~ubuntu-branches/ubuntu/trusty/gst-libav1.0/trusty-proposed : revision 19

gst-libs/ext/libav/libavfilter/af_amix.c

* Audio Mix Filter

* This file is part of Libav.

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

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

* License as published by the Free Software Foundation; either

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

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

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

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

* Lesser General Public License for more details.

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

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

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

/**

* @file

* Audio Mix Filter

* Mixes audio from multiple sources into a single output. The channel layout,

* sample rate, and sample format will be the same for all inputs and the

* output.

#include "libavutil/audio_fifo.h"

#include "libavutil/avassert.h"

#include "libavutil/avstring.h"

#include "libavutil/channel_layout.h"

#include "libavutil/common.h"

#include "libavutil/float_dsp.h"

#include "libavutil/mathematics.h"

#include "libavutil/opt.h"

#include "libavutil/samplefmt.h"

#include "audio.h"

#include "avfilter.h"

#include "formats.h"

#include "internal.h"

#define INPUT_OFF 0 /**< input has reached EOF */

#define INPUT_ON 1 /**< input is active */

#define INPUT_INACTIVE 2 /**< input is on, but is currently inactive */

#define DURATION_LONGEST 0

#define DURATION_SHORTEST 1

#define DURATION_FIRST 2

typedef struct FrameInfo {

int nb_samples;

int64_t pts;

struct FrameInfo *next;

} FrameInfo;

/**

* Linked list used to store timestamps and frame sizes of all frames in the

* FIFO for the first input.

* This is needed to keep timestamps synchronized for the case where multiple

* input frames are pushed to the filter for processing before a frame is

* requested by the output link.

typedef struct FrameList {

int nb_frames;

int nb_samples;

FrameInfo *list;

FrameInfo *end;

} FrameList;

static void frame_list_clear(FrameList *frame_list)

{

if (frame_list) {

while (frame_list->list) {

FrameInfo *info = frame_list->list;

frame_list->list = info->next;

av_free(info);

}

frame_list->nb_frames = 0;

frame_list->nb_samples = 0;

frame_list->end = NULL;

}

static int frame_list_next_frame_size(FrameList *frame_list)

{

if (!frame_list->list)

return 0;

return frame_list->list->nb_samples;

}

static int64_t frame_list_next_pts(FrameList *frame_list)

{

if (!frame_list->list)

100

return AV_NOPTS_VALUE;

101

return frame_list->list->pts;

102

}

103

104

static void frame_list_remove_samples(FrameList *frame_list, int nb_samples)

105

{

106

if (nb_samples >= frame_list->nb_samples) {

107

frame_list_clear(frame_list);

108

} else {

109

int samples = nb_samples;

110

while (samples > 0) {

111

FrameInfo *info = frame_list->list;

112

av_assert0(info != NULL);

113

if (info->nb_samples <= samples) {

114

samples -= info->nb_samples;

115

frame_list->list = info->next;

116

if (!frame_list->list)

117

frame_list->end = NULL;

118

frame_list->nb_frames--;

119

frame_list->nb_samples -= info->nb_samples;

120

av_free(info);

121

} else {

122

info->nb_samples -= samples;

123

info->pts += samples;

124

frame_list->nb_samples -= samples;

125

samples = 0;

126

}

127

}

128

}

129

}

130

131

static int frame_list_add_frame(FrameList *frame_list, int nb_samples, int64_t pts)

132

{

133

FrameInfo *info = av_malloc(sizeof(*info));

134

if (!info)

135

return AVERROR(ENOMEM);

136

info->nb_samples = nb_samples;

137

info->pts = pts;

138

info->next = NULL;

139

140

if (!frame_list->list) {

141

frame_list->list = info;

142

frame_list->end = info;

143

} else {

144

av_assert0(frame_list->end != NULL);

145

frame_list->end->next = info;

146

frame_list->end = info;

147

}

148

frame_list->nb_frames++;

149

frame_list->nb_samples += nb_samples;

150

151

return 0;

152

}

153

154

155

typedef struct MixContext {

156

const AVClass *class; /**< class for AVOptions */

157

AVFloatDSPContext fdsp;

158

159

int nb_inputs; /**< number of inputs */

160

int active_inputs; /**< number of input currently active */

161

int duration_mode; /**< mode for determining duration */

162

float dropout_transition; /**< transition time when an input drops out */

163

164

int nb_channels; /**< number of channels */

165

int sample_rate; /**< sample rate */

166

int planar;

167

AVAudioFifo **fifos; /**< audio fifo for each input */

168

uint8_t *input_state; /**< current state of each input */

169

float *input_scale; /**< mixing scale factor for each input */

170

float scale_norm; /**< normalization factor for all inputs */

171

int64_t next_pts; /**< calculated pts for next output frame */

172

FrameList *frame_list; /**< list of frame info for the first input */

173

} MixContext;

174

175

#define OFFSET(x) offsetof(MixContext, x)

176

#define A AV_OPT_FLAG_AUDIO_PARAM

177

static const AVOption options[] = {

178

{ "inputs", "Number of inputs.",

179

OFFSET(nb_inputs), AV_OPT_TYPE_INT, { .i64 = 2 }, 1, 32, A },

180

{ "duration", "How to determine the end-of-stream.",

181

OFFSET(duration_mode), AV_OPT_TYPE_INT, { .i64 = DURATION_LONGEST }, 0, 2, A, "duration" },

182

{ "longest", "Duration of longest input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_LONGEST }, INT_MIN, INT_MAX, A, "duration" },

183

{ "shortest", "Duration of shortest input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_SHORTEST }, INT_MIN, INT_MAX, A, "duration" },

184

{ "first", "Duration of first input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_FIRST }, INT_MIN, INT_MAX, A, "duration" },

185

{ "dropout_transition", "Transition time, in seconds, for volume "

186

"renormalization when an input stream ends.",

187

OFFSET(dropout_transition), AV_OPT_TYPE_FLOAT, { .dbl = 2.0 }, 0, INT_MAX, A },

188

{ NULL },

189

};

190

191

static const AVClass amix_class = {

192

.class_name = "amix filter",

193

.item_name = av_default_item_name,

194

.option = options,

195

.version = LIBAVUTIL_VERSION_INT,

196

};

197

198

199

/**

200

* Update the scaling factors to apply to each input during mixing.

201

202

* This balances the full volume range between active inputs and handles

203

* volume transitions when EOF is encountered on an input but mixing continues

204

* with the remaining inputs.

205

206

static void calculate_scales(MixContext *s, int nb_samples)

207

{

208

int i;

209

210

if (s->scale_norm > s->active_inputs) {

211

s->scale_norm -= nb_samples / (s->dropout_transition * s->sample_rate);

212

s->scale_norm = FFMAX(s->scale_norm, s->active_inputs);

213

}

214

215

for (i = 0; i < s->nb_inputs; i++) {

216

if (s->input_state[i] == INPUT_ON)

217

s->input_scale[i] = 1.0f / s->scale_norm;

218

else

219

s->input_scale[i] = 0.0f;

220

}

221

}

222

223

static int config_output(AVFilterLink *outlink)

224

{

225

AVFilterContext *ctx = outlink->src;

226

MixContext *s = ctx->priv;

227

int i;

228

char buf[64];

229

230

s->planar = av_sample_fmt_is_planar(outlink->format);

231

s->sample_rate = outlink->sample_rate;

232

outlink->time_base = (AVRational){ 1, outlink->sample_rate };

233

s->next_pts = AV_NOPTS_VALUE;

234

235

s->frame_list = av_mallocz(sizeof(*s->frame_list));

236

if (!s->frame_list)

237

return AVERROR(ENOMEM);

238

239

s->fifos = av_mallocz(s->nb_inputs * sizeof(*s->fifos));

240

if (!s->fifos)

241

return AVERROR(ENOMEM);

242

243

s->nb_channels = av_get_channel_layout_nb_channels(outlink->channel_layout);

244

for (i = 0; i < s->nb_inputs; i++) {

245

s->fifos[i] = av_audio_fifo_alloc(outlink->format, s->nb_channels, 1024);

246

if (!s->fifos[i])

247

return AVERROR(ENOMEM);

248

}

249

250

s->input_state = av_malloc(s->nb_inputs);

251

if (!s->input_state)

252

return AVERROR(ENOMEM);

253

memset(s->input_state, INPUT_ON, s->nb_inputs);

254

s->active_inputs = s->nb_inputs;

255

256

s->input_scale = av_mallocz(s->nb_inputs * sizeof(*s->input_scale));

257

if (!s->input_scale)

258

return AVERROR(ENOMEM);

259

s->scale_norm = s->active_inputs;

260

calculate_scales(s, 0);

261

262

av_get_channel_layout_string(buf, sizeof(buf), -1, outlink->channel_layout);

263

264

av_log(ctx, AV_LOG_VERBOSE,

265

"inputs:%d fmt:%s srate:%d cl:%s\n", s->nb_inputs,

266

av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf);

267

268

return 0;

269

}

270

271

/**

272

* Read samples from the input FIFOs, mix, and write to the output link.

273

274

static int output_frame(AVFilterLink *outlink, int nb_samples)

275

{

276

AVFilterContext *ctx = outlink->src;

277

MixContext *s = ctx->priv;

278

AVFilterBufferRef *out_buf, *in_buf;

279

int i;

280

281

calculate_scales(s, nb_samples);

282

283

out_buf = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);

284

if (!out_buf)

285

return AVERROR(ENOMEM);

286

287

in_buf = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);

288

if (!in_buf) {

289

avfilter_unref_buffer(out_buf);

290

return AVERROR(ENOMEM);

291

}

292

293

for (i = 0; i < s->nb_inputs; i++) {

294

if (s->input_state[i] == INPUT_ON) {

295

int planes, plane_size, p;

296

297

av_audio_fifo_read(s->fifos[i], (void **)in_buf->extended_data,

298

nb_samples);

299

300

planes = s->planar ? s->nb_channels : 1;

301

plane_size = nb_samples * (s->planar ? 1 : s->nb_channels);

302

plane_size = FFALIGN(plane_size, 16);

303

304

for (p = 0; p < planes; p++) {

305

s->fdsp.vector_fmac_scalar((float *)out_buf->extended_data[p],

306

(float *) in_buf->extended_data[p],

307

s->input_scale[i], plane_size);

308

}

309

}

310

}

311

avfilter_unref_buffer(in_buf);

312

313

out_buf->pts = s->next_pts;

314

if (s->next_pts != AV_NOPTS_VALUE)

315

s->next_pts += nb_samples;

316

317

return ff_filter_frame(outlink, out_buf);

318

}

319

320

/**

321

* Returns the smallest number of samples available in the input FIFOs other

322

* than that of the first input.

323

324

static int get_available_samples(MixContext *s)

325

{

326

int i;

327

int available_samples = INT_MAX;

328

329

av_assert0(s->nb_inputs > 1);

330

331

for (i = 1; i < s->nb_inputs; i++) {

332

int nb_samples;

333

if (s->input_state[i] == INPUT_OFF)

334

continue;

335

nb_samples = av_audio_fifo_size(s->fifos[i]);

336

available_samples = FFMIN(available_samples, nb_samples);

337

}

338

if (available_samples == INT_MAX)

339

return 0;

340

return available_samples;

341

}

342

343

/**

344

* Requests a frame, if needed, from each input link other than the first.

345

346

static int request_samples(AVFilterContext *ctx, int min_samples)

347

{

348

MixContext *s = ctx->priv;

349

int i, ret;

350

351

av_assert0(s->nb_inputs > 1);

352

353

for (i = 1; i < s->nb_inputs; i++) {

354

ret = 0;

355

if (s->input_state[i] == INPUT_OFF)

356

continue;

357

while (!ret && av_audio_fifo_size(s->fifos[i]) < min_samples)

358

ret = ff_request_frame(ctx->inputs[i]);

359

if (ret == AVERROR_EOF) {

360

if (av_audio_fifo_size(s->fifos[i]) == 0) {

361

s->input_state[i] = INPUT_OFF;

362

continue;

363

}

364

} else if (ret < 0)

365

return ret;

366

}

367

return 0;

368

}

369

370

/**

371

* Calculates the number of active inputs and determines EOF based on the

372

* duration option.

373

374

* @return 0 if mixing should continue, or AVERROR_EOF if mixing should stop.

375

376

static int calc_active_inputs(MixContext *s)

377

{

378

int i;

379

int active_inputs = 0;

380

for (i = 0; i < s->nb_inputs; i++)

381

active_inputs += !!(s->input_state[i] != INPUT_OFF);

382

s->active_inputs = active_inputs;

383

384

if (!active_inputs ||

385

(s->duration_mode == DURATION_FIRST && s->input_state[0] == INPUT_OFF) ||

386

(s->duration_mode == DURATION_SHORTEST && active_inputs != s->nb_inputs))

387

return AVERROR_EOF;

388

return 0;

389

}

390

391

static int request_frame(AVFilterLink *outlink)

392

{

393

AVFilterContext *ctx = outlink->src;

394

MixContext *s = ctx->priv;

395

int ret;

396

int wanted_samples, available_samples;

397

398

ret = calc_active_inputs(s);

399

if (ret < 0)

400

return ret;

401

402

if (s->input_state[0] == INPUT_OFF) {

403

ret = request_samples(ctx, 1);

404

if (ret < 0)

405

return ret;

406

407

ret = calc_active_inputs(s);

408

if (ret < 0)

409

return ret;

410

411

available_samples = get_available_samples(s);

412

if (!available_samples)

413

return AVERROR(EAGAIN);

414

415

return output_frame(outlink, available_samples);

416

}

417

418

if (s->frame_list->nb_frames == 0) {

419

ret = ff_request_frame(ctx->inputs[0]);

420

if (ret == AVERROR_EOF) {

421

s->input_state[0] = INPUT_OFF;

422

if (s->nb_inputs == 1)

423

return AVERROR_EOF;

424

else

425

return AVERROR(EAGAIN);

426

} else if (ret < 0)

427

return ret;

428

}

429

av_assert0(s->frame_list->nb_frames > 0);

430

431

wanted_samples = frame_list_next_frame_size(s->frame_list);

432

433

if (s->active_inputs > 1) {

434

ret = request_samples(ctx, wanted_samples);

435

if (ret < 0)

436

return ret;

437

438

ret = calc_active_inputs(s);

439

if (ret < 0)

440

return ret;

441

}

442

443

if (s->active_inputs > 1) {

444

available_samples = get_available_samples(s);

445

if (!available_samples)

446

return AVERROR(EAGAIN);

447

available_samples = FFMIN(available_samples, wanted_samples);

448

} else {

449

available_samples = wanted_samples;

450

}

451

452

s->next_pts = frame_list_next_pts(s->frame_list);

453

frame_list_remove_samples(s->frame_list, available_samples);

454

455

return output_frame(outlink, available_samples);

456

}

457

458

static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)

459

{

460

AVFilterContext *ctx = inlink->dst;

461

MixContext *s = ctx->priv;

462

AVFilterLink *outlink = ctx->outputs[0];

463

int i, ret = 0;

464

465

for (i = 0; i < ctx->nb_inputs; i++)

466

if (ctx->inputs[i] == inlink)

467

break;

468

if (i >= ctx->nb_inputs) {

469

av_log(ctx, AV_LOG_ERROR, "unknown input link\n");

470

ret = AVERROR(EINVAL);

471

goto fail;

472

}

473

474

if (i == 0) {

475

int64_t pts = av_rescale_q(buf->pts, inlink->time_base,

476

outlink->time_base);

477

ret = frame_list_add_frame(s->frame_list, buf->audio->nb_samples, pts);

478

if (ret < 0)

479

goto fail;

480

}

481

482

ret = av_audio_fifo_write(s->fifos[i], (void **)buf->extended_data,

483

buf->audio->nb_samples);

484

485

fail:

486

avfilter_unref_buffer(buf);

487

488

return ret;

489

}

490

491

static int init(AVFilterContext *ctx, const char *args)

492

{

493

MixContext *s = ctx->priv;

494

int i, ret;

495

496

s->class = &amix_class;

497

av_opt_set_defaults(s);

498

499

if ((ret = av_set_options_string(s, args, "=", ":")) < 0) {

500

av_log(ctx, AV_LOG_ERROR, "Error parsing options string '%s'.\n", args);

501

return ret;

502

}

503

av_opt_free(s);

504

505

for (i = 0; i < s->nb_inputs; i++) {

506

char name[32];

507

AVFilterPad pad = { 0 };

508

509

snprintf(name, sizeof(name), "input%d", i);

510

pad.type = AVMEDIA_TYPE_AUDIO;

511

pad.name = av_strdup(name);

512

pad.filter_frame = filter_frame;

513

514

ff_insert_inpad(ctx, i, &pad);

515

}

516

517

avpriv_float_dsp_init(&s->fdsp, 0);

518

519

return 0;

520

}

521

522

static void uninit(AVFilterContext *ctx)

523

{

524

int i;

525

MixContext *s = ctx->priv;

526

527

if (s->fifos) {

528

for (i = 0; i < s->nb_inputs; i++)

529

av_audio_fifo_free(s->fifos[i]);

530

av_freep(&s->fifos);

531

}

532

frame_list_clear(s->frame_list);

533

av_freep(&s->frame_list);

534

av_freep(&s->input_state);

535

av_freep(&s->input_scale);

536

537

for (i = 0; i < ctx->nb_inputs; i++)

538

av_freep(&ctx->input_pads[i].name);

539

}

540

541

static int query_formats(AVFilterContext *ctx)

542

{

543

AVFilterFormats *formats = NULL;

544

ff_add_format(&formats, AV_SAMPLE_FMT_FLT);

545

ff_add_format(&formats, AV_SAMPLE_FMT_FLTP);

546

ff_set_common_formats(ctx, formats);

547

ff_set_common_channel_layouts(ctx, ff_all_channel_layouts());

548

ff_set_common_samplerates(ctx, ff_all_samplerates());

549

return 0;

550

}

551

552

static const AVFilterPad avfilter_af_amix_outputs[] = {

553

{

554

.name = "default",

555

.type = AVMEDIA_TYPE_AUDIO,

556

.config_props = config_output,

557

.request_frame = request_frame

558

559

{ NULL }

560

};

561

562

AVFilter avfilter_af_amix = {

563

.name = "amix",

564

.description = NULL_IF_CONFIG_SMALL("Audio mixing."),

565

.priv_size = sizeof(MixContext),

566

567

.init = init,

568

.uninit = uninit,

569

.query_formats = query_formats,

570

571

.inputs = NULL,

572

.outputs = avfilter_af_amix_outputs,

573

};

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