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., 59 Temple Place - Suite 330,
20 * Boston, MA 02111-1307, 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 /* FIXME 0.11: suppress warnings for deprecated API such as GStaticRecMutex
38 * with newer GLib versions (>= 2.31.0) */
39 #define GLIB_DISABLE_DEPRECATION_WARNINGS
40 #include "gstaudiobasesink.h"
42 GST_DEBUG_CATEGORY_STATIC (gst_audio_base_sink_debug);
43 #define GST_CAT_DEFAULT gst_audio_base_sink_debug
45 #define GST_AUDIO_BASE_SINK_GET_PRIVATE(obj) \
46 (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_AUDIO_BASE_SINK, GstAudioBaseSinkPrivate))
48 struct _GstAudioBaseSinkPrivate
50 /* upstream latency */
51 GstClockTime us_latency;
52 /* the clock slaving algorithm in use */
53 GstAudioBaseSinkSlaveMethod slave_method;
54 /* running average of clock skew */
55 GstClockTimeDiff avg_skew;
56 /* the number of samples we aligned last time */
59 gboolean sync_latency;
61 GstClockTime eos_time;
63 /* number of microseconds we allow clock slaving to drift
65 guint64 drift_tolerance;
67 /* number of nanoseconds we allow timestamps to drift
69 GstClockTime alignment_threshold;
71 /* time of the previous detected discont candidate */
72 GstClockTime discont_time;
74 /* number of nanoseconds to wait until creating a discontinuity */
75 GstClockTime discont_wait;
78 /* BaseAudioSink signals and args */
85 /* FIXME: 0.11, store the buffer_time and latency_time in nanoseconds */
86 #define DEFAULT_BUFFER_TIME ((200 * GST_MSECOND) / GST_USECOND)
87 #define DEFAULT_LATENCY_TIME ((10 * GST_MSECOND) / GST_USECOND)
88 #define DEFAULT_PROVIDE_CLOCK TRUE
89 #define DEFAULT_SLAVE_METHOD GST_AUDIO_BASE_SINK_SLAVE_SKEW
91 /* FIXME, enable pull mode when clock slaving and trick modes are figured out */
92 #define DEFAULT_CAN_ACTIVATE_PULL FALSE
94 /* when timestamps drift for more than 40ms we resync. This should
95 * be anough to compensate for timestamp rounding errors. */
96 #define DEFAULT_ALIGNMENT_THRESHOLD (40 * GST_MSECOND)
98 /* when clock slaving drift for more than 40ms we resync. This is
99 * a reasonable default */
100 #define DEFAULT_DRIFT_TOLERANCE ((40 * GST_MSECOND) / GST_USECOND)
102 /* allow for one second before resyncing to see if the timestamps drift will
103 * fix itself, or is a permanent offset */
104 #define DEFAULT_DISCONT_WAIT (1 * GST_SECOND)
114 PROP_CAN_ACTIVATE_PULL,
115 PROP_ALIGNMENT_THRESHOLD,
116 PROP_DRIFT_TOLERANCE,
123 gst_audio_base_sink_slave_method_get_type (void)
125 static volatile gsize slave_method_type = 0;
126 static const GEnumValue slave_method[] = {
127 {GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE, "GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE",
129 {GST_AUDIO_BASE_SINK_SLAVE_SKEW, "GST_AUDIO_BASE_SINK_SLAVE_SKEW", "skew"},
130 {GST_AUDIO_BASE_SINK_SLAVE_NONE, "GST_AUDIO_BASE_SINK_SLAVE_NONE", "none"},
134 if (g_once_init_enter (&slave_method_type)) {
136 g_enum_register_static ("GstAudioBaseSinkSlaveMethod", slave_method);
137 g_once_init_leave (&slave_method_type, tmp);
140 return (GType) slave_method_type;
145 GST_DEBUG_CATEGORY_INIT (gst_audio_base_sink_debug, "audiobasesink", 0, "audiobasesink element");
146 #define gst_audio_base_sink_parent_class parent_class
147 G_DEFINE_TYPE_WITH_CODE (GstAudioBaseSink, gst_audio_base_sink,
148 GST_TYPE_BASE_SINK, _do_init);
150 static void gst_audio_base_sink_dispose (GObject * object);
152 static void gst_audio_base_sink_set_property (GObject * object, guint prop_id,
153 const GValue * value, GParamSpec * pspec);
154 static void gst_audio_base_sink_get_property (GObject * object, guint prop_id,
155 GValue * value, GParamSpec * pspec);
158 static GstStateChangeReturn gst_audio_base_sink_async_play (GstBaseSink *
161 static GstStateChangeReturn gst_audio_base_sink_change_state (GstElement *
162 element, GstStateChange transition);
163 static gboolean gst_audio_base_sink_activate_pull (GstBaseSink * basesink,
165 static gboolean gst_audio_base_sink_query (GstElement * element, GstQuery *
168 static GstClock *gst_audio_base_sink_provide_clock (GstElement * elem);
169 static GstClockTime gst_audio_base_sink_get_time (GstClock * clock,
170 GstAudioBaseSink * sink);
171 static void gst_audio_base_sink_callback (GstAudioRingBuffer * rbuf,
172 guint8 * data, guint len, gpointer user_data);
174 static GstFlowReturn gst_audio_base_sink_preroll (GstBaseSink * bsink,
176 static GstFlowReturn gst_audio_base_sink_render (GstBaseSink * bsink,
178 static gboolean gst_audio_base_sink_event (GstBaseSink * bsink,
180 static GstFlowReturn gst_audio_base_sink_wait_eos (GstBaseSink * bsink,
182 static void gst_audio_base_sink_get_times (GstBaseSink * bsink,
183 GstBuffer * buffer, GstClockTime * start, GstClockTime * end);
184 static gboolean gst_audio_base_sink_setcaps (GstBaseSink * bsink,
186 static void gst_audio_base_sink_fixate (GstBaseSink * bsink, GstCaps * caps);
188 static gboolean gst_audio_base_sink_query_pad (GstBaseSink * bsink,
192 /* static guint gst_audio_base_sink_signals[LAST_SIGNAL] = { 0 }; */
195 gst_audio_base_sink_class_init (GstAudioBaseSinkClass * klass)
197 GObjectClass *gobject_class;
198 GstElementClass *gstelement_class;
199 GstBaseSinkClass *gstbasesink_class;
201 gobject_class = (GObjectClass *) klass;
202 gstelement_class = (GstElementClass *) klass;
203 gstbasesink_class = (GstBaseSinkClass *) klass;
205 g_type_class_add_private (klass, sizeof (GstAudioBaseSinkPrivate));
207 gobject_class->set_property = gst_audio_base_sink_set_property;
208 gobject_class->get_property = gst_audio_base_sink_get_property;
209 gobject_class->dispose = gst_audio_base_sink_dispose;
211 g_object_class_install_property (gobject_class, PROP_BUFFER_TIME,
212 g_param_spec_int64 ("buffer-time", "Buffer Time",
213 "Size of audio buffer in microseconds", 1,
214 G_MAXINT64, DEFAULT_BUFFER_TIME,
215 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
217 g_object_class_install_property (gobject_class, PROP_LATENCY_TIME,
218 g_param_spec_int64 ("latency-time", "Latency Time",
219 "Audio latency in microseconds", 1,
220 G_MAXINT64, DEFAULT_LATENCY_TIME,
221 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
223 g_object_class_install_property (gobject_class, PROP_PROVIDE_CLOCK,
224 g_param_spec_boolean ("provide-clock", "Provide Clock",
225 "Provide a clock to be used as the global pipeline clock",
226 DEFAULT_PROVIDE_CLOCK, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
228 g_object_class_install_property (gobject_class, PROP_SLAVE_METHOD,
229 g_param_spec_enum ("slave-method", "Slave Method",
230 "Algorithm to use to match the rate of the masterclock",
231 GST_TYPE_AUDIO_BASE_SINK_SLAVE_METHOD, DEFAULT_SLAVE_METHOD,
232 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
234 g_object_class_install_property (gobject_class, PROP_CAN_ACTIVATE_PULL,
235 g_param_spec_boolean ("can-activate-pull", "Allow Pull Scheduling",
236 "Allow pull-based scheduling", DEFAULT_CAN_ACTIVATE_PULL,
237 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
239 * GstAudioBaseSink:drift-tolerance
241 * Controls the amount of time in microseconds that clocks are allowed
242 * to drift before resynchronisation happens.
246 g_object_class_install_property (gobject_class, PROP_DRIFT_TOLERANCE,
247 g_param_spec_int64 ("drift-tolerance", "Drift Tolerance",
248 "Tolerance for clock drift in microseconds", 1,
249 G_MAXINT64, DEFAULT_DRIFT_TOLERANCE,
250 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
252 * GstAudioBaseSink:alignment_threshold
254 * Controls the amount of time in nanoseconds that timestamps are allowed
255 * to drift from their ideal time before choosing not to align them.
259 g_object_class_install_property (gobject_class, PROP_ALIGNMENT_THRESHOLD,
260 g_param_spec_uint64 ("alignment-threshold", "Alignment Threshold",
261 "Timestamp alignment threshold in nanoseconds", 1,
262 G_MAXUINT64 - 1, DEFAULT_ALIGNMENT_THRESHOLD,
263 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
266 * GstAudioBaseSink:discont-wait
268 * A window of time in nanoseconds to wait before creating a discontinuity as
269 * a result of breaching the drift-tolerance.
273 g_object_class_install_property (gobject_class, PROP_DISCONT_WAIT,
274 g_param_spec_uint64 ("discont-wait", "Discont Wait",
275 "Window of time in nanoseconds to wait before "
276 "creating a discontinuity", 0,
277 G_MAXUINT64 - 1, DEFAULT_DISCONT_WAIT,
278 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
280 gstelement_class->change_state =
281 GST_DEBUG_FUNCPTR (gst_audio_base_sink_change_state);
282 gstelement_class->provide_clock =
283 GST_DEBUG_FUNCPTR (gst_audio_base_sink_provide_clock);
284 gstelement_class->query = GST_DEBUG_FUNCPTR (gst_audio_base_sink_query);
286 gstbasesink_class->fixate = GST_DEBUG_FUNCPTR (gst_audio_base_sink_fixate);
287 gstbasesink_class->set_caps = GST_DEBUG_FUNCPTR (gst_audio_base_sink_setcaps);
288 gstbasesink_class->event = GST_DEBUG_FUNCPTR (gst_audio_base_sink_event);
289 gstbasesink_class->wait_eos =
290 GST_DEBUG_FUNCPTR (gst_audio_base_sink_wait_eos);
291 gstbasesink_class->get_times =
292 GST_DEBUG_FUNCPTR (gst_audio_base_sink_get_times);
293 gstbasesink_class->preroll = GST_DEBUG_FUNCPTR (gst_audio_base_sink_preroll);
294 gstbasesink_class->render = GST_DEBUG_FUNCPTR (gst_audio_base_sink_render);
295 gstbasesink_class->query = GST_DEBUG_FUNCPTR (gst_audio_base_sink_query_pad);
296 gstbasesink_class->activate_pull =
297 GST_DEBUG_FUNCPTR (gst_audio_base_sink_activate_pull);
299 /* ref class from a thread-safe context to work around missing bit of
300 * thread-safety in GObject */
301 g_type_class_ref (GST_TYPE_AUDIO_CLOCK);
302 g_type_class_ref (GST_TYPE_AUDIO_RING_BUFFER);
307 gst_audio_base_sink_init (GstAudioBaseSink * audiobasesink)
309 GstBaseSink *basesink;
311 audiobasesink->priv = GST_AUDIO_BASE_SINK_GET_PRIVATE (audiobasesink);
313 audiobasesink->buffer_time = DEFAULT_BUFFER_TIME;
314 audiobasesink->latency_time = DEFAULT_LATENCY_TIME;
315 audiobasesink->priv->slave_method = DEFAULT_SLAVE_METHOD;
316 audiobasesink->priv->drift_tolerance = DEFAULT_DRIFT_TOLERANCE;
317 audiobasesink->priv->alignment_threshold = DEFAULT_ALIGNMENT_THRESHOLD;
318 audiobasesink->priv->discont_wait = DEFAULT_DISCONT_WAIT;
320 audiobasesink->provided_clock = gst_audio_clock_new ("GstAudioSinkClock",
321 (GstAudioClockGetTimeFunc) gst_audio_base_sink_get_time, audiobasesink,
324 basesink = GST_BASE_SINK_CAST (audiobasesink);
325 basesink->can_activate_push = TRUE;
326 basesink->can_activate_pull = DEFAULT_CAN_ACTIVATE_PULL;
328 gst_base_sink_set_last_sample_enabled (basesink, FALSE);
329 if (DEFAULT_PROVIDE_CLOCK)
330 GST_OBJECT_FLAG_SET (basesink, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
332 GST_OBJECT_FLAG_UNSET (basesink, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
336 gst_audio_base_sink_dispose (GObject * object)
338 GstAudioBaseSink *sink;
340 sink = GST_AUDIO_BASE_SINK (object);
342 if (sink->provided_clock) {
343 gst_audio_clock_invalidate (sink->provided_clock);
344 gst_object_unref (sink->provided_clock);
345 sink->provided_clock = NULL;
348 if (sink->ringbuffer) {
349 gst_object_unparent (GST_OBJECT_CAST (sink->ringbuffer));
350 sink->ringbuffer = NULL;
353 G_OBJECT_CLASS (parent_class)->dispose (object);
358 gst_audio_base_sink_provide_clock (GstElement * elem)
360 GstAudioBaseSink *sink;
363 sink = GST_AUDIO_BASE_SINK (elem);
365 /* we have no ringbuffer (must be NULL state) */
366 if (sink->ringbuffer == NULL)
369 if (!gst_audio_ring_buffer_is_acquired (sink->ringbuffer))
372 GST_OBJECT_LOCK (sink);
373 if (!GST_OBJECT_FLAG_IS_SET (sink, GST_ELEMENT_FLAG_PROVIDE_CLOCK))
376 clock = GST_CLOCK_CAST (gst_object_ref (sink->provided_clock));
377 GST_OBJECT_UNLOCK (sink);
384 GST_DEBUG_OBJECT (sink, "ringbuffer not acquired");
389 GST_DEBUG_OBJECT (sink, "clock provide disabled");
390 GST_OBJECT_UNLOCK (sink);
396 gst_audio_base_sink_query_pad (GstBaseSink * bsink, GstQuery * query)
398 gboolean res = FALSE;
399 GstAudioBaseSink *basesink;
401 basesink = GST_AUDIO_BASE_SINK (bsink);
403 switch (GST_QUERY_TYPE (query)) {
404 case GST_QUERY_CONVERT:
406 GstFormat src_fmt, dest_fmt;
407 gint64 src_val, dest_val;
409 GST_LOG_OBJECT (basesink, "query convert");
411 if (basesink->ringbuffer) {
412 gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, NULL);
414 gst_audio_ring_buffer_convert (basesink->ringbuffer, src_fmt,
415 src_val, dest_fmt, &dest_val);
417 gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
423 res = GST_BASE_SINK_CLASS (parent_class)->query (bsink, query);
430 gst_audio_base_sink_query (GstElement * element, GstQuery * query)
432 gboolean res = FALSE;
433 GstAudioBaseSink *basesink;
435 basesink = GST_AUDIO_BASE_SINK (element);
437 switch (GST_QUERY_TYPE (query)) {
438 case GST_QUERY_LATENCY:
440 gboolean live, us_live;
441 GstClockTime min_l, max_l;
443 GST_DEBUG_OBJECT (basesink, "latency query");
445 /* ask parent first, it will do an upstream query for us. */
447 gst_base_sink_query_latency (GST_BASE_SINK_CAST (basesink), &live,
448 &us_live, &min_l, &max_l))) {
449 GstClockTime base_latency, min_latency, max_latency;
451 /* we and upstream are both live, adjust the min_latency */
452 if (live && us_live) {
453 GstAudioRingBufferSpec *spec;
455 GST_OBJECT_LOCK (basesink);
456 if (!basesink->ringbuffer || !basesink->ringbuffer->spec.info.rate) {
457 GST_OBJECT_UNLOCK (basesink);
459 GST_DEBUG_OBJECT (basesink,
460 "we are not yet negotiated, can't report latency yet");
464 spec = &basesink->ringbuffer->spec;
466 basesink->priv->us_latency = min_l;
469 gst_util_uint64_scale_int (spec->seglatency * spec->segsize,
470 GST_SECOND, spec->info.rate * spec->info.bpf);
471 GST_OBJECT_UNLOCK (basesink);
473 /* we cannot go lower than the buffer size and the min peer latency */
474 min_latency = base_latency + min_l;
475 /* the max latency is the max of the peer, we can delay an infinite
477 max_latency = (max_l == -1) ? -1 : (base_latency + max_l);
479 GST_DEBUG_OBJECT (basesink,
480 "peer min %" GST_TIME_FORMAT ", our min latency: %"
481 GST_TIME_FORMAT, GST_TIME_ARGS (min_l),
482 GST_TIME_ARGS (min_latency));
483 GST_DEBUG_OBJECT (basesink,
484 "peer max %" GST_TIME_FORMAT ", our max latency: %"
485 GST_TIME_FORMAT, GST_TIME_ARGS (max_l),
486 GST_TIME_ARGS (max_latency));
488 GST_DEBUG_OBJECT (basesink,
489 "peer or we are not live, don't care about latency");
493 gst_query_set_latency (query, live, min_latency, max_latency);
497 case GST_QUERY_CONVERT:
499 GstFormat src_fmt, dest_fmt;
500 gint64 src_val, dest_val;
502 GST_LOG_OBJECT (basesink, "query convert");
504 if (basesink->ringbuffer) {
505 gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, NULL);
507 gst_audio_ring_buffer_convert (basesink->ringbuffer, src_fmt,
508 src_val, dest_fmt, &dest_val);
510 gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
516 res = GST_ELEMENT_CLASS (parent_class)->query (element, query);
526 gst_audio_base_sink_get_time (GstClock * clock, GstAudioBaseSink * sink)
528 guint64 raw, samples;
532 if (sink->ringbuffer == NULL || sink->ringbuffer->spec.info.rate == 0)
533 return GST_CLOCK_TIME_NONE;
535 /* our processed samples are always increasing */
536 raw = samples = gst_audio_ring_buffer_samples_done (sink->ringbuffer);
538 /* the number of samples not yet processed, this is still queued in the
539 * device (not played for playback). */
540 delay = gst_audio_ring_buffer_delay (sink->ringbuffer);
542 if (G_LIKELY (samples >= delay))
547 result = gst_util_uint64_scale_int (samples, GST_SECOND,
548 sink->ringbuffer->spec.info.rate);
550 GST_DEBUG_OBJECT (sink,
551 "processed samples: raw %" G_GUINT64_FORMAT ", delay %u, real %"
552 G_GUINT64_FORMAT ", time %" GST_TIME_FORMAT,
553 raw, delay, samples, GST_TIME_ARGS (result));
559 * gst_audio_base_sink_set_provide_clock:
560 * @sink: a #GstAudioBaseSink
561 * @provide: new state
563 * Controls whether @sink will provide a clock or not. If @provide is %TRUE,
564 * gst_element_provide_clock() will return a clock that reflects the datarate
565 * of @sink. If @provide is %FALSE, gst_element_provide_clock() will return NULL.
570 gst_audio_base_sink_set_provide_clock (GstAudioBaseSink * sink,
573 g_return_if_fail (GST_IS_AUDIO_BASE_SINK (sink));
575 GST_OBJECT_LOCK (sink);
577 GST_OBJECT_FLAG_SET (sink, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
579 GST_OBJECT_FLAG_UNSET (sink, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
580 GST_OBJECT_UNLOCK (sink);
584 * gst_audio_base_sink_get_provide_clock:
585 * @sink: a #GstAudioBaseSink
587 * Queries whether @sink will provide a clock or not. See also
588 * gst_audio_base_sink_set_provide_clock.
590 * Returns: %TRUE if @sink will provide a clock.
595 gst_audio_base_sink_get_provide_clock (GstAudioBaseSink * sink)
599 g_return_val_if_fail (GST_IS_AUDIO_BASE_SINK (sink), FALSE);
601 GST_OBJECT_LOCK (sink);
602 result = GST_OBJECT_FLAG_IS_SET (sink, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
603 GST_OBJECT_UNLOCK (sink);
609 * gst_audio_base_sink_set_slave_method:
610 * @sink: a #GstAudioBaseSink
611 * @method: the new slave method
613 * Controls how clock slaving will be performed in @sink.
618 gst_audio_base_sink_set_slave_method (GstAudioBaseSink * sink,
619 GstAudioBaseSinkSlaveMethod method)
621 g_return_if_fail (GST_IS_AUDIO_BASE_SINK (sink));
623 GST_OBJECT_LOCK (sink);
624 sink->priv->slave_method = method;
625 GST_OBJECT_UNLOCK (sink);
629 * gst_audio_base_sink_get_slave_method:
630 * @sink: a #GstAudioBaseSink
632 * Get the current slave method used by @sink.
634 * Returns: The current slave method used by @sink.
638 GstAudioBaseSinkSlaveMethod
639 gst_audio_base_sink_get_slave_method (GstAudioBaseSink * sink)
641 GstAudioBaseSinkSlaveMethod result;
643 g_return_val_if_fail (GST_IS_AUDIO_BASE_SINK (sink), -1);
645 GST_OBJECT_LOCK (sink);
646 result = sink->priv->slave_method;
647 GST_OBJECT_UNLOCK (sink);
654 * gst_audio_base_sink_set_drift_tolerance:
655 * @sink: a #GstAudioBaseSink
656 * @drift_tolerance: the new drift tolerance in microseconds
658 * Controls the sink's drift tolerance.
663 gst_audio_base_sink_set_drift_tolerance (GstAudioBaseSink * sink,
664 gint64 drift_tolerance)
666 g_return_if_fail (GST_IS_AUDIO_BASE_SINK (sink));
668 GST_OBJECT_LOCK (sink);
669 sink->priv->drift_tolerance = drift_tolerance;
670 GST_OBJECT_UNLOCK (sink);
674 * gst_audio_base_sink_get_drift_tolerance
675 * @sink: a #GstAudioBaseSink
677 * Get the current drift tolerance, in microseconds, used by @sink.
679 * Returns: The current drift tolerance used by @sink.
684 gst_audio_base_sink_get_drift_tolerance (GstAudioBaseSink * sink)
688 g_return_val_if_fail (GST_IS_AUDIO_BASE_SINK (sink), -1);
690 GST_OBJECT_LOCK (sink);
691 result = sink->priv->drift_tolerance;
692 GST_OBJECT_UNLOCK (sink);
698 * gst_audio_base_sink_set_alignment_threshold:
699 * @sink: a #GstAudioBaseSink
700 * @alignment_threshold: the new alignment threshold in nanoseconds
702 * Controls the sink's alignment threshold.
707 gst_audio_base_sink_set_alignment_threshold (GstAudioBaseSink * sink,
708 GstClockTime alignment_threshold)
710 g_return_if_fail (GST_IS_AUDIO_BASE_SINK (sink));
712 GST_OBJECT_LOCK (sink);
713 sink->priv->alignment_threshold = alignment_threshold;
714 GST_OBJECT_UNLOCK (sink);
718 * gst_audio_base_sink_get_alignment_threshold
719 * @sink: a #GstAudioBaseSink
721 * Get the current alignment threshold, in nanoseconds, used by @sink.
723 * Returns: The current alignment threshold used by @sink.
728 gst_audio_base_sink_get_alignment_threshold (GstAudioBaseSink * sink)
732 g_return_val_if_fail (GST_IS_AUDIO_BASE_SINK (sink), -1);
734 GST_OBJECT_LOCK (sink);
735 result = sink->priv->alignment_threshold;
736 GST_OBJECT_UNLOCK (sink);
742 * gst_audio_base_sink_set_discont_wait:
743 * @sink: a #GstAudioBaseSink
744 * @discont_wait: the new discont wait in nanoseconds
746 * Controls how long the sink will wait before creating a discontinuity.
751 gst_audio_base_sink_set_discont_wait (GstAudioBaseSink * sink,
752 GstClockTime discont_wait)
754 g_return_if_fail (GST_IS_AUDIO_BASE_SINK (sink));
756 GST_OBJECT_LOCK (sink);
757 sink->priv->discont_wait = discont_wait;
758 GST_OBJECT_UNLOCK (sink);
762 * gst_audio_base_sink_get_discont_wait
763 * @sink: a #GstAudioBaseSink
765 * Get the current discont wait, in nanoseconds, used by @sink.
767 * Returns: The current discont wait used by @sink.
772 gst_audio_base_sink_get_discont_wait (GstAudioBaseSink * sink)
776 g_return_val_if_fail (GST_IS_AUDIO_BASE_SINK (sink), -1);
778 GST_OBJECT_LOCK (sink);
779 result = sink->priv->discont_wait;
780 GST_OBJECT_UNLOCK (sink);
786 gst_audio_base_sink_set_property (GObject * object, guint prop_id,
787 const GValue * value, GParamSpec * pspec)
789 GstAudioBaseSink *sink;
791 sink = GST_AUDIO_BASE_SINK (object);
794 case PROP_BUFFER_TIME:
795 sink->buffer_time = g_value_get_int64 (value);
797 case PROP_LATENCY_TIME:
798 sink->latency_time = g_value_get_int64 (value);
800 case PROP_PROVIDE_CLOCK:
801 gst_audio_base_sink_set_provide_clock (sink, g_value_get_boolean (value));
803 case PROP_SLAVE_METHOD:
804 gst_audio_base_sink_set_slave_method (sink, g_value_get_enum (value));
806 case PROP_CAN_ACTIVATE_PULL:
807 GST_BASE_SINK (sink)->can_activate_pull = g_value_get_boolean (value);
809 case PROP_DRIFT_TOLERANCE:
810 gst_audio_base_sink_set_drift_tolerance (sink, g_value_get_int64 (value));
812 case PROP_ALIGNMENT_THRESHOLD:
813 gst_audio_base_sink_set_alignment_threshold (sink,
814 g_value_get_uint64 (value));
816 case PROP_DISCONT_WAIT:
817 gst_audio_base_sink_set_discont_wait (sink, g_value_get_uint64 (value));
820 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
826 gst_audio_base_sink_get_property (GObject * object, guint prop_id,
827 GValue * value, GParamSpec * pspec)
829 GstAudioBaseSink *sink;
831 sink = GST_AUDIO_BASE_SINK (object);
834 case PROP_BUFFER_TIME:
835 g_value_set_int64 (value, sink->buffer_time);
837 case PROP_LATENCY_TIME:
838 g_value_set_int64 (value, sink->latency_time);
840 case PROP_PROVIDE_CLOCK:
841 g_value_set_boolean (value, gst_audio_base_sink_get_provide_clock (sink));
843 case PROP_SLAVE_METHOD:
844 g_value_set_enum (value, gst_audio_base_sink_get_slave_method (sink));
846 case PROP_CAN_ACTIVATE_PULL:
847 g_value_set_boolean (value, GST_BASE_SINK (sink)->can_activate_pull);
849 case PROP_DRIFT_TOLERANCE:
850 g_value_set_int64 (value, gst_audio_base_sink_get_drift_tolerance (sink));
852 case PROP_ALIGNMENT_THRESHOLD:
853 g_value_set_uint64 (value,
854 gst_audio_base_sink_get_alignment_threshold (sink));
856 case PROP_DISCONT_WAIT:
857 g_value_set_uint64 (value, gst_audio_base_sink_get_discont_wait (sink));
860 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
866 gst_audio_base_sink_setcaps (GstBaseSink * bsink, GstCaps * caps)
868 GstAudioBaseSink *sink = GST_AUDIO_BASE_SINK (bsink);
869 GstAudioRingBufferSpec *spec;
871 GstClockTime crate_num, crate_denom;
873 if (!sink->ringbuffer)
876 spec = &sink->ringbuffer->spec;
878 GST_DEBUG_OBJECT (sink, "release old ringbuffer");
880 /* get current time, updates the last_time. When the subclass has a clock that
881 * restarts from 0 when a new format is negotiated, it will call
882 * gst_audio_clock_reset() which will use this last_time to create an offset
883 * so that time from the clock keeps on increasing monotonically. */
884 now = gst_clock_get_time (sink->provided_clock);
886 GST_DEBUG_OBJECT (sink, "time was %" GST_TIME_FORMAT, GST_TIME_ARGS (now));
888 /* release old ringbuffer */
889 gst_audio_ring_buffer_pause (sink->ringbuffer);
890 gst_audio_ring_buffer_activate (sink->ringbuffer, FALSE);
891 gst_audio_ring_buffer_release (sink->ringbuffer);
893 GST_DEBUG_OBJECT (sink, "parse caps");
895 spec->buffer_time = sink->buffer_time;
896 spec->latency_time = sink->latency_time;
899 if (!gst_audio_ring_buffer_parse_caps (spec, caps))
902 gst_audio_ring_buffer_debug_spec_buff (spec);
904 GST_DEBUG_OBJECT (sink, "acquire ringbuffer");
905 if (!gst_audio_ring_buffer_acquire (sink->ringbuffer, spec))
908 if (bsink->pad_mode == GST_PAD_MODE_PUSH) {
909 GST_DEBUG_OBJECT (sink, "activate ringbuffer");
910 gst_audio_ring_buffer_activate (sink->ringbuffer, TRUE);
913 /* due to possible changes in the spec file we should recalibrate the clock */
914 gst_clock_get_calibration (sink->provided_clock, NULL, NULL,
915 &crate_num, &crate_denom);
916 gst_clock_set_calibration (sink->provided_clock,
917 gst_clock_get_internal_time (sink->provided_clock), now, crate_num,
920 /* calculate actual latency and buffer times.
921 * FIXME: In 0.11, store the latency_time internally in ns */
922 spec->latency_time = gst_util_uint64_scale (spec->segsize,
923 (GST_SECOND / GST_USECOND), spec->info.rate * spec->info.bpf);
925 spec->buffer_time = spec->segtotal * spec->latency_time;
927 gst_audio_ring_buffer_debug_spec_buff (spec);
934 GST_DEBUG_OBJECT (sink, "could not parse caps");
935 GST_ELEMENT_ERROR (sink, STREAM, FORMAT,
936 (NULL), ("cannot parse audio format."));
941 GST_DEBUG_OBJECT (sink, "could not acquire ringbuffer");
947 gst_audio_base_sink_fixate (GstBaseSink * bsink, GstCaps * caps)
952 s = gst_caps_get_structure (caps, 0);
954 /* fields for all formats */
955 gst_structure_fixate_field_nearest_int (s, "rate", 44100);
956 gst_structure_fixate_field_nearest_int (s, "channels", 2);
957 gst_structure_fixate_field_nearest_int (s, "width", 16);
960 if (gst_structure_has_field (s, "depth")) {
961 gst_structure_get_int (s, "width", &width);
962 /* round width to nearest multiple of 8 for the depth */
963 depth = GST_ROUND_UP_8 (width);
964 gst_structure_fixate_field_nearest_int (s, "depth", depth);
966 if (gst_structure_has_field (s, "signed"))
967 gst_structure_fixate_field_boolean (s, "signed", TRUE);
968 if (gst_structure_has_field (s, "endianness"))
969 gst_structure_fixate_field_nearest_int (s, "endianness", G_BYTE_ORDER);
971 GST_BASE_SINK_CLASS (parent_class)->fixate (bsink, caps);
975 gst_audio_base_sink_get_times (GstBaseSink * bsink, GstBuffer * buffer,
976 GstClockTime * start, GstClockTime * end)
978 /* our clock sync is a bit too much for the base class to handle so
979 * we implement it ourselves. */
980 *start = GST_CLOCK_TIME_NONE;
981 *end = GST_CLOCK_TIME_NONE;
984 /* This waits for the drain to happen and can be canceled */
986 gst_audio_base_sink_drain (GstAudioBaseSink * sink)
988 if (!sink->ringbuffer)
990 if (!sink->ringbuffer->spec.info.rate)
993 /* if PLAYING is interrupted,
994 * arrange to have clock running when going to PLAYING again */
995 g_atomic_int_set (&sink->eos_rendering, 1);
997 /* need to start playback before we can drain, but only when
998 * we have successfully negotiated a format and thus acquired the
1000 if (gst_audio_ring_buffer_is_acquired (sink->ringbuffer))
1001 gst_audio_ring_buffer_start (sink->ringbuffer);
1003 if (sink->priv->eos_time != -1) {
1004 GST_DEBUG_OBJECT (sink,
1005 "last sample time %" GST_TIME_FORMAT,
1006 GST_TIME_ARGS (sink->priv->eos_time));
1008 /* wait for the EOS time to be reached, this is the time when the last
1009 * sample is played. */
1010 gst_base_sink_wait_eos (GST_BASE_SINK (sink), sink->priv->eos_time, NULL);
1012 GST_DEBUG_OBJECT (sink, "drained audio");
1014 g_atomic_int_set (&sink->eos_rendering, 0);
1018 static GstFlowReturn
1019 gst_audio_base_sink_wait_eos (GstBaseSink * bsink, GstEvent * event)
1021 GstAudioBaseSink *sink = GST_AUDIO_BASE_SINK (bsink);
1024 ret = GST_BASE_SINK_CLASS (parent_class)->event (bsink, event);
1025 if (ret != GST_FLOW_OK)
1028 /* now wait till we played everything */
1029 gst_audio_base_sink_drain (sink);
1035 gst_audio_base_sink_event (GstBaseSink * bsink, GstEvent * event)
1037 GstAudioBaseSink *sink = GST_AUDIO_BASE_SINK (bsink);
1039 switch (GST_EVENT_TYPE (event)) {
1040 case GST_EVENT_FLUSH_START:
1041 if (sink->ringbuffer)
1042 gst_audio_ring_buffer_set_flushing (sink->ringbuffer, TRUE);
1044 case GST_EVENT_FLUSH_STOP:
1045 /* always resync on sample after a flush */
1046 sink->priv->avg_skew = -1;
1047 sink->next_sample = -1;
1048 sink->priv->eos_time = -1;
1049 sink->priv->discont_time = -1;
1050 if (sink->ringbuffer)
1051 gst_audio_ring_buffer_set_flushing (sink->ringbuffer, FALSE);
1056 return GST_BASE_SINK_CLASS (parent_class)->event (bsink, event);
1059 static GstFlowReturn
1060 gst_audio_base_sink_preroll (GstBaseSink * bsink, GstBuffer * buffer)
1062 GstAudioBaseSink *sink = GST_AUDIO_BASE_SINK (bsink);
1064 if (!gst_audio_ring_buffer_is_acquired (sink->ringbuffer))
1067 /* we don't really do anything when prerolling. We could make a
1068 * property to play this buffer to have some sort of scrubbing
1074 GST_DEBUG_OBJECT (sink, "ringbuffer in wrong state");
1075 GST_ELEMENT_ERROR (sink, STREAM, FORMAT, (NULL), ("sink not negotiated."));
1076 return GST_FLOW_NOT_NEGOTIATED;
1081 gst_audio_base_sink_get_offset (GstAudioBaseSink * sink)
1084 gint writeseg, segdone, sps;
1087 /* assume we can append to the previous sample */
1088 sample = sink->next_sample;
1089 /* no previous sample, try to insert at position 0 */
1093 sps = sink->ringbuffer->samples_per_seg;
1095 /* figure out the segment and the offset inside the segment where
1096 * the sample should be written. */
1097 writeseg = sample / sps;
1099 /* get the currently processed segment */
1100 segdone = g_atomic_int_get (&sink->ringbuffer->segdone)
1101 - sink->ringbuffer->segbase;
1103 /* see how far away it is from the write segment */
1104 diff = writeseg - segdone;
1106 /* sample would be dropped, position to next playable position */
1107 sample = (segdone + 1) * sps;
1114 clock_convert_external (GstClockTime external, GstClockTime cinternal,
1115 GstClockTime cexternal, GstClockTime crate_num, GstClockTime crate_denom)
1117 /* adjust for rate and speed */
1118 if (external >= cexternal) {
1120 gst_util_uint64_scale (external - cexternal, crate_denom, crate_num);
1121 external += cinternal;
1124 gst_util_uint64_scale (cexternal - external, crate_denom, crate_num);
1125 if (cinternal > external)
1126 external = cinternal - external;
1133 /* algorithm to calculate sample positions that will result in resampling to
1134 * match the clock rate of the master */
1136 gst_audio_base_sink_resample_slaving (GstAudioBaseSink * sink,
1137 GstClockTime render_start, GstClockTime render_stop,
1138 GstClockTime * srender_start, GstClockTime * srender_stop)
1140 GstClockTime cinternal, cexternal;
1141 GstClockTime crate_num, crate_denom;
1143 /* FIXME, we can sample and add observations here or use the timeouts on the
1144 * clock. No idea which one is better or more stable. The timeout seems more
1145 * arbitrary but this one seems more demanding and does not work when there is
1146 * no data comming in to the sink. */
1148 GstClockTime etime, itime;
1151 /* sample clocks and figure out clock skew */
1152 etime = gst_clock_get_time (GST_ELEMENT_CLOCK (sink));
1153 itime = gst_audio_clock_get_time (sink->provided_clock);
1155 /* add new observation */
1156 gst_clock_add_observation (sink->provided_clock, itime, etime, &r_squared);
1159 /* get calibration parameters to compensate for speed and offset differences
1160 * when we are slaved */
1161 gst_clock_get_calibration (sink->provided_clock, &cinternal, &cexternal,
1162 &crate_num, &crate_denom);
1164 GST_DEBUG_OBJECT (sink, "internal %" GST_TIME_FORMAT " external %"
1165 GST_TIME_FORMAT " %" G_GUINT64_FORMAT "/%" G_GUINT64_FORMAT " = %f",
1166 GST_TIME_ARGS (cinternal), GST_TIME_ARGS (cexternal), crate_num,
1167 crate_denom, gst_guint64_to_gdouble (crate_num) /
1168 gst_guint64_to_gdouble (crate_denom));
1171 crate_denom = crate_num = 1;
1173 /* bring external time to internal time */
1174 render_start = clock_convert_external (render_start, cinternal, cexternal,
1175 crate_num, crate_denom);
1176 render_stop = clock_convert_external (render_stop, cinternal, cexternal,
1177 crate_num, crate_denom);
1179 GST_DEBUG_OBJECT (sink,
1180 "after slaving: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
1181 GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
1183 *srender_start = render_start;
1184 *srender_stop = render_stop;
1187 /* algorithm to calculate sample positions that will result in changing the
1188 * playout pointer to match the clock rate of the master */
1190 gst_audio_base_sink_skew_slaving (GstAudioBaseSink * sink,
1191 GstClockTime render_start, GstClockTime render_stop,
1192 GstClockTime * srender_start, GstClockTime * srender_stop)
1194 GstClockTime cinternal, cexternal, crate_num, crate_denom;
1195 GstClockTime etime, itime;
1196 GstClockTimeDiff skew, mdrift, mdrift2;
1200 /* get calibration parameters to compensate for offsets */
1201 gst_clock_get_calibration (sink->provided_clock, &cinternal, &cexternal,
1202 &crate_num, &crate_denom);
1204 /* sample clocks and figure out clock skew */
1205 etime = gst_clock_get_time (GST_ELEMENT_CLOCK (sink));
1206 itime = gst_audio_clock_get_time (sink->provided_clock);
1207 itime = gst_audio_clock_adjust (sink->provided_clock, itime);
1209 GST_DEBUG_OBJECT (sink,
1210 "internal %" GST_TIME_FORMAT " external %" GST_TIME_FORMAT
1211 " cinternal %" GST_TIME_FORMAT " cexternal %" GST_TIME_FORMAT,
1212 GST_TIME_ARGS (itime), GST_TIME_ARGS (etime),
1213 GST_TIME_ARGS (cinternal), GST_TIME_ARGS (cexternal));
1215 /* make sure we never go below 0 */
1216 etime = etime > cexternal ? etime - cexternal : 0;
1217 itime = itime > cinternal ? itime - cinternal : 0;
1219 /* do itime - etime.
1220 * positive value means external clock goes slower
1221 * negative value means external clock goes faster */
1222 skew = GST_CLOCK_DIFF (etime, itime);
1223 if (sink->priv->avg_skew == -1) {
1224 /* first observation */
1225 sink->priv->avg_skew = skew;
1227 /* next observations use a moving average */
1228 sink->priv->avg_skew = (31 * sink->priv->avg_skew + skew) / 32;
1231 GST_DEBUG_OBJECT (sink, "internal %" GST_TIME_FORMAT " external %"
1232 GST_TIME_FORMAT " skew %" G_GINT64_FORMAT " avg %" G_GINT64_FORMAT,
1233 GST_TIME_ARGS (itime), GST_TIME_ARGS (etime), skew, sink->priv->avg_skew);
1235 /* the max drift we allow */
1236 mdrift = sink->priv->drift_tolerance * 1000;
1237 mdrift2 = mdrift / 2;
1239 /* adjust playout pointer based on skew */
1240 if (sink->priv->avg_skew > mdrift2) {
1241 /* master is running slower, move internal time forward */
1242 GST_WARNING_OBJECT (sink,
1243 "correct clock skew %" G_GINT64_FORMAT " > %" G_GINT64_FORMAT,
1244 sink->priv->avg_skew, mdrift2);
1245 cexternal = cexternal > mdrift ? cexternal - mdrift : 0;
1246 sink->priv->avg_skew -= mdrift;
1248 driftsamples = (sink->ringbuffer->spec.info.rate * mdrift) / GST_SECOND;
1249 last_align = sink->priv->last_align;
1251 /* if we were aligning in the wrong direction or we aligned more than what we
1252 * will correct, resync */
1253 if (last_align < 0 || last_align > driftsamples)
1254 sink->next_sample = -1;
1256 GST_DEBUG_OBJECT (sink,
1257 "last_align %" G_GINT64_FORMAT " driftsamples %u, next %"
1258 G_GUINT64_FORMAT, last_align, driftsamples, sink->next_sample);
1260 gst_clock_set_calibration (sink->provided_clock, cinternal, cexternal,
1261 crate_num, crate_denom);
1262 } else if (sink->priv->avg_skew < -mdrift2) {
1263 /* master is running faster, move external time forwards */
1264 GST_WARNING_OBJECT (sink,
1265 "correct clock skew %" G_GINT64_FORMAT " < %" G_GINT64_FORMAT,
1266 sink->priv->avg_skew, -mdrift2);
1267 cexternal += mdrift;
1268 sink->priv->avg_skew += mdrift;
1270 driftsamples = (sink->ringbuffer->spec.info.rate * mdrift) / GST_SECOND;
1271 last_align = sink->priv->last_align;
1273 /* if we were aligning in the wrong direction or we aligned more than what we
1274 * will correct, resync */
1275 if (last_align > 0 || -last_align > driftsamples)
1276 sink->next_sample = -1;
1278 GST_DEBUG_OBJECT (sink,
1279 "last_align %" G_GINT64_FORMAT " driftsamples %u, next %"
1280 G_GUINT64_FORMAT, last_align, driftsamples, sink->next_sample);
1282 gst_clock_set_calibration (sink->provided_clock, cinternal, cexternal,
1283 crate_num, crate_denom);
1286 /* convert, ignoring speed */
1287 render_start = clock_convert_external (render_start, cinternal, cexternal,
1288 crate_num, crate_denom);
1289 render_stop = clock_convert_external (render_stop, cinternal, cexternal,
1290 crate_num, crate_denom);
1292 *srender_start = render_start;
1293 *srender_stop = render_stop;
1296 /* apply the clock offset but do no slaving otherwise */
1298 gst_audio_base_sink_none_slaving (GstAudioBaseSink * sink,
1299 GstClockTime render_start, GstClockTime render_stop,
1300 GstClockTime * srender_start, GstClockTime * srender_stop)
1302 GstClockTime cinternal, cexternal, crate_num, crate_denom;
1304 /* get calibration parameters to compensate for offsets */
1305 gst_clock_get_calibration (sink->provided_clock, &cinternal, &cexternal,
1306 &crate_num, &crate_denom);
1308 /* convert, ignoring speed */
1309 render_start = clock_convert_external (render_start, cinternal, cexternal,
1310 crate_num, crate_denom);
1311 render_stop = clock_convert_external (render_stop, cinternal, cexternal,
1312 crate_num, crate_denom);
1314 *srender_start = render_start;
1315 *srender_stop = render_stop;
1318 /* converts render_start and render_stop to their slaved values */
1320 gst_audio_base_sink_handle_slaving (GstAudioBaseSink * sink,
1321 GstClockTime render_start, GstClockTime render_stop,
1322 GstClockTime * srender_start, GstClockTime * srender_stop)
1324 switch (sink->priv->slave_method) {
1325 case GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE:
1326 gst_audio_base_sink_resample_slaving (sink, render_start, render_stop,
1327 srender_start, srender_stop);
1329 case GST_AUDIO_BASE_SINK_SLAVE_SKEW:
1330 gst_audio_base_sink_skew_slaving (sink, render_start, render_stop,
1331 srender_start, srender_stop);
1333 case GST_AUDIO_BASE_SINK_SLAVE_NONE:
1334 gst_audio_base_sink_none_slaving (sink, render_start, render_stop,
1335 srender_start, srender_stop);
1338 g_warning ("unknown slaving method %d", sink->priv->slave_method);
1343 /* must be called with LOCK */
1344 static GstFlowReturn
1345 gst_audio_base_sink_sync_latency (GstBaseSink * bsink, GstMiniObject * obj)
1348 GstClockReturn status;
1349 GstClockTime time, render_delay;
1351 GstAudioBaseSink *sink;
1352 GstClockTime itime, etime;
1353 GstClockTime rate_num, rate_denom;
1354 GstClockTimeDiff jitter;
1356 sink = GST_AUDIO_BASE_SINK (bsink);
1358 clock = GST_ELEMENT_CLOCK (sink);
1359 if (G_UNLIKELY (clock == NULL))
1362 /* we provided the global clock, don't need to do anything special */
1363 if (clock == sink->provided_clock)
1366 GST_OBJECT_UNLOCK (sink);
1369 GST_DEBUG_OBJECT (sink, "checking preroll");
1371 ret = gst_base_sink_do_preroll (bsink, obj);
1372 if (ret != GST_FLOW_OK)
1375 GST_OBJECT_LOCK (sink);
1376 time = sink->priv->us_latency;
1377 GST_OBJECT_UNLOCK (sink);
1379 /* Renderdelay is added onto our own latency, and needs
1380 * to be subtracted as well */
1381 render_delay = gst_base_sink_get_render_delay (bsink);
1383 if (G_LIKELY (time > render_delay))
1384 time -= render_delay;
1388 /* preroll done, we can sync since we are in PLAYING now. */
1389 GST_DEBUG_OBJECT (sink, "possibly waiting for clock to reach %"
1390 GST_TIME_FORMAT, GST_TIME_ARGS (time));
1392 /* wait for the clock, this can be interrupted because we got shut down or
1394 status = gst_base_sink_wait_clock (bsink, time, &jitter);
1396 GST_DEBUG_OBJECT (sink, "clock returned %d %" GST_TIME_FORMAT, status,
1397 GST_TIME_ARGS (jitter));
1399 /* invalid time, no clock or sync disabled, just continue then */
1400 if (status == GST_CLOCK_BADTIME)
1403 /* waiting could have been interrupted and we can be flushing now */
1404 if (G_UNLIKELY (bsink->flushing))
1407 /* retry if we got unscheduled, which means we did not reach the timeout
1408 * yet. if some other error occures, we continue. */
1409 } while (status == GST_CLOCK_UNSCHEDULED);
1411 GST_OBJECT_LOCK (sink);
1412 GST_DEBUG_OBJECT (sink, "latency synced");
1414 /* when we prerolled in time, we can accurately set the calibration,
1415 * our internal clock should exactly have been the latency (== the running
1416 * time of the external clock) */
1417 etime = GST_ELEMENT_CAST (sink)->base_time + time;
1418 itime = gst_audio_clock_get_time (sink->provided_clock);
1419 itime = gst_audio_clock_adjust (sink->provided_clock, itime);
1421 if (status == GST_CLOCK_EARLY) {
1422 /* when we prerolled late, we have to take into account the lateness */
1423 GST_DEBUG_OBJECT (sink, "late preroll, adding jitter");
1427 /* start ringbuffer so we can start slaving right away when we need to */
1428 gst_audio_ring_buffer_start (sink->ringbuffer);
1430 GST_DEBUG_OBJECT (sink,
1431 "internal time: %" GST_TIME_FORMAT " external time: %" GST_TIME_FORMAT,
1432 GST_TIME_ARGS (itime), GST_TIME_ARGS (etime));
1434 /* copy the original calibrated rate but update the internal and external
1436 gst_clock_get_calibration (sink->provided_clock, NULL, NULL, &rate_num,
1438 gst_clock_set_calibration (sink->provided_clock, itime, etime,
1439 rate_num, rate_denom);
1441 switch (sink->priv->slave_method) {
1442 case GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE:
1443 /* only set as master when we are resampling */
1444 GST_DEBUG_OBJECT (sink, "Setting clock as master");
1445 gst_clock_set_master (sink->provided_clock, clock);
1447 case GST_AUDIO_BASE_SINK_SLAVE_SKEW:
1448 case GST_AUDIO_BASE_SINK_SLAVE_NONE:
1453 sink->priv->avg_skew = -1;
1454 sink->next_sample = -1;
1455 sink->priv->eos_time = -1;
1456 sink->priv->discont_time = -1;
1463 GST_DEBUG_OBJECT (sink, "we have no clock");
1468 GST_DEBUG_OBJECT (sink, "we are not slaved");
1473 GST_DEBUG_OBJECT (sink, "we are flushing");
1474 GST_OBJECT_LOCK (sink);
1475 return GST_FLOW_WRONG_STATE;
1480 gst_audio_base_sink_get_alignment (GstAudioBaseSink * sink,
1481 GstClockTime sample_offset)
1483 GstAudioRingBuffer *ringbuf = sink->ringbuffer;
1486 gint64 max_sample_diff;
1487 gint segdone = g_atomic_int_get (&ringbuf->segdone) - ringbuf->segbase;
1488 gint64 samples_done = segdone * ringbuf->samples_per_seg;
1489 gint64 headroom = sample_offset - samples_done;
1490 gboolean allow_align = TRUE;
1491 gboolean discont = FALSE;
1494 /* now try to align the sample to the previous one, first see how big the
1496 if (sample_offset >= sink->next_sample)
1497 sample_diff = sample_offset - sink->next_sample;
1499 sample_diff = sink->next_sample - sample_offset;
1501 rate = GST_AUDIO_INFO_RATE (&ringbuf->spec.info);
1503 /* calculate the max allowed drift in units of samples. */
1504 max_sample_diff = gst_util_uint64_scale_int (sink->priv->alignment_threshold,
1507 /* calc align with previous sample */
1508 align = sink->next_sample - sample_offset;
1510 /* don't align if it means writing behind the read-segment */
1511 if (sample_diff > headroom && align < 0)
1512 allow_align = FALSE;
1514 if (G_UNLIKELY (sample_diff >= max_sample_diff)) {
1515 /* wait before deciding to make a discontinuity */
1516 if (sink->priv->discont_wait > 0) {
1517 GstClockTime time = gst_util_uint64_scale_int (sample_offset,
1519 if (sink->priv->discont_time == -1) {
1520 /* discont candidate */
1521 sink->priv->discont_time = time;
1522 } else if (time - sink->priv->discont_time >= sink->priv->discont_wait) {
1523 /* discont_wait expired, discontinuity detected */
1525 sink->priv->discont_time = -1;
1530 } else if (G_UNLIKELY (sink->priv->discont_time != -1)) {
1531 /* we have had a discont, but are now back on track! */
1532 sink->priv->discont_time = -1;
1535 if (G_LIKELY (!discont && allow_align)) {
1536 GST_DEBUG_OBJECT (sink,
1537 "align with prev sample, ABS (%" G_GINT64_FORMAT ") < %"
1538 G_GINT64_FORMAT, align, max_sample_diff);
1540 gint64 diff_s G_GNUC_UNUSED;
1542 /* calculate sample diff in seconds for error message */
1543 diff_s = gst_util_uint64_scale_int (sample_diff, GST_SECOND, rate);
1545 /* timestamps drifted apart from previous samples too much, we need to
1546 * resync. We log this as an element warning. */
1547 GST_WARNING_OBJECT (sink,
1548 "Unexpected discontinuity in audio timestamps of "
1549 "%s%" GST_TIME_FORMAT ", resyncing",
1550 sample_offset > sink->next_sample ? "+" : "-", GST_TIME_ARGS (diff_s));
1557 static GstFlowReturn
1558 gst_audio_base_sink_render (GstBaseSink * bsink, GstBuffer * buf)
1561 GstClockTime time, stop, render_start, render_stop, sample_offset;
1562 GstClockTimeDiff sync_offset, ts_offset;
1563 GstAudioBaseSinkClass *bclass;
1564 GstAudioBaseSink *sink;
1565 GstAudioRingBuffer *ringbuf;
1567 guint64 ctime, cstop;
1571 guint samples, written;
1575 GstClockTime base_time, render_delay, latency;
1577 gboolean sync, slaved, align_next;
1579 GstSegment clip_seg;
1581 GstBuffer *out = NULL;
1583 sink = GST_AUDIO_BASE_SINK (bsink);
1584 bclass = GST_AUDIO_BASE_SINK_GET_CLASS (sink);
1586 ringbuf = sink->ringbuffer;
1588 /* can't do anything when we don't have the device */
1589 if (G_UNLIKELY (!gst_audio_ring_buffer_is_acquired (ringbuf)))
1592 /* Wait for upstream latency before starting the ringbuffer, we do this so
1593 * that we can align the first sample of the ringbuffer to the base_time +
1595 GST_OBJECT_LOCK (sink);
1596 base_time = GST_ELEMENT_CAST (sink)->base_time;
1597 if (G_UNLIKELY (sink->priv->sync_latency)) {
1598 ret = gst_audio_base_sink_sync_latency (bsink, GST_MINI_OBJECT_CAST (buf));
1599 GST_OBJECT_UNLOCK (sink);
1600 if (G_UNLIKELY (ret != GST_FLOW_OK))
1601 goto sync_latency_failed;
1602 /* only do this once until we are set back to PLAYING */
1603 sink->priv->sync_latency = FALSE;
1605 GST_OBJECT_UNLOCK (sink);
1608 /* Before we go on, let's see if we need to payload the data. If yes, we also
1609 * need to unref the output buffer before leaving. */
1610 if (bclass->payload) {
1611 out = bclass->payload (sink, buf);
1614 goto payload_failed;
1619 bpf = GST_AUDIO_INFO_BPF (&ringbuf->spec.info);
1620 rate = GST_AUDIO_INFO_RATE (&ringbuf->spec.info);
1622 size = gst_buffer_get_size (buf);
1623 if (G_UNLIKELY (size % bpf) != 0)
1626 samples = size / bpf;
1627 out_samples = samples;
1629 in_offset = GST_BUFFER_OFFSET (buf);
1630 time = GST_BUFFER_TIMESTAMP (buf);
1632 GST_DEBUG_OBJECT (sink,
1633 "time %" GST_TIME_FORMAT ", offset %" G_GUINT64_FORMAT ", start %"
1634 GST_TIME_FORMAT ", samples %u", GST_TIME_ARGS (time), in_offset,
1635 GST_TIME_ARGS (bsink->segment.start), samples);
1639 /* if not valid timestamp or we can't clip or sync, try to play
1641 if (!GST_CLOCK_TIME_IS_VALID (time)) {
1642 render_start = gst_audio_base_sink_get_offset (sink);
1643 render_stop = render_start + samples;
1644 GST_DEBUG_OBJECT (sink, "Buffer of size %" G_GSIZE_FORMAT " has no time."
1645 " Using render_start=%" G_GUINT64_FORMAT, size, render_start);
1646 /* we don't have a start so we don't know stop either */
1651 /* let's calc stop based on the number of samples in the buffer instead
1652 * of trusting the DURATION */
1653 stop = time + gst_util_uint64_scale_int (samples, GST_SECOND, rate);
1655 /* prepare the clipping segment. Since we will be subtracting ts-offset and
1656 * device-delay later we scale the start and stop with those values so that we
1657 * can correctly clip them */
1658 clip_seg.format = GST_FORMAT_TIME;
1659 clip_seg.start = bsink->segment.start;
1660 clip_seg.stop = bsink->segment.stop;
1661 clip_seg.duration = -1;
1663 /* the sync offset is the combination of ts-offset and device-delay */
1664 latency = gst_base_sink_get_latency (bsink);
1665 ts_offset = gst_base_sink_get_ts_offset (bsink);
1666 render_delay = gst_base_sink_get_render_delay (bsink);
1667 sync_offset = ts_offset - render_delay + latency;
1669 GST_DEBUG_OBJECT (sink,
1670 "sync-offset %" G_GINT64_FORMAT ", render-delay %" GST_TIME_FORMAT
1671 ", ts-offset %" G_GINT64_FORMAT, sync_offset,
1672 GST_TIME_ARGS (render_delay), ts_offset);
1674 /* compensate for ts-offset and device-delay when negative we need to
1676 if (sync_offset < 0) {
1677 clip_seg.start += -sync_offset;
1678 if (clip_seg.stop != -1)
1679 clip_seg.stop += -sync_offset;
1682 /* samples should be rendered based on their timestamp. All samples
1683 * arriving before the segment.start or after segment.stop are to be
1684 * thrown away. All samples should also be clipped to the segment
1686 if (!gst_segment_clip (&clip_seg, GST_FORMAT_TIME, time, stop, &ctime,
1688 goto out_of_segment;
1690 /* see if some clipping happened */
1691 diff = ctime - time;
1693 /* bring clipped time to samples */
1694 diff = gst_util_uint64_scale_int (diff, rate, GST_SECOND);
1695 GST_DEBUG_OBJECT (sink, "clipping start to %" GST_TIME_FORMAT " %"
1696 G_GUINT64_FORMAT " samples", GST_TIME_ARGS (ctime), diff);
1698 offset += diff * bpf;
1701 diff = stop - cstop;
1703 /* bring clipped time to samples */
1704 diff = gst_util_uint64_scale_int (diff, rate, GST_SECOND);
1705 GST_DEBUG_OBJECT (sink, "clipping stop to %" GST_TIME_FORMAT " %"
1706 G_GUINT64_FORMAT " samples", GST_TIME_ARGS (cstop), diff);
1711 /* figure out how to sync */
1712 if ((clock = GST_ELEMENT_CLOCK (bsink)))
1718 /* no sync needed, play sample ASAP */
1719 render_start = gst_audio_base_sink_get_offset (sink);
1720 render_stop = render_start + samples;
1721 GST_DEBUG_OBJECT (sink,
1722 "no sync needed. Using render_start=%" G_GUINT64_FORMAT, render_start);
1726 /* bring buffer start and stop times to running time */
1728 gst_segment_to_running_time (&bsink->segment, GST_FORMAT_TIME, time);
1730 gst_segment_to_running_time (&bsink->segment, GST_FORMAT_TIME, stop);
1732 GST_DEBUG_OBJECT (sink,
1733 "running: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
1734 GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
1736 /* store the time of the last sample, we'll use this to perform sync on the
1737 * last sample when draining the buffer */
1738 if (bsink->segment.rate >= 0.0) {
1739 sink->priv->eos_time = render_stop;
1741 sink->priv->eos_time = render_start;
1744 /* compensate for ts-offset and delay we know this will not underflow because we
1746 GST_DEBUG_OBJECT (sink,
1747 "compensating for sync-offset %" GST_TIME_FORMAT,
1748 GST_TIME_ARGS (sync_offset));
1749 render_start += sync_offset;
1750 render_stop += sync_offset;
1752 GST_DEBUG_OBJECT (sink, "adding base_time %" GST_TIME_FORMAT,
1753 GST_TIME_ARGS (base_time));
1755 /* add base time to sync against the clock */
1756 render_start += base_time;
1757 render_stop += base_time;
1759 GST_DEBUG_OBJECT (sink,
1760 "after compensation: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
1761 GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
1763 if ((slaved = clock != sink->provided_clock)) {
1764 /* handle clock slaving */
1765 gst_audio_base_sink_handle_slaving (sink, render_start, render_stop,
1766 &render_start, &render_stop);
1768 /* no slaving needed but we need to adapt to the clock calibration
1770 gst_audio_base_sink_none_slaving (sink, render_start, render_stop,
1771 &render_start, &render_stop);
1774 GST_DEBUG_OBJECT (sink,
1775 "final timestamps: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
1776 GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
1778 /* bring to position in the ringbuffer */
1779 time_offset = GST_AUDIO_CLOCK_CAST (sink->provided_clock)->time_offset;
1780 GST_DEBUG_OBJECT (sink,
1781 "time offset %" GST_TIME_FORMAT, GST_TIME_ARGS (time_offset));
1782 if (render_start > time_offset)
1783 render_start -= time_offset;
1786 if (render_stop > time_offset)
1787 render_stop -= time_offset;
1791 /* in some clock slaving cases, all late samples end up at 0 first,
1792 * and subsequent ones align with that until threshold exceeded,
1793 * and then sync back to 0 and so on, so avoid that altogether */
1794 if (G_UNLIKELY (render_start == 0 && render_stop == 0))
1797 /* and bring the time to the rate corrected offset in the buffer */
1798 render_start = gst_util_uint64_scale_int (render_start, rate, GST_SECOND);
1799 render_stop = gst_util_uint64_scale_int (render_stop, rate, GST_SECOND);
1801 /* positive playback rate, first sample is render_start, negative rate, first
1802 * sample is render_stop. When no rate conversion is active, render exactly
1803 * the amount of input samples to avoid aligning to rounding errors. */
1804 if (bsink->segment.rate >= 0.0) {
1805 sample_offset = render_start;
1806 if (bsink->segment.rate == 1.0)
1807 render_stop = sample_offset + samples;
1809 sample_offset = render_stop;
1810 if (bsink->segment.rate == -1.0)
1811 render_start = sample_offset + samples;
1814 /* always resync after a discont */
1815 if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT))) {
1816 GST_DEBUG_OBJECT (sink, "resync after discont");
1820 /* resync when we don't know what to align the sample with */
1821 if (G_UNLIKELY (sink->next_sample == -1)) {
1822 GST_DEBUG_OBJECT (sink,
1823 "no align possible: no previous sample position known");
1827 align = gst_audio_base_sink_get_alignment (sink, sample_offset);
1828 sink->priv->last_align = align;
1830 /* apply alignment */
1831 render_start += align;
1833 /* only align stop if we are not slaved to resample */
1834 if (slaved && sink->priv->slave_method == GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE) {
1835 GST_DEBUG_OBJECT (sink, "no stop time align needed: we are slaved");
1838 render_stop += align;
1841 /* number of target samples is difference between start and stop */
1842 out_samples = render_stop - render_start;
1845 /* we render the first or last sample first, depending on the rate */
1846 if (bsink->segment.rate >= 0.0)
1847 sample_offset = render_start;
1849 sample_offset = render_stop;
1851 GST_DEBUG_OBJECT (sink, "rendering at %" G_GUINT64_FORMAT " %d/%d",
1852 sample_offset, samples, out_samples);
1854 /* we need to accumulate over different runs for when we get interrupted */
1857 data = gst_buffer_map (buf, &size, NULL, GST_MAP_READ);
1860 gst_audio_ring_buffer_commit (ringbuf, &sample_offset,
1861 data + offset, samples, out_samples, &accum);
1863 GST_DEBUG_OBJECT (sink, "wrote %u of %u", written, samples);
1864 /* if we wrote all, we're done */
1865 if (written == samples)
1868 /* else something interrupted us and we wait for preroll. */
1869 if ((ret = gst_base_sink_wait_preroll (bsink)) != GST_FLOW_OK)
1872 /* if we got interrupted, we cannot assume that the next sample should
1873 * be aligned to this one */
1876 /* update the output samples. FIXME, this will just skip them when pausing
1877 * during trick mode */
1878 if (out_samples > written) {
1879 out_samples -= written;
1885 offset += written * bpf;
1887 gst_buffer_unmap (buf, data, size);
1890 sink->next_sample = sample_offset;
1892 sink->next_sample = -1;
1894 GST_DEBUG_OBJECT (sink, "next sample expected at %" G_GUINT64_FORMAT,
1897 if (GST_CLOCK_TIME_IS_VALID (stop) && stop >= bsink->segment.stop) {
1898 GST_DEBUG_OBJECT (sink,
1899 "start playback because we are at the end of segment");
1900 gst_audio_ring_buffer_start (ringbuf);
1907 gst_buffer_unref (out);
1914 GST_DEBUG_OBJECT (sink,
1915 "dropping sample out of segment time %" GST_TIME_FORMAT ", start %"
1916 GST_TIME_FORMAT, GST_TIME_ARGS (time),
1917 GST_TIME_ARGS (bsink->segment.start));
1923 GST_DEBUG_OBJECT (sink, "dropping late sample");
1930 GST_ELEMENT_ERROR (sink, STREAM, FORMAT, (NULL), ("failed to payload."));
1931 ret = GST_FLOW_ERROR;
1936 GST_DEBUG_OBJECT (sink, "ringbuffer not negotiated");
1937 GST_ELEMENT_ERROR (sink, STREAM, FORMAT, (NULL), ("sink not negotiated."));
1938 ret = GST_FLOW_NOT_NEGOTIATED;
1943 GST_DEBUG_OBJECT (sink, "wrong size");
1944 GST_ELEMENT_ERROR (sink, STREAM, WRONG_TYPE,
1945 (NULL), ("sink received buffer of wrong size."));
1946 ret = GST_FLOW_ERROR;
1951 GST_DEBUG_OBJECT (sink, "preroll got interrupted: %d (%s)", ret,
1952 gst_flow_get_name (ret));
1953 gst_buffer_unmap (buf, data, size);
1956 sync_latency_failed:
1958 GST_DEBUG_OBJECT (sink, "failed waiting for latency");
1964 * gst_audio_base_sink_create_ringbuffer:
1965 * @sink: a #GstAudioBaseSink.
1967 * Create and return the #GstAudioRingBuffer for @sink. This function will call the
1968 * ::create_ringbuffer vmethod and will set @sink as the parent of the returned
1969 * buffer (see gst_object_set_parent()).
1971 * Returns: The new ringbuffer of @sink.
1973 GstAudioRingBuffer *
1974 gst_audio_base_sink_create_ringbuffer (GstAudioBaseSink * sink)
1976 GstAudioBaseSinkClass *bclass;
1977 GstAudioRingBuffer *buffer = NULL;
1979 bclass = GST_AUDIO_BASE_SINK_GET_CLASS (sink);
1980 if (bclass->create_ringbuffer)
1981 buffer = bclass->create_ringbuffer (sink);
1984 gst_object_set_parent (GST_OBJECT (buffer), GST_OBJECT (sink));
1990 gst_audio_base_sink_callback (GstAudioRingBuffer * rbuf, guint8 * data,
1991 guint len, gpointer user_data)
1993 GstBaseSink *basesink;
1994 GstAudioBaseSink *sink;
1999 basesink = GST_BASE_SINK (user_data);
2000 sink = GST_AUDIO_BASE_SINK (user_data);
2002 GST_PAD_STREAM_LOCK (basesink->sinkpad);
2004 /* would be nice to arrange for pad_alloc_buffer to return data -- as it is we
2005 will copy twice, once into data, once into DMA */
2006 GST_LOG_OBJECT (basesink, "pulling %u bytes offset %" G_GUINT64_FORMAT
2007 " to fill audio buffer", len, basesink->offset);
2009 gst_pad_pull_range (basesink->sinkpad, basesink->segment.position, len,
2012 if (ret != GST_FLOW_OK) {
2013 if (ret == GST_FLOW_EOS)
2019 GST_BASE_SINK_PREROLL_LOCK (basesink);
2020 if (basesink->flushing)
2023 /* complete preroll and wait for PLAYING */
2024 ret = gst_base_sink_do_preroll (basesink, GST_MINI_OBJECT_CAST (buf));
2025 if (ret != GST_FLOW_OK)
2028 size = gst_buffer_get_size (buf);
2031 GST_INFO_OBJECT (basesink,
2032 "got different size than requested from sink pad: %u"
2033 " != %" G_GSIZE_FORMAT, len, size);
2034 len = MIN (size, len);
2037 basesink->segment.position += len;
2039 gst_buffer_extract (buf, 0, data, len);
2040 GST_BASE_SINK_PREROLL_UNLOCK (basesink);
2042 GST_PAD_STREAM_UNLOCK (basesink->sinkpad);
2048 GST_WARNING_OBJECT (basesink, "Got flow '%s' but can't return it: %d",
2049 gst_flow_get_name (ret), ret);
2050 gst_audio_ring_buffer_pause (rbuf);
2051 GST_PAD_STREAM_UNLOCK (basesink->sinkpad);
2056 /* FIXME: this is not quite correct; we'll be called endlessly until
2057 * the sink gets shut down; maybe we should set a flag somewhere, or
2058 * set segment.stop and segment.duration to the last sample or so */
2059 GST_DEBUG_OBJECT (sink, "EOS");
2060 gst_audio_base_sink_drain (sink);
2061 gst_audio_ring_buffer_pause (rbuf);
2062 gst_element_post_message (GST_ELEMENT_CAST (sink),
2063 gst_message_new_eos (GST_OBJECT_CAST (sink)));
2064 GST_PAD_STREAM_UNLOCK (basesink->sinkpad);
2068 GST_DEBUG_OBJECT (sink, "we are flushing");
2069 gst_audio_ring_buffer_pause (rbuf);
2070 GST_BASE_SINK_PREROLL_UNLOCK (basesink);
2071 GST_PAD_STREAM_UNLOCK (basesink->sinkpad);
2076 GST_DEBUG_OBJECT (sink, "error %s", gst_flow_get_name (ret));
2077 gst_audio_ring_buffer_pause (rbuf);
2078 GST_BASE_SINK_PREROLL_UNLOCK (basesink);
2079 GST_PAD_STREAM_UNLOCK (basesink->sinkpad);
2085 gst_audio_base_sink_activate_pull (GstBaseSink * basesink, gboolean active)
2088 GstAudioBaseSink *sink = GST_AUDIO_BASE_SINK (basesink);
2091 GST_DEBUG_OBJECT (basesink, "activating pull");
2093 gst_audio_ring_buffer_set_callback (sink->ringbuffer,
2094 gst_audio_base_sink_callback, sink);
2096 ret = gst_audio_ring_buffer_activate (sink->ringbuffer, TRUE);
2098 GST_DEBUG_OBJECT (basesink, "deactivating pull");
2099 gst_audio_ring_buffer_set_callback (sink->ringbuffer, NULL, NULL);
2100 ret = gst_audio_ring_buffer_activate (sink->ringbuffer, FALSE);
2107 /* should be called with the LOCK */
2108 static GstStateChangeReturn
2109 gst_audio_base_sink_async_play (GstBaseSink * basesink)
2111 GstAudioBaseSink *sink;
2113 sink = GST_AUDIO_BASE_SINK (basesink);
2115 GST_DEBUG_OBJECT (sink, "ringbuffer may start now");
2116 sink->priv->sync_latency = TRUE;
2117 gst_audio_ring_buffer_may_start (sink->ringbuffer, TRUE);
2118 if (basesink->pad_mode == GST_PAD_MODE_PULL) {
2119 /* we always start the ringbuffer in pull mode immediatly */
2120 gst_audio_ring_buffer_start (sink->ringbuffer);
2123 return GST_STATE_CHANGE_SUCCESS;
2127 static GstStateChangeReturn
2128 gst_audio_base_sink_change_state (GstElement * element,
2129 GstStateChange transition)
2131 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
2132 GstAudioBaseSink *sink = GST_AUDIO_BASE_SINK (element);
2134 switch (transition) {
2135 case GST_STATE_CHANGE_NULL_TO_READY:
2136 if (sink->ringbuffer == NULL) {
2137 gst_audio_clock_reset (GST_AUDIO_CLOCK (sink->provided_clock), 0);
2138 sink->ringbuffer = gst_audio_base_sink_create_ringbuffer (sink);
2140 if (!gst_audio_ring_buffer_open_device (sink->ringbuffer))
2143 case GST_STATE_CHANGE_READY_TO_PAUSED:
2144 sink->next_sample = -1;
2145 sink->priv->last_align = -1;
2146 sink->priv->eos_time = -1;
2147 sink->priv->discont_time = -1;
2148 gst_audio_ring_buffer_set_flushing (sink->ringbuffer, FALSE);
2149 gst_audio_ring_buffer_may_start (sink->ringbuffer, FALSE);
2151 /* Only post clock-provide messages if this is the clock that
2152 * we've created. If the subclass has overriden it the subclass
2153 * should post this messages whenever necessary */
2154 if (sink->provided_clock && GST_IS_AUDIO_CLOCK (sink->provided_clock) &&
2155 GST_AUDIO_CLOCK_CAST (sink->provided_clock)->func ==
2156 (GstAudioClockGetTimeFunc) gst_audio_base_sink_get_time)
2157 gst_element_post_message (element,
2158 gst_message_new_clock_provide (GST_OBJECT_CAST (element),
2159 sink->provided_clock, TRUE));
2161 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
2165 GST_OBJECT_LOCK (sink);
2166 GST_DEBUG_OBJECT (sink, "ringbuffer may start now");
2167 sink->priv->sync_latency = TRUE;
2168 eos = GST_BASE_SINK (sink)->eos;
2169 GST_OBJECT_UNLOCK (sink);
2171 gst_audio_ring_buffer_may_start (sink->ringbuffer, TRUE);
2172 if (GST_BASE_SINK_CAST (sink)->pad_mode == GST_PAD_MODE_PULL ||
2173 g_atomic_int_get (&sink->eos_rendering) || eos) {
2174 /* we always start the ringbuffer in pull mode immediatly */
2175 /* sync rendering on eos needs running clock,
2176 * and others need running clock when finished rendering eos */
2177 gst_audio_ring_buffer_start (sink->ringbuffer);
2181 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
2182 /* ringbuffer cannot start anymore */
2183 gst_audio_ring_buffer_may_start (sink->ringbuffer, FALSE);
2184 gst_audio_ring_buffer_pause (sink->ringbuffer);
2186 GST_OBJECT_LOCK (sink);
2187 sink->priv->sync_latency = FALSE;
2188 GST_OBJECT_UNLOCK (sink);
2190 case GST_STATE_CHANGE_PAUSED_TO_READY:
2191 /* Only post clock-lost messages if this is the clock that
2192 * we've created. If the subclass has overriden it the subclass
2193 * should post this messages whenever necessary */
2194 if (sink->provided_clock && GST_IS_AUDIO_CLOCK (sink->provided_clock) &&
2195 GST_AUDIO_CLOCK_CAST (sink->provided_clock)->func ==
2196 (GstAudioClockGetTimeFunc) gst_audio_base_sink_get_time)
2197 gst_element_post_message (element,
2198 gst_message_new_clock_lost (GST_OBJECT_CAST (element),
2199 sink->provided_clock));
2201 /* make sure we unblock before calling the parent state change
2202 * so it can grab the STREAM_LOCK */
2203 gst_audio_ring_buffer_set_flushing (sink->ringbuffer, TRUE);
2209 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
2211 switch (transition) {
2212 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
2213 /* stop slaving ourselves to the master, if any */
2214 gst_clock_set_master (sink->provided_clock, NULL);
2216 case GST_STATE_CHANGE_PAUSED_TO_READY:
2217 gst_audio_ring_buffer_activate (sink->ringbuffer, FALSE);
2218 gst_audio_ring_buffer_release (sink->ringbuffer);
2220 case GST_STATE_CHANGE_READY_TO_NULL:
2221 /* we release again here because the aqcuire happens when setting the
2222 * caps, which happens before we commit the state to PAUSED and thus the
2223 * PAUSED->READY state change (see above, where we release the ringbuffer)
2224 * might not be called when we get here. */
2225 gst_audio_ring_buffer_activate (sink->ringbuffer, FALSE);
2226 gst_audio_ring_buffer_release (sink->ringbuffer);
2227 gst_audio_ring_buffer_close_device (sink->ringbuffer);
2228 GST_OBJECT_LOCK (sink);
2229 gst_object_unparent (GST_OBJECT_CAST (sink->ringbuffer));
2230 sink->ringbuffer = NULL;
2231 GST_OBJECT_UNLOCK (sink);
2242 /* subclass must post a meaningful error message */
2243 GST_DEBUG_OBJECT (sink, "open failed");
2244 return GST_STATE_CHANGE_FAILURE;