/* GStreamer * Copyright (C) 1999,2000 Erik Walthinsen * 2000,2005 Wim Taymans * * 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: * * random access sources like files * seekable sources * live sources * * * 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. * * #GstBaseSrc always supports push mode scheduling. If the following * conditions are met, it also supports pull mode scheduling: * * The format is set to #GST_FORMAT_BYTES (default). * * #GstBaseSrc::is_seekable returns %TRUE. * * * * 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. * * 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: * * * #GstBaseSrc::is_seekable returns %TRUE. * * * #GstBaseSrc::query can convert all supported seek formats to the * internal format as set with gst_base_src_set_format(). * * * #GstBaseSrc::do_seek is implemented, performs the seek and returns %TRUE. * * * * 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. * * #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. * * 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. * * 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. * * 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. * * 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. * * 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. * * 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_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); * } * * * * Controlled shutdown of live sources in applications * * 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. * * 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) * * */ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include #include #include "gstbasesrc.h" #include "gsttypefindhelper.h" #include #include GST_DEBUG_CATEGORY_STATIC (gst_base_src_debug); #define GST_CAT_DEFAULT gst_base_src_debug #define GST_LIVE_GET_LOCK(elem) (GST_BASE_SRC_CAST(elem)->live_lock) #define GST_LIVE_LOCK(elem) g_mutex_lock(GST_LIVE_GET_LOCK(elem)) #define GST_LIVE_TRYLOCK(elem) g_mutex_trylock(GST_LIVE_GET_LOCK(elem)) #define GST_LIVE_UNLOCK(elem) g_mutex_unlock(GST_LIVE_GET_LOCK(elem)) #define GST_LIVE_GET_COND(elem) (GST_BASE_SRC_CAST(elem)->live_cond) #define GST_LIVE_WAIT(elem) g_cond_wait (GST_LIVE_GET_COND (elem), GST_LIVE_GET_LOCK (elem)) #define GST_LIVE_TIMED_WAIT(elem, timeval) g_cond_timed_wait (GST_LIVE_GET_COND (elem), GST_LIVE_GET_LOCK (elem),\ timeval) #define GST_LIVE_SIGNAL(elem) g_cond_signal (GST_LIVE_GET_COND (elem)); #define GST_LIVE_BROADCAST(elem) g_cond_broadcast (GST_LIVE_GET_COND (elem)); /* BaseSrc signals and args */ enum { /* FILL ME */ LAST_SIGNAL }; #define DEFAULT_BLOCKSIZE 4096 #define DEFAULT_NUM_BUFFERS -1 #define DEFAULT_TYPEFIND FALSE #define DEFAULT_DO_TIMESTAMP FALSE enum { PROP_0, PROP_BLOCKSIZE, PROP_NUM_BUFFERS, PROP_TYPEFIND, PROP_DO_TIMESTAMP }; #define GST_BASE_SRC_GET_PRIVATE(obj) \ (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_BASE_SRC, GstBaseSrcPrivate)) struct _GstBaseSrcPrivate { gboolean last_sent_eos; /* last thing we did was send an EOS (we set this * to avoid the sending of two EOS in some cases) */ gboolean discont; gboolean flushing; /* two segments to be sent in the streaming thread with STREAM_LOCK */ GstEvent *close_segment; GstEvent *start_segment; /* if EOS is pending (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 * 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; /* stream sequence number */ guint32 seqnum; /* pending tags to be pushed in the data stream */ GList *pending_tags; }; static GstElementClass *parent_class = NULL; static void gst_base_src_base_init (gpointer g_class); static void gst_base_src_class_init (GstBaseSrcClass * klass); static void gst_base_src_init (GstBaseSrc * src, gpointer g_class); static void gst_base_src_finalize (GObject * object); GType gst_base_src_get_type (void) { static volatile gsize 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, NULL, (GClassInitFunc) gst_base_src_class_init, NULL, NULL, sizeof (GstBaseSrc), 0, (GInstanceInitFunc) gst_base_src_init, }; _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_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 gboolean gst_base_src_seekable (GstBaseSrc * src); 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_base_src_finalize; gobject_class->set_property = gst_base_src_set_property; gobject_class->get_property = 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 (-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 = 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_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_STATIC_STRINGS)); 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); if (basesrc->priv->pending_tags) { g_list_foreach (basesrc->priv->pending_tags, (GFunc) gst_event_unref, NULL); g_list_free (basesrc->priv->pending_tags); } G_OBJECT_CLASS (parent_class)->finalize (object); } /** * gst_base_src_wait_playing: * @src: the src * * If the #GstBaseSrcClass::create method performs its own synchronisation against * the clock it must unblock when going from PLAYING to the PAUSED state and call * this method before continuing to produce the remaining data. * * This function will block until a state change to PLAYING happens (in which * case this function returns #GST_FLOW_OK) or the processing must be stopped due * to a state change to READY or a FLUSH event (in which case this function * returns #GST_FLOW_WRONG_STATE). * * Since: 0.10.12 * * Returns: #GST_FLOW_OK if @src is PLAYING and processing can * continue. Any other return value should be returned from the create vmethod. */ GstFlowReturn gst_base_src_wait_playing (GstBaseSrc * src) { g_return_val_if_fail (GST_IS_BASE_SRC (src), GST_FLOW_ERROR); /* block until the state changes, or we get a flush, or something */ GST_DEBUG_OBJECT (src, "live source waiting for running state"); GST_LIVE_WAIT (src); if (src->priv->flushing) goto flushing; GST_DEBUG_OBJECT (src, "live source unlocked"); return GST_FLOW_OK; /* ERRORS */ flushing: { GST_DEBUG_OBJECT (src, "we are flushing"); return GST_FLOW_WRONG_STATE; } } /** * gst_base_src_set_live: * @src: base source instance * @live: new live-mode * * If the element listens to a live source, @live should * be set to %TRUE. * * A live source will not produce data in the PAUSED state and * will therefore not be able to participate in the PREROLL phase * of a pipeline. To signal this fact to the application and the * pipeline, the state change return value of the live source will * be GST_STATE_CHANGE_NO_PREROLL. */ void gst_base_src_set_live (GstBaseSrc * src, gboolean live) { g_return_if_fail (GST_IS_BASE_SRC (src)); GST_OBJECT_LOCK (src); src->is_live = live; GST_OBJECT_UNLOCK (src); } /** * gst_base_src_is_live: * @src: base source instance * * Check if an element is in live mode. * * Returns: %TRUE if element is in live mode. */ gboolean gst_base_src_is_live (GstBaseSrc * src) { gboolean result; g_return_val_if_fail (GST_IS_BASE_SRC (src), FALSE); GST_OBJECT_LOCK (src); result = src->is_live; GST_OBJECT_UNLOCK (src); return result; } /** * gst_base_src_set_format: * @src: base source instance * @format: the format to use * * Sets the default format of the source. This will be the format used * for sending NEW_SEGMENT events and for performing seeks. * * If a format of GST_FORMAT_BYTES is set, the element will be able to * operate in pull mode if the #GstBaseSrc::is_seekable returns TRUE. * * Since: 0.10.1 */ void gst_base_src_set_format (GstBaseSrc * src, GstFormat format) { g_return_if_fail (GST_IS_BASE_SRC (src)); gst_segment_init (&src->segment, format); } /** * gst_base_src_query_latency: * @src: the source * @live: if the source is live * @min_latency: the min latency of the source * @max_latency: the max latency of the source * * Query the source for the latency parameters. @live will be TRUE when @src is * configured as a live source. @min_latency will be set to the difference * between the running time and the timestamp of the first buffer. * @max_latency is always the undefined value of -1. * * This function is mostly used by subclasses. * * Returns: TRUE if the query succeeded. * * Since: 0.10.13 */ gboolean gst_base_src_query_latency (GstBaseSrc * src, gboolean * live, GstClockTime * min_latency, GstClockTime * max_latency) { GstClockTime min; g_return_val_if_fail (GST_IS_BASE_SRC (src), FALSE); 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_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 */ 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 */ 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 * * Configure @src to automatically timestamp outgoing buffers based on the * current running_time of the pipeline. This property is mostly useful for live * sources. * * Since: 0.10.15 */ void gst_base_src_set_do_timestamp (GstBaseSrc * src, gboolean timestamp) { g_return_if_fail (GST_IS_BASE_SRC (src)); GST_OBJECT_LOCK (src); src->priv->do_timestamp = timestamp; GST_OBJECT_UNLOCK (src); } /** * gst_base_src_get_do_timestamp: * @src: the source * * Query if @src timestamps outgoing buffers based on the current running_time. * * Returns: %TRUE if the base class will automatically timestamp outgoing buffers. * * Since: 0.10.15 */ gboolean gst_base_src_get_do_timestamp (GstBaseSrc * src) { gboolean res; 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; } 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: { GstFormat format; gst_query_parse_seeking (query, &format, NULL, NULL, NULL); if (format == src->segment.format) { gst_query_set_seeking (query, src->segment.format, gst_base_src_seekable (src), 0, src->segment.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; /* 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: res = FALSE; break; case GST_QUERY_BUFFERING: { GstFormat 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)); 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; } /* 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, src->segment.format, stop, &format, &stop); } else { res = TRUE; } if (res && format != GST_FORMAT_PERCENT && start != -1) res = gst_pad_query_convert (src->srcpad, src->segment.format, start, &format, &start); gst_query_set_buffering_range (query, format, start, stop, estimated); break; } 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; } else { res = FALSE; GST_INFO_OBJECT (src, "Can't do a default seek"); } 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, tres; 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; guint32 seqnum; GstEvent *tevent; 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 last configures segment * start and stop positions */ 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; seqnum = gst_event_get_seqnum (event); } else { flush = FALSE; /* get next seqnum */ seqnum = gst_util_seqnum_next (); } /* send flush start */ 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. */ 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); 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); } 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) { 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, tevent); } 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); gst_event_set_seqnum (src->priv->close_segment, seqnum); } /* 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) { GstMessage *message; message = gst_message_new_segment_start (GST_OBJECT (src), src->segment.format, src->segment.last_stop); gst_message_set_seqnum (message, seqnum); gst_element_post_message (GST_ELEMENT (src), message); } /* 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 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_event_set_seqnum (src->priv->start_segment, seqnum); } 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. */ 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); 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); GST_DEBUG_OBJECT (src, "reveived %s event", GST_EVENT_TYPE_NAME (event)); 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: { 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 immediatly. 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: /* Insert tag in the dataflow */ GST_OBJECT_LOCK (src); src->priv->pending_tags = g_list_append (src->priv->pending_tags, event); GST_OBJECT_UNLOCK (src); event = NULL; result = TRUE; 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) { 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 { GstEvent **event_p; /* 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); /* 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_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 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 (!gst_base_src_seekable (src)) 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: gst_base_src_set_blocksize (src, 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: gst_base_src_set_do_timestamp (src, 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, gst_base_src_get_blocksize (src)); 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, gst_base_src_get_do_timestamp (src)); 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); again: 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--; } /* 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 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; /* set pad caps on the buffer if the buffer had no caps */ if (GST_BUFFER_CAPS (*buf) == NULL) 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. */ 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. In any case, get rid of * the buffer. */ gst_buffer_unref (*buf); *buf = NULL; 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 */ 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"); 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; 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 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; GList *tags, *tmp; eos = FALSE; 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) { 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; 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 */ 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; } GST_OBJECT_LOCK (src); /* take the tags */ tags = src->priv->pending_tags; src->priv->pending_tags = NULL; GST_OBJECT_UNLOCK (src); /* Push out pending tags if any */ if (G_UNLIKELY (tags != NULL)) { for (tmp = tags; tmp; tmp = g_list_next (tmp)) { GstEvent *ev = (GstEvent *) tmp->data; gst_pad_push_event (pad, ev); } g_list_free (tags); } /* 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; } 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) { GstMessage *message; message = gst_message_new_segment_done (GST_OBJECT_CAST (src), src->segment.format, src->segment.last_stop); gst_message_set_seqnum (message, src->priv->seqnum); gst_element_post_message (GST_ELEMENT_CAST (src), message); } else { 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 { 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. */ 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; } 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_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_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); 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, 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; no_nego_needed: { GST_DEBUG_OBJECT (basesrc, "no negotiation needed"); if (thiscaps) 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 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; gboolean seekable; 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); seekable = gst_base_src_seekable (basesrc); GST_DEBUG_OBJECT (basesrc, "is seekable: %d", seekable); /* update for random access flag */ basesrc->random_access = 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; 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 (!(result = gst_base_src_negotiate (basesrc))) goto could_not_negotiate; } return result; /* 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; } 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 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 */ 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; } 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, *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 EOS event to the element */ if (!basesrc->priv->last_sent_eos) { GST_DEBUG_OBJECT (basesrc, "Sending EOS event"); 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; 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; } }