/* GStreamer * Copyright (C) <2005> Philippe Khalaf * Copyright (C) <2005> Nokia Corporation * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ /** * SECTION:gstrtpbasedepayload * @title: GstRTPBaseDepayload * @short_description: Base class for RTP depayloader * * Provides a base class for RTP depayloaders */ #include "gstrtpbasedepayload.h" GST_DEBUG_CATEGORY_STATIC (rtpbasedepayload_debug); #define GST_CAT_DEFAULT (rtpbasedepayload_debug) #define GST_RTP_BASE_DEPAYLOAD_GET_PRIVATE(obj) \ (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTP_BASE_DEPAYLOAD, GstRTPBaseDepayloadPrivate)) struct _GstRTPBaseDepayloadPrivate { GstClockTime npt_start; GstClockTime npt_stop; gdouble play_speed; gdouble play_scale; guint clock_base; gboolean discont; GstClockTime pts; GstClockTime dts; GstClockTime duration; guint32 last_ssrc; guint32 last_seqnum; guint32 last_rtptime; guint32 next_seqnum; gboolean negotiated; GstCaps *last_caps; GstEvent *segment_event; }; /* Filter signals and args */ enum { /* FILL ME */ LAST_SIGNAL }; enum { PROP_0, PROP_STATS, PROP_LAST }; static void gst_rtp_base_depayload_finalize (GObject * object); static void gst_rtp_base_depayload_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_rtp_base_depayload_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static GstFlowReturn gst_rtp_base_depayload_chain (GstPad * pad, GstObject * parent, GstBuffer * in); static GstFlowReturn gst_rtp_base_depayload_chain_list (GstPad * pad, GstObject * parent, GstBufferList * list); static gboolean gst_rtp_base_depayload_handle_sink_event (GstPad * pad, GstObject * parent, GstEvent * event); static GstStateChangeReturn gst_rtp_base_depayload_change_state (GstElement * element, GstStateChange transition); static gboolean gst_rtp_base_depayload_packet_lost (GstRTPBaseDepayload * filter, GstEvent * event); static gboolean gst_rtp_base_depayload_handle_event (GstRTPBaseDepayload * filter, GstEvent * event); static GstElementClass *parent_class = NULL; static void gst_rtp_base_depayload_class_init (GstRTPBaseDepayloadClass * klass); static void gst_rtp_base_depayload_init (GstRTPBaseDepayload * rtpbasepayload, GstRTPBaseDepayloadClass * klass); static GstEvent *create_segment_event (GstRTPBaseDepayload * filter, guint rtptime, GstClockTime position); GType gst_rtp_base_depayload_get_type (void) { static GType rtp_base_depayload_type = 0; if (g_once_init_enter ((gsize *) & rtp_base_depayload_type)) { static const GTypeInfo rtp_base_depayload_info = { sizeof (GstRTPBaseDepayloadClass), NULL, NULL, (GClassInitFunc) gst_rtp_base_depayload_class_init, NULL, NULL, sizeof (GstRTPBaseDepayload), 0, (GInstanceInitFunc) gst_rtp_base_depayload_init, }; g_once_init_leave ((gsize *) & rtp_base_depayload_type, g_type_register_static (GST_TYPE_ELEMENT, "GstRTPBaseDepayload", &rtp_base_depayload_info, G_TYPE_FLAG_ABSTRACT)); } return rtp_base_depayload_type; } static void gst_rtp_base_depayload_class_init (GstRTPBaseDepayloadClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; gobject_class = G_OBJECT_CLASS (klass); gstelement_class = (GstElementClass *) klass; parent_class = g_type_class_peek_parent (klass); g_type_class_add_private (klass, sizeof (GstRTPBaseDepayloadPrivate)); gobject_class->finalize = gst_rtp_base_depayload_finalize; gobject_class->set_property = gst_rtp_base_depayload_set_property; gobject_class->get_property = gst_rtp_base_depayload_get_property; /** * GstRTPBaseDepayload:stats: * * Various depayloader statistics retrieved atomically (and are therefore * synchroized with each other). This property return a GstStructure named * application/x-rtp-depayload-stats containing the following fields relating to * the last processed buffer and current state of the stream being depayloaded: * * * `clock-rate`: #G_TYPE_UINT, clock-rate of the stream * * `npt-start`: #G_TYPE_UINT64, time of playback start * * `npt-stop`: #G_TYPE_UINT64, time of playback stop * * `play-speed`: #G_TYPE_DOUBLE, the playback speed * * `play-scale`: #G_TYPE_DOUBLE, the playback scale * * `running-time-dts`: #G_TYPE_UINT64, the last running-time of the * last DTS * * `running-time-pts`: #G_TYPE_UINT64, the last running-time of the * last PTS * * `seqnum`: #G_TYPE_UINT, the last seen seqnum * * `timestamp`: #G_TYPE_UINT, the last seen RTP timestamp **/ g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_STATS, g_param_spec_boxed ("stats", "Statistics", "Various statistics", GST_TYPE_STRUCTURE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); gstelement_class->change_state = gst_rtp_base_depayload_change_state; klass->packet_lost = gst_rtp_base_depayload_packet_lost; klass->handle_event = gst_rtp_base_depayload_handle_event; GST_DEBUG_CATEGORY_INIT (rtpbasedepayload_debug, "rtpbasedepayload", 0, "Base class for RTP Depayloaders"); } static void gst_rtp_base_depayload_init (GstRTPBaseDepayload * filter, GstRTPBaseDepayloadClass * klass) { GstPadTemplate *pad_template; GstRTPBaseDepayloadPrivate *priv; priv = GST_RTP_BASE_DEPAYLOAD_GET_PRIVATE (filter); filter->priv = priv; GST_DEBUG_OBJECT (filter, "init"); pad_template = gst_element_class_get_pad_template (GST_ELEMENT_CLASS (klass), "sink"); g_return_if_fail (pad_template != NULL); filter->sinkpad = gst_pad_new_from_template (pad_template, "sink"); gst_pad_set_chain_function (filter->sinkpad, gst_rtp_base_depayload_chain); gst_pad_set_chain_list_function (filter->sinkpad, gst_rtp_base_depayload_chain_list); gst_pad_set_event_function (filter->sinkpad, gst_rtp_base_depayload_handle_sink_event); gst_element_add_pad (GST_ELEMENT (filter), filter->sinkpad); pad_template = gst_element_class_get_pad_template (GST_ELEMENT_CLASS (klass), "src"); g_return_if_fail (pad_template != NULL); filter->srcpad = gst_pad_new_from_template (pad_template, "src"); gst_pad_use_fixed_caps (filter->srcpad); gst_element_add_pad (GST_ELEMENT (filter), filter->srcpad); priv->npt_start = 0; priv->npt_stop = -1; priv->play_speed = 1.0; priv->play_scale = 1.0; priv->clock_base = -1; priv->dts = -1; priv->pts = -1; priv->duration = -1; gst_segment_init (&filter->segment, GST_FORMAT_UNDEFINED); } static void gst_rtp_base_depayload_finalize (GObject * object) { G_OBJECT_CLASS (parent_class)->finalize (object); } static gboolean gst_rtp_base_depayload_setcaps (GstRTPBaseDepayload * filter, GstCaps * caps) { GstRTPBaseDepayloadClass *bclass; GstRTPBaseDepayloadPrivate *priv; gboolean res; GstStructure *caps_struct; const GValue *value; priv = filter->priv; bclass = GST_RTP_BASE_DEPAYLOAD_GET_CLASS (filter); GST_DEBUG_OBJECT (filter, "Set caps %" GST_PTR_FORMAT, caps); if (priv->last_caps) { if (gst_caps_is_equal (priv->last_caps, caps)) { res = TRUE; goto caps_not_changed; } else { gst_caps_unref (priv->last_caps); priv->last_caps = NULL; } } caps_struct = gst_caps_get_structure (caps, 0); /* get other values for newsegment */ value = gst_structure_get_value (caps_struct, "npt-start"); if (value && G_VALUE_HOLDS_UINT64 (value)) priv->npt_start = g_value_get_uint64 (value); else priv->npt_start = 0; GST_DEBUG_OBJECT (filter, "NPT start %" G_GUINT64_FORMAT, priv->npt_start); value = gst_structure_get_value (caps_struct, "npt-stop"); if (value && G_VALUE_HOLDS_UINT64 (value)) priv->npt_stop = g_value_get_uint64 (value); else priv->npt_stop = -1; GST_DEBUG_OBJECT (filter, "NPT stop %" G_GUINT64_FORMAT, priv->npt_stop); value = gst_structure_get_value (caps_struct, "play-speed"); if (value && G_VALUE_HOLDS_DOUBLE (value)) priv->play_speed = g_value_get_double (value); else priv->play_speed = 1.0; value = gst_structure_get_value (caps_struct, "play-scale"); if (value && G_VALUE_HOLDS_DOUBLE (value)) priv->play_scale = g_value_get_double (value); else priv->play_scale = 1.0; value = gst_structure_get_value (caps_struct, "clock-base"); if (value && G_VALUE_HOLDS_UINT (value)) priv->clock_base = g_value_get_uint (value); else priv->clock_base = -1; if (bclass->set_caps) { res = bclass->set_caps (filter, caps); if (!res) { GST_WARNING_OBJECT (filter, "Subclass rejected caps %" GST_PTR_FORMAT, caps); } } else { res = TRUE; } priv->negotiated = res; if (priv->negotiated) priv->last_caps = gst_caps_ref (caps); return res; caps_not_changed: { GST_DEBUG_OBJECT (filter, "Caps did not change"); return res; } } /* takes ownership of the input buffer */ static GstFlowReturn gst_rtp_base_depayload_handle_buffer (GstRTPBaseDepayload * filter, GstRTPBaseDepayloadClass * bclass, GstBuffer * in) { GstBuffer *(*process_rtp_packet_func) (GstRTPBaseDepayload * base, GstRTPBuffer * rtp_buffer); GstBuffer *(*process_func) (GstRTPBaseDepayload * base, GstBuffer * in); GstRTPBaseDepayloadPrivate *priv; GstFlowReturn ret = GST_FLOW_OK; GstBuffer *out_buf; guint32 ssrc; guint16 seqnum; guint32 rtptime; gboolean discont, buf_discont; gint gap; GstRTPBuffer rtp = { NULL }; priv = filter->priv; process_func = bclass->process; process_rtp_packet_func = bclass->process_rtp_packet; /* we must have a setcaps first */ if (G_UNLIKELY (!priv->negotiated)) goto not_negotiated; if (G_UNLIKELY (!gst_rtp_buffer_map (in, GST_MAP_READ, &rtp))) goto invalid_buffer; buf_discont = GST_BUFFER_IS_DISCONT (in); priv->pts = GST_BUFFER_PTS (in); priv->dts = GST_BUFFER_DTS (in); priv->duration = GST_BUFFER_DURATION (in); ssrc = gst_rtp_buffer_get_ssrc (&rtp); seqnum = gst_rtp_buffer_get_seq (&rtp); rtptime = gst_rtp_buffer_get_timestamp (&rtp); priv->last_seqnum = seqnum; priv->last_rtptime = rtptime; discont = buf_discont; GST_LOG_OBJECT (filter, "discont %d, seqnum %u, rtptime %u, pts %" GST_TIME_FORMAT ", dts %" GST_TIME_FORMAT, buf_discont, seqnum, rtptime, GST_TIME_ARGS (priv->pts), GST_TIME_ARGS (priv->dts)); /* Check seqnum. This is a very simple check that makes sure that the seqnums * are strictly increasing, dropping anything that is out of the ordinary. We * can only do this when the next_seqnum is known. */ if (G_LIKELY (priv->next_seqnum != -1)) { if (ssrc != priv->last_ssrc) { GST_LOG_OBJECT (filter, "New ssrc %u (current ssrc %u), sender restarted", ssrc, priv->last_ssrc); discont = TRUE; } else { gap = gst_rtp_buffer_compare_seqnum (seqnum, priv->next_seqnum); /* if we have no gap, all is fine */ if (G_UNLIKELY (gap != 0)) { GST_LOG_OBJECT (filter, "got packet %u, expected %u, gap %d", seqnum, priv->next_seqnum, gap); if (gap < 0) { /* seqnum > next_seqnum, we are missing some packets, this is always a * DISCONT. */ GST_LOG_OBJECT (filter, "%d missing packets", gap); discont = TRUE; } else { /* seqnum < next_seqnum, we have seen this packet before or the sender * could be restarted. If the packet is not too old, we throw it away as * a duplicate, otherwise we mark discont and continue. 100 misordered * packets is a good threshold. See also RFC 4737. */ if (gap < 100) goto dropping; GST_LOG_OBJECT (filter, "%d > 100, packet too old, sender likely restarted", gap); discont = TRUE; } } } } priv->next_seqnum = (seqnum + 1) & 0xffff; priv->last_ssrc = ssrc; if (G_UNLIKELY (discont)) { priv->discont = TRUE; if (!buf_discont) { gpointer old_inbuf = in; /* we detected a seqnum discont but the buffer was not flagged with a discont, * set the discont flag so that the subclass can throw away old data. */ GST_LOG_OBJECT (filter, "mark DISCONT on input buffer"); in = gst_buffer_make_writable (in); GST_BUFFER_FLAG_SET (in, GST_BUFFER_FLAG_DISCONT); /* depayloaders will check flag on rtpbuffer->buffer, so if the input * buffer was not writable already we need to remap to make our * newly-flagged buffer current on the rtpbuffer */ if (in != old_inbuf) { gst_rtp_buffer_unmap (&rtp); if (G_UNLIKELY (!gst_rtp_buffer_map (in, GST_MAP_READ, &rtp))) goto invalid_buffer; } } } /* prepare segment event if needed */ if (filter->need_newsegment) { priv->segment_event = create_segment_event (filter, rtptime, GST_BUFFER_PTS (in)); filter->need_newsegment = FALSE; } if (process_rtp_packet_func != NULL) { out_buf = process_rtp_packet_func (filter, &rtp); gst_rtp_buffer_unmap (&rtp); } else if (process_func != NULL) { gst_rtp_buffer_unmap (&rtp); out_buf = process_func (filter, in); } else { goto no_process; } /* let's send it out to processing */ if (out_buf) { ret = gst_rtp_base_depayload_push (filter, out_buf); } gst_buffer_unref (in); return ret; /* 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->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); 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; 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; } /* send GAP event */ sevent = gst_event_new_gap (timestamp, duration); return gst_pad_push_event (filter->srcpad, sevent); } 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; 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; } }