gstfaac.c 19.2 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
/* GStreamer FAAC (Free AAC Encoder) plugin
 * Copyright (C) 2003 Ronald Bultje <rbultje@ronald.bitfreak.net>
 *
 * 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., 59 Temple Place - Suite 330,
 * Boston, MA 02111-1307, USA.
 */

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

#include "gstfaac.h"

26 27
static GstStaticPadTemplate src_template =
GST_STATIC_PAD_TEMPLATE (
28 29 30
  "src",
  GST_PAD_SRC,
  GST_PAD_ALWAYS,
31 32 33 34 35
  GST_STATIC_CAPS (
    "audio/mpeg, "
      "mpegversion = (int) { 4, 2 }, "
      "channels = (int) [ 1, 6 ], "
      "rate = (int) [ 8000, 96000 ]"
36 37 38
  )
);

39 40
static GstStaticPadTemplate sink_template =
GST_STATIC_PAD_TEMPLATE (
41 42 43
  "sink",
  GST_PAD_SINK,
  GST_PAD_ALWAYS,
44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63
  GST_STATIC_CAPS (
    "audio/x-raw-int, "
      "endianness = (int) BYTE_ORDER, "
      "signed = (boolean) TRUE, "
      "width = (int) 16, "
      "depth = (int) 16, "
      "rate = (int) [ 8000, 96000 ], "
      "channels = (int) [ 1, 6]; "
    "audio/x-raw-int, "
      "endianness = (int) BYTE_ORDER, "
      "signed = (boolean) TRUE, "
      "width = (int) 32, "
      "depth = (int) 24, "
      "rate = (int) [ 8000, 96000], "
      "channels = (int) [ 1, 6]; "
    "audio/x-raw-float, "
      "endianness = (int) BYTE_ORDER, "
      "depth = (int) 32, " /* sizeof (gfloat) */
      "rate = (int) [ 8000, 96000], "
      "channels = (int) [ 1, 6]"
64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91
  )
);

enum {
  ARG_0,
  ARG_BITRATE,
  ARG_PROFILE,
  ARG_TNS,
  ARG_MIDSIDE,
  ARG_SHORTCTL
  /* FILL ME */
};

static void     gst_faac_base_init    (GstFaacClass *klass);
static void     gst_faac_class_init   (GstFaacClass *klass);
static void     gst_faac_init         (GstFaac      *faac);

static void     gst_faac_set_property (GObject      *object,
				       guint         prop_id, 
				       const GValue *value,
				       GParamSpec   *pspec);
static void     gst_faac_get_property (GObject      *object,
				       guint         prop_id, 
				       GValue       *value,
				       GParamSpec   *pspec);

static GstPadLinkReturn
                gst_faac_sinkconnect  (GstPad       *pad,
92
				       const GstCaps *caps);
93 94
static GstPadLinkReturn
                gst_faac_srcconnect   (GstPad       *pad,
95
				       const GstCaps *caps);
96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141
static void     gst_faac_chain        (GstPad       *pad,
				       GstData      *data);
static GstElementStateReturn
                gst_faac_change_state (GstElement   *element);

static GstElementClass *parent_class = NULL;
/* static guint gst_faac_signals[LAST_SIGNAL] = { 0 }; */

GType
gst_faac_get_type (void)
{
  static GType gst_faac_type = 0;

  if (!gst_faac_type) {
    static const GTypeInfo gst_faac_info = {
      sizeof (GstFaacClass),      
      (GBaseInitFunc) gst_faac_base_init,
      NULL,
      (GClassInitFunc) gst_faac_class_init,
      NULL,
      NULL,
      sizeof(GstFaac),
      0,
      (GInstanceInitFunc) gst_faac_init,
    };

    gst_faac_type = g_type_register_static (GST_TYPE_ELEMENT,
					    "GstFaac",
					    &gst_faac_info, 0);
  }

  return gst_faac_type;
}

static void
gst_faac_base_init (GstFaacClass *klass)
{
  GstElementDetails gst_faac_details = {
    "Free AAC Encoder (FAAC)",
    "Codec/Audio/Encoder",
    "Free MPEG-2/4 AAC encoder",
    "Ronald Bultje <rbultje@ronald.bitfreak.net>",
  };
  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);

  gst_element_class_add_pad_template (element_class,
142
	gst_static_pad_template_get (&src_template));
143
  gst_element_class_add_pad_template (element_class,
144
	gst_static_pad_template_get (&sink_template));
145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 171 172 173 174 175 176 177 178 179 180 181 182 183 184 185 186 187 188 189 190 191 192 193 194 195 196 197 198 199 200 201 202 203 204 205 206 207 208 209 210 211 212 213 214 215 216 217 218 219 220 221 222 223 224 225 226 227 228 229 230 231 232 233 234 235 236 237

  gst_element_class_set_details (element_class, &gst_faac_details);
}

#define GST_TYPE_FAAC_PROFILE (gst_faac_profile_get_type ())
static GType
gst_faac_profile_get_type (void)
{
  static GType gst_faac_profile_type = 0;

  if (!gst_faac_profile_type) {
    static GEnumValue gst_faac_profile[] = {
      { MAIN, "MAIN", "Main profile"                   },
      { LOW,  "LOW",  "Low complexity profile"         },
      { SSR,  "SSR",  "Scalable sampling rate profile" },
      { LTP,  "LTP",  "Long term prediction profile"   },
      { 0, NULL, NULL },
    };

    gst_faac_profile_type = g_enum_register_static ("GstFaacProfile",
						    gst_faac_profile);
  }

  return gst_faac_profile_type;
}

#define GST_TYPE_FAAC_SHORTCTL (gst_faac_shortctl_get_type ())
static GType
gst_faac_shortctl_get_type (void)
{
  static GType gst_faac_shortctl_type = 0;

  if (!gst_faac_shortctl_type) {
    static GEnumValue gst_faac_shortctl[] = {
      { SHORTCTL_NORMAL,  "SHORTCTL_NORMAL",  "Normal block type" },
      { SHORTCTL_NOSHORT, "SHORTCTL_NOSHORT", "No short blocks"   },
      { SHORTCTL_NOLONG,  "SHORTCTL_NOLONG",  "No long blocks"    },
      { 0, NULL, NULL },
    };

    gst_faac_shortctl_type = g_enum_register_static ("GstFaacShortCtl",
						    gst_faac_shortctl);
  }

  return gst_faac_shortctl_type;
}

static void
gst_faac_class_init (GstFaacClass *klass)
{
  GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
  GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);

  parent_class = g_type_class_ref (GST_TYPE_ELEMENT);

  /* properties */
  g_object_class_install_property (gobject_class, ARG_BITRATE,
    g_param_spec_int ("bitrate", "Bitrate (bps)", "Bitrate in bits/sec",
                      8 * 1024, 320 * 1024, 128 * 1024, G_PARAM_READWRITE));
  g_object_class_install_property (gobject_class, ARG_PROFILE,
    g_param_spec_enum ("profile", "Profile", "MPEG/AAC encoding profile",
                       GST_TYPE_FAAC_PROFILE, MAIN, G_PARAM_READWRITE));
  g_object_class_install_property (gobject_class, ARG_TNS,
    g_param_spec_boolean ("tns", "TNS", "Use temporal noise shaping",
                          FALSE, G_PARAM_READWRITE));
  g_object_class_install_property (gobject_class, ARG_MIDSIDE,
    g_param_spec_boolean ("midside", "Midside", "Allow mid/side encoding",
                          TRUE, G_PARAM_READWRITE));
  g_object_class_install_property (gobject_class, ARG_SHORTCTL,
    g_param_spec_enum ("shortctl", "Block type",
		       "Block type encorcing",
                       GST_TYPE_FAAC_SHORTCTL, MAIN, G_PARAM_READWRITE));

  /* virtual functions */
  gstelement_class->change_state = gst_faac_change_state;

  gobject_class->set_property = gst_faac_set_property;
  gobject_class->get_property = gst_faac_get_property;
}

static void
gst_faac_init (GstFaac *faac)
{
  faac->handle = NULL;
  faac->samplerate = -1;
  faac->channels = -1;
  faac->cache = NULL;
  faac->cache_time = GST_CLOCK_TIME_NONE;
  faac->cache_duration = 0;

  GST_FLAG_SET (faac, GST_ELEMENT_EVENT_AWARE);

  faac->sinkpad = gst_pad_new_from_template (
238
	gst_static_pad_template_get (&sink_template), "sink");
239 240 241 242 243
  gst_element_add_pad (GST_ELEMENT (faac), faac->sinkpad);
  gst_pad_set_chain_function (faac->sinkpad, gst_faac_chain);
  gst_pad_set_link_function (faac->sinkpad, gst_faac_sinkconnect);

  faac->srcpad = gst_pad_new_from_template (
244
	gst_static_pad_template_get (&src_template), "src");
245 246 247 248 249 250 251 252 253 254 255 256
  gst_element_add_pad (GST_ELEMENT (faac), faac->srcpad);
  gst_pad_set_link_function (faac->srcpad, gst_faac_srcconnect);

  /* default properties */
  faac->bitrate = 1024 * 128;
  faac->profile = MAIN;
  faac->shortctl = SHORTCTL_NORMAL;
  faac->tns = FALSE;
  faac->midside = TRUE;
}

static GstPadLinkReturn
257 258
gst_faac_sinkconnect (GstPad        *pad,
		      const GstCaps *caps)
259 260
{
  GstFaac *faac = GST_FAAC (gst_pad_get_parent (pad));
261 262 263 264
  GstStructure *structure = gst_caps_get_structure (caps, 0);
  faacEncHandle *handle;
  gint channels, samplerate, depth;
  gulong samples, bytes, fmt = 0, bps = 0;
265

266
  if (!gst_caps_is_fixed (caps))
267 268 269 270 271 272 273 274 275 276 277
    return GST_PAD_LINK_DELAYED;

  if (faac->handle) {
    faacEncClose (faac->handle);
    faac->handle = NULL;
  }
  if (faac->cache) {
    gst_buffer_unref (faac->cache);
    faac->cache = NULL;
  }

278 279 280 281 282 283 284 285 286 287 288 289 290 291 292 293 294 295 296 297 298 299 300
  gst_structure_get_int (structure, "channels", &channels);
  gst_structure_get_int (structure, "rate", &samplerate);
  gst_structure_get_int (structure, "depth", &depth);

  /* open a new handle to the encoder */
  if (!(handle = faacEncOpen (samplerate, channels,
			      &samples, &bytes)))
    return GST_PAD_LINK_REFUSED;

  switch (depth) {
    case 16:
      fmt = FAAC_INPUT_16BIT;
      bps = 2;
      break;
    case 24:
      fmt = FAAC_INPUT_32BIT; /* 24-in-32, actually */
      bps = 4;
      break;
    case 32:
      fmt = FAAC_INPUT_FLOAT; /* see template, this is right */
      bps = 4;
      break;
  }
301

302 303 304
  if (!fmt) {
    faacEncClose (handle);
    return GST_PAD_LINK_REFUSED;
305 306
  }

307 308 309 310 311 312 313 314 315 316 317 318 319 320 321
  faac->format = fmt;
  faac->bps = bps;
  faac->handle = handle;
  faac->bytes = bytes;
  faac->samples = samples;
  faac->channels = channels;
  faac->samplerate = samplerate;

  /* if the other side was already set-up, redo that */
  if (GST_PAD_CAPS (faac->srcpad))
    return gst_faac_srcconnect (faac->srcpad,
				gst_pad_get_caps (GST_PAD_PEER (faac->srcpad)));

  /* else, that'll be done later */
  return GST_PAD_LINK_OK;
322 323 324
}

static GstPadLinkReturn
325 326
gst_faac_srcconnect (GstPad        *pad,
		     const GstCaps *caps)
327 328
{
  GstFaac *faac = GST_FAAC (gst_pad_get_parent (pad));
329
  gint n;
330 331 332 333 334 335 336

  if (!faac->handle ||
      (faac->samplerate == -1 || faac->channels == -1)) {
    return GST_PAD_LINK_DELAYED;
  }

  /* we do samplerate/channels ourselves */
337 338 339 340
  for (n = 0; n < gst_caps_get_size (caps); n++) {
    GstStructure *structure = gst_caps_get_structure (caps, n);
    gst_structure_remove_field (structure, "rate");
    gst_structure_remove_field (structure, "channels");
341 342 343 344
  }

  /* go through list */
  caps = gst_caps_normalize (caps);
345 346
  for (n = 0; n < gst_caps_get_size (caps); n++) {
    GstStructure *structure = gst_caps_get_structure (caps, n);
347 348 349 350 351
    faacEncConfiguration *conf;
    gint mpegversion = 0;
    GstCaps *newcaps;
    GstPadLinkReturn ret;

352
    gst_structure_get_int (structure, "mpegversion", &mpegversion);
353 354 355 356 357 358 359 360 361 362 363 364 365 366 367 368 369 370 371 372 373 374 375 376 377 378

    /* new conf */
    conf = faacEncGetCurrentConfiguration (faac->handle);
    conf->mpegVersion = (mpegversion == 4) ? MPEG4 : MPEG2;
    conf->aacObjectType = faac->profile;
    conf->allowMidside = faac->midside;
    conf->useLfe = 0;
    conf->useTns = faac->tns;
    conf->bitRate = faac->bitrate;
    conf->inputFormat = faac->format;

    /* FIXME: this one here means that we do not support direct
     * "MPEG audio file" output (like mp3). This means we can
     * only mux this into mov/qt (mp4a) or matroska or so. If
     * we want to support direct AAC file output, we need ADTS
     * headers, and we need to find a way in the caps to detect
     * that (that the next element is filesink or any element
     * that does want ADTS headers). */

    conf->outputFormat = 0; /* raw, no ADTS headers */
    conf->shortctl = faac->shortctl;
    if (!faacEncSetConfiguration (faac->handle, conf)) {
      GST_WARNING ("Faac doesn't support the current conf");
      continue;
    }

379 380 381 382 383
    newcaps = gst_caps_new_simple ("audio/mpeg",
				   "mpegversion", G_TYPE_INT, mpegversion,
				   "channels",    G_TYPE_INT, faac->channels,
				   "rate",        G_TYPE_INT, faac->samplerate,
				   NULL);
384 385 386 387 388 389 390 391 392 393 394 395 396 397 398 399 400 401 402 403 404 405 406 407 408 409 410 411 412 413 414 415 416 417 418 419 420 421 422 423 424 425 426 427 428 429 430 431 432 433 434 435 436 437 438 439 440 441 442 443 444 445 446 447 448 449 450 451 452 453 454 455 456 457 458 459 460 461 462 463 464 465 466 467 468 469 470 471 472 473 474 475 476 477 478 479 480 481 482 483 484 485 486 487 488 489 490 491 492 493 494 495 496 497 498 499 500 501 502 503 504 505 506 507 508 509 510 511 512 513 514 515 516 517 518 519 520 521 522 523 524 525 526 527 528 529 530 531 532 533 534 535 536 537 538 539 540 541 542 543 544 545 546 547 548 549 550 551 552 553 554 555 556 557 558 559 560 561 562 563 564 565 566 567 568 569 570 571 572 573 574 575 576 577 578 579 580 581 582 583 584 585 586 587 588 589 590 591 592 593 594 595 596 597 598 599 600 601 602 603 604 605 606 607 608 609 610 611 612 613 614 615 616 617 618 619 620 621 622 623 624 625 626 627 628 629 630 631 632 633 634 635 636 637 638 639 640 641 642 643 644 645 646 647 648 649 650 651 652 653 654 655 656 657 658 659 660 661 662 663 664 665 666
    ret = gst_pad_try_set_caps (faac->srcpad, newcaps);

    switch (ret) {
      case GST_PAD_LINK_OK:
      case GST_PAD_LINK_DONE:
        return GST_PAD_LINK_DONE;
      case GST_PAD_LINK_DELAYED:
        return GST_PAD_LINK_DELAYED;
      default:
        break;
    }
  }

  return GST_PAD_LINK_REFUSED;
}

static void
gst_faac_chain (GstPad  *pad,
		GstData *data)
{
  GstFaac *faac = GST_FAAC (gst_pad_get_parent (pad));
  GstBuffer *inbuf, *outbuf, *subbuf;
  guint size, ret_size, in_size, frame_size;

  if (GST_IS_EVENT (data)) {
    GstEvent *event = GST_EVENT (data);

    switch (GST_EVENT_TYPE (event)) {
      case GST_EVENT_EOS:
        /* flush first */
        while (1) {
          outbuf = gst_buffer_new_and_alloc (faac->bytes);
          if ((ret_size = faacEncEncode (faac->handle,
					 NULL, 0,
					 GST_BUFFER_DATA (outbuf),
					 faac->bytes)) < 0) {
            gst_element_error (GST_ELEMENT (faac), "Error during AAC encoding");
            gst_event_unref (event);
            gst_buffer_unref (outbuf);
            return;
          }

          if (ret_size > 0) {
            GST_BUFFER_SIZE (outbuf) = ret_size;
            GST_BUFFER_TIMESTAMP (outbuf) = 0;
            GST_BUFFER_DURATION (outbuf) = 0;
            gst_pad_push (faac->srcpad, GST_DATA (outbuf));
          } else {
            break;
          }
        }

        gst_element_set_eos (GST_ELEMENT (faac));
        gst_pad_push (faac->srcpad, data);
        return;
      default:
	gst_pad_event_default (pad, event);
        return;
    }
  }

  inbuf = GST_BUFFER (data);

  if (!faac->handle) {
    gst_element_error (GST_ELEMENT (faac),
		       "No input format negotiated");
    gst_buffer_unref (inbuf);
    return;
  }

  if (!GST_PAD_CAPS (faac->srcpad)) {
    if (gst_faac_srcconnect (faac->srcpad,
			     gst_pad_get_allowed_caps (faac->srcpad)) <= 0) {
      gst_element_error (GST_ELEMENT (faac),
			 "Failed to negotiate MPEG/AAC format with next element");
      gst_buffer_unref (inbuf);
      return;
    }
  }

  size = GST_BUFFER_SIZE (inbuf);
  in_size = size;
  if (faac->cache)
    in_size += GST_BUFFER_SIZE (faac->cache);
  frame_size = faac->samples * faac->bps;

  while (1) {
    /* do we have enough data for one frame? */
    if (in_size / faac->bps < faac->samples) {
      if (in_size > size) {
        /* this is panic! we got a buffer, but still don't have enough
         * data. Merge them and retry in the next cycle... */
        faac->cache = gst_buffer_merge (faac->cache, inbuf);
      } else if (in_size == size) {
        /* this shouldn't happen, but still... */
        faac->cache = inbuf;
      } else if (in_size > 0) {
        faac->cache = gst_buffer_create_sub (inbuf, size - in_size,
					     in_size);
        GST_BUFFER_DURATION (faac->cache) =
	  GST_BUFFER_DURATION (inbuf) * GST_BUFFER_SIZE (faac->cache) / size;
        GST_BUFFER_TIMESTAMP (faac->cache) =
	  GST_BUFFER_TIMESTAMP (inbuf) + (GST_BUFFER_DURATION (inbuf) *
	    (size - in_size) / size);
        gst_buffer_unref (inbuf);
      } else {
        gst_buffer_unref (inbuf);
      }
          
      return;
    }

    /* create the frame */
    if (in_size > size) {
      /* merge */
      subbuf = gst_buffer_create_sub (inbuf, 0, frame_size - (in_size - size));
      GST_BUFFER_DURATION (subbuf) =
	GST_BUFFER_DURATION (inbuf) * GST_BUFFER_SIZE (subbuf) / size;
      subbuf = gst_buffer_merge (faac->cache, subbuf);
      faac->cache = NULL;
    } else {
      subbuf = gst_buffer_create_sub (inbuf, size - in_size, frame_size);
      GST_BUFFER_DURATION (subbuf) =
	GST_BUFFER_DURATION (inbuf) * GST_BUFFER_SIZE (subbuf) / size;
      GST_BUFFER_TIMESTAMP (subbuf) =
	GST_BUFFER_TIMESTAMP (inbuf) + (GST_BUFFER_DURATION (inbuf) *
	  (size - in_size) / size);
    }

    outbuf = gst_buffer_new_and_alloc (faac->bytes);
    if ((ret_size = faacEncEncode (faac->handle,
				   (gint32 *) GST_BUFFER_DATA (subbuf),
				   GST_BUFFER_SIZE (subbuf) / faac->bps,
				   GST_BUFFER_DATA (outbuf),
				   faac->bytes)) < 0) {
      gst_element_error (GST_ELEMENT (faac), "Error during AAC encoding");
      gst_buffer_unref (inbuf);
      gst_buffer_unref (subbuf);
      return;
    }

    if (ret_size > 0) {
      GST_BUFFER_SIZE (outbuf) = ret_size;
      if (faac->cache_time != GST_CLOCK_TIME_NONE) {
        GST_BUFFER_TIMESTAMP (outbuf) = faac->cache_time;
        faac->cache_time = GST_CLOCK_TIME_NONE;
      } else
        GST_BUFFER_TIMESTAMP (outbuf) = GST_BUFFER_TIMESTAMP (subbuf);
      GST_BUFFER_DURATION (outbuf) = GST_BUFFER_DURATION (subbuf);
      if (faac->cache_duration) {
        GST_BUFFER_DURATION (outbuf) += faac->cache_duration;
        faac->cache_duration = 0;
      }
      gst_pad_push (faac->srcpad, GST_DATA (outbuf));
    } else {
      /* FIXME: what I'm doing here isn't fully correct, but there
       * really isn't a better way yet.
       * Problem is that libfaac caches buffers (for encoding
       * purposes), so the timestamp of the outgoing buffer isn't
       * the same as the timestamp of the data that I pushed in.
       * However, I don't know the delay between those two so I
       * cannot really say aything about it. This is a bad guess. */

      gst_buffer_unref (outbuf);
      if (faac->cache_time != GST_CLOCK_TIME_NONE)
        faac->cache_time = GST_BUFFER_TIMESTAMP (subbuf);
      faac->cache_duration += GST_BUFFER_DURATION (subbuf);
    }

    in_size -= frame_size;
    gst_buffer_unref (subbuf);
  }
}

static void
gst_faac_set_property (GObject      *object,
		       guint         prop_id, 
		       const GValue *value,
		       GParamSpec   *pspec)
{
  GstFaac *faac = GST_FAAC (object);

  switch (prop_id) {
    case ARG_BITRATE:
      faac->bitrate = g_value_get_int (value);
      break;
    case ARG_PROFILE:
      faac->profile = g_value_get_enum (value);
      break;
    case ARG_TNS:
      faac->tns = g_value_get_boolean (value);
      break;
    case ARG_MIDSIDE:
      faac->midside = g_value_get_boolean (value);
      break;
    case ARG_SHORTCTL:
      faac->shortctl = g_value_get_enum (value);
      break;
    default:
      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
      break;
  }
}

static void
gst_faac_get_property (GObject    *object,
		       guint       prop_id, 
		       GValue     *value,
		       GParamSpec *pspec)
{
  GstFaac *faac = GST_FAAC (object);

  switch (prop_id) {
    case ARG_BITRATE:
      g_value_set_int (value, faac->bitrate);
      break;
    case ARG_PROFILE:
      g_value_set_enum (value, faac->profile);
      break;
    case ARG_TNS:
      g_value_set_boolean (value, faac->tns);
      break;
    case ARG_MIDSIDE:
      g_value_set_boolean (value, faac->midside);
      break;
    case ARG_SHORTCTL:
      g_value_set_enum (value, faac->shortctl);
      break;
    default:
      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
      break;
  }
}

static GstElementStateReturn
gst_faac_change_state (GstElement *element)
{
  GstFaac *faac = GST_FAAC (element);

  switch (GST_STATE_TRANSITION (element)) {
    case GST_STATE_PAUSED_TO_READY:
      if (faac->handle) {
        faacEncClose (faac->handle);
        faac->handle = NULL;
      }
      if (faac->cache) {
        gst_buffer_unref (faac->cache);
        faac->cache = NULL;
      }
      faac->cache_time = GST_CLOCK_TIME_NONE;
      faac->cache_duration = 0;
      faac->samplerate = -1;
      faac->channels = -1;
      break;
    default:
      break;
  }

  if (GST_ELEMENT_CLASS (parent_class)->change_state)
    return GST_ELEMENT_CLASS (parent_class)->change_state (element);

  return GST_STATE_SUCCESS;
}

static gboolean
plugin_init (GstPlugin *plugin)
{
  return gst_element_register (plugin, "faac",
			       GST_RANK_NONE,
			       GST_TYPE_FAAC);
}

GST_PLUGIN_DEFINE (
  GST_VERSION_MAJOR,
  GST_VERSION_MINOR,
  "faac",
  "Free AAC Encoder (FAAC)",
  plugin_init,
  VERSION,
  "LGPL",
  GST_PACKAGE,
  GST_ORIGIN
)