Changeset 4598946 for src

Timestamp:

Sep 2, 2015, 11:22:24 AM (9 years ago)

Author:

Paul Brossier <piem@piem.org>

Branches:

feature/autosink, feature/cnn, feature/cnn_org, feature/constantq, feature/crepe, feature/crepe_org, feature/pitchshift, feature/pydocstrings, feature/timestretch, fix/ffmpeg5, master, pitchshift, sampler, timestretch, yinfft+

Children:

0683ee2

Parents:

f549b98d

Message:

src/temporal/*.h: remove trailing spaces, update copyrights

Location:

src/temporal

Files:

• a_weighting.h (modified) (5 diffs)
• biquad.h (modified) (2 diffs)
• c_weighting.h (modified) (5 diffs)
• filter.h (modified) (7 diffs)
• resampler.h (modified) (3 diffs)

src/temporal/a_weighting.h

@@ -1 +1 @@
 /*
-  Copyright (C) 2003-2013 Paul Brossier <piem@aubio.org>
+  Copyright (C) 2003-2015 Paul Brossier <piem@aubio.org>
 
   This file is part of aubio.
@@ -25 +25 @@
 
   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>
@@ -43 +43 @@
     - <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
@@ -63 +63 @@
 /** 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
@@ -75 +75 @@
 
   \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

src/temporal/biquad.h

@@ -1 +1 @@
 /*
-  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 +22 @@
 #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$
 

src/temporal/c_weighting.h

@@ -1 +1 @@
 /*
-  Copyright (C) 2003-2013 Paul Brossier <piem@aubio.org>
+  Copyright (C) 2003-2015 Paul Brossier <piem@aubio.org>
 
   This file is part of aubio.
@@ -25 +25 @@
 
   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>
@@ -43 +43 @@
     - <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
@@ -63 +63 @@
 /** 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
@@ -75 +75 @@
 
   \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

src/temporal/filter.h

@@ -1 +1 @@
 /*
-  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 +22 @@
 #define _AUBIO_FILTER_H
 
-/** \file 
+/** \file
 
   Digital filter
@@ -28 +28 @@
   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
@@ -41 +41 @@
   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] \\
@@ -54 +54 @@
   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().
@@ -60 +60 @@
 
   \example temporal/test-filter.c
- 
+
 */
 
@@ -164 +164 @@
 
 /** delete a filter object
- 
+
   \param f filter object to delete
 

src/temporal/resampler.h

@@ -1 +1 @@
 /*
-  Copyright (C) 2003-2013 Paul Brossier <piem@aubio.org>
+  Copyright (C) 2003-2015 Paul Brossier <piem@aubio.org>
 
   This file is part of aubio.
@@ -23 +23 @@
 
 /** \file
- 
+
  Resampling object
 
  This object resamples an input vector into an output vector using
  libsamplerate. See http://www.mega-nerd.com/SRC/
- 
+
 */
 
@@ -38 +38 @@
 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