/* GStreamer
- * Copyright (C) <2007> Wim Taymans <wim@fluendo.com>
+ * 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
*/
/**
- * SECTION:element-rtpbin
+ * SECTION:element-gstrtpbin
* @short_description: handle media from one RTP bin
- * @see_also: rtpjitterbuffer, rtpclient, rtpsession
+ * @see_also: gstrtpjitterbuffer, gstrtpsession, gstrtpptdemux, gstrtpssrcdemux
*
* <refsect2>
* <para>
+ * RTP bin combines the functions of gstrtpsession, gstrtpssrcdemux, gstrtpjitterbuffer
+ * and gstrtpptdemux in one element. It allows for multiple RTP sessions that will
+ * be synchronized together using RTCP SR packets.
+ * </para>
+ * <para>
+ * gstrtpbin is configured with a number of request pads that define the
+ * functionality that is activated, similar to the gstrtpsession element.
+ * </para>
+ * <para>
+ * To use gstrtpbin as an RTP receiver, request a recv_rtp_sink_%%d pad. The session
+ * number must be specified in the pad name.
+ * Data received on the recv_rtp_sink_%%d pad will be processed in the gstrtpsession
+ * manager and after being validated forwarded on gstrtpssrcdemuxer element. Each
+ * RTP stream is demuxed based on the SSRC and send to a gstrtpjitterbuffer. After
+ * the packets are released from the jitterbuffer, they will be forwarded to a
+ * gstrtpptdemuxer element. The gstrtpptdemuxer element will demux the packets based
+ * on the payload type and will create a unique pad recv_rtp_src_%%d_%%d_%%d on
+ * gstrtpbin with the session number, SSRC and payload type respectively as the pad
+ * name.
+ * </para>
+ * <para>
+ * To also use gstrtpbin as an RTCP receiver, request a recv_rtcp_sink_%%d pad. The
+ * session number must be specified in the pad name.
+ * </para>
+ * <para>
+ * If you want the session manager to generate and send RTCP packets, request
+ * the send_rtcp_src_%%d pad with the session number in the pad name. Packet pushed
+ * on this pad contain SR/RR RTCP reports that should be sent to all participants
+ * in the session.
+ * </para>
+ * <para>
+ * To use gstrtpbin as a sender, request a send_rtp_sink_%%d pad, which will
+ * automatically create a send_rtp_src_%%d pad. If the session number is not provided,
+ * the pad from the lowest available session will be returned. The session manager will modify the
+ * SSRC in the RTP packets to its own SSRC and wil forward the packets on the
+ * send_rtp_src_%%d pad after updating its internal state.
+ * </para>
+ * <para>
+ * 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.
* </para>
* <title>Example pipelines</title>
* <para>
* <programlisting>
- * gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! alsasink
+ * gst-launch udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink_0 \
+ * gstrtpbin ! rtptheoradepay ! theoradec ! xvimagesink
* </programlisting>
+ * Receive RTP data from port 5000 and send to the session 0 in gstrtpbin.
+ * </para>
+ * <para>
+ * <programlisting>
+ * gst-launch gstrtpbin name=rtpbin \
+ * v4l2src ! ffmpegcolorspace ! ffenc_h263 ! rtph263ppay ! rtpbin.send_rtp_sink_0 \
+ * rtpbin.send_rtp_src_0 ! udpsink port=5000 \
+ * rtpbin.send_rtcp_src_0 ! udpsink port=5001 sync=false async=false \
+ * udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 \
+ * audiotestsrc ! amrnbenc ! rtpamrpay ! rtpbin.send_rtp_sink_1 \
+ * rtpbin.send_rtp_src_1 ! udpsink port=5002 \
+ * rtpbin.send_rtcp_src_1 ! udpsink port=5003 sync=false async=false \
+ * udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1
+ * </programlisting>
+ * Encode and payload H263 video captured from a v4l2src. Encode and payload AMR
+ * audio generated from audiotestsrc. The video is sent to session 0 in rtpbin
+ * and the audio is sent to session 1. Video packets are sent on UDP port 5000
+ * and audio packets on port 5002. The video RTCP packets for session 0 are sent
+ * on port 5001 and the audio RTCP packets for session 0 are sent on port 5003.
+ * RTCP packets for session 0 are received on port 5005 and RTCP for session 1
+ * is received on port 5007. Since RTCP packets from the sender should be sent
+ * as soon as possible and do not participate in preroll, sync=false and
+ * async=false is configured on udpsink
+ * </para>
+ * <para>
+ * <programlisting>
+ * gst-launch -v gstrtpbin name=rtpbin \
+ * udpsrc caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H263-1998" \
+ * port=5000 ! rtpbin.recv_rtp_sink_0 \
+ * rtpbin. ! rtph263pdepay ! ffdec_h263 ! xvimagesink \
+ * udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \
+ * rtpbin.send_rtcp_src_0 ! udpsink port=5005 sync=false async=false \
+ * udpsrc caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)AMR,encoding-params=(string)1,octet-align=(string)1" \
+ * port=5002 ! rtpbin.recv_rtp_sink_1 \
+ * rtpbin. ! rtpamrdepay ! amrnbdec ! alsasink \
+ * udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \
+ * rtpbin.send_rtcp_src_1 ! udpsink port=5007 sync=false async=false
+ * </programlisting>
+ * Receive H263 on port 5000, send it through rtpbin in session 0, depayload,
+ * decode and display the video.
+ * Receive AMR on port 5002, send it through rtpbin in session 1, depayload,
+ * decode and play the audio.
+ * Receive server RTCP packets for session 0 on port 5001 and RTCP packets for
+ * session 1 on port 5003. These packets will be used for session management and
+ * synchronisation.
+ * Send RTCP reports for session 0 on port 5005 and RTCP reports for session 1
+ * on port 5007.
* </para>
* </refsect2>
*
- * Last reviewed on 2007-04-02 (0.10.6)
+ * Last reviewed on 2007-08-30 (0.10.6)
*/
#ifdef HAVE_CONFIG_H
#endif
#include <string.h>
+#include <gst/rtp/gstrtpbuffer.h>
+#include <gst/rtp/gstrtcpbuffer.h>
+
+#include "gstrtpbin-marshal.h"
#include "gstrtpbin.h"
GST_DEBUG_CATEGORY_STATIC (gst_rtp_bin_debug);
#define GST_CAT_DEFAULT gst_rtp_bin_debug
-
/* elementfactory information */
static const GstElementDetails rtpbin_details = GST_ELEMENT_DETAILS ("RTP Bin",
- "Filter/Editor/Video",
+ "Filter/Network/RTP",
"Implement an RTP bin",
- "Wim Taymans <wim@fluendo.com>");
+ "Wim Taymans <wim.taymans@gmail.com>");
/* sink pads */
static GstStaticPadTemplate rtpbin_recv_rtp_sink_template =
GST_STATIC_CAPS ("application/x-rtp")
);
-static GstStaticPadTemplate rtpbin_rtcp_src_template =
-GST_STATIC_PAD_TEMPLATE ("rtcp_src_%d",
+static GstStaticPadTemplate rtpbin_send_rtcp_src_template =
+GST_STATIC_PAD_TEMPLATE ("send_rtcp_src_%d",
GST_PAD_SRC,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("application/x-rtcp")
GST_STATIC_CAPS ("application/x-rtp")
);
+/* padtemplate for the internal pad */
+static GstStaticPadTemplate rtpbin_sync_sink_template =
+GST_STATIC_PAD_TEMPLATE ("sink_%d",
+ GST_PAD_SINK,
+ GST_PAD_SOMETIMES,
+ GST_STATIC_CAPS ("application/x-rtcp")
+ );
+
#define GST_RTP_BIN_GET_PRIVATE(obj) \
- (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTP_BIN, GstRTPBinPrivate))
+ (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)
-struct _GstRTPBinPrivate
+struct _GstRtpBinPrivate
{
- guint foo;
+ GMutex *bin_lock;
+
+ GstClockTime ntp_ns_base;
};
/* signals and args */
enum
{
- /* FILL ME */
+ SIGNAL_REQUEST_PT_MAP,
+ SIGNAL_CLEAR_PT_MAP,
+
+ SIGNAL_ON_NEW_SSRC,
+ SIGNAL_ON_SSRC_COLLISION,
+ SIGNAL_ON_SSRC_VALIDATED,
+ SIGNAL_ON_SSRC_ACTIVE,
+ SIGNAL_ON_SSRC_SDES,
+ SIGNAL_ON_BYE_SSRC,
+ SIGNAL_ON_BYE_TIMEOUT,
+ SIGNAL_ON_TIMEOUT,
LAST_SIGNAL
};
+#define DEFAULT_LATENCY_MS 200
+#define DEFAULT_SDES_CNAME NULL
+#define DEFAULT_SDES_NAME NULL
+#define DEFAULT_SDES_EMAIL NULL
+#define DEFAULT_SDES_PHONE NULL
+#define DEFAULT_SDES_LOCATION NULL
+#define DEFAULT_SDES_TOOL NULL
+#define DEFAULT_SDES_NOTE NULL
+#define DEFAULT_DO_LOST FALSE
+
enum
{
- PROP_0
+ PROP_0,
+ PROP_LATENCY,
+ PROP_SDES_CNAME,
+ PROP_SDES_NAME,
+ PROP_SDES_EMAIL,
+ PROP_SDES_PHONE,
+ PROP_SDES_LOCATION,
+ PROP_SDES_TOOL,
+ PROP_SDES_NOTE,
+ PROP_DO_LOST,
+ PROP_LAST
};
/* helper objects */
-typedef struct _GstRTPBinSession GstRTPBinSession;
-typedef struct _GstRTPBinStream GstRTPBinStream;
-typedef struct _GstRTPBinClient GstRTPBinClient;
+typedef struct _GstRtpBinSession GstRtpBinSession;
+typedef struct _GstRtpBinStream GstRtpBinStream;
+typedef struct _GstRtpBinClient GstRtpBinClient;
+
+static guint gst_rtp_bin_signals[LAST_SIGNAL] = { 0 };
+
+static GstCaps *pt_map_requested (GstElement * element, guint pt,
+ GstRtpBinSession * session);
+static const gchar *sdes_type_to_name (GstRTCPSDESType type);
+static void gst_rtp_bin_set_sdes_string (GstRtpBin * bin,
+ GstRTCPSDESType type, const gchar * data);
+
+static void free_stream (GstRtpBinStream * stream);
/* Manages the RTP stream for one SSRC.
*
* We pipe the stream (comming from the SSRC demuxer) into a jitterbuffer.
* If we see an SDES RTCP packet that links multiple SSRCs together based on a
- * common CNAME, we create a GstRTPBinClient structure to group the SSRCs
+ * common CNAME, we create a GstRtpBinClient structure to group the SSRCs
* together (see below).
*/
-struct _GstRTPBinStream
+struct _GstRtpBinStream
{
/* the SSRC of this stream */
guint32 ssrc;
+
/* parent bin */
- GstRTPBin *bin;
+ GstRtpBin *bin;
+
/* the session this SSRC belongs to */
- GstRTPBinSession *session;
+ GstRtpBinSession *session;
+
/* the jitterbuffer of the SSRC */
GstElement *buffer;
+
/* the PT demuxer of the SSRC */
GstElement *demux;
gulong demux_newpad_sig;
+ gulong demux_ptreq_sig;
+ gulong demux_pt_change_sig;
+
+ /* the internal pad we use to get RTCP sync messages */
+ GstPad *sync_pad;
+ gboolean have_sync;
+ guint64 last_unix;
+ guint64 last_extrtptime;
+
+ /* mapping to local RTP and NTP time */
+ guint64 local_rtp;
+ guint64 local_unix;
+ gint64 unix_delta;
+
+ /* for lip-sync */
+ guint64 clock_base;
+ guint64 clock_base_time;
+ gint clock_rate;
+ gint64 ts_offset;
+ gint64 prev_ts_offset;
+ gint last_pt;
};
+#define GST_RTP_SESSION_LOCK(sess) g_mutex_lock ((sess)->lock)
+#define GST_RTP_SESSION_UNLOCK(sess) g_mutex_unlock ((sess)->lock)
+
/* Manages the receiving end of the packets.
*
* There is one such structure for each RTP session (audio/video/...).
* We get the RTP/RTCP packets and stuff them into the session manager. From
* 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).
+ * the GstRtpBinStream above).
*/
-struct _GstRTPBinSession
+struct _GstRtpBinSession
{
/* session id */
gint id;
/* the parent bin */
- GstRTPBin *bin;
+ GstRtpBin *bin;
/* the session element */
GstElement *session;
/* the SSRC demuxer */
GstElement *demux;
gulong demux_newpad_sig;
- /* list of GstRTPBinStream */
+ GMutex *lock;
+
+ /* list of GstRtpBinStream */
GSList *streams;
+ /* mapping of payload type to caps */
+ GHashTable *ptmap;
+
/* the pads of the session */
GstPad *recv_rtp_sink;
GstPad *recv_rtp_src;
GstPad *recv_rtcp_sink;
- GstPad *recv_rtcp_src;
+ GstPad *sync_src;
GstPad *send_rtp_sink;
GstPad *send_rtp_src;
- GstPad *rtcp_src;
+ GstPad *send_rtcp_src;
+};
+
+/* Manages the RTP streams that come from one client and should therefore be
+ * synchronized.
+ */
+struct _GstRtpBinClient
+{
+ /* the common CNAME for the streams */
+ gchar *cname;
+ guint cname_len;
+
+ /* the streams */
+ guint nstreams;
+ GSList *streams;
+
+ gint64 min_delta;
};
-/* find a session with the given id */
-static GstRTPBinSession *
-find_session_by_id (GstRTPBin * rtpbin, gint id)
+/* find a session with the given id. Must be called with RTP_BIN_LOCK */
+static GstRtpBinSession *
+find_session_by_id (GstRtpBin * rtpbin, gint id)
{
GSList *walk;
for (walk = rtpbin->sessions; walk; walk = g_slist_next (walk)) {
- GstRTPBinSession *sess = (GstRTPBinSession *) walk->data;
+ GstRtpBinSession *sess = (GstRtpBinSession *) walk->data;
if (sess->id == id)
return sess;
return NULL;
}
-/* create a session with the given id */
-static GstRTPBinSession *
-create_session (GstRTPBin * rtpbin, gint id)
+static void
+on_new_ssrc (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
{
- GstRTPBinSession *sess;
- GstElement *elem, *demux;
+ g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_NEW_SSRC], 0,
+ sess->id, ssrc);
+}
+
+static void
+on_ssrc_collision (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
+{
+ g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SSRC_COLLISION], 0,
+ sess->id, ssrc);
+}
+
+static void
+on_ssrc_validated (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
+{
+ g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SSRC_VALIDATED], 0,
+ sess->id, ssrc);
+}
+
+static void
+on_ssrc_active (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
+{
+ g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SSRC_ACTIVE], 0,
+ sess->id, ssrc);
+}
+
+static void
+on_ssrc_sdes (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
+{
+ g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SSRC_SDES], 0,
+ sess->id, ssrc);
+}
+
+static void
+on_bye_ssrc (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
+{
+ g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_BYE_SSRC], 0,
+ sess->id, ssrc);
+}
- if (!(elem = gst_element_factory_make ("rtpsession", NULL)))
+static void
+on_bye_timeout (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
+{
+ g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_BYE_TIMEOUT], 0,
+ sess->id, ssrc);
+}
+
+static void
+on_timeout (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
+{
+ g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_TIMEOUT], 0,
+ sess->id, ssrc);
+}
+
+/* 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;
+ gint i;
+
+ if (!(session = gst_element_factory_make ("gstrtpsession", NULL)))
goto no_session;
- if (!(demux = gst_element_factory_make ("rtpssrcdemux", NULL)))
+ if (!(demux = gst_element_factory_make ("gstrtpssrcdemux", NULL)))
goto no_demux;
- sess = g_new0 (GstRTPBinSession, 1);
+ sess = g_new0 (GstRtpBinSession, 1);
+ sess->lock = g_mutex_new ();
sess->id = id;
sess->bin = rtpbin;
- sess->session = elem;
+ sess->session = session;
sess->demux = demux;
+ sess->ptmap = g_hash_table_new_full (NULL, NULL, NULL,
+ (GDestroyNotify) gst_caps_unref);
rtpbin->sessions = g_slist_prepend (rtpbin->sessions, sess);
- gst_bin_add (GST_BIN_CAST (rtpbin), elem);
- gst_element_set_state (elem, GST_STATE_PLAYING);
+ /* set NTP base or new session */
+ g_object_set (session, "ntp-ns-base", rtpbin->priv->ntp_ns_base, NULL);
+ /* configure SDES items */
+ GST_OBJECT_LOCK (rtpbin);
+ for (i = GST_RTCP_SDES_CNAME; i < GST_RTCP_SDES_PRIV; i++) {
+ g_object_set (session, sdes_type_to_name (i), rtpbin->sdes[i], NULL);
+ }
+ GST_OBJECT_UNLOCK (rtpbin);
+
+ /* provide clock_rate to the session manager when needed */
+ g_signal_connect (session, "request-pt-map",
+ (GCallback) pt_map_requested, sess);
+
+ g_signal_connect (sess->session, "on-new-ssrc",
+ (GCallback) on_new_ssrc, sess);
+ g_signal_connect (sess->session, "on-ssrc-collision",
+ (GCallback) on_ssrc_collision, sess);
+ g_signal_connect (sess->session, "on-ssrc-validated",
+ (GCallback) on_ssrc_validated, sess);
+ g_signal_connect (sess->session, "on-ssrc-active",
+ (GCallback) on_ssrc_active, sess);
+ g_signal_connect (sess->session, "on-ssrc-sdes",
+ (GCallback) on_ssrc_sdes, sess);
+ g_signal_connect (sess->session, "on-bye-ssrc",
+ (GCallback) on_bye_ssrc, sess);
+ g_signal_connect (sess->session, "on-bye-timeout",
+ (GCallback) on_bye_timeout, sess);
+ g_signal_connect (sess->session, "on-timeout", (GCallback) on_timeout, sess);
+
+ /* FIXME, change state only to what's needed */
+ gst_bin_add (GST_BIN_CAST (rtpbin), session);
+ gst_element_set_state (session, GST_STATE_PLAYING);
gst_bin_add (GST_BIN_CAST (rtpbin), demux);
gst_element_set_state (demux, GST_STATE_PLAYING);
/* ERRORS */
no_session:
{
- g_warning ("rtpbin: could not create rtpsession element");
+ g_warning ("gstrtpbin: could not create gstrtpsession element");
return NULL;
}
no_demux:
{
- gst_object_unref (elem);
- g_warning ("rtpbin: could not create rtpssrcdemux element");
+ gst_object_unref (session);
+ g_warning ("gstrtpbin: could not create gstrtpssrcdemux element");
return NULL;
}
}
+static void
+free_session (GstRtpBinSession * sess)
+{
+ GstRtpBin *bin;
+
+ bin = sess->bin;
+
+ gst_element_set_state (sess->session, GST_STATE_NULL);
+ gst_element_set_state (sess->demux, GST_STATE_NULL);
+
+ if (sess->recv_rtp_sink != NULL)
+ gst_element_release_request_pad (sess->session, sess->recv_rtp_sink);
+ if (sess->recv_rtp_src != NULL)
+ gst_object_unref (sess->recv_rtp_src);
+ if (sess->recv_rtcp_sink != NULL)
+ gst_element_release_request_pad (sess->session, sess->recv_rtcp_sink);
+ if (sess->sync_src != NULL)
+ gst_object_unref (sess->sync_src);
+ if (sess->send_rtp_sink != NULL)
+ gst_element_release_request_pad (sess->session, sess->send_rtp_sink);
+ if (sess->send_rtp_src != NULL)
+ gst_object_unref (sess->send_rtp_src);
+ if (sess->send_rtcp_src != NULL)
+ gst_element_release_request_pad (sess->session, sess->send_rtcp_src);
+
+ gst_bin_remove (GST_BIN_CAST (bin), sess->session);
+ gst_bin_remove (GST_BIN_CAST (bin), sess->demux);
+
+ g_slist_foreach (sess->streams, (GFunc) free_stream, NULL);
+ g_slist_free (sess->streams);
+
+ g_mutex_free (sess->lock);
+ g_hash_table_destroy (sess->ptmap);
+
+ bin->sessions = g_slist_remove (bin->sessions, sess);
+
+ g_free (sess);
+}
+
#if 0
-static GstRTPBinStream *
-find_stream_by_ssrc (GstRTPBinSession * session, guint32 ssrc)
+static GstRtpBinStream *
+find_stream_by_ssrc (GstRtpBinSession * session, guint32 ssrc)
{
GSList *walk;
for (walk = session->streams; walk; walk = g_slist_next (walk)) {
- GstRTPBinStream *stream = (GstRTPBinStream *) walk->data;
+ GstRtpBinStream *stream = (GstRtpBinStream *) walk->data;
if (stream->ssrc == ssrc)
return stream;
}
#endif
-static GstRTPBinStream *
-create_stream (GstRTPBinSession * session, guint32 ssrc)
+/* get the payload type caps for the specific payload @pt in @session */
+static GstCaps *
+get_pt_map (GstRtpBinSession * session, guint pt)
+{
+ GstCaps *caps = NULL;
+ GstRtpBin *bin;
+ GValue ret = { 0 };
+ GValue args[3] = { {0}, {0}, {0} };
+
+ GST_DEBUG ("searching pt %d in cache", pt);
+
+ GST_RTP_SESSION_LOCK (session);
+
+ /* first look in the cache */
+ caps = g_hash_table_lookup (session->ptmap, GINT_TO_POINTER (pt));
+ if (caps) {
+ gst_caps_ref (caps);
+ goto done;
+ }
+
+ bin = session->bin;
+
+ GST_DEBUG ("emiting signal for pt %d in session %d", pt, session->id);
+
+ /* not in cache, send signal to request caps */
+ g_value_init (&args[0], GST_TYPE_ELEMENT);
+ g_value_set_object (&args[0], bin);
+ g_value_init (&args[1], G_TYPE_UINT);
+ g_value_set_uint (&args[1], session->id);
+ g_value_init (&args[2], G_TYPE_UINT);
+ g_value_set_uint (&args[2], pt);
+
+ g_value_init (&ret, GST_TYPE_CAPS);
+ g_value_set_boxed (&ret, NULL);
+
+ g_signal_emitv (args, gst_rtp_bin_signals[SIGNAL_REQUEST_PT_MAP], 0, &ret);
+
+ g_value_unset (&args[0]);
+ g_value_unset (&args[1]);
+ g_value_unset (&args[2]);
+ caps = (GstCaps *) g_value_dup_boxed (&ret);
+ g_value_unset (&ret);
+ if (!caps)
+ goto no_caps;
+
+ GST_DEBUG ("caching pt %d as %" GST_PTR_FORMAT, pt, caps);
+
+ /* store in cache, take additional ref */
+ g_hash_table_insert (session->ptmap, GINT_TO_POINTER (pt),
+ gst_caps_ref (caps));
+
+done:
+ GST_RTP_SESSION_UNLOCK (session);
+
+ return caps;
+
+ /* ERRORS */
+no_caps:
+ {
+ GST_RTP_SESSION_UNLOCK (session);
+ GST_DEBUG ("no pt map could be obtained");
+ return NULL;
+ }
+}
+
+static gboolean
+return_true (gpointer key, gpointer value, gpointer user_data)
+{
+ return TRUE;
+}
+
+static void
+gst_rtp_bin_clear_pt_map (GstRtpBin * bin)
+{
+ GSList *sessions, *streams;
+
+ GST_RTP_BIN_LOCK (bin);
+ GST_DEBUG_OBJECT (bin, "clearing pt map");
+ for (sessions = bin->sessions; sessions; sessions = g_slist_next (sessions)) {
+ GstRtpBinSession *session = (GstRtpBinSession *) sessions->data;
+
+ GST_DEBUG_OBJECT (bin, "clearing session %p", session);
+ g_signal_emit_by_name (session->session, "clear-pt-map", NULL);
+
+ GST_RTP_SESSION_LOCK (session);
+ g_hash_table_foreach_remove (session->ptmap, return_true, NULL);
+
+ for (streams = session->streams; streams; streams = g_slist_next (streams)) {
+ GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
+
+ GST_DEBUG_OBJECT (bin, "clearing stream %p", stream);
+ g_signal_emit_by_name (stream->buffer, "clear-pt-map", NULL);
+ g_signal_emit_by_name (stream->demux, "clear-pt-map", NULL);
+ }
+ GST_RTP_SESSION_UNLOCK (session);
+ }
+ GST_RTP_BIN_UNLOCK (bin);
+}
+
+static void
+gst_rtp_bin_propagate_property_to_jitterbuffer (GstRtpBin * bin,
+ const gchar * name, const GValue * value)
+{
+ GSList *sessions, *streams;
+
+ GST_RTP_BIN_LOCK (bin);
+ for (sessions = bin->sessions; sessions; sessions = g_slist_next (sessions)) {
+ GstRtpBinSession *session = (GstRtpBinSession *) sessions->data;
+
+ GST_RTP_SESSION_LOCK (session);
+ for (streams = session->streams; streams; streams = g_slist_next (streams)) {
+ GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
+
+ g_object_set_property (G_OBJECT (stream->buffer), name, value);
+ }
+ GST_RTP_SESSION_UNLOCK (session);
+ }
+ GST_RTP_BIN_UNLOCK (bin);
+}
+
+/* get a client with the given SDES name. Must be called with RTP_BIN_LOCK */
+static GstRtpBinClient *
+get_client (GstRtpBin * bin, guint8 len, guint8 * data, gboolean * created)
+{
+ GstRtpBinClient *result = NULL;
+ GSList *walk;
+
+ for (walk = bin->clients; walk; walk = g_slist_next (walk)) {
+ GstRtpBinClient *client = (GstRtpBinClient *) walk->data;
+
+ if (len != client->cname_len)
+ continue;
+
+ if (!strncmp ((gchar *) data, client->cname, client->cname_len)) {
+ GST_DEBUG_OBJECT (bin, "found existing client %p with CNAME %s", client,
+ client->cname);
+ result = client;
+ break;
+ }
+ }
+
+ /* nothing found, create one */
+ if (result == NULL) {
+ result = g_new0 (GstRtpBinClient, 1);
+ result->cname = g_strndup ((gchar *) data, len);
+ result->cname_len = len;
+ result->min_delta = G_MAXINT64;
+ bin->clients = g_slist_prepend (bin->clients, result);
+ GST_DEBUG_OBJECT (bin, "created new client %p with CNAME %s", result,
+ result->cname);
+ }
+ return result;
+}
+
+static void
+free_client (GstRtpBinClient * client)
+{
+ g_slist_free (client->streams);
+ g_free (client->cname);
+ g_free (client);
+}
+
+/* associate a stream to the given CNAME. This will make sure all streams for
+ * that CNAME are synchronized together. */
+static void
+gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream, guint8 len,
+ guint8 * data)
+{
+ GstRtpBinClient *client;
+ gboolean created;
+ GSList *walk;
+
+ /* first find or create the CNAME */
+ GST_RTP_BIN_LOCK (bin);
+ client = get_client (bin, len, data, &created);
+
+ /* find stream in the client */
+ for (walk = client->streams; walk; walk = g_slist_next (walk)) {
+ GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data;
+
+ if (ostream == stream)
+ break;
+ }
+ /* not found, add it to the list */
+ if (walk == NULL) {
+ GST_DEBUG_OBJECT (bin,
+ "new association of SSRC %08x with client %p with CNAME %s",
+ stream->ssrc, client, client->cname);
+ client->streams = g_slist_prepend (client->streams, stream);
+ client->nstreams++;
+ } else {
+ GST_DEBUG_OBJECT (bin,
+ "found association of SSRC %08x with client %p with CNAME %s",
+ stream->ssrc, client, client->cname);
+ }
+
+ /* we can only continue if we know the local clock-base and clock-rate */
+ if (stream->clock_base == -1)
+ goto no_clock_base;
+
+ if (stream->clock_rate <= 0) {
+ gint pt = -1;
+ GstCaps *caps = NULL;
+ GstStructure *s = NULL;
+
+ GST_RTP_SESSION_LOCK (stream->session);
+ pt = stream->last_pt;
+ GST_RTP_SESSION_UNLOCK (stream->session);
+
+ if (pt < 0)
+ goto no_clock_rate;
+
+ caps = get_pt_map (stream->session, pt);
+ if (!caps)
+ goto no_clock_rate;
+
+ s = gst_caps_get_structure (caps, 0);
+ gst_structure_get_int (s, "clock-rate", &stream->clock_rate);
+ gst_caps_unref (caps);
+
+ if (stream->clock_rate <= 0)
+ goto no_clock_rate;
+ }
+
+ /* map last RTP time to local timeline using our clock-base */
+ stream->local_rtp = stream->last_extrtptime - stream->clock_base;
+
+ GST_DEBUG_OBJECT (bin,
+ "base %" G_GUINT64_FORMAT ", extrtptime %" G_GUINT64_FORMAT
+ ", local RTP %" G_GUINT64_FORMAT ", clock-rate %d", stream->clock_base,
+ stream->last_extrtptime, stream->local_rtp, stream->clock_rate);
+
+ /* calculate local NTP time in gstreamer timestamp */
+ stream->local_unix =
+ gst_util_uint64_scale_int (stream->local_rtp, GST_SECOND,
+ stream->clock_rate);
+ stream->local_unix += stream->clock_base_time;
+ /* calculate delta between server and receiver */
+ stream->unix_delta = stream->last_unix - stream->local_unix;
+
+ GST_DEBUG_OBJECT (bin,
+ "local UNIX %" G_GUINT64_FORMAT ", remote UNIX %" G_GUINT64_FORMAT
+ ", delta %" G_GINT64_FORMAT, stream->local_unix, stream->last_unix,
+ stream->unix_delta);
+
+ /* recalc inter stream playout offset, but only if there are more than one
+ * stream. */
+ if (client->nstreams > 1) {
+ gint64 min;
+
+ /* calculate the min of all deltas */
+ min = G_MAXINT64;
+ for (walk = client->streams; walk; walk = g_slist_next (walk)) {
+ GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data;
+
+ if (ostream->unix_delta && ostream->unix_delta < min)
+ min = ostream->unix_delta;
+ }
+
+ GST_DEBUG_OBJECT (bin, "client %p min delta %" G_GINT64_FORMAT, client,
+ min);
+
+ /* calculate offsets for each stream */
+ for (walk = client->streams; walk; walk = g_slist_next (walk)) {
+ GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data;
+
+ if (ostream->unix_delta == 0)
+ continue;
+
+ ostream->ts_offset = ostream->unix_delta - min;
+
+ /* delta changed, see how much */
+ if (ostream->prev_ts_offset != ostream->ts_offset) {
+ gint64 diff;
+
+ if (ostream->prev_ts_offset > ostream->ts_offset)
+ diff = ostream->prev_ts_offset - ostream->ts_offset;
+ else
+ diff = ostream->ts_offset - ostream->prev_ts_offset;
+
+ /* only change diff when it changed more than 1 millisecond. This
+ * compensates for rounding errors in NTP to RTP timestamp
+ * conversions */
+ if (diff > GST_MSECOND)
+ g_object_set (ostream->buffer, "ts-offset", ostream->ts_offset, NULL);
+
+ ostream->prev_ts_offset = ostream->ts_offset;
+ }
+ GST_DEBUG_OBJECT (bin, "stream SSRC %08x, delta %" G_GINT64_FORMAT,
+ ostream->ssrc, ostream->ts_offset);
+ }
+ }
+ GST_RTP_BIN_UNLOCK (bin);
+ return;
+
+no_clock_base:
+ {
+ GST_WARNING_OBJECT (bin, "we have no clock-base");
+ GST_RTP_BIN_UNLOCK (bin);
+ return;
+ }
+no_clock_rate:
+ {
+ GST_WARNING_OBJECT (bin, "we have no clock-rate");
+ GST_RTP_BIN_UNLOCK (bin);
+ return;
+ }
+}
+
+#define GST_RTCP_BUFFER_FOR_PACKETS(b,buffer,packet) \
+ for ((b) = gst_rtcp_buffer_get_first_packet ((buffer), (packet)); (b); \
+ (b) = gst_rtcp_packet_move_to_next ((packet)))
+
+#define GST_RTCP_SDES_FOR_ITEMS(b,packet) \
+ for ((b) = gst_rtcp_packet_sdes_first_item ((packet)); (b); \
+ (b) = gst_rtcp_packet_sdes_next_item ((packet)))
+
+#define GST_RTCP_SDES_FOR_ENTRIES(b,packet) \
+ for ((b) = gst_rtcp_packet_sdes_first_entry ((packet)); (b); \
+ (b) = gst_rtcp_packet_sdes_next_entry ((packet)))
+
+static GstFlowReturn
+gst_rtp_bin_sync_chain (GstPad * pad, GstBuffer * buffer)
+{
+ GstFlowReturn ret = GST_FLOW_OK;
+ GstRtpBinStream *stream;
+ GstRtpBin *bin;
+ GstRTCPPacket packet;
+ guint32 ssrc;
+ guint64 ntptime;
+ guint32 rtptime;
+ gboolean have_sr, have_sdes;
+ gboolean more;
+
+ stream = gst_pad_get_element_private (pad);
+ bin = stream->bin;
+
+ GST_DEBUG_OBJECT (bin, "received sync packet");
+
+ if (!gst_rtcp_buffer_validate (buffer))
+ goto invalid_rtcp;
+
+ have_sr = FALSE;
+ have_sdes = FALSE;
+ GST_RTCP_BUFFER_FOR_PACKETS (more, buffer, &packet) {
+ /* first packet must be SR or RR or else the validate would have failed */
+ switch (gst_rtcp_packet_get_type (&packet)) {
+ case GST_RTCP_TYPE_SR:
+ /* only parse first. There is only supposed to be one SR in the packet
+ * but we will deal with malformed packets gracefully */
+ if (have_sr)
+ break;
+ /* get NTP and RTP times */
+ gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, &ntptime, &rtptime,
+ NULL, NULL);
+
+ GST_DEBUG_OBJECT (bin, "received sync packet from SSRC %08x", ssrc);
+ /* ignore SR that is not ours */
+ if (ssrc != stream->ssrc)
+ continue;
+
+ have_sr = TRUE;
+
+ /* store values in the stream */
+ stream->have_sync = TRUE;
+ stream->last_unix = gst_rtcp_ntp_to_unix (ntptime);
+ /* use extended timestamp */
+ gst_rtp_buffer_ext_timestamp (&stream->last_extrtptime, rtptime);
+ break;
+ case GST_RTCP_TYPE_SDES:
+ {
+ gboolean more_items, more_entries;
+
+ /* only deal with first SDES, there is only supposed to be one SDES in
+ * the RTCP packet but we deal with bad packets gracefully. Also bail
+ * out if we have not seen an SR item yet. */
+ if (have_sdes || !have_sr)
+ break;
+
+ GST_RTCP_SDES_FOR_ITEMS (more_items, &packet) {
+ /* skip items that are not about the SSRC of the sender */
+ if (gst_rtcp_packet_sdes_get_ssrc (&packet) != ssrc)
+ continue;
+
+ /* find the CNAME entry */
+ GST_RTCP_SDES_FOR_ENTRIES (more_entries, &packet) {
+ GstRTCPSDESType type;
+ guint8 len;
+ guint8 *data;
+
+ gst_rtcp_packet_sdes_get_entry (&packet, &type, &len, &data);
+
+ if (type == GST_RTCP_SDES_CNAME) {
+ stream->clock_base = GST_BUFFER_OFFSET (buffer);
+ stream->clock_base_time = GST_BUFFER_OFFSET_END (buffer);
+ /* associate the stream to CNAME */
+ gst_rtp_bin_associate (bin, stream, len, data);
+ }
+ }
+ }
+ have_sdes = TRUE;
+ break;
+ }
+ default:
+ /* we can ignore these packets */
+ break;
+ }
+ }
+
+ gst_buffer_unref (buffer);
+
+ return ret;
+
+ /* ERRORS */
+invalid_rtcp:
+ {
+ /* this is fatal and should be filtered earlier */
+ GST_ELEMENT_ERROR (bin, STREAM, DECODE, (NULL),
+ ("invalid RTCP packet received"));
+ gst_buffer_unref (buffer);
+ return GST_FLOW_ERROR;
+ }
+}
+
+/* create a new stream with @ssrc in @session. Must be called with
+ * RTP_SESSION_LOCK. */
+static GstRtpBinStream *
+create_stream (GstRtpBinSession * session, guint32 ssrc)
{
GstElement *buffer, *demux;
- GstRTPBinStream *stream;
+ GstRtpBinStream *stream;
+ GstPadTemplate *templ;
+ gchar *padname;
- if (!(buffer = gst_element_factory_make ("rtpjitterbuffer", NULL)))
+ if (!(buffer = gst_element_factory_make ("gstrtpjitterbuffer", NULL)))
goto no_jitterbuffer;
- if (!(demux = gst_element_factory_make ("rtpptdemux", NULL)))
+ if (!(demux = gst_element_factory_make ("gstrtpptdemux", NULL)))
goto no_demux;
- stream = g_new0 (GstRTPBinStream, 1);
+ stream = g_new0 (GstRtpBinStream, 1);
stream->ssrc = ssrc;
stream->bin = session->bin;
stream->session = session;
stream->buffer = buffer;
stream->demux = demux;
+ stream->last_extrtptime = -1;
+ stream->last_pt = -1;
+ stream->have_sync = FALSE;
session->streams = g_slist_prepend (session->streams, stream);
+ /* make an internal sinkpad for RTCP sync packets. Take ownership of the
+ * pad. We will link this pad later. */
+ padname = g_strdup_printf ("sync_%d", ssrc);
+ templ = gst_static_pad_template_get (&rtpbin_sync_sink_template);
+ stream->sync_pad = gst_pad_new_from_template (templ, padname);
+ gst_object_unref (templ);
+ g_free (padname);
+ gst_object_ref (stream->sync_pad);
+ gst_object_sink (stream->sync_pad);
+ gst_pad_set_element_private (stream->sync_pad, stream);
+ gst_pad_set_chain_function (stream->sync_pad, gst_rtp_bin_sync_chain);
+ gst_pad_set_active (stream->sync_pad, TRUE);
+
+ /* provide clock_rate to the jitterbuffer when needed */
+ g_signal_connect (buffer, "request-pt-map",
+ (GCallback) pt_map_requested, session);
+
+ /* configure latency and packet lost */
+ g_object_set (buffer, "latency", session->bin->latency, NULL);
+ g_object_set (buffer, "do-lost", session->bin->do_lost, NULL);
+
gst_bin_add (GST_BIN_CAST (session->bin), buffer);
gst_element_set_state (buffer, GST_STATE_PLAYING);
gst_bin_add (GST_BIN_CAST (session->bin), demux);
/* ERRORS */
no_jitterbuffer:
{
- g_warning ("rtpbin: could not create rtpjitterbuffer element");
+ g_warning ("gstrtpbin: could not create gstrtpjitterbuffer element");
return NULL;
}
no_demux:
{
gst_object_unref (buffer);
- g_warning ("rtpbin: could not create rtpptdemux element");
+ g_warning ("gstrtpbin: could not create gstrtpptdemux element");
return NULL;
}
}
-/* Manages the RTP streams that come from one client and should therefore be
- * synchronized.
- */
-struct _GstRTPBinClient
+static void
+free_stream (GstRtpBinStream * stream)
{
- /* the common CNAME for the streams */
- gchar *cname;
- /* the streams */
- GSList *streams;
-};
+ GstRtpBinSession *session;
+
+ session = stream->session;
+
+ gst_element_set_state (stream->buffer, GST_STATE_NULL);
+ gst_element_set_state (stream->demux, GST_STATE_NULL);
+
+ gst_bin_remove (GST_BIN_CAST (session->bin), stream->buffer);
+ gst_bin_remove (GST_BIN_CAST (session->bin), stream->demux);
+
+ gst_object_unref (stream->sync_pad);
+
+ session->streams = g_slist_remove (session->streams, stream);
+
+ g_free (stream);
+}
/* GObject vmethods */
+static void gst_rtp_bin_dispose (GObject * object);
static void gst_rtp_bin_finalize (GObject * object);
static void gst_rtp_bin_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static GstPad *gst_rtp_bin_request_new_pad (GstElement * element,
GstPadTemplate * templ, const gchar * name);
static void gst_rtp_bin_release_pad (GstElement * element, GstPad * pad);
+static void gst_rtp_bin_handle_message (GstBin * bin, GstMessage * message);
+static void gst_rtp_bin_clear_pt_map (GstRtpBin * bin);
-/*static guint gst_rtp_bin_signals[LAST_SIGNAL] = { 0 }; */
-
-GST_BOILERPLATE (GstRTPBin, gst_rtp_bin, GstBin, GST_TYPE_BIN);
+GST_BOILERPLATE (GstRtpBin, gst_rtp_bin, GstBin, GST_TYPE_BIN);
static void
gst_rtp_bin_base_init (gpointer klass)
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&rtpbin_recv_rtp_src_template));
gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&rtpbin_rtcp_src_template));
+ gst_static_pad_template_get (&rtpbin_send_rtcp_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&rtpbin_send_rtp_src_template));
}
static void
-gst_rtp_bin_class_init (GstRTPBinClass * klass)
+gst_rtp_bin_class_init (GstRtpBinClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
+ GstBinClass *gstbin_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
+ gstbin_class = (GstBinClass *) klass;
- g_type_class_add_private (klass, sizeof (GstRTPBinPrivate));
+ 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;
gobject_class->get_property = gst_rtp_bin_get_property;
+ g_object_class_install_property (gobject_class, PROP_LATENCY,
+ g_param_spec_uint ("latency", "Buffer latency in ms",
+ "Default amount of ms to buffer in the jitterbuffers", 0,
+ G_MAXUINT, DEFAULT_LATENCY_MS, G_PARAM_READWRITE));
+
+ /**
+ * GstRtpBin::request-pt-map:
+ * @rtpbin: the object which received the signal
+ * @session: the session
+ * @pt: the pt
+ *
+ * Request the payload type as #GstCaps for @pt in @session.
+ */
+ gst_rtp_bin_signals[SIGNAL_REQUEST_PT_MAP] =
+ g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, request_pt_map),
+ NULL, NULL, gst_rtp_bin_marshal_BOXED__UINT_UINT, GST_TYPE_CAPS, 2,
+ G_TYPE_UINT, G_TYPE_UINT);
+ /**
+ * GstRtpBin::clear-pt-map:
+ * @rtpbin: the object which received the signal
+ *
+ * Clear all previously cached pt-mapping obtained with
+ * GstRtpBin::request-pt-map.
+ */
+ gst_rtp_bin_signals[SIGNAL_CLEAR_PT_MAP] =
+ g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
+ clear_pt_map), NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE,
+ 0, G_TYPE_NONE);
+
+ /**
+ * GstRtpBin::on-new-ssrc:
+ * @rtpbin: the object which received the signal
+ * @session: the session
+ * @ssrc: the SSRC
+ *
+ * Notify of a new SSRC that entered @session.
+ */
+ gst_rtp_bin_signals[SIGNAL_ON_NEW_SSRC] =
+ g_signal_new ("on-new-ssrc", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_new_ssrc),
+ NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
+ G_TYPE_UINT, G_TYPE_UINT);
+ /**
+ * GstRtpBin::on-ssrc-collision:
+ * @rtpbin: the object which received the signal
+ * @session: the session
+ * @ssrc: the SSRC
+ *
+ * Notify when we have an SSRC collision
+ */
+ gst_rtp_bin_signals[SIGNAL_ON_SSRC_COLLISION] =
+ g_signal_new ("on-ssrc-collision", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_collision),
+ NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
+ G_TYPE_UINT, G_TYPE_UINT);
+ /**
+ * GstRtpBin::on-ssrc-validated:
+ * @rtpbin: the object which received the signal
+ * @session: the session
+ * @ssrc: the SSRC
+ *
+ * Notify of a new SSRC that became validated.
+ */
+ gst_rtp_bin_signals[SIGNAL_ON_SSRC_VALIDATED] =
+ g_signal_new ("on-ssrc-validated", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_validated),
+ NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
+ G_TYPE_UINT, G_TYPE_UINT);
+ /**
+ * GstRtpBin::on-ssrc-active:
+ * @rtpbin: the object which received the signal
+ * @session: the session
+ * @ssrc: the SSRC
+ *
+ * Notify of a SSRC that is active, i.e., sending RTCP.
+ */
+ gst_rtp_bin_signals[SIGNAL_ON_SSRC_ACTIVE] =
+ g_signal_new ("on-ssrc-active", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_active),
+ NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
+ G_TYPE_UINT, G_TYPE_UINT);
+ /**
+ * GstRtpBin::on-ssrc-sdes:
+ * @rtpbin: the object which received the signal
+ * @session: the session
+ * @ssrc: the SSRC
+ *
+ * Notify of a SSRC that is active, i.e., sending RTCP.
+ */
+ gst_rtp_bin_signals[SIGNAL_ON_SSRC_SDES] =
+ g_signal_new ("on-ssrc-sdes", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_sdes),
+ NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
+ G_TYPE_UINT, G_TYPE_UINT);
+
+ /**
+ * GstRtpBin::on-bye-ssrc:
+ * @rtpbin: the object which received the signal
+ * @session: the session
+ * @ssrc: the SSRC
+ *
+ * Notify of an SSRC that became inactive because of a BYE packet.
+ */
+ gst_rtp_bin_signals[SIGNAL_ON_BYE_SSRC] =
+ g_signal_new ("on-bye-ssrc", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_bye_ssrc),
+ NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
+ G_TYPE_UINT, G_TYPE_UINT);
+ /**
+ * GstRtpBin::on-bye-timeout:
+ * @rtpbin: the object which received the signal
+ * @session: the session
+ * @ssrc: the SSRC
+ *
+ * Notify of an SSRC that has timed out because of BYE
+ */
+ gst_rtp_bin_signals[SIGNAL_ON_BYE_TIMEOUT] =
+ g_signal_new ("on-bye-timeout", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_bye_timeout),
+ NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
+ G_TYPE_UINT, G_TYPE_UINT);
+ /**
+ * GstRtpBin::on-timeout:
+ * @rtpbin: the object which received the signal
+ * @session: the session
+ * @ssrc: the SSRC
+ *
+ * Notify of an SSRC that has timed out
+ */
+ gst_rtp_bin_signals[SIGNAL_ON_TIMEOUT] =
+ g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_timeout),
+ NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
+ G_TYPE_UINT, G_TYPE_UINT);
+
+ g_object_class_install_property (gobject_class, PROP_SDES_CNAME,
+ g_param_spec_string ("sdes-cname", "SDES CNAME",
+ "The CNAME to put in SDES messages of this session",
+ DEFAULT_SDES_CNAME, G_PARAM_READWRITE));
+
+ g_object_class_install_property (gobject_class, PROP_SDES_NAME,
+ g_param_spec_string ("sdes-name", "SDES NAME",
+ "The NAME to put in SDES messages of this session",
+ DEFAULT_SDES_NAME, G_PARAM_READWRITE));
+
+ g_object_class_install_property (gobject_class, PROP_SDES_EMAIL,
+ g_param_spec_string ("sdes-email", "SDES EMAIL",
+ "The EMAIL to put in SDES messages of this session",
+ DEFAULT_SDES_EMAIL, G_PARAM_READWRITE));
+
+ g_object_class_install_property (gobject_class, PROP_SDES_PHONE,
+ g_param_spec_string ("sdes-phone", "SDES PHONE",
+ "The PHONE to put in SDES messages of this session",
+ DEFAULT_SDES_PHONE, G_PARAM_READWRITE));
+
+ g_object_class_install_property (gobject_class, PROP_SDES_LOCATION,
+ g_param_spec_string ("sdes-location", "SDES LOCATION",
+ "The LOCATION to put in SDES messages of this session",
+ DEFAULT_SDES_LOCATION, G_PARAM_READWRITE));
+
+ g_object_class_install_property (gobject_class, PROP_SDES_TOOL,
+ g_param_spec_string ("sdes-tool", "SDES TOOL",
+ "The TOOL to put in SDES messages of this session",
+ DEFAULT_SDES_TOOL, G_PARAM_READWRITE));
+
+ g_object_class_install_property (gobject_class, PROP_SDES_NOTE,
+ g_param_spec_string ("sdes-note", "SDES NOTE",
+ "The NOTE to put in SDES messages of this session",
+ DEFAULT_SDES_NOTE, G_PARAM_READWRITE));
+
+ g_object_class_install_property (gobject_class, PROP_DO_LOST,
+ g_param_spec_boolean ("do-lost", "Do Lost",
+ "Send an event downstream when a packet is lost", DEFAULT_DO_LOST,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
gstelement_class->provide_clock =
GST_DEBUG_FUNCPTR (gst_rtp_bin_provide_clock);
gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_rtp_bin_change_state);
GST_DEBUG_FUNCPTR (gst_rtp_bin_request_new_pad);
gstelement_class->release_pad = GST_DEBUG_FUNCPTR (gst_rtp_bin_release_pad);
+ gstbin_class->handle_message = GST_DEBUG_FUNCPTR (gst_rtp_bin_handle_message);
+
+ klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_bin_clear_pt_map);
+
GST_DEBUG_CATEGORY_INIT (gst_rtp_bin_debug, "rtpbin", 0, "RTP bin");
}
static void
-gst_rtp_bin_init (GstRTPBin * rtpbin, GstRTPBinClass * klass)
+gst_rtp_bin_init (GstRtpBin * rtpbin, GstRtpBinClass * klass)
{
+ gchar *str;
+
rtpbin->priv = GST_RTP_BIN_GET_PRIVATE (rtpbin);
+ rtpbin->priv->bin_lock = g_mutex_new ();
rtpbin->provided_clock = gst_system_clock_obtain ();
+
+ rtpbin->latency = DEFAULT_LATENCY_MS;
+ rtpbin->do_lost = DEFAULT_DO_LOST;
+
+ /* some default SDES entries */
+ str = g_strdup_printf ("%s@%s", g_get_user_name (), g_get_host_name ());
+ gst_rtp_bin_set_sdes_string (rtpbin, GST_RTCP_SDES_CNAME, str);
+ g_free (str);
+
+ gst_rtp_bin_set_sdes_string (rtpbin, GST_RTCP_SDES_NAME, g_get_real_name ());
+ gst_rtp_bin_set_sdes_string (rtpbin, GST_RTCP_SDES_TOOL, "GStreamer");
+}
+
+static void
+gst_rtp_bin_dispose (GObject * object)
+{
+ GstRtpBin *rtpbin;
+
+ rtpbin = GST_RTP_BIN (object);
+
+ g_slist_foreach (rtpbin->sessions, (GFunc) free_session, NULL);
+ g_slist_free (rtpbin->sessions);
+ rtpbin->sessions = NULL;
+ g_slist_foreach (rtpbin->clients, (GFunc) free_client, NULL);
+ g_slist_free (rtpbin->clients);
+ rtpbin->clients = NULL;
+
+ G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_rtp_bin_finalize (GObject * object)
{
- GstRTPBin *rtpbin;
+ GstRtpBin *rtpbin;
+ gint i;
rtpbin = GST_RTP_BIN (object);
+ for (i = 0; i < 9; i++)
+ g_free (rtpbin->sdes[i]);
+
+ g_mutex_free (rtpbin->priv->bin_lock);
+ gst_object_unref (rtpbin->provided_clock);
+
G_OBJECT_CLASS (parent_class)->finalize (object);
}
+static const gchar *
+sdes_type_to_name (GstRTCPSDESType type)
+{
+ const gchar *result;
+
+ switch (type) {
+ case GST_RTCP_SDES_CNAME:
+ result = "sdes-cname";
+ break;
+ case GST_RTCP_SDES_NAME:
+ result = "sdes-name";
+ break;
+ case GST_RTCP_SDES_EMAIL:
+ result = "sdes-email";
+ break;
+ case GST_RTCP_SDES_PHONE:
+ result = "sdes-phone";
+ break;
+ case GST_RTCP_SDES_LOC:
+ result = "sdes-location";
+ break;
+ case GST_RTCP_SDES_TOOL:
+ result = "sdes-tool";
+ break;
+ case GST_RTCP_SDES_NOTE:
+ result = "sdes-note";
+ break;
+ case GST_RTCP_SDES_PRIV:
+ result = "sdes-priv";
+ break;
+ default:
+ result = NULL;
+ break;
+ }
+ return result;
+}
+
+static void
+gst_rtp_bin_set_sdes_string (GstRtpBin * bin, GstRTCPSDESType type,
+ const gchar * data)
+{
+ GSList *item;
+ const gchar *name;
+
+ if (type < 0 || type > 8)
+ return;
+
+ GST_OBJECT_LOCK (bin);
+ g_free (bin->sdes[type]);
+ bin->sdes[type] = g_strdup (data);
+ name = sdes_type_to_name (type);
+ /* store in all sessions */
+ for (item = bin->sessions; item; item = g_slist_next (item))
+ g_object_set (item->data, name, bin->sdes[type], NULL);
+ GST_OBJECT_UNLOCK (bin);
+}
+
+static gchar *
+gst_rtp_bin_get_sdes_string (GstRtpBin * bin, GstRTCPSDESType type)
+{
+ gchar *result;
+
+ if (type < 0 || type > 8)
+ return NULL;
+
+ GST_OBJECT_LOCK (bin);
+ result = g_strdup (bin->sdes[type]);
+ GST_OBJECT_UNLOCK (bin);
+
+ return result;
+}
+
static void
gst_rtp_bin_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
- GstRTPBin *rtpbin;
+ GstRtpBin *rtpbin;
rtpbin = GST_RTP_BIN (object);
switch (prop_id) {
+ case PROP_LATENCY:
+ GST_RTP_BIN_LOCK (rtpbin);
+ rtpbin->latency = g_value_get_uint (value);
+ GST_RTP_BIN_UNLOCK (rtpbin);
+ /* propegate the property down to the jitterbuffer */
+ gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin, "latency", value);
+ break;
+ case PROP_SDES_CNAME:
+ gst_rtp_bin_set_sdes_string (rtpbin, GST_RTCP_SDES_CNAME,
+ g_value_get_string (value));
+ break;
+ case PROP_SDES_NAME:
+ gst_rtp_bin_set_sdes_string (rtpbin, GST_RTCP_SDES_NAME,
+ g_value_get_string (value));
+ break;
+ case PROP_SDES_EMAIL:
+ gst_rtp_bin_set_sdes_string (rtpbin, GST_RTCP_SDES_EMAIL,
+ g_value_get_string (value));
+ break;
+ case PROP_SDES_PHONE:
+ gst_rtp_bin_set_sdes_string (rtpbin, GST_RTCP_SDES_PHONE,
+ g_value_get_string (value));
+ break;
+ case PROP_SDES_LOCATION:
+ gst_rtp_bin_set_sdes_string (rtpbin, GST_RTCP_SDES_LOC,
+ g_value_get_string (value));
+ break;
+ case PROP_SDES_TOOL:
+ gst_rtp_bin_set_sdes_string (rtpbin, GST_RTCP_SDES_TOOL,
+ g_value_get_string (value));
+ break;
+ case PROP_SDES_NOTE:
+ gst_rtp_bin_set_sdes_string (rtpbin, GST_RTCP_SDES_NOTE,
+ g_value_get_string (value));
+ break;
+ case PROP_DO_LOST:
+ GST_RTP_BIN_LOCK (rtpbin);
+ rtpbin->do_lost = g_value_get_boolean (value);
+ GST_RTP_BIN_UNLOCK (rtpbin);
+ gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin, "do-lost", value);
+ break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
gst_rtp_bin_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
- GstRTPBin *rtpbin;
+ GstRtpBin *rtpbin;
rtpbin = GST_RTP_BIN (object);
switch (prop_id) {
+ case PROP_LATENCY:
+ GST_RTP_BIN_LOCK (rtpbin);
+ g_value_set_uint (value, rtpbin->latency);
+ GST_RTP_BIN_UNLOCK (rtpbin);
+ break;
+ case PROP_SDES_CNAME:
+ g_value_take_string (value, gst_rtp_bin_get_sdes_string (rtpbin,
+ GST_RTCP_SDES_CNAME));
+ break;
+ case PROP_SDES_NAME:
+ g_value_take_string (value, gst_rtp_bin_get_sdes_string (rtpbin,
+ GST_RTCP_SDES_NAME));
+ break;
+ case PROP_SDES_EMAIL:
+ g_value_take_string (value, gst_rtp_bin_get_sdes_string (rtpbin,
+ GST_RTCP_SDES_EMAIL));
+ break;
+ case PROP_SDES_PHONE:
+ g_value_take_string (value, gst_rtp_bin_get_sdes_string (rtpbin,
+ GST_RTCP_SDES_PHONE));
+ break;
+ case PROP_SDES_LOCATION:
+ g_value_take_string (value, gst_rtp_bin_get_sdes_string (rtpbin,
+ GST_RTCP_SDES_LOC));
+ break;
+ case PROP_SDES_TOOL:
+ g_value_take_string (value, gst_rtp_bin_get_sdes_string (rtpbin,
+ GST_RTCP_SDES_TOOL));
+ break;
+ case PROP_SDES_NOTE:
+ g_value_take_string (value, gst_rtp_bin_get_sdes_string (rtpbin,
+ GST_RTCP_SDES_NOTE));
+ break;
+ case PROP_DO_LOST:
+ GST_RTP_BIN_LOCK (rtpbin);
+ g_value_set_boolean (value, rtpbin->do_lost);
+ GST_RTP_BIN_UNLOCK (rtpbin);
+ break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
static GstClock *
gst_rtp_bin_provide_clock (GstElement * element)
{
- GstRTPBin *rtpbin;
+ GstRtpBin *rtpbin;
rtpbin = GST_RTP_BIN (element);
return GST_CLOCK_CAST (gst_object_ref (rtpbin->provided_clock));
}
+static void
+gst_rtp_bin_handle_message (GstBin * bin, GstMessage * message)
+{
+ GstRtpBin *rtpbin;
+
+ rtpbin = GST_RTP_BIN (bin);
+
+ switch (GST_MESSAGE_TYPE (message)) {
+ case GST_MESSAGE_ELEMENT:
+ {
+ const GstStructure *s = gst_message_get_structure (message);
+
+ /* we change the structure name and add the session ID to it */
+ if (gst_structure_has_name (s, "GstRTPSessionSDES")) {
+ GSList *walk;
+
+ /* find the session, the message source has it */
+ for (walk = rtpbin->sessions; walk; walk = g_slist_next (walk)) {
+ GstRtpBinSession *sess = (GstRtpBinSession *) walk->data;
+
+ /* if we found the session, change message. else we exit the loop and
+ * leave the message unchanged */
+ if (GST_OBJECT_CAST (sess->session) == GST_MESSAGE_SRC (message)) {
+ message = gst_message_make_writable (message);
+ s = gst_message_get_structure (message);
+
+ gst_structure_set_name ((GstStructure *) s, "GstRTPBinSDES");
+
+ gst_structure_set ((GstStructure *) s, "session", G_TYPE_UINT,
+ sess->id, NULL);
+ break;
+ }
+ }
+ }
+ /* fallthrough to forward the modified message to the parent */
+ }
+ default:
+ {
+ GST_BIN_CLASS (parent_class)->handle_message (bin, message);
+ break;
+ }
+ }
+}
+
+static void
+calc_ntp_ns_base (GstRtpBin * bin)
+{
+ GstClockTime now;
+ GTimeVal current;
+ GSList *walk;
+
+ /* get the current time and convert it to NTP time in nanoseconds */
+ g_get_current_time (¤t);
+ now = GST_TIMEVAL_TO_TIME (current);
+ now += (2208988800LL * GST_SECOND);
+
+ GST_RTP_BIN_LOCK (bin);
+ bin->priv->ntp_ns_base = now;
+ for (walk = bin->sessions; walk; walk = g_slist_next (walk)) {
+ GstRtpBinSession *session = (GstRtpBinSession *) walk->data;
+
+ g_object_set (session->session, "ntp-ns-base", now, NULL);
+ }
+ GST_RTP_BIN_UNLOCK (bin);
+
+ return;
+}
+
static GstStateChangeReturn
gst_rtp_bin_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn res;
- GstRTPBin *rtpbin;
+ GstRtpBin *rtpbin;
rtpbin = GST_RTP_BIN (element);
case GST_STATE_CHANGE_READY_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
+ calc_ntp_ns_base (rtpbin);
break;
default:
break;
/* a new pad (SSRC) was created in @session */
static void
new_payload_found (GstElement * element, guint pt, GstPad * pad,
- GstRTPBinStream * stream)
+ GstRtpBinStream * stream)
{
- GstRTPBin *rtpbin;
+ GstRtpBin *rtpbin;
GstElementClass *klass;
GstPadTemplate *templ;
gchar *padname;
gpad = gst_ghost_pad_new_from_template (padname, pad, templ);
g_free (padname);
+ gst_pad_set_caps (gpad, GST_PAD_CAPS (pad));
gst_pad_set_active (gpad, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), gpad);
}
+static GstCaps *
+pt_map_requested (GstElement * element, guint pt, GstRtpBinSession * session)
+{
+ GstRtpBin *rtpbin;
+ GstCaps *caps;
+
+ rtpbin = session->bin;
+
+ GST_DEBUG_OBJECT (rtpbin, "payload map requested for pt %d in session %d", pt,
+ session->id);
+
+ caps = get_pt_map (session, pt);
+ if (!caps)
+ goto no_caps;
+
+ return caps;
+
+ /* ERRORS */
+no_caps:
+ {
+ GST_DEBUG_OBJECT (rtpbin, "could not get caps");
+ return NULL;
+ }
+}
+
+/* emited when caps changed for the session */
+static void
+caps_changed (GstPad * pad, GParamSpec * pspec, GstRtpBinSession * session)
+{
+ GstRtpBin *bin;
+ GstCaps *caps;
+ gint payload;
+ const GstStructure *s;
+
+ bin = session->bin;
+
+ g_object_get (pad, "caps", &caps, NULL);
+
+ if (caps == NULL)
+ return;
+
+ GST_DEBUG_OBJECT (bin, "got caps %" GST_PTR_FORMAT, caps);
+
+ s = gst_caps_get_structure (caps, 0);
+
+ /* get payload, finish when it's not there */
+ if (!gst_structure_get_int (s, "payload", &payload))
+ return;
+
+ GST_RTP_SESSION_LOCK (session);
+ GST_DEBUG_OBJECT (bin, "insert caps for payload %d", payload);
+ g_hash_table_insert (session->ptmap, GINT_TO_POINTER (payload), caps);
+ GST_RTP_SESSION_UNLOCK (session);
+}
+
+/* Stores the last payload type received on a particular stream */
+static void
+payload_type_change (GstElement * element, guint pt, GstRtpBinStream * stream)
+{
+ GST_RTP_SESSION_LOCK (stream->session);
+ stream->last_pt = pt;
+ GST_RTP_SESSION_UNLOCK (stream->session);
+}
+
/* a new pad (SSRC) was created in @session */
static void
new_ssrc_pad_found (GstElement * element, guint ssrc, GstPad * pad,
- GstRTPBinSession * session)
+ GstRtpBinSession * session)
{
- GstRTPBinStream *stream;
- GstPad *sinkpad;
+ GstRtpBinStream *stream;
+ GstPad *sinkpad, *srcpad;
+ gchar *padname;
+ GstCaps *caps;
GST_DEBUG_OBJECT (session->bin, "new SSRC pad %08x", ssrc);
+ GST_RTP_SESSION_LOCK (session);
+
/* create new stream */
stream = create_stream (session, ssrc);
if (!stream)
goto no_stream;
+ /* get the caps of the pad, we need the clock-rate and base_time if any. */
+ if ((caps = gst_pad_get_caps (pad))) {
+ const GstStructure *s;
+ guint val;
+
+ GST_DEBUG_OBJECT (session->bin, "pad has caps %" GST_PTR_FORMAT, caps);
+
+ s = gst_caps_get_structure (caps, 0);
+
+ if (!gst_structure_get_int (s, "clock-rate", &stream->clock_rate)) {
+ stream->clock_rate = -1;
+
+ GST_WARNING_OBJECT (session->bin,
+ "Caps have no clock rate %s from pad %s:%s",
+ gst_caps_to_string (caps), GST_DEBUG_PAD_NAME (pad));
+ }
+
+ if (gst_structure_get_uint (s, "clock-base", &val))
+ stream->clock_base = val;
+ else
+ stream->clock_base = -1;
+
+ gst_caps_unref (caps);
+ }
+
/* get pad and link */
GST_DEBUG_OBJECT (session->bin, "linking jitterbuffer");
+ padname = g_strdup_printf ("src_%d", ssrc);
+ srcpad = gst_element_get_static_pad (element, padname);
+ g_free (padname);
sinkpad = gst_element_get_static_pad (stream->buffer, "sink");
- gst_pad_link (pad, sinkpad);
+ gst_pad_link (srcpad, sinkpad);
gst_object_unref (sinkpad);
+ gst_object_unref (srcpad);
+
+ /* get the RTCP sync pad */
+ GST_DEBUG_OBJECT (session->bin, "linking sync pad");
+ padname = g_strdup_printf ("rtcp_src_%d", ssrc);
+ srcpad = gst_element_get_static_pad (element, padname);
+ g_free (padname);
+ gst_pad_link (srcpad, stream->sync_pad);
+ gst_object_unref (srcpad);
- /* connect to the new-pad signal of the payload demuxer */
+ /* connect to the new-pad signal of the payload demuxer, this will expose the
+ * new pad by ghosting it. */
stream->demux_newpad_sig = g_signal_connect (stream->demux,
"new-payload-type", (GCallback) new_payload_found, stream);
+ /* connect to the request-pt-map signal. This signal will be emited 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);
+ /* connect to the payload-type-change signal so that we can know which
+ * PT is the current PT so that the jitterbuffer can be matched to the right
+ * offset. */
+ stream->demux_pt_change_sig = g_signal_connect (stream->demux,
+ "payload-type-change", (GCallback) payload_type_change, stream);
+
+ GST_RTP_SESSION_UNLOCK (session);
return;
/* ERRORS */
no_stream:
{
- GST_DEBUG ("could not create stream");
+ GST_RTP_SESSION_UNLOCK (session);
+ GST_DEBUG_OBJECT (session->bin, "could not create stream");
return;
}
}
-/* Create a pad for receiving RTP for the session in @name
+/* 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)
+create_recv_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
{
GstPad *result, *sinkdpad;
guint sessid;
- GstRTPBinSession *session;
+ GstRtpBinSession *session;
GstPadLinkReturn lres;
/* first get the session number */
if (session == NULL)
goto create_error;
}
+
/* check if pad was requested */
if (session->recv_rtp_sink != NULL)
goto existed;
if (session->recv_rtp_sink == NULL)
goto pad_failed;
+ g_signal_connect (session->recv_rtp_sink, "notify::caps",
+ (GCallback) caps_changed, session);
+
GST_DEBUG_OBJECT (rtpbin, "getting RTP src pad");
/* get srcpad, link to SSRCDemux */
session->recv_rtp_src =
if (session->recv_rtp_src == NULL)
goto pad_failed;
- GST_DEBUG_OBJECT (rtpbin, "getting demuxer sink pad");
+ GST_DEBUG_OBJECT (rtpbin, "getting demuxer RTP sink pad");
sinkdpad = gst_element_get_static_pad (session->demux, "sink");
+ GST_DEBUG_OBJECT (rtpbin, "linking demuxer RTP sink pad");
lres = gst_pad_link (session->recv_rtp_src, sinkdpad);
gst_object_unref (sinkdpad);
if (lres != GST_PAD_LINK_OK)
goto link_failed;
- /* connect to the new-ssrc-pad signal of the demuxer */
+ /* connect to the new-ssrc-pad signal of the SSRC demuxer */
session->demux_newpad_sig = g_signal_connect (session->demux,
"new-ssrc-pad", (GCallback) new_ssrc_pad_found, session);
/* ERRORS */
no_name:
{
- g_warning ("rtpbin: invalid name given");
+ g_warning ("gstrtpbin: invalid name given");
return NULL;
}
create_error:
}
existed:
{
- g_warning ("rtpbin: recv_rtp pad already requested for session %d", sessid);
+ g_warning ("gstrtpbin: recv_rtp pad already requested for session %d",
+ sessid);
return NULL;
}
pad_failed:
{
- g_warning ("rtpbin: failed to get session pad");
+ g_warning ("gstrtpbin: failed to get session pad");
return NULL;
}
link_failed:
{
- g_warning ("rtpbin: failed to link pads");
+ g_warning ("gstrtpbin: failed to link pads");
return NULL;
}
}
-/* Create a pad for receiving RTCP for the session in @name
+/* 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,
+create_recv_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ,
const gchar * name)
{
GstPad *result;
guint sessid;
- GstRTPBinSession *session;
-
-#if 0
+ GstRtpBinSession *session;
GstPad *sinkdpad;
GstPadLinkReturn lres;
-#endif
/* first get the session number */
if (name == NULL || sscanf (name, "recv_rtcp_sink_%d", &sessid) != 1)
GST_DEBUG_OBJECT (rtpbin, "finding session %d", sessid);
- /* get the session, it must exist or we error */
+ /* get or create the session */
session = find_session_by_id (rtpbin, sessid);
- if (!session)
- goto no_session;
+ 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 != NULL)
goto existed;
- GST_DEBUG_OBJECT (rtpbin, "getting RTCP sink pad");
-
/* get recv_rtp pad and store */
+ GST_DEBUG_OBJECT (rtpbin, "getting RTCP sink pad");
session->recv_rtcp_sink =
gst_element_get_request_pad (session->session, "recv_rtcp_sink");
if (session->recv_rtcp_sink == NULL)
goto pad_failed;
-#if 0
/* get srcpad, link to SSRCDemux */
GST_DEBUG_OBJECT (rtpbin, "getting sync src pad");
- session->recv_rtcp_src =
- gst_element_get_static_pad (session->session, "sync_src");
- if (session->recv_rtcp_src == NULL)
+ session->sync_src = gst_element_get_static_pad (session->session, "sync_src");
+ if (session->sync_src == NULL)
goto pad_failed;
- GST_DEBUG_OBJECT (rtpbin, "linking sync to demux");
- sinkdpad = gst_element_get_static_pad (session->demux, "sink");
- lres = gst_pad_link (session->recv_rtcp_src, sinkdpad);
+ GST_DEBUG_OBJECT (rtpbin, "getting demuxer RTCP sink pad");
+ sinkdpad = gst_element_get_static_pad (session->demux, "rtcp_sink");
+ lres = gst_pad_link (session->sync_src, sinkdpad);
gst_object_unref (sinkdpad);
if (lres != GST_PAD_LINK_OK)
goto link_failed;
-#endif
result =
gst_ghost_pad_new_from_template (name, session->recv_rtcp_sink, templ);
/* ERRORS */
no_name:
{
- g_warning ("rtpbin: invalid name given");
+ g_warning ("gstrtpbin: invalid name given");
return NULL;
}
-no_session:
+create_error:
{
- g_warning ("rtpbin: no session with id %d", sessid);
+ /* create_session already warned */
return NULL;
}
existed:
{
- g_warning ("rtpbin: recv_rtcp pad already requested for session %d",
+ g_warning ("gstrtpbin: recv_rtcp pad already requested for session %d",
sessid);
return NULL;
}
pad_failed:
{
- g_warning ("rtpbin: failed to get session pad");
+ g_warning ("gstrtpbin: failed to get session pad");
return NULL;
}
-#if 0
link_failed:
{
- g_warning ("rtpbin: failed to link pads");
+ g_warning ("gstrtpbin: failed to link pads");
return NULL;
}
-#endif
}
-/* Create a pad for sending RTP for the session in @name
+/* Create a pad for sending RTP for the session in @name. Must be called with
+ * RTP_BIN_LOCK.
*/
static GstPad *
-create_send_rtp (GstRTPBin * rtpbin, GstPadTemplate * templ, const gchar * name)
+create_send_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
{
- GstPad *result, *srcpad, *srcghost;
+ GstPad *result, *srcghost;
gchar *gname;
guint sessid;
- GstRTPBinSession *session;
+ GstRtpBinSession *session;
GstElementClass *klass;
/* first get the session number */
if (session->send_rtp_sink != NULL)
goto existed;
- /* get recv_rtp pad and store */
+ /* get send_rtp pad and store */
session->send_rtp_sink =
gst_element_get_request_pad (session->session, "send_rtp_sink");
if (session->send_rtp_sink == NULL)
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), result);
/* get srcpad */
- srcpad = gst_element_get_pad (session->session, "send_rtp_src");
- if (srcpad == NULL)
+ session->send_rtp_src =
+ gst_element_get_static_pad (session->session, "send_rtp_src");
+ if (session->send_rtp_src == NULL)
goto no_srcpad;
/* ghost the new source pad */
gname = g_strdup_printf ("send_rtp_src_%d", sessid);
templ = gst_element_class_get_pad_template (klass, "send_rtp_src_%d");
srcghost =
- gst_ghost_pad_new_from_template (gname, session->send_rtp_sink, templ);
+ gst_ghost_pad_new_from_template (gname, session->send_rtp_src, templ);
gst_pad_set_active (srcghost, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), srcghost);
g_free (gname);
/* ERRORS */
no_name:
{
- g_warning ("rtpbin: invalid name given");
+ g_warning ("gstrtpbin: invalid name given");
return NULL;
}
create_error:
}
existed:
{
- g_warning ("rtpbin: send_rtp pad already requested for session %d", sessid);
+ g_warning ("gstrtpbin: send_rtp pad already requested for session %d",
+ sessid);
return NULL;
}
pad_failed:
{
- g_warning ("rtpbin: failed to get session pad for session %d", sessid);
+ g_warning ("gstrtpbin: failed to get session pad for session %d", sessid);
return NULL;
}
no_srcpad:
{
- g_warning ("rtpbin: failed to get rtp source pad for session %d", sessid);
+ g_warning ("gstrtpbin: failed to get rtp source pad for session %d",
+ sessid);
return NULL;
}
}
-/* Create a pad for sending RTCP for the session in @name
+/* Create a pad for sending RTCP for the session in @name. Must be called with
+ * RTP_BIN_LOCK.
*/
static GstPad *
-create_rtcp (GstRTPBin * rtpbin, GstPadTemplate * templ, const gchar * name)
+create_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
{
GstPad *result;
guint sessid;
- GstRTPBinSession *session;
+ GstRtpBinSession *session;
/* first get the session number */
- if (name == NULL || sscanf (name, "rtcp_src_%d", &sessid) != 1)
+ if (name == NULL || sscanf (name, "send_rtcp_src_%d", &sessid) != 1)
goto no_name;
/* get or create session */
goto no_session;
/* check if pad was requested */
- if (session->rtcp_src != NULL)
+ if (session->send_rtcp_src != NULL)
goto existed;
/* get rtcp_src pad and store */
- session->rtcp_src =
- gst_element_get_request_pad (session->session, "rtcp_src");
- if (session->rtcp_src == NULL)
+ session->send_rtcp_src =
+ gst_element_get_request_pad (session->session, "send_rtcp_src");
+ if (session->send_rtcp_src == NULL)
goto pad_failed;
- result = gst_ghost_pad_new_from_template (name, session->rtcp_src, templ);
+ result =
+ gst_ghost_pad_new_from_template (name, session->send_rtcp_src, templ);
gst_pad_set_active (result, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), result);
/* ERRORS */
no_name:
{
- g_warning ("rtpbin: invalid name given");
+ g_warning ("gstrtpbin: invalid name given");
return NULL;
}
no_session:
{
- g_warning ("rtpbin: session with id %d does not exist", sessid);
+ g_warning ("gstrtpbin: session with id %d does not exist", sessid);
return NULL;
}
existed:
{
- g_warning ("rtpbin: rtcp_src pad already requested for session %d", sessid);
+ g_warning ("gstrtpbin: send_rtcp_src pad already requested for session %d",
+ sessid);
return NULL;
}
pad_failed:
{
- g_warning ("rtpbin: failed to get rtcp pad for session %d", sessid);
+ g_warning ("gstrtpbin: failed to get rtcp pad for session %d", sessid);
return NULL;
}
}
+/* If the requested name is NULL we should create a name with
+ * the session number assuming we want the lowest posible session
+ * with a free pad like the template */
+static gchar *
+gst_rtp_bin_get_free_pad_name (GstElement * element, GstPadTemplate * templ)
+{
+ gboolean name_found = FALSE;
+ gint session = 0;
+ GstPad *pad = NULL;
+ GstIterator *pad_it = NULL;
+ gchar *pad_name = NULL;
+
+ GST_DEBUG_OBJECT (element, "find a free pad name for template");
+ while (!name_found) {
+ g_free (pad_name);
+ pad_name = g_strdup_printf (templ->name_template, session++);
+ pad_it = gst_element_iterate_pads (GST_ELEMENT (element));
+ name_found = TRUE;
+ while (gst_iterator_next (pad_it, (gpointer) & pad) == GST_ITERATOR_OK) {
+ gchar *name;
+
+ name = gst_pad_get_name (pad);
+ if (strcmp (name, pad_name) == 0)
+ name_found = FALSE;
+ g_free (name);
+ }
+ gst_iterator_free (pad_it);
+ }
+
+ GST_DEBUG_OBJECT (element, "free pad name found: '%s'", pad_name);
+ return pad_name;
+}
+
/*
*/
static GstPad *
gst_rtp_bin_request_new_pad (GstElement * element,
GstPadTemplate * templ, const gchar * name)
{
- GstRTPBin *rtpbin;
+ GstRtpBin *rtpbin;
GstElementClass *klass;
GstPad *result;
+ gchar *pad_name = NULL;
g_return_val_if_fail (templ != NULL, NULL);
g_return_val_if_fail (GST_IS_RTP_BIN (element), NULL);
rtpbin = GST_RTP_BIN (element);
klass = GST_ELEMENT_GET_CLASS (element);
+ GST_RTP_BIN_LOCK (rtpbin);
+
+ if (name == NULL) {
+ /* use a free pad name */
+ pad_name = gst_rtp_bin_get_free_pad_name (element, templ);
+ } else {
+ /* use the provided name */
+ pad_name = g_strdup (name);
+ }
+
+ GST_DEBUG ("Trying to request a pad with name %s", pad_name);
+
/* figure out the template */
if (templ == gst_element_class_get_pad_template (klass, "recv_rtp_sink_%d")) {
- result = create_recv_rtp (rtpbin, templ, name);
+ result = create_recv_rtp (rtpbin, templ, pad_name);
} else if (templ == gst_element_class_get_pad_template (klass,
"recv_rtcp_sink_%d")) {
- result = create_recv_rtcp (rtpbin, templ, name);
+ result = create_recv_rtcp (rtpbin, templ, pad_name);
} else if (templ == gst_element_class_get_pad_template (klass,
"send_rtp_sink_%d")) {
- result = create_send_rtp (rtpbin, templ, name);
- } else if (templ == gst_element_class_get_pad_template (klass, "rtcp_src_%d")) {
- result = create_rtcp (rtpbin, templ, name);
+ result = create_send_rtp (rtpbin, templ, pad_name);
+ } else if (templ == gst_element_class_get_pad_template (klass,
+ "send_rtcp_src_%d")) {
+ result = create_rtcp (rtpbin, templ, pad_name);
} else
goto wrong_template;
+ g_free (pad_name);
+ GST_RTP_BIN_UNLOCK (rtpbin);
+
return result;
/* ERRORS */
wrong_template:
{
- g_warning ("rtpbin: this is not our template");
+ g_free (pad_name);
+ GST_RTP_BIN_UNLOCK (rtpbin);
+ g_warning ("gstrtpbin: this is not our template");
return NULL;
}
}
projects / platform / upstream / gstreamer.git / blobdiff