2 * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
3 * 2005 Wim Taymans <wim@fluendo.com>
7 * This library is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Library General Public
9 * License as published by the Free Software Foundation; either
10 * version 2 of the License, or (at your option) any later version.
12 * This library is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Library General Public License for more details.
17 * You should have received a copy of the GNU Library General Public
18 * License along with this library; if not, write to the
19 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
20 * Boston, MA 02110-1301, USA.
24 * SECTION:gstaudiobasesink
25 * @short_description: Base class for audio sinks
26 * @see_also: #GstAudioSink, #GstAudioRingBuffer.
28 * This is the base class for audio sinks. Subclasses need to implement the
29 * ::create_ringbuffer vmethod. This base class will then take care of
30 * writing samples to the ringbuffer, synchronisation, clipping and flushing.
32 * Last reviewed on 2006-09-27 (0.10.12)
37 #include <gst/audio/audio.h>
38 #include "gstaudiobasesink.h"
40 GST_DEBUG_CATEGORY_STATIC (gst_audio_base_sink_debug);
41 #define GST_CAT_DEFAULT gst_audio_base_sink_debug
43 #define GST_AUDIO_BASE_SINK_GET_PRIVATE(obj) \
44 (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_AUDIO_BASE_SINK, GstAudioBaseSinkPrivate))
46 struct _GstAudioBaseSinkPrivate
48 /* upstream latency */
49 GstClockTime us_latency;
50 /* the clock slaving algorithm in use */
51 GstAudioBaseSinkSlaveMethod slave_method;
52 /* running average of clock skew */
53 GstClockTimeDiff avg_skew;
54 /* the number of samples we aligned last time */
57 gboolean sync_latency;
59 GstClockTime eos_time;
61 /* number of microseconds we allow clock slaving to drift
63 guint64 drift_tolerance;
65 /* number of nanoseconds we allow timestamps to drift
67 GstClockTime alignment_threshold;
69 /* time of the previous detected discont candidate */
70 GstClockTime discont_time;
72 /* number of nanoseconds to wait until creating a discontinuity */
73 GstClockTime discont_wait;
76 /* BaseAudioSink signals and args */
83 /* FIXME: 0.11, store the buffer_time and latency_time in nanoseconds */
84 #define DEFAULT_BUFFER_TIME ((200 * GST_MSECOND) / GST_USECOND)
85 #define DEFAULT_LATENCY_TIME ((10 * GST_MSECOND) / GST_USECOND)
86 #define DEFAULT_PROVIDE_CLOCK TRUE
87 #define DEFAULT_SLAVE_METHOD GST_AUDIO_BASE_SINK_SLAVE_SKEW
89 /* FIXME, enable pull mode when clock slaving and trick modes are figured out */
90 #define DEFAULT_CAN_ACTIVATE_PULL FALSE
92 /* when timestamps drift for more than 40ms we resync. This should
93 * be anough to compensate for timestamp rounding errors. */
94 #define DEFAULT_ALIGNMENT_THRESHOLD (40 * GST_MSECOND)
96 /* when clock slaving drift for more than 40ms we resync. This is
97 * a reasonable default */
98 #define DEFAULT_DRIFT_TOLERANCE ((40 * GST_MSECOND) / GST_USECOND)
100 /* allow for one second before resyncing to see if the timestamps drift will
101 * fix itself, or is a permanent offset */
102 #define DEFAULT_DISCONT_WAIT (1 * GST_SECOND)
112 PROP_CAN_ACTIVATE_PULL,
113 PROP_ALIGNMENT_THRESHOLD,
114 PROP_DRIFT_TOLERANCE,
121 gst_audio_base_sink_slave_method_get_type (void)
123 static volatile gsize slave_method_type = 0;
124 static const GEnumValue slave_method[] = {
125 {GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE, "GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE",
127 {GST_AUDIO_BASE_SINK_SLAVE_SKEW, "GST_AUDIO_BASE_SINK_SLAVE_SKEW", "skew"},
128 {GST_AUDIO_BASE_SINK_SLAVE_NONE, "GST_AUDIO_BASE_SINK_SLAVE_NONE", "none"},
132 if (g_once_init_enter (&slave_method_type)) {
134 g_enum_register_static ("GstAudioBaseSinkSlaveMethod", slave_method);
135 g_once_init_leave (&slave_method_type, tmp);
138 return (GType) slave_method_type;
143 GST_DEBUG_CATEGORY_INIT (gst_audio_base_sink_debug, "audiobasesink", 0, "audiobasesink element");
144 #define gst_audio_base_sink_parent_class parent_class
145 G_DEFINE_TYPE_WITH_CODE (GstAudioBaseSink, gst_audio_base_sink,
146 GST_TYPE_BASE_SINK, _do_init);
148 static void gst_audio_base_sink_dispose (GObject * object);
150 static void gst_audio_base_sink_set_property (GObject * object, guint prop_id,
151 const GValue * value, GParamSpec * pspec);
152 static void gst_audio_base_sink_get_property (GObject * object, guint prop_id,
153 GValue * value, GParamSpec * pspec);
156 static GstStateChangeReturn gst_audio_base_sink_async_play (GstBaseSink *
159 static GstStateChangeReturn gst_audio_base_sink_change_state (GstElement *
160 element, GstStateChange transition);
161 static gboolean gst_audio_base_sink_activate_pull (GstBaseSink * basesink,
163 static gboolean gst_audio_base_sink_query (GstElement * element, GstQuery *
166 static GstClock *gst_audio_base_sink_provide_clock (GstElement * elem);
167 static GstClockTime gst_audio_base_sink_get_time (GstClock * clock,
168 GstAudioBaseSink * sink);
169 static void gst_audio_base_sink_callback (GstAudioRingBuffer * rbuf,
170 guint8 * data, guint len, gpointer user_data);
172 static GstFlowReturn gst_audio_base_sink_preroll (GstBaseSink * bsink,
174 static GstFlowReturn gst_audio_base_sink_render (GstBaseSink * bsink,
176 static gboolean gst_audio_base_sink_event (GstBaseSink * bsink,
178 static GstFlowReturn gst_audio_base_sink_wait_event (GstBaseSink * bsink,
180 static void gst_audio_base_sink_get_times (GstBaseSink * bsink,
181 GstBuffer * buffer, GstClockTime * start, GstClockTime * end);
182 static gboolean gst_audio_base_sink_setcaps (GstBaseSink * bsink,
184 static GstCaps *gst_audio_base_sink_fixate (GstBaseSink * bsink,
187 static gboolean gst_audio_base_sink_query_pad (GstBaseSink * bsink,
191 /* static guint gst_audio_base_sink_signals[LAST_SIGNAL] = { 0 }; */
194 gst_audio_base_sink_class_init (GstAudioBaseSinkClass * klass)
196 GObjectClass *gobject_class;
197 GstElementClass *gstelement_class;
198 GstBaseSinkClass *gstbasesink_class;
200 gobject_class = (GObjectClass *) klass;
201 gstelement_class = (GstElementClass *) klass;
202 gstbasesink_class = (GstBaseSinkClass *) klass;
204 g_type_class_add_private (klass, sizeof (GstAudioBaseSinkPrivate));
206 gobject_class->set_property = gst_audio_base_sink_set_property;
207 gobject_class->get_property = gst_audio_base_sink_get_property;
208 gobject_class->dispose = gst_audio_base_sink_dispose;
210 g_object_class_install_property (gobject_class, PROP_BUFFER_TIME,
211 g_param_spec_int64 ("buffer-time", "Buffer Time",
212 "Size of audio buffer in microseconds, this is the minimum "
213 "latency that the sink reports", 1, G_MAXINT64, DEFAULT_BUFFER_TIME,
214 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
216 g_object_class_install_property (gobject_class, PROP_LATENCY_TIME,
217 g_param_spec_int64 ("latency-time", "Latency Time",
218 "The minimum amount of data to write in each iteration in microseconds",
219 1, G_MAXINT64, DEFAULT_LATENCY_TIME,
220 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
222 g_object_class_install_property (gobject_class, PROP_PROVIDE_CLOCK,
223 g_param_spec_boolean ("provide-clock", "Provide Clock",
224 "Provide a clock to be used as the global pipeline clock",
225 DEFAULT_PROVIDE_CLOCK, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
227 g_object_class_install_property (gobject_class, PROP_SLAVE_METHOD,
228 g_param_spec_enum ("slave-method", "Slave Method",
229 "Algorithm to use to match the rate of the masterclock",
230 GST_TYPE_AUDIO_BASE_SINK_SLAVE_METHOD, DEFAULT_SLAVE_METHOD,
231 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
233 g_object_class_install_property (gobject_class, PROP_CAN_ACTIVATE_PULL,
234 g_param_spec_boolean ("can-activate-pull", "Allow Pull Scheduling",
235 "Allow pull-based scheduling", DEFAULT_CAN_ACTIVATE_PULL,
236 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
238 * GstAudioBaseSink:drift-tolerance:
240 * Controls the amount of time in microseconds that clocks are allowed
241 * to drift before resynchronisation happens.
243 g_object_class_install_property (gobject_class, PROP_DRIFT_TOLERANCE,
244 g_param_spec_int64 ("drift-tolerance", "Drift Tolerance",
245 "Tolerance for clock drift in microseconds", 1,
246 G_MAXINT64, DEFAULT_DRIFT_TOLERANCE,
247 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
249 * GstAudioBaseSink:alignment_threshold:
251 * Controls the amount of time in nanoseconds that timestamps are allowed
252 * to drift from their ideal time before choosing not to align them.
254 g_object_class_install_property (gobject_class, PROP_ALIGNMENT_THRESHOLD,
255 g_param_spec_uint64 ("alignment-threshold", "Alignment Threshold",
256 "Timestamp alignment threshold in nanoseconds", 1,
257 G_MAXUINT64 - 1, DEFAULT_ALIGNMENT_THRESHOLD,
258 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
261 * GstAudioBaseSink:discont-wait:
263 * A window of time in nanoseconds to wait before creating a discontinuity as
264 * a result of breaching the drift-tolerance.
266 g_object_class_install_property (gobject_class, PROP_DISCONT_WAIT,
267 g_param_spec_uint64 ("discont-wait", "Discont Wait",
268 "Window of time in nanoseconds to wait before "
269 "creating a discontinuity", 0,
270 G_MAXUINT64 - 1, DEFAULT_DISCONT_WAIT,
271 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
273 gstelement_class->change_state =
274 GST_DEBUG_FUNCPTR (gst_audio_base_sink_change_state);
275 gstelement_class->provide_clock =
276 GST_DEBUG_FUNCPTR (gst_audio_base_sink_provide_clock);
277 gstelement_class->query = GST_DEBUG_FUNCPTR (gst_audio_base_sink_query);
279 gstbasesink_class->fixate = GST_DEBUG_FUNCPTR (gst_audio_base_sink_fixate);
280 gstbasesink_class->set_caps = GST_DEBUG_FUNCPTR (gst_audio_base_sink_setcaps);
281 gstbasesink_class->event = GST_DEBUG_FUNCPTR (gst_audio_base_sink_event);
282 gstbasesink_class->wait_event =
283 GST_DEBUG_FUNCPTR (gst_audio_base_sink_wait_event);
284 gstbasesink_class->get_times =
285 GST_DEBUG_FUNCPTR (gst_audio_base_sink_get_times);
286 gstbasesink_class->preroll = GST_DEBUG_FUNCPTR (gst_audio_base_sink_preroll);
287 gstbasesink_class->render = GST_DEBUG_FUNCPTR (gst_audio_base_sink_render);
288 gstbasesink_class->query = GST_DEBUG_FUNCPTR (gst_audio_base_sink_query_pad);
289 gstbasesink_class->activate_pull =
290 GST_DEBUG_FUNCPTR (gst_audio_base_sink_activate_pull);
292 /* ref class from a thread-safe context to work around missing bit of
293 * thread-safety in GObject */
294 g_type_class_ref (GST_TYPE_AUDIO_CLOCK);
295 g_type_class_ref (GST_TYPE_AUDIO_RING_BUFFER);
300 gst_audio_base_sink_init (GstAudioBaseSink * audiobasesink)
302 GstBaseSink *basesink;
304 audiobasesink->priv = GST_AUDIO_BASE_SINK_GET_PRIVATE (audiobasesink);
306 audiobasesink->buffer_time = DEFAULT_BUFFER_TIME;
307 audiobasesink->latency_time = DEFAULT_LATENCY_TIME;
308 audiobasesink->priv->slave_method = DEFAULT_SLAVE_METHOD;
309 audiobasesink->priv->drift_tolerance = DEFAULT_DRIFT_TOLERANCE;
310 audiobasesink->priv->alignment_threshold = DEFAULT_ALIGNMENT_THRESHOLD;
311 audiobasesink->priv->discont_wait = DEFAULT_DISCONT_WAIT;
313 audiobasesink->provided_clock = gst_audio_clock_new ("GstAudioSinkClock",
314 (GstAudioClockGetTimeFunc) gst_audio_base_sink_get_time, audiobasesink,
317 basesink = GST_BASE_SINK_CAST (audiobasesink);
318 basesink->can_activate_push = TRUE;
319 basesink->can_activate_pull = DEFAULT_CAN_ACTIVATE_PULL;
321 gst_base_sink_set_last_sample_enabled (basesink, FALSE);
322 if (DEFAULT_PROVIDE_CLOCK)
323 GST_OBJECT_FLAG_SET (basesink, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
325 GST_OBJECT_FLAG_UNSET (basesink, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
329 gst_audio_base_sink_dispose (GObject * object)
331 GstAudioBaseSink *sink;
333 sink = GST_AUDIO_BASE_SINK (object);
335 if (sink->provided_clock) {
336 gst_audio_clock_invalidate (sink->provided_clock);
337 gst_object_unref (sink->provided_clock);
338 sink->provided_clock = NULL;
341 if (sink->ringbuffer) {
342 gst_object_unparent (GST_OBJECT_CAST (sink->ringbuffer));
343 sink->ringbuffer = NULL;
346 G_OBJECT_CLASS (parent_class)->dispose (object);
351 gst_audio_base_sink_provide_clock (GstElement * elem)
353 GstAudioBaseSink *sink;
356 sink = GST_AUDIO_BASE_SINK (elem);
358 /* we have no ringbuffer (must be NULL state) */
359 if (sink->ringbuffer == NULL)
362 if (!gst_audio_ring_buffer_is_acquired (sink->ringbuffer))
365 GST_OBJECT_LOCK (sink);
366 if (!GST_OBJECT_FLAG_IS_SET (sink, GST_ELEMENT_FLAG_PROVIDE_CLOCK))
369 clock = GST_CLOCK_CAST (gst_object_ref (sink->provided_clock));
370 GST_OBJECT_UNLOCK (sink);
377 GST_DEBUG_OBJECT (sink, "ringbuffer not acquired");
382 GST_DEBUG_OBJECT (sink, "clock provide disabled");
383 GST_OBJECT_UNLOCK (sink);
389 gst_audio_base_sink_query_pad (GstBaseSink * bsink, GstQuery * query)
391 gboolean res = FALSE;
392 GstAudioBaseSink *basesink;
394 basesink = GST_AUDIO_BASE_SINK (bsink);
396 switch (GST_QUERY_TYPE (query)) {
397 case GST_QUERY_CONVERT:
399 GstFormat src_fmt, dest_fmt;
400 gint64 src_val, dest_val;
402 GST_LOG_OBJECT (basesink, "query convert");
404 if (basesink->ringbuffer) {
405 gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, NULL);
407 gst_audio_ring_buffer_convert (basesink->ringbuffer, src_fmt,
408 src_val, dest_fmt, &dest_val);
410 gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
416 res = GST_BASE_SINK_CLASS (parent_class)->query (bsink, query);
423 gst_audio_base_sink_query (GstElement * element, GstQuery * query)
425 gboolean res = FALSE;
426 GstAudioBaseSink *basesink;
428 basesink = GST_AUDIO_BASE_SINK (element);
430 switch (GST_QUERY_TYPE (query)) {
431 case GST_QUERY_LATENCY:
433 gboolean live, us_live;
434 GstClockTime min_l, max_l;
436 GST_DEBUG_OBJECT (basesink, "latency query");
438 /* ask parent first, it will do an upstream query for us. */
440 gst_base_sink_query_latency (GST_BASE_SINK_CAST (basesink), &live,
441 &us_live, &min_l, &max_l))) {
442 GstClockTime base_latency, min_latency, max_latency;
444 /* we and upstream are both live, adjust the min_latency */
445 if (live && us_live) {
446 GstAudioRingBufferSpec *spec;
448 GST_OBJECT_LOCK (basesink);
449 if (!basesink->ringbuffer || !basesink->ringbuffer->spec.info.rate) {
450 GST_OBJECT_UNLOCK (basesink);
452 GST_DEBUG_OBJECT (basesink,
453 "we are not yet negotiated, can't report latency yet");
457 spec = &basesink->ringbuffer->spec;
459 basesink->priv->us_latency = min_l;
462 gst_util_uint64_scale_int (spec->seglatency * spec->segsize,
463 GST_SECOND, spec->info.rate * spec->info.bpf);
464 GST_OBJECT_UNLOCK (basesink);
466 /* we cannot go lower than the buffer size and the min peer latency */
467 min_latency = base_latency + min_l;
468 /* the max latency is the max of the peer, we can delay an infinite
470 max_latency = (max_l == -1) ? -1 : (base_latency + max_l);
472 GST_DEBUG_OBJECT (basesink,
473 "peer min %" GST_TIME_FORMAT ", our min latency: %"
474 GST_TIME_FORMAT, GST_TIME_ARGS (min_l),
475 GST_TIME_ARGS (min_latency));
476 GST_DEBUG_OBJECT (basesink,
477 "peer max %" GST_TIME_FORMAT ", our max latency: %"
478 GST_TIME_FORMAT, GST_TIME_ARGS (max_l),
479 GST_TIME_ARGS (max_latency));
481 GST_DEBUG_OBJECT (basesink,
482 "peer or we are not live, don't care about latency");
486 gst_query_set_latency (query, live, min_latency, max_latency);
490 case GST_QUERY_CONVERT:
492 GstFormat src_fmt, dest_fmt;
493 gint64 src_val, dest_val;
495 GST_LOG_OBJECT (basesink, "query convert");
497 if (basesink->ringbuffer) {
498 gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, NULL);
500 gst_audio_ring_buffer_convert (basesink->ringbuffer, src_fmt,
501 src_val, dest_fmt, &dest_val);
503 gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
509 res = GST_ELEMENT_CLASS (parent_class)->query (element, query);
519 gst_audio_base_sink_get_time (GstClock * clock, GstAudioBaseSink * sink)
521 guint64 raw, samples;
525 if (sink->ringbuffer == NULL || sink->ringbuffer->spec.info.rate == 0)
526 return GST_CLOCK_TIME_NONE;
528 /* our processed samples are always increasing */
529 raw = samples = gst_audio_ring_buffer_samples_done (sink->ringbuffer);
531 /* the number of samples not yet processed, this is still queued in the
532 * device (not played for playback). */
533 delay = gst_audio_ring_buffer_delay (sink->ringbuffer);
535 if (G_LIKELY (samples >= delay))
540 result = gst_util_uint64_scale_int (samples, GST_SECOND,
541 sink->ringbuffer->spec.info.rate);
543 GST_DEBUG_OBJECT (sink,
544 "processed samples: raw %" G_GUINT64_FORMAT ", delay %u, real %"
545 G_GUINT64_FORMAT ", time %" GST_TIME_FORMAT,
546 raw, delay, samples, GST_TIME_ARGS (result));
552 * gst_audio_base_sink_set_provide_clock:
553 * @sink: a #GstAudioBaseSink
554 * @provide: new state
556 * Controls whether @sink will provide a clock or not. If @provide is %TRUE,
557 * gst_element_provide_clock() will return a clock that reflects the datarate
558 * of @sink. If @provide is %FALSE, gst_element_provide_clock() will return NULL.
561 gst_audio_base_sink_set_provide_clock (GstAudioBaseSink * sink,
564 g_return_if_fail (GST_IS_AUDIO_BASE_SINK (sink));
566 GST_OBJECT_LOCK (sink);
568 GST_OBJECT_FLAG_SET (sink, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
570 GST_OBJECT_FLAG_UNSET (sink, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
571 GST_OBJECT_UNLOCK (sink);
575 * gst_audio_base_sink_get_provide_clock:
576 * @sink: a #GstAudioBaseSink
578 * Queries whether @sink will provide a clock or not. See also
579 * gst_audio_base_sink_set_provide_clock.
581 * Returns: %TRUE if @sink will provide a clock.
584 gst_audio_base_sink_get_provide_clock (GstAudioBaseSink * sink)
588 g_return_val_if_fail (GST_IS_AUDIO_BASE_SINK (sink), FALSE);
590 GST_OBJECT_LOCK (sink);
591 result = GST_OBJECT_FLAG_IS_SET (sink, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
592 GST_OBJECT_UNLOCK (sink);
598 * gst_audio_base_sink_set_slave_method:
599 * @sink: a #GstAudioBaseSink
600 * @method: the new slave method
602 * Controls how clock slaving will be performed in @sink.
605 gst_audio_base_sink_set_slave_method (GstAudioBaseSink * sink,
606 GstAudioBaseSinkSlaveMethod method)
608 g_return_if_fail (GST_IS_AUDIO_BASE_SINK (sink));
610 GST_OBJECT_LOCK (sink);
611 sink->priv->slave_method = method;
612 GST_OBJECT_UNLOCK (sink);
616 * gst_audio_base_sink_get_slave_method:
617 * @sink: a #GstAudioBaseSink
619 * Get the current slave method used by @sink.
621 * Returns: The current slave method used by @sink.
623 GstAudioBaseSinkSlaveMethod
624 gst_audio_base_sink_get_slave_method (GstAudioBaseSink * sink)
626 GstAudioBaseSinkSlaveMethod result;
628 g_return_val_if_fail (GST_IS_AUDIO_BASE_SINK (sink), -1);
630 GST_OBJECT_LOCK (sink);
631 result = sink->priv->slave_method;
632 GST_OBJECT_UNLOCK (sink);
639 * gst_audio_base_sink_set_drift_tolerance:
640 * @sink: a #GstAudioBaseSink
641 * @drift_tolerance: the new drift tolerance in microseconds
643 * Controls the sink's drift tolerance.
646 gst_audio_base_sink_set_drift_tolerance (GstAudioBaseSink * sink,
647 gint64 drift_tolerance)
649 g_return_if_fail (GST_IS_AUDIO_BASE_SINK (sink));
651 GST_OBJECT_LOCK (sink);
652 sink->priv->drift_tolerance = drift_tolerance;
653 GST_OBJECT_UNLOCK (sink);
657 * gst_audio_base_sink_get_drift_tolerance:
658 * @sink: a #GstAudioBaseSink
660 * Get the current drift tolerance, in microseconds, used by @sink.
662 * Returns: The current drift tolerance used by @sink.
665 gst_audio_base_sink_get_drift_tolerance (GstAudioBaseSink * sink)
669 g_return_val_if_fail (GST_IS_AUDIO_BASE_SINK (sink), -1);
671 GST_OBJECT_LOCK (sink);
672 result = sink->priv->drift_tolerance;
673 GST_OBJECT_UNLOCK (sink);
679 * gst_audio_base_sink_set_alignment_threshold:
680 * @sink: a #GstAudioBaseSink
681 * @alignment_threshold: the new alignment threshold in nanoseconds
683 * Controls the sink's alignment threshold.
686 gst_audio_base_sink_set_alignment_threshold (GstAudioBaseSink * sink,
687 GstClockTime alignment_threshold)
689 g_return_if_fail (GST_IS_AUDIO_BASE_SINK (sink));
691 GST_OBJECT_LOCK (sink);
692 sink->priv->alignment_threshold = alignment_threshold;
693 GST_OBJECT_UNLOCK (sink);
697 * gst_audio_base_sink_get_alignment_threshold:
698 * @sink: a #GstAudioBaseSink
700 * Get the current alignment threshold, in nanoseconds, used by @sink.
702 * Returns: The current alignment threshold used by @sink.
705 gst_audio_base_sink_get_alignment_threshold (GstAudioBaseSink * sink)
709 g_return_val_if_fail (GST_IS_AUDIO_BASE_SINK (sink), GST_CLOCK_TIME_NONE);
711 GST_OBJECT_LOCK (sink);
712 result = sink->priv->alignment_threshold;
713 GST_OBJECT_UNLOCK (sink);
719 * gst_audio_base_sink_set_discont_wait:
720 * @sink: a #GstAudioBaseSink
721 * @discont_wait: the new discont wait in nanoseconds
723 * Controls how long the sink will wait before creating a discontinuity.
726 gst_audio_base_sink_set_discont_wait (GstAudioBaseSink * sink,
727 GstClockTime discont_wait)
729 g_return_if_fail (GST_IS_AUDIO_BASE_SINK (sink));
731 GST_OBJECT_LOCK (sink);
732 sink->priv->discont_wait = discont_wait;
733 GST_OBJECT_UNLOCK (sink);
737 * gst_audio_base_sink_get_discont_wait:
738 * @sink: a #GstAudioBaseSink
740 * Get the current discont wait, in nanoseconds, used by @sink.
742 * Returns: The current discont wait used by @sink.
745 gst_audio_base_sink_get_discont_wait (GstAudioBaseSink * sink)
749 g_return_val_if_fail (GST_IS_AUDIO_BASE_SINK (sink), -1);
751 GST_OBJECT_LOCK (sink);
752 result = sink->priv->discont_wait;
753 GST_OBJECT_UNLOCK (sink);
759 gst_audio_base_sink_set_property (GObject * object, guint prop_id,
760 const GValue * value, GParamSpec * pspec)
762 GstAudioBaseSink *sink;
764 sink = GST_AUDIO_BASE_SINK (object);
767 case PROP_BUFFER_TIME:
768 sink->buffer_time = g_value_get_int64 (value);
770 case PROP_LATENCY_TIME:
771 sink->latency_time = g_value_get_int64 (value);
773 case PROP_PROVIDE_CLOCK:
774 gst_audio_base_sink_set_provide_clock (sink, g_value_get_boolean (value));
776 case PROP_SLAVE_METHOD:
777 gst_audio_base_sink_set_slave_method (sink, g_value_get_enum (value));
779 case PROP_CAN_ACTIVATE_PULL:
780 GST_BASE_SINK (sink)->can_activate_pull = g_value_get_boolean (value);
782 case PROP_DRIFT_TOLERANCE:
783 gst_audio_base_sink_set_drift_tolerance (sink, g_value_get_int64 (value));
785 case PROP_ALIGNMENT_THRESHOLD:
786 gst_audio_base_sink_set_alignment_threshold (sink,
787 g_value_get_uint64 (value));
789 case PROP_DISCONT_WAIT:
790 gst_audio_base_sink_set_discont_wait (sink, g_value_get_uint64 (value));
793 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
799 gst_audio_base_sink_get_property (GObject * object, guint prop_id,
800 GValue * value, GParamSpec * pspec)
802 GstAudioBaseSink *sink;
804 sink = GST_AUDIO_BASE_SINK (object);
807 case PROP_BUFFER_TIME:
808 g_value_set_int64 (value, sink->buffer_time);
810 case PROP_LATENCY_TIME:
811 g_value_set_int64 (value, sink->latency_time);
813 case PROP_PROVIDE_CLOCK:
814 g_value_set_boolean (value, gst_audio_base_sink_get_provide_clock (sink));
816 case PROP_SLAVE_METHOD:
817 g_value_set_enum (value, gst_audio_base_sink_get_slave_method (sink));
819 case PROP_CAN_ACTIVATE_PULL:
820 g_value_set_boolean (value, GST_BASE_SINK (sink)->can_activate_pull);
822 case PROP_DRIFT_TOLERANCE:
823 g_value_set_int64 (value, gst_audio_base_sink_get_drift_tolerance (sink));
825 case PROP_ALIGNMENT_THRESHOLD:
826 g_value_set_uint64 (value,
827 gst_audio_base_sink_get_alignment_threshold (sink));
829 case PROP_DISCONT_WAIT:
830 g_value_set_uint64 (value, gst_audio_base_sink_get_discont_wait (sink));
833 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
839 gst_audio_base_sink_setcaps (GstBaseSink * bsink, GstCaps * caps)
841 GstAudioBaseSink *sink = GST_AUDIO_BASE_SINK (bsink);
842 GstAudioRingBufferSpec *spec;
844 GstClockTime crate_num, crate_denom;
846 if (!sink->ringbuffer)
849 spec = &sink->ringbuffer->spec;
851 if (G_UNLIKELY (spec->caps && gst_caps_is_equal (spec->caps, caps))) {
852 GST_DEBUG_OBJECT (sink,
853 "Ringbuffer caps haven't changed, skipping reconfiguration");
857 GST_DEBUG_OBJECT (sink, "release old ringbuffer");
859 /* get current time, updates the last_time. When the subclass has a clock that
860 * restarts from 0 when a new format is negotiated, it will call
861 * gst_audio_clock_reset() which will use this last_time to create an offset
862 * so that time from the clock keeps on increasing monotonically. */
863 now = gst_clock_get_time (sink->provided_clock);
865 GST_DEBUG_OBJECT (sink, "time was %" GST_TIME_FORMAT, GST_TIME_ARGS (now));
867 /* release old ringbuffer */
868 gst_audio_ring_buffer_pause (sink->ringbuffer);
869 gst_audio_ring_buffer_activate (sink->ringbuffer, FALSE);
870 gst_audio_ring_buffer_release (sink->ringbuffer);
872 GST_DEBUG_OBJECT (sink, "parse caps");
874 spec->buffer_time = sink->buffer_time;
875 spec->latency_time = sink->latency_time;
878 if (!gst_audio_ring_buffer_parse_caps (spec, caps))
881 gst_audio_ring_buffer_debug_spec_buff (spec);
883 GST_DEBUG_OBJECT (sink, "acquire ringbuffer");
884 if (!gst_audio_ring_buffer_acquire (sink->ringbuffer, spec))
887 /* We need to resync since the ringbuffer restarted */
888 sink->priv->avg_skew = -1;
889 sink->next_sample = -1;
890 sink->priv->eos_time = -1;
891 sink->priv->discont_time = -1;
893 if (bsink->pad_mode == GST_PAD_MODE_PUSH) {
894 GST_DEBUG_OBJECT (sink, "activate ringbuffer");
895 gst_audio_ring_buffer_activate (sink->ringbuffer, TRUE);
898 /* due to possible changes in the spec file we should recalibrate the clock */
899 gst_clock_get_calibration (sink->provided_clock, NULL, NULL,
900 &crate_num, &crate_denom);
901 gst_clock_set_calibration (sink->provided_clock,
902 gst_clock_get_internal_time (sink->provided_clock), now, crate_num,
905 /* calculate actual latency and buffer times.
906 * FIXME: In 0.11, store the latency_time internally in ns */
907 spec->latency_time = gst_util_uint64_scale (spec->segsize,
908 (GST_SECOND / GST_USECOND), spec->info.rate * spec->info.bpf);
910 spec->buffer_time = spec->segtotal * spec->latency_time;
912 gst_audio_ring_buffer_debug_spec_buff (spec);
919 GST_DEBUG_OBJECT (sink, "could not parse caps");
920 GST_ELEMENT_ERROR (sink, STREAM, FORMAT,
921 (NULL), ("cannot parse audio format."));
926 GST_DEBUG_OBJECT (sink, "could not acquire ringbuffer");
932 gst_audio_base_sink_fixate (GstBaseSink * bsink, GstCaps * caps)
937 caps = gst_caps_make_writable (caps);
939 s = gst_caps_get_structure (caps, 0);
941 /* fields for all formats */
942 gst_structure_fixate_field_nearest_int (s, "rate", 44100);
943 gst_structure_fixate_field_nearest_int (s, "channels", 2);
944 gst_structure_fixate_field_nearest_int (s, "width", 16);
947 if (gst_structure_has_field (s, "depth")) {
948 gst_structure_get_int (s, "width", &width);
949 /* round width to nearest multiple of 8 for the depth */
950 depth = GST_ROUND_UP_8 (width);
951 gst_structure_fixate_field_nearest_int (s, "depth", depth);
953 if (gst_structure_has_field (s, "signed"))
954 gst_structure_fixate_field_boolean (s, "signed", TRUE);
955 if (gst_structure_has_field (s, "endianness"))
956 gst_structure_fixate_field_nearest_int (s, "endianness", G_BYTE_ORDER);
958 caps = GST_BASE_SINK_CLASS (parent_class)->fixate (bsink, caps);
964 gst_audio_base_sink_get_times (GstBaseSink * bsink, GstBuffer * buffer,
965 GstClockTime * start, GstClockTime * end)
967 /* our clock sync is a bit too much for the base class to handle so
968 * we implement it ourselves. */
969 *start = GST_CLOCK_TIME_NONE;
970 *end = GST_CLOCK_TIME_NONE;
973 /* This waits for the drain to happen and can be canceled */
975 gst_audio_base_sink_drain (GstAudioBaseSink * sink)
977 if (!sink->ringbuffer)
979 if (!sink->ringbuffer->spec.info.rate)
982 /* if PLAYING is interrupted,
983 * arrange to have clock running when going to PLAYING again */
984 g_atomic_int_set (&sink->eos_rendering, 1);
986 /* need to start playback before we can drain, but only when
987 * we have successfully negotiated a format and thus acquired the
989 if (gst_audio_ring_buffer_is_acquired (sink->ringbuffer))
990 gst_audio_ring_buffer_start (sink->ringbuffer);
992 if (sink->priv->eos_time != -1) {
993 GST_DEBUG_OBJECT (sink,
994 "last sample time %" GST_TIME_FORMAT,
995 GST_TIME_ARGS (sink->priv->eos_time));
997 /* wait for the EOS time to be reached, this is the time when the last
998 * sample is played. */
999 gst_base_sink_wait (GST_BASE_SINK (sink), sink->priv->eos_time, NULL);
1001 GST_DEBUG_OBJECT (sink, "drained audio");
1003 g_atomic_int_set (&sink->eos_rendering, 0);
1007 static GstFlowReturn
1008 gst_audio_base_sink_wait_event (GstBaseSink * bsink, GstEvent * event)
1010 GstAudioBaseSink *sink = GST_AUDIO_BASE_SINK (bsink);
1013 ret = GST_BASE_SINK_CLASS (parent_class)->wait_event (bsink, event);
1014 if (ret != GST_FLOW_OK)
1017 switch (GST_EVENT_TYPE (event)) {
1018 case GST_EVENT_GAP:{
1019 GstClockTime timestamp, duration;
1020 GstAudioRingBufferSpec *spec;
1025 spec = &sink->ringbuffer->spec;
1026 gst_event_parse_gap (event, ×tamp, &duration);
1028 /* If the GAP event has a duration, handle it like a
1029 * silence buffer of that duration. Otherwise at least
1030 * start the ringbuffer to make sure the clock is running.
1032 if (duration != GST_CLOCK_TIME_NONE) {
1034 gst_util_uint64_scale_ceil (duration, spec->info.rate, GST_SECOND);
1035 buffer = gst_buffer_new_and_alloc (n_samples * spec->info.bpf);
1037 if (n_samples != 0) {
1038 gst_buffer_map (buffer, &minfo, GST_MAP_WRITE);
1039 gst_audio_format_fill_silence (spec->info.finfo, minfo.data,
1041 gst_buffer_unmap (buffer, &minfo);
1043 GST_BUFFER_PTS (buffer) = timestamp;
1044 GST_BUFFER_DURATION (buffer) = duration;
1045 GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_GAP);
1047 ret = gst_audio_base_sink_render (bsink, buffer);
1048 gst_buffer_unref (buffer);
1050 gst_audio_base_sink_drain (sink);
1055 /* now wait till we played everything */
1056 gst_audio_base_sink_drain (sink);
1065 gst_audio_base_sink_event (GstBaseSink * bsink, GstEvent * event)
1067 GstAudioBaseSink *sink = GST_AUDIO_BASE_SINK (bsink);
1069 switch (GST_EVENT_TYPE (event)) {
1070 case GST_EVENT_FLUSH_START:
1071 if (sink->ringbuffer)
1072 gst_audio_ring_buffer_set_flushing (sink->ringbuffer, TRUE);
1074 case GST_EVENT_FLUSH_STOP:
1075 /* always resync on sample after a flush */
1076 sink->priv->avg_skew = -1;
1077 sink->next_sample = -1;
1078 sink->priv->eos_time = -1;
1079 sink->priv->discont_time = -1;
1080 if (sink->ringbuffer)
1081 gst_audio_ring_buffer_set_flushing (sink->ringbuffer, FALSE);
1086 return GST_BASE_SINK_CLASS (parent_class)->event (bsink, event);
1089 static GstFlowReturn
1090 gst_audio_base_sink_preroll (GstBaseSink * bsink, GstBuffer * buffer)
1092 GstAudioBaseSink *sink = GST_AUDIO_BASE_SINK (bsink);
1094 if (!gst_audio_ring_buffer_is_acquired (sink->ringbuffer))
1097 /* we don't really do anything when prerolling. We could make a
1098 * property to play this buffer to have some sort of scrubbing
1104 GST_DEBUG_OBJECT (sink, "ringbuffer in wrong state");
1105 GST_ELEMENT_ERROR (sink, STREAM, FORMAT, (NULL), ("sink not negotiated."));
1106 return GST_FLOW_NOT_NEGOTIATED;
1111 gst_audio_base_sink_get_offset (GstAudioBaseSink * sink)
1114 gint writeseg, segdone, sps;
1117 /* assume we can append to the previous sample */
1118 sample = sink->next_sample;
1119 /* no previous sample, try to insert at position 0 */
1123 sps = sink->ringbuffer->samples_per_seg;
1125 /* figure out the segment and the offset inside the segment where
1126 * the sample should be written. */
1127 writeseg = sample / sps;
1129 /* get the currently processed segment */
1130 segdone = g_atomic_int_get (&sink->ringbuffer->segdone)
1131 - sink->ringbuffer->segbase;
1133 /* see how far away it is from the write segment */
1134 diff = writeseg - segdone;
1136 /* sample would be dropped, position to next playable position */
1137 sample = (segdone + 1) * sps;
1144 clock_convert_external (GstClockTime external, GstClockTime cinternal,
1145 GstClockTime cexternal, GstClockTime crate_num, GstClockTime crate_denom)
1147 /* adjust for rate and speed */
1148 if (external >= cexternal) {
1150 gst_util_uint64_scale (external - cexternal, crate_denom, crate_num);
1151 external += cinternal;
1154 gst_util_uint64_scale (cexternal - external, crate_denom, crate_num);
1155 if (cinternal > external)
1156 external = cinternal - external;
1163 /* algorithm to calculate sample positions that will result in resampling to
1164 * match the clock rate of the master */
1166 gst_audio_base_sink_resample_slaving (GstAudioBaseSink * sink,
1167 GstClockTime render_start, GstClockTime render_stop,
1168 GstClockTime * srender_start, GstClockTime * srender_stop)
1170 GstClockTime cinternal, cexternal;
1171 GstClockTime crate_num, crate_denom;
1173 /* FIXME, we can sample and add observations here or use the timeouts on the
1174 * clock. No idea which one is better or more stable. The timeout seems more
1175 * arbitrary but this one seems more demanding and does not work when there is
1176 * no data comming in to the sink. */
1178 GstClockTime etime, itime;
1181 /* sample clocks and figure out clock skew */
1182 etime = gst_clock_get_time (GST_ELEMENT_CLOCK (sink));
1183 itime = gst_audio_clock_get_time (sink->provided_clock);
1185 /* add new observation */
1186 gst_clock_add_observation (sink->provided_clock, itime, etime, &r_squared);
1189 /* get calibration parameters to compensate for speed and offset differences
1190 * when we are slaved */
1191 gst_clock_get_calibration (sink->provided_clock, &cinternal, &cexternal,
1192 &crate_num, &crate_denom);
1194 GST_DEBUG_OBJECT (sink, "internal %" GST_TIME_FORMAT " external %"
1195 GST_TIME_FORMAT " %" G_GUINT64_FORMAT "/%" G_GUINT64_FORMAT " = %f",
1196 GST_TIME_ARGS (cinternal), GST_TIME_ARGS (cexternal), crate_num,
1197 crate_denom, gst_guint64_to_gdouble (crate_num) /
1198 gst_guint64_to_gdouble (crate_denom));
1201 crate_denom = crate_num = 1;
1203 /* bring external time to internal time */
1204 render_start = clock_convert_external (render_start, cinternal, cexternal,
1205 crate_num, crate_denom);
1206 render_stop = clock_convert_external (render_stop, cinternal, cexternal,
1207 crate_num, crate_denom);
1209 GST_DEBUG_OBJECT (sink,
1210 "after slaving: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
1211 GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
1213 *srender_start = render_start;
1214 *srender_stop = render_stop;
1217 /* algorithm to calculate sample positions that will result in changing the
1218 * playout pointer to match the clock rate of the master */
1220 gst_audio_base_sink_skew_slaving (GstAudioBaseSink * sink,
1221 GstClockTime render_start, GstClockTime render_stop,
1222 GstClockTime * srender_start, GstClockTime * srender_stop)
1224 GstClockTime cinternal, cexternal, crate_num, crate_denom;
1225 GstClockTime etime, itime;
1226 GstClockTimeDiff skew, mdrift, mdrift2;
1230 /* get calibration parameters to compensate for offsets */
1231 gst_clock_get_calibration (sink->provided_clock, &cinternal, &cexternal,
1232 &crate_num, &crate_denom);
1234 /* sample clocks and figure out clock skew */
1235 etime = gst_clock_get_time (GST_ELEMENT_CLOCK (sink));
1236 itime = gst_audio_clock_get_time (sink->provided_clock);
1237 itime = gst_audio_clock_adjust (sink->provided_clock, itime);
1239 GST_DEBUG_OBJECT (sink,
1240 "internal %" GST_TIME_FORMAT " external %" GST_TIME_FORMAT
1241 " cinternal %" GST_TIME_FORMAT " cexternal %" GST_TIME_FORMAT,
1242 GST_TIME_ARGS (itime), GST_TIME_ARGS (etime),
1243 GST_TIME_ARGS (cinternal), GST_TIME_ARGS (cexternal));
1245 /* make sure we never go below 0 */
1246 etime = etime > cexternal ? etime - cexternal : 0;
1247 itime = itime > cinternal ? itime - cinternal : 0;
1249 /* do itime - etime.
1250 * positive value means external clock goes slower
1251 * negative value means external clock goes faster */
1252 skew = GST_CLOCK_DIFF (etime, itime);
1253 if (sink->priv->avg_skew == -1) {
1254 /* first observation */
1255 sink->priv->avg_skew = skew;
1257 /* next observations use a moving average */
1258 sink->priv->avg_skew = (31 * sink->priv->avg_skew + skew) / 32;
1261 GST_DEBUG_OBJECT (sink, "internal %" GST_TIME_FORMAT " external %"
1262 GST_TIME_FORMAT " skew %" G_GINT64_FORMAT " avg %" G_GINT64_FORMAT,
1263 GST_TIME_ARGS (itime), GST_TIME_ARGS (etime), skew, sink->priv->avg_skew);
1265 /* the max drift we allow */
1266 mdrift = sink->priv->drift_tolerance * 1000;
1267 mdrift2 = mdrift / 2;
1269 /* adjust playout pointer based on skew */
1270 if (sink->priv->avg_skew > mdrift2) {
1271 /* master is running slower, move internal time forward */
1272 GST_WARNING_OBJECT (sink,
1273 "correct clock skew %" G_GINT64_FORMAT " > %" G_GINT64_FORMAT,
1274 sink->priv->avg_skew, mdrift2);
1275 cexternal = cexternal > mdrift ? cexternal - mdrift : 0;
1276 sink->priv->avg_skew -= mdrift;
1278 driftsamples = (sink->ringbuffer->spec.info.rate * mdrift) / GST_SECOND;
1279 last_align = sink->priv->last_align;
1281 /* if we were aligning in the wrong direction or we aligned more than what we
1282 * will correct, resync */
1283 if (last_align < 0 || last_align > driftsamples)
1284 sink->next_sample = -1;
1286 GST_DEBUG_OBJECT (sink,
1287 "last_align %" G_GINT64_FORMAT " driftsamples %u, next %"
1288 G_GUINT64_FORMAT, last_align, driftsamples, sink->next_sample);
1290 gst_clock_set_calibration (sink->provided_clock, cinternal, cexternal,
1291 crate_num, crate_denom);
1292 } else if (sink->priv->avg_skew < -mdrift2) {
1293 /* master is running faster, move external time forwards */
1294 GST_WARNING_OBJECT (sink,
1295 "correct clock skew %" G_GINT64_FORMAT " < %" G_GINT64_FORMAT,
1296 sink->priv->avg_skew, -mdrift2);
1297 cexternal += mdrift;
1298 sink->priv->avg_skew += mdrift;
1300 driftsamples = (sink->ringbuffer->spec.info.rate * mdrift) / GST_SECOND;
1301 last_align = sink->priv->last_align;
1303 /* if we were aligning in the wrong direction or we aligned more than what we
1304 * will correct, resync */
1305 if (last_align > 0 || -last_align > driftsamples)
1306 sink->next_sample = -1;
1308 GST_DEBUG_OBJECT (sink,
1309 "last_align %" G_GINT64_FORMAT " driftsamples %u, next %"
1310 G_GUINT64_FORMAT, last_align, driftsamples, sink->next_sample);
1312 gst_clock_set_calibration (sink->provided_clock, cinternal, cexternal,
1313 crate_num, crate_denom);
1316 /* convert, ignoring speed */
1317 render_start = clock_convert_external (render_start, cinternal, cexternal,
1318 crate_num, crate_denom);
1319 render_stop = clock_convert_external (render_stop, cinternal, cexternal,
1320 crate_num, crate_denom);
1322 *srender_start = render_start;
1323 *srender_stop = render_stop;
1326 /* apply the clock offset but do no slaving otherwise */
1328 gst_audio_base_sink_none_slaving (GstAudioBaseSink * sink,
1329 GstClockTime render_start, GstClockTime render_stop,
1330 GstClockTime * srender_start, GstClockTime * srender_stop)
1332 GstClockTime cinternal, cexternal, crate_num, crate_denom;
1334 /* get calibration parameters to compensate for offsets */
1335 gst_clock_get_calibration (sink->provided_clock, &cinternal, &cexternal,
1336 &crate_num, &crate_denom);
1338 /* convert, ignoring speed */
1339 render_start = clock_convert_external (render_start, cinternal, cexternal,
1340 crate_num, crate_denom);
1341 render_stop = clock_convert_external (render_stop, cinternal, cexternal,
1342 crate_num, crate_denom);
1344 *srender_start = render_start;
1345 *srender_stop = render_stop;
1348 /* converts render_start and render_stop to their slaved values */
1350 gst_audio_base_sink_handle_slaving (GstAudioBaseSink * sink,
1351 GstClockTime render_start, GstClockTime render_stop,
1352 GstClockTime * srender_start, GstClockTime * srender_stop)
1354 switch (sink->priv->slave_method) {
1355 case GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE:
1356 gst_audio_base_sink_resample_slaving (sink, render_start, render_stop,
1357 srender_start, srender_stop);
1359 case GST_AUDIO_BASE_SINK_SLAVE_SKEW:
1360 gst_audio_base_sink_skew_slaving (sink, render_start, render_stop,
1361 srender_start, srender_stop);
1363 case GST_AUDIO_BASE_SINK_SLAVE_NONE:
1364 gst_audio_base_sink_none_slaving (sink, render_start, render_stop,
1365 srender_start, srender_stop);
1368 g_warning ("unknown slaving method %d", sink->priv->slave_method);
1373 /* must be called with LOCK */
1374 static GstFlowReturn
1375 gst_audio_base_sink_sync_latency (GstBaseSink * bsink, GstMiniObject * obj)
1378 GstClockReturn status;
1379 GstClockTime time, render_delay;
1381 GstAudioBaseSink *sink;
1382 GstClockTime itime, etime;
1383 GstClockTime rate_num, rate_denom;
1384 GstClockTimeDiff jitter;
1386 sink = GST_AUDIO_BASE_SINK (bsink);
1388 clock = GST_ELEMENT_CLOCK (sink);
1389 if (G_UNLIKELY (clock == NULL))
1392 /* we provided the global clock, don't need to do anything special */
1393 if (clock == sink->provided_clock)
1396 GST_OBJECT_UNLOCK (sink);
1399 GST_DEBUG_OBJECT (sink, "checking preroll");
1401 ret = gst_base_sink_do_preroll (bsink, obj);
1402 if (ret != GST_FLOW_OK)
1405 GST_OBJECT_LOCK (sink);
1406 time = sink->priv->us_latency;
1407 GST_OBJECT_UNLOCK (sink);
1409 /* Renderdelay is added onto our own latency, and needs
1410 * to be subtracted as well */
1411 render_delay = gst_base_sink_get_render_delay (bsink);
1413 if (G_LIKELY (time > render_delay))
1414 time -= render_delay;
1418 /* preroll done, we can sync since we are in PLAYING now. */
1419 GST_DEBUG_OBJECT (sink, "possibly waiting for clock to reach %"
1420 GST_TIME_FORMAT, GST_TIME_ARGS (time));
1422 /* wait for the clock, this can be interrupted because we got shut down or
1424 status = gst_base_sink_wait_clock (bsink, time, &jitter);
1426 GST_DEBUG_OBJECT (sink, "clock returned %d %" GST_TIME_FORMAT, status,
1427 GST_TIME_ARGS (jitter));
1429 /* invalid time, no clock or sync disabled, just continue then */
1430 if (status == GST_CLOCK_BADTIME)
1433 /* waiting could have been interrupted and we can be flushing now */
1434 if (G_UNLIKELY (bsink->flushing))
1437 /* retry if we got unscheduled, which means we did not reach the timeout
1438 * yet. if some other error occures, we continue. */
1439 } while (status == GST_CLOCK_UNSCHEDULED);
1441 GST_OBJECT_LOCK (sink);
1442 GST_DEBUG_OBJECT (sink, "latency synced");
1444 /* when we prerolled in time, we can accurately set the calibration,
1445 * our internal clock should exactly have been the latency (== the running
1446 * time of the external clock) */
1447 etime = GST_ELEMENT_CAST (sink)->base_time + time;
1448 itime = gst_audio_clock_get_time (sink->provided_clock);
1449 itime = gst_audio_clock_adjust (sink->provided_clock, itime);
1451 if (status == GST_CLOCK_EARLY) {
1452 /* when we prerolled late, we have to take into account the lateness */
1453 GST_DEBUG_OBJECT (sink, "late preroll, adding jitter");
1457 /* start ringbuffer so we can start slaving right away when we need to */
1458 gst_audio_ring_buffer_start (sink->ringbuffer);
1460 GST_DEBUG_OBJECT (sink,
1461 "internal time: %" GST_TIME_FORMAT " external time: %" GST_TIME_FORMAT,
1462 GST_TIME_ARGS (itime), GST_TIME_ARGS (etime));
1464 /* copy the original calibrated rate but update the internal and external
1466 gst_clock_get_calibration (sink->provided_clock, NULL, NULL, &rate_num,
1468 gst_clock_set_calibration (sink->provided_clock, itime, etime,
1469 rate_num, rate_denom);
1471 switch (sink->priv->slave_method) {
1472 case GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE:
1473 /* only set as master when we are resampling */
1474 GST_DEBUG_OBJECT (sink, "Setting clock as master");
1475 gst_clock_set_master (sink->provided_clock, clock);
1477 case GST_AUDIO_BASE_SINK_SLAVE_SKEW:
1478 case GST_AUDIO_BASE_SINK_SLAVE_NONE:
1483 sink->priv->avg_skew = -1;
1484 sink->next_sample = -1;
1485 sink->priv->eos_time = -1;
1486 sink->priv->discont_time = -1;
1493 GST_DEBUG_OBJECT (sink, "we have no clock");
1498 GST_DEBUG_OBJECT (sink, "we are not slaved");
1503 GST_DEBUG_OBJECT (sink, "we are flushing");
1504 GST_OBJECT_LOCK (sink);
1505 return GST_FLOW_FLUSHING;
1510 gst_audio_base_sink_get_alignment (GstAudioBaseSink * sink,
1511 GstClockTime sample_offset)
1513 GstAudioRingBuffer *ringbuf = sink->ringbuffer;
1516 gint64 max_sample_diff;
1517 gint segdone = g_atomic_int_get (&ringbuf->segdone) - ringbuf->segbase;
1518 gint64 samples_done = segdone * ringbuf->samples_per_seg;
1519 gint64 headroom = sample_offset - samples_done;
1520 gboolean allow_align = TRUE;
1521 gboolean discont = FALSE;
1524 /* now try to align the sample to the previous one, first see how big the
1526 if (sample_offset >= sink->next_sample)
1527 sample_diff = sample_offset - sink->next_sample;
1529 sample_diff = sink->next_sample - sample_offset;
1531 rate = GST_AUDIO_INFO_RATE (&ringbuf->spec.info);
1533 /* calculate the max allowed drift in units of samples. */
1534 max_sample_diff = gst_util_uint64_scale_int (sink->priv->alignment_threshold,
1537 /* calc align with previous sample */
1538 align = sink->next_sample - sample_offset;
1540 /* don't align if it means writing behind the read-segment */
1541 if (sample_diff > headroom && align < 0)
1542 allow_align = FALSE;
1544 if (G_UNLIKELY (sample_diff >= max_sample_diff)) {
1545 /* wait before deciding to make a discontinuity */
1546 if (sink->priv->discont_wait > 0) {
1547 GstClockTime time = gst_util_uint64_scale_int (sample_offset,
1549 if (sink->priv->discont_time == -1) {
1550 /* discont candidate */
1551 sink->priv->discont_time = time;
1552 } else if (time - sink->priv->discont_time >= sink->priv->discont_wait) {
1553 /* discont_wait expired, discontinuity detected */
1555 sink->priv->discont_time = -1;
1560 } else if (G_UNLIKELY (sink->priv->discont_time != -1)) {
1561 /* we have had a discont, but are now back on track! */
1562 sink->priv->discont_time = -1;
1565 if (G_LIKELY (!discont && allow_align)) {
1566 GST_DEBUG_OBJECT (sink,
1567 "align with prev sample, ABS (%" G_GINT64_FORMAT ") < %"
1568 G_GINT64_FORMAT, align, max_sample_diff);
1570 gint64 diff_s G_GNUC_UNUSED;
1572 /* calculate sample diff in seconds for error message */
1573 diff_s = gst_util_uint64_scale_int (sample_diff, GST_SECOND, rate);
1575 /* timestamps drifted apart from previous samples too much, we need to
1576 * resync. We log this as an element warning. */
1577 GST_WARNING_OBJECT (sink,
1578 "Unexpected discontinuity in audio timestamps of "
1579 "%s%" GST_TIME_FORMAT ", resyncing",
1580 sample_offset > sink->next_sample ? "+" : "-", GST_TIME_ARGS (diff_s));
1587 static GstFlowReturn
1588 gst_audio_base_sink_render (GstBaseSink * bsink, GstBuffer * buf)
1591 GstClockTime time, stop, render_start, render_stop, sample_offset;
1592 GstClockTimeDiff sync_offset, ts_offset;
1593 GstAudioBaseSinkClass *bclass;
1594 GstAudioBaseSink *sink;
1595 GstAudioRingBuffer *ringbuf;
1597 guint64 ctime, cstop;
1601 guint samples, written;
1605 GstClockTime base_time, render_delay, latency;
1607 gboolean sync, slaved, align_next;
1609 GstSegment clip_seg;
1611 GstBuffer *out = NULL;
1613 sink = GST_AUDIO_BASE_SINK (bsink);
1614 bclass = GST_AUDIO_BASE_SINK_GET_CLASS (sink);
1616 ringbuf = sink->ringbuffer;
1618 /* can't do anything when we don't have the device */
1619 if (G_UNLIKELY (!gst_audio_ring_buffer_is_acquired (ringbuf)))
1622 /* Wait for upstream latency before starting the ringbuffer, we do this so
1623 * that we can align the first sample of the ringbuffer to the base_time +
1625 GST_OBJECT_LOCK (sink);
1626 base_time = GST_ELEMENT_CAST (sink)->base_time;
1627 if (G_UNLIKELY (sink->priv->sync_latency)) {
1628 ret = gst_audio_base_sink_sync_latency (bsink, GST_MINI_OBJECT_CAST (buf));
1629 GST_OBJECT_UNLOCK (sink);
1630 if (G_UNLIKELY (ret != GST_FLOW_OK))
1631 goto sync_latency_failed;
1632 /* only do this once until we are set back to PLAYING */
1633 sink->priv->sync_latency = FALSE;
1635 GST_OBJECT_UNLOCK (sink);
1638 /* Before we go on, let's see if we need to payload the data. If yes, we also
1639 * need to unref the output buffer before leaving. */
1640 if (bclass->payload) {
1641 out = bclass->payload (sink, buf);
1644 goto payload_failed;
1649 bpf = GST_AUDIO_INFO_BPF (&ringbuf->spec.info);
1650 rate = GST_AUDIO_INFO_RATE (&ringbuf->spec.info);
1652 size = gst_buffer_get_size (buf);
1653 if (G_UNLIKELY (size % bpf) != 0)
1656 samples = size / bpf;
1657 out_samples = samples;
1659 in_offset = GST_BUFFER_OFFSET (buf);
1660 time = GST_BUFFER_TIMESTAMP (buf);
1662 GST_DEBUG_OBJECT (sink,
1663 "time %" GST_TIME_FORMAT ", offset %" G_GUINT64_FORMAT ", start %"
1664 GST_TIME_FORMAT ", samples %u", GST_TIME_ARGS (time), in_offset,
1665 GST_TIME_ARGS (bsink->segment.start), samples);
1669 /* if not valid timestamp or we can't clip or sync, try to play
1671 if (!GST_CLOCK_TIME_IS_VALID (time)) {
1672 render_start = gst_audio_base_sink_get_offset (sink);
1673 render_stop = render_start + samples;
1674 GST_DEBUG_OBJECT (sink, "Buffer of size %" G_GSIZE_FORMAT " has no time."
1675 " Using render_start=%" G_GUINT64_FORMAT, size, render_start);
1676 /* we don't have a start so we don't know stop either */
1681 /* let's calc stop based on the number of samples in the buffer instead
1682 * of trusting the DURATION */
1683 stop = time + gst_util_uint64_scale_int (samples, GST_SECOND, rate);
1685 /* prepare the clipping segment. Since we will be subtracting ts-offset and
1686 * device-delay later we scale the start and stop with those values so that we
1687 * can correctly clip them */
1688 clip_seg.format = GST_FORMAT_TIME;
1689 clip_seg.start = bsink->segment.start;
1690 clip_seg.stop = bsink->segment.stop;
1691 clip_seg.duration = -1;
1693 /* the sync offset is the combination of ts-offset and device-delay */
1694 latency = gst_base_sink_get_latency (bsink);
1695 ts_offset = gst_base_sink_get_ts_offset (bsink);
1696 render_delay = gst_base_sink_get_render_delay (bsink);
1697 sync_offset = ts_offset - render_delay + latency;
1699 GST_DEBUG_OBJECT (sink,
1700 "sync-offset %" G_GINT64_FORMAT ", render-delay %" GST_TIME_FORMAT
1701 ", ts-offset %" G_GINT64_FORMAT, sync_offset,
1702 GST_TIME_ARGS (render_delay), ts_offset);
1704 /* compensate for ts-offset and device-delay when negative we need to
1706 if (G_UNLIKELY (sync_offset < 0)) {
1707 clip_seg.start += -sync_offset;
1708 if (clip_seg.stop != -1)
1709 clip_seg.stop += -sync_offset;
1712 /* samples should be rendered based on their timestamp. All samples
1713 * arriving before the segment.start or after segment.stop are to be
1714 * thrown away. All samples should also be clipped to the segment
1716 if (G_UNLIKELY (!gst_segment_clip (&clip_seg, GST_FORMAT_TIME, time, stop,
1718 goto out_of_segment;
1720 /* see if some clipping happened */
1721 diff = ctime - time;
1722 if (G_UNLIKELY (diff > 0)) {
1723 /* bring clipped time to samples */
1724 diff = gst_util_uint64_scale_int (diff, rate, GST_SECOND);
1725 GST_DEBUG_OBJECT (sink, "clipping start to %" GST_TIME_FORMAT " %"
1726 G_GUINT64_FORMAT " samples", GST_TIME_ARGS (ctime), diff);
1728 offset += diff * bpf;
1731 diff = stop - cstop;
1732 if (G_UNLIKELY (diff > 0)) {
1733 /* bring clipped time to samples */
1734 diff = gst_util_uint64_scale_int (diff, rate, GST_SECOND);
1735 GST_DEBUG_OBJECT (sink, "clipping stop to %" GST_TIME_FORMAT " %"
1736 G_GUINT64_FORMAT " samples", GST_TIME_ARGS (cstop), diff);
1741 /* figure out how to sync */
1742 if (G_LIKELY ((clock = GST_ELEMENT_CLOCK (bsink))))
1747 if (G_UNLIKELY (!sync)) {
1748 /* no sync needed, play sample ASAP */
1749 render_start = gst_audio_base_sink_get_offset (sink);
1750 render_stop = render_start + samples;
1751 GST_DEBUG_OBJECT (sink,
1752 "no sync needed. Using render_start=%" G_GUINT64_FORMAT, render_start);
1756 /* bring buffer start and stop times to running time */
1758 gst_segment_to_running_time (&bsink->segment, GST_FORMAT_TIME, time);
1760 gst_segment_to_running_time (&bsink->segment, GST_FORMAT_TIME, stop);
1762 GST_DEBUG_OBJECT (sink,
1763 "running: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
1764 GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
1766 /* store the time of the last sample, we'll use this to perform sync on the
1767 * last sample when draining the buffer */
1768 if (G_LIKELY (bsink->segment.rate >= 0.0)) {
1769 sink->priv->eos_time = render_stop;
1771 sink->priv->eos_time = render_start;
1774 if (G_UNLIKELY (sync_offset != 0)) {
1775 /* compensate for ts-offset and delay we know this will not underflow because we
1777 GST_DEBUG_OBJECT (sink,
1778 "compensating for sync-offset %" GST_TIME_FORMAT,
1779 GST_TIME_ARGS (sync_offset));
1780 render_start += sync_offset;
1781 render_stop += sync_offset;
1784 if (base_time != 0) {
1785 GST_DEBUG_OBJECT (sink, "adding base_time %" GST_TIME_FORMAT,
1786 GST_TIME_ARGS (base_time));
1788 /* add base time to sync against the clock */
1789 render_start += base_time;
1790 render_stop += base_time;
1793 if (G_UNLIKELY ((slaved = (clock != sink->provided_clock)))) {
1794 /* handle clock slaving */
1795 gst_audio_base_sink_handle_slaving (sink, render_start, render_stop,
1796 &render_start, &render_stop);
1798 /* no slaving needed but we need to adapt to the clock calibration
1800 gst_audio_base_sink_none_slaving (sink, render_start, render_stop,
1801 &render_start, &render_stop);
1804 GST_DEBUG_OBJECT (sink,
1805 "final timestamps: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
1806 GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
1808 /* bring to position in the ringbuffer */
1809 time_offset = GST_AUDIO_CLOCK_CAST (sink->provided_clock)->time_offset;
1811 if (G_UNLIKELY (time_offset != 0)) {
1812 GST_DEBUG_OBJECT (sink,
1813 "apply time offset %" GST_TIME_FORMAT, GST_TIME_ARGS (time_offset));
1815 if (render_start > time_offset)
1816 render_start -= time_offset;
1819 if (render_stop > time_offset)
1820 render_stop -= time_offset;
1825 /* in some clock slaving cases, all late samples end up at 0 first,
1826 * and subsequent ones align with that until threshold exceeded,
1827 * and then sync back to 0 and so on, so avoid that altogether */
1828 if (G_UNLIKELY (render_start == 0 && render_stop == 0))
1831 /* and bring the time to the rate corrected offset in the buffer */
1832 render_start = gst_util_uint64_scale_int (render_start, rate, GST_SECOND);
1833 render_stop = gst_util_uint64_scale_int (render_stop, rate, GST_SECOND);
1835 /* positive playback rate, first sample is render_start, negative rate, first
1836 * sample is render_stop. When no rate conversion is active, render exactly
1837 * the amount of input samples to avoid aligning to rounding errors. */
1838 if (G_LIKELY (bsink->segment.rate >= 0.0)) {
1839 sample_offset = render_start;
1840 if (G_LIKELY (bsink->segment.rate == 1.0))
1841 render_stop = sample_offset + samples;
1843 sample_offset = render_stop;
1844 if (bsink->segment.rate == -1.0)
1845 render_start = sample_offset + samples;
1848 /* always resync after a discont */
1849 if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT))) {
1850 GST_DEBUG_OBJECT (sink, "resync after discont");
1854 /* resync when we don't know what to align the sample with */
1855 if (G_UNLIKELY (sink->next_sample == -1)) {
1856 GST_DEBUG_OBJECT (sink,
1857 "no align possible: no previous sample position known");
1861 align = gst_audio_base_sink_get_alignment (sink, sample_offset);
1862 sink->priv->last_align = align;
1864 /* apply alignment */
1865 render_start += align;
1867 /* only align stop if we are not slaved to resample */
1868 if (G_UNLIKELY (slaved
1869 && sink->priv->slave_method == GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE)) {
1870 GST_DEBUG_OBJECT (sink, "no stop time align needed: we are slaved");
1873 render_stop += align;
1876 /* number of target samples is difference between start and stop */
1877 out_samples = render_stop - render_start;
1879 /* we render the first or last sample first, depending on the rate */
1880 if (G_LIKELY (bsink->segment.rate >= 0.0))
1881 sample_offset = render_start;
1883 sample_offset = render_stop;
1885 GST_DEBUG_OBJECT (sink, "rendering at %" G_GUINT64_FORMAT " %d/%d",
1886 sample_offset, samples, out_samples);
1888 /* we need to accumulate over different runs for when we get interrupted */
1891 gst_buffer_map (buf, &info, GST_MAP_READ);
1894 gst_audio_ring_buffer_commit (ringbuf, &sample_offset,
1895 info.data + offset, samples, out_samples, &accum);
1897 GST_DEBUG_OBJECT (sink, "wrote %u of %u", written, samples);
1898 /* if we wrote all, we're done */
1899 if (G_LIKELY (written == samples))
1902 /* else something interrupted us and we wait for preroll. */
1903 if ((ret = gst_base_sink_wait_preroll (bsink)) != GST_FLOW_OK)
1906 /* if we got interrupted, we cannot assume that the next sample should
1907 * be aligned to this one */
1910 /* update the output samples. FIXME, this will just skip them when pausing
1911 * during trick mode */
1912 if (out_samples > written) {
1913 out_samples -= written;
1919 offset += written * bpf;
1921 gst_buffer_unmap (buf, &info);
1923 if (G_LIKELY (align_next))
1924 sink->next_sample = sample_offset;
1926 sink->next_sample = -1;
1928 GST_DEBUG_OBJECT (sink, "next sample expected at %" G_GUINT64_FORMAT,
1931 if (G_UNLIKELY (GST_CLOCK_TIME_IS_VALID (stop)
1932 && stop >= bsink->segment.stop)) {
1933 GST_DEBUG_OBJECT (sink,
1934 "start playback because we are at the end of segment");
1935 gst_audio_ring_buffer_start (ringbuf);
1942 gst_buffer_unref (out);
1949 GST_DEBUG_OBJECT (sink,
1950 "dropping sample out of segment time %" GST_TIME_FORMAT ", start %"
1951 GST_TIME_FORMAT, GST_TIME_ARGS (time),
1952 GST_TIME_ARGS (bsink->segment.start));
1958 GST_DEBUG_OBJECT (sink, "dropping late sample");
1965 GST_ELEMENT_ERROR (sink, STREAM, FORMAT, (NULL), ("failed to payload."));
1966 ret = GST_FLOW_ERROR;
1971 GST_DEBUG_OBJECT (sink, "ringbuffer not negotiated");
1972 GST_ELEMENT_ERROR (sink, STREAM, FORMAT, (NULL), ("sink not negotiated."));
1973 ret = GST_FLOW_NOT_NEGOTIATED;
1978 GST_DEBUG_OBJECT (sink, "wrong size");
1979 GST_ELEMENT_ERROR (sink, STREAM, WRONG_TYPE,
1980 (NULL), ("sink received buffer of wrong size."));
1981 ret = GST_FLOW_ERROR;
1986 GST_DEBUG_OBJECT (sink, "preroll got interrupted: %d (%s)", ret,
1987 gst_flow_get_name (ret));
1988 gst_buffer_unmap (buf, &info);
1991 sync_latency_failed:
1993 GST_DEBUG_OBJECT (sink, "failed waiting for latency");
1999 * gst_audio_base_sink_create_ringbuffer:
2000 * @sink: a #GstAudioBaseSink.
2002 * Create and return the #GstAudioRingBuffer for @sink. This function will call the
2003 * ::create_ringbuffer vmethod and will set @sink as the parent of the returned
2004 * buffer (see gst_object_set_parent()).
2006 * Returns: (transfer none): The new ringbuffer of @sink.
2008 GstAudioRingBuffer *
2009 gst_audio_base_sink_create_ringbuffer (GstAudioBaseSink * sink)
2011 GstAudioBaseSinkClass *bclass;
2012 GstAudioRingBuffer *buffer = NULL;
2014 bclass = GST_AUDIO_BASE_SINK_GET_CLASS (sink);
2015 if (bclass->create_ringbuffer)
2016 buffer = bclass->create_ringbuffer (sink);
2019 gst_object_set_parent (GST_OBJECT (buffer), GST_OBJECT (sink));
2025 gst_audio_base_sink_callback (GstAudioRingBuffer * rbuf, guint8 * data,
2026 guint len, gpointer user_data)
2028 GstBaseSink *basesink;
2029 GstAudioBaseSink *sink;
2030 GstBuffer *buf = NULL;
2034 basesink = GST_BASE_SINK (user_data);
2035 sink = GST_AUDIO_BASE_SINK (user_data);
2037 GST_PAD_STREAM_LOCK (basesink->sinkpad);
2039 /* would be nice to arrange for pad_alloc_buffer to return data -- as it is we
2040 will copy twice, once into data, once into DMA */
2041 GST_LOG_OBJECT (basesink, "pulling %u bytes offset %" G_GUINT64_FORMAT
2042 " to fill audio buffer", len, basesink->offset);
2044 gst_pad_pull_range (basesink->sinkpad, basesink->segment.position, len,
2047 if (ret != GST_FLOW_OK) {
2048 if (ret == GST_FLOW_EOS)
2054 GST_BASE_SINK_PREROLL_LOCK (basesink);
2055 if (basesink->flushing)
2058 /* complete preroll and wait for PLAYING */
2059 ret = gst_base_sink_do_preroll (basesink, GST_MINI_OBJECT_CAST (buf));
2060 if (ret != GST_FLOW_OK)
2063 size = gst_buffer_get_size (buf);
2066 GST_INFO_OBJECT (basesink,
2067 "got different size than requested from sink pad: %u"
2068 " != %" G_GSIZE_FORMAT, len, size);
2069 len = MIN (size, len);
2072 basesink->segment.position += len;
2074 gst_buffer_extract (buf, 0, data, len);
2075 GST_BASE_SINK_PREROLL_UNLOCK (basesink);
2077 GST_PAD_STREAM_UNLOCK (basesink->sinkpad);
2083 GST_WARNING_OBJECT (basesink, "Got flow '%s' but can't return it: %d",
2084 gst_flow_get_name (ret), ret);
2085 gst_audio_ring_buffer_pause (rbuf);
2086 GST_PAD_STREAM_UNLOCK (basesink->sinkpad);
2091 /* FIXME: this is not quite correct; we'll be called endlessly until
2092 * the sink gets shut down; maybe we should set a flag somewhere, or
2093 * set segment.stop and segment.duration to the last sample or so */
2094 GST_DEBUG_OBJECT (sink, "EOS");
2095 gst_audio_base_sink_drain (sink);
2096 gst_audio_ring_buffer_pause (rbuf);
2097 gst_element_post_message (GST_ELEMENT_CAST (sink),
2098 gst_message_new_eos (GST_OBJECT_CAST (sink)));
2099 GST_PAD_STREAM_UNLOCK (basesink->sinkpad);
2103 GST_DEBUG_OBJECT (sink, "we are flushing");
2104 gst_audio_ring_buffer_pause (rbuf);
2105 GST_BASE_SINK_PREROLL_UNLOCK (basesink);
2106 GST_PAD_STREAM_UNLOCK (basesink->sinkpad);
2111 GST_DEBUG_OBJECT (sink, "error %s", gst_flow_get_name (ret));
2112 gst_audio_ring_buffer_pause (rbuf);
2113 GST_BASE_SINK_PREROLL_UNLOCK (basesink);
2114 GST_PAD_STREAM_UNLOCK (basesink->sinkpad);
2120 gst_audio_base_sink_activate_pull (GstBaseSink * basesink, gboolean active)
2123 GstAudioBaseSink *sink = GST_AUDIO_BASE_SINK (basesink);
2126 GST_DEBUG_OBJECT (basesink, "activating pull");
2128 gst_audio_ring_buffer_set_callback (sink->ringbuffer,
2129 gst_audio_base_sink_callback, sink);
2131 ret = gst_audio_ring_buffer_activate (sink->ringbuffer, TRUE);
2133 GST_DEBUG_OBJECT (basesink, "deactivating pull");
2134 gst_audio_ring_buffer_set_callback (sink->ringbuffer, NULL, NULL);
2135 ret = gst_audio_ring_buffer_activate (sink->ringbuffer, FALSE);
2142 /* should be called with the LOCK */
2143 static GstStateChangeReturn
2144 gst_audio_base_sink_async_play (GstBaseSink * basesink)
2146 GstAudioBaseSink *sink;
2148 sink = GST_AUDIO_BASE_SINK (basesink);
2150 GST_DEBUG_OBJECT (sink, "ringbuffer may start now");
2151 sink->priv->sync_latency = TRUE;
2152 gst_audio_ring_buffer_may_start (sink->ringbuffer, TRUE);
2153 if (basesink->pad_mode == GST_PAD_MODE_PULL) {
2154 /* we always start the ringbuffer in pull mode immediatly */
2155 gst_audio_ring_buffer_start (sink->ringbuffer);
2158 return GST_STATE_CHANGE_SUCCESS;
2162 static GstStateChangeReturn
2163 gst_audio_base_sink_change_state (GstElement * element,
2164 GstStateChange transition)
2166 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
2167 GstAudioBaseSink *sink = GST_AUDIO_BASE_SINK (element);
2169 switch (transition) {
2170 case GST_STATE_CHANGE_NULL_TO_READY:
2171 if (sink->ringbuffer == NULL) {
2172 gst_audio_clock_reset (GST_AUDIO_CLOCK (sink->provided_clock), 0);
2173 sink->ringbuffer = gst_audio_base_sink_create_ringbuffer (sink);
2175 if (!gst_audio_ring_buffer_open_device (sink->ringbuffer))
2178 case GST_STATE_CHANGE_READY_TO_PAUSED:
2179 sink->next_sample = -1;
2180 sink->priv->last_align = -1;
2181 sink->priv->eos_time = -1;
2182 sink->priv->discont_time = -1;
2183 gst_audio_ring_buffer_set_flushing (sink->ringbuffer, FALSE);
2184 gst_audio_ring_buffer_may_start (sink->ringbuffer, FALSE);
2186 /* Only post clock-provide messages if this is the clock that
2187 * we've created. If the subclass has overriden it the subclass
2188 * should post this messages whenever necessary */
2189 if (sink->provided_clock && GST_IS_AUDIO_CLOCK (sink->provided_clock) &&
2190 GST_AUDIO_CLOCK_CAST (sink->provided_clock)->func ==
2191 (GstAudioClockGetTimeFunc) gst_audio_base_sink_get_time)
2192 gst_element_post_message (element,
2193 gst_message_new_clock_provide (GST_OBJECT_CAST (element),
2194 sink->provided_clock, TRUE));
2196 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
2200 GST_OBJECT_LOCK (sink);
2201 GST_DEBUG_OBJECT (sink, "ringbuffer may start now");
2202 sink->priv->sync_latency = TRUE;
2203 eos = GST_BASE_SINK (sink)->eos;
2204 GST_OBJECT_UNLOCK (sink);
2206 gst_audio_ring_buffer_may_start (sink->ringbuffer, TRUE);
2207 if (GST_BASE_SINK_CAST (sink)->pad_mode == GST_PAD_MODE_PULL ||
2208 g_atomic_int_get (&sink->eos_rendering) || eos) {
2209 /* we always start the ringbuffer in pull mode immediatly */
2210 /* sync rendering on eos needs running clock,
2211 * and others need running clock when finished rendering eos */
2212 gst_audio_ring_buffer_start (sink->ringbuffer);
2216 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
2217 /* ringbuffer cannot start anymore */
2218 gst_audio_ring_buffer_may_start (sink->ringbuffer, FALSE);
2219 gst_audio_ring_buffer_pause (sink->ringbuffer);
2221 GST_OBJECT_LOCK (sink);
2222 sink->priv->sync_latency = FALSE;
2223 GST_OBJECT_UNLOCK (sink);
2225 case GST_STATE_CHANGE_PAUSED_TO_READY:
2226 /* Only post clock-lost messages if this is the clock that
2227 * we've created. If the subclass has overriden it the subclass
2228 * should post this messages whenever necessary */
2229 if (sink->provided_clock && GST_IS_AUDIO_CLOCK (sink->provided_clock) &&
2230 GST_AUDIO_CLOCK_CAST (sink->provided_clock)->func ==
2231 (GstAudioClockGetTimeFunc) gst_audio_base_sink_get_time)
2232 gst_element_post_message (element,
2233 gst_message_new_clock_lost (GST_OBJECT_CAST (element),
2234 sink->provided_clock));
2236 /* make sure we unblock before calling the parent state change
2237 * so it can grab the STREAM_LOCK */
2238 gst_audio_ring_buffer_set_flushing (sink->ringbuffer, TRUE);
2244 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
2246 switch (transition) {
2247 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
2248 /* stop slaving ourselves to the master, if any */
2249 gst_clock_set_master (sink->provided_clock, NULL);
2251 case GST_STATE_CHANGE_PAUSED_TO_READY:
2252 gst_audio_ring_buffer_activate (sink->ringbuffer, FALSE);
2253 gst_audio_ring_buffer_release (sink->ringbuffer);
2255 case GST_STATE_CHANGE_READY_TO_NULL:
2256 /* we release again here because the aqcuire happens when setting the
2257 * caps, which happens before we commit the state to PAUSED and thus the
2258 * PAUSED->READY state change (see above, where we release the ringbuffer)
2259 * might not be called when we get here. */
2260 gst_audio_ring_buffer_activate (sink->ringbuffer, FALSE);
2261 gst_audio_ring_buffer_release (sink->ringbuffer);
2262 gst_audio_ring_buffer_close_device (sink->ringbuffer);
2263 GST_OBJECT_LOCK (sink);
2264 gst_object_unparent (GST_OBJECT_CAST (sink->ringbuffer));
2265 sink->ringbuffer = NULL;
2266 GST_OBJECT_UNLOCK (sink);
2277 /* subclass must post a meaningful error message */
2278 GST_DEBUG_OBJECT (sink, "open failed");
2279 return GST_STATE_CHANGE_FAILURE;