gstbaseaudiosrc.c 24.1 KB
Newer Older
Wim Taymans's avatar
Wim Taymans committed
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22
/* GStreamer
 * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
 *                    2005 Wim Taymans <wim@fluendo.com>
 *
 * gstbaseaudiosrc.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., 59 Temple Place - Suite 330,
 * Boston, MA 02111-1307, USA.
 */

Wim Taymans's avatar
Wim Taymans committed
23 24 25 26 27 28 29 30 31 32 33 34
/**
 * SECTION:gstbaseaudiosrc
 * @short_description: Base class for audio sources
 * @see_also: #GstAudioSrc, #GstRingBuffer.
 *
 * This is the base class for audio sources. Subclasses need to implement the
 * ::create_ringbuffer vmethod. This base class will then take care of
 * reading samples from the ringbuffer, synchronisation and flushing.
 *
 * Last reviewed on 2006-09-27 (0.10.12)
 */

35 36 37 38
#ifdef HAVE_CONFIG_H
#  include "config.h"
#endif

Wim Taymans's avatar
Wim Taymans committed
39 40 41 42
#include <string.h>

#include "gstbaseaudiosrc.h"

43 44
#include "gst/gst-i18n-plugin.h"

45 46
GST_DEBUG_CATEGORY_STATIC (gst_base_audio_src_debug);
#define GST_CAT_DEFAULT gst_base_audio_src_debug
Wim Taymans's avatar
Wim Taymans committed
47

48 49 50 51 52 53 54 55
#define GST_BASE_AUDIO_SRC_GET_PRIVATE(obj)  \
   (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_BASE_AUDIO_SRC, GstBaseAudioSrcPrivate))

struct _GstBaseAudioSrcPrivate
{
  gboolean provide_clock;
};

Wim Taymans's avatar
Wim Taymans committed
56 57 58 59 60 61 62
/* BaseAudioSrc signals and args */
enum
{
  /* FILL ME */
  LAST_SIGNAL
};

63 64
#define DEFAULT_BUFFER_TIME     ((200 * GST_MSECOND) / GST_USECOND)
#define DEFAULT_LATENCY_TIME    ((10 * GST_MSECOND) / GST_USECOND)
65
#define DEFAULT_PROVIDE_CLOCK   TRUE
66

Wim Taymans's avatar
Wim Taymans committed
67 68 69 70 71
enum
{
  PROP_0,
  PROP_BUFFER_TIME,
  PROP_LATENCY_TIME,
72
  PROP_PROVIDE_CLOCK
Wim Taymans's avatar
Wim Taymans committed
73 74
};

75 76 77 78 79 80 81 82 83 84 85 86
static void
_do_init (GType type)
{
  GST_DEBUG_CATEGORY_INIT (gst_base_audio_src_debug, "baseaudiosrc", 0,
      "baseaudiosrc element");

#ifdef ENABLE_NLS
  GST_DEBUG ("binding text domain %s to locale dir %s", GETTEXT_PACKAGE,
      LOCALEDIR);
  bindtextdomain (GETTEXT_PACKAGE, LOCALEDIR);
#endif /* ENABLE_NLS */
}
Wim Taymans's avatar
Wim Taymans committed
87

88
GST_BOILERPLATE_FULL (GstBaseAudioSrc, gst_base_audio_src, GstPushSrc,
89
    GST_TYPE_PUSH_SRC, _do_init);
Wim Taymans's avatar
Wim Taymans committed
90

91
static void gst_base_audio_src_set_property (GObject * object, guint prop_id,
Wim Taymans's avatar
Wim Taymans committed
92
    const GValue * value, GParamSpec * pspec);
93
static void gst_base_audio_src_get_property (GObject * object, guint prop_id,
Wim Taymans's avatar
Wim Taymans committed
94
    GValue * value, GParamSpec * pspec);
95
static void gst_base_audio_src_dispose (GObject * object);
Wim Taymans's avatar
Wim Taymans committed
96

97 98
static GstStateChangeReturn gst_base_audio_src_change_state (GstElement *
    element, GstStateChange transition);
Wim Taymans's avatar
Wim Taymans committed
99

100
static GstClock *gst_base_audio_src_provide_clock (GstElement * elem);
101
static GstClockTime gst_base_audio_src_get_time (GstClock * clock,
Wim Taymans's avatar
Wim Taymans committed
102 103
    GstBaseAudioSrc * src);

104 105 106
static GstFlowReturn gst_base_audio_src_create (GstBaseSrc * bsrc,
    guint64 offset, guint length, GstBuffer ** buf);
static gboolean gst_base_audio_src_check_get_range (GstBaseSrc * bsrc);
Wim Taymans's avatar
Wim Taymans committed
107

108 109
static gboolean gst_base_audio_src_event (GstBaseSrc * bsrc, GstEvent * event);
static void gst_base_audio_src_get_times (GstBaseSrc * bsrc,
Wim Taymans's avatar
Wim Taymans committed
110
    GstBuffer * buffer, GstClockTime * start, GstClockTime * end);
111
static gboolean gst_base_audio_src_setcaps (GstBaseSrc * bsrc, GstCaps * caps);
112
static gboolean gst_base_audio_src_query (GstBaseSrc * bsrc, GstQuery * query);
113
static void gst_base_audio_src_fixate (GstBaseSrc * bsrc, GstCaps * caps);
Wim Taymans's avatar
Wim Taymans committed
114

115
/* static guint gst_base_audio_src_signals[LAST_SIGNAL] = { 0 }; */
Wim Taymans's avatar
Wim Taymans committed
116 117

static void
118
gst_base_audio_src_base_init (gpointer g_class)
Wim Taymans's avatar
Wim Taymans committed
119 120 121 122
{
}

static void
123
gst_base_audio_src_class_init (GstBaseAudioSrcClass * klass)
Wim Taymans's avatar
Wim Taymans committed
124 125 126 127 128 129 130 131 132 133 134
{
  GObjectClass *gobject_class;
  GstElementClass *gstelement_class;
  GstBaseSrcClass *gstbasesrc_class;
  GstPushSrcClass *gstpushsrc_class;

  gobject_class = (GObjectClass *) klass;
  gstelement_class = (GstElementClass *) klass;
  gstbasesrc_class = (GstBaseSrcClass *) klass;
  gstpushsrc_class = (GstPushSrcClass *) klass;

135 136
  g_type_class_add_private (klass, sizeof (GstBaseAudioSrcPrivate));

Wim Taymans's avatar
Wim Taymans committed
137
  gobject_class->set_property =
138
      GST_DEBUG_FUNCPTR (gst_base_audio_src_set_property);
Wim Taymans's avatar
Wim Taymans committed
139
  gobject_class->get_property =
140
      GST_DEBUG_FUNCPTR (gst_base_audio_src_get_property);
141
  gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_base_audio_src_dispose);
Wim Taymans's avatar
Wim Taymans committed
142

143 144 145
  g_object_class_install_property (gobject_class, PROP_BUFFER_TIME,
      g_param_spec_int64 ("buffer-time", "Buffer Time",
          "Size of audio buffer in microseconds", 1,
146
          G_MAXINT64, DEFAULT_BUFFER_TIME, G_PARAM_READWRITE));
147 148 149 150

  g_object_class_install_property (gobject_class, PROP_LATENCY_TIME,
      g_param_spec_int64 ("latency-time", "Latency Time",
          "Audio latency in microseconds", 1,
151
          G_MAXINT64, DEFAULT_LATENCY_TIME, G_PARAM_READWRITE));
Wim Taymans's avatar
Wim Taymans committed
152

153 154 155 156 157
  g_object_class_install_property (gobject_class, PROP_PROVIDE_CLOCK,
      g_param_spec_boolean ("provide-clock", "Provide Clock",
          "Provide a clock to be used as the global pipeline clock",
          DEFAULT_PROVIDE_CLOCK, G_PARAM_READWRITE));

Wim Taymans's avatar
Wim Taymans committed
158
  gstelement_class->change_state =
159
      GST_DEBUG_FUNCPTR (gst_base_audio_src_change_state);
160 161
  gstelement_class->provide_clock =
      GST_DEBUG_FUNCPTR (gst_base_audio_src_provide_clock);
Wim Taymans's avatar
Wim Taymans committed
162

163 164
  gstbasesrc_class->set_caps = GST_DEBUG_FUNCPTR (gst_base_audio_src_setcaps);
  gstbasesrc_class->event = GST_DEBUG_FUNCPTR (gst_base_audio_src_event);
165
  gstbasesrc_class->query = GST_DEBUG_FUNCPTR (gst_base_audio_src_query);
166 167
  gstbasesrc_class->get_times =
      GST_DEBUG_FUNCPTR (gst_base_audio_src_get_times);
168 169 170
  gstbasesrc_class->create = GST_DEBUG_FUNCPTR (gst_base_audio_src_create);
  gstbasesrc_class->check_get_range =
      GST_DEBUG_FUNCPTR (gst_base_audio_src_check_get_range);
171
  gstbasesrc_class->fixate = GST_DEBUG_FUNCPTR (gst_base_audio_src_fixate);
Wim Taymans's avatar
Wim Taymans committed
172 173 174
}

static void
175 176
gst_base_audio_src_init (GstBaseAudioSrc * baseaudiosrc,
    GstBaseAudioSrcClass * g_class)
Wim Taymans's avatar
Wim Taymans committed
177
{
178 179
  baseaudiosrc->priv = GST_BASE_AUDIO_SRC_GET_PRIVATE (baseaudiosrc);

Wim Taymans's avatar
Wim Taymans committed
180 181
  baseaudiosrc->buffer_time = DEFAULT_BUFFER_TIME;
  baseaudiosrc->latency_time = DEFAULT_LATENCY_TIME;
182
  baseaudiosrc->priv->provide_clock = DEFAULT_PROVIDE_CLOCK;
183 184 185
  /* reset blocksize we use latency time to calculate a more useful 
   * value based on negotiated format. */
  GST_BASE_SRC (baseaudiosrc)->blocksize = 0;
Wim Taymans's avatar
Wim Taymans committed
186

187
  baseaudiosrc->clock = gst_audio_clock_new ("GstAudioSrcClock",
188
      (GstAudioClockGetTimeFunc) gst_base_audio_src_get_time, baseaudiosrc);
Wim Taymans's avatar
Wim Taymans committed
189

190
  /* we are always a live source */
191
  gst_base_src_set_live (GST_BASE_SRC (baseaudiosrc), TRUE);
Wim Taymans's avatar
Wim Taymans committed
192
  /* we operate in time */
193
  gst_base_src_set_format (GST_BASE_SRC (baseaudiosrc), GST_FORMAT_TIME);
Wim Taymans's avatar
Wim Taymans committed
194 195
}

196 197 198 199 200 201 202 203 204 205 206 207 208 209 210 211 212 213 214
static void
gst_base_audio_src_dispose (GObject * object)
{
  GstBaseAudioSrc *src;

  src = GST_BASE_AUDIO_SRC (object);

  if (src->clock)
    gst_object_unref (src->clock);
  src->clock = NULL;

  if (src->ringbuffer) {
    gst_object_unparent (GST_OBJECT_CAST (src->ringbuffer));
    src->ringbuffer = NULL;
  }

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

Wim Taymans's avatar
Wim Taymans committed
215
static GstClock *
216
gst_base_audio_src_provide_clock (GstElement * elem)
Wim Taymans's avatar
Wim Taymans committed
217 218
{
  GstBaseAudioSrc *src;
219
  GstClock *clock;
Wim Taymans's avatar
Wim Taymans committed
220

221
  src = GST_BASE_AUDIO_SRC (elem);
Wim Taymans's avatar
Wim Taymans committed
222

223 224 225 226 227 228 229
  /* we have no ringbuffer (must be NULL state) */
  if (src->ringbuffer == NULL)
    goto wrong_state;

  if (!gst_ring_buffer_is_acquired (src->ringbuffer))
    goto wrong_state;

230 231 232 233
  GST_OBJECT_LOCK (src);
  if (!src->priv->provide_clock)
    goto clock_disabled;

234
  clock = GST_CLOCK_CAST (gst_object_ref (src->clock));
235
  GST_OBJECT_UNLOCK (src);
236 237 238 239 240 241 242 243 244

  return clock;

  /* ERRORS */
wrong_state:
  {
    GST_DEBUG_OBJECT (src, "ringbuffer not acquired");
    return NULL;
  }
245 246 247 248 249 250
clock_disabled:
  {
    GST_DEBUG_OBJECT (src, "clock provide disabled");
    GST_OBJECT_UNLOCK (src);
    return NULL;
  }
Wim Taymans's avatar
Wim Taymans committed
251 252 253
}

static GstClockTime
254
gst_base_audio_src_get_time (GstClock * clock, GstBaseAudioSrc * src)
Wim Taymans's avatar
Wim Taymans committed
255
{
256 257
  guint64 raw, samples;
  guint delay;
Wim Taymans's avatar
Wim Taymans committed
258 259
  GstClockTime result;

260 261
  if (G_UNLIKELY (src->ringbuffer == NULL || src->ringbuffer->spec.rate == 0))
    return GST_CLOCK_TIME_NONE;
Wim Taymans's avatar
Wim Taymans committed
262

263 264 265 266 267 268 269
  raw = samples = gst_ring_buffer_samples_done (src->ringbuffer);

  /* the number of samples not yet processed, this is still queued in the
   * device (not yet read for capture). */
  delay = gst_ring_buffer_delay (src->ringbuffer);

  samples += delay;
Wim Taymans's avatar
Wim Taymans committed
270

271 272
  result = gst_util_uint64_scale_int (samples, GST_SECOND,
      src->ringbuffer->spec.rate);
Wim Taymans's avatar
Wim Taymans committed
273

274 275 276 277
  GST_DEBUG_OBJECT (src,
      "processed samples: raw %llu, delay %u, real %llu, time %"
      GST_TIME_FORMAT, raw, delay, samples, GST_TIME_ARGS (result));

Wim Taymans's avatar
Wim Taymans committed
278 279 280
  return result;
}

281 282 283 284 285 286 287 288 289 290
static gboolean
gst_base_audio_src_check_get_range (GstBaseSrc * bsrc)
{
  /* we allow limited pull base operation of which the details
   * will eventually exposed in an as of yet non-existing query.
   * Basically pulling can be done on any number of bytes as long
   * as the offset is -1 or sequentially increasing. */
  return TRUE;
}

291 292 293 294 295 296 297 298 299 300 301 302 303 304 305 306 307 308 309 310 311 312 313 314 315 316 317 318 319 320 321 322 323 324 325 326 327 328 329 330 331 332 333 334 335 336
/**
 * gst_base_audio_src_set_provide_clock:
 * @src: a #GstBaseAudioSrc
 * @provide: new state
 *
 * Controls whether @src will provide a clock or not. If @provide is %TRUE, 
 * gst_element_provide_clock() will return a clock that reflects the datarate
 * of @src. If @provide is %FALSE, gst_element_provide_clock() will return NULL.
 *
 * Since: 0.10.16
 */
void
gst_base_audio_src_set_provide_clock (GstBaseAudioSrc * src, gboolean provide)
{
  g_return_if_fail (GST_IS_BASE_AUDIO_SRC (src));

  GST_OBJECT_LOCK (src);
  src->priv->provide_clock = provide;
  GST_OBJECT_UNLOCK (src);
}

/**
 * gst_base_audio_src_get_provide_clock:
 * @src: a #GstBaseAudioSrc
 *
 * Queries whether @src will provide a clock or not. See also
 * gst_base_audio_src_set_provide_clock.
 *
 * Returns: %TRUE if @src will provide a clock.
 *
 * Since: 0.10.16
 */
gboolean
gst_base_audio_src_get_provide_clock (GstBaseAudioSrc * src)
{
  gboolean result;

  g_return_val_if_fail (GST_IS_BASE_AUDIO_SRC (src), FALSE);

  GST_OBJECT_LOCK (src);
  result = src->priv->provide_clock;
  GST_OBJECT_UNLOCK (src);

  return result;
}

Wim Taymans's avatar
Wim Taymans committed
337
static void
338
gst_base_audio_src_set_property (GObject * object, guint prop_id,
Wim Taymans's avatar
Wim Taymans committed
339 340 341 342
    const GValue * value, GParamSpec * pspec)
{
  GstBaseAudioSrc *src;

343
  src = GST_BASE_AUDIO_SRC (object);
Wim Taymans's avatar
Wim Taymans committed
344 345 346 347 348 349 350 351

  switch (prop_id) {
    case PROP_BUFFER_TIME:
      src->buffer_time = g_value_get_int64 (value);
      break;
    case PROP_LATENCY_TIME:
      src->latency_time = g_value_get_int64 (value);
      break;
352
    case PROP_PROVIDE_CLOCK:
353
      gst_base_audio_src_set_provide_clock (src, g_value_get_boolean (value));
354
      break;
Wim Taymans's avatar
Wim Taymans committed
355 356 357 358 359 360 361
    default:
      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
      break;
  }
}

static void
362 363
gst_base_audio_src_get_property (GObject * object, guint prop_id,
    GValue * value, GParamSpec * pspec)
Wim Taymans's avatar
Wim Taymans committed
364 365 366
{
  GstBaseAudioSrc *src;

367
  src = GST_BASE_AUDIO_SRC (object);
Wim Taymans's avatar
Wim Taymans committed
368 369 370 371 372 373 374 375

  switch (prop_id) {
    case PROP_BUFFER_TIME:
      g_value_set_int64 (value, src->buffer_time);
      break;
    case PROP_LATENCY_TIME:
      g_value_set_int64 (value, src->latency_time);
      break;
376
    case PROP_PROVIDE_CLOCK:
377
      g_value_set_boolean (value, gst_base_audio_src_get_provide_clock (src));
378
      break;
Wim Taymans's avatar
Wim Taymans committed
379 380 381 382 383 384 385
    default:
      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
      break;
  }
}

static void
386
gst_base_audio_src_fixate (GstBaseSrc * bsrc, GstCaps * caps)
Wim Taymans's avatar
Wim Taymans committed
387 388
{
  GstStructure *s;
389
  gint width, depth;
Wim Taymans's avatar
Wim Taymans committed
390 391 392

  s = gst_caps_get_structure (caps, 0);

393
  /* fields for all formats */
394 395 396
  gst_structure_fixate_field_nearest_int (s, "rate", 44100);
  gst_structure_fixate_field_nearest_int (s, "channels", 2);
  gst_structure_fixate_field_nearest_int (s, "width", 16);
397 398 399 400 401 402 403 404 405 406

  /* fields for int */
  if (gst_structure_has_field (s, "depth")) {
    gst_structure_get_int (s, "width", &width);
    /* round width to nearest multiple of 8 for the depth */
    depth = GST_ROUND_UP_8 (width);
    gst_structure_fixate_field_nearest_int (s, "depth", depth);
  }
  if (gst_structure_has_field (s, "signed"))
    gst_structure_fixate_field_boolean (s, "signed", TRUE);
407
  if (gst_structure_has_field (s, "endianness"))
408
    gst_structure_fixate_field_nearest_int (s, "endianness", G_BYTE_ORDER);
Wim Taymans's avatar
Wim Taymans committed
409 410 411
}

static gboolean
412
gst_base_audio_src_setcaps (GstBaseSrc * bsrc, GstCaps * caps)
Wim Taymans's avatar
Wim Taymans committed
413
{
414
  GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (bsrc);
Wim Taymans's avatar
Wim Taymans committed
415 416 417 418 419 420 421
  GstRingBufferSpec *spec;

  spec = &src->ringbuffer->spec;

  spec->buffer_time = src->buffer_time;
  spec->latency_time = src->latency_time;

422
  if (!gst_ring_buffer_parse_caps (spec, caps))
Wim Taymans's avatar
Wim Taymans committed
423 424 425 426 427 428 429 430 431
    goto parse_error;

  /* calculate suggested segsize and segtotal */
  spec->segsize =
      spec->rate * spec->bytes_per_sample * spec->latency_time / GST_MSECOND;
  spec->segtotal = spec->buffer_time / spec->latency_time;

  GST_DEBUG ("release old ringbuffer");

432
  gst_ring_buffer_release (src->ringbuffer);
Wim Taymans's avatar
Wim Taymans committed
433

434
  gst_ring_buffer_debug_spec_buff (spec);
Wim Taymans's avatar
Wim Taymans committed
435 436 437

  GST_DEBUG ("acquire new ringbuffer");

438
  if (!gst_ring_buffer_acquire (src->ringbuffer, spec))
Wim Taymans's avatar
Wim Taymans committed
439 440 441 442 443 444 445 446 447
    goto acquire_error;

  /* calculate actual latency and buffer times */
  spec->latency_time =
      spec->segsize * GST_MSECOND / (spec->rate * spec->bytes_per_sample);
  spec->buffer_time =
      spec->segtotal * spec->segsize * GST_MSECOND / (spec->rate *
      spec->bytes_per_sample);

448
  gst_ring_buffer_debug_spec_buff (spec);
Wim Taymans's avatar
Wim Taymans committed
449 450 451 452 453 454 455 456 457 458 459 460 461 462 463 464 465

  return TRUE;

  /* ERRORS */
parse_error:
  {
    GST_DEBUG ("could not parse caps");
    return FALSE;
  }
acquire_error:
  {
    GST_DEBUG ("could not acquire ringbuffer");
    return FALSE;
  }
}

static void
466
gst_base_audio_src_get_times (GstBaseSrc * bsrc, GstBuffer * buffer,
Wim Taymans's avatar
Wim Taymans committed
467 468
    GstClockTime * start, GstClockTime * end)
{
469 470
  /* no need to sync to a clock here, we schedule the samples based
   * on our own clock for the moment. */
Wim Taymans's avatar
Wim Taymans committed
471 472 473 474
  *start = GST_CLOCK_TIME_NONE;
  *end = GST_CLOCK_TIME_NONE;
}

475 476 477 478 479 480 481 482 483 484 485 486 487 488 489 490 491 492
static gboolean
gst_base_audio_src_query (GstBaseSrc * bsrc, GstQuery * query)
{
  GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (bsrc);
  gboolean res = FALSE;

  switch (GST_QUERY_TYPE (query)) {
    case GST_QUERY_LATENCY:
    {
      GstClockTime min_latency, max_latency;
      GstRingBufferSpec *spec;

      if (G_UNLIKELY (src->ringbuffer == NULL
              || src->ringbuffer->spec.rate == 0))
        goto done;

      spec = &src->ringbuffer->spec;

493
      /* we have at least 1 segment of latency */
494 495 496
      min_latency =
          gst_util_uint64_scale_int (spec->segsize, GST_SECOND,
          spec->rate * spec->bytes_per_sample);
497
      /* we cannot delay more than the buffersize else we lose data */
498 499 500 501 502 503 504 505 506 507 508 509 510 511 512 513 514 515 516 517 518 519 520
      max_latency =
          gst_util_uint64_scale_int (spec->segtotal * spec->segsize, GST_SECOND,
          spec->rate * spec->bytes_per_sample);

      GST_DEBUG_OBJECT (src,
          "report latency min %" GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
          GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));

      /* we are always live, the min latency is 1 segment and the max latency is
       * the complete buffer of segments. */
      gst_query_set_latency (query, TRUE, min_latency, max_latency);

      res = TRUE;
      break;
    }
    default:
      res = GST_BASE_SRC_CLASS (parent_class)->query (bsrc, query);
      break;
  }
done:
  return res;
}

Wim Taymans's avatar
Wim Taymans committed
521
static gboolean
522
gst_base_audio_src_event (GstBaseSrc * bsrc, GstEvent * event)
Wim Taymans's avatar
Wim Taymans committed
523
{
524
  GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (bsrc);
Wim Taymans's avatar
Wim Taymans committed
525 526

  switch (GST_EVENT_TYPE (event)) {
527 528
    case GST_EVENT_FLUSH_START:
      gst_ring_buffer_pause (src->ringbuffer);
529
      gst_ring_buffer_clear_all (src->ringbuffer);
530 531
      break;
    case GST_EVENT_FLUSH_STOP:
532 533 534
      /* always resync on sample after a flush */
      src->next_sample = -1;
      gst_ring_buffer_clear_all (src->ringbuffer);
Wim Taymans's avatar
Wim Taymans committed
535 536 537 538 539 540 541
      break;
    default:
      break;
  }
  return TRUE;
}

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
/* get the next offset in the ringbuffer for reading samples.
 * If the next sample is too far away, this function will position itself to the
 * next most recent sample, creating discontinuity */
static guint64
gst_base_audio_src_get_offset (GstBaseAudioSrc * src)
{
  guint64 sample;
  gint readseg, segdone, segtotal, sps;
  gint diff;

  /* assume we can append to the previous sample */
  sample = src->next_sample;
  /* no previous sample, try to read from position 0 */
  if (sample == -1)
    sample = 0;

  sps = src->ringbuffer->samples_per_seg;
  segtotal = src->ringbuffer->spec.segtotal;

  /* figure out the segment and the offset inside the segment where
   * the sample should be read from. */
  readseg = sample / sps;

  /* get the currently processed segment */
  segdone = g_atomic_int_get (&src->ringbuffer->segdone)
      - src->ringbuffer->segbase;

569 570
  GST_DEBUG_OBJECT (src, "reading from %d, we are at %d", readseg, segdone);

571 572 573 574 575
  /* see how far away it is from the read segment, normally segdone (where new
   * data is written in the ringbuffer) is bigger than readseg (where we are
   * reading). */
  diff = segdone - readseg;
  if (diff >= segtotal) {
576
    GST_DEBUG_OBJECT (src, "dropped, align to segment %d", segdone);
577 578 579 580 581 582 583
    /* sample would be dropped, position to next playable position */
    sample = (segdone - segtotal + 1) * sps;
  }

  return sample;
}

Wim Taymans's avatar
Wim Taymans committed
584
static GstFlowReturn
585 586
gst_base_audio_src_create (GstBaseSrc * bsrc, guint64 offset, guint length,
    GstBuffer ** outbuf)
Wim Taymans's avatar
Wim Taymans committed
587
{
588
  GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (bsrc);
Wim Taymans's avatar
Wim Taymans committed
589 590
  GstBuffer *buf;
  guchar *data;
591
  guint samples, total_samples;
592
  guint64 sample;
593
  gint bps;
594
  GstRingBuffer *ringbuffer;
595
  GstRingBufferSpec *spec;
596
  guint read;
Wim Taymans's avatar
Wim Taymans committed
597
  GstClockTime timestamp;
598
  GstClock *clock;
599 600

  ringbuffer = src->ringbuffer;
601
  spec = &ringbuffer->spec;
Wim Taymans's avatar
Wim Taymans committed
602

603
  if (G_UNLIKELY (!gst_ring_buffer_is_acquired (ringbuffer)))
Wim Taymans's avatar
Wim Taymans committed
604 605
    goto wrong_state;

606
  bps = spec->bytes_per_sample;
Wim Taymans's avatar
Wim Taymans committed
607

608 609
  if ((length == 0 && bsrc->blocksize == 0) || length == -1)
    /* no length given, use the default segment size */
610
    length = spec->segsize;
611 612 613
  else
    /* make sure we round down to an integral number of samples */
    length -= length % bps;
Wim Taymans's avatar
Wim Taymans committed
614

615
  /* figure out the offset in the ringbuffer */
616
  if (G_UNLIKELY (offset != -1)) {
617 618 619 620
    sample = offset / bps;
    /* if a specific offset was given it must be the next sequential
     * offset we expect or we fail for now. */
    if (src->next_sample != -1 && sample != src->next_sample)
621
      goto wrong_offset;
622 623 624 625
  } else {
    /* calculate the sequentially next sample we need to read. This can jump and
     * create a DISCONT. */
    sample = gst_base_audio_src_get_offset (src);
626 627
  }

Wim Taymans's avatar
Wim Taymans committed
628 629
  GST_DEBUG_OBJECT (src, "reading from sample %" G_GUINT64_FORMAT, sample);

630
  /* get the number of samples to read */
631
  total_samples = samples = length / bps;
632 633 634 635

  /* FIXME, using a bufferpool would be nice here */
  buf = gst_buffer_new_and_alloc (length);
  data = GST_BUFFER_DATA (buf);
Wim Taymans's avatar
Wim Taymans committed
636

637 638 639 640 641 642 643 644
  do {
    read = gst_ring_buffer_read (ringbuffer, sample, data, samples);
    GST_DEBUG_OBJECT (src, "read %u of %u", read, samples);
    /* if we read all, we're done */
    if (read == samples)
      break;

    /* else something interrupted us and we wait for playing again. */
645
    GST_DEBUG_OBJECT (src, "wait playing");
646 647 648
    if (gst_base_src_wait_playing (bsrc) != GST_FLOW_OK)
      goto stopped;

649 650
    GST_DEBUG_OBJECT (src, "continue playing");

651 652 653 654 655
    /* read next samples */
    sample += read;
    samples -= read;
    data += read * bps;
  } while (TRUE);
Wim Taymans's avatar
Wim Taymans committed
656

657 658 659 660 661
  /* mark discontinuity if needed */
  if (G_UNLIKELY (sample != src->next_sample) && src->next_sample != -1) {
    GST_WARNING_OBJECT (src,
        "create DISCONT of %" G_GUINT64_FORMAT " samples at sample %"
        G_GUINT64_FORMAT, sample - src->next_sample, sample);
662 663
    GST_ELEMENT_WARNING (src, CORE, CLOCK,
        (_("Can't record audio fast enough")),
664
        ("dropped %" G_GUINT64_FORMAT " samples", sample - src->next_sample));
665 666 667
    GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
  }

668 669
  GST_OBJECT_LOCK (src);
  clock = GST_ELEMENT_CLOCK (src);
670 671 672
  if (clock == NULL || clock == src->clock) {
    /* timestamp against our own clock. We do this also when no external clock
     * was provided to us. */
673 674 675 676
    timestamp = gst_util_uint64_scale_int (sample, GST_SECOND, spec->rate);
  } else {
    GstClockTime base_time, latency;

677 678 679
    /* We are slaved to another clock, take running time of the clock and just
     * timestamp against it. Somebody else in the pipeline should figure out the
     * clock drift, for now. */
680 681 682 683 684 685 686 687 688 689 690 691 692 693 694 695
    timestamp = gst_clock_get_time (clock);
    base_time = GST_ELEMENT_CAST (src)->base_time;

    if (timestamp > base_time)
      timestamp -= base_time;
    else
      timestamp = 0;

    /* subtract latency */
    latency = gst_util_uint64_scale_int (total_samples, GST_SECOND, spec->rate);
    if (timestamp > latency)
      timestamp -= latency;
    else
      timestamp = 0;
  }
  GST_OBJECT_UNLOCK (src);
Wim Taymans's avatar
Wim Taymans committed
696 697

  GST_BUFFER_TIMESTAMP (buf) = timestamp;
Wim Taymans's avatar
Wim Taymans committed
698
  src->next_sample = sample + samples;
699
  GST_BUFFER_DURATION (buf) = gst_util_uint64_scale_int (src->next_sample,
700
      GST_SECOND, spec->rate) - GST_BUFFER_TIMESTAMP (buf);
701 702
  GST_BUFFER_OFFSET (buf) = sample;
  GST_BUFFER_OFFSET_END (buf) = sample + samples;
703

704
  gst_buffer_set_caps (buf, GST_PAD_CAPS (GST_BASE_SRC_PAD (bsrc)));
Wim Taymans's avatar
Wim Taymans committed
705 706 707 708 709

  *outbuf = buf;

  return GST_FLOW_OK;

710
  /* ERRORS */
Wim Taymans's avatar
Wim Taymans committed
711 712
wrong_state:
  {
713
    GST_DEBUG_OBJECT (src, "ringbuffer in wrong state");
Wim Taymans's avatar
Wim Taymans committed
714 715
    return GST_FLOW_WRONG_STATE;
  }
716 717 718 719 720 721 722
wrong_offset:
  {
    GST_ELEMENT_ERROR (src, RESOURCE, SEEK,
        (NULL), ("resource can only be operated on sequentially but offset %"
            G_GUINT64_FORMAT " was given", offset));
    return GST_FLOW_ERROR;
  }
Wim Taymans's avatar
Wim Taymans committed
723 724 725
stopped:
  {
    gst_buffer_unref (buf);
726
    GST_DEBUG_OBJECT (src, "ringbuffer stopped");
Wim Taymans's avatar
Wim Taymans committed
727 728 729 730
    return GST_FLOW_WRONG_STATE;
  }
}

Wim Taymans's avatar
Wim Taymans committed
731 732 733 734 735 736 737 738 739 740
/**
 * gst_base_audio_src_create_ringbuffer:
 * @src: a #GstBaseAudioSrc.
 *
 * Create and return the #GstRingBuffer for @src. This function will call the
 * ::create_ringbuffer vmethod and will set @src as the parent of the returned
 * buffer (see gst_object_set_parent()).
 *
 * Returns: The new ringbuffer of @src.
 */
Wim Taymans's avatar
Wim Taymans committed
741
GstRingBuffer *
742
gst_base_audio_src_create_ringbuffer (GstBaseAudioSrc * src)
Wim Taymans's avatar
Wim Taymans committed
743 744 745 746
{
  GstBaseAudioSrcClass *bclass;
  GstRingBuffer *buffer = NULL;

747
  bclass = GST_BASE_AUDIO_SRC_GET_CLASS (src);
Wim Taymans's avatar
Wim Taymans committed
748 749 750
  if (bclass->create_ringbuffer)
    buffer = bclass->create_ringbuffer (src);

751 752
  if (G_LIKELY (buffer))
    gst_object_set_parent (GST_OBJECT_CAST (buffer), GST_OBJECT_CAST (src));
Wim Taymans's avatar
Wim Taymans committed
753 754 755 756

  return buffer;
}

757 758 759
static GstStateChangeReturn
gst_base_audio_src_change_state (GstElement * element,
    GstStateChange transition)
Wim Taymans's avatar
Wim Taymans committed
760
{
761
  GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
762
  GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (element);
Wim Taymans's avatar
Wim Taymans committed
763 764

  switch (transition) {
765
    case GST_STATE_CHANGE_NULL_TO_READY:
766
      GST_DEBUG_OBJECT (src, "NULL->READY");
767 768 769 770
      if (src->ringbuffer == NULL) {
        src->ringbuffer = gst_base_audio_src_create_ringbuffer (src);
      }
      if (!gst_ring_buffer_open_device (src->ringbuffer))
771
        goto open_failed;
Wim Taymans's avatar
Wim Taymans committed
772
      break;
773
    case GST_STATE_CHANGE_READY_TO_PAUSED:
774
      GST_DEBUG_OBJECT (src, "READY->PAUSED");
775
      src->next_sample = -1;
776
      gst_ring_buffer_set_flushing (src->ringbuffer, FALSE);
777
      gst_ring_buffer_may_start (src->ringbuffer, FALSE);
Wim Taymans's avatar
Wim Taymans committed
778
      break;
779
    case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
780
      GST_DEBUG_OBJECT (src, "PAUSED->PLAYING");
781
      gst_ring_buffer_may_start (src->ringbuffer, TRUE);
Wim Taymans's avatar
Wim Taymans committed
782
      break;
783 784 785 786 787
    case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
      GST_DEBUG_OBJECT (src, "PLAYING->PAUSED");
      gst_ring_buffer_may_start (src->ringbuffer, FALSE);
      gst_ring_buffer_pause (src->ringbuffer);
      break;
788
    case GST_STATE_CHANGE_PAUSED_TO_READY:
789
      GST_DEBUG_OBJECT (src, "PAUSED->READY");
790 791
      gst_ring_buffer_set_flushing (src->ringbuffer, TRUE);
      break;
Wim Taymans's avatar
Wim Taymans committed
792 793 794 795
    default:
      break;
  }

796
  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
Wim Taymans's avatar
Wim Taymans committed
797 798

  switch (transition) {
799
    case GST_STATE_CHANGE_PAUSED_TO_READY:
800
      GST_DEBUG_OBJECT (src, "PAUSED->READY");
801
      gst_ring_buffer_release (src->ringbuffer);
Wim Taymans's avatar
Wim Taymans committed
802
      break;
803
    case GST_STATE_CHANGE_READY_TO_NULL:
804
      GST_DEBUG_OBJECT (src, "READY->NULL");
805
      gst_ring_buffer_close_device (src->ringbuffer);
806
      gst_object_unparent (GST_OBJECT_CAST (src->ringbuffer));
807
      src->ringbuffer = NULL;
Wim Taymans's avatar
Wim Taymans committed
808 809 810 811 812 813
      break;
    default:
      break;
  }

  return ret;
814 815 816 817 818 819 820 821 822

  /* ERRORS */
open_failed:
  {
    /* subclass must post a meaningfull error message */
    GST_DEBUG_OBJECT (src, "open failed");
    return GST_STATE_CHANGE_FAILURE;
  }

Wim Taymans's avatar
Wim Taymans committed
823
}