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527da05476
Return FALSE in basesrc's default query handler when we get a SEEKING query for a format that's not the one the source operates in. Previously (ie. before, in the git version) we would return TRUE in that case and seekable=FALSE, which is more correct, but causes backwards compatibility problems. (Before that we would change the format of the query when answering, which was completely broken since callers don't expect that or check for it). Since the SEEKING query is a fairly recent addition, not all demuxers, parsers and decoders implement it yet, in which case any SEEKING query by an application will just be passed upstream where it will then be handled by basesrc. Now, if e.g. totem does a SEEKING query for TIME format and we have a demuxer that doesn't implement the query, basesrc would answer it with seekable=FALSE in most cases, and totem can only take that as authoritative answer, not knowing that the demuxer doesn't implement the SEEKING query. To avoid this, we make basesrc return FALSE to SEEKING queries in unhandled formats. That way applications like totem can fall back on assuming seekability depending on whether a duration is available, or somesuch. Downstream elements doing such queries are likely to equate an unhandled query with a non-seekable response as well, so this should be an acceptable fix for the time being. See #584838, #588944, #589423 and #589424.
2933 lines
88 KiB
C
2933 lines
88 KiB
C
/* GStreamer
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* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
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* 2000,2005 Wim Taymans <wim@fluendo.com>
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*
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* gstbasesrc.c:
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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:gstbasesrc
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* @short_description: Base class for getrange based source elements
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* @see_also: #GstPushSrc, #GstBaseTransform, #GstBaseSink
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*
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* This is a generice base class for source elements. The following
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* types of sources are supported:
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* <itemizedlist>
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* <listitem><para>random access sources like files</para></listitem>
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* <listitem><para>seekable sources</para></listitem>
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* <listitem><para>live sources</para></listitem>
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* </itemizedlist>
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*
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* The source can be configured to operate in any #GstFormat with the
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* gst_base_src_set_format() method. The currently set format determines
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* the format of the internal #GstSegment and any #GST_EVENT_NEWSEGMENT
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* events. The default format for #GstBaseSrc is #GST_FORMAT_BYTES.
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*
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* #GstBaseSrc always supports push mode scheduling. If the following
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* conditions are met, it also supports pull mode scheduling:
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* <itemizedlist>
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* <listitem><para>The format is set to #GST_FORMAT_BYTES (default).</para>
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* </listitem>
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* <listitem><para>#GstBaseSrc::is_seekable returns %TRUE.</para>
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* </listitem>
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* </itemizedlist>
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*
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* Since 0.10.9, any #GstBaseSrc can enable pull based scheduling at any
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* time by overriding #GstBaseSrc::check_get_range so that it returns %TRUE.
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*
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* If all the conditions are met for operating in pull mode, #GstBaseSrc is
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* automatically seekable in push mode as well. The following conditions must
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* be met to make the element seekable in push mode when the format is not
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* #GST_FORMAT_BYTES:
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* <itemizedlist>
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* <listitem><para>
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* #GstBaseSrc::is_seekable returns %TRUE.
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* </para></listitem>
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* <listitem><para>
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* #GstBaseSrc::query can convert all supported seek formats to the
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* internal format as set with gst_base_src_set_format().
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* </para></listitem>
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* <listitem><para>
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* #GstBaseSrc::do_seek is implemented, performs the seek and returns %TRUE.
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* </para></listitem>
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* </itemizedlist>
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*
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* When the element does not meet the requirements to operate in pull mode,
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* the offset and length in the #GstBaseSrc::create method should be ignored.
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* It is recommended to subclass #GstPushSrc instead, in this situation. If the
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* element can operate in pull mode but only with specific offsets and
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* lengths, it is allowed to generate an error when the wrong values are passed
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* to the #GstBaseSrc::create function.
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*
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* #GstBaseSrc has support for live sources. Live sources are sources that when
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* paused discard data, such as audio or video capture devices. A typical live
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* source also produces data at a fixed rate and thus provides a clock to publish
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* this rate.
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* Use gst_base_src_set_live() to activate the live source mode.
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*
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* A live source does not produce data in the PAUSED state. This means that the
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* #GstBaseSrc::create method will not be called in PAUSED but only in PLAYING.
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* To signal the pipeline that the element will not produce data, the return
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* value from the READY to PAUSED state will be #GST_STATE_CHANGE_NO_PREROLL.
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*
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* A typical live source will timestamp the buffers it creates with the
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* current running time of the pipeline. This is one reason why a live source
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* can only produce data in the PLAYING state, when the clock is actually
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* distributed and running.
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*
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* Live sources that synchronize and block on the clock (an audio source, for
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* example) can since 0.10.12 use gst_base_src_wait_playing() when the ::create
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* function was interrupted by a state change to PAUSED.
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*
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* The #GstBaseSrc::get_times method can be used to implement pseudo-live
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* sources.
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* It only makes sense to implement the ::get_times function if the source is
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* a live source. The ::get_times function should return timestamps starting
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* from 0, as if it were a non-live source. The base class will make sure that
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* the timestamps are transformed into the current running_time.
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* The base source will then wait for the calculated running_time before pushing
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* out the buffer.
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*
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* For live sources, the base class will by default report a latency of 0.
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* For pseudo live sources, the base class will by default measure the difference
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* between the first buffer timestamp and the start time of get_times and will
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* report this value as the latency.
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* Subclasses should override the query function when this behaviour is not
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* acceptable.
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*
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* There is only support in #GstBaseSrc for exactly one source pad, which
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* should be named "src". A source implementation (subclass of #GstBaseSrc)
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* should install a pad template in its class_init function, like so:
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* <programlisting>
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* static void
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* my_element_class_init (GstMyElementClass *klass)
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* {
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* GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
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* // srctemplate should be a #GstStaticPadTemplate with direction
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* // #GST_PAD_SRC and name "src"
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* gst_element_class_add_pad_template (gstelement_class,
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* gst_static_pad_template_get (&srctemplate));
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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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* <refsect2>
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* <title>Controlled shutdown of live sources in applications</title>
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* <para>
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* Applications that record from a live source may want to stop recording
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* in a controlled way, so that the recording is stopped, but the data
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* already in the pipeline is processed to the end (remember that many live
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* sources would go on recording forever otherwise). For that to happen the
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* application needs to make the source stop recording and send an EOS
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* event down the pipeline. The application would then wait for an
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* EOS message posted on the pipeline's bus to know when all data has
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* been processed and the pipeline can safely be stopped.
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*
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* Since GStreamer 0.10.16 an application may send an EOS event to a source
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* element to make it perform the EOS logic (send EOS event downstream or post a
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* #GST_MESSAGE_SEGMENT_DONE on the bus). This can typically be done
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* with the gst_element_send_event() function on the element or its parent bin.
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*
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* After the EOS has been sent to the element, the application should wait for
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* an EOS message to be posted on the pipeline's bus. Once this EOS message is
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* received, it may safely shut down the entire pipeline.
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*
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* The old behaviour for controlled shutdown introduced since GStreamer 0.10.3
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* is still available but deprecated as it is dangerous and less flexible.
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*
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* Last reviewed on 2007-12-19 (0.10.16)
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* </para>
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* </refsect2>
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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 <stdlib.h>
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#include <string.h>
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#include "gstbasesrc.h"
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#include "gsttypefindhelper.h"
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#include <gst/gstmarshal.h>
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#include <gst/gst-i18n-lib.h>
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GST_DEBUG_CATEGORY_STATIC (gst_base_src_debug);
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#define GST_CAT_DEFAULT gst_base_src_debug
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#define GST_LIVE_GET_LOCK(elem) (GST_BASE_SRC_CAST(elem)->live_lock)
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#define GST_LIVE_LOCK(elem) g_mutex_lock(GST_LIVE_GET_LOCK(elem))
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#define GST_LIVE_TRYLOCK(elem) g_mutex_trylock(GST_LIVE_GET_LOCK(elem))
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#define GST_LIVE_UNLOCK(elem) g_mutex_unlock(GST_LIVE_GET_LOCK(elem))
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#define GST_LIVE_GET_COND(elem) (GST_BASE_SRC_CAST(elem)->live_cond)
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#define GST_LIVE_WAIT(elem) g_cond_wait (GST_LIVE_GET_COND (elem), GST_LIVE_GET_LOCK (elem))
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#define GST_LIVE_TIMED_WAIT(elem, timeval) g_cond_timed_wait (GST_LIVE_GET_COND (elem), GST_LIVE_GET_LOCK (elem),\
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timeval)
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#define GST_LIVE_SIGNAL(elem) g_cond_signal (GST_LIVE_GET_COND (elem));
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#define GST_LIVE_BROADCAST(elem) g_cond_broadcast (GST_LIVE_GET_COND (elem));
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/* BaseSrc signals and args */
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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#define DEFAULT_BLOCKSIZE 4096
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#define DEFAULT_NUM_BUFFERS -1
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#define DEFAULT_TYPEFIND FALSE
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#define DEFAULT_DO_TIMESTAMP FALSE
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enum
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{
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PROP_0,
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PROP_BLOCKSIZE,
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PROP_NUM_BUFFERS,
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PROP_TYPEFIND,
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PROP_DO_TIMESTAMP
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};
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#define GST_BASE_SRC_GET_PRIVATE(obj) \
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(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_BASE_SRC, GstBaseSrcPrivate))
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struct _GstBaseSrcPrivate
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{
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gboolean last_sent_eos; /* last thing we did was send an EOS (we set this
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* to avoid the sending of two EOS in some cases) */
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gboolean discont;
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gboolean flushing;
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/* two segments to be sent in the streaming thread with STREAM_LOCK */
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GstEvent *close_segment;
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GstEvent *start_segment;
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/* if EOS is pending (atomic) */
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gint pending_eos;
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/* startup latency is the time it takes between going to PLAYING and producing
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* the first BUFFER with running_time 0. This value is included in the latency
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* reporting. */
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GstClockTime latency;
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/* timestamp offset, this is the offset add to the values of gst_times for
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* pseudo live sources */
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GstClockTimeDiff ts_offset;
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gboolean do_timestamp;
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/* stream sequence number */
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guint32 seqnum;
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/* pending tags to be pushed in the data stream */
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GList *pending_tags;
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};
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static GstElementClass *parent_class = NULL;
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static void gst_base_src_base_init (gpointer g_class);
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static void gst_base_src_class_init (GstBaseSrcClass * klass);
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static void gst_base_src_init (GstBaseSrc * src, gpointer g_class);
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static void gst_base_src_finalize (GObject * object);
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GType
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gst_base_src_get_type (void)
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{
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static volatile gsize base_src_type = 0;
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if (g_once_init_enter (&base_src_type)) {
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GType _type;
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static const GTypeInfo base_src_info = {
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sizeof (GstBaseSrcClass),
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(GBaseInitFunc) gst_base_src_base_init,
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NULL,
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(GClassInitFunc) gst_base_src_class_init,
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NULL,
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NULL,
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sizeof (GstBaseSrc),
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0,
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(GInstanceInitFunc) gst_base_src_init,
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};
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_type = g_type_register_static (GST_TYPE_ELEMENT,
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"GstBaseSrc", &base_src_info, G_TYPE_FLAG_ABSTRACT);
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g_once_init_leave (&base_src_type, _type);
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}
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return base_src_type;
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}
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static GstCaps *gst_base_src_getcaps (GstPad * pad);
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static gboolean gst_base_src_setcaps (GstPad * pad, GstCaps * caps);
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static void gst_base_src_fixate (GstPad * pad, GstCaps * caps);
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static gboolean gst_base_src_activate_push (GstPad * pad, gboolean active);
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static gboolean gst_base_src_activate_pull (GstPad * pad, gboolean active);
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static void gst_base_src_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_base_src_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static gboolean gst_base_src_event_handler (GstPad * pad, GstEvent * event);
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static gboolean gst_base_src_send_event (GstElement * elem, GstEvent * event);
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static gboolean gst_base_src_default_event (GstBaseSrc * src, GstEvent * event);
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static const GstQueryType *gst_base_src_get_query_types (GstElement * element);
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static gboolean gst_base_src_query (GstPad * pad, GstQuery * query);
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static gboolean gst_base_src_default_negotiate (GstBaseSrc * basesrc);
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static gboolean gst_base_src_default_do_seek (GstBaseSrc * src,
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GstSegment * segment);
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static gboolean gst_base_src_default_query (GstBaseSrc * src, GstQuery * query);
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static gboolean gst_base_src_default_prepare_seek_segment (GstBaseSrc * src,
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GstEvent * event, GstSegment * segment);
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static gboolean gst_base_src_set_flushing (GstBaseSrc * basesrc,
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gboolean flushing, gboolean live_play, gboolean unlock, gboolean * playing);
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static gboolean gst_base_src_start (GstBaseSrc * basesrc);
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static gboolean gst_base_src_stop (GstBaseSrc * basesrc);
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static GstStateChangeReturn gst_base_src_change_state (GstElement * element,
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GstStateChange transition);
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static void gst_base_src_loop (GstPad * pad);
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static gboolean gst_base_src_pad_check_get_range (GstPad * pad);
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static gboolean gst_base_src_default_check_get_range (GstBaseSrc * bsrc);
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static GstFlowReturn gst_base_src_pad_get_range (GstPad * pad, guint64 offset,
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guint length, GstBuffer ** buf);
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static GstFlowReturn gst_base_src_get_range (GstBaseSrc * src, guint64 offset,
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guint length, GstBuffer ** buf);
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static gboolean gst_base_src_seekable (GstBaseSrc * src);
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static void
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gst_base_src_base_init (gpointer g_class)
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{
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GST_DEBUG_CATEGORY_INIT (gst_base_src_debug, "basesrc", 0, "basesrc element");
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}
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static void
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gst_base_src_class_init (GstBaseSrcClass * 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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g_type_class_add_private (klass, sizeof (GstBaseSrcPrivate));
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parent_class = g_type_class_peek_parent (klass);
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gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_base_src_finalize);
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gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_base_src_set_property);
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gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_base_src_get_property);
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g_object_class_install_property (gobject_class, PROP_BLOCKSIZE,
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g_param_spec_ulong ("blocksize", "Block size",
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"Size in bytes to read per buffer (-1 = default)", 0, G_MAXULONG,
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DEFAULT_BLOCKSIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_NUM_BUFFERS,
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g_param_spec_int ("num-buffers", "num-buffers",
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"Number of buffers to output before sending EOS (-1 = unlimited)",
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-1, G_MAXINT, DEFAULT_NUM_BUFFERS, G_PARAM_READWRITE |
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G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_TYPEFIND,
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g_param_spec_boolean ("typefind", "Typefind",
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"Run typefind before negotiating", DEFAULT_TYPEFIND,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_DO_TIMESTAMP,
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g_param_spec_boolean ("do-timestamp", "Do timestamp",
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"Apply current stream time to buffers", DEFAULT_DO_TIMESTAMP,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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gstelement_class->change_state =
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GST_DEBUG_FUNCPTR (gst_base_src_change_state);
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gstelement_class->send_event = GST_DEBUG_FUNCPTR (gst_base_src_send_event);
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gstelement_class->get_query_types =
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GST_DEBUG_FUNCPTR (gst_base_src_get_query_types);
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klass->negotiate = GST_DEBUG_FUNCPTR (gst_base_src_default_negotiate);
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klass->event = GST_DEBUG_FUNCPTR (gst_base_src_default_event);
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klass->do_seek = GST_DEBUG_FUNCPTR (gst_base_src_default_do_seek);
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klass->query = GST_DEBUG_FUNCPTR (gst_base_src_default_query);
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klass->check_get_range =
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GST_DEBUG_FUNCPTR (gst_base_src_default_check_get_range);
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klass->prepare_seek_segment =
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GST_DEBUG_FUNCPTR (gst_base_src_default_prepare_seek_segment);
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}
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static void
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gst_base_src_init (GstBaseSrc * basesrc, gpointer g_class)
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{
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GstPad *pad;
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GstPadTemplate *pad_template;
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basesrc->priv = GST_BASE_SRC_GET_PRIVATE (basesrc);
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basesrc->is_live = FALSE;
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basesrc->live_lock = g_mutex_new ();
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basesrc->live_cond = g_cond_new ();
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basesrc->num_buffers = DEFAULT_NUM_BUFFERS;
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basesrc->num_buffers_left = -1;
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basesrc->can_activate_push = TRUE;
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basesrc->pad_mode = GST_ACTIVATE_NONE;
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pad_template =
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gst_element_class_get_pad_template (GST_ELEMENT_CLASS (g_class), "src");
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g_return_if_fail (pad_template != NULL);
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GST_DEBUG_OBJECT (basesrc, "creating src pad");
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pad = gst_pad_new_from_template (pad_template, "src");
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GST_DEBUG_OBJECT (basesrc, "setting functions on src pad");
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gst_pad_set_activatepush_function (pad,
|
|
GST_DEBUG_FUNCPTR (gst_base_src_activate_push));
|
|
gst_pad_set_activatepull_function (pad,
|
|
GST_DEBUG_FUNCPTR (gst_base_src_activate_pull));
|
|
gst_pad_set_event_function (pad,
|
|
GST_DEBUG_FUNCPTR (gst_base_src_event_handler));
|
|
gst_pad_set_query_function (pad, GST_DEBUG_FUNCPTR (gst_base_src_query));
|
|
gst_pad_set_checkgetrange_function (pad,
|
|
GST_DEBUG_FUNCPTR (gst_base_src_pad_check_get_range));
|
|
gst_pad_set_getrange_function (pad,
|
|
GST_DEBUG_FUNCPTR (gst_base_src_pad_get_range));
|
|
gst_pad_set_getcaps_function (pad, GST_DEBUG_FUNCPTR (gst_base_src_getcaps));
|
|
gst_pad_set_setcaps_function (pad, GST_DEBUG_FUNCPTR (gst_base_src_setcaps));
|
|
gst_pad_set_fixatecaps_function (pad,
|
|
GST_DEBUG_FUNCPTR (gst_base_src_fixate));
|
|
|
|
/* hold pointer to pad */
|
|
basesrc->srcpad = pad;
|
|
GST_DEBUG_OBJECT (basesrc, "adding src pad");
|
|
gst_element_add_pad (GST_ELEMENT (basesrc), pad);
|
|
|
|
basesrc->blocksize = DEFAULT_BLOCKSIZE;
|
|
basesrc->clock_id = NULL;
|
|
/* we operate in BYTES by default */
|
|
gst_base_src_set_format (basesrc, GST_FORMAT_BYTES);
|
|
basesrc->data.ABI.typefind = DEFAULT_TYPEFIND;
|
|
basesrc->priv->do_timestamp = DEFAULT_DO_TIMESTAMP;
|
|
|
|
GST_OBJECT_FLAG_UNSET (basesrc, GST_BASE_SRC_STARTED);
|
|
|
|
GST_DEBUG_OBJECT (basesrc, "init done");
|
|
}
|
|
|
|
static void
|
|
gst_base_src_finalize (GObject * object)
|
|
{
|
|
GstBaseSrc *basesrc;
|
|
GstEvent **event_p;
|
|
|
|
basesrc = GST_BASE_SRC (object);
|
|
|
|
g_mutex_free (basesrc->live_lock);
|
|
g_cond_free (basesrc->live_cond);
|
|
|
|
event_p = &basesrc->data.ABI.pending_seek;
|
|
gst_event_replace (event_p, NULL);
|
|
|
|
if (basesrc->priv->pending_tags) {
|
|
g_list_foreach (basesrc->priv->pending_tags, (GFunc) gst_event_unref, NULL);
|
|
g_list_free (basesrc->priv->pending_tags);
|
|
}
|
|
|
|
G_OBJECT_CLASS (parent_class)->finalize (object);
|
|
}
|
|
|
|
/**
|
|
* gst_base_src_wait_playing:
|
|
* @src: the src
|
|
*
|
|
* If the #GstBaseSrcClass::create method performs its own synchronisation against
|
|
* the clock it must unblock when going from PLAYING to the PAUSED state and call
|
|
* this method before continuing to produce the remaining data.
|
|
*
|
|
* This function will block until a state change to PLAYING happens (in which
|
|
* case this function returns #GST_FLOW_OK) or the processing must be stopped due
|
|
* to a state change to READY or a FLUSH event (in which case this function
|
|
* returns #GST_FLOW_WRONG_STATE).
|
|
*
|
|
* Since: 0.10.12
|
|
*
|
|
* Returns: #GST_FLOW_OK if @src is PLAYING and processing can
|
|
* continue. Any other return value should be returned from the create vmethod.
|
|
*/
|
|
GstFlowReturn
|
|
gst_base_src_wait_playing (GstBaseSrc * src)
|
|
{
|
|
g_return_val_if_fail (GST_IS_BASE_SRC (src), GST_FLOW_ERROR);
|
|
|
|
/* block until the state changes, or we get a flush, or something */
|
|
GST_DEBUG_OBJECT (src, "live source waiting for running state");
|
|
GST_LIVE_WAIT (src);
|
|
if (src->priv->flushing)
|
|
goto flushing;
|
|
GST_DEBUG_OBJECT (src, "live source unlocked");
|
|
|
|
return GST_FLOW_OK;
|
|
|
|
/* ERRORS */
|
|
flushing:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "we are flushing");
|
|
return GST_FLOW_WRONG_STATE;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_base_src_set_live:
|
|
* @src: base source instance
|
|
* @live: new live-mode
|
|
*
|
|
* If the element listens to a live source, @live should
|
|
* be set to %TRUE.
|
|
*
|
|
* A live source will not produce data in the PAUSED state and
|
|
* will therefore not be able to participate in the PREROLL phase
|
|
* of a pipeline. To signal this fact to the application and the
|
|
* pipeline, the state change return value of the live source will
|
|
* be GST_STATE_CHANGE_NO_PREROLL.
|
|
*/
|
|
void
|
|
gst_base_src_set_live (GstBaseSrc * src, gboolean live)
|
|
{
|
|
g_return_if_fail (GST_IS_BASE_SRC (src));
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
src->is_live = live;
|
|
GST_OBJECT_UNLOCK (src);
|
|
}
|
|
|
|
/**
|
|
* gst_base_src_is_live:
|
|
* @src: base source instance
|
|
*
|
|
* Check if an element is in live mode.
|
|
*
|
|
* Returns: %TRUE if element is in live mode.
|
|
*/
|
|
gboolean
|
|
gst_base_src_is_live (GstBaseSrc * src)
|
|
{
|
|
gboolean result;
|
|
|
|
g_return_val_if_fail (GST_IS_BASE_SRC (src), FALSE);
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
result = src->is_live;
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
return result;
|
|
}
|
|
|
|
/**
|
|
* gst_base_src_set_format:
|
|
* @src: base source instance
|
|
* @format: the format to use
|
|
*
|
|
* Sets the default format of the source. This will be the format used
|
|
* for sending NEW_SEGMENT events and for performing seeks.
|
|
*
|
|
* If a format of GST_FORMAT_BYTES is set, the element will be able to
|
|
* operate in pull mode if the #GstBaseSrc::is_seekable returns TRUE.
|
|
*
|
|
* Since: 0.10.1
|
|
*/
|
|
void
|
|
gst_base_src_set_format (GstBaseSrc * src, GstFormat format)
|
|
{
|
|
g_return_if_fail (GST_IS_BASE_SRC (src));
|
|
|
|
gst_segment_init (&src->segment, format);
|
|
}
|
|
|
|
/**
|
|
* gst_base_src_query_latency:
|
|
* @src: the source
|
|
* @live: if the source is live
|
|
* @min_latency: the min latency of the source
|
|
* @max_latency: the max latency of the source
|
|
*
|
|
* Query the source for the latency parameters. @live will be TRUE when @src is
|
|
* configured as a live source. @min_latency will be set to the difference
|
|
* between the running time and the timestamp of the first buffer.
|
|
* @max_latency is always the undefined value of -1.
|
|
*
|
|
* This function is mostly used by subclasses.
|
|
*
|
|
* Returns: TRUE if the query succeeded.
|
|
*
|
|
* Since: 0.10.13
|
|
*/
|
|
gboolean
|
|
gst_base_src_query_latency (GstBaseSrc * src, gboolean * live,
|
|
GstClockTime * min_latency, GstClockTime * max_latency)
|
|
{
|
|
GstClockTime min;
|
|
|
|
g_return_val_if_fail (GST_IS_BASE_SRC (src), FALSE);
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
if (live)
|
|
*live = src->is_live;
|
|
|
|
/* if we have a startup latency, report this one, else report 0. Subclasses
|
|
* are supposed to override the query function if they want something
|
|
* else. */
|
|
if (src->priv->latency != -1)
|
|
min = src->priv->latency;
|
|
else
|
|
min = 0;
|
|
|
|
if (min_latency)
|
|
*min_latency = min;
|
|
if (max_latency)
|
|
*max_latency = -1;
|
|
|
|
GST_LOG_OBJECT (src, "latency: live %d, min %" GST_TIME_FORMAT
|
|
", max %" GST_TIME_FORMAT, src->is_live, GST_TIME_ARGS (min),
|
|
GST_TIME_ARGS (-1));
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/**
|
|
* gst_base_src_set_blocksize:
|
|
* @src: the source
|
|
* @blocksize: the new blocksize in bytes
|
|
*
|
|
* Set the number of bytes that @src will push out with each buffer. When
|
|
* @blocksize is set to -1, a default length will be used.
|
|
*
|
|
* Since: 0.10.22
|
|
*/
|
|
void
|
|
gst_base_src_set_blocksize (GstBaseSrc * src, gulong blocksize)
|
|
{
|
|
g_return_if_fail (GST_IS_BASE_SRC (src));
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
src->blocksize = blocksize;
|
|
GST_OBJECT_UNLOCK (src);
|
|
}
|
|
|
|
/**
|
|
* gst_base_src_get_blocksize:
|
|
* @src: the source
|
|
*
|
|
* Get the number of bytes that @src will push out with each buffer.
|
|
*
|
|
* Returns: the number of bytes pushed with each buffer.
|
|
*
|
|
* Since: 0.10.22
|
|
*/
|
|
gulong
|
|
gst_base_src_get_blocksize (GstBaseSrc * src)
|
|
{
|
|
gulong res;
|
|
|
|
g_return_val_if_fail (GST_IS_BASE_SRC (src), 0);
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
res = src->blocksize;
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
return res;
|
|
}
|
|
|
|
|
|
/**
|
|
* gst_base_src_set_do_timestamp:
|
|
* @src: the source
|
|
* @timestamp: enable or disable timestamping
|
|
*
|
|
* Configure @src to automatically timestamp outgoing buffers based on the
|
|
* current running_time of the pipeline. This property is mostly useful for live
|
|
* sources.
|
|
*
|
|
* Since: 0.10.15
|
|
*/
|
|
void
|
|
gst_base_src_set_do_timestamp (GstBaseSrc * src, gboolean timestamp)
|
|
{
|
|
g_return_if_fail (GST_IS_BASE_SRC (src));
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
src->priv->do_timestamp = timestamp;
|
|
GST_OBJECT_UNLOCK (src);
|
|
}
|
|
|
|
/**
|
|
* gst_base_src_get_do_timestamp:
|
|
* @src: the source
|
|
*
|
|
* Query if @src timestamps outgoing buffers based on the current running_time.
|
|
*
|
|
* Returns: %TRUE if the base class will automatically timestamp outgoing buffers.
|
|
*
|
|
* Since: 0.10.15
|
|
*/
|
|
gboolean
|
|
gst_base_src_get_do_timestamp (GstBaseSrc * src)
|
|
{
|
|
gboolean res;
|
|
|
|
g_return_val_if_fail (GST_IS_BASE_SRC (src), FALSE);
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
res = src->priv->do_timestamp;
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
return res;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_setcaps (GstPad * pad, GstCaps * caps)
|
|
{
|
|
GstBaseSrcClass *bclass;
|
|
GstBaseSrc *bsrc;
|
|
gboolean res = TRUE;
|
|
|
|
bsrc = GST_BASE_SRC (GST_PAD_PARENT (pad));
|
|
bclass = GST_BASE_SRC_GET_CLASS (bsrc);
|
|
|
|
if (bclass->set_caps)
|
|
res = bclass->set_caps (bsrc, caps);
|
|
|
|
return res;
|
|
}
|
|
|
|
static GstCaps *
|
|
gst_base_src_getcaps (GstPad * pad)
|
|
{
|
|
GstBaseSrcClass *bclass;
|
|
GstBaseSrc *bsrc;
|
|
GstCaps *caps = NULL;
|
|
|
|
bsrc = GST_BASE_SRC (GST_PAD_PARENT (pad));
|
|
bclass = GST_BASE_SRC_GET_CLASS (bsrc);
|
|
if (bclass->get_caps)
|
|
caps = bclass->get_caps (bsrc);
|
|
|
|
if (caps == NULL) {
|
|
GstPadTemplate *pad_template;
|
|
|
|
pad_template =
|
|
gst_element_class_get_pad_template (GST_ELEMENT_CLASS (bclass), "src");
|
|
if (pad_template != NULL) {
|
|
caps = gst_caps_ref (gst_pad_template_get_caps (pad_template));
|
|
}
|
|
}
|
|
return caps;
|
|
}
|
|
|
|
static void
|
|
gst_base_src_fixate (GstPad * pad, GstCaps * caps)
|
|
{
|
|
GstBaseSrcClass *bclass;
|
|
GstBaseSrc *bsrc;
|
|
|
|
bsrc = GST_BASE_SRC (gst_pad_get_parent (pad));
|
|
bclass = GST_BASE_SRC_GET_CLASS (bsrc);
|
|
|
|
if (bclass->fixate)
|
|
bclass->fixate (bsrc, caps);
|
|
|
|
gst_object_unref (bsrc);
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_default_query (GstBaseSrc * src, GstQuery * query)
|
|
{
|
|
gboolean res;
|
|
|
|
switch (GST_QUERY_TYPE (query)) {
|
|
case GST_QUERY_POSITION:
|
|
{
|
|
GstFormat format;
|
|
|
|
gst_query_parse_position (query, &format, NULL);
|
|
switch (format) {
|
|
case GST_FORMAT_PERCENT:
|
|
{
|
|
gint64 percent;
|
|
gint64 position;
|
|
gint64 duration;
|
|
|
|
position = src->segment.last_stop;
|
|
duration = src->segment.duration;
|
|
|
|
if (position != -1 && duration != -1) {
|
|
if (position < duration)
|
|
percent = gst_util_uint64_scale (GST_FORMAT_PERCENT_MAX, position,
|
|
duration);
|
|
else
|
|
percent = GST_FORMAT_PERCENT_MAX;
|
|
} else
|
|
percent = -1;
|
|
|
|
gst_query_set_position (query, GST_FORMAT_PERCENT, percent);
|
|
res = TRUE;
|
|
break;
|
|
}
|
|
default:
|
|
{
|
|
gint64 position;
|
|
|
|
position = src->segment.last_stop;
|
|
|
|
if (position != -1) {
|
|
/* convert to requested format */
|
|
res =
|
|
gst_pad_query_convert (src->srcpad, src->segment.format,
|
|
position, &format, &position);
|
|
} else
|
|
res = TRUE;
|
|
|
|
gst_query_set_position (query, format, position);
|
|
break;
|
|
}
|
|
}
|
|
break;
|
|
}
|
|
case GST_QUERY_DURATION:
|
|
{
|
|
GstFormat format;
|
|
|
|
gst_query_parse_duration (query, &format, NULL);
|
|
|
|
GST_DEBUG_OBJECT (src, "duration query in format %s",
|
|
gst_format_get_name (format));
|
|
|
|
switch (format) {
|
|
case GST_FORMAT_PERCENT:
|
|
gst_query_set_duration (query, GST_FORMAT_PERCENT,
|
|
GST_FORMAT_PERCENT_MAX);
|
|
res = TRUE;
|
|
break;
|
|
default:
|
|
{
|
|
gint64 duration;
|
|
|
|
/* this is the duration as configured by the subclass. */
|
|
duration = src->segment.duration;
|
|
|
|
if (duration != -1) {
|
|
/* convert to requested format, if this fails, we have a duration
|
|
* but we cannot answer the query, we must return FALSE. */
|
|
res =
|
|
gst_pad_query_convert (src->srcpad, src->segment.format,
|
|
duration, &format, &duration);
|
|
} else {
|
|
/* The subclass did not configure a duration, we assume that the
|
|
* media has an unknown duration then and we return TRUE to report
|
|
* this. Note that this is not the same as returning FALSE, which
|
|
* means that we cannot report the duration at all. */
|
|
res = TRUE;
|
|
}
|
|
gst_query_set_duration (query, format, duration);
|
|
break;
|
|
}
|
|
}
|
|
break;
|
|
}
|
|
|
|
case GST_QUERY_SEEKING:
|
|
{
|
|
GstFormat format;
|
|
|
|
gst_query_parse_seeking (query, &format, NULL, NULL, NULL);
|
|
if (format == src->segment.format) {
|
|
gst_query_set_seeking (query, src->segment.format,
|
|
gst_base_src_seekable (src), 0, src->segment.duration);
|
|
res = TRUE;
|
|
} else {
|
|
/* FIXME 0.11: return TRUE + seekable=FALSE for SEEKING query here */
|
|
/* Don't reply to the query to make up for demuxers which don't
|
|
* handle the SEEKING query yet. Players like Totem will fall back
|
|
* to the duration when the SEEKING query isn't answered. */
|
|
res = FALSE;
|
|
}
|
|
break;
|
|
}
|
|
case GST_QUERY_SEGMENT:
|
|
{
|
|
gint64 start, stop;
|
|
|
|
/* no end segment configured, current duration then */
|
|
if ((stop = src->segment.stop) == -1)
|
|
stop = src->segment.duration;
|
|
start = src->segment.start;
|
|
|
|
/* adjust to stream time */
|
|
if (src->segment.time != -1) {
|
|
start -= src->segment.time;
|
|
if (stop != -1)
|
|
stop -= src->segment.time;
|
|
}
|
|
gst_query_set_segment (query, src->segment.rate, src->segment.format,
|
|
start, stop);
|
|
res = TRUE;
|
|
break;
|
|
}
|
|
|
|
case GST_QUERY_FORMATS:
|
|
{
|
|
gst_query_set_formats (query, 3, GST_FORMAT_DEFAULT,
|
|
GST_FORMAT_BYTES, GST_FORMAT_PERCENT);
|
|
res = TRUE;
|
|
break;
|
|
}
|
|
case GST_QUERY_CONVERT:
|
|
{
|
|
GstFormat src_fmt, dest_fmt;
|
|
gint64 src_val, dest_val;
|
|
|
|
gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
|
|
|
|
/* we can only convert between equal formats... */
|
|
if (src_fmt == dest_fmt) {
|
|
dest_val = src_val;
|
|
res = TRUE;
|
|
} else
|
|
res = FALSE;
|
|
|
|
gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
|
|
break;
|
|
}
|
|
case GST_QUERY_LATENCY:
|
|
{
|
|
GstClockTime min, max;
|
|
gboolean live;
|
|
|
|
/* Subclasses should override and implement something usefull */
|
|
res = gst_base_src_query_latency (src, &live, &min, &max);
|
|
|
|
GST_LOG_OBJECT (src, "report latency: live %d, min %" GST_TIME_FORMAT
|
|
", max %" GST_TIME_FORMAT, live, GST_TIME_ARGS (min),
|
|
GST_TIME_ARGS (max));
|
|
|
|
gst_query_set_latency (query, live, min, max);
|
|
break;
|
|
}
|
|
case GST_QUERY_JITTER:
|
|
case GST_QUERY_RATE:
|
|
res = FALSE;
|
|
break;
|
|
case GST_QUERY_BUFFERING:
|
|
{
|
|
GstFormat format;
|
|
gint64 start, stop, estimated;
|
|
|
|
gst_query_parse_buffering_range (query, &format, NULL, NULL, NULL);
|
|
|
|
GST_DEBUG_OBJECT (src, "buffering query in format %s",
|
|
gst_format_get_name (format));
|
|
|
|
if (src->random_access) {
|
|
estimated = 0;
|
|
start = 0;
|
|
if (format == GST_FORMAT_PERCENT)
|
|
stop = GST_FORMAT_PERCENT_MAX;
|
|
else
|
|
stop = src->segment.duration;
|
|
} else {
|
|
estimated = -1;
|
|
start = -1;
|
|
stop = -1;
|
|
}
|
|
/* convert to required format. When the conversion fails, we can't answer
|
|
* the query. When the value is unknown, we can don't perform conversion
|
|
* but report TRUE. */
|
|
if (format != GST_FORMAT_PERCENT && stop != -1) {
|
|
res = gst_pad_query_convert (src->srcpad, src->segment.format,
|
|
stop, &format, &stop);
|
|
} else {
|
|
res = TRUE;
|
|
}
|
|
if (res && format != GST_FORMAT_PERCENT && start != -1)
|
|
res = gst_pad_query_convert (src->srcpad, src->segment.format,
|
|
start, &format, &start);
|
|
|
|
gst_query_set_buffering_range (query, format, start, stop, estimated);
|
|
break;
|
|
}
|
|
default:
|
|
res = FALSE;
|
|
break;
|
|
}
|
|
GST_DEBUG_OBJECT (src, "query %s returns %d", GST_QUERY_TYPE_NAME (query),
|
|
res);
|
|
return res;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_query (GstPad * pad, GstQuery * query)
|
|
{
|
|
GstBaseSrc *src;
|
|
GstBaseSrcClass *bclass;
|
|
gboolean result = FALSE;
|
|
|
|
src = GST_BASE_SRC (gst_pad_get_parent (pad));
|
|
|
|
bclass = GST_BASE_SRC_GET_CLASS (src);
|
|
|
|
if (bclass->query)
|
|
result = bclass->query (src, query);
|
|
else
|
|
result = gst_pad_query_default (pad, query);
|
|
|
|
gst_object_unref (src);
|
|
|
|
return result;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_default_do_seek (GstBaseSrc * src, GstSegment * segment)
|
|
{
|
|
gboolean res = TRUE;
|
|
|
|
/* update our offset if the start/stop position was updated */
|
|
if (segment->format == GST_FORMAT_BYTES) {
|
|
segment->time = segment->start;
|
|
} else if (segment->start == 0) {
|
|
/* seek to start, we can implement a default for this. */
|
|
segment->time = 0;
|
|
} else {
|
|
res = FALSE;
|
|
GST_INFO_OBJECT (src, "Can't do a default seek");
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_do_seek (GstBaseSrc * src, GstSegment * segment)
|
|
{
|
|
GstBaseSrcClass *bclass;
|
|
gboolean result = FALSE;
|
|
|
|
bclass = GST_BASE_SRC_GET_CLASS (src);
|
|
|
|
if (bclass->do_seek)
|
|
result = bclass->do_seek (src, segment);
|
|
|
|
return result;
|
|
}
|
|
|
|
#define SEEK_TYPE_IS_RELATIVE(t) (((t) != GST_SEEK_TYPE_NONE) && ((t) != GST_SEEK_TYPE_SET))
|
|
|
|
static gboolean
|
|
gst_base_src_default_prepare_seek_segment (GstBaseSrc * src, GstEvent * event,
|
|
GstSegment * segment)
|
|
{
|
|
/* By default, we try one of 2 things:
|
|
* - For absolute seek positions, convert the requested position to our
|
|
* configured processing format and place it in the output segment \
|
|
* - For relative seek positions, convert our current (input) values to the
|
|
* seek format, adjust by the relative seek offset and then convert back to
|
|
* the processing format
|
|
*/
|
|
GstSeekType cur_type, stop_type;
|
|
gint64 cur, stop;
|
|
GstSeekFlags flags;
|
|
GstFormat seek_format, dest_format;
|
|
gdouble rate;
|
|
gboolean update;
|
|
gboolean res = TRUE;
|
|
|
|
gst_event_parse_seek (event, &rate, &seek_format, &flags,
|
|
&cur_type, &cur, &stop_type, &stop);
|
|
dest_format = segment->format;
|
|
|
|
if (seek_format == dest_format) {
|
|
gst_segment_set_seek (segment, rate, seek_format, flags,
|
|
cur_type, cur, stop_type, stop, &update);
|
|
return TRUE;
|
|
}
|
|
|
|
if (cur_type != GST_SEEK_TYPE_NONE) {
|
|
/* FIXME: Handle seek_cur & seek_end by converting the input segment vals */
|
|
res =
|
|
gst_pad_query_convert (src->srcpad, seek_format, cur, &dest_format,
|
|
&cur);
|
|
cur_type = GST_SEEK_TYPE_SET;
|
|
}
|
|
|
|
if (res && stop_type != GST_SEEK_TYPE_NONE) {
|
|
/* FIXME: Handle seek_cur & seek_end by converting the input segment vals */
|
|
res =
|
|
gst_pad_query_convert (src->srcpad, seek_format, stop, &dest_format,
|
|
&stop);
|
|
stop_type = GST_SEEK_TYPE_SET;
|
|
}
|
|
|
|
/* And finally, configure our output segment in the desired format */
|
|
gst_segment_set_seek (segment, rate, dest_format, flags, cur_type, cur,
|
|
stop_type, stop, &update);
|
|
|
|
if (!res)
|
|
goto no_format;
|
|
|
|
return res;
|
|
|
|
no_format:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "undefined format given, seek aborted.");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_prepare_seek_segment (GstBaseSrc * src, GstEvent * event,
|
|
GstSegment * seeksegment)
|
|
{
|
|
GstBaseSrcClass *bclass;
|
|
gboolean result = FALSE;
|
|
|
|
bclass = GST_BASE_SRC_GET_CLASS (src);
|
|
|
|
if (bclass->prepare_seek_segment)
|
|
result = bclass->prepare_seek_segment (src, event, seeksegment);
|
|
|
|
return result;
|
|
}
|
|
|
|
/* this code implements the seeking. It is a good example
|
|
* handling all cases.
|
|
*
|
|
* A seek updates the currently configured segment.start
|
|
* and segment.stop values based on the SEEK_TYPE. If the
|
|
* segment.start value is updated, a seek to this new position
|
|
* should be performed.
|
|
*
|
|
* The seek can only be executed when we are not currently
|
|
* streaming any data, to make sure that this is the case, we
|
|
* acquire the STREAM_LOCK which is taken when we are in the
|
|
* _loop() function or when a getrange() is called. Normally
|
|
* we will not receive a seek if we are operating in pull mode
|
|
* though. When we operate as a live source we might block on the live
|
|
* cond, which does not release the STREAM_LOCK. Therefore we will try
|
|
* to grab the LIVE_LOCK instead of the STREAM_LOCK to make sure it is
|
|
* safe to perform the seek.
|
|
*
|
|
* When we are in the loop() function, we might be in the middle
|
|
* of pushing a buffer, which might block in a sink. To make sure
|
|
* that the push gets unblocked we push out a FLUSH_START event.
|
|
* Our loop function will get a WRONG_STATE return value from
|
|
* the push and will pause, effectively releasing the STREAM_LOCK.
|
|
*
|
|
* For a non-flushing seek, we pause the task, which might eventually
|
|
* release the STREAM_LOCK. We say eventually because when the sink
|
|
* blocks on the sample we might wait a very long time until the sink
|
|
* unblocks the sample. In any case we acquire the STREAM_LOCK and
|
|
* can continue the seek. A non-flushing seek is normally done in a
|
|
* running pipeline to perform seamless playback, this means that the sink is
|
|
* PLAYING and will return from its chain function.
|
|
* In the case of a non-flushing seek we need to make sure that the
|
|
* data we output after the seek is continuous with the previous data,
|
|
* this is because a non-flushing seek does not reset the running-time
|
|
* to 0. We do this by closing the currently running segment, ie. sending
|
|
* a new_segment event with the stop position set to the last processed
|
|
* position.
|
|
*
|
|
* After updating the segment.start/stop values, we prepare for
|
|
* streaming again. We push out a FLUSH_STOP to make the peer pad
|
|
* accept data again and we start our task again.
|
|
*
|
|
* A segment seek posts a message on the bus saying that the playback
|
|
* of the segment started. We store the segment flag internally because
|
|
* when we reach the segment.stop we have to post a segment.done
|
|
* instead of EOS when doing a segment seek.
|
|
*/
|
|
/* FIXME (0.11), we have the unlock gboolean here because most current
|
|
* implementations (fdsrc, -base/gst/tcp/, ...) unconditionally unlock, even when
|
|
* the streaming thread isn't running, resulting in bogus unlocks later when it
|
|
* starts. This is fixed by adding unlock_stop, but we should still avoid unlocking
|
|
* unnecessarily for backwards compatibility. Ergo, the unlock variable stays
|
|
* until 0.11
|
|
*/
|
|
static gboolean
|
|
gst_base_src_perform_seek (GstBaseSrc * src, GstEvent * event, gboolean unlock)
|
|
{
|
|
gboolean res = TRUE, tres;
|
|
gdouble rate;
|
|
GstFormat seek_format, dest_format;
|
|
GstSeekFlags flags;
|
|
GstSeekType cur_type, stop_type;
|
|
gint64 cur, stop;
|
|
gboolean flush, playing;
|
|
gboolean update;
|
|
gboolean relative_seek = FALSE;
|
|
gboolean seekseg_configured = FALSE;
|
|
GstSegment seeksegment;
|
|
guint32 seqnum;
|
|
GstEvent *tevent;
|
|
|
|
GST_DEBUG_OBJECT (src, "doing seek");
|
|
|
|
dest_format = src->segment.format;
|
|
|
|
if (event) {
|
|
gst_event_parse_seek (event, &rate, &seek_format, &flags,
|
|
&cur_type, &cur, &stop_type, &stop);
|
|
|
|
relative_seek = SEEK_TYPE_IS_RELATIVE (cur_type) ||
|
|
SEEK_TYPE_IS_RELATIVE (stop_type);
|
|
|
|
if (dest_format != seek_format && !relative_seek) {
|
|
/* If we have an ABSOLUTE position (SEEK_SET only), we can convert it
|
|
* here before taking the stream lock, otherwise we must convert it later,
|
|
* once we have the stream lock and can read the last configures segment
|
|
* start and stop positions */
|
|
gst_segment_init (&seeksegment, dest_format);
|
|
|
|
if (!gst_base_src_prepare_seek_segment (src, event, &seeksegment))
|
|
goto prepare_failed;
|
|
|
|
seekseg_configured = TRUE;
|
|
}
|
|
|
|
flush = flags & GST_SEEK_FLAG_FLUSH;
|
|
seqnum = gst_event_get_seqnum (event);
|
|
} else {
|
|
flush = FALSE;
|
|
/* get next seqnum */
|
|
seqnum = gst_util_seqnum_next ();
|
|
}
|
|
|
|
/* send flush start */
|
|
if (flush) {
|
|
tevent = gst_event_new_flush_start ();
|
|
gst_event_set_seqnum (tevent, seqnum);
|
|
gst_pad_push_event (src->srcpad, tevent);
|
|
} else
|
|
gst_pad_pause_task (src->srcpad);
|
|
|
|
/* unblock streaming thread. */
|
|
gst_base_src_set_flushing (src, TRUE, FALSE, unlock, &playing);
|
|
|
|
/* grab streaming lock, this should eventually be possible, either
|
|
* because the task is paused, our streaming thread stopped
|
|
* or because our peer is flushing. */
|
|
GST_PAD_STREAM_LOCK (src->srcpad);
|
|
if (G_UNLIKELY (src->priv->seqnum == seqnum)) {
|
|
/* we have seen this event before, issue a warning for now */
|
|
GST_WARNING_OBJECT (src, "duplicate event found %" G_GUINT32_FORMAT,
|
|
seqnum);
|
|
} else {
|
|
src->priv->seqnum = seqnum;
|
|
GST_DEBUG_OBJECT (src, "seek with seqnum %" G_GUINT32_FORMAT, seqnum);
|
|
}
|
|
|
|
gst_base_src_set_flushing (src, FALSE, playing, unlock, NULL);
|
|
|
|
/* If we configured the seeksegment above, don't overwrite it now. Otherwise
|
|
* copy the current segment info into the temp segment that we can actually
|
|
* attempt the seek with. We only update the real segment if the seek suceeds. */
|
|
if (!seekseg_configured) {
|
|
memcpy (&seeksegment, &src->segment, sizeof (GstSegment));
|
|
|
|
/* now configure the final seek segment */
|
|
if (event) {
|
|
if (src->segment.format != seek_format) {
|
|
/* OK, here's where we give the subclass a chance to convert the relative
|
|
* seek into an absolute one in the processing format. We set up any
|
|
* absolute seek above, before taking the stream lock. */
|
|
if (!gst_base_src_prepare_seek_segment (src, event, &seeksegment)) {
|
|
GST_DEBUG_OBJECT (src, "Preparing the seek failed after flushing. "
|
|
"Aborting seek");
|
|
res = FALSE;
|
|
}
|
|
} else {
|
|
/* The seek format matches our processing format, no need to ask the
|
|
* the subclass to configure the segment. */
|
|
gst_segment_set_seek (&seeksegment, rate, seek_format, flags,
|
|
cur_type, cur, stop_type, stop, &update);
|
|
}
|
|
}
|
|
/* Else, no seek event passed, so we're just (re)starting the
|
|
current segment. */
|
|
}
|
|
|
|
if (res) {
|
|
GST_DEBUG_OBJECT (src, "segment configured from %" G_GINT64_FORMAT
|
|
" to %" G_GINT64_FORMAT ", position %" G_GINT64_FORMAT,
|
|
seeksegment.start, seeksegment.stop, seeksegment.last_stop);
|
|
|
|
/* do the seek, segment.last_stop contains the new position. */
|
|
res = gst_base_src_do_seek (src, &seeksegment);
|
|
}
|
|
|
|
/* and prepare to continue streaming */
|
|
if (flush) {
|
|
tevent = gst_event_new_flush_stop ();
|
|
gst_event_set_seqnum (tevent, seqnum);
|
|
/* send flush stop, peer will accept data and events again. We
|
|
* are not yet providing data as we still have the STREAM_LOCK. */
|
|
gst_pad_push_event (src->srcpad, tevent);
|
|
} else if (res && src->data.ABI.running) {
|
|
/* we are running the current segment and doing a non-flushing seek,
|
|
* close the segment first based on the last_stop. */
|
|
GST_DEBUG_OBJECT (src, "closing running segment %" G_GINT64_FORMAT
|
|
" to %" G_GINT64_FORMAT, src->segment.start, src->segment.last_stop);
|
|
|
|
/* queue the segment for sending in the stream thread */
|
|
if (src->priv->close_segment)
|
|
gst_event_unref (src->priv->close_segment);
|
|
src->priv->close_segment =
|
|
gst_event_new_new_segment_full (TRUE,
|
|
src->segment.rate, src->segment.applied_rate, src->segment.format,
|
|
src->segment.start, src->segment.last_stop, src->segment.time);
|
|
gst_event_set_seqnum (src->priv->close_segment, seqnum);
|
|
}
|
|
|
|
/* The subclass must have converted the segment to the processing format
|
|
* by now */
|
|
if (res && seeksegment.format != dest_format) {
|
|
GST_DEBUG_OBJECT (src, "Subclass failed to prepare a seek segment "
|
|
"in the correct format. Aborting seek.");
|
|
res = FALSE;
|
|
}
|
|
|
|
/* if successfull seek, we update our real segment and push
|
|
* out the new segment. */
|
|
if (res) {
|
|
memcpy (&src->segment, &seeksegment, sizeof (GstSegment));
|
|
|
|
if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) {
|
|
GstMessage *message;
|
|
|
|
message = gst_message_new_segment_start (GST_OBJECT (src),
|
|
src->segment.format, src->segment.last_stop);
|
|
gst_message_set_seqnum (message, seqnum);
|
|
|
|
gst_element_post_message (GST_ELEMENT (src), message);
|
|
}
|
|
|
|
/* for deriving a stop position for the playback segment from the seek
|
|
* segment, we must take the duration when the stop is not set */
|
|
if ((stop = src->segment.stop) == -1)
|
|
stop = src->segment.duration;
|
|
|
|
GST_DEBUG_OBJECT (src, "Sending newsegment from %" G_GINT64_FORMAT
|
|
" to %" G_GINT64_FORMAT, src->segment.start, stop);
|
|
|
|
/* now replace the old segment so that we send it in the stream thread the
|
|
* next time it is scheduled. */
|
|
if (src->priv->start_segment)
|
|
gst_event_unref (src->priv->start_segment);
|
|
if (src->segment.rate >= 0.0) {
|
|
/* forward, we send data from last_stop to stop */
|
|
src->priv->start_segment =
|
|
gst_event_new_new_segment_full (FALSE,
|
|
src->segment.rate, src->segment.applied_rate, src->segment.format,
|
|
src->segment.last_stop, stop, src->segment.time);
|
|
} else {
|
|
/* reverse, we send data from last_stop to start */
|
|
src->priv->start_segment =
|
|
gst_event_new_new_segment_full (FALSE,
|
|
src->segment.rate, src->segment.applied_rate, src->segment.format,
|
|
src->segment.start, src->segment.last_stop, src->segment.time);
|
|
}
|
|
gst_event_set_seqnum (src->priv->start_segment, seqnum);
|
|
}
|
|
|
|
src->priv->discont = TRUE;
|
|
src->data.ABI.running = TRUE;
|
|
/* and restart the task in case it got paused explicitely or by
|
|
* the FLUSH_START event we pushed out. */
|
|
tres = gst_pad_start_task (src->srcpad, (GstTaskFunction) gst_base_src_loop,
|
|
src->srcpad);
|
|
if (res && !tres)
|
|
res = FALSE;
|
|
|
|
/* and release the lock again so we can continue streaming */
|
|
GST_PAD_STREAM_UNLOCK (src->srcpad);
|
|
|
|
return res;
|
|
|
|
/* ERROR */
|
|
prepare_failed:
|
|
GST_DEBUG_OBJECT (src, "Preparing the seek failed before flushing. "
|
|
"Aborting seek");
|
|
return FALSE;
|
|
}
|
|
|
|
static const GstQueryType *
|
|
gst_base_src_get_query_types (GstElement * element)
|
|
{
|
|
static const GstQueryType query_types[] = {
|
|
GST_QUERY_DURATION,
|
|
GST_QUERY_POSITION,
|
|
GST_QUERY_SEEKING,
|
|
GST_QUERY_SEGMENT,
|
|
GST_QUERY_FORMATS,
|
|
GST_QUERY_LATENCY,
|
|
GST_QUERY_JITTER,
|
|
GST_QUERY_RATE,
|
|
GST_QUERY_CONVERT,
|
|
0
|
|
};
|
|
|
|
return query_types;
|
|
}
|
|
|
|
/* all events send to this element directly. This is mainly done from the
|
|
* application.
|
|
*/
|
|
static gboolean
|
|
gst_base_src_send_event (GstElement * element, GstEvent * event)
|
|
{
|
|
GstBaseSrc *src;
|
|
gboolean result = FALSE;
|
|
|
|
src = GST_BASE_SRC (element);
|
|
|
|
GST_DEBUG_OBJECT (src, "reveived %s event", GST_EVENT_TYPE_NAME (event));
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
/* bidirectional events */
|
|
case GST_EVENT_FLUSH_START:
|
|
case GST_EVENT_FLUSH_STOP:
|
|
/* sending random flushes downstream can break stuff,
|
|
* especially sync since all segment info will get flushed */
|
|
break;
|
|
|
|
/* downstream serialized events */
|
|
case GST_EVENT_EOS:
|
|
{
|
|
GstBaseSrcClass *bclass;
|
|
|
|
bclass = GST_BASE_SRC_GET_CLASS (src);
|
|
|
|
/* queue EOS and make sure the task or pull function performs the EOS
|
|
* actions.
|
|
*
|
|
* We have two possibilities:
|
|
*
|
|
* - Before we are to enter the _create function, we check the pending_eos
|
|
* first and do EOS instead of entering it.
|
|
* - If we are in the _create function or we did not manage to set the
|
|
* flag fast enough and we are about to enter the _create function,
|
|
* we unlock it so that we exit with WRONG_STATE immediatly. We then
|
|
* check the EOS flag and do the EOS logic.
|
|
*/
|
|
g_atomic_int_set (&src->priv->pending_eos, TRUE);
|
|
GST_DEBUG_OBJECT (src, "EOS marked, calling unlock");
|
|
|
|
/* unlock the _create function so that we can check the pending_eos flag
|
|
* and we can do EOS. This will eventually release the LIVE_LOCK again so
|
|
* that we can grab it and stop the unlock again. We don't take the stream
|
|
* lock so that this operation is guaranteed to never block. */
|
|
if (bclass->unlock)
|
|
bclass->unlock (src);
|
|
|
|
GST_DEBUG_OBJECT (src, "unlock called, waiting for LIVE_LOCK");
|
|
|
|
GST_LIVE_LOCK (src);
|
|
GST_DEBUG_OBJECT (src, "LIVE_LOCK acquired, calling unlock_stop");
|
|
/* now stop the unlock of the streaming thread again. Grabbing the live
|
|
* lock is enough because that protects the create function. */
|
|
if (bclass->unlock_stop)
|
|
bclass->unlock_stop (src);
|
|
GST_LIVE_UNLOCK (src);
|
|
|
|
result = TRUE;
|
|
break;
|
|
}
|
|
case GST_EVENT_NEWSEGMENT:
|
|
/* sending random NEWSEGMENT downstream can break sync. */
|
|
break;
|
|
case GST_EVENT_TAG:
|
|
/* Insert tag in the dataflow */
|
|
GST_OBJECT_LOCK (src);
|
|
src->priv->pending_tags = g_list_append (src->priv->pending_tags, event);
|
|
GST_OBJECT_UNLOCK (src);
|
|
event = NULL;
|
|
result = TRUE;
|
|
break;
|
|
case GST_EVENT_BUFFERSIZE:
|
|
/* does not seem to make much sense currently */
|
|
break;
|
|
|
|
/* upstream events */
|
|
case GST_EVENT_QOS:
|
|
/* elements should override send_event and do something */
|
|
break;
|
|
case GST_EVENT_SEEK:
|
|
{
|
|
gboolean started;
|
|
|
|
GST_OBJECT_LOCK (src->srcpad);
|
|
if (GST_PAD_ACTIVATE_MODE (src->srcpad) == GST_ACTIVATE_PULL)
|
|
goto wrong_mode;
|
|
started = GST_PAD_ACTIVATE_MODE (src->srcpad) == GST_ACTIVATE_PUSH;
|
|
GST_OBJECT_UNLOCK (src->srcpad);
|
|
|
|
if (started) {
|
|
GST_DEBUG_OBJECT (src, "performing seek");
|
|
/* when we are running in push mode, we can execute the
|
|
* seek right now, we need to unlock. */
|
|
result = gst_base_src_perform_seek (src, event, TRUE);
|
|
} else {
|
|
GstEvent **event_p;
|
|
|
|
/* else we store the event and execute the seek when we
|
|
* get activated */
|
|
GST_OBJECT_LOCK (src);
|
|
GST_DEBUG_OBJECT (src, "queueing seek");
|
|
event_p = &src->data.ABI.pending_seek;
|
|
gst_event_replace ((GstEvent **) event_p, event);
|
|
GST_OBJECT_UNLOCK (src);
|
|
/* assume the seek will work */
|
|
result = TRUE;
|
|
}
|
|
break;
|
|
}
|
|
case GST_EVENT_NAVIGATION:
|
|
/* could make sense for elements that do something with navigation events
|
|
* but then they would need to override the send_event function */
|
|
break;
|
|
case GST_EVENT_LATENCY:
|
|
/* does not seem to make sense currently */
|
|
break;
|
|
|
|
/* custom events */
|
|
case GST_EVENT_CUSTOM_UPSTREAM:
|
|
/* override send_event if you want this */
|
|
break;
|
|
case GST_EVENT_CUSTOM_DOWNSTREAM:
|
|
case GST_EVENT_CUSTOM_BOTH:
|
|
/* FIXME, insert event in the dataflow */
|
|
break;
|
|
case GST_EVENT_CUSTOM_DOWNSTREAM_OOB:
|
|
case GST_EVENT_CUSTOM_BOTH_OOB:
|
|
/* insert a random custom event into the pipeline */
|
|
GST_DEBUG_OBJECT (src, "pushing custom OOB event downstream");
|
|
result = gst_pad_push_event (src->srcpad, event);
|
|
/* we gave away the ref to the event in the push */
|
|
event = NULL;
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
done:
|
|
/* if we still have a ref to the event, unref it now */
|
|
if (event)
|
|
gst_event_unref (event);
|
|
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
wrong_mode:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "cannot perform seek when operating in pull mode");
|
|
GST_OBJECT_UNLOCK (src->srcpad);
|
|
result = FALSE;
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_seekable (GstBaseSrc * src)
|
|
{
|
|
GstBaseSrcClass *bclass;
|
|
bclass = GST_BASE_SRC_GET_CLASS (src);
|
|
if (bclass->is_seekable)
|
|
return bclass->is_seekable (src);
|
|
else
|
|
return FALSE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_default_event (GstBaseSrc * src, GstEvent * event)
|
|
{
|
|
gboolean result;
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_SEEK:
|
|
/* is normally called when in push mode */
|
|
if (!gst_base_src_seekable (src))
|
|
goto not_seekable;
|
|
|
|
result = gst_base_src_perform_seek (src, event, TRUE);
|
|
break;
|
|
case GST_EVENT_FLUSH_START:
|
|
/* cancel any blocking getrange, is normally called
|
|
* when in pull mode. */
|
|
result = gst_base_src_set_flushing (src, TRUE, FALSE, TRUE, NULL);
|
|
break;
|
|
case GST_EVENT_FLUSH_STOP:
|
|
result = gst_base_src_set_flushing (src, FALSE, TRUE, TRUE, NULL);
|
|
break;
|
|
default:
|
|
result = TRUE;
|
|
break;
|
|
}
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
not_seekable:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "is not seekable");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_event_handler (GstPad * pad, GstEvent * event)
|
|
{
|
|
GstBaseSrc *src;
|
|
GstBaseSrcClass *bclass;
|
|
gboolean result = FALSE;
|
|
|
|
src = GST_BASE_SRC (gst_pad_get_parent (pad));
|
|
bclass = GST_BASE_SRC_GET_CLASS (src);
|
|
|
|
if (bclass->event) {
|
|
if (!(result = bclass->event (src, event)))
|
|
goto subclass_failed;
|
|
}
|
|
|
|
done:
|
|
gst_event_unref (event);
|
|
gst_object_unref (src);
|
|
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
subclass_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "subclass refused event");
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_base_src_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstBaseSrc *src;
|
|
|
|
src = GST_BASE_SRC (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_BLOCKSIZE:
|
|
gst_base_src_set_blocksize (src, g_value_get_ulong (value));
|
|
break;
|
|
case PROP_NUM_BUFFERS:
|
|
src->num_buffers = g_value_get_int (value);
|
|
break;
|
|
case PROP_TYPEFIND:
|
|
src->data.ABI.typefind = g_value_get_boolean (value);
|
|
break;
|
|
case PROP_DO_TIMESTAMP:
|
|
gst_base_src_set_do_timestamp (src, g_value_get_boolean (value));
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_base_src_get_property (GObject * object, guint prop_id, GValue * value,
|
|
GParamSpec * pspec)
|
|
{
|
|
GstBaseSrc *src;
|
|
|
|
src = GST_BASE_SRC (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_BLOCKSIZE:
|
|
g_value_set_ulong (value, gst_base_src_get_blocksize (src));
|
|
break;
|
|
case PROP_NUM_BUFFERS:
|
|
g_value_set_int (value, src->num_buffers);
|
|
break;
|
|
case PROP_TYPEFIND:
|
|
g_value_set_boolean (value, src->data.ABI.typefind);
|
|
break;
|
|
case PROP_DO_TIMESTAMP:
|
|
g_value_set_boolean (value, gst_base_src_get_do_timestamp (src));
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
/* with STREAM_LOCK and LOCK */
|
|
static GstClockReturn
|
|
gst_base_src_wait (GstBaseSrc * basesrc, GstClock * clock, GstClockTime time)
|
|
{
|
|
GstClockReturn ret;
|
|
GstClockID id;
|
|
|
|
id = gst_clock_new_single_shot_id (clock, time);
|
|
|
|
basesrc->clock_id = id;
|
|
/* release the live lock while waiting */
|
|
GST_LIVE_UNLOCK (basesrc);
|
|
|
|
ret = gst_clock_id_wait (id, NULL);
|
|
|
|
GST_LIVE_LOCK (basesrc);
|
|
gst_clock_id_unref (id);
|
|
basesrc->clock_id = NULL;
|
|
|
|
return ret;
|
|
}
|
|
|
|
/* perform synchronisation on a buffer.
|
|
* with STREAM_LOCK.
|
|
*/
|
|
static GstClockReturn
|
|
gst_base_src_do_sync (GstBaseSrc * basesrc, GstBuffer * buffer)
|
|
{
|
|
GstClockReturn result;
|
|
GstClockTime start, end;
|
|
GstBaseSrcClass *bclass;
|
|
GstClockTime base_time;
|
|
GstClock *clock;
|
|
GstClockTime now = GST_CLOCK_TIME_NONE, timestamp;
|
|
gboolean do_timestamp, first, pseudo_live;
|
|
|
|
bclass = GST_BASE_SRC_GET_CLASS (basesrc);
|
|
|
|
start = end = -1;
|
|
if (bclass->get_times)
|
|
bclass->get_times (basesrc, buffer, &start, &end);
|
|
|
|
/* get buffer timestamp */
|
|
timestamp = GST_BUFFER_TIMESTAMP (buffer);
|
|
|
|
/* grab the lock to prepare for clocking and calculate the startup
|
|
* latency. */
|
|
GST_OBJECT_LOCK (basesrc);
|
|
|
|
/* if we are asked to sync against the clock we are a pseudo live element */
|
|
pseudo_live = (start != -1 && basesrc->is_live);
|
|
/* check for the first buffer */
|
|
first = (basesrc->priv->latency == -1);
|
|
|
|
if (timestamp != -1 && pseudo_live) {
|
|
GstClockTime latency;
|
|
|
|
/* we have a timestamp and a sync time, latency is the diff */
|
|
if (timestamp <= start)
|
|
latency = start - timestamp;
|
|
else
|
|
latency = 0;
|
|
|
|
if (first) {
|
|
GST_DEBUG_OBJECT (basesrc, "pseudo_live with latency %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (latency));
|
|
/* first time we calculate latency, just configure */
|
|
basesrc->priv->latency = latency;
|
|
} else {
|
|
if (basesrc->priv->latency != latency) {
|
|
/* we have a new latency, FIXME post latency message */
|
|
basesrc->priv->latency = latency;
|
|
GST_DEBUG_OBJECT (basesrc, "latency changed to %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (latency));
|
|
}
|
|
}
|
|
} else if (first) {
|
|
GST_DEBUG_OBJECT (basesrc, "no latency needed, live %d, sync %d",
|
|
basesrc->is_live, start != -1);
|
|
basesrc->priv->latency = 0;
|
|
}
|
|
|
|
/* get clock, if no clock, we can't sync or do timestamps */
|
|
if ((clock = GST_ELEMENT_CLOCK (basesrc)) == NULL)
|
|
goto no_clock;
|
|
|
|
base_time = GST_ELEMENT_CAST (basesrc)->base_time;
|
|
|
|
do_timestamp = basesrc->priv->do_timestamp;
|
|
|
|
/* first buffer, calculate the timestamp offset */
|
|
if (first) {
|
|
GstClockTime running_time;
|
|
|
|
now = gst_clock_get_time (clock);
|
|
running_time = now - base_time;
|
|
|
|
GST_LOG_OBJECT (basesrc,
|
|
"startup timestamp: %" GST_TIME_FORMAT ", running_time %"
|
|
GST_TIME_FORMAT, GST_TIME_ARGS (timestamp),
|
|
GST_TIME_ARGS (running_time));
|
|
|
|
if (pseudo_live && timestamp != -1) {
|
|
/* live source and we need to sync, add startup latency to all timestamps
|
|
* to get the real running_time. Live sources should always timestamp
|
|
* according to the current running time. */
|
|
basesrc->priv->ts_offset = GST_CLOCK_DIFF (timestamp, running_time);
|
|
|
|
GST_LOG_OBJECT (basesrc, "live with sync, ts_offset %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (basesrc->priv->ts_offset));
|
|
} else {
|
|
basesrc->priv->ts_offset = 0;
|
|
GST_LOG_OBJECT (basesrc, "no timestamp offset needed");
|
|
}
|
|
|
|
if (!GST_CLOCK_TIME_IS_VALID (timestamp)) {
|
|
if (do_timestamp)
|
|
timestamp = running_time;
|
|
else
|
|
timestamp = 0;
|
|
|
|
GST_BUFFER_TIMESTAMP (buffer) = timestamp;
|
|
|
|
GST_LOG_OBJECT (basesrc, "created timestamp: %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (timestamp));
|
|
}
|
|
|
|
/* add the timestamp offset we need for sync */
|
|
timestamp += basesrc->priv->ts_offset;
|
|
} else {
|
|
/* not the first buffer, the timestamp is the diff between the clock and
|
|
* base_time */
|
|
if (do_timestamp && !GST_CLOCK_TIME_IS_VALID (timestamp)) {
|
|
now = gst_clock_get_time (clock);
|
|
|
|
GST_BUFFER_TIMESTAMP (buffer) = now - base_time;
|
|
|
|
GST_LOG_OBJECT (basesrc, "created timestamp: %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (now - base_time));
|
|
}
|
|
}
|
|
|
|
/* if we don't have a buffer timestamp, we don't sync */
|
|
if (!GST_CLOCK_TIME_IS_VALID (start))
|
|
goto no_sync;
|
|
|
|
if (basesrc->is_live && GST_CLOCK_TIME_IS_VALID (timestamp)) {
|
|
/* for pseudo live sources, add our ts_offset to the timestamp */
|
|
GST_BUFFER_TIMESTAMP (buffer) += basesrc->priv->ts_offset;
|
|
start += basesrc->priv->ts_offset;
|
|
}
|
|
|
|
GST_LOG_OBJECT (basesrc,
|
|
"waiting for clock, base time %" GST_TIME_FORMAT
|
|
", stream_start %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (base_time), GST_TIME_ARGS (start));
|
|
GST_OBJECT_UNLOCK (basesrc);
|
|
|
|
result = gst_base_src_wait (basesrc, clock, start + base_time);
|
|
|
|
GST_LOG_OBJECT (basesrc, "clock entry done: %d", result);
|
|
|
|
return result;
|
|
|
|
/* special cases */
|
|
no_clock:
|
|
{
|
|
GST_DEBUG_OBJECT (basesrc, "we have no clock");
|
|
GST_OBJECT_UNLOCK (basesrc);
|
|
return GST_CLOCK_OK;
|
|
}
|
|
no_sync:
|
|
{
|
|
GST_DEBUG_OBJECT (basesrc, "no sync needed");
|
|
GST_OBJECT_UNLOCK (basesrc);
|
|
return GST_CLOCK_OK;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_update_length (GstBaseSrc * src, guint64 offset, guint * length)
|
|
{
|
|
guint64 size, maxsize;
|
|
GstBaseSrcClass *bclass;
|
|
|
|
bclass = GST_BASE_SRC_GET_CLASS (src);
|
|
|
|
/* only operate if we are working with bytes */
|
|
if (src->segment.format != GST_FORMAT_BYTES)
|
|
return TRUE;
|
|
|
|
/* get total file size */
|
|
size = (guint64) src->segment.duration;
|
|
|
|
/* the max amount of bytes to read is the total size or
|
|
* up to the segment.stop if present. */
|
|
if (src->segment.stop != -1)
|
|
maxsize = MIN (size, src->segment.stop);
|
|
else
|
|
maxsize = size;
|
|
|
|
GST_DEBUG_OBJECT (src,
|
|
"reading offset %" G_GUINT64_FORMAT ", length %u, size %" G_GINT64_FORMAT
|
|
", segment.stop %" G_GINT64_FORMAT ", maxsize %" G_GINT64_FORMAT, offset,
|
|
*length, size, src->segment.stop, maxsize);
|
|
|
|
/* check size if we have one */
|
|
if (maxsize != -1) {
|
|
/* if we run past the end, check if the file became bigger and
|
|
* retry. */
|
|
if (G_UNLIKELY (offset + *length >= maxsize)) {
|
|
/* see if length of the file changed */
|
|
if (bclass->get_size)
|
|
if (!bclass->get_size (src, &size))
|
|
size = -1;
|
|
|
|
gst_segment_set_duration (&src->segment, GST_FORMAT_BYTES, size);
|
|
|
|
/* make sure we don't exceed the configured segment stop
|
|
* if it was set */
|
|
if (src->segment.stop != -1)
|
|
maxsize = MIN (size, src->segment.stop);
|
|
else
|
|
maxsize = size;
|
|
|
|
/* if we are at or past the end, EOS */
|
|
if (G_UNLIKELY (offset >= maxsize))
|
|
goto unexpected_length;
|
|
|
|
/* else we can clip to the end */
|
|
if (G_UNLIKELY (offset + *length >= maxsize))
|
|
*length = maxsize - offset;
|
|
|
|
}
|
|
}
|
|
|
|
/* keep track of current position. segment is in bytes, we checked
|
|
* that above. */
|
|
gst_segment_set_last_stop (&src->segment, GST_FORMAT_BYTES, offset);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
unexpected_length:
|
|
{
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/* must be called with LIVE_LOCK */
|
|
static GstFlowReturn
|
|
gst_base_src_get_range (GstBaseSrc * src, guint64 offset, guint length,
|
|
GstBuffer ** buf)
|
|
{
|
|
GstFlowReturn ret;
|
|
GstBaseSrcClass *bclass;
|
|
GstClockReturn status;
|
|
|
|
bclass = GST_BASE_SRC_GET_CLASS (src);
|
|
|
|
if (src->is_live) {
|
|
while (G_UNLIKELY (!src->live_running)) {
|
|
ret = gst_base_src_wait_playing (src);
|
|
if (ret != GST_FLOW_OK)
|
|
goto stopped;
|
|
}
|
|
}
|
|
|
|
if (G_UNLIKELY (!GST_OBJECT_FLAG_IS_SET (src, GST_BASE_SRC_STARTED)))
|
|
goto not_started;
|
|
|
|
if (G_UNLIKELY (!bclass->create))
|
|
goto no_function;
|
|
|
|
if (G_UNLIKELY (!gst_base_src_update_length (src, offset, &length)))
|
|
goto unexpected_length;
|
|
|
|
/* normally we don't count buffers */
|
|
if (G_UNLIKELY (src->num_buffers_left >= 0)) {
|
|
if (src->num_buffers_left == 0)
|
|
goto reached_num_buffers;
|
|
else
|
|
src->num_buffers_left--;
|
|
}
|
|
|
|
/* don't enter the create function if a pending EOS event was set. For the
|
|
* logic of the pending_eos, check the event function of this class. */
|
|
if (G_UNLIKELY (g_atomic_int_get (&src->priv->pending_eos)))
|
|
goto eos;
|
|
|
|
GST_DEBUG_OBJECT (src,
|
|
"calling create offset %" G_GUINT64_FORMAT " length %u, time %"
|
|
G_GINT64_FORMAT, offset, length, src->segment.time);
|
|
|
|
ret = bclass->create (src, offset, length, buf);
|
|
|
|
/* The create function could be unlocked because we have a pending EOS. It's
|
|
* possible that we have a valid buffer from create that we need to
|
|
* discard when the create function returned _OK. */
|
|
if (G_UNLIKELY (g_atomic_int_get (&src->priv->pending_eos))) {
|
|
if (ret == GST_FLOW_OK) {
|
|
gst_buffer_unref (*buf);
|
|
*buf = NULL;
|
|
}
|
|
goto eos;
|
|
}
|
|
|
|
if (G_UNLIKELY (ret != GST_FLOW_OK))
|
|
goto not_ok;
|
|
|
|
/* no timestamp set and we are at offset 0, we can timestamp with 0 */
|
|
if (offset == 0 && src->segment.time == 0
|
|
&& GST_BUFFER_TIMESTAMP (*buf) == -1)
|
|
GST_BUFFER_TIMESTAMP (*buf) = 0;
|
|
|
|
/* set pad caps on the buffer if the buffer had no caps */
|
|
if (GST_BUFFER_CAPS (*buf) == NULL)
|
|
gst_buffer_set_caps (*buf, GST_PAD_CAPS (src->srcpad));
|
|
|
|
/* now sync before pushing the buffer */
|
|
status = gst_base_src_do_sync (src, *buf);
|
|
|
|
/* waiting for the clock could have made us flushing */
|
|
if (G_UNLIKELY (src->priv->flushing))
|
|
goto flushing;
|
|
|
|
switch (status) {
|
|
case GST_CLOCK_EARLY:
|
|
/* the buffer is too late. We currently don't drop the buffer. */
|
|
GST_DEBUG_OBJECT (src, "buffer too late!, returning anyway");
|
|
break;
|
|
case GST_CLOCK_OK:
|
|
/* buffer synchronised properly */
|
|
GST_DEBUG_OBJECT (src, "buffer ok");
|
|
break;
|
|
case GST_CLOCK_UNSCHEDULED:
|
|
/* this case is triggered when we were waiting for the clock and
|
|
* it got unlocked because we did a state change. We return
|
|
* WRONG_STATE in this case to stop the dataflow also get rid of the
|
|
* produced buffer. */
|
|
GST_DEBUG_OBJECT (src,
|
|
"clock was unscheduled (%d), returning WRONG_STATE", status);
|
|
gst_buffer_unref (*buf);
|
|
*buf = NULL;
|
|
ret = GST_FLOW_WRONG_STATE;
|
|
break;
|
|
default:
|
|
/* all other result values are unexpected and errors */
|
|
GST_ELEMENT_ERROR (src, CORE, CLOCK,
|
|
(_("Internal clock error.")),
|
|
("clock returned unexpected return value %d", status));
|
|
gst_buffer_unref (*buf);
|
|
*buf = NULL;
|
|
ret = GST_FLOW_ERROR;
|
|
break;
|
|
}
|
|
return ret;
|
|
|
|
/* ERROR */
|
|
stopped:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "wait_playing returned %d (%s)", ret,
|
|
gst_flow_get_name (ret));
|
|
return ret;
|
|
}
|
|
not_ok:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "create returned %d (%s)", ret,
|
|
gst_flow_get_name (ret));
|
|
return ret;
|
|
}
|
|
not_started:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "getrange but not started");
|
|
return GST_FLOW_WRONG_STATE;
|
|
}
|
|
no_function:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "no create function");
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
unexpected_length:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "unexpected length %u (offset=%" G_GUINT64_FORMAT
|
|
", size=%" G_GINT64_FORMAT ")", length, offset, src->segment.duration);
|
|
return GST_FLOW_UNEXPECTED;
|
|
}
|
|
reached_num_buffers:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "sent all buffers");
|
|
return GST_FLOW_UNEXPECTED;
|
|
}
|
|
flushing:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "we are flushing");
|
|
gst_buffer_unref (*buf);
|
|
*buf = NULL;
|
|
return GST_FLOW_WRONG_STATE;
|
|
}
|
|
eos:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "we are EOS");
|
|
return GST_FLOW_UNEXPECTED;
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_base_src_pad_get_range (GstPad * pad, guint64 offset, guint length,
|
|
GstBuffer ** buf)
|
|
{
|
|
GstBaseSrc *src;
|
|
GstFlowReturn res;
|
|
|
|
src = GST_BASE_SRC (gst_pad_get_parent (pad));
|
|
|
|
GST_LIVE_LOCK (src);
|
|
if (G_UNLIKELY (src->priv->flushing))
|
|
goto flushing;
|
|
|
|
res = gst_base_src_get_range (src, offset, length, buf);
|
|
|
|
done:
|
|
GST_LIVE_UNLOCK (src);
|
|
|
|
gst_object_unref (src);
|
|
|
|
return res;
|
|
|
|
/* ERRORS */
|
|
flushing:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "we are flushing");
|
|
res = GST_FLOW_WRONG_STATE;
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_default_check_get_range (GstBaseSrc * src)
|
|
{
|
|
gboolean res;
|
|
|
|
if (!GST_OBJECT_FLAG_IS_SET (src, GST_BASE_SRC_STARTED)) {
|
|
GST_LOG_OBJECT (src, "doing start/stop to check get_range support");
|
|
if (G_LIKELY (gst_base_src_start (src)))
|
|
gst_base_src_stop (src);
|
|
}
|
|
|
|
/* we can operate in getrange mode if the native format is bytes
|
|
* and we are seekable, this condition is set in the random_access
|
|
* flag and is set in the _start() method. */
|
|
res = src->random_access;
|
|
|
|
return res;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_check_get_range (GstBaseSrc * src)
|
|
{
|
|
GstBaseSrcClass *bclass;
|
|
gboolean res;
|
|
|
|
bclass = GST_BASE_SRC_GET_CLASS (src);
|
|
|
|
if (bclass->check_get_range == NULL)
|
|
goto no_function;
|
|
|
|
res = bclass->check_get_range (src);
|
|
GST_LOG_OBJECT (src, "%s() returned %d",
|
|
GST_DEBUG_FUNCPTR_NAME (bclass->check_get_range), (gint) res);
|
|
|
|
return res;
|
|
|
|
/* ERRORS */
|
|
no_function:
|
|
{
|
|
GST_WARNING_OBJECT (src, "no check_get_range function set");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_pad_check_get_range (GstPad * pad)
|
|
{
|
|
GstBaseSrc *src;
|
|
gboolean res;
|
|
|
|
src = GST_BASE_SRC (GST_OBJECT_PARENT (pad));
|
|
|
|
res = gst_base_src_check_get_range (src);
|
|
|
|
return res;
|
|
}
|
|
|
|
static void
|
|
gst_base_src_loop (GstPad * pad)
|
|
{
|
|
GstBaseSrc *src;
|
|
GstBuffer *buf = NULL;
|
|
GstFlowReturn ret;
|
|
gint64 position;
|
|
gboolean eos;
|
|
gulong blocksize;
|
|
GList *tags, *tmp;
|
|
|
|
eos = FALSE;
|
|
|
|
src = GST_BASE_SRC (GST_OBJECT_PARENT (pad));
|
|
|
|
GST_LIVE_LOCK (src);
|
|
|
|
if (G_UNLIKELY (src->priv->flushing))
|
|
goto flushing;
|
|
|
|
src->priv->last_sent_eos = FALSE;
|
|
|
|
blocksize = src->blocksize;
|
|
|
|
/* if we operate in bytes, we can calculate an offset */
|
|
if (src->segment.format == GST_FORMAT_BYTES) {
|
|
position = src->segment.last_stop;
|
|
/* for negative rates, start with subtracting the blocksize */
|
|
if (src->segment.rate < 0.0) {
|
|
/* we cannot go below segment.start */
|
|
if (position > src->segment.start + blocksize)
|
|
position -= blocksize;
|
|
else {
|
|
/* last block, remainder up to segment.start */
|
|
blocksize = position - src->segment.start;
|
|
position = src->segment.start;
|
|
}
|
|
}
|
|
} else
|
|
position = -1;
|
|
|
|
GST_LOG_OBJECT (src, "next_ts %" GST_TIME_FORMAT " size %lu",
|
|
GST_TIME_ARGS (position), blocksize);
|
|
|
|
ret = gst_base_src_get_range (src, position, blocksize, &buf);
|
|
if (G_UNLIKELY (ret != GST_FLOW_OK)) {
|
|
GST_INFO_OBJECT (src, "pausing after gst_base_src_get_range() = %s",
|
|
gst_flow_get_name (ret));
|
|
GST_LIVE_UNLOCK (src);
|
|
goto pause;
|
|
}
|
|
/* this should not happen */
|
|
if (G_UNLIKELY (buf == NULL))
|
|
goto null_buffer;
|
|
|
|
/* push events to close/start our segment before we push the buffer. */
|
|
if (G_UNLIKELY (src->priv->close_segment)) {
|
|
gst_pad_push_event (pad, src->priv->close_segment);
|
|
src->priv->close_segment = NULL;
|
|
}
|
|
if (G_UNLIKELY (src->priv->start_segment)) {
|
|
gst_pad_push_event (pad, src->priv->start_segment);
|
|
src->priv->start_segment = NULL;
|
|
}
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
/* take the tags */
|
|
tags = src->priv->pending_tags;
|
|
src->priv->pending_tags = NULL;
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
/* Push out pending tags if any */
|
|
if (G_UNLIKELY (tags != NULL)) {
|
|
for (tmp = tags; tmp; tmp = g_list_next (tmp)) {
|
|
GstEvent *ev = (GstEvent *) tmp->data;
|
|
gst_pad_push_event (pad, ev);
|
|
}
|
|
g_list_free (tags);
|
|
}
|
|
|
|
/* figure out the new position */
|
|
switch (src->segment.format) {
|
|
case GST_FORMAT_BYTES:
|
|
{
|
|
guint bufsize = GST_BUFFER_SIZE (buf);
|
|
|
|
/* we subtracted above for negative rates */
|
|
if (src->segment.rate >= 0.0)
|
|
position += bufsize;
|
|
break;
|
|
}
|
|
case GST_FORMAT_TIME:
|
|
{
|
|
GstClockTime start, duration;
|
|
|
|
start = GST_BUFFER_TIMESTAMP (buf);
|
|
duration = GST_BUFFER_DURATION (buf);
|
|
|
|
if (GST_CLOCK_TIME_IS_VALID (start))
|
|
position = start;
|
|
else
|
|
position = src->segment.last_stop;
|
|
|
|
if (GST_CLOCK_TIME_IS_VALID (duration)) {
|
|
if (src->segment.rate >= 0.0)
|
|
position += duration;
|
|
else if (position > duration)
|
|
position -= duration;
|
|
else
|
|
position = 0;
|
|
}
|
|
break;
|
|
}
|
|
case GST_FORMAT_DEFAULT:
|
|
if (src->segment.rate >= 0.0)
|
|
position = GST_BUFFER_OFFSET_END (buf);
|
|
else
|
|
position = GST_BUFFER_OFFSET (buf);
|
|
break;
|
|
default:
|
|
position = -1;
|
|
break;
|
|
}
|
|
if (position != -1) {
|
|
if (src->segment.rate >= 0.0) {
|
|
/* positive rate, check if we reached the stop */
|
|
if (src->segment.stop != -1) {
|
|
if (position >= src->segment.stop) {
|
|
eos = TRUE;
|
|
position = src->segment.stop;
|
|
}
|
|
}
|
|
} else {
|
|
/* negative rate, check if we reached the start. start is always set to
|
|
* something different from -1 */
|
|
if (position <= src->segment.start) {
|
|
eos = TRUE;
|
|
position = src->segment.start;
|
|
}
|
|
/* when going reverse, all buffers are DISCONT */
|
|
src->priv->discont = TRUE;
|
|
}
|
|
gst_segment_set_last_stop (&src->segment, src->segment.format, position);
|
|
}
|
|
|
|
if (G_UNLIKELY (src->priv->discont)) {
|
|
buf = gst_buffer_make_metadata_writable (buf);
|
|
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
|
|
src->priv->discont = FALSE;
|
|
}
|
|
GST_LIVE_UNLOCK (src);
|
|
|
|
ret = gst_pad_push (pad, buf);
|
|
if (G_UNLIKELY (ret != GST_FLOW_OK)) {
|
|
GST_INFO_OBJECT (src, "pausing after gst_pad_push() = %s",
|
|
gst_flow_get_name (ret));
|
|
goto pause;
|
|
}
|
|
|
|
if (G_UNLIKELY (eos)) {
|
|
GST_INFO_OBJECT (src, "pausing after end of segment");
|
|
ret = GST_FLOW_UNEXPECTED;
|
|
goto pause;
|
|
}
|
|
|
|
done:
|
|
return;
|
|
|
|
/* special cases */
|
|
flushing:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "we are flushing");
|
|
GST_LIVE_UNLOCK (src);
|
|
ret = GST_FLOW_WRONG_STATE;
|
|
goto pause;
|
|
}
|
|
pause:
|
|
{
|
|
const gchar *reason = gst_flow_get_name (ret);
|
|
GstEvent *event;
|
|
|
|
GST_DEBUG_OBJECT (src, "pausing task, reason %s", reason);
|
|
src->data.ABI.running = FALSE;
|
|
gst_pad_pause_task (pad);
|
|
if (GST_FLOW_IS_FATAL (ret) || ret == GST_FLOW_NOT_LINKED) {
|
|
if (ret == GST_FLOW_UNEXPECTED) {
|
|
/* perform EOS logic */
|
|
if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) {
|
|
GstMessage *message;
|
|
|
|
message = gst_message_new_segment_done (GST_OBJECT_CAST (src),
|
|
src->segment.format, src->segment.last_stop);
|
|
gst_message_set_seqnum (message, src->priv->seqnum);
|
|
gst_element_post_message (GST_ELEMENT_CAST (src), message);
|
|
} else {
|
|
event = gst_event_new_eos ();
|
|
gst_event_set_seqnum (event, src->priv->seqnum);
|
|
gst_pad_push_event (pad, event);
|
|
src->priv->last_sent_eos = TRUE;
|
|
}
|
|
} else {
|
|
event = gst_event_new_eos ();
|
|
gst_event_set_seqnum (event, src->priv->seqnum);
|
|
/* for fatal errors we post an error message, post the error
|
|
* first so the app knows about the error first. */
|
|
GST_ELEMENT_ERROR (src, STREAM, FAILED,
|
|
(_("Internal data flow error.")),
|
|
("streaming task paused, reason %s (%d)", reason, ret));
|
|
gst_pad_push_event (pad, event);
|
|
src->priv->last_sent_eos = TRUE;
|
|
}
|
|
}
|
|
goto done;
|
|
}
|
|
null_buffer:
|
|
{
|
|
GST_ELEMENT_ERROR (src, STREAM, FAILED,
|
|
(_("Internal data flow error.")), ("element returned NULL buffer"));
|
|
GST_LIVE_UNLOCK (src);
|
|
/* we finished the segment on error */
|
|
ret = GST_FLOW_ERROR;
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
/* default negotiation code.
|
|
*
|
|
* Take intersection between src and sink pads, take first
|
|
* caps and fixate.
|
|
*/
|
|
static gboolean
|
|
gst_base_src_default_negotiate (GstBaseSrc * basesrc)
|
|
{
|
|
GstCaps *thiscaps;
|
|
GstCaps *caps = NULL;
|
|
GstCaps *peercaps = NULL;
|
|
gboolean result = FALSE;
|
|
|
|
/* first see what is possible on our source pad */
|
|
thiscaps = gst_pad_get_caps (GST_BASE_SRC_PAD (basesrc));
|
|
GST_DEBUG_OBJECT (basesrc, "caps of src: %" GST_PTR_FORMAT, thiscaps);
|
|
/* nothing or anything is allowed, we're done */
|
|
if (thiscaps == NULL || gst_caps_is_any (thiscaps))
|
|
goto no_nego_needed;
|
|
|
|
if (G_UNLIKELY (gst_caps_is_empty (thiscaps)))
|
|
goto no_caps;
|
|
|
|
/* get the peer caps */
|
|
peercaps = gst_pad_peer_get_caps (GST_BASE_SRC_PAD (basesrc));
|
|
GST_DEBUG_OBJECT (basesrc, "caps of peer: %" GST_PTR_FORMAT, peercaps);
|
|
if (peercaps) {
|
|
GstCaps *icaps;
|
|
|
|
/* get intersection */
|
|
icaps = gst_caps_intersect (thiscaps, peercaps);
|
|
GST_DEBUG_OBJECT (basesrc, "intersect: %" GST_PTR_FORMAT, icaps);
|
|
gst_caps_unref (thiscaps);
|
|
gst_caps_unref (peercaps);
|
|
if (icaps) {
|
|
/* take first (and best, since they are sorted) possibility */
|
|
caps = gst_caps_copy_nth (icaps, 0);
|
|
gst_caps_unref (icaps);
|
|
}
|
|
} else {
|
|
/* no peer, work with our own caps then */
|
|
caps = thiscaps;
|
|
}
|
|
if (caps) {
|
|
caps = gst_caps_make_writable (caps);
|
|
gst_caps_truncate (caps);
|
|
|
|
/* now fixate */
|
|
if (!gst_caps_is_empty (caps)) {
|
|
gst_pad_fixate_caps (GST_BASE_SRC_PAD (basesrc), caps);
|
|
GST_DEBUG_OBJECT (basesrc, "fixated to: %" GST_PTR_FORMAT, caps);
|
|
|
|
if (gst_caps_is_any (caps)) {
|
|
/* hmm, still anything, so element can do anything and
|
|
* nego is not needed */
|
|
result = TRUE;
|
|
} else if (gst_caps_is_fixed (caps)) {
|
|
/* yay, fixed caps, use those then, it's possible that the subclass does
|
|
* not accept this caps after all and we have to fail. */
|
|
result = gst_pad_set_caps (GST_BASE_SRC_PAD (basesrc), caps);
|
|
}
|
|
}
|
|
gst_caps_unref (caps);
|
|
} else {
|
|
GST_DEBUG_OBJECT (basesrc, "no common caps");
|
|
}
|
|
return result;
|
|
|
|
no_nego_needed:
|
|
{
|
|
GST_DEBUG_OBJECT (basesrc, "no negotiation needed");
|
|
if (thiscaps)
|
|
gst_caps_unref (thiscaps);
|
|
return TRUE;
|
|
}
|
|
no_caps:
|
|
{
|
|
GST_ELEMENT_ERROR (basesrc, STREAM, FORMAT,
|
|
("No supported formats found"),
|
|
("This element did not produce valid caps"));
|
|
if (thiscaps)
|
|
gst_caps_unref (thiscaps);
|
|
return TRUE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_negotiate (GstBaseSrc * basesrc)
|
|
{
|
|
GstBaseSrcClass *bclass;
|
|
gboolean result = TRUE;
|
|
|
|
bclass = GST_BASE_SRC_GET_CLASS (basesrc);
|
|
|
|
if (bclass->negotiate)
|
|
result = bclass->negotiate (basesrc);
|
|
|
|
return result;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_start (GstBaseSrc * basesrc)
|
|
{
|
|
GstBaseSrcClass *bclass;
|
|
gboolean result;
|
|
guint64 size;
|
|
gboolean seekable;
|
|
|
|
if (GST_OBJECT_FLAG_IS_SET (basesrc, GST_BASE_SRC_STARTED))
|
|
return TRUE;
|
|
|
|
GST_DEBUG_OBJECT (basesrc, "starting source");
|
|
|
|
basesrc->num_buffers_left = basesrc->num_buffers;
|
|
|
|
gst_segment_init (&basesrc->segment, basesrc->segment.format);
|
|
basesrc->data.ABI.running = FALSE;
|
|
|
|
bclass = GST_BASE_SRC_GET_CLASS (basesrc);
|
|
if (bclass->start)
|
|
result = bclass->start (basesrc);
|
|
else
|
|
result = TRUE;
|
|
|
|
if (!result)
|
|
goto could_not_start;
|
|
|
|
GST_OBJECT_FLAG_SET (basesrc, GST_BASE_SRC_STARTED);
|
|
|
|
/* figure out the size */
|
|
if (basesrc->segment.format == GST_FORMAT_BYTES) {
|
|
if (bclass->get_size) {
|
|
if (!(result = bclass->get_size (basesrc, &size)))
|
|
size = -1;
|
|
} else {
|
|
result = FALSE;
|
|
size = -1;
|
|
}
|
|
GST_DEBUG_OBJECT (basesrc, "setting size %" G_GUINT64_FORMAT, size);
|
|
/* only update the size when operating in bytes, subclass is supposed
|
|
* to set duration in the start method for other formats */
|
|
gst_segment_set_duration (&basesrc->segment, GST_FORMAT_BYTES, size);
|
|
} else {
|
|
size = -1;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (basesrc,
|
|
"format: %d, have size: %d, size: %" G_GUINT64_FORMAT ", duration: %"
|
|
G_GINT64_FORMAT, basesrc->segment.format, result, size,
|
|
basesrc->segment.duration);
|
|
|
|
seekable = gst_base_src_seekable (basesrc);
|
|
GST_DEBUG_OBJECT (basesrc, "is seekable: %d", seekable);
|
|
|
|
/* update for random access flag */
|
|
basesrc->random_access = seekable &&
|
|
basesrc->segment.format == GST_FORMAT_BYTES;
|
|
|
|
GST_DEBUG_OBJECT (basesrc, "is random_access: %d", basesrc->random_access);
|
|
|
|
/* run typefind if we are random_access and the typefinding is enabled. */
|
|
if (basesrc->random_access && basesrc->data.ABI.typefind && size != -1) {
|
|
GstCaps *caps;
|
|
|
|
if (!(caps = gst_type_find_helper (basesrc->srcpad, size)))
|
|
goto typefind_failed;
|
|
|
|
gst_pad_set_caps (basesrc->srcpad, caps);
|
|
gst_caps_unref (caps);
|
|
} else {
|
|
/* use class or default negotiate function */
|
|
if (!gst_base_src_negotiate (basesrc))
|
|
goto could_not_negotiate;
|
|
}
|
|
|
|
return TRUE;
|
|
|
|
/* ERROR */
|
|
could_not_start:
|
|
{
|
|
GST_DEBUG_OBJECT (basesrc, "could not start");
|
|
/* subclass is supposed to post a message. We don't have to call _stop. */
|
|
return FALSE;
|
|
}
|
|
could_not_negotiate:
|
|
{
|
|
GST_DEBUG_OBJECT (basesrc, "could not negotiate, stopping");
|
|
GST_ELEMENT_ERROR (basesrc, STREAM, FORMAT,
|
|
("Could not negotiate format"), ("Check your filtered caps, if any"));
|
|
/* we must call stop */
|
|
gst_base_src_stop (basesrc);
|
|
return FALSE;
|
|
}
|
|
typefind_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (basesrc, "could not typefind, stopping");
|
|
GST_ELEMENT_ERROR (basesrc, STREAM, TYPE_NOT_FOUND, (NULL), (NULL));
|
|
/* we must call stop */
|
|
gst_base_src_stop (basesrc);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_stop (GstBaseSrc * basesrc)
|
|
{
|
|
GstBaseSrcClass *bclass;
|
|
gboolean result = TRUE;
|
|
|
|
if (!GST_OBJECT_FLAG_IS_SET (basesrc, GST_BASE_SRC_STARTED))
|
|
return TRUE;
|
|
|
|
GST_DEBUG_OBJECT (basesrc, "stopping source");
|
|
|
|
bclass = GST_BASE_SRC_GET_CLASS (basesrc);
|
|
if (bclass->stop)
|
|
result = bclass->stop (basesrc);
|
|
|
|
if (result)
|
|
GST_OBJECT_FLAG_UNSET (basesrc, GST_BASE_SRC_STARTED);
|
|
|
|
return result;
|
|
}
|
|
|
|
/* start or stop flushing dataprocessing
|
|
*/
|
|
static gboolean
|
|
gst_base_src_set_flushing (GstBaseSrc * basesrc,
|
|
gboolean flushing, gboolean live_play, gboolean unlock, gboolean * playing)
|
|
{
|
|
GstBaseSrcClass *bclass;
|
|
|
|
bclass = GST_BASE_SRC_GET_CLASS (basesrc);
|
|
|
|
if (flushing && unlock) {
|
|
/* unlock any subclasses, we need to do this before grabbing the
|
|
* LIVE_LOCK since we hold this lock before going into ::create. We pass an
|
|
* unlock to the params because of backwards compat (see seek handler)*/
|
|
if (bclass->unlock)
|
|
bclass->unlock (basesrc);
|
|
}
|
|
|
|
/* the live lock is released when we are blocked, waiting for playing or
|
|
* when we sync to the clock. */
|
|
GST_LIVE_LOCK (basesrc);
|
|
if (playing)
|
|
*playing = basesrc->live_running;
|
|
basesrc->priv->flushing = flushing;
|
|
if (flushing) {
|
|
/* if we are locked in the live lock, signal it to make it flush */
|
|
basesrc->live_running = TRUE;
|
|
|
|
/* clear pending EOS if any */
|
|
g_atomic_int_set (&basesrc->priv->pending_eos, FALSE);
|
|
|
|
/* step 1, now that we have the LIVE lock, clear our unlock request */
|
|
if (bclass->unlock_stop)
|
|
bclass->unlock_stop (basesrc);
|
|
|
|
/* step 2, unblock clock sync (if any) or any other blocking thing */
|
|
if (basesrc->clock_id)
|
|
gst_clock_id_unschedule (basesrc->clock_id);
|
|
} else {
|
|
/* signal the live source that it can start playing */
|
|
basesrc->live_running = live_play;
|
|
}
|
|
GST_LIVE_SIGNAL (basesrc);
|
|
GST_LIVE_UNLOCK (basesrc);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/* the purpose of this function is to make sure that a live source blocks in the
|
|
* LIVE lock or leaves the LIVE lock and continues playing. */
|
|
static gboolean
|
|
gst_base_src_set_playing (GstBaseSrc * basesrc, gboolean live_play)
|
|
{
|
|
GstBaseSrcClass *bclass;
|
|
|
|
bclass = GST_BASE_SRC_GET_CLASS (basesrc);
|
|
|
|
/* unlock subclasses locked in ::create, we only do this when we stop playing. */
|
|
if (!live_play) {
|
|
GST_DEBUG_OBJECT (basesrc, "unlock");
|
|
if (bclass->unlock)
|
|
bclass->unlock (basesrc);
|
|
}
|
|
|
|
/* we are now able to grab the LIVE lock, when we get it, we can be
|
|
* waiting for PLAYING while blocked in the LIVE cond or we can be waiting
|
|
* for the clock. */
|
|
GST_LIVE_LOCK (basesrc);
|
|
GST_DEBUG_OBJECT (basesrc, "unschedule clock");
|
|
|
|
/* unblock clock sync (if any) */
|
|
if (basesrc->clock_id)
|
|
gst_clock_id_unschedule (basesrc->clock_id);
|
|
|
|
/* configure what to do when we get to the LIVE lock. */
|
|
GST_DEBUG_OBJECT (basesrc, "live running %d", live_play);
|
|
basesrc->live_running = live_play;
|
|
|
|
if (live_play) {
|
|
gboolean start;
|
|
|
|
/* clear our unlock request when going to PLAYING */
|
|
GST_DEBUG_OBJECT (basesrc, "unlock stop");
|
|
if (bclass->unlock_stop)
|
|
bclass->unlock_stop (basesrc);
|
|
|
|
/* for live sources we restart the timestamp correction */
|
|
basesrc->priv->latency = -1;
|
|
/* have to restart the task in case it stopped because of the unlock when
|
|
* we went to PAUSED. Only do this if we operating in push mode. */
|
|
GST_OBJECT_LOCK (basesrc->srcpad);
|
|
start = (GST_PAD_ACTIVATE_MODE (basesrc->srcpad) == GST_ACTIVATE_PUSH);
|
|
GST_OBJECT_UNLOCK (basesrc->srcpad);
|
|
if (start)
|
|
gst_pad_start_task (basesrc->srcpad, (GstTaskFunction) gst_base_src_loop,
|
|
basesrc->srcpad);
|
|
GST_DEBUG_OBJECT (basesrc, "signal");
|
|
GST_LIVE_SIGNAL (basesrc);
|
|
}
|
|
GST_LIVE_UNLOCK (basesrc);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_activate_push (GstPad * pad, gboolean active)
|
|
{
|
|
GstBaseSrc *basesrc;
|
|
GstEvent *event;
|
|
|
|
basesrc = GST_BASE_SRC (GST_OBJECT_PARENT (pad));
|
|
|
|
/* prepare subclass first */
|
|
if (active) {
|
|
GST_DEBUG_OBJECT (basesrc, "Activating in push mode");
|
|
|
|
if (G_UNLIKELY (!basesrc->can_activate_push))
|
|
goto no_push_activation;
|
|
|
|
if (G_UNLIKELY (!gst_base_src_start (basesrc)))
|
|
goto error_start;
|
|
|
|
basesrc->priv->last_sent_eos = FALSE;
|
|
basesrc->priv->discont = TRUE;
|
|
gst_base_src_set_flushing (basesrc, FALSE, FALSE, FALSE, NULL);
|
|
|
|
/* do initial seek, which will start the task */
|
|
GST_OBJECT_LOCK (basesrc);
|
|
event = basesrc->data.ABI.pending_seek;
|
|
basesrc->data.ABI.pending_seek = NULL;
|
|
GST_OBJECT_UNLOCK (basesrc);
|
|
|
|
/* no need to unlock anything, the task is certainly
|
|
* not running here. The perform seek code will start the task when
|
|
* finished. */
|
|
if (G_UNLIKELY (!gst_base_src_perform_seek (basesrc, event, FALSE)))
|
|
goto seek_failed;
|
|
|
|
if (event)
|
|
gst_event_unref (event);
|
|
} else {
|
|
GST_DEBUG_OBJECT (basesrc, "Deactivating in push mode");
|
|
/* flush all */
|
|
gst_base_src_set_flushing (basesrc, TRUE, FALSE, TRUE, NULL);
|
|
/* stop the task */
|
|
gst_pad_stop_task (pad);
|
|
/* now we can stop the source */
|
|
if (G_UNLIKELY (!gst_base_src_stop (basesrc)))
|
|
goto error_stop;
|
|
}
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
no_push_activation:
|
|
{
|
|
GST_WARNING_OBJECT (basesrc, "Subclass disabled push-mode activation");
|
|
return FALSE;
|
|
}
|
|
error_start:
|
|
{
|
|
GST_WARNING_OBJECT (basesrc, "Failed to start in push mode");
|
|
return FALSE;
|
|
}
|
|
seek_failed:
|
|
{
|
|
GST_ERROR_OBJECT (basesrc, "Failed to perform initial seek");
|
|
gst_base_src_stop (basesrc);
|
|
if (event)
|
|
gst_event_unref (event);
|
|
return FALSE;
|
|
}
|
|
error_stop:
|
|
{
|
|
GST_DEBUG_OBJECT (basesrc, "Failed to stop in push mode");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_activate_pull (GstPad * pad, gboolean active)
|
|
{
|
|
GstBaseSrc *basesrc;
|
|
|
|
basesrc = GST_BASE_SRC (GST_OBJECT_PARENT (pad));
|
|
|
|
/* prepare subclass first */
|
|
if (active) {
|
|
GST_DEBUG_OBJECT (basesrc, "Activating in pull mode");
|
|
if (G_UNLIKELY (!gst_base_src_start (basesrc)))
|
|
goto error_start;
|
|
|
|
/* if not random_access, we cannot operate in pull mode for now */
|
|
if (G_UNLIKELY (!gst_base_src_check_get_range (basesrc)))
|
|
goto no_get_range;
|
|
|
|
/* stop flushing now but for live sources, still block in the LIVE lock when
|
|
* we are not yet PLAYING */
|
|
gst_base_src_set_flushing (basesrc, FALSE, FALSE, FALSE, NULL);
|
|
} else {
|
|
GST_DEBUG_OBJECT (basesrc, "Deactivating in pull mode");
|
|
/* flush all, there is no task to stop */
|
|
gst_base_src_set_flushing (basesrc, TRUE, FALSE, TRUE, NULL);
|
|
|
|
/* don't send EOS when going from PAUSED => READY when in pull mode */
|
|
basesrc->priv->last_sent_eos = TRUE;
|
|
|
|
if (G_UNLIKELY (!gst_base_src_stop (basesrc)))
|
|
goto error_stop;
|
|
}
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
error_start:
|
|
{
|
|
GST_ERROR_OBJECT (basesrc, "Failed to start in pull mode");
|
|
return FALSE;
|
|
}
|
|
no_get_range:
|
|
{
|
|
GST_ERROR_OBJECT (basesrc, "Cannot operate in pull mode, stopping");
|
|
gst_base_src_stop (basesrc);
|
|
return FALSE;
|
|
}
|
|
error_stop:
|
|
{
|
|
GST_ERROR_OBJECT (basesrc, "Failed to stop in pull mode");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_base_src_change_state (GstElement * element, GstStateChange transition)
|
|
{
|
|
GstBaseSrc *basesrc;
|
|
GstStateChangeReturn result;
|
|
gboolean no_preroll = FALSE;
|
|
|
|
basesrc = GST_BASE_SRC (element);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_NULL_TO_READY:
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
no_preroll = gst_base_src_is_live (basesrc);
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
|
GST_DEBUG_OBJECT (basesrc, "PAUSED->PLAYING");
|
|
if (gst_base_src_is_live (basesrc)) {
|
|
/* now we can start playback */
|
|
gst_base_src_set_playing (basesrc, TRUE);
|
|
}
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
if ((result =
|
|
GST_ELEMENT_CLASS (parent_class)->change_state (element,
|
|
transition)) == GST_STATE_CHANGE_FAILURE)
|
|
goto failure;
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
|
GST_DEBUG_OBJECT (basesrc, "PLAYING->PAUSED");
|
|
if (gst_base_src_is_live (basesrc)) {
|
|
/* make sure we block in the live lock in PAUSED */
|
|
gst_base_src_set_playing (basesrc, FALSE);
|
|
no_preroll = TRUE;
|
|
}
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
{
|
|
GstEvent **event_p, *event;
|
|
|
|
/* we don't need to unblock anything here, the pad deactivation code
|
|
* already did this */
|
|
|
|
/* FIXME, deprecate this behaviour, it is very dangerous.
|
|
* the prefered way of sending EOS downstream is by sending
|
|
* the EOS event to the element */
|
|
if (!basesrc->priv->last_sent_eos) {
|
|
GST_DEBUG_OBJECT (basesrc, "Sending EOS event");
|
|
event = gst_event_new_eos ();
|
|
gst_event_set_seqnum (event, basesrc->priv->seqnum);
|
|
gst_pad_push_event (basesrc->srcpad, event);
|
|
basesrc->priv->last_sent_eos = TRUE;
|
|
}
|
|
g_atomic_int_set (&basesrc->priv->pending_eos, FALSE);
|
|
event_p = &basesrc->data.ABI.pending_seek;
|
|
gst_event_replace (event_p, NULL);
|
|
event_p = &basesrc->priv->close_segment;
|
|
gst_event_replace (event_p, NULL);
|
|
event_p = &basesrc->priv->start_segment;
|
|
gst_event_replace (event_p, NULL);
|
|
break;
|
|
}
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
if (no_preroll && result == GST_STATE_CHANGE_SUCCESS)
|
|
result = GST_STATE_CHANGE_NO_PREROLL;
|
|
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
failure:
|
|
{
|
|
GST_DEBUG_OBJECT (basesrc, "parent failed state change");
|
|
return result;
|
|
}
|
|
}
|