FFmpeg: libavfilter/af_volume.c Source File

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libavfilter

af_volume.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 /**

23 * @file

24 * audio volume filter

25 */

27 #include "libavutil/channel_layout.h"

28 #include "libavutil/common.h"

29 #include "libavutil/eval.h"

30 #include "libavutil/float_dsp.h"

31 #include "libavutil/intreadwrite.h"

32 #include "libavutil/opt.h"

33 #include "libavutil/replaygain.h"

35 #include "audio.h"

36 #include "avfilter.h"

37 #include "formats.h"

38 #include "internal.h"

39 #include "af_volume.h"

41 static const char * const precision_str[] = {

42 "fixed", "float", "double"

43 };

45 static const char *const var_names[] = {

46 "n", ///< frame number (starting at zero)

47 "nb_channels", ///< number of channels

48 "nb_consumed_samples", ///< number of samples consumed by the filter

49 "nb_samples", ///< number of samples in the current frame

50 "pos", ///< position in the file of the frame

51 "pts", ///< frame presentation timestamp

52 "sample_rate", ///< sample rate

53 "startpts", ///< PTS at start of stream

54 "startt", ///< time at start of stream

55 "t", ///< time in the file of the frame

56 "tb", ///< timebase

57 "volume", ///< last set value

58 NULL

59 };

61 #define OFFSET(x) offsetof(VolumeContext, x)

62 #define A AV_OPT_FLAG_AUDIO_PARAM

63 #define F AV_OPT_FLAG_FILTERING_PARAM

65 static const AVOption volume_options[] = {

66 { "volume", "set volume adjustment expression",

67 OFFSET(volume_expr), AV_OPT_TYPE_STRING, { .str = "1.0" }, .flags = A|F },

68 { "precision", "select mathematical precision",

69 OFFSET(precision), AV_OPT_TYPE_INT, { .i64 = PRECISION_FLOAT }, PRECISION_FIXED, PRECISION_DOUBLE, A|F, "precision" },

70 { "fixed", "select 8-bit fixed-point", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_FIXED }, INT_MIN, INT_MAX, A|F, "precision" },

71 { "float", "select 32-bit floating-point", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_FLOAT }, INT_MIN, INT_MAX, A|F, "precision" },

72 { "double", "select 64-bit floating-point", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_DOUBLE }, INT_MIN, INT_MAX, A|F, "precision" },

73 { "eval", "specify when to evaluate expressions", OFFSET(eval_mode), AV_OPT_TYPE_INT, {.i64 = EVAL_MODE_ONCE}, 0, EVAL_MODE_NB-1, .flags = A|F, "eval" },

74 { "once", "eval volume expression once", 0, AV_OPT_TYPE_CONST, {.i64=EVAL_MODE_ONCE}, .flags = A|F, .unit = "eval" },

75 { "frame", "eval volume expression per-frame", 0, AV_OPT_TYPE_CONST, {.i64=EVAL_MODE_FRAME}, .flags = A|F, .unit = "eval" },

76 { "replaygain", "Apply replaygain side data when present",

77 OFFSET(replaygain), AV_OPT_TYPE_INT, { .i64 = REPLAYGAIN_DROP }, REPLAYGAIN_DROP, REPLAYGAIN_ALBUM, A, "replaygain" },

78 { "drop", "replaygain side data is dropped", 0, AV_OPT_TYPE_CONST, { .i64 = REPLAYGAIN_DROP }, 0, 0, A, "replaygain" },

79 { "ignore", "replaygain side data is ignored", 0, AV_OPT_TYPE_CONST, { .i64 = REPLAYGAIN_IGNORE }, 0, 0, A, "replaygain" },

80 { "track", "track gain is preferred", 0, AV_OPT_TYPE_CONST, { .i64 = REPLAYGAIN_TRACK }, 0, 0, A, "replaygain" },

81 { "album", "album gain is preferred", 0, AV_OPT_TYPE_CONST, { .i64 = REPLAYGAIN_ALBUM }, 0, 0, A, "replaygain" },

82 { "replaygain_preamp", "Apply replaygain pre-amplification",

83 OFFSET(replaygain_preamp), AV_OPT_TYPE_DOUBLE, { .dbl = 0.0 }, -15.0, 15.0, A },

84 { "replaygain_noclip", "Apply replaygain clipping prevention",

85 OFFSET(replaygain_noclip), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, A },

86 { NULL },

87 };

89 AVFILTER_DEFINE_CLASS(volume);

91 static int set_expr(AVExpr **pexpr, const char *expr, void *log_ctx)

92 {

93 int ret;

94 AVExpr *old = NULL;

96 if (*pexpr)

97 old = *pexpr;

98 ret = av_expr_parse(pexpr, expr, var_names,

99 NULL, NULL, NULL, NULL, 0, log_ctx);

100 if (ret < 0) {

101 av_log(log_ctx, AV_LOG_ERROR,

102 "Error when evaluating the volume expression '%s'\n", expr);

103 *pexpr = old;

104 return ret;

105 }

106

107 av_expr_free(old);

108 return 0;

109 }

110

111 static av_cold int init(AVFilterContext *ctx)

112 {

113 VolumeContext *vol = ctx->priv;

114

115 vol->fdsp = avpriv_float_dsp_alloc(0);

116 if (!vol->fdsp)

117 return AVERROR(ENOMEM);

118

119 return set_expr(&vol->volume_pexpr, vol->volume_expr, ctx);

120 }

121

122 static av_cold void uninit(AVFilterContext *ctx)

123 {

124 VolumeContext *vol = ctx->priv;

125 av_expr_free(vol->volume_pexpr);

126 av_opt_free(vol);

127 av_freep(&vol->fdsp);

128 }

129

130 static int query_formats(AVFilterContext *ctx)

131 {

132 VolumeContext *vol = ctx->priv;

133 AVFilterFormats *formats = NULL;

134 AVFilterChannelLayouts *layouts;

135 static const enum AVSampleFormat sample_fmts[][7] = {

136 [PRECISION_FIXED] = {

137 AV_SAMPLE_FMT_U8,

138 AV_SAMPLE_FMT_U8P,

139 AV_SAMPLE_FMT_S16,

140 AV_SAMPLE_FMT_S16P,

141 AV_SAMPLE_FMT_S32,

142 AV_SAMPLE_FMT_S32P,

143 AV_SAMPLE_FMT_NONE

144 },

145 [PRECISION_FLOAT] = {

146 AV_SAMPLE_FMT_FLT,

147 AV_SAMPLE_FMT_FLTP,

148 AV_SAMPLE_FMT_NONE

149 },

150 [PRECISION_DOUBLE] = {

151 AV_SAMPLE_FMT_DBL,

152 AV_SAMPLE_FMT_DBLP,

153 AV_SAMPLE_FMT_NONE

154 }

155 };

156

157 layouts = ff_all_channel_counts();

158 if (!layouts)

159 return AVERROR(ENOMEM);

160 ff_set_common_channel_layouts(ctx, layouts);

161

162 formats = ff_make_format_list(sample_fmts[vol->precision]);

163 if (!formats)

164 return AVERROR(ENOMEM);

165 ff_set_common_formats(ctx, formats);

166

167 formats = ff_all_samplerates();

168 if (!formats)

169 return AVERROR(ENOMEM);

170 ff_set_common_samplerates(ctx, formats);

171

172 return 0;

173 }

174

175 static inline void scale_samples_u8(uint8_t *dst, const uint8_t *src,

176 int nb_samples, int volume)

177 {

178 int i;

179 for (i = 0; i < nb_samples; i++)

180 dst[i] = av_clip_uint8(((((int64_t)src[i] - 128) * volume + 128) >> 8) + 128);

181 }

182

183 static inline void scale_samples_u8_small(uint8_t *dst, const uint8_t *src,

184 int nb_samples, int volume)

185 {

186 int i;

187 for (i = 0; i < nb_samples; i++)

188 dst[i] = av_clip_uint8((((src[i] - 128) * volume + 128) >> 8) + 128);

189 }

190

191 static inline void scale_samples_s16(uint8_t *dst, const uint8_t *src,

192 int nb_samples, int volume)

193 {

194 int i;

195 int16_t *smp_dst = (int16_t *)dst;

196 const int16_t *smp_src = (const int16_t *)src;

197 for (i = 0; i < nb_samples; i++)

198 smp_dst[i] = av_clip_int16(((int64_t)smp_src[i] * volume + 128) >> 8);

199 }

200

201 static inline void scale_samples_s16_small(uint8_t *dst, const uint8_t *src,

202 int nb_samples, int volume)

203 {

204 int i;

205 int16_t *smp_dst = (int16_t *)dst;

206 const int16_t *smp_src = (const int16_t *)src;

207 for (i = 0; i < nb_samples; i++)

208 smp_dst[i] = av_clip_int16((smp_src[i] * volume + 128) >> 8);

209 }

210

211 static inline void scale_samples_s32(uint8_t *dst, const uint8_t *src,

212 int nb_samples, int volume)

213 {

214 int i;

215 int32_t *smp_dst = (int32_t *)dst;

216 const int32_t *smp_src = (const int32_t *)src;

217 for (i = 0; i < nb_samples; i++)

218 smp_dst[i] = av_clipl_int32((((int64_t)smp_src[i] * volume + 128) >> 8));

219 }

220

221 static av_cold void volume_init(VolumeContext *vol)

222 {

223 vol->samples_align = 1;

224

225 switch (av_get_packed_sample_fmt(vol->sample_fmt)) {

226 case AV_SAMPLE_FMT_U8:

227 if (vol->volume_i < 0x1000000)

228 vol->scale_samples = scale_samples_u8_small;

229 else

230 vol->scale_samples = scale_samples_u8;

231 break;

232 case AV_SAMPLE_FMT_S16:

233 if (vol->volume_i < 0x10000)

234 vol->scale_samples = scale_samples_s16_small;

235 else

236 vol->scale_samples = scale_samples_s16;

237 break;

238 case AV_SAMPLE_FMT_S32:

239 vol->scale_samples = scale_samples_s32;

240 break;

241 case AV_SAMPLE_FMT_FLT:

242 vol->samples_align = 4;

243 break;

244 case AV_SAMPLE_FMT_DBL:

245 vol->samples_align = 8;

246 break;

247 }

248

249 if (ARCH_X86)

250 ff_volume_init_x86(vol);

251 }

252

253 static int set_volume(AVFilterContext *ctx)

254 {

255 VolumeContext *vol = ctx->priv;

256

257 vol->volume = av_expr_eval(vol->volume_pexpr, vol->var_values, NULL);

258 if (isnan(vol->volume)) {

259 if (vol->eval_mode == EVAL_MODE_ONCE) {

260 av_log(ctx, AV_LOG_ERROR, "Invalid value NaN for volume\n");

261 return AVERROR(EINVAL);

262 } else {

263 av_log(ctx, AV_LOG_WARNING, "Invalid value NaN for volume, setting to 0\n");

264 vol->volume = 0;

265 }

266 }

267 vol->var_values[VAR_VOLUME] = vol->volume;

268

269 av_log(ctx, AV_LOG_VERBOSE, "n:%f t:%f pts:%f precision:%s ",

270 vol->var_values[VAR_N], vol->var_values[VAR_T], vol->var_values[VAR_PTS],

271 precision_str[vol->precision]);

272

273 if (vol->precision == PRECISION_FIXED) {

274 vol->volume_i = (int)(vol->volume * 256 + 0.5);

275 vol->volume = vol->volume_i / 256.0;

276 av_log(ctx, AV_LOG_VERBOSE, "volume_i:%d/255 ", vol->volume_i);

277 }

278 av_log(ctx, AV_LOG_VERBOSE, "volume:%f volume_dB:%f\n",

279 vol->volume, 20.0*log(vol->volume)/M_LN10);

280

281 volume_init(vol);

282 return 0;

283 }

284

285 static int config_output(AVFilterLink *outlink)

286 {

287 AVFilterContext *ctx = outlink->src;

288 VolumeContext *vol = ctx->priv;

289 AVFilterLink *inlink = ctx->inputs[0];

290

291 vol->sample_fmt = inlink->format;

292 vol->channels = inlink->channels;

293 vol->planes = av_sample_fmt_is_planar(inlink->format) ? vol->channels : 1;

294

295 vol->var_values[VAR_N] =

296 vol->var_values[VAR_NB_CONSUMED_SAMPLES] =

297 vol->var_values[VAR_NB_SAMPLES] =

298 vol->var_values[VAR_POS] =

299 vol->var_values[VAR_PTS] =

300 vol->var_values[VAR_STARTPTS] =

301 vol->var_values[VAR_STARTT] =

302 vol->var_values[VAR_T] =

303 vol->var_values[VAR_VOLUME] = NAN;

304

305 vol->var_values[VAR_NB_CHANNELS] = inlink->channels;

306 vol->var_values[VAR_TB] = av_q2d(inlink->time_base);

307 vol->var_values[VAR_SAMPLE_RATE] = inlink->sample_rate;

308

309 av_log(inlink->src, AV_LOG_VERBOSE, "tb:%f sample_rate:%f nb_channels:%f\n",

310 vol->var_values[VAR_TB],

311 vol->var_values[VAR_SAMPLE_RATE],

312 vol->var_values[VAR_NB_CHANNELS]);

313

314 return set_volume(ctx);

315 }

316

317 static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,

318 char *res, int res_len, int flags)

319 {

320 VolumeContext *vol = ctx->priv;

321 int ret = AVERROR(ENOSYS);

322

323 if (!strcmp(cmd, "volume")) {

324 if ((ret = set_expr(&vol->volume_pexpr, args, ctx)) < 0)

325 return ret;

326 if (vol->eval_mode == EVAL_MODE_ONCE)

327 set_volume(ctx);

328 }

329

330 return ret;

331 }

332

333 #define D2TS(d) (isnan(d) ? AV_NOPTS_VALUE : (int64_t)(d))

334 #define TS2D(ts) ((ts) == AV_NOPTS_VALUE ? NAN : (double)(ts))

335 #define TS2T(ts, tb) ((ts) == AV_NOPTS_VALUE ? NAN : (double)(ts)*av_q2d(tb))

336

337 static int filter_frame(AVFilterLink *inlink, AVFrame *buf)

338 {

339 AVFilterContext *ctx = inlink->dst;

340 VolumeContext *vol = inlink->dst->priv;

341 AVFilterLink *outlink = inlink->dst->outputs[0];

342 int nb_samples = buf->nb_samples;

343 AVFrame *out_buf;

344 int64_t pos;

345 AVFrameSideData *sd = av_frame_get_side_data(buf, AV_FRAME_DATA_REPLAYGAIN);

346 int ret;

347

348 if (sd && vol->replaygain != REPLAYGAIN_IGNORE) {

349 if (vol->replaygain != REPLAYGAIN_DROP) {

350 AVReplayGain *replaygain = (AVReplayGain*)sd->data;

351 int32_t gain = 100000;

352 uint32_t peak = 100000;

353 float g, p;

354

355 if (vol->replaygain == REPLAYGAIN_TRACK &&

356 replaygain->track_gain != INT32_MIN) {

357 gain = replaygain->track_gain;

358

359 if (replaygain->track_peak != 0)

360 peak = replaygain->track_peak;

361 } else if (replaygain->album_gain != INT32_MIN) {

362 gain = replaygain->album_gain;

363

364 if (replaygain->album_peak != 0)

365 peak = replaygain->album_peak;

366 } else {

367 av_log(inlink->dst, AV_LOG_WARNING, "Both ReplayGain gain "

368 "values are unknown.\n");

369 }

370 g = gain / 100000.0f;

371 p = peak / 100000.0f;

372

373 av_log(inlink->dst, AV_LOG_VERBOSE,

374 "Using gain %f dB from replaygain side data.\n", g);

375

376 vol->volume = pow(10, (g + vol->replaygain_preamp) / 20);

377 if (vol->replaygain_noclip)

378 vol->volume = FFMIN(vol->volume, 1.0 / p);

379 vol->volume_i = (int)(vol->volume * 256 + 0.5);

380

381 volume_init(vol);

382 }

383 av_frame_remove_side_data(buf, AV_FRAME_DATA_REPLAYGAIN);

384 }

385

386 if (isnan(vol->var_values[VAR_STARTPTS])) {

387 vol->var_values[VAR_STARTPTS] = TS2D(buf->pts);

388 vol->var_values[VAR_STARTT ] = TS2T(buf->pts, inlink->time_base);

389 }

390 vol->var_values[VAR_PTS] = TS2D(buf->pts);

391 vol->var_values[VAR_T ] = TS2T(buf->pts, inlink->time_base);

392 vol->var_values[VAR_N ] = inlink->frame_count;

393

394 pos = av_frame_get_pkt_pos(buf);

395 vol->var_values[VAR_POS] = pos == -1 ? NAN : pos;

396 if (vol->eval_mode == EVAL_MODE_FRAME)

397 set_volume(ctx);

398

399 if (vol->volume == 1.0 || vol->volume_i == 256) {

400 out_buf = buf;

401 goto end;

402 }

403

404 /* do volume scaling in-place if input buffer is writable */

405 if (av_frame_is_writable(buf)

406 && (vol->precision != PRECISION_FIXED || vol->volume_i > 0)) {

407 out_buf = buf;

408 } else {

409 out_buf = ff_get_audio_buffer(inlink, nb_samples);

410 if (!out_buf)

411 return AVERROR(ENOMEM);

412 ret = av_frame_copy_props(out_buf, buf);

413 if (ret < 0) {

414 av_frame_free(&out_buf);

415 av_frame_free(&buf);

416 return ret;

417 }

418 }

419

420 if (vol->precision != PRECISION_FIXED || vol->volume_i > 0) {

421 int p, plane_samples;

422

423 if (av_sample_fmt_is_planar(buf->format))

424 plane_samples = FFALIGN(nb_samples, vol->samples_align);

425 else

426 plane_samples = FFALIGN(nb_samples * vol->channels, vol->samples_align);

427

428 if (vol->precision == PRECISION_FIXED) {

429 for (p = 0; p < vol->planes; p++) {

430 vol->scale_samples(out_buf->extended_data[p],

431 buf->extended_data[p], plane_samples,

432 vol->volume_i);

433 }

434 } else if (av_get_packed_sample_fmt(vol->sample_fmt) == AV_SAMPLE_FMT_FLT) {

435 for (p = 0; p < vol->planes; p++) {

436 vol->fdsp->vector_fmul_scalar((float *)out_buf->extended_data[p],

437 (const float *)buf->extended_data[p],

438 vol->volume, plane_samples);

439 }

440 } else {

441 for (p = 0; p < vol->planes; p++) {

442 vol->fdsp->vector_dmul_scalar((double *)out_buf->extended_data[p],

443 (const double *)buf->extended_data[p],

444 vol->volume, plane_samples);

445 }

446 }

447 }

448

449 emms_c();

450

451 if (buf != out_buf)

452 av_frame_free(&buf);

453

454 end:

455 vol->var_values[VAR_NB_CONSUMED_SAMPLES] += out_buf->nb_samples;

456 return ff_filter_frame(outlink, out_buf);

457 }

458

459 static const AVFilterPad avfilter_af_volume_inputs[] = {

460 {

461 .name = "default",

462 .type = AVMEDIA_TYPE_AUDIO,

463 .filter_frame = filter_frame,

464 },

465 { NULL }

466 };

467

468 static const AVFilterPad avfilter_af_volume_outputs[] = {

469 {

470 .name = "default",

471 .type = AVMEDIA_TYPE_AUDIO,

472 .config_props = config_output,

473 },

474 { NULL }

475 };

476

477 AVFilter ff_af_volume = {

478 .name = "volume",

479 .description = NULL_IF_CONFIG_SMALL("Change input volume."),

480 .query_formats = query_formats,

481 .priv_size = sizeof(VolumeContext),

482 .priv_class = &volume_class,

483 .init = init,

484 .uninit = uninit,

485 .inputs = avfilter_af_volume_inputs,

486 .outputs = avfilter_af_volume_outputs,

487 .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC,

488 .process_command = process_command,

489 };

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