/* 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, #GstBaseSource * * #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 base_init function, like so: * * static void * my_element_base_init (gpointer g_class) * { * GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class); * * // 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)); * // see #GstElementDetails * gst_element_class_set_details (gstelement_class, &details); * } * * * #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 * #GstBaseSink::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 ::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 "sync" * property to %FALSE. * * After synchronisation the virtual method #GstBaseSink::render will be called. * Subclasses should minimally implement this method. * * Since 0.10.3 subclasses that synchronise on the clock in the ::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. Since 0.10.11 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 * ::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_NEWSEGMENT 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 ::set_caps function will be called when the subclass should configure * itself to process a specific media type. * * The ::start and ::stop virtual methods will be called when resources should * be allocated. Any ::preroll, ::render and ::set_caps function will be * called between the ::start and ::stop calls. * * The ::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. * * #GstBaseSink provides an overridable ::buffer_alloc function that can be * used by sinks that want to do reverse negotiation or to provide * custom buffers (hardware buffers for example) to upstream elements. * * The ::unlock method is called when the elements should unblock any blocking * operations they perform in the ::render method. This is mostly useful when * the ::render method performs a blocking write on a file descriptor, for * example. * * The 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 ::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 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. * * Since 0.10.15 the 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 "gstbasesink.h" #include #include #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)) /* FIXME, some stuff in ABI.data and other in Private... * Make up your mind please. */ 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; /* EOS sync time in running time */ GstClockTime eos_rtime; /* last buffer that arrived in time, running time */ GstClockTime last_in_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; /* 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; /* 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 */ GstBuffer *last_buffer; }; #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_SIZE 1024 #define DEFAULT_CAN_ACTIVATE_PULL FALSE /* fixme: enable me */ #define DEFAULT_CAN_ACTIVATE_PUSH TRUE #define DEFAULT_PREROLL_QUEUE_LEN 0 #define DEFAULT_SYNC TRUE #define DEFAULT_MAX_LATENESS -1 #define DEFAULT_QOS FALSE #define DEFAULT_ASYNC TRUE #define DEFAULT_TS_OFFSET 0 enum { PROP_0, PROP_PREROLL_QUEUE_LEN, PROP_SYNC, PROP_MAX_LATENESS, PROP_QOS, PROP_ASYNC, PROP_TS_OFFSET, PROP_LAST_BUFFER, 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 GType base_sink_type = 0; if (G_UNLIKELY (base_sink_type == 0)) { 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, }; base_sink_type = g_type_register_static (GST_TYPE_ELEMENT, "GstBaseSink", &base_sink_info, G_TYPE_FLAG_ABSTRACT); } 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 gst_base_sink_query (GstElement * element, GstQuery * query); static GstCaps *gst_base_sink_get_caps (GstBaseSink * sink); static gboolean gst_base_sink_set_caps (GstBaseSink * sink, GstCaps * caps); static GstFlowReturn gst_base_sink_buffer_alloc (GstBaseSink * sink, guint64 offset, guint size, GstCaps * caps, GstBuffer ** buf); static void gst_base_sink_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 GstStateChangeReturn gst_base_sink_change_state (GstElement * element, GstStateChange transition); static GstFlowReturn gst_base_sink_chain (GstPad * pad, GstBuffer * buffer); static void gst_base_sink_loop (GstPad * pad); static gboolean gst_base_sink_pad_activate (GstPad * pad); static gboolean gst_base_sink_pad_activate_push (GstPad * pad, gboolean active); static gboolean gst_base_sink_pad_activate_pull (GstPad * pad, gboolean active); static gboolean gst_base_sink_event (GstPad * pad, GstEvent * event); static gboolean gst_base_sink_peer_query (GstBaseSink * sink, GstQuery * query); /* check if an object was too late */ static gboolean gst_base_sink_is_too_late (GstBaseSink * basesink, GstMiniObject * obj, GstClockTime start, GstClockTime stop, 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_DEBUG_FUNCPTR (gst_base_sink_finalize); gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_base_sink_set_property); gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_base_sink_get_property); /* FIXME, this next value should be configured using an event from the * upstream element, ie, the BUFFER_SIZE event. */ g_object_class_install_property (gobject_class, PROP_PREROLL_QUEUE_LEN, g_param_spec_uint ("preroll-queue-len", "Preroll queue length", "Number of buffers to queue during preroll", 0, G_MAXUINT, DEFAULT_PREROLL_QUEUE_LEN, G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS)); 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. * * Since: 0.10.15 */ 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. * * Since: 0.10.15 */ 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:last-buffer * * 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. * * Since: 0.10.15 */ g_object_class_install_property (gobject_class, PROP_LAST_BUFFER, gst_param_spec_mini_object ("last-buffer", "Last Buffer", "The last buffer received in the sink", GST_TYPE_BUFFER, G_PARAM_READABLE | 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 (gst_base_sink_query); klass->get_caps = GST_DEBUG_FUNCPTR (gst_base_sink_get_caps); klass->set_caps = GST_DEBUG_FUNCPTR (gst_base_sink_set_caps); klass->buffer_alloc = GST_DEBUG_FUNCPTR (gst_base_sink_buffer_alloc); klass->get_times = GST_DEBUG_FUNCPTR (gst_base_sink_get_times); klass->activate_pull = GST_DEBUG_FUNCPTR (gst_base_sink_default_activate_pull); } static GstCaps * gst_base_sink_pad_getcaps (GstPad * pad) { GstBaseSinkClass *bclass; GstBaseSink *bsink; GstCaps *caps = NULL; bsink = GST_BASE_SINK (gst_pad_get_parent (pad)); bclass = GST_BASE_SINK_GET_CLASS (bsink); if (bclass->get_caps) caps = bclass->get_caps (bsink); 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_caps_ref (gst_pad_template_get_caps (pad_template)); } } gst_object_unref (bsink); return caps; } static gboolean gst_base_sink_pad_setcaps (GstPad * pad, GstCaps * caps) { GstBaseSinkClass *bclass; GstBaseSink *bsink; gboolean res = TRUE; bsink = GST_BASE_SINK (gst_pad_get_parent (pad)); bclass = GST_BASE_SINK_GET_CLASS (bsink); if (bsink->pad_mode == GST_ACTIVATE_PULL) { GstPad *peer = gst_pad_get_peer (pad); if (peer) res = gst_pad_set_caps (peer, caps); else res = FALSE; if (!res) GST_DEBUG_OBJECT (bsink, "peer setcaps() failed"); } if (res && bclass->set_caps) res = bclass->set_caps (bsink, caps); gst_object_unref (bsink); return res; } static void gst_base_sink_pad_fixate (GstPad * pad, GstCaps * caps) { GstBaseSinkClass *bclass; GstBaseSink *bsink; bsink = GST_BASE_SINK (gst_pad_get_parent (pad)); bclass = GST_BASE_SINK_GET_CLASS (bsink); if (bclass->fixate) bclass->fixate (bsink, caps); gst_object_unref (bsink); } static GstFlowReturn gst_base_sink_pad_buffer_alloc (GstPad * pad, guint64 offset, guint size, GstCaps * caps, GstBuffer ** buf) { GstBaseSinkClass *bclass; GstBaseSink *bsink; GstFlowReturn result = GST_FLOW_OK; bsink = GST_BASE_SINK (gst_pad_get_parent (pad)); bclass = GST_BASE_SINK_GET_CLASS (bsink); if (bclass->buffer_alloc) result = bclass->buffer_alloc (bsink, offset, size, caps, buf); else *buf = NULL; /* fallback in gstpad.c will allocate generic buffer */ gst_object_unref (bsink); return result; } 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_getcaps_function (basesink->sinkpad, GST_DEBUG_FUNCPTR (gst_base_sink_pad_getcaps)); gst_pad_set_setcaps_function (basesink->sinkpad, GST_DEBUG_FUNCPTR (gst_base_sink_pad_setcaps)); gst_pad_set_fixatecaps_function (basesink->sinkpad, GST_DEBUG_FUNCPTR (gst_base_sink_pad_fixate)); gst_pad_set_bufferalloc_function (basesink->sinkpad, GST_DEBUG_FUNCPTR (gst_base_sink_pad_buffer_alloc)); gst_pad_set_activate_function (basesink->sinkpad, GST_DEBUG_FUNCPTR (gst_base_sink_pad_activate)); gst_pad_set_activatepush_function (basesink->sinkpad, GST_DEBUG_FUNCPTR (gst_base_sink_pad_activate_push)); gst_pad_set_activatepull_function (basesink->sinkpad, GST_DEBUG_FUNCPTR (gst_base_sink_pad_activate_pull)); gst_pad_set_event_function (basesink->sinkpad, GST_DEBUG_FUNCPTR (gst_base_sink_event)); gst_pad_set_chain_function (basesink->sinkpad, GST_DEBUG_FUNCPTR (gst_base_sink_chain)); gst_element_add_pad (GST_ELEMENT_CAST (basesink), basesink->sinkpad); basesink->pad_mode = GST_ACTIVATE_NONE; basesink->preroll_queue = g_queue_new (); basesink->abidata.ABI.clip_segment = gst_segment_new (); 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->abidata.ABI.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 = 0; GST_OBJECT_FLAG_SET (basesink, GST_ELEMENT_IS_SINK); } static void gst_base_sink_finalize (GObject * object) { GstBaseSink *basesink; basesink = GST_BASE_SINK (object); g_queue_free (basesink->preroll_queue); gst_segment_free (basesink->abidata.ABI.clip_segment); 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, incomming 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. * * Since: 0.10.4 */ 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. * * Since: 0.10.4 */ 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. * * Since: 0.10.4 */ 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->abidata.ABI.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. * * Since: 0.10.4 */ 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->abidata.ABI.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. * * Since: 0.10.5 */ 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. * * Since: 0.10.5 */ 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 immediatly go to PAUSED instead of waiting for a * preroll buffer. This feature is usefull if the sink does not synchronize * against the clock or when it is dealing with sparse streams. * * Since: 0.10.15 */ void gst_base_sink_set_async_enabled (GstBaseSink * sink, gboolean enabled) { g_return_if_fail (GST_IS_BASE_SINK (sink)); GST_PAD_PREROLL_LOCK (sink->sinkpad); sink->priv->async_enabled = enabled; GST_LOG_OBJECT (sink, "set async enabled to %d", enabled); GST_PAD_PREROLL_UNLOCK (sink->sinkpad); } /** * 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. * * Since: 0.10.15 */ gboolean gst_base_sink_is_async_enabled (GstBaseSink * sink) { gboolean res; g_return_val_if_fail (GST_IS_BASE_SINK (sink), FALSE); GST_PAD_PREROLL_LOCK (sink->sinkpad); res = sink->priv->async_enabled; GST_PAD_PREROLL_UNLOCK (sink->sinkpad); 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. * * Since: 0.10.15 */ 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. * * Since: 0.10.15 */ 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_buffer: * @sink: the sink * * Get the last buffer 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 buffer can be used to determine the type of the buffer. * * Returns: a #GstBuffer. gst_buffer_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. * * Since: 0.10.15 */ GstBuffer * gst_base_sink_get_last_buffer (GstBaseSink * sink) { GstBuffer *res; g_return_val_if_fail (GST_IS_BASE_SINK (sink), NULL); GST_OBJECT_LOCK (sink); if ((res = sink->priv->last_buffer)) gst_buffer_ref (res); GST_OBJECT_UNLOCK (sink); return res; } static void gst_base_sink_set_last_buffer (GstBaseSink * sink, GstBuffer * buffer) { GstBuffer *old; if (buffer) gst_buffer_ref (buffer); GST_OBJECT_LOCK (sink); old = sink->priv->last_buffer; sink->priv->last_buffer = buffer; GST_OBJECT_UNLOCK (sink); if (old) gst_buffer_unref (old); } /** * gst_base_sink_get_latency: * @sink: the sink * * Get the currently configured latency. * * Returns: The configured latency. * * Since: 0.10.12 */ 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: if the sink is live * @upstream_live: if an upstream element is live * @min_latency: the min latency of the upstream elements * @max_latency: 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. * * Since: 0.10.12 */ 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_base_sink_peer_query (sink, 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. * * Since: 0.10.21 */ void gst_base_sink_set_render_delay (GstBaseSink * sink, GstClockTime delay) { g_return_if_fail (GST_IS_BASE_SINK (sink)); GST_OBJECT_LOCK (sink); sink->priv->render_delay = delay; GST_LOG_OBJECT (sink, "set render delay to %" GST_TIME_FORMAT, GST_TIME_ARGS (delay)); GST_OBJECT_UNLOCK (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. * * Since: 0.10.21 */ 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; } 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_PREROLL_QUEUE_LEN: /* preroll lock necessary to serialize with finish_preroll */ GST_PAD_PREROLL_LOCK (sink->sinkpad); sink->preroll_queue_max_len = g_value_get_uint (value); GST_PAD_PREROLL_UNLOCK (sink->sinkpad); break; 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; 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_PREROLL_QUEUE_LEN: GST_PAD_PREROLL_LOCK (sink->sinkpad); g_value_set_uint (value, sink->preroll_queue_max_len); GST_PAD_PREROLL_UNLOCK (sink->sinkpad); break; 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_BUFFER: gst_value_take_buffer (value, gst_base_sink_get_last_buffer (sink)); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static GstCaps * gst_base_sink_get_caps (GstBaseSink * sink) { return NULL; } static gboolean gst_base_sink_set_caps (GstBaseSink * sink, GstCaps * caps) { return TRUE; } static GstFlowReturn gst_base_sink_buffer_alloc (GstBaseSink * sink, guint64 offset, guint size, GstCaps * caps, GstBuffer ** buf) { *buf = NULL; return GST_FLOW_OK; } /* with PREROLL_LOCK, STREAM_LOCK */ static void gst_base_sink_preroll_queue_flush (GstBaseSink * basesink, GstPad * pad) { GstMiniObject *obj; GST_DEBUG_OBJECT (basesink, "flushing queue %p", basesink); while ((obj = g_queue_pop_head (basesink->preroll_queue))) { GST_DEBUG_OBJECT (basesink, "popped %p", obj); gst_mini_object_unref (obj); } /* we can't have EOS anymore now */ basesink->eos = FALSE; basesink->priv->received_eos = FALSE; basesink->have_preroll = FALSE; basesink->eos_queued = FALSE; basesink->preroll_queued = 0; basesink->buffers_queued = 0; basesink->events_queued = 0; /* 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_PAD_PREROLL_SIGNAL (pad); } /* with STREAM_LOCK, configures given segment with the event information. */ static void gst_base_sink_configure_segment (GstBaseSink * basesink, GstPad * pad, GstEvent * event, GstSegment * segment) { gboolean update; gdouble rate, arate; GstFormat format; gint64 start; gint64 stop; gint64 time; /* 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_parse_new_segment_full (event, &update, &rate, &arate, &format, &start, &stop, &time); /* 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); gst_segment_set_newsegment_full (segment, update, rate, arate, format, start, stop, time); if (format == GST_FORMAT_TIME) { GST_DEBUG_OBJECT (basesink, "configured NEWSEGMENT update %d, rate %lf, applied rate %lf, " "format GST_FORMAT_TIME, " "%" GST_TIME_FORMAT " -- %" GST_TIME_FORMAT ", time %" GST_TIME_FORMAT ", accum %" GST_TIME_FORMAT, update, rate, arate, GST_TIME_ARGS (segment->start), GST_TIME_ARGS (segment->stop), GST_TIME_ARGS (segment->time), GST_TIME_ARGS (segment->accum)); } else { GST_DEBUG_OBJECT (basesink, "configured NEWSEGMENT update %d, rate %lf, applied rate %lf, " "format %d, " "%" G_GINT64_FORMAT " -- %" G_GINT64_FORMAT ", time %" G_GINT64_FORMAT ", accum %" G_GINT64_FORMAT, update, rate, arate, segment->format, segment->start, segment->stop, segment->time, segment->accum); } GST_OBJECT_UNLOCK (basesink); } /* 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; gboolean sync; /* 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; sync = basesink->sync; switch (pending) { case GST_STATE_PLAYING: { GstBaseSinkClass *bclass; GstStateChangeReturn ret; bclass = GST_BASE_SINK_GET_CLASS (basesink); 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; } /* make sure we notify the subclass of async playing */ if (bclass->async_play) { ret = bclass->async_play (basesink); if (ret == GST_STATE_CHANGE_FAILURE) goto async_failed; } 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))); } 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; } async_failed: { GST_DEBUG_OBJECT (basesink, "async commit failed"); GST_STATE_RETURN (basesink) = GST_STATE_CHANGE_FAILURE; GST_OBJECT_UNLOCK (basesink); return FALSE; } } /* 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, GstSegment * segment) { GstBaseSinkClass *bclass; GstBuffer *buffer; GstClockTime start, stop; /* raw start/stop timestamps */ gint64 cstart, cstop; /* clipped raw timestamps */ gint64 rstart, rstop; /* clipped timestamps converted to running time */ GstClockTime sstart, sstop; /* clipped timestamps converted to stream time */ GstFormat format; GstBaseSinkPrivate *priv; priv = basesink->priv; /* start with nothing */ start = stop = sstart = sstop = rstart = rstop = -1; 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 (basesink->segment.rate >= 0.0) { sstart = sstop = priv->current_sstop; if (sstart == -1) { /* we have not seen a buffer yet, use the segment values */ sstart = sstop = gst_segment_to_stream_time (&basesink->segment, basesink->segment.format, basesink->segment.stop); } } else { sstart = sstop = priv->current_sstart; if (sstart == -1) { /* we have not seen a buffer yet, use the segment values */ sstart = sstop = gst_segment_to_stream_time (&basesink->segment, basesink->segment.format, basesink->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)); goto done; } default: /* other events do not need syncing */ /* FIXME, maybe NEWSEGMENT might need synchronisation * since the POSITION query depends on accumulated times and * we cannot accumulate the current segment before the previous * one completed. */ return FALSE; } } /* else do buffer sync code */ buffer = GST_BUFFER_CAST (obj); bclass = GST_BASE_SINK_GET_CLASS (basesink); /* just get the times to see if we need syncing */ if (bclass->get_times) bclass->get_times (basesink, buffer, &start, &stop); if (start == -1) { gst_base_sink_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; /* no timestamp clipping if we did not * get a TIME segment format */ if (G_UNLIKELY (format != GST_FORMAT_TIME)) { cstart = start; cstop = stop; /* do running and stream time in TIME format */ format = GST_FORMAT_TIME; goto do_times; } /* clip */ if (G_UNLIKELY (!gst_segment_clip (segment, GST_FORMAT_TIME, (gint64) start, (gint64) stop, &cstart, &cstop))) 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 (cstop != GST_CLOCK_TIME_NONE)) gst_segment_set_last_stop (segment, GST_FORMAT_TIME, cstop); else gst_segment_set_last_stop (segment, GST_FORMAT_TIME, cstart); do_times: /* 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); rstart = gst_segment_to_running_time (segment, format, cstart); rstop = gst_segment_to_running_time (segment, format, cstop); done: /* 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: { /* should not happen since we clip them in the chain function already, * we return FALSE so that we don't try to sync on it. */ GST_ELEMENT_WARNING (basesink, STREAM, FAILED, (NULL), ("unexpected buffer out of segment found.")); 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 */ 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; return time; } /* gst_base_sink_wait_clock: * @sink: the sink * @time: the running_time to be reached * @jitter: the jitter to be filled with time diff (can be 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 ::event vmethod or when receiving a buffer in * the ::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. * * Since 0.10.20 * * Returns: #GstClockReturn */ GstClockReturn gst_base_sink_wait_clock (GstBaseSink * basesink, GstClockTime time, GstClockTimeDiff * jitter) { GstClockID id; GstClockReturn ret; GstClock *clock; if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (time))) goto invalid_time; GST_OBJECT_LOCK (basesink); if (G_UNLIKELY (!basesink->sync)) goto no_sync; if (G_UNLIKELY ((clock = GST_ELEMENT_CLOCK (basesink)) == NULL)) goto no_clock; /* add base_time to running_time to get the time against the clock */ time += GST_ELEMENT_CAST (basesink)->base_time; id = gst_clock_new_single_shot_id (clock, time); GST_OBJECT_UNLOCK (basesink); /* 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. */ basesink->clock_id = id; /* release the preroll lock while waiting */ GST_PAD_PREROLL_UNLOCK (basesink->sinkpad); ret = gst_clock_id_wait (id, jitter); GST_PAD_PREROLL_LOCK (basesink->sinkpad); gst_clock_id_unref (id); basesink->clock_id = NULL; return ret; /* no syncing needed */ invalid_time: { GST_DEBUG_OBJECT (basesink, "time not valid, no sync needed"); return GST_CLOCK_BADTIME; } no_sync: { GST_DEBUG_OBJECT (basesink, "sync disabled"); GST_OBJECT_UNLOCK (basesink); return GST_CLOCK_BADTIME; } no_clock: { GST_DEBUG_OBJECT (basesink, "no clock, can't sync"); GST_OBJECT_UNLOCK (basesink); 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_WRONG_STATE). * * Since: 0.10.11 * * 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_PAD_PREROLL_WAIT (sink->sinkpad); sink->have_preroll = FALSE; if (G_UNLIKELY (sink->flushing)) goto stopping; GST_DEBUG_OBJECT (sink, "continue after preroll"); return GST_FLOW_OK; /* ERRORS */ stopping: { GST_DEBUG_OBJECT (sink, "preroll interrupted"); return GST_FLOW_WRONG_STATE; } } /** * gst_base_sink_wait_eos: * @sink: the sink * @time: the running_time to be reached * @jitter: the jitter to be filled with time diff (can be NULL) * * This function will block until @time is reached. It is usually called by * subclasses that use their own internal synchronisation but want to let the * 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. * * The @time argument should be the running_time of when the EOS should happen * and will be adjusted with any latency and offset configured in the sink. * * Since 0.10.15 * * Returns: #GstFlowReturn */ GstFlowReturn gst_base_sink_wait_eos (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 */ if (G_UNLIKELY (sink->need_preroll)) { ret = gst_base_sink_wait_preroll (sink); if (ret != GST_FLOW_OK) 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 and ts_offset. We don't adjust for render delay * because we don't interact with the device on EOS normally. */ 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_WRONG_STATE; } } /* 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 * immediatly try to re-preroll. * * Some objects do not need synchronisation (most events) and so this function * immediatly 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, GstPad * pad, GstMiniObject * obj, gboolean * late) { GstClockTimeDiff jitter; gboolean syncable; GstClockReturn status = GST_CLOCK_OK; GstClockTime rstart, rstop, sstart, sstop, stime; gboolean do_sync; GstBaseSinkPrivate *priv; priv = basesink->priv; sstart = sstop = rstart = rstop = -1; do_sync = TRUE; priv->current_rstart = -1; /* 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, &basesink->segment); /* 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 = (rstop != -1 ? rstop : rstart); /* save sync time for eos when the previous object needed sync */ priv->eos_rtime = (do_sync ? priv->current_rstop : -1); 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. */ while (G_UNLIKELY (basesink->need_preroll)) { GST_DEBUG_OBJECT (basesink, "prerolling object %p", obj); if (G_LIKELY (basesink->playing_async)) { /* commit state */ if (G_UNLIKELY (!gst_base_sink_commit_state (basesink))) goto stopping; } /* need to recheck here because the commit state could have * made us not need the preroll anymore */ if (G_LIKELY (basesink->need_preroll)) { /* block until the state changes, or we get a flush, or something */ if (gst_base_sink_wait_preroll (basesink) != GST_FLOW_OK) goto flushing; } } /* 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 = (sstop != -1 ? sstop : sstart); GST_OBJECT_UNLOCK (basesink); if (!do_sync) goto done; /* adjust for latency */ stime = gst_base_sink_adjust_time (basesink, rstart); /* adjust for render-delay, avoid underflows */ if (stime != -1) { if (stime > priv->render_delay) stime -= priv->render_delay; else stime = 0; } /* 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 immediatly 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", status); /* 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"); 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 */ not_syncable: { GST_DEBUG_OBJECT (basesink, "non syncable object %p", obj); return GST_FLOW_OK; } flushing: { GST_DEBUG_OBJECT (basesink, "we are flushing"); return GST_FLOW_WRONG_STATE; } stopping: { GST_DEBUG_OBJECT (basesink, "stopping while commiting state"); return GST_FLOW_WRONG_STATE; } } static gboolean gst_base_sink_send_qos (GstBaseSink * basesink, gdouble proportion, GstClockTime time, GstClockTimeDiff diff) { GstEvent *event; gboolean res; /* generate Quality-of-Service event */ GST_CAT_DEBUG_OBJECT (GST_CAT_QOS, basesink, "qos: proportion: %lf, diff %" G_GINT64_FORMAT ", timestamp %" GST_TIME_FORMAT, proportion, diff, GST_TIME_ARGS (time)); event = gst_event_new_qos (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 Quality-of-Service disabled, do nothing */ if (!g_atomic_int_get (&priv->qos_enabled) || start == -1) 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 (stop != -1) duration = stop - start; else duration = -1; /* if we have the time when the last buffer left us, calculate * processing time */ if (priv->last_left != -1) { 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 ", 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 (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 (priv->avg_duration == -1) priv->avg_duration = duration; else priv->avg_duration = UPDATE_RUNNING_AVG (priv->avg_duration, duration); if (priv->avg_pt == -1) 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 = 0.0; if (priv->last_left != -1) { 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) { /* 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; } gst_base_sink_send_qos (sink, priv->avg_rate, priv->current_rstart, priv->current_jitter); } /* 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_in_time = -1; priv->last_left = -1; priv->avg_duration = -1; priv->avg_pt = -1; priv->avg_rate = -1.0; priv->avg_render = -1; priv->rendered = 0; priv->dropped = 0; } /* Checks if the object was scheduled too late. * * start/stop contain the raw timestamp 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 start, GstClockTime stop, GstClockReturn status, GstClockTimeDiff jitter) { gboolean late; gint64 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->abidata.ABI.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 (start == -1)) goto no_timestamp; /* we can add a valid stop time */ if (stop != -1) max_lateness += stop; else max_lateness += start; /* if the jitter bigger than duration and lateness we are too late */ if ((late = start + jitter > max_lateness)) { GST_DEBUG_OBJECT (basesink, "buffer is too late %" GST_TIME_FORMAT " > %" GST_TIME_FORMAT, GST_TIME_ARGS (start + 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 (priv->last_in_time && start - priv->last_in_time > GST_SECOND) { late = FALSE; GST_DEBUG_OBJECT (basesink, "**emergency** last buffer at %" GST_TIME_FORMAT " > GST_SECOND", GST_TIME_ARGS (priv->last_in_time)); } } done: if (!late) { priv->last_in_time = start; } 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 (priv->avg_render == -1) 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)); } } /* with STREAM_LOCK, PREROLL_LOCK, * * Synchronize the object on the clock and then render it. * * takes ownership of obj. */ static GstFlowReturn gst_base_sink_render_object (GstBaseSink * basesink, GstPad * pad, GstMiniObject * obj) { GstFlowReturn ret = GST_FLOW_OK; GstBaseSinkClass *bclass; gboolean late = FALSE; GstBaseSinkPrivate *priv; priv = basesink->priv; /* synchronize this object, non syncable objects return OK * immediatly. */ ret = gst_base_sink_do_sync (basesink, pad, obj, &late); if (G_UNLIKELY (ret != GST_FLOW_OK)) goto sync_failed; /* and now render, event or buffer. */ if (G_LIKELY (GST_IS_BUFFER (obj))) { GstBuffer *buf; /* drop late buffers unconditionally, let's hope it's unlikely */ if (G_UNLIKELY (late)) goto dropped; buf = GST_BUFFER_CAST (obj); gst_base_sink_set_last_buffer (basesink, buf); bclass = GST_BASE_SINK_GET_CLASS (basesink); if (G_LIKELY (bclass->render)) { gint do_qos; /* read once, to get same value before and after */ do_qos = g_atomic_int_get (&priv->qos_enabled); GST_DEBUG_OBJECT (basesink, "rendering buffer %p", obj); /* record rendering time for QoS and stats */ if (do_qos) gst_base_sink_do_render_stats (basesink, TRUE); ret = bclass->render (basesink, buf); priv->rendered++; if (do_qos) gst_base_sink_do_render_stats (basesink, FALSE); } } else { GstEvent *event = GST_EVENT_CAST (obj); gboolean event_res = TRUE; GstEventType type; bclass = GST_BASE_SINK_GET_CLASS (basesink); type = GST_EVENT_TYPE (event); GST_DEBUG_OBJECT (basesink, "rendering event %p, type %s", obj, gst_event_type_get_name (type)); if (bclass->event) event_res = bclass->event (basesink, event); /* when we get here we could be flushing again when the event handler calls * _wait_eos(). We have to ignore this object in that case. */ if (G_UNLIKELY (basesink->flushing)) goto flushing; if (G_LIKELY (event_res)) { switch (type) { case GST_EVENT_EOS: /* 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"); gst_element_post_message (GST_ELEMENT_CAST (basesink), gst_message_new_eos (GST_OBJECT_CAST (basesink))); break; case GST_EVENT_NEWSEGMENT: /* configure the segment */ gst_base_sink_configure_segment (basesink, pad, event, &basesink->segment); break; default: break; } } } done: gst_base_sink_perform_qos (basesink, late); GST_DEBUG_OBJECT (basesink, "object unref after render %p", obj); gst_mini_object_unref (obj); return ret; /* ERRORS */ 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"); goto done; } flushing: { GST_DEBUG_OBJECT (basesink, "we are flushing, ignore object"); gst_mini_object_unref (obj); return GST_FLOW_WRONG_STATE; } } /* with STREAM_LOCK, PREROLL_LOCK * * Perform preroll on the given object. For buffers this means * calling the preroll subclass method. * If that succeeds, the state will be commited. * * function does not take ownership of obj. */ static GstFlowReturn gst_base_sink_preroll_object (GstBaseSink * basesink, GstPad * pad, GstMiniObject * obj) { GstFlowReturn ret; GST_DEBUG_OBJECT (basesink, "do preroll %p", obj); /* if it's a buffer, we need to call the preroll method */ if (G_LIKELY (GST_IS_BUFFER (obj))) { GstBaseSinkClass *bclass; GstBuffer *buf; GstClockTime timestamp; buf = GST_BUFFER_CAST (obj); timestamp = GST_BUFFER_TIMESTAMP (buf); GST_DEBUG_OBJECT (basesink, "preroll buffer %" GST_TIME_FORMAT, GST_TIME_ARGS (timestamp)); gst_base_sink_set_last_buffer (basesink, buf); bclass = GST_BASE_SINK_GET_CLASS (basesink); if (bclass->preroll) if ((ret = bclass->preroll (basesink, buf)) != GST_FLOW_OK) goto preroll_failed; } /* commit state */ if (G_LIKELY (basesink->playing_async)) { if (G_UNLIKELY (!gst_base_sink_commit_state (basesink))) goto stopping; } return GST_FLOW_OK; /* ERRORS */ preroll_failed: { GST_DEBUG_OBJECT (basesink, "preroll failed, abort state"); gst_element_abort_state (GST_ELEMENT_CAST (basesink)); return ret; } stopping: { GST_DEBUG_OBJECT (basesink, "stopping while commiting state"); return GST_FLOW_WRONG_STATE; } } /* with STREAM_LOCK, PREROLL_LOCK * * Queue an object for rendering. * The first prerollable object queued will complete the preroll. If the * preroll queue if filled, we render all the objects in the queue. * * This function takes ownership of the object. */ static GstFlowReturn gst_base_sink_queue_object_unlocked (GstBaseSink * basesink, GstPad * pad, GstMiniObject * obj, gboolean prerollable) { GstFlowReturn ret = GST_FLOW_OK; gint length; GQueue *q; if (G_UNLIKELY (basesink->need_preroll)) { if (G_LIKELY (prerollable)) basesink->preroll_queued++; length = basesink->preroll_queued; GST_DEBUG_OBJECT (basesink, "now %d prerolled items", length); /* first prerollable item needs to finish the preroll */ if (length == 1) { ret = gst_base_sink_preroll_object (basesink, pad, obj); if (G_UNLIKELY (ret != GST_FLOW_OK)) goto preroll_failed; } /* need to recheck if we need preroll, commmit state during preroll * could have made us not need more preroll. */ if (G_UNLIKELY (basesink->need_preroll)) { /* see if we can render now, if we can't add the object to the preroll * queue. */ if (G_UNLIKELY (length <= basesink->preroll_queue_max_len)) goto more_preroll; } } /* we can start rendering (or blocking) the queued object * if any. */ q = basesink->preroll_queue; while (G_UNLIKELY (!g_queue_is_empty (q))) { GstMiniObject *o; o = g_queue_pop_head (q); GST_DEBUG_OBJECT (basesink, "rendering queued object %p", o); /* do something with the return value */ ret = gst_base_sink_render_object (basesink, pad, o); if (ret != GST_FLOW_OK) goto dequeue_failed; } /* now render the object */ ret = gst_base_sink_render_object (basesink, pad, obj); basesink->preroll_queued = 0; return ret; /* special cases */ preroll_failed: { GST_DEBUG_OBJECT (basesink, "preroll failed, reason %s", gst_flow_get_name (ret)); gst_mini_object_unref (obj); return ret; } more_preroll: { /* add object to the queue and return */ GST_DEBUG_OBJECT (basesink, "need more preroll data %d <= %d", length, basesink->preroll_queue_max_len); g_queue_push_tail (basesink->preroll_queue, obj); return GST_FLOW_OK; } dequeue_failed: { GST_DEBUG_OBJECT (basesink, "rendering queued objects failed, reason %s", gst_flow_get_name (ret)); gst_mini_object_unref (obj); return ret; } } /* with STREAM_LOCK * * This function grabs the PREROLL_LOCK and adds the object to * the queue. * * This function takes ownership of obj. */ static GstFlowReturn gst_base_sink_queue_object (GstBaseSink * basesink, GstPad * pad, GstMiniObject * obj, gboolean prerollable) { GstFlowReturn ret; GST_PAD_PREROLL_LOCK (pad); if (G_UNLIKELY (basesink->flushing)) goto flushing; if (G_UNLIKELY (basesink->priv->received_eos)) goto was_eos; ret = gst_base_sink_queue_object_unlocked (basesink, pad, obj, prerollable); GST_PAD_PREROLL_UNLOCK (pad); return ret; /* ERRORS */ flushing: { GST_DEBUG_OBJECT (basesink, "sink is flushing"); GST_PAD_PREROLL_UNLOCK (pad); gst_mini_object_unref (obj); return GST_FLOW_WRONG_STATE; } was_eos: { GST_DEBUG_OBJECT (basesink, "we are EOS, dropping object, return UNEXPECTED"); GST_PAD_PREROLL_UNLOCK (pad); gst_mini_object_unref (obj); return GST_FLOW_UNEXPECTED; } } static gboolean gst_base_sink_event (GstPad * pad, GstEvent * event) { GstBaseSink *basesink; gboolean result = TRUE; GstBaseSinkClass *bclass; basesink = GST_BASE_SINK (gst_pad_get_parent (pad)); bclass = GST_BASE_SINK_GET_CLASS (basesink); GST_DEBUG_OBJECT (basesink, "event %p (%s)", event, GST_EVENT_TYPE_NAME (event)); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_EOS: { GstFlowReturn ret; GST_PAD_PREROLL_LOCK (pad); if (G_UNLIKELY (basesink->flushing)) goto flushing; if (G_UNLIKELY (basesink->priv->received_eos)) { /* we can't accept anything when we are EOS */ result = FALSE; gst_event_unref (event); } else { /* we set the received EOS flag here so that we can use it when testing if * we are prerolled and to refure more buffers. */ basesink->priv->received_eos = TRUE; /* EOS is a prerollable object, we call the unlocked version because it * does not check the received_eos flag. */ ret = gst_base_sink_queue_object_unlocked (basesink, pad, GST_MINI_OBJECT_CAST (event), TRUE); if (G_UNLIKELY (ret != GST_FLOW_OK)) result = FALSE; } GST_PAD_PREROLL_UNLOCK (pad); break; } case GST_EVENT_NEWSEGMENT: { GstFlowReturn ret; GST_DEBUG_OBJECT (basesink, "newsegment %p", event); GST_PAD_PREROLL_LOCK (pad); if (G_UNLIKELY (basesink->flushing)) goto flushing; if (G_UNLIKELY (basesink->priv->received_eos)) { /* we can't accept anything when we are EOS */ result = FALSE; gst_event_unref (event); } else { /* the new segment is a non prerollable item and does not block anything, * we need to configure the current clipping segment and insert the event * in the queue to serialize it with the buffers for rendering. */ gst_base_sink_configure_segment (basesink, pad, event, basesink->abidata.ABI.clip_segment); ret = gst_base_sink_queue_object_unlocked (basesink, pad, GST_MINI_OBJECT_CAST (event), FALSE); if (G_UNLIKELY (ret != GST_FLOW_OK)) result = FALSE; else { GST_OBJECT_LOCK (basesink); basesink->have_newsegment = TRUE; GST_OBJECT_UNLOCK (basesink); } } GST_PAD_PREROLL_UNLOCK (pad); break; } case GST_EVENT_FLUSH_START: if (bclass->event) bclass->event (basesink, event); GST_DEBUG_OBJECT (basesink, "flush-start %p", event); /* 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); if (basesink->priv->async_enabled) { /* and we need to commit our state again on the next * prerolled buffer */ basesink->playing_async = TRUE; gst_element_lost_state (GST_ELEMENT_CAST (basesink)); } else { basesink->priv->have_latency = TRUE; basesink->need_preroll = FALSE; } gst_base_sink_set_last_buffer (basesink, NULL); GST_PAD_STREAM_UNLOCK (pad); gst_event_unref (event); break; case GST_EVENT_FLUSH_STOP: if (bclass->event) bclass->event (basesink, event); GST_DEBUG_OBJECT (basesink, "flush-stop %p", event); /* 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 = -1; basesink->priv->current_sstop = -1; basesink->priv->eos_rtime = -1; basesink->have_newsegment = FALSE; GST_OBJECT_UNLOCK (basesink); /* we need new segment info after the flush. */ gst_segment_init (&basesink->segment, GST_FORMAT_UNDEFINED); gst_segment_init (basesink->abidata.ABI.clip_segment, GST_FORMAT_UNDEFINED); gst_event_unref (event); break; default: /* other events are sent to queue or subclass depending on if they * are serialized. */ if (GST_EVENT_IS_SERIALIZED (event)) { gst_base_sink_queue_object (basesink, pad, GST_MINI_OBJECT_CAST (event), FALSE); } else { if (bclass->event) bclass->event (basesink, event); gst_event_unref (event); } break; } done: gst_object_unref (basesink); return result; /* ERRORS */ flushing: { GST_DEBUG_OBJECT (basesink, "we are flushing"); GST_PAD_PREROLL_UNLOCK (pad); result = FALSE; gst_event_unref (event); goto done; } } /* default implementation to calculate the start and end * timestamps on a buffer, subclasses can override */ static void gst_base_sink_get_times (GstBaseSink * basesink, GstBuffer * buffer, GstClockTime * start, GstClockTime * end) { GstClockTime timestamp, duration; timestamp = GST_BUFFER_TIMESTAMP (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 && basesink->pad_mode != GST_ACTIVATE_PULL; 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 queue the buffer for preroll and rendering. * * This function takes ownership of the buffer. */ static GstFlowReturn gst_base_sink_chain_unlocked (GstBaseSink * basesink, GstPad * pad, GstBuffer * buf) { GstBaseSinkClass *bclass; GstFlowReturn result; GstClockTime start = GST_CLOCK_TIME_NONE, end = GST_CLOCK_TIME_NONE; GstSegment *clip_segment; if (G_UNLIKELY (basesink->flushing)) goto flushing; if (G_UNLIKELY (basesink->priv->received_eos)) goto was_eos; /* for code clarity */ clip_segment = basesink->abidata.ABI.clip_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); clip_segment->start = 0; clip_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, buf, &start, &end); if (start == -1) { /* 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_get_times (basesink, 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) && (clip_segment->format == GST_FORMAT_TIME)) { if (G_UNLIKELY (!gst_segment_clip (clip_segment, GST_FORMAT_TIME, (gint64) start, (gint64) end, NULL, NULL))) goto out_of_segment; } /* now we can process the buffer in the queue, this function takes ownership * of the buffer */ result = gst_base_sink_queue_object_unlocked (basesink, pad, GST_MINI_OBJECT_CAST (buf), TRUE); return result; /* ERRORS */ flushing: { GST_DEBUG_OBJECT (basesink, "sink is flushing"); gst_buffer_unref (buf); return GST_FLOW_WRONG_STATE; } was_eos: { GST_DEBUG_OBJECT (basesink, "we are EOS, dropping object, return UNEXPECTED"); gst_buffer_unref (buf); return GST_FLOW_UNEXPECTED; } out_of_segment: { GST_DEBUG_OBJECT (basesink, "dropping buffer, out of clipping segment"); gst_buffer_unref (buf); return GST_FLOW_OK; } } /* with STREAM_LOCK */ static GstFlowReturn gst_base_sink_chain (GstPad * pad, GstBuffer * buf) { GstBaseSink *basesink; GstFlowReturn result; basesink = GST_BASE_SINK (GST_OBJECT_PARENT (pad)); if (G_UNLIKELY (basesink->pad_mode != GST_ACTIVATE_PUSH)) goto wrong_mode; GST_PAD_PREROLL_LOCK (pad); result = gst_base_sink_chain_unlocked (basesink, pad, buf); GST_PAD_PREROLL_UNLOCK (pad); 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_buffer_unref (buf); /* we don't post an error message this will signal to the peer * pushing that EOS is reached. */ result = GST_FLOW_UNEXPECTED; goto done; } } /* with STREAM_LOCK */ static void gst_base_sink_loop (GstPad * pad) { GstBaseSink *basesink; GstBuffer *buf = NULL; GstFlowReturn result; basesink = GST_BASE_SINK (GST_OBJECT_PARENT (pad)); g_assert (basesink->pad_mode == GST_ACTIVATE_PULL); GST_DEBUG_OBJECT (basesink, "pulling %" G_GUINT64_FORMAT ", %u", basesink->offset, (guint) DEFAULT_SIZE); result = gst_pad_pull_range (pad, basesink->offset, DEFAULT_SIZE, &buf); if (G_UNLIKELY (result != GST_FLOW_OK)) goto paused; if (G_UNLIKELY (buf == NULL)) goto no_buffer; basesink->offset += GST_BUFFER_SIZE (buf); GST_PAD_PREROLL_LOCK (pad); result = gst_base_sink_chain_unlocked (basesink, pad, buf); GST_PAD_PREROLL_UNLOCK (pad); 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); /* fatal errors and NOT_LINKED cause EOS */ if (GST_FLOW_IS_FATAL (result) || result == GST_FLOW_NOT_LINKED) { /* FIXME, we shouldn't post EOS when we are operating in segment mode */ gst_base_sink_event (pad, gst_event_new_eos ()); /* EOS does not cause an ERROR message */ if (result != GST_FLOW_UNEXPECTED) { GST_ELEMENT_ERROR (basesink, STREAM, FAILED, (_("Internal data stream error.")), ("stream stopped, reason %s", gst_flow_get_name (result))); } } return; } no_buffer: { GST_LOG_OBJECT (basesink, "no buffer, pausing"); 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_PAD_PREROLL_LOCK (pad); 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"); gst_base_sink_preroll_queue_flush (basesink, pad); } GST_PAD_PREROLL_UNLOCK (pad); 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); } 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) { gboolean result = FALSE; GstBaseSink *basesink; basesink = GST_BASE_SINK (gst_pad_get_parent (pad)); GST_DEBUG_OBJECT (basesink, "Trying pull mode first"); gst_base_sink_set_flushing (basesink, pad, FALSE); if (basesink->can_activate_pull && gst_pad_check_pull_range (pad) && gst_pad_activate_pull (pad, TRUE)) { GST_DEBUG_OBJECT (basesink, "Success activating pull mode"); result = TRUE; } else { GST_DEBUG_OBJECT (basesink, "Falling back to push mode"); if (gst_pad_activate_push (pad, TRUE)) { GST_DEBUG_OBJECT (basesink, "Success activating push mode"); result = TRUE; } } if (!result) { GST_WARNING_OBJECT (basesink, "Could not activate pad in either mode"); gst_base_sink_set_flushing (basesink, pad, TRUE); } gst_object_unref (basesink); return result; } static gboolean gst_base_sink_pad_activate_push (GstPad * pad, gboolean active) { gboolean result; GstBaseSink *basesink; basesink = GST_BASE_SINK (gst_pad_get_parent (pad)); if (active) { if (!basesink->can_activate_push) { result = FALSE; basesink->pad_mode = GST_ACTIVATE_NONE; } else { result = TRUE; basesink->pad_mode = GST_ACTIVATE_PUSH; } } else { if (G_UNLIKELY (basesink->pad_mode != GST_ACTIVATE_PUSH)) { g_warning ("Internal GStreamer activation error!!!"); result = FALSE; } else { gst_base_sink_set_flushing (basesink, pad, TRUE); result = TRUE; basesink->pad_mode = GST_ACTIVATE_NONE; } } gst_object_unref (basesink); 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); caps = gst_caps_make_writable (caps); /* get the first (prefered) format */ gst_caps_truncate (caps); /* try to fixate */ gst_pad_fixate_caps (GST_BASE_SINK_PAD (basesink), caps); GST_DEBUG_OBJECT (basesink, "fixated to: %" 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 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, gboolean active) { gboolean result = FALSE; GstBaseSink *basesink; GstBaseSinkClass *bclass; basesink = GST_BASE_SINK (gst_pad_get_parent (pad)); bclass = GST_BASE_SINK_GET_CLASS (basesink); if (active) { if (!basesink->can_activate_pull) { result = FALSE; basesink->pad_mode = GST_ACTIVATE_NONE; } else { GstPad *peer = gst_pad_get_peer (pad); if (G_UNLIKELY (peer == NULL)) { g_warning ("Trying to activate pad in pull mode, but no peer"); result = FALSE; basesink->pad_mode = GST_ACTIVATE_NONE; } else { if (gst_pad_activate_pull (peer, TRUE)) { /* 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_UNDEFINED); gst_segment_init (basesink->abidata.ABI.clip_segment, GST_FORMAT_UNDEFINED); GST_OBJECT_LOCK (basesink); basesink->have_newsegment = TRUE; GST_OBJECT_UNLOCK (basesink); /* 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 function checks this */ basesink->pad_mode = GST_ACTIVATE_PULL; if ((result = gst_base_sink_negotiate_pull (basesink))) { if (bclass->activate_pull) result = bclass->activate_pull (basesink, TRUE); else result = FALSE; } /* but if starting the thread fails, set it back */ if (!result) basesink->pad_mode = GST_ACTIVATE_NONE; } else { GST_DEBUG_OBJECT (pad, "Failed to activate peer in pull mode"); result = FALSE; basesink->pad_mode = GST_ACTIVATE_NONE; } gst_object_unref (peer); } } } else { if (G_UNLIKELY (basesink->pad_mode != GST_ACTIVATE_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_ACTIVATE_NONE; } } gst_object_unref (basesink); return result; } /* 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; /* only push UPSTREAM events upstream */ forward = GST_EVENT_IS_UPSTREAM (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; GST_OBJECT_UNLOCK (element); GST_DEBUG_OBJECT (basesink, "latency set to %" GST_TIME_FORMAT, GST_TIME_ARGS (latency)); /* don't forward, yet. FIXME. The latency event should likely be forwarded * to upstream element so that they can configure themselves. Each element * would subtract the amount of LATENCY it can maximally compensate for. * It's currently not very useful; even if this sink cannot compensate for * all the latency, upstream will block while this sink waits which will * trigger implicit buffering and latency there. */ forward = FALSE; break; } default: break; } if (forward) { GST_OBJECT_LOCK (element); pad = gst_object_ref (basesink->sinkpad); GST_OBJECT_UNLOCK (element); result = gst_pad_push_event (pad, event); gst_object_unref (pad); } else { /* not forwarded, unref the event */ gst_event_unref (event); } return result; } static gboolean gst_base_sink_peer_query (GstBaseSink * sink, GstQuery * query) { GstPad *peer; gboolean res = FALSE; if ((peer = gst_pad_get_peer (sink->sinkpad))) { res = gst_pad_query (peer, query); gst_object_unref (peer); } return res; } /* get the end position of the last seen object, this is used * for EOS and for making sure that we don't report a position we * have not reached yet. */ static gboolean gst_base_sink_get_position_last (GstBaseSink * basesink, gint64 * cur) { /* return last observed stream time */ *cur = basesink->priv->current_sstop; GST_DEBUG_OBJECT (basesink, "POSITION: %" GST_TIME_FORMAT, GST_TIME_ARGS (*cur)); return TRUE; } /* get the position when we are PAUSED, this is the stream time of the buffer * that prerolled. If no buffer is prerolled (we are still flushing), this * value will be -1. */ static gboolean gst_base_sink_get_position_paused (GstBaseSink * basesink, gint64 * cur) { gboolean res; gint64 time; GstSegment *segment; *cur = basesink->priv->current_sstart; segment = basesink->abidata.ABI.clip_segment; time = segment->time; if (*cur != -1) { *cur = MAX (*cur, time); GST_DEBUG_OBJECT (basesink, "POSITION as max: %" GST_TIME_FORMAT ", time %" GST_TIME_FORMAT, GST_TIME_ARGS (*cur), GST_TIME_ARGS (time)); } else { /* we have no buffer, use the segment times. */ if (segment->rate >= 0.0) { /* forward, next position is always the time of the segment */ *cur = time; GST_DEBUG_OBJECT (basesink, "POSITION as time: %" GST_TIME_FORMAT, GST_TIME_ARGS (*cur)); } else { /* reverse, next expected timestamp is segment->stop. We use the function * to get things right for negative applied_rates. */ *cur = gst_segment_to_stream_time (segment, GST_FORMAT_TIME, segment->stop); GST_DEBUG_OBJECT (basesink, "reverse POSITION: %" GST_TIME_FORMAT, GST_TIME_ARGS (*cur)); } } res = (*cur != -1); return res; } static gboolean gst_base_sink_get_position (GstBaseSink * basesink, GstFormat format, gint64 * cur) { GstClock *clock; gboolean res = FALSE; switch (format) { /* we can answer time format */ case GST_FORMAT_TIME: { GstClockTime now, base, latency; gint64 time, accum, duration; gdouble rate; gint64 last; GST_OBJECT_LOCK (basesink); /* can only give answer based on the clock if not EOS */ if (G_UNLIKELY (basesink->eos)) goto in_eos; /* we can only get the segment when we are not NULL or READY */ if (!basesink->have_newsegment) goto wrong_state; /* 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) goto in_pause; /* we need to sync on the clock. */ if (basesink->sync == FALSE) goto no_sync; /* and we need a clock */ if (G_UNLIKELY ((clock = GST_ELEMENT_CLOCK (basesink)) == NULL)) goto no_sync; /* collect all data we need holding the lock */ if (GST_CLOCK_TIME_IS_VALID (basesink->segment.time)) time = basesink->segment.time; else time = 0; if (GST_CLOCK_TIME_IS_VALID (basesink->segment.stop)) duration = basesink->segment.stop - basesink->segment.start; else duration = 0; base = GST_ELEMENT_CAST (basesink)->base_time; accum = basesink->segment.accum; rate = basesink->segment.rate * basesink->segment.applied_rate; gst_base_sink_get_position_last (basesink, &last); latency = basesink->priv->latency; gst_object_ref (clock); /* 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); now = gst_clock_get_time (clock); /* subtract base time and accumulated 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 += accum; base += latency; base = MIN (now, base); /* for negative rates we need to count back from from the segment * duration. */ if (rate < 0.0) time += duration; *cur = time + gst_guint64_to_gdouble (now - base) * rate; /* never report more than last seen position */ if (last != -1) *cur = MIN (last, *cur); gst_object_unref (clock); res = TRUE; GST_DEBUG_OBJECT (basesink, "now %" GST_TIME_FORMAT " - base %" GST_TIME_FORMAT " - accum %" GST_TIME_FORMAT " + time %" GST_TIME_FORMAT, GST_TIME_ARGS (now), GST_TIME_ARGS (base), GST_TIME_ARGS (accum), GST_TIME_ARGS (time)); break; } default: /* cannot answer other than TIME, we return FALSE, which will * send the query upstream. */ break; } done: GST_DEBUG_OBJECT (basesink, "res: %d, POSITION: %" GST_TIME_FORMAT, res, GST_TIME_ARGS (*cur)); return res; /* special cases */ in_eos: { GST_DEBUG_OBJECT (basesink, "position in EOS"); res = gst_base_sink_get_position_last (basesink, cur); GST_OBJECT_UNLOCK (basesink); goto done; } in_pause: { GST_DEBUG_OBJECT (basesink, "position in PAUSED"); res = gst_base_sink_get_position_paused (basesink, cur); GST_OBJECT_UNLOCK (basesink); goto done; } wrong_state: { /* in NULL or READY we always return 0 */ GST_DEBUG_OBJECT (basesink, "position in wrong state, return -1"); res = FALSE; *cur = -1; GST_OBJECT_UNLOCK (basesink); goto done; } no_sync: { /* report last seen timestamp if any, else return FALSE so * that upstream can answer */ if ((*cur = basesink->priv->current_sstart) != -1) res = TRUE; GST_DEBUG_OBJECT (basesink, "no sync, res %d, POSITION %" GST_TIME_FORMAT, res, GST_TIME_ARGS (*cur)); GST_OBJECT_UNLOCK (basesink); return res; } } static gboolean gst_base_sink_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; gst_query_parse_position (query, &format, NULL); GST_DEBUG_OBJECT (basesink, "position format %d", format); /* first try to get the position based on the clock */ if ((res = gst_base_sink_get_position (basesink, format, &cur))) { gst_query_set_position (query, format, cur); } else { /* fallback to peer query */ res = gst_base_sink_peer_query (basesink, query); } break; } case GST_QUERY_DURATION: GST_DEBUG_OBJECT (basesink, "duration query"); res = gst_base_sink_peer_query (basesink, query); 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: { /* FIXME, bring start/stop to stream time */ gst_query_set_segment (query, basesink->segment.rate, GST_FORMAT_TIME, basesink->segment.start, basesink->segment.stop); break; } case GST_QUERY_SEEKING: case GST_QUERY_CONVERT: case GST_QUERY_FORMATS: default: res = gst_base_sink_peer_query (basesink, query); break; } 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_PAD_PREROLL_LOCK (basesink->sinkpad); GST_DEBUG_OBJECT (basesink, "READY to PAUSED"); basesink->have_newsegment = FALSE; gst_segment_init (&basesink->segment, GST_FORMAT_UNDEFINED); gst_segment_init (basesink->abidata.ABI.clip_segment, GST_FORMAT_UNDEFINED); basesink->offset = 0; basesink->have_preroll = FALSE; basesink->need_preroll = TRUE; basesink->playing_async = TRUE; priv->current_sstart = -1; priv->current_sstop = -1; priv->eos_rtime = -1; priv->latency = 0; basesink->eos = FALSE; priv->received_eos = FALSE; gst_base_sink_reset_qos (basesink); priv->commited = 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), FALSE)); } else { priv->have_latency = TRUE; } GST_PAD_PREROLL_UNLOCK (basesink->sinkpad); break; case GST_STATE_CHANGE_PAUSED_TO_PLAYING: GST_PAD_PREROLL_LOCK (basesink->sinkpad); 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) { /* need to post EOS message here */ GST_DEBUG_OBJECT (basesink, "Now posting EOS"); gst_element_post_message (GST_ELEMENT_CAST (basesink), gst_message_new_eos (GST_OBJECT_CAST (basesink))); } else { GST_DEBUG_OBJECT (basesink, "signal preroll"); GST_PAD_PREROLL_SIGNAL (basesink->sinkpad); } } else { GST_DEBUG_OBJECT (basesink, "PAUSED to PLAYING, we are not prerolled"); basesink->need_preroll = TRUE; basesink->playing_async = 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), FALSE)); } } GST_PAD_PREROLL_UNLOCK (basesink->sinkpad); 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_READY_TO_PAUSED: /* note that this is the upward case, which doesn't follow most patterns */ if (basesink->pad_mode == GST_ACTIVATE_PULL) { GST_DEBUG_OBJECT (basesink, "basesink activated in pull mode, " "returning SUCCESS directly"); GST_PAD_PREROLL_LOCK (basesink->sinkpad); gst_element_post_message (GST_ELEMENT_CAST (basesink), gst_message_new_async_done (GST_OBJECT_CAST (basesink))); GST_PAD_PREROLL_UNLOCK (basesink->sinkpad); ret = GST_STATE_CHANGE_SUCCESS; } break; case GST_STATE_CHANGE_PLAYING_TO_PAUSED: 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_PAD_PREROLL_LOCK (basesink->sinkpad); /* 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_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) { ret = GST_STATE_CHANGE_SUCCESS; } else { GST_DEBUG_OBJECT (basesink, "PLAYING to PAUSED, we are not prerolled"); basesink->playing_async = 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), FALSE)); } } } GST_DEBUG_OBJECT (basesink, "rendered: %" G_GUINT64_FORMAT ", dropped: %" G_GUINT64_FORMAT, priv->rendered, priv->dropped); gst_base_sink_reset_qos (basesink); GST_PAD_PREROLL_UNLOCK (basesink->sinkpad); break; case GST_STATE_CHANGE_PAUSED_TO_READY: GST_PAD_PREROLL_LOCK (basesink->sinkpad); /* start by reseting 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 = -1; priv->current_sstop = -1; priv->have_latency = FALSE; GST_OBJECT_UNLOCK (basesink); gst_base_sink_set_last_buffer (basesink, NULL); 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))); } priv->commited = TRUE; } else { GST_DEBUG_OBJECT (basesink, "PAUSED to READY, don't need_preroll"); } GST_PAD_PREROLL_UNLOCK (basesink->sinkpad); 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); 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; } }