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gstreamer/libs/gst/base/gstbasesink.c
Wim Taymans 6624a8de12 Add method to allow sinks to specify additional delay between the sync times and the actual rendering of the data. 
Original commit message from CVS:
* docs/libs/gstreamer-libs-sections.txt:
* libs/gst/base/gstbasesink.c: (gst_base_sink_init),
(gst_base_sink_query_latency), (gst_base_sink_set_render_delay),
(gst_base_sink_get_render_delay), (gst_base_sink_wait_eos),
(gst_base_sink_do_sync):
* libs/gst/base/gstbasesink.h:
* win32/common/libgstbase.def:
Add method to allow sinks to specify additional delay between the sync
times and the actual rendering of the data.
API: gst_base_sink_set_render_delay()
API: gst_base_sink_get_render_delay()
2008-06-20 08:54:45 +00:00

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/* 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;
GstClockTime render_delay;
/* 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;
priv->render_delay = 0;
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_LOG_OBJECT (sink, "set async enabled to %d", 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_LOG_OBJECT (sink, "set time offset to %" G_GINT64_FORMAT, 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, render_delay;
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;
render_delay = sink->priv->render_delay;
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;
}
if (l) {
/* we need to add the render delay if we are live */
if (min != -1)
min += render_delay;
if (max != -1)
max += render_delay;
}
}
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;
}
/**
* gst_base_sink_set_render_delay:
* @sink: a #GstBaseSink
* @delay: the new delay
*
* Set the render delay in @sink to @delay. The render delay is the time
* between actual rendering of a buffer and its synchronisation time. Some
* devices might delay media rendering which can be compensated for with this
* function.
*
* After calling this function, this sink will report additional latency and
* other sinks will adjust their latency to delay the rendering of their media.
*
* This function is usually called by subclasses.
*
* Since: 0.10.21
*/
void
gst_base_sink_set_render_delay (GstBaseSink * sink, GstClockTime delay)
{
g_return_if_fail (GST_IS_BASE_SINK (sink));
GST_OBJECT_LOCK (sink);
sink->priv->render_delay = delay;
GST_LOG_OBJECT (sink, "set render delay to %" GST_TIME_FORMAT,
GST_TIME_ARGS (delay));
GST_OBJECT_UNLOCK (sink);
}
/**
* gst_base_sink_get_render_delay:
* @sink: a #GstBaseSink
*
* Get the render delay of @sink. see gst_base_sink_set_render_delay() for more
* information about the render delay.
*
* Returns: the render delay of @sink.
*
* Since: 0.10.21
*/
GstClockTime
gst_base_sink_get_render_delay (GstBaseSink * sink)
{
GstClockTimeDiff res;
g_return_val_if_fail (GST_IS_BASE_SINK (sink), 0);
GST_OBJECT_LOCK (sink);
res = sink->priv->render_delay;
GST_OBJECT_UNLOCK (sink);
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:
{
if (basesink->segment.rate >= 0.0) {
sstart = sstop = priv->current_sstop;
if (sstart == -1) {
/* we have not seen a buffer yet, use the segment values */
sstart = sstop = gst_segment_to_stream_time (&basesink->segment,
basesink->segment.format, basesink->segment.stop);
}
} else {
sstart = sstop = priv->current_sstart;
if (sstart == -1) {
/* we have not seen a buffer yet, use the segment values */
sstart = sstop = gst_segment_to_stream_time (&basesink->segment,
basesink->segment.format, basesink->segment.start);
}
}
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;
}
default:
/* 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.
*/
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 */
if (G_LIKELY (cstop != GST_CLOCK_TIME_NONE))
gst_segment_set_last_stop (segment, GST_FORMAT_TIME, cstop);
else
gst_segment_set_last_stop (segment, GST_FORMAT_TIME, cstart);
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, LOCK
* adjust a timestamp with the latency and timestamp offset */
static GstClockTime
gst_base_sink_adjust_time (GstBaseSink * basesink, GstClockTime time)
{
GstClockTimeDiff ts_offset;
/* don't do anything funny with invalid timestamps */
if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (time)))
return 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;
return time;
}
/* gst_base_sink_wait_clock:
* @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.
*
* If @time is not valid, no sycnhronisation is done and #GST_CLOCK_BADTIME is
* returned. Likewise, if synchronisation is disabled in the element or there
* is no clock, no synchronisation is done and #GST_CLOCK_BADTIME is returned.
*
* This function should only be called with the PREROLL_LOCK held, like when
* receiving an EOS event in the ::event vmethod or when receiving a buffer in
* the ::render vmethod.
*
* The @time argument should be the running_time of when this method should
* return and is not adjusted with any latency or offset configured in the
* sink.
*
* Since 0.10.20
*
* Returns: #GstClockReturn
*/
GstClockReturn
gst_base_sink_wait_clock (GstBaseSink * basesink, GstClockTime time,
GstClockTimeDiff * jitter)
{
GstClockID id;
GstClockReturn ret;
GstClock *clock;
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 to running_time to get the time against the clock */
time += GST_ELEMENT_CAST (basesink)->base_time;
id = gst_clock_new_single_shot_id (clock, time);
GST_OBJECT_UNLOCK (basesink);
/* 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. */
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.
*
* The @time argument should be the running_time of when the EOS should happen
* and will be adjusted with any latency and offset configured in the sink.
*
* Since 0.10.15
*
* Returns: #GstFlowReturn
*/
GstFlowReturn
gst_base_sink_wait_eos (GstBaseSink * sink, GstClockTime time,
GstClockTimeDiff * jitter)
{
GstClockReturn status;
GstFlowReturn ret;
do {
GstClockTime stime;
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));
/* compensate for latency and ts_offset. We don't adjust for render delay
* because we don't interact with the device on EOS normally. */
stime = gst_base_sink_adjust_time (sink, time);
/* wait for the clock, this can be interrupted because we got shut down or
* we PAUSED. */
status = gst_base_sink_wait_clock (sink, stime, 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, stime;
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;
/* adjust for latency */
stime = gst_base_sink_adjust_time (basesink, rstart);
/* adjust for render-delay, avoid underflows */
if (stime != -1) {
if (stime > priv->render_delay)
stime -= priv->render_delay;
else
stime = 0;
}
/* preroll done, we can sync since we are in PLAYING now. */
GST_DEBUG_OBJECT (basesink, "possibly waiting for clock to reach %"
GST_TIME_FORMAT ", adjusted %" GST_TIME_FORMAT,
GST_TIME_ARGS (rstart), GST_TIME_ARGS (stime));
/* 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, stime, &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;
if (jitter < 0) {
/* this is the time the buffer entered the sink */
if (start < -jitter)
entered = 0;
else
entered = start + jitter;
left = start;
} else {
/* this is the time the buffer entered the sink */
entered = start + jitter;
/* this is the time the buffer left the sink */
left = start + 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 (priv->avg_rate >= 0.0) {
/* if we have a valid rate, start sending QoS messages */
if (priv->current_jitter < 0) {
/* make sure we never go below 0 when adding the jitter to the
* timestamp. */
if (priv->current_rstart < -priv->current_jitter)
priv->current_jitter = -priv->current_rstart;
}
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;
basesink->priv->eos_rtime = -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)
{
GstBaseSinkClass *bclass;
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;
}
bclass = GST_BASE_SINK_GET_CLASS (basesink);
/* check if the buffer needs to be dropped, we first ask the subclass for the
* start and end */
if (bclass->get_times)
bclass->get_times (basesink, buf, &start, &end);
if (start == -1) {
/* if the subclass does not want sync, we use our own values so that we at
* least clip the buffer to the segment */
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, this will
* also flush out the EOS state */
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;
}
}
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