-/* GStreamer
- * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.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.
- */
+ /* GStreamer
+ * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.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.
+ */
/**
* SECTION:element-rtpbin
* 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)
#define GST_RTP_BIN_SHUTDOWN_UNLOCK(bin) \
GST_RTP_BIN_DYN_UNLOCK (bin); \
+/* Minimum time offset to apply. This compensates for rounding errors in NTP to
+ * RTP timestamp conversions */
+#define MIN_TS_OFFSET (4 * GST_MSECOND)
+
struct _GstRtpBinPrivate
{
GMutex bin_lock;
SIGNAL_RESET_SYNC,
SIGNAL_GET_SESSION,
SIGNAL_GET_INTERNAL_SESSION,
+ SIGNAL_GET_STORAGE,
+ SIGNAL_GET_INTERNAL_STORAGE,
SIGNAL_ON_NEW_SSRC,
SIGNAL_ON_SSRC_COLLISION,
SIGNAL_REQUEST_RTCP_ENCODER,
SIGNAL_REQUEST_RTCP_DECODER,
+ SIGNAL_REQUEST_FEC_DECODER,
+ SIGNAL_REQUEST_FEC_ENCODER,
+
SIGNAL_NEW_JITTERBUFFER,
+ SIGNAL_NEW_STORAGE,
SIGNAL_REQUEST_AUX_SENDER,
SIGNAL_REQUEST_AUX_RECEIVER,
#define DEFAULT_MAX_MISORDER_TIME 2000
#define DEFAULT_RFC7273_SYNC FALSE
#define DEFAULT_MAX_STREAMS G_MAXUINT
+#define DEFAULT_MAX_TS_OFFSET_ADJUSTMENT G_GUINT64_CONSTANT(0)
+#define DEFAULT_MAX_TS_OFFSET G_GINT64_CONSTANT(3000000000)
enum
{
PROP_MAX_DROPOUT_TIME,
PROP_MAX_MISORDER_TIME,
PROP_RFC7273_SYNC,
- PROP_MAX_STREAMS
+ PROP_MAX_STREAMS,
+ PROP_MAX_TS_OFFSET_ADJUSTMENT,
+ PROP_MAX_TS_OFFSET,
};
#define GST_RTP_BIN_RTCP_SYNC_TYPE (gst_rtp_bin_rtcp_sync_get_type())
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.
*
* there they are pushed into an SSRC demuxer that splits the stream based on
* SSRC. Each of the SSRC streams go into their own jitterbuffer (managed with
* the GstRtpBinStream above).
+ *
+ * Before the SSRC demuxer, a storage element may be inserted for the purpose
+ * of Forward Error Correction.
*/
struct _GstRtpBinSession
{
gulong demux_newpad_sig;
gulong demux_padremoved_sig;
- /* Bundling support */
- GstElement *rtp_funnel;
- GstElement *rtcp_funnel;
- GstElement *bundle_demux;
- gulong bundle_demux_newpad_sig;
+ /* Fec support */
+ GstElement *storage;
GMutex lock;
GstPad *sync_src;
GstPad *send_rtp_sink;
GstPad *send_rtp_sink_ghost;
- GstPad *send_rtp_src;
GstPad *send_rtp_src_ghost;
GstPad *send_rtcp_src;
GstPad *send_rtcp_src_ghost;
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)
{
GstRtpBinSession *sess;
GstElement *session, *demux;
+ GstElement *storage = NULL;
GstState target;
if (!(session = gst_element_factory_make ("rtpsession", NULL)))
if (!(demux = gst_element_factory_make ("rtpssrcdemux", NULL)))
goto no_demux;
+ 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 = g_new0 (GstRtpBinSession, 1);
g_mutex_init (&sess->lock);
sess->id = 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->storage = storage;
sess->ptmap = g_hash_table_new_full (NULL, NULL, NULL,
(GDestroyNotify) gst_caps_unref);
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);
/* 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;
g_warning ("rtpbin: could not create rtpssrcdemux element");
return NULL;
}
+no_storage:
+ {
+ gst_object_unref (session);
+ gst_object_unref (demux);
+ g_warning ("rtpbin: could not create rtpstorage element");
+ return NULL;
+ }
}
static gboolean
GST_DEBUG_OBJECT (bin, "requested element %p already in bin", element);
} else {
GST_DEBUG_OBJECT (bin, "adding requested element %p", element);
+
+ if (g_object_is_floating (element))
+ element = gst_object_ref_sink (element);
+
if (!gst_bin_add (GST_BIN_CAST (bin), element))
goto add_failed;
if (!gst_element_sync_state_with_parent (element))
add_failed:
{
GST_WARNING_OBJECT (bin, "unable to add element");
+ gst_object_unref (element);
return FALSE;
}
}
if (find) {
priv->elements = g_list_delete_link (priv->elements, find);
- if (!g_list_find (priv->elements, element))
+ if (!g_list_find (priv->elements, element)) {
+ gst_element_set_locked_state (element, TRUE);
gst_bin_remove (GST_BIN_CAST (bin), element);
- else
- gst_object_unref (element);
+ gst_element_set_state (element, GST_STATE_NULL);
+ }
+
+ gst_object_unref (element);
}
}
gst_element_set_locked_state (sess->demux, TRUE);
gst_element_set_locked_state (sess->session, TRUE);
+ gst_element_set_locked_state (sess->storage, TRUE);
gst_element_set_state (sess->demux, GST_STATE_NULL);
gst_element_set_state (sess->session, GST_STATE_NULL);
+ gst_element_set_state (sess->storage, GST_STATE_NULL);
remove_recv_rtp (bin, sess);
remove_recv_rtcp (bin, sess);
gst_bin_remove (GST_BIN_CAST (bin), sess->session);
gst_bin_remove (GST_BIN_CAST (bin), sess->demux);
+ gst_bin_remove (GST_BIN_CAST (bin), sess->storage);
g_slist_foreach (sess->elements, (GFunc) remove_bin_element, bin);
g_slist_free (sess->elements);
}
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)
+{
+ GObject *internal_storage = NULL;
+ GstRtpBinSession *session;
+
+ 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) {
+ g_object_get (session->storage, "internal-storage", &internal_storage,
+ NULL);
+ }
+ GST_RTP_BIN_UNLOCK (bin);
+
+ return internal_storage;
+}
+
+static GstElement *
gst_rtp_bin_request_encoder (GstRtpBin * bin, guint session_id)
{
GST_DEBUG_OBJECT (bin, "return NULL encoder");
static void
stream_set_ts_offset (GstRtpBin * bin, GstRtpBinStream * stream,
- gint64 ts_offset, gboolean check)
+ gint64 ts_offset, gint64 max_ts_offset, gint64 min_ts_offset,
+ gboolean allow_positive_ts_offset)
{
gint64 prev_ts_offset;
"ts-offset %" G_GINT64_FORMAT ", prev %" G_GINT64_FORMAT
", diff: %" G_GINT64_FORMAT, ts_offset, prev_ts_offset, diff);
- if (check) {
- /* only change diff when it changed more than 4 milliseconds. This
- * compensates for rounding errors in NTP to RTP timestamp
- * conversions */
- if (ABS (diff) < 4 * GST_MSECOND) {
- GST_DEBUG_OBJECT (bin, "offset too small, ignoring");
+ /* ignore minor offsets */
+ if (ABS (diff) < min_ts_offset) {
+ GST_DEBUG_OBJECT (bin, "offset too small, ignoring");
+ return;
+ }
+
+ /* sanity check offset */
+ if (max_ts_offset > 0) {
+ if (ts_offset > 0 && !allow_positive_ts_offset) {
+ GST_DEBUG_OBJECT (bin,
+ "offset is positive (clocks are out of sync), ignoring");
return;
}
- if (ABS (diff) > (3 * GST_SECOND)) {
- GST_WARNING_OBJECT (bin, "offset unusually large, ignoring");
+ if (ABS (ts_offset) > max_ts_offset) {
+ GST_DEBUG_OBJECT (bin, "offset too large, ignoring");
return;
}
}
+
g_object_set (stream->buffer, "ts-offset", ts_offset, NULL);
}
GST_DEBUG_OBJECT (bin, "stream SSRC %08x, delta %" G_GINT64_FORMAT,
"local NTP time %" G_GUINT64_FORMAT ", SR NTP time %" G_GUINT64_FORMAT,
local_ntpnstime, ntpnstime);
GST_DEBUG_OBJECT (bin,
+ "local running time %" G_GUINT64_FORMAT ", SR RTP running time %"
+ G_GUINT64_FORMAT, local_running_time, running_time);
+ GST_DEBUG_OBJECT (bin,
"NTP diff %" G_GINT64_FORMAT ", RT diff %" G_GINT64_FORMAT, ntpdiff,
rtdiff);
/* combine to get the final diff to apply to the running_time */
stream->rt_delta = rtdiff - ntpdiff;
- stream_set_ts_offset (bin, stream, stream->rt_delta, FALSE);
+ stream_set_ts_offset (bin, stream, stream->rt_delta, bin->max_ts_offset,
+ 0, FALSE);
} else {
gint64 min, rtp_min, clock_base = stream->clock_base;
gboolean all_sync, use_rtp;
else
ts_offset = ostream->rt_delta - min;
- stream_set_ts_offset (bin, ostream, ts_offset, TRUE);
+ stream_set_ts_offset (bin, ostream, ts_offset, bin->max_ts_offset,
+ MIN_TS_OFFSET, TRUE);
}
}
gst_rtp_bin_send_sync_event (stream);
if (!(buffer = gst_element_factory_make ("rtpjitterbuffer", NULL)))
goto no_jitterbuffer;
- if (!rtpbin->ignore_pt)
+ if (!rtpbin->ignore_pt) {
if (!(demux = gst_element_factory_make ("rtpptdemux", NULL)))
goto no_demux;
+ }
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
if (session->bin->buffer_mode == RTP_JITTER_BUFFER_MODE_SLAVE)
if (!(queue2 = gst_element_factory_make ("queue2", NULL)))
g_object_set (buffer, "max-dropout-time", rtpbin->max_dropout_time,
"max-misorder-time", rtpbin->max_misorder_time, NULL);
g_object_set (buffer, "rfc7273-sync", rtpbin->rfc7273_sync, NULL);
+ g_object_set (buffer, "max-ts-offset-adjustment",
+ rtpbin->max_ts_offset_adjustment, NULL);
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
/* configure queue2 to use live buffering */
g_object_set (queue2, "buffer-mode", GST_BUFFERING_LIVE, NULL);
}
#endif
-
+ /* 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), buffer);
+ /* unref the jitterbuffer again, the bin has a reference now and
+ * we don't need it anymore */
+ gst_object_unref (buffer);
+
/* link stuff */
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
if (queue2) {
gst_element_link_pads_full (queue2, "src", demux, "sink",
GST_PAD_LINK_CHECK_NOTHING);
}
- } else if (demux)
- gst_element_link_pads_full (buffer, "src", demux, "sink",
- GST_PAD_LINK_CHECK_NOTHING);
-
+ } else if (demux) {
+ gst_element_link_pads_full (buffer, "src", demux, "sink",
+ GST_PAD_LINK_CHECK_NOTHING);
+ }
#else
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;
RTP_TYPE_SESSION, 1, G_TYPE_UINT);
/**
+ * GstRtpBin::get-internal-storage:
+ * @rtpbin: the object which received the signal
+ * @id: the session id
+ *
+ * Request the internal RTPStorage object as #GObject in session @id.
+ *
+ * Since: 1.14
+ */
+ gst_rtp_bin_signals[SIGNAL_GET_INTERNAL_STORAGE] =
+ g_signal_new ("get-internal-storage", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
+ get_internal_storage), NULL, NULL, g_cclosure_marshal_generic,
+ 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
G_TYPE_NONE, 3, GST_TYPE_ELEMENT, G_TYPE_UINT, G_TYPE_UINT);
/**
+ * GstRtpBin::new-storage:
+ * @rtpbin: the object which received the signal
+ * @storage: the new storage
+ * @session: the session
+ *
+ * Notify that a new @storage was created for @session.
+ * This signal can, for example, be used to configure @storage.
+ *
+ * Since: 1.14
+ */
+ gst_rtp_bin_signals[SIGNAL_NEW_STORAGE] =
+ g_signal_new ("new-storage", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
+ new_storage), NULL, NULL, g_cclosure_marshal_generic,
+ G_TYPE_NONE, 2, GST_TYPE_ELEMENT, G_TYPE_UINT);
+
+ /**
* GstRtpBin::request-aux-sender:
* @rtpbin: the object which received the signal
* @session: the session
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
request_aux_sender), _gst_element_accumulator, NULL,
g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
+
/**
* GstRtpBin::request-aux-receiver:
* @rtpbin: the object which received the signal
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
request_aux_receiver), _gst_element_accumulator, NULL,
g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
+
+ /**
+ * GstRtpBin::request-fec-decoder:
+ * @rtpbin: the object which received the signal
+ * @session: the session index
+ *
+ * Request a FEC decoder element for the given @session. The element
+ * will be added to the bin after the pt demuxer.
+ *
+ * If no handler is connected, no FEC decoder will be used.
+ *
+ * Since: 1.14
+ */
+ gst_rtp_bin_signals[SIGNAL_REQUEST_FEC_DECODER] =
+ g_signal_new ("request-fec-decoder", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
+ request_fec_decoder), _gst_element_accumulator, NULL,
+ g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
+
+ /**
+ * GstRtpBin::request-fec-encoder:
+ * @rtpbin: the object which received the signal
+ * @session: the session index
+ *
+ * Request a FEC encoder element for the given @session. The element
+ * will be added to the bin after the RTPSession.
+ *
+ * If no handler is connected, no FEC encoder will be used.
+ *
+ * Since: 1.14
+ */
+ gst_rtp_bin_signals[SIGNAL_REQUEST_FEC_ENCODER] =
+ g_signal_new ("request-fec-encoder", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
+ request_fec_encoder), _gst_element_accumulator, NULL,
+ g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
+
/**
* GstRtpBin::on-new-sender-ssrc:
* @rtpbin: the object which received the signal
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",
0, G_MAXUINT, DEFAULT_MAX_STREAMS,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ /**
+ * GstRtpBin:max-ts-offset-adjustment:
+ *
+ * Syncing time stamps to NTP time adds a time offset. This parameter
+ * specifies the maximum number of nanoseconds per frame that this time offset
+ * may be adjusted with. This is used to avoid sudden large changes to time
+ * stamps.
+ *
+ * Since: 1.14
+ */
+ g_object_class_install_property (gobject_class, PROP_MAX_TS_OFFSET_ADJUSTMENT,
+ g_param_spec_uint64 ("max-ts-offset-adjustment",
+ "Max Timestamp Offset Adjustment",
+ "The maximum number of nanoseconds per frame that time stamp offsets "
+ "may be adjusted (0 = no limit).", 0, G_MAXUINT64,
+ DEFAULT_MAX_TS_OFFSET_ADJUSTMENT, G_PARAM_READWRITE |
+ G_PARAM_STATIC_STRINGS));
+
+ /**
+ * GstRtpBin:max-ts-offset:
+ *
+ * Used to set an upper limit of how large a time offset may be. This
+ * is used to protect against unrealistic values as a result of either
+ * client,server or clock issues.
+ *
+ * Since: 1.14
+ */
+ g_object_class_install_property (gobject_class, PROP_MAX_TS_OFFSET,
+ g_param_spec_int64 ("max-ts-offset", "Max TS Offset",
+ "The maximum absolute value of the time offset in (nanoseconds). "
+ "Note, if the ntp-sync parameter is set the default value is "
+ "changed to 0 (no limit)", 0, G_MAXINT64, DEFAULT_MAX_TS_OFFSET,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_rtp_bin_change_state);
gstelement_class->request_new_pad =
GST_DEBUG_FUNCPTR (gst_rtp_bin_request_new_pad);
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);
klass->request_rtp_decoder = GST_DEBUG_FUNCPTR (gst_rtp_bin_request_decoder);
klass->request_rtcp_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);
rtpbin->max_misorder_time = DEFAULT_MAX_MISORDER_TIME;
rtpbin->rfc7273_sync = DEFAULT_RFC7273_SYNC;
rtpbin->max_streams = DEFAULT_MAX_STREAMS;
+ rtpbin->max_ts_offset_adjustment = DEFAULT_MAX_TS_OFFSET_ADJUSTMENT;
+ rtpbin->max_ts_offset = DEFAULT_MAX_TS_OFFSET;
+ rtpbin->max_ts_offset_is_set = FALSE;
/* some default SDES entries */
cname = g_strdup_printf ("user%u@host-%x", g_random_int (), g_random_int ());
break;
case PROP_NTP_SYNC:
rtpbin->ntp_sync = g_value_get_boolean (value);
+ /* The default value of max_ts_offset depends on ntp_sync. If user
+ * hasn't set it then change default value */
+ if (!rtpbin->max_ts_offset_is_set) {
+ if (rtpbin->ntp_sync) {
+ rtpbin->max_ts_offset = 0;
+ } else {
+ rtpbin->max_ts_offset = DEFAULT_MAX_TS_OFFSET;
+ }
+ }
break;
case PROP_RTCP_SYNC:
g_atomic_int_set (&rtpbin->rtcp_sync, g_value_get_enum (value));
case PROP_MAX_STREAMS:
rtpbin->max_streams = g_value_get_uint (value);
break;
+ case PROP_MAX_TS_OFFSET_ADJUSTMENT:
+ rtpbin->max_ts_offset_adjustment = g_value_get_uint64 (value);
+ gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin,
+ "max-ts-offset-adjustment", value);
+ break;
+ case PROP_MAX_TS_OFFSET:
+ rtpbin->max_ts_offset = g_value_get_int64 (value);
+ rtpbin->max_ts_offset_is_set = TRUE;
+ break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
case PROP_MAX_STREAMS:
g_value_set_uint (value, rtpbin->max_streams);
break;
+ case PROP_MAX_TS_OFFSET_ADJUSTMENT:
+ g_value_set_uint64 (value, rtpbin->max_ts_offset_adjustment);
+ break;
+ case PROP_MAX_TS_OFFSET:
+ g_value_set_int64 (value, rtpbin->max_ts_offset);
+ break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
GstStructure *caps_structure;
const gchar *caps_str_media = NULL;
temp_pad_src = gst_element_get_static_pad (stream->buffer, "src");
- temp_caps_src = gst_pad_get_current_caps(temp_pad_src);
- GST_DEBUG_OBJECT (bin, "stream %p percent %d : temp_caps_src=%"GST_PTR_FORMAT, stream,stream->percent,temp_caps_src);
- if (temp_caps_src)
- {
+ temp_caps_src = gst_pad_get_current_caps (temp_pad_src);
+ GST_DEBUG_OBJECT (bin,
+ "stream %p percent %d : temp_caps_src=%" GST_PTR_FORMAT,
+ stream, stream->percent, temp_caps_src);
+ if (temp_caps_src) {
caps_structure = gst_caps_get_structure (temp_caps_src, 0);
- caps_str_media = gst_structure_get_string (caps_structure, "media");
- if (caps_str_media != NULL)
- {
- if ((strcmp(caps_str_media,"video") != 0)&&(strcmp(caps_str_media,"audio") != 0))
- {
- GST_DEBUG_OBJECT (bin, "Non Audio/Video Stream.. ignoring the same !!");
- gst_caps_unref( temp_caps_src );
- gst_object_unref( temp_pad_src );
+ caps_str_media =
+ gst_structure_get_string (caps_structure, "media");
+ if (caps_str_media != NULL) {
+ if ((strcmp (caps_str_media, "video") != 0)
+ && (strcmp (caps_str_media, "audio") != 0)) {
+ GST_DEBUG_OBJECT (bin,
+ "Non Audio/Video Stream.. ignoring the same !!");
+ gst_caps_unref (temp_caps_src);
+ gst_object_unref (temp_pad_src);
continue;
- }
- else if(stream->percent >= 100)
- {
+ } else if (stream->percent >= 100) {
/* Most of the time buffering icon displays in rtsp playback.
- Optimizing the buffering updation code. Whenever any stream percentage
- reaches 100 do not post buffering messages.*/
- if(stream->prev_percent < 100)
- {
+ Optimizing the buffering updation code. Whenever any stream percentage
+ reaches 100 do not post buffering messages. */
+ if (stream->prev_percent < 100)
buffering_flag = TRUE;
- }
else
- {
update_buffering_status = FALSE;
- }
}
}
- gst_caps_unref( temp_caps_src );
+ gst_caps_unref (temp_caps_src);
}
- gst_object_unref( temp_pad_src );
+ gst_object_unref (temp_pad_src);
#else
GST_DEBUG_OBJECT (bin, "stream %p percent %d", stream,
stream->percent);
gst_message_unref (message);
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- if (rtpbin->buffer_mode == RTP_JITTER_BUFFER_MODE_SLAVE)
- {
- if(update_buffering_status==FALSE)
- {
- break;
- }
- if(buffering_flag)
- {
- min_percent=100;
- GST_DEBUG_OBJECT (bin, "forcefully change min_percent to 100!!!");
- }
- }
+ if (rtpbin->buffer_mode == RTP_JITTER_BUFFER_MODE_SLAVE) {
+ if (update_buffering_status == FALSE)
+ break;
+ if (buffering_flag) {
+ min_percent = 100;
+ GST_DEBUG_OBJECT (bin, "forcefully change min_percent to 100!!!");
+ }
+ }
#endif
/* make a new buffering message with the min value */
message =
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_object_ref (pad);
- GST_DEBUG_OBJECT (rtpbin, "new payload pad %u", pt);
+ if (stream->session->storage) {
+ GstElement *fec_decoder =
+ session_request_element (stream->session, SIGNAL_REQUEST_FEC_DECODER);
+
+ if (fec_decoder) {
+ GstPad *sinkpad, *srcpad;
+ GstPadLinkReturn ret;
+
+ sinkpad = gst_element_get_static_pad (fec_decoder, "sink");
+
+ if (!sinkpad)
+ goto fec_decoder_sink_failed;
+
+ ret = gst_pad_link (pad, sinkpad);
+ gst_object_unref (sinkpad);
+
+ if (ret != GST_PAD_LINK_OK)
+ goto fec_decoder_link_failed;
+
+ srcpad = gst_element_get_static_pad (fec_decoder, "src");
+
+ if (!srcpad)
+ goto fec_decoder_src_failed;
+
+ gst_pad_sticky_events_foreach (pad, copy_sticky_events, srcpad);
+ gst_object_unref (pad);
+ pad = srcpad;
+ }
+ }
GST_RTP_BIN_SHUTDOWN_LOCK (rtpbin, shutdown);
gst_pad_sticky_events_foreach (pad, copy_sticky_events, gpad);
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), gpad);
+done:
+ gst_object_unref (pad);
+
return;
shutdown:
{
GST_DEBUG ("ignoring, we are shutting down");
- return;
+ goto done;
+ }
+fec_decoder_sink_failed:
+ {
+ g_warning ("rtpbin: failed to get fec encoder sink pad for session %u",
+ stream->session->id);
+ goto done;
+ }
+fec_decoder_src_failed:
+ {
+ g_warning ("rtpbin: failed to get fec encoder src pad for session %u",
+ stream->session->id);
+ goto done;
+ }
+fec_decoder_link_failed:
+ {
+ g_warning ("rtpbin: failed to link fec decoder for session %u",
+ stream->session->id);
+ goto done;
}
}
+/* 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)
}
}
+static GstCaps *
+ptdemux_pt_map_requested (GstElement * element, guint pt,
+ GstRtpBinSession * session)
+{
+ GstCaps *ret = pt_map_requested (element, pt, session);
+
+ if (ret && gst_caps_get_size (ret) == 1) {
+ const GstStructure *s = gst_caps_get_structure (ret, 0);
+ gboolean is_fec;
+
+ if (gst_structure_get_boolean (s, "is-fec", &is_fec) && is_fec) {
+ GValue v = G_VALUE_INIT;
+ GValue v2 = G_VALUE_INIT;
+
+ GST_INFO_OBJECT (session->bin, "Will ignore FEC pt %u in session %u", pt,
+ session->id);
+ g_value_init (&v, GST_TYPE_ARRAY);
+ g_value_init (&v2, G_TYPE_INT);
+ g_object_get_property (G_OBJECT (element), "ignored-payload-types", &v);
+ g_value_set_int (&v2, pt);
+ gst_value_array_append_value (&v, &v2);
+ g_value_unset (&v2);
+ g_object_set_property (G_OBJECT (element), "ignored-payload-types", &v);
+ g_value_unset (&v);
+ }
+ }
+
+ return ret;
+}
+
static void
payload_type_change (GstElement * element, guint pt, GstRtpBinSession * session)
{
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,
- "request-pt-map", (GCallback) pt_map_requested, session);
+ "request-pt-map", (GCallback) ptdemux_pt_map_requested, session);
/* 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 */
if (session->recv_rtp_src == NULL)
goto pad_failed;
- /* find out if we need AUX elements or if we can go into the SSRC demuxer
- * directly */
+ /* find out if we need AUX elements */
aux = session_request_element (session, SIGNAL_REQUEST_AUX_RECEIVER);
if (aux) {
gchar *pname;
if (ret != GST_PAD_LINK_OK)
goto aux_link_failed;
- /* this can be NULL when this AUX element is not to be linked to
- * an SSRC demuxer */
+ /* this can be NULL when this AUX element is not to be linked any further */
pname = g_strdup_printf ("src_%u", sessid);
recv_rtp_src = gst_element_get_static_pad (aux, pname);
g_free (pname);
recv_rtp_src = gst_object_ref (session->recv_rtp_src);
}
+ /* Add a storage element if needed */
+ if (recv_rtp_src && session->storage) {
+ GstPadLinkReturn ret;
+ GstPad *sinkpad = gst_element_get_static_pad (session->storage, "sink");
+
+ ret = gst_pad_link (recv_rtp_src, sinkpad);
+
+ gst_object_unref (sinkpad);
+ gst_object_unref (recv_rtp_src);
+
+ if (ret != GST_PAD_LINK_OK)
+ goto storage_link_failed;
+
+ recv_rtp_src = gst_element_get_static_pad (session->storage, "src");
+ }
+
if (recv_rtp_src) {
GstPad *sinkdpad;
g_warning ("rtpbin: failed to link AUX pad to session %u", sessid);
return;
}
+storage_link_failed:
+ {
+ g_warning ("rtpbin: failed to link storage");
+ return;
+ }
}
/* Create a pad for receiving RTP for the session in @name. Must be called with
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;
GstElement *encoder;
GstElementClass *klass;
GstPadTemplate *templ;
+ gboolean ret = FALSE;
/* get srcpad */
- session->send_rtp_src =
- gst_element_get_static_pad (session->session, "send_rtp_src");
- if (session->send_rtp_src == NULL)
+ send_rtp_src = gst_element_get_static_pad (session->session, "send_rtp_src");
+
+ if (send_rtp_src == NULL)
goto no_srcpad;
GST_DEBUG_OBJECT (rtpbin, "getting RTP encoder");
if (encsrc == NULL)
goto enc_src_failed;
- send_rtp_src = encsrc;
-
ename = g_strdup_printf ("rtp_sink_%u", sessid);
encsink = gst_element_get_static_pad (encoder, ename);
g_free (ename);
if (encsink == NULL)
goto enc_sink_failed;
- ret = gst_pad_link (session->send_rtp_src, encsink);
+ ret = gst_pad_link (send_rtp_src, encsink);
gst_object_unref (encsink);
+ gst_object_unref (send_rtp_src);
+
+ send_rtp_src = encsrc;
if (ret != GST_PAD_LINK_OK)
goto enc_link_failed;
} else {
GST_DEBUG_OBJECT (rtpbin, "no RTP encoder given");
- send_rtp_src = gst_object_ref (session->send_rtp_src);
}
/* ghost the new source pad */
templ = gst_element_class_get_pad_template (klass, "send_rtp_src_%u");
session->send_rtp_src_ghost =
gst_ghost_pad_new_from_template (gname, send_rtp_src, templ);
- gst_object_unref (send_rtp_src);
gst_pad_set_active (session->send_rtp_src_ghost, TRUE);
gst_pad_sticky_events_foreach (send_rtp_src, copy_sticky_events,
session->send_rtp_src_ghost);
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->send_rtp_src_ghost);
g_free (gname);
- return TRUE;
+ ret = TRUE;
+
+done:
+ if (send_rtp_src)
+ gst_object_unref (send_rtp_src);
+
+ return ret;
/* ERRORS */
no_srcpad:
{
g_warning ("rtpbin: failed to get rtp source pad for session %u", sessid);
- return FALSE;
+ goto done;
}
enc_src_failed:
{
- g_warning ("rtpbin: failed to get encoder src pad for session %u", sessid);
- return FALSE;
+ g_warning ("rtpbin: failed to get %" GST_PTR_FORMAT
+ " src pad for session %u", encoder, sessid);
+ goto done;
}
enc_sink_failed:
{
- g_warning ("rtpbin: failed to get encoder sink pad for session %u", sessid);
- gst_object_unref (send_rtp_src);
- return FALSE;
+ g_warning ("rtpbin: failed to get %" GST_PTR_FORMAT
+ " sink pad for session %u", encoder, sessid);
+ goto done;
}
enc_link_failed:
{
- g_warning ("rtpbin: failed to link rtp encoder for session %u", sessid);
- gst_object_unref (send_rtp_src);
- return FALSE;
+ g_warning ("rtpbin: failed to link %" GST_PTR_FORMAT " for session %u",
+ encoder, sessid);
+ goto done;
}
}
}
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:
{
guint sessid;
GstPad *send_rtp_sink;
GstElement *aux;
+ GstElement *encoder;
+ GstElement *prev = NULL;
GstRtpBinSession *session;
/* first get the session number */
if (session->send_rtp_sink != NULL)
goto existing_session;
+ encoder = session_request_element (session, SIGNAL_REQUEST_FEC_ENCODER);
+
+ if (encoder) {
+ GST_DEBUG_OBJECT (rtpbin, "Linking FEC encoder");
+
+ send_rtp_sink = gst_element_get_static_pad (encoder, "sink");
+
+ if (!send_rtp_sink)
+ goto enc_sink_failed;
+
+ prev = encoder;
+ }
+
GST_DEBUG_OBJECT (rtpbin, "getting RTP AUX sender");
aux = session_request_element (session, SIGNAL_REQUEST_AUX_SENDER);
if (aux) {
+ GstPad *sinkpad;
GST_DEBUG_OBJECT (rtpbin, "linking AUX sender");
if (!setup_aux_sender (rtpbin, session, aux))
goto aux_session_failed;
pname = g_strdup_printf ("sink_%u", sessid);
- send_rtp_sink = gst_element_get_static_pad (aux, pname);
+ sinkpad = gst_element_get_static_pad (aux, pname);
g_free (pname);
- if (send_rtp_sink == NULL)
+ if (sinkpad == NULL)
goto aux_sink_failed;
+
+ if (!prev) {
+ send_rtp_sink = sinkpad;
+ } else {
+ GstPad *srcpad = gst_element_get_static_pad (prev, "src");
+ GstPadLinkReturn ret;
+
+ ret = gst_pad_link (srcpad, sinkpad);
+ gst_object_unref (srcpad);
+ if (ret != GST_PAD_LINK_OK) {
+ goto aux_link_failed;
+ }
+ }
+ prev = aux;
} else {
/* get send_rtp pad and store */
session->send_rtp_sink =
if (!complete_session_src (rtpbin, session))
goto session_src_failed;
- send_rtp_sink = gst_object_ref (session->send_rtp_sink);
+ if (!prev) {
+ send_rtp_sink = gst_object_ref (session->send_rtp_sink);
+ } else {
+ GstPad *srcpad = gst_element_get_static_pad (prev, "src");
+ GstPadLinkReturn ret;
+
+ ret = gst_pad_link (srcpad, session->send_rtp_sink);
+ gst_object_unref (srcpad);
+ if (ret != GST_PAD_LINK_OK)
+ goto session_link_failed;
+ }
}
session->send_rtp_sink_ghost =
g_warning ("rtpbin: failed to get AUX sink pad for session %u", sessid);
return NULL;
}
+aux_link_failed:
+ {
+ g_warning ("rtpbin: failed to link %" GST_PTR_FORMAT " for session %u",
+ aux, sessid);
+ return NULL;
+ }
pad_failed:
{
g_warning ("rtpbin: failed to get session pad for session %u", sessid);
g_warning ("rtpbin: failed to setup source pads for session %u", sessid);
return NULL;
}
+session_link_failed:
+ {
+ g_warning ("rtpbin: failed to link %" GST_PTR_FORMAT " for session %u",
+ session, sessid);
+ return NULL;
+ }
+enc_sink_failed:
+ {
+ g_warning ("rtpbin: failed to get %" GST_PTR_FORMAT
+ " sink pad for session %u", encoder, sessid);
+ return NULL;
+ }
}
static void
session->send_rtp_src_ghost);
session->send_rtp_src_ghost = NULL;
}
- if (session->send_rtp_src) {
- gst_object_unref (session->send_rtp_src);
- session->send_rtp_src = NULL;
- }
if (session->send_rtp_sink) {
gst_element_release_request_pad (GST_ELEMENT_CAST (session->session),
session->send_rtp_sink);
* RTP_BIN_LOCK.
*/
static GstPad *
-create_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
+create_send_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ,
+ const gchar * name)
{
guint sessid;
GstPad *encsrc;
/* get or create session */
session = find_session_by_id (rtpbin, sessid);
- if (!session)
- goto no_session;
+ 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->send_rtcp_src_ghost != NULL)
g_warning ("rtpbin: invalid name given");
return NULL;
}
-no_session:
+create_error:
{
- g_warning ("rtpbin: session with id %d does not exist", sessid);
+ /* create_session already warned */
return NULL;
}
pad_failed:
result = create_send_rtp (rtpbin, templ, pad_name);
} else if (templ == gst_element_class_get_pad_template (klass,
"send_rtcp_src_%u")) {
- result = create_rtcp (rtpbin, templ, pad_name);
+ result = create_send_rtcp (rtpbin, templ, pad_name);
} else
goto wrong_template;
projects / platform / upstream / gst-plugins-good.git / blobdiff