mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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9f42bd643e
Use a new GCond, protected with the object lock, to signal completion of the async state change. We can't reuse the live lock because that one can be locked when the create function blocks. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686723
3687 lines
106 KiB
C
3687 lines
106 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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* 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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* #GstBaseSrcClass.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 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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*
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* gst_element_class_set_static_metadata (gstelement_class,
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* "Source name",
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* "Source",
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* "My Source element",
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* "The author <my.sink@my.email>");
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* }
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* ]|
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*
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* <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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* An application may send an EOS event to a source element to make it
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* 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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* 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 <gst/gst_private.h>
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#include <gst/glib-compat-private.h>
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#include "gstbasesrc.h"
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#include "gsttypefindhelper.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_WAIT_UNTIL(elem, end_time) g_cond_timed_wait (GST_LIVE_GET_COND (elem), GST_LIVE_GET_LOCK (elem), end_time)
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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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#define GST_ASYNC_GET_COND(elem) (&GST_BASE_SRC_CAST(elem)->priv->async_cond)
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#define GST_ASYNC_WAIT(elem) g_cond_wait (GST_ASYNC_GET_COND (elem), GST_OBJECT_GET_LOCK (elem))
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#define GST_ASYNC_SIGNAL(elem) g_cond_signal (GST_ASYNC_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 discont;
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gboolean flushing;
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GstFlowReturn start_result;
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gboolean async;
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/* if a stream-start event should be sent */
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gboolean stream_start_pending;
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/* if segment should be sent */
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gboolean segment_pending;
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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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volatile gint dynamic_size;
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/* stream sequence number */
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guint32 seqnum;
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/* pending events (TAG, CUSTOM_BOTH, CUSTOM_DOWNSTREAM) to be
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* pushed in the data stream */
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GList *pending_events;
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volatile gint have_events;
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/* QoS *//* with LOCK */
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gboolean qos_enabled;
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gdouble proportion;
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GstClockTime earliest_time;
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GstBufferPool *pool;
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GstAllocator *allocator;
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GstAllocationParams params;
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GCond async_cond;
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};
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static GstElementClass *parent_class = NULL;
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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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NULL,
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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_default_get_caps (GstBaseSrc * bsrc,
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GstCaps * filter);
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static GstCaps *gst_base_src_default_fixate (GstBaseSrc * src, GstCaps * caps);
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static GstCaps *gst_base_src_fixate (GstBaseSrc * src, GstCaps * caps);
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static gboolean gst_base_src_is_random_access (GstBaseSrc * src);
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static gboolean gst_base_src_activate_mode (GstPad * pad, GstObject * parent,
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GstPadMode mode, 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 (GstPad * pad, GstObject * parent,
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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 gboolean gst_base_src_query (GstPad * pad, GstObject * parent,
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GstQuery * query);
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static gboolean gst_base_src_activate_pool (GstBaseSrc * basesrc,
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gboolean active);
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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 GstFlowReturn gst_base_src_default_create (GstBaseSrc * basesrc,
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guint64 offset, guint size, GstBuffer ** buf);
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static GstFlowReturn gst_base_src_default_alloc (GstBaseSrc * basesrc,
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guint64 offset, guint size, GstBuffer ** buf);
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static gboolean gst_base_src_decide_allocation_default (GstBaseSrc * basesrc,
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GstQuery * query);
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static gboolean gst_base_src_set_flushing (GstBaseSrc * basesrc,
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gboolean flushing, gboolean live_play, 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 GstFlowReturn gst_base_src_getrange (GstPad * pad, GstObject * parent,
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guint64 offset, 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 gboolean gst_base_src_negotiate (GstBaseSrc * basesrc);
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static gboolean gst_base_src_update_length (GstBaseSrc * src, guint64 offset,
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guint * length);
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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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GST_DEBUG_CATEGORY_INIT (gst_base_src_debug, "basesrc", 0, "basesrc element");
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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_uint ("blocksize", "Block size",
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"Size in bytes to read per buffer (-1 = default)", 0, G_MAXUINT,
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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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klass->get_caps = GST_DEBUG_FUNCPTR (gst_base_src_default_get_caps);
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klass->negotiate = GST_DEBUG_FUNCPTR (gst_base_src_default_negotiate);
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klass->fixate = GST_DEBUG_FUNCPTR (gst_base_src_default_fixate);
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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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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->event = GST_DEBUG_FUNCPTR (gst_base_src_default_event);
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klass->create = GST_DEBUG_FUNCPTR (gst_base_src_default_create);
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klass->alloc = GST_DEBUG_FUNCPTR (gst_base_src_default_alloc);
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klass->decide_allocation =
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GST_DEBUG_FUNCPTR (gst_base_src_decide_allocation_default);
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/* Registering debug symbols for function pointers */
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GST_DEBUG_REGISTER_FUNCPTR (gst_base_src_activate_mode);
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GST_DEBUG_REGISTER_FUNCPTR (gst_base_src_event);
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GST_DEBUG_REGISTER_FUNCPTR (gst_base_src_query);
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GST_DEBUG_REGISTER_FUNCPTR (gst_base_src_getrange);
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GST_DEBUG_REGISTER_FUNCPTR (gst_base_src_fixate);
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}
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static void
|
|
gst_base_src_init (GstBaseSrc * basesrc, gpointer g_class)
|
|
{
|
|
GstPad *pad;
|
|
GstPadTemplate *pad_template;
|
|
|
|
basesrc->priv = GST_BASE_SRC_GET_PRIVATE (basesrc);
|
|
|
|
basesrc->is_live = FALSE;
|
|
g_mutex_init (&basesrc->live_lock);
|
|
g_cond_init (&basesrc->live_cond);
|
|
basesrc->num_buffers = DEFAULT_NUM_BUFFERS;
|
|
basesrc->num_buffers_left = -1;
|
|
|
|
basesrc->can_activate_push = TRUE;
|
|
|
|
pad_template =
|
|
gst_element_class_get_pad_template (GST_ELEMENT_CLASS (g_class), "src");
|
|
g_return_if_fail (pad_template != NULL);
|
|
|
|
GST_DEBUG_OBJECT (basesrc, "creating src pad");
|
|
pad = gst_pad_new_from_template (pad_template, "src");
|
|
|
|
GST_DEBUG_OBJECT (basesrc, "setting functions on src pad");
|
|
gst_pad_set_activatemode_function (pad, gst_base_src_activate_mode);
|
|
gst_pad_set_event_function (pad, gst_base_src_event);
|
|
gst_pad_set_query_function (pad, gst_base_src_query);
|
|
gst_pad_set_getrange_function (pad, gst_base_src_getrange);
|
|
|
|
/* 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->typefind = DEFAULT_TYPEFIND;
|
|
basesrc->priv->do_timestamp = DEFAULT_DO_TIMESTAMP;
|
|
g_atomic_int_set (&basesrc->priv->have_events, FALSE);
|
|
|
|
g_cond_init (&basesrc->priv->async_cond);
|
|
basesrc->priv->start_result = GST_FLOW_FLUSHING;
|
|
GST_OBJECT_FLAG_UNSET (basesrc, GST_BASE_SRC_FLAG_STARTED);
|
|
GST_OBJECT_FLAG_UNSET (basesrc, GST_BASE_SRC_FLAG_STARTING);
|
|
GST_OBJECT_FLAG_SET (basesrc, GST_ELEMENT_FLAG_SOURCE);
|
|
|
|
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_clear (&basesrc->live_lock);
|
|
g_cond_clear (&basesrc->live_cond);
|
|
g_cond_clear (&basesrc->priv->async_cond);
|
|
|
|
event_p = &basesrc->pending_seek;
|
|
gst_event_replace (event_p, NULL);
|
|
|
|
if (basesrc->priv->pending_events) {
|
|
g_list_foreach (basesrc->priv->pending_events, (GFunc) gst_event_unref,
|
|
NULL);
|
|
g_list_free (basesrc->priv->pending_events);
|
|
}
|
|
|
|
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_FLUSHING).
|
|
*
|
|
* 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);
|
|
|
|
do {
|
|
/* 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);
|
|
GST_DEBUG_OBJECT (src, "live source unlocked");
|
|
if (src->priv->flushing)
|
|
goto flushing;
|
|
} while (G_UNLIKELY (!src->live_running));
|
|
|
|
return GST_FLOW_OK;
|
|
|
|
/* ERRORS */
|
|
flushing:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "we are flushing");
|
|
return GST_FLOW_FLUSHING;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* 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 #GstBaseSrcClass.is_seekable() returns TRUE.
|
|
*
|
|
* This function must only be called in states < %GST_STATE_PAUSED.
|
|
*/
|
|
void
|
|
gst_base_src_set_format (GstBaseSrc * src, GstFormat format)
|
|
{
|
|
g_return_if_fail (GST_IS_BASE_SRC (src));
|
|
g_return_if_fail (GST_STATE (src) <= GST_STATE_READY);
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
gst_segment_init (&src->segment, format);
|
|
GST_OBJECT_UNLOCK (src);
|
|
}
|
|
|
|
/**
|
|
* gst_base_src_set_dynamic_size:
|
|
* @src: base source instance
|
|
* @dynamic: new dynamic size mode
|
|
*
|
|
* If not @dynamic, size is only updated when needed, such as when trying to
|
|
* read past current tracked size. Otherwise, size is checked for upon each
|
|
* read.
|
|
*/
|
|
void
|
|
gst_base_src_set_dynamic_size (GstBaseSrc * src, gboolean dynamic)
|
|
{
|
|
g_return_if_fail (GST_IS_BASE_SRC (src));
|
|
|
|
g_atomic_int_set (&src->priv->dynamic_size, dynamic);
|
|
}
|
|
|
|
/**
|
|
* gst_base_src_set_async:
|
|
* @src: base source instance
|
|
* @async: new async mode
|
|
*
|
|
* Configure async behaviour in @src, no state change will block. The open,
|
|
* close, start, stop, play and pause virtual methods will be executed in a
|
|
* different thread and are thus allowed to perform blocking operations. Any
|
|
* blocking operation should be unblocked with the unlock vmethod.
|
|
*/
|
|
void
|
|
gst_base_src_set_async (GstBaseSrc * src, gboolean async)
|
|
{
|
|
g_return_if_fail (GST_IS_BASE_SRC (src));
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
src->priv->async = async;
|
|
GST_OBJECT_UNLOCK (src);
|
|
}
|
|
|
|
/**
|
|
* gst_base_src_is_async:
|
|
* @src: base source instance
|
|
*
|
|
* Get the current async behaviour of @src. See also gst_base_src_set_async().
|
|
*
|
|
* Returns: %TRUE if @src is operating in async mode.
|
|
*/
|
|
gboolean
|
|
gst_base_src_is_async (GstBaseSrc * src)
|
|
{
|
|
gboolean res;
|
|
|
|
g_return_val_if_fail (GST_IS_BASE_SRC (src), FALSE);
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
res = src->priv->async;
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
return res;
|
|
}
|
|
|
|
|
|
/**
|
|
* gst_base_src_query_latency:
|
|
* @src: the source
|
|
* @live: (out) (allow-none): if the source is live
|
|
* @min_latency: (out) (allow-none): the min latency of the source
|
|
* @max_latency: (out) (allow-none): 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.
|
|
*/
|
|
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.
|
|
*/
|
|
void
|
|
gst_base_src_set_blocksize (GstBaseSrc * src, guint 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.
|
|
*/
|
|
guint
|
|
gst_base_src_get_blocksize (GstBaseSrc * src)
|
|
{
|
|
gint 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.
|
|
*/
|
|
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.
|
|
*/
|
|
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
|
|
* @time: The new time value for the start of 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.
|
|
*/
|
|
gboolean
|
|
gst_base_src_new_seamless_segment (GstBaseSrc * src, gint64 start, gint64 stop,
|
|
gint64 time)
|
|
{
|
|
gboolean res = TRUE;
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
|
|
src->segment.base = gst_segment_to_running_time (&src->segment,
|
|
src->segment.format, src->segment.position);
|
|
src->segment.position = src->segment.start = start;
|
|
src->segment.stop = stop;
|
|
src->segment.time = time;
|
|
|
|
/* Mark pending segment. Will be sent before next data */
|
|
src->priv->segment_pending = TRUE;
|
|
|
|
GST_DEBUG_OBJECT (src,
|
|
"Starting new seamless segment. Start %" GST_TIME_FORMAT " stop %"
|
|
GST_TIME_FORMAT " time %" GST_TIME_FORMAT " base %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (start), GST_TIME_ARGS (stop), GST_TIME_ARGS (time),
|
|
GST_TIME_ARGS (src->segment.base));
|
|
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
src->priv->discont = TRUE;
|
|
src->running = TRUE;
|
|
|
|
return res;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_send_stream_start (GstBaseSrc * src)
|
|
{
|
|
gboolean ret = TRUE;
|
|
|
|
if (src->priv->stream_start_pending) {
|
|
gchar *stream_id;
|
|
|
|
stream_id =
|
|
gst_pad_create_stream_id (src->srcpad, GST_ELEMENT_CAST (src), NULL);
|
|
|
|
GST_DEBUG_OBJECT (src, "Pushing STREAM_START");
|
|
ret =
|
|
gst_pad_push_event (src->srcpad,
|
|
gst_event_new_stream_start (stream_id));
|
|
src->priv->stream_start_pending = FALSE;
|
|
g_free (stream_id);
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
/**
|
|
* gst_base_src_set_caps:
|
|
* @src: a #GstBaseSrc
|
|
* @caps: a #GstCaps
|
|
*
|
|
* Set new caps on the basesrc source pad.
|
|
*
|
|
* Returns: %TRUE if the caps could be set
|
|
*/
|
|
gboolean
|
|
gst_base_src_set_caps (GstBaseSrc * src, GstCaps * caps)
|
|
{
|
|
GstBaseSrcClass *bclass;
|
|
gboolean res = TRUE;
|
|
|
|
bclass = GST_BASE_SRC_GET_CLASS (src);
|
|
|
|
gst_base_src_send_stream_start (src);
|
|
|
|
if (bclass->set_caps)
|
|
res = bclass->set_caps (src, caps);
|
|
|
|
if (res)
|
|
res = gst_pad_set_caps (src->srcpad, caps);
|
|
|
|
return res;
|
|
}
|
|
|
|
static GstCaps *
|
|
gst_base_src_default_get_caps (GstBaseSrc * bsrc, GstCaps * filter)
|
|
{
|
|
GstCaps *caps = NULL;
|
|
GstPadTemplate *pad_template;
|
|
GstBaseSrcClass *bclass;
|
|
|
|
bclass = GST_BASE_SRC_GET_CLASS (bsrc);
|
|
|
|
pad_template =
|
|
gst_element_class_get_pad_template (GST_ELEMENT_CLASS (bclass), "src");
|
|
|
|
if (pad_template != NULL) {
|
|
caps = gst_pad_template_get_caps (pad_template);
|
|
|
|
if (filter) {
|
|
GstCaps *intersection;
|
|
|
|
intersection =
|
|
gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
|
|
gst_caps_unref (caps);
|
|
caps = intersection;
|
|
}
|
|
}
|
|
return caps;
|
|
}
|
|
|
|
static GstCaps *
|
|
gst_base_src_default_fixate (GstBaseSrc * bsrc, GstCaps * caps)
|
|
{
|
|
GST_DEBUG_OBJECT (bsrc, "using default caps fixate function");
|
|
return gst_caps_fixate (caps);
|
|
}
|
|
|
|
static GstCaps *
|
|
gst_base_src_fixate (GstBaseSrc * bsrc, GstCaps * caps)
|
|
{
|
|
GstBaseSrcClass *bclass;
|
|
|
|
bclass = GST_BASE_SRC_GET_CLASS (bsrc);
|
|
|
|
if (bclass->fixate)
|
|
caps = bclass->fixate (bsrc, caps);
|
|
|
|
return caps;
|
|
}
|
|
|
|
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);
|
|
|
|
GST_DEBUG_OBJECT (src, "position query in format %s",
|
|
gst_format_get_name (format));
|
|
|
|
switch (format) {
|
|
case GST_FORMAT_PERCENT:
|
|
{
|
|
gint64 percent;
|
|
gint64 position;
|
|
gint64 duration;
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
position = src->segment.position;
|
|
duration = src->segment.duration;
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
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;
|
|
GstFormat seg_format;
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
position =
|
|
gst_segment_to_stream_time (&src->segment, src->segment.format,
|
|
src->segment.position);
|
|
seg_format = src->segment.format;
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
if (position != -1) {
|
|
/* convert to requested format */
|
|
res =
|
|
gst_pad_query_convert (src->srcpad, seg_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;
|
|
GstFormat seg_format;
|
|
guint length = 0;
|
|
|
|
/* may have to refresh duration */
|
|
if (g_atomic_int_get (&src->priv->dynamic_size))
|
|
gst_base_src_update_length (src, 0, &length);
|
|
|
|
/* this is the duration as configured by the subclass. */
|
|
GST_OBJECT_LOCK (src);
|
|
duration = src->segment.duration;
|
|
seg_format = src->segment.format;
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
GST_LOG_OBJECT (src, "duration %" G_GINT64_FORMAT ", format %s",
|
|
duration, gst_format_get_name (seg_format));
|
|
|
|
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, seg_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, seg_format;
|
|
gint64 duration;
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
duration = src->segment.duration;
|
|
seg_format = src->segment.format;
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
gst_query_parse_seeking (query, &format, NULL, NULL, NULL);
|
|
if (format == seg_format) {
|
|
gst_query_set_seeking (query, seg_format,
|
|
gst_base_src_seekable (src), 0, 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;
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
/* 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);
|
|
GST_OBJECT_UNLOCK (src);
|
|
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 useful */
|
|
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, seg_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));
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
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;
|
|
}
|
|
seg_format = src->segment.format;
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
/* 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, seg_format,
|
|
stop, format, &stop);
|
|
} else {
|
|
res = TRUE;
|
|
}
|
|
if (res && format != GST_FORMAT_PERCENT && start != -1)
|
|
res = gst_pad_query_convert (src->srcpad, seg_format,
|
|
start, format, &start);
|
|
|
|
gst_query_set_buffering_range (query, format, start, stop, estimated);
|
|
break;
|
|
}
|
|
case GST_QUERY_SCHEDULING:
|
|
{
|
|
gboolean random_access;
|
|
|
|
random_access = gst_base_src_is_random_access (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. */
|
|
gst_query_set_scheduling (query, GST_SCHEDULING_FLAG_SEEKABLE, 1, -1, 0);
|
|
if (random_access)
|
|
gst_query_add_scheduling_mode (query, GST_PAD_MODE_PULL);
|
|
gst_query_add_scheduling_mode (query, GST_PAD_MODE_PUSH);
|
|
|
|
res = TRUE;
|
|
break;
|
|
}
|
|
case GST_QUERY_CAPS:
|
|
{
|
|
GstBaseSrcClass *bclass;
|
|
GstCaps *caps, *filter;
|
|
|
|
bclass = GST_BASE_SRC_GET_CLASS (src);
|
|
if (bclass->get_caps) {
|
|
gst_query_parse_caps (query, &filter);
|
|
if ((caps = bclass->get_caps (src, filter))) {
|
|
gst_query_set_caps_result (query, caps);
|
|
gst_caps_unref (caps);
|
|
res = TRUE;
|
|
} else {
|
|
res = FALSE;
|
|
}
|
|
} else
|
|
res = FALSE;
|
|
break;
|
|
}
|
|
case GST_QUERY_URI:{
|
|
if (GST_IS_URI_HANDLER (src)) {
|
|
gchar *uri = gst_uri_handler_get_uri (GST_URI_HANDLER (src));
|
|
|
|
if (uri != NULL) {
|
|
gst_query_set_uri (query, uri);
|
|
g_free (uri);
|
|
res = TRUE;
|
|
} else {
|
|
res = FALSE;
|
|
}
|
|
} else {
|
|
res = FALSE;
|
|
}
|
|
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, GstObject * parent, GstQuery * query)
|
|
{
|
|
GstBaseSrc *src;
|
|
GstBaseSrcClass *bclass;
|
|
gboolean result = FALSE;
|
|
|
|
src = GST_BASE_SRC (parent);
|
|
bclass = GST_BASE_SRC_GET_CLASS (src);
|
|
|
|
if (bclass->query)
|
|
result = bclass->query (src, query);
|
|
|
|
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 start_type, stop_type;
|
|
gint64 start, stop;
|
|
GstSeekFlags flags;
|
|
GstFormat seek_format, dest_format;
|
|
gdouble rate;
|
|
gboolean update;
|
|
gboolean res = TRUE;
|
|
|
|
gst_event_parse_seek (event, &rate, &seek_format, &flags,
|
|
&start_type, &start, &stop_type, &stop);
|
|
dest_format = segment->format;
|
|
|
|
if (seek_format == dest_format) {
|
|
gst_segment_do_seek (segment, rate, seek_format, flags,
|
|
start_type, start, stop_type, stop, &update);
|
|
return TRUE;
|
|
}
|
|
|
|
if (start_type != GST_SEEK_TYPE_NONE) {
|
|
/* FIXME: Handle seek_end by converting the input segment vals */
|
|
res =
|
|
gst_pad_query_convert (src->srcpad, seek_format, start, dest_format,
|
|
&start);
|
|
start_type = GST_SEEK_TYPE_SET;
|
|
}
|
|
|
|
if (res && stop_type != GST_SEEK_TYPE_NONE) {
|
|
/* FIXME: Handle seek_end by converting the input segment vals */
|
|
res =
|
|
gst_pad_query_convert (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_do_seek (segment, rate, dest_format, flags, start_type, start,
|
|
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;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_base_src_default_alloc (GstBaseSrc * src, guint64 offset,
|
|
guint size, GstBuffer ** buffer)
|
|
{
|
|
GstFlowReturn ret;
|
|
GstBaseSrcPrivate *priv = src->priv;
|
|
|
|
if (priv->pool) {
|
|
ret = gst_buffer_pool_acquire_buffer (priv->pool, buffer, NULL);
|
|
} else if (size != -1) {
|
|
*buffer = gst_buffer_new_allocate (priv->allocator, size, &priv->params);
|
|
if (G_UNLIKELY (*buffer == NULL))
|
|
goto alloc_failed;
|
|
|
|
ret = GST_FLOW_OK;
|
|
} else {
|
|
GST_WARNING_OBJECT (src, "Not trying to alloc %u bytes. Blocksize not set?",
|
|
size);
|
|
goto alloc_failed;
|
|
}
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
alloc_failed:
|
|
{
|
|
GST_ERROR_OBJECT (src, "Failed to allocate %u bytes", size);
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_base_src_default_create (GstBaseSrc * src, guint64 offset,
|
|
guint size, GstBuffer ** buffer)
|
|
{
|
|
GstBaseSrcClass *bclass;
|
|
GstFlowReturn ret;
|
|
GstBuffer *res_buf;
|
|
|
|
bclass = GST_BASE_SRC_GET_CLASS (src);
|
|
|
|
if (G_UNLIKELY (!bclass->alloc))
|
|
goto no_function;
|
|
if (G_UNLIKELY (!bclass->fill))
|
|
goto no_function;
|
|
|
|
if (*buffer == NULL) {
|
|
/* downstream did not provide us with a buffer to fill, allocate one
|
|
* ourselves */
|
|
ret = bclass->alloc (src, offset, size, &res_buf);
|
|
if (G_UNLIKELY (ret != GST_FLOW_OK))
|
|
goto alloc_failed;
|
|
} else {
|
|
res_buf = *buffer;
|
|
}
|
|
|
|
if (G_LIKELY (size > 0)) {
|
|
/* only call fill when there is a size */
|
|
ret = bclass->fill (src, offset, size, res_buf);
|
|
if (G_UNLIKELY (ret != GST_FLOW_OK))
|
|
goto not_ok;
|
|
}
|
|
|
|
*buffer = res_buf;
|
|
|
|
return GST_FLOW_OK;
|
|
|
|
/* ERRORS */
|
|
no_function:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "no fill or alloc function");
|
|
return GST_FLOW_NOT_SUPPORTED;
|
|
}
|
|
alloc_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "Failed to allocate buffer of %u bytes", size);
|
|
return ret;
|
|
}
|
|
not_ok:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "fill returned %d (%s)", ret,
|
|
gst_flow_get_name (ret));
|
|
if (*buffer == NULL)
|
|
gst_buffer_unref (res_buf);
|
|
return ret;
|
|
}
|
|
}
|
|
|
|
/* 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 FLUSHING 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.
|
|
*/
|
|
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 start_type, stop_type;
|
|
gint64 start, 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: %" GST_PTR_FORMAT, event);
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
dest_format = src->segment.format;
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
if (event) {
|
|
gst_event_parse_seek (event, &rate, &seek_format, &flags,
|
|
&start_type, &start, &stop_type, &stop);
|
|
|
|
relative_seek = SEEK_TYPE_IS_RELATIVE (start_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. */
|
|
if (unlock)
|
|
gst_base_src_set_flushing (src, TRUE, FALSE, &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);
|
|
}
|
|
|
|
if (unlock)
|
|
gst_base_src_set_flushing (src, FALSE, playing, 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 succeeds. */
|
|
if (!seekseg_configured) {
|
|
memcpy (&seeksegment, &src->segment, sizeof (GstSegment));
|
|
|
|
/* now configure the final seek segment */
|
|
if (event) {
|
|
if (seeksegment.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_do_seek (&seeksegment, rate, seek_format, flags,
|
|
start_type, start, stop_type, stop, &update);
|
|
}
|
|
}
|
|
/* Else, no seek event passed, so we're just (re)starting the
|
|
current segment. */
|
|
}
|
|
|
|
if (res) {
|
|
GST_DEBUG_OBJECT (src, "segment configured from %" G_GINT64_FORMAT
|
|
" to %" G_GINT64_FORMAT ", position %" G_GINT64_FORMAT,
|
|
seeksegment.start, seeksegment.stop, seeksegment.position);
|
|
|
|
/* do the seek, segment.position contains the new position. */
|
|
res = gst_base_src_do_seek (src, &seeksegment);
|
|
}
|
|
|
|
/* and prepare to continue streaming */
|
|
if (flush) {
|
|
tevent = gst_event_new_flush_stop (TRUE);
|
|
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);
|
|
}
|
|
|
|
/* 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) {
|
|
GST_OBJECT_LOCK (src);
|
|
memcpy (&src->segment, &seeksegment, sizeof (GstSegment));
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
if (seeksegment.flags & GST_SEGMENT_FLAG_SEGMENT) {
|
|
GstMessage *message;
|
|
|
|
message = gst_message_new_segment_start (GST_OBJECT (src),
|
|
seeksegment.format, seeksegment.position);
|
|
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 = seeksegment.stop) == -1)
|
|
stop = seeksegment.duration;
|
|
|
|
src->priv->segment_pending = TRUE;
|
|
}
|
|
|
|
src->priv->discont = TRUE;
|
|
src->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, NULL);
|
|
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;
|
|
}
|
|
|
|
/* 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, "handling event %p %" GST_PTR_FORMAT, event, event);
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
/* bidirectional events */
|
|
case GST_EVENT_FLUSH_START:
|
|
GST_DEBUG_OBJECT (src, "pushing flush-start event downstream");
|
|
result = gst_pad_push_event (src->srcpad, event);
|
|
event = NULL;
|
|
break;
|
|
case GST_EVENT_FLUSH_STOP:
|
|
GST_LIVE_LOCK (src->srcpad);
|
|
src->priv->segment_pending = TRUE;
|
|
/* sending random flushes downstream can break stuff,
|
|
* especially sync since all segment info will get flushed */
|
|
GST_DEBUG_OBJECT (src, "pushing flush-stop event downstream");
|
|
result = gst_pad_push_event (src->srcpad, event);
|
|
GST_LIVE_UNLOCK (src->srcpad);
|
|
event = NULL;
|
|
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 FLUSHING immediately. 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. */
|
|
gst_base_src_activate_pool (src, FALSE);
|
|
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_base_src_activate_pool (src, TRUE);
|
|
GST_LIVE_UNLOCK (src);
|
|
|
|
result = TRUE;
|
|
break;
|
|
}
|
|
case GST_EVENT_SEGMENT:
|
|
/* sending random SEGMENT downstream can break sync. */
|
|
break;
|
|
case GST_EVENT_TAG:
|
|
case GST_EVENT_CUSTOM_DOWNSTREAM:
|
|
case GST_EVENT_CUSTOM_BOTH:
|
|
/* Insert TAG, CUSTOM_DOWNSTREAM, CUSTOM_BOTH in the dataflow */
|
|
GST_OBJECT_LOCK (src);
|
|
src->priv->pending_events =
|
|
g_list_append (src->priv->pending_events, event);
|
|
g_atomic_int_set (&src->priv->have_events, TRUE);
|
|
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_MODE (src->srcpad) == GST_PAD_MODE_PULL)
|
|
goto wrong_mode;
|
|
started = GST_PAD_MODE (src->srcpad) == GST_PAD_MODE_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. */
|
|
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->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_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 void
|
|
gst_base_src_update_qos (GstBaseSrc * src,
|
|
gdouble proportion, GstClockTimeDiff diff, GstClockTime timestamp)
|
|
{
|
|
GST_CAT_DEBUG_OBJECT (GST_CAT_QOS, src,
|
|
"qos: proportion: %lf, diff %" G_GINT64_FORMAT ", timestamp %"
|
|
GST_TIME_FORMAT, proportion, diff, GST_TIME_ARGS (timestamp));
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
src->priv->proportion = proportion;
|
|
src->priv->earliest_time = timestamp + diff;
|
|
GST_OBJECT_UNLOCK (src);
|
|
}
|
|
|
|
|
|
static gboolean
|
|
gst_base_src_default_event (GstBaseSrc * src, GstEvent * event)
|
|
{
|
|
gboolean result;
|
|
|
|
GST_DEBUG_OBJECT (src, "handle event %" GST_PTR_FORMAT, event);
|
|
|
|
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, NULL);
|
|
break;
|
|
case GST_EVENT_FLUSH_STOP:
|
|
result = gst_base_src_set_flushing (src, FALSE, TRUE, NULL);
|
|
break;
|
|
case GST_EVENT_QOS:
|
|
{
|
|
gdouble proportion;
|
|
GstClockTimeDiff diff;
|
|
GstClockTime timestamp;
|
|
|
|
gst_event_parse_qos (event, NULL, &proportion, &diff, ×tamp);
|
|
gst_base_src_update_qos (src, proportion, diff, timestamp);
|
|
result = TRUE;
|
|
break;
|
|
}
|
|
case GST_EVENT_RECONFIGURE:
|
|
result = TRUE;
|
|
break;
|
|
case GST_EVENT_LATENCY:
|
|
result = TRUE;
|
|
break;
|
|
default:
|
|
result = FALSE;
|
|
break;
|
|
}
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
not_seekable:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "is not seekable");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_event (GstPad * pad, GstObject * parent, GstEvent * event)
|
|
{
|
|
GstBaseSrc *src;
|
|
GstBaseSrcClass *bclass;
|
|
gboolean result = FALSE;
|
|
|
|
src = GST_BASE_SRC (parent);
|
|
bclass = GST_BASE_SRC_GET_CLASS (src);
|
|
|
|
if (bclass->event) {
|
|
if (!(result = bclass->event (src, event)))
|
|
goto subclass_failed;
|
|
}
|
|
|
|
done:
|
|
gst_event_unref (event);
|
|
|
|
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_uint (value));
|
|
break;
|
|
case PROP_NUM_BUFFERS:
|
|
src->num_buffers = g_value_get_int (value);
|
|
break;
|
|
case PROP_TYPEFIND:
|
|
src->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_uint (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->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, pts, dts, timestamp;
|
|
gboolean do_timestamp, first, pseudo_live, is_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 */
|
|
dts = GST_BUFFER_DTS (buffer);
|
|
pts = GST_BUFFER_PTS (buffer);
|
|
|
|
if (GST_CLOCK_TIME_IS_VALID (dts))
|
|
timestamp = dts;
|
|
else
|
|
timestamp = pts;
|
|
|
|
/* grab the lock to prepare for clocking and calculate the startup
|
|
* latency. */
|
|
GST_OBJECT_LOCK (basesrc);
|
|
|
|
is_live = basesrc->is_live;
|
|
/* if we are asked to sync against the clock we are a pseudo live element */
|
|
pseudo_live = (start != -1 && 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",
|
|
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;
|
|
else
|
|
gst_object_ref (clock);
|
|
|
|
base_time = GST_ELEMENT_CAST (basesrc)->base_time;
|
|
|
|
do_timestamp = basesrc->priv->do_timestamp;
|
|
GST_OBJECT_UNLOCK (basesrc);
|
|
|
|
/* 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 PTS: %" GST_TIME_FORMAT ", DTS %" GST_TIME_FORMAT
|
|
", running_time %" GST_TIME_FORMAT, GST_TIME_ARGS (pts),
|
|
GST_TIME_ARGS (dts), 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 (dts)) {
|
|
if (do_timestamp) {
|
|
dts = running_time;
|
|
} else {
|
|
dts = 0;
|
|
}
|
|
GST_BUFFER_DTS (buffer) = dts;
|
|
|
|
GST_LOG_OBJECT (basesrc, "created DTS %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (dts));
|
|
}
|
|
} else {
|
|
/* not the first buffer, the timestamp is the diff between the clock and
|
|
* base_time */
|
|
if (do_timestamp && !GST_CLOCK_TIME_IS_VALID (dts)) {
|
|
now = gst_clock_get_time (clock);
|
|
|
|
dts = now - base_time;
|
|
GST_BUFFER_DTS (buffer) = dts;
|
|
|
|
GST_LOG_OBJECT (basesrc, "created DTS %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (dts));
|
|
}
|
|
}
|
|
if (!GST_CLOCK_TIME_IS_VALID (pts)) {
|
|
if (!GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DELTA_UNIT))
|
|
pts = dts;
|
|
|
|
GST_BUFFER_PTS (buffer) = dts;
|
|
|
|
GST_LOG_OBJECT (basesrc, "created PTS %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (pts));
|
|
}
|
|
|
|
/* if we don't have a buffer timestamp, we don't sync */
|
|
if (!GST_CLOCK_TIME_IS_VALID (start))
|
|
goto no_sync;
|
|
|
|
if (is_live) {
|
|
/* for pseudo live sources, add our ts_offset to the timestamp */
|
|
if (GST_CLOCK_TIME_IS_VALID (pts))
|
|
GST_BUFFER_PTS (buffer) += basesrc->priv->ts_offset;
|
|
if (GST_CLOCK_TIME_IS_VALID (dts))
|
|
GST_BUFFER_DTS (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));
|
|
|
|
result = gst_base_src_wait (basesrc, clock, start + base_time);
|
|
|
|
gst_object_unref (clock);
|
|
|
|
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_unref (clock);
|
|
return GST_CLOCK_OK;
|
|
}
|
|
}
|
|
|
|
/* Called with STREAM_LOCK and LIVE_LOCK */
|
|
static gboolean
|
|
gst_base_src_update_length (GstBaseSrc * src, guint64 offset, guint * length)
|
|
{
|
|
guint64 size, maxsize;
|
|
GstBaseSrcClass *bclass;
|
|
GstFormat format;
|
|
gint64 stop;
|
|
gboolean dynamic;
|
|
|
|
bclass = GST_BASE_SRC_GET_CLASS (src);
|
|
|
|
format = src->segment.format;
|
|
stop = src->segment.stop;
|
|
/* get total file size */
|
|
size = src->segment.duration;
|
|
|
|
/* only operate if we are working with bytes */
|
|
if (format != GST_FORMAT_BYTES)
|
|
return TRUE;
|
|
|
|
/* the max amount of bytes to read is the total size or
|
|
* up to the segment.stop if present. */
|
|
if (stop != -1)
|
|
maxsize = MIN (size, 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, stop, maxsize);
|
|
|
|
dynamic = g_atomic_int_get (&src->priv->dynamic_size);
|
|
GST_DEBUG_OBJECT (src, "dynamic size: %d", dynamic);
|
|
|
|
/* 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 || dynamic)) {
|
|
/* see if length of the file changed */
|
|
if (bclass->get_size)
|
|
if (!bclass->get_size (src, &size))
|
|
size = -1;
|
|
|
|
/* make sure we don't exceed the configured segment stop
|
|
* if it was set */
|
|
if (stop != -1)
|
|
maxsize = MIN (size, 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 duration.
|
|
* segment is in bytes, we checked that above. */
|
|
GST_OBJECT_LOCK (src);
|
|
src->segment.duration = size;
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
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;
|
|
GstBuffer *res_buf;
|
|
GstBuffer *in_buf;
|
|
|
|
bclass = GST_BASE_SRC_GET_CLASS (src);
|
|
|
|
again:
|
|
if (src->is_live) {
|
|
if (G_UNLIKELY (!src->live_running)) {
|
|
ret = gst_base_src_wait_playing (src);
|
|
if (ret != GST_FLOW_OK)
|
|
goto stopped;
|
|
}
|
|
}
|
|
|
|
if (G_UNLIKELY (!GST_BASE_SRC_IS_STARTED (src)
|
|
&& !GST_BASE_SRC_IS_STARTING (src)))
|
|
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;
|
|
|
|
/* track position */
|
|
GST_OBJECT_LOCK (src);
|
|
if (src->segment.format == GST_FORMAT_BYTES)
|
|
src->segment.position = offset;
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
/* 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);
|
|
|
|
res_buf = in_buf = *buf;
|
|
|
|
ret = bclass->create (src, offset, length, &res_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) {
|
|
if (*buf == NULL)
|
|
gst_buffer_unref (res_buf);
|
|
}
|
|
goto eos;
|
|
}
|
|
|
|
if (G_UNLIKELY (ret != GST_FLOW_OK))
|
|
goto not_ok;
|
|
|
|
/* fallback in case the create function didn't fill a provided buffer */
|
|
if (in_buf != NULL && res_buf != in_buf) {
|
|
GstMapInfo info;
|
|
gsize copied_size;
|
|
|
|
GST_CAT_DEBUG_OBJECT (GST_CAT_PERFORMANCE, src, "create function didn't "
|
|
"fill the provided buffer, copying");
|
|
|
|
gst_buffer_map (in_buf, &info, GST_MAP_WRITE);
|
|
copied_size = gst_buffer_extract (res_buf, 0, info.data, info.size);
|
|
gst_buffer_unmap (in_buf, &info);
|
|
gst_buffer_set_size (in_buf, copied_size);
|
|
|
|
gst_buffer_copy_into (in_buf, res_buf, GST_BUFFER_COPY_METADATA, 0, -1);
|
|
|
|
gst_buffer_unref (res_buf);
|
|
res_buf = in_buf;
|
|
}
|
|
|
|
/* no timestamp set and we are at offset 0, we can timestamp with 0 */
|
|
if (offset == 0 && src->segment.time == 0
|
|
&& GST_BUFFER_DTS (res_buf) == -1 && !src->is_live) {
|
|
GST_DEBUG_OBJECT (src, "setting first timestamp to 0");
|
|
res_buf = gst_buffer_make_writable (res_buf);
|
|
GST_BUFFER_DTS (res_buf) = 0;
|
|
}
|
|
|
|
/* now sync before pushing the buffer */
|
|
status = gst_base_src_do_sync (src, res_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. */
|
|
if (*buf == NULL)
|
|
gst_buffer_unref (res_buf);
|
|
|
|
if (!src->live_running) {
|
|
/* We return FLUSHING 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 FLUSHING", status);
|
|
ret = GST_FLOW_FLUSHING;
|
|
} 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));
|
|
if (*buf == NULL)
|
|
gst_buffer_unref (res_buf);
|
|
ret = GST_FLOW_ERROR;
|
|
break;
|
|
}
|
|
if (G_LIKELY (ret == GST_FLOW_OK))
|
|
*buf = res_buf;
|
|
|
|
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_FLUSHING;
|
|
}
|
|
no_function:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "no create function");
|
|
return GST_FLOW_NOT_SUPPORTED;
|
|
}
|
|
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_EOS;
|
|
}
|
|
reached_num_buffers:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "sent all buffers");
|
|
return GST_FLOW_EOS;
|
|
}
|
|
flushing:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "we are flushing");
|
|
if (*buf == NULL)
|
|
gst_buffer_unref (res_buf);
|
|
return GST_FLOW_FLUSHING;
|
|
}
|
|
eos:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "we are EOS");
|
|
return GST_FLOW_EOS;
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_base_src_getrange (GstPad * pad, GstObject * parent, guint64 offset,
|
|
guint length, GstBuffer ** buf)
|
|
{
|
|
GstBaseSrc *src;
|
|
GstFlowReturn res;
|
|
|
|
src = GST_BASE_SRC_CAST (parent);
|
|
|
|
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);
|
|
|
|
return res;
|
|
|
|
/* ERRORS */
|
|
flushing:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "we are flushing");
|
|
res = GST_FLOW_FLUSHING;
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_is_random_access (GstBaseSrc * src)
|
|
{
|
|
/* we need to start the basesrc to check random access */
|
|
if (!GST_BASE_SRC_IS_STARTED (src)) {
|
|
GST_LOG_OBJECT (src, "doing start/stop to check get_range support");
|
|
if (G_LIKELY (gst_base_src_start (src))) {
|
|
if (gst_base_src_start_wait (src) != GST_FLOW_OK)
|
|
goto start_failed;
|
|
gst_base_src_stop (src);
|
|
}
|
|
}
|
|
|
|
return src->random_access;
|
|
|
|
/* ERRORS */
|
|
start_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "failed to start");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_base_src_loop (GstPad * pad)
|
|
{
|
|
GstBaseSrc *src;
|
|
GstBuffer *buf = NULL;
|
|
GstFlowReturn ret;
|
|
gint64 position;
|
|
gboolean eos;
|
|
guint blocksize;
|
|
GList *pending_events = NULL, *tmp;
|
|
|
|
eos = FALSE;
|
|
|
|
src = GST_BASE_SRC (GST_OBJECT_PARENT (pad));
|
|
|
|
gst_base_src_send_stream_start (src);
|
|
|
|
/* check if we need to renegotiate */
|
|
if (gst_pad_check_reconfigure (pad)) {
|
|
if (!gst_base_src_negotiate (src))
|
|
goto not_negotiated;
|
|
}
|
|
|
|
GST_LIVE_LOCK (src);
|
|
|
|
if (G_UNLIKELY (src->priv->flushing))
|
|
goto flushing;
|
|
|
|
blocksize = src->blocksize;
|
|
|
|
/* if we operate in bytes, we can calculate an offset */
|
|
if (src->segment.format == GST_FORMAT_BYTES) {
|
|
position = src->segment.position;
|
|
/* 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 %u",
|
|
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->segment_pending)) {
|
|
gst_pad_push_event (pad, gst_event_new_segment (&src->segment));
|
|
src->priv->segment_pending = FALSE;
|
|
}
|
|
|
|
if (g_atomic_int_get (&src->priv->have_events)) {
|
|
GST_OBJECT_LOCK (src);
|
|
/* take the events */
|
|
pending_events = src->priv->pending_events;
|
|
src->priv->pending_events = NULL;
|
|
g_atomic_int_set (&src->priv->have_events, FALSE);
|
|
GST_OBJECT_UNLOCK (src);
|
|
}
|
|
|
|
/* Push out pending events if any */
|
|
if (G_UNLIKELY (pending_events != NULL)) {
|
|
for (tmp = pending_events; tmp; tmp = g_list_next (tmp)) {
|
|
GstEvent *ev = (GstEvent *) tmp->data;
|
|
gst_pad_push_event (pad, ev);
|
|
}
|
|
g_list_free (pending_events);
|
|
}
|
|
|
|
/* figure out the new position */
|
|
switch (src->segment.format) {
|
|
case GST_FORMAT_BYTES:
|
|
{
|
|
guint bufsize = gst_buffer_get_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.position;
|
|
|
|
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_OBJECT_LOCK (src);
|
|
src->segment.position = position;
|
|
GST_OBJECT_UNLOCK (src);
|
|
}
|
|
|
|
if (G_UNLIKELY (src->priv->discont)) {
|
|
GST_INFO_OBJECT (src, "marking pending DISCONT");
|
|
buf = gst_buffer_make_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)) {
|
|
if (ret == GST_FLOW_NOT_NEGOTIATED) {
|
|
goto not_negotiated;
|
|
}
|
|
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_EOS;
|
|
goto pause;
|
|
}
|
|
|
|
done:
|
|
return;
|
|
|
|
/* special cases */
|
|
not_negotiated:
|
|
{
|
|
if (gst_pad_needs_reconfigure (pad)) {
|
|
GST_DEBUG_OBJECT (src, "Retrying to renegotiate");
|
|
return;
|
|
}
|
|
GST_DEBUG_OBJECT (src, "Failed to renegotiate");
|
|
ret = GST_FLOW_NOT_NEGOTIATED;
|
|
goto pause;
|
|
}
|
|
flushing:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "we are flushing");
|
|
GST_LIVE_UNLOCK (src);
|
|
ret = GST_FLOW_FLUSHING;
|
|
goto pause;
|
|
}
|
|
pause:
|
|
{
|
|
const gchar *reason = gst_flow_get_name (ret);
|
|
GstEvent *event;
|
|
|
|
GST_DEBUG_OBJECT (src, "pausing task, reason %s", reason);
|
|
src->running = FALSE;
|
|
gst_pad_pause_task (pad);
|
|
if (ret == GST_FLOW_EOS) {
|
|
gboolean flag_segment;
|
|
GstFormat format;
|
|
gint64 position;
|
|
|
|
/* perform EOS logic */
|
|
flag_segment = (src->segment.flags & GST_SEGMENT_FLAG_SEGMENT) != 0;
|
|
format = src->segment.format;
|
|
position = src->segment.position;
|
|
|
|
if (flag_segment) {
|
|
GstMessage *message;
|
|
|
|
message = gst_message_new_segment_done (GST_OBJECT_CAST (src),
|
|
format, position);
|
|
gst_message_set_seqnum (message, src->priv->seqnum);
|
|
gst_element_post_message (GST_ELEMENT_CAST (src), message);
|
|
event = gst_event_new_segment_done (format, position);
|
|
gst_event_set_seqnum (event, src->priv->seqnum);
|
|
gst_pad_push_event (pad, event);
|
|
} else {
|
|
event = gst_event_new_eos ();
|
|
gst_event_set_seqnum (event, src->priv->seqnum);
|
|
gst_pad_push_event (pad, event);
|
|
}
|
|
} else if (ret == GST_FLOW_NOT_LINKED || ret <= GST_FLOW_EOS) {
|
|
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.
|
|
* Also don't do this for FLUSHING because it happens
|
|
* due to flushing and posting an error message because of
|
|
* that is the wrong thing to do, e.g. when we're doing
|
|
* a flushing seek. */
|
|
GST_ELEMENT_ERROR (src, STREAM, FAILED,
|
|
(_("Internal data flow error.")),
|
|
("streaming task paused, reason %s (%d)", reason, ret));
|
|
gst_pad_push_event (pad, event);
|
|
}
|
|
goto done;
|
|
}
|
|
null_buffer:
|
|
{
|
|
GST_ELEMENT_ERROR (src, STREAM, FAILED,
|
|
(_("Internal data flow error.")), ("element returned NULL buffer"));
|
|
GST_LIVE_UNLOCK (src);
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_set_allocation (GstBaseSrc * basesrc, GstBufferPool * pool,
|
|
GstAllocator * allocator, GstAllocationParams * params)
|
|
{
|
|
GstAllocator *oldalloc;
|
|
GstBufferPool *oldpool;
|
|
GstBaseSrcPrivate *priv = basesrc->priv;
|
|
|
|
if (pool) {
|
|
GST_DEBUG_OBJECT (basesrc, "activate pool");
|
|
if (!gst_buffer_pool_set_active (pool, TRUE))
|
|
goto activate_failed;
|
|
}
|
|
|
|
GST_OBJECT_LOCK (basesrc);
|
|
oldpool = priv->pool;
|
|
priv->pool = pool;
|
|
|
|
oldalloc = priv->allocator;
|
|
priv->allocator = allocator;
|
|
|
|
if (params)
|
|
priv->params = *params;
|
|
else
|
|
gst_allocation_params_init (&priv->params);
|
|
GST_OBJECT_UNLOCK (basesrc);
|
|
|
|
if (oldpool) {
|
|
/* only deactivate if the pool is not the one we're using */
|
|
if (oldpool != pool) {
|
|
GST_DEBUG_OBJECT (basesrc, "deactivate old pool");
|
|
gst_buffer_pool_set_active (oldpool, FALSE);
|
|
}
|
|
gst_object_unref (oldpool);
|
|
}
|
|
if (oldalloc) {
|
|
gst_object_unref (oldalloc);
|
|
}
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
activate_failed:
|
|
{
|
|
GST_ERROR_OBJECT (basesrc, "failed to activate bufferpool.");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_activate_pool (GstBaseSrc * basesrc, gboolean active)
|
|
{
|
|
GstBaseSrcPrivate *priv = basesrc->priv;
|
|
GstBufferPool *pool;
|
|
gboolean res = TRUE;
|
|
|
|
GST_OBJECT_LOCK (basesrc);
|
|
if ((pool = priv->pool))
|
|
pool = gst_object_ref (pool);
|
|
GST_OBJECT_UNLOCK (basesrc);
|
|
|
|
if (pool) {
|
|
res = gst_buffer_pool_set_active (pool, active);
|
|
gst_object_unref (pool);
|
|
}
|
|
return res;
|
|
}
|
|
|
|
|
|
static gboolean
|
|
gst_base_src_decide_allocation_default (GstBaseSrc * basesrc, GstQuery * query)
|
|
{
|
|
GstCaps *outcaps;
|
|
GstBufferPool *pool;
|
|
guint size, min, max;
|
|
GstAllocator *allocator;
|
|
GstAllocationParams params;
|
|
GstStructure *config;
|
|
gboolean update_allocator;
|
|
|
|
gst_query_parse_allocation (query, &outcaps, NULL);
|
|
|
|
/* we got configuration from our peer or the decide_allocation method,
|
|
* parse them */
|
|
if (gst_query_get_n_allocation_params (query) > 0) {
|
|
/* try the allocator */
|
|
gst_query_parse_nth_allocation_param (query, 0, &allocator, ¶ms);
|
|
update_allocator = TRUE;
|
|
} else {
|
|
allocator = NULL;
|
|
gst_allocation_params_init (¶ms);
|
|
update_allocator = FALSE;
|
|
}
|
|
|
|
if (gst_query_get_n_allocation_pools (query) > 0) {
|
|
gst_query_parse_nth_allocation_pool (query, 0, &pool, &size, &min, &max);
|
|
|
|
if (pool == NULL) {
|
|
/* no pool, we can make our own */
|
|
GST_DEBUG_OBJECT (basesrc, "no pool, making new pool");
|
|
pool = gst_buffer_pool_new ();
|
|
}
|
|
} else {
|
|
pool = NULL;
|
|
size = min = max = 0;
|
|
}
|
|
|
|
/* now configure */
|
|
if (pool) {
|
|
config = gst_buffer_pool_get_config (pool);
|
|
gst_buffer_pool_config_set_params (config, outcaps, size, min, max);
|
|
gst_buffer_pool_config_set_allocator (config, allocator, ¶ms);
|
|
gst_buffer_pool_set_config (pool, config);
|
|
}
|
|
|
|
if (update_allocator)
|
|
gst_query_set_nth_allocation_param (query, 0, allocator, ¶ms);
|
|
else
|
|
gst_query_add_allocation_param (query, allocator, ¶ms);
|
|
if (allocator)
|
|
gst_object_unref (allocator);
|
|
|
|
if (pool) {
|
|
gst_query_set_nth_allocation_pool (query, 0, pool, size, min, max);
|
|
gst_object_unref (pool);
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_prepare_allocation (GstBaseSrc * basesrc, GstCaps * caps)
|
|
{
|
|
GstBaseSrcClass *bclass;
|
|
gboolean result = TRUE;
|
|
GstQuery *query;
|
|
GstBufferPool *pool = NULL;
|
|
GstAllocator *allocator = NULL;
|
|
GstAllocationParams params;
|
|
|
|
bclass = GST_BASE_SRC_GET_CLASS (basesrc);
|
|
|
|
/* make query and let peer pad answer, we don't really care if it worked or
|
|
* not, if it failed, the allocation query would contain defaults and the
|
|
* subclass would then set better values if needed */
|
|
query = gst_query_new_allocation (caps, TRUE);
|
|
if (!gst_pad_peer_query (basesrc->srcpad, query)) {
|
|
/* not a problem, just debug a little */
|
|
GST_DEBUG_OBJECT (basesrc, "peer ALLOCATION query failed");
|
|
}
|
|
|
|
g_assert (bclass->decide_allocation != NULL);
|
|
result = bclass->decide_allocation (basesrc, query);
|
|
|
|
GST_DEBUG_OBJECT (basesrc, "ALLOCATION (%d) params: %" GST_PTR_FORMAT, result,
|
|
query);
|
|
|
|
if (!result)
|
|
goto no_decide_allocation;
|
|
|
|
/* we got configuration from our peer or the decide_allocation method,
|
|
* parse them */
|
|
if (gst_query_get_n_allocation_params (query) > 0) {
|
|
gst_query_parse_nth_allocation_param (query, 0, &allocator, ¶ms);
|
|
} else {
|
|
allocator = NULL;
|
|
gst_allocation_params_init (¶ms);
|
|
}
|
|
|
|
if (gst_query_get_n_allocation_pools (query) > 0)
|
|
gst_query_parse_nth_allocation_pool (query, 0, &pool, NULL, NULL, NULL);
|
|
|
|
result = gst_base_src_set_allocation (basesrc, pool, allocator, ¶ms);
|
|
|
|
gst_query_unref (query);
|
|
|
|
return result;
|
|
|
|
/* Errors */
|
|
no_decide_allocation:
|
|
{
|
|
GST_WARNING_OBJECT (basesrc, "Subclass failed to decide allocation");
|
|
gst_query_unref (query);
|
|
|
|
return result;
|
|
}
|
|
}
|
|
|
|
/* 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_query_caps (GST_BASE_SRC_PAD (basesrc), NULL);
|
|
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_query_caps (GST_BASE_SRC_PAD (basesrc), thiscaps);
|
|
GST_DEBUG_OBJECT (basesrc, "caps of peer: %" GST_PTR_FORMAT, peercaps);
|
|
if (peercaps) {
|
|
/* The result is already a subset of our caps */
|
|
caps = peercaps;
|
|
gst_caps_unref (thiscaps);
|
|
} else {
|
|
/* no peer, work with our own caps then */
|
|
caps = thiscaps;
|
|
}
|
|
if (caps && !gst_caps_is_empty (caps)) {
|
|
/* now fixate */
|
|
GST_DEBUG_OBJECT (basesrc, "have caps: %" GST_PTR_FORMAT, caps);
|
|
if (gst_caps_is_any (caps)) {
|
|
GST_DEBUG_OBJECT (basesrc, "any caps, we stop");
|
|
/* hmm, still anything, so element can do anything and
|
|
* nego is not needed */
|
|
result = TRUE;
|
|
} else {
|
|
caps = gst_base_src_fixate (basesrc, caps);
|
|
GST_DEBUG_OBJECT (basesrc, "fixated to: %" GST_PTR_FORMAT, caps);
|
|
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_base_src_set_caps (basesrc, caps);
|
|
}
|
|
}
|
|
gst_caps_unref (caps);
|
|
} else {
|
|
if (caps)
|
|
gst_caps_unref (caps);
|
|
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;
|
|
|
|
bclass = GST_BASE_SRC_GET_CLASS (basesrc);
|
|
|
|
GST_DEBUG_OBJECT (basesrc, "starting negotiation");
|
|
|
|
if (G_LIKELY (bclass->negotiate))
|
|
result = bclass->negotiate (basesrc);
|
|
else
|
|
result = TRUE;
|
|
|
|
if (G_LIKELY (result)) {
|
|
GstCaps *caps;
|
|
|
|
caps = gst_pad_get_current_caps (basesrc->srcpad);
|
|
|
|
result = gst_base_src_prepare_allocation (basesrc, caps);
|
|
|
|
if (caps)
|
|
gst_caps_unref (caps);
|
|
}
|
|
return result;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_start (GstBaseSrc * basesrc)
|
|
{
|
|
GstBaseSrcClass *bclass;
|
|
gboolean result;
|
|
|
|
GST_LIVE_LOCK (basesrc);
|
|
|
|
GST_OBJECT_LOCK (basesrc);
|
|
if (GST_BASE_SRC_IS_STARTING (basesrc))
|
|
goto was_starting;
|
|
if (GST_BASE_SRC_IS_STARTED (basesrc))
|
|
goto was_started;
|
|
|
|
basesrc->priv->start_result = GST_FLOW_FLUSHING;
|
|
GST_OBJECT_FLAG_SET (basesrc, GST_BASE_SRC_FLAG_STARTING);
|
|
gst_segment_init (&basesrc->segment, basesrc->segment.format);
|
|
GST_OBJECT_UNLOCK (basesrc);
|
|
|
|
basesrc->num_buffers_left = basesrc->num_buffers;
|
|
basesrc->running = FALSE;
|
|
basesrc->priv->segment_pending = FALSE;
|
|
GST_LIVE_UNLOCK (basesrc);
|
|
|
|
bclass = GST_BASE_SRC_GET_CLASS (basesrc);
|
|
if (bclass->start)
|
|
result = bclass->start (basesrc);
|
|
else
|
|
result = TRUE;
|
|
|
|
if (!result)
|
|
goto could_not_start;
|
|
|
|
if (!gst_base_src_is_async (basesrc)) {
|
|
gst_base_src_start_complete (basesrc, GST_FLOW_OK);
|
|
/* not really waiting here, we call this to get the result
|
|
* from the start_complete call */
|
|
result = gst_base_src_start_wait (basesrc) == GST_FLOW_OK;
|
|
}
|
|
|
|
return result;
|
|
|
|
/* ERROR */
|
|
was_starting:
|
|
{
|
|
GST_DEBUG_OBJECT (basesrc, "was starting");
|
|
GST_OBJECT_UNLOCK (basesrc);
|
|
GST_LIVE_UNLOCK (basesrc);
|
|
return TRUE;
|
|
}
|
|
was_started:
|
|
{
|
|
GST_DEBUG_OBJECT (basesrc, "was started");
|
|
GST_OBJECT_UNLOCK (basesrc);
|
|
GST_LIVE_UNLOCK (basesrc);
|
|
return TRUE;
|
|
}
|
|
could_not_start:
|
|
{
|
|
GST_DEBUG_OBJECT (basesrc, "could not start");
|
|
/* subclass is supposed to post a message. We don't have to call _stop. */
|
|
gst_base_src_start_complete (basesrc, GST_FLOW_ERROR);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_base_src_start_complete:
|
|
* @basesrc: base source instance
|
|
* @ret: a #GstFlowReturn
|
|
*
|
|
* Complete an asynchronous start operation. When the subclass overrides the
|
|
* start method, it should call gst_base_src_start_complete() when the start
|
|
* operation completes either from the same thread or from an asynchronous
|
|
* helper thread.
|
|
*/
|
|
void
|
|
gst_base_src_start_complete (GstBaseSrc * basesrc, GstFlowReturn ret)
|
|
{
|
|
gboolean have_size;
|
|
guint64 size;
|
|
gboolean seekable;
|
|
GstFormat format;
|
|
GstPadMode mode;
|
|
GstEvent *event;
|
|
|
|
if (ret != GST_FLOW_OK)
|
|
goto error;
|
|
|
|
GST_DEBUG_OBJECT (basesrc, "starting source");
|
|
format = basesrc->segment.format;
|
|
|
|
/* figure out the size */
|
|
have_size = FALSE;
|
|
size = -1;
|
|
if (format == GST_FORMAT_BYTES) {
|
|
GstBaseSrcClass *bclass = GST_BASE_SRC_GET_CLASS (basesrc);
|
|
|
|
if (bclass->get_size) {
|
|
if (!(have_size = bclass->get_size (basesrc, &size)))
|
|
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_OBJECT_LOCK (basesrc);
|
|
basesrc->segment.duration = size;
|
|
GST_OBJECT_UNLOCK (basesrc);
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (basesrc,
|
|
"format: %s, have size: %d, size: %" G_GUINT64_FORMAT ", duration: %"
|
|
G_GINT64_FORMAT, gst_format_get_name (format), have_size, 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 && format == GST_FORMAT_BYTES;
|
|
|
|
GST_DEBUG_OBJECT (basesrc, "is random_access: %d", basesrc->random_access);
|
|
|
|
/* 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, NULL);
|
|
|
|
gst_pad_mark_reconfigure (GST_BASE_SRC_PAD (basesrc));
|
|
|
|
GST_OBJECT_LOCK (basesrc->srcpad);
|
|
mode = GST_PAD_MODE (basesrc->srcpad);
|
|
GST_OBJECT_UNLOCK (basesrc->srcpad);
|
|
|
|
/* take the stream lock here, we only want to let the task run when we have
|
|
* set the STARTED flag */
|
|
GST_PAD_STREAM_LOCK (basesrc->srcpad);
|
|
if (mode == GST_PAD_MODE_PUSH) {
|
|
/* do initial seek, which will start the task */
|
|
GST_OBJECT_LOCK (basesrc);
|
|
event = basesrc->pending_seek;
|
|
basesrc->pending_seek = NULL;
|
|
GST_OBJECT_UNLOCK (basesrc);
|
|
|
|
/* The perform seek code will start the task when finished. We don't have to
|
|
* unlock the streaming thread because it is not running yet */
|
|
if (G_UNLIKELY (!gst_base_src_perform_seek (basesrc, event, FALSE)))
|
|
goto seek_failed;
|
|
|
|
if (event)
|
|
gst_event_unref (event);
|
|
} else {
|
|
/* if not random_access, we cannot operate in pull mode for now */
|
|
if (G_UNLIKELY (!basesrc->random_access))
|
|
goto no_get_range;
|
|
}
|
|
|
|
GST_OBJECT_LOCK (basesrc);
|
|
GST_OBJECT_FLAG_SET (basesrc, GST_BASE_SRC_FLAG_STARTED);
|
|
GST_OBJECT_FLAG_UNSET (basesrc, GST_BASE_SRC_FLAG_STARTING);
|
|
basesrc->priv->start_result = ret;
|
|
GST_ASYNC_SIGNAL (basesrc);
|
|
GST_OBJECT_UNLOCK (basesrc);
|
|
|
|
GST_PAD_STREAM_UNLOCK (basesrc->srcpad);
|
|
|
|
return;
|
|
|
|
seek_failed:
|
|
{
|
|
GST_PAD_STREAM_UNLOCK (basesrc->srcpad);
|
|
GST_ERROR_OBJECT (basesrc, "Failed to perform initial seek");
|
|
gst_base_src_set_flushing (basesrc, TRUE, FALSE, NULL);
|
|
if (event)
|
|
gst_event_unref (event);
|
|
ret = GST_FLOW_ERROR;
|
|
goto error;
|
|
}
|
|
no_get_range:
|
|
{
|
|
GST_PAD_STREAM_UNLOCK (basesrc->srcpad);
|
|
gst_base_src_set_flushing (basesrc, TRUE, FALSE, NULL);
|
|
GST_ERROR_OBJECT (basesrc, "Cannot operate in pull mode, stopping");
|
|
ret = GST_FLOW_ERROR;
|
|
goto error;
|
|
}
|
|
error:
|
|
{
|
|
GST_OBJECT_LOCK (basesrc);
|
|
basesrc->priv->start_result = ret;
|
|
GST_OBJECT_FLAG_UNSET (basesrc, GST_BASE_SRC_FLAG_STARTING);
|
|
GST_ASYNC_SIGNAL (basesrc);
|
|
GST_OBJECT_UNLOCK (basesrc);
|
|
return;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_base_src_start_wait:
|
|
* @basesrc: base source instance
|
|
*
|
|
* Wait until the start operation completes.
|
|
*
|
|
* Returns: a #GstFlowReturn.
|
|
*/
|
|
GstFlowReturn
|
|
gst_base_src_start_wait (GstBaseSrc * basesrc)
|
|
{
|
|
GstFlowReturn result;
|
|
|
|
GST_OBJECT_LOCK (basesrc);
|
|
while (GST_BASE_SRC_IS_STARTING (basesrc)) {
|
|
GST_ASYNC_WAIT (basesrc);
|
|
}
|
|
result = basesrc->priv->start_result;
|
|
GST_OBJECT_UNLOCK (basesrc);
|
|
|
|
GST_DEBUG_OBJECT (basesrc, "got %s", gst_flow_get_name (result));
|
|
|
|
return result;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_stop (GstBaseSrc * basesrc)
|
|
{
|
|
GstBaseSrcClass *bclass;
|
|
gboolean result = TRUE;
|
|
|
|
GST_DEBUG_OBJECT (basesrc, "stopping source");
|
|
|
|
/* flush all */
|
|
gst_base_src_set_flushing (basesrc, TRUE, FALSE, NULL);
|
|
/* stop the task */
|
|
gst_pad_stop_task (basesrc->srcpad);
|
|
|
|
GST_OBJECT_LOCK (basesrc);
|
|
if (!GST_BASE_SRC_IS_STARTED (basesrc) && !GST_BASE_SRC_IS_STARTING (basesrc))
|
|
goto was_stopped;
|
|
|
|
GST_OBJECT_FLAG_UNSET (basesrc, GST_BASE_SRC_FLAG_STARTING);
|
|
GST_OBJECT_FLAG_UNSET (basesrc, GST_BASE_SRC_FLAG_STARTED);
|
|
basesrc->priv->start_result = GST_FLOW_FLUSHING;
|
|
GST_ASYNC_SIGNAL (basesrc);
|
|
GST_OBJECT_UNLOCK (basesrc);
|
|
|
|
bclass = GST_BASE_SRC_GET_CLASS (basesrc);
|
|
if (bclass->stop)
|
|
result = bclass->stop (basesrc);
|
|
|
|
gst_base_src_set_allocation (basesrc, NULL, NULL, NULL);
|
|
|
|
return result;
|
|
|
|
was_stopped:
|
|
{
|
|
GST_DEBUG_OBJECT (basesrc, "was started");
|
|
GST_OBJECT_UNLOCK (basesrc);
|
|
return TRUE;
|
|
}
|
|
}
|
|
|
|
/* start or stop flushing dataprocessing
|
|
*/
|
|
static gboolean
|
|
gst_base_src_set_flushing (GstBaseSrc * basesrc,
|
|
gboolean flushing, gboolean live_play, gboolean * playing)
|
|
{
|
|
GstBaseSrcClass *bclass;
|
|
|
|
bclass = GST_BASE_SRC_GET_CLASS (basesrc);
|
|
|
|
GST_DEBUG_OBJECT (basesrc, "flushing %d, live_play %d", flushing, live_play);
|
|
|
|
if (flushing) {
|
|
gst_base_src_activate_pool (basesrc, FALSE);
|
|
/* 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_base_src_activate_pool (basesrc, TRUE);
|
|
|
|
/* Drop all delayed events */
|
|
GST_OBJECT_LOCK (basesrc);
|
|
if (basesrc->priv->pending_events) {
|
|
g_list_foreach (basesrc->priv->pending_events, (GFunc) gst_event_unref,
|
|
NULL);
|
|
g_list_free (basesrc->priv->pending_events);
|
|
basesrc->priv->pending_events = NULL;
|
|
g_atomic_int_set (&basesrc->priv->have_events, FALSE);
|
|
}
|
|
GST_OBJECT_UNLOCK (basesrc);
|
|
}
|
|
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_MODE (basesrc->srcpad) == GST_PAD_MODE_PUSH);
|
|
GST_OBJECT_UNLOCK (basesrc->srcpad);
|
|
if (start)
|
|
gst_pad_start_task (basesrc->srcpad, (GstTaskFunction) gst_base_src_loop,
|
|
basesrc->srcpad, NULL);
|
|
GST_DEBUG_OBJECT (basesrc, "signal");
|
|
GST_LIVE_SIGNAL (basesrc);
|
|
}
|
|
GST_LIVE_UNLOCK (basesrc);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_activate_push (GstPad * pad, GstObject * parent, gboolean active)
|
|
{
|
|
GstBaseSrc *basesrc;
|
|
|
|
basesrc = GST_BASE_SRC (parent);
|
|
|
|
/* 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;
|
|
} else {
|
|
GST_DEBUG_OBJECT (basesrc, "Deactivating in push mode");
|
|
/* 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;
|
|
}
|
|
error_stop:
|
|
{
|
|
GST_DEBUG_OBJECT (basesrc, "Failed to stop in push mode");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_activate_pull (GstPad * pad, GstObject * parent, gboolean active)
|
|
{
|
|
GstBaseSrc *basesrc;
|
|
|
|
basesrc = GST_BASE_SRC (parent);
|
|
|
|
/* prepare subclass first */
|
|
if (active) {
|
|
GST_DEBUG_OBJECT (basesrc, "Activating in pull mode");
|
|
if (G_UNLIKELY (!gst_base_src_start (basesrc)))
|
|
goto error_start;
|
|
} else {
|
|
GST_DEBUG_OBJECT (basesrc, "Deactivating in pull mode");
|
|
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;
|
|
}
|
|
error_stop:
|
|
{
|
|
GST_ERROR_OBJECT (basesrc, "Failed to stop in pull mode");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_activate_mode (GstPad * pad, GstObject * parent,
|
|
GstPadMode mode, gboolean active)
|
|
{
|
|
gboolean res;
|
|
GstBaseSrc *src = GST_BASE_SRC (parent);
|
|
|
|
src->priv->stream_start_pending = FALSE;
|
|
|
|
switch (mode) {
|
|
case GST_PAD_MODE_PULL:
|
|
res = gst_base_src_activate_pull (pad, parent, active);
|
|
break;
|
|
case GST_PAD_MODE_PUSH:
|
|
src->priv->stream_start_pending = active;
|
|
res = gst_base_src_activate_push (pad, parent, active);
|
|
break;
|
|
default:
|
|
GST_LOG_OBJECT (pad, "unknown activation mode %d", mode);
|
|
res = FALSE;
|
|
break;
|
|
}
|
|
return res;
|
|
}
|
|
|
|
|
|
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:
|
|
{
|
|
/* we don't need to unblock anything here, the pad deactivation code
|
|
* already did this */
|
|
g_atomic_int_set (&basesrc->priv->pending_eos, FALSE);
|
|
gst_event_replace (&basesrc->pending_seek, 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;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_base_src_get_buffer_pool:
|
|
* @src: a #GstBaseSrc
|
|
*
|
|
* Returns: (transfer full): the instance of the #GstBufferPool used
|
|
* by the src; free it after use it
|
|
*/
|
|
GstBufferPool *
|
|
gst_base_src_get_buffer_pool (GstBaseSrc * src)
|
|
{
|
|
g_return_val_if_fail (GST_IS_BASE_SRC (src), NULL);
|
|
|
|
if (src->priv->pool)
|
|
return gst_object_ref (src->priv->pool);
|
|
|
|
return NULL;
|
|
}
|
|
|
|
/**
|
|
* gst_base_src_get_allocator:
|
|
* @src: a #GstBaseSrc
|
|
* @allocator: (out) (allow-none) (transfer full): the #GstAllocator
|
|
* used
|
|
* @params: (out) (allow-none) (transfer full): the
|
|
* #GstAllocatorParams of @allocator
|
|
*
|
|
* Lets #GstBaseSrc sub-classes to know the memory @allocator
|
|
* used by the base class and its @params.
|
|
*
|
|
* Unref the @allocator after use it.
|
|
*/
|
|
void
|
|
gst_base_src_get_allocator (GstBaseSrc * src,
|
|
GstAllocator ** allocator, GstAllocationParams * params)
|
|
{
|
|
g_return_if_fail (GST_IS_BASE_SRC (src));
|
|
|
|
if (allocator)
|
|
*allocator = src->priv->allocator ?
|
|
gst_object_ref (src->priv->allocator) : NULL;
|
|
|
|
if (params)
|
|
*params = src->priv->params;
|
|
}
|