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_FLT, 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_FLT, 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);
157 static void free_temp(AudioData *a){
159 memset(a, 0, sizeof(*a));
162 void swr_free(SwrContext **ss){
165 free_temp(&s->postin);
166 free_temp(&s->midbuf);
167 free_temp(&s->preout);
168 free_temp(&s->in_buffer);
169 free_temp(&s->dither);
170 swri_audio_convert_free(&s-> in_convert);
171 swri_audio_convert_free(&s->out_convert);
172 swri_audio_convert_free(&s->full_convert);
173 swri_resample_free(&s->resample);
179 int swr_init(struct SwrContext *s){
180 s->in_buffer_index= 0;
181 s->in_buffer_count= 0;
182 s->resample_in_constraint= 0;
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);
194 s-> in.planar= av_sample_fmt_is_planar(s-> in_sample_fmt);
195 s->out.planar= av_sample_fmt_is_planar(s->out_sample_fmt);
196 s-> in_sample_fmt= av_get_alt_sample_fmt(s-> in_sample_fmt, 0);
197 s->out_sample_fmt= av_get_alt_sample_fmt(s->out_sample_fmt, 0);
199 if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
200 av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt);
201 return AVERROR(EINVAL);
203 if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){
204 av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt);
205 return AVERROR(EINVAL);
208 //FIXME should we allow/support using FLT on material that doesnt need it ?
209 if(s->in_sample_fmt <= AV_SAMPLE_FMT_S16 || s->int_sample_fmt==AV_SAMPLE_FMT_S16){
210 s->int_sample_fmt= AV_SAMPLE_FMT_S16;
212 s->int_sample_fmt= AV_SAMPLE_FMT_FLT;
214 if( s->int_sample_fmt != AV_SAMPLE_FMT_S16
215 &&s->int_sample_fmt != AV_SAMPLE_FMT_S32
216 &&s->int_sample_fmt != AV_SAMPLE_FMT_FLT){
217 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));
218 return AVERROR(EINVAL);
221 if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
222 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);
224 swri_resample_free(&s->resample);
225 if( s->int_sample_fmt != AV_SAMPLE_FMT_S16
226 && s->int_sample_fmt != AV_SAMPLE_FMT_S32
227 && s->int_sample_fmt != AV_SAMPLE_FMT_FLT
229 av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt\n");
233 if(!s->used_ch_count)
234 s->used_ch_count= s->in.ch_count;
236 if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){
237 av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");
241 if(!s-> in_ch_layout)
242 s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count);
243 if(!s->out_ch_layout)
244 s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count);
246 s->rematrix= s->out_ch_layout !=s->in_ch_layout || s->rematrix_volume!=1.0 ||
249 #define RSC 1 //FIXME finetune
251 s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
252 if(!s->used_ch_count)
253 s->used_ch_count= s->in.ch_count;
255 s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
257 if(!s-> in.ch_count){
258 av_assert0(!s->in_ch_layout);
259 av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
263 if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {
264 av_log(s, AV_LOG_ERROR, "Rematrix is needed but there is not enough information to do it\n");
268 av_assert0(s->used_ch_count);
269 av_assert0(s->out.ch_count);
270 s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
272 s-> in.bps= av_get_bytes_per_sample(s-> in_sample_fmt);
273 s->int_bps= av_get_bytes_per_sample(s->int_sample_fmt);
274 s->out.bps= av_get_bytes_per_sample(s->out_sample_fmt);
277 if(!s->resample && !s->rematrix && !s->channel_map){
278 s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,
279 s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
283 s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt,
284 s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
285 s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt,
286 s->int_sample_fmt, s->out.ch_count, NULL, 0);
295 s->midbuf.ch_count= s->used_ch_count;
297 s->in_buffer.ch_count= s->used_ch_count;
299 if(!s->resample_first){
300 s->midbuf.ch_count= s->out.ch_count;
302 s->in_buffer.ch_count = s->out.ch_count;
305 s->postin.bps = s->midbuf.bps = s->preout.bps = s->int_bps;
306 s->postin.planar = s->midbuf.planar = s->preout.planar = 1;
309 s->in_buffer.bps = s->int_bps;
310 s->in_buffer.planar = 1;
313 s->dither = s->preout;
316 return swri_rematrix_init(s);
321 static int realloc_audio(AudioData *a, int count){
325 if(a->count >= count)
330 countb= FFALIGN(count*a->bps, 32);
334 av_assert0(a->ch_count);
336 a->data= av_malloc(countb*a->ch_count);
338 return AVERROR(ENOMEM);
339 for(i=0; i<a->ch_count; i++){
340 a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
341 if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
343 if(!a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);
350 static void copy(AudioData *out, AudioData *in,
352 av_assert0(out->planar == in->planar);
353 av_assert0(out->bps == in->bps);
354 av_assert0(out->ch_count == in->ch_count);
357 for(ch=0; ch<out->ch_count; ch++)
358 memcpy(out->ch[ch], in->ch[ch], count*out->bps);
360 memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
363 static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
366 for(i=0; i<out->ch_count; i++)
367 out->ch[i]= in_arg[i];
369 for(i=0; i<out->ch_count; i++)
370 out->ch[i]= in_arg[0] + i*out->bps;
376 * out may be equal in.
378 static void buf_set(AudioData *out, AudioData *in, int count){
381 for(ch=0; ch<out->ch_count; ch++)
382 out->ch[ch]= in->ch[ch] + count*out->bps;
384 for(ch=0; ch<out->ch_count; ch++)
385 out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps;
391 * @return number of samples output per channel
393 static int resample(SwrContext *s, AudioData *out_param, int out_count,
394 const AudioData * in_param, int in_count){
395 AudioData in, out, tmp;
403 int ret, size, consumed;
404 if(!s->resample_in_constraint && s->in_buffer_count){
405 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
406 ret= swri_multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
409 buf_set(&out, &out, ret);
410 s->in_buffer_count -= consumed;
411 s->in_buffer_index += consumed;
415 if(s->in_buffer_count <= border){
416 buf_set(&in, &in, -s->in_buffer_count);
417 in_count += s->in_buffer_count;
418 s->in_buffer_count=0;
419 s->in_buffer_index=0;
424 if(in_count && !s->in_buffer_count){
425 s->in_buffer_index=0;
426 ret= swri_multiple_resample(s->resample, &out, out_count, &in, in_count, &consumed);
429 buf_set(&out, &out, ret);
430 in_count -= consumed;
431 buf_set(&in, &in, consumed);
434 //TODO is this check sane considering the advanced copy avoidance below
435 size= s->in_buffer_index + s->in_buffer_count + in_count;
436 if( size > s->in_buffer.count
437 && s->in_buffer_count + in_count <= s->in_buffer_index){
438 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
439 copy(&s->in_buffer, &tmp, s->in_buffer_count);
440 s->in_buffer_index=0;
442 if((ret=realloc_audio(&s->in_buffer, size)) < 0)
447 if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
449 buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
450 copy(&tmp, &in, /*in_*/count);
451 s->in_buffer_count += count;
454 buf_set(&in, &in, count);
455 s->resample_in_constraint= 0;
456 if(s->in_buffer_count != count || in_count)
462 s->resample_in_constraint= !!out_count;
467 static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,
468 AudioData *in , int in_count){
469 AudioData *postin, *midbuf, *preout;
471 AudioData preout_tmp, midbuf_tmp;
474 av_assert0(!s->resample);
475 swri_audio_convert(s->full_convert, out, in, in_count);
479 // in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
480 // in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
482 if((ret=realloc_audio(&s->postin, in_count))<0)
484 if(s->resample_first){
485 av_assert0(s->midbuf.ch_count == s->used_ch_count);
486 if((ret=realloc_audio(&s->midbuf, out_count))<0)
489 av_assert0(s->midbuf.ch_count == s->out.ch_count);
490 if((ret=realloc_audio(&s->midbuf, in_count))<0)
493 if((ret=realloc_audio(&s->preout, out_count))<0)
498 midbuf_tmp= s->midbuf;
500 preout_tmp= s->preout;
503 if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar)
506 if(s->resample_first ? !s->resample : !s->rematrix)
509 if(s->resample_first ? !s->rematrix : !s->resample)
512 if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar){
514 out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant
515 av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
516 copy(out, in, out_count);
519 else if(preout==postin) preout= midbuf= postin= out;
520 else if(preout==midbuf) preout= midbuf= out;
525 swri_audio_convert(s->in_convert, postin, in, in_count);
528 if(s->resample_first){
530 out_count= resample(s, midbuf, out_count, postin, in_count);
532 swri_rematrix(s, preout, midbuf, out_count, preout==out);
535 swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
537 out_count= resample(s, preout, out_count, midbuf, in_count);
540 if(preout != out && out_count){
541 if(s->dither_method){
543 int dither_count= FFMAX(out_count, 1<<16);
544 av_assert0(preout != in);
546 if((ret=realloc_audio(&s->dither, dither_count))<0)
549 for(ch=0; ch<s->dither.ch_count; ch++)
550 swri_get_dither(s, s->dither.ch[ch], s->dither.count, 12345678913579<<ch, s->out_sample_fmt, s->int_sample_fmt);
551 av_assert0(s->dither.ch_count == preout->ch_count);
553 if(s->dither_pos + out_count > s->dither.count)
555 for(ch=0; ch<preout->ch_count; ch++)
556 swri_sum2(s->int_sample_fmt, preout->ch[ch], preout->ch[ch], s->dither.ch[ch] + s->dither.bps * s->dither_pos, 1, 1, out_count);
558 s->dither_pos += out_count;
560 //FIXME packed doesnt need more than 1 chan here!
561 swri_audio_convert(s->out_convert, out, preout, out_count);
566 int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
567 const uint8_t *in_arg [SWR_CH_MAX], int in_count){
568 AudioData * in= &s->in;
569 AudioData *out= &s->out;
572 if(s->in_buffer_count){
573 if (s->resample && !s->flushed) {
574 AudioData *a= &s->in_buffer;
576 if((ret=realloc_audio(a, s->in_buffer_index + 2*s->in_buffer_count)) < 0)
578 av_assert0(a->planar);
579 for(i=0; i<a->ch_count; i++){
580 for(j=0; j<s->in_buffer_count; j++){
581 memcpy(a->ch[i] + (s->in_buffer_index+s->in_buffer_count+j )*a->bps,
582 a->ch[i] + (s->in_buffer_index+s->in_buffer_count-j-1)*a->bps, a->bps);
585 s->in_buffer_count += (s->in_buffer_count+1)/2;
586 s->resample_in_constraint = 0;
593 fill_audiodata(in , (void*)in_arg);
595 fill_audiodata(out, out_arg);
598 return swr_convert_internal(s, out, out_count, in, in_count);
603 size = FFMIN(out_count, s->in_buffer_count);
605 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
606 ret= swr_convert_internal(s, out, size, &tmp, size);
610 s->in_buffer_count -= ret;
611 s->in_buffer_index += ret;
612 buf_set(out, out, ret);
614 if(!s->in_buffer_count)
615 s->in_buffer_index = 0;
619 size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
621 if(in_count > out_count) { //FIXME move after swr_convert_internal
622 if( size > s->in_buffer.count
623 && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
624 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
625 copy(&s->in_buffer, &tmp, s->in_buffer_count);
626 s->in_buffer_index=0;
628 if((ret=realloc_audio(&s->in_buffer, size)) < 0)
633 size = FFMIN(in_count, out_count);
634 ret= swr_convert_internal(s, out, size, in, size);
637 buf_set(in, in, ret);
642 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
643 copy(&tmp, in, in_count);
644 s->in_buffer_count += in_count;