swresample.c 33 KB

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  1. /*
  2. * Copyright (C) 2011-2013 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/channel_layout.h"
  25. #include "libavutil/internal.h"
  26. #include <float.h>
  27. #define ALIGN 32
  28. #include "libavutil/ffversion.h"
  29. const char swr_ffversion[] = "FFmpeg version " FFMPEG_VERSION;
  30. unsigned swresample_version(void)
  31. {
  32. av_assert0(LIBSWRESAMPLE_VERSION_MICRO >= 100);
  33. return LIBSWRESAMPLE_VERSION_INT;
  34. }
  35. const char *swresample_configuration(void)
  36. {
  37. return FFMPEG_CONFIGURATION;
  38. }
  39. const char *swresample_license(void)
  40. {
  41. #define LICENSE_PREFIX "libswresample license: "
  42. return &LICENSE_PREFIX FFMPEG_LICENSE[sizeof(LICENSE_PREFIX) - 1];
  43. }
  44. int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map){
  45. if(!s || s->in_convert) // s needs to be allocated but not initialized
  46. return AVERROR(EINVAL);
  47. s->channel_map = channel_map;
  48. return 0;
  49. }
  50. struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,
  51. int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
  52. int64_t in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate,
  53. int log_offset, void *log_ctx){
  54. if(!s) s= swr_alloc();
  55. if(!s) return NULL;
  56. s->log_level_offset= log_offset;
  57. s->log_ctx= log_ctx;
  58. if (av_opt_set_int(s, "ocl", out_ch_layout, 0) < 0)
  59. goto fail;
  60. if (av_opt_set_int(s, "osf", out_sample_fmt, 0) < 0)
  61. goto fail;
  62. if (av_opt_set_int(s, "osr", out_sample_rate, 0) < 0)
  63. goto fail;
  64. if (av_opt_set_int(s, "icl", in_ch_layout, 0) < 0)
  65. goto fail;
  66. if (av_opt_set_int(s, "isf", in_sample_fmt, 0) < 0)
  67. goto fail;
  68. if (av_opt_set_int(s, "isr", in_sample_rate, 0) < 0)
  69. goto fail;
  70. if (av_opt_set_int(s, "ich", av_get_channel_layout_nb_channels(s-> user_in_ch_layout), 0) < 0)
  71. goto fail;
  72. if (av_opt_set_int(s, "och", av_get_channel_layout_nb_channels(s->user_out_ch_layout), 0) < 0)
  73. goto fail;
  74. av_opt_set_int(s, "uch", 0, 0);
  75. return s;
  76. fail:
  77. av_log(s, AV_LOG_ERROR, "Failed to set option\n");
  78. swr_free(&s);
  79. return NULL;
  80. }
  81. static void set_audiodata_fmt(AudioData *a, enum AVSampleFormat fmt){
  82. a->fmt = fmt;
  83. a->bps = av_get_bytes_per_sample(fmt);
  84. a->planar= av_sample_fmt_is_planar(fmt);
  85. if (a->ch_count == 1)
  86. a->planar = 1;
  87. }
  88. static void free_temp(AudioData *a){
  89. av_free(a->data);
  90. memset(a, 0, sizeof(*a));
  91. }
  92. static void clear_context(SwrContext *s){
  93. s->in_buffer_index= 0;
  94. s->in_buffer_count= 0;
  95. s->resample_in_constraint= 0;
  96. memset(s->in.ch, 0, sizeof(s->in.ch));
  97. memset(s->out.ch, 0, sizeof(s->out.ch));
  98. free_temp(&s->postin);
  99. free_temp(&s->midbuf);
  100. free_temp(&s->preout);
  101. free_temp(&s->in_buffer);
  102. free_temp(&s->silence);
  103. free_temp(&s->drop_temp);
  104. free_temp(&s->dither.noise);
  105. free_temp(&s->dither.temp);
  106. swri_audio_convert_free(&s-> in_convert);
  107. swri_audio_convert_free(&s->out_convert);
  108. swri_audio_convert_free(&s->full_convert);
  109. swri_rematrix_free(s);
  110. s->delayed_samples_fixup = 0;
  111. s->flushed = 0;
  112. }
  113. av_cold void swr_free(SwrContext **ss){
  114. SwrContext *s= *ss;
  115. if(s){
  116. clear_context(s);
  117. if (s->resampler)
  118. s->resampler->free(&s->resample);
  119. }
  120. av_freep(ss);
  121. }
  122. av_cold void swr_close(SwrContext *s){
  123. clear_context(s);
  124. }
  125. av_cold int swr_init(struct SwrContext *s){
  126. int ret;
  127. char l1[1024], l2[1024];
  128. clear_context(s);
  129. if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
  130. av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt);
  131. return AVERROR(EINVAL);
  132. }
  133. if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){
  134. av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt);
  135. return AVERROR(EINVAL);
  136. }
  137. if(s-> in_sample_rate <= 0){
  138. av_log(s, AV_LOG_ERROR, "Requested input sample rate %d is invalid\n", s->in_sample_rate);
  139. return AVERROR(EINVAL);
  140. }
  141. if(s->out_sample_rate <= 0){
  142. av_log(s, AV_LOG_ERROR, "Requested output sample rate %d is invalid\n", s->out_sample_rate);
  143. return AVERROR(EINVAL);
  144. }
  145. s->out.ch_count = s-> user_out_ch_count;
  146. s-> in.ch_count = s-> user_in_ch_count;
  147. s->used_ch_count = s->user_used_ch_count;
  148. s-> in_ch_layout = s-> user_in_ch_layout;
  149. s->out_ch_layout = s->user_out_ch_layout;
  150. s->int_sample_fmt= s->user_int_sample_fmt;
  151. s->dither.method = s->user_dither_method;
  152. if(av_get_channel_layout_nb_channels(s-> in_ch_layout) > SWR_CH_MAX) {
  153. av_log(s, AV_LOG_WARNING, "Input channel layout 0x%"PRIx64" is invalid or unsupported.\n", s-> in_ch_layout);
  154. s->in_ch_layout = 0;
  155. }
  156. if(av_get_channel_layout_nb_channels(s->out_ch_layout) > SWR_CH_MAX) {
  157. av_log(s, AV_LOG_WARNING, "Output channel layout 0x%"PRIx64" is invalid or unsupported.\n", s->out_ch_layout);
  158. s->out_ch_layout = 0;
  159. }
  160. switch(s->engine){
  161. #if CONFIG_LIBSOXR
  162. case SWR_ENGINE_SOXR: s->resampler = &swri_soxr_resampler; break;
  163. #endif
  164. case SWR_ENGINE_SWR : s->resampler = &swri_resampler; break;
  165. default:
  166. av_log(s, AV_LOG_ERROR, "Requested resampling engine is unavailable\n");
  167. return AVERROR(EINVAL);
  168. }
  169. if(!s->used_ch_count)
  170. s->used_ch_count= s->in.ch_count;
  171. if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){
  172. av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");
  173. s-> in_ch_layout= 0;
  174. }
  175. if(!s-> in_ch_layout)
  176. s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count);
  177. if(!s->out_ch_layout)
  178. s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count);
  179. s->rematrix= s->out_ch_layout !=s->in_ch_layout || s->rematrix_volume!=1.0 ||
  180. s->rematrix_custom;
  181. if(s->int_sample_fmt == AV_SAMPLE_FMT_NONE){
  182. if( av_get_bytes_per_sample(s-> in_sample_fmt) <= 2
  183. && av_get_bytes_per_sample(s->out_sample_fmt) <= 2){
  184. s->int_sample_fmt= AV_SAMPLE_FMT_S16P;
  185. }else if( av_get_bytes_per_sample(s-> in_sample_fmt) <= 2
  186. && !s->rematrix
  187. && s->out_sample_rate==s->in_sample_rate
  188. && !(s->flags & SWR_FLAG_RESAMPLE)){
  189. s->int_sample_fmt= AV_SAMPLE_FMT_S16P;
  190. }else if( av_get_planar_sample_fmt(s-> in_sample_fmt) == AV_SAMPLE_FMT_S32P
  191. && av_get_planar_sample_fmt(s->out_sample_fmt) == AV_SAMPLE_FMT_S32P
  192. && !s->rematrix
  193. && s->out_sample_rate == s->in_sample_rate
  194. && !(s->flags & SWR_FLAG_RESAMPLE)
  195. && s->engine != SWR_ENGINE_SOXR){
  196. s->int_sample_fmt= AV_SAMPLE_FMT_S32P;
  197. }else if(av_get_bytes_per_sample(s->in_sample_fmt) <= 4){
  198. s->int_sample_fmt= AV_SAMPLE_FMT_FLTP;
  199. }else{
  200. s->int_sample_fmt= AV_SAMPLE_FMT_DBLP;
  201. }
  202. }
  203. av_log(s, AV_LOG_DEBUG, "Using %s internally between filters\n", av_get_sample_fmt_name(s->int_sample_fmt));
  204. if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P
  205. &&s->int_sample_fmt != AV_SAMPLE_FMT_S32P
  206. &&s->int_sample_fmt != AV_SAMPLE_FMT_S64P
  207. &&s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
  208. &&s->int_sample_fmt != AV_SAMPLE_FMT_DBLP){
  209. av_log(s, AV_LOG_ERROR, "Requested sample format %s is not supported internally, s16p/s32p/s64p/fltp/dblp are supported\n", av_get_sample_fmt_name(s->int_sample_fmt));
  210. return AVERROR(EINVAL);
  211. }
  212. set_audiodata_fmt(&s-> in, s-> in_sample_fmt);
  213. set_audiodata_fmt(&s->out, s->out_sample_fmt);
  214. if (s->firstpts_in_samples != AV_NOPTS_VALUE) {
  215. if (!s->async && s->min_compensation >= FLT_MAX/2)
  216. s->async = 1;
  217. s->firstpts =
  218. s->outpts = s->firstpts_in_samples * s->out_sample_rate;
  219. } else
  220. s->firstpts = AV_NOPTS_VALUE;
  221. if (s->async) {
  222. if (s->min_compensation >= FLT_MAX/2)
  223. s->min_compensation = 0.001;
  224. if (s->async > 1.0001) {
  225. s->max_soft_compensation = s->async / (double) s->in_sample_rate;
  226. }
  227. }
  228. if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
  229. 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, s->exact_rational);
  230. if (!s->resample) {
  231. av_log(s, AV_LOG_ERROR, "Failed to initialize resampler\n");
  232. return AVERROR(ENOMEM);
  233. }
  234. }else
  235. s->resampler->free(&s->resample);
  236. if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P
  237. && s->int_sample_fmt != AV_SAMPLE_FMT_S32P
  238. && s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
  239. && s->int_sample_fmt != AV_SAMPLE_FMT_DBLP
  240. && s->resample){
  241. av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16p/s32p/fltp/dblp\n");
  242. ret = AVERROR(EINVAL);
  243. goto fail;
  244. }
  245. #define RSC 1 //FIXME finetune
  246. if(!s-> in.ch_count)
  247. s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
  248. if(!s->used_ch_count)
  249. s->used_ch_count= s->in.ch_count;
  250. if(!s->out.ch_count)
  251. s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
  252. if(!s-> in.ch_count){
  253. av_assert0(!s->in_ch_layout);
  254. av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
  255. ret = AVERROR(EINVAL);
  256. goto fail;
  257. }
  258. av_get_channel_layout_string(l1, sizeof(l1), s-> in.ch_count, s-> in_ch_layout);
  259. av_get_channel_layout_string(l2, sizeof(l2), s->out.ch_count, s->out_ch_layout);
  260. if (s->out_ch_layout && s->out.ch_count != av_get_channel_layout_nb_channels(s->out_ch_layout)) {
  261. av_log(s, AV_LOG_ERROR, "Output channel layout %s mismatches specified channel count %d\n", l2, s->out.ch_count);
  262. ret = AVERROR(EINVAL);
  263. goto fail;
  264. }
  265. if (s->in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s->in_ch_layout)) {
  266. av_log(s, AV_LOG_ERROR, "Input channel layout %s mismatches specified channel count %d\n", l1, s->used_ch_count);
  267. ret = AVERROR(EINVAL);
  268. goto fail;
  269. }
  270. if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {
  271. av_log(s, AV_LOG_ERROR, "Rematrix is needed between %s and %s "
  272. "but there is not enough information to do it\n", l1, l2);
  273. ret = AVERROR(EINVAL);
  274. goto fail;
  275. }
  276. av_assert0(s->used_ch_count);
  277. av_assert0(s->out.ch_count);
  278. s->resample_first= RSC*s->out.ch_count/s->used_ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
  279. s->in_buffer= s->in;
  280. s->silence = s->in;
  281. s->drop_temp= s->out;
  282. if ((ret = swri_dither_init(s, s->out_sample_fmt, s->int_sample_fmt)) < 0)
  283. goto fail;
  284. if(!s->resample && !s->rematrix && !s->channel_map && !s->dither.method){
  285. s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,
  286. s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
  287. return 0;
  288. }
  289. s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt,
  290. s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
  291. s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt,
  292. s->int_sample_fmt, s->out.ch_count, NULL, 0);
  293. if (!s->in_convert || !s->out_convert) {
  294. ret = AVERROR(ENOMEM);
  295. goto fail;
  296. }
  297. s->postin= s->in;
  298. s->preout= s->out;
  299. s->midbuf= s->in;
  300. if(s->channel_map){
  301. s->postin.ch_count=
  302. s->midbuf.ch_count= s->used_ch_count;
  303. if(s->resample)
  304. s->in_buffer.ch_count= s->used_ch_count;
  305. }
  306. if(!s->resample_first){
  307. s->midbuf.ch_count= s->out.ch_count;
  308. if(s->resample)
  309. s->in_buffer.ch_count = s->out.ch_count;
  310. }
  311. set_audiodata_fmt(&s->postin, s->int_sample_fmt);
  312. set_audiodata_fmt(&s->midbuf, s->int_sample_fmt);
  313. set_audiodata_fmt(&s->preout, s->int_sample_fmt);
  314. if(s->resample){
  315. set_audiodata_fmt(&s->in_buffer, s->int_sample_fmt);
  316. }
  317. av_assert0(!s->preout.count);
  318. s->dither.noise = s->preout;
  319. s->dither.temp = s->preout;
  320. if (s->dither.method > SWR_DITHER_NS) {
  321. s->dither.noise.bps = 4;
  322. s->dither.noise.fmt = AV_SAMPLE_FMT_FLTP;
  323. s->dither.noise_scale = 1;
  324. }
  325. if(s->rematrix || s->dither.method) {
  326. ret = swri_rematrix_init(s);
  327. if (ret < 0)
  328. goto fail;
  329. }
  330. return 0;
  331. fail:
  332. swr_close(s);
  333. return ret;
  334. }
  335. int swri_realloc_audio(AudioData *a, int count){
  336. int i, countb;
  337. AudioData old;
  338. if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)
  339. return AVERROR(EINVAL);
  340. if(a->count >= count)
  341. return 0;
  342. count*=2;
  343. countb= FFALIGN(count*a->bps, ALIGN);
  344. old= *a;
  345. av_assert0(a->bps);
  346. av_assert0(a->ch_count);
  347. a->data = av_calloc(countb, a->ch_count);
  348. if(!a->data)
  349. return AVERROR(ENOMEM);
  350. for(i=0; i<a->ch_count; i++){
  351. a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
  352. if(a->count && a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
  353. }
  354. if(a->count && !a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);
  355. av_freep(&old.data);
  356. a->count= count;
  357. return 1;
  358. }
  359. static void copy(AudioData *out, AudioData *in,
  360. int count){
  361. av_assert0(out->planar == in->planar);
  362. av_assert0(out->bps == in->bps);
  363. av_assert0(out->ch_count == in->ch_count);
  364. if(out->planar){
  365. int ch;
  366. for(ch=0; ch<out->ch_count; ch++)
  367. memcpy(out->ch[ch], in->ch[ch], count*out->bps);
  368. }else
  369. memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
  370. }
  371. static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
  372. int i;
  373. if(!in_arg){
  374. memset(out->ch, 0, sizeof(out->ch));
  375. }else if(out->planar){
  376. for(i=0; i<out->ch_count; i++)
  377. out->ch[i]= in_arg[i];
  378. }else{
  379. for(i=0; i<out->ch_count; i++)
  380. out->ch[i]= in_arg[0] + i*out->bps;
  381. }
  382. }
  383. static void reversefill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
  384. int i;
  385. if(out->planar){
  386. for(i=0; i<out->ch_count; i++)
  387. in_arg[i]= out->ch[i];
  388. }else{
  389. in_arg[0]= out->ch[0];
  390. }
  391. }
  392. /**
  393. *
  394. * out may be equal in.
  395. */
  396. static void buf_set(AudioData *out, AudioData *in, int count){
  397. int ch;
  398. if(in->planar){
  399. for(ch=0; ch<out->ch_count; ch++)
  400. out->ch[ch]= in->ch[ch] + count*out->bps;
  401. }else{
  402. for(ch=out->ch_count-1; ch>=0; ch--)
  403. out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps;
  404. }
  405. }
  406. /**
  407. *
  408. * @return number of samples output per channel
  409. */
  410. static int resample(SwrContext *s, AudioData *out_param, int out_count,
  411. const AudioData * in_param, int in_count){
  412. AudioData in, out, tmp;
  413. int ret_sum=0;
  414. int border=0;
  415. int padless = ARCH_X86 && s->engine == SWR_ENGINE_SWR ? 7 : 0;
  416. av_assert1(s->in_buffer.ch_count == in_param->ch_count);
  417. av_assert1(s->in_buffer.planar == in_param->planar);
  418. av_assert1(s->in_buffer.fmt == in_param->fmt);
  419. tmp=out=*out_param;
  420. in = *in_param;
  421. border = s->resampler->invert_initial_buffer(s->resample, &s->in_buffer,
  422. &in, in_count, &s->in_buffer_index, &s->in_buffer_count);
  423. if (border == INT_MAX) {
  424. return 0;
  425. } else if (border < 0) {
  426. return border;
  427. } else if (border) {
  428. buf_set(&in, &in, border);
  429. in_count -= border;
  430. s->resample_in_constraint = 0;
  431. }
  432. do{
  433. int ret, size, consumed;
  434. if(!s->resample_in_constraint && s->in_buffer_count){
  435. buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
  436. ret= s->resampler->multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
  437. out_count -= ret;
  438. ret_sum += ret;
  439. buf_set(&out, &out, ret);
  440. s->in_buffer_count -= consumed;
  441. s->in_buffer_index += consumed;
  442. if(!in_count)
  443. break;
  444. if(s->in_buffer_count <= border){
  445. buf_set(&in, &in, -s->in_buffer_count);
  446. in_count += s->in_buffer_count;
  447. s->in_buffer_count=0;
  448. s->in_buffer_index=0;
  449. border = 0;
  450. }
  451. }
  452. if((s->flushed || in_count > padless) && !s->in_buffer_count){
  453. s->in_buffer_index=0;
  454. ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, FFMAX(in_count-padless, 0), &consumed);
  455. out_count -= ret;
  456. ret_sum += ret;
  457. buf_set(&out, &out, ret);
  458. in_count -= consumed;
  459. buf_set(&in, &in, consumed);
  460. }
  461. //TODO is this check sane considering the advanced copy avoidance below
  462. size= s->in_buffer_index + s->in_buffer_count + in_count;
  463. if( size > s->in_buffer.count
  464. && s->in_buffer_count + in_count <= s->in_buffer_index){
  465. buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
  466. copy(&s->in_buffer, &tmp, s->in_buffer_count);
  467. s->in_buffer_index=0;
  468. }else
  469. if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
  470. return ret;
  471. if(in_count){
  472. int count= in_count;
  473. if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
  474. buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
  475. copy(&tmp, &in, /*in_*/count);
  476. s->in_buffer_count += count;
  477. in_count -= count;
  478. border += count;
  479. buf_set(&in, &in, count);
  480. s->resample_in_constraint= 0;
  481. if(s->in_buffer_count != count || in_count)
  482. continue;
  483. if (padless) {
  484. padless = 0;
  485. continue;
  486. }
  487. }
  488. break;
  489. }while(1);
  490. s->resample_in_constraint= !!out_count;
  491. return ret_sum;
  492. }
  493. static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,
  494. AudioData *in , int in_count){
  495. AudioData *postin, *midbuf, *preout;
  496. int ret/*, in_max*/;
  497. AudioData preout_tmp, midbuf_tmp;
  498. if(s->full_convert){
  499. av_assert0(!s->resample);
  500. swri_audio_convert(s->full_convert, out, in, in_count);
  501. return out_count;
  502. }
  503. // in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
  504. // in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
  505. if((ret=swri_realloc_audio(&s->postin, in_count))<0)
  506. return ret;
  507. if(s->resample_first){
  508. av_assert0(s->midbuf.ch_count == s->used_ch_count);
  509. if((ret=swri_realloc_audio(&s->midbuf, out_count))<0)
  510. return ret;
  511. }else{
  512. av_assert0(s->midbuf.ch_count == s->out.ch_count);
  513. if((ret=swri_realloc_audio(&s->midbuf, in_count))<0)
  514. return ret;
  515. }
  516. if((ret=swri_realloc_audio(&s->preout, out_count))<0)
  517. return ret;
  518. postin= &s->postin;
  519. midbuf_tmp= s->midbuf;
  520. midbuf= &midbuf_tmp;
  521. preout_tmp= s->preout;
  522. preout= &preout_tmp;
  523. if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map)
  524. postin= in;
  525. if(s->resample_first ? !s->resample : !s->rematrix)
  526. midbuf= postin;
  527. if(s->resample_first ? !s->rematrix : !s->resample)
  528. preout= midbuf;
  529. if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar
  530. && !(s->out_sample_fmt==AV_SAMPLE_FMT_S32P && (s->dither.output_sample_bits&31))){
  531. if(preout==in){
  532. out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant
  533. av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
  534. copy(out, in, out_count);
  535. return out_count;
  536. }
  537. else if(preout==postin) preout= midbuf= postin= out;
  538. else if(preout==midbuf) preout= midbuf= out;
  539. else preout= out;
  540. }
  541. if(in != postin){
  542. swri_audio_convert(s->in_convert, postin, in, in_count);
  543. }
  544. if(s->resample_first){
  545. if(postin != midbuf)
  546. if ((out_count = resample(s, midbuf, out_count, postin, in_count)) < 0)
  547. return out_count;
  548. if(midbuf != preout)
  549. swri_rematrix(s, preout, midbuf, out_count, preout==out);
  550. }else{
  551. if(postin != midbuf)
  552. swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
  553. if(midbuf != preout)
  554. if ((out_count = resample(s, preout, out_count, midbuf, in_count)) < 0)
  555. return out_count;
  556. }
  557. if(preout != out && out_count){
  558. AudioData *conv_src = preout;
  559. if(s->dither.method){
  560. int ch;
  561. int dither_count= FFMAX(out_count, 1<<16);
  562. if (preout == in) {
  563. conv_src = &s->dither.temp;
  564. if((ret=swri_realloc_audio(&s->dither.temp, dither_count))<0)
  565. return ret;
  566. }
  567. if((ret=swri_realloc_audio(&s->dither.noise, dither_count))<0)
  568. return ret;
  569. if(ret)
  570. for(ch=0; ch<s->dither.noise.ch_count; ch++)
  571. if((ret=swri_get_dither(s, s->dither.noise.ch[ch], s->dither.noise.count, (12345678913579ULL*ch + 3141592) % 2718281828U, s->dither.noise.fmt))<0)
  572. return ret;
  573. av_assert0(s->dither.noise.ch_count == preout->ch_count);
  574. if(s->dither.noise_pos + out_count > s->dither.noise.count)
  575. s->dither.noise_pos = 0;
  576. if (s->dither.method < SWR_DITHER_NS){
  577. if (s->mix_2_1_simd) {
  578. int len1= out_count&~15;
  579. int off = len1 * preout->bps;
  580. if(len1)
  581. for(ch=0; ch<preout->ch_count; ch++)
  582. 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);
  583. if(out_count != len1)
  584. for(ch=0; ch<preout->ch_count; ch++)
  585. 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, s->native_one, 0, 0, out_count - len1);
  586. } else {
  587. for(ch=0; ch<preout->ch_count; ch++)
  588. 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);
  589. }
  590. } else {
  591. switch(s->int_sample_fmt) {
  592. case AV_SAMPLE_FMT_S16P :swri_noise_shaping_int16(s, conv_src, preout, &s->dither.noise, out_count); break;
  593. case AV_SAMPLE_FMT_S32P :swri_noise_shaping_int32(s, conv_src, preout, &s->dither.noise, out_count); break;
  594. case AV_SAMPLE_FMT_FLTP :swri_noise_shaping_float(s, conv_src, preout, &s->dither.noise, out_count); break;
  595. case AV_SAMPLE_FMT_DBLP :swri_noise_shaping_double(s,conv_src, preout, &s->dither.noise, out_count); break;
  596. }
  597. }
  598. s->dither.noise_pos += out_count;
  599. }
  600. //FIXME packed doesn't need more than 1 chan here!
  601. swri_audio_convert(s->out_convert, out, conv_src, out_count);
  602. }
  603. return out_count;
  604. }
  605. int swr_is_initialized(struct SwrContext *s) {
  606. return !!s->in_buffer.ch_count;
  607. }
  608. int attribute_align_arg swr_convert(struct SwrContext *s,
  609. uint8_t **out_arg, int out_count,
  610. const uint8_t **in_arg, int in_count)
  611. {
  612. AudioData * in= &s->in;
  613. AudioData *out= &s->out;
  614. int av_unused max_output;
  615. if (!swr_is_initialized(s)) {
  616. av_log(s, AV_LOG_ERROR, "Context has not been initialized\n");
  617. return AVERROR(EINVAL);
  618. }
  619. #if defined(ASSERT_LEVEL) && ASSERT_LEVEL >1
  620. max_output = swr_get_out_samples(s, in_count);
  621. #endif
  622. while(s->drop_output > 0){
  623. int ret;
  624. uint8_t *tmp_arg[SWR_CH_MAX];
  625. #define MAX_DROP_STEP 16384
  626. if((ret=swri_realloc_audio(&s->drop_temp, FFMIN(s->drop_output, MAX_DROP_STEP)))<0)
  627. return ret;
  628. reversefill_audiodata(&s->drop_temp, tmp_arg);
  629. s->drop_output *= -1; //FIXME find a less hackish solution
  630. 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
  631. s->drop_output *= -1;
  632. in_count = 0;
  633. if(ret>0) {
  634. s->drop_output -= ret;
  635. if (!s->drop_output && !out_arg)
  636. return 0;
  637. continue;
  638. }
  639. av_assert0(s->drop_output);
  640. return 0;
  641. }
  642. if(!in_arg){
  643. if(s->resample){
  644. if (!s->flushed)
  645. s->resampler->flush(s);
  646. s->resample_in_constraint = 0;
  647. s->flushed = 1;
  648. }else if(!s->in_buffer_count){
  649. return 0;
  650. }
  651. }else
  652. fill_audiodata(in , (void*)in_arg);
  653. fill_audiodata(out, out_arg);
  654. if(s->resample){
  655. int ret = swr_convert_internal(s, out, out_count, in, in_count);
  656. if(ret>0 && !s->drop_output)
  657. s->outpts += ret * (int64_t)s->in_sample_rate;
  658. av_assert2(max_output < 0 || ret <= max_output);
  659. return ret;
  660. }else{
  661. AudioData tmp= *in;
  662. int ret2=0;
  663. int ret, size;
  664. size = FFMIN(out_count, s->in_buffer_count);
  665. if(size){
  666. buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
  667. ret= swr_convert_internal(s, out, size, &tmp, size);
  668. if(ret<0)
  669. return ret;
  670. ret2= ret;
  671. s->in_buffer_count -= ret;
  672. s->in_buffer_index += ret;
  673. buf_set(out, out, ret);
  674. out_count -= ret;
  675. if(!s->in_buffer_count)
  676. s->in_buffer_index = 0;
  677. }
  678. if(in_count){
  679. size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
  680. if(in_count > out_count) { //FIXME move after swr_convert_internal
  681. if( size > s->in_buffer.count
  682. && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
  683. buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
  684. copy(&s->in_buffer, &tmp, s->in_buffer_count);
  685. s->in_buffer_index=0;
  686. }else
  687. if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
  688. return ret;
  689. }
  690. if(out_count){
  691. size = FFMIN(in_count, out_count);
  692. ret= swr_convert_internal(s, out, size, in, size);
  693. if(ret<0)
  694. return ret;
  695. buf_set(in, in, ret);
  696. in_count -= ret;
  697. ret2 += ret;
  698. }
  699. if(in_count){
  700. buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
  701. copy(&tmp, in, in_count);
  702. s->in_buffer_count += in_count;
  703. }
  704. }
  705. if(ret2>0 && !s->drop_output)
  706. s->outpts += ret2 * (int64_t)s->in_sample_rate;
  707. av_assert2(max_output < 0 || ret2 < 0 || ret2 <= max_output);
  708. return ret2;
  709. }
  710. }
  711. int swr_drop_output(struct SwrContext *s, int count){
  712. const uint8_t *tmp_arg[SWR_CH_MAX];
  713. s->drop_output += count;
  714. if(s->drop_output <= 0)
  715. return 0;
  716. av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);
  717. return swr_convert(s, NULL, s->drop_output, tmp_arg, 0);
  718. }
  719. int swr_inject_silence(struct SwrContext *s, int count){
  720. int ret, i;
  721. uint8_t *tmp_arg[SWR_CH_MAX];
  722. if(count <= 0)
  723. return 0;
  724. #define MAX_SILENCE_STEP 16384
  725. while (count > MAX_SILENCE_STEP) {
  726. if ((ret = swr_inject_silence(s, MAX_SILENCE_STEP)) < 0)
  727. return ret;
  728. count -= MAX_SILENCE_STEP;
  729. }
  730. if((ret=swri_realloc_audio(&s->silence, count))<0)
  731. return ret;
  732. if(s->silence.planar) for(i=0; i<s->silence.ch_count; i++) {
  733. memset(s->silence.ch[i], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps);
  734. } else
  735. memset(s->silence.ch[0], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps*s->silence.ch_count);
  736. reversefill_audiodata(&s->silence, tmp_arg);
  737. av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);
  738. ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);
  739. return ret;
  740. }
  741. int64_t swr_get_delay(struct SwrContext *s, int64_t base){
  742. if (s->resampler && s->resample){
  743. return s->resampler->get_delay(s, base);
  744. }else{
  745. return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate;
  746. }
  747. }
  748. int swr_get_out_samples(struct SwrContext *s, int in_samples)
  749. {
  750. int64_t out_samples;
  751. if (in_samples < 0)
  752. return AVERROR(EINVAL);
  753. if (s->resampler && s->resample) {
  754. if (!s->resampler->get_out_samples)
  755. return AVERROR(ENOSYS);
  756. out_samples = s->resampler->get_out_samples(s, in_samples);
  757. } else {
  758. out_samples = s->in_buffer_count + in_samples;
  759. av_assert0(s->out_sample_rate == s->in_sample_rate);
  760. }
  761. if (out_samples > INT_MAX)
  762. return AVERROR(EINVAL);
  763. return out_samples;
  764. }
  765. int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){
  766. int ret;
  767. if (!s || compensation_distance < 0)
  768. return AVERROR(EINVAL);
  769. if (!compensation_distance && sample_delta)
  770. return AVERROR(EINVAL);
  771. if (!s->resample) {
  772. s->flags |= SWR_FLAG_RESAMPLE;
  773. ret = swr_init(s);
  774. if (ret < 0)
  775. return ret;
  776. }
  777. if (!s->resampler->set_compensation){
  778. return AVERROR(EINVAL);
  779. }else{
  780. return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance);
  781. }
  782. }
  783. int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
  784. if(pts == INT64_MIN)
  785. return s->outpts;
  786. if (s->firstpts == AV_NOPTS_VALUE)
  787. s->outpts = s->firstpts = pts;
  788. if(s->min_compensation >= FLT_MAX) {
  789. return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
  790. } else {
  791. 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;
  792. double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
  793. if(fabs(fdelta) > s->min_compensation) {
  794. if(s->outpts == s->firstpts || fabs(fdelta) > s->min_hard_compensation){
  795. int ret;
  796. if(delta > 0) ret = swr_inject_silence(s, delta / s->out_sample_rate);
  797. else ret = swr_drop_output (s, -delta / s-> in_sample_rate);
  798. if(ret<0){
  799. av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
  800. }
  801. } else if(s->soft_compensation_duration && s->max_soft_compensation) {
  802. int duration = s->out_sample_rate * s->soft_compensation_duration;
  803. double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1);
  804. int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;
  805. av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
  806. swr_set_compensation(s, comp, duration);
  807. }
  808. }
  809. return s->outpts;
  810. }
  811. }