/* * 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. */ #ifndef MODULES_AUDIO_CODING_NETEQ_DSP_HELPER_H_ #define MODULES_AUDIO_CODING_NETEQ_DSP_HELPER_H_ #include #include #include "modules/audio_coding/neteq/audio_multi_vector.h" #include "modules/audio_coding/neteq/audio_vector.h" namespace webrtc { // This class contains various signal processing functions, all implemented as // static methods. class DspHelper { public: // Filter coefficients used when downsampling from the indicated sample rates // (8, 16, 32, 48 kHz) to 4 kHz. Coefficients are in Q12. static const int16_t kDownsample8kHzTbl[3]; static const int16_t kDownsample16kHzTbl[5]; static const int16_t kDownsample32kHzTbl[7]; static const int16_t kDownsample48kHzTbl[7]; // Constants used to mute and unmute over 5 samples. The coefficients are // in Q15. static const int kMuteFactorStart8kHz = 27307; static const int kMuteFactorIncrement8kHz = -5461; static const int kUnmuteFactorStart8kHz = 5461; static const int kUnmuteFactorIncrement8kHz = 5461; static const int kMuteFactorStart16kHz = 29789; static const int kMuteFactorIncrement16kHz = -2979; static const int kUnmuteFactorStart16kHz = 2979; static const int kUnmuteFactorIncrement16kHz = 2979; static const int kMuteFactorStart32kHz = 31208; static const int kMuteFactorIncrement32kHz = -1560; static const int kUnmuteFactorStart32kHz = 1560; static const int kUnmuteFactorIncrement32kHz = 1560; static const int kMuteFactorStart48kHz = 31711; static const int kMuteFactorIncrement48kHz = -1057; static const int kUnmuteFactorStart48kHz = 1057; static const int kUnmuteFactorIncrement48kHz = 1057; // Multiplies the signal with a gradually changing factor. // The first sample is multiplied with `factor` (in Q14). For each sample, // `factor` is increased (additive) by the `increment` (in Q20), which can // be negative. Returns the scale factor after the last increment. static int RampSignal(const int16_t* input, size_t length, int factor, int increment, int16_t* output); // Same as above, but with the samples of `signal` being modified in-place. static int RampSignal(int16_t* signal, size_t length, int factor, int increment); // Same as above, but processes `length` samples from `signal`, starting at // `start_index`. static int RampSignal(AudioVector* signal, size_t start_index, size_t length, int factor, int increment); // Same as above, but for an AudioMultiVector. static int RampSignal(AudioMultiVector* signal, size_t start_index, size_t length, int factor, int increment); // Peak detection with parabolic fit. Looks for `num_peaks` maxima in `data`, // having length `data_length` and sample rate multiplier `fs_mult`. The peak // locations and values are written to the arrays `peak_index` and // `peak_value`, respectively. Both arrays must hold at least `num_peaks` // elements. static void PeakDetection(int16_t* data, size_t data_length, size_t num_peaks, int fs_mult, size_t* peak_index, int16_t* peak_value); // Estimates the height and location of a maximum. The three values in the // array `signal_points` are used as basis for a parabolic fit, which is then // used to find the maximum in an interpolated signal. The `signal_points` are // assumed to be from a 4 kHz signal, while the maximum, written to // `peak_index` and `peak_value` is given in the full sample rate, as // indicated by the sample rate multiplier `fs_mult`. static void ParabolicFit(int16_t* signal_points, int fs_mult, size_t* peak_index, int16_t* peak_value); // Calculates the sum-abs-diff for `signal` when compared to a displaced // version of itself. Returns the displacement lag that results in the minimum // distortion. The resulting distortion is written to `distortion_value`. // The values of `min_lag` and `max_lag` are boundaries for the search. static size_t MinDistortion(const int16_t* signal, size_t min_lag, size_t max_lag, size_t length, int32_t* distortion_value); // Mixes `length` samples from `input1` and `input2` together and writes the // result to `output`. The gain for `input1` starts at `mix_factor` (Q14) and // is decreased by `factor_decrement` (Q14) for each sample. The gain for // `input2` is the complement 16384 - mix_factor. static void CrossFade(const int16_t* input1, const int16_t* input2, size_t length, int16_t* mix_factor, int16_t factor_decrement, int16_t* output); // Scales `input` with an increasing gain. Applies `factor` (Q14) to the first // sample and increases the gain by `increment` (Q20) for each sample. The // result is written to `output`. `length` samples are processed. static void UnmuteSignal(const int16_t* input, size_t length, int16_t* factor, int increment, int16_t* output); // Starts at unity gain and gradually fades out `signal`. For each sample, // the gain is reduced by `mute_slope` (Q14). `length` samples are processed. static void MuteSignal(int16_t* signal, int mute_slope, size_t length); // Downsamples `input` from `sample_rate_hz` to 4 kHz sample rate. The input // has `input_length` samples, and the method will write `output_length` // samples to `output`. Compensates for the phase delay of the downsampling // filters if `compensate_delay` is true. Returns -1 if the input is too short // to produce `output_length` samples, otherwise 0. static int DownsampleTo4kHz(const int16_t* input, size_t input_length, size_t output_length, int input_rate_hz, bool compensate_delay, int16_t* output); DspHelper(const DspHelper&) = delete; DspHelper& operator=(const DspHelper&) = delete; private: // Table of constants used in method DspHelper::ParabolicFit(). static const int16_t kParabolaCoefficients[17][3]; }; } // namespace webrtc #endif // MODULES_AUDIO_CODING_NETEQ_DSP_HELPER_H_