| Index: webrtc/modules/audio_processing/low_cut_filter.cc
|
| diff --git a/webrtc/modules/audio_processing/high_pass_filter_impl.cc b/webrtc/modules/audio_processing/low_cut_filter.cc
|
| similarity index 63%
|
| rename from webrtc/modules/audio_processing/high_pass_filter_impl.cc
|
| rename to webrtc/modules/audio_processing/low_cut_filter.cc
|
| index d33ec78dcbca29b7214d338600864812a0d05e03..77dab9af43c735e328077d4eb030104ca9006c7b 100644
|
| --- a/webrtc/modules/audio_processing/high_pass_filter_impl.cc
|
| +++ b/webrtc/modules/audio_processing/low_cut_filter.cc
|
| @@ -8,7 +8,7 @@
|
| * 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"
|
| @@ -20,16 +20,12 @@ 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_));
|
| }
|
| @@ -44,11 +40,11 @@ class HighPassFilterImpl::BiquadFilter {
|
| // 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]
|
| @@ -70,8 +66,7 @@ class HighPassFilterImpl::BiquadFilter {
|
| 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.
|
| @@ -85,29 +80,17 @@ class HighPassFilterImpl::BiquadFilter {
|
| 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<std::unique_ptr<BiquadFilter>> new_filters(channels);
|
| +LowCutFilter::LowCutFilter(size_t channels, int sample_rate_hz) {
|
| + filters_.resize(channels);
|
| for (size_t i = 0; i < channels; i++) {
|
| - new_filters[i].reset(new BiquadFilter(sample_rate_hz));
|
| + 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;
|
| - }
|
| +LowCutFilter::~LowCutFilter() {}
|
|
|
| +void LowCutFilter::Process(AudioBuffer* audio) {
|
| + RTC_DCHECK(audio);
|
| 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++) {
|
| @@ -116,19 +99,4 @@ void HighPassFilterImpl::ProcessCaptureAudio(AudioBuffer* audio) {
|
| }
|
| }
|
|
|
| -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
|
|
|