FFmpeg: libswresample/resample.c Source File

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libswresample

resample.c

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1 /*

2 * audio resampling

4 *

5 * This file is part of FFmpeg.

6 *

7 * FFmpeg is free software; you can redistribute it and/or

8 * modify it under the terms of the GNU Lesser General Public

9 * License as published by the Free Software Foundation; either

10 * version 2.1 of the License, or (at your option) any later version.

11 *

12 * FFmpeg is distributed in the hope that it will be useful,

13 * but WITHOUT ANY WARRANTY; without even the implied warranty of

14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU

15 * Lesser General Public License for more details.

16 *

17 * You should have received a copy of the GNU Lesser General Public

18 * License along with FFmpeg; if not, write to the Free Software

19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA

20 */

22 /**

23 * @file

24 * audio resampling

25 * @author Michael Niedermayer <michaelni@gmx.at>

26 */

28 #include "libavutil/avassert.h"

29 #include "resample.h"

31 /**

32 * 0th order modified bessel function of the first kind.

33 */

34 static double bessel(double x){

35 double v=1;

36 double lastv=0;

37 double t=1;

38 int i;

39 static const double inv[100]={

40 1.0/( 1* 1), 1.0/( 2* 2), 1.0/( 3* 3), 1.0/( 4* 4), 1.0/( 5* 5), 1.0/( 6* 6), 1.0/( 7* 7), 1.0/( 8* 8), 1.0/( 9* 9), 1.0/(10*10),

41 1.0/(11*11), 1.0/(12*12), 1.0/(13*13), 1.0/(14*14), 1.0/(15*15), 1.0/(16*16), 1.0/(17*17), 1.0/(18*18), 1.0/(19*19), 1.0/(20*20),

42 1.0/(21*21), 1.0/(22*22), 1.0/(23*23), 1.0/(24*24), 1.0/(25*25), 1.0/(26*26), 1.0/(27*27), 1.0/(28*28), 1.0/(29*29), 1.0/(30*30),

43 1.0/(31*31), 1.0/(32*32), 1.0/(33*33), 1.0/(34*34), 1.0/(35*35), 1.0/(36*36), 1.0/(37*37), 1.0/(38*38), 1.0/(39*39), 1.0/(40*40),

44 1.0/(41*41), 1.0/(42*42), 1.0/(43*43), 1.0/(44*44), 1.0/(45*45), 1.0/(46*46), 1.0/(47*47), 1.0/(48*48), 1.0/(49*49), 1.0/(50*50),

45 1.0/(51*51), 1.0/(52*52), 1.0/(53*53), 1.0/(54*54), 1.0/(55*55), 1.0/(56*56), 1.0/(57*57), 1.0/(58*58), 1.0/(59*59), 1.0/(60*60),

46 1.0/(61*61), 1.0/(62*62), 1.0/(63*63), 1.0/(64*64), 1.0/(65*65), 1.0/(66*66), 1.0/(67*67), 1.0/(68*68), 1.0/(69*69), 1.0/(70*70),

47 1.0/(71*71), 1.0/(72*72), 1.0/(73*73), 1.0/(74*74), 1.0/(75*75), 1.0/(76*76), 1.0/(77*77), 1.0/(78*78), 1.0/(79*79), 1.0/(80*80),

48 1.0/(81*81), 1.0/(82*82), 1.0/(83*83), 1.0/(84*84), 1.0/(85*85), 1.0/(86*86), 1.0/(87*87), 1.0/(88*88), 1.0/(89*89), 1.0/(90*90),

49 1.0/(91*91), 1.0/(92*92), 1.0/(93*93), 1.0/(94*94), 1.0/(95*95), 1.0/(96*96), 1.0/(97*97), 1.0/(98*98), 1.0/(99*99), 1.0/(10000)

50 };

52 x= x*x/4;

53 for(i=0; v != lastv; i++){

54 lastv=v;

55 t *= x*inv[i];

56 v += t;

57 av_assert2(i<99);

58 }

59 return v;

60 }

62 /**

63 * builds a polyphase filterbank.

64 * @param factor resampling factor

65 * @param scale wanted sum of coefficients for each filter

66 * @param filter_type filter type

67 * @param kaiser_beta kaiser window beta

68 * @return 0 on success, negative on error

69 */

70 static int build_filter(ResampleContext *c, void *filter, double factor, int tap_count, int alloc, int phase_count, int scale,

71 int filter_type, int kaiser_beta){

72 int ph, i;

73 double x, y, w;

74 double *tab = av_malloc_array(tap_count, sizeof(*tab));

75 const int center= (tap_count-1)/2;

77 if (!tab)

78 return AVERROR(ENOMEM);

80 /* if upsampling, only need to interpolate, no filter */

81 if (factor > 1.0)

82 factor = 1.0;

84 for(ph=0;ph<phase_count;ph++) {

85 double norm = 0;

86 for(i=0;i<tap_count;i++) {

87 x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;

88 if (x == 0) y = 1.0;

89 else y = sin(x) / x;

90 switch(filter_type){

91 case SWR_FILTER_TYPE_CUBIC:{

92 const float d= -0.5; //first order derivative = -0.5

93 x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);

94 if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x);

95 else y= d*(-4 + 8*x - 5*x*x + x*x*x);

96 break;}

97 case SWR_FILTER_TYPE_BLACKMAN_NUTTALL:

98 w = 2.0*x / (factor*tap_count) + M_PI;

99 y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w);

100 break;

101 case SWR_FILTER_TYPE_KAISER:

102 w = 2.0*x / (factor*tap_count*M_PI);

103 y *= bessel(kaiser_beta*sqrt(FFMAX(1-w*w, 0)));

104 break;

105 default:

106 av_assert0(0);

107 }

108

109 tab[i] = y;

110 norm += y;

111 }

112

113 /* normalize so that an uniform color remains the same */

114 switch(c->format){

115 case AV_SAMPLE_FMT_S16P:

116 for(i=0;i<tap_count;i++)

117 ((int16_t*)filter)[ph * alloc + i] = av_clip(lrintf(tab[i] * scale / norm), INT16_MIN, INT16_MAX);

118 break;

119 case AV_SAMPLE_FMT_S32P:

120 for(i=0;i<tap_count;i++)

121 ((int32_t*)filter)[ph * alloc + i] = av_clipl_int32(llrint(tab[i] * scale / norm));

122 break;

123 case AV_SAMPLE_FMT_FLTP:

124 for(i=0;i<tap_count;i++)

125 ((float*)filter)[ph * alloc + i] = tab[i] * scale / norm;

126 break;

127 case AV_SAMPLE_FMT_DBLP:

128 for(i=0;i<tap_count;i++)

129 ((double*)filter)[ph * alloc + i] = tab[i] * scale / norm;

130 break;

131 }

132 }

133 #if 0

134 {

135 #define LEN 1024

136 int j,k;

137 double sine[LEN + tap_count];

138 double filtered[LEN];

139 double maxff=-2, minff=2, maxsf=-2, minsf=2;

140 for(i=0; i<LEN; i++){

141 double ss=0, sf=0, ff=0;

142 for(j=0; j<LEN+tap_count; j++)

143 sine[j]= cos(i*j*M_PI/LEN);

144 for(j=0; j<LEN; j++){

145 double sum=0;

146 ph=0;

147 for(k=0; k<tap_count; k++)

148 sum += filter[ph * tap_count + k] * sine[k+j];

149 filtered[j]= sum / (1<<FILTER_SHIFT);

150 ss+= sine[j + center] * sine[j + center];

151 ff+= filtered[j] * filtered[j];

152 sf+= sine[j + center] * filtered[j];

153 }

154 ss= sqrt(2*ss/LEN);

155 ff= sqrt(2*ff/LEN);

156 sf= 2*sf/LEN;

157 maxff= FFMAX(maxff, ff);

158 minff= FFMIN(minff, ff);

159 maxsf= FFMAX(maxsf, sf);

160 minsf= FFMIN(minsf, sf);

161 if(i%11==0){

162 av_log(NULL, AV_LOG_ERROR, "i:%4d ss:%f ff:%13.6e-%13.6e sf:%13.6e-%13.6e\n", i, ss, maxff, minff, maxsf, minsf);

163 minff=minsf= 2;

164 maxff=maxsf= -2;

165 }

166 }

167 }

168 #endif

169

170 av_free(tab);

171 return 0;

172 }

173

174 static ResampleContext *resample_init(ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear,

175 double cutoff0, enum AVSampleFormat format, enum SwrFilterType filter_type, int kaiser_beta,

176 double precision, int cheby)

177 {

178 double cutoff = cutoff0? cutoff0 : 0.97;

179 double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);

180 int phase_count= 1<<phase_shift;

181

182 if (!c || c->phase_shift != phase_shift || c->linear!=linear || c->factor != factor

183 || c->filter_length != FFMAX((int)ceil(filter_size/factor), 1) || c->format != format

184 || c->filter_type != filter_type || c->kaiser_beta != kaiser_beta) {

185 c = av_mallocz(sizeof(*c));

186 if (!c)

187 return NULL;

188

189 c->format= format;

190

191 c->felem_size= av_get_bytes_per_sample(c->format);

192

193 switch(c->format){

194 case AV_SAMPLE_FMT_S16P:

195 c->filter_shift = 15;

196 break;

197 case AV_SAMPLE_FMT_S32P:

198 c->filter_shift = 30;

199 break;

200 case AV_SAMPLE_FMT_FLTP:

201 case AV_SAMPLE_FMT_DBLP:

202 c->filter_shift = 0;

203 break;

204 default:

205 av_log(NULL, AV_LOG_ERROR, "Unsupported sample format\n");

206 av_assert0(0);

207 }

208

209 if (filter_size/factor > INT32_MAX/256) {

210 av_log(NULL, AV_LOG_ERROR, "Filter length too large\n");

211 goto error;

212 }

213

214 c->phase_shift = phase_shift;

215 c->phase_mask = phase_count - 1;

216 c->linear = linear;

217 c->factor = factor;

218 c->filter_length = FFMAX((int)ceil(filter_size/factor), 1);

219 c->filter_alloc = FFALIGN(c->filter_length, 8);

220 c->filter_bank = av_calloc(c->filter_alloc, (phase_count+1)*c->felem_size);

221 c->filter_type = filter_type;

222 c->kaiser_beta = kaiser_beta;

223 if (!c->filter_bank)

224 goto error;

225 if (build_filter(c, (void*)c->filter_bank, factor, c->filter_length, c->filter_alloc, phase_count, 1<<c->filter_shift, filter_type, kaiser_beta))

226 goto error;

227 memcpy(c->filter_bank + (c->filter_alloc*phase_count+1)*c->felem_size, c->filter_bank, (c->filter_alloc-1)*c->felem_size);

228 memcpy(c->filter_bank + (c->filter_alloc*phase_count )*c->felem_size, c->filter_bank + (c->filter_alloc - 1)*c->felem_size, c->felem_size);

229 }

230

231 c->compensation_distance= 0;

232 if(!av_reduce(&c->src_incr, &c->dst_incr, out_rate, in_rate * (int64_t)phase_count, INT32_MAX/2))

233 goto error;

234 c->ideal_dst_incr = c->dst_incr;

235 c->dst_incr_div = c->dst_incr / c->src_incr;

236 c->dst_incr_mod = c->dst_incr % c->src_incr;

237

238 c->index= -phase_count*((c->filter_length-1)/2);

239 c->frac= 0;

240

241 swri_resample_dsp_init(c);

242

243 return c;

244 error:

245 av_freep(&c->filter_bank);

246 av_free(c);

247 return NULL;

248 }

249

250 static void resample_free(ResampleContext **c){

251 if(!*c)

252 return;

253 av_freep(&(*c)->filter_bank);

254 av_freep(c);

255 }

256

257 static int set_compensation(ResampleContext *c, int sample_delta, int compensation_distance){

258 c->compensation_distance= compensation_distance;

259 if (compensation_distance)

260 c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance;

261 else

262 c->dst_incr = c->ideal_dst_incr;

263

264 c->dst_incr_div = c->dst_incr / c->src_incr;

265 c->dst_incr_mod = c->dst_incr % c->src_incr;

266

267 return 0;

268 }

269

270 static int swri_resample(ResampleContext *c,

271 uint8_t *dst, const uint8_t *src, int *consumed,

272 int src_size, int dst_size, int update_ctx)

273 {

274 if (c->filter_length == 1 && c->phase_shift == 0) {

275 int index= c->index;

276 int frac= c->frac;

277 int64_t index2= (1LL<<32)*c->frac/c->src_incr + (1LL<<32)*index;

278 int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr;

279 int new_size = (src_size * (int64_t)c->src_incr - frac + c->dst_incr - 1) / c->dst_incr;

280

281 dst_size= FFMIN(dst_size, new_size);

282 c->dsp.resample_one(dst, src, dst_size, index2, incr);

283

284 index += dst_size * c->dst_incr_div;

285 index += (frac + dst_size * (int64_t)c->dst_incr_mod) / c->src_incr;

286 av_assert2(index >= 0);

287 *consumed= index;

288 if (update_ctx) {

289 c->frac = (frac + dst_size * (int64_t)c->dst_incr_mod) % c->src_incr;

290 c->index = 0;

291 }

292 } else {

293 int64_t end_index = (1LL + src_size - c->filter_length) << c->phase_shift;

294 int64_t delta_frac = (end_index - c->index) * c->src_incr - c->frac;

295 int delta_n = (delta_frac + c->dst_incr - 1) / c->dst_incr;

296

297 dst_size = FFMIN(dst_size, delta_n);

298 if (dst_size > 0) {

299 *consumed = c->dsp.resample(c, dst, src, dst_size, update_ctx);

300 } else {

301 *consumed = 0;

302 }

303 }

304

305 return dst_size;

306 }

307

308 static int multiple_resample(ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed){

309 int i, ret= -1;

310 int av_unused mm_flags = av_get_cpu_flags();

311 int need_emms = c->format == AV_SAMPLE_FMT_S16P && ARCH_X86_32 &&

312 (mm_flags & (AV_CPU_FLAG_MMX2 | AV_CPU_FLAG_SSE2)) == AV_CPU_FLAG_MMX2;

313 int64_t max_src_size = (INT64_MAX >> (c->phase_shift+1)) / c->src_incr;

314

315 if (c->compensation_distance)

316 dst_size = FFMIN(dst_size, c->compensation_distance);

317 src_size = FFMIN(src_size, max_src_size);

318

319 for(i=0; i<dst->ch_count; i++){

320 ret= swri_resample(c, dst->ch[i], src->ch[i],

321 consumed, src_size, dst_size, i+1==dst->ch_count);

322 }

323 if(need_emms)

324 emms_c();

325

326 if (c->compensation_distance) {

327 c->compensation_distance -= ret;

328 if (!c->compensation_distance) {

329 c->dst_incr = c->ideal_dst_incr;

330 c->dst_incr_div = c->dst_incr / c->src_incr;

331 c->dst_incr_mod = c->dst_incr % c->src_incr;

332 }

333 }

334

335 return ret;

336 }

337

338 static int64_t get_delay(struct SwrContext *s, int64_t base){

339 ResampleContext *c = s->resample;

340 int64_t num = s->in_buffer_count - (c->filter_length-1)/2;

341 num <<= c->phase_shift;

342 num -= c->index;

343 num *= c->src_incr;

344 num -= c->frac;

345 return av_rescale(num, base, s->in_sample_rate*(int64_t)c->src_incr << c->phase_shift);

346 }

347

348 static int resample_flush(struct SwrContext *s) {

349 AudioData *a= &s->in_buffer;

350 int i, j, ret;

351 if((ret = swri_realloc_audio(a, s->in_buffer_index + 2*s->in_buffer_count)) < 0)

352 return ret;

353 av_assert0(a->planar);

354 for(i=0; i<a->ch_count; i++){

355 for(j=0; j<s->in_buffer_count; j++){

356 memcpy(a->ch[i] + (s->in_buffer_index+s->in_buffer_count+j )*a->bps,

357 a->ch[i] + (s->in_buffer_index+s->in_buffer_count-j-1)*a->bps, a->bps);

358 }

359 }

360 s->in_buffer_count += (s->in_buffer_count+1)/2;

361 return 0;

362 }

363

364 // in fact the whole handle multiple ridiculously small buffers might need more thinking...

365 static int invert_initial_buffer(ResampleContext *c, AudioData *dst, const AudioData *src,

366 int in_count, int *out_idx, int *out_sz)

367 {

368 int n, ch, num = FFMIN(in_count + *out_sz, c->filter_length + 1), res;

369

370 if (c->index >= 0)

371 return 0;

372

373 if ((res = swri_realloc_audio(dst, c->filter_length * 2 + 1)) < 0)

374 return res;

375

376 // copy

377 for (n = *out_sz; n < num; n++) {

378 for (ch = 0; ch < src->ch_count; ch++) {

379 memcpy(dst->ch[ch] + ((c->filter_length + n) * c->felem_size),

380 src->ch[ch] + ((n - *out_sz) * c->felem_size), c->felem_size);

381 }

382 }

383

384 // if not enough data is in, return and wait for more

385 if (num < c->filter_length + 1) {

386 *out_sz = num;

387 *out_idx = c->filter_length;

388 return INT_MAX;

389 }

390

391 // else invert

392 for (n = 1; n <= c->filter_length; n++) {

393 for (ch = 0; ch < src->ch_count; ch++) {

394 memcpy(dst->ch[ch] + ((c->filter_length - n) * c->felem_size),

395 dst->ch[ch] + ((c->filter_length + n) * c->felem_size),

396 c->felem_size);

397 }

398 }

399

400 res = num - *out_sz;

401 *out_idx = c->filter_length + (c->index >> c->phase_shift);

402 *out_sz = FFMAX(*out_sz + c->filter_length,

403 1 + c->filter_length * 2) - *out_idx;

404 c->index &= c->phase_mask;

405

406 return FFMAX(res, 0);

407 }

408

409 struct Resampler const swri_resampler={

410 resample_init,

411 resample_free,

412 multiple_resample,

413 resample_flush,

414 set_compensation,

415 get_delay,

416 invert_initial_buffer,

417 };

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