FFmpeg: libavresample/resample.c Source File

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libavresample

resample.c

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1 /*

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 #include "libavutil/common.h"

23 #include "libavutil/libm.h"

24 #include "libavutil/log.h"

25 #include "internal.h"

26 #include "resample.h"

27 #include "audio_data.h"

29 struct ResampleContext {

30 AVAudioResampleContext *avr;

31 AudioData *buffer;

32 uint8_t *filter_bank;

33 int filter_length;

34 int ideal_dst_incr;

35 int dst_incr;

36 int index;

37 int frac;

38 int src_incr;

39 int compensation_distance;

40 int phase_shift;

41 int phase_mask;

42 int linear;

43 enum AVResampleFilterType filter_type;

44 int kaiser_beta;

45 double factor;

46 void (*set_filter)(void *filter, double *tab, int phase, int tap_count);

47 void (*resample_one)(struct ResampleContext *c, int no_filter, void *dst0,

48 int dst_index, const void *src0, int src_size,

49 int index, int frac);

50 };

53 /* double template */

54 #define CONFIG_RESAMPLE_DBL

55 #include "resample_template.c"

56 #undef CONFIG_RESAMPLE_DBL

58 /* float template */

59 #define CONFIG_RESAMPLE_FLT

60 #include "resample_template.c"

61 #undef CONFIG_RESAMPLE_FLT

63 /* s32 template */

64 #define CONFIG_RESAMPLE_S32

65 #include "resample_template.c"

66 #undef CONFIG_RESAMPLE_S32

68 /* s16 template */

69 #include "resample_template.c"

72 /* 0th order modified bessel function of the first kind. */

73 static double bessel(double x)

74 {

75 double v = 1;

76 double lastv = 0;

77 double t = 1;

78 int i;

80 x = x * x / 4;

81 for (i = 1; v != lastv; i++) {

82 lastv = v;

83 t *= x / (i * i);

84 v += t;

85 }

86 return v;

87 }

89 /* Build a polyphase filterbank. */

90 static int build_filter(ResampleContext *c)

91 {

92 int ph, i;

93 double x, y, w, factor;

94 double *tab;

95 int tap_count = c->filter_length;

96 int phase_count = 1 << c->phase_shift;

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

99 tab = av_malloc(tap_count * sizeof(*tab));

100 if (!tab)

101 return AVERROR(ENOMEM);

102

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

104 factor = FFMIN(c->factor, 1.0);

105

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

107 double norm = 0;

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

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

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

111 else y = sin(x) / x;

112 switch (c->filter_type) {

113 case AV_RESAMPLE_FILTER_TYPE_CUBIC: {

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

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

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

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

118 break;

119 }

120 case AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL:

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

122 y *= 0.3635819 - 0.4891775 * cos( w) +

123 0.1365995 * cos(2 * w) -

124 0.0106411 * cos(3 * w);

125 break;

126 case AV_RESAMPLE_FILTER_TYPE_KAISER:

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

128 y *= bessel(c->kaiser_beta * sqrt(FFMAX(1 - w * w, 0)));

129 break;

130 }

131

132 tab[i] = y;

133 norm += y;

134 }

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

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

137 tab[i] = tab[i] / norm;

138

139 c->set_filter(c->filter_bank, tab, ph, tap_count);

140 }

141

142 av_free(tab);

143 return 0;

144 }

145

146 ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr)

147 {

148 ResampleContext *c;

149 int out_rate = avr->out_sample_rate;

150 int in_rate = avr->in_sample_rate;

151 double factor = FFMIN(out_rate * avr->cutoff / in_rate, 1.0);

152 int phase_count = 1 << avr->phase_shift;

153 int felem_size;

154

155 if (avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P &&

156 avr->internal_sample_fmt != AV_SAMPLE_FMT_S32P &&

157 avr->internal_sample_fmt != AV_SAMPLE_FMT_FLTP &&

158 avr->internal_sample_fmt != AV_SAMPLE_FMT_DBLP) {

159 av_log(avr, AV_LOG_ERROR, "Unsupported internal format for "

160 "resampling: %s\n",

161 av_get_sample_fmt_name(avr->internal_sample_fmt));

162 return NULL;

163 }

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

165 if (!c)

166 return NULL;

167

168 c->avr = avr;

169 c->phase_shift = avr->phase_shift;

170 c->phase_mask = phase_count - 1;

171 c->linear = avr->linear_interp;

172 c->factor = factor;

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

174 c->filter_type = avr->filter_type;

175 c->kaiser_beta = avr->kaiser_beta;

176

177 switch (avr->internal_sample_fmt) {

178 case AV_SAMPLE_FMT_DBLP:

179 c->resample_one = resample_one_dbl;

180 c->set_filter = set_filter_dbl;

181 break;

182 case AV_SAMPLE_FMT_FLTP:

183 c->resample_one = resample_one_flt;

184 c->set_filter = set_filter_flt;

185 break;

186 case AV_SAMPLE_FMT_S32P:

187 c->resample_one = resample_one_s32;

188 c->set_filter = set_filter_s32;

189 break;

190 case AV_SAMPLE_FMT_S16P:

191 c->resample_one = resample_one_s16;

192 c->set_filter = set_filter_s16;

193 break;

194 }

195

196 felem_size = av_get_bytes_per_sample(avr->internal_sample_fmt);

197 c->filter_bank = av_mallocz(c->filter_length * (phase_count + 1) * felem_size);

198 if (!c->filter_bank)

199 goto error;

200

201 if (build_filter(c) < 0)

202 goto error;

203

204 memcpy(&c->filter_bank[(c->filter_length * phase_count + 1) * felem_size],

205 c->filter_bank, (c->filter_length - 1) * felem_size);

206 memcpy(&c->filter_bank[c->filter_length * phase_count * felem_size],

207 &c->filter_bank[(c->filter_length - 1) * felem_size], felem_size);

208

209 c->compensation_distance = 0;

210 if (!av_reduce(&c->src_incr, &c->dst_incr, out_rate,

211 in_rate * (int64_t)phase_count, INT32_MAX / 2))

212 goto error;

213 c->ideal_dst_incr = c->dst_incr;

214

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

216 c->frac = 0;

217

218 /* allocate internal buffer */

219 c->buffer = ff_audio_data_alloc(avr->resample_channels, 0,

220 avr->internal_sample_fmt,

221 "resample buffer");

222 if (!c->buffer)

223 goto error;

224

225 av_log(avr, AV_LOG_DEBUG, "resample: %s from %d Hz to %d Hz\n",

226 av_get_sample_fmt_name(avr->internal_sample_fmt),

227 avr->in_sample_rate, avr->out_sample_rate);

228

229 return c;

230

231 error:

232 ff_audio_data_free(&c->buffer);

233 av_free(c->filter_bank);

234 av_free(c);

235 return NULL;

236 }

237

238 void ff_audio_resample_free(ResampleContext **c)

239 {

240 if (!*c)

241 return;

242 ff_audio_data_free(&(*c)->buffer);

243 av_free((*c)->filter_bank);

244 av_freep(c);

245 }

246

247 int avresample_set_compensation(AVAudioResampleContext *avr, int sample_delta,

248 int compensation_distance)

249 {

250 ResampleContext *c;

251 AudioData *fifo_buf = NULL;

252 int ret = 0;

253

254 if (compensation_distance < 0)

255 return AVERROR(EINVAL);

256 if (!compensation_distance && sample_delta)

257 return AVERROR(EINVAL);

258

259 if (!avr->resample_needed) {

260 #if FF_API_RESAMPLE_CLOSE_OPEN

261 /* if resampling was not enabled previously, re-initialize the

262 AVAudioResampleContext and force resampling */

263 int fifo_samples;

264 int restore_matrix = 0;

265 double matrix[AVRESAMPLE_MAX_CHANNELS * AVRESAMPLE_MAX_CHANNELS] = { 0 };

266

267 /* buffer any remaining samples in the output FIFO before closing */

268 fifo_samples = av_audio_fifo_size(avr->out_fifo);

269 if (fifo_samples > 0) {

270 fifo_buf = ff_audio_data_alloc(avr->out_channels, fifo_samples,

271 avr->out_sample_fmt, NULL);

272 if (!fifo_buf)

273 return AVERROR(EINVAL);

274 ret = ff_audio_data_read_from_fifo(avr->out_fifo, fifo_buf,

275 fifo_samples);

276 if (ret < 0)

277 goto reinit_fail;

278 }

279 /* save the channel mixing matrix */

280 if (avr->am) {

281 ret = avresample_get_matrix(avr, matrix, AVRESAMPLE_MAX_CHANNELS);

282 if (ret < 0)

283 goto reinit_fail;

284 restore_matrix = 1;

285 }

286

287 /* close the AVAudioResampleContext */

288 avresample_close(avr);

289

290 avr->force_resampling = 1;

291

292 /* restore the channel mixing matrix */

293 if (restore_matrix) {

294 ret = avresample_set_matrix(avr, matrix, AVRESAMPLE_MAX_CHANNELS);

295 if (ret < 0)

296 goto reinit_fail;

297 }

298

299 /* re-open the AVAudioResampleContext */

300 ret = avresample_open(avr);

301 if (ret < 0)

302 goto reinit_fail;

303

304 /* restore buffered samples to the output FIFO */

305 if (fifo_samples > 0) {

306 ret = ff_audio_data_add_to_fifo(avr->out_fifo, fifo_buf, 0,

307 fifo_samples);

308 if (ret < 0)

309 goto reinit_fail;

310 ff_audio_data_free(&fifo_buf);

311 }

312 #else

313 av_log(avr, AV_LOG_ERROR, "Unable to set resampling compensation\n");

314 return AVERROR(EINVAL);

315 #endif

316 }

317 c = avr->resample;

318 c->compensation_distance = compensation_distance;

319 if (compensation_distance) {

320 c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr *

321 (int64_t)sample_delta / compensation_distance;

322 } else {

323 c->dst_incr = c->ideal_dst_incr;

324 }

325 return 0;

326

327 reinit_fail:

328 ff_audio_data_free(&fifo_buf);

329 return ret;

330 }

331

332 static int resample(ResampleContext *c, void *dst, const void *src,

333 int *consumed, int src_size, int dst_size, int update_ctx)

334 {

335 int dst_index;

336 int index = c->index;

337 int frac = c->frac;

338 int dst_incr_frac = c->dst_incr % c->src_incr;

339 int dst_incr = c->dst_incr / c->src_incr;

340 int compensation_distance = c->compensation_distance;

341

342 if (!dst != !src)

343 return AVERROR(EINVAL);

344

345 if (compensation_distance == 0 && c->filter_length == 1 &&

346 c->phase_shift == 0) {

347 int64_t index2 = ((int64_t)index) << 32;

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

349 dst_size = FFMIN(dst_size,

350 (src_size-1-index) * (int64_t)c->src_incr /

351 c->dst_incr);

352

353 if (dst) {

354 for(dst_index = 0; dst_index < dst_size; dst_index++) {

355 c->resample_one(c, 1, dst, dst_index, src, 0, index2 >> 32, 0);

356 index2 += incr;

357 }

358 } else {

359 dst_index = dst_size;

360 }

361 index += dst_index * dst_incr;

362 index += (frac + dst_index * (int64_t)dst_incr_frac) / c->src_incr;

363 frac = (frac + dst_index * (int64_t)dst_incr_frac) % c->src_incr;

364 } else {

365 for (dst_index = 0; dst_index < dst_size; dst_index++) {

366 int sample_index = index >> c->phase_shift;

367

368 if (sample_index + c->filter_length > src_size ||

369 -sample_index >= src_size)

370 break;

371

372 if (dst)

373 c->resample_one(c, 0, dst, dst_index, src, src_size, index, frac);

374

375 frac += dst_incr_frac;

376 index += dst_incr;

377 if (frac >= c->src_incr) {

378 frac -= c->src_incr;

379 index++;

380 }

381 if (dst_index + 1 == compensation_distance) {

382 compensation_distance = 0;

383 dst_incr_frac = c->ideal_dst_incr % c->src_incr;

384 dst_incr = c->ideal_dst_incr / c->src_incr;

385 }

386 }

387 }

388 if (consumed)

389 *consumed = FFMAX(index, 0) >> c->phase_shift;

390

391 if (update_ctx) {

392 if (index >= 0)

393 index &= c->phase_mask;

394

395 if (compensation_distance) {

396 compensation_distance -= dst_index;

397 if (compensation_distance <= 0)

398 return AVERROR_BUG;

399 }

400 c->frac = frac;

401 c->index = index;

402 c->dst_incr = dst_incr_frac + c->src_incr*dst_incr;

403 c->compensation_distance = compensation_distance;

404 }

405

406 return dst_index;

407 }

408

409 int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src)

410 {

411 int ch, in_samples, in_leftover, consumed = 0, out_samples = 0;

412 int ret = AVERROR(EINVAL);

413

414 in_samples = src ? src->nb_samples : 0;

415 in_leftover = c->buffer->nb_samples;

416

417 /* add input samples to the internal buffer */

418 if (src) {

419 ret = ff_audio_data_combine(c->buffer, in_leftover, src, 0, in_samples);

420 if (ret < 0)

421 return ret;

422 } else if (!in_leftover) {

423 /* no remaining samples to flush */

424 return 0;

425 } else {

426 /* TODO: pad buffer to flush completely */

427 }

428

429 /* calculate output size and reallocate output buffer if needed */

430 /* TODO: try to calculate this without the dummy resample() run */

431 if (!dst->read_only && dst->allow_realloc) {

432 out_samples = resample(c, NULL, NULL, NULL, c->buffer->nb_samples,

433 INT_MAX, 0);

434 ret = ff_audio_data_realloc(dst, out_samples);

435 if (ret < 0) {

436 av_log(c->avr, AV_LOG_ERROR, "error reallocating output\n");

437 return ret;

438 }

439 }

440

441 /* resample each channel plane */

442 for (ch = 0; ch < c->buffer->channels; ch++) {

443 out_samples = resample(c, (void *)dst->data[ch],

444 (const void *)c->buffer->data[ch], &consumed,

445 c->buffer->nb_samples, dst->allocated_samples,

446 ch + 1 == c->buffer->channels);

447 }

448 if (out_samples < 0) {

449 av_log(c->avr, AV_LOG_ERROR, "error during resampling\n");

450 return out_samples;

451 }

452

453 /* drain consumed samples from the internal buffer */

454 ff_audio_data_drain(c->buffer, consumed);

455

456 av_dlog(c->avr, "resampled %d in + %d leftover to %d out + %d leftover\n",

457 in_samples, in_leftover, out_samples, c->buffer->nb_samples);

458

459 dst->nb_samples = out_samples;

460 return 0;

461 }

462

463 int avresample_get_delay(AVAudioResampleContext *avr)

464 {

465 if (!avr->resample_needed || !avr->resample)

466 return 0;

467

468 return avr->resample->buffer->nb_samples;

469 }

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