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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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524ea147cc
Remove some unused variables from the inner product functions. Make filter coefficients by interpolating if required. Rename some fields. Try hard to not recalculate filters when just chaging the rate. Add more proprties to audioresample.
1821 lines
64 KiB
C
1821 lines
64 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 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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gint channels;
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gint in_rate;
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gint out_rate;
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gint bps;
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gint ostride;
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GstAudioResamplerFilterMode filter_mode;
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guint filter_threshold;
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GstAudioResamplerFilterInterpolation filter_interpolation;
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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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/* temp taps */
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gpointer tmp_taps;
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/* oversampled main filter table */
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gint oversample;
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guint n_taps;
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gpointer taps;
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gpointer taps_mem;
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gsize taps_stride;
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gint n_phases;
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guint alloc_taps;
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guint alloc_phases;
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/* cached taps */
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gpointer *cached_phases;
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gpointer cached_taps;
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gpointer cached_taps_mem;
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gsize cached_taps_stride;
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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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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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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_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_nearest_tap (gdouble x, gint n_taps)
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{
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gdouble a = fabs (x), res;;
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if (a < 0.5)
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res = 1.0;
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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_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 PRECISION_S16 15
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#define PRECISION_S32 31
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static inline gdouble
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make_taps (GstAudioResampler * resampler,
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gdouble * tmp_taps, gdouble x, gint n_taps, gint oversample)
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{
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gdouble weight = 0.0;
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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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for (i = 0; i < n_taps; i++)
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weight += tmp_taps[i] = get_nearest_tap (x + i, resampler->n_taps);
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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 / (double) oversample,
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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 / (double) oversample, resampler->n_taps,
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resampler->cutoff, resampler->kaiser_beta);
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break;
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default:
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break;
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}
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return weight;
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}
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#define MAKE_CONVERT_TAPS_INT_FUNC(type, precision) \
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static inline void \
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convert_taps_##type (gdouble *tmp_taps, type *taps, \
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gdouble weight, gint n_taps) \
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{ \
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gint64 one = (1L << precision) - 1; \
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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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taps[j] = floor (offset + tmp_taps[j] * multiplier / weight); \
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if (!exact) \
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GST_WARNING ("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 inline void \
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convert_taps_##type (gdouble *tmp_taps, type *taps, \
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gdouble weight, gint n_taps) \
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{ \
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gint i; \
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for (i = 0; i < n_taps; i++) \
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taps[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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#define MAKE_EXTRACT_TAPS_FUNC(type) \
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static inline void \
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extract_taps_##type (GstAudioResampler * resampler, type *tmp_taps, \
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gint n_taps, gint oversample, gint mult) \
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{ \
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gint i, j, k; \
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for (i = 0; i < oversample; i++) { \
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type *taps = (type *) ((gint8*)resampler->taps + \
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i * resampler->taps_stride); \
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for (j = 0; j < n_taps; j++) { \
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for (k = 0; k < mult; k++) { \
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*taps++ = tmp_taps[i + j*oversample + k]; \
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} \
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} \
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} \
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}
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MAKE_EXTRACT_TAPS_FUNC (gint16);
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MAKE_EXTRACT_TAPS_FUNC (gint32);
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MAKE_EXTRACT_TAPS_FUNC (gfloat);
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MAKE_EXTRACT_TAPS_FUNC (gdouble);
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typedef void (*InterpolateFunc) (gdouble * o, const gdouble * a, gint len,
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const gdouble * icoeff);
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static void
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interpolate_gdouble_linear_c (gdouble * o, const gdouble * a, gint len,
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const gdouble * ic)
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{
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gint i;
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for (i = 0; i < len; i++)
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o[i] = (a[2 * i + 0] - a[2 * i + 1]) * ic[0] + a[2 * i + 1];
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}
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static void
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interpolate_gdouble_cubic_c (gdouble * o, const gdouble * a, gint len,
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const gdouble * ic)
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{
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gint i;
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for (i = 0; i < len; i++)
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o[i] =
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a[4 * i + 0] * ic[0] + a[4 * i + 1] * ic[1] + a[4 * i + 2] * ic[2] +
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a[4 * i + 3] * ic[3];
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}
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static InterpolateFunc interpolate_funcs[] = {
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interpolate_gdouble_linear_c,
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interpolate_gdouble_cubic_c
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};
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#define interpolate_gdouble_linear interpolate_funcs[0]
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#define interpolate_gdouble_cubic interpolate_funcs[1]
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#define MAKE_COEFF_LINEAR_INT_FUNC(type,prec) \
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static inline void \
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make_coeff_##type##_linear (gint frac, gint out_rate, type *icoeff) \
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{ \
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type x = ((gint64)frac << prec) / out_rate; \
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icoeff[0] = icoeff[2] = x; \
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icoeff[1] = icoeff[3] = (1L << 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 \
|
|
make_coeff_##type##_linear (gint frac, gint out_rate, type *icoeff) \
|
|
{ \
|
|
type x = (type)frac / out_rate; \
|
|
icoeff[0] = icoeff[2] = x; \
|
|
icoeff[1] = icoeff[3] = 1.0 - x; \
|
|
}
|
|
MAKE_COEFF_LINEAR_INT_FUNC (gint16, PRECISION_S16);
|
|
MAKE_COEFF_LINEAR_INT_FUNC (gint32, PRECISION_S32);
|
|
MAKE_COEFF_LINEAR_FLOAT_FUNC (gfloat);
|
|
MAKE_COEFF_LINEAR_FLOAT_FUNC (gdouble);
|
|
|
|
#define MAKE_COEFF_CUBIC_INT_FUNC(type,type2,prec) \
|
|
static inline void \
|
|
make_coeff_##type##_cubic (gint frac, gint out_rate, type *icoeff) \
|
|
{ \
|
|
type2 one = (1L << prec) - 1; \
|
|
type2 x = ((gint64) frac << prec) / out_rate; \
|
|
type2 x2 = (x * x) >> prec; \
|
|
type2 x3 = (x2 * x) >> prec; \
|
|
icoeff[0] = (((x3 - x) << prec) / 6) >> prec; \
|
|
icoeff[1] = x + ((x2 - x3) >> 1); \
|
|
icoeff[3] = -(((x << prec) / 3) >> prec) + \
|
|
(x2 >> 1) - (((x3 << prec) / 6) >> prec); \
|
|
icoeff[2] = one - icoeff[0] - icoeff[1] - icoeff[3]; \
|
|
}
|
|
#define MAKE_COEFF_CUBIC_FLOAT_FUNC(type) \
|
|
static inline void \
|
|
make_coeff_##type##_cubic (gint frac, gint out_rate, type *icoeff) \
|
|
{ \
|
|
type x = (type) frac / out_rate, x2 = x * x, x3 = x2 * x; \
|
|
icoeff[0] = 0.16667f * (x3 - x); \
|
|
icoeff[1] = x + 0.5f * (x2 - x3); \
|
|
icoeff[3] = -0.33333f * x + 0.5f * x2 - 0.16667f * x3; \
|
|
icoeff[2] = 1. - icoeff[0] - icoeff[1] - icoeff[3]; \
|
|
}
|
|
MAKE_COEFF_CUBIC_INT_FUNC (gint16, gint32, PRECISION_S16);
|
|
MAKE_COEFF_CUBIC_INT_FUNC (gint32, gint64, PRECISION_S32);
|
|
MAKE_COEFF_CUBIC_FLOAT_FUNC (gfloat);
|
|
MAKE_COEFF_CUBIC_FLOAT_FUNC (gdouble);
|
|
|
|
static inline gdouble
|
|
fill_taps (GstAudioResampler * resampler,
|
|
gdouble * tmp_taps, gint phase, gint n_phases, gint n_taps)
|
|
{
|
|
gdouble res;
|
|
|
|
if (resampler->filter_interpolation ==
|
|
GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_NONE) {
|
|
gdouble x = 1.0 - n_taps / 2 - (gdouble) phase / n_phases;
|
|
res = make_taps (resampler, tmp_taps, x, n_taps, 1);
|
|
} else {
|
|
gint out_rate = resampler->out_rate;
|
|
gint offset, pos, frac;
|
|
gint oversample = resampler->oversample;
|
|
gint taps_stride = resampler->taps_stride;
|
|
gdouble ic[4], *taps;
|
|
|
|
pos = phase * oversample;
|
|
offset = (oversample - 1) - (pos / out_rate);
|
|
frac = pos % out_rate;
|
|
|
|
taps = (gdouble *) ((gint8 *) resampler->taps + offset * taps_stride);
|
|
|
|
switch (resampler->filter_interpolation) {
|
|
case GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_LINEAR:
|
|
make_coeff_gdouble_linear (frac, out_rate, ic);
|
|
interpolate_gdouble_linear (tmp_taps, taps, n_taps, ic);
|
|
break;
|
|
case GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC:
|
|
make_coeff_gdouble_cubic (frac, out_rate, ic);
|
|
interpolate_gdouble_cubic (tmp_taps, taps, n_taps, ic);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
res = 1.0;
|
|
}
|
|
return res;
|
|
}
|
|
|
|
#define GET_TAPS_NONE_FUNC(type) \
|
|
static inline gpointer \
|
|
get_taps_##type##_none (GstAudioResampler * resampler, \
|
|
gint *samp_index, gint *samp_phase, type icoeff[4]) \
|
|
{ \
|
|
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)) { \
|
|
gdouble weight; \
|
|
gdouble *tmp_taps = resampler->tmp_taps; \
|
|
gint n_taps = resampler->n_taps; \
|
|
\
|
|
res = (gint8 *) resampler->cached_taps + \
|
|
phase * resampler->cached_taps_stride; \
|
|
\
|
|
weight = fill_taps (resampler, tmp_taps, phase, n_phases, n_taps); \
|
|
convert_taps_##type (tmp_taps, res, weight, n_taps); \
|
|
\
|
|
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_NONE_FUNC (gint16);
|
|
GET_TAPS_NONE_FUNC (gint32);
|
|
GET_TAPS_NONE_FUNC (gfloat);
|
|
GET_TAPS_NONE_FUNC (gdouble);
|
|
|
|
#define GET_TAPS_INTERPOLATE_FUNC(type,inter) \
|
|
static inline gpointer \
|
|
get_taps_##type##_##inter (GstAudioResampler * resampler, \
|
|
gint *samp_index, gint *samp_phase, type icoeff[4]) \
|
|
{ \
|
|
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_INT_NONE_FUNC(type,type2,prec,limit) \
|
|
static inline void \
|
|
inner_product_##type##_none_1_c (type * o, const type * a, \
|
|
const type * b, gint len, const type *ic) \
|
|
{ \
|
|
gint i; \
|
|
type2 res = 0; \
|
|
\
|
|
for (i = 0; i < len; i++) \
|
|
res += (type2) a[i] * (type2) b[i]; \
|
|
\
|
|
res = (res + (1L << ((prec) - 1))) >> (prec); \
|
|
*o = CLAMP (res, -(limit), (limit) - 1); \
|
|
}
|
|
|
|
INNER_PRODUCT_INT_NONE_FUNC (gint16, gint32, PRECISION_S16, 1L << 15);
|
|
INNER_PRODUCT_INT_NONE_FUNC (gint32, gint64, PRECISION_S32, 1L << 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 i; \
|
|
type2 res[2] = { 0, 0 }; \
|
|
\
|
|
for (i = 0; i < len; i++) { \
|
|
res[0] += (type2) a[i] * (type2) b[2 * i + 0]; \
|
|
res[1] += (type2) a[i] * (type2) b[2 * i + 1]; \
|
|
} \
|
|
res[0] = (res[0] >> (prec)) * (type2) ic[0] + \
|
|
(res[1] >> (prec)) * (type2) ic[1]; \
|
|
res[0] = (res[0] + (1L << ((prec) - 1))) >> (prec); \
|
|
*o = CLAMP (res[0], -(limit), (limit) - 1); \
|
|
}
|
|
|
|
INNER_PRODUCT_INT_LINEAR_FUNC (gint16, gint32, PRECISION_S16, 1L << 15);
|
|
INNER_PRODUCT_INT_LINEAR_FUNC (gint32, gint64, PRECISION_S32, 1L << 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 i; \
|
|
type2 res[4] = { 0, 0, 0, 0 }; \
|
|
\
|
|
for (i = 0; i < len; i++) { \
|
|
res[0] += (type2) a[i] * (type2) b[4 * i + 0]; \
|
|
res[1] += (type2) a[i] * (type2) b[4 * i + 1]; \
|
|
res[2] += (type2) a[i] * (type2) b[4 * i + 2]; \
|
|
res[3] += (type2) a[i] * (type2) b[4 * i + 3]; \
|
|
} \
|
|
res[0] = (res[0] >> (prec)) * (type2) ic[0] + \
|
|
(res[1] >> (prec)) * (type2) ic[1] + \
|
|
(res[2] >> (prec)) * (type2) ic[2] + \
|
|
(res[3] >> (prec)) * (type2) ic[3]; \
|
|
res[0] = (res[0] + (1L << ((prec) - 1))) >> (prec); \
|
|
*o = CLAMP (res[0], -(limit), (limit) - 1); \
|
|
}
|
|
|
|
INNER_PRODUCT_INT_CUBIC_FUNC (gint16, gint32, PRECISION_S16, 1L << 15);
|
|
INNER_PRODUCT_INT_CUBIC_FUNC (gint32, gint64, PRECISION_S32, 1L << 31);
|
|
|
|
#define INNER_PRODUCT_FLOAT_NONE_FUNC(type) \
|
|
static inline void \
|
|
inner_product_##type##_none_1_c (type * o, const type * a, \
|
|
const type * b, gint len, const type *ic) \
|
|
{ \
|
|
gint i; \
|
|
type res = 0.0; \
|
|
\
|
|
for (i = 0; i < len; i++) \
|
|
res += a[i] * b[i]; \
|
|
\
|
|
*o = res; \
|
|
}
|
|
|
|
INNER_PRODUCT_FLOAT_NONE_FUNC (gfloat);
|
|
INNER_PRODUCT_FLOAT_NONE_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 i; \
|
|
type res[2] = { 0.0, 0.0 }; \
|
|
\
|
|
for (i = 0; i < len; i++) { \
|
|
res[0] += a[i] * b[2 * i + 0]; \
|
|
res[1] += a[i] * b[2 * i + 1]; \
|
|
} \
|
|
*o = res[0] * ic[0] + res[1] * ic[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 i; \
|
|
type res[4] = { 0.0, 0.0, 0.0, 0.0 }; \
|
|
\
|
|
for (i = 0; i < len; i++) { \
|
|
res[0] += a[i] * b[4 * i + 0]; \
|
|
res[1] += a[i] * b[4 * i + 1]; \
|
|
res[2] += a[i] * b[4 * i + 2]; \
|
|
res[3] += a[i] * b[4 * i + 3]; \
|
|
} \
|
|
*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);
|
|
|
|
#define MAKE_RESAMPLE_FUNC(type,inter,channels,arch) \
|
|
static void \
|
|
resample_ ##type## _ ##inter## _ ##channels## _ ##arch (GstAudioResampler * resampler, \
|
|
gpointer in[], gsize in_len, gpointer out[], gsize out_len, \
|
|
gsize * consumed) \
|
|
{ \
|
|
gint c, di = 0; \
|
|
gint n_taps = resampler->n_taps; \
|
|
gint blocks = resampler->blocks; \
|
|
gint ostride = resampler->ostride; \
|
|
gint samp_index = 0; \
|
|
gint samp_phase = 0; \
|
|
\
|
|
for (c = 0; c < blocks; c++) { \
|
|
type *ip = in[c]; \
|
|
type *op = ostride == 1 ? out[c] : (type *)out[0] + c; \
|
|
\
|
|
samp_index = resampler->samp_index; \
|
|
samp_phase = resampler->samp_phase; \
|
|
\
|
|
for (di = 0; di < out_len; di++) { \
|
|
type *ipp, icoeff[4], *taps; \
|
|
\
|
|
ipp = &ip[samp_index * channels]; \
|
|
\
|
|
taps = get_taps_ ##type##_##inter \
|
|
(resampler, &samp_index, &samp_phase, icoeff); \
|
|
inner_product_ ##type##_##inter##_##channels##_##arch \
|
|
(op, ipp, taps, n_taps, icoeff); \
|
|
op += ostride; \
|
|
} \
|
|
if (in_len > samp_index) \
|
|
memmove (ip, &ip[samp_index * channels], \
|
|
(in_len - samp_index) * sizeof(type) * channels); \
|
|
} \
|
|
*consumed = samp_index - resampler->samp_index; \
|
|
\
|
|
resampler->samp_index = 0; \
|
|
resampler->samp_phase = samp_phase; \
|
|
}
|
|
|
|
MAKE_RESAMPLE_FUNC (gint16, none, 1, c);
|
|
MAKE_RESAMPLE_FUNC (gint32, none, 1, c);
|
|
MAKE_RESAMPLE_FUNC (gfloat, none, 1, c);
|
|
MAKE_RESAMPLE_FUNC (gdouble, none, 1, c);
|
|
|
|
MAKE_RESAMPLE_FUNC (gint16, linear, 1, c);
|
|
MAKE_RESAMPLE_FUNC (gint32, linear, 1, c);
|
|
MAKE_RESAMPLE_FUNC (gfloat, linear, 1, c);
|
|
MAKE_RESAMPLE_FUNC (gdouble, linear, 1, c);
|
|
|
|
MAKE_RESAMPLE_FUNC (gint16, cubic, 1, c);
|
|
MAKE_RESAMPLE_FUNC (gint32, cubic, 1, c);
|
|
MAKE_RESAMPLE_FUNC (gfloat, cubic, 1, c);
|
|
MAKE_RESAMPLE_FUNC (gdouble, cubic, 1, c);
|
|
|
|
static ResampleFunc resample_funcs[] = {
|
|
resample_gint16_none_1_c,
|
|
resample_gint32_none_1_c,
|
|
resample_gfloat_none_1_c,
|
|
resample_gdouble_none_1_c,
|
|
NULL,
|
|
NULL,
|
|
NULL,
|
|
NULL,
|
|
|
|
resample_gint16_linear_1_c,
|
|
resample_gint32_linear_1_c,
|
|
resample_gfloat_linear_1_c,
|
|
resample_gdouble_linear_1_c,
|
|
NULL,
|
|
NULL,
|
|
NULL,
|
|
NULL,
|
|
|
|
resample_gint16_cubic_1_c,
|
|
resample_gint32_cubic_1_c,
|
|
resample_gfloat_cubic_1_c,
|
|
resample_gdouble_cubic_1_c,
|
|
NULL,
|
|
NULL,
|
|
NULL,
|
|
NULL,
|
|
};
|
|
|
|
#define resample_gint16_none_1 resample_funcs[0]
|
|
#define resample_gint32_none_1 resample_funcs[1]
|
|
#define resample_gfloat_none_1 resample_funcs[2]
|
|
#define resample_gdouble_none_1 resample_funcs[3]
|
|
#define resample_gint16_none_2 resample_funcs[4]
|
|
#define resample_gint32_none_2 resample_funcs[5]
|
|
#define resample_gfloat_none_2 resample_funcs[6]
|
|
#define resample_gdouble_none_2 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[16]
|
|
#define resample_gint32_cubic_1 resample_funcs[17]
|
|
#define resample_gfloat_cubic_1 resample_funcs[18]
|
|
#define resample_gdouble_cubic_1 resample_funcs[19]
|
|
|
|
#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 (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
|
|
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
|
|
alloc_taps_mem (GstAudioResampler * resampler, gint bps, gint n_taps,
|
|
gint n_phases, gint n_mult)
|
|
{
|
|
if (resampler->alloc_taps >= n_taps && resampler->alloc_phases >= n_phases)
|
|
return;
|
|
|
|
GST_DEBUG ("allocate n_taps %d n_phases %d n_mult %d", n_taps, n_phases,
|
|
n_mult);
|
|
|
|
resampler->tmp_taps =
|
|
g_realloc_n (resampler->tmp_taps, n_taps, sizeof (gdouble));
|
|
|
|
resampler->taps_stride =
|
|
GST_ROUND_UP_32 (bps * (n_mult * 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)
|
|
{
|
|
gboolean non_interleaved;
|
|
gint n_taps, index;
|
|
DeinterleaveFunc deinterleave;
|
|
ResampleFunc resample, resample_2;
|
|
|
|
n_taps = resampler->n_taps;
|
|
non_interleaved =
|
|
(resampler->flags & GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED);
|
|
|
|
resampler->ostride = non_interleaved ? 1 : resampler->channels;
|
|
|
|
switch (resampler->format) {
|
|
case GST_AUDIO_FORMAT_S16:
|
|
GST_DEBUG ("using S16 functions");
|
|
index = 0;
|
|
break;
|
|
case GST_AUDIO_FORMAT_S32:
|
|
GST_DEBUG ("using S32 functions");
|
|
index = 1;
|
|
break;
|
|
case GST_AUDIO_FORMAT_F32:
|
|
GST_DEBUG ("using F32 functions");
|
|
index = 2;
|
|
break;
|
|
case GST_AUDIO_FORMAT_F64:
|
|
GST_DEBUG ("using F64 functions");
|
|
index = 3;
|
|
break;
|
|
default:
|
|
g_assert_not_reached ();
|
|
break;
|
|
}
|
|
deinterleave = deinterleave_funcs[index];
|
|
|
|
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:
|
|
switch (resampler->filter_interpolation) {
|
|
case GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_LINEAR:
|
|
GST_DEBUG ("using linear interpolation filter function");
|
|
index += 8;
|
|
break;
|
|
case GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC:
|
|
GST_DEBUG ("using cubic interpolation filter function");
|
|
index += 16;
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
break;
|
|
}
|
|
resample = resample_funcs[index];
|
|
resample_2 = resample_funcs[index + 4];
|
|
|
|
if (!non_interleaved && resampler->channels == 2 && n_taps >= 4 && resample_2) {
|
|
/* we resample 2 channels in parallel */
|
|
resampler->resample = resample_2;
|
|
resampler->deinterleave = deinterleave_copy;
|
|
resampler->blocks = 1;
|
|
resampler->inc = resampler->channels;;
|
|
GST_DEBUG ("resample 2 channels at a time");
|
|
} else {
|
|
/* we resample each channel separately */
|
|
resampler->resample = resample;
|
|
resampler->deinterleave = deinterleave;
|
|
resampler->blocks = resampler->channels;
|
|
resampler->inc = 1;
|
|
GST_DEBUG ("resample 1 channel at a time");
|
|
}
|
|
}
|
|
|
|
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);
|
|
|
|
/* 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;
|
|
} 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 <<= 5;
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
} else {
|
|
oversample = 1;
|
|
}
|
|
resampler->oversample = oversample;
|
|
resampler->filter_interpolation = filter_interpolation;
|
|
|
|
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) {
|
|
/* 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 {
|
|
GST_DEBUG ("automatically selected interpolated filter");
|
|
resampler->filter_mode = GST_AUDIO_RESAMPLER_FILTER_MODE_INTERPOLATED;
|
|
}
|
|
}
|
|
|
|
if (resampler->filter_mode == GST_AUDIO_RESAMPLER_FILTER_MODE_FULL) {
|
|
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 otaps, isize;
|
|
gdouble x, weight, *tmp_taps;
|
|
GstAudioFormat format;
|
|
gpointer taps;
|
|
|
|
switch (resampler->filter_interpolation) {
|
|
default:
|
|
case GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_LINEAR:
|
|
isize = 2;
|
|
break;
|
|
case GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC:
|
|
isize = 4;
|
|
break;
|
|
}
|
|
otaps = oversample * n_taps + isize - 1;
|
|
|
|
if (resampler->filter_mode == GST_AUDIO_RESAMPLER_FILTER_MODE_FULL) {
|
|
format = GST_AUDIO_FORMAT_F64;
|
|
bps = sizeof (gdouble);
|
|
} else
|
|
format = resampler->format;
|
|
|
|
alloc_taps_mem (resampler, bps, otaps, oversample, isize);
|
|
|
|
taps = tmp_taps = resampler->tmp_taps;
|
|
x = 1.0 - n_taps / 2;
|
|
weight = make_taps (resampler, tmp_taps, x, otaps, oversample);
|
|
|
|
switch (format) {
|
|
case GST_AUDIO_FORMAT_S16:
|
|
convert_taps_gint16 (tmp_taps, taps, weight / oversample, otaps);
|
|
extract_taps_gint16 (resampler, taps, n_taps, oversample, isize);
|
|
break;
|
|
case GST_AUDIO_FORMAT_S32:
|
|
convert_taps_gint32 (tmp_taps, taps, weight / oversample, otaps);
|
|
extract_taps_gint32 (resampler, taps, n_taps, oversample, isize);
|
|
break;
|
|
case GST_AUDIO_FORMAT_F32:
|
|
convert_taps_gfloat (tmp_taps, taps, weight / oversample, otaps);
|
|
extract_taps_gfloat (resampler, taps, n_taps, oversample, isize);
|
|
break;
|
|
default:
|
|
case GST_AUDIO_FORMAT_F64:
|
|
convert_taps_gdouble (tmp_taps, taps, weight / oversample, otaps);
|
|
extract_taps_gdouble (resampler, taps, n_taps, oversample, isize);
|
|
break;
|
|
}
|
|
}
|
|
setup_functions (resampler);
|
|
}
|
|
|
|
#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##_none (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:
|
|
* @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, gint channels,
|
|
gint in_rate, gint out_rate, GstStructure * options)
|
|
{
|
|
GstAudioResampler *resampler;
|
|
const GstAudioFormatInfo *info;
|
|
GstStructure *def_options = NULL;
|
|
|
|
g_return_val_if_fail (channels > 0, FALSE);
|
|
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->sbuf = g_malloc0 (sizeof (gpointer) * channels);
|
|
|
|
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);
|
|
|
|
/* half of the filter is filled with 0 */
|
|
resampler->samp_index = 0;
|
|
resampler->samples_avail = resampler->n_taps / 2 - 1;
|
|
|
|
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)) {
|
|
guint 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;
|
|
}
|
|
}
|
|
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->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_slice_free (GstAudioResampler, resampler);
|
|
}
|
|
|
|
static inline gsize
|
|
calc_out (GstAudioResampler * resampler, gsize in)
|
|
{
|
|
gsize out;
|
|
|
|
out = in * resampler->out_rate;
|
|
if (out < 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) 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;
|
|
}
|
|
|
|
/**
|
|
* 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 ", avail %" 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;
|
|
}
|
|
}
|
|
}
|