</SIGNAL>
<SIGNAL>
+<NAME>GstRtpBin::on-bundled-ssrc</NAME>
+<RETURNS>guint</RETURNS>
+<FLAGS>l</FLAGS>
+GstRtpBin *gstrtpbin
+guint arg1
+</SIGNAL>
+
+<SIGNAL>
<NAME>GstRtpJitterBuffer::clear-pt-map</NAME>
<RETURNS>void</RETURNS>
<FLAGS>la</FLAGS>
* SSRC in the RTP packets to its own SSRC and wil forward the packets on the
* send_rtp_src_\%u pad after updating its internal state.
*
+ * #GstRtpBin can also demultiplex incoming bundled streams. The first
+ * #GstRtpSession will have a #GstRtpSsrcDemux element splitting the streams
+ * based on their SSRC and potentially dispatched to a different #GstRtpSession.
+ * Because retransmission SSRCs need to be merged with the corresponding media
+ * stream the #GstRtpBin::on-bundled-ssrc signal is emitted so that the
+ * application can find out to which session the SSRC belongs.
+ *
* The session manager needs the clock-rate of the payload types it is handling
* and will signal the #GstRtpSession::request-pt-map signal when it needs such a
* mapping. One can clear the cached values with the #GstRtpSession::clear-pt-map
SIGNAL_ON_NEW_SENDER_SSRC,
SIGNAL_ON_SENDER_SSRC_ACTIVE,
+ SIGNAL_ON_BUNDLED_SSRC,
+
LAST_SIGNAL
};
static void remove_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session);
static void free_client (GstRtpBinClient * client, GstRtpBin * bin);
static void free_stream (GstRtpBinStream * stream, GstRtpBin * bin);
+static GstRtpBinSession *create_session (GstRtpBin * rtpbin, gint id);
+static GstPad *complete_session_sink (GstRtpBin * rtpbin,
+ GstRtpBinSession * session, gboolean bundle_demuxer_needed);
+static void
+complete_session_receiver (GstRtpBin * rtpbin, GstRtpBinSession * session,
+ guint sessid);
+static GstPad *complete_session_rtcp (GstRtpBin * rtpbin,
+ GstRtpBinSession * session, guint sessid, gboolean bundle_demuxer_needed);
/* Manages the RTP stream for one SSRC.
*
gulong demux_newpad_sig;
gulong demux_padremoved_sig;
+ /* Bundling support */
+ GstElement *rtp_funnel;
+ GstElement *rtcp_funnel;
+ GstElement *bundle_demux;
+ gulong bundle_demux_newpad_sig;
+
GMutex lock;
/* list of GstRtpBinStream */
GST_RTP_BIN_UNLOCK (rtpbin);
}
+static void
+new_bundled_ssrc_pad_found (GstElement * element, guint ssrc, GstPad * pad,
+ GstRtpBinSession * session)
+{
+ GValue result = G_VALUE_INIT;
+ GValue params[2] = { G_VALUE_INIT, G_VALUE_INIT };
+ guint session_id = 0;
+ GstRtpBinSession *target_session = NULL;
+ GstRtpBin *rtpbin = session->bin;
+ gchar *name;
+ GstPad *src_pad;
+ GstPad *recv_rtp_sink = NULL;
+ GstPad *recv_rtcp_sink = NULL;
+ GstPadLinkReturn ret;
+
+ GST_RTP_BIN_DYN_LOCK (rtpbin);
+ GST_DEBUG_OBJECT (rtpbin, "new bundled SSRC pad %08x, %s:%s", ssrc,
+ GST_DEBUG_PAD_NAME (pad));
+
+ g_value_init (&result, G_TYPE_UINT);
+ g_value_init (¶ms[0], GST_TYPE_ELEMENT);
+ g_value_set_object (¶ms[0], rtpbin);
+ g_value_init (¶ms[1], G_TYPE_UINT);
+ g_value_set_uint (¶ms[1], ssrc);
+
+ g_signal_emitv (params,
+ gst_rtp_bin_signals[SIGNAL_ON_BUNDLED_SSRC], 0, &result);
+ g_value_unset (¶ms[0]);
+
+ session_id = g_value_get_uint (&result);
+ if (session_id == 0) {
+ target_session = session;
+ } else {
+ target_session = find_session_by_id (rtpbin, (gint) session_id);
+ if (!target_session) {
+ target_session = create_session (rtpbin, session_id);
+ }
+ if (!target_session->recv_rtp_sink) {
+ recv_rtp_sink = complete_session_sink (rtpbin, target_session, FALSE);
+ }
+
+ if (!target_session->recv_rtp_src)
+ complete_session_receiver (rtpbin, target_session, session_id);
+
+ if (!target_session->recv_rtcp_sink) {
+ recv_rtcp_sink =
+ complete_session_rtcp (rtpbin, target_session, session_id, FALSE);
+ }
+ }
+
+ GST_DEBUG_OBJECT (rtpbin, "Assigning bundled ssrc %u to session %u", ssrc,
+ session_id);
+
+ if (!recv_rtp_sink) {
+ recv_rtp_sink =
+ gst_element_get_request_pad (target_session->rtp_funnel, "sink_%u");
+ }
+
+ if (!recv_rtcp_sink) {
+ recv_rtcp_sink =
+ gst_element_get_request_pad (target_session->rtcp_funnel, "sink_%u");
+ }
+
+ name = g_strdup_printf ("src_%u", ssrc);
+ src_pad = gst_element_get_static_pad (element, name);
+ ret = gst_pad_link (src_pad, recv_rtp_sink);
+ g_free (name);
+ gst_object_unref (src_pad);
+ gst_object_unref (recv_rtp_sink);
+ if (ret != GST_PAD_LINK_OK) {
+ g_warning
+ ("rtpbin: failed to link bundle demuxer to receive rtp funnel for session %u",
+ session_id);
+ }
+
+ name = g_strdup_printf ("rtcp_src_%u", ssrc);
+ src_pad = gst_element_get_static_pad (element, name);
+ gst_pad_link (src_pad, recv_rtcp_sink);
+ g_free (name);
+ gst_object_unref (src_pad);
+ gst_object_unref (recv_rtcp_sink);
+ if (ret != GST_PAD_LINK_OK) {
+ g_warning
+ ("rtpbin: failed to link bundle demuxer to receive rtcp sink pad for session %u",
+ session_id);
+ }
+
+ GST_RTP_BIN_DYN_UNLOCK (rtpbin);
+}
+
/* create a session with the given id. Must be called with RTP_BIN_LOCK */
static GstRtpBinSession *
create_session (GstRtpBin * rtpbin, gint id)
sess->bin = rtpbin;
sess->session = session;
sess->demux = demux;
+
+ sess->rtp_funnel = gst_element_factory_make ("funnel", NULL);
+ sess->rtcp_funnel = gst_element_factory_make ("funnel", NULL);
+
sess->ptmap = g_hash_table_new_full (NULL, NULL, NULL,
(GDestroyNotify) gst_caps_unref);
rtpbin->sessions = g_slist_prepend (rtpbin->sessions, sess);
gst_bin_add (GST_BIN_CAST (rtpbin), session);
gst_bin_add (GST_BIN_CAST (rtpbin), demux);
+ gst_bin_add (GST_BIN_CAST (rtpbin), sess->rtp_funnel);
+ gst_bin_add (GST_BIN_CAST (rtpbin), sess->rtcp_funnel);
GST_OBJECT_LOCK (rtpbin);
target = GST_STATE_TARGET (rtpbin);
/* change state only to what's needed */
gst_element_set_state (demux, target);
gst_element_set_state (session, target);
+ gst_element_set_state (sess->rtp_funnel, target);
+ gst_element_set_state (sess->rtcp_funnel, target);
return sess;
GValue ret = { 0 };
GValue args[3] = { {0}, {0}, {0} };
- GST_DEBUG ("searching pt %d in cache", pt);
+ GST_DEBUG ("searching pt %u in cache", pt);
GST_RTP_SESSION_LOCK (session);
bin = session->bin;
- GST_DEBUG ("emiting signal for pt %d in session %d", pt, session->id);
+ GST_DEBUG ("emiting signal for pt %u in session %u", pt, session->id);
/* not in cache, send signal to request caps */
g_value_init (&args[0], GST_TYPE_ELEMENT);
if (!caps)
goto no_caps;
- GST_DEBUG ("caching pt %d as %" GST_PTR_FORMAT, pt, caps);
+ GST_DEBUG ("caching pt %u as %" GST_PTR_FORMAT, pt, caps);
/* store in cache, take additional ref */
g_hash_table_insert (session->ptmap, GINT_TO_POINTER (pt),
GstElement *ret = NULL;
GST_RTP_BIN_LOCK (bin);
- GST_DEBUG_OBJECT (bin, "retrieving GstRtpSession, index: %d", session_id);
+ GST_DEBUG_OBJECT (bin, "retrieving GstRtpSession, index: %u", session_id);
session = find_session_by_id (bin, (gint) session_id);
if (session) {
ret = gst_object_ref (session->session);
GstRtpBinSession *session;
GST_RTP_BIN_LOCK (bin);
- GST_DEBUG_OBJECT (bin, "retrieving internal RTPSession object, index: %d",
+ GST_DEBUG_OBJECT (bin, "retrieving internal RTPSession object, index: %u",
session_id);
session = find_session_by_id (bin, (gint) session_id);
if (session) {
on_sender_ssrc_active), NULL, NULL, g_cclosure_marshal_generic,
G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT);
+
+ /**
+ * GstRtpBin::on-bundled-ssrc:
+ * @rtpbin: the object which received the signal
+ * @ssrc: the bundled SSRC
+ *
+ * Notify of a new incoming bundled SSRC. If no handler is connected to the
+ * signal then the #GstRtpSession created for the recv_rtp_sink_\%u
+ * request pad will be managing this new SSRC. However if there is a handler
+ * connected then the application can decided to dispatch this new stream to
+ * another session by providing its ID as return value of the handler. This
+ * can be particularly useful to keep retransmission SSRCs grouped with the
+ * session for which they handle retransmission.
+ *
+ * Since: 1.12
+ */
+ gst_rtp_bin_signals[SIGNAL_ON_BUNDLED_SSRC] =
+ g_signal_new ("on-bundled-ssrc", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
+ on_bundled_ssrc), NULL, NULL,
+ g_cclosure_marshal_generic, G_TYPE_UINT, 1, G_TYPE_UINT);
+
+
g_object_class_install_property (gobject_class, PROP_SDES,
g_param_spec_boxed ("sdes", "SDES",
"The SDES items of this session",
rtpbin = stream->bin;
- GST_DEBUG ("new payload pad %d", pt);
+ GST_DEBUG_OBJECT (rtpbin, "new payload pad %u", pt);
GST_RTP_BIN_SHUTDOWN_LOCK (rtpbin, shutdown);
rtpbin = session->bin;
- GST_DEBUG_OBJECT (rtpbin, "payload map requested for pt %d in session %d", pt,
+ GST_DEBUG_OBJECT (rtpbin, "payload map requested for pt %u in session %u", pt,
session->id);
caps = get_pt_map (session, pt);
payload_type_change (GstElement * element, guint pt, GstRtpBinSession * session)
{
GST_DEBUG_OBJECT (session->bin,
- "emiting signal for pt type changed to %d in session %d", pt,
+ "emiting signal for pt type changed to %u in session %u", pt,
session->id);
g_signal_emit (session->bin, gst_rtp_bin_signals[SIGNAL_PAYLOAD_TYPE_CHANGE],
}
}
-static gboolean
-complete_session_sink (GstRtpBin * rtpbin, GstRtpBinSession * session)
+static void
+session_maybe_create_bundle_demuxer (GstRtpBinSession * session)
+{
+ GstRtpBin *rtpbin;
+
+ if (session->bundle_demux)
+ return;
+
+ rtpbin = session->bin;
+ if (g_signal_has_handler_pending (rtpbin,
+ gst_rtp_bin_signals[SIGNAL_ON_BUNDLED_SSRC], 0, TRUE)) {
+ GST_DEBUG_OBJECT (rtpbin, "Adding a bundle SSRC demuxer to session %u",
+ session->id);
+ session->bundle_demux = gst_element_factory_make ("rtpssrcdemux", NULL);
+ session->bundle_demux_newpad_sig = g_signal_connect (session->bundle_demux,
+ "new-ssrc-pad", (GCallback) new_bundled_ssrc_pad_found, session);
+
+ gst_bin_add (GST_BIN_CAST (rtpbin), session->bundle_demux);
+ gst_element_sync_state_with_parent (session->bundle_demux);
+ } else {
+ GST_DEBUG_OBJECT (rtpbin,
+ "No handler for the on-bundled-ssrc signal so no need for a bundle SSRC demuxer in session %u",
+ session->id);
+ }
+}
+
+static GstPad *
+complete_session_sink (GstRtpBin * rtpbin, GstRtpBinSession * session,
+ gboolean bundle_demuxer_needed)
{
- gchar *gname;
guint sessid = session->id;
GstPad *recv_rtp_sink;
+ GstPad *funnel_src;
GstElement *decoder;
- GstElementClass *klass;
- GstPadTemplate *templ;
+
+ g_assert (!session->recv_rtp_sink);
/* get recv_rtp pad and store */
session->recv_rtp_sink =
g_signal_connect (session->recv_rtp_sink, "notify::caps",
(GCallback) caps_changed, session);
+ if (bundle_demuxer_needed)
+ session_maybe_create_bundle_demuxer (session);
+
GST_DEBUG_OBJECT (rtpbin, "requesting RTP decoder");
decoder = session_request_element (session, SIGNAL_REQUEST_RTP_DECODER);
if (decoder) {
if (decsrc == NULL)
goto dec_src_failed;
- ret = gst_pad_link (decsrc, session->recv_rtp_sink);
+ if (session->bundle_demux) {
+ GstPad *demux_sink;
+ demux_sink = gst_element_get_static_pad (session->bundle_demux, "sink");
+ ret = gst_pad_link (decsrc, demux_sink);
+ gst_object_unref (demux_sink);
+ } else {
+ ret = gst_pad_link (decsrc, session->recv_rtp_sink);
+ }
gst_object_unref (decsrc);
if (ret != GST_PAD_LINK_OK)
} else {
GST_DEBUG_OBJECT (rtpbin, "no RTP decoder given");
- recv_rtp_sink = gst_object_ref (session->recv_rtp_sink);
+ if (session->bundle_demux) {
+ recv_rtp_sink =
+ gst_element_get_static_pad (session->bundle_demux, "sink");
+ } else {
+ recv_rtp_sink =
+ gst_element_get_request_pad (session->rtp_funnel, "sink_%u");
+ }
}
- GST_DEBUG_OBJECT (rtpbin, "ghosting session sink pad");
- klass = GST_ELEMENT_GET_CLASS (rtpbin);
- gname = g_strdup_printf ("recv_rtp_sink_%u", sessid);
- templ = gst_element_class_get_pad_template (klass, "recv_rtp_sink_%u");
- session->recv_rtp_sink_ghost =
- gst_ghost_pad_new_from_template (gname, recv_rtp_sink, templ);
- gst_object_unref (recv_rtp_sink);
- gst_pad_set_active (session->recv_rtp_sink_ghost, TRUE);
- gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->recv_rtp_sink_ghost);
- g_free (gname);
+ funnel_src = gst_element_get_static_pad (session->rtp_funnel, "src");
+ gst_pad_link (funnel_src, session->recv_rtp_sink);
+ gst_object_unref (funnel_src);
- return TRUE;
+ return recv_rtp_sink;
/* ERRORS */
pad_failed:
{
g_warning ("rtpbin: failed to get session recv_rtp_sink pad");
- return FALSE;
+ return NULL;
}
dec_sink_failed:
{
- g_warning ("rtpbin: failed to get decoder sink pad for session %d", sessid);
- return FALSE;
+ g_warning ("rtpbin: failed to get decoder sink pad for session %u", sessid);
+ return NULL;
}
dec_src_failed:
{
- g_warning ("rtpbin: failed to get decoder src pad for session %d", sessid);
+ g_warning ("rtpbin: failed to get decoder src pad for session %u", sessid);
gst_object_unref (recv_rtp_sink);
- return FALSE;
+ return NULL;
}
dec_link_failed:
{
- g_warning ("rtpbin: failed to link rtp decoder for session %d", sessid);
+ g_warning ("rtpbin: failed to link rtp decoder for session %u", sessid);
gst_object_unref (recv_rtp_sink);
- return FALSE;
+ return NULL;
}
}
-/* Create a pad for receiving RTP for the session in @name. Must be called with
- * RTP_BIN_LOCK.
- */
-static GstPad *
-create_recv_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
+static void
+complete_session_receiver (GstRtpBin * rtpbin, GstRtpBinSession * session,
+ guint sessid)
{
- guint sessid;
GstElement *aux;
GstPad *recv_rtp_src;
- GstRtpBinSession *session;
-
- /* first get the session number */
- if (name == NULL || sscanf (name, "recv_rtp_sink_%u", &sessid) != 1)
- goto no_name;
-
- GST_DEBUG_OBJECT (rtpbin, "finding session %d", sessid);
- /* get or create session */
- session = find_session_by_id (rtpbin, sessid);
- if (!session) {
- GST_DEBUG_OBJECT (rtpbin, "creating session %d", sessid);
- /* create session now */
- session = create_session (rtpbin, sessid);
- if (session == NULL)
- goto create_error;
- }
-
- /* check if pad was requested */
- if (session->recv_rtp_sink_ghost != NULL)
- return session->recv_rtp_sink_ghost;
-
- /* setup the session sink pad */
- if (!complete_session_sink (rtpbin, session))
- goto session_sink_failed;
+ g_assert (!session->recv_rtp_src);
session->recv_rtp_src =
gst_element_get_static_pad (session->session, "recv_rtp_src");
GST_DEBUG_OBJECT (rtpbin, "linking AUX receiver");
- pname = g_strdup_printf ("sink_%d", sessid);
+ pname = g_strdup_printf ("sink_%u", sessid);
auxsink = gst_element_get_static_pad (aux, pname);
g_free (pname);
if (auxsink == NULL)
/* this can be NULL when this AUX element is not to be linked to
* an SSRC demuxer */
- pname = g_strdup_printf ("src_%d", sessid);
+ pname = g_strdup_printf ("src_%u", sessid);
recv_rtp_src = gst_element_get_static_pad (aux, pname);
g_free (pname);
} else {
sinkdpad = gst_element_get_static_pad (session->demux, "sink");
GST_DEBUG_OBJECT (rtpbin, "linking demuxer RTP sink pad");
gst_pad_link_full (recv_rtp_src, sinkdpad, GST_PAD_LINK_CHECK_NOTHING);
- gst_object_unref (recv_rtp_src);
gst_object_unref (sinkdpad);
+ gst_object_unref (recv_rtp_src);
/* connect to the new-ssrc-pad signal of the SSRC demuxer */
session->demux_newpad_sig = g_signal_connect (session->demux,
session->demux_padremoved_sig = g_signal_connect (session->demux,
"removed-ssrc-pad", (GCallback) ssrc_demux_pad_removed, session);
}
+
+ return;
+
+pad_failed:
+ {
+ g_warning ("rtpbin: failed to get session recv_rtp_src pad");
+ return;
+ }
+aux_sink_failed:
+ {
+ g_warning ("rtpbin: failed to get AUX sink pad for session %u", sessid);
+ return;
+ }
+aux_link_failed:
+ {
+ g_warning ("rtpbin: failed to link AUX pad to session %u", sessid);
+ return;
+ }
+}
+
+/* Create a pad for receiving RTP for the session in @name. Must be called with
+ * RTP_BIN_LOCK.
+ */
+static GstPad *
+create_recv_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
+{
+ guint sessid;
+ GstRtpBinSession *session;
+ GstPad *recv_rtp_sink;
+
+ /* first get the session number */
+ if (name == NULL || sscanf (name, "recv_rtp_sink_%u", &sessid) != 1)
+ goto no_name;
+
+ GST_DEBUG_OBJECT (rtpbin, "finding session %u", sessid);
+
+ /* get or create session */
+ session = find_session_by_id (rtpbin, sessid);
+ if (!session) {
+ GST_DEBUG_OBJECT (rtpbin, "creating session %u", sessid);
+ /* create session now */
+ session = create_session (rtpbin, sessid);
+ if (session == NULL)
+ goto create_error;
+ }
+
+ /* check if pad was requested */
+ if (session->recv_rtp_sink_ghost != NULL)
+ return session->recv_rtp_sink_ghost;
+
+ /* setup the session sink pad */
+ recv_rtp_sink = complete_session_sink (rtpbin, session, TRUE);
+ if (!recv_rtp_sink)
+ goto session_sink_failed;
+
+
+ GST_DEBUG_OBJECT (rtpbin, "ghosting session sink pad");
+ session->recv_rtp_sink_ghost =
+ gst_ghost_pad_new_from_template (name, recv_rtp_sink, templ);
+ gst_object_unref (recv_rtp_sink);
+ gst_pad_set_active (session->recv_rtp_sink_ghost, TRUE);
+ gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->recv_rtp_sink_ghost);
+
+ complete_session_receiver (rtpbin, session, sessid);
+
return session->recv_rtp_sink_ghost;
/* ERRORS */
/* warning already done */
return NULL;
}
-pad_failed:
- {
- g_warning ("rtpbin: failed to get session recv_rtp_src pad");
- return NULL;
- }
-aux_sink_failed:
- {
- g_warning ("rtpbin: failed to get AUX sink pad for session %d", sessid);
- return NULL;
- }
-aux_link_failed:
- {
- g_warning ("rtpbin: failed to link AUX pad to session %d", sessid);
- return NULL;
- }
}
static void
g_signal_handler_disconnect (session->demux, session->demux_padremoved_sig);
session->demux_padremoved_sig = 0;
}
+ if (session->bundle_demux_newpad_sig) {
+ g_signal_handler_disconnect (session->bundle_demux,
+ session->bundle_demux_newpad_sig);
+ session->bundle_demux_newpad_sig = 0;
+ }
if (session->recv_rtp_src) {
gst_object_unref (session->recv_rtp_src);
session->recv_rtp_src = NULL;
}
}
-/* Create a pad for receiving RTCP for the session in @name. Must be called with
- * RTP_BIN_LOCK.
- */
static GstPad *
-create_recv_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ,
- const gchar * name)
+complete_session_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session,
+ guint sessid, gboolean bundle_demuxer_needed)
{
- guint sessid;
GstElement *decoder;
- GstRtpBinSession *session;
- GstPad *sinkdpad, *decsink;
-
- /* first get the session number */
- if (name == NULL || sscanf (name, "recv_rtcp_sink_%u", &sessid) != 1)
- goto no_name;
-
- GST_DEBUG_OBJECT (rtpbin, "finding session %d", sessid);
-
- /* get or create the session */
- session = find_session_by_id (rtpbin, sessid);
- if (!session) {
- GST_DEBUG_OBJECT (rtpbin, "creating session %d", sessid);
- /* create session now */
- session = create_session (rtpbin, sessid);
- if (session == NULL)
- goto create_error;
- }
-
- /* check if pad was requested */
- if (session->recv_rtcp_sink_ghost != NULL)
- return session->recv_rtcp_sink_ghost;
+ GstPad *sinkdpad;
+ GstPad *decsink = NULL;
+ GstPad *funnel_src;
/* get recv_rtp pad and store */
GST_DEBUG_OBJECT (rtpbin, "getting RTCP sink pad");
if (session->recv_rtcp_sink == NULL)
goto pad_failed;
+ if (bundle_demuxer_needed)
+ session_maybe_create_bundle_demuxer (session);
+
GST_DEBUG_OBJECT (rtpbin, "getting RTCP decoder");
decoder = session_request_element (session, SIGNAL_REQUEST_RTCP_DECODER);
if (decoder) {
if (decsrc == NULL)
goto dec_src_failed;
- ret = gst_pad_link (decsrc, session->recv_rtcp_sink);
+ if (session->bundle_demux) {
+ GstPad *demux_sink;
+ demux_sink =
+ gst_element_get_static_pad (session->bundle_demux, "rtcp_sink");
+ ret = gst_pad_link (decsrc, demux_sink);
+ gst_object_unref (demux_sink);
+ } else {
+ ret = gst_pad_link (decsrc, session->recv_rtcp_sink);
+ }
gst_object_unref (decsrc);
if (ret != GST_PAD_LINK_OK)
goto dec_link_failed;
} else {
GST_DEBUG_OBJECT (rtpbin, "no RTCP decoder given");
- decsink = gst_object_ref (session->recv_rtcp_sink);
+ if (session->bundle_demux) {
+ decsink = gst_element_get_static_pad (session->bundle_demux, "rtcp_sink");
+ } else {
+ decsink = gst_element_get_request_pad (session->rtcp_funnel, "sink_%u");
+ }
}
/* get srcpad, link to SSRCDemux */
gst_pad_link_full (session->sync_src, sinkdpad, GST_PAD_LINK_CHECK_NOTHING);
gst_object_unref (sinkdpad);
- session->recv_rtcp_sink_ghost =
- gst_ghost_pad_new_from_template (name, decsink, templ);
- gst_object_unref (decsink);
- gst_pad_set_active (session->recv_rtcp_sink_ghost, TRUE);
- gst_element_add_pad (GST_ELEMENT_CAST (rtpbin),
- session->recv_rtcp_sink_ghost);
+ funnel_src = gst_element_get_static_pad (session->rtcp_funnel, "src");
+ gst_pad_link (funnel_src, session->recv_rtcp_sink);
+ gst_object_unref (funnel_src);
- return session->recv_rtcp_sink_ghost;
+ return decsink;
- /* ERRORS */
-no_name:
- {
- g_warning ("rtpbin: invalid name given");
- return NULL;
- }
-create_error:
- {
- /* create_session already warned */
- return NULL;
- }
pad_failed:
{
g_warning ("rtpbin: failed to get session rtcp_sink pad");
}
dec_sink_failed:
{
- g_warning ("rtpbin: failed to get decoder sink pad for session %d", sessid);
+ g_warning ("rtpbin: failed to get decoder sink pad for session %u", sessid);
return NULL;
}
dec_src_failed:
{
- g_warning ("rtpbin: failed to get decoder src pad for session %d", sessid);
- gst_object_unref (decsink);
- return NULL;
+ g_warning ("rtpbin: failed to get decoder src pad for session %u", sessid);
+ goto cleanup;
}
dec_link_failed:
{
- g_warning ("rtpbin: failed to link rtcp decoder for session %d", sessid);
- gst_object_unref (decsink);
- return NULL;
+ g_warning ("rtpbin: failed to link rtcp decoder for session %u", sessid);
+ goto cleanup;
}
src_pad_failed:
{
g_warning ("rtpbin: failed to get session sync_src pad");
- gst_object_unref (decsink);
+ }
+
+cleanup:
+ gst_object_unref (decsink);
+ return NULL;
+}
+
+/* Create a pad for receiving RTCP for the session in @name. Must be called with
+ * RTP_BIN_LOCK.
+ */
+static GstPad *
+create_recv_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ,
+ const gchar * name)
+{
+ guint sessid;
+ GstRtpBinSession *session;
+ GstPad *decsink = NULL;
+
+ /* first get the session number */
+ if (name == NULL || sscanf (name, "recv_rtcp_sink_%u", &sessid) != 1)
+ goto no_name;
+
+ GST_DEBUG_OBJECT (rtpbin, "finding session %u", sessid);
+
+ /* get or create the session */
+ session = find_session_by_id (rtpbin, sessid);
+ if (!session) {
+ GST_DEBUG_OBJECT (rtpbin, "creating session %u", sessid);
+ /* create session now */
+ session = create_session (rtpbin, sessid);
+ if (session == NULL)
+ goto create_error;
+ }
+
+ /* check if pad was requested */
+ if (session->recv_rtcp_sink_ghost != NULL)
+ return session->recv_rtcp_sink_ghost;
+
+ decsink = complete_session_rtcp (rtpbin, session, sessid, TRUE);
+ if (!decsink)
+ goto create_error;
+
+ session->recv_rtcp_sink_ghost =
+ gst_ghost_pad_new_from_template (name, decsink, templ);
+ gst_object_unref (decsink);
+ gst_pad_set_active (session->recv_rtcp_sink_ghost, TRUE);
+ gst_element_add_pad (GST_ELEMENT_CAST (rtpbin),
+ session->recv_rtcp_sink_ghost);
+
+ return session->recv_rtcp_sink_ghost;
+
+ /* ERRORS */
+no_name:
+ {
+ g_warning ("rtpbin: invalid name given");
+ return NULL;
+ }
+create_error:
+ {
+ /* create_session already warned */
return NULL;
}
}
GstPadLinkReturn ret;
GST_DEBUG_OBJECT (rtpbin, "linking RTP encoder");
- ename = g_strdup_printf ("rtp_src_%d", sessid);
+ ename = g_strdup_printf ("rtp_src_%u", sessid);
encsrc = gst_element_get_static_pad (encoder, ename);
g_free (ename);
send_rtp_src = encsrc;
- ename = g_strdup_printf ("rtp_sink_%d", sessid);
+ ename = g_strdup_printf ("rtp_sink_%u", sessid);
encsink = gst_element_get_static_pad (encoder, ename);
g_free (ename);
if (encsink == NULL)
/* ERRORS */
no_srcpad:
{
- g_warning ("rtpbin: failed to get rtp source pad for session %d", sessid);
+ g_warning ("rtpbin: failed to get rtp source pad for session %u", sessid);
return FALSE;
}
enc_src_failed:
{
- g_warning ("rtpbin: failed to get encoder src pad for session %d", sessid);
+ g_warning ("rtpbin: failed to get encoder src pad for session %u", sessid);
return FALSE;
}
enc_sink_failed:
{
- g_warning ("rtpbin: failed to get encoder sink pad for session %d", sessid);
+ g_warning ("rtpbin: failed to get encoder sink pad for session %u", sessid);
gst_object_unref (send_rtp_src);
return FALSE;
}
enc_link_failed:
{
- g_warning ("rtpbin: failed to link rtp encoder for session %d", sessid);
+ g_warning ("rtpbin: failed to link rtp encoder for session %u", sessid);
gst_object_unref (send_rtp_src);
return FALSE;
}
}
existing_session:
{
- g_warning ("rtpbin: session %d is already a sender", sessid);
+ g_warning ("rtpbin: session %u is already a sender", sessid);
return FALSE;
}
pad_failed:
{
- g_warning ("rtpbin: failed to get session pad for session %d", sessid);
+ g_warning ("rtpbin: failed to get session pad for session %u", sessid);
return FALSE;
}
aux_link_failed:
{
- g_warning ("rtpbin: failed to link AUX for session %d", sessid);
+ g_warning ("rtpbin: failed to link AUX for session %u", sessid);
return FALSE;
}
session_src_failed:
{
- g_warning ("rtpbin: failed to complete AUX for session %d", sessid);
+ g_warning ("rtpbin: failed to complete AUX for session %u", sessid);
return FALSE;
}
}
if (!setup_aux_sender (rtpbin, session, aux))
goto aux_session_failed;
- pname = g_strdup_printf ("sink_%d", sessid);
+ pname = g_strdup_printf ("sink_%u", sessid);
send_rtp_sink = gst_element_get_static_pad (aux, pname);
g_free (pname);
}
existing_session:
{
- g_warning ("rtpbin: session %d is already in use", sessid);
+ g_warning ("rtpbin: session %u is already in use", sessid);
return NULL;
}
aux_session_failed:
{
- g_warning ("rtpbin: failed to get AUX sink pad for session %d", sessid);
+ g_warning ("rtpbin: failed to get AUX sink pad for session %u", sessid);
return NULL;
}
aux_sink_failed:
{
- g_warning ("rtpbin: failed to get AUX sink pad for session %d", sessid);
+ g_warning ("rtpbin: failed to get AUX sink pad for session %u", sessid);
return NULL;
}
pad_failed:
{
- g_warning ("rtpbin: failed to get session pad for session %d", sessid);
+ g_warning ("rtpbin: failed to get session pad for session %u", sessid);
return NULL;
}
session_src_failed:
{
- g_warning ("rtpbin: failed to setup source pads for session %d", sessid);
+ g_warning ("rtpbin: failed to setup source pads for session %u", sessid);
return NULL;
}
}
GST_DEBUG_OBJECT (rtpbin, "linking RTCP encoder");
- ename = g_strdup_printf ("rtcp_src_%d", sessid);
+ ename = g_strdup_printf ("rtcp_src_%u", sessid);
encsrc = gst_element_get_static_pad (encoder, ename);
g_free (ename);
if (encsrc == NULL)
goto enc_src_failed;
- ename = g_strdup_printf ("rtcp_sink_%d", sessid);
+ ename = g_strdup_printf ("rtcp_sink_%u", sessid);
encsink = gst_element_get_static_pad (encoder, ename);
g_free (ename);
if (encsink == NULL)
}
pad_failed:
{
- g_warning ("rtpbin: failed to get rtcp pad for session %d", sessid);
+ g_warning ("rtpbin: failed to get rtcp pad for session %u", sessid);
return NULL;
}
enc_src_failed:
{
- g_warning ("rtpbin: failed to get encoder src pad for session %d", sessid);
+ g_warning ("rtpbin: failed to get encoder src pad for session %u", sessid);
return NULL;
}
enc_sink_failed:
{
- g_warning ("rtpbin: failed to get encoder sink pad for session %d", sessid);
+ g_warning ("rtpbin: failed to get encoder sink pad for session %u", sessid);
gst_object_unref (encsrc);
return NULL;
}
enc_link_failed:
{
- g_warning ("rtpbin: failed to link rtcp encoder for session %d", sessid);
+ g_warning ("rtpbin: failed to link rtcp encoder for session %u", sessid);
gst_object_unref (encsrc);
return NULL;
}
void (*on_new_sender_ssrc) (GstRtpBin *rtpbin, guint session, guint32 ssrc);
void (*on_sender_ssrc_active) (GstRtpBin *rtpbin, guint session, guint32 ssrc);
+
+ guint (*on_bundled_ssrc) (GstRtpBin *rtpbin, guint ssrc);
};
GType gst_rtp_bin_get_type (void);
elements/rtpaux \
elements/rtpbin \
elements/rtpbin_buffer_list \
+ elements/rtpbundle \
elements/rtpcollision \
elements/rtpjitterbuffer \
elements/rtpmux \
elements_rtpaux_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(CFLAGS) $(AM_CFLAGS)
elements_rtpaux_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstrtp-$(GST_API_VERSION) $(LDADD)
+elements_rtpbundle_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(CFLAGS) $(AM_CFLAGS)
+elements_rtpbundle_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstrtp-$(GST_API_VERSION) $(LDADD)
+
# FIXME: configure should check for gdk-pixbuf not gtk
# only need video.h header, not the lib
elements_gdkpixbufsink_CFLAGS = \
rtpaux
rtpbin
rtpbin_buffer_list
+rtpbundle
rtpcollision
rtph261
rtph263
--- /dev/null
+/* GStreamer
+ *
+ * Copyright (C) 2016 Igalia S.L.
+ * @author Philippe Normand <philn@igalia.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., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/check/gstcheck.h>
+#include <gst/check/gstconsistencychecker.h>
+#include <gst/check/gsttestclock.h>
+#include <gst/rtp/gstrtpbuffer.h>
+
+static GMainLoop *main_loop;
+
+static void
+message_received (GstBus * bus, GstMessage * message, GstPipeline * bin)
+{
+ GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT,
+ GST_MESSAGE_SRC (message), message);
+
+ switch (message->type) {
+ case GST_MESSAGE_EOS:
+ g_main_loop_quit (main_loop);
+ break;
+ case GST_MESSAGE_WARNING:{
+ GError *gerror;
+ gchar *debug;
+
+ gst_message_parse_warning (message, &gerror, &debug);
+ gst_object_default_error (GST_MESSAGE_SRC (message), gerror, debug);
+ g_error_free (gerror);
+ g_free (debug);
+ break;
+ }
+ case GST_MESSAGE_ERROR:{
+ GError *gerror;
+ gchar *debug;
+
+ gst_message_parse_error (message, &gerror, &debug);
+ gst_object_default_error (GST_MESSAGE_SRC (message), gerror, debug);
+ g_error_free (gerror);
+ g_free (debug);
+ fail ("Error!");
+ break;
+ }
+ default:
+ break;
+ }
+}
+
+static void
+on_rtpbinreceive_pad_added (GstElement * element, GstPad * new_pad,
+ gpointer data)
+{
+ GstElement *pipeline = GST_ELEMENT (data);
+ gchar *pad_name = gst_pad_get_name (new_pad);
+
+ if (g_str_has_prefix (pad_name, "recv_rtp_src_")) {
+ GstCaps *caps = gst_pad_get_current_caps (new_pad);
+ GstStructure *s = gst_caps_get_structure (caps, 0);
+ const gchar *media_type = gst_structure_get_string (s, "media");
+ gchar *depayloader_name = g_strdup_printf ("%s_rtpdepayloader", media_type);
+ GstElement *rtpdepayloader =
+ gst_bin_get_by_name (GST_BIN (pipeline), depayloader_name);
+ GstPad *sinkpad;
+
+ g_free (depayloader_name);
+ fail_unless (rtpdepayloader != NULL, NULL);
+
+ sinkpad = gst_element_get_static_pad (rtpdepayloader, "sink");
+ gst_pad_link (new_pad, sinkpad);
+ gst_object_unref (sinkpad);
+ gst_object_unref (rtpdepayloader);
+
+ gst_caps_unref (caps);
+ }
+ g_free (pad_name);
+}
+
+static guint
+on_bundled_ssrc (GstElement * rtpbin, guint ssrc, gpointer user_data)
+{
+ static gboolean create_session = FALSE;
+ guint session_id = 0;
+
+ if (create_session) {
+ session_id = 1;
+ } else {
+ create_session = TRUE;
+ /* use existing session 0, a new session will be created for the next discovered bundled SSRC */
+ }
+ return session_id;
+}
+
+static GstCaps *
+on_request_pt_map (GstElement * rtpbin, guint session_id, guint pt,
+ gpointer user_data)
+{
+ GstCaps *caps = NULL;
+ if (pt == 96) {
+ caps =
+ gst_caps_from_string
+ ("application/x-rtp,media=(string)audio,encoding-name=(string)PCMA,clock-rate=(int)8000");
+ } else if (pt == 100) {
+ caps =
+ gst_caps_from_string
+ ("application/x-rtp,media=(string)video,encoding-name=(string)RAW,clock-rate=(int)90000,sampling=(string)\"YCbCr-4:2:0\",depth=(string)8,width=(string)320,height=(string)240");
+ }
+ return caps;
+}
+
+
+static GstElement *
+create_pipeline (gboolean send)
+{
+ GstElement *pipeline, *rtpbin, *audiosrc, *audio_encoder,
+ *audio_rtppayloader, *sendrtp_udpsink, *recv_rtp_udpsrc,
+ *send_rtcp_udpsink, *recv_rtcp_udpsrc, *sendrtcp_funnel, *sendrtp_funnel;
+ GstElement *audio_rtpdepayloader, *audio_decoder, *audio_sink;
+ GstElement *videosrc, *video_rtppayloader, *video_rtpdepayloader, *video_sink;
+ gboolean res;
+ GstPad *funnel_pad, *rtp_src_pad;
+ GstCaps *rtpcaps;
+ gint rtp_udp_port = 5001;
+ gint rtcp_udp_port = 5002;
+
+ pipeline = gst_pipeline_new (send ? "pipeline_send" : "pipeline_receive");
+
+ rtpbin =
+ gst_element_factory_make ("rtpbin",
+ send ? "rtpbin_send" : "rtpbin_receive");
+ g_object_set (rtpbin, "latency", 200, NULL);
+
+ if (!send) {
+ g_signal_connect (rtpbin, "on-bundled-ssrc",
+ G_CALLBACK (on_bundled_ssrc), NULL);
+ g_signal_connect (rtpbin, "request-pt-map",
+ G_CALLBACK (on_request_pt_map), NULL);
+ }
+
+ g_signal_connect (rtpbin, "pad-added",
+ G_CALLBACK (on_rtpbinreceive_pad_added), pipeline);
+
+ gst_bin_add (GST_BIN (pipeline), rtpbin);
+
+ if (send) {
+ audiosrc = gst_element_factory_make ("audiotestsrc", NULL);
+ audio_encoder = gst_element_factory_make ("alawenc", NULL);
+ audio_rtppayloader = gst_element_factory_make ("rtppcmapay", NULL);
+ g_object_set (audio_rtppayloader, "pt", 96, NULL);
+ g_object_set (audio_rtppayloader, "seqnum-offset", 1, NULL);
+
+ videosrc = gst_element_factory_make ("videotestsrc", NULL);
+ video_rtppayloader = gst_element_factory_make ("rtpvrawpay", NULL);
+ g_object_set (video_rtppayloader, "pt", 100, "seqnum-offset", 1, NULL);
+
+ g_object_set (audiosrc, "num-buffers", 5, NULL);
+ g_object_set (videosrc, "num-buffers", 5, NULL);
+
+ /* muxed rtcp */
+ sendrtcp_funnel = gst_element_factory_make ("funnel", "send_rtcp_funnel");
+ send_rtcp_udpsink = gst_element_factory_make ("udpsink", NULL);
+ g_object_set (send_rtcp_udpsink, "host", "127.0.0.1", NULL);
+ g_object_set (send_rtcp_udpsink, "port", rtcp_udp_port, NULL);
+ g_object_set (send_rtcp_udpsink, "sync", FALSE, NULL);
+ g_object_set (send_rtcp_udpsink, "async", FALSE, NULL);
+
+ /* outgoing bundled stream */
+ sendrtp_funnel = gst_element_factory_make ("funnel", "send_rtp_funnel");
+ sendrtp_udpsink = gst_element_factory_make ("udpsink", NULL);
+ g_object_set (sendrtp_udpsink, "host", "127.0.0.1", NULL);
+ g_object_set (sendrtp_udpsink, "port", rtp_udp_port, NULL);
+
+ gst_bin_add_many (GST_BIN (pipeline), audiosrc, audio_encoder,
+ audio_rtppayloader, sendrtp_udpsink, send_rtcp_udpsink,
+ sendrtp_funnel, sendrtcp_funnel, videosrc, video_rtppayloader, NULL);
+
+ res = gst_element_link (audiosrc, audio_encoder);
+ fail_unless (res == TRUE, NULL);
+ res = gst_element_link (audio_encoder, audio_rtppayloader);
+ fail_unless (res == TRUE, NULL);
+ res =
+ gst_element_link_pads_full (audio_rtppayloader, "src", rtpbin,
+ "send_rtp_sink_0", GST_PAD_LINK_CHECK_NOTHING);
+ fail_unless (res == TRUE, NULL);
+
+ res = gst_element_link (videosrc, video_rtppayloader);
+ fail_unless (res == TRUE, NULL);
+ res =
+ gst_element_link_pads_full (video_rtppayloader, "src", rtpbin,
+ "send_rtp_sink_1", GST_PAD_LINK_CHECK_NOTHING);
+ fail_unless (res == TRUE, NULL);
+
+ res =
+ gst_element_link_pads_full (sendrtp_funnel, "src", sendrtp_udpsink,
+ "sink", GST_PAD_LINK_CHECK_NOTHING);
+ fail_unless (res == TRUE, NULL);
+
+ funnel_pad = gst_element_get_request_pad (sendrtp_funnel, "sink_%u");
+ rtp_src_pad = gst_element_get_static_pad (rtpbin, "send_rtp_src_0");
+ res = gst_pad_link (rtp_src_pad, funnel_pad);
+ gst_object_unref (funnel_pad);
+ gst_object_unref (rtp_src_pad);
+
+ funnel_pad = gst_element_get_request_pad (sendrtp_funnel, "sink_%u");
+ rtp_src_pad = gst_element_get_static_pad (rtpbin, "send_rtp_src_1");
+ res = gst_pad_link (rtp_src_pad, funnel_pad);
+ gst_object_unref (funnel_pad);
+ gst_object_unref (rtp_src_pad);
+
+ res =
+ gst_element_link_pads_full (sendrtcp_funnel, "src", send_rtcp_udpsink,
+ "sink", GST_PAD_LINK_CHECK_NOTHING);
+ fail_unless (res == TRUE, NULL);
+
+ funnel_pad = gst_element_get_request_pad (sendrtcp_funnel, "sink_%u");
+ rtp_src_pad = gst_element_get_request_pad (rtpbin, "send_rtcp_src_0");
+ res =
+ gst_pad_link_full (rtp_src_pad, funnel_pad, GST_PAD_LINK_CHECK_NOTHING);
+ gst_object_unref (funnel_pad);
+ gst_object_unref (rtp_src_pad);
+
+ funnel_pad = gst_element_get_request_pad (sendrtcp_funnel, "sink_%u");
+ rtp_src_pad = gst_element_get_request_pad (rtpbin, "send_rtcp_src_1");
+ res =
+ gst_pad_link_full (rtp_src_pad, funnel_pad, GST_PAD_LINK_CHECK_NOTHING);
+ gst_object_unref (funnel_pad);
+ gst_object_unref (rtp_src_pad);
+
+ } else {
+ recv_rtp_udpsrc = gst_element_factory_make ("udpsrc", NULL);
+ g_object_set (recv_rtp_udpsrc, "port", rtp_udp_port, NULL);
+ rtpcaps = gst_caps_from_string ("application/x-rtp");
+ g_object_set (recv_rtp_udpsrc, "caps", rtpcaps, NULL);
+ gst_caps_unref (rtpcaps);
+
+ recv_rtcp_udpsrc = gst_element_factory_make ("udpsrc", NULL);
+ g_object_set (recv_rtcp_udpsrc, "port", rtcp_udp_port, NULL);
+
+ audio_rtpdepayloader =
+ gst_element_factory_make ("rtppcmadepay", "audio_rtpdepayloader");
+ audio_decoder = gst_element_factory_make ("alawdec", NULL);
+ audio_sink = gst_element_factory_make ("fakesink", NULL);
+ g_object_set (audio_sink, "sync", TRUE, NULL);
+
+ video_rtpdepayloader =
+ gst_element_factory_make ("rtpvrawdepay", "video_rtpdepayloader");
+ video_sink = gst_element_factory_make ("fakesink", NULL);
+ g_object_set (video_sink, "sync", TRUE, NULL);
+
+ gst_bin_add_many (GST_BIN (pipeline), recv_rtp_udpsrc, recv_rtcp_udpsrc,
+ audio_rtpdepayloader, audio_decoder, audio_sink, video_rtpdepayloader,
+ video_sink, NULL);
+
+ res =
+ gst_element_link_pads_full (audio_rtpdepayloader, "src", audio_decoder,
+ "sink", GST_PAD_LINK_CHECK_NOTHING);
+ fail_unless (res == TRUE, NULL);
+ res = gst_element_link (audio_decoder, audio_sink);
+ fail_unless (res == TRUE, NULL);
+
+ res =
+ gst_element_link_pads_full (video_rtpdepayloader, "src", video_sink,
+ "sink", GST_PAD_LINK_CHECK_NOTHING);
+ fail_unless (res == TRUE, NULL);
+
+ /* request a single receiving RTP session. */
+ res =
+ gst_element_link_pads_full (recv_rtcp_udpsrc, "src", rtpbin,
+ "recv_rtcp_sink_0", GST_PAD_LINK_CHECK_NOTHING);
+ fail_unless (res == TRUE, NULL);
+ res =
+ gst_element_link_pads_full (recv_rtp_udpsrc, "src", rtpbin,
+ "recv_rtp_sink_0", GST_PAD_LINK_CHECK_NOTHING);
+ fail_unless (res == TRUE, NULL);
+ }
+
+ return pipeline;
+}
+
+GST_START_TEST (test_simple_rtpbin_bundle)
+{
+ GstElement *send_pipeline, *recv_pipeline;
+ GstBus *send_bus, *recv_bus;
+ GstStateChangeReturn state_res = GST_STATE_CHANGE_FAILURE;
+ GstElement *rtpbin_receive;
+ GObject *rtp_session;
+
+ main_loop = g_main_loop_new (NULL, FALSE);
+
+ send_pipeline = create_pipeline (TRUE);
+ recv_pipeline = create_pipeline (FALSE);
+
+ send_bus = gst_element_get_bus (send_pipeline);
+ gst_bus_add_signal_watch_full (send_bus, G_PRIORITY_HIGH);
+
+ g_signal_connect (send_bus, "message::error", (GCallback) message_received,
+ send_pipeline);
+ g_signal_connect (send_bus, "message::warning", (GCallback) message_received,
+ send_pipeline);
+ g_signal_connect (send_bus, "message::eos", (GCallback) message_received,
+ send_pipeline);
+
+ recv_bus = gst_element_get_bus (recv_pipeline);
+ gst_bus_add_signal_watch_full (recv_bus, G_PRIORITY_HIGH);
+
+ g_signal_connect (recv_bus, "message::error", (GCallback) message_received,
+ recv_pipeline);
+ g_signal_connect (recv_bus, "message::warning", (GCallback) message_received,
+ recv_pipeline);
+ g_signal_connect (recv_bus, "message::eos", (GCallback) message_received,
+ recv_pipeline);
+
+ state_res = gst_element_set_state (recv_pipeline, GST_STATE_PLAYING);
+ ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
+
+ state_res = gst_element_set_state (send_pipeline, GST_STATE_PLAYING);
+ ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
+
+ GST_INFO ("enter mainloop");
+ g_main_loop_run (main_loop);
+ GST_INFO ("exit mainloop");
+
+ rtpbin_receive =
+ gst_bin_get_by_name (GST_BIN (recv_pipeline), "rtpbin_receive");
+ fail_if (rtpbin_receive == NULL, NULL);
+
+ /* Check that 2 RTP sessions where created while only one was explicitely requested. */
+ g_signal_emit_by_name (rtpbin_receive, "get-internal-session", 0,
+ &rtp_session);
+ fail_if (rtp_session == NULL, NULL);
+ g_object_unref (rtp_session);
+ g_signal_emit_by_name (rtpbin_receive, "get-internal-session", 1,
+ &rtp_session);
+ fail_if (rtp_session == NULL, NULL);
+ g_object_unref (rtp_session);
+
+ gst_object_unref (rtpbin_receive);
+
+ state_res = gst_element_set_state (send_pipeline, GST_STATE_NULL);
+ ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
+
+ state_res = gst_element_set_state (recv_pipeline, GST_STATE_NULL);
+ ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
+
+ /* cleanup */
+ g_main_loop_unref (main_loop);
+
+ gst_bus_remove_signal_watch (send_bus);
+ gst_object_unref (send_bus);
+ gst_object_unref (send_pipeline);
+
+ gst_bus_remove_signal_watch (recv_bus);
+ gst_object_unref (recv_bus);
+ gst_object_unref (recv_pipeline);
+
+}
+
+GST_END_TEST;
+
+static Suite *
+rtpbundle_suite (void)
+{
+ Suite *s = suite_create ("rtpbundle");
+ TCase *tc_chain = tcase_create ("general");
+
+ tcase_set_timeout (tc_chain, 10000);
+
+ suite_add_tcase (s, tc_chain);
+
+ tcase_add_test (tc_chain, test_simple_rtpbin_bundle);
+
+ return s;
+}
+
+GST_CHECK_MAIN (rtpbundle);
[ 'elements/rtpaux' ],
[ 'elements/rtpbin' ],
[ 'elements/rtpbin_buffer_list' ],
+ [ 'elements/rtpbundle' ],
[ 'elements/rtpcollision' ],
[ 'elements/rtpjitterbuffer' ],
[ 'elements/rtpmux' ],
server-alsasrc-PCMA
client-rtpaux
server-rtpaux
+client-rtpbundle
+server-rtpbundle
noinst_PROGRAMS = server-alsasrc-PCMA client-PCMA \
- client-rtpaux server-rtpaux
+ client-rtpaux server-rtpaux client-rtpbundle server-rtpbundle
# FIXME 0.11: ignore GValueArray warnings for now until this is sorted
ERROR_CFLAGS=
client_rtpaux_CFLAGS = $(GST_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS)
client_rtpaux_LDADD = $(GST_LIBS)
+server_rtpbundle_SOURCES = server-rtpbundle.c
+server_rtpbundle_CFLAGS = $(GST_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS)
+server_rtpbundle_LDADD = $(GST_LIBS)
+
+client_rtpbundle_SOURCES = client-rtpbundle.c
+client_rtpbundle_CFLAGS = $(GST_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS)
+client_rtpbundle_LDADD = $(GST_LIBS)
+
server_alsasrc_PCMA_SOURCES = server-alsasrc-PCMA.c
server_alsasrc_PCMA_CFLAGS = $(GST_CFLAGS)
server_alsasrc_PCMA_LDADD = $(GST_LIBS) $(LIBM)
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2016 Igalia S.L
+ * @author Philippe Normand <philn@igalia.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., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+/*
+ * RTP bundle receiver
+ *
+ * In this example we initially create one RTP session but the incoming RTP
+ * and RTCP streams actually bundle 2 different media type, one audio stream
+ * and one video stream. We are notified of the discovery of the streams by
+ * the on-bundled-ssrc rtpbin signal. In the handler we decide to assign the
+ * first SSRC to the (existing) audio session and the second SSRC to a new
+ * session (id: 1).
+ *
+ * .-------. .----------. .-----------. .-------. .-------------.
+ * RTP |udpsrc | | rtpbin | | pcmadepay | |alawdec| |autoaudiosink|
+ * port=5001 | src->recv_rtp_0 recv_rtp_0->sink src->sink src->sink |
+ * '-------' | | '-----------' '-------' '-------------'
+ * | |
+ * | | .-------.
+ * | | |udpsink| RTCP
+ * | send_rtcp_0->sink | port=5003
+ * .-------. | | '-------' sync=false
+ * RTCP |udpsrc | | | async=false
+ * port=5002 | src->recv_rtcp_0 |
+ * '-------' | |
+ * | |
+ * | | .---------. .-------------.
+ * | | |vrawdepay| |autovideosink|
+ * | recv_rtp_1->sink src->sink |
+ * | | '---------' '-------------'
+ * | |
+ * | | .-------.
+ * | | |udpsink| RTCP
+ * | send_rtcp_1->sink | port=5004
+ * | | '-------' sync=false
+ * | | async=false
+ * | |
+ * '----------'
+ *
+ */
+
+static gboolean
+plug_video_rtcp_sender (gpointer user_data)
+{
+ gint send_video_rtcp_port = 5004;
+ GstElement *rtpbin = GST_ELEMENT_CAST (user_data);
+ GstElement *send_video_rtcp_udpsink;
+ GstElement *pipeline =
+ GST_ELEMENT_CAST (gst_object_get_parent (GST_OBJECT (rtpbin)));
+
+ send_video_rtcp_udpsink = gst_element_factory_make ("udpsink", NULL);
+ g_object_set (send_video_rtcp_udpsink, "host", "127.0.0.1", NULL);
+ g_object_set (send_video_rtcp_udpsink, "port", send_video_rtcp_port, NULL);
+ g_object_set (send_video_rtcp_udpsink, "sync", FALSE, NULL);
+ g_object_set (send_video_rtcp_udpsink, "async", FALSE, NULL);
+ gst_bin_add (GST_BIN (pipeline), send_video_rtcp_udpsink);
+ gst_element_link_pads (rtpbin, "send_rtcp_src_1", send_video_rtcp_udpsink,
+ "sink");
+ gst_element_sync_state_with_parent (send_video_rtcp_udpsink);
+
+ gst_object_unref (pipeline);
+ gst_object_unref (rtpbin);
+ return G_SOURCE_REMOVE;
+}
+
+static void
+on_rtpbinreceive_pad_added (GstElement * rtpbin, GstPad * new_pad,
+ gpointer data)
+{
+ GstElement *pipeline = GST_ELEMENT (data);
+ gchar *pad_name = gst_pad_get_name (new_pad);
+
+ if (g_str_has_prefix (pad_name, "recv_rtp_src_")) {
+ GstCaps *caps = gst_pad_get_current_caps (new_pad);
+ GstStructure *s = gst_caps_get_structure (caps, 0);
+ const gchar *media_type = gst_structure_get_string (s, "media");
+ gchar *depayloader_name = g_strdup_printf ("%s_rtpdepayloader", media_type);
+ GstElement *rtpdepayloader =
+ gst_bin_get_by_name (GST_BIN (pipeline), depayloader_name);
+ GstPad *sinkpad;
+
+ g_free (depayloader_name);
+
+ sinkpad = gst_element_get_static_pad (rtpdepayloader, "sink");
+ gst_pad_link (new_pad, sinkpad);
+ gst_object_unref (sinkpad);
+ gst_object_unref (rtpdepayloader);
+
+ gst_caps_unref (caps);
+
+ if (g_str_has_prefix (pad_name, "recv_rtp_src_1")) {
+ g_timeout_add (0, plug_video_rtcp_sender, gst_object_ref (rtpbin));
+ }
+ }
+ g_free (pad_name);
+}
+
+static guint
+on_bundled_ssrc (GstElement * rtpbin, guint ssrc, gpointer user_data)
+{
+ static gboolean create_session = FALSE;
+ guint session_id = 0;
+
+ if (create_session) {
+ session_id = 1;
+ } else {
+ create_session = TRUE;
+ /* use existing session 0, a new session will be created for the next discovered bundled SSRC */
+ }
+ return session_id;
+}
+
+static GstCaps *
+on_request_pt_map (GstElement * rtpbin, guint session_id, guint pt,
+ gpointer user_data)
+{
+ GstCaps *caps = NULL;
+ if (pt == 96) {
+ caps =
+ gst_caps_from_string
+ ("application/x-rtp,media=(string)audio,encoding-name=(string)PCMA,clock-rate=(int)8000");
+ } else if (pt == 100) {
+ caps =
+ gst_caps_from_string
+ ("application/x-rtp,media=(string)video,encoding-name=(string)RAW,clock-rate=(int)90000,sampling=(string)\"YCbCr-4:2:0\",depth=(string)8,width=(string)320,height=(string)240");
+ }
+ return caps;
+}
+
+static GstElement *
+create_pipeline (void)
+{
+ GstElement *pipeline, *rtpbin, *recv_rtp_udpsrc, *recv_rtcp_udpsrc,
+ *audio_rtpdepayloader, *audio_decoder, *audio_sink, *video_rtpdepayloader,
+ *video_sink, *send_audio_rtcp_udpsink;
+ GstCaps *rtpcaps;
+ gint rtp_udp_port = 5001;
+ gint rtcp_udp_port = 5002;
+ gint send_audio_rtcp_port = 5003;
+
+ pipeline = gst_pipeline_new (NULL);
+
+ rtpbin = gst_element_factory_make ("rtpbin", NULL);
+ g_object_set (rtpbin, "latency", 200, NULL);
+
+ g_signal_connect (rtpbin, "on-bundled-ssrc",
+ G_CALLBACK (on_bundled_ssrc), NULL);
+ g_signal_connect (rtpbin, "request-pt-map",
+ G_CALLBACK (on_request_pt_map), NULL);
+
+ g_signal_connect (rtpbin, "pad-added",
+ G_CALLBACK (on_rtpbinreceive_pad_added), pipeline);
+
+ gst_bin_add (GST_BIN (pipeline), rtpbin);
+
+ recv_rtp_udpsrc = gst_element_factory_make ("udpsrc", NULL);
+ g_object_set (recv_rtp_udpsrc, "port", rtp_udp_port, NULL);
+ rtpcaps = gst_caps_from_string ("application/x-rtp");
+ g_object_set (recv_rtp_udpsrc, "caps", rtpcaps, NULL);
+ gst_caps_unref (rtpcaps);
+
+ recv_rtcp_udpsrc = gst_element_factory_make ("udpsrc", NULL);
+ g_object_set (recv_rtcp_udpsrc, "port", rtcp_udp_port, NULL);
+
+ audio_rtpdepayloader =
+ gst_element_factory_make ("rtppcmadepay", "audio_rtpdepayloader");
+ audio_decoder = gst_element_factory_make ("alawdec", NULL);
+ audio_sink = gst_element_factory_make ("autoaudiosink", NULL);
+
+ video_rtpdepayloader =
+ gst_element_factory_make ("rtpvrawdepay", "video_rtpdepayloader");
+ video_sink = gst_element_factory_make ("autovideosink", NULL);
+
+ gst_bin_add_many (GST_BIN (pipeline), recv_rtp_udpsrc, recv_rtcp_udpsrc,
+ audio_rtpdepayloader, audio_decoder, audio_sink, video_rtpdepayloader,
+ video_sink, NULL);
+
+ gst_element_link_pads (audio_rtpdepayloader, "src", audio_decoder, "sink");
+ gst_element_link (audio_decoder, audio_sink);
+
+ gst_element_link_pads (video_rtpdepayloader, "src", video_sink, "sink");
+
+ /* request a single receiving RTP session. */
+ gst_element_link_pads (recv_rtcp_udpsrc, "src", rtpbin, "recv_rtcp_sink_0");
+ gst_element_link_pads (recv_rtp_udpsrc, "src", rtpbin, "recv_rtp_sink_0");
+
+ send_audio_rtcp_udpsink = gst_element_factory_make ("udpsink", NULL);
+ g_object_set (send_audio_rtcp_udpsink, "host", "127.0.0.1", NULL);
+ g_object_set (send_audio_rtcp_udpsink, "port", send_audio_rtcp_port, NULL);
+ g_object_set (send_audio_rtcp_udpsink, "sync", FALSE, NULL);
+ g_object_set (send_audio_rtcp_udpsink, "async", FALSE, NULL);
+ gst_bin_add (GST_BIN (pipeline), send_audio_rtcp_udpsink);
+ gst_element_link_pads (rtpbin, "send_rtcp_src_0", send_audio_rtcp_udpsink,
+ "sink");
+
+ return pipeline;
+}
+
+/*
+ * Used to generate informative messages during pipeline startup
+ */
+static void
+cb_state (GstBus * bus, GstMessage * message, gpointer data)
+{
+ GstObject *pipe = GST_OBJECT (data);
+ GstState old, new, pending;
+ gst_message_parse_state_changed (message, &old, &new, &pending);
+ if (message->src == pipe) {
+ g_print ("Pipeline %s changed state from %s to %s\n",
+ GST_OBJECT_NAME (message->src),
+ gst_element_state_get_name (old), gst_element_state_get_name (new));
+ if (old == GST_STATE_PAUSED && new == GST_STATE_PLAYING)
+ GST_DEBUG_BIN_TO_DOT_FILE (GST_BIN (pipe), GST_DEBUG_GRAPH_SHOW_ALL,
+ GST_OBJECT_NAME (message->src));
+ }
+}
+
+int
+main (int argc, char **argv)
+{
+ GstElement *pipe;
+ GstBus *bus;
+ GMainLoop *loop;
+
+ gst_init (&argc, &argv);
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ pipe = create_pipeline ();
+ bus = gst_element_get_bus (pipe);
+ g_signal_connect (bus, "message::state-changed", G_CALLBACK (cb_state), pipe);
+ gst_bus_add_signal_watch (bus);
+ gst_object_unref (bus);
+
+ g_print ("starting server pipeline\n");
+ gst_element_set_state (pipe, GST_STATE_PLAYING);
+
+ g_main_loop_run (loop);
+
+ g_print ("stopping server pipeline\n");
+ gst_element_set_state (pipe, GST_STATE_NULL);
+
+ gst_object_unref (pipe);
+ g_main_loop_unref (loop);
+
+ return 0;
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2016 Igalia S.L
+ * @author Philippe Normand <philn@igalia.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., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+#include <gst/gst.h>
+
+/*
+ * An bundling RTP server
+ * creates two sessions and streams audio on one, video on the other, with RTCP
+ * on both sessions. The destination is 127.0.0.1.
+ *
+ * The RTP streams are bundled to a single outgoing connection. Same for the RTCP streams.
+ *
+ * .-------. .-------. .-------. .------------. .------.
+ * |audiots| |alawenc| |pcmapay| | rtpbin | |funnel|
+ * | src->sink src->sink src->send_rtp_0 send_rtp_0--->sink_0 | .-------.
+ * '-------' '-------' '-------' | | | | |udpsink|
+ * | | | src->sink |
+ * .-------. .---------. | | | | '-------'
+ * |videots| | vrawpay | | | | |
+ * | src------------>sink src->send_rtp_1 send_rtp_1--->sink_1 |
+ * '-------' '---------' | | '------'
+ * | |
+ * .------. | |
+ * |udpsrc| | | .------.
+ * | src->recv_rtcp_0 | |funnel|
+ * '------' | send_rtcp_0-->sink_0 | .-------.
+ * | | | | |udpsink|
+ * .------. | | | src->sink |
+ * |udpsrc| | | | | '-------'
+ * | src->recv_rtcp_1 | | |
+ * '------' | send_rtcp_1-->sink_1 |
+ * '------------' '------'
+ *
+ */
+
+static GstElement *
+create_pipeline (void)
+{
+ GstElement *pipeline, *rtpbin, *audiosrc, *audio_encoder,
+ *audio_rtppayloader, *sendrtp_udpsink,
+ *send_rtcp_udpsink, *sendrtcp_funnel, *sendrtp_funnel;
+ GstElement *videosrc, *video_rtppayloader, *time_overlay;
+ gint rtp_udp_port = 5001;
+ gint rtcp_udp_port = 5002;
+ gint recv_audio_rtcp_port = 5003;
+ gint recv_video_rtcp_port = 5004;
+ GstElement *audio_rtcp_udpsrc, *video_rtcp_udpsrc;
+
+ pipeline = gst_pipeline_new (NULL);
+
+ rtpbin = gst_element_factory_make ("rtpbin", NULL);
+
+ audiosrc = gst_element_factory_make ("audiotestsrc", NULL);
+ g_object_set (audiosrc, "is-live", TRUE, NULL);
+ audio_encoder = gst_element_factory_make ("alawenc", NULL);
+ audio_rtppayloader = gst_element_factory_make ("rtppcmapay", NULL);
+ g_object_set (audio_rtppayloader, "pt", 96, NULL);
+
+ videosrc = gst_element_factory_make ("videotestsrc", NULL);
+ g_object_set (videosrc, "is-live", TRUE, NULL);
+ time_overlay = gst_element_factory_make ("timeoverlay", NULL);
+ video_rtppayloader = gst_element_factory_make ("rtpvrawpay", NULL);
+ g_object_set (video_rtppayloader, "pt", 100, NULL);
+
+ /* muxed rtcp */
+ sendrtcp_funnel = gst_element_factory_make ("funnel", "send_rtcp_funnel");
+ send_rtcp_udpsink = gst_element_factory_make ("udpsink", NULL);
+ g_object_set (send_rtcp_udpsink, "host", "127.0.0.1", NULL);
+ g_object_set (send_rtcp_udpsink, "port", rtcp_udp_port, NULL);
+ g_object_set (send_rtcp_udpsink, "sync", FALSE, NULL);
+ g_object_set (send_rtcp_udpsink, "async", FALSE, NULL);
+
+ /* outgoing bundled stream */
+ sendrtp_funnel = gst_element_factory_make ("funnel", "send_rtp_funnel");
+ sendrtp_udpsink = gst_element_factory_make ("udpsink", NULL);
+ g_object_set (sendrtp_udpsink, "host", "127.0.0.1", NULL);
+ g_object_set (sendrtp_udpsink, "port", rtp_udp_port, NULL);
+ g_object_set (sendrtp_udpsink, "sync", FALSE, NULL);
+ g_object_set (sendrtp_udpsink, "async", FALSE, NULL);
+
+ gst_bin_add_many (GST_BIN (pipeline), rtpbin, audiosrc, audio_encoder,
+ audio_rtppayloader, sendrtp_udpsink, send_rtcp_udpsink,
+ sendrtp_funnel, sendrtcp_funnel, videosrc, video_rtppayloader, NULL);
+
+ if (time_overlay)
+ gst_bin_add (GST_BIN (pipeline), time_overlay);
+
+ gst_element_link_many (audiosrc, audio_encoder, audio_rtppayloader, NULL);
+ gst_element_link_pads (audio_rtppayloader, "src", rtpbin, "send_rtp_sink_0");
+
+ if (time_overlay) {
+ gst_element_link_many (videosrc, time_overlay, video_rtppayloader, NULL);
+ } else {
+ gst_element_link (videosrc, video_rtppayloader);
+ }
+
+ gst_element_link_pads (video_rtppayloader, "src", rtpbin, "send_rtp_sink_1");
+
+ gst_element_link_pads (sendrtp_funnel, "src", sendrtp_udpsink, "sink");
+ gst_element_link_pads (rtpbin, "send_rtp_src_0", sendrtp_funnel, "sink_%u");
+ gst_element_link_pads (rtpbin, "send_rtp_src_1", sendrtp_funnel, "sink_%u");
+ gst_element_link_pads (sendrtcp_funnel, "src", send_rtcp_udpsink, "sink");
+ gst_element_link_pads (rtpbin, "send_rtcp_src_0", sendrtcp_funnel, "sink_%u");
+ gst_element_link_pads (rtpbin, "send_rtcp_src_1", sendrtcp_funnel, "sink_%u");
+
+ audio_rtcp_udpsrc = gst_element_factory_make ("udpsrc", NULL);
+ g_object_set (audio_rtcp_udpsrc, "port", recv_audio_rtcp_port, NULL);
+ video_rtcp_udpsrc = gst_element_factory_make ("udpsrc", NULL);
+ g_object_set (video_rtcp_udpsrc, "port", recv_video_rtcp_port, NULL);
+ gst_bin_add_many (GST_BIN (pipeline), audio_rtcp_udpsrc, video_rtcp_udpsrc,
+ NULL);
+ gst_element_link_pads (audio_rtcp_udpsrc, "src", rtpbin, "recv_rtcp_sink_0");
+ gst_element_link_pads (video_rtcp_udpsrc, "src", rtpbin, "recv_rtcp_sink_1");
+
+ return pipeline;
+}
+
+/*
+ * Used to generate informative messages during pipeline startup
+ */
+static void
+cb_state (GstBus * bus, GstMessage * message, gpointer data)
+{
+ GstObject *pipe = GST_OBJECT (data);
+ GstState old, new, pending;
+ gst_message_parse_state_changed (message, &old, &new, &pending);
+ if (message->src == pipe) {
+ g_print ("Pipeline %s changed state from %s to %s\n",
+ GST_OBJECT_NAME (message->src),
+ gst_element_state_get_name (old), gst_element_state_get_name (new));
+ }
+}
+
+int
+main (int argc, char **argv)
+{
+ GstElement *pipe;
+ GstBus *bus;
+ GMainLoop *loop;
+
+ gst_init (&argc, &argv);
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ pipe = create_pipeline ();
+ bus = gst_element_get_bus (pipe);
+ g_signal_connect (bus, "message::state-changed", G_CALLBACK (cb_state), pipe);
+ gst_bus_add_signal_watch (bus);
+ gst_object_unref (bus);
+
+ g_print ("starting server pipeline\n");
+ gst_element_set_state (pipe, GST_STATE_PLAYING);
+
+ g_main_loop_run (loop);
+
+ g_print ("stopping server pipeline\n");
+ gst_element_set_state (pipe, GST_STATE_NULL);
+
+ gst_object_unref (pipe);
+ g_main_loop_unref (loop);
+
+ return 0;
+}
projects / platform / upstream / gst-plugins-good.git / commitdiff