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 #include "libavutil/ffversion.h"
32 const char swr_ffversion[] = "FFmpeg version " FFMPEG_VERSION;
34 unsigned swresample_version(void)
36 av_assert0(LIBSWRESAMPLE_VERSION_MICRO >= 100);
37 return LIBSWRESAMPLE_VERSION_INT;
40 const char *swresample_configuration(void)
42 return FFMPEG_CONFIGURATION;
45 const char *swresample_license(void)
47 #define LICENSE_PREFIX "libswresample license: "
48 return LICENSE_PREFIX FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1;
51 int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map){
52 if(!s || s->in_convert) // s needs to be allocated but not initialized
53 return AVERROR(EINVAL);
54 s->channel_map = channel_map;
58 struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,
59 int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
60 int64_t in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate,
61 int log_offset, void *log_ctx){
62 if(!s) s= swr_alloc();
65 s->log_level_offset= log_offset;
68 if (av_opt_set_int(s, "ocl", out_ch_layout, 0) < 0)
71 if (av_opt_set_int(s, "osf", out_sample_fmt, 0) < 0)
74 if (av_opt_set_int(s, "osr", out_sample_rate, 0) < 0)
77 if (av_opt_set_int(s, "icl", in_ch_layout, 0) < 0)
80 if (av_opt_set_int(s, "isf", in_sample_fmt, 0) < 0)
83 if (av_opt_set_int(s, "isr", in_sample_rate, 0) < 0)
86 if (av_opt_set_int(s, "tsf", AV_SAMPLE_FMT_NONE, 0) < 0)
89 if (av_opt_set_int(s, "ich", av_get_channel_layout_nb_channels(s-> in_ch_layout), 0) < 0)
92 if (av_opt_set_int(s, "och", av_get_channel_layout_nb_channels(s->out_ch_layout), 0) < 0)
95 av_opt_set_int(s, "uch", 0, 0);
98 av_log(s, AV_LOG_ERROR, "Failed to set option\n");
103 static void set_audiodata_fmt(AudioData *a, enum AVSampleFormat fmt){
105 a->bps = av_get_bytes_per_sample(fmt);
106 a->planar= av_sample_fmt_is_planar(fmt);
107 if (a->ch_count == 1)
111 static void free_temp(AudioData *a){
113 memset(a, 0, sizeof(*a));
116 static void clear_context(SwrContext *s){
117 s->in_buffer_index= 0;
118 s->in_buffer_count= 0;
119 s->resample_in_constraint= 0;
120 memset(s->in.ch, 0, sizeof(s->in.ch));
121 memset(s->out.ch, 0, sizeof(s->out.ch));
122 free_temp(&s->postin);
123 free_temp(&s->midbuf);
124 free_temp(&s->preout);
125 free_temp(&s->in_buffer);
126 free_temp(&s->silence);
127 free_temp(&s->drop_temp);
128 free_temp(&s->dither.noise);
129 free_temp(&s->dither.temp);
130 swri_audio_convert_free(&s-> in_convert);
131 swri_audio_convert_free(&s->out_convert);
132 swri_audio_convert_free(&s->full_convert);
133 swri_rematrix_free(s);
138 av_cold void swr_free(SwrContext **ss){
143 s->resampler->free(&s->resample);
149 av_cold void swr_close(SwrContext *s){
153 av_cold int swr_init(struct SwrContext *s){
158 if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
159 av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt);
160 return AVERROR(EINVAL);
162 if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){
163 av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt);
164 return AVERROR(EINVAL);
167 if(av_get_channel_layout_nb_channels(s-> in_ch_layout) > SWR_CH_MAX) {
168 av_log(s, AV_LOG_WARNING, "Input channel layout 0x%"PRIx64" is invalid or unsupported.\n", s-> in_ch_layout);
172 if(av_get_channel_layout_nb_channels(s->out_ch_layout) > SWR_CH_MAX) {
173 av_log(s, AV_LOG_WARNING, "Output channel layout 0x%"PRIx64" is invalid or unsupported.\n", s->out_ch_layout);
174 s->out_ch_layout = 0;
179 extern struct Resampler const soxr_resampler;
180 case SWR_ENGINE_SOXR: s->resampler = &soxr_resampler; break;
182 case SWR_ENGINE_SWR : s->resampler = &swri_resampler; break;
184 av_log(s, AV_LOG_ERROR, "Requested resampling engine is unavailable\n");
185 return AVERROR(EINVAL);
188 if(!s->used_ch_count)
189 s->used_ch_count= s->in.ch_count;
191 if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){
192 av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");
196 if(!s-> in_ch_layout)
197 s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count);
198 if(!s->out_ch_layout)
199 s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count);
201 s->rematrix= s->out_ch_layout !=s->in_ch_layout || s->rematrix_volume!=1.0 ||
204 if(s->int_sample_fmt == AV_SAMPLE_FMT_NONE){
205 if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_S16P){
206 s->int_sample_fmt= AV_SAMPLE_FMT_S16P;
207 }else if( av_get_planar_sample_fmt(s-> in_sample_fmt) == AV_SAMPLE_FMT_S32P
208 && av_get_planar_sample_fmt(s->out_sample_fmt) == AV_SAMPLE_FMT_S32P
210 && s->engine != SWR_ENGINE_SOXR){
211 s->int_sample_fmt= AV_SAMPLE_FMT_S32P;
212 }else if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_FLTP){
213 s->int_sample_fmt= AV_SAMPLE_FMT_FLTP;
215 av_log(s, AV_LOG_DEBUG, "Using double precision mode\n");
216 s->int_sample_fmt= AV_SAMPLE_FMT_DBLP;
220 if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P
221 &&s->int_sample_fmt != AV_SAMPLE_FMT_S32P
222 &&s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
223 &&s->int_sample_fmt != AV_SAMPLE_FMT_DBLP){
224 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));
225 return AVERROR(EINVAL);
228 set_audiodata_fmt(&s-> in, s-> in_sample_fmt);
229 set_audiodata_fmt(&s->out, s->out_sample_fmt);
231 if (s->firstpts_in_samples != AV_NOPTS_VALUE) {
232 if (!s->async && s->min_compensation >= FLT_MAX/2)
235 s->outpts = s->firstpts_in_samples * s->out_sample_rate;
237 s->firstpts = AV_NOPTS_VALUE;
240 if (s->min_compensation >= FLT_MAX/2)
241 s->min_compensation = 0.001;
242 if (s->async > 1.0001) {
243 s->max_soft_compensation = s->async / (double) s->in_sample_rate;
247 if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
248 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);
250 s->resampler->free(&s->resample);
251 if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P
252 && s->int_sample_fmt != AV_SAMPLE_FMT_S32P
253 && s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
254 && s->int_sample_fmt != AV_SAMPLE_FMT_DBLP
256 av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt/dbl\n");
260 #define RSC 1 //FIXME finetune
262 s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
263 if(!s->used_ch_count)
264 s->used_ch_count= s->in.ch_count;
266 s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
268 if(!s-> in.ch_count){
269 av_assert0(!s->in_ch_layout);
270 av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
274 if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {
275 char l1[1024], l2[1024];
276 av_get_channel_layout_string(l1, sizeof(l1), s-> in.ch_count, s-> in_ch_layout);
277 av_get_channel_layout_string(l2, sizeof(l2), s->out.ch_count, s->out_ch_layout);
278 av_log(s, AV_LOG_ERROR, "Rematrix is needed between %s and %s "
279 "but there is not enough information to do it\n", l1, l2);
283 av_assert0(s->used_ch_count);
284 av_assert0(s->out.ch_count);
285 s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
289 s->drop_temp= s->out;
291 if(!s->resample && !s->rematrix && !s->channel_map && !s->dither.method){
292 s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,
293 s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
297 s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt,
298 s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
299 s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt,
300 s->int_sample_fmt, s->out.ch_count, NULL, 0);
302 if (!s->in_convert || !s->out_convert)
303 return AVERROR(ENOMEM);
311 s->midbuf.ch_count= s->used_ch_count;
313 s->in_buffer.ch_count= s->used_ch_count;
315 if(!s->resample_first){
316 s->midbuf.ch_count= s->out.ch_count;
318 s->in_buffer.ch_count = s->out.ch_count;
321 set_audiodata_fmt(&s->postin, s->int_sample_fmt);
322 set_audiodata_fmt(&s->midbuf, s->int_sample_fmt);
323 set_audiodata_fmt(&s->preout, s->int_sample_fmt);
326 set_audiodata_fmt(&s->in_buffer, s->int_sample_fmt);
329 if ((ret = swri_dither_init(s, s->out_sample_fmt, s->int_sample_fmt)) < 0)
332 if(s->rematrix || s->dither.method)
333 return swri_rematrix_init(s);
338 int swri_realloc_audio(AudioData *a, int count){
342 if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)
343 return AVERROR(EINVAL);
345 if(a->count >= count)
350 countb= FFALIGN(count*a->bps, ALIGN);
354 av_assert0(a->ch_count);
356 a->data= av_mallocz(countb*a->ch_count);
358 return AVERROR(ENOMEM);
359 for(i=0; i<a->ch_count; i++){
360 a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
361 if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
363 if(!a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);
370 static void copy(AudioData *out, AudioData *in,
372 av_assert0(out->planar == in->planar);
373 av_assert0(out->bps == in->bps);
374 av_assert0(out->ch_count == in->ch_count);
377 for(ch=0; ch<out->ch_count; ch++)
378 memcpy(out->ch[ch], in->ch[ch], count*out->bps);
380 memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
383 static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
386 memset(out->ch, 0, sizeof(out->ch));
387 }else if(out->planar){
388 for(i=0; i<out->ch_count; i++)
389 out->ch[i]= in_arg[i];
391 for(i=0; i<out->ch_count; i++)
392 out->ch[i]= in_arg[0] + i*out->bps;
396 static void reversefill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
399 for(i=0; i<out->ch_count; i++)
400 in_arg[i]= out->ch[i];
402 in_arg[0]= out->ch[0];
408 * out may be equal in.
410 static void buf_set(AudioData *out, AudioData *in, int count){
413 for(ch=0; ch<out->ch_count; ch++)
414 out->ch[ch]= in->ch[ch] + count*out->bps;
416 for(ch=out->ch_count-1; ch>=0; ch--)
417 out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps;
423 * @return number of samples output per channel
425 static int resample(SwrContext *s, AudioData *out_param, int out_count,
426 const AudioData * in_param, int in_count){
427 AudioData in, out, tmp;
430 int padless = ARCH_X86 && s->engine == SWR_ENGINE_SWR ? 7 : 0;
432 av_assert1(s->in_buffer.ch_count == in_param->ch_count);
433 av_assert1(s->in_buffer.planar == in_param->planar);
434 av_assert1(s->in_buffer.fmt == in_param->fmt);
439 border = s->resampler->invert_initial_buffer(s->resample, &s->in_buffer,
440 &in, in_count, &s->in_buffer_index, &s->in_buffer_count);
441 if (border == INT_MAX) {
443 } else if (border < 0) {
446 buf_set(&in, &in, border);
448 s->resample_in_constraint = 0;
452 int ret, size, consumed;
453 if(!s->resample_in_constraint && s->in_buffer_count){
454 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
455 ret= s->resampler->multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
458 buf_set(&out, &out, ret);
459 s->in_buffer_count -= consumed;
460 s->in_buffer_index += consumed;
464 if(s->in_buffer_count <= border){
465 buf_set(&in, &in, -s->in_buffer_count);
466 in_count += s->in_buffer_count;
467 s->in_buffer_count=0;
468 s->in_buffer_index=0;
473 if((s->flushed || in_count > padless) && !s->in_buffer_count){
474 s->in_buffer_index=0;
475 ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, FFMAX(in_count-padless, 0), &consumed);
478 buf_set(&out, &out, ret);
479 in_count -= consumed;
480 buf_set(&in, &in, consumed);
483 //TODO is this check sane considering the advanced copy avoidance below
484 size= s->in_buffer_index + s->in_buffer_count + in_count;
485 if( size > s->in_buffer.count
486 && s->in_buffer_count + in_count <= s->in_buffer_index){
487 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
488 copy(&s->in_buffer, &tmp, s->in_buffer_count);
489 s->in_buffer_index=0;
491 if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
496 if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
498 buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
499 copy(&tmp, &in, /*in_*/count);
500 s->in_buffer_count += count;
503 buf_set(&in, &in, count);
504 s->resample_in_constraint= 0;
505 if(s->in_buffer_count != count || in_count)
515 s->resample_in_constraint= !!out_count;
520 static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,
521 AudioData *in , int in_count){
522 AudioData *postin, *midbuf, *preout;
524 AudioData preout_tmp, midbuf_tmp;
527 av_assert0(!s->resample);
528 swri_audio_convert(s->full_convert, out, in, in_count);
532 // in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
533 // in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
535 if((ret=swri_realloc_audio(&s->postin, in_count))<0)
537 if(s->resample_first){
538 av_assert0(s->midbuf.ch_count == s->used_ch_count);
539 if((ret=swri_realloc_audio(&s->midbuf, out_count))<0)
542 av_assert0(s->midbuf.ch_count == s->out.ch_count);
543 if((ret=swri_realloc_audio(&s->midbuf, in_count))<0)
546 if((ret=swri_realloc_audio(&s->preout, out_count))<0)
551 midbuf_tmp= s->midbuf;
553 preout_tmp= s->preout;
556 if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map)
559 if(s->resample_first ? !s->resample : !s->rematrix)
562 if(s->resample_first ? !s->rematrix : !s->resample)
565 if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar
566 && !(s->out_sample_fmt==AV_SAMPLE_FMT_S32P && (s->dither.output_sample_bits&31))){
568 out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant
569 av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
570 copy(out, in, out_count);
573 else if(preout==postin) preout= midbuf= postin= out;
574 else if(preout==midbuf) preout= midbuf= out;
579 swri_audio_convert(s->in_convert, postin, in, in_count);
582 if(s->resample_first){
584 out_count= resample(s, midbuf, out_count, postin, in_count);
586 swri_rematrix(s, preout, midbuf, out_count, preout==out);
589 swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
591 out_count= resample(s, preout, out_count, midbuf, in_count);
594 if(preout != out && out_count){
595 AudioData *conv_src = preout;
596 if(s->dither.method){
598 int dither_count= FFMAX(out_count, 1<<16);
601 conv_src = &s->dither.temp;
602 if((ret=swri_realloc_audio(&s->dither.temp, dither_count))<0)
606 if((ret=swri_realloc_audio(&s->dither.noise, dither_count))<0)
609 for(ch=0; ch<s->dither.noise.ch_count; ch++)
610 swri_get_dither(s, s->dither.noise.ch[ch], s->dither.noise.count, 12345678913579<<ch, s->dither.noise.fmt);
611 av_assert0(s->dither.noise.ch_count == preout->ch_count);
613 if(s->dither.noise_pos + out_count > s->dither.noise.count)
614 s->dither.noise_pos = 0;
616 if (s->dither.method < SWR_DITHER_NS){
617 if (s->mix_2_1_simd) {
618 int len1= out_count&~15;
619 int off = len1 * preout->bps;
622 for(ch=0; ch<preout->ch_count; ch++)
623 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);
624 if(out_count != len1)
625 for(ch=0; ch<preout->ch_count; ch++)
626 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);
628 for(ch=0; ch<preout->ch_count; ch++)
629 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);
632 switch(s->int_sample_fmt) {
633 case AV_SAMPLE_FMT_S16P :swri_noise_shaping_int16(s, conv_src, preout, &s->dither.noise, out_count); break;
634 case AV_SAMPLE_FMT_S32P :swri_noise_shaping_int32(s, conv_src, preout, &s->dither.noise, out_count); break;
635 case AV_SAMPLE_FMT_FLTP :swri_noise_shaping_float(s, conv_src, preout, &s->dither.noise, out_count); break;
636 case AV_SAMPLE_FMT_DBLP :swri_noise_shaping_double(s,conv_src, preout, &s->dither.noise, out_count); break;
639 s->dither.noise_pos += out_count;
641 //FIXME packed doesn't need more than 1 chan here!
642 swri_audio_convert(s->out_convert, out, conv_src, out_count);
647 int swr_is_initialized(struct SwrContext *s) {
648 return !!s->in_buffer.ch_count;
651 int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
652 const uint8_t *in_arg [SWR_CH_MAX], int in_count){
653 AudioData * in= &s->in;
654 AudioData *out= &s->out;
656 if (!swr_is_initialized(s)) {
657 av_log(s, AV_LOG_ERROR, "Context has not been initialized\n");
658 return AVERROR(EINVAL);
661 while(s->drop_output > 0){
663 uint8_t *tmp_arg[SWR_CH_MAX];
664 #define MAX_DROP_STEP 16384
665 if((ret=swri_realloc_audio(&s->drop_temp, FFMIN(s->drop_output, MAX_DROP_STEP)))<0)
668 reversefill_audiodata(&s->drop_temp, tmp_arg);
669 s->drop_output *= -1; //FIXME find a less hackish solution
670 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
671 s->drop_output *= -1;
674 s->drop_output -= ret;
675 if (!s->drop_output && !out_arg)
680 av_assert0(s->drop_output);
687 s->resampler->flush(s);
688 s->resample_in_constraint = 0;
690 }else if(!s->in_buffer_count){
694 fill_audiodata(in , (void*)in_arg);
696 fill_audiodata(out, out_arg);
699 int ret = swr_convert_internal(s, out, out_count, in, in_count);
700 if(ret>0 && !s->drop_output)
701 s->outpts += ret * (int64_t)s->in_sample_rate;
707 size = FFMIN(out_count, s->in_buffer_count);
709 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
710 ret= swr_convert_internal(s, out, size, &tmp, size);
714 s->in_buffer_count -= ret;
715 s->in_buffer_index += ret;
716 buf_set(out, out, ret);
718 if(!s->in_buffer_count)
719 s->in_buffer_index = 0;
723 size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
725 if(in_count > out_count) { //FIXME move after swr_convert_internal
726 if( size > s->in_buffer.count
727 && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
728 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
729 copy(&s->in_buffer, &tmp, s->in_buffer_count);
730 s->in_buffer_index=0;
732 if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
737 size = FFMIN(in_count, out_count);
738 ret= swr_convert_internal(s, out, size, in, size);
741 buf_set(in, in, ret);
746 buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
747 copy(&tmp, in, in_count);
748 s->in_buffer_count += in_count;
751 if(ret2>0 && !s->drop_output)
752 s->outpts += ret2 * (int64_t)s->in_sample_rate;
757 int swr_drop_output(struct SwrContext *s, int count){
758 const uint8_t *tmp_arg[SWR_CH_MAX];
759 s->drop_output += count;
761 if(s->drop_output <= 0)
764 av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);
765 return swr_convert(s, NULL, s->drop_output, tmp_arg, 0);
768 int swr_inject_silence(struct SwrContext *s, int count){
770 uint8_t *tmp_arg[SWR_CH_MAX];
775 #define MAX_SILENCE_STEP 16384
776 while (count > MAX_SILENCE_STEP) {
777 if ((ret = swr_inject_silence(s, MAX_SILENCE_STEP)) < 0)
779 count -= MAX_SILENCE_STEP;
782 if((ret=swri_realloc_audio(&s->silence, count))<0)
785 if(s->silence.planar) for(i=0; i<s->silence.ch_count; i++) {
786 memset(s->silence.ch[i], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps);
788 memset(s->silence.ch[0], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps*s->silence.ch_count);
790 reversefill_audiodata(&s->silence, tmp_arg);
791 av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);
792 ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);
796 int64_t swr_get_delay(struct SwrContext *s, int64_t base){
797 if (s->resampler && s->resample){
798 return s->resampler->get_delay(s, base);
800 return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate;
804 int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){
807 if (!s || compensation_distance < 0)
808 return AVERROR(EINVAL);
809 if (!compensation_distance && sample_delta)
810 return AVERROR(EINVAL);
812 s->flags |= SWR_FLAG_RESAMPLE;
817 if (!s->resampler->set_compensation){
818 return AVERROR(EINVAL);
820 return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance);
824 int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
828 if (s->firstpts == AV_NOPTS_VALUE)
829 s->outpts = s->firstpts = pts;
831 if(s->min_compensation >= FLT_MAX) {
832 return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
834 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;
835 double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
837 if(fabs(fdelta) > s->min_compensation) {
838 if(s->outpts == s->firstpts || fabs(fdelta) > s->min_hard_compensation){
840 if(delta > 0) ret = swr_inject_silence(s, delta / s->out_sample_rate);
841 else ret = swr_drop_output (s, -delta / s-> in_sample_rate);
843 av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
845 } else if(s->soft_compensation_duration && s->max_soft_compensation) {
846 int duration = s->out_sample_rate * s->soft_compensation_duration;
847 double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1);
848 int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;
849 av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
850 swr_set_compensation(s, comp, duration);