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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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1808 lines
69 KiB
C
1808 lines
69 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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#include "audio-resampler-private.h"
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#include "audio-resampler-macros.h"
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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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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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* @title: 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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static const gint oversample_qualities[] = {
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4, 4, 4, 8, 8, 16, 16, 16, 16, 32, 32
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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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gint 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_RESAMPLER_METHOD GST_AUDIO_RESAMPLER_METHOD_KAISER
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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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#define DEFAULT_OPT_FILTER_MODE GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO
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#define DEFAULT_OPT_FILTER_MODE_THRESHOLD 1048576
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#define DEFAULT_OPT_FILTER_INTERPOLATION GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC
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#define DEFAULT_OPT_FILTER_OVERSAMPLE 8
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#define DEFAULT_OPT_MAX_PHASE_ERROR 0.1
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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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static gint
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get_opt_enum (GstStructure * options, const gchar * name, GType type, gint def)
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{
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gint res;
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if (!options || !gst_structure_get_enum (options, name, type, &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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#define GET_OPT_FILTER_MODE(options) get_opt_enum(options, \
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GST_AUDIO_RESAMPLER_OPT_FILTER_MODE, GST_TYPE_AUDIO_RESAMPLER_FILTER_MODE, \
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DEFAULT_OPT_FILTER_MODE)
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#define GET_OPT_FILTER_MODE_THRESHOLD(options) get_opt_int(options, \
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GST_AUDIO_RESAMPLER_OPT_FILTER_MODE_THRESHOLD, DEFAULT_OPT_FILTER_MODE_THRESHOLD)
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#define GET_OPT_FILTER_INTERPOLATION(options) get_opt_enum(options, \
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GST_AUDIO_RESAMPLER_OPT_FILTER_INTERPOLATION, GST_TYPE_AUDIO_RESAMPLER_FILTER_INTERPOLATION, \
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DEFAULT_OPT_FILTER_INTERPOLATION)
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#define GET_OPT_FILTER_OVERSAMPLE(options) get_opt_int(options, \
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GST_AUDIO_RESAMPLER_OPT_FILTER_OVERSAMPLE, DEFAULT_OPT_FILTER_OVERSAMPLE)
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#define GET_OPT_MAX_PHASE_ERROR(options) get_opt_double(options, \
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GST_AUDIO_RESAMPLER_OPT_MAX_PHASE_ERROR, DEFAULT_OPT_MAX_PHASE_ERROR)
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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_linear_tap (gdouble x, gint n_taps)
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{
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gdouble res = GST_ROUND_UP_2 (n_taps) / 2 - fabs (x);
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return res;
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}
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static inline gdouble
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get_cubic_tap (gdouble x, gint n_taps, gdouble b, gdouble c)
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{
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gdouble res, a, a2, a3;
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a = fabs (x * 4.0) / n_taps;
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a2 = a * a;
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a3 = a2 * a;
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if (a <= 1.0)
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res = ((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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res = ((-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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res = 0.0;
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return res;
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}
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static inline gdouble
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get_blackman_nuttall_tap (gdouble x, gint n_taps, gdouble Fc)
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{
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gdouble s, y, w;
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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 / 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 (gdouble x, gint n_taps, gdouble Fc, gdouble beta)
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{
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gdouble s, y, w;
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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 / n_taps;
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return s * bessel (beta * sqrt (MAX (1 - w * w, 0)));
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}
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#define MAKE_CONVERT_TAPS_INT_FUNC(type, precision) \
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static void \
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convert_taps_##type##_c (gdouble *tmp_taps, gpointer taps, \
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gdouble weight, gint n_taps) \
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{ \
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gint64 one = (1LL << precision) - 1; \
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type *t = taps; \
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gdouble multiplier = one; \
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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 += floor (offset + tmp_taps[j] * multiplier / weight); \
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if (sum == one) { \
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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 < one) { \
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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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for (j = 0; j < n_taps; j++) \
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t[j] = floor (offset + tmp_taps[j] * multiplier / weight); \
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if (!exact) \
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GST_DEBUG ("can't find exact taps"); \
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}
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#define MAKE_CONVERT_TAPS_FLOAT_FUNC(type) \
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static void \
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convert_taps_##type##_c (gdouble *tmp_taps, gpointer taps, \
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gdouble weight, gint n_taps) \
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{ \
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gint i; \
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type *t = taps; \
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for (i = 0; i < n_taps; i++) \
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t[i] = tmp_taps[i] / weight; \
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}
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MAKE_CONVERT_TAPS_INT_FUNC (gint16, PRECISION_S16);
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MAKE_CONVERT_TAPS_INT_FUNC (gint32, PRECISION_S32);
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MAKE_CONVERT_TAPS_FLOAT_FUNC (gfloat);
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MAKE_CONVERT_TAPS_FLOAT_FUNC (gdouble);
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static ConvertTapsFunc convert_taps_funcs[] = {
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convert_taps_gint16_c,
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convert_taps_gint32_c,
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convert_taps_gfloat_c,
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convert_taps_gdouble_c
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};
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#define convert_taps_gint16 convert_taps_funcs[0]
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#define convert_taps_gint32 convert_taps_funcs[1]
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#define convert_taps_gfloat convert_taps_funcs[2]
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#define convert_taps_gdouble convert_taps_funcs[3]
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static void
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make_taps (GstAudioResampler * resampler, gdouble * res, gdouble x, gint n_taps)
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{
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gdouble weight = 0.0, *tmp_taps = resampler->tmp_taps;
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gint i;
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switch (resampler->method) {
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case GST_AUDIO_RESAMPLER_METHOD_NEAREST:
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break;
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case GST_AUDIO_RESAMPLER_METHOD_LINEAR:
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for (i = 0; i < n_taps; i++)
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weight += tmp_taps[i] = get_linear_tap (x + i, resampler->n_taps);
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break;
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case GST_AUDIO_RESAMPLER_METHOD_CUBIC:
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for (i = 0; i < n_taps; i++)
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weight += tmp_taps[i] = get_cubic_tap (x + i, resampler->n_taps,
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resampler->b, resampler->c);
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break;
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case GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL:
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for (i = 0; i < n_taps; i++)
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weight += tmp_taps[i] =
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get_blackman_nuttall_tap (x + i,
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resampler->n_taps, resampler->cutoff);
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break;
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case GST_AUDIO_RESAMPLER_METHOD_KAISER:
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for (i = 0; i < n_taps; i++)
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weight += tmp_taps[i] =
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get_kaiser_tap (x + i, resampler->n_taps,
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resampler->cutoff, resampler->kaiser_beta);
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break;
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}
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resampler->convert_taps (tmp_taps, res, weight, n_taps);
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}
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#define MAKE_COEFF_LINEAR_INT_FUNC(type,type2,prec) \
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static inline void \
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make_coeff_##type##_linear (gint num, gint denom, type *icoeff) \
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{ \
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type x = ((gint64)num << prec) / denom; \
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icoeff[0] = icoeff[2] = x; \
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icoeff[1] = icoeff[3] = (type)(((type2)1 << prec)-1) - x; \
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}
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#define MAKE_COEFF_LINEAR_FLOAT_FUNC(type) \
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static inline void \
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make_coeff_##type##_linear (gint num, gint denom, type *icoeff) \
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{ \
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type x = (type)num / denom; \
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icoeff[0] = icoeff[2] = x; \
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icoeff[1] = icoeff[3] = (type)1.0 - x; \
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}
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MAKE_COEFF_LINEAR_INT_FUNC (gint16, gint32, PRECISION_S16);
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MAKE_COEFF_LINEAR_INT_FUNC (gint32, gint64, PRECISION_S32);
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MAKE_COEFF_LINEAR_FLOAT_FUNC (gfloat);
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MAKE_COEFF_LINEAR_FLOAT_FUNC (gdouble);
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#define MAKE_COEFF_CUBIC_INT_FUNC(type,type2,prec) \
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static inline void \
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make_coeff_##type##_cubic (gint num, gint denom, type *icoeff) \
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{ \
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type2 one = ((type2)1 << prec) - 1; \
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type2 x = ((gint64) num << prec) / denom; \
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type2 x2 = (x * x) >> prec; \
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type2 x3 = (x2 * x) >> prec; \
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icoeff[0] = (((x3 - x) << prec) / 6) >> prec; \
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icoeff[1] = x + ((x2 - x3) >> 1); \
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icoeff[3] = -(((x << prec) / 3) >> prec) + \
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(x2 >> 1) - (((x3 << prec) / 6) >> prec); \
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icoeff[2] = one - icoeff[0] - icoeff[1] - icoeff[3]; \
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}
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#define MAKE_COEFF_CUBIC_FLOAT_FUNC(type) \
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static inline void \
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make_coeff_##type##_cubic (gint num, gint denom, type *icoeff) \
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{ \
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type x = (type) num / denom, x2 = x * x, x3 = x2 * x; \
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icoeff[0] = 0.16667f * (x3 - x); \
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icoeff[1] = x + 0.5f * (x2 - x3); \
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icoeff[3] = -0.33333f * x + 0.5f * x2 - 0.16667f * x3; \
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icoeff[2] = (type)1.0 - icoeff[0] - icoeff[1] - icoeff[3]; \
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}
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MAKE_COEFF_CUBIC_INT_FUNC (gint16, gint32, PRECISION_S16);
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MAKE_COEFF_CUBIC_INT_FUNC (gint32, gint64, PRECISION_S32);
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MAKE_COEFF_CUBIC_FLOAT_FUNC (gfloat);
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MAKE_COEFF_CUBIC_FLOAT_FUNC (gdouble);
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#define INTERPOLATE_INT_LINEAR_FUNC(type,type2,prec,limit) \
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static inline void \
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interpolate_##type##_linear_c (gpointer op, const gpointer ap, \
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gint len, const gpointer icp, gint astride) \
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{ \
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gint i; \
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type *o = op, *a = ap, *ic = icp; \
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type2 tmp, c0 = ic[0]; \
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const type *c[2] = {(type*)((gint8*)a + 0*astride), \
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(type*)((gint8*)a + 1*astride)}; \
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\
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for (i = 0; i < len; i++) { \
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tmp = ((type2)c[0][i] - (type2)c[1][i]) * c0 + \
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(((type2)c[1][i]) << (prec)); \
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o[i] = (tmp + ((type2)1 << ((prec) - 1))) >> (prec); \
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} \
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}
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#define INTERPOLATE_FLOAT_LINEAR_FUNC(type) \
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static inline void \
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interpolate_##type##_linear_c (gpointer op, const gpointer ap, \
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gint len, const gpointer icp, gint astride) \
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{ \
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gint i; \
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type *o = op, *a = ap, *ic = icp; \
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type c0 = ic[0]; \
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const type *c[2] = {(type*)((gint8*)a + 0*astride), \
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(type*)((gint8*)a + 1*astride)}; \
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\
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for (i = 0; i < len; i++) { \
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o[i] = (c[0][i] - c[1][i]) * c0 + c[1][i]; \
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} \
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}
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INTERPOLATE_INT_LINEAR_FUNC (gint16, gint32, PRECISION_S16, (gint32) 1 << 15);
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INTERPOLATE_INT_LINEAR_FUNC (gint32, gint64, PRECISION_S32, (gint64) 1 << 31);
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INTERPOLATE_FLOAT_LINEAR_FUNC (gfloat);
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INTERPOLATE_FLOAT_LINEAR_FUNC (gdouble);
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|
|
#define INTERPOLATE_INT_CUBIC_FUNC(type,type2,prec,limit) \
|
|
static inline void \
|
|
interpolate_##type##_cubic_c (gpointer op, const gpointer ap, \
|
|
gint len, const gpointer icp, gint astride) \
|
|
{ \
|
|
gint i; \
|
|
type *o = op, *a = ap, *ic = icp; \
|
|
type2 tmp, c0 = ic[0], c1 = ic[1], c2 = ic[2], c3 = ic[3]; \
|
|
const type *c[4] = {(type*)((gint8*)a + 0*astride), \
|
|
(type*)((gint8*)a + 1*astride), \
|
|
(type*)((gint8*)a + 2*astride), \
|
|
(type*)((gint8*)a + 3*astride)}; \
|
|
\
|
|
for (i = 0; i < len; i++) { \
|
|
tmp = (type2)c[0][i] * c0 + (type2)c[1][i] * c1 + \
|
|
(type2)c[2][i] * c2 + (type2)c[3][i] * c3; \
|
|
tmp = (tmp + ((type2)1 << ((prec) - 1))) >> (prec); \
|
|
o[i] = CLAMP (tmp, -(limit), (limit) - 1); \
|
|
} \
|
|
}
|
|
#define INTERPOLATE_FLOAT_CUBIC_FUNC(type) \
|
|
static inline void \
|
|
interpolate_##type##_cubic_c (gpointer op, const gpointer ap, \
|
|
gint len, const gpointer icp, gint astride) \
|
|
{ \
|
|
gint i; \
|
|
type *o = op, *a = ap, *ic = icp; \
|
|
type c0 = ic[0], c1 = ic[1], c2 = ic[2], c3 = ic[3]; \
|
|
const type *c[4] = {(type*)((gint8*)a + 0*astride), \
|
|
(type*)((gint8*)a + 1*astride), \
|
|
(type*)((gint8*)a + 2*astride), \
|
|
(type*)((gint8*)a + 3*astride)}; \
|
|
\
|
|
for (i = 0; i < len; i++) { \
|
|
o[i] = c[0][i] * c0 + c[1][i] * c1 + \
|
|
c[2][i] * c2 + c[3][i] * c3; \
|
|
} \
|
|
}
|
|
|
|
INTERPOLATE_INT_CUBIC_FUNC (gint16, gint32, PRECISION_S16, (gint32) 1 << 15);
|
|
INTERPOLATE_INT_CUBIC_FUNC (gint32, gint64, PRECISION_S32, (gint64) 1 << 31);
|
|
INTERPOLATE_FLOAT_CUBIC_FUNC (gfloat);
|
|
INTERPOLATE_FLOAT_CUBIC_FUNC (gdouble);
|
|
|
|
static InterpolateFunc interpolate_funcs[] = {
|
|
interpolate_gint16_linear_c,
|
|
interpolate_gint32_linear_c,
|
|
interpolate_gfloat_linear_c,
|
|
interpolate_gdouble_linear_c,
|
|
|
|
interpolate_gint16_cubic_c,
|
|
interpolate_gint32_cubic_c,
|
|
interpolate_gfloat_cubic_c,
|
|
interpolate_gdouble_cubic_c,
|
|
};
|
|
|
|
#define interpolate_gint16_linear interpolate_funcs[0]
|
|
#define interpolate_gint32_linear interpolate_funcs[1]
|
|
#define interpolate_gfloat_linear interpolate_funcs[2]
|
|
#define interpolate_gdouble_linear interpolate_funcs[3]
|
|
|
|
#define interpolate_gint16_cubic interpolate_funcs[4]
|
|
#define interpolate_gint32_cubic interpolate_funcs[5]
|
|
#define interpolate_gfloat_cubic interpolate_funcs[6]
|
|
#define interpolate_gdouble_cubic interpolate_funcs[7]
|
|
|
|
#define GET_TAPS_NEAREST_FUNC(type) \
|
|
static inline gpointer \
|
|
get_taps_##type##_nearest (GstAudioResampler * resampler, \
|
|
gint *samp_index, gint *samp_phase, type icoeff[4]) \
|
|
{ \
|
|
gint out_rate = resampler->out_rate; \
|
|
*samp_index += resampler->samp_inc; \
|
|
*samp_phase += resampler->samp_frac; \
|
|
if (*samp_phase >= out_rate) { \
|
|
*samp_phase -= out_rate; \
|
|
*samp_index += 1; \
|
|
} \
|
|
return NULL; \
|
|
}
|
|
GET_TAPS_NEAREST_FUNC (gint16);
|
|
GET_TAPS_NEAREST_FUNC (gint32);
|
|
GET_TAPS_NEAREST_FUNC (gfloat);
|
|
GET_TAPS_NEAREST_FUNC (gdouble);
|
|
|
|
#define get_taps_gint16_nearest get_taps_gint16_nearest
|
|
#define get_taps_gint32_nearest get_taps_gint32_nearest
|
|
#define get_taps_gfloat_nearest get_taps_gfloat_nearest
|
|
#define get_taps_gdouble_nearest get_taps_gdouble_nearest
|
|
|
|
#define GET_TAPS_FULL_FUNC(type) \
|
|
DECL_GET_TAPS_FULL_FUNC(type) \
|
|
{ \
|
|
gpointer res; \
|
|
gint out_rate = resampler->out_rate; \
|
|
gint n_phases = resampler->n_phases; \
|
|
gint phase = (n_phases == out_rate ? *samp_phase : \
|
|
((gint64)*samp_phase * n_phases) / out_rate); \
|
|
\
|
|
res = resampler->cached_phases[phase]; \
|
|
if (G_UNLIKELY (res == NULL)) { \
|
|
res = (gint8 *) resampler->cached_taps + \
|
|
phase * resampler->cached_taps_stride; \
|
|
switch (resampler->filter_interpolation) { \
|
|
case GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_NONE: \
|
|
{ \
|
|
gdouble x; \
|
|
gint n_taps = resampler->n_taps; \
|
|
\
|
|
x = 1.0 - n_taps / 2 - (gdouble) phase / n_phases; \
|
|
make_taps (resampler, res, x, n_taps); \
|
|
break; \
|
|
} \
|
|
default: \
|
|
{ \
|
|
gint offset, pos, frac; \
|
|
gint oversample = resampler->oversample; \
|
|
gint taps_stride = resampler->taps_stride; \
|
|
gint n_taps = resampler->n_taps; \
|
|
type ic[4], *taps; \
|
|
\
|
|
pos = phase * oversample; \
|
|
offset = (oversample - 1) - pos / n_phases; \
|
|
frac = pos % n_phases; \
|
|
\
|
|
taps = (type *) ((gint8 *) resampler->taps + offset * taps_stride); \
|
|
\
|
|
switch (resampler->filter_interpolation) { \
|
|
default: \
|
|
case GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_LINEAR: \
|
|
make_coeff_##type##_linear (frac, n_phases, ic); \
|
|
break; \
|
|
case GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC: \
|
|
make_coeff_##type##_cubic (frac, n_phases, ic); \
|
|
break; \
|
|
} \
|
|
resampler->interpolate (res, taps, n_taps, ic, taps_stride); \
|
|
} \
|
|
} \
|
|
resampler->cached_phases[phase] = res; \
|
|
} \
|
|
*samp_index += resampler->samp_inc; \
|
|
*samp_phase += resampler->samp_frac; \
|
|
if (*samp_phase >= out_rate) { \
|
|
*samp_phase -= out_rate; \
|
|
*samp_index += 1; \
|
|
} \
|
|
return res; \
|
|
}
|
|
GET_TAPS_FULL_FUNC (gint16);
|
|
GET_TAPS_FULL_FUNC (gint32);
|
|
GET_TAPS_FULL_FUNC (gfloat);
|
|
GET_TAPS_FULL_FUNC (gdouble);
|
|
|
|
#define GET_TAPS_INTERPOLATE_FUNC(type,inter) \
|
|
DECL_GET_TAPS_INTERPOLATE_FUNC (type, inter) \
|
|
{ \
|
|
gpointer res; \
|
|
gint out_rate = resampler->out_rate; \
|
|
gint offset, frac, pos; \
|
|
gint oversample = resampler->oversample; \
|
|
gint taps_stride = resampler->taps_stride; \
|
|
\
|
|
pos = *samp_phase * oversample; \
|
|
offset = (oversample - 1) - pos / out_rate; \
|
|
frac = pos % out_rate; \
|
|
\
|
|
res = (gint8 *) resampler->taps + offset * taps_stride; \
|
|
make_coeff_##type##_##inter (frac, out_rate, icoeff); \
|
|
\
|
|
*samp_index += resampler->samp_inc; \
|
|
*samp_phase += resampler->samp_frac; \
|
|
if (*samp_phase >= out_rate) { \
|
|
*samp_phase -= out_rate; \
|
|
*samp_index += 1; \
|
|
} \
|
|
return res; \
|
|
}
|
|
|
|
GET_TAPS_INTERPOLATE_FUNC (gint16, linear);
|
|
GET_TAPS_INTERPOLATE_FUNC (gint32, linear);
|
|
GET_TAPS_INTERPOLATE_FUNC (gfloat, linear);
|
|
GET_TAPS_INTERPOLATE_FUNC (gdouble, linear);
|
|
|
|
GET_TAPS_INTERPOLATE_FUNC (gint16, cubic);
|
|
GET_TAPS_INTERPOLATE_FUNC (gint32, cubic);
|
|
GET_TAPS_INTERPOLATE_FUNC (gfloat, cubic);
|
|
GET_TAPS_INTERPOLATE_FUNC (gdouble, cubic);
|
|
|
|
#define INNER_PRODUCT_NEAREST_FUNC(type) \
|
|
static inline void \
|
|
inner_product_##type##_nearest_1_c (type * o, const type * a, \
|
|
const type * b, gint len, const type *ic, gint bstride) \
|
|
{ \
|
|
*o = *a; \
|
|
}
|
|
INNER_PRODUCT_NEAREST_FUNC (gint16);
|
|
INNER_PRODUCT_NEAREST_FUNC (gint32);
|
|
INNER_PRODUCT_NEAREST_FUNC (gfloat);
|
|
INNER_PRODUCT_NEAREST_FUNC (gdouble);
|
|
|
|
#define INNER_PRODUCT_INT_FULL_FUNC(type,type2,prec,limit) \
|
|
static inline void \
|
|
inner_product_##type##_full_1_c (type * o, const type * a, \
|
|
const type * b, gint len, const type *ic, gint bstride) \
|
|
{ \
|
|
gint i; \
|
|
type2 res[4] = { 0, 0, 0, 0 }; \
|
|
\
|
|
for (i = 0; i < len; i += 4) { \
|
|
res[0] += (type2) a[i + 0] * (type2) b[i + 0]; \
|
|
res[1] += (type2) a[i + 1] * (type2) b[i + 1]; \
|
|
res[2] += (type2) a[i + 2] * (type2) b[i + 2]; \
|
|
res[3] += (type2) a[i + 3] * (type2) b[i + 3]; \
|
|
} \
|
|
res[0] = res[0] + res[1] + res[2] + res[3]; \
|
|
res[0] = (res[0] + ((type2)1 << ((prec) - 1))) >> (prec); \
|
|
*o = CLAMP (res[0], -(limit), (limit) - 1); \
|
|
}
|
|
|
|
INNER_PRODUCT_INT_FULL_FUNC (gint16, gint32, PRECISION_S16, (gint32) 1 << 15);
|
|
INNER_PRODUCT_INT_FULL_FUNC (gint32, gint64, PRECISION_S32, (gint64) 1 << 31);
|
|
|
|
#define INNER_PRODUCT_INT_LINEAR_FUNC(type,type2,prec,limit) \
|
|
static inline void \
|
|
inner_product_##type##_linear_1_c (type * o, const type * a, \
|
|
const type * b, gint len, const type *ic, gint bstride) \
|
|
{ \
|
|
gint i; \
|
|
type2 res[4] = { 0, 0, 0, 0 }, c0 = ic[0]; \
|
|
const type *c[2] = {(type*)((gint8*)b + 0*bstride), \
|
|
(type*)((gint8*)b + 1*bstride)}; \
|
|
\
|
|
for (i = 0; i < len; i += 2) { \
|
|
res[0] += (type2) a[i + 0] * (type2) c[0][i + 0]; \
|
|
res[1] += (type2) a[i + 0] * (type2) c[1][i + 0]; \
|
|
res[2] += (type2) a[i + 1] * (type2) c[0][i + 1]; \
|
|
res[3] += (type2) a[i + 1] * (type2) c[1][i + 1]; \
|
|
} \
|
|
res[0] = (res[0] + res[2]) >> (prec); \
|
|
res[1] = (res[1] + res[3]) >> (prec); \
|
|
res[0] = ((type2)(type)res[0] - (type2)(type)res[1]) * c0 + \
|
|
((type2)(type)res[1] << (prec)); \
|
|
res[0] = (res[0] + ((type2)1 << ((prec) - 1))) >> (prec); \
|
|
*o = CLAMP (res[0], -(limit), (limit) - 1); \
|
|
}
|
|
|
|
INNER_PRODUCT_INT_LINEAR_FUNC (gint16, gint32, PRECISION_S16, (gint32) 1 << 15);
|
|
INNER_PRODUCT_INT_LINEAR_FUNC (gint32, gint64, PRECISION_S32, (gint64) 1 << 31);
|
|
|
|
#define INNER_PRODUCT_INT_CUBIC_FUNC(type,type2,prec,limit) \
|
|
static inline void \
|
|
inner_product_##type##_cubic_1_c (type * o, const type * a, \
|
|
const type * b, gint len, const type *ic, gint bstride) \
|
|
{ \
|
|
gint i; \
|
|
type2 res[4] = { 0, 0, 0, 0 }; \
|
|
const type *c[4] = {(type*)((gint8*)b + 0*bstride), \
|
|
(type*)((gint8*)b + 1*bstride), \
|
|
(type*)((gint8*)b + 2*bstride), \
|
|
(type*)((gint8*)b + 3*bstride)}; \
|
|
\
|
|
for (i = 0; i < len; i++) { \
|
|
res[0] += (type2) a[i] * (type2) c[0][i]; \
|
|
res[1] += (type2) a[i] * (type2) c[1][i]; \
|
|
res[2] += (type2) a[i] * (type2) c[2][i]; \
|
|
res[3] += (type2) a[i] * (type2) c[3][i]; \
|
|
} \
|
|
res[0] = (type2)(type)(res[0] >> (prec)) * (type2) ic[0] + \
|
|
(type2)(type)(res[1] >> (prec)) * (type2) ic[1] + \
|
|
(type2)(type)(res[2] >> (prec)) * (type2) ic[2] + \
|
|
(type2)(type)(res[3] >> (prec)) * (type2) ic[3]; \
|
|
res[0] = (res[0] + ((type2)1 << ((prec) - 1))) >> (prec); \
|
|
*o = CLAMP (res[0], -(limit), (limit) - 1); \
|
|
}
|
|
|
|
INNER_PRODUCT_INT_CUBIC_FUNC (gint16, gint32, PRECISION_S16, (gint32) 1 << 15);
|
|
INNER_PRODUCT_INT_CUBIC_FUNC (gint32, gint64, PRECISION_S32, (gint64) 1 << 31);
|
|
|
|
#define INNER_PRODUCT_FLOAT_FULL_FUNC(type) \
|
|
static inline void \
|
|
inner_product_##type##_full_1_c (type * o, const type * a, \
|
|
const type * b, gint len, const type *ic, gint bstride) \
|
|
{ \
|
|
gint i; \
|
|
type res[4] = { 0.0, 0.0, 0.0, 0.0 }; \
|
|
\
|
|
for (i = 0; i < len; i += 4) { \
|
|
res[0] += a[i + 0] * b[i + 0]; \
|
|
res[1] += a[i + 1] * b[i + 1]; \
|
|
res[2] += a[i + 2] * b[i + 2]; \
|
|
res[3] += a[i + 3] * b[i + 3]; \
|
|
} \
|
|
*o = res[0] + res[1] + res[2] + res[3]; \
|
|
}
|
|
|
|
INNER_PRODUCT_FLOAT_FULL_FUNC (gfloat);
|
|
INNER_PRODUCT_FLOAT_FULL_FUNC (gdouble);
|
|
|
|
#define INNER_PRODUCT_FLOAT_LINEAR_FUNC(type) \
|
|
static inline void \
|
|
inner_product_##type##_linear_1_c (type * o, const type * a, \
|
|
const type * b, gint len, const type *ic, gint bstride) \
|
|
{ \
|
|
gint i; \
|
|
type res[4] = { 0.0, 0.0, 0.0, 0.0 }; \
|
|
const type *c[2] = {(type*)((gint8*)b + 0*bstride), \
|
|
(type*)((gint8*)b + 1*bstride)}; \
|
|
\
|
|
for (i = 0; i < len; i += 2) { \
|
|
res[0] += a[i + 0] * c[0][i + 0]; \
|
|
res[1] += a[i + 0] * c[1][i + 0]; \
|
|
res[2] += a[i + 1] * c[0][i + 1]; \
|
|
res[3] += a[i + 1] * c[1][i + 1]; \
|
|
} \
|
|
res[0] += res[2]; \
|
|
res[1] += res[3]; \
|
|
*o = (res[0] - res[1]) * ic[0] + res[1]; \
|
|
}
|
|
INNER_PRODUCT_FLOAT_LINEAR_FUNC (gfloat);
|
|
INNER_PRODUCT_FLOAT_LINEAR_FUNC (gdouble);
|
|
|
|
#define INNER_PRODUCT_FLOAT_CUBIC_FUNC(type) \
|
|
static inline void \
|
|
inner_product_##type##_cubic_1_c (type * o, const type * a, \
|
|
const type * b, gint len, const type *ic, gint bstride) \
|
|
{ \
|
|
gint i; \
|
|
type res[4] = { 0.0, 0.0, 0.0, 0.0 }; \
|
|
const type *c[4] = {(type*)((gint8*)b + 0*bstride), \
|
|
(type*)((gint8*)b + 1*bstride), \
|
|
(type*)((gint8*)b + 2*bstride), \
|
|
(type*)((gint8*)b + 3*bstride)}; \
|
|
\
|
|
for (i = 0; i < len; i++) { \
|
|
res[0] += a[i] * c[0][i]; \
|
|
res[1] += a[i] * c[1][i]; \
|
|
res[2] += a[i] * c[2][i]; \
|
|
res[3] += a[i] * c[3][i]; \
|
|
} \
|
|
*o = res[0] * ic[0] + res[1] * ic[1] + \
|
|
res[2] * ic[2] + res[3] * ic[3]; \
|
|
}
|
|
INNER_PRODUCT_FLOAT_CUBIC_FUNC (gfloat);
|
|
INNER_PRODUCT_FLOAT_CUBIC_FUNC (gdouble);
|
|
|
|
MAKE_RESAMPLE_FUNC_STATIC (gint16, nearest, 1, c);
|
|
MAKE_RESAMPLE_FUNC_STATIC (gint32, nearest, 1, c);
|
|
MAKE_RESAMPLE_FUNC_STATIC (gfloat, nearest, 1, c);
|
|
MAKE_RESAMPLE_FUNC_STATIC (gdouble, nearest, 1, c);
|
|
|
|
MAKE_RESAMPLE_FUNC_STATIC (gint16, full, 1, c);
|
|
MAKE_RESAMPLE_FUNC_STATIC (gint32, full, 1, c);
|
|
MAKE_RESAMPLE_FUNC_STATIC (gfloat, full, 1, c);
|
|
MAKE_RESAMPLE_FUNC_STATIC (gdouble, full, 1, c);
|
|
|
|
MAKE_RESAMPLE_FUNC_STATIC (gint16, linear, 1, c);
|
|
MAKE_RESAMPLE_FUNC_STATIC (gint32, linear, 1, c);
|
|
MAKE_RESAMPLE_FUNC_STATIC (gfloat, linear, 1, c);
|
|
MAKE_RESAMPLE_FUNC_STATIC (gdouble, linear, 1, c);
|
|
|
|
MAKE_RESAMPLE_FUNC_STATIC (gint16, cubic, 1, c);
|
|
MAKE_RESAMPLE_FUNC_STATIC (gint32, cubic, 1, c);
|
|
MAKE_RESAMPLE_FUNC_STATIC (gfloat, cubic, 1, c);
|
|
MAKE_RESAMPLE_FUNC_STATIC (gdouble, cubic, 1, c);
|
|
|
|
static ResampleFunc resample_funcs[] = {
|
|
resample_gint16_nearest_1_c,
|
|
resample_gint32_nearest_1_c,
|
|
resample_gfloat_nearest_1_c,
|
|
resample_gdouble_nearest_1_c,
|
|
|
|
resample_gint16_full_1_c,
|
|
resample_gint32_full_1_c,
|
|
resample_gfloat_full_1_c,
|
|
resample_gdouble_full_1_c,
|
|
|
|
resample_gint16_linear_1_c,
|
|
resample_gint32_linear_1_c,
|
|
resample_gfloat_linear_1_c,
|
|
resample_gdouble_linear_1_c,
|
|
|
|
resample_gint16_cubic_1_c,
|
|
resample_gint32_cubic_1_c,
|
|
resample_gfloat_cubic_1_c,
|
|
resample_gdouble_cubic_1_c,
|
|
};
|
|
|
|
#define resample_gint16_nearest_1 resample_funcs[0]
|
|
#define resample_gint32_nearest_1 resample_funcs[1]
|
|
#define resample_gfloat_nearest_1 resample_funcs[2]
|
|
#define resample_gdouble_nearest_1 resample_funcs[3]
|
|
|
|
#define resample_gint16_full_1 resample_funcs[4]
|
|
#define resample_gint32_full_1 resample_funcs[5]
|
|
#define resample_gfloat_full_1 resample_funcs[6]
|
|
#define resample_gdouble_full_1 resample_funcs[7]
|
|
|
|
#define resample_gint16_linear_1 resample_funcs[8]
|
|
#define resample_gint32_linear_1 resample_funcs[9]
|
|
#define resample_gfloat_linear_1 resample_funcs[10]
|
|
#define resample_gdouble_linear_1 resample_funcs[11]
|
|
|
|
#define resample_gint16_cubic_1 resample_funcs[12]
|
|
#define resample_gint32_cubic_1 resample_funcs[13]
|
|
#define resample_gfloat_cubic_1 resample_funcs[14]
|
|
#define resample_gdouble_cubic_1 resample_funcs[15]
|
|
|
|
#if defined HAVE_ORC && !defined DISABLE_ORC
|
|
# if defined (HAVE_ARM_NEON)
|
|
# define CHECK_NEON
|
|
# include "audio-resampler-neon.h"
|
|
# endif
|
|
# 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) {
|
|
const gchar *name;
|
|
unsigned int flags = orc_target_get_default_flags (target);
|
|
|
|
for (i = -1; i < 32; ++i) {
|
|
if (i == -1) {
|
|
name = orc_target_get_name (target);
|
|
GST_DEBUG ("target %s, default flags %08x", name, flags);
|
|
} else if (flags & (1U << i)) {
|
|
name = orc_target_get_flag_name (target, i);
|
|
GST_DEBUG ("target flag %s", name);
|
|
} else
|
|
name = NULL;
|
|
|
|
if (name) {
|
|
#ifdef CHECK_X86
|
|
audio_resampler_check_x86 (name);
|
|
#endif
|
|
#ifdef CHECK_NEON
|
|
audio_resampler_check_neon (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) \
|
|
{ \
|
|
gint 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 (gint16);
|
|
MAKE_DEINTERLEAVE_FUNC (gint32);
|
|
MAKE_DEINTERLEAVE_FUNC (gfloat);
|
|
MAKE_DEINTERLEAVE_FUNC (gdouble);
|
|
|
|
static DeinterleaveFunc deinterleave_funcs[] = {
|
|
deinterleave_gint16,
|
|
deinterleave_gint32,
|
|
deinterleave_gfloat,
|
|
deinterleave_gdouble
|
|
};
|
|
|
|
static void
|
|
copy_func (GstAudioResampler * resampler, gpointer sbuf[],
|
|
gpointer in[], gsize in_frames)
|
|
{
|
|
gint c, channels = resampler->channels;
|
|
gsize samples_avail = resampler->samples_avail;
|
|
for (c = 0; c < channels; c++) {
|
|
guint8 *s = ((guint8 *) sbuf[c]) + (samples_avail * resampler->bps);
|
|
if (G_UNLIKELY (in == NULL)) {
|
|
memset (s, 0, in_frames * resampler->bps);
|
|
} else {
|
|
memcpy (s, in[c], in_frames * resampler->bps);
|
|
}
|
|
}
|
|
}
|
|
|
|
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
|
|
alloc_taps_mem (GstAudioResampler * resampler, gint bps, gint n_taps,
|
|
gint n_phases)
|
|
{
|
|
if (resampler->alloc_taps >= n_taps && resampler->alloc_phases >= n_phases)
|
|
return;
|
|
|
|
GST_DEBUG ("allocate bps %d n_taps %d n_phases %d", bps, n_taps, n_phases);
|
|
|
|
resampler->tmp_taps =
|
|
g_realloc_n (resampler->tmp_taps, n_taps, sizeof (gdouble));
|
|
|
|
resampler->taps_stride = GST_ROUND_UP_32 (bps * (n_taps + TAPS_OVERREAD));
|
|
|
|
g_free (resampler->taps_mem);
|
|
resampler->taps_mem =
|
|
g_malloc0 (n_phases * resampler->taps_stride + ALIGN - 1);
|
|
resampler->taps = MEM_ALIGN ((gint8 *) resampler->taps_mem, ALIGN);
|
|
resampler->alloc_taps = n_taps;
|
|
resampler->alloc_phases = n_phases;
|
|
}
|
|
|
|
static void
|
|
alloc_cache_mem (GstAudioResampler * resampler, gint bps, gint n_taps,
|
|
gint n_phases)
|
|
{
|
|
gsize phases_size;
|
|
|
|
resampler->tmp_taps =
|
|
g_realloc_n (resampler->tmp_taps, n_taps, sizeof (gdouble));
|
|
|
|
resampler->cached_taps_stride =
|
|
GST_ROUND_UP_32 (bps * (n_taps + TAPS_OVERREAD));
|
|
|
|
phases_size = sizeof (gpointer) * n_phases;
|
|
|
|
g_free (resampler->cached_taps_mem);
|
|
resampler->cached_taps_mem =
|
|
g_malloc0 (phases_size + n_phases * resampler->cached_taps_stride +
|
|
ALIGN - 1);
|
|
resampler->cached_taps =
|
|
MEM_ALIGN ((gint8 *) resampler->cached_taps_mem + phases_size, ALIGN);
|
|
resampler->cached_phases = resampler->cached_taps_mem;
|
|
}
|
|
|
|
static void
|
|
setup_functions (GstAudioResampler * resampler)
|
|
{
|
|
gint index, fidx;
|
|
|
|
index = resampler->format_index;
|
|
|
|
if (resampler->in_rate == resampler->out_rate)
|
|
resampler->resample = resample_funcs[index];
|
|
else {
|
|
switch (resampler->filter_interpolation) {
|
|
default:
|
|
case GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_NONE:
|
|
fidx = 0;
|
|
break;
|
|
case GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_LINEAR:
|
|
GST_DEBUG ("using linear interpolation for filter coefficients");
|
|
fidx = 0;
|
|
break;
|
|
case GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC:
|
|
GST_DEBUG ("using cubic interpolation for filter coefficients");
|
|
fidx = 4;
|
|
break;
|
|
}
|
|
GST_DEBUG ("using filter interpolate function %d", index + fidx);
|
|
resampler->interpolate = interpolate_funcs[index + fidx];
|
|
|
|
switch (resampler->method) {
|
|
case GST_AUDIO_RESAMPLER_METHOD_NEAREST:
|
|
GST_DEBUG ("using nearest filter function");
|
|
break;
|
|
default:
|
|
index += 4;
|
|
switch (resampler->filter_mode) {
|
|
default:
|
|
case GST_AUDIO_RESAMPLER_FILTER_MODE_FULL:
|
|
GST_DEBUG ("using full filter function");
|
|
break;
|
|
case GST_AUDIO_RESAMPLER_FILTER_MODE_INTERPOLATED:
|
|
index += 4 + fidx;
|
|
GST_DEBUG ("using interpolated filter function");
|
|
break;
|
|
}
|
|
break;
|
|
}
|
|
GST_DEBUG ("using resample function %d", index);
|
|
resampler->resample = resample_funcs[index];
|
|
}
|
|
}
|
|
|
|
static void
|
|
resampler_calculate_taps (GstAudioResampler * resampler)
|
|
{
|
|
gint bps;
|
|
gint n_taps, oversample;
|
|
gint in_rate, out_rate;
|
|
gboolean scale = TRUE, sinc_table = FALSE;
|
|
GstAudioResamplerFilterInterpolation filter_interpolation;
|
|
|
|
switch (resampler->method) {
|
|
case GST_AUDIO_RESAMPLER_METHOD_NEAREST:
|
|
resampler->n_taps = 2;
|
|
scale = FALSE;
|
|
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);
|
|
sinc_table = TRUE;
|
|
break;
|
|
}
|
|
case GST_AUDIO_RESAMPLER_METHOD_KAISER:
|
|
calculate_kaiser_params (resampler);
|
|
sinc_table = TRUE;
|
|
break;
|
|
}
|
|
|
|
in_rate = resampler->in_rate;
|
|
out_rate = resampler->out_rate;
|
|
|
|
if (out_rate < in_rate && scale) {
|
|
resampler->cutoff = resampler->cutoff * out_rate / in_rate;
|
|
resampler->n_taps =
|
|
gst_util_uint64_scale_int (resampler->n_taps, in_rate, out_rate);
|
|
}
|
|
|
|
if (sinc_table) {
|
|
resampler->n_taps = GST_ROUND_UP_8 (resampler->n_taps);
|
|
resampler->filter_mode = GET_OPT_FILTER_MODE (resampler->options);
|
|
resampler->filter_threshold =
|
|
GET_OPT_FILTER_MODE_THRESHOLD (resampler->options);
|
|
filter_interpolation = GET_OPT_FILTER_INTERPOLATION (resampler->options);
|
|
|
|
} else {
|
|
resampler->filter_mode = GST_AUDIO_RESAMPLER_FILTER_MODE_FULL;
|
|
filter_interpolation = GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_NONE;
|
|
}
|
|
|
|
/* calculate oversampling for interpolated filter */
|
|
if (filter_interpolation != GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_NONE) {
|
|
gint mult = 2;
|
|
|
|
oversample = GET_OPT_FILTER_OVERSAMPLE (resampler->options);
|
|
while (oversample > 1) {
|
|
if (mult * out_rate >= in_rate)
|
|
break;
|
|
|
|
mult *= 2;
|
|
oversample >>= 1;
|
|
}
|
|
|
|
switch (filter_interpolation) {
|
|
case GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_LINEAR:
|
|
oversample *= 11;
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
} else {
|
|
oversample = 1;
|
|
}
|
|
resampler->oversample = oversample;
|
|
|
|
n_taps = resampler->n_taps;
|
|
bps = resampler->bps;
|
|
|
|
GST_LOG ("using n_taps %d cutoff %f oversample %d", n_taps, resampler->cutoff,
|
|
oversample);
|
|
|
|
if (resampler->filter_mode == GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO) {
|
|
if (out_rate <= oversample
|
|
&& !(resampler->flags & GST_AUDIO_RESAMPLER_FLAG_VARIABLE_RATE)) {
|
|
/* don't interpolate if we need to calculate at least the same amount
|
|
* of filter coefficients than the full table case */
|
|
resampler->filter_mode = GST_AUDIO_RESAMPLER_FILTER_MODE_FULL;
|
|
GST_DEBUG ("automatically selected full filter, %d <= %d", out_rate,
|
|
oversample);
|
|
} else if (bps * n_taps * out_rate < resampler->filter_threshold) {
|
|
/* switch to full filter when memory is below threshold */
|
|
resampler->filter_mode = GST_AUDIO_RESAMPLER_FILTER_MODE_FULL;
|
|
GST_DEBUG ("automatically selected full filter, memory %d <= %d",
|
|
bps * n_taps * out_rate, resampler->filter_threshold);
|
|
} else {
|
|
GST_DEBUG ("automatically selected interpolated filter");
|
|
resampler->filter_mode = GST_AUDIO_RESAMPLER_FILTER_MODE_INTERPOLATED;
|
|
}
|
|
}
|
|
/* interpolated table but no interpolation given, assume default */
|
|
if (resampler->filter_mode != GST_AUDIO_RESAMPLER_FILTER_MODE_FULL &&
|
|
filter_interpolation == GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_NONE)
|
|
filter_interpolation = DEFAULT_OPT_FILTER_INTERPOLATION;
|
|
|
|
resampler->filter_interpolation = filter_interpolation;
|
|
|
|
if (resampler->filter_mode == GST_AUDIO_RESAMPLER_FILTER_MODE_FULL &&
|
|
resampler->method != GST_AUDIO_RESAMPLER_METHOD_NEAREST) {
|
|
GST_DEBUG ("setting up filter cache");
|
|
resampler->n_phases = out_rate;
|
|
alloc_cache_mem (resampler, bps, n_taps, out_rate);
|
|
}
|
|
|
|
if (resampler->filter_interpolation !=
|
|
GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_NONE) {
|
|
gint i, isize;
|
|
gdouble x;
|
|
gpointer taps;
|
|
|
|
switch (resampler->filter_interpolation) {
|
|
default:
|
|
case GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_LINEAR:
|
|
GST_DEBUG ("using linear interpolation to build filter");
|
|
isize = 2;
|
|
break;
|
|
case GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC:
|
|
GST_DEBUG ("using cubic interpolation to build filter");
|
|
isize = 4;
|
|
break;
|
|
}
|
|
|
|
alloc_taps_mem (resampler, bps, n_taps, oversample + isize);
|
|
|
|
for (i = 0; i < oversample + isize; i++) {
|
|
x = -(n_taps / 2) + i / (gdouble) oversample;
|
|
taps = (gint8 *) resampler->taps + i * resampler->taps_stride;
|
|
make_taps (resampler, taps, x, n_taps);
|
|
}
|
|
}
|
|
}
|
|
|
|
#define PRINT_TAPS(type,print) \
|
|
G_STMT_START { \
|
|
type sum = 0.0, *taps; \
|
|
type icoeff[4]; \
|
|
gint samp_index = 0, samp_phase = i; \
|
|
\
|
|
taps = get_taps_##type##_full (resampler, &samp_index,\
|
|
&samp_phase, icoeff); \
|
|
\
|
|
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;
|
|
|
|
//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, gint in_rate, gint out_rate, GstStructure * options)
|
|
{
|
|
g_return_if_fail (options != NULL);
|
|
g_return_if_fail (quality <= GST_AUDIO_RESAMPLER_QUALITY_MAX);
|
|
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_structure_set (options,
|
|
GST_AUDIO_RESAMPLER_OPT_FILTER_OVERSAMPLE, G_TYPE_INT,
|
|
oversample_qualities[quality], NULL);
|
|
}
|
|
|
|
/**
|
|
* gst_audio_resampler_new:
|
|
* @method: a #GstAudioResamplerMethod
|
|
* @flags: #GstAudioResamplerFlags
|
|
* @format: the #GstAudioFormat
|
|
* @channels: the number of channels
|
|
* @in_rate: input rate
|
|
* @out_rate: output rate
|
|
* @options: extra options
|
|
*
|
|
* Make a new resampler.
|
|
*
|
|
* Returns: (skip) (transfer full): The new #GstAudioResampler.
|
|
*/
|
|
GstAudioResampler *
|
|
gst_audio_resampler_new (GstAudioResamplerMethod method,
|
|
GstAudioResamplerFlags flags,
|
|
GstAudioFormat format, gint channels,
|
|
gint in_rate, gint out_rate, GstStructure * options)
|
|
{
|
|
gboolean non_interleaved_in, non_interleaved_out;
|
|
GstAudioResampler *resampler;
|
|
const GstAudioFormatInfo *info;
|
|
GstStructure *def_options = NULL;
|
|
|
|
g_return_val_if_fail (method >= GST_AUDIO_RESAMPLER_METHOD_NEAREST
|
|
&& method <= GST_AUDIO_RESAMPLER_METHOD_KAISER, NULL);
|
|
g_return_val_if_fail (format == GST_AUDIO_FORMAT_S16 ||
|
|
format == GST_AUDIO_FORMAT_S32 || format == GST_AUDIO_FORMAT_F32 ||
|
|
format == GST_AUDIO_FORMAT_F64, NULL);
|
|
g_return_val_if_fail (channels > 0, NULL);
|
|
g_return_val_if_fail (in_rate > 0, NULL);
|
|
g_return_val_if_fail (out_rate > 0, NULL);
|
|
|
|
audio_resampler_init ();
|
|
|
|
resampler = g_new0 (GstAudioResampler, 1);
|
|
resampler->method = method;
|
|
resampler->flags = flags;
|
|
resampler->format = format;
|
|
resampler->channels = channels;
|
|
|
|
switch (format) {
|
|
case GST_AUDIO_FORMAT_S16:
|
|
resampler->format_index = 0;
|
|
break;
|
|
case GST_AUDIO_FORMAT_S32:
|
|
resampler->format_index = 1;
|
|
break;
|
|
case GST_AUDIO_FORMAT_F32:
|
|
resampler->format_index = 2;
|
|
break;
|
|
case GST_AUDIO_FORMAT_F64:
|
|
resampler->format_index = 3;
|
|
break;
|
|
default:
|
|
g_assert_not_reached ();
|
|
break;
|
|
}
|
|
|
|
info = gst_audio_format_get_info (format);
|
|
resampler->bps = GST_AUDIO_FORMAT_INFO_WIDTH (info) / 8;
|
|
resampler->sbuf = g_malloc0 (sizeof (gpointer) * channels);
|
|
|
|
non_interleaved_in =
|
|
(resampler->flags & GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED_IN);
|
|
non_interleaved_out =
|
|
(resampler->flags & GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED_OUT);
|
|
|
|
/* we resample each channel separately */
|
|
resampler->blocks = resampler->channels;
|
|
resampler->inc = 1;
|
|
resampler->ostride = non_interleaved_out ? 1 : resampler->channels;
|
|
resampler->deinterleave = non_interleaved_in ?
|
|
copy_func : deinterleave_funcs[resampler->format_index];
|
|
resampler->convert_taps = convert_taps_funcs[resampler->format_index];
|
|
|
|
GST_DEBUG ("method %d, bps %d, channels %d", method, resampler->bps,
|
|
resampler->channels);
|
|
|
|
if (options == NULL) {
|
|
options = def_options =
|
|
gst_structure_new_empty ("GstAudioResampler.options");
|
|
gst_audio_resampler_options_set_quality (DEFAULT_RESAMPLER_METHOD,
|
|
GST_AUDIO_RESAMPLER_QUALITY_DEFAULT, in_rate, out_rate, options);
|
|
}
|
|
|
|
gst_audio_resampler_update (resampler, in_rate, out_rate, options);
|
|
gst_audio_resampler_reset (resampler);
|
|
|
|
if (def_options)
|
|
gst_structure_free (def_options);
|
|
|
|
return resampler;
|
|
}
|
|
|
|
/* 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)) {
|
|
gint c, blocks = resampler->blocks;
|
|
gsize bytes, to_move = 0;
|
|
gint8 *ptr, *samples;
|
|
|
|
GST_LOG ("realloc %d -> %d", (gint) resampler->samples_len, (gint) need);
|
|
|
|
bytes = GST_ROUND_UP_N (need * resampler->bps * resampler->inc, ALIGN);
|
|
|
|
samples = g_malloc0 (blocks * bytes + ALIGN - 1);
|
|
ptr = MEM_ALIGN (samples, ALIGN);
|
|
|
|
/* if we had some data, move history */
|
|
if (resampler->samples_len > 0)
|
|
to_move = resampler->samples_avail * resampler->bps * resampler->inc;
|
|
|
|
/* set up new pointers */
|
|
for (c = 0; c < blocks; c++) {
|
|
memcpy (ptr + (c * bytes), resampler->sbuf[c], to_move);
|
|
resampler->sbuf[c] = ptr + (c * bytes);
|
|
}
|
|
g_free (resampler->samples);
|
|
resampler->samples = samples;
|
|
resampler->samples_len = need;
|
|
}
|
|
return resampler->sbuf;
|
|
}
|
|
|
|
/**
|
|
* gst_audio_resampler_reset:
|
|
* @resampler: a #GstAudioResampler
|
|
*
|
|
* Reset @resampler to the state it was when it was first created, discarding
|
|
* all sample history.
|
|
*/
|
|
void
|
|
gst_audio_resampler_reset (GstAudioResampler * resampler)
|
|
{
|
|
g_return_if_fail (resampler != NULL);
|
|
|
|
if (resampler->samples) {
|
|
gsize bytes;
|
|
gint c, blocks, bpf;
|
|
|
|
bpf = resampler->bps * resampler->inc;
|
|
bytes = (resampler->n_taps / 2) * bpf;
|
|
blocks = resampler->blocks;
|
|
|
|
for (c = 0; c < blocks; c++)
|
|
memset (resampler->sbuf[c], 0, bytes);
|
|
}
|
|
/* half of the filter is filled with 0 */
|
|
resampler->samp_index = 0;
|
|
resampler->samples_avail = resampler->n_taps / 2 - 1;
|
|
}
|
|
|
|
/**
|
|
* 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.
|
|
*
|
|
* When @options is %NULL, the previously configured options are reused.
|
|
*
|
|
* Returns: %TRUE if the new parameters could be set
|
|
*/
|
|
gboolean
|
|
gst_audio_resampler_update (GstAudioResampler * resampler,
|
|
gint in_rate, gint out_rate, GstStructure * options)
|
|
{
|
|
gint gcd, samp_phase, old_n_taps;
|
|
gdouble max_error;
|
|
|
|
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;
|
|
|
|
if (resampler->out_rate > 0) {
|
|
GST_INFO ("old phase %d/%d", resampler->samp_phase, resampler->out_rate);
|
|
samp_phase =
|
|
gst_util_uint64_scale_int (resampler->samp_phase, out_rate,
|
|
resampler->out_rate);
|
|
} else
|
|
samp_phase = 0;
|
|
|
|
gcd = gst_util_greatest_common_divisor (in_rate, out_rate);
|
|
|
|
max_error = GET_OPT_MAX_PHASE_ERROR (resampler->options);
|
|
|
|
if (max_error < 1.0e-8) {
|
|
GST_INFO ("using exact phase divider");
|
|
gcd = gst_util_greatest_common_divisor (gcd, samp_phase);
|
|
} else {
|
|
while (gcd > 1) {
|
|
gdouble ph1 = (gdouble) samp_phase / out_rate;
|
|
gint factor = 2;
|
|
|
|
/* reduce the factor until we have a phase error of less than 10% */
|
|
gdouble ph2 = (gdouble) (samp_phase / gcd) / (out_rate / gcd);
|
|
|
|
if (fabs (ph1 - ph2) < max_error)
|
|
break;
|
|
|
|
while (gcd % factor != 0)
|
|
factor++;
|
|
gcd /= factor;
|
|
|
|
GST_INFO ("divide by factor %d, gcd %d", factor, gcd);
|
|
}
|
|
}
|
|
|
|
GST_INFO ("phase %d out_rate %d, in_rate %d, gcd %d", samp_phase, out_rate,
|
|
in_rate, gcd);
|
|
|
|
resampler->samp_phase = samp_phase /= gcd;
|
|
resampler->in_rate = in_rate /= gcd;
|
|
resampler->out_rate = out_rate /= gcd;
|
|
|
|
GST_INFO ("new phase %d/%d", resampler->samp_phase, resampler->out_rate);
|
|
|
|
resampler->samp_inc = in_rate / out_rate;
|
|
resampler->samp_frac = in_rate % out_rate;
|
|
|
|
if (options) {
|
|
GST_INFO ("have new options, reconfigure filter");
|
|
|
|
if (resampler->options)
|
|
gst_structure_free (resampler->options);
|
|
resampler->options = gst_structure_copy (options);
|
|
|
|
old_n_taps = resampler->n_taps;
|
|
|
|
resampler_calculate_taps (resampler);
|
|
resampler_dump (resampler);
|
|
|
|
if (old_n_taps > 0 && old_n_taps != resampler->n_taps) {
|
|
gpointer *sbuf;
|
|
gint i, bpf, bytes, soff, doff, diff;
|
|
|
|
sbuf = get_sample_bufs (resampler, resampler->n_taps);
|
|
|
|
bpf = resampler->bps * resampler->inc;
|
|
bytes = resampler->samples_avail * bpf;
|
|
soff = doff = resampler->samp_index * bpf;
|
|
|
|
diff = ((gint) resampler->n_taps - old_n_taps) / 2;
|
|
|
|
GST_DEBUG ("taps %d->%d, %d", old_n_taps, resampler->n_taps, diff);
|
|
|
|
if (diff < 0) {
|
|
/* diff < 0, decrease taps, adjust source */
|
|
soff += -diff * bpf;
|
|
bytes -= -diff * bpf;
|
|
} else {
|
|
/* diff > 0, increase taps, adjust dest */
|
|
doff += diff * bpf;
|
|
}
|
|
|
|
/* now shrink or enlarge the history buffer, when we enlarge we
|
|
* just leave the old samples in there. FIXME, probably do something better
|
|
* like mirror or fill with zeroes. */
|
|
for (i = 0; i < resampler->blocks; i++)
|
|
memmove ((gint8 *) sbuf[i] + doff, (gint8 *) sbuf[i] + soff, bytes);
|
|
|
|
resampler->samples_avail += diff;
|
|
}
|
|
} else if (resampler->filter_mode == GST_AUDIO_RESAMPLER_FILTER_MODE_FULL) {
|
|
GST_DEBUG ("setting up filter cache");
|
|
resampler->n_phases = resampler->out_rate;
|
|
alloc_cache_mem (resampler, resampler->bps, resampler->n_taps,
|
|
resampler->n_phases);
|
|
}
|
|
setup_functions (resampler);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/**
|
|
* gst_audio_resampler_free:
|
|
* @resampler: a #GstAudioResampler
|
|
*
|
|
* Free a previously allocated #GstAudioResampler @resampler.
|
|
*/
|
|
void
|
|
gst_audio_resampler_free (GstAudioResampler * resampler)
|
|
{
|
|
g_return_if_fail (resampler != NULL);
|
|
|
|
g_free (resampler->cached_taps_mem);
|
|
g_free (resampler->taps_mem);
|
|
g_free (resampler->tmp_taps);
|
|
g_free (resampler->samples);
|
|
g_free (resampler->sbuf);
|
|
if (resampler->options)
|
|
gst_structure_free (resampler->options);
|
|
g_free (resampler);
|
|
}
|
|
|
|
/**
|
|
* 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 available after giving
|
|
* @in_frames as input to @resampler.
|
|
*/
|
|
gsize
|
|
gst_audio_resampler_get_out_frames (GstAudioResampler * resampler,
|
|
gsize in_frames)
|
|
{
|
|
gsize need, avail, out;
|
|
|
|
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) {
|
|
GST_LOG ("avail %d < need %d", (int) avail, (int) need);
|
|
return 0;
|
|
}
|
|
|
|
out = (avail - need) * resampler->out_rate;
|
|
if (out < resampler->samp_phase) {
|
|
GST_LOG ("out %d < samp_phase %d", (int) out, (int) resampler->samp_phase);
|
|
return 0;
|
|
}
|
|
|
|
out = ((out - resampler->samp_phase) / resampler->in_rate) + 1;
|
|
GST_LOG ("out %d = ((%d * %d - %d) / %d) + 1", (gint) out,
|
|
(gint) (avail - need), resampler->out_rate, resampler->samp_phase,
|
|
resampler->in_rate);
|
|
|
|
return out;
|
|
}
|
|
|
|
/**
|
|
* 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;
|
|
}
|
|
|
|
/**
|
|
* gst_audio_resampler_resample:
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* @resampler: a #GstAudioResampler
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* @in: input samples
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* @in_frames: number of input frames
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* @out: output samples
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* @out_frames: number of output frames
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*
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* Perform resampling on @in_frames frames in @in and write @out_frames to @out.
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*
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* In case the samples are interleaved, @in and @out must point to an
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* array with a single element pointing to a block of interleaved samples.
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*
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* If non-interleaved samples are used, @in and @out must point to an
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* array with pointers to memory blocks, one for each channel.
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*
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* @in may be %NULL, in which case @in_frames of silence samples are pushed
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* into the resampler.
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*
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* This function always produces @out_frames of output and consumes @in_frames of
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* input. Use gst_audio_resampler_get_out_frames() and
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* gst_audio_resampler_get_in_frames() to make sure @in_frames and @out_frames
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* are matching and @in and @out point to enough memory.
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*/
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void
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gst_audio_resampler_resample (GstAudioResampler * resampler,
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gpointer in[], gsize in_frames, gpointer out[], gsize out_frames)
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{
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gsize samples_avail;
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gsize need, consumed;
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gpointer *sbuf;
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/* do sample skipping */
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if (G_UNLIKELY (resampler->skip >= in_frames)) {
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/* we need tp skip all input */
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resampler->skip -= in_frames;
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return;
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}
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/* skip the last samples by advancing the sample index */
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resampler->samp_index += resampler->skip;
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samples_avail = resampler->samples_avail;
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/* make sure we have enough space to copy our samples */
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sbuf = get_sample_bufs (resampler, in_frames + samples_avail);
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/* copy/deinterleave the samples */
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resampler->deinterleave (resampler, sbuf, in, in_frames);
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/* update new amount of samples in our buffer */
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resampler->samples_avail = samples_avail += in_frames;
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need = resampler->n_taps + resampler->samp_index;
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if (G_UNLIKELY (samples_avail < need || out_frames == 0)) {
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GST_LOG ("not enough samples to start: need %" G_GSIZE_FORMAT ", avail %"
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G_GSIZE_FORMAT ", out %" G_GSIZE_FORMAT, need, samples_avail,
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out_frames);
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/* not enough samples to start */
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return;
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}
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/* resample all channels */
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resampler->resample (resampler, sbuf, samples_avail, out, out_frames,
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&consumed);
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GST_LOG ("in %" G_GSIZE_FORMAT ", avail %" G_GSIZE_FORMAT ", consumed %"
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G_GSIZE_FORMAT, in_frames, samples_avail, consumed);
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/* update pointers */
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if (G_LIKELY (consumed > 0)) {
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gssize left = samples_avail - consumed;
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if (left > 0) {
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/* we consumed part of our samples */
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resampler->samples_avail = left;
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} else {
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/* we consumed all our samples, empty our buffers */
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resampler->samples_avail = 0;
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resampler->skip = -left;
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}
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}
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}
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