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