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gstreamer/gst/rtpmanager/gstrtpbin.c
Sebastian Dröge 2f47105129 rtpbin: Don't leak caps
2014-06-29 23:55:19 +02:00

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/* GStreamer
* Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-rtpbin
* @see_also: rtpjitterbuffer, rtpsession, rtpptdemux, rtpssrcdemux
*
* 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_\%u pad. The session
* number must be specified in the pad name.
* Data received on the recv_rtp_sink_\%u 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
* #GstRtpPtDemux element. The #GstRtpPtDemux element will demux the packets based
* on the payload type and will create a unique pad recv_rtp_src_\%u_\%u_\%u on
* rtpbin 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_\%u 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_\%u 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_\%u pad, which will
* automatically create a send_rtp_src_\%u 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_\%u 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.
*
* Access to the internal statistics of rtpbin is provided with the
* get-internal-session property. This action signal gives access to the
* RTPSession object which further provides action signals to retrieve the
* internal source and other sources.
*
* #GstRtpBin also has signals (#GstRtpBin::request-rtp-encoder,
* #GstRtpBin::request-rtp-decoder, #GstRtpBin::request-rtcp-encoder and
* #GstRtpBin::request-rtp-decoder) to dynamically request for RTP and RTCP encoders
* and decoders in order to support SRTP. The encoders must provide the pads
* rtp_sink_\%u and rtp_src_\%u for RTP and rtcp_sink_\%u and rtcp_src_\%u for
* RTCP. The session number will be used in the pad name. The decoders must provide
* rtp_sink and rtp_src for RTP and rtcp_sink and rtcp_src for RTCP. The decoders will
* be placed before the #GstRtpSession element, thus they must support SSRC demuxing
* internally.
*
* #GstRtpBin has signals (#GstRtpBin::request-aux-sender and
* #GstRtpBin::request-aux-receiver to dynamically request an element that can be
* used to create or merge additional RTP streams. AUX elements are needed to
* implement FEC or retransmission (such as RFC 4588). An AUX sender must have one
* sink_\%u pad that matches the sessionid in the signal and it should have 1 or
* more src_\%u pads. For each src_%\u pad, a session will be made (if needed)
* and the pad will be linked to the session send_rtp_sink pad. Each session will
* then expose its source pad ad send_rtp_src_\%u on #GstRtpBin.
* An AUX receiver has 1 src_\%u pad that much match the sessionid in the signal
* and 1 or more sink_\%u pads. A session will be made for each sink_\%u pad
* when the corresponding recv_rtp_sink_\%u pad is requested on #GstRtpBin.
*
* <refsect2>
* <title>Example pipelines</title>
* |[
* gst-launch-1.0 udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink_0 \
* rtpbin ! rtptheoradepay ! theoradec ! xvimagesink
* ]| Receive RTP data from port 5000 and send to the session 0 in rtpbin.
* |[
* gst-launch-1.0 rtpbin name=rtpbin \
* v4l2src ! videoconvert ! 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-1.0 -v rtpbin 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>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <stdio.h>
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/rtp/gstrtcpbuffer.h>
#include "gstrtpbin.h"
#include "rtpsession.h"
#include "gstrtpsession.h"
#include "gstrtpjitterbuffer.h"
#include <gst/glib-compat-private.h>
GST_DEBUG_CATEGORY_STATIC (gst_rtp_bin_debug);
#define GST_CAT_DEFAULT gst_rtp_bin_debug
/* sink pads */
static GstStaticPadTemplate rtpbin_recv_rtp_sink_template =
GST_STATIC_PAD_TEMPLATE ("recv_rtp_sink_%u",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("application/x-rtp;application/x-srtp")
);
static GstStaticPadTemplate rtpbin_recv_rtcp_sink_template =
GST_STATIC_PAD_TEMPLATE ("recv_rtcp_sink_%u",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("application/x-rtcp;application/x-srtcp")
);
static GstStaticPadTemplate rtpbin_send_rtp_sink_template =
GST_STATIC_PAD_TEMPLATE ("send_rtp_sink_%u",
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_%u_%u_%u",
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_%u",
GST_PAD_SRC,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("application/x-rtcp;application/x-srtcp")
);
static GstStaticPadTemplate rtpbin_send_rtp_src_template =
GST_STATIC_PAD_TEMPLATE ("send_rtp_src_%u",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS ("application/x-rtp;application/x-srtp")
);
#define GST_RTP_BIN_GET_PRIVATE(obj) \
(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTP_BIN, GstRtpBinPrivate))
#define GST_RTP_BIN_LOCK(bin) g_mutex_lock (&(bin)->priv->bin_lock)
#define GST_RTP_BIN_UNLOCK(bin) g_mutex_unlock (&(bin)->priv->bin_lock)
/* 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;
/* if we are shutting down or not */
gint shutdown;
gboolean autoremove;
/* UNIX (ntp) time of last SR sync used */
guint64 last_unix;
/* list of extra elements */
GList *elements;
};
/* signals and args */
enum
{
SIGNAL_REQUEST_PT_MAP,
SIGNAL_PAYLOAD_TYPE_CHANGE,
SIGNAL_CLEAR_PT_MAP,
SIGNAL_RESET_SYNC,
SIGNAL_GET_INTERNAL_SESSION,
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,
SIGNAL_ON_NPT_STOP,
SIGNAL_REQUEST_RTP_ENCODER,
SIGNAL_REQUEST_RTP_DECODER,
SIGNAL_REQUEST_RTCP_ENCODER,
SIGNAL_REQUEST_RTCP_DECODER,
SIGNAL_NEW_JITTERBUFFER,
SIGNAL_REQUEST_AUX_SENDER,
SIGNAL_REQUEST_AUX_RECEIVER,
LAST_SIGNAL
};
#define DEFAULT_LATENCY_MS 200
#define DEFAULT_DROP_ON_LATENCY FALSE
#define DEFAULT_SDES NULL
#define DEFAULT_DO_LOST FALSE
#define DEFAULT_IGNORE_PT FALSE
#define DEFAULT_NTP_SYNC FALSE
#define DEFAULT_AUTOREMOVE FALSE
#define DEFAULT_BUFFER_MODE RTP_JITTER_BUFFER_MODE_SLAVE
#define DEFAULT_USE_PIPELINE_CLOCK FALSE
#define DEFAULT_RTCP_SYNC GST_RTP_BIN_RTCP_SYNC_ALWAYS
#define DEFAULT_RTCP_SYNC_INTERVAL 0
#define DEFAULT_DO_SYNC_EVENT FALSE
#define DEFAULT_DO_RETRANSMISSION FALSE
enum
{
PROP_0,
PROP_LATENCY,
PROP_DROP_ON_LATENCY,
PROP_SDES,
PROP_DO_LOST,
PROP_IGNORE_PT,
PROP_NTP_SYNC,
PROP_RTCP_SYNC,
PROP_RTCP_SYNC_INTERVAL,
PROP_AUTOREMOVE,
PROP_BUFFER_MODE,
PROP_USE_PIPELINE_CLOCK,
PROP_DO_SYNC_EVENT,
PROP_DO_RETRANSMISSION,
PROP_LAST
};
enum
{
GST_RTP_BIN_RTCP_SYNC_ALWAYS,
GST_RTP_BIN_RTCP_SYNC_INITIAL,
GST_RTP_BIN_RTCP_SYNC_RTP
};
#define GST_RTP_BIN_RTCP_SYNC_TYPE (gst_rtp_bin_rtcp_sync_get_type())
static GType
gst_rtp_bin_rtcp_sync_get_type (void)
{
static GType rtcp_sync_type = 0;
static const GEnumValue rtcp_sync_types[] = {
{GST_RTP_BIN_RTCP_SYNC_ALWAYS, "always", "always"},
{GST_RTP_BIN_RTCP_SYNC_INITIAL, "initial", "initial"},
{GST_RTP_BIN_RTCP_SYNC_RTP, "rtp-info", "rtp-info"},
{0, NULL, NULL},
};
if (!rtcp_sync_type) {
rtcp_sync_type = g_enum_register_static ("GstRTCPSync", rtcp_sync_types);
}
return rtcp_sync_type;
}
/* 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 payload_type_change (GstElement * element, guint pt,
GstRtpBinSession * session);
static void remove_recv_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session);
static void remove_recv_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session);
static void remove_send_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session);
static void remove_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session);
static void free_client (GstRtpBinClient * client, GstRtpBin * bin);
static void free_stream (GstRtpBinStream * stream, GstRtpBin * bin);
/* 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;
gulong buffer_handlesync_sig;
gulong buffer_ptreq_sig;
gulong buffer_ntpstop_sig;
gint percent;
/* the PT demuxer of the SSRC */
GstElement *demux;
gulong demux_newpad_sig;
gulong demux_padremoved_sig;
gulong demux_ptreq_sig;
gulong demux_ptchange_sig;
/* if we have calculated a valid rt_delta for this stream */
gboolean have_sync;
/* mapping to local RTP and NTP time */
gint64 rt_delta;
gint64 rtp_delta;
/* base rtptime in gst time */
gint64 clock_base;
};
#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;
gulong demux_padremoved_sig;
GMutex lock;
/* list of GstRtpBinStream */
GSList *streams;
/* list of elements */
GSList *elements;
/* mapping of payload type to caps */
GHashTable *ptmap;
/* the pads of the session */
GstPad *recv_rtp_sink;
GstPad *recv_rtp_sink_ghost;
GstPad *recv_rtp_src;
GstPad *recv_rtcp_sink;
GstPad *recv_rtcp_sink_ghost;
GstPad *sync_src;
GstPad *send_rtp_sink;
GstPad *send_rtp_sink_ghost;
GstPad *send_rtp_src;
GstPad *send_rtp_src_ghost;
GstPad *send_rtcp_src;
GstPad *send_rtcp_src_ghost;
};
/* 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;
};
/* 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;
}
/* find a session with the given request pad. Must be called with RTP_BIN_LOCK */
static GstRtpBinSession *
find_session_by_pad (GstRtpBin * rtpbin, GstPad * pad)
{
GSList *walk;
for (walk = rtpbin->sessions; walk; walk = g_slist_next (walk)) {
GstRtpBinSession *sess = (GstRtpBinSession *) walk->data;
if ((sess->recv_rtp_sink_ghost == pad) ||
(sess->recv_rtcp_sink_ghost == pad) ||
(sess->send_rtp_sink_ghost == pad)
|| (sess->send_rtcp_src_ghost == pad))
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);
if (sess->bin->priv->autoremove)
g_signal_emit_by_name (sess->demux, "clear-ssrc", ssrc, NULL);
}
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);
if (sess->bin->priv->autoremove)
g_signal_emit_by_name (sess->demux, "clear-ssrc", ssrc, NULL);
}
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);
}
static void
on_npt_stop (GstElement * jbuf, GstRtpBinStream * stream)
{
g_signal_emit (stream->bin, gst_rtp_bin_signals[SIGNAL_ON_NPT_STOP], 0,
stream->session->id, stream->ssrc);
}
/* must be called with the SESSION lock */
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;
}
static void
ssrc_demux_pad_removed (GstElement * element, guint ssrc, GstPad * pad,
GstRtpBinSession * session)
{
GstRtpBinStream *stream = NULL;
GstRtpBin *rtpbin;
rtpbin = session->bin;
GST_RTP_BIN_LOCK (rtpbin);
GST_RTP_SESSION_LOCK (session);
if ((stream = find_stream_by_ssrc (session, ssrc)))
session->streams = g_slist_remove (session->streams, stream);
GST_RTP_SESSION_UNLOCK (session);
if (stream)
free_stream (stream, rtpbin);
GST_RTP_BIN_UNLOCK (rtpbin);
}
/* create a session with the given id. Must be called with RTP_BIN_LOCK */
static GstRtpBinSession *
create_session (GstRtpBin * rtpbin, gint id)
{
GstRtpBinSession *sess;
GstElement *session, *demux;
GstState target;
if (!(session = gst_element_factory_make ("rtpsession", NULL)))
goto no_session;
if (!(demux = gst_element_factory_make ("rtpssrcdemux", NULL)))
goto no_demux;
sess = g_new0 (GstRtpBinSession, 1);
g_mutex_init (&sess->lock);
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);
/* configure SDES items */
GST_OBJECT_LOCK (rtpbin);
g_object_set (session, "sdes", rtpbin->sdes, "use-pipeline-clock",
rtpbin->use_pipeline_clock, 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);
gst_bin_add (GST_BIN_CAST (rtpbin), session);
gst_bin_add (GST_BIN_CAST (rtpbin), demux);
GST_OBJECT_LOCK (rtpbin);
target = GST_STATE_TARGET (rtpbin);
GST_OBJECT_UNLOCK (rtpbin);
/* change state only to what's needed */
gst_element_set_state (demux, target);
gst_element_set_state (session, target);
return sess;
/* ERRORS */
no_session:
{
g_warning ("rtpbin: could not create rtpsession element");
return NULL;
}
no_demux:
{
gst_object_unref (session);
g_warning ("rtpbin: could not create rtpssrcdemux element");
return NULL;
}
}
static gboolean
bin_manage_element (GstRtpBin * bin, GstElement * element)
{
GstRtpBinPrivate *priv = bin->priv;
if (g_list_find (priv->elements, element)) {
GST_DEBUG_OBJECT (bin, "requested element %p already in bin", element);
} else {
GST_DEBUG_OBJECT (bin, "adding requested element %p", element);
if (!gst_bin_add (GST_BIN_CAST (bin), element))
goto add_failed;
if (!gst_element_sync_state_with_parent (element))
GST_WARNING_OBJECT (bin, "unable to sync element state with rtpbin");
}
/* we add the element multiple times, each we need an equal number of
* removes to really remove the element from the bin */
priv->elements = g_list_prepend (priv->elements, element);
return TRUE;
/* ERRORS */
add_failed:
{
GST_WARNING_OBJECT (bin, "unable to add element");
return FALSE;
}
}
static void
remove_bin_element (GstElement * element, GstRtpBin * bin)
{
GstRtpBinPrivate *priv = bin->priv;
GList *find;
find = g_list_find (priv->elements, element);
if (find) {
priv->elements = g_list_delete_link (priv->elements, find);
if (!g_list_find (priv->elements, element))
gst_bin_remove (GST_BIN_CAST (bin), element);
else
gst_object_unref (element);
}
}
/* called with RTP_BIN_LOCK */
static void
free_session (GstRtpBinSession * sess, GstRtpBin * bin)
{
GST_DEBUG_OBJECT (bin, "freeing session %p", sess);
gst_element_set_locked_state (sess->demux, TRUE);
gst_element_set_locked_state (sess->session, TRUE);
gst_element_set_state (sess->demux, GST_STATE_NULL);
gst_element_set_state (sess->session, GST_STATE_NULL);
remove_recv_rtp (bin, sess);
remove_recv_rtcp (bin, sess);
remove_send_rtp (bin, sess);
remove_rtcp (bin, sess);
gst_bin_remove (GST_BIN_CAST (bin), sess->session);
gst_bin_remove (GST_BIN_CAST (bin), sess->demux);
g_slist_foreach (sess->elements, (GFunc) remove_bin_element, bin);
g_slist_free (sess->elements);
g_slist_foreach (sess->streams, (GFunc) free_stream, bin);
g_slist_free (sess->streams);
g_mutex_clear (&sess->lock);
g_hash_table_destroy (sess->ptmap);
g_free (sess);
}
/* 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_reset_sync (GstRtpBin * rtpbin)
{
GSList *clients, *streams;
GST_DEBUG_OBJECT (rtpbin, "Reset sync on all clients");
GST_RTP_BIN_LOCK (rtpbin);
for (clients = rtpbin->clients; clients; clients = g_slist_next (clients)) {
GstRtpBinClient *client = (GstRtpBinClient *) clients->data;
/* reset sync on all streams for this client */
for (streams = client->streams; streams; streams = g_slist_next (streams)) {
GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
/* make use require a new SR packet for this stream before we attempt new
* lip-sync */
stream->have_sync = FALSE;
stream->rt_delta = 0;
stream->rtp_delta = 0;
stream->clock_base = -100 * GST_SECOND;
}
}
GST_RTP_BIN_UNLOCK (rtpbin);
}
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);
if (stream->demux)
g_signal_emit_by_name (stream->demux, "clear-pt-map", NULL);
}
GST_RTP_SESSION_UNLOCK (session);
}
GST_RTP_BIN_UNLOCK (bin);
/* reset sync too */
gst_rtp_bin_reset_sync (bin);
}
static RTPSession *
gst_rtp_bin_get_internal_session (GstRtpBin * bin, guint session_id)
{
RTPSession *internal_session = NULL;
GstRtpBinSession *session;
GST_RTP_BIN_LOCK (bin);
GST_DEBUG_OBJECT (bin, "retrieving internal RTPSession object, index: %d",
session_id);
session = find_session_by_id (bin, (gint) session_id);
if (session) {
g_object_get (session->session, "internal-session", &internal_session,
NULL);
}
GST_RTP_BIN_UNLOCK (bin);
return internal_session;
}
static GstElement *
gst_rtp_bin_request_encoder (GstRtpBin * bin, guint session_id)
{
GST_DEBUG_OBJECT (bin, "return NULL encoder");
return NULL;
}
static GstElement *
gst_rtp_bin_request_decoder (GstRtpBin * bin, guint session_id)
{
GST_DEBUG_OBJECT (bin, "return NULL decoder");
return NULL;
}
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;
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)
{
GST_DEBUG_OBJECT (bin, "freeing client %p", client);
g_slist_free (client->streams);
g_free (client->cname);
g_free (client);
}
static void
get_current_times (GstRtpBin * bin, GstClockTime * running_time,
guint64 * ntpnstime)
{
guint64 ntpns;
GstClock *clock;
GstClockTime base_time, rt, clock_time;
GST_OBJECT_LOCK (bin);
if ((clock = GST_ELEMENT_CLOCK (bin))) {
base_time = GST_ELEMENT_CAST (bin)->base_time;
gst_object_ref (clock);
GST_OBJECT_UNLOCK (bin);
clock_time = gst_clock_get_time (clock);
if (bin->use_pipeline_clock) {
ntpns = clock_time - base_time;
} else {
GTimeVal current;
/* get current NTP time */
g_get_current_time (&current);
ntpns = GST_TIMEVAL_TO_TIME (current);
}
/* add constant to convert from 1970 based time to 1900 based time */
ntpns += (2208988800LL * GST_SECOND);
/* get current clock time and convert to running time */
rt = clock_time - base_time;
gst_object_unref (clock);
} else {
GST_OBJECT_UNLOCK (bin);
rt = -1;
ntpns = -1;
}
if (running_time)
*running_time = rt;
if (ntpnstime)
*ntpnstime = ntpns;
}
static void
stream_set_ts_offset (GstRtpBin * bin, GstRtpBinStream * stream,
gint64 ts_offset, gboolean check)
{
gint64 prev_ts_offset;
g_object_get (stream->buffer, "ts-offset", &prev_ts_offset, NULL);
/* delta changed, see how much */
if (prev_ts_offset != ts_offset) {
gint64 diff;
diff = prev_ts_offset - ts_offset;
GST_DEBUG_OBJECT (bin,
"ts-offset %" G_GINT64_FORMAT ", prev %" G_GINT64_FORMAT
", diff: %" G_GINT64_FORMAT, ts_offset, prev_ts_offset, diff);
if (check) {
/* only change diff when it changed more than 4 milliseconds. This
* compensates for rounding errors in NTP to RTP timestamp
* conversions */
if (ABS (diff) < 4 * GST_MSECOND) {
GST_DEBUG_OBJECT (bin, "offset too small, ignoring");
return;
}
if (ABS (diff) > (3 * GST_SECOND)) {
GST_WARNING_OBJECT (bin, "offset unusually large, ignoring");
return;
}
}
g_object_set (stream->buffer, "ts-offset", ts_offset, NULL);
}
GST_DEBUG_OBJECT (bin, "stream SSRC %08x, delta %" G_GINT64_FORMAT,
stream->ssrc, ts_offset);
}
static void
gst_rtp_bin_send_sync_event (GstRtpBinStream * stream)
{
if (stream->bin->send_sync_event) {
GstEvent *event;
GstPad *srcpad;
GST_DEBUG_OBJECT (stream->bin,
"sending GstRTCPSRReceived event downstream");
event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM,
gst_structure_new_empty ("GstRTCPSRReceived"));
srcpad = gst_element_get_static_pad (stream->buffer, "src");
gst_pad_push_event (srcpad, event);
gst_object_unref (srcpad);
}
}
/* associate a stream to the given CNAME. This will make sure all streams for
* that CNAME are synchronized together.
* Must be called with GST_RTP_BIN_LOCK */
static void
gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream, guint8 len,
guint8 * data, guint64 ntptime, guint64 last_extrtptime,
guint64 base_rtptime, guint64 base_time, guint clock_rate,
gint64 rtp_clock_base)
{
GstRtpBinClient *client;
gboolean created;
GSList *walk;
guint64 local_rt;
guint64 local_rtp;
GstClockTime running_time;
guint64 ntpnstime;
gint64 ntpdiff, rtdiff;
guint64 last_unix;
/* 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);
}
if (!GST_CLOCK_TIME_IS_VALID (last_extrtptime)) {
GST_DEBUG_OBJECT (bin, "invalidated sync data");
if (bin->rtcp_sync == GST_RTP_BIN_RTCP_SYNC_RTP) {
/* we don't need that data, so carry on,
* but make some values look saner */
last_extrtptime = base_rtptime;
} else {
/* nothing we can do with this data in this case */
GST_DEBUG_OBJECT (bin, "bailing out");
return;
}
}
/* 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 so we will calculate what running_time corresponds to the RTP
* timestamp in the SR packet. */
local_rtp = last_extrtptime - base_rtptime;
GST_DEBUG_OBJECT (bin,
"base %" G_GUINT64_FORMAT ", extrtptime %" G_GUINT64_FORMAT
", local RTP %" G_GUINT64_FORMAT ", clock-rate %d, "
"clock-base %" G_GINT64_FORMAT, base_rtptime,
last_extrtptime, local_rtp, clock_rate, rtp_clock_base);
/* calculate local RTP 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. Note that the base_time also
* contains the drift between sender and receiver. */
local_rt = gst_util_uint64_scale_int (local_rtp, GST_SECOND, clock_rate);
local_rt += base_time;
/* convert ntptime to unix time since 1900 */
last_unix = gst_util_uint64_scale (ntptime, GST_SECOND,
(G_GINT64_CONSTANT (1) << 32));
stream->have_sync = TRUE;
GST_DEBUG_OBJECT (bin,
"local UNIX %" G_GUINT64_FORMAT ", remote UNIX %" G_GUINT64_FORMAT,
local_rt, last_unix);
/* recalc inter stream playout offset, but only if there is more than one
* stream or we're doing NTP sync. */
if (bin->ntp_sync) {
/* For NTP sync we need to first get a snapshot of running_time and NTP
* time. We know at what running_time we play a certain RTP time, we also
* calculated when we would play the RTP time in the SR packet. Now we need
* to know how the running_time and the NTP time relate to eachother. */
get_current_times (bin, &running_time, &ntpnstime);
/* see how far away the NTP time is. This is the difference between the
* current NTP time and the NTP time in the last SR packet. */
ntpdiff = ntpnstime - last_unix;
/* see how far away the running_time is. This is the difference between the
* current running_time and the running_time of the RTP timestamp in the
* last SR packet. */
rtdiff = running_time - local_rt;
GST_DEBUG_OBJECT (bin,
"NTP time %" G_GUINT64_FORMAT ", last unix %" G_GUINT64_FORMAT,
ntpnstime, last_unix);
GST_DEBUG_OBJECT (bin,
"NTP diff %" G_GINT64_FORMAT ", RT diff %" G_GINT64_FORMAT, ntpdiff,
rtdiff);
/* combine to get the final diff to apply to the running_time */
stream->rt_delta = rtdiff - ntpdiff;
stream_set_ts_offset (bin, stream, stream->rt_delta, FALSE);
} else {
gint64 min, rtp_min, clock_base = stream->clock_base;
gboolean all_sync, use_rtp;
gboolean rtcp_sync = g_atomic_int_get (&bin->rtcp_sync);
/* 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. The
* difference in itself doesn't mean much but we can combine the delta of
* multiple streams to create a stream specific offset. */
stream->rt_delta = last_unix - local_rt;
/* calculate the min of all deltas, ignoring streams that did not yet have a
* valid rt_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 smallest diff is selected as the reference stream,
* all other streams will have a positive offset to this difference. */
/* some alternative setting allow ignoring RTCP as much as possible,
* for servers generating bogus ntp timeline */
min = rtp_min = G_MAXINT64;
use_rtp = FALSE;
if (rtcp_sync == GST_RTP_BIN_RTCP_SYNC_RTP) {
guint64 ext_base;
use_rtp = TRUE;
/* signed version for convienience */
clock_base = base_rtptime;
/* deal with possible wrap-around */
ext_base = base_rtptime;
rtp_clock_base = gst_rtp_buffer_ext_timestamp (&ext_base, rtp_clock_base);
/* sanity check; base rtp and provided clock_base should be close */
if (rtp_clock_base >= clock_base) {
if (rtp_clock_base - clock_base < 10 * clock_rate) {
rtp_clock_base = base_time +
gst_util_uint64_scale_int (rtp_clock_base - clock_base,
GST_SECOND, clock_rate);
} else {
use_rtp = FALSE;
}
} else {
if (clock_base - rtp_clock_base < 10 * clock_rate) {
rtp_clock_base = base_time -
gst_util_uint64_scale_int (clock_base - rtp_clock_base,
GST_SECOND, clock_rate);
} else {
use_rtp = FALSE;
}
}
/* warn and bail for clarity out if no sane values */
if (!use_rtp) {
GST_WARNING_OBJECT (bin, "unable to sync to provided rtptime");
return;
}
/* store to track changes */
clock_base = rtp_clock_base;
/* generate a fake as before,
* now equating rtptime obtained from RTP-Info,
* where the large time represent the otherwise irrelevant npt/ntp time */
stream->rtp_delta = (GST_SECOND << 28) - rtp_clock_base;
} else {
clock_base = rtp_clock_base;
}
all_sync = TRUE;
for (walk = client->streams; walk; walk = g_slist_next (walk)) {
GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data;
if (!ostream->have_sync) {
all_sync = FALSE;
continue;
}
/* change in current stream's base from previously init'ed value
* leads to reset of all stream's base */
if (stream != ostream && stream->clock_base >= 0 &&
(stream->clock_base != clock_base)) {
GST_DEBUG_OBJECT (bin, "reset upon clock base change");
ostream->clock_base = -100 * GST_SECOND;
ostream->rtp_delta = 0;
}
if (ostream->rt_delta < min)
min = ostream->rt_delta;
if (ostream->rtp_delta < rtp_min)
rtp_min = ostream->rtp_delta;
}
/* arrange to re-sync for each stream upon significant change,
* e.g. post-seek */
all_sync = all_sync && (stream->clock_base == clock_base);
stream->clock_base = clock_base;
/* may need init performed above later on, but nothing more to do now */
if (client->nstreams <= 1)
return;
GST_DEBUG_OBJECT (bin, "client %p min delta %" G_GINT64_FORMAT
" all sync %d", client, min, all_sync);
GST_DEBUG_OBJECT (bin, "rtcp sync mode %d, use_rtp %d", rtcp_sync, use_rtp);
switch (rtcp_sync) {
case GST_RTP_BIN_RTCP_SYNC_RTP:
if (!use_rtp)
break;
GST_DEBUG_OBJECT (bin, "using rtp generated reports; "
"client %p min rtp delta %" G_GINT64_FORMAT, client, rtp_min);
/* fall-through */
case GST_RTP_BIN_RTCP_SYNC_INITIAL:
/* if all have been synced already, do not bother further */
if (all_sync) {
GST_DEBUG_OBJECT (bin, "all streams already synced; done");
return;
}
break;
default:
break;
}
/* bail out if we adjusted recently enough */
if (all_sync && (last_unix - bin->priv->last_unix) <
bin->rtcp_sync_interval * GST_MSECOND) {
GST_DEBUG_OBJECT (bin, "discarding RTCP sender packet for sync; "
"previous sender info too recent "
"(previous UNIX %" G_GUINT64_FORMAT ")", bin->priv->last_unix);
return;
}
bin->priv->last_unix = last_unix;
/* calculate offsets for each stream */
for (walk = client->streams; walk; walk = g_slist_next (walk)) {
GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data;
gint64 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. */
if (use_rtp)
ts_offset = ostream->rtp_delta - rtp_min;
else
ts_offset = ostream->rt_delta - min;
stream_set_ts_offset (bin, ostream, ts_offset, TRUE);
}
}
gst_rtp_bin_send_sync_event (stream);
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 void
gst_rtp_bin_handle_sync (GstElement * jitterbuffer, GstStructure * s,
GstRtpBinStream * stream)
{
GstRtpBin *bin;
GstRTCPPacket packet;
guint32 ssrc;
guint64 ntptime;
gboolean have_sr, have_sdes;
gboolean more;
guint64 base_rtptime;
guint64 base_time;
guint clock_rate;
guint64 clock_base;
guint64 extrtptime;
GstBuffer *buffer;
GstRTCPBuffer rtcp = { NULL, };
bin = stream->bin;
GST_DEBUG_OBJECT (bin, "sync handler called");
/* 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. */
base_rtptime =
g_value_get_uint64 (gst_structure_get_value (s, "base-rtptime"));
base_time = g_value_get_uint64 (gst_structure_get_value (s, "base-time"));
clock_rate = g_value_get_uint (gst_structure_get_value (s, "clock-rate"));
clock_base = g_value_get_uint64 (gst_structure_get_value (s, "clock-base"));
extrtptime =
g_value_get_uint64 (gst_structure_get_value (s, "sr-ext-rtptime"));
buffer = gst_value_get_buffer (gst_structure_get_value (s, "sr-buffer"));
have_sr = FALSE;
have_sdes = FALSE;
gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
GST_RTCP_BUFFER_FOR_PACKETS (more, &rtcp, &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, NULL,
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;
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) {
GST_RTP_BIN_LOCK (bin);
/* associate the stream to CNAME */
gst_rtp_bin_associate (bin, stream, len, data,
ntptime, extrtptime, base_rtptime, base_time, clock_rate,
clock_base);
GST_RTP_BIN_UNLOCK (bin);
}
}
}
have_sdes = TRUE;
break;
}
default:
/* we can ignore these packets */
break;
}
}
gst_rtcp_buffer_unmap (&rtcp);
}
/* 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 = NULL;
GstRtpBinStream *stream;
GstRtpBin *rtpbin;
GstState target;
rtpbin = session->bin;
if (!(buffer = gst_element_factory_make ("rtpjitterbuffer", NULL)))
goto no_jitterbuffer;
if (!rtpbin->ignore_pt)
if (!(demux = gst_element_factory_make ("rtpptdemux", NULL)))
goto no_demux;
stream = g_new0 (GstRtpBinStream, 1);
stream->ssrc = ssrc;
stream->bin = rtpbin;
stream->session = session;
stream->buffer = buffer;
stream->demux = demux;
stream->have_sync = FALSE;
stream->rt_delta = 0;
stream->rtp_delta = 0;
stream->percent = 100;
stream->clock_base = -100 * GST_SECOND;
session->streams = g_slist_prepend (session->streams, stream);
/* provide clock_rate to the jitterbuffer when needed */
stream->buffer_ptreq_sig = g_signal_connect (buffer, "request-pt-map",
(GCallback) pt_map_requested, session);
stream->buffer_ntpstop_sig = g_signal_connect (buffer, "on-npt-stop",
(GCallback) on_npt_stop, stream);
g_object_set_data (G_OBJECT (buffer), "GstRTPBin.session", session);
g_object_set_data (G_OBJECT (buffer), "GstRTPBin.stream", stream);
/* configure latency and packet lost */
g_object_set (buffer, "latency", rtpbin->latency_ms, NULL);
g_object_set (buffer, "drop-on-latency", rtpbin->drop_on_latency, NULL);
g_object_set (buffer, "do-lost", rtpbin->do_lost, NULL);
g_object_set (buffer, "mode", rtpbin->buffer_mode, NULL);
g_object_set (buffer, "do-retransmission", rtpbin->do_retransmission, NULL);
g_signal_emit (rtpbin, gst_rtp_bin_signals[SIGNAL_NEW_JITTERBUFFER], 0,
buffer, session->id, ssrc);
if (!rtpbin->ignore_pt)
gst_bin_add (GST_BIN_CAST (rtpbin), demux);
gst_bin_add (GST_BIN_CAST (rtpbin), buffer);
/* link stuff */
if (demux)
gst_element_link_pads_full (buffer, "src", demux, "sink",
GST_PAD_LINK_CHECK_NOTHING);
if (rtpbin->buffering) {
guint64 last_out;
GST_INFO_OBJECT (rtpbin,
"bin is buffering, set jitterbuffer as not active");
g_signal_emit_by_name (buffer, "set-active", FALSE, (gint64) 0, &last_out);
}
GST_OBJECT_LOCK (rtpbin);
target = GST_STATE_TARGET (rtpbin);
GST_OBJECT_UNLOCK (rtpbin);
/* from sink to source */
if (demux)
gst_element_set_state (demux, target);
gst_element_set_state (buffer, target);
return stream;
/* ERRORS */
no_jitterbuffer:
{
g_warning ("rtpbin: could not create rtpjitterbuffer element");
return NULL;
}
no_demux:
{
gst_object_unref (buffer);
g_warning ("rtpbin: could not create rtpptdemux element");
return NULL;
}
}
/* called with RTP_BIN_LOCK */
static void
free_stream (GstRtpBinStream * stream, GstRtpBin * bin)
{
GSList *clients, *next_client;
GST_DEBUG_OBJECT (bin, "freeing stream %p", stream);
if (stream->demux) {
g_signal_handler_disconnect (stream->demux, stream->demux_newpad_sig);
g_signal_handler_disconnect (stream->demux, stream->demux_ptreq_sig);
g_signal_handler_disconnect (stream->demux, stream->demux_ptchange_sig);
}
g_signal_handler_disconnect (stream->buffer, stream->buffer_handlesync_sig);
g_signal_handler_disconnect (stream->buffer, stream->buffer_ptreq_sig);
g_signal_handler_disconnect (stream->buffer, stream->buffer_ntpstop_sig);
if (stream->demux)
gst_element_set_locked_state (stream->demux, TRUE);
gst_element_set_locked_state (stream->buffer, TRUE);
if (stream->demux)
gst_element_set_state (stream->demux, GST_STATE_NULL);
gst_element_set_state (stream->buffer, GST_STATE_NULL);
/* now remove this signal, we need this while going to NULL because it to
* do some cleanups */
if (stream->demux)
g_signal_handler_disconnect (stream->demux, stream->demux_padremoved_sig);
gst_bin_remove (GST_BIN_CAST (bin), stream->buffer);
if (stream->demux)
gst_bin_remove (GST_BIN_CAST (bin), stream->demux);
for (clients = bin->clients; clients; clients = next_client) {
GstRtpBinClient *client = (GstRtpBinClient *) clients->data;
GSList *streams, *next_stream;
next_client = g_slist_next (clients);
for (streams = client->streams; streams; streams = next_stream) {
GstRtpBinStream *ostream = (GstRtpBinStream *) streams->data;
next_stream = g_slist_next (streams);
if (ostream == stream) {
client->streams = g_slist_delete_link (client->streams, streams);
/* If this was the last stream belonging to this client,
* clean up the client. */
if (--client->nstreams == 0) {
bin->clients = g_slist_delete_link (bin->clients, clients);
free_client (client, bin);
break;
}
}
}
}
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 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, const GstCaps * caps);
static void gst_rtp_bin_release_pad (GstElement * element, GstPad * pad);
static void gst_rtp_bin_handle_message (GstBin * bin, GstMessage * message);
#define gst_rtp_bin_parent_class parent_class
G_DEFINE_TYPE (GstRtpBin, gst_rtp_bin, GST_TYPE_BIN);
static gboolean
_gst_element_accumulator (GSignalInvocationHint * ihint,
GValue * return_accu, const GValue * handler_return, gpointer dummy)
{
GstElement *element;
element = g_value_get_object (handler_return);
GST_DEBUG ("got element %" GST_PTR_FORMAT, element);
if (!(ihint->run_type & G_SIGNAL_RUN_CLEANUP))
g_value_set_object (return_accu, element);
/* stop emission if we have an element */
return (element == NULL);
}
static gboolean
_gst_caps_accumulator (GSignalInvocationHint * ihint,
GValue * return_accu, const GValue * handler_return, gpointer dummy)
{
GstCaps *caps;
caps = g_value_get_boxed (handler_return);
GST_DEBUG ("got caps %" GST_PTR_FORMAT, caps);
if (!(ihint->run_type & G_SIGNAL_RUN_CLEANUP))
g_value_set_boxed (return_accu, caps);
/* stop emission if we have a caps */
return (caps == NULL);
}
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 | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
g_param_spec_boolean ("drop-on-latency",
"Drop buffers when maximum latency is reached",
"Tells the jitterbuffer to never exceed the given latency in size",
DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* 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),
_gst_caps_accumulator, NULL, g_cclosure_marshal_generic, GST_TYPE_CAPS,
2, G_TYPE_UINT, G_TYPE_UINT);
/**
* GstRtpBin::payload-type-change:
* @rtpbin: the object which received the signal
* @session: the session
* @pt: the pt
*
* Signal that the current payload type changed to @pt in @session.
*/
gst_rtp_bin_signals[SIGNAL_PAYLOAD_TYPE_CHANGE] =
g_signal_new ("payload-type-change", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, payload_type_change),
NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 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::reset-sync:
* @rtpbin: the object which received the signal
*
* Reset all currently configured lip-sync parameters and require new SR
* packets for all streams before lip-sync is attempted again.
*/
gst_rtp_bin_signals[SIGNAL_RESET_SYNC] =
g_signal_new ("reset-sync", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
reset_sync), NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE,
0, G_TYPE_NONE);
/**
* GstRtpBin::get-internal-session:
* @rtpbin: the object which received the signal
* @id: the session id
*
* Request the internal RTPSession object as #GObject in session @id.
*/
gst_rtp_bin_signals[SIGNAL_GET_INTERNAL_SESSION] =
g_signal_new ("get-internal-session", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
get_internal_session), NULL, NULL, g_cclosure_marshal_generic,
RTP_TYPE_SESSION, 1, G_TYPE_UINT);
/**
* 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, g_cclosure_marshal_generic, 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, g_cclosure_marshal_generic, 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, g_cclosure_marshal_generic, 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, g_cclosure_marshal_generic, 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, g_cclosure_marshal_generic, 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, g_cclosure_marshal_generic, 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, g_cclosure_marshal_generic, 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, g_cclosure_marshal_generic, 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, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
G_TYPE_UINT);
/**
* GstRtpBin::on-npt-stop:
* @rtpbin: the object which received the signal
* @session: the session
* @ssrc: the SSRC
*
* Notify that SSRC sender has sent data up to the configured NPT stop time.
*/
gst_rtp_bin_signals[SIGNAL_ON_NPT_STOP] =
g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_npt_stop),
NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
G_TYPE_UINT);
/**
* GstRtpBin::request-rtp-encoder:
* @rtpbin: the object which received the signal
* @session: the session
*
* Request an RTP encoder element for the given @session. The encoder
* element will be added to the bin if not previously added.
*
* If no handler is connected, no encoder will be used.
*
* Since: 1.4
*/
gst_rtp_bin_signals[SIGNAL_REQUEST_RTP_ENCODER] =
g_signal_new ("request-rtp-encoder", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
request_rtp_encoder), _gst_element_accumulator, NULL,
g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
/**
* GstRtpBin::request-rtp-decoder:
* @rtpbin: the object which received the signal
* @session: the session
*
* Request an RTP decoder element for the given @session. The decoder
* element will be added to the bin if not previously added.
*
* If no handler is connected, no encoder will be used.
*
* Since: 1.4
*/
gst_rtp_bin_signals[SIGNAL_REQUEST_RTP_DECODER] =
g_signal_new ("request-rtp-decoder", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
request_rtp_decoder), _gst_element_accumulator, NULL,
g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
/**
* GstRtpBin::request-rtcp-encoder:
* @rtpbin: the object which received the signal
* @session: the session
*
* Request an RTCP encoder element for the given @session. The encoder
* element will be added to the bin if not previously added.
*
* If no handler is connected, no encoder will be used.
*
* Since: 1.4
*/
gst_rtp_bin_signals[SIGNAL_REQUEST_RTCP_ENCODER] =
g_signal_new ("request-rtcp-encoder", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
request_rtcp_encoder), _gst_element_accumulator, NULL,
g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
/**
* GstRtpBin::request-rtcp-decoder:
* @rtpbin: the object which received the signal
* @session: the session
*
* Request an RTCP decoder element for the given @session. The decoder
* element will be added to the bin if not previously added.
*
* If no handler is connected, no encoder will be used.
*
* Since: 1.4
*/
gst_rtp_bin_signals[SIGNAL_REQUEST_RTCP_DECODER] =
g_signal_new ("request-rtcp-decoder", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
request_rtcp_decoder), _gst_element_accumulator, NULL,
g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
/**
* GstRtpBin::new-jitterbuffer:
* @rtpbin: the object which received the signal
* @jitterbuffer: the new jitterbuffer
* @session: the session
* @ssrc: the SSRC
*
* Notify that a new @jitterbuffer was created for @session and @ssrc.
* This signal can, for example, be used to configure @jitterbuffer.
*
* Since: 1.4
*/
gst_rtp_bin_signals[SIGNAL_NEW_JITTERBUFFER] =
g_signal_new ("new-jitterbuffer", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
new_jitterbuffer), NULL, NULL, g_cclosure_marshal_generic,
G_TYPE_NONE, 3, GST_TYPE_ELEMENT, G_TYPE_UINT, G_TYPE_UINT);
/**
* GstRtpBin::request-aux-sender:
* @rtpbin: the object which received the signal
* @session: the session
*
* Request an AUX sender element for the given @session. The AUX
* element will be added to the bin.
*
* If no handler is connected, no AUX element will be used.
*
* Since: 1.4
*/
gst_rtp_bin_signals[SIGNAL_REQUEST_AUX_SENDER] =
g_signal_new ("request-aux-sender", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
request_aux_sender), _gst_element_accumulator, NULL,
g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
/**
* GstRtpBin::request-aux-receiver:
* @rtpbin: the object which received the signal
* @session: the session
*
* Request an AUX receiver element for the given @session. The AUX
* element will be added to the bin.
*
* If no handler is connected, no AUX element will be used.
*
* Since: 1.4
*/
gst_rtp_bin_signals[SIGNAL_REQUEST_AUX_RECEIVER] =
g_signal_new ("request-aux-receiver", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
request_aux_receiver), _gst_element_accumulator, NULL,
g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
g_object_class_install_property (gobject_class, PROP_SDES,
g_param_spec_boxed ("sdes", "SDES",
"The SDES items of this session",
GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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));
g_object_class_install_property (gobject_class, PROP_AUTOREMOVE,
g_param_spec_boolean ("autoremove", "Auto Remove",
"Automatically remove timed out sources", DEFAULT_AUTOREMOVE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_IGNORE_PT,
g_param_spec_boolean ("ignore-pt", "Ignore PT",
"Do not demultiplex based on PT values", DEFAULT_IGNORE_PT,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_USE_PIPELINE_CLOCK,
g_param_spec_boolean ("use-pipeline-clock", "Use pipeline clock",
"Use the pipeline running-time to set the NTP time in the RTCP SR messages",
DEFAULT_USE_PIPELINE_CLOCK,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRtpBin:buffer-mode:
*
* Control the buffering and timestamping mode used by the jitterbuffer.
*/
g_object_class_install_property (gobject_class, PROP_BUFFER_MODE,
g_param_spec_enum ("buffer-mode", "Buffer Mode",
"Control the buffering algorithm in use", RTP_TYPE_JITTER_BUFFER_MODE,
DEFAULT_BUFFER_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRtpBin:ntp-sync:
*
* Set the NTP time from the sender reports as the running-time on the
* buffers. When both the sender and receiver have sychronized
* running-time, i.e. when the clock and base-time is shared
* between the receivers and the and the senders, this option can be
* used to synchronize receivers on multiple machines.
*/
g_object_class_install_property (gobject_class, PROP_NTP_SYNC,
g_param_spec_boolean ("ntp-sync", "Sync on NTP clock",
"Synchronize received streams to the NTP clock", DEFAULT_NTP_SYNC,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRtpBin:rtcp-sync:
*
* If not synchronizing (directly) to the NTP clock, determines how to sync
* the various streams.
*/
g_object_class_install_property (gobject_class, PROP_RTCP_SYNC,
g_param_spec_enum ("rtcp-sync", "RTCP Sync",
"Use of RTCP SR in synchronization", GST_RTP_BIN_RTCP_SYNC_TYPE,
DEFAULT_RTCP_SYNC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRtpBin:rtcp-sync-interval:
*
* Determines how often to sync streams using RTCP data.
*/
g_object_class_install_property (gobject_class, PROP_RTCP_SYNC_INTERVAL,
g_param_spec_uint ("rtcp-sync-interval", "RTCP Sync Interval",
"RTCP SR interval synchronization (ms) (0 = always)",
0, G_MAXUINT, DEFAULT_RTCP_SYNC_INTERVAL,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_DO_SYNC_EVENT,
g_param_spec_boolean ("do-sync-event", "Do Sync Event",
"Send event downstream when a stream is synchronized to the sender",
DEFAULT_DO_SYNC_EVENT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_DO_RETRANSMISSION,
g_param_spec_boolean ("do-retransmission", "Do retransmission",
"Send an event downstream to request packet retransmission",
DEFAULT_DO_RETRANSMISSION,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_rtp_bin_change_state);
gstelement_class->request_new_pad =
GST_DEBUG_FUNCPTR (gst_rtp_bin_request_new_pad);
gstelement_class->release_pad = GST_DEBUG_FUNCPTR (gst_rtp_bin_release_pad);
/* sink pads */
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&rtpbin_recv_rtp_sink_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&rtpbin_recv_rtcp_sink_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&rtpbin_send_rtp_sink_template));
/* src pads */
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&rtpbin_recv_rtp_src_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&rtpbin_send_rtcp_src_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&rtpbin_send_rtp_src_template));
gst_element_class_set_static_metadata (gstelement_class, "RTP Bin",
"Filter/Network/RTP",
"Real-Time Transport Protocol bin",
"Wim Taymans <wim.taymans@gmail.com>");
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);
klass->reset_sync = GST_DEBUG_FUNCPTR (gst_rtp_bin_reset_sync);
klass->get_internal_session =
GST_DEBUG_FUNCPTR (gst_rtp_bin_get_internal_session);
klass->request_rtp_encoder = GST_DEBUG_FUNCPTR (gst_rtp_bin_request_encoder);
klass->request_rtp_decoder = GST_DEBUG_FUNCPTR (gst_rtp_bin_request_decoder);
klass->request_rtcp_encoder = GST_DEBUG_FUNCPTR (gst_rtp_bin_request_encoder);
klass->request_rtcp_decoder = GST_DEBUG_FUNCPTR (gst_rtp_bin_request_decoder);
GST_DEBUG_CATEGORY_INIT (gst_rtp_bin_debug, "rtpbin", 0, "RTP bin");
}
static void
gst_rtp_bin_init (GstRtpBin * rtpbin)
{
gchar *cname;
rtpbin->priv = GST_RTP_BIN_GET_PRIVATE (rtpbin);
g_mutex_init (&rtpbin->priv->bin_lock);
g_mutex_init (&rtpbin->priv->dyn_lock);
rtpbin->latency_ms = DEFAULT_LATENCY_MS;
rtpbin->latency_ns = DEFAULT_LATENCY_MS * GST_MSECOND;
rtpbin->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
rtpbin->do_lost = DEFAULT_DO_LOST;
rtpbin->ignore_pt = DEFAULT_IGNORE_PT;
rtpbin->ntp_sync = DEFAULT_NTP_SYNC;
rtpbin->rtcp_sync = DEFAULT_RTCP_SYNC;
rtpbin->rtcp_sync_interval = DEFAULT_RTCP_SYNC_INTERVAL;
rtpbin->priv->autoremove = DEFAULT_AUTOREMOVE;
rtpbin->buffer_mode = DEFAULT_BUFFER_MODE;
rtpbin->use_pipeline_clock = DEFAULT_USE_PIPELINE_CLOCK;
rtpbin->send_sync_event = DEFAULT_DO_SYNC_EVENT;
rtpbin->do_retransmission = DEFAULT_DO_RETRANSMISSION;
/* some default SDES entries */
cname = g_strdup_printf ("user%u@host-%x", g_random_int (), g_random_int ());
rtpbin->sdes = gst_structure_new ("application/x-rtp-source-sdes",
"cname", G_TYPE_STRING, cname, "tool", G_TYPE_STRING, "GStreamer", NULL);
g_free (cname);
}
static void
gst_rtp_bin_dispose (GObject * object)
{
GstRtpBin *rtpbin;
rtpbin = GST_RTP_BIN (object);
GST_RTP_BIN_LOCK (rtpbin);
GST_DEBUG_OBJECT (object, "freeing sessions");
g_slist_foreach (rtpbin->sessions, (GFunc) free_session, rtpbin);
g_slist_free (rtpbin->sessions);
rtpbin->sessions = NULL;
GST_RTP_BIN_UNLOCK (rtpbin);
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_rtp_bin_finalize (GObject * object)
{
GstRtpBin *rtpbin;
rtpbin = GST_RTP_BIN (object);
if (rtpbin->sdes)
gst_structure_free (rtpbin->sdes);
g_mutex_clear (&rtpbin->priv->bin_lock);
g_mutex_clear (&rtpbin->priv->dyn_lock);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_rtp_bin_set_sdes_struct (GstRtpBin * bin, const GstStructure * sdes)
{
GSList *item;
if (sdes == NULL)
return;
GST_RTP_BIN_LOCK (bin);
GST_OBJECT_LOCK (bin);
if (bin->sdes)
gst_structure_free (bin->sdes);
bin->sdes = gst_structure_copy (sdes);
GST_OBJECT_UNLOCK (bin);
/* store in all sessions */
for (item = bin->sessions; item; item = g_slist_next (item)) {
GstRtpBinSession *session = item->data;
g_object_set (session->session, "sdes", sdes, NULL);
}
GST_RTP_BIN_UNLOCK (bin);
}
static GstStructure *
gst_rtp_bin_get_sdes_struct (GstRtpBin * bin)
{
GstStructure *result;
GST_OBJECT_LOCK (bin);
result = gst_structure_copy (bin->sdes);
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_ms = g_value_get_uint (value);
rtpbin->latency_ns = rtpbin->latency_ms * GST_MSECOND;
GST_RTP_BIN_UNLOCK (rtpbin);
/* propagate the property down to the jitterbuffer */
gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin, "latency", value);
break;
case PROP_DROP_ON_LATENCY:
GST_RTP_BIN_LOCK (rtpbin);
rtpbin->drop_on_latency = g_value_get_boolean (value);
GST_RTP_BIN_UNLOCK (rtpbin);
/* propagate the property down to the jitterbuffer */
gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin,
"drop-on-latency", value);
break;
case PROP_SDES:
gst_rtp_bin_set_sdes_struct (rtpbin, g_value_get_boxed (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;
case PROP_NTP_SYNC:
rtpbin->ntp_sync = g_value_get_boolean (value);
break;
case PROP_RTCP_SYNC:
g_atomic_int_set (&rtpbin->rtcp_sync, g_value_get_enum (value));
break;
case PROP_RTCP_SYNC_INTERVAL:
rtpbin->rtcp_sync_interval = g_value_get_uint (value);
break;
case PROP_IGNORE_PT:
rtpbin->ignore_pt = g_value_get_boolean (value);
break;
case PROP_AUTOREMOVE:
rtpbin->priv->autoremove = g_value_get_boolean (value);
break;
case PROP_USE_PIPELINE_CLOCK:
{
GSList *sessions;
GST_RTP_BIN_LOCK (rtpbin);
rtpbin->use_pipeline_clock = g_value_get_boolean (value);
for (sessions = rtpbin->sessions; sessions;
sessions = g_slist_next (sessions)) {
GstRtpBinSession *session = (GstRtpBinSession *) sessions->data;
g_object_set (G_OBJECT (session->session),
"use-pipeline-clock", rtpbin->use_pipeline_clock, NULL);
}
GST_RTP_BIN_UNLOCK (rtpbin);
}
break;
case PROP_DO_SYNC_EVENT:
rtpbin->send_sync_event = g_value_get_boolean (value);
break;
case PROP_BUFFER_MODE:
GST_RTP_BIN_LOCK (rtpbin);
rtpbin->buffer_mode = g_value_get_enum (value);
GST_RTP_BIN_UNLOCK (rtpbin);
/* propagate the property down to the jitterbuffer */
gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin, "mode", value);
break;
case PROP_DO_RETRANSMISSION:
GST_RTP_BIN_LOCK (rtpbin);
rtpbin->do_retransmission = g_value_get_boolean (value);
GST_RTP_BIN_UNLOCK (rtpbin);
gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin,
"do-retransmission", 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_ms);
GST_RTP_BIN_UNLOCK (rtpbin);
break;
case PROP_DROP_ON_LATENCY:
GST_RTP_BIN_LOCK (rtpbin);
g_value_set_boolean (value, rtpbin->drop_on_latency);
GST_RTP_BIN_UNLOCK (rtpbin);
break;
case PROP_SDES:
g_value_take_boxed (value, gst_rtp_bin_get_sdes_struct (rtpbin));
break;
case PROP_DO_LOST:
GST_RTP_BIN_LOCK (rtpbin);
g_value_set_boolean (value, rtpbin->do_lost);
GST_RTP_BIN_UNLOCK (rtpbin);
break;
case PROP_IGNORE_PT:
g_value_set_boolean (value, rtpbin->ignore_pt);
break;
case PROP_NTP_SYNC:
g_value_set_boolean (value, rtpbin->ntp_sync);
break;
case PROP_RTCP_SYNC:
g_value_set_enum (value, g_atomic_int_get (&rtpbin->rtcp_sync));
break;
case PROP_RTCP_SYNC_INTERVAL:
g_value_set_uint (value, rtpbin->rtcp_sync_interval);
break;
case PROP_AUTOREMOVE:
g_value_set_boolean (value, rtpbin->priv->autoremove);
break;
case PROP_BUFFER_MODE:
g_value_set_enum (value, rtpbin->buffer_mode);
break;
case PROP_USE_PIPELINE_CLOCK:
g_value_set_boolean (value, rtpbin->use_pipeline_clock);
break;
case PROP_DO_SYNC_EVENT:
g_value_set_boolean (value, rtpbin->send_sync_event);
break;
case PROP_DO_RETRANSMISSION:
GST_RTP_BIN_LOCK (rtpbin);
g_value_set_boolean (value, rtpbin->do_retransmission);
GST_RTP_BIN_UNLOCK (rtpbin);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
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, "application/x-rtp-source-sdes")) {
GstRtpBinSession *sess;
/* find the session we set it as object data */
sess = g_object_get_data (G_OBJECT (GST_MESSAGE_SRC (message)),
"GstRTPBin.session");
if (G_LIKELY (sess)) {
message = gst_message_make_writable (message);
s = gst_message_get_structure (message);
gst_structure_set ((GstStructure *) s, "session", G_TYPE_UINT,
sess->id, NULL);
}
}
GST_BIN_CLASS (parent_class)->handle_message (bin, message);
break;
}
case GST_MESSAGE_BUFFERING:
{
gint percent;
gint min_percent = 100;
GSList *sessions, *streams;
GstRtpBinStream *stream;
gboolean change = FALSE, active = FALSE;
GstClockTime min_out_time;
GstBufferingMode mode;
gint avg_in, avg_out;
gint64 buffering_left;
gst_message_parse_buffering (message, &percent);
gst_message_parse_buffering_stats (message, &mode, &avg_in, &avg_out,
&buffering_left);
stream =
g_object_get_data (G_OBJECT (GST_MESSAGE_SRC (message)),
"GstRTPBin.stream");
GST_DEBUG_OBJECT (bin, "got percent %d from stream %p", percent, stream);
/* get the stream */
if (G_LIKELY (stream)) {
GST_RTP_BIN_LOCK (rtpbin);
/* fill in the percent */
stream->percent = percent;
/* calculate the min value for all streams */
for (sessions = rtpbin->sessions; sessions;
sessions = g_slist_next (sessions)) {
GstRtpBinSession *session = (GstRtpBinSession *) sessions->data;
GST_RTP_SESSION_LOCK (session);
if (session->streams) {
for (streams = session->streams; streams;
streams = g_slist_next (streams)) {
GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
GST_DEBUG_OBJECT (bin, "stream %p percent %d", stream,
stream->percent);
/* find min percent */
if (min_percent > stream->percent)
min_percent = stream->percent;
}
} else {
GST_INFO_OBJECT (bin,
"session has no streams, setting min_percent to 0");
min_percent = 0;
}
GST_RTP_SESSION_UNLOCK (session);
}
GST_DEBUG_OBJECT (bin, "min percent %d", min_percent);
if (rtpbin->buffering) {
if (min_percent == 100) {
rtpbin->buffering = FALSE;
active = TRUE;
change = TRUE;
}
} else {
if (min_percent < 100) {
/* pause the streams */
rtpbin->buffering = TRUE;
active = FALSE;
change = TRUE;
}
}
GST_RTP_BIN_UNLOCK (rtpbin);
gst_message_unref (message);
/* make a new buffering message with the min value */
message =
gst_message_new_buffering (GST_OBJECT_CAST (bin), min_percent);
gst_message_set_buffering_stats (message, mode, avg_in, avg_out,
buffering_left);
if (G_UNLIKELY (change)) {
GstClock *clock;
guint64 running_time = 0;
guint64 offset = 0;
/* figure out the running time when we have a clock */
if (G_LIKELY ((clock =
gst_element_get_clock (GST_ELEMENT_CAST (bin))))) {
guint64 now, base_time;
now = gst_clock_get_time (clock);
base_time = gst_element_get_base_time (GST_ELEMENT_CAST (bin));
running_time = now - base_time;
gst_object_unref (clock);
}
GST_DEBUG_OBJECT (bin,
"running time now %" GST_TIME_FORMAT,
GST_TIME_ARGS (running_time));
GST_RTP_BIN_LOCK (rtpbin);
/* when we reactivate, calculate the offsets so that all streams have
* an output time that is at least as big as the running_time */
offset = 0;
if (active) {
if (running_time > rtpbin->buffer_start) {
offset = running_time - rtpbin->buffer_start;
if (offset >= rtpbin->latency_ns)
offset -= rtpbin->latency_ns;
else
offset = 0;
}
}
/* pause all streams */
min_out_time = -1;
for (sessions = rtpbin->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;
GstElement *element = stream->buffer;
guint64 last_out;
g_signal_emit_by_name (element, "set-active", active, offset,
&last_out);
if (!active) {
g_object_get (element, "percent", &stream->percent, NULL);
if (last_out == -1)
last_out = 0;
if (min_out_time == -1 || last_out < min_out_time)
min_out_time = last_out;
}
GST_DEBUG_OBJECT (bin,
"setting %p to %d, offset %" GST_TIME_FORMAT ", last %"
GST_TIME_FORMAT ", percent %d", element, active,
GST_TIME_ARGS (offset), GST_TIME_ARGS (last_out),
stream->percent);
}
GST_RTP_SESSION_UNLOCK (session);
}
GST_DEBUG_OBJECT (bin,
"min out time %" GST_TIME_FORMAT, GST_TIME_ARGS (min_out_time));
/* the buffer_start is the min out time of all paused jitterbuffers */
if (!active)
rtpbin->buffer_start = min_out_time;
GST_RTP_BIN_UNLOCK (rtpbin);
}
}
GST_BIN_CLASS (parent_class)->handle_message (bin, message);
break;
}
default:
{
GST_BIN_CLASS (parent_class)->handle_message (bin, message);
break;
}
}
}
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:
priv->last_unix = 0;
GST_LOG_OBJECT (rtpbin, "clearing shutdown flag");
g_atomic_int_set (&priv->shutdown, 0);
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;
}
static GstElement *
session_request_element (GstRtpBinSession * session, guint signal)
{
GstElement *element = NULL;
GstRtpBin *bin = session->bin;
g_signal_emit (bin, gst_rtp_bin_signals[signal], 0, session->id, &element);
if (element) {
if (!bin_manage_element (bin, element))
goto manage_failed;
session->elements = g_slist_prepend (session->elements, element);
}
return element;
/* ERRORS */
manage_failed:
{
GST_WARNING_OBJECT (bin, "unable to manage element");
gst_object_unref (element);
return NULL;
}
}
static gboolean
copy_sticky_events (GstPad * pad, GstEvent ** event, gpointer user_data)
{
GstPad *gpad = GST_PAD_CAST (user_data);
GST_DEBUG_OBJECT (gpad, "store sticky event %" GST_PTR_FORMAT, *event);
gst_pad_store_sticky_event (gpad, *event);
return TRUE;
}
/* a new pad (SSRC) was created in @session. This signal is emited from the
* payload demuxer. */
static void
new_payload_found (GstElement * element, guint pt, GstPad * pad,
GstRtpBinStream * stream)
{
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_%u_%u_%u");
padname = g_strdup_printf ("recv_rtp_src_%u_%u_%u",
stream->session->id, stream->ssrc, pt);
gpad = gst_ghost_pad_new_from_template (padname, pad, templ);
g_free (padname);
g_object_set_data (G_OBJECT (pad), "GstRTPBin.ghostpad", gpad);
gst_pad_set_active (gpad, TRUE);
GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin);
gst_pad_sticky_events_foreach (pad, copy_sticky_events, gpad);
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), gpad);
return;
shutdown:
{
GST_DEBUG ("ignoring, we are shutting down");
return;
}
}
static void
payload_pad_removed (GstElement * element, GstPad * pad,
GstRtpBinStream * stream)
{
GstRtpBin *rtpbin;
GstPad *gpad;
rtpbin = stream->bin;
GST_DEBUG ("payload pad removed");
GST_RTP_BIN_DYN_LOCK (rtpbin);
if ((gpad = g_object_get_data (G_OBJECT (pad), "GstRTPBin.ghostpad"))) {
g_object_set_data (G_OBJECT (pad), "GstRTPBin.ghostpad", NULL);
gst_pad_set_active (gpad, FALSE);
gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin), gpad);
}
GST_RTP_BIN_DYN_UNLOCK (rtpbin);
}
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;
}
}
static void
payload_type_change (GstElement * element, guint pt, GstRtpBinSession * session)
{
GST_DEBUG_OBJECT (session->bin,
"emiting signal for pt type changed to %d in session %d", pt,
session->id);
g_signal_emit (session->bin, gst_rtp_bin_signals[SIGNAL_PAYLOAD_TYPE_CHANGE],
0, session->id, pt);
}
/* 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)) {
gst_caps_unref (caps);
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)
{
GstRtpBin *rtpbin;
GstRtpBinStream *stream;
GstPad *sinkpad, *srcpad;
gchar *padname;
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 pad and link */
GST_DEBUG_OBJECT (rtpbin, "linking jitterbuffer RTP");
padname = g_strdup_printf ("src_%u", ssrc);
srcpad = gst_element_get_static_pad (element, padname);
g_free (padname);
sinkpad = gst_element_get_static_pad (stream->buffer, "sink");
gst_pad_link_full (srcpad, sinkpad, GST_PAD_LINK_CHECK_NOTHING);
gst_object_unref (sinkpad);
gst_object_unref (srcpad);
GST_DEBUG_OBJECT (rtpbin, "linking jitterbuffer RTCP");
padname = g_strdup_printf ("rtcp_src_%u", ssrc);
srcpad = gst_element_get_static_pad (element, padname);
g_free (padname);
sinkpad = gst_element_get_request_pad (stream->buffer, "sink_rtcp");
gst_pad_link_full (srcpad, sinkpad, GST_PAD_LINK_CHECK_NOTHING);
gst_object_unref (sinkpad);
gst_object_unref (srcpad);
/* connect to the RTCP sync signal from the jitterbuffer */
GST_DEBUG_OBJECT (rtpbin, "connecting sync signal");
stream->buffer_handlesync_sig = g_signal_connect (stream->buffer,
"handle-sync", (GCallback) gst_rtp_bin_handle_sync, stream);
if (stream->demux) {
/* 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);
stream->demux_padremoved_sig = g_signal_connect (stream->demux,
"pad-removed", (GCallback) payload_pad_removed, stream);
/* connect to the request-pt-map signal. This signal will be emited by the
* 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 signal so it can be forwarded. */
stream->demux_ptchange_sig = g_signal_connect (stream->demux,
"payload-type-change", (GCallback) payload_type_change, session);
} else {
/* add rtpjitterbuffer src pad to pads */
GstElementClass *klass;
GstPadTemplate *templ;
gchar *padname;
GstPad *gpad, *pad;
pad = gst_element_get_static_pad (stream->buffer, "src");
/* ghost the pad to the parent */
klass = GST_ELEMENT_GET_CLASS (rtpbin);
templ = gst_element_class_get_pad_template (klass, "recv_rtp_src_%u_%u_%u");
padname = g_strdup_printf ("recv_rtp_src_%u_%u_%u",
stream->session->id, stream->ssrc, 255);
gpad = gst_ghost_pad_new_from_template (padname, pad, templ);
g_free (padname);
gst_pad_set_active (gpad, TRUE);
gst_pad_sticky_events_foreach (pad, copy_sticky_events, gpad);
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), gpad);
gst_object_unref (pad);
}
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;
}
}
static gboolean
complete_session_sink (GstRtpBin * rtpbin, GstRtpBinSession * session)
{
gchar *gname;
guint sessid = session->id;
GstPad *recv_rtp_sink;
GstElement *decoder;
GstElementClass *klass;
GstPadTemplate *templ;
/* 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, "requesting RTP decoder");
decoder = session_request_element (session, SIGNAL_REQUEST_RTP_DECODER);
if (decoder) {
GstPad *decsrc, *decsink;
GstPadLinkReturn ret;
GST_DEBUG_OBJECT (rtpbin, "linking RTP decoder");
decsink = gst_element_get_static_pad (decoder, "rtp_sink");
if (decsink == NULL)
goto dec_sink_failed;
recv_rtp_sink = decsink;
decsrc = gst_element_get_static_pad (decoder, "rtp_src");
if (decsrc == NULL)
goto dec_src_failed;
ret = gst_pad_link (decsrc, session->recv_rtp_sink);
gst_object_unref (decsrc);
if (ret != GST_PAD_LINK_OK)
goto dec_link_failed;
} else {
GST_DEBUG_OBJECT (rtpbin, "no RTP decoder given");
recv_rtp_sink = gst_object_ref (session->recv_rtp_sink);
}
GST_DEBUG_OBJECT (rtpbin, "ghosting session sink pad");
klass = GST_ELEMENT_GET_CLASS (rtpbin);
gname = g_strdup_printf ("recv_rtp_sink_%u", sessid);
templ = gst_element_class_get_pad_template (klass, "recv_rtp_sink_%u");
session->recv_rtp_sink_ghost =
gst_ghost_pad_new_from_template (gname, recv_rtp_sink, templ);
gst_object_unref (recv_rtp_sink);
gst_pad_set_active (session->recv_rtp_sink_ghost, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->recv_rtp_sink_ghost);
g_free (gname);
return TRUE;
/* ERRORS */
pad_failed:
{
g_warning ("rtpbin: failed to get session recv_rtp_sink pad");
return FALSE;
}
dec_sink_failed:
{
g_warning ("rtpbin: failed to get decoder sink pad for session %d", sessid);
return FALSE;
}
dec_src_failed:
{
g_warning ("rtpbin: failed to get decoder src pad for session %d", sessid);
gst_object_unref (recv_rtp_sink);
return FALSE;
}
dec_link_failed:
{
g_warning ("rtpbin: failed to link rtp decoder for session %d", sessid);
gst_object_unref (recv_rtp_sink);
return FALSE;
}
}
/* 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)
{
guint sessid;
GstElement *aux;
GstPad *recv_rtp_src;
GstRtpBinSession *session;
/* first get the session number */
if (name == NULL || sscanf (name, "recv_rtp_sink_%u", &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_ghost != NULL)
return session->recv_rtp_sink_ghost;
/* setup the session sink pad */
if (!complete_session_sink (rtpbin, session))
goto session_sink_failed;
session->recv_rtp_src =
gst_element_get_static_pad (session->session, "recv_rtp_src");
if (session->recv_rtp_src == NULL)
goto pad_failed;
/* find out if we need AUX elements or if we can go into the SSRC demuxer
* directly */
aux = session_request_element (session, SIGNAL_REQUEST_AUX_RECEIVER);
if (aux) {
gchar *pname;
GstPad *auxsink;
GstPadLinkReturn ret;
GST_DEBUG_OBJECT (rtpbin, "linking AUX receiver");
pname = g_strdup_printf ("sink_%d", sessid);
auxsink = gst_element_get_static_pad (aux, pname);
g_free (pname);
if (auxsink == NULL)
goto aux_sink_failed;
ret = gst_pad_link (session->recv_rtp_src, auxsink);
gst_object_unref (auxsink);
if (ret != GST_PAD_LINK_OK)
goto aux_link_failed;
/* this can be NULL when this AUX element is not to be linked to
* an SSRC demuxer */
pname = g_strdup_printf ("src_%d", sessid);
recv_rtp_src = gst_element_get_static_pad (aux, pname);
g_free (pname);
} else {
recv_rtp_src = gst_object_ref (session->recv_rtp_src);
}
if (recv_rtp_src) {
GstPad *sinkdpad;
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");
gst_pad_link_full (recv_rtp_src, sinkdpad, GST_PAD_LINK_CHECK_NOTHING);
gst_object_unref (recv_rtp_src);
gst_object_unref (sinkdpad);
/* 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);
session->demux_padremoved_sig = g_signal_connect (session->demux,
"removed-ssrc-pad", (GCallback) ssrc_demux_pad_removed, session);
}
return session->recv_rtp_sink_ghost;
/* ERRORS */
no_name:
{
g_warning ("rtpbin: invalid name given");
return NULL;
}
create_error:
{
/* create_session already warned */
return NULL;
}
session_sink_failed:
{
/* warning already done */
return NULL;
}
pad_failed:
{
g_warning ("rtpbin: failed to get session recv_rtp_src pad");
return NULL;
}
aux_sink_failed:
{
g_warning ("rtpbin: failed to get AUX sink pad for session %d", sessid);
return NULL;
}
aux_link_failed:
{
g_warning ("rtpbin: failed to link AUX pad to session %d", sessid);
return NULL;
}
}
static void
remove_recv_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session)
{
if (session->demux_newpad_sig) {
g_signal_handler_disconnect (session->demux, session->demux_newpad_sig);
session->demux_newpad_sig = 0;
}
if (session->demux_padremoved_sig) {
g_signal_handler_disconnect (session->demux, session->demux_padremoved_sig);
session->demux_padremoved_sig = 0;
}
if (session->recv_rtp_src) {
gst_object_unref (session->recv_rtp_src);
session->recv_rtp_src = NULL;
}
if (session->recv_rtp_sink) {
gst_element_release_request_pad (session->session, session->recv_rtp_sink);
gst_object_unref (session->recv_rtp_sink);
session->recv_rtp_sink = NULL;
}
if (session->recv_rtp_sink_ghost) {
gst_pad_set_active (session->recv_rtp_sink_ghost, FALSE);
gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin),
session->recv_rtp_sink_ghost);
session->recv_rtp_sink_ghost = 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)
{
guint sessid;
GstElement *decoder;
GstRtpBinSession *session;
GstPad *sinkdpad, *decsink;
/* first get the session number */
if (name == NULL || sscanf (name, "recv_rtcp_sink_%u", &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_ghost != NULL)
return session->recv_rtcp_sink_ghost;
/* 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;
GST_DEBUG_OBJECT (rtpbin, "getting RTCP decoder");
decoder = session_request_element (session, SIGNAL_REQUEST_RTCP_DECODER);
if (decoder) {
GstPad *decsrc;
GstPadLinkReturn ret;
GST_DEBUG_OBJECT (rtpbin, "linking RTCP decoder");
decsink = gst_element_get_static_pad (decoder, "rtcp_sink");
decsrc = gst_element_get_static_pad (decoder, "rtcp_src");
if (decsink == NULL)
goto dec_sink_failed;
if (decsrc == NULL)
goto dec_src_failed;
ret = gst_pad_link (decsrc, session->recv_rtcp_sink);
gst_object_unref (decsrc);
if (ret != GST_PAD_LINK_OK)
goto dec_link_failed;
} else {
GST_DEBUG_OBJECT (rtpbin, "no RTCP decoder given");
decsink = gst_object_ref (session->recv_rtcp_sink);
}
/* 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 src_pad_failed;
GST_DEBUG_OBJECT (rtpbin, "getting demuxer RTCP sink pad");
sinkdpad = gst_element_get_static_pad (session->demux, "rtcp_sink");
gst_pad_link_full (session->sync_src, sinkdpad, GST_PAD_LINK_CHECK_NOTHING);
gst_object_unref (sinkdpad);
session->recv_rtcp_sink_ghost =
gst_ghost_pad_new_from_template (name, decsink, templ);
gst_object_unref (decsink);
gst_pad_set_active (session->recv_rtcp_sink_ghost, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin),
session->recv_rtcp_sink_ghost);
return session->recv_rtcp_sink_ghost;
/* ERRORS */
no_name:
{
g_warning ("rtpbin: invalid name given");
return NULL;
}
create_error:
{
/* create_session already warned */
return NULL;
}
pad_failed:
{
g_warning ("rtpbin: failed to get session rtcp_sink pad");
return NULL;
}
dec_sink_failed:
{
g_warning ("rtpbin: failed to get decoder sink pad for session %d", sessid);
return NULL;
}
dec_src_failed:
{
g_warning ("rtpbin: failed to get decoder src pad for session %d", sessid);
gst_object_unref (decsink);
return NULL;
}
dec_link_failed:
{
g_warning ("rtpbin: failed to link rtcp decoder for session %d", sessid);
gst_object_unref (decsink);
return NULL;
}
src_pad_failed:
{
g_warning ("rtpbin: failed to get session sync_src pad");
gst_object_unref (decsink);
return NULL;
}
}
static void
remove_recv_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session)
{
if (session->recv_rtcp_sink_ghost) {
gst_pad_set_active (session->recv_rtcp_sink_ghost, FALSE);
gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin),
session->recv_rtcp_sink_ghost);
session->recv_rtcp_sink_ghost = NULL;
}
if (session->sync_src) {
/* releasing the request pad should also unref the sync pad */
gst_object_unref (session->sync_src);
session->sync_src = NULL;
}
if (session->recv_rtcp_sink) {
gst_element_release_request_pad (session->session, session->recv_rtcp_sink);
gst_object_unref (session->recv_rtcp_sink);
session->recv_rtcp_sink = NULL;
}
}
static gboolean
complete_session_src (GstRtpBin * rtpbin, GstRtpBinSession * session)
{
gchar *gname;
guint sessid = session->id;
GstPad *send_rtp_src;
GstElement *encoder;
GstElementClass *klass;
GstPadTemplate *templ;
/* 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;
GST_DEBUG_OBJECT (rtpbin, "getting RTP encoder");
encoder = session_request_element (session, SIGNAL_REQUEST_RTP_ENCODER);
if (encoder) {
gchar *ename;
GstPad *encsrc, *encsink;
GstPadLinkReturn ret;
GST_DEBUG_OBJECT (rtpbin, "linking RTP encoder");
ename = g_strdup_printf ("rtp_src_%d", sessid);
encsrc = gst_element_get_static_pad (encoder, ename);
g_free (ename);
if (encsrc == NULL)
goto enc_src_failed;
send_rtp_src = encsrc;
ename = g_strdup_printf ("rtp_sink_%d", sessid);
encsink = gst_element_get_static_pad (encoder, ename);
g_free (ename);
if (encsink == NULL)
goto enc_sink_failed;
ret = gst_pad_link (session->send_rtp_src, encsink);
gst_object_unref (encsink);
if (ret != GST_PAD_LINK_OK)
goto enc_link_failed;
} else {
GST_DEBUG_OBJECT (rtpbin, "no RTP encoder given");
send_rtp_src = gst_object_ref (session->send_rtp_src);
}
/* ghost the new source pad */
klass = GST_ELEMENT_GET_CLASS (rtpbin);
gname = g_strdup_printf ("send_rtp_src_%u", sessid);
templ = gst_element_class_get_pad_template (klass, "send_rtp_src_%u");
session->send_rtp_src_ghost =
gst_ghost_pad_new_from_template (gname, send_rtp_src, templ);
gst_object_unref (send_rtp_src);
gst_pad_set_active (session->send_rtp_src_ghost, TRUE);
gst_pad_sticky_events_foreach (send_rtp_src, copy_sticky_events,
session->send_rtp_src_ghost);
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->send_rtp_src_ghost);
g_free (gname);
return TRUE;
/* ERRORS */
no_srcpad:
{
g_warning ("rtpbin: failed to get rtp source pad for session %d", sessid);
return FALSE;
}
enc_src_failed:
{
g_warning ("rtpbin: failed to get encoder src pad for session %d", sessid);
return FALSE;
}
enc_sink_failed:
{
g_warning ("rtpbin: failed to get encoder sink pad for session %d", sessid);
gst_object_unref (send_rtp_src);
return FALSE;
}
enc_link_failed:
{
g_warning ("rtpbin: failed to link rtp encoder for session %d", sessid);
gst_object_unref (send_rtp_src);
return FALSE;
}
}
static gboolean
setup_aux_sender_fold (const GValue * item, GValue * result, gpointer user_data)
{
GstPad *pad;
gchar *name;
guint sessid;
GstRtpBinSession *session = user_data, *newsess;
GstRtpBin *rtpbin = session->bin;
GstPadLinkReturn ret;
pad = g_value_get_object (item);
name = gst_pad_get_name (pad);
if (name == NULL || sscanf (name, "src_%u", &sessid) != 1)
goto no_name;
g_free (name);
newsess = find_session_by_id (rtpbin, sessid);
if (newsess == NULL) {
/* create new session */
newsess = create_session (rtpbin, sessid);
if (newsess == NULL)
goto create_error;
} else if (newsess->send_rtp_sink != NULL)
goto existing_session;
/* get send_rtp pad and store */
newsess->send_rtp_sink =
gst_element_get_request_pad (newsess->session, "send_rtp_sink");
if (newsess->send_rtp_sink == NULL)
goto pad_failed;
ret = gst_pad_link (pad, newsess->send_rtp_sink);
if (ret != GST_PAD_LINK_OK)
goto aux_link_failed;
if (!complete_session_src (rtpbin, newsess))
goto session_src_failed;
return TRUE;
/* ERRORS */
no_name:
{
GST_WARNING ("ignoring invalid pad name %s", GST_STR_NULL (name));
g_free (name);
return TRUE;
}
create_error:
{
/* create_session already warned */
return FALSE;
}
existing_session:
{
g_warning ("rtpbin: session %d is already a sender", sessid);
return FALSE;
}
pad_failed:
{
g_warning ("rtpbin: failed to get session pad for session %d", sessid);
return FALSE;
}
aux_link_failed:
{
g_warning ("rtpbin: failed to link AUX for session %d", sessid);
return FALSE;
}
session_src_failed:
{
g_warning ("rtpbin: failed to complete AUX for session %d", sessid);
return FALSE;
}
}
static gboolean
setup_aux_sender (GstRtpBin * rtpbin, GstRtpBinSession * session,
GstElement * aux)
{
GstIterator *it;
GValue result = { 0, };
GstIteratorResult res;
it = gst_element_iterate_src_pads (aux);
res = gst_iterator_fold (it, setup_aux_sender_fold, &result, session);
gst_iterator_free (it);
return res == GST_ITERATOR_DONE;
}
/* 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)
{
gchar *pname;
guint sessid;
GstPad *send_rtp_sink;
GstElement *aux;
GstRtpBinSession *session;
/* first get the session number */
if (name == NULL || sscanf (name, "send_rtp_sink_%u", &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_ghost != NULL)
return session->send_rtp_sink_ghost;
/* check if we are already using this session as a sender */
if (session->send_rtp_sink != NULL)
goto existing_session;
GST_DEBUG_OBJECT (rtpbin, "getting RTP AUX sender");
aux = session_request_element (session, SIGNAL_REQUEST_AUX_SENDER);
if (aux) {
GST_DEBUG_OBJECT (rtpbin, "linking AUX sender");
if (!setup_aux_sender (rtpbin, session, aux))
goto aux_session_failed;
pname = g_strdup_printf ("sink_%d", sessid);
send_rtp_sink = gst_element_get_static_pad (aux, pname);
g_free (pname);
if (send_rtp_sink == NULL)
goto aux_sink_failed;
} else {
/* 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;
if (!complete_session_src (rtpbin, session))
goto session_src_failed;
send_rtp_sink = gst_object_ref (session->send_rtp_sink);
}
session->send_rtp_sink_ghost =
gst_ghost_pad_new_from_template (name, send_rtp_sink, templ);
gst_object_unref (send_rtp_sink);
gst_pad_set_active (session->send_rtp_sink_ghost, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->send_rtp_sink_ghost);
return session->send_rtp_sink_ghost;
/* ERRORS */
no_name:
{
g_warning ("rtpbin: invalid name given");
return NULL;
}
create_error:
{
/* create_session already warned */
return NULL;
}
existing_session:
{
g_warning ("rtpbin: session %d is already in use", sessid);
return NULL;
}
aux_session_failed:
{
g_warning ("rtpbin: failed to get AUX sink pad for session %d", sessid);
return NULL;
}
aux_sink_failed:
{
g_warning ("rtpbin: failed to get AUX sink pad for session %d", sessid);
return NULL;
}
pad_failed:
{
g_warning ("rtpbin: failed to get session pad for session %d", sessid);
return NULL;
}
session_src_failed:
{
g_warning ("rtpbin: failed to setup source pads for session %d", sessid);
return NULL;
}
}
static void
remove_send_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session)
{
if (session->send_rtp_src_ghost) {
gst_pad_set_active (session->send_rtp_src_ghost, FALSE);
gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin),
session->send_rtp_src_ghost);
session->send_rtp_src_ghost = NULL;
}
if (session->send_rtp_src) {
gst_object_unref (session->send_rtp_src);
session->send_rtp_src = NULL;
}
if (session->send_rtp_sink) {
gst_element_release_request_pad (GST_ELEMENT_CAST (session->session),
session->send_rtp_sink);
gst_object_unref (session->send_rtp_sink);
session->send_rtp_sink = NULL;
}
if (session->send_rtp_sink_ghost) {
gst_pad_set_active (session->send_rtp_sink_ghost, FALSE);
gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin),
session->send_rtp_sink_ghost);
session->send_rtp_sink_ghost = 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)
{
guint sessid;
GstPad *encsrc;
GstElement *encoder;
GstRtpBinSession *session;
/* first get the session number */
if (name == NULL || sscanf (name, "send_rtcp_src_%u", &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_ghost != NULL)
return session->send_rtcp_src_ghost;
/* 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;
GST_DEBUG_OBJECT (rtpbin, "getting RTCP encoder");
encoder = session_request_element (session, SIGNAL_REQUEST_RTCP_ENCODER);
if (encoder) {
gchar *ename;
GstPad *encsink;
GstPadLinkReturn ret;
GST_DEBUG_OBJECT (rtpbin, "linking RTCP encoder");
ename = g_strdup_printf ("rtcp_sink_%d", sessid);
encsink = gst_element_get_static_pad (encoder, ename);
g_free (ename);
ename = g_strdup_printf ("rtcp_src_%d", sessid);
encsrc = gst_element_get_static_pad (encoder, ename);
g_free (ename);
if (encsrc == NULL)
goto enc_src_failed;
if (encsink == NULL)
goto enc_sink_failed;
ret = gst_pad_link (session->send_rtcp_src, encsink);
gst_object_unref (encsink);
if (ret != GST_PAD_LINK_OK)
goto enc_link_failed;
} else {
GST_DEBUG_OBJECT (rtpbin, "no RTCP encoder given");
encsrc = gst_object_ref (session->send_rtcp_src);
}
session->send_rtcp_src_ghost =
gst_ghost_pad_new_from_template (name, encsrc, templ);
gst_object_unref (encsrc);
gst_pad_set_active (session->send_rtcp_src_ghost, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->send_rtcp_src_ghost);
return session->send_rtcp_src_ghost;
/* ERRORS */
no_name:
{
g_warning ("rtpbin: invalid name given");
return NULL;
}
no_session:
{
g_warning ("rtpbin: session with id %d does not exist", sessid);
return NULL;
}
pad_failed:
{
g_warning ("rtpbin: failed to get rtcp pad for session %d", sessid);
return NULL;
}
enc_src_failed:
{
g_warning ("rtpbin: failed to get encoder src pad for session %d", sessid);
return NULL;
}
enc_sink_failed:
{
g_warning ("rtpbin: failed to get encoder sink pad for session %d", sessid);
gst_object_unref (encsrc);
return NULL;
}
enc_link_failed:
{
g_warning ("rtpbin: failed to link rtcp encoder for session %d", sessid);
gst_object_unref (encsrc);
return NULL;
}
}
static void
remove_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session)
{
if (session->send_rtcp_src_ghost) {
gst_pad_set_active (session->send_rtcp_src_ghost, FALSE);
gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin),
session->send_rtcp_src_ghost);
session->send_rtcp_src_ghost = NULL;
}
if (session->send_rtcp_src) {
gst_element_release_request_pad (session->session, session->send_rtcp_src);
gst_object_unref (session->send_rtcp_src);
session->send_rtcp_src = 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;
GstIterator *pad_it = NULL;
gchar *pad_name = NULL;
GValue data = { 0, };
GST_DEBUG_OBJECT (element, "find a free pad name for template");
while (!name_found) {
gboolean done = FALSE;
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 (!done) {
switch (gst_iterator_next (pad_it, &data)) {
case GST_ITERATOR_OK:
{
GstPad *pad;
gchar *name;
pad = g_value_get_object (&data);
name = gst_pad_get_name (pad);
if (strcmp (name, pad_name) == 0) {
done = TRUE;
name_found = FALSE;
}
g_free (name);
g_value_reset (&data);
break;
}
case GST_ITERATOR_ERROR:
case GST_ITERATOR_RESYNC:
/* restart iteration */
done = TRUE;
name_found = FALSE;
session = 0;
break;
case GST_ITERATOR_DONE:
done = TRUE;
break;
}
}
g_value_unset (&data);
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, const GstCaps * caps)
{
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_OBJECT (rtpbin, "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_%u")) {
result = create_recv_rtp (rtpbin, templ, pad_name);
} else if (templ == gst_element_class_get_pad_template (klass,
"recv_rtcp_sink_%u")) {
result = create_recv_rtcp (rtpbin, templ, pad_name);
} else if (templ == gst_element_class_get_pad_template (klass,
"send_rtp_sink_%u")) {
result = create_send_rtp (rtpbin, templ, pad_name);
} else if (templ == gst_element_class_get_pad_template (klass,
"send_rtcp_src_%u")) {
result = create_rtcp (rtpbin, templ, pad_name);
} 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 ("rtpbin: this is not our template");
return NULL;
}
}
static void
gst_rtp_bin_release_pad (GstElement * element, GstPad * pad)
{
GstRtpBinSession *session;
GstRtpBin *rtpbin;
g_return_if_fail (GST_IS_GHOST_PAD (pad));
g_return_if_fail (GST_IS_RTP_BIN (element));
rtpbin = GST_RTP_BIN (element);
GST_RTP_BIN_LOCK (rtpbin);
GST_DEBUG_OBJECT (rtpbin, "Trying to release pad %s:%s",
GST_DEBUG_PAD_NAME (pad));
if (!(session = find_session_by_pad (rtpbin, pad)))
goto unknown_pad;
if (session->recv_rtp_sink_ghost == pad) {
remove_recv_rtp (rtpbin, session);
} else if (session->recv_rtcp_sink_ghost == pad) {
remove_recv_rtcp (rtpbin, session);
} else if (session->send_rtp_sink_ghost == pad) {
remove_send_rtp (rtpbin, session);
} else if (session->send_rtcp_src_ghost == pad) {
remove_rtcp (rtpbin, session);
}
/* no more request pads, free the complete session */
if (session->recv_rtp_sink_ghost == NULL
&& session->recv_rtcp_sink_ghost == NULL
&& session->send_rtp_sink_ghost == NULL
&& session->send_rtcp_src_ghost == NULL) {
GST_DEBUG_OBJECT (rtpbin, "no more pads for session %p", session);
rtpbin->sessions = g_slist_remove (rtpbin->sessions, session);
free_session (session, rtpbin);
}
GST_RTP_BIN_UNLOCK (rtpbin);
return;
/* ERROR */
unknown_pad:
{
GST_RTP_BIN_UNLOCK (rtpbin);
g_warning ("rtpbin: %s:%s is not one of our request pads",
GST_DEBUG_PAD_NAME (pad));
return;
}
}
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