diff options
Diffstat (limited to 'webrtc/modules/audio_processing/high_pass_filter_impl.cc')
-rw-r--r-- | webrtc/modules/audio_processing/high_pass_filter_impl.cc | 222 |
1 files changed, 94 insertions, 128 deletions
diff --git a/webrtc/modules/audio_processing/high_pass_filter_impl.cc b/webrtc/modules/audio_processing/high_pass_filter_impl.cc index 29e482078e..375d58febb 100644 --- a/webrtc/modules/audio_processing/high_pass_filter_impl.cc +++ b/webrtc/modules/audio_processing/high_pass_filter_impl.cc @@ -10,159 +10,125 @@ #include "webrtc/modules/audio_processing/high_pass_filter_impl.h" -#include <assert.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" -#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); +const int16_t kFilterCoefficients8kHz[5] = {3798, -7596, 3798, 7807, -3733}; +const int16_t kFilterCoefficients[5] = {4012, -8024, 4012, 8002, -3913}; +} // namespace - if (sample_rate_hz == AudioProcessing::kSampleRate8kHz) { - hpf->ba = kFilterCoefficients8kHz; - } else { - hpf->ba = kFilterCoefficients; +class HighPassFilterImpl::BiquadFilter { + public: + explicit BiquadFilter(int sample_rate_hz) : + ba_(sample_rate_hz == AudioProcessing::kSampleRate8kHz ? + kFilterCoefficients8kHz : kFilterCoefficients) + { + Reset(); } - 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<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] = (int16_t)(tmp_int32 >> 12); + void Reset() { + std::memset(x_, 0, sizeof(x_)); + std::memset(y_, 0, sizeof(y_)); } - return AudioProcessing::kNoError; -} -} // namespace + 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); + } + } -typedef FilterState Handle; + private: + const int16_t* const ba_ = nullptr; + int16_t x_[2]; + int16_t y_[4]; +}; -HighPassFilterImpl::HighPassFilterImpl(const AudioProcessing* apm, - CriticalSectionWrapper* crit) - : ProcessingComponent(), - apm_(apm), - crit_(crit) {} +HighPassFilterImpl::HighPassFilterImpl(rtc::CriticalSection* crit) + : crit_(crit) { + RTC_DCHECK(crit_); +} HighPassFilterImpl::~HighPassFilterImpl() {} -int HighPassFilterImpl::ProcessCaptureAudio(AudioBuffer* audio) { - int err = apm_->kNoError; - - if (!is_component_enabled()) { - return apm_->kNoError; +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); +} - assert(audio->num_frames_per_band() <= 160); - - for (int i = 0; i < num_handles(); i++) { - Handle* my_handle = static_cast<Handle*>(handle(i)); - err = Filter(my_handle, - audio->split_bands(i)[kBand0To8kHz], - audio->num_frames_per_band()); - - if (err != apm_->kNoError) { - return GetHandleError(my_handle); - } +void HighPassFilterImpl::ProcessCaptureAudio(AudioBuffer* audio) { + RTC_DCHECK(audio); + rtc::CritScope cs(crit_); + if (!enabled_) { + return; } - return apm_->kNoError; + 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) { - CriticalSectionScoped crit_scoped(crit_); - return EnableComponent(enable); + rtc::CritScope cs(crit_); + if (!enabled_ && enable) { + for (auto& filter : filters_) { + filter->Reset(); + } + } + enabled_ = enable; + return AudioProcessing::kNoError; } 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*>(handle); -} - -int HighPassFilterImpl::InitializeHandle(void* handle) const { - return InitializeFilter(static_cast<Handle*>(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; + rtc::CritScope cs(crit_); + return enabled_; } } // namespace webrtc |