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  1. /*
  2. * Copyright (C) 2011 Michael Niedermayer (michaelni@gmx.at)
  3. *
  4. * This file is part of libswresample
  5. *
  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.
  10. *
  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.
  15. *
  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
  19. */
  20. #include "libavutil/opt.h"
  21. #include "swresample_internal.h"
  22. #include "audioconvert.h"
  23. #include "libavutil/avassert.h"
  24. #include "libavutil/audioconvert.h"
  25. #define C30DB M_SQRT2
  26. #define C15DB 1.189207115
  27. #define C__0DB 1.0
  28. #define C_15DB 0.840896415
  29. #define C_30DB M_SQRT1_2
  30. #define C_45DB 0.594603558
  31. #define C_60DB 0.5
  32. //TODO split options array out?
  33. #define OFFSET(x) offsetof(SwrContext,x)
  34. static const AVOption options[]={
  35. {"ich", "input channel count", OFFSET( in.ch_count ), AV_OPT_TYPE_INT, {.dbl=2}, 0, SWR_CH_MAX, 0},
  36. {"och", "output channel count", OFFSET(out.ch_count ), AV_OPT_TYPE_INT, {.dbl=2}, 0, SWR_CH_MAX, 0},
  37. {"uch", "used channel count", OFFSET(used_ch_count ), AV_OPT_TYPE_INT, {.dbl=0}, 0, SWR_CH_MAX, 0},
  38. {"isr", "input sample rate" , OFFSET( in_sample_rate), AV_OPT_TYPE_INT, {.dbl=48000}, 1, INT_MAX, 0},
  39. {"osr", "output sample rate" , OFFSET(out_sample_rate), AV_OPT_TYPE_INT, {.dbl=48000}, 1, INT_MAX, 0},
  40. //{"ip" , "input planar" , OFFSET( in.planar ), AV_OPT_TYPE_INT, {.dbl=0}, 0, 1, 0},
  41. //{"op" , "output planar" , OFFSET(out.planar ), AV_OPT_TYPE_INT, {.dbl=0}, 0, 1, 0},
  42. {"isf", "input sample format", OFFSET( in_sample_fmt ), AV_OPT_TYPE_INT, {.dbl=AV_SAMPLE_FMT_S16}, 0, AV_SAMPLE_FMT_NB-1+256, 0},
  43. {"osf", "output sample format", OFFSET(out_sample_fmt ), AV_OPT_TYPE_INT, {.dbl=AV_SAMPLE_FMT_S16}, 0, AV_SAMPLE_FMT_NB-1+256, 0},
  44. {"tsf", "internal sample format", OFFSET(int_sample_fmt ), AV_OPT_TYPE_INT, {.dbl=AV_SAMPLE_FMT_NONE}, -1, AV_SAMPLE_FMT_FLT, 0},
  45. {"icl", "input channel layout" , OFFSET( in_ch_layout), AV_OPT_TYPE_INT64, {.dbl=0}, 0, INT64_MAX, 0, "channel_layout"},
  46. {"ocl", "output channel layout", OFFSET(out_ch_layout), AV_OPT_TYPE_INT64, {.dbl=0}, 0, INT64_MAX, 0, "channel_layout"},
  47. {"clev", "center mix level" , OFFSET(clev) , AV_OPT_TYPE_FLOAT, {.dbl=C_30DB}, 0, 4, 0},
  48. {"slev", "sourround mix level" , OFFSET(slev) , AV_OPT_TYPE_FLOAT, {.dbl=C_30DB}, 0, 4, 0},
  49. {"rmvol", "rematrix volume" , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0}, -1000, 1000, 0},
  50. {"flags", NULL , OFFSET(flags) , AV_OPT_TYPE_FLAGS, {.dbl=0}, 0, UINT_MAX, 0, "flags"},
  51. {"res", "force resampling", 0, AV_OPT_TYPE_CONST, {.dbl=SWR_FLAG_RESAMPLE}, INT_MIN, INT_MAX, 0, "flags"},
  52. {"dither", "dither method" , OFFSET(dither_method), AV_OPT_TYPE_INT, {.dbl=0}, 0, SWR_DITHER_NB-1, 0, "dither_method"},
  53. {"rectangular", "rectangular dither", 0, AV_OPT_TYPE_CONST, {.dbl=SWR_DITHER_RECTANGULAR}, INT_MIN, INT_MAX, 0, "dither_method"},
  54. {"triangular" , "triangular dither" , 0, AV_OPT_TYPE_CONST, {.dbl=SWR_DITHER_TRIANGULAR }, INT_MIN, INT_MAX, 0, "dither_method"},
  55. {0}
  56. };
  57. static const char* context_to_name(void* ptr) {
  58. return "SWR";
  59. }
  60. static const AVClass av_class = {
  61. .class_name = "SwrContext",
  62. .item_name = context_to_name,
  63. .option = options,
  64. .version = LIBAVUTIL_VERSION_INT,
  65. .log_level_offset_offset = OFFSET(log_level_offset),
  66. .parent_log_context_offset = OFFSET(log_ctx),
  67. };
  68. unsigned swresample_version(void)
  69. {
  70. av_assert0(LIBSWRESAMPLE_VERSION_MICRO >= 100);
  71. return LIBSWRESAMPLE_VERSION_INT;
  72. }
  73. const char *swresample_configuration(void)
  74. {
  75. return FFMPEG_CONFIGURATION;
  76. }
  77. const char *swresample_license(void)
  78. {
  79. #define LICENSE_PREFIX "libswresample license: "
  80. return LICENSE_PREFIX FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1;
  81. }
  82. int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map){
  83. if(!s || s->in_convert) // s needs to be allocated but not initialized
  84. return AVERROR(EINVAL);
  85. s->channel_map = channel_map;
  86. return 0;
  87. }
  88. const AVClass *swr_get_class(void)
  89. {
  90. return &av_class;
  91. }
  92. struct SwrContext *swr_alloc(void){
  93. SwrContext *s= av_mallocz(sizeof(SwrContext));
  94. if(s){
  95. s->av_class= &av_class;
  96. av_opt_set_defaults(s);
  97. }
  98. return s;
  99. }
  100. struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,
  101. int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
  102. int64_t in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate,
  103. int log_offset, void *log_ctx){
  104. if(!s) s= swr_alloc();
  105. if(!s) return NULL;
  106. s->log_level_offset= log_offset;
  107. s->log_ctx= log_ctx;
  108. av_opt_set_int(s, "ocl", out_ch_layout, 0);
  109. av_opt_set_int(s, "osf", out_sample_fmt, 0);
  110. av_opt_set_int(s, "osr", out_sample_rate, 0);
  111. av_opt_set_int(s, "icl", in_ch_layout, 0);
  112. av_opt_set_int(s, "isf", in_sample_fmt, 0);
  113. av_opt_set_int(s, "isr", in_sample_rate, 0);
  114. av_opt_set_int(s, "tsf", AV_SAMPLE_FMT_NONE, 0);
  115. av_opt_set_int(s, "ich", av_get_channel_layout_nb_channels(s-> in_ch_layout), 0);
  116. av_opt_set_int(s, "och", av_get_channel_layout_nb_channels(s->out_ch_layout), 0);
  117. av_opt_set_int(s, "uch", 0, 0);
  118. return s;
  119. }
  120. static void free_temp(AudioData *a){
  121. av_free(a->data);
  122. memset(a, 0, sizeof(*a));
  123. }
  124. void swr_free(SwrContext **ss){
  125. SwrContext *s= *ss;
  126. if(s){
  127. free_temp(&s->postin);
  128. free_temp(&s->midbuf);
  129. free_temp(&s->preout);
  130. free_temp(&s->in_buffer);
  131. free_temp(&s->dither);
  132. swri_audio_convert_free(&s-> in_convert);
  133. swri_audio_convert_free(&s->out_convert);
  134. swri_audio_convert_free(&s->full_convert);
  135. swri_resample_free(&s->resample);
  136. }
  137. av_freep(ss);
  138. }
  139. int swr_init(struct SwrContext *s){
  140. s->in_buffer_index= 0;
  141. s->in_buffer_count= 0;
  142. s->resample_in_constraint= 0;
  143. free_temp(&s->postin);
  144. free_temp(&s->midbuf);
  145. free_temp(&s->preout);
  146. free_temp(&s->in_buffer);
  147. free_temp(&s->dither);
  148. swri_audio_convert_free(&s-> in_convert);
  149. swri_audio_convert_free(&s->out_convert);
  150. swri_audio_convert_free(&s->full_convert);
  151. s->flushed = 0;
  152. s-> in.planar= av_sample_fmt_is_planar(s-> in_sample_fmt);
  153. s->out.planar= av_sample_fmt_is_planar(s->out_sample_fmt);
  154. s-> in_sample_fmt= av_get_alt_sample_fmt(s-> in_sample_fmt, 0);
  155. s->out_sample_fmt= av_get_alt_sample_fmt(s->out_sample_fmt, 0);
  156. if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
  157. av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt);
  158. return AVERROR(EINVAL);
  159. }
  160. if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){
  161. av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt);
  162. return AVERROR(EINVAL);
  163. }
  164. //FIXME should we allow/support using FLT on material that doesnt need it ?
  165. if(s->in_sample_fmt <= AV_SAMPLE_FMT_S16 || s->int_sample_fmt==AV_SAMPLE_FMT_S16){
  166. s->int_sample_fmt= AV_SAMPLE_FMT_S16;
  167. }else
  168. s->int_sample_fmt= AV_SAMPLE_FMT_FLT;
  169. if( s->int_sample_fmt != AV_SAMPLE_FMT_S16
  170. &&s->int_sample_fmt != AV_SAMPLE_FMT_S32
  171. &&s->int_sample_fmt != AV_SAMPLE_FMT_FLT){
  172. 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));
  173. return AVERROR(EINVAL);
  174. }
  175. if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
  176. s->resample = swri_resample_init(s->resample, s->out_sample_rate, s->in_sample_rate, 16, 10, 0, 0.8, s->int_sample_fmt);
  177. }else
  178. swri_resample_free(&s->resample);
  179. if( s->int_sample_fmt != AV_SAMPLE_FMT_S16
  180. && s->int_sample_fmt != AV_SAMPLE_FMT_S32
  181. && s->int_sample_fmt != AV_SAMPLE_FMT_FLT
  182. && s->resample){
  183. av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt\n");
  184. return -1;
  185. }
  186. if(!s->used_ch_count)
  187. 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");
  190. s-> in_ch_layout= 0;
  191. }
  192. if(!s-> in_ch_layout)
  193. s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count);
  194. if(!s->out_ch_layout)
  195. s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count);
  196. s->rematrix= s->out_ch_layout !=s->in_ch_layout || s->rematrix_volume!=1.0 ||
  197. s->rematrix_custom;
  198. #define RSC 1 //FIXME finetune
  199. if(!s-> in.ch_count)
  200. s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
  201. if(!s->used_ch_count)
  202. s->used_ch_count= s->in.ch_count;
  203. if(!s->out.ch_count)
  204. s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
  205. if(!s-> in.ch_count){
  206. av_assert0(!s->in_ch_layout);
  207. av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
  208. return -1;
  209. }
  210. if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {
  211. av_log(s, AV_LOG_ERROR, "Rematrix is needed but there is not enough information to do it\n");
  212. return -1;
  213. }
  214. av_assert0(s->used_ch_count);
  215. av_assert0(s->out.ch_count);
  216. s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
  217. s-> in.bps= av_get_bytes_per_sample(s-> in_sample_fmt);
  218. s->int_bps= av_get_bytes_per_sample(s->int_sample_fmt);
  219. s->out.bps= av_get_bytes_per_sample(s->out_sample_fmt);
  220. s->in_buffer= s->in;
  221. if(!s->resample && !s->rematrix && !s->channel_map){
  222. s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,
  223. s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
  224. return 0;
  225. }
  226. s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt,
  227. s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
  228. s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt,
  229. s->int_sample_fmt, s->out.ch_count, NULL, 0);
  230. s->postin= s->in;
  231. s->preout= s->out;
  232. s->midbuf= s->in;
  233. if(s->channel_map){
  234. s->postin.ch_count=
  235. s->midbuf.ch_count= s->used_ch_count;
  236. if(s->resample)
  237. s->in_buffer.ch_count= s->used_ch_count;
  238. }
  239. if(!s->resample_first){
  240. s->midbuf.ch_count= s->out.ch_count;
  241. if(s->resample)
  242. s->in_buffer.ch_count = s->out.ch_count;
  243. }
  244. s->postin.bps = s->midbuf.bps = s->preout.bps = s->int_bps;
  245. s->postin.planar = s->midbuf.planar = s->preout.planar = 1;
  246. if(s->resample){
  247. s->in_buffer.bps = s->int_bps;
  248. s->in_buffer.planar = 1;
  249. }
  250. s->dither = s->preout;
  251. if(s->rematrix)
  252. return swri_rematrix_init(s);
  253. return 0;
  254. }
  255. static int realloc_audio(AudioData *a, int count){
  256. int i, countb;
  257. AudioData old;
  258. if(a->count >= count)
  259. return 0;
  260. count*=2;
  261. countb= FFALIGN(count*a->bps, 32);
  262. old= *a;
  263. av_assert0(a->bps);
  264. av_assert0(a->ch_count);
  265. a->data= av_malloc(countb*a->ch_count);
  266. if(!a->data)
  267. return AVERROR(ENOMEM);
  268. for(i=0; i<a->ch_count; i++){
  269. a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
  270. if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
  271. }
  272. if(!a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);
  273. av_free(old.data);
  274. a->count= count;
  275. return 1;
  276. }
  277. static void copy(AudioData *out, AudioData *in,
  278. int count){
  279. av_assert0(out->planar == in->planar);
  280. av_assert0(out->bps == in->bps);
  281. av_assert0(out->ch_count == in->ch_count);
  282. if(out->planar){
  283. int ch;
  284. for(ch=0; ch<out->ch_count; ch++)
  285. memcpy(out->ch[ch], in->ch[ch], count*out->bps);
  286. }else
  287. memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
  288. }
  289. static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
  290. int i;
  291. if(out->planar){
  292. for(i=0; i<out->ch_count; i++)
  293. out->ch[i]= in_arg[i];
  294. }else{
  295. for(i=0; i<out->ch_count; i++)
  296. out->ch[i]= in_arg[0] + i*out->bps;
  297. }
  298. }
  299. /**
  300. *
  301. * out may be equal in.
  302. */
  303. static void buf_set(AudioData *out, AudioData *in, int count){
  304. if(in->planar){
  305. int ch;
  306. for(ch=0; ch<out->ch_count; ch++)
  307. out->ch[ch]= in->ch[ch] + count*out->bps;
  308. }else
  309. out->ch[0]= in->ch[0] + count*out->ch_count*out->bps;
  310. }
  311. /**
  312. *
  313. * @return number of samples output per channel
  314. */
  315. static int resample(SwrContext *s, AudioData *out_param, int out_count,
  316. const AudioData * in_param, int in_count){
  317. AudioData in, out, tmp;
  318. int ret_sum=0;
  319. int border=0;
  320. tmp=out=*out_param;
  321. in = *in_param;
  322. do{
  323. int ret, size, consumed;
  324. if(!s->resample_in_constraint && s->in_buffer_count){
  325. buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
  326. ret= swri_multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
  327. out_count -= ret;
  328. ret_sum += ret;
  329. buf_set(&out, &out, ret);
  330. s->in_buffer_count -= consumed;
  331. s->in_buffer_index += consumed;
  332. if(!in_count)
  333. break;
  334. if(s->in_buffer_count <= border){
  335. buf_set(&in, &in, -s->in_buffer_count);
  336. in_count += s->in_buffer_count;
  337. s->in_buffer_count=0;
  338. s->in_buffer_index=0;
  339. border = 0;
  340. }
  341. }
  342. if(in_count && !s->in_buffer_count){
  343. s->in_buffer_index=0;
  344. ret= swri_multiple_resample(s->resample, &out, out_count, &in, in_count, &consumed);
  345. out_count -= ret;
  346. ret_sum += ret;
  347. buf_set(&out, &out, ret);
  348. in_count -= consumed;
  349. buf_set(&in, &in, consumed);
  350. }
  351. //TODO is this check sane considering the advanced copy avoidance below
  352. size= s->in_buffer_index + s->in_buffer_count + in_count;
  353. if( size > s->in_buffer.count
  354. && s->in_buffer_count + in_count <= s->in_buffer_index){
  355. buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
  356. copy(&s->in_buffer, &tmp, s->in_buffer_count);
  357. s->in_buffer_index=0;
  358. }else
  359. if((ret=realloc_audio(&s->in_buffer, size)) < 0)
  360. return ret;
  361. if(in_count){
  362. int count= in_count;
  363. if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
  364. buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
  365. copy(&tmp, &in, /*in_*/count);
  366. s->in_buffer_count += count;
  367. in_count -= count;
  368. border += count;
  369. buf_set(&in, &in, count);
  370. s->resample_in_constraint= 0;
  371. if(s->in_buffer_count != count || in_count)
  372. continue;
  373. }
  374. break;
  375. }while(1);
  376. s->resample_in_constraint= !!out_count;
  377. return ret_sum;
  378. }
  379. static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,
  380. AudioData *in , int in_count){
  381. AudioData *postin, *midbuf, *preout;
  382. int ret/*, in_max*/;
  383. AudioData preout_tmp, midbuf_tmp;
  384. if(s->full_convert){
  385. av_assert0(!s->resample);
  386. swri_audio_convert(s->full_convert, out, in, in_count);
  387. return out_count;
  388. }
  389. // in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
  390. // in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
  391. if((ret=realloc_audio(&s->postin, in_count))<0)
  392. return ret;
  393. if(s->resample_first){
  394. av_assert0(s->midbuf.ch_count == s->used_ch_count);
  395. if((ret=realloc_audio(&s->midbuf, out_count))<0)
  396. return ret;
  397. }else{
  398. av_assert0(s->midbuf.ch_count == s->out.ch_count);
  399. if((ret=realloc_audio(&s->midbuf, in_count))<0)
  400. return ret;
  401. }
  402. if((ret=realloc_audio(&s->preout, out_count))<0)
  403. return ret;
  404. postin= &s->postin;
  405. midbuf_tmp= s->midbuf;
  406. midbuf= &midbuf_tmp;
  407. preout_tmp= s->preout;
  408. preout= &preout_tmp;
  409. if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar)
  410. postin= in;
  411. if(s->resample_first ? !s->resample : !s->rematrix)
  412. midbuf= postin;
  413. if(s->resample_first ? !s->rematrix : !s->resample)
  414. preout= midbuf;
  415. if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar){
  416. if(preout==in){
  417. out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant
  418. av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
  419. copy(out, in, out_count);
  420. return out_count;
  421. }
  422. else if(preout==postin) preout= midbuf= postin= out;
  423. else if(preout==midbuf) preout= midbuf= out;
  424. else preout= out;
  425. }
  426. if(in != postin){
  427. swri_audio_convert(s->in_convert, postin, in, in_count);
  428. }
  429. if(s->resample_first){
  430. if(postin != midbuf)
  431. out_count= resample(s, midbuf, out_count, postin, in_count);
  432. if(midbuf != preout)
  433. swri_rematrix(s, preout, midbuf, out_count, preout==out);
  434. }else{
  435. if(postin != midbuf)
  436. swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
  437. if(midbuf != preout)
  438. out_count= resample(s, preout, out_count, midbuf, in_count);
  439. }
  440. if(preout != out && out_count){
  441. if(s->dither_method){
  442. int ch;
  443. av_assert0(preout != in);
  444. if((ret=realloc_audio(&s->dither, out_count))<0)
  445. return ret;
  446. if(ret)
  447. for(ch=0; ch<s->dither.ch_count; ch++)
  448. swri_get_dither(s->dither.ch[ch], s->dither.count, 12345678913579<<ch, s->out_sample_fmt, s->int_sample_fmt, s->dither_method);
  449. av_assert0(s->dither.ch_count == preout->ch_count);
  450. for(ch=0; ch<preout->ch_count; ch++){
  451. swri_sum2(s->int_sample_fmt, preout->ch[ch], preout->ch[ch], s->dither.ch[ch], 1, 1, out_count);
  452. }
  453. }
  454. //FIXME packed doesnt need more than 1 chan here!
  455. swri_audio_convert(s->out_convert, out, preout, out_count);
  456. }
  457. return out_count;
  458. }
  459. int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
  460. const uint8_t *in_arg [SWR_CH_MAX], int in_count){
  461. AudioData * in= &s->in;
  462. AudioData *out= &s->out;
  463. if(!in_arg){
  464. if(s->in_buffer_count){
  465. if (s->resample && !s->flushed) {
  466. AudioData *a= &s->in_buffer;
  467. int i, j, ret;
  468. if((ret=realloc_audio(a, s->in_buffer_index + 2*s->in_buffer_count)) < 0)
  469. return ret;
  470. av_assert0(a->planar);
  471. for(i=0; i<a->ch_count; i++){
  472. for(j=0; j<s->in_buffer_count; j++){
  473. memcpy(a->ch[i] + (s->in_buffer_index+s->in_buffer_count+j )*a->bps,
  474. a->ch[i] + (s->in_buffer_index+s->in_buffer_count-j-1)*a->bps, a->bps);
  475. }
  476. }
  477. s->in_buffer_count += (s->in_buffer_count+1)/2;
  478. s->resample_in_constraint = 0;
  479. s->flushed = 1;
  480. }
  481. }else{
  482. return 0;
  483. }
  484. }else
  485. fill_audiodata(in , (void*)in_arg);
  486. fill_audiodata(out, out_arg);
  487. if(s->resample){
  488. return swr_convert_internal(s, out, out_count, in, in_count);
  489. }else{
  490. AudioData tmp= *in;
  491. int ret2=0;
  492. int ret, size;
  493. size = FFMIN(out_count, s->in_buffer_count);
  494. if(size){
  495. buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
  496. ret= swr_convert_internal(s, out, size, &tmp, size);
  497. if(ret<0)
  498. return ret;
  499. ret2= ret;
  500. s->in_buffer_count -= ret;
  501. s->in_buffer_index += ret;
  502. buf_set(out, out, ret);
  503. out_count -= ret;
  504. if(!s->in_buffer_count)
  505. s->in_buffer_index = 0;
  506. }
  507. if(in_count){
  508. size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
  509. if(in_count > out_count) { //FIXME move after swr_convert_internal
  510. if( size > s->in_buffer.count
  511. && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
  512. buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
  513. copy(&s->in_buffer, &tmp, s->in_buffer_count);
  514. s->in_buffer_index=0;
  515. }else
  516. if((ret=realloc_audio(&s->in_buffer, size)) < 0)
  517. return ret;
  518. }
  519. if(out_count){
  520. size = FFMIN(in_count, out_count);
  521. ret= swr_convert_internal(s, out, size, in, size);
  522. if(ret<0)
  523. return ret;
  524. buf_set(in, in, ret);
  525. in_count -= ret;
  526. ret2 += ret;
  527. }
  528. if(in_count){
  529. buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
  530. copy(&tmp, in, in_count);
  531. s->in_buffer_count += in_count;
  532. }
  533. }
  534. return ret2;
  535. }
  536. }