Introduced the new parameter setting scheme for activating the high-pass filter in APM

webrtc/modules/audio_processing/low_cut_filter.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_processing/high_pass_filter_impl.h"
+#include "webrtc/modules/audio_processing/low_cut_filter.h"
 
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_processing/audio_buffer.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 namespace {
 const int16_t kFilterCoefficients8kHz[5] = {3798, -7596, 3798, 7807, -3733};
 const int16_t kFilterCoefficients[5] = {4012, -8024, 4012, 8002, -3913};
 }  // namespace
 
-class HighPassFilterImpl::BiquadFilter {
+class LowCutFilter::BiquadFilter {
  public:
-  explicit BiquadFilter(int sample_rate_hz) :
-      ba_(sample_rate_hz == AudioProcessing::kSampleRate8kHz ?
-          kFilterCoefficients8kHz : kFilterCoefficients)
-  {
-    Reset();
-  }
-
-  void Reset() {
+  explicit BiquadFilter(int sample_rate_hz)
+      : ba_(sample_rate_hz == AudioProcessing::kSampleRate8kHz
+                ? kFilterCoefficients8kHz
+                : kFilterCoefficients) {
     std::memset(x_, 0, sizeof(x_));
     std::memset(y_, 0, sizeof(y_));
   }
 
   void Process(int16_t* data, size_t length) {
     const int16_t* const ba = ba_;
     int16_t* x = x_;
     int16_t* y = y_;
     int32_t tmp_int32 = 0;
 
     for (size_t i = 0; i < length; i++) {
       //  y[i] = b[0] * x[i] +  b[1] * x[i-1] +  b[2] * x[i-2]
       //                     + -a[1] * y[i-1] + -a[2] * y[i-2];
 
-      tmp_int32 = y[1] * ba[3];      // -a[1] * y[i-1] (low part)
-      tmp_int32 += y[3] * ba[4];     // -a[2] * y[i-2] (low part)
+      tmp_int32 = y[1] * ba[3];   // -a[1] * y[i-1] (low part)
+      tmp_int32 += y[3] * ba[4];  // -a[2] * y[i-2] (low part)
       tmp_int32 = (tmp_int32 >> 15);
-      tmp_int32 += y[0] * ba[3];     // -a[1] * y[i-1] (high part)
-      tmp_int32 += y[2] * ba[4];     // -a[2] * y[i-2] (high part)
+      tmp_int32 += y[0] * ba[3];  // -a[1] * y[i-1] (high part)
+      tmp_int32 += y[2] * ba[4];  // -a[2] * y[i-2] (high part)
       tmp_int32 = (tmp_int32 << 1);
 
       tmp_int32 += data[i] * ba[0];  // b[0] * x[0]
       tmp_int32 += x[0] * ba[1];     // b[1] * x[i-1]
       tmp_int32 += x[1] * ba[2];     // b[2] * x[i-2]
 
       // Update state (input part).
       x[1] = x[0];
       x[0] = data[i];
 
       // Update state (filtered part).
       y[2] = y[0];
       y[3] = y[1];
       y[0] = static_cast<int16_t>(tmp_int32 >> 13);
       y[1] = static_cast<int16_t>(
           (tmp_int32 - (static_cast<int32_t>(y[0]) << 13)) << 2);
 
       // Rounding in Q12, i.e. add 2^11.
       tmp_int32 += 2048;
 
       // Saturate (to 2^27) so that the HP filtered signal does not overflow.
-      tmp_int32 = WEBRTC_SPL_SAT(static_cast<int32_t>(134217727),
-                                 tmp_int32,
+      tmp_int32 = WEBRTC_SPL_SAT(static_cast<int32_t>(134217727), tmp_int32,
                                  static_cast<int32_t>(-134217728));
 
       // Convert back to Q0 and use rounding.
       data[i] = static_cast<int16_t>(tmp_int32 >> 12);
     }
   }
 
  private:
   const int16_t* const ba_ = nullptr;
   int16_t x_[2];
   int16_t y_[4];
 };
 
-HighPassFilterImpl::HighPassFilterImpl(rtc::CriticalSection* crit)
-    : crit_(crit) {
-  RTC_DCHECK(crit_);
+LowCutFilter::LowCutFilter(size_t channels, int sample_rate_hz) {
+  filters_.resize(channels);
+  for (size_t i = 0; i < channels; i++) {
+    filters_[i].reset(new BiquadFilter(sample_rate_hz));
+  }
 }
 
-HighPassFilterImpl::~HighPassFilterImpl() {}
+LowCutFilter::~LowCutFilter() {}
 
-void HighPassFilterImpl::Initialize(size_t channels, int sample_rate_hz) {
-  std::vector<std::unique_ptr<BiquadFilter>> new_filters(channels);
-  for (size_t i = 0; i < channels; i++) {
-    new_filters[i].reset(new BiquadFilter(sample_rate_hz));
-  }
-  rtc::CritScope cs(crit_);
-  filters_.swap(new_filters);
-}
-
-void HighPassFilterImpl::ProcessCaptureAudio(AudioBuffer* audio) {
+void LowCutFilter::Process(AudioBuffer* audio) {
   RTC_DCHECK(audio);
-  rtc::CritScope cs(crit_);
-  if (!enabled_) {
-    return;
-  }
-
   RTC_DCHECK_GE(160u, audio->num_frames_per_band());
   RTC_DCHECK_EQ(filters_.size(), audio->num_channels());
   for (size_t i = 0; i < filters_.size(); i++) {
     filters_[i]->Process(audio->split_bands(i)[kBand0To8kHz],
                          audio->num_frames_per_band());
   }
 }
 
-int HighPassFilterImpl::Enable(bool enable) {
-  rtc::CritScope cs(crit_);
-  if (!enabled_ && enable) {
-    for (auto& filter : filters_) {
-      filter->Reset();
-    }
-  }
-  enabled_ = enable;
-  return AudioProcessing::kNoError;
-}
-
-bool HighPassFilterImpl::is_enabled() const {
-  rtc::CritScope cs(crit_);
-  return enabled_;
-}
 }  // namespace webrtc