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 "modules/audio_processing/low_cut_filter.h"

#include <stdint.h>
#include <cstring>

#include "common_audio/signal_processing/include/signal_processing_library.h"
#include "modules/audio_processing/audio_buffer.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "rtc_base/checks.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 LowCutFilter::BiquadFilter {
 public:
  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 = (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 *= 2;

      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 & 0x00001FFF) * 4);

      // 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,
                                 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_;
  int16_t x_[2];
  int16_t y_[4];
};

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));
  }
}

LowCutFilter::~LowCutFilter() {}

void LowCutFilter::Process(AudioBuffer* audio) {
  RTC_DCHECK(audio);
  RTC_DCHECK_GE(160, 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());
  }
}

}  // namespace webrtc