2 * Copyright (C) 2011-2013 Michael Niedermayer (michaelni@gmx.at)
4 * This file is part of libswresample
6 * libswresample is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Lesser General Public
8 * License as published by the Free Software Foundation; either
9 * version 2.1 of the License, or (at your option) any later version.
11 * libswresample is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Lesser General Public License for more details.
16 * You should have received a copy of the GNU Lesser General Public
17 * License along with libswresample; if not, write to the Free Software
18 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
21 #include "libavutil/opt.h"
22 #include "swresample_internal.h"
23 #include "audioconvert.h"
24 #include "libavutil/avassert.h"
25 #include "libavutil/channel_layout.h"
31 unsigned swresample_version(void)
33 av_assert0(LIBSWRESAMPLE_VERSION_MICRO >= 100);
34 return LIBSWRESAMPLE_VERSION_INT;
37 const char *swresample_configuration(void)
39 return FFMPEG_CONFIGURATION;
42 const char *swresample_license(void)
44 #define LICENSE_PREFIX "libswresample license: "
45 return LICENSE_PREFIX FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1;
48 int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map){
49 if(!s || s->in_convert) // s needs to be allocated but not initialized
50 return AVERROR(EINVAL);
51 s->channel_map = channel_map;
55 struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,
56 int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
57 int64_t in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate,
58 int log_offset, void *log_ctx){
59 if(!s) s= swr_alloc();
62 s->log_level_offset= log_offset;
65 if (av_opt_set_int(s, "ocl", out_ch_layout, 0) < 0)
68 if (av_opt_set_int(s, "osf", out_sample_fmt, 0) < 0)
71 if (av_opt_set_int(s, "osr", out_sample_rate, 0) < 0)
74 if (av_opt_set_int(s, "icl", in_ch_layout, 0) < 0)
77 if (av_opt_set_int(s, "isf", in_sample_fmt, 0) < 0)
80 if (av_opt_set_int(s, "isr", in_sample_rate, 0) < 0)
83 if (av_opt_set_int(s, "tsf", AV_SAMPLE_FMT_NONE, 0) < 0)
86 if (av_opt_set_int(s, "ich", av_get_channel_layout_nb_channels(s-> in_ch_layout), 0) < 0)
89 if (av_opt_set_int(s, "och", av_get_channel_layout_nb_channels(s->out_ch_layout), 0) < 0)
92 av_opt_set_int(s, "uch", 0, 0);
95 av_log(s, AV_LOG_ERROR, "Failed to set option\n");
100 static void set_audiodata_fmt(AudioData *a, enum AVSampleFormat fmt){
102 a->bps = av_get_bytes_per_sample(fmt);
103 a->planar= av_sample_fmt_is_planar(fmt);
104 if (a->ch_count == 1)
108 static void free_temp(AudioData *a){
110 memset(a, 0, sizeof(*a));
113 static void clear_context(SwrContext *s){
114 s->in_buffer_index= 0;
115 s->in_buffer_count= 0;
116 s->resample_in_constraint= 0;
117 memset(s->in.ch, 0, sizeof(s->in.ch));
118 memset(s->out.ch, 0, sizeof(s->out.ch));
119 free_temp(&s->postin);
120 free_temp(&s->midbuf);
121 free_temp(&s->preout);
122 free_temp(&s->in_buffer);
123 free_temp(&s->silence);
124 free_temp(&s->drop_temp);
125 free_temp(&s->dither.noise);
126 free_temp(&s->dither.temp);
127 swri_audio_convert_free(&s-> in_convert);
128 swri_audio_convert_free(&s->out_convert);
129 swri_audio_convert_free(&s->full_convert);
130 swri_rematrix_free(s);
135 av_cold void swr_free(SwrContext **ss){
140 s->resampler->free(&s->resample);
146 av_cold void swr_close(SwrContext *s){
150 av_cold int swr_init(struct SwrContext *s){
155 if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
156 av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt);
157 return AVERROR(EINVAL);
159 if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){
160 av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt);
161 return AVERROR(EINVAL);
164 if(av_get_channel_layout_nb_channels(s-> in_ch_layout) > SWR_CH_MAX) {
165 av_log(s, AV_LOG_WARNING, "Input channel layout 0x%"PRIx64" is invalid or unsupported.\n", s-> in_ch_layout);
169 if(av_get_channel_layout_nb_channels(s->out_ch_layout) > SWR_CH_MAX) {
170 av_log(s, AV_LOG_WARNING, "Output channel layout 0x%"PRIx64" is invalid or unsupported.\n", s->out_ch_layout);
171 s->out_ch_layout = 0;
176 extern struct Resampler const soxr_resampler;
177 case SWR_ENGINE_SOXR: s->resampler = &soxr_resampler; break;
179 case SWR_ENGINE_SWR : s->resampler = &swri_resampler; break;
181 av_log(s, AV_LOG_ERROR, "Requested resampling engine is unavailable\n");
182 return AVERROR(EINVAL);
185 if(!s->used_ch_count)
186 s->used_ch_count= s->in.ch_count;
188 if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){
189 av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");
193 if(!s-> in_ch_layout)
194 s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count);
195 if(!s->out_ch_layout)
196 s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count);
198 s->rematrix= s->out_ch_layout !=s->in_ch_layout || s->rematrix_volume!=1.0 ||
201 if(s->int_sample_fmt == AV_SAMPLE_FMT_NONE){
202 if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_S16P){
203 s->int_sample_fmt= AV_SAMPLE_FMT_S16P;
204 }else if( av_get_planar_sample_fmt(s-> in_sample_fmt) == AV_SAMPLE_FMT_S32P
205 && av_get_planar_sample_fmt(s->out_sample_fmt) == AV_SAMPLE_FMT_S32P
207 && s->engine != SWR_ENGINE_SOXR){
208 s->int_sample_fmt= AV_SAMPLE_FMT_S32P;
209 }else if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_FLTP){
210 s->int_sample_fmt= AV_SAMPLE_FMT_FLTP;
212 av_log(s, AV_LOG_DEBUG, "Using double precision mode\n");
213 s->int_sample_fmt= AV_SAMPLE_FMT_DBLP;
217 if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P
218 &&s->int_sample_fmt != AV_SAMPLE_FMT_S32P
219 &&s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
220 &&s->int_sample_fmt != AV_SAMPLE_FMT_DBLP){
221 av_log(s, AV_LOG_ERROR, "Requested sample format %s is not supported internally, S16/S32/FLT/DBL is supported\n", av_get_sample_fmt_name(s->int_sample_fmt));
222 return AVERROR(EINVAL);
225 set_audiodata_fmt(&s-> in, s-> in_sample_fmt);
226 set_audiodata_fmt(&s->out, s->out_sample_fmt);
228 if (s->firstpts_in_samples != AV_NOPTS_VALUE) {
229 if (!s->async && s->min_compensation >= FLT_MAX/2)
232 s->outpts = s->firstpts_in_samples * s->out_sample_rate;
234 s->firstpts = AV_NOPTS_VALUE;
237 if (s->min_compensation >= FLT_MAX/2)
238 s->min_compensation = 0.001;
239 if (s->async > 1.0001) {
240 s->max_soft_compensation = s->async / (double) s->in_sample_rate;
244 if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
245 s->resample = s->resampler->init(s->resample, s->out_sample_rate, s->in_sample_rate, s->filter_size, s->phase_shift, s->linear_interp, s->cutoff, s->int_sample_fmt, s->filter_type, s->kaiser_beta, s->precision, s->cheby);
247 s->resampler->free(&s->resample);
248 if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P
249 && s->int_sample_fmt != AV_SAMPLE_FMT_S32P
250 && s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
251 && s->int_sample_fmt != AV_SAMPLE_FMT_DBLP
253 av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt/dbl\n");
257 #define RSC 1 //FIXME finetune
259 s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
260 if(!s->used_ch_count)
261 s->used_ch_count= s->in.ch_count;
263 s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
265 if(!s-> in.ch_count){
266 av_assert0(!s->in_ch_layout);
267 av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
271 if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {
272 char l1[1024], l2[1024];
273 av_get_channel_layout_string(l1, sizeof(l1), s-> in.ch_count, s-> in_ch_layout);
274 av_get_channel_layout_string(l2, sizeof(l2), s->out.ch_count, s->out_ch_layout);
275 av_log(s, AV_LOG_ERROR, "Rematrix is needed between %s and %s "
276 "but there is not enough information to do it\n", l1, l2);
280 av_assert0(s->used_ch_count);
281 av_assert0(s->out.ch_count);
282 s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
286 s->drop_temp= s->out;
288 if(!s->resample && !s->rematrix && !s->channel_map && !s->dither.method){
289 s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,
290 s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
294 s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt,
295 s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
296 s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt,
297 s->int_sample_fmt, s->out.ch_count, NULL, 0);
299 if (!s->in_convert || !s->out_convert)
300 return AVERROR(ENOMEM);
308 s->midbuf.ch_count= s->used_ch_count;
310 s->in_buffer.ch_count= s->used_ch_count;
312 if(!s->resample_first){
313 s->midbuf.ch_count= s->out.ch_count;
315 s->in_buffer.ch_count = s->out.ch_count;
318 set_audiodata_fmt(&s->postin, s->int_sample_fmt);
319 set_audiodata_fmt(&s->midbuf, s->int_sample_fmt);
320 set_audiodata_fmt(&s->preout, s->int_sample_fmt);
323 set_audiodata_fmt(&s->in_buffer, s->int_sample_fmt);
326 if ((ret = swri_dither_init(s, s->out_sample_fmt, s->int_sample_fmt)) < 0)
329 if(s->rematrix || s->dither.method)
330 return swri_rematrix_init(s);
335 int swri_realloc_audio(AudioData *a, int count){
339 if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)
340 return AVERROR(EINVAL);
342 if(a->count >= count)
347 countb= FFALIGN(count*a->bps, ALIGN);
351 av_assert0(a->ch_count);
353 a->data= av_mallocz(countb*a->ch_count);
355 return AVERROR(ENOMEM);
356 for(i=0; i<a->ch_count; i++){
357 a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
358 if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
360 if(!a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);
367 static void copy(AudioData *out, AudioData *in,
369 av_assert0(out->planar == in->planar);
370 av_assert0(out->bps == in->bps);
371 av_assert0(out->ch_count == in->ch_count);
374 for(ch=0; ch<out->ch_count; ch++)
375 memcpy(out->ch[ch], in->ch[ch], count*out->bps);
377 memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
380 static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
383 memset(out->ch, 0, sizeof(out->ch));
384 }else if(out->planar){
385 for(i=0; i<out->ch_count; i++)
386 out->ch[i]= in_arg[i];
388 for(i=0; i<out->ch_count; i++)
389 out->ch[i]= in_arg[0] + i*out->bps;
393 static void reversefill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
396 for(i=0; i<out->ch_count; i++)
397 in_arg[i]= out->ch[i];
399 in_arg[0]= out->ch[0];
405 * out may be equal in.
407 static void buf_set(AudioData *out, AudioData *in, int count){
410 for(ch=0; ch<out->ch_count; ch++)
411 out->ch[ch]= in->ch[ch] + count*out->bps;
413 for(ch=out->ch_count-1; ch>=0; ch--)
414 out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps;
420 * @return number of samples output per channel
422 static int resample(SwrContext *s, AudioData *out_param, int out_count,
423 const AudioData * in_param, int in_count){
424 AudioData in, out, tmp;
427 int padless = ARCH_X86 && s->engine == SWR_ENGINE_SWR ? 7 : 0;
429 av_assert1(s->in_buffer.ch_count == in_param->ch_count);
430 av_assert1(s->in_buffer.planar == in_param->planar);
431 av_assert1(s->in_buffer.fmt == in_param->fmt);
436 border = s->resampler->invert_initial_buffer(s->resample, &s->in_buffer,
437 &in, in_count, &s->in_buffer_index, &s->in_buffer_count);
438 if (border == INT_MAX) {
440 } else if (border < 0) {
443 buf_set(&in, &in, border);
445 s->resample_in_constraint = 0;
449 int ret, size, consumed;
450 if(!s->resample_in_constraint && s->in_buffer_count){
451 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
452 ret= s->resampler->multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
455 buf_set(&out, &out, ret);
456 s->in_buffer_count -= consumed;
457 s->in_buffer_index += consumed;
461 if(s->in_buffer_count <= border){
462 buf_set(&in, &in, -s->in_buffer_count);
463 in_count += s->in_buffer_count;
464 s->in_buffer_count=0;
465 s->in_buffer_index=0;
470 if((s->flushed || in_count > padless) && !s->in_buffer_count){
471 s->in_buffer_index=0;
472 ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, FFMAX(in_count-padless, 0), &consumed);
475 buf_set(&out, &out, ret);
476 in_count -= consumed;
477 buf_set(&in, &in, consumed);
480 //TODO is this check sane considering the advanced copy avoidance below
481 size= s->in_buffer_index + s->in_buffer_count + in_count;
482 if( size > s->in_buffer.count
483 && s->in_buffer_count + in_count <= s->in_buffer_index){
484 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
485 copy(&s->in_buffer, &tmp, s->in_buffer_count);
486 s->in_buffer_index=0;
488 if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
493 if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
495 buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
496 copy(&tmp, &in, /*in_*/count);
497 s->in_buffer_count += count;
500 buf_set(&in, &in, count);
501 s->resample_in_constraint= 0;
502 if(s->in_buffer_count != count || in_count)
512 s->resample_in_constraint= !!out_count;
517 static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,
518 AudioData *in , int in_count){
519 AudioData *postin, *midbuf, *preout;
521 AudioData preout_tmp, midbuf_tmp;
524 av_assert0(!s->resample);
525 swri_audio_convert(s->full_convert, out, in, in_count);
529 // in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
530 // in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
532 if((ret=swri_realloc_audio(&s->postin, in_count))<0)
534 if(s->resample_first){
535 av_assert0(s->midbuf.ch_count == s->used_ch_count);
536 if((ret=swri_realloc_audio(&s->midbuf, out_count))<0)
539 av_assert0(s->midbuf.ch_count == s->out.ch_count);
540 if((ret=swri_realloc_audio(&s->midbuf, in_count))<0)
543 if((ret=swri_realloc_audio(&s->preout, out_count))<0)
548 midbuf_tmp= s->midbuf;
550 preout_tmp= s->preout;
553 if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map)
556 if(s->resample_first ? !s->resample : !s->rematrix)
559 if(s->resample_first ? !s->rematrix : !s->resample)
562 if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar
563 && !(s->out_sample_fmt==AV_SAMPLE_FMT_S32P && (s->dither.output_sample_bits&31))){
565 out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant
566 av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
567 copy(out, in, out_count);
570 else if(preout==postin) preout= midbuf= postin= out;
571 else if(preout==midbuf) preout= midbuf= out;
576 swri_audio_convert(s->in_convert, postin, in, in_count);
579 if(s->resample_first){
581 out_count= resample(s, midbuf, out_count, postin, in_count);
583 swri_rematrix(s, preout, midbuf, out_count, preout==out);
586 swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
588 out_count= resample(s, preout, out_count, midbuf, in_count);
591 if(preout != out && out_count){
592 AudioData *conv_src = preout;
593 if(s->dither.method){
595 int dither_count= FFMAX(out_count, 1<<16);
598 conv_src = &s->dither.temp;
599 if((ret=swri_realloc_audio(&s->dither.temp, dither_count))<0)
603 if((ret=swri_realloc_audio(&s->dither.noise, dither_count))<0)
606 for(ch=0; ch<s->dither.noise.ch_count; ch++)
607 swri_get_dither(s, s->dither.noise.ch[ch], s->dither.noise.count, 12345678913579<<ch, s->dither.noise.fmt);
608 av_assert0(s->dither.noise.ch_count == preout->ch_count);
610 if(s->dither.noise_pos + out_count > s->dither.noise.count)
611 s->dither.noise_pos = 0;
613 if (s->dither.method < SWR_DITHER_NS){
614 if (s->mix_2_1_simd) {
615 int len1= out_count&~15;
616 int off = len1 * preout->bps;
619 for(ch=0; ch<preout->ch_count; ch++)
620 s->mix_2_1_simd(conv_src->ch[ch], preout->ch[ch], s->dither.noise.ch[ch] + s->dither.noise.bps * s->dither.noise_pos, s->native_simd_one, 0, 0, len1);
621 if(out_count != len1)
622 for(ch=0; ch<preout->ch_count; ch++)
623 s->mix_2_1_f(conv_src->ch[ch] + off, preout->ch[ch] + off, s->dither.noise.ch[ch] + s->dither.noise.bps * s->dither.noise_pos + off + len1, s->native_one, 0, 0, out_count - len1);
625 for(ch=0; ch<preout->ch_count; ch++)
626 s->mix_2_1_f(conv_src->ch[ch], preout->ch[ch], s->dither.noise.ch[ch] + s->dither.noise.bps * s->dither.noise_pos, s->native_one, 0, 0, out_count);
629 switch(s->int_sample_fmt) {
630 case AV_SAMPLE_FMT_S16P :swri_noise_shaping_int16(s, conv_src, preout, &s->dither.noise, out_count); break;
631 case AV_SAMPLE_FMT_S32P :swri_noise_shaping_int32(s, conv_src, preout, &s->dither.noise, out_count); break;
632 case AV_SAMPLE_FMT_FLTP :swri_noise_shaping_float(s, conv_src, preout, &s->dither.noise, out_count); break;
633 case AV_SAMPLE_FMT_DBLP :swri_noise_shaping_double(s,conv_src, preout, &s->dither.noise, out_count); break;
636 s->dither.noise_pos += out_count;
638 //FIXME packed doesn't need more than 1 chan here!
639 swri_audio_convert(s->out_convert, out, conv_src, out_count);
644 int swr_is_initialized(struct SwrContext *s) {
645 return !!s->in_buffer.ch_count;
648 int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
649 const uint8_t *in_arg [SWR_CH_MAX], int in_count){
650 AudioData * in= &s->in;
651 AudioData *out= &s->out;
653 if (!swr_is_initialized(s)) {
654 av_log(s, AV_LOG_ERROR, "Context has not been initialized\n");
655 return AVERROR(EINVAL);
658 while(s->drop_output > 0){
660 uint8_t *tmp_arg[SWR_CH_MAX];
661 #define MAX_DROP_STEP 16384
662 if((ret=swri_realloc_audio(&s->drop_temp, FFMIN(s->drop_output, MAX_DROP_STEP)))<0)
665 reversefill_audiodata(&s->drop_temp, tmp_arg);
666 s->drop_output *= -1; //FIXME find a less hackish solution
667 ret = swr_convert(s, tmp_arg, FFMIN(-s->drop_output, MAX_DROP_STEP), in_arg, in_count); //FIXME optimize but this is as good as never called so maybe it doesn't matter
668 s->drop_output *= -1;
671 s->drop_output -= ret;
672 if (!s->drop_output && !out_arg)
677 av_assert0(s->drop_output);
684 s->resampler->flush(s);
685 s->resample_in_constraint = 0;
687 }else if(!s->in_buffer_count){
691 fill_audiodata(in , (void*)in_arg);
693 fill_audiodata(out, out_arg);
696 int ret = swr_convert_internal(s, out, out_count, in, in_count);
697 if(ret>0 && !s->drop_output)
698 s->outpts += ret * (int64_t)s->in_sample_rate;
704 size = FFMIN(out_count, s->in_buffer_count);
706 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
707 ret= swr_convert_internal(s, out, size, &tmp, size);
711 s->in_buffer_count -= ret;
712 s->in_buffer_index += ret;
713 buf_set(out, out, ret);
715 if(!s->in_buffer_count)
716 s->in_buffer_index = 0;
720 size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
722 if(in_count > out_count) { //FIXME move after swr_convert_internal
723 if( size > s->in_buffer.count
724 && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
725 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
726 copy(&s->in_buffer, &tmp, s->in_buffer_count);
727 s->in_buffer_index=0;
729 if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
734 size = FFMIN(in_count, out_count);
735 ret= swr_convert_internal(s, out, size, in, size);
738 buf_set(in, in, ret);
743 buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
744 copy(&tmp, in, in_count);
745 s->in_buffer_count += in_count;
748 if(ret2>0 && !s->drop_output)
749 s->outpts += ret2 * (int64_t)s->in_sample_rate;
754 int swr_drop_output(struct SwrContext *s, int count){
755 s->drop_output += count;
757 if(s->drop_output <= 0)
760 av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);
761 return swr_convert(s, NULL, s->drop_output, NULL, 0);
764 int swr_inject_silence(struct SwrContext *s, int count){
766 uint8_t *tmp_arg[SWR_CH_MAX];
771 #define MAX_SILENCE_STEP 16384
772 while (count > MAX_SILENCE_STEP) {
773 if ((ret = swr_inject_silence(s, MAX_SILENCE_STEP)) < 0)
775 count -= MAX_SILENCE_STEP;
778 if((ret=swri_realloc_audio(&s->silence, count))<0)
781 if(s->silence.planar) for(i=0; i<s->silence.ch_count; i++) {
782 memset(s->silence.ch[i], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps);
784 memset(s->silence.ch[0], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps*s->silence.ch_count);
786 reversefill_audiodata(&s->silence, tmp_arg);
787 av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);
788 ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);
792 int64_t swr_get_delay(struct SwrContext *s, int64_t base){
793 if (s->resampler && s->resample){
794 return s->resampler->get_delay(s, base);
796 return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate;
800 int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){
803 if (!s || compensation_distance < 0)
804 return AVERROR(EINVAL);
805 if (!compensation_distance && sample_delta)
806 return AVERROR(EINVAL);
808 s->flags |= SWR_FLAG_RESAMPLE;
813 if (!s->resampler->set_compensation){
814 return AVERROR(EINVAL);
816 return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance);
820 int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
824 if (s->firstpts == AV_NOPTS_VALUE)
825 s->outpts = s->firstpts = pts;
827 if(s->min_compensation >= FLT_MAX) {
828 return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
830 int64_t delta = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate) - s->outpts + s->drop_output*(int64_t)s->in_sample_rate;
831 double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
833 if(fabs(fdelta) > s->min_compensation) {
834 if(s->outpts == s->firstpts || fabs(fdelta) > s->min_hard_compensation){
836 if(delta > 0) ret = swr_inject_silence(s, delta / s->out_sample_rate);
837 else ret = swr_drop_output (s, -delta / s-> in_sample_rate);
839 av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
841 } else if(s->soft_compensation_duration && s->max_soft_compensation) {
842 int duration = s->out_sample_rate * s->soft_compensation_duration;
843 double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1);
844 int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;
845 av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
846 swr_set_compensation(s, comp, duration);