2 * Copyright (C) 2011 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/audioconvert.h"
30 #define C15DB 1.189207115
32 #define C_15DB 0.840896415
33 #define C_30DB M_SQRT1_2
34 #define C_45DB 0.594603558
39 //TODO split options array out?
40 #define OFFSET(x) offsetof(SwrContext,x)
41 #define PARAM AV_OPT_FLAG_AUDIO_PARAM
43 static const AVOption options[]={
44 {"ich" , "Input Channel Count" , OFFSET( in.ch_count ), AV_OPT_TYPE_INT , {.dbl=2 }, 0 , SWR_CH_MAX, PARAM},
45 {"in_channel_count" , "Input Channel Count" , OFFSET( in.ch_count ), AV_OPT_TYPE_INT , {.dbl=2 }, 0 , SWR_CH_MAX, PARAM},
46 {"och" , "Output Channel Count" , OFFSET(out.ch_count ), AV_OPT_TYPE_INT , {.dbl=2 }, 0 , SWR_CH_MAX, PARAM},
47 {"out_channel_count" , "Output Channel Count" , OFFSET(out.ch_count ), AV_OPT_TYPE_INT , {.dbl=2 }, 0 , SWR_CH_MAX, PARAM},
48 {"uch" , "Used Channel Count" , OFFSET(used_ch_count ), AV_OPT_TYPE_INT , {.dbl=0 }, 0 , SWR_CH_MAX, PARAM},
49 {"used_channel_count" , "Used Channel Count" , OFFSET(used_ch_count ), AV_OPT_TYPE_INT , {.dbl=0 }, 0 , SWR_CH_MAX, PARAM},
50 {"isr" , "Input Sample Rate" , OFFSET( in_sample_rate), AV_OPT_TYPE_INT , {.dbl=0 }, 0 , INT_MAX , PARAM},
51 {"in_sample_rate" , "Input Sample Rate" , OFFSET( in_sample_rate), AV_OPT_TYPE_INT , {.dbl=0 }, 0 , INT_MAX , PARAM},
52 {"osr" , "Output Sample Rate" , OFFSET(out_sample_rate), AV_OPT_TYPE_INT , {.dbl=0 }, 0 , INT_MAX , PARAM},
53 {"out_sample_rate" , "Output Sample Rate" , OFFSET(out_sample_rate), AV_OPT_TYPE_INT , {.dbl=0 }, 0 , INT_MAX , PARAM},
54 {"isf" , "Input Sample Format" , OFFSET( in_sample_fmt ), AV_OPT_TYPE_INT , {.dbl=AV_SAMPLE_FMT_NONE }, -1 , AV_SAMPLE_FMT_NB-1+256, PARAM},
55 {"in_sample_fmt" , "Input Sample Format" , OFFSET( in_sample_fmt ), AV_OPT_TYPE_INT , {.dbl=AV_SAMPLE_FMT_NONE }, -1 , AV_SAMPLE_FMT_NB-1+256, PARAM},
56 {"osf" , "Output Sample Format" , OFFSET(out_sample_fmt ), AV_OPT_TYPE_INT , {.dbl=AV_SAMPLE_FMT_NONE }, -1 , AV_SAMPLE_FMT_NB-1+256, PARAM},
57 {"out_sample_fmt" , "Output Sample Format" , OFFSET(out_sample_fmt ), AV_OPT_TYPE_INT , {.dbl=AV_SAMPLE_FMT_NONE }, -1 , AV_SAMPLE_FMT_NB-1+256, PARAM},
58 {"tsf" , "Internal Sample Format" , OFFSET(int_sample_fmt ), AV_OPT_TYPE_INT , {.dbl=AV_SAMPLE_FMT_NONE }, -1 , AV_SAMPLE_FMT_FLTP, PARAM},
59 {"internal_sample_fmt" , "Internal Sample Format" , OFFSET(int_sample_fmt ), AV_OPT_TYPE_INT , {.dbl=AV_SAMPLE_FMT_NONE }, -1 , AV_SAMPLE_FMT_FLTP, PARAM},
60 {"icl" , "Input Channel Layout" , OFFSET( in_ch_layout ), AV_OPT_TYPE_INT64, {.dbl=0 }, 0 , INT64_MAX , PARAM, "channel_layout"},
61 {"in_channel_layout" , "Input Channel Layout" , OFFSET( in_ch_layout ), AV_OPT_TYPE_INT64, {.dbl=0 }, 0 , INT64_MAX , PARAM, "channel_layout"},
62 {"ocl" , "Output Channel Layout" , OFFSET(out_ch_layout ), AV_OPT_TYPE_INT64, {.dbl=0 }, 0 , INT64_MAX , PARAM, "channel_layout"},
63 {"out_channel_layout" , "Output Channel Layout" , OFFSET(out_ch_layout ), AV_OPT_TYPE_INT64, {.dbl=0 }, 0 , INT64_MAX , PARAM, "channel_layout"},
64 {"clev" , "Center Mix Level" , OFFSET(clev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM},
65 {"center_mix_level" , "Center Mix Level" , OFFSET(clev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM},
66 {"slev" , "Sourround Mix Level" , OFFSET(slev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM},
67 {"surround_mix_level" , "Sourround Mix Level" , OFFSET(slev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM},
68 {"lfe_mix_level" , "LFE Mix Level" , OFFSET(lfe_mix_level ), AV_OPT_TYPE_FLOAT, {.dbl=0 }, -32 , 32 , PARAM},
69 {"rmvol" , "Rematrix Volume" , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0 }, -1000 , 1000 , PARAM},
70 {"rematrix_volume" , "Rematrix Volume" , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0 }, -1000 , 1000 , PARAM},
71 {"flags" , NULL , OFFSET(flags ), AV_OPT_TYPE_FLAGS, {.dbl=0 }, 0 , UINT_MAX , PARAM, "flags"},
72 {"swr_flags" , NULL , OFFSET(flags ), AV_OPT_TYPE_FLAGS, {.dbl=0 }, 0 , UINT_MAX , PARAM, "flags"},
73 {"res" , "Force Resampling" , 0 , AV_OPT_TYPE_CONST, {.dbl=SWR_FLAG_RESAMPLE }, INT_MIN, INT_MAX , PARAM, "flags"},
74 {"dither_scale" , "Dither Scale" , OFFSET(dither_scale ), AV_OPT_TYPE_FLOAT, {.dbl=1 }, 0 , INT_MAX , PARAM},
75 {"dither_method" , "Dither Method" , OFFSET(dither_method ), AV_OPT_TYPE_INT , {.dbl=0 }, 0 , SWR_DITHER_NB-1, PARAM, "dither_method"},
76 {"rectangular" , "Rectangular Dither" , 0 , AV_OPT_TYPE_CONST, {.dbl=SWR_DITHER_RECTANGULAR}, INT_MIN, INT_MAX , PARAM, "dither_method"},
77 {"triangular" , "Triangular Dither" , 0 , AV_OPT_TYPE_CONST, {.dbl=SWR_DITHER_TRIANGULAR }, INT_MIN, INT_MAX , PARAM, "dither_method"},
78 {"triangular_hp" , "Triangular Dither With High Pass" , 0 , AV_OPT_TYPE_CONST, {.dbl=SWR_DITHER_TRIANGULAR_HIGHPASS }, INT_MIN, INT_MAX, PARAM, "dither_method"},
79 {"filter_size" , "Resampling Filter Size" , OFFSET(filter_size) , AV_OPT_TYPE_INT , {.dbl=16 }, 0 , INT_MAX , PARAM },
80 {"phase_shift" , "Resampling Phase Shift" , OFFSET(phase_shift) , AV_OPT_TYPE_INT , {.dbl=10 }, 0 , 30 , PARAM },
81 {"linear_interp" , "Use Linear Interpolation" , OFFSET(linear_interp) , AV_OPT_TYPE_INT , {.dbl=0 }, 0 , 1 , PARAM },
82 {"cutoff" , "Cutoff Frequency Ratio" , OFFSET(cutoff) , AV_OPT_TYPE_DOUBLE,{.dbl=0.8 }, 0 , 1 , PARAM },
83 {"min_comp" , "Minimum difference between timestamps and audio data (in seconds) below which no timestamp compensation of either kind is applied"
84 , OFFSET(min_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=FLT_MAX }, 0 , FLT_MAX , PARAM },
85 {"min_hard_comp" , "Minimum difference between timestamps and audio data (in seconds) to trigger padding/trimming the data."
86 , OFFSET(min_hard_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=0.1 }, 0 , INT_MAX , PARAM },
87 {"comp_duration" , "Duration (in seconds) over which data is stretched/squeezeed to make it match the timestamps."
88 , OFFSET(soft_compensation_duration),AV_OPT_TYPE_FLOAT ,{.dbl=1 }, 0 , INT_MAX , PARAM },
89 {"max_soft_comp" , "Maximum factor by which data is stretched/squeezeed to make it match the timestamps."
90 , OFFSET(max_soft_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=0 }, INT_MIN, INT_MAX , PARAM },
95 static const char* context_to_name(void* ptr) {
99 static const AVClass av_class = {
100 .class_name = "SwrContext",
101 .item_name = context_to_name,
103 .version = LIBAVUTIL_VERSION_INT,
104 .log_level_offset_offset = OFFSET(log_level_offset),
105 .parent_log_context_offset = OFFSET(log_ctx),
108 unsigned swresample_version(void)
110 av_assert0(LIBSWRESAMPLE_VERSION_MICRO >= 100);
111 return LIBSWRESAMPLE_VERSION_INT;
114 const char *swresample_configuration(void)
116 return FFMPEG_CONFIGURATION;
119 const char *swresample_license(void)
121 #define LICENSE_PREFIX "libswresample license: "
122 return LICENSE_PREFIX FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1;
125 int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map){
126 if(!s || s->in_convert) // s needs to be allocated but not initialized
127 return AVERROR(EINVAL);
128 s->channel_map = channel_map;
132 const AVClass *swr_get_class(void)
137 struct SwrContext *swr_alloc(void){
138 SwrContext *s= av_mallocz(sizeof(SwrContext));
140 s->av_class= &av_class;
141 av_opt_set_defaults(s);
146 struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,
147 int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
148 int64_t in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate,
149 int log_offset, void *log_ctx){
150 if(!s) s= swr_alloc();
153 s->log_level_offset= log_offset;
156 av_opt_set_int(s, "ocl", out_ch_layout, 0);
157 av_opt_set_int(s, "osf", out_sample_fmt, 0);
158 av_opt_set_int(s, "osr", out_sample_rate, 0);
159 av_opt_set_int(s, "icl", in_ch_layout, 0);
160 av_opt_set_int(s, "isf", in_sample_fmt, 0);
161 av_opt_set_int(s, "isr", in_sample_rate, 0);
162 av_opt_set_int(s, "tsf", AV_SAMPLE_FMT_NONE, 0);
163 av_opt_set_int(s, "ich", av_get_channel_layout_nb_channels(s-> in_ch_layout), 0);
164 av_opt_set_int(s, "och", av_get_channel_layout_nb_channels(s->out_ch_layout), 0);
165 av_opt_set_int(s, "uch", 0, 0);
169 static void set_audiodata_fmt(AudioData *a, enum AVSampleFormat fmt){
171 a->bps = av_get_bytes_per_sample(fmt);
172 a->planar= av_sample_fmt_is_planar(fmt);
175 static void free_temp(AudioData *a){
177 memset(a, 0, sizeof(*a));
180 void swr_free(SwrContext **ss){
183 free_temp(&s->postin);
184 free_temp(&s->midbuf);
185 free_temp(&s->preout);
186 free_temp(&s->in_buffer);
187 free_temp(&s->dither);
188 swri_audio_convert_free(&s-> in_convert);
189 swri_audio_convert_free(&s->out_convert);
190 swri_audio_convert_free(&s->full_convert);
191 swri_resample_free(&s->resample);
192 swri_rematrix_free(s);
198 int swr_init(struct SwrContext *s){
199 s->in_buffer_index= 0;
200 s->in_buffer_count= 0;
201 s->resample_in_constraint= 0;
202 free_temp(&s->postin);
203 free_temp(&s->midbuf);
204 free_temp(&s->preout);
205 free_temp(&s->in_buffer);
206 free_temp(&s->dither);
207 swri_audio_convert_free(&s-> in_convert);
208 swri_audio_convert_free(&s->out_convert);
209 swri_audio_convert_free(&s->full_convert);
210 swri_rematrix_free(s);
214 if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
215 av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt);
216 return AVERROR(EINVAL);
218 if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){
219 av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt);
220 return AVERROR(EINVAL);
223 if(s->int_sample_fmt == AV_SAMPLE_FMT_NONE){
224 if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_S16P){
225 s->int_sample_fmt= AV_SAMPLE_FMT_S16P;
226 }else if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_FLTP){
227 s->int_sample_fmt= AV_SAMPLE_FMT_FLTP;
229 av_log(s, AV_LOG_DEBUG, "Using double precission mode\n");
230 s->int_sample_fmt= AV_SAMPLE_FMT_DBLP;
234 if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P
235 &&s->int_sample_fmt != AV_SAMPLE_FMT_S32P
236 &&s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
237 &&s->int_sample_fmt != AV_SAMPLE_FMT_DBLP){
238 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));
239 return AVERROR(EINVAL);
242 set_audiodata_fmt(&s-> in, s-> in_sample_fmt);
243 set_audiodata_fmt(&s->out, s->out_sample_fmt);
245 if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
246 s->resample = swri_resample_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);
248 swri_resample_free(&s->resample);
249 if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P
250 && s->int_sample_fmt != AV_SAMPLE_FMT_S32P
251 && s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
252 && s->int_sample_fmt != AV_SAMPLE_FMT_DBLP
254 av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt/dbl\n");
258 if(!s->used_ch_count)
259 s->used_ch_count= s->in.ch_count;
261 if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){
262 av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");
266 if(!s-> in_ch_layout)
267 s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count);
268 if(!s->out_ch_layout)
269 s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count);
271 s->rematrix= s->out_ch_layout !=s->in_ch_layout || s->rematrix_volume!=1.0 ||
274 #define RSC 1 //FIXME finetune
276 s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
277 if(!s->used_ch_count)
278 s->used_ch_count= s->in.ch_count;
280 s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
282 if(!s-> in.ch_count){
283 av_assert0(!s->in_ch_layout);
284 av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
288 if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {
289 av_log(s, AV_LOG_ERROR, "Rematrix is needed but there is not enough information to do it\n");
293 av_assert0(s->used_ch_count);
294 av_assert0(s->out.ch_count);
295 s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
299 if(!s->resample && !s->rematrix && !s->channel_map && !s->dither_method){
300 s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,
301 s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
305 s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt,
306 s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
307 s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt,
308 s->int_sample_fmt, s->out.ch_count, NULL, 0);
317 s->midbuf.ch_count= s->used_ch_count;
319 s->in_buffer.ch_count= s->used_ch_count;
321 if(!s->resample_first){
322 s->midbuf.ch_count= s->out.ch_count;
324 s->in_buffer.ch_count = s->out.ch_count;
327 set_audiodata_fmt(&s->postin, s->int_sample_fmt);
328 set_audiodata_fmt(&s->midbuf, s->int_sample_fmt);
329 set_audiodata_fmt(&s->preout, s->int_sample_fmt);
332 set_audiodata_fmt(&s->in_buffer, s->int_sample_fmt);
335 s->dither = s->preout;
337 if(s->rematrix || s->dither_method)
338 return swri_rematrix_init(s);
343 static int realloc_audio(AudioData *a, int count){
347 if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)
348 return AVERROR(EINVAL);
350 if(a->count >= count)
355 countb= FFALIGN(count*a->bps, ALIGN);
359 av_assert0(a->ch_count);
361 a->data= av_malloc(countb*a->ch_count);
363 return AVERROR(ENOMEM);
364 for(i=0; i<a->ch_count; i++){
365 a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
366 if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
368 if(!a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);
375 static void copy(AudioData *out, AudioData *in,
377 av_assert0(out->planar == in->planar);
378 av_assert0(out->bps == in->bps);
379 av_assert0(out->ch_count == in->ch_count);
382 for(ch=0; ch<out->ch_count; ch++)
383 memcpy(out->ch[ch], in->ch[ch], count*out->bps);
385 memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
388 static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
391 memset(out->ch, 0, sizeof(out->ch));
392 }else if(out->planar){
393 for(i=0; i<out->ch_count; i++)
394 out->ch[i]= in_arg[i];
396 for(i=0; i<out->ch_count; i++)
397 out->ch[i]= in_arg[0] + i*out->bps;
401 static void reversefill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
404 for(i=0; i<out->ch_count; i++)
405 in_arg[i]= out->ch[i];
407 in_arg[0]= out->ch[0];
413 * out may be equal in.
415 static void buf_set(AudioData *out, AudioData *in, int count){
418 for(ch=0; ch<out->ch_count; ch++)
419 out->ch[ch]= in->ch[ch] + count*out->bps;
421 for(ch=out->ch_count-1; ch>=0; ch--)
422 out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps;
428 * @return number of samples output per channel
430 static int resample(SwrContext *s, AudioData *out_param, int out_count,
431 const AudioData * in_param, int in_count){
432 AudioData in, out, tmp;
436 av_assert1(s->in_buffer.ch_count == in_param->ch_count);
437 av_assert1(s->in_buffer.planar == in_param->planar);
438 av_assert1(s->in_buffer.fmt == in_param->fmt);
444 int ret, size, consumed;
445 if(!s->resample_in_constraint && s->in_buffer_count){
446 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
447 ret= swri_multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
450 buf_set(&out, &out, ret);
451 s->in_buffer_count -= consumed;
452 s->in_buffer_index += consumed;
456 if(s->in_buffer_count <= border){
457 buf_set(&in, &in, -s->in_buffer_count);
458 in_count += s->in_buffer_count;
459 s->in_buffer_count=0;
460 s->in_buffer_index=0;
465 if(in_count && !s->in_buffer_count){
466 s->in_buffer_index=0;
467 ret= swri_multiple_resample(s->resample, &out, out_count, &in, in_count, &consumed);
470 buf_set(&out, &out, ret);
471 in_count -= consumed;
472 buf_set(&in, &in, consumed);
475 //TODO is this check sane considering the advanced copy avoidance below
476 size= s->in_buffer_index + s->in_buffer_count + in_count;
477 if( size > s->in_buffer.count
478 && s->in_buffer_count + in_count <= s->in_buffer_index){
479 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
480 copy(&s->in_buffer, &tmp, s->in_buffer_count);
481 s->in_buffer_index=0;
483 if((ret=realloc_audio(&s->in_buffer, size)) < 0)
488 if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
490 buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
491 copy(&tmp, &in, /*in_*/count);
492 s->in_buffer_count += count;
495 buf_set(&in, &in, count);
496 s->resample_in_constraint= 0;
497 if(s->in_buffer_count != count || in_count)
503 s->resample_in_constraint= !!out_count;
508 static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,
509 AudioData *in , int in_count){
510 AudioData *postin, *midbuf, *preout;
512 AudioData preout_tmp, midbuf_tmp;
515 av_assert0(!s->resample);
516 swri_audio_convert(s->full_convert, out, in, in_count);
520 // in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
521 // in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
523 if((ret=realloc_audio(&s->postin, in_count))<0)
525 if(s->resample_first){
526 av_assert0(s->midbuf.ch_count == s->used_ch_count);
527 if((ret=realloc_audio(&s->midbuf, out_count))<0)
530 av_assert0(s->midbuf.ch_count == s->out.ch_count);
531 if((ret=realloc_audio(&s->midbuf, in_count))<0)
534 if((ret=realloc_audio(&s->preout, out_count))<0)
539 midbuf_tmp= s->midbuf;
541 preout_tmp= s->preout;
544 if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar)
547 if(s->resample_first ? !s->resample : !s->rematrix)
550 if(s->resample_first ? !s->rematrix : !s->resample)
553 if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar){
555 out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant
556 av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
557 copy(out, in, out_count);
560 else if(preout==postin) preout= midbuf= postin= out;
561 else if(preout==midbuf) preout= midbuf= out;
566 swri_audio_convert(s->in_convert, postin, in, in_count);
569 if(s->resample_first){
571 out_count= resample(s, midbuf, out_count, postin, in_count);
573 swri_rematrix(s, preout, midbuf, out_count, preout==out);
576 swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
578 out_count= resample(s, preout, out_count, midbuf, in_count);
581 if(preout != out && out_count){
582 if(s->dither_method){
584 int dither_count= FFMAX(out_count, 1<<16);
585 av_assert0(preout != in);
587 if((ret=realloc_audio(&s->dither, dither_count))<0)
590 for(ch=0; ch<s->dither.ch_count; ch++)
591 swri_get_dither(s, s->dither.ch[ch], s->dither.count, 12345678913579<<ch, s->out_sample_fmt, s->int_sample_fmt);
592 av_assert0(s->dither.ch_count == preout->ch_count);
594 if(s->dither_pos + out_count > s->dither.count)
597 for(ch=0; ch<preout->ch_count; ch++)
598 s->mix_2_1_f(preout->ch[ch], preout->ch[ch], s->dither.ch[ch] + s->dither.bps * s->dither_pos, s->native_one, 0, 0, out_count);
600 s->dither_pos += out_count;
602 //FIXME packed doesnt need more than 1 chan here!
603 swri_audio_convert(s->out_convert, out, preout, out_count);
608 int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
609 const uint8_t *in_arg [SWR_CH_MAX], int in_count){
610 AudioData * in= &s->in;
611 AudioData *out= &s->out;
613 if(s->drop_output > 0){
615 AudioData tmp = s->out;
616 uint8_t *tmp_arg[SWR_CH_MAX];
619 if((ret=realloc_audio(&tmp, s->drop_output))<0)
622 reversefill_audiodata(&tmp, tmp_arg);
623 s->drop_output *= -1; //FIXME find a less hackish solution
624 ret = swr_convert(s, tmp_arg, -s->drop_output, in_arg, in_count); //FIXME optimize but this is as good as never called so maybe it doesnt matter
625 s->drop_output *= -1;
627 s->drop_output -= ret;
630 if(s->drop_output || !out_arg)
636 if(s->in_buffer_count){
637 if (s->resample && !s->flushed) {
638 AudioData *a= &s->in_buffer;
640 if((ret=realloc_audio(a, s->in_buffer_index + 2*s->in_buffer_count)) < 0)
642 av_assert0(a->planar);
643 for(i=0; i<a->ch_count; i++){
644 for(j=0; j<s->in_buffer_count; j++){
645 memcpy(a->ch[i] + (s->in_buffer_index+s->in_buffer_count+j )*a->bps,
646 a->ch[i] + (s->in_buffer_index+s->in_buffer_count-j-1)*a->bps, a->bps);
649 s->in_buffer_count += (s->in_buffer_count+1)/2;
650 s->resample_in_constraint = 0;
657 fill_audiodata(in , (void*)in_arg);
659 fill_audiodata(out, out_arg);
662 int ret = swr_convert_internal(s, out, out_count, in, in_count);
663 if(ret>0 && !s->drop_output)
664 s->outpts += ret * (int64_t)s->in_sample_rate;
670 size = FFMIN(out_count, s->in_buffer_count);
672 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
673 ret= swr_convert_internal(s, out, size, &tmp, size);
677 s->in_buffer_count -= ret;
678 s->in_buffer_index += ret;
679 buf_set(out, out, ret);
681 if(!s->in_buffer_count)
682 s->in_buffer_index = 0;
686 size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
688 if(in_count > out_count) { //FIXME move after swr_convert_internal
689 if( size > s->in_buffer.count
690 && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
691 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
692 copy(&s->in_buffer, &tmp, s->in_buffer_count);
693 s->in_buffer_index=0;
695 if((ret=realloc_audio(&s->in_buffer, size)) < 0)
700 size = FFMIN(in_count, out_count);
701 ret= swr_convert_internal(s, out, size, in, size);
704 buf_set(in, in, ret);
709 buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
710 copy(&tmp, in, in_count);
711 s->in_buffer_count += in_count;
714 if(ret2>0 && !s->drop_output)
715 s->outpts += ret2 * (int64_t)s->in_sample_rate;
720 int swr_drop_output(struct SwrContext *s, int count){
721 s->drop_output += count;
723 if(s->drop_output <= 0)
726 av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);
727 return swr_convert(s, NULL, s->drop_output, NULL, 0);
730 int swr_inject_silence(struct SwrContext *s, int count){
732 AudioData silence = s->out;
733 uint8_t *tmp_arg[SWR_CH_MAX];
740 if((ret=realloc_audio(&silence, count))<0)
743 if(silence.planar) for(i=0; i<silence.ch_count; i++) {
744 memset(silence.ch[i], silence.bps==1 ? 0x80 : 0, count*silence.bps);
746 memset(silence.ch[0], silence.bps==1 ? 0x80 : 0, count*silence.bps*silence.ch_count);
748 reversefill_audiodata(&silence, tmp_arg);
749 av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);
750 ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);
751 av_freep(&silence.data);
755 int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
758 if(s->min_compensation >= FLT_MAX) {
759 return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
761 int64_t delta = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate) - s->outpts;
762 double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
764 if(fabs(fdelta) > s->min_compensation) {
765 if(!s->outpts || fabs(fdelta) > s->min_hard_compensation){
767 if(delta > 0) ret = swr_inject_silence(s, delta / s->out_sample_rate);
768 else ret = swr_drop_output (s, -delta / s-> in_sample_rate);
770 av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
772 } else if(s->soft_compensation_duration && s->max_soft_compensation) {
773 int duration = s->out_sample_rate * s->soft_compensation_duration;
774 double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1);
775 int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;
776 av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
777 swr_set_compensation(s, comp, duration);