| /* |
| * Copyright (c) 2013 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_TRANSIENT_TRANSIENT_SUPPRESSOR_H_ |
| #define WEBRTC_MODULES_AUDIO_PROCESSING_TRANSIENT_TRANSIENT_SUPPRESSOR_H_ |
| |
| #include <deque> |
| #include <memory> |
| #include <set> |
| |
| #include "webrtc/rtc_base/gtest_prod_util.h" |
| #include "webrtc/typedefs.h" |
| |
| namespace webrtc { |
| |
| class TransientDetector; |
| |
| // Detects transients in an audio stream and suppress them using a simple |
| // restoration algorithm that attenuates unexpected spikes in the spectrum. |
| class TransientSuppressor { |
| public: |
| TransientSuppressor(); |
| ~TransientSuppressor(); |
| |
| int Initialize(int sample_rate_hz, int detector_rate_hz, int num_channels); |
| |
| // Processes a |data| chunk, and returns it with keystrokes suppressed from |
| // it. The float format is assumed to be int16 ranged. If there are more than |
| // one channel, the chunks are concatenated one after the other in |data|. |
| // |data_length| must be equal to |data_length_|. |
| // |num_channels| must be equal to |num_channels_|. |
| // A sub-band, ideally the higher, can be used as |detection_data|. If it is |
| // NULL, |data| is used for the detection too. The |detection_data| is always |
| // assumed mono. |
| // If a reference signal (e.g. keyboard microphone) is available, it can be |
| // passed in as |reference_data|. It is assumed mono and must have the same |
| // length as |data|. NULL is accepted if unavailable. |
| // This suppressor performs better if voice information is available. |
| // |voice_probability| is the probability of voice being present in this chunk |
| // of audio. If voice information is not available, |voice_probability| must |
| // always be set to 1. |
| // |key_pressed| determines if a key was pressed on this audio chunk. |
| // Returns 0 on success and -1 otherwise. |
| int Suppress(float* data, |
| size_t data_length, |
| int num_channels, |
| const float* detection_data, |
| size_t detection_length, |
| const float* reference_data, |
| size_t reference_length, |
| float voice_probability, |
| bool key_pressed); |
| |
| private: |
| FRIEND_TEST_ALL_PREFIXES(TransientSuppressorTest, |
| TypingDetectionLogicWorksAsExpectedForMono); |
| void Suppress(float* in_ptr, float* spectral_mean, float* out_ptr); |
| |
| void UpdateKeypress(bool key_pressed); |
| void UpdateRestoration(float voice_probability); |
| |
| void UpdateBuffers(float* data); |
| |
| void HardRestoration(float* spectral_mean); |
| void SoftRestoration(float* spectral_mean); |
| |
| std::unique_ptr<TransientDetector> detector_; |
| |
| size_t data_length_; |
| size_t detection_length_; |
| size_t analysis_length_; |
| size_t buffer_delay_; |
| size_t complex_analysis_length_; |
| int num_channels_; |
| // Input buffer where the original samples are stored. |
| std::unique_ptr<float[]> in_buffer_; |
| std::unique_ptr<float[]> detection_buffer_; |
| // Output buffer where the restored samples are stored. |
| std::unique_ptr<float[]> out_buffer_; |
| |
| // Arrays for fft. |
| std::unique_ptr<size_t[]> ip_; |
| std::unique_ptr<float[]> wfft_; |
| |
| std::unique_ptr<float[]> spectral_mean_; |
| |
| // Stores the data for the fft. |
| std::unique_ptr<float[]> fft_buffer_; |
| |
| std::unique_ptr<float[]> magnitudes_; |
| |
| const float* window_; |
| |
| std::unique_ptr<float[]> mean_factor_; |
| |
| float detector_smoothed_; |
| |
| int keypress_counter_; |
| int chunks_since_keypress_; |
| bool detection_enabled_; |
| bool suppression_enabled_; |
| |
| bool use_hard_restoration_; |
| int chunks_since_voice_change_; |
| |
| uint32_t seed_; |
| |
| bool using_reference_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_MODULES_AUDIO_PROCESSING_TRANSIENT_TRANSIENT_SUPPRESSOR_H_ |