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gstreamer/gst/rtpmanager/gstrtpbin.c
Nicolas Dufresne 2ff7519d73 rtpbin: Allow reusing the sender AUX bin 
This is needed for the case you don't know in advance all the sessions
you will be using, but would like to place all the related AUX element
in the same GstBin. As per current implementation, each time an sender
AUX bin is requested and returned, RTPBin will walk the src pads and
create sessions for these pads.

In the current implementation, if a src pad already have a sessions, it
returns an error and stops. As a side effect, if an AUX bin is reused in
a following AUX bin request, it can only work if the pads are created on
the last request.

This change simply relax the restriction in order to keep walking, and
just ensure that all newly created pads have a sessions.
2019-03-21 21:10:43 +00:00

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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 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_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_STORAGE,
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);
static void
complete_session_receiver (GstRtpBin * rtpbin, GstRtpBinSession * session,
guint sessid);
static GstPad *complete_session_rtcp (GstRtpBin * rtpbin,
GstRtpBinSession * session, guint sessid);
/* 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;
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);
}
/* 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;
/* need to sink the storage or otherwise signal handlers from bindings will
* take ownership of it and we don't own it anymore */
gst_object_ref_sink (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->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), storage);
/* unref the storage again, the bin has a reference now and
* we don't need it anymore */
gst_object_unref (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 (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 GstElement *
gst_rtp_bin_get_storage (GstRtpBin * bin, guint session_id)
{
GstRtpBinSession *session;
GstElement *res = NULL;
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) {
res = gst_object_ref (session->storage);
}
GST_RTP_BIN_UNLOCK (bin);
return res;
}
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);
/* need to sink the jitterbufer or otherwise signal handlers from bindings will
* take ownership of it and we don't own it anymore */
gst_object_ref_sink (buffer);
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);
/* unref the jitterbuffer again, the bin has a reference now and
* we don't need it anymore */
gst_object_unref (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_WITH_PRIVATE (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;
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::get-storage:
* @rtpbin: the object which received the signal
* @id: the session id
*
* Request the RTPStorage element as #GObject in session @id.
*
* Since: 1.16
*/
gst_rtp_bin_signals[SIGNAL_GET_STORAGE] =
g_signal_new ("get-storage", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
get_storage), NULL, NULL, g_cclosure_marshal_generic,
GST_TYPE_ELEMENT, 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);
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_storage = GST_DEBUG_FUNCPTR (gst_rtp_bin_get_storage);
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_instance_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 */
if (!rtpbin->max_ts_offset_is_set) {
if (rtpbin->ntp_sync) {
rtpbin->max_ts_offset = 0;
} else {
rtpbin->max_ts_offset = DEFAULT_MAX_TS_OFFSET;
}
}
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;
case PROP_RTP_PROFILE:
rtpbin->rtp_profile = g_value_get_enum (value);
break;
case PROP_NTP_TIME_SOURCE:{
GSList *sessions;
GST_RTP_BIN_LOCK (rtpbin);
rtpbin->ntp_time_source = g_value_get_enum (value);
for (sessions = rtpbin->sessions; sessions;
sessions = g_slist_next (sessions)) {
GstRtpBinSession *session = (GstRtpBinSession *) sessions->data;
g_object_set (G_OBJECT (session->session),
"ntp-time-source", rtpbin->ntp_time_source, NULL);
}
GST_RTP_BIN_UNLOCK (rtpbin);
break;
}
case PROP_RTCP_SYNC_SEND_TIME:{
GSList *sessions;
GST_RTP_BIN_LOCK (rtpbin);
rtpbin->rtcp_sync_send_time = 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),
"rtcp-sync-send-time", rtpbin->rtcp_sync_send_time, NULL);
}
GST_RTP_BIN_UNLOCK (rtpbin);
break;
}
case PROP_MAX_RTCP_RTP_TIME_DIFF:
GST_RTP_BIN_LOCK (rtpbin);
rtpbin->max_rtcp_rtp_time_diff = g_value_get_int (value);
GST_RTP_BIN_UNLOCK (rtpbin);
gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin,
"max-rtcp-rtp-time-diff", value);
break;
case PROP_MAX_DROPOUT_TIME:
GST_RTP_BIN_LOCK (rtpbin);
rtpbin->max_dropout_time = g_value_get_uint (value);
GST_RTP_BIN_UNLOCK (rtpbin);
gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin,
"max-dropout-time", value);
gst_rtp_bin_propagate_property_to_session (rtpbin, "max-dropout-time",
value);
break;
case PROP_MAX_MISORDER_TIME:
GST_RTP_BIN_LOCK (rtpbin);
rtpbin->max_misorder_time = g_value_get_uint (value);
GST_RTP_BIN_UNLOCK (rtpbin);
gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin,
"max-misorder-time", value);
gst_rtp_bin_propagate_property_to_session (rtpbin, "max-misorder-time",
value);
break;
case PROP_RFC7273_SYNC:
rtpbin->rfc7273_sync = g_value_get_boolean (value);
gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin,
"rfc7273-sync", value);
break;
case PROP_MAX_STREAMS:
rtpbin->max_streams = g_value_get_uint (value);
break;
case PROP_MAX_TS_OFFSET_ADJUSTMENT:
rtpbin->max_ts_offset_adjustment = g_value_get_uint64 (value);
gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin,
"max-ts-offset-adjustment", value);
break;
case PROP_MAX_TS_OFFSET:
rtpbin->max_ts_offset = g_value_get_int64 (value);
rtpbin->max_ts_offset_is_set = TRUE;
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;
case PROP_RTP_PROFILE:
g_value_set_enum (value, rtpbin->rtp_profile);
break;
case PROP_NTP_TIME_SOURCE:
g_value_set_enum (value, rtpbin->ntp_time_source);
break;
case PROP_RTCP_SYNC_SEND_TIME:
g_value_set_boolean (value, rtpbin->rtcp_sync_send_time);
break;
case PROP_MAX_RTCP_RTP_TIME_DIFF:
GST_RTP_BIN_LOCK (rtpbin);
g_value_set_int (value, rtpbin->max_rtcp_rtp_time_diff);
GST_RTP_BIN_UNLOCK (rtpbin);
break;
case PROP_MAX_DROPOUT_TIME:
g_value_set_uint (value, rtpbin->max_dropout_time);
break;
case PROP_MAX_MISORDER_TIME:
g_value_set_uint (value, rtpbin->max_misorder_time);
break;
case PROP_RFC7273_SYNC:
g_value_set_boolean (value, rtpbin->rfc7273_sync);
break;
case PROP_MAX_STREAMS:
g_value_set_uint (value, rtpbin->max_streams);
break;
case PROP_MAX_TS_OFFSET_ADJUSTMENT:
g_value_set_uint64 (value, rtpbin->max_ts_offset_adjustment);
break;
case PROP_MAX_TS_OFFSET:
g_value_set_int64 (value, rtpbin->max_ts_offset);
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_ntpnstime = 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 emitted 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_OBJECT (rtpbin, "new payload pad %u", pt);
pad = gst_object_ref (pad);
if (stream->session->storage) {
GstElement *fec_decoder =
session_request_element (stream->session, SIGNAL_REQUEST_FEC_DECODER);
if (fec_decoder) {
GstPad *sinkpad, *srcpad;
GstPadLinkReturn ret;
sinkpad = gst_element_get_static_pad (fec_decoder, "sink");
if (!sinkpad)
goto fec_decoder_sink_failed;
ret = gst_pad_link (pad, sinkpad);
gst_object_unref (sinkpad);
if (ret != GST_PAD_LINK_OK)
goto fec_decoder_link_failed;
srcpad = gst_element_get_static_pad (fec_decoder, "src");
if (!srcpad)
goto fec_decoder_src_failed;
gst_pad_sticky_events_foreach (pad, copy_sticky_events, srcpad);
gst_object_unref (pad);
pad = srcpad;
}
}
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);
done:
gst_object_unref (pad);
return;
shutdown:
{
GST_DEBUG ("ignoring, we are shutting down");
goto done;
}
fec_decoder_sink_failed:
{
g_warning ("rtpbin: failed to get fec encoder sink pad for session %u",
stream->session->id);
goto done;
}
fec_decoder_src_failed:
{
g_warning ("rtpbin: failed to get fec encoder src pad for session %u",
stream->session->id);
goto done;
}
fec_decoder_link_failed:
{
g_warning ("rtpbin: failed to link fec decoder for session %u",
stream->session->id);
goto done;
}
}
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 %u in session %u", 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 GstCaps *
ptdemux_pt_map_requested (GstElement * element, guint pt,
GstRtpBinSession * session)
{
GstCaps *ret = pt_map_requested (element, pt, session);
if (ret && gst_caps_get_size (ret) == 1) {
const GstStructure *s = gst_caps_get_structure (ret, 0);
gboolean is_fec;
if (gst_structure_get_boolean (s, "is-fec", &is_fec) && is_fec) {
GValue v = G_VALUE_INIT;
GValue v2 = G_VALUE_INIT;
GST_INFO_OBJECT (session->bin, "Will ignore FEC pt %u in session %u", pt,
session->id);
g_value_init (&v, GST_TYPE_ARRAY);
g_value_init (&v2, G_TYPE_INT);
g_object_get_property (G_OBJECT (element), "ignored-payload-types", &v);
g_value_set_int (&v2, pt);
gst_value_array_append_value (&v, &v2);
g_value_unset (&v2);
g_object_set_property (G_OBJECT (element), "ignored-payload-types", &v);
g_value_unset (&v);
}
}
return ret;
}
static void
payload_type_change (GstElement * element, guint pt, GstRtpBinSession * session)
{
GST_DEBUG_OBJECT (session->bin,
"emiting signal for pt type changed to %u in session %u", pt,
session->id);
g_signal_emit (session->bin, gst_rtp_bin_signals[SIGNAL_PAYLOAD_TYPE_CHANGE],
0, session->id, pt);
}
/* emitted 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 emitted 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) ptdemux_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 GstPad *
complete_session_sink (GstRtpBin * rtpbin, GstRtpBinSession * session)
{
guint sessid = session->id;
GstPad *recv_rtp_sink;
GstElement *decoder;
g_assert (!session->recv_rtp_sink);
/* 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);
}
return recv_rtp_sink;
/* ERRORS */
pad_failed:
{
g_warning ("rtpbin: failed to get session recv_rtp_sink pad");
return NULL;
}
dec_sink_failed:
{
g_warning ("rtpbin: failed to get decoder sink pad for session %u", sessid);
return NULL;
}
dec_src_failed:
{
g_warning ("rtpbin: failed to get decoder src pad for session %u", sessid);
gst_object_unref (recv_rtp_sink);
return NULL;
}
dec_link_failed:
{
g_warning ("rtpbin: failed to link rtp decoder for session %u", sessid);
gst_object_unref (recv_rtp_sink);
return NULL;
}
}
static void
complete_session_receiver (GstRtpBin * rtpbin, GstRtpBinSession * session,
guint sessid)
{
GstElement *aux;
GstPad *recv_rtp_src;
g_assert (!session->recv_rtp_src);
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 */
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_%u", 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 any further */
pname = g_strdup_printf ("src_%u", 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);
}
/* Add a storage element if needed */
if (recv_rtp_src && session->storage) {
GstPadLinkReturn ret;
GstPad *sinkpad = gst_element_get_static_pad (session->storage, "sink");
ret = gst_pad_link (recv_rtp_src, sinkpad);
gst_object_unref (sinkpad);
gst_object_unref (recv_rtp_src);
if (ret != GST_PAD_LINK_OK)
goto storage_link_failed;
recv_rtp_src = gst_element_get_static_pad (session->storage, "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 (sinkdpad);
gst_object_unref (recv_rtp_src);
/* 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;
pad_failed:
{
g_warning ("rtpbin: failed to get session recv_rtp_src pad");
return;
}
aux_sink_failed:
{
g_warning ("rtpbin: failed to get AUX sink pad for session %u", sessid);
return;
}
aux_link_failed:
{
g_warning ("rtpbin: failed to link AUX pad to session %u", sessid);
return;
}
storage_link_failed:
{
g_warning ("rtpbin: failed to link storage");
return;
}
}
/* Create a pad for receiving RTP for the session in @name. Must be called with
* RTP_BIN_LOCK.
*/
static GstPad *
create_recv_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
{
guint sessid;
GstRtpBinSession *session;
GstPad *recv_rtp_sink;
/* 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 %u", sessid);
/* get or create session */
session = find_session_by_id (rtpbin, sessid);
if (!session) {
GST_DEBUG_OBJECT (rtpbin, "creating session %u", 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 */
recv_rtp_sink = complete_session_sink (rtpbin, session);
if (!recv_rtp_sink)
goto session_sink_failed;
GST_DEBUG_OBJECT (rtpbin, "ghosting session sink pad");
session->recv_rtp_sink_ghost =
gst_ghost_pad_new_from_template (name, 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);
complete_session_receiver (rtpbin, session, sessid);
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;
}
}
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;
}
}
static GstPad *
complete_session_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session,
guint sessid)
{
GstElement *decoder;
GstPad *sinkdpad;
GstPad *decsink = NULL;
/* 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);
return decsink;
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 %u", sessid);
return NULL;
}
dec_src_failed:
{
g_warning ("rtpbin: failed to get decoder src pad for session %u", sessid);
goto cleanup;
}
dec_link_failed:
{
g_warning ("rtpbin: failed to link rtcp decoder for session %u", sessid);
goto cleanup;
}
src_pad_failed:
{
g_warning ("rtpbin: failed to get session sync_src pad");
}
cleanup:
gst_object_unref (decsink);
return NULL;
}
/* Create a pad for receiving RTCP for the session in @name. Must be called with
* RTP_BIN_LOCK.
*/
static GstPad *
create_recv_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ,
const gchar * name)
{
guint sessid;
GstRtpBinSession *session;
GstPad *decsink = NULL;
/* 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 %u", sessid);
/* get or create the session */
session = find_session_by_id (rtpbin, sessid);
if (!session) {
GST_DEBUG_OBJECT (rtpbin, "creating session %u", 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;
decsink = complete_session_rtcp (rtpbin, session, sessid);
if (!decsink)
goto create_error;
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;
}
}
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;
gboolean ret = FALSE;
/* get srcpad */
send_rtp_src = gst_element_get_static_pad (session->session, "send_rtp_src");
if (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_%u", sessid);
encsrc = gst_element_get_static_pad (encoder, ename);
g_free (ename);
if (encsrc == NULL)
goto enc_src_failed;
ename = g_strdup_printf ("rtp_sink_%u", sessid);
encsink = gst_element_get_static_pad (encoder, ename);
g_free (ename);
if (encsink == NULL)
goto enc_sink_failed;
ret = gst_pad_link (send_rtp_src, encsink);
gst_object_unref (encsink);
gst_object_unref (send_rtp_src);
send_rtp_src = encsrc;
if (ret != GST_PAD_LINK_OK)
goto enc_link_failed;
} else {
GST_DEBUG_OBJECT (rtpbin, "no RTP encoder given");
}
/* 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_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);
ret = TRUE;
done:
if (send_rtp_src)
gst_object_unref (send_rtp_src);
return ret;
/* ERRORS */
no_srcpad:
{
g_warning ("rtpbin: failed to get rtp source pad for session %u", sessid);
goto done;
}
enc_src_failed:
{
g_warning ("rtpbin: failed to get %" GST_PTR_FORMAT
" src pad for session %u", encoder, sessid);
goto done;
}
enc_sink_failed:
{
g_warning ("rtpbin: failed to get %" GST_PTR_FORMAT
" sink pad for session %u", encoder, sessid);
goto done;
}
enc_link_failed:
{
g_warning ("rtpbin: failed to link %" GST_PTR_FORMAT " for session %u",
encoder, sessid);
goto done;
}
}
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:
{
GST_DEBUG_OBJECT (rtpbin,
"skipping src_%i setup, since it is already configured.", sessid);
return TRUE;
}
pad_failed:
{
g_warning ("rtpbin: failed to get session pad for session %u", sessid);
return FALSE;
}
aux_link_failed:
{
g_warning ("rtpbin: failed to link AUX for session %u", sessid);
return FALSE;
}
session_src_failed:
{
g_warning ("rtpbin: failed to complete AUX for session %u", 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;
GstElement *encoder;
GstElement *prev = NULL;
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;
encoder = session_request_element (session, SIGNAL_REQUEST_FEC_ENCODER);
if (encoder) {
GST_DEBUG_OBJECT (rtpbin, "Linking FEC encoder");
send_rtp_sink = gst_element_get_static_pad (encoder, "sink");
if (!send_rtp_sink)
goto enc_sink_failed;
prev = encoder;
}
GST_DEBUG_OBJECT (rtpbin, "getting RTP AUX sender");
aux = session_request_element (session, SIGNAL_REQUEST_AUX_SENDER);
if (aux) {
GstPad *sinkpad;
GST_DEBUG_OBJECT (rtpbin, "linking AUX sender");
if (!setup_aux_sender (rtpbin, session, aux))
goto aux_session_failed;
pname = g_strdup_printf ("sink_%u", sessid);
sinkpad = gst_element_get_static_pad (aux, pname);
g_free (pname);
if (sinkpad == NULL)
goto aux_sink_failed;
if (!prev) {
send_rtp_sink = sinkpad;
} else {
GstPad *srcpad = gst_element_get_static_pad (prev, "src");
GstPadLinkReturn ret;
ret = gst_pad_link (srcpad, sinkpad);
gst_object_unref (srcpad);
if (ret != GST_PAD_LINK_OK) {
goto aux_link_failed;
}
}
prev = aux;
} 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;
if (!prev) {
send_rtp_sink = gst_object_ref (session->send_rtp_sink);
} else {
GstPad *srcpad = gst_element_get_static_pad (prev, "src");
GstPadLinkReturn ret;
ret = gst_pad_link (srcpad, session->send_rtp_sink);
gst_object_unref (srcpad);
if (ret != GST_PAD_LINK_OK)
goto session_link_failed;
}
}
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 %u is already in use", sessid);
return NULL;
}
aux_session_failed:
{
g_warning ("rtpbin: failed to get AUX sink pad for session %u", sessid);
return NULL;
}
aux_sink_failed:
{
g_warning ("rtpbin: failed to get AUX sink pad for session %u", sessid);
return NULL;
}
aux_link_failed:
{
g_warning ("rtpbin: failed to link %" GST_PTR_FORMAT " for session %u",
aux, sessid);
return NULL;
}
pad_failed:
{
g_warning ("rtpbin: failed to get session pad for session %u", sessid);
return NULL;
}
session_src_failed:
{
g_warning ("rtpbin: failed to setup source pads for session %u", sessid);
return NULL;
}
session_link_failed:
{
g_warning ("rtpbin: failed to link %" GST_PTR_FORMAT " for session %u",
session, sessid);
return NULL;
}
enc_sink_failed:
{
g_warning ("rtpbin: failed to get %" GST_PTR_FORMAT
" sink pad for session %u", encoder, 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_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_send_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) {
GST_DEBUG_OBJECT (rtpbin, "creating session %u", sessid);
/* create session now */
session = create_session (rtpbin, sessid);
if (session == NULL)
goto create_error;
}
/* 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_src_%u", sessid);
encsrc = gst_element_get_static_pad (encoder, ename);
g_free (ename);
if (encsrc == NULL)
goto enc_src_failed;
ename = g_strdup_printf ("rtcp_sink_%u", 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_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;
}
create_error:
{
/* create_session already warned */
return NULL;
}
pad_failed:
{
g_warning ("rtpbin: failed to get rtcp pad for session %u", sessid);
return NULL;
}
enc_src_failed:
{
g_warning ("rtpbin: failed to get encoder src pad for session %u", sessid);
return NULL;
}
enc_sink_failed:
{
g_warning ("rtpbin: failed to get encoder sink pad for session %u", sessid);
gst_object_unref (encsrc);
return NULL;
}
enc_link_failed:
{
g_warning ("rtpbin: failed to link rtcp encoder for session %u", 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_send_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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