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"
28 #define C15DB 1.189207115
30 #define C_15DB 0.840896415
31 #define C_30DB M_SQRT1_2
32 #define C_45DB 0.594603558
36 //TODO split options array out?
37 #define OFFSET(x) offsetof(SwrContext,x)
38 #define PARAM AV_OPT_FLAG_AUDIO_PARAM
40 static const AVOption options[]={
41 {"ich" , "Input Channel Count" , OFFSET( in.ch_count ), AV_OPT_TYPE_INT , {.dbl=2 }, 0 , SWR_CH_MAX, PARAM},
42 {"in_channel_count" , "Input Channel Count" , OFFSET( in.ch_count ), AV_OPT_TYPE_INT , {.dbl=2 }, 0 , SWR_CH_MAX, PARAM},
43 {"och" , "Output Channel Count" , OFFSET(out.ch_count ), AV_OPT_TYPE_INT , {.dbl=2 }, 0 , SWR_CH_MAX, PARAM},
44 {"out_channel_count" , "Output Channel Count" , OFFSET(out.ch_count ), AV_OPT_TYPE_INT , {.dbl=2 }, 0 , SWR_CH_MAX, PARAM},
45 {"uch" , "Used Channel Count" , OFFSET(used_ch_count ), AV_OPT_TYPE_INT , {.dbl=0 }, 0 , SWR_CH_MAX, PARAM},
46 {"used_channel_count" , "Used Channel Count" , OFFSET(used_ch_count ), AV_OPT_TYPE_INT , {.dbl=0 }, 0 , SWR_CH_MAX, PARAM},
47 {"isr" , "Input Sample Rate" , OFFSET( in_sample_rate), AV_OPT_TYPE_INT , {.dbl=48000 }, 1 , INT_MAX , PARAM},
48 {"in_sample_rate" , "Input Sample Rate" , OFFSET( in_sample_rate), AV_OPT_TYPE_INT , {.dbl=48000 }, 1 , INT_MAX , PARAM},
49 {"osr" , "Output Sample Rate" , OFFSET(out_sample_rate), AV_OPT_TYPE_INT , {.dbl=48000 }, 1 , INT_MAX , PARAM},
50 {"out_sample_rate" , "Output Sample Rate" , OFFSET(out_sample_rate), AV_OPT_TYPE_INT , {.dbl=48000 }, 1 , INT_MAX , PARAM},
51 {"isf" , "Input Sample Format" , OFFSET( in_sample_fmt ), AV_OPT_TYPE_INT , {.dbl=AV_SAMPLE_FMT_S16 }, 0 , AV_SAMPLE_FMT_NB-1+256, PARAM},
52 {"in_sample_fmt" , "Input Sample Format" , OFFSET( in_sample_fmt ), AV_OPT_TYPE_INT , {.dbl=AV_SAMPLE_FMT_S16 }, 0 , AV_SAMPLE_FMT_NB-1+256, PARAM},
53 {"osf" , "Output Sample Format" , OFFSET(out_sample_fmt ), AV_OPT_TYPE_INT , {.dbl=AV_SAMPLE_FMT_S16 }, 0 , AV_SAMPLE_FMT_NB-1+256, PARAM},
54 {"out_sample_fmt" , "Output Sample Format" , OFFSET(out_sample_fmt ), AV_OPT_TYPE_INT , {.dbl=AV_SAMPLE_FMT_S16 }, 0 , AV_SAMPLE_FMT_NB-1+256, PARAM},
55 {"tsf" , "Internal Sample Format" , OFFSET(int_sample_fmt ), AV_OPT_TYPE_INT , {.dbl=AV_SAMPLE_FMT_NONE }, -1 , AV_SAMPLE_FMT_FLTP, PARAM},
56 {"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},
57 {"icl" , "Input Channel Layout" , OFFSET( in_ch_layout ), AV_OPT_TYPE_INT64, {.dbl=0 }, 0 , INT64_MAX , PARAM, "channel_layout"},
58 {"in_channel_layout" , "Input Channel Layout" , OFFSET( in_ch_layout ), AV_OPT_TYPE_INT64, {.dbl=0 }, 0 , INT64_MAX , PARAM, "channel_layout"},
59 {"ocl" , "Output Channel Layout" , OFFSET(out_ch_layout ), AV_OPT_TYPE_INT64, {.dbl=0 }, 0 , INT64_MAX , PARAM, "channel_layout"},
60 {"out_channel_layout" , "Output Channel Layout" , OFFSET(out_ch_layout ), AV_OPT_TYPE_INT64, {.dbl=0 }, 0 , INT64_MAX , PARAM, "channel_layout"},
61 {"clev" , "Center Mix Level" , OFFSET(clev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM},
62 {"center_mix_level" , "Center Mix Level" , OFFSET(clev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM},
63 {"slev" , "Sourround Mix Level" , OFFSET(slev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM},
64 {"surround_mix_level" , "Sourround Mix Level" , OFFSET(slev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM},
65 {"rmvol" , "Rematrix Volume" , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0 }, -1000 , 1000 , PARAM},
66 {"rematrix_volume" , "Rematrix Volume" , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0 }, -1000 , 1000 , PARAM},
67 {"flags" , NULL , OFFSET(flags ), AV_OPT_TYPE_FLAGS, {.dbl=0 }, 0 , UINT_MAX , PARAM, "flags"},
68 {"swr_flags" , NULL , OFFSET(flags ), AV_OPT_TYPE_FLAGS, {.dbl=0 }, 0 , UINT_MAX , PARAM, "flags"},
69 {"res" , "Force Resampling" , 0 , AV_OPT_TYPE_CONST, {.dbl=SWR_FLAG_RESAMPLE }, INT_MIN, INT_MAX , PARAM, "flags"},
70 {"dither_scale" , "Dither Scale" , OFFSET(dither_scale ), AV_OPT_TYPE_FLOAT, {.dbl=1 }, 0 , INT_MAX , PARAM},
71 {"dither_method" , "Dither Method" , OFFSET(dither_method ), AV_OPT_TYPE_INT , {.dbl=0 }, 0 , SWR_DITHER_NB-1, PARAM, "dither_method"},
72 {"rectangular" , "Rectangular Dither" , 0 , AV_OPT_TYPE_CONST, {.dbl=SWR_DITHER_RECTANGULAR}, INT_MIN, INT_MAX , PARAM, "dither_method"},
73 {"triangular" , "Triangular Dither" , 0 , AV_OPT_TYPE_CONST, {.dbl=SWR_DITHER_TRIANGULAR }, INT_MIN, INT_MAX , PARAM, "dither_method"},
74 {"triangular_hp" , "Triangular Dither With High Pass" , 0 , AV_OPT_TYPE_CONST, {.dbl=SWR_DITHER_TRIANGULAR_HIGHPASS }, INT_MIN, INT_MAX, PARAM, "dither_method"},
75 {"filter_size" , "Resampling Filter Size" , OFFSET(filter_size) , AV_OPT_TYPE_INT , {.dbl=16 }, 0 , INT_MAX , PARAM },
76 {"phase_shift" , "Resampling Phase Shift" , OFFSET(phase_shift) , AV_OPT_TYPE_INT , {.dbl=10 }, 0 , 30 , PARAM },
77 {"linear_interp" , "Use Linear Interpolation" , OFFSET(linear_interp) , AV_OPT_TYPE_INT , {.dbl=0 }, 0 , 1 , PARAM },
78 {"cutoff" , "Cutoff Frequency Ratio" , OFFSET(cutoff) , AV_OPT_TYPE_DOUBLE,{.dbl=0.8 }, 0 , 1 , PARAM },
82 static const char* context_to_name(void* ptr) {
86 static const AVClass av_class = {
87 .class_name = "SwrContext",
88 .item_name = context_to_name,
90 .version = LIBAVUTIL_VERSION_INT,
91 .log_level_offset_offset = OFFSET(log_level_offset),
92 .parent_log_context_offset = OFFSET(log_ctx),
95 unsigned swresample_version(void)
97 av_assert0(LIBSWRESAMPLE_VERSION_MICRO >= 100);
98 return LIBSWRESAMPLE_VERSION_INT;
101 const char *swresample_configuration(void)
103 return FFMPEG_CONFIGURATION;
106 const char *swresample_license(void)
108 #define LICENSE_PREFIX "libswresample license: "
109 return LICENSE_PREFIX FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1;
112 int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map){
113 if(!s || s->in_convert) // s needs to be allocated but not initialized
114 return AVERROR(EINVAL);
115 s->channel_map = channel_map;
119 const AVClass *swr_get_class(void)
124 struct SwrContext *swr_alloc(void){
125 SwrContext *s= av_mallocz(sizeof(SwrContext));
127 s->av_class= &av_class;
128 av_opt_set_defaults(s);
133 struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,
134 int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
135 int64_t in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate,
136 int log_offset, void *log_ctx){
137 if(!s) s= swr_alloc();
140 s->log_level_offset= log_offset;
143 av_opt_set_int(s, "ocl", out_ch_layout, 0);
144 av_opt_set_int(s, "osf", out_sample_fmt, 0);
145 av_opt_set_int(s, "osr", out_sample_rate, 0);
146 av_opt_set_int(s, "icl", in_ch_layout, 0);
147 av_opt_set_int(s, "isf", in_sample_fmt, 0);
148 av_opt_set_int(s, "isr", in_sample_rate, 0);
149 av_opt_set_int(s, "tsf", AV_SAMPLE_FMT_NONE, 0);
150 av_opt_set_int(s, "ich", av_get_channel_layout_nb_channels(s-> in_ch_layout), 0);
151 av_opt_set_int(s, "och", av_get_channel_layout_nb_channels(s->out_ch_layout), 0);
152 av_opt_set_int(s, "uch", 0, 0);
156 static void set_audiodata_fmt(AudioData *a, enum AVSampleFormat fmt){
158 a->bps = av_get_bytes_per_sample(fmt);
159 a->planar= av_sample_fmt_is_planar(fmt);
162 static void free_temp(AudioData *a){
164 memset(a, 0, sizeof(*a));
167 void swr_free(SwrContext **ss){
170 free_temp(&s->postin);
171 free_temp(&s->midbuf);
172 free_temp(&s->preout);
173 free_temp(&s->in_buffer);
174 free_temp(&s->dither);
175 swri_audio_convert_free(&s-> in_convert);
176 swri_audio_convert_free(&s->out_convert);
177 swri_audio_convert_free(&s->full_convert);
178 swri_resample_free(&s->resample);
179 swri_rematrix_free(s);
185 int swr_init(struct SwrContext *s){
186 s->in_buffer_index= 0;
187 s->in_buffer_count= 0;
188 s->resample_in_constraint= 0;
189 free_temp(&s->postin);
190 free_temp(&s->midbuf);
191 free_temp(&s->preout);
192 free_temp(&s->in_buffer);
193 free_temp(&s->dither);
194 swri_audio_convert_free(&s-> in_convert);
195 swri_audio_convert_free(&s->out_convert);
196 swri_audio_convert_free(&s->full_convert);
197 swri_rematrix_free(s);
201 if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
202 av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt);
203 return AVERROR(EINVAL);
205 if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){
206 av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt);
207 return AVERROR(EINVAL);
210 //FIXME should we allow/support using FLT on material that doesnt need it ?
211 if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_S16P || s->int_sample_fmt==AV_SAMPLE_FMT_S16P){
212 s->int_sample_fmt= AV_SAMPLE_FMT_S16P;
214 s->int_sample_fmt= AV_SAMPLE_FMT_FLTP;
216 if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P
217 &&s->int_sample_fmt != AV_SAMPLE_FMT_S32P
218 &&s->int_sample_fmt != AV_SAMPLE_FMT_FLTP){
219 av_log(s, AV_LOG_ERROR, "Requested sample format %s is not supported internally, S16/S32/FLT is supported\n", av_get_sample_fmt_name(s->int_sample_fmt));
220 return AVERROR(EINVAL);
223 set_audiodata_fmt(&s-> in, s-> in_sample_fmt);
224 set_audiodata_fmt(&s->out, s->out_sample_fmt);
226 if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
227 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);
229 swri_resample_free(&s->resample);
230 if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P
231 && s->int_sample_fmt != AV_SAMPLE_FMT_S32P
232 && s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
234 av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt\n");
238 if(!s->used_ch_count)
239 s->used_ch_count= s->in.ch_count;
241 if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){
242 av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");
246 if(!s-> in_ch_layout)
247 s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count);
248 if(!s->out_ch_layout)
249 s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count);
251 s->rematrix= s->out_ch_layout !=s->in_ch_layout || s->rematrix_volume!=1.0 ||
254 #define RSC 1 //FIXME finetune
256 s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
257 if(!s->used_ch_count)
258 s->used_ch_count= s->in.ch_count;
260 s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
262 if(!s-> in.ch_count){
263 av_assert0(!s->in_ch_layout);
264 av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
268 if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {
269 av_log(s, AV_LOG_ERROR, "Rematrix is needed but there is not enough information to do it\n");
273 av_assert0(s->used_ch_count);
274 av_assert0(s->out.ch_count);
275 s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
279 if(!s->resample && !s->rematrix && !s->channel_map && !s->dither_method){
280 s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,
281 s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
285 s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt,
286 s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
287 s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt,
288 s->int_sample_fmt, s->out.ch_count, NULL, 0);
297 s->midbuf.ch_count= s->used_ch_count;
299 s->in_buffer.ch_count= s->used_ch_count;
301 if(!s->resample_first){
302 s->midbuf.ch_count= s->out.ch_count;
304 s->in_buffer.ch_count = s->out.ch_count;
307 set_audiodata_fmt(&s->postin, s->int_sample_fmt);
308 set_audiodata_fmt(&s->midbuf, s->int_sample_fmt);
309 set_audiodata_fmt(&s->preout, s->int_sample_fmt);
312 set_audiodata_fmt(&s->in_buffer, s->int_sample_fmt);
315 s->dither = s->preout;
317 if(s->rematrix || s->dither_method)
318 return swri_rematrix_init(s);
323 static int realloc_audio(AudioData *a, int count){
327 if(a->count >= count)
332 countb= FFALIGN(count*a->bps, 32);
336 av_assert0(a->ch_count);
338 a->data= av_malloc(countb*a->ch_count);
340 return AVERROR(ENOMEM);
341 for(i=0; i<a->ch_count; i++){
342 a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
343 if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
345 if(!a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);
352 static void copy(AudioData *out, AudioData *in,
354 av_assert0(out->planar == in->planar);
355 av_assert0(out->bps == in->bps);
356 av_assert0(out->ch_count == in->ch_count);
359 for(ch=0; ch<out->ch_count; ch++)
360 memcpy(out->ch[ch], in->ch[ch], count*out->bps);
362 memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
365 static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
368 for(i=0; i<out->ch_count; i++)
369 out->ch[i]= in_arg[i];
371 for(i=0; i<out->ch_count; i++)
372 out->ch[i]= in_arg[0] + i*out->bps;
378 * out may be equal in.
380 static void buf_set(AudioData *out, AudioData *in, int count){
383 for(ch=0; ch<out->ch_count; ch++)
384 out->ch[ch]= in->ch[ch] + count*out->bps;
386 for(ch=0; ch<out->ch_count; ch++)
387 out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps;
393 * @return number of samples output per channel
395 static int resample(SwrContext *s, AudioData *out_param, int out_count,
396 const AudioData * in_param, int in_count){
397 AudioData in, out, tmp;
405 int ret, size, consumed;
406 if(!s->resample_in_constraint && s->in_buffer_count){
407 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
408 ret= swri_multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
411 buf_set(&out, &out, ret);
412 s->in_buffer_count -= consumed;
413 s->in_buffer_index += consumed;
417 if(s->in_buffer_count <= border){
418 buf_set(&in, &in, -s->in_buffer_count);
419 in_count += s->in_buffer_count;
420 s->in_buffer_count=0;
421 s->in_buffer_index=0;
426 if(in_count && !s->in_buffer_count){
427 s->in_buffer_index=0;
428 ret= swri_multiple_resample(s->resample, &out, out_count, &in, in_count, &consumed);
431 buf_set(&out, &out, ret);
432 in_count -= consumed;
433 buf_set(&in, &in, consumed);
436 //TODO is this check sane considering the advanced copy avoidance below
437 size= s->in_buffer_index + s->in_buffer_count + in_count;
438 if( size > s->in_buffer.count
439 && s->in_buffer_count + in_count <= s->in_buffer_index){
440 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
441 copy(&s->in_buffer, &tmp, s->in_buffer_count);
442 s->in_buffer_index=0;
444 if((ret=realloc_audio(&s->in_buffer, size)) < 0)
449 if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
451 buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
452 copy(&tmp, &in, /*in_*/count);
453 s->in_buffer_count += count;
456 buf_set(&in, &in, count);
457 s->resample_in_constraint= 0;
458 if(s->in_buffer_count != count || in_count)
464 s->resample_in_constraint= !!out_count;
469 static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,
470 AudioData *in , int in_count){
471 AudioData *postin, *midbuf, *preout;
473 AudioData preout_tmp, midbuf_tmp;
476 av_assert0(!s->resample);
477 swri_audio_convert(s->full_convert, out, in, in_count);
481 // in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
482 // in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
484 if((ret=realloc_audio(&s->postin, in_count))<0)
486 if(s->resample_first){
487 av_assert0(s->midbuf.ch_count == s->used_ch_count);
488 if((ret=realloc_audio(&s->midbuf, out_count))<0)
491 av_assert0(s->midbuf.ch_count == s->out.ch_count);
492 if((ret=realloc_audio(&s->midbuf, in_count))<0)
495 if((ret=realloc_audio(&s->preout, out_count))<0)
500 midbuf_tmp= s->midbuf;
502 preout_tmp= s->preout;
505 if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar)
508 if(s->resample_first ? !s->resample : !s->rematrix)
511 if(s->resample_first ? !s->rematrix : !s->resample)
514 if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar){
516 out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant
517 av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
518 copy(out, in, out_count);
521 else if(preout==postin) preout= midbuf= postin= out;
522 else if(preout==midbuf) preout= midbuf= out;
527 swri_audio_convert(s->in_convert, postin, in, in_count);
530 if(s->resample_first){
532 out_count= resample(s, midbuf, out_count, postin, in_count);
534 swri_rematrix(s, preout, midbuf, out_count, preout==out);
537 swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
539 out_count= resample(s, preout, out_count, midbuf, in_count);
542 if(preout != out && out_count){
543 if(s->dither_method){
545 int dither_count= FFMAX(out_count, 1<<16);
546 av_assert0(preout != in);
548 if((ret=realloc_audio(&s->dither, dither_count))<0)
551 for(ch=0; ch<s->dither.ch_count; ch++)
552 swri_get_dither(s, s->dither.ch[ch], s->dither.count, 12345678913579<<ch, s->out_sample_fmt, s->int_sample_fmt);
553 av_assert0(s->dither.ch_count == preout->ch_count);
555 if(s->dither_pos + out_count > s->dither.count)
558 for(ch=0; ch<preout->ch_count; ch++)
559 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);
561 s->dither_pos += out_count;
563 //FIXME packed doesnt need more than 1 chan here!
564 swri_audio_convert(s->out_convert, out, preout, out_count);
569 int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
570 const uint8_t *in_arg [SWR_CH_MAX], int in_count){
571 AudioData * in= &s->in;
572 AudioData *out= &s->out;
575 if(s->in_buffer_count){
576 if (s->resample && !s->flushed) {
577 AudioData *a= &s->in_buffer;
579 if((ret=realloc_audio(a, s->in_buffer_index + 2*s->in_buffer_count)) < 0)
581 av_assert0(a->planar);
582 for(i=0; i<a->ch_count; i++){
583 for(j=0; j<s->in_buffer_count; j++){
584 memcpy(a->ch[i] + (s->in_buffer_index+s->in_buffer_count+j )*a->bps,
585 a->ch[i] + (s->in_buffer_index+s->in_buffer_count-j-1)*a->bps, a->bps);
588 s->in_buffer_count += (s->in_buffer_count+1)/2;
589 s->resample_in_constraint = 0;
596 fill_audiodata(in , (void*)in_arg);
598 fill_audiodata(out, out_arg);
601 return swr_convert_internal(s, out, out_count, in, in_count);
606 size = FFMIN(out_count, s->in_buffer_count);
608 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
609 ret= swr_convert_internal(s, out, size, &tmp, size);
613 s->in_buffer_count -= ret;
614 s->in_buffer_index += ret;
615 buf_set(out, out, ret);
617 if(!s->in_buffer_count)
618 s->in_buffer_index = 0;
622 size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
624 if(in_count > out_count) { //FIXME move after swr_convert_internal
625 if( size > s->in_buffer.count
626 && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
627 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
628 copy(&s->in_buffer, &tmp, s->in_buffer_count);
629 s->in_buffer_index=0;
631 if((ret=realloc_audio(&s->in_buffer, size)) < 0)
636 size = FFMIN(in_count, out_count);
637 ret= swr_convert_internal(s, out, size, in, size);
640 buf_set(in, in, ret);
645 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
646 copy(&tmp, in, in_count);
647 s->in_buffer_count += in_count;