diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc index 48f23b1856..a38d73e2dc 100644 --- a/webrtc/modules/audio_processing/audio_processing_impl.cc +++ b/webrtc/modules/audio_processing/audio_processing_impl.cc @@ -81,7 +81,6 @@ struct AudioProcessingImpl::ApmPublicSubmodules { : echo_cancellation(nullptr), echo_control_mobile(nullptr), gain_control(nullptr), - high_pass_filter(nullptr), level_estimator(nullptr), noise_suppression(nullptr), voice_detection(nullptr) {} @@ -89,7 +88,7 @@ struct AudioProcessingImpl::ApmPublicSubmodules { EchoCancellationImpl* echo_cancellation; EchoControlMobileImpl* echo_control_mobile; GainControlImpl* gain_control; - HighPassFilterImpl* high_pass_filter; + rtc::scoped_ptr high_pass_filter; LevelEstimatorImpl* level_estimator; NoiseSuppressionImpl* noise_suppression; VoiceDetectionImpl* voice_detection; @@ -243,8 +242,8 @@ AudioProcessingImpl::AudioProcessingImpl(const Config& config, new EchoControlMobileImpl(this, &crit_render_, &crit_capture_); public_submodules_->gain_control = new GainControlImpl(this, &crit_capture_, &crit_capture_); - public_submodules_->high_pass_filter = - new HighPassFilterImpl(this, &crit_capture_); + public_submodules_->high_pass_filter.reset( + new HighPassFilterImpl(&crit_capture_)); public_submodules_->level_estimator = new LevelEstimatorImpl(this, &crit_capture_); public_submodules_->noise_suppression = @@ -260,8 +259,6 @@ AudioProcessingImpl::AudioProcessingImpl(const Config& config, public_submodules_->echo_control_mobile); private_submodules_->component_list.push_back( public_submodules_->gain_control); - private_submodules_->component_list.push_back( - public_submodules_->high_pass_filter); private_submodules_->component_list.push_back( public_submodules_->level_estimator); private_submodules_->component_list.push_back( @@ -406,6 +403,8 @@ int AudioProcessingImpl::InitializeLocked() { InitializeIntelligibility(); + InitializeHighPassFilter(); + #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP if (debug_dump_.debug_file->Open()) { int err = WriteInitMessage(); @@ -767,7 +766,7 @@ int AudioProcessingImpl::ProcessStreamLocked() { ca->set_num_channels(1); } - RETURN_ON_ERR(public_submodules_->high_pass_filter->ProcessCaptureAudio(ca)); + public_submodules_->high_pass_filter->ProcessCaptureAudio(ca); RETURN_ON_ERR(public_submodules_->gain_control->AnalyzeCaptureAudio(ca)); RETURN_ON_ERR(public_submodules_->noise_suppression->AnalyzeCaptureAudio(ca)); RETURN_ON_ERR(public_submodules_->echo_cancellation->ProcessCaptureAudio(ca)); @@ -1147,7 +1146,7 @@ GainControl* AudioProcessingImpl::gain_control() const { HighPassFilter* AudioProcessingImpl::high_pass_filter() const { // Adding a lock here has no effect as it allows any access to the submodule // from the returned pointer. - return public_submodules_->high_pass_filter; + return public_submodules_->high_pass_filter.get(); } LevelEstimator* AudioProcessingImpl::level_estimator() const { @@ -1179,6 +1178,9 @@ bool AudioProcessingImpl::is_data_processed() const { enabled_count++; } } + if (public_submodules_->high_pass_filter->is_enabled()) { + enabled_count++; + } // Data is unchanged if no components are enabled, or if only // public_submodules_->level_estimator @@ -1293,6 +1295,11 @@ void AudioProcessingImpl::InitializeIntelligibility() { } } +void AudioProcessingImpl::InitializeHighPassFilter() { + public_submodules_->high_pass_filter->Initialize(num_output_channels(), + proc_sample_rate_hz()); +} + void AudioProcessingImpl::MaybeUpdateHistograms() { static const int kMinDiffDelayMs = 60; diff --git a/webrtc/modules/audio_processing/audio_processing_impl.h b/webrtc/modules/audio_processing/audio_processing_impl.h index 4a290a8be7..1ca5a8c19e 100644 --- a/webrtc/modules/audio_processing/audio_processing_impl.h +++ b/webrtc/modules/audio_processing/audio_processing_impl.h @@ -193,6 +193,8 @@ class AudioProcessingImpl : public AudioProcessing { EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_); void InitializeIntelligibility() EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_); + void InitializeHighPassFilter() + EXCLUSIVE_LOCKS_REQUIRED(crit_capture_); int InitializeLocked(const ProcessingConfig& config) EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_); diff --git a/webrtc/modules/audio_processing/high_pass_filter_impl.cc b/webrtc/modules/audio_processing/high_pass_filter_impl.cc index 2ad0a5098c..795dcbd21c 100644 --- a/webrtc/modules/audio_processing/high_pass_filter_impl.cc +++ b/webrtc/modules/audio_processing/high_pass_filter_impl.cc @@ -10,165 +10,115 @@ #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; -} +const int16_t kFilterCoefficients8kHz[5] = {3798, -7596, 3798, 7807, -3733}; +const int16_t kFilterCoefficients[5] = {4012, -8024, 4012, 8002, -3913}; } // namespace -typedef FilterState Handle; +class HighPassFilterImpl::BiquadFilter { + public: + 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_)); + } -HighPassFilterImpl::HighPassFilterImpl(const AudioProcessing* apm, - rtc::CriticalSection* crit) - : ProcessingComponent(), apm_(apm), crit_(crit) { - RTC_DCHECK(apm); - RTC_DCHECK(crit); + 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(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] = static_cast(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() {} -int HighPassFilterImpl::ProcessCaptureAudio(AudioBuffer* audio) { +void HighPassFilterImpl::Initialize(int channels, int sample_rate_hz) { + std::vector> new_filters(channels); + for (int i = 0; i < channels; i++) { + new_filters[i].reset(new BiquadFilter(sample_rate_hz)); + } rtc::CritScope cs(crit_); - int err = AudioProcessing::kNoError; + filters_.swap(new_filters); +} - if (!is_component_enabled()) { - return AudioProcessing::kNoError; +void HighPassFilterImpl::ProcessCaptureAudio(AudioBuffer* audio) { + rtc::CritScope cs(crit_); + if (!enabled_) { + return; } - 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 != AudioProcessing::kNoError) { - return GetHandleError(my_handle); - } + RTC_DCHECK_GE(160u, audio->num_frames_per_band()); + RTC_DCHECK_EQ(filters_.size(), static_cast(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()); } - - return AudioProcessing::kNoError; } int HighPassFilterImpl::Enable(bool enable) { rtc::CritScope cs(crit_); - return EnableComponent(enable); + enabled_ = enable; + return AudioProcessing::kNoError; } bool HighPassFilterImpl::is_enabled() const { rtc::CritScope cs(crit_); - 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 { - // TODO(peah): Remove dependency on apm for the - // capture side sample rate. - rtc::CritScope cs(crit_); - return InitializeFilter(static_cast(handle), - apm_->proc_sample_rate_hz()); -} - -int HighPassFilterImpl::ConfigureHandle(void* /*handle*/) const { - return AudioProcessing::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 AudioProcessing::kUnspecifiedError; + return enabled_; } } // namespace webrtc diff --git a/webrtc/modules/audio_processing/high_pass_filter_impl.h b/webrtc/modules/audio_processing/high_pass_filter_impl.h index 6f8079e32c..b2a0717374 100644 --- a/webrtc/modules/audio_processing/high_pass_filter_impl.h +++ b/webrtc/modules/audio_processing/high_pass_filter_impl.h @@ -12,39 +12,31 @@ #define WEBRTC_MODULES_AUDIO_PROCESSING_HIGH_PASS_FILTER_IMPL_H_ #include "webrtc/base/criticalsection.h" +#include "webrtc/base/scoped_ptr.h" #include "webrtc/modules/audio_processing/include/audio_processing.h" -#include "webrtc/modules/audio_processing/processing_component.h" namespace webrtc { class AudioBuffer; -class HighPassFilterImpl : public HighPassFilter, - public ProcessingComponent { +class HighPassFilterImpl : public HighPassFilter { public: - HighPassFilterImpl(const AudioProcessing* apm, rtc::CriticalSection* crit); - virtual ~HighPassFilterImpl(); + explicit HighPassFilterImpl(rtc::CriticalSection* crit); + ~HighPassFilterImpl() override; - int ProcessCaptureAudio(AudioBuffer* audio); + // TODO(peah): Fold into ctor, once public API is removed. + void Initialize(int channels, int sample_rate_hz); + void ProcessCaptureAudio(AudioBuffer* audio); // HighPassFilter implementation. + int Enable(bool enable) override; bool is_enabled() const override; private: - // HighPassFilter implementation. - int Enable(bool enable) override; - - // ProcessingComponent implementation. - void* CreateHandle() const override; - int InitializeHandle(void* handle) const override; - int ConfigureHandle(void* handle) const override; - void DestroyHandle(void* handle) const override; - int num_handles_required() const override; - int GetHandleError(void* handle) const override; - - const AudioProcessing* apm_; - - rtc::CriticalSection* const crit_; + class BiquadFilter; + rtc::CriticalSection* const crit_ = nullptr; + bool enabled_ GUARDED_BY(crit_) = false; + std::vector> filters_ GUARDED_BY(crit_); }; } // namespace webrtc