gstreamer/libs/gst/base/gstbasesink.c
Wim Taymans d8d6ab79cb libs/gst/base/gstbasesink.c: Rearrange the latency query code. We always want to do the upstream query, even if we ar...
Original commit message from CVS:
* libs/gst/base/gstbasesink.c: (gst_base_sink_query_latency),
(gst_base_sink_send_event):
Rearrange the latency query code. We always want to do the upstream
query, even if we are not live so that the upstream elements can get the
latency results too. If we fail doing the query and we are live, we
return TRUE afterwards.
2008-04-25 07:22:16 +00:00

3449 lines
102 KiB
C

/* GStreamer
* Copyright (C) 2005-2007 Wim Taymans <wim.taymans@gmail.com>
*
* gstbasesink.c: Base class for sink elements
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:gstbasesink
* @short_description: Base class for sink elements
* @see_also: #GstBaseTransform, #GstBaseSource
*
* #GstBaseSink is the base class for sink elements in GStreamer, such as
* xvimagesink or filesink. It is a layer on top of #GstElement that provides a
* simplified interface to plugin writers. #GstBaseSink handles many details
* for you, for example: preroll, clock synchronization, state changes,
* activation in push or pull mode, and queries.
*
* In most cases, when writing sink elements, there is no need to implement
* class methods from #GstElement or to set functions on pads, because the
* #GstBaseSink infrastructure should be sufficient.
*
* #GstBaseSink provides support for exactly one sink pad, which should be
* named "sink". A sink implementation (subclass of #GstBaseSink) should
* install a pad template in its base_init function, like so:
* <programlisting>
* static void
* my_element_base_init (gpointer g_class)
* {
* GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class);
*
* // sinktemplate should be a #GstStaticPadTemplate with direction
* // #GST_PAD_SINK and name "sink"
* gst_element_class_add_pad_template (gstelement_class,
* gst_static_pad_template_get (&amp;sinktemplate));
* // see #GstElementDetails
* gst_element_class_set_details (gstelement_class, &amp;details);
* }
* </programlisting>
*
* #GstBaseSink will handle the prerolling correctly. This means that it will
* return #GST_STATE_CHANGE_ASYNC from a state change to PAUSED until the first
* buffer arrives in this element. The base class will call the
* #GstBaseSink::preroll vmethod with this preroll buffer and will then commit
* the state change to the next asynchronously pending state.
*
* When the element is set to PLAYING, #GstBaseSink will synchronise on the
* clock using the times returned from ::get_times. If this function returns
* #GST_CLOCK_TIME_NONE for the start time, no synchronisation will be done.
* Synchronisation can be disabled entirely by setting the object "sync"
* property to %FALSE.
*
* After synchronisation the virtual method #GstBaseSink::render will be called.
* Subclasses should minimally implement this method.
*
* Since 0.10.3 subclasses that synchronise on the clock in the ::render method
* are supported as well. These classes typically receive a buffer in the render
* method and can then potentially block on the clock while rendering. A typical
* example is an audiosink. Since 0.10.11 these subclasses can use
* gst_base_sink_wait_preroll() to perform the blocking wait.
*
* Upon receiving the EOS event in the PLAYING state, #GstBaseSink will wait
* for the clock to reach the time indicated by the stop time of the last
* ::get_times call before posting an EOS message. When the element receives
* EOS in PAUSED, preroll completes, the event is queued and an EOS message is
* posted when going to PLAYING.
*
* #GstBaseSink will internally use the #GST_EVENT_NEWSEGMENT events to schedule
* synchronisation and clipping of buffers. Buffers that fall completely outside
* of the current segment are dropped. Buffers that fall partially in the
* segment are rendered (and prerolled). Subclasses should do any subbuffer
* clipping themselves when needed.
*
* #GstBaseSink will by default report the current playback position in
* #GST_FORMAT_TIME based on the current clock time and segment information.
* If no clock has been set on the element, the query will be forwarded
* upstream.
*
* The ::set_caps function will be called when the subclass should configure
* itself to process a specific media type.
*
* The ::start and ::stop virtual methods will be called when resources should
* be allocated. Any ::preroll, ::render and ::set_caps function will be
* called between the ::start and ::stop calls.
*
* The ::event virtual method will be called when an event is received by
* #GstBaseSink. Normally this method should only be overriden by very specific
* elements (such as file sinks) which need to handle the newsegment event
* specially.
*
* #GstBaseSink provides an overridable ::buffer_alloc function that can be
* used by sinks that want to do reverse negotiation or to provide
* custom buffers (hardware buffers for example) to upstream elements.
*
* The ::unlock method is called when the elements should unblock any blocking
* operations they perform in the ::render method. This is mostly useful when
* the ::render method performs a blocking write on a file descriptor, for
* example.
*
* The max-lateness property affects how the sink deals with buffers that
* arrive too late in the sink. A buffer arrives too late in the sink when
* the presentation time (as a combination of the last segment, buffer
* timestamp and element base_time) plus the duration is before the current
* time of the clock.
* If the frame is later than max-lateness, the sink will drop the buffer
* without calling the render method.
* This feature is disabled if sync is disabled, the ::get-times method does
* not return a valid start time or max-lateness is set to -1 (the default).
* Subclasses can use gst_base_sink_set_max_lateness() to configure the
* max-lateness value.
*
* The qos property will enable the quality-of-service features of the basesink
* which gather statistics about the real-time performance of the clock
* synchronisation. For each buffer received in the sink, statistics are
* gathered and a QOS event is sent upstream with these numbers. This
* information can then be used by upstream elements to reduce their processing
* rate, for example.
*
* Since 0.10.15 the async property can be used to instruct the sink to never
* perform an ASYNC state change. This feature is mostly usable when dealing
* with non-synchronized streams or sparse streams.
*
* Last reviewed on 2007-08-29 (0.10.15)
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include "gstbasesink.h"
#include <gst/gstmarshal.h>
#include <gst/gst_private.h>
#include <gst/gst-i18n-lib.h>
GST_DEBUG_CATEGORY_STATIC (gst_base_sink_debug);
#define GST_CAT_DEFAULT gst_base_sink_debug
#define GST_BASE_SINK_GET_PRIVATE(obj) \
(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_BASE_SINK, GstBaseSinkPrivate))
/* FIXME, some stuff in ABI.data and other in Private...
* Make up your mind please.
*/
struct _GstBaseSinkPrivate
{
gint qos_enabled; /* ATOMIC */
gboolean async_enabled;
GstClockTimeDiff ts_offset;
/* start, stop of current buffer, stream time, used to report position */
GstClockTime current_sstart;
GstClockTime current_sstop;
/* start, stop and jitter of current buffer, running time */
GstClockTime current_rstart;
GstClockTime current_rstop;
GstClockTimeDiff current_jitter;
/* EOS sync time in running time */
GstClockTime eos_rtime;
/* last buffer that arrived in time, running time */
GstClockTime last_in_time;
/* when the last buffer left the sink, running time */
GstClockTime last_left;
/* running averages go here these are done on running time */
GstClockTime avg_pt;
GstClockTime avg_duration;
gdouble avg_rate;
/* these are done on system time. avg_jitter and avg_render are
* compared to eachother to see if the rendering time takes a
* huge amount of the processing, If so we are flooded with
* buffers. */
GstClockTime last_left_systime;
GstClockTime avg_jitter;
GstClockTime start, stop;
GstClockTime avg_render;
/* number of rendered and dropped frames */
guint64 rendered;
guint64 dropped;
/* latency stuff */
GstClockTime latency;
/* if we already commited the state */
gboolean commited;
/* when we received EOS */
gboolean received_eos;
/* when we are prerolled and able to report latency */
gboolean have_latency;
/* the last buffer we prerolled or rendered. Useful for making snapshots */
GstBuffer *last_buffer;
};
#define DO_RUNNING_AVG(avg,val,size) (((val) + ((size)-1) * (avg)) / (size))
/* generic running average, this has a neutral window size */
#define UPDATE_RUNNING_AVG(avg,val) DO_RUNNING_AVG(avg,val,8)
/* the windows for these running averages are experimentally obtained.
* possitive values get averaged more while negative values use a small
* window so we can react faster to badness. */
#define UPDATE_RUNNING_AVG_P(avg,val) DO_RUNNING_AVG(avg,val,16)
#define UPDATE_RUNNING_AVG_N(avg,val) DO_RUNNING_AVG(avg,val,4)
/* BaseSink properties */
#define DEFAULT_SIZE 1024
#define DEFAULT_CAN_ACTIVATE_PULL FALSE /* fixme: enable me */
#define DEFAULT_CAN_ACTIVATE_PUSH TRUE
#define DEFAULT_PREROLL_QUEUE_LEN 0
#define DEFAULT_SYNC TRUE
#define DEFAULT_MAX_LATENESS -1
#define DEFAULT_QOS FALSE
#define DEFAULT_ASYNC TRUE
#define DEFAULT_TS_OFFSET 0
enum
{
PROP_0,
PROP_PREROLL_QUEUE_LEN,
PROP_SYNC,
PROP_MAX_LATENESS,
PROP_QOS,
PROP_ASYNC,
PROP_TS_OFFSET,
PROP_LAST_BUFFER,
PROP_LAST
};
static GstElementClass *parent_class = NULL;
static void gst_base_sink_class_init (GstBaseSinkClass * klass);
static void gst_base_sink_init (GstBaseSink * trans, gpointer g_class);
static void gst_base_sink_finalize (GObject * object);
GType
gst_base_sink_get_type (void)
{
static GType base_sink_type = 0;
if (G_UNLIKELY (base_sink_type == 0)) {
static const GTypeInfo base_sink_info = {
sizeof (GstBaseSinkClass),
NULL,
NULL,
(GClassInitFunc) gst_base_sink_class_init,
NULL,
NULL,
sizeof (GstBaseSink),
0,
(GInstanceInitFunc) gst_base_sink_init,
};
base_sink_type = g_type_register_static (GST_TYPE_ELEMENT,
"GstBaseSink", &base_sink_info, G_TYPE_FLAG_ABSTRACT);
}
return base_sink_type;
}
static void gst_base_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_base_sink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static gboolean gst_base_sink_send_event (GstElement * element,
GstEvent * event);
static gboolean gst_base_sink_query (GstElement * element, GstQuery * query);
static GstCaps *gst_base_sink_get_caps (GstBaseSink * sink);
static gboolean gst_base_sink_set_caps (GstBaseSink * sink, GstCaps * caps);
static GstFlowReturn gst_base_sink_buffer_alloc (GstBaseSink * sink,
guint64 offset, guint size, GstCaps * caps, GstBuffer ** buf);
static void gst_base_sink_get_times (GstBaseSink * basesink, GstBuffer * buffer,
GstClockTime * start, GstClockTime * end);
static gboolean gst_base_sink_set_flushing (GstBaseSink * basesink,
GstPad * pad, gboolean flushing);
static gboolean gst_base_sink_default_activate_pull (GstBaseSink * basesink,
gboolean active);
static GstStateChangeReturn gst_base_sink_change_state (GstElement * element,
GstStateChange transition);
static GstFlowReturn gst_base_sink_chain (GstPad * pad, GstBuffer * buffer);
static void gst_base_sink_loop (GstPad * pad);
static gboolean gst_base_sink_pad_activate (GstPad * pad);
static gboolean gst_base_sink_pad_activate_push (GstPad * pad, gboolean active);
static gboolean gst_base_sink_pad_activate_pull (GstPad * pad, gboolean active);
static gboolean gst_base_sink_event (GstPad * pad, GstEvent * event);
static gboolean gst_base_sink_peer_query (GstBaseSink * sink, GstQuery * query);
/* check if an object was too late */
static gboolean gst_base_sink_is_too_late (GstBaseSink * basesink,
GstMiniObject * obj, GstClockTime start, GstClockTime stop,
GstClockReturn status, GstClockTimeDiff jitter);
static void
gst_base_sink_class_init (GstBaseSinkClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = G_OBJECT_CLASS (klass);
gstelement_class = GST_ELEMENT_CLASS (klass);
GST_DEBUG_CATEGORY_INIT (gst_base_sink_debug, "basesink", 0,
"basesink element");
g_type_class_add_private (klass, sizeof (GstBaseSinkPrivate));
parent_class = g_type_class_peek_parent (klass);
gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_base_sink_finalize);
gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_base_sink_set_property);
gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_base_sink_get_property);
/* FIXME, this next value should be configured using an event from the
* upstream element, ie, the BUFFER_SIZE event. */
g_object_class_install_property (gobject_class, PROP_PREROLL_QUEUE_LEN,
g_param_spec_uint ("preroll-queue-len", "Preroll queue length",
"Number of buffers to queue during preroll", 0, G_MAXUINT,
DEFAULT_PREROLL_QUEUE_LEN,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_SYNC,
g_param_spec_boolean ("sync", "Sync", "Sync on the clock", DEFAULT_SYNC,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_MAX_LATENESS,
g_param_spec_int64 ("max-lateness", "Max Lateness",
"Maximum number of nanoseconds that a buffer can be late before it "
"is dropped (-1 unlimited)", -1, G_MAXINT64, DEFAULT_MAX_LATENESS,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_QOS,
g_param_spec_boolean ("qos", "Qos",
"Generate Quality-of-Service events upstream", DEFAULT_QOS,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstBaseSink:async
*
* If set to #TRUE, the basesink will perform asynchronous state changes.
* When set to #FALSE, the sink will not signal the parent when it prerolls.
* Use this option when dealing with sparse streams or when synchronisation is
* not required.
*
* Since: 0.10.15
*/
g_object_class_install_property (gobject_class, PROP_ASYNC,
g_param_spec_boolean ("async", "Async",
"Go asynchronously to PAUSED", DEFAULT_ASYNC,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstBaseSink:ts-offset
*
* Controls the final synchronisation, a negative value will render the buffer
* earlier while a positive value delays playback. This property can be
* used to fix synchronisation in bad files.
*
* Since: 0.10.15
*/
g_object_class_install_property (gobject_class, PROP_TS_OFFSET,
g_param_spec_int64 ("ts-offset", "TS Offset",
"Timestamp offset in nanoseconds", G_MININT64, G_MAXINT64,
DEFAULT_TS_OFFSET, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstBaseSink:last-buffer
*
* The last buffer that arrived in the sink and was used for preroll or for
* rendering. This property can be used to generate thumbnails. This property
* can be NULL when the sink has not yet received a bufer.
*
* Since: 0.10.15
*/
g_object_class_install_property (gobject_class, PROP_LAST_BUFFER,
gst_param_spec_mini_object ("last-buffer", "Last Buffer",
"The last buffer received in the sink", GST_TYPE_BUFFER,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_base_sink_change_state);
gstelement_class->send_event = GST_DEBUG_FUNCPTR (gst_base_sink_send_event);
gstelement_class->query = GST_DEBUG_FUNCPTR (gst_base_sink_query);
klass->get_caps = GST_DEBUG_FUNCPTR (gst_base_sink_get_caps);
klass->set_caps = GST_DEBUG_FUNCPTR (gst_base_sink_set_caps);
klass->buffer_alloc = GST_DEBUG_FUNCPTR (gst_base_sink_buffer_alloc);
klass->get_times = GST_DEBUG_FUNCPTR (gst_base_sink_get_times);
klass->activate_pull =
GST_DEBUG_FUNCPTR (gst_base_sink_default_activate_pull);
}
static GstCaps *
gst_base_sink_pad_getcaps (GstPad * pad)
{
GstBaseSinkClass *bclass;
GstBaseSink *bsink;
GstCaps *caps = NULL;
bsink = GST_BASE_SINK (gst_pad_get_parent (pad));
bclass = GST_BASE_SINK_GET_CLASS (bsink);
if (bclass->get_caps)
caps = bclass->get_caps (bsink);
if (caps == NULL) {
GstPadTemplate *pad_template;
pad_template =
gst_element_class_get_pad_template (GST_ELEMENT_CLASS (bclass), "sink");
if (pad_template != NULL) {
caps = gst_caps_ref (gst_pad_template_get_caps (pad_template));
}
}
gst_object_unref (bsink);
return caps;
}
static gboolean
gst_base_sink_pad_setcaps (GstPad * pad, GstCaps * caps)
{
GstBaseSinkClass *bclass;
GstBaseSink *bsink;
gboolean res = TRUE;
bsink = GST_BASE_SINK (gst_pad_get_parent (pad));
bclass = GST_BASE_SINK_GET_CLASS (bsink);
if (bsink->pad_mode == GST_ACTIVATE_PULL) {
GstPad *peer = gst_pad_get_peer (pad);
if (peer)
res = gst_pad_set_caps (peer, caps);
else
res = FALSE;
if (!res)
GST_DEBUG_OBJECT (bsink, "peer setcaps() failed");
}
if (res && bclass->set_caps)
res = bclass->set_caps (bsink, caps);
gst_object_unref (bsink);
return res;
}
static void
gst_base_sink_pad_fixate (GstPad * pad, GstCaps * caps)
{
GstBaseSinkClass *bclass;
GstBaseSink *bsink;
bsink = GST_BASE_SINK (gst_pad_get_parent (pad));
bclass = GST_BASE_SINK_GET_CLASS (bsink);
if (bclass->fixate)
bclass->fixate (bsink, caps);
gst_object_unref (bsink);
}
static GstFlowReturn
gst_base_sink_pad_buffer_alloc (GstPad * pad, guint64 offset, guint size,
GstCaps * caps, GstBuffer ** buf)
{
GstBaseSinkClass *bclass;
GstBaseSink *bsink;
GstFlowReturn result = GST_FLOW_OK;
bsink = GST_BASE_SINK (gst_pad_get_parent (pad));
bclass = GST_BASE_SINK_GET_CLASS (bsink);
if (bclass->buffer_alloc)
result = bclass->buffer_alloc (bsink, offset, size, caps, buf);
else
*buf = NULL; /* fallback in gstpad.c will allocate generic buffer */
gst_object_unref (bsink);
return result;
}
static void
gst_base_sink_init (GstBaseSink * basesink, gpointer g_class)
{
GstPadTemplate *pad_template;
GstBaseSinkPrivate *priv;
basesink->priv = priv = GST_BASE_SINK_GET_PRIVATE (basesink);
pad_template =
gst_element_class_get_pad_template (GST_ELEMENT_CLASS (g_class), "sink");
g_return_if_fail (pad_template != NULL);
basesink->sinkpad = gst_pad_new_from_template (pad_template, "sink");
gst_pad_set_getcaps_function (basesink->sinkpad,
GST_DEBUG_FUNCPTR (gst_base_sink_pad_getcaps));
gst_pad_set_setcaps_function (basesink->sinkpad,
GST_DEBUG_FUNCPTR (gst_base_sink_pad_setcaps));
gst_pad_set_fixatecaps_function (basesink->sinkpad,
GST_DEBUG_FUNCPTR (gst_base_sink_pad_fixate));
gst_pad_set_bufferalloc_function (basesink->sinkpad,
GST_DEBUG_FUNCPTR (gst_base_sink_pad_buffer_alloc));
gst_pad_set_activate_function (basesink->sinkpad,
GST_DEBUG_FUNCPTR (gst_base_sink_pad_activate));
gst_pad_set_activatepush_function (basesink->sinkpad,
GST_DEBUG_FUNCPTR (gst_base_sink_pad_activate_push));
gst_pad_set_activatepull_function (basesink->sinkpad,
GST_DEBUG_FUNCPTR (gst_base_sink_pad_activate_pull));
gst_pad_set_event_function (basesink->sinkpad,
GST_DEBUG_FUNCPTR (gst_base_sink_event));
gst_pad_set_chain_function (basesink->sinkpad,
GST_DEBUG_FUNCPTR (gst_base_sink_chain));
gst_element_add_pad (GST_ELEMENT_CAST (basesink), basesink->sinkpad);
basesink->pad_mode = GST_ACTIVATE_NONE;
basesink->preroll_queue = g_queue_new ();
basesink->abidata.ABI.clip_segment = gst_segment_new ();
priv->have_latency = FALSE;
basesink->can_activate_push = DEFAULT_CAN_ACTIVATE_PUSH;
basesink->can_activate_pull = DEFAULT_CAN_ACTIVATE_PULL;
basesink->sync = DEFAULT_SYNC;
basesink->abidata.ABI.max_lateness = DEFAULT_MAX_LATENESS;
g_atomic_int_set (&priv->qos_enabled, DEFAULT_QOS);
priv->async_enabled = DEFAULT_ASYNC;
priv->ts_offset = DEFAULT_TS_OFFSET;
GST_OBJECT_FLAG_SET (basesink, GST_ELEMENT_IS_SINK);
}
static void
gst_base_sink_finalize (GObject * object)
{
GstBaseSink *basesink;
basesink = GST_BASE_SINK (object);
g_queue_free (basesink->preroll_queue);
gst_segment_free (basesink->abidata.ABI.clip_segment);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
/**
* gst_base_sink_set_sync:
* @sink: the sink
* @sync: the new sync value.
*
* Configures @sink to synchronize on the clock or not. When
* @sync is FALSE, incomming samples will be played as fast as
* possible. If @sync is TRUE, the timestamps of the incomming
* buffers will be used to schedule the exact render time of its
* contents.
*
* Since: 0.10.4
*/
void
gst_base_sink_set_sync (GstBaseSink * sink, gboolean sync)
{
g_return_if_fail (GST_IS_BASE_SINK (sink));
GST_OBJECT_LOCK (sink);
sink->sync = sync;
GST_OBJECT_UNLOCK (sink);
}
/**
* gst_base_sink_get_sync:
* @sink: the sink
*
* Checks if @sink is currently configured to synchronize against the
* clock.
*
* Returns: TRUE if the sink is configured to synchronize against the clock.
*
* Since: 0.10.4
*/
gboolean
gst_base_sink_get_sync (GstBaseSink * sink)
{
gboolean res;
g_return_val_if_fail (GST_IS_BASE_SINK (sink), FALSE);
GST_OBJECT_LOCK (sink);
res = sink->sync;
GST_OBJECT_UNLOCK (sink);
return res;
}
/**
* gst_base_sink_set_max_lateness:
* @sink: the sink
* @max_lateness: the new max lateness value.
*
* Sets the new max lateness value to @max_lateness. This value is
* used to decide if a buffer should be dropped or not based on the
* buffer timestamp and the current clock time. A value of -1 means
* an unlimited time.
*
* Since: 0.10.4
*/
void
gst_base_sink_set_max_lateness (GstBaseSink * sink, gint64 max_lateness)
{
g_return_if_fail (GST_IS_BASE_SINK (sink));
GST_OBJECT_LOCK (sink);
sink->abidata.ABI.max_lateness = max_lateness;
GST_OBJECT_UNLOCK (sink);
}
/**
* gst_base_sink_get_max_lateness:
* @sink: the sink
*
* Gets the max lateness value. See gst_base_sink_set_max_lateness for
* more details.
*
* Returns: The maximum time in nanoseconds that a buffer can be late
* before it is dropped and not rendered. A value of -1 means an
* unlimited time.
*
* Since: 0.10.4
*/
gint64
gst_base_sink_get_max_lateness (GstBaseSink * sink)
{
gint64 res;
g_return_val_if_fail (GST_IS_BASE_SINK (sink), -1);
GST_OBJECT_LOCK (sink);
res = sink->abidata.ABI.max_lateness;
GST_OBJECT_UNLOCK (sink);
return res;
}
/**
* gst_base_sink_set_qos_enabled:
* @sink: the sink
* @enabled: the new qos value.
*
* Configures @sink to send Quality-of-Service events upstream.
*
* Since: 0.10.5
*/
void
gst_base_sink_set_qos_enabled (GstBaseSink * sink, gboolean enabled)
{
g_return_if_fail (GST_IS_BASE_SINK (sink));
g_atomic_int_set (&sink->priv->qos_enabled, enabled);
}
/**
* gst_base_sink_is_qos_enabled:
* @sink: the sink
*
* Checks if @sink is currently configured to send Quality-of-Service events
* upstream.
*
* Returns: TRUE if the sink is configured to perform Quality-of-Service.
*
* Since: 0.10.5
*/
gboolean
gst_base_sink_is_qos_enabled (GstBaseSink * sink)
{
gboolean res;
g_return_val_if_fail (GST_IS_BASE_SINK (sink), FALSE);
res = g_atomic_int_get (&sink->priv->qos_enabled);
return res;
}
/**
* gst_base_sink_set_async_enabled:
* @sink: the sink
* @enabled: the new async value.
*
* Configures @sink to perform all state changes asynchronusly. When async is
* disabled, the sink will immediatly go to PAUSED instead of waiting for a
* preroll buffer. This feature is usefull if the sink does not synchronize
* against the clock or when it is dealing with sparse streams.
*
* Since: 0.10.15
*/
void
gst_base_sink_set_async_enabled (GstBaseSink * sink, gboolean enabled)
{
g_return_if_fail (GST_IS_BASE_SINK (sink));
GST_PAD_PREROLL_LOCK (sink->sinkpad);
sink->priv->async_enabled = enabled;
GST_PAD_PREROLL_UNLOCK (sink->sinkpad);
}
/**
* gst_base_sink_is_async_enabled:
* @sink: the sink
*
* Checks if @sink is currently configured to perform asynchronous state
* changes to PAUSED.
*
* Returns: TRUE if the sink is configured to perform asynchronous state
* changes.
*
* Since: 0.10.15
*/
gboolean
gst_base_sink_is_async_enabled (GstBaseSink * sink)
{
gboolean res;
g_return_val_if_fail (GST_IS_BASE_SINK (sink), FALSE);
GST_PAD_PREROLL_LOCK (sink->sinkpad);
res = sink->priv->async_enabled;
GST_PAD_PREROLL_UNLOCK (sink->sinkpad);
return res;
}
/**
* gst_base_sink_set_ts_offset:
* @sink: the sink
* @offset: the new offset
*
* Adjust the synchronisation of @sink with @offset. A negative value will
* render buffers earlier than their timestamp. A positive value will delay
* rendering. This function can be used to fix playback of badly timestamped
* buffers.
*
* Since: 0.10.15
*/
void
gst_base_sink_set_ts_offset (GstBaseSink * sink, GstClockTimeDiff offset)
{
g_return_if_fail (GST_IS_BASE_SINK (sink));
GST_OBJECT_LOCK (sink);
sink->priv->ts_offset = offset;
GST_OBJECT_UNLOCK (sink);
}
/**
* gst_base_sink_get_ts_offset:
* @sink: the sink
*
* Get the synchronisation offset of @sink.
*
* Returns: The synchronisation offset.
*
* Since: 0.10.15
*/
GstClockTimeDiff
gst_base_sink_get_ts_offset (GstBaseSink * sink)
{
GstClockTimeDiff res;
g_return_val_if_fail (GST_IS_BASE_SINK (sink), 0);
GST_OBJECT_LOCK (sink);
res = sink->priv->ts_offset;
GST_OBJECT_UNLOCK (sink);
return res;
}
/**
* gst_base_sink_get_last_buffer:
* @sink: the sink
*
* Get the last buffer that arrived in the sink and was used for preroll or for
* rendering. This property can be used to generate thumbnails.
*
* The #GstCaps on the buffer can be used to determine the type of the buffer.
*
* Returns: a #GstBuffer. gst_buffer_unref() after usage. This function returns
* NULL when no buffer has arrived in the sink yet or when the sink is not in
* PAUSED or PLAYING.
*
* Since: 0.10.15
*/
GstBuffer *
gst_base_sink_get_last_buffer (GstBaseSink * sink)
{
GstBuffer *res;
g_return_val_if_fail (GST_IS_BASE_SINK (sink), NULL);
GST_OBJECT_LOCK (sink);
if ((res = sink->priv->last_buffer))
gst_buffer_ref (res);
GST_OBJECT_UNLOCK (sink);
return res;
}
static void
gst_base_sink_set_last_buffer (GstBaseSink * sink, GstBuffer * buffer)
{
GstBuffer *old;
if (buffer)
gst_buffer_ref (buffer);
GST_OBJECT_LOCK (sink);
old = sink->priv->last_buffer;
sink->priv->last_buffer = buffer;
GST_OBJECT_UNLOCK (sink);
if (old)
gst_buffer_unref (old);
}
/**
* gst_base_sink_get_latency:
* @sink: the sink
*
* Get the currently configured latency.
*
* Returns: The configured latency.
*
* Since: 0.10.12
*/
GstClockTime
gst_base_sink_get_latency (GstBaseSink * sink)
{
GstClockTime res;
GST_OBJECT_LOCK (sink);
res = sink->priv->latency;
GST_OBJECT_UNLOCK (sink);
return res;
}
/**
* gst_base_sink_query_latency:
* @sink: the sink
* @live: if the sink is live
* @upstream_live: if an upstream element is live
* @min_latency: the min latency of the upstream elements
* @max_latency: the max latency of the upstream elements
*
* Query the sink for the latency parameters. The latency will be queried from
* the upstream elements. @live will be TRUE if @sink is configured to
* synchronize against the clock. @upstream_live will be TRUE if an upstream
* element is live.
*
* If both @live and @upstream_live are TRUE, the sink will want to compensate
* for the latency introduced by the upstream elements by setting the
* @min_latency to a strictly possitive value.
*
* This function is mostly used by subclasses.
*
* Returns: TRUE if the query succeeded.
*
* Since: 0.10.12
*/
gboolean
gst_base_sink_query_latency (GstBaseSink * sink, gboolean * live,
gboolean * upstream_live, GstClockTime * min_latency,
GstClockTime * max_latency)
{
gboolean l, us_live, res, have_latency;
GstClockTime min, max;
GstQuery *query;
GstClockTime us_min, us_max;
/* we are live when we sync to the clock */
GST_OBJECT_LOCK (sink);
l = sink->sync;
have_latency = sink->priv->have_latency;
GST_OBJECT_UNLOCK (sink);
/* assume no latency */
min = 0;
max = -1;
us_live = FALSE;
if (have_latency) {
GST_DEBUG_OBJECT (sink, "we are ready for LATENCY query");
/* we are ready for a latency query this is when we preroll or when we are
* not async. */
query = gst_query_new_latency ();
/* ask the peer for the latency */
if ((res = gst_base_sink_peer_query (sink, query))) {
/* get upstream min and max latency */
gst_query_parse_latency (query, &us_live, &us_min, &us_max);
if (us_live) {
/* upstream live, use its latency, subclasses should use these
* values to create the complete latency. */
min = us_min;
max = us_max;
}
}
gst_query_unref (query);
} else {
GST_DEBUG_OBJECT (sink, "we are not yet ready for LATENCY query");
res = FALSE;
}
/* not live, we tried to do the query, if it failed we return TRUE anyway */
if (!res) {
if (!l) {
res = TRUE;
GST_DEBUG_OBJECT (sink, "latency query failed but we are not live");
} else {
GST_DEBUG_OBJECT (sink, "latency query failed and we are live");
}
}
if (res) {
GST_DEBUG_OBJECT (sink, "latency query: live: %d, have_latency %d,"
" upstream: %d, min %" GST_TIME_FORMAT ", max %" GST_TIME_FORMAT, l,
have_latency, us_live, GST_TIME_ARGS (min), GST_TIME_ARGS (max));
if (live)
*live = l;
if (upstream_live)
*upstream_live = us_live;
if (min_latency)
*min_latency = min;
if (max_latency)
*max_latency = max;
}
return res;
}
static void
gst_base_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstBaseSink *sink = GST_BASE_SINK (object);
switch (prop_id) {
case PROP_PREROLL_QUEUE_LEN:
/* preroll lock necessary to serialize with finish_preroll */
GST_PAD_PREROLL_LOCK (sink->sinkpad);
sink->preroll_queue_max_len = g_value_get_uint (value);
GST_PAD_PREROLL_UNLOCK (sink->sinkpad);
break;
case PROP_SYNC:
gst_base_sink_set_sync (sink, g_value_get_boolean (value));
break;
case PROP_MAX_LATENESS:
gst_base_sink_set_max_lateness (sink, g_value_get_int64 (value));
break;
case PROP_QOS:
gst_base_sink_set_qos_enabled (sink, g_value_get_boolean (value));
break;
case PROP_ASYNC:
gst_base_sink_set_async_enabled (sink, g_value_get_boolean (value));
break;
case PROP_TS_OFFSET:
gst_base_sink_set_ts_offset (sink, g_value_get_int64 (value));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_base_sink_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstBaseSink *sink = GST_BASE_SINK (object);
switch (prop_id) {
case PROP_PREROLL_QUEUE_LEN:
GST_PAD_PREROLL_LOCK (sink->sinkpad);
g_value_set_uint (value, sink->preroll_queue_max_len);
GST_PAD_PREROLL_UNLOCK (sink->sinkpad);
break;
case PROP_SYNC:
g_value_set_boolean (value, gst_base_sink_get_sync (sink));
break;
case PROP_MAX_LATENESS:
g_value_set_int64 (value, gst_base_sink_get_max_lateness (sink));
break;
case PROP_QOS:
g_value_set_boolean (value, gst_base_sink_is_qos_enabled (sink));
break;
case PROP_ASYNC:
g_value_set_boolean (value, gst_base_sink_is_async_enabled (sink));
break;
case PROP_TS_OFFSET:
g_value_set_int64 (value, gst_base_sink_get_ts_offset (sink));
break;
case PROP_LAST_BUFFER:
gst_value_take_buffer (value, gst_base_sink_get_last_buffer (sink));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstCaps *
gst_base_sink_get_caps (GstBaseSink * sink)
{
return NULL;
}
static gboolean
gst_base_sink_set_caps (GstBaseSink * sink, GstCaps * caps)
{
return TRUE;
}
static GstFlowReturn
gst_base_sink_buffer_alloc (GstBaseSink * sink, guint64 offset, guint size,
GstCaps * caps, GstBuffer ** buf)
{
*buf = NULL;
return GST_FLOW_OK;
}
/* with PREROLL_LOCK, STREAM_LOCK */
static void
gst_base_sink_preroll_queue_flush (GstBaseSink * basesink, GstPad * pad)
{
GstMiniObject *obj;
GST_DEBUG_OBJECT (basesink, "flushing queue %p", basesink);
while ((obj = g_queue_pop_head (basesink->preroll_queue))) {
GST_DEBUG_OBJECT (basesink, "popped %p", obj);
gst_mini_object_unref (obj);
}
/* we can't have EOS anymore now */
basesink->eos = FALSE;
basesink->priv->received_eos = FALSE;
basesink->have_preroll = FALSE;
basesink->eos_queued = FALSE;
basesink->preroll_queued = 0;
basesink->buffers_queued = 0;
basesink->events_queued = 0;
/* can't report latency anymore until we preroll again */
if (basesink->priv->async_enabled) {
GST_OBJECT_LOCK (basesink);
basesink->priv->have_latency = FALSE;
GST_OBJECT_UNLOCK (basesink);
}
/* and signal any waiters now */
GST_PAD_PREROLL_SIGNAL (pad);
}
/* with STREAM_LOCK, configures given segment with the event information. */
static void
gst_base_sink_configure_segment (GstBaseSink * basesink, GstPad * pad,
GstEvent * event, GstSegment * segment)
{
gboolean update;
gdouble rate, arate;
GstFormat format;
gint64 start;
gint64 stop;
gint64 time;
/* the newsegment event is needed to bring the buffer timestamps to the
* stream time and to drop samples outside of the playback segment. */
gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
&start, &stop, &time);
/* The segment is protected with both the STREAM_LOCK and the OBJECT_LOCK.
* We protect with the OBJECT_LOCK so that we can use the values to
* safely answer a POSITION query. */
GST_OBJECT_LOCK (basesink);
gst_segment_set_newsegment_full (segment, update, rate, arate, format, start,
stop, time);
if (format == GST_FORMAT_TIME) {
GST_DEBUG_OBJECT (basesink,
"configured NEWSEGMENT update %d, rate %lf, applied rate %lf, "
"format GST_FORMAT_TIME, "
"%" GST_TIME_FORMAT " -- %" GST_TIME_FORMAT
", time %" GST_TIME_FORMAT ", accum %" GST_TIME_FORMAT,
update, rate, arate, GST_TIME_ARGS (segment->start),
GST_TIME_ARGS (segment->stop), GST_TIME_ARGS (segment->time),
GST_TIME_ARGS (segment->accum));
} else {
GST_DEBUG_OBJECT (basesink,
"configured NEWSEGMENT update %d, rate %lf, applied rate %lf, "
"format %d, "
"%" G_GINT64_FORMAT " -- %" G_GINT64_FORMAT ", time %"
G_GINT64_FORMAT ", accum %" G_GINT64_FORMAT, update, rate, arate,
segment->format, segment->start, segment->stop, segment->time,
segment->accum);
}
GST_OBJECT_UNLOCK (basesink);
}
/* with PREROLL_LOCK, STREAM_LOCK */
static gboolean
gst_base_sink_commit_state (GstBaseSink * basesink)
{
/* commit state and proceed to next pending state */
GstState current, next, pending, post_pending;
gboolean post_paused = FALSE;
gboolean post_async_done = FALSE;
gboolean post_playing = FALSE;
gboolean sync;
/* we are certainly not playing async anymore now */
basesink->playing_async = FALSE;
GST_OBJECT_LOCK (basesink);
current = GST_STATE (basesink);
next = GST_STATE_NEXT (basesink);
pending = GST_STATE_PENDING (basesink);
post_pending = pending;
sync = basesink->sync;
switch (pending) {
case GST_STATE_PLAYING:
{
GstBaseSinkClass *bclass;
GstStateChangeReturn ret;
bclass = GST_BASE_SINK_GET_CLASS (basesink);
GST_DEBUG_OBJECT (basesink, "commiting state to PLAYING");
basesink->need_preroll = FALSE;
post_async_done = TRUE;
basesink->priv->commited = TRUE;
post_playing = TRUE;
/* post PAUSED too when we were READY */
if (current == GST_STATE_READY) {
post_paused = TRUE;
}
/* make sure we notify the subclass of async playing */
if (bclass->async_play) {
ret = bclass->async_play (basesink);
if (ret == GST_STATE_CHANGE_FAILURE)
goto async_failed;
}
break;
}
case GST_STATE_PAUSED:
GST_DEBUG_OBJECT (basesink, "commiting state to PAUSED");
post_paused = TRUE;
post_async_done = TRUE;
basesink->priv->commited = TRUE;
post_pending = GST_STATE_VOID_PENDING;
break;
case GST_STATE_READY:
case GST_STATE_NULL:
goto stopping;
case GST_STATE_VOID_PENDING:
goto nothing_pending;
default:
break;
}
/* we can report latency queries now */
basesink->priv->have_latency = TRUE;
GST_STATE (basesink) = pending;
GST_STATE_NEXT (basesink) = GST_STATE_VOID_PENDING;
GST_STATE_PENDING (basesink) = GST_STATE_VOID_PENDING;
GST_STATE_RETURN (basesink) = GST_STATE_CHANGE_SUCCESS;
GST_OBJECT_UNLOCK (basesink);
if (post_paused) {
GST_DEBUG_OBJECT (basesink, "posting PAUSED state change message");
gst_element_post_message (GST_ELEMENT_CAST (basesink),
gst_message_new_state_changed (GST_OBJECT_CAST (basesink),
current, next, post_pending));
}
if (post_async_done) {
GST_DEBUG_OBJECT (basesink, "posting async-done message");
gst_element_post_message (GST_ELEMENT_CAST (basesink),
gst_message_new_async_done (GST_OBJECT_CAST (basesink)));
}
if (post_playing) {
GST_DEBUG_OBJECT (basesink, "posting PLAYING state change message");
gst_element_post_message (GST_ELEMENT_CAST (basesink),
gst_message_new_state_changed (GST_OBJECT_CAST (basesink),
next, pending, GST_STATE_VOID_PENDING));
}
GST_STATE_BROADCAST (basesink);
return TRUE;
nothing_pending:
{
/* Depending on the state, set our vars. We get in this situation when the
* state change function got a change to update the state vars before the
* streaming thread did. This is fine but we need to make sure that we
* update the need_preroll var since it was TRUE when we got here and might
* become FALSE if we got to PLAYING. */
GST_DEBUG_OBJECT (basesink, "nothing to commit, now in %s",
gst_element_state_get_name (current));
switch (current) {
case GST_STATE_PLAYING:
basesink->need_preroll = FALSE;
break;
case GST_STATE_PAUSED:
basesink->need_preroll = TRUE;
break;
default:
basesink->need_preroll = FALSE;
basesink->flushing = TRUE;
break;
}
/* we can report latency queries now */
basesink->priv->have_latency = TRUE;
GST_OBJECT_UNLOCK (basesink);
return TRUE;
}
stopping:
{
/* app is going to READY */
GST_DEBUG_OBJECT (basesink, "stopping");
basesink->need_preroll = FALSE;
basesink->flushing = TRUE;
GST_OBJECT_UNLOCK (basesink);
return FALSE;
}
async_failed:
{
GST_DEBUG_OBJECT (basesink, "async commit failed");
GST_STATE_RETURN (basesink) = GST_STATE_CHANGE_FAILURE;
GST_OBJECT_UNLOCK (basesink);
return FALSE;
}
}
/* with STREAM_LOCK, PREROLL_LOCK
*
* Returns TRUE if the object needs synchronisation and takes therefore
* part in prerolling.
*
* rsstart/rsstop contain the start/stop in stream time.
* rrstart/rrstop contain the start/stop in running time.
*/
static gboolean
gst_base_sink_get_sync_times (GstBaseSink * basesink, GstMiniObject * obj,
GstClockTime * rsstart, GstClockTime * rsstop,
GstClockTime * rrstart, GstClockTime * rrstop, gboolean * do_sync,
GstSegment * segment)
{
GstBaseSinkClass *bclass;
GstBuffer *buffer;
GstClockTime start, stop; /* raw start/stop timestamps */
gint64 cstart, cstop; /* clipped raw timestamps */
gint64 rstart, rstop; /* clipped timestamps converted to running time */
GstClockTime sstart, sstop; /* clipped timestamps converted to stream time */
GstFormat format;
GstBaseSinkPrivate *priv;
priv = basesink->priv;
/* start with nothing */
start = stop = sstart = sstop = rstart = rstop = -1;
if (G_UNLIKELY (GST_IS_EVENT (obj))) {
GstEvent *event = GST_EVENT_CAST (obj);
switch (GST_EVENT_TYPE (event)) {
/* EOS event needs syncing */
case GST_EVENT_EOS:
sstart = sstop = priv->current_sstop;
rstart = rstop = priv->eos_rtime;
*do_sync = rstart != -1;
GST_DEBUG_OBJECT (basesink, "sync times for EOS %" GST_TIME_FORMAT,
GST_TIME_ARGS (rstart));
goto done;
/* other events do not need syncing */
/* FIXME, maybe NEWSEGMENT might need synchronisation
* since the POSITION query depends on accumulated times and
* we cannot accumulate the current segment before the previous
* one completed.
*/
default:
return FALSE;
}
}
/* else do buffer sync code */
buffer = GST_BUFFER_CAST (obj);
bclass = GST_BASE_SINK_GET_CLASS (basesink);
/* just get the times to see if we need syncing */
if (bclass->get_times)
bclass->get_times (basesink, buffer, &start, &stop);
if (start == -1) {
gst_base_sink_get_times (basesink, buffer, &start, &stop);
*do_sync = FALSE;
} else {
*do_sync = TRUE;
}
GST_DEBUG_OBJECT (basesink, "got times start: %" GST_TIME_FORMAT
", stop: %" GST_TIME_FORMAT ", do_sync %d", GST_TIME_ARGS (start),
GST_TIME_ARGS (stop), *do_sync);
/* collect segment and format for code clarity */
format = segment->format;
/* no timestamp clipping if we did not * get a TIME segment format */
if (G_UNLIKELY (format != GST_FORMAT_TIME)) {
cstart = start;
cstop = stop;
/* do running and stream time in TIME format */
format = GST_FORMAT_TIME;
goto do_times;
}
/* clip */
if (G_UNLIKELY (!gst_segment_clip (segment, GST_FORMAT_TIME,
(gint64) start, (gint64) stop, &cstart, &cstop)))
goto out_of_segment;
if (G_UNLIKELY (start != cstart || stop != cstop)) {
GST_DEBUG_OBJECT (basesink, "clipped to: start %" GST_TIME_FORMAT
", stop: %" GST_TIME_FORMAT, GST_TIME_ARGS (cstart),
GST_TIME_ARGS (cstop));
}
/* set last stop position */
gst_segment_set_last_stop (segment, GST_FORMAT_TIME, cstop);
do_times:
/* this can produce wrong values if we accumulated non-TIME segments. If this happens,
* upstream is behaving very badly */
sstart = gst_segment_to_stream_time (segment, format, cstart);
sstop = gst_segment_to_stream_time (segment, format, cstop);
rstart = gst_segment_to_running_time (segment, format, cstart);
rstop = gst_segment_to_running_time (segment, format, cstop);
done:
/* save times */
*rsstart = sstart;
*rsstop = sstop;
*rrstart = rstart;
*rrstop = rstop;
/* buffers and EOS always need syncing and preroll */
return TRUE;
/* special cases */
out_of_segment:
{
/* should not happen since we clip them in the chain function already,
* we return FALSE so that we don't try to sync on it. */
GST_ELEMENT_WARNING (basesink, STREAM, FAILED,
(NULL), ("unexpected buffer out of segment found."));
GST_LOG_OBJECT (basesink, "buffer skipped, not in segment");
return FALSE;
}
}
/* with STREAM_LOCK, PREROLL_LOCK
*
* Waits for the clock to reach @time. If @time is not valid, no
* synchronisation is done and BADTIME is returned.
* If synchronisation is disabled in the element or there is no
* clock, no synchronisation is done and BADTIME is returned.
*
* Else a blocking wait is performed on the clock. We save the ClockID
* so we can unlock the entry at any time. While we are blocking, we
* release the PREROLL_LOCK so that other threads can interrupt the entry.
*
* @time is expressed in running time.
*/
static GstClockReturn
gst_base_sink_wait_clock (GstBaseSink * basesink, GstClockTime time,
GstClockTimeDiff * jitter)
{
GstClockID id;
GstClockReturn ret;
GstClock *clock;
GstClockTimeDiff ts_offset;
if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (time)))
goto invalid_time;
GST_OBJECT_LOCK (basesink);
if (G_UNLIKELY (!basesink->sync))
goto no_sync;
if (G_UNLIKELY ((clock = GST_ELEMENT_CLOCK (basesink)) == NULL))
goto no_clock;
/* add base time and latency */
time += GST_ELEMENT_CAST (basesink)->base_time;
time += basesink->priv->latency;
/* apply offset, be carefull for underflows */
ts_offset = basesink->priv->ts_offset;
if (ts_offset < 0) {
ts_offset = -ts_offset;
if (ts_offset < time)
time -= ts_offset;
else
time = 0;
} else
time += ts_offset;
id = gst_clock_new_single_shot_id (clock, time);
GST_OBJECT_UNLOCK (basesink);
basesink->clock_id = id;
/* release the preroll lock while waiting */
GST_PAD_PREROLL_UNLOCK (basesink->sinkpad);
ret = gst_clock_id_wait (id, jitter);
GST_PAD_PREROLL_LOCK (basesink->sinkpad);
gst_clock_id_unref (id);
basesink->clock_id = NULL;
return ret;
/* no syncing needed */
invalid_time:
{
GST_DEBUG_OBJECT (basesink, "time not valid, no sync needed");
return GST_CLOCK_BADTIME;
}
no_sync:
{
GST_DEBUG_OBJECT (basesink, "sync disabled");
GST_OBJECT_UNLOCK (basesink);
return GST_CLOCK_BADTIME;
}
no_clock:
{
GST_DEBUG_OBJECT (basesink, "no clock, can't sync");
GST_OBJECT_UNLOCK (basesink);
return GST_CLOCK_BADTIME;
}
}
/**
* gst_base_sink_wait_preroll:
* @sink: the sink
*
* If the #GstBaseSinkClass::render method performs its own synchronisation against
* the clock it must unblock when going from PLAYING to the PAUSED state and call
* this method before continuing to render the remaining data.
*
* This function will block until a state change to PLAYING happens (in which
* case this function returns #GST_FLOW_OK) or the processing must be stopped due
* to a state change to READY or a FLUSH event (in which case this function
* returns #GST_FLOW_WRONG_STATE).
*
* Since: 0.10.11
*
* Returns: #GST_FLOW_OK if the preroll completed and processing can
* continue. Any other return value should be returned from the render vmethod.
*/
GstFlowReturn
gst_base_sink_wait_preroll (GstBaseSink * sink)
{
sink->have_preroll = TRUE;
GST_DEBUG_OBJECT (sink, "waiting in preroll for flush or PLAYING");
/* block until the state changes, or we get a flush, or something */
GST_PAD_PREROLL_WAIT (sink->sinkpad);
sink->have_preroll = FALSE;
if (G_UNLIKELY (sink->flushing))
goto stopping;
GST_DEBUG_OBJECT (sink, "continue after preroll");
return GST_FLOW_OK;
/* ERRORS */
stopping:
{
GST_DEBUG_OBJECT (sink, "preroll interrupted");
return GST_FLOW_WRONG_STATE;
}
}
/**
* gst_base_sink_wait_eos:
* @sink: the sink
* @time: the running_time to be reached
* @jitter: the jitter to be filled with time diff (can be NULL)
*
* This function will block until @time is reached. It is usually called by
* subclasses that use their own internal synchronisation but want to let the
* EOS be handled by the base class.
*
* This function should only be called with the PREROLL_LOCK held, like when
* receiving an EOS event in the ::event vmethod.
*
* Since 0.10.15
*
* Returns: #GstFlowReturn
*/
GstFlowReturn
gst_base_sink_wait_eos (GstBaseSink * sink, GstClockTime time,
GstClockTimeDiff * jitter)
{
GstClockReturn status;
GstFlowReturn ret;
do {
GST_DEBUG_OBJECT (sink, "checking preroll");
/* first wait for the playing state before we can continue */
if (G_UNLIKELY (sink->need_preroll)) {
ret = gst_base_sink_wait_preroll (sink);
if (ret != GST_FLOW_OK)
goto flushing;
}
/* preroll done, we can sync since we are in PLAYING now. */
GST_DEBUG_OBJECT (sink, "possibly waiting for clock to reach %"
GST_TIME_FORMAT, GST_TIME_ARGS (time));
/* wait for the clock, this can be interrupted because we got shut down or
* we PAUSED. */
status = gst_base_sink_wait_clock (sink, time, jitter);
GST_DEBUG_OBJECT (sink, "clock returned %d", status);
/* invalid time, no clock or sync disabled, just continue then */
if (status == GST_CLOCK_BADTIME)
break;
/* waiting could have been interrupted and we can be flushing now */
if (G_UNLIKELY (sink->flushing))
goto flushing;
/* retry if we got unscheduled, which means we did not reach the timeout
* yet. if some other error occures, we continue. */
} while (status == GST_CLOCK_UNSCHEDULED);
GST_DEBUG_OBJECT (sink, "end of stream");
return GST_FLOW_OK;
/* ERRORS */
flushing:
{
GST_DEBUG_OBJECT (sink, "we are flushing");
return GST_FLOW_WRONG_STATE;
}
}
/* with STREAM_LOCK, PREROLL_LOCK
*
* Make sure we are in PLAYING and synchronize an object to the clock.
*
* If we need preroll, we are not in PLAYING. We try to commit the state
* if needed and then block if we still are not PLAYING.
*
* We start waiting on the clock in PLAYING. If we got interrupted, we
* immediatly try to re-preroll.
*
* Some objects do not need synchronisation (most events) and so this function
* immediatly returns GST_FLOW_OK.
*
* for objects that arrive later than max-lateness to be synchronized to the
* clock have the @late boolean set to TRUE.
*
* This function keeps a running average of the jitter (the diff between the
* clock time and the requested sync time). The jitter is negative for
* objects that arrive in time and positive for late buffers.
*
* does not take ownership of obj.
*/
static GstFlowReturn
gst_base_sink_do_sync (GstBaseSink * basesink, GstPad * pad,
GstMiniObject * obj, gboolean * late)
{
GstClockTimeDiff jitter;
gboolean syncable;
GstClockReturn status = GST_CLOCK_OK;
GstClockTime rstart, rstop, sstart, sstop;
gboolean do_sync;
GstBaseSinkPrivate *priv;
priv = basesink->priv;
sstart = sstop = rstart = rstop = -1;
do_sync = TRUE;
priv->current_rstart = -1;
/* get timing information for this object against the render segment */
syncable = gst_base_sink_get_sync_times (basesink, obj,
&sstart, &sstop, &rstart, &rstop, &do_sync, &basesink->segment);
/* a syncable object needs to participate in preroll and
* clocking. All buffers and EOS are syncable. */
if (G_UNLIKELY (!syncable))
goto not_syncable;
/* store timing info for current object */
priv->current_rstart = rstart;
priv->current_rstop = (rstop != -1 ? rstop : rstart);
/* save sync time for eos when the previous object needed sync */
priv->eos_rtime = (do_sync ? priv->current_rstop : -1);
again:
/* first do preroll, this makes sure we commit our state
* to PAUSED and can continue to PLAYING. We cannot perform
* any clock sync in PAUSED because there is no clock.
*/
while (G_UNLIKELY (basesink->need_preroll)) {
GST_DEBUG_OBJECT (basesink, "prerolling object %p", obj);
if (G_LIKELY (basesink->playing_async)) {
/* commit state */
if (G_UNLIKELY (!gst_base_sink_commit_state (basesink)))
goto stopping;
}
/* need to recheck here because the commit state could have
* made us not need the preroll anymore */
if (G_LIKELY (basesink->need_preroll)) {
/* block until the state changes, or we get a flush, or something */
if (gst_base_sink_wait_preroll (basesink) != GST_FLOW_OK)
goto flushing;
}
}
/* After rendering we store the position of the last buffer so that we can use
* it to report the position. We need to take the lock here. */
GST_OBJECT_LOCK (basesink);
priv->current_sstart = sstart;
priv->current_sstop = (sstop != -1 ? sstop : sstart);
GST_OBJECT_UNLOCK (basesink);
if (!do_sync)
goto done;
/* preroll done, we can sync since we are in PLAYING now. */
GST_DEBUG_OBJECT (basesink, "possibly waiting for clock to reach %"
GST_TIME_FORMAT, GST_TIME_ARGS (rstart));
/* this function will return immediatly if start == -1, no clock
* or sync is disabled with GST_CLOCK_BADTIME. */
status = gst_base_sink_wait_clock (basesink, rstart, &jitter);
GST_DEBUG_OBJECT (basesink, "clock returned %d", status);
/* invalid time, no clock or sync disabled, just render */
if (status == GST_CLOCK_BADTIME)
goto done;
/* waiting could have been interrupted and we can be flushing now */
if (G_UNLIKELY (basesink->flushing))
goto flushing;
/* check for unlocked by a state change, we are not flushing so
* we can try to preroll on the current buffer. */
if (G_UNLIKELY (status == GST_CLOCK_UNSCHEDULED)) {
GST_DEBUG_OBJECT (basesink, "unscheduled, waiting some more");
goto again;
}
/* successful syncing done, record observation */
priv->current_jitter = jitter;
/* check if the object should be dropped */
*late = gst_base_sink_is_too_late (basesink, obj, rstart, rstop,
status, jitter);
done:
return GST_FLOW_OK;
/* ERRORS */
not_syncable:
{
GST_DEBUG_OBJECT (basesink, "non syncable object %p", obj);
return GST_FLOW_OK;
}
flushing:
{
GST_DEBUG_OBJECT (basesink, "we are flushing");
return GST_FLOW_WRONG_STATE;
}
stopping:
{
GST_DEBUG_OBJECT (basesink, "stopping while commiting state");
return GST_FLOW_WRONG_STATE;
}
}
static gboolean
gst_base_sink_send_qos (GstBaseSink * basesink,
gdouble proportion, GstClockTime time, GstClockTimeDiff diff)
{
GstEvent *event;
gboolean res;
/* generate Quality-of-Service event */
GST_CAT_DEBUG_OBJECT (GST_CAT_QOS, basesink,
"qos: proportion: %lf, diff %" G_GINT64_FORMAT ", timestamp %"
GST_TIME_FORMAT, proportion, diff, GST_TIME_ARGS (time));
event = gst_event_new_qos (proportion, diff, time);
/* send upstream */
res = gst_pad_push_event (basesink->sinkpad, event);
return res;
}
static void
gst_base_sink_perform_qos (GstBaseSink * sink, gboolean dropped)
{
GstBaseSinkPrivate *priv;
GstClockTime start, stop;
GstClockTimeDiff jitter;
GstClockTime pt, entered, left;
GstClockTime duration;
gdouble rate;
priv = sink->priv;
start = priv->current_rstart;
/* if Quality-of-Service disabled, do nothing */
if (!g_atomic_int_get (&priv->qos_enabled) || start == -1)
return;
stop = priv->current_rstop;
jitter = priv->current_jitter;
/* this is the time the buffer entered the sink */
entered = start + jitter;
/* this is the time the buffer left the sink */
left = start + (jitter < 0 ? 0 : jitter);
/* calculate duration of the buffer */
if (stop != -1)
duration = stop - start;
else
duration = -1;
/* if we have the time when the last buffer left us, calculate
* processing time */
if (priv->last_left != -1) {
if (entered > priv->last_left) {
pt = entered - priv->last_left;
} else {
pt = 0;
}
} else {
pt = priv->avg_pt;
}
GST_CAT_DEBUG_OBJECT (GST_CAT_QOS, sink, "start: %" GST_TIME_FORMAT
", entered %" GST_TIME_FORMAT ", left %" GST_TIME_FORMAT ", pt: %"
GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT ",jitter %"
G_GINT64_FORMAT, GST_TIME_ARGS (start), GST_TIME_ARGS (entered),
GST_TIME_ARGS (left), GST_TIME_ARGS (pt), GST_TIME_ARGS (duration),
jitter);
GST_CAT_DEBUG_OBJECT (GST_CAT_QOS, sink, "avg_duration: %" GST_TIME_FORMAT
", avg_pt: %" GST_TIME_FORMAT ", avg_rate: %g",
GST_TIME_ARGS (priv->avg_duration), GST_TIME_ARGS (priv->avg_pt),
priv->avg_rate);
/* collect running averages. for first observations, we copy the
* values */
if (priv->avg_duration == -1)
priv->avg_duration = duration;
else
priv->avg_duration = UPDATE_RUNNING_AVG (priv->avg_duration, duration);
if (priv->avg_pt == -1)
priv->avg_pt = pt;
else
priv->avg_pt = UPDATE_RUNNING_AVG (priv->avg_pt, pt);
if (priv->avg_duration != 0)
rate =
gst_guint64_to_gdouble (priv->avg_pt) /
gst_guint64_to_gdouble (priv->avg_duration);
else
rate = 0.0;
if (priv->last_left != -1) {
if (dropped || priv->avg_rate < 0.0) {
priv->avg_rate = rate;
} else {
if (rate > 1.0)
priv->avg_rate = UPDATE_RUNNING_AVG_N (priv->avg_rate, rate);
else
priv->avg_rate = UPDATE_RUNNING_AVG_P (priv->avg_rate, rate);
}
}
GST_CAT_DEBUG_OBJECT (GST_CAT_QOS, sink,
"updated: avg_duration: %" GST_TIME_FORMAT ", avg_pt: %" GST_TIME_FORMAT
", avg_rate: %g", GST_TIME_ARGS (priv->avg_duration),
GST_TIME_ARGS (priv->avg_pt), priv->avg_rate);
/* if we have a valid rate, start sending QoS messages */
if (priv->avg_rate >= 0.0) {
gst_base_sink_send_qos (sink, priv->avg_rate, priv->current_rstart,
priv->current_jitter);
}
/* record when this buffer will leave us */
priv->last_left = left;
}
/* reset all qos measuring */
static void
gst_base_sink_reset_qos (GstBaseSink * sink)
{
GstBaseSinkPrivate *priv;
priv = sink->priv;
priv->last_in_time = -1;
priv->last_left = -1;
priv->avg_duration = -1;
priv->avg_pt = -1;
priv->avg_rate = -1.0;
priv->avg_render = -1;
priv->rendered = 0;
priv->dropped = 0;
}
/* Checks if the object was scheduled too late.
*
* start/stop contain the raw timestamp start and stop values
* of the object.
*
* status and jitter contain the return values from the clock wait.
*
* returns TRUE if the buffer was too late.
*/
static gboolean
gst_base_sink_is_too_late (GstBaseSink * basesink, GstMiniObject * obj,
GstClockTime start, GstClockTime stop,
GstClockReturn status, GstClockTimeDiff jitter)
{
gboolean late;
gint64 max_lateness;
GstBaseSinkPrivate *priv;
priv = basesink->priv;
late = FALSE;
/* only for objects that were too late */
if (G_LIKELY (status != GST_CLOCK_EARLY))
goto in_time;
max_lateness = basesink->abidata.ABI.max_lateness;
/* check if frame dropping is enabled */
if (max_lateness == -1)
goto no_drop;
/* only check for buffers */
if (G_UNLIKELY (!GST_IS_BUFFER (obj)))
goto not_buffer;
/* can't do check if we don't have a timestamp */
if (G_UNLIKELY (start == -1))
goto no_timestamp;
/* we can add a valid stop time */
if (stop != -1)
max_lateness += stop;
else
max_lateness += start;
/* if the jitter bigger than duration and lateness we are too late */
if ((late = start + jitter > max_lateness)) {
GST_DEBUG_OBJECT (basesink, "buffer is too late %" GST_TIME_FORMAT
" > %" GST_TIME_FORMAT, GST_TIME_ARGS (start + jitter),
GST_TIME_ARGS (max_lateness));
/* !!emergency!!, if we did not receive anything valid for more than a
* second, render it anyway so the user sees something */
if (priv->last_in_time && start - priv->last_in_time > GST_SECOND) {
late = FALSE;
GST_DEBUG_OBJECT (basesink,
"**emergency** last buffer at %" GST_TIME_FORMAT " > GST_SECOND",
GST_TIME_ARGS (priv->last_in_time));
}
}
done:
if (!late) {
priv->last_in_time = start;
}
return late;
/* all is fine */
in_time:
{
GST_DEBUG_OBJECT (basesink, "object was scheduled in time");
goto done;
}
no_drop:
{
GST_DEBUG_OBJECT (basesink, "frame dropping disabled");
goto done;
}
not_buffer:
{
GST_DEBUG_OBJECT (basesink, "object is not a buffer");
return FALSE;
}
no_timestamp:
{
GST_DEBUG_OBJECT (basesink, "buffer has no timestamp");
return FALSE;
}
}
static void
gst_base_sink_do_render_stats (GstBaseSink * basesink, gboolean start)
{
GstBaseSinkPrivate *priv;
priv = basesink->priv;
if (start) {
priv->start = gst_util_get_timestamp ();
} else {
GstClockTime elapsed;
priv->stop = gst_util_get_timestamp ();
elapsed = GST_CLOCK_DIFF (priv->start, priv->stop);
if (priv->avg_render == -1)
priv->avg_render = elapsed;
else
priv->avg_render = UPDATE_RUNNING_AVG (priv->avg_render, elapsed);
GST_CAT_DEBUG_OBJECT (GST_CAT_QOS, basesink,
"avg_render: %" GST_TIME_FORMAT, GST_TIME_ARGS (priv->avg_render));
}
}
/* with STREAM_LOCK, PREROLL_LOCK,
*
* Synchronize the object on the clock and then render it.
*
* takes ownership of obj.
*/
static GstFlowReturn
gst_base_sink_render_object (GstBaseSink * basesink, GstPad * pad,
GstMiniObject * obj)
{
GstFlowReturn ret = GST_FLOW_OK;
GstBaseSinkClass *bclass;
gboolean late = FALSE;
GstBaseSinkPrivate *priv;
priv = basesink->priv;
/* synchronize this object, non syncable objects return OK
* immediatly. */
ret = gst_base_sink_do_sync (basesink, pad, obj, &late);
if (G_UNLIKELY (ret != GST_FLOW_OK))
goto sync_failed;
/* and now render, event or buffer. */
if (G_LIKELY (GST_IS_BUFFER (obj))) {
GstBuffer *buf;
/* drop late buffers unconditionally, let's hope it's unlikely */
if (G_UNLIKELY (late))
goto dropped;
buf = GST_BUFFER_CAST (obj);
gst_base_sink_set_last_buffer (basesink, buf);
bclass = GST_BASE_SINK_GET_CLASS (basesink);
if (G_LIKELY (bclass->render)) {
gint do_qos;
/* read once, to get same value before and after */
do_qos = g_atomic_int_get (&priv->qos_enabled);
GST_DEBUG_OBJECT (basesink, "rendering buffer %p", obj);
/* record rendering time for QoS and stats */
if (do_qos)
gst_base_sink_do_render_stats (basesink, TRUE);
ret = bclass->render (basesink, buf);
priv->rendered++;
if (do_qos)
gst_base_sink_do_render_stats (basesink, FALSE);
}
} else {
GstEvent *event = GST_EVENT_CAST (obj);
gboolean event_res = TRUE;
GstEventType type;
bclass = GST_BASE_SINK_GET_CLASS (basesink);
type = GST_EVENT_TYPE (event);
GST_DEBUG_OBJECT (basesink, "rendering event %p, type %s", obj,
gst_event_type_get_name (type));
if (bclass->event)
event_res = bclass->event (basesink, event);
if (G_LIKELY (event_res)) {
switch (type) {
case GST_EVENT_EOS:
/* the EOS event is completely handled so we mark
* ourselves as being in the EOS state. eos is also
* protected by the object lock so we can read it when
* answering the POSITION query. */
GST_OBJECT_LOCK (basesink);
basesink->eos = TRUE;
GST_OBJECT_UNLOCK (basesink);
/* ok, now we can post the message */
GST_DEBUG_OBJECT (basesink, "Now posting EOS");
gst_element_post_message (GST_ELEMENT_CAST (basesink),
gst_message_new_eos (GST_OBJECT_CAST (basesink)));
break;
case GST_EVENT_NEWSEGMENT:
/* configure the segment */
gst_base_sink_configure_segment (basesink, pad, event,
&basesink->segment);
break;
default:
break;
}
}
}
done:
gst_base_sink_perform_qos (basesink, late);
GST_DEBUG_OBJECT (basesink, "object unref after render %p", obj);
gst_mini_object_unref (obj);
return ret;
/* ERRORS */
sync_failed:
{
GST_DEBUG_OBJECT (basesink, "do_sync returned %s", gst_flow_get_name (ret));
goto done;
}
dropped:
{
priv->dropped++;
GST_DEBUG_OBJECT (basesink, "buffer late, dropping");
goto done;
}
}
/* with STREAM_LOCK, PREROLL_LOCK
*
* Perform preroll on the given object. For buffers this means
* calling the preroll subclass method.
* If that succeeds, the state will be commited.
*
* function does not take ownership of obj.
*/
static GstFlowReturn
gst_base_sink_preroll_object (GstBaseSink * basesink, GstPad * pad,
GstMiniObject * obj)
{
GstFlowReturn ret;
GST_DEBUG_OBJECT (basesink, "do preroll %p", obj);
/* if it's a buffer, we need to call the preroll method */
if (G_LIKELY (GST_IS_BUFFER (obj))) {
GstBaseSinkClass *bclass;
GstBuffer *buf;
GstClockTime timestamp;
buf = GST_BUFFER_CAST (obj);
timestamp = GST_BUFFER_TIMESTAMP (buf);
GST_DEBUG_OBJECT (basesink, "preroll buffer %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp));
gst_base_sink_set_last_buffer (basesink, buf);
bclass = GST_BASE_SINK_GET_CLASS (basesink);
if (bclass->preroll)
if ((ret = bclass->preroll (basesink, buf)) != GST_FLOW_OK)
goto preroll_failed;
}
/* commit state */
if (G_LIKELY (basesink->playing_async)) {
if (G_UNLIKELY (!gst_base_sink_commit_state (basesink)))
goto stopping;
}
return GST_FLOW_OK;
/* ERRORS */
preroll_failed:
{
GST_DEBUG_OBJECT (basesink, "preroll failed, abort state");
gst_element_abort_state (GST_ELEMENT_CAST (basesink));
return ret;
}
stopping:
{
GST_DEBUG_OBJECT (basesink, "stopping while commiting state");
return GST_FLOW_WRONG_STATE;
}
}
/* with STREAM_LOCK, PREROLL_LOCK
*
* Queue an object for rendering.
* The first prerollable object queued will complete the preroll. If the
* preroll queue if filled, we render all the objects in the queue.
*
* This function takes ownership of the object.
*/
static GstFlowReturn
gst_base_sink_queue_object_unlocked (GstBaseSink * basesink, GstPad * pad,
GstMiniObject * obj, gboolean prerollable)
{
GstFlowReturn ret = GST_FLOW_OK;
gint length;
GQueue *q;
if (G_UNLIKELY (basesink->need_preroll)) {
if (G_LIKELY (prerollable))
basesink->preroll_queued++;
length = basesink->preroll_queued;
GST_DEBUG_OBJECT (basesink, "now %d prerolled items", length);
/* first prerollable item needs to finish the preroll */
if (length == 1) {
ret = gst_base_sink_preroll_object (basesink, pad, obj);
if (G_UNLIKELY (ret != GST_FLOW_OK))
goto preroll_failed;
}
/* need to recheck if we need preroll, commmit state during preroll
* could have made us not need more preroll. */
if (G_UNLIKELY (basesink->need_preroll)) {
/* see if we can render now, if we can't add the object to the preroll
* queue. */
if (G_UNLIKELY (length <= basesink->preroll_queue_max_len))
goto more_preroll;
}
}
/* we can start rendering (or blocking) the queued object
* if any. */
q = basesink->preroll_queue;
while (G_UNLIKELY (!g_queue_is_empty (q))) {
GstMiniObject *o;
o = g_queue_pop_head (q);
GST_DEBUG_OBJECT (basesink, "rendering queued object %p", o);
/* do something with the return value */
ret = gst_base_sink_render_object (basesink, pad, o);
if (ret != GST_FLOW_OK)
goto dequeue_failed;
}
/* now render the object */
ret = gst_base_sink_render_object (basesink, pad, obj);
basesink->preroll_queued = 0;
return ret;
/* special cases */
preroll_failed:
{
GST_DEBUG_OBJECT (basesink, "preroll failed, reason %s",
gst_flow_get_name (ret));
gst_mini_object_unref (obj);
return ret;
}
more_preroll:
{
/* add object to the queue and return */
GST_DEBUG_OBJECT (basesink, "need more preroll data %d <= %d",
length, basesink->preroll_queue_max_len);
g_queue_push_tail (basesink->preroll_queue, obj);
return GST_FLOW_OK;
}
dequeue_failed:
{
GST_DEBUG_OBJECT (basesink, "rendering queued objects failed, reason %s",
gst_flow_get_name (ret));
gst_mini_object_unref (obj);
return ret;
}
}
/* with STREAM_LOCK
*
* This function grabs the PREROLL_LOCK and adds the object to
* the queue.
*
* This function takes ownership of obj.
*/
static GstFlowReturn
gst_base_sink_queue_object (GstBaseSink * basesink, GstPad * pad,
GstMiniObject * obj, gboolean prerollable)
{
GstFlowReturn ret;
GST_PAD_PREROLL_LOCK (pad);
if (G_UNLIKELY (basesink->flushing))
goto flushing;
if (G_UNLIKELY (basesink->priv->received_eos))
goto was_eos;
ret = gst_base_sink_queue_object_unlocked (basesink, pad, obj, prerollable);
GST_PAD_PREROLL_UNLOCK (pad);
return ret;
/* ERRORS */
flushing:
{
GST_DEBUG_OBJECT (basesink, "sink is flushing");
GST_PAD_PREROLL_UNLOCK (pad);
gst_mini_object_unref (obj);
return GST_FLOW_WRONG_STATE;
}
was_eos:
{
GST_DEBUG_OBJECT (basesink,
"we are EOS, dropping object, return UNEXPECTED");
GST_PAD_PREROLL_UNLOCK (pad);
gst_mini_object_unref (obj);
return GST_FLOW_UNEXPECTED;
}
}
static gboolean
gst_base_sink_event (GstPad * pad, GstEvent * event)
{
GstBaseSink *basesink;
gboolean result = TRUE;
GstBaseSinkClass *bclass;
basesink = GST_BASE_SINK (gst_pad_get_parent (pad));
bclass = GST_BASE_SINK_GET_CLASS (basesink);
GST_DEBUG_OBJECT (basesink, "event %p (%s)", event,
GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:
{
GstFlowReturn ret;
GST_PAD_PREROLL_LOCK (pad);
if (G_UNLIKELY (basesink->flushing))
goto flushing;
if (G_UNLIKELY (basesink->priv->received_eos)) {
/* we can't accept anything when we are EOS */
result = FALSE;
gst_event_unref (event);
} else {
/* we set the received EOS flag here so that we can use it when testing if
* we are prerolled and to refure more buffers. */
basesink->priv->received_eos = TRUE;
/* EOS is a prerollable object, we call the unlocked version because it
* does not check the received_eos flag. */
ret = gst_base_sink_queue_object_unlocked (basesink, pad,
GST_MINI_OBJECT_CAST (event), TRUE);
if (G_UNLIKELY (ret != GST_FLOW_OK))
result = FALSE;
}
GST_PAD_PREROLL_UNLOCK (pad);
break;
}
case GST_EVENT_NEWSEGMENT:
{
GstFlowReturn ret;
GST_DEBUG_OBJECT (basesink, "newsegment %p", event);
GST_PAD_PREROLL_LOCK (pad);
if (G_UNLIKELY (basesink->flushing))
goto flushing;
if (G_UNLIKELY (basesink->priv->received_eos)) {
/* we can't accept anything when we are EOS */
result = FALSE;
gst_event_unref (event);
} else {
/* the new segment is a non prerollable item and does not block anything,
* we need to configure the current clipping segment and insert the event
* in the queue to serialize it with the buffers for rendering. */
gst_base_sink_configure_segment (basesink, pad, event,
basesink->abidata.ABI.clip_segment);
ret =
gst_base_sink_queue_object_unlocked (basesink, pad,
GST_MINI_OBJECT_CAST (event), FALSE);
if (G_UNLIKELY (ret != GST_FLOW_OK))
result = FALSE;
else
basesink->have_newsegment = TRUE;
}
GST_PAD_PREROLL_UNLOCK (pad);
break;
}
case GST_EVENT_FLUSH_START:
if (bclass->event)
bclass->event (basesink, event);
GST_DEBUG_OBJECT (basesink, "flush-start %p", event);
/* make sure we are not blocked on the clock also clear any pending
* eos state. */
gst_base_sink_set_flushing (basesink, pad, TRUE);
/* we grab the stream lock but that is not needed since setting the
* sink to flushing would make sure no state commit is being done
* anymore */
GST_PAD_STREAM_LOCK (pad);
gst_base_sink_reset_qos (basesink);
if (basesink->priv->async_enabled) {
/* and we need to commit our state again on the next
* prerolled buffer */
basesink->playing_async = TRUE;
gst_element_lost_state (GST_ELEMENT_CAST (basesink));
} else {
basesink->priv->have_latency = TRUE;
basesink->need_preroll = FALSE;
}
gst_base_sink_set_last_buffer (basesink, NULL);
GST_PAD_STREAM_UNLOCK (pad);
gst_event_unref (event);
break;
case GST_EVENT_FLUSH_STOP:
if (bclass->event)
bclass->event (basesink, event);
GST_DEBUG_OBJECT (basesink, "flush-stop %p", event);
/* unset flushing so we can accept new data, this also flushes out any EOS
* event. */
gst_base_sink_set_flushing (basesink, pad, FALSE);
/* we need new segment info after the flush. */
gst_segment_init (&basesink->segment, GST_FORMAT_UNDEFINED);
gst_segment_init (basesink->abidata.ABI.clip_segment,
GST_FORMAT_UNDEFINED);
basesink->have_newsegment = FALSE;
/* for position reporting */
GST_OBJECT_LOCK (basesink);
basesink->priv->current_sstart = -1;
basesink->priv->current_sstop = -1;
GST_OBJECT_UNLOCK (basesink);
gst_event_unref (event);
break;
default:
/* other events are sent to queue or subclass depending on if they
* are serialized. */
if (GST_EVENT_IS_SERIALIZED (event)) {
gst_base_sink_queue_object (basesink, pad,
GST_MINI_OBJECT_CAST (event), FALSE);
} else {
if (bclass->event)
bclass->event (basesink, event);
gst_event_unref (event);
}
break;
}
done:
gst_object_unref (basesink);
return result;
/* ERRORS */
flushing:
{
GST_DEBUG_OBJECT (basesink, "we are flushing");
GST_PAD_PREROLL_UNLOCK (pad);
result = FALSE;
gst_event_unref (event);
goto done;
}
}
/* default implementation to calculate the start and end
* timestamps on a buffer, subclasses can override
*/
static void
gst_base_sink_get_times (GstBaseSink * basesink, GstBuffer * buffer,
GstClockTime * start, GstClockTime * end)
{
GstClockTime timestamp, duration;
timestamp = GST_BUFFER_TIMESTAMP (buffer);
if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
/* get duration to calculate end time */
duration = GST_BUFFER_DURATION (buffer);
if (GST_CLOCK_TIME_IS_VALID (duration)) {
*end = timestamp + duration;
}
*start = timestamp;
}
}
/* must be called with PREROLL_LOCK */
static gboolean
gst_base_sink_needs_preroll (GstBaseSink * basesink)
{
gboolean is_prerolled, res;
/* we have 2 cases where the PREROLL_LOCK is released:
* 1) we are blocking in the PREROLL_LOCK and thus are prerolled.
* 2) we are syncing on the clock
*/
is_prerolled = basesink->have_preroll || basesink->priv->received_eos;
res = !is_prerolled && basesink->pad_mode != GST_ACTIVATE_PULL;
GST_DEBUG_OBJECT (basesink, "have_preroll: %d, EOS: %d => needs preroll: %d",
basesink->have_preroll, basesink->priv->received_eos, res);
return res;
}
/* with STREAM_LOCK, PREROLL_LOCK
*
* Takes a buffer and compare the timestamps with the last segment.
* If the buffer falls outside of the segment boundaries, drop it.
* Else queue the buffer for preroll and rendering.
*
* This function takes ownership of the buffer.
*/
static GstFlowReturn
gst_base_sink_chain_unlocked (GstBaseSink * basesink, GstPad * pad,
GstBuffer * buf)
{
GstFlowReturn result;
GstClockTime start = GST_CLOCK_TIME_NONE, end = GST_CLOCK_TIME_NONE;
GstSegment *clip_segment;
if (G_UNLIKELY (basesink->flushing))
goto flushing;
if (G_UNLIKELY (basesink->priv->received_eos))
goto was_eos;
/* for code clarity */
clip_segment = basesink->abidata.ABI.clip_segment;
if (G_UNLIKELY (!basesink->have_newsegment)) {
gboolean sync;
sync = gst_base_sink_get_sync (basesink);
if (sync) {
GST_ELEMENT_WARNING (basesink, STREAM, FAILED,
(_("Internal data flow problem.")),
("Received buffer without a new-segment. Assuming timestamps start from 0."));
}
basesink->have_newsegment = TRUE;
/* this means this sink will assume timestamps start from 0 */
clip_segment->start = 0;
clip_segment->stop = -1;
basesink->segment.start = 0;
basesink->segment.stop = -1;
}
/* check if the buffer needs to be dropped */
/* we don't use the subclassed method as it may not return
* valid values for our purpose here */
gst_base_sink_get_times (basesink, buf, &start, &end);
GST_DEBUG_OBJECT (basesink, "got times start: %" GST_TIME_FORMAT
", end: %" GST_TIME_FORMAT, GST_TIME_ARGS (start), GST_TIME_ARGS (end));
/* a dropped buffer does not participate in anything */
if (GST_CLOCK_TIME_IS_VALID (start) &&
(clip_segment->format == GST_FORMAT_TIME)) {
if (G_UNLIKELY (!gst_segment_clip (clip_segment,
GST_FORMAT_TIME, (gint64) start, (gint64) end, NULL, NULL)))
goto out_of_segment;
}
/* now we can process the buffer in the queue, this function takes ownership
* of the buffer */
result = gst_base_sink_queue_object_unlocked (basesink, pad,
GST_MINI_OBJECT_CAST (buf), TRUE);
return result;
/* ERRORS */
flushing:
{
GST_DEBUG_OBJECT (basesink, "sink is flushing");
gst_buffer_unref (buf);
return GST_FLOW_WRONG_STATE;
}
was_eos:
{
GST_DEBUG_OBJECT (basesink,
"we are EOS, dropping object, return UNEXPECTED");
gst_buffer_unref (buf);
return GST_FLOW_UNEXPECTED;
}
out_of_segment:
{
GST_DEBUG_OBJECT (basesink, "dropping buffer, out of clipping segment");
gst_buffer_unref (buf);
return GST_FLOW_OK;
}
}
/* with STREAM_LOCK
*/
static GstFlowReturn
gst_base_sink_chain (GstPad * pad, GstBuffer * buf)
{
GstBaseSink *basesink;
GstFlowReturn result;
basesink = GST_BASE_SINK (GST_OBJECT_PARENT (pad));
if (G_UNLIKELY (basesink->pad_mode != GST_ACTIVATE_PUSH))
goto wrong_mode;
GST_PAD_PREROLL_LOCK (pad);
result = gst_base_sink_chain_unlocked (basesink, pad, buf);
GST_PAD_PREROLL_UNLOCK (pad);
done:
return result;
/* ERRORS */
wrong_mode:
{
GST_OBJECT_LOCK (pad);
GST_WARNING_OBJECT (basesink,
"Push on pad %s:%s, but it was not activated in push mode",
GST_DEBUG_PAD_NAME (pad));
GST_OBJECT_UNLOCK (pad);
gst_buffer_unref (buf);
/* we don't post an error message this will signal to the peer
* pushing that EOS is reached. */
result = GST_FLOW_UNEXPECTED;
goto done;
}
}
/* with STREAM_LOCK
*/
static void
gst_base_sink_loop (GstPad * pad)
{
GstBaseSink *basesink;
GstBuffer *buf = NULL;
GstFlowReturn result;
basesink = GST_BASE_SINK (GST_OBJECT_PARENT (pad));
g_assert (basesink->pad_mode == GST_ACTIVATE_PULL);
GST_DEBUG_OBJECT (basesink, "pulling %" G_GUINT64_FORMAT ", %u",
basesink->offset, (guint) DEFAULT_SIZE);
result = gst_pad_pull_range (pad, basesink->offset, DEFAULT_SIZE, &buf);
if (G_UNLIKELY (result != GST_FLOW_OK))
goto paused;
if (G_UNLIKELY (buf == NULL))
goto no_buffer;
basesink->offset += GST_BUFFER_SIZE (buf);
GST_PAD_PREROLL_LOCK (pad);
result = gst_base_sink_chain_unlocked (basesink, pad, buf);
GST_PAD_PREROLL_UNLOCK (pad);
if (G_UNLIKELY (result != GST_FLOW_OK))
goto paused;
return;
/* ERRORS */
paused:
{
GST_LOG_OBJECT (basesink, "pausing task, reason %s",
gst_flow_get_name (result));
gst_pad_pause_task (pad);
/* fatal errors and NOT_LINKED cause EOS */
if (GST_FLOW_IS_FATAL (result) || result == GST_FLOW_NOT_LINKED) {
/* FIXME, we shouldn't post EOS when we are operating in segment mode */
gst_base_sink_event (pad, gst_event_new_eos ());
/* EOS does not cause an ERROR message */
if (result != GST_FLOW_UNEXPECTED) {
GST_ELEMENT_ERROR (basesink, STREAM, FAILED,
(_("Internal data stream error.")),
("stream stopped, reason %s", gst_flow_get_name (result)));
}
}
return;
}
no_buffer:
{
GST_LOG_OBJECT (basesink, "no buffer, pausing");
result = GST_FLOW_ERROR;
goto paused;
}
}
static gboolean
gst_base_sink_set_flushing (GstBaseSink * basesink, GstPad * pad,
gboolean flushing)
{
GstBaseSinkClass *bclass;
bclass = GST_BASE_SINK_GET_CLASS (basesink);
if (flushing) {
/* unlock any subclasses, we need to do this before grabbing the
* PREROLL_LOCK since we hold this lock before going into ::render. */
if (bclass->unlock)
bclass->unlock (basesink);
}
GST_PAD_PREROLL_LOCK (pad);
basesink->flushing = flushing;
if (flushing) {
/* step 1, now that we have the PREROLL lock, clear our unlock request */
if (bclass->unlock_stop)
bclass->unlock_stop (basesink);
/* set need_preroll before we unblock the clock. If the clock is unblocked
* before timing out, we can reuse the buffer for preroll. */
basesink->need_preroll = TRUE;
/* step 2, unblock clock sync (if any) or any other blocking thing */
if (basesink->clock_id) {
gst_clock_id_unschedule (basesink->clock_id);
}
/* flush out the data thread if it's locked in finish_preroll */
GST_DEBUG_OBJECT (basesink,
"flushing out data thread, need preroll to TRUE");
gst_base_sink_preroll_queue_flush (basesink, pad);
}
GST_PAD_PREROLL_UNLOCK (pad);
return TRUE;
}
static gboolean
gst_base_sink_default_activate_pull (GstBaseSink * basesink, gboolean active)
{
gboolean result;
if (active) {
/* start task */
result = gst_pad_start_task (basesink->sinkpad,
(GstTaskFunction) gst_base_sink_loop, basesink->sinkpad);
} else {
/* step 2, make sure streaming finishes */
result = gst_pad_stop_task (basesink->sinkpad);
}
return result;
}
static gboolean
gst_base_sink_pad_activate (GstPad * pad)
{
gboolean result = FALSE;
GstBaseSink *basesink;
basesink = GST_BASE_SINK (gst_pad_get_parent (pad));
GST_DEBUG_OBJECT (basesink, "Trying pull mode first");
gst_base_sink_set_flushing (basesink, pad, FALSE);
if (basesink->can_activate_pull && gst_pad_check_pull_range (pad)
&& gst_pad_activate_pull (pad, TRUE)) {
GST_DEBUG_OBJECT (basesink, "Success activating pull mode");
result = TRUE;
} else {
GST_DEBUG_OBJECT (basesink, "Falling back to push mode");
if (gst_pad_activate_push (pad, TRUE)) {
GST_DEBUG_OBJECT (basesink, "Success activating push mode");
result = TRUE;
}
}
if (!result) {
GST_WARNING_OBJECT (basesink, "Could not activate pad in either mode");
gst_base_sink_set_flushing (basesink, pad, TRUE);
}
gst_object_unref (basesink);
return result;
}
static gboolean
gst_base_sink_pad_activate_push (GstPad * pad, gboolean active)
{
gboolean result;
GstBaseSink *basesink;
basesink = GST_BASE_SINK (gst_pad_get_parent (pad));
if (active) {
if (!basesink->can_activate_push) {
result = FALSE;
basesink->pad_mode = GST_ACTIVATE_NONE;
} else {
result = TRUE;
basesink->pad_mode = GST_ACTIVATE_PUSH;
}
} else {
if (G_UNLIKELY (basesink->pad_mode != GST_ACTIVATE_PUSH)) {
g_warning ("Internal GStreamer activation error!!!");
result = FALSE;
} else {
gst_base_sink_set_flushing (basesink, pad, TRUE);
result = TRUE;
basesink->pad_mode = GST_ACTIVATE_NONE;
}
}
gst_object_unref (basesink);
return result;
}
static gboolean
gst_base_sink_negotiate_pull (GstBaseSink * basesink)
{
GstCaps *caps;
GstPad *pad;
GST_OBJECT_LOCK (basesink);
pad = basesink->sinkpad;
gst_object_ref (pad);
GST_OBJECT_UNLOCK (basesink);
caps = gst_pad_get_allowed_caps (pad);
if (gst_caps_is_empty (caps))
goto no_caps_possible;
caps = gst_caps_make_writable (caps);
gst_caps_truncate (caps);
gst_pad_fixate_caps (pad, caps);
if (gst_caps_is_any (caps)) {
GST_DEBUG_OBJECT (basesink, "caps were ANY after fixating, "
"allowing pull()");
/* neither side has template caps in this case, so they are prepared for
pull() without setcaps() */
} else {
if (!gst_pad_set_caps (pad, caps))
goto could_not_set_caps;
}
gst_caps_unref (caps);
gst_object_unref (pad);
return TRUE;
no_caps_possible:
{
GST_INFO_OBJECT (basesink, "Pipeline could not agree on caps");
GST_DEBUG_OBJECT (basesink, "get_allowed_caps() returned EMPTY");
gst_object_unref (pad);
return FALSE;
}
could_not_set_caps:
{
GST_INFO_OBJECT (basesink, "Could not set caps: %" GST_PTR_FORMAT, caps);
gst_caps_unref (caps);
gst_object_unref (pad);
return FALSE;
}
}
/* this won't get called until we implement an activate function */
static gboolean
gst_base_sink_pad_activate_pull (GstPad * pad, gboolean active)
{
gboolean result = FALSE;
GstBaseSink *basesink;
GstBaseSinkClass *bclass;
basesink = GST_BASE_SINK (gst_pad_get_parent (pad));
bclass = GST_BASE_SINK_GET_CLASS (basesink);
if (active) {
if (!basesink->can_activate_pull) {
result = FALSE;
basesink->pad_mode = GST_ACTIVATE_NONE;
} else {
GstPad *peer = gst_pad_get_peer (pad);
if (G_UNLIKELY (peer == NULL)) {
g_warning ("Trying to activate pad in pull mode, but no peer");
result = FALSE;
basesink->pad_mode = GST_ACTIVATE_NONE;
} else {
if (gst_pad_activate_pull (peer, TRUE)) {
/* we mark we have a newsegment here because pull based
* mode works just fine without having a newsegment before the
* first buffer */
gst_segment_init (&basesink->segment, GST_FORMAT_UNDEFINED);
gst_segment_init (basesink->abidata.ABI.clip_segment,
GST_FORMAT_UNDEFINED);
basesink->have_newsegment = TRUE;
/* set the pad mode before starting the task so that it's in the
correct state for the new thread. also the sink set_caps function
checks this */
basesink->pad_mode = GST_ACTIVATE_PULL;
if ((result = gst_base_sink_negotiate_pull (basesink))) {
if (bclass->activate_pull)
result = bclass->activate_pull (basesink, TRUE);
else
result = FALSE;
}
/* but if starting the thread fails, set it back */
if (!result)
basesink->pad_mode = GST_ACTIVATE_NONE;
} else {
GST_DEBUG_OBJECT (pad, "Failed to activate peer in pull mode");
result = FALSE;
basesink->pad_mode = GST_ACTIVATE_NONE;
}
gst_object_unref (peer);
}
}
} else {
if (G_UNLIKELY (basesink->pad_mode != GST_ACTIVATE_PULL)) {
g_warning ("Internal GStreamer activation error!!!");
result = FALSE;
} else {
result = gst_base_sink_set_flushing (basesink, pad, TRUE);
if (bclass->activate_pull)
result &= bclass->activate_pull (basesink, FALSE);
basesink->pad_mode = GST_ACTIVATE_NONE;
}
}
gst_object_unref (basesink);
return result;
}
/* send an event to our sinkpad peer. */
static gboolean
gst_base_sink_send_event (GstElement * element, GstEvent * event)
{
GstPad *pad;
GstBaseSink *basesink = GST_BASE_SINK (element);
gboolean forward, result = TRUE;
/* only push UPSTREAM events upstream */
forward = GST_EVENT_IS_UPSTREAM (event);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_LATENCY:
{
GstClockTime latency;
gst_event_parse_latency (event, &latency);
/* store the latency. We use this to adjust the running_time before syncing
* it to the clock. */
GST_OBJECT_LOCK (element);
basesink->priv->latency = latency;
GST_OBJECT_UNLOCK (element);
GST_DEBUG_OBJECT (basesink, "latency set to %" GST_TIME_FORMAT,
GST_TIME_ARGS (latency));
/* don't forward, yet. FIXME. The latency event should likely be forwarded
* to upstream element so that they can configure themselves. Each element
* would subtract the amount of LATENCY it can maximally compensate for.
* It's currently not very useful; even if this sink cannot compensate for
* all the latency, upstream will block while this sink waits which will
* trigger implicit buffering and latency there. */
forward = FALSE;
break;
}
default:
break;
}
if (forward) {
GST_OBJECT_LOCK (element);
pad = gst_object_ref (basesink->sinkpad);
GST_OBJECT_UNLOCK (element);
result = gst_pad_push_event (pad, event);
gst_object_unref (pad);
} else {
/* not forwarded, unref the event */
gst_event_unref (event);
}
return result;
}
static gboolean
gst_base_sink_peer_query (GstBaseSink * sink, GstQuery * query)
{
GstPad *peer;
gboolean res = FALSE;
if ((peer = gst_pad_get_peer (sink->sinkpad))) {
res = gst_pad_query (peer, query);
gst_object_unref (peer);
}
return res;
}
/* get the end position of the last seen object, this is used
* for EOS and for making sure that we don't report a position we
* have not reached yet. */
static gboolean
gst_base_sink_get_position_last (GstBaseSink * basesink, gint64 * cur)
{
/* return last observed stream time */
*cur = basesink->priv->current_sstop;
GST_DEBUG_OBJECT (basesink, "POSITION: %" GST_TIME_FORMAT,
GST_TIME_ARGS (*cur));
return TRUE;
}
/* get the position when we are PAUSED, this is the stream time of the buffer
* that prerolled. If no buffer is prerolled (we are still flushing), this
* value will be -1. */
static gboolean
gst_base_sink_get_position_paused (GstBaseSink * basesink, gint64 * cur)
{
gboolean res;
gint64 time;
GstSegment *segment;
*cur = basesink->priv->current_sstart;
segment = basesink->abidata.ABI.clip_segment;
time = segment->time;
if (*cur != -1) {
*cur = MAX (*cur, time);
GST_DEBUG_OBJECT (basesink, "POSITION as max: %" GST_TIME_FORMAT
", time %" GST_TIME_FORMAT, GST_TIME_ARGS (*cur), GST_TIME_ARGS (time));
} else {
/* we have no buffer, use the segment times. */
if (segment->rate >= 0.0) {
/* forward, next position is always the time of the segment */
*cur = time;
GST_DEBUG_OBJECT (basesink, "POSITION as time: %" GST_TIME_FORMAT,
GST_TIME_ARGS (*cur));
} else {
/* reverse, next expected timestamp is segment->stop. We use the function
* to get things right for negative applied_rates. */
*cur =
gst_segment_to_stream_time (segment, GST_FORMAT_TIME, segment->stop);
GST_DEBUG_OBJECT (basesink, "reverse POSITION: %" GST_TIME_FORMAT,
GST_TIME_ARGS (*cur));
}
}
res = (*cur != -1);
return res;
}
static gboolean
gst_base_sink_get_position (GstBaseSink * basesink, GstFormat format,
gint64 * cur)
{
GstClock *clock;
gboolean res = FALSE;
switch (format) {
/* we can answer time format */
case GST_FORMAT_TIME:
{
GstClockTime now, base, latency;
gint64 time, accum, duration;
gdouble rate;
gint64 last;
GST_OBJECT_LOCK (basesink);
/* can only give answer based on the clock if not EOS */
if (G_UNLIKELY (basesink->eos))
goto in_eos;
/* in PAUSE we cannot read from the clock so we
* report time based on the last seen timestamp. */
if (GST_STATE (basesink) == GST_STATE_PAUSED)
goto in_pause;
/* We get position from clock only in PLAYING, we checked
* the PAUSED case above, so this is check is to test
* READY and NULL, where the position is always 0 */
if (GST_STATE (basesink) != GST_STATE_PLAYING)
goto wrong_state;
/* we need to sync on the clock. */
if (basesink->sync == FALSE)
goto no_sync;
/* and we need a clock */
if (G_UNLIKELY ((clock = GST_ELEMENT_CLOCK (basesink)) == NULL))
goto no_sync;
/* collect all data we need holding the lock */
if (GST_CLOCK_TIME_IS_VALID (basesink->segment.time))
time = basesink->segment.time;
else
time = 0;
if (GST_CLOCK_TIME_IS_VALID (basesink->segment.stop))
duration = basesink->segment.stop - basesink->segment.start;
else
duration = 0;
base = GST_ELEMENT_CAST (basesink)->base_time;
accum = basesink->segment.accum;
rate = basesink->segment.rate * basesink->segment.applied_rate;
gst_base_sink_get_position_last (basesink, &last);
latency = basesink->priv->latency;
gst_object_ref (clock);
/* need to release the object lock before we can get the time,
* a clock might take the LOCK of the provider, which could be
* a basesink subclass. */
GST_OBJECT_UNLOCK (basesink);
now = gst_clock_get_time (clock);
/* subtract base time and accumulated time from the clock time.
* Make sure we don't go negative. This is the current time in
* the segment which we need to scale with the combined
* rate and applied rate. */
base += accum;
base += latency;
base = MIN (now, base);
/* for negative rates we need to count back from from the segment
* duration. */
if (rate < 0.0)
time += duration;
*cur = time + gst_guint64_to_gdouble (now - base) * rate;
/* never report more than last seen position */
if (last != -1)
*cur = MIN (last, *cur);
gst_object_unref (clock);
res = TRUE;
GST_DEBUG_OBJECT (basesink,
"now %" GST_TIME_FORMAT " - base %" GST_TIME_FORMAT " - accum %"
GST_TIME_FORMAT " + time %" GST_TIME_FORMAT,
GST_TIME_ARGS (now), GST_TIME_ARGS (base),
GST_TIME_ARGS (accum), GST_TIME_ARGS (time));
break;
}
default:
/* cannot answer other than TIME, we return FALSE, which will
* send the query upstream. */
break;
}
done:
GST_DEBUG_OBJECT (basesink, "res: %d, POSITION: %" GST_TIME_FORMAT,
res, GST_TIME_ARGS (*cur));
return res;
/* special cases */
in_eos:
{
GST_DEBUG_OBJECT (basesink, "position in EOS");
res = gst_base_sink_get_position_last (basesink, cur);
GST_OBJECT_UNLOCK (basesink);
goto done;
}
in_pause:
{
GST_DEBUG_OBJECT (basesink, "position in PAUSED");
res = gst_base_sink_get_position_paused (basesink, cur);
GST_OBJECT_UNLOCK (basesink);
goto done;
}
wrong_state:
{
/* in NULL or READY we always return 0 */
GST_DEBUG_OBJECT (basesink, "position in wrong state, return -1");
res = FALSE;
*cur = -1;
GST_OBJECT_UNLOCK (basesink);
goto done;
}
no_sync:
{
/* report last seen timestamp if any, else return FALSE so
* that upstream can answer */
if ((*cur = basesink->priv->current_sstart) != -1)
res = TRUE;
GST_DEBUG_OBJECT (basesink, "no sync, res %d, POSITION %" GST_TIME_FORMAT,
res, GST_TIME_ARGS (*cur));
GST_OBJECT_UNLOCK (basesink);
return res;
}
}
static gboolean
gst_base_sink_query (GstElement * element, GstQuery * query)
{
gboolean res = FALSE;
GstBaseSink *basesink = GST_BASE_SINK (element);
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_POSITION:
{
gint64 cur = 0;
GstFormat format;
gst_query_parse_position (query, &format, NULL);
GST_DEBUG_OBJECT (basesink, "position format %d", format);
/* first try to get the position based on the clock */
if ((res = gst_base_sink_get_position (basesink, format, &cur))) {
gst_query_set_position (query, format, cur);
} else {
/* fallback to peer query */
res = gst_base_sink_peer_query (basesink, query);
}
break;
}
case GST_QUERY_DURATION:
GST_DEBUG_OBJECT (basesink, "duration query");
res = gst_base_sink_peer_query (basesink, query);
break;
case GST_QUERY_LATENCY:
{
gboolean live, us_live;
GstClockTime min, max;
if ((res = gst_base_sink_query_latency (basesink, &live, &us_live, &min,
&max))) {
gst_query_set_latency (query, live, min, max);
}
break;
}
case GST_QUERY_JITTER:
break;
case GST_QUERY_RATE:
/* gst_query_set_rate (query, basesink->segment_rate); */
res = TRUE;
break;
case GST_QUERY_SEGMENT:
{
/* FIXME, bring start/stop to stream time */
gst_query_set_segment (query, basesink->segment.rate,
GST_FORMAT_TIME, basesink->segment.start, basesink->segment.stop);
break;
}
case GST_QUERY_SEEKING:
case GST_QUERY_CONVERT:
case GST_QUERY_FORMATS:
default:
res = gst_base_sink_peer_query (basesink, query);
break;
}
return res;
}
static GstStateChangeReturn
gst_base_sink_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GstBaseSink *basesink = GST_BASE_SINK (element);
GstBaseSinkClass *bclass;
GstBaseSinkPrivate *priv;
priv = basesink->priv;
bclass = GST_BASE_SINK_GET_CLASS (basesink);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
if (bclass->start)
if (!bclass->start (basesink))
goto start_failed;
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
/* need to complete preroll before this state change completes, there
* is no data flow in READY so we can safely assume we need to preroll. */
GST_PAD_PREROLL_LOCK (basesink->sinkpad);
GST_DEBUG_OBJECT (basesink, "READY to PAUSED");
gst_segment_init (&basesink->segment, GST_FORMAT_UNDEFINED);
gst_segment_init (basesink->abidata.ABI.clip_segment,
GST_FORMAT_UNDEFINED);
basesink->have_newsegment = FALSE;
basesink->offset = 0;
basesink->have_preroll = FALSE;
basesink->need_preroll = TRUE;
basesink->playing_async = TRUE;
priv->current_sstart = -1;
priv->current_sstop = -1;
priv->eos_rtime = -1;
priv->latency = 0;
basesink->eos = FALSE;
priv->received_eos = FALSE;
gst_base_sink_reset_qos (basesink);
priv->commited = FALSE;
if (priv->async_enabled) {
GST_DEBUG_OBJECT (basesink, "doing async state change");
/* when async enabled, post async-start message and return ASYNC from
* the state change function */
ret = GST_STATE_CHANGE_ASYNC;
gst_element_post_message (GST_ELEMENT_CAST (basesink),
gst_message_new_async_start (GST_OBJECT_CAST (basesink), FALSE));
} else {
priv->have_latency = TRUE;
}
GST_PAD_PREROLL_UNLOCK (basesink->sinkpad);
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
GST_PAD_PREROLL_LOCK (basesink->sinkpad);
if (!gst_base_sink_needs_preroll (basesink)) {
GST_DEBUG_OBJECT (basesink, "PAUSED to PLAYING, don't need preroll");
/* no preroll needed anymore now. */
basesink->playing_async = FALSE;
basesink->need_preroll = FALSE;
if (basesink->eos) {
/* need to post EOS message here */
GST_DEBUG_OBJECT (basesink, "Now posting EOS");
gst_element_post_message (GST_ELEMENT_CAST (basesink),
gst_message_new_eos (GST_OBJECT_CAST (basesink)));
} else {
GST_DEBUG_OBJECT (basesink, "signal preroll");
GST_PAD_PREROLL_SIGNAL (basesink->sinkpad);
}
} else {
GST_DEBUG_OBJECT (basesink, "PAUSED to PLAYING, we are not prerolled");
basesink->need_preroll = TRUE;
basesink->playing_async = TRUE;
priv->commited = FALSE;
if (priv->async_enabled) {
GST_DEBUG_OBJECT (basesink, "doing async state change");
ret = GST_STATE_CHANGE_ASYNC;
gst_element_post_message (GST_ELEMENT_CAST (basesink),
gst_message_new_async_start (GST_OBJECT_CAST (basesink), FALSE));
}
}
GST_PAD_PREROLL_UNLOCK (basesink->sinkpad);
break;
default:
break;
}
{
GstStateChangeReturn bret;
bret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
if (G_UNLIKELY (bret == GST_STATE_CHANGE_FAILURE))
goto activate_failed;
}
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
/* note that this is the upward case, which doesn't follow most
patterns */
if (basesink->pad_mode == GST_ACTIVATE_PULL) {
GST_DEBUG_OBJECT (basesink, "basesink activated in pull mode, "
"returning SUCCESS directly");
GST_PAD_PREROLL_LOCK (basesink->sinkpad);
gst_element_post_message (GST_ELEMENT_CAST (basesink),
gst_message_new_async_done (GST_OBJECT_CAST (basesink)));
GST_PAD_PREROLL_UNLOCK (basesink->sinkpad);
ret = GST_STATE_CHANGE_SUCCESS;
}
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
GST_DEBUG_OBJECT (basesink, "PLAYING to PAUSED");
/* FIXME, make sure we cannot enter _render first */
/* we need to call ::unlock before locking PREROLL_LOCK
* since we lock it before going into ::render */
if (bclass->unlock)
bclass->unlock (basesink);
GST_PAD_PREROLL_LOCK (basesink->sinkpad);
/* now that we have the PREROLL lock, clear our unlock request */
if (bclass->unlock_stop)
bclass->unlock_stop (basesink);
/* we need preroll again and we set the flag before unlocking the clockid
* because if the clockid is unlocked before a current buffer expired, we
* can use that buffer to preroll with */
basesink->need_preroll = TRUE;
if (basesink->clock_id) {
gst_clock_id_unschedule (basesink->clock_id);
}
/* if we don't have a preroll buffer we need to wait for a preroll and
* return ASYNC. */
if (!gst_base_sink_needs_preroll (basesink)) {
GST_DEBUG_OBJECT (basesink, "PLAYING to PAUSED, we are prerolled");
basesink->playing_async = FALSE;
} else {
if (GST_STATE_TARGET (GST_ELEMENT (basesink)) <= GST_STATE_READY) {
ret = GST_STATE_CHANGE_SUCCESS;
} else {
GST_DEBUG_OBJECT (basesink,
"PLAYING to PAUSED, we are not prerolled");
basesink->playing_async = TRUE;
priv->commited = FALSE;
if (priv->async_enabled) {
GST_DEBUG_OBJECT (basesink, "doing async state change");
ret = GST_STATE_CHANGE_ASYNC;
gst_element_post_message (GST_ELEMENT_CAST (basesink),
gst_message_new_async_start (GST_OBJECT_CAST (basesink),
FALSE));
}
}
}
GST_DEBUG_OBJECT (basesink, "rendered: %" G_GUINT64_FORMAT
", dropped: %" G_GUINT64_FORMAT, priv->rendered, priv->dropped);
gst_base_sink_reset_qos (basesink);
GST_PAD_PREROLL_UNLOCK (basesink->sinkpad);
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
GST_PAD_PREROLL_LOCK (basesink->sinkpad);
if (!priv->commited) {
if (priv->async_enabled) {
GST_DEBUG_OBJECT (basesink, "PAUSED to READY, posting async-done");
gst_element_post_message (GST_ELEMENT_CAST (basesink),
gst_message_new_state_changed (GST_OBJECT_CAST (basesink),
GST_STATE_PLAYING, GST_STATE_PAUSED, GST_STATE_READY));
gst_element_post_message (GST_ELEMENT_CAST (basesink),
gst_message_new_async_done (GST_OBJECT_CAST (basesink)));
}
priv->commited = TRUE;
} else {
GST_DEBUG_OBJECT (basesink, "PAUSED to READY, don't need_preroll");
}
priv->current_sstart = -1;
priv->current_sstop = -1;
priv->have_latency = FALSE;
gst_base_sink_set_last_buffer (basesink, NULL);
GST_PAD_PREROLL_UNLOCK (basesink->sinkpad);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
if (bclass->stop) {
if (!bclass->stop (basesink)) {
GST_WARNING_OBJECT (basesink, "failed to stop");
}
}
gst_base_sink_set_last_buffer (basesink, NULL);
break;
default:
break;
}
return ret;
/* ERRORS */
start_failed:
{
GST_DEBUG_OBJECT (basesink, "failed to start");
return GST_STATE_CHANGE_FAILURE;
}
activate_failed:
{
GST_DEBUG_OBJECT (basesink,
"element failed to change states -- activation problem?");
return GST_STATE_CHANGE_FAILURE;
}
}