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b0346dff44
Original commit message from CVS: * docs/pwg/advanced-dparams.xml: * docs/pwg/building-props.xml: * docs/pwg/other-source.xml: * gst/glib-compat.h: * gst/gstbin.c: (gst_bin_class_init): * gst/gstclock.c: (gst_clock_class_init): * gst/gstindex.c: (gst_index_class_init): * gst/gstobject.c: (gst_object_class_init): * gst/gstpad.c: (gst_pad_class_init): * gst/gstpipeline.c: (gst_pipeline_class_init): * libs/gst/base/gstbasesink.c: (gst_base_sink_class_init): * libs/gst/base/gstbasesrc.c: (gst_base_src_class_init): * libs/gst/base/gstbasetransform.c: (gst_base_transform_class_init): * libs/gst/base/gstdataqueue.c: (gst_data_queue_class_init): * libs/gst/check/gstcheck.c: (_gst_check_fault_handler_restore), (_gst_check_fault_handler_sighandler), (_gst_check_fault_handler_setup), (gst_check_init): * libs/gst/controller/gstcontroller.c: (_gst_controller_class_init): * libs/gst/controller/gstlfocontrolsource.c: (gst_lfo_control_source_class_init): * libs/gst/net/gstnetclientclock.c: (gst_net_client_clock_class_init): * libs/gst/net/gstnettimeprovider.c: (gst_net_time_provider_class_init): * plugins/elements/gstcapsfilter.c: (gst_capsfilter_class_init): * plugins/elements/gstfakesink.c: (gst_fake_sink_class_init): * plugins/elements/gstfakesrc.c: (gst_fake_src_class_init): * plugins/elements/gstfdsink.c: (gst_fd_sink_class_init): * plugins/elements/gstfdsrc.c: (gst_fd_src_class_init): * plugins/elements/gstfilesink.c: (gst_file_sink_class_init): * plugins/elements/gstfilesrc.c: (gst_file_src_class_init): * plugins/elements/gstidentity.c: (gst_identity_class_init): * plugins/elements/gstmultiqueue.c: (gst_multi_queue_class_init): * plugins/elements/gstqueue.c: (gst_queue_class_init): * plugins/elements/gsttee.c: (gst_tee_class_init): * plugins/elements/gsttypefindelement.c: (gst_type_find_element_class_init): * plugins/indexers/gstfileindex.c: (gst_file_index_class_init): Define G_PARAM_STATIC_STRINGS if it's undefined (GLib < 2.13.0) and use it everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory usage, fewer allocations and thus less memory defragmentation. Fixes bug #523806.
3444 lines
101 KiB
C
3444 lines
101 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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, 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, #GstBaseSource
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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 base_init function, like so:
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* <programlisting>
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* static void
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* my_element_base_init (gpointer g_class)
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* {
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* GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class);
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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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* // see #GstElementDetails
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* gst_element_class_set_details (gstelement_class, &details);
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* }
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* </programlisting>
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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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* #GstBaseSink::preroll vmethod with this preroll buffer and will then commit
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* 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 ::get_times. If this function returns
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* #GST_CLOCK_TIME_NONE for the start time, no synchronisation will be done.
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* Synchronisation can be disabled entirely by setting the object "sync"
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* property to %FALSE.
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*
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* After synchronisation the virtual method #GstBaseSink::render will be called.
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* Subclasses should minimally implement this method.
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*
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* Since 0.10.3 subclasses that synchronise on the clock in the ::render method
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* are supported as well. These classes typically receive a buffer in the render
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* method and can then potentially block on the clock while rendering. A typical
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* example is an audiosink. Since 0.10.11 these subclasses can use
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* gst_base_sink_wait_preroll() to perform the 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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* ::get_times call before posting an EOS message. When the element receives
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* EOS in PAUSED, preroll completes, the event is queued and an EOS message is
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* posted when going to PLAYING.
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*
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* #GstBaseSink will internally use the #GST_EVENT_NEWSEGMENT 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 ::set_caps function will be called when the subclass should configure
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* itself to process a specific media type.
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*
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* The ::start and ::stop virtual methods will be called when resources should
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* be allocated. Any ::preroll, ::render and ::set_caps function will be
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* called between the ::start and ::stop calls.
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*
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* The ::event virtual method will be called when an event is received by
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* #GstBaseSink. Normally this method should only be overriden by very specific
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* elements (such as file sinks) which need to handle the newsegment event
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* specially.
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*
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* #GstBaseSink provides an overridable ::buffer_alloc function that can be
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* used by sinks that want to do reverse negotiation or to provide
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* custom buffers (hardware buffers for example) to upstream elements.
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*
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* The ::unlock method is called when the elements should unblock any blocking
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* operations they perform in the ::render method. This is mostly useful when
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* the ::render method performs a blocking write on a file descriptor, for
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* example.
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*
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* The max-lateness property affects how the sink deals with buffers that
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* arrive too late in the sink. A buffer arrives too late in the sink when
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* 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 ::get-times method does
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* not return a valid start time or 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 qos property will enable the quality-of-service features of the basesink
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* which gather statistics about the real-time performance of the clock
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* 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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* Since 0.10.15 the 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 "gstbasesink.h"
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#include <gst/gstmarshal.h>
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#include <gst/gst_private.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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/* FIXME, some stuff in ABI.data and other in Private...
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* Make up your mind please.
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*/
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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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/* 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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/* 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_in_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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/* 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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/* 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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GstBuffer *last_buffer;
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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_SIZE 1024
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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_PREROLL_QUEUE_LEN 0
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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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enum
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{
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PROP_0,
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PROP_PREROLL_QUEUE_LEN,
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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_LAST_BUFFER,
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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 GType base_sink_type = 0;
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if (G_UNLIKELY (base_sink_type == 0)) {
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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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base_sink_type = g_type_register_static (GST_TYPE_ELEMENT,
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"GstBaseSink", &base_sink_info, G_TYPE_FLAG_ABSTRACT);
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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 gst_base_sink_query (GstElement * element, GstQuery * query);
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static GstCaps *gst_base_sink_get_caps (GstBaseSink * sink);
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static gboolean gst_base_sink_set_caps (GstBaseSink * sink, GstCaps * caps);
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static GstFlowReturn gst_base_sink_buffer_alloc (GstBaseSink * sink,
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guint64 offset, guint size, GstCaps * caps, GstBuffer ** buf);
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static void gst_base_sink_get_times (GstBaseSink * basesink, GstBuffer * buffer,
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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 GstStateChangeReturn gst_base_sink_change_state (GstElement * element,
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GstStateChange transition);
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static GstFlowReturn gst_base_sink_chain (GstPad * pad, GstBuffer * buffer);
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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);
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static gboolean gst_base_sink_pad_activate_push (GstPad * pad, gboolean active);
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static gboolean gst_base_sink_pad_activate_pull (GstPad * pad, gboolean active);
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static gboolean gst_base_sink_event (GstPad * pad, GstEvent * event);
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static gboolean gst_base_sink_peer_query (GstBaseSink * sink, GstQuery * query);
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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 start, GstClockTime stop,
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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_DEBUG_FUNCPTR (gst_base_sink_finalize);
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gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_base_sink_set_property);
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gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_base_sink_get_property);
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/* FIXME, this next value should be configured using an event from the
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* upstream element, ie, the BUFFER_SIZE event. */
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g_object_class_install_property (gobject_class, PROP_PREROLL_QUEUE_LEN,
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g_param_spec_uint ("preroll-queue-len", "Preroll queue length",
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"Number of buffers to queue during preroll", 0, G_MAXUINT,
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DEFAULT_PREROLL_QUEUE_LEN,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
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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,
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g_param_spec_boolean ("qos", "Qos",
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"Generate Quality-of-Service events upstream", DEFAULT_QOS,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstBaseSink:async
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*
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* If set to #TRUE, the basesink will perform asynchronous state changes.
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* When set to #FALSE, the sink will not signal the parent when it prerolls.
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* Use this option when dealing with sparse streams or when synchronisation is
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* not required.
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*
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* Since: 0.10.15
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*/
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g_object_class_install_property (gobject_class, PROP_ASYNC,
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g_param_spec_boolean ("async", "Async",
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"Go asynchronously to PAUSED", DEFAULT_ASYNC,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstBaseSink:ts-offset
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*
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* Controls the final synchronisation, a negative value will render the buffer
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* earlier while a positive value delays playback. This property can be
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* used to fix synchronisation in bad files.
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*
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* Since: 0.10.15
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*/
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g_object_class_install_property (gobject_class, PROP_TS_OFFSET,
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g_param_spec_int64 ("ts-offset", "TS Offset",
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"Timestamp offset in nanoseconds", G_MININT64, G_MAXINT64,
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|
DEFAULT_TS_OFFSET, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstBaseSink:last-buffer
|
|
*
|
|
* The last buffer that arrived in the sink and was used for preroll or for
|
|
* rendering. This property can be used to generate thumbnails. This property
|
|
* can be NULL when the sink has not yet received a bufer.
|
|
*
|
|
* Since: 0.10.15
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_LAST_BUFFER,
|
|
gst_param_spec_mini_object ("last-buffer", "Last Buffer",
|
|
"The last buffer received in the sink", GST_TYPE_BUFFER,
|
|
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
|
|
|
|
gstelement_class->change_state =
|
|
GST_DEBUG_FUNCPTR (gst_base_sink_change_state);
|
|
gstelement_class->send_event = GST_DEBUG_FUNCPTR (gst_base_sink_send_event);
|
|
gstelement_class->query = GST_DEBUG_FUNCPTR (gst_base_sink_query);
|
|
|
|
klass->get_caps = GST_DEBUG_FUNCPTR (gst_base_sink_get_caps);
|
|
klass->set_caps = GST_DEBUG_FUNCPTR (gst_base_sink_set_caps);
|
|
klass->buffer_alloc = GST_DEBUG_FUNCPTR (gst_base_sink_buffer_alloc);
|
|
klass->get_times = GST_DEBUG_FUNCPTR (gst_base_sink_get_times);
|
|
klass->activate_pull =
|
|
GST_DEBUG_FUNCPTR (gst_base_sink_default_activate_pull);
|
|
}
|
|
|
|
static GstCaps *
|
|
gst_base_sink_pad_getcaps (GstPad * pad)
|
|
{
|
|
GstBaseSinkClass *bclass;
|
|
GstBaseSink *bsink;
|
|
GstCaps *caps = NULL;
|
|
|
|
bsink = GST_BASE_SINK (gst_pad_get_parent (pad));
|
|
bclass = GST_BASE_SINK_GET_CLASS (bsink);
|
|
if (bclass->get_caps)
|
|
caps = bclass->get_caps (bsink);
|
|
|
|
if (caps == NULL) {
|
|
GstPadTemplate *pad_template;
|
|
|
|
pad_template =
|
|
gst_element_class_get_pad_template (GST_ELEMENT_CLASS (bclass), "sink");
|
|
if (pad_template != NULL) {
|
|
caps = gst_caps_ref (gst_pad_template_get_caps (pad_template));
|
|
}
|
|
}
|
|
gst_object_unref (bsink);
|
|
|
|
return caps;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_sink_pad_setcaps (GstPad * pad, GstCaps * caps)
|
|
{
|
|
GstBaseSinkClass *bclass;
|
|
GstBaseSink *bsink;
|
|
gboolean res = TRUE;
|
|
|
|
bsink = GST_BASE_SINK (gst_pad_get_parent (pad));
|
|
bclass = GST_BASE_SINK_GET_CLASS (bsink);
|
|
|
|
if (bsink->pad_mode == GST_ACTIVATE_PULL) {
|
|
GstPad *peer = gst_pad_get_peer (pad);
|
|
|
|
if (peer)
|
|
res = gst_pad_set_caps (peer, caps);
|
|
else
|
|
res = FALSE;
|
|
|
|
if (!res)
|
|
GST_DEBUG_OBJECT (bsink, "peer setcaps() failed");
|
|
}
|
|
|
|
if (res && bclass->set_caps)
|
|
res = bclass->set_caps (bsink, caps);
|
|
|
|
gst_object_unref (bsink);
|
|
|
|
return res;
|
|
}
|
|
|
|
static void
|
|
gst_base_sink_pad_fixate (GstPad * pad, GstCaps * caps)
|
|
{
|
|
GstBaseSinkClass *bclass;
|
|
GstBaseSink *bsink;
|
|
|
|
bsink = GST_BASE_SINK (gst_pad_get_parent (pad));
|
|
bclass = GST_BASE_SINK_GET_CLASS (bsink);
|
|
|
|
if (bclass->fixate)
|
|
bclass->fixate (bsink, caps);
|
|
|
|
gst_object_unref (bsink);
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_base_sink_pad_buffer_alloc (GstPad * pad, guint64 offset, guint size,
|
|
GstCaps * caps, GstBuffer ** buf)
|
|
{
|
|
GstBaseSinkClass *bclass;
|
|
GstBaseSink *bsink;
|
|
GstFlowReturn result = GST_FLOW_OK;
|
|
|
|
bsink = GST_BASE_SINK (gst_pad_get_parent (pad));
|
|
bclass = GST_BASE_SINK_GET_CLASS (bsink);
|
|
|
|
if (bclass->buffer_alloc)
|
|
result = bclass->buffer_alloc (bsink, offset, size, caps, buf);
|
|
else
|
|
*buf = NULL; /* fallback in gstpad.c will allocate generic buffer */
|
|
|
|
gst_object_unref (bsink);
|
|
|
|
return result;
|
|
}
|
|
|
|
static void
|
|
gst_base_sink_init (GstBaseSink * basesink, gpointer g_class)
|
|
{
|
|
GstPadTemplate *pad_template;
|
|
GstBaseSinkPrivate *priv;
|
|
|
|
basesink->priv = priv = GST_BASE_SINK_GET_PRIVATE (basesink);
|
|
|
|
pad_template =
|
|
gst_element_class_get_pad_template (GST_ELEMENT_CLASS (g_class), "sink");
|
|
g_return_if_fail (pad_template != NULL);
|
|
|
|
basesink->sinkpad = gst_pad_new_from_template (pad_template, "sink");
|
|
|
|
gst_pad_set_getcaps_function (basesink->sinkpad,
|
|
GST_DEBUG_FUNCPTR (gst_base_sink_pad_getcaps));
|
|
gst_pad_set_setcaps_function (basesink->sinkpad,
|
|
GST_DEBUG_FUNCPTR (gst_base_sink_pad_setcaps));
|
|
gst_pad_set_fixatecaps_function (basesink->sinkpad,
|
|
GST_DEBUG_FUNCPTR (gst_base_sink_pad_fixate));
|
|
gst_pad_set_bufferalloc_function (basesink->sinkpad,
|
|
GST_DEBUG_FUNCPTR (gst_base_sink_pad_buffer_alloc));
|
|
gst_pad_set_activate_function (basesink->sinkpad,
|
|
GST_DEBUG_FUNCPTR (gst_base_sink_pad_activate));
|
|
gst_pad_set_activatepush_function (basesink->sinkpad,
|
|
GST_DEBUG_FUNCPTR (gst_base_sink_pad_activate_push));
|
|
gst_pad_set_activatepull_function (basesink->sinkpad,
|
|
GST_DEBUG_FUNCPTR (gst_base_sink_pad_activate_pull));
|
|
gst_pad_set_event_function (basesink->sinkpad,
|
|
GST_DEBUG_FUNCPTR (gst_base_sink_event));
|
|
gst_pad_set_chain_function (basesink->sinkpad,
|
|
GST_DEBUG_FUNCPTR (gst_base_sink_chain));
|
|
gst_element_add_pad (GST_ELEMENT_CAST (basesink), basesink->sinkpad);
|
|
|
|
basesink->pad_mode = GST_ACTIVATE_NONE;
|
|
basesink->preroll_queue = g_queue_new ();
|
|
basesink->abidata.ABI.clip_segment = gst_segment_new ();
|
|
priv->have_latency = FALSE;
|
|
|
|
basesink->can_activate_push = DEFAULT_CAN_ACTIVATE_PUSH;
|
|
basesink->can_activate_pull = DEFAULT_CAN_ACTIVATE_PULL;
|
|
|
|
basesink->sync = DEFAULT_SYNC;
|
|
basesink->abidata.ABI.max_lateness = DEFAULT_MAX_LATENESS;
|
|
gst_atomic_int_set (&priv->qos_enabled, DEFAULT_QOS);
|
|
priv->async_enabled = DEFAULT_ASYNC;
|
|
priv->ts_offset = DEFAULT_TS_OFFSET;
|
|
|
|
GST_OBJECT_FLAG_SET (basesink, GST_ELEMENT_IS_SINK);
|
|
}
|
|
|
|
static void
|
|
gst_base_sink_finalize (GObject * object)
|
|
{
|
|
GstBaseSink *basesink;
|
|
|
|
basesink = GST_BASE_SINK (object);
|
|
|
|
g_queue_free (basesink->preroll_queue);
|
|
gst_segment_free (basesink->abidata.ABI.clip_segment);
|
|
|
|
G_OBJECT_CLASS (parent_class)->finalize (object);
|
|
}
|
|
|
|
/**
|
|
* gst_base_sink_set_sync:
|
|
* @sink: the sink
|
|
* @sync: the new sync value.
|
|
*
|
|
* Configures @sink to synchronize on the clock or not. When
|
|
* @sync is FALSE, incomming samples will be played as fast as
|
|
* possible. If @sync is TRUE, the timestamps of the incomming
|
|
* buffers will be used to schedule the exact render time of its
|
|
* contents.
|
|
*
|
|
* Since: 0.10.4
|
|
*/
|
|
void
|
|
gst_base_sink_set_sync (GstBaseSink * sink, gboolean sync)
|
|
{
|
|
g_return_if_fail (GST_IS_BASE_SINK (sink));
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
sink->sync = sync;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
}
|
|
|
|
/**
|
|
* gst_base_sink_get_sync:
|
|
* @sink: the sink
|
|
*
|
|
* Checks if @sink is currently configured to synchronize against the
|
|
* clock.
|
|
*
|
|
* Returns: TRUE if the sink is configured to synchronize against the clock.
|
|
*
|
|
* Since: 0.10.4
|
|
*/
|
|
gboolean
|
|
gst_base_sink_get_sync (GstBaseSink * sink)
|
|
{
|
|
gboolean res;
|
|
|
|
g_return_val_if_fail (GST_IS_BASE_SINK (sink), FALSE);
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
res = sink->sync;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_base_sink_set_max_lateness:
|
|
* @sink: the sink
|
|
* @max_lateness: the new max lateness value.
|
|
*
|
|
* Sets the new max lateness value to @max_lateness. This value is
|
|
* used to decide if a buffer should be dropped or not based on the
|
|
* buffer timestamp and the current clock time. A value of -1 means
|
|
* an unlimited time.
|
|
*
|
|
* Since: 0.10.4
|
|
*/
|
|
void
|
|
gst_base_sink_set_max_lateness (GstBaseSink * sink, gint64 max_lateness)
|
|
{
|
|
g_return_if_fail (GST_IS_BASE_SINK (sink));
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
sink->abidata.ABI.max_lateness = max_lateness;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
}
|
|
|
|
/**
|
|
* gst_base_sink_get_max_lateness:
|
|
* @sink: the sink
|
|
*
|
|
* Gets the max lateness value. See gst_base_sink_set_max_lateness for
|
|
* more details.
|
|
*
|
|
* Returns: The maximum time in nanoseconds that a buffer can be late
|
|
* before it is dropped and not rendered. A value of -1 means an
|
|
* unlimited time.
|
|
*
|
|
* Since: 0.10.4
|
|
*/
|
|
gint64
|
|
gst_base_sink_get_max_lateness (GstBaseSink * sink)
|
|
{
|
|
gint64 res;
|
|
|
|
g_return_val_if_fail (GST_IS_BASE_SINK (sink), -1);
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
res = sink->abidata.ABI.max_lateness;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_base_sink_set_qos_enabled:
|
|
* @sink: the sink
|
|
* @enabled: the new qos value.
|
|
*
|
|
* Configures @sink to send Quality-of-Service events upstream.
|
|
*
|
|
* Since: 0.10.5
|
|
*/
|
|
void
|
|
gst_base_sink_set_qos_enabled (GstBaseSink * sink, gboolean enabled)
|
|
{
|
|
g_return_if_fail (GST_IS_BASE_SINK (sink));
|
|
|
|
gst_atomic_int_set (&sink->priv->qos_enabled, enabled);
|
|
}
|
|
|
|
/**
|
|
* gst_base_sink_is_qos_enabled:
|
|
* @sink: the sink
|
|
*
|
|
* Checks if @sink is currently configured to send Quality-of-Service events
|
|
* upstream.
|
|
*
|
|
* Returns: TRUE if the sink is configured to perform Quality-of-Service.
|
|
*
|
|
* Since: 0.10.5
|
|
*/
|
|
gboolean
|
|
gst_base_sink_is_qos_enabled (GstBaseSink * sink)
|
|
{
|
|
gboolean res;
|
|
|
|
g_return_val_if_fail (GST_IS_BASE_SINK (sink), FALSE);
|
|
|
|
res = g_atomic_int_get (&sink->priv->qos_enabled);
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_base_sink_set_async_enabled:
|
|
* @sink: the sink
|
|
* @enabled: the new async value.
|
|
*
|
|
* Configures @sink to perform all state changes asynchronusly. When async is
|
|
* disabled, the sink will immediatly go to PAUSED instead of waiting for a
|
|
* preroll buffer. This feature is usefull if the sink does not synchronize
|
|
* against the clock or when it is dealing with sparse streams.
|
|
*
|
|
* Since: 0.10.15
|
|
*/
|
|
void
|
|
gst_base_sink_set_async_enabled (GstBaseSink * sink, gboolean enabled)
|
|
{
|
|
g_return_if_fail (GST_IS_BASE_SINK (sink));
|
|
|
|
GST_PAD_PREROLL_LOCK (sink->sinkpad);
|
|
sink->priv->async_enabled = enabled;
|
|
GST_PAD_PREROLL_UNLOCK (sink->sinkpad);
|
|
}
|
|
|
|
/**
|
|
* gst_base_sink_is_async_enabled:
|
|
* @sink: the sink
|
|
*
|
|
* Checks if @sink is currently configured to perform asynchronous state
|
|
* changes to PAUSED.
|
|
*
|
|
* Returns: TRUE if the sink is configured to perform asynchronous state
|
|
* changes.
|
|
*
|
|
* Since: 0.10.15
|
|
*/
|
|
gboolean
|
|
gst_base_sink_is_async_enabled (GstBaseSink * sink)
|
|
{
|
|
gboolean res;
|
|
|
|
g_return_val_if_fail (GST_IS_BASE_SINK (sink), FALSE);
|
|
|
|
GST_PAD_PREROLL_LOCK (sink->sinkpad);
|
|
res = sink->priv->async_enabled;
|
|
GST_PAD_PREROLL_UNLOCK (sink->sinkpad);
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_base_sink_set_ts_offset:
|
|
* @sink: the sink
|
|
* @offset: the new offset
|
|
*
|
|
* Adjust the synchronisation of @sink with @offset. A negative value will
|
|
* render buffers earlier than their timestamp. A positive value will delay
|
|
* rendering. This function can be used to fix playback of badly timestamped
|
|
* buffers.
|
|
*
|
|
* Since: 0.10.15
|
|
*/
|
|
void
|
|
gst_base_sink_set_ts_offset (GstBaseSink * sink, GstClockTimeDiff offset)
|
|
{
|
|
g_return_if_fail (GST_IS_BASE_SINK (sink));
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
sink->priv->ts_offset = offset;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
}
|
|
|
|
/**
|
|
* gst_base_sink_get_ts_offset:
|
|
* @sink: the sink
|
|
*
|
|
* Get the synchronisation offset of @sink.
|
|
*
|
|
* Returns: The synchronisation offset.
|
|
*
|
|
* Since: 0.10.15
|
|
*/
|
|
GstClockTimeDiff
|
|
gst_base_sink_get_ts_offset (GstBaseSink * sink)
|
|
{
|
|
GstClockTimeDiff res;
|
|
|
|
g_return_val_if_fail (GST_IS_BASE_SINK (sink), 0);
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
res = sink->priv->ts_offset;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_base_sink_get_last_buffer:
|
|
* @sink: the sink
|
|
*
|
|
* Get the last buffer that arrived in the sink and was used for preroll or for
|
|
* rendering. This property can be used to generate thumbnails.
|
|
*
|
|
* The #GstCaps on the buffer can be used to determine the type of the buffer.
|
|
*
|
|
* Returns: a #GstBuffer. gst_buffer_unref() after usage. This function returns
|
|
* NULL when no buffer has arrived in the sink yet or when the sink is not in
|
|
* PAUSED or PLAYING.
|
|
*
|
|
* Since: 0.10.15
|
|
*/
|
|
GstBuffer *
|
|
gst_base_sink_get_last_buffer (GstBaseSink * sink)
|
|
{
|
|
GstBuffer *res;
|
|
|
|
g_return_val_if_fail (GST_IS_BASE_SINK (sink), NULL);
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
if ((res = sink->priv->last_buffer))
|
|
gst_buffer_ref (res);
|
|
GST_OBJECT_UNLOCK (sink);
|
|
|
|
return res;
|
|
}
|
|
|
|
static void
|
|
gst_base_sink_set_last_buffer (GstBaseSink * sink, GstBuffer * buffer)
|
|
{
|
|
GstBuffer *old;
|
|
|
|
if (buffer)
|
|
gst_buffer_ref (buffer);
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
old = sink->priv->last_buffer;
|
|
sink->priv->last_buffer = buffer;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
|
|
if (old)
|
|
gst_buffer_unref (old);
|
|
}
|
|
|
|
/**
|
|
* gst_base_sink_get_latency:
|
|
* @sink: the sink
|
|
*
|
|
* Get the currently configured latency.
|
|
*
|
|
* Returns: The configured latency.
|
|
*
|
|
* Since: 0.10.12
|
|
*/
|
|
GstClockTime
|
|
gst_base_sink_get_latency (GstBaseSink * sink)
|
|
{
|
|
GstClockTime res;
|
|
|
|
GST_OBJECT_LOCK (sink);
|
|
res = sink->priv->latency;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_base_sink_query_latency:
|
|
* @sink: the sink
|
|
* @live: if the sink is live
|
|
* @upstream_live: if an upstream element is live
|
|
* @min_latency: the min latency of the upstream elements
|
|
* @max_latency: the max latency of the upstream elements
|
|
*
|
|
* Query the sink for the latency parameters. The latency will be queried from
|
|
* the upstream elements. @live will be TRUE if @sink is configured to
|
|
* synchronize against the clock. @upstream_live will be TRUE if an upstream
|
|
* element is live.
|
|
*
|
|
* If both @live and @upstream_live are TRUE, the sink will want to compensate
|
|
* for the latency introduced by the upstream elements by setting the
|
|
* @min_latency to a strictly possitive value.
|
|
*
|
|
* This function is mostly used by subclasses.
|
|
*
|
|
* Returns: TRUE if the query succeeded.
|
|
*
|
|
* Since: 0.10.12
|
|
*/
|
|
gboolean
|
|
gst_base_sink_query_latency (GstBaseSink * sink, gboolean * live,
|
|
gboolean * upstream_live, GstClockTime * min_latency,
|
|
GstClockTime * max_latency)
|
|
{
|
|
gboolean l, us_live, res, have_latency;
|
|
GstClockTime min, max;
|
|
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;
|
|
GST_OBJECT_UNLOCK (sink);
|
|
|
|
/* assume no latency */
|
|
min = 0;
|
|
max = -1;
|
|
us_live = FALSE;
|
|
|
|
/* we are live */
|
|
if (l) {
|
|
if (have_latency) {
|
|
/* we are live and ready for a latency query */
|
|
query = gst_query_new_latency ();
|
|
|
|
/* ask the peer for the latency */
|
|
if (!(res = gst_base_sink_peer_query (sink, query)))
|
|
goto query_failed;
|
|
|
|
/* get upstream min and max latency */
|
|
gst_query_parse_latency (query, &us_live, &us_min, &us_max);
|
|
gst_query_unref (query);
|
|
|
|
if (us_live) {
|
|
/* upstream live, use its latency, subclasses should use these
|
|
* values to create the complete latency. */
|
|
min = us_min;
|
|
max = us_max;
|
|
}
|
|
} else {
|
|
/* we are live but are not yet ready for a latency query */
|
|
res = FALSE;
|
|
}
|
|
} else {
|
|
/* not live, result is always TRUE */
|
|
res = TRUE;
|
|
}
|
|
|
|
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;
|
|
|
|
/* ERRORS */
|
|
query_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (sink, "latency query failed");
|
|
gst_query_unref (query);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_base_sink_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstBaseSink *sink = GST_BASE_SINK (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_PREROLL_QUEUE_LEN:
|
|
/* preroll lock necessary to serialize with finish_preroll */
|
|
GST_PAD_PREROLL_LOCK (sink->sinkpad);
|
|
sink->preroll_queue_max_len = g_value_get_uint (value);
|
|
GST_PAD_PREROLL_UNLOCK (sink->sinkpad);
|
|
break;
|
|
case PROP_SYNC:
|
|
gst_base_sink_set_sync (sink, g_value_get_boolean (value));
|
|
break;
|
|
case PROP_MAX_LATENESS:
|
|
gst_base_sink_set_max_lateness (sink, g_value_get_int64 (value));
|
|
break;
|
|
case PROP_QOS:
|
|
gst_base_sink_set_qos_enabled (sink, g_value_get_boolean (value));
|
|
break;
|
|
case PROP_ASYNC:
|
|
gst_base_sink_set_async_enabled (sink, g_value_get_boolean (value));
|
|
break;
|
|
case PROP_TS_OFFSET:
|
|
gst_base_sink_set_ts_offset (sink, g_value_get_int64 (value));
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_base_sink_get_property (GObject * object, guint prop_id, GValue * value,
|
|
GParamSpec * pspec)
|
|
{
|
|
GstBaseSink *sink = GST_BASE_SINK (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_PREROLL_QUEUE_LEN:
|
|
GST_PAD_PREROLL_LOCK (sink->sinkpad);
|
|
g_value_set_uint (value, sink->preroll_queue_max_len);
|
|
GST_PAD_PREROLL_UNLOCK (sink->sinkpad);
|
|
break;
|
|
case PROP_SYNC:
|
|
g_value_set_boolean (value, gst_base_sink_get_sync (sink));
|
|
break;
|
|
case PROP_MAX_LATENESS:
|
|
g_value_set_int64 (value, gst_base_sink_get_max_lateness (sink));
|
|
break;
|
|
case PROP_QOS:
|
|
g_value_set_boolean (value, gst_base_sink_is_qos_enabled (sink));
|
|
break;
|
|
case PROP_ASYNC:
|
|
g_value_set_boolean (value, gst_base_sink_is_async_enabled (sink));
|
|
break;
|
|
case PROP_TS_OFFSET:
|
|
g_value_set_int64 (value, gst_base_sink_get_ts_offset (sink));
|
|
break;
|
|
case PROP_LAST_BUFFER:
|
|
gst_value_take_buffer (value, gst_base_sink_get_last_buffer (sink));
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
|
|
static GstCaps *
|
|
gst_base_sink_get_caps (GstBaseSink * sink)
|
|
{
|
|
return NULL;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_sink_set_caps (GstBaseSink * sink, GstCaps * caps)
|
|
{
|
|
return TRUE;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_base_sink_buffer_alloc (GstBaseSink * sink, guint64 offset, guint size,
|
|
GstCaps * caps, GstBuffer ** buf)
|
|
{
|
|
*buf = NULL;
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
/* with PREROLL_LOCK, STREAM_LOCK */
|
|
static void
|
|
gst_base_sink_preroll_queue_flush (GstBaseSink * basesink, GstPad * pad)
|
|
{
|
|
GstMiniObject *obj;
|
|
|
|
GST_DEBUG_OBJECT (basesink, "flushing queue %p", basesink);
|
|
while ((obj = g_queue_pop_head (basesink->preroll_queue))) {
|
|
GST_DEBUG_OBJECT (basesink, "popped %p", obj);
|
|
gst_mini_object_unref (obj);
|
|
}
|
|
/* we can't have EOS anymore now */
|
|
basesink->eos = FALSE;
|
|
basesink->priv->received_eos = FALSE;
|
|
basesink->have_preroll = FALSE;
|
|
basesink->eos_queued = FALSE;
|
|
basesink->preroll_queued = 0;
|
|
basesink->buffers_queued = 0;
|
|
basesink->events_queued = 0;
|
|
/* can't report latency anymore until we preroll again */
|
|
if (basesink->priv->async_enabled) {
|
|
GST_OBJECT_LOCK (basesink);
|
|
basesink->priv->have_latency = FALSE;
|
|
GST_OBJECT_UNLOCK (basesink);
|
|
}
|
|
/* and signal any waiters now */
|
|
GST_PAD_PREROLL_SIGNAL (pad);
|
|
}
|
|
|
|
/* with STREAM_LOCK, configures given segment with the event information. */
|
|
static void
|
|
gst_base_sink_configure_segment (GstBaseSink * basesink, GstPad * pad,
|
|
GstEvent * event, GstSegment * segment)
|
|
{
|
|
gboolean update;
|
|
gdouble rate, arate;
|
|
GstFormat format;
|
|
gint64 start;
|
|
gint64 stop;
|
|
gint64 time;
|
|
|
|
/* the newsegment event is needed to bring the buffer timestamps to the
|
|
* stream time and to drop samples outside of the playback segment. */
|
|
gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
|
|
&start, &stop, &time);
|
|
|
|
/* The segment is protected with both the STREAM_LOCK and the OBJECT_LOCK.
|
|
* We protect with the OBJECT_LOCK so that we can use the values to
|
|
* safely answer a POSITION query. */
|
|
GST_OBJECT_LOCK (basesink);
|
|
gst_segment_set_newsegment_full (segment, update, rate, arate, format, start,
|
|
stop, time);
|
|
|
|
if (format == GST_FORMAT_TIME) {
|
|
GST_DEBUG_OBJECT (basesink,
|
|
"configured NEWSEGMENT update %d, rate %lf, applied rate %lf, "
|
|
"format GST_FORMAT_TIME, "
|
|
"%" GST_TIME_FORMAT " -- %" GST_TIME_FORMAT
|
|
", time %" GST_TIME_FORMAT ", accum %" GST_TIME_FORMAT,
|
|
update, rate, arate, GST_TIME_ARGS (segment->start),
|
|
GST_TIME_ARGS (segment->stop), GST_TIME_ARGS (segment->time),
|
|
GST_TIME_ARGS (segment->accum));
|
|
} else {
|
|
GST_DEBUG_OBJECT (basesink,
|
|
"configured NEWSEGMENT update %d, rate %lf, applied rate %lf, "
|
|
"format %d, "
|
|
"%" G_GINT64_FORMAT " -- %" G_GINT64_FORMAT ", time %"
|
|
G_GINT64_FORMAT ", accum %" G_GINT64_FORMAT, update, rate, arate,
|
|
segment->format, segment->start, segment->stop, segment->time,
|
|
segment->accum);
|
|
}
|
|
GST_OBJECT_UNLOCK (basesink);
|
|
}
|
|
|
|
/* with PREROLL_LOCK, STREAM_LOCK */
|
|
static gboolean
|
|
gst_base_sink_commit_state (GstBaseSink * basesink)
|
|
{
|
|
/* commit state and proceed to next pending state */
|
|
GstState current, next, pending, post_pending;
|
|
gboolean post_paused = FALSE;
|
|
gboolean post_async_done = FALSE;
|
|
gboolean post_playing = FALSE;
|
|
gboolean sync;
|
|
|
|
/* we are certainly not playing async anymore now */
|
|
basesink->playing_async = FALSE;
|
|
|
|
GST_OBJECT_LOCK (basesink);
|
|
current = GST_STATE (basesink);
|
|
next = GST_STATE_NEXT (basesink);
|
|
pending = GST_STATE_PENDING (basesink);
|
|
post_pending = pending;
|
|
sync = basesink->sync;
|
|
|
|
switch (pending) {
|
|
case GST_STATE_PLAYING:
|
|
{
|
|
GstBaseSinkClass *bclass;
|
|
GstStateChangeReturn ret;
|
|
|
|
bclass = GST_BASE_SINK_GET_CLASS (basesink);
|
|
|
|
GST_DEBUG_OBJECT (basesink, "commiting state to PLAYING");
|
|
|
|
basesink->need_preroll = FALSE;
|
|
post_async_done = TRUE;
|
|
basesink->priv->commited = TRUE;
|
|
post_playing = TRUE;
|
|
/* post PAUSED too when we were READY */
|
|
if (current == GST_STATE_READY) {
|
|
post_paused = TRUE;
|
|
}
|
|
|
|
/* make sure we notify the subclass of async playing */
|
|
if (bclass->async_play) {
|
|
ret = bclass->async_play (basesink);
|
|
if (ret == GST_STATE_CHANGE_FAILURE)
|
|
goto async_failed;
|
|
}
|
|
break;
|
|
}
|
|
case GST_STATE_PAUSED:
|
|
GST_DEBUG_OBJECT (basesink, "commiting state to PAUSED");
|
|
post_paused = TRUE;
|
|
post_async_done = TRUE;
|
|
basesink->priv->commited = TRUE;
|
|
post_pending = GST_STATE_VOID_PENDING;
|
|
break;
|
|
case GST_STATE_READY:
|
|
case GST_STATE_NULL:
|
|
goto stopping;
|
|
case GST_STATE_VOID_PENDING:
|
|
goto nothing_pending;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
/* we can report latency queries now */
|
|
basesink->priv->have_latency = TRUE;
|
|
|
|
GST_STATE (basesink) = pending;
|
|
GST_STATE_NEXT (basesink) = GST_STATE_VOID_PENDING;
|
|
GST_STATE_PENDING (basesink) = GST_STATE_VOID_PENDING;
|
|
GST_STATE_RETURN (basesink) = GST_STATE_CHANGE_SUCCESS;
|
|
GST_OBJECT_UNLOCK (basesink);
|
|
|
|
if (post_paused) {
|
|
GST_DEBUG_OBJECT (basesink, "posting PAUSED state change message");
|
|
gst_element_post_message (GST_ELEMENT_CAST (basesink),
|
|
gst_message_new_state_changed (GST_OBJECT_CAST (basesink),
|
|
current, next, post_pending));
|
|
}
|
|
if (post_async_done) {
|
|
GST_DEBUG_OBJECT (basesink, "posting async-done message");
|
|
gst_element_post_message (GST_ELEMENT_CAST (basesink),
|
|
gst_message_new_async_done (GST_OBJECT_CAST (basesink)));
|
|
}
|
|
if (post_playing) {
|
|
GST_DEBUG_OBJECT (basesink, "posting PLAYING state change message");
|
|
gst_element_post_message (GST_ELEMENT_CAST (basesink),
|
|
gst_message_new_state_changed (GST_OBJECT_CAST (basesink),
|
|
next, pending, GST_STATE_VOID_PENDING));
|
|
}
|
|
|
|
GST_STATE_BROADCAST (basesink);
|
|
|
|
return TRUE;
|
|
|
|
nothing_pending:
|
|
{
|
|
/* Depending on the state, set our vars. We get in this situation when the
|
|
* state change function got a change to update the state vars before the
|
|
* streaming thread did. This is fine but we need to make sure that we
|
|
* update the need_preroll var since it was TRUE when we got here and might
|
|
* become FALSE if we got to PLAYING. */
|
|
GST_DEBUG_OBJECT (basesink, "nothing to commit, now in %s",
|
|
gst_element_state_get_name (current));
|
|
switch (current) {
|
|
case GST_STATE_PLAYING:
|
|
basesink->need_preroll = FALSE;
|
|
break;
|
|
case GST_STATE_PAUSED:
|
|
basesink->need_preroll = TRUE;
|
|
break;
|
|
default:
|
|
basesink->need_preroll = FALSE;
|
|
basesink->flushing = TRUE;
|
|
break;
|
|
}
|
|
/* we can report latency queries now */
|
|
basesink->priv->have_latency = TRUE;
|
|
GST_OBJECT_UNLOCK (basesink);
|
|
return TRUE;
|
|
}
|
|
stopping:
|
|
{
|
|
/* app is going to READY */
|
|
GST_DEBUG_OBJECT (basesink, "stopping");
|
|
basesink->need_preroll = FALSE;
|
|
basesink->flushing = TRUE;
|
|
GST_OBJECT_UNLOCK (basesink);
|
|
return FALSE;
|
|
}
|
|
async_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "async commit failed");
|
|
GST_STATE_RETURN (basesink) = GST_STATE_CHANGE_FAILURE;
|
|
GST_OBJECT_UNLOCK (basesink);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
|
|
/* with STREAM_LOCK, PREROLL_LOCK
|
|
*
|
|
* Returns TRUE if the object needs synchronisation and takes therefore
|
|
* part in prerolling.
|
|
*
|
|
* rsstart/rsstop contain the start/stop in stream time.
|
|
* rrstart/rrstop contain the start/stop in running time.
|
|
*/
|
|
static gboolean
|
|
gst_base_sink_get_sync_times (GstBaseSink * basesink, GstMiniObject * obj,
|
|
GstClockTime * rsstart, GstClockTime * rsstop,
|
|
GstClockTime * rrstart, GstClockTime * rrstop, gboolean * do_sync,
|
|
GstSegment * segment)
|
|
{
|
|
GstBaseSinkClass *bclass;
|
|
GstBuffer *buffer;
|
|
GstClockTime start, stop; /* raw start/stop timestamps */
|
|
gint64 cstart, cstop; /* clipped raw timestamps */
|
|
gint64 rstart, rstop; /* clipped timestamps converted to running time */
|
|
GstClockTime sstart, sstop; /* clipped timestamps converted to stream time */
|
|
GstFormat format;
|
|
GstBaseSinkPrivate *priv;
|
|
|
|
priv = basesink->priv;
|
|
|
|
/* start with nothing */
|
|
start = stop = sstart = sstop = rstart = rstop = -1;
|
|
|
|
if (G_UNLIKELY (GST_IS_EVENT (obj))) {
|
|
GstEvent *event = GST_EVENT_CAST (obj);
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
/* EOS event needs syncing */
|
|
case GST_EVENT_EOS:
|
|
sstart = sstop = priv->current_sstop;
|
|
rstart = rstop = priv->eos_rtime;
|
|
*do_sync = rstart != -1;
|
|
GST_DEBUG_OBJECT (basesink, "sync times for EOS %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (rstart));
|
|
goto done;
|
|
/* other events do not need syncing */
|
|
/* FIXME, maybe NEWSEGMENT might need synchronisation
|
|
* since the POSITION query depends on accumulated times and
|
|
* we cannot accumulate the current segment before the previous
|
|
* one completed.
|
|
*/
|
|
default:
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/* else do buffer sync code */
|
|
buffer = GST_BUFFER_CAST (obj);
|
|
|
|
bclass = GST_BASE_SINK_GET_CLASS (basesink);
|
|
|
|
/* just get the times to see if we need syncing */
|
|
if (bclass->get_times)
|
|
bclass->get_times (basesink, buffer, &start, &stop);
|
|
|
|
if (start == -1) {
|
|
gst_base_sink_get_times (basesink, buffer, &start, &stop);
|
|
*do_sync = FALSE;
|
|
} else {
|
|
*do_sync = TRUE;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (basesink, "got times start: %" GST_TIME_FORMAT
|
|
", stop: %" GST_TIME_FORMAT ", do_sync %d", GST_TIME_ARGS (start),
|
|
GST_TIME_ARGS (stop), *do_sync);
|
|
|
|
/* collect segment and format for code clarity */
|
|
format = segment->format;
|
|
|
|
/* no timestamp clipping if we did not * get a TIME segment format */
|
|
if (G_UNLIKELY (format != GST_FORMAT_TIME)) {
|
|
cstart = start;
|
|
cstop = stop;
|
|
/* do running and stream time in TIME format */
|
|
format = GST_FORMAT_TIME;
|
|
goto do_times;
|
|
}
|
|
|
|
/* clip */
|
|
if (G_UNLIKELY (!gst_segment_clip (segment, GST_FORMAT_TIME,
|
|
(gint64) start, (gint64) stop, &cstart, &cstop)))
|
|
goto out_of_segment;
|
|
|
|
if (G_UNLIKELY (start != cstart || stop != cstop)) {
|
|
GST_DEBUG_OBJECT (basesink, "clipped to: start %" GST_TIME_FORMAT
|
|
", stop: %" GST_TIME_FORMAT, GST_TIME_ARGS (cstart),
|
|
GST_TIME_ARGS (cstop));
|
|
}
|
|
|
|
/* set last stop position */
|
|
gst_segment_set_last_stop (segment, GST_FORMAT_TIME, cstop);
|
|
|
|
do_times:
|
|
/* this can produce wrong values if we accumulated non-TIME segments. If this happens,
|
|
* upstream is behaving very badly */
|
|
sstart = gst_segment_to_stream_time (segment, format, cstart);
|
|
sstop = gst_segment_to_stream_time (segment, format, cstop);
|
|
rstart = gst_segment_to_running_time (segment, format, cstart);
|
|
rstop = gst_segment_to_running_time (segment, format, cstop);
|
|
|
|
done:
|
|
/* save times */
|
|
*rsstart = sstart;
|
|
*rsstop = sstop;
|
|
*rrstart = rstart;
|
|
*rrstop = rstop;
|
|
|
|
/* buffers and EOS always need syncing and preroll */
|
|
return TRUE;
|
|
|
|
/* special cases */
|
|
out_of_segment:
|
|
{
|
|
/* should not happen since we clip them in the chain function already,
|
|
* we return FALSE so that we don't try to sync on it. */
|
|
GST_ELEMENT_WARNING (basesink, STREAM, FAILED,
|
|
(NULL), ("unexpected buffer out of segment found."));
|
|
GST_LOG_OBJECT (basesink, "buffer skipped, not in segment");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/* with STREAM_LOCK, PREROLL_LOCK
|
|
*
|
|
* Waits for the clock to reach @time. If @time is not valid, no
|
|
* synchronisation is done and BADTIME is returned.
|
|
* If synchronisation is disabled in the element or there is no
|
|
* clock, no synchronisation is done and BADTIME is returned.
|
|
*
|
|
* Else 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.
|
|
*
|
|
* @time is expressed in running time.
|
|
*/
|
|
static GstClockReturn
|
|
gst_base_sink_wait_clock (GstBaseSink * basesink, GstClockTime time,
|
|
GstClockTimeDiff * jitter)
|
|
{
|
|
GstClockID id;
|
|
GstClockReturn ret;
|
|
GstClock *clock;
|
|
GstClockTimeDiff ts_offset;
|
|
|
|
if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (time)))
|
|
goto invalid_time;
|
|
|
|
GST_OBJECT_LOCK (basesink);
|
|
if (G_UNLIKELY (!basesink->sync))
|
|
goto no_sync;
|
|
|
|
if (G_UNLIKELY ((clock = GST_ELEMENT_CLOCK (basesink)) == NULL))
|
|
goto no_clock;
|
|
|
|
/* add base time and latency */
|
|
time += GST_ELEMENT_CAST (basesink)->base_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;
|
|
|
|
id = gst_clock_new_single_shot_id (clock, time);
|
|
GST_OBJECT_UNLOCK (basesink);
|
|
|
|
basesink->clock_id = id;
|
|
/* release the preroll lock while waiting */
|
|
GST_PAD_PREROLL_UNLOCK (basesink->sinkpad);
|
|
|
|
ret = gst_clock_id_wait (id, jitter);
|
|
|
|
GST_PAD_PREROLL_LOCK (basesink->sinkpad);
|
|
gst_clock_id_unref (id);
|
|
basesink->clock_id = NULL;
|
|
|
|
return ret;
|
|
|
|
/* no syncing needed */
|
|
invalid_time:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "time not valid, no sync needed");
|
|
return GST_CLOCK_BADTIME;
|
|
}
|
|
no_sync:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "sync disabled");
|
|
GST_OBJECT_UNLOCK (basesink);
|
|
return GST_CLOCK_BADTIME;
|
|
}
|
|
no_clock:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "no clock, can't sync");
|
|
GST_OBJECT_UNLOCK (basesink);
|
|
return GST_CLOCK_BADTIME;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_base_sink_wait_preroll:
|
|
* @sink: the sink
|
|
*
|
|
* If the #GstBaseSinkClass::render method performs its own synchronisation against
|
|
* the clock it must unblock when going from PLAYING to the PAUSED state and call
|
|
* this method before continuing to render the remaining data.
|
|
*
|
|
* This function will block until a state change to PLAYING happens (in which
|
|
* case this function returns #GST_FLOW_OK) or the processing must be stopped due
|
|
* to a state change to READY or a FLUSH event (in which case this function
|
|
* returns #GST_FLOW_WRONG_STATE).
|
|
*
|
|
* Since: 0.10.11
|
|
*
|
|
* Returns: #GST_FLOW_OK if the preroll completed and processing can
|
|
* continue. Any other return value should be returned from the render vmethod.
|
|
*/
|
|
GstFlowReturn
|
|
gst_base_sink_wait_preroll (GstBaseSink * sink)
|
|
{
|
|
sink->have_preroll = TRUE;
|
|
GST_DEBUG_OBJECT (sink, "waiting in preroll for flush or PLAYING");
|
|
/* block until the state changes, or we get a flush, or something */
|
|
GST_PAD_PREROLL_WAIT (sink->sinkpad);
|
|
sink->have_preroll = FALSE;
|
|
if (G_UNLIKELY (sink->flushing))
|
|
goto stopping;
|
|
GST_DEBUG_OBJECT (sink, "continue after preroll");
|
|
|
|
return GST_FLOW_OK;
|
|
|
|
/* ERRORS */
|
|
stopping:
|
|
{
|
|
GST_DEBUG_OBJECT (sink, "preroll interrupted");
|
|
return GST_FLOW_WRONG_STATE;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_base_sink_wait_eos:
|
|
* @sink: the sink
|
|
* @time: the running_time to be reached
|
|
* @jitter: the jitter to be filled with time diff (can be NULL)
|
|
*
|
|
* This function will block until @time is reached. It is usually called by
|
|
* subclasses that use their own internal synchronisation but want to let the
|
|
* EOS be handled by the base class.
|
|
*
|
|
* This function should only be called with the PREROLL_LOCK held, like when
|
|
* receiving an EOS event in the ::event vmethod.
|
|
*
|
|
* Since 0.10.15
|
|
*
|
|
* Returns: #GstFlowReturn
|
|
*/
|
|
GstFlowReturn
|
|
gst_base_sink_wait_eos (GstBaseSink * sink, GstClockTime time,
|
|
GstClockTimeDiff * jitter)
|
|
{
|
|
GstClockReturn status;
|
|
GstFlowReturn ret;
|
|
|
|
do {
|
|
GST_DEBUG_OBJECT (sink, "checking preroll");
|
|
|
|
/* first wait for the playing state before we can continue */
|
|
if (G_UNLIKELY (sink->need_preroll)) {
|
|
ret = gst_base_sink_wait_preroll (sink);
|
|
if (ret != GST_FLOW_OK)
|
|
goto flushing;
|
|
}
|
|
|
|
/* preroll done, we can sync since we are in PLAYING now. */
|
|
GST_DEBUG_OBJECT (sink, "possibly waiting for clock to reach %"
|
|
GST_TIME_FORMAT, GST_TIME_ARGS (time));
|
|
|
|
/* wait for the clock, this can be interrupted because we got shut down or
|
|
* we PAUSED. */
|
|
status = gst_base_sink_wait_clock (sink, time, jitter);
|
|
|
|
GST_DEBUG_OBJECT (sink, "clock returned %d", status);
|
|
|
|
/* invalid time, no clock or sync disabled, just continue then */
|
|
if (status == GST_CLOCK_BADTIME)
|
|
break;
|
|
|
|
/* waiting could have been interrupted and we can be flushing now */
|
|
if (G_UNLIKELY (sink->flushing))
|
|
goto flushing;
|
|
|
|
/* retry if we got unscheduled, which means we did not reach the timeout
|
|
* yet. if some other error occures, we continue. */
|
|
} while (status == GST_CLOCK_UNSCHEDULED);
|
|
|
|
GST_DEBUG_OBJECT (sink, "end of stream");
|
|
|
|
return GST_FLOW_OK;
|
|
|
|
/* ERRORS */
|
|
flushing:
|
|
{
|
|
GST_DEBUG_OBJECT (sink, "we are flushing");
|
|
return GST_FLOW_WRONG_STATE;
|
|
}
|
|
}
|
|
|
|
/* with STREAM_LOCK, PREROLL_LOCK
|
|
*
|
|
* Make sure we are in PLAYING and synchronize an object to the clock.
|
|
*
|
|
* If we need preroll, we are not in PLAYING. We try to commit the state
|
|
* if needed and then block if we still are not PLAYING.
|
|
*
|
|
* We start waiting on the clock in PLAYING. If we got interrupted, we
|
|
* immediatly try to re-preroll.
|
|
*
|
|
* Some objects do not need synchronisation (most events) and so this function
|
|
* immediatly returns GST_FLOW_OK.
|
|
*
|
|
* for objects that arrive later than max-lateness to be synchronized to the
|
|
* clock have the @late boolean set to TRUE.
|
|
*
|
|
* This function keeps a running average of the jitter (the diff between the
|
|
* clock time and the requested sync time). The jitter is negative for
|
|
* objects that arrive in time and positive for late buffers.
|
|
*
|
|
* does not take ownership of obj.
|
|
*/
|
|
static GstFlowReturn
|
|
gst_base_sink_do_sync (GstBaseSink * basesink, GstPad * pad,
|
|
GstMiniObject * obj, gboolean * late)
|
|
{
|
|
GstClockTimeDiff jitter;
|
|
gboolean syncable;
|
|
GstClockReturn status = GST_CLOCK_OK;
|
|
GstClockTime rstart, rstop, sstart, sstop;
|
|
gboolean do_sync;
|
|
GstBaseSinkPrivate *priv;
|
|
|
|
priv = basesink->priv;
|
|
|
|
sstart = sstop = rstart = rstop = -1;
|
|
do_sync = TRUE;
|
|
|
|
priv->current_rstart = -1;
|
|
|
|
/* get timing information for this object against the render segment */
|
|
syncable = gst_base_sink_get_sync_times (basesink, obj,
|
|
&sstart, &sstop, &rstart, &rstop, &do_sync, &basesink->segment);
|
|
|
|
/* a syncable object needs to participate in preroll and
|
|
* clocking. All buffers and EOS are syncable. */
|
|
if (G_UNLIKELY (!syncable))
|
|
goto not_syncable;
|
|
|
|
/* store timing info for current object */
|
|
priv->current_rstart = rstart;
|
|
priv->current_rstop = (rstop != -1 ? rstop : rstart);
|
|
/* save sync time for eos when the previous object needed sync */
|
|
priv->eos_rtime = (do_sync ? priv->current_rstop : -1);
|
|
|
|
again:
|
|
/* first do preroll, this makes sure we commit our state
|
|
* to PAUSED and can continue to PLAYING. We cannot perform
|
|
* any clock sync in PAUSED because there is no clock.
|
|
*/
|
|
while (G_UNLIKELY (basesink->need_preroll)) {
|
|
GST_DEBUG_OBJECT (basesink, "prerolling object %p", obj);
|
|
|
|
if (G_LIKELY (basesink->playing_async)) {
|
|
/* commit state */
|
|
if (G_UNLIKELY (!gst_base_sink_commit_state (basesink)))
|
|
goto stopping;
|
|
}
|
|
|
|
/* need to recheck here because the commit state could have
|
|
* made us not need the preroll anymore */
|
|
if (G_LIKELY (basesink->need_preroll)) {
|
|
/* block until the state changes, or we get a flush, or something */
|
|
if (gst_base_sink_wait_preroll (basesink) != GST_FLOW_OK)
|
|
goto flushing;
|
|
}
|
|
}
|
|
|
|
/* After rendering we store the position of the last buffer so that we can use
|
|
* it to report the position. We need to take the lock here. */
|
|
GST_OBJECT_LOCK (basesink);
|
|
priv->current_sstart = sstart;
|
|
priv->current_sstop = (sstop != -1 ? sstop : sstart);
|
|
GST_OBJECT_UNLOCK (basesink);
|
|
|
|
if (!do_sync)
|
|
goto done;
|
|
|
|
/* preroll done, we can sync since we are in PLAYING now. */
|
|
GST_DEBUG_OBJECT (basesink, "possibly waiting for clock to reach %"
|
|
GST_TIME_FORMAT, GST_TIME_ARGS (rstart));
|
|
|
|
/* this function will return immediatly if start == -1, no clock
|
|
* or sync is disabled with GST_CLOCK_BADTIME. */
|
|
status = gst_base_sink_wait_clock (basesink, rstart, &jitter);
|
|
|
|
GST_DEBUG_OBJECT (basesink, "clock returned %d", status);
|
|
|
|
/* invalid time, no clock or sync disabled, just render */
|
|
if (status == GST_CLOCK_BADTIME)
|
|
goto done;
|
|
|
|
/* waiting could have been interrupted and we can be flushing now */
|
|
if (G_UNLIKELY (basesink->flushing))
|
|
goto flushing;
|
|
|
|
/* check for unlocked by a state change, we are not flushing so
|
|
* we can try to preroll on the current buffer. */
|
|
if (G_UNLIKELY (status == GST_CLOCK_UNSCHEDULED)) {
|
|
GST_DEBUG_OBJECT (basesink, "unscheduled, waiting some more");
|
|
goto again;
|
|
}
|
|
|
|
/* successful syncing done, record observation */
|
|
priv->current_jitter = jitter;
|
|
|
|
/* check if the object should be dropped */
|
|
*late = gst_base_sink_is_too_late (basesink, obj, rstart, rstop,
|
|
status, jitter);
|
|
|
|
done:
|
|
return GST_FLOW_OK;
|
|
|
|
/* ERRORS */
|
|
not_syncable:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "non syncable object %p", obj);
|
|
return GST_FLOW_OK;
|
|
}
|
|
flushing:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "we are flushing");
|
|
return GST_FLOW_WRONG_STATE;
|
|
}
|
|
stopping:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "stopping while commiting state");
|
|
return GST_FLOW_WRONG_STATE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_sink_send_qos (GstBaseSink * basesink,
|
|
gdouble proportion, GstClockTime time, GstClockTimeDiff diff)
|
|
{
|
|
GstEvent *event;
|
|
gboolean res;
|
|
|
|
/* generate Quality-of-Service event */
|
|
GST_CAT_DEBUG_OBJECT (GST_CAT_QOS, basesink,
|
|
"qos: proportion: %lf, diff %" G_GINT64_FORMAT ", timestamp %"
|
|
GST_TIME_FORMAT, proportion, diff, GST_TIME_ARGS (time));
|
|
|
|
event = gst_event_new_qos (proportion, diff, time);
|
|
|
|
/* send upstream */
|
|
res = gst_pad_push_event (basesink->sinkpad, event);
|
|
|
|
return res;
|
|
}
|
|
|
|
static void
|
|
gst_base_sink_perform_qos (GstBaseSink * sink, gboolean dropped)
|
|
{
|
|
GstBaseSinkPrivate *priv;
|
|
GstClockTime start, stop;
|
|
GstClockTimeDiff jitter;
|
|
GstClockTime pt, entered, left;
|
|
GstClockTime duration;
|
|
gdouble rate;
|
|
|
|
priv = sink->priv;
|
|
|
|
start = priv->current_rstart;
|
|
|
|
/* if Quality-of-Service disabled, do nothing */
|
|
if (!g_atomic_int_get (&priv->qos_enabled) || start == -1)
|
|
return;
|
|
|
|
stop = priv->current_rstop;
|
|
jitter = priv->current_jitter;
|
|
|
|
/* this is the time the buffer entered the sink */
|
|
entered = start + jitter;
|
|
/* this is the time the buffer left the sink */
|
|
left = start + (jitter < 0 ? 0 : jitter);
|
|
|
|
/* calculate duration of the buffer */
|
|
if (stop != -1)
|
|
duration = stop - start;
|
|
else
|
|
duration = -1;
|
|
|
|
/* if we have the time when the last buffer left us, calculate
|
|
* processing time */
|
|
if (priv->last_left != -1) {
|
|
if (entered > priv->last_left) {
|
|
pt = entered - priv->last_left;
|
|
} else {
|
|
pt = 0;
|
|
}
|
|
} else {
|
|
pt = priv->avg_pt;
|
|
}
|
|
|
|
GST_CAT_DEBUG_OBJECT (GST_CAT_QOS, sink, "start: %" GST_TIME_FORMAT
|
|
", entered %" GST_TIME_FORMAT ", left %" GST_TIME_FORMAT ", pt: %"
|
|
GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT ",jitter %"
|
|
G_GINT64_FORMAT, GST_TIME_ARGS (start), GST_TIME_ARGS (entered),
|
|
GST_TIME_ARGS (left), GST_TIME_ARGS (pt), GST_TIME_ARGS (duration),
|
|
jitter);
|
|
|
|
GST_CAT_DEBUG_OBJECT (GST_CAT_QOS, sink, "avg_duration: %" GST_TIME_FORMAT
|
|
", avg_pt: %" GST_TIME_FORMAT ", avg_rate: %g",
|
|
GST_TIME_ARGS (priv->avg_duration), GST_TIME_ARGS (priv->avg_pt),
|
|
priv->avg_rate);
|
|
|
|
/* collect running averages. for first observations, we copy the
|
|
* values */
|
|
if (priv->avg_duration == -1)
|
|
priv->avg_duration = duration;
|
|
else
|
|
priv->avg_duration = UPDATE_RUNNING_AVG (priv->avg_duration, duration);
|
|
|
|
if (priv->avg_pt == -1)
|
|
priv->avg_pt = pt;
|
|
else
|
|
priv->avg_pt = UPDATE_RUNNING_AVG (priv->avg_pt, pt);
|
|
|
|
if (priv->avg_duration != 0)
|
|
rate =
|
|
gst_guint64_to_gdouble (priv->avg_pt) /
|
|
gst_guint64_to_gdouble (priv->avg_duration);
|
|
else
|
|
rate = 0.0;
|
|
|
|
if (priv->last_left != -1) {
|
|
if (dropped || priv->avg_rate < 0.0) {
|
|
priv->avg_rate = rate;
|
|
} else {
|
|
if (rate > 1.0)
|
|
priv->avg_rate = UPDATE_RUNNING_AVG_N (priv->avg_rate, rate);
|
|
else
|
|
priv->avg_rate = UPDATE_RUNNING_AVG_P (priv->avg_rate, rate);
|
|
}
|
|
}
|
|
|
|
GST_CAT_DEBUG_OBJECT (GST_CAT_QOS, sink,
|
|
"updated: avg_duration: %" GST_TIME_FORMAT ", avg_pt: %" GST_TIME_FORMAT
|
|
", avg_rate: %g", GST_TIME_ARGS (priv->avg_duration),
|
|
GST_TIME_ARGS (priv->avg_pt), priv->avg_rate);
|
|
|
|
|
|
/* if we have a valid rate, start sending QoS messages */
|
|
if (priv->avg_rate >= 0.0) {
|
|
gst_base_sink_send_qos (sink, priv->avg_rate, priv->current_rstart,
|
|
priv->current_jitter);
|
|
}
|
|
|
|
/* record when this buffer will leave us */
|
|
priv->last_left = left;
|
|
}
|
|
|
|
/* reset all qos measuring */
|
|
static void
|
|
gst_base_sink_reset_qos (GstBaseSink * sink)
|
|
{
|
|
GstBaseSinkPrivate *priv;
|
|
|
|
priv = sink->priv;
|
|
|
|
priv->last_in_time = -1;
|
|
priv->last_left = -1;
|
|
priv->avg_duration = -1;
|
|
priv->avg_pt = -1;
|
|
priv->avg_rate = -1.0;
|
|
priv->avg_render = -1;
|
|
priv->rendered = 0;
|
|
priv->dropped = 0;
|
|
|
|
}
|
|
|
|
/* Checks if the object was scheduled too late.
|
|
*
|
|
* start/stop contain the raw timestamp start and stop values
|
|
* of the object.
|
|
*
|
|
* status and jitter contain the return values from the clock wait.
|
|
*
|
|
* returns TRUE if the buffer was too late.
|
|
*/
|
|
static gboolean
|
|
gst_base_sink_is_too_late (GstBaseSink * basesink, GstMiniObject * obj,
|
|
GstClockTime start, GstClockTime stop,
|
|
GstClockReturn status, GstClockTimeDiff jitter)
|
|
{
|
|
gboolean late;
|
|
gint64 max_lateness;
|
|
GstBaseSinkPrivate *priv;
|
|
|
|
priv = basesink->priv;
|
|
|
|
late = FALSE;
|
|
|
|
/* only for objects that were too late */
|
|
if (G_LIKELY (status != GST_CLOCK_EARLY))
|
|
goto in_time;
|
|
|
|
max_lateness = basesink->abidata.ABI.max_lateness;
|
|
|
|
/* check if frame dropping is enabled */
|
|
if (max_lateness == -1)
|
|
goto no_drop;
|
|
|
|
/* only check for buffers */
|
|
if (G_UNLIKELY (!GST_IS_BUFFER (obj)))
|
|
goto not_buffer;
|
|
|
|
/* can't do check if we don't have a timestamp */
|
|
if (G_UNLIKELY (start == -1))
|
|
goto no_timestamp;
|
|
|
|
/* we can add a valid stop time */
|
|
if (stop != -1)
|
|
max_lateness += stop;
|
|
else
|
|
max_lateness += start;
|
|
|
|
/* if the jitter bigger than duration and lateness we are too late */
|
|
if ((late = start + jitter > max_lateness)) {
|
|
GST_DEBUG_OBJECT (basesink, "buffer is too late %" GST_TIME_FORMAT
|
|
" > %" GST_TIME_FORMAT, GST_TIME_ARGS (start + jitter),
|
|
GST_TIME_ARGS (max_lateness));
|
|
/* !!emergency!!, if we did not receive anything valid for more than a
|
|
* second, render it anyway so the user sees something */
|
|
if (priv->last_in_time && start - priv->last_in_time > GST_SECOND) {
|
|
late = FALSE;
|
|
GST_DEBUG_OBJECT (basesink,
|
|
"**emergency** last buffer at %" GST_TIME_FORMAT " > GST_SECOND",
|
|
GST_TIME_ARGS (priv->last_in_time));
|
|
}
|
|
}
|
|
|
|
done:
|
|
if (!late) {
|
|
priv->last_in_time = start;
|
|
}
|
|
return late;
|
|
|
|
/* all is fine */
|
|
in_time:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "object was scheduled in time");
|
|
goto done;
|
|
}
|
|
no_drop:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "frame dropping disabled");
|
|
goto done;
|
|
}
|
|
not_buffer:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "object is not a buffer");
|
|
return FALSE;
|
|
}
|
|
no_timestamp:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "buffer has no timestamp");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_base_sink_do_render_stats (GstBaseSink * basesink, gboolean start)
|
|
{
|
|
GstBaseSinkPrivate *priv;
|
|
|
|
priv = basesink->priv;
|
|
|
|
if (start) {
|
|
priv->start = gst_util_get_timestamp ();
|
|
} else {
|
|
GstClockTime elapsed;
|
|
|
|
priv->stop = gst_util_get_timestamp ();
|
|
|
|
elapsed = GST_CLOCK_DIFF (priv->start, priv->stop);
|
|
|
|
if (priv->avg_render == -1)
|
|
priv->avg_render = elapsed;
|
|
else
|
|
priv->avg_render = UPDATE_RUNNING_AVG (priv->avg_render, elapsed);
|
|
|
|
GST_CAT_DEBUG_OBJECT (GST_CAT_QOS, basesink,
|
|
"avg_render: %" GST_TIME_FORMAT, GST_TIME_ARGS (priv->avg_render));
|
|
}
|
|
}
|
|
|
|
/* with STREAM_LOCK, PREROLL_LOCK,
|
|
*
|
|
* Synchronize the object on the clock and then render it.
|
|
*
|
|
* takes ownership of obj.
|
|
*/
|
|
static GstFlowReturn
|
|
gst_base_sink_render_object (GstBaseSink * basesink, GstPad * pad,
|
|
GstMiniObject * obj)
|
|
{
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
GstBaseSinkClass *bclass;
|
|
gboolean late = FALSE;
|
|
GstBaseSinkPrivate *priv;
|
|
|
|
priv = basesink->priv;
|
|
|
|
/* synchronize this object, non syncable objects return OK
|
|
* immediatly. */
|
|
ret = gst_base_sink_do_sync (basesink, pad, obj, &late);
|
|
if (G_UNLIKELY (ret != GST_FLOW_OK))
|
|
goto sync_failed;
|
|
|
|
/* and now render, event or buffer. */
|
|
if (G_LIKELY (GST_IS_BUFFER (obj))) {
|
|
GstBuffer *buf;
|
|
|
|
/* drop late buffers unconditionally, let's hope it's unlikely */
|
|
if (G_UNLIKELY (late))
|
|
goto dropped;
|
|
|
|
buf = GST_BUFFER_CAST (obj);
|
|
|
|
gst_base_sink_set_last_buffer (basesink, buf);
|
|
|
|
bclass = GST_BASE_SINK_GET_CLASS (basesink);
|
|
|
|
if (G_LIKELY (bclass->render)) {
|
|
gint do_qos;
|
|
|
|
/* read once, to get same value before and after */
|
|
do_qos = g_atomic_int_get (&priv->qos_enabled);
|
|
|
|
GST_DEBUG_OBJECT (basesink, "rendering buffer %p", obj);
|
|
|
|
/* record rendering time for QoS and stats */
|
|
if (do_qos)
|
|
gst_base_sink_do_render_stats (basesink, TRUE);
|
|
|
|
ret = bclass->render (basesink, buf);
|
|
|
|
priv->rendered++;
|
|
|
|
if (do_qos)
|
|
gst_base_sink_do_render_stats (basesink, FALSE);
|
|
}
|
|
} else {
|
|
GstEvent *event = GST_EVENT_CAST (obj);
|
|
gboolean event_res = TRUE;
|
|
GstEventType type;
|
|
|
|
bclass = GST_BASE_SINK_GET_CLASS (basesink);
|
|
|
|
type = GST_EVENT_TYPE (event);
|
|
|
|
GST_DEBUG_OBJECT (basesink, "rendering event %p, type %s", obj,
|
|
gst_event_type_get_name (type));
|
|
|
|
if (bclass->event)
|
|
event_res = bclass->event (basesink, event);
|
|
|
|
if (G_LIKELY (event_res)) {
|
|
switch (type) {
|
|
case GST_EVENT_EOS:
|
|
/* the EOS event is completely handled so we mark
|
|
* ourselves as being in the EOS state. eos is also
|
|
* protected by the object lock so we can read it when
|
|
* answering the POSITION query. */
|
|
GST_OBJECT_LOCK (basesink);
|
|
basesink->eos = TRUE;
|
|
GST_OBJECT_UNLOCK (basesink);
|
|
/* ok, now we can post the message */
|
|
GST_DEBUG_OBJECT (basesink, "Now posting EOS");
|
|
gst_element_post_message (GST_ELEMENT_CAST (basesink),
|
|
gst_message_new_eos (GST_OBJECT_CAST (basesink)));
|
|
break;
|
|
case GST_EVENT_NEWSEGMENT:
|
|
/* configure the segment */
|
|
gst_base_sink_configure_segment (basesink, pad, event,
|
|
&basesink->segment);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
|
|
done:
|
|
gst_base_sink_perform_qos (basesink, late);
|
|
|
|
GST_DEBUG_OBJECT (basesink, "object unref after render %p", obj);
|
|
gst_mini_object_unref (obj);
|
|
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
sync_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "do_sync returned %s", gst_flow_get_name (ret));
|
|
goto done;
|
|
}
|
|
dropped:
|
|
{
|
|
priv->dropped++;
|
|
GST_DEBUG_OBJECT (basesink, "buffer late, dropping");
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
/* with STREAM_LOCK, PREROLL_LOCK
|
|
*
|
|
* Perform preroll on the given object. For buffers this means
|
|
* calling the preroll subclass method.
|
|
* If that succeeds, the state will be commited.
|
|
*
|
|
* function does not take ownership of obj.
|
|
*/
|
|
static GstFlowReturn
|
|
gst_base_sink_preroll_object (GstBaseSink * basesink, GstPad * pad,
|
|
GstMiniObject * obj)
|
|
{
|
|
GstFlowReturn ret;
|
|
|
|
GST_DEBUG_OBJECT (basesink, "do preroll %p", obj);
|
|
|
|
/* if it's a buffer, we need to call the preroll method */
|
|
if (G_LIKELY (GST_IS_BUFFER (obj))) {
|
|
GstBaseSinkClass *bclass;
|
|
GstBuffer *buf;
|
|
GstClockTime timestamp;
|
|
|
|
buf = GST_BUFFER_CAST (obj);
|
|
timestamp = GST_BUFFER_TIMESTAMP (buf);
|
|
|
|
GST_DEBUG_OBJECT (basesink, "preroll buffer %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (timestamp));
|
|
|
|
gst_base_sink_set_last_buffer (basesink, buf);
|
|
|
|
bclass = GST_BASE_SINK_GET_CLASS (basesink);
|
|
if (bclass->preroll)
|
|
if ((ret = bclass->preroll (basesink, buf)) != GST_FLOW_OK)
|
|
goto preroll_failed;
|
|
}
|
|
|
|
/* commit state */
|
|
if (G_LIKELY (basesink->playing_async)) {
|
|
if (G_UNLIKELY (!gst_base_sink_commit_state (basesink)))
|
|
goto stopping;
|
|
}
|
|
|
|
return GST_FLOW_OK;
|
|
|
|
/* ERRORS */
|
|
preroll_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "preroll failed, abort state");
|
|
gst_element_abort_state (GST_ELEMENT_CAST (basesink));
|
|
return ret;
|
|
}
|
|
stopping:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "stopping while commiting state");
|
|
return GST_FLOW_WRONG_STATE;
|
|
}
|
|
}
|
|
|
|
/* with STREAM_LOCK, PREROLL_LOCK
|
|
*
|
|
* Queue an object for rendering.
|
|
* The first prerollable object queued will complete the preroll. If the
|
|
* preroll queue if filled, we render all the objects in the queue.
|
|
*
|
|
* This function takes ownership of the object.
|
|
*/
|
|
static GstFlowReturn
|
|
gst_base_sink_queue_object_unlocked (GstBaseSink * basesink, GstPad * pad,
|
|
GstMiniObject * obj, gboolean prerollable)
|
|
{
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
gint length;
|
|
GQueue *q;
|
|
|
|
if (G_UNLIKELY (basesink->need_preroll)) {
|
|
if (G_LIKELY (prerollable))
|
|
basesink->preroll_queued++;
|
|
|
|
length = basesink->preroll_queued;
|
|
|
|
GST_DEBUG_OBJECT (basesink, "now %d prerolled items", length);
|
|
|
|
/* first prerollable item needs to finish the preroll */
|
|
if (length == 1) {
|
|
ret = gst_base_sink_preroll_object (basesink, pad, obj);
|
|
if (G_UNLIKELY (ret != GST_FLOW_OK))
|
|
goto preroll_failed;
|
|
}
|
|
/* need to recheck if we need preroll, commmit state during preroll
|
|
* could have made us not need more preroll. */
|
|
if (G_UNLIKELY (basesink->need_preroll)) {
|
|
/* see if we can render now, if we can't add the object to the preroll
|
|
* queue. */
|
|
if (G_UNLIKELY (length <= basesink->preroll_queue_max_len))
|
|
goto more_preroll;
|
|
}
|
|
}
|
|
|
|
/* we can start rendering (or blocking) the queued object
|
|
* if any. */
|
|
q = basesink->preroll_queue;
|
|
while (G_UNLIKELY (!g_queue_is_empty (q))) {
|
|
GstMiniObject *o;
|
|
|
|
o = g_queue_pop_head (q);
|
|
GST_DEBUG_OBJECT (basesink, "rendering queued object %p", o);
|
|
|
|
/* do something with the return value */
|
|
ret = gst_base_sink_render_object (basesink, pad, o);
|
|
if (ret != GST_FLOW_OK)
|
|
goto dequeue_failed;
|
|
}
|
|
|
|
/* now render the object */
|
|
ret = gst_base_sink_render_object (basesink, pad, obj);
|
|
basesink->preroll_queued = 0;
|
|
|
|
return ret;
|
|
|
|
/* special cases */
|
|
preroll_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "preroll failed, reason %s",
|
|
gst_flow_get_name (ret));
|
|
gst_mini_object_unref (obj);
|
|
return ret;
|
|
}
|
|
more_preroll:
|
|
{
|
|
/* add object to the queue and return */
|
|
GST_DEBUG_OBJECT (basesink, "need more preroll data %d <= %d",
|
|
length, basesink->preroll_queue_max_len);
|
|
g_queue_push_tail (basesink->preroll_queue, obj);
|
|
return GST_FLOW_OK;
|
|
}
|
|
dequeue_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "rendering queued objects failed, reason %s",
|
|
gst_flow_get_name (ret));
|
|
gst_mini_object_unref (obj);
|
|
return ret;
|
|
}
|
|
}
|
|
|
|
/* with STREAM_LOCK
|
|
*
|
|
* This function grabs the PREROLL_LOCK and adds the object to
|
|
* the queue.
|
|
*
|
|
* This function takes ownership of obj.
|
|
*/
|
|
static GstFlowReturn
|
|
gst_base_sink_queue_object (GstBaseSink * basesink, GstPad * pad,
|
|
GstMiniObject * obj, gboolean prerollable)
|
|
{
|
|
GstFlowReturn ret;
|
|
|
|
GST_PAD_PREROLL_LOCK (pad);
|
|
if (G_UNLIKELY (basesink->flushing))
|
|
goto flushing;
|
|
|
|
if (G_UNLIKELY (basesink->priv->received_eos))
|
|
goto was_eos;
|
|
|
|
ret = gst_base_sink_queue_object_unlocked (basesink, pad, obj, prerollable);
|
|
GST_PAD_PREROLL_UNLOCK (pad);
|
|
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
flushing:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "sink is flushing");
|
|
GST_PAD_PREROLL_UNLOCK (pad);
|
|
gst_mini_object_unref (obj);
|
|
return GST_FLOW_WRONG_STATE;
|
|
}
|
|
was_eos:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink,
|
|
"we are EOS, dropping object, return UNEXPECTED");
|
|
GST_PAD_PREROLL_UNLOCK (pad);
|
|
gst_mini_object_unref (obj);
|
|
return GST_FLOW_UNEXPECTED;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_sink_event (GstPad * pad, GstEvent * event)
|
|
{
|
|
GstBaseSink *basesink;
|
|
gboolean result = TRUE;
|
|
GstBaseSinkClass *bclass;
|
|
|
|
basesink = GST_BASE_SINK (gst_pad_get_parent (pad));
|
|
|
|
bclass = GST_BASE_SINK_GET_CLASS (basesink);
|
|
|
|
GST_DEBUG_OBJECT (basesink, "event %p (%s)", event,
|
|
GST_EVENT_TYPE_NAME (event));
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_EOS:
|
|
{
|
|
GstFlowReturn ret;
|
|
|
|
GST_PAD_PREROLL_LOCK (pad);
|
|
if (G_UNLIKELY (basesink->flushing))
|
|
goto flushing;
|
|
|
|
if (G_UNLIKELY (basesink->priv->received_eos)) {
|
|
/* we can't accept anything when we are EOS */
|
|
result = FALSE;
|
|
gst_event_unref (event);
|
|
} else {
|
|
/* we set the received EOS flag here so that we can use it when testing if
|
|
* we are prerolled and to refure more buffers. */
|
|
basesink->priv->received_eos = TRUE;
|
|
|
|
/* EOS is a prerollable object, we call the unlocked version because it
|
|
* does not check the received_eos flag. */
|
|
ret = gst_base_sink_queue_object_unlocked (basesink, pad,
|
|
GST_MINI_OBJECT_CAST (event), TRUE);
|
|
if (G_UNLIKELY (ret != GST_FLOW_OK))
|
|
result = FALSE;
|
|
}
|
|
GST_PAD_PREROLL_UNLOCK (pad);
|
|
break;
|
|
}
|
|
case GST_EVENT_NEWSEGMENT:
|
|
{
|
|
GstFlowReturn ret;
|
|
|
|
GST_DEBUG_OBJECT (basesink, "newsegment %p", event);
|
|
|
|
GST_PAD_PREROLL_LOCK (pad);
|
|
if (G_UNLIKELY (basesink->flushing))
|
|
goto flushing;
|
|
|
|
if (G_UNLIKELY (basesink->priv->received_eos)) {
|
|
/* we can't accept anything when we are EOS */
|
|
result = FALSE;
|
|
gst_event_unref (event);
|
|
} else {
|
|
/* the new segment is a non prerollable item and does not block anything,
|
|
* we need to configure the current clipping segment and insert the event
|
|
* in the queue to serialize it with the buffers for rendering. */
|
|
gst_base_sink_configure_segment (basesink, pad, event,
|
|
basesink->abidata.ABI.clip_segment);
|
|
|
|
ret =
|
|
gst_base_sink_queue_object_unlocked (basesink, pad,
|
|
GST_MINI_OBJECT_CAST (event), FALSE);
|
|
if (G_UNLIKELY (ret != GST_FLOW_OK))
|
|
result = FALSE;
|
|
else
|
|
basesink->have_newsegment = TRUE;
|
|
}
|
|
GST_PAD_PREROLL_UNLOCK (pad);
|
|
break;
|
|
}
|
|
case GST_EVENT_FLUSH_START:
|
|
if (bclass->event)
|
|
bclass->event (basesink, event);
|
|
|
|
GST_DEBUG_OBJECT (basesink, "flush-start %p", event);
|
|
|
|
/* make sure we are not blocked on the clock also clear any pending
|
|
* eos state. */
|
|
gst_base_sink_set_flushing (basesink, pad, TRUE);
|
|
|
|
/* we grab the stream lock but that is not needed since setting the
|
|
* sink to flushing would make sure no state commit is being done
|
|
* anymore */
|
|
GST_PAD_STREAM_LOCK (pad);
|
|
gst_base_sink_reset_qos (basesink);
|
|
if (basesink->priv->async_enabled) {
|
|
/* and we need to commit our state again on the next
|
|
* prerolled buffer */
|
|
basesink->playing_async = TRUE;
|
|
gst_element_lost_state (GST_ELEMENT_CAST (basesink));
|
|
} else {
|
|
basesink->priv->have_latency = TRUE;
|
|
basesink->need_preroll = FALSE;
|
|
}
|
|
gst_base_sink_set_last_buffer (basesink, NULL);
|
|
GST_PAD_STREAM_UNLOCK (pad);
|
|
|
|
gst_event_unref (event);
|
|
break;
|
|
case GST_EVENT_FLUSH_STOP:
|
|
if (bclass->event)
|
|
bclass->event (basesink, event);
|
|
|
|
GST_DEBUG_OBJECT (basesink, "flush-stop %p", event);
|
|
|
|
/* unset flushing so we can accept new data, this also flushes out any EOS
|
|
* event. */
|
|
gst_base_sink_set_flushing (basesink, pad, FALSE);
|
|
|
|
/* we need new segment info after the flush. */
|
|
gst_segment_init (&basesink->segment, GST_FORMAT_UNDEFINED);
|
|
gst_segment_init (basesink->abidata.ABI.clip_segment,
|
|
GST_FORMAT_UNDEFINED);
|
|
basesink->have_newsegment = FALSE;
|
|
|
|
/* for position reporting */
|
|
GST_OBJECT_LOCK (basesink);
|
|
basesink->priv->current_sstart = -1;
|
|
basesink->priv->current_sstop = -1;
|
|
GST_OBJECT_UNLOCK (basesink);
|
|
|
|
gst_event_unref (event);
|
|
break;
|
|
default:
|
|
/* other events are sent to queue or subclass depending on if they
|
|
* are serialized. */
|
|
if (GST_EVENT_IS_SERIALIZED (event)) {
|
|
gst_base_sink_queue_object (basesink, pad,
|
|
GST_MINI_OBJECT_CAST (event), FALSE);
|
|
} else {
|
|
if (bclass->event)
|
|
bclass->event (basesink, event);
|
|
gst_event_unref (event);
|
|
}
|
|
break;
|
|
}
|
|
done:
|
|
gst_object_unref (basesink);
|
|
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
flushing:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "we are flushing");
|
|
GST_PAD_PREROLL_UNLOCK (pad);
|
|
result = FALSE;
|
|
gst_event_unref (event);
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
/* default implementation to calculate the start and end
|
|
* timestamps on a buffer, subclasses can override
|
|
*/
|
|
static void
|
|
gst_base_sink_get_times (GstBaseSink * basesink, GstBuffer * buffer,
|
|
GstClockTime * start, GstClockTime * end)
|
|
{
|
|
GstClockTime timestamp, duration;
|
|
|
|
timestamp = GST_BUFFER_TIMESTAMP (buffer);
|
|
if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
|
|
|
|
/* get duration to calculate end time */
|
|
duration = GST_BUFFER_DURATION (buffer);
|
|
if (GST_CLOCK_TIME_IS_VALID (duration)) {
|
|
*end = timestamp + duration;
|
|
}
|
|
*start = timestamp;
|
|
}
|
|
}
|
|
|
|
/* must be called with PREROLL_LOCK */
|
|
static gboolean
|
|
gst_base_sink_needs_preroll (GstBaseSink * basesink)
|
|
{
|
|
gboolean is_prerolled, res;
|
|
|
|
/* we have 2 cases where the PREROLL_LOCK is released:
|
|
* 1) we are blocking in the PREROLL_LOCK and thus are prerolled.
|
|
* 2) we are syncing on the clock
|
|
*/
|
|
is_prerolled = basesink->have_preroll || basesink->priv->received_eos;
|
|
res = !is_prerolled && basesink->pad_mode != GST_ACTIVATE_PULL;
|
|
GST_DEBUG_OBJECT (basesink, "have_preroll: %d, EOS: %d => needs preroll: %d",
|
|
basesink->have_preroll, basesink->priv->received_eos, res);
|
|
|
|
return res;
|
|
}
|
|
|
|
/* with STREAM_LOCK, PREROLL_LOCK
|
|
*
|
|
* Takes a buffer and compare the timestamps with the last segment.
|
|
* If the buffer falls outside of the segment boundaries, drop it.
|
|
* Else queue the buffer for preroll and rendering.
|
|
*
|
|
* This function takes ownership of the buffer.
|
|
*/
|
|
static GstFlowReturn
|
|
gst_base_sink_chain_unlocked (GstBaseSink * basesink, GstPad * pad,
|
|
GstBuffer * buf)
|
|
{
|
|
GstFlowReturn result;
|
|
GstClockTime start = GST_CLOCK_TIME_NONE, end = GST_CLOCK_TIME_NONE;
|
|
GstSegment *clip_segment;
|
|
|
|
if (G_UNLIKELY (basesink->flushing))
|
|
goto flushing;
|
|
|
|
if (G_UNLIKELY (basesink->priv->received_eos))
|
|
goto was_eos;
|
|
|
|
/* for code clarity */
|
|
clip_segment = basesink->abidata.ABI.clip_segment;
|
|
|
|
if (G_UNLIKELY (!basesink->have_newsegment)) {
|
|
gboolean sync;
|
|
|
|
sync = gst_base_sink_get_sync (basesink);
|
|
if (sync) {
|
|
GST_ELEMENT_WARNING (basesink, STREAM, FAILED,
|
|
(_("Internal data flow problem.")),
|
|
("Received buffer without a new-segment. Assuming timestamps start from 0."));
|
|
}
|
|
|
|
basesink->have_newsegment = TRUE;
|
|
/* this means this sink will assume timestamps start from 0 */
|
|
clip_segment->start = 0;
|
|
clip_segment->stop = -1;
|
|
basesink->segment.start = 0;
|
|
basesink->segment.stop = -1;
|
|
}
|
|
|
|
/* check if the buffer needs to be dropped */
|
|
/* we don't use the subclassed method as it may not return
|
|
* valid values for our purpose here */
|
|
gst_base_sink_get_times (basesink, buf, &start, &end);
|
|
|
|
GST_DEBUG_OBJECT (basesink, "got times start: %" GST_TIME_FORMAT
|
|
", end: %" GST_TIME_FORMAT, GST_TIME_ARGS (start), GST_TIME_ARGS (end));
|
|
|
|
/* a dropped buffer does not participate in anything */
|
|
if (GST_CLOCK_TIME_IS_VALID (start) &&
|
|
(clip_segment->format == GST_FORMAT_TIME)) {
|
|
if (G_UNLIKELY (!gst_segment_clip (clip_segment,
|
|
GST_FORMAT_TIME, (gint64) start, (gint64) end, NULL, NULL)))
|
|
goto out_of_segment;
|
|
}
|
|
|
|
/* now we can process the buffer in the queue, this function takes ownership
|
|
* of the buffer */
|
|
result = gst_base_sink_queue_object_unlocked (basesink, pad,
|
|
GST_MINI_OBJECT_CAST (buf), TRUE);
|
|
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
flushing:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "sink is flushing");
|
|
gst_buffer_unref (buf);
|
|
return GST_FLOW_WRONG_STATE;
|
|
}
|
|
was_eos:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink,
|
|
"we are EOS, dropping object, return UNEXPECTED");
|
|
gst_buffer_unref (buf);
|
|
return GST_FLOW_UNEXPECTED;
|
|
}
|
|
out_of_segment:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "dropping buffer, out of clipping segment");
|
|
gst_buffer_unref (buf);
|
|
return GST_FLOW_OK;
|
|
}
|
|
}
|
|
|
|
/* with STREAM_LOCK
|
|
*/
|
|
static GstFlowReturn
|
|
gst_base_sink_chain (GstPad * pad, GstBuffer * buf)
|
|
{
|
|
GstBaseSink *basesink;
|
|
GstFlowReturn result;
|
|
|
|
basesink = GST_BASE_SINK (GST_OBJECT_PARENT (pad));
|
|
|
|
if (G_UNLIKELY (basesink->pad_mode != GST_ACTIVATE_PUSH))
|
|
goto wrong_mode;
|
|
|
|
GST_PAD_PREROLL_LOCK (pad);
|
|
result = gst_base_sink_chain_unlocked (basesink, pad, buf);
|
|
GST_PAD_PREROLL_UNLOCK (pad);
|
|
|
|
done:
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
wrong_mode:
|
|
{
|
|
GST_OBJECT_LOCK (pad);
|
|
GST_WARNING_OBJECT (basesink,
|
|
"Push on pad %s:%s, but it was not activated in push mode",
|
|
GST_DEBUG_PAD_NAME (pad));
|
|
GST_OBJECT_UNLOCK (pad);
|
|
gst_buffer_unref (buf);
|
|
/* we don't post an error message this will signal to the peer
|
|
* pushing that EOS is reached. */
|
|
result = GST_FLOW_UNEXPECTED;
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
/* with STREAM_LOCK
|
|
*/
|
|
static void
|
|
gst_base_sink_loop (GstPad * pad)
|
|
{
|
|
GstBaseSink *basesink;
|
|
GstBuffer *buf = NULL;
|
|
GstFlowReturn result;
|
|
|
|
basesink = GST_BASE_SINK (GST_OBJECT_PARENT (pad));
|
|
|
|
g_assert (basesink->pad_mode == GST_ACTIVATE_PULL);
|
|
|
|
GST_DEBUG_OBJECT (basesink, "pulling %" G_GUINT64_FORMAT ", %u",
|
|
basesink->offset, (guint) DEFAULT_SIZE);
|
|
|
|
result = gst_pad_pull_range (pad, basesink->offset, DEFAULT_SIZE, &buf);
|
|
if (G_UNLIKELY (result != GST_FLOW_OK))
|
|
goto paused;
|
|
|
|
if (G_UNLIKELY (buf == NULL))
|
|
goto no_buffer;
|
|
|
|
basesink->offset += GST_BUFFER_SIZE (buf);
|
|
|
|
GST_PAD_PREROLL_LOCK (pad);
|
|
result = gst_base_sink_chain_unlocked (basesink, pad, buf);
|
|
GST_PAD_PREROLL_UNLOCK (pad);
|
|
if (G_UNLIKELY (result != GST_FLOW_OK))
|
|
goto paused;
|
|
|
|
return;
|
|
|
|
/* ERRORS */
|
|
paused:
|
|
{
|
|
GST_LOG_OBJECT (basesink, "pausing task, reason %s",
|
|
gst_flow_get_name (result));
|
|
gst_pad_pause_task (pad);
|
|
/* fatal errors and NOT_LINKED cause EOS */
|
|
if (GST_FLOW_IS_FATAL (result) || result == GST_FLOW_NOT_LINKED) {
|
|
/* FIXME, we shouldn't post EOS when we are operating in segment mode */
|
|
gst_base_sink_event (pad, gst_event_new_eos ());
|
|
/* EOS does not cause an ERROR message */
|
|
if (result != GST_FLOW_UNEXPECTED) {
|
|
GST_ELEMENT_ERROR (basesink, STREAM, FAILED,
|
|
(_("Internal data stream error.")),
|
|
("stream stopped, reason %s", gst_flow_get_name (result)));
|
|
}
|
|
}
|
|
return;
|
|
}
|
|
no_buffer:
|
|
{
|
|
GST_LOG_OBJECT (basesink, "no buffer, pausing");
|
|
result = GST_FLOW_ERROR;
|
|
goto paused;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_sink_set_flushing (GstBaseSink * basesink, GstPad * pad,
|
|
gboolean flushing)
|
|
{
|
|
GstBaseSinkClass *bclass;
|
|
|
|
bclass = GST_BASE_SINK_GET_CLASS (basesink);
|
|
|
|
if (flushing) {
|
|
/* unlock any subclasses, we need to do this before grabbing the
|
|
* PREROLL_LOCK since we hold this lock before going into ::render. */
|
|
if (bclass->unlock)
|
|
bclass->unlock (basesink);
|
|
}
|
|
|
|
GST_PAD_PREROLL_LOCK (pad);
|
|
basesink->flushing = flushing;
|
|
if (flushing) {
|
|
/* step 1, now that we have the PREROLL lock, clear our unlock request */
|
|
if (bclass->unlock_stop)
|
|
bclass->unlock_stop (basesink);
|
|
|
|
/* set need_preroll before we unblock the clock. If the clock is unblocked
|
|
* before timing out, we can reuse the buffer for preroll. */
|
|
basesink->need_preroll = TRUE;
|
|
|
|
/* step 2, unblock clock sync (if any) or any other blocking thing */
|
|
if (basesink->clock_id) {
|
|
gst_clock_id_unschedule (basesink->clock_id);
|
|
}
|
|
|
|
/* flush out the data thread if it's locked in finish_preroll */
|
|
GST_DEBUG_OBJECT (basesink,
|
|
"flushing out data thread, need preroll to TRUE");
|
|
gst_base_sink_preroll_queue_flush (basesink, pad);
|
|
}
|
|
GST_PAD_PREROLL_UNLOCK (pad);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_sink_default_activate_pull (GstBaseSink * basesink, gboolean active)
|
|
{
|
|
gboolean result;
|
|
|
|
if (active) {
|
|
/* start task */
|
|
result = gst_pad_start_task (basesink->sinkpad,
|
|
(GstTaskFunction) gst_base_sink_loop, basesink->sinkpad);
|
|
} else {
|
|
/* step 2, make sure streaming finishes */
|
|
result = gst_pad_stop_task (basesink->sinkpad);
|
|
}
|
|
|
|
return result;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_sink_pad_activate (GstPad * pad)
|
|
{
|
|
gboolean result = FALSE;
|
|
GstBaseSink *basesink;
|
|
|
|
basesink = GST_BASE_SINK (gst_pad_get_parent (pad));
|
|
|
|
GST_DEBUG_OBJECT (basesink, "Trying pull mode first");
|
|
|
|
gst_base_sink_set_flushing (basesink, pad, FALSE);
|
|
|
|
if (basesink->can_activate_pull && gst_pad_check_pull_range (pad)
|
|
&& gst_pad_activate_pull (pad, TRUE)) {
|
|
GST_DEBUG_OBJECT (basesink, "Success activating pull mode");
|
|
result = TRUE;
|
|
} else {
|
|
GST_DEBUG_OBJECT (basesink, "Falling back to push mode");
|
|
if (gst_pad_activate_push (pad, TRUE)) {
|
|
GST_DEBUG_OBJECT (basesink, "Success activating push mode");
|
|
result = TRUE;
|
|
}
|
|
}
|
|
|
|
if (!result) {
|
|
GST_WARNING_OBJECT (basesink, "Could not activate pad in either mode");
|
|
gst_base_sink_set_flushing (basesink, pad, TRUE);
|
|
}
|
|
|
|
gst_object_unref (basesink);
|
|
|
|
return result;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_sink_pad_activate_push (GstPad * pad, gboolean active)
|
|
{
|
|
gboolean result;
|
|
GstBaseSink *basesink;
|
|
|
|
basesink = GST_BASE_SINK (gst_pad_get_parent (pad));
|
|
|
|
if (active) {
|
|
if (!basesink->can_activate_push) {
|
|
result = FALSE;
|
|
basesink->pad_mode = GST_ACTIVATE_NONE;
|
|
} else {
|
|
result = TRUE;
|
|
basesink->pad_mode = GST_ACTIVATE_PUSH;
|
|
}
|
|
} else {
|
|
if (G_UNLIKELY (basesink->pad_mode != GST_ACTIVATE_PUSH)) {
|
|
g_warning ("Internal GStreamer activation error!!!");
|
|
result = FALSE;
|
|
} else {
|
|
gst_base_sink_set_flushing (basesink, pad, TRUE);
|
|
result = TRUE;
|
|
basesink->pad_mode = GST_ACTIVATE_NONE;
|
|
}
|
|
}
|
|
|
|
gst_object_unref (basesink);
|
|
|
|
return result;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_sink_negotiate_pull (GstBaseSink * basesink)
|
|
{
|
|
GstCaps *caps;
|
|
GstPad *pad;
|
|
|
|
GST_OBJECT_LOCK (basesink);
|
|
pad = basesink->sinkpad;
|
|
gst_object_ref (pad);
|
|
GST_OBJECT_UNLOCK (basesink);
|
|
|
|
caps = gst_pad_get_allowed_caps (pad);
|
|
if (gst_caps_is_empty (caps))
|
|
goto no_caps_possible;
|
|
|
|
caps = gst_caps_make_writable (caps);
|
|
gst_caps_truncate (caps);
|
|
gst_pad_fixate_caps (pad, 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() */
|
|
} else {
|
|
if (!gst_pad_set_caps (pad, caps))
|
|
goto could_not_set_caps;
|
|
}
|
|
|
|
gst_caps_unref (caps);
|
|
gst_object_unref (pad);
|
|
|
|
return TRUE;
|
|
|
|
no_caps_possible:
|
|
{
|
|
GST_INFO_OBJECT (basesink, "Pipeline could not agree on caps");
|
|
GST_DEBUG_OBJECT (basesink, "get_allowed_caps() returned EMPTY");
|
|
gst_object_unref (pad);
|
|
return FALSE;
|
|
}
|
|
could_not_set_caps:
|
|
{
|
|
GST_INFO_OBJECT (basesink, "Could not set caps: %" GST_PTR_FORMAT, caps);
|
|
gst_caps_unref (caps);
|
|
gst_object_unref (pad);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/* this won't get called until we implement an activate function */
|
|
static gboolean
|
|
gst_base_sink_pad_activate_pull (GstPad * pad, gboolean active)
|
|
{
|
|
gboolean result = FALSE;
|
|
GstBaseSink *basesink;
|
|
GstBaseSinkClass *bclass;
|
|
|
|
basesink = GST_BASE_SINK (gst_pad_get_parent (pad));
|
|
bclass = GST_BASE_SINK_GET_CLASS (basesink);
|
|
|
|
if (active) {
|
|
if (!basesink->can_activate_pull) {
|
|
result = FALSE;
|
|
basesink->pad_mode = GST_ACTIVATE_NONE;
|
|
} else {
|
|
GstPad *peer = gst_pad_get_peer (pad);
|
|
|
|
if (G_UNLIKELY (peer == NULL)) {
|
|
g_warning ("Trying to activate pad in pull mode, but no peer");
|
|
result = FALSE;
|
|
basesink->pad_mode = GST_ACTIVATE_NONE;
|
|
} else {
|
|
if (gst_pad_activate_pull (peer, TRUE)) {
|
|
/* we mark we have a newsegment here because pull based
|
|
* mode works just fine without having a newsegment before the
|
|
* first buffer */
|
|
gst_segment_init (&basesink->segment, GST_FORMAT_UNDEFINED);
|
|
gst_segment_init (basesink->abidata.ABI.clip_segment,
|
|
GST_FORMAT_UNDEFINED);
|
|
basesink->have_newsegment = TRUE;
|
|
|
|
/* set the pad mode before starting the task so that it's in the
|
|
correct state for the new thread. also the sink set_caps function
|
|
checks this */
|
|
basesink->pad_mode = GST_ACTIVATE_PULL;
|
|
if ((result = gst_base_sink_negotiate_pull (basesink))) {
|
|
if (bclass->activate_pull)
|
|
result = bclass->activate_pull (basesink, TRUE);
|
|
else
|
|
result = FALSE;
|
|
}
|
|
/* but if starting the thread fails, set it back */
|
|
if (!result)
|
|
basesink->pad_mode = GST_ACTIVATE_NONE;
|
|
} else {
|
|
GST_DEBUG_OBJECT (pad, "Failed to activate peer in pull mode");
|
|
result = FALSE;
|
|
basesink->pad_mode = GST_ACTIVATE_NONE;
|
|
}
|
|
gst_object_unref (peer);
|
|
}
|
|
}
|
|
} else {
|
|
if (G_UNLIKELY (basesink->pad_mode != GST_ACTIVATE_PULL)) {
|
|
g_warning ("Internal GStreamer activation error!!!");
|
|
result = FALSE;
|
|
} else {
|
|
result = gst_base_sink_set_flushing (basesink, pad, TRUE);
|
|
if (bclass->activate_pull)
|
|
result &= bclass->activate_pull (basesink, FALSE);
|
|
basesink->pad_mode = GST_ACTIVATE_NONE;
|
|
}
|
|
}
|
|
|
|
gst_object_unref (basesink);
|
|
|
|
return result;
|
|
}
|
|
|
|
/* send an event to our sinkpad peer. */
|
|
static gboolean
|
|
gst_base_sink_send_event (GstElement * element, GstEvent * event)
|
|
{
|
|
GstPad *pad;
|
|
GstBaseSink *basesink = GST_BASE_SINK (element);
|
|
gboolean forward, result = TRUE;
|
|
|
|
/* only push UPSTREAM events upstream */
|
|
forward = GST_EVENT_IS_UPSTREAM (event);
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_LATENCY:
|
|
{
|
|
GstClockTime latency;
|
|
|
|
gst_event_parse_latency (event, &latency);
|
|
|
|
GST_OBJECT_LOCK (element);
|
|
basesink->priv->latency = latency;
|
|
GST_OBJECT_UNLOCK (element);
|
|
GST_DEBUG_OBJECT (basesink, "latency set to %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (latency));
|
|
|
|
/* don't forward, yet */
|
|
forward = FALSE;
|
|
break;
|
|
}
|
|
default:
|
|
break;
|
|
}
|
|
|
|
if (forward) {
|
|
GST_OBJECT_LOCK (element);
|
|
pad = gst_object_ref (basesink->sinkpad);
|
|
GST_OBJECT_UNLOCK (element);
|
|
|
|
result = gst_pad_push_event (pad, event);
|
|
|
|
gst_object_unref (pad);
|
|
} else {
|
|
/* not forwarded, unref the event */
|
|
gst_event_unref (event);
|
|
}
|
|
return result;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_sink_peer_query (GstBaseSink * sink, GstQuery * query)
|
|
{
|
|
GstPad *peer;
|
|
gboolean res = FALSE;
|
|
|
|
if ((peer = gst_pad_get_peer (sink->sinkpad))) {
|
|
res = gst_pad_query (peer, query);
|
|
gst_object_unref (peer);
|
|
}
|
|
return res;
|
|
}
|
|
|
|
/* get the end position of the last seen object, this is used
|
|
* for EOS and for making sure that we don't report a position we
|
|
* have not reached yet. */
|
|
static gboolean
|
|
gst_base_sink_get_position_last (GstBaseSink * basesink, gint64 * cur)
|
|
{
|
|
/* return last observed stream time */
|
|
*cur = basesink->priv->current_sstop;
|
|
GST_DEBUG_OBJECT (basesink, "POSITION: %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (*cur));
|
|
return TRUE;
|
|
}
|
|
|
|
/* get the position when we are PAUSED, this is the stream time of the buffer
|
|
* that prerolled. If no buffer is prerolled (we are still flushing), this
|
|
* value will be -1. */
|
|
static gboolean
|
|
gst_base_sink_get_position_paused (GstBaseSink * basesink, gint64 * cur)
|
|
{
|
|
gboolean res;
|
|
gint64 time;
|
|
GstSegment *segment;
|
|
|
|
*cur = basesink->priv->current_sstart;
|
|
segment = basesink->abidata.ABI.clip_segment;
|
|
|
|
time = segment->time;
|
|
|
|
if (*cur != -1) {
|
|
*cur = MAX (*cur, time);
|
|
GST_DEBUG_OBJECT (basesink, "POSITION as max: %" GST_TIME_FORMAT
|
|
", time %" GST_TIME_FORMAT, GST_TIME_ARGS (*cur), GST_TIME_ARGS (time));
|
|
} else {
|
|
/* we have no buffer, use the segment times. */
|
|
if (segment->rate >= 0.0) {
|
|
/* forward, next position is always the time of the segment */
|
|
*cur = time;
|
|
GST_DEBUG_OBJECT (basesink, "POSITION as time: %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (*cur));
|
|
} else {
|
|
/* reverse, next expected timestamp is segment->stop. We use the function
|
|
* to get things right for negative applied_rates. */
|
|
*cur =
|
|
gst_segment_to_stream_time (segment, GST_FORMAT_TIME, segment->stop);
|
|
GST_DEBUG_OBJECT (basesink, "reverse POSITION: %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (*cur));
|
|
}
|
|
}
|
|
res = (*cur != -1);
|
|
|
|
return res;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_sink_get_position (GstBaseSink * basesink, GstFormat format,
|
|
gint64 * cur)
|
|
{
|
|
GstClock *clock;
|
|
gboolean res = FALSE;
|
|
|
|
switch (format) {
|
|
/* we can answer time format */
|
|
case GST_FORMAT_TIME:
|
|
{
|
|
GstClockTime now, base, latency;
|
|
gint64 time, accum, duration;
|
|
gdouble rate;
|
|
gint64 last;
|
|
|
|
GST_OBJECT_LOCK (basesink);
|
|
|
|
/* can only give answer based on the clock if not EOS */
|
|
if (G_UNLIKELY (basesink->eos))
|
|
goto in_eos;
|
|
|
|
/* in PAUSE we cannot read from the clock so we
|
|
* report time based on the last seen timestamp. */
|
|
if (GST_STATE (basesink) == GST_STATE_PAUSED)
|
|
goto in_pause;
|
|
|
|
/* We get position from clock only in PLAYING, we checked
|
|
* the PAUSED case above, so this is check is to test
|
|
* READY and NULL, where the position is always 0 */
|
|
if (GST_STATE (basesink) != GST_STATE_PLAYING)
|
|
goto wrong_state;
|
|
|
|
/* we need to sync on the clock. */
|
|
if (basesink->sync == FALSE)
|
|
goto no_sync;
|
|
|
|
/* and we need a clock */
|
|
if (G_UNLIKELY ((clock = GST_ELEMENT_CLOCK (basesink)) == NULL))
|
|
goto no_sync;
|
|
|
|
/* collect all data we need holding the lock */
|
|
if (GST_CLOCK_TIME_IS_VALID (basesink->segment.time))
|
|
time = basesink->segment.time;
|
|
else
|
|
time = 0;
|
|
|
|
if (GST_CLOCK_TIME_IS_VALID (basesink->segment.stop))
|
|
duration = basesink->segment.stop - basesink->segment.start;
|
|
else
|
|
duration = 0;
|
|
|
|
base = GST_ELEMENT_CAST (basesink)->base_time;
|
|
accum = basesink->segment.accum;
|
|
rate = basesink->segment.rate * basesink->segment.applied_rate;
|
|
gst_base_sink_get_position_last (basesink, &last);
|
|
latency = basesink->priv->latency;
|
|
|
|
gst_object_ref (clock);
|
|
/* need to release the object lock before we can get the time,
|
|
* a clock might take the LOCK of the provider, which could be
|
|
* a basesink subclass. */
|
|
GST_OBJECT_UNLOCK (basesink);
|
|
|
|
now = gst_clock_get_time (clock);
|
|
|
|
/* subtract base time and accumulated time from the clock time.
|
|
* Make sure we don't go negative. This is the current time in
|
|
* the segment which we need to scale with the combined
|
|
* rate and applied rate. */
|
|
base += accum;
|
|
base += latency;
|
|
base = MIN (now, base);
|
|
|
|
/* for negative rates we need to count back from from the segment
|
|
* duration. */
|
|
if (rate < 0.0)
|
|
time += duration;
|
|
*cur = time + gst_guint64_to_gdouble (now - base) * rate;
|
|
|
|
/* never report more than last seen position */
|
|
if (last != -1)
|
|
*cur = MIN (last, *cur);
|
|
|
|
gst_object_unref (clock);
|
|
|
|
res = TRUE;
|
|
|
|
GST_DEBUG_OBJECT (basesink,
|
|
"now %" GST_TIME_FORMAT " - base %" GST_TIME_FORMAT " - accum %"
|
|
GST_TIME_FORMAT " + time %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (now), GST_TIME_ARGS (base),
|
|
GST_TIME_ARGS (accum), GST_TIME_ARGS (time));
|
|
break;
|
|
}
|
|
default:
|
|
/* cannot answer other than TIME, we return FALSE, which will
|
|
* send the query upstream. */
|
|
break;
|
|
}
|
|
|
|
done:
|
|
GST_DEBUG_OBJECT (basesink, "res: %d, POSITION: %" GST_TIME_FORMAT,
|
|
res, GST_TIME_ARGS (*cur));
|
|
return res;
|
|
|
|
/* special cases */
|
|
in_eos:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "position in EOS");
|
|
res = gst_base_sink_get_position_last (basesink, cur);
|
|
GST_OBJECT_UNLOCK (basesink);
|
|
goto done;
|
|
}
|
|
in_pause:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "position in PAUSED");
|
|
res = gst_base_sink_get_position_paused (basesink, cur);
|
|
GST_OBJECT_UNLOCK (basesink);
|
|
goto done;
|
|
}
|
|
wrong_state:
|
|
{
|
|
/* in NULL or READY we always return 0 */
|
|
GST_DEBUG_OBJECT (basesink, "position in wrong state, return -1");
|
|
res = FALSE;
|
|
*cur = -1;
|
|
GST_OBJECT_UNLOCK (basesink);
|
|
goto done;
|
|
}
|
|
no_sync:
|
|
{
|
|
/* report last seen timestamp if any, else return FALSE so
|
|
* that upstream can answer */
|
|
if ((*cur = basesink->priv->current_sstart) != -1)
|
|
res = TRUE;
|
|
GST_DEBUG_OBJECT (basesink, "no sync, res %d, POSITION %" GST_TIME_FORMAT,
|
|
res, GST_TIME_ARGS (*cur));
|
|
GST_OBJECT_UNLOCK (basesink);
|
|
return res;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_sink_query (GstElement * element, GstQuery * query)
|
|
{
|
|
gboolean res = FALSE;
|
|
|
|
GstBaseSink *basesink = GST_BASE_SINK (element);
|
|
|
|
switch (GST_QUERY_TYPE (query)) {
|
|
case GST_QUERY_POSITION:
|
|
{
|
|
gint64 cur = 0;
|
|
GstFormat format;
|
|
|
|
gst_query_parse_position (query, &format, NULL);
|
|
|
|
GST_DEBUG_OBJECT (basesink, "position format %d", format);
|
|
|
|
/* first try to get the position based on the clock */
|
|
if ((res = gst_base_sink_get_position (basesink, format, &cur))) {
|
|
gst_query_set_position (query, format, cur);
|
|
} else {
|
|
/* fallback to peer query */
|
|
res = gst_base_sink_peer_query (basesink, query);
|
|
}
|
|
break;
|
|
}
|
|
case GST_QUERY_DURATION:
|
|
GST_DEBUG_OBJECT (basesink, "duration query");
|
|
res = gst_base_sink_peer_query (basesink, query);
|
|
break;
|
|
case GST_QUERY_LATENCY:
|
|
{
|
|
gboolean live, us_live;
|
|
GstClockTime min, max;
|
|
|
|
if ((res = gst_base_sink_query_latency (basesink, &live, &us_live, &min,
|
|
&max))) {
|
|
gst_query_set_latency (query, live, min, max);
|
|
}
|
|
break;
|
|
}
|
|
case GST_QUERY_JITTER:
|
|
break;
|
|
case GST_QUERY_RATE:
|
|
/* gst_query_set_rate (query, basesink->segment_rate); */
|
|
res = TRUE;
|
|
break;
|
|
case GST_QUERY_SEGMENT:
|
|
{
|
|
/* FIXME, bring start/stop to stream time */
|
|
gst_query_set_segment (query, basesink->segment.rate,
|
|
GST_FORMAT_TIME, basesink->segment.start, basesink->segment.stop);
|
|
break;
|
|
}
|
|
case GST_QUERY_SEEKING:
|
|
case GST_QUERY_CONVERT:
|
|
case GST_QUERY_FORMATS:
|
|
default:
|
|
res = gst_base_sink_peer_query (basesink, query);
|
|
break;
|
|
}
|
|
return res;
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_base_sink_change_state (GstElement * element, GstStateChange transition)
|
|
{
|
|
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
|
|
GstBaseSink *basesink = GST_BASE_SINK (element);
|
|
GstBaseSinkClass *bclass;
|
|
GstBaseSinkPrivate *priv;
|
|
|
|
priv = basesink->priv;
|
|
|
|
bclass = GST_BASE_SINK_GET_CLASS (basesink);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_NULL_TO_READY:
|
|
if (bclass->start)
|
|
if (!bclass->start (basesink))
|
|
goto start_failed;
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
/* need to complete preroll before this state change completes, there
|
|
* is no data flow in READY so we can safely assume we need to preroll. */
|
|
GST_PAD_PREROLL_LOCK (basesink->sinkpad);
|
|
GST_DEBUG_OBJECT (basesink, "READY to PAUSED");
|
|
gst_segment_init (&basesink->segment, GST_FORMAT_UNDEFINED);
|
|
gst_segment_init (basesink->abidata.ABI.clip_segment,
|
|
GST_FORMAT_UNDEFINED);
|
|
basesink->have_newsegment = FALSE;
|
|
basesink->offset = 0;
|
|
basesink->have_preroll = FALSE;
|
|
basesink->need_preroll = TRUE;
|
|
basesink->playing_async = TRUE;
|
|
priv->current_sstart = -1;
|
|
priv->current_sstop = -1;
|
|
priv->eos_rtime = -1;
|
|
priv->latency = 0;
|
|
basesink->eos = FALSE;
|
|
priv->received_eos = FALSE;
|
|
gst_base_sink_reset_qos (basesink);
|
|
priv->commited = FALSE;
|
|
if (priv->async_enabled) {
|
|
GST_DEBUG_OBJECT (basesink, "doing async state change");
|
|
/* when async enabled, post async-start message and return ASYNC from
|
|
* the state change function */
|
|
ret = GST_STATE_CHANGE_ASYNC;
|
|
gst_element_post_message (GST_ELEMENT_CAST (basesink),
|
|
gst_message_new_async_start (GST_OBJECT_CAST (basesink), FALSE));
|
|
} else {
|
|
priv->have_latency = TRUE;
|
|
}
|
|
GST_PAD_PREROLL_UNLOCK (basesink->sinkpad);
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
|
GST_PAD_PREROLL_LOCK (basesink->sinkpad);
|
|
if (!gst_base_sink_needs_preroll (basesink)) {
|
|
GST_DEBUG_OBJECT (basesink, "PAUSED to PLAYING, don't need preroll");
|
|
/* no preroll needed anymore now. */
|
|
basesink->playing_async = FALSE;
|
|
basesink->need_preroll = FALSE;
|
|
if (basesink->eos) {
|
|
/* need to post EOS message here */
|
|
GST_DEBUG_OBJECT (basesink, "Now posting EOS");
|
|
gst_element_post_message (GST_ELEMENT_CAST (basesink),
|
|
gst_message_new_eos (GST_OBJECT_CAST (basesink)));
|
|
} else {
|
|
GST_DEBUG_OBJECT (basesink, "signal preroll");
|
|
GST_PAD_PREROLL_SIGNAL (basesink->sinkpad);
|
|
}
|
|
} else {
|
|
GST_DEBUG_OBJECT (basesink, "PAUSED to PLAYING, we are not prerolled");
|
|
basesink->need_preroll = TRUE;
|
|
basesink->playing_async = TRUE;
|
|
priv->commited = FALSE;
|
|
if (priv->async_enabled) {
|
|
GST_DEBUG_OBJECT (basesink, "doing async state change");
|
|
ret = GST_STATE_CHANGE_ASYNC;
|
|
gst_element_post_message (GST_ELEMENT_CAST (basesink),
|
|
gst_message_new_async_start (GST_OBJECT_CAST (basesink), FALSE));
|
|
}
|
|
}
|
|
GST_PAD_PREROLL_UNLOCK (basesink->sinkpad);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
{
|
|
GstStateChangeReturn bret;
|
|
|
|
bret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
if (G_UNLIKELY (bret == GST_STATE_CHANGE_FAILURE))
|
|
goto activate_failed;
|
|
}
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
/* note that this is the upward case, which doesn't follow most
|
|
patterns */
|
|
if (basesink->pad_mode == GST_ACTIVATE_PULL) {
|
|
GST_DEBUG_OBJECT (basesink, "basesink activated in pull mode, "
|
|
"returning SUCCESS directly");
|
|
GST_PAD_PREROLL_LOCK (basesink->sinkpad);
|
|
gst_element_post_message (GST_ELEMENT_CAST (basesink),
|
|
gst_message_new_async_done (GST_OBJECT_CAST (basesink)));
|
|
GST_PAD_PREROLL_UNLOCK (basesink->sinkpad);
|
|
ret = GST_STATE_CHANGE_SUCCESS;
|
|
}
|
|
break;
|
|
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
|
GST_DEBUG_OBJECT (basesink, "PLAYING to PAUSED");
|
|
/* FIXME, make sure we cannot enter _render first */
|
|
|
|
/* we need to call ::unlock before locking PREROLL_LOCK
|
|
* since we lock it before going into ::render */
|
|
if (bclass->unlock)
|
|
bclass->unlock (basesink);
|
|
|
|
GST_PAD_PREROLL_LOCK (basesink->sinkpad);
|
|
/* now that we have the PREROLL lock, clear our unlock request */
|
|
if (bclass->unlock_stop)
|
|
bclass->unlock_stop (basesink);
|
|
|
|
/* we need preroll again and we set the flag before unlocking the clockid
|
|
* because if the clockid is unlocked before a current buffer expired, we
|
|
* can use that buffer to preroll with */
|
|
basesink->need_preroll = TRUE;
|
|
|
|
if (basesink->clock_id) {
|
|
gst_clock_id_unschedule (basesink->clock_id);
|
|
}
|
|
|
|
/* if we don't have a preroll buffer we need to wait for a preroll and
|
|
* return ASYNC. */
|
|
if (!gst_base_sink_needs_preroll (basesink)) {
|
|
GST_DEBUG_OBJECT (basesink, "PLAYING to PAUSED, we are prerolled");
|
|
basesink->playing_async = FALSE;
|
|
} else {
|
|
if (GST_STATE_TARGET (GST_ELEMENT (basesink)) <= GST_STATE_READY) {
|
|
ret = GST_STATE_CHANGE_SUCCESS;
|
|
} else {
|
|
GST_DEBUG_OBJECT (basesink,
|
|
"PLAYING to PAUSED, we are not prerolled");
|
|
basesink->playing_async = TRUE;
|
|
priv->commited = FALSE;
|
|
if (priv->async_enabled) {
|
|
GST_DEBUG_OBJECT (basesink, "doing async state change");
|
|
ret = GST_STATE_CHANGE_ASYNC;
|
|
gst_element_post_message (GST_ELEMENT_CAST (basesink),
|
|
gst_message_new_async_start (GST_OBJECT_CAST (basesink),
|
|
FALSE));
|
|
}
|
|
}
|
|
}
|
|
GST_DEBUG_OBJECT (basesink, "rendered: %" G_GUINT64_FORMAT
|
|
", dropped: %" G_GUINT64_FORMAT, priv->rendered, priv->dropped);
|
|
|
|
gst_base_sink_reset_qos (basesink);
|
|
GST_PAD_PREROLL_UNLOCK (basesink->sinkpad);
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
GST_PAD_PREROLL_LOCK (basesink->sinkpad);
|
|
if (!priv->commited) {
|
|
if (priv->async_enabled) {
|
|
GST_DEBUG_OBJECT (basesink, "PAUSED to READY, posting async-done");
|
|
|
|
gst_element_post_message (GST_ELEMENT_CAST (basesink),
|
|
gst_message_new_state_changed (GST_OBJECT_CAST (basesink),
|
|
GST_STATE_PLAYING, GST_STATE_PAUSED, GST_STATE_READY));
|
|
|
|
gst_element_post_message (GST_ELEMENT_CAST (basesink),
|
|
gst_message_new_async_done (GST_OBJECT_CAST (basesink)));
|
|
}
|
|
priv->commited = TRUE;
|
|
} else {
|
|
GST_DEBUG_OBJECT (basesink, "PAUSED to READY, don't need_preroll");
|
|
}
|
|
priv->current_sstart = -1;
|
|
priv->current_sstop = -1;
|
|
priv->have_latency = FALSE;
|
|
gst_base_sink_set_last_buffer (basesink, NULL);
|
|
GST_PAD_PREROLL_UNLOCK (basesink->sinkpad);
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
if (bclass->stop) {
|
|
if (!bclass->stop (basesink)) {
|
|
GST_WARNING_OBJECT (basesink, "failed to stop");
|
|
}
|
|
}
|
|
gst_base_sink_set_last_buffer (basesink, NULL);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
start_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink, "failed to start");
|
|
return GST_STATE_CHANGE_FAILURE;
|
|
}
|
|
activate_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (basesink,
|
|
"element failed to change states -- activation problem?");
|
|
return GST_STATE_CHANGE_FAILURE;
|
|
}
|
|
}
|