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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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2ea32f3ab5
After a buffer or GAP without duration, an EOS event should be rendered immediately instead of waiting for the end of the segment. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692868
5003 lines
151 KiB
C
5003 lines
151 KiB
C
/* GStreamer
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* Copyright (C) 2005-2007 Wim Taymans <wim.taymans@gmail.com>
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*
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* gstbasesink.c: Base class for sink elements
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:gstbasesink
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* @short_description: Base class for sink elements
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* @see_also: #GstBaseTransform, #GstBaseSrc
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*
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* #GstBaseSink is the base class for sink elements in GStreamer, such as
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* xvimagesink or filesink. It is a layer on top of #GstElement that provides a
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* simplified interface to plugin writers. #GstBaseSink handles many details
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* for you, for example: preroll, clock synchronization, state changes,
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* activation in push or pull mode, and queries.
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*
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* In most cases, when writing sink elements, there is no need to implement
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* class methods from #GstElement or to set functions on pads, because the
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* #GstBaseSink infrastructure should be sufficient.
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*
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* #GstBaseSink provides support for exactly one sink pad, which should be
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* named "sink". A sink implementation (subclass of #GstBaseSink) should
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* install a pad template in its class_init function, like so:
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* |[
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* static void
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* my_element_class_init (GstMyElementClass *klass)
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* {
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* GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
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*
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* // sinktemplate should be a #GstStaticPadTemplate with direction
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* // #GST_PAD_SINK and name "sink"
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* gst_element_class_add_pad_template (gstelement_class,
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* gst_static_pad_template_get (&sinktemplate));
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*
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* gst_element_class_set_static_metadata (gstelement_class,
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* "Sink name",
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* "Sink",
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* "My Sink element",
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* "The author <my.sink@my.email>");
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* }
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* ]|
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*
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* #GstBaseSink will handle the prerolling correctly. This means that it will
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* return #GST_STATE_CHANGE_ASYNC from a state change to PAUSED until the first
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* buffer arrives in this element. The base class will call the
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* #GstBaseSinkClass.preroll() vmethod with this preroll buffer and will then
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* commit the state change to the next asynchronously pending state.
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*
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* When the element is set to PLAYING, #GstBaseSink will synchronise on the
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* clock using the times returned from #GstBaseSinkClass.get_times(). If this
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* function returns #GST_CLOCK_TIME_NONE for the start time, no synchronisation
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* will be done. Synchronisation can be disabled entirely by setting the object
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* #GstBaseSink:sync property to %FALSE.
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*
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* After synchronisation the virtual method #GstBaseSinkClass.render() will be
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* called. Subclasses should minimally implement this method.
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*
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* Subclasses that synchronise on the clock in the #GstBaseSinkClass.render()
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* method are supported as well. These classes typically receive a buffer in
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* the render method and can then potentially block on the clock while
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* rendering. A typical example is an audiosink.
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* These subclasses can use gst_base_sink_wait_preroll() to perform the
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* blocking wait.
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*
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* Upon receiving the EOS event in the PLAYING state, #GstBaseSink will wait
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* for the clock to reach the time indicated by the stop time of the last
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* #GstBaseSinkClass.get_times() call before posting an EOS message. When the
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* element receives EOS in PAUSED, preroll completes, the event is queued and an
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* EOS message is posted when going to PLAYING.
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*
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* #GstBaseSink will internally use the #GST_EVENT_SEGMENT events to schedule
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* synchronisation and clipping of buffers. Buffers that fall completely outside
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* of the current segment are dropped. Buffers that fall partially in the
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* segment are rendered (and prerolled). Subclasses should do any subbuffer
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* clipping themselves when needed.
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*
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* #GstBaseSink will by default report the current playback position in
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* #GST_FORMAT_TIME based on the current clock time and segment information.
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* If no clock has been set on the element, the query will be forwarded
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* upstream.
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*
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* The #GstBaseSinkClass.set_caps() function will be called when the subclass
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* should configure itself to process a specific media type.
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*
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* The #GstBaseSinkClass.start() and #GstBaseSinkClass.stop() virtual methods
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* will be called when resources should be allocated. Any
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* #GstBaseSinkClass.preroll(), #GstBaseSinkClass.render() and
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* #GstBaseSinkClass.set_caps() function will be called between the
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* #GstBaseSinkClass.start() and #GstBaseSinkClass.stop() calls.
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*
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* The #GstBaseSinkClass.event() virtual method will be called when an event is
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* received by #GstBaseSink. Normally this method should only be overriden by
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* very specific elements (such as file sinks) which need to handle the
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* newsegment event specially.
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*
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* The #GstBaseSinkClass.unlock() method is called when the elements should
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* unblock any blocking operations they perform in the
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* #GstBaseSinkClass.render() method. This is mostly useful when the
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* #GstBaseSinkClass.render() method performs a blocking write on a file
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* descriptor, for example.
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*
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* The #GstBaseSink:max-lateness property affects how the sink deals with
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* buffers that arrive too late in the sink. A buffer arrives too late in the
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* sink when the presentation time (as a combination of the last segment, buffer
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* timestamp and element base_time) plus the duration is before the current
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* time of the clock.
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* If the frame is later than max-lateness, the sink will drop the buffer
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* without calling the render method.
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* This feature is disabled if sync is disabled, the
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* #GstBaseSinkClass.get_times() method does not return a valid start time or
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* max-lateness is set to -1 (the default).
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* Subclasses can use gst_base_sink_set_max_lateness() to configure the
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* max-lateness value.
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*
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* The #GstBaseSink:qos property will enable the quality-of-service features of
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* the basesink which gather statistics about the real-time performance of the
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* clock synchronisation. For each buffer received in the sink, statistics are
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* gathered and a QOS event is sent upstream with these numbers. This
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* information can then be used by upstream elements to reduce their processing
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* rate, for example.
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*
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* The #GstBaseSink:async property can be used to instruct the sink to never
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* perform an ASYNC state change. This feature is mostly usable when dealing
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* with non-synchronized streams or sparse streams.
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*
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* Last reviewed on 2007-08-29 (0.10.15)
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include <gst/gst_private.h>
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#include "gstbasesink.h"
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#include <gst/gst-i18n-lib.h>
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GST_DEBUG_CATEGORY_STATIC (gst_base_sink_debug);
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#define GST_CAT_DEFAULT gst_base_sink_debug
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#define GST_BASE_SINK_GET_PRIVATE(obj) \
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(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_BASE_SINK, GstBaseSinkPrivate))
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#define GST_FLOW_STEP GST_FLOW_CUSTOM_ERROR
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typedef struct
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{
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gboolean valid; /* if this info is valid */
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guint32 seqnum; /* the seqnum of the STEP event */
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GstFormat format; /* the format of the amount */
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guint64 amount; /* the total amount of data to skip */
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guint64 position; /* the position in the stepped data */
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guint64 duration; /* the duration in time of the skipped data */
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guint64 start; /* running_time of the start */
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gdouble rate; /* rate of skipping */
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gdouble start_rate; /* rate before skipping */
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guint64 start_start; /* start position skipping */
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guint64 start_stop; /* stop position skipping */
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gboolean flush; /* if this was a flushing step */
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gboolean intermediate; /* if this is an intermediate step */
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gboolean need_preroll; /* if we need preroll after this step */
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} GstStepInfo;
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struct _GstBaseSinkPrivate
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{
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gint qos_enabled; /* ATOMIC */
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gboolean async_enabled;
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GstClockTimeDiff ts_offset;
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GstClockTime render_delay;
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/* start, stop of current buffer, stream time, used to report position */
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GstClockTime current_sstart;
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GstClockTime current_sstop;
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/* start, stop and jitter of current buffer, running time */
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GstClockTime current_rstart;
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GstClockTime current_rstop;
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GstClockTimeDiff current_jitter;
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/* the running time of the previous buffer */
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GstClockTime prev_rstart;
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/* EOS sync time in running time */
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GstClockTime eos_rtime;
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/* last buffer that arrived in time, running time */
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GstClockTime last_render_time;
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/* when the last buffer left the sink, running time */
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GstClockTime last_left;
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/* running averages go here these are done on running time */
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GstClockTime avg_pt;
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GstClockTime avg_duration;
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gdouble avg_rate;
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GstClockTime avg_in_diff;
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/* these are done on system time. avg_jitter and avg_render are
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* compared to eachother to see if the rendering time takes a
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* huge amount of the processing, If so we are flooded with
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* buffers. */
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GstClockTime last_left_systime;
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GstClockTime avg_jitter;
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GstClockTime start, stop;
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GstClockTime avg_render;
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/* number of rendered and dropped frames */
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guint64 rendered;
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guint64 dropped;
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/* latency stuff */
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GstClockTime latency;
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/* if we already commited the state */
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gboolean commited;
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/* state change to playing ongoing */
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gboolean to_playing;
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/* when we received EOS */
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gboolean received_eos;
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/* when we are prerolled and able to report latency */
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gboolean have_latency;
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/* the last buffer we prerolled or rendered. Useful for making snapshots */
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gint enable_last_sample; /* atomic */
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GstBuffer *last_buffer;
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GstCaps *last_caps;
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/* negotiated caps */
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GstCaps *caps;
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/* blocksize for pulling */
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guint blocksize;
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gboolean discont;
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/* seqnum of the stream */
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guint32 seqnum;
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gboolean call_preroll;
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gboolean step_unlock;
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/* we have a pending and a current step operation */
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GstStepInfo current_step;
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GstStepInfo pending_step;
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/* Cached GstClockID */
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GstClockID cached_clock_id;
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/* for throttling and QoS */
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GstClockTime earliest_in_time;
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GstClockTime throttle_time;
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/* for rate control */
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guint64 max_bitrate;
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GstClockTime rc_time;
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GstClockTime rc_next;
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gsize rc_accumulated;
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};
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#define DO_RUNNING_AVG(avg,val,size) (((val) + ((size)-1) * (avg)) / (size))
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/* generic running average, this has a neutral window size */
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#define UPDATE_RUNNING_AVG(avg,val) DO_RUNNING_AVG(avg,val,8)
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/* the windows for these running averages are experimentally obtained.
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* possitive values get averaged more while negative values use a small
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* window so we can react faster to badness. */
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#define UPDATE_RUNNING_AVG_P(avg,val) DO_RUNNING_AVG(avg,val,16)
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#define UPDATE_RUNNING_AVG_N(avg,val) DO_RUNNING_AVG(avg,val,4)
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/* BaseSink properties */
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#define DEFAULT_CAN_ACTIVATE_PULL FALSE /* fixme: enable me */
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#define DEFAULT_CAN_ACTIVATE_PUSH TRUE
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#define DEFAULT_SYNC TRUE
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#define DEFAULT_MAX_LATENESS -1
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#define DEFAULT_QOS FALSE
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#define DEFAULT_ASYNC TRUE
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#define DEFAULT_TS_OFFSET 0
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#define DEFAULT_BLOCKSIZE 4096
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#define DEFAULT_RENDER_DELAY 0
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#define DEFAULT_ENABLE_LAST_SAMPLE TRUE
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#define DEFAULT_THROTTLE_TIME 0
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#define DEFAULT_MAX_BITRATE 0
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enum
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{
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PROP_0,
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PROP_SYNC,
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PROP_MAX_LATENESS,
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PROP_QOS,
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PROP_ASYNC,
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PROP_TS_OFFSET,
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PROP_ENABLE_LAST_SAMPLE,
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PROP_LAST_SAMPLE,
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PROP_BLOCKSIZE,
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PROP_RENDER_DELAY,
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PROP_THROTTLE_TIME,
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PROP_MAX_BITRATE,
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PROP_LAST
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};
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static GstElementClass *parent_class = NULL;
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static void gst_base_sink_class_init (GstBaseSinkClass * klass);
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static void gst_base_sink_init (GstBaseSink * trans, gpointer g_class);
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static void gst_base_sink_finalize (GObject * object);
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GType
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gst_base_sink_get_type (void)
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{
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static volatile gsize base_sink_type = 0;
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if (g_once_init_enter (&base_sink_type)) {
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GType _type;
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static const GTypeInfo base_sink_info = {
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sizeof (GstBaseSinkClass),
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NULL,
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NULL,
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(GClassInitFunc) gst_base_sink_class_init,
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NULL,
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NULL,
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sizeof (GstBaseSink),
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0,
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(GInstanceInitFunc) gst_base_sink_init,
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};
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_type = g_type_register_static (GST_TYPE_ELEMENT,
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"GstBaseSink", &base_sink_info, G_TYPE_FLAG_ABSTRACT);
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g_once_init_leave (&base_sink_type, _type);
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}
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return base_sink_type;
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}
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static void gst_base_sink_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_base_sink_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static gboolean gst_base_sink_send_event (GstElement * element,
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GstEvent * event);
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static gboolean default_element_query (GstElement * element, GstQuery * query);
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static GstCaps *gst_base_sink_default_get_caps (GstBaseSink * sink,
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GstCaps * caps);
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static gboolean gst_base_sink_default_set_caps (GstBaseSink * sink,
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GstCaps * caps);
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static void gst_base_sink_default_get_times (GstBaseSink * basesink,
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GstBuffer * buffer, GstClockTime * start, GstClockTime * end);
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static gboolean gst_base_sink_set_flushing (GstBaseSink * basesink,
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GstPad * pad, gboolean flushing);
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static gboolean gst_base_sink_default_activate_pull (GstBaseSink * basesink,
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gboolean active);
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static gboolean gst_base_sink_default_do_seek (GstBaseSink * sink,
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GstSegment * segment);
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static gboolean gst_base_sink_default_prepare_seek_segment (GstBaseSink * sink,
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GstEvent * event, GstSegment * segment);
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static GstStateChangeReturn gst_base_sink_change_state (GstElement * element,
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GstStateChange transition);
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static gboolean gst_base_sink_sink_query (GstPad * pad, GstObject * parent,
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GstQuery * query);
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static GstFlowReturn gst_base_sink_chain (GstPad * pad, GstObject * parent,
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GstBuffer * buffer);
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static GstFlowReturn gst_base_sink_chain_list (GstPad * pad, GstObject * parent,
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GstBufferList * list);
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static void gst_base_sink_loop (GstPad * pad);
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static gboolean gst_base_sink_pad_activate (GstPad * pad, GstObject * parent);
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static gboolean gst_base_sink_pad_activate_mode (GstPad * pad,
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GstObject * parent, GstPadMode mode, gboolean active);
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static gboolean gst_base_sink_default_event (GstBaseSink * basesink,
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GstEvent * event);
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static GstFlowReturn gst_base_sink_default_wait_event (GstBaseSink * basesink,
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GstEvent * event);
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static gboolean gst_base_sink_event (GstPad * pad, GstObject * parent,
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GstEvent * event);
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static gboolean gst_base_sink_default_query (GstBaseSink * sink,
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GstQuery * query);
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static gboolean gst_base_sink_negotiate_pull (GstBaseSink * basesink);
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static GstCaps *gst_base_sink_default_fixate (GstBaseSink * bsink,
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GstCaps * caps);
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static GstCaps *gst_base_sink_fixate (GstBaseSink * bsink, GstCaps * caps);
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/* check if an object was too late */
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static gboolean gst_base_sink_is_too_late (GstBaseSink * basesink,
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GstMiniObject * obj, GstClockTime rstart, GstClockTime rstop,
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GstClockReturn status, GstClockTimeDiff jitter);
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static void
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gst_base_sink_class_init (GstBaseSinkClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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gobject_class = G_OBJECT_CLASS (klass);
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gstelement_class = GST_ELEMENT_CLASS (klass);
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GST_DEBUG_CATEGORY_INIT (gst_base_sink_debug, "basesink", 0,
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"basesink element");
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g_type_class_add_private (klass, sizeof (GstBaseSinkPrivate));
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parent_class = g_type_class_peek_parent (klass);
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gobject_class->finalize = gst_base_sink_finalize;
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gobject_class->set_property = gst_base_sink_set_property;
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gobject_class->get_property = gst_base_sink_get_property;
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g_object_class_install_property (gobject_class, PROP_SYNC,
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g_param_spec_boolean ("sync", "Sync", "Sync on the clock", DEFAULT_SYNC,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_MAX_LATENESS,
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g_param_spec_int64 ("max-lateness", "Max Lateness",
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"Maximum number of nanoseconds that a buffer can be late before it "
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"is dropped (-1 unlimited)", -1, G_MAXINT64, DEFAULT_MAX_LATENESS,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_QOS,
|
|
g_param_spec_boolean ("qos", "Qos",
|
|
"Generate Quality-of-Service events upstream", DEFAULT_QOS,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
/**
|
|
* GstBaseSink:async:
|
|
*
|
|
* If set to #TRUE, the basesink will perform asynchronous state changes.
|
|
* When set to #FALSE, the sink will not signal the parent when it prerolls.
|
|
* Use this option when dealing with sparse streams or when synchronisation is
|
|
* not required.
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_ASYNC,
|
|
g_param_spec_boolean ("async", "Async",
|
|
"Go asynchronously to PAUSED", DEFAULT_ASYNC,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
/**
|
|
* GstBaseSink:ts-offset:
|
|
*
|
|
* Controls the final synchronisation, a negative value will render the buffer
|
|
* earlier while a positive value delays playback. This property can be
|
|
* used to fix synchronisation in bad files.
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_TS_OFFSET,
|
|
g_param_spec_int64 ("ts-offset", "TS Offset",
|
|
"Timestamp offset in nanoseconds", G_MININT64, G_MAXINT64,
|
|
DEFAULT_TS_OFFSET, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstBaseSink:enable-last-sample:
|
|
*
|
|
* Enable the last-sample property. If FALSE, basesink doesn't keep a
|
|
* reference to the last buffer arrived and the last-sample property is always
|
|
* set to NULL. This can be useful if you need buffers to be released as soon
|
|
* as possible, eg. if you're using a buffer pool.
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_ENABLE_LAST_SAMPLE,
|
|
g_param_spec_boolean ("enable-last-sample", "Enable Last Buffer",
|
|
"Enable the last-sample property", DEFAULT_ENABLE_LAST_SAMPLE,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstBaseSink:last-sample:
|
|
*
|
|
* The last buffer that arrived in the sink and was used for preroll or for
|
|
* rendering. This property can be used to generate thumbnails. This property
|
|
* can be NULL when the sink has not yet received a bufer.
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_LAST_SAMPLE,
|
|
g_param_spec_boxed ("last-sample", "Last Sample",
|
|
"The last sample received in the sink", GST_TYPE_SAMPLE,
|
|
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
|
|
/**
|
|
* GstBaseSink:blocksize:
|
|
*
|
|
* The amount of bytes to pull when operating in pull mode.
|
|
*/
|
|
/* FIXME 0.11: blocksize property should be int, otherwise min>max.. */
|
|
g_object_class_install_property (gobject_class, PROP_BLOCKSIZE,
|
|
g_param_spec_uint ("blocksize", "Block size",
|
|
"Size in bytes to pull per buffer (0 = default)", 0, G_MAXUINT,
|
|
DEFAULT_BLOCKSIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
/**
|
|
* GstBaseSink:render-delay:
|
|
*
|
|
* The additional delay between synchronisation and actual rendering of the
|
|
* media. This property will add additional latency to the device in order to
|
|
* make other sinks compensate for the delay.
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_RENDER_DELAY,
|
|
g_param_spec_uint64 ("render-delay", "Render Delay",
|
|
"Additional render delay of the sink in nanoseconds", 0, G_MAXUINT64,
|
|
DEFAULT_RENDER_DELAY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
/**
|
|
* GstBaseSink:throttle-time:
|
|
*
|
|
* The time to insert between buffers. This property can be used to control
|
|
* the maximum amount of buffers per second to render. Setting this property
|
|
* to a value bigger than 0 will make the sink create THROTTLE QoS events.
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_THROTTLE_TIME,
|
|
g_param_spec_uint64 ("throttle-time", "Throttle time",
|
|
"The time to keep between rendered buffers (0 = disabled)", 0,
|
|
G_MAXUINT64, DEFAULT_THROTTLE_TIME,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
/**
|
|
* GstBaseSink:max-bitrate:
|
|
*
|
|
* Control the maximum amount of bits that will be rendered per second.
|
|
* Setting this property to a value bigger than 0 will make the sink delay
|
|
* rendering of the buffers when it would exceed to max-bitrate.
|
|
*
|
|
* Since: 1.1.1
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_MAX_BITRATE,
|
|
g_param_spec_uint64 ("max-bitrate", "Max Bitrate",
|
|
"The maximum bits per second to render (0 = disabled)", 0,
|
|
G_MAXUINT64, DEFAULT_MAX_BITRATE,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
gstelement_class->change_state =
|
|
GST_DEBUG_FUNCPTR (gst_base_sink_change_state);
|
|
gstelement_class->send_event = GST_DEBUG_FUNCPTR (gst_base_sink_send_event);
|
|
gstelement_class->query = GST_DEBUG_FUNCPTR (default_element_query);
|
|
|
|
klass->get_caps = GST_DEBUG_FUNCPTR (gst_base_sink_default_get_caps);
|
|
klass->set_caps = GST_DEBUG_FUNCPTR (gst_base_sink_default_set_caps);
|
|
klass->fixate = GST_DEBUG_FUNCPTR (gst_base_sink_default_fixate);
|
|
klass->activate_pull =
|
|
GST_DEBUG_FUNCPTR (gst_base_sink_default_activate_pull);
|
|
klass->get_times = GST_DEBUG_FUNCPTR (gst_base_sink_default_get_times);
|
|
klass->query = GST_DEBUG_FUNCPTR (gst_base_sink_default_query);
|
|
klass->event = GST_DEBUG_FUNCPTR (gst_base_sink_default_event);
|
|
klass->wait_event = GST_DEBUG_FUNCPTR (gst_base_sink_default_wait_event);
|
|
|
|
/* Registering debug symbols for function pointers */
|
|
GST_DEBUG_REGISTER_FUNCPTR (gst_base_sink_fixate);
|
|
GST_DEBUG_REGISTER_FUNCPTR (gst_base_sink_pad_activate);
|
|
GST_DEBUG_REGISTER_FUNCPTR (gst_base_sink_pad_activate_mode);
|
|
GST_DEBUG_REGISTER_FUNCPTR (gst_base_sink_event);
|
|
GST_DEBUG_REGISTER_FUNCPTR (gst_base_sink_chain);
|
|
GST_DEBUG_REGISTER_FUNCPTR (gst_base_sink_chain_list);
|
|
GST_DEBUG_REGISTER_FUNCPTR (gst_base_sink_sink_query);
|
|
}
|
|
|
|
static GstCaps *
|
|
gst_base_sink_query_caps (GstBaseSink * bsink, GstPad * pad, GstCaps * filter)
|
|
{
|
|
GstBaseSinkClass *bclass;
|
|
GstCaps *caps = NULL;
|
|
gboolean fixed;
|
|
|
|
bclass = GST_BASE_SINK_GET_CLASS (bsink);
|
|
fixed = GST_PAD_IS_FIXED_CAPS (pad);
|
|
|
|
if (fixed || bsink->pad_mode == GST_PAD_MODE_PULL) {
|
|
/* if we are operating in pull mode or fixed caps, we only accept the
|
|
* currently negotiated caps */
|
|
caps = gst_pad_get_current_caps (pad);
|
|
}
|
|
if (caps == NULL) {
|
|
if (bclass->get_caps)
|
|
caps = bclass->get_caps (bsink, filter);
|
|
|
|
if (caps == NULL) {
|
|
GstPadTemplate *pad_template;
|
|
|
|
pad_template =
|
|
gst_element_class_get_pad_template (GST_ELEMENT_CLASS (bclass),
|
|
"sink");
|
|
if (pad_template != NULL) {
|
|
caps = gst_pad_template_get_caps (pad_template);
|
|
|
|
if (filter) {
|
|
GstCaps *intersection;
|
|
|
|
intersection =
|
|
gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
|
|
gst_caps_unref (caps);
|
|
caps = intersection;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
return caps;
|
|
}
|
|
|
|
static GstCaps *
|
|
gst_base_sink_default_fixate (GstBaseSink * bsink, GstCaps * caps)
|
|
{
|
|
GST_DEBUG_OBJECT (bsink, "using default caps fixate function");
|
|
return gst_caps_fixate (caps);
|
|
}
|
|
|
|
static GstCaps *
|
|
gst_base_sink_fixate (GstBaseSink * bsink, GstCaps * caps)
|
|
{
|
|
GstBaseSinkClass *bclass;
|
|
|
|
bclass = GST_BASE_SINK_GET_CLASS (bsink);
|
|
|
|
if (bclass->fixate)
|
|
caps = bclass->fixate (bsink, caps);
|
|
|
|
return caps;
|
|
}
|
|
|
|
static void
|
|
gst_base_sink_init (GstBaseSink * basesink, gpointer g_class)
|
|
{
|
|
GstPadTemplate *pad_template;
|
|
GstBaseSinkPrivate *priv;
|
|
|
|
basesink->priv = priv = GST_BASE_SINK_GET_PRIVATE (basesink);
|
|
|
|
pad_template =
|
|
gst_element_class_get_pad_template (GST_ELEMENT_CLASS (g_class), "sink");
|
|
g_return_if_fail (pad_template != NULL);
|
|
|
|
basesink->sinkpad = gst_pad_new_from_template (pad_template, "sink");
|
|
|
|
gst_pad_set_activate_function (basesink->sinkpad, gst_base_sink_pad_activate);
|
|
gst_pad_set_activatemode_function (basesink->sinkpad,
|
|
gst_base_sink_pad_activate_mode);
|
|
gst_pad_set_query_function (basesink->sinkpad, gst_base_sink_sink_query);
|
|
gst_pad_set_event_function (basesink->sinkpad, gst_base_sink_event);
|
|
gst_pad_set_chain_function (basesink->sinkpad, gst_base_sink_chain);
|
|
gst_pad_set_chain_list_function (basesink->sinkpad, gst_base_sink_chain_list);
|
|
gst_element_add_pad (GST_ELEMENT_CAST (basesink), basesink->sinkpad);
|
|
|
|
basesink->pad_mode = GST_PAD_MODE_NONE;
|
|
g_mutex_init (&basesink->preroll_lock);
|
|
g_cond_init (&basesink->preroll_cond);
|
|
priv->have_latency = FALSE;
|
|
|
|
basesink->can_activate_push = DEFAULT_CAN_ACTIVATE_PUSH;
|
|
basesink->can_activate_pull = DEFAULT_CAN_ACTIVATE_PULL;
|
|
|
|
basesink->sync = DEFAULT_SYNC;
|
|
basesink->max_lateness = DEFAULT_MAX_LATENESS;
|
|
g_atomic_int_set (&priv->qos_enabled, DEFAULT_QOS);
|
|
priv->async_enabled = DEFAULT_ASYNC;
|
|
priv->ts_offset = DEFAULT_TS_OFFSET;
|
|
priv->render_delay = DEFAULT_RENDER_DELAY;
|
|
priv->blocksize = DEFAULT_BLOCKSIZE;
|
|
priv->cached_clock_id = NULL;
|
|
g_atomic_int_set (&priv->enable_last_sample, DEFAULT_ENABLE_LAST_SAMPLE);
|
|
priv->throttle_time = DEFAULT_THROTTLE_TIME;
|
|
priv->max_bitrate = DEFAULT_MAX_BITRATE;
|
|
|
|
GST_OBJECT_FLAG_SET (basesink, GST_ELEMENT_FLAG_SINK);
|
|
}
|
|
|
|
static void
|
|
gst_base_sink_finalize (GObject * object)
|
|
{
|
|
GstBaseSink *basesink;
|
|
|
|
basesink = GST_BASE_SINK (object);
|
|
|
|
g_mutex_clear (&basesink->preroll_lock);
|
|
g_cond_clear (&basesink->preroll_cond);
|
|
|
|
G_OBJECT_CLASS (parent_class)->finalize (object);
|
|
}
|
|
|
|
/**
|
|
* gst_base_sink_set_sync:
|
|
* @sink: the sink
|
|
* @sync: the new sync value.
|
|
*
|
|
* Configures @sink to synchronize on the clock or not. When
|
|
* @sync is FALSE, incoming samples will be played as fast as
|
|
* possible. If @sync is TRUE, the timestamps of the incomming
|
|
* buffers will be used to schedule the exact render time of its
|
|
* contents.
|
|
*/
|
|
void
|
|
gst_base_sink_set_sync (GstBaseSink * sink, gboolean sync)
|
|
{
|
|
g_return_if_fail (GST_IS_BASE_SINK (sink));
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
sink->sync = sync;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
}
|
|
|
|
/**
|
|
* gst_base_sink_get_sync:
|
|
* @sink: the sink
|
|
*
|
|
* Checks if @sink is currently configured to synchronize against the
|
|
* clock.
|
|
*
|
|
* Returns: TRUE if the sink is configured to synchronize against the clock.
|
|
*/
|
|
gboolean
|
|
gst_base_sink_get_sync (GstBaseSink * sink)
|
|
{
|
|
gboolean res;
|
|
|
|
g_return_val_if_fail (GST_IS_BASE_SINK (sink), FALSE);
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
res = sink->sync;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_base_sink_set_max_lateness:
|
|
* @sink: the sink
|
|
* @max_lateness: the new max lateness value.
|
|
*
|
|
* Sets the new max lateness value to @max_lateness. This value is
|
|
* used to decide if a buffer should be dropped or not based on the
|
|
* buffer timestamp and the current clock time. A value of -1 means
|
|
* an unlimited time.
|
|
*/
|
|
void
|
|
gst_base_sink_set_max_lateness (GstBaseSink * sink, gint64 max_lateness)
|
|
{
|
|
g_return_if_fail (GST_IS_BASE_SINK (sink));
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
sink->max_lateness = max_lateness;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
}
|
|
|
|
/**
|
|
* gst_base_sink_get_max_lateness:
|
|
* @sink: the sink
|
|
*
|
|
* Gets the max lateness value. See gst_base_sink_set_max_lateness for
|
|
* more details.
|
|
*
|
|
* Returns: The maximum time in nanoseconds that a buffer can be late
|
|
* before it is dropped and not rendered. A value of -1 means an
|
|
* unlimited time.
|
|
*/
|
|
gint64
|
|
gst_base_sink_get_max_lateness (GstBaseSink * sink)
|
|
{
|
|
gint64 res;
|
|
|
|
g_return_val_if_fail (GST_IS_BASE_SINK (sink), -1);
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
res = sink->max_lateness;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_base_sink_set_qos_enabled:
|
|
* @sink: the sink
|
|
* @enabled: the new qos value.
|
|
*
|
|
* Configures @sink to send Quality-of-Service events upstream.
|
|
*/
|
|
void
|
|
gst_base_sink_set_qos_enabled (GstBaseSink * sink, gboolean enabled)
|
|
{
|
|
g_return_if_fail (GST_IS_BASE_SINK (sink));
|
|
|
|
g_atomic_int_set (&sink->priv->qos_enabled, enabled);
|
|
}
|
|
|
|
/**
|
|
* gst_base_sink_is_qos_enabled:
|
|
* @sink: the sink
|
|
*
|
|
* Checks if @sink is currently configured to send Quality-of-Service events
|
|
* upstream.
|
|
*
|
|
* Returns: TRUE if the sink is configured to perform Quality-of-Service.
|
|
*/
|
|
gboolean
|
|
gst_base_sink_is_qos_enabled (GstBaseSink * sink)
|
|
{
|
|
gboolean res;
|
|
|
|
g_return_val_if_fail (GST_IS_BASE_SINK (sink), FALSE);
|
|
|
|
res = g_atomic_int_get (&sink->priv->qos_enabled);
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_base_sink_set_async_enabled:
|
|
* @sink: the sink
|
|
* @enabled: the new async value.
|
|
*
|
|
* Configures @sink to perform all state changes asynchronusly. When async is
|
|
* disabled, the sink will immediately go to PAUSED instead of waiting for a
|
|
* preroll buffer. This feature is useful if the sink does not synchronize
|
|
* against the clock or when it is dealing with sparse streams.
|
|
*/
|
|
void
|
|
gst_base_sink_set_async_enabled (GstBaseSink * sink, gboolean enabled)
|
|
{
|
|
g_return_if_fail (GST_IS_BASE_SINK (sink));
|
|
|
|
GST_BASE_SINK_PREROLL_LOCK (sink);
|
|
g_atomic_int_set (&sink->priv->async_enabled, enabled);
|
|
GST_LOG_OBJECT (sink, "set async enabled to %d", enabled);
|
|
GST_BASE_SINK_PREROLL_UNLOCK (sink);
|
|
}
|
|
|
|
/**
|
|
* gst_base_sink_is_async_enabled:
|
|
* @sink: the sink
|
|
*
|
|
* Checks if @sink is currently configured to perform asynchronous state
|
|
* changes to PAUSED.
|
|
*
|
|
* Returns: TRUE if the sink is configured to perform asynchronous state
|
|
* changes.
|
|
*/
|
|
gboolean
|
|
gst_base_sink_is_async_enabled (GstBaseSink * sink)
|
|
{
|
|
gboolean res;
|
|
|
|
g_return_val_if_fail (GST_IS_BASE_SINK (sink), FALSE);
|
|
|
|
res = g_atomic_int_get (&sink->priv->async_enabled);
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_base_sink_set_ts_offset:
|
|
* @sink: the sink
|
|
* @offset: the new offset
|
|
*
|
|
* Adjust the synchronisation of @sink with @offset. A negative value will
|
|
* render buffers earlier than their timestamp. A positive value will delay
|
|
* rendering. This function can be used to fix playback of badly timestamped
|
|
* buffers.
|
|
*/
|
|
void
|
|
gst_base_sink_set_ts_offset (GstBaseSink * sink, GstClockTimeDiff offset)
|
|
{
|
|
g_return_if_fail (GST_IS_BASE_SINK (sink));
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
sink->priv->ts_offset = offset;
|
|
GST_LOG_OBJECT (sink, "set time offset to %" G_GINT64_FORMAT, offset);
|
|
GST_OBJECT_UNLOCK (sink);
|
|
}
|
|
|
|
/**
|
|
* gst_base_sink_get_ts_offset:
|
|
* @sink: the sink
|
|
*
|
|
* Get the synchronisation offset of @sink.
|
|
*
|
|
* Returns: The synchronisation offset.
|
|
*/
|
|
GstClockTimeDiff
|
|
gst_base_sink_get_ts_offset (GstBaseSink * sink)
|
|
{
|
|
GstClockTimeDiff res;
|
|
|
|
g_return_val_if_fail (GST_IS_BASE_SINK (sink), 0);
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
res = sink->priv->ts_offset;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_base_sink_get_last_sample:
|
|
* @sink: the sink
|
|
*
|
|
* Get the last sample that arrived in the sink and was used for preroll or for
|
|
* rendering. This property can be used to generate thumbnails.
|
|
*
|
|
* The #GstCaps on the sample can be used to determine the type of the buffer.
|
|
*
|
|
* Free-function: gst_sample_unref
|
|
*
|
|
* Returns: (transfer full): a #GstSample. gst_sample_unref() after usage.
|
|
* This function returns NULL when no buffer has arrived in the sink yet
|
|
* or when the sink is not in PAUSED or PLAYING.
|
|
*/
|
|
GstSample *
|
|
gst_base_sink_get_last_sample (GstBaseSink * sink)
|
|
{
|
|
GstSample *res = NULL;
|
|
|
|
g_return_val_if_fail (GST_IS_BASE_SINK (sink), NULL);
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
if (sink->priv->last_buffer) {
|
|
res = gst_sample_new (sink->priv->last_buffer,
|
|
sink->priv->last_caps, &sink->segment, NULL);
|
|
}
|
|
GST_OBJECT_UNLOCK (sink);
|
|
|
|
return res;
|
|
}
|
|
|
|
/* with OBJECT_LOCK */
|
|
static void
|
|
gst_base_sink_set_last_buffer_unlocked (GstBaseSink * sink, GstBuffer * buffer)
|
|
{
|
|
GstBuffer *old;
|
|
|
|
old = sink->priv->last_buffer;
|
|
if (G_LIKELY (old != buffer)) {
|
|
GST_DEBUG_OBJECT (sink, "setting last buffer to %p", buffer);
|
|
if (G_LIKELY (buffer))
|
|
gst_buffer_ref (buffer);
|
|
sink->priv->last_buffer = buffer;
|
|
if (buffer)
|
|
/* copy over the caps */
|
|
gst_caps_replace (&sink->priv->last_caps, sink->priv->caps);
|
|
else
|
|
gst_caps_replace (&sink->priv->last_caps, NULL);
|
|
} else {
|
|
old = NULL;
|
|
}
|
|
/* avoid unreffing with the lock because cleanup code might want to take the
|
|
* lock too */
|
|
if (G_LIKELY (old)) {
|
|
GST_OBJECT_UNLOCK (sink);
|
|
gst_buffer_unref (old);
|
|
GST_OBJECT_LOCK (sink);
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_base_sink_set_last_buffer (GstBaseSink * sink, GstBuffer * buffer)
|
|
{
|
|
if (!g_atomic_int_get (&sink->priv->enable_last_sample))
|
|
return;
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
gst_base_sink_set_last_buffer_unlocked (sink, buffer);
|
|
GST_OBJECT_UNLOCK (sink);
|
|
}
|
|
|
|
/**
|
|
* gst_base_sink_set_last_sample_enabled:
|
|
* @sink: the sink
|
|
* @enabled: the new enable-last-sample value.
|
|
*
|
|
* Configures @sink to store the last received sample in the last-sample
|
|
* property.
|
|
*/
|
|
void
|
|
gst_base_sink_set_last_sample_enabled (GstBaseSink * sink, gboolean enabled)
|
|
{
|
|
g_return_if_fail (GST_IS_BASE_SINK (sink));
|
|
|
|
/* Only take lock if we change the value */
|
|
if (g_atomic_int_compare_and_exchange (&sink->priv->enable_last_sample,
|
|
!enabled, enabled) && !enabled) {
|
|
GST_OBJECT_LOCK (sink);
|
|
gst_base_sink_set_last_buffer_unlocked (sink, NULL);
|
|
GST_OBJECT_UNLOCK (sink);
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_base_sink_is_last_sample_enabled:
|
|
* @sink: the sink
|
|
*
|
|
* Checks if @sink is currently configured to store the last received sample in
|
|
* the last-sample property.
|
|
*
|
|
* Returns: TRUE if the sink is configured to store the last received sample.
|
|
*/
|
|
gboolean
|
|
gst_base_sink_is_last_sample_enabled (GstBaseSink * sink)
|
|
{
|
|
g_return_val_if_fail (GST_IS_BASE_SINK (sink), FALSE);
|
|
|
|
return g_atomic_int_get (&sink->priv->enable_last_sample);
|
|
}
|
|
|
|
/**
|
|
* gst_base_sink_get_latency:
|
|
* @sink: the sink
|
|
*
|
|
* Get the currently configured latency.
|
|
*
|
|
* Returns: The configured latency.
|
|
*/
|
|
GstClockTime
|
|
gst_base_sink_get_latency (GstBaseSink * sink)
|
|
{
|
|
GstClockTime res;
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
res = sink->priv->latency;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_base_sink_query_latency:
|
|
* @sink: the sink
|
|
* @live: (out) (allow-none): if the sink is live
|
|
* @upstream_live: (out) (allow-none): if an upstream element is live
|
|
* @min_latency: (out) (allow-none): the min latency of the upstream elements
|
|
* @max_latency: (out) (allow-none): the max latency of the upstream elements
|
|
*
|
|
* Query the sink for the latency parameters. The latency will be queried from
|
|
* the upstream elements. @live will be TRUE if @sink is configured to
|
|
* synchronize against the clock. @upstream_live will be TRUE if an upstream
|
|
* element is live.
|
|
*
|
|
* If both @live and @upstream_live are TRUE, the sink will want to compensate
|
|
* for the latency introduced by the upstream elements by setting the
|
|
* @min_latency to a strictly possitive value.
|
|
*
|
|
* This function is mostly used by subclasses.
|
|
*
|
|
* Returns: TRUE if the query succeeded.
|
|
*/
|
|
gboolean
|
|
gst_base_sink_query_latency (GstBaseSink * sink, gboolean * live,
|
|
gboolean * upstream_live, GstClockTime * min_latency,
|
|
GstClockTime * max_latency)
|
|
{
|
|
gboolean l, us_live, res, have_latency;
|
|
GstClockTime min, max, render_delay;
|
|
GstQuery *query;
|
|
GstClockTime us_min, us_max;
|
|
|
|
/* we are live when we sync to the clock */
|
|
GST_OBJECT_LOCK (sink);
|
|
l = sink->sync;
|
|
have_latency = sink->priv->have_latency;
|
|
render_delay = sink->priv->render_delay;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
|
|
/* assume no latency */
|
|
min = 0;
|
|
max = -1;
|
|
us_live = FALSE;
|
|
|
|
if (have_latency) {
|
|
GST_DEBUG_OBJECT (sink, "we are ready for LATENCY query");
|
|
/* we are ready for a latency query this is when we preroll or when we are
|
|
* not async. */
|
|
query = gst_query_new_latency ();
|
|
|
|
/* ask the peer for the latency */
|
|
if ((res = gst_pad_peer_query (sink->sinkpad, query))) {
|
|
/* get upstream min and max latency */
|
|
gst_query_parse_latency (query, &us_live, &us_min, &us_max);
|
|
|
|
if (us_live) {
|
|
/* upstream live, use its latency, subclasses should use these
|
|
* values to create the complete latency. */
|
|
min = us_min;
|
|
max = us_max;
|
|
}
|
|
if (l) {
|
|
/* we need to add the render delay if we are live */
|
|
if (min != -1)
|
|
min += render_delay;
|
|
if (max != -1)
|
|
max += render_delay;
|
|
}
|
|
}
|
|
gst_query_unref (query);
|
|
} else {
|
|
GST_DEBUG_OBJECT (sink, "we are not yet ready for LATENCY query");
|
|
res = FALSE;
|
|
}
|
|
|
|
/* not live, we tried to do the query, if it failed we return TRUE anyway */
|
|
if (!res) {
|
|
if (!l) {
|
|
res = TRUE;
|
|
GST_DEBUG_OBJECT (sink, "latency query failed but we are not live");
|
|
} else {
|
|
GST_DEBUG_OBJECT (sink, "latency query failed and we are live");
|
|
}
|
|
}
|
|
|
|
if (res) {
|
|
GST_DEBUG_OBJECT (sink, "latency query: live: %d, have_latency %d,"
|
|
" upstream: %d, min %" GST_TIME_FORMAT ", max %" GST_TIME_FORMAT, l,
|
|
have_latency, us_live, GST_TIME_ARGS (min), GST_TIME_ARGS (max));
|
|
|
|
if (live)
|
|
*live = l;
|
|
if (upstream_live)
|
|
*upstream_live = us_live;
|
|
if (min_latency)
|
|
*min_latency = min;
|
|
if (max_latency)
|
|
*max_latency = max;
|
|
}
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_base_sink_set_render_delay:
|
|
* @sink: a #GstBaseSink
|
|
* @delay: the new delay
|
|
*
|
|
* Set the render delay in @sink to @delay. The render delay is the time
|
|
* between actual rendering of a buffer and its synchronisation time. Some
|
|
* devices might delay media rendering which can be compensated for with this
|
|
* function.
|
|
*
|
|
* After calling this function, this sink will report additional latency and
|
|
* other sinks will adjust their latency to delay the rendering of their media.
|
|
*
|
|
* This function is usually called by subclasses.
|
|
*/
|
|
void
|
|
gst_base_sink_set_render_delay (GstBaseSink * sink, GstClockTime delay)
|
|
{
|
|
GstClockTime old_render_delay;
|
|
|
|
g_return_if_fail (GST_IS_BASE_SINK (sink));
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
old_render_delay = sink->priv->render_delay;
|
|
sink->priv->render_delay = delay;
|
|
GST_LOG_OBJECT (sink, "set render delay to %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (delay));
|
|
GST_OBJECT_UNLOCK (sink);
|
|
|
|
if (delay != old_render_delay) {
|
|
GST_DEBUG_OBJECT (sink, "posting latency changed");
|
|
gst_element_post_message (GST_ELEMENT_CAST (sink),
|
|
gst_message_new_latency (GST_OBJECT_CAST (sink)));
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_base_sink_get_render_delay:
|
|
* @sink: a #GstBaseSink
|
|
*
|
|
* Get the render delay of @sink. see gst_base_sink_set_render_delay() for more
|
|
* information about the render delay.
|
|
*
|
|
* Returns: the render delay of @sink.
|
|
*/
|
|
GstClockTime
|
|
gst_base_sink_get_render_delay (GstBaseSink * sink)
|
|
{
|
|
GstClockTimeDiff res;
|
|
|
|
g_return_val_if_fail (GST_IS_BASE_SINK (sink), 0);
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
res = sink->priv->render_delay;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_base_sink_set_blocksize:
|
|
* @sink: a #GstBaseSink
|
|
* @blocksize: the blocksize in bytes
|
|
*
|
|
* Set the number of bytes that the sink will pull when it is operating in pull
|
|
* mode.
|
|
*/
|
|
/* FIXME 0.11: blocksize property should be int, otherwise min>max.. */
|
|
void
|
|
gst_base_sink_set_blocksize (GstBaseSink * sink, guint blocksize)
|
|
{
|
|
g_return_if_fail (GST_IS_BASE_SINK (sink));
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
sink->priv->blocksize = blocksize;
|
|
GST_LOG_OBJECT (sink, "set blocksize to %u", blocksize);
|
|
GST_OBJECT_UNLOCK (sink);
|
|
}
|
|
|
|
/**
|
|
* gst_base_sink_get_blocksize:
|
|
* @sink: a #GstBaseSink
|
|
*
|
|
* Get the number of bytes that the sink will pull when it is operating in pull
|
|
* mode.
|
|
*
|
|
* Returns: the number of bytes @sink will pull in pull mode.
|
|
*/
|
|
/* FIXME 0.11: blocksize property should be int, otherwise min>max.. */
|
|
guint
|
|
gst_base_sink_get_blocksize (GstBaseSink * sink)
|
|
{
|
|
guint res;
|
|
|
|
g_return_val_if_fail (GST_IS_BASE_SINK (sink), 0);
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
res = sink->priv->blocksize;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_base_sink_set_throttle_time:
|
|
* @sink: a #GstBaseSink
|
|
* @throttle: the throttle time in nanoseconds
|
|
*
|
|
* Set the time that will be inserted between rendered buffers. This
|
|
* can be used to control the maximum buffers per second that the sink
|
|
* will render.
|
|
*/
|
|
void
|
|
gst_base_sink_set_throttle_time (GstBaseSink * sink, guint64 throttle)
|
|
{
|
|
g_return_if_fail (GST_IS_BASE_SINK (sink));
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
sink->priv->throttle_time = throttle;
|
|
GST_LOG_OBJECT (sink, "set throttle_time to %" G_GUINT64_FORMAT, throttle);
|
|
GST_OBJECT_UNLOCK (sink);
|
|
}
|
|
|
|
/**
|
|
* gst_base_sink_get_throttle_time:
|
|
* @sink: a #GstBaseSink
|
|
*
|
|
* Get the time that will be inserted between frames to control the
|
|
* maximum buffers per second.
|
|
*
|
|
* Returns: the number of nanoseconds @sink will put between frames.
|
|
*/
|
|
guint64
|
|
gst_base_sink_get_throttle_time (GstBaseSink * sink)
|
|
{
|
|
guint64 res;
|
|
|
|
g_return_val_if_fail (GST_IS_BASE_SINK (sink), 0);
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
res = sink->priv->throttle_time;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_base_sink_set_max_bitrate:
|
|
* @sink: a #GstBaseSink
|
|
* @max_bitrate: the max_bitrate in bits per second
|
|
*
|
|
* Set the maximum amount of bits per second that the sink will render.
|
|
*
|
|
* Since: 1.1.1
|
|
*/
|
|
void
|
|
gst_base_sink_set_max_bitrate (GstBaseSink * sink, guint64 max_bitrate)
|
|
{
|
|
g_return_if_fail (GST_IS_BASE_SINK (sink));
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
sink->priv->max_bitrate = max_bitrate;
|
|
GST_LOG_OBJECT (sink, "set max_bitrate to %" G_GUINT64_FORMAT, max_bitrate);
|
|
GST_OBJECT_UNLOCK (sink);
|
|
}
|
|
|
|
/**
|
|
* gst_base_sink_get_max_bitrate:
|
|
* @sink: a #GstBaseSink
|
|
*
|
|
* Get the maximum amount of bits per second that the sink will render.
|
|
*
|
|
* Returns: the maximum number of bits per second @sink will render.
|
|
*
|
|
* Since: 1.1.1
|
|
*/
|
|
guint64
|
|
gst_base_sink_get_max_bitrate (GstBaseSink * sink)
|
|
{
|
|
guint64 res;
|
|
|
|
g_return_val_if_fail (GST_IS_BASE_SINK (sink), 0);
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
res = sink->priv->max_bitrate;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
|
|
return res;
|
|
}
|
|
|
|
static void
|
|
gst_base_sink_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstBaseSink *sink = GST_BASE_SINK (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_SYNC:
|
|
gst_base_sink_set_sync (sink, g_value_get_boolean (value));
|
|
break;
|
|
case PROP_MAX_LATENESS:
|
|
gst_base_sink_set_max_lateness (sink, g_value_get_int64 (value));
|
|
break;
|
|
case PROP_QOS:
|
|
gst_base_sink_set_qos_enabled (sink, g_value_get_boolean (value));
|
|
break;
|
|
case PROP_ASYNC:
|
|
gst_base_sink_set_async_enabled (sink, g_value_get_boolean (value));
|
|
break;
|
|
case PROP_TS_OFFSET:
|
|
gst_base_sink_set_ts_offset (sink, g_value_get_int64 (value));
|
|
break;
|
|
case PROP_BLOCKSIZE:
|
|
gst_base_sink_set_blocksize (sink, g_value_get_uint (value));
|
|
break;
|
|
case PROP_RENDER_DELAY:
|
|
gst_base_sink_set_render_delay (sink, g_value_get_uint64 (value));
|
|
break;
|
|
case PROP_ENABLE_LAST_SAMPLE:
|
|
gst_base_sink_set_last_sample_enabled (sink, g_value_get_boolean (value));
|
|
break;
|
|
case PROP_THROTTLE_TIME:
|
|
gst_base_sink_set_throttle_time (sink, g_value_get_uint64 (value));
|
|
break;
|
|
case PROP_MAX_BITRATE:
|
|
gst_base_sink_set_max_bitrate (sink, g_value_get_uint64 (value));
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_base_sink_get_property (GObject * object, guint prop_id, GValue * value,
|
|
GParamSpec * pspec)
|
|
{
|
|
GstBaseSink *sink = GST_BASE_SINK (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_SYNC:
|
|
g_value_set_boolean (value, gst_base_sink_get_sync (sink));
|
|
break;
|
|
case PROP_MAX_LATENESS:
|
|
g_value_set_int64 (value, gst_base_sink_get_max_lateness (sink));
|
|
break;
|
|
case PROP_QOS:
|
|
g_value_set_boolean (value, gst_base_sink_is_qos_enabled (sink));
|
|
break;
|
|
case PROP_ASYNC:
|
|
g_value_set_boolean (value, gst_base_sink_is_async_enabled (sink));
|
|
break;
|
|
case PROP_TS_OFFSET:
|
|
g_value_set_int64 (value, gst_base_sink_get_ts_offset (sink));
|
|
break;
|
|
case PROP_LAST_SAMPLE:
|
|
gst_value_take_buffer (value, gst_base_sink_get_last_sample (sink));
|
|
break;
|
|
case PROP_ENABLE_LAST_SAMPLE:
|
|
g_value_set_boolean (value, gst_base_sink_is_last_sample_enabled (sink));
|
|
break;
|
|
case PROP_BLOCKSIZE:
|
|
g_value_set_uint (value, gst_base_sink_get_blocksize (sink));
|
|
break;
|
|
case PROP_RENDER_DELAY:
|
|
g_value_set_uint64 (value, gst_base_sink_get_render_delay (sink));
|
|
break;
|
|
case PROP_THROTTLE_TIME:
|
|
g_value_set_uint64 (value, gst_base_sink_get_throttle_time (sink));
|
|
break;
|
|
case PROP_MAX_BITRATE:
|
|
g_value_set_uint64 (value, gst_base_sink_get_max_bitrate (sink));
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
|
|
static GstCaps *
|
|
gst_base_sink_default_get_caps (GstBaseSink * sink, GstCaps * filter)
|
|
{
|
|
return NULL;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_sink_default_set_caps (GstBaseSink * sink, GstCaps * caps)
|
|
{
|
|
return TRUE;
|
|
}
|
|
|
|
/* with PREROLL_LOCK, STREAM_LOCK */
|
|
static gboolean
|
|
gst_base_sink_commit_state (GstBaseSink * basesink)
|
|
{
|
|
/* commit state and proceed to next pending state */
|
|
GstState current, next, pending, post_pending;
|
|
gboolean post_paused = FALSE;
|
|
gboolean post_async_done = FALSE;
|
|
gboolean post_playing = FALSE;
|
|
|
|
/* we are certainly not playing async anymore now */
|
|
basesink->playing_async = FALSE;
|
|
|
|
GST_OBJECT_LOCK (basesink);
|
|
current = GST_STATE (basesink);
|
|
next = GST_STATE_NEXT (basesink);
|
|
pending = GST_STATE_PENDING (basesink);
|
|
post_pending = pending;
|
|
|
|
switch (pending) {
|
|
case GST_STATE_PLAYING:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "commiting state to PLAYING");
|
|
|
|
basesink->need_preroll = FALSE;
|
|
post_async_done = TRUE;
|
|
basesink->priv->commited = TRUE;
|
|
post_playing = TRUE;
|
|
/* post PAUSED too when we were READY */
|
|
if (current == GST_STATE_READY) {
|
|
post_paused = TRUE;
|
|
}
|
|
break;
|
|
}
|
|
case GST_STATE_PAUSED:
|
|
GST_DEBUG_OBJECT (basesink, "commiting state to PAUSED");
|
|
post_paused = TRUE;
|
|
post_async_done = TRUE;
|
|
basesink->priv->commited = TRUE;
|
|
post_pending = GST_STATE_VOID_PENDING;
|
|
break;
|
|
case GST_STATE_READY:
|
|
case GST_STATE_NULL:
|
|
goto stopping;
|
|
case GST_STATE_VOID_PENDING:
|
|
goto nothing_pending;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
/* we can report latency queries now */
|
|
basesink->priv->have_latency = TRUE;
|
|
|
|
GST_STATE (basesink) = pending;
|
|
GST_STATE_NEXT (basesink) = GST_STATE_VOID_PENDING;
|
|
GST_STATE_PENDING (basesink) = GST_STATE_VOID_PENDING;
|
|
GST_STATE_RETURN (basesink) = GST_STATE_CHANGE_SUCCESS;
|
|
GST_OBJECT_UNLOCK (basesink);
|
|
|
|
if (post_paused) {
|
|
GST_DEBUG_OBJECT (basesink, "posting PAUSED state change message");
|
|
gst_element_post_message (GST_ELEMENT_CAST (basesink),
|
|
gst_message_new_state_changed (GST_OBJECT_CAST (basesink),
|
|
current, next, post_pending));
|
|
}
|
|
if (post_async_done) {
|
|
GST_DEBUG_OBJECT (basesink, "posting async-done message");
|
|
gst_element_post_message (GST_ELEMENT_CAST (basesink),
|
|
gst_message_new_async_done (GST_OBJECT_CAST (basesink),
|
|
GST_CLOCK_TIME_NONE));
|
|
}
|
|
if (post_playing) {
|
|
GST_DEBUG_OBJECT (basesink, "posting PLAYING state change message");
|
|
gst_element_post_message (GST_ELEMENT_CAST (basesink),
|
|
gst_message_new_state_changed (GST_OBJECT_CAST (basesink),
|
|
next, pending, GST_STATE_VOID_PENDING));
|
|
}
|
|
|
|
GST_STATE_BROADCAST (basesink);
|
|
|
|
return TRUE;
|
|
|
|
nothing_pending:
|
|
{
|
|
/* Depending on the state, set our vars. We get in this situation when the
|
|
* state change function got a change to update the state vars before the
|
|
* streaming thread did. This is fine but we need to make sure that we
|
|
* update the need_preroll var since it was TRUE when we got here and might
|
|
* become FALSE if we got to PLAYING. */
|
|
GST_DEBUG_OBJECT (basesink, "nothing to commit, now in %s",
|
|
gst_element_state_get_name (current));
|
|
switch (current) {
|
|
case GST_STATE_PLAYING:
|
|
basesink->need_preroll = FALSE;
|
|
break;
|
|
case GST_STATE_PAUSED:
|
|
basesink->need_preroll = TRUE;
|
|
break;
|
|
default:
|
|
basesink->need_preroll = FALSE;
|
|
basesink->flushing = TRUE;
|
|
break;
|
|
}
|
|
/* we can report latency queries now */
|
|
basesink->priv->have_latency = TRUE;
|
|
GST_OBJECT_UNLOCK (basesink);
|
|
return TRUE;
|
|
}
|
|
stopping:
|
|
{
|
|
/* app is going to READY */
|
|
GST_DEBUG_OBJECT (basesink, "stopping");
|
|
basesink->need_preroll = FALSE;
|
|
basesink->flushing = TRUE;
|
|
GST_OBJECT_UNLOCK (basesink);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static void
|
|
start_stepping (GstBaseSink * sink, GstSegment * segment,
|
|
GstStepInfo * pending, GstStepInfo * current)
|
|
{
|
|
gint64 end;
|
|
GstMessage *message;
|
|
|
|
GST_DEBUG_OBJECT (sink, "update pending step");
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
memcpy (current, pending, sizeof (GstStepInfo));
|
|
pending->valid = FALSE;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
|
|
/* post message first */
|
|
message =
|
|
gst_message_new_step_start (GST_OBJECT (sink), TRUE, current->format,
|
|
current->amount, current->rate, current->flush, current->intermediate);
|
|
gst_message_set_seqnum (message, current->seqnum);
|
|
gst_element_post_message (GST_ELEMENT (sink), message);
|
|
|
|
/* get the running time of where we paused and remember it */
|
|
current->start = gst_element_get_start_time (GST_ELEMENT_CAST (sink));
|
|
gst_segment_set_running_time (segment, GST_FORMAT_TIME, current->start);
|
|
|
|
/* set the new rate for the remainder of the segment */
|
|
current->start_rate = segment->rate;
|
|
segment->rate *= current->rate;
|
|
|
|
/* save values */
|
|
if (segment->rate > 0.0)
|
|
current->start_stop = segment->stop;
|
|
else
|
|
current->start_start = segment->start;
|
|
|
|
if (current->format == GST_FORMAT_TIME) {
|
|
/* calculate the running-time when the step operation should stop */
|
|
if (current->amount != -1)
|
|
end = current->start + current->amount;
|
|
else
|
|
end = -1;
|
|
|
|
if (!current->flush) {
|
|
gint64 position;
|
|
|
|
/* update the segment clipping regions for non-flushing seeks */
|
|
if (segment->rate > 0.0) {
|
|
if (end != -1)
|
|
position = gst_segment_to_position (segment, GST_FORMAT_TIME, end);
|
|
else
|
|
position = segment->stop;
|
|
|
|
segment->stop = position;
|
|
segment->position = position;
|
|
} else {
|
|
if (end != -1)
|
|
position = gst_segment_to_position (segment, GST_FORMAT_TIME, end);
|
|
else
|
|
position = segment->start;
|
|
|
|
segment->time = position;
|
|
segment->start = position;
|
|
segment->position = position;
|
|
}
|
|
}
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (sink, "segment now %" GST_SEGMENT_FORMAT, segment);
|
|
GST_DEBUG_OBJECT (sink, "step started at running_time %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (current->start));
|
|
|
|
GST_DEBUG_OBJECT (sink, "step amount: %" G_GUINT64_FORMAT ", format: %s, "
|
|
"rate: %f", current->amount, gst_format_get_name (current->format),
|
|
current->rate);
|
|
}
|
|
|
|
static void
|
|
stop_stepping (GstBaseSink * sink, GstSegment * segment,
|
|
GstStepInfo * current, gint64 rstart, gint64 rstop, gboolean eos)
|
|
{
|
|
gint64 stop, position;
|
|
GstMessage *message;
|
|
|
|
GST_DEBUG_OBJECT (sink, "step complete");
|
|
|
|
if (segment->rate > 0.0)
|
|
stop = rstart;
|
|
else
|
|
stop = rstop;
|
|
|
|
GST_DEBUG_OBJECT (sink,
|
|
"step stop at running_time %" GST_TIME_FORMAT, GST_TIME_ARGS (stop));
|
|
|
|
if (stop == -1)
|
|
current->duration = current->position;
|
|
else
|
|
current->duration = stop - current->start;
|
|
|
|
GST_DEBUG_OBJECT (sink, "step elapsed running_time %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (current->duration));
|
|
|
|
position = current->start + current->duration;
|
|
|
|
/* now move the segment to the new running time */
|
|
gst_segment_set_running_time (segment, GST_FORMAT_TIME, position);
|
|
|
|
if (current->flush) {
|
|
/* and remove the time we flushed, start time did not change */
|
|
segment->base = current->start;
|
|
} else {
|
|
/* start time is now the stepped position */
|
|
gst_element_set_start_time (GST_ELEMENT_CAST (sink), position);
|
|
}
|
|
|
|
/* restore the previous rate */
|
|
segment->rate = current->start_rate;
|
|
|
|
if (segment->rate > 0.0)
|
|
segment->stop = current->start_stop;
|
|
else
|
|
segment->start = current->start_start;
|
|
|
|
/* post the step done when we know the stepped duration in TIME */
|
|
message =
|
|
gst_message_new_step_done (GST_OBJECT_CAST (sink), current->format,
|
|
current->amount, current->rate, current->flush, current->intermediate,
|
|
current->duration, eos);
|
|
gst_message_set_seqnum (message, current->seqnum);
|
|
gst_element_post_message (GST_ELEMENT_CAST (sink), message);
|
|
|
|
if (!current->intermediate)
|
|
sink->need_preroll = current->need_preroll;
|
|
|
|
/* and the current step info finished and becomes invalid */
|
|
current->valid = FALSE;
|
|
}
|
|
|
|
static gboolean
|
|
handle_stepping (GstBaseSink * sink, GstSegment * segment,
|
|
GstStepInfo * current, guint64 * cstart, guint64 * cstop, guint64 * rstart,
|
|
guint64 * rstop)
|
|
{
|
|
gboolean step_end = FALSE;
|
|
|
|
/* stepping never stops */
|
|
if (current->amount == -1)
|
|
return FALSE;
|
|
|
|
/* see if we need to skip this buffer because of stepping */
|
|
switch (current->format) {
|
|
case GST_FORMAT_TIME:
|
|
{
|
|
guint64 end;
|
|
guint64 first, last;
|
|
gdouble abs_rate;
|
|
|
|
if (segment->rate > 0.0) {
|
|
if (segment->stop == *cstop)
|
|
*rstop = *rstart + current->amount;
|
|
|
|
first = *rstart;
|
|
last = *rstop;
|
|
} else {
|
|
if (segment->start == *cstart)
|
|
*rstart = *rstop + current->amount;
|
|
|
|
first = *rstop;
|
|
last = *rstart;
|
|
}
|
|
|
|
end = current->start + current->amount;
|
|
current->position = first - current->start;
|
|
|
|
abs_rate = ABS (segment->rate);
|
|
if (G_UNLIKELY (abs_rate != 1.0))
|
|
current->position /= abs_rate;
|
|
|
|
GST_DEBUG_OBJECT (sink,
|
|
"buffer: %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (first), GST_TIME_ARGS (last));
|
|
GST_DEBUG_OBJECT (sink,
|
|
"got time step %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT "/%"
|
|
GST_TIME_FORMAT, GST_TIME_ARGS (current->position),
|
|
GST_TIME_ARGS (last - current->start),
|
|
GST_TIME_ARGS (current->amount));
|
|
|
|
if ((current->flush && current->position >= current->amount)
|
|
|| last >= end) {
|
|
GST_DEBUG_OBJECT (sink, "step ended, we need clipping");
|
|
step_end = TRUE;
|
|
if (segment->rate > 0.0) {
|
|
*rstart = end;
|
|
*cstart = gst_segment_to_position (segment, GST_FORMAT_TIME, end);
|
|
} else {
|
|
*rstop = end;
|
|
*cstop = gst_segment_to_position (segment, GST_FORMAT_TIME, end);
|
|
}
|
|
}
|
|
GST_DEBUG_OBJECT (sink,
|
|
"cstart %" GST_TIME_FORMAT ", rstart %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (*cstart), GST_TIME_ARGS (*rstart));
|
|
GST_DEBUG_OBJECT (sink,
|
|
"cstop %" GST_TIME_FORMAT ", rstop %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (*cstop), GST_TIME_ARGS (*rstop));
|
|
break;
|
|
}
|
|
case GST_FORMAT_BUFFERS:
|
|
GST_DEBUG_OBJECT (sink,
|
|
"got default step %" G_GUINT64_FORMAT "/%" G_GUINT64_FORMAT,
|
|
current->position, current->amount);
|
|
|
|
if (current->position < current->amount) {
|
|
current->position++;
|
|
} else {
|
|
step_end = TRUE;
|
|
}
|
|
break;
|
|
case GST_FORMAT_DEFAULT:
|
|
default:
|
|
GST_DEBUG_OBJECT (sink,
|
|
"got unknown step %" G_GUINT64_FORMAT "/%" G_GUINT64_FORMAT,
|
|
current->position, current->amount);
|
|
break;
|
|
}
|
|
return step_end;
|
|
}
|
|
|
|
/* with STREAM_LOCK, PREROLL_LOCK
|
|
*
|
|
* Returns TRUE if the object needs synchronisation and takes therefore
|
|
* part in prerolling.
|
|
*
|
|
* rsstart/rsstop contain the start/stop in stream time.
|
|
* rrstart/rrstop contain the start/stop in running time.
|
|
*/
|
|
static gboolean
|
|
gst_base_sink_get_sync_times (GstBaseSink * basesink, GstMiniObject * obj,
|
|
GstClockTime * rsstart, GstClockTime * rsstop,
|
|
GstClockTime * rrstart, GstClockTime * rrstop, GstClockTime * rrnext,
|
|
gboolean * do_sync, gboolean * stepped, GstStepInfo * step,
|
|
gboolean * step_end)
|
|
{
|
|
GstBaseSinkClass *bclass;
|
|
GstClockTime start, stop; /* raw start/stop timestamps */
|
|
guint64 cstart, cstop; /* clipped raw timestamps */
|
|
guint64 rstart, rstop, rnext; /* clipped timestamps converted to running time */
|
|
GstClockTime sstart, sstop; /* clipped timestamps converted to stream time */
|
|
GstFormat format;
|
|
GstBaseSinkPrivate *priv;
|
|
GstSegment *segment;
|
|
gboolean eos;
|
|
|
|
priv = basesink->priv;
|
|
segment = &basesink->segment;
|
|
|
|
bclass = GST_BASE_SINK_GET_CLASS (basesink);
|
|
|
|
again:
|
|
/* start with nothing */
|
|
start = stop = GST_CLOCK_TIME_NONE;
|
|
eos = FALSE;
|
|
|
|
if (G_UNLIKELY (GST_IS_EVENT (obj))) {
|
|
GstEvent *event = GST_EVENT_CAST (obj);
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
/* EOS event needs syncing */
|
|
case GST_EVENT_EOS:
|
|
{
|
|
if (segment->rate >= 0.0) {
|
|
sstart = sstop = priv->current_sstop;
|
|
if (!GST_CLOCK_TIME_IS_VALID (sstart)) {
|
|
/* we have not seen a buffer yet, use the segment values */
|
|
sstart = sstop = gst_segment_to_stream_time (segment,
|
|
segment->format, segment->stop);
|
|
}
|
|
} else {
|
|
sstart = sstop = priv->current_sstart;
|
|
if (!GST_CLOCK_TIME_IS_VALID (sstart)) {
|
|
/* we have not seen a buffer yet, use the segment values */
|
|
sstart = sstop = gst_segment_to_stream_time (segment,
|
|
segment->format, segment->start);
|
|
}
|
|
}
|
|
|
|
rstart = rstop = rnext = priv->eos_rtime;
|
|
*do_sync = rstart != -1;
|
|
GST_DEBUG_OBJECT (basesink, "sync times for EOS %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (rstart));
|
|
/* if we are stepping, we end now */
|
|
*step_end = step->valid;
|
|
eos = TRUE;
|
|
goto eos_done;
|
|
}
|
|
case GST_EVENT_GAP:
|
|
{
|
|
GstClockTime timestamp, duration;
|
|
gst_event_parse_gap (event, ×tamp, &duration);
|
|
|
|
GST_DEBUG_OBJECT (basesink, "Got Gap time %" GST_TIME_FORMAT
|
|
" duration %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (timestamp), GST_TIME_ARGS (duration));
|
|
|
|
if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
|
|
start = timestamp;
|
|
if (GST_CLOCK_TIME_IS_VALID (duration))
|
|
stop = start + duration;
|
|
}
|
|
*do_sync = TRUE;
|
|
break;
|
|
}
|
|
default:
|
|
/* other events do not need syncing */
|
|
return FALSE;
|
|
}
|
|
} else {
|
|
/* else do buffer sync code */
|
|
GstBuffer *buffer = GST_BUFFER_CAST (obj);
|
|
|
|
/* just get the times to see if we need syncing, if the start returns -1 we
|
|
* don't sync. */
|
|
if (bclass->get_times)
|
|
bclass->get_times (basesink, buffer, &start, &stop);
|
|
|
|
if (!GST_CLOCK_TIME_IS_VALID (start)) {
|
|
/* we don't need to sync but we still want to get the timestamps for
|
|
* tracking the position */
|
|
gst_base_sink_default_get_times (basesink, buffer, &start, &stop);
|
|
*do_sync = FALSE;
|
|
} else {
|
|
*do_sync = TRUE;
|
|
}
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (basesink, "got times start: %" GST_TIME_FORMAT
|
|
", stop: %" GST_TIME_FORMAT ", do_sync %d", GST_TIME_ARGS (start),
|
|
GST_TIME_ARGS (stop), *do_sync);
|
|
|
|
/* collect segment and format for code clarity */
|
|
format = segment->format;
|
|
|
|
/* clip */
|
|
if (G_UNLIKELY (!gst_segment_clip (segment, format,
|
|
start, stop, &cstart, &cstop))) {
|
|
if (step->valid) {
|
|
GST_DEBUG_OBJECT (basesink, "step out of segment");
|
|
/* when we are stepping, pretend we're at the end of the segment */
|
|
if (segment->rate > 0.0) {
|
|
cstart = segment->stop;
|
|
cstop = segment->stop;
|
|
} else {
|
|
cstart = segment->start;
|
|
cstop = segment->start;
|
|
}
|
|
goto do_times;
|
|
}
|
|
goto out_of_segment;
|
|
}
|
|
|
|
if (G_UNLIKELY (start != cstart || stop != cstop)) {
|
|
GST_DEBUG_OBJECT (basesink, "clipped to: start %" GST_TIME_FORMAT
|
|
", stop: %" GST_TIME_FORMAT, GST_TIME_ARGS (cstart),
|
|
GST_TIME_ARGS (cstop));
|
|
}
|
|
|
|
/* set last stop position */
|
|
if (G_LIKELY (stop != GST_CLOCK_TIME_NONE && cstop != GST_CLOCK_TIME_NONE))
|
|
segment->position = cstop;
|
|
else
|
|
segment->position = cstart;
|
|
|
|
do_times:
|
|
rstart = gst_segment_to_running_time (segment, format, cstart);
|
|
rstop = gst_segment_to_running_time (segment, format, cstop);
|
|
|
|
if (GST_CLOCK_TIME_IS_VALID (stop))
|
|
rnext = rstop;
|
|
else
|
|
rnext = rstart;
|
|
|
|
if (G_UNLIKELY (step->valid)) {
|
|
if (!(*step_end = handle_stepping (basesink, segment, step, &cstart, &cstop,
|
|
&rstart, &rstop))) {
|
|
/* step is still busy, we discard data when we are flushing */
|
|
*stepped = step->flush;
|
|
GST_DEBUG_OBJECT (basesink, "stepping busy");
|
|
}
|
|
}
|
|
/* this can produce wrong values if we accumulated non-TIME segments. If this happens,
|
|
* upstream is behaving very badly */
|
|
sstart = gst_segment_to_stream_time (segment, format, cstart);
|
|
sstop = gst_segment_to_stream_time (segment, format, cstop);
|
|
|
|
eos_done:
|
|
/* eos_done label only called when doing EOS, we also stop stepping then */
|
|
if (*step_end && step->flush) {
|
|
GST_DEBUG_OBJECT (basesink, "flushing step ended");
|
|
stop_stepping (basesink, segment, step, rstart, rstop, eos);
|
|
*step_end = FALSE;
|
|
/* re-determine running start times for adjusted segment
|
|
* (which has a flushed amount of running/accumulated time removed) */
|
|
if (!GST_IS_EVENT (obj)) {
|
|
GST_DEBUG_OBJECT (basesink, "refresh sync times");
|
|
goto again;
|
|
}
|
|
}
|
|
|
|
/* save times */
|
|
*rsstart = sstart;
|
|
*rsstop = sstop;
|
|
*rrstart = rstart;
|
|
*rrstop = rstop;
|
|
*rrnext = rnext;
|
|
|
|
/* buffers and EOS always need syncing and preroll */
|
|
return TRUE;
|
|
|
|
/* special cases */
|
|
out_of_segment:
|
|
{
|
|
/* we usually clip in the chain function already but stepping could cause
|
|
* the segment to be updated later. we return FALSE so that we don't try
|
|
* to sync on it. */
|
|
GST_LOG_OBJECT (basesink, "buffer skipped, not in segment");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/* with STREAM_LOCK, PREROLL_LOCK, LOCK
|
|
* adjust a timestamp with the latency and timestamp offset. This function does
|
|
* not adjust for the render delay. */
|
|
static GstClockTime
|
|
gst_base_sink_adjust_time (GstBaseSink * basesink, GstClockTime time)
|
|
{
|
|
GstClockTimeDiff ts_offset;
|
|
|
|
/* don't do anything funny with invalid timestamps */
|
|
if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (time)))
|
|
return time;
|
|
|
|
time += basesink->priv->latency;
|
|
|
|
/* apply offset, be carefull for underflows */
|
|
ts_offset = basesink->priv->ts_offset;
|
|
if (ts_offset < 0) {
|
|
ts_offset = -ts_offset;
|
|
if (ts_offset < time)
|
|
time -= ts_offset;
|
|
else
|
|
time = 0;
|
|
} else
|
|
time += ts_offset;
|
|
|
|
/* subtract the render delay again, which was included in the latency */
|
|
if (time > basesink->priv->render_delay)
|
|
time -= basesink->priv->render_delay;
|
|
else
|
|
time = 0;
|
|
|
|
return time;
|
|
}
|
|
|
|
/**
|
|
* gst_base_sink_wait_clock:
|
|
* @sink: the sink
|
|
* @time: the running_time to be reached
|
|
* @jitter: (out) (allow-none): the jitter to be filled with time diff, or NULL
|
|
*
|
|
* This function will block until @time is reached. It is usually called by
|
|
* subclasses that use their own internal synchronisation.
|
|
*
|
|
* If @time is not valid, no sycnhronisation is done and #GST_CLOCK_BADTIME is
|
|
* returned. Likewise, if synchronisation is disabled in the element or there
|
|
* is no clock, no synchronisation is done and #GST_CLOCK_BADTIME is returned.
|
|
*
|
|
* This function should only be called with the PREROLL_LOCK held, like when
|
|
* receiving an EOS event in the #GstBaseSinkClass.event() vmethod or when
|
|
* receiving a buffer in
|
|
* the #GstBaseSinkClass.render() vmethod.
|
|
*
|
|
* The @time argument should be the running_time of when this method should
|
|
* return and is not adjusted with any latency or offset configured in the
|
|
* sink.
|
|
*
|
|
* Returns: #GstClockReturn
|
|
*/
|
|
GstClockReturn
|
|
gst_base_sink_wait_clock (GstBaseSink * sink, GstClockTime time,
|
|
GstClockTimeDiff * jitter)
|
|
{
|
|
GstClockReturn ret;
|
|
GstClock *clock;
|
|
GstClockTime base_time;
|
|
|
|
if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (time)))
|
|
goto invalid_time;
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
if (G_UNLIKELY (!sink->sync))
|
|
goto no_sync;
|
|
|
|
if (G_UNLIKELY ((clock = GST_ELEMENT_CLOCK (sink)) == NULL))
|
|
goto no_clock;
|
|
|
|
base_time = GST_ELEMENT_CAST (sink)->base_time;
|
|
GST_LOG_OBJECT (sink,
|
|
"time %" GST_TIME_FORMAT ", base_time %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (time), GST_TIME_ARGS (base_time));
|
|
|
|
/* add base_time to running_time to get the time against the clock */
|
|
time += base_time;
|
|
|
|
/* Re-use existing clockid if available */
|
|
/* FIXME: Casting to GstClockEntry only works because the types
|
|
* are the same */
|
|
if (G_LIKELY (sink->priv->cached_clock_id != NULL
|
|
&& GST_CLOCK_ENTRY_CLOCK ((GstClockEntry *) sink->
|
|
priv->cached_clock_id) == clock)) {
|
|
if (!gst_clock_single_shot_id_reinit (clock, sink->priv->cached_clock_id,
|
|
time)) {
|
|
gst_clock_id_unref (sink->priv->cached_clock_id);
|
|
sink->priv->cached_clock_id = gst_clock_new_single_shot_id (clock, time);
|
|
}
|
|
} else {
|
|
if (sink->priv->cached_clock_id != NULL)
|
|
gst_clock_id_unref (sink->priv->cached_clock_id);
|
|
sink->priv->cached_clock_id = gst_clock_new_single_shot_id (clock, time);
|
|
}
|
|
GST_OBJECT_UNLOCK (sink);
|
|
|
|
/* A blocking wait is performed on the clock. We save the ClockID
|
|
* so we can unlock the entry at any time. While we are blocking, we
|
|
* release the PREROLL_LOCK so that other threads can interrupt the
|
|
* entry. */
|
|
sink->clock_id = sink->priv->cached_clock_id;
|
|
/* release the preroll lock while waiting */
|
|
GST_BASE_SINK_PREROLL_UNLOCK (sink);
|
|
|
|
ret = gst_clock_id_wait (sink->priv->cached_clock_id, jitter);
|
|
|
|
GST_BASE_SINK_PREROLL_LOCK (sink);
|
|
sink->clock_id = NULL;
|
|
|
|
return ret;
|
|
|
|
/* no syncing needed */
|
|
invalid_time:
|
|
{
|
|
GST_DEBUG_OBJECT (sink, "time not valid, no sync needed");
|
|
return GST_CLOCK_BADTIME;
|
|
}
|
|
no_sync:
|
|
{
|
|
GST_DEBUG_OBJECT (sink, "sync disabled");
|
|
GST_OBJECT_UNLOCK (sink);
|
|
return GST_CLOCK_BADTIME;
|
|
}
|
|
no_clock:
|
|
{
|
|
GST_DEBUG_OBJECT (sink, "no clock, can't sync");
|
|
GST_OBJECT_UNLOCK (sink);
|
|
return GST_CLOCK_BADTIME;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_base_sink_wait_preroll:
|
|
* @sink: the sink
|
|
*
|
|
* If the #GstBaseSinkClass.render() method performs its own synchronisation
|
|
* against the clock it must unblock when going from PLAYING to the PAUSED state
|
|
* and call this method before continuing to render the remaining data.
|
|
*
|
|
* This function will block until a state change to PLAYING happens (in which
|
|
* case this function returns #GST_FLOW_OK) or the processing must be stopped due
|
|
* to a state change to READY or a FLUSH event (in which case this function
|
|
* returns #GST_FLOW_FLUSHING).
|
|
*
|
|
* This function should only be called with the PREROLL_LOCK held, like in the
|
|
* render function.
|
|
*
|
|
* Returns: #GST_FLOW_OK if the preroll completed and processing can
|
|
* continue. Any other return value should be returned from the render vmethod.
|
|
*/
|
|
GstFlowReturn
|
|
gst_base_sink_wait_preroll (GstBaseSink * sink)
|
|
{
|
|
sink->have_preroll = TRUE;
|
|
GST_DEBUG_OBJECT (sink, "waiting in preroll for flush or PLAYING");
|
|
/* block until the state changes, or we get a flush, or something */
|
|
GST_BASE_SINK_PREROLL_WAIT (sink);
|
|
sink->have_preroll = FALSE;
|
|
if (G_UNLIKELY (sink->flushing))
|
|
goto stopping;
|
|
if (G_UNLIKELY (sink->priv->step_unlock))
|
|
goto step_unlocked;
|
|
GST_DEBUG_OBJECT (sink, "continue after preroll");
|
|
|
|
return GST_FLOW_OK;
|
|
|
|
/* ERRORS */
|
|
stopping:
|
|
{
|
|
GST_DEBUG_OBJECT (sink, "preroll interrupted because of flush");
|
|
return GST_FLOW_FLUSHING;
|
|
}
|
|
step_unlocked:
|
|
{
|
|
sink->priv->step_unlock = FALSE;
|
|
GST_DEBUG_OBJECT (sink, "preroll interrupted because of step");
|
|
return GST_FLOW_STEP;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_base_sink_do_preroll:
|
|
* @sink: the sink
|
|
* @obj: (transfer none): the mini object that caused the preroll
|
|
*
|
|
* If the @sink spawns its own thread for pulling buffers from upstream it
|
|
* should call this method after it has pulled a buffer. If the element needed
|
|
* to preroll, this function will perform the preroll and will then block
|
|
* until the element state is changed.
|
|
*
|
|
* This function should be called with the PREROLL_LOCK held.
|
|
*
|
|
* Returns: #GST_FLOW_OK if the preroll completed and processing can
|
|
* continue. Any other return value should be returned from the render vmethod.
|
|
*/
|
|
GstFlowReturn
|
|
gst_base_sink_do_preroll (GstBaseSink * sink, GstMiniObject * obj)
|
|
{
|
|
GstFlowReturn ret;
|
|
|
|
while (G_UNLIKELY (sink->need_preroll)) {
|
|
GST_DEBUG_OBJECT (sink, "prerolling object %p", obj);
|
|
|
|
/* if it's a buffer, we need to call the preroll method */
|
|
if (sink->priv->call_preroll) {
|
|
GstBaseSinkClass *bclass;
|
|
GstBuffer *buf;
|
|
|
|
if (GST_IS_BUFFER_LIST (obj)) {
|
|
buf = gst_buffer_list_get (GST_BUFFER_LIST_CAST (obj), 0);
|
|
g_assert (NULL != buf);
|
|
} else if (GST_IS_BUFFER (obj)) {
|
|
buf = GST_BUFFER_CAST (obj);
|
|
/* For buffer lists do not set last buffer for now */
|
|
gst_base_sink_set_last_buffer (sink, buf);
|
|
} else
|
|
buf = NULL;
|
|
|
|
if (buf) {
|
|
GST_DEBUG_OBJECT (sink, "preroll buffer %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
|
|
|
|
bclass = GST_BASE_SINK_GET_CLASS (sink);
|
|
|
|
if (bclass->prepare)
|
|
if ((ret = bclass->prepare (sink, buf)) != GST_FLOW_OK)
|
|
goto prepare_canceled;
|
|
|
|
if (bclass->preroll)
|
|
if ((ret = bclass->preroll (sink, buf)) != GST_FLOW_OK)
|
|
goto preroll_canceled;
|
|
|
|
sink->priv->call_preroll = FALSE;
|
|
}
|
|
}
|
|
|
|
/* commit state */
|
|
if (G_LIKELY (sink->playing_async)) {
|
|
if (G_UNLIKELY (!gst_base_sink_commit_state (sink)))
|
|
goto stopping;
|
|
}
|
|
|
|
/* need to recheck here because the commit state could have
|
|
* made us not need the preroll anymore */
|
|
if (G_LIKELY (sink->need_preroll)) {
|
|
/* block until the state changes, or we get a flush, or something */
|
|
ret = gst_base_sink_wait_preroll (sink);
|
|
if ((ret != GST_FLOW_OK) && (ret != GST_FLOW_STEP))
|
|
goto preroll_failed;
|
|
}
|
|
}
|
|
return GST_FLOW_OK;
|
|
|
|
/* ERRORS */
|
|
prepare_canceled:
|
|
{
|
|
GST_DEBUG_OBJECT (sink, "prepare failed, abort state");
|
|
gst_element_abort_state (GST_ELEMENT_CAST (sink));
|
|
return ret;
|
|
}
|
|
preroll_canceled:
|
|
{
|
|
GST_DEBUG_OBJECT (sink, "preroll failed, abort state");
|
|
gst_element_abort_state (GST_ELEMENT_CAST (sink));
|
|
return ret;
|
|
}
|
|
stopping:
|
|
{
|
|
GST_DEBUG_OBJECT (sink, "stopping while commiting state");
|
|
return GST_FLOW_FLUSHING;
|
|
}
|
|
preroll_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (sink, "preroll failed: %s", gst_flow_get_name (ret));
|
|
return ret;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_base_sink_wait:
|
|
* @sink: the sink
|
|
* @time: the running_time to be reached
|
|
* @jitter: (out) (allow-none): the jitter to be filled with time diff, or NULL
|
|
*
|
|
* This function will wait for preroll to complete and will then block until @time
|
|
* is reached. It is usually called by subclasses that use their own internal
|
|
* synchronisation but want to let some synchronization (like EOS) be handled
|
|
* by the base class.
|
|
*
|
|
* This function should only be called with the PREROLL_LOCK held (like when
|
|
* receiving an EOS event in the ::event vmethod or when handling buffers in
|
|
* ::render).
|
|
*
|
|
* The @time argument should be the running_time of when the timeout should happen
|
|
* and will be adjusted with any latency and offset configured in the sink.
|
|
*
|
|
* Returns: #GstFlowReturn
|
|
*/
|
|
GstFlowReturn
|
|
gst_base_sink_wait (GstBaseSink * sink, GstClockTime time,
|
|
GstClockTimeDiff * jitter)
|
|
{
|
|
GstClockReturn status;
|
|
GstFlowReturn ret;
|
|
|
|
do {
|
|
GstClockTime stime;
|
|
|
|
GST_DEBUG_OBJECT (sink, "checking preroll");
|
|
|
|
/* first wait for the playing state before we can continue */
|
|
while (G_UNLIKELY (sink->need_preroll)) {
|
|
ret = gst_base_sink_wait_preroll (sink);
|
|
if ((ret != GST_FLOW_OK) && (ret != GST_FLOW_STEP))
|
|
goto flushing;
|
|
}
|
|
|
|
/* preroll done, we can sync since we are in PLAYING now. */
|
|
GST_DEBUG_OBJECT (sink, "possibly waiting for clock to reach %"
|
|
GST_TIME_FORMAT, GST_TIME_ARGS (time));
|
|
|
|
/* compensate for latency, ts_offset and render delay */
|
|
stime = gst_base_sink_adjust_time (sink, time);
|
|
|
|
/* wait for the clock, this can be interrupted because we got shut down or
|
|
* we PAUSED. */
|
|
status = gst_base_sink_wait_clock (sink, stime, jitter);
|
|
|
|
GST_DEBUG_OBJECT (sink, "clock returned %d", status);
|
|
|
|
/* invalid time, no clock or sync disabled, just continue then */
|
|
if (status == GST_CLOCK_BADTIME)
|
|
break;
|
|
|
|
/* waiting could have been interrupted and we can be flushing now */
|
|
if (G_UNLIKELY (sink->flushing))
|
|
goto flushing;
|
|
|
|
/* retry if we got unscheduled, which means we did not reach the timeout
|
|
* yet. if some other error occures, we continue. */
|
|
} while (status == GST_CLOCK_UNSCHEDULED);
|
|
|
|
GST_DEBUG_OBJECT (sink, "end of stream");
|
|
|
|
return GST_FLOW_OK;
|
|
|
|
/* ERRORS */
|
|
flushing:
|
|
{
|
|
GST_DEBUG_OBJECT (sink, "we are flushing");
|
|
return GST_FLOW_FLUSHING;
|
|
}
|
|
}
|
|
|
|
/* with STREAM_LOCK, PREROLL_LOCK
|
|
*
|
|
* Make sure we are in PLAYING and synchronize an object to the clock.
|
|
*
|
|
* If we need preroll, we are not in PLAYING. We try to commit the state
|
|
* if needed and then block if we still are not PLAYING.
|
|
*
|
|
* We start waiting on the clock in PLAYING. If we got interrupted, we
|
|
* immediately try to re-preroll.
|
|
*
|
|
* Some objects do not need synchronisation (most events) and so this function
|
|
* immediately returns GST_FLOW_OK.
|
|
*
|
|
* for objects that arrive later than max-lateness to be synchronized to the
|
|
* clock have the @late boolean set to TRUE.
|
|
*
|
|
* This function keeps a running average of the jitter (the diff between the
|
|
* clock time and the requested sync time). The jitter is negative for
|
|
* objects that arrive in time and positive for late buffers.
|
|
*
|
|
* does not take ownership of obj.
|
|
*/
|
|
static GstFlowReturn
|
|
gst_base_sink_do_sync (GstBaseSink * basesink,
|
|
GstMiniObject * obj, gboolean * late, gboolean * step_end)
|
|
{
|
|
GstClockTimeDiff jitter = 0;
|
|
gboolean syncable;
|
|
GstClockReturn status = GST_CLOCK_OK;
|
|
GstClockTime rstart, rstop, rnext, sstart, sstop, stime;
|
|
gboolean do_sync;
|
|
GstBaseSinkPrivate *priv;
|
|
GstFlowReturn ret;
|
|
GstStepInfo *current, *pending;
|
|
gboolean stepped;
|
|
|
|
priv = basesink->priv;
|
|
|
|
do_step:
|
|
sstart = sstop = rstart = rstop = rnext = GST_CLOCK_TIME_NONE;
|
|
do_sync = TRUE;
|
|
stepped = FALSE;
|
|
|
|
priv->current_rstart = GST_CLOCK_TIME_NONE;
|
|
|
|
/* get stepping info */
|
|
current = &priv->current_step;
|
|
pending = &priv->pending_step;
|
|
|
|
/* get timing information for this object against the render segment */
|
|
syncable = gst_base_sink_get_sync_times (basesink, obj,
|
|
&sstart, &sstop, &rstart, &rstop, &rnext, &do_sync, &stepped, current,
|
|
step_end);
|
|
|
|
if (G_UNLIKELY (stepped))
|
|
goto step_skipped;
|
|
|
|
/* a syncable object needs to participate in preroll and
|
|
* clocking. All buffers and EOS are syncable. */
|
|
if (G_UNLIKELY (!syncable))
|
|
goto not_syncable;
|
|
|
|
/* store timing info for current object */
|
|
priv->current_rstart = rstart;
|
|
priv->current_rstop = (GST_CLOCK_TIME_IS_VALID (rstop) ? rstop : rstart);
|
|
|
|
/* save sync time for eos when the previous object needed sync */
|
|
priv->eos_rtime = (do_sync ? rnext : GST_CLOCK_TIME_NONE);
|
|
|
|
/* calculate inter frame spacing */
|
|
if (G_UNLIKELY (priv->prev_rstart != -1 && priv->prev_rstart < rstart)) {
|
|
GstClockTime in_diff;
|
|
|
|
in_diff = rstart - priv->prev_rstart;
|
|
|
|
if (priv->avg_in_diff == -1)
|
|
priv->avg_in_diff = in_diff;
|
|
else
|
|
priv->avg_in_diff = UPDATE_RUNNING_AVG (priv->avg_in_diff, in_diff);
|
|
|
|
GST_LOG_OBJECT (basesink, "avg frame diff %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (priv->avg_in_diff));
|
|
|
|
}
|
|
priv->prev_rstart = rstart;
|
|
|
|
if (G_UNLIKELY (priv->earliest_in_time != -1
|
|
&& rstart < priv->earliest_in_time))
|
|
goto qos_dropped;
|
|
|
|
again:
|
|
/* first do preroll, this makes sure we commit our state
|
|
* to PAUSED and can continue to PLAYING. We cannot perform
|
|
* any clock sync in PAUSED because there is no clock. */
|
|
ret = gst_base_sink_do_preroll (basesink, obj);
|
|
if (G_UNLIKELY (ret != GST_FLOW_OK))
|
|
goto preroll_failed;
|
|
|
|
/* update the segment with a pending step if the current one is invalid and we
|
|
* have a new pending one. We only accept new step updates after a preroll */
|
|
if (G_UNLIKELY (pending->valid && !current->valid)) {
|
|
start_stepping (basesink, &basesink->segment, pending, current);
|
|
goto do_step;
|
|
}
|
|
|
|
/* After rendering we store the position of the last buffer so that we can use
|
|
* it to report the position. We need to take the lock here. */
|
|
GST_OBJECT_LOCK (basesink);
|
|
priv->current_sstart = sstart;
|
|
priv->current_sstop = (GST_CLOCK_TIME_IS_VALID (sstop) ? sstop : sstart);
|
|
GST_OBJECT_UNLOCK (basesink);
|
|
|
|
if (!do_sync)
|
|
goto done;
|
|
|
|
/* adjust for latency */
|
|
stime = gst_base_sink_adjust_time (basesink, rstart);
|
|
|
|
/* adjust for rate control */
|
|
if (priv->rc_next == -1 || (stime != -1 && stime >= priv->rc_next)) {
|
|
GST_DEBUG_OBJECT (basesink, "reset rc_time to time %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (stime));
|
|
priv->rc_time = stime;
|
|
priv->rc_accumulated = 0;
|
|
} else {
|
|
GST_DEBUG_OBJECT (basesink, "rate control next %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (priv->rc_next));
|
|
stime = priv->rc_next;
|
|
}
|
|
|
|
/* preroll done, we can sync since we are in PLAYING now. */
|
|
GST_DEBUG_OBJECT (basesink, "possibly waiting for clock to reach %"
|
|
GST_TIME_FORMAT ", adjusted %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (rstart), GST_TIME_ARGS (stime));
|
|
|
|
/* This function will return immediately if start == -1, no clock
|
|
* or sync is disabled with GST_CLOCK_BADTIME. */
|
|
status = gst_base_sink_wait_clock (basesink, stime, &jitter);
|
|
|
|
GST_DEBUG_OBJECT (basesink, "clock returned %d, jitter %c%" GST_TIME_FORMAT,
|
|
status, (jitter < 0 ? '-' : ' '), GST_TIME_ARGS (ABS (jitter)));
|
|
|
|
/* invalid time, no clock or sync disabled, just render */
|
|
if (status == GST_CLOCK_BADTIME)
|
|
goto done;
|
|
|
|
/* waiting could have been interrupted and we can be flushing now */
|
|
if (G_UNLIKELY (basesink->flushing))
|
|
goto flushing;
|
|
|
|
/* check for unlocked by a state change, we are not flushing so
|
|
* we can try to preroll on the current buffer. */
|
|
if (G_UNLIKELY (status == GST_CLOCK_UNSCHEDULED)) {
|
|
GST_DEBUG_OBJECT (basesink, "unscheduled, waiting some more");
|
|
priv->call_preroll = TRUE;
|
|
goto again;
|
|
}
|
|
|
|
/* successful syncing done, record observation */
|
|
priv->current_jitter = jitter;
|
|
|
|
/* check if the object should be dropped */
|
|
*late = gst_base_sink_is_too_late (basesink, obj, rstart, rstop,
|
|
status, jitter);
|
|
|
|
done:
|
|
return GST_FLOW_OK;
|
|
|
|
/* ERRORS */
|
|
step_skipped:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "skipped stepped object %p", obj);
|
|
*late = TRUE;
|
|
return GST_FLOW_OK;
|
|
}
|
|
not_syncable:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "non syncable object %p", obj);
|
|
return GST_FLOW_OK;
|
|
}
|
|
qos_dropped:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "dropped because of QoS %p", obj);
|
|
*late = TRUE;
|
|
return GST_FLOW_OK;
|
|
}
|
|
flushing:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "we are flushing");
|
|
return GST_FLOW_FLUSHING;
|
|
}
|
|
preroll_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "preroll failed");
|
|
*step_end = FALSE;
|
|
return ret;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_sink_send_qos (GstBaseSink * basesink, GstQOSType type,
|
|
gdouble proportion, GstClockTime time, GstClockTimeDiff diff)
|
|
{
|
|
GstEvent *event;
|
|
gboolean res;
|
|
|
|
/* generate Quality-of-Service event */
|
|
GST_CAT_DEBUG_OBJECT (GST_CAT_QOS, basesink,
|
|
"qos: type %d, proportion: %lf, diff %" G_GINT64_FORMAT ", timestamp %"
|
|
GST_TIME_FORMAT, type, proportion, diff, GST_TIME_ARGS (time));
|
|
|
|
event = gst_event_new_qos (type, proportion, diff, time);
|
|
|
|
/* send upstream */
|
|
res = gst_pad_push_event (basesink->sinkpad, event);
|
|
|
|
return res;
|
|
}
|
|
|
|
static void
|
|
gst_base_sink_perform_qos (GstBaseSink * sink, gboolean dropped)
|
|
{
|
|
GstBaseSinkPrivate *priv;
|
|
GstClockTime start, stop;
|
|
GstClockTimeDiff jitter;
|
|
GstClockTime pt, entered, left;
|
|
GstClockTime duration;
|
|
gdouble rate;
|
|
|
|
priv = sink->priv;
|
|
|
|
start = priv->current_rstart;
|
|
|
|
if (priv->current_step.valid)
|
|
return;
|
|
|
|
/* if Quality-of-Service disabled, do nothing */
|
|
if (!g_atomic_int_get (&priv->qos_enabled) ||
|
|
!GST_CLOCK_TIME_IS_VALID (start))
|
|
return;
|
|
|
|
stop = priv->current_rstop;
|
|
jitter = priv->current_jitter;
|
|
|
|
if (jitter < 0) {
|
|
/* this is the time the buffer entered the sink */
|
|
if (start < -jitter)
|
|
entered = 0;
|
|
else
|
|
entered = start + jitter;
|
|
left = start;
|
|
} else {
|
|
/* this is the time the buffer entered the sink */
|
|
entered = start + jitter;
|
|
/* this is the time the buffer left the sink */
|
|
left = start + jitter;
|
|
}
|
|
|
|
/* calculate duration of the buffer */
|
|
if (GST_CLOCK_TIME_IS_VALID (stop) && stop != start)
|
|
duration = stop - start;
|
|
else
|
|
duration = priv->avg_in_diff;
|
|
|
|
/* if we have the time when the last buffer left us, calculate
|
|
* processing time */
|
|
if (GST_CLOCK_TIME_IS_VALID (priv->last_left)) {
|
|
if (entered > priv->last_left) {
|
|
pt = entered - priv->last_left;
|
|
} else {
|
|
pt = 0;
|
|
}
|
|
} else {
|
|
pt = priv->avg_pt;
|
|
}
|
|
|
|
GST_CAT_DEBUG_OBJECT (GST_CAT_QOS, sink, "start: %" GST_TIME_FORMAT
|
|
", stop %" GST_TIME_FORMAT ", entered %" GST_TIME_FORMAT ", left %"
|
|
GST_TIME_FORMAT ", pt: %" GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT
|
|
",jitter %" G_GINT64_FORMAT, GST_TIME_ARGS (start), GST_TIME_ARGS (stop),
|
|
GST_TIME_ARGS (entered), GST_TIME_ARGS (left), GST_TIME_ARGS (pt),
|
|
GST_TIME_ARGS (duration), jitter);
|
|
|
|
GST_CAT_DEBUG_OBJECT (GST_CAT_QOS, sink, "avg_duration: %" GST_TIME_FORMAT
|
|
", avg_pt: %" GST_TIME_FORMAT ", avg_rate: %g",
|
|
GST_TIME_ARGS (priv->avg_duration), GST_TIME_ARGS (priv->avg_pt),
|
|
priv->avg_rate);
|
|
|
|
/* collect running averages. for first observations, we copy the
|
|
* values */
|
|
if (!GST_CLOCK_TIME_IS_VALID (priv->avg_duration))
|
|
priv->avg_duration = duration;
|
|
else
|
|
priv->avg_duration = UPDATE_RUNNING_AVG (priv->avg_duration, duration);
|
|
|
|
if (!GST_CLOCK_TIME_IS_VALID (priv->avg_pt))
|
|
priv->avg_pt = pt;
|
|
else
|
|
priv->avg_pt = UPDATE_RUNNING_AVG (priv->avg_pt, pt);
|
|
|
|
if (priv->avg_duration != 0)
|
|
rate =
|
|
gst_guint64_to_gdouble (priv->avg_pt) /
|
|
gst_guint64_to_gdouble (priv->avg_duration);
|
|
else
|
|
rate = 1.0;
|
|
|
|
if (GST_CLOCK_TIME_IS_VALID (priv->last_left)) {
|
|
if (dropped || priv->avg_rate < 0.0) {
|
|
priv->avg_rate = rate;
|
|
} else {
|
|
if (rate > 1.0)
|
|
priv->avg_rate = UPDATE_RUNNING_AVG_N (priv->avg_rate, rate);
|
|
else
|
|
priv->avg_rate = UPDATE_RUNNING_AVG_P (priv->avg_rate, rate);
|
|
}
|
|
}
|
|
|
|
GST_CAT_DEBUG_OBJECT (GST_CAT_QOS, sink,
|
|
"updated: avg_duration: %" GST_TIME_FORMAT ", avg_pt: %" GST_TIME_FORMAT
|
|
", avg_rate: %g", GST_TIME_ARGS (priv->avg_duration),
|
|
GST_TIME_ARGS (priv->avg_pt), priv->avg_rate);
|
|
|
|
|
|
if (priv->avg_rate >= 0.0) {
|
|
GstQOSType type;
|
|
GstClockTimeDiff diff;
|
|
|
|
/* if we have a valid rate, start sending QoS messages */
|
|
if (priv->current_jitter < 0) {
|
|
/* make sure we never go below 0 when adding the jitter to the
|
|
* timestamp. */
|
|
if (priv->current_rstart < -priv->current_jitter)
|
|
priv->current_jitter = -priv->current_rstart;
|
|
}
|
|
|
|
if (priv->throttle_time > 0) {
|
|
diff = priv->throttle_time;
|
|
type = GST_QOS_TYPE_THROTTLE;
|
|
} else {
|
|
diff = priv->current_jitter;
|
|
if (diff <= 0)
|
|
type = GST_QOS_TYPE_OVERFLOW;
|
|
else
|
|
type = GST_QOS_TYPE_UNDERFLOW;
|
|
}
|
|
|
|
gst_base_sink_send_qos (sink, type, priv->avg_rate, priv->current_rstart,
|
|
diff);
|
|
}
|
|
|
|
/* record when this buffer will leave us */
|
|
priv->last_left = left;
|
|
}
|
|
|
|
/* reset all qos measuring */
|
|
static void
|
|
gst_base_sink_reset_qos (GstBaseSink * sink)
|
|
{
|
|
GstBaseSinkPrivate *priv;
|
|
|
|
priv = sink->priv;
|
|
|
|
priv->last_render_time = GST_CLOCK_TIME_NONE;
|
|
priv->prev_rstart = GST_CLOCK_TIME_NONE;
|
|
priv->earliest_in_time = GST_CLOCK_TIME_NONE;
|
|
priv->last_left = GST_CLOCK_TIME_NONE;
|
|
priv->avg_duration = GST_CLOCK_TIME_NONE;
|
|
priv->avg_pt = GST_CLOCK_TIME_NONE;
|
|
priv->avg_rate = -1.0;
|
|
priv->avg_render = GST_CLOCK_TIME_NONE;
|
|
priv->avg_in_diff = GST_CLOCK_TIME_NONE;
|
|
priv->rendered = 0;
|
|
priv->dropped = 0;
|
|
|
|
}
|
|
|
|
/* Checks if the object was scheduled too late.
|
|
*
|
|
* rstart/rstop contain the running_time start and stop values
|
|
* of the object.
|
|
*
|
|
* status and jitter contain the return values from the clock wait.
|
|
*
|
|
* returns TRUE if the buffer was too late.
|
|
*/
|
|
static gboolean
|
|
gst_base_sink_is_too_late (GstBaseSink * basesink, GstMiniObject * obj,
|
|
GstClockTime rstart, GstClockTime rstop,
|
|
GstClockReturn status, GstClockTimeDiff jitter)
|
|
{
|
|
gboolean late;
|
|
guint64 max_lateness;
|
|
GstBaseSinkPrivate *priv;
|
|
|
|
priv = basesink->priv;
|
|
|
|
late = FALSE;
|
|
|
|
/* only for objects that were too late */
|
|
if (G_LIKELY (status != GST_CLOCK_EARLY))
|
|
goto in_time;
|
|
|
|
max_lateness = basesink->max_lateness;
|
|
|
|
/* check if frame dropping is enabled */
|
|
if (max_lateness == -1)
|
|
goto no_drop;
|
|
|
|
/* only check for buffers */
|
|
if (G_UNLIKELY (!GST_IS_BUFFER (obj)))
|
|
goto not_buffer;
|
|
|
|
/* can't do check if we don't have a timestamp */
|
|
if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (rstart)))
|
|
goto no_timestamp;
|
|
|
|
/* we can add a valid stop time */
|
|
if (GST_CLOCK_TIME_IS_VALID (rstop))
|
|
max_lateness += rstop;
|
|
else {
|
|
max_lateness += rstart;
|
|
/* no stop time, use avg frame diff */
|
|
if (priv->avg_in_diff != -1)
|
|
max_lateness += priv->avg_in_diff;
|
|
}
|
|
|
|
/* if the jitter bigger than duration and lateness we are too late */
|
|
if ((late = rstart + jitter > max_lateness)) {
|
|
GST_CAT_DEBUG_OBJECT (GST_CAT_PERFORMANCE, basesink,
|
|
"buffer is too late %" GST_TIME_FORMAT
|
|
" > %" GST_TIME_FORMAT, GST_TIME_ARGS (rstart + jitter),
|
|
GST_TIME_ARGS (max_lateness));
|
|
/* !!emergency!!, if we did not receive anything valid for more than a
|
|
* second, render it anyway so the user sees something */
|
|
if (GST_CLOCK_TIME_IS_VALID (priv->last_render_time) &&
|
|
rstart - priv->last_render_time > GST_SECOND) {
|
|
late = FALSE;
|
|
GST_ELEMENT_WARNING (basesink, CORE, CLOCK,
|
|
(_("A lot of buffers are being dropped.")),
|
|
("There may be a timestamping problem, or this computer is too slow."));
|
|
GST_CAT_DEBUG_OBJECT (GST_CAT_PERFORMANCE, basesink,
|
|
"**emergency** last buffer at %" GST_TIME_FORMAT " > GST_SECOND",
|
|
GST_TIME_ARGS (priv->last_render_time));
|
|
}
|
|
}
|
|
|
|
done:
|
|
if (!late || !GST_CLOCK_TIME_IS_VALID (priv->last_render_time)) {
|
|
priv->last_render_time = rstart;
|
|
/* the next allowed input timestamp */
|
|
if (priv->throttle_time > 0)
|
|
priv->earliest_in_time = rstart + priv->throttle_time;
|
|
}
|
|
return late;
|
|
|
|
/* all is fine */
|
|
in_time:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "object was scheduled in time");
|
|
goto done;
|
|
}
|
|
no_drop:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "frame dropping disabled");
|
|
goto done;
|
|
}
|
|
not_buffer:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "object is not a buffer");
|
|
return FALSE;
|
|
}
|
|
no_timestamp:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "buffer has no timestamp");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/* called before and after calling the render vmethod. It keeps track of how
|
|
* much time was spent in the render method and is used to check if we are
|
|
* flooded */
|
|
static void
|
|
gst_base_sink_do_render_stats (GstBaseSink * basesink, gboolean start)
|
|
{
|
|
GstBaseSinkPrivate *priv;
|
|
|
|
priv = basesink->priv;
|
|
|
|
if (start) {
|
|
priv->start = gst_util_get_timestamp ();
|
|
} else {
|
|
GstClockTime elapsed;
|
|
|
|
priv->stop = gst_util_get_timestamp ();
|
|
|
|
elapsed = GST_CLOCK_DIFF (priv->start, priv->stop);
|
|
|
|
if (!GST_CLOCK_TIME_IS_VALID (priv->avg_render))
|
|
priv->avg_render = elapsed;
|
|
else
|
|
priv->avg_render = UPDATE_RUNNING_AVG (priv->avg_render, elapsed);
|
|
|
|
GST_CAT_DEBUG_OBJECT (GST_CAT_QOS, basesink,
|
|
"avg_render: %" GST_TIME_FORMAT, GST_TIME_ARGS (priv->avg_render));
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_base_sink_flush_start (GstBaseSink * basesink, GstPad * pad)
|
|
{
|
|
/* make sure we are not blocked on the clock also clear any pending
|
|
* eos state. */
|
|
gst_base_sink_set_flushing (basesink, pad, TRUE);
|
|
|
|
/* we grab the stream lock but that is not needed since setting the
|
|
* sink to flushing would make sure no state commit is being done
|
|
* anymore */
|
|
GST_PAD_STREAM_LOCK (pad);
|
|
gst_base_sink_reset_qos (basesink);
|
|
/* and we need to commit our state again on the next
|
|
* prerolled buffer */
|
|
basesink->playing_async = TRUE;
|
|
if (basesink->priv->async_enabled) {
|
|
gst_element_lost_state (GST_ELEMENT_CAST (basesink));
|
|
} else {
|
|
/* start time reset in above case as well;
|
|
* arranges for a.o. proper position reporting when flushing in PAUSED */
|
|
gst_element_set_start_time (GST_ELEMENT_CAST (basesink), 0);
|
|
basesink->priv->have_latency = TRUE;
|
|
}
|
|
gst_base_sink_set_last_buffer (basesink, NULL);
|
|
GST_PAD_STREAM_UNLOCK (pad);
|
|
}
|
|
|
|
static void
|
|
gst_base_sink_flush_stop (GstBaseSink * basesink, GstPad * pad,
|
|
gboolean reset_time)
|
|
{
|
|
/* unset flushing so we can accept new data, this also flushes out any EOS
|
|
* event. */
|
|
gst_base_sink_set_flushing (basesink, pad, FALSE);
|
|
|
|
/* for position reporting */
|
|
GST_OBJECT_LOCK (basesink);
|
|
basesink->priv->current_sstart = GST_CLOCK_TIME_NONE;
|
|
basesink->priv->current_sstop = GST_CLOCK_TIME_NONE;
|
|
basesink->priv->eos_rtime = GST_CLOCK_TIME_NONE;
|
|
basesink->priv->call_preroll = TRUE;
|
|
basesink->priv->current_step.valid = FALSE;
|
|
basesink->priv->pending_step.valid = FALSE;
|
|
if (basesink->pad_mode == GST_PAD_MODE_PUSH) {
|
|
/* we need new segment info after the flush. */
|
|
basesink->have_newsegment = FALSE;
|
|
if (reset_time) {
|
|
gst_segment_init (&basesink->segment, GST_FORMAT_UNDEFINED);
|
|
GST_ELEMENT_START_TIME (basesink) = 0;
|
|
}
|
|
}
|
|
GST_OBJECT_UNLOCK (basesink);
|
|
|
|
if (reset_time) {
|
|
GST_DEBUG_OBJECT (basesink, "posting reset-time message");
|
|
gst_element_post_message (GST_ELEMENT_CAST (basesink),
|
|
gst_message_new_reset_time (GST_OBJECT_CAST (basesink), 0));
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_base_sink_default_wait_event (GstBaseSink * basesink, GstEvent * event)
|
|
{
|
|
GstFlowReturn ret;
|
|
gboolean late, step_end;
|
|
|
|
ret = gst_base_sink_do_sync (basesink, GST_MINI_OBJECT_CAST (event),
|
|
&late, &step_end);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_base_sink_wait_event (GstBaseSink * basesink, GstEvent * event)
|
|
{
|
|
GstFlowReturn ret;
|
|
GstBaseSinkClass *bclass;
|
|
|
|
bclass = GST_BASE_SINK_GET_CLASS (basesink);
|
|
|
|
if (G_LIKELY (bclass->wait_event))
|
|
ret = bclass->wait_event (basesink, event);
|
|
else
|
|
ret = GST_FLOW_NOT_SUPPORTED;
|
|
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_sink_default_event (GstBaseSink * basesink, GstEvent * event)
|
|
{
|
|
gboolean result = TRUE;
|
|
GstBaseSinkClass *bclass;
|
|
|
|
bclass = GST_BASE_SINK_GET_CLASS (basesink);
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_FLUSH_START:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "flush-start %p", event);
|
|
gst_base_sink_flush_start (basesink, basesink->sinkpad);
|
|
break;
|
|
}
|
|
case GST_EVENT_FLUSH_STOP:
|
|
{
|
|
gboolean reset_time;
|
|
|
|
gst_event_parse_flush_stop (event, &reset_time);
|
|
GST_DEBUG_OBJECT (basesink, "flush-stop %p, reset_time: %d", event,
|
|
reset_time);
|
|
gst_base_sink_flush_stop (basesink, basesink->sinkpad, reset_time);
|
|
break;
|
|
}
|
|
case GST_EVENT_EOS:
|
|
{
|
|
GstMessage *message;
|
|
guint32 seqnum;
|
|
|
|
/* we set the received EOS flag here so that we can use it when testing if
|
|
* we are prerolled and to refuse more buffers. */
|
|
basesink->priv->received_eos = TRUE;
|
|
|
|
/* wait for EOS */
|
|
if (G_UNLIKELY (gst_base_sink_wait_event (basesink,
|
|
event) != GST_FLOW_OK)) {
|
|
result = FALSE;
|
|
goto done;
|
|
}
|
|
|
|
/* the EOS event is completely handled so we mark
|
|
* ourselves as being in the EOS state. eos is also
|
|
* protected by the object lock so we can read it when
|
|
* answering the POSITION query. */
|
|
GST_OBJECT_LOCK (basesink);
|
|
basesink->eos = TRUE;
|
|
GST_OBJECT_UNLOCK (basesink);
|
|
|
|
/* ok, now we can post the message */
|
|
GST_DEBUG_OBJECT (basesink, "Now posting EOS");
|
|
|
|
seqnum = basesink->priv->seqnum = gst_event_get_seqnum (event);
|
|
GST_DEBUG_OBJECT (basesink, "Got seqnum #%" G_GUINT32_FORMAT, seqnum);
|
|
|
|
message = gst_message_new_eos (GST_OBJECT_CAST (basesink));
|
|
gst_message_set_seqnum (message, seqnum);
|
|
gst_element_post_message (GST_ELEMENT_CAST (basesink), message);
|
|
break;
|
|
}
|
|
case GST_EVENT_STREAM_START:
|
|
{
|
|
GstMessage *message;
|
|
guint32 seqnum;
|
|
|
|
seqnum = gst_event_get_seqnum (event);
|
|
GST_DEBUG_OBJECT (basesink, "Now posting STREAM_START (seqnum:%d)",
|
|
seqnum);
|
|
message = gst_message_new_stream_start (GST_OBJECT_CAST (basesink));
|
|
gst_message_set_seqnum (message, seqnum);
|
|
gst_element_post_message (GST_ELEMENT_CAST (basesink), message);
|
|
break;
|
|
}
|
|
case GST_EVENT_CAPS:
|
|
{
|
|
GstCaps *caps;
|
|
|
|
GST_DEBUG_OBJECT (basesink, "caps %p", event);
|
|
|
|
gst_event_parse_caps (event, &caps);
|
|
if (bclass->set_caps)
|
|
result = bclass->set_caps (basesink, caps);
|
|
|
|
if (result) {
|
|
GST_OBJECT_LOCK (basesink);
|
|
gst_caps_replace (&basesink->priv->caps, caps);
|
|
GST_OBJECT_UNLOCK (basesink);
|
|
}
|
|
break;
|
|
}
|
|
case GST_EVENT_SEGMENT:
|
|
/* configure the segment */
|
|
/* The segment is protected with both the STREAM_LOCK and the OBJECT_LOCK.
|
|
* We protect with the OBJECT_LOCK so that we can use the values to
|
|
* safely answer a POSITION query. */
|
|
GST_OBJECT_LOCK (basesink);
|
|
/* the newsegment event is needed to bring the buffer timestamps to the
|
|
* stream time and to drop samples outside of the playback segment. */
|
|
gst_event_copy_segment (event, &basesink->segment);
|
|
GST_DEBUG_OBJECT (basesink, "configured SEGMENT %" GST_SEGMENT_FORMAT,
|
|
&basesink->segment);
|
|
basesink->have_newsegment = TRUE;
|
|
GST_OBJECT_UNLOCK (basesink);
|
|
break;
|
|
case GST_EVENT_GAP:
|
|
{
|
|
if (G_UNLIKELY (gst_base_sink_wait_event (basesink,
|
|
event) != GST_FLOW_OK))
|
|
result = FALSE;
|
|
break;
|
|
}
|
|
case GST_EVENT_TAG:
|
|
{
|
|
GstTagList *taglist;
|
|
|
|
gst_event_parse_tag (event, &taglist);
|
|
|
|
gst_element_post_message (GST_ELEMENT_CAST (basesink),
|
|
gst_message_new_tag (GST_OBJECT_CAST (basesink),
|
|
gst_tag_list_copy (taglist)));
|
|
break;
|
|
}
|
|
case GST_EVENT_TOC:
|
|
{
|
|
GstToc *toc;
|
|
gboolean updated;
|
|
|
|
gst_event_parse_toc (event, &toc, &updated);
|
|
|
|
gst_element_post_message (GST_ELEMENT_CAST (basesink),
|
|
gst_message_new_toc (GST_OBJECT_CAST (basesink), toc, updated));
|
|
|
|
gst_toc_unref (toc);
|
|
break;
|
|
}
|
|
case GST_EVENT_SINK_MESSAGE:
|
|
{
|
|
GstMessage *msg = NULL;
|
|
|
|
gst_event_parse_sink_message (event, &msg);
|
|
if (msg)
|
|
gst_element_post_message (GST_ELEMENT_CAST (basesink), msg);
|
|
break;
|
|
}
|
|
default:
|
|
break;
|
|
}
|
|
done:
|
|
gst_event_unref (event);
|
|
|
|
return result;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
|
|
{
|
|
GstBaseSink *basesink;
|
|
gboolean result = TRUE;
|
|
GstBaseSinkClass *bclass;
|
|
|
|
basesink = GST_BASE_SINK_CAST (parent);
|
|
bclass = GST_BASE_SINK_GET_CLASS (basesink);
|
|
|
|
GST_DEBUG_OBJECT (basesink, "received event %p %" GST_PTR_FORMAT, event,
|
|
event);
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_FLUSH_STOP:
|
|
/* special case for this serialized event because we don't want to grab
|
|
* the PREROLL lock or check if we were flushing */
|
|
if (bclass->event)
|
|
result = bclass->event (basesink, event);
|
|
break;
|
|
default:
|
|
if (GST_EVENT_IS_SERIALIZED (event)) {
|
|
GST_BASE_SINK_PREROLL_LOCK (basesink);
|
|
if (G_UNLIKELY (basesink->flushing))
|
|
goto flushing;
|
|
|
|
if (G_UNLIKELY (basesink->priv->received_eos))
|
|
goto after_eos;
|
|
|
|
if (bclass->event)
|
|
result = bclass->event (basesink, event);
|
|
|
|
GST_BASE_SINK_PREROLL_UNLOCK (basesink);
|
|
} else {
|
|
if (bclass->event)
|
|
result = bclass->event (basesink, event);
|
|
}
|
|
break;
|
|
}
|
|
done:
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
flushing:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "we are flushing");
|
|
GST_BASE_SINK_PREROLL_UNLOCK (basesink);
|
|
gst_event_unref (event);
|
|
result = FALSE;
|
|
goto done;
|
|
}
|
|
|
|
after_eos:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "Event received after EOS, dropping");
|
|
GST_BASE_SINK_PREROLL_UNLOCK (basesink);
|
|
gst_event_unref (event);
|
|
result = FALSE;
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
/* default implementation to calculate the start and end
|
|
* timestamps on a buffer, subclasses can override
|
|
*/
|
|
static void
|
|
gst_base_sink_default_get_times (GstBaseSink * basesink, GstBuffer * buffer,
|
|
GstClockTime * start, GstClockTime * end)
|
|
{
|
|
GstClockTime timestamp, duration;
|
|
|
|
/* first sync on DTS, else use PTS */
|
|
timestamp = GST_BUFFER_DTS (buffer);
|
|
if (!GST_CLOCK_TIME_IS_VALID (timestamp))
|
|
timestamp = GST_BUFFER_PTS (buffer);
|
|
|
|
if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
|
|
/* get duration to calculate end time */
|
|
duration = GST_BUFFER_DURATION (buffer);
|
|
if (GST_CLOCK_TIME_IS_VALID (duration)) {
|
|
*end = timestamp + duration;
|
|
}
|
|
*start = timestamp;
|
|
}
|
|
}
|
|
|
|
/* must be called with PREROLL_LOCK */
|
|
static gboolean
|
|
gst_base_sink_needs_preroll (GstBaseSink * basesink)
|
|
{
|
|
gboolean is_prerolled, res;
|
|
|
|
/* we have 2 cases where the PREROLL_LOCK is released:
|
|
* 1) we are blocking in the PREROLL_LOCK and thus are prerolled.
|
|
* 2) we are syncing on the clock
|
|
*/
|
|
is_prerolled = basesink->have_preroll || basesink->priv->received_eos;
|
|
res = !is_prerolled;
|
|
|
|
GST_DEBUG_OBJECT (basesink, "have_preroll: %d, EOS: %d => needs preroll: %d",
|
|
basesink->have_preroll, basesink->priv->received_eos, res);
|
|
|
|
return res;
|
|
}
|
|
|
|
static gboolean
|
|
count_list_bytes (GstBuffer ** buffer, guint idx, GstBaseSinkPrivate * priv)
|
|
{
|
|
priv->rc_accumulated += gst_buffer_get_size (*buffer);
|
|
return TRUE;
|
|
}
|
|
|
|
/* with STREAM_LOCK, PREROLL_LOCK
|
|
*
|
|
* Takes a buffer and compare the timestamps with the last segment.
|
|
* If the buffer falls outside of the segment boundaries, drop it.
|
|
* Else send the buffer for preroll and rendering.
|
|
*
|
|
* This function takes ownership of the buffer.
|
|
*/
|
|
static GstFlowReturn
|
|
gst_base_sink_chain_unlocked (GstBaseSink * basesink, GstPad * pad,
|
|
gpointer obj, gboolean is_list)
|
|
{
|
|
GstBaseSinkClass *bclass;
|
|
GstBaseSinkPrivate *priv = basesink->priv;
|
|
GstFlowReturn ret;
|
|
GstClockTime start = GST_CLOCK_TIME_NONE, end = GST_CLOCK_TIME_NONE;
|
|
GstSegment *segment;
|
|
GstBuffer *sync_buf;
|
|
gint do_qos;
|
|
gboolean late, step_end;
|
|
|
|
if (G_UNLIKELY (basesink->flushing))
|
|
goto flushing;
|
|
|
|
if (G_UNLIKELY (priv->received_eos))
|
|
goto was_eos;
|
|
|
|
if (is_list) {
|
|
sync_buf = gst_buffer_list_get (GST_BUFFER_LIST_CAST (obj), 0);
|
|
g_assert (NULL != sync_buf);
|
|
} else {
|
|
sync_buf = GST_BUFFER_CAST (obj);
|
|
}
|
|
|
|
/* for code clarity */
|
|
segment = &basesink->segment;
|
|
|
|
if (G_UNLIKELY (!basesink->have_newsegment)) {
|
|
gboolean sync;
|
|
|
|
sync = gst_base_sink_get_sync (basesink);
|
|
if (sync) {
|
|
GST_ELEMENT_WARNING (basesink, STREAM, FAILED,
|
|
(_("Internal data flow problem.")),
|
|
("Received buffer without a new-segment. Assuming timestamps start from 0."));
|
|
}
|
|
|
|
/* this means this sink will assume timestamps start from 0 */
|
|
GST_OBJECT_LOCK (basesink);
|
|
segment->start = 0;
|
|
segment->stop = -1;
|
|
basesink->segment.start = 0;
|
|
basesink->segment.stop = -1;
|
|
basesink->have_newsegment = TRUE;
|
|
GST_OBJECT_UNLOCK (basesink);
|
|
}
|
|
|
|
bclass = GST_BASE_SINK_GET_CLASS (basesink);
|
|
|
|
/* check if the buffer needs to be dropped, we first ask the subclass for the
|
|
* start and end */
|
|
if (bclass->get_times)
|
|
bclass->get_times (basesink, sync_buf, &start, &end);
|
|
|
|
if (!GST_CLOCK_TIME_IS_VALID (start)) {
|
|
/* if the subclass does not want sync, we use our own values so that we at
|
|
* least clip the buffer to the segment */
|
|
gst_base_sink_default_get_times (basesink, sync_buf, &start, &end);
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (basesink, "got times start: %" GST_TIME_FORMAT
|
|
", end: %" GST_TIME_FORMAT, GST_TIME_ARGS (start), GST_TIME_ARGS (end));
|
|
|
|
/* a dropped buffer does not participate in anything */
|
|
if (GST_CLOCK_TIME_IS_VALID (start) && (segment->format == GST_FORMAT_TIME)) {
|
|
if (G_UNLIKELY (!gst_segment_clip (segment,
|
|
GST_FORMAT_TIME, start, end, NULL, NULL)))
|
|
goto out_of_segment;
|
|
}
|
|
|
|
if (bclass->prepare || bclass->prepare_list) {
|
|
gboolean late = FALSE;
|
|
gboolean do_sync = TRUE, stepped = FALSE, step_end = FALSE, syncable = TRUE;
|
|
GstClockTime sstart, sstop, rstart, rstop, rnext;
|
|
GstStepInfo *current;
|
|
|
|
current = &priv->current_step;
|
|
syncable =
|
|
gst_base_sink_get_sync_times (basesink, obj, &sstart, &sstop, &rstart,
|
|
&rstop, &rnext, &do_sync, &stepped, current, &step_end);
|
|
|
|
if (!stepped && syncable && do_sync)
|
|
late =
|
|
gst_base_sink_is_too_late (basesink, obj, rstart, rstop,
|
|
GST_CLOCK_EARLY, 0);
|
|
if (late)
|
|
goto dropped;
|
|
|
|
if (!is_list) {
|
|
if (bclass->prepare) {
|
|
ret = bclass->prepare (basesink, GST_BUFFER_CAST (obj));
|
|
if (G_UNLIKELY (ret != GST_FLOW_OK))
|
|
goto prepare_failed;
|
|
}
|
|
} else {
|
|
if (bclass->prepare_list) {
|
|
ret = bclass->prepare_list (basesink, GST_BUFFER_LIST_CAST (obj));
|
|
if (G_UNLIKELY (ret != GST_FLOW_OK))
|
|
goto prepare_failed;
|
|
}
|
|
}
|
|
}
|
|
|
|
again:
|
|
late = FALSE;
|
|
step_end = FALSE;
|
|
|
|
/* synchronize this object, non syncable objects return OK
|
|
* immediately. */
|
|
ret = gst_base_sink_do_sync (basesink, GST_MINI_OBJECT_CAST (sync_buf),
|
|
&late, &step_end);
|
|
if (G_UNLIKELY (ret != GST_FLOW_OK))
|
|
goto sync_failed;
|
|
|
|
/* drop late buffers unconditionally, let's hope it's unlikely */
|
|
if (G_UNLIKELY (late))
|
|
goto dropped;
|
|
|
|
if (priv->max_bitrate) {
|
|
if (is_list) {
|
|
gst_buffer_list_foreach (GST_BUFFER_LIST_CAST (obj),
|
|
(GstBufferListFunc) count_list_bytes, priv);
|
|
} else {
|
|
priv->rc_accumulated += gst_buffer_get_size (GST_BUFFER_CAST (obj));
|
|
}
|
|
priv->rc_next = priv->rc_time + gst_util_uint64_scale (priv->rc_accumulated,
|
|
8 * GST_SECOND, priv->max_bitrate);
|
|
}
|
|
|
|
/* read once, to get same value before and after */
|
|
do_qos = g_atomic_int_get (&priv->qos_enabled);
|
|
|
|
GST_DEBUG_OBJECT (basesink, "rendering object %p", obj);
|
|
|
|
/* record rendering time for QoS and stats */
|
|
if (do_qos)
|
|
gst_base_sink_do_render_stats (basesink, TRUE);
|
|
|
|
if (!is_list) {
|
|
/* For buffer lists do not set last buffer for now. */
|
|
gst_base_sink_set_last_buffer (basesink, GST_BUFFER_CAST (obj));
|
|
|
|
if (bclass->render)
|
|
ret = bclass->render (basesink, GST_BUFFER_CAST (obj));
|
|
} else {
|
|
if (bclass->render_list)
|
|
ret = bclass->render_list (basesink, GST_BUFFER_LIST_CAST (obj));
|
|
}
|
|
|
|
if (do_qos)
|
|
gst_base_sink_do_render_stats (basesink, FALSE);
|
|
|
|
if (ret == GST_FLOW_STEP)
|
|
goto again;
|
|
|
|
if (G_UNLIKELY (basesink->flushing))
|
|
goto flushing;
|
|
|
|
priv->rendered++;
|
|
|
|
done:
|
|
if (step_end) {
|
|
/* the step ended, check if we need to activate a new step */
|
|
GST_DEBUG_OBJECT (basesink, "step ended");
|
|
stop_stepping (basesink, &basesink->segment, &priv->current_step,
|
|
priv->current_rstart, priv->current_rstop, basesink->eos);
|
|
goto again;
|
|
}
|
|
|
|
gst_base_sink_perform_qos (basesink, late);
|
|
|
|
GST_DEBUG_OBJECT (basesink, "object unref after render %p", obj);
|
|
gst_mini_object_unref (GST_MINI_OBJECT_CAST (obj));
|
|
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
flushing:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "sink is flushing");
|
|
gst_mini_object_unref (GST_MINI_OBJECT_CAST (obj));
|
|
return GST_FLOW_FLUSHING;
|
|
}
|
|
was_eos:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "we are EOS, dropping object, return EOS");
|
|
gst_mini_object_unref (GST_MINI_OBJECT_CAST (obj));
|
|
return GST_FLOW_EOS;
|
|
}
|
|
out_of_segment:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "dropping buffer, out of clipping segment");
|
|
gst_mini_object_unref (GST_MINI_OBJECT_CAST (obj));
|
|
return GST_FLOW_OK;
|
|
}
|
|
prepare_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "prepare buffer failed %s",
|
|
gst_flow_get_name (ret));
|
|
gst_mini_object_unref (GST_MINI_OBJECT_CAST (obj));
|
|
return ret;
|
|
}
|
|
sync_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "do_sync returned %s", gst_flow_get_name (ret));
|
|
goto done;
|
|
}
|
|
dropped:
|
|
{
|
|
priv->dropped++;
|
|
GST_DEBUG_OBJECT (basesink, "buffer late, dropping");
|
|
|
|
if (g_atomic_int_get (&priv->qos_enabled)) {
|
|
GstMessage *qos_msg;
|
|
GstClockTime timestamp, duration;
|
|
|
|
timestamp = GST_BUFFER_TIMESTAMP (GST_BUFFER_CAST (sync_buf));
|
|
duration = GST_BUFFER_DURATION (GST_BUFFER_CAST (sync_buf));
|
|
|
|
GST_CAT_DEBUG_OBJECT (GST_CAT_QOS, basesink,
|
|
"qos: dropped buffer rt %" GST_TIME_FORMAT ", st %" GST_TIME_FORMAT
|
|
", ts %" GST_TIME_FORMAT ", dur %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (priv->current_rstart),
|
|
GST_TIME_ARGS (priv->current_sstart), GST_TIME_ARGS (timestamp),
|
|
GST_TIME_ARGS (duration));
|
|
GST_CAT_DEBUG_OBJECT (GST_CAT_QOS, basesink,
|
|
"qos: rendered %" G_GUINT64_FORMAT ", dropped %" G_GUINT64_FORMAT,
|
|
priv->rendered, priv->dropped);
|
|
|
|
qos_msg =
|
|
gst_message_new_qos (GST_OBJECT_CAST (basesink), basesink->sync,
|
|
priv->current_rstart, priv->current_sstart, timestamp, duration);
|
|
gst_message_set_qos_values (qos_msg, priv->current_jitter, priv->avg_rate,
|
|
1000000);
|
|
gst_message_set_qos_stats (qos_msg, GST_FORMAT_BUFFERS, priv->rendered,
|
|
priv->dropped);
|
|
gst_element_post_message (GST_ELEMENT_CAST (basesink), qos_msg);
|
|
}
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
/* with STREAM_LOCK
|
|
*/
|
|
static GstFlowReturn
|
|
gst_base_sink_chain_main (GstBaseSink * basesink, GstPad * pad, gpointer obj,
|
|
gboolean is_list)
|
|
{
|
|
GstFlowReturn result;
|
|
|
|
if (G_UNLIKELY (basesink->pad_mode != GST_PAD_MODE_PUSH))
|
|
goto wrong_mode;
|
|
|
|
GST_BASE_SINK_PREROLL_LOCK (basesink);
|
|
result = gst_base_sink_chain_unlocked (basesink, pad, obj, is_list);
|
|
GST_BASE_SINK_PREROLL_UNLOCK (basesink);
|
|
|
|
done:
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
wrong_mode:
|
|
{
|
|
GST_OBJECT_LOCK (pad);
|
|
GST_WARNING_OBJECT (basesink,
|
|
"Push on pad %s:%s, but it was not activated in push mode",
|
|
GST_DEBUG_PAD_NAME (pad));
|
|
GST_OBJECT_UNLOCK (pad);
|
|
gst_mini_object_unref (GST_MINI_OBJECT_CAST (obj));
|
|
/* we don't post an error message this will signal to the peer
|
|
* pushing that EOS is reached. */
|
|
result = GST_FLOW_EOS;
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_base_sink_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
|
|
{
|
|
GstBaseSink *basesink;
|
|
|
|
basesink = GST_BASE_SINK (parent);
|
|
|
|
return gst_base_sink_chain_main (basesink, pad, buf, FALSE);
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_base_sink_chain_list (GstPad * pad, GstObject * parent,
|
|
GstBufferList * list)
|
|
{
|
|
GstBaseSink *basesink;
|
|
GstBaseSinkClass *bclass;
|
|
GstFlowReturn result;
|
|
|
|
basesink = GST_BASE_SINK (parent);
|
|
bclass = GST_BASE_SINK_GET_CLASS (basesink);
|
|
|
|
if (G_LIKELY (bclass->render_list)) {
|
|
result = gst_base_sink_chain_main (basesink, pad, list, TRUE);
|
|
} else {
|
|
guint i, len;
|
|
GstBuffer *buffer;
|
|
|
|
GST_INFO_OBJECT (pad, "chaining each buffer in list");
|
|
|
|
len = gst_buffer_list_length (list);
|
|
|
|
result = GST_FLOW_OK;
|
|
for (i = 0; i < len; i++) {
|
|
buffer = gst_buffer_list_get (list, i);
|
|
result = gst_base_sink_chain_main (basesink, pad,
|
|
gst_buffer_ref (buffer), FALSE);
|
|
if (result != GST_FLOW_OK)
|
|
break;
|
|
}
|
|
gst_buffer_list_unref (list);
|
|
}
|
|
return result;
|
|
}
|
|
|
|
|
|
static gboolean
|
|
gst_base_sink_default_do_seek (GstBaseSink * sink, GstSegment * segment)
|
|
{
|
|
gboolean res = TRUE;
|
|
|
|
/* update our offset if the start/stop position was updated */
|
|
if (segment->format == GST_FORMAT_BYTES) {
|
|
segment->time = segment->start;
|
|
} else if (segment->start == 0) {
|
|
/* seek to start, we can implement a default for this. */
|
|
segment->time = 0;
|
|
} else {
|
|
res = FALSE;
|
|
GST_INFO_OBJECT (sink, "Can't do a default seek");
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
#define SEEK_TYPE_IS_RELATIVE(t) (((t) != GST_SEEK_TYPE_NONE) && ((t) != GST_SEEK_TYPE_SET))
|
|
|
|
static gboolean
|
|
gst_base_sink_default_prepare_seek_segment (GstBaseSink * sink,
|
|
GstEvent * event, GstSegment * segment)
|
|
{
|
|
/* By default, we try one of 2 things:
|
|
* - For absolute seek positions, convert the requested position to our
|
|
* configured processing format and place it in the output segment \
|
|
* - For relative seek positions, convert our current (input) values to the
|
|
* seek format, adjust by the relative seek offset and then convert back to
|
|
* the processing format
|
|
*/
|
|
GstSeekType start_type, stop_type;
|
|
gint64 start, stop;
|
|
GstSeekFlags flags;
|
|
GstFormat seek_format;
|
|
gdouble rate;
|
|
gboolean update;
|
|
gboolean res = TRUE;
|
|
|
|
gst_event_parse_seek (event, &rate, &seek_format, &flags,
|
|
&start_type, &start, &stop_type, &stop);
|
|
|
|
if (seek_format == segment->format) {
|
|
gst_segment_do_seek (segment, rate, seek_format, flags,
|
|
start_type, start, stop_type, stop, &update);
|
|
return TRUE;
|
|
}
|
|
|
|
if (start_type != GST_SEEK_TYPE_NONE) {
|
|
/* FIXME: Handle seek_end by converting the input segment vals */
|
|
res =
|
|
gst_pad_query_convert (sink->sinkpad, seek_format, start,
|
|
segment->format, &start);
|
|
start_type = GST_SEEK_TYPE_SET;
|
|
}
|
|
|
|
if (res && stop_type != GST_SEEK_TYPE_NONE) {
|
|
/* FIXME: Handle seek_end by converting the input segment vals */
|
|
res =
|
|
gst_pad_query_convert (sink->sinkpad, seek_format, stop,
|
|
segment->format, &stop);
|
|
stop_type = GST_SEEK_TYPE_SET;
|
|
}
|
|
|
|
/* And finally, configure our output segment in the desired format */
|
|
gst_segment_do_seek (segment, rate, segment->format, flags, start_type, start,
|
|
stop_type, stop, &update);
|
|
|
|
if (!res)
|
|
goto no_format;
|
|
|
|
return res;
|
|
|
|
no_format:
|
|
{
|
|
GST_DEBUG_OBJECT (sink, "undefined format given, seek aborted.");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/* perform a seek, only executed in pull mode */
|
|
static gboolean
|
|
gst_base_sink_perform_seek (GstBaseSink * sink, GstPad * pad, GstEvent * event)
|
|
{
|
|
gboolean flush;
|
|
gdouble rate;
|
|
GstFormat seek_format, dest_format;
|
|
GstSeekFlags flags;
|
|
GstSeekType start_type, stop_type;
|
|
gboolean seekseg_configured = FALSE;
|
|
gint64 start, stop;
|
|
gboolean update, res = TRUE;
|
|
GstSegment seeksegment;
|
|
|
|
dest_format = sink->segment.format;
|
|
|
|
if (event) {
|
|
GST_DEBUG_OBJECT (sink, "performing seek with event %p", event);
|
|
gst_event_parse_seek (event, &rate, &seek_format, &flags,
|
|
&start_type, &start, &stop_type, &stop);
|
|
|
|
flush = flags & GST_SEEK_FLAG_FLUSH;
|
|
} else {
|
|
GST_DEBUG_OBJECT (sink, "performing seek without event");
|
|
flush = FALSE;
|
|
}
|
|
|
|
if (flush) {
|
|
GST_DEBUG_OBJECT (sink, "flushing upstream");
|
|
gst_pad_push_event (pad, gst_event_new_flush_start ());
|
|
gst_base_sink_flush_start (sink, pad);
|
|
} else {
|
|
GST_DEBUG_OBJECT (sink, "pausing pulling thread");
|
|
}
|
|
|
|
GST_PAD_STREAM_LOCK (pad);
|
|
|
|
/* If we configured the seeksegment above, don't overwrite it now. Otherwise
|
|
* copy the current segment info into the temp segment that we can actually
|
|
* attempt the seek with. We only update the real segment if the seek succeeds. */
|
|
if (!seekseg_configured) {
|
|
memcpy (&seeksegment, &sink->segment, sizeof (GstSegment));
|
|
|
|
/* now configure the final seek segment */
|
|
if (event) {
|
|
if (sink->segment.format != seek_format) {
|
|
/* OK, here's where we give the subclass a chance to convert the relative
|
|
* seek into an absolute one in the processing format. We set up any
|
|
* absolute seek above, before taking the stream lock. */
|
|
if (!gst_base_sink_default_prepare_seek_segment (sink, event,
|
|
&seeksegment)) {
|
|
GST_DEBUG_OBJECT (sink,
|
|
"Preparing the seek failed after flushing. " "Aborting seek");
|
|
res = FALSE;
|
|
}
|
|
} else {
|
|
/* The seek format matches our processing format, no need to ask the
|
|
* the subclass to configure the segment. */
|
|
gst_segment_do_seek (&seeksegment, rate, seek_format, flags,
|
|
start_type, start, stop_type, stop, &update);
|
|
}
|
|
}
|
|
/* Else, no seek event passed, so we're just (re)starting the
|
|
current segment. */
|
|
}
|
|
|
|
if (res) {
|
|
GST_DEBUG_OBJECT (sink, "segment configured from %" G_GINT64_FORMAT
|
|
" to %" G_GINT64_FORMAT ", position %" G_GINT64_FORMAT,
|
|
seeksegment.start, seeksegment.stop, seeksegment.position);
|
|
|
|
/* do the seek, segment.position contains the new position. */
|
|
res = gst_base_sink_default_do_seek (sink, &seeksegment);
|
|
}
|
|
|
|
|
|
if (flush) {
|
|
GST_DEBUG_OBJECT (sink, "stop flushing upstream");
|
|
gst_pad_push_event (pad, gst_event_new_flush_stop (TRUE));
|
|
gst_base_sink_flush_stop (sink, pad, TRUE);
|
|
} else if (res && sink->running) {
|
|
/* we are running the current segment and doing a non-flushing seek,
|
|
* close the segment first based on the position. */
|
|
GST_DEBUG_OBJECT (sink, "closing running segment %" G_GINT64_FORMAT
|
|
" to %" G_GINT64_FORMAT, sink->segment.start, sink->segment.position);
|
|
}
|
|
|
|
/* The subclass must have converted the segment to the processing format
|
|
* by now */
|
|
if (res && seeksegment.format != dest_format) {
|
|
GST_DEBUG_OBJECT (sink, "Subclass failed to prepare a seek segment "
|
|
"in the correct format. Aborting seek.");
|
|
res = FALSE;
|
|
}
|
|
|
|
/* if successful seek, we update our real segment and push
|
|
* out the new segment. */
|
|
if (res) {
|
|
gst_segment_copy_into (&seeksegment, &sink->segment);
|
|
|
|
if (sink->segment.flags & GST_SEGMENT_FLAG_SEGMENT) {
|
|
gst_element_post_message (GST_ELEMENT (sink),
|
|
gst_message_new_segment_start (GST_OBJECT (sink),
|
|
sink->segment.format, sink->segment.position));
|
|
}
|
|
}
|
|
|
|
sink->priv->discont = TRUE;
|
|
sink->running = TRUE;
|
|
|
|
GST_PAD_STREAM_UNLOCK (pad);
|
|
|
|
return res;
|
|
}
|
|
|
|
static void
|
|
set_step_info (GstBaseSink * sink, GstStepInfo * current, GstStepInfo * pending,
|
|
guint seqnum, GstFormat format, guint64 amount, gdouble rate,
|
|
gboolean flush, gboolean intermediate)
|
|
{
|
|
GST_OBJECT_LOCK (sink);
|
|
pending->seqnum = seqnum;
|
|
pending->format = format;
|
|
pending->amount = amount;
|
|
pending->position = 0;
|
|
pending->rate = rate;
|
|
pending->flush = flush;
|
|
pending->intermediate = intermediate;
|
|
pending->valid = TRUE;
|
|
/* flush invalidates the current stepping segment */
|
|
if (flush)
|
|
current->valid = FALSE;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_sink_perform_step (GstBaseSink * sink, GstPad * pad, GstEvent * event)
|
|
{
|
|
GstBaseSinkPrivate *priv;
|
|
GstBaseSinkClass *bclass;
|
|
gboolean flush, intermediate;
|
|
gdouble rate;
|
|
GstFormat format;
|
|
guint64 amount;
|
|
guint seqnum;
|
|
GstStepInfo *pending, *current;
|
|
GstMessage *message;
|
|
|
|
bclass = GST_BASE_SINK_GET_CLASS (sink);
|
|
priv = sink->priv;
|
|
|
|
GST_DEBUG_OBJECT (sink, "performing step with event %p", event);
|
|
|
|
gst_event_parse_step (event, &format, &amount, &rate, &flush, &intermediate);
|
|
seqnum = gst_event_get_seqnum (event);
|
|
|
|
pending = &priv->pending_step;
|
|
current = &priv->current_step;
|
|
|
|
/* post message first */
|
|
message = gst_message_new_step_start (GST_OBJECT (sink), FALSE, format,
|
|
amount, rate, flush, intermediate);
|
|
gst_message_set_seqnum (message, seqnum);
|
|
gst_element_post_message (GST_ELEMENT (sink), message);
|
|
|
|
if (flush) {
|
|
/* we need to call ::unlock before locking PREROLL_LOCK
|
|
* since we lock it before going into ::render */
|
|
if (bclass->unlock)
|
|
bclass->unlock (sink);
|
|
|
|
GST_BASE_SINK_PREROLL_LOCK (sink);
|
|
/* now that we have the PREROLL lock, clear our unlock request */
|
|
if (bclass->unlock_stop)
|
|
bclass->unlock_stop (sink);
|
|
|
|
/* update the stepinfo and make it valid */
|
|
set_step_info (sink, current, pending, seqnum, format, amount, rate, flush,
|
|
intermediate);
|
|
|
|
if (sink->priv->async_enabled) {
|
|
/* and we need to commit our state again on the next
|
|
* prerolled buffer */
|
|
sink->playing_async = TRUE;
|
|
priv->pending_step.need_preroll = TRUE;
|
|
sink->need_preroll = FALSE;
|
|
gst_element_lost_state (GST_ELEMENT_CAST (sink));
|
|
} else {
|
|
sink->priv->have_latency = TRUE;
|
|
sink->need_preroll = FALSE;
|
|
}
|
|
priv->current_sstart = GST_CLOCK_TIME_NONE;
|
|
priv->current_sstop = GST_CLOCK_TIME_NONE;
|
|
priv->eos_rtime = GST_CLOCK_TIME_NONE;
|
|
priv->call_preroll = TRUE;
|
|
gst_base_sink_set_last_buffer (sink, NULL);
|
|
gst_base_sink_reset_qos (sink);
|
|
|
|
if (sink->clock_id) {
|
|
gst_clock_id_unschedule (sink->clock_id);
|
|
}
|
|
|
|
if (sink->have_preroll) {
|
|
GST_DEBUG_OBJECT (sink, "signal waiter");
|
|
priv->step_unlock = TRUE;
|
|
GST_BASE_SINK_PREROLL_SIGNAL (sink);
|
|
}
|
|
GST_BASE_SINK_PREROLL_UNLOCK (sink);
|
|
} else {
|
|
/* update the stepinfo and make it valid */
|
|
set_step_info (sink, current, pending, seqnum, format, amount, rate, flush,
|
|
intermediate);
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/* with STREAM_LOCK
|
|
*/
|
|
static void
|
|
gst_base_sink_loop (GstPad * pad)
|
|
{
|
|
GstObject *parent;
|
|
GstBaseSink *basesink;
|
|
GstBuffer *buf = NULL;
|
|
GstFlowReturn result;
|
|
guint blocksize;
|
|
guint64 offset;
|
|
|
|
parent = GST_OBJECT_PARENT (pad);
|
|
basesink = GST_BASE_SINK (parent);
|
|
|
|
g_assert (basesink->pad_mode == GST_PAD_MODE_PULL);
|
|
|
|
if ((blocksize = basesink->priv->blocksize) == 0)
|
|
blocksize = -1;
|
|
|
|
offset = basesink->segment.position;
|
|
|
|
GST_DEBUG_OBJECT (basesink, "pulling %" G_GUINT64_FORMAT ", %u",
|
|
offset, blocksize);
|
|
|
|
result = gst_pad_pull_range (pad, offset, blocksize, &buf);
|
|
if (G_UNLIKELY (result != GST_FLOW_OK))
|
|
goto paused;
|
|
|
|
if (G_UNLIKELY (buf == NULL))
|
|
goto no_buffer;
|
|
|
|
offset += gst_buffer_get_size (buf);
|
|
|
|
basesink->segment.position = offset;
|
|
|
|
GST_BASE_SINK_PREROLL_LOCK (basesink);
|
|
result = gst_base_sink_chain_unlocked (basesink, pad, buf, FALSE);
|
|
GST_BASE_SINK_PREROLL_UNLOCK (basesink);
|
|
if (G_UNLIKELY (result != GST_FLOW_OK))
|
|
goto paused;
|
|
|
|
return;
|
|
|
|
/* ERRORS */
|
|
paused:
|
|
{
|
|
GST_LOG_OBJECT (basesink, "pausing task, reason %s",
|
|
gst_flow_get_name (result));
|
|
gst_pad_pause_task (pad);
|
|
if (result == GST_FLOW_EOS) {
|
|
/* perform EOS logic */
|
|
if (basesink->segment.flags & GST_SEGMENT_FLAG_SEGMENT) {
|
|
gst_element_post_message (GST_ELEMENT_CAST (basesink),
|
|
gst_message_new_segment_done (GST_OBJECT_CAST (basesink),
|
|
basesink->segment.format, basesink->segment.position));
|
|
gst_base_sink_event (pad, parent,
|
|
gst_event_new_segment_done (basesink->segment.format,
|
|
basesink->segment.position));
|
|
} else {
|
|
gst_base_sink_event (pad, parent, gst_event_new_eos ());
|
|
}
|
|
} else if (result == GST_FLOW_NOT_LINKED || result <= GST_FLOW_EOS) {
|
|
/* for fatal errors we post an error message, post the error
|
|
* first so the app knows about the error first.
|
|
* wrong-state is not a fatal error because it happens due to
|
|
* flushing and posting an error message in that case is the
|
|
* wrong thing to do, e.g. when basesrc is doing a flushing
|
|
* seek. */
|
|
GST_ELEMENT_ERROR (basesink, STREAM, FAILED,
|
|
(_("Internal data stream error.")),
|
|
("stream stopped, reason %s", gst_flow_get_name (result)));
|
|
gst_base_sink_event (pad, parent, gst_event_new_eos ());
|
|
}
|
|
return;
|
|
}
|
|
no_buffer:
|
|
{
|
|
GST_LOG_OBJECT (basesink, "no buffer, pausing");
|
|
GST_ELEMENT_ERROR (basesink, STREAM, FAILED,
|
|
(_("Internal data flow error.")), ("element returned NULL buffer"));
|
|
result = GST_FLOW_ERROR;
|
|
goto paused;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_sink_set_flushing (GstBaseSink * basesink, GstPad * pad,
|
|
gboolean flushing)
|
|
{
|
|
GstBaseSinkClass *bclass;
|
|
|
|
bclass = GST_BASE_SINK_GET_CLASS (basesink);
|
|
|
|
if (flushing) {
|
|
/* unlock any subclasses, we need to do this before grabbing the
|
|
* PREROLL_LOCK since we hold this lock before going into ::render. */
|
|
if (bclass->unlock)
|
|
bclass->unlock (basesink);
|
|
}
|
|
|
|
GST_BASE_SINK_PREROLL_LOCK (basesink);
|
|
basesink->flushing = flushing;
|
|
if (flushing) {
|
|
/* step 1, now that we have the PREROLL lock, clear our unlock request */
|
|
if (bclass->unlock_stop)
|
|
bclass->unlock_stop (basesink);
|
|
|
|
/* set need_preroll before we unblock the clock. If the clock is unblocked
|
|
* before timing out, we can reuse the buffer for preroll. */
|
|
basesink->need_preroll = TRUE;
|
|
|
|
/* step 2, unblock clock sync (if any) or any other blocking thing */
|
|
if (basesink->clock_id) {
|
|
gst_clock_id_unschedule (basesink->clock_id);
|
|
}
|
|
|
|
/* flush out the data thread if it's locked in finish_preroll, this will
|
|
* also flush out the EOS state */
|
|
GST_DEBUG_OBJECT (basesink,
|
|
"flushing out data thread, need preroll to TRUE");
|
|
|
|
/* we can't have EOS anymore now */
|
|
basesink->eos = FALSE;
|
|
basesink->priv->received_eos = FALSE;
|
|
basesink->have_preroll = FALSE;
|
|
basesink->priv->step_unlock = FALSE;
|
|
/* can't report latency anymore until we preroll again */
|
|
if (basesink->priv->async_enabled) {
|
|
GST_OBJECT_LOCK (basesink);
|
|
basesink->priv->have_latency = FALSE;
|
|
GST_OBJECT_UNLOCK (basesink);
|
|
}
|
|
/* and signal any waiters now */
|
|
GST_BASE_SINK_PREROLL_SIGNAL (basesink);
|
|
}
|
|
GST_BASE_SINK_PREROLL_UNLOCK (basesink);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_sink_default_activate_pull (GstBaseSink * basesink, gboolean active)
|
|
{
|
|
gboolean result;
|
|
|
|
if (active) {
|
|
/* start task */
|
|
result = gst_pad_start_task (basesink->sinkpad,
|
|
(GstTaskFunction) gst_base_sink_loop, basesink->sinkpad, NULL);
|
|
} else {
|
|
/* step 2, make sure streaming finishes */
|
|
result = gst_pad_stop_task (basesink->sinkpad);
|
|
}
|
|
|
|
return result;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_sink_pad_activate (GstPad * pad, GstObject * parent)
|
|
{
|
|
gboolean result = FALSE;
|
|
GstBaseSink *basesink;
|
|
GstQuery *query;
|
|
gboolean pull_mode;
|
|
|
|
basesink = GST_BASE_SINK (parent);
|
|
|
|
GST_DEBUG_OBJECT (basesink, "Trying pull mode first");
|
|
|
|
gst_base_sink_set_flushing (basesink, pad, FALSE);
|
|
|
|
/* we need to have the pull mode enabled */
|
|
if (!basesink->can_activate_pull) {
|
|
GST_DEBUG_OBJECT (basesink, "pull mode disabled");
|
|
goto fallback;
|
|
}
|
|
|
|
/* check if downstreams supports pull mode at all */
|
|
query = gst_query_new_scheduling ();
|
|
|
|
if (!gst_pad_peer_query (pad, query)) {
|
|
gst_query_unref (query);
|
|
GST_DEBUG_OBJECT (basesink, "peer query faild, no pull mode");
|
|
goto fallback;
|
|
}
|
|
|
|
/* parse result of the query */
|
|
pull_mode = gst_query_has_scheduling_mode (query, GST_PAD_MODE_PULL);
|
|
gst_query_unref (query);
|
|
|
|
if (!pull_mode) {
|
|
GST_DEBUG_OBJECT (basesink, "pull mode not supported");
|
|
goto fallback;
|
|
}
|
|
|
|
/* set the pad mode before starting the task so that it's in the
|
|
* correct state for the new thread. also the sink set_caps and get_caps
|
|
* function checks this */
|
|
basesink->pad_mode = GST_PAD_MODE_PULL;
|
|
|
|
/* we first try to negotiate a format so that when we try to activate
|
|
* downstream, it knows about our format */
|
|
if (!gst_base_sink_negotiate_pull (basesink)) {
|
|
GST_DEBUG_OBJECT (basesink, "failed to negotiate in pull mode");
|
|
goto fallback;
|
|
}
|
|
|
|
/* ok activate now */
|
|
if (!gst_pad_activate_mode (pad, GST_PAD_MODE_PULL, TRUE)) {
|
|
/* clear any pending caps */
|
|
GST_OBJECT_LOCK (basesink);
|
|
gst_caps_replace (&basesink->priv->caps, NULL);
|
|
GST_OBJECT_UNLOCK (basesink);
|
|
GST_DEBUG_OBJECT (basesink, "failed to activate in pull mode");
|
|
goto fallback;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (basesink, "Success activating pull mode");
|
|
result = TRUE;
|
|
goto done;
|
|
|
|
/* push mode fallback */
|
|
fallback:
|
|
GST_DEBUG_OBJECT (basesink, "Falling back to push mode");
|
|
if ((result = gst_pad_activate_mode (pad, GST_PAD_MODE_PUSH, TRUE))) {
|
|
GST_DEBUG_OBJECT (basesink, "Success activating push mode");
|
|
}
|
|
|
|
done:
|
|
if (!result) {
|
|
GST_WARNING_OBJECT (basesink, "Could not activate pad in either mode");
|
|
gst_base_sink_set_flushing (basesink, pad, TRUE);
|
|
}
|
|
|
|
return result;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_sink_pad_activate_push (GstPad * pad, GstObject * parent,
|
|
gboolean active)
|
|
{
|
|
gboolean result;
|
|
GstBaseSink *basesink;
|
|
|
|
basesink = GST_BASE_SINK (parent);
|
|
|
|
if (active) {
|
|
if (!basesink->can_activate_push) {
|
|
result = FALSE;
|
|
basesink->pad_mode = GST_PAD_MODE_NONE;
|
|
} else {
|
|
result = TRUE;
|
|
basesink->pad_mode = GST_PAD_MODE_PUSH;
|
|
}
|
|
} else {
|
|
if (G_UNLIKELY (basesink->pad_mode != GST_PAD_MODE_PUSH)) {
|
|
g_warning ("Internal GStreamer activation error!!!");
|
|
result = FALSE;
|
|
} else {
|
|
gst_base_sink_set_flushing (basesink, pad, TRUE);
|
|
result = TRUE;
|
|
basesink->pad_mode = GST_PAD_MODE_NONE;
|
|
}
|
|
}
|
|
|
|
return result;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_sink_negotiate_pull (GstBaseSink * basesink)
|
|
{
|
|
GstCaps *caps;
|
|
gboolean result;
|
|
|
|
result = FALSE;
|
|
|
|
/* this returns the intersection between our caps and the peer caps. If there
|
|
* is no peer, it returns NULL and we can't operate in pull mode so we can
|
|
* fail the negotiation. */
|
|
caps = gst_pad_get_allowed_caps (GST_BASE_SINK_PAD (basesink));
|
|
if (caps == NULL || gst_caps_is_empty (caps))
|
|
goto no_caps_possible;
|
|
|
|
GST_DEBUG_OBJECT (basesink, "allowed caps: %" GST_PTR_FORMAT, caps);
|
|
|
|
if (gst_caps_is_any (caps)) {
|
|
GST_DEBUG_OBJECT (basesink, "caps were ANY after fixating, "
|
|
"allowing pull()");
|
|
/* neither side has template caps in this case, so they are prepared for
|
|
pull() without setcaps() */
|
|
result = TRUE;
|
|
} else {
|
|
/* try to fixate */
|
|
caps = gst_base_sink_fixate (basesink, caps);
|
|
GST_DEBUG_OBJECT (basesink, "fixated to: %" GST_PTR_FORMAT, caps);
|
|
|
|
if (gst_caps_is_fixed (caps)) {
|
|
if (!gst_pad_set_caps (GST_BASE_SINK_PAD (basesink), caps))
|
|
goto could_not_set_caps;
|
|
|
|
result = TRUE;
|
|
}
|
|
}
|
|
|
|
gst_caps_unref (caps);
|
|
|
|
return result;
|
|
|
|
no_caps_possible:
|
|
{
|
|
GST_INFO_OBJECT (basesink, "Pipeline could not agree on caps");
|
|
GST_DEBUG_OBJECT (basesink, "get_allowed_caps() returned EMPTY");
|
|
if (caps)
|
|
gst_caps_unref (caps);
|
|
return FALSE;
|
|
}
|
|
could_not_set_caps:
|
|
{
|
|
GST_INFO_OBJECT (basesink, "Could not set caps: %" GST_PTR_FORMAT, caps);
|
|
gst_caps_unref (caps);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/* this won't get called until we implement an activate function */
|
|
static gboolean
|
|
gst_base_sink_pad_activate_pull (GstPad * pad, GstObject * parent,
|
|
gboolean active)
|
|
{
|
|
gboolean result = FALSE;
|
|
GstBaseSink *basesink;
|
|
GstBaseSinkClass *bclass;
|
|
|
|
basesink = GST_BASE_SINK (parent);
|
|
bclass = GST_BASE_SINK_GET_CLASS (basesink);
|
|
|
|
if (active) {
|
|
gint64 duration;
|
|
|
|
/* we mark we have a newsegment here because pull based
|
|
* mode works just fine without having a newsegment before the
|
|
* first buffer */
|
|
gst_segment_init (&basesink->segment, GST_FORMAT_BYTES);
|
|
GST_OBJECT_LOCK (basesink);
|
|
basesink->have_newsegment = TRUE;
|
|
GST_OBJECT_UNLOCK (basesink);
|
|
|
|
/* get the peer duration in bytes */
|
|
result = gst_pad_peer_query_duration (pad, GST_FORMAT_BYTES, &duration);
|
|
if (result) {
|
|
GST_DEBUG_OBJECT (basesink,
|
|
"setting duration in bytes to %" G_GINT64_FORMAT, duration);
|
|
basesink->segment.duration = duration;
|
|
} else {
|
|
GST_DEBUG_OBJECT (basesink, "unknown duration");
|
|
}
|
|
|
|
if (bclass->activate_pull)
|
|
result = bclass->activate_pull (basesink, TRUE);
|
|
else
|
|
result = FALSE;
|
|
|
|
if (!result)
|
|
goto activate_failed;
|
|
|
|
} else {
|
|
if (G_UNLIKELY (basesink->pad_mode != GST_PAD_MODE_PULL)) {
|
|
g_warning ("Internal GStreamer activation error!!!");
|
|
result = FALSE;
|
|
} else {
|
|
result = gst_base_sink_set_flushing (basesink, pad, TRUE);
|
|
if (bclass->activate_pull)
|
|
result &= bclass->activate_pull (basesink, FALSE);
|
|
basesink->pad_mode = GST_PAD_MODE_NONE;
|
|
}
|
|
}
|
|
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
activate_failed:
|
|
{
|
|
/* reset, as starting the thread failed */
|
|
basesink->pad_mode = GST_PAD_MODE_NONE;
|
|
|
|
GST_ERROR_OBJECT (basesink, "subclass failed to activate in pull mode");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_sink_pad_activate_mode (GstPad * pad, GstObject * parent,
|
|
GstPadMode mode, gboolean active)
|
|
{
|
|
gboolean res;
|
|
|
|
switch (mode) {
|
|
case GST_PAD_MODE_PULL:
|
|
res = gst_base_sink_pad_activate_pull (pad, parent, active);
|
|
break;
|
|
case GST_PAD_MODE_PUSH:
|
|
res = gst_base_sink_pad_activate_push (pad, parent, active);
|
|
break;
|
|
default:
|
|
GST_LOG_OBJECT (pad, "unknown activation mode %d", mode);
|
|
res = FALSE;
|
|
break;
|
|
}
|
|
return res;
|
|
}
|
|
|
|
/* send an event to our sinkpad peer. */
|
|
static gboolean
|
|
gst_base_sink_send_event (GstElement * element, GstEvent * event)
|
|
{
|
|
GstPad *pad;
|
|
GstBaseSink *basesink = GST_BASE_SINK (element);
|
|
gboolean forward, result = TRUE;
|
|
GstPadMode mode;
|
|
|
|
GST_OBJECT_LOCK (element);
|
|
/* get the pad and the scheduling mode */
|
|
pad = gst_object_ref (basesink->sinkpad);
|
|
mode = basesink->pad_mode;
|
|
GST_OBJECT_UNLOCK (element);
|
|
|
|
/* only push UPSTREAM events upstream */
|
|
forward = GST_EVENT_IS_UPSTREAM (event);
|
|
|
|
GST_DEBUG_OBJECT (basesink, "handling event %p %" GST_PTR_FORMAT, event,
|
|
event);
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_LATENCY:
|
|
{
|
|
GstClockTime latency;
|
|
|
|
gst_event_parse_latency (event, &latency);
|
|
|
|
/* store the latency. We use this to adjust the running_time before syncing
|
|
* it to the clock. */
|
|
GST_OBJECT_LOCK (element);
|
|
basesink->priv->latency = latency;
|
|
if (!basesink->priv->have_latency)
|
|
forward = FALSE;
|
|
GST_OBJECT_UNLOCK (element);
|
|
GST_DEBUG_OBJECT (basesink, "latency set to %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (latency));
|
|
|
|
/* We forward this event so that all elements know about the global pipeline
|
|
* latency. This is interesting for an element when it wants to figure out
|
|
* when a particular piece of data will be rendered. */
|
|
break;
|
|
}
|
|
case GST_EVENT_SEEK:
|
|
/* in pull mode we will execute the seek */
|
|
if (mode == GST_PAD_MODE_PULL)
|
|
result = gst_base_sink_perform_seek (basesink, pad, event);
|
|
break;
|
|
case GST_EVENT_STEP:
|
|
result = gst_base_sink_perform_step (basesink, pad, event);
|
|
forward = FALSE;
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
if (forward) {
|
|
result = gst_pad_push_event (pad, event);
|
|
} else {
|
|
/* not forwarded, unref the event */
|
|
gst_event_unref (event);
|
|
}
|
|
|
|
gst_object_unref (pad);
|
|
|
|
GST_DEBUG_OBJECT (basesink, "handled event %p %" GST_PTR_FORMAT ": %d", event,
|
|
event, result);
|
|
|
|
return result;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_sink_get_position (GstBaseSink * basesink, GstFormat format,
|
|
gint64 * cur, gboolean * upstream)
|
|
{
|
|
GstClock *clock = NULL;
|
|
gboolean res = FALSE;
|
|
GstFormat oformat;
|
|
GstSegment *segment;
|
|
GstClockTime now, latency;
|
|
GstClockTimeDiff base_time;
|
|
gint64 time, base, duration;
|
|
gdouble rate;
|
|
gint64 last;
|
|
gboolean last_seen, with_clock, in_paused;
|
|
|
|
GST_OBJECT_LOCK (basesink);
|
|
/* we can only get the segment when we are not NULL or READY */
|
|
if (!basesink->have_newsegment)
|
|
goto wrong_state;
|
|
|
|
in_paused = FALSE;
|
|
/* when not in PLAYING or when we're busy with a state change, we
|
|
* cannot read from the clock so we report time based on the
|
|
* last seen timestamp. */
|
|
if (GST_STATE (basesink) != GST_STATE_PLAYING ||
|
|
GST_STATE_PENDING (basesink) != GST_STATE_VOID_PENDING) {
|
|
in_paused = TRUE;
|
|
}
|
|
|
|
segment = &basesink->segment;
|
|
|
|
/* get the format in the segment */
|
|
oformat = segment->format;
|
|
|
|
/* report with last seen position when EOS */
|
|
last_seen = basesink->eos;
|
|
|
|
/* assume we will use the clock for getting the current position */
|
|
with_clock = TRUE;
|
|
if (basesink->sync == FALSE)
|
|
with_clock = FALSE;
|
|
|
|
/* and we need a clock */
|
|
if (G_UNLIKELY ((clock = GST_ELEMENT_CLOCK (basesink)) == NULL))
|
|
with_clock = FALSE;
|
|
else
|
|
gst_object_ref (clock);
|
|
|
|
/* mainloop might be querying position when going to playing async,
|
|
* while (audio) rendering might be quickly advancing stream position,
|
|
* so use clock asap rather than last reported position */
|
|
if (in_paused && with_clock && g_atomic_int_get (&basesink->priv->to_playing)) {
|
|
GST_DEBUG_OBJECT (basesink, "going to PLAYING, so not PAUSED");
|
|
in_paused = FALSE;
|
|
}
|
|
|
|
/* collect all data we need holding the lock */
|
|
if (GST_CLOCK_TIME_IS_VALID (segment->time))
|
|
time = segment->time;
|
|
else
|
|
time = 0;
|
|
|
|
if (GST_CLOCK_TIME_IS_VALID (segment->stop))
|
|
duration = segment->stop - segment->start;
|
|
else
|
|
duration = 0;
|
|
|
|
base = segment->base;
|
|
rate = segment->rate * segment->applied_rate;
|
|
latency = basesink->priv->latency;
|
|
|
|
if (oformat == GST_FORMAT_TIME) {
|
|
gint64 start, stop;
|
|
|
|
start = basesink->priv->current_sstart;
|
|
stop = basesink->priv->current_sstop;
|
|
|
|
if (in_paused) {
|
|
/* in paused we use the last position as a lower bound */
|
|
if (stop == -1 || segment->rate > 0.0)
|
|
last = start;
|
|
else
|
|
last = stop;
|
|
|
|
GST_DEBUG_OBJECT (basesink, "in PAUSED using last %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (last));
|
|
} else {
|
|
/* in playing, use last stop time as upper bound */
|
|
if (start == -1 || segment->rate > 0.0)
|
|
last = stop;
|
|
else
|
|
last = start;
|
|
|
|
GST_DEBUG_OBJECT (basesink, "in PLAYING using last %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (last));
|
|
}
|
|
} else {
|
|
/* convert last stop to stream time */
|
|
last = gst_segment_to_stream_time (segment, oformat, segment->position);
|
|
|
|
GST_DEBUG_OBJECT (basesink, "in using last %" G_GINT64_FORMAT, last);
|
|
}
|
|
|
|
if (in_paused) {
|
|
/* in paused, use start_time */
|
|
base_time = GST_ELEMENT_START_TIME (basesink);
|
|
GST_DEBUG_OBJECT (basesink, "in paused, using start time %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (base_time));
|
|
} else if (with_clock) {
|
|
/* else use clock when needed */
|
|
base_time = GST_ELEMENT_CAST (basesink)->base_time;
|
|
GST_DEBUG_OBJECT (basesink, "using clock and base time %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (base_time));
|
|
} else {
|
|
/* else, no sync or clock -> no base time */
|
|
GST_DEBUG_OBJECT (basesink, "no sync or no clock");
|
|
base_time = -1;
|
|
}
|
|
|
|
/* no base_time, we can't calculate running_time, use last seem timestamp to report
|
|
* time */
|
|
if (base_time == -1)
|
|
last_seen = TRUE;
|
|
|
|
/* need to release the object lock before we can get the time,
|
|
* a clock might take the LOCK of the provider, which could be
|
|
* a basesink subclass. */
|
|
GST_OBJECT_UNLOCK (basesink);
|
|
|
|
if (last_seen) {
|
|
/* in EOS or when no valid stream_time, report the value of last seen
|
|
* timestamp */
|
|
if (last == -1) {
|
|
/* no timestamp, we need to ask upstream */
|
|
GST_DEBUG_OBJECT (basesink, "no last seen timestamp, asking upstream");
|
|
res = FALSE;
|
|
*upstream = TRUE;
|
|
goto done;
|
|
}
|
|
GST_DEBUG_OBJECT (basesink, "using last seen timestamp %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (last));
|
|
*cur = last;
|
|
} else {
|
|
if (oformat != GST_FORMAT_TIME) {
|
|
/* convert base, time and duration to time */
|
|
if (!gst_pad_query_convert (basesink->sinkpad, oformat, base,
|
|
GST_FORMAT_TIME, &base))
|
|
goto convert_failed;
|
|
if (!gst_pad_query_convert (basesink->sinkpad, oformat, duration,
|
|
GST_FORMAT_TIME, &duration))
|
|
goto convert_failed;
|
|
if (!gst_pad_query_convert (basesink->sinkpad, oformat, time,
|
|
GST_FORMAT_TIME, &time))
|
|
goto convert_failed;
|
|
if (!gst_pad_query_convert (basesink->sinkpad, oformat, last,
|
|
GST_FORMAT_TIME, &last))
|
|
goto convert_failed;
|
|
|
|
/* assume time format from now on */
|
|
oformat = GST_FORMAT_TIME;
|
|
}
|
|
|
|
if (!in_paused && with_clock) {
|
|
now = gst_clock_get_time (clock);
|
|
} else {
|
|
now = base_time;
|
|
base_time = 0;
|
|
}
|
|
|
|
/* subtract base time and base time from the clock time.
|
|
* Make sure we don't go negative. This is the current time in
|
|
* the segment which we need to scale with the combined
|
|
* rate and applied rate. */
|
|
base_time += base;
|
|
base_time += latency;
|
|
if (GST_CLOCK_DIFF (base_time, now) < 0)
|
|
base_time = now;
|
|
|
|
/* for negative rates we need to count back from the segment
|
|
* duration. */
|
|
if (rate < 0.0)
|
|
time += duration;
|
|
|
|
*cur = time + gst_guint64_to_gdouble (now - base_time) * rate;
|
|
|
|
if (in_paused) {
|
|
/* never report less than segment values in paused */
|
|
if (last != -1)
|
|
*cur = MAX (last, *cur);
|
|
} else {
|
|
/* never report more than last seen position in playing */
|
|
if (last != -1)
|
|
*cur = MIN (last, *cur);
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (basesink,
|
|
"now %" GST_TIME_FORMAT " - base_time %" GST_TIME_FORMAT " - base %"
|
|
GST_TIME_FORMAT " + time %" GST_TIME_FORMAT " last %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (now), GST_TIME_ARGS (base_time), GST_TIME_ARGS (base),
|
|
GST_TIME_ARGS (time), GST_TIME_ARGS (last));
|
|
}
|
|
|
|
if (oformat != format) {
|
|
/* convert to final format */
|
|
if (!gst_pad_query_convert (basesink->sinkpad, oformat, *cur, format, cur))
|
|
goto convert_failed;
|
|
}
|
|
|
|
res = TRUE;
|
|
|
|
done:
|
|
GST_DEBUG_OBJECT (basesink, "res: %d, POSITION: %" GST_TIME_FORMAT,
|
|
res, GST_TIME_ARGS (*cur));
|
|
|
|
if (clock)
|
|
gst_object_unref (clock);
|
|
|
|
return res;
|
|
|
|
/* special cases */
|
|
wrong_state:
|
|
{
|
|
/* in NULL or READY we always return FALSE and -1 */
|
|
GST_DEBUG_OBJECT (basesink, "position in wrong state, return -1");
|
|
res = FALSE;
|
|
*cur = -1;
|
|
GST_OBJECT_UNLOCK (basesink);
|
|
goto done;
|
|
}
|
|
convert_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "convert failed, try upstream");
|
|
*upstream = TRUE;
|
|
res = FALSE;
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_sink_get_duration (GstBaseSink * basesink, GstFormat format,
|
|
gint64 * dur, gboolean * upstream)
|
|
{
|
|
gboolean res = FALSE;
|
|
|
|
if (basesink->pad_mode == GST_PAD_MODE_PULL) {
|
|
gint64 uduration;
|
|
|
|
/* get the duration in bytes, in pull mode that's all we are sure to
|
|
* know. We have to explicitly get this value from upstream instead of
|
|
* using our cached value because it might change. Duration caching
|
|
* should be done at a higher level. */
|
|
res =
|
|
gst_pad_peer_query_duration (basesink->sinkpad, GST_FORMAT_BYTES,
|
|
&uduration);
|
|
if (res) {
|
|
basesink->segment.duration = uduration;
|
|
if (format != GST_FORMAT_BYTES) {
|
|
/* convert to the requested format */
|
|
res =
|
|
gst_pad_query_convert (basesink->sinkpad, GST_FORMAT_BYTES,
|
|
uduration, format, dur);
|
|
} else {
|
|
*dur = uduration;
|
|
}
|
|
}
|
|
*upstream = FALSE;
|
|
} else {
|
|
*upstream = TRUE;
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
static gboolean
|
|
default_element_query (GstElement * element, GstQuery * query)
|
|
{
|
|
gboolean res = FALSE;
|
|
|
|
GstBaseSink *basesink = GST_BASE_SINK (element);
|
|
|
|
switch (GST_QUERY_TYPE (query)) {
|
|
case GST_QUERY_POSITION:
|
|
{
|
|
gint64 cur = 0;
|
|
GstFormat format;
|
|
gboolean upstream = FALSE;
|
|
|
|
gst_query_parse_position (query, &format, NULL);
|
|
|
|
GST_DEBUG_OBJECT (basesink, "position query in format %s",
|
|
gst_format_get_name (format));
|
|
|
|
/* first try to get the position based on the clock */
|
|
if ((res =
|
|
gst_base_sink_get_position (basesink, format, &cur, &upstream))) {
|
|
gst_query_set_position (query, format, cur);
|
|
} else if (upstream) {
|
|
/* fallback to peer query */
|
|
res = gst_pad_peer_query (basesink->sinkpad, query);
|
|
}
|
|
if (!res) {
|
|
/* we can handle a few things if upstream failed */
|
|
if (format == GST_FORMAT_PERCENT) {
|
|
gint64 dur = 0;
|
|
|
|
res = gst_base_sink_get_position (basesink, GST_FORMAT_TIME, &cur,
|
|
&upstream);
|
|
if (!res && upstream) {
|
|
res =
|
|
gst_pad_peer_query_position (basesink->sinkpad, GST_FORMAT_TIME,
|
|
&cur);
|
|
}
|
|
if (res) {
|
|
res = gst_base_sink_get_duration (basesink, GST_FORMAT_TIME, &dur,
|
|
&upstream);
|
|
if (!res && upstream) {
|
|
res =
|
|
gst_pad_peer_query_duration (basesink->sinkpad,
|
|
GST_FORMAT_TIME, &dur);
|
|
}
|
|
}
|
|
if (res) {
|
|
gint64 pos;
|
|
|
|
pos = gst_util_uint64_scale (100 * GST_FORMAT_PERCENT_SCALE, cur,
|
|
dur);
|
|
gst_query_set_position (query, GST_FORMAT_PERCENT, pos);
|
|
}
|
|
}
|
|
}
|
|
break;
|
|
}
|
|
case GST_QUERY_DURATION:
|
|
{
|
|
gint64 dur = 0;
|
|
GstFormat format;
|
|
gboolean upstream = FALSE;
|
|
|
|
gst_query_parse_duration (query, &format, NULL);
|
|
|
|
GST_DEBUG_OBJECT (basesink, "duration query in format %s",
|
|
gst_format_get_name (format));
|
|
|
|
if ((res =
|
|
gst_base_sink_get_duration (basesink, format, &dur, &upstream))) {
|
|
gst_query_set_duration (query, format, dur);
|
|
} else if (upstream) {
|
|
/* fallback to peer query */
|
|
res = gst_pad_peer_query (basesink->sinkpad, query);
|
|
}
|
|
if (!res) {
|
|
/* we can handle a few things if upstream failed */
|
|
if (format == GST_FORMAT_PERCENT) {
|
|
gst_query_set_duration (query, GST_FORMAT_PERCENT,
|
|
GST_FORMAT_PERCENT_MAX);
|
|
res = TRUE;
|
|
}
|
|
}
|
|
break;
|
|
}
|
|
case GST_QUERY_LATENCY:
|
|
{
|
|
gboolean live, us_live;
|
|
GstClockTime min, max;
|
|
|
|
if ((res = gst_base_sink_query_latency (basesink, &live, &us_live, &min,
|
|
&max))) {
|
|
gst_query_set_latency (query, live, min, max);
|
|
}
|
|
break;
|
|
}
|
|
case GST_QUERY_JITTER:
|
|
break;
|
|
case GST_QUERY_RATE:
|
|
/* gst_query_set_rate (query, basesink->segment_rate); */
|
|
res = TRUE;
|
|
break;
|
|
case GST_QUERY_SEGMENT:
|
|
{
|
|
if (basesink->pad_mode == GST_PAD_MODE_PULL) {
|
|
gst_query_set_segment (query, basesink->segment.rate,
|
|
GST_FORMAT_TIME, basesink->segment.start, basesink->segment.stop);
|
|
res = TRUE;
|
|
} else {
|
|
res = gst_pad_peer_query (basesink->sinkpad, query);
|
|
}
|
|
break;
|
|
}
|
|
case GST_QUERY_SEEKING:
|
|
case GST_QUERY_CONVERT:
|
|
case GST_QUERY_FORMATS:
|
|
default:
|
|
res = gst_pad_peer_query (basesink->sinkpad, query);
|
|
break;
|
|
}
|
|
GST_DEBUG_OBJECT (basesink, "query %s returns %d",
|
|
GST_QUERY_TYPE_NAME (query), res);
|
|
return res;
|
|
}
|
|
|
|
|
|
static gboolean
|
|
gst_base_sink_default_query (GstBaseSink * basesink, GstQuery * query)
|
|
{
|
|
gboolean res;
|
|
GstBaseSinkClass *bclass;
|
|
|
|
bclass = GST_BASE_SINK_GET_CLASS (basesink);
|
|
|
|
switch (GST_QUERY_TYPE (query)) {
|
|
case GST_QUERY_ALLOCATION:
|
|
{
|
|
if (bclass->propose_allocation)
|
|
res = bclass->propose_allocation (basesink, query);
|
|
else
|
|
res = FALSE;
|
|
break;
|
|
}
|
|
case GST_QUERY_CAPS:
|
|
{
|
|
GstCaps *caps, *filter;
|
|
|
|
gst_query_parse_caps (query, &filter);
|
|
caps = gst_base_sink_query_caps (basesink, basesink->sinkpad, filter);
|
|
gst_query_set_caps_result (query, caps);
|
|
gst_caps_unref (caps);
|
|
res = TRUE;
|
|
break;
|
|
}
|
|
case GST_QUERY_ACCEPT_CAPS:
|
|
{
|
|
GstCaps *caps, *allowed;
|
|
gboolean subset;
|
|
|
|
/* slightly faster than the default implementation */
|
|
gst_query_parse_accept_caps (query, &caps);
|
|
allowed = gst_base_sink_query_caps (basesink, basesink->sinkpad, NULL);
|
|
subset = gst_caps_is_subset (caps, allowed);
|
|
gst_caps_unref (allowed);
|
|
gst_query_set_accept_caps_result (query, subset);
|
|
res = TRUE;
|
|
break;
|
|
}
|
|
case GST_QUERY_DRAIN:
|
|
res = TRUE;
|
|
break;
|
|
default:
|
|
res =
|
|
gst_pad_query_default (basesink->sinkpad, GST_OBJECT_CAST (basesink),
|
|
query);
|
|
break;
|
|
}
|
|
return res;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_sink_sink_query (GstPad * pad, GstObject * parent, GstQuery * query)
|
|
{
|
|
GstBaseSink *basesink;
|
|
GstBaseSinkClass *bclass;
|
|
gboolean res;
|
|
|
|
basesink = GST_BASE_SINK_CAST (parent);
|
|
bclass = GST_BASE_SINK_GET_CLASS (basesink);
|
|
|
|
if (bclass->query)
|
|
res = bclass->query (basesink, query);
|
|
else
|
|
res = FALSE;
|
|
|
|
return res;
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_base_sink_change_state (GstElement * element, GstStateChange transition)
|
|
{
|
|
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
|
|
GstBaseSink *basesink = GST_BASE_SINK (element);
|
|
GstBaseSinkClass *bclass;
|
|
GstBaseSinkPrivate *priv;
|
|
|
|
priv = basesink->priv;
|
|
|
|
bclass = GST_BASE_SINK_GET_CLASS (basesink);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_NULL_TO_READY:
|
|
if (bclass->start)
|
|
if (!bclass->start (basesink))
|
|
goto start_failed;
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
/* need to complete preroll before this state change completes, there
|
|
* is no data flow in READY so we can safely assume we need to preroll. */
|
|
GST_BASE_SINK_PREROLL_LOCK (basesink);
|
|
GST_DEBUG_OBJECT (basesink, "READY to PAUSED");
|
|
basesink->have_newsegment = FALSE;
|
|
gst_segment_init (&basesink->segment, GST_FORMAT_UNDEFINED);
|
|
basesink->offset = 0;
|
|
basesink->have_preroll = FALSE;
|
|
priv->step_unlock = FALSE;
|
|
basesink->need_preroll = TRUE;
|
|
basesink->playing_async = TRUE;
|
|
priv->current_sstart = GST_CLOCK_TIME_NONE;
|
|
priv->current_sstop = GST_CLOCK_TIME_NONE;
|
|
priv->eos_rtime = GST_CLOCK_TIME_NONE;
|
|
priv->latency = 0;
|
|
basesink->eos = FALSE;
|
|
priv->received_eos = FALSE;
|
|
gst_base_sink_reset_qos (basesink);
|
|
priv->rc_next = -1;
|
|
priv->commited = FALSE;
|
|
priv->call_preroll = TRUE;
|
|
priv->current_step.valid = FALSE;
|
|
priv->pending_step.valid = FALSE;
|
|
if (priv->async_enabled) {
|
|
GST_DEBUG_OBJECT (basesink, "doing async state change");
|
|
/* when async enabled, post async-start message and return ASYNC from
|
|
* the state change function */
|
|
ret = GST_STATE_CHANGE_ASYNC;
|
|
gst_element_post_message (GST_ELEMENT_CAST (basesink),
|
|
gst_message_new_async_start (GST_OBJECT_CAST (basesink)));
|
|
} else {
|
|
priv->have_latency = TRUE;
|
|
}
|
|
GST_BASE_SINK_PREROLL_UNLOCK (basesink);
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
|
GST_BASE_SINK_PREROLL_LOCK (basesink);
|
|
g_atomic_int_set (&basesink->priv->to_playing, TRUE);
|
|
if (!gst_base_sink_needs_preroll (basesink)) {
|
|
GST_DEBUG_OBJECT (basesink, "PAUSED to PLAYING, don't need preroll");
|
|
/* no preroll needed anymore now. */
|
|
basesink->playing_async = FALSE;
|
|
basesink->need_preroll = FALSE;
|
|
if (basesink->eos) {
|
|
GstMessage *message;
|
|
|
|
/* need to post EOS message here */
|
|
GST_DEBUG_OBJECT (basesink, "Now posting EOS");
|
|
message = gst_message_new_eos (GST_OBJECT_CAST (basesink));
|
|
gst_message_set_seqnum (message, basesink->priv->seqnum);
|
|
gst_element_post_message (GST_ELEMENT_CAST (basesink), message);
|
|
} else {
|
|
GST_DEBUG_OBJECT (basesink, "signal preroll");
|
|
GST_BASE_SINK_PREROLL_SIGNAL (basesink);
|
|
}
|
|
} else {
|
|
GST_DEBUG_OBJECT (basesink, "PAUSED to PLAYING, we are not prerolled");
|
|
basesink->need_preroll = TRUE;
|
|
basesink->playing_async = TRUE;
|
|
priv->call_preroll = TRUE;
|
|
priv->commited = FALSE;
|
|
if (priv->async_enabled) {
|
|
GST_DEBUG_OBJECT (basesink, "doing async state change");
|
|
ret = GST_STATE_CHANGE_ASYNC;
|
|
gst_element_post_message (GST_ELEMENT_CAST (basesink),
|
|
gst_message_new_async_start (GST_OBJECT_CAST (basesink)));
|
|
}
|
|
}
|
|
GST_BASE_SINK_PREROLL_UNLOCK (basesink);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
{
|
|
GstStateChangeReturn bret;
|
|
|
|
bret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
if (G_UNLIKELY (bret == GST_STATE_CHANGE_FAILURE))
|
|
goto activate_failed;
|
|
}
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
|
/* completed transition, so need not be marked any longer
|
|
* And it should be unmarked, since e.g. losing our position upon flush
|
|
* does not really change state to PAUSED ... */
|
|
g_atomic_int_set (&basesink->priv->to_playing, FALSE);
|
|
break;
|
|
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
|
g_atomic_int_set (&basesink->priv->to_playing, FALSE);
|
|
GST_DEBUG_OBJECT (basesink, "PLAYING to PAUSED");
|
|
/* FIXME, make sure we cannot enter _render first */
|
|
|
|
/* we need to call ::unlock before locking PREROLL_LOCK
|
|
* since we lock it before going into ::render */
|
|
if (bclass->unlock)
|
|
bclass->unlock (basesink);
|
|
|
|
GST_BASE_SINK_PREROLL_LOCK (basesink);
|
|
GST_DEBUG_OBJECT (basesink, "got preroll lock");
|
|
/* now that we have the PREROLL lock, clear our unlock request */
|
|
if (bclass->unlock_stop)
|
|
bclass->unlock_stop (basesink);
|
|
|
|
/* we need preroll again and we set the flag before unlocking the clockid
|
|
* because if the clockid is unlocked before a current buffer expired, we
|
|
* can use that buffer to preroll with */
|
|
basesink->need_preroll = TRUE;
|
|
|
|
if (basesink->clock_id) {
|
|
GST_DEBUG_OBJECT (basesink, "unschedule clock");
|
|
gst_clock_id_unschedule (basesink->clock_id);
|
|
}
|
|
|
|
/* if we don't have a preroll buffer we need to wait for a preroll and
|
|
* return ASYNC. */
|
|
if (!gst_base_sink_needs_preroll (basesink)) {
|
|
GST_DEBUG_OBJECT (basesink, "PLAYING to PAUSED, we are prerolled");
|
|
basesink->playing_async = FALSE;
|
|
} else {
|
|
if (GST_STATE_TARGET (GST_ELEMENT (basesink)) <= GST_STATE_READY) {
|
|
GST_DEBUG_OBJECT (basesink, "element is <= READY");
|
|
ret = GST_STATE_CHANGE_SUCCESS;
|
|
} else {
|
|
GST_DEBUG_OBJECT (basesink,
|
|
"PLAYING to PAUSED, we are not prerolled");
|
|
basesink->playing_async = TRUE;
|
|
priv->commited = FALSE;
|
|
priv->call_preroll = TRUE;
|
|
if (priv->async_enabled) {
|
|
GST_DEBUG_OBJECT (basesink, "doing async state change");
|
|
ret = GST_STATE_CHANGE_ASYNC;
|
|
gst_element_post_message (GST_ELEMENT_CAST (basesink),
|
|
gst_message_new_async_start (GST_OBJECT_CAST (basesink)));
|
|
}
|
|
}
|
|
}
|
|
GST_DEBUG_OBJECT (basesink, "rendered: %" G_GUINT64_FORMAT
|
|
", dropped: %" G_GUINT64_FORMAT, priv->rendered, priv->dropped);
|
|
|
|
gst_base_sink_reset_qos (basesink);
|
|
GST_BASE_SINK_PREROLL_UNLOCK (basesink);
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
GST_BASE_SINK_PREROLL_LOCK (basesink);
|
|
/* start by resetting our position state with the object lock so that the
|
|
* position query gets the right idea. We do this before we post the
|
|
* messages so that the message handlers pick this up. */
|
|
GST_OBJECT_LOCK (basesink);
|
|
basesink->have_newsegment = FALSE;
|
|
priv->current_sstart = GST_CLOCK_TIME_NONE;
|
|
priv->current_sstop = GST_CLOCK_TIME_NONE;
|
|
priv->have_latency = FALSE;
|
|
if (priv->cached_clock_id) {
|
|
gst_clock_id_unref (priv->cached_clock_id);
|
|
priv->cached_clock_id = NULL;
|
|
}
|
|
gst_caps_replace (&basesink->priv->caps, NULL);
|
|
GST_OBJECT_UNLOCK (basesink);
|
|
|
|
gst_base_sink_set_last_buffer (basesink, NULL);
|
|
priv->call_preroll = FALSE;
|
|
|
|
if (!priv->commited) {
|
|
if (priv->async_enabled) {
|
|
GST_DEBUG_OBJECT (basesink, "PAUSED to READY, posting async-done");
|
|
|
|
gst_element_post_message (GST_ELEMENT_CAST (basesink),
|
|
gst_message_new_state_changed (GST_OBJECT_CAST (basesink),
|
|
GST_STATE_PLAYING, GST_STATE_PAUSED, GST_STATE_READY));
|
|
|
|
gst_element_post_message (GST_ELEMENT_CAST (basesink),
|
|
gst_message_new_async_done (GST_OBJECT_CAST (basesink),
|
|
GST_CLOCK_TIME_NONE));
|
|
}
|
|
priv->commited = TRUE;
|
|
} else {
|
|
GST_DEBUG_OBJECT (basesink, "PAUSED to READY, don't need_preroll");
|
|
}
|
|
GST_BASE_SINK_PREROLL_UNLOCK (basesink);
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
if (bclass->stop) {
|
|
if (!bclass->stop (basesink)) {
|
|
GST_WARNING_OBJECT (basesink, "failed to stop");
|
|
}
|
|
}
|
|
gst_base_sink_set_last_buffer (basesink, NULL);
|
|
priv->call_preroll = FALSE;
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
start_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "failed to start");
|
|
return GST_STATE_CHANGE_FAILURE;
|
|
}
|
|
activate_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink,
|
|
"element failed to change states -- activation problem?");
|
|
return GST_STATE_CHANGE_FAILURE;
|
|
}
|
|
}
|