/*
* 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 <stdint.h>
#include <string.h>
#include "modules/audio_coding/neteq/audio_multi_vector.h"
#include "modules/audio_coding/neteq/audio_vector.h"
#include "rtc_base/constructor_magic.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);
private:
// Table of constants used in method DspHelper::ParabolicFit().
static const int16_t kParabolaCoefficients[17][3];
RTC_DISALLOW_COPY_AND_ASSIGN(DspHelper);
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_NETEQ_DSP_HELPER_H_