--- /dev/null Thu Jan 01 00:00:00 1970 +0000
+++ b/gstreamer_core/libs/gst/base/gstbasesrc.c Thu Dec 17 08:53:32 2009 +0200
@@ -0,0 +1,2738 @@
+/* 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 (&srctemplate));
+ * // see #GstElementDetails
+ * gst_element_class_set_details (gstelement_class, &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
+
+#ifdef __SYMBIAN32__
+#include <gst_global.h>
+#endif
+#include <stdlib.h>
+#include <string.h>
+
+#include "gstbasesrc.h"
+#include "gsttypefindhelper.h"
+#include <gst/gstmarshal.h>
+#include <gst/gst-i18n-lib.h>
+
+#ifdef __SYMBIAN32__
+#include <glib_global.h>
+#include <gobject_global.h>
+#endif
+GST_DEBUG_CATEGORY_STATIC (gst_base_src_debug);
+#ifndef __SYMBIAN32__
+#define GST_CAT_DEFAULT gst_base_src_debug
+#endif
+
+#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);
+#ifdef __SYMBIAN32__
+EXPORT_C
+#endif
+
+
+
+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.
+ */
+#ifdef __SYMBIAN32__
+EXPORT_C
+#endif
+
+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.
+ */
+#ifdef __SYMBIAN32__
+EXPORT_C
+#endif
+
+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.
+ */
+#ifdef __SYMBIAN32__
+EXPORT_C
+#endif
+
+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
+ */
+#ifdef __SYMBIAN32__
+EXPORT_C
+#endif
+
+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
+ */
+#ifdef __SYMBIAN32__
+EXPORT_C
+#endif
+
+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
+ */
+#ifdef __SYMBIAN32__
+EXPORT_C
+#endif
+
+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
+ */
+#ifdef __SYMBIAN32__
+EXPORT_C
+#endif
+
+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;
+ }
+}