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a6d8c4109e
Original commit message from CVS: * gst-libs/gst/rtp/gstbasertpdepayload.c: (gst_base_rtp_depayload_set_gst_timestamp): Use G_GINT64_CONSTANT macro for int64 constant. * win32/common/libgstinterfaces.def: * win32/common/libgsttag.def: Add new exported functions.
769 lines
22 KiB
C
769 lines
22 KiB
C
/* GStreamer
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* Copyright (C) <2005> Philippe Khalaf <burger@speedy.org>
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* Copyright (C) <2005> Nokia Corporation <kai.vehmanen@nokia.com>
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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:gstbasertpdepayload
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* @short_description: Base class for RTP depayloader
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*
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* <refsect2>
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* <para>
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* Provides a base class for RTP depayloaders
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* </para>
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* </refsect2>
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*/
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#include "gstbasertpdepayload.h"
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GST_DEBUG_CATEGORY_STATIC (basertpdepayload_debug);
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#define GST_CAT_DEFAULT (basertpdepayload_debug)
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#define GST_BASE_RTP_DEPAYLOAD_GET_PRIVATE(obj) \
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(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_BASE_RTP_DEPAYLOAD, GstBaseRTPDepayloadPrivate))
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struct _GstBaseRTPDepayloadPrivate
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{
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guint64 clock_base;
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GstClockTime npt_start;
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GstClockTime npt_stop;
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gdouble play_speed;
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gdouble play_scale;
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GstClockTime ts_wraparound;
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GstClockTime prev_timestamp;
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};
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/* Filter 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_QUEUE_DELAY 0
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enum
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{
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PROP_0,
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PROP_QUEUE_DELAY,
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};
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static void gst_base_rtp_depayload_finalize (GObject * object);
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static void gst_base_rtp_depayload_set_property (GObject * object,
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guint prop_id, const GValue * value, GParamSpec * pspec);
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static void gst_base_rtp_depayload_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec);
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static gboolean gst_base_rtp_depayload_setcaps (GstPad * pad, GstCaps * caps);
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static GstFlowReturn gst_base_rtp_depayload_chain (GstPad * pad,
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GstBuffer * in);
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static gboolean gst_base_rtp_depayload_handle_sink_event (GstPad * pad,
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GstEvent * event);
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static GstStateChangeReturn gst_base_rtp_depayload_change_state (GstElement *
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element, GstStateChange transition);
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static GstFlowReturn gst_base_rtp_depayload_add_to_queue (GstBaseRTPDepayload *
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filter, GstBuffer * in);
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static GstFlowReturn gst_base_rtp_depayload_process (GstBaseRTPDepayload *
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filter, GstBuffer * rtp_buf);
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static void gst_base_rtp_depayload_set_gst_timestamp
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(GstBaseRTPDepayload * filter, guint32 timestamp, GstBuffer * buf);
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static void gst_base_rtp_depayload_wait (GstBaseRTPDepayload * filter,
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GstClockTime time);
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GST_BOILERPLATE (GstBaseRTPDepayload, gst_base_rtp_depayload, GstElement,
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GST_TYPE_ELEMENT);
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static void
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gst_base_rtp_depayload_base_init (gpointer klass)
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{
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/*GstElementClass *element_class = GST_ELEMENT_CLASS (klass); */
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}
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static void
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gst_base_rtp_depayload_class_init (GstBaseRTPDepayloadClass * 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 = (GstElementClass *) klass;
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parent_class = g_type_class_peek_parent (klass);
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g_type_class_add_private (klass, sizeof (GstBaseRTPDepayloadPrivate));
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gobject_class->finalize = gst_base_rtp_depayload_finalize;
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gobject_class->set_property = gst_base_rtp_depayload_set_property;
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gobject_class->get_property = gst_base_rtp_depayload_get_property;
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g_object_class_install_property (gobject_class, PROP_QUEUE_DELAY,
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g_param_spec_uint ("queue_delay", "Queue Delay",
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"Amount of ms to queue/buffer", 0, G_MAXUINT, DEFAULT_QUEUE_DELAY,
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G_PARAM_READWRITE));
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gstelement_class->change_state = gst_base_rtp_depayload_change_state;
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klass->add_to_queue = gst_base_rtp_depayload_add_to_queue;
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klass->set_gst_timestamp = gst_base_rtp_depayload_set_gst_timestamp;
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GST_DEBUG_CATEGORY_INIT (basertpdepayload_debug, "basertpdepayload", 0,
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"Base class for RTP Depayloaders");
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}
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static void
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gst_base_rtp_depayload_init (GstBaseRTPDepayload * filter,
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GstBaseRTPDepayloadClass * klass)
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{
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GstPadTemplate *pad_template;
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GstBaseRTPDepayloadPrivate *priv;
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priv = GST_BASE_RTP_DEPAYLOAD_GET_PRIVATE (filter);
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filter->priv = priv;
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GST_DEBUG_OBJECT (filter, "init");
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pad_template =
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gst_element_class_get_pad_template (GST_ELEMENT_CLASS (klass), "sink");
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g_return_if_fail (pad_template != NULL);
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filter->sinkpad = gst_pad_new_from_template (pad_template, "sink");
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gst_pad_set_setcaps_function (filter->sinkpad,
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gst_base_rtp_depayload_setcaps);
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gst_pad_set_chain_function (filter->sinkpad, gst_base_rtp_depayload_chain);
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gst_pad_set_event_function (filter->sinkpad,
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gst_base_rtp_depayload_handle_sink_event);
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gst_element_add_pad (GST_ELEMENT (filter), filter->sinkpad);
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pad_template =
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gst_element_class_get_pad_template (GST_ELEMENT_CLASS (klass), "src");
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g_return_if_fail (pad_template != NULL);
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filter->srcpad = gst_pad_new_from_template (pad_template, "src");
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gst_pad_use_fixed_caps (filter->srcpad);
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gst_element_add_pad (GST_ELEMENT (filter), filter->srcpad);
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filter->queue = g_queue_new ();
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filter->queue_delay = DEFAULT_QUEUE_DELAY;
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}
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static void
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gst_base_rtp_depayload_finalize (GObject * object)
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{
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GstBuffer *buf;
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GstBaseRTPDepayload *filter = GST_BASE_RTP_DEPAYLOAD (object);
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while ((buf = g_queue_pop_head (filter->queue)))
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gst_buffer_unref (buf);
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g_queue_free (filter->queue);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static gboolean
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gst_base_rtp_depayload_setcaps (GstPad * pad, GstCaps * caps)
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{
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GstBaseRTPDepayload *filter;
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GstBaseRTPDepayloadClass *bclass;
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GstBaseRTPDepayloadPrivate *priv;
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gboolean res;
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GstStructure *caps_struct;
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const GValue *value;
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filter = GST_BASE_RTP_DEPAYLOAD (gst_pad_get_parent (pad));
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priv = filter->priv;
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bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
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GST_DEBUG_OBJECT (filter, "Set caps");
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caps_struct = gst_caps_get_structure (caps, 0);
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/* get clock base if any, we need this for the newsegment */
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value = gst_structure_get_value (caps_struct, "clock-base");
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if (value && G_VALUE_HOLDS_UINT (value))
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priv->clock_base = g_value_get_uint (value);
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else
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priv->clock_base = -1;
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/* get other values for newsegment */
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value = gst_structure_get_value (caps_struct, "npt-start");
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if (value && G_VALUE_HOLDS_UINT64 (value))
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priv->npt_start = g_value_get_uint64 (value);
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else
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priv->npt_start = 0;
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value = gst_structure_get_value (caps_struct, "npt-stop");
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if (value && G_VALUE_HOLDS_UINT64 (value))
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priv->npt_stop = g_value_get_uint64 (value);
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else
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priv->npt_stop = -1;
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value = gst_structure_get_value (caps_struct, "play-speed");
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if (value && G_VALUE_HOLDS_DOUBLE (value))
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priv->play_speed = g_value_get_double (value);
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else
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priv->play_speed = 1.0;
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value = gst_structure_get_value (caps_struct, "play-scale");
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if (value && G_VALUE_HOLDS_DOUBLE (value))
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priv->play_scale = g_value_get_double (value);
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else
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priv->play_scale = 1.0;
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priv->prev_timestamp = -1;
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if (bclass->set_caps)
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res = bclass->set_caps (filter, caps);
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else
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res = TRUE;
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gst_object_unref (filter);
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return res;
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}
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static GstFlowReturn
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gst_base_rtp_depayload_chain (GstPad * pad, GstBuffer * in)
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{
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GstBaseRTPDepayload *filter;
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GstBaseRTPDepayloadClass *bclass;
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GstFlowReturn ret = GST_FLOW_OK;
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filter = GST_BASE_RTP_DEPAYLOAD (GST_OBJECT_PARENT (pad));
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if (filter->clock_rate == 0)
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goto not_configured;
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bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
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if (filter->queue_delay == 0) {
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GST_DEBUG_OBJECT (filter, "Pushing directly!");
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ret = gst_base_rtp_depayload_process (filter, in);
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} else {
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if (bclass->add_to_queue)
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ret = bclass->add_to_queue (filter, in);
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else
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goto no_delay;
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}
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return ret;
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/* ERRORS */
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not_configured:
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{
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GST_ELEMENT_ERROR (filter, STREAM, FORMAT,
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(NULL), ("no clock rate was specified, likely incomplete input caps"));
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gst_buffer_unref (in);
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return GST_FLOW_NOT_NEGOTIATED;
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}
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no_delay:
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{
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GST_ELEMENT_ERROR (filter, STREAM, NOT_IMPLEMENTED,
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(NULL), ("This element cannot operate with delay"));
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gst_buffer_unref (in);
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return GST_FLOW_NOT_SUPPORTED;
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}
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}
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static gboolean
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gst_base_rtp_depayload_handle_sink_event (GstPad * pad, GstEvent * event)
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{
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GstBaseRTPDepayload *filter =
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GST_BASE_RTP_DEPAYLOAD (GST_OBJECT_PARENT (pad));
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gboolean res = TRUE;
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_NEWSEGMENT:
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{
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/* intercept NEWSEGMENT events only if the packet scheduler thread
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is active */
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if (filter->thread) {
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GST_DEBUG_OBJECT (filter,
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"Upstream sent a NEWSEGMENT, handle in worker thread.");
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/* the worker thread will assign a new RTP-TS<->GST-TS mapping
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* based on the next processed RTP packet */
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filter->need_newsegment = TRUE;
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gst_event_unref (event);
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break;
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} else {
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GstFormat format;
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gst_event_parse_new_segment (event, NULL, NULL, &format, NULL, NULL,
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NULL);
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if (format != GST_FORMAT_TIME)
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goto wrong_format;
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GST_DEBUG_OBJECT (filter,
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"Upstream sent a NEWSEGMENT, passing through.");
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}
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/* note: pass through to default if no thread running */
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}
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default:
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/* pass other events forward */
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res = gst_pad_push_event (filter->srcpad, event);
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break;
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}
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return res;
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/* ERRORS */
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wrong_format:
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{
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GST_DEBUG_OBJECT (filter,
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"Upstream sent a NEWSEGMENT in wrong format, dropping.");
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gst_event_unref (event);
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return TRUE;
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}
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}
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static GstFlowReturn
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gst_base_rtp_depayload_add_to_queue (GstBaseRTPDepayload * filter,
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GstBuffer * in)
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{
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GQueue *queue = filter->queue;
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int i;
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/* our first packet, just push it */
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QUEUE_LOCK (filter);
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if (g_queue_is_empty (queue)) {
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g_queue_push_tail (queue, in);
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QUEUE_UNLOCK (filter);
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} else {
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guint16 seqnum, queueseq;
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guint32 timestamp;
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seqnum = gst_rtp_buffer_get_seq (in);
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queueseq = gst_rtp_buffer_get_seq (GST_BUFFER (g_queue_peek_head (queue)));
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/* look for right place to insert it */
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i = 0;
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/* Check for seqnum wraparound.
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* Seqnums in the lowest quadrant of the 0-65535 space are considered to
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* be greater than seqnums in the highest quadrant of this space. */
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while (seqnum > queueseq || (seqnum < 16384 && queueseq > 49150)) {
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gpointer data;
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i++;
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data = g_queue_peek_nth (queue, i);
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if (!data)
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break;
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queueseq = gst_rtp_buffer_get_seq (GST_BUFFER (data));
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}
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/* now insert it at that place */
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g_queue_push_nth (queue, in, i);
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QUEUE_UNLOCK (filter);
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timestamp = gst_rtp_buffer_get_timestamp (in);
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GST_DEBUG_OBJECT (filter,
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"Packet added to queue %d at pos %d timestamp %u sn %d",
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g_queue_get_length (queue), i, timestamp, seqnum);
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}
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return GST_FLOW_OK;
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}
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static GstFlowReturn
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gst_base_rtp_depayload_push_full (GstBaseRTPDepayload * filter,
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gboolean do_ts, guint32 timestamp, GstBuffer * out_buf)
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{
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GstFlowReturn ret;
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GstCaps *srccaps;
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GstBaseRTPDepayloadClass *bclass;
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/* set the caps if any */
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srccaps = GST_PAD_CAPS (filter->srcpad);
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if (srccaps)
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gst_buffer_set_caps (out_buf, srccaps);
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bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
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/* set the timestamp if we must and can */
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if (bclass->set_gst_timestamp && do_ts)
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bclass->set_gst_timestamp (filter, timestamp, out_buf);
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/* push it */
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GST_LOG_OBJECT (filter, "Pushing buffer size %d, timestamp %" GST_TIME_FORMAT,
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GST_BUFFER_SIZE (out_buf),
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GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (out_buf)));
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ret = gst_pad_push (filter->srcpad, out_buf);
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GST_LOG_OBJECT (filter, "Pushed buffer: %s", gst_flow_get_name (ret));
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return ret;
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}
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/**
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* gst_base_rtp_depayload_push_ts:
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* @filter: a #GstBaseRTPDepayload
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* @timestamp: an RTP timestamp to apply
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* @out_buf: a #GstBuffer
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*
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* Push @out_buf to the peer of @filter. This function takes ownership of
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* @out_buf.
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*
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* Unlike gst_base_rtp_depayload_push(), this function will apply @timestamp
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* on the outgoing buffer, using the configured clock_rate to convert the
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* timestamp to a valid GStreamer clock time.
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*
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* Returns: a #GstFlowReturn.
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*/
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GstFlowReturn
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gst_base_rtp_depayload_push_ts (GstBaseRTPDepayload * filter, guint32 timestamp,
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GstBuffer * out_buf)
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{
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return gst_base_rtp_depayload_push_full (filter, TRUE, timestamp, out_buf);
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}
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/**
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* gst_base_rtp_depayload_push:
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* @filter: a #GstBaseRTPDepayload
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* @out_buf: a #GstBuffer
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*
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* Push @out_buf to the peer of @filter. This function takes ownership of
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* @out_buf.
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*
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* Unlike gst_base_rtp_depayload_push_ts(), this function will not apply
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* any timestamp on the outgoing buffer.
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*
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* Returns: a #GstFlowReturn.
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*/
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GstFlowReturn
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gst_base_rtp_depayload_push (GstBaseRTPDepayload * filter, GstBuffer * out_buf)
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{
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return gst_base_rtp_depayload_push_full (filter, FALSE, 0, out_buf);
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}
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static GstFlowReturn
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gst_base_rtp_depayload_process (GstBaseRTPDepayload * filter,
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GstBuffer * rtp_buf)
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{
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GstBaseRTPDepayloadClass *bclass;
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GstBuffer *out_buf;
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GstFlowReturn ret = GST_FLOW_OK;
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bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
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/* let's send it out to processing */
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out_buf = bclass->process (filter, rtp_buf);
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if (out_buf) {
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guint32 timestamp = gst_rtp_buffer_get_timestamp (rtp_buf);
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/* push buffer with timestamp
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* We are assuming here that the timestamp of the last RTP buffer
|
|
* is the same as the timestamp wanted on the collector. If this is not a
|
|
* desired result, the process function should push itself with another
|
|
* timestamp and return NULL.
|
|
*/
|
|
ret = gst_base_rtp_depayload_push_ts (filter, timestamp, out_buf);
|
|
}
|
|
gst_buffer_unref (rtp_buf);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static void
|
|
gst_base_rtp_depayload_set_gst_timestamp (GstBaseRTPDepayload * filter,
|
|
guint32 timestamp, GstBuffer * buf)
|
|
{
|
|
GstClockTime ts, adjusted, exttimestamp;
|
|
GstBaseRTPDepayloadPrivate *priv;
|
|
guint64 diff;
|
|
|
|
priv = filter->priv;
|
|
|
|
/* no clock-base set, take first timestamp as base */
|
|
if (priv->clock_base == -1)
|
|
priv->clock_base = timestamp;
|
|
|
|
if (priv->prev_timestamp == -1) {
|
|
priv->prev_timestamp = timestamp;
|
|
priv->ts_wraparound = 0;
|
|
}
|
|
|
|
/* check for timestamp wraparound */
|
|
exttimestamp = timestamp + priv->ts_wraparound;
|
|
|
|
if (exttimestamp < priv->prev_timestamp)
|
|
diff = priv->prev_timestamp - exttimestamp;
|
|
else
|
|
diff = exttimestamp - priv->prev_timestamp;
|
|
|
|
if (diff > G_MAXINT32) {
|
|
/* timestamp went backwards more than allowed, we wrap around and get
|
|
* updated extended timestamp. */
|
|
priv->ts_wraparound += (G_GINT64_CONSTANT (1) << 32);
|
|
exttimestamp = timestamp + priv->ts_wraparound;
|
|
}
|
|
priv->prev_timestamp = exttimestamp;
|
|
|
|
/* rtp timestamps are based on the clock_rate
|
|
* gst timesamps are in nanoseconds */
|
|
ts = gst_util_uint64_scale_int (exttimestamp, GST_SECOND, filter->clock_rate);
|
|
|
|
GST_DEBUG_OBJECT (filter,
|
|
"timestamp: %u, wrap %" G_GUINT64_FORMAT ", clockrate : %u", timestamp,
|
|
priv->ts_wraparound, filter->clock_rate);
|
|
|
|
/* add delay to timestamp */
|
|
adjusted = ts + (filter->queue_delay * GST_MSECOND);
|
|
|
|
GST_DEBUG_OBJECT (filter, "RTP: %u, GST: %" GST_TIME_FORMAT ", adjusted %"
|
|
GST_TIME_FORMAT, timestamp, GST_TIME_ARGS (ts), GST_TIME_ARGS (adjusted));
|
|
|
|
GST_BUFFER_TIMESTAMP (buf) = adjusted;
|
|
|
|
/* if this is the first buf send a NEWSEGMENT */
|
|
if (filter->need_newsegment) {
|
|
GstEvent *event;
|
|
GstClockTime start, stop, position;
|
|
|
|
start = gst_util_uint64_scale_int (priv->clock_base, GST_SECOND,
|
|
filter->clock_rate);
|
|
|
|
if (priv->npt_stop != -1)
|
|
stop = priv->npt_stop - priv->npt_start + start;
|
|
else
|
|
stop = -1;
|
|
|
|
position = priv->npt_start;
|
|
|
|
event =
|
|
gst_event_new_new_segment_full (FALSE, priv->play_speed,
|
|
priv->play_scale, GST_FORMAT_TIME, start, stop, position);
|
|
|
|
gst_pad_push_event (filter->srcpad, event);
|
|
|
|
filter->need_newsegment = FALSE;
|
|
GST_DEBUG_OBJECT (filter, "Pushed newsegment event on this first buffer");
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_base_rtp_depayload_queue_release (GstBaseRTPDepayload * filter)
|
|
{
|
|
GQueue *queue = filter->queue;
|
|
guint32 headts, tailts;
|
|
GstBaseRTPDepayloadClass *bclass;
|
|
gfloat q_size_secs;
|
|
guint maxtsunits;
|
|
|
|
if (g_queue_is_empty (queue))
|
|
return;
|
|
|
|
/* if our queue is getting to big (more than RTP_QUEUEDELAY ms of data)
|
|
* release heading buffers
|
|
*/
|
|
/*GST_DEBUG_OBJECT (filter, "clockrate %d, queue_delay %d", filter->clock_rate,
|
|
filter->queue_delay); */
|
|
q_size_secs = (gfloat) filter->queue_delay / 1000;
|
|
maxtsunits = (gfloat) filter->clock_rate * q_size_secs;
|
|
|
|
QUEUE_LOCK (filter);
|
|
headts =
|
|
gst_rtp_buffer_get_timestamp (GST_BUFFER (g_queue_peek_head (queue)));
|
|
tailts =
|
|
gst_rtp_buffer_get_timestamp (GST_BUFFER (g_queue_peek_tail (queue)));
|
|
|
|
bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
|
|
|
|
/*GST_DEBUG("maxtsunit is %u %u %u %u", maxtsunits, headts, tailts, headts - tailts); */
|
|
while (headts - tailts > maxtsunits) {
|
|
GST_DEBUG_OBJECT (filter, "Poping packet from queue");
|
|
if (bclass->process) {
|
|
GstBuffer *in = g_queue_pop_head (queue);
|
|
|
|
gst_base_rtp_depayload_process (filter, in);
|
|
}
|
|
headts =
|
|
gst_rtp_buffer_get_timestamp (GST_BUFFER (g_queue_peek_head (queue)));
|
|
}
|
|
QUEUE_UNLOCK (filter);
|
|
}
|
|
|
|
|
|
static gpointer
|
|
gst_base_rtp_depayload_thread (GstBaseRTPDepayload * filter)
|
|
{
|
|
while (filter->thread_running) {
|
|
gst_base_rtp_depayload_queue_release (filter);
|
|
/* sleep for 5msec (XXX: 5msec is a value that works for audio and video,
|
|
* should be adjusted based on frequency of incoming packet,
|
|
* or by data comsumption rate of the sink (depends on how
|
|
* clock-drift compensation is implemented) */
|
|
gst_base_rtp_depayload_wait (filter, GST_MSECOND * 5);
|
|
}
|
|
return NULL;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_rtp_depayload_start_thread (GstBaseRTPDepayload * filter)
|
|
{
|
|
/* only launch the thread if processing is needed */
|
|
if (filter->queue_delay) {
|
|
GST_DEBUG_OBJECT (filter, "Starting queue release thread");
|
|
filter->thread_running = TRUE;
|
|
filter->thread =
|
|
g_thread_create ((GThreadFunc) gst_base_rtp_depayload_thread, filter,
|
|
TRUE, NULL);
|
|
GST_DEBUG_OBJECT (filter, "Started queue release thread");
|
|
}
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_rtp_depayload_stop_thread (GstBaseRTPDepayload * filter)
|
|
{
|
|
filter->thread_running = FALSE;
|
|
|
|
if (filter->thread) {
|
|
g_thread_join (filter->thread);
|
|
filter->thread = NULL;
|
|
}
|
|
QUEUE_LOCK_FREE (filter);
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
gst_base_rtp_depayload_wait (GstBaseRTPDepayload * filter, GstClockTime time)
|
|
{
|
|
GstClockID id;
|
|
GstClock *clock;
|
|
GstClockTime base;
|
|
|
|
g_return_if_fail (GST_CLOCK_TIME_IS_VALID (time));
|
|
|
|
GST_OBJECT_LOCK (filter);
|
|
if ((clock = GST_ELEMENT_CLOCK (filter)) == NULL)
|
|
goto no_clock;
|
|
gst_object_ref (clock);
|
|
GST_OBJECT_UNLOCK (filter);
|
|
|
|
base = gst_clock_get_time (clock);
|
|
id = gst_clock_new_single_shot_id (clock, base + time);
|
|
|
|
gst_object_unref (clock);
|
|
|
|
gst_clock_id_wait (id, NULL);
|
|
gst_clock_id_unref (id);
|
|
|
|
return;
|
|
|
|
no_clock:
|
|
{
|
|
GST_DEBUG_OBJECT (filter, "No clock given yet");
|
|
GST_OBJECT_UNLOCK (filter);
|
|
return;
|
|
}
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_base_rtp_depayload_change_state (GstElement * element,
|
|
GstStateChange transition)
|
|
{
|
|
GstBaseRTPDepayload *filter;
|
|
GstStateChangeReturn ret;
|
|
|
|
filter = GST_BASE_RTP_DEPAYLOAD (element);
|
|
|
|
/* we disallow changing the state from the thread */
|
|
if (g_thread_self () == filter->thread)
|
|
goto wrong_thread;
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_NULL_TO_READY:
|
|
if (!gst_base_rtp_depayload_start_thread (filter))
|
|
goto start_failed;
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
/* clock_rate needs to be overwritten by child */
|
|
filter->clock_rate = 0;
|
|
filter->priv->clock_base = -1;
|
|
filter->priv->ts_wraparound = 0;
|
|
filter->need_newsegment = TRUE;
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
gst_base_rtp_depayload_stop_thread (filter);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
wrong_thread:
|
|
{
|
|
GST_ELEMENT_ERROR (filter, CORE, STATE_CHANGE,
|
|
(NULL), ("cannot perform a state change from this thread"));
|
|
return GST_STATE_CHANGE_FAILURE;
|
|
}
|
|
start_failed:
|
|
{
|
|
/* start method should have posted an error message */
|
|
return GST_STATE_CHANGE_FAILURE;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_base_rtp_depayload_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstBaseRTPDepayload *filter;
|
|
|
|
filter = GST_BASE_RTP_DEPAYLOAD (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_QUEUE_DELAY:
|
|
filter->queue_delay = g_value_get_uint (value);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_base_rtp_depayload_get_property (GObject * object, guint prop_id,
|
|
GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstBaseRTPDepayload *filter;
|
|
|
|
filter = GST_BASE_RTP_DEPAYLOAD (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_QUEUE_DELAY:
|
|
g_value_set_uint (value, filter->queue_delay);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|