* 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
GST_STATIC_CAPS ("application/x-rtp;application/x-srtp")
);
-#define GST_RTP_BIN_GET_PRIVATE(obj) \
- (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTP_BIN, GstRtpBinPrivate))
-
#define GST_RTP_BIN_LOCK(bin) g_mutex_lock (&(bin)->priv->bin_lock)
#define GST_RTP_BIN_UNLOCK(bin) g_mutex_unlock (&(bin)->priv->bin_lock)
SIGNAL_RESET_SYNC,
SIGNAL_GET_SESSION,
SIGNAL_GET_INTERNAL_SESSION,
+ SIGNAL_GET_STORAGE,
SIGNAL_GET_INTERNAL_STORAGE,
SIGNAL_ON_NEW_SSRC,
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);
+ GstRtpBinSession * session);
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);
+ GstRtpBinSession * session, guint sessid);
/* Manages the RTP stream for one SSRC.
*
/* Fec support */
GstElement *storage;
- /* 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) {
- /* create_session() warned already */
- GST_RTP_BIN_DYN_UNLOCK (rtpbin);
- return;
- }
-
- 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)
if (!(storage = gst_element_factory_make ("rtpstorage", NULL)))
goto no_storage;
+ /* need to sink the storage or otherwise signal handlers from bindings will
+ * take ownership of it and we don't own it anymore */
+ gst_object_ref_sink (storage);
g_signal_emit (rtpbin, gst_rtp_bin_signals[SIGNAL_NEW_STORAGE], 0, storage,
id);
sess->demux = demux;
sess->storage = storage;
- 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_bin_add (GST_BIN_CAST (rtpbin), storage);
+ /* unref the storage again, the bin has a reference now and
+ * we don't need it anymore */
+ gst_object_unref (storage);
+
GST_OBJECT_LOCK (rtpbin);
target = GST_STATE_TARGET (rtpbin);
GST_OBJECT_UNLOCK (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);
gst_element_set_state (storage, target);
return sess;
return internal_session;
}
+static GstElement *
+gst_rtp_bin_get_storage (GstRtpBin * bin, guint session_id)
+{
+ GstRtpBinSession *session;
+ GstElement *res = NULL;
+
+ GST_RTP_BIN_LOCK (bin);
+ GST_DEBUG_OBJECT (bin, "retrieving internal storage object, index: %u",
+ session_id);
+ session = find_session_by_id (bin, (gint) session_id);
+ if (session && session->storage) {
+ res = gst_object_ref (session->storage);
+ }
+ GST_RTP_BIN_UNLOCK (bin);
+
+ return res;
+}
+
static GObject *
gst_rtp_bin_get_internal_storage (GstRtpBin * bin, guint session_id)
{
g_object_set (buffer, "max-ts-offset-adjustment",
rtpbin->max_ts_offset_adjustment, NULL);
+ /* need to sink the jitterbufer or otherwise signal handlers from bindings will
+ * take ownership of it and we don't own it anymore */
+ gst_object_ref_sink (buffer);
g_signal_emit (rtpbin, gst_rtp_bin_signals[SIGNAL_NEW_JITTERBUFFER], 0,
buffer, session->id, ssrc);
gst_bin_add (GST_BIN_CAST (rtpbin), demux);
gst_bin_add (GST_BIN_CAST (rtpbin), buffer);
+ /* unref the jitterbuffer again, the bin has a reference now and
+ * we don't need it anymore */
+ gst_object_unref (buffer);
+
/* link stuff */
if (demux)
gst_element_link_pads_full (buffer, "src", demux, "sink",
static void gst_rtp_bin_handle_message (GstBin * bin, GstMessage * message);
#define gst_rtp_bin_parent_class parent_class
-G_DEFINE_TYPE (GstRtpBin, gst_rtp_bin, GST_TYPE_BIN);
+G_DEFINE_TYPE_WITH_PRIVATE (GstRtpBin, gst_rtp_bin, GST_TYPE_BIN);
static gboolean
_gst_element_accumulator (GSignalInvocationHint * ihint,
gstelement_class = (GstElementClass *) klass;
gstbin_class = (GstBinClass *) klass;
- g_type_class_add_private (klass, sizeof (GstRtpBinPrivate));
-
gobject_class->dispose = gst_rtp_bin_dispose;
gobject_class->finalize = gst_rtp_bin_finalize;
gobject_class->set_property = gst_rtp_bin_set_property;
G_TYPE_OBJECT, 1, G_TYPE_UINT);
/**
+ * GstRtpBin::get-storage:
+ * @rtpbin: the object which received the signal
+ * @id: the session id
+ *
+ * Request the RTPStorage element as #GObject in session @id.
+ *
+ * Since: 1.16
+ */
+ gst_rtp_bin_signals[SIGNAL_GET_STORAGE] =
+ g_signal_new ("get-storage", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
+ get_storage), NULL, NULL, g_cclosure_marshal_generic,
+ GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
+
+ /**
* GstRtpBin::on-new-ssrc:
* @rtpbin: the object which received the signal
* @session: the 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",
klass->get_session = GST_DEBUG_FUNCPTR (gst_rtp_bin_get_session);
klass->get_internal_session =
GST_DEBUG_FUNCPTR (gst_rtp_bin_get_internal_session);
+ klass->get_storage = GST_DEBUG_FUNCPTR (gst_rtp_bin_get_storage);
klass->get_internal_storage =
GST_DEBUG_FUNCPTR (gst_rtp_bin_get_internal_storage);
klass->request_rtp_encoder = GST_DEBUG_FUNCPTR (gst_rtp_bin_request_encoder);
{
gchar *cname;
- rtpbin->priv = GST_RTP_BIN_GET_PRIVATE (rtpbin);
+ rtpbin->priv = gst_rtp_bin_get_instance_private (rtpbin);
g_mutex_init (&rtpbin->priv->bin_lock);
g_mutex_init (&rtpbin->priv->dyn_lock);
return TRUE;
}
-/* a new pad (SSRC) was created in @session. This signal is emited from the
- * payload demuxer. */
static void
-new_payload_found (GstElement * element, guint pt, GstPad * pad,
- GstRtpBinStream * stream)
+expose_recv_src_pad (GstRtpBin * rtpbin, GstPad * pad, GstRtpBinStream * stream,
+ guint8 pt)
{
- GstRtpBin *rtpbin;
GstElementClass *klass;
GstPadTemplate *templ;
gchar *padname;
GstPad *gpad;
- rtpbin = stream->bin;
-
- GST_DEBUG_OBJECT (rtpbin, "new payload pad %u", pt);
-
- pad = gst_object_ref (pad);
+ gst_object_ref (pad);
if (stream->session->storage) {
GstElement *fec_decoder =
}
}
+/* a new pad (SSRC) was created in @session. This signal is emited from the
+ * payload demuxer. */
+static void
+new_payload_found (GstElement * element, guint pt, GstPad * pad,
+ GstRtpBinStream * stream)
+{
+ GstRtpBin *rtpbin;
+
+ rtpbin = stream->bin;
+
+ GST_DEBUG_OBJECT (rtpbin, "new payload pad %u", pt);
+
+ expose_recv_src_pad (rtpbin, pad, stream, pt);
+}
+
static void
payload_pad_removed (GstElement * element, GstPad * pad,
GstRtpBinStream * stream)
0, session->id, pt);
}
-/* emited when caps changed for the session */
+/* emitted when caps changed for the session */
static void
caps_changed (GstPad * pad, GParamSpec * pspec, GstRtpBinSession * session)
{
stream->demux_padremoved_sig = g_signal_connect (stream->demux,
"pad-removed", (GCallback) payload_pad_removed, stream);
- /* connect to the request-pt-map signal. This signal will be emited by the
+ /* connect to the request-pt-map signal. This signal will be emitted by the
* demuxer so that it can apply a proper caps on the buffers for the
* depayloaders. */
stream->demux_ptreq_sig = g_signal_connect (stream->demux,
/* connect to the signal so it can be forwarded. */
stream->demux_ptchange_sig = g_signal_connect (stream->demux,
"payload-type-change", (GCallback) payload_type_change, session);
+
+ GST_RTP_SESSION_UNLOCK (session);
+ GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin);
} else {
/* add rtpjitterbuffer src pad to pads */
- GstElementClass *klass;
- GstPadTemplate *templ;
- gchar *padname;
- GstPad *gpad, *pad;
+ GstPad *pad;
pad = gst_element_get_static_pad (stream->buffer, "src");
- /* ghost the pad to the parent */
- klass = GST_ELEMENT_GET_CLASS (rtpbin);
- templ = gst_element_class_get_pad_template (klass, "recv_rtp_src_%u_%u_%u");
- padname = g_strdup_printf ("recv_rtp_src_%u_%u_%u",
- stream->session->id, stream->ssrc, 255);
- gpad = gst_ghost_pad_new_from_template (padname, pad, templ);
- g_free (padname);
+ GST_RTP_SESSION_UNLOCK (session);
+ GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin);
- gst_pad_set_active (gpad, TRUE);
- gst_pad_sticky_events_foreach (pad, copy_sticky_events, gpad);
- gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), gpad);
+ expose_recv_src_pad (rtpbin, pad, stream, 255);
gst_object_unref (pad);
}
- GST_RTP_SESSION_UNLOCK (session);
- GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin);
-
return;
/* ERRORS */
}
}
-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)
+complete_session_sink (GstRtpBin * rtpbin, GstRtpBinSession * session)
{
guint sessid = session->id;
GstPad *recv_rtp_sink;
- GstPad *funnel_src;
GstElement *decoder;
g_assert (!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;
- 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);
- }
+ 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");
- 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");
- }
+ recv_rtp_sink = gst_object_ref (session->recv_rtp_sink);
}
- 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 recv_rtp_sink;
/* ERRORS */
return session->recv_rtp_sink_ghost;
/* setup the session sink pad */
- recv_rtp_sink = complete_session_sink (rtpbin, session, TRUE);
+ recv_rtp_sink = complete_session_sink (rtpbin, session);
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);
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;
static GstPad *
complete_session_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session,
- guint sessid, gboolean bundle_demuxer_needed)
+ guint sessid)
{
GstElement *decoder;
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;
- 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);
- }
+ 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");
- 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");
- }
+ decsink = gst_object_ref (session->recv_rtcp_sink);
}
/* get srcpad, link to SSRCDemux */
gst_pad_link_full (session->sync_src, sinkdpad, GST_PAD_LINK_CHECK_NOTHING);
gst_object_unref (sinkdpad);
- 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 decsink;
pad_failed:
if (session->recv_rtcp_sink_ghost != NULL)
return session->recv_rtcp_sink_ghost;
- decsink = complete_session_rtcp (rtpbin, session, sessid, TRUE);
+ decsink = complete_session_rtcp (rtpbin, session, sessid);
if (!decsink)
goto create_error;
}
existing_session:
{
- g_warning ("rtpbin: session %u is already a sender", sessid);
- return FALSE;
+ GST_DEBUG_OBJECT (rtpbin,
+ "skipping src_%i setup, since it is already configured.", sessid);
+ return TRUE;
}
pad_failed:
{
projects / platform / upstream / gst-plugins-good.git / blobdiff