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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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f490508490
Set the pad flushing and stop the pad task when the initial seek fails during activation. Avoids racy calls into the _create() function when BaseSrc::stop() has already run. Fixes: #603059 Also, fix some misspelled comments.
3018 lines
91 KiB
C
3018 lines
91 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>#GstBaseSrcClass.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 time
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* by overriding #GstBaseSrcClass.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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* #GstBaseSrcClass.is_seekable() returns %TRUE.
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* </para></listitem>
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* <listitem><para>
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* #GstBaseSrc:Class.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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* #GstBaseSrcClass.do_seek() is implemented, performs the seek and returns
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* %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, the
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* offset and length in the #GstBaseSrcClass.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 #GstBaseSrcClass.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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* #GstBaseSrcClass.create() method will not be called in PAUSED but only in
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* PLAYING. To signal the pipeline that the element will not produce data, the
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* return value from the READY to PAUSED state will be
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* #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
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* #GstBaseSrcClass.create() function was interrupted by a state change to
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* PAUSED.
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*
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* The #GstBaseSrcClass.get_times() method can be used to implement pseudo-live
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* sources. It only makes sense to implement the #GstBaseSrcClass.get_times()
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* function if the source is a live source. The #GstBaseSrcClass.get_times()
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* function should return timestamps starting from 0, as if it were a non-live
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* source. The base class will make sure that the timestamps are transformed
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* into the current running_time. The base source will then wait for the
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* calculated running_time before pushing 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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* |[
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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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* ]|
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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_base_src_finalize;
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gobject_class->set_property = gst_base_src_set_property;
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gobject_class->get_property = 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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/* Registering debug symbols for function pointers */
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GST_DEBUG_REGISTER_FUNCPTR (gst_base_src_activate_push);
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GST_DEBUG_REGISTER_FUNCPTR (gst_base_src_activate_pull);
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GST_DEBUG_REGISTER_FUNCPTR (gst_base_src_event_handler);
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GST_DEBUG_REGISTER_FUNCPTR (gst_base_src_query);
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GST_DEBUG_REGISTER_FUNCPTR (gst_base_src_pad_get_range);
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GST_DEBUG_REGISTER_FUNCPTR (gst_base_src_pad_check_get_range);
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GST_DEBUG_REGISTER_FUNCPTR (gst_base_src_getcaps);
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GST_DEBUG_REGISTER_FUNCPTR (gst_base_src_setcaps);
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GST_DEBUG_REGISTER_FUNCPTR (gst_base_src_fixate);
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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");
|
|
|
|
GST_DEBUG_OBJECT (basesrc, "setting functions on src pad");
|
|
gst_pad_set_activatepush_function (pad, gst_base_src_activate_push);
|
|
gst_pad_set_activatepull_function (pad, gst_base_src_activate_pull);
|
|
gst_pad_set_event_function (pad, gst_base_src_event_handler);
|
|
gst_pad_set_query_function (pad, gst_base_src_query);
|
|
gst_pad_set_checkgetrange_function (pad, gst_base_src_pad_check_get_range);
|
|
gst_pad_set_getrange_function (pad, gst_base_src_pad_get_range);
|
|
gst_pad_set_getcaps_function (pad, gst_base_src_getcaps);
|
|
gst_pad_set_setcaps_function (pad, gst_base_src_setcaps);
|
|
gst_pad_set_fixatecaps_function (pad, 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;
|
|
}
|
|
|
|
/**
|
|
* gst_base_src_new_seamless_segment:
|
|
* @src: The source
|
|
* @start: The new start value for the segment
|
|
* @stop: Stop value for the new segment
|
|
* @position: The position value for the new segent
|
|
*
|
|
* Prepare a new seamless segment for emission downstream. This function must
|
|
* only be called by derived sub-classes, and only from the create() function,
|
|
* as the stream-lock needs to be held.
|
|
*
|
|
* The format for the new segment will be the current format of the source, as
|
|
* configured with gst_base_src_set_format()
|
|
*
|
|
* Returns: %TRUE if preparation of the seamless segment succeeded.
|
|
*
|
|
* Since: 0.10.26
|
|
*/
|
|
gboolean
|
|
gst_base_src_new_seamless_segment (GstBaseSrc * src, gint64 start, gint64 stop,
|
|
gint64 position)
|
|
{
|
|
gboolean res = TRUE;
|
|
|
|
GST_DEBUG_OBJECT (src,
|
|
"Starting new seamless segment. Start %" GST_TIME_FORMAT " stop %"
|
|
GST_TIME_FORMAT " position %" GST_TIME_FORMAT, GST_TIME_ARGS (start),
|
|
GST_TIME_ARGS (stop), GST_TIME_ARGS (position));
|
|
|
|
if (src->data.ABI.running) {
|
|
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_segment_set_newsegment_full (&src->segment, FALSE, src->segment.rate,
|
|
src->segment.applied_rate, src->segment.format, start, stop, position);
|
|
|
|
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);
|
|
}
|
|
|
|
src->priv->discont = TRUE;
|
|
src->data.ABI.running = TRUE;
|
|
|
|
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 the seek was successful, 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 explicitly 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);
|
|
|
|
again:
|
|
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. In any case, get rid of
|
|
* the buffer. */
|
|
gst_buffer_unref (*buf);
|
|
*buf = NULL;
|
|
if (!src->live_running) {
|
|
/* We return WRONG_STATE when we are not running to stop the dataflow also
|
|
* get rid of the produced buffer. */
|
|
GST_DEBUG_OBJECT (src,
|
|
"clock was unscheduled (%d), returning WRONG_STATE", status);
|
|
ret = GST_FLOW_WRONG_STATE;
|
|
} else {
|
|
/* If we are running when this happens, we quickly switched between
|
|
* pause and playing. We try to produce a new buffer */
|
|
GST_DEBUG_OBJECT (src,
|
|
"clock was unscheduled (%d), but we are running", status);
|
|
goto again;
|
|
}
|
|
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_reffed (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_reffed (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;
|
|
|
|
result = gst_pad_set_caps (basesrc->srcpad, caps);
|
|
gst_caps_unref (caps);
|
|
} else {
|
|
/* use class or default negotiate function */
|
|
if (!(result = gst_base_src_negotiate (basesrc)))
|
|
goto could_not_negotiate;
|
|
}
|
|
|
|
return result;
|
|
|
|
/* 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");
|
|
/* flush all */
|
|
gst_base_src_set_flushing (basesrc, TRUE, FALSE, TRUE, NULL);
|
|
/* stop the task */
|
|
gst_pad_stop_task (pad);
|
|
/* Stop the basesrc */
|
|
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;
|
|
}
|
|
}
|