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gstreamer/libs/gst/base/gstbasesrc.c
Sebastian Dröge 2faa733276 Revert previous changes to the behaviour of GstPadTemplates, etc and the possiblity to call them in class_init as it ... 
Original commit message from CVS:
* gst/gstbin.c: (gst_bin_get_type), (gst_bin_base_init),
(gst_bin_class_init):
* gst/gstelement.c: (gst_element_base_class_init),
(gst_element_class_add_pad_template):
* gst/gstpadtemplate.c: (gst_pad_template_init):
* gst/gstpipeline.c: (gst_pipeline_get_type),
(gst_pipeline_base_init), (gst_pipeline_class_init):
* libs/gst/base/gstbasesink.c:
* libs/gst/base/gstbasesrc.c: (gst_base_src_get_type),
(gst_base_src_base_init), (gst_base_src_class_init):
* plugins/elements/gstcapsfilter.c: (gst_capsfilter_base_init),
(gst_capsfilter_class_init):
* plugins/elements/gstfakesink.c: (gst_fake_sink_base_init),
(gst_fake_sink_class_init):
* plugins/elements/gstfakesrc.c: (gst_fake_src_base_init),
(gst_fake_src_class_init):
* plugins/elements/gstfdsink.c: (gst_fd_sink_base_init),
(gst_fd_sink_class_init):
* plugins/elements/gstfdsrc.c: (gst_fd_src_base_init),
(gst_fd_src_class_init):
* plugins/elements/gstfilesink.c: (gst_file_sink_base_init),
(gst_file_sink_class_init):
* plugins/elements/gstfilesrc.c: (gst_file_src_base_init),
(gst_file_src_class_init):
* plugins/elements/gstidentity.c: (gst_identity_base_init),
(gst_identity_class_init):
* plugins/elements/gstmultiqueue.c: (gst_multi_queue_base_init),
(gst_multi_queue_class_init):
* plugins/elements/gstqueue.c: (gst_queue_base_init),
(gst_queue_class_init):
* plugins/elements/gsttee.c: (gst_tee_base_init),
(gst_tee_class_init):
* plugins/elements/gsttypefindelement.c:
(gst_type_find_element_base_init),
(gst_type_find_element_class_init):
* tests/check/gst/gstelement.c: (gst_element_suite):
Revert previous changes to the behaviour of GstPadTemplates, etc
and the possiblity to call them in class_init as it breaks too
many elements. Reopens bug #491501.
Should be applied again for 0.11, thus added a few FIXME 0.11 at
several places.
2008-02-05 14:15:15 +00:00

2697 lines
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/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2000,2005 Wim Taymans <wim@fluendo.com>
*
* gstbasesrc.c:
*
* 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:gstbasesrc
* @short_description: Base class for getrange based source elements
* @see_also: #GstPushSrc, #GstBaseTransform, #GstBaseSink
*
* This is a generice base class for source elements. The following
* types of sources are supported:
* <itemizedlist>
* <listitem><para>random access sources like files</para></listitem>
* <listitem><para>seekable sources</para></listitem>
* <listitem><para>live sources</para></listitem>
* </itemizedlist>
*
* <refsect2>
* <para>
* The source can be configured to operate in any #GstFormat with the
* gst_base_src_set_format() method. The currently set format determines
* the format of the internal #GstSegment and any #GST_EVENT_NEWSEGMENT
* events. The default format for #GstBaseSrc is #GST_FORMAT_BYTES.
* </para>
* <para>
* #GstBaseSrc always supports push mode scheduling. If the following
* conditions are met, it also supports pull mode scheduling:
* <itemizedlist>
* <listitem><para>The format is set to #GST_FORMAT_BYTES (default).</para>
* </listitem>
* <listitem><para>#GstBaseSrc::is_seekable returns %TRUE.</para>
* </listitem>
* </itemizedlist>
* </para>
* <para>
* Since 0.10.9, any #GstBaseSrc can enable pull based scheduling at any
* time by overriding #GstBaseSrc::check_get_range so that it returns %TRUE.
* </para>
* <para>
* If all the conditions are met for operating in pull mode, #GstBaseSrc is
* automatically seekable in push mode as well. The following conditions must
* be met to make the element seekable in push mode when the format is not
* #GST_FORMAT_BYTES:
* <itemizedlist>
* <listitem><para>
* #GstBaseSrc::is_seekable returns %TRUE.
* </para></listitem>
* <listitem><para>
* #GstBaseSrc::query can convert all supported seek formats to the
* internal format as set with gst_base_src_set_format().
* </para></listitem>
* <listitem><para>
* #GstBaseSrc::do_seek is implemented, performs the seek and returns %TRUE.
* </para></listitem>
* </itemizedlist>
* </para>
* <para>
* When the element does not meet the requirements to operate in pull mode,
* the offset and length in the #GstBaseSrc::create method should be ignored.
* It is recommended to subclass #GstPushSrc instead, in this situation. If the
* element can operate in pull mode but only with specific offsets and
* lengths, it is allowed to generate an error when the wrong values are passed
* to the #GstBaseSrc::create function.
* </para>
* <para>
* #GstBaseSrc has support for live sources. Live sources are sources that when
* paused discard data, such as audio or video capture devices. A typical live
* source also produces data at a fixed rate and thus provides a clock to publish
* this rate.
* Use gst_base_src_set_live() to activate the live source mode.
* </para>
* <para>
* A live source does not produce data in the PAUSED state. This means that the
* #GstBaseSrc::create method will not be called in PAUSED but only in PLAYING.
* To signal the pipeline that the element will not produce data, the return
* value from the READY to PAUSED state will be #GST_STATE_CHANGE_NO_PREROLL.
* </para>
* <para>
* A typical live source will timestamp the buffers it creates with the
* current running time of the pipeline. This is one reason why a live source
* can only produce data in the PLAYING state, when the clock is actually
* distributed and running.
* </para>
* <para>
* Live sources that synchronize and block on the clock (an audio source, for
* example) can since 0.10.12 use gst_base_src_wait_playing() when the ::create
* function was interrupted by a state change to PAUSED.
* </para>
* <para>
* The #GstBaseSrc::get_times method can be used to implement pseudo-live
* sources.
* It only makes sense to implement the ::get_times function if the source is
* a live source. The ::get_times function should return timestamps starting
* from 0, as if it were a non-live source. The base class will make sure that
* the timestamps are transformed into the current running_time.
* The base source will then wait for the calculated running_time before pushing
* out the buffer.
* </para>
* <para>
* For live sources, the base class will by default report a latency of 0.
* For pseudo live sources, the base class will by default measure the difference
* between the first buffer timestamp and the start time of get_times and will
* report this value as the latency.
* Subclasses should override the query function when this behaviour is not
* acceptable.
* </para>
* <para>
* There is only support in #GstBaseSrc for exactly one source pad, which
* should be named "src". A source implementation (subclass of #GstBaseSrc)
* should install a pad template in its class_init function, like so:
* </para>
* <para>
* <programlisting>
* static void
* my_element_class_init (GstMyElementClass *klass)
* {
* GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
* // srctemplate should be a #GstStaticPadTemplate with direction
* // #GST_PAD_SRC and name "src"
* gst_element_class_add_pad_template (gstelement_class,
* gst_static_pad_template_get (&amp;srctemplate));
* // see #GstElementDetails
* gst_element_class_set_details (gstelement_class, &amp;details);
* }
* </programlisting>
* </para>
* <title>Controlled shutdown of live sources in applications</title>
* <para>
* Applications that record from a live source may want to stop recording
* in a controlled way, so that the recording is stopped, but the data
* already in the pipeline is processed to the end (remember that many live
* sources would go on recording forever otherwise). For that to happen the
* application needs to make the source stop recording and send an EOS
* event down the pipeline. The application would then wait for an
* EOS message posted on the pipeline's bus to know when all data has
* been processed and the pipeline can safely be stopped.
* </para>
* <para>
* Since GStreamer 0.10.16 an application may send an EOS event to a source
* element to make it send an EOS event downstream. This can typically be done
* with the gst_element_send_event() function on the element or its parent bin.
* </para>
* <para>
* After the EOS has been sent to the element, the application should wait for
* an EOS message to be posted on the pipeline's bus. Once this EOS message is
* received, it may safely shut down the entire pipeline.
* </para>
* <para>
* The old behaviour for controlled shutdown introduced since GStreamer 0.10.3
* is still available but deprecated as it is dangerous and less flexible.
* </para>
* <para>
* Last reviewed on 2007-12-19 (0.10.16)
* </para>
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <stdlib.h>
#include <string.h>
#include "gstbasesrc.h"
#include "gsttypefindhelper.h"
#include <gst/gstmarshal.h>
#include <gst/gst-i18n-lib.h>
GST_DEBUG_CATEGORY_STATIC (gst_base_src_debug);
#define GST_CAT_DEFAULT gst_base_src_debug
#define GST_LIVE_GET_LOCK(elem) (GST_BASE_SRC_CAST(elem)->live_lock)
#define GST_LIVE_LOCK(elem) g_mutex_lock(GST_LIVE_GET_LOCK(elem))
#define GST_LIVE_TRYLOCK(elem) g_mutex_trylock(GST_LIVE_GET_LOCK(elem))
#define GST_LIVE_UNLOCK(elem) g_mutex_unlock(GST_LIVE_GET_LOCK(elem))
#define GST_LIVE_GET_COND(elem) (GST_BASE_SRC_CAST(elem)->live_cond)
#define GST_LIVE_WAIT(elem) g_cond_wait (GST_LIVE_GET_COND (elem), GST_LIVE_GET_LOCK (elem))
#define GST_LIVE_TIMED_WAIT(elem, timeval) g_cond_timed_wait (GST_LIVE_GET_COND (elem), GST_LIVE_GET_LOCK (elem),\
timeval)
#define GST_LIVE_SIGNAL(elem) g_cond_signal (GST_LIVE_GET_COND (elem));
#define GST_LIVE_BROADCAST(elem) g_cond_broadcast (GST_LIVE_GET_COND (elem));
/* BaseSrc signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
#define DEFAULT_BLOCKSIZE 4096
#define DEFAULT_NUM_BUFFERS -1
#define DEFAULT_TYPEFIND FALSE
#define DEFAULT_DO_TIMESTAMP FALSE
enum
{
PROP_0,
PROP_BLOCKSIZE,
PROP_NUM_BUFFERS,
PROP_TYPEFIND,
PROP_DO_TIMESTAMP
};
#define GST_BASE_SRC_GET_PRIVATE(obj) \
(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_BASE_SRC, GstBaseSrcPrivate))
struct _GstBaseSrcPrivate
{
gboolean last_sent_eos; /* last thing we did was send an EOS (we set this
* to avoid the sending of two EOS in some cases) */
gboolean discont;
gboolean flushing;
/* two segments to be sent in the streaming thread with STREAM_LOCK */
GstEvent *close_segment;
GstEvent *start_segment;
/* if EOS is pending */
gboolean pending_eos;
/* startup latency is the time it takes between going to PLAYING and producing
* the first BUFFER with running_time 0. This value is included in the latency
* reporting. */
GstClockTime latency;
/* timestamp offset, this is the offset add to the values of gst_times for
* pseudo live sources */
GstClockTimeDiff ts_offset;
gboolean do_timestamp;
};
static GstElementClass *parent_class = NULL;
static void gst_base_src_base_init (gpointer g_class);
static void gst_base_src_class_init (GstBaseSrcClass * klass);
static void gst_base_src_init (GstBaseSrc * src, gpointer g_class);
static void gst_base_src_finalize (GObject * object);
GType
gst_base_src_get_type (void)
{
static GType base_src_type = 0;
if (G_UNLIKELY (base_src_type == 0)) {
static const GTypeInfo base_src_info = {
sizeof (GstBaseSrcClass),
(GBaseInitFunc) gst_base_src_base_init,
NULL,
(GClassInitFunc) gst_base_src_class_init,
NULL,
NULL,
sizeof (GstBaseSrc),
0,
(GInstanceInitFunc) gst_base_src_init,
};
base_src_type = g_type_register_static (GST_TYPE_ELEMENT,
"GstBaseSrc", &base_src_info, G_TYPE_FLAG_ABSTRACT);
}
return base_src_type;
}
static GstCaps *gst_base_src_getcaps (GstPad * pad);
static gboolean gst_base_src_setcaps (GstPad * pad, GstCaps * caps);
static void gst_base_src_fixate (GstPad * pad, GstCaps * caps);
static gboolean gst_base_src_activate_push (GstPad * pad, gboolean active);
static gboolean gst_base_src_activate_pull (GstPad * pad, gboolean active);
static void gst_base_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_base_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static gboolean gst_base_src_event_handler (GstPad * pad, GstEvent * event);
static gboolean gst_base_src_send_event (GstElement * elem, GstEvent * event);
static gboolean gst_base_src_default_event (GstBaseSrc * src, GstEvent * event);
static const GstQueryType *gst_base_src_get_query_types (GstElement * element);
static gboolean gst_base_src_query (GstPad * pad, GstQuery * query);
static gboolean gst_base_src_default_negotiate (GstBaseSrc * basesrc);
static gboolean gst_base_src_default_do_seek (GstBaseSrc * src,
GstSegment * segment);
static gboolean gst_base_src_default_query (GstBaseSrc * src, GstQuery * query);
static gboolean gst_base_src_default_prepare_seek_segment (GstBaseSrc * src,
GstEvent * event, GstSegment * segment);
static gboolean gst_base_src_set_flushing (GstBaseSrc * basesrc,
gboolean flushing, gboolean live_play, gboolean unlock, gboolean * playing);
static gboolean gst_base_src_start (GstBaseSrc * basesrc);
static gboolean gst_base_src_stop (GstBaseSrc * basesrc);
static GstStateChangeReturn gst_base_src_change_state (GstElement * element,
GstStateChange transition);
static void gst_base_src_loop (GstPad * pad);
static gboolean gst_base_src_pad_check_get_range (GstPad * pad);
static gboolean gst_base_src_default_check_get_range (GstBaseSrc * bsrc);
static GstFlowReturn gst_base_src_pad_get_range (GstPad * pad, guint64 offset,
guint length, GstBuffer ** buf);
static GstFlowReturn gst_base_src_get_range (GstBaseSrc * src, guint64 offset,
guint length, GstBuffer ** buf);
static void
gst_base_src_base_init (gpointer g_class)
{
GST_DEBUG_CATEGORY_INIT (gst_base_src_debug, "basesrc", 0, "basesrc element");
}
static void
gst_base_src_class_init (GstBaseSrcClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = G_OBJECT_CLASS (klass);
gstelement_class = GST_ELEMENT_CLASS (klass);
g_type_class_add_private (klass, sizeof (GstBaseSrcPrivate));
parent_class = g_type_class_peek_parent (klass);
gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_base_src_finalize);
gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_base_src_set_property);
gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_base_src_get_property);
g_object_class_install_property (gobject_class, PROP_BLOCKSIZE,
g_param_spec_ulong ("blocksize", "Block size",
"Size in bytes to read per buffer (0 = default)", 0, G_MAXULONG,
DEFAULT_BLOCKSIZE, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, PROP_NUM_BUFFERS,
g_param_spec_int ("num-buffers", "num-buffers",
"Number of buffers to output before sending EOS", -1, G_MAXINT,
DEFAULT_NUM_BUFFERS, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, PROP_TYPEFIND,
g_param_spec_boolean ("typefind", "Typefind",
"Run typefind before negotiating", DEFAULT_TYPEFIND,
G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, PROP_DO_TIMESTAMP,
g_param_spec_boolean ("do-timestamp", "Do timestamp",
"Apply current stream time to buffers", DEFAULT_DO_TIMESTAMP,
G_PARAM_READWRITE));
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_base_src_change_state);
gstelement_class->send_event = GST_DEBUG_FUNCPTR (gst_base_src_send_event);
gstelement_class->get_query_types =
GST_DEBUG_FUNCPTR (gst_base_src_get_query_types);
klass->negotiate = GST_DEBUG_FUNCPTR (gst_base_src_default_negotiate);
klass->event = GST_DEBUG_FUNCPTR (gst_base_src_default_event);
klass->do_seek = GST_DEBUG_FUNCPTR (gst_base_src_default_do_seek);
klass->query = GST_DEBUG_FUNCPTR (gst_base_src_default_query);
klass->check_get_range =
GST_DEBUG_FUNCPTR (gst_base_src_default_check_get_range);
klass->prepare_seek_segment =
GST_DEBUG_FUNCPTR (gst_base_src_default_prepare_seek_segment);
}
static void
gst_base_src_init (GstBaseSrc * basesrc, gpointer g_class)
{
GstPad *pad;
GstPadTemplate *pad_template;
basesrc->priv = GST_BASE_SRC_GET_PRIVATE (basesrc);
basesrc->is_live = FALSE;
basesrc->live_lock = g_mutex_new ();
basesrc->live_cond = g_cond_new ();
basesrc->num_buffers = DEFAULT_NUM_BUFFERS;
basesrc->num_buffers_left = -1;
basesrc->can_activate_push = TRUE;
basesrc->pad_mode = GST_ACTIVATE_NONE;
pad_template =
gst_element_class_get_pad_template (GST_ELEMENT_CLASS (g_class), "src");
g_return_if_fail (pad_template != NULL);
GST_DEBUG_OBJECT (basesrc, "creating src pad");
pad = gst_pad_new_from_template (pad_template, "src");
GST_DEBUG_OBJECT (basesrc, "setting functions on src pad");
gst_pad_set_activatepush_function (pad,
GST_DEBUG_FUNCPTR (gst_base_src_activate_push));
gst_pad_set_activatepull_function (pad,
GST_DEBUG_FUNCPTR (gst_base_src_activate_pull));
gst_pad_set_event_function (pad,
GST_DEBUG_FUNCPTR (gst_base_src_event_handler));
gst_pad_set_query_function (pad, GST_DEBUG_FUNCPTR (gst_base_src_query));
gst_pad_set_checkgetrange_function (pad,
GST_DEBUG_FUNCPTR (gst_base_src_pad_check_get_range));
gst_pad_set_getrange_function (pad,
GST_DEBUG_FUNCPTR (gst_base_src_pad_get_range));
gst_pad_set_getcaps_function (pad, GST_DEBUG_FUNCPTR (gst_base_src_getcaps));
gst_pad_set_setcaps_function (pad, GST_DEBUG_FUNCPTR (gst_base_src_setcaps));
gst_pad_set_fixatecaps_function (pad,
GST_DEBUG_FUNCPTR (gst_base_src_fixate));
/* hold pointer to pad */
basesrc->srcpad = pad;
GST_DEBUG_OBJECT (basesrc, "adding src pad");
gst_element_add_pad (GST_ELEMENT (basesrc), pad);
basesrc->blocksize = DEFAULT_BLOCKSIZE;
basesrc->clock_id = NULL;
/* we operate in BYTES by default */
gst_base_src_set_format (basesrc, GST_FORMAT_BYTES);
basesrc->data.ABI.typefind = DEFAULT_TYPEFIND;
basesrc->priv->do_timestamp = DEFAULT_DO_TIMESTAMP;
GST_OBJECT_FLAG_UNSET (basesrc, GST_BASE_SRC_STARTED);
GST_DEBUG_OBJECT (basesrc, "init done");
}
static void
gst_base_src_finalize (GObject * object)
{
GstBaseSrc *basesrc;
GstEvent **event_p;
basesrc = GST_BASE_SRC (object);
g_mutex_free (basesrc->live_lock);
g_cond_free (basesrc->live_cond);
event_p = &basesrc->data.ABI.pending_seek;
gst_event_replace (event_p, NULL);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
/**
* gst_base_src_wait_playing:
* @src: the src
*
* If the #GstBaseSrcClass::create 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 produce 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: 0.10.12
*
* Returns: #GST_FLOW_OK if @src is PLAYING and processing can
* continue. Any other return value should be returned from the create vmethod.
*/
GstFlowReturn
gst_base_src_wait_playing (GstBaseSrc * src)
{
/* block until the state changes, or we get a flush, or something */
GST_DEBUG_OBJECT (src, "live source waiting for running state");
GST_LIVE_WAIT (src);
if (src->priv->flushing)
goto flushing;
GST_DEBUG_OBJECT (src, "live source unlocked");
return GST_FLOW_OK;
/* ERRORS */
flushing:
{
GST_DEBUG_OBJECT (src, "we are flushing");
return GST_FLOW_WRONG_STATE;
}
}
/**
* gst_base_src_set_live:
* @src: base source instance
* @live: new live-mode
*
* If the element listens to a live source, @live should
* be set to %TRUE.
*
* A live source will not produce data in the PAUSED state and
* will therefore not be able to participate in the PREROLL phase
* of a pipeline. To signal this fact to the application and the
* pipeline, the state change return value of the live source will
* be GST_STATE_CHANGE_NO_PREROLL.
*/
void
gst_base_src_set_live (GstBaseSrc * src, gboolean live)
{
GST_OBJECT_LOCK (src);
src->is_live = live;
GST_OBJECT_UNLOCK (src);
}
/**
* gst_base_src_is_live:
* @src: base source instance
*
* Check if an element is in live mode.
*
* Returns: %TRUE if element is in live mode.
*/
gboolean
gst_base_src_is_live (GstBaseSrc * src)
{
gboolean result;
GST_OBJECT_LOCK (src);
result = src->is_live;
GST_OBJECT_UNLOCK (src);
return result;
}
/**
* gst_base_src_set_format:
* @src: base source instance
* @format: the format to use
*
* Sets the default format of the source. This will be the format used
* for sending NEW_SEGMENT events and for performing seeks.
*
* If a format of GST_FORMAT_BYTES is set, the element will be able to
* operate in pull mode if the #GstBaseSrc::is_seekable returns TRUE.
*
* Since: 0.10.1
*/
void
gst_base_src_set_format (GstBaseSrc * src, GstFormat format)
{
gst_segment_init (&src->segment, format);
}
/**
* gst_base_src_query_latency:
* @src: the source
* @live: if the source is live
* @min_latency: the min latency of the source
* @max_latency: the max latency of the source
*
* Query the source for the latency parameters. @live will be TRUE when @src is
* configured as a live source. @min_latency will be set to the difference
* between the running time and the timestamp of the first buffer.
* @max_latency is always the undefined value of -1.
*
* This function is mostly used by subclasses.
*
* Returns: TRUE if the query succeeded.
*
* Since: 0.10.13
*/
gboolean
gst_base_src_query_latency (GstBaseSrc * src, gboolean * live,
GstClockTime * min_latency, GstClockTime * max_latency)
{
GstClockTime min;
GST_OBJECT_LOCK (src);
if (live)
*live = src->is_live;
/* if we have a startup latency, report this one, else report 0. Subclasses
* are supposed to override the query function if they want something
* else. */
if (src->priv->latency != -1)
min = src->priv->latency;
else
min = 0;
if (min_latency)
*min_latency = min;
if (max_latency)
*max_latency = -1;
GST_LOG_OBJECT (src, "latency: live %d, min %" GST_TIME_FORMAT
", max %" GST_TIME_FORMAT, src->is_live, GST_TIME_ARGS (min),
GST_TIME_ARGS (-1));
GST_OBJECT_UNLOCK (src);
return TRUE;
}
/**
* gst_base_src_set_do_timestamp:
* @src: the source
* @timestamp: enable or disable timestamping
*
* Configure @src to automatically timestamp outgoing buffers based on the
* current running_time of the pipeline. This property is mostly useful for live
* sources.
*
* Since: 0.10.15
*/
void
gst_base_src_set_do_timestamp (GstBaseSrc * src, gboolean timestamp)
{
GST_OBJECT_LOCK (src);
src->priv->do_timestamp = timestamp;
GST_OBJECT_UNLOCK (src);
}
/**
* gst_base_src_get_do_timestamp:
* @src: the source
*
* Query if @src timestamps outgoing buffers based on the current running_time.
*
* Returns: %TRUE if the base class will automatically timestamp outgoing buffers.
*
* Since: 0.10.15
*/
gboolean
gst_base_src_get_do_timestamp (GstBaseSrc * src)
{
gboolean res;
GST_OBJECT_LOCK (src);
res = src->priv->do_timestamp;
GST_OBJECT_UNLOCK (src);
return res;
}
static gboolean
gst_base_src_setcaps (GstPad * pad, GstCaps * caps)
{
GstBaseSrcClass *bclass;
GstBaseSrc *bsrc;
gboolean res = TRUE;
bsrc = GST_BASE_SRC (GST_PAD_PARENT (pad));
bclass = GST_BASE_SRC_GET_CLASS (bsrc);
if (bclass->set_caps)
res = bclass->set_caps (bsrc, caps);
return res;
}
static GstCaps *
gst_base_src_getcaps (GstPad * pad)
{
GstBaseSrcClass *bclass;
GstBaseSrc *bsrc;
GstCaps *caps = NULL;
bsrc = GST_BASE_SRC (GST_PAD_PARENT (pad));
bclass = GST_BASE_SRC_GET_CLASS (bsrc);
if (bclass->get_caps)
caps = bclass->get_caps (bsrc);
if (caps == NULL) {
GstPadTemplate *pad_template;
pad_template =
gst_element_class_get_pad_template (GST_ELEMENT_CLASS (bclass), "src");
if (pad_template != NULL) {
caps = gst_caps_ref (gst_pad_template_get_caps (pad_template));
}
}
return caps;
}
static void
gst_base_src_fixate (GstPad * pad, GstCaps * caps)
{
GstBaseSrcClass *bclass;
GstBaseSrc *bsrc;
bsrc = GST_BASE_SRC (gst_pad_get_parent (pad));
bclass = GST_BASE_SRC_GET_CLASS (bsrc);
if (bclass->fixate)
bclass->fixate (bsrc, caps);
gst_object_unref (bsrc);
}
static gboolean
gst_base_src_default_query (GstBaseSrc * src, GstQuery * query)
{
gboolean res;
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_POSITION:
{
GstFormat format;
gst_query_parse_position (query, &format, NULL);
switch (format) {
case GST_FORMAT_PERCENT:
{
gint64 percent;
gint64 position;
gint64 duration;
position = src->segment.last_stop;
duration = src->segment.duration;
if (position != -1 && duration != -1) {
if (position < duration)
percent = gst_util_uint64_scale (GST_FORMAT_PERCENT_MAX, position,
duration);
else
percent = GST_FORMAT_PERCENT_MAX;
} else
percent = -1;
gst_query_set_position (query, GST_FORMAT_PERCENT, percent);
res = TRUE;
break;
}
default:
{
gint64 position;
position = src->segment.last_stop;
if (position != -1) {
/* convert to requested format */
res =
gst_pad_query_convert (src->srcpad, src->segment.format,
position, &format, &position);
} else
res = TRUE;
gst_query_set_position (query, format, position);
break;
}
}
break;
}
case GST_QUERY_DURATION:
{
GstFormat format;
gst_query_parse_duration (query, &format, NULL);
GST_DEBUG_OBJECT (src, "duration query in format %s",
gst_format_get_name (format));
switch (format) {
case GST_FORMAT_PERCENT:
gst_query_set_duration (query, GST_FORMAT_PERCENT,
GST_FORMAT_PERCENT_MAX);
res = TRUE;
break;
default:
{
gint64 duration;
/* this is the duration as configured by the subclass. */
duration = src->segment.duration;
if (duration != -1) {
/* convert to requested format, if this fails, we have a duration
* but we cannot answer the query, we must return FALSE. */
res =
gst_pad_query_convert (src->srcpad, src->segment.format,
duration, &format, &duration);
} else {
/* The subclass did not configure a duration, we assume that the
* media has an unknown duration then and we return TRUE to report
* this. Note that this is not the same as returning FALSE, which
* means that we cannot report the duration at all. */
res = TRUE;
}
gst_query_set_duration (query, format, duration);
break;
}
}
break;
}
case GST_QUERY_SEEKING:
{
gst_query_set_seeking (query, src->segment.format,
src->seekable, 0, src->segment.duration);
res = TRUE;
break;
}
case GST_QUERY_SEGMENT:
{
gint64 start, stop;
/* no end segment configured, current duration then */
if ((stop = src->segment.stop) == -1)
stop = src->segment.duration;
start = src->segment.start;
/* adjust to stream time */
if (src->segment.time != -1) {
start -= src->segment.time;
if (stop != -1)
stop -= src->segment.time;
}
gst_query_set_segment (query, src->segment.rate, src->segment.format,
start, stop);
res = TRUE;
break;
}
case GST_QUERY_FORMATS:
{
gst_query_set_formats (query, 3, GST_FORMAT_DEFAULT,
GST_FORMAT_BYTES, GST_FORMAT_PERCENT);
res = TRUE;
break;
}
case GST_QUERY_CONVERT:
{
GstFormat src_fmt, dest_fmt;
gint64 src_val, dest_val;
gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
/* we can only convert between equal formats... */
if (src_fmt == dest_fmt) {
dest_val = src_val;
res = TRUE;
} else
res = FALSE;
gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
break;
}
case GST_QUERY_LATENCY:
{
GstClockTime min, max;
gboolean live;
/* Subclasses should override and implement something usefull */
res = gst_base_src_query_latency (src, &live, &min, &max);
GST_LOG_OBJECT (src, "report latency: live %d, min %" GST_TIME_FORMAT
", max %" GST_TIME_FORMAT, live, GST_TIME_ARGS (min),
GST_TIME_ARGS (max));
gst_query_set_latency (query, live, min, max);
break;
}
case GST_QUERY_JITTER:
case GST_QUERY_RATE:
default:
res = FALSE;
break;
}
GST_DEBUG_OBJECT (src, "query %s returns %d", GST_QUERY_TYPE_NAME (query),
res);
return res;
}
static gboolean
gst_base_src_query (GstPad * pad, GstQuery * query)
{
GstBaseSrc *src;
GstBaseSrcClass *bclass;
gboolean result = FALSE;
src = GST_BASE_SRC (gst_pad_get_parent (pad));
bclass = GST_BASE_SRC_GET_CLASS (src);
if (bclass->query)
result = bclass->query (src, query);
else
result = gst_pad_query_default (pad, query);
gst_object_unref (src);
return result;
}
static gboolean
gst_base_src_default_do_seek (GstBaseSrc * src, GstSegment * segment)
{
gboolean res = TRUE;
/* update our offset if the start/stop position was updated */
if (segment->format == GST_FORMAT_BYTES) {
segment->time = segment->start;
} else if (segment->start == 0) {
/* seek to start, we can implement a default for this. */
segment->time = 0;
res = TRUE;
} else
res = FALSE;
return res;
}
static gboolean
gst_base_src_do_seek (GstBaseSrc * src, GstSegment * segment)
{
GstBaseSrcClass *bclass;
gboolean result = FALSE;
bclass = GST_BASE_SRC_GET_CLASS (src);
if (bclass->do_seek)
result = bclass->do_seek (src, segment);
return result;
}
#define SEEK_TYPE_IS_RELATIVE(t) (((t) != GST_SEEK_TYPE_NONE) && ((t) != GST_SEEK_TYPE_SET))
static gboolean
gst_base_src_default_prepare_seek_segment (GstBaseSrc * src, GstEvent * event,
GstSegment * segment)
{
/* By default, we try one of 2 things:
* - For absolute seek positions, convert the requested position to our
* configured processing format and place it in the output segment \
* - For relative seek positions, convert our current (input) values to the
* seek format, adjust by the relative seek offset and then convert back to
* the processing format
*/
GstSeekType cur_type, stop_type;
gint64 cur, stop;
GstSeekFlags flags;
GstFormat seek_format, dest_format;
gdouble rate;
gboolean update;
gboolean res = TRUE;
gst_event_parse_seek (event, &rate, &seek_format, &flags,
&cur_type, &cur, &stop_type, &stop);
dest_format = segment->format;
if (seek_format == dest_format) {
gst_segment_set_seek (segment, rate, seek_format, flags,
cur_type, cur, stop_type, stop, &update);
return TRUE;
}
if (cur_type != GST_SEEK_TYPE_NONE) {
/* FIXME: Handle seek_cur & seek_end by converting the input segment vals */
res =
gst_pad_query_convert (src->srcpad, seek_format, cur, &dest_format,
&cur);
cur_type = GST_SEEK_TYPE_SET;
}
if (res && stop_type != GST_SEEK_TYPE_NONE) {
/* FIXME: Handle seek_cur & seek_end by converting the input segment vals */
res =
gst_pad_query_convert (src->srcpad, seek_format, stop, &dest_format,
&stop);
stop_type = GST_SEEK_TYPE_SET;
}
/* And finally, configure our output segment in the desired format */
gst_segment_set_seek (segment, rate, dest_format, flags, cur_type, cur,
stop_type, stop, &update);
if (!res)
goto no_format;
return res;
no_format:
{
GST_DEBUG_OBJECT (src, "undefined format given, seek aborted.");
return FALSE;
}
}
static gboolean
gst_base_src_prepare_seek_segment (GstBaseSrc * src, GstEvent * event,
GstSegment * seeksegment)
{
GstBaseSrcClass *bclass;
gboolean result = FALSE;
bclass = GST_BASE_SRC_GET_CLASS (src);
if (bclass->prepare_seek_segment)
result = bclass->prepare_seek_segment (src, event, seeksegment);
return result;
}
/* this code implements the seeking. It is a good example
* handling all cases.
*
* A seek updates the currently configured segment.start
* and segment.stop values based on the SEEK_TYPE. If the
* segment.start value is updated, a seek to this new position
* should be performed.
*
* The seek can only be executed when we are not currently
* streaming any data, to make sure that this is the case, we
* acquire the STREAM_LOCK which is taken when we are in the
* _loop() function or when a getrange() is called. Normally
* we will not receive a seek if we are operating in pull mode
* though. When we operate as a live source we might block on the live
* cond, which does not release the STREAM_LOCK. Therefore we will try
* to grab the LIVE_LOCK instead of the STREAM_LOCK to make sure it is
* safe to perform the seek.
*
* When we are in the loop() function, we might be in the middle
* of pushing a buffer, which might block in a sink. To make sure
* that the push gets unblocked we push out a FLUSH_START event.
* Our loop function will get a WRONG_STATE return value from
* the push and will pause, effectively releasing the STREAM_LOCK.
*
* For a non-flushing seek, we pause the task, which might eventually
* release the STREAM_LOCK. We say eventually because when the sink
* blocks on the sample we might wait a very long time until the sink
* unblocks the sample. In any case we acquire the STREAM_LOCK and
* can continue the seek. A non-flushing seek is normally done in a
* running pipeline to perform seamless playback, this means that the sink is
* PLAYING and will return from its chain function.
* In the case of a non-flushing seek we need to make sure that the
* data we output after the seek is continuous with the previous data,
* this is because a non-flushing seek does not reset the running-time
* to 0. We do this by closing the currently running segment, ie. sending
* a new_segment event with the stop position set to the last processed
* position.
*
* After updating the segment.start/stop values, we prepare for
* streaming again. We push out a FLUSH_STOP to make the peer pad
* accept data again and we start our task again.
*
* A segment seek posts a message on the bus saying that the playback
* of the segment started. We store the segment flag internally because
* when we reach the segment.stop we have to post a segment.done
* instead of EOS when doing a segment seek.
*/
/* FIXME (0.11), we have the unlock gboolean here because most current
* implementations (fdsrc, -base/gst/tcp/, ...) unconditionally unlock, even when
* the streaming thread isn't running, resulting in bogus unlocks later when it
* starts. This is fixed by adding unlock_stop, but we should still avoid unlocking
* unnecessarily for backwards compatibility. Ergo, the unlock variable stays
* until 0.11
*/
static gboolean
gst_base_src_perform_seek (GstBaseSrc * src, GstEvent * event, gboolean unlock)
{
gboolean res = TRUE;
gdouble rate;
GstFormat seek_format, dest_format;
GstSeekFlags flags;
GstSeekType cur_type, stop_type;
gint64 cur, stop;
gboolean flush, playing;
gboolean update;
gboolean relative_seek = FALSE;
gboolean seekseg_configured = FALSE;
GstSegment seeksegment;
GST_DEBUG_OBJECT (src, "doing seek");
dest_format = src->segment.format;
if (event) {
gst_event_parse_seek (event, &rate, &seek_format, &flags,
&cur_type, &cur, &stop_type, &stop);
relative_seek = SEEK_TYPE_IS_RELATIVE (cur_type) ||
SEEK_TYPE_IS_RELATIVE (stop_type);
if (dest_format != seek_format && !relative_seek) {
/* If we have an ABSOLUTE position (SEEK_SET only), we can convert it
* here before taking the stream lock, otherwise we must convert it later,
* once we have the stream lock and can read the current position */
gst_segment_init (&seeksegment, dest_format);
if (!gst_base_src_prepare_seek_segment (src, event, &seeksegment))
goto prepare_failed;
seekseg_configured = TRUE;
}
flush = flags & GST_SEEK_FLAG_FLUSH;
} else {
flush = FALSE;
}
/* send flush start */
if (flush)
gst_pad_push_event (src->srcpad, gst_event_new_flush_start ());
else
gst_pad_pause_task (src->srcpad);
/* unblock streaming thread. */
gst_base_src_set_flushing (src, TRUE, FALSE, unlock, &playing);
/* grab streaming lock, this should eventually be possible, either
* because the task is paused, our streaming thread stopped
* or because our peer is flushing. */
GST_PAD_STREAM_LOCK (src->srcpad);
gst_base_src_set_flushing (src, FALSE, playing, unlock, NULL);
/* If we configured the seeksegment above, don't overwrite it now. Otherwise
* copy the current segment info into the temp segment that we can actually
* attempt the seek with. We only update the real segment if the seek suceeds. */
if (!seekseg_configured) {
memcpy (&seeksegment, &src->segment, sizeof (GstSegment));
/* now configure the final seek segment */
if (event) {
if (src->segment.format != seek_format) {
/* OK, here's where we give the subclass a chance to convert the relative
* seek into an absolute one in the processing format. We set up any
* absolute seek above, before taking the stream lock. */
if (!gst_base_src_prepare_seek_segment (src, event, &seeksegment)) {
GST_DEBUG_OBJECT (src, "Preparing the seek failed after flushing. "
"Aborting seek");
res = FALSE;
}
} else {
/* The seek format matches our processing format, no need to ask the
* the subclass to configure the segment. */
gst_segment_set_seek (&seeksegment, rate, seek_format, flags,
cur_type, cur, stop_type, stop, &update);
}
}
/* Else, no seek event passed, so we're just (re)starting the
current segment. */
}
if (res) {
GST_DEBUG_OBJECT (src, "segment configured from %" G_GINT64_FORMAT
" to %" G_GINT64_FORMAT ", position %" G_GINT64_FORMAT,
seeksegment.start, seeksegment.stop, seeksegment.last_stop);
/* do the seek, segment.last_stop contains the new position. */
res = gst_base_src_do_seek (src, &seeksegment);
}
/* and prepare to continue streaming */
if (flush) {
/* send flush stop, peer will accept data and events again. We
* are not yet providing data as we still have the STREAM_LOCK. */
gst_pad_push_event (src->srcpad, gst_event_new_flush_stop ());
} else if (res && src->data.ABI.running) {
/* we are running the current segment and doing a non-flushing seek,
* close the segment first based on the last_stop. */
GST_DEBUG_OBJECT (src, "closing running segment %" G_GINT64_FORMAT
" to %" G_GINT64_FORMAT, src->segment.start, src->segment.last_stop);
/* queue the segment for sending in the stream thread */
if (src->priv->close_segment)
gst_event_unref (src->priv->close_segment);
src->priv->close_segment =
gst_event_new_new_segment_full (TRUE,
src->segment.rate, src->segment.applied_rate, src->segment.format,
src->segment.start, src->segment.last_stop, src->segment.time);
}
/* The subclass must have converted the segment to the processing format
* by now */
if (res && seeksegment.format != dest_format) {
GST_DEBUG_OBJECT (src, "Subclass failed to prepare a seek segment "
"in the correct format. Aborting seek.");
res = FALSE;
}
/* if successfull seek, we update our real segment and push
* out the new segment. */
if (res) {
memcpy (&src->segment, &seeksegment, sizeof (GstSegment));
if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) {
gst_element_post_message (GST_ELEMENT (src),
gst_message_new_segment_start (GST_OBJECT (src),
src->segment.format, src->segment.last_stop));
}
/* for deriving a stop position for the playback segment from the seek
* segment, we must take the duration when the stop is not set */
if ((stop = src->segment.stop) == -1)
stop = src->segment.duration;
GST_DEBUG_OBJECT (src, "Sending newsegment from %" G_GINT64_FORMAT
" to %" G_GINT64_FORMAT, src->segment.start, stop);
/* now replace the old segment so that we send it in the stream thread the
* next time it is scheduled. */
if (src->priv->start_segment)
gst_event_unref (src->priv->start_segment);
if (src->segment.rate >= 0.0) {
/* forward, we send data from last_stop to stop */
src->priv->start_segment =
gst_event_new_new_segment_full (FALSE,
src->segment.rate, src->segment.applied_rate, src->segment.format,
src->segment.last_stop, stop, src->segment.time);
} else {
/* reverse, we send data from stop to last_stop */
src->priv->start_segment =
gst_event_new_new_segment_full (FALSE,
src->segment.rate, src->segment.applied_rate, src->segment.format,
src->segment.start, src->segment.last_stop, src->segment.time);
}
}
src->priv->discont = TRUE;
src->data.ABI.running = TRUE;
/* and restart the task in case it got paused explicitely or by
* the FLUSH_START event we pushed out. */
gst_pad_start_task (src->srcpad, (GstTaskFunction) gst_base_src_loop,
src->srcpad);
/* and release the lock again so we can continue streaming */
GST_PAD_STREAM_UNLOCK (src->srcpad);
return res;
/* ERROR */
prepare_failed:
GST_DEBUG_OBJECT (src, "Preparing the seek failed before flushing. "
"Aborting seek");
return FALSE;
}
static const GstQueryType *
gst_base_src_get_query_types (GstElement * element)
{
static const GstQueryType query_types[] = {
GST_QUERY_DURATION,
GST_QUERY_POSITION,
GST_QUERY_SEEKING,
GST_QUERY_SEGMENT,
GST_QUERY_FORMATS,
GST_QUERY_LATENCY,
GST_QUERY_JITTER,
GST_QUERY_RATE,
GST_QUERY_CONVERT,
0
};
return query_types;
}
/* all events send to this element directly. This is mainly done from the
* application.
*/
static gboolean
gst_base_src_send_event (GstElement * element, GstEvent * event)
{
GstBaseSrc *src;
gboolean result = FALSE;
src = GST_BASE_SRC (element);
switch (GST_EVENT_TYPE (event)) {
/* bidirectional events */
case GST_EVENT_FLUSH_START:
case GST_EVENT_FLUSH_STOP:
/* sending random flushes downstream can break stuff,
* especially sync since all segment info will get flushed */
break;
/* downstream serialized events */
case GST_EVENT_EOS:
/* queue EOS and make sure the task or pull function
* performs the EOS actions. */
GST_LIVE_LOCK (src);
src->priv->pending_eos = TRUE;
GST_LIVE_UNLOCK (src);
result = TRUE;
break;
case GST_EVENT_NEWSEGMENT:
/* sending random NEWSEGMENT downstream can break sync. */
break;
case GST_EVENT_TAG:
/* sending tags could be useful, FIXME insert in dataflow */
break;
case GST_EVENT_BUFFERSIZE:
/* does not seem to make much sense currently */
break;
/* upstream events */
case GST_EVENT_QOS:
/* elements should override send_event and do something */
break;
case GST_EVENT_SEEK:
{
gboolean started;
GST_OBJECT_LOCK (src->srcpad);
if (GST_PAD_ACTIVATE_MODE (src->srcpad) == GST_ACTIVATE_PULL)
goto wrong_mode;
started = GST_PAD_ACTIVATE_MODE (src->srcpad) == GST_ACTIVATE_PUSH;
GST_OBJECT_UNLOCK (src->srcpad);
if (started) {
/* when we are running in push mode, we can execute the
* seek right now, we need to unlock. */
result = gst_base_src_perform_seek (src, event, TRUE);
} else {
GstEvent **event_p;
/* else we store the event and execute the seek when we
* get activated */
GST_OBJECT_LOCK (src);
event_p = &src->data.ABI.pending_seek;
gst_event_replace ((GstEvent **) event_p, event);
GST_OBJECT_UNLOCK (src);
/* assume the seek will work */
result = TRUE;
}
break;
}
case GST_EVENT_NAVIGATION:
/* could make sense for elements that do something with navigation events
* but then they would need to override the send_event function */
break;
case GST_EVENT_LATENCY:
/* does not seem to make sense currently */
break;
/* custom events */
case GST_EVENT_CUSTOM_UPSTREAM:
/* override send_event if you want this */
break;
case GST_EVENT_CUSTOM_DOWNSTREAM:
case GST_EVENT_CUSTOM_BOTH:
/* FIXME, insert event in the dataflow */
break;
case GST_EVENT_CUSTOM_DOWNSTREAM_OOB:
case GST_EVENT_CUSTOM_BOTH_OOB:
/* insert a random custom event into the pipeline */
GST_DEBUG_OBJECT (src, "pushing custom OOB event downstream");
result = gst_pad_push_event (src->srcpad, event);
/* we gave away the ref to the event in the push */
event = NULL;
break;
default:
break;
}
done:
/* if we still have a ref to the event, unref it now */
if (event)
gst_event_unref (event);
return result;
/* ERRORS */
wrong_mode:
{
GST_DEBUG_OBJECT (src, "cannot perform seek when operating in pull mode");
GST_OBJECT_UNLOCK (src->srcpad);
result = FALSE;
goto done;
}
}
static gboolean
gst_base_src_default_event (GstBaseSrc * src, GstEvent * event)
{
gboolean result;
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEEK:
/* is normally called when in push mode */
if (!src->seekable)
goto not_seekable;
result = gst_base_src_perform_seek (src, event, TRUE);
break;
case GST_EVENT_FLUSH_START:
/* cancel any blocking getrange, is normally called
* when in pull mode. */
result = gst_base_src_set_flushing (src, TRUE, FALSE, TRUE, NULL);
break;
case GST_EVENT_FLUSH_STOP:
result = gst_base_src_set_flushing (src, FALSE, TRUE, TRUE, NULL);
break;
default:
result = TRUE;
break;
}
return result;
/* ERRORS */
not_seekable:
{
GST_DEBUG_OBJECT (src, "is not seekable");
return FALSE;
}
}
static gboolean
gst_base_src_event_handler (GstPad * pad, GstEvent * event)
{
GstBaseSrc *src;
GstBaseSrcClass *bclass;
gboolean result = FALSE;
src = GST_BASE_SRC (gst_pad_get_parent (pad));
bclass = GST_BASE_SRC_GET_CLASS (src);
if (bclass->event) {
if (!(result = bclass->event (src, event)))
goto subclass_failed;
}
done:
gst_event_unref (event);
gst_object_unref (src);
return result;
/* ERRORS */
subclass_failed:
{
GST_DEBUG_OBJECT (src, "subclass refused event");
goto done;
}
}
static void
gst_base_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstBaseSrc *src;
src = GST_BASE_SRC (object);
switch (prop_id) {
case PROP_BLOCKSIZE:
src->blocksize = g_value_get_ulong (value);
break;
case PROP_NUM_BUFFERS:
src->num_buffers = g_value_get_int (value);
break;
case PROP_TYPEFIND:
src->data.ABI.typefind = g_value_get_boolean (value);
break;
case PROP_DO_TIMESTAMP:
src->priv->do_timestamp = g_value_get_boolean (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_base_src_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstBaseSrc *src;
src = GST_BASE_SRC (object);
switch (prop_id) {
case PROP_BLOCKSIZE:
g_value_set_ulong (value, src->blocksize);
break;
case PROP_NUM_BUFFERS:
g_value_set_int (value, src->num_buffers);
break;
case PROP_TYPEFIND:
g_value_set_boolean (value, src->data.ABI.typefind);
break;
case PROP_DO_TIMESTAMP:
g_value_set_boolean (value, src->priv->do_timestamp);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
/* with STREAM_LOCK and LOCK */
static GstClockReturn
gst_base_src_wait (GstBaseSrc * basesrc, GstClock * clock, GstClockTime time)
{
GstClockReturn ret;
GstClockID id;
id = gst_clock_new_single_shot_id (clock, time);
basesrc->clock_id = id;
/* release the live lock while waiting */
GST_LIVE_UNLOCK (basesrc);
ret = gst_clock_id_wait (id, NULL);
GST_LIVE_LOCK (basesrc);
gst_clock_id_unref (id);
basesrc->clock_id = NULL;
return ret;
}
/* perform synchronisation on a buffer.
* with STREAM_LOCK.
*/
static GstClockReturn
gst_base_src_do_sync (GstBaseSrc * basesrc, GstBuffer * buffer)
{
GstClockReturn result;
GstClockTime start, end;
GstBaseSrcClass *bclass;
GstClockTime base_time;
GstClock *clock;
GstClockTime now = GST_CLOCK_TIME_NONE, timestamp;
gboolean do_timestamp, first, pseudo_live;
bclass = GST_BASE_SRC_GET_CLASS (basesrc);
start = end = -1;
if (bclass->get_times)
bclass->get_times (basesrc, buffer, &start, &end);
/* get buffer timestamp */
timestamp = GST_BUFFER_TIMESTAMP (buffer);
/* grab the lock to prepare for clocking and calculate the startup
* latency. */
GST_OBJECT_LOCK (basesrc);
/* if we are asked to sync against the clock we are a pseudo live element */
pseudo_live = (start != -1 && basesrc->is_live);
/* check for the first buffer */
first = (basesrc->priv->latency == -1);
if (timestamp != -1 && pseudo_live) {
GstClockTime latency;
/* we have a timestamp and a sync time, latency is the diff */
if (timestamp <= start)
latency = start - timestamp;
else
latency = 0;
if (first) {
GST_DEBUG_OBJECT (basesrc, "pseudo_live with latency %" GST_TIME_FORMAT,
GST_TIME_ARGS (latency));
/* first time we calculate latency, just configure */
basesrc->priv->latency = latency;
} else {
if (basesrc->priv->latency != latency) {
/* we have a new latency, FIXME post latency message */
basesrc->priv->latency = latency;
GST_DEBUG_OBJECT (basesrc, "latency changed to %" GST_TIME_FORMAT,
GST_TIME_ARGS (latency));
}
}
} else if (first) {
GST_DEBUG_OBJECT (basesrc, "no latency needed, live %d, sync %d",
basesrc->is_live, start != -1);
basesrc->priv->latency = 0;
}
/* get clock, if no clock, we can't sync or do timestamps */
if ((clock = GST_ELEMENT_CLOCK (basesrc)) == NULL)
goto no_clock;
base_time = GST_ELEMENT_CAST (basesrc)->base_time;
do_timestamp = basesrc->priv->do_timestamp;
/* first buffer, calculate the timestamp offset */
if (first) {
GstClockTime running_time;
now = gst_clock_get_time (clock);
running_time = now - base_time;
GST_LOG_OBJECT (basesrc,
"startup timestamp: %" GST_TIME_FORMAT ", running_time %"
GST_TIME_FORMAT, GST_TIME_ARGS (timestamp),
GST_TIME_ARGS (running_time));
if (pseudo_live && timestamp != -1) {
/* live source and we need to sync, add startup latency to all timestamps
* to get the real running_time. Live sources should always timestamp
* according to the current running time. */
basesrc->priv->ts_offset = GST_CLOCK_DIFF (timestamp, running_time);
GST_LOG_OBJECT (basesrc, "live with sync, ts_offset %" GST_TIME_FORMAT,
GST_TIME_ARGS (basesrc->priv->ts_offset));
} else {
basesrc->priv->ts_offset = 0;
GST_LOG_OBJECT (basesrc, "no timestamp offset needed");
}
if (!GST_CLOCK_TIME_IS_VALID (timestamp)) {
if (do_timestamp)
timestamp = running_time;
else
timestamp = 0;
GST_BUFFER_TIMESTAMP (buffer) = timestamp;
GST_LOG_OBJECT (basesrc, "created timestamp: %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp));
}
/* add the timestamp offset we need for sync */
timestamp += basesrc->priv->ts_offset;
} else {
/* not the first buffer, the timestamp is the diff between the clock and
* base_time */
if (do_timestamp && !GST_CLOCK_TIME_IS_VALID (timestamp)) {
now = gst_clock_get_time (clock);
GST_BUFFER_TIMESTAMP (buffer) = now - base_time;
GST_LOG_OBJECT (basesrc, "created timestamp: %" GST_TIME_FORMAT,
GST_TIME_ARGS (now - base_time));
}
}
/* if we don't have a buffer timestamp, we don't sync */
if (!GST_CLOCK_TIME_IS_VALID (start))
goto no_sync;
if (basesrc->is_live && GST_CLOCK_TIME_IS_VALID (timestamp)) {
/* for pseudo live sources, add our ts_offset to the timestamp */
GST_BUFFER_TIMESTAMP (buffer) += basesrc->priv->ts_offset;
start += basesrc->priv->ts_offset;
}
GST_LOG_OBJECT (basesrc,
"waiting for clock, base time %" GST_TIME_FORMAT
", stream_start %" GST_TIME_FORMAT,
GST_TIME_ARGS (base_time), GST_TIME_ARGS (start));
GST_OBJECT_UNLOCK (basesrc);
result = gst_base_src_wait (basesrc, clock, start + base_time);
GST_LOG_OBJECT (basesrc, "clock entry done: %d", result);
return result;
/* special cases */
no_clock:
{
GST_DEBUG_OBJECT (basesrc, "we have no clock");
GST_OBJECT_UNLOCK (basesrc);
return GST_CLOCK_OK;
}
no_sync:
{
GST_DEBUG_OBJECT (basesrc, "no sync needed");
GST_OBJECT_UNLOCK (basesrc);
return GST_CLOCK_OK;
}
}
static gboolean
gst_base_src_update_length (GstBaseSrc * src, guint64 offset, guint * length)
{
guint64 size, maxsize;
GstBaseSrcClass *bclass;
bclass = GST_BASE_SRC_GET_CLASS (src);
/* only operate if we are working with bytes */
if (src->segment.format != GST_FORMAT_BYTES)
return TRUE;
/* get total file size */
size = (guint64) src->segment.duration;
/* the max amount of bytes to read is the total size or
* up to the segment.stop if present. */
if (src->segment.stop != -1)
maxsize = MIN (size, src->segment.stop);
else
maxsize = size;
GST_DEBUG_OBJECT (src,
"reading offset %" G_GUINT64_FORMAT ", length %u, size %" G_GINT64_FORMAT
", segment.stop %" G_GINT64_FORMAT ", maxsize %" G_GINT64_FORMAT, offset,
*length, size, src->segment.stop, maxsize);
/* check size if we have one */
if (maxsize != -1) {
/* if we run past the end, check if the file became bigger and
* retry. */
if (G_UNLIKELY (offset + *length >= maxsize)) {
/* see if length of the file changed */
if (bclass->get_size)
if (!bclass->get_size (src, &size))
size = -1;
gst_segment_set_duration (&src->segment, GST_FORMAT_BYTES, size);
/* make sure we don't exceed the configured segment stop
* if it was set */
if (src->segment.stop != -1)
maxsize = MIN (size, src->segment.stop);
else
maxsize = size;
/* if we are at or past the end, EOS */
if (G_UNLIKELY (offset >= maxsize))
goto unexpected_length;
/* else we can clip to the end */
if (G_UNLIKELY (offset + *length >= maxsize))
*length = maxsize - offset;
}
}
/* keep track of current position. segment is in bytes, we checked
* that above. */
gst_segment_set_last_stop (&src->segment, GST_FORMAT_BYTES, offset);
return TRUE;
/* ERRORS */
unexpected_length:
{
return FALSE;
}
}
/* must be called with LIVE_LOCK */
static GstFlowReturn
gst_base_src_get_range (GstBaseSrc * src, guint64 offset, guint length,
GstBuffer ** buf)
{
GstFlowReturn ret;
GstBaseSrcClass *bclass;
GstClockReturn status;
bclass = GST_BASE_SRC_GET_CLASS (src);
if (src->is_live) {
while (G_UNLIKELY (!src->live_running)) {
ret = gst_base_src_wait_playing (src);
if (ret != GST_FLOW_OK)
goto stopped;
}
}
if (G_UNLIKELY (!GST_OBJECT_FLAG_IS_SET (src, GST_BASE_SRC_STARTED)))
goto not_started;
if (G_UNLIKELY (!bclass->create))
goto no_function;
if (G_UNLIKELY (!gst_base_src_update_length (src, offset, &length)))
goto unexpected_length;
/* normally we don't count buffers */
if (G_UNLIKELY (src->num_buffers_left >= 0)) {
if (src->num_buffers_left == 0)
goto reached_num_buffers;
else
src->num_buffers_left--;
}
GST_DEBUG_OBJECT (src,
"calling create offset %" G_GUINT64_FORMAT " length %u, time %"
G_GINT64_FORMAT, offset, length, src->segment.time);
ret = bclass->create (src, offset, length, buf);
if (G_UNLIKELY (ret != GST_FLOW_OK))
goto not_ok;
/* no timestamp set and we are at offset 0, we can timestamp with 0 */
if (offset == 0 && src->segment.time == 0
&& GST_BUFFER_TIMESTAMP (*buf) == -1)
GST_BUFFER_TIMESTAMP (*buf) = 0;
/* now sync before pushing the buffer */
status = gst_base_src_do_sync (src, *buf);
/* waiting for the clock could have made us flushing */
if (G_UNLIKELY (src->priv->flushing))
goto flushing;
if (G_UNLIKELY (src->priv->pending_eos))
goto eos;
switch (status) {
case GST_CLOCK_EARLY:
/* the buffer is too late. We currently don't drop the buffer. */
GST_DEBUG_OBJECT (src, "buffer too late!, returning anyway");
break;
case GST_CLOCK_OK:
/* buffer synchronised properly */
GST_DEBUG_OBJECT (src, "buffer ok");
break;
case GST_CLOCK_UNSCHEDULED:
/* this case is triggered when we were waiting for the clock and
* it got unlocked because we did a state change. We return
* WRONG_STATE in this case to stop the dataflow also get rid of the
* produced buffer. */
GST_DEBUG_OBJECT (src,
"clock was unscheduled (%d), returning WRONG_STATE", status);
gst_buffer_unref (*buf);
*buf = NULL;
ret = GST_FLOW_WRONG_STATE;
break;
default:
/* all other result values are unexpected and errors */
GST_ELEMENT_ERROR (src, CORE, CLOCK,
(_("Internal clock error.")),
("clock returned unexpected return value %d", status));
gst_buffer_unref (*buf);
*buf = NULL;
ret = GST_FLOW_ERROR;
break;
}
return ret;
/* ERROR */
stopped:
{
GST_DEBUG_OBJECT (src, "wait_playing returned %d (%s)", ret,
gst_flow_get_name (ret));
return ret;
}
not_ok:
{
GST_DEBUG_OBJECT (src, "create returned %d (%s)", ret,
gst_flow_get_name (ret));
return ret;
}
not_started:
{
GST_DEBUG_OBJECT (src, "getrange but not started");
return GST_FLOW_WRONG_STATE;
}
no_function:
{
GST_DEBUG_OBJECT (src, "no create function");
return GST_FLOW_ERROR;
}
unexpected_length:
{
GST_DEBUG_OBJECT (src, "unexpected length %u (offset=%" G_GUINT64_FORMAT
", size=%" G_GINT64_FORMAT ")", length, offset, src->segment.duration);
return GST_FLOW_UNEXPECTED;
}
reached_num_buffers:
{
GST_DEBUG_OBJECT (src, "sent all buffers");
return GST_FLOW_UNEXPECTED;
}
flushing:
{
GST_DEBUG_OBJECT (src, "we are flushing");
gst_buffer_unref (*buf);
*buf = NULL;
return GST_FLOW_WRONG_STATE;
}
eos:
{
GST_DEBUG_OBJECT (src, "we are EOS");
gst_buffer_unref (*buf);
*buf = NULL;
return GST_FLOW_UNEXPECTED;
}
}
static GstFlowReturn
gst_base_src_pad_get_range (GstPad * pad, guint64 offset, guint length,
GstBuffer ** buf)
{
GstBaseSrc *src;
GstFlowReturn res;
src = GST_BASE_SRC (gst_pad_get_parent (pad));
GST_LIVE_LOCK (src);
if (G_UNLIKELY (src->priv->flushing))
goto flushing;
/* if we're EOS, return right away */
if (G_UNLIKELY (src->priv->pending_eos))
goto eos;
res = gst_base_src_get_range (src, offset, length, buf);
done:
GST_LIVE_UNLOCK (src);
gst_object_unref (src);
return res;
/* ERRORS */
flushing:
{
GST_DEBUG_OBJECT (src, "we are flushing");
res = GST_FLOW_WRONG_STATE;
goto done;
}
eos:
{
GST_DEBUG_OBJECT (src, "we are EOS");
res = GST_FLOW_UNEXPECTED;
goto done;
}
}
static gboolean
gst_base_src_default_check_get_range (GstBaseSrc * src)
{
gboolean res;
if (!GST_OBJECT_FLAG_IS_SET (src, GST_BASE_SRC_STARTED)) {
GST_LOG_OBJECT (src, "doing start/stop to check get_range support");
if (G_LIKELY (gst_base_src_start (src)))
gst_base_src_stop (src);
}
/* we can operate in getrange mode if the native format is bytes
* and we are seekable, this condition is set in the random_access
* flag and is set in the _start() method. */
res = src->random_access;
return res;
}
static gboolean
gst_base_src_check_get_range (GstBaseSrc * src)
{
GstBaseSrcClass *bclass;
gboolean res;
bclass = GST_BASE_SRC_GET_CLASS (src);
if (bclass->check_get_range == NULL)
goto no_function;
res = bclass->check_get_range (src);
GST_LOG_OBJECT (src, "%s() returned %d",
GST_DEBUG_FUNCPTR_NAME (bclass->check_get_range), (gint) res);
return res;
/* ERRORS */
no_function:
{
GST_WARNING_OBJECT (src, "no check_get_range function set");
return FALSE;
}
}
static gboolean
gst_base_src_pad_check_get_range (GstPad * pad)
{
GstBaseSrc *src;
gboolean res;
src = GST_BASE_SRC (GST_OBJECT_PARENT (pad));
res = gst_base_src_check_get_range (src);
return res;
}
static void
gst_base_src_loop (GstPad * pad)
{
GstBaseSrc *src;
GstBuffer *buf = NULL;
GstFlowReturn ret;
gint64 position;
gboolean eos;
gulong blocksize;
eos = FALSE;
src = GST_BASE_SRC (GST_OBJECT_PARENT (pad));
GST_LIVE_LOCK (src);
if (G_UNLIKELY (src->priv->flushing))
goto flushing;
/* if we're EOS, return right away */
if (G_UNLIKELY (src->priv->pending_eos))
goto eos;
src->priv->last_sent_eos = FALSE;
blocksize = src->blocksize;
/* if we operate in bytes, we can calculate an offset */
if (src->segment.format == GST_FORMAT_BYTES) {
position = src->segment.last_stop;
/* for negative rates, start with subtracting the blocksize */
if (src->segment.rate < 0.0) {
/* we cannot go below segment.start */
if (position > src->segment.start + blocksize)
position -= blocksize;
else {
/* last block, remainder up to segment.start */
blocksize = position - src->segment.start;
position = src->segment.start;
}
}
} else
position = -1;
ret = gst_base_src_get_range (src, position, blocksize, &buf);
if (G_UNLIKELY (ret != GST_FLOW_OK)) {
GST_INFO_OBJECT (src, "pausing after gst_base_src_get_range() = %s",
gst_flow_get_name (ret));
GST_LIVE_UNLOCK (src);
goto pause;
}
/* this should not happen */
if (G_UNLIKELY (buf == NULL))
goto null_buffer;
/* push events to close/start our segment before we push the buffer. */
if (G_UNLIKELY (src->priv->close_segment)) {
gst_pad_push_event (pad, src->priv->close_segment);
src->priv->close_segment = NULL;
}
if (G_UNLIKELY (src->priv->start_segment)) {
gst_pad_push_event (pad, src->priv->start_segment);
src->priv->start_segment = NULL;
}
/* figure out the new position */
switch (src->segment.format) {
case GST_FORMAT_BYTES:
{
guint bufsize = GST_BUFFER_SIZE (buf);
/* we subtracted above for negative rates */
if (src->segment.rate >= 0.0)
position += bufsize;
break;
}
case GST_FORMAT_TIME:
{
GstClockTime start, duration;
start = GST_BUFFER_TIMESTAMP (buf);
duration = GST_BUFFER_DURATION (buf);
if (GST_CLOCK_TIME_IS_VALID (start))
position = start;
else
position = src->segment.last_stop;
if (GST_CLOCK_TIME_IS_VALID (duration)) {
if (src->segment.rate >= 0.0)
position += duration;
else if (position > duration)
position -= duration;
else
position = 0;
}
break;
}
case GST_FORMAT_DEFAULT:
if (src->segment.rate >= 0.0)
position = GST_BUFFER_OFFSET_END (buf);
else
position = GST_BUFFER_OFFSET (buf);
break;
default:
position = -1;
break;
}
if (position != -1) {
if (src->segment.rate >= 0.0) {
/* positive rate, check if we reached the stop */
if (src->segment.stop != -1) {
if (position >= src->segment.stop) {
eos = TRUE;
position = src->segment.stop;
}
}
} else {
/* negative rate, check if we reached the start. start is always set to
* something different from -1 */
if (position <= src->segment.start) {
eos = TRUE;
position = src->segment.start;
}
/* when going reverse, all buffers are DISCONT */
src->priv->discont = TRUE;
}
gst_segment_set_last_stop (&src->segment, src->segment.format, position);
}
if (G_UNLIKELY (src->priv->discont)) {
buf = gst_buffer_make_metadata_writable (buf);
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
src->priv->discont = FALSE;
}
GST_LIVE_UNLOCK (src);
ret = gst_pad_push (pad, buf);
if (G_UNLIKELY (ret != GST_FLOW_OK)) {
GST_INFO_OBJECT (src, "pausing after gst_pad_push() = %s",
gst_flow_get_name (ret));
goto pause;
}
if (G_UNLIKELY (eos)) {
GST_INFO_OBJECT (src, "pausing after end of segment");
ret = GST_FLOW_UNEXPECTED;
goto pause;
}
done:
return;
/* special cases */
flushing:
{
GST_DEBUG_OBJECT (src, "we are flushing");
GST_LIVE_UNLOCK (src);
ret = GST_FLOW_WRONG_STATE;
goto pause;
}
eos:
{
GST_DEBUG_OBJECT (src, "we are EOS");
GST_LIVE_UNLOCK (src);
ret = GST_FLOW_UNEXPECTED;
goto pause;
}
pause:
{
const gchar *reason = gst_flow_get_name (ret);
GST_DEBUG_OBJECT (src, "pausing task, reason %s", reason);
src->data.ABI.running = FALSE;
gst_pad_pause_task (pad);
if (GST_FLOW_IS_FATAL (ret) || ret == GST_FLOW_NOT_LINKED) {
if (ret == GST_FLOW_UNEXPECTED) {
/* perform EOS logic */
if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) {
gst_element_post_message (GST_ELEMENT_CAST (src),
gst_message_new_segment_done (GST_OBJECT_CAST (src),
src->segment.format, src->segment.last_stop));
} else {
gst_pad_push_event (pad, gst_event_new_eos ());
src->priv->last_sent_eos = TRUE;
}
} else {
/* for fatal errors we post an error message, post the error
* first so the app knows about the error first. */
GST_ELEMENT_ERROR (src, STREAM, FAILED,
(_("Internal data flow error.")),
("streaming task paused, reason %s (%d)", reason, ret));
gst_pad_push_event (pad, gst_event_new_eos ());
src->priv->last_sent_eos = TRUE;
}
}
goto done;
}
null_buffer:
{
GST_ELEMENT_ERROR (src, STREAM, FAILED,
(_("Internal data flow error.")), ("element returned NULL buffer"));
GST_LIVE_UNLOCK (src);
/* we finished the segment on error */
ret = GST_FLOW_ERROR;
goto done;
}
}
/* default negotiation code.
*
* Take intersection between src and sink pads, take first
* caps and fixate.
*/
static gboolean
gst_base_src_default_negotiate (GstBaseSrc * basesrc)
{
GstCaps *thiscaps;
GstCaps *caps = NULL;
GstCaps *peercaps = NULL;
gboolean result = FALSE;
/* first see what is possible on our source pad */
thiscaps = gst_pad_get_caps (GST_BASE_SRC_PAD (basesrc));
GST_DEBUG_OBJECT (basesrc, "caps of src: %" GST_PTR_FORMAT, thiscaps);
/* nothing or anything is allowed, we're done */
if (thiscaps == NULL || gst_caps_is_any (thiscaps))
goto no_nego_needed;
/* get the peer caps */
peercaps = gst_pad_peer_get_caps (GST_BASE_SRC_PAD (basesrc));
GST_DEBUG_OBJECT (basesrc, "caps of peer: %" GST_PTR_FORMAT, peercaps);
if (peercaps) {
GstCaps *icaps;
/* get intersection */
icaps = gst_caps_intersect (thiscaps, peercaps);
GST_DEBUG_OBJECT (basesrc, "intersect: %" GST_PTR_FORMAT, icaps);
gst_caps_unref (thiscaps);
gst_caps_unref (peercaps);
if (icaps) {
/* take first (and best, since they are sorted) possibility */
caps = gst_caps_copy_nth (icaps, 0);
gst_caps_unref (icaps);
}
} else {
/* no peer, work with our own caps then */
caps = thiscaps;
}
if (caps) {
caps = gst_caps_make_writable (caps);
gst_caps_truncate (caps);
/* now fixate */
if (!gst_caps_is_empty (caps)) {
gst_pad_fixate_caps (GST_BASE_SRC_PAD (basesrc), caps);
GST_DEBUG_OBJECT (basesrc, "fixated to: %" GST_PTR_FORMAT, caps);
if (gst_caps_is_any (caps)) {
/* hmm, still anything, so element can do anything and
* nego is not needed */
result = TRUE;
} else if (gst_caps_is_fixed (caps)) {
/* yay, fixed caps, use those then */
gst_pad_set_caps (GST_BASE_SRC_PAD (basesrc), caps);
result = TRUE;
}
}
gst_caps_unref (caps);
}
return result;
no_nego_needed:
{
GST_DEBUG_OBJECT (basesrc, "no negotiation needed");
if (thiscaps)
gst_caps_unref (thiscaps);
return TRUE;
}
}
static gboolean
gst_base_src_negotiate (GstBaseSrc * basesrc)
{
GstBaseSrcClass *bclass;
gboolean result = TRUE;
bclass = GST_BASE_SRC_GET_CLASS (basesrc);
if (bclass->negotiate)
result = bclass->negotiate (basesrc);
return result;
}
static gboolean
gst_base_src_start (GstBaseSrc * basesrc)
{
GstBaseSrcClass *bclass;
gboolean result;
guint64 size;
if (GST_OBJECT_FLAG_IS_SET (basesrc, GST_BASE_SRC_STARTED))
return TRUE;
GST_DEBUG_OBJECT (basesrc, "starting source");
basesrc->num_buffers_left = basesrc->num_buffers;
gst_segment_init (&basesrc->segment, basesrc->segment.format);
basesrc->data.ABI.running = FALSE;
bclass = GST_BASE_SRC_GET_CLASS (basesrc);
if (bclass->start)
result = bclass->start (basesrc);
else
result = TRUE;
if (!result)
goto could_not_start;
GST_OBJECT_FLAG_SET (basesrc, GST_BASE_SRC_STARTED);
/* figure out the size */
if (basesrc->segment.format == GST_FORMAT_BYTES) {
if (bclass->get_size) {
if (!(result = bclass->get_size (basesrc, &size)))
size = -1;
} else {
result = FALSE;
size = -1;
}
GST_DEBUG_OBJECT (basesrc, "setting size %" G_GUINT64_FORMAT, size);
/* only update the size when operating in bytes, subclass is supposed
* to set duration in the start method for other formats */
gst_segment_set_duration (&basesrc->segment, GST_FORMAT_BYTES, size);
} else {
size = -1;
}
GST_DEBUG_OBJECT (basesrc,
"format: %d, have size: %d, size: %" G_GUINT64_FORMAT ", duration: %"
G_GINT64_FORMAT, basesrc->segment.format, result, size,
basesrc->segment.duration);
/* check if we can seek */
if (bclass->is_seekable)
basesrc->seekable = bclass->is_seekable (basesrc);
else
basesrc->seekable = FALSE;
GST_DEBUG_OBJECT (basesrc, "is seekable: %d", basesrc->seekable);
/* update for random access flag */
basesrc->random_access = basesrc->seekable &&
basesrc->segment.format == GST_FORMAT_BYTES;
GST_DEBUG_OBJECT (basesrc, "is random_access: %d", basesrc->random_access);
/* run typefind if we are random_access and the typefinding is enabled. */
if (basesrc->random_access && basesrc->data.ABI.typefind && size != -1) {
GstCaps *caps;
caps = gst_type_find_helper (basesrc->srcpad, size);
gst_pad_set_caps (basesrc->srcpad, caps);
gst_caps_unref (caps);
} else {
/* use class or default negotiate function */
if (!gst_base_src_negotiate (basesrc))
goto could_not_negotiate;
}
return TRUE;
/* ERROR */
could_not_start:
{
GST_DEBUG_OBJECT (basesrc, "could not start");
/* subclass is supposed to post a message. We don't have to call _stop. */
return FALSE;
}
could_not_negotiate:
{
GST_DEBUG_OBJECT (basesrc, "could not negotiate, stopping");
GST_ELEMENT_ERROR (basesrc, STREAM, FORMAT,
("Could not negotiate format"), ("Check your filtered caps, if any"));
/* we must call stop */
gst_base_src_stop (basesrc);
return FALSE;
}
}
static gboolean
gst_base_src_stop (GstBaseSrc * basesrc)
{
GstBaseSrcClass *bclass;
gboolean result = TRUE;
if (!GST_OBJECT_FLAG_IS_SET (basesrc, GST_BASE_SRC_STARTED))
return TRUE;
GST_DEBUG_OBJECT (basesrc, "stopping source");
bclass = GST_BASE_SRC_GET_CLASS (basesrc);
if (bclass->stop)
result = bclass->stop (basesrc);
if (result)
GST_OBJECT_FLAG_UNSET (basesrc, GST_BASE_SRC_STARTED);
return result;
}
/* start or stop flushing dataprocessing
*/
static gboolean
gst_base_src_set_flushing (GstBaseSrc * basesrc,
gboolean flushing, gboolean live_play, gboolean unlock, gboolean * playing)
{
GstBaseSrcClass *bclass;
bclass = GST_BASE_SRC_GET_CLASS (basesrc);
if (flushing && unlock) {
/* unlock any subclasses, we need to do this before grabbing the
* LIVE_LOCK since we hold this lock before going into ::create. We pass an
* unlock to the params because of backwards compat (see seek handler)*/
if (bclass->unlock)
bclass->unlock (basesrc);
}
/* the live lock is released when we are blocked, waiting for playing or
* when we sync to the clock. */
GST_LIVE_LOCK (basesrc);
if (playing)
*playing = basesrc->live_running;
basesrc->priv->flushing = flushing;
if (flushing) {
/* if we are locked in the live lock, signal it to make it flush */
basesrc->live_running = TRUE;
/* clear pending EOS if any */
basesrc->priv->pending_eos = FALSE;
/* step 1, now that we have the LIVE lock, clear our unlock request */
if (bclass->unlock_stop)
bclass->unlock_stop (basesrc);
/* step 2, unblock clock sync (if any) or any other blocking thing */
if (basesrc->clock_id)
gst_clock_id_unschedule (basesrc->clock_id);
} else {
/* signal the live source that it can start playing */
basesrc->live_running = live_play;
}
GST_LIVE_SIGNAL (basesrc);
GST_LIVE_UNLOCK (basesrc);
return TRUE;
}
/* the purpose of this function is to make sure that a live source blocks in the
* LIVE lock or leaves the LIVE lock and continues playing. */
static gboolean
gst_base_src_set_playing (GstBaseSrc * basesrc, gboolean live_play)
{
GstBaseSrcClass *bclass;
bclass = GST_BASE_SRC_GET_CLASS (basesrc);
/* unlock subclasses locked in ::create, we only do this when we stop playing. */
if (!live_play) {
GST_DEBUG_OBJECT (basesrc, "unlock");
if (bclass->unlock)
bclass->unlock (basesrc);
}
/* we are now able to grab the LIVE lock, when we get it, we can be
* waiting for PLAYING while blocked in the LIVE cond or we can be waiting
* for the clock. */
GST_LIVE_LOCK (basesrc);
GST_DEBUG_OBJECT (basesrc, "unschedule clock");
/* unblock clock sync (if any) */
if (basesrc->clock_id)
gst_clock_id_unschedule (basesrc->clock_id);
/* configure what to do when we get to the LIVE lock. */
GST_DEBUG_OBJECT (basesrc, "live running %d", live_play);
basesrc->live_running = live_play;
if (live_play) {
gboolean start;
/* clear our unlock request when going to PLAYING */
GST_DEBUG_OBJECT (basesrc, "unlock stop");
if (bclass->unlock_stop)
bclass->unlock_stop (basesrc);
/* for live sources we restart the timestamp correction */
basesrc->priv->latency = -1;
/* have to restart the task in case it stopped because of the unlock when
* we went to PAUSED. Only do this if we operating in push mode. */
GST_OBJECT_LOCK (basesrc->srcpad);
start = (GST_PAD_ACTIVATE_MODE (basesrc->srcpad) == GST_ACTIVATE_PUSH);
GST_OBJECT_UNLOCK (basesrc->srcpad);
if (start)
gst_pad_start_task (basesrc->srcpad, (GstTaskFunction) gst_base_src_loop,
basesrc->srcpad);
GST_DEBUG_OBJECT (basesrc, "signal");
GST_LIVE_SIGNAL (basesrc);
}
GST_LIVE_UNLOCK (basesrc);
return TRUE;
}
static gboolean
gst_base_src_activate_push (GstPad * pad, gboolean active)
{
GstBaseSrc *basesrc;
GstEvent *event;
basesrc = GST_BASE_SRC (GST_OBJECT_PARENT (pad));
/* prepare subclass first */
if (active) {
GST_DEBUG_OBJECT (basesrc, "Activating in push mode");
if (G_UNLIKELY (!basesrc->can_activate_push))
goto no_push_activation;
if (G_UNLIKELY (!gst_base_src_start (basesrc)))
goto error_start;
basesrc->priv->last_sent_eos = FALSE;
basesrc->priv->discont = TRUE;
gst_base_src_set_flushing (basesrc, FALSE, FALSE, FALSE, NULL);
/* do initial seek, which will start the task */
GST_OBJECT_LOCK (basesrc);
event = basesrc->data.ABI.pending_seek;
basesrc->data.ABI.pending_seek = NULL;
GST_OBJECT_UNLOCK (basesrc);
/* no need to unlock anything, the task is certainly
* not running here. The perform seek code will start the task when
* finished. */
if (G_UNLIKELY (!gst_base_src_perform_seek (basesrc, event, FALSE)))
goto seek_failed;
if (event)
gst_event_unref (event);
} else {
GST_DEBUG_OBJECT (basesrc, "Deactivating in push mode");
/* flush all */
gst_base_src_set_flushing (basesrc, TRUE, FALSE, TRUE, NULL);
/* stop the task */
gst_pad_stop_task (pad);
/* now we can stop the source */
if (G_UNLIKELY (!gst_base_src_stop (basesrc)))
goto error_stop;
}
return TRUE;
/* ERRORS */
no_push_activation:
{
GST_WARNING_OBJECT (basesrc, "Subclass disabled push-mode activation");
return FALSE;
}
error_start:
{
GST_WARNING_OBJECT (basesrc, "Failed to start in push mode");
return FALSE;
}
seek_failed:
{
GST_ERROR_OBJECT (basesrc, "Failed to perform initial seek");
gst_base_src_stop (basesrc);
if (event)
gst_event_unref (event);
return FALSE;
}
error_stop:
{
GST_DEBUG_OBJECT (basesrc, "Failed to stop in push mode");
return FALSE;
}
}
static gboolean
gst_base_src_activate_pull (GstPad * pad, gboolean active)
{
GstBaseSrc *basesrc;
basesrc = GST_BASE_SRC (GST_OBJECT_PARENT (pad));
/* prepare subclass first */
if (active) {
GST_DEBUG_OBJECT (basesrc, "Activating in pull mode");
if (G_UNLIKELY (!gst_base_src_start (basesrc)))
goto error_start;
/* if not random_access, we cannot operate in pull mode for now */
if (G_UNLIKELY (!gst_base_src_check_get_range (basesrc)))
goto no_get_range;
/* stop flushing now but for live sources, still block in the LIVE lock when
* we are not yet PLAYING */
gst_base_src_set_flushing (basesrc, FALSE, FALSE, FALSE, NULL);
} else {
GST_DEBUG_OBJECT (basesrc, "Deactivating in pull mode");
/* flush all, there is no task to stop */
gst_base_src_set_flushing (basesrc, TRUE, FALSE, TRUE, NULL);
/* don't send EOS when going from PAUSED => READY when in pull mode */
basesrc->priv->last_sent_eos = TRUE;
if (G_UNLIKELY (!gst_base_src_stop (basesrc)))
goto error_stop;
}
return TRUE;
/* ERRORS */
error_start:
{
GST_ERROR_OBJECT (basesrc, "Failed to start in pull mode");
return FALSE;
}
no_get_range:
{
GST_ERROR_OBJECT (basesrc, "Cannot operate in pull mode, stopping");
gst_base_src_stop (basesrc);
return FALSE;
}
error_stop:
{
GST_ERROR_OBJECT (basesrc, "Failed to stop in pull mode");
return FALSE;
}
}
static GstStateChangeReturn
gst_base_src_change_state (GstElement * element, GstStateChange transition)
{
GstBaseSrc *basesrc;
GstStateChangeReturn result;
gboolean no_preroll = FALSE;
basesrc = GST_BASE_SRC (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
no_preroll = gst_base_src_is_live (basesrc);
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
GST_DEBUG_OBJECT (basesrc, "PAUSED->PLAYING");
if (gst_base_src_is_live (basesrc)) {
/* now we can start playback */
gst_base_src_set_playing (basesrc, TRUE);
}
break;
default:
break;
}
if ((result =
GST_ELEMENT_CLASS (parent_class)->change_state (element,
transition)) == GST_STATE_CHANGE_FAILURE)
goto failure;
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
GST_DEBUG_OBJECT (basesrc, "PLAYING->PAUSED");
if (gst_base_src_is_live (basesrc)) {
/* make sure we block in the live lock in PAUSED */
gst_base_src_set_playing (basesrc, FALSE);
no_preroll = TRUE;
}
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
{
GstEvent **event_p;
/* we don't need to unblock anything here, the pad deactivation code
* already did this */
/* FIXME, deprecate this behaviour, it is very dangerous.
* the prefered way of sending EOS downstream is by sending
* the EOS event to the element */
if (!basesrc->priv->last_sent_eos) {
GST_DEBUG_OBJECT (basesrc, "Sending EOS event");
gst_pad_push_event (basesrc->srcpad, gst_event_new_eos ());
basesrc->priv->last_sent_eos = TRUE;
}
basesrc->priv->pending_eos = FALSE;
event_p = &basesrc->data.ABI.pending_seek;
gst_event_replace (event_p, NULL);
event_p = &basesrc->priv->close_segment;
gst_event_replace (event_p, NULL);
event_p = &basesrc->priv->start_segment;
gst_event_replace (event_p, NULL);
break;
}
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
if (no_preroll && result == GST_STATE_CHANGE_SUCCESS)
result = GST_STATE_CHANGE_NO_PREROLL;
return result;
/* ERRORS */
failure:
{
GST_DEBUG_OBJECT (basesrc, "parent failed state change");
return result;
}
}
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