From 45989463296ac74bd151affc216eeb83a93c919e Mon Sep 17 00:00:00 2001 From: Paul Brossier Date: Wed, 2 Sep 2015 11:22:24 +0200 Subject: [PATCH] src/temporal/*.h: remove trailing spaces, update copyrights --- src/temporal/a_weighting.h | 16 ++++++++-------- src/temporal/biquad.h | 6 +++--- src/temporal/c_weighting.h | 16 ++++++++-------- src/temporal/filter.h | 16 ++++++++-------- src/temporal/resampler.h | 10 +++++----- 5 files changed, 32 insertions(+), 32 deletions(-) diff --git a/src/temporal/a_weighting.h b/src/temporal/a_weighting.h index 1a423c00..72f02b8d 100644 --- a/src/temporal/a_weighting.h +++ b/src/temporal/a_weighting.h @@ -1,5 +1,5 @@ /* - Copyright (C) 2003-2013 Paul Brossier + Copyright (C) 2003-2015 Paul Brossier This file is part of aubio. @@ -24,25 +24,25 @@ /** \file A-weighting filter coefficients - + This file creates an A-weighting digital filter, which reduces low and high frequencies and enhance the middle ones to reflect the ability of the human hearing. - + The implementation is based on the following standard: - IEC/CD 1672: Electroacoustics-Sound Level Meters, IEC, Geneva, Nov. 1996, for A- and C-weighting filters. - + See also: - + - A-Weighting on Wikipedia - Weighting filter on Wikipedia - Christophe Couvreur's 'octave' toolbox - + The coefficients in this file have been computed using Christophe Couvreur's scripts in octave 3.0 (debian package 1:3.0.5-6+b2 with octave-signal 1.0.9-1+b1 on i386), with 
 [b, a] = adsign(1/Fs)
for various @@ -62,7 +62,7 @@ extern "C" { /** create new A-design filter - \param samplerate sampling frequency of the signal to filter. Should be one of + \param samplerate sampling frequency of the signal to filter. Should be one of 8000, 11025, 16000, 22050, 24000, 32000, 44100, 48000, 88200, 96000, and 192000 Hz @@ -74,7 +74,7 @@ aubio_filter_t *new_aubio_filter_a_weighting (uint_t samplerate); /** set feedback and feedforward coefficients of a A-weighting filter \param f filter object to get coefficients from - \param samplerate sampling frequency of the signal to filter. Should be one of + \param samplerate sampling frequency of the signal to filter. Should be one of 8000, 11025, 16000, 22050, 24000, 32000, 44100, 48000, 88200, 96000, and 192000 Hz diff --git a/src/temporal/biquad.h b/src/temporal/biquad.h index c5305c38..1b5147a4 100644 --- a/src/temporal/biquad.h +++ b/src/temporal/biquad.h @@ -1,5 +1,5 @@ /* - Copyright (C) 2003-2013 Paul Brossier + Copyright (C) 2003-2015 Paul Brossier This file is part of aubio. @@ -21,12 +21,12 @@ #ifndef _AUBIO_FILTER_BIQUAD_H #define _AUBIO_FILTER_BIQUAD_H -/** \file +/** \file Second order Infinite Impulse Response filter This file implements a normalised biquad filter (second order IIR): - + \f$ y[n] = b_0 x[n] + b_1 x[n-1] + b_2 x[n-2] - a_1 y[n-1] - a_2 y[n-2] \f$ The filtfilt version runs the filter twice, forward and backward, to diff --git a/src/temporal/c_weighting.h b/src/temporal/c_weighting.h index 3290be83..5ef3874e 100644 --- a/src/temporal/c_weighting.h +++ b/src/temporal/c_weighting.h @@ -1,5 +1,5 @@ /* - Copyright (C) 2003-2013 Paul Brossier + Copyright (C) 2003-2015 Paul Brossier This file is part of aubio. @@ -24,25 +24,25 @@ /** \file C-weighting filter coefficients - + This file creates a C-weighting digital filter, which reduces low and high frequencies and enhance the middle ones to reflect the ability of the human hearing. - + The implementation is based on the following standard: - IEC/CD 1672: Electroacoustics-Sound Level Meters, IEC, Geneva, Nov. 1996, for A- and C-weighting filters. - + See also: - + - A-Weighting on Wikipedia - Weighting filter on Wikipedia - Christophe Couvreur's 'octave' toolbox - + The coefficients in this file have been computed using Christophe Couvreur's scripts in octave 3.0 (debian package 1:3.0.5-6+b2 with octave-signal 1.0.9-1+b1 on i386), with 
 [b, a] = cdsign(1/Fs)
for various @@ -62,7 +62,7 @@ extern "C" { /** create new C-design filter - \param samplerate sampling frequency of the signal to filter. Should be one of + \param samplerate sampling frequency of the signal to filter. Should be one of 8000, 11025, 16000, 22050, 24000, 32000, 44100, 48000, 88200, 96000, and 192000 Hz @@ -74,7 +74,7 @@ aubio_filter_t *new_aubio_filter_c_weighting (uint_t samplerate); /** set feedback and feedforward coefficients of a C-weighting filter \param f filter object to get coefficients from - \param samplerate sampling frequency of the signal to filter. Should be one of + \param samplerate sampling frequency of the signal to filter. Should be one of 8000, 11025, 16000, 22050, 24000, 32000, 44100, 48000, 88200, 96000, and 192000 Hz diff --git a/src/temporal/filter.h b/src/temporal/filter.h index 1fa38963..13c7cba5 100644 --- a/src/temporal/filter.h +++ b/src/temporal/filter.h @@ -1,5 +1,5 @@ /* - Copyright (C) 2003-2013 Paul Brossier + Copyright (C) 2003-2015 Paul Brossier This file is part of aubio. @@ -21,14 +21,14 @@ #ifndef _AUBIO_FILTER_H #define _AUBIO_FILTER_H -/** \file +/** \file Digital filter This object stores a digital filter of order \f$n\f$. It contains the following data: - - \f$ n*1 b_i \f$ feedforward coefficients - - \f$ n*1 a_i \f$ feedback coefficients + - \f$ n*1 b_i \f$ feedforward coefficients + - \f$ n*1 a_i \f$ feedback coefficients - \f$ n*c x_i \f$ input signal - \f$ n*c y_i \f$ output signal @@ -40,7 +40,7 @@ The function aubio_filter_do_outplace() computes the following output signal \f$ y[n] \f$ from the input signal \f$ x[n] \f$: - + \f{eqnarray*}{ y[n] = b_0 x[n] & + & b_1 x[n-1] + b_2 x[n-2] + ... + b_P x[n-P] \\ & - & a_1 y[n-1] - a_2 y[n-2] - ... - a_P y[n-P] \\ @@ -53,13 +53,13 @@ forward then backward, to compensate with the phase shifting of the forward operation. - Some convenience functions are provided: + Some convenience functions are provided: - new_aubio_filter_a_weighting() and aubio_filter_set_a_weighting(), - new_aubio_filter_c_weighting() and aubio_filter_set_c_weighting(). - new_aubio_filter_biquad() and aubio_filter_set_biquad(). \example temporal/test-filter.c - + */ #ifdef __cplusplus @@ -163,7 +163,7 @@ void aubio_filter_do_reset (aubio_filter_t * f); aubio_filter_t *new_aubio_filter (uint_t order); /** delete a filter object - + \param f filter object to delete */ diff --git a/src/temporal/resampler.h b/src/temporal/resampler.h index 2c233594..1684f752 100644 --- a/src/temporal/resampler.h +++ b/src/temporal/resampler.h @@ -1,5 +1,5 @@ /* - Copyright (C) 2003-2013 Paul Brossier + Copyright (C) 2003-2015 Paul Brossier This file is part of aubio. @@ -22,12 +22,12 @@ #define _AUBIO_RESAMPLER_H /** \file - + Resampling object This object resamples an input vector into an output vector using libsamplerate. See http://www.mega-nerd.com/SRC/ - + */ #ifdef __cplusplus @@ -37,9 +37,9 @@ extern "C" { /** resampler object */ typedef struct _aubio_resampler_t aubio_resampler_t; -/** create resampler object +/** create resampler object - \param ratio output_sample_rate / input_sample_rate + \param ratio output_sample_rate / input_sample_rate \param type libsamplerate resampling type, see http://www.mega-nerd.com/SRC/api_misc.html#Converters */ -- 2.11.0