mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-22 08:17:01 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/gst/rtpmanager/gstrtpbin.c
Wim Taymans 5e98fa572f gst/rtpmanager/gstrtpbin.c: Do not try to adjust the offset of streams for which we have not yet seen an SR packet. A... 
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_session),
(gst_rtp_bin_associate), (gst_rtp_bin_sync_chain):
Do not try to adjust the offset of streams for which we have not yet
seen an SR packet. Avoids large ts-offsets in some cases.
2009-08-11 02:30:37 +01:00

2443 lines
74 KiB
C
Raw Blame History

/* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:element-gstrtpbin
* @see_also: gstrtpjitterbuffer, gstrtpsession, gstrtpptdemux, gstrtpssrcdemux
*
* 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.
*
* #GstRtpBin is configured with a number of request pads that define the
* functionality that is activated, similar to the #GstRtpSession element.
*
* 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 #GstRtpsSrcDemux 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
* #GstRtpsSrcDemux element. The #GstRtpsSrcDemux 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.
*
* 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.
*
* 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.
*
* 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.
*
* 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.
*
* <refsect2>
* <title>Example pipelines</title>
* |[
* gst-launch udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink_0 \
* gstrtpbin ! rtptheoradepay ! theoradec ! xvimagesink
* ]| Receive RTP data from port 5000 and send to the session 0 in gstrtpbin.
* |[
* 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
* ]| 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
* |[
* 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
* ]| 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.
* </refsect2>
*
* Last reviewed on 2007-08-30 (0.10.6)
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/rtp/gstrtcpbuffer.h>
#include "gstrtpbin-marshal.h"
#include "gstrtpbin.h"
#include "gstrtpsession.h"
#include "gstrtpjitterbuffer.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/Network/RTP",
"Implement an RTP bin",
"Wim Taymans <wim.taymans@gmail.com>");
/* sink pads */
static GstStaticPadTemplate rtpbin_recv_rtp_sink_template =
GST_STATIC_PAD_TEMPLATE ("recv_rtp_sink_%d",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("application/x-rtp")
);
static GstStaticPadTemplate rtpbin_recv_rtcp_sink_template =
GST_STATIC_PAD_TEMPLATE ("recv_rtcp_sink_%d",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("application/x-rtcp")
);
static GstStaticPadTemplate rtpbin_send_rtp_sink_template =
GST_STATIC_PAD_TEMPLATE ("send_rtp_sink_%d",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("application/x-rtp")
);
/* src pads */
static GstStaticPadTemplate rtpbin_recv_rtp_src_template =
GST_STATIC_PAD_TEMPLATE ("recv_rtp_src_%d_%d_%d",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS ("application/x-rtp")
);
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")
);
static GstStaticPadTemplate rtpbin_send_rtp_src_template =
GST_STATIC_PAD_TEMPLATE ("send_rtp_src_%d",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
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))
#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)
/* lock to protect dynamic callbacks, like pad-added and new ssrc. */
#define GST_RTP_BIN_DYN_LOCK(bin) g_mutex_lock ((bin)->priv->dyn_lock)
#define GST_RTP_BIN_DYN_UNLOCK(bin) g_mutex_unlock ((bin)->priv->dyn_lock)
/* lock for shutdown */
#define GST_RTP_BIN_SHUTDOWN_LOCK(bin,label) \
G_STMT_START { \
if (g_atomic_int_get (&bin->priv->shutdown)) \
goto label; \
GST_RTP_BIN_DYN_LOCK (bin); \
if (g_atomic_int_get (&bin->priv->shutdown)) { \
GST_RTP_BIN_DYN_UNLOCK (bin); \
goto label; \
} \
} G_STMT_END
/* unlock for shutdown */
#define GST_RTP_BIN_SHUTDOWN_UNLOCK(bin) \
GST_RTP_BIN_DYN_UNLOCK (bin); \
struct _GstRtpBinPrivate
{
GMutex *bin_lock;
/* lock protecting dynamic adding/removing */
GMutex *dyn_lock;
/* the time when we went to playing */
GstClockTime ntp_ns_base;
/* if we are shutting down or not */
gint shutdown;
};
/* signals and args */
enum
{
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,
SIGNAL_ON_SENDER_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_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;
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
* together (see below).
*/
struct _GstRtpBinStream
{
/* the SSRC of this stream */
guint32 ssrc;
/* parent bin */
GstRtpBin *bin;
/* the session this SSRC belongs to */
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 last_clock_base;
guint64 clock_base;
guint64 clock_base_time;
gint clock_rate;
gint64 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).
*/
struct _GstRtpBinSession
{
/* session id */
gint id;
/* the parent bin */
GstRtpBin *bin;
/* the session element */
GstElement *session;
/* the SSRC demuxer */
GstElement *demux;
gulong demux_newpad_sig;
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 *sync_src;
GstPad *send_rtp_sink;
GstPad *send_rtp_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. 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;
if (sess->id == id)
return sess;
}
return NULL;
}
static void
on_new_ssrc (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
{
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);
}
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);
}
static void
on_sender_timeout (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
{
g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SENDER_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 ("gstrtpssrcdemux", NULL)))
goto no_demux;
sess = g_new0 (GstRtpBinSession, 1);
sess->lock = g_mutex_new ();
sess->id = id;
sess->bin = rtpbin;
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);
/* 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);
g_signal_connect (sess->session, "on-sender-timeout",
(GCallback) on_sender_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);
return sess;
/* ERRORS */
no_session:
{
g_warning ("gstrtpbin: could not create gstrtpsession element");
return NULL;
}
no_demux:
{
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)
{
GSList *walk;
for (walk = session->streams; walk; walk = g_slist_next (walk)) {
GstRtpBinStream *stream = (GstRtpBinStream *) walk->data;
if (stream->ssrc == ssrc)
return stream;
}
return NULL;
}
#endif
/* 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);
GST_RTP_SESSION_UNLOCK (session);
g_signal_emitv (args, gst_rtp_bin_signals[SIGNAL_REQUEST_PT_MAP], 0, &ret);
GST_RTP_SESSION_LOCK (session);
g_value_unset (&args[0]);
g_value_unset (&args[1]);
g_value_unset (&args[2]);
/* look in the cache again because we let the lock go */
caps = g_hash_table_lookup (session->ptmap, GINT_TO_POINTER (pt));
if (caps) {
gst_caps_ref (caps);
g_value_unset (&ret);
goto done;
}
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;
}
/* take the extended rtptime we found in the SR packet and map it to the
* local rtptime. The local rtp time is used to construct timestamps on the
* buffers. */
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, we essentially perform the
* same conversion that a jitterbuffer would use to convert an rtp timestamp
* into a corresponding 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. last_unix is created by
* converting the ntptime in the last SR packet to a gstreamer timestamp. This
* delta expresses the difference to our timeline and the server timeline. */
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 is more than one
* stream. */
if (client->nstreams > 1) {
gint64 min;
/* calculate the min of all deltas, ignoring streams that did not yet have a
* valid unix_delta because we did not yet receive an SR packet for those
* streams.
* We calculate the mininum because we would like to only apply positive
* offsets to streams, delaying their playback instead of trying to speed up
* other streams (which might be imposible when we have to create negative
* latencies).
* The stream that has the smalest diff is selected as the reference stream,
* all other streams will have a positive offset to this difference. */
min = G_MAXINT64;
for (walk = client->streams; walk; walk = g_slist_next (walk)) {
GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data;
if (!ostream->have_sync)
continue;
if (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;
gint64 prev_ts_offset;
/* ignore streams for which we didn't receive an SR packet yet, we
* can't synchronize them yet. We can however sync other streams just
* fine. */
if (!ostream->have_sync)
continue;
/* calculate offset to our reference stream, this should always give a
* positive number. */
ostream->ts_offset = ostream->unix_delta - min;
g_object_get (ostream->buffer, "ts-offset", &prev_ts_offset, NULL);
/* delta changed, see how much */
if (prev_ts_offset != ostream->ts_offset) {
gint64 diff;
if (prev_ts_offset > ostream->ts_offset)
diff = prev_ts_offset - ostream->ts_offset;
else
diff = ostream->ts_offset - prev_ts_offset;
GST_DEBUG_OBJECT (bin,
"ts-offset %" G_GUINT64_FORMAT ", prev %" G_GUINT64_FORMAT
", diff: %" G_GINT64_FORMAT, ostream->ts_offset, prev_ts_offset,
diff);
/* only change diff when it changed more than 4 milliseconds. This
* compensates for rounding errors in NTP to RTP timestamp
* conversions */
if (diff > 4 * GST_MSECOND && diff < (3 * GST_SECOND)) {
g_object_set (ostream->buffer, "ts-offset", ostream->ts_offset, NULL);
}
}
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;
guint64 clock_base;
guint64 clock_base_time;
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;
/* get the last relation between the rtp timestamps and the gstreamer
* timestamps. We get this info directly from the jitterbuffer which
* constructs gstreamer timestamps from rtp timestamps and so it know exactly
* what the current situation is. */
gst_rtp_jitter_buffer_get_sync (GST_RTP_JITTER_BUFFER (stream->buffer),
&clock_base, &clock_base_time);
/* clock base changes when there is a huge gap in the timestamps or seqnum.
* When this happens we don't want to calculate the extended timestamp based
* on the previous one but reset the calculation. */
if (stream->last_clock_base != clock_base) {
stream->last_extrtptime = -1;
stream->last_clock_base = clock_base;
}
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 = clock_base;
stream->clock_base_time = clock_base_time;
/* 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;
GstPadTemplate *templ;
gchar *padname;
if (!(buffer = gst_element_factory_make ("gstrtpjitterbuffer", NULL)))
goto no_jitterbuffer;
if (!(demux = gst_element_factory_make ("gstrtpptdemux", NULL)))
goto no_demux;
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);
gst_element_set_state (demux, GST_STATE_PLAYING);
/* link stuff */
gst_element_link (buffer, demux);
return stream;
/* ERRORS */
no_jitterbuffer:
{
g_warning ("gstrtpbin: could not create gstrtpjitterbuffer element");
return NULL;
}
no_demux:
{
gst_object_unref (buffer);
g_warning ("gstrtpbin: could not create gstrtpptdemux element");
return NULL;
}
}
static void
free_stream (GstRtpBinStream * stream)
{
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 void gst_rtp_bin_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
/* GstElement vmethods */
static GstClock *gst_rtp_bin_provide_clock (GstElement * element);
static GstStateChangeReturn gst_rtp_bin_change_state (GstElement * element,
GstStateChange transition);
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);
GST_BOILERPLATE (GstRtpBin, gst_rtp_bin, GstBin, GST_TYPE_BIN);
static void
gst_rtp_bin_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
/* sink pads */
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&rtpbin_recv_rtp_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&rtpbin_recv_rtcp_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&rtpbin_send_rtp_sink_template));
/* src pads */
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_send_rtcp_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&rtpbin_send_rtp_src_template));
gst_element_class_set_details (element_class, &rtpbin_details);
}
static void
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));
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);
/**
* GstRtpBin::on-sender-timeout:
* @rtpbin: the object which received the signal
* @session: the session
* @ssrc: the SSRC
*
* Notify of a sender SSRC that has timed out and became a receiver
*/
gst_rtp_bin_signals[SIGNAL_ON_SENDER_TIMEOUT] =
g_signal_new ("on-sender-timeout", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_sender_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);
gstelement_class->request_new_pad =
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)
{
gchar *str;
rtpbin->priv = GST_RTP_BIN_GET_PRIVATE (rtpbin);
rtpbin->priv->bin_lock = g_mutex_new ();
rtpbin->priv->dyn_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;
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);
g_mutex_free (rtpbin->priv->dyn_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;
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;
}
}
static void
gst_rtp_bin_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
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;
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 (&current);
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;
GstRtpBinPrivate *priv;
rtpbin = GST_RTP_BIN (element);
priv = rtpbin->priv;
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
GST_LOG_OBJECT (rtpbin, "clearing shutdown flag");
g_atomic_int_set (&priv->shutdown, 0);
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
calc_ntp_ns_base (rtpbin);
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
GST_LOG_OBJECT (rtpbin, "setting shutdown flag");
g_atomic_int_set (&priv->shutdown, 1);
/* wait for all callbacks to end by taking the lock. No new callbacks will
* be able to happen as we set the shutdown flag. */
GST_RTP_BIN_DYN_LOCK (rtpbin);
GST_LOG_OBJECT (rtpbin, "dynamic lock taken, we can continue shutdown");
GST_RTP_BIN_DYN_UNLOCK (rtpbin);
break;
default:
break;
}
res = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
return res;
}
/* 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;
GstElementClass *klass;
GstPadTemplate *templ;
gchar *padname;
GstPad *gpad;
rtpbin = stream->bin;
GST_DEBUG ("new payload pad %d", pt);
GST_RTP_BIN_SHUTDOWN_LOCK (rtpbin, shutdown);
/* ghost the pad to the parent */
klass = GST_ELEMENT_GET_CLASS (rtpbin);
templ = gst_element_class_get_pad_template (klass, "recv_rtp_src_%d_%d_%d");
padname = g_strdup_printf ("recv_rtp_src_%d_%u_%d",
stream->session->id, stream->ssrc, pt);
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);
GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin);
return;
shutdown:
{
GST_DEBUG ("ignoring, we are shutting down");
return;
}
}
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)
{
GstRtpBin *rtpbin;
GstRtpBinStream *stream;
GstPad *sinkpad, *srcpad;
gchar *padname;
GstCaps *caps;
rtpbin = session->bin;
GST_DEBUG_OBJECT (rtpbin, "new SSRC pad %08x, %s:%s", ssrc,
GST_DEBUG_PAD_NAME (pad));
GST_RTP_BIN_SHUTDOWN_LOCK (rtpbin, shutdown);
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 (rtpbin, "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 (rtpbin,
"Caps have no clock rate %s from pad %s:%s",
gst_caps_to_string (caps), GST_DEBUG_PAD_NAME (pad));
}
stream->last_clock_base = -1;
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 (rtpbin, "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 (srcpad, sinkpad);
gst_object_unref (sinkpad);
gst_object_unref (srcpad);
/* get the RTCP sync pad */
GST_DEBUG_OBJECT (rtpbin, "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, 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);
GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin);
return;
/* ERRORS */
shutdown:
{
GST_DEBUG_OBJECT (rtpbin, "we are shutting down");
return;
}
no_stream:
{
GST_RTP_SESSION_UNLOCK (session);
GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin);
GST_DEBUG_OBJECT (rtpbin, "could not create stream");
return;
}
}
/* Create a pad for receiving RTP for the session in @name. Must be called with
* RTP_BIN_LOCK.
*/
static GstPad *
create_recv_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
{
GstPad *result, *sinkdpad;
guint sessid;
GstRtpBinSession *session;
GstPadLinkReturn lres;
/* first get the session number */
if (name == NULL || sscanf (name, "recv_rtp_sink_%d", &sessid) != 1)
goto no_name;
GST_DEBUG_OBJECT (rtpbin, "finding session %d", sessid);
/* get or create session */
session = find_session_by_id (rtpbin, sessid);
if (!session) {
GST_DEBUG_OBJECT (rtpbin, "creating session %d", sessid);
/* create session now */
session = create_session (rtpbin, sessid);
if (session == NULL)
goto create_error;
}
/* check if pad was requested */
if (session->recv_rtp_sink != NULL)
goto existed;
GST_DEBUG_OBJECT (rtpbin, "getting RTP sink pad");
/* get recv_rtp pad and store */
session->recv_rtp_sink =
gst_element_get_request_pad (session->session, "recv_rtp_sink");
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 =
gst_element_get_static_pad (session->session, "recv_rtp_src");
if (session->recv_rtp_src == NULL)
goto pad_failed;
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 SSRC demuxer */
session->demux_newpad_sig = g_signal_connect (session->demux,
"new-ssrc-pad", (GCallback) new_ssrc_pad_found, session);
GST_DEBUG_OBJECT (rtpbin, "ghosting session sink pad");
result =
gst_ghost_pad_new_from_template (name, session->recv_rtp_sink, templ);
gst_pad_set_active (result, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), result);
return result;
/* ERRORS */
no_name:
{
g_warning ("gstrtpbin: invalid name given");
return NULL;
}
create_error:
{
/* create_session already warned */
return NULL;
}
existed:
{
g_warning ("gstrtpbin: recv_rtp pad already requested for session %d",
sessid);
return NULL;
}
pad_failed:
{
g_warning ("gstrtpbin: failed to get session pad");
return NULL;
}
link_failed:
{
g_warning ("gstrtpbin: failed to link pads");
return NULL;
}
}
/* Create a pad for receiving RTCP for the session in @name. Must be called with
* RTP_BIN_LOCK.
*/
static GstPad *
create_recv_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ,
const gchar * name)
{
GstPad *result;
guint sessid;
GstRtpBinSession *session;
GstPad *sinkdpad;
GstPadLinkReturn lres;
/* first get the session number */
if (name == NULL || sscanf (name, "recv_rtcp_sink_%d", &sessid) != 1)
goto no_name;
GST_DEBUG_OBJECT (rtpbin, "finding session %d", sessid);
/* get or create the session */
session = find_session_by_id (rtpbin, sessid);
if (!session) {
GST_DEBUG_OBJECT (rtpbin, "creating session %d", sessid);
/* create session now */
session = create_session (rtpbin, sessid);
if (session == NULL)
goto create_error;
}
/* check if pad was requested */
if (session->recv_rtcp_sink != NULL)
goto existed;
/* 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;
/* get srcpad, link to SSRCDemux */
GST_DEBUG_OBJECT (rtpbin, "getting sync src pad");
session->sync_src = gst_element_get_static_pad (session->session, "sync_src");
if (session->sync_src == NULL)
goto pad_failed;
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;
result =
gst_ghost_pad_new_from_template (name, session->recv_rtcp_sink, templ);
gst_pad_set_active (result, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), result);
return result;
/* ERRORS */
no_name:
{
g_warning ("gstrtpbin: invalid name given");
return NULL;
}
create_error:
{
/* create_session already warned */
return NULL;
}
existed:
{
g_warning ("gstrtpbin: recv_rtcp pad already requested for session %d",
sessid);
return NULL;
}
pad_failed:
{
g_warning ("gstrtpbin: failed to get session pad");
return NULL;
}
link_failed:
{
g_warning ("gstrtpbin: failed to link pads");
return NULL;
}
}
/* 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)
{
GstPad *result, *srcghost;
gchar *gname;
guint sessid;
GstRtpBinSession *session;
GstElementClass *klass;
/* first get the session number */
if (name == NULL || sscanf (name, "send_rtp_sink_%d", &sessid) != 1)
goto no_name;
/* get or create session */
session = find_session_by_id (rtpbin, sessid);
if (!session) {
/* create session now */
session = create_session (rtpbin, sessid);
if (session == NULL)
goto create_error;
}
/* check if pad was requested */
if (session->send_rtp_sink != NULL)
goto existed;
/* 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)
goto pad_failed;
result =
gst_ghost_pad_new_from_template (name, session->send_rtp_sink, templ);
gst_pad_set_active (result, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), result);
/* get srcpad */
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 */
klass = GST_ELEMENT_GET_CLASS (rtpbin);
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_src, templ);
gst_pad_set_active (srcghost, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), srcghost);
g_free (gname);
return result;
/* ERRORS */
no_name:
{
g_warning ("gstrtpbin: invalid name given");
return NULL;
}
create_error:
{
/* create_session already warned */
return NULL;
}
existed:
{
g_warning ("gstrtpbin: send_rtp pad already requested for session %d",
sessid);
return NULL;
}
pad_failed:
{
g_warning ("gstrtpbin: failed to get session pad for session %d", sessid);
return NULL;
}
no_srcpad:
{
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. Must be called with
* RTP_BIN_LOCK.
*/
static GstPad *
create_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
{
GstPad *result;
guint sessid;
GstRtpBinSession *session;
/* first get the session number */
if (name == NULL || sscanf (name, "send_rtcp_src_%d", &sessid) != 1)
goto no_name;
/* get or create session */
session = find_session_by_id (rtpbin, sessid);
if (!session)
goto no_session;
/* check if pad was requested */
if (session->send_rtcp_src != NULL)
goto existed;
/* get rtcp_src pad and store */
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->send_rtcp_src, templ);
gst_pad_set_active (result, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), result);
return result;
/* ERRORS */
no_name:
{
g_warning ("gstrtpbin: invalid name given");
return NULL;
}
no_session:
{
g_warning ("gstrtpbin: session with id %d does not exist", sessid);
return NULL;
}
existed:
{
g_warning ("gstrtpbin: send_rtcp_src pad already requested for session %d",
sessid);
return NULL;
}
pad_failed:
{
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;
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, pad_name);
} else if (templ == gst_element_class_get_pad_template (klass,
"recv_rtcp_sink_%d")) {
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, 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_free (pad_name);
GST_RTP_BIN_UNLOCK (rtpbin);
g_warning ("gstrtpbin: this is not our template");
return NULL;
}
}
static void
gst_rtp_bin_release_pad (GstElement * element, GstPad * pad)
{
}
Reference in a new issue View git blame Copy permalink