447 lines
15 KiB
C
447 lines
15 KiB
C
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/*
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* Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
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* Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg 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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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "libavutil/common.h"
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#include "libavutil/libm.h"
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#include "libavutil/log.h"
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#include "internal.h"
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#include "resample.h"
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#include "audio_data.h"
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/* double template */
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#define CONFIG_RESAMPLE_DBL
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#include "resample_template.c"
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#undef CONFIG_RESAMPLE_DBL
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/* float template */
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#define CONFIG_RESAMPLE_FLT
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#include "resample_template.c"
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#undef CONFIG_RESAMPLE_FLT
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/* s32 template */
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#define CONFIG_RESAMPLE_S32
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#include "resample_template.c"
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#undef CONFIG_RESAMPLE_S32
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/* s16 template */
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#include "resample_template.c"
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/* 0th order modified Bessel function of the first kind. */
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static double bessel(double x)
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{
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double v = 1;
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double lastv = 0;
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double t = 1;
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int i;
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x = x * x / 4;
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for (i = 1; v != lastv; i++) {
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lastv = v;
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t *= x / (i * i);
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v += t;
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}
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return v;
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}
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/* Build a polyphase filterbank. */
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static int build_filter(ResampleContext *c, double factor)
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{
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int ph, i;
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double x, y, w;
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double *tab;
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int tap_count = c->filter_length;
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int phase_count = 1 << c->phase_shift;
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const int center = (tap_count - 1) / 2;
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tab = av_malloc(tap_count * sizeof(*tab));
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if (!tab)
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return AVERROR(ENOMEM);
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for (ph = 0; ph < phase_count; ph++) {
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double norm = 0;
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for (i = 0; i < tap_count; i++) {
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x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
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if (x == 0) y = 1.0;
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else y = sin(x) / x;
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switch (c->filter_type) {
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case AV_RESAMPLE_FILTER_TYPE_CUBIC: {
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const float d = -0.5; //first order derivative = -0.5
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x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
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if (x < 1.0) y = 1 - 3 * x*x + 2 * x*x*x + d * ( -x*x + x*x*x);
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else y = d * (-4 + 8 * x - 5 * x*x + x*x*x);
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break;
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}
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case AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL:
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w = 2.0 * x / (factor * tap_count) + M_PI;
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y *= 0.3635819 - 0.4891775 * cos( w) +
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0.1365995 * cos(2 * w) -
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0.0106411 * cos(3 * w);
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break;
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case AV_RESAMPLE_FILTER_TYPE_KAISER:
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w = 2.0 * x / (factor * tap_count * M_PI);
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y *= bessel(c->kaiser_beta * sqrt(FFMAX(1 - w * w, 0)));
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break;
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}
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tab[i] = y;
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norm += y;
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}
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/* normalize so that an uniform color remains the same */
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for (i = 0; i < tap_count; i++)
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tab[i] = tab[i] / norm;
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c->set_filter(c->filter_bank, tab, ph, tap_count);
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}
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av_free(tab);
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return 0;
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}
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ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr)
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{
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ResampleContext *c;
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int out_rate = avr->out_sample_rate;
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int in_rate = avr->in_sample_rate;
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double factor = FFMIN(out_rate * avr->cutoff / in_rate, 1.0);
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int phase_count = 1 << avr->phase_shift;
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int felem_size;
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if (avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P &&
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avr->internal_sample_fmt != AV_SAMPLE_FMT_S32P &&
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avr->internal_sample_fmt != AV_SAMPLE_FMT_FLTP &&
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avr->internal_sample_fmt != AV_SAMPLE_FMT_DBLP) {
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av_log(avr, AV_LOG_ERROR, "Unsupported internal format for "
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"resampling: %s\n",
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av_get_sample_fmt_name(avr->internal_sample_fmt));
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return NULL;
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}
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c = av_mallocz(sizeof(*c));
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if (!c)
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return NULL;
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c->avr = avr;
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c->phase_shift = avr->phase_shift;
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c->phase_mask = phase_count - 1;
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c->linear = avr->linear_interp;
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c->filter_length = FFMAX((int)ceil(avr->filter_size / factor), 1);
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c->filter_type = avr->filter_type;
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c->kaiser_beta = avr->kaiser_beta;
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switch (avr->internal_sample_fmt) {
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case AV_SAMPLE_FMT_DBLP:
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c->resample_one = c->linear ? resample_linear_dbl : resample_one_dbl;
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c->resample_nearest = resample_nearest_dbl;
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c->set_filter = set_filter_dbl;
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break;
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case AV_SAMPLE_FMT_FLTP:
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c->resample_one = c->linear ? resample_linear_flt : resample_one_flt;
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c->resample_nearest = resample_nearest_flt;
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c->set_filter = set_filter_flt;
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break;
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case AV_SAMPLE_FMT_S32P:
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c->resample_one = c->linear ? resample_linear_s32 : resample_one_s32;
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c->resample_nearest = resample_nearest_s32;
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c->set_filter = set_filter_s32;
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break;
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case AV_SAMPLE_FMT_S16P:
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c->resample_one = c->linear ? resample_linear_s16 : resample_one_s16;
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c->resample_nearest = resample_nearest_s16;
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c->set_filter = set_filter_s16;
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break;
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}
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if (ARCH_AARCH64)
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ff_audio_resample_init_aarch64(c, avr->internal_sample_fmt);
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if (ARCH_ARM)
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ff_audio_resample_init_arm(c, avr->internal_sample_fmt);
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felem_size = av_get_bytes_per_sample(avr->internal_sample_fmt);
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c->filter_bank = av_mallocz(c->filter_length * (phase_count + 1) * felem_size);
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if (!c->filter_bank)
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goto error;
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if (build_filter(c, factor) < 0)
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goto error;
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memcpy(&c->filter_bank[(c->filter_length * phase_count + 1) * felem_size],
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c->filter_bank, (c->filter_length - 1) * felem_size);
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memcpy(&c->filter_bank[c->filter_length * phase_count * felem_size],
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&c->filter_bank[(c->filter_length - 1) * felem_size], felem_size);
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c->compensation_distance = 0;
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if (!av_reduce(&c->src_incr, &c->dst_incr, out_rate,
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in_rate * (int64_t)phase_count, INT32_MAX / 2))
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goto error;
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c->ideal_dst_incr = c->dst_incr;
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c->padding_size = (c->filter_length - 1) / 2;
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c->initial_padding_filled = 0;
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c->index = 0;
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c->frac = 0;
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/* allocate internal buffer */
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c->buffer = ff_audio_data_alloc(avr->resample_channels, c->padding_size,
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avr->internal_sample_fmt,
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"resample buffer");
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if (!c->buffer)
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goto error;
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c->buffer->nb_samples = c->padding_size;
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c->initial_padding_samples = c->padding_size;
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av_log(avr, AV_LOG_DEBUG, "resample: %s from %d Hz to %d Hz\n",
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av_get_sample_fmt_name(avr->internal_sample_fmt),
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avr->in_sample_rate, avr->out_sample_rate);
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return c;
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error:
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ff_audio_data_free(&c->buffer);
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av_free(c->filter_bank);
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av_free(c);
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return NULL;
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}
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void ff_audio_resample_free(ResampleContext **c)
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{
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if (!*c)
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return;
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ff_audio_data_free(&(*c)->buffer);
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av_free((*c)->filter_bank);
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av_freep(c);
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}
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int avresample_set_compensation(AVAudioResampleContext *avr, int sample_delta,
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int compensation_distance)
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{
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ResampleContext *c;
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if (compensation_distance < 0)
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return AVERROR(EINVAL);
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if (!compensation_distance && sample_delta)
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return AVERROR(EINVAL);
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if (!avr->resample_needed) {
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av_log(avr, AV_LOG_ERROR, "Unable to set resampling compensation\n");
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return AVERROR(EINVAL);
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}
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c = avr->resample;
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c->compensation_distance = compensation_distance;
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if (compensation_distance) {
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c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr *
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(int64_t)sample_delta / compensation_distance;
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} else {
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c->dst_incr = c->ideal_dst_incr;
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}
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return 0;
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}
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static int resample(ResampleContext *c, void *dst, const void *src,
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int *consumed, int src_size, int dst_size, int update_ctx,
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int nearest_neighbour)
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{
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int dst_index;
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unsigned int index = c->index;
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int frac = c->frac;
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int dst_incr_frac = c->dst_incr % c->src_incr;
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int dst_incr = c->dst_incr / c->src_incr;
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int compensation_distance = c->compensation_distance;
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if (!dst != !src)
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return AVERROR(EINVAL);
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if (nearest_neighbour) {
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uint64_t index2 = ((uint64_t)index) << 32;
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int64_t incr = (1LL << 32) * c->dst_incr / c->src_incr;
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dst_size = FFMIN(dst_size,
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(src_size-1-index) * (int64_t)c->src_incr /
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c->dst_incr);
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if (dst) {
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for(dst_index = 0; dst_index < dst_size; dst_index++) {
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c->resample_nearest(dst, dst_index, src, index2 >> 32);
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index2 += incr;
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}
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} else {
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dst_index = dst_size;
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}
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index += dst_index * dst_incr;
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index += (frac + dst_index * (int64_t)dst_incr_frac) / c->src_incr;
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frac = (frac + dst_index * (int64_t)dst_incr_frac) % c->src_incr;
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} else {
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for (dst_index = 0; dst_index < dst_size; dst_index++) {
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int sample_index = index >> c->phase_shift;
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if (sample_index + c->filter_length > src_size)
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break;
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if (dst)
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c->resample_one(c, dst, dst_index, src, index, frac);
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frac += dst_incr_frac;
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index += dst_incr;
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if (frac >= c->src_incr) {
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frac -= c->src_incr;
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index++;
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}
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if (dst_index + 1 == compensation_distance) {
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compensation_distance = 0;
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dst_incr_frac = c->ideal_dst_incr % c->src_incr;
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dst_incr = c->ideal_dst_incr / c->src_incr;
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}
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}
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}
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if (consumed)
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*consumed = index >> c->phase_shift;
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if (update_ctx) {
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index &= c->phase_mask;
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if (compensation_distance) {
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compensation_distance -= dst_index;
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if (compensation_distance <= 0)
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return AVERROR_BUG;
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}
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c->frac = frac;
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c->index = index;
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c->dst_incr = dst_incr_frac + c->src_incr*dst_incr;
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c->compensation_distance = compensation_distance;
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}
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return dst_index;
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}
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int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src)
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{
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int ch, in_samples, in_leftover, consumed = 0, out_samples = 0;
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int ret = AVERROR(EINVAL);
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int nearest_neighbour = (c->compensation_distance == 0 &&
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c->filter_length == 1 &&
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c->phase_shift == 0);
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in_samples = src ? src->nb_samples : 0;
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in_leftover = c->buffer->nb_samples;
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/* add input samples to the internal buffer */
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if (src) {
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ret = ff_audio_data_combine(c->buffer, in_leftover, src, 0, in_samples);
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if (ret < 0)
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return ret;
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} else if (in_leftover <= c->final_padding_samples) {
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/* no remaining samples to flush */
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return 0;
|
||
|
|
}
|
||
|
|
|
||
|
|
if (!c->initial_padding_filled) {
|
||
|
|
int bps = av_get_bytes_per_sample(c->avr->internal_sample_fmt);
|
||
|
|
int i;
|
||
|
|
|
||
|
|
if (src && c->buffer->nb_samples < 2 * c->padding_size)
|
||
|
|
return 0;
|
||
|
|
|
||
|
|
for (i = 0; i < c->padding_size; i++)
|
||
|
|
for (ch = 0; ch < c->buffer->channels; ch++) {
|
||
|
|
if (c->buffer->nb_samples > 2 * c->padding_size - i) {
|
||
|
|
memcpy(c->buffer->data[ch] + bps * i,
|
||
|
|
c->buffer->data[ch] + bps * (2 * c->padding_size - i), bps);
|
||
|
|
} else {
|
||
|
|
memset(c->buffer->data[ch] + bps * i, 0, bps);
|
||
|
|
}
|
||
|
|
}
|
||
|
|
c->initial_padding_filled = 1;
|
||
|
|
}
|
||
|
|
|
||
|
|
if (!src && !c->final_padding_filled) {
|
||
|
|
int bps = av_get_bytes_per_sample(c->avr->internal_sample_fmt);
|
||
|
|
int i;
|
||
|
|
|
||
|
|
ret = ff_audio_data_realloc(c->buffer,
|
||
|
|
FFMAX(in_samples, in_leftover) +
|
||
|
|
c->padding_size);
|
||
|
|
if (ret < 0) {
|
||
|
|
av_log(c->avr, AV_LOG_ERROR, "Error reallocating resampling buffer\n");
|
||
|
|
return AVERROR(ENOMEM);
|
||
|
|
}
|
||
|
|
|
||
|
|
for (i = 0; i < c->padding_size; i++)
|
||
|
|
for (ch = 0; ch < c->buffer->channels; ch++) {
|
||
|
|
if (in_leftover > i) {
|
||
|
|
memcpy(c->buffer->data[ch] + bps * (in_leftover + i),
|
||
|
|
c->buffer->data[ch] + bps * (in_leftover - i - 1),
|
||
|
|
bps);
|
||
|
|
} else {
|
||
|
|
memset(c->buffer->data[ch] + bps * (in_leftover + i),
|
||
|
|
0, bps);
|
||
|
|
}
|
||
|
|
}
|
||
|
|
c->buffer->nb_samples += c->padding_size;
|
||
|
|
c->final_padding_samples = c->padding_size;
|
||
|
|
c->final_padding_filled = 1;
|
||
|
|
}
|
||
|
|
|
||
|
|
|
||
|
|
/* calculate output size and reallocate output buffer if needed */
|
||
|
|
/* TODO: try to calculate this without the dummy resample() run */
|
||
|
|
if (!dst->read_only && dst->allow_realloc) {
|
||
|
|
out_samples = resample(c, NULL, NULL, NULL, c->buffer->nb_samples,
|
||
|
|
INT_MAX, 0, nearest_neighbour);
|
||
|
|
ret = ff_audio_data_realloc(dst, out_samples);
|
||
|
|
if (ret < 0) {
|
||
|
|
av_log(c->avr, AV_LOG_ERROR, "error reallocating output\n");
|
||
|
|
return ret;
|
||
|
|
}
|
||
|
|
}
|
||
|
|
|
||
|
|
/* resample each channel plane */
|
||
|
|
for (ch = 0; ch < c->buffer->channels; ch++) {
|
||
|
|
out_samples = resample(c, (void *)dst->data[ch],
|
||
|
|
(const void *)c->buffer->data[ch], &consumed,
|
||
|
|
c->buffer->nb_samples, dst->allocated_samples,
|
||
|
|
ch + 1 == c->buffer->channels, nearest_neighbour);
|
||
|
|
}
|
||
|
|
if (out_samples < 0) {
|
||
|
|
av_log(c->avr, AV_LOG_ERROR, "error during resampling\n");
|
||
|
|
return out_samples;
|
||
|
|
}
|
||
|
|
|
||
|
|
/* drain consumed samples from the internal buffer */
|
||
|
|
ff_audio_data_drain(c->buffer, consumed);
|
||
|
|
c->initial_padding_samples = FFMAX(c->initial_padding_samples - consumed, 0);
|
||
|
|
|
||
|
|
av_log(c->avr, AV_LOG_TRACE, "resampled %d in + %d leftover to %d out + %d leftover\n",
|
||
|
|
in_samples, in_leftover, out_samples, c->buffer->nb_samples);
|
||
|
|
|
||
|
|
dst->nb_samples = out_samples;
|
||
|
|
return 0;
|
||
|
|
}
|
||
|
|
|
||
|
|
int avresample_get_delay(AVAudioResampleContext *avr)
|
||
|
|
{
|
||
|
|
ResampleContext *c = avr->resample;
|
||
|
|
|
||
|
|
if (!avr->resample_needed || !avr->resample)
|
||
|
|
return 0;
|
||
|
|
|
||
|
|
return FFMAX(c->buffer->nb_samples - c->padding_size, 0);
|
||
|
|
}
|