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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.
*
* #GstRtpBin can also demultiplex incoming bundled streams. The first
* #GstRtpSession will have a #GstRtpSsrcDemux element splitting the streams
* based on their SSRC and potentially dispatched to a different #GstRtpSession.
* Because retransmission SSRCs need to be merged with the corresponding media
* stream the #GstRtpBin::on-bundled-ssrc signal is emitted so that the
* application can find out to which session the SSRC belongs.
*
* The session manager needs the clock-rate of the payload types it is handling
* and will signal the #GstRtpSession::request-pt-map signal when it needs such a
* mapping. One can clear the cached values with the #GstRtpSession::clear-pt-map
* 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 as 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); \
/* Minimum time offset to apply. This compensates for rounding errors in NTP to
* RTP timestamp conversions */
#define MIN_TS_OFFSET (4 * GST_MSECOND)
struct _GstRtpBinPrivate
{
GMutex bin_lock;
/* lock protecting dynamic adding/removing */
GMutex dyn_lock;
/* if we are shutting down or not */
gint shutdown;
gboolean autoremove;
/* NTP time in ns of last SR sync used */
guint64 last_ntpnstime;
/* 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_SESSION,
SIGNAL_GET_INTERNAL_SESSION,
SIGNAL_GET_INTERNAL_STORAGE,
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_REQUEST_FEC_DECODER,
SIGNAL_REQUEST_FEC_ENCODER,
SIGNAL_NEW_JITTERBUFFER,
SIGNAL_NEW_STORAGE,
SIGNAL_REQUEST_AUX_SENDER,
SIGNAL_REQUEST_AUX_RECEIVER,
SIGNAL_ON_NEW_SENDER_SSRC,
SIGNAL_ON_SENDER_SSRC_ACTIVE,
SIGNAL_ON_BUNDLED_SSRC,
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
#define DEFAULT_RTP_PROFILE GST_RTP_PROFILE_AVP
#define DEFAULT_NTP_TIME_SOURCE GST_RTP_NTP_TIME_SOURCE_NTP
#define DEFAULT_RTCP_SYNC_SEND_TIME TRUE
#define DEFAULT_MAX_RTCP_RTP_TIME_DIFF 1000
#define DEFAULT_MAX_DROPOUT_TIME 60000
#define DEFAULT_MAX_MISORDER_TIME 2000
#define DEFAULT_RFC7273_SYNC FALSE
#define DEFAULT_MAX_STREAMS G_MAXUINT
#define DEFAULT_MAX_TS_OFFSET_ADJUSTMENT G_GUINT64_CONSTANT(0)
#define DEFAULT_MAX_TS_OFFSET G_GINT64_CONSTANT(3000000000)
enum
{
PROP_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_RTP_PROFILE,
PROP_NTP_TIME_SOURCE,
PROP_RTCP_SYNC_SEND_TIME,
PROP_MAX_RTCP_RTP_TIME_DIFF,
PROP_MAX_DROPOUT_TIME,
PROP_MAX_MISORDER_TIME,
PROP_RFC7273_SYNC,
PROP_MAX_STREAMS,
PROP_MAX_TS_OFFSET_ADJUSTMENT,
PROP_MAX_TS_OFFSET,
};
#define GST_RTP_BIN_RTCP_SYNC_TYPE (gst_rtp_bin_rtcp_sync_get_type())
static 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);
static GstRtpBinSession *create_session (GstRtpBin * rtpbin, gint id);
static GstPad *complete_session_sink (GstRtpBin * rtpbin,
GstRtpBinSession * session, gboolean bundle_demuxer_needed);
static void
complete_session_receiver (GstRtpBin * rtpbin, GstRtpBinSession * session,
guint sessid);
static GstPad *complete_session_rtcp (GstRtpBin * rtpbin,
GstRtpBinSession * session, guint sessid, gboolean bundle_demuxer_needed);
/* 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).
*
* Before the SSRC demuxer, a storage element may be inserted for the purpose
* of Forward Error Correction.
*/
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;
/* Fec support */
GstElement *storage;
/* Bundling support */
GstElement *rtp_funnel;
GstElement *rtcp_funnel;
GstElement *bundle_demux;
gulong bundle_demux_newpad_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_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);
}
static void
on_new_sender_ssrc (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
{
g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_NEW_SENDER_SSRC], 0,
sess->id, ssrc);
}
static void
on_sender_ssrc_active (GstElement * session, guint32 ssrc,
GstRtpBinSession * sess)
{
g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SENDER_SSRC_ACTIVE],
0, sess->id, 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);
}
static void
new_bundled_ssrc_pad_found (GstElement * element, guint ssrc, GstPad * pad,
GstRtpBinSession * session)
{
GValue result = G_VALUE_INIT;
GValue params[2] = { G_VALUE_INIT, G_VALUE_INIT };
guint session_id = 0;
GstRtpBinSession *target_session = NULL;
GstRtpBin *rtpbin = session->bin;
gchar *name;
GstPad *src_pad;
GstPad *recv_rtp_sink = NULL;
GstPad *recv_rtcp_sink = NULL;
GstPadLinkReturn ret;
GST_RTP_BIN_DYN_LOCK (rtpbin);
GST_DEBUG_OBJECT (rtpbin, "new bundled SSRC pad %08x, %s:%s", ssrc,
GST_DEBUG_PAD_NAME (pad));
g_value_init (&result, G_TYPE_UINT);
g_value_init (&params[0], GST_TYPE_ELEMENT);
g_value_set_object (&params[0], rtpbin);
g_value_init (&params[1], G_TYPE_UINT);
g_value_set_uint (&params[1], ssrc);
g_signal_emitv (params,
gst_rtp_bin_signals[SIGNAL_ON_BUNDLED_SSRC], 0, &result);
g_value_unset (&params[0]);
session_id = g_value_get_uint (&result);
if (session_id == 0) {
target_session = session;
} else {
target_session = find_session_by_id (rtpbin, (gint) session_id);
if (!target_session) {
target_session = create_session (rtpbin, session_id);
}
if (!target_session) {
/* create_session() warned already */
GST_RTP_BIN_DYN_UNLOCK (rtpbin);
return;
}
if (!target_session->recv_rtp_sink) {
recv_rtp_sink = complete_session_sink (rtpbin, target_session, FALSE);
}
if (!target_session->recv_rtp_src)
complete_session_receiver (rtpbin, target_session, session_id);
if (!target_session->recv_rtcp_sink) {
recv_rtcp_sink =
complete_session_rtcp (rtpbin, target_session, session_id, FALSE);
}
}
GST_DEBUG_OBJECT (rtpbin, "Assigning bundled ssrc %u to session %u", ssrc,
session_id);
if (!recv_rtp_sink) {
recv_rtp_sink =
gst_element_get_request_pad (target_session->rtp_funnel, "sink_%u");
}
if (!recv_rtcp_sink) {
recv_rtcp_sink =
gst_element_get_request_pad (target_session->rtcp_funnel, "sink_%u");
}
name = g_strdup_printf ("src_%u", ssrc);
src_pad = gst_element_get_static_pad (element, name);
ret = gst_pad_link (src_pad, recv_rtp_sink);
g_free (name);
gst_object_unref (src_pad);
gst_object_unref (recv_rtp_sink);
if (ret != GST_PAD_LINK_OK) {
g_warning
("rtpbin: failed to link bundle demuxer to receive rtp funnel for session %u",
session_id);
}
name = g_strdup_printf ("rtcp_src_%u", ssrc);
src_pad = gst_element_get_static_pad (element, name);
gst_pad_link (src_pad, recv_rtcp_sink);
g_free (name);
gst_object_unref (src_pad);
gst_object_unref (recv_rtcp_sink);
if (ret != GST_PAD_LINK_OK) {
g_warning
("rtpbin: failed to link bundle demuxer to receive rtcp sink pad for session %u",
session_id);
}
GST_RTP_BIN_DYN_UNLOCK (rtpbin);
}
/* create a session with the given id. Must be called with RTP_BIN_LOCK */
static GstRtpBinSession *
create_session (GstRtpBin * rtpbin, gint id)
{
GstRtpBinSession *sess;
GstElement *session, *demux;
GstElement *storage = NULL;
GstState target;
if (!(session = gst_element_factory_make ("rtpsession", NULL)))
goto no_session;
if (!(demux = gst_element_factory_make ("rtpssrcdemux", NULL)))
goto no_demux;
if (!(storage = gst_element_factory_make ("rtpstorage", NULL)))
goto no_storage;
g_signal_emit (rtpbin, gst_rtp_bin_signals[SIGNAL_NEW_STORAGE], 0, storage,
id);
sess = g_new0 (GstRtpBinSession, 1);
g_mutex_init (&sess->lock);
sess->id = id;
sess->bin = rtpbin;
sess->session = session;
sess->demux = demux;
sess->storage = storage;
sess->rtp_funnel = gst_element_factory_make ("funnel", NULL);
sess->rtcp_funnel = gst_element_factory_make ("funnel", NULL);
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, "rtp-profile",
rtpbin->rtp_profile, "rtcp-sync-send-time", rtpbin->rtcp_sync_send_time,
NULL);
if (rtpbin->use_pipeline_clock)
g_object_set (session, "use-pipeline-clock", rtpbin->use_pipeline_clock,
NULL);
else
g_object_set (session, "ntp-time-source", rtpbin->ntp_time_source, NULL);
g_object_set (session, "max-dropout-time", rtpbin->max_dropout_time,
"max-misorder-time", rtpbin->max_misorder_time, 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);
g_signal_connect (sess->session, "on-new-sender-ssrc",
(GCallback) on_new_sender_ssrc, sess);
g_signal_connect (sess->session, "on-sender-ssrc-active",
(GCallback) on_sender_ssrc_active, sess);
gst_bin_add (GST_BIN_CAST (rtpbin), session);
gst_bin_add (GST_BIN_CAST (rtpbin), demux);
gst_bin_add (GST_BIN_CAST (rtpbin), sess->rtp_funnel);
gst_bin_add (GST_BIN_CAST (rtpbin), sess->rtcp_funnel);
gst_bin_add (GST_BIN_CAST (rtpbin), storage);
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);
gst_element_set_state (sess->rtp_funnel, target);
gst_element_set_state (sess->rtcp_funnel, target);
gst_element_set_state (storage, 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;
}
no_storage:
{
gst_object_unref (session);
gst_object_unref (demux);
g_warning ("rtpbin: could not create rtpstorage 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 (g_object_is_floating (element))
element = gst_object_ref_sink (element);
if (!gst_bin_add (GST_BIN_CAST (bin), element))
goto add_failed;
if (!gst_element_sync_state_with_parent (element))
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");
gst_object_unref (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_element_set_locked_state (element, TRUE);
gst_bin_remove (GST_BIN_CAST (bin), element);
gst_element_set_state (element, GST_STATE_NULL);
}
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 %u 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 %u in session %u", 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 %u 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 GstElement *
gst_rtp_bin_get_session (GstRtpBin * bin, guint session_id)
{
GstRtpBinSession *session;
GstElement *ret = NULL;
GST_RTP_BIN_LOCK (bin);
GST_DEBUG_OBJECT (bin, "retrieving GstRtpSession, index: %u", session_id);
session = find_session_by_id (bin, (gint) session_id);
if (session) {
ret = gst_object_ref (session->session);
}
GST_RTP_BIN_UNLOCK (bin);
return ret;
}
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: %u",
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 GObject *
gst_rtp_bin_get_internal_storage (GstRtpBin * bin, guint session_id)
{
GObject *internal_storage = NULL;
GstRtpBinSession *session;
GST_RTP_BIN_LOCK (bin);
GST_DEBUG_OBJECT (bin, "retrieving internal storage object, index: %u",
session_id);
session = find_session_by_id (bin, (gint) session_id);
if (session && session->storage) {
g_object_get (session->storage, "internal-storage", &internal_storage,
NULL);
}
GST_RTP_BIN_UNLOCK (bin);
return internal_storage;
}
static GstElement *
gst_rtp_bin_request_encoder (GstRtpBin * bin, guint session_id)
{
GST_DEBUG_OBJECT (bin, "return NULL encoder");
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);
}
static void
gst_rtp_bin_propagate_property_to_session (GstRtpBin * bin,
const gchar * name, const GValue * value)
{
GSList *sessions;
GST_RTP_BIN_LOCK (bin);
for (sessions = bin->sessions; sessions; sessions = g_slist_next (sessions)) {
GstRtpBinSession *sess = (GstRtpBinSession *) sessions->data;
g_object_set_property (G_OBJECT (sess->session), name, value);
}
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 = -1;
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);
/* get current clock time and convert to running time */
clock_time = gst_clock_get_time (clock);
rt = clock_time - base_time;
if (bin->use_pipeline_clock) {
ntpns = rt;
/* add constant to convert from 1970 based time to 1900 based time */
ntpns += (2208988800LL * GST_SECOND);
} else {
switch (bin->ntp_time_source) {
case GST_RTP_NTP_TIME_SOURCE_NTP:
case GST_RTP_NTP_TIME_SOURCE_UNIX:{
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 */
if (bin->ntp_time_source == GST_RTP_NTP_TIME_SOURCE_NTP)
ntpns += (2208988800LL * GST_SECOND);
break;
}
case GST_RTP_NTP_TIME_SOURCE_RUNNING_TIME:
ntpns = rt;
break;
case GST_RTP_NTP_TIME_SOURCE_CLOCK_TIME:
ntpns = clock_time;
break;
default:
ntpns = -1; /* Fix uninited compiler warning */
g_assert_not_reached ();
break;
}
}
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, gint64 max_ts_offset, gint64 min_ts_offset,
gboolean allow_positive_ts_offset)
{
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);
/* ignore minor offsets */
if (ABS (diff) < min_ts_offset) {
GST_DEBUG_OBJECT (bin, "offset too small, ignoring");
return;
}
/* sanity check offset */
if (max_ts_offset > 0) {
if (ts_offset > 0 && !allow_positive_ts_offset) {
GST_DEBUG_OBJECT (bin,
"offset is positive (clocks are out of sync), ignoring");
return;
}
if (ABS (ts_offset) > max_ts_offset) {
GST_DEBUG_OBJECT (bin, "offset too large, ignoring");
return;
}
}
g_object_set (stream->buffer, "ts-offset", ts_offset, NULL);
}
GST_DEBUG_OBJECT (bin, "stream SSRC %08x, delta %" G_GINT64_FORMAT,
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;
GstClockTime running_time, running_time_rtp;
guint64 ntpnstime;
/* 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. */
running_time_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, running_time_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. */
running_time =
gst_util_uint64_scale_int (running_time_rtp, GST_SECOND, clock_rate);
running_time += base_time;
/* convert ntptime to nanoseconds */
ntpnstime = gst_util_uint64_scale (ntptime, GST_SECOND,
(G_GINT64_CONSTANT (1) << 32));
stream->have_sync = TRUE;
GST_DEBUG_OBJECT (bin,
"SR RTP running time %" G_GUINT64_FORMAT ", SR NTP %" G_GUINT64_FORMAT,
running_time, ntpnstime);
/* recalc inter stream playout offset, but only if there is more than one
* stream or we're doing NTP sync. */
if (bin->ntp_sync) {
gint64 ntpdiff, rtdiff;
guint64 local_ntpnstime;
GstClockTime local_running_time;
/* 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, &local_running_time, &local_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 = local_ntpnstime - ntpnstime;
/* 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 = local_running_time - running_time;
GST_DEBUG_OBJECT (bin,
"local NTP time %" G_GUINT64_FORMAT ", SR NTP time %" G_GUINT64_FORMAT,
local_ntpnstime, ntpnstime);
GST_DEBUG_OBJECT (bin,
"local running time %" G_GUINT64_FORMAT ", SR RTP running time %"
G_GUINT64_FORMAT, local_running_time, running_time);
GST_DEBUG_OBJECT (bin,
"NTP diff %" G_GINT64_FORMAT ", RT diff %" G_GINT64_FORMAT, ntpdiff,
rtdiff);
/* combine to get the final diff to apply to the running_time */
stream->rt_delta = rtdiff - ntpdiff;
stream_set_ts_offset (bin, stream, stream->rt_delta, bin->max_ts_offset,
0, 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. ntpnstime 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 = ntpnstime - running_time;
/* 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 && (ntpnstime - bin->priv->last_ntpnstime) <
bin->rtcp_sync_interval * GST_MSECOND) {
GST_DEBUG_OBJECT (bin, "discarding RTCP sender packet for sync; "
"previous sender info too recent "
"(previous NTP %" G_GUINT64_FORMAT ")", bin->priv->last_ntpnstime);
return;
}
bin->priv->last_ntpnstime = ntpnstime;
/* 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, bin->max_ts_offset,
MIN_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 (g_slist_length (session->streams) >= rtpbin->max_streams)
goto max_streams;
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_object_set (buffer, "max-rtcp-rtp-time-diff",
rtpbin->max_rtcp_rtp_time_diff, NULL);
g_object_set (buffer, "max-dropout-time", rtpbin->max_dropout_time,
"max-misorder-time", rtpbin->max_misorder_time, NULL);
g_object_set (buffer, "rfc7273-sync", rtpbin->rfc7273_sync, NULL);
g_object_set (buffer, "max-ts-offset-adjustment",
rtpbin->max_ts_offset_adjustment, NULL);
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 */
max_streams:
{
GST_WARNING_OBJECT (rtpbin, "stream exeeds maximum (%d)",
rtpbin->max_streams);
return NULL;
}
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-session:
* @rtpbin: the object which received the signal
* @id: the session id
*
* Request the related GstRtpSession as #GstElement related with session @id.
*
* Since: 1.8
*/
gst_rtp_bin_signals[SIGNAL_GET_SESSION] =
g_signal_new ("get-session", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
get_session), NULL, NULL, g_cclosure_marshal_generic,
GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
/**
* 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::get-internal-storage:
* @rtpbin: the object which received the signal
* @id: the session id
*
* Request the internal RTPStorage object as #GObject in session @id.
*
* Since: 1.14
*/
gst_rtp_bin_signals[SIGNAL_GET_INTERNAL_STORAGE] =
g_signal_new ("get-internal-storage", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
get_internal_storage), NULL, NULL, g_cclosure_marshal_generic,
G_TYPE_OBJECT, 1, G_TYPE_UINT);
/**
* GstRtpBin::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::new-storage:
* @rtpbin: the object which received the signal
* @storage: the new storage
* @session: the session
*
* Notify that a new @storage was created for @session.
* This signal can, for example, be used to configure @storage.
*
* Since: 1.14
*/
gst_rtp_bin_signals[SIGNAL_NEW_STORAGE] =
g_signal_new ("new-storage", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
new_storage), NULL, NULL, g_cclosure_marshal_generic,
G_TYPE_NONE, 2, GST_TYPE_ELEMENT, G_TYPE_UINT);
/**
* GstRtpBin::request-aux-sender:
* @rtpbin: the object which received the signal
* @session: the session
*
* 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);
/**
* GstRtpBin::request-fec-decoder:
* @rtpbin: the object which received the signal
* @session: the session index
*
* Request a FEC decoder element for the given @session. The element
* will be added to the bin after the pt demuxer.
*
* If no handler is connected, no FEC decoder will be used.
*
* Since: 1.14
*/
gst_rtp_bin_signals[SIGNAL_REQUEST_FEC_DECODER] =
g_signal_new ("request-fec-decoder", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
request_fec_decoder), _gst_element_accumulator, NULL,
g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
/**
* GstRtpBin::request-fec-encoder:
* @rtpbin: the object which received the signal
* @session: the session index
*
* Request a FEC encoder element for the given @session. The element
* will be added to the bin after the RTPSession.
*
* If no handler is connected, no FEC encoder will be used.
*
* Since: 1.14
*/
gst_rtp_bin_signals[SIGNAL_REQUEST_FEC_ENCODER] =
g_signal_new ("request-fec-encoder", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
request_fec_encoder), _gst_element_accumulator, NULL,
g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
/**
* GstRtpBin::on-new-sender-ssrc:
* @rtpbin: the object which received the signal
* @session: the session
* @ssrc: the sender SSRC
*
* Notify of a new sender SSRC that entered @session.
*
* Since: 1.8
*/
gst_rtp_bin_signals[SIGNAL_ON_NEW_SENDER_SSRC] =
g_signal_new ("on-new-sender-ssrc", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_new_sender_ssrc),
NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
G_TYPE_UINT);
/**
* GstRtpBin::on-sender-ssrc-active:
* @rtpbin: the object which received the signal
* @session: the session
* @ssrc: the sender SSRC
*
* Notify of a sender SSRC that is active, i.e., sending RTCP.
*
* Since: 1.8
*/
gst_rtp_bin_signals[SIGNAL_ON_SENDER_SSRC_ACTIVE] =
g_signal_new ("on-sender-ssrc-active", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
on_sender_ssrc_active), NULL, NULL, g_cclosure_marshal_generic,
G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT);
/**
* GstRtpBin::on-bundled-ssrc:
* @rtpbin: the object which received the signal
* @ssrc: the bundled SSRC
*
* Notify of a new incoming bundled SSRC. If no handler is connected to the
* signal then the #GstRtpSession created for the recv_rtp_sink_\%u
* request pad will be managing this new SSRC. However if there is a handler
* connected then the application can decided to dispatch this new stream to
* another session by providing its ID as return value of the handler. This
* can be particularly useful to keep retransmission SSRCs grouped with the
* session for which they handle retransmission.
*
* Since: 1.12
*/
gst_rtp_bin_signals[SIGNAL_ON_BUNDLED_SSRC] =
g_signal_new ("on-bundled-ssrc", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
on_bundled_ssrc), NULL, NULL,
g_cclosure_marshal_generic, G_TYPE_UINT, 1, G_TYPE_UINT);
g_object_class_install_property (gobject_class, PROP_SDES,
g_param_spec_boxed ("sdes", "SDES",
"The SDES items of this session",
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 "
"(DEPRECATED: Use ntp-time-source property)",
DEFAULT_USE_PIPELINE_CLOCK,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | G_PARAM_DEPRECATED));
/**
* 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));
/**
* GstRtpBin:do-retransmission:
*
* Enables RTP retransmission on all streams. To control retransmission on
* a per-SSRC basis, connect to the #GstRtpBin::new-jitterbuffer signal and
* set the #GstRtpJitterBuffer::do-retransmission property on the
* #GstRtpJitterBuffer object instead.
*/
g_object_class_install_property (gobject_class, PROP_DO_RETRANSMISSION,
g_param_spec_boolean ("do-retransmission", "Do retransmission",
"Enable retransmission on all streams",
DEFAULT_DO_RETRANSMISSION,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRtpBin:rtp-profile:
*
* Sets the default RTP profile of newly created RTP sessions. The
* profile can be changed afterwards on a per-session basis.
*/
g_object_class_install_property (gobject_class, PROP_RTP_PROFILE,
g_param_spec_enum ("rtp-profile", "RTP Profile",
"Default RTP profile of newly created sessions",
GST_TYPE_RTP_PROFILE, DEFAULT_RTP_PROFILE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_NTP_TIME_SOURCE,
g_param_spec_enum ("ntp-time-source", "NTP Time Source",
"NTP time source for RTCP packets",
gst_rtp_ntp_time_source_get_type (), DEFAULT_NTP_TIME_SOURCE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_RTCP_SYNC_SEND_TIME,
g_param_spec_boolean ("rtcp-sync-send-time", "RTCP Sync Send Time",
"Use send time or capture time for RTCP sync "
"(TRUE = send time, FALSE = capture time)",
DEFAULT_RTCP_SYNC_SEND_TIME,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_MAX_RTCP_RTP_TIME_DIFF,
g_param_spec_int ("max-rtcp-rtp-time-diff", "Max RTCP RTP Time Diff",
"Maximum amount of time in ms that the RTP time in RTCP SRs "
"is allowed to be ahead (-1 disabled)", -1, G_MAXINT,
DEFAULT_MAX_RTCP_RTP_TIME_DIFF,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_MAX_DROPOUT_TIME,
g_param_spec_uint ("max-dropout-time", "Max dropout time",
"The maximum time (milliseconds) of missing packets tolerated.",
0, G_MAXUINT, DEFAULT_MAX_DROPOUT_TIME,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_MAX_MISORDER_TIME,
g_param_spec_uint ("max-misorder-time", "Max misorder time",
"The maximum time (milliseconds) of misordered packets tolerated.",
0, G_MAXUINT, DEFAULT_MAX_MISORDER_TIME,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_RFC7273_SYNC,
g_param_spec_boolean ("rfc7273-sync", "Sync on RFC7273 clock",
"Synchronize received streams to the RFC7273 clock "
"(requires clock and offset to be provided)", DEFAULT_RFC7273_SYNC,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_MAX_STREAMS,
g_param_spec_uint ("max-streams", "Max Streams",
"The maximum number of streams to create for one session",
0, G_MAXUINT, DEFAULT_MAX_STREAMS,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRtpBin:max-ts-offset-adjustment:
*
* Syncing time stamps to NTP time adds a time offset. This parameter
* specifies the maximum number of nanoseconds per frame that this time offset
* may be adjusted with. This is used to avoid sudden large changes to time
* stamps.
*
* Since: 1.14
*/
g_object_class_install_property (gobject_class, PROP_MAX_TS_OFFSET_ADJUSTMENT,
g_param_spec_uint64 ("max-ts-offset-adjustment",
"Max Timestamp Offset Adjustment",
"The maximum number of nanoseconds per frame that time stamp offsets "
"may be adjusted (0 = no limit).", 0, G_MAXUINT64,
DEFAULT_MAX_TS_OFFSET_ADJUSTMENT, G_PARAM_READWRITE |
G_PARAM_STATIC_STRINGS));
/**
* GstRtpBin:max-ts-offset:
*
* Used to set an upper limit of how large a time offset may be. This
* is used to protect against unrealistic values as a result of either
* client,server or clock issues.
*
* Since: 1.14
*/
g_object_class_install_property (gobject_class, PROP_MAX_TS_OFFSET,
g_param_spec_int64 ("max-ts-offset", "Max TS Offset",
"The maximum absolute value of the time offset in (nanoseconds). "
"Note, if the ntp-sync parameter is set the default value is "
"changed to 0 (no limit)", 0, G_MAXINT64, DEFAULT_MAX_TS_OFFSET,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_rtp_bin_change_state);
gstelement_class->request_new_pad =
GST_DEBUG_FUNCPTR (gst_rtp_bin_request_new_pad);
gstelement_class->release_pad = GST_DEBUG_FUNCPTR (gst_rtp_bin_release_pad);
/* sink pads */
gst_element_class_add_static_pad_template (gstelement_class,
&rtpbin_recv_rtp_sink_template);
gst_element_class_add_static_pad_template (gstelement_class,
&rtpbin_recv_rtcp_sink_template);
gst_element_class_add_static_pad_template (gstelement_class,
&rtpbin_send_rtp_sink_template);
/* src pads */
gst_element_class_add_static_pad_template (gstelement_class,
&rtpbin_recv_rtp_src_template);
gst_element_class_add_static_pad_template (gstelement_class,
&rtpbin_send_rtcp_src_template);
gst_element_class_add_static_pad_template (gstelement_class,
&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_session = GST_DEBUG_FUNCPTR (gst_rtp_bin_get_session);
klass->get_internal_session =
GST_DEBUG_FUNCPTR (gst_rtp_bin_get_internal_session);
klass->get_internal_storage =
GST_DEBUG_FUNCPTR (gst_rtp_bin_get_internal_storage);
klass->request_rtp_encoder = GST_DEBUG_FUNCPTR (gst_rtp_bin_request_encoder);
klass->request_rtp_decoder = GST_DEBUG_FUNCPTR (gst_rtp_bin_request_decoder);
klass->request_rtcp_encoder = GST_DEBUG_FUNCPTR (gst_rtp_bin_request_encoder);
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;
rtpbin->rtp_profile = DEFAULT_RTP_PROFILE;
rtpbin->ntp_time_source = DEFAULT_NTP_TIME_SOURCE;
rtpbin->rtcp_sync_send_time = DEFAULT_RTCP_SYNC_SEND_TIME;
rtpbin->max_rtcp_rtp_time_diff = DEFAULT_MAX_RTCP_RTP_TIME_DIFF;
rtpbin->max_dropout_time = DEFAULT_MAX_DROPOUT_TIME;
rtpbin->max_misorder_time = DEFAULT_MAX_MISORDER_TIME;
rtpbin->rfc7273_sync = DEFAULT_RFC7273_SYNC;
rtpbin->max_streams = DEFAULT_MAX_STREAMS;
rtpbin->max_ts_offset_adjustment = DEFAULT_MAX_TS_OFFSET_ADJUSTMENT;
rtpbin->max_ts_offset = DEFAULT_MAX_TS_OFFSET;
rtpbin->max_ts_offset_is_set = FALSE;
/* some default SDES entries */
cname = g_strdup_printf ("user%u@host-%x", g_random_int (), g_random_int ());
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);
/* The default value of max_ts_offset depends on ntp_sync. If user
* hasn't set it then change default value */