| /* |
| * Copyright (c) 2014 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. |
| */ |
| |
| #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_INTELLIGIBILITY_INTELLIGIBILITY_ENHANCER_H_ |
| #define WEBRTC_MODULES_AUDIO_PROCESSING_INTELLIGIBILITY_INTELLIGIBILITY_ENHANCER_H_ |
| |
| #include <complex> |
| #include <memory> |
| #include <vector> |
| |
| #include "webrtc/base/swap_queue.h" |
| #include "webrtc/common_audio/channel_buffer.h" |
| #include "webrtc/common_audio/lapped_transform.h" |
| #include "webrtc/modules/audio_processing/audio_buffer.h" |
| #include "webrtc/modules/audio_processing/intelligibility/intelligibility_utils.h" |
| #include "webrtc/modules/audio_processing/render_queue_item_verifier.h" |
| #include "webrtc/modules/audio_processing/vad/voice_activity_detector.h" |
| |
| namespace webrtc { |
| |
| // Speech intelligibility enhancement module. Reads render and capture |
| // audio streams and modifies the render stream with a set of gains per |
| // frequency bin to enhance speech against the noise background. |
| // Details of the model and algorithm can be found in the original paper: |
| // http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=&arnumber=6882788 |
| class IntelligibilityEnhancer : public LappedTransform::Callback { |
| public: |
| IntelligibilityEnhancer(int sample_rate_hz, |
| size_t num_render_channels, |
| size_t num_bands, |
| size_t num_noise_bins); |
| |
| ~IntelligibilityEnhancer() override; |
| |
| // Sets the capture noise magnitude spectrum estimate. |
| void SetCaptureNoiseEstimate(std::vector<float> noise, float gain); |
| |
| // Reads chunk of speech in time domain and updates with modified signal. |
| void ProcessRenderAudio(AudioBuffer* audio); |
| bool active() const; |
| |
| protected: |
| // All in frequency domain, receives input |in_block|, applies |
| // intelligibility enhancement, and writes result to |out_block|. |
| void ProcessAudioBlock(const std::complex<float>* const* in_block, |
| size_t in_channels, |
| size_t frames, |
| size_t out_channels, |
| std::complex<float>* const* out_block) override; |
| |
| private: |
| FRIEND_TEST_ALL_PREFIXES(IntelligibilityEnhancerTest, TestRenderUpdate); |
| FRIEND_TEST_ALL_PREFIXES(IntelligibilityEnhancerTest, TestErbCreation); |
| FRIEND_TEST_ALL_PREFIXES(IntelligibilityEnhancerTest, TestSolveForGains); |
| FRIEND_TEST_ALL_PREFIXES(IntelligibilityEnhancerTest, |
| TestNoiseGainHasExpectedResult); |
| FRIEND_TEST_ALL_PREFIXES(IntelligibilityEnhancerTest, |
| TestAllBandsHaveSameDelay); |
| |
| // Updates the SNR estimation and enables or disables this component using a |
| // hysteresis. |
| void SnrBasedEffectActivation(); |
| |
| // Bisection search for optimal |lambda|. |
| void SolveForLambda(float power_target); |
| |
| // Transforms freq gains to ERB gains. |
| void UpdateErbGains(); |
| |
| // Returns number of ERB filters. |
| static size_t GetBankSize(int sample_rate, size_t erb_resolution); |
| |
| // Initializes ERB filterbank. |
| std::vector<std::vector<float>> CreateErbBank(size_t num_freqs); |
| |
| // Analytically solves quadratic for optimal gains given |lambda|. |
| // Negative gains are set to 0. Stores the results in |sols|. |
| void SolveForGainsGivenLambda(float lambda, size_t start_freq, float* sols); |
| |
| // Returns true if the audio is speech. |
| bool IsSpeech(const float* audio); |
| |
| // Delays the high bands to compensate for the processing delay in the low |
| // band. |
| void DelayHighBands(AudioBuffer* audio); |
| |
| static const size_t kMaxNumNoiseEstimatesToBuffer = 5; |
| |
| const size_t freqs_; // Num frequencies in frequency domain. |
| const size_t num_noise_bins_; |
| const size_t chunk_length_; // Chunk size in samples. |
| const size_t bank_size_; // Num ERB filters. |
| const int sample_rate_hz_; |
| const size_t num_render_channels_; |
| |
| intelligibility::PowerEstimator<std::complex<float>> clear_power_estimator_; |
| intelligibility::PowerEstimator<float> noise_power_estimator_; |
| std::vector<float> filtered_clear_pow_; |
| std::vector<float> filtered_noise_pow_; |
| std::vector<float> center_freqs_; |
| std::vector<std::vector<float>> capture_filter_bank_; |
| std::vector<std::vector<float>> render_filter_bank_; |
| size_t start_freq_; |
| |
| std::vector<float> gains_eq_; // Pre-filter modified gains. |
| intelligibility::GainApplier gain_applier_; |
| |
| std::unique_ptr<LappedTransform> render_mangler_; |
| |
| VoiceActivityDetector vad_; |
| std::vector<int16_t> audio_s16_; |
| size_t chunks_since_voice_; |
| bool is_speech_; |
| float snr_; |
| bool is_active_; |
| |
| unsigned long int num_chunks_; |
| unsigned long int num_active_chunks_; |
| |
| std::vector<float> noise_estimation_buffer_; |
| SwapQueue<std::vector<float>, RenderQueueItemVerifier<float>> |
| noise_estimation_queue_; |
| |
| std::vector<std::unique_ptr<intelligibility::DelayBuffer>> |
| high_bands_buffers_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_MODULES_AUDIO_PROCESSING_INTELLIGIBILITY_INTELLIGIBILITY_ENHANCER_H_ |