* <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
+ * 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><para>#GstBaseSrcClass.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>
+ *
+ * Since 0.10.9, any #GstBaseSrc can enable pull based scheduling at any time
+ * by overriding #GstBaseSrcClass.check_get_range() so that it returns %TRUE.
+ *
* If all the conditions are met for operating in pull mode, #GstBaseSrc is
- * automatically seekable in push mode as well. The following conditions must
+ * 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.
+ * #GstBaseSrcClass.is_seekable() returns %TRUE.
* </para></listitem>
* <listitem><para>
- * #GstBaseSrc::query can convert all supported seek formats to the
+ * #GstBaseSrcClass.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.
+ * #GstBaseSrcClass.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.
+ *
+ * When the element does not meet the requirements to operate in pull mode, the
+ * offset and length in the #GstBaseSrcClass.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>
+ * to the #GstBaseSrcClass.create() function.
+ *
* #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
+ *
+ * A live source does not produce data in the PAUSED state. This means that the
+ * #GstBaseSrcClass.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.
+ *
+ * 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>
+ * can only produce data in the PLAYING state, when the clock is actually
+ * distributed and running.
+ *
* 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>
+ * example) can since 0.10.12 use gst_base_src_wait_playing() when the
+ * #GstBaseSrcClass.create() function was interrupted by a state change to
+ * PAUSED.
+ *
+ * The #GstBaseSrcClass.get_times() method can be used to implement pseudo-live
+ * sources. It only makes sense to implement the #GstBaseSrcClass.get_times()
+ * function if the source is a live source. The #GstBaseSrcClass.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.
+ *
* 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.
+ * 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 base_init function, like so:
- * </para>
- * <para>
- * <programlisting>
+ *
+ * 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:
+ * |[
* static void
- * my_element_base_init (gpointer g_class)
+ * my_element_class_init (GstMyElementClass *klass)
* {
- * GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class);
+ * 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,
* // see #GstElementDetails
* gst_element_class_set_details (gstelement_class, &details);
* }
- * </programlisting>
- * </para>
+ * ]|
+ *
+ * <refsect2>
* <title>Controlled shutdown of live sources in applications</title>
* <para>
* Applications that record from a live source may want to stop recording
* 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 0.10.3 an application may simply set the source
- * element to NULL or READY state to make it send an EOS event downstream.
- * The application should lock the state of the source afterwards, so that
- * shutting down the pipeline from PLAYING doesn't temporarily start up the
- * source element for a second time:
- * <programlisting>
- * ...
- * // stop recording
- * gst_element_set_state (audio_source, #GST_STATE_NULL);
- * gst_element_set_locked_state (audio_source, %TRUE);
- * ...
- * </programlisting>
- * Now the application should wait for an EOS message
- * to be posted on the pipeline's bus. Once it has received
- * an EOS message, it may safely shut down the entire pipeline:
- * <programlisting>
- * ...
- * // everything done - shut down pipeline
- * gst_element_set_state (pipeline, #GST_STATE_NULL);
- * gst_element_set_locked_state (audio_source, %FALSE);
- * ...
- * </programlisting>
- * </para>
- * <para>
- * Note that setting the source element to NULL or READY when the
- * pipeline is in the PAUSED state may cause a deadlock since the streaming
- * thread might be blocked in PREROLL.
- * </para>
- * <para>
- * Last reviewed on 2007-09-13 (0.10.15)
+ *
+ * Since GStreamer 0.10.16 an application may send an EOS event to a source
+ * element to make it perform the EOS logic (send EOS event downstream or post a
+ * #GST_MESSAGE_SEGMENT_DONE on the bus). This can typically be done
+ * with the gst_element_send_event() function on the element or its parent bin.
+ *
+ * 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.
+ *
+ * The old behaviour for controlled shutdown introduced since GStreamer 0.10.3
+ * is still available but deprecated as it is dangerous and less flexible.
+ *
+ * Last reviewed on 2007-12-19 (0.10.16)
* </para>
* </refsect2>
*/
#include <stdlib.h>
#include <string.h>
+#include <gst/gst_private.h>
+
#include "gstbasesrc.h"
#include "gsttypefindhelper.h"
#include <gst/gstmarshal.h>
#define DEFAULT_BLOCKSIZE 4096
#define DEFAULT_NUM_BUFFERS -1
-#define DEFAULT_TYPEFIND FALSE
-#define DEFAULT_DO_TIMESTAMP FALSE
+#define DEFAULT_TYPEFIND FALSE
+#define DEFAULT_DO_TIMESTAMP FALSE
enum
{
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;
+ gboolean newsegment_pending;
+
+ /* if EOS is pending (atomic) */
+ gint 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
GstClockTimeDiff ts_offset;
gboolean do_timestamp;
+ volatile gint dynamic_size;
+
+ /* stream sequence number */
+ guint32 seqnum;
+
+ /* pending events (TAG, CUSTOM_BOTH, CUSTOM_DOWNSTREAM) to be
+ * pushed in the data stream */
+ GList *pending_events;
+ volatile gint have_events;
+
+ /* QoS *//* with LOCK */
+ gboolean qos_enabled;
+ gdouble proportion;
+ GstClockTime earliest_time;
};
static GstElementClass *parent_class = NULL;
GType
gst_base_src_get_type (void)
{
- static GType base_src_type = 0;
+ static volatile gsize base_src_type = 0;
- if (G_UNLIKELY (base_src_type == 0)) {
+ if (g_once_init_enter (&base_src_type)) {
+ GType _type;
static const GTypeInfo base_src_info = {
sizeof (GstBaseSrcClass),
(GBaseInitFunc) gst_base_src_base_init,
(GInstanceInitFunc) gst_base_src_init,
};
- base_src_type = g_type_register_static (GST_TYPE_ELEMENT,
+ _type = g_type_register_static (GST_TYPE_ELEMENT,
"GstBaseSrc", &base_src_info, G_TYPE_FLAG_ABSTRACT);
+ g_once_init_leave (&base_src_type, _type);
}
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_default_prepare_seek_segment (GstBaseSrc * src,
GstEvent * event, GstSegment * segment);
-static gboolean gst_base_src_unlock (GstBaseSrc * basesrc);
-static gboolean gst_base_src_unlock_stop (GstBaseSrc * basesrc);
+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);
guint length, GstBuffer ** buf);
static GstFlowReturn gst_base_src_get_range (GstBaseSrc * src, guint64 offset,
guint length, GstBuffer ** buf);
+static gboolean gst_base_src_seekable (GstBaseSrc * src);
+static gboolean gst_base_src_update_length (GstBaseSrc * src, guint64 offset,
+ guint * length);
static void
gst_base_src_base_init (gpointer g_class)
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);
+ gobject_class->finalize = gst_base_src_finalize;
+ gobject_class->set_property = gst_base_src_set_property;
+ gobject_class->get_property = gst_base_src_get_property;
+/* FIXME 0.11: blocksize property should be int, not ulong (min is >max here) */
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));
+ "Size in bytes to read per buffer (-1 = default)", 0, G_MAXULONG,
+ DEFAULT_BLOCKSIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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));
+ "Number of buffers to output before sending EOS (-1 = unlimited)",
+ -1, G_MAXINT, DEFAULT_NUM_BUFFERS, G_PARAM_READWRITE |
+ G_PARAM_STATIC_STRINGS));
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_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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));
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_base_src_change_state);
GST_DEBUG_FUNCPTR (gst_base_src_default_check_get_range);
klass->prepare_seek_segment =
GST_DEBUG_FUNCPTR (gst_base_src_default_prepare_seek_segment);
+
+ /* Registering debug symbols for function pointers */
+ GST_DEBUG_REGISTER_FUNCPTR (gst_base_src_activate_push);
+ GST_DEBUG_REGISTER_FUNCPTR (gst_base_src_activate_pull);
+ GST_DEBUG_REGISTER_FUNCPTR (gst_base_src_event_handler);
+ GST_DEBUG_REGISTER_FUNCPTR (gst_base_src_query);
+ GST_DEBUG_REGISTER_FUNCPTR (gst_base_src_pad_get_range);
+ GST_DEBUG_REGISTER_FUNCPTR (gst_base_src_pad_check_get_range);
+ GST_DEBUG_REGISTER_FUNCPTR (gst_base_src_getcaps);
+ GST_DEBUG_REGISTER_FUNCPTR (gst_base_src_setcaps);
+ GST_DEBUG_REGISTER_FUNCPTR (gst_base_src_fixate);
}
static void
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));
+ gst_pad_set_activatepush_function (pad, gst_base_src_activate_push);
+ gst_pad_set_activatepull_function (pad, gst_base_src_activate_pull);
+ gst_pad_set_event_function (pad, gst_base_src_event_handler);
+ gst_pad_set_query_function (pad, gst_base_src_query);
+ gst_pad_set_checkgetrange_function (pad, gst_base_src_pad_check_get_range);
+ gst_pad_set_getrange_function (pad, gst_base_src_pad_get_range);
+ gst_pad_set_getcaps_function (pad, gst_base_src_getcaps);
+ gst_pad_set_setcaps_function (pad, gst_base_src_setcaps);
+ gst_pad_set_fixatecaps_function (pad, gst_base_src_fixate);
/* hold pointer to pad */
basesrc->srcpad = pad;
gst_base_src_set_format (basesrc, GST_FORMAT_BYTES);
basesrc->data.ABI.typefind = DEFAULT_TYPEFIND;
basesrc->priv->do_timestamp = DEFAULT_DO_TIMESTAMP;
+ g_atomic_int_set (&basesrc->priv->have_events, FALSE);
GST_OBJECT_FLAG_UNSET (basesrc, GST_BASE_SRC_STARTED);
+ GST_OBJECT_FLAG_SET (basesrc, GST_ELEMENT_IS_SOURCE);
GST_DEBUG_OBJECT (basesrc, "init done");
}
g_cond_free (basesrc->live_cond);
event_p = &basesrc->data.ABI.pending_seek;
- gst_event_replace ((GstEvent **) event_p, NULL);
+ gst_event_replace (event_p, NULL);
+
+ if (basesrc->priv->pending_events) {
+ g_list_foreach (basesrc->priv->pending_events, (GFunc) gst_event_unref,
+ NULL);
+ g_list_free (basesrc->priv->pending_events);
+ }
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.
+ * 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
GstFlowReturn
gst_base_src_wait_playing (GstBaseSrc * src)
{
- /* block until the state changes, or we get a flush, or something */
- GST_LIVE_LOCK (src);
- if (src->is_live) {
- while (G_UNLIKELY (!src->live_running)) {
- GST_DEBUG ("live source signal waiting");
- GST_LIVE_SIGNAL (src);
- GST_DEBUG ("live source waiting for running state");
- GST_LIVE_WAIT (src);
- GST_DEBUG ("live source unlocked");
- }
- /* FIXME, use another variable to signal stopping so that we don't
- * have to grab another lock. */
- GST_OBJECT_LOCK (src->srcpad);
- if (G_UNLIKELY (GST_PAD_IS_FLUSHING (src->srcpad)))
+ g_return_val_if_fail (GST_IS_BASE_SRC (src), GST_FLOW_ERROR);
+
+ do {
+ /* 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);
+ GST_DEBUG_OBJECT (src, "live source unlocked");
+ if (src->priv->flushing)
goto flushing;
- GST_OBJECT_UNLOCK (src->srcpad);
- }
- GST_LIVE_UNLOCK (src);
+ } while (G_UNLIKELY (!src->live_running));
return GST_FLOW_OK;
/* ERRORS */
flushing:
{
- GST_DEBUG_OBJECT (src, "pad is flushing");
- GST_OBJECT_UNLOCK (src->srcpad);
- GST_LIVE_UNLOCK (src);
+ GST_DEBUG_OBJECT (src, "we are flushing");
return GST_FLOW_WRONG_STATE;
}
}
* @live: new live-mode
*
* If the element listens to a live source, @live should
- * be set to %TRUE.
+ * 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
+ * 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_LIVE_LOCK (src);
+ g_return_if_fail (GST_IS_BASE_SRC (src));
+
+ GST_OBJECT_LOCK (src);
src->is_live = live;
- GST_LIVE_UNLOCK (src);
+ GST_OBJECT_UNLOCK (src);
}
/**
{
gboolean result;
- GST_LIVE_LOCK (src);
+ g_return_val_if_fail (GST_IS_BASE_SRC (src), FALSE);
+
+ GST_OBJECT_LOCK (src);
result = src->is_live;
- GST_LIVE_UNLOCK (src);
+ GST_OBJECT_UNLOCK (src);
return result;
}
* 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.
+ * operate in pull mode if the #GstBaseSrcClass.is_seekable() returns TRUE.
+ *
+ * This function must only be called in states < %GST_STATE_PAUSED.
*
- * @Since: 0.10.1
+ * Since: 0.10.1
*/
void
gst_base_src_set_format (GstBaseSrc * src, GstFormat format)
{
+ g_return_if_fail (GST_IS_BASE_SRC (src));
+ g_return_if_fail (GST_STATE (src) <= GST_STATE_READY);
+
+ GST_OBJECT_LOCK (src);
gst_segment_init (&src->segment, format);
+ GST_OBJECT_UNLOCK (src);
+}
+
+/**
+ * gst_base_src_set_dynamic_size:
+ * @src: base source instance
+ * @dynamic: new dynamic size mode
+ *
+ * If not @dynamic, size is only updated when needed, such as when trying to
+ * read past current tracked size. Otherwise, size is checked for upon each
+ * read.
+ *
+ * Since: 0.10.35
+ */
+void
+gst_base_src_set_dynamic_size (GstBaseSrc * src, gboolean dynamic)
+{
+ g_return_if_fail (GST_IS_BASE_SRC (src));
+
+ g_atomic_int_set (&src->priv->dynamic_size, dynamic);
}
/**
* 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
+ * @live: (out) (allow-none): if the source is live
+ * @min_latency: (out) (allow-none): the min latency of the source
+ * @max_latency: (out) (allow-none): 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.
+ * This function is mostly used by subclasses.
*
* Returns: TRUE if the query succeeded.
*
{
GstClockTime min;
- GST_LIVE_LOCK (src);
+ g_return_val_if_fail (GST_IS_BASE_SRC (src), FALSE);
+
+ GST_OBJECT_LOCK (src);
if (live)
*live = src->is_live;
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_LIVE_UNLOCK (src);
+ GST_OBJECT_UNLOCK (src);
return TRUE;
}
/**
+ * gst_base_src_set_blocksize:
+ * @src: the source
+ * @blocksize: the new blocksize in bytes
+ *
+ * Set the number of bytes that @src will push out with each buffer. When
+ * @blocksize is set to -1, a default length will be used.
+ *
+ * Since: 0.10.22
+ */
+/* FIXME 0.11: blocksize property should be int, not ulong */
+void
+gst_base_src_set_blocksize (GstBaseSrc * src, gulong blocksize)
+{
+ g_return_if_fail (GST_IS_BASE_SRC (src));
+
+ GST_OBJECT_LOCK (src);
+ src->blocksize = blocksize;
+ GST_OBJECT_UNLOCK (src);
+}
+
+/**
+ * gst_base_src_get_blocksize:
+ * @src: the source
+ *
+ * Get the number of bytes that @src will push out with each buffer.
+ *
+ * Returns: the number of bytes pushed with each buffer.
+ *
+ * Since: 0.10.22
+ */
+/* FIXME 0.11: blocksize property should be int, not ulong */
+gulong
+gst_base_src_get_blocksize (GstBaseSrc * src)
+{
+ gulong res;
+
+ g_return_val_if_fail (GST_IS_BASE_SRC (src), 0);
+
+ GST_OBJECT_LOCK (src);
+ res = src->blocksize;
+ GST_OBJECT_UNLOCK (src);
+
+ return res;
+}
+
+
+/**
* gst_base_src_set_do_timestamp:
* @src: the source
* @timestamp: enable or disable timestamping
void
gst_base_src_set_do_timestamp (GstBaseSrc * src, gboolean timestamp)
{
+ g_return_if_fail (GST_IS_BASE_SRC (src));
+
GST_OBJECT_LOCK (src);
src->priv->do_timestamp = timestamp;
GST_OBJECT_UNLOCK (src);
{
gboolean res;
+ g_return_val_if_fail (GST_IS_BASE_SRC (src), FALSE);
+
GST_OBJECT_LOCK (src);
res = src->priv->do_timestamp;
GST_OBJECT_UNLOCK (src);
return res;
}
+/**
+ * gst_base_src_new_seamless_segment:
+ * @src: The source
+ * @start: The new start value for the segment
+ * @stop: Stop value for the new segment
+ * @position: The position value for the new segent
+ *
+ * Prepare a new seamless segment for emission downstream. This function must
+ * only be called by derived sub-classes, and only from the create() function,
+ * as the stream-lock needs to be held.
+ *
+ * The format for the new segment will be the current format of the source, as
+ * configured with gst_base_src_set_format()
+ *
+ * Returns: %TRUE if preparation of the seamless segment succeeded.
+ *
+ * Since: 0.10.26
+ */
+gboolean
+gst_base_src_new_seamless_segment (GstBaseSrc * src, gint64 start, gint64 stop,
+ gint64 position)
+{
+ gboolean res = TRUE;
+
+ GST_DEBUG_OBJECT (src,
+ "Starting new seamless segment. Start %" GST_TIME_FORMAT " stop %"
+ GST_TIME_FORMAT " position %" GST_TIME_FORMAT, GST_TIME_ARGS (start),
+ GST_TIME_ARGS (stop), GST_TIME_ARGS (position));
+
+ GST_OBJECT_LOCK (src);
+ if (src->data.ABI.running && !src->priv->newsegment_pending) {
+ 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);
+ }
+
+ gst_segment_set_newsegment_full (&src->segment, FALSE, src->segment.rate,
+ src->segment.applied_rate, src->segment.format, start, stop, position);
+
+ 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 last_stop to start */
+ 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);
+ }
+ GST_OBJECT_UNLOCK (src);
+
+ src->priv->discont = TRUE;
+ src->data.ABI.running = TRUE;
+
+ return res;
+}
+
static gboolean
gst_base_src_setcaps (GstPad * pad, GstCaps * caps)
{
GstFormat format;
gst_query_parse_position (query, &format, NULL);
+
+ GST_DEBUG_OBJECT (src, "position query in format %s",
+ gst_format_get_name (format));
+
switch (format) {
case GST_FORMAT_PERCENT:
{
gint64 position;
gint64 duration;
+ GST_OBJECT_LOCK (src);
position = src->segment.last_stop;
duration = src->segment.duration;
+ GST_OBJECT_UNLOCK (src);
if (position != -1 && duration != -1) {
if (position < duration)
default:
{
gint64 position;
+ GstFormat seg_format;
- position = src->segment.last_stop;
+ GST_OBJECT_LOCK (src);
+ position =
+ gst_segment_to_stream_time (&src->segment, src->segment.format,
+ src->segment.last_stop);
+ seg_format = src->segment.format;
+ GST_OBJECT_UNLOCK (src);
if (position != -1) {
/* convert to requested format */
res =
- gst_pad_query_convert (src->srcpad, src->segment.format,
+ gst_pad_query_convert (src->srcpad, seg_format,
position, &format, &position);
} else
res = TRUE;
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,
default:
{
gint64 duration;
+ GstFormat seg_format;
+ guint length = 0;
+
+ /* may have to refresh duration */
+ if (g_atomic_int_get (&src->priv->dynamic_size))
+ gst_base_src_update_length (src, 0, &length);
+ /* this is the duration as configured by the subclass. */
+ GST_OBJECT_LOCK (src);
duration = src->segment.duration;
+ seg_format = src->segment.format;
+ GST_OBJECT_UNLOCK (src);
+
+ GST_LOG_OBJECT (src, "duration %" G_GINT64_FORMAT ", format %s",
+ duration, gst_format_get_name (seg_format));
if (duration != -1) {
- /* convert to requested format */
+ /* 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,
+ gst_pad_query_convert (src->srcpad, seg_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);
case GST_QUERY_SEEKING:
{
- gst_query_set_seeking (query, src->segment.format,
- src->seekable, 0, src->segment.duration);
- res = TRUE;
+ GstFormat format, seg_format;
+ gint64 duration;
+
+ GST_OBJECT_LOCK (src);
+ duration = src->segment.duration;
+ seg_format = src->segment.format;
+ GST_OBJECT_UNLOCK (src);
+
+ gst_query_parse_seeking (query, &format, NULL, NULL, NULL);
+ if (format == seg_format) {
+ gst_query_set_seeking (query, seg_format,
+ gst_base_src_seekable (src), 0, duration);
+ res = TRUE;
+ } else {
+ /* FIXME 0.11: return TRUE + seekable=FALSE for SEEKING query here */
+ /* Don't reply to the query to make up for demuxers which don't
+ * handle the SEEKING query yet. Players like Totem will fall back
+ * to the duration when the SEEKING query isn't answered. */
+ res = FALSE;
+ }
break;
}
case GST_QUERY_SEGMENT:
{
gint64 start, stop;
+ GST_OBJECT_LOCK (src);
/* no end segment configured, current duration then */
if ((stop = src->segment.stop) == -1)
stop = src->segment.duration;
if (stop != -1)
stop -= src->segment.time;
}
+
gst_query_set_segment (query, src->segment.rate, src->segment.format,
start, stop);
+ GST_OBJECT_UNLOCK (src);
res = TRUE;
break;
}
GstClockTime min, max;
gboolean live;
- /* Subclasses should override and implement something usefull */
+ /* Subclasses should override and implement something useful */
res = gst_base_src_query_latency (src, &live, &min, &max);
GST_LOG_OBJECT (src, "report latency: live %d, min %" GST_TIME_FORMAT
}
case GST_QUERY_JITTER:
case GST_QUERY_RATE:
+ res = FALSE;
+ break;
+ case GST_QUERY_BUFFERING:
+ {
+ GstFormat format, seg_format;
+ gint64 start, stop, estimated;
+
+ gst_query_parse_buffering_range (query, &format, NULL, NULL, NULL);
+
+ GST_DEBUG_OBJECT (src, "buffering query in format %s",
+ gst_format_get_name (format));
+
+ GST_OBJECT_LOCK (src);
+ if (src->random_access) {
+ estimated = 0;
+ start = 0;
+ if (format == GST_FORMAT_PERCENT)
+ stop = GST_FORMAT_PERCENT_MAX;
+ else
+ stop = src->segment.duration;
+ } else {
+ estimated = -1;
+ start = -1;
+ stop = -1;
+ }
+ seg_format = src->segment.format;
+ GST_OBJECT_UNLOCK (src);
+
+ /* convert to required format. When the conversion fails, we can't answer
+ * the query. When the value is unknown, we can don't perform conversion
+ * but report TRUE. */
+ if (format != GST_FORMAT_PERCENT && stop != -1) {
+ res = gst_pad_query_convert (src->srcpad, seg_format,
+ stop, &format, &stop);
+ } else {
+ res = TRUE;
+ }
+ if (res && format != GST_FORMAT_PERCENT && start != -1)
+ res = gst_pad_query_convert (src->srcpad, seg_format,
+ start, &format, &start);
+
+ gst_query_set_buffering_range (query, format, start, stop, estimated);
+ break;
+ }
default:
res = FALSE;
break;
gboolean result = FALSE;
src = GST_BASE_SRC (gst_pad_get_parent (pad));
+ if (G_UNLIKELY (src == NULL))
+ return FALSE;
bclass = GST_BASE_SRC_GET_CLASS (src);
/* update our offset if the start/stop position was updated */
if (segment->format == GST_FORMAT_BYTES) {
- segment->last_stop = segment->start;
segment->time = segment->start;
} else if (segment->start == 0) {
/* seek to start, we can implement a default for this. */
- segment->last_stop = 0;
segment->time = 0;
- res = TRUE;
- } else
+ } else {
res = FALSE;
+ GST_INFO_OBJECT (src, "Can't do a default seek");
+ }
return res;
}
GstSegment * segment)
{
/* By default, we try one of 2 things:
- * - For absolute seek positions, convert the requested position to our
+ * - 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
* 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.
+ * 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
* 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.
+ * 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 stream-time
+ * 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
+ * a new_segment event with the stop position set to the last processed
* position.
*
* After updating the segment.start/stop values, we prepare for
* 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
+/* 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
+ * 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;
+ gboolean res = TRUE, tres;
gdouble rate;
GstFormat seek_format, dest_format;
GstSeekFlags flags;
GstSeekType cur_type, stop_type;
gint64 cur, stop;
- gboolean flush;
+ gboolean flush, playing;
gboolean update;
gboolean relative_seek = FALSE;
gboolean seekseg_configured = FALSE;
GstSegment seeksegment;
+ guint32 seqnum;
+ GstEvent *tevent;
- GST_DEBUG_OBJECT (src, "doing seek");
+ GST_DEBUG_OBJECT (src, "doing seek: %" GST_PTR_FORMAT, event);
+ GST_OBJECT_LOCK (src);
dest_format = src->segment.format;
+ GST_OBJECT_UNLOCK (src);
if (event) {
gst_event_parse_seek (event, &rate, &seek_format, &flags,
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 */
+ * once we have the stream lock and can read the last configures segment
+ * start and stop positions */
gst_segment_init (&seeksegment, dest_format);
if (!gst_base_src_prepare_seek_segment (src, event, &seeksegment))
}
flush = flags & GST_SEEK_FLAG_FLUSH;
+ seqnum = gst_event_get_seqnum (event);
} else {
flush = FALSE;
+ /* get next seqnum */
+ seqnum = gst_util_seqnum_next ();
}
/* send flush start */
- if (flush)
- gst_pad_push_event (src->srcpad, gst_event_new_flush_start ());
- else
+ if (flush) {
+ tevent = gst_event_new_flush_start ();
+ gst_event_set_seqnum (tevent, seqnum);
+ gst_pad_push_event (src->srcpad, tevent);
+ } else
gst_pad_pause_task (src->srcpad);
- /* unblock streaming thread */
- if (unlock)
- gst_base_src_unlock (src);
+ /* 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 or our streaming thread stopped
- * because our peer is flushing. */
+ * because the task is paused, our streaming thread stopped
+ * or because our peer is flushing. */
GST_PAD_STREAM_LOCK (src->srcpad);
+ if (G_UNLIKELY (src->priv->seqnum == seqnum)) {
+ /* we have seen this event before, issue a warning for now */
+ GST_WARNING_OBJECT (src, "duplicate event found %" G_GUINT32_FORMAT,
+ seqnum);
+ } else {
+ src->priv->seqnum = seqnum;
+ GST_DEBUG_OBJECT (src, "seek with seqnum %" G_GUINT32_FORMAT, seqnum);
+ }
- if (unlock)
- gst_base_src_unlock_stop (src);
+ 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. */
+ * attempt the seek with. We only update the real segment if the seek succeeds. */
if (!seekseg_configured) {
memcpy (&seeksegment, &src->segment, sizeof (GstSegment));
/* now configure the final seek segment */
if (event) {
- if (src->segment.format != seek_format) {
+ if (seeksegment.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. */
cur_type, cur, stop_type, stop, &update);
}
}
- /* Else, no seek event passed, so we're just (re)starting the
+ /* Else, no seek event passed, so we're just (re)starting the
current segment. */
}
/* and prepare to continue streaming */
if (flush) {
+ tevent = gst_event_new_flush_stop ();
+ gst_event_set_seqnum (tevent, seqnum);
/* 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 ());
+ gst_pad_push_event (src->srcpad, tevent);
} else if (res && src->data.ABI.running) {
- /* we are running the current segment and doing a non-flushing seek,
+ /* 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);
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);
+ gst_event_set_seqnum (src->priv->close_segment, seqnum);
}
- /* The subclass must have converted the segment to the processing format
+ /* 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 "
res = FALSE;
}
- /* if successfull seek, we update our real segment and push
+ /* if the seek was successful, we update our real segment and push
* out the new segment. */
if (res) {
+ GST_OBJECT_LOCK (src);
memcpy (&src->segment, &seeksegment, sizeof (GstSegment));
+ GST_OBJECT_UNLOCK (src);
+
+ if (seeksegment.flags & GST_SEEK_FLAG_SEGMENT) {
+ GstMessage *message;
+
+ message = gst_message_new_segment_start (GST_OBJECT (src),
+ seeksegment.format, seeksegment.last_stop);
+ gst_message_set_seqnum (message, seqnum);
- 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));
+ gst_element_post_message (GST_ELEMENT (src), message);
}
- /* for deriving a stop position for the playback segment form the seek
+ /* 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;
+ if ((stop = seeksegment.stop) == -1)
+ stop = seeksegment.duration;
GST_DEBUG_OBJECT (src, "Sending newsegment from %" G_GINT64_FORMAT
- " to %" G_GINT64_FORMAT, src->segment.start, stop);
+ " to %" G_GINT64_FORMAT, seeksegment.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);
- 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);
+ if (seeksegment.rate >= 0.0) {
+ /* forward, we send data from last_stop to stop */
+ src->priv->start_segment =
+ gst_event_new_new_segment_full (FALSE,
+ seeksegment.rate, seeksegment.applied_rate, seeksegment.format,
+ seeksegment.last_stop, stop, seeksegment.time);
+ } else {
+ /* reverse, we send data from last_stop to start */
+ src->priv->start_segment =
+ gst_event_new_new_segment_full (FALSE,
+ seeksegment.rate, seeksegment.applied_rate, seeksegment.format,
+ seeksegment.start, seeksegment.last_stop, seeksegment.time);
+ }
+ gst_event_set_seqnum (src->priv->start_segment, seqnum);
+ src->priv->newsegment_pending = TRUE;
}
src->priv->discont = TRUE;
src->data.ABI.running = TRUE;
- /* and restart the task in case it got paused explicitely or by
+ /* and restart the task in case it got paused explicitly or by
* the FLUSH_START event we pushed out. */
- gst_pad_start_task (src->srcpad, (GstTaskFunction) gst_base_src_loop,
+ tres = gst_pad_start_task (src->srcpad, (GstTaskFunction) gst_base_src_loop,
src->srcpad);
+ if (res && !tres)
+ res = FALSE;
/* and release the lock again so we can continue streaming */
GST_PAD_STREAM_UNLOCK (src->srcpad);
src = GST_BASE_SRC (element);
+ GST_DEBUG_OBJECT (src, "handling event %p %" GST_PTR_FORMAT, event, event);
+
switch (GST_EVENT_TYPE (event)) {
/* bidirectional events */
case GST_EVENT_FLUSH_START:
/* downstream serialized events */
case GST_EVENT_EOS:
- /* FIXME, queue EOS and make sure the task or pull function
- * perform the EOS actions. */
+ {
+ GstBaseSrcClass *bclass;
+
+ bclass = GST_BASE_SRC_GET_CLASS (src);
+
+ /* queue EOS and make sure the task or pull function performs the EOS
+ * actions.
+ *
+ * We have two possibilities:
+ *
+ * - Before we are to enter the _create function, we check the pending_eos
+ * first and do EOS instead of entering it.
+ * - If we are in the _create function or we did not manage to set the
+ * flag fast enough and we are about to enter the _create function,
+ * we unlock it so that we exit with WRONG_STATE immediately. We then
+ * check the EOS flag and do the EOS logic.
+ */
+ g_atomic_int_set (&src->priv->pending_eos, TRUE);
+ GST_DEBUG_OBJECT (src, "EOS marked, calling unlock");
+
+ /* unlock the _create function so that we can check the pending_eos flag
+ * and we can do EOS. This will eventually release the LIVE_LOCK again so
+ * that we can grab it and stop the unlock again. We don't take the stream
+ * lock so that this operation is guaranteed to never block. */
+ if (bclass->unlock)
+ bclass->unlock (src);
+
+ GST_DEBUG_OBJECT (src, "unlock called, waiting for LIVE_LOCK");
+
+ GST_LIVE_LOCK (src);
+ GST_DEBUG_OBJECT (src, "LIVE_LOCK acquired, calling unlock_stop");
+ /* now stop the unlock of the streaming thread again. Grabbing the live
+ * lock is enough because that protects the create function. */
+ if (bclass->unlock_stop)
+ bclass->unlock_stop (src);
+ 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 */
+ case GST_EVENT_CUSTOM_DOWNSTREAM:
+ case GST_EVENT_CUSTOM_BOTH:
+ /* Insert TAG, CUSTOM_DOWNSTREAM, CUSTOM_BOTH in the dataflow */
+ GST_OBJECT_LOCK (src);
+ src->priv->pending_events =
+ g_list_append (src->priv->pending_events, event);
+ g_atomic_int_set (&src->priv->have_events, TRUE);
+ GST_OBJECT_UNLOCK (src);
+ event = NULL;
+ result = TRUE;
break;
case GST_EVENT_BUFFERSIZE:
/* does not seem to make much sense currently */
GST_OBJECT_UNLOCK (src->srcpad);
if (started) {
+ GST_DEBUG_OBJECT (src, "performing seek");
/* 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 we store the event and execute the seek when we
* get activated */
GST_OBJECT_LOCK (src);
+ GST_DEBUG_OBJECT (src, "queueing seek");
event_p = &src->data.ABI.pending_seek;
gst_event_replace ((GstEvent **) event_p, event);
GST_OBJECT_UNLOCK (src);
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 */
}
static gboolean
+gst_base_src_seekable (GstBaseSrc * src)
+{
+ GstBaseSrcClass *bclass;
+ bclass = GST_BASE_SRC_GET_CLASS (src);
+ if (bclass->is_seekable)
+ return bclass->is_seekable (src);
+ else
+ return FALSE;
+}
+
+static void
+gst_base_src_update_qos (GstBaseSrc * src,
+ gdouble proportion, GstClockTimeDiff diff, GstClockTime timestamp)
+{
+ GST_CAT_DEBUG_OBJECT (GST_CAT_QOS, src,
+ "qos: proportion: %lf, diff %" G_GINT64_FORMAT ", timestamp %"
+ GST_TIME_FORMAT, proportion, diff, GST_TIME_ARGS (timestamp));
+
+ GST_OBJECT_LOCK (src);
+ src->priv->proportion = proportion;
+ src->priv->earliest_time = timestamp + diff;
+ GST_OBJECT_UNLOCK (src);
+}
+
+
+static gboolean
gst_base_src_default_event (GstBaseSrc * src, GstEvent * event)
{
gboolean result;
+ GST_DEBUG_OBJECT (src, "handle event %" GST_PTR_FORMAT, event);
+
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEEK:
/* is normally called when in push mode */
- if (!src->seekable)
+ if (!gst_base_src_seekable (src))
goto not_seekable;
result = gst_base_src_perform_seek (src, event, TRUE);
case GST_EVENT_FLUSH_START:
/* cancel any blocking getrange, is normally called
* when in pull mode. */
- result = gst_base_src_unlock (src);
+ result = gst_base_src_set_flushing (src, TRUE, FALSE, TRUE, NULL);
break;
case GST_EVENT_FLUSH_STOP:
- result = gst_base_src_unlock_stop (src);
+ result = gst_base_src_set_flushing (src, FALSE, TRUE, TRUE, NULL);
break;
- default:
+ case GST_EVENT_QOS:
+ {
+ gdouble proportion;
+ GstClockTimeDiff diff;
+ GstClockTime timestamp;
+
+ gst_event_parse_qos (event, &proportion, &diff, ×tamp);
+ gst_base_src_update_qos (src, proportion, diff, timestamp);
result = TRUE;
break;
+ }
+ default:
+ result = FALSE;
+ break;
}
return result;
gboolean result = FALSE;
src = GST_BASE_SRC (gst_pad_get_parent (pad));
+ if (G_UNLIKELY (src == NULL)) {
+ gst_event_unref (event);
+ return FALSE;
+ }
+
bclass = GST_BASE_SRC_GET_CLASS (src);
if (bclass->event) {
switch (prop_id) {
case PROP_BLOCKSIZE:
- src->blocksize = g_value_get_ulong (value);
+ gst_base_src_set_blocksize (src, g_value_get_ulong (value));
break;
case PROP_NUM_BUFFERS:
src->num_buffers = g_value_get_int (value);
src->data.ABI.typefind = g_value_get_boolean (value);
break;
case PROP_DO_TIMESTAMP:
- src->priv->do_timestamp = g_value_get_boolean (value);
+ gst_base_src_set_do_timestamp (src, g_value_get_boolean (value));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
switch (prop_id) {
case PROP_BLOCKSIZE:
- g_value_set_ulong (value, src->blocksize);
+ g_value_set_ulong (value, gst_base_src_get_blocksize (src));
break;
case PROP_NUM_BUFFERS:
g_value_set_int (value, src->num_buffers);
g_value_set_boolean (value, src->data.ABI.typefind);
break;
case PROP_DO_TIMESTAMP:
- g_value_set_boolean (value, src->priv->do_timestamp);
+ g_value_set_boolean (value, gst_base_src_get_do_timestamp (src));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
id = gst_clock_new_single_shot_id (clock, time);
basesrc->clock_id = id;
- /* release the object lock while waiting */
- GST_OBJECT_UNLOCK (basesrc);
+ /* release the live lock while waiting */
+ GST_LIVE_UNLOCK (basesrc);
ret = gst_clock_id_wait (id, NULL);
- GST_OBJECT_LOCK (basesrc);
+ GST_LIVE_LOCK (basesrc);
gst_clock_id_unref (id);
basesrc->clock_id = NULL;
return ret;
}
-/* perform synchronisation on a buffer.
+/* perform synchronisation on a buffer.
* with STREAM_LOCK.
*/
static GstClockReturn
/* get buffer timestamp */
timestamp = GST_BUFFER_TIMESTAMP (buffer);
- /* grab the lock to prepare for clocking and calculate the startup
+ /* grab the lock to prepare for clocking and calculate the startup
* latency. */
GST_OBJECT_LOCK (basesrc);
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));
}
}
"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_OBJECT_UNLOCK (basesrc);
GST_LOG_OBJECT (basesrc, "clock entry done: %d", result);
}
}
+/* Called with STREAM_LOCK and LIVE_LOCK */
static gboolean
gst_base_src_update_length (GstBaseSrc * src, guint64 offset, guint * length)
{
guint64 size, maxsize;
GstBaseSrcClass *bclass;
+ GstFormat format;
+ gint64 stop;
+ gboolean dynamic;
bclass = GST_BASE_SRC_GET_CLASS (src);
- /* only operate if we are working with bytes */
- if (src->segment.format != GST_FORMAT_BYTES)
- return TRUE;
-
+ format = src->segment.format;
+ stop = src->segment.stop;
/* get total file size */
size = (guint64) src->segment.duration;
+ /* only operate if we are working with bytes */
+ if (format != GST_FORMAT_BYTES)
+ return TRUE;
+
/* 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);
+ if (stop != -1)
+ maxsize = MIN (size, 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);
+ *length, size, stop, maxsize);
+
+ dynamic = g_atomic_int_get (&src->priv->dynamic_size);
+ GST_DEBUG_OBJECT (src, "dynamic size: %d", dynamic);
/* check size if we have one */
if (maxsize != -1) {
- /* if we run past the end, check if the file became bigger and
+ /* if we run past the end, check if the file became bigger and
* retry. */
- if (G_UNLIKELY (offset + *length >= maxsize)) {
+ if (G_UNLIKELY (offset + *length >= maxsize || dynamic)) {
/* 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);
+ if (stop != -1)
+ maxsize = MIN (size, stop);
else
maxsize = size;
}
}
- /* keep track of current position. segment is in bytes, we checked
- * that above. */
- gst_segment_set_last_stop (&src->segment, GST_FORMAT_BYTES, offset);
+ /* keep track of current position and update duration.
+ * segment is in bytes, we checked that above. */
+ GST_OBJECT_LOCK (src);
+ gst_segment_set_duration (&src->segment, GST_FORMAT_BYTES, size);
+ GST_OBJECT_UNLOCK (src);
return TRUE;
}
}
+/* must be called with LIVE_LOCK */
static GstFlowReturn
gst_base_src_get_range (GstBaseSrc * src, guint64 offset, guint length,
GstBuffer ** buf)
bclass = GST_BASE_SRC_GET_CLASS (src);
- ret = gst_base_src_wait_playing (src);
- if (ret != GST_FLOW_OK)
- goto stopped;
+again:
+ if (src->is_live) {
+ if (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 (!gst_base_src_update_length (src, offset, &length)))
goto unexpected_length;
+ /* track position */
+ GST_OBJECT_LOCK (src);
+ if (src->segment.format == GST_FORMAT_BYTES)
+ gst_segment_set_last_stop (&src->segment, GST_FORMAT_BYTES, offset);
+ GST_OBJECT_UNLOCK (src);
+
/* normally we don't count buffers */
if (G_UNLIKELY (src->num_buffers_left >= 0)) {
if (src->num_buffers_left == 0)
src->num_buffers_left--;
}
+ /* don't enter the create function if a pending EOS event was set. For the
+ * logic of the pending_eos, check the event function of this class. */
+ if (G_UNLIKELY (g_atomic_int_get (&src->priv->pending_eos)))
+ goto eos;
+
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);
+
+ /* The create function could be unlocked because we have a pending EOS. It's
+ * possible that we have a valid buffer from create that we need to
+ * discard when the create function returned _OK. */
+ if (G_UNLIKELY (g_atomic_int_get (&src->priv->pending_eos))) {
+ if (ret == GST_FLOW_OK) {
+ gst_buffer_unref (*buf);
+ *buf = NULL;
+ }
+ goto eos;
+ }
+
if (G_UNLIKELY (ret != GST_FLOW_OK))
- goto done;
+ 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) == -1 && !src->is_live) {
+ *buf = gst_buffer_make_metadata_writable (*buf);
GST_BUFFER_TIMESTAMP (*buf) = 0;
+ }
+
+ /* set pad caps on the buffer if the buffer had no caps */
+ if (GST_BUFFER_CAPS (*buf) == NULL) {
+ *buf = gst_buffer_make_metadata_writable (*buf);
+ gst_buffer_set_caps (*buf, GST_PAD_CAPS (src->srcpad));
+ }
/* 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;
+
switch (status) {
case GST_CLOCK_EARLY:
/* the buffer is too late. We currently don't drop the buffer. */
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);
+ * it got unlocked because we did a state change. In any case, get rid of
+ * the buffer. */
gst_buffer_unref (*buf);
*buf = NULL;
- ret = GST_FLOW_WRONG_STATE;
+ if (!src->live_running) {
+ /* We return WRONG_STATE when we are not running to stop the dataflow also
+ * get rid of the produced buffer. */
+ GST_DEBUG_OBJECT (src,
+ "clock was unscheduled (%d), returning WRONG_STATE", status);
+ ret = GST_FLOW_WRONG_STATE;
+ } else {
+ /* If we are running when this happens, we quickly switched between
+ * pause and playing. We try to produce a new buffer */
+ GST_DEBUG_OBJECT (src,
+ "clock was unscheduled (%d), but we are running", status);
+ goto again;
+ }
break;
default:
/* all other result values are unexpected and errors */
ret = GST_FLOW_ERROR;
break;
}
-done:
return ret;
/* ERROR */
stopped:
{
- GST_DEBUG_OBJECT (src, "wait_playing returned %d", ret);
+ 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, "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");
+ return GST_FLOW_UNEXPECTED;
+ }
}
static GstFlowReturn
GstBaseSrc *src;
GstFlowReturn res;
- src = GST_BASE_SRC (gst_pad_get_parent (pad));
+ src = GST_BASE_SRC_CAST (gst_object_ref (GST_OBJECT_PARENT (pad)));
+
+ GST_LIVE_LOCK (src);
+ if (G_UNLIKELY (src->priv->flushing))
+ goto flushing;
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;
+ }
}
static gboolean
GstBaseSrc *src;
gboolean res;
- src = GST_BASE_SRC (gst_pad_get_parent (pad));
+ src = GST_BASE_SRC (GST_OBJECT_PARENT (pad));
res = gst_base_src_check_get_range (src);
- gst_object_unref (src);
-
return res;
}
GstFlowReturn ret;
gint64 position;
gboolean eos;
+ gulong blocksize;
+ GList *pending_events = NULL, *tmp;
eos = FALSE;
- src = GST_BASE_SRC (gst_pad_get_parent (pad));
+ src = GST_BASE_SRC (GST_OBJECT_PARENT (pad));
+
+ GST_LIVE_LOCK (src);
+
+ if (G_UNLIKELY (src->priv->flushing))
+ goto flushing;
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)
+ if (src->segment.format == GST_FORMAT_BYTES) {
position = src->segment.last_stop;
- else
+ /* 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, src->blocksize, &buf);
+ GST_LOG_OBJECT (src, "next_ts %" GST_TIME_FORMAT " size %lu",
+ GST_TIME_ARGS (position), blocksize);
+
+ 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 */
goto null_buffer;
/* push events to close/start our segment before we push the buffer. */
- if (src->priv->close_segment) {
+ if (G_UNLIKELY (src->priv->close_segment)) {
gst_pad_push_event (pad, src->priv->close_segment);
src->priv->close_segment = NULL;
}
- if (src->priv->start_segment) {
+ if (G_UNLIKELY (src->priv->start_segment)) {
gst_pad_push_event (pad, src->priv->start_segment);
src->priv->start_segment = NULL;
}
+ src->priv->newsegment_pending = FALSE;
+
+ if (g_atomic_int_get (&src->priv->have_events)) {
+ GST_OBJECT_LOCK (src);
+ /* take the events */
+ pending_events = src->priv->pending_events;
+ src->priv->pending_events = NULL;
+ g_atomic_int_set (&src->priv->have_events, FALSE);
+ GST_OBJECT_UNLOCK (src);
+ }
+
+ /* Push out pending events if any */
+ if (G_UNLIKELY (pending_events != NULL)) {
+ for (tmp = pending_events; tmp; tmp = g_list_next (tmp)) {
+ GstEvent *ev = (GstEvent *) tmp->data;
+ gst_pad_push_event (pad, ev);
+ }
+ g_list_free (pending_events);
+ }
/* figure out the new position */
switch (src->segment.format) {
case GST_FORMAT_BYTES:
- position += GST_BUFFER_SIZE (buf);
+ {
+ 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;
else
position = src->segment.last_stop;
- if (GST_CLOCK_TIME_IS_VALID (duration))
- position += duration;
+ 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:
- position = GST_BUFFER_OFFSET_END (buf);
+ 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.stop != -1) {
- if (position >= src->segment.stop) {
+ 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.stop;
+ position = src->segment.start;
}
+ /* when going reverse, all buffers are DISCONT */
+ src->priv->discont = TRUE;
}
+ GST_OBJECT_LOCK (src);
gst_segment_set_last_stop (&src->segment, src->segment.format, position);
+ GST_OBJECT_UNLOCK (src);
}
if (G_UNLIKELY (src->priv->discont)) {
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)) {
goto pause;
}
- if (eos) {
- GST_INFO_OBJECT (src, "pausing after EOS");
+ if (G_UNLIKELY (eos)) {
+ GST_INFO_OBJECT (src, "pausing after end of segment");
ret = GST_FLOW_UNEXPECTED;
goto pause;
}
done:
- gst_object_unref (src);
return;
/* special cases */
+flushing:
+ {
+ GST_DEBUG_OBJECT (src, "we are flushing");
+ GST_LIVE_UNLOCK (src);
+ ret = GST_FLOW_WRONG_STATE;
+ goto pause;
+ }
pause:
{
const gchar *reason = gst_flow_get_name (ret);
+ GstEvent *event;
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;
- }
+ if (ret == GST_FLOW_UNEXPECTED) {
+ gboolean flag_segment;
+ GstFormat format;
+ gint64 last_stop;
+
+ /* perform EOS logic */
+ flag_segment = (src->segment.flags & GST_SEEK_FLAG_SEGMENT) != 0;
+ format = src->segment.format;
+ last_stop = src->segment.last_stop;
+
+ if (flag_segment) {
+ GstMessage *message;
+
+ message = gst_message_new_segment_done (GST_OBJECT_CAST (src),
+ format, last_stop);
+ gst_message_set_seqnum (message, src->priv->seqnum);
+ gst_element_post_message (GST_ELEMENT_CAST (src), message);
} 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 ());
+ event = gst_event_new_eos ();
+ gst_event_set_seqnum (event, src->priv->seqnum);
+ gst_pad_push_event (pad, event);
src->priv->last_sent_eos = TRUE;
}
+ } else if (ret == GST_FLOW_NOT_LINKED || ret <= GST_FLOW_UNEXPECTED) {
+ event = gst_event_new_eos ();
+ gst_event_set_seqnum (event, src->priv->seqnum);
+ /* for fatal errors we post an error message, post the error
+ * first so the app knows about the error first.
+ * Also don't do this for WRONG_STATE because it happens
+ * due to flushing and posting an error message because of
+ * that is the wrong thing to do, e.g. when we're doing
+ * a flushing seek. */
+ GST_ELEMENT_ERROR (src, STREAM, FAILED,
+ (_("Internal data flow error.")),
+ ("streaming task paused, reason %s (%d)", reason, ret));
+ gst_pad_push_event (pad, event);
+ src->priv->last_sent_eos = TRUE;
}
goto done;
}
{
GST_ELEMENT_ERROR (src, STREAM, FAILED,
(_("Internal data flow error.")), ("element returned NULL buffer"));
- /* we finished the segment on error */
- src->data.ABI.running = FALSE;
- gst_pad_pause_task (pad);
- gst_pad_push_event (pad, gst_event_new_eos ());
- src->priv->last_sent_eos = TRUE;
+ GST_LIVE_UNLOCK (src);
goto done;
}
}
-/* this will always be called between start() and stop(). So you can rely on
- * resources allocated by start() and freed from stop(). This needs to be added
- * to the docs at some point. */
-static gboolean
-gst_base_src_unlock (GstBaseSrc * basesrc)
-{
- GstBaseSrcClass *bclass;
- gboolean result = TRUE;
-
- GST_DEBUG ("unlock");
- /* unblock whatever the subclass is doing */
- bclass = GST_BASE_SRC_GET_CLASS (basesrc);
- if (bclass->unlock)
- result = bclass->unlock (basesrc);
-
- GST_DEBUG ("unschedule clock");
- /* and unblock the clock as well, if any */
- GST_OBJECT_LOCK (basesrc);
- if (basesrc->clock_id) {
- gst_clock_id_unschedule (basesrc->clock_id);
- }
- GST_OBJECT_UNLOCK (basesrc);
-
- GST_DEBUG ("unlock done");
-
- return result;
-}
-
-/* this will always be called between start() and stop(). So you can rely on
- * resources allocated by start() and freed from stop(). This needs to be added
- * to the docs at some point. */
-static gboolean
-gst_base_src_unlock_stop (GstBaseSrc * basesrc)
-{
- GstBaseSrcClass *bclass;
- gboolean result = TRUE;
-
- GST_DEBUG_OBJECT (basesrc, "unlock stop");
-
- /* Finish a previous unblock request, allowing subclasses to flush command
- * queues or whatever they need to do */
- bclass = GST_BASE_SRC_GET_CLASS (basesrc);
- if (bclass->unlock_stop)
- result = bclass->unlock_stop (basesrc);
-
- GST_DEBUG_OBJECT (basesrc, "unlock stop done");
-
- return result;
-}
-
-/* default negotiation code.
+/* default negotiation code.
*
* Take intersection between src and sink pads, take first
- * caps and fixate.
+ * caps and fixate.
*/
static gboolean
gst_base_src_default_negotiate (GstBaseSrc * basesrc)
gboolean result = FALSE;
/* first see what is possible on our source pad */
- thiscaps = gst_pad_get_caps (GST_BASE_SRC_PAD (basesrc));
+ thiscaps = gst_pad_get_caps_reffed (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;
+ if (G_UNLIKELY (gst_caps_is_empty (thiscaps)))
+ goto no_caps;
+
/* get the peer caps */
- peercaps = gst_pad_peer_get_caps (GST_BASE_SRC_PAD (basesrc));
+ peercaps = gst_pad_peer_get_caps_reffed (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);
+ caps =
+ gst_caps_intersect_full (peercaps, thiscaps, GST_CAPS_INTERSECT_FIRST);
+ GST_DEBUG_OBJECT (basesrc, "intersect: %" GST_PTR_FORMAT, caps);
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;
+ caps = gst_caps_copy (thiscaps);
}
+ gst_caps_unref (thiscaps);
if (caps) {
- caps = gst_caps_make_writable (caps);
+ /* take first (and best, since they are sorted) possibility */
gst_caps_truncate (caps);
/* now fixate */
* 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;
+ /* yay, fixed caps, use those then, it's possible that the subclass does
+ * not accept this caps after all and we have to fail. */
+ result = gst_pad_set_caps (GST_BASE_SRC_PAD (basesrc), caps);
}
}
gst_caps_unref (caps);
+ } else {
+ GST_DEBUG_OBJECT (basesrc, "no common caps");
}
return result;
gst_caps_unref (thiscaps);
return TRUE;
}
+no_caps:
+ {
+ GST_ELEMENT_ERROR (basesrc, STREAM, FORMAT,
+ ("No supported formats found"),
+ ("This element did not produce valid caps"));
+ if (thiscaps)
+ gst_caps_unref (thiscaps);
+ return TRUE;
+ }
}
static gboolean
GstBaseSrcClass *bclass;
gboolean result;
guint64 size;
+ gboolean seekable;
+ GstFormat format;
if (GST_OBJECT_FLAG_IS_SET (basesrc, GST_BASE_SRC_STARTED))
return TRUE;
basesrc->num_buffers_left = basesrc->num_buffers;
+ GST_OBJECT_LOCK (basesrc);
gst_segment_init (&basesrc->segment, basesrc->segment.format);
+ GST_OBJECT_UNLOCK (basesrc);
+
basesrc->data.ABI.running = FALSE;
+ basesrc->priv->newsegment_pending = FALSE;
bclass = GST_BASE_SRC_GET_CLASS (basesrc);
if (bclass->start)
GST_OBJECT_FLAG_SET (basesrc, GST_BASE_SRC_STARTED);
+ format = basesrc->segment.format;
+
/* figure out the size */
- if (basesrc->segment.format == GST_FORMAT_BYTES) {
+ if (format == GST_FORMAT_BYTES) {
if (bclass->get_size) {
if (!(result = bclass->get_size (basesrc, &size)))
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_OBJECT_LOCK (basesrc);
gst_segment_set_duration (&basesrc->segment, GST_FORMAT_BYTES, size);
+ GST_OBJECT_UNLOCK (basesrc);
} 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,
+ "format: %s, have size: %d, size: %" G_GUINT64_FORMAT ", duration: %"
+ G_GINT64_FORMAT, gst_format_get_name (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);
+ seekable = gst_base_src_seekable (basesrc);
+ GST_DEBUG_OBJECT (basesrc, "is seekable: %d", seekable);
/* update for random access flag */
- basesrc->random_access = basesrc->seekable &&
- basesrc->segment.format == GST_FORMAT_BYTES;
+ basesrc->random_access = seekable && format == GST_FORMAT_BYTES;
GST_DEBUG_OBJECT (basesrc, "is random_access: %d", basesrc->random_access);
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);
+ if (!(caps = gst_type_find_helper (basesrc->srcpad, size)))
+ goto typefind_failed;
+
+ result = gst_pad_set_caps (basesrc->srcpad, caps);
gst_caps_unref (caps);
} else {
/* use class or default negotiate function */
- if (!gst_base_src_negotiate (basesrc))
+ if (!(result = gst_base_src_negotiate (basesrc)))
goto could_not_negotiate;
}
- return TRUE;
+ return result;
/* ERROR */
could_not_start:
gst_base_src_stop (basesrc);
return FALSE;
}
+typefind_failed:
+ {
+ GST_DEBUG_OBJECT (basesrc, "could not typefind, stopping");
+ GST_ELEMENT_ERROR (basesrc, STREAM, TYPE_NOT_FOUND, (NULL), (NULL));
+ /* we must call stop */
+ gst_base_src_stop (basesrc);
+ return FALSE;
+ }
}
static gboolean
return result;
}
+/* start or stop flushing dataprocessing
+ */
static gboolean
-gst_base_src_deactivate (GstBaseSrc * basesrc, GstPad * pad)
+gst_base_src_set_flushing (GstBaseSrc * basesrc,
+ gboolean flushing, gboolean live_play, gboolean unlock, gboolean * playing)
{
- gboolean result;
+ 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);
- basesrc->live_running = TRUE;
+ 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 */
+ g_atomic_int_set (&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;
+
+ /* When unlocking drop all delayed events */
+ if (unlock) {
+ GST_OBJECT_LOCK (basesrc);
+ if (basesrc->priv->pending_events) {
+ g_list_foreach (basesrc->priv->pending_events, (GFunc) gst_event_unref,
+ NULL);
+ g_list_free (basesrc->priv->pending_events);
+ basesrc->priv->pending_events = NULL;
+ g_atomic_int_set (&basesrc->priv->have_events, FALSE);
+ }
+ GST_OBJECT_UNLOCK (basesrc);
+ }
+ }
GST_LIVE_SIGNAL (basesrc);
GST_LIVE_UNLOCK (basesrc);
- /* step 1, unblock clock sync (if any) */
- result = gst_base_src_unlock (basesrc);
+ return TRUE;
+}
- /* step 2, make sure streaming finishes */
- result &= gst_pad_stop_task (pad);
+/* 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;
- /* step 3, clear the unblock condition */
- result &= gst_base_src_unlock_stop (basesrc);
+ bclass = GST_BASE_SRC_GET_CLASS (basesrc);
- return result;
+ /* 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
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);
gst_event_unref (event);
} else {
GST_DEBUG_OBJECT (basesrc, "Deactivating in push mode");
- /* call the unlock function and stop the task */
- if (G_UNLIKELY (!gst_base_src_deactivate (basesrc, pad)))
- goto deactivate_failed;
-
+ /* 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;
seek_failed:
{
GST_ERROR_OBJECT (basesrc, "Failed to perform initial seek");
+ /* flush all */
+ gst_base_src_set_flushing (basesrc, TRUE, FALSE, TRUE, NULL);
+ /* stop the task */
+ gst_pad_stop_task (pad);
+ /* Stop the basesrc */
gst_base_src_stop (basesrc);
if (event)
gst_event_unref (event);
return FALSE;
}
-deactivate_failed:
- {
- GST_ERROR_OBJECT (basesrc, "Failed to deactivate in push mode");
- return FALSE;
- }
error_stop:
{
GST_DEBUG_OBJECT (basesrc, "Failed to stop in push mode");
/* 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");
- /* call the unlock function. We have no task to stop. */
- if (G_UNLIKELY (!gst_base_src_deactivate (basesrc, pad)))
- goto deactivate_failed;
+ /* 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;
gst_base_src_stop (basesrc);
return FALSE;
}
-deactivate_failed:
- {
- GST_ERROR_OBJECT (basesrc, "Failed to deactivate in pull mode");
- return FALSE;
- }
error_stop:
{
GST_ERROR_OBJECT (basesrc, "Failed to stop in pull mode");
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
- GST_LIVE_LOCK (element);
- if (basesrc->is_live) {
- no_preroll = TRUE;
- basesrc->live_running = FALSE;
- }
- basesrc->priv->last_sent_eos = FALSE;
- basesrc->priv->discont = TRUE;
- GST_LIVE_UNLOCK (element);
+ no_preroll = gst_base_src_is_live (basesrc);
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
- GST_LIVE_LOCK (element);
- basesrc->priv->latency = -1;
- if (basesrc->is_live) {
- basesrc->live_running = TRUE;
- GST_LIVE_SIGNAL (element);
+ 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);
}
- GST_LIVE_UNLOCK (element);
break;
default:
break;
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
- GST_LIVE_LOCK (element);
- if (basesrc->is_live) {
+ 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;
- basesrc->live_running = FALSE;
}
- GST_LIVE_UNLOCK (element);
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
{
- GstEvent **event_p;
+ GstEvent **event_p, *event;
+
+ /* 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 preferred 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 ());
+ event = gst_event_new_eos ();
+ gst_event_set_seqnum (event, basesrc->priv->seqnum);
+ gst_pad_push_event (basesrc->srcpad, event);
basesrc->priv->last_sent_eos = TRUE;
}
+ g_atomic_int_set (&basesrc->priv->pending_eos, FALSE);
event_p = &basesrc->data.ABI.pending_seek;
gst_event_replace (event_p, NULL);
event_p = &basesrc->priv->close_segment;
projects / platform / upstream / gstreamer.git / blobdiff