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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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:gstbasesink
* @short_description: Base class for sink elements
* @see_also: #GstBaseTransform, #GstBaseSrc
*
* #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 class_init function, like so:
* |[
* static void
* my_element_class_init (GstMyElementClass *klass)
* {
* GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
*
* // 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));
*
* gst_element_class_set_static_metadata (gstelement_class,
* "Sink name",
* "Sink",
* "My Sink element",
* "The author &lt;my.sink@my.email&gt;");
* }
* ]|
*
* #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
* #GstBaseSinkClass.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 #GstBaseSinkClass.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
* #GstBaseSink:sync property to %FALSE.
*
* After synchronisation the virtual method #GstBaseSinkClass.render() will be
* called. Subclasses should minimally implement this method.
*
* Subclasses that synchronise on the clock in the #GstBaseSinkClass.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.
* 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
* #GstBaseSinkClass.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_SEGMENT 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 #GstBaseSinkClass.set_caps() function will be called when the subclass
* should configure itself to process a specific media type.
*
* The #GstBaseSinkClass.start() and #GstBaseSinkClass.stop() virtual methods
* will be called when resources should be allocated. Any
* #GstBaseSinkClass.preroll(), #GstBaseSinkClass.render() and
* #GstBaseSinkClass.set_caps() function will be called between the
* #GstBaseSinkClass.start() and #GstBaseSinkClass.stop() calls.
*
* The #GstBaseSinkClass.event() virtual method will be called when an event is
* received by #GstBaseSink. Normally this method should only be overridden by
* very specific elements (such as file sinks) which need to handle the
* newsegment event specially.
*
* The #GstBaseSinkClass.unlock() method is called when the elements should
* unblock any blocking operations they perform in the
* #GstBaseSinkClass.render() method. This is mostly useful when the
* #GstBaseSinkClass.render() method performs a blocking write on a file
* descriptor, for example.
*
* The #GstBaseSink: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
* #GstBaseSinkClass.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 #GstBaseSink: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.
*
* The #GstBaseSink: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.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <gst/gst_private.h>
#include "gstbasesink.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))
#define GST_FLOW_STEP GST_FLOW_CUSTOM_ERROR
typedef struct
{
gboolean valid; /* if this info is valid */
guint32 seqnum; /* the seqnum of the STEP event */
GstFormat format; /* the format of the amount */
guint64 amount; /* the total amount of data to skip */
guint64 position; /* the position in the stepped data */
guint64 duration; /* the duration in time of the skipped data */
guint64 start; /* running_time of the start */
gdouble rate; /* rate of skipping */
gdouble start_rate; /* rate before skipping */
guint64 start_start; /* start position skipping */
guint64 start_stop; /* stop position skipping */
gboolean flush; /* if this was a flushing step */
gboolean intermediate; /* if this is an intermediate step */
gboolean need_preroll; /* if we need preroll after this step */
} GstStepInfo;
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;
/* the running time of the previous buffer */
GstClockTime prev_rstart;
/* EOS sync time in running time */
GstClockTime eos_rtime;
/* last buffer that arrived in time, running time */
GstClockTime last_render_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;
GstClockTime avg_in_diff;
/* 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;
/* state change to playing ongoing */
gboolean to_playing;
/* 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 */
gint enable_last_sample; /* atomic */
GstBuffer *last_buffer;
GstCaps *last_caps;
/* negotiated caps */
GstCaps *caps;
/* blocksize for pulling */
guint blocksize;
gboolean discont;
/* seqnum of the stream */
guint32 seqnum;
gboolean call_preroll;
gboolean step_unlock;
/* we have a pending and a current step operation */
GstStepInfo current_step;
GstStepInfo pending_step;
/* Cached GstClockID */
GstClockID cached_clock_id;
/* for throttling and QoS */
GstClockTime earliest_in_time;
GstClockTime throttle_time;
/* for rate control */
guint64 max_bitrate;
GstClockTime rc_time;
GstClockTime rc_next;
gsize rc_accumulated;
};
#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.
* positive 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_CAN_ACTIVATE_PULL FALSE /* fixme: enable me */
#define DEFAULT_CAN_ACTIVATE_PUSH TRUE
#define DEFAULT_SYNC TRUE
#define DEFAULT_MAX_LATENESS -1
#define DEFAULT_QOS FALSE
#define DEFAULT_ASYNC TRUE
#define DEFAULT_TS_OFFSET 0
#define DEFAULT_BLOCKSIZE 4096
#define DEFAULT_RENDER_DELAY 0
#define DEFAULT_ENABLE_LAST_SAMPLE TRUE
#define DEFAULT_THROTTLE_TIME 0
#define DEFAULT_MAX_BITRATE 0
enum
{
PROP_0,
PROP_SYNC,
PROP_MAX_LATENESS,
PROP_QOS,
PROP_ASYNC,
PROP_TS_OFFSET,
PROP_ENABLE_LAST_SAMPLE,
PROP_LAST_SAMPLE,
PROP_BLOCKSIZE,
PROP_RENDER_DELAY,
PROP_THROTTLE_TIME,
PROP_MAX_BITRATE,
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 volatile gsize base_sink_type = 0;
if (g_once_init_enter (&base_sink_type)) {
GType _type;
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,
};
_type = g_type_register_static (GST_TYPE_ELEMENT,
"GstBaseSink", &base_sink_info, G_TYPE_FLAG_ABSTRACT);
g_once_init_leave (&base_sink_type, _type);
}
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 default_element_query (GstElement * element, GstQuery * query);
static GstCaps *gst_base_sink_default_get_caps (GstBaseSink * sink,
GstCaps * caps);
static gboolean gst_base_sink_default_set_caps (GstBaseSink * sink,
GstCaps * caps);
static void gst_base_sink_default_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 gboolean gst_base_sink_default_do_seek (GstBaseSink * sink,
GstSegment * segment);
static gboolean gst_base_sink_default_prepare_seek_segment (GstBaseSink * sink,
GstEvent * event, GstSegment * segment);
static GstStateChangeReturn gst_base_sink_change_state (GstElement * element,
GstStateChange transition);
static gboolean gst_base_sink_sink_query (GstPad * pad, GstObject * parent,
GstQuery * query);
static GstFlowReturn gst_base_sink_chain (GstPad * pad, GstObject * parent,
GstBuffer * buffer);
static GstFlowReturn gst_base_sink_chain_list (GstPad * pad, GstObject * parent,
GstBufferList * list);
static void gst_base_sink_loop (GstPad * pad);
static gboolean gst_base_sink_pad_activate (GstPad * pad, GstObject * parent);
static gboolean gst_base_sink_pad_activate_mode (GstPad * pad,
GstObject * parent, GstPadMode mode, gboolean active);
static gboolean gst_base_sink_default_event (GstBaseSink * basesink,
GstEvent * event);
static GstFlowReturn gst_base_sink_default_wait_event (GstBaseSink * basesink,
GstEvent * event);
static gboolean gst_base_sink_event (GstPad * pad, GstObject * parent,
GstEvent * event);
static gboolean gst_base_sink_default_query (GstBaseSink * sink,
GstQuery * query);
static gboolean gst_base_sink_negotiate_pull (GstBaseSink * basesink);
static GstCaps *gst_base_sink_default_fixate (GstBaseSink * bsink,
GstCaps * caps);
static GstCaps *gst_base_sink_fixate (GstBaseSink * bsink, GstCaps * caps);
/* check if an object was too late */
static gboolean gst_base_sink_is_too_late (GstBaseSink * basesink,
GstMiniObject * obj, GstClockTime rstart, GstClockTime rstop,
GstClockReturn status, GstClockTimeDiff jitter, gboolean render);
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_base_sink_finalize;
gobject_class->set_property = gst_base_sink_set_property;
gobject_class->get_property = gst_base_sink_get_property;
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.
*/
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.
*/
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:enable-last-sample:
*
* Enable the last-sample property. If %FALSE, basesink doesn't keep a
* reference to the last buffer arrived and the last-sample property is always
* set to %NULL. This can be useful if you need buffers to be released as soon
* as possible, eg. if you're using a buffer pool.
*/
g_object_class_install_property (gobject_class, PROP_ENABLE_LAST_SAMPLE,
g_param_spec_boolean ("enable-last-sample", "Enable Last Buffer",
"Enable the last-sample property", DEFAULT_ENABLE_LAST_SAMPLE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstBaseSink:last-sample:
*
* 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 buffer.
*/
g_object_class_install_property (gobject_class, PROP_LAST_SAMPLE,
g_param_spec_boxed ("last-sample", "Last Sample",
"The last sample received in the sink", GST_TYPE_SAMPLE,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
/**
* GstBaseSink:blocksize:
*
* The amount of bytes to pull when operating in pull mode.
*/
/* FIXME 2.0: blocksize property should be int, otherwise min>max.. */
g_object_class_install_property (gobject_class, PROP_BLOCKSIZE,
g_param_spec_uint ("blocksize", "Block size",
"Size in bytes to pull per buffer (0 = default)", 0, G_MAXUINT,
DEFAULT_BLOCKSIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstBaseSink:render-delay:
*
* The additional delay between synchronisation and actual rendering of the
* media. This property will add additional latency to the device in order to
* make other sinks compensate for the delay.
*/
g_object_class_install_property (gobject_class, PROP_RENDER_DELAY,
g_param_spec_uint64 ("render-delay", "Render Delay",
"Additional render delay of the sink in nanoseconds", 0, G_MAXUINT64,
DEFAULT_RENDER_DELAY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstBaseSink:throttle-time:
*
* The time to insert between buffers. This property can be used to control
* the maximum amount of buffers per second to render. Setting this property
* to a value bigger than 0 will make the sink create THROTTLE QoS events.
*/
g_object_class_install_property (gobject_class, PROP_THROTTLE_TIME,
g_param_spec_uint64 ("throttle-time", "Throttle time",
"The time to keep between rendered buffers (0 = disabled)", 0,
G_MAXUINT64, DEFAULT_THROTTLE_TIME,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstBaseSink:max-bitrate:
*
* Control the maximum amount of bits that will be rendered per second.
* Setting this property to a value bigger than 0 will make the sink delay
* rendering of the buffers when it would exceed to max-bitrate.
*
* Since: 1.2
*/
g_object_class_install_property (gobject_class, PROP_MAX_BITRATE,
g_param_spec_uint64 ("max-bitrate", "Max Bitrate",
"The maximum bits per second to render (0 = disabled)", 0,
G_MAXUINT64, DEFAULT_MAX_BITRATE,
G_PARAM_READWRITE | 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 (default_element_query);
klass->get_caps = GST_DEBUG_FUNCPTR (gst_base_sink_default_get_caps);
klass->set_caps = GST_DEBUG_FUNCPTR (gst_base_sink_default_set_caps);
klass->fixate = GST_DEBUG_FUNCPTR (gst_base_sink_default_fixate);
klass->activate_pull =
GST_DEBUG_FUNCPTR (gst_base_sink_default_activate_pull);
klass->get_times = GST_DEBUG_FUNCPTR (gst_base_sink_default_get_times);
klass->query = GST_DEBUG_FUNCPTR (gst_base_sink_default_query);
klass->event = GST_DEBUG_FUNCPTR (gst_base_sink_default_event);
klass->wait_event = GST_DEBUG_FUNCPTR (gst_base_sink_default_wait_event);
/* Registering debug symbols for function pointers */
GST_DEBUG_REGISTER_FUNCPTR (gst_base_sink_fixate);
GST_DEBUG_REGISTER_FUNCPTR (gst_base_sink_pad_activate);
GST_DEBUG_REGISTER_FUNCPTR (gst_base_sink_pad_activate_mode);
GST_DEBUG_REGISTER_FUNCPTR (gst_base_sink_event);
GST_DEBUG_REGISTER_FUNCPTR (gst_base_sink_chain);
GST_DEBUG_REGISTER_FUNCPTR (gst_base_sink_chain_list);
GST_DEBUG_REGISTER_FUNCPTR (gst_base_sink_sink_query);
}
static GstCaps *
gst_base_sink_query_caps (GstBaseSink * bsink, GstPad * pad, GstCaps * filter)
{
GstBaseSinkClass *bclass;
GstCaps *caps = NULL;
gboolean fixed;
bclass = GST_BASE_SINK_GET_CLASS (bsink);
fixed = GST_PAD_IS_FIXED_CAPS (pad);
if (fixed || bsink->pad_mode == GST_PAD_MODE_PULL) {
/* if we are operating in pull mode or fixed caps, we only accept the
* currently negotiated caps */
caps = gst_pad_get_current_caps (pad);
}
if (caps == NULL) {
if (bclass->get_caps)
caps = bclass->get_caps (bsink, filter);
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_pad_template_get_caps (pad_template);
if (filter) {
GstCaps *intersection;
intersection =
gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (caps);
caps = intersection;
}
}
}
}
return caps;
}
static GstCaps *
gst_base_sink_default_fixate (GstBaseSink * bsink, GstCaps * caps)
{
GST_DEBUG_OBJECT (bsink, "using default caps fixate function");
return gst_caps_fixate (caps);
}
static GstCaps *
gst_base_sink_fixate (GstBaseSink * bsink, GstCaps * caps)
{
GstBaseSinkClass *bclass;
bclass = GST_BASE_SINK_GET_CLASS (bsink);
if (bclass->fixate)
caps = bclass->fixate (bsink, caps);
return caps;
}
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_activate_function (basesink->sinkpad, gst_base_sink_pad_activate);
gst_pad_set_activatemode_function (basesink->sinkpad,
gst_base_sink_pad_activate_mode);
gst_pad_set_query_function (basesink->sinkpad, gst_base_sink_sink_query);
gst_pad_set_event_function (basesink->sinkpad, gst_base_sink_event);
gst_pad_set_chain_function (basesink->sinkpad, gst_base_sink_chain);
gst_pad_set_chain_list_function (basesink->sinkpad, gst_base_sink_chain_list);
gst_element_add_pad (GST_ELEMENT_CAST (basesink), basesink->sinkpad);
basesink->pad_mode = GST_PAD_MODE_NONE;
g_mutex_init (&basesink->preroll_lock);
g_cond_init (&basesink->preroll_cond);
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->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 = DEFAULT_RENDER_DELAY;
priv->blocksize = DEFAULT_BLOCKSIZE;
priv->cached_clock_id = NULL;
g_atomic_int_set (&priv->enable_last_sample, DEFAULT_ENABLE_LAST_SAMPLE);
priv->throttle_time = DEFAULT_THROTTLE_TIME;
priv->max_bitrate = DEFAULT_MAX_BITRATE;
GST_OBJECT_FLAG_SET (basesink, GST_ELEMENT_FLAG_SINK);
}
static void
gst_base_sink_finalize (GObject * object)
{
GstBaseSink *basesink;
basesink = GST_BASE_SINK (object);
g_mutex_clear (&basesink->preroll_lock);
g_cond_clear (&basesink->preroll_cond);
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, incoming samples will be played as fast as
* possible. If @sync is %TRUE, the timestamps of the incoming
* buffers will be used to schedule the exact render time of its
* contents.
*/
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.
*/
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.
*/
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->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.
*/
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->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.
*/
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.
*/
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 asynchronously. When async is
* disabled, the sink will immediately go to PAUSED instead of waiting for a
* preroll buffer. This feature is useful if the sink does not synchronize
* against the clock or when it is dealing with sparse streams.
*/
void
gst_base_sink_set_async_enabled (GstBaseSink * sink, gboolean enabled)
{
g_return_if_fail (GST_IS_BASE_SINK (sink));
GST_BASE_SINK_PREROLL_LOCK (sink);
g_atomic_int_set (&sink->priv->async_enabled, enabled);
GST_LOG_OBJECT (sink, "set async enabled to %d", enabled);
GST_BASE_SINK_PREROLL_UNLOCK (sink);
}
/**
* 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.
*/
gboolean
gst_base_sink_is_async_enabled (GstBaseSink * sink)
{
gboolean res;
g_return_val_if_fail (GST_IS_BASE_SINK (sink), FALSE);
res = g_atomic_int_get (&sink->priv->async_enabled);
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.
*/
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.
*/
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_sample:
* @sink: the sink
*
* Get the last sample 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 sample can be used to determine the type of the buffer.
*
* Free-function: gst_sample_unref
*
* Returns: (transfer full) (nullable): a #GstSample. gst_sample_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.
*/
GstSample *
gst_base_sink_get_last_sample (GstBaseSink * sink)
{
GstSample *res = NULL;
g_return_val_if_fail (GST_IS_BASE_SINK (sink), NULL);
GST_OBJECT_LOCK (sink);
if (sink->priv->last_buffer) {
res = gst_sample_new (sink->priv->last_buffer,
sink->priv->last_caps, &sink->segment, NULL);
}
GST_OBJECT_UNLOCK (sink);
return res;
}
/* with OBJECT_LOCK */
static void
gst_base_sink_set_last_buffer_unlocked (GstBaseSink * sink, GstBuffer * buffer)
{
GstBuffer *old;
old = sink->priv->last_buffer;
if (G_LIKELY (old != buffer)) {
GST_DEBUG_OBJECT (sink, "setting last buffer to %p", buffer);
if (G_LIKELY (buffer))
gst_buffer_ref (buffer);
sink->priv->last_buffer = buffer;
if (buffer)
/* copy over the caps */
gst_caps_replace (&sink->priv->last_caps, sink->priv->caps);
else
gst_caps_replace (&sink->priv->last_caps, NULL);
} else {
old = NULL;
}
/* avoid unreffing with the lock because cleanup code might want to take the
* lock too */
if (G_LIKELY (old)) {
GST_OBJECT_UNLOCK (sink);
gst_buffer_unref (old);
GST_OBJECT_LOCK (sink);
}
}
static void
gst_base_sink_set_last_buffer (GstBaseSink * sink, GstBuffer * buffer)
{
if (!g_atomic_int_get (&sink->priv->enable_last_sample))
return;
GST_OBJECT_LOCK (sink);
gst_base_sink_set_last_buffer_unlocked (sink, buffer);
GST_OBJECT_UNLOCK (sink);
}
/**
* gst_base_sink_set_last_sample_enabled:
* @sink: the sink
* @enabled: the new enable-last-sample value.
*
* Configures @sink to store the last received sample in the last-sample
* property.
*/
void
gst_base_sink_set_last_sample_enabled (GstBaseSink * sink, gboolean enabled)
{
g_return_if_fail (GST_IS_BASE_SINK (sink));
/* Only take lock if we change the value */
if (g_atomic_int_compare_and_exchange (&sink->priv->enable_last_sample,
!enabled, enabled) && !enabled) {
GST_OBJECT_LOCK (sink);
gst_base_sink_set_last_buffer_unlocked (sink, NULL);
GST_OBJECT_UNLOCK (sink);
}
}
/**
* gst_base_sink_is_last_sample_enabled:
* @sink: the sink
*
* Checks if @sink is currently configured to store the last received sample in
* the last-sample property.
*
* Returns: %TRUE if the sink is configured to store the last received sample.
*/
gboolean
gst_base_sink_is_last_sample_enabled (GstBaseSink * sink)
{
g_return_val_if_fail (GST_IS_BASE_SINK (sink), FALSE);
return g_atomic_int_get (&sink->priv->enable_last_sample);
}
/**
* gst_base_sink_get_latency:
* @sink: the sink
*
* Get the currently configured latency.
*
* Returns: The configured latency.
*/
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: (out) (allow-none): if the sink is live
* @upstream_live: (out) (allow-none): if an upstream element is live
* @min_latency: (out) (allow-none): the min latency of the upstream elements
* @max_latency: (out) (allow-none): 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 positive value.
*
* This function is mostly used by subclasses.
*
* Returns: %TRUE if the query succeeded.
*/
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_pad_peer_query (sink->sinkpad, 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 */
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.
*/
void
gst_base_sink_set_render_delay (GstBaseSink * sink, GstClockTime delay)
{
GstClockTime old_render_delay;
g_return_if_fail (GST_IS_BASE_SINK (sink));
GST_OBJECT_LOCK (sink);
old_render_delay = sink->priv->render_delay;
sink->priv->render_delay = delay;
GST_LOG_OBJECT (sink, "set render delay to %" GST_TIME_FORMAT,
GST_TIME_ARGS (delay));
GST_OBJECT_UNLOCK (sink);
if (delay != old_render_delay) {
GST_DEBUG_OBJECT (sink, "posting latency changed");
gst_element_post_message (GST_ELEMENT_CAST (sink),
gst_message_new_latency (GST_OBJECT_CAST (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.
*/
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;
}
/**
* gst_base_sink_set_blocksize:
* @sink: a #GstBaseSink
* @blocksize: the blocksize in bytes
*
* Set the number of bytes that the sink will pull when it is operating in pull
* mode.
*/
/* FIXME 2.0: blocksize property should be int, otherwise min>max.. */
void
gst_base_sink_set_blocksize (GstBaseSink * sink, guint blocksize)
{
g_return_if_fail (GST_IS_BASE_SINK (sink));
GST_OBJECT_LOCK (sink);
sink->priv->blocksize = blocksize;
GST_LOG_OBJECT (sink, "set blocksize to %u", blocksize);
GST_OBJECT_UNLOCK (sink);
}
/**
* gst_base_sink_get_blocksize:
* @sink: a #GstBaseSink
*
* Get the number of bytes that the sink will pull when it is operating in pull
* mode.
*
* Returns: the number of bytes @sink will pull in pull mode.
*/
/* FIXME 2.0: blocksize property should be int, otherwise min>max.. */
guint
gst_base_sink_get_blocksize (GstBaseSink * sink)
{
guint res;
g_return_val_if_fail (GST_IS_BASE_SINK (sink), 0);
GST_OBJECT_LOCK (sink);
res = sink->priv->blocksize;
GST_OBJECT_UNLOCK (sink);
return res;
}
/**
* gst_base_sink_set_throttle_time:
* @sink: a #GstBaseSink
* @throttle: the throttle time in nanoseconds
*
* Set the time that will be inserted between rendered buffers. This
* can be used to control the maximum buffers per second that the sink
* will render.
*/
void
gst_base_sink_set_throttle_time (GstBaseSink * sink, guint64 throttle)
{
g_return_if_fail (GST_IS_BASE_SINK (sink));
GST_OBJECT_LOCK (sink);
sink->priv->throttle_time = throttle;
GST_LOG_OBJECT (sink, "set throttle_time to %" G_GUINT64_FORMAT, throttle);
GST_OBJECT_UNLOCK (sink);
}
/**
* gst_base_sink_get_throttle_time:
* @sink: a #GstBaseSink
*
* Get the time that will be inserted between frames to control the
* maximum buffers per second.
*
* Returns: the number of nanoseconds @sink will put between frames.
*/
guint64
gst_base_sink_get_throttle_time (GstBaseSink * sink)
{
guint64 res;
g_return_val_if_fail (GST_IS_BASE_SINK (sink), 0);
GST_OBJECT_LOCK (sink);
res = sink->priv->throttle_time;
GST_OBJECT_UNLOCK (sink);
return res;
}
/**
* gst_base_sink_set_max_bitrate:
* @sink: a #GstBaseSink
* @max_bitrate: the max_bitrate in bits per second
*
* Set the maximum amount of bits per second that the sink will render.
*
* Since: 1.2
*/
void
gst_base_sink_set_max_bitrate (GstBaseSink * sink, guint64 max_bitrate)
{
g_return_if_fail (GST_IS_BASE_SINK (sink));
GST_OBJECT_LOCK (sink);
sink->priv->max_bitrate = max_bitrate;
GST_LOG_OBJECT (sink, "set max_bitrate to %" G_GUINT64_FORMAT, max_bitrate);
GST_OBJECT_UNLOCK (sink);
}
/**
* gst_base_sink_get_max_bitrate:
* @sink: a #GstBaseSink
*
* Get the maximum amount of bits per second that the sink will render.
*
* Returns: the maximum number of bits per second @sink will render.
*
* Since: 1.2
*/
guint64
gst_base_sink_get_max_bitrate (GstBaseSink * sink)
{
guint64 res;
g_return_val_if_fail (GST_IS_BASE_SINK (sink), 0);
GST_OBJECT_LOCK (sink);
res = sink->priv->max_bitrate;
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_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;
case PROP_BLOCKSIZE:
gst_base_sink_set_blocksize (sink, g_value_get_uint (value));
break;
case PROP_RENDER_DELAY:
gst_base_sink_set_render_delay (sink, g_value_get_uint64 (value));
break;
case PROP_ENABLE_LAST_SAMPLE:
gst_base_sink_set_last_sample_enabled (sink, g_value_get_boolean (value));
break;
case PROP_THROTTLE_TIME:
gst_base_sink_set_throttle_time (sink, g_value_get_uint64 (value));
break;
case PROP_MAX_BITRATE:
gst_base_sink_set_max_bitrate (sink, g_value_get_uint64 (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_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_SAMPLE:
gst_value_take_sample (value, gst_base_sink_get_last_sample (sink));
break;
case PROP_ENABLE_LAST_SAMPLE:
g_value_set_boolean (value, gst_base_sink_is_last_sample_enabled (sink));
break;
case PROP_BLOCKSIZE:
g_value_set_uint (value, gst_base_sink_get_blocksize (sink));
break;
case PROP_RENDER_DELAY:
g_value_set_uint64 (value, gst_base_sink_get_render_delay (sink));
break;
case PROP_THROTTLE_TIME:
g_value_set_uint64 (value, gst_base_sink_get_throttle_time (sink));
break;
case PROP_MAX_BITRATE:
g_value_set_uint64 (value, gst_base_sink_get_max_bitrate (sink));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstCaps *
gst_base_sink_default_get_caps (GstBaseSink * sink, GstCaps * filter)
{
return NULL;
}
static gboolean
gst_base_sink_default_set_caps (GstBaseSink * sink, GstCaps * caps)
{
return TRUE;
}
/* 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;
/* 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;
switch (pending) {
case GST_STATE_PLAYING:
{
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;
}
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),
GST_CLOCK_TIME_NONE));
}
if (post_playing) {
if (post_paused) {
GstElementClass *klass;
klass = GST_ELEMENT_GET_CLASS (basesink);
basesink->have_preroll = TRUE;
/* after releasing this lock, the state change function
* can execute concurrently with this thread. There is nothing we do to
* prevent this for now. subclasses should be prepared to handle it. */
GST_BASE_SINK_PREROLL_UNLOCK (basesink);
if (klass->change_state)
klass->change_state (GST_ELEMENT_CAST (basesink),
GST_STATE_CHANGE_PAUSED_TO_PLAYING);
GST_BASE_SINK_PREROLL_LOCK (basesink);
/* state change function could have been executed and we could be
* flushing now */
if (G_UNLIKELY (basesink->flushing))
goto stopping_unlocked;
}
GST_DEBUG_OBJECT (basesink, "posting PLAYING state change message");
/* FIXME, we released the PREROLL lock above, it's possible that this
* message is not correct anymore when the element went back to PAUSED */
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_unlocked:
{
GST_OBJECT_LOCK (basesink);
goto stopping;
}
stopping:
{
/* app is going to READY */
GST_DEBUG_OBJECT (basesink, "stopping");
basesink->need_preroll = FALSE;
basesink->flushing = TRUE;
GST_OBJECT_UNLOCK (basesink);
return FALSE;
}
}
static void
start_stepping (GstBaseSink * sink, GstSegment * segment,
GstStepInfo * pending, GstStepInfo * current)
{
gint64 end;
GstMessage *message;
GST_DEBUG_OBJECT (sink, "update pending step");
GST_OBJECT_LOCK (sink);
memcpy (current, pending, sizeof (GstStepInfo));
pending->valid = FALSE;
GST_OBJECT_UNLOCK (sink);
/* post message first */
message =
gst_message_new_step_start (GST_OBJECT (sink), TRUE, current->format,
current->amount, current->rate, current->flush, current->intermediate);
gst_message_set_seqnum (message, current->seqnum);
gst_element_post_message (GST_ELEMENT (sink), message);
/* get the running time of where we paused and remember it */
current->start = gst_element_get_start_time (GST_ELEMENT_CAST (sink));
gst_segment_set_running_time (segment, GST_FORMAT_TIME, current->start);
/* set the new rate for the remainder of the segment */
current->start_rate = segment->rate;
segment->rate *= current->rate;
/* save values */
if (segment->rate > 0.0)
current->start_stop = segment->stop;
else
current->start_start = segment->start;
if (current->format == GST_FORMAT_TIME) {
/* calculate the running-time when the step operation should stop */
if (current->amount != -1)
end = current->start + current->amount;
else
end = -1;
if (!current->flush) {
gint64 position;
/* update the segment clipping regions for non-flushing seeks */
if (segment->rate > 0.0) {
if (end != -1)
position = gst_segment_to_position (segment, GST_FORMAT_TIME, end);
else
position = segment->stop;
segment->stop = position;
segment->position = position;
} else {
if (end != -1)
position = gst_segment_to_position (segment, GST_FORMAT_TIME, end);
else
position = segment->start;
segment->time = position;
segment->start = position;
segment->position = position;
}
}
}
GST_DEBUG_OBJECT (sink, "segment now %" GST_SEGMENT_FORMAT, segment);
GST_DEBUG_OBJECT (sink, "step started at running_time %" GST_TIME_FORMAT,
GST_TIME_ARGS (current->start));
GST_DEBUG_OBJECT (sink, "step amount: %" G_GUINT64_FORMAT ", format: %s, "
"rate: %f", current->amount, gst_format_get_name (current->format),
current->rate);
}
static void
stop_stepping (GstBaseSink * sink, GstSegment * segment,
GstStepInfo * current, gint64 rstart, gint64 rstop, gboolean eos)
{
gint64 stop, position;
GstMessage *message;
GST_DEBUG_OBJECT (sink, "step complete");
if (segment->rate > 0.0)
stop = rstart;
else
stop = rstop;
GST_DEBUG_OBJECT (sink,
"step stop at running_time %" GST_TIME_FORMAT, GST_TIME_ARGS (stop));
if (stop == -1)
current->duration = current->position;
else
current->duration = stop - current->start;
GST_DEBUG_OBJECT (sink, "step elapsed running_time %" GST_TIME_FORMAT,
GST_TIME_ARGS (current->duration));
position = current->start + current->duration;
/* now move the segment to the new running time */
gst_segment_set_running_time (segment, GST_FORMAT_TIME, position);
if (current->flush) {
/* and remove the time we flushed, start time did not change */
segment->base = current->start;
} else {
/* start time is now the stepped position */
gst_element_set_start_time (GST_ELEMENT_CAST (sink), position);
}
/* restore the previous rate */
segment->rate = current->start_rate;
if (segment->rate > 0.0)
segment->stop = current->start_stop;
else
segment->start = current->start_start;
/* post the step done when we know the stepped duration in TIME */
message =
gst_message_new_step_done (GST_OBJECT_CAST (sink), current->format,
current->amount, current->rate, current->flush, current->intermediate,
current->duration, eos);
gst_message_set_seqnum (message, current->seqnum);
gst_element_post_message (GST_ELEMENT_CAST (sink), message);
if (!current->intermediate)
sink->need_preroll = current->need_preroll;
/* and the current step info finished and becomes invalid */
current->valid = FALSE;
}
static gboolean
handle_stepping (GstBaseSink * sink, GstSegment * segment,
GstStepInfo * current, guint64 * cstart, guint64 * cstop, guint64 * rstart,
guint64 * rstop)
{
gboolean step_end = FALSE;
/* stepping never stops */
if (current->amount == -1)
return FALSE;
/* see if we need to skip this buffer because of stepping */
switch (current->format) {
case GST_FORMAT_TIME:
{
guint64 end;
guint64 first, last;
gdouble abs_rate;
if (segment->rate > 0.0) {
if (segment->stop == *cstop)
*rstop = *rstart + current->amount;
first = *rstart;
last = *rstop;
} else {
if (segment->start == *cstart)
*rstart = *rstop + current->amount;
first = *rstop;
last = *rstart;
}
end = current->start + current->amount;
current->position = first - current->start;
abs_rate = ABS (segment->rate);
if (G_UNLIKELY (abs_rate != 1.0))
current->position /= abs_rate;
GST_DEBUG_OBJECT (sink,
"buffer: %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT,
GST_TIME_ARGS (first), GST_TIME_ARGS (last));
GST_DEBUG_OBJECT (sink,
"got time step %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT "/%"
GST_TIME_FORMAT, GST_TIME_ARGS (current->position),
GST_TIME_ARGS (last - current->start),
GST_TIME_ARGS (current->amount));
if ((current->flush && current->position >= current->amount)
|| last >= end) {
GST_DEBUG_OBJECT (sink, "step ended, we need clipping");
step_end = TRUE;
if (segment->rate > 0.0) {
*rstart = end;
*cstart = gst_segment_to_position (segment, GST_FORMAT_TIME, end);
} else {
*rstop = end;
*cstop = gst_segment_to_position (segment, GST_FORMAT_TIME, end);
}
}
GST_DEBUG_OBJECT (sink,
"cstart %" GST_TIME_FORMAT ", rstart %" GST_TIME_FORMAT,
GST_TIME_ARGS (*cstart), GST_TIME_ARGS (*rstart));
GST_DEBUG_OBJECT (sink,
"cstop %" GST_TIME_FORMAT ", rstop %" GST_TIME_FORMAT,
GST_TIME_ARGS (*cstop), GST_TIME_ARGS (*rstop));
break;
}
case GST_FORMAT_BUFFERS:
GST_DEBUG_OBJECT (sink,
"got default step %" G_GUINT64_FORMAT "/%" G_GUINT64_FORMAT,
current->position, current->amount);
if (current->position < current->amount) {
current->position++;
} else {
step_end = TRUE;
}
break;
case GST_FORMAT_DEFAULT:
default:
GST_DEBUG_OBJECT (sink,
"got unknown step %" G_GUINT64_FORMAT "/%" G_GUINT64_FORMAT,
current->position, current->amount);
break;
}
return step_end;
}
/* 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, GstClockTime * rrnext,
gboolean * do_sync, gboolean * stepped, GstStepInfo * step,
gboolean * step_end)
{
GstBaseSinkClass *bclass;
GstClockTime start, stop; /* raw start/stop timestamps */
guint64 cstart, cstop; /* clipped raw timestamps */
guint64 rstart, rstop, rnext; /* clipped timestamps converted to running time */
GstClockTime sstart, sstop; /* clipped timestamps converted to stream time */
GstFormat format;
GstBaseSinkPrivate *priv;
GstSegment *segment;
gboolean eos;
priv = basesink->priv;
segment = &basesink->segment;
bclass = GST_BASE_SINK_GET_CLASS (basesink);
again:
/* start with nothing */
start = stop = GST_CLOCK_TIME_NONE;
eos = FALSE;
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 (segment->rate >= 0.0) {
sstart = sstop = priv->current_sstop;
if (!GST_CLOCK_TIME_IS_VALID (sstart)) {
/* we have not seen a buffer yet, use the segment values */
sstart = sstop = gst_segment_to_stream_time (segment,
segment->format, segment->stop);
}
} else {
sstart = sstop = priv->current_sstart;
if (!GST_CLOCK_TIME_IS_VALID (sstart)) {
/* we have not seen a buffer yet, use the segment values */
sstart = sstop = gst_segment_to_stream_time (segment,
segment->format, segment->start);
}
}
rstart = rstop = rnext = priv->eos_rtime;
*do_sync = rstart != -1;
GST_DEBUG_OBJECT (basesink, "sync times for EOS %" GST_TIME_FORMAT,
GST_TIME_ARGS (rstart));
/* if we are stepping, we end now */
*step_end = step->valid;
eos = TRUE;
goto eos_done;
}
case GST_EVENT_GAP:
{
GstClockTime timestamp, duration;
gst_event_parse_gap (event, &timestamp, &duration);
GST_DEBUG_OBJECT (basesink, "Got Gap time %" GST_TIME_FORMAT
" duration %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp), GST_TIME_ARGS (duration));
if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
start = timestamp;
if (GST_CLOCK_TIME_IS_VALID (duration))
stop = start + duration;
}
*do_sync = TRUE;
break;
}
default:
/* other events do not need syncing */
return FALSE;
}
} else {
/* else do buffer sync code */
GstBuffer *buffer = GST_BUFFER_CAST (obj);
/* just get the times to see if we need syncing, if the start returns -1 we
* don't sync. */
if (bclass->get_times)
bclass->get_times (basesink, buffer, &start, &stop);
if (!GST_CLOCK_TIME_IS_VALID (start)) {
/* we don't need to sync but we still want to get the timestamps for
* tracking the position */
gst_base_sink_default_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;
/* clip */
if (G_UNLIKELY (!gst_segment_clip (segment, format,
start, stop, &cstart, &cstop))) {
if (step->valid) {
GST_DEBUG_OBJECT (basesink, "step out of segment");
/* when we are stepping, pretend we're at the end of the segment */
if (segment->rate > 0.0) {
cstart = segment->stop;
cstop = segment->stop;
} else {
cstart = segment->start;
cstop = segment->start;
}
goto do_times;
}
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 (stop != GST_CLOCK_TIME_NONE && cstop != GST_CLOCK_TIME_NONE))
segment->position = cstop;
else
segment->position = cstart;
do_times:
rstart = gst_segment_to_running_time (segment, format, cstart);
rstop = gst_segment_to_running_time (segment, format, cstop);
if (GST_CLOCK_TIME_IS_VALID (stop))
rnext = rstop;
else
rnext = rstart;
if (G_UNLIKELY (step->valid)) {
if (!(*step_end = handle_stepping (basesink, segment, step, &cstart, &cstop,
&rstart, &rstop))) {
/* step is still busy, we discard data when we are flushing */
*stepped = step->flush;
GST_DEBUG_OBJECT (basesink, "stepping busy");
}
}
/* 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);
eos_done:
/* eos_done label only called when doing EOS, we also stop stepping then */
if (*step_end && step->flush) {
GST_DEBUG_OBJECT (basesink, "flushing step ended");
stop_stepping (basesink, segment, step, rstart, rstop, eos);
*step_end = FALSE;
/* re-determine running start times for adjusted segment
* (which has a flushed amount of running/accumulated time removed) */
if (!GST_IS_EVENT (obj)) {
GST_DEBUG_OBJECT (basesink, "refresh sync times");
goto again;
}
}
/* save times */
*rsstart = sstart;
*rsstop = sstop;
*rrstart = rstart;
*rrstop = rstop;
*rrnext = rnext;
/* buffers and EOS always need syncing and preroll */
return TRUE;
/* special cases */
out_of_segment:
{
/* we usually clip in the chain function already but stepping could cause
* the segment to be updated later. we return %FALSE so that we don't try
* to sync on it. */
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. This function does
* not adjust for the render delay. */
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 careful 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;
/* subtract the render delay again, which was included in the latency */
if (time > basesink->priv->render_delay)
time -= basesink->priv->render_delay;
else
time = 0;
return time;
}
/**
* gst_base_sink_wait_clock:
* @sink: the sink
* @time: the running_time to be reached
* @jitter: (out) (allow-none): the jitter to be filled with time diff, or %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 synchronisation 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 #GstBaseSinkClass.event() vmethod or when
* receiving a buffer in
* the #GstBaseSinkClass.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.
*
* Returns: #GstClockReturn
*/
GstClockReturn
gst_base_sink_wait_clock (GstBaseSink * sink, GstClockTime time,
GstClockTimeDiff * jitter)
{
GstClockReturn ret;
GstClock *clock;
GstClockTime base_time;
if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (time)))
goto invalid_time;
GST_OBJECT_LOCK (sink);
if (G_UNLIKELY (!sink->sync))
goto no_sync;
if (G_UNLIKELY ((clock = GST_ELEMENT_CLOCK (sink)) == NULL))
goto no_clock;
base_time = GST_ELEMENT_CAST (sink)->base_time;
GST_LOG_OBJECT (sink,
"time %" GST_TIME_FORMAT ", base_time %" GST_TIME_FORMAT,
GST_TIME_ARGS (time), GST_TIME_ARGS (base_time));
/* add base_time to running_time to get the time against the clock */
time += base_time;
/* Re-use existing clockid if available */
/* FIXME: Casting to GstClockEntry only works because the types
* are the same */
if (G_LIKELY (sink->priv->cached_clock_id != NULL
&& GST_CLOCK_ENTRY_CLOCK ((GstClockEntry *) sink->
priv->cached_clock_id) == clock)) {
if (!gst_clock_single_shot_id_reinit (clock, sink->priv->cached_clock_id,
time)) {
gst_clock_id_unref (sink->priv->cached_clock_id);
sink->priv->cached_clock_id = gst_clock_new_single_shot_id (clock, time);
}
} else {
if (sink->priv->cached_clock_id != NULL)
gst_clock_id_unref (sink->priv->cached_clock_id);
sink->priv->cached_clock_id = gst_clock_new_single_shot_id (clock, time);
}
GST_OBJECT_UNLOCK (sink);
/* 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. */
sink->clock_id = sink->priv->cached_clock_id;
/* release the preroll lock while waiting */
GST_BASE_SINK_PREROLL_UNLOCK (sink);
ret = gst_clock_id_wait (sink->priv->cached_clock_id, jitter);
GST_BASE_SINK_PREROLL_LOCK (sink);
sink->clock_id = NULL;
return ret;
/* no syncing needed */
invalid_time:
{
GST_DEBUG_OBJECT (sink, "time not valid, no sync needed");
return GST_CLOCK_BADTIME;
}
no_sync:
{
GST_DEBUG_OBJECT (sink, "sync disabled");
GST_OBJECT_UNLOCK (sink);
return GST_CLOCK_BADTIME;
}
no_clock:
{
GST_DEBUG_OBJECT (sink, "no clock, can't sync");
GST_OBJECT_UNLOCK (sink);
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_FLUSHING).
*
* This function should only be called with the PREROLL_LOCK held, like in the
* render function.
*
* 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_BASE_SINK_PREROLL_WAIT (sink);
sink->have_preroll = FALSE;
if (G_UNLIKELY (sink->flushing))
goto stopping;
if (G_UNLIKELY (sink->priv->step_unlock))
goto step_unlocked;
GST_DEBUG_OBJECT (sink, "continue after preroll");
return GST_FLOW_OK;
/* ERRORS */
stopping:
{
GST_DEBUG_OBJECT (sink, "preroll interrupted because of flush");
return GST_FLOW_FLUSHING;
}
step_unlocked:
{
sink->priv->step_unlock = FALSE;
GST_DEBUG_OBJECT (sink, "preroll interrupted because of step");
return GST_FLOW_STEP;
}
}
/**
* gst_base_sink_do_preroll:
* @sink: the sink
* @obj: (transfer none): the mini object that caused the preroll
*
* If the @sink spawns its own thread for pulling buffers from upstream it
* should call this method after it has pulled a buffer. If the element needed
* to preroll, this function will perform the preroll and will then block
* until the element state is changed.
*
* This function should be called with the PREROLL_LOCK held.
*
* 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_do_preroll (GstBaseSink * sink, GstMiniObject * obj)
{
GstFlowReturn ret;
while (G_UNLIKELY (sink->need_preroll)) {
GST_DEBUG_OBJECT (sink, "prerolling object %p", obj);
/* if it's a buffer, we need to call the preroll method */
if (sink->priv->call_preroll) {
GstBaseSinkClass *bclass;
GstBuffer *buf;
if (GST_IS_BUFFER_LIST (obj)) {
buf = gst_buffer_list_get (GST_BUFFER_LIST_CAST (obj), 0);
g_assert (NULL != buf);
} else if (GST_IS_BUFFER (obj)) {
buf = GST_BUFFER_CAST (obj);
/* For buffer lists do not set last buffer for now */
gst_base_sink_set_last_buffer (sink, buf);
} else
buf = NULL;
if (buf) {
GST_DEBUG_OBJECT (sink, "preroll buffer %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
bclass = GST_BASE_SINK_GET_CLASS (sink);
if (bclass->prepare)
if ((ret = bclass->prepare (sink, buf)) != GST_FLOW_OK)
goto prepare_canceled;
if (bclass->preroll)
if ((ret = bclass->preroll (sink, buf)) != GST_FLOW_OK)
goto preroll_canceled;
sink->priv->call_preroll = FALSE;
}
}
/* commit state */
if (G_LIKELY (sink->playing_async)) {
if (G_UNLIKELY (!gst_base_sink_commit_state (sink)))
goto stopping;
}
/* need to recheck here because the commit state could have
* made us not need the preroll anymore */
if (G_LIKELY (sink->need_preroll)) {
/* block until the state changes, or we get a flush, or something */
ret = gst_base_sink_wait_preroll (sink);
if ((ret != GST_FLOW_OK) && (ret != GST_FLOW_STEP))
goto preroll_failed;
}
}
return GST_FLOW_OK;
/* ERRORS */
prepare_canceled:
{
GST_DEBUG_OBJECT (sink, "prepare failed, abort state");
gst_element_abort_state (GST_ELEMENT_CAST (sink));
return ret;
}
preroll_canceled:
{
GST_DEBUG_OBJECT (sink, "preroll failed, abort state");
gst_element_abort_state (GST_ELEMENT_CAST (sink));
return ret;
}
stopping:
{
GST_DEBUG_OBJECT (sink, "stopping while commiting state");
return GST_FLOW_FLUSHING;
}
preroll_failed:
{
GST_DEBUG_OBJECT (sink, "preroll failed: %s", gst_flow_get_name (ret));
return ret;
}
}
/**
* gst_base_sink_wait:
* @sink: the sink
* @time: the running_time to be reached
* @jitter: (out) (allow-none): the jitter to be filled with time diff, or %NULL
*
* This function will wait for preroll to complete and will then block until @time
* is reached. It is usually called by subclasses that use their own internal
* synchronisation but want to let some synchronization (like 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 or when handling buffers in
* ::render).
*
* The @time argument should be the running_time of when the timeout should happen
* and will be adjusted with any latency and offset configured in the sink.
*
* Returns: #GstFlowReturn
*/
GstFlowReturn
gst_base_sink_wait (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 */
while (G_UNLIKELY (sink->need_preroll)) {
ret = gst_base_sink_wait_preroll (sink);
if ((ret != GST_FLOW_OK) && (ret != GST_FLOW_STEP))
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, ts_offset and render delay */
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_FLUSHING;
}
}
/* 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
* immediately try to re-preroll.
*
* Some objects do not need synchronisation (most events) and so this function
* immediately 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,
GstMiniObject * obj, gboolean * late, gboolean * step_end)
{
GstClockTimeDiff jitter = 0;
gboolean syncable;
GstClockReturn status = GST_CLOCK_OK;
GstClockTime rstart, rstop, rnext, sstart, sstop, stime;
gboolean do_sync;
GstBaseSinkPrivate *priv;
GstFlowReturn ret;
GstStepInfo *current, *pending;
gboolean stepped;
priv = basesink->priv;
do_step:
sstart = sstop = rstart = rstop = rnext = GST_CLOCK_TIME_NONE;
do_sync = TRUE;
stepped = FALSE;
priv->current_rstart = GST_CLOCK_TIME_NONE;
/* get stepping info */
current = &priv->current_step;
pending = &priv->pending_step;
/* get timing information for this object against the render segment */
syncable = gst_base_sink_get_sync_times (basesink, obj,
&sstart, &sstop, &rstart, &rstop, &rnext, &do_sync, &stepped, current,
step_end);
if (G_UNLIKELY (stepped))
goto step_skipped;
/* 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 = (GST_CLOCK_TIME_IS_VALID (rstop) ? rstop : rstart);
/* save sync time for eos when the previous object needed sync */
priv->eos_rtime = (do_sync ? rnext : GST_CLOCK_TIME_NONE);
/* calculate inter frame spacing */
if (G_UNLIKELY (priv->prev_rstart != -1 && priv->prev_rstart < rstart)) {
GstClockTime in_diff;
in_diff = rstart - priv->prev_rstart;
if (priv->avg_in_diff == -1)
priv->avg_in_diff = in_diff;
else
priv->avg_in_diff = UPDATE_RUNNING_AVG (priv->avg_in_diff, in_diff);
GST_LOG_OBJECT (basesink, "avg frame diff %" GST_TIME_FORMAT,
GST_TIME_ARGS (priv->avg_in_diff));
}
priv->prev_rstart = rstart;
if (G_UNLIKELY (priv->earliest_in_time != -1
&& rstart < priv->earliest_in_time))
goto qos_dropped;
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. */
ret = gst_base_sink_do_preroll (basesink, obj);
if (G_UNLIKELY (ret != GST_FLOW_OK))
goto preroll_failed;
/* update the segment with a pending step if the current one is invalid and we
* have a new pending one. We only accept new step updates after a preroll */
if (G_UNLIKELY (pending->valid && !current->valid)) {
start_stepping (basesink, &basesink->segment, pending, current);
goto do_step;
}
/* 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 = (GST_CLOCK_TIME_IS_VALID (sstop) ? sstop : sstart);
GST_OBJECT_UNLOCK (basesink);
if (!do_sync)
goto done;
/* adjust for latency */
stime = gst_base_sink_adjust_time (basesink, rstart);
/* adjust for rate control */
if (priv->rc_next == -1 || (stime != -1 && stime >= priv->rc_next)) {
GST_DEBUG_OBJECT (basesink, "reset rc_time to time %" GST_TIME_FORMAT,
GST_TIME_ARGS (stime));
priv->rc_time = stime;
priv->rc_accumulated = 0;
} else {
GST_DEBUG_OBJECT (basesink, "rate control next %" GST_TIME_FORMAT,
GST_TIME_ARGS (priv->rc_next));
stime = priv->rc_next;
}
/* 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 immediately 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, jitter %c%" GST_TIME_FORMAT,
status, (jitter < 0 ? '-' : ' '), GST_TIME_ARGS (ABS (jitter)));
/* 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");
priv->call_preroll = TRUE;
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, TRUE);
done:
return GST_FLOW_OK;
/* ERRORS */
step_skipped:
{
GST_DEBUG_OBJECT (basesink, "skipped stepped object %p", obj);
*late = TRUE;
return GST_FLOW_OK;
}
not_syncable:
{
GST_DEBUG_OBJECT (basesink, "non syncable object %p", obj);
return GST_FLOW_OK;
}
qos_dropped:
{
GST_DEBUG_OBJECT (basesink, "dropped because of QoS %p", obj);
*late = TRUE;
return GST_FLOW_OK;
}
flushing:
{
GST_DEBUG_OBJECT (basesink, "we are flushing");
return GST_FLOW_FLUSHING;
}
preroll_failed:
{
GST_DEBUG_OBJECT (basesink, "preroll failed");
*step_end = FALSE;
return ret;
}
}
static gboolean
gst_base_sink_send_qos (GstBaseSink * basesink, GstQOSType type,
gdouble proportion, GstClockTime time, GstClockTimeDiff diff)
{
GstEvent *event;
gboolean res;
/* generate Quality-of-Service event */
GST_CAT_DEBUG_OBJECT (GST_CAT_QOS, basesink,
"qos: type %d, proportion: %lf, diff %" G_GINT64_FORMAT ", timestamp %"
GST_TIME_FORMAT, type, proportion, diff, GST_TIME_ARGS (time));
event = gst_event_new_qos (type, 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 (priv->current_step.valid)
return;
/* if Quality-of-Service disabled, do nothing */
if (!g_atomic_int_get (&priv->qos_enabled) ||
!GST_CLOCK_TIME_IS_VALID (start))
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 (GST_CLOCK_TIME_IS_VALID (stop) && stop != start)
duration = stop - start;
else
duration = priv->avg_in_diff;
/* if we have the time when the last buffer left us, calculate
* processing time */
if (GST_CLOCK_TIME_IS_VALID (priv->last_left)) {
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
", stop %" 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 (stop),
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 (!GST_CLOCK_TIME_IS_VALID (priv->avg_duration))
priv->avg_duration = duration;
else
priv->avg_duration = UPDATE_RUNNING_AVG (priv->avg_duration, duration);
if (!GST_CLOCK_TIME_IS_VALID (priv->avg_pt))
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 = 1.0;
if (GST_CLOCK_TIME_IS_VALID (priv->last_left)) {
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) {
GstQOSType type;
GstClockTimeDiff diff;
/* 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;
}
if (priv->throttle_time > 0) {
diff = priv->throttle_time;
type = GST_QOS_TYPE_THROTTLE;
} else {
diff = priv->current_jitter;
if (diff <= 0)
type = GST_QOS_TYPE_OVERFLOW;
else
type = GST_QOS_TYPE_UNDERFLOW;
}
gst_base_sink_send_qos (sink, type, priv->avg_rate, priv->current_rstart,
diff);
}
/* 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_render_time = GST_CLOCK_TIME_NONE;
priv->prev_rstart = GST_CLOCK_TIME_NONE;
priv->earliest_in_time = GST_CLOCK_TIME_NONE;
priv->last_left = GST_CLOCK_TIME_NONE;
priv->avg_duration = GST_CLOCK_TIME_NONE;
priv->avg_pt = GST_CLOCK_TIME_NONE;
priv->avg_rate = -1.0;
priv->avg_render = GST_CLOCK_TIME_NONE;
priv->avg_in_diff = GST_CLOCK_TIME_NONE;
priv->rendered = 0;
priv->dropped = 0;
}
/* Checks if the object was scheduled too late.
*
* rstart/rstop contain the running_time 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 rstart, GstClockTime rstop,
GstClockReturn status, GstClockTimeDiff jitter, gboolean render)
{
gboolean late;
guint64 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->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 (!GST_CLOCK_TIME_IS_VALID (rstart)))
goto no_timestamp;
/* we can add a valid stop time */
if (GST_CLOCK_TIME_IS_VALID (rstop))
max_lateness += rstop;
else {
max_lateness += rstart;
/* no stop time, use avg frame diff */
if (priv->avg_in_diff != -1)
max_lateness += priv->avg_in_diff;
}
/* if the jitter bigger than duration and lateness we are too late */
if ((late = rstart + jitter > max_lateness)) {
GST_CAT_DEBUG_OBJECT (GST_CAT_PERFORMANCE, basesink,
"buffer is too late %" GST_TIME_FORMAT
" > %" GST_TIME_FORMAT, GST_TIME_ARGS (rstart + 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 (GST_CLOCK_TIME_IS_VALID (priv->last_render_time) &&
rstart - priv->last_render_time > GST_SECOND) {
late = FALSE;
GST_ELEMENT_WARNING (basesink, CORE, CLOCK,
(_("A lot of buffers are being dropped.")),
("There may be a timestamping problem, or this computer is too slow."));
GST_CAT_DEBUG_OBJECT (GST_CAT_PERFORMANCE, basesink,
"**emergency** last buffer at %" GST_TIME_FORMAT " > GST_SECOND",
GST_TIME_ARGS (priv->last_render_time));
}
}
done:
if (render && (!late || !GST_CLOCK_TIME_IS_VALID (priv->last_render_time))) {
priv->last_render_time = rstart;
/* the next allowed input timestamp */
if (priv->throttle_time > 0)
priv->earliest_in_time = rstart + priv->throttle_time;
}
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;
}
}
/* called before and after calling the render vmethod. It keeps track of how
* much time was spent in the render method and is used to check if we are
* flooded */
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 (!GST_CLOCK_TIME_IS_VALID (priv->avg_render))
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));
}
}
static void
gst_base_sink_update_start_time (GstBaseSink * basesink)
{
GstClock *clock;
GST_OBJECT_LOCK (basesink);
if ((clock = GST_ELEMENT_CLOCK (basesink))) {
GstClockTime now;
gst_object_ref (clock);
GST_OBJECT_UNLOCK (basesink);
/* calculate the time when we stopped */
now = gst_clock_get_time (clock);
gst_object_unref (clock);
GST_OBJECT_LOCK (basesink);
/* store the current running time */
if (GST_ELEMENT_START_TIME (basesink) != GST_CLOCK_TIME_NONE) {
if (now != GST_CLOCK_TIME_NONE)
GST_ELEMENT_START_TIME (basesink) =
now - GST_ELEMENT_CAST (basesink)->base_time;
else
GST_WARNING_OBJECT (basesink,
"Clock %s returned invalid time, can't calculate "
"running_time when going to the PAUSED state",
GST_OBJECT_NAME (clock));
}
GST_DEBUG_OBJECT (basesink,
"start_time=%" GST_TIME_FORMAT ", now=%" GST_TIME_FORMAT
", base_time %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_ELEMENT_START_TIME (basesink)),
GST_TIME_ARGS (now),
GST_TIME_ARGS (GST_ELEMENT_CAST (basesink)->base_time));
}
GST_OBJECT_UNLOCK (basesink);
}
static void
gst_base_sink_flush_start (GstBaseSink * basesink, GstPad * pad)
{
/* 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);
/* and we need to commit our state again on the next
* prerolled buffer */
basesink->playing_async = TRUE;
if (basesink->priv->async_enabled) {
gst_base_sink_update_start_time (basesink);
gst_element_lost_state (GST_ELEMENT_CAST (basesink));
} else {
/* start time reset in above case as well;
* arranges for a.o. proper position reporting when flushing in PAUSED */
gst_element_set_start_time (GST_ELEMENT_CAST (basesink), 0);
basesink->priv->have_latency = TRUE;
}
gst_base_sink_set_last_buffer (basesink, NULL);
GST_PAD_STREAM_UNLOCK (pad);
}
static void
gst_base_sink_flush_stop (GstBaseSink * basesink, GstPad * pad,
gboolean reset_time)
{
/* unset flushing so we can accept new data, this also flushes out any EOS
* event. */
gst_base_sink_set_flushing (basesink, pad, FALSE);
/* for position reporting */
GST_OBJECT_LOCK (basesink);
basesink->priv->current_sstart = GST_CLOCK_TIME_NONE;
basesink->priv->current_sstop = GST_CLOCK_TIME_NONE;
basesink->priv->eos_rtime = GST_CLOCK_TIME_NONE;
basesink->priv->call_preroll = TRUE;
basesink->priv->current_step.valid = FALSE;
basesink->priv->pending_step.valid = FALSE;
if (basesink->pad_mode == GST_PAD_MODE_PUSH) {
/* we need new segment info after the flush. */
basesink->have_newsegment = FALSE;
if (reset_time) {
gst_segment_init (&basesink->segment, GST_FORMAT_UNDEFINED);
GST_ELEMENT_START_TIME (basesink) = 0;
}
}
GST_OBJECT_UNLOCK (basesink);
if (reset_time) {