mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-09 19:09:41 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/gst/rtpmanager/gstrtpbin.c
Wim Taymans 6494828ef2 gst/rtpmanager/gstrtpbin.c: Also set NTP base time on new sessions. 
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_session):
Also set NTP base time on new sessions.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_query),
(gst_rtp_jitter_buffer_set_property),
(gst_rtp_jitter_buffer_get_property):
Use the right lock to protect our variables.
Fix some comment.
* gst/rtpmanager/gstrtpsession.c:
(gst_rtp_session_getcaps_send_rtp),
(gst_rtp_session_chain_send_rtp), (create_send_rtp_sink):
Implement getcaps on the sender sinkpad so that payloaders can negotiate
the right SSRC.
2007-09-15 18:48:03 +00:00

1821 lines
53 KiB
C
Raw Blame History

/* GStreamer
* Copyright (C) <2007> Wim Taymans <wim@fluendo.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
* @short_description: handle media from one RTP bin
* @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. The session number must be specified when
* requesting the sink pad. 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 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 2007-08-30 (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"
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@fluendo.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)
struct _GstRtpBinPrivate
{
GMutex *bin_lock;
GstClockTime ntp_ns_base;
};
/* 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_BYE_SSRC,
SIGNAL_ON_BYE_TIMEOUT,
SIGNAL_ON_TIMEOUT,
LAST_SIGNAL
};
#define DEFAULT_LATENCY_MS 200
enum
{
PROP_0,
PROP_LATENCY
};
/* 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 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;
/* 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;
gint clock_rate;
gint64 ts_offset;
gint64 prev_ts_offset;
};
#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_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);
}
/* 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;
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 (NULL, NULL);
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);
/* 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-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);
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);
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)
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);
caps = (GstCaps *) g_value_get_boxed (&ret);
if (!caps)
goto no_caps;
GST_DEBUG ("caching pt %d as %" GST_PTR_FORMAT, pt, caps);
/* store in cache */
g_hash_table_insert (session->ptmap, GINT_TO_POINTER (pt), 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 *walk;
GST_RTP_BIN_LOCK (bin);
GST_DEBUG_OBJECT (bin, "clearing pt map");
for (walk = bin->sessions; walk; walk = g_slist_next (walk)) {
GstRtpBinSession *session = (GstRtpBinSession *) walk->data;
GST_RTP_SESSION_LOCK (session);
#if 0
/* This requires GLib 2.12 */
g_hash_table_remove_all (session->ptmap);
#else
g_hash_table_foreach_remove (session->ptmap, return_true, NULL);
#endif
GST_RTP_SESSION_UNLOCK (session);
}
GST_RTP_BIN_UNLOCK (bin);
}
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, GstRtpBin * bin)
{
bin->clients = g_slist_remove (bin->clients, client);
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 */
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)
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);
/* 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 < 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;
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);
}
}
return;
no_clock_base:
{
GST_WARNING_OBJECT (bin, "we have no clock-base");
return;
}
no_clock_rate:
{
GST_WARNING_OBJECT (bin, "we have no clock-rate");
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);
/* 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->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);
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 */
g_object_set (buffer, "latency", session->bin->latency, 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_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;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) 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_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-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);
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);
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)
{
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;
}
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_foreach (rtpbin->clients, (GFunc) free_client, NULL);
g_slist_free (rtpbin->sessions);
rtpbin->sessions = NULL;
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_rtp_bin_finalize (GObject * object)
{
GstRtpBin *rtpbin;
rtpbin = GST_RTP_BIN (object);
g_mutex_free (rtpbin->priv->bin_lock);
gst_object_unref (rtpbin->provided_clock);
g_slist_free (rtpbin->sessions);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
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:
rtpbin->latency = g_value_get_uint (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:
g_value_set_uint (value, rtpbin->latency);
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
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;
rtpbin = GST_RTP_BIN (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
calc_ntp_ns_base (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 */
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);
/* 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);
}
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);
}
/* a new pad (SSRC) was created in @session */
static void
new_ssrc_pad_found (GstElement * element, guint ssrc, GstPad * pad,
GstRtpBinSession * session)
{
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;
if (gst_structure_get_uint (s, "clock-base", &val))
stream->clock_base = val;
else
stream->clock_base = -1;
}
/* get pad and link */
GST_DEBUG_OBJECT (session->bin, "linking jitterbuffer");
sinkpad = gst_element_get_static_pad (stream->buffer, "sink");
gst_pad_link (pad, sinkpad);
gst_object_unref (sinkpad);
/* 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_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);
GST_RTP_SESSION_UNLOCK (session);
return;
/* ERRORS */
no_stream:
{
GST_RTP_SESSION_UNLOCK (session);
GST_DEBUG ("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;
}
}
/*
*/
static GstPad *
gst_rtp_bin_request_new_pad (GstElement * element,
GstPadTemplate * templ, const gchar * name)
{
GstRtpBin *rtpbin;
GstElementClass *klass;
GstPad *result;
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);
/* figure out the template */
if (templ == gst_element_class_get_pad_template (klass, "recv_rtp_sink_%d")) {
result = create_recv_rtp (rtpbin, templ, name);
} else if (templ == gst_element_class_get_pad_template (klass,
"recv_rtcp_sink_%d")) {
result = create_recv_rtcp (rtpbin, templ, 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,
"send_rtcp_src_%d")) {
result = create_rtcp (rtpbin, templ, name);
} else
goto wrong_template;
GST_RTP_BIN_UNLOCK (rtpbin);
return result;
/* ERRORS */
wrong_template:
{
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