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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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1169 lines
36 KiB
C
1169 lines
36 KiB
C
/* GStreamer
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* Copyright (C) <2015> Wim Taymans <wim.taymans@gmail.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include <string.h>
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#include <stdio.h>
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#include <math.h>
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#ifdef HAVE_ORC
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#include <orc/orc.h>
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#endif
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#include "audio-resampler.h"
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typedef struct _Tap
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{
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gpointer taps;
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gint sample_inc;
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gint next_phase;
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gint size;
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} Tap;
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typedef void (*MakeTapsFunc) (GstAudioResampler * resampler, Tap * t, gint j);
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typedef void (*ResampleFunc) (GstAudioResampler * resampler, gpointer in[],
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gsize in_len, gpointer out[], gsize out_len, gsize * consumed);
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typedef void (*DeinterleaveFunc) (GstAudioResampler * resampler,
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gpointer * sbuf, gpointer in[], gsize in_frames);
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#define MEM_ALIGN(m,a) ((gint8 *)((guintptr)((gint8 *)(m) + ((a)-1)) & ~((a)-1)))
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#define ALIGN 16
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#define TAPS_OVERREAD 16
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struct _GstAudioResampler
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{
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GstAudioResamplerMethod method;
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GstAudioResamplerFlags flags;
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GstAudioFormat format;
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GstStructure *options;
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guint channels;
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gint in_rate;
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gint out_rate;
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gint bps, bpf;
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gint ostride;
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gdouble cutoff;
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gdouble kaiser_beta;
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/* for cubic */
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gdouble b, c;
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guint n_taps;
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Tap *taps;
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gpointer coeff;
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gpointer coeffmem;
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gsize cstride;
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gpointer tmpcoeff;
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DeinterleaveFunc deinterleave;
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ResampleFunc resample;
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guint blocks;
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guint inc;
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gint samp_inc;
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gint samp_frac;
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gint samp_index;
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gint samp_phase;
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gint skip;
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gpointer samples;
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gsize samples_len;
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gsize samples_avail;
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gpointer *sbuf;
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};
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GST_DEBUG_CATEGORY_STATIC (audio_resampler_debug);
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#define GST_CAT_DEFAULT audio_resampler_debug
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/**
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* SECTION:gstaudioresampler
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* @short_description: Utility structure for resampler information
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*
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* #GstAudioResampler is a structure which holds the information
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* required to perform various kinds of resampling filtering.
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*
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*/
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typedef struct
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{
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gdouble cutoff;
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gdouble downsample_cutoff_factor;
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gdouble stopband_attenuation;
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gdouble transition_bandwidth;
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} KaiserQualityMap;
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static const KaiserQualityMap kaiser_qualities[] = {
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{0.860, 0.96511, 60, 0.7}, /* 8 taps */
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{0.880, 0.96591, 65, 0.29}, /* 16 taps */
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{0.910, 0.96923, 70, 0.145}, /* 32 taps */
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{0.920, 0.97600, 80, 0.105}, /* 48 taps */
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{0.940, 0.97979, 85, 0.087}, /* 64 taps default quality */
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{0.940, 0.98085, 95, 0.077}, /* 80 taps */
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{0.945, 0.99471, 100, 0.068}, /* 96 taps */
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{0.950, 1.0, 105, 0.055}, /* 128 taps */
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{0.960, 1.0, 110, 0.045}, /* 160 taps */
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{0.968, 1.0, 115, 0.039}, /* 192 taps */
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{0.975, 1.0, 120, 0.0305} /* 256 taps */
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};
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typedef struct
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{
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guint n_taps;
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gdouble cutoff;
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} BlackmanQualityMap;
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static const BlackmanQualityMap blackman_qualities[] = {
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{8, 0.5,},
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{16, 0.6,},
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{24, 0.72,},
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{32, 0.8,},
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{48, 0.85,}, /* default */
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{64, 0.90,},
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{80, 0.92,},
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{96, 0.933,},
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{128, 0.950,},
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{148, 0.955,},
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{160, 0.960,}
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};
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#define DEFAULT_QUALITY GST_AUDIO_RESAMPLER_QUALITY_DEFAULT
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#define DEFAULT_OPT_CUBIC_B 1.0
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#define DEFAULT_OPT_CUBIC_C 0.0
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static gdouble
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get_opt_double (GstStructure * options, const gchar * name, gdouble def)
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{
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gdouble res;
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if (!options || !gst_structure_get_double (options, name, &res))
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res = def;
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return res;
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}
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static gint
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get_opt_int (GstStructure * options, const gchar * name, gint def)
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{
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gint res;
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if (!options || !gst_structure_get_int (options, name, &res))
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res = def;
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return res;
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}
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#define GET_OPT_CUTOFF(options,def) get_opt_double(options, \
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GST_AUDIO_RESAMPLER_OPT_CUTOFF,def)
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#define GET_OPT_DOWN_CUTOFF_FACTOR(options,def) get_opt_double(options, \
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GST_AUDIO_RESAMPLER_OPT_DOWN_CUTOFF_FACTOR, def)
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#define GET_OPT_STOP_ATTENUATION(options,def) get_opt_double(options, \
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GST_AUDIO_RESAMPLER_OPT_STOP_ATTENUATION, def)
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#define GET_OPT_TRANSITION_BANDWIDTH(options,def) get_opt_double(options, \
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GST_AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH, def)
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#define GET_OPT_CUBIC_B(options) get_opt_double(options, \
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GST_AUDIO_RESAMPLER_OPT_CUBIC_B, DEFAULT_OPT_CUBIC_B)
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#define GET_OPT_CUBIC_C(options) get_opt_double(options, \
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GST_AUDIO_RESAMPLER_OPT_CUBIC_C, DEFAULT_OPT_CUBIC_C)
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#define GET_OPT_N_TAPS(options,def) get_opt_int(options, \
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GST_AUDIO_RESAMPLER_OPT_N_TAPS, def)
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#include "dbesi0.c"
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#define bessel dbesi0
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static inline gdouble
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get_nearest_tap (GstAudioResampler * resampler, gdouble x)
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{
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gdouble a = fabs (x);
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if (a < 0.5)
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return 1.0;
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else
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return 0.0;
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}
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static inline gdouble
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get_linear_tap (GstAudioResampler * resampler, gdouble x)
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{
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gdouble a;
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a = fabs (x) / resampler->n_taps;
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if (a < 1.0)
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return 1.0 - a;
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else
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return 0.0;
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}
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static inline gdouble
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get_cubic_tap (GstAudioResampler * resampler, gdouble x)
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{
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gdouble a, a2, a3, b, c;
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a = fabs (x * 4.0) / resampler->n_taps;
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a2 = a * a;
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a3 = a2 * a;
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b = resampler->b;
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c = resampler->c;
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if (a <= 1.0)
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return ((12.0 - 9.0 * b - 6.0 * c) * a3 +
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(-18.0 + 12.0 * b + 6.0 * c) * a2 + (6.0 - 2.0 * b)) / 6.0;
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else if (a <= 2.0)
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return ((-b - 6.0 * c) * a3 +
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(6.0 * b + 30.0 * c) * a2 +
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(-12.0 * b - 48.0 * c) * a + (8.0 * b + 24.0 * c)) / 6.0;
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else
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return 0.0;
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}
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static inline gdouble
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get_blackman_nuttall_tap (GstAudioResampler * resampler, gdouble x)
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{
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gdouble s, y, w, Fc = resampler->cutoff;
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y = G_PI * x;
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s = (y == 0.0 ? Fc : sin (y * Fc) / y);
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w = 2.0 * y / resampler->n_taps + G_PI;
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return s * (0.3635819 - 0.4891775 * cos (w) + 0.1365995 * cos (2 * w) -
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0.0106411 * cos (3 * w));
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}
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static inline gdouble
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get_kaiser_tap (GstAudioResampler * resampler, gdouble x)
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{
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gdouble s, y, w, Fc = resampler->cutoff;
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y = G_PI * x;
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s = (y == 0.0 ? Fc : sin (y * Fc) / y);
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w = 2.0 * x / resampler->n_taps;
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return s * bessel (resampler->kaiser_beta * sqrt (MAX (1 - w * w, 0)));
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}
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#define CONVERT_TAPS(type, precision) \
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G_STMT_START { \
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type *taps = t->taps = (type *) ((gint8*)resampler->coeff + j * resampler->cstride); \
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gdouble multiplier = (1 << precision); \
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gint i, j; \
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gdouble offset, l_offset, h_offset; \
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gboolean exact = FALSE; \
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/* Round to integer, but with an adjustable bias that we use to */ \
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/* eliminate the DC error. */ \
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l_offset = 0.0; \
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h_offset = 1.0; \
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offset = 0.5; \
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for (i = 0; i < 32; i++) { \
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gint64 sum = 0; \
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for (j = 0; j < n_taps; j++) \
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sum += taps[j] = floor (offset + tmpcoeff[j] * multiplier / weight); \
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if (sum == (1 << precision)) { \
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exact = TRUE; \
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break; \
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} \
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if (l_offset == h_offset) \
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break; \
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if (sum < (1 << precision)) { \
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if (offset > l_offset) \
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l_offset = offset; \
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offset += (h_offset - l_offset) / 2; \
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} else { \
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if (offset < h_offset) \
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h_offset = offset; \
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offset -= (h_offset - l_offset) / 2; \
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} \
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} \
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if (!exact) \
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GST_WARNING ("can't find exact taps"); \
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} G_STMT_END
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#define PRECISION_S16 15
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#define PRECISION_S32 30
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static void
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make_taps (GstAudioResampler * resampler, Tap * t, gint j)
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{
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gint n_taps = resampler->n_taps;
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gdouble x, weight = 0.0;
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gdouble *tmpcoeff = resampler->tmpcoeff;
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gint tap_offs = n_taps / 2;
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gint out_rate = resampler->out_rate;
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gint l;
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x = ((double) (1.0 - tap_offs) - (double) j / out_rate);
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switch (resampler->method) {
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case GST_AUDIO_RESAMPLER_METHOD_NEAREST:
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for (l = 0; l < n_taps; l++, x += 1.0)
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weight += tmpcoeff[l] = get_nearest_tap (resampler, x);
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break;
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case GST_AUDIO_RESAMPLER_METHOD_LINEAR:
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for (l = 0; l < n_taps; l++, x += 1.0)
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weight += tmpcoeff[l] = get_linear_tap (resampler, x);
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break;
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case GST_AUDIO_RESAMPLER_METHOD_CUBIC:
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for (l = 0; l < n_taps; l++, x += 1.0)
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weight += tmpcoeff[l] = get_cubic_tap (resampler, x);
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break;
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case GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL:
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for (l = 0; l < n_taps; l++, x += 1.0)
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weight += tmpcoeff[l] = get_blackman_nuttall_tap (resampler, x);
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break;
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case GST_AUDIO_RESAMPLER_METHOD_KAISER:
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for (l = 0; l < n_taps; l++, x += 1.0)
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weight += tmpcoeff[l] = get_kaiser_tap (resampler, x);
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break;
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default:
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break;
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}
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switch (resampler->format) {
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case GST_AUDIO_FORMAT_F64:
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{
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gdouble *taps = t->taps =
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(gdouble *) ((gint8 *) resampler->coeff + j * resampler->cstride);
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for (l = 0; l < n_taps; l++)
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taps[l] = tmpcoeff[l] / weight;
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break;
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}
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case GST_AUDIO_FORMAT_F32:
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{
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gfloat *taps = t->taps =
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(gfloat *) ((gint8 *) resampler->coeff + j * resampler->cstride);
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for (l = 0; l < n_taps; l++)
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taps[l] = tmpcoeff[l] / weight;
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break;
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}
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case GST_AUDIO_FORMAT_S32:
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CONVERT_TAPS (gint32, PRECISION_S32);
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break;
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case GST_AUDIO_FORMAT_S16:
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CONVERT_TAPS (gint16, PRECISION_S16);
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break;
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default:
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break;
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}
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}
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static inline void
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inner_product_gint16_1_c (gint16 * o, const gint16 * a, const gint16 * b,
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gint len)
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{
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gint i;
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gint32 res = 0;
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for (i = 0; i < len; i++)
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res += (gint32) a[i] * (gint32) b[i];
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res = (res + (1 << (PRECISION_S16 - 1))) >> PRECISION_S16;
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*o = CLAMP (res, -(1L << 15), (1L << 15) - 1);
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}
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static inline void
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inner_product_gint32_1_c (gint32 * o, const gint32 * a, const gint32 * b,
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gint len)
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{
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gint i;
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gint64 res = 0;
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for (i = 0; i < len; i++)
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res += (gint64) a[i] * (gint64) b[i];
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res = (res + (1 << (PRECISION_S32 - 1))) >> PRECISION_S32;
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*o = CLAMP (res, -(1L << 31), (1L << 31) - 1);
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}
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static inline void
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inner_product_gfloat_1_c (gfloat * o, const gfloat * a, const gfloat * b,
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gint len)
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{
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gint i;
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gfloat res = 0.0;
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for (i = 0; i < len; i++)
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res += a[i] * b[i];
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*o = res;
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}
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static inline void
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inner_product_gdouble_1_c (gdouble * o, const gdouble * a, const gdouble * b,
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gint len)
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{
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gint i;
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gdouble res = 0.0;
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for (i = 0; i < len; i++)
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res += a[i] * b[i];
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*o = res;
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}
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#define MAKE_RESAMPLE_FUNC(type,channels,arch) \
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static void \
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resample_ ##type## _ ##channels## _ ##arch (GstAudioResampler * resampler, \
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gpointer in[], gsize in_len, gpointer out[], gsize out_len, \
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gsize * consumed) \
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{ \
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gint c, di = 0; \
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gint n_taps = resampler->n_taps; \
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gint blocks = resampler->blocks; \
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gint ostride = resampler->ostride; \
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gint samp_index = 0; \
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gint samp_phase = 0; \
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\
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for (c = 0; c < blocks; c++) { \
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type *ip = in[c]; \
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type *op = ostride == 1 ? out[c] : (type *)out[0] + c; \
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\
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samp_index = resampler->samp_index; \
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samp_phase = resampler->samp_phase; \
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\
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for (di = 0; di < out_len; di++) { \
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Tap *t = &resampler->taps[samp_phase]; \
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type *ipp = &ip[samp_index * channels]; \
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\
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if (G_UNLIKELY (t->taps == NULL)) \
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make_taps (resampler, t, samp_phase); \
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\
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inner_product_ ##type## _##channels##_##arch (op, ipp, t->taps, n_taps); \
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op += ostride; \
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\
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samp_phase = t->next_phase; \
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samp_index += t->sample_inc; \
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} \
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memmove (ip, &ip[samp_index * channels], \
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(in_len - samp_index) * sizeof(type) * channels); \
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} \
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*consumed = samp_index - resampler->samp_index; \
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\
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resampler->samp_index = 0; \
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resampler->samp_phase = samp_phase; \
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}
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MAKE_RESAMPLE_FUNC (gint16, 1, c);
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MAKE_RESAMPLE_FUNC (gint32, 1, c);
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MAKE_RESAMPLE_FUNC (gfloat, 1, c);
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MAKE_RESAMPLE_FUNC (gdouble, 1, c);
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static ResampleFunc resample_funcs[] = {
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resample_gint16_1_c,
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resample_gint32_1_c,
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resample_gfloat_1_c,
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resample_gdouble_1_c,
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NULL,
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NULL,
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NULL,
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NULL,
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};
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#define resample_gint16_1 resample_funcs[0]
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#define resample_gint32_1 resample_funcs[1]
|
|
#define resample_gfloat_1 resample_funcs[2]
|
|
#define resample_gdouble_1 resample_funcs[3]
|
|
#define resample_gint16_2 resample_funcs[4]
|
|
#define resample_gint32_2 resample_funcs[5]
|
|
#define resample_gfloat_2 resample_funcs[6]
|
|
#define resample_gdouble_2 resample_funcs[7]
|
|
|
|
#if defined HAVE_ORC && !defined DISABLE_ORC
|
|
# if defined (__i386__) || defined (__x86_64__)
|
|
# define CHECK_X86
|
|
# include "audio-resampler-x86.h"
|
|
# endif
|
|
#endif
|
|
|
|
static void
|
|
audio_resampler_init (void)
|
|
{
|
|
static gsize init_gonce = 0;
|
|
|
|
if (g_once_init_enter (&init_gonce)) {
|
|
|
|
GST_DEBUG_CATEGORY_INIT (audio_resampler_debug, "audio-resampler", 0,
|
|
"audio-resampler object");
|
|
|
|
#if defined HAVE_ORC && !defined DISABLE_ORC
|
|
orc_init ();
|
|
{
|
|
OrcTarget *target = orc_target_get_default ();
|
|
gint i;
|
|
|
|
if (target) {
|
|
unsigned int flags = orc_target_get_default_flags (target);
|
|
const gchar *name;
|
|
|
|
name = orc_target_get_name (target);
|
|
GST_DEBUG ("target %s, default flags %08x", name, flags);
|
|
|
|
for (i = 0; i < 32; ++i) {
|
|
if (flags & (1U << i)) {
|
|
name = orc_target_get_flag_name (target, i);
|
|
GST_DEBUG ("target flag %s", name);
|
|
#ifdef CHECK_X86
|
|
audio_resampler_check_x86 (name);
|
|
#endif
|
|
}
|
|
}
|
|
}
|
|
}
|
|
#endif
|
|
g_once_init_leave (&init_gonce, 1);
|
|
}
|
|
}
|
|
|
|
#define MAKE_DEINTERLEAVE_FUNC(type) \
|
|
static void \
|
|
deinterleave_ ##type (GstAudioResampler * resampler, gpointer sbuf[], \
|
|
gpointer in[], gsize in_frames) \
|
|
{ \
|
|
guint i, c, channels = resampler->channels; \
|
|
gsize samples_avail = resampler->samples_avail; \
|
|
for (c = 0; c < channels; c++) { \
|
|
type *s = (type *) sbuf[c] + samples_avail; \
|
|
if (G_UNLIKELY (in == NULL)) { \
|
|
for (i = 0; i < in_frames; i++) \
|
|
s[i] = 0; \
|
|
} else { \
|
|
type *ip = (type *) in[0] + c; \
|
|
for (i = 0; i < in_frames; i++, ip += channels) \
|
|
s[i] = *ip; \
|
|
} \
|
|
} \
|
|
}
|
|
|
|
MAKE_DEINTERLEAVE_FUNC (gdouble);
|
|
MAKE_DEINTERLEAVE_FUNC (gfloat);
|
|
MAKE_DEINTERLEAVE_FUNC (gint32);
|
|
MAKE_DEINTERLEAVE_FUNC (gint16);
|
|
|
|
static void
|
|
deinterleave_copy (GstAudioResampler * resampler, gpointer sbuf[],
|
|
gpointer in[], gsize in_frames)
|
|
{
|
|
guint c, blocks = resampler->blocks;
|
|
gsize bytes_avail, in_bytes, bpf;
|
|
|
|
bpf = resampler->bps * resampler->inc;
|
|
bytes_avail = resampler->samples_avail * bpf;
|
|
in_bytes = in_frames * bpf;
|
|
|
|
for (c = 0; c < blocks; c++) {
|
|
if (G_UNLIKELY (in == NULL))
|
|
memset ((guint8 *) sbuf[c] + bytes_avail, 0, in_bytes);
|
|
else
|
|
memcpy ((guint8 *) sbuf[c] + bytes_avail, in[c], in_bytes);
|
|
}
|
|
}
|
|
|
|
static void
|
|
calculate_kaiser_params (GstAudioResampler * resampler)
|
|
{
|
|
gdouble A, B, dw, tr_bw, Fc;
|
|
gint n;
|
|
const KaiserQualityMap *q = &kaiser_qualities[DEFAULT_QUALITY];
|
|
|
|
/* default cutoff */
|
|
Fc = q->cutoff;
|
|
if (resampler->out_rate < resampler->in_rate)
|
|
Fc *= q->downsample_cutoff_factor;
|
|
|
|
Fc = GET_OPT_CUTOFF (resampler->options, Fc);
|
|
A = GET_OPT_STOP_ATTENUATION (resampler->options, q->stopband_attenuation);
|
|
tr_bw =
|
|
GET_OPT_TRANSITION_BANDWIDTH (resampler->options,
|
|
q->transition_bandwidth);
|
|
|
|
GST_LOG ("Fc %f, A %f, tr_bw %f", Fc, A, tr_bw);
|
|
|
|
/* calculate Beta */
|
|
if (A > 50)
|
|
B = 0.1102 * (A - 8.7);
|
|
else if (A >= 21)
|
|
B = 0.5842 * pow (A - 21, 0.4) + 0.07886 * (A - 21);
|
|
else
|
|
B = 0.0;
|
|
/* calculate transition width in radians */
|
|
dw = 2 * G_PI * (tr_bw);
|
|
/* order of the filter */
|
|
n = (A - 8.0) / (2.285 * dw);
|
|
|
|
resampler->kaiser_beta = B;
|
|
resampler->n_taps = n + 1;
|
|
resampler->cutoff = Fc;
|
|
|
|
GST_LOG ("using Beta %f n_taps %d cutoff %f", resampler->kaiser_beta,
|
|
resampler->n_taps, resampler->cutoff);
|
|
}
|
|
|
|
static void
|
|
resampler_calculate_taps (GstAudioResampler * resampler)
|
|
{
|
|
gint bps;
|
|
gint j;
|
|
gint n_taps;
|
|
gint out_rate;
|
|
gint in_rate;
|
|
gboolean non_interleaved;
|
|
DeinterleaveFunc deinterleave;
|
|
ResampleFunc resample, resample_2;
|
|
|
|
|
|
switch (resampler->method) {
|
|
case GST_AUDIO_RESAMPLER_METHOD_NEAREST:
|
|
resampler->n_taps = 2;
|
|
break;
|
|
case GST_AUDIO_RESAMPLER_METHOD_LINEAR:
|
|
resampler->n_taps = GET_OPT_N_TAPS (resampler->options, 2);
|
|
break;
|
|
case GST_AUDIO_RESAMPLER_METHOD_CUBIC:
|
|
resampler->n_taps = GET_OPT_N_TAPS (resampler->options, 4);
|
|
resampler->b = GET_OPT_CUBIC_B (resampler->options);
|
|
resampler->c = GET_OPT_CUBIC_C (resampler->options);;
|
|
break;
|
|
case GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL:
|
|
{
|
|
const BlackmanQualityMap *q = &blackman_qualities[DEFAULT_QUALITY];
|
|
resampler->n_taps = GET_OPT_N_TAPS (resampler->options, q->n_taps);
|
|
resampler->cutoff = GET_OPT_CUTOFF (resampler->options, q->cutoff);
|
|
break;
|
|
}
|
|
case GST_AUDIO_RESAMPLER_METHOD_KAISER:
|
|
calculate_kaiser_params (resampler);
|
|
break;
|
|
}
|
|
|
|
in_rate = resampler->in_rate;
|
|
out_rate = resampler->out_rate;
|
|
|
|
if (out_rate < in_rate) {
|
|
resampler->cutoff = resampler->cutoff * out_rate / in_rate;
|
|
resampler->n_taps = resampler->n_taps * in_rate / out_rate;
|
|
}
|
|
/* only round up for bigger taps, the small taps are used for nearest,
|
|
* linear and cubic and we want to use less taps for those. */
|
|
if (resampler->n_taps > 4)
|
|
resampler->n_taps = GST_ROUND_UP_8 (resampler->n_taps);
|
|
|
|
n_taps = resampler->n_taps;
|
|
bps = resampler->bps;
|
|
|
|
GST_LOG ("using n_taps %d cutoff %f", n_taps, resampler->cutoff);
|
|
|
|
resampler->taps = g_realloc_n (resampler->taps, out_rate, sizeof (Tap));
|
|
|
|
resampler->cstride = GST_ROUND_UP_32 (bps * (n_taps + TAPS_OVERREAD));
|
|
g_free (resampler->coeffmem);
|
|
resampler->coeffmem = g_malloc0 (out_rate * resampler->cstride + ALIGN - 1);
|
|
resampler->coeff = MEM_ALIGN (resampler->coeffmem, ALIGN);
|
|
|
|
resampler->tmpcoeff =
|
|
g_realloc_n (resampler->tmpcoeff, n_taps, sizeof (gdouble));
|
|
|
|
resampler->samp_inc = in_rate / out_rate;
|
|
resampler->samp_frac = in_rate % out_rate;
|
|
|
|
for (j = 0; j < out_rate; j++) {
|
|
Tap *t = &resampler->taps[j];
|
|
t->taps = NULL;
|
|
t->sample_inc = (j + in_rate) / out_rate;
|
|
t->next_phase = (j + in_rate) % out_rate;
|
|
}
|
|
|
|
non_interleaved =
|
|
(resampler->flags & GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED);
|
|
|
|
resampler->ostride = non_interleaved ? 1 : resampler->channels;
|
|
|
|
/* we resample each channel separately */
|
|
resampler->blocks = resampler->channels;
|
|
resampler->inc = 1;
|
|
|
|
switch (resampler->format) {
|
|
case GST_AUDIO_FORMAT_S16:
|
|
resample = resample_gint16_1;
|
|
resample_2 = resample_gint16_2;
|
|
deinterleave = deinterleave_gint16;
|
|
break;
|
|
case GST_AUDIO_FORMAT_S32:
|
|
resample = resample_gint32_1;
|
|
resample_2 = resample_gint32_2;
|
|
deinterleave = deinterleave_gint32;
|
|
break;
|
|
case GST_AUDIO_FORMAT_F32:
|
|
resample = resample_gfloat_1;
|
|
resample_2 = resample_gfloat_2;
|
|
deinterleave = deinterleave_gfloat;
|
|
break;
|
|
case GST_AUDIO_FORMAT_F64:
|
|
resample = resample_gdouble_1;
|
|
resample_2 = resample_gdouble_2;
|
|
deinterleave = deinterleave_gdouble;
|
|
break;
|
|
default:
|
|
g_assert_not_reached ();
|
|
break;
|
|
}
|
|
if (!non_interleaved && resampler->channels == 2 && n_taps >= 4 && resample_2) {
|
|
resampler->resample = resample_2;
|
|
resampler->deinterleave = deinterleave_copy;
|
|
resampler->blocks = 1;
|
|
resampler->inc = resampler->channels;;
|
|
} else {
|
|
resampler->resample = resample;
|
|
resampler->deinterleave = deinterleave;
|
|
}
|
|
}
|
|
|
|
#define PRINT_TAPS(type,print) \
|
|
G_STMT_START { \
|
|
type sum = 0.0, *taps; \
|
|
\
|
|
if (t->taps == NULL) \
|
|
make_taps (resampler, t, i); \
|
|
\
|
|
taps = t->taps; \
|
|
for (j = 0; j < n_taps; j++) { \
|
|
type tap = taps[j]; \
|
|
fprintf (stderr, "\t%" print " ", tap); \
|
|
sum += tap; \
|
|
} \
|
|
fprintf (stderr, "\t: sum %" print "\n", sum);\
|
|
} G_STMT_END
|
|
|
|
static void
|
|
resampler_dump (GstAudioResampler * resampler)
|
|
{
|
|
#if 0
|
|
gint i, n_taps, out_rate;
|
|
gint64 a;
|
|
|
|
out_rate = resampler->out_rate;
|
|
n_taps = resampler->n_taps;
|
|
|
|
fprintf (stderr, "out size %d, max taps %d\n", out_rate, n_taps);
|
|
|
|
a = g_get_monotonic_time ();
|
|
|
|
for (i = 0; i < out_rate; i++) {
|
|
gint j;
|
|
Tap *t = &resampler->taps[i];
|
|
|
|
fprintf (stderr, "%u: %d %d\t ", i, t->sample_inc, t->next_phase);
|
|
switch (resampler->format) {
|
|
case GST_AUDIO_FORMAT_F64:
|
|
PRINT_TAPS (gdouble, "f");
|
|
break;
|
|
case GST_AUDIO_FORMAT_F32:
|
|
PRINT_TAPS (gfloat, "f");
|
|
break;
|
|
case GST_AUDIO_FORMAT_S32:
|
|
PRINT_TAPS (gint32, "d");
|
|
break;
|
|
case GST_AUDIO_FORMAT_S16:
|
|
PRINT_TAPS (gint16, "d");
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
fprintf (stderr, "time %" G_GUINT64_FORMAT "\n", g_get_monotonic_time () - a);
|
|
#endif
|
|
}
|
|
|
|
/**
|
|
* gst_audio_resampler_options_set_quality:
|
|
* @method: a #GstAudioResamplerMethod
|
|
* @quality: the quality
|
|
* @in_rate: the input rate
|
|
* @out_rate: the output rate
|
|
* @options: a #GstStructure
|
|
*
|
|
* Set the parameters for resampling from @in_rate to @out_rate using @method
|
|
* for @quality in @options.
|
|
*/
|
|
void
|
|
gst_audio_resampler_options_set_quality (GstAudioResamplerMethod method,
|
|
guint quality, guint in_rate, guint out_rate, GstStructure * options)
|
|
{
|
|
g_return_if_fail (options != NULL);
|
|
g_return_if_fail (quality < 11);
|
|
g_return_if_fail (in_rate != 0 && out_rate != 0);
|
|
|
|
switch (method) {
|
|
case GST_AUDIO_RESAMPLER_METHOD_NEAREST:
|
|
break;
|
|
case GST_AUDIO_RESAMPLER_METHOD_LINEAR:
|
|
gst_structure_set (options,
|
|
GST_AUDIO_RESAMPLER_OPT_N_TAPS, G_TYPE_INT, 2, NULL);
|
|
break;
|
|
case GST_AUDIO_RESAMPLER_METHOD_CUBIC:
|
|
gst_structure_set (options,
|
|
GST_AUDIO_RESAMPLER_OPT_N_TAPS, G_TYPE_INT, 4,
|
|
GST_AUDIO_RESAMPLER_OPT_CUBIC_B, G_TYPE_DOUBLE, DEFAULT_OPT_CUBIC_B,
|
|
GST_AUDIO_RESAMPLER_OPT_CUBIC_C, G_TYPE_DOUBLE, DEFAULT_OPT_CUBIC_C,
|
|
NULL);
|
|
break;
|
|
case GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL:
|
|
{
|
|
const BlackmanQualityMap *map = &blackman_qualities[quality];
|
|
gst_structure_set (options,
|
|
GST_AUDIO_RESAMPLER_OPT_N_TAPS, G_TYPE_INT, map->n_taps,
|
|
GST_AUDIO_RESAMPLER_OPT_CUTOFF, G_TYPE_DOUBLE, map->cutoff, NULL);
|
|
break;
|
|
}
|
|
case GST_AUDIO_RESAMPLER_METHOD_KAISER:
|
|
{
|
|
const KaiserQualityMap *map = &kaiser_qualities[quality];
|
|
gdouble cutoff;
|
|
|
|
cutoff = map->cutoff;
|
|
if (out_rate < in_rate)
|
|
cutoff *= map->downsample_cutoff_factor;
|
|
|
|
gst_structure_set (options,
|
|
GST_AUDIO_RESAMPLER_OPT_CUTOFF, G_TYPE_DOUBLE, cutoff,
|
|
GST_AUDIO_RESAMPLER_OPT_STOP_ATTENUATION, G_TYPE_DOUBLE,
|
|
map->stopband_attenuation,
|
|
GST_AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH, G_TYPE_DOUBLE,
|
|
map->transition_bandwidth, NULL);
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_audio_resampler_new:
|
|
* @resampler: a #GstAudioResampler
|
|
* @method: a #GstAudioResamplerMethod
|
|
* @flags: #GstAudioResamplerFlags
|
|
* @in_rate: input rate
|
|
* @out_rate: output rate
|
|
* @options: extra options
|
|
*
|
|
* Make a new resampler.
|
|
*
|
|
* Returns: %TRUE on success
|
|
*/
|
|
GstAudioResampler *
|
|
gst_audio_resampler_new (GstAudioResamplerMethod method,
|
|
GstAudioResamplerFlags flags,
|
|
GstAudioFormat format, guint channels,
|
|
guint in_rate, guint out_rate, GstStructure * options)
|
|
{
|
|
GstAudioResampler *resampler;
|
|
const GstAudioFormatInfo *info;
|
|
|
|
g_return_val_if_fail (in_rate != 0, FALSE);
|
|
g_return_val_if_fail (out_rate != 0, FALSE);
|
|
|
|
audio_resampler_init ();
|
|
|
|
resampler = g_slice_new0 (GstAudioResampler);
|
|
resampler->method = method;
|
|
resampler->flags = flags;
|
|
resampler->format = format;
|
|
resampler->channels = channels;
|
|
|
|
info = gst_audio_format_get_info (format);
|
|
resampler->bps = GST_AUDIO_FORMAT_INFO_WIDTH (info) / 8;
|
|
resampler->bpf = resampler->bps * channels;
|
|
resampler->sbuf = g_malloc0 (sizeof (gpointer) * channels);
|
|
|
|
GST_DEBUG ("method %d, bps %d, bpf %d", method, resampler->bps,
|
|
resampler->bpf);
|
|
|
|
gst_audio_resampler_update (resampler, in_rate, out_rate, options);
|
|
|
|
return resampler;
|
|
}
|
|
|
|
/**
|
|
* gst_audio_resampler_update:
|
|
* @resampler: a #GstAudioResampler
|
|
* @in_rate: new input rate
|
|
* @out_rate: new output rate
|
|
* @options: new options or %NULL
|
|
*
|
|
* Update the resampler parameters for @resampler. This function should
|
|
* not be called concurrently with any other function on @resampler.
|
|
*
|
|
* When @in_rate or @out_rate is 0, its value is unchanged.
|
|
*
|
|
* Returns: %TRUE if the new parameters could be set
|
|
*/
|
|
gboolean
|
|
gst_audio_resampler_update (GstAudioResampler * resampler,
|
|
guint in_rate, guint out_rate, GstStructure * options)
|
|
{
|
|
gint gcd;
|
|
|
|
g_return_val_if_fail (resampler != NULL, FALSE);
|
|
|
|
if (in_rate == 0)
|
|
in_rate = resampler->in_rate;
|
|
if (out_rate == 0)
|
|
out_rate = resampler->out_rate;
|
|
|
|
gcd = gst_util_greatest_common_divisor (in_rate, out_rate);
|
|
in_rate /= gcd;
|
|
out_rate /= gcd;
|
|
|
|
resampler->in_rate = in_rate;
|
|
resampler->out_rate = out_rate;
|
|
if (options) {
|
|
if (resampler->options)
|
|
gst_structure_free (resampler->options);
|
|
resampler->options = gst_structure_copy (options);
|
|
}
|
|
|
|
GST_DEBUG ("%u->%u", in_rate, out_rate);
|
|
|
|
resampler_calculate_taps (resampler);
|
|
resampler_dump (resampler);
|
|
|
|
resampler->samp_index = 0;
|
|
resampler->samp_phase = 0;
|
|
resampler->samples_avail = resampler->n_taps / 2 - 1;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/**
|
|
* gst_audio_resampler_free:
|
|
* @resampler: a #GstAudioResampler
|
|
*
|
|
* Free a previously allocated #GstAudioResampler @resampler.
|
|
*
|
|
* Since: 1.6
|
|
*/
|
|
void
|
|
gst_audio_resampler_free (GstAudioResampler * resampler)
|
|
{
|
|
g_return_if_fail (resampler != NULL);
|
|
|
|
g_free (resampler->taps);
|
|
g_free (resampler->coeffmem);
|
|
g_free (resampler->tmpcoeff);
|
|
g_free (resampler->samples);
|
|
g_free (resampler->sbuf);
|
|
if (resampler->options)
|
|
gst_structure_free (resampler->options);
|
|
g_slice_free (GstAudioResampler, resampler);
|
|
}
|
|
|
|
static inline gsize
|
|
calc_out (GstAudioResampler * resampler, gsize in)
|
|
{
|
|
gsize out;
|
|
|
|
out = ((in * resampler->out_rate -
|
|
resampler->samp_phase) / resampler->in_rate) + 1;
|
|
GST_LOG ("out %d = ((%d * %d - %d) / %d) + 1", (gint) out,
|
|
(gint) in, resampler->out_rate, resampler->samp_phase,
|
|
resampler->in_rate);
|
|
return out;
|
|
}
|
|
|
|
/**
|
|
* gst_audio_resampler_get_out_frames:
|
|
* @resampler: a #GstAudioResampler
|
|
* @in_frames: number of input frames
|
|
*
|
|
* Get the number of output frames that would be currently available when
|
|
* @in_frames are given to @resampler.
|
|
*
|
|
* Returns: The number of frames that would be availabe after giving
|
|
* @in_frames as input to @resampler.
|
|
*/
|
|
gsize
|
|
gst_audio_resampler_get_out_frames (GstAudioResampler * resampler,
|
|
gsize in_frames)
|
|
{
|
|
gsize need, avail;
|
|
|
|
g_return_val_if_fail (resampler != NULL, 0);
|
|
|
|
need = resampler->n_taps + resampler->samp_index + resampler->skip;
|
|
avail = resampler->samples_avail + in_frames;
|
|
GST_LOG ("need %d = %d + %d + %d, avail %d = %d + %d", (gint) need,
|
|
resampler->n_taps, resampler->samp_index, resampler->skip,
|
|
(gint) avail, (gint) resampler->samples_avail, (gint) in_frames);
|
|
if (avail < need)
|
|
return 0;
|
|
|
|
return calc_out (resampler, avail - need);
|
|
}
|
|
|
|
/**
|
|
* gst_audio_resampler_get_in_frames:
|
|
* @resampler: a #GstAudioResampler
|
|
* @out_frames: number of input frames
|
|
*
|
|
* Get the number of input frames that would currently be needed
|
|
* to produce @out_frames from @resampler.
|
|
*
|
|
* Returns: The number of input frames needed for producing
|
|
* @out_frames of data from @resampler.
|
|
*/
|
|
gsize
|
|
gst_audio_resampler_get_in_frames (GstAudioResampler * resampler,
|
|
gsize out_frames)
|
|
{
|
|
gsize in_frames;
|
|
|
|
g_return_val_if_fail (resampler != NULL, 0);
|
|
|
|
in_frames =
|
|
(resampler->samp_phase +
|
|
out_frames * resampler->samp_frac) / resampler->out_rate;
|
|
in_frames += out_frames * resampler->samp_inc;
|
|
|
|
return in_frames;
|
|
}
|
|
|
|
/**
|
|
* gst_audio_resampler_get_max_latency:
|
|
* @resampler: a #GstAudioResampler
|
|
*
|
|
* Get the maximum number of input samples that the resampler would
|
|
* need before producing output.
|
|
*
|
|
* Returns: the latency of @resampler as expressed in the number of
|
|
* frames.
|
|
*/
|
|
gsize
|
|
gst_audio_resampler_get_max_latency (GstAudioResampler * resampler)
|
|
{
|
|
g_return_val_if_fail (resampler != NULL, 0);
|
|
|
|
return resampler->n_taps / 2;
|
|
}
|
|
|
|
/* make the buffers to hold the (deinterleaved) samples */
|
|
static inline gpointer *
|
|
get_sample_bufs (GstAudioResampler * resampler, gsize need)
|
|
{
|
|
if (G_LIKELY (resampler->samples_len < need)) {
|
|
guint c, blocks = resampler->blocks;
|
|
gsize bytes;
|
|
gint8 *ptr;
|
|
|
|
GST_LOG ("realloc %d -> %d", (gint) resampler->samples_len, (gint) need);
|
|
|
|
bytes = GST_ROUND_UP_N (need * resampler->bps * resampler->inc, ALIGN);
|
|
|
|
/* FIXME, move history */
|
|
resampler->samples =
|
|
g_realloc (resampler->samples, resampler->blocks * bytes + ALIGN - 1);
|
|
resampler->samples_len = need;
|
|
|
|
ptr = MEM_ALIGN (resampler->samples, ALIGN);
|
|
|
|
/* set up new pointers */
|
|
for (c = 0; c < blocks; c++) {
|
|
resampler->sbuf[c] = ptr + (c * bytes);
|
|
}
|
|
}
|
|
return resampler->sbuf;
|
|
}
|
|
|
|
/**
|
|
* gst_audio_resampler_resample:
|
|
* @resampler: a #GstAudioResampler
|
|
* @in: input samples
|
|
* @in_frames: number of input frames
|
|
* @out: output samples
|
|
* @out_frames: number of output frames
|
|
*
|
|
* Perform resampling on @in_frames frames in @in and write @out_frames to @out.
|
|
*
|
|
* In case the samples are interleaved, @in and @out must point to an
|
|
* array with a single element pointing to a block of interleaved samples.
|
|
*
|
|
* If non-interleaved samples are used, @in and @out must point to an
|
|
* array with pointers to memory blocks, one for each channel.
|
|
*
|
|
* @in may be %NULL, in which case @in_frames of silence samples are pushed
|
|
* into the resampler.
|
|
*
|
|
* This function always produces @out_frames of output and consumes @in_frames of
|
|
* input. Use gst_audio_resampler_get_out_frames() and
|
|
* gst_audio_resampler_get_in_frames() to make sure @in_frames and @out_frames
|
|
* are matching and @in and @out point to enough memory.
|
|
*/
|
|
void
|
|
gst_audio_resampler_resample (GstAudioResampler * resampler,
|
|
gpointer in[], gsize in_frames, gpointer out[], gsize out_frames)
|
|
{
|
|
gsize samples_avail;
|
|
gsize need, consumed;
|
|
gpointer *sbuf;
|
|
|
|
/* do sample skipping */
|
|
if (G_UNLIKELY (resampler->skip >= in_frames)) {
|
|
/* we need tp skip all input */
|
|
resampler->skip -= in_frames;
|
|
return;
|
|
}
|
|
/* skip the last samples by advancing the sample index */
|
|
resampler->samp_index += resampler->skip;
|
|
|
|
samples_avail = resampler->samples_avail;
|
|
|
|
/* make sure we have enough space to copy our samples */
|
|
sbuf = get_sample_bufs (resampler, in_frames + samples_avail);
|
|
|
|
/* copy/deinterleave the samples */
|
|
resampler->deinterleave (resampler, sbuf, in, in_frames);
|
|
|
|
/* update new amount of samples in our buffer */
|
|
resampler->samples_avail = samples_avail += in_frames;
|
|
|
|
need = resampler->n_taps + resampler->samp_index;
|
|
if (G_UNLIKELY (samples_avail < need)) {
|
|
/* not enough samples to start */
|
|
return;
|
|
}
|
|
|
|
/* resample all channels */
|
|
resampler->resample (resampler, sbuf, samples_avail, out, out_frames,
|
|
&consumed);
|
|
|
|
GST_LOG ("in %" G_GSIZE_FORMAT ", used %" G_GSIZE_FORMAT ", consumed %"
|
|
G_GSIZE_FORMAT, in_frames, samples_avail, consumed);
|
|
|
|
/* update pointers */
|
|
if (G_LIKELY (consumed > 0)) {
|
|
gssize left = samples_avail - consumed;
|
|
if (left > 0) {
|
|
/* we consumed part of our samples */
|
|
resampler->samples_avail = left;
|
|
} else {
|
|
/* we consumed all our samples, empty our buffers */
|
|
resampler->samples_avail = 0;
|
|
resampler->skip = -left;
|
|
}
|
|
}
|
|
}
|