mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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983 lines
29 KiB
C
983 lines
29 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., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:gstrtpbasedepayload
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* @title: GstRTPBaseDepayload
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* @short_description: Base class for RTP depayloader
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*
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* Provides a base class for RTP depayloaders
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*/
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#include "gstrtpbasedepayload.h"
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GST_DEBUG_CATEGORY_STATIC (rtpbasedepayload_debug);
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#define GST_CAT_DEFAULT (rtpbasedepayload_debug)
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struct _GstRTPBaseDepayloadPrivate
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{
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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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guint clock_base;
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gboolean discont;
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GstClockTime pts;
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GstClockTime dts;
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GstClockTime duration;
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guint32 last_ssrc;
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guint32 last_seqnum;
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guint32 last_rtptime;
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guint32 next_seqnum;
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gboolean negotiated;
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GstCaps *last_caps;
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GstEvent *segment_event;
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guint32 segment_seqnum; /* Note: this is a GstEvent seqnum */
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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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enum
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{
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PROP_0,
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PROP_STATS,
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PROP_LAST
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};
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static void gst_rtp_base_depayload_finalize (GObject * object);
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static void gst_rtp_base_depayload_set_property (GObject * object,
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guint prop_id, const GValue * value, GParamSpec * pspec);
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static void gst_rtp_base_depayload_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec);
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static GstFlowReturn gst_rtp_base_depayload_chain (GstPad * pad,
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GstObject * parent, GstBuffer * in);
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static GstFlowReturn gst_rtp_base_depayload_chain_list (GstPad * pad,
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GstObject * parent, GstBufferList * list);
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static gboolean gst_rtp_base_depayload_handle_sink_event (GstPad * pad,
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GstObject * parent, GstEvent * event);
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static GstStateChangeReturn gst_rtp_base_depayload_change_state (GstElement *
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element, GstStateChange transition);
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static gboolean gst_rtp_base_depayload_packet_lost (GstRTPBaseDepayload *
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filter, GstEvent * event);
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static gboolean gst_rtp_base_depayload_handle_event (GstRTPBaseDepayload *
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filter, GstEvent * event);
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static GstElementClass *parent_class = NULL;
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static gint private_offset = 0;
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static void gst_rtp_base_depayload_class_init (GstRTPBaseDepayloadClass *
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klass);
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static void gst_rtp_base_depayload_init (GstRTPBaseDepayload * rtpbasepayload,
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GstRTPBaseDepayloadClass * klass);
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static GstEvent *create_segment_event (GstRTPBaseDepayload * filter,
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guint rtptime, GstClockTime position);
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GType
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gst_rtp_base_depayload_get_type (void)
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{
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static GType rtp_base_depayload_type = 0;
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if (g_once_init_enter ((gsize *) & rtp_base_depayload_type)) {
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static const GTypeInfo rtp_base_depayload_info = {
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sizeof (GstRTPBaseDepayloadClass),
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NULL,
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NULL,
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(GClassInitFunc) gst_rtp_base_depayload_class_init,
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NULL,
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NULL,
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sizeof (GstRTPBaseDepayload),
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0,
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(GInstanceInitFunc) gst_rtp_base_depayload_init,
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};
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GType _type;
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_type = g_type_register_static (GST_TYPE_ELEMENT, "GstRTPBaseDepayload",
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&rtp_base_depayload_info, G_TYPE_FLAG_ABSTRACT);
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private_offset =
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g_type_add_instance_private (_type,
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sizeof (GstRTPBaseDepayloadPrivate));
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g_once_init_leave ((gsize *) & rtp_base_depayload_type, _type);
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}
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return rtp_base_depayload_type;
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}
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static inline GstRTPBaseDepayloadPrivate *
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gst_rtp_base_depayload_get_instance_private (GstRTPBaseDepayload * self)
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{
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return (G_STRUCT_MEMBER_P (self, private_offset));
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}
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static void
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gst_rtp_base_depayload_class_init (GstRTPBaseDepayloadClass * 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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if (private_offset != 0)
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g_type_class_adjust_private_offset (klass, &private_offset);
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gobject_class->finalize = gst_rtp_base_depayload_finalize;
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gobject_class->set_property = gst_rtp_base_depayload_set_property;
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gobject_class->get_property = gst_rtp_base_depayload_get_property;
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/**
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* GstRTPBaseDepayload:stats:
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*
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* Various depayloader statistics retrieved atomically (and are therefore
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* synchroized with each other). This property return a GstStructure named
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* application/x-rtp-depayload-stats containing the following fields relating to
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* the last processed buffer and current state of the stream being depayloaded:
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*
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* * `clock-rate`: #G_TYPE_UINT, clock-rate of the stream
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* * `npt-start`: #G_TYPE_UINT64, time of playback start
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* * `npt-stop`: #G_TYPE_UINT64, time of playback stop
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* * `play-speed`: #G_TYPE_DOUBLE, the playback speed
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* * `play-scale`: #G_TYPE_DOUBLE, the playback scale
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* * `running-time-dts`: #G_TYPE_UINT64, the last running-time of the
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* last DTS
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* * `running-time-pts`: #G_TYPE_UINT64, the last running-time of the
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* last PTS
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* * `seqnum`: #G_TYPE_UINT, the last seen seqnum
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* * `timestamp`: #G_TYPE_UINT, the last seen RTP timestamp
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**/
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_STATS,
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g_param_spec_boxed ("stats", "Statistics", "Various statistics",
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GST_TYPE_STRUCTURE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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gstelement_class->change_state = gst_rtp_base_depayload_change_state;
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klass->packet_lost = gst_rtp_base_depayload_packet_lost;
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klass->handle_event = gst_rtp_base_depayload_handle_event;
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GST_DEBUG_CATEGORY_INIT (rtpbasedepayload_debug, "rtpbasedepayload", 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_rtp_base_depayload_init (GstRTPBaseDepayload * filter,
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GstRTPBaseDepayloadClass * klass)
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{
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GstPadTemplate *pad_template;
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GstRTPBaseDepayloadPrivate *priv;
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priv = gst_rtp_base_depayload_get_instance_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_chain_function (filter->sinkpad, gst_rtp_base_depayload_chain);
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gst_pad_set_chain_list_function (filter->sinkpad,
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gst_rtp_base_depayload_chain_list);
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gst_pad_set_event_function (filter->sinkpad,
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gst_rtp_base_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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priv->npt_start = 0;
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priv->npt_stop = -1;
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priv->play_speed = 1.0;
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priv->play_scale = 1.0;
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priv->clock_base = -1;
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priv->dts = -1;
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priv->pts = -1;
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priv->duration = -1;
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gst_segment_init (&filter->segment, GST_FORMAT_UNDEFINED);
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}
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static void
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gst_rtp_base_depayload_finalize (GObject * object)
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{
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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_rtp_base_depayload_setcaps (GstRTPBaseDepayload * filter, GstCaps * caps)
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{
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GstRTPBaseDepayloadClass *bclass;
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GstRTPBaseDepayloadPrivate *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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priv = filter->priv;
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bclass = GST_RTP_BASE_DEPAYLOAD_GET_CLASS (filter);
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GST_DEBUG_OBJECT (filter, "Set caps %" GST_PTR_FORMAT, caps);
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if (priv->last_caps) {
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if (gst_caps_is_equal (priv->last_caps, caps)) {
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res = TRUE;
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goto caps_not_changed;
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} else {
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gst_caps_unref (priv->last_caps);
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priv->last_caps = NULL;
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}
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}
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caps_struct = gst_caps_get_structure (caps, 0);
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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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GST_DEBUG_OBJECT (filter, "NPT start %" G_GUINT64_FORMAT, priv->npt_start);
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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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GST_DEBUG_OBJECT (filter, "NPT stop %" G_GUINT64_FORMAT, priv->npt_stop);
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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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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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if (bclass->set_caps) {
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res = bclass->set_caps (filter, caps);
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if (!res) {
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GST_WARNING_OBJECT (filter, "Subclass rejected caps %" GST_PTR_FORMAT,
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caps);
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}
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} else {
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res = TRUE;
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}
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priv->negotiated = res;
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if (priv->negotiated)
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priv->last_caps = gst_caps_ref (caps);
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return res;
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caps_not_changed:
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{
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GST_DEBUG_OBJECT (filter, "Caps did not change");
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return res;
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}
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}
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/* takes ownership of the input buffer */
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static GstFlowReturn
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gst_rtp_base_depayload_handle_buffer (GstRTPBaseDepayload * filter,
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GstRTPBaseDepayloadClass * bclass, GstBuffer * in)
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{
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GstBuffer *(*process_rtp_packet_func) (GstRTPBaseDepayload * base,
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GstRTPBuffer * rtp_buffer);
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GstBuffer *(*process_func) (GstRTPBaseDepayload * base, GstBuffer * in);
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GstRTPBaseDepayloadPrivate *priv;
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GstFlowReturn ret = GST_FLOW_OK;
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GstBuffer *out_buf;
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guint32 ssrc;
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guint16 seqnum;
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guint32 rtptime;
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gboolean discont, buf_discont;
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gint gap;
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GstRTPBuffer rtp = { NULL };
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priv = filter->priv;
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process_func = bclass->process;
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process_rtp_packet_func = bclass->process_rtp_packet;
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/* we must have a setcaps first */
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if (G_UNLIKELY (!priv->negotiated))
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goto not_negotiated;
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if (G_UNLIKELY (!gst_rtp_buffer_map (in, GST_MAP_READ, &rtp)))
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goto invalid_buffer;
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buf_discont = GST_BUFFER_IS_DISCONT (in);
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priv->pts = GST_BUFFER_PTS (in);
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priv->dts = GST_BUFFER_DTS (in);
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priv->duration = GST_BUFFER_DURATION (in);
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ssrc = gst_rtp_buffer_get_ssrc (&rtp);
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seqnum = gst_rtp_buffer_get_seq (&rtp);
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rtptime = gst_rtp_buffer_get_timestamp (&rtp);
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priv->last_seqnum = seqnum;
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priv->last_rtptime = rtptime;
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discont = buf_discont;
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GST_LOG_OBJECT (filter, "discont %d, seqnum %u, rtptime %u, pts %"
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GST_TIME_FORMAT ", dts %" GST_TIME_FORMAT, buf_discont, seqnum, rtptime,
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GST_TIME_ARGS (priv->pts), GST_TIME_ARGS (priv->dts));
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/* Check seqnum. This is a very simple check that makes sure that the seqnums
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* are strictly increasing, dropping anything that is out of the ordinary. We
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* can only do this when the next_seqnum is known. */
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if (G_LIKELY (priv->next_seqnum != -1)) {
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if (ssrc != priv->last_ssrc) {
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GST_LOG_OBJECT (filter,
|
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"New ssrc %u (current ssrc %u), sender restarted",
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ssrc, priv->last_ssrc);
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discont = TRUE;
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} else {
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gap = gst_rtp_buffer_compare_seqnum (seqnum, priv->next_seqnum);
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|
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/* if we have no gap, all is fine */
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if (G_UNLIKELY (gap != 0)) {
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GST_LOG_OBJECT (filter, "got packet %u, expected %u, gap %d", seqnum,
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priv->next_seqnum, gap);
|
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if (gap < 0) {
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/* seqnum > next_seqnum, we are missing some packets, this is always a
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* DISCONT. */
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GST_LOG_OBJECT (filter, "%d missing packets", gap);
|
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discont = TRUE;
|
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} else {
|
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/* seqnum < next_seqnum, we have seen this packet before or the sender
|
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* could be restarted. If the packet is not too old, we throw it away as
|
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* a duplicate, otherwise we mark discont and continue. 100 misordered
|
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* packets is a good threshold. See also RFC 4737. */
|
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if (gap < 100)
|
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goto dropping;
|
|
|
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GST_LOG_OBJECT (filter,
|
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"%d > 100, packet too old, sender likely restarted", gap);
|
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discont = TRUE;
|
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}
|
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}
|
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}
|
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}
|
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priv->next_seqnum = (seqnum + 1) & 0xffff;
|
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priv->last_ssrc = ssrc;
|
|
|
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if (G_UNLIKELY (discont)) {
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priv->discont = TRUE;
|
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if (!buf_discont) {
|
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gpointer old_inbuf = in;
|
|
|
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/* we detected a seqnum discont but the buffer was not flagged with a discont,
|
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* set the discont flag so that the subclass can throw away old data. */
|
|
GST_LOG_OBJECT (filter, "mark DISCONT on input buffer");
|
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in = gst_buffer_make_writable (in);
|
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GST_BUFFER_FLAG_SET (in, GST_BUFFER_FLAG_DISCONT);
|
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/* depayloaders will check flag on rtpbuffer->buffer, so if the input
|
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* buffer was not writable already we need to remap to make our
|
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* newly-flagged buffer current on the rtpbuffer */
|
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if (in != old_inbuf) {
|
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gst_rtp_buffer_unmap (&rtp);
|
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if (G_UNLIKELY (!gst_rtp_buffer_map (in, GST_MAP_READ, &rtp)))
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goto invalid_buffer;
|
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}
|
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}
|
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}
|
|
|
|
/* prepare segment event if needed */
|
|
if (filter->need_newsegment) {
|
|
priv->segment_event = create_segment_event (filter, rtptime,
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GST_BUFFER_PTS (in));
|
|
filter->need_newsegment = FALSE;
|
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}
|
|
|
|
if (process_rtp_packet_func != NULL) {
|
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out_buf = process_rtp_packet_func (filter, &rtp);
|
|
gst_rtp_buffer_unmap (&rtp);
|
|
} else if (process_func != NULL) {
|
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gst_rtp_buffer_unmap (&rtp);
|
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out_buf = process_func (filter, in);
|
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} else {
|
|
goto no_process;
|
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}
|
|
|
|
/* let's send it out to processing */
|
|
if (out_buf) {
|
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ret = gst_rtp_base_depayload_push (filter, out_buf);
|
|
}
|
|
|
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gst_buffer_unref (in);
|
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|
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return ret;
|
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|
|
/* ERRORS */
|
|
not_negotiated:
|
|
{
|
|
/* this is not fatal but should be filtered earlier */
|
|
GST_ELEMENT_ERROR (filter, CORE, NEGOTIATION,
|
|
("No RTP format was negotiated."),
|
|
("Input buffers need to have RTP caps set on them. This is usually "
|
|
"achieved by setting the 'caps' property of the upstream source "
|
|
"element (often udpsrc or appsrc), or by putting a capsfilter "
|
|
"element before the depayloader and setting the 'caps' property "
|
|
"on that. Also see http://cgit.freedesktop.org/gstreamer/"
|
|
"gst-plugins-good/tree/gst/rtp/README"));
|
|
gst_buffer_unref (in);
|
|
return GST_FLOW_NOT_NEGOTIATED;
|
|
}
|
|
invalid_buffer:
|
|
{
|
|
/* this is not fatal but should be filtered earlier */
|
|
GST_ELEMENT_WARNING (filter, STREAM, DECODE, (NULL),
|
|
("Received invalid RTP payload, dropping"));
|
|
gst_buffer_unref (in);
|
|
return GST_FLOW_OK;
|
|
}
|
|
dropping:
|
|
{
|
|
gst_rtp_buffer_unmap (&rtp);
|
|
GST_WARNING_OBJECT (filter, "%d <= 100, dropping old packet", gap);
|
|
gst_buffer_unref (in);
|
|
return GST_FLOW_OK;
|
|
}
|
|
no_process:
|
|
{
|
|
gst_rtp_buffer_unmap (&rtp);
|
|
/* this is not fatal but should be filtered earlier */
|
|
GST_ELEMENT_ERROR (filter, STREAM, NOT_IMPLEMENTED, (NULL),
|
|
("The subclass does not have a process or process_rtp_packet method"));
|
|
gst_buffer_unref (in);
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_rtp_base_depayload_chain (GstPad * pad, GstObject * parent, GstBuffer * in)
|
|
{
|
|
GstRTPBaseDepayloadClass *bclass;
|
|
GstRTPBaseDepayload *basedepay;
|
|
GstFlowReturn flow_ret;
|
|
|
|
basedepay = GST_RTP_BASE_DEPAYLOAD_CAST (parent);
|
|
|
|
bclass = GST_RTP_BASE_DEPAYLOAD_GET_CLASS (basedepay);
|
|
|
|
flow_ret = gst_rtp_base_depayload_handle_buffer (basedepay, bclass, in);
|
|
|
|
return flow_ret;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_rtp_base_depayload_chain_list (GstPad * pad, GstObject * parent,
|
|
GstBufferList * list)
|
|
{
|
|
GstRTPBaseDepayloadClass *bclass;
|
|
GstRTPBaseDepayload *basedepay;
|
|
GstFlowReturn flow_ret;
|
|
GstBuffer *buffer;
|
|
guint i, len;
|
|
|
|
basedepay = GST_RTP_BASE_DEPAYLOAD_CAST (parent);
|
|
|
|
bclass = GST_RTP_BASE_DEPAYLOAD_GET_CLASS (basedepay);
|
|
|
|
flow_ret = GST_FLOW_OK;
|
|
|
|
/* chain each buffer in list individually */
|
|
len = gst_buffer_list_length (list);
|
|
|
|
if (len == 0)
|
|
goto done;
|
|
|
|
for (i = 0; i < len; i++) {
|
|
buffer = gst_buffer_list_get (list, i);
|
|
|
|
/* handle_buffer takes ownership of input buffer */
|
|
/* FIXME: add a way to steal buffers from list as we will unref it anyway */
|
|
gst_buffer_ref (buffer);
|
|
|
|
/* Should we fix up any missing timestamps for list buffers here
|
|
* (e.g. set to first or previous timestamp in list) or just assume
|
|
* the's a jitterbuffer that will have done that for us? */
|
|
flow_ret = gst_rtp_base_depayload_handle_buffer (basedepay, bclass, buffer);
|
|
if (flow_ret != GST_FLOW_OK)
|
|
break;
|
|
}
|
|
|
|
done:
|
|
|
|
gst_buffer_list_unref (list);
|
|
|
|
return flow_ret;
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_base_depayload_handle_event (GstRTPBaseDepayload * filter,
|
|
GstEvent * event)
|
|
{
|
|
gboolean res = TRUE;
|
|
gboolean forward = TRUE;
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_FLUSH_STOP:
|
|
GST_OBJECT_LOCK (filter);
|
|
gst_segment_init (&filter->segment, GST_FORMAT_UNDEFINED);
|
|
GST_OBJECT_UNLOCK (filter);
|
|
|
|
filter->need_newsegment = TRUE;
|
|
filter->priv->next_seqnum = -1;
|
|
gst_event_replace (&filter->priv->segment_event, NULL);
|
|
break;
|
|
case GST_EVENT_CAPS:
|
|
{
|
|
GstCaps *caps;
|
|
|
|
gst_event_parse_caps (event, &caps);
|
|
|
|
res = gst_rtp_base_depayload_setcaps (filter, caps);
|
|
forward = FALSE;
|
|
break;
|
|
}
|
|
case GST_EVENT_SEGMENT:
|
|
{
|
|
GstSegment segment;
|
|
|
|
GST_OBJECT_LOCK (filter);
|
|
gst_event_copy_segment (event, &segment);
|
|
|
|
if (segment.format != GST_FORMAT_TIME) {
|
|
GST_ERROR_OBJECT (filter, "Segment with non-TIME format not supported");
|
|
res = FALSE;
|
|
}
|
|
filter->priv->segment_seqnum = gst_event_get_seqnum (event);
|
|
filter->segment = segment;
|
|
GST_OBJECT_UNLOCK (filter);
|
|
|
|
/* don't pass the event downstream, we generate our own segment including
|
|
* the NTP time and other things we receive in caps */
|
|
forward = FALSE;
|
|
break;
|
|
}
|
|
case GST_EVENT_CUSTOM_DOWNSTREAM:
|
|
{
|
|
GstRTPBaseDepayloadClass *bclass;
|
|
|
|
bclass = GST_RTP_BASE_DEPAYLOAD_GET_CLASS (filter);
|
|
|
|
if (gst_event_has_name (event, "GstRTPPacketLost")) {
|
|
/* we get this event from the jitterbuffer when it considers a packet as
|
|
* being lost. We send it to our packet_lost vmethod. The default
|
|
* implementation will make time progress by pushing out a GAP event.
|
|
* Subclasses can override and do one of the following:
|
|
* - Adjust timestamp/duration to something more accurate before
|
|
* calling the parent (default) packet_lost method.
|
|
* - do some more advanced error concealing on the already received
|
|
* (fragmented) packets.
|
|
* - ignore the packet lost.
|
|
*/
|
|
if (bclass->packet_lost)
|
|
res = bclass->packet_lost (filter, event);
|
|
forward = FALSE;
|
|
}
|
|
break;
|
|
}
|
|
default:
|
|
break;
|
|
}
|
|
|
|
if (forward)
|
|
res = gst_pad_push_event (filter->srcpad, event);
|
|
else
|
|
gst_event_unref (event);
|
|
|
|
return res;
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_base_depayload_handle_sink_event (GstPad * pad, GstObject * parent,
|
|
GstEvent * event)
|
|
{
|
|
gboolean res = FALSE;
|
|
GstRTPBaseDepayload *filter;
|
|
GstRTPBaseDepayloadClass *bclass;
|
|
|
|
filter = GST_RTP_BASE_DEPAYLOAD (parent);
|
|
bclass = GST_RTP_BASE_DEPAYLOAD_GET_CLASS (filter);
|
|
if (bclass->handle_event)
|
|
res = bclass->handle_event (filter, event);
|
|
else
|
|
gst_event_unref (event);
|
|
|
|
return res;
|
|
}
|
|
|
|
static GstEvent *
|
|
create_segment_event (GstRTPBaseDepayload * filter, guint rtptime,
|
|
GstClockTime position)
|
|
{
|
|
GstEvent *event;
|
|
GstClockTime start, stop, running_time;
|
|
GstRTPBaseDepayloadPrivate *priv;
|
|
GstSegment segment;
|
|
|
|
priv = filter->priv;
|
|
|
|
/* We don't need the object lock around - the segment
|
|
* can't change here while we're holding the STREAM_LOCK
|
|
*/
|
|
|
|
/* determining the start of the segment */
|
|
start = filter->segment.start;
|
|
if (priv->clock_base != -1 && position != -1) {
|
|
GstClockTime exttime, gap;
|
|
|
|
exttime = priv->clock_base;
|
|
gst_rtp_buffer_ext_timestamp (&exttime, rtptime);
|
|
gap = gst_util_uint64_scale_int (exttime - priv->clock_base,
|
|
filter->clock_rate, GST_SECOND);
|
|
|
|
/* account for lost packets */
|
|
if (position > gap) {
|
|
GST_DEBUG_OBJECT (filter,
|
|
"Found gap of %" GST_TIME_FORMAT ", adjusting start: %"
|
|
GST_TIME_FORMAT " = %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (gap), GST_TIME_ARGS (position - gap),
|
|
GST_TIME_ARGS (position), GST_TIME_ARGS (gap));
|
|
start = position - gap;
|
|
}
|
|
}
|
|
|
|
/* determining the stop of the segment */
|
|
stop = filter->segment.stop;
|
|
if (priv->npt_stop != -1)
|
|
stop = start + (priv->npt_stop - priv->npt_start);
|
|
|
|
if (position == -1)
|
|
position = start;
|
|
|
|
running_time = gst_segment_to_running_time (&filter->segment,
|
|
GST_FORMAT_TIME, start);
|
|
|
|
gst_segment_init (&segment, GST_FORMAT_TIME);
|
|
segment.rate = priv->play_speed;
|
|
segment.applied_rate = priv->play_scale;
|
|
segment.start = start;
|
|
segment.stop = stop;
|
|
segment.time = priv->npt_start;
|
|
segment.position = position;
|
|
segment.base = running_time;
|
|
|
|
GST_DEBUG_OBJECT (filter, "Creating segment event %" GST_SEGMENT_FORMAT,
|
|
&segment);
|
|
event = gst_event_new_segment (&segment);
|
|
if (filter->priv->segment_seqnum != GST_SEQNUM_INVALID)
|
|
gst_event_set_seqnum (event, filter->priv->segment_seqnum);
|
|
|
|
return event;
|
|
}
|
|
|
|
static gboolean
|
|
set_headers (GstBuffer ** buffer, guint idx, GstRTPBaseDepayload * depayload)
|
|
{
|
|
GstRTPBaseDepayloadPrivate *priv = depayload->priv;
|
|
GstClockTime pts, dts, duration;
|
|
|
|
*buffer = gst_buffer_make_writable (*buffer);
|
|
|
|
pts = GST_BUFFER_PTS (*buffer);
|
|
dts = GST_BUFFER_DTS (*buffer);
|
|
duration = GST_BUFFER_DURATION (*buffer);
|
|
|
|
/* apply last incomming timestamp and duration to outgoing buffer if
|
|
* not otherwise set. */
|
|
if (!GST_CLOCK_TIME_IS_VALID (pts))
|
|
GST_BUFFER_PTS (*buffer) = priv->pts;
|
|
if (!GST_CLOCK_TIME_IS_VALID (dts))
|
|
GST_BUFFER_DTS (*buffer) = priv->dts;
|
|
if (!GST_CLOCK_TIME_IS_VALID (duration))
|
|
GST_BUFFER_DURATION (*buffer) = priv->duration;
|
|
|
|
if (G_UNLIKELY (depayload->priv->discont)) {
|
|
GST_LOG_OBJECT (depayload, "Marking DISCONT on output buffer");
|
|
GST_BUFFER_FLAG_SET (*buffer, GST_BUFFER_FLAG_DISCONT);
|
|
depayload->priv->discont = FALSE;
|
|
}
|
|
|
|
/* make sure we only set the timestamp on the first packet */
|
|
priv->pts = GST_CLOCK_TIME_NONE;
|
|
priv->dts = GST_CLOCK_TIME_NONE;
|
|
priv->duration = GST_CLOCK_TIME_NONE;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_rtp_base_depayload_prepare_push (GstRTPBaseDepayload * filter,
|
|
gboolean is_list, gpointer obj)
|
|
{
|
|
if (is_list) {
|
|
GstBufferList **blist = obj;
|
|
gst_buffer_list_foreach (*blist, (GstBufferListFunc) set_headers, filter);
|
|
} else {
|
|
GstBuffer **buf = obj;
|
|
set_headers (buf, 0, filter);
|
|
}
|
|
|
|
/* if this is the first buffer send a NEWSEGMENT */
|
|
if (G_UNLIKELY (filter->priv->segment_event)) {
|
|
gst_pad_push_event (filter->srcpad, filter->priv->segment_event);
|
|
filter->priv->segment_event = NULL;
|
|
GST_DEBUG_OBJECT (filter, "Pushed newsegment event on this first buffer");
|
|
}
|
|
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
/**
|
|
* gst_rtp_base_depayload_push:
|
|
* @filter: a #GstRTPBaseDepayload
|
|
* @out_buf: a #GstBuffer
|
|
*
|
|
* Push @out_buf to the peer of @filter. This function takes ownership of
|
|
* @out_buf.
|
|
*
|
|
* This function will by default apply the last incomming timestamp on
|
|
* the outgoing buffer when it didn't have a timestamp already.
|
|
*
|
|
* Returns: a #GstFlowReturn.
|
|
*/
|
|
GstFlowReturn
|
|
gst_rtp_base_depayload_push (GstRTPBaseDepayload * filter, GstBuffer * out_buf)
|
|
{
|
|
GstFlowReturn res;
|
|
|
|
res = gst_rtp_base_depayload_prepare_push (filter, FALSE, &out_buf);
|
|
|
|
if (G_LIKELY (res == GST_FLOW_OK))
|
|
res = gst_pad_push (filter->srcpad, out_buf);
|
|
else
|
|
gst_buffer_unref (out_buf);
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_rtp_base_depayload_push_list:
|
|
* @filter: a #GstRTPBaseDepayload
|
|
* @out_list: a #GstBufferList
|
|
*
|
|
* Push @out_list to the peer of @filter. This function takes ownership of
|
|
* @out_list.
|
|
*
|
|
* Returns: a #GstFlowReturn.
|
|
*/
|
|
GstFlowReturn
|
|
gst_rtp_base_depayload_push_list (GstRTPBaseDepayload * filter,
|
|
GstBufferList * out_list)
|
|
{
|
|
GstFlowReturn res;
|
|
|
|
res = gst_rtp_base_depayload_prepare_push (filter, TRUE, &out_list);
|
|
|
|
if (G_LIKELY (res == GST_FLOW_OK))
|
|
res = gst_pad_push_list (filter->srcpad, out_list);
|
|
else
|
|
gst_buffer_list_unref (out_list);
|
|
|
|
return res;
|
|
}
|
|
|
|
/* convert the PacketLost event from a jitterbuffer to a GAP event.
|
|
* subclasses can override this. */
|
|
static gboolean
|
|
gst_rtp_base_depayload_packet_lost (GstRTPBaseDepayload * filter,
|
|
GstEvent * event)
|
|
{
|
|
GstClockTime timestamp, duration;
|
|
GstEvent *sevent;
|
|
const GstStructure *s;
|
|
gboolean might_have_been_fec;
|
|
gboolean res = TRUE;
|
|
|
|
s = gst_event_get_structure (event);
|
|
|
|
/* first start by parsing the timestamp and duration */
|
|
timestamp = -1;
|
|
duration = -1;
|
|
|
|
if (!gst_structure_get_clock_time (s, "timestamp", ×tamp) ||
|
|
!gst_structure_get_clock_time (s, "duration", &duration)) {
|
|
GST_ERROR_OBJECT (filter,
|
|
"Packet loss event without timestamp or duration");
|
|
return FALSE;
|
|
}
|
|
|
|
if (!gst_structure_get_boolean (s, "might-have-been-fec",
|
|
&might_have_been_fec) || !might_have_been_fec) {
|
|
/* send GAP event */
|
|
sevent = gst_event_new_gap (timestamp, duration);
|
|
res = gst_pad_push_event (filter->srcpad, sevent);
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_rtp_base_depayload_change_state (GstElement * element,
|
|
GstStateChange transition)
|
|
{
|
|
GstRTPBaseDepayload *filter;
|
|
GstRTPBaseDepayloadPrivate *priv;
|
|
GstStateChangeReturn ret;
|
|
|
|
filter = GST_RTP_BASE_DEPAYLOAD (element);
|
|
priv = filter->priv;
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_NULL_TO_READY:
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
filter->need_newsegment = TRUE;
|
|
priv->npt_start = 0;
|
|
priv->npt_stop = -1;
|
|
priv->play_speed = 1.0;
|
|
priv->play_scale = 1.0;
|
|
priv->clock_base = -1;
|
|
priv->next_seqnum = -1;
|
|
priv->negotiated = FALSE;
|
|
priv->discont = FALSE;
|
|
priv->segment_seqnum = GST_SEQNUM_INVALID;
|
|
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:
|
|
gst_caps_replace (&priv->last_caps, NULL);
|
|
gst_event_replace (&priv->segment_event, NULL);
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
return ret;
|
|
}
|
|
|
|
static GstStructure *
|
|
gst_rtp_base_depayload_create_stats (GstRTPBaseDepayload * depayload)
|
|
{
|
|
GstRTPBaseDepayloadPrivate *priv;
|
|
GstStructure *s;
|
|
GstClockTime pts = GST_CLOCK_TIME_NONE, dts = GST_CLOCK_TIME_NONE;
|
|
|
|
priv = depayload->priv;
|
|
|
|
GST_OBJECT_LOCK (depayload);
|
|
if (depayload->segment.format != GST_FORMAT_UNDEFINED) {
|
|
pts = gst_segment_to_running_time (&depayload->segment, GST_FORMAT_TIME,
|
|
priv->pts);
|
|
dts = gst_segment_to_running_time (&depayload->segment, GST_FORMAT_TIME,
|
|
priv->dts);
|
|
}
|
|
GST_OBJECT_UNLOCK (depayload);
|
|
|
|
s = gst_structure_new ("application/x-rtp-depayload-stats",
|
|
"clock_rate", G_TYPE_UINT, depayload->clock_rate,
|
|
"npt-start", G_TYPE_UINT64, priv->npt_start,
|
|
"npt-stop", G_TYPE_UINT64, priv->npt_stop,
|
|
"play-speed", G_TYPE_DOUBLE, priv->play_speed,
|
|
"play-scale", G_TYPE_DOUBLE, priv->play_scale,
|
|
"running-time-dts", G_TYPE_UINT64, dts,
|
|
"running-time-pts", G_TYPE_UINT64, pts,
|
|
"seqnum", G_TYPE_UINT, (guint) priv->last_seqnum,
|
|
"timestamp", G_TYPE_UINT, (guint) priv->last_rtptime, NULL);
|
|
|
|
return s;
|
|
}
|
|
|
|
|
|
static void
|
|
gst_rtp_base_depayload_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
switch (prop_id) {
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtp_base_depayload_get_property (GObject * object, guint prop_id,
|
|
GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstRTPBaseDepayload *depayload;
|
|
|
|
depayload = GST_RTP_BASE_DEPAYLOAD (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_STATS:
|
|
g_value_take_boxed (value,
|
|
gst_rtp_base_depayload_create_stats (depayload));
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|