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configure.ac: Disable rtpmanager for now because it depends on CVS -base.

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Original commit message from CVS:
* configure.ac:
Disable rtpmanager for now because it depends on CVS -base.
* gst/rtpmanager/Makefile.am:
Added new files for session manager.
* gst/rtpmanager/gstrtpjitterbuffer.h:
* gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map),
(create_stream), (pt_map_requested), (new_ssrc_pad_found):
Some cleanups.
the session manager can now also request a pt-map.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_base_init),
(gst_rtp_session_class_init), (gst_rtp_session_init),
(gst_rtp_session_finalize), (rtcp_thread), (start_rtcp_thread),
(stop_rtcp_thread), (gst_rtp_session_change_state),
(gst_rtp_session_process_rtp), (gst_rtp_session_send_rtp),
(gst_rtp_session_send_rtcp), (gst_rtp_session_clock_rate),
(gst_rtp_session_get_time), (gst_rtp_session_event_recv_rtp_sink),
(gst_rtp_session_chain_recv_rtp),
(gst_rtp_session_event_recv_rtcp_sink),
(gst_rtp_session_chain_recv_rtcp),
(gst_rtp_session_event_send_rtp_sink),
(gst_rtp_session_chain_send_rtp), (create_send_rtcp_src),
(gst_rtp_session_request_new_pad):
* gst/rtpmanager/gstrtpsession.h:
We can ask for pt-map now too when the session manager needs it.
Hook up to the new session manager, implement the needed callbacks for
pushing data, getting clock time and requesting clock-rates.
Rename rtcp_src to send_rtcp_src to make it clear that this RTCP is to
be send to clients.
Add code to start and stop the thread that will schedule RTCP through
the session manager.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(rtp_session_init), (rtp_session_finalize),
(rtp_session_set_property), (rtp_session_get_property),
(on_new_ssrc), (on_ssrc_collision), (on_ssrc_validated),
(on_bye_ssrc), (rtp_session_new), (rtp_session_set_callbacks),
(rtp_session_set_bandwidth), (rtp_session_get_bandwidth),
(rtp_session_set_rtcp_bandwidth), (rtp_session_get_rtcp_bandwidth),
(source_push_rtp), (source_clock_rate), (check_collision),
(obtain_source), (rtp_session_add_source),
(rtp_session_get_num_sources),
(rtp_session_get_num_active_sources),
(rtp_session_get_source_by_ssrc),
(rtp_session_get_source_by_cname), (rtp_session_create_source),
(update_arrival_stats), (rtp_session_process_rtp),
(rtp_session_process_sr), (rtp_session_process_rr),
(rtp_session_process_sdes), (rtp_session_process_bye),
(rtp_session_process_app), (rtp_session_process_rtcp),
(rtp_session_send_rtp), (rtp_session_get_rtcp_interval),
(rtp_session_produce_rtcp):
* gst/rtpmanager/rtpsession.h:
The advanced beginnings of the main session manager that handles the
participant database of RTPSources, SSRC probation, SSRC collisions,
parse RTCP to update source stats. etc..
* gst/rtpmanager/rtpsource.c: (rtp_source_class_init),
(rtp_source_init), (rtp_source_finalize), (rtp_source_new),
(rtp_source_set_callbacks), (rtp_source_set_as_csrc),
(rtp_source_set_rtp_from), (rtp_source_set_rtcp_from),
(push_packet), (get_clock_rate), (calculate_jitter),
(rtp_source_process_rtp), (rtp_source_process_bye),
(rtp_source_send_rtp), (rtp_source_process_sr),
(rtp_source_process_rb):
* gst/rtpmanager/rtpsource.h:
Object that encapsulates an SSRC and its state in the database.
Calculates the jitter and transit times of data packets.
* gst/rtpmanager/rtpstats.c: (rtp_stats_init_defaults),
(rtp_stats_calculate_rtcp_interval), (rtp_stats_add_rtcp_jitter):
* gst/rtpmanager/rtpstats.h:
Various stats regarding the session and sources.
Used to calculate the RTCP interval.
This commit is contained in:
Wim Taymans 2007-04-18 18:58:53 +00:00
parent 6cbfc31aae
commit 1d75a69ccf
14 changed files with 2556 additions and 37 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

78
ChangeLog
View file

@ -1,3 +1,81 @@
2007-04-18 Wim Taymans <wim@fluendo.com>
* configure.ac:
Disable rtpmanager for now because it depends on CVS -base.
* gst/rtpmanager/Makefile.am:
Added new files for session manager.
* gst/rtpmanager/gstrtpjitterbuffer.h:
* gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map),
(create_stream), (pt_map_requested), (new_ssrc_pad_found):
Some cleanups.
the session manager can now also request a pt-map.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_base_init),
(gst_rtp_session_class_init), (gst_rtp_session_init),
(gst_rtp_session_finalize), (rtcp_thread), (start_rtcp_thread),
(stop_rtcp_thread), (gst_rtp_session_change_state),
(gst_rtp_session_process_rtp), (gst_rtp_session_send_rtp),
(gst_rtp_session_send_rtcp), (gst_rtp_session_clock_rate),
(gst_rtp_session_get_time), (gst_rtp_session_event_recv_rtp_sink),
(gst_rtp_session_chain_recv_rtp),
(gst_rtp_session_event_recv_rtcp_sink),
(gst_rtp_session_chain_recv_rtcp),
(gst_rtp_session_event_send_rtp_sink),
(gst_rtp_session_chain_send_rtp), (create_send_rtcp_src),
(gst_rtp_session_request_new_pad):
* gst/rtpmanager/gstrtpsession.h:
We can ask for pt-map now too when the session manager needs it.
Hook up to the new session manager, implement the needed callbacks for
pushing data, getting clock time and requesting clock-rates.
Rename rtcp_src to send_rtcp_src to make it clear that this RTCP is to
be send to clients.
Add code to start and stop the thread that will schedule RTCP through
the session manager.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(rtp_session_init), (rtp_session_finalize),
(rtp_session_set_property), (rtp_session_get_property),
(on_new_ssrc), (on_ssrc_collision), (on_ssrc_validated),
(on_bye_ssrc), (rtp_session_new), (rtp_session_set_callbacks),
(rtp_session_set_bandwidth), (rtp_session_get_bandwidth),
(rtp_session_set_rtcp_bandwidth), (rtp_session_get_rtcp_bandwidth),
(source_push_rtp), (source_clock_rate), (check_collision),
(obtain_source), (rtp_session_add_source),
(rtp_session_get_num_sources),
(rtp_session_get_num_active_sources),
(rtp_session_get_source_by_ssrc),
(rtp_session_get_source_by_cname), (rtp_session_create_source),
(update_arrival_stats), (rtp_session_process_rtp),
(rtp_session_process_sr), (rtp_session_process_rr),
(rtp_session_process_sdes), (rtp_session_process_bye),
(rtp_session_process_app), (rtp_session_process_rtcp),
(rtp_session_send_rtp), (rtp_session_get_rtcp_interval),
(rtp_session_produce_rtcp):
* gst/rtpmanager/rtpsession.h:
The advanced beginnings of the main session manager that handles the
participant database of RTPSources, SSRC probation, SSRC collisions,
parse RTCP to update source stats. etc..
* gst/rtpmanager/rtpsource.c: (rtp_source_class_init),
(rtp_source_init), (rtp_source_finalize), (rtp_source_new),
(rtp_source_set_callbacks), (rtp_source_set_as_csrc),
(rtp_source_set_rtp_from), (rtp_source_set_rtcp_from),
(push_packet), (get_clock_rate), (calculate_jitter),
(rtp_source_process_rtp), (rtp_source_process_bye),
(rtp_source_send_rtp), (rtp_source_process_sr),
(rtp_source_process_rb):
* gst/rtpmanager/rtpsource.h:
Object that encapsulates an SSRC and its state in the database.
Calculates the jitter and transit times of data packets.
* gst/rtpmanager/rtpstats.c: (rtp_stats_init_defaults),
(rtp_stats_calculate_rtcp_interval), (rtp_stats_add_rtcp_jitter):
* gst/rtpmanager/rtpstats.h:
Various stats regarding the session and sources.
Used to calculate the RTCP interval.
2007-04-17 Tim-Philipp Müller <tim at centricular dot net>
* gst/app/Makefile.am:

2
common

@ -1 +1 @@
Subproject commit 9097e252e477e18182f08a032d8860bdee9a0416
Subproject commit e05f45f13961b851501ca8938aa2049fa96c7b11

1
configure.ac
View file

@ -95,7 +95,6 @@ GST_PLUGINS_ALL="\
nuvdemux \
real \
replaygain \
rtpmanager \
spectrum \
speed \
qtdemux \

9
gst/rtpmanager/Makefile.am
View file

@ -17,6 +17,9 @@ libgstrtpmanager_la_SOURCES = gstrtpmanager.c \
gstrtpjitterbuffer.c \
gstrtpptdemux.c \
gstrtpssrcdemux.c \
rtpsession.c \
rtpsource.c \
rtpstats.c \
gstrtpsession.c
nodist_libgstrtpmanager_la_SOURCES = \
@ -28,11 +31,15 @@ noinst_HEADERS = gstrtpbin.h \
gstrtpjitterbuffer.h \
gstrtpptdemux.h \
gstrtpssrcdemux.h \
rtpsession.h \
rtpsource.h \
rtpstats.h \
gstrtpsession.h
libgstrtpmanager_la_CFLAGS = $(GST_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS) $(ERROR_CFLAGS)
libgstrtpmanager_la_LIBADD = $(GST_LIBS_LIBS)
libgstrtpmanager_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) $(GST_BASE_LIBS) $(GST_PLUGINS_BASE_LIBS) -lgstrtp-@GST_MAJORMINOR@
libgstrtpmanager_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) $(GST_BASE_LIBS) $(GST_PLUGINS_BASE_LIBS) -lgstrtp-@GST_MAJORMINOR@ \
-lgstnetbuffer-@GST_MAJORMINOR@
CLEANFILES = $(BUILT_SOURCES)

28
gst/rtpmanager/gstrtpbin.c
View file

@ -129,7 +129,7 @@ typedef struct _GstRTPBinClient GstRTPBinClient;
static guint gst_rtp_bin_signals[LAST_SIGNAL] = { 0 };
static GstCaps *pt_map_requested (GstElement * element, guint pt,
GstRTPBinStream * stream);
GstRTPBinSession * session);
/* Manages the RTP stream for one SSRC.
*
@ -215,9 +215,9 @@ static GstRTPBinSession *
create_session (GstRTPBin * rtpbin, gint id)
{
GstRTPBinSession *sess;
GstElement *elem, *demux;
GstElement *session, *demux;
if (!(elem = gst_element_factory_make ("rtpsession", NULL)))
if (!(session = gst_element_factory_make ("rtpsession", NULL)))
goto no_session;
if (!(demux = gst_element_factory_make ("rtpssrcdemux", NULL)))
@ -227,13 +227,17 @@ create_session (GstRTPBin * rtpbin, gint id)
sess->lock = g_mutex_new ();
sess->id = id;
sess->bin = rtpbin;
sess->session = elem;
sess->session = session;
sess->demux = demux;
sess->ptmap = g_hash_table_new (NULL, NULL);
rtpbin->sessions = g_slist_prepend (rtpbin->sessions, sess);
gst_bin_add (GST_BIN_CAST (rtpbin), elem);
gst_element_set_state (elem, GST_STATE_PLAYING);
/* provide clock_rate to the session manager when needed */
g_signal_connect (session, "request-pt-map",
(GCallback) pt_map_requested, sess);
gst_bin_add (GST_BIN_CAST (rtpbin), session);
gst_element_set_state (session, GST_STATE_PLAYING);
gst_bin_add (GST_BIN_CAST (rtpbin), demux);
gst_element_set_state (demux, GST_STATE_PLAYING);
@ -247,7 +251,7 @@ no_session:
}
no_demux:
{
gst_object_unref (elem);
gst_object_unref (session);
g_warning ("rtpbin: could not create rtpssrcdemux element");
return NULL;
}
@ -351,7 +355,7 @@ create_stream (GstRTPBinSession * session, guint32 ssrc)
/* provide clock_rate to the jitterbuffer when needed */
g_signal_connect (buffer, "request-pt-map",
(GCallback) pt_map_requested, stream);
(GCallback) pt_map_requested, session);
gst_bin_add (GST_BIN_CAST (session->bin), buffer);
gst_element_set_state (buffer, GST_STATE_PLAYING);
@ -590,14 +594,12 @@ new_payload_found (GstElement * element, guint pt, GstPad * pad,
}
static GstCaps *
pt_map_requested (GstElement * element, guint pt, GstRTPBinStream * stream)
pt_map_requested (GstElement * element, guint pt, GstRTPBinSession * session)
{
GstRTPBin *rtpbin;
GstRTPBinSession *session;
GstCaps *caps;
rtpbin = stream->bin;
session = stream->session;
rtpbin = session->bin;
GST_DEBUG_OBJECT (rtpbin, "payload map requested for pt %d in session %d", pt,
session->id);
@ -647,7 +649,7 @@ new_ssrc_pad_found (GstElement * element, guint ssrc, GstPad * pad,
* demuxer so that it can apply a proper caps on the buffers for the
* depayloaders. */
stream->demux_ptreq_sig = g_signal_connect (stream->demux,
"request-pt-map", (GCallback) pt_map_requested, stream);
"request-pt-map", (GCallback) pt_map_requested, session);
GST_RTP_SESSION_UNLOCK (session);

1
gst/rtpmanager/gstrtpjitterbuffer.h
View file

@ -63,6 +63,7 @@ struct _GstRTPJitterBufferClass
{
GstElementClass parent_class;
/* signals */
GstCaps* (*request_pt_map) (GstRTPJitterBuffer *buffer, guint pt);
/*< private > */

325
gst/rtpmanager/gstrtpsession.c
View file

@ -39,7 +39,10 @@
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstrtpbin-marshal.h"
#include "gstrtpsession.h"
#include "rtpsession.h"
GST_DEBUG_CATEGORY_STATIC (gst_rtp_session_debug);
#define GST_CAT_DEFAULT gst_rtp_session_debug
@ -95,8 +98,8 @@ GST_STATIC_PAD_TEMPLATE ("send_rtp_src",
GST_STATIC_CAPS ("application/x-rtp")
);
static GstStaticPadTemplate rtpsession_rtcp_src_template =
GST_STATIC_PAD_TEMPLATE ("rtcp_src",
static GstStaticPadTemplate rtpsession_send_rtcp_src_template =
GST_STATIC_PAD_TEMPLATE ("send_rtcp_src",
GST_PAD_SRC,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("application/x-rtcp")
@ -105,7 +108,7 @@ GST_STATIC_PAD_TEMPLATE ("rtcp_src",
/* signals and args */
enum
{
/* FILL ME */
SIGNAL_REQUEST_PT_MAP,
LAST_SIGNAL
};
@ -123,6 +126,31 @@ enum
struct _GstRTPSessionPrivate
{
GMutex *lock;
RTPSession *session;
/* thread for sending out RTCP */
GstClockID id;
gboolean stop_thread;
GThread *thread;
};
/* callbacks to handle actions from the session manager */
static GstFlowReturn gst_rtp_session_process_rtp (RTPSession * sess,
RTPSource * src, GstBuffer * buffer, gpointer user_data);
static GstFlowReturn gst_rtp_session_send_rtp (RTPSession * sess,
RTPSource * src, GstBuffer * buffer, gpointer user_data);
static GstFlowReturn gst_rtp_session_send_rtcp (RTPSession * sess,
RTPSource * src, GstBuffer * buffer, gpointer user_data);
static gint gst_rtp_session_clock_rate (RTPSession * sess, guint8 payload,
gpointer user_data);
static GstClockTime gst_rtp_session_get_time (RTPSession * sess,
gpointer user_data);
static RTPSessionCallbacks callbacks = {
gst_rtp_session_process_rtp,
gst_rtp_session_send_rtp,
gst_rtp_session_send_rtcp,
gst_rtp_session_clock_rate,
gst_rtp_session_get_time
};
/* GObject vmethods */
@ -139,7 +167,7 @@ static GstPad *gst_rtp_session_request_new_pad (GstElement * element,
GstPadTemplate * templ, const gchar * name);
static void gst_rtp_session_release_pad (GstElement * element, GstPad * pad);
/*static guint gst_rtp_session_signals[LAST_SIGNAL] = { 0 }; */
static guint gst_rtp_session_signals[LAST_SIGNAL] = { 0 };
GST_BOILERPLATE (GstRTPSession, gst_rtp_session, GstElement, GST_TYPE_ELEMENT);
@ -164,7 +192,7 @@ gst_rtp_session_base_init (gpointer klass)
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_rtcp_src_template));
gst_static_pad_template_get (&rtpsession_send_rtcp_src_template));
gst_element_class_set_details (element_class, &rtpsession_details);
}
@ -184,6 +212,19 @@ gst_rtp_session_class_init (GstRTPSessionClass * klass)
gobject_class->set_property = gst_rtp_session_set_property;
gobject_class->get_property = gst_rtp_session_get_property;
/**
* GstRTPSession::request-pt-map:
* @sess: the object which received the signal
* @pt: the pt
*
* Request the payload type as #GstCaps for @pt.
*/
gst_rtp_session_signals[SIGNAL_REQUEST_PT_MAP] =
g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPSessionClass, request_pt_map),
NULL, NULL, gst_rtp_bin_marshal_BOXED__UINT, GST_TYPE_CAPS, 1,
G_TYPE_UINT);
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_rtp_session_change_state);
gstelement_class->request_new_pad =
@ -200,6 +241,9 @@ gst_rtp_session_init (GstRTPSession * rtpsession, GstRTPSessionClass * klass)
{
rtpsession->priv = GST_RTP_SESSION_GET_PRIVATE (rtpsession);
rtpsession->priv->lock = g_mutex_new ();
rtpsession->priv->session = rtp_session_new ();
/* configure callbacks */
rtp_session_set_callbacks (rtpsession->priv->session, &callbacks, rtpsession);
}
static void
@ -209,6 +253,7 @@ gst_rtp_session_finalize (GObject * object)
rtpsession = GST_RTP_SESSION (object);
g_mutex_free (rtpsession->priv->lock);
g_object_unref (rtpsession->priv->session);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
@ -243,6 +288,87 @@ gst_rtp_session_get_property (GObject * object, guint prop_id,
}
}
static void
rtcp_thread (GstRTPSession * rtpsession)
{
GstClock *clock;
GstClockID id;
clock = gst_element_get_clock (GST_ELEMENT_CAST (rtpsession));
if (clock == NULL)
return;
GST_DEBUG_OBJECT (rtpsession, "entering RTCP thread");
GST_RTP_SESSION_LOCK (rtpsession);
while (!rtpsession->priv->stop_thread) {
gdouble timeout;
GstClockTime target;
timeout = rtp_session_get_rtcp_interval (rtpsession->priv->session);
GST_DEBUG_OBJECT (rtpsession, "next RTCP timeout: %lf", timeout);
target = gst_clock_get_time (clock);
target += GST_SECOND * timeout;
id = rtpsession->priv->id = gst_clock_new_single_shot_id (clock, target);
GST_RTP_SESSION_UNLOCK (rtpsession);
gst_clock_id_wait (id, NULL);
GST_DEBUG_OBJECT (rtpsession, "got RTCP timeout");
/* make the session manager produce RTCP, we ignore the result. */
rtp_session_produce_rtcp (rtpsession->priv->session);
GST_RTP_SESSION_LOCK (rtpsession);
gst_clock_id_unref (id);
rtpsession->priv->id = NULL;
}
GST_RTP_SESSION_UNLOCK (rtpsession);
gst_object_unref (clock);
GST_DEBUG_OBJECT (rtpsession, "leaving RTCP thread");
}
static gboolean
start_rtcp_thread (GstRTPSession * rtpsession)
{
GError *error = NULL;
gboolean res;
GST_DEBUG_OBJECT (rtpsession, "starting RTCP thread");
GST_RTP_SESSION_LOCK (rtpsession);
rtpsession->priv->stop_thread = FALSE;
rtpsession->priv->thread =
g_thread_create ((GThreadFunc) rtcp_thread, rtpsession, TRUE, &error);
GST_RTP_SESSION_UNLOCK (rtpsession);
if (error != NULL) {
res = FALSE;
GST_DEBUG_OBJECT (rtpsession, "failed to start thread, %s", error->message);
g_error_free (error);
} else {
res = TRUE;
}
return res;
}
static void
stop_rtcp_thread (GstRTPSession * rtpsession)
{
GST_DEBUG_OBJECT (rtpsession, "stopping RTCP thread");
GST_RTP_SESSION_LOCK (rtpsession);
rtpsession->priv->stop_thread = TRUE;
if (rtpsession->priv->id)
gst_clock_id_unschedule (rtpsession->priv->id);
GST_RTP_SESSION_UNLOCK (rtpsession);
g_thread_join (rtpsession->priv->thread);
}
static GstStateChangeReturn
gst_rtp_session_change_state (GstElement * element, GstStateChange transition)
{
@ -258,6 +384,8 @@ gst_rtp_session_change_state (GstElement * element, GstStateChange transition)
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
stop_rtcp_thread (rtpsession);
default:
break;
}
@ -265,6 +393,10 @@ gst_rtp_session_change_state (GstElement * element, GstStateChange transition)
res = parent_class->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
if (!start_rtcp_thread (rtpsession))
goto failed_thread;
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
@ -275,15 +407,158 @@ gst_rtp_session_change_state (GstElement * element, GstStateChange transition)
break;
}
return res;
/* ERRORS */
failed_thread:
{
return GST_STATE_CHANGE_FAILURE;
}
}
/* called when the session manager has an RTP packet ready for further
* processing */
static GstFlowReturn
gst_rtp_session_process_rtp (RTPSession * sess, RTPSource * src,
GstBuffer * buffer, gpointer user_data)
{
GstFlowReturn result;
GstRTPSession *rtpsession;
GstRTPSessionPrivate *priv;
rtpsession = GST_RTP_SESSION (user_data);
priv = rtpsession->priv;
if (rtpsession->recv_rtp_src) {
result = gst_pad_push (rtpsession->recv_rtp_src, buffer);
} else {
gst_buffer_unref (buffer);
result = GST_FLOW_OK;
}
return result;
}
/* called when the session manager has an RTP packet ready for further
* sending */
static GstFlowReturn
gst_rtp_session_send_rtp (RTPSession * sess, RTPSource * src,
GstBuffer * buffer, gpointer user_data)
{
GstFlowReturn result;
GstRTPSession *rtpsession;
GstRTPSessionPrivate *priv;
rtpsession = GST_RTP_SESSION (user_data);
priv = rtpsession->priv;
if (rtpsession->send_rtp_src) {
result = gst_pad_push (rtpsession->send_rtp_src, buffer);
} else {
gst_buffer_unref (buffer);
result = GST_FLOW_OK;
}
return result;
}
/* called when the session manager has an RTCP packet ready for further
* sending */
static GstFlowReturn
gst_rtp_session_send_rtcp (RTPSession * sess, RTPSource * src,
GstBuffer * buffer, gpointer user_data)
{
GstFlowReturn result;
GstRTPSession *rtpsession;
GstRTPSessionPrivate *priv;
rtpsession = GST_RTP_SESSION (user_data);
priv = rtpsession->priv;
if (rtpsession->send_rtcp_src) {
result = gst_pad_push (rtpsession->send_rtcp_src, buffer);
} else {
gst_buffer_unref (buffer);
result = GST_FLOW_OK;
}
return result;
}
/* called when the session manager needs the clock rate */
static gint
gst_rtp_session_clock_rate (RTPSession * sess, guint8 payload,
gpointer user_data)
{
gint result = -1;
GstRTPSession *rtpsession;
GValue ret = { 0 };
GValue args[2] = { {0}, {0} };
GstCaps *caps;
const GstStructure *caps_struct;
rtpsession = GST_RTP_SESSION_CAST (user_data);
g_value_init (&args[0], GST_TYPE_ELEMENT);
g_value_set_object (&args[0], rtpsession);
g_value_init (&args[1], G_TYPE_UINT);
g_value_set_uint (&args[1], payload);
g_value_init (&ret, GST_TYPE_CAPS);
g_value_set_boxed (&ret, NULL);
g_signal_emitv (args, gst_rtp_session_signals[SIGNAL_REQUEST_PT_MAP], 0,
&ret);
caps = (GstCaps *) g_value_get_boxed (&ret);
if (!caps)
goto no_caps;
caps_struct = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (caps_struct, "clock-rate", &result))
goto no_clock_rate;
return result;
/* ERRORS */
no_caps:
{
GST_DEBUG_OBJECT (rtpsession, "could not get caps");
return -1;
}
no_clock_rate:
{
GST_DEBUG_OBJECT (rtpsession, "could not clock-rate from caps");
return -1;
}
}
/* called when the session manager needs the time of clock */
static GstClockTime
gst_rtp_session_get_time (RTPSession * sess, gpointer user_data)
{
GstClockTime result;
GstRTPSession *rtpsession;
GstClock *clock;
rtpsession = GST_RTP_SESSION_CAST (user_data);
clock = gst_element_get_clock (GST_ELEMENT_CAST (rtpsession));
if (clock) {
result = gst_clock_get_time (clock);
gst_object_unref (clock);
} else
result = GST_CLOCK_TIME_NONE;
return result;
}
static GstFlowReturn
gst_rtp_session_event_recv_rtp_sink (GstPad * pad, GstEvent * event)
{
GstRTPSession *rtpsession;
GstRTPSessionPrivate *priv;
gboolean ret = FALSE;
rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
priv = rtpsession->priv;
GST_DEBUG_OBJECT (rtpsession, "received event %s",
GST_EVENT_TYPE_NAME (event));
@ -305,14 +580,15 @@ static GstFlowReturn
gst_rtp_session_chain_recv_rtp (GstPad * pad, GstBuffer * buffer)
{
GstRTPSession *rtpsession;
GstRTPSessionPrivate *priv;
GstFlowReturn ret;
rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
priv = rtpsession->priv;
GST_DEBUG_OBJECT (rtpsession, "received RTP packet");
/* FIXME, do something */
ret = gst_pad_push (rtpsession->recv_rtp_src, buffer);
ret = rtp_session_process_rtp (priv->session, buffer);
gst_object_unref (rtpsession);
@ -323,9 +599,11 @@ static GstFlowReturn
gst_rtp_session_event_recv_rtcp_sink (GstPad * pad, GstEvent * event)
{
GstRTPSession *rtpsession;
GstRTPSessionPrivate *priv;
gboolean ret = FALSE;
rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
priv = rtpsession->priv;
GST_DEBUG_OBJECT (rtpsession, "received event %s",
GST_EVENT_TYPE_NAME (event));
@ -347,14 +625,15 @@ static GstFlowReturn
gst_rtp_session_chain_recv_rtcp (GstPad * pad, GstBuffer * buffer)
{
GstRTPSession *rtpsession;
GstRTPSessionPrivate *priv;
GstFlowReturn ret;
rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
priv = rtpsession->priv;
/* FIXME, do something */
GST_DEBUG_OBJECT (rtpsession, "received RTCP packet");
ret = gst_pad_push (rtpsession->sync_src, buffer);
ret = rtp_session_process_rtcp (priv->session, buffer);
gst_object_unref (rtpsession);
@ -365,9 +644,11 @@ static GstFlowReturn
gst_rtp_session_event_send_rtp_sink (GstPad * pad, GstEvent * event)
{
GstRTPSession *rtpsession;
GstRTPSessionPrivate *priv;
gboolean ret = FALSE;
rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
priv = rtpsession->priv;
GST_DEBUG_OBJECT (rtpsession, "received event");
@ -388,14 +669,15 @@ static GstFlowReturn
gst_rtp_session_chain_send_rtp (GstPad * pad, GstBuffer * buffer)
{
GstRTPSession *rtpsession;
GstRTPSessionPrivate *priv;
GstFlowReturn ret;
rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
priv = rtpsession->priv;
GST_DEBUG_OBJECT (rtpsession, "received RTP packet");
/* FIXME, do something */
ret = gst_pad_push (rtpsession->send_rtp_src, buffer);
ret = rtp_session_send_rtp (priv->session, buffer);
gst_object_unref (rtpsession);
@ -494,16 +776,18 @@ create_send_rtp_sink (GstRTPSession * rtpsession)
* RTCP packets.
*/
static GstPad *
create_rtcp_src (GstRTPSession * rtpsession)
create_send_rtcp_src (GstRTPSession * rtpsession)
{
GST_DEBUG_OBJECT (rtpsession, "creating pad");
rtpsession->rtcp_src =
gst_pad_new_from_static_template (&rtpsession_rtcp_src_template, NULL);
gst_pad_set_active (rtpsession->rtcp_src, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->rtcp_src);
rtpsession->send_rtcp_src =
gst_pad_new_from_static_template (&rtpsession_send_rtcp_src_template,
NULL);
gst_pad_set_active (rtpsession->send_rtcp_src, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpsession),
rtpsession->send_rtcp_src);
return rtpsession->rtcp_src;
return rtpsession->send_rtcp_src;
}
static GstPad *
@ -542,11 +826,12 @@ gst_rtp_session_request_new_pad (GstElement * element,
goto exists;
result = create_send_rtp_sink (rtpsession);
} else if (templ == gst_element_class_get_pad_template (klass, "rtcp_src")) {
if (rtpsession->rtcp_src != NULL)
} else if (templ == gst_element_class_get_pad_template (klass,
"send_rtcp_src")) {
if (rtpsession->send_rtcp_src != NULL)
goto exists;
result = create_rtcp_src (rtpsession);
result = create_send_rtcp_src (rtpsession);
} else
goto wrong_template;

6
gst/rtpmanager/gstrtpsession.h
View file

@ -32,6 +32,7 @@
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_SESSION))
#define GST_IS_RTP_SESSION_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_SESSION))
#define GST_RTP_SESSION_CAST(obj) ((GstRTPSession *)(obj))
typedef struct _GstRTPSession GstRTPSession;
typedef struct _GstRTPSessionClass GstRTPSessionClass;
@ -48,13 +49,16 @@ struct _GstRTPSession {
GstPad *recv_rtp_src;
GstPad *sync_src;
GstPad *send_rtp_src;
GstPad *rtcp_src;
GstPad *send_rtcp_src;
GstRTPSessionPrivate *priv;
};
struct _GstRTPSessionClass {
GstElementClass parent_class;
/* signals */
GstCaps* (*request_pt_map) (GstRTPSession *sess, guint pt);
};
GType gst_rtp_session_get_type (void);

1026
gst/rtpmanager/rtpsession.c Normal file
View file

File diff suppressed because it is too large Load diff

206
gst/rtpmanager/rtpsession.h Normal file
View file

@ -0,0 +1,206 @@
/* GStreamer
* Copyright (C) <2007> Wim Taymans <wim@fluendo.com>
*
* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifndef __RTP_SESSION_H__
#define __RTP_SESSION_H__
#include <gst/gst.h>
#include <gst/netbuffer/gstnetbuffer.h>
#include "rtpsource.h"
typedef struct _RTPSession RTPSession;
typedef struct _RTPSessionClass RTPSessionClass;
#define RTP_TYPE_SESSION (rtp_session_get_type())
#define RTP_SESSION(sess) (G_TYPE_CHECK_INSTANCE_CAST((sess),RTP_TYPE_SESSION,RTPSession))
#define RTP_SESSION_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),RTP_TYPE_SESSION,RTPSessionClass))
#define RTP_IS_SESSION(sess) (G_TYPE_CHECK_INSTANCE_TYPE((sess),RTP_TYPE_SESSION))
#define RTP_IS_SESSION_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),RTP_TYPE_SESSION))
#define RTP_SESSION_CAST(sess) ((RTPSession *)(sess))
#define RTP_SESSION_LOCK(sess) (g_mutex_lock ((sess)->lock))
#define RTP_SESSION_UNLOCK(sess) (g_mutex_unlock ((sess)->lock))
/**
* RTPSessionProcessRTP:
* @sess: an #RTPSession
* @src: the #RTPSource
* @buffer: the RTP buffer ready for processing
* @user_data: user data specified when registering
*
* This callback will be called when @sess has @buffer ready for further
* processing. Processing the buffer typically includes decoding and displaying
* the buffer.
*
* Returns: a #GstFlowReturn.
*/
typedef GstFlowReturn (*RTPSessionProcessRTP) (RTPSession *sess, RTPSource *src, GstBuffer *buffer, gpointer user_data);
/**
* RTPSessionSendRTP:
* @sess: an #RTPSession
* @src: the #RTPSource
* @buffer: the RTP buffer ready for sending
* @user_data: user data specified when registering
*
* This callback will be called when @sess has @buffer ready for sending to
* all listening participants in this session.
*
* Returns: a #GstFlowReturn.
*/
typedef GstFlowReturn (*RTPSessionSendRTP) (RTPSession *sess, RTPSource *src, GstBuffer *buffer, gpointer user_data);
/**
* RTPSessionSendRTCP:
* @sess: an #RTPSession
* @src: the #RTPSource
* @buffer: the RTCP buffer ready for sending
* @user_data: user data specified when registering
*
* This callback will be called when @sess has @buffer ready for sending to
* all listening participants in this session.
*
* Returns: a #GstFlowReturn.
*/
typedef GstFlowReturn (*RTPSessionSendRTCP) (RTPSession *sess, RTPSource *src, GstBuffer *buffer, gpointer user_data);
/**
* RTPSessionClockRate:
* @sess: an #RTPSession
* @payload: the payload
* @user_data: user data specified when registering
*
* This callback will be called when @sess needs the clock-rate of @payload.
*
* Returns: the clock-rate of @pt.
*/
typedef gint (*RTPSessionClockRate) (RTPSession *sess, guint8 payload, gpointer user_data);
/**
* RTPSessionGetTime:
* @sess: an #RTPSession
* @user_data: user data specified when registering
*
* This callback will be called when @sess needs the current time in
* nanoseconds.
*
* Returns: a #GstClockTime with the current time in nanoseconds.
*/
typedef GstClockTime (*RTPSessionGetTime) (RTPSession *sess, gpointer user_data);
/**
* RTPSessionCallbacks:
* @RTPSessionProcessRTP: callback to process RTP packets
* @RTPSessionSendRTP: callback for sending RTP packets
* @RTPSessionSendRTCP: callback for sending RTCP packets
* @RTPSessionGetTime: callback for returning the current time
*
* These callbacks can be installed on the session manager to get notification
* when RTP and RTCP packets are ready for further processing. These callbacks
* are not implemented with signals for performance reasons.
*/
typedef struct {
RTPSessionProcessRTP process_rtp;
RTPSessionSendRTP send_rtp;
RTPSessionSendRTCP send_rtcp;
RTPSessionClockRate clock_rate;
RTPSessionGetTime get_time;
} RTPSessionCallbacks;
/**
* RTPSession:
* @lock: lock to protect the session
* @source: the source of this session
* @ssrcs: Hashtable of sources indexed by SSRC
* @cnames: Hashtable of sources indexed by CNAME
* @num_sources: the number of sources
* @activecount: the number of active sources
* @callbacks: callbacks
* @user_data: user data passed in callbacks
*
* The RTP session manager object
*/
struct _RTPSession {
GObject object;
GMutex *lock;
guint header_len;
RTPSource *source;
GHashTable *ssrcs;
GHashTable *cnames;
guint total_sources;
RTPSessionCallbacks callbacks;
gpointer user_data;
RTPSessionStats stats;
};
/**
* RTPSessionClass:
* @on_new_ssrc: emited when a new source is found
* @on_bye_ssrc: emited when a source is gone
*
* The session class.
*/
struct _RTPSessionClass {
GObjectClass parent_class;
/* signals */
void (*on_new_ssrc) (RTPSession *sess, RTPSource *source);
void (*on_ssrc_collision) (RTPSession *sess, RTPSource *source);
void (*on_ssrc_validated) (RTPSession *sess, RTPSource *source);
void (*on_bye_ssrc) (RTPSession *sess, RTPSource *source);
};
GType rtp_session_get_type (void);
/* create and configure */
RTPSession* rtp_session_new (void);
void rtp_session_set_callbacks (RTPSession *sess,
RTPSessionCallbacks *callbacks,
gpointer user_data);
void rtp_session_set_bandwidth (RTPSession *sess, gdouble bandwidth);
gdouble rtp_session_get_bandwidth (RTPSession *sess);
void rtp_session_set_rtcp_fraction (RTPSession *sess, gdouble fraction);
gdouble rtp_session_get_rtcp_fraction (RTPSession *sess);
/* handling sources */
gboolean rtp_session_add_source (RTPSession *sess, RTPSource *src);
gint rtp_session_get_num_sources (RTPSession *sess);
gint rtp_session_get_num_active_sources (RTPSession *sess);
RTPSource* rtp_session_get_source_by_ssrc (RTPSession *sess, guint32 ssrc);
RTPSource* rtp_session_get_source_by_cname (RTPSession *sess, const gchar *cname);
RTPSource* rtp_session_create_source (RTPSession *sess);
/* processing packets from receivers */
GstFlowReturn rtp_session_process_rtp (RTPSession *sess, GstBuffer *buffer);
GstFlowReturn rtp_session_process_rtcp (RTPSession *sess, GstBuffer *buffer);
/* processing packets for sending */
GstFlowReturn rtp_session_send_rtp (RTPSession *sess, GstBuffer *buffer);
/* get interval for next RTCP interval */
gdouble rtp_session_get_rtcp_interval (RTPSession *sess);
GstFlowReturn rtp_session_produce_rtcp (RTPSession *sess);
#endif /* __RTP_SESSION_H__ */

477
gst/rtpmanager/rtpsource.c Normal file
View file

@ -0,0 +1,477 @@
/* GStreamer
* Copyright (C) <2007> Wim Taymans <wim@fluendo.com>
*
* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/rtp/gstrtcpbuffer.h>
#include "rtpsource.h"
GST_DEBUG_CATEGORY_STATIC (rtp_source_debug);
#define GST_CAT_DEFAULT rtp_source_debug
#define RTP_MAX_PROBATION_LEN 32
/* signals and args */
enum
{
LAST_SIGNAL
};
enum
{
PROP_0
};
/* GObject vmethods */
static void rtp_source_finalize (GObject * object);
/* static guint rtp_source_signals[LAST_SIGNAL] = { 0 }; */
G_DEFINE_TYPE (RTPSource, rtp_source, G_TYPE_OBJECT);
static void
rtp_source_class_init (RTPSourceClass * klass)
{
GObjectClass *gobject_class;
gobject_class = (GObjectClass *) klass;
gobject_class->finalize = rtp_source_finalize;
GST_DEBUG_CATEGORY_INIT (rtp_source_debug, "rtpsource", 0, "RTP Source");
}
static void
rtp_source_init (RTPSource * src)
{
/* sources are initialy on probation until we receive enough valid RTP
* packets or a valid RTCP packet */
src->validated = FALSE;
src->probation = RTP_DEFAULT_PROBATION;
src->payload = 0;
src->clock_rate = -1;
src->packets = g_queue_new ();
src->stats.jitter = 0;
src->stats.transit = -1;
src->stats.curr_sr = 0;
src->stats.curr_rr = 0;
}
static void
rtp_source_finalize (GObject * object)
{
RTPSource *src;
GstBuffer *buffer;
src = RTP_SOURCE_CAST (object);
while ((buffer = g_queue_pop_head (src->packets)))
gst_buffer_unref (buffer);
g_queue_free (src->packets);
G_OBJECT_CLASS (rtp_source_parent_class)->finalize (object);
}
/**
* rtp_source_new:
* @ssrc: an SSRC
*
* Create a #RTPSource with @ssrc.
*
* Returns: a new #RTPSource. Use g_object_unref() after usage.
*/
RTPSource *
rtp_source_new (guint32 ssrc)
{
RTPSource *src;
src = g_object_new (RTP_TYPE_SOURCE, NULL);
src->ssrc = ssrc;
return src;
}
/**
* rtp_source_set_callbacks:
* @src: an #RTPSource
* @cb: callback functions
* @user_data: user data
*
* Set the callbacks for the source.
*/
void
rtp_source_set_callbacks (RTPSource * src, RTPSourceCallbacks * cb,
gpointer user_data)
{
g_return_if_fail (RTP_IS_SOURCE (src));
src->callbacks.push_rtp = cb->push_rtp;
src->callbacks.clock_rate = cb->clock_rate;
src->user_data = user_data;
}
/**
* rtp_source_set_as_csrc:
* @src: an #RTPSource
*
* Configure @src as a CSRC, this will validate the RTpSource.
*/
void
rtp_source_set_as_csrc (RTPSource * src)
{
g_return_if_fail (RTP_IS_SOURCE (src));
src->validated = TRUE;
src->is_csrc = TRUE;
}
/**
* rtp_source_set_rtp_from:
* @src: an #RTPSource
* @address: the RTP address to set
*
* Set that @src is receiving RTP packets from @address. This is used for
* collistion checking.
*/
void
rtp_source_set_rtp_from (RTPSource * src, GstNetAddress * address)
{
g_return_if_fail (RTP_IS_SOURCE (src));
src->have_rtp_from = TRUE;
memcpy (&src->rtp_from, address, sizeof (GstNetAddress));
}
/**
* rtp_source_set_rtcp_from:
* @src: an #RTPSource
* @address: the RTCP address to set
*
* Set that @src is receiving RTCP packets from @address. This is used for
* collistion checking.
*/
void
rtp_source_set_rtcp_from (RTPSource * src, GstNetAddress * address)
{
g_return_if_fail (RTP_IS_SOURCE (src));
src->have_rtcp_from = TRUE;
memcpy (&src->rtcp_from, address, sizeof (GstNetAddress));
}
static GstFlowReturn
push_packet (RTPSource * src, GstBuffer * buffer)
{
GstFlowReturn ret = GST_FLOW_OK;
/* push queued packets first if any */
while (!g_queue_is_empty (src->packets)) {
GstBuffer *buffer = GST_BUFFER_CAST (g_queue_pop_head (src->packets));
GST_DEBUG ("pushing queued packet");
if (src->callbacks.push_rtp)
src->callbacks.push_rtp (src, buffer, src->user_data);
else
gst_buffer_unref (buffer);
}
GST_DEBUG ("pushing new packet");
/* push packet */
if (src->callbacks.push_rtp)
ret = src->callbacks.push_rtp (src, buffer, src->user_data);
else
gst_buffer_unref (buffer);
return ret;
}
static gint
get_clock_rate (RTPSource * src, guint8 payload)
{
if (payload != src->payload) {
gint clock_rate = -1;
if (src->callbacks.clock_rate)
clock_rate = src->callbacks.clock_rate (src, payload, src->user_data);
GST_DEBUG ("new payload %d, got clock-rate %d", payload, clock_rate);
src->clock_rate = clock_rate;
src->payload = payload;
}
return src->clock_rate;
}
static void
calculate_jitter (RTPSource * src, GstBuffer * buffer,
RTPArrivalStats * arrival)
{
GstClockTime current;
guint32 rtparrival, transit, rtptime;
gint32 diff;
gint clock_rate;
guint8 pt;
/* get arrival time */
if ((current = arrival->time) == GST_CLOCK_TIME_NONE)
goto no_time;
pt = gst_rtp_buffer_get_payload_type (buffer);
/* get clockrate */
if ((clock_rate = get_clock_rate (src, pt)) == -1)
goto no_clock_rate;
rtptime = gst_rtp_buffer_get_timestamp (buffer);
/* convert arrival time to RTP timestamp units */
rtparrival = gst_util_uint64_scale_int (current, clock_rate, GST_SECOND);
/* transit time is difference with RTP timestamp */
transit = rtparrival - rtptime;
/* get diff with previous transit time */
if (src->stats.transit != -1)
diff = transit - src->stats.transit;
else
diff = 0;
src->stats.transit = transit;
if (diff < 0)
diff = -diff;
/* update jitter */
src->stats.jitter += diff - ((src->stats.jitter + 8) >> 4);
src->stats.prev_rtptime = src->stats.last_rtptime;
src->stats.last_rtptime = rtparrival;
GST_DEBUG ("rtparrival %u, rtptime %u, clock-rate %d, diff %d, jitter: %u",
rtparrival, rtptime, clock_rate, diff, src->stats.jitter);
return;
/* ERRORS */
no_time:
{
GST_WARNING ("cannot get current time");
return;
}
no_clock_rate:
{
GST_WARNING ("cannot get clock-rate for pt %d", pt);
return;
}
}
/**
* rtp_source_process_rtp:
* @src: an #RTPSource
* @buffer: an RTP buffer
*
* Let @src handle the incomming RTP @buffer.
*
* Returns: a #GstFlowReturn.
*/
GstFlowReturn
rtp_source_process_rtp (RTPSource * src, GstBuffer * buffer,
RTPArrivalStats * arrival)
{
GstFlowReturn result = GST_FLOW_OK;
g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
/* if we are still on probation, check seqnum */
if (src->probation) {
guint16 seqnr, expected;
expected = src->stats.max_seqnr + 1;
/* when in probation, we require consecutive seqnums */
seqnr = gst_rtp_buffer_get_seq (buffer);
if (seqnr == expected) {
/* expected packet */
src->probation--;
src->stats.max_seqnr = seqnr;
GST_DEBUG ("probation: seqnr %d == expected %d", seqnr, expected);
} else {
GST_DEBUG ("probation: seqnr %d != expected %d", seqnr, expected);
src->probation = RTP_DEFAULT_PROBATION;
src->stats.max_seqnr = seqnr;
}
}
if (src->probation) {
GstBuffer *q;
GST_DEBUG ("probation %d: queue buffer", src->probation);
/* when still in probation, keep packets in a list. */
g_queue_push_tail (src->packets, buffer);
/* remove packets from queue if there are too many */
while (g_queue_get_length (src->packets) > RTP_MAX_PROBATION_LEN) {
q = g_queue_pop_head (src->packets);
gst_object_unref (q);
}
} else {
/* we are not in probation */
src->stats.octetsreceived += arrival->payload_len;
src->stats.bytesreceived += arrival->bytes;
src->stats.packetsreceived++;
src->is_sender = TRUE;
GST_DEBUG ("PC: %" G_GUINT64_FORMAT ", OC: %" G_GUINT64_FORMAT,
src->stats.packetsreceived, src->stats.octetsreceived);
/* calculate jitter */
calculate_jitter (src, buffer, arrival);
/* we're ready to push the RTP packet now */
result = push_packet (src, buffer);
}
return result;
}
/**
* rtp_source_process_bye:
* @src: an #RTPSource
* @reason: the reason for leaving
*
* Notify @src that a BYE packet has been received. This will make the source
* inactive.
*/
void
rtp_source_process_bye (RTPSource * src, const gchar * reason)
{
g_return_if_fail (RTP_IS_SOURCE (src));
GST_DEBUG ("marking SSRC %08x as BYE, reason: %s", src->ssrc,
GST_STR_NULL (reason));
/* copy the reason and mark as received_bye */
g_free (src->bye_reason);
src->bye_reason = g_strdup (reason);
src->received_bye = TRUE;
}
/**
* rtp_source_send_rtp:
* @src: an #RTPSource
* @buffer: an RTP buffer
*
* Send an RTP @buffer originating from @src. This will make @src a sender.
*
* Returns: a #GstFlowReturn.
*/
GstFlowReturn
rtp_source_send_rtp (RTPSource * src, GstBuffer * buffer)
{
GstFlowReturn result = GST_FLOW_OK;
g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
/* we are a sender now */
src->is_sender = TRUE;
/* push packet */
if (src->callbacks.push_rtp)
result = src->callbacks.push_rtp (src, buffer, src->user_data);
else
gst_buffer_unref (buffer);
return result;
}
/**
* rtp_source_process_sr:
* @src: an #RTPSource
* @ntptime: the NTP time
* @rtptime: the RTP time
* @packet_count: the packet count
* @octet_count: the octect count
*
* Update the sender report in @src.
*/
void
rtp_source_process_sr (RTPSource * src, guint64 ntptime, guint32 rtptime,
guint32 packet_count, guint32 octet_count)
{
RTPSenderReport *curr;
gint curridx;
g_return_if_fail (RTP_IS_SOURCE (src));
GST_DEBUG ("got SR packet: SSRC %08x, NTP %" G_GUINT64_FORMAT
", RTP %u, PC %u, OC %u", src->ssrc, ntptime, rtptime, packet_count,
octet_count);
curridx = src->stats.curr_sr ^ 1;
curr = &src->stats.sr[curridx];
/* update current */
curr->is_valid = TRUE;
curr->ntptime = ntptime;
curr->rtptime = rtptime;
curr->packet_count = packet_count;
curr->octet_count = octet_count;
/* make current */
src->stats.curr_sr = curridx;
}
/**
* rtp_source_process_rb:
* @src: an #RTPSource
* @fractionlost: fraction lost since last SR/RR
* @packetslost: the cumululative number of packets lost
* @exthighestseq: the extended last sequence number received
* @jitter: the interarrival jitter
* @lsr: the last SR packet from this source
* @dlsr: the delay since last SR packet
*
* Update the report block in @src.
*/
void
rtp_source_process_rb (RTPSource * src, guint8 fractionlost, gint32 packetslost,
guint32 exthighestseq, guint32 jitter, guint32 lsr, guint32 dlsr)
{
RTPReceiverReport *curr;
gint curridx;
g_return_if_fail (RTP_IS_SOURCE (src));
GST_DEBUG ("got RB packet %d: SSRC %08x, FL %u"
", PL %u, HS %u, JITTER %u, LSR %u, DLSR %u", src->ssrc, fractionlost,
packetslost, exthighestseq, jitter, lsr, dlsr);
curridx = src->stats.curr_rr ^ 1;
curr = &src->stats.rr[curridx];
/* update current */
curr->is_valid = TRUE;
curr->fractionlost = fractionlost;
curr->packetslost = packetslost;
curr->exthighestseq = exthighestseq;
curr->jitter = jitter;
curr->lsr = lsr;
curr->dlsr = dlsr;
/* make current */
src->stats.curr_rr = curridx;
}

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/* GStreamer
* Copyright (C) <2007> Wim Taymans <wim@fluendo.com>
*
* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifndef __RTP_SOURCE_H__
#define __RTP_SOURCE_H__
#include <gst/gst.h>
#include <gst/rtp/gstrtcpbuffer.h>
#include <gst/netbuffer/gstnetbuffer.h>
#include "rtpstats.h"
/* the default number of consecutive RTP packets we need to receive before the
* source is considered valid */
#define RTP_NO_PROBATION 0
#define RTP_DEFAULT_PROBATION 2
typedef struct _RTPSource RTPSource;
typedef struct _RTPSourceClass RTPSourceClass;
#define RTP_TYPE_SOURCE (rtp_source_get_type())
#define RTP_SOURCE(src) (G_TYPE_CHECK_INSTANCE_CAST((src),RTP_TYPE_SOURCE,RTPSource))
#define RTP_SOURCE_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),RTP_TYPE_SOURCE,RTPSourceClass))
#define RTP_IS_SOURCE(src) (G_TYPE_CHECK_INSTANCE_TYPE((src),RTP_TYPE_SOURCE))
#define RTP_IS_SOURCE_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),RTP_TYPE_SOURCE))
#define RTP_SOURCE_CAST(src) ((RTPSource *)(src))
/**
* RTP_SOURCE_IS_ACTIVE:
* @src: an #RTPSource
*
* Check if @src is active. A source is active when it has been validated
* and has not yet received a BYE packet.
*/
#define RTP_SOURCE_IS_ACTIVE(src) (src->validated && !src->received_bye)
/**
* RTP_SOURCE_IS_SENDER:
* @src: an #RTPSource
*
* Check if @src is a sender.
*/
#define RTP_SOURCE_IS_SENDER(src) (src->is_sender)
/**
* RTPSourcePushRTP:
* @src: an #RTPSource
* @buffer: the RTP buffer ready for processing
* @user_data: user data specified when registering
*
* This callback will be called when @src has @buffer ready for further
* processing.
*
* Returns: a #GstFlowReturn.
*/
typedef GstFlowReturn (*RTPSourcePushRTP) (RTPSource *src, GstBuffer *buffer, gpointer user_data);
/**
* RTPSourceClockRate:
* @src: an #RTPSource
* @payload: a payload type
* @user_data: user data specified when registering
*
* This callback will be called when @src needs the clock-rate of the
* @payload.
*
* Returns: a clock-rate for @payload.
*/
typedef gint (*RTPSourceClockRate) (RTPSource *src, guint8 payload, gpointer user_data);
/**
* RTPSourceCallbacks:
* @push_rtp: a packet becomes available for handling
* @clock_rate: a clock-rate is requested
* @get_time: the current clock time is requested
*
* Callbacks performed by #RTPSource when actions need to be performed.
*/
typedef struct {
RTPSourcePushRTP push_rtp;
RTPSourceClockRate clock_rate;
} RTPSourceCallbacks;
/**
* RTPSource:
*
* A source in the #RTPSession
*/
struct _RTPSource {
GObject object;
/*< private >*/
RTPSourceCallbacks callbacks;
gpointer user_data;
guint32 ssrc;
gchar *cname;
gint probation;
gboolean validated;
gboolean received_bye;
gchar *bye_reason;
gboolean is_csrc;
gboolean is_sender;
gboolean have_rtp_from;
GstNetAddress rtp_from;
gboolean have_rtcp_from;
GstNetAddress rtcp_from;
guint8 payload;
gint clock_rate;
GQueue *packets;
RTPSourceStats stats;
};
struct _RTPSourceClass {
GObjectClass parent_class;
};
GType rtp_source_get_type (void);
/* managing lifetime of sources */
RTPSource* rtp_source_new (guint32 ssrc);
void rtp_source_set_callbacks (RTPSource *src, RTPSourceCallbacks *cb, gpointer data);
void rtp_source_set_as_csrc (RTPSource *src);
void rtp_source_set_rtp_from (RTPSource *src, GstNetAddress *address);
void rtp_source_set_rtcp_from (RTPSource *src, GstNetAddress *address);
GstFlowReturn rtp_source_process_rtp (RTPSource *src, GstBuffer *buffer, RTPArrivalStats *arrival);
GstFlowReturn rtp_source_send_rtp (RTPSource *src, GstBuffer *buffer);
/* RTCP messages */
void rtp_source_process_bye (RTPSource *src, const gchar *reason);
void rtp_source_process_sr (RTPSource *src, guint64 ntptime, guint32 rtptime,
guint32 packet_count, guint32 octet_count);
void rtp_source_process_rb (RTPSource *src, guint8 fractionlost, gint32 packetslost,
guint32 exthighestseq, guint32 jitter,
guint32 lsr, guint32 dlsr);
#endif /* __RTP_SOURCE_H__ */

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/* GStreamer
* Copyright (C) <2007> Wim Taymans <wim@fluendo.com>
*
* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#include "rtpstats.h"
/**
* rtp_stats_init_defaults:
* @stats: an #RTPSessionStats struct
*
* Initialize @stats with its default values.
*/
void
rtp_stats_init_defaults (RTPSessionStats * stats)
{
stats->bandwidth = RTP_STATS_BANDWIDTH;
stats->sender_fraction = RTP_STATS_SENDER_FRACTION;
stats->receiver_fraction = RTP_STATS_RECEIVER_FRACTION;
stats->rtcp_bandwidth = RTP_STATS_RTCP_BANDWIDTH;
stats->min_interval = RTP_STATS_MIN_INTERVAL;
}
/**
* rtp_stats_calculate_rtcp_interval:
* @stats: an #RTPSessionStats struct
*
* Calculate the RTCP interval. The result of this function is the amount of
* time to wait (in seconds) before sender a new RTCP message.
*
* Returns: the RTCP interval.
*/
gdouble
rtp_stats_calculate_rtcp_interval (RTPSessionStats * stats, gboolean sender)
{
gdouble active, senders, receivers, sfraction;
gboolean avg_rtcp;
gdouble interval;
active = stats->active_sources;
/* Try to avoid division by zero */
if (stats->active_sources == 0)
active += 1.0;
senders = (gdouble) stats->sender_sources;
receivers = (gdouble) (active - senders);
avg_rtcp = (gdouble) stats->avg_rtcp_packet_size;
sfraction = senders / active;
GST_DEBUG ("senders: %f, receivers %f, avg_rtcp %f, sfraction %f",
senders, receivers, avg_rtcp, sfraction);
if (sfraction <= stats->sender_fraction) {
if (sender) {
interval =
(avg_rtcp * senders) / (stats->sender_fraction *
stats->rtcp_bandwidth);
} else {
interval =
(avg_rtcp * receivers) / ((1.0 -
stats->sender_fraction) * stats->rtcp_bandwidth);
}
} else {
interval = (avg_rtcp * active) / stats->rtcp_bandwidth;
}
if (interval < stats->min_interval)
interval = stats->min_interval;
if (!stats->sent_rtcp)
interval /= 2.0;
return interval;
}
/**
* rtp_stats_calculate_rtcp_interval:
* @stats: an #RTPSessionStats struct
* @interval: an RTCP interval
*
* Apply a random jitter to the @interval. @interval is typically obtained with
* rtp_stats_calculate_rtcp_interval().
*
* Returns: the new RTCP interval.
*/
gdouble
rtp_stats_add_rtcp_jitter (RTPSessionStats * stats, gdouble interval)
{
/* see RFC 3550 p 30
* To compensate for "unconditional reconsideration" converging to a
* value below the intended average.
*/
#define COMPENSATION (2.71828 - 1.5);
return (interval * g_random_double_range (0.5, 1.5)) / COMPENSATION;
}

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/* GStreamer
* Copyright (C) <2007> Wim Taymans <wim@fluendo.com>
*
* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifndef __RTP_STATS_H__
#define __RTP_STATS_H__
#include <gst/gst.h>
#include <gst/netbuffer/gstnetbuffer.h>
/**
* RTPSenderReport:
*
* A sender report structure.
*/
typedef struct {
gboolean is_valid;
guint64 ntptime;
guint32 rtptime;
guint32 packet_count;
guint32 octet_count;
} RTPSenderReport;
/**
* RTPReceiverReport:
*
* A receiver report structure.
*/
typedef struct {
gboolean is_valid;
guint32 ssrc; /* who the report is from */
guint8 fractionlost;
guint32 packetslost;
guint32 exthighestseq;
guint32 jitter;
guint32 lsr;
guint32 dlsr;
} RTPReceiverReport;
/**
* RTPArrivalStats:
* @time: arrival time of a packet
* @address: address of the sender of the packet
* @bytes: bytes of the packet including lowlevel overhead
* @payload_len: bytes of the RTP payload
*
* Structure holding information about the arrival stats of a packet.
*/
typedef struct {
GstClockTime time;
gboolean have_address;
GstNetAddress address;
guint bytes;
guint payload_len;
} RTPArrivalStats;
/**
* RTPSourceStats:
* @packetsreceived: number of received packets in total
* @prevpacketsreceived: number of packets received in previous reporting
* interval
* @octetsreceived: number of payload bytes received
* @bytesreceived: number of total bytes received including headers and lower
* protocol level overhead
* @max_seqnr: highest sequence number received
* @transit: previous transit time used for calculating @jitter
* @jitter: current jitter
* @prev_rtptime: previous time when an RTP packet was received
* @prev_rtcptime: previous time when an RTCP packet was received
* @last_rtptime: time when last RTP packet received
* @last_rtcptime: time when last RTCP packet received
* @curr_rr: index of current @rr block
* @rr: previous and current receiver report block
* @curr_sr: index of current @sr block
* @sr: previous and current sender report block
*
* Stats about a source.
*/
typedef struct {
guint64 packetsreceived;
guint64 prevpacketsreceived;
guint64 octetsreceived;
guint64 bytesreceived;
guint16 max_seqnr;
guint32 transit;
guint32 jitter;
/* when we received stuff */
GstClockTime prev_rtptime;
GstClockTime prev_rtcptime;
GstClockTime last_rtptime;
GstClockTime last_rtcptime;
/* sender and receiver reports */
gint curr_rr;
RTPReceiverReport rr[2];
gint curr_sr;
RTPSenderReport sr[2];
} RTPSourceStats;
#define RTP_STATS_BANDWIDTH 64000.0
#define RTP_STATS_RTCP_BANDWIDTH 3000.0
/*
* Minimum average time between RTCP packets from this site (in
* seconds). This time prevents the reports from `clumping' when
* sessions are small and the law of large numbers isn't helping
* to smooth out the traffic. It also keeps the report interval
* from becoming ridiculously small during transient outages like
* a network partition.
*/
#define RTP_STATS_MIN_INTERVAL 5.0
/*
* Fraction of the RTCP bandwidth to be shared among active
* senders. (This fraction was chosen so that in a typical
* session with one or two active senders, the computed report
* time would be roughly equal to the minimum report time so that
* we don't unnecessarily slow down receiver reports.) The
* receiver fraction must be 1 - the sender fraction.
*/
#define RTP_STATS_SENDER_FRACTION (0.25)
#define RTP_STATS_RECEIVER_FRACTION (1.0 - RTP_STATS_SENDER_FRACTION)
/**
* RTPSessionStats:
*
* Stats kept for a session and used to produce RTCP packet timeouts.
*/
typedef struct {
gdouble bandwidth;
gdouble sender_fraction;
gdouble receiver_fraction;
gdouble rtcp_bandwidth;
gdouble min_interval;
guint sender_sources;
guint active_sources;
guint avg_rtcp_packet_size;
guint avg_bye_packet_size;
gboolean sent_rtcp;
} RTPSessionStats;
void rtp_stats_init_defaults (RTPSessionStats *stats);
gdouble rtp_stats_calculate_rtcp_interval (RTPSessionStats *stats, gboolean sender);
gdouble rtp_stats_add_rtcp_jitter (RTPSessionStats *stats, gdouble interval);
#endif /* __RTP_STATS_H__ */