/* 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., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, 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_SEGMENT * 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). * * #GstBaseSrcClass.is_seekable() 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: * * * #GstBaseSrcClass.is_seekable() returns %TRUE. * * * #GstBaseSrcClass.query() can convert all supported seek formats to the * internal format as set with gst_base_src_set_format(). * * * #GstBaseSrcClass.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 #GstBaseSrcClass.create() method should be ignored. * It is recommended to subclass #GstPushSrc instead, in this situation. If the * element can operate in pull mode but only with specific offsets and * lengths, it is allowed to generate an error when the wrong values are passed * to the #GstBaseSrcClass.create() function. * * #GstBaseSrc has support for live sources. Live sources are sources that when * paused discard data, such as audio or video capture devices. A typical live * source also produces data at a fixed rate and thus provides a clock to publish * this rate. * Use gst_base_src_set_live() to activate the live source mode. * * A live source does not produce data in the PAUSED state. This means that the * #GstBaseSrcClass.create() method will not be called in PAUSED but only in * PLAYING. To signal the pipeline that the element will not produce data, the * return value from the READY to PAUSED state will be * #GST_STATE_CHANGE_NO_PREROLL. * * A typical live source will timestamp the buffers it creates with the * current running time of the pipeline. This is one reason why a live source * can only produce data in the PLAYING state, when the clock is actually * distributed and running. * * Live sources that synchronize and block on the clock (an audio source, for * example) can use gst_base_src_wait_playing() when the * #GstBaseSrcClass.create() function was interrupted by a state change to * PAUSED. * * The #GstBaseSrcClass.get_times() method can be used to implement pseudo-live * sources. It only makes sense to implement the #GstBaseSrcClass.get_times() * function if the source is a live source. The #GstBaseSrcClass.get_times() * function should return timestamps starting from 0, as if it were a non-live * source. The base class will make sure that the timestamps are transformed * into the current running_time. The base source will then wait for the * calculated running_time before pushing out the buffer. * * For live sources, the base class will by default report a latency of 0. * For pseudo live sources, the base class will by default measure the difference * between the first buffer timestamp and the start time of get_times and will * report this value as the latency. * 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)); * * gst_element_class_set_static_metadata (gstelement_class, * "Source name", * "Source", * "My Source element", * "The author <my.sink@my.email>"); * } * ]| * * * 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. * * 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. * * Last reviewed on 2007-12-19 (0.10.16) * * */ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include #include #include #include #include "gstbasesrc.h" #include "gsttypefindhelper.h" #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_WAIT_UNTIL(elem, end_time) g_cond_timed_wait (GST_LIVE_GET_COND (elem), GST_LIVE_GET_LOCK (elem), end_time) #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)); #define GST_ASYNC_GET_COND(elem) (&GST_BASE_SRC_CAST(elem)->priv->async_cond) #define GST_ASYNC_WAIT(elem) g_cond_wait (GST_ASYNC_GET_COND (elem), GST_OBJECT_GET_LOCK (elem)) #define GST_ASYNC_SIGNAL(elem) g_cond_signal (GST_ASYNC_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 discont; gboolean flushing; GstFlowReturn start_result; gboolean async; /* if a stream-start event should be sent */ gboolean stream_start_pending; /* if segment should be sent and a * seqnum if it was originated by a seek */ gboolean segment_pending; guint32 segment_seqnum; /* 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; volatile gint dynamic_size; /* stream sequence number */ guint32 seqnum; /* pending events (TAG, CUSTOM_BOTH, CUSTOM_DOWNSTREAM) to be * pushed in the data stream */ GList *pending_events; volatile gint have_events; /* QoS *//* with LOCK */ gboolean qos_enabled; gdouble proportion; GstClockTime earliest_time; GstBufferPool *pool; GstAllocator *allocator; GstAllocationParams params; GCond async_cond; }; static GstElementClass *parent_class = NULL; 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), NULL, 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_default_get_caps (GstBaseSrc * bsrc, GstCaps * filter); static GstCaps *gst_base_src_default_fixate (GstBaseSrc * src, GstCaps * caps); static GstCaps *gst_base_src_fixate (GstBaseSrc * src, GstCaps * caps); static gboolean gst_base_src_is_random_access (GstBaseSrc * src); static gboolean gst_base_src_activate_mode (GstPad * pad, GstObject * parent, GstPadMode mode, 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 (GstPad * pad, GstObject * parent, 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 gboolean gst_base_src_query (GstPad * pad, GstObject * parent, GstQuery * query); static gboolean gst_base_src_activate_pool (GstBaseSrc * basesrc, gboolean active); 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 GstFlowReturn gst_base_src_default_create (GstBaseSrc * basesrc, guint64 offset, guint size, GstBuffer ** buf); static GstFlowReturn gst_base_src_default_alloc (GstBaseSrc * basesrc, guint64 offset, guint size, GstBuffer ** buf); static gboolean gst_base_src_decide_allocation_default (GstBaseSrc * basesrc, GstQuery * query); static gboolean gst_base_src_set_flushing (GstBaseSrc * basesrc, gboolean flushing, gboolean live_play, 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 GstFlowReturn gst_base_src_getrange (GstPad * pad, GstObject * parent, 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 gboolean gst_base_src_negotiate (GstBaseSrc * basesrc); static gboolean gst_base_src_update_length (GstBaseSrc * src, guint64 offset, guint * length, gboolean force); 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); GST_DEBUG_CATEGORY_INIT (gst_base_src_debug, "basesrc", 0, "basesrc element"); 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_uint ("blocksize", "Block size", "Size in bytes to read per buffer (-1 = default)", 0, G_MAXUINT, 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); klass->get_caps = GST_DEBUG_FUNCPTR (gst_base_src_default_get_caps); klass->negotiate = GST_DEBUG_FUNCPTR (gst_base_src_default_negotiate); klass->fixate = GST_DEBUG_FUNCPTR (gst_base_src_default_fixate); klass->prepare_seek_segment = GST_DEBUG_FUNCPTR (gst_base_src_default_prepare_seek_segment); klass->do_seek = GST_DEBUG_FUNCPTR (gst_base_src_default_do_seek); klass->query = GST_DEBUG_FUNCPTR (gst_base_src_default_query); klass->event = GST_DEBUG_FUNCPTR (gst_base_src_default_event); klass->create = GST_DEBUG_FUNCPTR (gst_base_src_default_create); klass->alloc = GST_DEBUG_FUNCPTR (gst_base_src_default_alloc); klass->decide_allocation = GST_DEBUG_FUNCPTR (gst_base_src_decide_allocation_default); /* Registering debug symbols for function pointers */ GST_DEBUG_REGISTER_FUNCPTR (gst_base_src_activate_mode); GST_DEBUG_REGISTER_FUNCPTR (gst_base_src_event); GST_DEBUG_REGISTER_FUNCPTR (gst_base_src_query); GST_DEBUG_REGISTER_FUNCPTR (gst_base_src_getrange); GST_DEBUG_REGISTER_FUNCPTR (gst_base_src_fixate); } 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; g_mutex_init (&basesrc->live_lock); g_cond_init (&basesrc->live_cond); basesrc->num_buffers = DEFAULT_NUM_BUFFERS; basesrc->num_buffers_left = -1; basesrc->can_activate_push = TRUE; 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_activatemode_function (pad, gst_base_src_activate_mode); gst_pad_set_event_function (pad, gst_base_src_event); gst_pad_set_query_function (pad, gst_base_src_query); gst_pad_set_getrange_function (pad, gst_base_src_getrange); /* 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->typefind = DEFAULT_TYPEFIND; basesrc->priv->do_timestamp = DEFAULT_DO_TIMESTAMP; g_atomic_int_set (&basesrc->priv->have_events, FALSE); g_cond_init (&basesrc->priv->async_cond); basesrc->priv->start_result = GST_FLOW_FLUSHING; GST_OBJECT_FLAG_UNSET (basesrc, GST_BASE_SRC_FLAG_STARTED); GST_OBJECT_FLAG_UNSET (basesrc, GST_BASE_SRC_FLAG_STARTING); GST_OBJECT_FLAG_SET (basesrc, GST_ELEMENT_FLAG_SOURCE); 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_clear (&basesrc->live_lock); g_cond_clear (&basesrc->live_cond); g_cond_clear (&basesrc->priv->async_cond); event_p = &basesrc->pending_seek; gst_event_replace (event_p, NULL); if (basesrc->priv->pending_events) { g_list_foreach (basesrc->priv->pending_events, (GFunc) gst_event_unref, NULL); g_list_free (basesrc->priv->pending_events); } G_OBJECT_CLASS (parent_class)->finalize (object); } /** * gst_base_src_wait_playing: * @src: the src * * If the #GstBaseSrcClass.create() method performs its own synchronisation * against the clock it must unblock when going from PLAYING to the PAUSED state * and call this method before continuing to produce the remaining data. * * This function will block until a state change to PLAYING happens (in which * case this function returns #GST_FLOW_OK) or the processing must be stopped due * to a state change to READY or a FLUSH event (in which case this function * returns #GST_FLOW_FLUSHING). * * 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); do { /* block until the state changes, or we get a flush, or something */ GST_DEBUG_OBJECT (src, "live source waiting for running state"); GST_LIVE_WAIT (src); GST_DEBUG_OBJECT (src, "live source unlocked"); if (src->priv->flushing) goto flushing; } while (G_UNLIKELY (!src->live_running)); return GST_FLOW_OK; /* ERRORS */ flushing: { GST_DEBUG_OBJECT (src, "we are flushing"); return GST_FLOW_FLUSHING; } } /** * 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 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 #GstBaseSrcClass.is_seekable() returns TRUE. * * This function must only be called in states < %GST_STATE_PAUSED. */ void gst_base_src_set_format (GstBaseSrc * src, GstFormat format) { g_return_if_fail (GST_IS_BASE_SRC (src)); g_return_if_fail (GST_STATE (src) <= GST_STATE_READY); GST_OBJECT_LOCK (src); gst_segment_init (&src->segment, format); GST_OBJECT_UNLOCK (src); } /** * gst_base_src_set_dynamic_size: * @src: base source instance * @dynamic: new dynamic size mode * * If not @dynamic, size is only updated when needed, such as when trying to * read past current tracked size. Otherwise, size is checked for upon each * read. */ void gst_base_src_set_dynamic_size (GstBaseSrc * src, gboolean dynamic) { g_return_if_fail (GST_IS_BASE_SRC (src)); g_atomic_int_set (&src->priv->dynamic_size, dynamic); } /** * gst_base_src_set_async: * @src: base source instance * @async: new async mode * * Configure async behaviour in @src, no state change will block. The open, * close, start, stop, play and pause virtual methods will be executed in a * different thread and are thus allowed to perform blocking operations. Any * blocking operation should be unblocked with the unlock vmethod. */ void gst_base_src_set_async (GstBaseSrc * src, gboolean async) { g_return_if_fail (GST_IS_BASE_SRC (src)); GST_OBJECT_LOCK (src); src->priv->async = async; GST_OBJECT_UNLOCK (src); } /** * gst_base_src_is_async: * @src: base source instance * * Get the current async behaviour of @src. See also gst_base_src_set_async(). * * Returns: %TRUE if @src is operating in async mode. */ gboolean gst_base_src_is_async (GstBaseSrc * src) { gboolean res; g_return_val_if_fail (GST_IS_BASE_SRC (src), FALSE); GST_OBJECT_LOCK (src); res = src->priv->async; GST_OBJECT_UNLOCK (src); return res; } /** * gst_base_src_query_latency: * @src: the source * @live: (out) (allow-none): if the source is live * @min_latency: (out) (allow-none): the min latency of the source * @max_latency: (out) (allow-none): the max latency of the source * * Query the source for the latency parameters. @live will be TRUE when @src is * configured as a live source. @min_latency will be set to the difference * between the running time and the timestamp of the first buffer. * @max_latency is always the undefined value of -1. * * This function is mostly used by subclasses. * * Returns: TRUE if the query succeeded. */ 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. */ void gst_base_src_set_blocksize (GstBaseSrc * src, guint 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. */ guint gst_base_src_get_blocksize (GstBaseSrc * src) { gint 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. */ 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. */ 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; } /** * gst_base_src_new_seamless_segment: * @src: The source * @start: The new start value for the segment * @stop: Stop value for the new segment * @time: The new time value for the start of the new segent * * Prepare a new seamless segment for emission downstream. This function must * only be called by derived sub-classes, and only from the create() function, * as the stream-lock needs to be held. * * The format for the new segment will be the current format of the source, as * configured with gst_base_src_set_format() * * Returns: %TRUE if preparation of the seamless segment succeeded. */ gboolean gst_base_src_new_seamless_segment (GstBaseSrc * src, gint64 start, gint64 stop, gint64 time) { gboolean res = TRUE; GST_OBJECT_LOCK (src); src->segment.base = gst_segment_to_running_time (&src->segment, src->segment.format, src->segment.position); src->segment.position = src->segment.start = start; src->segment.stop = stop; src->segment.time = time; /* Mark pending segment. Will be sent before next data */ src->priv->segment_pending = TRUE; src->priv->segment_seqnum = gst_util_seqnum_next (); GST_DEBUG_OBJECT (src, "Starting new seamless segment. Start %" GST_TIME_FORMAT " stop %" GST_TIME_FORMAT " time %" GST_TIME_FORMAT " base %" GST_TIME_FORMAT, GST_TIME_ARGS (start), GST_TIME_ARGS (stop), GST_TIME_ARGS (time), GST_TIME_ARGS (src->segment.base)); GST_OBJECT_UNLOCK (src); src->priv->discont = TRUE; src->running = TRUE; return res; } static gboolean gst_base_src_send_stream_start (GstBaseSrc * src) { gboolean ret = TRUE; if (src->priv->stream_start_pending) { gchar *stream_id; GstEvent *event; stream_id = gst_pad_create_stream_id (src->srcpad, GST_ELEMENT_CAST (src), NULL); GST_DEBUG_OBJECT (src, "Pushing STREAM_START"); event = gst_event_new_stream_start (stream_id); gst_event_set_group_id (event, gst_util_group_id_next ()); ret = gst_pad_push_event (src->srcpad, event); src->priv->stream_start_pending = FALSE; g_free (stream_id); } return ret; } /** * gst_base_src_set_caps: * @src: a #GstBaseSrc * @caps: a #GstCaps * * Set new caps on the basesrc source pad. * * Returns: %TRUE if the caps could be set */ gboolean gst_base_src_set_caps (GstBaseSrc * src, GstCaps * caps) { GstBaseSrcClass *bclass; gboolean res = TRUE; bclass = GST_BASE_SRC_GET_CLASS (src); gst_base_src_send_stream_start (src); if (bclass->set_caps) res = bclass->set_caps (src, caps); if (res) res = gst_pad_push_event (src->srcpad, gst_event_new_caps (caps)); return res; } static GstCaps * gst_base_src_default_get_caps (GstBaseSrc * bsrc, GstCaps * filter) { GstCaps *caps = NULL; GstPadTemplate *pad_template; GstBaseSrcClass *bclass; bclass = GST_BASE_SRC_GET_CLASS (bsrc); pad_template = gst_element_class_get_pad_template (GST_ELEMENT_CLASS (bclass), "src"); if (pad_template != NULL) { caps = gst_pad_template_get_caps (pad_template); if (filter) { GstCaps *intersection; intersection = gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST); gst_caps_unref (caps); caps = intersection; } } return caps; } static GstCaps * gst_base_src_default_fixate (GstBaseSrc * bsrc, GstCaps * caps) { GST_DEBUG_OBJECT (bsrc, "using default caps fixate function"); return gst_caps_fixate (caps); } static GstCaps * gst_base_src_fixate (GstBaseSrc * bsrc, GstCaps * caps) { GstBaseSrcClass *bclass; bclass = GST_BASE_SRC_GET_CLASS (bsrc); if (bclass->fixate) caps = bclass->fixate (bsrc, caps); return caps; } 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); GST_DEBUG_OBJECT (src, "position query in format %s", gst_format_get_name (format)); switch (format) { case GST_FORMAT_PERCENT: { gint64 percent; gint64 position; gint64 duration; GST_OBJECT_LOCK (src); position = src->segment.position; duration = src->segment.duration; GST_OBJECT_UNLOCK (src); 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; GstFormat seg_format; GST_OBJECT_LOCK (src); position = gst_segment_to_stream_time (&src->segment, src->segment.format, src->segment.position); seg_format = src->segment.format; GST_OBJECT_UNLOCK (src); if (position != -1) { /* convert to requested format */ res = gst_pad_query_convert (src->srcpad, seg_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; GstFormat seg_format; guint length = 0; /* may have to refresh duration */ gst_base_src_update_length (src, 0, &length, g_atomic_int_get (&src->priv->dynamic_size)); /* this is the duration as configured by the subclass. */ GST_OBJECT_LOCK (src); duration = src->segment.duration; seg_format = src->segment.format; GST_OBJECT_UNLOCK (src); GST_LOG_OBJECT (src, "duration %" G_GINT64_FORMAT ", format %s", duration, gst_format_get_name (seg_format)); if (duration != -1) { /* convert to requested format, if this fails, we have a duration * but we cannot answer the query, we must return FALSE. */ res = gst_pad_query_convert (src->srcpad, seg_format, duration, format, &duration); } else { /* The subclass did not configure a duration, we assume that the * media has an unknown duration then and we return TRUE to report * this. Note that this is not the same as returning FALSE, which * means that we cannot report the duration at all. */ res = TRUE; } gst_query_set_duration (query, format, duration); break; } } break; } case GST_QUERY_SEEKING: { GstFormat format, seg_format; gint64 duration; GST_OBJECT_LOCK (src); duration = src->segment.duration; seg_format = src->segment.format; GST_OBJECT_UNLOCK (src); gst_query_parse_seeking (query, &format, NULL, NULL, NULL); if (format == seg_format) { gst_query_set_seeking (query, seg_format, gst_base_src_seekable (src), 0, duration); res = TRUE; } else { /* FIXME 0.11: return TRUE + seekable=FALSE for SEEKING query here */ /* Don't reply to the query to make up for demuxers which don't * handle the SEEKING query yet. Players like Totem will fall back * to the duration when the SEEKING query isn't answered. */ res = FALSE; } break; } case GST_QUERY_SEGMENT: { GstFormat format; gint64 start, stop; GST_OBJECT_LOCK (src); format = src->segment.format; start = gst_segment_to_stream_time (&src->segment, format, src->segment.start); if ((stop = src->segment.stop) == -1) stop = src->segment.duration; else stop = gst_segment_to_stream_time (&src->segment, format, stop); gst_query_set_segment (query, src->segment.rate, format, start, stop); GST_OBJECT_UNLOCK (src); 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 useful */ 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, seg_format; gint64 start, stop, estimated; gst_query_parse_buffering_range (query, &format, NULL, NULL, NULL); GST_DEBUG_OBJECT (src, "buffering query in format %s", gst_format_get_name (format)); GST_OBJECT_LOCK (src); if (src->random_access) { estimated = 0; start = 0; if (format == GST_FORMAT_PERCENT) stop = GST_FORMAT_PERCENT_MAX; else stop = src->segment.duration; } else { estimated = -1; start = -1; stop = -1; } seg_format = src->segment.format; GST_OBJECT_UNLOCK (src); /* convert to required format. When the conversion fails, we can't answer * the query. When the value is unknown, we can don't perform conversion * but report TRUE. */ if (format != GST_FORMAT_PERCENT && stop != -1) { res = gst_pad_query_convert (src->srcpad, seg_format, stop, format, &stop); } else { res = TRUE; } if (res && format != GST_FORMAT_PERCENT && start != -1) res = gst_pad_query_convert (src->srcpad, seg_format, start, format, &start); gst_query_set_buffering_range (query, format, start, stop, estimated); break; } case GST_QUERY_SCHEDULING: { gboolean random_access; random_access = gst_base_src_is_random_access (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. */ gst_query_set_scheduling (query, GST_SCHEDULING_FLAG_SEEKABLE, 1, -1, 0); if (random_access) gst_query_add_scheduling_mode (query, GST_PAD_MODE_PULL); gst_query_add_scheduling_mode (query, GST_PAD_MODE_PUSH); res = TRUE; break; } case GST_QUERY_CAPS: { GstBaseSrcClass *bclass; GstCaps *caps, *filter; bclass = GST_BASE_SRC_GET_CLASS (src); if (bclass->get_caps) { gst_query_parse_caps (query, &filter); if ((caps = bclass->get_caps (src, filter))) { gst_query_set_caps_result (query, caps); gst_caps_unref (caps); res = TRUE; } else { res = FALSE; } } else res = FALSE; break; } case GST_QUERY_URI:{ if (GST_IS_URI_HANDLER (src)) { gchar *uri = gst_uri_handler_get_uri (GST_URI_HANDLER (src)); if (uri != NULL) { gst_query_set_uri (query, uri); g_free (uri); res = TRUE; } else { res = FALSE; } } else { res = FALSE; } 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, GstObject * parent, GstQuery * query) { GstBaseSrc *src; GstBaseSrcClass *bclass; gboolean result = FALSE; src = GST_BASE_SRC (parent); bclass = GST_BASE_SRC_GET_CLASS (src); if (bclass->query) result = bclass->query (src, query); 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); GST_INFO_OBJECT (src, "seeking: %" GST_SEGMENT_FORMAT, segment); 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 start_type, stop_type; gint64 start, stop; GstSeekFlags flags; GstFormat seek_format, dest_format; gdouble rate; gboolean update; gboolean res = TRUE; gst_event_parse_seek (event, &rate, &seek_format, &flags, &start_type, &start, &stop_type, &stop); dest_format = segment->format; if (seek_format == dest_format) { gst_segment_do_seek (segment, rate, seek_format, flags, start_type, start, stop_type, stop, &update); return TRUE; } if (start_type != GST_SEEK_TYPE_NONE) { /* FIXME: Handle seek_end by converting the input segment vals */ res = gst_pad_query_convert (src->srcpad, seek_format, start, dest_format, &start); start_type = GST_SEEK_TYPE_SET; } if (res && stop_type != GST_SEEK_TYPE_NONE) { /* FIXME: Handle 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_do_seek (segment, rate, dest_format, flags, start_type, start, 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; } static GstFlowReturn gst_base_src_default_alloc (GstBaseSrc * src, guint64 offset, guint size, GstBuffer ** buffer) { GstFlowReturn ret; GstBaseSrcPrivate *priv = src->priv; if (priv->pool) { ret = gst_buffer_pool_acquire_buffer (priv->pool, buffer, NULL); } else if (size != -1) { *buffer = gst_buffer_new_allocate (priv->allocator, size, &priv->params); if (G_UNLIKELY (*buffer == NULL)) goto alloc_failed; ret = GST_FLOW_OK; } else { GST_WARNING_OBJECT (src, "Not trying to alloc %u bytes. Blocksize not set?", size); goto alloc_failed; } return ret; /* ERRORS */ alloc_failed: { GST_ERROR_OBJECT (src, "Failed to allocate %u bytes", size); return GST_FLOW_ERROR; } } static GstFlowReturn gst_base_src_default_create (GstBaseSrc * src, guint64 offset, guint size, GstBuffer ** buffer) { GstBaseSrcClass *bclass; GstFlowReturn ret; GstBuffer *res_buf; bclass = GST_BASE_SRC_GET_CLASS (src); if (G_UNLIKELY (!bclass->alloc)) goto no_function; if (G_UNLIKELY (!bclass->fill)) goto no_function; if (*buffer == NULL) { /* downstream did not provide us with a buffer to fill, allocate one * ourselves */ ret = bclass->alloc (src, offset, size, &res_buf); if (G_UNLIKELY (ret != GST_FLOW_OK)) goto alloc_failed; } else { res_buf = *buffer; } if (G_LIKELY (size > 0)) { /* only call fill when there is a size */ ret = bclass->fill (src, offset, size, res_buf); if (G_UNLIKELY (ret != GST_FLOW_OK)) goto not_ok; } *buffer = res_buf; return GST_FLOW_OK; /* ERRORS */ no_function: { GST_DEBUG_OBJECT (src, "no fill or alloc function"); return GST_FLOW_NOT_SUPPORTED; } alloc_failed: { GST_DEBUG_OBJECT (src, "Failed to allocate buffer of %u bytes", size); return ret; } not_ok: { GST_DEBUG_OBJECT (src, "fill returned %d (%s)", ret, gst_flow_get_name (ret)); if (*buffer == NULL) gst_buffer_unref (res_buf); return ret; } } /* 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 FLUSHING 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. */ 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 start_type, stop_type; gint64 start, 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: %" GST_PTR_FORMAT, event); GST_OBJECT_LOCK (src); dest_format = src->segment.format; GST_OBJECT_UNLOCK (src); if (event) { gst_event_parse_seek (event, &rate, &seek_format, &flags, &start_type, &start, &stop_type, &stop); relative_seek = SEEK_TYPE_IS_RELATIVE (start_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. */ if (unlock) gst_base_src_set_flushing (src, TRUE, FALSE, &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); } if (unlock) gst_base_src_set_flushing (src, FALSE, playing, 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 succeeds. */ if (!seekseg_configured) { memcpy (&seeksegment, &src->segment, sizeof (GstSegment)); /* now configure the final seek segment */ if (event) { if (seeksegment.format != seek_format) { /* OK, here's where we give the subclass a chance to convert the relative * seek into an absolute one in the processing format. We set up any * absolute seek above, before taking the stream lock. */ 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_do_seek (&seeksegment, rate, seek_format, flags, start_type, start, 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.position); /* do the seek, segment.position 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 (TRUE); 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); } /* 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 the seek was successful, we update our real segment and push * out the new segment. */ if (res) { GST_OBJECT_LOCK (src); memcpy (&src->segment, &seeksegment, sizeof (GstSegment)); GST_OBJECT_UNLOCK (src); if (seeksegment.flags & GST_SEGMENT_FLAG_SEGMENT) { GstMessage *message; message = gst_message_new_segment_start (GST_OBJECT (src), seeksegment.format, seeksegment.position); 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 */ /* FIXME: This is never used below */ if ((stop = seeksegment.stop) == -1) stop = seeksegment.duration; src->priv->segment_pending = TRUE; src->priv->segment_seqnum = seqnum; } src->priv->discont = TRUE; src->running = TRUE; /* and restart the task in case it got paused explicitly or by * the FLUSH_START event we pushed out. */ tres = gst_pad_start_task (src->srcpad, (GstTaskFunction) gst_base_src_loop, src->srcpad, NULL); 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; } /* 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; GstBaseSrcClass *bclass; src = GST_BASE_SRC (element); bclass = GST_BASE_SRC_GET_CLASS (src); GST_DEBUG_OBJECT (src, "handling event %p %" GST_PTR_FORMAT, event, event); switch (GST_EVENT_TYPE (event)) { /* bidirectional events */ case GST_EVENT_FLUSH_START: GST_DEBUG_OBJECT (src, "pushing flush-start event downstream"); result = gst_pad_push_event (src->srcpad, event); /* also unblock the create function */ gst_base_src_activate_pool (src, FALSE); /* 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 (src); /* the live lock is released when we are blocked, waiting for playing or * when we sync to the clock. */ GST_LIVE_LOCK (src); src->priv->flushing = TRUE; /* clear pending EOS if any */ g_atomic_int_set (&src->priv->pending_eos, FALSE); if (bclass->unlock_stop) bclass->unlock_stop (src); if (src->clock_id) gst_clock_id_unschedule (src->clock_id); GST_DEBUG_OBJECT (src, "signal"); GST_LIVE_SIGNAL (src); GST_LIVE_UNLOCK (src); event = NULL; break; case GST_EVENT_FLUSH_STOP: { gboolean start; GST_LIVE_LOCK (src); src->priv->segment_pending = TRUE; src->priv->flushing = FALSE; GST_DEBUG_OBJECT (src, "pushing flush-stop event downstream"); result = gst_pad_push_event (src->srcpad, event); gst_base_src_activate_pool (src, TRUE); GST_OBJECT_LOCK (src->srcpad); start = (GST_PAD_MODE (src->srcpad) == GST_PAD_MODE_PUSH); GST_OBJECT_UNLOCK (src->srcpad); if (start) gst_pad_start_task (src->srcpad, (GstTaskFunction) gst_base_src_loop, src->srcpad, NULL); GST_LIVE_UNLOCK (src); event = NULL; break; } /* downstream serialized events */ case GST_EVENT_EOS: { /* 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 FLUSHING immediately. We then * check the EOS flag and do the EOS logic. */ g_atomic_int_set (&src->priv->pending_eos, TRUE); GST_DEBUG_OBJECT (src, "EOS marked, calling unlock"); /* unlock the _create function so that we can check the pending_eos flag * and we can do EOS. This will eventually release the LIVE_LOCK again so * that we can grab it and stop the unlock again. We don't take the stream * lock so that this operation is guaranteed to never block. */ gst_base_src_activate_pool (src, FALSE); 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_base_src_activate_pool (src, TRUE); GST_LIVE_UNLOCK (src); result = TRUE; break; } case GST_EVENT_SEGMENT: /* sending random SEGMENT downstream can break sync. */ break; case GST_EVENT_TAG: case GST_EVENT_CUSTOM_DOWNSTREAM: case GST_EVENT_CUSTOM_BOTH: /* Insert TAG, CUSTOM_DOWNSTREAM, CUSTOM_BOTH in the dataflow */ GST_OBJECT_LOCK (src); src->priv->pending_events = g_list_append (src->priv->pending_events, event); g_atomic_int_set (&src->priv->have_events, TRUE); GST_OBJECT_UNLOCK (src); event = NULL; result = TRUE; break; case GST_EVENT_BUFFERSIZE: /* does not seem to make much sense currently */ 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_MODE (src->srcpad) == GST_PAD_MODE_PULL) goto wrong_mode; started = GST_PAD_MODE (src->srcpad) == GST_PAD_MODE_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. */ 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->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_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 void gst_base_src_update_qos (GstBaseSrc * src, gdouble proportion, GstClockTimeDiff diff, GstClockTime timestamp) { GST_CAT_DEBUG_OBJECT (GST_CAT_QOS, src, "qos: proportion: %lf, diff %" G_GINT64_FORMAT ", timestamp %" GST_TIME_FORMAT, proportion, diff, GST_TIME_ARGS (timestamp)); GST_OBJECT_LOCK (src); src->priv->proportion = proportion; src->priv->earliest_time = timestamp + diff; GST_OBJECT_UNLOCK (src); } static gboolean gst_base_src_default_event (GstBaseSrc * src, GstEvent * event) { gboolean result; GST_DEBUG_OBJECT (src, "handle event %" GST_PTR_FORMAT, event); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_SEEK: /* is normally called when in push mode */ if (!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, NULL); break; case GST_EVENT_FLUSH_STOP: result = gst_base_src_set_flushing (src, FALSE, TRUE, NULL); break; case GST_EVENT_QOS: { gdouble proportion; GstClockTimeDiff diff; GstClockTime timestamp; gst_event_parse_qos (event, NULL, &proportion, &diff, ×tamp); gst_base_src_update_qos (src, proportion, diff, timestamp); result = TRUE; break; } case GST_EVENT_RECONFIGURE: result = TRUE; break; case GST_EVENT_LATENCY: result = TRUE; break; default: result = FALSE; break; } return result; /* ERRORS */ not_seekable: { GST_DEBUG_OBJECT (src, "is not seekable"); return FALSE; } } static gboolean gst_base_src_event (GstPad * pad, GstObject * parent, GstEvent * event) { GstBaseSrc *src; GstBaseSrcClass *bclass; gboolean result = FALSE; src = GST_BASE_SRC (parent); bclass = GST_BASE_SRC_GET_CLASS (src); if (bclass->event) { if (!(result = bclass->event (src, event))) goto subclass_failed; } done: gst_event_unref (event); 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_uint (value)); break; case PROP_NUM_BUFFERS: src->num_buffers = g_value_get_int (value); break; case PROP_TYPEFIND: src->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_uint (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->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, pts, dts, timestamp; gboolean do_timestamp, first, pseudo_live, is_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 */ dts = GST_BUFFER_DTS (buffer); pts = GST_BUFFER_PTS (buffer); if (GST_CLOCK_TIME_IS_VALID (dts)) timestamp = dts; else timestamp = pts; /* grab the lock to prepare for clocking and calculate the startup * latency. */ GST_OBJECT_LOCK (basesrc); is_live = basesrc->is_live; /* if we are asked to sync against the clock we are a pseudo live element */ pseudo_live = (start != -1 && 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", 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; else gst_object_ref (clock); base_time = GST_ELEMENT_CAST (basesrc)->base_time; do_timestamp = basesrc->priv->do_timestamp; GST_OBJECT_UNLOCK (basesrc); /* 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 PTS: %" GST_TIME_FORMAT ", DTS %" GST_TIME_FORMAT ", running_time %" GST_TIME_FORMAT, GST_TIME_ARGS (pts), GST_TIME_ARGS (dts), 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 (dts)) { if (do_timestamp) { dts = running_time; } else { dts = 0; } GST_BUFFER_DTS (buffer) = dts; GST_LOG_OBJECT (basesrc, "created DTS %" GST_TIME_FORMAT, GST_TIME_ARGS (dts)); } } else { /* not the first buffer, the timestamp is the diff between the clock and * base_time */ if (do_timestamp && !GST_CLOCK_TIME_IS_VALID (dts)) { now = gst_clock_get_time (clock); dts = now - base_time; GST_BUFFER_DTS (buffer) = dts; GST_LOG_OBJECT (basesrc, "created DTS %" GST_TIME_FORMAT, GST_TIME_ARGS (dts)); } } if (!GST_CLOCK_TIME_IS_VALID (pts)) { if (!GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DELTA_UNIT)) pts = dts; GST_BUFFER_PTS (buffer) = dts; GST_LOG_OBJECT (basesrc, "created PTS %" GST_TIME_FORMAT, GST_TIME_ARGS (pts)); } /* if we don't have a buffer timestamp, we don't sync */ if (!GST_CLOCK_TIME_IS_VALID (start)) goto no_sync; if (is_live) { /* for pseudo live sources, add our ts_offset to the timestamp */ if (GST_CLOCK_TIME_IS_VALID (pts)) GST_BUFFER_PTS (buffer) += basesrc->priv->ts_offset; if (GST_CLOCK_TIME_IS_VALID (dts)) GST_BUFFER_DTS (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)); result = gst_base_src_wait (basesrc, clock, start + base_time); gst_object_unref (clock); 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_unref (clock); return GST_CLOCK_OK; } } /* Called with STREAM_LOCK and LIVE_LOCK */ static gboolean gst_base_src_update_length (GstBaseSrc * src, guint64 offset, guint * length, gboolean force) { guint64 size, maxsize; GstBaseSrcClass *bclass; GstFormat format; gint64 stop; bclass = GST_BASE_SRC_GET_CLASS (src); format = src->segment.format; stop = src->segment.stop; /* get total file size */ size = src->segment.duration; /* only operate if we are working with bytes */ if (format != GST_FORMAT_BYTES) return TRUE; /* the max amount of bytes to read is the total size or * up to the segment.stop if present. */ if (stop != -1) maxsize = MIN (size, stop); else maxsize = size; GST_DEBUG_OBJECT (src, "reading offset %" G_GUINT64_FORMAT ", length %u, size %" G_GINT64_FORMAT ", segment.stop %" G_GINT64_FORMAT ", maxsize %" G_GINT64_FORMAT, offset, *length, size, 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 || force)) { /* see if length of the file changed */ if (bclass->get_size) if (!bclass->get_size (src, &size)) size = -1; /* make sure we don't exceed the configured segment stop * if it was set */ if (stop != -1) maxsize = MIN (size, 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 duration. * segment is in bytes, we checked that above. */ GST_OBJECT_LOCK (src); src->segment.duration = size; GST_OBJECT_UNLOCK (src); 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; GstBuffer *res_buf; GstBuffer *in_buf; bclass = GST_BASE_SRC_GET_CLASS (src); again: if (src->is_live) { if (G_UNLIKELY (!src->live_running)) { ret = gst_base_src_wait_playing (src); if (ret != GST_FLOW_OK) goto stopped; } } if (G_UNLIKELY (!GST_BASE_SRC_IS_STARTED (src) && !GST_BASE_SRC_IS_STARTING (src))) goto not_started; if (G_UNLIKELY (!bclass->create)) goto no_function; if (G_UNLIKELY (!gst_base_src_update_length (src, offset, &length, FALSE))) goto unexpected_length; /* track position */ GST_OBJECT_LOCK (src); if (src->segment.format == GST_FORMAT_BYTES) src->segment.position = offset; GST_OBJECT_UNLOCK (src); /* normally we don't count buffers */ if (G_UNLIKELY (src->num_buffers_left >= 0)) { if (src->num_buffers_left == 0) 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); res_buf = in_buf = *buf; ret = bclass->create (src, offset, length, &res_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) { if (*buf == NULL) gst_buffer_unref (res_buf); } goto eos; } if (G_UNLIKELY (ret != GST_FLOW_OK)) goto not_ok; /* fallback in case the create function didn't fill a provided buffer */ if (in_buf != NULL && res_buf != in_buf) { GstMapInfo info; gsize copied_size; GST_CAT_DEBUG_OBJECT (GST_CAT_PERFORMANCE, src, "create function didn't " "fill the provided buffer, copying"); if (!gst_buffer_map (in_buf, &info, GST_MAP_WRITE)) goto map_failed; copied_size = gst_buffer_extract (res_buf, 0, info.data, info.size); gst_buffer_unmap (in_buf, &info); gst_buffer_set_size (in_buf, copied_size); gst_buffer_copy_into (in_buf, res_buf, GST_BUFFER_COPY_METADATA, 0, -1); gst_buffer_unref (res_buf); res_buf = in_buf; } /* no timestamp set and we are at offset 0, we can timestamp with 0 */ if (offset == 0 && src->segment.time == 0 && GST_BUFFER_DTS (res_buf) == -1 && !src->is_live) { GST_DEBUG_OBJECT (src, "setting first timestamp to 0"); res_buf = gst_buffer_make_writable (res_buf); GST_BUFFER_DTS (res_buf) = 0; } /* now sync before pushing the buffer */ status = gst_base_src_do_sync (src, res_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. */ if (*buf == NULL) gst_buffer_unref (res_buf); if (!src->live_running) { /* We return FLUSHING 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 FLUSHING", status); ret = GST_FLOW_FLUSHING; } 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)); if (*buf == NULL) gst_buffer_unref (res_buf); ret = GST_FLOW_ERROR; break; } if (G_LIKELY (ret == GST_FLOW_OK)) *buf = res_buf; 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; } map_failed: { GST_ELEMENT_ERROR (src, RESOURCE, BUSY, (_("Failed to map buffer.")), ("failed to map result buffer in WRITE mode")); if (*buf == NULL) gst_buffer_unref (res_buf); return GST_FLOW_ERROR; } not_started: { GST_DEBUG_OBJECT (src, "getrange but not started"); return GST_FLOW_FLUSHING; } no_function: { GST_DEBUG_OBJECT (src, "no create function"); return GST_FLOW_NOT_SUPPORTED; } 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_EOS; } reached_num_buffers: { GST_DEBUG_OBJECT (src, "sent all buffers"); return GST_FLOW_EOS; } flushing: { GST_DEBUG_OBJECT (src, "we are flushing"); if (*buf == NULL) gst_buffer_unref (res_buf); return GST_FLOW_FLUSHING; } eos: { GST_DEBUG_OBJECT (src, "we are EOS"); return GST_FLOW_EOS; } } static GstFlowReturn gst_base_src_getrange (GstPad * pad, GstObject * parent, guint64 offset, guint length, GstBuffer ** buf) { GstBaseSrc *src; GstFlowReturn res; src = GST_BASE_SRC_CAST (parent); 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); return res; /* ERRORS */ flushing: { GST_DEBUG_OBJECT (src, "we are flushing"); res = GST_FLOW_FLUSHING; goto done; } } static gboolean gst_base_src_is_random_access (GstBaseSrc * src) { /* we need to start the basesrc to check random access */ if (!GST_BASE_SRC_IS_STARTED (src)) { GST_LOG_OBJECT (src, "doing start/stop to check get_range support"); if (G_LIKELY (gst_base_src_start (src))) { if (gst_base_src_start_wait (src) != GST_FLOW_OK) goto start_failed; gst_base_src_stop (src); } } return src->random_access; /* ERRORS */ start_failed: { GST_DEBUG_OBJECT (src, "failed to start"); return FALSE; } } static void gst_base_src_loop (GstPad * pad) { GstBaseSrc *src; GstBuffer *buf = NULL; GstFlowReturn ret; gint64 position; gboolean eos; guint blocksize; GList *pending_events = NULL, *tmp; eos = FALSE; src = GST_BASE_SRC (GST_OBJECT_PARENT (pad)); /* Just leave immediately if we're flushing */ GST_LIVE_LOCK (src); if (G_UNLIKELY (src->priv->flushing || GST_PAD_IS_FLUSHING (pad))) goto flushing; GST_LIVE_UNLOCK (src); gst_base_src_send_stream_start (src); /* The stream-start event could've caused something to flush us */ GST_LIVE_LOCK (src); if (G_UNLIKELY (src->priv->flushing || GST_PAD_IS_FLUSHING (pad))) goto flushing; GST_LIVE_UNLOCK (src); /* check if we need to renegotiate */ if (gst_pad_check_reconfigure (pad)) { if (!gst_base_src_negotiate (src)) { gst_pad_mark_reconfigure (pad); if (GST_PAD_IS_FLUSHING (pad)) goto flushing; else goto negotiate_failed; } } GST_LIVE_LOCK (src); if (G_UNLIKELY (src->priv->flushing || GST_PAD_IS_FLUSHING (pad))) goto flushing; blocksize = src->blocksize; /* if we operate in bytes, we can calculate an offset */ if (src->segment.format == GST_FORMAT_BYTES) { position = src->segment.position; /* 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 %u", 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->segment_pending)) { GstEvent *seg_event = gst_event_new_segment (&src->segment); gst_event_set_seqnum (seg_event, src->priv->segment_seqnum); src->priv->segment_seqnum = gst_util_seqnum_next (); gst_pad_push_event (pad, seg_event); src->priv->segment_pending = FALSE; } if (g_atomic_int_get (&src->priv->have_events)) { GST_OBJECT_LOCK (src); /* take the events */ pending_events = src->priv->pending_events; src->priv->pending_events = NULL; g_atomic_int_set (&src->priv->have_events, FALSE); GST_OBJECT_UNLOCK (src); } /* Push out pending events if any */ if (G_UNLIKELY (pending_events != NULL)) { for (tmp = pending_events; tmp; tmp = g_list_next (tmp)) { GstEvent *ev = (GstEvent *) tmp->data; gst_pad_push_event (pad, ev); } g_list_free (pending_events); } /* figure out the new position */ switch (src->segment.format) { case GST_FORMAT_BYTES: { guint bufsize = gst_buffer_get_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.position; 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_OBJECT_LOCK (src); src->segment.position = position; GST_OBJECT_UNLOCK (src); } if (G_UNLIKELY (src->priv->discont)) { GST_INFO_OBJECT (src, "marking pending DISCONT"); buf = gst_buffer_make_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)) { if (ret == GST_FLOW_NOT_NEGOTIATED) { goto not_negotiated; } 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_EOS; goto pause; } done: return; /* special cases */ not_negotiated: { if (gst_pad_needs_reconfigure (pad)) { GST_DEBUG_OBJECT (src, "Retrying to renegotiate"); return; } /* fallthrough when push returns NOT_NEGOTIATED and we don't have * a pending negotiation request on our srcpad */ } negotiate_failed: { GST_DEBUG_OBJECT (src, "Not negotiated"); ret = GST_FLOW_NOT_NEGOTIATED; goto pause; } flushing: { GST_DEBUG_OBJECT (src, "we are flushing"); GST_LIVE_UNLOCK (src); ret = GST_FLOW_FLUSHING; goto pause; } pause: { const gchar *reason = gst_flow_get_name (ret); GstEvent *event; GST_DEBUG_OBJECT (src, "pausing task, reason %s", reason); src->running = FALSE; gst_pad_pause_task (pad); if (ret == GST_FLOW_EOS) { gboolean flag_segment; GstFormat format; gint64 position; /* perform EOS logic */ flag_segment = (src->segment.flags & GST_SEGMENT_FLAG_SEGMENT) != 0; format = src->segment.format; position = src->segment.position; if (flag_segment) { GstMessage *message; message = gst_message_new_segment_done (GST_OBJECT_CAST (src), format, position); gst_message_set_seqnum (message, src->priv->seqnum); gst_element_post_message (GST_ELEMENT_CAST (src), message); event = gst_event_new_segment_done (format, position); gst_event_set_seqnum (event, src->priv->seqnum); gst_pad_push_event (pad, event); } else { event = gst_event_new_eos (); gst_event_set_seqnum (event, src->priv->seqnum); gst_pad_push_event (pad, event); } } else if (ret == GST_FLOW_NOT_LINKED || ret <= GST_FLOW_EOS) { event = gst_event_new_eos (); gst_event_set_seqnum (event, src->priv->seqnum); /* for fatal errors we post an error message, post the error * first so the app knows about the error first. * Also don't do this for FLUSHING because it happens * due to flushing and posting an error message because of * that is the wrong thing to do, e.g. when we're doing * a flushing seek. */ GST_ELEMENT_ERROR (src, STREAM, FAILED, (_("Internal data flow error.")), ("streaming task paused, reason %s (%d)", reason, ret)); gst_pad_push_event (pad, event); } goto done; } null_buffer: { GST_ELEMENT_ERROR (src, STREAM, FAILED, (_("Internal data flow error.")), ("element returned NULL buffer")); GST_LIVE_UNLOCK (src); goto done; } } static gboolean gst_base_src_set_allocation (GstBaseSrc * basesrc, GstBufferPool * pool, GstAllocator * allocator, GstAllocationParams * params) { GstAllocator *oldalloc; GstBufferPool *oldpool; GstBaseSrcPrivate *priv = basesrc->priv; if (pool) { GST_DEBUG_OBJECT (basesrc, "activate pool"); if (!gst_buffer_pool_set_active (pool, TRUE)) goto activate_failed; } GST_OBJECT_LOCK (basesrc); oldpool = priv->pool; priv->pool = pool; oldalloc = priv->allocator; priv->allocator = allocator; if (params) priv->params = *params; else gst_allocation_params_init (&priv->params); GST_OBJECT_UNLOCK (basesrc); if (oldpool) { /* only deactivate if the pool is not the one we're using */ if (oldpool != pool) { GST_DEBUG_OBJECT (basesrc, "deactivate old pool"); gst_buffer_pool_set_active (oldpool, FALSE); } gst_object_unref (oldpool); } if (oldalloc) { gst_object_unref (oldalloc); } return TRUE; /* ERRORS */ activate_failed: { GST_ERROR_OBJECT (basesrc, "failed to activate bufferpool."); return FALSE; } } static gboolean gst_base_src_activate_pool (GstBaseSrc * basesrc, gboolean active) { GstBaseSrcPrivate *priv = basesrc->priv; GstBufferPool *pool; gboolean res = TRUE; GST_OBJECT_LOCK (basesrc); if ((pool = priv->pool)) pool = gst_object_ref (pool); GST_OBJECT_UNLOCK (basesrc); if (pool) { res = gst_buffer_pool_set_active (pool, active); gst_object_unref (pool); } return res; } static gboolean gst_base_src_decide_allocation_default (GstBaseSrc * basesrc, GstQuery * query) { GstCaps *outcaps; GstBufferPool *pool; guint size, min, max; GstAllocator *allocator; GstAllocationParams params; GstStructure *config; gboolean update_allocator; gst_query_parse_allocation (query, &outcaps, NULL); /* we got configuration from our peer or the decide_allocation method, * parse them */ if (gst_query_get_n_allocation_params (query) > 0) { /* try the allocator */ gst_query_parse_nth_allocation_param (query, 0, &allocator, ¶ms); update_allocator = TRUE; } else { allocator = NULL; gst_allocation_params_init (¶ms); update_allocator = FALSE; } if (gst_query_get_n_allocation_pools (query) > 0) { gst_query_parse_nth_allocation_pool (query, 0, &pool, &size, &min, &max); if (pool == NULL) { /* no pool, we can make our own */ GST_DEBUG_OBJECT (basesrc, "no pool, making new pool"); pool = gst_buffer_pool_new (); } } else { pool = NULL; size = min = max = 0; } /* now configure */ if (pool) { config = gst_buffer_pool_get_config (pool); gst_buffer_pool_config_set_params (config, outcaps, size, min, max); gst_buffer_pool_config_set_allocator (config, allocator, ¶ms); gst_buffer_pool_set_config (pool, config); } if (update_allocator) gst_query_set_nth_allocation_param (query, 0, allocator, ¶ms); else gst_query_add_allocation_param (query, allocator, ¶ms); if (allocator) gst_object_unref (allocator); if (pool) { gst_query_set_nth_allocation_pool (query, 0, pool, size, min, max); gst_object_unref (pool); } return TRUE; } static gboolean gst_base_src_prepare_allocation (GstBaseSrc * basesrc, GstCaps * caps) { GstBaseSrcClass *bclass; gboolean result = TRUE; GstQuery *query; GstBufferPool *pool = NULL; GstAllocator *allocator = NULL; GstAllocationParams params; bclass = GST_BASE_SRC_GET_CLASS (basesrc); /* make query and let peer pad answer, we don't really care if it worked or * not, if it failed, the allocation query would contain defaults and the * subclass would then set better values if needed */ query = gst_query_new_allocation (caps, TRUE); if (!gst_pad_peer_query (basesrc->srcpad, query)) { /* not a problem, just debug a little */ GST_DEBUG_OBJECT (basesrc, "peer ALLOCATION query failed"); } g_assert (bclass->decide_allocation != NULL); result = bclass->decide_allocation (basesrc, query); GST_DEBUG_OBJECT (basesrc, "ALLOCATION (%d) params: %" GST_PTR_FORMAT, result, query); if (!result) goto no_decide_allocation; /* we got configuration from our peer or the decide_allocation method, * parse them */ if (gst_query_get_n_allocation_params (query) > 0) { gst_query_parse_nth_allocation_param (query, 0, &allocator, ¶ms); } else { allocator = NULL; gst_allocation_params_init (¶ms); } if (gst_query_get_n_allocation_pools (query) > 0) gst_query_parse_nth_allocation_pool (query, 0, &pool, NULL, NULL, NULL); result = gst_base_src_set_allocation (basesrc, pool, allocator, ¶ms); gst_query_unref (query); return result; /* Errors */ no_decide_allocation: { GST_WARNING_OBJECT (basesrc, "Subclass failed to decide allocation"); gst_query_unref (query); return result; } } /* 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_query_caps (GST_BASE_SRC_PAD (basesrc), NULL); 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_query_caps (GST_BASE_SRC_PAD (basesrc), thiscaps); GST_DEBUG_OBJECT (basesrc, "caps of peer: %" GST_PTR_FORMAT, peercaps); if (peercaps) { /* The result is already a subset of our caps */ caps = peercaps; gst_caps_unref (thiscaps); } else { /* no peer, work with our own caps then */ caps = thiscaps; } if (caps && !gst_caps_is_empty (caps)) { /* now fixate */ GST_DEBUG_OBJECT (basesrc, "have caps: %" GST_PTR_FORMAT, caps); if (gst_caps_is_any (caps)) { GST_DEBUG_OBJECT (basesrc, "any caps, we stop"); /* hmm, still anything, so element can do anything and * nego is not needed */ result = TRUE; } else { caps = gst_base_src_fixate (basesrc, caps); GST_DEBUG_OBJECT (basesrc, "fixated to: %" GST_PTR_FORMAT, caps); 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_base_src_set_caps (basesrc, caps); } } gst_caps_unref (caps); } else { if (caps) gst_caps_unref (caps); 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; bclass = GST_BASE_SRC_GET_CLASS (basesrc); GST_DEBUG_OBJECT (basesrc, "starting negotiation"); if (G_LIKELY (bclass->negotiate)) result = bclass->negotiate (basesrc); else result = TRUE; if (G_LIKELY (result)) { GstCaps *caps; caps = gst_pad_get_current_caps (basesrc->srcpad); result = gst_base_src_prepare_allocation (basesrc, caps); if (caps) gst_caps_unref (caps); } return result; } static gboolean gst_base_src_start (GstBaseSrc * basesrc) { GstBaseSrcClass *bclass; gboolean result; GST_LIVE_LOCK (basesrc); GST_OBJECT_LOCK (basesrc); if (GST_BASE_SRC_IS_STARTING (basesrc)) goto was_starting; if (GST_BASE_SRC_IS_STARTED (basesrc)) goto was_started; basesrc->priv->start_result = GST_FLOW_FLUSHING; GST_OBJECT_FLAG_SET (basesrc, GST_BASE_SRC_FLAG_STARTING); gst_segment_init (&basesrc->segment, basesrc->segment.format); GST_OBJECT_UNLOCK (basesrc); basesrc->num_buffers_left = basesrc->num_buffers; basesrc->running = FALSE; basesrc->priv->segment_pending = FALSE; basesrc->priv->segment_seqnum = gst_util_seqnum_next (); GST_LIVE_UNLOCK (basesrc); bclass = GST_BASE_SRC_GET_CLASS (basesrc); if (bclass->start) result = bclass->start (basesrc); else result = TRUE; if (!result) goto could_not_start; if (!gst_base_src_is_async (basesrc)) { gst_base_src_start_complete (basesrc, GST_FLOW_OK); /* not really waiting here, we call this to get the result * from the start_complete call */ result = gst_base_src_start_wait (basesrc) == GST_FLOW_OK; } return result; /* ERROR */ was_starting: { GST_DEBUG_OBJECT (basesrc, "was starting"); GST_OBJECT_UNLOCK (basesrc); GST_LIVE_UNLOCK (basesrc); return TRUE; } was_started: { GST_DEBUG_OBJECT (basesrc, "was started"); GST_OBJECT_UNLOCK (basesrc); GST_LIVE_UNLOCK (basesrc); return TRUE; } could_not_start: { GST_DEBUG_OBJECT (basesrc, "could not start"); /* subclass is supposed to post a message. We don't have to call _stop. */ gst_base_src_start_complete (basesrc, GST_FLOW_ERROR); return FALSE; } } /** * gst_base_src_start_complete: * @basesrc: base source instance * @ret: a #GstFlowReturn * * Complete an asynchronous start operation. When the subclass overrides the * start method, it should call gst_base_src_start_complete() when the start * operation completes either from the same thread or from an asynchronous * helper thread. */ void gst_base_src_start_complete (GstBaseSrc * basesrc, GstFlowReturn ret) { gboolean have_size; guint64 size; gboolean seekable; GstFormat format; GstPadMode mode; GstEvent *event; if (ret != GST_FLOW_OK) goto error; GST_DEBUG_OBJECT (basesrc, "starting source"); format = basesrc->segment.format; /* figure out the size */ have_size = FALSE; size = -1; if (format == GST_FORMAT_BYTES) { GstBaseSrcClass *bclass = GST_BASE_SRC_GET_CLASS (basesrc); if (bclass->get_size) { if (!(have_size = bclass->get_size (basesrc, &size))) size = -1; } GST_DEBUG_OBJECT (basesrc, "setting size %" G_GUINT64_FORMAT, size); /* only update the size when operating in bytes, subclass is supposed * to set duration in the start method for other formats */ GST_OBJECT_LOCK (basesrc); basesrc->segment.duration = size; GST_OBJECT_UNLOCK (basesrc); } GST_DEBUG_OBJECT (basesrc, "format: %s, have size: %d, size: %" G_GUINT64_FORMAT ", duration: %" G_GINT64_FORMAT, gst_format_get_name (format), have_size, 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 && format == GST_FORMAT_BYTES; GST_DEBUG_OBJECT (basesrc, "is random_access: %d", basesrc->random_access); /* 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, NULL); gst_pad_mark_reconfigure (GST_BASE_SRC_PAD (basesrc)); GST_OBJECT_LOCK (basesrc->srcpad); mode = GST_PAD_MODE (basesrc->srcpad); GST_OBJECT_UNLOCK (basesrc->srcpad); /* take the stream lock here, we only want to let the task run when we have * set the STARTED flag */ GST_PAD_STREAM_LOCK (basesrc->srcpad); if (mode == GST_PAD_MODE_PUSH) { /* do initial seek, which will start the task */ GST_OBJECT_LOCK (basesrc); event = basesrc->pending_seek; basesrc->pending_seek = NULL; GST_OBJECT_UNLOCK (basesrc); /* The perform seek code will start the task when finished. We don't have to * unlock the streaming thread because it is not running yet */ if (G_UNLIKELY (!gst_base_src_perform_seek (basesrc, event, FALSE))) goto seek_failed; if (event) gst_event_unref (event); } else { /* if not random_access, we cannot operate in pull mode for now */ if (G_UNLIKELY (!basesrc->random_access)) goto no_get_range; } GST_OBJECT_LOCK (basesrc); GST_OBJECT_FLAG_SET (basesrc, GST_BASE_SRC_FLAG_STARTED); GST_OBJECT_FLAG_UNSET (basesrc, GST_BASE_SRC_FLAG_STARTING); basesrc->priv->start_result = ret; GST_ASYNC_SIGNAL (basesrc); GST_OBJECT_UNLOCK (basesrc); GST_PAD_STREAM_UNLOCK (basesrc->srcpad); return; seek_failed: { GST_PAD_STREAM_UNLOCK (basesrc->srcpad); GST_ERROR_OBJECT (basesrc, "Failed to perform initial seek"); gst_base_src_stop (basesrc); if (event) gst_event_unref (event); ret = GST_FLOW_ERROR; goto error; } no_get_range: { GST_PAD_STREAM_UNLOCK (basesrc->srcpad); gst_base_src_stop (basesrc); GST_ERROR_OBJECT (basesrc, "Cannot operate in pull mode, stopping"); ret = GST_FLOW_ERROR; goto error; } error: { GST_OBJECT_LOCK (basesrc); basesrc->priv->start_result = ret; GST_OBJECT_FLAG_UNSET (basesrc, GST_BASE_SRC_FLAG_STARTING); GST_ASYNC_SIGNAL (basesrc); GST_OBJECT_UNLOCK (basesrc); return; } } /** * gst_base_src_start_wait: * @basesrc: base source instance * * Wait until the start operation completes. * * Returns: a #GstFlowReturn. */ GstFlowReturn gst_base_src_start_wait (GstBaseSrc * basesrc) { GstFlowReturn result; GST_OBJECT_LOCK (basesrc); while (GST_BASE_SRC_IS_STARTING (basesrc)) { GST_ASYNC_WAIT (basesrc); } result = basesrc->priv->start_result; GST_OBJECT_UNLOCK (basesrc); GST_DEBUG_OBJECT (basesrc, "got %s", gst_flow_get_name (result)); return result; } static gboolean gst_base_src_stop (GstBaseSrc * basesrc) { GstBaseSrcClass *bclass; gboolean result = TRUE; GST_DEBUG_OBJECT (basesrc, "stopping source"); /* flush all */ gst_base_src_set_flushing (basesrc, TRUE, FALSE, NULL); /* stop the task */ gst_pad_stop_task (basesrc->srcpad); GST_OBJECT_LOCK (basesrc); if (!GST_BASE_SRC_IS_STARTED (basesrc) && !GST_BASE_SRC_IS_STARTING (basesrc)) goto was_stopped; GST_OBJECT_FLAG_UNSET (basesrc, GST_BASE_SRC_FLAG_STARTING); GST_OBJECT_FLAG_UNSET (basesrc, GST_BASE_SRC_FLAG_STARTED); basesrc->priv->start_result = GST_FLOW_FLUSHING; GST_ASYNC_SIGNAL (basesrc); GST_OBJECT_UNLOCK (basesrc); bclass = GST_BASE_SRC_GET_CLASS (basesrc); if (bclass->stop) result = bclass->stop (basesrc); gst_base_src_set_allocation (basesrc, NULL, NULL, NULL); return result; was_stopped: { GST_DEBUG_OBJECT (basesrc, "was stopped"); GST_OBJECT_UNLOCK (basesrc); return TRUE; } } /* start or stop flushing dataprocessing */ static gboolean gst_base_src_set_flushing (GstBaseSrc * basesrc, gboolean flushing, gboolean live_play, gboolean * playing) { GstBaseSrcClass *bclass; bclass = GST_BASE_SRC_GET_CLASS (basesrc); GST_DEBUG_OBJECT (basesrc, "flushing %d, live_play %d", flushing, live_play); if (flushing) { gst_base_src_activate_pool (basesrc, FALSE); /* 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_base_src_activate_pool (basesrc, TRUE); /* Drop all delayed events */ GST_OBJECT_LOCK (basesrc); if (basesrc->priv->pending_events) { g_list_foreach (basesrc->priv->pending_events, (GFunc) gst_event_unref, NULL); g_list_free (basesrc->priv->pending_events); basesrc->priv->pending_events = NULL; g_atomic_int_set (&basesrc->priv->have_events, FALSE); } GST_OBJECT_UNLOCK (basesrc); } GST_LIVE_SIGNAL (basesrc); GST_LIVE_UNLOCK (basesrc); 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_MODE (basesrc->srcpad) == GST_PAD_MODE_PUSH); GST_OBJECT_UNLOCK (basesrc->srcpad); if (start) gst_pad_start_task (basesrc->srcpad, (GstTaskFunction) gst_base_src_loop, basesrc->srcpad, NULL); GST_DEBUG_OBJECT (basesrc, "signal"); GST_LIVE_SIGNAL (basesrc); } GST_LIVE_UNLOCK (basesrc); return TRUE; } static gboolean gst_base_src_activate_push (GstPad * pad, GstObject * parent, gboolean active) { GstBaseSrc *basesrc; basesrc = GST_BASE_SRC (parent); /* 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; } else { GST_DEBUG_OBJECT (basesrc, "Deactivating in push mode"); /* 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; } error_stop: { GST_DEBUG_OBJECT (basesrc, "Failed to stop in push mode"); return FALSE; } } static gboolean gst_base_src_activate_pull (GstPad * pad, GstObject * parent, gboolean active) { GstBaseSrc *basesrc; basesrc = GST_BASE_SRC (parent); /* prepare subclass first */ if (active) { GST_DEBUG_OBJECT (basesrc, "Activating in pull mode"); if (G_UNLIKELY (!gst_base_src_start (basesrc))) goto error_start; } else { GST_DEBUG_OBJECT (basesrc, "Deactivating in pull mode"); 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; } error_stop: { GST_ERROR_OBJECT (basesrc, "Failed to stop in pull mode"); return FALSE; } } static gboolean gst_base_src_activate_mode (GstPad * pad, GstObject * parent, GstPadMode mode, gboolean active) { gboolean res; GstBaseSrc *src = GST_BASE_SRC (parent); src->priv->stream_start_pending = FALSE; switch (mode) { case GST_PAD_MODE_PULL: res = gst_base_src_activate_pull (pad, parent, active); break; case GST_PAD_MODE_PUSH: src->priv->stream_start_pending = active; res = gst_base_src_activate_push (pad, parent, active); break; default: GST_LOG_OBJECT (pad, "unknown activation mode %d", mode); res = FALSE; break; } return res; } 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: { /* we don't need to unblock anything here, the pad deactivation code * already did this */ g_atomic_int_set (&basesrc->priv->pending_eos, FALSE); gst_event_replace (&basesrc->pending_seek, 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; } } /** * gst_base_src_get_buffer_pool: * @src: a #GstBaseSrc * * Returns: (transfer full): the instance of the #GstBufferPool used * by the src; free it after use it */ GstBufferPool * gst_base_src_get_buffer_pool (GstBaseSrc * src) { g_return_val_if_fail (GST_IS_BASE_SRC (src), NULL); if (src->priv->pool) return gst_object_ref (src->priv->pool); return NULL; } /** * gst_base_src_get_allocator: * @src: a #GstBaseSrc * @allocator: (out) (allow-none) (transfer full): the #GstAllocator * used * @params: (out) (allow-none) (transfer full): the * #GstAllocatorParams of @allocator * * Lets #GstBaseSrc sub-classes to know the memory @allocator * used by the base class and its @params. * * Unref the @allocator after use it. */ void gst_base_src_get_allocator (GstBaseSrc * src, GstAllocator ** allocator, GstAllocationParams * params) { g_return_if_fail (GST_IS_BASE_SRC (src)); if (allocator) *allocator = src->priv->allocator ? gst_object_ref (src->priv->allocator) : NULL; if (params) *params = src->priv->params; }