gstaudioaggregator.c 63.1 KB
Newer Older
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27
/* GStreamer
 * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
 *                    2001 Thomas <thomas@apestaart.org>
 *               2005,2006 Wim Taymans <wim@fluendo.com>
 *                    2013 Sebastian Dröge <sebastian@centricular.com>
 *                    2014 Collabora
 *                             Olivier Crete <olivier.crete@collabora.com>
 *
 * gstaudioaggregator.c:
 *
 * This library is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Library General Public
 * License as published by the Free Software Foundation; either
 * version 2 of the License, or (at your option) any later version.
 *
 * This library 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
 * Library General Public License for more details.
 *
 * You should have received a copy of the GNU Library General Public
 * License along with this library; if not, write to the
 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
 * Boston, MA 02110-1301, USA.
 */
/**
 * SECTION: gstaudioaggregator
28 29 30
 * @title: GstAudioAggregator
 * @short_description: Base class that manages a set of audio input pads
 * with the purpose of aggregating or mixing their raw audio input buffers
31
 * @see_also: #GstAggregator, #GstAudioMixer
32
 *
33 34 35 36
 * Subclasses must use (a subclass of) #GstAudioAggregatorPad for both
 * their source and sink pads,
 * gst_element_class_add_static_pad_template_with_gtype() is a convenient
 * helper.
37
 *
38 39 40 41 42 43
 * #GstAudioAggregator can perform conversion on the data arriving
 * on its sink pads, based on the format expected downstream: in order
 * to enable that behaviour, the GType of the sink pads must either be
 * a (subclass of) #GstAudioAggregatorConvertPad to use the default
 * #GstAudioConverter implementation, or a subclass of #GstAudioAggregatorPad
 * implementing #GstAudioAggregatorPad.convert_buffer.
44
 *
45 46
 * To allow for the output caps to change, the mechanism is the same as
 * above, with the GType of the source pad.
47
 *
48
 * See #GstAudioMixer for an example.
49 50 51 52 53 54 55 56 57 58 59 60 61 62
 *
 * When conversion is enabled, #GstAudioAggregator will accept
 * any type of raw audio caps and perform conversion
 * on the data arriving on its sink pads, with whatever downstream
 * expects as the target format.
 *
 * In case downstream caps are not fully fixated, it will use
 * the first configured sink pad to finish fixating its source pad
 * caps.
 *
 * A notable exception for now is the sample rate, sink pads must
 * have the same sample rate as either the downstream requirement,
 * or the first configured pad, or a combination of both (when
 * downstream specifies a range or a set of acceptable rates).
63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80
 */


#ifdef HAVE_CONFIG_H
#  include "config.h"
#endif

#include "gstaudioaggregator.h"

#include <string.h>

GST_DEBUG_CATEGORY_STATIC (audio_aggregator_debug);
#define GST_CAT_DEFAULT audio_aggregator_debug

struct _GstAudioAggregatorPadPrivate
{
  /* All members are protected by the pad object lock */

81
  GstBuffer *buffer;            /* current buffer we're mixing, for
82 83
                                   comparison with a new input buffer from
                                   aggregator to see if we need to update our
84
                                   cached values. */
85 86 87

  guint position, size;         /* position in the input buffer and size of the
                                   input buffer in number of samples */
88

89 90
  GstBuffer *input_buffer;

91
  guint64 output_offset;        /* Sample offset in output segment relative to
92 93
                                   pad.segment.start that position refers to
                                   in the current buffer. */
94

95 96
  guint64 next_offset;          /* Next expected sample offset relative to
                                   pad.segment.start */
97 98 99 100 101 102 103 104 105 106 107 108 109 110 111

  /* Last time we noticed a discont */
  GstClockTime discont_time;

  /* A new unhandled segment event has been received */
  gboolean new_segment;
};


/*****************************************
 * GstAudioAggregatorPad implementation  *
 *****************************************/
G_DEFINE_TYPE (GstAudioAggregatorPad, gst_audio_aggregator_pad,
    GST_TYPE_AGGREGATOR_PAD);

112 113 114 115 116 117
enum
{
  PROP_PAD_0,
  PROP_PAD_CONVERTER_CONFIG,
};

118
static GstFlowReturn
119 120 121
gst_audio_aggregator_pad_flush_pad (GstAggregatorPad * aggpad,
    GstAggregator * aggregator);

122 123 124 125 126 127
static void
gst_audio_aggregator_pad_finalize (GObject * object)
{
  GstAudioAggregatorPad *pad = (GstAudioAggregatorPad *) object;

  gst_buffer_replace (&pad->priv->buffer, NULL);
128
  gst_buffer_replace (&pad->priv->input_buffer, NULL);
129 130 131 132

  G_OBJECT_CLASS (gst_audio_aggregator_pad_parent_class)->finalize (object);
}

133 134 135
static void
gst_audio_aggregator_pad_class_init (GstAudioAggregatorPadClass * klass)
{
136
  GObjectClass *gobject_class = (GObjectClass *) klass;
137 138 139 140
  GstAggregatorPadClass *aggpadclass = (GstAggregatorPadClass *) klass;

  g_type_class_add_private (klass, sizeof (GstAudioAggregatorPadPrivate));

141
  gobject_class->finalize = gst_audio_aggregator_pad_finalize;
142 143 144 145 146 147 148 149 150 151 152 153 154
  aggpadclass->flush = GST_DEBUG_FUNCPTR (gst_audio_aggregator_pad_flush_pad);
}

static void
gst_audio_aggregator_pad_init (GstAudioAggregatorPad * pad)
{
  pad->priv =
      G_TYPE_INSTANCE_GET_PRIVATE (pad, GST_TYPE_AUDIO_AGGREGATOR_PAD,
      GstAudioAggregatorPadPrivate);

  gst_audio_info_init (&pad->info);

  pad->priv->buffer = NULL;
155
  pad->priv->input_buffer = NULL;
156 157 158 159 160 161 162 163
  pad->priv->position = 0;
  pad->priv->size = 0;
  pad->priv->output_offset = -1;
  pad->priv->next_offset = -1;
  pad->priv->discont_time = GST_CLOCK_TIME_NONE;
}


164
static GstFlowReturn
165 166 167 168 169 170 171 172 173 174
gst_audio_aggregator_pad_flush_pad (GstAggregatorPad * aggpad,
    GstAggregator * aggregator)
{
  GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (aggpad);

  GST_OBJECT_LOCK (aggpad);
  pad->priv->position = pad->priv->size = 0;
  pad->priv->output_offset = pad->priv->next_offset = -1;
  pad->priv->discont_time = GST_CLOCK_TIME_NONE;
  gst_buffer_replace (&pad->priv->buffer, NULL);
175
  gst_buffer_replace (&pad->priv->input_buffer, NULL);
176 177
  GST_OBJECT_UNLOCK (aggpad);

178
  return GST_FLOW_OK;
179 180
}

181 182 183 184 185 186 187 188 189 190 191 192 193 194 195 196 197 198 199 200 201 202 203 204 205 206 207 208 209 210 211 212 213 214 215 216 217 218 219 220 221 222 223
struct _GstAudioAggregatorConvertPadPrivate
{
  /* All members are protected by the pad object lock */
  GstAudioConverter *converter;
  GstStructure *converter_config;
  gboolean converter_config_changed;
};


G_DEFINE_TYPE (GstAudioAggregatorConvertPad, gst_audio_aggregator_convert_pad,
    GST_TYPE_AUDIO_AGGREGATOR_PAD);

static void
gst_audio_aggregator_convert_pad_update_converter (GstAudioAggregatorConvertPad
    * aaggcpad, GstAudioInfo * in_info, GstAudioInfo * out_info)
{
  if (!aaggcpad->priv->converter_config_changed)
    return;

  if (aaggcpad->priv->converter) {
    gst_audio_converter_free (aaggcpad->priv->converter);
    aaggcpad->priv->converter = NULL;
  }

  if (gst_audio_info_is_equal (in_info, out_info) ||
      in_info->finfo->format == GST_AUDIO_FORMAT_UNKNOWN) {
    if (aaggcpad->priv->converter) {
      gst_audio_converter_free (aaggcpad->priv->converter);
      aaggcpad->priv->converter = NULL;
    }
  } else {
    /* If we haven't received caps yet, this pad should not have
     * a buffer to convert anyway */
    aaggcpad->priv->converter =
        gst_audio_converter_new (GST_AUDIO_CONVERTER_FLAG_NONE,
        in_info, out_info,
        aaggcpad->priv->converter_config ? gst_structure_copy (aaggcpad->
            priv->converter_config) : NULL);
  }

  aaggcpad->priv->converter_config_changed = FALSE;
}

224 225 226 227 228 229 230 231
static void
gst_audio_aggregator_pad_update_conversion_info (GstAudioAggregatorPad *
    aaggpad)
{
  GST_AUDIO_AGGREGATOR_CONVERT_PAD (aaggpad)->priv->converter_config_changed =
      TRUE;
}

232
static GstBuffer *
233 234
gst_audio_aggregator_convert_pad_convert_buffer (GstAudioAggregatorPad *
    aaggpad, GstAudioInfo * in_info, GstAudioInfo * out_info,
235 236 237
    GstBuffer * input_buffer)
{
  GstBuffer *res;
238 239
  GstAudioAggregatorConvertPad *aaggcpad =
      GST_AUDIO_AGGREGATOR_CONVERT_PAD (aaggpad);
240 241 242 243 244 245 246 247 248 249 250 251 252 253 254 255 256 257 258 259 260 261 262 263 264 265 266 267 268 269 270 271 272 273 274 275 276 277 278 279 280 281 282 283 284 285 286 287 288 289 290 291 292 293 294 295 296 297 298 299 300 301 302 303 304 305 306 307 308 309 310 311 312 313 314 315 316

  gst_audio_aggregator_convert_pad_update_converter (aaggcpad, in_info,
      out_info);

  if (aaggcpad->priv->converter) {
    gint insize = gst_buffer_get_size (input_buffer);
    gsize insamples = insize / in_info->bpf;
    gsize outsamples =
        gst_audio_converter_get_out_frames (aaggcpad->priv->converter,
        insamples);
    gint outsize = outsamples * out_info->bpf;
    GstMapInfo inmap, outmap;

    res = gst_buffer_new_allocate (NULL, outsize, NULL);

    /* We create a perfectly similar buffer, except obviously for
     * its converted contents */
    gst_buffer_copy_into (res, input_buffer,
        GST_BUFFER_COPY_FLAGS | GST_BUFFER_COPY_TIMESTAMPS |
        GST_BUFFER_COPY_META, 0, -1);

    gst_buffer_map (input_buffer, &inmap, GST_MAP_READ);
    gst_buffer_map (res, &outmap, GST_MAP_WRITE);

    gst_audio_converter_samples (aaggcpad->priv->converter,
        GST_AUDIO_CONVERTER_FLAG_NONE,
        (gpointer *) & inmap.data, insamples,
        (gpointer *) & outmap.data, outsamples);

    gst_buffer_unmap (input_buffer, &inmap);
    gst_buffer_unmap (res, &outmap);
  } else {
    res = gst_buffer_ref (input_buffer);
  }

  return res;
}

static void
gst_audio_aggregator_convert_pad_finalize (GObject * object)
{
  GstAudioAggregatorConvertPad *pad = (GstAudioAggregatorConvertPad *) object;

  if (pad->priv->converter)
    gst_audio_converter_free (pad->priv->converter);

  if (pad->priv->converter_config)
    gst_structure_free (pad->priv->converter_config);

  G_OBJECT_CLASS (gst_audio_aggregator_convert_pad_parent_class)->finalize
      (object);
}

static void
gst_audio_aggregator_convert_pad_get_property (GObject * object, guint prop_id,
    GValue * value, GParamSpec * pspec)
{
  GstAudioAggregatorConvertPad *pad = GST_AUDIO_AGGREGATOR_CONVERT_PAD (object);

  switch (prop_id) {
    case PROP_PAD_CONVERTER_CONFIG:
      GST_OBJECT_LOCK (pad);
      if (pad->priv->converter_config)
        g_value_set_boxed (value, pad->priv->converter_config);
      GST_OBJECT_UNLOCK (pad);
      break;
    default:
      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
      break;
  }
}

static void
gst_audio_aggregator_convert_pad_set_property (GObject * object, guint prop_id,
    const GValue * value, GParamSpec * pspec)
{
  GstAudioAggregatorConvertPad *pad = GST_AUDIO_AGGREGATOR_CONVERT_PAD (object);
317

318 319 320 321 322 323 324 325 326 327 328 329 330 331 332 333 334 335 336 337
  switch (prop_id) {
    case PROP_PAD_CONVERTER_CONFIG:
      GST_OBJECT_LOCK (pad);
      if (pad->priv->converter_config)
        gst_structure_free (pad->priv->converter_config);
      pad->priv->converter_config = g_value_dup_boxed (value);
      pad->priv->converter_config_changed = TRUE;
      GST_OBJECT_UNLOCK (pad);
      break;
    default:
      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
      break;
  }
}

static void
gst_audio_aggregator_convert_pad_class_init (GstAudioAggregatorConvertPadClass *
    klass)
{
  GObjectClass *gobject_class = (GObjectClass *) klass;
338 339
  GstAudioAggregatorPadClass *aaggpad_class =
      (GstAudioAggregatorPadClass *) klass;
340 341 342 343 344 345 346 347 348 349 350 351
  g_type_class_add_private (klass,
      sizeof (GstAudioAggregatorConvertPadPrivate));

  gobject_class->set_property = gst_audio_aggregator_convert_pad_set_property;
  gobject_class->get_property = gst_audio_aggregator_convert_pad_get_property;

  g_object_class_install_property (gobject_class, PROP_PAD_CONVERTER_CONFIG,
      g_param_spec_boxed ("converter-config", "Converter configuration",
          "A GstStructure describing the configuration that should be used "
          "when converting this pad's audio buffers",
          GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

352 353 354 355 356 357
  aaggpad_class->convert_buffer =
      gst_audio_aggregator_convert_pad_convert_buffer;

  aaggpad_class->update_conversion_info =
      gst_audio_aggregator_pad_update_conversion_info;

358 359 360 361 362 363 364 365 366 367
  gobject_class->finalize = gst_audio_aggregator_convert_pad_finalize;
}

static void
gst_audio_aggregator_convert_pad_init (GstAudioAggregatorConvertPad * pad)
{
  pad->priv =
      G_TYPE_INSTANCE_GET_PRIVATE (pad, GST_TYPE_AUDIO_AGGREGATOR_CONVERT_PAD,
      GstAudioAggregatorConvertPadPrivate);
}
368 369 370 371 372 373 374 375 376 377 378 379 380 381 382 383

/**************************************
 * GstAudioAggregator implementation  *
 **************************************/

struct _GstAudioAggregatorPrivate
{
  GMutex mutex;

  /* All three properties are unprotected, can't be modified while streaming */
  /* Size in frames that is output per buffer */
  GstClockTime output_buffer_duration;
  GstClockTime alignment_threshold;
  GstClockTime discont_wait;

  /* Protected by srcpad stream clock */
384 385
  /* Output buffer starting at offset containing blocksize frames (calculated
   * from output_buffer_duration) */
386 387 388 389
  GstBuffer *current_buffer;

  /* counters to keep track of timestamps */
  /* Readable with object lock, writable with both aag lock and object lock */
390

391 392
  /* Sample offset starting from 0 at aggregator.segment.start */
  gint64 offset;
393 394 395 396 397 398 399 400 401 402 403 404 405 406 407 408 409
};

#define GST_AUDIO_AGGREGATOR_LOCK(self)   g_mutex_lock (&(self)->priv->mutex);
#define GST_AUDIO_AGGREGATOR_UNLOCK(self) g_mutex_unlock (&(self)->priv->mutex);

static void gst_audio_aggregator_set_property (GObject * object, guint prop_id,
    const GValue * value, GParamSpec * pspec);
static void gst_audio_aggregator_get_property (GObject * object, guint prop_id,
    GValue * value, GParamSpec * pspec);
static void gst_audio_aggregator_dispose (GObject * object);

static gboolean gst_audio_aggregator_src_event (GstAggregator * agg,
    GstEvent * event);
static gboolean gst_audio_aggregator_sink_event (GstAggregator * agg,
    GstAggregatorPad * aggpad, GstEvent * event);
static gboolean gst_audio_aggregator_src_query (GstAggregator * agg,
    GstQuery * query);
410 411 412
static gboolean
gst_audio_aggregator_sink_query (GstAggregator * agg, GstAggregatorPad * aggpad,
    GstQuery * query);
413 414 415 416 417 418
static gboolean gst_audio_aggregator_start (GstAggregator * agg);
static gboolean gst_audio_aggregator_stop (GstAggregator * agg);
static GstFlowReturn gst_audio_aggregator_flush (GstAggregator * agg);

static GstBuffer *gst_audio_aggregator_create_output_buffer (GstAudioAggregator
    * aagg, guint num_frames);
419 420
static GstBuffer *gst_audio_aggregator_do_clip (GstAggregator * agg,
    GstAggregatorPad * bpad, GstBuffer * buffer);
421 422
static GstFlowReturn gst_audio_aggregator_aggregate (GstAggregator * agg,
    gboolean timeout);
423
static gboolean sync_pad_values (GstElement * aagg, GstPad * pad, gpointer ud);
424 425
static gboolean gst_audio_aggregator_negotiated_src_caps (GstAggregator * agg,
    GstCaps * caps);
426 427 428 429 430
static GstFlowReturn
gst_audio_aggregator_update_src_caps (GstAggregator * agg,
    GstCaps * caps, GstCaps ** ret);
static GstCaps *gst_audio_aggregator_fixate_src_caps (GstAggregator * agg,
    GstCaps * caps);
431 432 433 434 435 436 437 438 439 440 441 442 443 444 445 446 447 448 449 450

#define DEFAULT_OUTPUT_BUFFER_DURATION (10 * GST_MSECOND)
#define DEFAULT_ALIGNMENT_THRESHOLD   (40 * GST_MSECOND)
#define DEFAULT_DISCONT_WAIT (1 * GST_SECOND)

enum
{
  PROP_0,
  PROP_OUTPUT_BUFFER_DURATION,
  PROP_ALIGNMENT_THRESHOLD,
  PROP_DISCONT_WAIT,
};

G_DEFINE_ABSTRACT_TYPE (GstAudioAggregator, gst_audio_aggregator,
    GST_TYPE_AGGREGATOR);

static GstClockTime
gst_audio_aggregator_get_next_time (GstAggregator * agg)
{
  GstClockTime next_time;
451
  GstSegment *segment = &GST_AGGREGATOR_PAD (agg->srcpad)->segment;
452 453

  GST_OBJECT_LOCK (agg);
454 455
  if (segment->position == -1 || segment->position < segment->start)
    next_time = segment->start;
456
  else
457
    next_time = segment->position;
458

459 460
  if (segment->stop != -1 && next_time > segment->stop)
    next_time = segment->stop;
461

462
  next_time = gst_segment_to_running_time (segment, GST_FORMAT_TIME, next_time);
463 464 465 466 467
  GST_OBJECT_UNLOCK (agg);

  return next_time;
}

468 469 470 471
static GstBuffer *
gst_audio_aggregator_convert_buffer (GstAudioAggregator * aagg, GstPad * pad,
    GstAudioInfo * in_info, GstAudioInfo * out_info, GstBuffer * buffer)
{
472 473
  GstAudioAggregatorPadClass *klass = GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (pad);
  GstAudioAggregatorPad *aaggpad = GST_AUDIO_AGGREGATOR_PAD (pad);
474 475 476

  g_assert (klass->convert_buffer);

477
  return klass->convert_buffer (aaggpad, in_info, out_info, buffer);
478 479
}

480 481 482 483 484 485 486 487 488 489 490 491 492 493 494 495 496 497
static void
gst_audio_aggregator_class_init (GstAudioAggregatorClass * klass)
{
  GObjectClass *gobject_class = (GObjectClass *) klass;
  GstAggregatorClass *gstaggregator_class = (GstAggregatorClass *) klass;

  g_type_class_add_private (klass, sizeof (GstAudioAggregatorPrivate));

  gobject_class->set_property = gst_audio_aggregator_set_property;
  gobject_class->get_property = gst_audio_aggregator_get_property;
  gobject_class->dispose = gst_audio_aggregator_dispose;

  gstaggregator_class->src_event =
      GST_DEBUG_FUNCPTR (gst_audio_aggregator_src_event);
  gstaggregator_class->sink_event =
      GST_DEBUG_FUNCPTR (gst_audio_aggregator_sink_event);
  gstaggregator_class->src_query =
      GST_DEBUG_FUNCPTR (gst_audio_aggregator_src_query);
498
  gstaggregator_class->sink_query = gst_audio_aggregator_sink_query;
499 500 501 502 503 504 505
  gstaggregator_class->start = gst_audio_aggregator_start;
  gstaggregator_class->stop = gst_audio_aggregator_stop;
  gstaggregator_class->flush = gst_audio_aggregator_flush;
  gstaggregator_class->aggregate =
      GST_DEBUG_FUNCPTR (gst_audio_aggregator_aggregate);
  gstaggregator_class->clip = GST_DEBUG_FUNCPTR (gst_audio_aggregator_do_clip);
  gstaggregator_class->get_next_time = gst_audio_aggregator_get_next_time;
506 507 508
  gstaggregator_class->update_src_caps =
      GST_DEBUG_FUNCPTR (gst_audio_aggregator_update_src_caps);
  gstaggregator_class->fixate_src_caps = gst_audio_aggregator_fixate_src_caps;
509 510
  gstaggregator_class->negotiated_src_caps =
      gst_audio_aggregator_negotiated_src_caps;
511 512 513 514 515 516 517 518 519 520 521 522 523 524 525 526 527 528 529 530 531 532 533 534 535 536 537 538 539 540 541 542 543 544 545 546 547 548 549 550 551 552 553 554 555 556 557 558 559 560 561 562 563 564 565 566 567 568 569 570 571 572 573 574 575 576 577 578 579 580 581 582 583 584 585 586 587 588 589 590 591 592 593 594 595 596 597 598 599 600 601 602 603 604 605 606 607 608 609 610 611 612 613 614

  klass->create_output_buffer = gst_audio_aggregator_create_output_buffer;

  GST_DEBUG_CATEGORY_INIT (audio_aggregator_debug, "audioaggregator",
      GST_DEBUG_FG_MAGENTA, "GstAudioAggregator");

  g_object_class_install_property (gobject_class, PROP_OUTPUT_BUFFER_DURATION,
      g_param_spec_uint64 ("output-buffer-duration", "Output Buffer Duration",
          "Output block size in nanoseconds", 1,
          G_MAXUINT64, DEFAULT_OUTPUT_BUFFER_DURATION,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (gobject_class, PROP_ALIGNMENT_THRESHOLD,
      g_param_spec_uint64 ("alignment-threshold", "Alignment Threshold",
          "Timestamp alignment threshold in nanoseconds", 0,
          G_MAXUINT64 - 1, DEFAULT_ALIGNMENT_THRESHOLD,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (gobject_class, PROP_DISCONT_WAIT,
      g_param_spec_uint64 ("discont-wait", "Discont Wait",
          "Window of time in nanoseconds to wait before "
          "creating a discontinuity", 0,
          G_MAXUINT64 - 1, DEFAULT_DISCONT_WAIT,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
}

static void
gst_audio_aggregator_init (GstAudioAggregator * aagg)
{
  aagg->priv =
      G_TYPE_INSTANCE_GET_PRIVATE (aagg, GST_TYPE_AUDIO_AGGREGATOR,
      GstAudioAggregatorPrivate);

  g_mutex_init (&aagg->priv->mutex);

  aagg->priv->output_buffer_duration = DEFAULT_OUTPUT_BUFFER_DURATION;
  aagg->priv->alignment_threshold = DEFAULT_ALIGNMENT_THRESHOLD;
  aagg->priv->discont_wait = DEFAULT_DISCONT_WAIT;

  aagg->current_caps = NULL;

  gst_aggregator_set_latency (GST_AGGREGATOR (aagg),
      aagg->priv->output_buffer_duration, aagg->priv->output_buffer_duration);
}

static void
gst_audio_aggregator_dispose (GObject * object)
{
  GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (object);

  gst_caps_replace (&aagg->current_caps, NULL);

  g_mutex_clear (&aagg->priv->mutex);

  G_OBJECT_CLASS (gst_audio_aggregator_parent_class)->dispose (object);
}

static void
gst_audio_aggregator_set_property (GObject * object, guint prop_id,
    const GValue * value, GParamSpec * pspec)
{
  GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (object);

  switch (prop_id) {
    case PROP_OUTPUT_BUFFER_DURATION:
      aagg->priv->output_buffer_duration = g_value_get_uint64 (value);
      gst_aggregator_set_latency (GST_AGGREGATOR (aagg),
          aagg->priv->output_buffer_duration,
          aagg->priv->output_buffer_duration);
      break;
    case PROP_ALIGNMENT_THRESHOLD:
      aagg->priv->alignment_threshold = g_value_get_uint64 (value);
      break;
    case PROP_DISCONT_WAIT:
      aagg->priv->discont_wait = g_value_get_uint64 (value);
      break;
    default:
      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
      break;
  }
}

static void
gst_audio_aggregator_get_property (GObject * object, guint prop_id,
    GValue * value, GParamSpec * pspec)
{
  GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (object);

  switch (prop_id) {
    case PROP_OUTPUT_BUFFER_DURATION:
      g_value_set_uint64 (value, aagg->priv->output_buffer_duration);
      break;
    case PROP_ALIGNMENT_THRESHOLD:
      g_value_set_uint64 (value, aagg->priv->alignment_threshold);
      break;
    case PROP_DISCONT_WAIT:
      g_value_set_uint64 (value, aagg->priv->discont_wait);
      break;
    default:
      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
      break;
  }
}

615 616 617 618 619 620 621 622 623 624 625 626 627 628 629 630 631 632 633 634 635 636 637 638 639 640 641 642 643 644 645 646 647 648 649 650 651 652
/* Caps negotiation */

/* Unref after usage */
static GstAudioAggregatorPad *
gst_audio_aggregator_get_first_configured_pad (GstAggregator * agg)
{
  GstAudioAggregatorPad *res = NULL;
  GList *l;

  GST_OBJECT_LOCK (agg);
  for (l = GST_ELEMENT (agg)->sinkpads; l; l = l->next) {
    GstAudioAggregatorPad *aaggpad = l->data;

    if (GST_AUDIO_INFO_FORMAT (&aaggpad->info) != GST_AUDIO_FORMAT_UNKNOWN) {
      res = gst_object_ref (aaggpad);
      break;
    }
  }
  GST_OBJECT_UNLOCK (agg);

  return res;
}

static GstCaps *
gst_audio_aggregator_sink_getcaps (GstPad * pad, GstAggregator * agg,
    GstCaps * filter)
{
  GstAudioAggregatorPad *first_configured_pad =
      gst_audio_aggregator_get_first_configured_pad (agg);
  GstCaps *sink_template_caps = gst_pad_get_pad_template_caps (pad);
  GstCaps *downstream_caps = gst_pad_get_allowed_caps (agg->srcpad);
  GstCaps *sink_caps;
  GstStructure *s, *s2;
  gint downstream_rate;

  sink_template_caps = gst_caps_make_writable (sink_template_caps);
  s = gst_caps_get_structure (sink_template_caps, 0);

653 654 655 656 657 658 659 660 661 662 663 664
  /* We will then use the rate in the first structure as the expected
   * rate, we want to make sure only the compatible structures remain
   * in downstream_caps
   */
  if (downstream_caps && filter) {
    GstCaps *tmp = gst_caps_intersect_full (downstream_caps, filter,
        GST_CAPS_INTERSECT_FIRST);

    gst_caps_unref (downstream_caps);
    downstream_caps = tmp;
  }

665 666 667 668 669 670 671 672 673 674 675 676 677 678 679 680 681 682 683 684 685 686 687 688 689 690 691 692 693 694 695 696 697 698 699 700 701 702 703 704 705 706 707 708 709 710 711 712 713
  if (downstream_caps && !gst_caps_is_empty (downstream_caps))
    s2 = gst_caps_get_structure (downstream_caps, 0);
  else
    s2 = NULL;

  if (s2 && gst_structure_get_int (s2, "rate", &downstream_rate)) {
    gst_structure_fixate_field_nearest_int (s, "rate", downstream_rate);
  } else if (first_configured_pad) {
    gst_structure_fixate_field_nearest_int (s, "rate",
        first_configured_pad->info.rate);
  }

  if (first_configured_pad)
    gst_object_unref (first_configured_pad);

  sink_caps = filter ? gst_caps_intersect (sink_template_caps,
      filter) : gst_caps_ref (sink_template_caps);

  GST_INFO_OBJECT (pad, "Getting caps with filter %" GST_PTR_FORMAT, filter);
  GST_DEBUG_OBJECT (pad, "sink template caps : %" GST_PTR_FORMAT,
      sink_template_caps);
  GST_DEBUG_OBJECT (pad, "downstream caps %" GST_PTR_FORMAT, downstream_caps);
  GST_INFO_OBJECT (pad, "returned sink caps : %" GST_PTR_FORMAT, sink_caps);

  gst_caps_unref (sink_template_caps);

  if (downstream_caps)
    gst_caps_unref (downstream_caps);

  return sink_caps;
}

static gboolean
gst_audio_aggregator_sink_setcaps (GstAudioAggregatorPad * aaggpad,
    GstAggregator * agg, GstCaps * caps)
{
  GstAudioAggregatorPad *first_configured_pad =
      gst_audio_aggregator_get_first_configured_pad (agg);
  GstCaps *downstream_caps = gst_pad_get_allowed_caps (agg->srcpad);
  GstAudioInfo info;
  gboolean ret = TRUE;
  gint downstream_rate;
  GstStructure *s;

  if (!downstream_caps || gst_caps_is_empty (downstream_caps)) {
    ret = FALSE;
    goto done;
  }

714 715 716 717
  if (!gst_audio_info_from_caps (&info, caps)) {
    GST_WARNING_OBJECT (agg, "Rejecting invalid caps: %" GST_PTR_FORMAT, caps);
    return FALSE;
  }
718 719 720 721 722 723 724 725 726 727 728 729
  s = gst_caps_get_structure (downstream_caps, 0);

  /* TODO: handle different rates on sinkpads, a bit complex
   * because offsets will have to be updated, and audio resampling
   * has a latency to take into account
   */
  if ((gst_structure_get_int (s, "rate", &downstream_rate)
          && info.rate != downstream_rate) || (first_configured_pad
          && info.rate != first_configured_pad->info.rate)) {
    gst_pad_push_event (GST_PAD (aaggpad), gst_event_new_reconfigure ());
    ret = FALSE;
  } else {
730 731
    GstAudioAggregatorPadClass *klass =
        GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (aaggpad);
732
    GST_OBJECT_LOCK (aaggpad);
733
    aaggpad->info = info;
734 735
    if (klass->update_conversion_info)
      klass->update_conversion_info (aaggpad);
736 737 738 739
    GST_OBJECT_UNLOCK (aaggpad);
  }

done:
740 741 742
  if (first_configured_pad)
    gst_object_unref (first_configured_pad);

743 744 745 746 747 748 749 750 751 752 753 754 755 756 757 758 759 760 761 762 763 764 765 766 767 768 769 770 771 772 773 774 775 776
  if (downstream_caps)
    gst_caps_unref (downstream_caps);

  return ret;
}

static GstFlowReturn
gst_audio_aggregator_update_src_caps (GstAggregator * agg,
    GstCaps * caps, GstCaps ** ret)
{
  GstCaps *src_template_caps = gst_pad_get_pad_template_caps (agg->srcpad);
  GstCaps *downstream_caps =
      gst_pad_peer_query_caps (agg->srcpad, src_template_caps);

  gst_caps_unref (src_template_caps);

  *ret = gst_caps_intersect (caps, downstream_caps);

  GST_INFO ("Updated src caps to %" GST_PTR_FORMAT, *ret);

  if (downstream_caps)
    gst_caps_unref (downstream_caps);

  return GST_FLOW_OK;
}

/* At that point if the caps are not fixed, this means downstream
 * didn't have fully specified requirements, we'll just go ahead
 * and fixate raw audio fields using our first configured pad, we don't for
 * now need a more complicated heuristic
 */
static GstCaps *
gst_audio_aggregator_fixate_src_caps (GstAggregator * agg, GstCaps * caps)
{
777
  GstAudioAggregatorPad *first_configured_pad = NULL;
778

779 780
  if (GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (agg->srcpad)->convert_buffer)
    first_configured_pad = gst_audio_aggregator_get_first_configured_pad (agg);
781

782 783
  caps = gst_caps_make_writable (caps);

784 785 786 787 788
  if (first_configured_pad) {
    GstStructure *s, *s2;
    GstCaps *first_configured_caps =
        gst_audio_info_to_caps (&first_configured_pad->info);
    gint first_configured_rate, first_configured_channels;
789
    gint channels;
790 791 792 793 794 795 796 797 798 799 800 801 802 803 804

    s = gst_caps_get_structure (caps, 0);
    s2 = gst_caps_get_structure (first_configured_caps, 0);

    gst_structure_get_int (s2, "rate", &first_configured_rate);
    gst_structure_get_int (s2, "channels", &first_configured_channels);

    gst_structure_fixate_field_string (s, "format",
        gst_structure_get_string (s2, "format"));
    gst_structure_fixate_field_string (s, "layout",
        gst_structure_get_string (s2, "layout"));
    gst_structure_fixate_field_nearest_int (s, "rate", first_configured_rate);
    gst_structure_fixate_field_nearest_int (s, "channels",
        first_configured_channels);

805 806 807 808 809 810 811 812 813 814 815 816
    gst_structure_get_int (s, "channels", &channels);

    if (!gst_structure_has_field (s, "channel-mask") && channels > 2) {
      guint64 mask;

      if (!gst_structure_get (s2, "channel-mask", GST_TYPE_BITMASK, &mask,
              NULL)) {
        mask = gst_audio_channel_get_fallback_mask (channels);
      }
      gst_structure_set (s, "channel-mask", GST_TYPE_BITMASK, mask, NULL);
    }

817 818
    gst_caps_unref (first_configured_caps);
    gst_object_unref (first_configured_pad);
819 820 821 822 823 824 825 826 827 828 829 830 831 832 833
  } else {
    GstStructure *s;
    gint channels;

    s = gst_caps_get_structure (caps, 0);

    gst_structure_fixate_field_nearest_int (s, "rate", GST_AUDIO_DEF_RATE);
    gst_structure_fixate_field_string (s, "format", GST_AUDIO_NE ("S16"));
    gst_structure_fixate_field_string (s, "layout", "interleaved");
    gst_structure_fixate_field_nearest_int (s, "channels", 2);

    if (gst_structure_get_int (s, "channels", &channels) && channels > 2) {
      if (!gst_structure_has_field_typed (s, "channel-mask", GST_TYPE_BITMASK))
        gst_structure_set (s, "channel-mask", GST_TYPE_BITMASK, 0ULL, NULL);
    }
834 835 836 837 838 839 840 841 842 843 844 845 846 847 848 849 850 851 852
  }

  if (!gst_caps_is_fixed (caps))
    caps = gst_caps_fixate (caps);

  GST_INFO_OBJECT (agg, "Fixated src caps to %" GST_PTR_FORMAT, caps);

  return caps;
}

/* Must be called with OBJECT_LOCK taken */
static void
gst_audio_aggregator_update_converters (GstAudioAggregator * aagg,
    GstAudioInfo * new_info)
{
  GList *l;

  for (l = GST_ELEMENT (aagg)->sinkpads; l; l = l->next) {
    GstAudioAggregatorPad *aaggpad = l->data;
853 854
    GstAudioAggregatorPadClass *klass =
        GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (aaggpad);
855

856 857
    if (klass->update_conversion_info)
      klass->update_conversion_info (aaggpad);
858 859 860 861 862 863 864 865

    /* If we currently were mixing a buffer, we need to convert it to the new
     * format */
    if (aaggpad->priv->buffer) {
      GstBuffer *new_converted_buffer =
          gst_audio_aggregator_convert_buffer (aagg, GST_PAD (aaggpad),
          &aaggpad->info, new_info, aaggpad->priv->input_buffer);
      gst_buffer_replace (&aaggpad->priv->buffer, new_converted_buffer);
866
      gst_buffer_unref (new_converted_buffer);
867 868 869 870 871 872 873 874 875 876
    }
  }
}

/* We now have our final output caps, we can create the required converters */
static gboolean
gst_audio_aggregator_negotiated_src_caps (GstAggregator * agg, GstCaps * caps)
{
  GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (agg);
  GstAudioInfo info;
877
  GstAudioAggregatorPad *srcpad = GST_AUDIO_AGGREGATOR_PAD (agg->srcpad);
878 879 880 881 882 883 884 885 886 887 888

  GST_INFO_OBJECT (agg, "src caps negotiated %" GST_PTR_FORMAT, caps);

  if (!gst_audio_info_from_caps (&info, caps)) {
    GST_WARNING_OBJECT (aagg, "Rejecting invalid caps: %" GST_PTR_FORMAT, caps);
    return FALSE;
  }

  GST_AUDIO_AGGREGATOR_LOCK (aagg);
  GST_OBJECT_LOCK (aagg);

889
  if (GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (agg->srcpad)->convert_buffer) {
890 891 892
    gst_audio_aggregator_update_converters (aagg, &info);

    if (aagg->priv->current_buffer
893
        && !gst_audio_info_is_equal (&srcpad->info, &info)) {
894 895 896 897 898 899 900 901
      GstBuffer *converted;
      GstAudioAggregatorPadClass *klass =
          GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (agg->srcpad);

      if (klass->update_conversion_info)
        klass->update_conversion_info (GST_AUDIO_AGGREGATOR_PAD (agg->srcpad));

      converted =
902
          gst_audio_aggregator_convert_buffer (aagg, agg->srcpad, &srcpad->info,
903 904 905 906 907 908
          &info, aagg->priv->current_buffer);
      gst_buffer_unref (aagg->priv->current_buffer);
      aagg->priv->current_buffer = converted;
    }
  }

909
  if (!gst_audio_info_is_equal (&info, &srcpad->info)) {
910 911 912
    GST_INFO_OBJECT (aagg, "setting caps to %" GST_PTR_FORMAT, caps);
    gst_caps_replace (&aagg->current_caps, caps);

913
    memcpy (&srcpad->info, &info, sizeof (info));
914 915 916 917 918 919 920 921 922
  }

  GST_OBJECT_UNLOCK (aagg);
  GST_AUDIO_AGGREGATOR_UNLOCK (aagg);

  return
      GST_AGGREGATOR_CLASS
      (gst_audio_aggregator_parent_class)->negotiated_src_caps (agg, caps);
}
923 924 925 926 927 928 929 930 931 932 933 934 935 936 937 938 939 940 941 942 943 944 945 946 947 948 949 950 951 952 953 954 955 956

/* event handling */

static gboolean
gst_audio_aggregator_src_event (GstAggregator * agg, GstEvent * event)
{
  gboolean result;

  GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (agg);
  GST_DEBUG_OBJECT (agg->srcpad, "Got %s event on src pad",
      GST_EVENT_TYPE_NAME (event));

  switch (GST_EVENT_TYPE (event)) {
    case GST_EVENT_QOS:
      /* QoS might be tricky */
      gst_event_unref (event);
      return FALSE;
    case GST_EVENT_NAVIGATION:
      /* navigation is rather pointless. */
      gst_event_unref (event);
      return FALSE;
      break;
    case GST_EVENT_SEEK:
    {
      GstSeekFlags flags;
      gdouble rate;
      GstSeekType start_type, stop_type;
      gint64 start, stop;
      GstFormat seek_format, dest_format;

      /* parse the seek parameters */
      gst_event_parse_seek (event, &rate, &seek_format, &flags, &start_type,
          &start, &stop_type, &stop);

Stefan Sauer's avatar
Stefan Sauer committed
957
      /* Check the seeking parameters before linking up */
958 959 960 961 962 963 964 965 966 967 968 969 970 971 972
      if ((start_type != GST_SEEK_TYPE_NONE)
          && (start_type != GST_SEEK_TYPE_SET)) {
        result = FALSE;
        GST_DEBUG_OBJECT (aagg,
            "seeking failed, unhandled seek type for start: %d", start_type);
        goto done;
      }
      if ((stop_type != GST_SEEK_TYPE_NONE) && (stop_type != GST_SEEK_TYPE_SET)) {
        result = FALSE;
        GST_DEBUG_OBJECT (aagg,
            "seeking failed, unhandled seek type for end: %d", stop_type);
        goto done;
      }

      GST_OBJECT_LOCK (agg);
973
      dest_format = GST_AGGREGATOR_PAD (agg->srcpad)->segment.format;
974 975 976 977 978 979 980 981 982 983 984 985 986 987 988 989 990 991 992 993 994 995 996 997 998 999 1000
      GST_OBJECT_UNLOCK (agg);
      if (seek_format != dest_format) {
        result = FALSE;
        GST_DEBUG_OBJECT (aagg,
            "seeking failed, unhandled seek format: %s",
            gst_format_get_name (seek_format));
        goto done;
      }
    }
      break;
    default:
      break;
  }

  return
      GST_AGGREGATOR_CLASS (gst_audio_aggregator_parent_class)->src_event (agg,
      event);

done:
  return result;
}


static gboolean
gst_audio_aggregator_sink_event (GstAggregator * agg,
    GstAggregatorPad * aggpad, GstEvent * event)
{
1001
  GstAudioAggregatorPad *aaggpad = GST_AUDIO_AGGREGATOR_PAD (aggpad);
1002 1003 1004 1005 1006 1007 1008 1009 1010 1011 1012 1013 1014 1015 1016 1017 1018 1019 1020 1021 1022 1023
  gboolean res = TRUE;

  GST_DEBUG_OBJECT (aggpad, "Got %s event on sink pad",
      GST_EVENT_TYPE_NAME (event));

  switch (GST_EVENT_TYPE (event)) {
    case GST_EVENT_SEGMENT:
    {
      const GstSegment *segment;
      gst_event_parse_segment (event, &segment);

      if (segment->format != GST_FORMAT_TIME) {
        GST_ERROR_OBJECT (agg, "Segment of type %s are not supported,"
            " only TIME segments are supported",
            gst_format_get_name (segment->format));
        gst_event_unref (event);
        event = NULL;
        res = FALSE;
        break;
      }

      GST_OBJECT_LOCK (agg);
1024
      if (segment->rate != GST_AGGREGATOR_PAD (agg->srcpad)->segment.rate) {
1025 1026
        GST_ERROR_OBJECT (aggpad,
            "Got segment event with wrong rate %lf, expected %lf",
1027
            segment->rate, GST_AGGREGATOR_PAD (agg->srcpad)->segment.rate);
1028 1029 1030 1031 1032 1033 1034 1035 1036 1037 1038 1039 1040 1041 1042 1043 1044 1045 1046
        res = FALSE;
        gst_event_unref (event);
        event = NULL;
      } else if (segment->rate < 0.0) {
        GST_ERROR_OBJECT (aggpad, "Negative rates not supported yet");
        res = FALSE;
        gst_event_unref (event);
        event = NULL;
      } else {
        GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (aggpad);

        GST_OBJECT_LOCK (pad);
        pad->priv->new_segment = TRUE;
        GST_OBJECT_UNLOCK (pad);
      }
      GST_OBJECT_UNLOCK (agg);

      break;
    }
1047 1048 1049 1050 1051 1052 1053 1054 1055 1056 1057
    case GST_EVENT_CAPS:
    {
      GstCaps *caps;

      gst_event_parse_caps (event, &caps);
      GST_INFO_OBJECT (aggpad, "Got caps %" GST_PTR_FORMAT, caps);
      res = gst_audio_aggregator_sink_setcaps (aaggpad, agg, caps);
      gst_event_unref (event);
      event = NULL;
      break;
    }
1058 1059 1060 1061 1062 1063 1064 1065 1066 1067 1068 1069
    default:
      break;
  }

  if (event != NULL)
    return
        GST_AGGREGATOR_CLASS (gst_audio_aggregator_parent_class)->sink_event
        (agg, aggpad, event);

  return res;
}

1070 1071 1072 1073 1074 1075 1076 1077 1078 1079 1080 1081 1082 1083 1084 1085 1086 1087 1088 1089 1090 1091 1092 1093 1094 1095 1096 1097 1098
static gboolean
gst_audio_aggregator_sink_query (GstAggregator * agg, GstAggregatorPad * aggpad,
    GstQuery * query)
{
  gboolean res = FALSE;

  switch (GST_QUERY_TYPE (query)) {
    case GST_QUERY_CAPS:
    {
      GstCaps *filter, *caps;

      gst_query_parse_caps (query, &filter);
      caps = gst_audio_aggregator_sink_getcaps (GST_PAD (aggpad), agg, filter);
      gst_query_set_caps_result (query, caps);
      gst_caps_unref (caps);
      res = TRUE;
      break;
    }
    default:
      res =
          GST_AGGREGATOR_CLASS (gst_audio_aggregator_parent_class)->sink_query
          (agg, aggpad, query);
      break;
  }

  return res;
}


1099 1100 1101 1102 1103 1104 1105 1106 1107 1108 1109 1110 1111 1112 1113 1114 1115 1116 1117 1118 1119 1120 1121 1122 1123 1124 1125 1126 1127 1128 1129 1130 1131 1132 1133 1134 1135 1136 1137 1138 1139 1140 1141 1142 1143 1144 1145 1146 1147 1148 1149 1150 1151 1152 1153 1154 1155 1156 1157 1158 1159 1160 1161 1162 1163 1164 1165 1166 1167 1168 1169 1170 1171 1172 1173 1174 1175 1176 1177 1178 1179 1180 1181 1182 1183 1184 1185 1186 1187 1188 1189 1190 1191
/* FIXME, the duration query should reflect how long you will produce
 * data, that is the amount of stream time until you will emit EOS.
 *
 * For synchronized mixing this is always the max of all the durations
 * of upstream since we emit EOS when all of them finished.
 *
 * We don't do synchronized mixing so this really depends on where the
 * streams where punched in and what their relative offsets are against
 * eachother which we can get from the first timestamps we see.
 *
 * When we add a new stream (or remove a stream) the duration might
 * also become invalid again and we need to post a new DURATION
 * message to notify this fact to the parent.
 * For now we take the max of all the upstream elements so the simple
 * cases work at least somewhat.
 */
static gboolean
gst_audio_aggregator_query_duration (GstAudioAggregator * aagg,
    GstQuery * query)
{
  gint64 max;
  gboolean res;
  GstFormat format;
  GstIterator *it;
  gboolean done;
  GValue item = { 0, };

  /* parse format */
  gst_query_parse_duration (query, &format, NULL);

  max = -1;
  res = TRUE;
  done = FALSE;

  it = gst_element_iterate_sink_pads (GST_ELEMENT_CAST (aagg));
  while (!done) {
    GstIteratorResult ires;

    ires = gst_iterator_next (it, &item);
    switch (ires) {
      case GST_ITERATOR_DONE:
        done = TRUE;
        break;
      case GST_ITERATOR_OK:
      {
        GstPad *pad = g_value_get_object (&item);
        gint64 duration;

        /* ask sink peer for duration */
        res &= gst_pad_peer_query_duration (pad, format, &duration);
        /* take max from all valid return values */
        if (res) {
          /* valid unknown length, stop searching */
          if (duration == -1) {
            max = duration;
            done = TRUE;
          }
          /* else see if bigger than current max */
          else if (duration > max)
            max = duration;
        }
        g_value_reset (&item);
        break;
      }
      case GST_ITERATOR_RESYNC:
        max = -1;
        res = TRUE;
        gst_iterator_resync (it);
        break;
      default:
        res = FALSE;
        done = TRUE;
        break;
    }
  }
  g_value_unset (&item);
  gst_iterator_free (it);

  if (res) {
    /* and store the max */
    GST_DEBUG_OBJECT (aagg, "Total duration in format %s: %"
        GST_TIME_FORMAT, gst_format_get_name (format), GST_TIME_ARGS (max));
    gst_query_set_duration (query, format, max);
  }

  return res;
}


static gboolean
gst_audio_aggregator_src_query (GstAggregator * agg, GstQuery * query)
{
  GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (agg);
1192
  GstAudioAggregatorPad *srcpad = GST_AUDIO_AGGREGATOR_PAD (agg->srcpad);
1193 1194 1195 1196 1197 1198 1199 1200 1201 1202 1203 1204 1205 1206 1207 1208
  gboolean res = FALSE;

  switch (GST_QUERY_TYPE (query)) {
    case GST_QUERY_DURATION:
      res = gst_audio_aggregator_query_duration (aagg, query);
      break;
    case GST_QUERY_POSITION:
    {
      GstFormat format;

      gst_query_parse_position (query, &format, NULL);

      GST_OBJECT_LOCK (aagg);

      switch (format) {
        case GST_FORMAT_TIME:
1209
          gst_query_set_position (query, format,
1210 1211 1212
              gst_segment_to_stream_time (&GST_AGGREGATOR_PAD (agg->
                      srcpad)->segment, GST_FORMAT_TIME,
                  GST_AGGREGATOR_PAD (agg->srcpad)->segment.position));
1213 1214 1215
          res = TRUE;
          break;
        case GST_FORMAT_BYTES:
1216
          if (GST_AUDIO_INFO_BPF (&srcpad->info)) {
1217
            gst_query_set_position (query, format, aagg->priv->offset *
1218
                GST_AUDIO_INFO_BPF (&srcpad->info));
1219 1220 1221 1222 1223 1224 1225 1226 1227 1228 1229 1230 1231 1232 1233 1234 1235 1236 1237 1238 1239 1240 1241 1242 1243 1244 1245 1246 1247 1248
            res = TRUE;
          }
          break;
        case GST_FORMAT_DEFAULT:
          gst_query_set_position (query, format, aagg->priv->offset);
          res = TRUE;
          break;
        default:
          break;
      }

      GST_OBJECT_UNLOCK (aagg);

      break;
    }
    default:
      res =
          GST_AGGREGATOR_CLASS (gst_audio_aggregator_parent_class)->src_query
          (agg, query);
      break;
  }

  return res;
}


void
gst_audio_aggregator_set_sink_caps (GstAudioAggregator * aagg,
    GstAudioAggregatorPad * pad, GstCaps * caps)
{
1249
#ifndef G_DISABLE_ASSERT
1250 1251
  gboolean valid;

1252
  GST_OBJECT_LOCK (pad);
1253 1254
  valid = gst_audio_info_from_caps (&pad->info, caps);
  g_assert (valid);
1255
  GST_OBJECT_UNLOCK (pad);
1256 1257 1258 1259 1260
#else
  GST_OBJECT_LOCK (pad);
  (void) gst_audio_info_from_caps (&pad->info, caps);
  GST_OBJECT_UNLOCK (pad);
#endif
1261 1262 1263 1264 1265 1266 1267 1268 1269 1270 1271
}

/* Must hold object lock and aagg lock to call */

static void
gst_audio_aggregator_reset (GstAudioAggregator * aagg)
{
  GstAggregator *agg = GST_AGGREGATOR (aagg);

  GST_AUDIO_AGGREGATOR_LOCK (aagg);
  GST_OBJECT_LOCK (aagg);
1272
  GST_AGGREGATOR_PAD (agg->srcpad)->segment.position = -1;
1273
  aagg->priv->offset = -1;
1274
  gst_audio_info_init (&GST_AUDIO_AGGREGATOR_PAD (agg->srcpad)->info);
1275 1276 1277 1278 1279 1280 1281 1282 1283 1284 1285 1286 1287 1288 1289 1290 1291 1292 1293 1294 1295 1296 1297 1298 1299 1300 1301 1302 1303 1304 1305 1306 1307
  gst_caps_replace (&aagg->current_caps, NULL);
  gst_buffer_replace (&aagg->priv->current_buffer, NULL);
  GST_OBJECT_UNLOCK (aagg);
  GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
}

static gboolean
gst_audio_aggregator_start (GstAggregator * agg)
{
  GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (agg);

  gst_audio_aggregator_reset (aagg);

  return TRUE;
}

static gboolean
gst_audio_aggregator_stop (GstAggregator * agg)
{
  GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (agg);

  gst_audio_aggregator_reset (aagg);

  return TRUE;
}

static GstFlowReturn
gst_audio_aggregator_flush (GstAggregator * agg)
{
  GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (agg);

  GST_AUDIO_AGGREGATOR_LOCK (aagg);
  GST_OBJECT_LOCK (aagg);
1308
  GST_AGGREGATOR_PAD (agg->srcpad)->segment.position = -1;
1309
  aagg->priv->offset = -1;
1310 1311 1312 1313 1314 1315 1316
  gst_buffer_replace (&aagg->priv->current_buffer, NULL);
  GST_OBJECT_UNLOCK (aagg);
  GST_AUDIO_AGGREGATOR_UNLOCK (aagg);

  return GST_FLOW_OK;
}

1317
static GstBuffer *
1318
gst_audio_aggregator_do_clip (GstAggregator * agg,
1319
    GstAggregatorPad * bpad, GstBuffer * buffer)
1320 1321 1322 1323 1324 1325 1326 1327
{
  GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (bpad);
  gint rate, bpf;

  rate = GST_AUDIO_INFO_RATE (&pad->info);
  bpf = GST_AUDIO_INFO_BPF (&pad->info);

  GST_OBJECT_LOCK (bpad);
1328
  buffer = gst_audio_buffer_clip (buffer, &bpad->segment, rate, bpf);
1329 1330
  GST_OBJECT_UNLOCK (bpad);

1331
  return buffer;
1332 1333 1334 1335
}

/* Called with the object lock for both the element and pad held,
 * as well as the aagg lock
1336 1337 1338
 *
 * Replace the current buffer with input and update GstAudioAggregatorPadPrivate
 * values.
1339 1340
 */
static gboolean
1341 1342
gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg,
    GstAudioAggregatorPad * pad)
1343 1344 1345 1346 1347 1348
{
  GstClockTime start_time, end_time;
  gboolean discont = FALSE;
  guint64 start_offset, end_offset;
  gint rate, bpf;

1349
  GstAggregator *agg = GST_AGGREGATOR (aagg);
1350
  GstAggregatorPad *aggpad = GST_AGGREGATOR_PAD (pad);
1351
  GstAudioAggregatorPad *srcpad = GST_AUDIO_AGGREGATOR_PAD (agg->srcpad);
1352

1353
  if (GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (pad)->convert_buffer) {
1354 1355
    rate = GST_AUDIO_INFO_RATE (&srcpad->info);
    bpf = GST_AUDIO_INFO_BPF (&srcpad->info);
1356 1357 1358 1359
  } else {
    rate = GST_AUDIO_INFO_RATE (&pad->info);
    bpf = GST_AUDIO_INFO_BPF (&pad->info);
  }
1360 1361

  pad->priv->position = 0;
1362
  pad->priv->size = gst_buffer_get_size (pad->priv->buffer) / bpf;
1363

1364
  if (pad->priv->size == 0) {
1365 1366
    if (!GST_BUFFER_DURATION_IS_VALID (pad->priv->buffer) ||
        !GST_BUFFER_FLAG_IS_SET (pad->priv->buffer, GST_BUFFER_FLAG_GAP)) {
1367
      GST_WARNING_OBJECT (pad, "Dropping 0-sized buffer missing either a"
1368
          " duration or a GAP flag: %" GST_PTR_FORMAT, pad->priv->buffer);
1369 1370 1371
      return FALSE;
    }

1372 1373
    pad->priv->size =
        gst_util_uint64_scale (GST_BUFFER_DURATION (pad->priv->buffer), rate,
1374 1375 1376
        GST_SECOND);
  }

1377
  if (!GST_BUFFER_PTS_IS_VALID (pad->priv->buffer)) {
1378 1379 1380 1381 1382 1383 1384 1385 1386
    if (pad->priv->output_offset == -1)
      pad->priv->output_offset = aagg->priv->offset;
    if (pad->priv->next_offset == -1)
      pad->priv->next_offset = pad->priv->size;
    else
      pad->priv->next_offset += pad->priv->size;
    goto done;
  }

1387
  start_time = GST_BUFFER_PTS (pad->priv->buffer);
1388 1389 1390 1391
  end_time =
      start_time + gst_util_uint64_scale_ceil (pad->priv->size, GST_SECOND,
      rate);

1392 1393 1394 1395 1396 1397
  /* Clipping should've ensured this */
  g_assert (start_time >= aggpad->segment.start);

  start_offset =
      gst_util_uint64_scale (start_time - aggpad->segment.start, rate,
      GST_SECOND);
1398 1399
  end_offset = start_offset + pad->priv->size;

1400 1401
  if (GST_BUFFER_IS_DISCONT (pad->priv->buffer)
      || GST_BUFFER_FLAG_IS_SET (pad->priv->buffer, GST_BUFFER_FLAG_RESYNC)
1402 1403 1404 1405 1406 1407 1408 1409 1410 1411 1412 1413 1414 1415 1416 1417 1418 1419 1420 1421 1422 1423 1424 1425 1426 1427 1428 1429 1430 1431 1432 1433 1434 1435 1436 1437 1438 1439
      || pad->priv->new_segment || pad->priv->next_offset == -1) {
    discont = TRUE;
    pad->priv->new_segment = FALSE;
  } else {
    guint64 diff, max_sample_diff;

    /* Check discont, based on audiobasesink */
    if (start_offset <= pad->priv->next_offset)
      diff = pad->priv->next_offset - start_offset;
    else
      diff = start_offset - pad->priv->next_offset;

    max_sample_diff =
        gst_util_uint64_scale_int (aagg->priv->alignment_threshold, rate,
        GST_SECOND);

    /* Discont! */
    if (G_UNLIKELY (diff >= max_sample_diff)) {
      if (aagg->priv->discont_wait > 0) {
        if (pad->priv->discont_time == GST_CLOCK_TIME_NONE) {
          pad->priv->discont_time = start_time;
        } else if (start_time - pad->priv->discont_time >=
            aagg->priv->discont_wait) {
          discont = TRUE;
          pad->priv->discont_time = GST_CLOCK_TIME_NONE;
        }
      } else {
        discont = TRUE;
      }
    } else if (G_UNLIKELY (pad->priv->discont_time != GST_CLOCK_TIME_NONE)) {
      /* we have had a discont, but are now back on track! */
      pad->priv->discont_time = GST_CLOCK_TIME_NONE;
    }
  }

  if (discont) {
    /* Have discont, need resync */
    if (pad->priv->next_offset != -1)
1440
      GST_DEBUG_OBJECT (pad, "Have discont. Expected %"
1441 1442 1443 1444 1445 1446 1447 1448 1449 1450 1451
          G_GUINT64_FORMAT ", got %" G_GUINT64_FORMAT,
          pad->priv->next_offset, start_offset);
    pad->priv->output_offset = -1;
    pad->priv->next_offset = end_offset;
  } else {
    pad->priv->next_offset += pad->priv->size;
  }

  if (pad->priv->output_offset == -1) {
    GstClockTime start_running_time;
    GstClockTime end_running_time;
1452 1453 1454
    GstClockTime segment_pos;
    guint64 start_output_offset = -1;
    guint64 end_output_offset = -1;
1455
    GstSegment *agg_segment = &GST_AGGREGATOR_PAD (agg->srcpad)->segment;
1456 1457 1458 1459 1460 1461 1462 1463

    start_running_time =
        gst_segment_to_running_time (&aggpad->segment,
        GST_FORMAT_TIME, start_time);
    end_running_time =
        gst_segment_to_running_time (&aggpad->segment,
        GST_FORMAT_TIME, end_time);

1464
    /* Convert to position in the output segment */
1465
    segment_pos =
1466
        gst_segment_position_from_running_time (agg_segment, GST_FORMAT_TIME,
1467
        start_running_time);
1468
    if (GST_CLOCK_TIME_IS_VALID (segment_pos))
1469
      start_output_offset =
1470
          gst_util_uint64_scale (segment_pos - agg_segment->start, rate,
1471 1472
          GST_SECOND);

1473
    segment_pos =
1474
        gst_segment_position_from_running_time (agg_segment, GST_FORMAT_TIME,
1475
        end_running_time);
1476
    if (GST_CLOCK_TIME_IS_VALID (segment_pos))
1477
      end_output_offset =
1478
          gst_util_uint64_scale (segment_pos - agg_segment->start, rate,
1479 1480 1481 1482 1483 1484 1485 1486 1487 1488 1489 1490 1491 1492 1493 1494
          GST_SECOND);

    if (start_output_offset == -1 && end_output_offset == -1) {
      /* Outside output segment, drop */
      pad->priv->position = 0;
      pad->priv->size = 0;
      pad->priv->output_offset = -1;
      GST_DEBUG_OBJECT (pad, "Buffer outside output segment");
      return FALSE;
    }

    /* Calculate end_output_offset if it was outside the output segment */
    if (end_output_offset == -1)
      end_output_offset = start_output_offset + pad->priv->size;

    if (end_output_offset < aagg->priv->offset) {
1495 1496 1497 1498 1499
      pad->priv->position = 0;
      pad->priv->size = 0;
      pad->priv->output_offset = -1;
      GST_DEBUG_OBJECT (pad,
          "Buffer before segment or current position: %" G_GUINT64_FORMAT " < %"
1500
          G_GINT64_FORMAT, end_output_offset, aagg->priv->offset);
1501 1502 1503
      return FALSE;
    }

1504 1505 1506 1507 1508 1509 1510 1511 1512 1513 1514 1515 1516 1517 1518
    if (start_output_offset == -1 || start_output_offset < aagg->priv->offset) {
      guint diff;

      if (start_output_offset == -1 && end_output_offset < pad->priv->size) {
        diff = pad->priv->size - end_output_offset + aagg->priv->offset;
      } else if (start_output_offset == -1) {
        start_output_offset = end_output_offset - pad->priv->size;

        if (start_output_offset < aagg->priv->offset)
          diff = aagg->priv->offset - start_output_offset;
        else
          diff = 0;
      } else {
        diff = aagg->priv->offset - start_output_offset;
      }
1519 1520 1521 1522 1523 1524 1525 1526 1527

      pad->priv->position += diff;
      if (pad->priv->position >= pad->priv->size) {
        /* Empty buffer, drop */
        pad->priv->position = 0;
        pad->priv->size = 0;
        pad->priv->output_offset = -1;
        GST_DEBUG_OBJECT (pad,
            "Buffer before segment or current position: %" G_GUINT64_FORMAT
1528
            " < %" G_GINT64_FORMAT, end_output_offset, aagg->priv->offset);
1529 1530 1531 1532
        return FALSE;
      }
    }

1533 1534 1535 1536 1537
    if (start_output_offset == -1 || start_output_offset < aagg->priv->offset)
      pad->priv->output_offset = aagg->priv->offset;
    else
      pad->priv->output_offset = start_output_offset;

1538 1539
    GST_DEBUG_OBJECT (pad,
        "Buffer resynced: Pad offset %" G_GUINT64_FORMAT
1540
        ", current audio aggregator offset %" G_GINT64_FORMAT,
1541 1542 1543 1544 1545 1546 1547 1548 1549 1550 1551 1552 1553 1554 1555 1556
        pad->priv->output_offset, aagg->priv->offset);
  }

done:

  GST_LOG_OBJECT (pad,
      "Queued new buffer at offset %" G_GUINT64_FORMAT,
      pad->priv->output_offset);

  return TRUE;
}

/* Called with pad object lock held */

static gboolean
gst_audio_aggregator_mix_buffer (GstAudioAggregator * aagg,
1557 1558
    GstAudioAggregatorPad * pad, GstBuffer * inbuf, GstBuffer * outbuf,
    guint blocksize)
1559 1560 1561 1562
{
  guint overlap;
  guint out_start;
  gboolean filled;
1563 1564
  guint in_offset;
  gboolean pad_changed = FALSE;
1565 1566 1567 1568 1569 1570 1571 1572 1573 1574 1575 1576 1577 1578

  /* Overlap => mix */
  if (aagg->priv->offset < pad->priv->output_offset)
    out_start = pad->priv->output_offset - aagg->priv->offset;
  else
    out_start = 0;

  overlap = pad->priv->size - pad->priv->position;
  if (overlap > blocksize - out_start)
    overlap = blocksize - out_start;

  if (GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP)) {
    /* skip gap buffer */
    GST_LOG_OBJECT (pad, "skipping GAP buffer");
1579
    pad->priv->output_offset += pad->priv->size - pad->priv->position;
1580 1581 1582
    pad->priv->position = pad->priv->size;

    gst_buffer_replace (&pad->priv->buffer, NULL);
1583
    gst_buffer_replace (&pad->priv->input_buffer, NULL);
1584 1585 1586
    return FALSE;
  }

1587 1588 1589 1590 1591
  gst_buffer_ref (inbuf);
  in_offset = pad->priv->position;
  GST_OBJECT_UNLOCK (pad);
  GST_OBJECT_UNLOCK (aagg);

1592
  filled = GST_AUDIO_AGGREGATOR_GET_CLASS (aagg)->aggregate_one_buffer (aagg,
1593 1594 1595 1596 1597 1598 1599
      pad, inbuf, in_offset, outbuf, out_start, overlap);

  GST_OBJECT_LOCK (aagg);
  GST_OBJECT_LOCK (pad);

  pad_changed = (inbuf != pad->priv->buffer);
  gst_buffer_unref (inbuf);
1600 1601 1602 1603

  if (filled)
    GST_BUFFER_FLAG_UNSET (outbuf, GST_BUFFER_FLAG_GAP);

1604 1605 1606
  if (pad_changed)
    return FALSE;

1607 1608 1609 1610 1611 1612
  pad->priv->position += overlap;
  pad->priv->output_offset += overlap;

  if (pad->priv->position == pad->priv->size) {
    /* Buffer done, drop it */
    gst_buffer_replace (&pad->priv->buffer, NULL);
1613
    gst_buffer_replace (&pad->priv->input_buffer, NULL);
1614
    GST_LOG_OBJECT (pad, "Finished mixing buffer, waiting for next");
1615 1616 1617 1618 1619 1620 1621 1622 1623 1624
    return FALSE;
  }

  return TRUE;
}

static GstBuffer *
gst_audio_aggregator_create_output_buffer (GstAudioAggregator * aagg,
    guint num_frames)
{
1625 1626 1627
  GstAllocator *allocator;
  GstAllocationParams params;
  GstBuffer *outbuf;
1628
  GstMapInfo outmap;
1629 1630
  GstAggregator *agg = GST_AGGREGATOR (aagg);
  GstAudioAggregatorPad *srcpad = GST_AUDIO_AGGREGATOR_PAD (agg->srcpad);
1631

1632 1633
  gst_aggregator_get_allocator (GST_AGGREGATOR (aagg), &allocator, &params);

1634
  GST_DEBUG ("Creating output buffer with size %d",
1635
      num_frames * GST_AUDIO_INFO_BPF (&srcpad->info));
1636

1637
  outbuf = gst_buffer_new_allocate (allocator, num_frames *
1638
      GST_AUDIO_INFO_BPF (&srcpad->info), &params);
1639 1640 1641 1642

  if (allocator)
    gst_object_unref (allocator);

1643
  gst_buffer_map (outbuf, &outmap, GST_MAP_WRITE);
1644
  gst_audio_format_fill_silence (srcpad->info.finfo, outmap.data, outmap.size);
1645 1646 1647 1648 1649
  gst_buffer_unmap (outbuf, &outmap);

  return outbuf;
}

1650
static gboolean
1651
sync_pad_values (GstElement * aagg, GstPad * pad, gpointer user_data)
1652
{
1653 1654
  GstAudioAggregatorPad *aapad = GST_AUDIO_AGGREGATOR_PAD (pad);
  GstAggregatorPad *bpad = GST_AGGREGATOR_PAD_CAST (pad);
1655 1656
  GstClockTime timestamp, stream_time;

1657
  if (aapad->priv->buffer == NULL)
1658 1659
    return TRUE;

1660
  timestamp = GST_BUFFER_PTS (aapad->priv->buffer);
1661 1662 1663 1664 1665 1666 1667 1668
  GST_OBJECT_LOCK (bpad);
  stream_time = gst_segment_to_stream_time (&bpad->segment, GST_FORMAT_TIME,
      timestamp);
  GST_OBJECT_UNLOCK (bpad);

  /* sync object properties on stream time */
  /* TODO: Ideally we would want to do that on every sample */
  if (GST_CLOCK_TIME_IS_VALID (stream_time))
1669
    gst_object_sync_values (GST_OBJECT_CAST (pad), stream_time);
1670 1671 1672 1673

  return TRUE;
}

1674 1675 1676
static GstFlowReturn
gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
{
1677 1678
  /* Calculate the current output offset/timestamp and offset_end/timestamp_end.
   * Allocate a silence buffer for this and store it.
1679 1680 1681 1682 1683 1684 1685 1686 1687 1688 1689 1690 1691 1692 1693 1694
   *
   * For all pads:
   * 1) Once per input buffer (cached)
   *   1) Check discont (flag and timestamp with tolerance)
   *   2) If discont or new, resync. That means:
   *     1) Drop all start data of the buffer that comes before
   *        the current position/offset.
   *     2) Calculate the offset (output segment!) that the first
   *        frame of the input buffer corresponds to. Base this on
   *        the running time.
   *
   * 2) If the current pad's offset/offset_end overlaps with the output
   *    offset/offset_end, mix it at the appropiate position in the output
   *    buffer and advance the pad's position. Remember if this pad needs
   *    a new buffer to advance behind the output offset_end.
   *
1695
   * If we had no pad with a buffer, go EOS.
1696
   *
1697 1698 1699
   * If we had at least one pad that did not advance behind output
   * offset_end, let aggregate be called again for the current
   * output offset/offset_end.
1700 1701 1702 1703 1704 1705 1706 1707 1708 1709 1710 1711 1712
   */
  GstElement *element;
  GstAudioAggregator *aagg;
  GList *iter;
  GstFlowReturn ret;
  GstBuffer *outbuf = NULL;
  gint64 next_offset;
  gint64 next_timestamp;
  gint rate, bpf;
  gboolean dropped = FALSE;
  gboolean is_eos = TRUE;
  gboolean is_done = TRUE;
  guint blocksize;
1713
  GstAudioAggregatorPad *srcpad = GST_AUDIO_AGGREGATOR_PAD (agg->srcpad);
1714
  GstSegment *agg_segment = &GST_AGGREGATOR_PAD (agg->srcpad)->segment;
1715 1716 1717 1718

  element = GST_ELEMENT (agg);
  aagg = GST_AUDIO_AGGREGATOR (agg);

1719
  /* Sync pad properties to the stream time */
1720
  gst_element_foreach_sink_pad (element, sync_pad_values, NULL);
1721

1722 1723 1724 1725
  GST_AUDIO_AGGREGATOR_LOCK (aagg);
  GST_OBJECT_LOCK (agg);

  /* Update position from the segment start/stop if needed */
1726 1727 1728
  if (agg_segment->position == -1) {
    if (agg_segment->rate > 0.0)
      agg_segment->position = agg_segment->start;
1729
    else
1730
      agg_segment->position = agg_segment->stop;
1731 1732
  }

1733
  if (G_UNLIKELY (srcpad->info.finfo->format == GST_AUDIO_FORMAT_UNKNOWN)) {
1734 1735 1736 1737 1738
    if (timeout) {
      GST_DEBUG_OBJECT (aagg,
          "Got timeout before receiving any caps, don't output anything");

      /* Advance position */
1739 1740 1741 1742
      if (agg_segment->rate > 0.0)
        agg_segment->position += aagg->priv->output_buffer_duration;
      else if (agg_segment->position > aagg->priv->output_buffer_duration)
        agg_segment->position -= aagg->priv->output_buffer_duration;
1743
      else
1744
        agg_segment->position = 0;
1745 1746 1747

      GST_OBJECT_UNLOCK (agg);
      GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
1748
      return GST_AGGREGATOR_FLOW_NEED_DATA;
1749 1750 1751 1752 1753 1754
    } else {
      GST_OBJECT_UNLOCK (agg);
      goto not_negotiated;
    }
  }

1755 1756
  rate = GST_AUDIO_INFO_RATE (&srcpad->info);
  bpf = GST_AUDIO_INFO_BPF (&srcpad->info);
1757

1758 1759
  if (aagg->priv->offset == -1) {
    aagg->priv->offset =
1760
        gst_util_uint64_scale (agg_segment->position - agg_segment->start, rate,
1761
        GST_SECOND);
1762 1763
    GST_DEBUG_OBJECT (aagg, "Starting at offset %" G_GINT64_FORMAT,
        aagg->priv->offset);
1764 1765
  }

1766
  blocksize = gst_util_uint64_scale (aagg->priv->output_buffer_duration,
1767
      rate, GST_SECOND);
1768
  blocksize = MAX (1, blocksize);
1769 1770

  /* FIXME: Reverse mixing does not work at all yet */
1771
  if (agg_segment->rate > 0.0) {
1772 1773 1774 1775 1776
    next_offset = aagg->priv->offset + blocksize;
  } else {
    next_offset = aagg->priv->offset - blocksize;
  }

1777
  /* Use the sample counter, which will never accumulate rounding errors */
1778
  next_timestamp =
1779
      agg_segment->start + gst_util_uint64_scale (next_offset, GST_SECOND,
1780
      rate);
1781 1782 1783 1784 1785 1786 1787 1788 1789 1790 1791 1792 1793

  if (aagg->priv->current_buffer == NULL) {
    GST_OBJECT_UNLOCK (agg);
    aagg->priv->current_buffer =
        GST_AUDIO_AGGREGATOR_GET_CLASS (aagg)->create_output_buffer (aagg,
        blocksize);
    /* Be careful, some things could have changed ? */
    GST_OBJECT_LOCK (agg);
    GST_BUFFER_FLAG_SET (aagg->priv->current_buffer, GST_BUFFER_FLAG_GAP);
  }
  outbuf = aagg->priv->current_buffer;

  GST_LOG_OBJECT (agg,
1794
      "Starting to mix %u samples for offset %" G_GINT64_FORMAT
1795
      " with timestamp %" GST_TIME_FORMAT, blocksize,
1796
      aagg->priv->offset, GST_TIME_ARGS (agg_segment->position));
1797 1798 1799 1800 1801 1802 1803 1804 1805

  for (iter = element->sinkpads; iter; iter = iter->next) {
    GstAudioAggregatorPad *pad = (GstAudioAggregatorPad *) iter->data;
    GstAggregatorPad *aggpad = (GstAggregatorPad *) iter->data;
    gboolean pad_eos = gst_aggregator_pad_is_eos (aggpad);

    if (!pad_eos)
      is_eos = FALSE;

1806
    pad->priv->input_buffer = gst_aggregator_pad_peek_buffer (aggpad);
1807 1808

    GST_OBJECT_LOCK (pad);
1809
    if (!pad->priv->input_buffer) {
1810 1811 1812
      if (timeout) {
        if (pad->priv->output_offset < next_offset) {
          gint64 diff = next_offset - pad->priv->output_offset;
1813 1814
          GST_DEBUG_OBJECT (pad, "Timeout, missing %" G_GINT64_FORMAT
              " frames (%" GST_TIME_FORMAT ")", diff,
1815
              GST_TIME_ARGS (gst_util_uint64_scale (diff, GST_SECOND,
1816
                      GST_AUDIO_INFO_RATE (&srcpad->info))));
1817 1818 1819 1820 1821 1822 1823 1824 1825 1826
        }
      } else if (!pad_eos) {
        is_done = FALSE;
      }
      GST_OBJECT_UNLOCK (pad);
      continue;
    }

    /* New buffer? */
    if (!pad->priv->buffer) {
1827
      if (GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (pad)->convert_buffer)
1828 1829
        pad->priv->buffer =
            gst_audio_aggregator_convert_buffer
1830
            (aagg, GST_PAD (pad), &pad->info, &srcpad->info,
1831 1832 1833 1834 1835 1836 1837 1838
            pad->priv->input_buffer);
      else
        pad->priv->buffer = gst_buffer_ref (pad->priv->input_buffer);

      if (!gst_audio_aggregator_fill_buffer (aagg, pad)) {
        gst_buffer_replace (&pad->priv->buffer, NULL);
        gst_buffer_replace (&pad->priv->input_buffer, NULL);
        pad->priv->buffer = NULL;
1839 1840
        dropped = TRUE;
        GST_OBJECT_UNLOCK (pad);
1841

1842 1843 1844 1845
        gst_aggregator_pad_drop_buffer (aggpad);
        continue;
      }
    } else {
1846
      gst_buffer_unref (pad->priv->input_buffer);
1847 1848 1849 1850 1851 1852 1853 1854 1855 1856
    }

    if (!pad->priv->buffer && !dropped && pad_eos) {
      GST_DEBUG_OBJECT (aggpad, "Pad is in EOS state");
      GST_OBJECT_UNLOCK (pad);
      continue;
    }

    g_assert (pad->priv->buffer);

1857
    /* This pad is lagging behind, we need to update the offset
1858 1859 1860
     * and maybe drop the current buffer */
    if (pad->priv->output_offset < aagg->priv->offset) {
      gint64 diff = aagg->priv->offset - pad->priv->output_offset;
1861
      gint64 odiff = diff;
1862 1863 1864 1865 1866 1867 1868

      if (pad->priv->position + diff > pad->priv->size)
        diff = pad->priv->size - pad->priv->position;
      pad->priv->position += diff;
      pad->priv->output_offset += diff;

      if (pad->priv->position == pad->priv->size) {
1869
        GST_DEBUG_OBJECT