LAST_SIGNAL
};
-GST_BOILERPLATE (GstRtpPtDemux, gst_rtp_pt_demux, GstElement, GST_TYPE_ELEMENT);
+#define gst_rtp_pt_demux_parent_class parent_class
+G_DEFINE_TYPE (GstRtpPtDemux, gst_rtp_pt_demux, GST_TYPE_ELEMENT);
static void gst_rtp_pt_demux_finalize (GObject * object);
static guint gst_rtp_pt_demux_signals[LAST_SIGNAL] = { 0 };
static void
-gst_rtp_pt_demux_base_init (gpointer g_class)
-{
- GstElementClass *gstelement_klass = GST_ELEMENT_CLASS (g_class);
-
- gst_element_class_add_pad_template (gstelement_klass,
- gst_static_pad_template_get (&rtp_pt_demux_sink_template));
- gst_element_class_add_pad_template (gstelement_klass,
- gst_static_pad_template_get (&rtp_pt_demux_src_template));
-
- gst_element_class_set_details_simple (gstelement_klass, "RTP Demux",
- "Demux/Network/RTP",
- "Parses codec streams transmitted in the same RTP session",
- "Kai Vehmanen <kai.vehmanen@nokia.com>");
-}
-
-static void
gst_rtp_pt_demux_class_init (GstRtpPtDemuxClass * klass)
{
GObjectClass *gobject_klass;
klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_pt_demux_clear_pt_map);
+ gst_element_class_add_pad_template (gstelement_klass,
+ gst_static_pad_template_get (&rtp_pt_demux_sink_template));
+ gst_element_class_add_pad_template (gstelement_klass,
+ gst_static_pad_template_get (&rtp_pt_demux_src_template));
+
+ gst_element_class_set_details_simple (gstelement_klass, "RTP Demux",
+ "Demux/Network/RTP",
+ "Parses codec streams transmitted in the same RTP session",
+ "Kai Vehmanen <kai.vehmanen@nokia.com>");
+
GST_DEBUG_CATEGORY_INIT (gst_rtp_pt_demux_debug,
"rtpptdemux", 0, "RTP codec demuxer");
}
static void
-gst_rtp_pt_demux_init (GstRtpPtDemux * ptdemux, GstRtpPtDemuxClass * g_class)
+gst_rtp_pt_demux_init (GstRtpPtDemux * ptdemux)
{
GstElementClass *klass = GST_ELEMENT_GET_CLASS (ptdemux);
caps = g_value_dup_boxed (&ret);
g_value_unset (&ret);
if (caps == NULL) {
- caps = GST_PAD_CAPS (rtpdemux->sink);
- if (caps)
- gst_caps_ref (caps);
+ caps = gst_pad_get_current_caps (rtpdemux->sink);
}
GST_DEBUG ("pt %d, got caps %" GST_PTR_FORMAT, pt, caps);
GstPad *srcpad;
GstRtpPtDemuxPad *rtpdemuxpad;
GstCaps *caps;
+ GstRTPBuffer rtp;
rtpdemux = GST_RTP_PT_DEMUX (GST_OBJECT_PARENT (pad));
if (!gst_rtp_buffer_validate (buf))
goto invalid_buffer;
- pt = gst_rtp_buffer_get_payload_type (buf);
+ gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
+ pt = gst_rtp_buffer_get_payload_type (&rtp);
+ gst_rtp_buffer_unmap (&rtp);
GST_DEBUG_OBJECT (rtpdemux, "received buffer for pt %d", pt);
rtpdemuxpad->newcaps = FALSE;
}
- gst_buffer_set_caps (buf, GST_PAD_CAPS (srcpad));
-
/* push to srcpad */
ret = gst_pad_push (srcpad, buf);
GstRtpPtDemuxPad *dpad = (GstRtpPtDemuxPad *) walk->data;
if (dpad->pad == pad) {
+ GstStructure *ws;
+
event =
GST_EVENT_CAST (gst_mini_object_make_writable
(GST_MINI_OBJECT_CAST (event)));
- gst_structure_set (event->structure,
- "payload", G_TYPE_UINT, dpad->pt, NULL);
+ ws = gst_event_writable_structure (event);
+ gst_structure_set (ws, "payload", G_TYPE_UINT, dpad->pt, NULL);
break;
}
}
static GstStateChangeReturn gst_rtp_session_change_state (GstElement * element,
GstStateChange transition);
static GstPad *gst_rtp_session_request_new_pad (GstElement * element,
- GstPadTemplate * templ, const gchar * name);
+ GstPadTemplate * templ, const gchar * name, const GstCaps * caps);
static void gst_rtp_session_release_pad (GstElement * element, GstPad * pad);
+static gboolean gst_rtp_session_sink_setcaps (GstPad * pad, GstCaps * caps);
+static gboolean gst_rtp_session_setcaps_send_rtp (GstPad * pad, GstCaps * caps);
+
static void gst_rtp_session_clear_pt_map (GstRtpSession * rtpsession);
static guint gst_rtp_session_signals[LAST_SIGNAL] = { 0 };
src->ssrc);
}
-GST_BOILERPLATE (GstRtpSession, gst_rtp_session, GstElement, GST_TYPE_ELEMENT);
-
-static void
-gst_rtp_session_base_init (gpointer klass)
-{
- GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
-
- /* sink pads */
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&rtpsession_recv_rtp_sink_template));
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&rtpsession_recv_rtcp_sink_template));
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&rtpsession_send_rtp_sink_template));
-
- /* src pads */
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&rtpsession_recv_rtp_src_template));
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&rtpsession_sync_src_template));
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&rtpsession_send_rtp_src_template));
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&rtpsession_send_rtcp_src_template));
-
- gst_element_class_set_details_simple (element_class, "RTP Session",
- "Filter/Network/RTP",
- "Implement an RTP session", "Wim Taymans <wim.taymans@gmail.com>");
-}
+#define gst_rtp_session_parent_class parent_class
+G_DEFINE_TYPE (GstRtpSession, gst_rtp_session, GST_TYPE_ELEMENT);
static void
gst_rtp_session_class_init (GstRtpSessionClass * klass)
klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_session_clear_pt_map);
+ /* sink pads */
+ gst_element_class_add_pad_template (gstelement_class,
+ gst_static_pad_template_get (&rtpsession_recv_rtp_sink_template));
+ gst_element_class_add_pad_template (gstelement_class,
+ gst_static_pad_template_get (&rtpsession_recv_rtcp_sink_template));
+ gst_element_class_add_pad_template (gstelement_class,
+ gst_static_pad_template_get (&rtpsession_send_rtp_sink_template));
+
+ /* src pads */
+ gst_element_class_add_pad_template (gstelement_class,
+ gst_static_pad_template_get (&rtpsession_recv_rtp_src_template));
+ gst_element_class_add_pad_template (gstelement_class,
+ gst_static_pad_template_get (&rtpsession_sync_src_template));
+ gst_element_class_add_pad_template (gstelement_class,
+ gst_static_pad_template_get (&rtpsession_send_rtp_src_template));
+ gst_element_class_add_pad_template (gstelement_class,
+ gst_static_pad_template_get (&rtpsession_send_rtcp_src_template));
+
+ gst_element_class_set_details_simple (gstelement_class, "RTP Session",
+ "Filter/Network/RTP",
+ "Implement an RTP session", "Wim Taymans <wim.taymans@gmail.com>");
+
GST_DEBUG_CATEGORY_INIT (gst_rtp_session_debug,
"rtpsession", 0, "RTP Session");
}
static void
-gst_rtp_session_init (GstRtpSession * rtpsession, GstRtpSessionClass * klass)
+gst_rtp_session_init (GstRtpSession * rtpsession)
{
rtpsession->priv = GST_RTP_SESSION_GET_PRIVATE (rtpsession);
rtpsession->priv->lock = g_mutex_new ();
break;
}
- res = parent_class->change_state (element, transition);
+ res = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
GstCaps *caps;
/* set rtcp caps on output pad */
- if (!(caps = GST_PAD_CAPS (rtcp_src))) {
+ if (!(caps = gst_pad_get_current_caps (rtcp_src))) {
caps = gst_caps_new_simple ("application/x-rtcp", NULL);
gst_pad_set_caps (rtcp_src, caps);
- } else
- gst_caps_ref (caps);
- gst_buffer_set_caps (buffer, caps);
+ }
gst_caps_unref (caps);
gst_object_ref (rtcp_src);
GstCaps *caps;
/* set rtcp caps on output pad */
- if (!(caps = GST_PAD_CAPS (sync_src))) {
+ if (!(caps = gst_pad_get_current_caps (sync_src))) {
caps = gst_caps_new_simple ("application/x-rtcp", NULL);
gst_pad_set_caps (sync_src, caps);
- } else
- gst_caps_ref (caps);
- gst_buffer_set_caps (buffer, caps);
+ }
gst_caps_unref (caps);
gst_object_ref (sync_src);
GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
+ case GST_EVENT_CAPS:
+ {
+ GstCaps *caps;
+
+ /* process */
+ gst_event_parse_caps (event, &caps);
+ gst_rtp_session_sink_setcaps (pad, caps);
+ /* and eat */
+ gst_event_unref (event);
+ break;
+ }
case GST_EVENT_FLUSH_STOP:
gst_segment_init (&rtpsession->recv_rtp_seg, GST_FORMAT_UNDEFINED);
ret = gst_pad_push_event (rtpsession->recv_rtp_src, event);
break;
- case GST_EVENT_NEWSEGMENT:
+ case GST_EVENT_SEGMENT:
{
- gboolean update;
- gdouble rate, arate;
- GstFormat format;
- gint64 start, stop, time;
- GstSegment *segment;
+ GstSegment *segment, in_segment;
segment = &rtpsession->recv_rtp_seg;
/* the newsegment event is needed to convert the RTP timestamp to
* running_time, which is needed to generate a mapping from RTP to NTP
* timestamps in SR reports */
- gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
- &start, &stop, &time);
-
- GST_DEBUG_OBJECT (rtpsession,
- "configured NEWSEGMENT update %d, rate %lf, applied rate %lf, "
- "format GST_FORMAT_TIME, "
- "%" GST_TIME_FORMAT " -- %" GST_TIME_FORMAT
- ", time %" GST_TIME_FORMAT ", accum %" GST_TIME_FORMAT,
- update, rate, arate, GST_TIME_ARGS (segment->start),
- GST_TIME_ARGS (segment->stop), GST_TIME_ARGS (segment->time),
- GST_TIME_ARGS (segment->accum));
+ gst_event_copy_segment (event, &in_segment);
+ GST_DEBUG_OBJECT (rtpsession, "received segment %" GST_SEGMENT_FORMAT,
+ in_segment);
- gst_segment_set_newsegment_full (segment, update, rate,
- arate, format, start, stop, time);
+ /* accept upstream */
+ gst_segment_copy_into (&in_segment, segment);
/* push event forward */
ret = gst_pad_push_event (rtpsession->recv_rtp_src, event);
GST_RTP_SESSION_UNLOCK (rtpsession);
if (otherpad) {
- it = gst_iterator_new_single (GST_TYPE_PAD, otherpad,
- (GstCopyFunction) gst_object_ref, (GFreeFunc) gst_object_unref);
+ GValue val = { 0, };
+
+ g_value_init (&val, GST_TYPE_PAD);
+ g_value_set_object (&val, otherpad);
+ it = gst_iterator_new_single (GST_TYPE_PAD, &val);
+ g_value_unset (&val);
gst_object_unref (otherpad);
}
GST_DEBUG_OBJECT (rtpsession, "received event");
switch (GST_EVENT_TYPE (event)) {
+ case GST_EVENT_CAPS:
+ {
+ GstCaps *caps;
+
+ /* process */
+ gst_event_parse_caps (event, &caps);
+ gst_rtp_session_setcaps_send_rtp (pad, caps);
+ /* and eat */
+ gst_event_unref (event);
+ break;
+ }
case GST_EVENT_FLUSH_STOP:
gst_segment_init (&rtpsession->send_rtp_seg, GST_FORMAT_UNDEFINED);
ret = gst_pad_push_event (rtpsession->send_rtp_src, event);
break;
- case GST_EVENT_NEWSEGMENT:{
- gboolean update;
- gdouble rate, arate;
- GstFormat format;
- gint64 start, stop, time;
- GstSegment *segment;
+ case GST_EVENT_SEGMENT:{
+ GstSegment *segment, in_segment;
segment = &rtpsession->send_rtp_seg;
/* the newsegment event is needed to convert the RTP timestamp to
* running_time, which is needed to generate a mapping from RTP to NTP
* timestamps in SR reports */
- gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
- &start, &stop, &time);
-
- GST_DEBUG_OBJECT (rtpsession,
- "configured NEWSEGMENT update %d, rate %lf, applied rate %lf, "
- "format GST_FORMAT_TIME, "
- "%" GST_TIME_FORMAT " -- %" GST_TIME_FORMAT
- ", time %" GST_TIME_FORMAT ", accum %" GST_TIME_FORMAT,
- update, rate, arate, GST_TIME_ARGS (segment->start),
- GST_TIME_ARGS (segment->stop), GST_TIME_ARGS (segment->time),
- GST_TIME_ARGS (segment->accum));
+ gst_event_copy_segment (event, &in_segment);
+ GST_DEBUG_OBJECT (rtpsession, "received segment %" GST_SEGMENT_FORMAT,
+ in_segment);
- gst_segment_set_newsegment_full (segment, update, rate,
- arate, format, start, stop, time);
+ /* accept upstream */
+ gst_segment_copy_into (&in_segment, segment);
/* push event forward */
ret = gst_pad_push_event (rtpsession->send_rtp_src, event);
}
static GstCaps *
-gst_rtp_session_getcaps_send_rtp (GstPad * pad)
+gst_rtp_session_getcaps_send_rtp (GstPad * pad, GstCaps * filter)
{
GstRtpSession *rtpsession;
GstRtpSessionPrivate *priv;
result = gst_caps_new_full (s1, s2, NULL);
+ if (filter) {
+ GstCaps *caps = result;
+
+ result = gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
+ gst_caps_unref (caps);
+ }
+
GST_DEBUG_OBJECT (rtpsession, "getting caps %" GST_PTR_FORMAT, result);
gst_object_unref (rtpsession);
/* All groups in an list have the same timestamp.
* So, just take it from the first group. */
- buffer = gst_buffer_list_get (GST_BUFFER_LIST_CAST (data), 0, 0);
+ buffer = gst_buffer_list_get (GST_BUFFER_LIST_CAST (data), 0);
if (buffer)
timestamp = GST_BUFFER_TIMESTAMP (buffer);
else
gst_rtp_session_chain_recv_rtp);
gst_pad_set_event_function (rtpsession->recv_rtp_sink,
(GstPadEventFunction) gst_rtp_session_event_recv_rtp_sink);
- gst_pad_set_setcaps_function (rtpsession->recv_rtp_sink,
- gst_rtp_session_sink_setcaps);
gst_pad_set_iterate_internal_links_function (rtpsession->recv_rtp_sink,
gst_rtp_session_iterate_internal_links);
gst_pad_set_active (rtpsession->recv_rtp_sink, TRUE);
gst_rtp_session_chain_send_rtp_list);
gst_pad_set_getcaps_function (rtpsession->send_rtp_sink,
gst_rtp_session_getcaps_send_rtp);
- gst_pad_set_setcaps_function (rtpsession->send_rtp_sink,
- gst_rtp_session_setcaps_send_rtp);
gst_pad_set_event_function (rtpsession->send_rtp_sink,
(GstPadEventFunction) gst_rtp_session_event_send_rtp_sink);
gst_pad_set_iterate_internal_links_function (rtpsession->send_rtp_sink,
static GstPad *
gst_rtp_session_request_new_pad (GstElement * element,
- GstPadTemplate * templ, const gchar * name)
+ GstPadTemplate * templ, const gchar * name, const GstCaps * caps)
{
GstRtpSession *rtpsession;
GstElementClass *klass;
LAST_SIGNAL
};
-GST_BOILERPLATE (GstRtpSsrcDemux, gst_rtp_ssrc_demux, GstElement,
- GST_TYPE_ELEMENT);
-
+#define gst_rtp_ssrc_demux_parent_class parent_class
+G_DEFINE_TYPE (GstRtpSsrcDemux, gst_rtp_ssrc_demux, GST_TYPE_ELEMENT);
/* GObject vmethods */
static void gst_rtp_ssrc_demux_dispose (GObject * object);
GstPadTemplate *templ;
gchar *padname;
GstRtpSsrcDemuxPad *demuxpad;
+ GstCaps *caps;
GST_DEBUG_OBJECT (demux, "creating pad for SSRC %08x", ssrc);
demux->srcpads = g_slist_prepend (demux->srcpads, demuxpad);
/* copy caps from input */
- gst_pad_set_caps (rtp_pad, GST_PAD_CAPS (demux->rtp_sink));
+ caps = gst_pad_get_current_caps (demux->rtp_sink);
+ gst_pad_set_caps (rtp_pad, caps);
+ gst_caps_unref (caps);
gst_pad_use_fixed_caps (rtp_pad);
- gst_pad_set_caps (rtcp_pad, GST_PAD_CAPS (demux->rtcp_sink));
+ caps = gst_pad_get_current_caps (demux->rtcp_sink);
+ gst_pad_set_caps (rtcp_pad, caps);
+ gst_caps_unref (caps);
gst_pad_use_fixed_caps (rtcp_pad);
gst_pad_set_event_function (rtp_pad, gst_rtp_ssrc_demux_src_event);
}
static void
-gst_rtp_ssrc_demux_base_init (gpointer g_class)
-{
- GstElementClass *gstelement_klass = GST_ELEMENT_CLASS (g_class);
-
- gst_element_class_add_pad_template (gstelement_klass,
- gst_static_pad_template_get (&rtp_ssrc_demux_sink_template));
- gst_element_class_add_pad_template (gstelement_klass,
- gst_static_pad_template_get (&rtp_ssrc_demux_rtcp_sink_template));
- gst_element_class_add_pad_template (gstelement_klass,
- gst_static_pad_template_get (&rtp_ssrc_demux_src_template));
- gst_element_class_add_pad_template (gstelement_klass,
- gst_static_pad_template_get (&rtp_ssrc_demux_rtcp_src_template));
-
- gst_element_class_set_details_simple (gstelement_klass, "RTP SSRC Demux",
- "Demux/Network/RTP",
- "Splits RTP streams based on the SSRC",
- "Wim Taymans <wim.taymans@gmail.com>");
-}
-
-static void
gst_rtp_ssrc_demux_class_init (GstRtpSsrcDemuxClass * klass)
{
GObjectClass *gobject_klass;
gstrtpssrcdemux_klass->clear_ssrc =
GST_DEBUG_FUNCPTR (gst_rtp_ssrc_demux_clear_ssrc);
+ gst_element_class_add_pad_template (gstelement_klass,
+ gst_static_pad_template_get (&rtp_ssrc_demux_sink_template));
+ gst_element_class_add_pad_template (gstelement_klass,
+ gst_static_pad_template_get (&rtp_ssrc_demux_rtcp_sink_template));
+ gst_element_class_add_pad_template (gstelement_klass,
+ gst_static_pad_template_get (&rtp_ssrc_demux_src_template));
+ gst_element_class_add_pad_template (gstelement_klass,
+ gst_static_pad_template_get (&rtp_ssrc_demux_rtcp_src_template));
+
+ gst_element_class_set_details_simple (gstelement_klass, "RTP SSRC Demux",
+ "Demux/Network/RTP",
+ "Splits RTP streams based on the SSRC",
+ "Wim Taymans <wim.taymans@gmail.com>");
+
GST_DEBUG_CATEGORY_INIT (gst_rtp_ssrc_demux_debug,
"rtpssrcdemux", 0, "RTP SSRC demuxer");
}
static void
-gst_rtp_ssrc_demux_init (GstRtpSsrcDemux * demux,
- GstRtpSsrcDemuxClass * g_class)
+gst_rtp_ssrc_demux_init (GstRtpSsrcDemux * demux)
{
GstElementClass *klass = GST_ELEMENT_GET_CLASS (demux);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_STOP:
gst_segment_init (&demux->segment, GST_FORMAT_UNDEFINED);
- case GST_EVENT_NEWSEGMENT:
default:
{
GSList *walk;
demux = GST_RTP_SSRC_DEMUX (gst_pad_get_parent (pad));
switch (GST_EVENT_TYPE (event)) {
- case GST_EVENT_NEWSEGMENT:
default:
{
GSList *walk;
GstRtpSsrcDemux *demux;
guint32 ssrc;
GstRtpSsrcDemuxPad *dpad;
+ GstRTPBuffer rtp;
demux = GST_RTP_SSRC_DEMUX (GST_OBJECT_PARENT (pad));
if (!gst_rtp_buffer_validate (buf))
goto invalid_payload;
- ssrc = gst_rtp_buffer_get_ssrc (buf);
+ gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
+ ssrc = gst_rtp_buffer_get_ssrc (&rtp);
+ gst_rtp_buffer_unmap (&rtp);
GST_DEBUG_OBJECT (demux, "received buffer of SSRC %08x", ssrc);
guint32 ssrc;
GstRtpSsrcDemuxPad *dpad;
GstRTCPPacket packet;
+ GstRTCPBuffer rtcp;
demux = GST_RTP_SSRC_DEMUX (GST_OBJECT_PARENT (pad));
if (!gst_rtcp_buffer_validate (buf))
goto invalid_rtcp;
- if (!gst_rtcp_buffer_get_first_packet (buf, &packet))
+ gst_rtcp_buffer_map (buf, GST_MAP_READ, &rtcp);
+ if (!gst_rtcp_buffer_get_first_packet (&rtcp, &packet)) {
+ gst_rtcp_buffer_unmap (&rtcp);
goto invalid_rtcp;
+ }
/* first packet must be SR or RR or else the validate would have failed */
switch (gst_rtcp_packet_get_type (&packet)) {
default:
goto unexpected_rtcp;
}
+ gst_rtcp_buffer_unmap (&rtcp);
GST_DEBUG_OBJECT (demux, "received RTCP of SSRC %08x", ssrc);
GstRtpSsrcDemuxPad *dpad = (GstRtpSsrcDemuxPad *) walk->data;
if (dpad->rtp_pad == pad || dpad->rtcp_pad == pad) {
+ GstStructure *ws;
+
event =
GST_EVENT_CAST (gst_mini_object_make_writable
(GST_MINI_OBJECT_CAST (event)));
- gst_structure_set (event->structure, "ssrc", G_TYPE_UINT,
- dpad->ssrc, NULL);
+ ws = gst_event_writable_structure (event);
+ gst_structure_set (ws, "ssrc", G_TYPE_UINT, dpad->ssrc, NULL);
break;
}
}
break;
}
}
- it = gst_iterator_new_single (GST_TYPE_PAD, otherpad,
- (GstCopyFunction) gst_object_ref, (GFreeFunc) gst_object_unref);
+ if (otherpad) {
+ GValue val = { 0, };
+
+ g_value_init (&val, GST_TYPE_PAD);
+ g_value_set_object (&val, otherpad);
+ it = gst_iterator_new_single (GST_TYPE_PAD, &val);
+ g_value_unset (&val);
+ gst_object_unref (otherpad);
+ }
GST_PAD_UNLOCK (demux);
gst_object_unref (demux);
GList *list;
guint32 rtptime;
guint16 seqnum;
+ GstRTPBuffer rtp;
g_return_val_if_fail (jbuf != NULL, FALSE);
g_return_val_if_fail (buf != NULL, FALSE);
- seqnum = gst_rtp_buffer_get_seq (buf);
+ gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
+
+ seqnum = gst_rtp_buffer_get_seq (&rtp);
/* loop the list to skip strictly smaller seqnum buffers */
for (list = jbuf->packets->head; list; list = g_list_next (list)) {
guint16 qseq;
gint gap;
+ GstRTPBuffer rtpb;
- qseq = gst_rtp_buffer_get_seq (GST_BUFFER_CAST (list->data));
+ gst_rtp_buffer_map (GST_BUFFER_CAST (list->data), GST_MAP_READ, &rtpb);
+ qseq = gst_rtp_buffer_get_seq (&rtpb);
+ gst_rtp_buffer_unmap (&rtpb);
/* compare the new seqnum to the one in the buffer */
gap = gst_rtp_buffer_compare_seqnum (seqnum, qseq);
break;
}
- rtptime = gst_rtp_buffer_get_timestamp (buf);
+ rtptime = gst_rtp_buffer_get_timestamp (&rtp);
switch (jbuf->mode) {
case RTP_JITTER_BUFFER_MODE_NONE:
case RTP_JITTER_BUFFER_MODE_BUFFER:
if (G_LIKELY (tail))
*tail = (list == NULL);
+ gst_rtp_buffer_unmap (&rtp);
+
return TRUE;
/* ERRORS */
duplicate:
{
+ gst_rtp_buffer_unmap (&rtp);
GST_WARNING ("duplicate packet %d found", (gint) seqnum);
return FALSE;
}
guint64 high_ts, low_ts;
GstBuffer *high_buf, *low_buf;
guint32 result;
+ GstRTPBuffer rtp;
g_return_val_if_fail (jbuf != NULL, 0);
if (!high_buf || !low_buf || high_buf == low_buf)
return 0;
- high_ts = gst_rtp_buffer_get_timestamp (high_buf);
- low_ts = gst_rtp_buffer_get_timestamp (low_buf);
+ gst_rtp_buffer_map (high_buf, GST_MAP_READ, &rtp);
+ high_ts = gst_rtp_buffer_get_timestamp (&rtp);
+ gst_rtp_buffer_unmap (&rtp);
+ gst_rtp_buffer_map (low_buf, GST_MAP_READ, &rtp);
+ low_ts = gst_rtp_buffer_get_timestamp (&rtp);
+ gst_rtp_buffer_unmap (&rtp);
/* it needs to work if ts wraps */
if (high_ts >= low_ts) {
cc->callback);
v_return = callback (data1,
- gst_value_get_mini_object (param_values + 1),
+ g_value_get_boxed (param_values + 1),
g_value_get_boolean (param_values + 2), data2);
g_value_set_boolean (return_value, v_return);
g_value_get_uint (param_values + 2),
g_value_get_uint (param_values + 3),
g_value_get_uint (param_values + 4),
- gst_value_get_mini_object (param_values + 5), data2);
+ g_value_get_boxed (param_values + 5), data2);
}
GstClockTime running_time, guint64 ntpnstime)
{
GstMetaNetAddress *meta;
+ GstRTPBuffer rtpb;
/* get time of arrival */
arrival->current_time = current_time;
arrival->ntpnstime = ntpnstime;
/* get packet size including header overhead */
- arrival->bytes = GST_BUFFER_SIZE (buffer) + sess->header_len;
+ arrival->bytes = gst_buffer_get_size (buffer) + sess->header_len;
if (rtp) {
- arrival->payload_len = gst_rtp_buffer_get_payload_len (buffer);
+ gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtpb);
+ arrival->payload_len = gst_rtp_buffer_get_payload_len (&rtpb);
+ gst_rtp_buffer_unmap (&rtpb);
} else {
arrival->payload_len = 0;
}
guint32 csrcs[16];
guint8 i, count;
guint64 oldrate;
+ GstRTPBuffer rtp;
g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
goto ignore;
/* get SSRC and look up in session database */
- ssrc = gst_rtp_buffer_get_ssrc (buffer);
+ gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp);
+ ssrc = gst_rtp_buffer_get_ssrc (&rtp);
source = obtain_source (sess, ssrc, &created, &arrival, TRUE);
- if (!source)
+ if (!source) {
+ gst_rtp_buffer_unmap (&rtp);
goto collision;
-
- prevsender = RTP_SOURCE_IS_SENDER (source);
- prevactive = RTP_SOURCE_IS_ACTIVE (source);
- oldrate = source->bitrate;
+ }
/* copy available csrc for later */
- count = gst_rtp_buffer_get_csrc_count (buffer);
+ count = gst_rtp_buffer_get_csrc_count (&rtp);
/* make sure to not overflow our array. An RTP buffer can maximally contain
* 16 CSRCs */
count = MIN (count, 16);
for (i = 0; i < count; i++)
- csrcs[i] = gst_rtp_buffer_get_csrc (buffer, i);
+ csrcs[i] = gst_rtp_buffer_get_csrc (&rtp, i);
+
+ gst_rtp_buffer_unmap (&rtp);
+
+ prevsender = RTP_SOURCE_IS_SENDER (source);
+ prevactive = RTP_SOURCE_IS_ACTIVE (source);
+ oldrate = source->bitrate;
/* let source process the packet */
result = rtp_source_process_rtp (source, buffer, &arrival);
GstBuffer *fci_buffer = NULL;
if (fci_length > 0) {
- fci_buffer = gst_buffer_create_sub (packet->buffer,
- fci_data - GST_BUFFER_DATA (packet->buffer), fci_length);
+ fci_buffer = gst_buffer_copy_region (packet->rtcp->buffer,
+ GST_BUFFER_COPY_MEMORY, fci_data - packet->rtcp->data, fci_length);
GST_BUFFER_TIMESTAMP (fci_buffer) = arrival->running_time;
}
gboolean more, is_bye = FALSE, do_sync = FALSE;
RTPArrivalStats arrival;
GstFlowReturn result = GST_FLOW_OK;
+ GstRTCPBuffer rtcp;
g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
goto ignore;
/* start processing the compound packet */
- more = gst_rtcp_buffer_get_first_packet (buffer, &packet);
+ gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
+ more = gst_rtcp_buffer_get_first_packet (&rtcp, &packet);
while (more) {
GstRTCPType type;
more = gst_rtcp_packet_move_to_next (&packet);
}
+ gst_rtcp_buffer_unmap (&rtcp);
+
/* if we are scheduling a BYE, we only want to count bye packets, else we
* count everything */
if (sess->source->received_bye) {
/* notify caller of sr packets in the callback */
if (do_sync && sess->callbacks.sync_rtcp) {
/* make writable, we might want to change the buffer */
- buffer = gst_buffer_make_metadata_writable (buffer);
+ buffer = gst_buffer_make_writable (buffer);
result = sess->callbacks.sync_rtcp (sess, sess->source, buffer,
sess->sync_rtcp_user_data);
g_return_val_if_fail (is_list || GST_IS_BUFFER (data), GST_FLOW_ERROR);
if (is_list) {
- valid_packet = gst_rtp_buffer_list_validate (GST_BUFFER_LIST_CAST (data));
+ GstBufferList *blist = GST_BUFFER_LIST_CAST (data);
+ gint i, len = gst_buffer_list_len (blist);
+
+ valid_packet = TRUE;
+ for (i = 0; i < len; i++)
+ valid_packet &= gst_rtp_buffer_validate (gst_buffer_list_get (blist, i));
} else {
valid_packet = gst_rtp_buffer_validate (GST_BUFFER_CAST (data));
}
{
GstRTCPPacket *packet = &data->packet;
RTPSource *own = sess->source;
+ GstRTCPBuffer rtcp;
data->rtcp = gst_rtcp_buffer_new (sess->mtu);
+ gst_rtcp_buffer_map (data->rtcp, GST_MAP_WRITE, &rtcp);
+
if (RTP_SOURCE_IS_SENDER (own)) {
guint64 ntptime;
guint32 rtptime;
/* we are a sender, create SR */
GST_DEBUG ("create SR for SSRC %08x", own->ssrc);
- gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_SR, packet);
+ gst_rtcp_buffer_add_packet (&rtcp, GST_RTCP_TYPE_SR, packet);
/* get latest stats */
rtp_source_get_new_sr (own, data->ntpnstime, data->running_time,
} else {
/* we are only receiver, create RR */
GST_DEBUG ("create RR for SSRC %08x", own->ssrc);
- gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_RR, packet);
+ gst_rtcp_buffer_add_packet (&rtcp, GST_RTCP_TYPE_RR, packet);
gst_rtcp_packet_rr_set_ssrc (packet, own->ssrc);
}
+
+ gst_rtcp_buffer_unmap (&rtcp);
}
/* construct a Sender or Receiver Report */
GstRTCPPacket *packet = &data->packet;
const GstStructure *sdes;
gint i, n_fields;
+ GstRTCPBuffer rtcp;
+
+ gst_rtcp_buffer_map (data->rtcp, GST_MAP_WRITE, &rtcp);
/* add SDES packet */
- gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_SDES, packet);
+ gst_rtcp_buffer_add_packet (&rtcp, GST_RTCP_TYPE_SDES, packet);
gst_rtcp_packet_sdes_add_item (packet, sess->source->ssrc);
}
data->has_sdes = TRUE;
+
+ gst_rtcp_buffer_unmap (&rtcp);
}
/* schedule a BYE packet */
session_bye (RTPSession * sess, ReportData * data)
{
GstRTCPPacket *packet = &data->packet;
+ GstRTCPBuffer rtcp;
/* open packet */
session_start_rtcp (sess, data);
/* add SDES */
session_sdes (sess, data);
+ gst_rtcp_buffer_map (data->rtcp, GST_MAP_WRITE, &rtcp);
+
/* add a BYE packet */
- gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_BYE, packet);
+ gst_rtcp_buffer_add_packet (&rtcp, GST_RTCP_TYPE_BYE, packet);
gst_rtcp_packet_bye_add_ssrc (packet, sess->source->ssrc);
if (sess->bye_reason)
gst_rtcp_packet_bye_set_reason (packet, sess->bye_reason);
/* we have a BYE packet now */
data->is_bye = TRUE;
+
+ gst_rtcp_buffer_unmap (&rtcp);
}
static gboolean
if (sess->callbacks.send_rtcp && (do_not_suppress || !data.may_suppress)) {
guint packet_size;
- /* close the RTCP packet */
- gst_rtcp_buffer_end (data.rtcp);
-
- packet_size = GST_BUFFER_SIZE (data.rtcp) + sess->header_len;
+ packet_size = gst_buffer_get_size (data.rtcp) + sess->header_len;
UPDATE_AVG (sess->stats.avg_rtcp_packet_size, packet_size);
GST_DEBUG ("%p, sending RTCP packet, avg size %u, %u", &sess->stats,
has_pli_compare_func (gconstpointer a, gconstpointer ignored)
{
GstRTCPPacket packet;
+ GstRTCPBuffer rtcp;
+ gboolean ret = FALSE;
- packet.buffer = (GstBuffer *) a;
- packet.offset = 0;
+ gst_rtcp_buffer_map ((GstBuffer *) a, GST_MAP_READ, &rtcp);
- if (gst_rtcp_packet_get_type (&packet) == GST_RTCP_TYPE_PSFB &&
- gst_rtcp_packet_fb_get_type (&packet) == GST_RTCP_PSFB_TYPE_PLI)
- return TRUE;
- else
- return FALSE;
+ if (gst_rtcp_buffer_get_first_packet (&rtcp, &packet)) {
+ if (gst_rtcp_packet_get_type (&packet) == GST_RTCP_TYPE_PSFB &&
+ gst_rtcp_packet_fb_get_type (&packet) == GST_RTCP_PSFB_TYPE_PLI)
+ ret = TRUE;
+ }
+
+ gst_rtcp_buffer_unmap (&rtcp);
+
+ return ret;
}
static gboolean
if (media_src && !rtp_source_has_retained (media_src,
has_pli_compare_func, NULL)) {
- if (gst_rtcp_buffer_add_packet (buffer, GST_RTCP_TYPE_PSFB, &rtcppacket)) {
+ GstRTCPBuffer rtcp;
+
+ gst_rtcp_buffer_map (buffer, GST_MAP_WRITE, &rtcp);
+ if (gst_rtcp_buffer_add_packet (&rtcp, GST_RTCP_TYPE_PSFB, &rtcppacket)) {
gst_rtcp_packet_fb_set_type (&rtcppacket, GST_RTCP_PSFB_TYPE_PLI);
gst_rtcp_packet_fb_set_sender_ssrc (&rtcppacket,
rtp_source_get_ssrc (sess->source));
gst_rtcp_packet_fb_set_media_ssrc (&rtcppacket, media_ssrc);
ret = TRUE;
+ gst_rtcp_buffer_unmap (&rtcp);
} else {
/* Break because the packet is full, will put next request in a
* further packet
*/
+ gst_rtcp_buffer_unmap (&rtcp);
break;
}
}
gint32 diff;
gint clock_rate;
guint8 pt;
+ GstRTPBuffer rtp;
/* get arrival time */
if ((running_time = arrival->running_time) == GST_CLOCK_TIME_NONE)
goto no_time;
- pt = gst_rtp_buffer_get_payload_type (buffer);
+ gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp);
+ pt = gst_rtp_buffer_get_payload_type (&rtp);
GST_LOG ("SSRC %08x got payload %d", src->ssrc, pt);
/* get clockrate */
- if ((clock_rate = get_clock_rate (src, pt)) == -1)
+ if ((clock_rate = get_clock_rate (src, pt)) == -1) {
+ gst_rtp_buffer_unmap (&rtp);
goto no_clock_rate;
+ }
- rtptime = gst_rtp_buffer_get_timestamp (buffer);
+ rtptime = gst_rtp_buffer_get_timestamp (&rtp);
/* convert arrival time to RTP timestamp units, truncate to 32 bits, we don't
* care about the absolute value, just the difference. */
GST_LOG ("rtparrival %u, rtptime %u, clock-rate %d, diff %d, jitter: %f",
rtparrival, rtptime, clock_rate, diff, (src->stats.jitter) / 16.0);
+ gst_rtp_buffer_unmap (&rtp);
return;
/* ERRORS */
guint16 seqnr, udelta;
RTPSourceStats *stats;
guint16 expected;
+ GstRTPBuffer rtp;
g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
stats = &src->stats;
- seqnr = gst_rtp_buffer_get_seq (buffer);
+ gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp);
+ seqnr = gst_rtp_buffer_get_seq (&rtp);
+ gst_rtp_buffer_unmap (&rtp);
- rtp_source_update_caps (src, GST_BUFFER_CAPS (buffer));
+ /* FIXME-0.11
+ * would be nice to be able to pass along with buffer */
+ g_assert_not_reached ();
+ /* rtp_source_update_caps (src, GST_BUFFER_CAPS (buffer)); */
if (stats->cycles == -1) {
GST_DEBUG ("received first buffer");
src->received_bye = TRUE;
}
-static GstBufferListItem
-set_ssrc (GstBuffer ** buffer, guint group, guint idx, RTPSource * src)
+static gboolean
+set_ssrc (GstBuffer ** buffer, guint idx, RTPSource * src)
{
+ GstRTPBuffer rtp;
+
*buffer = gst_buffer_make_writable (*buffer);
- gst_rtp_buffer_set_ssrc (*buffer, src->ssrc);
- return GST_BUFFER_LIST_SKIP_GROUP;
+ gst_rtp_buffer_map (*buffer, GST_MAP_WRITE, &rtp);
+ gst_rtp_buffer_set_ssrc (&rtp, src->ssrc);
+ gst_rtp_buffer_unmap (&rtp);
+ return TRUE;
}
/**
GstBuffer *buffer = NULL;
guint packets;
guint32 ssrc;
+ GstRTPBuffer rtp;
g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
g_return_val_if_fail (is_list || GST_IS_BUFFER (data), GST_FLOW_ERROR);
/* We can grab the caps from the first group, since all
* groups of a buffer list have same caps. */
- buffer = gst_buffer_list_get (list, 0, 0);
+ buffer = gst_buffer_list_get (list, 0);
if (!buffer)
goto no_buffer;
} else {
buffer = GST_BUFFER_CAST (data);
}
- rtp_source_update_caps (src, GST_BUFFER_CAPS (buffer));
+
+ /* FIXME-0.11 */
+ g_assert_not_reached ();
+ /* rtp_source_update_caps (src, GST_BUFFER_CAPS (buffer)); */
/* we are a sender now */
src->is_sender = TRUE;
if (is_list) {
+ gint i;
+
/* Each group makes up a network packet. */
- packets = gst_buffer_list_n_groups (list);
- len = gst_rtp_buffer_list_get_payload_len (list);
+ packets = gst_buffer_list_len (list);
+ for (i = 0, len = 0; i < packets; i++) {
+ gst_rtp_buffer_map (gst_buffer_list_get (list, i), GST_MAP_READ, &rtp);
+ len += gst_rtp_buffer_get_payload_len (&rtp);
+ gst_rtp_buffer_unmap (&rtp);
+ }
+ /* subsequent info taken from first list member */
+ gst_rtp_buffer_map (gst_buffer_list_get (list, 0), GST_MAP_READ, &rtp);
} else {
packets = 1;
- len = gst_rtp_buffer_get_payload_len (buffer);
+ gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp);
+ len = gst_rtp_buffer_get_payload_len (&rtp);
}
/* update stats for the SR */
do_bitrate_estimation (src, running_time, &src->bytes_sent);
- if (is_list) {
- rtptime = gst_rtp_buffer_list_get_timestamp (list);
- } else {
- rtptime = gst_rtp_buffer_get_timestamp (buffer);
- }
+ rtptime = gst_rtp_buffer_get_timestamp (&rtp);
ext_rtptime = src->last_rtptime;
ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
src->last_rtptime = ext_rtptime;
/* push packet */
- if (!src->callbacks.push_rtp)
+ if (!src->callbacks.push_rtp) {
+ gst_rtp_buffer_unmap (&rtp);
goto no_callback;
-
- if (is_list) {
- ssrc = gst_rtp_buffer_list_get_ssrc (list);
- } else {
- ssrc = gst_rtp_buffer_get_ssrc (buffer);
}
+ ssrc = gst_rtp_buffer_get_ssrc (&rtp);
+ gst_rtp_buffer_unmap (&rtp);
+
if (ssrc != src->ssrc) {
/* the SSRC of the packet is not correct, make a writable buffer and
* update the SSRC. This could involve a complete copy of the packet when
list = gst_buffer_list_make_writable (list);
gst_buffer_list_foreach (list, (GstBufferListFunc) set_ssrc, src);
} else {
- set_ssrc (&buffer, 0, 0, src);
+ set_ssrc (&buffer, 0, src);
}
}
GST_LOG ("pushing RTP %s %" G_GUINT64_FORMAT, is_list ? "list" : "packet",
{
GstBuffer *buffer;
- buffer = gst_buffer_create_sub (packet->buffer, packet->offset,
- (gst_rtcp_packet_get_length (packet) + 1) * 4);
+ buffer = gst_buffer_copy_region (packet->rtcp->buffer, GST_BUFFER_COPY_MEMORY,
+ packet->offset, (gst_rtcp_packet_get_length (packet) + 1) * 4);
GST_BUFFER_TIMESTAMP (buffer) = running_time;
projects / platform / upstream / gstreamer.git / commitdiff