From: Paul Brossier Date: Sat, 5 Sep 2015 10:27:38 +0000 (+0200) Subject: Merge branch 'develop' into notes X-Git-Tag: 0.4.4~301 X-Git-Url: https://git.aubio.org/?p=aubio.git;a=commitdiff_plain;h=60fc05b3fadd7da5957eb8c531bfa882d35e6568;hp=b257b60e3e1004a168b81130924fdd1ab4a9f0c3 Merge branch 'develop' into notes --- diff --git a/Makefile b/Makefile index 829101aa..f1258be7 100644 --- a/Makefile +++ b/Makefile @@ -7,7 +7,7 @@ getwaf: curl https://waf.io/waf-1.8.12 > waf @[ -d wafilb ] || rm -fr waflib @chmod +x waf && ./waf --help > /dev/null - @mv .waf-*/waflib . && rm -fr .waf-* + @mv .waf*/waflib . && rm -fr .waf-* @sed '/^#==>$$/,$$d' waf > waf2 && mv waf2 waf @chmod +x waf diff --git a/doc/aubiomfcc.txt b/doc/aubiomfcc.txt index 9084c4a0..afeafe38 100644 --- a/doc/aubiomfcc.txt +++ b/doc/aubiomfcc.txt @@ -1,5 +1,5 @@ NAME - aubiomfcc - a command line tool to compute Mel-frequency Cepstrum Coefficients + aubiomfcc - a command line tool to compute Mel-Frequency Cepstrum Coefficients SYNOPSIS @@ -10,7 +10,11 @@ SYNOPSIS DESCRIPTION - aubiomfcc compute the Mel Frequency Cepstrum Coefficients (MFCC). + aubiomfcc compute the Mel-Frequency Cepstrum Coefficients (MFCC). + + MFCCs are coefficients that make up for the mel-frequency spectrum, a + representation of the short-term power spectrum of a sound. By default, 13 + coefficents are computed using 40 filters. When started with an input source (-i/--input), the coefficients are given on the console, prefixed by their timestamps in seconds. diff --git a/src/aubio.h b/src/aubio.h index 29082876..0d2b416c 100644 --- a/src/aubio.h +++ b/src/aubio.h @@ -1,5 +1,5 @@ /* - Copyright (C) 2003-2013 Paul Brossier + Copyright (C) 2003-2015 Paul Brossier This file is part of aubio. @@ -18,18 +18,18 @@ */ -/** \mainpage - +/** \mainpage + \section introduction Introduction - + aubio is a library to extract annotations from audio signals: it provides a set of functions that take an input audio signal, and output pitch estimates, attack times (onset), beat location estimates, and other annotation tasks. - - \section basics Basics - + + \section basics Basics + All object structures in aubio share the same function prefixes and suffixes: - + - \p new_aubio_foo creates the object \p foo - \p aubio_foo_do executes the object \p foo - \p del_aubio_foo destroys the object \p foo @@ -37,7 +37,7 @@ All memory allocation and deallocation take place in the \p new_ and \p del_ functions. Optionally, more than one \p _do methods are available. Additional parameters can be adjusted and observed using: - + - \p aubio_foo_get_param, getter function, gets the value of a parameter - \p aubio_foo_set_param, setter function, changes the value of a parameter diff --git a/src/aubio_priv.h b/src/aubio_priv.h index 81cf24aa..83d36989 100644 --- a/src/aubio_priv.h +++ b/src/aubio_priv.h @@ -1,5 +1,5 @@ /* - Copyright (C) 2003-2009 Paul Brossier + Copyright (C) 2003-2015 Paul Brossier This file is part of aubio. @@ -20,7 +20,7 @@ /** @file * Private include file - * + * * This file is for inclusion from _within_ the library only. */ @@ -29,7 +29,7 @@ /********************* * - * External includes + * External includes * */ @@ -71,7 +71,7 @@ #include "mathutils.h" /**** - * + * * SYSTEM INTERFACE * */ @@ -196,10 +196,14 @@ typedef enum { /* handy shortcuts */ #define DB2LIN(g) (POW(10.0,(g)*0.05f)) #define LIN2DB(v) (20.0*LOG10(v)) -#define SQR(_a) (_a*_a) - -#define MAX(a,b) ( a > b ? a : b) -#define MIN(a,b) ( a < b ? a : b) +#define SQR(_a) ((_a)*(_a)) + +#ifndef MAX +#define MAX(a,b) (((a)>(b))?(a):(b)) +#endif /* MAX */ +#ifndef MIN +#define MIN(a,b) (((a)<(b))?(a):(b)) +#endif /* MIN */ #define ELEM_SWAP(a,b) { register smpl_t t=(a);(a)=(b);(b)=t; } diff --git a/src/cvec.h b/src/cvec.h index 5db26beb..c77853ff 100644 --- a/src/cvec.h +++ b/src/cvec.h @@ -1,5 +1,5 @@ /* - Copyright (C) 2003-2013 Paul Brossier + Copyright (C) 2003-2015 Paul Brossier This file is part of aubio. diff --git a/src/fmat.h b/src/fmat.h index 363965e4..f844832b 100644 --- a/src/fmat.h +++ b/src/fmat.h @@ -1,5 +1,5 @@ /* - Copyright (C) 2009-2013 Paul Brossier + Copyright (C) 2009-2015 Paul Brossier This file is part of aubio. @@ -62,17 +62,17 @@ void del_fmat(fmat_t *s); \param s vector to read from \param channel channel to read from - \param position sample position to read from + \param position sample position to read from */ smpl_t fmat_get_sample(fmat_t *s, uint_t channel, uint_t position); /** write sample value in a buffer - \param s vector to write to + \param s vector to write to \param data value to write in s->data[channel][position] - \param channel channel to write to - \param position sample position to write to + \param channel channel to write to + \param position sample position to write to */ void fmat_set_sample(fmat_t *s, smpl_t data, uint_t channel, uint_t position); @@ -101,9 +101,9 @@ smpl_t * fmat_get_channel_data (fmat_t *s, uint_t channel); */ smpl_t ** fmat_get_data(fmat_t *s); -/** print out fmat data +/** print out fmat data - \param s vector to print out + \param s vector to print out */ void fmat_print(fmat_t *s); @@ -116,14 +116,14 @@ void fmat_print(fmat_t *s); */ void fmat_set(fmat_t *s, smpl_t val); -/** set all elements to zero +/** set all elements to zero \param s vector to modify */ void fmat_zeros(fmat_t *s); -/** set all elements to ones +/** set all elements to ones \param s vector to modify @@ -148,7 +148,7 @@ void fmat_rev(fmat_t *s); */ void fmat_weight(fmat_t *s, fmat_t *weight); -/** make a copy of a matrix +/** make a copy of a matrix \param s source vector \param t vector to copy to diff --git a/src/fvec.c b/src/fvec.c index a96afd9c..abdf3d1b 100644 --- a/src/fvec.c +++ b/src/fvec.c @@ -21,6 +21,19 @@ #include "aubio_priv.h" #include "fvec.h" +#ifdef HAVE_ACCELERATE +#include +#if !HAVE_AUBIO_DOUBLE +#define aubio_vDSP_mmov vDSP_mmov +#define aubio_vDSP_vmul vDSP_vmul +#define aubio_vDSP_vfill vDSP_vfill +#else /* HAVE_AUBIO_DOUBLE */ +#define aubio_vDSP_mmov vDSP_mmovD +#define aubio_vDSP_vmul vDSP_vmulD +#define aubio_vDSP_vfill vDSP_vfillD +#endif /* HAVE_AUBIO_DOUBLE */ +#endif + fvec_t * new_fvec( uint_t length) { fvec_t * s; if ((sint_t)length <= 0) { @@ -60,17 +73,25 @@ void fvec_print(fvec_t *s) { } void fvec_set_all (fvec_t *s, smpl_t val) { +#ifndef HAVE_ACCELERATE uint_t j; for (j=0; j< s->length; j++) { s->data[j] = val; } +#else + aubio_vDSP_vfill(&val, s->data, 1, s->length); +#endif } void fvec_zeros(fvec_t *s) { -#if HAVE_MEMCPY_HACKS +#if !defined(HAVE_MEMCPY_HACKS) && !defined(HAVE_ACCELERATE) + fvec_set_all (s, 0.); +#else +#if defined(HAVE_MEMCPY_HACKS) memset(s->data, 0, s->length * sizeof(smpl_t)); #else - fvec_set_all (s, 0.); + aubio_vDSP_vclr(s->data, 1, s->length); +#endif #endif } @@ -86,11 +107,15 @@ void fvec_rev(fvec_t *s) { } void fvec_weight(fvec_t *s, fvec_t *weight) { +#ifndef HAVE_ACCELERATE uint_t j; uint_t length = MIN(s->length, weight->length); for (j=0; j< length; j++) { s->data[j] *= weight->data[j]; } +#else + aubio_vDSP_vmul(s->data, 1, weight->data, 1, s->data, 1, s->length); +#endif /* HAVE_ACCELERATE */ } void fvec_copy(fvec_t *s, fvec_t *t) { @@ -99,12 +124,16 @@ void fvec_copy(fvec_t *s, fvec_t *t) { s->length, t->length); return; } -#if HAVE_MEMCPY_HACKS - memcpy(t->data, s->data, t->length * sizeof(smpl_t)); -#else +#if !defined(HAVE_MEMCPY_HACKS) && !defined(HAVE_ACCELERATE) uint_t j; for (j=0; j< t->length; j++) { t->data[j] = s->data[j]; } +#else +#if defined(HAVE_MEMCPY_HACKS) + memcpy(t->data, s->data, t->length * sizeof(smpl_t)); +#else + aubio_vDSP_mmov(s->data, t->data, 1, s->length, 1, 1); +#endif #endif } diff --git a/src/fvec.h b/src/fvec.h index ee2f8739..c79295de 100644 --- a/src/fvec.h +++ b/src/fvec.h @@ -1,5 +1,5 @@ /* - Copyright (C) 2003-2013 Paul Brossier + Copyright (C) 2003-2015 Paul Brossier This file is part of aubio. @@ -86,16 +86,16 @@ void del_fvec(fvec_t *s); /** read sample value in a buffer \param s vector to read from - \param position sample position to read from + \param position sample position to read from */ smpl_t fvec_get_sample(fvec_t *s, uint_t position); /** write sample value in a buffer - \param s vector to write to + \param s vector to write to \param data value to write in s->data[position] - \param position sample position to write to + \param position sample position to write to */ void fvec_set_sample(fvec_t *s, smpl_t data, uint_t position); @@ -107,9 +107,9 @@ void fvec_set_sample(fvec_t *s, smpl_t data, uint_t position); */ smpl_t * fvec_get_data(fvec_t *s); -/** print out fvec data +/** print out fvec data - \param s vector to print out + \param s vector to print out */ void fvec_print(fvec_t *s); @@ -122,14 +122,14 @@ void fvec_print(fvec_t *s); */ void fvec_set_all (fvec_t *s, smpl_t val); -/** set all elements to zero +/** set all elements to zero \param s vector to modify */ void fvec_zeros(fvec_t *s); -/** set all elements to ones +/** set all elements to ones \param s vector to modify diff --git a/src/io/audio_unit.c b/src/io/audio_unit.c index eab299f3..a21906ae 100644 --- a/src/io/audio_unit.c +++ b/src/io/audio_unit.c @@ -19,7 +19,7 @@ */ #include "config.h" -#ifdef TARGET_OS_IPHONE +#ifdef HAVE_AUDIO_UNIT #include "aubio_priv.h" #include "fvec.h" @@ -774,4 +774,4 @@ uint_t del_aubio_audio_unit(aubio_audio_unit_t *o) return (int)err; } -#endif /* TARGET_OS_IPHONE */ +#endif /* HAVE_AUDIO_UNIT */ diff --git a/src/io/sink_sndfile.c b/src/io/sink_sndfile.c index 82d6167b..9fed7606 100644 --- a/src/io/sink_sndfile.c +++ b/src/io/sink_sndfile.c @@ -33,6 +33,12 @@ #define MAX_CHANNELS 6 #define MAX_SIZE 4096 +#if !HAVE_AUBIO_DOUBLE +#define aubio_sf_write_smpl sf_write_float +#else /* HAVE_AUBIO_DOUBLE */ +#define aubio_sf_write_smpl sf_write_double +#endif /* HAVE_AUBIO_DOUBLE */ + struct _aubio_sink_sndfile_t { uint_t samplerate; uint_t channels; @@ -125,7 +131,7 @@ uint_t aubio_sink_sndfile_open(aubio_sink_sndfile_t *s) { /* show libsndfile err msg */ AUBIO_ERR("sink_sndfile: Failed opening %s. %s\n", s->path, sf_strerror (NULL)); return AUBIO_FAIL; - } + } s->scratch_size = s->max_size*s->channels; /* allocate data for de/interleaving reallocated when needed. */ @@ -134,13 +140,13 @@ uint_t aubio_sink_sndfile_open(aubio_sink_sndfile_t *s) { s->max_size, s->channels, MAX_CHANNELS * MAX_CHANNELS); return AUBIO_FAIL; } - s->scratch_data = AUBIO_ARRAY(float,s->scratch_size); + s->scratch_data = AUBIO_ARRAY(smpl_t,s->scratch_size); return AUBIO_OK; } void aubio_sink_sndfile_do(aubio_sink_sndfile_t *s, fvec_t * write_data, uint_t write){ - uint_t i, j, channels = s->channels; + uint_t i, j, channels = s->channels; int nsamples = 0; smpl_t *pwrite; sf_count_t written_frames; @@ -161,7 +167,7 @@ void aubio_sink_sndfile_do(aubio_sink_sndfile_t *s, fvec_t * write_data, uint_t } } - written_frames = sf_write_float (s->handle, s->scratch_data, nsamples); + written_frames = aubio_sf_write_smpl (s->handle, s->scratch_data, nsamples); if (written_frames/channels != write) { AUBIO_WRN("sink_sndfile: trying to write %d frames to %s, but only %d could be written\n", write, s->path, (uint_t)written_frames); @@ -170,7 +176,7 @@ void aubio_sink_sndfile_do(aubio_sink_sndfile_t *s, fvec_t * write_data, uint_t } void aubio_sink_sndfile_do_multi(aubio_sink_sndfile_t *s, fmat_t * write_data, uint_t write){ - uint_t i, j, channels = s->channels; + uint_t i, j, channels = s->channels; int nsamples = 0; smpl_t *pwrite; sf_count_t written_frames; @@ -191,7 +197,7 @@ void aubio_sink_sndfile_do_multi(aubio_sink_sndfile_t *s, fmat_t * write_data, u } } - written_frames = sf_write_float (s->handle, s->scratch_data, nsamples); + written_frames = aubio_sf_write_smpl (s->handle, s->scratch_data, nsamples); if (written_frames/channels != write) { AUBIO_WRN("sink_sndfile: trying to write %d frames to %s, but only %d could be written\n", write, s->path, (uint_t)written_frames); diff --git a/src/io/source_avcodec.c b/src/io/source_avcodec.c index 22d22974..5cfd27de 100644 --- a/src/io/source_avcodec.c +++ b/src/io/source_avcodec.c @@ -23,6 +23,9 @@ #ifdef HAVE_LIBAV +// determine whether we use libavformat from ffmpe or libav +#define FFMPEG_LIBAVFORMAT (LIBAVFORMAT_VERSION_MICRO > 99) + #include #include #include @@ -100,7 +103,11 @@ aubio_source_avcodec_t * new_aubio_source_avcodec(char_t * path, uint_t samplera } // try to make sure max_analyze_duration is big enough for most songs +#if FFMPEG_LIBAVFORMAT + avFormatCtx->max_analyze_duration2 *= 100; +#else avFormatCtx->max_analyze_duration *= 100; +#endif // retrieve stream information if ( (err = avformat_find_stream_info(avFormatCtx, NULL)) < 0 ) { diff --git a/src/io/source_sndfile.c b/src/io/source_sndfile.c index e567a239..a2ef1f24 100644 --- a/src/io/source_sndfile.c +++ b/src/io/source_sndfile.c @@ -90,7 +90,7 @@ aubio_source_sndfile_t * new_aubio_source_sndfile(char_t * path, uint_t samplera /* show libsndfile err msg */ AUBIO_ERR("source_sndfile: Failed opening %s: %s\n", s->path, sf_strerror (NULL)); goto beach; - } + } /* get input specs */ s->input_samplerate = sfinfo.samplerate; diff --git a/src/lvec.h b/src/lvec.h index c5750f15..fc261be5 100644 --- a/src/lvec.h +++ b/src/lvec.h @@ -1,5 +1,5 @@ /* - Copyright (C) 2003-2013 Paul Brossier + Copyright (C) 2003-2015 Paul Brossier This file is part of aubio. @@ -61,16 +61,16 @@ void del_lvec(lvec_t *s); /** read sample value in a buffer \param s vector to read from - \param position sample position to read from + \param position sample position to read from */ lsmp_t lvec_get_sample(lvec_t *s, uint_t position); /** write sample value in a buffer - \param s vector to write to + \param s vector to write to \param data value to write in s->data[position] - \param position sample position to write to + \param position sample position to write to */ void lvec_set_sample(lvec_t *s, lsmp_t data, uint_t position); @@ -82,9 +82,9 @@ void lvec_set_sample(lvec_t *s, lsmp_t data, uint_t position); */ lsmp_t * lvec_get_data(lvec_t *s); -/** print out lvec data +/** print out lvec data - \param s vector to print out + \param s vector to print out */ void lvec_print(lvec_t *s); @@ -97,14 +97,14 @@ void lvec_print(lvec_t *s); */ void lvec_set_all(lvec_t *s, smpl_t val); -/** set all elements to zero +/** set all elements to zero \param s vector to modify */ void lvec_zeros(lvec_t *s); -/** set all elements to ones +/** set all elements to ones \param s vector to modify diff --git a/src/mathutils.c b/src/mathutils.c index 2298f468..4657bd8e 100644 --- a/src/mathutils.c +++ b/src/mathutils.c @@ -158,22 +158,39 @@ aubio_unwrap2pi (smpl_t phase) smpl_t fvec_mean (fvec_t * s) { - uint_t j; smpl_t tmp = 0.0; +#ifndef HAVE_ACCELERATE + uint_t j; for (j = 0; j < s->length; j++) { tmp += s->data[j]; } return tmp / (smpl_t) (s->length); +#else +#if !HAVE_AUBIO_DOUBLE + vDSP_meanv(s->data, 1, &tmp, s->length); +#else /* HAVE_AUBIO_DOUBLE */ + vDSP_meanvD(s->data, 1, &tmp, s->length); +#endif /* HAVE_AUBIO_DOUBLE */ + return tmp; +#endif /* HAVE_ACCELERATE */ } smpl_t fvec_sum (fvec_t * s) { - uint_t j; smpl_t tmp = 0.0; +#ifndef HAVE_ACCELERATE + uint_t j; for (j = 0; j < s->length; j++) { tmp += s->data[j]; } +#else +#if !HAVE_AUBIO_DOUBLE + vDSP_sve(s->data, 1, &tmp, s->length); +#else /* HAVE_AUBIO_DOUBLE */ + vDSP_sveD(s->data, 1, &tmp, s->length); +#endif /* HAVE_AUBIO_DOUBLE */ +#endif /* HAVE_ACCELERATE */ return tmp; } diff --git a/src/mathutils.h b/src/mathutils.h index fe04647e..77cc7d50 100644 --- a/src/mathutils.h +++ b/src/mathutils.h @@ -1,5 +1,5 @@ /* - Copyright (C) 2003-2014 Paul Brossier + Copyright (C) 2003-2015 Paul Brossier This file is part of aubio. @@ -82,20 +82,20 @@ uint_t fvec_min_elem (fvec_t * s); uint_t fvec_max_elem (fvec_t * s); /** swap the left and right halves of a vector - + This function swaps the left part of the signal with the right part of the signal. Therefore \f$ a[0], a[1], ..., a[\frac{N}{2}], a[\frac{N}{2}+1], ..., a[N-1], a[N] \f$ - + becomes - + \f$ a[\frac{N}{2}+1], ..., a[N-1], a[N], a[0], a[1], ..., a[\frac{N}{2}] \f$ This operation, known as 'fftshift' in the Matlab Signal Processing Toolbox, can be used before computing the FFT to simplify the phase relationship of the resulting spectrum. See Amalia de Götzen's paper referred to above. - + */ void fvec_shift (fvec_t * v); @@ -111,20 +111,20 @@ smpl_t fvec_sum (fvec_t * v); /** compute the High Frequency Content of a vector The High Frequency Content is defined as \f$ \sum_0^{N-1} (k+1) v[k] \f$. - - \param v vector to get the energy from + + \param v vector to get the energy from \return the HFC of v - + */ smpl_t fvec_local_hfc (fvec_t * v); -/** computes the p-norm of a vector - +/** computes the p-norm of a vector + Computes the p-norm of a vector for \f$ p = \alpha \f$ \f$ L^p = ||x||_p = (|x_1|^p + |x_2|^p + ... + |x_n|^p ) ^ \frac{1}{p} \f$ - + If p = 1, the result is the Manhattan distance. If p = 2, the result is the Euclidean distance. @@ -133,7 +133,7 @@ smpl_t fvec_local_hfc (fvec_t * v); input vector. References: - + - \f$L^p\f$ space on Wikipedia @@ -141,13 +141,13 @@ input vector. \param p order of the computed norm \return the p-norm of v - + */ smpl_t fvec_alpha_norm (fvec_t * v, smpl_t p); /** alpha normalisation - This function divides all elements of a vector by the p-norm as computed by + This function divides all elements of a vector by the p-norm as computed by fvec_alpha_norm(). \param v vector to compute norm from @@ -165,7 +165,7 @@ void fvec_alpha_normalise (fvec_t * v, smpl_t p); void fvec_add (fvec_t * v, smpl_t c); /** remove the minimum value of the vector to each elements - + \param v vector to remove minimum from */ @@ -176,14 +176,14 @@ void fvec_min_removal (fvec_t * v); This function computes the moving median threshold value of at the given position of a vector, taking the median among post elements before and up to pre elements after pos. - + \param v input vector \param tmp temporary vector of length post+1+pre - \param post length of causal part to take before pos + \param post length of causal part to take before pos \param pre length of anti-causal part to take after pos - \param pos index to compute threshold for + \param pos index to compute threshold for - \return moving median threshold value + \return moving median threshold value */ smpl_t fvec_moving_thres (fvec_t * v, fvec_t * tmp, uint_t post, uint_t pre, @@ -196,13 +196,13 @@ moving median threshold computed at p. \param v input vector \param tmp temporary vector of length post+1+pre - \param post length of causal part to take before pos + \param post length of causal part to take before pos \param pre length of anti-causal part to take after pos */ void fvec_adapt_thres (fvec_t * v, fvec_t * tmp, uint_t post, uint_t pre); -/** returns the median of a vector +/** returns the median of a vector The QuickSelect routine is based on the algorithm described in "Numerical recipes in C", Second Edition, Cambridge University Press, 1992, Section 8.5, @@ -215,7 +215,7 @@ and in the Public Domain. \param v vector to get median from \return the median of v - + */ smpl_t fvec_median (fvec_t * v); @@ -248,13 +248,13 @@ smpl_t fvec_quadratic_peak_pos (fvec_t * x, uint_t p); smpl_t fvec_quadratic_peak_mag (fvec_t * x, smpl_t p); /** Quadratic interpolation using Lagrange polynomial. - + Inspired from ``Comparison of interpolation algorithms in real-time sound -processing'', Vladimir Arnost, - - \param s0,s1,s2 are 3 consecutive samples of a curve +processing'', Vladimir Arnost, + + \param s0,s1,s2 are 3 consecutive samples of a curve \param pf is the floating point index [0;2] - + \return \f$ s0 + (pf/2.)*((pf-3.)*s0-2.*(pf-2.)*s1+(pf-1.)*s2); \f$ */ diff --git a/src/musicutils.h b/src/musicutils.h index 2f3103aa..14a05d89 100644 --- a/src/musicutils.h +++ b/src/musicutils.h @@ -1,5 +1,5 @@ /* - Copyright (C) 2003-2013 Paul Brossier + Copyright (C) 2003-2015 Paul Brossier This file is part of aubio. @@ -29,11 +29,11 @@ extern "C" { #endif -/** create window +/** create window \param window_type type of the window to create \param size length of the window to create (see fvec_set_window()) - + */ fvec_t *new_aubio_window (char_t * window_type, uint_t size); @@ -49,7 +49,7 @@ fvec_t *new_aubio_window (char_t * window_type, uint_t size); "default" is equivalent to "hanningz". References: - + - Window function on Wikipedia - Amalia de Götzen, Nicolas Bernardini, and Daniel Arfib. Traditional (?) @@ -68,7 +68,7 @@ uint_t fvec_set_window (fvec_t * window, char_t * window_type); range \f$ [-\pi, \pi] \f$. \param phase unwrapped phase to map to the unit circle - + \return equivalent phase wrapped to the unit circle */ @@ -93,7 +93,7 @@ smpl_t aubio_freqtomidi (smpl_t freq); smpl_t aubio_miditofreq (smpl_t midi); /** clean up cached memory at the end of program - + This function should be used at the end of programs to purge all cached memory. So far it is only useful to clean FFTW's cache. @@ -137,7 +137,7 @@ smpl_t aubio_level_lin (fvec_t * v); smpl_t aubio_db_spl (fvec_t * v); /** check if buffer level in dB SPL is under a given threshold - + \param v vector to get level from \param threshold threshold in dB SPL diff --git a/src/onset/onset.c b/src/onset/onset.c index 98660aad..d8f10d67 100644 --- a/src/onset/onset.c +++ b/src/onset/onset.c @@ -67,11 +67,16 @@ void aubio_onset_do (aubio_onset_t *o, fvec_t * input, fvec_t * onset) } } } else { - // we are at the beginning of the file, and we don't find silence - if (o->total_frames <= o->delay && o->last_onset < o ->minioi && aubio_silence_detection(input, o->silence) == 0) { - //AUBIO_DBG ("beginning of file is not silent, marking as onset\n"); - isonset = o->delay / o->hop_size; - o->last_onset = o->delay; + // we are at the beginning of the file + if (o->total_frames <= o->delay) { + // and we don't find silence + if (aubio_silence_detection(input, o->silence) == 0) { + uint_t new_onset = o->total_frames; + if (o->total_frames == 0 || o->last_onset + o->minioi < new_onset) { + isonset = o->delay / o->hop_size; + o->last_onset = o->total_frames + o->delay; + } + } } } onset->data[0] = isonset; diff --git a/src/onset/peakpicker.h b/src/onset/peakpicker.h index ece54048..c137cc86 100644 --- a/src/onset/peakpicker.h +++ b/src/onset/peakpicker.h @@ -18,12 +18,12 @@ */ -/** \file - +/** \file + Peak picking utilities function \example onset/test-peakpicker.c - + */ #ifndef _AUBIO_PEAKPICK_H diff --git a/src/pitch/pitchyinfft.h b/src/pitch/pitchyinfft.h index ee89d0f3..8f32ee91 100644 --- a/src/pitch/pitchyinfft.h +++ b/src/pitch/pitchyinfft.h @@ -1,5 +1,5 @@ /* - Copyright (C) 2003-2013 Paul Brossier + Copyright (C) 2003-2015 Paul Brossier This file is part of aubio. @@ -19,14 +19,14 @@ */ /** \file - + Pitch detection using a spectral implementation of the YIN algorithm - + This algorithm was derived from the YIN algorithm. In this implementation, a Fourier transform is used to compute a tapered square difference function, which allows spectral weighting. Because the difference function is tapered, the selection of the period is simplified. - + Paul Brossier, [Automatic annotation of musical audio for interactive systems](http://aubio.org/phd/), Chapter 3, Pitch Analysis, PhD thesis, Centre for Digital music, Queen Mary University of London, London, UK, 2006. @@ -45,40 +45,40 @@ extern "C" { /** pitch detection object */ typedef struct _aubio_pitchyinfft_t aubio_pitchyinfft_t; -/** execute pitch detection on an input buffer - +/** execute pitch detection on an input buffer + \param o pitch detection object as returned by new_aubio_pitchyinfft \param samples_in input signal vector (length as specified at creation time) \param cands_out pitch period candidates, in samples - + */ void aubio_pitchyinfft_do (aubio_pitchyinfft_t * o, fvec_t * samples_in, fvec_t * cands_out); /** creation of the pitch detection object - + \param samplerate samplerate of the input signal - \param buf_size size of the input buffer to analyse - + \param buf_size size of the input buffer to analyse + */ aubio_pitchyinfft_t *new_aubio_pitchyinfft (uint_t samplerate, uint_t buf_size); /** deletion of the pitch detection object - + \param o pitch detection object as returned by new_aubio_pitchyinfft() - + */ void del_aubio_pitchyinfft (aubio_pitchyinfft_t * o); -/** get tolerance parameter for YIN algorithm - - \param o YIN pitch detection object +/** get tolerance parameter for YIN algorithm + + \param o YIN pitch detection object \return tolerance parameter for minima selection [default 0.15] */ smpl_t aubio_pitchyinfft_get_tolerance (aubio_pitchyinfft_t * o); -/** set tolerance parameter for YIN algorithm - - \param o YIN pitch detection object +/** set tolerance parameter for YIN algorithm + + \param o YIN pitch detection object \param tol tolerance parameter for minima selection [default 0.15] */ diff --git a/src/spectral/fft.c b/src/spectral/fft.c index 3b29b8ee..8841ca2b 100644 --- a/src/spectral/fft.c +++ b/src/spectral/fft.c @@ -66,12 +66,12 @@ typedef FFTW_TYPE fft_data_t; #if HAVE_AUBIO_DOUBLE #warning "Using aubio in double precision with fftw3 in single precision" #endif /* HAVE_AUBIO_DOUBLE */ -#define real_t float +#define real_t float #else /* HAVE_FFTW3F */ #if !HAVE_AUBIO_DOUBLE #warning "Using aubio in single precision with fftw3 in double precision" #endif /* HAVE_AUBIO_DOUBLE */ -#define real_t double +#define real_t double #endif /* HAVE_FFTW3F */ // a global mutex for FFTW thread safety @@ -82,6 +82,36 @@ pthread_mutex_t aubio_fftw_mutex = PTHREAD_MUTEX_INITIALIZER; // https://developer.apple.com/library/mac/#documentation/Accelerate/Reference/vDSPRef/Reference/reference.html #include +#if !HAVE_AUBIO_DOUBLE +#define aubio_vDSP_ctoz vDSP_ctoz +#define aubio_vDSP_fft_zrip vDSP_fft_zrip +#define aubio_vDSP_ztoc vDSP_ztoc +#define aubio_vDSP_zvmags vDSP_zvmags +#define aubio_vDSP_zvphas vDSP_zvphas +#define aubio_vDSP_vsadd vDSP_vsadd +#define aubio_vDSP_vsmul vDSP_vsmul +#define aubio_vDSP_create_fftsetup vDSP_create_fftsetup +#define aubio_vDSP_destroy_fftsetup vDSP_destroy_fftsetup +#define aubio_DSPComplex DSPComplex +#define aubio_DSPSplitComplex DSPSplitComplex +#define aubio_FFTSetup FFTSetup +#define aubio_vvsqrt vvsqrtf +#else +#define aubio_vDSP_ctoz vDSP_ctozD +#define aubio_vDSP_fft_zrip vDSP_fft_zripD +#define aubio_vDSP_ztoc vDSP_ztocD +#define aubio_vDSP_zvmags vDSP_zvmagsD +#define aubio_vDSP_zvphas vDSP_zvphasD +#define aubio_vDSP_vsadd vDSP_vsaddD +#define aubio_vDSP_vsmul vDSP_vsmulD +#define aubio_vDSP_create_fftsetup vDSP_create_fftsetupD +#define aubio_vDSP_destroy_fftsetup vDSP_destroy_fftsetupD +#define aubio_DSPComplex DSPDoubleComplex +#define aubio_DSPSplitComplex DSPDoubleSplitComplex +#define aubio_FFTSetup FFTSetupD +#define aubio_vvsqrt vvsqrt +#endif /* HAVE_AUBIO_DOUBLE */ + #else // using OOURA // let's use ooura instead extern void rdft(int, int, smpl_t *, int *, smpl_t *); @@ -95,19 +125,13 @@ struct _aubio_fft_t { #ifdef HAVE_FFTW3 // using FFTW3 real_t *in, *out; fftw_plan pfw, pbw; - fft_data_t * specdata; /* complex spectral data */ + fft_data_t * specdata; /* complex spectral data */ #else #ifdef HAVE_ACCELERATE // using ACCELERATE int log2fftsize; -#if !HAVE_AUBIO_DOUBLE - FFTSetup fftSetup; - DSPSplitComplex spec; - float *in, *out; -#else - FFTSetupD fftSetup; - DSPDoubleSplitComplex spec; - double *in, *out; -#endif + aubio_FFTSetup fftSetup; + aubio_DSPSplitComplex spec; + smpl_t *in, *out; #else // using OOURA smpl_t *in, *out; smpl_t *w; @@ -119,6 +143,10 @@ struct _aubio_fft_t { aubio_fft_t * new_aubio_fft (uint_t winsize) { aubio_fft_t * s = AUBIO_NEW(aubio_fft_t); + if ((sint_t)winsize < 1) { + AUBIO_ERR("fft: got winsize %d, but can not be < 1\n", winsize); + goto beach; + } #ifdef HAVE_FFTW3 uint_t i; s->winsize = winsize; @@ -153,20 +181,17 @@ aubio_fft_t * new_aubio_fft (uint_t winsize) { s->fft_size = winsize; s->compspec = new_fvec(winsize); s->log2fftsize = (uint_t)log2f(s->fft_size); -#if !HAVE_AUBIO_DOUBLE - s->in = AUBIO_ARRAY(float, s->fft_size); - s->out = AUBIO_ARRAY(float, s->fft_size); - s->spec.realp = AUBIO_ARRAY(float, s->fft_size/2); - s->spec.imagp = AUBIO_ARRAY(float, s->fft_size/2); - s->fftSetup = vDSP_create_fftsetup(s->log2fftsize, FFT_RADIX2); -#else - s->in = AUBIO_ARRAY(double, s->fft_size); - s->out = AUBIO_ARRAY(double, s->fft_size); - s->spec.realp = AUBIO_ARRAY(double, s->fft_size/2); - s->spec.imagp = AUBIO_ARRAY(double, s->fft_size/2); - s->fftSetup = vDSP_create_fftsetupD(s->log2fftsize, FFT_RADIX2); -#endif + s->in = AUBIO_ARRAY(smpl_t, s->fft_size); + s->out = AUBIO_ARRAY(smpl_t, s->fft_size); + s->spec.realp = AUBIO_ARRAY(smpl_t, s->fft_size/2); + s->spec.imagp = AUBIO_ARRAY(smpl_t, s->fft_size/2); + s->fftSetup = aubio_vDSP_create_fftsetup(s->log2fftsize, FFT_RADIX2); #else // using OOURA + if (aubio_is_power_of_two(winsize) != 1) { + AUBIO_ERR("fft: can only create with sizes power of two," + " requested %d\n", winsize); + goto beach; + } s->winsize = winsize; s->fft_size = winsize / 2 + 1; s->compspec = new_fvec(winsize); @@ -178,6 +203,9 @@ aubio_fft_t * new_aubio_fft (uint_t winsize) { #endif /* HAVE_ACCELERATE */ #endif /* HAVE_FFTW3 */ return s; +beach: + AUBIO_FREE(s); + return NULL; } void del_aubio_fft(aubio_fft_t * s) { @@ -191,11 +219,7 @@ void del_aubio_fft(aubio_fft_t * s) { #ifdef HAVE_ACCELERATE // using ACCELERATE AUBIO_FREE(s->spec.realp); AUBIO_FREE(s->spec.imagp); -#if !HAVE_AUBIO_DOUBLE - vDSP_destroy_fftsetup(s->fftSetup); -#else - vDSP_destroy_fftsetupD(s->fftSetup); -#endif + aubio_vDSP_destroy_fftsetup(s->fftSetup); #else // using OOURA AUBIO_FREE(s->w); AUBIO_FREE(s->ip); @@ -218,9 +242,13 @@ void aubio_fft_rdo(aubio_fft_t * s, cvec_t * spectrum, fvec_t * output) { void aubio_fft_do_complex(aubio_fft_t * s, fvec_t * input, fvec_t * compspec) { uint_t i; +#ifndef HAVE_MEMCPY_HACKS for (i=0; i < s->winsize; i++) { s->in[i] = input->data[i]; } +#else + memcpy(s->in, input->data, s->winsize * sizeof(smpl_t)); +#endif /* HAVE_MEMCPY_HACKS */ #ifdef HAVE_FFTW3 // using FFTW3 fftw_execute(s->pfw); #ifdef HAVE_COMPLEX_H @@ -237,17 +265,10 @@ void aubio_fft_do_complex(aubio_fft_t * s, fvec_t * input, fvec_t * compspec) { #endif /* HAVE_COMPLEX_H */ #else /* HAVE_FFTW3 */ #ifdef HAVE_ACCELERATE // using ACCELERATE -#if !HAVE_AUBIO_DOUBLE - // convert real data to even/odd format used in vDSP - vDSP_ctoz((DSPComplex*)s->in, 2, &s->spec, 1, s->fft_size/2); - // compute the FFT - vDSP_fft_zrip(s->fftSetup, &s->spec, 1, s->log2fftsize, FFT_FORWARD); -#else // convert real data to even/odd format used in vDSP - vDSP_ctozD((DSPDoubleComplex*)s->in, 2, &s->spec, 1, s->fft_size/2); + aubio_vDSP_ctoz((aubio_DSPComplex*)s->in, 2, &s->spec, 1, s->fft_size/2); // compute the FFT - vDSP_fft_zripD(s->fftSetup, &s->spec, 1, s->log2fftsize, FFT_FORWARD); -#endif + aubio_vDSP_fft_zrip(s->fftSetup, &s->spec, 1, s->log2fftsize, FFT_FORWARD); // convert from vDSP complex split to [ r0, r1, ..., rN, iN-1, .., i2, i1] compspec->data[0] = s->spec.realp[0]; compspec->data[s->fft_size / 2] = s->spec.imagp[0]; @@ -257,11 +278,7 @@ void aubio_fft_do_complex(aubio_fft_t * s, fvec_t * input, fvec_t * compspec) { } // apply scaling smpl_t scale = 1./2.; -#if !HAVE_AUBIO_DOUBLE - vDSP_vsmul(compspec->data, 1, &scale, compspec->data, 1, s->fft_size); -#else - vDSP_vsmulD(compspec->data, 1, &scale, compspec->data, 1, s->fft_size); -#endif + aubio_vDSP_vsmul(compspec->data, 1, &scale, compspec->data, 1, s->fft_size); #else // using OOURA rdft(s->winsize, 1, s->in, s->ip, s->w); compspec->data[0] = s->in[0]; @@ -304,27 +321,15 @@ void aubio_fft_rdo_complex(aubio_fft_t * s, fvec_t * compspec, fvec_t * output) s->out[2 * i] = compspec->data[i]; s->out[2 * i + 1] = compspec->data[s->winsize - i]; } -#if !HAVE_AUBIO_DOUBLE - // convert to split complex format used in vDSP - vDSP_ctoz((DSPComplex*)s->out, 2, &s->spec, 1, s->fft_size/2); - // compute the FFT - vDSP_fft_zrip(s->fftSetup, &s->spec, 1, s->log2fftsize, FFT_INVERSE); - // convert result to real output - vDSP_ztoc(&s->spec, 1, (DSPComplex*)output->data, 2, s->fft_size/2); - // apply scaling - smpl_t scale = 1.0 / s->winsize; - vDSP_vsmul(output->data, 1, &scale, output->data, 1, s->fft_size); -#else // convert to split complex format used in vDSP - vDSP_ctozD((DSPDoubleComplex*)s->out, 2, &s->spec, 1, s->fft_size/2); + aubio_vDSP_ctoz((aubio_DSPComplex*)s->out, 2, &s->spec, 1, s->fft_size/2); // compute the FFT - vDSP_fft_zripD(s->fftSetup, &s->spec, 1, s->log2fftsize, FFT_INVERSE); + aubio_vDSP_fft_zrip(s->fftSetup, &s->spec, 1, s->log2fftsize, FFT_INVERSE); // convert result to real output - vDSP_ztocD(&s->spec, 1, (DSPDoubleComplex*)output->data, 2, s->fft_size/2); + aubio_vDSP_ztoc(&s->spec, 1, (aubio_DSPComplex*)output->data, 2, s->fft_size/2); // apply scaling smpl_t scale = 1.0 / s->winsize; - vDSP_vsmulD(output->data, 1, &scale, output->data, 1, s->fft_size); -#endif + aubio_vDSP_vsmul(output->data, 1, &scale, output->data, 1, s->fft_size); #else // using OOURA smpl_t scale = 2.0 / s->winsize; s->out[0] = compspec->data[0]; @@ -391,7 +396,7 @@ void aubio_fft_get_imag(cvec_t * spectrum, fvec_t * compspec) { void aubio_fft_get_real(cvec_t * spectrum, fvec_t * compspec) { uint_t i; for (i = 0; i < compspec->length / 2 + 1; i++) { - compspec->data[i] = + compspec->data[i] = spectrum->norm[i]*COS(spectrum->phas[i]); } } diff --git a/src/spectral/fft.h b/src/spectral/fft.h index 67e268f1..98b984e6 100644 --- a/src/spectral/fft.h +++ b/src/spectral/fft.h @@ -18,7 +18,7 @@ */ -/** \file +/** \file Fast Fourier Transform @@ -39,7 +39,7 @@ extern "C" { #endif /** FFT object - + This object computes forward and backward FFTs. */ @@ -51,7 +51,7 @@ typedef struct _aubio_fft_t aubio_fft_t; */ aubio_fft_t * new_aubio_fft (uint_t size); -/** delete FFT object +/** delete FFT object \param s fft object as returned by new_aubio_fft @@ -61,16 +61,16 @@ void del_aubio_fft(aubio_fft_t * s); /** compute forward FFT \param s fft object as returned by new_aubio_fft - \param input input signal - \param spectrum output spectrum + \param input input signal + \param spectrum output spectrum */ void aubio_fft_do (aubio_fft_t *s, fvec_t * input, cvec_t * spectrum); /** compute backward (inverse) FFT \param s fft object as returned by new_aubio_fft - \param spectrum input spectrum - \param output output signal + \param spectrum input spectrum + \param output output signal */ void aubio_fft_rdo (aubio_fft_t *s, cvec_t * spectrum, fvec_t * output); @@ -78,7 +78,7 @@ void aubio_fft_rdo (aubio_fft_t *s, cvec_t * spectrum, fvec_t * output); /** compute forward FFT \param s fft object as returned by new_aubio_fft - \param input real input signal + \param input real input signal \param compspec complex output fft real/imag */ @@ -86,53 +86,53 @@ void aubio_fft_do_complex (aubio_fft_t *s, fvec_t * input, fvec_t * compspec); /** compute backward (inverse) FFT from real/imag \param s fft object as returned by new_aubio_fft - \param compspec real/imag input fft array - \param output real output array + \param compspec real/imag input fft array + \param output real output array */ void aubio_fft_rdo_complex (aubio_fft_t *s, fvec_t * compspec, fvec_t * output); -/** convert real/imag spectrum to norm/phas spectrum +/** convert real/imag spectrum to norm/phas spectrum - \param compspec real/imag input fft array - \param spectrum cvec norm/phas output array + \param compspec real/imag input fft array + \param spectrum cvec norm/phas output array */ void aubio_fft_get_spectrum(fvec_t * compspec, cvec_t * spectrum); -/** convert real/imag spectrum to norm/phas spectrum +/** convert real/imag spectrum to norm/phas spectrum - \param compspec real/imag input fft array - \param spectrum cvec norm/phas output array + \param compspec real/imag input fft array + \param spectrum cvec norm/phas output array */ void aubio_fft_get_realimag(cvec_t * spectrum, fvec_t * compspec); -/** compute phas spectrum from real/imag parts +/** compute phas spectrum from real/imag parts - \param compspec real/imag input fft array - \param spectrum cvec norm/phas output array + \param compspec real/imag input fft array + \param spectrum cvec norm/phas output array */ void aubio_fft_get_phas(fvec_t * compspec, cvec_t * spectrum); -/** compute imaginary part from the norm/phas cvec +/** compute imaginary part from the norm/phas cvec - \param spectrum norm/phas input array - \param compspec real/imag output fft array + \param spectrum norm/phas input array + \param compspec real/imag output fft array */ void aubio_fft_get_imag(cvec_t * spectrum, fvec_t * compspec); -/** compute norm component from real/imag parts +/** compute norm component from real/imag parts - \param compspec real/imag input fft array - \param spectrum cvec norm/phas output array + \param compspec real/imag input fft array + \param spectrum cvec norm/phas output array */ void aubio_fft_get_norm(fvec_t * compspec, cvec_t * spectrum); -/** compute real part from norm/phas components +/** compute real part from norm/phas components - \param spectrum norm/phas input array - \param compspec real/imag output fft array + \param spectrum norm/phas input array + \param compspec real/imag output fft array */ void aubio_fft_get_real(cvec_t * spectrum, fvec_t * compspec); diff --git a/src/spectral/filterbank_mel.h b/src/spectral/filterbank_mel.h index 0d4a51fb..8fbb523e 100644 --- a/src/spectral/filterbank_mel.h +++ b/src/spectral/filterbank_mel.h @@ -58,7 +58,7 @@ uint_t aubio_filterbank_set_triangle_bands (aubio_filterbank_t * fb, \param samplerate audio sampling rate The filter coefficients are built according to Malcolm Slaney's Auditory - Toolbox, available at http://cobweb.ecn.purdue.edu/~malcolm/interval/1998-010/ + Toolbox, available at http://engineering.purdue.edu/~malcolm/interval/1998-010/ (see file mfcc.m). */ diff --git a/src/spectral/mfcc.h b/src/spectral/mfcc.h index 6527d448..552d141f 100644 --- a/src/spectral/mfcc.h +++ b/src/spectral/mfcc.h @@ -21,7 +21,14 @@ /** \file - Mel-frequency cepstrum coefficients object + Mel-Frequency Cepstrum Coefficients object + + This object computes MFCC coefficients on an input cvec_t. + + The implementation follows the specifications established by Malcolm Slaney + in its Auditory Toolbox, available online (see file mfcc.m). + + http://engineering.ecn.purdue.edu/~malcolm/interval/1998-010/ \example spectral/test-mfcc.c diff --git a/src/spectral/phasevoc.h b/src/spectral/phasevoc.h index 9b1a3b5f..7d9e6afe 100644 --- a/src/spectral/phasevoc.h +++ b/src/spectral/phasevoc.h @@ -56,14 +56,14 @@ aubio_pvoc_t * new_aubio_pvoc (uint_t win_s, uint_t hop_s); void del_aubio_pvoc(aubio_pvoc_t *pv); /** compute spectral frame - + This function accepts an input vector of size [hop_s]. The analysis buffer is rotated and filled with the new data. After windowing of this signal window, the Fourier transform is computed and returned in fftgrain as two vectors, magnitude and phase. \param pv phase vocoder object as returned by new_aubio_pvoc - \param in new input signal (hop_s long) + \param in new input signal (hop_s long) \param fftgrain output spectral frame */ @@ -74,10 +74,10 @@ void aubio_pvoc_do(aubio_pvoc_t *pv, fvec_t *in, cvec_t * fftgrain); [buf_s] and computes its inverse Fourier transform. Overlap-add synthesis is then computed using the previously synthetised frames, and the output stored in out. - + \param pv phase vocoder object as returned by new_aubio_pvoc \param fftgrain input spectral frame - \param out output signal (hop_s long) + \param out output signal (hop_s long) */ void aubio_pvoc_rdo(aubio_pvoc_t *pv, cvec_t * fftgrain, fvec_t *out); @@ -97,6 +97,6 @@ uint_t aubio_pvoc_get_hop(aubio_pvoc_t* pv); #ifdef __cplusplus } -#endif +#endif #endif /* _AUBIO_PHASEVOC_H */ diff --git a/src/spectral/tss.h b/src/spectral/tss.h index 610e7d63..51b35346 100644 --- a/src/spectral/tss.h +++ b/src/spectral/tss.h @@ -62,7 +62,7 @@ aubio_tss_t *new_aubio_tss (uint_t buf_size, uint_t hop_size); void del_aubio_tss (aubio_tss_t * o); /** split input into transient and steady states components - + \param o tss object as returned by new_aubio_tss() \param input input spectral frame \param trans output transient components @@ -72,8 +72,8 @@ void del_aubio_tss (aubio_tss_t * o); void aubio_tss_do (aubio_tss_t * o, cvec_t * input, cvec_t * trans, cvec_t * stead); -/** set transient / steady state separation threshold - +/** set transient / steady state separation threshold + \param o tss object as returned by new_aubio_tss() \param thrs new threshold value @@ -81,7 +81,7 @@ void aubio_tss_do (aubio_tss_t * o, cvec_t * input, cvec_t * trans, uint_t aubio_tss_set_threshold (aubio_tss_t * o, smpl_t thrs); /** set parameter a, defaults to 3 - + \param o tss object as returned by new_aubio_tss() \param alpha new value for alpha parameter @@ -89,7 +89,7 @@ uint_t aubio_tss_set_threshold (aubio_tss_t * o, smpl_t thrs); uint_t aubio_tss_set_alpha (aubio_tss_t * o, smpl_t alpha); /** set parameter b, defaults to 3 - + \param o tss object as returned by new_aubio_tss() \param beta new value for beta parameter diff --git a/src/tempo/beattracking.c b/src/tempo/beattracking.c index 3bfa1de4..c6bf4844 100644 --- a/src/tempo/beattracking.c +++ b/src/tempo/beattracking.c @@ -135,8 +135,8 @@ aubio_beattracking_do (aubio_beattracking_t * bt, fvec_t * dfframe, //number of harmonics in shift invariant comb filterbank uint_t numelem = 4; - smpl_t phase; // beat alignment (step - lastbeat) - smpl_t beat; // beat position + smpl_t phase; // beat alignment (step - lastbeat) + smpl_t beat; // beat position smpl_t bp; // beat period uint_t a, b; // used to build shift invariant comb filterbank uint_t kmax; // number of elements used to find beat phase @@ -226,7 +226,7 @@ aubio_beattracking_do (aubio_beattracking_t * bt, fvec_t * dfframe, i = 1; beat = bp - phase; - // AUBIO_DBG ("bp: %f, phase: %f, lastbeat: %f, step: %d, winlen: %d\n", + // AUBIO_DBG ("bp: %f, phase: %f, lastbeat: %f, step: %d, winlen: %d\n", // bp, phase, bt->lastbeat, step, winlen); /* the next beat will be earlier than 60% of the tempo period @@ -318,7 +318,7 @@ aubio_beattracking_checkstate (aubio_beattracking_t * bt) gp = 0; } - //now look for step change - i.e. a difference between gp and rp that + //now look for step change - i.e. a difference between gp and rp that // is greater than 2*constthresh - always true in first case, since gp = 0 if (counter == 0) { if (ABS (gp - rp) > 2. * bt->g_var) { @@ -341,7 +341,7 @@ aubio_beattracking_checkstate (aubio_beattracking_t * bt) counter = 2; // let it look next time } } else if (counter > 0) { - //if counter doesn't = 1, + //if counter doesn't = 1, counter = counter - 1; } diff --git a/src/tempo/beattracking.h b/src/tempo/beattracking.h index 80470e95..c3b6a231 100644 --- a/src/tempo/beattracking.h +++ b/src/tempo/beattracking.h @@ -1,5 +1,5 @@ /* - Copyright (C) 2003-2013 Matthew Davies and Paul Brossier + Copyright (C) 2003-2015 Matthew Davies and Paul Brossier This file is part of aubio. @@ -34,7 +34,7 @@ Engeeniring Society 118th Convention, Barcelona, Spain, May 2005. \example tempo/test-beattracking.c - + */ #ifndef _AUBIO_BEATTRACKING_H #define _AUBIO_BEATTRACKING_H @@ -49,19 +49,19 @@ typedef struct _aubio_beattracking_t aubio_beattracking_t; /** create beat tracking object \param winlen length of the onset detection window - \param hop_size number of onset detection samples [512] + \param hop_size number of onset detection samples [512] \param samplerate samplerate of the input signal */ aubio_beattracking_t * new_aubio_beattracking(uint_t winlen, uint_t hop_size, uint_t samplerate); -/** track the beat +/** track the beat \param bt beat tracking object \param dfframes current input detection function frame, smoothed by - adaptive median threshold. - \param out stored detected beat locations + adaptive median threshold. + \param out stored detected beat locations */ void aubio_beattracking_do (aubio_beattracking_t * bt, fvec_t * dfframes, @@ -77,7 +77,7 @@ void aubio_beattracking_do (aubio_beattracking_t * bt, fvec_t * dfframes, */ smpl_t aubio_beattracking_get_bpm(aubio_beattracking_t * bt); -/** get current tempo confidence +/** get current tempo confidence \param bt beat tracking object 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 */ diff --git a/src/types.h b/src/types.h index 9a4c56ca..4200ad21 100644 --- a/src/types.h +++ b/src/types.h @@ -1,5 +1,5 @@ /* - Copyright (C) 2003-2013 Paul Brossier + Copyright (C) 2003-2015 Paul Brossier This file is part of aubio. @@ -22,9 +22,9 @@ #define _AUBIO__TYPES_H /** \file - + Definition of data types used in aubio - + */ #ifdef __cplusplus diff --git a/src/utils/parameter.h b/src/utils/parameter.h index 3b7e1443..ef296d08 100644 --- a/src/utils/parameter.h +++ b/src/utils/parameter.h @@ -60,7 +60,7 @@ aubio_parameter_t * new_aubio_parameter(smpl_t min_value, smpl_t max_value, uint */ uint_t aubio_parameter_set_target_value ( aubio_parameter_t * param, smpl_t value ); -/** get next parameter +/** get next parameter \param param parameter, created by ::new_aubio_parameter diff --git a/src/utils/scale.h b/src/utils/scale.h index b52b610c..abcf53b7 100644 --- a/src/utils/scale.h +++ b/src/utils/scale.h @@ -20,13 +20,13 @@ /** \file - Vector scaling function - + Vector scaling function + This object, inspired from the scale object in FTS, the jMax engine, scales the values of a vector according to an affine function defined as follow: - - \f$ y = (x - ilow)*(ohig-olow)/(ihig-ilow) + olow \f$ - + + \f$ y = (x - ilow)*(ohig-olow)/(ihig-ilow) + olow \f$ + */ #ifndef _AUBIO_SCALE_H #define _AUBIO_SCALE_H @@ -39,7 +39,7 @@ extern "C" { typedef struct _aubio_scale_t aubio_scale_t; /** create a scale object - + \param flow lower value of output function \param fhig higher value of output function \param ilow lower value of input function @@ -48,7 +48,7 @@ typedef struct _aubio_scale_t aubio_scale_t; */ aubio_scale_t * new_aubio_scale(smpl_t flow, smpl_t fhig, smpl_t ilow, smpl_t ihig); -/** delete a scale object +/** delete a scale object \param s scale object as returned by new_aubio_scale @@ -75,6 +75,6 @@ uint_t aubio_scale_set_limits (aubio_scale_t *s, smpl_t ilow, smpl_t ihig, #ifdef __cplusplus } -#endif +#endif #endif /* _AUBIO_SCALE_H */ diff --git a/src/vecutils.h b/src/vecutils.h index 7e698492..d2cb119b 100644 --- a/src/vecutils.h +++ b/src/vecutils.h @@ -1,5 +1,5 @@ /* - Copyright (C) 2009-2013 Paul Brossier + Copyright (C) 2009-2015 Paul Brossier This file is part of aubio. diff --git a/tests/src/spectral/test-fft.c b/tests/src/spectral/test-fft.c index beef5ca3..72db5302 100644 --- a/tests/src/spectral/test-fft.c +++ b/tests/src/spectral/test-fft.c @@ -2,6 +2,7 @@ int main (void) { + int return_code = 0; uint_t i, n_iters = 100; // number of iterations uint_t win_s = 500; // window size fvec_t * in = new_fvec (win_s); // input buffer @@ -10,6 +11,11 @@ int main (void) // create fft object aubio_fft_t * fft = new_aubio_fft(win_s); + if (!fft) { + return_code = 1; + goto beach; + } + // fill input with some data in->data[0] = 1; in->data[1] = 2; @@ -33,9 +39,10 @@ int main (void) // cleam up //fvec_print(out); del_aubio_fft(fft); +beach: del_fvec(in); del_cvec(fftgrain); del_fvec(out); aubio_cleanup(); - return 0; + return return_code; } diff --git a/tests/src/test-delnull.c b/tests/src/test-delnull.c index 6a0afc41..bb245090 100644 --- a/tests/src/test-delnull.c +++ b/tests/src/test-delnull.c @@ -1,13 +1,24 @@ #include #include "aubio.h" -// Because aubio does not check for double free, this program will crash. -// Programs that call these functions should check for null pointers. +// When creating an aubio object, the user should check whether the object is +// set NULL, indicating the creation failed and the object was not allocated. int main (void) { - del_fvec(NULL); - del_lvec(NULL); - del_cvec(NULL); - return 0; + uint_t return_code = 0; + fvec_t *f = new_fvec(-12); + cvec_t *c = new_cvec(-12); + lvec_t *l = new_lvec(-12); + aubio_fft_t *fft = new_aubio_fft(-12); + if (f != NULL) { + return_code = 1; + } else if (c != NULL) { + return_code = 2; + } else if (l != NULL) { + return_code = 3; + } else if (fft != NULL) { + return_code = 3; + } + return return_code; } diff --git a/wscript b/wscript index 3157f13d..949fe7cc 100644 --- a/wscript +++ b/wscript @@ -75,6 +75,12 @@ def options(ctx): add_option_enable_disable(ctx, 'fat', default = False, help_str = 'build fat binaries (darwin only)', help_disable_str = 'do not build fat binaries (default)') + add_option_enable_disable(ctx, 'accelerate', default = None, + help_str = 'use Accelerate framework (darwin only) (auto)', + help_disable_str = 'do not use Accelerate framework') + add_option_enable_disable(ctx, 'apple-audio', default = None, + help_str = 'use CoreFoundation (darwin only) (auto)', + help_disable_str = 'do not use CoreFoundation framework') ctx.add_option('--with-target-platform', type='string', help='set target platform for cross-compilation', dest='target_platform') @@ -110,15 +116,20 @@ def configure(ctx): ctx.env.LINKFLAGS += ['-arch', 'i386', '-arch', 'x86_64'] if target_platform in [ 'darwin', 'ios', 'iosimulator']: - ctx.env.FRAMEWORK = ['CoreFoundation', 'AudioToolbox', 'Accelerate'] - ctx.define('HAVE_SOURCE_APPLE_AUDIO', 1) - ctx.define('HAVE_SINK_APPLE_AUDIO', 1) - ctx.define('HAVE_ACCELERATE', 1) + if (ctx.options.enable_apple_audio != False): + ctx.env.FRAMEWORK += ['CoreFoundation', 'AudioToolbox'] + ctx.define('HAVE_SOURCE_APPLE_AUDIO', 1) + ctx.define('HAVE_SINK_APPLE_AUDIO', 1) + if (ctx.options.enable_accelerate != False): + ctx.define('HAVE_ACCELERATE', 1) + ctx.env.FRAMEWORK += ['Accelerate'] if target_platform in [ 'ios', 'iosimulator' ]: - ctx.define('TARGET_OS_IPHONE', 1) MINSDKVER="6.1" ctx.env.CFLAGS += ['-std=c99'] + if (ctx.options.enable_audio_unit != False): + ctx.define('HAVE_AUDIO_UNIT', 1) + #ctx.env.FRAMEWORK += ['CoreFoundation', 'AudioToolbox'] if target_platform == 'ios': DEVROOT = "/Applications/Xcode.app/Contents" DEVROOT += "/Developer/Platforms/iPhoneOS.platform/Developer" @@ -238,7 +249,7 @@ def configure(ctx): ctx.check_cfg(package = 'libavresample', atleast_version = '1.0.1', args = '--cflags --libs', uselib_store = 'AVRESAMPLE', mandatory = ctx.options.enable_avcodec) - if all ( 'HAVE_' + i in ctx.env.define_key + if all ( 'HAVE_' + i in ctx.env for i in ['AVCODEC', 'AVFORMAT', 'AVUTIL', 'AVRESAMPLE'] ): ctx.define('HAVE_LIBAV', 1) ctx.msg('Checking for all libav libraries', 'yes')