/* GStreamer * Copyright (C) 2005-2007 Wim Taymans * * gstbasesink.c: Base class for sink elements * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ /** * SECTION:gstbasesink * @short_description: Base class for sink elements * @see_also: #GstBaseTransform, #GstBaseSrc * * #GstBaseSink is the base class for sink elements in GStreamer, such as * xvimagesink or filesink. It is a layer on top of #GstElement that provides a * simplified interface to plugin writers. #GstBaseSink handles many details * for you, for example: preroll, clock synchronization, state changes, * activation in push or pull mode, and queries. * * In most cases, when writing sink elements, there is no need to implement * class methods from #GstElement or to set functions on pads, because the * #GstBaseSink infrastructure should be sufficient. * * #GstBaseSink provides support for exactly one sink pad, which should be * named "sink". A sink implementation (subclass of #GstBaseSink) 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); * * // sinktemplate should be a #GstStaticPadTemplate with direction * // #GST_PAD_SINK and name "sink" * gst_element_class_add_pad_template (gstelement_class, * gst_static_pad_template_get (&sinktemplate)); * * gst_element_class_set_static_metadata (gstelement_class, * "Sink name", * "Sink", * "My Sink element", * "The author <my.sink@my.email>"); * } * ]| * * #GstBaseSink will handle the prerolling correctly. This means that it will * return #GST_STATE_CHANGE_ASYNC from a state change to PAUSED until the first * buffer arrives in this element. The base class will call the * #GstBaseSinkClass.preroll() vmethod with this preroll buffer and will then * commit the state change to the next asynchronously pending state. * * When the element is set to PLAYING, #GstBaseSink will synchronise on the * clock using the times returned from #GstBaseSinkClass.get_times(). If this * function returns #GST_CLOCK_TIME_NONE for the start time, no synchronisation * will be done. Synchronisation can be disabled entirely by setting the object * #GstBaseSink:sync property to %FALSE. * * After synchronisation the virtual method #GstBaseSinkClass.render() will be * called. Subclasses should minimally implement this method. * * Subclasses that synchronise on the clock in the #GstBaseSinkClass.render() * method are supported as well. These classes typically receive a buffer in * the render method and can then potentially block on the clock while * rendering. A typical example is an audiosink. * These subclasses can use gst_base_sink_wait_preroll() to perform the * blocking wait. * * Upon receiving the EOS event in the PLAYING state, #GstBaseSink will wait * for the clock to reach the time indicated by the stop time of the last * #GstBaseSinkClass.get_times() call before posting an EOS message. When the * element receives EOS in PAUSED, preroll completes, the event is queued and an * EOS message is posted when going to PLAYING. * * #GstBaseSink will internally use the #GST_EVENT_SEGMENT events to schedule * synchronisation and clipping of buffers. Buffers that fall completely outside * of the current segment are dropped. Buffers that fall partially in the * segment are rendered (and prerolled). Subclasses should do any subbuffer * clipping themselves when needed. * * #GstBaseSink will by default report the current playback position in * #GST_FORMAT_TIME based on the current clock time and segment information. * If no clock has been set on the element, the query will be forwarded * upstream. * * The #GstBaseSinkClass.set_caps() function will be called when the subclass * should configure itself to process a specific media type. * * The #GstBaseSinkClass.start() and #GstBaseSinkClass.stop() virtual methods * will be called when resources should be allocated. Any * #GstBaseSinkClass.preroll(), #GstBaseSinkClass.render() and * #GstBaseSinkClass.set_caps() function will be called between the * #GstBaseSinkClass.start() and #GstBaseSinkClass.stop() calls. * * The #GstBaseSinkClass.event() virtual method will be called when an event is * received by #GstBaseSink. Normally this method should only be overriden by * very specific elements (such as file sinks) which need to handle the * newsegment event specially. * * The #GstBaseSinkClass.unlock() method is called when the elements should * unblock any blocking operations they perform in the * #GstBaseSinkClass.render() method. This is mostly useful when the * #GstBaseSinkClass.render() method performs a blocking write on a file * descriptor, for example. * * The #GstBaseSink:max-lateness property affects how the sink deals with * buffers that arrive too late in the sink. A buffer arrives too late in the * sink when the presentation time (as a combination of the last segment, buffer * timestamp and element base_time) plus the duration is before the current * time of the clock. * If the frame is later than max-lateness, the sink will drop the buffer * without calling the render method. * This feature is disabled if sync is disabled, the * #GstBaseSinkClass.get_times() method does not return a valid start time or * max-lateness is set to -1 (the default). * Subclasses can use gst_base_sink_set_max_lateness() to configure the * max-lateness value. * * The #GstBaseSink:qos property will enable the quality-of-service features of * the basesink which gather statistics about the real-time performance of the * clock synchronisation. For each buffer received in the sink, statistics are * gathered and a QOS event is sent upstream with these numbers. This * information can then be used by upstream elements to reduce their processing * rate, for example. * * The #GstBaseSink:async property can be used to instruct the sink to never * perform an ASYNC state change. This feature is mostly usable when dealing * with non-synchronized streams or sparse streams. * * Last reviewed on 2007-08-29 (0.10.15) */ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include #include "gstbasesink.h" #include GST_DEBUG_CATEGORY_STATIC (gst_base_sink_debug); #define GST_CAT_DEFAULT gst_base_sink_debug #define GST_BASE_SINK_GET_PRIVATE(obj) \ (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_BASE_SINK, GstBaseSinkPrivate)) #define GST_FLOW_STEP GST_FLOW_CUSTOM_ERROR typedef struct { gboolean valid; /* if this info is valid */ guint32 seqnum; /* the seqnum of the STEP event */ GstFormat format; /* the format of the amount */ guint64 amount; /* the total amount of data to skip */ guint64 position; /* the position in the stepped data */ guint64 duration; /* the duration in time of the skipped data */ guint64 start; /* running_time of the start */ gdouble rate; /* rate of skipping */ gdouble start_rate; /* rate before skipping */ guint64 start_start; /* start position skipping */ guint64 start_stop; /* stop position skipping */ gboolean flush; /* if this was a flushing step */ gboolean intermediate; /* if this is an intermediate step */ gboolean need_preroll; /* if we need preroll after this step */ } GstStepInfo; struct _GstBaseSinkPrivate { gint qos_enabled; /* ATOMIC */ gboolean async_enabled; GstClockTimeDiff ts_offset; GstClockTime render_delay; /* start, stop of current buffer, stream time, used to report position */ GstClockTime current_sstart; GstClockTime current_sstop; /* start, stop and jitter of current buffer, running time */ GstClockTime current_rstart; GstClockTime current_rstop; GstClockTimeDiff current_jitter; /* the running time of the previous buffer */ GstClockTime prev_rstart; /* EOS sync time in running time */ GstClockTime eos_rtime; /* last buffer that arrived in time, running time */ GstClockTime last_render_time; /* when the last buffer left the sink, running time */ GstClockTime last_left; /* running averages go here these are done on running time */ GstClockTime avg_pt; GstClockTime avg_duration; gdouble avg_rate; GstClockTime avg_in_diff; /* these are done on system time. avg_jitter and avg_render are * compared to eachother to see if the rendering time takes a * huge amount of the processing, If so we are flooded with * buffers. */ GstClockTime last_left_systime; GstClockTime avg_jitter; GstClockTime start, stop; GstClockTime avg_render; /* number of rendered and dropped frames */ guint64 rendered; guint64 dropped; /* latency stuff */ GstClockTime latency; /* if we already commited the state */ gboolean commited; /* state change to playing ongoing */ gboolean to_playing; /* when we received EOS */ gboolean received_eos; /* when we are prerolled and able to report latency */ gboolean have_latency; /* the last buffer we prerolled or rendered. Useful for making snapshots */ gint enable_last_sample; /* atomic */ GstBuffer *last_buffer; GstCaps *last_caps; /* negotiated caps */ GstCaps *caps; /* blocksize for pulling */ guint blocksize; gboolean discont; /* seqnum of the stream */ guint32 seqnum; gboolean call_preroll; gboolean step_unlock; /* we have a pending and a current step operation */ GstStepInfo current_step; GstStepInfo pending_step; /* Cached GstClockID */ GstClockID cached_clock_id; /* for throttling and QoS */ GstClockTime earliest_in_time; GstClockTime throttle_time; }; #define DO_RUNNING_AVG(avg,val,size) (((val) + ((size)-1) * (avg)) / (size)) /* generic running average, this has a neutral window size */ #define UPDATE_RUNNING_AVG(avg,val) DO_RUNNING_AVG(avg,val,8) /* the windows for these running averages are experimentally obtained. * possitive values get averaged more while negative values use a small * window so we can react faster to badness. */ #define UPDATE_RUNNING_AVG_P(avg,val) DO_RUNNING_AVG(avg,val,16) #define UPDATE_RUNNING_AVG_N(avg,val) DO_RUNNING_AVG(avg,val,4) /* BaseSink properties */ #define DEFAULT_CAN_ACTIVATE_PULL FALSE /* fixme: enable me */ #define DEFAULT_CAN_ACTIVATE_PUSH TRUE #define DEFAULT_SYNC TRUE #define DEFAULT_MAX_LATENESS -1 #define DEFAULT_QOS FALSE #define DEFAULT_ASYNC TRUE #define DEFAULT_TS_OFFSET 0 #define DEFAULT_BLOCKSIZE 4096 #define DEFAULT_RENDER_DELAY 0 #define DEFAULT_ENABLE_LAST_SAMPLE TRUE #define DEFAULT_THROTTLE_TIME 0 enum { PROP_0, PROP_SYNC, PROP_MAX_LATENESS, PROP_QOS, PROP_ASYNC, PROP_TS_OFFSET, PROP_ENABLE_LAST_SAMPLE, PROP_LAST_SAMPLE, PROP_BLOCKSIZE, PROP_RENDER_DELAY, PROP_THROTTLE_TIME, PROP_LAST }; static GstElementClass *parent_class = NULL; static void gst_base_sink_class_init (GstBaseSinkClass * klass); static void gst_base_sink_init (GstBaseSink * trans, gpointer g_class); static void gst_base_sink_finalize (GObject * object); GType gst_base_sink_get_type (void) { static volatile gsize base_sink_type = 0; if (g_once_init_enter (&base_sink_type)) { GType _type; static const GTypeInfo base_sink_info = { sizeof (GstBaseSinkClass), NULL, NULL, (GClassInitFunc) gst_base_sink_class_init, NULL, NULL, sizeof (GstBaseSink), 0, (GInstanceInitFunc) gst_base_sink_init, }; _type = g_type_register_static (GST_TYPE_ELEMENT, "GstBaseSink", &base_sink_info, G_TYPE_FLAG_ABSTRACT); g_once_init_leave (&base_sink_type, _type); } return base_sink_type; } static void gst_base_sink_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_base_sink_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static gboolean gst_base_sink_send_event (GstElement * element, GstEvent * event); static gboolean default_element_query (GstElement * element, GstQuery * query); static GstCaps *gst_base_sink_default_get_caps (GstBaseSink * sink, GstCaps * caps); static gboolean gst_base_sink_default_set_caps (GstBaseSink * sink, GstCaps * caps); static void gst_base_sink_default_get_times (GstBaseSink * basesink, GstBuffer * buffer, GstClockTime * start, GstClockTime * end); static gboolean gst_base_sink_set_flushing (GstBaseSink * basesink, GstPad * pad, gboolean flushing); static gboolean gst_base_sink_default_activate_pull (GstBaseSink * basesink, gboolean active); static gboolean gst_base_sink_default_do_seek (GstBaseSink * sink, GstSegment * segment); static gboolean gst_base_sink_default_prepare_seek_segment (GstBaseSink * sink, GstEvent * event, GstSegment * segment); static GstStateChangeReturn gst_base_sink_change_state (GstElement * element, GstStateChange transition); static gboolean gst_base_sink_sink_query (GstPad * pad, GstObject * parent, GstQuery * query); static GstFlowReturn gst_base_sink_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer); static GstFlowReturn gst_base_sink_chain_list (GstPad * pad, GstObject * parent, GstBufferList * list); static void gst_base_sink_loop (GstPad * pad); static gboolean gst_base_sink_pad_activate (GstPad * pad, GstObject * parent); static gboolean gst_base_sink_pad_activate_mode (GstPad * pad, GstObject * parent, GstPadMode mode, gboolean active); static gboolean gst_base_sink_default_event (GstBaseSink * basesink, GstEvent * event); static GstFlowReturn gst_base_sink_default_wait_event (GstBaseSink * basesink, GstEvent * event); static gboolean gst_base_sink_event (GstPad * pad, GstObject * parent, GstEvent * event); static gboolean gst_base_sink_default_query (GstBaseSink * sink, GstQuery * query); static gboolean gst_base_sink_negotiate_pull (GstBaseSink * basesink); static GstCaps *gst_base_sink_default_fixate (GstBaseSink * bsink, GstCaps * caps); static GstCaps *gst_base_sink_fixate (GstBaseSink * bsink, GstCaps * caps); /* check if an object was too late */ static gboolean gst_base_sink_is_too_late (GstBaseSink * basesink, GstMiniObject * obj, GstClockTime rstart, GstClockTime rstop, GstClockReturn status, GstClockTimeDiff jitter); static void gst_base_sink_class_init (GstBaseSinkClass * 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_sink_debug, "basesink", 0, "basesink element"); g_type_class_add_private (klass, sizeof (GstBaseSinkPrivate)); parent_class = g_type_class_peek_parent (klass); gobject_class->finalize = gst_base_sink_finalize; gobject_class->set_property = gst_base_sink_set_property; gobject_class->get_property = gst_base_sink_get_property; g_object_class_install_property (gobject_class, PROP_SYNC, g_param_spec_boolean ("sync", "Sync", "Sync on the clock", DEFAULT_SYNC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_MAX_LATENESS, g_param_spec_int64 ("max-lateness", "Max Lateness", "Maximum number of nanoseconds that a buffer can be late before it " "is dropped (-1 unlimited)", -1, G_MAXINT64, DEFAULT_MAX_LATENESS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_QOS, g_param_spec_boolean ("qos", "Qos", "Generate Quality-of-Service events upstream", DEFAULT_QOS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstBaseSink:async: * * If set to #TRUE, the basesink will perform asynchronous state changes. * When set to #FALSE, the sink will not signal the parent when it prerolls. * Use this option when dealing with sparse streams or when synchronisation is * not required. */ g_object_class_install_property (gobject_class, PROP_ASYNC, g_param_spec_boolean ("async", "Async", "Go asynchronously to PAUSED", DEFAULT_ASYNC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstBaseSink:ts-offset: * * Controls the final synchronisation, a negative value will render the buffer * earlier while a positive value delays playback. This property can be * used to fix synchronisation in bad files. */ g_object_class_install_property (gobject_class, PROP_TS_OFFSET, g_param_spec_int64 ("ts-offset", "TS Offset", "Timestamp offset in nanoseconds", G_MININT64, G_MAXINT64, DEFAULT_TS_OFFSET, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstBaseSink:enable-last-sample: * * Enable the last-sample property. If FALSE, basesink doesn't keep a * reference to the last buffer arrived and the last-sample property is always * set to NULL. This can be useful if you need buffers to be released as soon * as possible, eg. if you're using a buffer pool. */ g_object_class_install_property (gobject_class, PROP_ENABLE_LAST_SAMPLE, g_param_spec_boolean ("enable-last-sample", "Enable Last Buffer", "Enable the last-sample property", DEFAULT_ENABLE_LAST_SAMPLE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstBaseSink:last-sample: * * The last buffer that arrived in the sink and was used for preroll or for * rendering. This property can be used to generate thumbnails. This property * can be NULL when the sink has not yet received a bufer. */ g_object_class_install_property (gobject_class, PROP_LAST_SAMPLE, g_param_spec_boxed ("last-sample", "Last Sample", "The last sample received in the sink", GST_TYPE_SAMPLE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); /** * GstBaseSink:blocksize: * * The amount of bytes to pull when operating in pull mode. */ /* FIXME 0.11: blocksize property should be int, otherwise min>max.. */ g_object_class_install_property (gobject_class, PROP_BLOCKSIZE, g_param_spec_uint ("blocksize", "Block size", "Size in bytes to pull per buffer (0 = default)", 0, G_MAXUINT, DEFAULT_BLOCKSIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstBaseSink:render-delay: * * The additional delay between synchronisation and actual rendering of the * media. This property will add additional latency to the device in order to * make other sinks compensate for the delay. */ g_object_class_install_property (gobject_class, PROP_RENDER_DELAY, g_param_spec_uint64 ("render-delay", "Render Delay", "Additional render delay of the sink in nanoseconds", 0, G_MAXUINT64, DEFAULT_RENDER_DELAY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstBaseSink:throttle-time: * * The time to insert between buffers. This property can be used to control * the maximum amount of buffers per second to render. Setting this property * to a value bigger than 0 will make the sink create THROTTLE QoS events. */ g_object_class_install_property (gobject_class, PROP_THROTTLE_TIME, g_param_spec_uint64 ("throttle-time", "Throttle time", "The time to keep between rendered buffers", 0, G_MAXUINT64, DEFAULT_THROTTLE_TIME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_base_sink_change_state); gstelement_class->send_event = GST_DEBUG_FUNCPTR (gst_base_sink_send_event); gstelement_class->query = GST_DEBUG_FUNCPTR (default_element_query); klass->get_caps = GST_DEBUG_FUNCPTR (gst_base_sink_default_get_caps); klass->set_caps = GST_DEBUG_FUNCPTR (gst_base_sink_default_set_caps); klass->fixate = GST_DEBUG_FUNCPTR (gst_base_sink_default_fixate); klass->activate_pull = GST_DEBUG_FUNCPTR (gst_base_sink_default_activate_pull); klass->get_times = GST_DEBUG_FUNCPTR (gst_base_sink_default_get_times); klass->query = GST_DEBUG_FUNCPTR (gst_base_sink_default_query); klass->event = GST_DEBUG_FUNCPTR (gst_base_sink_default_event); klass->wait_event = GST_DEBUG_FUNCPTR (gst_base_sink_default_wait_event); /* Registering debug symbols for function pointers */ GST_DEBUG_REGISTER_FUNCPTR (gst_base_sink_fixate); GST_DEBUG_REGISTER_FUNCPTR (gst_base_sink_pad_activate); GST_DEBUG_REGISTER_FUNCPTR (gst_base_sink_pad_activate_mode); GST_DEBUG_REGISTER_FUNCPTR (gst_base_sink_event); GST_DEBUG_REGISTER_FUNCPTR (gst_base_sink_chain); GST_DEBUG_REGISTER_FUNCPTR (gst_base_sink_chain_list); GST_DEBUG_REGISTER_FUNCPTR (gst_base_sink_sink_query); } static GstCaps * gst_base_sink_query_caps (GstBaseSink * bsink, GstPad * pad, GstCaps * filter) { GstBaseSinkClass *bclass; GstCaps *caps = NULL; gboolean fixed; bclass = GST_BASE_SINK_GET_CLASS (bsink); fixed = GST_PAD_IS_FIXED_CAPS (pad); if (fixed || bsink->pad_mode == GST_PAD_MODE_PULL) { /* if we are operating in pull mode or fixed caps, we only accept the * currently negotiated caps */ caps = gst_pad_get_current_caps (pad); } if (caps == NULL) { if (bclass->get_caps) caps = bclass->get_caps (bsink, filter); if (caps == NULL) { GstPadTemplate *pad_template; pad_template = gst_element_class_get_pad_template (GST_ELEMENT_CLASS (bclass), "sink"); 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_sink_default_fixate (GstBaseSink * bsink, GstCaps * caps) { GST_DEBUG_OBJECT (bsink, "using default caps fixate function"); return gst_caps_fixate (caps); } static GstCaps * gst_base_sink_fixate (GstBaseSink * bsink, GstCaps * caps) { GstBaseSinkClass *bclass; bclass = GST_BASE_SINK_GET_CLASS (bsink); if (bclass->fixate) caps = bclass->fixate (bsink, caps); return caps; } static void gst_base_sink_init (GstBaseSink * basesink, gpointer g_class) { GstPadTemplate *pad_template; GstBaseSinkPrivate *priv; basesink->priv = priv = GST_BASE_SINK_GET_PRIVATE (basesink); pad_template = gst_element_class_get_pad_template (GST_ELEMENT_CLASS (g_class), "sink"); g_return_if_fail (pad_template != NULL); basesink->sinkpad = gst_pad_new_from_template (pad_template, "sink"); gst_pad_set_activate_function (basesink->sinkpad, gst_base_sink_pad_activate); gst_pad_set_activatemode_function (basesink->sinkpad, gst_base_sink_pad_activate_mode); gst_pad_set_query_function (basesink->sinkpad, gst_base_sink_sink_query); gst_pad_set_event_function (basesink->sinkpad, gst_base_sink_event); gst_pad_set_chain_function (basesink->sinkpad, gst_base_sink_chain); gst_pad_set_chain_list_function (basesink->sinkpad, gst_base_sink_chain_list); gst_element_add_pad (GST_ELEMENT_CAST (basesink), basesink->sinkpad); basesink->pad_mode = GST_PAD_MODE_NONE; g_mutex_init (&basesink->preroll_lock); g_cond_init (&basesink->preroll_cond); priv->have_latency = FALSE; basesink->can_activate_push = DEFAULT_CAN_ACTIVATE_PUSH; basesink->can_activate_pull = DEFAULT_CAN_ACTIVATE_PULL; basesink->sync = DEFAULT_SYNC; basesink->max_lateness = DEFAULT_MAX_LATENESS; g_atomic_int_set (&priv->qos_enabled, DEFAULT_QOS); priv->async_enabled = DEFAULT_ASYNC; priv->ts_offset = DEFAULT_TS_OFFSET; priv->render_delay = DEFAULT_RENDER_DELAY; priv->blocksize = DEFAULT_BLOCKSIZE; priv->cached_clock_id = NULL; g_atomic_int_set (&priv->enable_last_sample, DEFAULT_ENABLE_LAST_SAMPLE); priv->throttle_time = DEFAULT_THROTTLE_TIME; GST_OBJECT_FLAG_SET (basesink, GST_ELEMENT_FLAG_SINK); } static void gst_base_sink_finalize (GObject * object) { GstBaseSink *basesink; basesink = GST_BASE_SINK (object); g_mutex_clear (&basesink->preroll_lock); g_cond_clear (&basesink->preroll_cond); G_OBJECT_CLASS (parent_class)->finalize (object); } /** * gst_base_sink_set_sync: * @sink: the sink * @sync: the new sync value. * * Configures @sink to synchronize on the clock or not. When * @sync is FALSE, incoming samples will be played as fast as * possible. If @sync is TRUE, the timestamps of the incomming * buffers will be used to schedule the exact render time of its * contents. */ void gst_base_sink_set_sync (GstBaseSink * sink, gboolean sync) { g_return_if_fail (GST_IS_BASE_SINK (sink)); GST_OBJECT_LOCK (sink); sink->sync = sync; GST_OBJECT_UNLOCK (sink); } /** * gst_base_sink_get_sync: * @sink: the sink * * Checks if @sink is currently configured to synchronize against the * clock. * * Returns: TRUE if the sink is configured to synchronize against the clock. */ gboolean gst_base_sink_get_sync (GstBaseSink * sink) { gboolean res; g_return_val_if_fail (GST_IS_BASE_SINK (sink), FALSE); GST_OBJECT_LOCK (sink); res = sink->sync; GST_OBJECT_UNLOCK (sink); return res; } /** * gst_base_sink_set_max_lateness: * @sink: the sink * @max_lateness: the new max lateness value. * * Sets the new max lateness value to @max_lateness. This value is * used to decide if a buffer should be dropped or not based on the * buffer timestamp and the current clock time. A value of -1 means * an unlimited time. */ void gst_base_sink_set_max_lateness (GstBaseSink * sink, gint64 max_lateness) { g_return_if_fail (GST_IS_BASE_SINK (sink)); GST_OBJECT_LOCK (sink); sink->max_lateness = max_lateness; GST_OBJECT_UNLOCK (sink); } /** * gst_base_sink_get_max_lateness: * @sink: the sink * * Gets the max lateness value. See gst_base_sink_set_max_lateness for * more details. * * Returns: The maximum time in nanoseconds that a buffer can be late * before it is dropped and not rendered. A value of -1 means an * unlimited time. */ gint64 gst_base_sink_get_max_lateness (GstBaseSink * sink) { gint64 res; g_return_val_if_fail (GST_IS_BASE_SINK (sink), -1); GST_OBJECT_LOCK (sink); res = sink->max_lateness; GST_OBJECT_UNLOCK (sink); return res; } /** * gst_base_sink_set_qos_enabled: * @sink: the sink * @enabled: the new qos value. * * Configures @sink to send Quality-of-Service events upstream. */ void gst_base_sink_set_qos_enabled (GstBaseSink * sink, gboolean enabled) { g_return_if_fail (GST_IS_BASE_SINK (sink)); g_atomic_int_set (&sink->priv->qos_enabled, enabled); } /** * gst_base_sink_is_qos_enabled: * @sink: the sink * * Checks if @sink is currently configured to send Quality-of-Service events * upstream. * * Returns: TRUE if the sink is configured to perform Quality-of-Service. */ gboolean gst_base_sink_is_qos_enabled (GstBaseSink * sink) { gboolean res; g_return_val_if_fail (GST_IS_BASE_SINK (sink), FALSE); res = g_atomic_int_get (&sink->priv->qos_enabled); return res; } /** * gst_base_sink_set_async_enabled: * @sink: the sink * @enabled: the new async value. * * Configures @sink to perform all state changes asynchronusly. When async is * disabled, the sink will immediately go to PAUSED instead of waiting for a * preroll buffer. This feature is useful if the sink does not synchronize * against the clock or when it is dealing with sparse streams. */ void gst_base_sink_set_async_enabled (GstBaseSink * sink, gboolean enabled) { g_return_if_fail (GST_IS_BASE_SINK (sink)); GST_BASE_SINK_PREROLL_LOCK (sink); g_atomic_int_set (&sink->priv->async_enabled, enabled); GST_LOG_OBJECT (sink, "set async enabled to %d", enabled); GST_BASE_SINK_PREROLL_UNLOCK (sink); } /** * gst_base_sink_is_async_enabled: * @sink: the sink * * Checks if @sink is currently configured to perform asynchronous state * changes to PAUSED. * * Returns: TRUE if the sink is configured to perform asynchronous state * changes. */ gboolean gst_base_sink_is_async_enabled (GstBaseSink * sink) { gboolean res; g_return_val_if_fail (GST_IS_BASE_SINK (sink), FALSE); res = g_atomic_int_get (&sink->priv->async_enabled); return res; } /** * gst_base_sink_set_ts_offset: * @sink: the sink * @offset: the new offset * * Adjust the synchronisation of @sink with @offset. A negative value will * render buffers earlier than their timestamp. A positive value will delay * rendering. This function can be used to fix playback of badly timestamped * buffers. */ void gst_base_sink_set_ts_offset (GstBaseSink * sink, GstClockTimeDiff offset) { g_return_if_fail (GST_IS_BASE_SINK (sink)); GST_OBJECT_LOCK (sink); sink->priv->ts_offset = offset; GST_LOG_OBJECT (sink, "set time offset to %" G_GINT64_FORMAT, offset); GST_OBJECT_UNLOCK (sink); } /** * gst_base_sink_get_ts_offset: * @sink: the sink * * Get the synchronisation offset of @sink. * * Returns: The synchronisation offset. */ GstClockTimeDiff gst_base_sink_get_ts_offset (GstBaseSink * sink) { GstClockTimeDiff res; g_return_val_if_fail (GST_IS_BASE_SINK (sink), 0); GST_OBJECT_LOCK (sink); res = sink->priv->ts_offset; GST_OBJECT_UNLOCK (sink); return res; } /** * gst_base_sink_get_last_sample: * @sink: the sink * * Get the last sample that arrived in the sink and was used for preroll or for * rendering. This property can be used to generate thumbnails. * * The #GstCaps on the sample can be used to determine the type of the buffer. * * Free-function: gst_sample_unref * * Returns: (transfer full): a #GstSample. gst_sample_unref() after usage. * This function returns NULL when no buffer has arrived in the sink yet * or when the sink is not in PAUSED or PLAYING. */ GstSample * gst_base_sink_get_last_sample (GstBaseSink * sink) { GstSample *res = NULL; g_return_val_if_fail (GST_IS_BASE_SINK (sink), NULL); GST_OBJECT_LOCK (sink); if (sink->priv->last_buffer) { res = gst_sample_new (sink->priv->last_buffer, sink->priv->last_caps, &sink->segment, NULL); } GST_OBJECT_UNLOCK (sink); return res; } /* with OBJECT_LOCK */ static void gst_base_sink_set_last_buffer_unlocked (GstBaseSink * sink, GstBuffer * buffer) { GstBuffer *old; old = sink->priv->last_buffer; if (G_LIKELY (old != buffer)) { GST_DEBUG_OBJECT (sink, "setting last buffer to %p", buffer); if (G_LIKELY (buffer)) gst_buffer_ref (buffer); sink->priv->last_buffer = buffer; if (buffer) /* copy over the caps */ gst_caps_replace (&sink->priv->last_caps, sink->priv->caps); else gst_caps_replace (&sink->priv->last_caps, NULL); } else { old = NULL; } /* avoid unreffing with the lock because cleanup code might want to take the * lock too */ if (G_LIKELY (old)) { GST_OBJECT_UNLOCK (sink); gst_buffer_unref (old); GST_OBJECT_LOCK (sink); } } static void gst_base_sink_set_last_buffer (GstBaseSink * sink, GstBuffer * buffer) { if (!g_atomic_int_get (&sink->priv->enable_last_sample)) return; GST_OBJECT_LOCK (sink); gst_base_sink_set_last_buffer_unlocked (sink, buffer); GST_OBJECT_UNLOCK (sink); } /** * gst_base_sink_set_last_sample_enabled: * @sink: the sink * @enabled: the new enable-last-sample value. * * Configures @sink to store the last received sample in the last-sample * property. */ void gst_base_sink_set_last_sample_enabled (GstBaseSink * sink, gboolean enabled) { g_return_if_fail (GST_IS_BASE_SINK (sink)); /* Only take lock if we change the value */ if (g_atomic_int_compare_and_exchange (&sink->priv->enable_last_sample, !enabled, enabled) && !enabled) { GST_OBJECT_LOCK (sink); gst_base_sink_set_last_buffer_unlocked (sink, NULL); GST_OBJECT_UNLOCK (sink); } } /** * gst_base_sink_is_last_sample_enabled: * @sink: the sink * * Checks if @sink is currently configured to store the last received sample in * the last-sample property. * * Returns: TRUE if the sink is configured to store the last received sample. */ gboolean gst_base_sink_is_last_sample_enabled (GstBaseSink * sink) { g_return_val_if_fail (GST_IS_BASE_SINK (sink), FALSE); return g_atomic_int_get (&sink->priv->enable_last_sample); } /** * gst_base_sink_get_latency: * @sink: the sink * * Get the currently configured latency. * * Returns: The configured latency. */ GstClockTime gst_base_sink_get_latency (GstBaseSink * sink) { GstClockTime res; GST_OBJECT_LOCK (sink); res = sink->priv->latency; GST_OBJECT_UNLOCK (sink); return res; } /** * gst_base_sink_query_latency: * @sink: the sink * @live: (out) (allow-none): if the sink is live * @upstream_live: (out) (allow-none): if an upstream element is live * @min_latency: (out) (allow-none): the min latency of the upstream elements * @max_latency: (out) (allow-none): the max latency of the upstream elements * * Query the sink for the latency parameters. The latency will be queried from * the upstream elements. @live will be TRUE if @sink is configured to * synchronize against the clock. @upstream_live will be TRUE if an upstream * element is live. * * If both @live and @upstream_live are TRUE, the sink will want to compensate * for the latency introduced by the upstream elements by setting the * @min_latency to a strictly possitive value. * * This function is mostly used by subclasses. * * Returns: TRUE if the query succeeded. */ gboolean gst_base_sink_query_latency (GstBaseSink * sink, gboolean * live, gboolean * upstream_live, GstClockTime * min_latency, GstClockTime * max_latency) { gboolean l, us_live, res, have_latency; GstClockTime min, max, render_delay; GstQuery *query; GstClockTime us_min, us_max; /* we are live when we sync to the clock */ GST_OBJECT_LOCK (sink); l = sink->sync; have_latency = sink->priv->have_latency; render_delay = sink->priv->render_delay; GST_OBJECT_UNLOCK (sink); /* assume no latency */ min = 0; max = -1; us_live = FALSE; if (have_latency) { GST_DEBUG_OBJECT (sink, "we are ready for LATENCY query"); /* we are ready for a latency query this is when we preroll or when we are * not async. */ query = gst_query_new_latency (); /* ask the peer for the latency */ if ((res = gst_pad_peer_query (sink->sinkpad, query))) { /* get upstream min and max latency */ gst_query_parse_latency (query, &us_live, &us_min, &us_max); if (us_live) { /* upstream live, use its latency, subclasses should use these * values to create the complete latency. */ min = us_min; max = us_max; } if (l) { /* we need to add the render delay if we are live */ if (min != -1) min += render_delay; if (max != -1) max += render_delay; } } gst_query_unref (query); } else { GST_DEBUG_OBJECT (sink, "we are not yet ready for LATENCY query"); res = FALSE; } /* not live, we tried to do the query, if it failed we return TRUE anyway */ if (!res) { if (!l) { res = TRUE; GST_DEBUG_OBJECT (sink, "latency query failed but we are not live"); } else { GST_DEBUG_OBJECT (sink, "latency query failed and we are live"); } } if (res) { GST_DEBUG_OBJECT (sink, "latency query: live: %d, have_latency %d," " upstream: %d, min %" GST_TIME_FORMAT ", max %" GST_TIME_FORMAT, l, have_latency, us_live, GST_TIME_ARGS (min), GST_TIME_ARGS (max)); if (live) *live = l; if (upstream_live) *upstream_live = us_live; if (min_latency) *min_latency = min; if (max_latency) *max_latency = max; } return res; } /** * gst_base_sink_set_render_delay: * @sink: a #GstBaseSink * @delay: the new delay * * Set the render delay in @sink to @delay. The render delay is the time * between actual rendering of a buffer and its synchronisation time. Some * devices might delay media rendering which can be compensated for with this * function. * * After calling this function, this sink will report additional latency and * other sinks will adjust their latency to delay the rendering of their media. * * This function is usually called by subclasses. */ void gst_base_sink_set_render_delay (GstBaseSink * sink, GstClockTime delay) { GstClockTime old_render_delay; g_return_if_fail (GST_IS_BASE_SINK (sink)); GST_OBJECT_LOCK (sink); old_render_delay = sink->priv->render_delay; sink->priv->render_delay = delay; GST_LOG_OBJECT (sink, "set render delay to %" GST_TIME_FORMAT, GST_TIME_ARGS (delay)); GST_OBJECT_UNLOCK (sink); if (delay != old_render_delay) { GST_DEBUG_OBJECT (sink, "posting latency changed"); gst_element_post_message (GST_ELEMENT_CAST (sink), gst_message_new_latency (GST_OBJECT_CAST (sink))); } } /** * gst_base_sink_get_render_delay: * @sink: a #GstBaseSink * * Get the render delay of @sink. see gst_base_sink_set_render_delay() for more * information about the render delay. * * Returns: the render delay of @sink. */ GstClockTime gst_base_sink_get_render_delay (GstBaseSink * sink) { GstClockTimeDiff res; g_return_val_if_fail (GST_IS_BASE_SINK (sink), 0); GST_OBJECT_LOCK (sink); res = sink->priv->render_delay; GST_OBJECT_UNLOCK (sink); return res; } /** * gst_base_sink_set_blocksize: * @sink: a #GstBaseSink * @blocksize: the blocksize in bytes * * Set the number of bytes that the sink will pull when it is operating in pull * mode. */ /* FIXME 0.11: blocksize property should be int, otherwise min>max.. */ void gst_base_sink_set_blocksize (GstBaseSink * sink, guint blocksize) { g_return_if_fail (GST_IS_BASE_SINK (sink)); GST_OBJECT_LOCK (sink); sink->priv->blocksize = blocksize; GST_LOG_OBJECT (sink, "set blocksize to %u", blocksize); GST_OBJECT_UNLOCK (sink); } /** * gst_base_sink_get_blocksize: * @sink: a #GstBaseSink * * Get the number of bytes that the sink will pull when it is operating in pull * mode. * * Returns: the number of bytes @sink will pull in pull mode. */ /* FIXME 0.11: blocksize property should be int, otherwise min>max.. */ guint gst_base_sink_get_blocksize (GstBaseSink * sink) { guint res; g_return_val_if_fail (GST_IS_BASE_SINK (sink), 0); GST_OBJECT_LOCK (sink); res = sink->priv->blocksize; GST_OBJECT_UNLOCK (sink); return res; } /** * gst_base_sink_set_throttle_time: * @sink: a #GstBaseSink * @throttle: the throttle time in nanoseconds * * Set the time that will be inserted between rendered buffers. This * can be used to control the maximum buffers per second that the sink * will render. */ void gst_base_sink_set_throttle_time (GstBaseSink * sink, guint64 throttle) { g_return_if_fail (GST_IS_BASE_SINK (sink)); GST_OBJECT_LOCK (sink); sink->priv->throttle_time = throttle; GST_LOG_OBJECT (sink, "set throttle_time to %" G_GUINT64_FORMAT, throttle); GST_OBJECT_UNLOCK (sink); } /** * gst_base_sink_get_throttle_time: * @sink: a #GstBaseSink * * Get the time that will be inserted between frames to control the * maximum buffers per second. * * Returns: the number of nanoseconds @sink will put between frames. */ guint64 gst_base_sink_get_throttle_time (GstBaseSink * sink) { guint64 res; g_return_val_if_fail (GST_IS_BASE_SINK (sink), 0); GST_OBJECT_LOCK (sink); res = sink->priv->throttle_time; GST_OBJECT_UNLOCK (sink); return res; } static void gst_base_sink_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstBaseSink *sink = GST_BASE_SINK (object); switch (prop_id) { case PROP_SYNC: gst_base_sink_set_sync (sink, g_value_get_boolean (value)); break; case PROP_MAX_LATENESS: gst_base_sink_set_max_lateness (sink, g_value_get_int64 (value)); break; case PROP_QOS: gst_base_sink_set_qos_enabled (sink, g_value_get_boolean (value)); break; case PROP_ASYNC: gst_base_sink_set_async_enabled (sink, g_value_get_boolean (value)); break; case PROP_TS_OFFSET: gst_base_sink_set_ts_offset (sink, g_value_get_int64 (value)); break; case PROP_BLOCKSIZE: gst_base_sink_set_blocksize (sink, g_value_get_uint (value)); break; case PROP_RENDER_DELAY: gst_base_sink_set_render_delay (sink, g_value_get_uint64 (value)); break; case PROP_ENABLE_LAST_SAMPLE: gst_base_sink_set_last_sample_enabled (sink, g_value_get_boolean (value)); break; case PROP_THROTTLE_TIME: gst_base_sink_set_throttle_time (sink, g_value_get_uint64 (value)); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_base_sink_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstBaseSink *sink = GST_BASE_SINK (object); switch (prop_id) { case PROP_SYNC: g_value_set_boolean (value, gst_base_sink_get_sync (sink)); break; case PROP_MAX_LATENESS: g_value_set_int64 (value, gst_base_sink_get_max_lateness (sink)); break; case PROP_QOS: g_value_set_boolean (value, gst_base_sink_is_qos_enabled (sink)); break; case PROP_ASYNC: g_value_set_boolean (value, gst_base_sink_is_async_enabled (sink)); break; case PROP_TS_OFFSET: g_value_set_int64 (value, gst_base_sink_get_ts_offset (sink)); break; case PROP_LAST_SAMPLE: gst_value_take_buffer (value, gst_base_sink_get_last_sample (sink)); break; case PROP_ENABLE_LAST_SAMPLE: g_value_set_boolean (value, gst_base_sink_is_last_sample_enabled (sink)); break; case PROP_BLOCKSIZE: g_value_set_uint (value, gst_base_sink_get_blocksize (sink)); break; case PROP_RENDER_DELAY: g_value_set_uint64 (value, gst_base_sink_get_render_delay (sink)); break; case PROP_THROTTLE_TIME: g_value_set_uint64 (value, gst_base_sink_get_throttle_time (sink)); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static GstCaps * gst_base_sink_default_get_caps (GstBaseSink * sink, GstCaps * filter) { return NULL; } static gboolean gst_base_sink_default_set_caps (GstBaseSink * sink, GstCaps * caps) { return TRUE; } /* with PREROLL_LOCK, STREAM_LOCK */ static gboolean gst_base_sink_commit_state (GstBaseSink * basesink) { /* commit state and proceed to next pending state */ GstState current, next, pending, post_pending; gboolean post_paused = FALSE; gboolean post_async_done = FALSE; gboolean post_playing = FALSE; /* we are certainly not playing async anymore now */ basesink->playing_async = FALSE; GST_OBJECT_LOCK (basesink); current = GST_STATE (basesink); next = GST_STATE_NEXT (basesink); pending = GST_STATE_PENDING (basesink); post_pending = pending; switch (pending) { case GST_STATE_PLAYING: { GST_DEBUG_OBJECT (basesink, "commiting state to PLAYING"); basesink->need_preroll = FALSE; post_async_done = TRUE; basesink->priv->commited = TRUE; post_playing = TRUE; /* post PAUSED too when we were READY */ if (current == GST_STATE_READY) { post_paused = TRUE; } break; } case GST_STATE_PAUSED: GST_DEBUG_OBJECT (basesink, "commiting state to PAUSED"); post_paused = TRUE; post_async_done = TRUE; basesink->priv->commited = TRUE; post_pending = GST_STATE_VOID_PENDING; break; case GST_STATE_READY: case GST_STATE_NULL: goto stopping; case GST_STATE_VOID_PENDING: goto nothing_pending; default: break; } /* we can report latency queries now */ basesink->priv->have_latency = TRUE; GST_STATE (basesink) = pending; GST_STATE_NEXT (basesink) = GST_STATE_VOID_PENDING; GST_STATE_PENDING (basesink) = GST_STATE_VOID_PENDING; GST_STATE_RETURN (basesink) = GST_STATE_CHANGE_SUCCESS; GST_OBJECT_UNLOCK (basesink); if (post_paused) { GST_DEBUG_OBJECT (basesink, "posting PAUSED state change message"); gst_element_post_message (GST_ELEMENT_CAST (basesink), gst_message_new_state_changed (GST_OBJECT_CAST (basesink), current, next, post_pending)); } if (post_async_done) { GST_DEBUG_OBJECT (basesink, "posting async-done message"); gst_element_post_message (GST_ELEMENT_CAST (basesink), gst_message_new_async_done (GST_OBJECT_CAST (basesink), GST_CLOCK_TIME_NONE)); } if (post_playing) { GST_DEBUG_OBJECT (basesink, "posting PLAYING state change message"); gst_element_post_message (GST_ELEMENT_CAST (basesink), gst_message_new_state_changed (GST_OBJECT_CAST (basesink), next, pending, GST_STATE_VOID_PENDING)); } GST_STATE_BROADCAST (basesink); return TRUE; nothing_pending: { /* Depending on the state, set our vars. We get in this situation when the * state change function got a change to update the state vars before the * streaming thread did. This is fine but we need to make sure that we * update the need_preroll var since it was TRUE when we got here and might * become FALSE if we got to PLAYING. */ GST_DEBUG_OBJECT (basesink, "nothing to commit, now in %s", gst_element_state_get_name (current)); switch (current) { case GST_STATE_PLAYING: basesink->need_preroll = FALSE; break; case GST_STATE_PAUSED: basesink->need_preroll = TRUE; break; default: basesink->need_preroll = FALSE; basesink->flushing = TRUE; break; } /* we can report latency queries now */ basesink->priv->have_latency = TRUE; GST_OBJECT_UNLOCK (basesink); return TRUE; } stopping: { /* app is going to READY */ GST_DEBUG_OBJECT (basesink, "stopping"); basesink->need_preroll = FALSE; basesink->flushing = TRUE; GST_OBJECT_UNLOCK (basesink); return FALSE; } } static void start_stepping (GstBaseSink * sink, GstSegment * segment, GstStepInfo * pending, GstStepInfo * current) { gint64 end; GstMessage *message; GST_DEBUG_OBJECT (sink, "update pending step"); GST_OBJECT_LOCK (sink); memcpy (current, pending, sizeof (GstStepInfo)); pending->valid = FALSE; GST_OBJECT_UNLOCK (sink); /* post message first */ message = gst_message_new_step_start (GST_OBJECT (sink), TRUE, current->format, current->amount, current->rate, current->flush, current->intermediate); gst_message_set_seqnum (message, current->seqnum); gst_element_post_message (GST_ELEMENT (sink), message); /* get the running time of where we paused and remember it */ current->start = gst_element_get_start_time (GST_ELEMENT_CAST (sink)); gst_segment_set_running_time (segment, GST_FORMAT_TIME, current->start); /* set the new rate for the remainder of the segment */ current->start_rate = segment->rate; segment->rate *= current->rate; /* save values */ if (segment->rate > 0.0) current->start_stop = segment->stop; else current->start_start = segment->start; if (current->format == GST_FORMAT_TIME) { /* calculate the running-time when the step operation should stop */ if (current->amount != -1) end = current->start + current->amount; else end = -1; if (!current->flush) { gint64 position; /* update the segment clipping regions for non-flushing seeks */ if (segment->rate > 0.0) { if (end != -1) position = gst_segment_to_position (segment, GST_FORMAT_TIME, end); else position = segment->stop; segment->stop = position; segment->position = position; } else { if (end != -1) position = gst_segment_to_position (segment, GST_FORMAT_TIME, end); else position = segment->start; segment->time = position; segment->start = position; segment->position = position; } } } GST_DEBUG_OBJECT (sink, "segment now %" GST_SEGMENT_FORMAT, segment); GST_DEBUG_OBJECT (sink, "step started at running_time %" GST_TIME_FORMAT, GST_TIME_ARGS (current->start)); GST_DEBUG_OBJECT (sink, "step amount: %" G_GUINT64_FORMAT ", format: %s, " "rate: %f", current->amount, gst_format_get_name (current->format), current->rate); } static void stop_stepping (GstBaseSink * sink, GstSegment * segment, GstStepInfo * current, gint64 rstart, gint64 rstop, gboolean eos) { gint64 stop, position; GstMessage *message; GST_DEBUG_OBJECT (sink, "step complete"); if (segment->rate > 0.0) stop = rstart; else stop = rstop; GST_DEBUG_OBJECT (sink, "step stop at running_time %" GST_TIME_FORMAT, GST_TIME_ARGS (stop)); if (stop == -1) current->duration = current->position; else current->duration = stop - current->start; GST_DEBUG_OBJECT (sink, "step elapsed running_time %" GST_TIME_FORMAT, GST_TIME_ARGS (current->duration)); position = current->start + current->duration; /* now move the segment to the new running time */ gst_segment_set_running_time (segment, GST_FORMAT_TIME, position); if (current->flush) { /* and remove the time we flushed, start time did not change */ segment->base = current->start; } else { /* start time is now the stepped position */ gst_element_set_start_time (GST_ELEMENT_CAST (sink), position); } /* restore the previous rate */ segment->rate = current->start_rate; if (segment->rate > 0.0) segment->stop = current->start_stop; else segment->start = current->start_start; /* post the step done when we know the stepped duration in TIME */ message = gst_message_new_step_done (GST_OBJECT_CAST (sink), current->format, current->amount, current->rate, current->flush, current->intermediate, current->duration, eos); gst_message_set_seqnum (message, current->seqnum); gst_element_post_message (GST_ELEMENT_CAST (sink), message); if (!current->intermediate) sink->need_preroll = current->need_preroll; /* and the current step info finished and becomes invalid */ current->valid = FALSE; } static gboolean handle_stepping (GstBaseSink * sink, GstSegment * segment, GstStepInfo * current, guint64 * cstart, guint64 * cstop, guint64 * rstart, guint64 * rstop) { gboolean step_end = FALSE; /* stepping never stops */ if (current->amount == -1) return FALSE; /* see if we need to skip this buffer because of stepping */ switch (current->format) { case GST_FORMAT_TIME: { guint64 end; guint64 first, last; gdouble abs_rate; if (segment->rate > 0.0) { if (segment->stop == *cstop) *rstop = *rstart + current->amount; first = *rstart; last = *rstop; } else { if (segment->start == *cstart) *rstart = *rstop + current->amount; first = *rstop; last = *rstart; } end = current->start + current->amount; current->position = first - current->start; abs_rate = ABS (segment->rate); if (G_UNLIKELY (abs_rate != 1.0)) current->position /= abs_rate; GST_DEBUG_OBJECT (sink, "buffer: %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT, GST_TIME_ARGS (first), GST_TIME_ARGS (last)); GST_DEBUG_OBJECT (sink, "got time step %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT "/%" GST_TIME_FORMAT, GST_TIME_ARGS (current->position), GST_TIME_ARGS (last - current->start), GST_TIME_ARGS (current->amount)); if ((current->flush && current->position >= current->amount) || last >= end) { GST_DEBUG_OBJECT (sink, "step ended, we need clipping"); step_end = TRUE; if (segment->rate > 0.0) { *rstart = end; *cstart = gst_segment_to_position (segment, GST_FORMAT_TIME, end); } else { *rstop = end; *cstop = gst_segment_to_position (segment, GST_FORMAT_TIME, end); } } GST_DEBUG_OBJECT (sink, "cstart %" GST_TIME_FORMAT ", rstart %" GST_TIME_FORMAT, GST_TIME_ARGS (*cstart), GST_TIME_ARGS (*rstart)); GST_DEBUG_OBJECT (sink, "cstop %" GST_TIME_FORMAT ", rstop %" GST_TIME_FORMAT, GST_TIME_ARGS (*cstop), GST_TIME_ARGS (*rstop)); break; } case GST_FORMAT_BUFFERS: GST_DEBUG_OBJECT (sink, "got default step %" G_GUINT64_FORMAT "/%" G_GUINT64_FORMAT, current->position, current->amount); if (current->position < current->amount) { current->position++; } else { step_end = TRUE; } break; case GST_FORMAT_DEFAULT: default: GST_DEBUG_OBJECT (sink, "got unknown step %" G_GUINT64_FORMAT "/%" G_GUINT64_FORMAT, current->position, current->amount); break; } return step_end; } /* with STREAM_LOCK, PREROLL_LOCK * * Returns TRUE if the object needs synchronisation and takes therefore * part in prerolling. * * rsstart/rsstop contain the start/stop in stream time. * rrstart/rrstop contain the start/stop in running time. */ static gboolean gst_base_sink_get_sync_times (GstBaseSink * basesink, GstMiniObject * obj, GstClockTime * rsstart, GstClockTime * rsstop, GstClockTime * rrstart, GstClockTime * rrstop, gboolean * do_sync, gboolean * stepped, GstStepInfo * step, gboolean * step_end) { GstBaseSinkClass *bclass; GstClockTime start, stop; /* raw start/stop timestamps */ guint64 cstart, cstop; /* clipped raw timestamps */ guint64 rstart, rstop; /* clipped timestamps converted to running time */ GstClockTime sstart, sstop; /* clipped timestamps converted to stream time */ GstFormat format; GstBaseSinkPrivate *priv; GstSegment *segment; gboolean eos; priv = basesink->priv; segment = &basesink->segment; bclass = GST_BASE_SINK_GET_CLASS (basesink); again: /* start with nothing */ start = stop = GST_CLOCK_TIME_NONE; eos = FALSE; if (G_UNLIKELY (GST_IS_EVENT (obj))) { GstEvent *event = GST_EVENT_CAST (obj); switch (GST_EVENT_TYPE (event)) { /* EOS event needs syncing */ case GST_EVENT_EOS: { if (segment->rate >= 0.0) { sstart = sstop = priv->current_sstop; if (!GST_CLOCK_TIME_IS_VALID (sstart)) { /* we have not seen a buffer yet, use the segment values */ sstart = sstop = gst_segment_to_stream_time (segment, segment->format, segment->stop); } } else { sstart = sstop = priv->current_sstart; if (!GST_CLOCK_TIME_IS_VALID (sstart)) { /* we have not seen a buffer yet, use the segment values */ sstart = sstop = gst_segment_to_stream_time (segment, segment->format, segment->start); } } rstart = rstop = priv->eos_rtime; *do_sync = rstart != -1; GST_DEBUG_OBJECT (basesink, "sync times for EOS %" GST_TIME_FORMAT, GST_TIME_ARGS (rstart)); /* if we are stepping, we end now */ *step_end = step->valid; eos = TRUE; goto eos_done; } case GST_EVENT_GAP: { GstClockTime timestamp, duration; gst_event_parse_gap (event, ×tamp, &duration); GST_DEBUG_OBJECT (basesink, "Got Gap time %" GST_TIME_FORMAT " duration %" GST_TIME_FORMAT, GST_TIME_ARGS (timestamp), GST_TIME_ARGS (duration)); if (GST_CLOCK_TIME_IS_VALID (timestamp)) { start = timestamp; if (GST_CLOCK_TIME_IS_VALID (duration)) stop = start + duration; } *do_sync = TRUE; break; } default: /* other events do not need syncing */ return FALSE; } } else { /* else do buffer sync code */ GstBuffer *buffer = GST_BUFFER_CAST (obj); /* just get the times to see if we need syncing, if the start returns -1 we * don't sync. */ if (bclass->get_times) bclass->get_times (basesink, buffer, &start, &stop); if (!GST_CLOCK_TIME_IS_VALID (start)) { /* we don't need to sync but we still want to get the timestamps for * tracking the position */ gst_base_sink_default_get_times (basesink, buffer, &start, &stop); *do_sync = FALSE; } else { *do_sync = TRUE; } } GST_DEBUG_OBJECT (basesink, "got times start: %" GST_TIME_FORMAT ", stop: %" GST_TIME_FORMAT ", do_sync %d", GST_TIME_ARGS (start), GST_TIME_ARGS (stop), *do_sync); /* collect segment and format for code clarity */ format = segment->format; /* clip */ if (G_UNLIKELY (!gst_segment_clip (segment, format, start, stop, &cstart, &cstop))) { if (step->valid) { GST_DEBUG_OBJECT (basesink, "step out of segment"); /* when we are stepping, pretend we're at the end of the segment */ if (segment->rate > 0.0) { cstart = segment->stop; cstop = segment->stop; } else { cstart = segment->start; cstop = segment->start; } goto do_times; } goto out_of_segment; } if (G_UNLIKELY (start != cstart || stop != cstop)) { GST_DEBUG_OBJECT (basesink, "clipped to: start %" GST_TIME_FORMAT ", stop: %" GST_TIME_FORMAT, GST_TIME_ARGS (cstart), GST_TIME_ARGS (cstop)); } /* set last stop position */ if (G_LIKELY (stop != GST_CLOCK_TIME_NONE && cstop != GST_CLOCK_TIME_NONE)) segment->position = cstop; else segment->position = cstart; do_times: rstart = gst_segment_to_running_time (segment, format, cstart); rstop = gst_segment_to_running_time (segment, format, cstop); if (G_UNLIKELY (step->valid)) { if (!(*step_end = handle_stepping (basesink, segment, step, &cstart, &cstop, &rstart, &rstop))) { /* step is still busy, we discard data when we are flushing */ *stepped = step->flush; GST_DEBUG_OBJECT (basesink, "stepping busy"); } } /* this can produce wrong values if we accumulated non-TIME segments. If this happens, * upstream is behaving very badly */ sstart = gst_segment_to_stream_time (segment, format, cstart); sstop = gst_segment_to_stream_time (segment, format, cstop); eos_done: /* eos_done label only called when doing EOS, we also stop stepping then */ if (*step_end && step->flush) { GST_DEBUG_OBJECT (basesink, "flushing step ended"); stop_stepping (basesink, segment, step, rstart, rstop, eos); *step_end = FALSE; /* re-determine running start times for adjusted segment * (which has a flushed amount of running/accumulated time removed) */ if (!GST_IS_EVENT (obj)) { GST_DEBUG_OBJECT (basesink, "refresh sync times"); goto again; } } /* save times */ *rsstart = sstart; *rsstop = sstop; *rrstart = rstart; *rrstop = rstop; /* buffers and EOS always need syncing and preroll */ return TRUE; /* special cases */ out_of_segment: { /* we usually clip in the chain function already but stepping could cause * the segment to be updated later. we return FALSE so that we don't try * to sync on it. */ GST_LOG_OBJECT (basesink, "buffer skipped, not in segment"); return FALSE; } } /* with STREAM_LOCK, PREROLL_LOCK, LOCK * adjust a timestamp with the latency and timestamp offset. This function does * not adjust for the render delay. */ static GstClockTime gst_base_sink_adjust_time (GstBaseSink * basesink, GstClockTime time) { GstClockTimeDiff ts_offset; /* don't do anything funny with invalid timestamps */ if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (time))) return time; time += basesink->priv->latency; /* apply offset, be carefull for underflows */ ts_offset = basesink->priv->ts_offset; if (ts_offset < 0) { ts_offset = -ts_offset; if (ts_offset < time) time -= ts_offset; else time = 0; } else time += ts_offset; /* subtract the render delay again, which was included in the latency */ if (time > basesink->priv->render_delay) time -= basesink->priv->render_delay; else time = 0; return time; } /** * gst_base_sink_wait_clock: * @sink: the sink * @time: the running_time to be reached * @jitter: (out) (allow-none): the jitter to be filled with time diff, or NULL * * This function will block until @time is reached. It is usually called by * subclasses that use their own internal synchronisation. * * If @time is not valid, no sycnhronisation is done and #GST_CLOCK_BADTIME is * returned. Likewise, if synchronisation is disabled in the element or there * is no clock, no synchronisation is done and #GST_CLOCK_BADTIME is returned. * * This function should only be called with the PREROLL_LOCK held, like when * receiving an EOS event in the #GstBaseSinkClass.event() vmethod or when * receiving a buffer in * the #GstBaseSinkClass.render() vmethod. * * The @time argument should be the running_time of when this method should * return and is not adjusted with any latency or offset configured in the * sink. * * Returns: #GstClockReturn */ GstClockReturn gst_base_sink_wait_clock (GstBaseSink * sink, GstClockTime time, GstClockTimeDiff * jitter) { GstClockReturn ret; GstClock *clock; GstClockTime base_time; if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (time))) goto invalid_time; GST_OBJECT_LOCK (sink); if (G_UNLIKELY (!sink->sync)) goto no_sync; if (G_UNLIKELY ((clock = GST_ELEMENT_CLOCK (sink)) == NULL)) goto no_clock; base_time = GST_ELEMENT_CAST (sink)->base_time; GST_LOG_OBJECT (sink, "time %" GST_TIME_FORMAT ", base_time %" GST_TIME_FORMAT, GST_TIME_ARGS (time), GST_TIME_ARGS (base_time)); /* add base_time to running_time to get the time against the clock */ time += base_time; /* Re-use existing clockid if available */ /* FIXME: Casting to GstClockEntry only works because the types * are the same */ if (G_LIKELY (sink->priv->cached_clock_id != NULL && GST_CLOCK_ENTRY_CLOCK ((GstClockEntry *) sink->priv-> cached_clock_id) == clock)) { if (!gst_clock_single_shot_id_reinit (clock, sink->priv->cached_clock_id, time)) { gst_clock_id_unref (sink->priv->cached_clock_id); sink->priv->cached_clock_id = gst_clock_new_single_shot_id (clock, time); } } else { if (sink->priv->cached_clock_id != NULL) gst_clock_id_unref (sink->priv->cached_clock_id); sink->priv->cached_clock_id = gst_clock_new_single_shot_id (clock, time); } GST_OBJECT_UNLOCK (sink); /* A blocking wait is performed on the clock. We save the ClockID * so we can unlock the entry at any time. While we are blocking, we * release the PREROLL_LOCK so that other threads can interrupt the * entry. */ sink->clock_id = sink->priv->cached_clock_id; /* release the preroll lock while waiting */ GST_BASE_SINK_PREROLL_UNLOCK (sink); ret = gst_clock_id_wait (sink->priv->cached_clock_id, jitter); GST_BASE_SINK_PREROLL_LOCK (sink); sink->clock_id = NULL; return ret; /* no syncing needed */ invalid_time: { GST_DEBUG_OBJECT (sink, "time not valid, no sync needed"); return GST_CLOCK_BADTIME; } no_sync: { GST_DEBUG_OBJECT (sink, "sync disabled"); GST_OBJECT_UNLOCK (sink); return GST_CLOCK_BADTIME; } no_clock: { GST_DEBUG_OBJECT (sink, "no clock, can't sync"); GST_OBJECT_UNLOCK (sink); return GST_CLOCK_BADTIME; } } /** * gst_base_sink_wait_preroll: * @sink: the sink * * If the #GstBaseSinkClass.render() 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 render 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). * * This function should only be called with the PREROLL_LOCK held, like in the * render function. * * Returns: #GST_FLOW_OK if the preroll completed and processing can * continue. Any other return value should be returned from the render vmethod. */ GstFlowReturn gst_base_sink_wait_preroll (GstBaseSink * sink) { sink->have_preroll = TRUE; GST_DEBUG_OBJECT (sink, "waiting in preroll for flush or PLAYING"); /* block until the state changes, or we get a flush, or something */ GST_BASE_SINK_PREROLL_WAIT (sink); sink->have_preroll = FALSE; if (G_UNLIKELY (sink->flushing)) goto stopping; if (G_UNLIKELY (sink->priv->step_unlock)) goto step_unlocked; GST_DEBUG_OBJECT (sink, "continue after preroll"); return GST_FLOW_OK; /* ERRORS */ stopping: { GST_DEBUG_OBJECT (sink, "preroll interrupted because of flush"); return GST_FLOW_FLUSHING; } step_unlocked: { sink->priv->step_unlock = FALSE; GST_DEBUG_OBJECT (sink, "preroll interrupted because of step"); return GST_FLOW_STEP; } } /** * gst_base_sink_do_preroll: * @sink: the sink * @obj: (transfer none): the mini object that caused the preroll * * If the @sink spawns its own thread for pulling buffers from upstream it * should call this method after it has pulled a buffer. If the element needed * to preroll, this function will perform the preroll and will then block * until the element state is changed. * * This function should be called with the PREROLL_LOCK held. * * Returns: #GST_FLOW_OK if the preroll completed and processing can * continue. Any other return value should be returned from the render vmethod. */ GstFlowReturn gst_base_sink_do_preroll (GstBaseSink * sink, GstMiniObject * obj) { GstFlowReturn ret; while (G_UNLIKELY (sink->need_preroll)) { GST_DEBUG_OBJECT (sink, "prerolling object %p", obj); /* if it's a buffer, we need to call the preroll method */ if (sink->priv->call_preroll) { GstBaseSinkClass *bclass; GstBuffer *buf; if (GST_IS_BUFFER_LIST (obj)) { buf = gst_buffer_list_get (GST_BUFFER_LIST_CAST (obj), 0); g_assert (NULL != buf); } else if (GST_IS_BUFFER (obj)) { buf = GST_BUFFER_CAST (obj); /* For buffer lists do not set last buffer for now */ gst_base_sink_set_last_buffer (sink, buf); } else buf = NULL; if (buf) { GST_DEBUG_OBJECT (sink, "preroll buffer %" GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf))); bclass = GST_BASE_SINK_GET_CLASS (sink); if (bclass->prepare) if ((ret = bclass->prepare (sink, buf)) != GST_FLOW_OK) goto prepare_canceled; if (bclass->preroll) if ((ret = bclass->preroll (sink, buf)) != GST_FLOW_OK) goto preroll_canceled; sink->priv->call_preroll = FALSE; } } /* commit state */ if (G_LIKELY (sink->playing_async)) { if (G_UNLIKELY (!gst_base_sink_commit_state (sink))) goto stopping; } /* need to recheck here because the commit state could have * made us not need the preroll anymore */ if (G_LIKELY (sink->need_preroll)) { /* block until the state changes, or we get a flush, or something */ ret = gst_base_sink_wait_preroll (sink); if ((ret != GST_FLOW_OK) && (ret != GST_FLOW_STEP)) goto preroll_failed; } } return GST_FLOW_OK; /* ERRORS */ prepare_canceled: { GST_DEBUG_OBJECT (sink, "prepare failed, abort state"); gst_element_abort_state (GST_ELEMENT_CAST (sink)); return ret; } preroll_canceled: { GST_DEBUG_OBJECT (sink, "preroll failed, abort state"); gst_element_abort_state (GST_ELEMENT_CAST (sink)); return ret; } stopping: { GST_DEBUG_OBJECT (sink, "stopping while commiting state"); return GST_FLOW_FLUSHING; } preroll_failed: { GST_DEBUG_OBJECT (sink, "preroll failed: %s", gst_flow_get_name (ret)); return ret; } } /** * gst_base_sink_wait: * @sink: the sink * @time: the running_time to be reached * @jitter: (out) (allow-none): the jitter to be filled with time diff, or NULL * * This function will wait for preroll to complete and will then block until @time * is reached. It is usually called by subclasses that use their own internal * synchronisation but want to let some synchronization (like EOS) be handled * by the base class. * * This function should only be called with the PREROLL_LOCK held (like when * receiving an EOS event in the ::event vmethod or when handling buffers in * ::render). * * The @time argument should be the running_time of when the timeout should happen * and will be adjusted with any latency and offset configured in the sink. * * Returns: #GstFlowReturn */ GstFlowReturn gst_base_sink_wait (GstBaseSink * sink, GstClockTime time, GstClockTimeDiff * jitter) { GstClockReturn status; GstFlowReturn ret; do { GstClockTime stime; GST_DEBUG_OBJECT (sink, "checking preroll"); /* first wait for the playing state before we can continue */ while (G_UNLIKELY (sink->need_preroll)) { ret = gst_base_sink_wait_preroll (sink); if ((ret != GST_FLOW_OK) && (ret != GST_FLOW_STEP)) goto flushing; } /* preroll done, we can sync since we are in PLAYING now. */ GST_DEBUG_OBJECT (sink, "possibly waiting for clock to reach %" GST_TIME_FORMAT, GST_TIME_ARGS (time)); /* compensate for latency, ts_offset and render delay */ stime = gst_base_sink_adjust_time (sink, time); /* wait for the clock, this can be interrupted because we got shut down or * we PAUSED. */ status = gst_base_sink_wait_clock (sink, stime, jitter); GST_DEBUG_OBJECT (sink, "clock returned %d", status); /* invalid time, no clock or sync disabled, just continue then */ if (status == GST_CLOCK_BADTIME) break; /* waiting could have been interrupted and we can be flushing now */ if (G_UNLIKELY (sink->flushing)) goto flushing; /* retry if we got unscheduled, which means we did not reach the timeout * yet. if some other error occures, we continue. */ } while (status == GST_CLOCK_UNSCHEDULED); GST_DEBUG_OBJECT (sink, "end of stream"); return GST_FLOW_OK; /* ERRORS */ flushing: { GST_DEBUG_OBJECT (sink, "we are flushing"); return GST_FLOW_FLUSHING; } } /* with STREAM_LOCK, PREROLL_LOCK * * Make sure we are in PLAYING and synchronize an object to the clock. * * If we need preroll, we are not in PLAYING. We try to commit the state * if needed and then block if we still are not PLAYING. * * We start waiting on the clock in PLAYING. If we got interrupted, we * immediately try to re-preroll. * * Some objects do not need synchronisation (most events) and so this function * immediately returns GST_FLOW_OK. * * for objects that arrive later than max-lateness to be synchronized to the * clock have the @late boolean set to TRUE. * * This function keeps a running average of the jitter (the diff between the * clock time and the requested sync time). The jitter is negative for * objects that arrive in time and positive for late buffers. * * does not take ownership of obj. */ static GstFlowReturn gst_base_sink_do_sync (GstBaseSink * basesink, GstMiniObject * obj, gboolean * late, gboolean * step_end) { GstClockTimeDiff jitter = 0; gboolean syncable; GstClockReturn status = GST_CLOCK_OK; GstClockTime rstart, rstop, sstart, sstop, stime; gboolean do_sync; GstBaseSinkPrivate *priv; GstFlowReturn ret; GstStepInfo *current, *pending; gboolean stepped; priv = basesink->priv; do_step: sstart = sstop = rstart = rstop = GST_CLOCK_TIME_NONE; do_sync = TRUE; stepped = FALSE; priv->current_rstart = GST_CLOCK_TIME_NONE; /* get stepping info */ current = &priv->current_step; pending = &priv->pending_step; /* get timing information for this object against the render segment */ syncable = gst_base_sink_get_sync_times (basesink, obj, &sstart, &sstop, &rstart, &rstop, &do_sync, &stepped, current, step_end); if (G_UNLIKELY (stepped)) goto step_skipped; /* a syncable object needs to participate in preroll and * clocking. All buffers and EOS are syncable. */ if (G_UNLIKELY (!syncable)) goto not_syncable; /* store timing info for current object */ priv->current_rstart = rstart; priv->current_rstop = (GST_CLOCK_TIME_IS_VALID (rstop) ? rstop : rstart); /* save sync time for eos when the previous object needed sync */ priv->eos_rtime = (do_sync ? priv->current_rstop : GST_CLOCK_TIME_NONE); /* calculate inter frame spacing */ if (G_UNLIKELY (priv->prev_rstart != -1 && priv->prev_rstart < rstart)) { GstClockTime in_diff; in_diff = rstart - priv->prev_rstart; if (priv->avg_in_diff == -1) priv->avg_in_diff = in_diff; else priv->avg_in_diff = UPDATE_RUNNING_AVG (priv->avg_in_diff, in_diff); GST_LOG_OBJECT (basesink, "avg frame diff %" GST_TIME_FORMAT, GST_TIME_ARGS (priv->avg_in_diff)); } priv->prev_rstart = rstart; if (G_UNLIKELY (priv->earliest_in_time != -1 && rstart < priv->earliest_in_time)) goto qos_dropped; again: /* first do preroll, this makes sure we commit our state * to PAUSED and can continue to PLAYING. We cannot perform * any clock sync in PAUSED because there is no clock. */ ret = gst_base_sink_do_preroll (basesink, obj); if (G_UNLIKELY (ret != GST_FLOW_OK)) goto preroll_failed; /* update the segment with a pending step if the current one is invalid and we * have a new pending one. We only accept new step updates after a preroll */ if (G_UNLIKELY (pending->valid && !current->valid)) { start_stepping (basesink, &basesink->segment, pending, current); goto do_step; } /* After rendering we store the position of the last buffer so that we can use * it to report the position. We need to take the lock here. */ GST_OBJECT_LOCK (basesink); priv->current_sstart = sstart; priv->current_sstop = (GST_CLOCK_TIME_IS_VALID (sstop) ? sstop : sstart); GST_OBJECT_UNLOCK (basesink); if (!do_sync) goto done; /* adjust for latency */ stime = gst_base_sink_adjust_time (basesink, rstart); /* preroll done, we can sync since we are in PLAYING now. */ GST_DEBUG_OBJECT (basesink, "possibly waiting for clock to reach %" GST_TIME_FORMAT ", adjusted %" GST_TIME_FORMAT, GST_TIME_ARGS (rstart), GST_TIME_ARGS (stime)); /* This function will return immediately if start == -1, no clock * or sync is disabled with GST_CLOCK_BADTIME. */ status = gst_base_sink_wait_clock (basesink, stime, &jitter); GST_DEBUG_OBJECT (basesink, "clock returned %d, jitter %c%" GST_TIME_FORMAT, status, (jitter < 0 ? '-' : ' '), GST_TIME_ARGS (ABS (jitter))); /* invalid time, no clock or sync disabled, just render */ if (status == GST_CLOCK_BADTIME) goto done; /* waiting could have been interrupted and we can be flushing now */ if (G_UNLIKELY (basesink->flushing)) goto flushing; /* check for unlocked by a state change, we are not flushing so * we can try to preroll on the current buffer. */ if (G_UNLIKELY (status == GST_CLOCK_UNSCHEDULED)) { GST_DEBUG_OBJECT (basesink, "unscheduled, waiting some more"); priv->call_preroll = TRUE; goto again; } /* successful syncing done, record observation */ priv->current_jitter = jitter; /* check if the object should be dropped */ *late = gst_base_sink_is_too_late (basesink, obj, rstart, rstop, status, jitter); done: return GST_FLOW_OK; /* ERRORS */ step_skipped: { GST_DEBUG_OBJECT (basesink, "skipped stepped object %p", obj); *late = TRUE; return GST_FLOW_OK; } not_syncable: { GST_DEBUG_OBJECT (basesink, "non syncable object %p", obj); return GST_FLOW_OK; } qos_dropped: { GST_DEBUG_OBJECT (basesink, "dropped because of QoS %p", obj); *late = TRUE; return GST_FLOW_OK; } flushing: { GST_DEBUG_OBJECT (basesink, "we are flushing"); return GST_FLOW_FLUSHING; } preroll_failed: { GST_DEBUG_OBJECT (basesink, "preroll failed"); *step_end = FALSE; return ret; } } static gboolean gst_base_sink_send_qos (GstBaseSink * basesink, GstQOSType type, gdouble proportion, GstClockTime time, GstClockTimeDiff diff) { GstEvent *event; gboolean res; /* generate Quality-of-Service event */ GST_CAT_DEBUG_OBJECT (GST_CAT_QOS, basesink, "qos: type %d, proportion: %lf, diff %" G_GINT64_FORMAT ", timestamp %" GST_TIME_FORMAT, type, proportion, diff, GST_TIME_ARGS (time)); event = gst_event_new_qos (type, proportion, diff, time); /* send upstream */ res = gst_pad_push_event (basesink->sinkpad, event); return res; } static void gst_base_sink_perform_qos (GstBaseSink * sink, gboolean dropped) { GstBaseSinkPrivate *priv; GstClockTime start, stop; GstClockTimeDiff jitter; GstClockTime pt, entered, left; GstClockTime duration; gdouble rate; priv = sink->priv; start = priv->current_rstart; if (priv->current_step.valid) return; /* if Quality-of-Service disabled, do nothing */ if (!g_atomic_int_get (&priv->qos_enabled) || !GST_CLOCK_TIME_IS_VALID (start)) return; stop = priv->current_rstop; jitter = priv->current_jitter; if (jitter < 0) { /* this is the time the buffer entered the sink */ if (start < -jitter) entered = 0; else entered = start + jitter; left = start; } else { /* this is the time the buffer entered the sink */ entered = start + jitter; /* this is the time the buffer left the sink */ left = start + jitter; } /* calculate duration of the buffer */ if (GST_CLOCK_TIME_IS_VALID (stop) && stop != start) duration = stop - start; else duration = priv->avg_in_diff; /* if we have the time when the last buffer left us, calculate * processing time */ if (GST_CLOCK_TIME_IS_VALID (priv->last_left)) { if (entered > priv->last_left) { pt = entered - priv->last_left; } else { pt = 0; } } else { pt = priv->avg_pt; } GST_CAT_DEBUG_OBJECT (GST_CAT_QOS, sink, "start: %" GST_TIME_FORMAT ", stop %" GST_TIME_FORMAT ", entered %" GST_TIME_FORMAT ", left %" GST_TIME_FORMAT ", pt: %" GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT ",jitter %" G_GINT64_FORMAT, GST_TIME_ARGS (start), GST_TIME_ARGS (stop), GST_TIME_ARGS (entered), GST_TIME_ARGS (left), GST_TIME_ARGS (pt), GST_TIME_ARGS (duration), jitter); GST_CAT_DEBUG_OBJECT (GST_CAT_QOS, sink, "avg_duration: %" GST_TIME_FORMAT ", avg_pt: %" GST_TIME_FORMAT ", avg_rate: %g", GST_TIME_ARGS (priv->avg_duration), GST_TIME_ARGS (priv->avg_pt), priv->avg_rate); /* collect running averages. for first observations, we copy the * values */ if (!GST_CLOCK_TIME_IS_VALID (priv->avg_duration)) priv->avg_duration = duration; else priv->avg_duration = UPDATE_RUNNING_AVG (priv->avg_duration, duration); if (!GST_CLOCK_TIME_IS_VALID (priv->avg_pt)) priv->avg_pt = pt; else priv->avg_pt = UPDATE_RUNNING_AVG (priv->avg_pt, pt); if (priv->avg_duration != 0) rate = gst_guint64_to_gdouble (priv->avg_pt) / gst_guint64_to_gdouble (priv->avg_duration); else rate = 1.0; if (GST_CLOCK_TIME_IS_VALID (priv->last_left)) { if (dropped || priv->avg_rate < 0.0) { priv->avg_rate = rate; } else { if (rate > 1.0) priv->avg_rate = UPDATE_RUNNING_AVG_N (priv->avg_rate, rate); else priv->avg_rate = UPDATE_RUNNING_AVG_P (priv->avg_rate, rate); } } GST_CAT_DEBUG_OBJECT (GST_CAT_QOS, sink, "updated: avg_duration: %" GST_TIME_FORMAT ", avg_pt: %" GST_TIME_FORMAT ", avg_rate: %g", GST_TIME_ARGS (priv->avg_duration), GST_TIME_ARGS (priv->avg_pt), priv->avg_rate); if (priv->avg_rate >= 0.0) { GstQOSType type; GstClockTimeDiff diff; /* if we have a valid rate, start sending QoS messages */ if (priv->current_jitter < 0) { /* make sure we never go below 0 when adding the jitter to the * timestamp. */ if (priv->current_rstart < -priv->current_jitter) priv->current_jitter = -priv->current_rstart; } if (priv->throttle_time > 0) { diff = priv->throttle_time; type = GST_QOS_TYPE_THROTTLE; } else { diff = priv->current_jitter; if (diff <= 0) type = GST_QOS_TYPE_OVERFLOW; else type = GST_QOS_TYPE_UNDERFLOW; } gst_base_sink_send_qos (sink, type, priv->avg_rate, priv->current_rstart, diff); } /* record when this buffer will leave us */ priv->last_left = left; } /* reset all qos measuring */ static void gst_base_sink_reset_qos (GstBaseSink * sink) { GstBaseSinkPrivate *priv; priv = sink->priv; priv->last_render_time = GST_CLOCK_TIME_NONE; priv->prev_rstart = GST_CLOCK_TIME_NONE; priv->earliest_in_time = GST_CLOCK_TIME_NONE; priv->last_left = GST_CLOCK_TIME_NONE; priv->avg_duration = GST_CLOCK_TIME_NONE; priv->avg_pt = GST_CLOCK_TIME_NONE; priv->avg_rate = -1.0; priv->avg_render = GST_CLOCK_TIME_NONE; priv->avg_in_diff = GST_CLOCK_TIME_NONE; priv->rendered = 0; priv->dropped = 0; } /* Checks if the object was scheduled too late. * * rstart/rstop contain the running_time start and stop values * of the object. * * status and jitter contain the return values from the clock wait. * * returns TRUE if the buffer was too late. */ static gboolean gst_base_sink_is_too_late (GstBaseSink * basesink, GstMiniObject * obj, GstClockTime rstart, GstClockTime rstop, GstClockReturn status, GstClockTimeDiff jitter) { gboolean late; guint64 max_lateness; GstBaseSinkPrivate *priv; priv = basesink->priv; late = FALSE; /* only for objects that were too late */ if (G_LIKELY (status != GST_CLOCK_EARLY)) goto in_time; max_lateness = basesink->max_lateness; /* check if frame dropping is enabled */ if (max_lateness == -1) goto no_drop; /* only check for buffers */ if (G_UNLIKELY (!GST_IS_BUFFER (obj))) goto not_buffer; /* can't do check if we don't have a timestamp */ if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (rstart))) goto no_timestamp; /* we can add a valid stop time */ if (GST_CLOCK_TIME_IS_VALID (rstop)) max_lateness += rstop; else { max_lateness += rstart; /* no stop time, use avg frame diff */ if (priv->avg_in_diff != -1) max_lateness += priv->avg_in_diff; } /* if the jitter bigger than duration and lateness we are too late */ if ((late = rstart + jitter > max_lateness)) { GST_CAT_DEBUG_OBJECT (GST_CAT_PERFORMANCE, basesink, "buffer is too late %" GST_TIME_FORMAT " > %" GST_TIME_FORMAT, GST_TIME_ARGS (rstart + jitter), GST_TIME_ARGS (max_lateness)); /* !!emergency!!, if we did not receive anything valid for more than a * second, render it anyway so the user sees something */ if (GST_CLOCK_TIME_IS_VALID (priv->last_render_time) && rstart - priv->last_render_time > GST_SECOND) { late = FALSE; GST_ELEMENT_WARNING (basesink, CORE, CLOCK, (_("A lot of buffers are being dropped.")), ("There may be a timestamping problem, or this computer is too slow.")); GST_CAT_DEBUG_OBJECT (GST_CAT_PERFORMANCE, basesink, "**emergency** last buffer at %" GST_TIME_FORMAT " > GST_SECOND", GST_TIME_ARGS (priv->last_render_time)); } } done: if (!late || !GST_CLOCK_TIME_IS_VALID (priv->last_render_time)) { priv->last_render_time = rstart; /* the next allowed input timestamp */ if (priv->throttle_time > 0) priv->earliest_in_time = rstart + priv->throttle_time; } return late; /* all is fine */ in_time: { GST_DEBUG_OBJECT (basesink, "object was scheduled in time"); goto done; } no_drop: { GST_DEBUG_OBJECT (basesink, "frame dropping disabled"); goto done; } not_buffer: { GST_DEBUG_OBJECT (basesink, "object is not a buffer"); return FALSE; } no_timestamp: { GST_DEBUG_OBJECT (basesink, "buffer has no timestamp"); return FALSE; } } /* called before and after calling the render vmethod. It keeps track of how * much time was spent in the render method and is used to check if we are * flooded */ static void gst_base_sink_do_render_stats (GstBaseSink * basesink, gboolean start) { GstBaseSinkPrivate *priv; priv = basesink->priv; if (start) { priv->start = gst_util_get_timestamp (); } else { GstClockTime elapsed; priv->stop = gst_util_get_timestamp (); elapsed = GST_CLOCK_DIFF (priv->start, priv->stop); if (!GST_CLOCK_TIME_IS_VALID (priv->avg_render)) priv->avg_render = elapsed; else priv->avg_render = UPDATE_RUNNING_AVG (priv->avg_render, elapsed); GST_CAT_DEBUG_OBJECT (GST_CAT_QOS, basesink, "avg_render: %" GST_TIME_FORMAT, GST_TIME_ARGS (priv->avg_render)); } } static void gst_base_sink_flush_start (GstBaseSink * basesink, GstPad * pad) { /* make sure we are not blocked on the clock also clear any pending * eos state. */ gst_base_sink_set_flushing (basesink, pad, TRUE); /* we grab the stream lock but that is not needed since setting the * sink to flushing would make sure no state commit is being done * anymore */ GST_PAD_STREAM_LOCK (pad); gst_base_sink_reset_qos (basesink); /* and we need to commit our state again on the next * prerolled buffer */ basesink->playing_async = TRUE; if (basesink->priv->async_enabled) { gst_element_lost_state (GST_ELEMENT_CAST (basesink)); } else { /* start time reset in above case as well; * arranges for a.o. proper position reporting when flushing in PAUSED */ gst_element_set_start_time (GST_ELEMENT_CAST (basesink), 0); basesink->priv->have_latency = TRUE; } gst_base_sink_set_last_buffer (basesink, NULL); GST_PAD_STREAM_UNLOCK (pad); } static void gst_base_sink_flush_stop (GstBaseSink * basesink, GstPad * pad, gboolean reset_time) { /* unset flushing so we can accept new data, this also flushes out any EOS * event. */ gst_base_sink_set_flushing (basesink, pad, FALSE); /* for position reporting */ GST_OBJECT_LOCK (basesink); basesink->priv->current_sstart = GST_CLOCK_TIME_NONE; basesink->priv->current_sstop = GST_CLOCK_TIME_NONE; basesink->priv->eos_rtime = GST_CLOCK_TIME_NONE; basesink->priv->call_preroll = TRUE; basesink->priv->current_step.valid = FALSE; basesink->priv->pending_step.valid = FALSE; if (basesink->pad_mode == GST_PAD_MODE_PUSH) { /* we need new segment info after the flush. */ basesink->have_newsegment = FALSE; if (reset_time) { gst_segment_init (&basesink->segment, GST_FORMAT_UNDEFINED); GST_ELEMENT_START_TIME (basesink) = 0; } } GST_OBJECT_UNLOCK (basesink); if (reset_time) { GST_DEBUG_OBJECT (basesink, "posting reset-time message"); gst_element_post_message (GST_ELEMENT_CAST (basesink), gst_message_new_reset_time (GST_OBJECT_CAST (basesink), 0)); } } static GstFlowReturn gst_base_sink_default_wait_event (GstBaseSink * basesink, GstEvent * event) { GstFlowReturn ret; gboolean late, step_end; ret = gst_base_sink_do_sync (basesink, GST_MINI_OBJECT_CAST (event), &late, &step_end); return ret; } static GstFlowReturn gst_base_sink_wait_event (GstBaseSink * basesink, GstEvent * event) { GstFlowReturn ret; GstBaseSinkClass *bclass; bclass = GST_BASE_SINK_GET_CLASS (basesink); if (G_LIKELY (bclass->wait_event)) ret = bclass->wait_event (basesink, event); else ret = GST_FLOW_NOT_SUPPORTED; return ret; } static gboolean gst_base_sink_default_event (GstBaseSink * basesink, GstEvent * event) { gboolean result = TRUE; GstBaseSinkClass *bclass; bclass = GST_BASE_SINK_GET_CLASS (basesink); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_FLUSH_START: { GST_DEBUG_OBJECT (basesink, "flush-start %p", event); gst_base_sink_flush_start (basesink, basesink->sinkpad); break; } case GST_EVENT_FLUSH_STOP: { gboolean reset_time; gst_event_parse_flush_stop (event, &reset_time); GST_DEBUG_OBJECT (basesink, "flush-stop %p, reset_time: %d", event, reset_time); gst_base_sink_flush_stop (basesink, basesink->sinkpad, reset_time); break; } case GST_EVENT_EOS: { GstMessage *message; guint32 seqnum; /* we set the received EOS flag here so that we can use it when testing if * we are prerolled and to refuse more buffers. */ basesink->priv->received_eos = TRUE; /* wait for EOS */ if (G_UNLIKELY (gst_base_sink_wait_event (basesink, event) != GST_FLOW_OK)) { result = FALSE; goto done; } /* the EOS event is completely handled so we mark * ourselves as being in the EOS state. eos is also * protected by the object lock so we can read it when * answering the POSITION query. */ GST_OBJECT_LOCK (basesink); basesink->eos = TRUE; GST_OBJECT_UNLOCK (basesink); /* ok, now we can post the message */ GST_DEBUG_OBJECT (basesink, "Now posting EOS"); seqnum = basesink->priv->seqnum = gst_event_get_seqnum (event); GST_DEBUG_OBJECT (basesink, "Got seqnum #%" G_GUINT32_FORMAT, seqnum); message = gst_message_new_eos (GST_OBJECT_CAST (basesink)); gst_message_set_seqnum (message, seqnum); gst_element_post_message (GST_ELEMENT_CAST (basesink), message); break; } case GST_EVENT_STREAM_START: { GstMessage *message; guint32 seqnum; seqnum = gst_event_get_seqnum (event); GST_DEBUG_OBJECT (basesink, "Now posting STREAM_START (seqnum:%d)", seqnum); message = gst_message_new_stream_start (GST_OBJECT_CAST (basesink)); gst_message_set_seqnum (message, seqnum); gst_element_post_message (GST_ELEMENT_CAST (basesink), message); break; } case GST_EVENT_CAPS: { GstCaps *caps; GST_DEBUG_OBJECT (basesink, "caps %p", event); gst_event_parse_caps (event, &caps); if (bclass->set_caps) result = bclass->set_caps (basesink, caps); if (result) { GST_OBJECT_LOCK (basesink); gst_caps_replace (&basesink->priv->caps, caps); GST_OBJECT_UNLOCK (basesink); } break; } case GST_EVENT_SEGMENT: /* configure the segment */ /* The segment is protected with both the STREAM_LOCK and the OBJECT_LOCK. * We protect with the OBJECT_LOCK so that we can use the values to * safely answer a POSITION query. */ GST_OBJECT_LOCK (basesink); /* the newsegment event is needed to bring the buffer timestamps to the * stream time and to drop samples outside of the playback segment. */ gst_event_copy_segment (event, &basesink->segment); GST_DEBUG_OBJECT (basesink, "configured SEGMENT %" GST_SEGMENT_FORMAT, &basesink->segment); basesink->have_newsegment = TRUE; GST_OBJECT_UNLOCK (basesink); break; case GST_EVENT_GAP: { if (G_UNLIKELY (gst_base_sink_wait_event (basesink, event) != GST_FLOW_OK)) result = FALSE; break; } case GST_EVENT_TAG: { GstTagList *taglist; gst_event_parse_tag (event, &taglist); gst_element_post_message (GST_ELEMENT_CAST (basesink), gst_message_new_tag (GST_OBJECT_CAST (basesink), gst_tag_list_copy (taglist))); break; } case GST_EVENT_TOC: { GstToc *toc; gboolean updated; gst_event_parse_toc (event, &toc, &updated); gst_element_post_message (GST_ELEMENT_CAST (basesink), gst_message_new_toc (GST_OBJECT_CAST (basesink), toc, updated)); gst_toc_unref (toc); break; } case GST_EVENT_SINK_MESSAGE: { GstMessage *msg = NULL; gst_event_parse_sink_message (event, &msg); if (msg) gst_element_post_message (GST_ELEMENT_CAST (basesink), msg); break; } default: break; } done: gst_event_unref (event); return result; } static gboolean gst_base_sink_event (GstPad * pad, GstObject * parent, GstEvent * event) { GstBaseSink *basesink; gboolean result = TRUE; GstBaseSinkClass *bclass; basesink = GST_BASE_SINK_CAST (parent); bclass = GST_BASE_SINK_GET_CLASS (basesink); GST_DEBUG_OBJECT (basesink, "received event %p %" GST_PTR_FORMAT, event, event); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_FLUSH_STOP: /* special case for this serialized event because we don't want to grab * the PREROLL lock or check if we were flushing */ if (bclass->event) result = bclass->event (basesink, event); break; default: if (GST_EVENT_IS_SERIALIZED (event)) { GST_BASE_SINK_PREROLL_LOCK (basesink); if (G_UNLIKELY (basesink->flushing)) goto flushing; if (G_UNLIKELY (basesink->priv->received_eos)) goto after_eos; if (bclass->event) result = bclass->event (basesink, event); GST_BASE_SINK_PREROLL_UNLOCK (basesink); } else { if (bclass->event) result = bclass->event (basesink, event); } break; } done: return result; /* ERRORS */ flushing: { GST_DEBUG_OBJECT (basesink, "we are flushing"); GST_BASE_SINK_PREROLL_UNLOCK (basesink); gst_event_unref (event); result = FALSE; goto done; } after_eos: { GST_DEBUG_OBJECT (basesink, "Event received after EOS, dropping"); GST_BASE_SINK_PREROLL_UNLOCK (basesink); gst_event_unref (event); result = FALSE; goto done; } } /* default implementation to calculate the start and end * timestamps on a buffer, subclasses can override */ static void gst_base_sink_default_get_times (GstBaseSink * basesink, GstBuffer * buffer, GstClockTime * start, GstClockTime * end) { GstClockTime timestamp, duration; /* first sync on DTS, else use PTS */ timestamp = GST_BUFFER_DTS (buffer); if (!GST_CLOCK_TIME_IS_VALID (timestamp)) timestamp = GST_BUFFER_PTS (buffer); if (GST_CLOCK_TIME_IS_VALID (timestamp)) { /* get duration to calculate end time */ duration = GST_BUFFER_DURATION (buffer); if (GST_CLOCK_TIME_IS_VALID (duration)) { *end = timestamp + duration; } *start = timestamp; } } /* must be called with PREROLL_LOCK */ static gboolean gst_base_sink_needs_preroll (GstBaseSink * basesink) { gboolean is_prerolled, res; /* we have 2 cases where the PREROLL_LOCK is released: * 1) we are blocking in the PREROLL_LOCK and thus are prerolled. * 2) we are syncing on the clock */ is_prerolled = basesink->have_preroll || basesink->priv->received_eos; res = !is_prerolled; GST_DEBUG_OBJECT (basesink, "have_preroll: %d, EOS: %d => needs preroll: %d", basesink->have_preroll, basesink->priv->received_eos, res); return res; } /* with STREAM_LOCK, PREROLL_LOCK * * Takes a buffer and compare the timestamps with the last segment. * If the buffer falls outside of the segment boundaries, drop it. * Else send the buffer for preroll and rendering. * * This function takes ownership of the buffer. */ static GstFlowReturn gst_base_sink_chain_unlocked (GstBaseSink * basesink, GstPad * pad, gpointer obj, gboolean is_list) { GstBaseSinkClass *bclass; GstBaseSinkPrivate *priv = basesink->priv; GstFlowReturn ret; GstClockTime start = GST_CLOCK_TIME_NONE, end = GST_CLOCK_TIME_NONE; GstSegment *segment; GstBuffer *sync_buf; gint do_qos; gboolean late, step_end; if (G_UNLIKELY (basesink->flushing)) goto flushing; if (G_UNLIKELY (priv->received_eos)) goto was_eos; if (is_list) { sync_buf = gst_buffer_list_get (GST_BUFFER_LIST_CAST (obj), 0); g_assert (NULL != sync_buf); } else { sync_buf = GST_BUFFER_CAST (obj); } /* for code clarity */ segment = &basesink->segment; if (G_UNLIKELY (!basesink->have_newsegment)) { gboolean sync; sync = gst_base_sink_get_sync (basesink); if (sync) { GST_ELEMENT_WARNING (basesink, STREAM, FAILED, (_("Internal data flow problem.")), ("Received buffer without a new-segment. Assuming timestamps start from 0.")); } /* this means this sink will assume timestamps start from 0 */ GST_OBJECT_LOCK (basesink); segment->start = 0; segment->stop = -1; basesink->segment.start = 0; basesink->segment.stop = -1; basesink->have_newsegment = TRUE; GST_OBJECT_UNLOCK (basesink); } bclass = GST_BASE_SINK_GET_CLASS (basesink); /* check if the buffer needs to be dropped, we first ask the subclass for the * start and end */ if (bclass->get_times) bclass->get_times (basesink, sync_buf, &start, &end); if (!GST_CLOCK_TIME_IS_VALID (start)) { /* if the subclass does not want sync, we use our own values so that we at * least clip the buffer to the segment */ gst_base_sink_default_get_times (basesink, sync_buf, &start, &end); } GST_DEBUG_OBJECT (basesink, "got times start: %" GST_TIME_FORMAT ", end: %" GST_TIME_FORMAT, GST_TIME_ARGS (start), GST_TIME_ARGS (end)); /* a dropped buffer does not participate in anything */ if (GST_CLOCK_TIME_IS_VALID (start) && (segment->format == GST_FORMAT_TIME)) { if (G_UNLIKELY (!gst_segment_clip (segment, GST_FORMAT_TIME, start, end, NULL, NULL))) goto out_of_segment; } if (!is_list) { if (bclass->prepare) { ret = bclass->prepare (basesink, GST_BUFFER_CAST (obj)); if (G_UNLIKELY (ret != GST_FLOW_OK)) goto prepare_failed; } } else { if (bclass->prepare_list) { ret = bclass->prepare_list (basesink, GST_BUFFER_LIST_CAST (obj)); if (G_UNLIKELY (ret != GST_FLOW_OK)) goto prepare_failed; } } again: late = FALSE; step_end = FALSE; /* synchronize this object, non syncable objects return OK * immediately. */ ret = gst_base_sink_do_sync (basesink, GST_MINI_OBJECT_CAST (sync_buf), &late, &step_end); if (G_UNLIKELY (ret != GST_FLOW_OK)) goto sync_failed; /* drop late buffers unconditionally, let's hope it's unlikely */ if (G_UNLIKELY (late)) goto dropped; /* read once, to get same value before and after */ do_qos = g_atomic_int_get (&priv->qos_enabled); GST_DEBUG_OBJECT (basesink, "rendering object %p", obj); /* record rendering time for QoS and stats */ if (do_qos) gst_base_sink_do_render_stats (basesink, TRUE); if (!is_list) { /* For buffer lists do not set last buffer for now. */ gst_base_sink_set_last_buffer (basesink, GST_BUFFER_CAST (obj)); if (bclass->render) ret = bclass->render (basesink, GST_BUFFER_CAST (obj)); } else { if (bclass->render_list) ret = bclass->render_list (basesink, GST_BUFFER_LIST_CAST (obj)); } if (do_qos) gst_base_sink_do_render_stats (basesink, FALSE); if (ret == GST_FLOW_STEP) goto again; if (G_UNLIKELY (basesink->flushing)) goto flushing; priv->rendered++; done: if (step_end) { /* the step ended, check if we need to activate a new step */ GST_DEBUG_OBJECT (basesink, "step ended"); stop_stepping (basesink, &basesink->segment, &priv->current_step, priv->current_rstart, priv->current_rstop, basesink->eos); goto again; } gst_base_sink_perform_qos (basesink, late); GST_DEBUG_OBJECT (basesink, "object unref after render %p", obj); gst_mini_object_unref (GST_MINI_OBJECT_CAST (obj)); return ret; /* ERRORS */ flushing: { GST_DEBUG_OBJECT (basesink, "sink is flushing"); gst_mini_object_unref (GST_MINI_OBJECT_CAST (obj)); return GST_FLOW_FLUSHING; } was_eos: { GST_DEBUG_OBJECT (basesink, "we are EOS, dropping object, return EOS"); gst_mini_object_unref (GST_MINI_OBJECT_CAST (obj)); return GST_FLOW_EOS; } out_of_segment: { GST_DEBUG_OBJECT (basesink, "dropping buffer, out of clipping segment"); gst_mini_object_unref (GST_MINI_OBJECT_CAST (obj)); return GST_FLOW_OK; } prepare_failed: { GST_DEBUG_OBJECT (basesink, "prepare buffer failed %s", gst_flow_get_name (ret)); gst_mini_object_unref (GST_MINI_OBJECT_CAST (obj)); return ret; } sync_failed: { GST_DEBUG_OBJECT (basesink, "do_sync returned %s", gst_flow_get_name (ret)); goto done; } dropped: { priv->dropped++; GST_DEBUG_OBJECT (basesink, "buffer late, dropping"); if (g_atomic_int_get (&priv->qos_enabled)) { GstMessage *qos_msg; GstClockTime timestamp, duration; timestamp = GST_BUFFER_TIMESTAMP (GST_BUFFER_CAST (sync_buf)); duration = GST_BUFFER_DURATION (GST_BUFFER_CAST (sync_buf)); GST_CAT_DEBUG_OBJECT (GST_CAT_QOS, basesink, "qos: dropped buffer rt %" GST_TIME_FORMAT ", st %" GST_TIME_FORMAT ", ts %" GST_TIME_FORMAT ", dur %" GST_TIME_FORMAT, GST_TIME_ARGS (priv->current_rstart), GST_TIME_ARGS (priv->current_sstart), GST_TIME_ARGS (timestamp), GST_TIME_ARGS (duration)); GST_CAT_DEBUG_OBJECT (GST_CAT_QOS, basesink, "qos: rendered %" G_GUINT64_FORMAT ", dropped %" G_GUINT64_FORMAT, priv->rendered, priv->dropped); qos_msg = gst_message_new_qos (GST_OBJECT_CAST (basesink), basesink->sync, priv->current_rstart, priv->current_sstart, timestamp, duration); gst_message_set_qos_values (qos_msg, priv->current_jitter, priv->avg_rate, 1000000); gst_message_set_qos_stats (qos_msg, GST_FORMAT_BUFFERS, priv->rendered, priv->dropped); gst_element_post_message (GST_ELEMENT_CAST (basesink), qos_msg); } goto done; } } /* with STREAM_LOCK */ static GstFlowReturn gst_base_sink_chain_main (GstBaseSink * basesink, GstPad * pad, gpointer obj, gboolean is_list) { GstFlowReturn result; if (G_UNLIKELY (basesink->pad_mode != GST_PAD_MODE_PUSH)) goto wrong_mode; GST_BASE_SINK_PREROLL_LOCK (basesink); result = gst_base_sink_chain_unlocked (basesink, pad, obj, is_list); GST_BASE_SINK_PREROLL_UNLOCK (basesink); done: return result; /* ERRORS */ wrong_mode: { GST_OBJECT_LOCK (pad); GST_WARNING_OBJECT (basesink, "Push on pad %s:%s, but it was not activated in push mode", GST_DEBUG_PAD_NAME (pad)); GST_OBJECT_UNLOCK (pad); gst_mini_object_unref (GST_MINI_OBJECT_CAST (obj)); /* we don't post an error message this will signal to the peer * pushing that EOS is reached. */ result = GST_FLOW_EOS; goto done; } } static GstFlowReturn gst_base_sink_chain (GstPad * pad, GstObject * parent, GstBuffer * buf) { GstBaseSink *basesink; basesink = GST_BASE_SINK (parent); return gst_base_sink_chain_main (basesink, pad, buf, FALSE); } static GstFlowReturn gst_base_sink_chain_list (GstPad * pad, GstObject * parent, GstBufferList * list) { GstBaseSink *basesink; GstBaseSinkClass *bclass; GstFlowReturn result; basesink = GST_BASE_SINK (parent); bclass = GST_BASE_SINK_GET_CLASS (basesink); if (G_LIKELY (bclass->render_list)) { result = gst_base_sink_chain_main (basesink, pad, list, TRUE); } else { guint i, len; GstBuffer *buffer; GST_INFO_OBJECT (pad, "chaining each buffer in list"); len = gst_buffer_list_length (list); result = GST_FLOW_OK; for (i = 0; i < len; i++) { buffer = gst_buffer_list_get (list, i); result = gst_base_sink_chain_main (basesink, pad, gst_buffer_ref (buffer), FALSE); if (result != GST_FLOW_OK) break; } gst_buffer_list_unref (list); } return result; } static gboolean gst_base_sink_default_do_seek (GstBaseSink * sink, 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 (sink, "Can't do a default seek"); } return res; } #define SEEK_TYPE_IS_RELATIVE(t) (((t) != GST_SEEK_TYPE_NONE) && ((t) != GST_SEEK_TYPE_SET)) static gboolean gst_base_sink_default_prepare_seek_segment (GstBaseSink * sink, 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; gdouble rate; gboolean update; gboolean res = TRUE; gst_event_parse_seek (event, &rate, &seek_format, &flags, &start_type, &start, &stop_type, &stop); if (seek_format == segment->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 (sink->sinkpad, seek_format, start, segment->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 (sink->sinkpad, seek_format, stop, segment->format, &stop); stop_type = GST_SEEK_TYPE_SET; } /* And finally, configure our output segment in the desired format */ gst_segment_do_seek (segment, rate, segment->format, flags, start_type, start, stop_type, stop, &update); if (!res) goto no_format; return res; no_format: { GST_DEBUG_OBJECT (sink, "undefined format given, seek aborted."); return FALSE; } } /* perform a seek, only executed in pull mode */ static gboolean gst_base_sink_perform_seek (GstBaseSink * sink, GstPad * pad, GstEvent * event) { gboolean flush; gdouble rate; GstFormat seek_format, dest_format; GstSeekFlags flags; GstSeekType start_type, stop_type; gboolean seekseg_configured = FALSE; gint64 start, stop; gboolean update, res = TRUE; GstSegment seeksegment; dest_format = sink->segment.format; if (event) { GST_DEBUG_OBJECT (sink, "performing seek with event %p", event); gst_event_parse_seek (event, &rate, &seek_format, &flags, &start_type, &start, &stop_type, &stop); flush = flags & GST_SEEK_FLAG_FLUSH; } else { GST_DEBUG_OBJECT (sink, "performing seek without event"); flush = FALSE; } if (flush) { GST_DEBUG_OBJECT (sink, "flushing upstream"); gst_pad_push_event (pad, gst_event_new_flush_start ()); gst_base_sink_flush_start (sink, pad); } else { GST_DEBUG_OBJECT (sink, "pausing pulling thread"); } GST_PAD_STREAM_LOCK (pad); /* 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, &sink->segment, sizeof (GstSegment)); /* now configure the final seek segment */ if (event) { if (sink->segment.format != seek_format) { /* OK, here's where we give the subclass a chance to convert the relative * seek into an absolute one in the processing format. We set up any * absolute seek above, before taking the stream lock. */ if (!gst_base_sink_default_prepare_seek_segment (sink, event, &seeksegment)) { GST_DEBUG_OBJECT (sink, "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 (sink, "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_sink_default_do_seek (sink, &seeksegment); } if (flush) { GST_DEBUG_OBJECT (sink, "stop flushing upstream"); gst_pad_push_event (pad, gst_event_new_flush_stop (TRUE)); gst_base_sink_flush_stop (sink, pad, TRUE); } else if (res && sink->running) { /* we are running the current segment and doing a non-flushing seek, * close the segment first based on the position. */ GST_DEBUG_OBJECT (sink, "closing running segment %" G_GINT64_FORMAT " to %" G_GINT64_FORMAT, sink->segment.start, sink->segment.position); } /* The subclass must have converted the segment to the processing format * by now */ if (res && seeksegment.format != dest_format) { GST_DEBUG_OBJECT (sink, "Subclass failed to prepare a seek segment " "in the correct format. Aborting seek."); res = FALSE; } /* if successful seek, we update our real segment and push * out the new segment. */ if (res) { gst_segment_copy_into (&seeksegment, &sink->segment); if (sink->segment.flags & GST_SEGMENT_FLAG_SEGMENT) { gst_element_post_message (GST_ELEMENT (sink), gst_message_new_segment_start (GST_OBJECT (sink), sink->segment.format, sink->segment.position)); } } sink->priv->discont = TRUE; sink->running = TRUE; GST_PAD_STREAM_UNLOCK (pad); return res; } static void set_step_info (GstBaseSink * sink, GstStepInfo * current, GstStepInfo * pending, guint seqnum, GstFormat format, guint64 amount, gdouble rate, gboolean flush, gboolean intermediate) { GST_OBJECT_LOCK (sink); pending->seqnum = seqnum; pending->format = format; pending->amount = amount; pending->position = 0; pending->rate = rate; pending->flush = flush; pending->intermediate = intermediate; pending->valid = TRUE; /* flush invalidates the current stepping segment */ if (flush) current->valid = FALSE; GST_OBJECT_UNLOCK (sink); } static gboolean gst_base_sink_perform_step (GstBaseSink * sink, GstPad * pad, GstEvent * event) { GstBaseSinkPrivate *priv; GstBaseSinkClass *bclass; gboolean flush, intermediate; gdouble rate; GstFormat format; guint64 amount; guint seqnum; GstStepInfo *pending, *current; GstMessage *message; bclass = GST_BASE_SINK_GET_CLASS (sink); priv = sink->priv; GST_DEBUG_OBJECT (sink, "performing step with event %p", event); gst_event_parse_step (event, &format, &amount, &rate, &flush, &intermediate); seqnum = gst_event_get_seqnum (event); pending = &priv->pending_step; current = &priv->current_step; /* post message first */ message = gst_message_new_step_start (GST_OBJECT (sink), FALSE, format, amount, rate, flush, intermediate); gst_message_set_seqnum (message, seqnum); gst_element_post_message (GST_ELEMENT (sink), message); if (flush) { /* we need to call ::unlock before locking PREROLL_LOCK * since we lock it before going into ::render */ if (bclass->unlock) bclass->unlock (sink); GST_BASE_SINK_PREROLL_LOCK (sink); /* now that we have the PREROLL lock, clear our unlock request */ if (bclass->unlock_stop) bclass->unlock_stop (sink); /* update the stepinfo and make it valid */ set_step_info (sink, current, pending, seqnum, format, amount, rate, flush, intermediate); if (sink->priv->async_enabled) { /* and we need to commit our state again on the next * prerolled buffer */ sink->playing_async = TRUE; priv->pending_step.need_preroll = TRUE; sink->need_preroll = FALSE; gst_element_lost_state (GST_ELEMENT_CAST (sink)); } else { sink->priv->have_latency = TRUE; sink->need_preroll = FALSE; } priv->current_sstart = GST_CLOCK_TIME_NONE; priv->current_sstop = GST_CLOCK_TIME_NONE; priv->eos_rtime = GST_CLOCK_TIME_NONE; priv->call_preroll = TRUE; gst_base_sink_set_last_buffer (sink, NULL); gst_base_sink_reset_qos (sink); if (sink->clock_id) { gst_clock_id_unschedule (sink->clock_id); } if (sink->have_preroll) { GST_DEBUG_OBJECT (sink, "signal waiter"); priv->step_unlock = TRUE; GST_BASE_SINK_PREROLL_SIGNAL (sink); } GST_BASE_SINK_PREROLL_UNLOCK (sink); } else { /* update the stepinfo and make it valid */ set_step_info (sink, current, pending, seqnum, format, amount, rate, flush, intermediate); } return TRUE; } /* with STREAM_LOCK */ static void gst_base_sink_loop (GstPad * pad) { GstObject *parent; GstBaseSink *basesink; GstBuffer *buf = NULL; GstFlowReturn result; guint blocksize; guint64 offset; parent = GST_OBJECT_PARENT (pad); basesink = GST_BASE_SINK (parent); g_assert (basesink->pad_mode == GST_PAD_MODE_PULL); if ((blocksize = basesink->priv->blocksize) == 0) blocksize = -1; offset = basesink->segment.position; GST_DEBUG_OBJECT (basesink, "pulling %" G_GUINT64_FORMAT ", %u", offset, blocksize); result = gst_pad_pull_range (pad, offset, blocksize, &buf); if (G_UNLIKELY (result != GST_FLOW_OK)) goto paused; if (G_UNLIKELY (buf == NULL)) goto no_buffer; offset += gst_buffer_get_size (buf); basesink->segment.position = offset; GST_BASE_SINK_PREROLL_LOCK (basesink); result = gst_base_sink_chain_unlocked (basesink, pad, buf, FALSE); GST_BASE_SINK_PREROLL_UNLOCK (basesink); if (G_UNLIKELY (result != GST_FLOW_OK)) goto paused; return; /* ERRORS */ paused: { GST_LOG_OBJECT (basesink, "pausing task, reason %s", gst_flow_get_name (result)); gst_pad_pause_task (pad); if (result == GST_FLOW_EOS) { /* perform EOS logic */ if (basesink->segment.flags & GST_SEGMENT_FLAG_SEGMENT) { gst_element_post_message (GST_ELEMENT_CAST (basesink), gst_message_new_segment_done (GST_OBJECT_CAST (basesink), basesink->segment.format, basesink->segment.position)); gst_base_sink_event (pad, parent, gst_event_new_segment_done (basesink->segment.format, basesink->segment.position)); } else { gst_base_sink_event (pad, parent, gst_event_new_eos ()); } } else if (result == GST_FLOW_NOT_LINKED || result <= GST_FLOW_EOS) { /* for fatal errors we post an error message, post the error * first so the app knows about the error first. * wrong-state is not a fatal error because it happens due to * flushing and posting an error message in that case is the * wrong thing to do, e.g. when basesrc is doing a flushing * seek. */ GST_ELEMENT_ERROR (basesink, STREAM, FAILED, (_("Internal data stream error.")), ("stream stopped, reason %s", gst_flow_get_name (result))); gst_base_sink_event (pad, parent, gst_event_new_eos ()); } return; } no_buffer: { GST_LOG_OBJECT (basesink, "no buffer, pausing"); GST_ELEMENT_ERROR (basesink, STREAM, FAILED, (_("Internal data flow error.")), ("element returned NULL buffer")); result = GST_FLOW_ERROR; goto paused; } } static gboolean gst_base_sink_set_flushing (GstBaseSink * basesink, GstPad * pad, gboolean flushing) { GstBaseSinkClass *bclass; bclass = GST_BASE_SINK_GET_CLASS (basesink); if (flushing) { /* unlock any subclasses, we need to do this before grabbing the * PREROLL_LOCK since we hold this lock before going into ::render. */ if (bclass->unlock) bclass->unlock (basesink); } GST_BASE_SINK_PREROLL_LOCK (basesink); basesink->flushing = flushing; if (flushing) { /* step 1, now that we have the PREROLL lock, clear our unlock request */ if (bclass->unlock_stop) bclass->unlock_stop (basesink); /* set need_preroll before we unblock the clock. If the clock is unblocked * before timing out, we can reuse the buffer for preroll. */ basesink->need_preroll = TRUE; /* step 2, unblock clock sync (if any) or any other blocking thing */ if (basesink->clock_id) { gst_clock_id_unschedule (basesink->clock_id); } /* flush out the data thread if it's locked in finish_preroll, this will * also flush out the EOS state */ GST_DEBUG_OBJECT (basesink, "flushing out data thread, need preroll to TRUE"); /* we can't have EOS anymore now */ basesink->eos = FALSE; basesink->priv->received_eos = FALSE; basesink->have_preroll = FALSE; basesink->priv->step_unlock = FALSE; /* can't report latency anymore until we preroll again */ if (basesink->priv->async_enabled) { GST_OBJECT_LOCK (basesink); basesink->priv->have_latency = FALSE; GST_OBJECT_UNLOCK (basesink); } /* and signal any waiters now */ GST_BASE_SINK_PREROLL_SIGNAL (basesink); } GST_BASE_SINK_PREROLL_UNLOCK (basesink); return TRUE; } static gboolean gst_base_sink_default_activate_pull (GstBaseSink * basesink, gboolean active) { gboolean result; if (active) { /* start task */ result = gst_pad_start_task (basesink->sinkpad, (GstTaskFunction) gst_base_sink_loop, basesink->sinkpad, NULL); } else { /* step 2, make sure streaming finishes */ result = gst_pad_stop_task (basesink->sinkpad); } return result; } static gboolean gst_base_sink_pad_activate (GstPad * pad, GstObject * parent) { gboolean result = FALSE; GstBaseSink *basesink; GstQuery *query; gboolean pull_mode; basesink = GST_BASE_SINK (parent); GST_DEBUG_OBJECT (basesink, "Trying pull mode first"); gst_base_sink_set_flushing (basesink, pad, FALSE); /* we need to have the pull mode enabled */ if (!basesink->can_activate_pull) { GST_DEBUG_OBJECT (basesink, "pull mode disabled"); goto fallback; } /* check if downstreams supports pull mode at all */ query = gst_query_new_scheduling (); if (!gst_pad_peer_query (pad, query)) { gst_query_unref (query); GST_DEBUG_OBJECT (basesink, "peer query faild, no pull mode"); goto fallback; } /* parse result of the query */ pull_mode = gst_query_has_scheduling_mode (query, GST_PAD_MODE_PULL); gst_query_unref (query); if (!pull_mode) { GST_DEBUG_OBJECT (basesink, "pull mode not supported"); goto fallback; } /* set the pad mode before starting the task so that it's in the * correct state for the new thread. also the sink set_caps and get_caps * function checks this */ basesink->pad_mode = GST_PAD_MODE_PULL; /* we first try to negotiate a format so that when we try to activate * downstream, it knows about our format */ if (!gst_base_sink_negotiate_pull (basesink)) { GST_DEBUG_OBJECT (basesink, "failed to negotiate in pull mode"); goto fallback; } /* ok activate now */ if (!gst_pad_activate_mode (pad, GST_PAD_MODE_PULL, TRUE)) { /* clear any pending caps */ GST_OBJECT_LOCK (basesink); gst_caps_replace (&basesink->priv->caps, NULL); GST_OBJECT_UNLOCK (basesink); GST_DEBUG_OBJECT (basesink, "failed to activate in pull mode"); goto fallback; } GST_DEBUG_OBJECT (basesink, "Success activating pull mode"); result = TRUE; goto done; /* push mode fallback */ fallback: GST_DEBUG_OBJECT (basesink, "Falling back to push mode"); if ((result = gst_pad_activate_mode (pad, GST_PAD_MODE_PUSH, TRUE))) { GST_DEBUG_OBJECT (basesink, "Success activating push mode"); } done: if (!result) { GST_WARNING_OBJECT (basesink, "Could not activate pad in either mode"); gst_base_sink_set_flushing (basesink, pad, TRUE); } return result; } static gboolean gst_base_sink_pad_activate_push (GstPad * pad, GstObject * parent, gboolean active) { gboolean result; GstBaseSink *basesink; basesink = GST_BASE_SINK (parent); if (active) { if (!basesink->can_activate_push) { result = FALSE; basesink->pad_mode = GST_PAD_MODE_NONE; } else { result = TRUE; basesink->pad_mode = GST_PAD_MODE_PUSH; } } else { if (G_UNLIKELY (basesink->pad_mode != GST_PAD_MODE_PUSH)) { g_warning ("Internal GStreamer activation error!!!"); result = FALSE; } else { gst_base_sink_set_flushing (basesink, pad, TRUE); result = TRUE; basesink->pad_mode = GST_PAD_MODE_NONE; } } return result; } static gboolean gst_base_sink_negotiate_pull (GstBaseSink * basesink) { GstCaps *caps; gboolean result; result = FALSE; /* this returns the intersection between our caps and the peer caps. If there * is no peer, it returns NULL and we can't operate in pull mode so we can * fail the negotiation. */ caps = gst_pad_get_allowed_caps (GST_BASE_SINK_PAD (basesink)); if (caps == NULL || gst_caps_is_empty (caps)) goto no_caps_possible; GST_DEBUG_OBJECT (basesink, "allowed caps: %" GST_PTR_FORMAT, caps); if (gst_caps_is_any (caps)) { GST_DEBUG_OBJECT (basesink, "caps were ANY after fixating, " "allowing pull()"); /* neither side has template caps in this case, so they are prepared for pull() without setcaps() */ result = TRUE; } else { /* try to fixate */ caps = gst_base_sink_fixate (basesink, caps); GST_DEBUG_OBJECT (basesink, "fixated to: %" GST_PTR_FORMAT, caps); if (gst_caps_is_fixed (caps)) { if (!gst_pad_set_caps (GST_BASE_SINK_PAD (basesink), caps)) goto could_not_set_caps; result = TRUE; } } gst_caps_unref (caps); return result; no_caps_possible: { GST_INFO_OBJECT (basesink, "Pipeline could not agree on caps"); GST_DEBUG_OBJECT (basesink, "get_allowed_caps() returned EMPTY"); if (caps) gst_caps_unref (caps); return FALSE; } could_not_set_caps: { GST_INFO_OBJECT (basesink, "Could not set caps: %" GST_PTR_FORMAT, caps); gst_caps_unref (caps); return FALSE; } } /* this won't get called until we implement an activate function */ static gboolean gst_base_sink_pad_activate_pull (GstPad * pad, GstObject * parent, gboolean active) { gboolean result = FALSE; GstBaseSink *basesink; GstBaseSinkClass *bclass; basesink = GST_BASE_SINK (parent); bclass = GST_BASE_SINK_GET_CLASS (basesink); if (active) { gint64 duration; /* we mark we have a newsegment here because pull based * mode works just fine without having a newsegment before the * first buffer */ gst_segment_init (&basesink->segment, GST_FORMAT_BYTES); GST_OBJECT_LOCK (basesink); basesink->have_newsegment = TRUE; GST_OBJECT_UNLOCK (basesink); /* get the peer duration in bytes */ result = gst_pad_peer_query_duration (pad, GST_FORMAT_BYTES, &duration); if (result) { GST_DEBUG_OBJECT (basesink, "setting duration in bytes to %" G_GINT64_FORMAT, duration); basesink->segment.duration = duration; } else { GST_DEBUG_OBJECT (basesink, "unknown duration"); } if (bclass->activate_pull) result = bclass->activate_pull (basesink, TRUE); else result = FALSE; if (!result) goto activate_failed; } else { if (G_UNLIKELY (basesink->pad_mode != GST_PAD_MODE_PULL)) { g_warning ("Internal GStreamer activation error!!!"); result = FALSE; } else { result = gst_base_sink_set_flushing (basesink, pad, TRUE); if (bclass->activate_pull) result &= bclass->activate_pull (basesink, FALSE); basesink->pad_mode = GST_PAD_MODE_NONE; } } return result; /* ERRORS */ activate_failed: { /* reset, as starting the thread failed */ basesink->pad_mode = GST_PAD_MODE_NONE; GST_ERROR_OBJECT (basesink, "subclass failed to activate in pull mode"); return FALSE; } } static gboolean gst_base_sink_pad_activate_mode (GstPad * pad, GstObject * parent, GstPadMode mode, gboolean active) { gboolean res; switch (mode) { case GST_PAD_MODE_PULL: res = gst_base_sink_pad_activate_pull (pad, parent, active); break; case GST_PAD_MODE_PUSH: res = gst_base_sink_pad_activate_push (pad, parent, active); break; default: GST_LOG_OBJECT (pad, "unknown activation mode %d", mode); res = FALSE; break; } return res; } /* send an event to our sinkpad peer. */ static gboolean gst_base_sink_send_event (GstElement * element, GstEvent * event) { GstPad *pad; GstBaseSink *basesink = GST_BASE_SINK (element); gboolean forward, result = TRUE; GstPadMode mode; GST_OBJECT_LOCK (element); /* get the pad and the scheduling mode */ pad = gst_object_ref (basesink->sinkpad); mode = basesink->pad_mode; GST_OBJECT_UNLOCK (element); /* only push UPSTREAM events upstream */ forward = GST_EVENT_IS_UPSTREAM (event); GST_DEBUG_OBJECT (basesink, "handling event %p %" GST_PTR_FORMAT, event, event); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_LATENCY: { GstClockTime latency; gst_event_parse_latency (event, &latency); /* store the latency. We use this to adjust the running_time before syncing * it to the clock. */ GST_OBJECT_LOCK (element); basesink->priv->latency = latency; if (!basesink->priv->have_latency) forward = FALSE; GST_OBJECT_UNLOCK (element); GST_DEBUG_OBJECT (basesink, "latency set to %" GST_TIME_FORMAT, GST_TIME_ARGS (latency)); /* We forward this event so that all elements know about the global pipeline * latency. This is interesting for an element when it wants to figure out * when a particular piece of data will be rendered. */ break; } case GST_EVENT_SEEK: /* in pull mode we will execute the seek */ if (mode == GST_PAD_MODE_PULL) result = gst_base_sink_perform_seek (basesink, pad, event); break; case GST_EVENT_STEP: result = gst_base_sink_perform_step (basesink, pad, event); forward = FALSE; break; default: break; } if (forward) { result = gst_pad_push_event (pad, event); } else { /* not forwarded, unref the event */ gst_event_unref (event); } gst_object_unref (pad); GST_DEBUG_OBJECT (basesink, "handled event %p %" GST_PTR_FORMAT ": %d", event, event, result); return result; } static gboolean gst_base_sink_get_position (GstBaseSink * basesink, GstFormat format, gint64 * cur, gboolean * upstream) { GstClock *clock = NULL; gboolean res = FALSE; GstFormat oformat; GstSegment *segment; GstClockTime now, latency; GstClockTimeDiff base_time; gint64 time, base, duration; gdouble rate; gint64 last; gboolean last_seen, with_clock, in_paused; GST_OBJECT_LOCK (basesink); /* we can only get the segment when we are not NULL or READY */ if (!basesink->have_newsegment) goto wrong_state; in_paused = FALSE; /* when not in PLAYING or when we're busy with a state change, we * cannot read from the clock so we report time based on the * last seen timestamp. */ if (GST_STATE (basesink) != GST_STATE_PLAYING || GST_STATE_PENDING (basesink) != GST_STATE_VOID_PENDING) { in_paused = TRUE; } segment = &basesink->segment; /* get the format in the segment */ oformat = segment->format; /* report with last seen position when EOS */ last_seen = basesink->eos; /* assume we will use the clock for getting the current position */ with_clock = TRUE; if (basesink->sync == FALSE) with_clock = FALSE; /* and we need a clock */ if (G_UNLIKELY ((clock = GST_ELEMENT_CLOCK (basesink)) == NULL)) with_clock = FALSE; else gst_object_ref (clock); /* mainloop might be querying position when going to playing async, * while (audio) rendering might be quickly advancing stream position, * so use clock asap rather than last reported position */ if (in_paused && with_clock && g_atomic_int_get (&basesink->priv->to_playing)) { GST_DEBUG_OBJECT (basesink, "going to PLAYING, so not PAUSED"); in_paused = FALSE; } /* collect all data we need holding the lock */ if (GST_CLOCK_TIME_IS_VALID (segment->time)) time = segment->time; else time = 0; if (GST_CLOCK_TIME_IS_VALID (segment->stop)) duration = segment->stop - segment->start; else duration = 0; base = segment->base; rate = segment->rate * segment->applied_rate; latency = basesink->priv->latency; if (oformat == GST_FORMAT_TIME) { gint64 start, stop; start = basesink->priv->current_sstart; stop = basesink->priv->current_sstop; if (in_paused) { /* in paused we use the last position as a lower bound */ if (stop == -1 || segment->rate > 0.0) last = start; else last = stop; GST_DEBUG_OBJECT (basesink, "in PAUSED using last %" GST_TIME_FORMAT, GST_TIME_ARGS (last)); } else { /* in playing, use last stop time as upper bound */ if (start == -1 || segment->rate > 0.0) last = stop; else last = start; GST_DEBUG_OBJECT (basesink, "in PLAYING using last %" GST_TIME_FORMAT, GST_TIME_ARGS (last)); } } else { /* convert last stop to stream time */ last = gst_segment_to_stream_time (segment, oformat, segment->position); GST_DEBUG_OBJECT (basesink, "in using last %" G_GINT64_FORMAT, last); } if (in_paused) { /* in paused, use start_time */ base_time = GST_ELEMENT_START_TIME (basesink); GST_DEBUG_OBJECT (basesink, "in paused, using start time %" GST_TIME_FORMAT, GST_TIME_ARGS (base_time)); } else if (with_clock) { /* else use clock when needed */ base_time = GST_ELEMENT_CAST (basesink)->base_time; GST_DEBUG_OBJECT (basesink, "using clock and base time %" GST_TIME_FORMAT, GST_TIME_ARGS (base_time)); } else { /* else, no sync or clock -> no base time */ GST_DEBUG_OBJECT (basesink, "no sync or no clock"); base_time = -1; } /* no base_time, we can't calculate running_time, use last seem timestamp to report * time */ if (base_time == -1) last_seen = TRUE; /* need to release the object lock before we can get the time, * a clock might take the LOCK of the provider, which could be * a basesink subclass. */ GST_OBJECT_UNLOCK (basesink); if (last_seen) { /* in EOS or when no valid stream_time, report the value of last seen * timestamp */ if (last == -1) { /* no timestamp, we need to ask upstream */ GST_DEBUG_OBJECT (basesink, "no last seen timestamp, asking upstream"); res = FALSE; *upstream = TRUE; goto done; } GST_DEBUG_OBJECT (basesink, "using last seen timestamp %" GST_TIME_FORMAT, GST_TIME_ARGS (last)); *cur = last; } else { if (oformat != GST_FORMAT_TIME) { /* convert base, time and duration to time */ if (!gst_pad_query_convert (basesink->sinkpad, oformat, base, GST_FORMAT_TIME, &base)) goto convert_failed; if (!gst_pad_query_convert (basesink->sinkpad, oformat, duration, GST_FORMAT_TIME, &duration)) goto convert_failed; if (!gst_pad_query_convert (basesink->sinkpad, oformat, time, GST_FORMAT_TIME, &time)) goto convert_failed; if (!gst_pad_query_convert (basesink->sinkpad, oformat, last, GST_FORMAT_TIME, &last)) goto convert_failed; /* assume time format from now on */ oformat = GST_FORMAT_TIME; } if (!in_paused && with_clock) { now = gst_clock_get_time (clock); } else { now = base_time; base_time = 0; } /* subtract base time and base time from the clock time. * Make sure we don't go negative. This is the current time in * the segment which we need to scale with the combined * rate and applied rate. */ base_time += base; base_time += latency; if (GST_CLOCK_DIFF (base_time, now) < 0) base_time = now; /* for negative rates we need to count back from the segment * duration. */ if (rate < 0.0) time += duration; *cur = time + gst_guint64_to_gdouble (now - base_time) * rate; if (in_paused) { /* never report less than segment values in paused */ if (last != -1) *cur = MAX (last, *cur); } else { /* never report more than last seen position in playing */ if (last != -1) *cur = MIN (last, *cur); } GST_DEBUG_OBJECT (basesink, "now %" GST_TIME_FORMAT " - base_time %" GST_TIME_FORMAT " - base %" GST_TIME_FORMAT " + time %" GST_TIME_FORMAT " last %" GST_TIME_FORMAT, GST_TIME_ARGS (now), GST_TIME_ARGS (base_time), GST_TIME_ARGS (base), GST_TIME_ARGS (time), GST_TIME_ARGS (last)); } if (oformat != format) { /* convert to final format */ if (!gst_pad_query_convert (basesink->sinkpad, oformat, *cur, format, cur)) goto convert_failed; } res = TRUE; done: GST_DEBUG_OBJECT (basesink, "res: %d, POSITION: %" GST_TIME_FORMAT, res, GST_TIME_ARGS (*cur)); if (clock) gst_object_unref (clock); return res; /* special cases */ wrong_state: { /* in NULL or READY we always return FALSE and -1 */ GST_DEBUG_OBJECT (basesink, "position in wrong state, return -1"); res = FALSE; *cur = -1; GST_OBJECT_UNLOCK (basesink); goto done; } convert_failed: { GST_DEBUG_OBJECT (basesink, "convert failed, try upstream"); *upstream = TRUE; res = FALSE; goto done; } } static gboolean gst_base_sink_get_duration (GstBaseSink * basesink, GstFormat format, gint64 * dur, gboolean * upstream) { gboolean res = FALSE; if (basesink->pad_mode == GST_PAD_MODE_PULL) { gint64 uduration; /* get the duration in bytes, in pull mode that's all we are sure to * know. We have to explicitly get this value from upstream instead of * using our cached value because it might change. Duration caching * should be done at a higher level. */ res = gst_pad_peer_query_duration (basesink->sinkpad, GST_FORMAT_BYTES, &uduration); if (res) { basesink->segment.duration = uduration; if (format != GST_FORMAT_BYTES) { /* convert to the requested format */ res = gst_pad_query_convert (basesink->sinkpad, GST_FORMAT_BYTES, uduration, format, dur); } else { *dur = uduration; } } *upstream = FALSE; } else { *upstream = TRUE; } return res; } static gboolean default_element_query (GstElement * element, GstQuery * query) { gboolean res = FALSE; GstBaseSink *basesink = GST_BASE_SINK (element); switch (GST_QUERY_TYPE (query)) { case GST_QUERY_POSITION: { gint64 cur = 0; GstFormat format; gboolean upstream = FALSE; gst_query_parse_position (query, &format, NULL); GST_DEBUG_OBJECT (basesink, "position query in format %s", gst_format_get_name (format)); /* first try to get the position based on the clock */ if ((res = gst_base_sink_get_position (basesink, format, &cur, &upstream))) { gst_query_set_position (query, format, cur); } else if (upstream) { /* fallback to peer query */ res = gst_pad_peer_query (basesink->sinkpad, query); } if (!res) { /* we can handle a few things if upstream failed */ if (format == GST_FORMAT_PERCENT) { gint64 dur = 0; res = gst_base_sink_get_position (basesink, GST_FORMAT_TIME, &cur, &upstream); if (!res && upstream) { res = gst_pad_peer_query_position (basesink->sinkpad, GST_FORMAT_TIME, &cur); } if (res) { res = gst_base_sink_get_duration (basesink, GST_FORMAT_TIME, &dur, &upstream); if (!res && upstream) { res = gst_pad_peer_query_duration (basesink->sinkpad, GST_FORMAT_TIME, &dur); } } if (res) { gint64 pos; pos = gst_util_uint64_scale (100 * GST_FORMAT_PERCENT_SCALE, cur, dur); gst_query_set_position (query, GST_FORMAT_PERCENT, pos); } } } break; } case GST_QUERY_DURATION: { gint64 dur = 0; GstFormat format; gboolean upstream = FALSE; gst_query_parse_duration (query, &format, NULL); GST_DEBUG_OBJECT (basesink, "duration query in format %s", gst_format_get_name (format)); if ((res = gst_base_sink_get_duration (basesink, format, &dur, &upstream))) { gst_query_set_duration (query, format, dur); } else if (upstream) { /* fallback to peer query */ res = gst_pad_peer_query (basesink->sinkpad, query); } if (!res) { /* we can handle a few things if upstream failed */ if (format == GST_FORMAT_PERCENT) { gst_query_set_duration (query, GST_FORMAT_PERCENT, GST_FORMAT_PERCENT_MAX); res = TRUE; } } break; } case GST_QUERY_LATENCY: { gboolean live, us_live; GstClockTime min, max; if ((res = gst_base_sink_query_latency (basesink, &live, &us_live, &min, &max))) { gst_query_set_latency (query, live, min, max); } break; } case GST_QUERY_JITTER: break; case GST_QUERY_RATE: /* gst_query_set_rate (query, basesink->segment_rate); */ res = TRUE; break; case GST_QUERY_SEGMENT: { if (basesink->pad_mode == GST_PAD_MODE_PULL) { gst_query_set_segment (query, basesink->segment.rate, GST_FORMAT_TIME, basesink->segment.start, basesink->segment.stop); res = TRUE; } else { res = gst_pad_peer_query (basesink->sinkpad, query); } break; } case GST_QUERY_SEEKING: case GST_QUERY_CONVERT: case GST_QUERY_FORMATS: default: res = gst_pad_peer_query (basesink->sinkpad, query); break; } GST_DEBUG_OBJECT (basesink, "query %s returns %d", GST_QUERY_TYPE_NAME (query), res); return res; } static gboolean gst_base_sink_default_query (GstBaseSink * basesink, GstQuery * query) { gboolean res; GstBaseSinkClass *bclass; bclass = GST_BASE_SINK_GET_CLASS (basesink); switch (GST_QUERY_TYPE (query)) { case GST_QUERY_ALLOCATION: { if (bclass->propose_allocation) res = bclass->propose_allocation (basesink, query); else res = FALSE; break; } case GST_QUERY_CAPS: { GstCaps *caps, *filter; gst_query_parse_caps (query, &filter); caps = gst_base_sink_query_caps (basesink, basesink->sinkpad, filter); gst_query_set_caps_result (query, caps); gst_caps_unref (caps); res = TRUE; break; } case GST_QUERY_ACCEPT_CAPS: { GstCaps *caps, *allowed; gboolean subset; /* slightly faster than the default implementation */ gst_query_parse_accept_caps (query, &caps); allowed = gst_base_sink_query_caps (basesink, basesink->sinkpad, NULL); subset = gst_caps_is_subset (caps, allowed); gst_caps_unref (allowed); gst_query_set_accept_caps_result (query, subset); res = TRUE; break; } case GST_QUERY_DRAIN: res = TRUE; break; default: res = gst_pad_query_default (basesink->sinkpad, GST_OBJECT_CAST (basesink), query); break; } return res; } static gboolean gst_base_sink_sink_query (GstPad * pad, GstObject * parent, GstQuery * query) { GstBaseSink *basesink; GstBaseSinkClass *bclass; gboolean res; basesink = GST_BASE_SINK_CAST (parent); bclass = GST_BASE_SINK_GET_CLASS (basesink); if (bclass->query) res = bclass->query (basesink, query); else res = FALSE; return res; } static GstStateChangeReturn gst_base_sink_change_state (GstElement * element, GstStateChange transition) { GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS; GstBaseSink *basesink = GST_BASE_SINK (element); GstBaseSinkClass *bclass; GstBaseSinkPrivate *priv; priv = basesink->priv; bclass = GST_BASE_SINK_GET_CLASS (basesink); switch (transition) { case GST_STATE_CHANGE_NULL_TO_READY: if (bclass->start) if (!bclass->start (basesink)) goto start_failed; break; case GST_STATE_CHANGE_READY_TO_PAUSED: /* need to complete preroll before this state change completes, there * is no data flow in READY so we can safely assume we need to preroll. */ GST_BASE_SINK_PREROLL_LOCK (basesink); GST_DEBUG_OBJECT (basesink, "READY to PAUSED"); basesink->have_newsegment = FALSE; gst_segment_init (&basesink->segment, GST_FORMAT_UNDEFINED); basesink->offset = 0; basesink->have_preroll = FALSE; priv->step_unlock = FALSE; basesink->need_preroll = TRUE; basesink->playing_async = TRUE; priv->current_sstart = GST_CLOCK_TIME_NONE; priv->current_sstop = GST_CLOCK_TIME_NONE; priv->eos_rtime = GST_CLOCK_TIME_NONE; priv->latency = 0; basesink->eos = FALSE; priv->received_eos = FALSE; gst_base_sink_reset_qos (basesink); priv->commited = FALSE; priv->call_preroll = TRUE; priv->current_step.valid = FALSE; priv->pending_step.valid = FALSE; if (priv->async_enabled) { GST_DEBUG_OBJECT (basesink, "doing async state change"); /* when async enabled, post async-start message and return ASYNC from * the state change function */ ret = GST_STATE_CHANGE_ASYNC; gst_element_post_message (GST_ELEMENT_CAST (basesink), gst_message_new_async_start (GST_OBJECT_CAST (basesink))); } else { priv->have_latency = TRUE; } GST_BASE_SINK_PREROLL_UNLOCK (basesink); break; case GST_STATE_CHANGE_PAUSED_TO_PLAYING: GST_BASE_SINK_PREROLL_LOCK (basesink); g_atomic_int_set (&basesink->priv->to_playing, TRUE); if (!gst_base_sink_needs_preroll (basesink)) { GST_DEBUG_OBJECT (basesink, "PAUSED to PLAYING, don't need preroll"); /* no preroll needed anymore now. */ basesink->playing_async = FALSE; basesink->need_preroll = FALSE; if (basesink->eos) { GstMessage *message; /* need to post EOS message here */ GST_DEBUG_OBJECT (basesink, "Now posting EOS"); message = gst_message_new_eos (GST_OBJECT_CAST (basesink)); gst_message_set_seqnum (message, basesink->priv->seqnum); gst_element_post_message (GST_ELEMENT_CAST (basesink), message); } else { GST_DEBUG_OBJECT (basesink, "signal preroll"); GST_BASE_SINK_PREROLL_SIGNAL (basesink); } } else { GST_DEBUG_OBJECT (basesink, "PAUSED to PLAYING, we are not prerolled"); basesink->need_preroll = TRUE; basesink->playing_async = TRUE; priv->call_preroll = TRUE; priv->commited = FALSE; if (priv->async_enabled) { GST_DEBUG_OBJECT (basesink, "doing async state change"); ret = GST_STATE_CHANGE_ASYNC; gst_element_post_message (GST_ELEMENT_CAST (basesink), gst_message_new_async_start (GST_OBJECT_CAST (basesink))); } } GST_BASE_SINK_PREROLL_UNLOCK (basesink); break; default: break; } { GstStateChangeReturn bret; bret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); if (G_UNLIKELY (bret == GST_STATE_CHANGE_FAILURE)) goto activate_failed; } switch (transition) { case GST_STATE_CHANGE_PAUSED_TO_PLAYING: /* completed transition, so need not be marked any longer * And it should be unmarked, since e.g. losing our position upon flush * does not really change state to PAUSED ... */ g_atomic_int_set (&basesink->priv->to_playing, FALSE); break; case GST_STATE_CHANGE_PLAYING_TO_PAUSED: g_atomic_int_set (&basesink->priv->to_playing, FALSE); GST_DEBUG_OBJECT (basesink, "PLAYING to PAUSED"); /* FIXME, make sure we cannot enter _render first */ /* we need to call ::unlock before locking PREROLL_LOCK * since we lock it before going into ::render */ if (bclass->unlock) bclass->unlock (basesink); GST_BASE_SINK_PREROLL_LOCK (basesink); GST_DEBUG_OBJECT (basesink, "got preroll lock"); /* now that we have the PREROLL lock, clear our unlock request */ if (bclass->unlock_stop) bclass->unlock_stop (basesink); /* we need preroll again and we set the flag before unlocking the clockid * because if the clockid is unlocked before a current buffer expired, we * can use that buffer to preroll with */ basesink->need_preroll = TRUE; if (basesink->clock_id) { GST_DEBUG_OBJECT (basesink, "unschedule clock"); gst_clock_id_unschedule (basesink->clock_id); } /* if we don't have a preroll buffer we need to wait for a preroll and * return ASYNC. */ if (!gst_base_sink_needs_preroll (basesink)) { GST_DEBUG_OBJECT (basesink, "PLAYING to PAUSED, we are prerolled"); basesink->playing_async = FALSE; } else { if (GST_STATE_TARGET (GST_ELEMENT (basesink)) <= GST_STATE_READY) { GST_DEBUG_OBJECT (basesink, "element is <= READY"); ret = GST_STATE_CHANGE_SUCCESS; } else { GST_DEBUG_OBJECT (basesink, "PLAYING to PAUSED, we are not prerolled"); basesink->playing_async = TRUE; priv->commited = FALSE; priv->call_preroll = TRUE; if (priv->async_enabled) { GST_DEBUG_OBJECT (basesink, "doing async state change"); ret = GST_STATE_CHANGE_ASYNC; gst_element_post_message (GST_ELEMENT_CAST (basesink), gst_message_new_async_start (GST_OBJECT_CAST (basesink))); } } } GST_DEBUG_OBJECT (basesink, "rendered: %" G_GUINT64_FORMAT ", dropped: %" G_GUINT64_FORMAT, priv->rendered, priv->dropped); gst_base_sink_reset_qos (basesink); GST_BASE_SINK_PREROLL_UNLOCK (basesink); break; case GST_STATE_CHANGE_PAUSED_TO_READY: GST_BASE_SINK_PREROLL_LOCK (basesink); /* start by resetting our position state with the object lock so that the * position query gets the right idea. We do this before we post the * messages so that the message handlers pick this up. */ GST_OBJECT_LOCK (basesink); basesink->have_newsegment = FALSE; priv->current_sstart = GST_CLOCK_TIME_NONE; priv->current_sstop = GST_CLOCK_TIME_NONE; priv->have_latency = FALSE; if (priv->cached_clock_id) { gst_clock_id_unref (priv->cached_clock_id); priv->cached_clock_id = NULL; } gst_caps_replace (&basesink->priv->caps, NULL); GST_OBJECT_UNLOCK (basesink); gst_base_sink_set_last_buffer (basesink, NULL); priv->call_preroll = FALSE; if (!priv->commited) { if (priv->async_enabled) { GST_DEBUG_OBJECT (basesink, "PAUSED to READY, posting async-done"); gst_element_post_message (GST_ELEMENT_CAST (basesink), gst_message_new_state_changed (GST_OBJECT_CAST (basesink), GST_STATE_PLAYING, GST_STATE_PAUSED, GST_STATE_READY)); gst_element_post_message (GST_ELEMENT_CAST (basesink), gst_message_new_async_done (GST_OBJECT_CAST (basesink), GST_CLOCK_TIME_NONE)); } priv->commited = TRUE; } else { GST_DEBUG_OBJECT (basesink, "PAUSED to READY, don't need_preroll"); } GST_BASE_SINK_PREROLL_UNLOCK (basesink); break; case GST_STATE_CHANGE_READY_TO_NULL: if (bclass->stop) { if (!bclass->stop (basesink)) { GST_WARNING_OBJECT (basesink, "failed to stop"); } } gst_base_sink_set_last_buffer (basesink, NULL); priv->call_preroll = FALSE; break; default: break; } return ret; /* ERRORS */ start_failed: { GST_DEBUG_OBJECT (basesink, "failed to start"); return GST_STATE_CHANGE_FAILURE; } activate_failed: { GST_DEBUG_OBJECT (basesink, "element failed to change states -- activation problem?"); return GST_STATE_CHANGE_FAILURE; } }