2 * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
3 * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
5 * This file is part of Libav.
7 * Libav is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
12 * Libav is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with Libav; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22 #include "libavutil/libm.h"
23 #include "libavutil/log.h"
25 #include "audio_data.h"
27 struct ResampleContext {
28 AVAudioResampleContext *avr;
37 int compensation_distance;
41 enum AVResampleFilterType filter_type;
44 void (*set_filter)(void *filter, double *tab, int phase, int tap_count);
45 void (*resample_one)(struct ResampleContext *c, int no_filter, void *dst0,
46 int dst_index, const void *src0, int src_size,
52 #define CONFIG_RESAMPLE_DBL
53 #include "resample_template.c"
54 #undef CONFIG_RESAMPLE_DBL
57 #define CONFIG_RESAMPLE_FLT
58 #include "resample_template.c"
59 #undef CONFIG_RESAMPLE_FLT
62 #define CONFIG_RESAMPLE_S32
63 #include "resample_template.c"
64 #undef CONFIG_RESAMPLE_S32
67 #include "resample_template.c"
70 /* 0th order modified bessel function of the first kind. */
71 static double bessel(double x)
79 for (i = 1; v != lastv; i++) {
87 /* Build a polyphase filterbank. */
88 static int build_filter(ResampleContext *c)
91 double x, y, w, factor;
93 int tap_count = c->filter_length;
94 int phase_count = 1 << c->phase_shift;
95 const int center = (tap_count - 1) / 2;
97 tab = av_malloc(tap_count * sizeof(*tab));
99 return AVERROR(ENOMEM);
101 /* if upsampling, only need to interpolate, no filter */
102 factor = FFMIN(c->factor, 1.0);
104 for (ph = 0; ph < phase_count; ph++) {
106 for (i = 0; i < tap_count; i++) {
107 x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
110 switch (c->filter_type) {
111 case AV_RESAMPLE_FILTER_TYPE_CUBIC: {
112 const float d = -0.5; //first order derivative = -0.5
113 x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
114 if (x < 1.0) y = 1 - 3 * x*x + 2 * x*x*x + d * ( -x*x + x*x*x);
115 else y = d * (-4 + 8 * x - 5 * x*x + x*x*x);
118 case AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL:
119 w = 2.0 * x / (factor * tap_count) + M_PI;
120 y *= 0.3635819 - 0.4891775 * cos( w) +
121 0.1365995 * cos(2 * w) -
122 0.0106411 * cos(3 * w);
124 case AV_RESAMPLE_FILTER_TYPE_KAISER:
125 w = 2.0 * x / (factor * tap_count * M_PI);
126 y *= bessel(c->kaiser_beta * sqrt(FFMAX(1 - w * w, 0)));
133 /* normalize so that an uniform color remains the same */
134 for (i = 0; i < tap_count; i++)
135 tab[i] = tab[i] / norm;
137 c->set_filter(c->filter_bank, tab, ph, tap_count);
144 ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr)
147 int out_rate = avr->out_sample_rate;
148 int in_rate = avr->in_sample_rate;
149 double factor = FFMIN(out_rate * avr->cutoff / in_rate, 1.0);
150 int phase_count = 1 << avr->phase_shift;
153 if (avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P &&
154 avr->internal_sample_fmt != AV_SAMPLE_FMT_S32P &&
155 avr->internal_sample_fmt != AV_SAMPLE_FMT_FLTP &&
156 avr->internal_sample_fmt != AV_SAMPLE_FMT_DBLP) {
157 av_log(avr, AV_LOG_ERROR, "Unsupported internal format for "
159 av_get_sample_fmt_name(avr->internal_sample_fmt));
162 c = av_mallocz(sizeof(*c));
167 c->phase_shift = avr->phase_shift;
168 c->phase_mask = phase_count - 1;
169 c->linear = avr->linear_interp;
171 c->filter_length = FFMAX((int)ceil(avr->filter_size / factor), 1);
172 c->filter_type = avr->filter_type;
173 c->kaiser_beta = avr->kaiser_beta;
175 switch (avr->internal_sample_fmt) {
176 case AV_SAMPLE_FMT_DBLP:
177 c->resample_one = resample_one_dbl;
178 c->set_filter = set_filter_dbl;
180 case AV_SAMPLE_FMT_FLTP:
181 c->resample_one = resample_one_flt;
182 c->set_filter = set_filter_flt;
184 case AV_SAMPLE_FMT_S32P:
185 c->resample_one = resample_one_s32;
186 c->set_filter = set_filter_s32;
188 case AV_SAMPLE_FMT_S16P:
189 c->resample_one = resample_one_s16;
190 c->set_filter = set_filter_s16;
194 felem_size = av_get_bytes_per_sample(avr->internal_sample_fmt);
195 c->filter_bank = av_mallocz(c->filter_length * (phase_count + 1) * felem_size);
199 if (build_filter(c) < 0)
202 memcpy(&c->filter_bank[(c->filter_length * phase_count + 1) * felem_size],
203 c->filter_bank, (c->filter_length - 1) * felem_size);
204 memcpy(&c->filter_bank[c->filter_length * phase_count * felem_size],
205 &c->filter_bank[(c->filter_length - 1) * felem_size], felem_size);
207 c->compensation_distance = 0;
208 if (!av_reduce(&c->src_incr, &c->dst_incr, out_rate,
209 in_rate * (int64_t)phase_count, INT32_MAX / 2))
211 c->ideal_dst_incr = c->dst_incr;
213 c->index = -phase_count * ((c->filter_length - 1) / 2);
216 /* allocate internal buffer */
217 c->buffer = ff_audio_data_alloc(avr->resample_channels, 0,
218 avr->internal_sample_fmt,
223 av_log(avr, AV_LOG_DEBUG, "resample: %s from %d Hz to %d Hz\n",
224 av_get_sample_fmt_name(avr->internal_sample_fmt),
225 avr->in_sample_rate, avr->out_sample_rate);
230 ff_audio_data_free(&c->buffer);
231 av_free(c->filter_bank);
236 void ff_audio_resample_free(ResampleContext **c)
240 ff_audio_data_free(&(*c)->buffer);
241 av_free((*c)->filter_bank);
245 int avresample_set_compensation(AVAudioResampleContext *avr, int sample_delta,
246 int compensation_distance)
249 AudioData *fifo_buf = NULL;
252 if (compensation_distance < 0)
253 return AVERROR(EINVAL);
254 if (!compensation_distance && sample_delta)
255 return AVERROR(EINVAL);
257 /* if resampling was not enabled previously, re-initialize the
258 AVAudioResampleContext and force resampling */
259 if (!avr->resample_needed) {
261 double matrix[AVRESAMPLE_MAX_CHANNELS * AVRESAMPLE_MAX_CHANNELS] = { 0 };
263 /* buffer any remaining samples in the output FIFO before closing */
264 fifo_samples = av_audio_fifo_size(avr->out_fifo);
265 if (fifo_samples > 0) {
266 fifo_buf = ff_audio_data_alloc(avr->out_channels, fifo_samples,
267 avr->out_sample_fmt, NULL);
269 return AVERROR(EINVAL);
270 ret = ff_audio_data_read_from_fifo(avr->out_fifo, fifo_buf,
275 /* save the channel mixing matrix */
276 ret = avresample_get_matrix(avr, matrix, AVRESAMPLE_MAX_CHANNELS);
280 /* close the AVAudioResampleContext */
281 avresample_close(avr);
283 avr->force_resampling = 1;
285 /* restore the channel mixing matrix */
286 ret = avresample_set_matrix(avr, matrix, AVRESAMPLE_MAX_CHANNELS);
290 /* re-open the AVAudioResampleContext */
291 ret = avresample_open(avr);
295 /* restore buffered samples to the output FIFO */
296 if (fifo_samples > 0) {
297 ret = ff_audio_data_add_to_fifo(avr->out_fifo, fifo_buf, 0,
301 ff_audio_data_free(&fifo_buf);
305 c->compensation_distance = compensation_distance;
306 if (compensation_distance) {
307 c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr *
308 (int64_t)sample_delta / compensation_distance;
310 c->dst_incr = c->ideal_dst_incr;
315 ff_audio_data_free(&fifo_buf);
319 static int resample(ResampleContext *c, void *dst, const void *src,
320 int *consumed, int src_size, int dst_size, int update_ctx)
323 int index = c->index;
325 int dst_incr_frac = c->dst_incr % c->src_incr;
326 int dst_incr = c->dst_incr / c->src_incr;
327 int compensation_distance = c->compensation_distance;
330 return AVERROR(EINVAL);
332 if (compensation_distance == 0 && c->filter_length == 1 &&
333 c->phase_shift == 0) {
334 int64_t index2 = ((int64_t)index) << 32;
335 int64_t incr = (1LL << 32) * c->dst_incr / c->src_incr;
336 dst_size = FFMIN(dst_size,
337 (src_size-1-index) * (int64_t)c->src_incr /
341 for(dst_index = 0; dst_index < dst_size; dst_index++) {
342 c->resample_one(c, 1, dst, dst_index, src, 0, index2 >> 32, 0);
346 dst_index = dst_size;
348 index += dst_index * dst_incr;
349 index += (frac + dst_index * (int64_t)dst_incr_frac) / c->src_incr;
350 frac = (frac + dst_index * (int64_t)dst_incr_frac) % c->src_incr;
352 for (dst_index = 0; dst_index < dst_size; dst_index++) {
353 int sample_index = index >> c->phase_shift;
355 if (sample_index + c->filter_length > src_size ||
356 -sample_index >= src_size)
360 c->resample_one(c, 0, dst, dst_index, src, src_size, index, frac);
362 frac += dst_incr_frac;
364 if (frac >= c->src_incr) {
368 if (dst_index + 1 == compensation_distance) {
369 compensation_distance = 0;
370 dst_incr_frac = c->ideal_dst_incr % c->src_incr;
371 dst_incr = c->ideal_dst_incr / c->src_incr;
376 *consumed = FFMAX(index, 0) >> c->phase_shift;
380 index &= c->phase_mask;
382 if (compensation_distance) {
383 compensation_distance -= dst_index;
384 if (compensation_distance <= 0)
389 c->dst_incr = dst_incr_frac + c->src_incr*dst_incr;
390 c->compensation_distance = compensation_distance;
396 int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src,
399 int ch, in_samples, in_leftover, out_samples = 0;
400 int ret = AVERROR(EINVAL);
402 in_samples = src ? src->nb_samples : 0;
403 in_leftover = c->buffer->nb_samples;
405 /* add input samples to the internal buffer */
407 ret = ff_audio_data_combine(c->buffer, in_leftover, src, 0, in_samples);
410 } else if (!in_leftover) {
411 /* no remaining samples to flush */
414 /* TODO: pad buffer to flush completely */
417 /* calculate output size and reallocate output buffer if needed */
418 /* TODO: try to calculate this without the dummy resample() run */
419 if (!dst->read_only && dst->allow_realloc) {
420 out_samples = resample(c, NULL, NULL, NULL, c->buffer->nb_samples,
422 ret = ff_audio_data_realloc(dst, out_samples);
424 av_log(c->avr, AV_LOG_ERROR, "error reallocating output\n");
429 /* resample each channel plane */
430 for (ch = 0; ch < c->buffer->channels; ch++) {
431 out_samples = resample(c, (void *)dst->data[ch],
432 (const void *)c->buffer->data[ch], consumed,
433 c->buffer->nb_samples, dst->allocated_samples,
434 ch + 1 == c->buffer->channels);
436 if (out_samples < 0) {
437 av_log(c->avr, AV_LOG_ERROR, "error during resampling\n");
441 /* drain consumed samples from the internal buffer */
442 ff_audio_data_drain(c->buffer, *consumed);
444 av_dlog(c->avr, "resampled %d in + %d leftover to %d out + %d leftover\n",
445 in_samples, in_leftover, out_samples, c->buffer->nb_samples);
447 dst->nb_samples = out_samples;
451 int avresample_get_delay(AVAudioResampleContext *avr)
453 if (!avr->resample_needed || !avr->resample)
456 return avr->resample->buffer->nb_samples;