ref: 45989463296ac74bd151affc216eeb83a93c919e
parent: f549b98dc7c35149bf9bece31343dff264f07c87
author: Paul Brossier <piem@piem.org>
date: Wed Sep 2 07:22:24 EDT 2015
src/temporal/*.h: remove trailing spaces, update copyrights
--- a/src/temporal/a_weighting.h
+++ b/src/temporal/a_weighting.h
@@ -1,5 +1,5 @@
/*
- Copyright (C) 2003-2013 Paul Brossier <piem@aubio.org>
+ Copyright (C) 2003-2015 Paul Brossier <piem@aubio.org>
This file is part of aubio.
@@ -24,18 +24,18 @@
/** \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 href="http://en.wikipedia.org/wiki/A-weighting">A-Weighting on
Wikipedia</a>
- <a href="http://en.wikipedia.org/wiki/Weighting_filter">Weighting filter on
@@ -42,7 +42,7 @@
Wikipedia</a>
- <a href="http://www.mathworks.com/matlabcentral/fileexchange/69">Christophe
Couvreur's 'octave' toolbox</a>
-
+
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 <pre> [b, a] = adsign(1/Fs) </pre> for various
@@ -62,7 +62,7 @@
/** 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 @@
/** 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
--- a/src/temporal/biquad.h
+++ b/src/temporal/biquad.h
@@ -1,5 +1,5 @@
/*
- Copyright (C) 2003-2013 Paul Brossier <piem@aubio.org>
+ Copyright (C) 2003-2015 Paul Brossier <piem@aubio.org>
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
--- a/src/temporal/c_weighting.h
+++ b/src/temporal/c_weighting.h
@@ -1,5 +1,5 @@
/*
- Copyright (C) 2003-2013 Paul Brossier <piem@aubio.org>
+ Copyright (C) 2003-2015 Paul Brossier <piem@aubio.org>
This file is part of aubio.
@@ -24,18 +24,18 @@
/** \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 href="http://en.wikipedia.org/wiki/A-weighting">A-Weighting on
Wikipedia</a>
- <a href="http://en.wikipedia.org/wiki/Weighting_filter">Weighting filter on
@@ -42,7 +42,7 @@
Wikipedia</a>
- <a href="http://www.mathworks.com/matlabcentral/fileexchange/69">Christophe
Couvreur's 'octave' toolbox</a>
-
+
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 <pre> [b, a] = cdsign(1/Fs) </pre> for various
@@ -62,7 +62,7 @@
/** 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 @@
/** 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
--- a/src/temporal/filter.h
+++ b/src/temporal/filter.h
@@ -1,5 +1,5 @@
/*
- Copyright (C) 2003-2013 Paul Brossier <piem@aubio.org>
+ Copyright (C) 2003-2015 Paul Brossier <piem@aubio.org>
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 @@
aubio_filter_t *new_aubio_filter (uint_t order);
/** delete a filter object
-
+
\param f filter object to delete
*/
--- a/src/temporal/resampler.h
+++ b/src/temporal/resampler.h
@@ -1,5 +1,5 @@
/*
- Copyright (C) 2003-2013 Paul Brossier <piem@aubio.org>
+ Copyright (C) 2003-2015 Paul Brossier <piem@aubio.org>
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 @@
/** 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
*/