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137 lines
6.4 KiB
C++
137 lines
6.4 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_DSP_HELPER_H_
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#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_DSP_HELPER_H_
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#include <string.h> // Access to size_t.
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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// This class contains various signal processing functions, all implemented as
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// static methods.
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class DspHelper {
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public:
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// Filter coefficients used when downsampling from the indicated sample rates
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// (8, 16, 32, 48 kHz) to 4 kHz. Coefficients are in Q12.
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static const int16_t kDownsample8kHzTbl[3];
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static const int16_t kDownsample16kHzTbl[5];
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static const int16_t kDownsample32kHzTbl[7];
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static const int16_t kDownsample48kHzTbl[7];
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// Constants used to mute and unmute over 5 samples. The coefficients are
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// in Q15.
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static const int kMuteFactorStart8kHz = 27307;
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static const int kMuteFactorIncrement8kHz = -5461;
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static const int kUnmuteFactorStart8kHz = 5461;
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static const int kUnmuteFactorIncrement8kHz = 5461;
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static const int kMuteFactorStart16kHz = 29789;
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static const int kMuteFactorIncrement16kHz = -2979;
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static const int kUnmuteFactorStart16kHz = 2979;
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static const int kUnmuteFactorIncrement16kHz = 2979;
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static const int kMuteFactorStart32kHz = 31208;
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static const int kMuteFactorIncrement32kHz = -1560;
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static const int kUnmuteFactorStart32kHz = 1560;
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static const int kUnmuteFactorIncrement32kHz = 1560;
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static const int kMuteFactorStart48kHz = 31711;
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static const int kMuteFactorIncrement48kHz = -1057;
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static const int kUnmuteFactorStart48kHz = 1057;
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static const int kUnmuteFactorIncrement48kHz = 1057;
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// Multiplies the signal with a gradually changing factor.
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// The first sample is multiplied with |factor| (in Q14). For each sample,
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// |factor| is increased (additive) by the |increment| (in Q20), which can
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// be negative. Returns the scale factor after the last increment.
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static int RampSignal(const int16_t* input,
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size_t length,
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int factor,
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int increment,
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int16_t* output);
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// Same as above, but with the samples of |signal| being modified in-place.
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static int RampSignal(int16_t* signal,
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size_t length,
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int factor,
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int increment);
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// Same as above, but processes |length| samples from |signal|, starting at
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// |start_index|.
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static int RampSignal(AudioMultiVector* signal,
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size_t start_index,
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size_t length,
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int factor,
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int increment);
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// Peak detection with parabolic fit. Looks for |num_peaks| maxima in |data|,
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// having length |data_length| and sample rate multiplier |fs_mult|. The peak
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// locations and values are written to the arrays |peak_index| and
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// |peak_value|, respectively. Both arrays must hold at least |num_peaks|
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// elements.
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static void PeakDetection(int16_t* data, int data_length,
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int num_peaks, int fs_mult,
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int* peak_index, int16_t* peak_value);
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// Estimates the height and location of a maximum. The three values in the
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// array |signal_points| are used as basis for a parabolic fit, which is then
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// used to find the maximum in an interpolated signal. The |signal_points| are
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// assumed to be from a 4 kHz signal, while the maximum, written to
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// |peak_index| and |peak_value| is given in the full sample rate, as
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// indicated by the sample rate multiplier |fs_mult|.
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static void ParabolicFit(int16_t* signal_points, int fs_mult,
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int* peak_index, int16_t* peak_value);
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// Calculates the sum-abs-diff for |signal| when compared to a displaced
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// version of itself. Returns the displacement lag that results in the minimum
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// distortion. The resulting distortion is written to |distortion_value|.
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// The values of |min_lag| and |max_lag| are boundaries for the search.
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static int MinDistortion(const int16_t* signal, int min_lag,
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int max_lag, int length, int32_t* distortion_value);
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// Mixes |length| samples from |input1| and |input2| together and writes the
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// result to |output|. The gain for |input1| starts at |mix_factor| (Q14) and
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// is decreased by |factor_decrement| (Q14) for each sample. The gain for
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// |input2| is the complement 16384 - mix_factor.
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static void CrossFade(const int16_t* input1, const int16_t* input2,
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size_t length, int16_t* mix_factor,
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int16_t factor_decrement, int16_t* output);
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// Scales |input| with an increasing gain. Applies |factor| (Q14) to the first
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// sample and increases the gain by |increment| (Q20) for each sample. The
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// result is written to |output|. |length| samples are processed.
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static void UnmuteSignal(const int16_t* input, size_t length, int16_t* factor,
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int16_t increment, int16_t* output);
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// Starts at unity gain and gradually fades out |signal|. For each sample,
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// the gain is reduced by |mute_slope| (Q14). |length| samples are processed.
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static void MuteSignal(int16_t* signal, int16_t mute_slope, size_t length);
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// Downsamples |input| from |sample_rate_hz| to 4 kHz sample rate. The input
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// has |input_length| samples, and the method will write |output_length|
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// samples to |output|. Compensates for the phase delay of the downsampling
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// filters if |compensate_delay| is true. Returns -1 if the input is too short
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// to produce |output_length| samples, otherwise 0.
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static int DownsampleTo4kHz(const int16_t* input, size_t input_length,
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int output_length, int input_rate_hz,
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bool compensate_delay, int16_t* output);
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private:
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// Table of constants used in method DspHelper::ParabolicFit().
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static const int16_t kParabolaCoefficients[17][3];
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DISALLOW_COPY_AND_ASSIGN(DspHelper);
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DSP_HELPER_H_
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