/* * 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 WEBRTC_MODULES_AUDIO_CODING_NETEQ_DSP_HELPER_H_ #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_DSP_HELPER_H_ #include // Access to size_t. #include "webrtc/base/constructormagic.h" #include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h" #include "webrtc/typedefs.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(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, int data_length, int num_peaks, int fs_mult, int* 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, int* 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 int MinDistortion(const int16_t* signal, int min_lag, int max_lag, int 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, int16_t 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, int16_t 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, int output_length, int input_rate_hz, bool compensate_delay, int16_t* output); private: // Table of constants used in method DspHelper::ParabolicFit(). static const int16_t kParabolaCoefficients[17][3]; DISALLOW_COPY_AND_ASSIGN(DspHelper); }; } // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DSP_HELPER_H_