/* GStreamer * Copyright (C) <2015> Wim Taymans * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include #include #include #ifdef HAVE_ORC #include #endif #include "audio-resampler.h" #include "audio-resampler-private.h" #include "audio-resampler-macros.h" #define MEM_ALIGN(m,a) ((gint8 *)((guintptr)((gint8 *)(m) + ((a)-1)) & ~((a)-1))) #define ALIGN 16 #define TAPS_OVERREAD 16 GST_DEBUG_CATEGORY_STATIC (audio_resampler_debug); #define GST_CAT_DEFAULT audio_resampler_debug /** * SECTION:gstaudioresampler * @title: GstAudioResampler * @short_description: Utility structure for resampler information * * #GstAudioResampler is a structure which holds the information * required to perform various kinds of resampling filtering. * */ static const gint oversample_qualities[] = { 4, 4, 4, 8, 8, 16, 16, 16, 16, 32, 32 }; typedef struct { gdouble cutoff; gdouble downsample_cutoff_factor; gdouble stopband_attenuation; gdouble transition_bandwidth; } KaiserQualityMap; static const KaiserQualityMap kaiser_qualities[] = { {0.860, 0.96511, 60, 0.7}, /* 8 taps */ {0.880, 0.96591, 65, 0.29}, /* 16 taps */ {0.910, 0.96923, 70, 0.145}, /* 32 taps */ {0.920, 0.97600, 80, 0.105}, /* 48 taps */ {0.940, 0.97979, 85, 0.087}, /* 64 taps default quality */ {0.940, 0.98085, 95, 0.077}, /* 80 taps */ {0.945, 0.99471, 100, 0.068}, /* 96 taps */ {0.950, 1.0, 105, 0.055}, /* 128 taps */ {0.960, 1.0, 110, 0.045}, /* 160 taps */ {0.968, 1.0, 115, 0.039}, /* 192 taps */ {0.975, 1.0, 120, 0.0305} /* 256 taps */ }; typedef struct { gint n_taps; gdouble cutoff; } BlackmanQualityMap; static const BlackmanQualityMap blackman_qualities[] = { {8, 0.5,}, {16, 0.6,}, {24, 0.72,}, {32, 0.8,}, {48, 0.85,}, /* default */ {64, 0.90,}, {80, 0.92,}, {96, 0.933,}, {128, 0.950,}, {148, 0.955,}, {160, 0.960,} }; #define DEFAULT_RESAMPLER_METHOD GST_AUDIO_RESAMPLER_METHOD_KAISER #define DEFAULT_QUALITY GST_AUDIO_RESAMPLER_QUALITY_DEFAULT #define DEFAULT_OPT_CUBIC_B 1.0 #define DEFAULT_OPT_CUBIC_C 0.0 #define DEFAULT_OPT_FILTER_MODE GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO #define DEFAULT_OPT_FILTER_MODE_THRESHOLD 1048576 #define DEFAULT_OPT_FILTER_INTERPOLATION GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC #define DEFAULT_OPT_FILTER_OVERSAMPLE 8 #define DEFAULT_OPT_MAX_PHASE_ERROR 0.1 static gdouble get_opt_double (GstStructure * options, const gchar * name, gdouble def) { gdouble res; if (!options || !gst_structure_get_double (options, name, &res)) res = def; return res; } static gint get_opt_int (GstStructure * options, const gchar * name, gint def) { gint res; if (!options || !gst_structure_get_int (options, name, &res)) res = def; return res; } static gint get_opt_enum (GstStructure * options, const gchar * name, GType type, gint def) { gint res; if (!options || !gst_structure_get_enum (options, name, type, &res)) res = def; return res; } #define GET_OPT_CUTOFF(options,def) get_opt_double(options, \ GST_AUDIO_RESAMPLER_OPT_CUTOFF,def) #define GET_OPT_DOWN_CUTOFF_FACTOR(options,def) get_opt_double(options, \ GST_AUDIO_RESAMPLER_OPT_DOWN_CUTOFF_FACTOR, def) #define GET_OPT_STOP_ATTENUATION(options,def) get_opt_double(options, \ GST_AUDIO_RESAMPLER_OPT_STOP_ATTENUATION, def) #define GET_OPT_TRANSITION_BANDWIDTH(options,def) get_opt_double(options, \ GST_AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH, def) #define GET_OPT_CUBIC_B(options) get_opt_double(options, \ GST_AUDIO_RESAMPLER_OPT_CUBIC_B, DEFAULT_OPT_CUBIC_B) #define GET_OPT_CUBIC_C(options) get_opt_double(options, \ GST_AUDIO_RESAMPLER_OPT_CUBIC_C, DEFAULT_OPT_CUBIC_C) #define GET_OPT_N_TAPS(options,def) get_opt_int(options, \ GST_AUDIO_RESAMPLER_OPT_N_TAPS, def) #define GET_OPT_FILTER_MODE(options) get_opt_enum(options, \ GST_AUDIO_RESAMPLER_OPT_FILTER_MODE, GST_TYPE_AUDIO_RESAMPLER_FILTER_MODE, \ DEFAULT_OPT_FILTER_MODE) #define GET_OPT_FILTER_MODE_THRESHOLD(options) get_opt_int(options, \ GST_AUDIO_RESAMPLER_OPT_FILTER_MODE_THRESHOLD, DEFAULT_OPT_FILTER_MODE_THRESHOLD) #define GET_OPT_FILTER_INTERPOLATION(options) get_opt_enum(options, \ GST_AUDIO_RESAMPLER_OPT_FILTER_INTERPOLATION, GST_TYPE_AUDIO_RESAMPLER_FILTER_INTERPOLATION, \ DEFAULT_OPT_FILTER_INTERPOLATION) #define GET_OPT_FILTER_OVERSAMPLE(options) get_opt_int(options, \ GST_AUDIO_RESAMPLER_OPT_FILTER_OVERSAMPLE, DEFAULT_OPT_FILTER_OVERSAMPLE) #define GET_OPT_MAX_PHASE_ERROR(options) get_opt_double(options, \ GST_AUDIO_RESAMPLER_OPT_MAX_PHASE_ERROR, DEFAULT_OPT_MAX_PHASE_ERROR) #include "dbesi0.c" #define bessel dbesi0 static inline gdouble get_linear_tap (gdouble x, gint n_taps) { gdouble res = GST_ROUND_UP_2 (n_taps) / 2 - fabs (x); return res; } static inline gdouble get_cubic_tap (gdouble x, gint n_taps, gdouble b, gdouble c) { gdouble res, a, a2, a3; a = fabs (x * 4.0) / n_taps; a2 = a * a; a3 = a2 * a; if (a <= 1.0) res = ((12.0 - 9.0 * b - 6.0 * c) * a3 + (-18.0 + 12.0 * b + 6.0 * c) * a2 + (6.0 - 2.0 * b)) / 6.0; else if (a <= 2.0) res = ((-b - 6.0 * c) * a3 + (6.0 * b + 30.0 * c) * a2 + (-12.0 * b - 48.0 * c) * a + (8.0 * b + 24.0 * c)) / 6.0; else res = 0.0; return res; } static inline gdouble get_blackman_nuttall_tap (gdouble x, gint n_taps, gdouble Fc) { gdouble s, y, w; y = G_PI * x; s = (y == 0.0 ? Fc : sin (y * Fc) / y); w = 2.0 * y / n_taps + G_PI; return s * (0.3635819 - 0.4891775 * cos (w) + 0.1365995 * cos (2 * w) - 0.0106411 * cos (3 * w)); } static inline gdouble get_kaiser_tap (gdouble x, gint n_taps, gdouble Fc, gdouble beta) { gdouble s, y, w; y = G_PI * x; s = (y == 0.0 ? Fc : sin (y * Fc) / y); w = 2.0 * x / n_taps; return s * bessel (beta * sqrt (MAX (1 - w * w, 0))); } #define MAKE_CONVERT_TAPS_INT_FUNC(type, precision) \ static void \ convert_taps_##type##_c (gdouble *tmp_taps, gpointer taps, \ gdouble weight, gint n_taps) \ { \ gint64 one = (1LL << precision) - 1; \ type *t = taps; \ gdouble multiplier = one; \ gint i, j; \ gdouble offset, l_offset, h_offset; \ gboolean exact = FALSE; \ /* Round to integer, but with an adjustable bias that we use to */ \ /* eliminate the DC error. */ \ l_offset = 0.0; \ h_offset = 1.0; \ offset = 0.5; \ for (i = 0; i < 32; i++) { \ gint64 sum = 0; \ for (j = 0; j < n_taps; j++) \ sum += floor (offset + tmp_taps[j] * multiplier / weight); \ if (sum == one) { \ exact = TRUE; \ break; \ } \ if (l_offset == h_offset) \ break; \ if (sum < one) { \ if (offset > l_offset) \ l_offset = offset; \ offset += (h_offset - l_offset) / 2; \ } else { \ if (offset < h_offset) \ h_offset = offset; \ offset -= (h_offset - l_offset) / 2; \ } \ } \ for (j = 0; j < n_taps; j++) \ t[j] = floor (offset + tmp_taps[j] * multiplier / weight); \ if (!exact) \ GST_DEBUG ("can't find exact taps"); \ } #define MAKE_CONVERT_TAPS_FLOAT_FUNC(type) \ static void \ convert_taps_##type##_c (gdouble *tmp_taps, gpointer taps, \ gdouble weight, gint n_taps) \ { \ gint i; \ type *t = taps; \ for (i = 0; i < n_taps; i++) \ t[i] = tmp_taps[i] / weight; \ } MAKE_CONVERT_TAPS_INT_FUNC (gint16, PRECISION_S16); MAKE_CONVERT_TAPS_INT_FUNC (gint32, PRECISION_S32); MAKE_CONVERT_TAPS_FLOAT_FUNC (gfloat); MAKE_CONVERT_TAPS_FLOAT_FUNC (gdouble); static ConvertTapsFunc convert_taps_funcs[] = { convert_taps_gint16_c, convert_taps_gint32_c, convert_taps_gfloat_c, convert_taps_gdouble_c }; #define convert_taps_gint16 convert_taps_funcs[0] #define convert_taps_gint32 convert_taps_funcs[1] #define convert_taps_gfloat convert_taps_funcs[2] #define convert_taps_gdouble convert_taps_funcs[3] static void make_taps (GstAudioResampler * resampler, gdouble * res, gdouble x, gint n_taps) { gdouble weight = 0.0, *tmp_taps = resampler->tmp_taps; gint i; switch (resampler->method) { case GST_AUDIO_RESAMPLER_METHOD_NEAREST: break; case GST_AUDIO_RESAMPLER_METHOD_LINEAR: for (i = 0; i < n_taps; i++) weight += tmp_taps[i] = get_linear_tap (x + i, resampler->n_taps); break; case GST_AUDIO_RESAMPLER_METHOD_CUBIC: for (i = 0; i < n_taps; i++) weight += tmp_taps[i] = get_cubic_tap (x + i, resampler->n_taps, resampler->b, resampler->c); break; case GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL: for (i = 0; i < n_taps; i++) weight += tmp_taps[i] = get_blackman_nuttall_tap (x + i, resampler->n_taps, resampler->cutoff); break; case GST_AUDIO_RESAMPLER_METHOD_KAISER: for (i = 0; i < n_taps; i++) weight += tmp_taps[i] = get_kaiser_tap (x + i, resampler->n_taps, resampler->cutoff, resampler->kaiser_beta); break; } resampler->convert_taps (tmp_taps, res, weight, n_taps); } #define MAKE_COEFF_LINEAR_INT_FUNC(type,type2,prec) \ static inline void \ make_coeff_##type##_linear (gint num, gint denom, type *icoeff) \ { \ type x = ((gint64)num << prec) / denom; \ icoeff[0] = icoeff[2] = x; \ icoeff[1] = icoeff[3] = (type)(((type2)1 << prec)-1) - x; \ } #define MAKE_COEFF_LINEAR_FLOAT_FUNC(type) \ static inline void \ make_coeff_##type##_linear (gint num, gint denom, type *icoeff) \ { \ type x = (type)num / denom; \ icoeff[0] = icoeff[2] = x; \ icoeff[1] = icoeff[3] = (type)1.0 - x; \ } MAKE_COEFF_LINEAR_INT_FUNC (gint16, gint32, PRECISION_S16); MAKE_COEFF_LINEAR_INT_FUNC (gint32, gint64, 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 num, gint denom, type *icoeff) \ { \ type2 one = ((type2)1 << prec) - 1; \ type2 x = ((gint64) num << prec) / denom; \ 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 num, gint denom, type *icoeff) \ { \ type x = (type) num / denom, 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] = (type)1.0 - 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); #define INTERPOLATE_INT_LINEAR_FUNC(type,type2,prec,limit) \ static inline void \ interpolate_##type##_linear_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]; \ const type *c[2] = {(type*)((gint8*)a + 0*astride), \ (type*)((gint8*)a + 1*astride)}; \ \ for (i = 0; i < len; i++) { \ tmp = ((type2)c[0][i] - (type2)c[1][i]) * c0 + \ (((type2)c[1][i]) << (prec)); \ o[i] = (tmp + ((type2)1 << ((prec) - 1))) >> (prec); \ } \ } #define INTERPOLATE_FLOAT_LINEAR_FUNC(type) \ static inline void \ interpolate_##type##_linear_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]; \ const type *c[2] = {(type*)((gint8*)a + 0*astride), \ (type*)((gint8*)a + 1*astride)}; \ \ for (i = 0; i < len; i++) { \ o[i] = (c[0][i] - c[1][i]) * c0 + c[1][i]; \ } \ } INTERPOLATE_INT_LINEAR_FUNC (gint16, gint32, PRECISION_S16, (gint32) 1 << 15); INTERPOLATE_INT_LINEAR_FUNC (gint32, gint64, PRECISION_S32, (gint64) 1 << 31); INTERPOLATE_FLOAT_LINEAR_FUNC (gfloat); INTERPOLATE_FLOAT_LINEAR_FUNC (gdouble); #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_slice_new0 (GstAudioResampler); 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_slice_free (GstAudioResampler, 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: * @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 || out_frames == 0)) { GST_LOG ("not enough samples to start: need %" G_GSIZE_FORMAT ", avail %" G_GSIZE_FORMAT ", out %" G_GSIZE_FORMAT, need, samples_avail, out_frames); /* 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; } } }