/* * 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 #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" #include "webrtc/typedefs.h" namespace webrtc { namespace { const int16_t kFilterCoefficients8kHz[5] = {3798, -7596, 3798, 7807, -3733}; const int16_t kFilterCoefficients[5] = {4012, -8024, 4012, 8002, -3913}; struct FilterState { int16_t y[4]; int16_t x[2]; const int16_t* ba; }; int InitializeFilter(FilterState* hpf, int sample_rate_hz) { assert(hpf != NULL); if (sample_rate_hz == AudioProcessing::kSampleRate8kHz) { hpf->ba = kFilterCoefficients8kHz; } else { hpf->ba = kFilterCoefficients; } WebRtcSpl_MemSetW16(hpf->x, 0, 2); WebRtcSpl_MemSetW16(hpf->y, 0, 4); return AudioProcessing::kNoError; } int Filter(FilterState* hpf, int16_t* data, size_t length) { assert(hpf != NULL); int32_t tmp_int32 = 0; int16_t* y = hpf->y; int16_t* x = hpf->x; const int16_t* ba = hpf->ba; 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(tmp_int32 >> 13); y[1] = static_cast( (tmp_int32 - (static_cast(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(134217727), tmp_int32, static_cast(-134217728)); // Convert back to Q0 and use rounding. data[i] = (int16_t)(tmp_int32 >> 12); } return AudioProcessing::kNoError; } } // namespace typedef FilterState Handle; HighPassFilterImpl::HighPassFilterImpl(const AudioProcessing* apm, CriticalSectionWrapper* crit) : ProcessingComponent(), apm_(apm), crit_(crit) {} HighPassFilterImpl::~HighPassFilterImpl() {} int HighPassFilterImpl::ProcessCaptureAudio(AudioBuffer* audio) { int err = apm_->kNoError; if (!is_component_enabled()) { return apm_->kNoError; } assert(audio->num_frames_per_band() <= 160); for (int i = 0; i < num_handles(); i++) { Handle* my_handle = static_cast(handle(i)); err = Filter(my_handle, audio->split_bands(i)[kBand0To8kHz], audio->num_frames_per_band()); if (err != apm_->kNoError) { return GetHandleError(my_handle); } } return apm_->kNoError; } int HighPassFilterImpl::Enable(bool enable) { CriticalSectionScoped crit_scoped(crit_); return EnableComponent(enable); } bool HighPassFilterImpl::is_enabled() const { return is_component_enabled(); } void* HighPassFilterImpl::CreateHandle() const { return new FilterState; } void HighPassFilterImpl::DestroyHandle(void* handle) const { delete static_cast(handle); } int HighPassFilterImpl::InitializeHandle(void* handle) const { return InitializeFilter(static_cast(handle), apm_->proc_sample_rate_hz()); } int HighPassFilterImpl::ConfigureHandle(void* /*handle*/) const { return apm_->kNoError; // Not configurable. } int HighPassFilterImpl::num_handles_required() const { return apm_->num_output_channels(); } int HighPassFilterImpl::GetHandleError(void* handle) const { // The component has no detailed errors. assert(handle != NULL); return apm_->kUnspecifiedError; } } // namespace webrtc