Revert of Allow intelligibility to compile in apm (patchset #1 id:1 of https://codereview.webrtc.org/1182323005/)

Reason for revert:
Breaking the build bots: http://build.chromium.org/p/client.webrtc/builders/Mac32%20Release%20%5Blarge%20tests%5D/builds/4544

Fails to compile with this error:

../../webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.cc:218:25: error: no member named 'accumulate' in namespace 'std'
    power_target = std::accumulate(clear_variance_.variance(),

Original issue's description:
> Allow intelligibility to compile in apm
>
> - Added files to gyp and BUILD
> - Made minor fixes to get everything to compile
>     and intelligibility_proc to run
> - Added comments
> - Auto-reformatting
>
> Original cl is at: https://webrtc-codereview.appspot.com/57579004/
>
> TBR=aluebs@webrtc.org
>
> Committed: b7553dfdbb

TBR=ekmeyerson@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1187033005

Cr-Commit-Position: refs/heads/master@{#9455}

webrtc/modules/audio_processing/BUILD.gn

@@ -89,10 +89,6 @@ source_set("audio_processing") {
     "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",

webrtc/modules/audio_processing/audio_processing.gypi

@@ -99,10 +99,6 @@
         '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',

webrtc/modules/audio_processing/audio_processing_tests.gypi

@@ -59,19 +59,6 @@
         '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',
-      ],
-      'sources': [
-        'intelligibility/intelligibility_proc.cc',
-      ],
-    }, # intelligibility_proc
   ],
   'conditions': [
     ['enable_protobuf==1', {

webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.cc

@@ -8,13 +8,6 @@
  *  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>
@@ -34,16 +27,13 @@ namespace webrtc {
 
 const int IntelligibilityEnhancer::kErbResolution = 2;
 const int IntelligibilityEnhancer::kWindowSizeMs = 2;
-const int IntelligibilityEnhancer::kChunkSizeMs = 10;  // Size provided by APM.
+// The size of the chunk provided by APM, in milliseconds.
+const int IntelligibilityEnhancer::kChunkSizeMs = 10;
 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;
@@ -51,14 +41,12 @@ using VarianceType = intelligibility::VarianceArray::StepType;
 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 in_channels, int frames, int /* out_channels */,
-    int frames,
-    int /* out_channels */,
     complex<float>* const* out_block) {
   DCHECK_EQ(parent_->freqs_, frames);
   for (int i = 0; i < in_channels; ++i) {
@@ -69,14 +57,13 @@ void IntelligibilityEnhancer::TransformCallback::ProcessAudioBlock(
 IntelligibilityEnhancer::IntelligibilityEnhancer(int erb_resolution,
                                                  int sample_rate_hz,
                                                  int channels,
-                                                 int cv_type,
+                                                 int cv_type, float cv_alpha,
-                                                 float cv_alpha,
                                                  int cv_win,
                                                  int analysis_rate,
                                                  int variance_rate,
                                                  float gain_limit)
-    : freqs_(RealFourier::ComplexLength(
+    : freqs_(RealFourier::ComplexLength(RealFourier::FftOrder(
-          RealFourier::FftOrder(sample_rate_hz * kWindowSizeMs / 1000))),
+          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)),
@@ -85,9 +72,7 @@ IntelligibilityEnhancer::IntelligibilityEnhancer(int erb_resolution,
       channels_(channels),
       analysis_rate_(analysis_rate),
       variance_rate_(variance_rate),
-      clear_variance_(freqs_,
+      clear_variance_(freqs_, static_cast<VarianceType>(cv_type), cv_win,
-                      static_cast<VarianceType>(cv_type),
-                      cv_win,
                       cv_alpha),
       noise_variance_(freqs_, VarianceType::kStepInfinite, 475, 0.01f),
       filtered_clear_var_(new float[bank_size_]),
@@ -98,51 +83,58 @@ IntelligibilityEnhancer::IntelligibilityEnhancer(int erb_resolution,
       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_(WebRtcVad_Create()),
+      vad_high_(nullptr),
-      vad_low_(WebRtcVad_Create()),
+      vad_low_(nullptr),
       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_set_mode(vad_high_, 0);  // high likelihood of speech
+  WebRtcVad_Create(&vad_low_);
   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**>(
+  temp_out_buffer_ = static_cast<float**>(malloc(
-      malloc(sizeof(*temp_out_buffer_) * channels_ +
+      sizeof(*temp_out_buffer_) * channels_ +
-             sizeof(**temp_out_buffer_) * chunk_length_ * channels_));
+      sizeof(**temp_out_buffer_) * chunk_length_ * channels_));
   for (int i = 0; i < channels_; ++i) {
-    temp_out_buffer_[i] =
+    temp_out_buffer_[i] = reinterpret_cast<float*>(temp_out_buffer_ + channels_)
-        reinterpret_cast<float*>(temp_out_buffer_ + channels_) +
+        + chunk_length_ * i;
-        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(
+  int erb_index = static_cast<int>(ceilf(11.17f * logf((freqs_khz + 0.312f) /
-      11.17f * logf((freqs_khz + 0.312f) / (freqs_khz + 14.6575f)) + 43.0f));
+                                                       (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(
+  render_mangler_.reset(new LappedTransform(channels_, channels_,
-      channels_, channels_, chunk_length_, kbd_window_.get(), window_size_,
+                                            chunk_length_,
-      window_size_ / 2, &render_callback_));
+                                            kbd_window_.get(),
-  capture_mangler_.reset(new LappedTransform(
+                                            window_size_,
-      channels_, channels_, chunk_length_, kbd_window_.get(), window_size_,
+                                            window_size_ / 2,
-      window_size_ / 2, &capture_callback_));
+                                            &render_callback_));
+  capture_mangler_.reset(new LappedTransform(channels_, channels_,
+                                             chunk_length_,
+                                             kbd_window_.get(),
+                                             window_size_,
+                                             window_size_ / 2,
+                                             &capture_callback_));
 }
 
 IntelligibilityEnhancer::~IntelligibilityEnhancer() {
@@ -158,9 +150,7 @@ void IntelligibilityEnhancer::ProcessRenderAudio(float* const* audio) {
   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_));
@@ -171,25 +161,21 @@ void IntelligibilityEnhancer::ProcessCaptureAudio(float* const* audio) {
   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,
+  // TODO(bercic): the VAD was always detecting voice in the noise stream,
-  // no matter what the aggressiveness, so it was temporarily disabled here.
+  // no matter what the aggressiveness, so it was temporarily disabled here
-
-  #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
 
+  //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");
   capture_mangler_->ProcessChunk(audio, temp_out_buffer_);
 }
 
 void IntelligibilityEnhancer::DispatchAudio(
     IntelligibilityEnhancer::AudioSource source,
-    const complex<float>* in_block,
+    const complex<float>* in_block, complex<float>* out_block) {
-    complex<float>* out_block) {
   switch (source) {
     case kRenderStream:
       ProcessClearBlock(in_block, out_block);
@@ -210,9 +196,6 @@ void IntelligibilityEnhancer::ProcessClearBlock(const complex<float>* in_block,
     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(),
@@ -238,25 +221,23 @@ void IntelligibilityEnhancer::AnalyzeClearBlock(float power_target) {
   FilterVariance(clear_variance_.variance(), filtered_clear_var_.get());
   FilterVariance(noise_variance_.variance(), filtered_noise_var_.get());
 
-  // Bisection search for optimal |lambda|
+  /* lambda binary search */
 
   float lambda_bot = -1.0f, lambda_top = -10e-18f, lambda;
   float power_bot, power_top, power;
-  SolveForGainsGivenLambda(lambda_top, start_freq_, gains_eq_.get());
+  SolveEquation14(lambda_top, start_freq_, gains_eq_.get());
-  power_top =
+  power_top = DotProduct(gains_eq_.get(), filtered_clear_var_.get(),
-      DotProduct(gains_eq_.get(), filtered_clear_var_.get(), bank_size_);
+                         bank_size_);
-  SolveForGainsGivenLambda(lambda_bot, start_freq_, gains_eq_.get());
+  SolveEquation14(lambda_bot, start_freq_, gains_eq_.get());
-  power_bot =
+  power_bot = DotProduct(gains_eq_.get(), filtered_clear_var_.get(),
-      DotProduct(gains_eq_.get(), filtered_clear_var_.get(), bank_size_);
+                         bank_size_);
   DCHECK(power_target >= power_bot && power_target <= power_top);
 
-  float power_ratio = 2.0f;  // Ratio of achieved power to target power.
+  float power_ratio = 2.0f;
-  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) > kConvergeThresh && iters <= kMaxIters) {
+  while (fabs(power_ratio - 1.0f) > 0.001f && iters <= 100) {
     lambda = lambda_bot + (lambda_top - lambda_bot) / 2.0f;
-    SolveForGainsGivenLambda(lambda, start_freq_, gains_eq_.get());
+    SolveEquation14(lambda, start_freq_, gains_eq_.get());
     power = DotProduct(gains_eq_.get(), filtered_clear_var_.get(), bank_size_);
     if (power < power_target) {
       lambda_bot = lambda;
@@ -267,7 +248,7 @@ void IntelligibilityEnhancer::AnalyzeClearBlock(float power_target) {
     ++iters;
   }
 
-  // (ERB gain) = filterbank' * (freq gain)
+  /* b = filterbank' * b */
   float* gains = gain_applier_.target();
   for (int i = 0; i < freqs_; ++i) {
     gains[i] = 0.0f;
@@ -284,8 +265,8 @@ void IntelligibilityEnhancer::ProcessNoiseBlock(const complex<float>* in_block,
 
 int IntelligibilityEnhancer::GetBankSize(int sample_rate, int erb_resolution) {
   float freq_limit = sample_rate / 2000.0f;
-  int erb_scale = ceilf(
+  int erb_scale = ceilf(11.17f * logf((freq_limit + 0.312f) /
-      11.17f * logf((freq_limit + 0.312f) / (freq_limit + 14.6575f)) + 43.0f);
+                                      (freq_limit + 14.6575f)) + 43.0f);
   return erb_scale * erb_resolution;
 }
 
@@ -302,29 +283,29 @@ void IntelligibilityEnhancer::CreateErbBank() {
     center_freqs_[i] *= 0.5f * sample_rate_hz_ / last_center_freq;
   }
 
-  filter_bank_ = static_cast<float**>(
+  filter_bank_ = static_cast<float**>(malloc(
-      malloc(sizeof(*filter_bank_) * bank_size_ +
+      sizeof(*filter_bank_) * bank_size_ +
-             sizeof(**filter_bank_) * freqs_ * bank_size_));
+      sizeof(**filter_bank_) * freqs_ * bank_size_));
   for (int i = 0; i < bank_size_; ++i) {
-    filter_bank_[i] =
+    filter_bank_[i] = reinterpret_cast<float*>(filter_bank_ + bank_size_) +
-        reinterpret_cast<float*>(filter_bank_ + bank_size_) + freqs_ * i;
+        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_));
+        (0.5f * sample_rate_hz_));
-    ll =
+    ll  = round(center_freqs_[max(1, i     ) - 1] * freqs_ /
-        round(center_freqs_[max(1, i) - 1] * freqs_ / (0.5f * sample_rate_hz_));
+        (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_));
+        (0.5f * sample_rate_hz_));
-    rr = round(center_freqs_[min(bank_size_, i + 1) - 1] * freqs_ /
+    rr  = round(center_freqs_[min(bank_size_, i + 1)  - 1] * freqs_ /
-               (0.5f * sample_rate_hz_));
+        (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;
 
@@ -357,9 +338,8 @@ void IntelligibilityEnhancer::CreateErbBank() {
   }
 }
 
-void IntelligibilityEnhancer::SolveForGainsGivenLambda(float lambda,
+void IntelligibilityEnhancer::SolveEquation14(float lambda, int start_freq,
-                                                       int start_freq,
+                                              float* sols) {
-                                                       float* sols) {
   bool quadratic = (kConfigRho < 1.0f);
   const float* var_x0 = filtered_clear_var_.get();
   const float* var_n0 = filtered_noise_var_.get();
@@ -367,17 +347,15 @@ void IntelligibilityEnhancer::SolveForGainsGivenLambda(float lambda,
   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] =
+      sols[n] = (-beta0 - sqrtf(beta0 * beta0 - 4 * alpha0 * gamma0))
-          (-beta0 - sqrtf(beta0 * beta0 - 4 * alpha0 * gamma0)) / (2 * alpha0);
+          / (2 * alpha0);
     } else {
       sols[n] = -gamma0 / beta0;
     }
@@ -391,9 +369,8 @@ void IntelligibilityEnhancer::FilterVariance(const float* var, float* result) {
   }
 }
 
-float IntelligibilityEnhancer::DotProduct(const float* a,
+float IntelligibilityEnhancer::DotProduct(const float* a, const float* b,
-                                          const float* b,
+                                               int length) {
-                                          int length) {
   float ret = 0.0f;
 
   for (int i = 0; i < length; ++i) {
@@ -403,3 +380,4 @@ float IntelligibilityEnhancer::DotProduct(const float* a,
 }
 
 }  // namespace webrtc
+

webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h

@@ -8,18 +8,14 @@
  *  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;
@@ -29,7 +25,6 @@ namespace webrtc {
 // 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,
@@ -38,43 +33,30 @@ class IntelligibilityEnhancer {
   // 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
-  // clear speech stream.
+  // lear 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,
+  IntelligibilityEnhancer(int erb_resolution, int sample_rate_hz, int channels,
-                          int sample_rate_hz,
+                          int cv_type, float cv_alpha, int cv_win,
-                          int channels,
+                          int analysis_rate, int variance_rate,
-                          int cv_type,
-                          float cv_alpha,
-                          int cv_win,
-                          int analysis_rate,
-                          int variance_rate,
                           float gain_limit);
   ~IntelligibilityEnhancer();
 
-  // 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);
+  void ProcessCaptureAudio(float* const* audio);
 
  private:
   enum AudioSource {
-    kRenderStream = 0,  // Clear speech stream.
+    kRenderStream = 0,
-    kCaptureStream,  // Noise stream.
+    kCaptureStream,
   };
 
-  // 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 in_channels, int frames,
-                                   int frames,
                                    int out_channels,
                                    std::complex<float>* const* out_block);
 
@@ -84,95 +66,72 @@ class IntelligibilityEnhancer {
   };
   friend class TransformCallback;
 
-  // Sends streams to ProcessClearBlock or ProcessNoiseBlock based on source.
+  void DispatchAudio(AudioSource source, const std::complex<float>* in_block,
-  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;   // Default for |analysis_rate_|.
+  static const int kAnalyzeRate;
-  static const int kVarianceRate;  // Default for |variance_rate_|.
+  static const int kVarianceRate;
   static const float kClipFreq;
-  static const float kConfigRho;  // Default production and interpretation SNR.
+  static const float kConfigRho;
   static const float kKbdAlpha;
   static const float kGainChangeLimit;
 
-  const int freqs_;         // Num frequencies in frequency domain.
+  const int freqs_;
-  const int window_size_;   // Window size in samples; also the block size.
+  const int window_size_;  // window size in samples; also the block size
-  const int chunk_length_;  // Chunk size in samples.
+  const int chunk_length_;  // chunk size in samples
-  const int bank_size_;     // Num ERB filters.
+  const int bank_size_;
   const int sample_rate_hz_;
   const int erb_resolution_;
-  const int channels_;       // Num channels.
+  const int channels_;
-  const int analysis_rate_;  // Num blocks before gains recalculated.
+  const int analysis_rate_;
-  const int variance_rate_;  // Num recalculations before history is cleared.
+  const int variance_rate_;
 
   intelligibility::VarianceArray clear_variance_;
   intelligibility::VarianceArray noise_variance_;
-  rtc::scoped_ptr<float[]> filtered_clear_var_;
+  scoped_ptr<float[]> filtered_clear_var_;
-  rtc::scoped_ptr<float[]> filtered_noise_var_;
+  scoped_ptr<float[]> filtered_noise_var_;
-  float** filter_bank_;  // TODO(ekmeyerson): Switch to using ChannelBuffer.
+  float** filter_bank_;
-  rtc::scoped_ptr<float[]> center_freqs_;
+  scoped_ptr<float[]> center_freqs_;
   int start_freq_;
-  rtc::scoped_ptr<float[]> rho_;  // Production and interpretation SNR.
+  scoped_ptr<float[]> rho_;
-                                  // for each ERB band.
+  scoped_ptr<float[]> gains_eq_;
-  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_;
-  rtc::scoped_ptr<float* []> input_audio_;
+  scoped_ptr<float[]> kbd_window_;
-  rtc::scoped_ptr<float[]> kbd_window_;
   TransformCallback render_callback_;
   TransformCallback capture_callback_;
-  rtc::scoped_ptr<LappedTransform> render_mangler_;
+  scoped_ptr<LappedTransform> render_mangler_;
-  rtc::scoped_ptr<LappedTransform> capture_mangler_;
+  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_;
-  rtc::scoped_ptr<int16_t[]> vad_tmp_buffer_;
+  scoped_ptr<int16_t[]> vad_tmp_buffer_;
-  bool has_voice_low_;  // Whether voice detected in speech stream.
+  bool has_voice_low_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_INTELLIGIBILITY_INTELLIGIBILITY_ENHANCER_H_
+

webrtc/modules/audio_processing/intelligibility/intelligibility_proc.cc

@@ -8,12 +8,6 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-//
-//  Command line tool for speech intelligibility enhancement. Provides for
-//  running and testing intelligibility_enhancer as an independent process.
-//  Use --help for options.
-//
-
 #include <arpa/inet.h>
 #include <fcntl.h>
 #include <stdint.h>
@@ -30,71 +24,53 @@
 #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/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"
 
-// PCM data simulating streams
 const int16_t* in_ipcm;
 int16_t* out_ipcm;
 const int16_t* noise_ipcm;
+
 float* in_fpcm;
 float* out_fpcm;
 float* noise_fpcm;
-
-// Current locations in streams
 float* noise_cursor;
 float* clear_cursor;
 
-int samples;        // Number of samples in input PCM file
+int samples;
-int fragment_size;  // Number of samples to process at a time
+int fragment_size;
-                    // to simulate APM stream processing
 
 using std::complex;
-
-namespace webrtc {
-
 using webrtc::RealFourier;
 using webrtc::IntelligibilityEnhancer;
 
-DEFINE_int32(clear_type,
+DEFINE_int32(clear_type, webrtc::intelligibility::VarianceArray::kStepInfinite,
-             webrtc::intelligibility::VarianceArray::kStepInfinite,
              "Variance algorithm for clear data.");
-DEFINE_double(clear_alpha, 0.9, "Variance decay factor for clear data.");
+DEFINE_double(clear_alpha, 0.9,
-DEFINE_int32(clear_window,
+              "Variance decay factor for clear data.");
-             475,
+DEFINE_int32(clear_window, 475,
              "Window size for windowed variance for clear data.");
-DEFINE_int32(sample_rate,
+DEFINE_int32(sample_rate, 16000,
-             16000,
              "Audio sample rate used in the input and output files.");
-DEFINE_int32(ana_rate,
+DEFINE_int32(ana_rate, 800,
-             800,
              "Analysis rate; gains recalculated every N blocks.");
-DEFINE_int32(
+DEFINE_int32(var_rate, 2,
-    var_rate,
+             "Variance clear rate; history is forgotten every N gain recalculations.");
-    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.pcm", "Input file with clear speech.");
 DEFINE_string(noise_file, "noise.pcm", "Input file with noise data.");
-DEFINE_string(out_file,
+DEFINE_string(out_file, "proc_enhanced.pcm", "Enhanced output. Use '-' to "
-              "proc_enhanced.pcm",
-              "Enhanced output. Use '-' to "
               "pipe through aplay internally.");
 
-// Constant IntelligibilityEnhancer constructor parameters.
+// Write an Sun AU-formatted audio chunk into file descriptor |fd|. Can be used
-const int kErbResolution = 2;
+// to pipe the audio stream directly into aplay.
-const int kNumChannels = 1;
-
-// Converts output stream to Sun AU format and writes to file descriptor |fd|.
-// Can be used to pipe output directly into aplay.
-// TODO(ekmeyerson): Modify to write WAV instead.
 void writeau(int fd) {
   uint32_t thing;
 
@@ -116,14 +92,12 @@ void writeau(int fd) {
   write(fd, out_ipcm, sizeof(*out_ipcm) * samples);
 }
 
-// void function for gtest
+int main(int argc, char* argv[]) {
-void void_main(int argc, char* argv[]) {
+  google::SetUsageMessage("\n\nVariance algorithm types are:\n"
-  google::SetUsageMessage(
+                          "  0 - infinite/normal,\n"
-      "\n\nVariance algorithm types are:\n"
+                          "  1 - exponentially decaying,\n"
-      "  0 - infinite/normal,\n"
+                          "  2 - rolling window.\n"
-      "  1 - exponentially decaying,\n"
+                          "\nInput files must be little-endian 16-bit signed raw PCM.\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();
@@ -133,15 +107,10 @@ void void_main(int argc, char* argv[]) {
   int in_fd, out_fd, noise_fd;
   FILE* aplay_file = nullptr;
 
-  // Load settings and set up PCMs.
+  fragment_size = FLAGS_sample_rate / 100;
-
-  fragment_size = FLAGS_sample_rate / 100;  // Mirror real time APM chunk size.
-                                            // Duplicates chunk_length_ in
-                                            // IntelligibilityEnhancer.
-
-  ASSERT_EQ(stat(in_name, &in_stat), 0) << "Empty speech input.";
-  ASSERT_EQ(stat(noise_name, &noise_stat), 0) << "Empty noise input.";
 
+  stat(in_name, &in_stat);
+  stat(noise_name, &noise_stat);
   samples = in_stat.st_size / sizeof(*in_ipcm);
 
   in_fd = open(in_name, O_RDONLY);
@@ -154,10 +123,10 @@ void void_main(int argc, char* argv[]) {
   }
   noise_fd = open(noise_name, O_RDONLY);
 
-  in_ipcm = static_cast<int16_t*>(
+  in_ipcm = static_cast<int16_t*>(mmap(nullptr, in_stat.st_size, PROT_READ,
-      mmap(nullptr, in_stat.st_size, PROT_READ, MAP_PRIVATE, in_fd, 0));
+                                       MAP_PRIVATE, in_fd, 0));
-  noise_ipcm = static_cast<int16_t*>(
+  noise_ipcm = static_cast<int16_t*>(mmap(nullptr, noise_stat.st_size,
-      mmap(nullptr, noise_stat.st_size, PROT_READ, MAP_PRIVATE, noise_fd, 0));
+                                          PROT_READ, MAP_PRIVATE, noise_fd, 0));
   out_ipcm = new int16_t[samples];
   out_fpcm = new float[samples];
   in_fpcm = new float[samples];
@@ -167,17 +136,18 @@ void void_main(int argc, char* argv[]) {
     noise_fpcm[i] = noise_ipcm[i % (noise_stat.st_size / sizeof(*noise_ipcm))];
   }
 
-  // Run intelligibility enhancement.
+  //feenableexcept(FE_INVALID | FE_OVERFLOW);
-
+  IntelligibilityEnhancer enh(2,
-  IntelligibilityEnhancer enh(
+                              FLAGS_sample_rate, 1,
-      kErbResolution,
+                              FLAGS_clear_type,
-      FLAGS_sample_rate,
+                              static_cast<float>(FLAGS_clear_alpha),
-      kNumChannels,
+                              FLAGS_clear_window,
-      FLAGS_clear_type, static_cast<float>(FLAGS_clear_alpha),
+                              FLAGS_ana_rate,
-      FLAGS_clear_window, FLAGS_ana_rate, FLAGS_var_rate, FLAGS_gain_limit);
+                              FLAGS_var_rate,
+                              FLAGS_gain_limit);
 
   // Slice the input into smaller chunks, as the APM would do, and feed them
-  // through the enhancer. Repeat indefinitely if FLAGS_repeat is set.
+  // into the enhancer. Repeat indefinitely if FLAGS_repeat is set.
   do {
     noise_cursor = noise_fpcm;
     clear_cursor = in_fpcm;
@@ -211,11 +181,7 @@ void void_main(int argc, char* argv[]) {
     close(out_fd);
   }
   close(in_fd);
-}
 
-}  // namespace webrtc
-
-int main(int argc, char* argv[]) {
-  webrtc::void_main(argc, argv);
   return 0;
 }
+

webrtc/modules/audio_processing/intelligibility/intelligibility_utils.cc

@@ -8,10 +8,6 @@
  *  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>
@@ -44,13 +40,10 @@ inline bool cplxnormal(complex<float> c) {
 // 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},
+  const static complex<float> fudge[7] = {
-                                          {0.008f, 0.001f},
+    {0.001f, 0.002f}, {0.008f, 0.001f}, {0.003f, 0.008f}, {0.0006f, 0.0009f},
-                                          {0.003f, 0.008f},
+    {0.001f, 0.004f}, {0.003f, 0.004f}, {0.002f, 0.009f}
-                                          {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;
@@ -61,9 +54,8 @@ inline complex<float> zerofudge(complex<float> c) {
 
 // Incremental mean computation. Return the mean of the series with the
 // mean |mean| with added |data|.
-inline complex<float> NewMean(complex<float> mean,
+inline complex<float> NewMean(complex<float> mean, complex<float> data,
-                              complex<float> data,
+                                 int count) {
-                              int count) {
   return mean + (data - mean) / static_cast<float>(count);
 }
 
@@ -81,9 +73,7 @@ namespace intelligibility {
 
 static const int kWindowBlockSize = 10;
 
-VarianceArray::VarianceArray(int freqs,
+VarianceArray::VarianceArray(int freqs, StepType type, int window_size,
-                             StepType type,
-                             int window_size,
                              float decay)
     : running_mean_(new complex<float>[freqs]()),
       running_mean_sq_(new complex<float>[freqs]()),
@@ -97,15 +87,15 @@ VarianceArray::VarianceArray(int freqs,
       history_cursor_(0),
       count_(0),
       array_mean_(0.0f) {
-  history_.reset(new rtc::scoped_ptr<complex<float>[]>[freqs_]());
+  history_.reset(new scoped_ptr<complex<float>[]>[freqs_]());
   for (int i = 0; i < freqs_; ++i) {
     history_[i].reset(new complex<float>[window_size_]());
   }
-  subhistory_.reset(new rtc::scoped_ptr<complex<float>[]>[freqs_]());
+  subhistory_.reset(new scoped_ptr<complex<float>[]>[freqs_]());
   for (int i = 0; i < freqs_; ++i) {
     subhistory_[i].reset(new complex<float>[window_size_]());
   }
-  subhistory_sq_.reset(new rtc::scoped_ptr<complex<float>[]>[freqs_]());
+  subhistory_sq_.reset(new scoped_ptr<complex<float>[]>[freqs_]());
   for (int i = 0; i < freqs_; ++i) {
     subhistory_sq_[i].reset(new complex<float>[window_size_]());
   }
@@ -141,15 +131,13 @@ void VarianceArray::InfiniteStep(const complex<float>* data, bool skip_fudge) {
     } else {
       float old_sum = conj_sum_[i];
       complex<float> old_mean = running_mean_[i];
-      running_mean_[i] =
+      running_mean_[i] = old_mean + (sample - old_mean) /
-          old_mean + (sample - old_mean) / static_cast<float>(count_);
+          static_cast<float>(count_);
-      conj_sum_[i] =
+      conj_sum_[i] = (old_sum + std::conj(sample - old_mean) *
-          (old_sum + std::conj(sample - old_mean) * (sample - running_mean_[i]))
+          (sample - running_mean_[i])).real();
-              .real();
+      variance_[i] = conj_sum_[i] / (count_ - 1); // + fudge[fudge_index].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);
@@ -173,13 +161,11 @@ void VarianceArray::DecayStep(const complex<float>* data, bool /*dummy*/) {
       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] =
+      running_mean_sq_[i] = decay_ * prev2 +
-          decay_ * prev2 + (1.0f - decay_) * sample * std::conj(sample);
+        (1.0f - decay_) * sample * std::conj(sample);
-      // variance_[i] = decay_ * variance_[i] + (1.0f - decay_) * (
+      //variance_[i] = decay_ * variance_[i] + (1.0f - decay_) * (
-      //  (sample - running_mean_[i]) * std::conj(sample -
+      //  (sample - running_mean_[i]) * std::conj(sample - running_mean_[i])).real();
-      //  running_mean_[i])).real();
+      variance_[i] = (running_mean_sq_[i] - running_mean_[i] * std::conj(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);
@@ -200,15 +186,15 @@ void VarianceArray::WindowedStep(const complex<float>* data, bool /*dummy*/) {
     mean = history_[i][history_cursor_];
     variance_[i] = 0.0f;
     for (int j = 1; j < num; ++j) {
-      complex<float> sample =
+      complex<float> sample = zerofudge(
-          zerofudge(history_[i][(history_cursor_ + j) % window_size_]);
+          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 =
+      conj_sum = (old_sum + std::conj(sample - old_mean) *
-          (old_sum + std::conj(sample - old_mean) * (sample - mean)).real();
+                                     (sample - mean)).real();
       variance_[i] = conj_sum / (j);
     }
     array_mean_ += (variance_[i] - array_mean_) / (i + 1);
@@ -231,11 +217,11 @@ void VarianceArray::BlockedStep(const complex<float>* data, bool /*dummy*/) {
     subhistory_[i][history_cursor_ % window_size_] = sub_running_mean_[i];
     subhistory_sq_[i][history_cursor_ % window_size_] = sub_running_mean_sq_[i];
 
-    variance_[i] =
+    variance_[i] = (NewMean(running_mean_sq_[i], sub_running_mean_sq_[i],
-        (NewMean(running_mean_sq_[i], sub_running_mean_sq_[i], blocks) -
+                            blocks) -
-         NewMean(running_mean_[i], sub_running_mean_[i], blocks) *
+                   NewMean(running_mean_[i], sub_running_mean_[i], blocks) *
-             std::conj(NewMean(running_mean_[i], sub_running_mean_[i], blocks)))
+                   std::conj(NewMean(running_mean_[i], sub_running_mean_[i],
-            .real();
+                                     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);
@@ -298,3 +284,4 @@ void GainApplier::Apply(const complex<float>* in_block,
 }  // namespace intelligibility
 
 }  // namespace webrtc
+

webrtc/modules/audio_processing/intelligibility/intelligibility_utils.h

@@ -8,16 +8,12 @@
  *  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/base/scoped_ptr.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 
 namespace webrtc {
 
@@ -67,10 +63,14 @@ class VarianceArray {
   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);
@@ -78,26 +78,23 @@ class VarianceArray {
   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.
-  rtc::scoped_ptr<std::complex<float>[]> running_mean_;
+  scoped_ptr<std::complex<float>[]> running_mean_;
-  rtc::scoped_ptr<std::complex<float>[]> running_mean_sq_;
+  scoped_ptr<std::complex<float>[]> running_mean_sq_;
 
   // Average X and X^2 for the current block in kStepBlocked.
-  rtc::scoped_ptr<std::complex<float>[]> sub_running_mean_;
+  scoped_ptr<std::complex<float>[]> sub_running_mean_;
-  rtc::scoped_ptr<std::complex<float>[]> sub_running_mean_sq_;
+  scoped_ptr<std::complex<float>[]> sub_running_mean_sq_;
 
   // Sample history for the rolling window in kStepWindowed and block-wise
   // histories for kStepBlocked.
-  rtc::scoped_ptr<rtc::scoped_ptr<std::complex<float>[]>[]> history_;
+  scoped_ptr<scoped_ptr<std::complex<float>[]>[]> history_;
-  rtc::scoped_ptr<rtc::scoped_ptr<std::complex<float>[]>[]> subhistory_;
+  scoped_ptr<scoped_ptr<std::complex<float>[]>[]> subhistory_;
-  rtc::scoped_ptr<rtc::scoped_ptr<std::complex<float>[]>[]> subhistory_sq_;
+  scoped_ptr<scoped_ptr<std::complex<float>[]>[]> subhistory_sq_;
 
   // The current set of variances and sums for Welford's algorithm.
-  rtc::scoped_ptr<float[]> variance_;
+  scoped_ptr<float[]> variance_;
-  rtc::scoped_ptr<float[]> conj_sum_;
+  scoped_ptr<float[]> conj_sum_;
 
   const int freqs_;
   const int window_size_;
@@ -121,13 +118,15 @@ class GainApplier {
              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_;
-  rtc::scoped_ptr<float[]> target_;
+  scoped_ptr<float[]> target_;
-  rtc::scoped_ptr<float[]> current_;
+  scoped_ptr<float[]> current_;
 };
 
 }  // namespace intelligibility
@@ -135,3 +134,4 @@ class GainApplier {
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_INTELLIGIBILITY_INTELLIGIBILITY_UTILS_H_
+