
+ ll = min(freqs_, max(ll, 1)) - 1;
rrr = round(center_freqs_[min(bank_size_, i + rf) - 1] * freqs_ /
- (0.5f * sample_rate_hz_));
- rr = round(center_freqs_[min(bank_size_, i + 1) - 1] * freqs_ /
- (0.5f * sample_rate_hz_));
+ (0.5f * sample_rate_hz_));
+ rr = round(center_freqs_[min(bank_size_, i + 1) - 1] * freqs_ /
+ (0.5f * sample_rate_hz_));
rrr = min(freqs_, max(rrr, 1)) - 1;
- rr = min(freqs_, max(rr, 1)) - 1;
+ rr = min(freqs_, max(rr, 1)) - 1;
float step, element;
@@ -338,8 +358,9 @@
}
}
-void IntelligibilityEnhancer::SolveEquation14(float lambda, int start_freq,
- float* sols) {
+void IntelligibilityEnhancer::SolveForGainsGivenLambda(float lambda,
+ int start_freq,
+ float* sols) {
bool quadratic = (kConfigRho < 1.0f);
const float* var_x0 = filtered_clear_var_.get();
const float* var_n0 = filtered_noise_var_.get();
@@ -347,15 +368,17 @@
for (int n = 0; n < start_freq; ++n) {
sols[n] = 1.0f;
}
+
+ // Analytic solution for optimal gains. See paper for derivation.
for (int n = start_freq - 1; n < bank_size_; ++n) {
float alpha0, beta0, gamma0;
gamma0 = 0.5f * rho_[n] * var_x0[n] * var_n0[n] +
- lambda * var_x0[n] * var_n0[n] * var_n0[n];
+ lambda * var_x0[n] * var_n0[n] * var_n0[n];
beta0 = lambda * var_x0[n] * (2 - rho_[n]) * var_x0[n] * var_n0[n];
if (quadratic) {
alpha0 = lambda * var_x0[n] * (1 - rho_[n]) * var_x0[n] * var_x0[n];
- sols[n] = (-beta0 - sqrtf(beta0 * beta0 - 4 * alpha0 * gamma0))
- / (2 * alpha0);
+ sols[n] =
+ (-beta0 - sqrtf(beta0 * beta0 - 4 * alpha0 * gamma0)) / (2 * alpha0);
} else {
sols[n] = -gamma0 / beta0;
}
@@ -369,8 +392,9 @@
}
}
-float IntelligibilityEnhancer::DotProduct(const float* a, const float* b,
- int length) {
+float IntelligibilityEnhancer::DotProduct(const float* a,
+ const float* b,
+ int length) {
float ret = 0.0f;
for (int i = 0; i < length; ++i) {
@@ -380,4 +404,3 @@
}
} // namespace webrtc
-
diff --git a/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h b/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h
index d0818f6..8125707 100644
--- a/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h
+++ b/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h
@@ -8,14 +8,18 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+//
+// Specifies core class for intelligbility enhancement.
+//
+
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_INTELLIGIBILITY_INTELLIGIBILITY_ENHANCER_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_INTELLIGIBILITY_INTELLIGIBILITY_ENHANCER_H_
#include <complex>
+#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_audio/lapped_transform.h"
#include "webrtc/modules/audio_processing/intelligibility/intelligibility_utils.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
struct WebRtcVadInst;
typedef struct WebRtcVadInst VadInst;
@@ -25,6 +29,7 @@
// Speech intelligibility enhancement module. Reads render and capture
// audio streams and modifies the render stream with a set of gains per
// frequency bin to enhance speech against the noise background.
+// Note: assumes speech and noise streams are already separated.
class IntelligibilityEnhancer {
public:
// Construct a new instance with the given filter bank resolution,
@@ -33,30 +38,43 @@
// to elapse before a new gain computation is made. |variance_rate| specifies
// the number of gain recomputations after which the variances are reset.
// |cv_*| are parameters for the VarianceArray constructor for the
- // lear speech stream.
+ // clear speech stream.
// TODO(bercic): the |cv_*|, |*_rate| and |gain_limit| parameters should
// probably go away once fine tuning is done. They override the internal
// constants in the class (kGainChangeLimit, kAnalyzeRate, kVarianceRate).
- IntelligibilityEnhancer(int erb_resolution, int sample_rate_hz, int channels,
- int cv_type, float cv_alpha, int cv_win,
- int analysis_rate, int variance_rate,
+ IntelligibilityEnhancer(int erb_resolution,
+ int sample_rate_hz,
+ int channels,
+ int cv_type,
+ float cv_alpha,
+ int cv_win,
+ int analysis_rate,
+ int variance_rate,
float gain_limit);
~IntelligibilityEnhancer();
- void ProcessRenderAudio(float* const* audio);
+ // Reads and processes chunk of noise stream in time domain.
void ProcessCaptureAudio(float* const* audio);
+ // Reads chunk of speech in time domain and updates with modified signal.
+ void ProcessRenderAudio(float* const* audio);
+
private:
enum AudioSource {
- kRenderStream = 0,
- kCaptureStream,
+ kRenderStream = 0, // Clear speech stream.
+ kCaptureStream, // Noise stream.
};
+ // Provides access point to the frequency domain.
class TransformCallback : public LappedTransform::Callback {
public:
TransformCallback(IntelligibilityEnhancer* parent, AudioSource source);
+
+ // All in frequency domain, receives input |in_block|, applies
+ // intelligibility enhancement, and writes result to |out_block|.
virtual void ProcessAudioBlock(const std::complex<float>* const* in_block,
- int in_channels, int frames,
+ int in_channels,
+ int frames,
int out_channels,
std::complex<float>* const* out_block);
@@ -66,72 +84,95 @@
};
friend class TransformCallback;
- void DispatchAudio(AudioSource source, const std::complex<float>* in_block,
+ // Sends streams to ProcessClearBlock or ProcessNoiseBlock based on source.
+ void DispatchAudio(AudioSource source,
+ const std::complex<float>* in_block,
std::complex<float>* out_block);
+
+ // Updates variance computation and analysis with |in_block_|,
+ // and writes modified speech to |out_block|.
void ProcessClearBlock(const std::complex<float>* in_block,
std::complex<float>* out_block);
+
+ // Computes and sets modified gains.
void AnalyzeClearBlock(float power_target);
+
+ // Updates variance calculation for noise input with |in_block|.
void ProcessNoiseBlock(const std::complex<float>* in_block,
std::complex<float>* out_block);
+ // Returns number of ERB filters.
static int GetBankSize(int sample_rate, int erb_resolution);
+
+ // Initializes ERB filterbank.
void CreateErbBank();
- void SolveEquation14(float lambda, int start_freq, float* sols);
+
+ // Analytically solves quadratic for optimal gains given |lambda|.
+ // Negative gains are set to 0. Stores the results in |sols|.
+ void SolveForGainsGivenLambda(float lambda, int start_freq, float* sols);
+
+ // Computes variance across ERB filters from freq variance |var|.
+ // Stores in |result|.
void FilterVariance(const float* var, float* result);
+
+ // Returns dot product of vectors specified by size |length| arrays |a|,|b|.
static float DotProduct(const float* a, const float* b, int length);
static const int kErbResolution;
static const int kWindowSizeMs;
static const int kChunkSizeMs;
- static const int kAnalyzeRate;
- static const int kVarianceRate;
+ static const int kAnalyzeRate; // Default for |analysis_rate_|.
+ static const int kVarianceRate; // Default for |variance_rate_|.
static const float kClipFreq;
- static const float kConfigRho;
+ static const float kConfigRho; // Default production and interpretation SNR.
static const float kKbdAlpha;
static const float kGainChangeLimit;
- const int freqs_;
- const int window_size_; // window size in samples; also the block size
- const int chunk_length_; // chunk size in samples
- const int bank_size_;
+ const int freqs_; // Num frequencies in frequency domain.
+ const int window_size_; // Window size in samples; also the block size.
+ const int chunk_length_; // Chunk size in samples.
+ const int bank_size_; // Num ERB filters.
const int sample_rate_hz_;
const int erb_resolution_;
- const int channels_;
- const int analysis_rate_;
- const int variance_rate_;
+ const int channels_; // Num channels.
+ const int analysis_rate_; // Num blocks before gains recalculated.
+ const int variance_rate_; // Num recalculations before history is cleared.
intelligibility::VarianceArray clear_variance_;
intelligibility::VarianceArray noise_variance_;
- scoped_ptr<float[]> filtered_clear_var_;
- scoped_ptr<float[]> filtered_noise_var_;
- float** filter_bank_;
- scoped_ptr<float[]> center_freqs_;
+ rtc::scoped_ptr<float[]> filtered_clear_var_;
+ rtc::scoped_ptr<float[]> filtered_noise_var_;
+ float** filter_bank_; // TODO(ekmeyerson): Switch to using ChannelBuffer.
+ rtc::scoped_ptr<float[]> center_freqs_;
int start_freq_;
- scoped_ptr<float[]> rho_;
- scoped_ptr<float[]> gains_eq_;
+ rtc::scoped_ptr<float[]> rho_; // Production and interpretation SNR.
+ // for each ERB band.
+ rtc::scoped_ptr<float[]> gains_eq_; // Pre-filter modified gains.
intelligibility::GainApplier gain_applier_;
// Destination buffer used to reassemble blocked chunks before overwriting
// the original input array with modifications.
+ // TODO(ekmeyerson): Switch to using ChannelBuffer.
float** temp_out_buffer_;
- scoped_ptr<float*[]> input_audio_;
- scoped_ptr<float[]> kbd_window_;
+
+ rtc::scoped_ptr<float* []> input_audio_;
+ rtc::scoped_ptr<float[]> kbd_window_;
TransformCallback render_callback_;
TransformCallback capture_callback_;
- scoped_ptr<LappedTransform> render_mangler_;
- scoped_ptr<LappedTransform> capture_mangler_;
+ rtc::scoped_ptr<LappedTransform> render_mangler_;
+ rtc::scoped_ptr<LappedTransform> capture_mangler_;
int block_count_;
int analysis_step_;
// TODO(bercic): Quick stopgap measure for voice detection in the clear
// and noise streams.
+ // Note: VAD currently does not affect anything in IntelligibilityEnhancer.
VadInst* vad_high_;
VadInst* vad_low_;
- scoped_ptr<int16_t[]> vad_tmp_buffer_;
- bool has_voice_low_;
+ rtc::scoped_ptr<int16_t[]> vad_tmp_buffer_;
+ bool has_voice_low_; // Whether voice detected in speech stream.
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_INTELLIGIBILITY_INTELLIGIBILITY_ENHANCER_H_
-
diff --git a/webrtc/modules/audio_processing/intelligibility/intelligibility_proc.cc b/webrtc/modules/audio_processing/intelligibility/intelligibility_proc.cc
index b0ea2df..9f7d84e 100644
--- a/webrtc/modules/audio_processing/intelligibility/intelligibility_proc.cc
+++ b/webrtc/modules/audio_processing/intelligibility/intelligibility_proc.cc
@@ -8,180 +8,138 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include <arpa/inet.h>
-#include <fcntl.h>
+//
+// Command line tool for speech intelligibility enhancement. Provides for
+// running and testing intelligibility_enhancer as an independent process.
+// Use --help for options.
+//
+
#include <stdint.h>
-#include <stdio.h>
#include <stdlib.h>
-#include <sys/mman.h>
+#include <string>
#include <sys/stat.h>
#include <sys/types.h>
-#include <unistd.h>
-
-#include <fenv.h>
-#include <limits>
-
-#include <complex>
#include "gflags/gflags.h"
+#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/checks.h"
#include "webrtc/common_audio/real_fourier.h"
+#include "webrtc/common_audio/wav_file.h"
#include "webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h"
#include "webrtc/modules/audio_processing/intelligibility/intelligibility_utils.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
-
-const int16_t* in_ipcm;
-int16_t* out_ipcm;
-const int16_t* noise_ipcm;
-
-float* in_fpcm;
-float* out_fpcm;
-float* noise_fpcm;
-float* noise_cursor;
-float* clear_cursor;
-
-int samples;
-int fragment_size;
+#include "webrtc/test/testsupport/fileutils.h"
using std::complex;
+
+namespace webrtc {
+
using webrtc::RealFourier;
using webrtc::IntelligibilityEnhancer;
-DEFINE_int32(clear_type, webrtc::intelligibility::VarianceArray::kStepInfinite,
+DEFINE_int32(clear_type,
+ webrtc::intelligibility::VarianceArray::kStepInfinite,
"Variance algorithm for clear data.");
-DEFINE_double(clear_alpha, 0.9,
- "Variance decay factor for clear data.");
-DEFINE_int32(clear_window, 475,
+DEFINE_double(clear_alpha, 0.9, "Variance decay factor for clear data.");
+DEFINE_int32(clear_window,
+ 475,
"Window size for windowed variance for clear data.");
-DEFINE_int32(sample_rate, 16000,
+DEFINE_int32(sample_rate,
+ 16000,
"Audio sample rate used in the input and output files.");
-DEFINE_int32(ana_rate, 800,
+DEFINE_int32(ana_rate,
+ 800,
"Analysis rate; gains recalculated every N blocks.");
-DEFINE_int32(var_rate, 2,
- "Variance clear rate; history is forgotten every N gain recalculations.");
+DEFINE_int32(
+ var_rate,
+ 2,
+ "Variance clear rate; history is forgotten every N gain recalculations.");
DEFINE_double(gain_limit, 1000.0, "Maximum gain change in one block.");
-DEFINE_bool(repeat, false, "Repeat input file ad nauseam.");
+DEFINE_string(clear_file, "speech.wav", "Input file with clear speech.");
+DEFINE_string(noise_file, "noise.wav", "Input file with noise data.");
+DEFINE_string(out_file,
+ "proc_enhanced.wav",
+ "Enhanced output. Use '-' to "
+ "play through aplay immediately.");
-DEFINE_string(clear_file, "speech.pcm", "Input file with clear speech.");
-DEFINE_string(noise_file, "noise.pcm", "Input file with noise data.");
-DEFINE_string(out_file, "proc_enhanced.pcm", "Enhanced output. Use '-' to "
- "pipe through aplay internally.");
+// Constant IntelligibilityEnhancer constructor parameters.
+const int kErbResolution = 2;
+const int kNumChannels = 1;
-// Write an Sun AU-formatted audio chunk into file descriptor |fd|. Can be used
-// to pipe the audio stream directly into aplay.
-void writeau(int fd) {
- uint32_t thing;
-
- write(fd, ".snd", 4);
- thing = htonl(24);
- write(fd, &thing, sizeof(thing));
- thing = htonl(0xffffffff);
- write(fd, &thing, sizeof(thing));
- thing = htonl(3);
- write(fd, &thing, sizeof(thing));
- thing = htonl(FLAGS_sample_rate);
- write(fd, &thing, sizeof(thing));
- thing = htonl(1);
- write(fd, &thing, sizeof(thing));
-
- for (int i = 0; i < samples; ++i) {
- out_ipcm[i] = htons(out_ipcm[i]);
- }
- write(fd, out_ipcm, sizeof(*out_ipcm) * samples);
-}
-
-int main(int argc, char* argv[]) {
- google::SetUsageMessage("\n\nVariance algorithm types are:\n"
- " 0 - infinite/normal,\n"
- " 1 - exponentially decaying,\n"
- " 2 - rolling window.\n"
- "\nInput files must be little-endian 16-bit signed raw PCM.\n");
+// void function for gtest
+void void_main(int argc, char* argv[]) {
+ google::SetUsageMessage(
+ "\n\nVariance algorithm types are:\n"
+ " 0 - infinite/normal,\n"
+ " 1 - exponentially decaying,\n"
+ " 2 - rolling window.\n"
+ "\nInput files must be little-endian 16-bit signed raw PCM.\n");
google::ParseCommandLineFlags(&argc, &argv, true);
- const char* in_name = FLAGS_clear_file.c_str();
- const char* out_name = FLAGS_out_file.c_str();
- const char* noise_name = FLAGS_noise_file.c_str();
+ size_t samples; // Number of samples in input PCM file
+ size_t fragment_size; // Number of samples to process at a time
+ // to simulate APM stream processing
+
+ // Load settings and wav input.
+
+ fragment_size = FLAGS_sample_rate / 100; // Mirror real time APM chunk size.
+ // Duplicates chunk_length_ in
+ // IntelligibilityEnhancer.
+
struct stat in_stat, noise_stat;
- int in_fd, out_fd, noise_fd;
- FILE* aplay_file = nullptr;
+ ASSERT_EQ(stat(FLAGS_clear_file.c_str(), &in_stat), 0)
+ << "Empty speech file.";
+ ASSERT_EQ(stat(FLAGS_noise_file.c_str(), &noise_stat), 0)
+ << "Empty noise file.";
- fragment_size = FLAGS_sample_rate / 100;
+ samples = std::min(in_stat.st_size, noise_stat.st_size) / 2;
- stat(in_name, &in_stat);
- stat(noise_name, &noise_stat);
- samples = in_stat.st_size / sizeof(*in_ipcm);
+ WavReader in_file(FLAGS_clear_file);
+ std::vector<float> in_fpcm(samples);
+ in_file.ReadSamples(samples, &in_fpcm[0]);
- in_fd = open(in_name, O_RDONLY);
- if (!strcmp(out_name, "-")) {
- aplay_file = popen("aplay -t au", "w");
- out_fd = fileno(aplay_file);
- } else {
- out_fd = open(out_name, O_WRONLY | O_CREAT | O_TRUNC,
- S_IRUSR | S_IWUSR | S_IRGRP | S_IWGRP | S_IROTH | S_IWOTH);
- }
- noise_fd = open(noise_name, O_RDONLY);
+ WavReader noise_file(FLAGS_noise_file);
+ std::vector<float> noise_fpcm(samples);
+ noise_file.ReadSamples(samples, &noise_fpcm[0]);
- in_ipcm = static_cast<int16_t*>(mmap(nullptr, in_stat.st_size, PROT_READ,
- MAP_PRIVATE, in_fd, 0));
- noise_ipcm = static_cast<int16_t*>(mmap(nullptr, noise_stat.st_size,
- PROT_READ, MAP_PRIVATE, noise_fd, 0));
- out_ipcm = new int16_t[samples];
- out_fpcm = new float[samples];
- in_fpcm = new float[samples];
- noise_fpcm = new float[samples];
+ // Run intelligibility enhancement.
- for (int i = 0; i < samples; ++i) {
- noise_fpcm[i] = noise_ipcm[i % (noise_stat.st_size / sizeof(*noise_ipcm))];
- }
-
- //feenableexcept(FE_INVALID | FE_OVERFLOW);
- IntelligibilityEnhancer enh(2,
- FLAGS_sample_rate, 1,
- FLAGS_clear_type,
- static_cast<float>(FLAGS_clear_alpha),
- FLAGS_clear_window,
- FLAGS_ana_rate,
- FLAGS_var_rate,
- FLAGS_gain_limit);
+ IntelligibilityEnhancer enh(
+ kErbResolution, FLAGS_sample_rate, kNumChannels, FLAGS_clear_type,
+ static_cast<float>(FLAGS_clear_alpha), FLAGS_clear_window, FLAGS_ana_rate,
+ FLAGS_var_rate, FLAGS_gain_limit);
// Slice the input into smaller chunks, as the APM would do, and feed them
- // into the enhancer. Repeat indefinitely if FLAGS_repeat is set.
- do {
- noise_cursor = noise_fpcm;
- clear_cursor = in_fpcm;
- for (int i = 0; i < samples; ++i) {
- in_fpcm[i] = in_ipcm[i];
- }
+ // through the enhancer.
+ float* clear_cursor = &in_fpcm[0];
+ float* noise_cursor = &noise_fpcm[0];
- for (int i = 0; i < samples; i += fragment_size) {
- enh.ProcessCaptureAudio(&noise_cursor);
- enh.ProcessRenderAudio(&clear_cursor);
- clear_cursor += fragment_size;
- noise_cursor += fragment_size;
- }
-
- for (int i = 0; i < samples; ++i) {
- out_ipcm[i] = static_cast<float>(in_fpcm[i]);
- }
- if (!strcmp(out_name, "-")) {
- writeau(out_fd);
- } else {
- write(out_fd, out_ipcm, samples * sizeof(*out_ipcm));
- }
- } while (FLAGS_repeat);
-
- munmap(const_cast<int16_t*>(noise_ipcm), noise_stat.st_size);
- munmap(const_cast<int16_t*>(in_ipcm), in_stat.st_size);
- close(noise_fd);
- if (aplay_file) {
- pclose(aplay_file);
- } else {
- close(out_fd);
+ for (size_t i = 0; i < samples; i += fragment_size) {
+ enh.ProcessCaptureAudio(&noise_cursor);
+ enh.ProcessRenderAudio(&clear_cursor);
+ clear_cursor += fragment_size;
+ noise_cursor += fragment_size;
}
- close(in_fd);
- return 0;
+ if (FLAGS_out_file.compare("-") == 0) {
+ const std::string temp_out_filename =
+ test::TempFilename(test::WorkingDir(), "temp_wav_file");
+ {
+ WavWriter out_file(temp_out_filename, FLAGS_sample_rate, kNumChannels);
+ out_file.WriteSamples(&in_fpcm[0], samples);
+ }
+ system(("aplay " + temp_out_filename).c_str());
+ system(("rm " + temp_out_filename).c_str());
+ } else {
+ WavWriter out_file(FLAGS_out_file, FLAGS_sample_rate, kNumChannels);
+ out_file.WriteSamples(&in_fpcm[0], samples);
+ }
}
+} // namespace webrtc
+
+int main(int argc, char* argv[]) {
+ webrtc::void_main(argc, argv);
+ return 0;
+}
diff --git a/webrtc/modules/audio_processing/intelligibility/intelligibility_utils.cc b/webrtc/modules/audio_processing/intelligibility/intelligibility_utils.cc
index e6fc3fa..145cc08 100644
--- a/webrtc/modules/audio_processing/intelligibility/intelligibility_utils.cc
+++ b/webrtc/modules/audio_processing/intelligibility/intelligibility_utils.cc
@@ -8,6 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+//
+// Implements helper functions and classes for intelligibility enhancement.
+//
+
#include "webrtc/modules/audio_processing/intelligibility/intelligibility_utils.h"
#include <algorithm>
@@ -40,10 +44,13 @@
// were chosen randomly, so that even a series of all zeroes has some small
// variability.
inline complex<float> zerofudge(complex<float> c) {
- const static complex<float> fudge[7] = {
- {0.001f, 0.002f}, {0.008f, 0.001f}, {0.003f, 0.008f}, {0.0006f, 0.0009f},
- {0.001f, 0.004f}, {0.003f, 0.004f}, {0.002f, 0.009f}
- };
+ const static complex<float> fudge[7] = {{0.001f, 0.002f},
+ {0.008f, 0.001f},
+ {0.003f, 0.008f},
+ {0.0006f, 0.0009f},
+ {0.001f, 0.004f},
+ {0.003f, 0.004f},
+ {0.002f, 0.009f}};
static int fudge_index = 0;
if (cplxfinite(c) && !cplxnormal(c)) {
fudge_index = (fudge_index + 1) % 7;
@@ -54,8 +61,9 @@
// Incremental mean computation. Return the mean of the series with the
// mean |mean| with added |data|.
-inline complex<float> NewMean(complex<float> mean, complex<float> data,
- int count) {
+inline complex<float> NewMean(complex<float> mean,
+ complex<float> data,
+ int count) {
return mean + (data - mean) / static_cast<float>(count);
}
@@ -73,7 +81,9 @@
static const int kWindowBlockSize = 10;
-VarianceArray::VarianceArray(int freqs, StepType type, int window_size,
+VarianceArray::VarianceArray(int freqs,
+ StepType type,
+ int window_size,
float decay)
: running_mean_(new complex<float>[freqs]()),
running_mean_sq_(new complex<float>[freqs]()),
@@ -87,15 +97,15 @@
history_cursor_(0),
count_(0),
array_mean_(0.0f) {
- history_.reset(new scoped_ptr<complex<float>[]>[freqs_]());
+ history_.reset(new rtc::scoped_ptr<complex<float>[]>[freqs_]());
for (int i = 0; i < freqs_; ++i) {
history_[i].reset(new complex<float>[window_size_]());
}
- subhistory_.reset(new scoped_ptr<complex<float>[]>[freqs_]());
+ subhistory_.reset(new rtc::scoped_ptr<complex<float>[]>[freqs_]());
for (int i = 0; i < freqs_; ++i) {
subhistory_[i].reset(new complex<float>[window_size_]());
}
- subhistory_sq_.reset(new scoped_ptr<complex<float>[]>[freqs_]());
+ subhistory_sq_.reset(new rtc::scoped_ptr<complex<float>[]>[freqs_]());
for (int i = 0; i < freqs_; ++i) {
subhistory_sq_[i].reset(new complex<float>[window_size_]());
}
@@ -131,13 +141,15 @@
} else {
float old_sum = conj_sum_[i];
complex<float> old_mean = running_mean_[i];
- running_mean_[i] = old_mean + (sample - old_mean) /
- static_cast<float>(count_);
- conj_sum_[i] = (old_sum + std::conj(sample - old_mean) *
- (sample - running_mean_[i])).real();
- variance_[i] = conj_sum_[i] / (count_ - 1); // + fudge[fudge_index].real();
+ running_mean_[i] =
+ old_mean + (sample - old_mean) / static_cast<float>(count_);
+ conj_sum_[i] =
+ (old_sum + std::conj(sample - old_mean) * (sample - running_mean_[i]))
+ .real();
+ variance_[i] =
+ conj_sum_[i] / (count_ - 1); // + fudge[fudge_index].real();
if (skip_fudge && false) {
- //variance_[i] -= fudge[fudge_index].real();
+ // variance_[i] -= fudge[fudge_index].real();
}
}
array_mean_ += (variance_[i] - array_mean_) / (i + 1);
@@ -161,11 +173,13 @@
complex<float> prev = running_mean_[i];
complex<float> prev2 = running_mean_sq_[i];
running_mean_[i] = decay_ * prev + (1.0f - decay_) * sample;
- running_mean_sq_[i] = decay_ * prev2 +
- (1.0f - decay_) * sample * std::conj(sample);
- //variance_[i] = decay_ * variance_[i] + (1.0f - decay_) * (
- // (sample - running_mean_[i]) * std::conj(sample - running_mean_[i])).real();
- variance_[i] = (running_mean_sq_[i] - running_mean_[i] * std::conj(running_mean_[i])).real();
+ running_mean_sq_[i] =
+ decay_ * prev2 + (1.0f - decay_) * sample * std::conj(sample);
+ // variance_[i] = decay_ * variance_[i] + (1.0f - decay_) * (
+ // (sample - running_mean_[i]) * std::conj(sample -
+ // running_mean_[i])).real();
+ variance_[i] = (running_mean_sq_[i] -
+ running_mean_[i] * std::conj(running_mean_[i])).real();
}
array_mean_ += (variance_[i] - array_mean_) / (i + 1);
@@ -186,15 +200,15 @@
mean = history_[i][history_cursor_];
variance_[i] = 0.0f;
for (int j = 1; j < num; ++j) {
- complex<float> sample = zerofudge(
- history_[i][(history_cursor_ + j) % window_size_]);
+ complex<float> sample =
+ zerofudge(history_[i][(history_cursor_ + j) % window_size_]);
sample = history_[i][(history_cursor_ + j) % window_size_];
float old_sum = conj_sum;
complex<float> old_mean = mean;
mean = old_mean + (sample - old_mean) / static_cast<float>(j + 1);
- conj_sum = (old_sum + std::conj(sample - old_mean) *
- (sample - mean)).real();
+ conj_sum =
+ (old_sum + std::conj(sample - old_mean) * (sample - mean)).real();
variance_[i] = conj_sum / (j);
}
array_mean_ += (variance_[i] - array_mean_) / (i + 1);
@@ -217,11 +231,11 @@
subhistory_[i][history_cursor_ % window_size_] = sub_running_mean_[i];
subhistory_sq_[i][history_cursor_ % window_size_] = sub_running_mean_sq_[i];
- variance_[i] = (NewMean(running_mean_sq_[i], sub_running_mean_sq_[i],
- blocks) -
- NewMean(running_mean_[i], sub_running_mean_[i], blocks) *
- std::conj(NewMean(running_mean_[i], sub_running_mean_[i],
- blocks))).real();
+ variance_[i] =
+ (NewMean(running_mean_sq_[i], sub_running_mean_sq_[i], blocks) -
+ NewMean(running_mean_[i], sub_running_mean_[i], blocks) *
+ std::conj(NewMean(running_mean_[i], sub_running_mean_[i], blocks)))
+ .real();
if (count_ == kWindowBlockSize - 1) {
sub_running_mean_[i] = complex<float>(0.0f, 0.0f);
sub_running_mean_sq_[i] = complex<float>(0.0f, 0.0f);
@@ -284,4 +298,3 @@
} // namespace intelligibility
} // namespace webrtc
-
diff --git a/webrtc/modules/audio_processing/intelligibility/intelligibility_utils.h b/webrtc/modules/audio_processing/intelligibility/intelligibility_utils.h
index 550f293..075b8ad 100644
--- a/webrtc/modules/audio_processing/intelligibility/intelligibility_utils.h
+++ b/webrtc/modules/audio_processing/intelligibility/intelligibility_utils.h
@@ -8,12 +8,16 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+//
+// Specifies helper classes for intelligibility enhancement.
+//
+
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_INTELLIGIBILITY_INTELLIGIBILITY_UTILS_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_INTELLIGIBILITY_INTELLIGIBILITY_UTILS_H_
#include <complex>
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/base/scoped_ptr.h"
namespace webrtc {
@@ -63,14 +67,10 @@
void ApplyScale(float scale);
// The current set of variances.
- const float* variance() const {
- return variance_.get();
- }
+ const float* variance() const { return variance_.get(); }
// The mean value of the current set of variances.
- float array_mean() const {
- return array_mean_;
- }
+ float array_mean() const { return array_mean_; }
private:
void InfiniteStep(const std::complex<float>* data, bool dummy);
@@ -78,23 +78,26 @@
void WindowedStep(const std::complex<float>* data, bool dummy);
void BlockedStep(const std::complex<float>* data, bool dummy);
+ // TODO(ekmeyerson): Switch the following running means
+ // and histories from rtc::scoped_ptr to std::vector.
+
// The current average X and X^2.
- scoped_ptr<std::complex<float>[]> running_mean_;
- scoped_ptr<std::complex<float>[]> running_mean_sq_;
+ rtc::scoped_ptr<std::complex<float>[]> running_mean_;
+ rtc::scoped_ptr<std::complex<float>[]> running_mean_sq_;
// Average X and X^2 for the current block in kStepBlocked.
- scoped_ptr<std::complex<float>[]> sub_running_mean_;
- scoped_ptr<std::complex<float>[]> sub_running_mean_sq_;
+ rtc::scoped_ptr<std::complex<float>[]> sub_running_mean_;
+ rtc::scoped_ptr<std::complex<float>[]> sub_running_mean_sq_;
// Sample history for the rolling window in kStepWindowed and block-wise
// histories for kStepBlocked.
- scoped_ptr<scoped_ptr<std::complex<float>[]>[]> history_;
- scoped_ptr<scoped_ptr<std::complex<float>[]>[]> subhistory_;
- scoped_ptr<scoped_ptr<std::complex<float>[]>[]> subhistory_sq_;
+ rtc::scoped_ptr<rtc::scoped_ptr<std::complex<float>[]>[]> history_;
+ rtc::scoped_ptr<rtc::scoped_ptr<std::complex<float>[]>[]> subhistory_;
+ rtc::scoped_ptr<rtc::scoped_ptr<std::complex<float>[]>[]> subhistory_sq_;
// The current set of variances and sums for Welford's algorithm.
- scoped_ptr<float[]> variance_;
- scoped_ptr<float[]> conj_sum_;
+ rtc::scoped_ptr<float[]> variance_;
+ rtc::scoped_ptr<float[]> conj_sum_;
const int freqs_;
const int window_size_;
@@ -118,15 +121,13 @@
std::complex<float>* out_block);
// Return the current target gain set. Modify this array to set the targets.
- float* target() const {
- return target_.get();
- }
+ float* target() const { return target_.get(); }
private:
const int freqs_;
const float change_limit_;
- scoped_ptr<float[]> target_;
- scoped_ptr<float[]> current_;
+ rtc::scoped_ptr<float[]> target_;
+ rtc::scoped_ptr<float[]> current_;
};
} // namespace intelligibility
@@ -134,4 +135,3 @@
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_INTELLIGIBILITY_INTELLIGIBILITY_UTILS_H_
-