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/*
* 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/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 {
public:
explicit BiquadFilter(int sample_rate_hz) :
ba_(sample_rate_hz == AudioProcessing::kSampleRate8kHz ?
kFilterCoefficients8kHz : kFilterCoefficients)
{
Reset();
}
void Reset() {
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 = (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,
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_);
}
HighPassFilterImpl::~HighPassFilterImpl() {}
void HighPassFilterImpl::Initialize(size_t channels, int sample_rate_hz) {
std::vector<rtc::scoped_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) {
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
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